diff -ruN linux-2.6.22/Documentation/sound/alsa/ALSA-Configuration.txt linux-2.6.22-alsa/Documentation/sound/alsa/ALSA-Configuration.txt --- linux-2.6.22/Documentation/sound/alsa/ALSA-Configuration.txt 2007-07-28 13:40:44.000000000 +0200 +++ linux-2.6.22-alsa/Documentation/sound/alsa/ALSA-Configuration.txt 2007-09-01 20:57:16.000000000 +0200 @@ -365,13 +365,15 @@ Module snd-cmipci ----------------- - Module for C-Media CMI8338 and 8738 PCI sound cards. + Module for C-Media CMI8338/8738/8768/8770 PCI sound cards. - mpu_port - 0x300,0x310,0x320,0x330 = legacy port, - 1 = integrated PCI port, + mpu_port - port address of MIDI interface: + 0x300,0x310,0x320,0x330 = legacy port, + 1 = integrated PCI port (8738 or later), 0 = disable (default) - fm_port - 0x388 = legacy port, - 1 = integrated PCI port (default), + fm_port - port address of OPL-3 FM synthesizer (8x38 only): + 0x388 = legacy port, + 1 = integrated PCI port (default on 8738), 0 = disable soft_ac3 - Software-conversion of raw SPDIF packets (model 033 only) (default = 1) @@ -467,7 +469,12 @@ above explicitly. The power-management is supported. - + + Module snd-cs5530 + _________________ + + Module for Cyrix/NatSemi Geode 5530 chip. + Module snd-cs5535audio ---------------------- @@ -759,9 +766,14 @@ model - force the model name position_fix - Fix DMA pointer (0 = auto, 1 = none, 2 = POSBUF, 3 = FIFO size) + probe_mask - Bitmask to probe codecs (default = -1, meaning all slots) single_cmd - Use single immediate commands to communicate with codecs (for debugging only) enable_msi - Enable Message Signaled Interrupt (MSI) (default = off) + power_save - Automatic power-saving timtout (in second, 0 = + disable, default = 10) + power_save_controller - Reset HD-audio controller in power-saving mode + (default = on) This module supports one card and autoprobe. @@ -803,6 +815,8 @@ hp-3013 HP machines (3013-variant) fujitsu Fujitsu S7020 acer Acer TravelMate + will Will laptops (PB V7900) + replacer Replacer 672V basic fixed pin assignment (old default model) auto auto-config reading BIOS (default) @@ -811,16 +825,34 @@ hp-bpc HP xw4400/6400/8400/9400 laptops hp-bpc-d7000 HP BPC D7000 benq Benq ED8 + benq-t31 Benq T31 hippo Hippo (ATI) with jack detection, Sony UX-90s hippo_1 Hippo (Benq) with jack detection + sony-assamd Sony ASSAMD basic fixed pin assignment w/o SPDIF auto auto-config reading BIOS (default) + ALC268 + 3stack 3-stack model + toshiba Toshiba A205 + acer Acer laptops + auto auto-config reading BIOS (default) + + ALC662 + 3stack-dig 3-stack (2-channel) with SPDIF + 3stack-6ch 3-stack (6-channel) + 3stack-6ch-dig 3-stack (6-channel) with SPDIF + 6stack-dig 6-stack with SPDIF + lenovo-101e Lenovo laptop + auto auto-config reading BIOS (default) + ALC882/885 3stack-dig 3-jack with SPDIF I/O 6stack-dig 6-jack digital with SPDIF I/O arima Arima W820Di1 macpro MacPro support + mbp3 Macbook Pro rev3 + imac24 iMac 24'' with jack detection w2jc ASUS W2JC auto auto-config reading BIOS (default) @@ -831,10 +863,18 @@ 3stack-6ch-dig 3-jack 6-channel with SPDIF I/O 6stack-dig-demo 6-jack digital for Intel demo board acer Acer laptops (Travelmate 3012WTMi, Aspire 5600, etc) + acer-aspire Acer Aspire 9810 medion Medion Laptops + medion-md2 Medion MD2 targa-dig Targa/MSI targa-2ch-dig Targs/MSI with 2-channel laptop-eapd 3-jack with SPDIF I/O and EAPD (Clevo M540JE, M550JE) + lenovo-101e Lenovo 101E + lenovo-nb0763 Lenovo NB0763 + lenovo-ms7195-dig Lenovo MS7195 + haier-w66 Haier W66 + 6stack-hp HP machines with 6stack (Nettle boards) + 3stack-hp HP machines with 3stack (Lucknow, Samba boards) auto auto-config reading BIOS (default) ALC861/660 @@ -853,7 +893,10 @@ 3stack-dig 3-jack with SPDIF OUT 6stack-dig 6-jack with SPDIF OUT 3stack-660 3-jack (for ALC660VD) + 3stack-660-digout 3-jack with SPDIF OUT (for ALC660VD) lenovo Lenovo 3000 C200 + dallas Dallas laptops + hp HP TX1000 auto auto-config reading BIOS (default) CMI9880 @@ -864,12 +907,26 @@ allout 5-jack in back, 2-jack in front, SPDIF out auto auto-config reading BIOS (default) + AD1882 + 3stack 3-stack mode (default) + 6stack 6-stack mode + + AD1884 + N/A + AD1981 basic 3-jack (default) hp HP nx6320 thinkpad Lenovo Thinkpad T60/X60/Z60 toshiba Toshiba U205 + AD1983 + N/A + + AD1984 + basic default configuration + thinkpad Lenovo Thinkpad T61/X61 + AD1986A 6stack 6-jack, separate surrounds (default) 3stack 3-stack, shared surrounds @@ -900,18 +957,41 @@ can be adjusted. Appearing only when compiled with $CONFIG_SND_DEBUG=y - STAC9200/9205/9254 + STAC9200 + ref Reference board + dell-m21 Dell Inspiron 630m, Dell Inspiron 640m + dell-m22 Dell Latitude D620, Dell Latitude D820 + dell-m23 Dell XPS M1710, Dell Precision M90 + dell-m24 Dell Latitude 120L + dell-m25 Dell Inspiron E1505n + dell-m26 Dell Inspiron 1501 + dell-m27 Dell Inspiron E1705/9400 + + STAC9205/9254 ref Reference board + dell-m42 Dell (unknown) + dell-m43 Dell Precision + dell-m44 Dell Inspiron STAC9220/9221 ref Reference board 3stack D945 3stack 5stack D945 5stack + SPDIF - macmini Intel Mac Mini - macbook Intel Mac Book - macbook-pro-v1 Intel Mac Book Pro 1st generation - macbook-pro Intel Mac Book Pro 2nd generation - imac-intel Intel iMac + intel-mac-v1 Intel Mac Type 1 + intel-mac-v2 Intel Mac Type 2 + intel-mac-v3 Intel Mac Type 3 + intel-mac-v4 Intel Mac Type 4 + intel-mac-v5 Intel Mac Type 5 + macmini Intel Mac Mini (equivalent with type 3) + macbook Intel Mac Book (eq. type 5) + macbook-pro-v1 Intel Mac Book Pro 1st generation (eq. type 3) + macbook-pro Intel Mac Book Pro 2nd generation (eq. type 3) + imac-intel Intel iMac (eq. type 2) + imac-intel-20 Intel iMac (newer version) (eq. type 3) + dell-d81 Dell (unknown) + dell-d82 Dell (unknown) + dell-m81 Dell (unknown) + dell-m82 Dell XPS M1210 STAC9202/9250/9251 ref Reference board, base config @@ -923,6 +1003,7 @@ ref Reference board 3stack D965 3stack 5stack D965 5stack + SPDIF + dell-3stack Dell Dimension E520 STAC9872 vaio Setup for VAIO FE550G/SZ110 @@ -937,6 +1018,12 @@ subsystem ID (output of "lspci -nv") to ALSA BTS or alsa-devel ML (see the section "Links and Addresses"). + When CONFIG_SND_HDA_POWER_SAVE is set, two options, power_save and + power_save_controller become available. power_save specifies the + time to turn off the power automatically at idle status. When + power_save_controller is true, the controller is also turned off. + This might result in more obvious click noise at turning on/off. + Note 2: If you get click noises on output, try the module option position_fix=1 or 2. position_fix=1 will use the SD_LPIB register value without FIFO size correction as the current @@ -956,6 +1043,17 @@ from the irq. Remember this is a last resort, and should be avoided as much as possible... + MORE NOTES ON "azx_get_response timeout" PROBLEMS: + On some hardwares, you may need to add a proper probe_mask option + to avoid the "azx_get_response timeout" problem above, instead. + This occurs when the access to non-existing or non-working codec slot + (likely a modem one) causes a stall of the communication via HD-audio + bus. You can see which codec slots are probed by enabling + CONFIG_SND_DEBUG_DETECT, or simply from the file name of the codec + proc files. Then limit the slots to probe by probe_mask option. + For example, probe_mask=1 means to probe only the first slot, and + probe_mask=4 means only the third slot. + The power-management is supported. Module snd-hdsp @@ -1634,8 +1732,52 @@ dma2 - DMA2 # for CS4232 PCM interface. isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) + The below are options for wavefront_synth features: + wf_raw - Assume that we need to boot the OS (default:no) + If yes, then during driver loading, the state of the board is + ignored, and we reset the board and load the firmware anyway. + fx_raw - Assume that the FX process needs help (default:yes) + If false, we'll leave the FX processor in whatever state it is + when the driver is loaded. The default is to download the + microprogram and associated coefficients to set it up for + "default" operation, whatever that means. + debug_default - Debug parameters for card initialization + wait_usecs - How long to wait without sleeping, usecs + (default:150) + This magic number seems to give pretty optimal throughput + based on my limited experimentation. + If you want to play around with it and find a better value, be + my guest. Remember, the idea is to get a number that causes us + to just busy wait for as many WaveFront commands as possible, + without coming up with a number so large that we hog the whole + CPU. + Specifically, with this number, out of about 134,000 status + waits, only about 250 result in a sleep. + sleep_interval - How long to sleep when waiting for reply + (default: 100) + sleep_tries - How many times to try sleeping during a wait + (default: 50) + ospath - Pathname to processed ICS2115 OS firmware + (default:wavefront.os) + The path name of the ISC2115 OS firmware. In the recent + version, it's handled via firmware loader framework, so it + must be installed in the proper path, typically, + /lib/firmware. + reset_time - How long to wait for a reset to take effect + (default:2) + ramcheck_time - How many seconds to wait for the RAM test + (default:20) + osrun_time - How many seconds to wait for the ICS2115 OS + (default:10) + This module supports multiple cards and ISA PnP. + Note: the firmware file "wavefront.os" was located in the earlier + version in /etc. Now it's loaded via firmware loader, and + must be in the proper firmware path, such as /lib/firmware. + Copy (or symlink) the file appropriately if you get an error + regarding firmware downloading after upgrading the kernel. + Module snd-sonicvibes --------------------- diff -ruN linux-2.6.22/Documentation/sound/alsa/Audiophile-Usb.txt linux-2.6.22-alsa/Documentation/sound/alsa/Audiophile-Usb.txt --- linux-2.6.22/Documentation/sound/alsa/Audiophile-Usb.txt 2007-07-28 13:40:44.000000000 +0200 +++ linux-2.6.22-alsa/Documentation/sound/alsa/Audiophile-Usb.txt 2007-09-01 20:57:16.000000000 +0200 @@ -1,4 +1,4 @@ - Guide to using M-Audio Audiophile USB with ALSA and Jack v1.3 + Guide to using M-Audio Audiophile USB with ALSA and Jack v1.5 ======================================================== Thibault Le Meur @@ -6,8 +6,19 @@ This document is a guide to using the M-Audio Audiophile USB (tm) device with ALSA and JACK. +History +======= +* v1.4 - Thibault Le Meur (2007-07-11) + - Added Low Endianness nature of 16bits-modes + found by Hakan Lennestal + - Modifying document structure +* v1.5 - Thibault Le Meur (2007-07-12) + - Added AC3/DTS passthru info + + 1 - Audiophile USB Specs and correct usage ========================================== + This part is a reminder of important facts about the functions and limitations of the device. @@ -25,18 +36,18 @@ The internal DAC/ADC has the following characteristics: * sample depth of 16 or 24 bits * sample rate from 8kHz to 96kHz -* Two ports can't use different sample depths at the same time. Moreover, the -Audiophile USB documentation gives the following Warning: "Please exit any -audio application running before switching between bit depths" +* Two interfaces can't use different sample depths at the same time. +Moreover, the Audiophile USB documentation gives the following Warning: +"Please exit any audio application running before switching between bit depths" Due to the USB 1.1 bandwidth limitation, a limited number of interfaces can be activated at the same time depending on the audio mode selected: - * 16-bit/48kHz ==> 4 channels in/4 channels out + * 16-bit/48kHz ==> 4 channels in + 4 channels out - Ai+Ao+Di+Do - * 24-bit/48kHz ==> 4 channels in/2 channels out, - or 2 channels in/4 channels out + * 24-bit/48kHz ==> 4 channels in + 2 channels out, + or 2 channels in + 4 channels out - Ai+Ao+Do or Ai+Di+Ao or Ai+Di+Do or Di+Ao+Do - * 24-bit/96kHz ==> 2 channels in, or 2 channels out (half duplex only) + * 24-bit/96kHz ==> 2 channels in _or_ 2 channels out (half duplex only) - Ai or Ao or Di or Do Important facts about the Digital interface: @@ -52,44 +63,56 @@ synchronization error (for instance sound played at an odd sample rate) -2 - Audiophile USB support in ALSA -================================== +2 - Audiophile USB MIDI support in ALSA +======================================= -2.1 - MIDI ports ----------------- -The Audiophile USB MIDI ports will be automatically supported once the +The Audiophile USB MIDI ports will be automatically supported once the following modules have been loaded: * snd-usb-audio * snd-seq-midi No additional setting is required. -2.2 - Audio ports ------------------ + +3 - Audiophile USB Audio support in ALSA +======================================== Audio functions of the Audiophile USB device are handled by the snd-usb-audio module. This module can work in a default mode (without any device-specific parameter), or in an "advanced" mode with the device-specific parameter called "device_setup". -2.2.1 - Default Alsa driver mode - -The default behavior of the snd-usb-audio driver is to parse the device -capabilities at startup and enable all functions inside the device (including -all ports at any supported sample rates and sample depths). This approach -has the advantage to let the driver easily switch from sample rates/depths -automatically according to the need of the application claiming the device. +3.1 - Default Alsa driver mode +------------------------------ -In this case the Audiophile ports are mapped to alsa pcm devices in the -following way (I suppose the device's index is 1): +The default behavior of the snd-usb-audio driver is to list the device +capabilities at startup and activate the required mode when required +by the applications: for instance if the user is recording in a +24bit-depth-mode and immediately after wants to switch to a 16bit-depth mode, +the snd-usb-audio module will reconfigure the device on the fly. + +This approach has the advantage to let the driver automatically switch from sample +rates/depths automatically according to the user's needs. However, those who +are using the device under windows know that this is not how the device is meant to +work: under windows applications must be closed before using the m-audio control +panel to switch the device working mode. Thus as we'll see in next section, this +Default Alsa driver mode can lead to device misconfigurations. + +Let's get back to the Default Alsa driver mode for now. In this case the +Audiophile interfaces are mapped to alsa pcm devices in the following +way (I suppose the device's index is 1): * hw:1,0 is Ao in playback and Di in capture * hw:1,1 is Do in playback and Ai in capture * hw:1,2 is Do in AC3/DTS passthrough mode -You must note as well that the device uses Big Endian byte encoding so that -supported audio format are S16_BE for 16-bit depth modes and S24_3BE for -24-bits depth mode. One exception is the hw:1,2 port which is Little Endian -compliant and thus uses S16_LE. +In this mode, the device uses Big Endian byte-encoding so that +supported audio format are S16_BE for 16-bit depth modes and S24_3BE for +24-bits depth mode. + +One exception is the hw:1,2 port which was reported to be Little Endian +compliant (supposedly supporting S16_LE) but processes in fact only S16_BE streams. +This has been fixed in kernel 2.6.23 and above and now the hw:1,2 interface +is reported to be big endian in this default driver mode. Examples: * playing a S24_3BE encoded raw file to the Ao port @@ -98,22 +121,26 @@ % arecord -D hw:1,1 -c2 -t raw -r48000 -fS24_3BE test.raw * playing a S16_BE encoded raw file to the Do port % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_BE test.raw + * playing an ac3 sample file to the Do port + % aplay -D hw:1,2 --channels=6 ac3_S16_BE_encoded_file.raw -If you're happy with the default Alsa driver setup and don't experience any +If you're happy with the default Alsa driver mode and don't experience any issue with this mode, then you can skip the following chapter. -2.2.2 - Advanced module setup +3.2 - Advanced module setup +--------------------------- Due to the hardware constraints described above, the device initialization made by the Alsa driver in default mode may result in a corrupted state of the device. For instance, a particularly annoying issue is that the sound captured -from the Ai port sounds distorted (as if boosted with an excessive high volume -gain). +from the Ai interface sounds distorted (as if boosted with an excessive high +volume gain). For people having this problem, the snd-usb-audio module has a new module -parameter called "device_setup". +parameter called "device_setup" (this parameter was introduced in kernel +release 2.6.17) -2.2.2.1 - Initializing the working mode of the Audiophile USB +3.2.1 - Initializing the working mode of the Audiophile USB As far as the Audiophile USB device is concerned, this value let the user specify: @@ -121,33 +148,57 @@ * the sample rate * whether the Di port is used or not -Here is a list of supported device_setup values for this device: - * device_setup=0x00 (or omitted) - - Alsa driver default mode - - maintains backward compatibility with setups that do not use this - parameter by not introducing any change - - results sometimes in corrupted sound as described earlier +When initialized with "device_setup=0x00", the snd-usb-audio module has +the same behaviour as when the parameter is omitted (see paragraph "Default +Alsa driver mode" above) + +Others modes are described in the following subsections. + +3.2.1.1 - 16-bit modes + +The two supported modes are: + * device_setup=0x01 - 16bits 48kHz mode with Di disabled - Ai,Ao,Do can be used at the same time - hw:1,0 is not available in capture mode - hw:1,2 is not available + * device_setup=0x11 - 16bits 48kHz mode with Di enabled - Ai,Ao,Di,Do can be used at the same time - hw:1,0 is available in capture mode - hw:1,2 is not available + +In this modes the device operates only at 16bits-modes. Before kernel 2.6.23, +the devices where reported to be Big-Endian when in fact they were Little-Endian +so that playing a file was a matter of using: + % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_BE test_S16_LE.raw +where "test_S16_LE.raw" was in fact a little-endian sample file. + +Thanks to Hakan Lennestal (who discovered the Little-Endiannes of the device in +these modes) a fix has been committed (expected in kernel 2.6.23) and +Alsa now reports Little-Endian interfaces. Thus playing a file now is as simple as +using: + % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_LE test_S16_LE.raw + +3.2.1.2 - 24-bit modes + +The three supported modes are: + * device_setup=0x09 - 24bits 48kHz mode with Di disabled - Ai,Ao,Do can be used at the same time - hw:1,0 is not available in capture mode - hw:1,2 is not available + * device_setup=0x19 - 24bits 48kHz mode with Di enabled - 3 ports from {Ai,Ao,Di,Do} can be used at the same time - hw:1,0 is available in capture mode and an active digital source must be connected to Di - hw:1,2 is not available + * device_setup=0x0D or 0x10 - 24bits 96kHz mode - Di is enabled by default for this mode but does not need to be connected @@ -155,34 +206,64 @@ - Only 1 port from {Ai,Ao,Di,Do} can be used at the same time - hw:1,0 is available in captured mode - hw:1,2 is not available + +In these modes the device is only Big-Endian compliant (see "Default Alsa driver +mode" above for an aplay command example) + +3.2.1.3 - AC3 w/ DTS passthru mode + +Thanks to Hakan Lennestal, I now have a report saying that this mode works. + * device_setup=0x03 - 16bits 48kHz mode with only the Do port enabled - - AC3 with DTS passthru (not tested) + - AC3 with DTS passthru - Caution with this setup the Do port is mapped to the pcm device hw:1,0 -2.2.2.2 - Setting and switching configurations with the device_setup parameter +The command line used to playback the AC3/DTS encoded .wav-files in this mode: + % aplay -D hw:1,0 --channels=6 ac3_S16_LE_encoded_file.raw + +3.2.2 - How to use the device_setup parameter +---------------------------------------------- The parameter can be given: + * By manually probing the device (as root): # modprobe -r snd-usb-audio # modprobe snd-usb-audio index=1 device_setup=0x09 + * Or while configuring the modules options in your modules configuration file - For Fedora distributions, edit the /etc/modprobe.conf file: alias snd-card-1 snd-usb-audio options snd-usb-audio index=1 device_setup=0x09 -IMPORTANT NOTE WHEN SWITCHING CONFIGURATION: -------------------------------------------- - * You may need to _first_ initialize the module with the correct device_setup - parameter and _only_after_ turn on the Audiophile USB device - * This is especially true when switching the sample depth: +CAUTION when initializaing the device +------------------------------------- + + * Correct initialization on the device requires that device_setup is given to + the module BEFORE the device is turned on. So, if you use the "manual probing" + method described above, take care to power-on the device AFTER this initialization. + + * Failing to respect this will lead in a misconfiguration of the device. In this case + turn off the device, unproble the snd-usb-audio module, then probe it again with + correct device_setup parameter and then (and only then) turn on the device again. + + * If you've correctly initialized the device in a valid mode and then want to switch + to another mode (possibly with another sample-depth), please use also the following + procedure: - first turn off the device - de-register the snd-usb-audio module (modprobe -r) - change the device_setup parameter by changing the device_setup option in /etc/modprobe.conf - turn on the device + * A workaround for this last issue has been applied to kernel 2.6.23, but it may not + be enough to ensure the 'stability' of the device initialization. -2.2.2.3 - Audiophile USB's device_setup structure +3.2.3 - Technical details for hackers +------------------------------------- +This section is for hackers, wanting to understand details about the device +internals and how Alsa supports it. + +3.2.3.1 - Audiophile USB's device_setup structure If you want to understand the device_setup magic numbers for the Audiophile USB, you need some very basic understanding of binary computation. However, @@ -228,12 +309,12 @@ - choosing b2 will prepare all interfaces for 24bits/96kHz but you'll only be able to use one at the same time -2.2.3 - USB implementation details for this device +3.2.3.2 - USB implementation details for this device You may safely skip this section if you're not interested in driver -development. +hacking. -This section describes some internal aspects of the device and summarize the +This section describes some internal aspects of the device and summarizes the data I got by usb-snooping the windows and Linux drivers. The M-Audio Audiophile USB has 7 USB Interfaces: @@ -293,43 +374,45 @@ "audiophile_skip_setting_quirk" in order to prevent AltSettings not corresponding to device_setup from being registered in the driver. -3 - Audiophile USB and Jack support +4 - Audiophile USB and Jack support =================================== This section deals with support of the Audiophile USB device in Jack. -The main issue regarding this support is that the device is Big Endian -compliant. -3.1 - Using the plug alsa plugin --------------------------------- +There are 2 main potential issues when using Jackd with the device: +* support for Big-Endian devices in 24-bit modes +* support for 4-in / 4-out channels + +4.1 - Direct support in Jackd +----------------------------- + +Jack supports big endian devices only in recent versions (thanks to +Andreas Steinmetz for his first big-endian patch). I can't remember +extacly when this support was released into jackd, let's just say that +with jackd version 0.103.0 it's almost ok (just a small bug is affecting +16bits Big-Endian devices, but since you've read carefully the above +paragraphs, you're now using kernel >= 2.6.23 and your 16bits devices +are now Little Endians ;-) ). -Jack doesn't directly support big endian devices. Thus, one way to have support -for this device with Alsa is to use the Alsa "plug" converter. +You can run jackd with the following command for playback with Ao and +record with Ai: + % jackd -R -dalsa -Phw:1,0 -r48000 -p128 -n2 -D -Chw:1,1 + +4.2 - Using Alsa plughw +----------------------- +If you don't have a recent Jackd installed, you can downgrade to using +the Alsa "plug" converter. For instance here is one way to run Jack with 2 playback channels on Ao and 2 capture channels from Ai: % jackd -R -dalsa -dplughw:1 -r48000 -p256 -n2 -D -Cplughw:1,1 - However you may see the following warning message: "You appear to be using the ALSA software "plug" layer, probably a result of using the "default" ALSA device. This is less efficient than it could be. Consider using a hardware device instead rather than using the plug layer." -3.2 - Patching alsa to use direct pcm device --------------------------------------------- -A patch for Jack by Andreas Steinmetz adds support for Big Endian devices. -However it has not been included in the CVS tree. - -You can find it at the following URL: -http://sourceforge.net/tracker/index.php?func=detail&aid=1289682&group_id=39687& -atid=425939 - -After having applied the patch you can run jackd with the following command -line: - % jackd -R -dalsa -Phw:1,0 -r48000 -p128 -n2 -D -Chw:1,1 - -3.2 - Getting 2 input and/or output interfaces in Jack +4.3 - Getting 2 input and/or output interfaces in Jack ------------------------------------------------------ As you can see, starting the Jack server this way will only enable 1 stereo @@ -339,6 +422,7 @@ * Jack can only open one capture device and one playback device at a time * The Audiophile USB is seen as 2 (or three) Alsa devices: hw:1,0, hw:1,1 (and optionally hw:1,2) + If you want to get Ai+Di and/or Ao+Do support with Jack, you would need to combine the Alsa devices into one logical "complex" device. @@ -348,13 +432,11 @@ the Audiophile USB. Enabling multiple Audiophile USB interfaces for Jackd will certainly require: -* patching Jack with the previously mentioned "Big Endian" patch -* patching Jackd with the MMAP_COMPLEX patch (see the ice1712 page) -* patching the alsa-lib/src/pcm/pcm_multi.c file (see the ice1712 page) +* Making sure your Jackd version has the MMAP_COMPLEX patch (see the ice1712 page) +* (maybe) patching the alsa-lib/src/pcm/pcm_multi.c file (see the ice1712 page) * define a multi device (combination of hw:1,0 and hw:1,1) in your .asoundrc file * start jackd with this device -I had no success in testing this for now, but this may be due to my OS -configuration. If you have any success with this kind of setup, please -drop me an email. +I had no success in testing this for now, if you have any success with this kind +of setup, please drop me an email. diff -ruN linux-2.6.22/Documentation/sound/alsa/CMIPCI.txt linux-2.6.22-alsa/Documentation/sound/alsa/CMIPCI.txt --- linux-2.6.22/Documentation/sound/alsa/CMIPCI.txt 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/Documentation/sound/alsa/CMIPCI.txt 2007-09-01 20:57:16.000000000 +0200 @@ -1,5 +1,5 @@ - Brief Notes on C-Media 8738/8338 Driver - ======================================= + Brief Notes on C-Media 8338/8738/8768/8770 Driver + ================================================= Takashi Iwai @@ -212,7 +212,9 @@ The MPU401-UART interface is disabled as default. You need to set module option "mpu_port" with a valid I/O port address to enable the MIDI support. The valid I/O ports are 0x300, 0x310, 0x320 and 0x330. -Choose the value which doesn't conflict with other cards. +Choose the value which doesn't conflict with other cards. With +CMI8738 and newer chips, you can use "mpu_port=1" to use a PCI port +address that does not conflict with any other card. There is _no_ hardware wavetable function on this chip (except for OPL3 synth below). @@ -230,6 +232,8 @@ The output quality of FM OPL/3 is, however, very weird. I don't know why.. +CMI8768 and newer chips do not have the FM synth. + Joystick and Modem ------------------ diff -ruN linux-2.6.22/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl linux-2.6.22-alsa/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl --- linux-2.6.22/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl 2007-09-01 20:57:16.000000000 +0200 @@ -18,8 +18,8 @@ - November 17, 2005 - 0.3.6 + July 26, 2007 + 0.3.6.1 @@ -405,8 +405,9 @@ /* definition of the chip-specific record */ struct mychip { struct snd_card *card; - // rest of implementation will be in the section - // "PCI Resource Managements" + /* rest of implementation will be in the section + * "PCI Resource Managements" + */ }; /* chip-specific destructor @@ -414,7 +415,7 @@ */ static int snd_mychip_free(struct mychip *chip) { - .... // will be implemented later... + .... /* will be implemented later... */ } /* component-destructor @@ -440,8 +441,9 @@ *rchip = NULL; - // check PCI availability here - // (see "PCI Resource Managements") + /* check PCI availability here + * (see "PCI Resource Managements") + */ .... /* allocate a chip-specific data with zero filled */ @@ -451,12 +453,13 @@ chip->card = card; - // rest of initialization here; will be implemented - // later, see "PCI Resource Managements" + /* rest of initialization here; will be implemented + * later, see "PCI Resource Managements" + */ .... - if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, - chip, &ops)) < 0) { + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { snd_mychip_free(chip); return err; } @@ -490,7 +493,8 @@ return -ENOMEM; /* (3) */ - if ((err = snd_mychip_create(card, pci, &chip)) < 0) { + err = snd_mychip_create(card, pci, &chip); + if (err < 0) { snd_card_free(card); return err; } @@ -502,10 +506,11 @@ card->shortname, chip->ioport, chip->irq); /* (5) */ - .... // implemented later + .... /* implemented later */ /* (6) */ - if ((err = snd_card_register(card)) < 0) { + err = snd_card_register(card); + if (err < 0) { snd_card_free(card); return err; } @@ -605,7 +610,8 @@ irq >= 0) @@ -1119,7 +1126,8 @@ *rchip = NULL; /* initialize the PCI entry */ - if ((err = pci_enable_device(pci)) < 0) + err = pci_enable_device(pci); + if (err < 0) return err; /* check PCI availability (28bit DMA) */ if (pci_set_dma_mask(pci, DMA_28BIT_MASK) < 0 || @@ -1141,7 +1149,8 @@ chip->irq = -1; /* (1) PCI resource allocation */ - if ((err = pci_request_regions(pci, "My Chip")) < 0) { + err = pci_request_regions(pci, "My Chip"); + if (err < 0) { kfree(chip); pci_disable_device(pci); return err; @@ -1156,10 +1165,10 @@ chip->irq = pci->irq; /* (2) initialization of the chip hardware */ - .... // (not implemented in this document) + .... /* (not implemented in this document) */ - if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, - chip, &ops)) < 0) { + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { snd_mychip_free(chip); return err; } @@ -1233,7 +1242,8 @@ irq, snd_mychip_interrupt, - IRQF_DISABLED|IRQF_SHARED, "My Chip", chip)) { + IRQF_SHARED, "My Chip", chip)) { printk(KERN_ERR "cannot grab irq %d\n", pci->irq); snd_mychip_free(chip); return -EBUSY; @@ -1773,7 +1784,8 @@ struct snd_pcm_runtime *runtime = substream->runtime; runtime->hw = snd_mychip_playback_hw; - // more hardware-initialization will be done here + /* more hardware-initialization will be done here */ + .... return 0; } @@ -1781,7 +1793,8 @@ static int snd_mychip_playback_close(struct snd_pcm_substream *substream) { struct mychip *chip = snd_pcm_substream_chip(substream); - // the hardware-specific codes will be here + /* the hardware-specific codes will be here */ + .... return 0; } @@ -1793,7 +1806,8 @@ struct snd_pcm_runtime *runtime = substream->runtime; runtime->hw = snd_mychip_capture_hw; - // more hardware-initialization will be done here + /* more hardware-initialization will be done here */ + .... return 0; } @@ -1801,7 +1815,8 @@ static int snd_mychip_capture_close(struct snd_pcm_substream *substream) { struct mychip *chip = snd_pcm_substream_chip(substream); - // the hardware-specific codes will be here + /* the hardware-specific codes will be here */ + .... return 0; } @@ -1844,10 +1859,12 @@ { switch (cmd) { case SNDRV_PCM_TRIGGER_START: - // do something to start the PCM engine + /* do something to start the PCM engine */ + .... break; case SNDRV_PCM_TRIGGER_STOP: - // do something to stop the PCM engine + /* do something to stop the PCM engine */ + .... break; default: return -EINVAL; @@ -1900,8 +1917,8 @@ struct snd_pcm *pcm; int err; - if ((err = snd_pcm_new(chip->card, "My Chip", 0, 1, 1, - &pcm)) < 0) + err = snd_pcm_new(chip->card, "My Chip", 0, 1, 1, &pcm); + if (err < 0) return err; pcm->private_data = chip; strcpy(pcm->name, "My Chip"); @@ -1939,8 +1956,8 @@ struct snd_pcm *pcm; int err; - if ((err = snd_pcm_new(chip->card, "My Chip", 0, 1, 1, - &pcm)) < 0) + err = snd_pcm_new(chip->card, "My Chip", 0, 1, 1, &pcm); + if (err < 0) return err; pcm->private_data = chip; strcpy(pcm->name, "My Chip"); @@ -2097,7 +2114,7 @@ struct mychip *chip = snd_pcm_chip(pcm); /* free your own data */ kfree(chip->my_private_pcm_data); - // do what you like else + /* do what you like else */ .... } @@ -2884,10 +2901,10 @@ lock); snd_pcm_period_elapsed(chip->substream); spin_lock(&chip->lock); - // acknowledge the interrupt if necessary + /* acknowledge the interrupt if necessary */ } .... spin_unlock(&chip->lock); @@ -3134,7 +3151,7 @@ snd_pcm_period_elapsed(substream); spin_lock(&chip->lock); } - // acknowledge the interrupt if necessary + /* acknowledge the interrupt if necessary */ } .... spin_unlock(&chip->lock); @@ -3604,7 +3621,7 @@ Example of info callback type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; @@ -3639,7 +3656,7 @@ + + + Some common info callbacks are prepared for easy use: + snd_ctl_boolean_mono_info() and + snd_ctl_boolean_stereo_info(). + Obviously, the former is an info callback for a mono channel + boolean item, just like snd_myctl_mono_info + above, and the latter is for a stereo channel boolean item. + +
@@ -3794,7 +3821,8 @@ @@ -3880,7 +3908,7 @@ { struct mychip *chip = ac97->private_data; .... - // read a register value here from the codec + /* read a register value here from the codec */ return the_register_value; } @@ -3889,7 +3917,7 @@ { struct mychip *chip = ac97->private_data; .... - // write the given register value to the codec + /* write the given register value to the codec */ } static int snd_mychip_ac97(struct mychip *chip) @@ -3902,7 +3930,8 @@ .read = snd_mychip_ac97_read, }; - if ((err = snd_ac97_bus(chip->card, 0, &ops, NULL, &bus)) < 0) + err = snd_ac97_bus(chip->card, 0, &ops, NULL, &bus); + if (err < 0) return err; memset(&ac97, 0, sizeof(ac97)); ac97.private_data = chip; @@ -4447,10 +4476,10 @@ streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substreams) { - substream = list_entry(list, struct snd_rawmidi_substream, list); + list_for_each_entry(substream, + &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substreams, + list { sprintf(substream->name, "My MIDI Port %d", substream->number + 1); } /* same for SNDRV_RAWMIDI_STREAM_INPUT */ diff -ruN linux-2.6.22/Documentation/sound/alsa/OSS-Emulation.txt linux-2.6.22-alsa/Documentation/sound/alsa/OSS-Emulation.txt --- linux-2.6.22/Documentation/sound/alsa/OSS-Emulation.txt 2007-07-28 13:40:44.000000000 +0200 +++ linux-2.6.22-alsa/Documentation/sound/alsa/OSS-Emulation.txt 2007-09-01 20:57:16.000000000 +0200 @@ -278,6 +278,21 @@ image. +Duplex Streams +============== + +Note that when attempting to use a single device file for playback and +capture, the OSS API provides no way to set the format, sample rate or +number of channels different in each direction. Thus + io_handle = open("device", O_RDWR) +will only function correctly if the values are the same in each direction. + +To use different values in the two directions, use both + input_handle = open("device", O_RDONLY) + output_handle = open("device", O_WRONLY) +and set the values for the corresponding handle. + + Unsupported Features ==================== @@ -288,10 +303,3 @@ the buffer as the conventional (mono or 2-channels, 8 or 16bit) format on OSS. -USB devices ------------ -Some USB devices support only 24bit format packed in 3bytes. This -format is not supported by OSS and no conversion is provided by kernel -OSS emulation. You can use the user-space OSS emulation via libaoss -instead. - diff -ruN linux-2.6.22/Documentation/sound/alsa/hda_codec.txt linux-2.6.22-alsa/Documentation/sound/alsa/hda_codec.txt --- linux-2.6.22/Documentation/sound/alsa/hda_codec.txt 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/Documentation/sound/alsa/hda_codec.txt 2007-09-01 20:57:16.000000000 +0200 @@ -49,6 +49,9 @@ unsigned int verb, unsigned int parm); unsigned int (*get_response)(struct hda_codec *codec); void (*private_free)(struct hda_bus *); +#ifdef CONFIG_SND_HDA_POWER_SAVE + void (*pm_notify)(struct hda_codec *codec); +#endif }; The command callback is called when the codec module needs to send a @@ -56,9 +59,16 @@ The get_response callback is called when the codec requires the answer for the last command. These two callbacks are mandatory and have to be given. -The last, private_free callback, is optional. It's called in the +The third, private_free callback, is optional. It's called in the destructor to release any necessary data in the lowlevel driver. +The pm_notify callback is available only with +CONFIG_SND_HDA_POWER_SAVE kconfig. It's called when the codec needs +to power up or may power down. The controller should check the all +belonging codecs on the bus whether they are actually powered off +(check codec->power_on), and optionally the driver may power down the +contoller side, too. + The bus instance is created via snd_hda_bus_new(). You need to pass the card instance, the template, and the pointer to store the resultant bus instance. @@ -86,10 +96,8 @@ The codec is stored in a linked list of bus instance. You can follow the codec list like: - struct list_head *p; struct hda_codec *codec; - list_for_each(p, &bus->codec_list) { - codec = list_entry(p, struct hda_codec, list); + list_for_each_entry(codec, &bus->codec_list, list) { ... } @@ -100,10 +108,15 @@ Codec Access ============ -To access codec, use snd_codec_read() and snd_codec_write(). +To access codec, use snd_hda_codec_read() and snd_hda_codec_write(). snd_hda_param_read() is for reading parameters. For writing a sequence of verbs, use snd_hda_sequence_write(). +There are variants of cached read/write, snd_hda_codec_write_cache(), +snd_hda_sequence_write_cache(). These are used for recording the +register states for the power-mangement resume. When no PM is needed, +these are equivalent with non-cached version. + To retrieve the number of sub nodes connected to the given node, use snd_hda_get_sub_nodes(). The connection list can be obtained via snd_hda_get_connections() call. @@ -239,6 +252,10 @@ int (*suspend)(struct hda_codec *codec, pm_message_t state); int (*resume)(struct hda_codec *codec); #endif + #ifdef CONFIG_SND_HDA_POWER_SAVE + int (*check_power_status)(struct hda_codec *codec, + hda_nid_t nid); + #endif }; The build_controls callback is called from snd_hda_build_controls(). @@ -251,6 +268,18 @@ received. The suspend and resume callbacks are for power management. +They can be NULL if no special sequence is required. When the resume +callback is NULL, the driver calls the init callback and resumes the +registers from the cache. If other handling is needed, you'd need to +write your own resume callback. There, the amp values can be resumed +via + void snd_hda_codec_resume_amp(struct hda_codec *codec); +and the other codec registers via + void snd_hda_codec_resume_cache(struct hda_codec *codec); + +The check_power_status callback is called when the amp value of the +given widget NID is changed. The codec code can turn on/off the power +appropriately from this information. Each entry can be NULL if not necessary to be called. @@ -267,8 +296,7 @@ =========== Call snd_hda_create_spdif_out_ctls() from the patch to create controls -related with SPDIF out. In the patch resume callback, call -snd_hda_resume_spdif(). +related with SPDIF out. Helper Functions @@ -284,12 +312,7 @@ is found, it returns the config field value. snd_hda_add_new_ctls() can be used to create and add control entries. -Pass the zero-terminated array of struct snd_kcontrol_new. The same array -can be passed to snd_hda_resume_ctls() for resume. -Note that this will call control->put callback of these entries. So, -put callback should check codec->in_resume and force to restore the -given value if it's non-zero even if the value is identical with the -cached value. +Pass the zero-terminated array of struct snd_kcontrol_new Macros HDA_CODEC_VOLUME(), HDA_CODEC_MUTE() and their variables can be used for the entry of struct snd_kcontrol_new. diff -ruN linux-2.6.22/include/sound/ak4xxx-adda.h linux-2.6.22-alsa/include/sound/ak4xxx-adda.h --- linux-2.6.22/include/sound/ak4xxx-adda.h 2007-07-28 13:40:44.000000000 +0200 +++ linux-2.6.22-alsa/include/sound/ak4xxx-adda.h 2007-09-01 20:56:34.000000000 +0200 @@ -43,6 +43,7 @@ struct snd_akm4xxx_dac_channel { char *name; /* mixer volume name */ unsigned int num_channels; + char *switch_name; /* mixer switch*/ }; /* ADC labels and channels */ diff -ruN linux-2.6.22/include/sound/asound.h linux-2.6.22-alsa/include/sound/asound.h --- linux-2.6.22/include/sound/asound.h 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/include/sound/asound.h 2007-09-01 20:56:34.000000000 +0200 @@ -92,6 +92,7 @@ SNDRV_HWDEP_IFACE_USX2Y_PCM, /* Tascam US122, US224 & US428 rawusb pcm */ SNDRV_HWDEP_IFACE_PCXHR, /* Digigram PCXHR */ SNDRV_HWDEP_IFACE_SB_RC, /* SB Extigy/Audigy2NX remote control */ + SNDRV_HWDEP_IFACE_HDA, /* HD-audio */ /* Don't forget to change the following: */ SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_SB_RC diff -ruN linux-2.6.22/include/sound/control.h linux-2.6.22-alsa/include/sound/control.h --- linux-2.6.22/include/sound/control.h 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/include/sound/control.h 2007-09-01 20:56:34.000000000 +0200 @@ -161,4 +161,12 @@ return dst_id; } +/* + * Frequently used control callbacks + */ +int snd_ctl_boolean_mono_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo); +int snd_ctl_boolean_stereo_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo); + #endif /* __SOUND_CONTROL_H */ diff -ruN linux-2.6.22/include/sound/cs4231.h linux-2.6.22-alsa/include/sound/cs4231.h --- linux-2.6.22/include/sound/cs4231.h 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/include/sound/cs4231.h 2007-09-01 20:56:34.000000000 +0200 @@ -210,7 +210,7 @@ #define CS4231_HW_CS4239 0x0404 /* CS4239 - Crystal Clear (tm) stereo enhancement */ /* compatible, but clones */ #define CS4231_HW_INTERWAVE 0x1000 /* InterWave chip */ -#define CS4231_HW_OPL3SA2 0x1001 /* OPL3-SA2 chip */ +#define CS4231_HW_OPL3SA2 0x1101 /* OPL3-SA2 chip, similar to cs4231 */ /* defines for codec.hwshare */ #define CS4231_HWSHARE_IRQ (1<<0) diff -ruN linux-2.6.22/include/sound/cs46xx.h linux-2.6.22-alsa/include/sound/cs46xx.h --- linux-2.6.22/include/sound/cs46xx.h 2007-07-28 13:40:44.000000000 +0200 +++ linux-2.6.22-alsa/include/sound/cs46xx.h 2007-09-01 20:56:34.000000000 +0200 @@ -1723,6 +1723,10 @@ struct snd_cs46xx_pcm *playback_pcm; unsigned int play_ctl; #endif + +#ifdef CONFIG_PM + u32 *saved_regs; +#endif }; int snd_cs46xx_create(struct snd_card *card, diff -ruN linux-2.6.22/include/sound/cs46xx_dsp_spos.h linux-2.6.22-alsa/include/sound/cs46xx_dsp_spos.h --- linux-2.6.22/include/sound/cs46xx_dsp_spos.h 2007-07-28 13:40:44.000000000 +0200 +++ linux-2.6.22-alsa/include/sound/cs46xx_dsp_spos.h 2007-09-01 20:56:34.000000000 +0200 @@ -107,6 +107,7 @@ char scb_name[DSP_MAX_SCB_NAME]; u32 address; int index; + u32 *data; struct dsp_scb_descriptor * sub_list_ptr; struct dsp_scb_descriptor * next_scb_ptr; @@ -127,6 +128,7 @@ int size; u32 address; int index; + u32 *data; }; struct dsp_pcm_channel_descriptor { diff -ruN linux-2.6.22/include/sound/emu10k1.h linux-2.6.22-alsa/include/sound/emu10k1.h --- linux-2.6.22/include/sound/emu10k1.h 2007-07-28 13:40:44.000000000 +0200 +++ linux-2.6.22-alsa/include/sound/emu10k1.h 2007-09-01 20:56:34.000000000 +0200 @@ -1120,6 +1120,16 @@ /************************************************************************************************/ /* EMU1010m HANA Destinations */ /************************************************************************************************/ +/* 32-bit destinations of signal in the Hana FPGA. Destinations are either + * physical outputs of Hana, or outputs going to Alice2 (audigy) for capture + * - 16 x EMU_DST_ALICE2_EMU32_X. + */ +/* EMU32 = 32-bit serial channel between Alice2 (audigy) and Hana (FPGA) */ +/* EMU_DST_ALICE2_EMU32_X - data channels from Hana to Alice2 used for capture. + * Which data is fed into a EMU_DST_ALICE2_EMU32_X channel in Hana depends on + * setup of mixer control for each destination - see emumixer.c - + * snd_emu1010_output_enum_ctls[], snd_emu1010_input_enum_ctls[] + */ #define EMU_DST_ALICE2_EMU32_0 0x000f /* 16 EMU32 channels to Alice2 +0 to +0xf */ #define EMU_DST_ALICE2_EMU32_1 0x0000 /* 16 EMU32 channels to Alice2 +0 to +0xf */ #define EMU_DST_ALICE2_EMU32_2 0x0001 /* 16 EMU32 channels to Alice2 +0 to +0xf */ @@ -1199,6 +1209,12 @@ /************************************************************************************************/ /* EMU1010m HANA Sources */ /************************************************************************************************/ +/* 32-bit sources of signal in the Hana FPGA. The sources are routed to + * destinations using mixer control for each destination - see emumixer.c + * Sources are either physical inputs of FPGA, + * or outputs from Alice (audigy) - 16 x EMU_SRC_ALICE_EMU32A + + * 16 x EMU_SRC_ALICE_EMU32B + */ #define EMU_SRC_SILENCE 0x0000 /* Silence */ #define EMU_SRC_DOCK_MIC_A1 0x0100 /* Audio Dock Mic A, 1st or 48kHz only */ #define EMU_SRC_DOCK_MIC_A2 0x0101 /* Audio Dock Mic A, 2nd or 96kHz */ @@ -1440,6 +1456,9 @@ unsigned int adc_pads; /* bit mask */ unsigned int dac_pads; /* bit mask */ unsigned int internal_clock; /* 44100 or 48000 */ + unsigned int optical_in; /* 0:SPDIF, 1:ADAT */ + unsigned int optical_out; /* 0:SPDIF, 1:ADAT */ + struct task_struct *firmware_thread; }; struct snd_emu10k1 { @@ -1583,9 +1602,9 @@ void snd_emu10k1_ptr20_write(struct snd_emu10k1 *emu, unsigned int reg, unsigned int chn, unsigned int data); int snd_emu10k1_spi_write(struct snd_emu10k1 * emu, unsigned int data); int snd_emu10k1_i2c_write(struct snd_emu10k1 *emu, u32 reg, u32 value); -int snd_emu1010_fpga_write(struct snd_emu10k1 * emu, int reg, int value); -int snd_emu1010_fpga_read(struct snd_emu10k1 * emu, int reg, int *value); -int snd_emu1010_fpga_link_dst_src_write(struct snd_emu10k1 * emu, int dst, int src); +int snd_emu1010_fpga_write(struct snd_emu10k1 * emu, u32 reg, u32 value); +int snd_emu1010_fpga_read(struct snd_emu10k1 * emu, u32 reg, u32 *value); +int snd_emu1010_fpga_link_dst_src_write(struct snd_emu10k1 * emu, u32 dst, u32 src); unsigned int snd_emu10k1_efx_read(struct snd_emu10k1 *emu, unsigned int pc); void snd_emu10k1_intr_enable(struct snd_emu10k1 *emu, unsigned int intrenb); void snd_emu10k1_intr_disable(struct snd_emu10k1 *emu, unsigned int intrenb); @@ -1730,6 +1749,8 @@ #define A_FXBUS2(x) (0x80 + (x)) /* x = 0x00 - 0x1f extra outs used for EFX capture -> A_FXWC2 */ #define A_EMU32OUTH(x) (0xa0 + (x)) /* x = 0x00 - 0x0f "EMU32_OUT_10 - _1F" - ??? */ #define A_EMU32OUTL(x) (0xb0 + (x)) /* x = 0x00 - 0x0f "EMU32_OUT_1 - _F" - ??? */ +#define A3_EMU32IN(x) (0x160 + (x)) /* x = 0x00 - 0x3f "EMU32_IN_00 - _3F" - Only when .device = 0x0008 */ +#define A3_EMU32OUT(x) (0x1E0 + (x)) /* x = 0x00 - 0x0f "EMU32_OUT_00 - _3F" - Only when .device = 0x0008 */ #define A_GPR(x) (A_FXGPREGBASE + (x)) /* cc_reg constants */ diff -ruN linux-2.6.22/include/sound/hda_hwdep.h linux-2.6.22-alsa/include/sound/hda_hwdep.h --- linux-2.6.22/include/sound/hda_hwdep.h 1970-01-01 01:00:00.000000000 +0100 +++ linux-2.6.22-alsa/include/sound/hda_hwdep.h 2007-09-01 20:56:34.000000000 +0200 @@ -0,0 +1,44 @@ +/* + * HWDEP Interface for HD-audio codec + * + * Copyright (c) 2007 Takashi Iwai + * + * This driver is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef __SOUND_HDA_HWDEP_H +#define __SOUND_HDA_HWDEP_H + +#define HDA_HWDEP_VERSION ((1 << 16) | (0 << 8) | (0 << 0)) /* 1.0.0 */ + +/* verb */ +#define HDA_REG_NID_SHIFT 24 +#define HDA_REG_VERB_SHIFT 8 +#define HDA_REG_VAL_SHIFT 0 +#define HDA_VERB(nid,verb,param) ((nid)<<24 | (verb)<<8 | (param)) + +struct hda_verb_ioctl { + u32 verb; /* HDA_VERB() */ + u32 res; /* response */ +}; + +/* + * ioctls + */ +#define HDA_IOCTL_PVERSION _IOR('H', 0x10, int) +#define HDA_IOCTL_VERB_WRITE _IOWR('H', 0x11, struct hda_verb_ioctl) +#define HDA_IOCTL_GET_WCAP _IOWR('H', 0x12, struct hda_verb_ioctl) + +#endif diff -ruN linux-2.6.22/include/sound/hdspm.h linux-2.6.22-alsa/include/sound/hdspm.h --- linux-2.6.22/include/sound/hdspm.h 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/include/sound/hdspm.h 2007-09-01 20:56:34.000000000 +0200 @@ -1,4 +1,4 @@ -#ifndef __SOUND_HDSPM_H /* -*- linux-c -*- */ +#ifndef __SOUND_HDSPM_H #define __SOUND_HDSPM_H /* * Copyright (C) 2003 Winfried Ritsch (IEM) @@ -61,7 +61,8 @@ }; /* use indirect access due to the limit of ioctl bit size */ -#define SNDRV_HDSPM_IOCTL_GET_PEAK_RMS _IOR('H', 0x40, struct hdspm_peak_rms_ioctl) +#define SNDRV_HDSPM_IOCTL_GET_PEAK_RMS \ + _IOR('H', 0x40, struct hdspm_peak_rms_ioctl) /* ------------ CONFIG block IOCTL ---------------------- */ @@ -79,7 +80,8 @@ unsigned int analog_out; }; -#define SNDRV_HDSPM_IOCTL_GET_CONFIG_INFO _IOR('H', 0x41, struct hdspm_config_info) +#define SNDRV_HDSPM_IOCTL_GET_CONFIG_INFO \ + _IOR('H', 0x41, struct hdspm_config_info) /* get Soundcard Version */ @@ -93,10 +95,14 @@ /* ------------- get Matrix Mixer IOCTL --------------- */ -/* MADI mixer: 64inputs+64playback in 64outputs = 8192 => *4Byte = 32768 Bytes */ +/* MADI mixer: 64inputs+64playback in 64outputs = 8192 => *4Byte = + * 32768 Bytes + */ /* organisation is 64 channelfader in a continous memory block */ -/* equivalent to hardware definition, maybe for future feature of mmap of them */ +/* equivalent to hardware definition, maybe for future feature of mmap of + * them + */ /* each of 64 outputs has 64 infader and 64 outfader: Ins to Outs mixer[out].in[in], Outstreams to Outs mixer[out].pb[pb] */ diff -ruN linux-2.6.22/include/sound/pcm.h linux-2.6.22-alsa/include/sound/pcm.h --- linux-2.6.22/include/sound/pcm.h 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/include/sound/pcm.h 2007-09-01 20:56:34.000000000 +0200 @@ -922,7 +922,10 @@ snd_pcm_sframes_t snd_pcm_lib_readv(struct snd_pcm_substream *substream, void __user **bufs, snd_pcm_uframes_t frames); +extern const struct snd_pcm_hw_constraint_list snd_pcm_known_rates; + int snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime); +unsigned int snd_pcm_rate_to_rate_bit(unsigned int rate); static inline void snd_pcm_set_runtime_buffer(struct snd_pcm_substream *substream, struct snd_dma_buffer *bufp) diff -ruN linux-2.6.22/include/sound/sb.h linux-2.6.22-alsa/include/sound/sb.h --- linux-2.6.22/include/sound/sb.h 2007-07-28 13:40:44.000000000 +0200 +++ linux-2.6.22-alsa/include/sound/sb.h 2007-09-01 20:56:34.000000000 +0200 @@ -38,6 +38,7 @@ SB_HW_ALS100, /* Avance Logic ALS100 chip */ SB_HW_ALS4000, /* Avance Logic ALS4000 chip */ SB_HW_DT019X, /* Diamond Tech. DT-019X / Avance Logic ALS-007 */ + SB_HW_CS5530, /* Cyrix/NatSemi 5530 VSA1 */ }; #define SB_OPEN_PCM 0x01 diff -ruN linux-2.6.22/include/sound/soc.h linux-2.6.22-alsa/include/sound/soc.h --- linux-2.6.22/include/sound/soc.h 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/include/sound/soc.h 2007-09-01 20:56:34.000000000 +0200 @@ -201,8 +201,7 @@ struct snd_ctl_elem_info *uinfo); int snd_soc_info_volsw_ext(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); -int snd_soc_info_bool_ext(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo); +#define snd_soc_info_bool_ext snd_ctl_boolean_mono int snd_soc_get_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, diff -ruN linux-2.6.22/sound/Kconfig linux-2.6.22-alsa/sound/Kconfig --- linux-2.6.22/sound/Kconfig 2007-07-28 13:40:44.000000000 +0200 +++ linux-2.6.22-alsa/sound/Kconfig 2007-09-01 20:55:22.000000000 +0200 @@ -63,8 +63,14 @@ source "sound/arm/Kconfig" +if SPI +source "sound/spi/Kconfig" +endif + source "sound/mips/Kconfig" +source "sound/sh/Kconfig" + # the following will depend on the order of config. # here assuming USB is defined before ALSA source "sound/usb/Kconfig" diff -ruN linux-2.6.22/sound/Makefile linux-2.6.22-alsa/sound/Makefile --- linux-2.6.22/sound/Makefile 2007-07-28 13:40:44.000000000 +0200 +++ linux-2.6.22-alsa/sound/Makefile 2007-09-01 20:55:22.000000000 +0200 @@ -5,7 +5,8 @@ obj-$(CONFIG_SOUND_PRIME) += sound_firmware.o obj-$(CONFIG_SOUND_PRIME) += oss/ obj-$(CONFIG_DMASOUND) += oss/ -obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ synth/ usb/ sparc/ parisc/ pcmcia/ mips/ soc/ +obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ sh/ synth/ usb/ \ + sparc/ spi/ parisc/ pcmcia/ mips/ soc/ obj-$(CONFIG_SND_AOA) += aoa/ # This one must be compilable even if sound is configured out diff -ruN linux-2.6.22/sound/aoa/codecs/snd-aoa-codec-onyx.c linux-2.6.22-alsa/sound/aoa/codecs/snd-aoa-codec-onyx.c --- linux-2.6.22/sound/aoa/codecs/snd-aoa-codec-onyx.c 2007-07-28 13:40:44.000000000 +0200 +++ linux-2.6.22-alsa/sound/aoa/codecs/snd-aoa-codec-onyx.c 2007-09-01 20:55:22.000000000 +0200 @@ -297,15 +297,7 @@ .put = onyx_snd_capture_source_put, }; -static int onyx_snd_mute_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 2; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define onyx_snd_mute_info snd_ctl_boolean_stereo_info static int onyx_snd_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -359,15 +351,7 @@ }; -static int onyx_snd_single_bit_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define onyx_snd_single_bit_info snd_ctl_boolean_mono_info #define FLAG_POLARITY_INVERT 1 #define FLAG_SPDIFLOCK 2 @@ -661,7 +645,7 @@ .tag = 2, }, #ifdef SNDRV_PCM_FMTBIT_COMPRESSED_16BE -Once alsa gets supports for this kind of thing we can add it... + /* Once alsa gets supports for this kind of thing we can add it... */ { /* digital compressed output */ .formats = SNDRV_PCM_FMTBIT_COMPRESSED_16BE, @@ -713,7 +697,7 @@ if (substream->runtime->format == SNDRV_PCM_FMTBIT_COMPRESSED_16BE) { /* mute and lock analog output */ onyx_read_register(onyx, ONYX_REG_DAC_CONTROL, &v); - if (onyx_write_register(onyx + if (onyx_write_register(onyx, ONYX_REG_DAC_CONTROL, v | ONYX_MUTE_RIGHT | ONYX_MUTE_LEFT)) goto out_unlock; diff -ruN linux-2.6.22/sound/aoa/codecs/snd-aoa-codec-tas.c linux-2.6.22-alsa/sound/aoa/codecs/snd-aoa-codec-tas.c --- linux-2.6.22/sound/aoa/codecs/snd-aoa-codec-tas.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/aoa/codecs/snd-aoa-codec-tas.c 2007-09-01 20:55:22.000000000 +0200 @@ -272,15 +272,7 @@ .put = tas_snd_vol_put, }; -static int tas_snd_mute_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 2; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define tas_snd_mute_info snd_ctl_boolean_stereo_info static int tas_snd_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -431,15 +423,7 @@ .put = tas_snd_drc_range_put, }; -static int tas_snd_drc_switch_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define tas_snd_drc_switch_info snd_ctl_boolean_mono_info static int tas_snd_drc_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -743,6 +727,7 @@ return 0; } +#ifdef CONFIG_PM /* we are controlled via i2c and assume that is always up * If that wasn't the case, we'd have to suspend once * our i2c device is suspended, and then take note of that! */ @@ -768,7 +753,6 @@ return 0; } -#ifdef CONFIG_PM static int _tas_suspend(struct codec_info_item *cii, pm_message_t state) { return tas_suspend(cii->codec_data); @@ -778,7 +762,10 @@ { return tas_resume(cii->codec_data); } -#endif +#else /* CONFIG_PM */ +#define _tas_suspend NULL +#define _tas_resume NULL +#endif /* CONFIG_PM */ static struct codec_info tas_codec_info = { .transfers = tas_transfers, @@ -791,10 +778,8 @@ .owner = THIS_MODULE, .usable = tas_usable, .switch_clock = tas_switch_clock, -#ifdef CONFIG_PM .suspend = _tas_suspend, .resume = _tas_resume, -#endif }; static int tas_init_codec(struct aoa_codec *codec) diff -ruN linux-2.6.22/sound/aoa/fabrics/snd-aoa-fabric-layout.c linux-2.6.22-alsa/sound/aoa/fabrics/snd-aoa-fabric-layout.c --- linux-2.6.22/sound/aoa/fabrics/snd-aoa-fabric-layout.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/aoa/fabrics/snd-aoa-fabric-layout.c 2007-09-01 20:55:22.000000000 +0200 @@ -582,15 +582,7 @@ * make the fabric handle all the card stuff, etc... */ static struct layout_dev *layout_device; -static int control_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define control_info snd_ctl_boolean_mono_info #define AMP_CONTROL(n, description) \ static int n##_control_get(struct snd_kcontrol *kcontrol, \ diff -ruN linux-2.6.22/sound/arm/sa11xx-uda1341.c linux-2.6.22-alsa/sound/arm/sa11xx-uda1341.c --- linux-2.6.22/sound/arm/sa11xx-uda1341.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/arm/sa11xx-uda1341.c 2007-09-01 20:55:22.000000000 +0200 @@ -79,12 +79,6 @@ #include #include -#ifdef CONFIG_H3600_HAL -#include -#include -#include -#endif - #include #include #include @@ -100,9 +94,6 @@ * We use DMA stuff from 2.4.18-rmk3-hh24 here to be able to compile this * module for Familiar 0.6.1 */ -#ifdef CONFIG_H3600_HAL -#define HH_VERSION 1 -#endif /* {{{ Type definitions */ @@ -238,11 +229,8 @@ rate = 8000; /* Set the external clock generator */ -#ifdef CONFIG_H3600_HAL - h3600_audio_clock(rate); -#else + sa11xx_uda1341_set_audio_clock(rate); -#endif /* Select the clock divisor */ switch (rate) { @@ -307,13 +295,10 @@ local_irq_restore(flags); /* Enable the audio power */ -#ifdef CONFIG_H3600_HAL - h3600_audio_power(AUDIO_RATE_DEFAULT); -#else + clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET); set_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON); set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE); -#endif /* Wait for the UDA1341 to wake up */ mdelay(1); //FIXME - was removed by Perex - Why? @@ -331,21 +316,13 @@ /* make the left and right channels unswapped (flip the WS latch) */ Ser4SSDR = 0; -#ifdef CONFIG_H3600_HAL - h3600_audio_mute(0); -#else - clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE); -#endif + clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE); } static void sa11xx_uda1341_audio_shutdown(struct sa11xx_uda1341 *sa11xx_uda1341) { /* mute on */ -#ifdef CONFIG_H3600_HAL - h3600_audio_mute(1); -#else set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE); -#endif /* disable the audio power and all signals leading to the audio chip */ l3_close(sa11xx_uda1341->uda1341); @@ -354,13 +331,9 @@ /* power off and mute off */ /* FIXME - is muting off necesary??? */ -#ifdef CONFIG_H3600_HAL - h3600_audio_power(0); - h3600_audio_mute(0); -#else + clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON); clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE); -#endif } /* }}} */ diff -ruN linux-2.6.22/sound/core/Makefile linux-2.6.22-alsa/sound/core/Makefile --- linux-2.6.22/sound/core/Makefile 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/core/Makefile 2007-09-01 20:55:22.000000000 +0200 @@ -3,18 +3,15 @@ # Copyright (c) 1999,2001 by Jaroslav Kysela # -snd-objs := sound.o init.o memory.o info.o control.o misc.o device.o -ifeq ($(CONFIG_ISA_DMA_API),y) -snd-objs += isadma.o -endif -ifeq ($(CONFIG_SND_OSSEMUL),y) -snd-objs += sound_oss.o info_oss.o -endif +snd-y := sound.o init.o memory.o info.o control.o misc.o device.o +snd-$(CONFIG_ISA_DMA_API) += isadma.o +snd-$(CONFIG_SND_OSSEMUL) += sound_oss.o info_oss.o snd-pcm-objs := pcm.o pcm_native.o pcm_lib.o pcm_timer.o pcm_misc.o \ pcm_memory.o -snd-page-alloc-objs := memalloc.o sgbuf.o +snd-page-alloc-y := memalloc.o +snd-page-alloc-$(CONFIG_HAS_DMA) += sgbuf.o snd-rawmidi-objs := rawmidi.o snd-timer-objs := timer.o diff -ruN linux-2.6.22/sound/core/control.c linux-2.6.22-alsa/sound/core/control.c --- linux-2.6.22/sound/core/control.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/core/control.c 2007-09-01 20:55:22.000000000 +0200 @@ -1486,3 +1486,30 @@ snd_assert(card != NULL, return -ENXIO); return snd_device_new(card, SNDRV_DEV_CONTROL, card, &ops); } + +/* + * Frequently used control callbacks + */ +int snd_ctl_boolean_mono_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +EXPORT_SYMBOL(snd_ctl_boolean_mono_info); + +int snd_ctl_boolean_stereo_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +EXPORT_SYMBOL(snd_ctl_boolean_stereo_info); diff -ruN linux-2.6.22/sound/core/memalloc.c linux-2.6.22-alsa/sound/core/memalloc.c --- linux-2.6.22/sound/core/memalloc.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/core/memalloc.c 2007-09-01 20:55:22.000000000 +0200 @@ -205,6 +205,7 @@ * */ +#ifdef CONFIG_HAS_DMA /* allocate the coherent DMA pages */ static void *snd_malloc_dev_pages(struct device *dev, size_t size, dma_addr_t *dma) { @@ -238,6 +239,7 @@ dec_snd_pages(pg); dma_free_coherent(dev, PAGE_SIZE << pg, ptr, dma); } +#endif /* CONFIG_HAS_DMA */ #ifdef CONFIG_SBUS @@ -311,12 +313,14 @@ dmab->area = snd_malloc_sbus_pages(device, size, &dmab->addr); break; #endif +#ifdef CONFIG_HAS_DMA case SNDRV_DMA_TYPE_DEV: dmab->area = snd_malloc_dev_pages(device, size, &dmab->addr); break; case SNDRV_DMA_TYPE_DEV_SG: snd_malloc_sgbuf_pages(device, size, dmab, NULL); break; +#endif default: printk(KERN_ERR "snd-malloc: invalid device type %d\n", type); dmab->area = NULL; @@ -382,12 +386,14 @@ snd_free_sbus_pages(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr); break; #endif +#ifdef CONFIG_HAS_DMA case SNDRV_DMA_TYPE_DEV: snd_free_dev_pages(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr); break; case SNDRV_DMA_TYPE_DEV_SG: snd_free_sgbuf_pages(dmab); break; +#endif default: printk(KERN_ERR "snd-malloc: invalid device type %d\n", dmab->dev.type); } diff -ruN linux-2.6.22/sound/core/oss/Makefile linux-2.6.22-alsa/sound/core/oss/Makefile --- linux-2.6.22/sound/core/oss/Makefile 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/core/oss/Makefile 2007-09-01 20:55:22.000000000 +0200 @@ -5,8 +5,9 @@ snd-mixer-oss-objs := mixer_oss.o -snd-pcm-oss-objs := pcm_oss.o pcm_plugin.o \ - io.o copy.o linear.o mulaw.o route.o rate.o +snd-pcm-oss-y := pcm_oss.o +snd-pcm-oss-$(CONFIG_SND_PCM_OSS_PLUGINS) += pcm_plugin.o \ + io.o copy.o linear.o mulaw.o route.o rate.o obj-$(CONFIG_SND_MIXER_OSS) += snd-mixer-oss.o obj-$(CONFIG_SND_PCM_OSS) += snd-pcm-oss.o diff -ruN linux-2.6.22/sound/core/oss/copy.c linux-2.6.22-alsa/sound/core/oss/copy.c --- linux-2.6.22/sound/core/oss/copy.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/core/oss/copy.c 2007-09-01 20:55:22.000000000 +0200 @@ -20,9 +20,6 @@ */ #include - -#ifdef CONFIG_SND_PCM_OSS_PLUGINS - #include #include #include @@ -88,5 +85,3 @@ *r_plugin = plugin; return 0; } - -#endif diff -ruN linux-2.6.22/sound/core/oss/io.c linux-2.6.22-alsa/sound/core/oss/io.c --- linux-2.6.22/sound/core/oss/io.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/core/oss/io.c 2007-09-01 20:55:22.000000000 +0200 @@ -20,9 +20,6 @@ */ #include - -#ifdef CONFIG_SND_PCM_OSS_PLUGINS - #include #include #include @@ -135,5 +132,3 @@ *r_plugin = plugin; return 0; } - -#endif diff -ruN linux-2.6.22/sound/core/oss/linear.c linux-2.6.22-alsa/sound/core/oss/linear.c --- linux-2.6.22/sound/core/oss/linear.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/core/oss/linear.c 2007-09-01 20:55:22.000000000 +0200 @@ -21,9 +21,6 @@ */ #include - -#ifdef CONFIG_SND_PCM_OSS_PLUGINS - #include #include #include @@ -34,19 +31,34 @@ */ struct linear_priv { - int conv; + int cvt_endian; /* need endian conversion? */ + unsigned int src_ofs; /* byte offset in source format */ + unsigned int dst_ofs; /* byte soffset in destination format */ + unsigned int copy_ofs; /* byte offset in temporary u32 data */ + unsigned int dst_bytes; /* byte size of destination format */ + unsigned int copy_bytes; /* bytes to copy per conversion */ + unsigned int flip; /* MSB flip for signeness, done after endian conv */ }; +static inline void do_convert(struct linear_priv *data, + unsigned char *dst, unsigned char *src) +{ + unsigned int tmp = 0; + unsigned char *p = (unsigned char *)&tmp; + + memcpy(p + data->copy_ofs, src + data->src_ofs, data->copy_bytes); + if (data->cvt_endian) + tmp = swab32(tmp); + tmp ^= data->flip; + memcpy(dst, p + data->dst_ofs, data->dst_bytes); +} + static void convert(struct snd_pcm_plugin *plugin, const struct snd_pcm_plugin_channel *src_channels, struct snd_pcm_plugin_channel *dst_channels, snd_pcm_uframes_t frames) { -#define CONV_LABELS -#include "plugin_ops.h" -#undef CONV_LABELS struct linear_priv *data = (struct linear_priv *)plugin->extra_data; - void *conv = conv_labels[data->conv]; int channel; int nchannels = plugin->src_format.channels; for (channel = 0; channel < nchannels; ++channel) { @@ -67,11 +79,7 @@ dst_step = dst_channels[channel].area.step / 8; frames1 = frames; while (frames1-- > 0) { - goto *conv; -#define CONV_END after -#include "plugin_ops.h" -#undef CONV_END - after: + do_convert(data, dst, src); src += src_step; dst += dst_step; } @@ -106,29 +114,36 @@ return frames; } -static int conv_index(int src_format, int dst_format) +static void init_data(struct linear_priv *data, int src_format, int dst_format) { - int src_endian, dst_endian, sign, src_width, dst_width; + int src_le, dst_le, src_bytes, dst_bytes; - sign = (snd_pcm_format_signed(src_format) != - snd_pcm_format_signed(dst_format)); -#ifdef SNDRV_LITTLE_ENDIAN - src_endian = snd_pcm_format_big_endian(src_format); - dst_endian = snd_pcm_format_big_endian(dst_format); -#else - src_endian = snd_pcm_format_little_endian(src_format); - dst_endian = snd_pcm_format_little_endian(dst_format); -#endif - - if (src_endian < 0) - src_endian = 0; - if (dst_endian < 0) - dst_endian = 0; - - src_width = snd_pcm_format_width(src_format) / 8 - 1; - dst_width = snd_pcm_format_width(dst_format) / 8 - 1; - - return src_width * 32 + src_endian * 16 + sign * 8 + dst_width * 2 + dst_endian; + src_bytes = snd_pcm_format_width(src_format) / 8; + dst_bytes = snd_pcm_format_width(dst_format) / 8; + src_le = snd_pcm_format_little_endian(src_format) > 0; + dst_le = snd_pcm_format_little_endian(dst_format) > 0; + + data->dst_bytes = dst_bytes; + data->cvt_endian = src_le != dst_le; + data->copy_bytes = src_bytes < dst_bytes ? src_bytes : dst_bytes; + if (src_le) { + data->copy_ofs = 4 - data->copy_bytes; + data->src_ofs = src_bytes - data->copy_bytes; + } else + data->src_ofs = snd_pcm_format_physical_width(src_format) / 8 - + src_bytes; + if (dst_le) + data->dst_ofs = 4 - data->dst_bytes; + else + data->dst_ofs = snd_pcm_format_physical_width(dst_format) / 8 - + dst_bytes; + if (snd_pcm_format_signed(src_format) != + snd_pcm_format_signed(dst_format)) { + if (dst_le) + data->flip = cpu_to_le32(0x80000000); + else + data->flip = cpu_to_be32(0x80000000); + } } int snd_pcm_plugin_build_linear(struct snd_pcm_substream *plug, @@ -154,10 +169,8 @@ if (err < 0) return err; data = (struct linear_priv *)plugin->extra_data; - data->conv = conv_index(src_format->format, dst_format->format); + init_data(data, src_format->format, dst_format->format); plugin->transfer = linear_transfer; *r_plugin = plugin; return 0; } - -#endif diff -ruN linux-2.6.22/sound/core/oss/mulaw.c linux-2.6.22-alsa/sound/core/oss/mulaw.c --- linux-2.6.22/sound/core/oss/mulaw.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/core/oss/mulaw.c 2007-09-01 20:55:22.000000000 +0200 @@ -22,9 +22,6 @@ */ #include - -#ifdef CONFIG_SND_PCM_OSS_PLUGINS - #include #include #include @@ -149,19 +146,32 @@ struct mulaw_priv { mulaw_f func; - int conv; + int cvt_endian; /* need endian conversion? */ + unsigned int native_ofs; /* byte offset in native format */ + unsigned int copy_ofs; /* byte offset in s16 format */ + unsigned int native_bytes; /* byte size of the native format */ + unsigned int copy_bytes; /* bytes to copy per conversion */ + u16 flip; /* MSB flip for signedness, done after endian conversion */ }; +static inline void cvt_s16_to_native(struct mulaw_priv *data, + unsigned char *dst, u16 sample) +{ + sample ^= data->flip; + if (data->cvt_endian) + sample = swab16(sample); + if (data->native_bytes > data->copy_bytes) + memset(dst, 0, data->native_bytes); + memcpy(dst + data->native_ofs, (char *)&sample + data->copy_ofs, + data->copy_bytes); +} + static void mulaw_decode(struct snd_pcm_plugin *plugin, const struct snd_pcm_plugin_channel *src_channels, struct snd_pcm_plugin_channel *dst_channels, snd_pcm_uframes_t frames) { -#define PUT_S16_LABELS -#include "plugin_ops.h" -#undef PUT_S16_LABELS struct mulaw_priv *data = (struct mulaw_priv *)plugin->extra_data; - void *put = put_s16_labels[data->conv]; int channel; int nchannels = plugin->src_format.channels; for (channel = 0; channel < nchannels; ++channel) { @@ -183,30 +193,33 @@ frames1 = frames; while (frames1-- > 0) { signed short sample = ulaw2linear(*src); - goto *put; -#define PUT_S16_END after -#include "plugin_ops.h" -#undef PUT_S16_END - after: + cvt_s16_to_native(data, dst, sample); src += src_step; dst += dst_step; } } } +static inline signed short cvt_native_to_s16(struct mulaw_priv *data, + unsigned char *src) +{ + u16 sample = 0; + memcpy((char *)&sample + data->copy_ofs, src + data->native_ofs, + data->copy_bytes); + if (data->cvt_endian) + sample = swab16(sample); + sample ^= data->flip; + return (signed short)sample; +} + static void mulaw_encode(struct snd_pcm_plugin *plugin, const struct snd_pcm_plugin_channel *src_channels, struct snd_pcm_plugin_channel *dst_channels, snd_pcm_uframes_t frames) { -#define GET_S16_LABELS -#include "plugin_ops.h" -#undef GET_S16_LABELS struct mulaw_priv *data = (struct mulaw_priv *)plugin->extra_data; - void *get = get_s16_labels[data->conv]; int channel; int nchannels = plugin->src_format.channels; - signed short sample = 0; for (channel = 0; channel < nchannels; ++channel) { char *src; char *dst; @@ -225,11 +238,7 @@ dst_step = dst_channels[channel].area.step / 8; frames1 = frames; while (frames1-- > 0) { - goto *get; -#define GET_S16_END after -#include "plugin_ops.h" -#undef GET_S16_END - after: + signed short sample = cvt_native_to_s16(data, src); *dst = linear2ulaw(sample); src += src_step; dst += dst_step; @@ -265,23 +274,25 @@ return frames; } -static int getput_index(int format) +static void init_data(struct mulaw_priv *data, int format) { - int sign, width, endian; - sign = !snd_pcm_format_signed(format); - width = snd_pcm_format_width(format) / 8 - 1; - if (width < 0 || width > 3) { - snd_printk(KERN_ERR "snd-pcm-oss: invalid format %d\n", format); - width = 0; - } #ifdef SNDRV_LITTLE_ENDIAN - endian = snd_pcm_format_big_endian(format); + data->cvt_endian = snd_pcm_format_big_endian(format) > 0; #else - endian = snd_pcm_format_little_endian(format); + data->cvt_endian = snd_pcm_format_little_endian(format) > 0; #endif - if (endian < 0) - endian = 0; - return width * 4 + endian * 2 + sign; + if (!snd_pcm_format_signed(format)) + data->flip = 0x8000; + data->native_bytes = snd_pcm_format_physical_width(format) / 8; + data->copy_bytes = data->native_bytes < 2 ? 1 : 2; + if (snd_pcm_format_little_endian(format)) { + data->native_ofs = data->native_bytes - data->copy_bytes; + data->copy_ofs = 2 - data->copy_bytes; + } else { + /* S24 in 4bytes need an 1 byte offset */ + data->native_ofs = data->native_bytes - + snd_pcm_format_width(format) / 8; + } } int snd_pcm_plugin_build_mulaw(struct snd_pcm_substream *plug, @@ -322,11 +333,8 @@ return err; data = (struct mulaw_priv *)plugin->extra_data; data->func = func; - data->conv = getput_index(format->format); - snd_assert(data->conv >= 0 && data->conv < 4*2*2, return -EINVAL); + init_data(data, format->format); plugin->transfer = mulaw_transfer; *r_plugin = plugin; return 0; } - -#endif diff -ruN linux-2.6.22/sound/core/oss/pcm_oss.c linux-2.6.22-alsa/sound/core/oss/pcm_oss.c --- linux-2.6.22/sound/core/oss/pcm_oss.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/core/oss/pcm_oss.c 2007-09-01 20:55:22.000000000 +0200 @@ -633,6 +633,22 @@ return bytes_to_frames(runtime, (buffer_size * bytes) / runtime->oss.buffer_bytes); } +/* define extended formats in the recent OSS versions (if any) */ +/* linear formats */ +#define AFMT_S32_LE 0x00001000 +#define AFMT_S32_BE 0x00002000 +#define AFMT_S24_LE 0x00008000 +#define AFMT_S24_BE 0x00010000 +#define AFMT_S24_PACKED 0x00040000 + +/* other supported formats */ +#define AFMT_FLOAT 0x00004000 +#define AFMT_SPDIF_RAW 0x00020000 + +/* unsupported formats */ +#define AFMT_AC3 0x00000400 +#define AFMT_VORBIS 0x00000800 + static int snd_pcm_oss_format_from(int format) { switch (format) { @@ -646,6 +662,13 @@ case AFMT_U16_LE: return SNDRV_PCM_FORMAT_U16_LE; case AFMT_U16_BE: return SNDRV_PCM_FORMAT_U16_BE; case AFMT_MPEG: return SNDRV_PCM_FORMAT_MPEG; + case AFMT_S32_LE: return SNDRV_PCM_FORMAT_S32_LE; + case AFMT_S32_BE: return SNDRV_PCM_FORMAT_S32_BE; + case AFMT_S24_LE: return SNDRV_PCM_FORMAT_S24_LE; + case AFMT_S24_BE: return SNDRV_PCM_FORMAT_S24_BE; + case AFMT_S24_PACKED: return SNDRV_PCM_FORMAT_S24_3LE; + case AFMT_FLOAT: return SNDRV_PCM_FORMAT_FLOAT; + case AFMT_SPDIF_RAW: return SNDRV_PCM_FORMAT_IEC958_SUBFRAME; default: return SNDRV_PCM_FORMAT_U8; } } @@ -663,6 +686,13 @@ case SNDRV_PCM_FORMAT_U16_LE: return AFMT_U16_LE; case SNDRV_PCM_FORMAT_U16_BE: return AFMT_U16_BE; case SNDRV_PCM_FORMAT_MPEG: return AFMT_MPEG; + case SNDRV_PCM_FORMAT_S32_LE: return AFMT_S32_LE; + case SNDRV_PCM_FORMAT_S32_BE: return AFMT_S32_BE; + case SNDRV_PCM_FORMAT_S24_LE: return AFMT_S24_LE; + case SNDRV_PCM_FORMAT_S24_BE: return AFMT_S24_BE; + case SNDRV_PCM_FORMAT_S24_3LE: return AFMT_S24_PACKED; + case SNDRV_PCM_FORMAT_FLOAT: return AFMT_FLOAT; + case SNDRV_PCM_FORMAT_IEC958_SUBFRAME: return AFMT_SPDIF_RAW; default: return -EINVAL; } } @@ -1725,7 +1755,10 @@ return AFMT_MU_LAW | AFMT_U8 | AFMT_S16_LE | AFMT_S16_BE | AFMT_S8 | AFMT_U16_LE | - AFMT_U16_BE; + AFMT_U16_BE | + AFMT_S32_LE | AFMT_S32_BE | + AFMT_S24_LE | AFMT_S24_LE | + AFMT_S24_PACKED; params = kmalloc(sizeof(*params), GFP_KERNEL); if (!params) return -ENOMEM; diff -ruN linux-2.6.22/sound/core/oss/pcm_plugin.c linux-2.6.22-alsa/sound/core/oss/pcm_plugin.c --- linux-2.6.22/sound/core/oss/pcm_plugin.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/core/oss/pcm_plugin.c 2007-09-01 20:55:22.000000000 +0200 @@ -25,9 +25,6 @@ #endif #include - -#ifdef CONFIG_SND_PCM_OSS_PLUGINS - #include #include #include @@ -267,6 +264,8 @@ SNDRV_PCM_FMTBIT_U16_BE | SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_U24_LE | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_U24_BE | SNDRV_PCM_FMTBIT_S24_BE | + SNDRV_PCM_FMTBIT_U24_3LE | SNDRV_PCM_FMTBIT_S24_3LE | + SNDRV_PCM_FMTBIT_U24_3BE | SNDRV_PCM_FMTBIT_S24_3BE | SNDRV_PCM_FMTBIT_U32_LE | SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_BE | SNDRV_PCM_FMTBIT_S32_BE); snd_mask_set(&formats, SNDRV_PCM_FORMAT_MU_LAW); @@ -283,6 +282,10 @@ SNDRV_PCM_FORMAT_S16_BE, SNDRV_PCM_FORMAT_U16_LE, SNDRV_PCM_FORMAT_U16_BE, + SNDRV_PCM_FORMAT_S24_3LE, + SNDRV_PCM_FORMAT_S24_3BE, + SNDRV_PCM_FORMAT_U24_3LE, + SNDRV_PCM_FORMAT_U24_3BE, SNDRV_PCM_FORMAT_S24_LE, SNDRV_PCM_FORMAT_S24_BE, SNDRV_PCM_FORMAT_U24_LE, @@ -297,41 +300,37 @@ int snd_pcm_plug_slave_format(int format, struct snd_mask *format_mask) { + int i; + if (snd_mask_test(format_mask, format)) return format; if (! snd_pcm_plug_formats(format_mask, format)) return -EINVAL; if (snd_pcm_format_linear(format)) { - int width = snd_pcm_format_width(format); - int unsignd = snd_pcm_format_unsigned(format); - int big = snd_pcm_format_big_endian(format); - int format1; - int wid, width1=width; - int dwidth1 = 8; - for (wid = 0; wid < 4; ++wid) { - int end, big1 = big; - for (end = 0; end < 2; ++end) { - int sgn, unsignd1 = unsignd; - for (sgn = 0; sgn < 2; ++sgn) { - format1 = snd_pcm_build_linear_format(width1, unsignd1, big1); - if (format1 >= 0 && - snd_mask_test(format_mask, format1)) - goto _found; - unsignd1 = !unsignd1; - } - big1 = !big1; - } - if (width1 == 32) { - dwidth1 = -dwidth1; - width1 = width; + unsigned int width = snd_pcm_format_width(format); + int unsignd = snd_pcm_format_unsigned(format) > 0; + int big = snd_pcm_format_big_endian(format) > 0; + unsigned int badness, best = -1; + int best_format = -1; + for (i = 0; i < ARRAY_SIZE(preferred_formats); i++) { + int f = preferred_formats[i]; + unsigned int w; + if (!snd_mask_test(format_mask, f)) + continue; + w = snd_pcm_format_width(f); + if (w >= width) + badness = w - width; + else + badness = width - w + 32; + badness += snd_pcm_format_unsigned(f) != unsignd; + badness += snd_pcm_format_big_endian(f) != big; + if (badness < best) { + best_format = f; + best = badness; } - width1 += dwidth1; } - return -EINVAL; - _found: - return format1; + return best_format >= 0 ? best_format : -EINVAL; } else { - unsigned int i; switch (format) { case SNDRV_PCM_FORMAT_MU_LAW: for (i = 0; i < ARRAY_SIZE(preferred_formats); ++i) { @@ -740,5 +739,3 @@ } return 0; } - -#endif diff -ruN linux-2.6.22/sound/core/oss/rate.c linux-2.6.22-alsa/sound/core/oss/rate.c --- linux-2.6.22/sound/core/oss/rate.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/core/oss/rate.c 2007-09-01 20:55:22.000000000 +0200 @@ -20,9 +20,6 @@ */ #include - -#ifdef CONFIG_SND_PCM_OSS_PLUGINS - #include #include #include @@ -340,5 +337,3 @@ *r_plugin = plugin; return 0; } - -#endif diff -ruN linux-2.6.22/sound/core/oss/route.c linux-2.6.22-alsa/sound/core/oss/route.c --- linux-2.6.22/sound/core/oss/route.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/core/oss/route.c 2007-09-01 20:55:22.000000000 +0200 @@ -20,9 +20,6 @@ */ #include - -#ifdef CONFIG_SND_PCM_OSS_PLUGINS - #include #include #include @@ -108,5 +105,3 @@ *r_plugin = plugin; return 0; } - -#endif diff -ruN linux-2.6.22/sound/core/pcm_misc.c linux-2.6.22-alsa/sound/core/pcm_misc.c --- linux-2.6.22/sound/core/pcm_misc.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/core/pcm_misc.c 2007-09-01 20:55:22.000000000 +0200 @@ -422,38 +422,6 @@ EXPORT_SYMBOL(snd_pcm_format_set_silence); -/* [width][unsigned][bigendian] */ -static int linear_formats[4][2][2] = { - {{ SNDRV_PCM_FORMAT_S8, SNDRV_PCM_FORMAT_S8}, - { SNDRV_PCM_FORMAT_U8, SNDRV_PCM_FORMAT_U8}}, - {{SNDRV_PCM_FORMAT_S16_LE, SNDRV_PCM_FORMAT_S16_BE}, - {SNDRV_PCM_FORMAT_U16_LE, SNDRV_PCM_FORMAT_U16_BE}}, - {{SNDRV_PCM_FORMAT_S24_LE, SNDRV_PCM_FORMAT_S24_BE}, - {SNDRV_PCM_FORMAT_U24_LE, SNDRV_PCM_FORMAT_U24_BE}}, - {{SNDRV_PCM_FORMAT_S32_LE, SNDRV_PCM_FORMAT_S32_BE}, - {SNDRV_PCM_FORMAT_U32_LE, SNDRV_PCM_FORMAT_U32_BE}} -}; - -/** - * snd_pcm_build_linear_format - return the suitable linear format for the given condition - * @width: the bit-width - * @unsignd: 1 if unsigned, 0 if signed. - * @big_endian: 1 if big-endian, 0 if little-endian - * - * Returns the suitable linear format for the given condition. - */ -snd_pcm_format_t snd_pcm_build_linear_format(int width, int unsignd, int big_endian) -{ - if (width & 7) - return SND_PCM_FORMAT_UNKNOWN; - width = (width / 8) - 1; - if (width < 0 || width >= 4) - return SND_PCM_FORMAT_UNKNOWN; - return linear_formats[width][!!unsignd][!!big_endian]; -} - -EXPORT_SYMBOL(snd_pcm_build_linear_format); - /** * snd_pcm_limit_hw_rates - determine rate_min/rate_max fields * @runtime: the runtime instance @@ -465,21 +433,16 @@ */ int snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime) { - static unsigned rates[] = { - /* ATTENTION: these values depend on the definition in pcm.h! */ - 5512, 8000, 11025, 16000, 22050, 32000, 44100, 48000, - 64000, 88200, 96000, 176400, 192000 - }; int i; - for (i = 0; i < (int)ARRAY_SIZE(rates); i++) { + for (i = 0; i < (int)snd_pcm_known_rates.count; i++) { if (runtime->hw.rates & (1 << i)) { - runtime->hw.rate_min = rates[i]; + runtime->hw.rate_min = snd_pcm_known_rates.list[i]; break; } } - for (i = (int)ARRAY_SIZE(rates) - 1; i >= 0; i--) { + for (i = (int)snd_pcm_known_rates.count - 1; i >= 0; i--) { if (runtime->hw.rates & (1 << i)) { - runtime->hw.rate_max = rates[i]; + runtime->hw.rate_max = snd_pcm_known_rates.list[i]; break; } } @@ -487,3 +450,21 @@ } EXPORT_SYMBOL(snd_pcm_limit_hw_rates); + +/** + * snd_pcm_rate_to_rate_bit - converts sample rate to SNDRV_PCM_RATE_xxx bit + * @rate: the sample rate to convert + * + * Returns the SNDRV_PCM_RATE_xxx flag that corresponds to the given rate, or + * SNDRV_PCM_RATE_KNOT for an unknown rate. + */ +unsigned int snd_pcm_rate_to_rate_bit(unsigned int rate) +{ + unsigned int i; + + for (i = 0; i < snd_pcm_known_rates.count; i++) + if (snd_pcm_known_rates.list[i] == rate) + return 1u << i; + return SNDRV_PCM_RATE_KNOT; +} +EXPORT_SYMBOL(snd_pcm_rate_to_rate_bit); diff -ruN linux-2.6.22/sound/core/pcm_native.c linux-2.6.22-alsa/sound/core/pcm_native.c --- linux-2.6.22/sound/core/pcm_native.c 2007-07-28 13:40:44.000000000 +0200 +++ linux-2.6.22-alsa/sound/core/pcm_native.c 2007-09-01 20:55:22.000000000 +0200 @@ -1487,7 +1487,7 @@ snd_pcm_stream_lock_irq(substream); /* resume pause */ - if (runtime->status->state == SNDRV_PCM_STATE_PAUSED) + if (substream->runtime->status->state == SNDRV_PCM_STATE_PAUSED) snd_pcm_pause(substream, 0); /* pre-start/stop - all running streams are changed to DRAINING state */ @@ -1787,12 +1787,18 @@ static unsigned int rates[] = { 5512, 8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000, 88200, 96000, 176400, 192000 }; +const struct snd_pcm_hw_constraint_list snd_pcm_known_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, +}; + static int snd_pcm_hw_rule_rate(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { struct snd_pcm_hardware *hw = rule->private; return snd_interval_list(hw_param_interval(params, rule->var), - ARRAY_SIZE(rates), rates, hw->rates); + snd_pcm_known_rates.count, + snd_pcm_known_rates.list, hw->rates); } static int snd_pcm_hw_rule_buffer_bytes_max(struct snd_pcm_hw_params *params, diff -ruN linux-2.6.22/sound/core/rawmidi.c linux-2.6.22-alsa/sound/core/rawmidi.c --- linux-2.6.22/sound/core/rawmidi.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/core/rawmidi.c 2007-09-01 20:55:22.000000000 +0200 @@ -30,7 +30,6 @@ #include #include #include -#include #include #include #include diff -ruN linux-2.6.22/sound/core/seq/oss/seq_oss_init.c linux-2.6.22-alsa/sound/core/seq/oss/seq_oss_init.c --- linux-2.6.22/sound/core/seq/oss/seq_oss_init.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/core/seq/oss/seq_oss_init.c 2007-09-01 20:55:22.000000000 +0200 @@ -176,29 +176,29 @@ int i, rc; struct seq_oss_devinfo *dp; - if ((dp = kzalloc(sizeof(*dp), GFP_KERNEL)) == NULL) { + dp = kzalloc(sizeof(*dp), GFP_KERNEL); + if (!dp) { snd_printk(KERN_ERR "can't malloc device info\n"); return -ENOMEM; } debug_printk(("oss_open: dp = %p\n", dp)); + dp->cseq = system_client; + dp->port = -1; + dp->queue = -1; + for (i = 0; i < SNDRV_SEQ_OSS_MAX_CLIENTS; i++) { if (client_table[i] == NULL) break; } + + dp->index = i; if (i >= SNDRV_SEQ_OSS_MAX_CLIENTS) { snd_printk(KERN_ERR "too many applications\n"); - kfree(dp); - return -ENOMEM; + rc = -ENOMEM; + goto _error; } - dp->index = i; - dp->cseq = system_client; - dp->port = -1; - dp->queue = -1; - dp->readq = NULL; - dp->writeq = NULL; - /* look up synth and midi devices */ snd_seq_oss_synth_setup(dp); snd_seq_oss_midi_setup(dp); @@ -211,14 +211,16 @@ /* create port */ debug_printk(("create new port\n")); - if ((rc = create_port(dp)) < 0) { + rc = create_port(dp); + if (rc < 0) { snd_printk(KERN_ERR "can't create port\n"); goto _error; } /* allocate queue */ debug_printk(("allocate queue\n")); - if ((rc = alloc_seq_queue(dp)) < 0) + rc = alloc_seq_queue(dp); + if (rc < 0) goto _error; /* set address */ @@ -235,7 +237,8 @@ /* initialize read queue */ debug_printk(("initialize read queue\n")); if (is_read_mode(dp->file_mode)) { - if ((dp->readq = snd_seq_oss_readq_new(dp, maxqlen)) == NULL) { + dp->readq = snd_seq_oss_readq_new(dp, maxqlen); + if (!dp->readq) { rc = -ENOMEM; goto _error; } @@ -245,7 +248,7 @@ debug_printk(("initialize write queue\n")); if (is_write_mode(dp->file_mode)) { dp->writeq = snd_seq_oss_writeq_new(dp, maxqlen); - if (dp->writeq == NULL) { + if (!dp->writeq) { rc = -ENOMEM; goto _error; } @@ -253,7 +256,8 @@ /* initialize timer */ debug_printk(("initialize timer\n")); - if ((dp->timer = snd_seq_oss_timer_new(dp)) == NULL) { + dp->timer = snd_seq_oss_timer_new(dp); + if (!dp->timer) { snd_printk(KERN_ERR "can't alloc timer\n"); rc = -ENOMEM; goto _error; @@ -276,11 +280,13 @@ return 0; _error: + snd_seq_oss_writeq_delete(dp->writeq); + snd_seq_oss_readq_delete(dp->readq); snd_seq_oss_synth_cleanup(dp); snd_seq_oss_midi_cleanup(dp); - i = dp->queue; delete_port(dp); - delete_seq_queue(i); + delete_seq_queue(dp->queue); + kfree(dp); return rc; } diff -ruN linux-2.6.22/sound/core/seq/oss/seq_oss_writeq.c linux-2.6.22-alsa/sound/core/seq/oss/seq_oss_writeq.c --- linux-2.6.22/sound/core/seq/oss/seq_oss_writeq.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/core/seq/oss/seq_oss_writeq.c 2007-09-01 20:55:22.000000000 +0200 @@ -63,8 +63,10 @@ void snd_seq_oss_writeq_delete(struct seq_oss_writeq *q) { - snd_seq_oss_writeq_clear(q); /* to be sure */ - kfree(q); + if (q) { + snd_seq_oss_writeq_clear(q); /* to be sure */ + kfree(q); + } } diff -ruN linux-2.6.22/sound/core/seq/seq_instr.c linux-2.6.22-alsa/sound/core/seq/seq_instr.c --- linux-2.6.22/sound/core/seq/seq_instr.c 2007-07-28 13:40:44.000000000 +0200 +++ linux-2.6.22-alsa/sound/core/seq/seq_instr.c 2007-09-01 20:55:22.000000000 +0200 @@ -109,7 +109,7 @@ spin_lock_irqsave(&list->lock, flags); while (instr->use) { spin_unlock_irqrestore(&list->lock, flags); - schedule_timeout_interruptible(1); + schedule_timeout(1); spin_lock_irqsave(&list->lock, flags); } spin_unlock_irqrestore(&list->lock, flags); @@ -199,7 +199,7 @@ instr = flist; flist = instr->next; while (instr->use) - schedule_timeout_interruptible(1); + schedule_timeout(1); if (snd_seq_instr_free(instr, atomic)<0) snd_printk(KERN_WARNING "instrument free problem\n"); instr = next; @@ -555,7 +555,7 @@ SNDRV_SEQ_INSTR_NOTIFY_REMOVE); while (instr->use) { spin_unlock_irqrestore(&list->lock, flags); - schedule_timeout_interruptible(1); + schedule_timeout(1); spin_lock_irqsave(&list->lock, flags); } spin_unlock_irqrestore(&list->lock, flags); diff -ruN linux-2.6.22/sound/core/seq/seq_midi_event.c linux-2.6.22-alsa/sound/core/seq/seq_midi_event.c --- linux-2.6.22/sound/core/seq/seq_midi_event.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/core/seq/seq_midi_event.c 2007-09-01 20:55:22.000000000 +0200 @@ -32,10 +32,9 @@ MODULE_DESCRIPTION("MIDI byte <-> sequencer event coder"); MODULE_LICENSE("GPL"); -/* queue type */ -/* from 0 to 7 are normal commands (note off, on, etc.) */ -#define ST_NOTEOFF 0 -#define ST_NOTEON 1 +/* event type, index into status_event[] */ +/* from 0 to 6 are normal commands (note off, on, etc.) for 0x9?-0xe? */ +#define ST_INVALID 7 #define ST_SPECIAL 8 #define ST_SYSEX ST_SPECIAL /* from 8 to 15 are events for 0xf0-0xf7 */ @@ -65,32 +64,33 @@ void (*encode)(struct snd_midi_event *dev, struct snd_seq_event *ev); void (*decode)(struct snd_seq_event *ev, unsigned char *buf); } status_event[] = { - /* 0x80 - 0xf0 */ - {SNDRV_SEQ_EVENT_NOTEOFF, 2, note_event, note_decode}, - {SNDRV_SEQ_EVENT_NOTEON, 2, note_event, note_decode}, - {SNDRV_SEQ_EVENT_KEYPRESS, 2, note_event, note_decode}, - {SNDRV_SEQ_EVENT_CONTROLLER, 2, two_param_ctrl_event, two_param_decode}, - {SNDRV_SEQ_EVENT_PGMCHANGE, 1, one_param_ctrl_event, one_param_decode}, - {SNDRV_SEQ_EVENT_CHANPRESS, 1, one_param_ctrl_event, one_param_decode}, - {SNDRV_SEQ_EVENT_PITCHBEND, 2, pitchbend_ctrl_event, pitchbend_decode}, - {SNDRV_SEQ_EVENT_NONE, 0, NULL, NULL}, /* 0xf0 */ + /* 0x80 - 0xef */ + {SNDRV_SEQ_EVENT_NOTEOFF, 2, note_event, note_decode}, + {SNDRV_SEQ_EVENT_NOTEON, 2, note_event, note_decode}, + {SNDRV_SEQ_EVENT_KEYPRESS, 2, note_event, note_decode}, + {SNDRV_SEQ_EVENT_CONTROLLER, 2, two_param_ctrl_event, two_param_decode}, + {SNDRV_SEQ_EVENT_PGMCHANGE, 1, one_param_ctrl_event, one_param_decode}, + {SNDRV_SEQ_EVENT_CHANPRESS, 1, one_param_ctrl_event, one_param_decode}, + {SNDRV_SEQ_EVENT_PITCHBEND, 2, pitchbend_ctrl_event, pitchbend_decode}, + /* invalid */ + {SNDRV_SEQ_EVENT_NONE, -1, NULL, NULL}, /* 0xf0 - 0xff */ - {SNDRV_SEQ_EVENT_SYSEX, 1, NULL, NULL}, /* sysex: 0xf0 */ - {SNDRV_SEQ_EVENT_QFRAME, 1, one_param_event, one_param_decode}, /* 0xf1 */ - {SNDRV_SEQ_EVENT_SONGPOS, 2, songpos_event, songpos_decode}, /* 0xf2 */ - {SNDRV_SEQ_EVENT_SONGSEL, 1, one_param_event, one_param_decode}, /* 0xf3 */ - {SNDRV_SEQ_EVENT_NONE, 0, NULL, NULL}, /* 0xf4 */ - {SNDRV_SEQ_EVENT_NONE, 0, NULL, NULL}, /* 0xf5 */ - {SNDRV_SEQ_EVENT_TUNE_REQUEST, 0, NULL, NULL}, /* 0xf6 */ - {SNDRV_SEQ_EVENT_NONE, 0, NULL, NULL}, /* 0xf7 */ - {SNDRV_SEQ_EVENT_CLOCK, 0, NULL, NULL}, /* 0xf8 */ - {SNDRV_SEQ_EVENT_NONE, 0, NULL, NULL}, /* 0xf9 */ - {SNDRV_SEQ_EVENT_START, 0, NULL, NULL}, /* 0xfa */ - {SNDRV_SEQ_EVENT_CONTINUE, 0, NULL, NULL}, /* 0xfb */ - {SNDRV_SEQ_EVENT_STOP, 0, NULL, NULL}, /* 0xfc */ - {SNDRV_SEQ_EVENT_NONE, 0, NULL, NULL}, /* 0xfd */ - {SNDRV_SEQ_EVENT_SENSING, 0, NULL, NULL}, /* 0xfe */ - {SNDRV_SEQ_EVENT_RESET, 0, NULL, NULL}, /* 0xff */ + {SNDRV_SEQ_EVENT_SYSEX, 1, NULL, NULL}, /* sysex: 0xf0 */ + {SNDRV_SEQ_EVENT_QFRAME, 1, one_param_event, one_param_decode}, /* 0xf1 */ + {SNDRV_SEQ_EVENT_SONGPOS, 2, songpos_event, songpos_decode}, /* 0xf2 */ + {SNDRV_SEQ_EVENT_SONGSEL, 1, one_param_event, one_param_decode}, /* 0xf3 */ + {SNDRV_SEQ_EVENT_NONE, -1, NULL, NULL}, /* 0xf4 */ + {SNDRV_SEQ_EVENT_NONE, -1, NULL, NULL}, /* 0xf5 */ + {SNDRV_SEQ_EVENT_TUNE_REQUEST, 0, NULL, NULL}, /* 0xf6 */ + {SNDRV_SEQ_EVENT_NONE, -1, NULL, NULL}, /* 0xf7 */ + {SNDRV_SEQ_EVENT_CLOCK, 0, NULL, NULL}, /* 0xf8 */ + {SNDRV_SEQ_EVENT_NONE, -1, NULL, NULL}, /* 0xf9 */ + {SNDRV_SEQ_EVENT_START, 0, NULL, NULL}, /* 0xfa */ + {SNDRV_SEQ_EVENT_CONTINUE, 0, NULL, NULL}, /* 0xfb */ + {SNDRV_SEQ_EVENT_STOP, 0, NULL, NULL}, /* 0xfc */ + {SNDRV_SEQ_EVENT_NONE, -1, NULL, NULL}, /* 0xfd */ + {SNDRV_SEQ_EVENT_SENSING, 0, NULL, NULL}, /* 0xfe */ + {SNDRV_SEQ_EVENT_RESET, 0, NULL, NULL}, /* 0xff */ }; static int extra_decode_ctrl14(struct snd_midi_event *dev, unsigned char *buf, int len, @@ -129,6 +129,7 @@ } dev->bufsize = bufsize; dev->lastcmd = 0xff; + dev->type = ST_INVALID; spin_lock_init(&dev->lock); *rdev = dev; return 0; @@ -149,7 +150,7 @@ { dev->read = 0; dev->qlen = 0; - dev->type = 0; + dev->type = ST_INVALID; } void snd_midi_event_reset_encode(struct snd_midi_event *dev) @@ -251,29 +252,31 @@ ev->type = status_event[ST_SPECIAL + c - 0xf0].event; ev->flags &= ~SNDRV_SEQ_EVENT_LENGTH_MASK; ev->flags |= SNDRV_SEQ_EVENT_LENGTH_FIXED; - return 1; + return ev->type != SNDRV_SEQ_EVENT_NONE; } spin_lock_irqsave(&dev->lock, flags); - if (dev->qlen > 0) { - /* rest of command */ - dev->buf[dev->read++] = c; - if (dev->type != ST_SYSEX) - dev->qlen--; - } else { + if ((c & 0x80) && + (c != MIDI_CMD_COMMON_SYSEX_END || dev->type != ST_SYSEX)) { /* new command */ + dev->buf[0] = c; + if ((c & 0xf0) == 0xf0) /* system messages */ + dev->type = (c & 0x0f) + ST_SPECIAL; + else + dev->type = (c >> 4) & 0x07; dev->read = 1; - if (c & 0x80) { - dev->buf[0] = c; - if ((c & 0xf0) == 0xf0) /* special events */ - dev->type = (c & 0x0f) + ST_SPECIAL; - else - dev->type = (c >> 4) & 0x07; - dev->qlen = status_event[dev->type].qlen; - } else { - /* process this byte as argument */ + dev->qlen = status_event[dev->type].qlen; + } else { + if (dev->qlen > 0) { + /* rest of command */ dev->buf[dev->read++] = c; + if (dev->type != ST_SYSEX) + dev->qlen--; + } else { + /* running status */ + dev->buf[1] = c; dev->qlen = status_event[dev->type].qlen - 1; + dev->read = 2; } } if (dev->qlen == 0) { @@ -282,6 +285,8 @@ ev->flags |= SNDRV_SEQ_EVENT_LENGTH_FIXED; if (status_event[dev->type].encode) /* set data values */ status_event[dev->type].encode(dev, ev); + if (dev->type >= ST_SPECIAL) + dev->type = ST_INVALID; rc = 1; } else if (dev->type == ST_SYSEX) { if (c == MIDI_CMD_COMMON_SYSEX_END || diff -ruN linux-2.6.22/sound/core/seq/seq_virmidi.c linux-2.6.22-alsa/sound/core/seq/seq_virmidi.c --- linux-2.6.22/sound/core/seq/seq_virmidi.c 2007-07-28 13:40:44.000000000 +0200 +++ linux-2.6.22-alsa/sound/core/seq/seq_virmidi.c 2007-09-01 20:55:22.000000000 +0200 @@ -363,7 +363,7 @@ if (rdev->client >= 0) return 0; - pinfo = kmalloc(sizeof(*pinfo), GFP_KERNEL); + pinfo = kzalloc(sizeof(*pinfo), GFP_KERNEL); if (!pinfo) { err = -ENOMEM; goto __error; @@ -380,7 +380,6 @@ rdev->client = client; /* create a port */ - memset(pinfo, 0, sizeof(*pinfo)); pinfo->addr.client = client; sprintf(pinfo->name, "VirMIDI %d-%d", rdev->card->number, rdev->device); /* set all capabilities */ diff -ruN linux-2.6.22/sound/core/sound.c linux-2.6.22-alsa/sound/core/sound.c --- linux-2.6.22/sound/core/sound.c 2007-07-28 13:40:44.000000000 +0200 +++ linux-2.6.22-alsa/sound/core/sound.c 2007-09-01 20:55:22.000000000 +0200 @@ -446,8 +446,7 @@ { snd_info_minor_unregister(); snd_info_done(); - if (unregister_chrdev(major, "alsa") != 0) - snd_printk(KERN_ERR "unable to unregister major device number %d\n", major); + unregister_chrdev(major, "alsa"); } module_init(alsa_sound_init) diff -ruN linux-2.6.22/sound/core/timer.c linux-2.6.22-alsa/sound/core/timer.c --- linux-2.6.22/sound/core/timer.c 2007-07-28 13:40:44.000000000 +0200 +++ linux-2.6.22-alsa/sound/core/timer.c 2007-09-01 20:55:22.000000000 +0200 @@ -1549,9 +1549,11 @@ int err = 0; tu = file->private_data; - snd_assert(tu->timeri != NULL, return -ENXIO); + if (!tu->timeri) + return -EBADFD; t = tu->timeri->timer; - snd_assert(t != NULL, return -ENXIO); + if (!t) + return -EBADFD; info = kzalloc(sizeof(*info), GFP_KERNEL); if (! info) @@ -1579,9 +1581,11 @@ int err; tu = file->private_data; - snd_assert(tu->timeri != NULL, return -ENXIO); + if (!tu->timeri) + return -EBADFD; t = tu->timeri->timer; - snd_assert(t != NULL, return -ENXIO); + if (!t) + return -EBADFD; if (copy_from_user(¶ms, _params, sizeof(params))) return -EFAULT; if (!(t->hw.flags & SNDRV_TIMER_HW_SLAVE) && params.ticks < 1) { @@ -1675,7 +1679,8 @@ struct snd_timer_status status; tu = file->private_data; - snd_assert(tu->timeri != NULL, return -ENXIO); + if (!tu->timeri) + return -EBADFD; memset(&status, 0, sizeof(status)); status.tstamp = tu->tstamp; status.resolution = snd_timer_resolution(tu->timeri); @@ -1695,7 +1700,8 @@ struct snd_timer_user *tu; tu = file->private_data; - snd_assert(tu->timeri != NULL, return -ENXIO); + if (!tu->timeri) + return -EBADFD; snd_timer_stop(tu->timeri); tu->timeri->lost = 0; tu->last_resolution = 0; @@ -1708,7 +1714,8 @@ struct snd_timer_user *tu; tu = file->private_data; - snd_assert(tu->timeri != NULL, return -ENXIO); + if (!tu->timeri) + return -EBADFD; return (err = snd_timer_stop(tu->timeri)) < 0 ? err : 0; } @@ -1718,7 +1725,8 @@ struct snd_timer_user *tu; tu = file->private_data; - snd_assert(tu->timeri != NULL, return -ENXIO); + if (!tu->timeri) + return -EBADFD; tu->timeri->lost = 0; return (err = snd_timer_continue(tu->timeri)) < 0 ? err : 0; } @@ -1729,7 +1737,8 @@ struct snd_timer_user *tu; tu = file->private_data; - snd_assert(tu->timeri != NULL, return -ENXIO); + if (!tu->timeri) + return -EBADFD; return (err = snd_timer_pause(tu->timeri)) < 0 ? err : 0; } diff -ruN linux-2.6.22/sound/drivers/dummy.c linux-2.6.22-alsa/sound/drivers/dummy.c --- linux-2.6.22/sound/drivers/dummy.c 2007-07-28 13:40:44.000000000 +0200 +++ linux-2.6.22-alsa/sound/drivers/dummy.c 2007-09-01 20:55:22.000000000 +0200 @@ -510,15 +510,7 @@ .get = snd_dummy_capsrc_get, .put = snd_dummy_capsrc_put, \ .private_value = addr } -static int snd_dummy_capsrc_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 2; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_dummy_capsrc_info snd_ctl_boolean_stereo_info static int snd_dummy_capsrc_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -659,7 +651,7 @@ }, }; -static void __init_or_module snd_dummy_unregister_all(void) +static void snd_dummy_unregister_all(void) { int i; diff -ruN linux-2.6.22/sound/drivers/mpu401/mpu401.c linux-2.6.22-alsa/sound/drivers/mpu401/mpu401.c --- linux-2.6.22/sound/drivers/mpu401/mpu401.c 2007-07-28 13:40:44.000000000 +0200 +++ linux-2.6.22-alsa/sound/drivers/mpu401/mpu401.c 2007-09-01 20:55:22.000000000 +0200 @@ -228,7 +228,7 @@ static struct pnp_driver snd_mpu401_pnp_driver; #endif -static void __init_or_module snd_mpu401_unregister_all(void) +static void snd_mpu401_unregister_all(void) { int i; diff -ruN linux-2.6.22/sound/drivers/mts64.c linux-2.6.22-alsa/sound/drivers/mts64.c --- linux-2.6.22/sound/drivers/mts64.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/drivers/mts64.c 2007-09-01 20:55:22.000000000 +0200 @@ -440,15 +440,7 @@ *********************************************************************/ /* SMPTE Switch */ -static int snd_mts64_ctl_smpte_switch_info(struct snd_kcontrol *kctl, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_mts64_ctl_smpte_switch_info snd_ctl_boolean_mono_info static int snd_mts64_ctl_smpte_switch_get(struct snd_kcontrol* kctl, struct snd_ctl_elem_value *uctl) diff -ruN linux-2.6.22/sound/drivers/opl3/Makefile linux-2.6.22-alsa/sound/drivers/opl3/Makefile --- linux-2.6.22/sound/drivers/opl3/Makefile 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/drivers/opl3/Makefile 2007-09-01 20:55:22.000000000 +0200 @@ -4,10 +4,8 @@ # snd-opl3-lib-objs := opl3_lib.o opl3_synth.o -snd-opl3-synth-objs := opl3_seq.o opl3_midi.o opl3_drums.o -ifeq ($(CONFIG_SND_SEQUENCER_OSS),y) -snd-opl3-synth-objs += opl3_oss.o -endif +snd-opl3-synth-y := opl3_seq.o opl3_midi.o opl3_drums.o +snd-opl3-synth-$(CONFIG_SND_SEQUENCER_OSS) += opl3_oss.o # # this function returns: diff -ruN linux-2.6.22/sound/drivers/portman2x4.c linux-2.6.22-alsa/sound/drivers/portman2x4.c --- linux-2.6.22/sound/drivers/portman2x4.c 2007-07-28 13:40:44.000000000 +0200 +++ linux-2.6.22-alsa/sound/drivers/portman2x4.c 2007-09-01 20:55:22.000000000 +0200 @@ -833,7 +833,7 @@ /********************************************************************* * module init stuff *********************************************************************/ -static void __init_or_module snd_portman_unregister_all(void) +static void snd_portman_unregister_all(void) { int i; diff -ruN linux-2.6.22/sound/drivers/serial-u16550.c linux-2.6.22-alsa/sound/drivers/serial-u16550.c --- linux-2.6.22/sound/drivers/serial-u16550.c 2007-07-28 13:40:44.000000000 +0200 +++ linux-2.6.22-alsa/sound/drivers/serial-u16550.c 2007-09-01 20:55:22.000000000 +0200 @@ -998,7 +998,7 @@ }, }; -static void __init_or_module snd_serial_unregister_all(void) +static void snd_serial_unregister_all(void) { int i; diff -ruN linux-2.6.22/sound/drivers/virmidi.c linux-2.6.22-alsa/sound/drivers/virmidi.c --- linux-2.6.22/sound/drivers/virmidi.c 2007-07-28 13:40:44.000000000 +0200 +++ linux-2.6.22-alsa/sound/drivers/virmidi.c 2007-09-01 20:55:22.000000000 +0200 @@ -145,7 +145,7 @@ }, }; -static void __init_or_module snd_virmidi_unregister_all(void) +static void snd_virmidi_unregister_all(void) { int i; diff -ruN linux-2.6.22/sound/drivers/vx/vx_mixer.c linux-2.6.22-alsa/sound/drivers/vx/vx_mixer.c --- linux-2.6.22/sound/drivers/vx/vx_mixer.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/drivers/vx/vx_mixer.c 2007-09-01 20:55:22.000000000 +0200 @@ -647,14 +647,7 @@ return 0; } -static int vx_audio_sw_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 2; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define vx_audio_sw_info snd_ctl_boolean_stereo_info static int vx_audio_sw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -865,14 +858,7 @@ return 0; } -static int vx_saturation_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 2; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define vx_saturation_info snd_ctl_boolean_stereo_info static int vx_saturation_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { diff -ruN linux-2.6.22/sound/i2c/Makefile linux-2.6.22-alsa/sound/i2c/Makefile --- linux-2.6.22/sound/i2c/Makefile 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/i2c/Makefile 2007-09-01 20:55:22.000000000 +0200 @@ -7,9 +7,7 @@ snd-cs8427-objs := cs8427.o snd-tea6330t-objs := tea6330t.o -ifeq ($(subst m,y,$(CONFIG_L3)),y) - obj-$(CONFIG_L3) += l3/ -endif +obj-$(CONFIG_L3) += l3/ obj-$(CONFIG_SND) += other/ diff -ruN linux-2.6.22/sound/i2c/other/ak4114.c linux-2.6.22-alsa/sound/i2c/other/ak4114.c --- linux-2.6.22/sound/i2c/other/ak4114.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/i2c/other/ak4114.c 2007-09-01 20:55:22.000000000 +0200 @@ -200,15 +200,7 @@ return 0; } -static int snd_ak4114_in_bit_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_ak4114_in_bit_info snd_ctl_boolean_mono_info static int snd_ak4114_in_bit_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) diff -ruN linux-2.6.22/sound/i2c/other/ak4117.c linux-2.6.22-alsa/sound/i2c/other/ak4117.c --- linux-2.6.22/sound/i2c/other/ak4117.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/i2c/other/ak4117.c 2007-09-01 20:55:22.000000000 +0200 @@ -181,15 +181,7 @@ return 0; } -static int snd_ak4117_in_bit_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_ak4117_in_bit_info snd_ctl_boolean_mono_info static int snd_ak4117_in_bit_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) diff -ruN linux-2.6.22/sound/i2c/other/ak4xxx-adda.c linux-2.6.22-alsa/sound/i2c/other/ak4xxx-adda.c --- linux-2.6.22/sound/i2c/other/ak4xxx-adda.c 2007-07-28 13:40:44.000000000 +0200 +++ linux-2.6.22-alsa/sound/i2c/other/ak4xxx-adda.c 2007-09-01 20:55:22.000000000 +0200 @@ -463,15 +463,7 @@ return change; } -static int ak4xxx_switch_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define ak4xxx_switch_info snd_ctl_boolean_mono_info static int ak4xxx_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -481,8 +473,8 @@ int addr = AK_GET_ADDR(kcontrol->private_value); int shift = AK_GET_SHIFT(kcontrol->private_value); int invert = AK_GET_INVERT(kcontrol->private_value); - unsigned char val = snd_akm4xxx_get(ak, chip, addr); - + /* we observe the (1<value.integer.value[0] = (val & (1<num_dacs; ) { + /* mute control for Revolution 7.1 - AK4381 */ + if (ak->type == SND_AK4381 + && ak->dac_info[mixer_ch].switch_name) { + memset(&knew, 0, sizeof(knew)); + knew.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + knew.count = 1; + knew.access = SNDRV_CTL_ELEM_ACCESS_READWRITE; + knew.name = ak->dac_info[mixer_ch].switch_name; + knew.info = ak4xxx_switch_info; + knew.get = ak4xxx_switch_get; + knew.put = ak4xxx_switch_put; + knew.access = 0; + /* register 1, bit 0 (SMUTE): 0 = normal operation, + 1 = mute */ + knew.private_value = + AK_COMPOSE(idx/2, 1, 0, 0) | AK_INVERT; + err = snd_ctl_add(ak->card, snd_ctl_new1(&knew, ak)); + if (err < 0) + return err; + } memset(&knew, 0, sizeof(knew)); if (! ak->dac_info || ! ak->dac_info[mixer_ch].name) { knew.name = "DAC Volume"; diff -ruN linux-2.6.22/sound/i2c/other/pt2258.c linux-2.6.22-alsa/sound/i2c/other/pt2258.c --- linux-2.6.22/sound/i2c/other/pt2258.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/i2c/other/pt2258.c 2007-09-01 20:55:22.000000000 +0200 @@ -140,15 +140,7 @@ return -EIO; } -static int pt2258_switch_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define pt2258_switch_info snd_ctl_boolean_mono_info static int pt2258_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) diff -ruN linux-2.6.22/sound/i2c/tea6330t.c linux-2.6.22-alsa/sound/i2c/tea6330t.c --- linux-2.6.22/sound/i2c/tea6330t.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/i2c/tea6330t.c 2007-09-01 20:55:22.000000000 +0200 @@ -142,15 +142,7 @@ .info = snd_tea6330t_info_master_switch, \ .get = snd_tea6330t_get_master_switch, .put = snd_tea6330t_put_master_switch } -static int snd_tea6330t_info_master_switch(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 2; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_tea6330t_info_master_switch snd_ctl_boolean_stereo_info static int snd_tea6330t_get_master_switch(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) diff -ruN linux-2.6.22/sound/isa/Kconfig linux-2.6.22-alsa/sound/isa/Kconfig --- linux-2.6.22/sound/isa/Kconfig 2007-07-28 13:40:44.000000000 +0200 +++ linux-2.6.22-alsa/sound/isa/Kconfig 2007-09-01 20:55:22.000000000 +0200 @@ -1,8 +1,5 @@ # ALSA ISA drivers -menu "ISA devices" - depends on SND!=n && ISA && ISA_DMA_API - config SND_AD1848_LIB tristate select SND_PCM @@ -11,6 +8,22 @@ tristate select SND_PCM +config SND_SB_COMMON + tristate + +config SND_SB8_DSP + tristate + select SND_PCM + select SND_SB_COMMON + +config SND_SB16_DSP + tristate + select SND_PCM + select SND_SB_COMMON + +menu "ISA devices" + depends on SND!=n && ISA && ISA_DMA_API + config SND_ADLIB tristate "AdLib FM card" depends on SND @@ -55,7 +68,7 @@ select ISAPNP select SND_OPL3_LIB select SND_MPU401_UART - select SND_PCM + select SND_SB16_DSP help Say Y here to include support for soundcards based on Avance Logic ALS100, ALS110, ALS120 and ALS200 chips. @@ -81,6 +94,7 @@ tristate "C-Media CMI8330" depends on SND select SND_AD1848_LIB + select SND_SB16_DSP help Say Y here to include support for soundcards based on the C-Media CMI8330 chip. @@ -132,7 +146,7 @@ select ISAPNP select SND_OPL3_LIB select SND_MPU401_UART - select SND_PCM + select SND_SB16_DSP help Say Y here to include support for soundcards based on the Diamond Technologies DT-019X or Avance Logic ALS-007 chips. @@ -145,7 +159,7 @@ depends on SND && PNP && ISA select ISAPNP select SND_MPU401_UART - select SND_PCM + select SND_SB8_DSP help Say Y here to include support for ESS AudioDrive ES968 chips. @@ -321,7 +335,7 @@ depends on SND select SND_OPL3_LIB select SND_RAWMIDI - select SND_PCM + select SND_SB8_DSP help Say Y here to include support for Creative Sound Blaster 1.0/ 2.0/Pro (8-bit) or 100% compatible soundcards. @@ -334,7 +348,7 @@ depends on SND select SND_OPL3_LIB select SND_MPU401_UART - select SND_PCM + select SND_SB16_DSP help Say Y here to include support for Sound Blaster 16 soundcards (including the Plug and Play version). @@ -347,7 +361,7 @@ depends on SND select SND_OPL3_LIB select SND_MPU401_UART - select SND_PCM + select SND_SB16_DSP help Say Y here to include support for Sound Blaster AWE soundcards (including the Plug and Play version). @@ -400,7 +414,7 @@ config SND_WAVEFRONT tristate "Turtle Beach Maui,Tropez,Tropez+ (Wavefront)" depends on SND - select FW_LOADER if !SND_WAVEFRONT_FIRMWARE_IN_KERNEL + select FW_LOADER select SND_OPL3_LIB select SND_MPU401_UART select SND_CS4231_LIB @@ -416,8 +430,9 @@ depends on SND_WAVEFRONT default y help - Say Y here to include the static firmware built in the kernel - for the Wavefront driver. If you choose N here, you need to - install the firmware files from the alsa-firmware package. + Say Y here to include the static firmware for FX DSP built in + the kernel for the Wavefront driver. If you choose N here, + you need to install the firmware files from the + alsa-firmware package. endmenu diff -ruN linux-2.6.22/sound/isa/ad1816a/ad1816a_lib.c linux-2.6.22-alsa/sound/isa/ad1816a/ad1816a_lib.c --- linux-2.6.22/sound/isa/ad1816a/ad1816a_lib.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/isa/ad1816a/ad1816a_lib.c 2007-09-01 20:55:22.000000000 +0200 @@ -453,7 +453,6 @@ if ((error = snd_ad1816a_open(chip, AD1816A_MODE_PLAYBACK)) < 0) return error; - snd_pcm_set_sync(substream); runtime->hw = snd_ad1816a_playback; snd_pcm_limit_isa_dma_size(chip->dma1, &runtime->hw.buffer_bytes_max); snd_pcm_limit_isa_dma_size(chip->dma1, &runtime->hw.period_bytes_max); @@ -469,7 +468,6 @@ if ((error = snd_ad1816a_open(chip, AD1816A_MODE_CAPTURE)) < 0) return error; - snd_pcm_set_sync(substream); runtime->hw = snd_ad1816a_capture; snd_pcm_limit_isa_dma_size(chip->dma2, &runtime->hw.buffer_bytes_max); snd_pcm_limit_isa_dma_size(chip->dma2, &runtime->hw.period_bytes_max); diff -ruN linux-2.6.22/sound/isa/ad1848/Makefile linux-2.6.22-alsa/sound/isa/ad1848/Makefile --- linux-2.6.22/sound/isa/ad1848/Makefile 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/isa/ad1848/Makefile 2007-09-01 20:55:22.000000000 +0200 @@ -7,9 +7,6 @@ snd-ad1848-objs := ad1848.o # Toplevel Module Dependency -obj-$(CONFIG_SND_CMI8330) += snd-ad1848-lib.o -obj-$(CONFIG_SND_SGALAXY) += snd-ad1848-lib.o -obj-$(CONFIG_SND_AD1848) += snd-ad1848.o snd-ad1848-lib.o -obj-$(CONFIG_SND_OPTI92X_AD1848) += snd-ad1848-lib.o +obj-$(CONFIG_SND_AD1848) += snd-ad1848.o +obj-$(CONFIG_SND_AD1848_LIB) += snd-ad1848-lib.o -obj-m := $(sort $(obj-m)) diff -ruN linux-2.6.22/sound/isa/ad1848/ad1848_lib.c linux-2.6.22-alsa/sound/isa/ad1848/ad1848_lib.c --- linux-2.6.22/sound/isa/ad1848/ad1848_lib.c 2007-07-28 13:40:44.000000000 +0200 +++ linux-2.6.22-alsa/sound/isa/ad1848/ad1848_lib.c 2007-09-01 20:55:22.000000000 +0200 @@ -245,7 +245,7 @@ snd_printk(KERN_ERR "mce_down - auto calibration time out (2)\n"); return; } - time = schedule_timeout_interruptible(time); + time = schedule_timeout(time); spin_lock_irqsave(&chip->reg_lock, flags); } #if 0 @@ -258,7 +258,7 @@ snd_printk(KERN_ERR "mce_down - auto calibration time out (3)\n"); return; } - time = schedule_timeout_interruptible(time); + time = schedule_timeout(time); spin_lock_irqsave(&chip->reg_lock, flags); } spin_unlock_irqrestore(&chip->reg_lock, flags); diff -ruN linux-2.6.22/sound/isa/cs423x/Makefile linux-2.6.22-alsa/sound/isa/cs423x/Makefile --- linux-2.6.22/sound/isa/cs423x/Makefile 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/isa/cs423x/Makefile 2007-09-01 20:55:22.000000000 +0200 @@ -10,17 +10,8 @@ snd-cs4236-objs := cs4236.o # Toplevel Module Dependency -obj-$(CONFIG_SND_AZT2320) += snd-cs4231-lib.o -obj-$(CONFIG_SND_MIRO) += snd-cs4231-lib.o -obj-$(CONFIG_SND_OPL3SA2) += snd-cs4231-lib.o -obj-$(CONFIG_SND_CS4231) += snd-cs4231.o snd-cs4231-lib.o -obj-$(CONFIG_SND_CS4232) += snd-cs4232.o snd-cs4231-lib.o -obj-$(CONFIG_SND_CS4236) += snd-cs4236.o snd-cs4236-lib.o snd-cs4231-lib.o -obj-$(CONFIG_SND_GUSMAX) += snd-cs4231-lib.o -obj-$(CONFIG_SND_INTERWAVE) += snd-cs4231-lib.o -obj-$(CONFIG_SND_INTERWAVE_STB) += snd-cs4231-lib.o -obj-$(CONFIG_SND_OPTI92X_CS4231) += snd-cs4231-lib.o -obj-$(CONFIG_SND_WAVEFRONT) += snd-cs4231-lib.o -obj-$(CONFIG_SND_SSCAPE) += snd-cs4231-lib.o +obj-$(CONFIG_SND_CS4231_LIB) += snd-cs4231-lib.o +obj-$(CONFIG_SND_CS4231) += snd-cs4231.o +obj-$(CONFIG_SND_CS4232) += snd-cs4232.o +obj-$(CONFIG_SND_CS4236) += snd-cs4236.o snd-cs4236-lib.o -obj-m := $(sort $(obj-m)) diff -ruN linux-2.6.22/sound/isa/cs423x/cs4231_lib.c linux-2.6.22-alsa/sound/isa/cs423x/cs4231_lib.c --- linux-2.6.22/sound/isa/cs423x/cs4231_lib.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/isa/cs423x/cs4231_lib.c 2007-09-01 20:55:22.000000000 +0200 @@ -555,6 +555,8 @@ snd_cs4231_out(chip, CS4231_PLAYBK_FORMAT, chip->image[CS4231_PLAYBK_FORMAT] = pdfr); } spin_unlock_irqrestore(&chip->reg_lock, flags); + if (chip->hardware == CS4231_HW_OPL3SA2) + udelay(100); /* this seems to help */ snd_cs4231_mce_down(chip); } snd_cs4231_calibrate_mute(chip, 0); diff -ruN linux-2.6.22/sound/isa/es18xx.c linux-2.6.22-alsa/sound/isa/es18xx.c --- linux-2.6.22/sound/isa/es18xx.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/isa/es18xx.c 2007-09-01 20:55:22.000000000 +0200 @@ -1071,14 +1071,7 @@ return (snd_es18xx_mixer_bits(chip, 0x1c, 0x07, val) != val) || retVal; } -static int snd_es18xx_info_spatializer_enable(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_es18xx_info_spatializer_enable snd_ctl_boolean_mono_info static int snd_es18xx_get_spatializer_enable(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -1120,14 +1113,7 @@ return 0; } -static int snd_es18xx_info_hw_switch(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 2; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_es18xx_info_hw_switch snd_ctl_boolean_stereo_info static int snd_es18xx_get_hw_switch(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -2042,6 +2028,7 @@ static struct pnp_device_id snd_audiodrive_pnpbiosids[] = { { .id = "ESS1869" }, + { .id = "ESS1879" }, { .id = "" } /* end */ }; diff -ruN linux-2.6.22/sound/isa/gus/gus_mixer.c linux-2.6.22-alsa/sound/isa/gus/gus_mixer.c --- linux-2.6.22/sound/isa/gus/gus_mixer.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/isa/gus/gus_mixer.c 2007-09-01 20:55:22.000000000 +0200 @@ -36,14 +36,7 @@ .get = snd_gf1_get_single, .put = snd_gf1_put_single, \ .private_value = shift | (invert << 8) } -static int snd_gf1_info_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_gf1_info_single snd_ctl_boolean_mono_info static int snd_gf1_get_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { diff -ruN linux-2.6.22/sound/isa/opl3sa2.c linux-2.6.22-alsa/sound/isa/opl3sa2.c --- linux-2.6.22/sound/isa/opl3sa2.c 2007-07-28 13:40:44.000000000 +0200 +++ linux-2.6.22-alsa/sound/isa/opl3sa2.c 2007-09-01 20:55:22.000000000 +0200 @@ -164,6 +164,8 @@ { .id = "YMH0801", .devs = { { "YMH0021" } } }, /* NeoMagic MagicWave 3DX */ { .id = "NMX2200", .devs = { { "YMH2210" } } }, + /* NeoMagic MagicWave 3D */ + { .id = "NMX2200", .devs = { { "NMX2210" } } }, /* --- */ { .id = "" } /* end */ }; @@ -251,6 +253,7 @@ /* 0x03 - YM715B */ /* 0x04 - YM719 - OPL-SA4? */ /* 0x05 - OPL3-SA3 - Libretto 100 */ + /* 0x07 - unknown - Neomagic MagicWave 3D */ break; } str[0] = chip->version + '0'; diff -ruN linux-2.6.22/sound/isa/opti9xx/miro.c linux-2.6.22-alsa/sound/isa/opti9xx/miro.c --- linux-2.6.22/sound/isa/opti9xx/miro.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/isa/opti9xx/miro.c 2007-09-01 20:55:22.000000000 +0200 @@ -242,14 +242,7 @@ * MIXER part */ -static int snd_miro_info_capture(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - - return 0; -} +#define snd_miro_info_capture snd_ctl_boolean_mono_info static int snd_miro_get_capture(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -344,14 +337,7 @@ return change; } -static int snd_miro_info_amp(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - - return 0; -} +#define snd_miro_info_amp snd_ctl_boolean_mono_info static int snd_miro_get_amp(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) diff -ruN linux-2.6.22/sound/isa/opti9xx/opti92x-ad1848.c linux-2.6.22-alsa/sound/isa/opti9xx/opti92x-ad1848.c --- linux-2.6.22/sound/isa/opti9xx/opti92x-ad1848.c 2007-07-28 13:40:44.000000000 +0200 +++ linux-2.6.22-alsa/sound/isa/opti9xx/opti92x-ad1848.c 2007-09-01 20:55:22.000000000 +0200 @@ -1927,10 +1927,12 @@ static int __devinit snd_opti9xx_isa_match(struct device *devptr, unsigned int dev) { +#ifdef CONFIG_PNP if (snd_opti9xx_pnp_is_probed) return 0; if (isapnp) return 0; +#endif return 1; } @@ -2096,6 +2098,7 @@ pnp_register_card_driver(&opti9xx_pnpc_driver); if (snd_opti9xx_pnp_is_probed) return 0; + pnp_unregister_card_driver(&opti9xx_pnpc_driver); #endif return isa_register_driver(&snd_opti9xx_driver, 1); } diff -ruN linux-2.6.22/sound/isa/sb/Makefile linux-2.6.22-alsa/sound/isa/sb/Makefile --- linux-2.6.22/sound/isa/sb/Makefile 2007-07-28 13:40:44.000000000 +0200 +++ linux-2.6.22-alsa/sound/isa/sb/Makefile 2007-09-01 20:55:22.000000000 +0200 @@ -22,14 +22,13 @@ sequencer = $(if $(subst y,,$(CONFIG_SND_SEQUENCER)),$(if $(1),m),$(if $(CONFIG_SND_SEQUENCER),$(1))) # Toplevel Module Dependency -obj-$(CONFIG_SND_ALS100) += snd-sb16-dsp.o snd-sb-common.o -obj-$(CONFIG_SND_CMI8330) += snd-sb16-dsp.o snd-sb-common.o -obj-$(CONFIG_SND_DT019X) += snd-sb16-dsp.o snd-sb-common.o -obj-$(CONFIG_SND_SB8) += snd-sb8.o snd-sb8-dsp.o snd-sb-common.o -obj-$(CONFIG_SND_SB16) += snd-sb16.o snd-sb16-dsp.o snd-sb-common.o -obj-$(CONFIG_SND_SBAWE) += snd-sbawe.o snd-sb16-dsp.o snd-sb-common.o -obj-$(CONFIG_SND_ES968) += snd-es968.o snd-sb8-dsp.o snd-sb-common.o -obj-$(CONFIG_SND_ALS4000) += snd-sb-common.o +obj-$(CONFIG_SND_SB_COMMON) += snd-sb-common.o +obj-$(CONFIG_SND_SB16_DSP) += snd-sb16-dsp.o +obj-$(CONFIG_SND_SB8_DSP) += snd-sb8-dsp.o +obj-$(CONFIG_SND_SB8) += snd-sb8.o +obj-$(CONFIG_SND_SB16) += snd-sb16.o +obj-$(CONFIG_SND_SBAWE) += snd-sbawe.o +obj-$(CONFIG_SND_ES968) += snd-es968.o ifeq ($(CONFIG_SND_SB16_CSP),y) obj-$(CONFIG_SND_SB16) += snd-sb16-csp.o obj-$(CONFIG_SND_SBAWE) += snd-sb16-csp.o diff -ruN linux-2.6.22/sound/isa/sb/sb16_csp.c linux-2.6.22-alsa/sound/isa/sb/sb16_csp.c --- linux-2.6.22/sound/isa/sb/sb16_csp.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/isa/sb/sb16_csp.c 2007-09-01 20:55:22.000000000 +0200 @@ -979,14 +979,7 @@ * QSound mixer control for PCM */ -static int snd_sb_qsound_switch_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_sb_qsound_switch_info snd_ctl_boolean_mono_info static int snd_sb_qsound_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { diff -ruN linux-2.6.22/sound/isa/sb/sb16_main.c linux-2.6.22-alsa/sound/isa/sb/sb16_main.c --- linux-2.6.22/sound/isa/sb/sb16_main.c 2007-07-28 13:40:44.000000000 +0200 +++ linux-2.6.22-alsa/sound/isa/sb/sb16_main.c 2007-09-01 20:55:22.000000000 +0200 @@ -563,6 +563,11 @@ __open_ok: if (chip->hardware == SB_HW_ALS100) runtime->hw.rate_max = 48000; + if (chip->hardware == SB_HW_CS5530) { + runtime->hw.buffer_bytes_max = 32 * 1024; + runtime->hw.periods_min = 2; + runtime->hw.rate_min = 44100; + } if (chip->mode & SB_RATE_LOCK) runtime->hw.rate_min = runtime->hw.rate_max = chip->locked_rate; chip->playback_substream = substream; @@ -633,6 +638,11 @@ __open_ok: if (chip->hardware == SB_HW_ALS100) runtime->hw.rate_max = 48000; + if (chip->hardware == SB_HW_CS5530) { + runtime->hw.buffer_bytes_max = 32 * 1024; + runtime->hw.periods_min = 2; + runtime->hw.rate_min = 44100; + } if (chip->mode & SB_RATE_LOCK) runtime->hw.rate_min = runtime->hw.rate_max = chip->locked_rate; chip->capture_substream = substream; diff -ruN linux-2.6.22/sound/isa/sb/sb_common.c linux-2.6.22-alsa/sound/isa/sb/sb_common.c --- linux-2.6.22/sound/isa/sb/sb_common.c 2007-07-28 13:40:44.000000000 +0200 +++ linux-2.6.22-alsa/sound/isa/sb/sb_common.c 2007-09-01 20:55:22.000000000 +0200 @@ -128,7 +128,7 @@ minor = version & 0xff; snd_printdd("SB [0x%lx]: DSP chip found, version = %i.%i\n", chip->port, major, minor); - + switch (chip->hardware) { case SB_HW_AUTO: switch (major) { @@ -168,6 +168,9 @@ case SB_HW_DT019X: str = "(DT019X/ALS007)"; break; + case SB_HW_CS5530: + str = "16 (CS5530)"; + break; default: return -ENODEV; } @@ -231,7 +234,9 @@ chip->dma16 = -1; chip->port = port; - if (request_irq(irq, irq_handler, hardware == SB_HW_ALS4000 ? + if (request_irq(irq, irq_handler, + (hardware == SB_HW_ALS4000 || + hardware == SB_HW_CS5530) ? IRQF_SHARED : IRQF_DISABLED, "SoundBlaster", (void *) chip)) { snd_printk(KERN_ERR "sb: can't grab irq %d\n", irq); diff -ruN linux-2.6.22/sound/isa/sb/sb_mixer.c linux-2.6.22-alsa/sound/isa/sb/sb_mixer.c --- linux-2.6.22/sound/isa/sb/sb_mixer.c 2007-07-28 13:40:44.000000000 +0200 +++ linux-2.6.22-alsa/sound/isa/sb/sb_mixer.c 2007-09-01 20:55:22.000000000 +0200 @@ -821,6 +821,7 @@ break; case SB_HW_16: case SB_HW_ALS100: + case SB_HW_CS5530: if ((err = snd_sbmixer_init(chip, snd_sb16_controls, ARRAY_SIZE(snd_sb16_controls), @@ -950,6 +951,7 @@ break; case SB_HW_16: case SB_HW_ALS100: + case SB_HW_CS5530: save_mixer(chip, sb16_saved_regs, ARRAY_SIZE(sb16_saved_regs)); break; case SB_HW_ALS4000: @@ -975,6 +977,7 @@ break; case SB_HW_16: case SB_HW_ALS100: + case SB_HW_CS5530: restore_mixer(chip, sb16_saved_regs, ARRAY_SIZE(sb16_saved_regs)); break; case SB_HW_ALS4000: diff -ruN linux-2.6.22/sound/isa/sscape.c linux-2.6.22-alsa/sound/isa/sscape.c --- linux-2.6.22/sound/isa/sscape.c 2007-07-28 13:40:44.000000000 +0200 +++ linux-2.6.22-alsa/sound/isa/sscape.c 2007-09-01 20:55:22.000000000 +0200 @@ -382,7 +382,7 @@ unsigned long flags; unsigned char x; - schedule_timeout_interruptible(1); + schedule_timeout(1); spin_lock_irqsave(&s->lock, flags); x = inb(HOST_DATA_IO(s->io_base)); @@ -409,7 +409,7 @@ unsigned long flags; unsigned char x; - schedule_timeout_interruptible(1); + schedule_timeout(1); spin_lock_irqsave(&s->lock, flags); x = inb(HOST_DATA_IO(s->io_base)); diff -ruN linux-2.6.22/sound/isa/wavefront/wavefront_synth.c linux-2.6.22-alsa/sound/isa/wavefront/wavefront_synth.c --- linux-2.6.22/sound/isa/wavefront/wavefront_synth.c 2007-07-28 13:40:44.000000000 +0200 +++ linux-2.6.22-alsa/sound/isa/wavefront/wavefront_synth.c 2007-09-01 20:55:22.000000000 +0200 @@ -27,6 +27,7 @@ #include #include #include +#include #include #include #include @@ -53,9 +54,8 @@ /* XXX this needs to be made firmware and hardware version dependent */ -static char *ospath = "/etc/sound/wavefront.os"; /* where to find a processed - version of the WaveFront OS - */ +#define DEFAULT_OSPATH "wavefront.os" +static char *ospath = DEFAULT_OSPATH; /* the firmware file name */ static int wait_usecs = 150; /* This magic number seems to give pretty optimal throughput based on my limited experimentation. @@ -97,7 +97,7 @@ module_param(sleep_tries, int, 0444); MODULE_PARM_DESC(sleep_tries, "how many times to try sleeping during a wait"); module_param(ospath, charp, 0444); -MODULE_PARM_DESC(ospath, "full pathname to processed ICS2115 OS firmware"); +MODULE_PARM_DESC(ospath, "pathname to processed ICS2115 OS firmware"); module_param(reset_time, int, 0444); MODULE_PARM_DESC(reset_time, "how long to wait for a reset to take effect"); module_param(ramcheck_time, int, 0444); @@ -1780,7 +1780,7 @@ outb (val,port); spin_unlock_irq(&dev->irq_lock); while (1) { - if ((timeout = schedule_timeout_interruptible(timeout)) == 0) + if ((timeout = schedule_timeout(timeout)) == 0) return; if (dev->irq_ok) return; @@ -1938,111 +1938,75 @@ return (1); } -#include -#include -#include -#include -#include -#include - - static int __devinit wavefront_download_firmware (snd_wavefront_t *dev, char *path) { - unsigned char section[WF_SECTION_MAX]; - signed char section_length; /* yes, just a char; max value is WF_SECTION_MAX */ + unsigned char *buf; + int len, err; int section_cnt_downloaded = 0; - int fd; - int c; - int i; - mm_segment_t fs; + const struct firmware *firmware; - /* This tries to be a bit cleverer than the stuff Alan Cox did for - the generic sound firmware, in that it actually knows - something about the structure of the Motorola firmware. In - particular, it uses a version that has been stripped of the - 20K of useless header information, and had section lengths - added, making it possible to load the entire OS without any - [kv]malloc() activity, since the longest entity we ever read is - 42 bytes (well, WF_SECTION_MAX) long. - */ - - fs = get_fs(); - set_fs (get_ds()); - - if ((fd = sys_open ((char __user *) path, 0, 0)) < 0) { - snd_printk ("Unable to load \"%s\".\n", - path); + err = request_firmware(&firmware, path, dev->card->dev); + if (err < 0) { + snd_printk(KERN_ERR "firmware (%s) download failed!!!\n", path); return 1; } - while (1) { - int x; - - if ((x = sys_read (fd, (char __user *) §ion_length, sizeof (section_length))) != - sizeof (section_length)) { - snd_printk ("firmware read error.\n"); - goto failure; - } - - if (section_length == 0) { + len = 0; + buf = firmware->data; + for (;;) { + int section_length = *(signed char *)buf; + if (section_length == 0) break; - } - if (section_length < 0 || section_length > WF_SECTION_MAX) { - snd_printk ("invalid firmware section length %d\n", - section_length); + snd_printk(KERN_ERR + "invalid firmware section length %d\n", + section_length); goto failure; } + buf++; + len++; - if (sys_read (fd, (char __user *) section, section_length) != section_length) { - snd_printk ("firmware section " - "read error.\n"); + if (firmware->size < len + section_length) { + snd_printk(KERN_ERR "firmware section read error.\n"); goto failure; } /* Send command */ - - if (wavefront_write (dev, WFC_DOWNLOAD_OS)) { + if (wavefront_write(dev, WFC_DOWNLOAD_OS)) goto failure; - } - for (i = 0; i < section_length; i++) { - if (wavefront_write (dev, section[i])) { + for (; section_length; section_length--) { + if (wavefront_write(dev, *buf)) goto failure; - } + buf++; + len++; } /* get ACK */ - - if (wavefront_wait (dev, STAT_CAN_READ)) { - - if ((c = inb (dev->data_port)) != WF_ACK) { - - snd_printk ("download " - "of section #%d not " - "acknowledged, ack = 0x%x\n", - section_cnt_downloaded + 1, c); - goto failure; - - } - - } else { - snd_printk ("time out for firmware ACK.\n"); + if (!wavefront_wait(dev, STAT_CAN_READ)) { + snd_printk(KERN_ERR "time out for firmware ACK.\n"); + goto failure; + } + err = inb(dev->data_port); + if (err != WF_ACK) { + snd_printk(KERN_ERR + "download of section #%d not " + "acknowledged, ack = 0x%x\n", + section_cnt_downloaded + 1, err); goto failure; } + section_cnt_downloaded++; } - sys_close (fd); - set_fs (fs); + release_firmware(firmware); return 0; failure: - sys_close (fd); - set_fs (fs); - snd_printk ("firmware download failed!!!\n"); + release_firmware(firmware); + snd_printk(KERN_ERR "firmware download failed!!!\n"); return 1; } @@ -2232,3 +2196,5 @@ return 0; } + +MODULE_FIRMWARE(DEFAULT_OSPATH); diff -ruN linux-2.6.22/sound/oss/Makefile linux-2.6.22-alsa/sound/oss/Makefile --- linux-2.6.22/sound/oss/Makefile 2007-07-28 13:40:44.000000000 +0200 +++ linux-2.6.22-alsa/sound/oss/Makefile 2007-09-01 20:55:22.000000000 +0200 @@ -15,42 +15,69 @@ obj-$(CONFIG_SOUND_AEDSP16) += aedsp16.o obj-$(CONFIG_SOUND_PSS) += pss.o ad1848.o mpu401.o obj-$(CONFIG_SOUND_TRIX) += trix.o ad1848.o sb_lib.o uart401.o +obj-$(CONFIG_SOUND_OPL3SA1) += opl3sa.o ad1848.o uart401.o obj-$(CONFIG_SOUND_SSCAPE) += sscape.o ad1848.o mpu401.o +obj-$(CONFIG_SOUND_MAD16) += mad16.o ad1848.o sb_lib.o uart401.o obj-$(CONFIG_SOUND_CS4232) += cs4232.o uart401.o obj-$(CONFIG_SOUND_MSS) += ad1848.o obj-$(CONFIG_SOUND_OPL3SA2) += opl3sa2.o ad1848.o mpu401.o obj-$(CONFIG_SOUND_PAS) += pas2.o sb.o sb_lib.o uart401.o obj-$(CONFIG_SOUND_SB) += sb.o sb_lib.o uart401.o obj-$(CONFIG_SOUND_KAHLUA) += kahlua.o +obj-$(CONFIG_SOUND_WAVEFRONT) += wavefront.o +obj-$(CONFIG_SOUND_MAUI) += maui.o mpu401.o obj-$(CONFIG_SOUND_MPU401) += mpu401.o obj-$(CONFIG_SOUND_UART6850) += uart6850.o +obj-$(CONFIG_SOUND_GUS) += gus.o ad1848.o obj-$(CONFIG_SOUND_ADLIB) += adlib_card.o opl3.o obj-$(CONFIG_SOUND_YM3812) += opl3.o obj-$(CONFIG_SOUND_VMIDI) += v_midi.o obj-$(CONFIG_SOUND_VIDC) += vidc_mod.o obj-$(CONFIG_SOUND_WAVEARTIST) += waveartist.o +obj-$(CONFIG_SOUND_SGALAXY) += sgalaxy.o ad1848.o obj-$(CONFIG_SOUND_AD1816) += ad1816.o obj-$(CONFIG_SOUND_AD1889) += ad1889.o ac97_codec.o obj-$(CONFIG_SOUND_ACI_MIXER) += aci.o +obj-$(CONFIG_SOUND_AWE32_SYNTH) += awe_wave.o obj-$(CONFIG_SOUND_VIA82CXXX) += via82cxxx_audio.o ac97_codec.o ifeq ($(CONFIG_MIDI_VIA82CXXX),y) obj-$(CONFIG_SOUND_VIA82CXXX) += sound.o uart401.o endif +obj-$(CONFIG_SOUND_YMFPCI) += ymfpci.o ac97_codec.o +ifeq ($(CONFIG_SOUND_YMFPCI_LEGACY),y) + obj-$(CONFIG_SOUND_YMFPCI) += opl3.o uart401.o +endif obj-$(CONFIG_SOUND_MSNDCLAS) += msnd.o msnd_classic.o obj-$(CONFIG_SOUND_MSNDPIN) += msnd.o msnd_pinnacle.o obj-$(CONFIG_SOUND_VWSND) += vwsnd.o obj-$(CONFIG_SOUND_NM256) += nm256_audio.o ac97.o obj-$(CONFIG_SOUND_ICH) += i810_audio.o ac97_codec.o +obj-$(CONFIG_SOUND_SONICVIBES) += sonicvibes.o +obj-$(CONFIG_SOUND_CMPCI) += cmpci.o +ifeq ($(CONFIG_SOUND_CMPCI_FM),y) + obj-$(CONFIG_SOUND_CMPCI) += sound.o opl3.o +endif +ifeq ($(CONFIG_SOUND_CMPCI_MIDI),y) + obj-$(CONFIG_SOUND_CMPCI) += sound.o mpu401.o +endif +obj-$(CONFIG_SOUND_ES1370) += es1370.o obj-$(CONFIG_SOUND_ES1371) += es1371.o ac97_codec.o obj-$(CONFIG_SOUND_VRC5477) += nec_vrc5477.o ac97_codec.o -obj-$(CONFIG_SOUND_AU1550_AC97) += au1550_ac97.o ac97_codec.o +obj-$(CONFIG_SOUND_ESSSOLO1) += esssolo1.o obj-$(CONFIG_SOUND_FUSION) += cs46xx.o ac97_codec.o +obj-$(CONFIG_SOUND_MAESTRO) += maestro.o +obj-$(CONFIG_SOUND_MAESTRO3) += maestro3.o ac97_codec.o obj-$(CONFIG_SOUND_TRIDENT) += trident.o ac97_codec.o +obj-$(CONFIG_SOUND_HARMONY) += harmony.o obj-$(CONFIG_SOUND_EMU10K1) += ac97_codec.o -obj-$(CONFIG_SOUND_BCM_CS4297A) += swarm_cs4297a.o +obj-$(CONFIG_SOUND_RME96XX) += rme96xx.o obj-$(CONFIG_SOUND_BT878) += btaudio.o +obj-$(CONFIG_SOUND_ALI5455) += ali5455.o ac97_codec.o +obj-$(CONFIG_SOUND_IT8172) += ite8172.o ac97_codec.o +obj-$(CONFIG_SOUND_FORTE) += forte.o ac97_codec.o +obj-$(CONFIG_SOUND_AD1980) += ac97_plugin_ad1980.o obj-$(CONFIG_SOUND_WM97XX) += ac97_plugin_wm97xx.o ifeq ($(CONFIG_MIDI_EMU10K1),y) @@ -63,20 +90,22 @@ # Declare multi-part drivers. sound-objs := \ - dev_table.o soundcard.o \ - audio.o dmabuf.o \ - midi_synth.o midibuf.o \ - sequencer.o sound_timer.o sys_timer.o + dev_table.o soundcard.o sound_syms.o \ + audio.o audio_syms.o dmabuf.o \ + midi_syms.o midi_synth.o midibuf.o \ + sequencer.o sequencer_syms.o sound_timer.o sys_timer.o +gus-objs := gus_card.o gus_midi.o gus_vol.o gus_wave.o ics2101.o pas2-objs := pas2_card.o pas2_midi.o pas2_mixer.o pas2_pcm.o sb-objs := sb_card.o sb_lib-objs := sb_common.o sb_audio.o sb_midi.o sb_mixer.o sb_ess.o vidc_mod-objs := vidc.o vidc_fill.o +wavefront-objs := wavfront.o wf_midi.o yss225.o -hostprogs-y := bin2hex hex2hex +host-progs := bin2hex hex2hex # Files generated that shall be removed upon make clean -clean-files := msndperm.c msndinit.c pndsperm.c pndspini.c \ +clean-files := maui_boot.h msndperm.c msndinit.c pndsperm.c pndspini.c \ pss_boot.h trix_boot.h # Firmware files that need translation diff -ruN linux-2.6.22/sound/pci/Kconfig linux-2.6.22-alsa/sound/pci/Kconfig --- linux-2.6.22/sound/pci/Kconfig 2007-07-28 13:40:44.000000000 +0200 +++ linux-2.6.22-alsa/sound/pci/Kconfig 2007-09-01 20:55:23.000000000 +0200 @@ -33,6 +33,7 @@ select SND_OPL3_LIB select SND_MPU401_UART select SND_PCM + select SND_SB_COMMON help Say Y here to include support for soundcards based on Avance Logic ALS4000 chips. @@ -169,14 +170,14 @@ will be called snd-ca0106. config SND_CMIPCI - tristate "C-Media 8738, 8338" + tristate "C-Media 8338, 8738, 8768, 8770" depends on SND select SND_OPL3_LIB select SND_MPU401_UART select SND_PCM help - If you want to use soundcards based on C-Media CMI8338 or CMI8738 - chips, say Y here and read + If you want to use soundcards based on C-Media CMI8338, CMI8738, + CMI8768 or CMI8770 chips, say Y here and read . To compile this driver as a module, choose M here: the module @@ -215,6 +216,16 @@ This works better than the old code, so say Y. +config SND_CS5530 + tristate "CS5530 Audio" + depends on SND && ISA_DMA_API + select SND_SB16_DSP + help + Say Y here to include support for audio on Cyrix/NatSemi CS5530 chips. + + To compile this driver as a module, choose M here: the module + will be called snd-cs5530. + config SND_CS5535AUDIO tristate "CS5535/CS5536 Audio" depends on SND && X86 && !X86_64 @@ -489,6 +500,95 @@ To compile this driver as a module, choose M here: the module will be called snd-hda-intel. +config SND_HDA_HWDEP + bool "Build hwdep interface for HD-audio driver" + depends on SND_HDA_INTEL + select SND_HWDEP + help + Say Y here to build a hwdep interface for HD-audio driver. + This interface can be used for out-of-bound communication + with codecs for debugging purposes. + +config SND_HDA_CODEC_REALTEK + bool "Build Realtek HD-audio codec support" + depends on SND_HDA_INTEL + default y + help + Say Y here to include Realtek HD-audio codec support in + snd-hda-intel driver, such as ALC880. + +config SND_HDA_CODEC_ANALOG + bool "Build Analog Device HD-audio codec support" + depends on SND_HDA_INTEL + default y + help + Say Y here to include Analog Device HD-audio codec support in + snd-hda-intel driver, such as AD1986A. + +config SND_HDA_CODEC_SIGMATEL + bool "Build IDT/Sigmatel HD-audio codec support" + depends on SND_HDA_INTEL + default y + help + Say Y here to include IDT (Sigmatel) HD-audio codec support in + snd-hda-intel driver, such as STAC9200. + +config SND_HDA_CODEC_VIA + bool "Build VIA HD-audio codec support" + depends on SND_HDA_INTEL + default y + help + Say Y here to include VIA HD-audio codec support in + snd-hda-intel driver, such as VT1708. + +config SND_HDA_CODEC_ATIHDMI + bool "Build ATI HDMI HD-audio codec support" + depends on SND_HDA_INTEL + default y + help + Say Y here to include ATI HDMI HD-audio codec support in + snd-hda-intel driver, such as ATI RS600 HDMI. + +config SND_HDA_CODEC_CONEXANT + bool "Build Conexant HD-audio codec support" + depends on SND_HDA_INTEL + default y + help + Say Y here to include Conexant HD-audio codec support in + snd-hda-intel driver, such as CX20549. + +config SND_HDA_CODEC_CMEDIA + bool "Build C-Media HD-audio codec support" + depends on SND_HDA_INTEL + default y + help + Say Y here to include C-Media HD-audio codec support in + snd-hda-intel driver, such as CMI9880. + +config SND_HDA_CODEC_SI3054 + bool "Build Silicon Labs 3054 HD-modem codec support" + depends on SND_HDA_INTEL + default y + help + Say Y here to include Silicon Labs 3054 HD-modem codec + (and compatibles) support in snd-hda-intel driver. + +config SND_HDA_GENERIC + bool "Enable generic HD-audio codec parser" + depends on SND_HDA_INTEL + default y + help + Say Y here to enable the generic HD-audio codec parser + in snd-hda-intel driver. + +config SND_HDA_POWER_SAVE + bool "Aggressive power-saving on HD-audio" + depends on SND_HDA_INTEL && EXPERIMENTAL + help + Say Y here to enable more aggressive power-saving mode on + HD-audio driver. The power-saving timeout can be configured + via power_save option or over sysfs on-the-fly. + config SND_HDSP tristate "RME Hammerfall DSP Audio" depends on SND diff -ruN linux-2.6.22/sound/pci/Makefile linux-2.6.22-alsa/sound/pci/Makefile --- linux-2.6.22/sound/pci/Makefile 2007-07-28 13:40:44.000000000 +0200 +++ linux-2.6.22-alsa/sound/pci/Makefile 2007-09-01 20:55:23.000000000 +0200 @@ -12,6 +12,7 @@ snd-bt87x-objs := bt87x.o snd-cmipci-objs := cmipci.o snd-cs4281-objs := cs4281.o +snd-cs5530-objs := cs5530.o snd-ens1370-objs := ens1370.o snd-ens1371-objs := ens1371.o snd-es1938-objs := es1938.o @@ -36,6 +37,7 @@ obj-$(CONFIG_SND_BT87X) += snd-bt87x.o obj-$(CONFIG_SND_CMIPCI) += snd-cmipci.o obj-$(CONFIG_SND_CS4281) += snd-cs4281.o +obj-$(CONFIG_SND_CS5530) += snd-cs5530.o obj-$(CONFIG_SND_ENS1370) += snd-ens1370.o obj-$(CONFIG_SND_ENS1371) += snd-ens1371.o obj-$(CONFIG_SND_ES1938) += snd-es1938.o diff -ruN linux-2.6.22/sound/pci/ac97/ac97_codec.c linux-2.6.22-alsa/sound/pci/ac97/ac97_codec.c --- linux-2.6.22/sound/pci/ac97/ac97_codec.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/pci/ac97/ac97_codec.c 2007-09-01 20:55:22.000000000 +0200 @@ -2036,11 +2036,12 @@ else { udelay(50); if (ac97->scaps & AC97_SCAP_SKIP_AUDIO) - err = ac97_reset_wait(ac97, HZ/2, 1); + err = ac97_reset_wait(ac97, msecs_to_jiffies(500), 1); else { - err = ac97_reset_wait(ac97, HZ/2, 0); + err = ac97_reset_wait(ac97, msecs_to_jiffies(500), 0); if (err < 0) - err = ac97_reset_wait(ac97, HZ/2, 1); + err = ac97_reset_wait(ac97, + msecs_to_jiffies(500), 1); } if (err < 0) { snd_printk(KERN_WARNING "AC'97 %d does not respond - RESET\n", ac97->num); @@ -2104,7 +2105,7 @@ } /* nothing should be in powerdown mode */ snd_ac97_write_cache(ac97, AC97_GENERAL_PURPOSE, 0); - end_time = jiffies + (HZ / 10); + end_time = jiffies + msecs_to_jiffies(100); do { if ((snd_ac97_read(ac97, AC97_POWERDOWN) & 0x0f) == 0x0f) goto __ready_ok; @@ -2136,7 +2137,7 @@ udelay(100); /* nothing should be in powerdown mode */ snd_ac97_write_cache(ac97, AC97_EXTENDED_MSTATUS, 0); - end_time = jiffies + (HZ / 10); + end_time = jiffies + msecs_to_jiffies(100); do { if ((snd_ac97_read(ac97, AC97_EXTENDED_MSTATUS) & tmp) == tmp) goto __ready_ok; @@ -2354,7 +2355,8 @@ * (for avoiding loud click noises for many (OSS) apps * that open/close frequently) */ - schedule_delayed_work(&ac97->power_work, HZ*2); + schedule_delayed_work(&ac97->power_work, + msecs_to_jiffies(2000)); else { cancel_delayed_work(&ac97->power_work); update_power_regs(ac97); @@ -2436,7 +2438,7 @@ /* * restore ac97 status */ -void snd_ac97_restore_status(struct snd_ac97 *ac97) +static void snd_ac97_restore_status(struct snd_ac97 *ac97) { int i; @@ -2457,7 +2459,7 @@ /* * restore IEC958 status */ -void snd_ac97_restore_iec958(struct snd_ac97 *ac97) +static void snd_ac97_restore_iec958(struct snd_ac97 *ac97) { if (ac97->ext_id & AC97_EI_SPDIF) { if (ac97->regs[AC97_EXTENDED_STATUS] & AC97_EA_SPDIF) { @@ -2494,7 +2496,10 @@ snd_ac97_write(ac97, AC97_POWERDOWN, 0); if (! (ac97->flags & AC97_DEFAULT_POWER_OFF)) { - snd_ac97_write(ac97, AC97_RESET, 0); + if (!(ac97->scaps & AC97_SCAP_SKIP_AUDIO)) + snd_ac97_write(ac97, AC97_RESET, 0); + else if (!(ac97->scaps & AC97_SCAP_SKIP_MODEM)) + snd_ac97_write(ac97, AC97_EXTENDED_MID, 0); udelay(100); snd_ac97_write(ac97, AC97_POWERDOWN, 0); } diff -ruN linux-2.6.22/sound/pci/ac97/ac97_patch.c linux-2.6.22-alsa/sound/pci/ac97/ac97_patch.c --- linux-2.6.22/sound/pci/ac97/ac97_patch.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/pci/ac97/ac97_patch.c 2007-09-01 20:55:22.000000000 +0200 @@ -1880,14 +1880,7 @@ return 0; } -static int snd_ac97_ad1888_lohpsel_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_ac97_ad1888_lohpsel_info snd_ctl_boolean_mono_info static int snd_ac97_ad1888_lohpsel_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -2186,15 +2179,7 @@ return 0; } -static int snd_ac97_ad1986_bool_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_ac97_ad1986_bool_info snd_ctl_boolean_mono_info static int snd_ac97_ad1986_lososel_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) diff -ruN linux-2.6.22/sound/pci/ali5451/ali5451.c linux-2.6.22-alsa/sound/pci/ali5451/ali5451.c --- linux-2.6.22/sound/pci/ali5451/ali5451.c 2007-07-28 13:40:44.000000000 +0200 +++ linux-2.6.22-alsa/sound/pci/ali5451/ali5451.c 2007-09-02 13:30:43.000000000 +0200 @@ -239,7 +239,7 @@ struct snd_ali { - unsigned long irq; + int irq; unsigned long port; unsigned char revision; @@ -731,8 +731,7 @@ return; } - count = 0; - while (count++ <= 50000) { + for (count = 0; count <= 50000; count++) { snd_ali_delay(codec, 6); bval = inb(ALI_REG(codec,ALI_SPDIF_CTRL + 1)); R2 = bval & 0x1F; @@ -1805,15 +1804,7 @@ .info = snd_ali5451_spdif_info, .get = snd_ali5451_spdif_get, \ .put = snd_ali5451_spdif_put, .private_value = value} -static int snd_ali5451_spdif_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_ali5451_spdif_info snd_ctl_boolean_mono_info static int snd_ali5451_spdif_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -2343,7 +2334,7 @@ strcpy(card->driver, "ALI5451"); strcpy(card->shortname, "ALI 5451"); - sprintf(card->longname, "%s at 0x%lx, irq %li", + sprintf(card->longname, "%s at 0x%lx, irq %i", card->shortname, codec->port, codec->irq); snd_ali_printk("register card.\n"); diff -ruN linux-2.6.22/sound/pci/ali5451/ali5451.c.orig linux-2.6.22-alsa/sound/pci/ali5451/ali5451.c.orig --- linux-2.6.22/sound/pci/ali5451/ali5451.c.orig 1970-01-01 01:00:00.000000000 +0100 +++ linux-2.6.22-alsa/sound/pci/ali5451/ali5451.c.orig 2007-09-01 20:55:23.000000000 +0200 @@ -0,0 +1,2381 @@ +/* + * Matt Wu + * Apr 26, 2001 + * Routines for control of ALi pci audio M5451 + * + * BUGS: + * -- + * + * TODO: + * -- + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public Lcodecnse as published by + * the Free Software Foundation; either version 2 of the Lcodecnse, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public Lcodecnse for more details. + * + * You should have received a copy of the GNU General Public Lcodecnse + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +MODULE_AUTHOR("Matt Wu "); +MODULE_DESCRIPTION("ALI M5451"); +MODULE_LICENSE("GPL"); +MODULE_SUPPORTED_DEVICE("{{ALI,M5451,pci},{ALI,M5451}}"); + +static int index = SNDRV_DEFAULT_IDX1; /* Index */ +static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ +static int pcm_channels = 32; +static int spdif; + +module_param(index, int, 0444); +MODULE_PARM_DESC(index, "Index value for ALI M5451 PCI Audio."); +module_param(id, charp, 0444); +MODULE_PARM_DESC(id, "ID string for ALI M5451 PCI Audio."); +module_param(pcm_channels, int, 0444); +MODULE_PARM_DESC(pcm_channels, "PCM Channels"); +module_param(spdif, bool, 0444); +MODULE_PARM_DESC(spdif, "Support SPDIF I/O"); + +/* just for backward compatibility */ +static int enable; +module_param(enable, bool, 0444); + + +/* + * Debug part definitions + */ + +/* #define ALI_DEBUG */ + +#ifdef ALI_DEBUG +#define snd_ali_printk(format, args...) printk(KERN_DEBUG format, ##args); +#else +#define snd_ali_printk(format, args...) +#endif + +/* + * Constants definition + */ + +#define DEVICE_ID_ALI5451 ((PCI_VENDOR_ID_AL<<16)|PCI_DEVICE_ID_AL_M5451) + + +#define ALI_CHANNELS 32 + +#define ALI_PCM_IN_CHANNEL 31 +#define ALI_SPDIF_IN_CHANNEL 19 +#define ALI_SPDIF_OUT_CHANNEL 15 +#define ALI_CENTER_CHANNEL 24 +#define ALI_LEF_CHANNEL 23 +#define ALI_SURR_LEFT_CHANNEL 26 +#define ALI_SURR_RIGHT_CHANNEL 25 +#define ALI_MODEM_IN_CHANNEL 21 +#define ALI_MODEM_OUT_CHANNEL 20 + +#define SNDRV_ALI_VOICE_TYPE_PCM 01 +#define SNDRV_ALI_VOICE_TYPE_OTH 02 + +#define ALI_5451_V02 0x02 + +/* + * Direct Registers + */ + +#define ALI_LEGACY_DMAR0 0x00 /* ADR0 */ +#define ALI_LEGACY_DMAR4 0x04 /* CNT0 */ +#define ALI_LEGACY_DMAR11 0x0b /* MOD */ +#define ALI_LEGACY_DMAR15 0x0f /* MMR */ +#define ALI_MPUR0 0x20 +#define ALI_MPUR1 0x21 +#define ALI_MPUR2 0x22 +#define ALI_MPUR3 0x23 + +#define ALI_AC97_WRITE 0x40 +#define ALI_AC97_READ 0x44 + +#define ALI_SCTRL 0x48 +#define ALI_SPDIF_OUT_ENABLE 0x20 +#define ALI_SCTRL_LINE_IN2 (1 << 9) +#define ALI_SCTRL_GPIO_IN2 (1 << 13) +#define ALI_SCTRL_LINE_OUT_EN (1 << 20) +#define ALI_SCTRL_GPIO_OUT_EN (1 << 23) +#define ALI_SCTRL_CODEC1_READY (1 << 24) +#define ALI_SCTRL_CODEC2_READY (1 << 25) +#define ALI_AC97_GPIO 0x4c +#define ALI_AC97_GPIO_ENABLE 0x8000 +#define ALI_AC97_GPIO_DATA_SHIFT 16 +#define ALI_SPDIF_CS 0x70 +#define ALI_SPDIF_CTRL 0x74 +#define ALI_SPDIF_IN_FUNC_ENABLE 0x02 +#define ALI_SPDIF_IN_CH_STATUS 0x40 +#define ALI_SPDIF_OUT_CH_STATUS 0xbf +#define ALI_START 0x80 +#define ALI_STOP 0x84 +#define ALI_CSPF 0x90 +#define ALI_AINT 0x98 +#define ALI_GC_CIR 0xa0 + #define ENDLP_IE 0x00001000 + #define MIDLP_IE 0x00002000 +#define ALI_AINTEN 0xa4 +#define ALI_VOLUME 0xa8 +#define ALI_SBDELTA_DELTA_R 0xac +#define ALI_MISCINT 0xb0 + #define ADDRESS_IRQ 0x00000020 + #define TARGET_REACHED 0x00008000 + #define MIXER_OVERFLOW 0x00000800 + #define MIXER_UNDERFLOW 0x00000400 + #define GPIO_IRQ 0x01000000 +#define ALI_SBBL_SBCL 0xc0 +#define ALI_SBCTRL_SBE2R_SBDD 0xc4 +#define ALI_STIMER 0xc8 +#define ALI_GLOBAL_CONTROL 0xd4 +#define ALI_SPDIF_OUT_SEL_PCM 0x00000400 /* bit 10 */ +#define ALI_SPDIF_IN_SUPPORT 0x00000800 /* bit 11 */ +#define ALI_SPDIF_OUT_CH_ENABLE 0x00008000 /* bit 15 */ +#define ALI_SPDIF_IN_CH_ENABLE 0x00080000 /* bit 19 */ +#define ALI_PCM_IN_ENABLE 0x80000000 /* bit 31 */ + +#define ALI_CSO_ALPHA_FMS 0xe0 +#define ALI_LBA 0xe4 +#define ALI_ESO_DELTA 0xe8 +#define ALI_GVSEL_PAN_VOC_CTRL_EC 0xf0 +#define ALI_EBUF1 0xf4 +#define ALI_EBUF2 0xf8 + +#define ALI_REG(codec, x) ((codec)->port + x) + +#define MAX_CODECS 2 + + +struct snd_ali; +struct snd_ali_voice; + +struct snd_ali_channel_control { + /* register data */ + struct REGDATA { + unsigned int start; + unsigned int stop; + unsigned int aint; + unsigned int ainten; + } data; + + /* register addresses */ + struct REGS { + unsigned int start; + unsigned int stop; + unsigned int aint; + unsigned int ainten; + unsigned int ac97read; + unsigned int ac97write; + } regs; + +}; + +struct snd_ali_voice { + unsigned int number; + unsigned int use :1, + pcm :1, + midi :1, + mode :1, + synth :1, + running :1; + + /* PCM data */ + struct snd_ali *codec; + struct snd_pcm_substream *substream; + struct snd_ali_voice *extra; + + int eso; /* final ESO value for channel */ + int count; /* runtime->period_size */ + + /* --- */ + + void *private_data; + void (*private_free)(void *private_data); +}; + + +struct snd_alidev { + + struct snd_ali_voice voices[ALI_CHANNELS]; + + unsigned int chcnt; /* num of opened channels */ + unsigned int chmap; /* bitmap for opened channels */ + unsigned int synthcount; + +}; + + +#define ALI_GLOBAL_REGS 56 +#define ALI_CHANNEL_REGS 8 +struct snd_ali_image { + u32 regs[ALI_GLOBAL_REGS]; + u32 channel_regs[ALI_CHANNELS][ALI_CHANNEL_REGS]; +}; + + +struct snd_ali { + int irq; + unsigned long port; + unsigned char revision; + + unsigned int hw_initialized :1; + unsigned int spdif_support :1; + + struct pci_dev *pci; + struct pci_dev *pci_m1533; + struct pci_dev *pci_m7101; + + struct snd_card *card; + struct snd_pcm *pcm[MAX_CODECS]; + struct snd_alidev synth; + struct snd_ali_channel_control chregs; + + /* S/PDIF Mask */ + unsigned int spdif_mask; + + unsigned int spurious_irq_count; + unsigned int spurious_irq_max_delta; + + unsigned int num_of_codecs; + + struct snd_ac97_bus *ac97_bus; + struct snd_ac97 *ac97[MAX_CODECS]; + unsigned short ac97_ext_id; + unsigned short ac97_ext_status; + + spinlock_t reg_lock; + spinlock_t voice_alloc; + +#ifdef CONFIG_PM + struct snd_ali_image *image; +#endif +}; + +static struct pci_device_id snd_ali_ids[] = { + {PCI_DEVICE(PCI_VENDOR_ID_AL, PCI_DEVICE_ID_AL_M5451), 0, 0, 0}, + {0, } +}; +MODULE_DEVICE_TABLE(pci, snd_ali_ids); + +static void snd_ali_clear_voices(struct snd_ali *, unsigned int, unsigned int); +static unsigned short snd_ali_codec_peek(struct snd_ali *, int, unsigned short); +static void snd_ali_codec_poke(struct snd_ali *, int, unsigned short, + unsigned short); + +/* + * AC97 ACCESS + */ + +static inline unsigned int snd_ali_5451_peek(struct snd_ali *codec, + unsigned int port) +{ + return (unsigned int)inl(ALI_REG(codec, port)); +} + +static inline void snd_ali_5451_poke(struct snd_ali *codec, + unsigned int port, + unsigned int val) +{ + outl((unsigned int)val, ALI_REG(codec, port)); +} + +static int snd_ali_codec_ready(struct snd_ali *codec, + unsigned int port) +{ + unsigned long end_time; + unsigned int res; + + end_time = jiffies + msecs_to_jiffies(250); + do { + res = snd_ali_5451_peek(codec,port); + if (!(res & 0x8000)) + return 0; + schedule_timeout_uninterruptible(1); + } while (time_after_eq(end_time, jiffies)); + snd_ali_5451_poke(codec, port, res & ~0x8000); + snd_printdd("ali_codec_ready: codec is not ready.\n "); + return -EIO; +} + +static int snd_ali_stimer_ready(struct snd_ali *codec) +{ + unsigned long end_time; + unsigned long dwChk1,dwChk2; + + dwChk1 = snd_ali_5451_peek(codec, ALI_STIMER); + dwChk2 = snd_ali_5451_peek(codec, ALI_STIMER); + + end_time = jiffies + msecs_to_jiffies(250); + do { + dwChk2 = snd_ali_5451_peek(codec, ALI_STIMER); + if (dwChk2 != dwChk1) + return 0; + schedule_timeout_uninterruptible(1); + } while (time_after_eq(end_time, jiffies)); + snd_printk(KERN_ERR "ali_stimer_read: stimer is not ready.\n"); + return -EIO; +} + +static void snd_ali_codec_poke(struct snd_ali *codec,int secondary, + unsigned short reg, + unsigned short val) +{ + unsigned int dwVal; + unsigned int port; + + if (reg >= 0x80) { + snd_printk(KERN_ERR "ali_codec_poke: reg(%xh) invalid.\n", reg); + return; + } + + port = codec->chregs.regs.ac97write; + + if (snd_ali_codec_ready(codec, port) < 0) + return; + if (snd_ali_stimer_ready(codec) < 0) + return; + + dwVal = (unsigned int) (reg & 0xff); + dwVal |= 0x8000 | (val << 16); + if (secondary) + dwVal |= 0x0080; + if (codec->revision == ALI_5451_V02) + dwVal |= 0x0100; + + snd_ali_5451_poke(codec, port, dwVal); + + return ; +} + +static unsigned short snd_ali_codec_peek(struct snd_ali *codec, + int secondary, + unsigned short reg) +{ + unsigned int dwVal; + unsigned int port; + + if (reg >= 0x80) { + snd_printk(KERN_ERR "ali_codec_peek: reg(%xh) invalid.\n", reg); + return ~0; + } + + port = codec->chregs.regs.ac97read; + + if (snd_ali_codec_ready(codec, port) < 0) + return ~0; + if (snd_ali_stimer_ready(codec) < 0) + return ~0; + + dwVal = (unsigned int) (reg & 0xff); + dwVal |= 0x8000; /* bit 15*/ + if (secondary) + dwVal |= 0x0080; + + snd_ali_5451_poke(codec, port, dwVal); + + if (snd_ali_stimer_ready(codec) < 0) + return ~0; + if (snd_ali_codec_ready(codec, port) < 0) + return ~0; + + return (snd_ali_5451_peek(codec, port) & 0xffff0000) >> 16; +} + +static void snd_ali_codec_write(struct snd_ac97 *ac97, + unsigned short reg, + unsigned short val ) +{ + struct snd_ali *codec = ac97->private_data; + + snd_ali_printk("codec_write: reg=%xh data=%xh.\n", reg, val); + if (reg == AC97_GPIO_STATUS) { + outl((val << ALI_AC97_GPIO_DATA_SHIFT) | ALI_AC97_GPIO_ENABLE, + ALI_REG(codec, ALI_AC97_GPIO)); + return; + } + snd_ali_codec_poke(codec, ac97->num, reg, val); + return ; +} + + +static unsigned short snd_ali_codec_read(struct snd_ac97 *ac97, + unsigned short reg) +{ + struct snd_ali *codec = ac97->private_data; + + snd_ali_printk("codec_read reg=%xh.\n", reg); + return snd_ali_codec_peek(codec, ac97->num, reg); +} + +/* + * AC97 Reset + */ + +static int snd_ali_reset_5451(struct snd_ali *codec) +{ + struct pci_dev *pci_dev; + unsigned short wCount, wReg; + unsigned int dwVal; + + pci_dev = codec->pci_m1533; + if (pci_dev) { + pci_read_config_dword(pci_dev, 0x7c, &dwVal); + pci_write_config_dword(pci_dev, 0x7c, dwVal | 0x08000000); + udelay(5000); + pci_read_config_dword(pci_dev, 0x7c, &dwVal); + pci_write_config_dword(pci_dev, 0x7c, dwVal & 0xf7ffffff); + udelay(5000); + } + + pci_dev = codec->pci; + pci_read_config_dword(pci_dev, 0x44, &dwVal); + pci_write_config_dword(pci_dev, 0x44, dwVal | 0x000c0000); + udelay(500); + pci_read_config_dword(pci_dev, 0x44, &dwVal); + pci_write_config_dword(pci_dev, 0x44, dwVal & 0xfffbffff); + udelay(5000); + + wCount = 200; + while(wCount--) { + wReg = snd_ali_codec_peek(codec, 0, AC97_POWERDOWN); + if ((wReg & 0x000f) == 0x000f) + return 0; + udelay(5000); + } + + /* non-fatal if you have a non PM capable codec */ + /* snd_printk(KERN_WARNING "ali5451: reset time out\n"); */ + return 0; +} + +#ifdef CODEC_RESET + +static int snd_ali_reset_codec(struct snd_ali *codec) +{ + struct pci_dev *pci_dev; + unsigned char bVal; + unsigned int dwVal; + unsigned short wCount, wReg; + + pci_dev = codec->pci_m1533; + + pci_read_config_dword(pci_dev, 0x7c, &dwVal); + pci_write_config_dword(pci_dev, 0x7c, dwVal | 0x08000000); + udelay(5000); + pci_read_config_dword(pci_dev, 0x7c, &dwVal); + pci_write_config_dword(pci_dev, 0x7c, dwVal & 0xf7ffffff); + udelay(5000); + + bVal = inb(ALI_REG(codec,ALI_SCTRL)); + bVal |= 0x02; + outb(ALI_REG(codec,ALI_SCTRL),bVal); + udelay(5000); + bVal = inb(ALI_REG(codec,ALI_SCTRL)); + bVal &= 0xfd; + outb(ALI_REG(codec,ALI_SCTRL),bVal); + udelay(15000); + + wCount = 200; + while (wCount--) { + wReg = snd_ali_codec_read(codec->ac97, AC97_POWERDOWN); + if ((wReg & 0x000f) == 0x000f) + return 0; + udelay(5000); + } + return -1; +} + +#endif + +/* + * ALI 5451 Controller + */ + +static void snd_ali_enable_special_channel(struct snd_ali *codec, + unsigned int channel) +{ + unsigned long dwVal; + + dwVal = inl(ALI_REG(codec, ALI_GLOBAL_CONTROL)); + dwVal |= 1 << (channel & 0x0000001f); + outl(dwVal, ALI_REG(codec, ALI_GLOBAL_CONTROL)); +} + +static void snd_ali_disable_special_channel(struct snd_ali *codec, + unsigned int channel) +{ + unsigned long dwVal; + + dwVal = inl(ALI_REG(codec, ALI_GLOBAL_CONTROL)); + dwVal &= ~(1 << (channel & 0x0000001f)); + outl(dwVal, ALI_REG(codec, ALI_GLOBAL_CONTROL)); +} + +static void snd_ali_enable_address_interrupt(struct snd_ali *codec) +{ + unsigned int gc; + + gc = inl(ALI_REG(codec, ALI_GC_CIR)); + gc |= ENDLP_IE; + gc |= MIDLP_IE; + outl( gc, ALI_REG(codec, ALI_GC_CIR)); +} + +static void snd_ali_disable_address_interrupt(struct snd_ali *codec) +{ + unsigned int gc; + + gc = inl(ALI_REG(codec, ALI_GC_CIR)); + gc &= ~ENDLP_IE; + gc &= ~MIDLP_IE; + outl(gc, ALI_REG(codec, ALI_GC_CIR)); +} + +#if 0 /* not used */ +static void snd_ali_enable_voice_irq(struct snd_ali *codec, + unsigned int channel) +{ + unsigned int mask; + struct snd_ali_channel_control *pchregs = &(codec->chregs); + + snd_ali_printk("enable_voice_irq channel=%d\n",channel); + + mask = 1 << (channel & 0x1f); + pchregs->data.ainten = inl(ALI_REG(codec, pchregs->regs.ainten)); + pchregs->data.ainten |= mask; + outl(pchregs->data.ainten, ALI_REG(codec, pchregs->regs.ainten)); +} +#endif + +static void snd_ali_disable_voice_irq(struct snd_ali *codec, + unsigned int channel) +{ + unsigned int mask; + struct snd_ali_channel_control *pchregs = &(codec->chregs); + + snd_ali_printk("disable_voice_irq channel=%d\n",channel); + + mask = 1 << (channel & 0x1f); + pchregs->data.ainten = inl(ALI_REG(codec, pchregs->regs.ainten)); + pchregs->data.ainten &= ~mask; + outl(pchregs->data.ainten, ALI_REG(codec, pchregs->regs.ainten)); +} + +static int snd_ali_alloc_pcm_channel(struct snd_ali *codec, int channel) +{ + unsigned int idx = channel & 0x1f; + + if (codec->synth.chcnt >= ALI_CHANNELS){ + snd_printk(KERN_ERR + "ali_alloc_pcm_channel: no free channels.\n"); + return -1; + } + + if (!(codec->synth.chmap & (1 << idx))) { + codec->synth.chmap |= 1 << idx; + codec->synth.chcnt++; + snd_ali_printk("alloc_pcm_channel no. %d.\n",idx); + return idx; + } + return -1; +} + +static int snd_ali_find_free_channel(struct snd_ali * codec, int rec) +{ + int idx; + int result = -1; + + snd_ali_printk("find_free_channel: for %s\n",rec ? "rec" : "pcm"); + + /* recording */ + if (rec) { + if (codec->spdif_support && + (inl(ALI_REG(codec, ALI_GLOBAL_CONTROL)) & + ALI_SPDIF_IN_SUPPORT)) + idx = ALI_SPDIF_IN_CHANNEL; + else + idx = ALI_PCM_IN_CHANNEL; + + result = snd_ali_alloc_pcm_channel(codec, idx); + if (result >= 0) + return result; + else { + snd_printk(KERN_ERR "ali_find_free_channel: " + "record channel is busy now.\n"); + return -1; + } + } + + /* playback... */ + if (codec->spdif_support && + (inl(ALI_REG(codec, ALI_GLOBAL_CONTROL)) & + ALI_SPDIF_OUT_CH_ENABLE)) { + idx = ALI_SPDIF_OUT_CHANNEL; + result = snd_ali_alloc_pcm_channel(codec, idx); + if (result >= 0) + return result; + else + snd_printk(KERN_ERR "ali_find_free_channel: " + "S/PDIF out channel is in busy now.\n"); + } + + for (idx = 0; idx < ALI_CHANNELS; idx++) { + result = snd_ali_alloc_pcm_channel(codec, idx); + if (result >= 0) + return result; + } + snd_printk(KERN_ERR "ali_find_free_channel: no free channels.\n"); + return -1; +} + +static void snd_ali_free_channel_pcm(struct snd_ali *codec, int channel) +{ + unsigned int idx = channel & 0x0000001f; + + snd_ali_printk("free_channel_pcm channel=%d\n",channel); + + if (channel < 0 || channel >= ALI_CHANNELS) + return; + + if (!(codec->synth.chmap & (1 << idx))) { + snd_printk(KERN_ERR "ali_free_channel_pcm: " + "channel %d is not in use.\n", channel); + return; + } else { + codec->synth.chmap &= ~(1 << idx); + codec->synth.chcnt--; + } +} + +#if 0 /* not used */ +static void snd_ali_start_voice(struct snd_ali *codec, unsigned int channel) +{ + unsigned int mask = 1 << (channel & 0x1f); + + snd_ali_printk("start_voice: channel=%d\n",channel); + outl(mask, ALI_REG(codec,codec->chregs.regs.start)); +} +#endif + +static void snd_ali_stop_voice(struct snd_ali *codec, unsigned int channel) +{ + unsigned int mask = 1 << (channel & 0x1f); + + snd_ali_printk("stop_voice: channel=%d\n",channel); + outl(mask, ALI_REG(codec, codec->chregs.regs.stop)); +} + +/* + * S/PDIF Part + */ + +static void snd_ali_delay(struct snd_ali *codec,int interval) +{ + unsigned long begintimer,currenttimer; + + begintimer = inl(ALI_REG(codec, ALI_STIMER)); + currenttimer = inl(ALI_REG(codec, ALI_STIMER)); + + while (currenttimer < begintimer + interval) { + if (snd_ali_stimer_ready(codec) < 0) + break; + currenttimer = inl(ALI_REG(codec, ALI_STIMER)); + cpu_relax(); + } +} + +static void snd_ali_detect_spdif_rate(struct snd_ali *codec) +{ + u16 wval; + u16 count = 0; + u8 bval, R1 = 0, R2; + + bval = inb(ALI_REG(codec, ALI_SPDIF_CTRL + 1)); + bval |= 0x1F; + outb(bval, ALI_REG(codec, ALI_SPDIF_CTRL + 1)); + + while ((R1 < 0x0b || R1 > 0x0e) && R1 != 0x12 && count <= 50000) { + count ++; + snd_ali_delay(codec, 6); + bval = inb(ALI_REG(codec, ALI_SPDIF_CTRL + 1)); + R1 = bval & 0x1F; + } + + if (count > 50000) { + snd_printk(KERN_ERR "ali_detect_spdif_rate: timeout!\n"); + return; + } + + for (count = 0; count <= 50000; count++) { + snd_ali_delay(codec, 6); + bval = inb(ALI_REG(codec,ALI_SPDIF_CTRL + 1)); + R2 = bval & 0x1F; + if (R2 != R1) + R1 = R2; + else + break; + } + + if (count > 50000) { + snd_printk(KERN_ERR "ali_detect_spdif_rate: timeout!\n"); + return; + } + + if (R2 >= 0x0b && R2 <= 0x0e) { + wval = inw(ALI_REG(codec, ALI_SPDIF_CTRL + 2)); + wval &= 0xe0f0; + wval |= (0x09 << 8) | 0x05; + outw(wval, ALI_REG(codec, ALI_SPDIF_CTRL + 2)); + + bval = inb(ALI_REG(codec, ALI_SPDIF_CS + 3)) & 0xf0; + outb(bval | 0x02, ALI_REG(codec, ALI_SPDIF_CS + 3)); + } else if (R2 == 0x12) { + wval = inw(ALI_REG(codec, ALI_SPDIF_CTRL + 2)); + wval &= 0xe0f0; + wval |= (0x0e << 8) | 0x08; + outw(wval, ALI_REG(codec, ALI_SPDIF_CTRL + 2)); + + bval = inb(ALI_REG(codec,ALI_SPDIF_CS + 3)) & 0xf0; + outb(bval | 0x03, ALI_REG(codec, ALI_SPDIF_CS + 3)); + } +} + +static unsigned int snd_ali_get_spdif_in_rate(struct snd_ali *codec) +{ + u32 dwRate; + u8 bval; + + bval = inb(ALI_REG(codec, ALI_SPDIF_CTRL)); + bval &= 0x7f; + bval |= 0x40; + outb(bval, ALI_REG(codec, ALI_SPDIF_CTRL)); + + snd_ali_detect_spdif_rate(codec); + + bval = inb(ALI_REG(codec, ALI_SPDIF_CS + 3)); + bval &= 0x0f; + + switch (bval) { + case 0: dwRate = 44100; break; + case 1: dwRate = 48000; break; + case 2: dwRate = 32000; break; + default: dwRate = 0; break; + } + + return dwRate; +} + +static void snd_ali_enable_spdif_in(struct snd_ali *codec) +{ + unsigned int dwVal; + + dwVal = inl(ALI_REG(codec, ALI_GLOBAL_CONTROL)); + dwVal |= ALI_SPDIF_IN_SUPPORT; + outl(dwVal, ALI_REG(codec, ALI_GLOBAL_CONTROL)); + + dwVal = inb(ALI_REG(codec, ALI_SPDIF_CTRL)); + dwVal |= 0x02; + outb(dwVal, ALI_REG(codec, ALI_SPDIF_CTRL)); + + snd_ali_enable_special_channel(codec, ALI_SPDIF_IN_CHANNEL); +} + +static void snd_ali_disable_spdif_in(struct snd_ali *codec) +{ + unsigned int dwVal; + + dwVal = inl(ALI_REG(codec, ALI_GLOBAL_CONTROL)); + dwVal &= ~ALI_SPDIF_IN_SUPPORT; + outl(dwVal, ALI_REG(codec, ALI_GLOBAL_CONTROL)); + + snd_ali_disable_special_channel(codec, ALI_SPDIF_IN_CHANNEL); +} + + +static void snd_ali_set_spdif_out_rate(struct snd_ali *codec, unsigned int rate) +{ + unsigned char bVal; + unsigned int dwRate; + + switch (rate) { + case 32000: dwRate = 0x300; break; + case 48000: dwRate = 0x200; break; + default: dwRate = 0; break; + } + + bVal = inb(ALI_REG(codec, ALI_SPDIF_CTRL)); + bVal &= (unsigned char)(~(1<<6)); + + bVal |= 0x80; /* select right */ + outb(bVal, ALI_REG(codec, ALI_SPDIF_CTRL)); + outb(dwRate | 0x20, ALI_REG(codec, ALI_SPDIF_CS + 2)); + + bVal &= ~0x80; /* select left */ + outb(bVal, ALI_REG(codec, ALI_SPDIF_CTRL)); + outw(rate | 0x10, ALI_REG(codec, ALI_SPDIF_CS + 2)); +} + +static void snd_ali_enable_spdif_out(struct snd_ali *codec) +{ + unsigned short wVal; + unsigned char bVal; + struct pci_dev *pci_dev; + + pci_dev = codec->pci_m1533; + if (pci_dev == NULL) + return; + pci_read_config_byte(pci_dev, 0x61, &bVal); + bVal |= 0x40; + pci_write_config_byte(pci_dev, 0x61, bVal); + pci_read_config_byte(pci_dev, 0x7d, &bVal); + bVal |= 0x01; + pci_write_config_byte(pci_dev, 0x7d, bVal); + + pci_read_config_byte(pci_dev, 0x7e, &bVal); + bVal &= (~0x20); + bVal |= 0x10; + pci_write_config_byte(pci_dev, 0x7e, bVal); + + bVal = inb(ALI_REG(codec, ALI_SCTRL)); + outb(bVal | ALI_SPDIF_OUT_ENABLE, ALI_REG(codec, ALI_SCTRL)); + + bVal = inb(ALI_REG(codec, ALI_SPDIF_CTRL)); + outb(bVal & ALI_SPDIF_OUT_CH_STATUS, ALI_REG(codec, ALI_SPDIF_CTRL)); + + wVal = inw(ALI_REG(codec, ALI_GLOBAL_CONTROL)); + wVal |= ALI_SPDIF_OUT_SEL_PCM; + outw(wVal, ALI_REG(codec, ALI_GLOBAL_CONTROL)); + snd_ali_disable_special_channel(codec, ALI_SPDIF_OUT_CHANNEL); +} + +static void snd_ali_enable_spdif_chnout(struct snd_ali *codec) +{ + unsigned short wVal; + + wVal = inw(ALI_REG(codec, ALI_GLOBAL_CONTROL)); + wVal &= ~ALI_SPDIF_OUT_SEL_PCM; + outw(wVal, ALI_REG(codec, ALI_GLOBAL_CONTROL)); +/* + wVal = inw(ALI_REG(codec, ALI_SPDIF_CS)); + if (flag & ALI_SPDIF_OUT_NON_PCM) + wVal |= 0x0002; + else + wVal &= (~0x0002); + outw(wVal, ALI_REG(codec, ALI_SPDIF_CS)); +*/ + snd_ali_enable_special_channel(codec, ALI_SPDIF_OUT_CHANNEL); +} + +static void snd_ali_disable_spdif_chnout(struct snd_ali *codec) +{ + unsigned short wVal; + + wVal = inw(ALI_REG(codec, ALI_GLOBAL_CONTROL)); + wVal |= ALI_SPDIF_OUT_SEL_PCM; + outw(wVal, ALI_REG(codec, ALI_GLOBAL_CONTROL)); + + snd_ali_enable_special_channel(codec, ALI_SPDIF_OUT_CHANNEL); +} + +static void snd_ali_disable_spdif_out(struct snd_ali *codec) +{ + unsigned char bVal; + + bVal = inb(ALI_REG(codec, ALI_SCTRL)); + outb(bVal & ~ALI_SPDIF_OUT_ENABLE, ALI_REG(codec, ALI_SCTRL)); + + snd_ali_disable_spdif_chnout(codec); +} + +static void snd_ali_update_ptr(struct snd_ali *codec, int channel) +{ + struct snd_ali_voice *pvoice; + struct snd_pcm_runtime *runtime; + struct snd_ali_channel_control *pchregs; + unsigned int old, mask; +#ifdef ALI_DEBUG + unsigned int temp, cspf; +#endif + + pchregs = &(codec->chregs); + + /* check if interrupt occurred for channel */ + old = pchregs->data.aint; + mask = 1U << (channel & 0x1f); + + if (!(old & mask)) + return; + + pvoice = &codec->synth.voices[channel]; + runtime = pvoice->substream->runtime; + + udelay(100); + spin_lock(&codec->reg_lock); + + if (pvoice->pcm && pvoice->substream) { + /* pcm interrupt */ +#ifdef ALI_DEBUG + outb((u8)(pvoice->number), ALI_REG(codec, ALI_GC_CIR)); + temp = inw(ALI_REG(codec, ALI_CSO_ALPHA_FMS + 2)); + cspf = (inl(ALI_REG(codec, ALI_CSPF)) & mask) == mask; +#endif + if (pvoice->running) { + snd_ali_printk("update_ptr: cso=%4.4x cspf=%d.\n", + (u16)temp, cspf); + spin_unlock(&codec->reg_lock); + snd_pcm_period_elapsed(pvoice->substream); + spin_lock(&codec->reg_lock); + } else { + snd_ali_stop_voice(codec, channel); + snd_ali_disable_voice_irq(codec, channel); + } + } else if (codec->synth.voices[channel].synth) { + /* synth interrupt */ + } else if (codec->synth.voices[channel].midi) { + /* midi interrupt */ + } else { + /* unknown interrupt */ + snd_ali_stop_voice(codec, channel); + snd_ali_disable_voice_irq(codec, channel); + } + spin_unlock(&codec->reg_lock); + outl(mask,ALI_REG(codec,pchregs->regs.aint)); + pchregs->data.aint = old & (~mask); +} + +static irqreturn_t snd_ali_card_interrupt(int irq, void *dev_id) +{ + struct snd_ali *codec = dev_id; + int channel; + unsigned int audio_int; + struct snd_ali_channel_control *pchregs; + + if (codec == NULL || !codec->hw_initialized) + return IRQ_NONE; + + audio_int = inl(ALI_REG(codec, ALI_MISCINT)); + if (!audio_int) + return IRQ_NONE; + + pchregs = &(codec->chregs); + if (audio_int & ADDRESS_IRQ) { + /* get interrupt status for all channels */ + pchregs->data.aint = inl(ALI_REG(codec, pchregs->regs.aint)); + for (channel = 0; channel < ALI_CHANNELS; channel++) + snd_ali_update_ptr(codec, channel); + } + outl((TARGET_REACHED | MIXER_OVERFLOW | MIXER_UNDERFLOW), + ALI_REG(codec, ALI_MISCINT)); + + return IRQ_HANDLED; +} + + +static struct snd_ali_voice *snd_ali_alloc_voice(struct snd_ali * codec, + int type, int rec, int channel) +{ + struct snd_ali_voice *pvoice; + int idx; + + snd_ali_printk("alloc_voice: type=%d rec=%d\n", type, rec); + + spin_lock_irq(&codec->voice_alloc); + if (type == SNDRV_ALI_VOICE_TYPE_PCM) { + idx = channel > 0 ? snd_ali_alloc_pcm_channel(codec, channel) : + snd_ali_find_free_channel(codec,rec); + if (idx < 0) { + snd_printk(KERN_ERR "ali_alloc_voice: err.\n"); + spin_unlock_irq(&codec->voice_alloc); + return NULL; + } + pvoice = &(codec->synth.voices[idx]); + pvoice->codec = codec; + pvoice->use = 1; + pvoice->pcm = 1; + pvoice->mode = rec; + spin_unlock_irq(&codec->voice_alloc); + return pvoice; + } + spin_unlock_irq(&codec->voice_alloc); + return NULL; +} + + +static void snd_ali_free_voice(struct snd_ali * codec, + struct snd_ali_voice *pvoice) +{ + void (*private_free)(void *); + void *private_data; + + snd_ali_printk("free_voice: channel=%d\n",pvoice->number); + if (pvoice == NULL || !pvoice->use) + return; + snd_ali_clear_voices(codec, pvoice->number, pvoice->number); + spin_lock_irq(&codec->voice_alloc); + private_free = pvoice->private_free; + private_data = pvoice->private_data; + pvoice->private_free = NULL; + pvoice->private_data = NULL; + if (pvoice->pcm) + snd_ali_free_channel_pcm(codec, pvoice->number); + pvoice->use = pvoice->pcm = pvoice->synth = 0; + pvoice->substream = NULL; + spin_unlock_irq(&codec->voice_alloc); + if (private_free) + private_free(private_data); +} + + +static void snd_ali_clear_voices(struct snd_ali *codec, + unsigned int v_min, + unsigned int v_max) +{ + unsigned int i; + + for (i = v_min; i <= v_max; i++) { + snd_ali_stop_voice(codec, i); + snd_ali_disable_voice_irq(codec, i); + } +} + +static void snd_ali_write_voice_regs(struct snd_ali *codec, + unsigned int Channel, + unsigned int LBA, + unsigned int CSO, + unsigned int ESO, + unsigned int DELTA, + unsigned int ALPHA_FMS, + unsigned int GVSEL, + unsigned int PAN, + unsigned int VOL, + unsigned int CTRL, + unsigned int EC) +{ + unsigned int ctlcmds[4]; + + outb((unsigned char)(Channel & 0x001f), ALI_REG(codec, ALI_GC_CIR)); + + ctlcmds[0] = (CSO << 16) | (ALPHA_FMS & 0x0000ffff); + ctlcmds[1] = LBA; + ctlcmds[2] = (ESO << 16) | (DELTA & 0x0ffff); + ctlcmds[3] = (GVSEL << 31) | + ((PAN & 0x0000007f) << 24) | + ((VOL & 0x000000ff) << 16) | + ((CTRL & 0x0000000f) << 12) | + (EC & 0x00000fff); + + outb(Channel, ALI_REG(codec, ALI_GC_CIR)); + + outl(ctlcmds[0], ALI_REG(codec, ALI_CSO_ALPHA_FMS)); + outl(ctlcmds[1], ALI_REG(codec, ALI_LBA)); + outl(ctlcmds[2], ALI_REG(codec, ALI_ESO_DELTA)); + outl(ctlcmds[3], ALI_REG(codec, ALI_GVSEL_PAN_VOC_CTRL_EC)); + + outl(0x30000000, ALI_REG(codec, ALI_EBUF1)); /* Still Mode */ + outl(0x30000000, ALI_REG(codec, ALI_EBUF2)); /* Still Mode */ +} + +static unsigned int snd_ali_convert_rate(unsigned int rate, int rec) +{ + unsigned int delta; + + if (rate < 4000) + rate = 4000; + if (rate > 48000) + rate = 48000; + + if (rec) { + if (rate == 44100) + delta = 0x116a; + else if (rate == 8000) + delta = 0x6000; + else if (rate == 48000) + delta = 0x1000; + else + delta = ((48000 << 12) / rate) & 0x0000ffff; + } else { + if (rate == 44100) + delta = 0xeb3; + else if (rate == 8000) + delta = 0x2ab; + else if (rate == 48000) + delta = 0x1000; + else + delta = (((rate << 12) + rate) / 48000) & 0x0000ffff; + } + + return delta; +} + +static unsigned int snd_ali_control_mode(struct snd_pcm_substream *substream) +{ + unsigned int CTRL; + struct snd_pcm_runtime *runtime = substream->runtime; + + /* set ctrl mode + CTRL default: 8-bit (unsigned) mono, loop mode enabled + */ + CTRL = 0x00000001; + if (snd_pcm_format_width(runtime->format) == 16) + CTRL |= 0x00000008; /* 16-bit data */ + if (!snd_pcm_format_unsigned(runtime->format)) + CTRL |= 0x00000002; /* signed data */ + if (runtime->channels > 1) + CTRL |= 0x00000004; /* stereo data */ + return CTRL; +} + +/* + * PCM part + */ + +static int snd_ali_trigger(struct snd_pcm_substream *substream, + int cmd) + +{ + struct snd_ali *codec = snd_pcm_substream_chip(substream); + struct snd_pcm_substream *s; + unsigned int what, whati, capture_flag; + struct snd_ali_voice *pvoice, *evoice; + unsigned int val; + int do_start; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + do_start = 1; + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + do_start = 0; + break; + default: + return -EINVAL; + } + + what = whati = capture_flag = 0; + snd_pcm_group_for_each_entry(s, substream) { + if ((struct snd_ali *) snd_pcm_substream_chip(s) == codec) { + pvoice = s->runtime->private_data; + evoice = pvoice->extra; + what |= 1 << (pvoice->number & 0x1f); + if (evoice == NULL) + whati |= 1 << (pvoice->number & 0x1f); + else { + whati |= 1 << (evoice->number & 0x1f); + what |= 1 << (evoice->number & 0x1f); + } + if (do_start) { + pvoice->running = 1; + if (evoice != NULL) + evoice->running = 1; + } else { + pvoice->running = 0; + if (evoice != NULL) + evoice->running = 0; + } + snd_pcm_trigger_done(s, substream); + if (pvoice->mode) + capture_flag = 1; + } + } + spin_lock(&codec->reg_lock); + if (!do_start) + outl(what, ALI_REG(codec, ALI_STOP)); + val = inl(ALI_REG(codec, ALI_AINTEN)); + if (do_start) + val |= whati; + else + val &= ~whati; + outl(val, ALI_REG(codec, ALI_AINTEN)); + if (do_start) + outl(what, ALI_REG(codec, ALI_START)); + snd_ali_printk("trigger: what=%xh whati=%xh\n", what, whati); + spin_unlock(&codec->reg_lock); + + return 0; +} + +static int snd_ali_playback_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_ali *codec = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_ali_voice *pvoice = runtime->private_data; + struct snd_ali_voice *evoice = pvoice->extra; + int err; + + err = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + if (err < 0) + return err; + + /* voice management */ + + if (params_buffer_size(hw_params) / 2 != + params_period_size(hw_params)) { + if (!evoice) { + evoice = snd_ali_alloc_voice(codec, + SNDRV_ALI_VOICE_TYPE_PCM, + 0, -1); + if (!evoice) + return -ENOMEM; + pvoice->extra = evoice; + evoice->substream = substream; + } + } else { + if (evoice) { + snd_ali_free_voice(codec, evoice); + pvoice->extra = evoice = NULL; + } + } + + return 0; +} + +static int snd_ali_playback_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_ali *codec = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_ali_voice *pvoice = runtime->private_data; + struct snd_ali_voice *evoice = pvoice ? pvoice->extra : NULL; + + snd_pcm_lib_free_pages(substream); + if (evoice) { + snd_ali_free_voice(codec, evoice); + pvoice->extra = NULL; + } + return 0; +} + +static int snd_ali_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); +} + +static int snd_ali_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static int snd_ali_playback_prepare(struct snd_pcm_substream *substream) +{ + struct snd_ali *codec = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_ali_voice *pvoice = runtime->private_data; + struct snd_ali_voice *evoice = pvoice->extra; + + unsigned int LBA; + unsigned int Delta; + unsigned int ESO; + unsigned int CTRL; + unsigned int GVSEL; + unsigned int PAN; + unsigned int VOL; + unsigned int EC; + + snd_ali_printk("playback_prepare ...\n"); + + spin_lock_irq(&codec->reg_lock); + + /* set Delta (rate) value */ + Delta = snd_ali_convert_rate(runtime->rate, 0); + + if (pvoice->number == ALI_SPDIF_IN_CHANNEL || + pvoice->number == ALI_PCM_IN_CHANNEL) + snd_ali_disable_special_channel(codec, pvoice->number); + else if (codec->spdif_support && + (inl(ALI_REG(codec, ALI_GLOBAL_CONTROL)) & + ALI_SPDIF_OUT_CH_ENABLE) + && pvoice->number == ALI_SPDIF_OUT_CHANNEL) { + snd_ali_set_spdif_out_rate(codec, runtime->rate); + Delta = 0x1000; + } + + /* set Loop Back Address */ + LBA = runtime->dma_addr; + + /* set interrupt count size */ + pvoice->count = runtime->period_size; + + /* set target ESO for channel */ + pvoice->eso = runtime->buffer_size; + + snd_ali_printk("playback_prepare: eso=%xh count=%xh\n", + pvoice->eso, pvoice->count); + + /* set ESO to capture first MIDLP interrupt */ + ESO = pvoice->eso -1; + /* set ctrl mode */ + CTRL = snd_ali_control_mode(substream); + + GVSEL = 1; + PAN = 0; + VOL = 0; + EC = 0; + snd_ali_printk("playback_prepare:\n"); + snd_ali_printk("ch=%d, Rate=%d Delta=%xh,GVSEL=%xh,PAN=%xh,CTRL=%xh\n", + pvoice->number,runtime->rate,Delta,GVSEL,PAN,CTRL); + snd_ali_write_voice_regs(codec, + pvoice->number, + LBA, + 0, /* cso */ + ESO, + Delta, + 0, /* alpha */ + GVSEL, + PAN, + VOL, + CTRL, + EC); + if (evoice) { + evoice->count = pvoice->count; + evoice->eso = pvoice->count << 1; + ESO = evoice->eso - 1; + snd_ali_write_voice_regs(codec, + evoice->number, + LBA, + 0, /* cso */ + ESO, + Delta, + 0, /* alpha */ + GVSEL, + 0x7f, + 0x3ff, + CTRL, + EC); + } + spin_unlock_irq(&codec->reg_lock); + return 0; +} + + +static int snd_ali_prepare(struct snd_pcm_substream *substream) +{ + struct snd_ali *codec = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_ali_voice *pvoice = runtime->private_data; + unsigned int LBA; + unsigned int Delta; + unsigned int ESO; + unsigned int CTRL; + unsigned int GVSEL; + unsigned int PAN; + unsigned int VOL; + unsigned int EC; + u8 bValue; + + spin_lock_irq(&codec->reg_lock); + + snd_ali_printk("ali_prepare...\n"); + + snd_ali_enable_special_channel(codec,pvoice->number); + + Delta = (pvoice->number == ALI_MODEM_IN_CHANNEL || + pvoice->number == ALI_MODEM_OUT_CHANNEL) ? + 0x1000 : snd_ali_convert_rate(runtime->rate, pvoice->mode); + + /* Prepare capture intr channel */ + if (pvoice->number == ALI_SPDIF_IN_CHANNEL) { + + unsigned int rate; + + spin_unlock_irq(&codec->reg_lock); + if (codec->revision != ALI_5451_V02) + return -1; + + rate = snd_ali_get_spdif_in_rate(codec); + if (rate == 0) { + snd_printk(KERN_WARNING "ali_capture_preapre: " + "spdif rate detect err!\n"); + rate = 48000; + } + spin_lock_irq(&codec->reg_lock); + bValue = inb(ALI_REG(codec,ALI_SPDIF_CTRL)); + if (bValue & 0x10) { + outb(bValue,ALI_REG(codec,ALI_SPDIF_CTRL)); + printk(KERN_WARNING "clear SPDIF parity error flag.\n"); + } + + if (rate != 48000) + Delta = ((rate << 12) / runtime->rate) & 0x00ffff; + } + + /* set target ESO for channel */ + pvoice->eso = runtime->buffer_size; + + /* set interrupt count size */ + pvoice->count = runtime->period_size; + + /* set Loop Back Address */ + LBA = runtime->dma_addr; + + /* set ESO to capture first MIDLP interrupt */ + ESO = pvoice->eso - 1; + CTRL = snd_ali_control_mode(substream); + GVSEL = 0; + PAN = 0x00; + VOL = 0x00; + EC = 0; + + snd_ali_write_voice_regs( codec, + pvoice->number, + LBA, + 0, /* cso */ + ESO, + Delta, + 0, /* alpha */ + GVSEL, + PAN, + VOL, + CTRL, + EC); + + spin_unlock_irq(&codec->reg_lock); + + return 0; +} + + +static snd_pcm_uframes_t +snd_ali_playback_pointer(struct snd_pcm_substream *substream) +{ + struct snd_ali *codec = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_ali_voice *pvoice = runtime->private_data; + unsigned int cso; + + spin_lock(&codec->reg_lock); + if (!pvoice->running) { + spin_unlock(&codec->reg_lock); + return 0; + } + outb(pvoice->number, ALI_REG(codec, ALI_GC_CIR)); + cso = inw(ALI_REG(codec, ALI_CSO_ALPHA_FMS + 2)); + spin_unlock(&codec->reg_lock); + snd_ali_printk("playback pointer returned cso=%xh.\n", cso); + + return cso; +} + + +static snd_pcm_uframes_t snd_ali_pointer(struct snd_pcm_substream *substream) +{ + struct snd_ali *codec = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_ali_voice *pvoice = runtime->private_data; + unsigned int cso; + + spin_lock(&codec->reg_lock); + if (!pvoice->running) { + spin_unlock_irq(&codec->reg_lock); + return 0; + } + outb(pvoice->number, ALI_REG(codec, ALI_GC_CIR)); + cso = inw(ALI_REG(codec, ALI_CSO_ALPHA_FMS + 2)); + spin_unlock(&codec->reg_lock); + + return cso; +} + +static struct snd_pcm_hardware snd_ali_playback = +{ + .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_SYNC_START), + .formats = (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U16_LE), + .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000, + .rate_min = 4000, + .rate_max = 48000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = (256*1024), + .period_bytes_min = 64, + .period_bytes_max = (256*1024), + .periods_min = 1, + .periods_max = 1024, + .fifo_size = 0, +}; + +/* + * Capture support device description + */ + +static struct snd_pcm_hardware snd_ali_capture = +{ + .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_SYNC_START), + .formats = (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U16_LE), + .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000, + .rate_min = 4000, + .rate_max = 48000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = (128*1024), + .period_bytes_min = 64, + .period_bytes_max = (128*1024), + .periods_min = 1, + .periods_max = 1024, + .fifo_size = 0, +}; + +static void snd_ali_pcm_free_substream(struct snd_pcm_runtime *runtime) +{ + struct snd_ali_voice *pvoice = runtime->private_data; + struct snd_ali *codec; + + if (pvoice) { + codec = pvoice->codec; + snd_ali_free_voice(pvoice->codec, pvoice); + } +} + +static int snd_ali_open(struct snd_pcm_substream *substream, int rec, + int channel, struct snd_pcm_hardware *phw) +{ + struct snd_ali *codec = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_ali_voice *pvoice; + + pvoice = snd_ali_alloc_voice(codec, SNDRV_ALI_VOICE_TYPE_PCM, rec, + channel); + if (!pvoice) + return -EAGAIN; + + pvoice->substream = substream; + runtime->private_data = pvoice; + runtime->private_free = snd_ali_pcm_free_substream; + + runtime->hw = *phw; + snd_pcm_set_sync(substream); + snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + 0, 64*1024); + return 0; +} + +static int snd_ali_playback_open(struct snd_pcm_substream *substream) +{ + return snd_ali_open(substream, 0, -1, &snd_ali_playback); +} + +static int snd_ali_capture_open(struct snd_pcm_substream *substream) +{ + return snd_ali_open(substream, 1, -1, &snd_ali_capture); +} + +static int snd_ali_playback_close(struct snd_pcm_substream *substream) +{ + return 0; +} + +static int snd_ali_close(struct snd_pcm_substream *substream) +{ + struct snd_ali *codec = snd_pcm_substream_chip(substream); + struct snd_ali_voice *pvoice = substream->runtime->private_data; + + snd_ali_disable_special_channel(codec,pvoice->number); + + return 0; +} + +static struct snd_pcm_ops snd_ali_playback_ops = { + .open = snd_ali_playback_open, + .close = snd_ali_playback_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_ali_playback_hw_params, + .hw_free = snd_ali_playback_hw_free, + .prepare = snd_ali_playback_prepare, + .trigger = snd_ali_trigger, + .pointer = snd_ali_playback_pointer, +}; + +static struct snd_pcm_ops snd_ali_capture_ops = { + .open = snd_ali_capture_open, + .close = snd_ali_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_ali_hw_params, + .hw_free = snd_ali_hw_free, + .prepare = snd_ali_prepare, + .trigger = snd_ali_trigger, + .pointer = snd_ali_pointer, +}; + +/* + * Modem PCM + */ + +static int snd_ali_modem_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_ali *chip = snd_pcm_substream_chip(substream); + unsigned int modem_num = chip->num_of_codecs - 1; + snd_ac97_write(chip->ac97[modem_num], AC97_LINE1_RATE, + params_rate(hw_params)); + snd_ac97_write(chip->ac97[modem_num], AC97_LINE1_LEVEL, 0); + return snd_ali_hw_params(substream, hw_params); +} + +static struct snd_pcm_hardware snd_ali_modem = +{ + .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_SYNC_START), + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rates = (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_8000 | + SNDRV_PCM_RATE_16000), + .rate_min = 8000, + .rate_max = 16000, + .channels_min = 1, + .channels_max = 1, + .buffer_bytes_max = (256*1024), + .period_bytes_min = 64, + .period_bytes_max = (256*1024), + .periods_min = 1, + .periods_max = 1024, + .fifo_size = 0, +}; + +static int snd_ali_modem_open(struct snd_pcm_substream *substream, int rec, + int channel) +{ + static unsigned int rates[] = {8000, 9600, 12000, 16000}; + static struct snd_pcm_hw_constraint_list hw_constraint_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, + }; + int err = snd_ali_open(substream, rec, channel, &snd_ali_modem); + + if (err) + return err; + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &hw_constraint_rates); +} + +static int snd_ali_modem_playback_open(struct snd_pcm_substream *substream) +{ + return snd_ali_modem_open(substream, 0, ALI_MODEM_OUT_CHANNEL); +} + +static int snd_ali_modem_capture_open(struct snd_pcm_substream *substream) +{ + return snd_ali_modem_open(substream, 1, ALI_MODEM_IN_CHANNEL); +} + +static struct snd_pcm_ops snd_ali_modem_playback_ops = { + .open = snd_ali_modem_playback_open, + .close = snd_ali_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_ali_modem_hw_params, + .hw_free = snd_ali_hw_free, + .prepare = snd_ali_prepare, + .trigger = snd_ali_trigger, + .pointer = snd_ali_pointer, +}; + +static struct snd_pcm_ops snd_ali_modem_capture_ops = { + .open = snd_ali_modem_capture_open, + .close = snd_ali_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_ali_modem_hw_params, + .hw_free = snd_ali_hw_free, + .prepare = snd_ali_prepare, + .trigger = snd_ali_trigger, + .pointer = snd_ali_pointer, +}; + + +struct ali_pcm_description { + char *name; + unsigned int playback_num; + unsigned int capture_num; + struct snd_pcm_ops *playback_ops; + struct snd_pcm_ops *capture_ops; + unsigned short class; +}; + + +static void snd_ali_pcm_free(struct snd_pcm *pcm) +{ + struct snd_ali *codec = pcm->private_data; + codec->pcm[pcm->device] = NULL; +} + + +static int __devinit snd_ali_pcm(struct snd_ali * codec, int device, + struct ali_pcm_description *desc) +{ + struct snd_pcm *pcm; + int err; + + err = snd_pcm_new(codec->card, desc->name, device, + desc->playback_num, desc->capture_num, &pcm); + if (err < 0) { + snd_printk(KERN_ERR "snd_ali_pcm: err called snd_pcm_new.\n"); + return err; + } + pcm->private_data = codec; + pcm->private_free = snd_ali_pcm_free; + if (desc->playback_ops) + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + desc->playback_ops); + if (desc->capture_ops) + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + desc->capture_ops); + + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(codec->pci), + 64*1024, 128*1024); + + pcm->info_flags = 0; + pcm->dev_class = desc->class; + pcm->dev_subclass = SNDRV_PCM_SUBCLASS_GENERIC_MIX; + strcpy(pcm->name, desc->name); + codec->pcm[0] = pcm; + return 0; +} + +static struct ali_pcm_description ali_pcms[] = { + { .name = "ALI 5451", + .playback_num = ALI_CHANNELS, + .capture_num = 1, + .playback_ops = &snd_ali_playback_ops, + .capture_ops = &snd_ali_capture_ops + }, + { .name = "ALI 5451 modem", + .playback_num = 1, + .capture_num = 1, + .playback_ops = &snd_ali_modem_playback_ops, + .capture_ops = &snd_ali_modem_capture_ops, + .class = SNDRV_PCM_CLASS_MODEM + } +}; + +static int __devinit snd_ali_build_pcms(struct snd_ali *codec) +{ + int i, err; + for (i = 0; i < codec->num_of_codecs && i < ARRAY_SIZE(ali_pcms); i++) { + err = snd_ali_pcm(codec, i, &ali_pcms[i]); + if (err < 0) + return err; + } + return 0; +} + + +#define ALI5451_SPDIF(xname, xindex, value) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex,\ +.info = snd_ali5451_spdif_info, .get = snd_ali5451_spdif_get, \ +.put = snd_ali5451_spdif_put, .private_value = value} + +#define snd_ali5451_spdif_info snd_ctl_boolean_mono_info + +static int snd_ali5451_spdif_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ali *codec = kcontrol->private_data; + unsigned int enable; + + enable = ucontrol->value.integer.value[0] ? 1 : 0; + + spin_lock_irq(&codec->reg_lock); + switch (kcontrol->private_value) { + case 0: + enable = (codec->spdif_mask & 0x02) ? 1 : 0; + break; + case 1: + enable = ((codec->spdif_mask & 0x02) && + (codec->spdif_mask & 0x04)) ? 1 : 0; + break; + case 2: + enable = (codec->spdif_mask & 0x01) ? 1 : 0; + break; + default: + break; + } + ucontrol->value.integer.value[0] = enable; + spin_unlock_irq(&codec->reg_lock); + return 0; +} + +static int snd_ali5451_spdif_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ali *codec = kcontrol->private_data; + unsigned int change = 0, enable = 0; + + enable = ucontrol->value.integer.value[0] ? 1 : 0; + + spin_lock_irq(&codec->reg_lock); + switch (kcontrol->private_value) { + case 0: + change = (codec->spdif_mask & 0x02) ? 1 : 0; + change = change ^ enable; + if (change) { + if (enable) { + codec->spdif_mask |= 0x02; + snd_ali_enable_spdif_out(codec); + } else { + codec->spdif_mask &= ~(0x02); + codec->spdif_mask &= ~(0x04); + snd_ali_disable_spdif_out(codec); + } + } + break; + case 1: + change = (codec->spdif_mask & 0x04) ? 1 : 0; + change = change ^ enable; + if (change && (codec->spdif_mask & 0x02)) { + if (enable) { + codec->spdif_mask |= 0x04; + snd_ali_enable_spdif_chnout(codec); + } else { + codec->spdif_mask &= ~(0x04); + snd_ali_disable_spdif_chnout(codec); + } + } + break; + case 2: + change = (codec->spdif_mask & 0x01) ? 1 : 0; + change = change ^ enable; + if (change) { + if (enable) { + codec->spdif_mask |= 0x01; + snd_ali_enable_spdif_in(codec); + } else { + codec->spdif_mask &= ~(0x01); + snd_ali_disable_spdif_in(codec); + } + } + break; + default: + break; + } + spin_unlock_irq(&codec->reg_lock); + + return change; +} + +static struct snd_kcontrol_new snd_ali5451_mixer_spdif[] __devinitdata = { + /* spdif aplayback switch */ + /* FIXME: "IEC958 Playback Switch" may conflict with one on ac97_codec */ + ALI5451_SPDIF(SNDRV_CTL_NAME_IEC958("Output ",NONE,SWITCH), 0, 0), + /* spdif out to spdif channel */ + ALI5451_SPDIF(SNDRV_CTL_NAME_IEC958("Channel Output ",NONE,SWITCH), 0, 1), + /* spdif in from spdif channel */ + ALI5451_SPDIF(SNDRV_CTL_NAME_IEC958("",CAPTURE,SWITCH), 0, 2) +}; + +static int __devinit snd_ali_mixer(struct snd_ali * codec) +{ + struct snd_ac97_template ac97; + unsigned int idx; + int i, err; + static struct snd_ac97_bus_ops ops = { + .write = snd_ali_codec_write, + .read = snd_ali_codec_read, + }; + + err = snd_ac97_bus(codec->card, 0, &ops, codec, &codec->ac97_bus); + if (err < 0) + return err; + + memset(&ac97, 0, sizeof(ac97)); + ac97.private_data = codec; + + for (i = 0; i < codec->num_of_codecs; i++) { + ac97.num = i; + err = snd_ac97_mixer(codec->ac97_bus, &ac97, &codec->ac97[i]); + if (err < 0) { + snd_printk(KERN_ERR + "ali mixer %d creating error.\n", i); + if (i == 0) + return err; + codec->num_of_codecs = 1; + break; + } + } + + if (codec->spdif_support) { + for (idx = 0; idx < ARRAY_SIZE(snd_ali5451_mixer_spdif); idx++) { + err = snd_ctl_add(codec->card, + snd_ctl_new1(&snd_ali5451_mixer_spdif[idx], codec)); + if (err < 0) + return err; + } + } + return 0; +} + +#ifdef CONFIG_PM +static int ali_suspend(struct pci_dev *pci, pm_message_t state) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct snd_ali *chip = card->private_data; + struct snd_ali_image *im; + int i, j; + + im = chip->image; + if (!im) + return 0; + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + for (i = 0; i < chip->num_of_codecs; i++) { + snd_pcm_suspend_all(chip->pcm[i]); + snd_ac97_suspend(chip->ac97[i]); + } + + spin_lock_irq(&chip->reg_lock); + + im->regs[ALI_MISCINT >> 2] = inl(ALI_REG(chip, ALI_MISCINT)); + /* im->regs[ALI_START >> 2] = inl(ALI_REG(chip, ALI_START)); */ + im->regs[ALI_STOP >> 2] = inl(ALI_REG(chip, ALI_STOP)); + + /* disable all IRQ bits */ + outl(0, ALI_REG(chip, ALI_MISCINT)); + + for (i = 0; i < ALI_GLOBAL_REGS; i++) { + if ((i*4 == ALI_MISCINT) || (i*4 == ALI_STOP)) + continue; + im->regs[i] = inl(ALI_REG(chip, i*4)); + } + + for (i = 0; i < ALI_CHANNELS; i++) { + outb(i, ALI_REG(chip, ALI_GC_CIR)); + for (j = 0; j < ALI_CHANNEL_REGS; j++) + im->channel_regs[i][j] = inl(ALI_REG(chip, j*4 + 0xe0)); + } + + /* stop all HW channel */ + outl(0xffffffff, ALI_REG(chip, ALI_STOP)); + + spin_unlock_irq(&chip->reg_lock); + + pci_disable_device(pci); + pci_save_state(pci); + pci_set_power_state(pci, pci_choose_state(pci, state)); + return 0; +} + +static int ali_resume(struct pci_dev *pci) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct snd_ali *chip = card->private_data; + struct snd_ali_image *im; + int i, j; + + im = chip->image; + if (!im) + return 0; + + pci_set_power_state(pci, PCI_D0); + pci_restore_state(pci); + if (pci_enable_device(pci) < 0) { + printk(KERN_ERR "ali5451: pci_enable_device failed, " + "disabling device\n"); + snd_card_disconnect(card); + return -EIO; + } + pci_set_master(pci); + + spin_lock_irq(&chip->reg_lock); + + for (i = 0; i < ALI_CHANNELS; i++) { + outb(i, ALI_REG(chip, ALI_GC_CIR)); + for (j = 0; j < ALI_CHANNEL_REGS; j++) + outl(im->channel_regs[i][j], ALI_REG(chip, j*4 + 0xe0)); + } + + for (i = 0; i < ALI_GLOBAL_REGS; i++) { + if ((i*4 == ALI_MISCINT) || (i*4 == ALI_STOP) || + (i*4 == ALI_START)) + continue; + outl(im->regs[i], ALI_REG(chip, i*4)); + } + + /* start HW channel */ + outl(im->regs[ALI_START >> 2], ALI_REG(chip, ALI_START)); + /* restore IRQ enable bits */ + outl(im->regs[ALI_MISCINT >> 2], ALI_REG(chip, ALI_MISCINT)); + + spin_unlock_irq(&chip->reg_lock); + + for (i = 0 ; i < chip->num_of_codecs; i++) + snd_ac97_resume(chip->ac97[i]); + + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + return 0; +} +#endif /* CONFIG_PM */ + +static int snd_ali_free(struct snd_ali * codec) +{ + if (codec->hw_initialized) + snd_ali_disable_address_interrupt(codec); + if (codec->irq >= 0) { + synchronize_irq(codec->irq); + free_irq(codec->irq, codec); + } + if (codec->port) + pci_release_regions(codec->pci); + pci_disable_device(codec->pci); +#ifdef CONFIG_PM + kfree(codec->image); +#endif + pci_dev_put(codec->pci_m1533); + pci_dev_put(codec->pci_m7101); + kfree(codec); + return 0; +} + +static int snd_ali_chip_init(struct snd_ali *codec) +{ + unsigned int legacy; + unsigned char temp; + struct pci_dev *pci_dev; + + snd_ali_printk("chip initializing ... \n"); + + if (snd_ali_reset_5451(codec)) { + snd_printk(KERN_ERR "ali_chip_init: reset 5451 error.\n"); + return -1; + } + + if (codec->revision == ALI_5451_V02) { + pci_dev = codec->pci_m1533; + pci_read_config_byte(pci_dev, 0x59, &temp); + temp |= 0x80; + pci_write_config_byte(pci_dev, 0x59, temp); + + pci_dev = codec->pci_m7101; + pci_read_config_byte(pci_dev, 0xb8, &temp); + temp |= 0x20; + pci_write_config_byte(pci_dev, 0xB8, temp); + } + + pci_read_config_dword(codec->pci, 0x44, &legacy); + legacy &= 0xff00ff00; + legacy |= 0x000800aa; + pci_write_config_dword(codec->pci, 0x44, legacy); + + outl(0x80000001, ALI_REG(codec, ALI_GLOBAL_CONTROL)); + outl(0x00000000, ALI_REG(codec, ALI_AINTEN)); + outl(0xffffffff, ALI_REG(codec, ALI_AINT)); + outl(0x00000000, ALI_REG(codec, ALI_VOLUME)); + outb(0x10, ALI_REG(codec, ALI_MPUR2)); + + codec->ac97_ext_id = snd_ali_codec_peek(codec, 0, AC97_EXTENDED_ID); + codec->ac97_ext_status = snd_ali_codec_peek(codec, 0, + AC97_EXTENDED_STATUS); + if (codec->spdif_support) { + snd_ali_enable_spdif_out(codec); + codec->spdif_mask = 0x00000002; + } + + codec->num_of_codecs = 1; + + /* secondary codec - modem */ + if (inl(ALI_REG(codec, ALI_SCTRL)) & ALI_SCTRL_CODEC2_READY) { + codec->num_of_codecs++; + outl(inl(ALI_REG(codec, ALI_SCTRL)) | + (ALI_SCTRL_LINE_IN2 | ALI_SCTRL_GPIO_IN2 | + ALI_SCTRL_LINE_OUT_EN), + ALI_REG(codec, ALI_SCTRL)); + } + + snd_ali_printk("chip initialize succeed.\n"); + return 0; + +} + +/* proc for register dump */ +static void snd_ali_proc_read(struct snd_info_entry *entry, + struct snd_info_buffer *buf) +{ + struct snd_ali *codec = entry->private_data; + int i; + for (i = 0; i < 256 ; i+= 4) + snd_iprintf(buf, "%02x: %08x\n", i, inl(ALI_REG(codec, i))); +} + +static void __devinit snd_ali_proc_init(struct snd_ali *codec) +{ + struct snd_info_entry *entry; + if (!snd_card_proc_new(codec->card, "ali5451", &entry)) + snd_info_set_text_ops(entry, codec, snd_ali_proc_read); +} + +static int __devinit snd_ali_resources(struct snd_ali *codec) +{ + int err; + + snd_ali_printk("resouces allocation ...\n"); + err = pci_request_regions(codec->pci, "ALI 5451"); + if (err < 0) + return err; + codec->port = pci_resource_start(codec->pci, 0); + + if (request_irq(codec->pci->irq, snd_ali_card_interrupt, + IRQF_SHARED, "ALI 5451", codec)) { + snd_printk(KERN_ERR "Unable to request irq.\n"); + return -EBUSY; + } + codec->irq = codec->pci->irq; + snd_ali_printk("resouces allocated.\n"); + return 0; +} +static int snd_ali_dev_free(struct snd_device *device) +{ + struct snd_ali *codec = device->device_data; + snd_ali_free(codec); + return 0; +} + +static int __devinit snd_ali_create(struct snd_card *card, + struct pci_dev *pci, + int pcm_streams, + int spdif_support, + struct snd_ali ** r_ali) +{ + struct snd_ali *codec; + int i, err; + unsigned short cmdw; + static struct snd_device_ops ops = { + .dev_free = snd_ali_dev_free, + }; + + *r_ali = NULL; + + snd_ali_printk("creating ...\n"); + + /* enable PCI device */ + err = pci_enable_device(pci); + if (err < 0) + return err; + /* check, if we can restrict PCI DMA transfers to 31 bits */ + if (pci_set_dma_mask(pci, DMA_31BIT_MASK) < 0 || + pci_set_consistent_dma_mask(pci, DMA_31BIT_MASK) < 0) { + snd_printk(KERN_ERR "architecture does not support " + "31bit PCI busmaster DMA\n"); + pci_disable_device(pci); + return -ENXIO; + } + + codec = kzalloc(sizeof(*codec), GFP_KERNEL); + if (!codec) { + pci_disable_device(pci); + return -ENOMEM; + } + + spin_lock_init(&codec->reg_lock); + spin_lock_init(&codec->voice_alloc); + + codec->card = card; + codec->pci = pci; + codec->irq = -1; + codec->revision = pci->revision; + codec->spdif_support = spdif_support; + + if (pcm_streams < 1) + pcm_streams = 1; + if (pcm_streams > 32) + pcm_streams = 32; + + pci_set_master(pci); + pci_read_config_word(pci, PCI_COMMAND, &cmdw); + if ((cmdw & PCI_COMMAND_IO) != PCI_COMMAND_IO) { + cmdw |= PCI_COMMAND_IO; + pci_write_config_word(pci, PCI_COMMAND, cmdw); + } + pci_set_master(pci); + + if (snd_ali_resources(codec)) { + snd_ali_free(codec); + return -EBUSY; + } + + synchronize_irq(pci->irq); + + codec->synth.chmap = 0; + codec->synth.chcnt = 0; + codec->spdif_mask = 0; + codec->synth.synthcount = 0; + + if (codec->revision == ALI_5451_V02) + codec->chregs.regs.ac97read = ALI_AC97_WRITE; + else + codec->chregs.regs.ac97read = ALI_AC97_READ; + codec->chregs.regs.ac97write = ALI_AC97_WRITE; + + codec->chregs.regs.start = ALI_START; + codec->chregs.regs.stop = ALI_STOP; + codec->chregs.regs.aint = ALI_AINT; + codec->chregs.regs.ainten = ALI_AINTEN; + + codec->chregs.data.start = 0x00; + codec->chregs.data.stop = 0x00; + codec->chregs.data.aint = 0x00; + codec->chregs.data.ainten = 0x00; + + /* M1533: southbridge */ + codec->pci_m1533 = pci_get_device(0x10b9, 0x1533, NULL); + if (!codec->pci_m1533) { + snd_printk(KERN_ERR "ali5451: cannot find ALi 1533 chip.\n"); + snd_ali_free(codec); + return -ENODEV; + } + /* M7101: power management */ + codec->pci_m7101 = pci_get_device(0x10b9, 0x7101, NULL); + if (!codec->pci_m7101 && codec->revision == ALI_5451_V02) { + snd_printk(KERN_ERR "ali5451: cannot find ALi 7101 chip.\n"); + snd_ali_free(codec); + return -ENODEV; + } + + snd_ali_printk("snd_device_new is called.\n"); + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, codec, &ops); + if (err < 0) { + snd_ali_free(codec); + return err; + } + + snd_card_set_dev(card, &pci->dev); + + /* initialise synth voices*/ + for (i = 0; i < ALI_CHANNELS; i++) + codec->synth.voices[i].number = i; + + err = snd_ali_chip_init(codec); + if (err < 0) { + snd_printk(KERN_ERR "ali create: chip init error.\n"); + return err; + } + +#ifdef CONFIG_PM + codec->image = kmalloc(sizeof(*codec->image), GFP_KERNEL); + if (!codec->image) + snd_printk(KERN_WARNING "can't allocate apm buffer\n"); +#endif + + snd_ali_enable_address_interrupt(codec); + codec->hw_initialized = 1; + + *r_ali = codec; + snd_ali_printk("created.\n"); + return 0; +} + +static int __devinit snd_ali_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) +{ + struct snd_card *card; + struct snd_ali *codec; + int err; + + snd_ali_printk("probe ...\n"); + + card = snd_card_new(index, id, THIS_MODULE, 0); + if (!card) + return -ENOMEM; + + err = snd_ali_create(card, pci, pcm_channels, spdif, &codec); + if (err < 0) + goto error; + card->private_data = codec; + + snd_ali_printk("mixer building ...\n"); + err = snd_ali_mixer(codec); + if (err < 0) + goto error; + + snd_ali_printk("pcm building ...\n"); + err = snd_ali_build_pcms(codec); + if (err < 0) + goto error; + + snd_ali_proc_init(codec); + + strcpy(card->driver, "ALI5451"); + strcpy(card->shortname, "ALI 5451"); + + sprintf(card->longname, "%s at 0x%lx, irq %i", + card->shortname, codec->port, codec->irq); + + snd_ali_printk("register card.\n"); + err = snd_card_register(card); + if (err < 0) + goto error; + + pci_set_drvdata(pci, card); + return 0; + + error: + snd_card_free(card); + return err; +} + +static void __devexit snd_ali_remove(struct pci_dev *pci) +{ + snd_card_free(pci_get_drvdata(pci)); + pci_set_drvdata(pci, NULL); +} + +static struct pci_driver driver = { + .name = "ALI 5451", + .id_table = snd_ali_ids, + .probe = snd_ali_probe, + .remove = __devexit_p(snd_ali_remove), +#ifdef CONFIG_PM + .suspend = ali_suspend, + .resume = ali_resume, +#endif +}; + +static int __init alsa_card_ali_init(void) +{ + return pci_register_driver(&driver); +} + +static void __exit alsa_card_ali_exit(void) +{ + pci_unregister_driver(&driver); +} + +module_init(alsa_card_ali_init) +module_exit(alsa_card_ali_exit) diff -ruN linux-2.6.22/sound/pci/als300.c linux-2.6.22-alsa/sound/pci/als300.c --- linux-2.6.22/sound/pci/als300.c 2007-07-28 13:40:44.000000000 +0200 +++ linux-2.6.22-alsa/sound/pci/als300.c 2007-09-01 20:55:23.000000000 +0200 @@ -88,8 +88,8 @@ #define PLAYBACK_BLOCK_COUNTER 0x9A #define RECORD_BLOCK_COUNTER 0x9B -#define DEBUG_CALLS 1 -#define DEBUG_PLAY_REC 1 +#define DEBUG_CALLS 0 +#define DEBUG_PLAY_REC 0 #if DEBUG_CALLS #define snd_als300_dbgcalls(format, args...) printk(format, ##args) @@ -733,7 +733,8 @@ snd_als300_init(chip); - if (snd_als300_ac97(chip) < 0) { + err = snd_als300_ac97(chip); + if (err < 0) { snd_printk(KERN_WARNING "Could not create ac97\n"); snd_als300_free(chip); return err; diff -ruN linux-2.6.22/sound/pci/au88x0/au88x0.c linux-2.6.22-alsa/sound/pci/au88x0/au88x0.c --- linux-2.6.22/sound/pci/au88x0/au88x0.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/pci/au88x0/au88x0.c 2007-09-02 13:30:43.000000000 +0200 @@ -232,6 +232,7 @@ pci_disable_device(chip->pci_dev); //FIXME: this not the right place to unregister the gameport vortex_gameport_unregister(chip); + kfree(chip); return err; } diff -ruN linux-2.6.22/sound/pci/au88x0/au88x0.c.orig linux-2.6.22-alsa/sound/pci/au88x0/au88x0.c.orig --- linux-2.6.22/sound/pci/au88x0/au88x0.c.orig 1970-01-01 01:00:00.000000000 +0100 +++ linux-2.6.22-alsa/sound/pci/au88x0/au88x0.c.orig 2007-09-01 20:55:23.000000000 +0200 @@ -0,0 +1,400 @@ +/* + * ALSA driver for the Aureal Vortex family of soundprocessors. + * Author: Manuel Jander (mjander@embedded.cl) + * + * This driver is the result of the OpenVortex Project from Savannah + * (savannah.nongnu.org/projects/openvortex). I would like to thank + * the developers of OpenVortex, Jeff Muizelaar and Kester Maddock, from + * whom i got plenty of help, and their codebase was invaluable. + * Thanks to the ALSA developers, they helped a lot working out + * the ALSA part. + * Thanks also to Sourceforge for maintaining the old binary drivers, + * and the forum, where developers could comunicate. + * + * Now at least i can play Legacy DOOM with MIDI music :-) + */ + +#include "au88x0.h" +#include +#include +#include +#include +#include +#include +#include + +// module parameters (see "Module Parameters") +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; +static int pcifix[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 255 }; + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for " CARD_NAME " soundcard."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard."); +module_param_array(pcifix, int, NULL, 0444); +MODULE_PARM_DESC(pcifix, "Enable VIA-workaround for " CARD_NAME " soundcard."); + +MODULE_DESCRIPTION("Aureal vortex"); +MODULE_LICENSE("GPL"); +MODULE_SUPPORTED_DEVICE("{{Aureal Semiconductor Inc., Aureal Vortex Sound Processor}}"); + +MODULE_DEVICE_TABLE(pci, snd_vortex_ids); + +static void vortex_fix_latency(struct pci_dev *vortex) +{ + int rc; + if (!(rc = pci_write_config_byte(vortex, 0x40, 0xff))) { + printk(KERN_INFO CARD_NAME + ": vortex latency is 0xff\n"); + } else { + printk(KERN_WARNING CARD_NAME + ": could not set vortex latency: pci error 0x%x\n", rc); + } +} + +static void vortex_fix_agp_bridge(struct pci_dev *via) +{ + int rc; + u8 value; + + /* + * only set the bit (Extend PCI#2 Internal Master for + * Efficient Handling of Dummy Requests) if the can + * read the config and it is not already set + */ + + if (!(rc = pci_read_config_byte(via, 0x42, &value)) + && ((value & 0x10) + || !(rc = pci_write_config_byte(via, 0x42, value | 0x10)))) { + printk(KERN_INFO CARD_NAME + ": bridge config is 0x%x\n", value | 0x10); + } else { + printk(KERN_WARNING CARD_NAME + ": could not set vortex latency: pci error 0x%x\n", rc); + } +} + +static void __devinit snd_vortex_workaround(struct pci_dev *vortex, int fix) +{ + struct pci_dev *via = NULL; + + /* autodetect if workarounds are required */ + if (fix == 255) { + /* VIA KT133 */ + via = pci_get_device(PCI_VENDOR_ID_VIA, + PCI_DEVICE_ID_VIA_8365_1, NULL); + /* VIA Apollo */ + if (via == NULL) { + via = pci_get_device(PCI_VENDOR_ID_VIA, + PCI_DEVICE_ID_VIA_82C598_1, NULL); + /* AMD Irongate */ + if (via == NULL) + via = pci_get_device(PCI_VENDOR_ID_AMD, + PCI_DEVICE_ID_AMD_FE_GATE_7007, NULL); + } + if (via) { + printk(KERN_INFO CARD_NAME ": Activating latency workaround...\n"); + vortex_fix_latency(vortex); + vortex_fix_agp_bridge(via); + } + } else { + if (fix & 0x1) + vortex_fix_latency(vortex); + if ((fix & 0x2) && (via = pci_get_device(PCI_VENDOR_ID_VIA, + PCI_DEVICE_ID_VIA_8365_1, NULL))) + vortex_fix_agp_bridge(via); + if ((fix & 0x4) && (via = pci_get_device(PCI_VENDOR_ID_VIA, + PCI_DEVICE_ID_VIA_82C598_1, NULL))) + vortex_fix_agp_bridge(via); + if ((fix & 0x8) && (via = pci_get_device(PCI_VENDOR_ID_AMD, + PCI_DEVICE_ID_AMD_FE_GATE_7007, NULL))) + vortex_fix_agp_bridge(via); + } + pci_dev_put(via); +} + +// component-destructor +// (see "Management of Cards and Components") +static int snd_vortex_dev_free(struct snd_device *device) +{ + vortex_t *vortex = device->device_data; + + vortex_gameport_unregister(vortex); + vortex_core_shutdown(vortex); + // Take down PCI interface. + synchronize_irq(vortex->irq); + free_irq(vortex->irq, vortex); + iounmap(vortex->mmio); + pci_release_regions(vortex->pci_dev); + pci_disable_device(vortex->pci_dev); + kfree(vortex); + + return 0; +} + +// chip-specific constructor +// (see "Management of Cards and Components") +static int __devinit +snd_vortex_create(struct snd_card *card, struct pci_dev *pci, vortex_t ** rchip) +{ + vortex_t *chip; + int err; + static struct snd_device_ops ops = { + .dev_free = snd_vortex_dev_free, + }; + + *rchip = NULL; + + // check PCI availability (DMA). + if ((err = pci_enable_device(pci)) < 0) + return err; + if (pci_set_dma_mask(pci, DMA_32BIT_MASK) < 0 || + pci_set_consistent_dma_mask(pci, DMA_32BIT_MASK) < 0) { + printk(KERN_ERR "error to set DMA mask\n"); + pci_disable_device(pci); + return -ENXIO; + } + + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + if (chip == NULL) { + pci_disable_device(pci); + return -ENOMEM; + } + + chip->card = card; + + // initialize the stuff + chip->pci_dev = pci; + chip->io = pci_resource_start(pci, 0); + chip->vendor = pci->vendor; + chip->device = pci->device; + chip->card = card; + chip->irq = -1; + + // (1) PCI resource allocation + // Get MMIO area + // + if ((err = pci_request_regions(pci, CARD_NAME_SHORT)) != 0) + goto regions_out; + + chip->mmio = ioremap_nocache(pci_resource_start(pci, 0), + pci_resource_len(pci, 0)); + if (!chip->mmio) { + printk(KERN_ERR "MMIO area remap failed.\n"); + err = -ENOMEM; + goto ioremap_out; + } + + /* Init audio core. + * This must be done before we do request_irq otherwise we can get spurious + * interupts that we do not handle properly and make a mess of things */ + if ((err = vortex_core_init(chip)) != 0) { + printk(KERN_ERR "hw core init failed\n"); + goto core_out; + } + + if ((err = request_irq(pci->irq, vortex_interrupt, + IRQF_SHARED, CARD_NAME_SHORT, + chip)) != 0) { + printk(KERN_ERR "cannot grab irq\n"); + goto irq_out; + } + chip->irq = pci->irq; + + pci_set_master(pci); + // End of PCI setup. + + // Register alsa root device. + if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) { + goto alloc_out; + } + + snd_card_set_dev(card, &pci->dev); + + *rchip = chip; + + return 0; + + alloc_out: + synchronize_irq(chip->irq); + free_irq(chip->irq, chip); + irq_out: + vortex_core_shutdown(chip); + core_out: + iounmap(chip->mmio); + ioremap_out: + pci_release_regions(chip->pci_dev); + regions_out: + pci_disable_device(chip->pci_dev); + //FIXME: this not the right place to unregister the gameport + vortex_gameport_unregister(chip); + kfree(chip); + return err; +} + +// constructor -- see "Constructor" sub-section +static int __devinit +snd_vortex_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) +{ + static int dev; + struct snd_card *card; + vortex_t *chip; + int err; + + // (1) + if (dev >= SNDRV_CARDS) + return -ENODEV; + if (!enable[dev]) { + dev++; + return -ENOENT; + } + // (2) + card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); + if (card == NULL) + return -ENOMEM; + + // (3) + if ((err = snd_vortex_create(card, pci, &chip)) < 0) { + snd_card_free(card); + return err; + } + snd_vortex_workaround(pci, pcifix[dev]); + + // Card details needed in snd_vortex_midi + strcpy(card->driver, CARD_NAME_SHORT); + sprintf(card->shortname, "Aureal Vortex %s", CARD_NAME_SHORT); + sprintf(card->longname, "%s at 0x%lx irq %i", + card->shortname, chip->io, chip->irq); + + // (4) Alloc components. + // ADB pcm. + if ((err = snd_vortex_new_pcm(chip, VORTEX_PCM_ADB, NR_ADB)) < 0) { + snd_card_free(card); + return err; + } +#ifndef CHIP_AU8820 + // ADB SPDIF + if ((err = snd_vortex_new_pcm(chip, VORTEX_PCM_SPDIF, 1)) < 0) { + snd_card_free(card); + return err; + } + // A3D + if ((err = snd_vortex_new_pcm(chip, VORTEX_PCM_A3D, NR_A3D)) < 0) { + snd_card_free(card); + return err; + } +#endif + /* + // ADB I2S + if ((err = snd_vortex_new_pcm(chip, VORTEX_PCM_I2S, 1)) < 0) { + snd_card_free(card); + return err; + } + */ +#ifndef CHIP_AU8810 + // WT pcm. + if ((err = snd_vortex_new_pcm(chip, VORTEX_PCM_WT, NR_WT)) < 0) { + snd_card_free(card); + return err; + } +#endif + // snd_ac97_mixer and Vortex mixer. + if ((err = snd_vortex_mixer(chip)) < 0) { + snd_card_free(card); + return err; + } + if ((err = snd_vortex_midi(chip)) < 0) { + snd_card_free(card); + return err; + } + + vortex_gameport_register(chip); + +#if 0 + if (snd_seq_device_new(card, 1, SNDRV_SEQ_DEV_ID_VORTEX_SYNTH, + sizeof(snd_vortex_synth_arg_t), &wave) < 0 + || wave == NULL) { + snd_printk(KERN_ERR "Can't initialize Aureal wavetable synth\n"); + } else { + snd_vortex_synth_arg_t *arg; + + arg = SNDRV_SEQ_DEVICE_ARGPTR(wave); + strcpy(wave->name, "Aureal Synth"); + arg->hwptr = vortex; + arg->index = 1; + arg->seq_ports = seq_ports[dev]; + arg->max_voices = max_synth_voices[dev]; + } +#endif + + // (5) + if ((err = pci_read_config_word(pci, PCI_DEVICE_ID, + &(chip->device))) < 0) { + snd_card_free(card); + return err; + } + if ((err = pci_read_config_word(pci, PCI_VENDOR_ID, + &(chip->vendor))) < 0) { + snd_card_free(card); + return err; + } + chip->rev = pci->revision; +#ifdef CHIP_AU8830 + if ((chip->rev) != 0xfe && (chip->rev) != 0xfa) { + printk(KERN_ALERT + "vortex: The revision (%x) of your card has not been seen before.\n", + chip->rev); + printk(KERN_ALERT + "vortex: Please email the results of 'lspci -vv' to openvortex-dev@nongnu.org.\n"); + snd_card_free(card); + err = -ENODEV; + return err; + } +#endif + + // (6) + if ((err = snd_card_register(card)) < 0) { + snd_card_free(card); + return err; + } + // (7) + pci_set_drvdata(pci, card); + dev++; + vortex_connect_default(chip, 1); + vortex_enable_int(chip); + return 0; +} + +// destructor -- see "Destructor" sub-section +static void __devexit snd_vortex_remove(struct pci_dev *pci) +{ + snd_card_free(pci_get_drvdata(pci)); + pci_set_drvdata(pci, NULL); +} + +// pci_driver definition +static struct pci_driver driver = { + .name = CARD_NAME_SHORT, + .id_table = snd_vortex_ids, + .probe = snd_vortex_probe, + .remove = __devexit_p(snd_vortex_remove), +}; + +// initialization of the module +static int __init alsa_card_vortex_init(void) +{ + return pci_register_driver(&driver); +} + +// clean up the module +static void __exit alsa_card_vortex_exit(void) +{ + pci_unregister_driver(&driver); +} + +module_init(alsa_card_vortex_init) +module_exit(alsa_card_vortex_exit) diff -ruN linux-2.6.22/sound/pci/au88x0/au88x0_eq.c linux-2.6.22-alsa/sound/pci/au88x0/au88x0_eq.c --- linux-2.6.22/sound/pci/au88x0/au88x0_eq.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/pci/au88x0/au88x0_eq.c 2007-09-01 20:55:23.000000000 +0200 @@ -728,15 +728,7 @@ /* ALSA interface */ /* Control interface */ -static int -snd_vortex_eqtoggle_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_vortex_eqtoggle_info snd_ctl_boolean_mono_info static int snd_vortex_eqtoggle_get(struct snd_kcontrol *kcontrol, diff -ruN linux-2.6.22/sound/pci/bt87x.c linux-2.6.22-alsa/sound/pci/bt87x.c --- linux-2.6.22/sound/pci/bt87x.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/pci/bt87x.c 2007-09-01 20:55:23.000000000 +0200 @@ -340,28 +340,9 @@ static int snd_bt87x_set_digital_hw(struct snd_bt87x *chip, struct snd_pcm_runtime *runtime) { - static struct { - int rate; - unsigned int bit; - } ratebits[] = { - {8000, SNDRV_PCM_RATE_8000}, - {11025, SNDRV_PCM_RATE_11025}, - {16000, SNDRV_PCM_RATE_16000}, - {22050, SNDRV_PCM_RATE_22050}, - {32000, SNDRV_PCM_RATE_32000}, - {44100, SNDRV_PCM_RATE_44100}, - {48000, SNDRV_PCM_RATE_48000} - }; - int i; - chip->reg_control |= CTL_DA_IOM_DA; runtime->hw = snd_bt87x_digital_hw; - runtime->hw.rates = SNDRV_PCM_RATE_KNOT; - for (i = 0; i < ARRAY_SIZE(ratebits); ++i) - if (chip->dig_rate == ratebits[i].rate) { - runtime->hw.rates = ratebits[i].bit; - break; - } + runtime->hw.rates = snd_pcm_rate_to_rate_bit(chip->dig_rate); runtime->hw.rate_min = chip->dig_rate; runtime->hw.rate_max = chip->dig_rate; return 0; @@ -569,15 +550,7 @@ .put = snd_bt87x_capture_volume_put, }; -static int snd_bt87x_capture_boost_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *info) -{ - info->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - info->count = 1; - info->value.integer.min = 0; - info->value.integer.max = 1; - return 0; -} +#define snd_bt87x_capture_boost_info snd_ctl_boolean_mono_info static int snd_bt87x_capture_boost_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *value) @@ -781,6 +754,8 @@ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_879, 0x0070, 0x13eb, 32000), /* Viewcast Osprey 200 */ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x0070, 0xff01, 44100), + /* Viewcast Osprey 440 (rate is configurable via gpio) */ + BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x0070, 0xff07, 32000), /* ATI TV-Wonder */ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x1002, 0x0001, 32000), /* Leadtek Winfast tv 2000xp delux */ @@ -789,6 +764,10 @@ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x121a, 0x3000, 32000), /* AVerMedia Studio No. 103, 203, ...? */ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x1461, 0x0003, 48000), + /* Prolink PixelView PV-M4900 */ + BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x1554, 0x4011, 32000), + /* Pinnacle Studio PCTV rave */ + BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0xbd11, 0x1200, 32000), { } }; MODULE_DEVICE_TABLE(pci, snd_bt87x_ids); diff -ruN linux-2.6.22/sound/pci/ca0106/ca0106.h linux-2.6.22-alsa/sound/pci/ca0106/ca0106.h --- linux-2.6.22/sound/pci/ca0106/ca0106.h 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/pci/ca0106/ca0106.h 2007-09-02 13:30:43.000000000 +0200 @@ -1,7 +1,7 @@ /* * Copyright (c) 2004 James Courtier-Dutton * Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit - * Version: 0.0.21 + * Version: 0.0.22 * * FEATURES currently supported: * See ca0106_main.c for features. @@ -47,6 +47,8 @@ * Added GPIO info for SB Live 24bit. * 0.0.21 * Implement support for Line-in capture on SB Live 24bit. + * 0.0.22 + * Add support for mute control on SB Live 24bit (cards w/ SPI DAC) * * * This code was initally based on code from ALSA's emu10k1x.c which is: @@ -552,6 +554,95 @@ #define CONTROL_CENTER_LFE_CHANNEL 1 #define CONTROL_UNKNOWN_CHANNEL 2 + +/* Based on WM8768 Datasheet Rev 4.2 page 32 */ +#define SPI_REG_MASK 0x1ff /* 16-bit SPI writes have a 7-bit address */ +#define SPI_REG_SHIFT 9 /* followed by 9 bits of data */ + +#define SPI_LDA1_REG 0 /* digital attenuation */ +#define SPI_RDA1_REG 1 +#define SPI_LDA2_REG 4 +#define SPI_RDA2_REG 5 +#define SPI_LDA3_REG 6 +#define SPI_RDA3_REG 7 +#define SPI_LDA4_REG 13 +#define SPI_RDA4_REG 14 +#define SPI_MASTDA_REG 8 + +#define SPI_DA_BIT_UPDATE (1<<8) /* update attenuation values */ +#define SPI_DA_BIT_0dB 0xff /* 0 dB */ +#define SPI_DA_BIT_infdB 0x00 /* inf dB attenuation (mute) */ + +#define SPI_PL_REG 2 +#define SPI_PL_BIT_L_M (0<<5) /* left channel = mute */ +#define SPI_PL_BIT_L_L (1<<5) /* left channel = left */ +#define SPI_PL_BIT_L_R (2<<5) /* left channel = right */ +#define SPI_PL_BIT_L_C (3<<5) /* left channel = (L+R)/2 */ +#define SPI_PL_BIT_R_M (0<<7) /* right channel = mute */ +#define SPI_PL_BIT_R_L (1<<7) /* right channel = left */ +#define SPI_PL_BIT_R_R (2<<7) /* right channel = right */ +#define SPI_PL_BIT_R_C (3<<7) /* right channel = (L+R)/2 */ +#define SPI_IZD_REG 2 +#define SPI_IZD_BIT (1<<4) /* infinite zero detect */ + +#define SPI_FMT_REG 3 +#define SPI_FMT_BIT_RJ (0<<0) /* right justified mode */ +#define SPI_FMT_BIT_LJ (1<<0) /* left justified mode */ +#define SPI_FMT_BIT_I2S (2<<0) /* I2S mode */ +#define SPI_FMT_BIT_DSP (3<<0) /* DSP Modes A or B */ +#define SPI_LRP_REG 3 +#define SPI_LRP_BIT (1<<2) /* invert LRCLK polarity */ +#define SPI_BCP_REG 3 +#define SPI_BCP_BIT (1<<3) /* invert BCLK polarity */ +#define SPI_IWL_REG 3 +#define SPI_IWL_BIT_16 (0<<4) /* 16-bit world length */ +#define SPI_IWL_BIT_20 (1<<4) /* 20-bit world length */ +#define SPI_IWL_BIT_24 (2<<4) /* 24-bit world length */ +#define SPI_IWL_BIT_32 (3<<4) /* 32-bit world length */ + +#define SPI_MS_REG 10 +#define SPI_MS_BIT (1<<5) /* master mode */ +#define SPI_RATE_REG 10 /* only applies in master mode */ +#define SPI_RATE_BIT_128 (0<<6) /* MCLK = LRCLK * 128 */ +#define SPI_RATE_BIT_192 (1<<6) +#define SPI_RATE_BIT_256 (2<<6) +#define SPI_RATE_BIT_384 (3<<6) +#define SPI_RATE_BIT_512 (4<<6) +#define SPI_RATE_BIT_768 (5<<6) + +/* They really do label the bit for the 4th channel "4" and not "3" */ +#define SPI_DMUTE0_REG 9 +#define SPI_DMUTE1_REG 9 +#define SPI_DMUTE2_REG 9 +#define SPI_DMUTE4_REG 15 +#define SPI_DMUTE0_BIT (1<<3) +#define SPI_DMUTE1_BIT (1<<4) +#define SPI_DMUTE2_BIT (1<<5) +#define SPI_DMUTE4_BIT (1<<2) + +#define SPI_PHASE0_REG 3 +#define SPI_PHASE1_REG 3 +#define SPI_PHASE2_REG 3 +#define SPI_PHASE4_REG 15 +#define SPI_PHASE0_BIT (1<<6) +#define SPI_PHASE1_BIT (1<<7) +#define SPI_PHASE2_BIT (1<<8) +#define SPI_PHASE4_BIT (1<<3) + +#define SPI_PDWN_REG 2 /* power down all DACs */ +#define SPI_PDWN_BIT (1<<2) +#define SPI_DACD0_REG 10 /* power down individual DACs */ +#define SPI_DACD1_REG 10 +#define SPI_DACD2_REG 10 +#define SPI_DACD4_REG 15 +#define SPI_DACD0_BIT (1<<1) +#define SPI_DACD1_BIT (1<<2) +#define SPI_DACD2_BIT (1<<3) +#define SPI_DACD4_BIT (1<<0) /* datasheet error says it's 1 */ + +#define SPI_PWRDNALL_REG 10 /* power down everything */ +#define SPI_PWRDNALL_BIT (1<<4) + #include "ca_midi.h" struct snd_ca0106; @@ -612,6 +703,8 @@ struct snd_ca_midi midi; struct snd_ca_midi midi2; + + u16 spi_dac_reg[16]; }; int snd_ca0106_mixer(struct snd_ca0106 *emu); @@ -628,4 +721,5 @@ int snd_ca0106_i2c_write(struct snd_ca0106 *emu, u32 reg, u32 value); - +int snd_ca0106_spi_write(struct snd_ca0106 * emu, + unsigned int data); diff -ruN linux-2.6.22/sound/pci/ca0106/ca0106.h.orig linux-2.6.22-alsa/sound/pci/ca0106/ca0106.h.orig --- linux-2.6.22/sound/pci/ca0106/ca0106.h.orig 1970-01-01 01:00:00.000000000 +0100 +++ linux-2.6.22-alsa/sound/pci/ca0106/ca0106.h.orig 2007-09-01 20:55:23.000000000 +0200 @@ -0,0 +1,724 @@ +/* + * Copyright (c) 2004 James Courtier-Dutton + * Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit + * Version: 0.0.22 + * + * FEATURES currently supported: + * See ca0106_main.c for features. + * + * Changelog: + * Support interrupts per period. + * Removed noise from Center/LFE channel when in Analog mode. + * Rename and remove mixer controls. + * 0.0.6 + * Use separate card based DMA buffer for periods table list. + * 0.0.7 + * Change remove and rename ctrls into lists. + * 0.0.8 + * Try to fix capture sources. + * 0.0.9 + * Fix AC3 output. + * Enable S32_LE format support. + * 0.0.10 + * Enable playback 48000 and 96000 rates. (Rates other that these do not work, even with "plug:front".) + * 0.0.11 + * Add Model name recognition. + * 0.0.12 + * Correct interrupt timing. interrupt at end of period, instead of in the middle of a playback period. + * Remove redundent "voice" handling. + * 0.0.13 + * Single trigger call for multi channels. + * 0.0.14 + * Set limits based on what the sound card hardware can do. + * playback periods_min=2, periods_max=8 + * capture hw constraints require period_size = n * 64 bytes. + * playback hw constraints require period_size = n * 64 bytes. + * 0.0.15 + * Separated ca0106.c into separate functional .c files. + * 0.0.16 + * Implement 192000 sample rate. + * 0.0.17 + * Add support for SB0410 and SB0413. + * 0.0.18 + * Modified Copyright message. + * 0.0.19 + * Added I2C and SPI registers. Filled in interrupt enable. + * 0.0.20 + * Added GPIO info for SB Live 24bit. + * 0.0.21 + * Implement support for Line-in capture on SB Live 24bit. + * 0.0.22 + * Add support for mute control on SB Live 24bit (cards w/ SPI DAC) + * + * + * This code was initally based on code from ALSA's emu10k1x.c which is: + * Copyright (c) by Francisco Moraes + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +/************************************************************************************************/ +/* PCI function 0 registers, address = + PCIBASE0 */ +/************************************************************************************************/ + +#define PTR 0x00 /* Indexed register set pointer register */ + /* NOTE: The CHANNELNUM and ADDRESS words can */ + /* be modified independently of each other. */ + /* CNL[1:0], ADDR[27:16] */ + +#define DATA 0x04 /* Indexed register set data register */ + /* DATA[31:0] */ + +#define IPR 0x08 /* Global interrupt pending register */ + /* Clear pending interrupts by writing a 1 to */ + /* the relevant bits and zero to the other bits */ +#define IPR_MIDI_RX_B 0x00020000 /* MIDI UART-B Receive buffer non-empty */ +#define IPR_MIDI_TX_B 0x00010000 /* MIDI UART-B Transmit buffer empty */ +#define IPR_SPDIF_IN_USER 0x00004000 /* SPDIF input user data has 16 more bits */ +#define IPR_SPDIF_OUT_USER 0x00002000 /* SPDIF output user data needs 16 more bits */ +#define IPR_SPDIF_OUT_FRAME 0x00001000 /* SPDIF frame about to start */ +#define IPR_SPI 0x00000800 /* SPI transaction completed */ +#define IPR_I2C_EEPROM 0x00000400 /* I2C EEPROM transaction completed */ +#define IPR_I2C_DAC 0x00000200 /* I2C DAC transaction completed */ +#define IPR_AI 0x00000100 /* Audio pending register changed. See PTR reg 0x76 */ +#define IPR_GPI 0x00000080 /* General Purpose input changed */ +#define IPR_SRC_LOCKED 0x00000040 /* SRC lock status changed */ +#define IPR_SPDIF_STATUS 0x00000020 /* SPDIF status changed */ +#define IPR_TIMER2 0x00000010 /* 192000Hz Timer */ +#define IPR_TIMER1 0x00000008 /* 44100Hz Timer */ +#define IPR_MIDI_RX_A 0x00000004 /* MIDI UART-A Receive buffer non-empty */ +#define IPR_MIDI_TX_A 0x00000002 /* MIDI UART-A Transmit buffer empty */ +#define IPR_PCI 0x00000001 /* PCI Bus error */ + +#define INTE 0x0c /* Interrupt enable register */ + +#define INTE_MIDI_RX_B 0x00020000 /* MIDI UART-B Receive buffer non-empty */ +#define INTE_MIDI_TX_B 0x00010000 /* MIDI UART-B Transmit buffer empty */ +#define INTE_SPDIF_IN_USER 0x00004000 /* SPDIF input user data has 16 more bits */ +#define INTE_SPDIF_OUT_USER 0x00002000 /* SPDIF output user data needs 16 more bits */ +#define INTE_SPDIF_OUT_FRAME 0x00001000 /* SPDIF frame about to start */ +#define INTE_SPI 0x00000800 /* SPI transaction completed */ +#define INTE_I2C_EEPROM 0x00000400 /* I2C EEPROM transaction completed */ +#define INTE_I2C_DAC 0x00000200 /* I2C DAC transaction completed */ +#define INTE_AI 0x00000100 /* Audio pending register changed. See PTR reg 0x75 */ +#define INTE_GPI 0x00000080 /* General Purpose input changed */ +#define INTE_SRC_LOCKED 0x00000040 /* SRC lock status changed */ +#define INTE_SPDIF_STATUS 0x00000020 /* SPDIF status changed */ +#define INTE_TIMER2 0x00000010 /* 192000Hz Timer */ +#define INTE_TIMER1 0x00000008 /* 44100Hz Timer */ +#define INTE_MIDI_RX_A 0x00000004 /* MIDI UART-A Receive buffer non-empty */ +#define INTE_MIDI_TX_A 0x00000002 /* MIDI UART-A Transmit buffer empty */ +#define INTE_PCI 0x00000001 /* PCI Bus error */ + +#define UNKNOWN10 0x10 /* Unknown ??. Defaults to 0 */ +#define HCFG 0x14 /* Hardware config register */ + /* 0x1000 causes AC3 to fails. It adds a dither bit. */ + +#define HCFG_STAC 0x10000000 /* Special mode for STAC9460 Codec. */ +#define HCFG_CAPTURE_I2S_BYPASS 0x08000000 /* 1 = bypass I2S input async SRC. */ +#define HCFG_CAPTURE_SPDIF_BYPASS 0x04000000 /* 1 = bypass SPDIF input async SRC. */ +#define HCFG_PLAYBACK_I2S_BYPASS 0x02000000 /* 0 = I2S IN mixer output, 1 = I2S IN1. */ +#define HCFG_FORCE_LOCK 0x01000000 /* For test only. Force input SRC tracker to lock. */ +#define HCFG_PLAYBACK_ATTENUATION 0x00006000 /* Playback attenuation mask. 0 = 0dB, 1 = 6dB, 2 = 12dB, 3 = Mute. */ +#define HCFG_PLAYBACK_DITHER 0x00001000 /* 1 = Add dither bit to all playback channels. */ +#define HCFG_PLAYBACK_S32_LE 0x00000800 /* 1 = S32_LE, 0 = S16_LE */ +#define HCFG_CAPTURE_S32_LE 0x00000400 /* 1 = S32_LE, 0 = S16_LE (S32_LE current not working) */ +#define HCFG_8_CHANNEL_PLAY 0x00000200 /* 1 = 8 channels, 0 = 2 channels per substream.*/ +#define HCFG_8_CHANNEL_CAPTURE 0x00000100 /* 1 = 8 channels, 0 = 2 channels per substream.*/ +#define HCFG_MONO 0x00000080 /* 1 = I2S Input mono */ +#define HCFG_I2S_OUTPUT 0x00000010 /* 1 = I2S Output disabled */ +#define HCFG_AC97 0x00000008 /* 0 = AC97 1.0, 1 = AC97 2.0 */ +#define HCFG_LOCK_PLAYBACK_CACHE 0x00000004 /* 1 = Cancel bustmaster accesses to soundcache */ + /* NOTE: This should generally never be used. */ +#define HCFG_LOCK_CAPTURE_CACHE 0x00000002 /* 1 = Cancel bustmaster accesses to soundcache */ + /* NOTE: This should generally never be used. */ +#define HCFG_AUDIOENABLE 0x00000001 /* 0 = CODECs transmit zero-valued samples */ + /* Should be set to 1 when the EMU10K1 is */ + /* completely initialized. */ +#define GPIO 0x18 /* Defaults: 005f03a3-Analog, 005f02a2-SPDIF. */ + /* Here pins 0,1,2,3,4,,6 are output. 5,7 are input */ + /* For the Audigy LS, pin 0 (or bit 8) controls the SPDIF/Analog jack. */ + /* SB Live 24bit: + * bit 8 0 = SPDIF in and out / 1 = Analog (Mic or Line)-in. + * bit 9 0 = Mute / 1 = Analog out. + * bit 10 0 = Line-in / 1 = Mic-in. + * bit 11 0 = ? / 1 = ? + * bit 12 0 = 48 Khz / 1 = 96 Khz Analog out on SB Live 24bit. + * bit 13 0 = ? / 1 = ? + * bit 14 0 = Mute / 1 = Analog out + * bit 15 0 = ? / 1 = ? + * Both bit 9 and bit 14 have to be set for analog sound to work on the SB Live 24bit. + */ + /* 8 general purpose programmable In/Out pins. + * GPI [8:0] Read only. Default 0. + * GPO [15:8] Default 0x9. (Default to SPDIF jack enabled for SPDIF) + * GPO Enable [23:16] Default 0x0f. Setting a bit to 1, causes the pin to be an output pin. + */ +#define AC97DATA 0x1c /* AC97 register set data register (16 bit) */ + +#define AC97ADDRESS 0x1e /* AC97 register set address register (8 bit) */ + +/********************************************************************************************************/ +/* CA0106 pointer-offset register set, accessed through the PTR and DATA registers */ +/********************************************************************************************************/ + +/* Initally all registers from 0x00 to 0x3f have zero contents. */ +#define PLAYBACK_LIST_ADDR 0x00 /* Base DMA address of a list of pointers to each period/size */ + /* One list entry: 4 bytes for DMA address, + * 4 bytes for period_size << 16. + * One list entry is 8 bytes long. + * One list entry for each period in the buffer. + */ + /* ADDR[31:0], Default: 0x0 */ +#define PLAYBACK_LIST_SIZE 0x01 /* Size of list in bytes << 16. E.g. 8 periods -> 0x00380000 */ + /* SIZE[21:16], Default: 0x8 */ +#define PLAYBACK_LIST_PTR 0x02 /* Pointer to the current period being played */ + /* PTR[5:0], Default: 0x0 */ +#define PLAYBACK_UNKNOWN3 0x03 /* Not used ?? */ +#define PLAYBACK_DMA_ADDR 0x04 /* Playback DMA addresss */ + /* DMA[31:0], Default: 0x0 */ +#define PLAYBACK_PERIOD_SIZE 0x05 /* Playback period size. win2000 uses 0x04000000 */ + /* SIZE[31:16], Default: 0x0 */ +#define PLAYBACK_POINTER 0x06 /* Playback period pointer. Used with PLAYBACK_LIST_PTR to determine buffer position currently in DAC */ + /* POINTER[15:0], Default: 0x0 */ +#define PLAYBACK_PERIOD_END_ADDR 0x07 /* Playback fifo end address */ + /* END_ADDR[15:0], FLAG[16] 0 = don't stop, 1 = stop */ +#define PLAYBACK_FIFO_OFFSET_ADDRESS 0x08 /* Current fifo offset address [21:16] */ + /* Cache size valid [5:0] */ +#define PLAYBACK_UNKNOWN9 0x09 /* 0x9 to 0xf Unused */ +#define CAPTURE_DMA_ADDR 0x10 /* Capture DMA address */ + /* DMA[31:0], Default: 0x0 */ +#define CAPTURE_BUFFER_SIZE 0x11 /* Capture buffer size */ + /* SIZE[31:16], Default: 0x0 */ +#define CAPTURE_POINTER 0x12 /* Capture buffer pointer. Sample currently in ADC */ + /* POINTER[15:0], Default: 0x0 */ +#define CAPTURE_FIFO_OFFSET_ADDRESS 0x13 /* Current fifo offset address [21:16] */ + /* Cache size valid [5:0] */ +#define PLAYBACK_LAST_SAMPLE 0x20 /* The sample currently being played */ +/* 0x21 - 0x3f unused */ +#define BASIC_INTERRUPT 0x40 /* Used by both playback and capture interrupt handler */ + /* Playback (0x1< Center Speaker, 2 -> Sub Woofer, 3 -> Ground, 4 -> Ground + * For Digital: 1 -> Front SPDIF, 2 -> Rear SPDIF, 3 -> Center/Subwoofer SPDIF, 4 -> Ground. + * Standard 4 pole Video A/V cable with RCA outputs: 1 -> White, 2 -> Yellow, 3 -> Sheild on all three, 4 -> Red. + * So, from this you can see that you cannot use a Standard 4 pole Video A/V cable with the SB Audigy LS card. + */ +/* The Front SPDIF PCM gets mixed with samples from the AC97 codec, so can only work for Stereo PCM and not AC3/DTS + * The Rear SPDIF can be used for Stereo PCM and also AC3/DTS + * The Center/LFE SPDIF cannot be used for AC3/DTS, but can be used for Stereo PCM. + * Summary: For ALSA we use the Rear channel for SPDIF Digital AC3/DTS output + */ +/* A standard 2 pole mono mini-jack to RCA plug can be used for SPDIF Stereo PCM output from the Front channel. + * A standard 3 pole stereo mini-jack to 2 RCA plugs can be used for SPDIF AC3/DTS and Stereo PCM output utilising the Rear channel and just one of the RCA plugs. + */ +#define SPCS0 0x41 /* SPDIF output Channel Status 0 register. For Rear. default=0x02108004, non-audio=0x02108006 */ +#define SPCS1 0x42 /* SPDIF output Channel Status 1 register. For Front */ +#define SPCS2 0x43 /* SPDIF output Channel Status 2 register. For Center/LFE */ +#define SPCS3 0x44 /* SPDIF output Channel Status 3 register. Unknown */ + /* When Channel set to 0: */ +#define SPCS_CLKACCYMASK 0x30000000 /* Clock accuracy */ +#define SPCS_CLKACCY_1000PPM 0x00000000 /* 1000 parts per million */ +#define SPCS_CLKACCY_50PPM 0x10000000 /* 50 parts per million */ +#define SPCS_CLKACCY_VARIABLE 0x20000000 /* Variable accuracy */ +#define SPCS_SAMPLERATEMASK 0x0f000000 /* Sample rate */ +#define SPCS_SAMPLERATE_44 0x00000000 /* 44.1kHz sample rate */ +#define SPCS_SAMPLERATE_48 0x02000000 /* 48kHz sample rate */ +#define SPCS_SAMPLERATE_32 0x03000000 /* 32kHz sample rate */ +#define SPCS_CHANNELNUMMASK 0x00f00000 /* Channel number */ +#define SPCS_CHANNELNUM_UNSPEC 0x00000000 /* Unspecified channel number */ +#define SPCS_CHANNELNUM_LEFT 0x00100000 /* Left channel */ +#define SPCS_CHANNELNUM_RIGHT 0x00200000 /* Right channel */ +#define SPCS_SOURCENUMMASK 0x000f0000 /* Source number */ +#define SPCS_SOURCENUM_UNSPEC 0x00000000 /* Unspecified source number */ +#define SPCS_GENERATIONSTATUS 0x00008000 /* Originality flag (see IEC-958 spec) */ +#define SPCS_CATEGORYCODEMASK 0x00007f00 /* Category code (see IEC-958 spec) */ +#define SPCS_MODEMASK 0x000000c0 /* Mode (see IEC-958 spec) */ +#define SPCS_EMPHASISMASK 0x00000038 /* Emphasis */ +#define SPCS_EMPHASIS_NONE 0x00000000 /* No emphasis */ +#define SPCS_EMPHASIS_50_15 0x00000008 /* 50/15 usec 2 channel */ +#define SPCS_COPYRIGHT 0x00000004 /* Copyright asserted flag -- do not modify */ +#define SPCS_NOTAUDIODATA 0x00000002 /* 0 = Digital audio, 1 = not audio */ +#define SPCS_PROFESSIONAL 0x00000001 /* 0 = Consumer (IEC-958), 1 = pro (AES3-1992) */ + + /* When Channel set to 1: */ +#define SPCS_WORD_LENGTH_MASK 0x0000000f /* Word Length Mask */ +#define SPCS_WORD_LENGTH_16 0x00000008 /* Word Length 16 bit */ +#define SPCS_WORD_LENGTH_17 0x00000006 /* Word Length 17 bit */ +#define SPCS_WORD_LENGTH_18 0x00000004 /* Word Length 18 bit */ +#define SPCS_WORD_LENGTH_19 0x00000002 /* Word Length 19 bit */ +#define SPCS_WORD_LENGTH_20A 0x0000000a /* Word Length 20 bit */ +#define SPCS_WORD_LENGTH_20 0x00000009 /* Word Length 20 bit (both 0xa and 0x9 are 20 bit) */ +#define SPCS_WORD_LENGTH_21 0x00000007 /* Word Length 21 bit */ +#define SPCS_WORD_LENGTH_21 0x00000007 /* Word Length 21 bit */ +#define SPCS_WORD_LENGTH_22 0x00000005 /* Word Length 22 bit */ +#define SPCS_WORD_LENGTH_23 0x00000003 /* Word Length 23 bit */ +#define SPCS_WORD_LENGTH_24 0x0000000b /* Word Length 24 bit */ +#define SPCS_ORIGINAL_SAMPLE_RATE_MASK 0x000000f0 /* Original Sample rate */ +#define SPCS_ORIGINAL_SAMPLE_RATE_NONE 0x00000000 /* Original Sample rate not indicated */ +#define SPCS_ORIGINAL_SAMPLE_RATE_16000 0x00000010 /* Original Sample rate */ +#define SPCS_ORIGINAL_SAMPLE_RATE_RES1 0x00000020 /* Original Sample rate */ +#define SPCS_ORIGINAL_SAMPLE_RATE_32000 0x00000030 /* Original Sample rate */ +#define SPCS_ORIGINAL_SAMPLE_RATE_12000 0x00000040 /* Original Sample rate */ +#define SPCS_ORIGINAL_SAMPLE_RATE_11025 0x00000050 /* Original Sample rate */ +#define SPCS_ORIGINAL_SAMPLE_RATE_8000 0x00000060 /* Original Sample rate */ +#define SPCS_ORIGINAL_SAMPLE_RATE_RES2 0x00000070 /* Original Sample rate */ +#define SPCS_ORIGINAL_SAMPLE_RATE_192000 0x00000080 /* Original Sample rate */ +#define SPCS_ORIGINAL_SAMPLE_RATE_24000 0x00000090 /* Original Sample rate */ +#define SPCS_ORIGINAL_SAMPLE_RATE_96000 0x000000a0 /* Original Sample rate */ +#define SPCS_ORIGINAL_SAMPLE_RATE_48000 0x000000b0 /* Original Sample rate */ +#define SPCS_ORIGINAL_SAMPLE_RATE_176400 0x000000c0 /* Original Sample rate */ +#define SPCS_ORIGINAL_SAMPLE_RATE_22050 0x000000d0 /* Original Sample rate */ +#define SPCS_ORIGINAL_SAMPLE_RATE_88200 0x000000e0 /* Original Sample rate */ +#define SPCS_ORIGINAL_SAMPLE_RATE_44100 0x000000f0 /* Original Sample rate */ + +#define SPDIF_SELECT1 0x45 /* Enables SPDIF or Analogue outputs 0-SPDIF, 0xf00-Analogue */ + /* 0x100 - Front, 0x800 - Rear, 0x200 - Center/LFE. + * But as the jack is shared, use 0xf00. + * The Windows2000 driver uses 0x0000000f for both digital and analog. + * 0xf00 introduces interesting noises onto the Center/LFE. + * If you turn the volume up, you hear computer noise, + * e.g. mouse moving, changing between app windows etc. + * So, I am going to set this to 0x0000000f all the time now, + * same as the windows driver does. + * Use register SPDIF_SELECT2(0x72) to switch between SPDIF and Analog. + */ + /* When Channel = 0: + * Wide SPDIF format [3:0] (one bit for each channel) (0=20bit, 1=24bit) + * Tristate SPDIF Output [11:8] (one bit for each channel) (0=Not tristate, 1=Tristate) + * SPDIF Bypass enable [19:16] (one bit for each channel) (0=Not bypass, 1=Bypass) + */ + /* When Channel = 1: + * SPDIF 0 User data [7:0] + * SPDIF 1 User data [15:8] + * SPDIF 0 User data [23:16] + * SPDIF 0 User data [31:24] + * User data can be sent by using the SPDIF output frame pending and SPDIF output user bit interrupts. + */ +#define WATERMARK 0x46 /* Test bit to indicate cache usage level */ +#define SPDIF_INPUT_STATUS 0x49 /* SPDIF Input status register. Bits the same as SPCS. + * When Channel = 0: Bits the same as SPCS channel 0. + * When Channel = 1: Bits the same as SPCS channel 1. + * When Channel = 2: + * SPDIF Input User data [16:0] + * SPDIF Input Frame count [21:16] + */ +#define CAPTURE_CACHE_DATA 0x50 /* 0x50-0x5f Recorded samples. */ +#define CAPTURE_SOURCE 0x60 /* Capture Source 0 = MIC */ +#define CAPTURE_SOURCE_CHANNEL0 0xf0000000 /* Mask for selecting the Capture sources */ +#define CAPTURE_SOURCE_CHANNEL1 0x0f000000 /* 0 - SPDIF mixer output. */ +#define CAPTURE_SOURCE_CHANNEL2 0x00f00000 /* 1 - What you hear or . 2 - ?? */ +#define CAPTURE_SOURCE_CHANNEL3 0x000f0000 /* 3 - Mic in, Line in, TAD in, Aux in. */ +#define CAPTURE_SOURCE_RECORD_MAP 0x0000ffff /* Default 0x00e4 */ + /* Record Map [7:0] (2 bits per channel) 0=mapped to channel 0, 1=mapped to channel 1, 2=mapped to channel2, 3=mapped to channel3 + * Record source select for channel 0 [18:16] + * Record source select for channel 1 [22:20] + * Record source select for channel 2 [26:24] + * Record source select for channel 3 [30:28] + * 0 - SPDIF mixer output. + * 1 - i2s mixer output. + * 2 - SPDIF input. + * 3 - i2s input. + * 4 - AC97 capture. + * 5 - SRC output. + */ +#define CAPTURE_VOLUME1 0x61 /* Capture volume per channel 0-3 */ +#define CAPTURE_VOLUME2 0x62 /* Capture volume per channel 4-7 */ + +#define PLAYBACK_ROUTING1 0x63 /* Playback routing of channels 0-7. Effects AC3 output. Default 0x32765410 */ +#define ROUTING1_REAR 0x77000000 /* Channel_id 0 sends to 10, Channel_id 1 sends to 32 */ +#define ROUTING1_NULL 0x00770000 /* Channel_id 2 sends to 54, Channel_id 3 sends to 76 */ +#define ROUTING1_CENTER_LFE 0x00007700 /* 0x32765410 means, send Channel_id 0 to FRONT, Channel_id 1 to REAR */ +#define ROUTING1_FRONT 0x00000077 /* Channel_id 2 to CENTER_LFE, Channel_id 3 to NULL. */ + /* Channel_id's handle stereo channels. Channel X is a single mono channel */ + /* Host is input from the PCI bus. */ + /* Host channel 0 [2:0] -> SPDIF Mixer/Router channel 0-7. + * Host channel 1 [6:4] -> SPDIF Mixer/Router channel 0-7. + * Host channel 2 [10:8] -> SPDIF Mixer/Router channel 0-7. + * Host channel 3 [14:12] -> SPDIF Mixer/Router channel 0-7. + * Host channel 4 [18:16] -> SPDIF Mixer/Router channel 0-7. + * Host channel 5 [22:20] -> SPDIF Mixer/Router channel 0-7. + * Host channel 6 [26:24] -> SPDIF Mixer/Router channel 0-7. + * Host channel 7 [30:28] -> SPDIF Mixer/Router channel 0-7. + */ + +#define PLAYBACK_ROUTING2 0x64 /* Playback Routing . Feeding Capture channels back into Playback. Effects AC3 output. Default 0x76767676 */ + /* SRC is input from the capture inputs. */ + /* SRC channel 0 [2:0] -> SPDIF Mixer/Router channel 0-7. + * SRC channel 1 [6:4] -> SPDIF Mixer/Router channel 0-7. + * SRC channel 2 [10:8] -> SPDIF Mixer/Router channel 0-7. + * SRC channel 3 [14:12] -> SPDIF Mixer/Router channel 0-7. + * SRC channel 4 [18:16] -> SPDIF Mixer/Router channel 0-7. + * SRC channel 5 [22:20] -> SPDIF Mixer/Router channel 0-7. + * SRC channel 6 [26:24] -> SPDIF Mixer/Router channel 0-7. + * SRC channel 7 [30:28] -> SPDIF Mixer/Router channel 0-7. + */ + +#define PLAYBACK_MUTE 0x65 /* Unknown. While playing 0x0, while silent 0x00fc0000 */ + /* SPDIF Mixer input control: + * Invert SRC to SPDIF Mixer [7-0] (One bit per channel) + * Invert Host to SPDIF Mixer [15:8] (One bit per channel) + * SRC to SPDIF Mixer disable [23:16] (One bit per channel) + * Host to SPDIF Mixer disable [31:24] (One bit per channel) + */ +#define PLAYBACK_VOLUME1 0x66 /* Playback SPDIF volume per channel. Set to the same PLAYBACK_VOLUME(0x6a) */ + /* PLAYBACK_VOLUME1 must be set to 30303030 for SPDIF AC3 Playback */ + /* SPDIF mixer input volume. 0=12dB, 0x30=0dB, 0xFE=-51.5dB, 0xff=Mute */ + /* One register for each of the 4 stereo streams. */ + /* SRC Right volume [7:0] + * SRC Left volume [15:8] + * Host Right volume [23:16] + * Host Left volume [31:24] + */ +#define CAPTURE_ROUTING1 0x67 /* Capture Routing. Default 0x32765410 */ + /* Similar to register 0x63, except that the destination is the I2S mixer instead of the SPDIF mixer. I.E. Outputs to the Analog outputs instead of SPDIF. */ +#define CAPTURE_ROUTING2 0x68 /* Unknown Routing. Default 0x76767676 */ + /* Similar to register 0x64, except that the destination is the I2S mixer instead of the SPDIF mixer. I.E. Outputs to the Analog outputs instead of SPDIF. */ +#define CAPTURE_MUTE 0x69 /* Unknown. While capturing 0x0, while silent 0x00fc0000 */ + /* Similar to register 0x65, except that the destination is the I2S mixer instead of the SPDIF mixer. I.E. Outputs to the Analog outputs instead of SPDIF. */ +#define PLAYBACK_VOLUME2 0x6a /* Playback Analog volume per channel. Does not effect AC3 output */ + /* Similar to register 0x66, except that the destination is the I2S mixer instead of the SPDIF mixer. I.E. Outputs to the Analog outputs instead of SPDIF. */ +#define UNKNOWN6b 0x6b /* Unknown. Readonly. Default 00400000 00400000 00400000 00400000 */ +#define MIDI_UART_A_DATA 0x6c /* Midi Uart A Data */ +#define MIDI_UART_A_CMD 0x6d /* Midi Uart A Command/Status */ +#define MIDI_UART_B_DATA 0x6e /* Midi Uart B Data (currently unused) */ +#define MIDI_UART_B_CMD 0x6f /* Midi Uart B Command/Status (currently unused) */ + +/* unique channel identifier for midi->channel */ + +#define CA0106_MIDI_CHAN_A 0x1 +#define CA0106_MIDI_CHAN_B 0x2 + +/* from mpu401 */ + +#define CA0106_MIDI_INPUT_AVAIL 0x80 +#define CA0106_MIDI_OUTPUT_READY 0x40 +#define CA0106_MPU401_RESET 0xff +#define CA0106_MPU401_ENTER_UART 0x3f +#define CA0106_MPU401_ACK 0xfe + +#define SAMPLE_RATE_TRACKER_STATUS 0x70 /* Readonly. Default 00108000 00108000 00500000 00500000 */ + /* Estimated sample rate [19:0] Relative to 48kHz. 0x8000 = 1.0 + * Rate Locked [20] + * SPDIF Locked [21] For SPDIF channel only. + * Valid Audio [22] For SPDIF channel only. + */ +#define CAPTURE_CONTROL 0x71 /* Some sort of routing. default = 40c81000 30303030 30300000 00700000 */ + /* Channel_id 0: 0x40c81000 must be changed to 0x40c80000 for SPDIF AC3 input or output. */ + /* Channel_id 1: 0xffffffff(mute) 0x30303030(max) controls CAPTURE feedback into PLAYBACK. */ + /* Sample rate output control register Channel=0 + * Sample output rate [1:0] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz) + * Sample input rate [3:2] (0=48kHz, 1=Not available, 2=96kHz, 3=192Khz) + * SRC input source select [4] 0=Audio from digital mixer, 1=Audio from analog source. + * Record rate [9:8] (0=48kHz, 1=Not available, 2=96kHz, 3=192Khz) + * Record mixer output enable [12:10] + * I2S input rate master mode [15:14] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz) + * I2S output rate [17:16] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz) + * I2S output source select [18] (0=Audio from host, 1=Audio from SRC) + * Record mixer I2S enable [20:19] (enable/disable i2sin1 and i2sin0) + * I2S output master clock select [21] (0=256*I2S output rate, 1=512*I2S output rate.) + * I2S input master clock select [22] (0=256*I2S input rate, 1=512*I2S input rate.) + * I2S input mode [23] (0=Slave, 1=Master) + * SPDIF output rate [25:24] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz) + * SPDIF output source select [26] (0=host, 1=SRC) + * Not used [27] + * Record Source 0 input [29:28] (0=SPDIF in, 1=I2S in, 2=AC97 Mic, 3=AC97 PCM) + * Record Source 1 input [31:30] (0=SPDIF in, 1=I2S in, 2=AC97 Mic, 3=AC97 PCM) + */ + /* Sample rate output control register Channel=1 + * I2S Input 0 volume Right [7:0] + * I2S Input 0 volume Left [15:8] + * I2S Input 1 volume Right [23:16] + * I2S Input 1 volume Left [31:24] + */ + /* Sample rate output control register Channel=2 + * SPDIF Input volume Right [23:16] + * SPDIF Input volume Left [31:24] + */ + /* Sample rate output control register Channel=3 + * No used + */ +#define SPDIF_SELECT2 0x72 /* Some sort of routing. Channel_id 0 only. default = 0x0f0f003f. Analog 0x000b0000, Digital 0x0b000000 */ +#define ROUTING2_FRONT_MASK 0x00010000 /* Enable for Front speakers. */ +#define ROUTING2_CENTER_LFE_MASK 0x00020000 /* Enable for Center/LFE speakers. */ +#define ROUTING2_REAR_MASK 0x00080000 /* Enable for Rear speakers. */ + /* Audio output control + * AC97 output enable [5:0] + * I2S output enable [19:16] + * SPDIF output enable [27:24] + */ +#define UNKNOWN73 0x73 /* Unknown. Readonly. Default 0x0 */ +#define CHIP_VERSION 0x74 /* P17 Chip version. Channel_id 0 only. Default 00000071 */ +#define EXTENDED_INT_MASK 0x75 /* Used by both playback and capture interrupt handler */ + /* Sets which Interrupts are enabled. */ + /* 0x00000001 = Half period. Playback. + * 0x00000010 = Full period. Playback. + * 0x00000100 = Half buffer. Playback. + * 0x00001000 = Full buffer. Playback. + * 0x00010000 = Half buffer. Capture. + * 0x00100000 = Full buffer. Capture. + * Capture can only do 2 periods. + * 0x01000000 = End audio. Playback. + * 0x40000000 = Half buffer Playback,Caputre xrun. + * 0x80000000 = Full buffer Playback,Caputre xrun. + */ +#define EXTENDED_INT 0x76 /* Used by both playback and capture interrupt handler */ + /* Shows which interrupts are active at the moment. */ + /* Same bit layout as EXTENDED_INT_MASK */ +#define COUNTER77 0x77 /* Counter range 0 to 0x3fffff, 192000 counts per second. */ +#define COUNTER78 0x78 /* Counter range 0 to 0x3fffff, 44100 counts per second. */ +#define EXTENDED_INT_TIMER 0x79 /* Channel_id 0 only. Used by both playback and capture interrupt handler */ + /* Causes interrupts based on timer intervals. */ +#define SPI 0x7a /* SPI: Serial Interface Register */ +#define I2C_A 0x7b /* I2C Address. 32 bit */ +#define I2C_D0 0x7c /* I2C Data Port 0. 32 bit */ +#define I2C_D1 0x7d /* I2C Data Port 1. 32 bit */ +//I2C values +#define I2C_A_ADC_ADD_MASK 0x000000fe //The address is a 7 bit address +#define I2C_A_ADC_RW_MASK 0x00000001 //bit mask for R/W +#define I2C_A_ADC_TRANS_MASK 0x00000010 //Bit mask for I2c address DAC value +#define I2C_A_ADC_ABORT_MASK 0x00000020 //Bit mask for I2C transaction abort flag +#define I2C_A_ADC_LAST_MASK 0x00000040 //Bit mask for Last word transaction +#define I2C_A_ADC_BYTE_MASK 0x00000080 //Bit mask for Byte Mode + +#define I2C_A_ADC_ADD 0x00000034 //This is the Device address for ADC +#define I2C_A_ADC_READ 0x00000001 //To perform a read operation +#define I2C_A_ADC_START 0x00000100 //Start I2C transaction +#define I2C_A_ADC_ABORT 0x00000200 //I2C transaction abort +#define I2C_A_ADC_LAST 0x00000400 //I2C last transaction +#define I2C_A_ADC_BYTE 0x00000800 //I2C one byte mode + +#define I2C_D_ADC_REG_MASK 0xfe000000 //ADC address register +#define I2C_D_ADC_DAT_MASK 0x01ff0000 //ADC data register + +#define ADC_TIMEOUT 0x00000007 //ADC Timeout Clock Disable +#define ADC_IFC_CTRL 0x0000000b //ADC Interface Control +#define ADC_MASTER 0x0000000c //ADC Master Mode Control +#define ADC_POWER 0x0000000d //ADC PowerDown Control +#define ADC_ATTEN_ADCL 0x0000000e //ADC Attenuation ADCL +#define ADC_ATTEN_ADCR 0x0000000f //ADC Attenuation ADCR +#define ADC_ALC_CTRL1 0x00000010 //ADC ALC Control 1 +#define ADC_ALC_CTRL2 0x00000011 //ADC ALC Control 2 +#define ADC_ALC_CTRL3 0x00000012 //ADC ALC Control 3 +#define ADC_NOISE_CTRL 0x00000013 //ADC Noise Gate Control +#define ADC_LIMIT_CTRL 0x00000014 //ADC Limiter Control +#define ADC_MUX 0x00000015 //ADC Mux offset + +#if 0 +/* FIXME: Not tested yet. */ +#define ADC_GAIN_MASK 0x000000ff //Mask for ADC Gain +#define ADC_ZERODB 0x000000cf //Value to set ADC to 0dB +#define ADC_MUTE_MASK 0x000000c0 //Mask for ADC mute +#define ADC_MUTE 0x000000c0 //Value to mute ADC +#define ADC_OSR 0x00000008 //Mask for ADC oversample rate select +#define ADC_TIMEOUT_DISABLE 0x00000008 //Value and mask to disable Timeout clock +#define ADC_HPF_DISABLE 0x00000100 //Value and mask to disable High pass filter +#define ADC_TRANWIN_MASK 0x00000070 //Mask for Length of Transient Window +#endif + +#define ADC_MUX_MASK 0x0000000f //Mask for ADC Mux +#define ADC_MUX_PHONE 0x00000001 //Value to select TAD at ADC Mux (Not used) +#define ADC_MUX_MIC 0x00000002 //Value to select Mic at ADC Mux +#define ADC_MUX_LINEIN 0x00000004 //Value to select LineIn at ADC Mux +#define ADC_MUX_AUX 0x00000008 //Value to select Aux at ADC Mux + +#define SET_CHANNEL 0 /* Testing channel outputs 0=Front, 1=Center/LFE, 2=Unknown, 3=Rear */ +#define PCM_FRONT_CHANNEL 0 +#define PCM_REAR_CHANNEL 1 +#define PCM_CENTER_LFE_CHANNEL 2 +#define PCM_UNKNOWN_CHANNEL 3 +#define CONTROL_FRONT_CHANNEL 0 +#define CONTROL_REAR_CHANNEL 3 +#define CONTROL_CENTER_LFE_CHANNEL 1 +#define CONTROL_UNKNOWN_CHANNEL 2 + + +/* Based on WM8768 Datasheet Rev 4.2 page 32 */ +#define SPI_REG_MASK 0x1ff /* 16-bit SPI writes have a 7-bit address */ +#define SPI_REG_SHIFT 9 /* followed by 9 bits of data */ + +#define SPI_LDA1_REG 0 /* digital attenuation */ +#define SPI_RDA1_REG 1 +#define SPI_LDA2_REG 4 +#define SPI_RDA2_REG 5 +#define SPI_LDA3_REG 6 +#define SPI_RDA3_REG 7 +#define SPI_LDA4_REG 13 +#define SPI_RDA4_REG 14 +#define SPI_MASTDA_REG 8 + +#define SPI_DA_BIT_UPDATE (1<<8) /* update attenuation values */ +#define SPI_DA_BIT_0dB 0xff /* 0 dB */ +#define SPI_DA_BIT_infdB 0x00 /* inf dB attenuation (mute) */ + +#define SPI_PL_REG 2 +#define SPI_PL_BIT_L_M (0<<5) /* left channel = mute */ +#define SPI_PL_BIT_L_L (1<<5) /* left channel = left */ +#define SPI_PL_BIT_L_R (2<<5) /* left channel = right */ +#define SPI_PL_BIT_L_C (3<<5) /* left channel = (L+R)/2 */ +#define SPI_PL_BIT_R_M (0<<7) /* right channel = mute */ +#define SPI_PL_BIT_R_L (1<<7) /* right channel = left */ +#define SPI_PL_BIT_R_R (2<<7) /* right channel = right */ +#define SPI_PL_BIT_R_C (3<<7) /* right channel = (L+R)/2 */ +#define SPI_IZD_REG 2 +#define SPI_IZD_BIT (1<<4) /* infinite zero detect */ + +#define SPI_FMT_REG 3 +#define SPI_FMT_BIT_RJ (0<<0) /* right justified mode */ +#define SPI_FMT_BIT_LJ (1<<0) /* left justified mode */ +#define SPI_FMT_BIT_I2S (2<<0) /* I2S mode */ +#define SPI_FMT_BIT_DSP (3<<0) /* DSP Modes A or B */ +#define SPI_LRP_REG 3 +#define SPI_LRP_BIT (1<<2) /* invert LRCLK polarity */ +#define SPI_BCP_REG 3 +#define SPI_BCP_BIT (1<<3) /* invert BCLK polarity */ +#define SPI_IWL_REG 3 +#define SPI_IWL_BIT_16 (0<<4) /* 16-bit world length */ +#define SPI_IWL_BIT_20 (1<<4) /* 20-bit world length */ +#define SPI_IWL_BIT_24 (2<<4) /* 24-bit world length */ +#define SPI_IWL_BIT_32 (3<<4) /* 32-bit world length */ + +#define SPI_MS_REG 10 +#define SPI_MS_BIT (1<<5) /* master mode */ +#define SPI_RATE_REG 10 /* only applies in master mode */ +#define SPI_RATE_BIT_128 (0<<6) /* MCLK = LRCLK * 128 */ +#define SPI_RATE_BIT_192 (1<<6) +#define SPI_RATE_BIT_256 (2<<6) +#define SPI_RATE_BIT_384 (3<<6) +#define SPI_RATE_BIT_512 (4<<6) +#define SPI_RATE_BIT_768 (5<<6) + +/* They really do label the bit for the 4th channel "4" and not "3" */ +#define SPI_DMUTE0_REG 9 +#define SPI_DMUTE1_REG 9 +#define SPI_DMUTE2_REG 9 +#define SPI_DMUTE4_REG 15 +#define SPI_DMUTE0_BIT (1<<3) +#define SPI_DMUTE1_BIT (1<<4) +#define SPI_DMUTE2_BIT (1<<5) +#define SPI_DMUTE4_BIT (1<<2) + +#define SPI_PHASE0_REG 3 +#define SPI_PHASE1_REG 3 +#define SPI_PHASE2_REG 3 +#define SPI_PHASE4_REG 15 +#define SPI_PHASE0_BIT (1<<6) +#define SPI_PHASE1_BIT (1<<7) +#define SPI_PHASE2_BIT (1<<8) +#define SPI_PHASE4_BIT (1<<3) + +#define SPI_PDWN_REG 2 /* power down all DACs */ +#define SPI_PDWN_BIT (1<<2) +#define SPI_DACD0_REG 10 /* power down individual DACs */ +#define SPI_DACD1_REG 10 +#define SPI_DACD2_REG 10 +#define SPI_DACD4_REG 15 +#define SPI_DACD0_BIT (1<<1) +#define SPI_DACD1_BIT (1<<2) +#define SPI_DACD2_BIT (1<<3) +#define SPI_DACD4_BIT (1<<0) /* datasheet error says it's 1 */ + +#define SPI_PWRDNALL_REG 10 /* power down everything */ +#define SPI_PWRDNALL_BIT (1<<4) + +#include "ca_midi.h" + +struct snd_ca0106; + +struct snd_ca0106_channel { + struct snd_ca0106 *emu; + int number; + int use; + void (*interrupt)(struct snd_ca0106 *emu, struct snd_ca0106_channel *channel); + struct snd_ca0106_pcm *epcm; +}; + +struct snd_ca0106_pcm { + struct snd_ca0106 *emu; + struct snd_pcm_substream *substream; + int channel_id; + unsigned short running; +}; + +struct snd_ca0106_details { + u32 serial; + char * name; + int ac97; + int gpio_type; + int i2c_adc; + int spi_dac; +}; + +// definition of the chip-specific record +struct snd_ca0106 { + struct snd_card *card; + struct snd_ca0106_details *details; + struct pci_dev *pci; + + unsigned long port; + struct resource *res_port; + int irq; + + unsigned int serial; /* serial number */ + unsigned short model; /* subsystem id */ + + spinlock_t emu_lock; + + struct snd_ac97 *ac97; + struct snd_pcm *pcm; + + struct snd_ca0106_channel playback_channels[4]; + struct snd_ca0106_channel capture_channels[4]; + u32 spdif_bits[4]; /* s/pdif out setup */ + int spdif_enable; + int capture_source; + int i2c_capture_source; + u8 i2c_capture_volume[4][2]; + int capture_mic_line_in; + + struct snd_dma_buffer buffer; + + struct snd_ca_midi midi; + struct snd_ca_midi midi2; + + u16 spi_dac_reg[16]; +}; + +int snd_ca0106_mixer(struct snd_ca0106 *emu); +int snd_ca0106_proc_init(struct snd_ca0106 * emu); + +unsigned int snd_ca0106_ptr_read(struct snd_ca0106 * emu, + unsigned int reg, + unsigned int chn); + +void snd_ca0106_ptr_write(struct snd_ca0106 *emu, + unsigned int reg, + unsigned int chn, + unsigned int data); + +int snd_ca0106_i2c_write(struct snd_ca0106 *emu, u32 reg, u32 value); + +int snd_ca0106_spi_write(struct snd_ca0106 * emu, + unsigned int data); diff -ruN linux-2.6.22/sound/pci/ca0106/ca0106_main.c linux-2.6.22-alsa/sound/pci/ca0106/ca0106_main.c --- linux-2.6.22/sound/pci/ca0106/ca0106_main.c 2007-07-28 13:40:44.000000000 +0200 +++ linux-2.6.22-alsa/sound/pci/ca0106/ca0106_main.c 2007-09-02 13:30:43.000000000 +0200 @@ -1,7 +1,7 @@ /* * Copyright (c) 2004 James Courtier-Dutton * Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit - * Version: 0.0.23 + * Version: 0.0.25 * * FEATURES currently supported: * Front, Rear and Center/LFE. @@ -79,6 +79,10 @@ * Add support for MSI K8N Diamond Motherboard with onboard SB Live 24bit without AC97. From kiksen, bug #901 * 0.0.23 * Implement support for Line-in capture on SB Live 24bit. + * 0.0.24 + * Add support for mute control on SB Live 24bit (cards w/ SPI DAC) + * 0.0.25 + * Powerdown SPI DAC channels when not in use * * BUGS: * Some stability problems when unloading the snd-ca0106 kernel module. @@ -168,6 +172,34 @@ #include "ca0106.h" static struct snd_ca0106_details ca0106_chip_details[] = { + /* Sound Blaster X-Fi Extreme Audio. This does not have an AC97. 53SB079000000 */ + /* It is really just a normal SB Live 24bit. */ + /* Tested: + * See ALSA bug#3251 + */ + { .serial = 0x10131102, + .name = "X-Fi Extreme Audio [SBxxxx]", + .gpio_type = 1, + .i2c_adc = 1 } , + /* Sound Blaster X-Fi Extreme Audio. This does not have an AC97. 53SB079000000 */ + /* It is really just a normal SB Live 24bit. */ + /* + * CTRL:CA0111-WTLF + * ADC: WM8775SEDS + * DAC: CS4382-KQZ + */ + /* Tested: + * Playback on front, rear, center/lfe speakers + * Capture from Mic in. + * Not-Tested: + * Capture from Line in. + * Playback to digital out. + */ + { .serial = 0x10121102, + .name = "X-Fi Extreme Audio [SB0790]", + .gpio_type = 1, + .i2c_adc = 1 } , + /* New Dell Sound Blaster Live! 7.1 24bit. This does not have an AC97. */ /* AudigyLS[SB0310] */ { .serial = 0x10021102, .name = "AudigyLS [SB0310]", @@ -242,10 +274,11 @@ /* hardware definition */ static struct snd_pcm_hardware snd_ca0106_playback_hw = { - .info = (SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP_VALID), + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_SYNC_START, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, .rates = (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000), @@ -428,6 +461,19 @@ kfree(runtime->private_data); } +static const int spi_dacd_reg[] = { + [PCM_FRONT_CHANNEL] = SPI_DACD4_REG, + [PCM_REAR_CHANNEL] = SPI_DACD0_REG, + [PCM_CENTER_LFE_CHANNEL]= SPI_DACD2_REG, + [PCM_UNKNOWN_CHANNEL] = SPI_DACD1_REG, +}; +static const int spi_dacd_bit[] = { + [PCM_FRONT_CHANNEL] = SPI_DACD4_BIT, + [PCM_REAR_CHANNEL] = SPI_DACD0_BIT, + [PCM_CENTER_LFE_CHANNEL]= SPI_DACD2_BIT, + [PCM_UNKNOWN_CHANNEL] = SPI_DACD1_BIT, +}; + /* open_playback callback */ static int snd_ca0106_pcm_open_playback_channel(struct snd_pcm_substream *substream, int channel_id) @@ -462,6 +508,17 @@ return err; if ((err = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64)) < 0) return err; + snd_pcm_set_sync(substream); + + if (chip->details->spi_dac && channel_id != PCM_FRONT_CHANNEL) { + const int reg = spi_dacd_reg[channel_id]; + + /* Power up dac */ + chip->spi_dac_reg[reg] &= ~spi_dacd_bit[channel_id]; + err = snd_ca0106_spi_write(chip, chip->spi_dac_reg[reg]); + if (err < 0) + return err; + } return 0; } @@ -472,6 +529,14 @@ struct snd_pcm_runtime *runtime = substream->runtime; struct snd_ca0106_pcm *epcm = runtime->private_data; chip->playback_channels[epcm->channel_id].use = 0; + + if (chip->details->spi_dac && epcm->channel_id != PCM_FRONT_CHANNEL) { + const int reg = spi_dacd_reg[epcm->channel_id]; + + /* Power down DAC */ + chip->spi_dac_reg[reg] |= spi_dacd_bit[epcm->channel_id]; + snd_ca0106_spi_write(chip, chip->spi_dac_reg[reg]); + } /* FIXME: maybe zero others */ return 0; } @@ -790,6 +855,9 @@ break; } snd_pcm_group_for_each_entry(s, substream) { + if (snd_pcm_substream_chip(s) != emu || + s->stream != SNDRV_PCM_STREAM_PLAYBACK) + continue; runtime = s->runtime; epcm = runtime->private_data; channel = epcm->channel_id; @@ -1195,28 +1263,23 @@ return 0; } +#define SPI_REG(reg, value) (((reg) << SPI_REG_SHIFT) | (value)) static unsigned int spi_dac_init[] = { - 0x00ff, - 0x02ff, - 0x0400, - 0x0520, - 0x0620, /* Set 24 bit. Was 0x0600 */ - 0x08ff, - 0x0aff, - 0x0cff, - 0x0eff, - 0x10ff, - 0x1200, - 0x1400, - 0x1480, - 0x1800, - 0x1aff, - 0x1cff, - 0x1e00, - 0x0530, - 0x0602, - 0x0622, - 0x1400, + SPI_REG(SPI_LDA1_REG, SPI_DA_BIT_0dB), /* 0dB dig. attenuation */ + SPI_REG(SPI_RDA1_REG, SPI_DA_BIT_0dB), + SPI_REG(SPI_PL_REG, SPI_PL_BIT_L_L | SPI_PL_BIT_R_R | SPI_IZD_BIT), + SPI_REG(SPI_FMT_REG, SPI_FMT_BIT_I2S | SPI_IWL_BIT_24), + SPI_REG(SPI_LDA2_REG, SPI_DA_BIT_0dB), + SPI_REG(SPI_RDA2_REG, SPI_DA_BIT_0dB), + SPI_REG(SPI_LDA3_REG, SPI_DA_BIT_0dB), + SPI_REG(SPI_RDA3_REG, SPI_DA_BIT_0dB), + SPI_REG(SPI_MASTDA_REG, SPI_DA_BIT_0dB), + SPI_REG(9, 0x00), + SPI_REG(SPI_MS_REG, SPI_DACD0_BIT | SPI_DACD1_BIT | SPI_DACD2_BIT), + SPI_REG(12, 0x00), + SPI_REG(SPI_LDA4_REG, SPI_DA_BIT_0dB), + SPI_REG(SPI_RDA4_REG, SPI_DA_BIT_0dB | SPI_DA_BIT_UPDATE), + SPI_REG(SPI_DACD4_REG, 0x00), }; static unsigned int i2c_adc_init[][2] = { @@ -1457,8 +1520,13 @@ int size, n; size = ARRAY_SIZE(spi_dac_init); - for (n=0; n < size; n++) + for (n = 0; n < size; n++) { + int reg = spi_dac_init[n] >> SPI_REG_SHIFT; + snd_ca0106_spi_write(chip, spi_dac_init[n]); + if (reg < ARRAY_SIZE(chip->spi_dac_reg)) + chip->spi_dac_reg[reg] = spi_dac_init[n]; + } } if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, diff -ruN linux-2.6.22/sound/pci/ca0106/ca0106_main.c.orig linux-2.6.22-alsa/sound/pci/ca0106/ca0106_main.c.orig --- linux-2.6.22/sound/pci/ca0106/ca0106_main.c.orig 1970-01-01 01:00:00.000000000 +0100 +++ linux-2.6.22-alsa/sound/pci/ca0106/ca0106_main.c.orig 2007-09-01 20:55:23.000000000 +0200 @@ -0,0 +1,1732 @@ +/* + * Copyright (c) 2004 James Courtier-Dutton + * Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit + * Version: 0.0.25 + * + * FEATURES currently supported: + * Front, Rear and Center/LFE. + * Surround40 and Surround51. + * Capture from MIC an LINE IN input. + * SPDIF digital playback of PCM stereo and AC3/DTS works. + * (One can use a standard mono mini-jack to one RCA plugs cable. + * or one can use a standard stereo mini-jack to two RCA plugs cable. + * Plug one of the RCA plugs into the Coax input of the external decoder/receiver.) + * ( In theory one could output 3 different AC3 streams at once, to 3 different SPDIF outputs. ) + * Notes on how to capture sound: + * The AC97 is used in the PLAYBACK direction. + * The output from the AC97 chip, instead of reaching the speakers, is fed into the Philips 1361T ADC. + * So, to record from the MIC, set the MIC Playback volume to max, + * unmute the MIC and turn up the MASTER Playback volume. + * So, to prevent feedback when capturing, minimise the "Capture feedback into Playback" volume. + * + * The only playback controls that currently do anything are: - + * Analog Front + * Analog Rear + * Analog Center/LFE + * SPDIF Front + * SPDIF Rear + * SPDIF Center/LFE + * + * For capture from Mic in or Line in. + * Digital/Analog ( switch must be in Analog mode for CAPTURE. ) + * + * CAPTURE feedback into PLAYBACK + * + * Changelog: + * Support interrupts per period. + * Removed noise from Center/LFE channel when in Analog mode. + * Rename and remove mixer controls. + * 0.0.6 + * Use separate card based DMA buffer for periods table list. + * 0.0.7 + * Change remove and rename ctrls into lists. + * 0.0.8 + * Try to fix capture sources. + * 0.0.9 + * Fix AC3 output. + * Enable S32_LE format support. + * 0.0.10 + * Enable playback 48000 and 96000 rates. (Rates other that these do not work, even with "plug:front".) + * 0.0.11 + * Add Model name recognition. + * 0.0.12 + * Correct interrupt timing. interrupt at end of period, instead of in the middle of a playback period. + * Remove redundent "voice" handling. + * 0.0.13 + * Single trigger call for multi channels. + * 0.0.14 + * Set limits based on what the sound card hardware can do. + * playback periods_min=2, periods_max=8 + * capture hw constraints require period_size = n * 64 bytes. + * playback hw constraints require period_size = n * 64 bytes. + * 0.0.15 + * Minor updates. + * 0.0.16 + * Implement 192000 sample rate. + * 0.0.17 + * Add support for SB0410 and SB0413. + * 0.0.18 + * Modified Copyright message. + * 0.0.19 + * Finally fix support for SB Live 24 bit. SB0410 and SB0413. + * The output codec needs resetting, otherwise all output is muted. + * 0.0.20 + * Merge "pci_disable_device(pci);" fixes. + * 0.0.21 + * Add 4 capture channels. (SPDIF only comes in on channel 0. ) + * Add SPDIF capture using optional digital I/O module for SB Live 24bit. (Analog capture does not yet work.) + * 0.0.22 + * Add support for MSI K8N Diamond Motherboard with onboard SB Live 24bit without AC97. From kiksen, bug #901 + * 0.0.23 + * Implement support for Line-in capture on SB Live 24bit. + * 0.0.24 + * Add support for mute control on SB Live 24bit (cards w/ SPI DAC) + * 0.0.25 + * Powerdown SPI DAC channels when not in use + * + * BUGS: + * Some stability problems when unloading the snd-ca0106 kernel module. + * -- + * + * TODO: + * 4 Capture channels, only one implemented so far. + * Other capture rates apart from 48khz not implemented. + * MIDI + * -- + * GENERAL INFO: + * Model: SB0310 + * P17 Chip: CA0106-DAT + * AC97 Codec: STAC 9721 + * ADC: Philips 1361T (Stereo 24bit) + * DAC: WM8746EDS (6-channel, 24bit, 192Khz) + * + * GENERAL INFO: + * Model: SB0410 + * P17 Chip: CA0106-DAT + * AC97 Codec: None + * ADC: WM8775EDS (4 Channel) + * DAC: CS4382 (114 dB, 24-Bit, 192 kHz, 8-Channel D/A Converter with DSD Support) + * SPDIF Out control switches between Mic in and SPDIF out. + * No sound out or mic input working yet. + * + * GENERAL INFO: + * Model: SB0413 + * P17 Chip: CA0106-DAT + * AC97 Codec: None. + * ADC: Unknown + * DAC: Unknown + * Trying to handle it like the SB0410. + * + * This code was initally based on code from ALSA's emu10k1x.c which is: + * Copyright (c) by Francisco Moraes + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +MODULE_AUTHOR("James Courtier-Dutton "); +MODULE_DESCRIPTION("CA0106"); +MODULE_LICENSE("GPL"); +MODULE_SUPPORTED_DEVICE("{{Creative,SB CA0106 chip}}"); + +// module parameters (see "Module Parameters") +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; +static uint subsystem[SNDRV_CARDS]; /* Force card subsystem model */ + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for the CA0106 soundcard."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for the CA0106 soundcard."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable the CA0106 soundcard."); +module_param_array(subsystem, uint, NULL, 0444); +MODULE_PARM_DESC(subsystem, "Force card subsystem model."); + +#include "ca0106.h" + +static struct snd_ca0106_details ca0106_chip_details[] = { + /* Sound Blaster X-Fi Extreme Audio. This does not have an AC97. 53SB079000000 */ + /* It is really just a normal SB Live 24bit. */ + /* Tested: + * See ALSA bug#3251 + */ + { .serial = 0x10131102, + .name = "X-Fi Extreme Audio [SBxxxx]", + .gpio_type = 1, + .i2c_adc = 1 } , + /* Sound Blaster X-Fi Extreme Audio. This does not have an AC97. 53SB079000000 */ + /* It is really just a normal SB Live 24bit. */ + /* + * CTRL:CA0111-WTLF + * ADC: WM8775SEDS + * DAC: CS4382-KQZ + */ + /* Tested: + * Playback on front, rear, center/lfe speakers + * Capture from Mic in. + * Not-Tested: + * Capture from Line in. + * Playback to digital out. + */ + { .serial = 0x10121102, + .name = "X-Fi Extreme Audio [SB0790]", + .gpio_type = 1, + .i2c_adc = 1 } , + /* New Dell Sound Blaster Live! 7.1 24bit. This does not have an AC97. */ + /* AudigyLS[SB0310] */ + { .serial = 0x10021102, + .name = "AudigyLS [SB0310]", + .ac97 = 1 } , + /* Unknown AudigyLS that also says SB0310 on it */ + { .serial = 0x10051102, + .name = "AudigyLS [SB0310b]", + .ac97 = 1 } , + /* New Sound Blaster Live! 7.1 24bit. This does not have an AC97. 53SB041000001 */ + { .serial = 0x10061102, + .name = "Live! 7.1 24bit [SB0410]", + .gpio_type = 1, + .i2c_adc = 1 } , + /* New Dell Sound Blaster Live! 7.1 24bit. This does not have an AC97. */ + { .serial = 0x10071102, + .name = "Live! 7.1 24bit [SB0413]", + .gpio_type = 1, + .i2c_adc = 1 } , + /* New Audigy SE. Has a different DAC. */ + /* SB0570: + * CTRL:CA0106-DAT + * ADC: WM8775EDS + * DAC: WM8768GEDS + */ + { .serial = 0x100a1102, + .name = "Audigy SE [SB0570]", + .gpio_type = 1, + .i2c_adc = 1, + .spi_dac = 1 } , + /* New Audigy LS. Has a different DAC. */ + /* SB0570: + * CTRL:CA0106-DAT + * ADC: WM8775EDS + * DAC: WM8768GEDS + */ + { .serial = 0x10111102, + .name = "Audigy SE OEM [SB0570a]", + .gpio_type = 1, + .i2c_adc = 1, + .spi_dac = 1 } , + /* MSI K8N Diamond Motherboard with onboard SB Live 24bit without AC97 */ + /* SB0438 + * CTRL:CA0106-DAT + * ADC: WM8775SEDS + * DAC: CS4382-KQZ + */ + { .serial = 0x10091462, + .name = "MSI K8N Diamond MB [SB0438]", + .gpio_type = 2, + .i2c_adc = 1 } , + /* Shuttle XPC SD31P which has an onboard Creative Labs + * Sound Blaster Live! 24-bit EAX + * high-definition 7.1 audio processor". + * Added using info from andrewvegan in alsa bug #1298 + */ + { .serial = 0x30381297, + .name = "Shuttle XPC SD31P [SD31P]", + .gpio_type = 1, + .i2c_adc = 1 } , + /* Shuttle XPC SD11G5 which has an onboard Creative Labs + * Sound Blaster Live! 24-bit EAX + * high-definition 7.1 audio processor". + * Fixes ALSA bug#1600 + */ + { .serial = 0x30411297, + .name = "Shuttle XPC SD11G5 [SD11G5]", + .gpio_type = 1, + .i2c_adc = 1 } , + { .serial = 0, + .name = "AudigyLS [Unknown]" } +}; + +/* hardware definition */ +static struct snd_pcm_hardware snd_ca0106_playback_hw = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_SYNC_START, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, + .rates = (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_192000), + .rate_min = 48000, + .rate_max = 192000, + .channels_min = 2, //1, + .channels_max = 2, //6, + .buffer_bytes_max = ((65536 - 64) * 8), + .period_bytes_min = 64, + .period_bytes_max = (65536 - 64), + .periods_min = 2, + .periods_max = 8, + .fifo_size = 0, +}; + +static struct snd_pcm_hardware snd_ca0106_capture_hw = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID), + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, + .rates = (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000), + .rate_min = 44100, + .rate_max = 192000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = ((65536 - 64) * 8), + .period_bytes_min = 64, + .period_bytes_max = (65536 - 64), + .periods_min = 2, + .periods_max = 2, + .fifo_size = 0, +}; + +unsigned int snd_ca0106_ptr_read(struct snd_ca0106 * emu, + unsigned int reg, + unsigned int chn) +{ + unsigned long flags; + unsigned int regptr, val; + + regptr = (reg << 16) | chn; + + spin_lock_irqsave(&emu->emu_lock, flags); + outl(regptr, emu->port + PTR); + val = inl(emu->port + DATA); + spin_unlock_irqrestore(&emu->emu_lock, flags); + return val; +} + +void snd_ca0106_ptr_write(struct snd_ca0106 *emu, + unsigned int reg, + unsigned int chn, + unsigned int data) +{ + unsigned int regptr; + unsigned long flags; + + regptr = (reg << 16) | chn; + + spin_lock_irqsave(&emu->emu_lock, flags); + outl(regptr, emu->port + PTR); + outl(data, emu->port + DATA); + spin_unlock_irqrestore(&emu->emu_lock, flags); +} + +int snd_ca0106_spi_write(struct snd_ca0106 * emu, + unsigned int data) +{ + unsigned int reset, set; + unsigned int reg, tmp; + int n, result; + reg = SPI; + if (data > 0xffff) /* Only 16bit values allowed */ + return 1; + tmp = snd_ca0106_ptr_read(emu, reg, 0); + reset = (tmp & ~0x3ffff) | 0x20000; /* Set xxx20000 */ + set = reset | 0x10000; /* Set xxx1xxxx */ + snd_ca0106_ptr_write(emu, reg, 0, reset | data); + tmp = snd_ca0106_ptr_read(emu, reg, 0); /* write post */ + snd_ca0106_ptr_write(emu, reg, 0, set | data); + result = 1; + /* Wait for status bit to return to 0 */ + for (n = 0; n < 100; n++) { + udelay(10); + tmp = snd_ca0106_ptr_read(emu, reg, 0); + if (!(tmp & 0x10000)) { + result = 0; + break; + } + } + if (result) /* Timed out */ + return 1; + snd_ca0106_ptr_write(emu, reg, 0, reset | data); + tmp = snd_ca0106_ptr_read(emu, reg, 0); /* Write post */ + return 0; +} + +/* The ADC does not support i2c read, so only write is implemented */ +int snd_ca0106_i2c_write(struct snd_ca0106 *emu, + u32 reg, + u32 value) +{ + u32 tmp; + int timeout = 0; + int status; + int retry; + if ((reg > 0x7f) || (value > 0x1ff)) { + snd_printk(KERN_ERR "i2c_write: invalid values.\n"); + return -EINVAL; + } + + tmp = reg << 25 | value << 16; + // snd_printk("I2C-write:reg=0x%x, value=0x%x\n", reg, value); + /* Not sure what this I2C channel controls. */ + /* snd_ca0106_ptr_write(emu, I2C_D0, 0, tmp); */ + + /* This controls the I2C connected to the WM8775 ADC Codec */ + snd_ca0106_ptr_write(emu, I2C_D1, 0, tmp); + + for (retry = 0; retry < 10; retry++) { + /* Send the data to i2c */ + //tmp = snd_ca0106_ptr_read(emu, I2C_A, 0); + //tmp = tmp & ~(I2C_A_ADC_READ|I2C_A_ADC_LAST|I2C_A_ADC_START|I2C_A_ADC_ADD_MASK); + tmp = 0; + tmp = tmp | (I2C_A_ADC_LAST|I2C_A_ADC_START|I2C_A_ADC_ADD); + snd_ca0106_ptr_write(emu, I2C_A, 0, tmp); + + /* Wait till the transaction ends */ + while (1) { + status = snd_ca0106_ptr_read(emu, I2C_A, 0); + //snd_printk("I2C:status=0x%x\n", status); + timeout++; + if ((status & I2C_A_ADC_START) == 0) + break; + + if (timeout > 1000) + break; + } + //Read back and see if the transaction is successful + if ((status & I2C_A_ADC_ABORT) == 0) + break; + } + + if (retry == 10) { + snd_printk(KERN_ERR "Writing to ADC failed!\n"); + return -EINVAL; + } + + return 0; +} + + +static void snd_ca0106_intr_enable(struct snd_ca0106 *emu, unsigned int intrenb) +{ + unsigned long flags; + unsigned int enable; + + spin_lock_irqsave(&emu->emu_lock, flags); + enable = inl(emu->port + INTE) | intrenb; + outl(enable, emu->port + INTE); + spin_unlock_irqrestore(&emu->emu_lock, flags); +} + +static void snd_ca0106_intr_disable(struct snd_ca0106 *emu, unsigned int intrenb) +{ + unsigned long flags; + unsigned int enable; + + spin_lock_irqsave(&emu->emu_lock, flags); + enable = inl(emu->port + INTE) & ~intrenb; + outl(enable, emu->port + INTE); + spin_unlock_irqrestore(&emu->emu_lock, flags); +} + + +static void snd_ca0106_pcm_free_substream(struct snd_pcm_runtime *runtime) +{ + kfree(runtime->private_data); +} + +static const int spi_dacd_reg[] = { + [PCM_FRONT_CHANNEL] = SPI_DACD4_REG, + [PCM_REAR_CHANNEL] = SPI_DACD0_REG, + [PCM_CENTER_LFE_CHANNEL]= SPI_DACD2_REG, + [PCM_UNKNOWN_CHANNEL] = SPI_DACD1_REG, +}; +static const int spi_dacd_bit[] = { + [PCM_FRONT_CHANNEL] = SPI_DACD4_BIT, + [PCM_REAR_CHANNEL] = SPI_DACD0_BIT, + [PCM_CENTER_LFE_CHANNEL]= SPI_DACD2_BIT, + [PCM_UNKNOWN_CHANNEL] = SPI_DACD1_BIT, +}; + +/* open_playback callback */ +static int snd_ca0106_pcm_open_playback_channel(struct snd_pcm_substream *substream, + int channel_id) +{ + struct snd_ca0106 *chip = snd_pcm_substream_chip(substream); + struct snd_ca0106_channel *channel = &(chip->playback_channels[channel_id]); + struct snd_ca0106_pcm *epcm; + struct snd_pcm_runtime *runtime = substream->runtime; + int err; + + epcm = kzalloc(sizeof(*epcm), GFP_KERNEL); + + if (epcm == NULL) + return -ENOMEM; + epcm->emu = chip; + epcm->substream = substream; + epcm->channel_id=channel_id; + + runtime->private_data = epcm; + runtime->private_free = snd_ca0106_pcm_free_substream; + + runtime->hw = snd_ca0106_playback_hw; + + channel->emu = chip; + channel->number = channel_id; + + channel->use = 1; + //printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel); + //channel->interrupt = snd_ca0106_pcm_channel_interrupt; + channel->epcm = epcm; + if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) + return err; + if ((err = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64)) < 0) + return err; + snd_pcm_set_sync(substream); + + if (chip->details->spi_dac && channel_id != PCM_FRONT_CHANNEL) { + const int reg = spi_dacd_reg[channel_id]; + + /* Power up dac */ + chip->spi_dac_reg[reg] &= ~spi_dacd_bit[channel_id]; + err = snd_ca0106_spi_write(chip, chip->spi_dac_reg[reg]); + if (err < 0) + return err; + } + return 0; +} + +/* close callback */ +static int snd_ca0106_pcm_close_playback(struct snd_pcm_substream *substream) +{ + struct snd_ca0106 *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_ca0106_pcm *epcm = runtime->private_data; + chip->playback_channels[epcm->channel_id].use = 0; + + if (chip->details->spi_dac && epcm->channel_id != PCM_FRONT_CHANNEL) { + const int reg = spi_dacd_reg[epcm->channel_id]; + + /* Power down DAC */ + chip->spi_dac_reg[reg] |= spi_dacd_bit[epcm->channel_id]; + snd_ca0106_spi_write(chip, chip->spi_dac_reg[reg]); + } + /* FIXME: maybe zero others */ + return 0; +} + +static int snd_ca0106_pcm_open_playback_front(struct snd_pcm_substream *substream) +{ + return snd_ca0106_pcm_open_playback_channel(substream, PCM_FRONT_CHANNEL); +} + +static int snd_ca0106_pcm_open_playback_center_lfe(struct snd_pcm_substream *substream) +{ + return snd_ca0106_pcm_open_playback_channel(substream, PCM_CENTER_LFE_CHANNEL); +} + +static int snd_ca0106_pcm_open_playback_unknown(struct snd_pcm_substream *substream) +{ + return snd_ca0106_pcm_open_playback_channel(substream, PCM_UNKNOWN_CHANNEL); +} + +static int snd_ca0106_pcm_open_playback_rear(struct snd_pcm_substream *substream) +{ + return snd_ca0106_pcm_open_playback_channel(substream, PCM_REAR_CHANNEL); +} + +/* open_capture callback */ +static int snd_ca0106_pcm_open_capture_channel(struct snd_pcm_substream *substream, + int channel_id) +{ + struct snd_ca0106 *chip = snd_pcm_substream_chip(substream); + struct snd_ca0106_channel *channel = &(chip->capture_channels[channel_id]); + struct snd_ca0106_pcm *epcm; + struct snd_pcm_runtime *runtime = substream->runtime; + int err; + + epcm = kzalloc(sizeof(*epcm), GFP_KERNEL); + if (epcm == NULL) { + snd_printk(KERN_ERR "open_capture_channel: failed epcm alloc\n"); + return -ENOMEM; + } + epcm->emu = chip; + epcm->substream = substream; + epcm->channel_id=channel_id; + + runtime->private_data = epcm; + runtime->private_free = snd_ca0106_pcm_free_substream; + + runtime->hw = snd_ca0106_capture_hw; + + channel->emu = chip; + channel->number = channel_id; + + channel->use = 1; + //printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel); + //channel->interrupt = snd_ca0106_pcm_channel_interrupt; + channel->epcm = epcm; + if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) + return err; + //snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, &hw_constraints_capture_period_sizes); + if ((err = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64)) < 0) + return err; + return 0; +} + +/* close callback */ +static int snd_ca0106_pcm_close_capture(struct snd_pcm_substream *substream) +{ + struct snd_ca0106 *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_ca0106_pcm *epcm = runtime->private_data; + chip->capture_channels[epcm->channel_id].use = 0; + /* FIXME: maybe zero others */ + return 0; +} + +static int snd_ca0106_pcm_open_0_capture(struct snd_pcm_substream *substream) +{ + return snd_ca0106_pcm_open_capture_channel(substream, 0); +} + +static int snd_ca0106_pcm_open_1_capture(struct snd_pcm_substream *substream) +{ + return snd_ca0106_pcm_open_capture_channel(substream, 1); +} + +static int snd_ca0106_pcm_open_2_capture(struct snd_pcm_substream *substream) +{ + return snd_ca0106_pcm_open_capture_channel(substream, 2); +} + +static int snd_ca0106_pcm_open_3_capture(struct snd_pcm_substream *substream) +{ + return snd_ca0106_pcm_open_capture_channel(substream, 3); +} + +/* hw_params callback */ +static int snd_ca0106_pcm_hw_params_playback(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); +} + +/* hw_free callback */ +static int snd_ca0106_pcm_hw_free_playback(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +/* hw_params callback */ +static int snd_ca0106_pcm_hw_params_capture(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); +} + +/* hw_free callback */ +static int snd_ca0106_pcm_hw_free_capture(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +/* prepare playback callback */ +static int snd_ca0106_pcm_prepare_playback(struct snd_pcm_substream *substream) +{ + struct snd_ca0106 *emu = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_ca0106_pcm *epcm = runtime->private_data; + int channel = epcm->channel_id; + u32 *table_base = (u32 *)(emu->buffer.area+(8*16*channel)); + u32 period_size_bytes = frames_to_bytes(runtime, runtime->period_size); + u32 hcfg_mask = HCFG_PLAYBACK_S32_LE; + u32 hcfg_set = 0x00000000; + u32 hcfg; + u32 reg40_mask = 0x30000 << (channel<<1); + u32 reg40_set = 0; + u32 reg40; + /* FIXME: Depending on mixer selection of SPDIF out or not, select the spdif rate or the DAC rate. */ + u32 reg71_mask = 0x03030000 ; /* Global. Set SPDIF rate. We only support 44100 to spdif, not to DAC. */ + u32 reg71_set = 0; + u32 reg71; + int i; + + //snd_printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, periods=%u, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, runtime->periods, frames_to_bytes(runtime, 1)); + //snd_printk("dma_addr=%x, dma_area=%p, table_base=%p\n",runtime->dma_addr, runtime->dma_area, table_base); + //snd_printk("dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",emu->buffer.addr, emu->buffer.area, emu->buffer.bytes); + /* Rate can be set per channel. */ + /* reg40 control host to fifo */ + /* reg71 controls DAC rate. */ + switch (runtime->rate) { + case 44100: + reg40_set = 0x10000 << (channel<<1); + reg71_set = 0x01010000; + break; + case 48000: + reg40_set = 0; + reg71_set = 0; + break; + case 96000: + reg40_set = 0x20000 << (channel<<1); + reg71_set = 0x02020000; + break; + case 192000: + reg40_set = 0x30000 << (channel<<1); + reg71_set = 0x03030000; + break; + default: + reg40_set = 0; + reg71_set = 0; + break; + } + /* Format is a global setting */ + /* FIXME: Only let the first channel accessed set this. */ + switch (runtime->format) { + case SNDRV_PCM_FORMAT_S16_LE: + hcfg_set = 0; + break; + case SNDRV_PCM_FORMAT_S32_LE: + hcfg_set = HCFG_PLAYBACK_S32_LE; + break; + default: + hcfg_set = 0; + break; + } + hcfg = inl(emu->port + HCFG) ; + hcfg = (hcfg & ~hcfg_mask) | hcfg_set; + outl(hcfg, emu->port + HCFG); + reg40 = snd_ca0106_ptr_read(emu, 0x40, 0); + reg40 = (reg40 & ~reg40_mask) | reg40_set; + snd_ca0106_ptr_write(emu, 0x40, 0, reg40); + reg71 = snd_ca0106_ptr_read(emu, 0x71, 0); + reg71 = (reg71 & ~reg71_mask) | reg71_set; + snd_ca0106_ptr_write(emu, 0x71, 0, reg71); + + /* FIXME: Check emu->buffer.size before actually writing to it. */ + for(i=0; i < runtime->periods; i++) { + table_base[i*2] = runtime->dma_addr + (i * period_size_bytes); + table_base[i*2+1] = period_size_bytes << 16; + } + + snd_ca0106_ptr_write(emu, PLAYBACK_LIST_ADDR, channel, emu->buffer.addr+(8*16*channel)); + snd_ca0106_ptr_write(emu, PLAYBACK_LIST_SIZE, channel, (runtime->periods - 1) << 19); + snd_ca0106_ptr_write(emu, PLAYBACK_LIST_PTR, channel, 0); + snd_ca0106_ptr_write(emu, PLAYBACK_DMA_ADDR, channel, runtime->dma_addr); + snd_ca0106_ptr_write(emu, PLAYBACK_PERIOD_SIZE, channel, frames_to_bytes(runtime, runtime->period_size)<<16); // buffer size in bytes + /* FIXME test what 0 bytes does. */ + snd_ca0106_ptr_write(emu, PLAYBACK_PERIOD_SIZE, channel, 0); // buffer size in bytes + snd_ca0106_ptr_write(emu, PLAYBACK_POINTER, channel, 0); + snd_ca0106_ptr_write(emu, 0x07, channel, 0x0); + snd_ca0106_ptr_write(emu, 0x08, channel, 0); + snd_ca0106_ptr_write(emu, PLAYBACK_MUTE, 0x0, 0x0); /* Unmute output */ +#if 0 + snd_ca0106_ptr_write(emu, SPCS0, 0, + SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 | + SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC | + SPCS_GENERATIONSTATUS | 0x00001200 | + 0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT ); + } +#endif + + return 0; +} + +/* prepare capture callback */ +static int snd_ca0106_pcm_prepare_capture(struct snd_pcm_substream *substream) +{ + struct snd_ca0106 *emu = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_ca0106_pcm *epcm = runtime->private_data; + int channel = epcm->channel_id; + u32 hcfg_mask = HCFG_CAPTURE_S32_LE; + u32 hcfg_set = 0x00000000; + u32 hcfg; + u32 over_sampling=0x2; + u32 reg71_mask = 0x0000c000 ; /* Global. Set ADC rate. */ + u32 reg71_set = 0; + u32 reg71; + + //snd_printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, periods=%u, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, runtime->periods, frames_to_bytes(runtime, 1)); + //snd_printk("dma_addr=%x, dma_area=%p, table_base=%p\n",runtime->dma_addr, runtime->dma_area, table_base); + //snd_printk("dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",emu->buffer.addr, emu->buffer.area, emu->buffer.bytes); + /* reg71 controls ADC rate. */ + switch (runtime->rate) { + case 44100: + reg71_set = 0x00004000; + break; + case 48000: + reg71_set = 0; + break; + case 96000: + reg71_set = 0x00008000; + over_sampling=0xa; + break; + case 192000: + reg71_set = 0x0000c000; + over_sampling=0xa; + break; + default: + reg71_set = 0; + break; + } + /* Format is a global setting */ + /* FIXME: Only let the first channel accessed set this. */ + switch (runtime->format) { + case SNDRV_PCM_FORMAT_S16_LE: + hcfg_set = 0; + break; + case SNDRV_PCM_FORMAT_S32_LE: + hcfg_set = HCFG_CAPTURE_S32_LE; + break; + default: + hcfg_set = 0; + break; + } + hcfg = inl(emu->port + HCFG) ; + hcfg = (hcfg & ~hcfg_mask) | hcfg_set; + outl(hcfg, emu->port + HCFG); + reg71 = snd_ca0106_ptr_read(emu, 0x71, 0); + reg71 = (reg71 & ~reg71_mask) | reg71_set; + snd_ca0106_ptr_write(emu, 0x71, 0, reg71); + if (emu->details->i2c_adc == 1) { /* The SB0410 and SB0413 use I2C to control ADC. */ + snd_ca0106_i2c_write(emu, ADC_MASTER, over_sampling); /* Adjust the over sampler to better suit the capture rate. */ + } + + + //printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, frames_to_bytes(runtime, 1)); + snd_ca0106_ptr_write(emu, 0x13, channel, 0); + snd_ca0106_ptr_write(emu, CAPTURE_DMA_ADDR, channel, runtime->dma_addr); + snd_ca0106_ptr_write(emu, CAPTURE_BUFFER_SIZE, channel, frames_to_bytes(runtime, runtime->buffer_size)<<16); // buffer size in bytes + snd_ca0106_ptr_write(emu, CAPTURE_POINTER, channel, 0); + + return 0; +} + +/* trigger_playback callback */ +static int snd_ca0106_pcm_trigger_playback(struct snd_pcm_substream *substream, + int cmd) +{ + struct snd_ca0106 *emu = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime; + struct snd_ca0106_pcm *epcm; + int channel; + int result = 0; + struct snd_pcm_substream *s; + u32 basic = 0; + u32 extended = 0; + int running=0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + running=1; + break; + case SNDRV_PCM_TRIGGER_STOP: + default: + running=0; + break; + } + snd_pcm_group_for_each_entry(s, substream) { + if (snd_pcm_substream_chip(s) != emu || + s->stream != SNDRV_PCM_STREAM_PLAYBACK) + continue; + runtime = s->runtime; + epcm = runtime->private_data; + channel = epcm->channel_id; + //snd_printk("channel=%d\n",channel); + epcm->running = running; + basic |= (0x1<runtime; + struct snd_ca0106_pcm *epcm = runtime->private_data; + int channel = epcm->channel_id; + int result = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + snd_ca0106_ptr_write(emu, EXTENDED_INT_MASK, 0, snd_ca0106_ptr_read(emu, EXTENDED_INT_MASK, 0) | (0x110000<running = 1; + break; + case SNDRV_PCM_TRIGGER_STOP: + snd_ca0106_ptr_write(emu, BASIC_INTERRUPT, 0, snd_ca0106_ptr_read(emu, BASIC_INTERRUPT, 0) & ~(0x100<running = 0; + break; + default: + result = -EINVAL; + break; + } + return result; +} + +/* pointer_playback callback */ +static snd_pcm_uframes_t +snd_ca0106_pcm_pointer_playback(struct snd_pcm_substream *substream) +{ + struct snd_ca0106 *emu = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_ca0106_pcm *epcm = runtime->private_data; + snd_pcm_uframes_t ptr, ptr1, ptr2,ptr3,ptr4 = 0; + int channel = epcm->channel_id; + + if (!epcm->running) + return 0; + + ptr3 = snd_ca0106_ptr_read(emu, PLAYBACK_LIST_PTR, channel); + ptr1 = snd_ca0106_ptr_read(emu, PLAYBACK_POINTER, channel); + ptr4 = snd_ca0106_ptr_read(emu, PLAYBACK_LIST_PTR, channel); + if (ptr3 != ptr4) ptr1 = snd_ca0106_ptr_read(emu, PLAYBACK_POINTER, channel); + ptr2 = bytes_to_frames(runtime, ptr1); + ptr2+= (ptr4 >> 3) * runtime->period_size; + ptr=ptr2; + if (ptr >= runtime->buffer_size) + ptr -= runtime->buffer_size; + //printk("ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n", ptr1, ptr2, ptr, (int)runtime->buffer_size, (int)runtime->period_size, (int)runtime->frame_bits, (int)runtime->rate); + + return ptr; +} + +/* pointer_capture callback */ +static snd_pcm_uframes_t +snd_ca0106_pcm_pointer_capture(struct snd_pcm_substream *substream) +{ + struct snd_ca0106 *emu = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_ca0106_pcm *epcm = runtime->private_data; + snd_pcm_uframes_t ptr, ptr1, ptr2 = 0; + int channel = channel=epcm->channel_id; + + if (!epcm->running) + return 0; + + ptr1 = snd_ca0106_ptr_read(emu, CAPTURE_POINTER, channel); + ptr2 = bytes_to_frames(runtime, ptr1); + ptr=ptr2; + if (ptr >= runtime->buffer_size) + ptr -= runtime->buffer_size; + //printk("ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n", ptr1, ptr2, ptr, (int)runtime->buffer_size, (int)runtime->period_size, (int)runtime->frame_bits, (int)runtime->rate); + + return ptr; +} + +/* operators */ +static struct snd_pcm_ops snd_ca0106_playback_front_ops = { + .open = snd_ca0106_pcm_open_playback_front, + .close = snd_ca0106_pcm_close_playback, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_ca0106_pcm_hw_params_playback, + .hw_free = snd_ca0106_pcm_hw_free_playback, + .prepare = snd_ca0106_pcm_prepare_playback, + .trigger = snd_ca0106_pcm_trigger_playback, + .pointer = snd_ca0106_pcm_pointer_playback, +}; + +static struct snd_pcm_ops snd_ca0106_capture_0_ops = { + .open = snd_ca0106_pcm_open_0_capture, + .close = snd_ca0106_pcm_close_capture, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_ca0106_pcm_hw_params_capture, + .hw_free = snd_ca0106_pcm_hw_free_capture, + .prepare = snd_ca0106_pcm_prepare_capture, + .trigger = snd_ca0106_pcm_trigger_capture, + .pointer = snd_ca0106_pcm_pointer_capture, +}; + +static struct snd_pcm_ops snd_ca0106_capture_1_ops = { + .open = snd_ca0106_pcm_open_1_capture, + .close = snd_ca0106_pcm_close_capture, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_ca0106_pcm_hw_params_capture, + .hw_free = snd_ca0106_pcm_hw_free_capture, + .prepare = snd_ca0106_pcm_prepare_capture, + .trigger = snd_ca0106_pcm_trigger_capture, + .pointer = snd_ca0106_pcm_pointer_capture, +}; + +static struct snd_pcm_ops snd_ca0106_capture_2_ops = { + .open = snd_ca0106_pcm_open_2_capture, + .close = snd_ca0106_pcm_close_capture, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_ca0106_pcm_hw_params_capture, + .hw_free = snd_ca0106_pcm_hw_free_capture, + .prepare = snd_ca0106_pcm_prepare_capture, + .trigger = snd_ca0106_pcm_trigger_capture, + .pointer = snd_ca0106_pcm_pointer_capture, +}; + +static struct snd_pcm_ops snd_ca0106_capture_3_ops = { + .open = snd_ca0106_pcm_open_3_capture, + .close = snd_ca0106_pcm_close_capture, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_ca0106_pcm_hw_params_capture, + .hw_free = snd_ca0106_pcm_hw_free_capture, + .prepare = snd_ca0106_pcm_prepare_capture, + .trigger = snd_ca0106_pcm_trigger_capture, + .pointer = snd_ca0106_pcm_pointer_capture, +}; + +static struct snd_pcm_ops snd_ca0106_playback_center_lfe_ops = { + .open = snd_ca0106_pcm_open_playback_center_lfe, + .close = snd_ca0106_pcm_close_playback, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_ca0106_pcm_hw_params_playback, + .hw_free = snd_ca0106_pcm_hw_free_playback, + .prepare = snd_ca0106_pcm_prepare_playback, + .trigger = snd_ca0106_pcm_trigger_playback, + .pointer = snd_ca0106_pcm_pointer_playback, +}; + +static struct snd_pcm_ops snd_ca0106_playback_unknown_ops = { + .open = snd_ca0106_pcm_open_playback_unknown, + .close = snd_ca0106_pcm_close_playback, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_ca0106_pcm_hw_params_playback, + .hw_free = snd_ca0106_pcm_hw_free_playback, + .prepare = snd_ca0106_pcm_prepare_playback, + .trigger = snd_ca0106_pcm_trigger_playback, + .pointer = snd_ca0106_pcm_pointer_playback, +}; + +static struct snd_pcm_ops snd_ca0106_playback_rear_ops = { + .open = snd_ca0106_pcm_open_playback_rear, + .close = snd_ca0106_pcm_close_playback, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_ca0106_pcm_hw_params_playback, + .hw_free = snd_ca0106_pcm_hw_free_playback, + .prepare = snd_ca0106_pcm_prepare_playback, + .trigger = snd_ca0106_pcm_trigger_playback, + .pointer = snd_ca0106_pcm_pointer_playback, +}; + + +static unsigned short snd_ca0106_ac97_read(struct snd_ac97 *ac97, + unsigned short reg) +{ + struct snd_ca0106 *emu = ac97->private_data; + unsigned long flags; + unsigned short val; + + spin_lock_irqsave(&emu->emu_lock, flags); + outb(reg, emu->port + AC97ADDRESS); + val = inw(emu->port + AC97DATA); + spin_unlock_irqrestore(&emu->emu_lock, flags); + return val; +} + +static void snd_ca0106_ac97_write(struct snd_ac97 *ac97, + unsigned short reg, unsigned short val) +{ + struct snd_ca0106 *emu = ac97->private_data; + unsigned long flags; + + spin_lock_irqsave(&emu->emu_lock, flags); + outb(reg, emu->port + AC97ADDRESS); + outw(val, emu->port + AC97DATA); + spin_unlock_irqrestore(&emu->emu_lock, flags); +} + +static int snd_ca0106_ac97(struct snd_ca0106 *chip) +{ + struct snd_ac97_bus *pbus; + struct snd_ac97_template ac97; + int err; + static struct snd_ac97_bus_ops ops = { + .write = snd_ca0106_ac97_write, + .read = snd_ca0106_ac97_read, + }; + + if ((err = snd_ac97_bus(chip->card, 0, &ops, NULL, &pbus)) < 0) + return err; + pbus->no_vra = 1; /* we don't need VRA */ + + memset(&ac97, 0, sizeof(ac97)); + ac97.private_data = chip; + ac97.scaps = AC97_SCAP_NO_SPDIF; + return snd_ac97_mixer(pbus, &ac97, &chip->ac97); +} + +static int snd_ca0106_free(struct snd_ca0106 *chip) +{ + if (chip->res_port != NULL) { /* avoid access to already used hardware */ + // disable interrupts + snd_ca0106_ptr_write(chip, BASIC_INTERRUPT, 0, 0); + outl(0, chip->port + INTE); + snd_ca0106_ptr_write(chip, EXTENDED_INT_MASK, 0, 0); + udelay(1000); + // disable audio + //outl(HCFG_LOCKSOUNDCACHE, chip->port + HCFG); + outl(0, chip->port + HCFG); + /* FIXME: We need to stop and DMA transfers here. + * But as I am not sure how yet, we cannot from the dma pages. + * So we can fix: snd-malloc: Memory leak? pages not freed = 8 + */ + } + // release the data +#if 1 + if (chip->buffer.area) + snd_dma_free_pages(&chip->buffer); +#endif + + // release the i/o port + release_and_free_resource(chip->res_port); + + // release the irq + if (chip->irq >= 0) + free_irq(chip->irq, chip); + pci_disable_device(chip->pci); + kfree(chip); + return 0; +} + +static int snd_ca0106_dev_free(struct snd_device *device) +{ + struct snd_ca0106 *chip = device->device_data; + return snd_ca0106_free(chip); +} + +static irqreturn_t snd_ca0106_interrupt(int irq, void *dev_id) +{ + unsigned int status; + + struct snd_ca0106 *chip = dev_id; + int i; + int mask; + unsigned int stat76; + struct snd_ca0106_channel *pchannel; + + status = inl(chip->port + IPR); + if (! status) + return IRQ_NONE; + + stat76 = snd_ca0106_ptr_read(chip, EXTENDED_INT, 0); + //snd_printk("interrupt status = 0x%08x, stat76=0x%08x\n", status, stat76); + //snd_printk("ptr=0x%08x\n",snd_ca0106_ptr_read(chip, PLAYBACK_POINTER, 0)); + mask = 0x11; /* 0x1 for one half, 0x10 for the other half period. */ + for(i = 0; i < 4; i++) { + pchannel = &(chip->playback_channels[i]); + if (stat76 & mask) { +/* FIXME: Select the correct substream for period elapsed */ + if(pchannel->use) { + snd_pcm_period_elapsed(pchannel->epcm->substream); + //printk(KERN_INFO "interrupt [%d] used\n", i); + } + } + //printk(KERN_INFO "channel=%p\n",pchannel); + //printk(KERN_INFO "interrupt stat76[%d] = %08x, use=%d, channel=%d\n", i, stat76, pchannel->use, pchannel->number); + mask <<= 1; + } + mask = 0x110000; /* 0x1 for one half, 0x10 for the other half period. */ + for(i = 0; i < 4; i++) { + pchannel = &(chip->capture_channels[i]); + if (stat76 & mask) { +/* FIXME: Select the correct substream for period elapsed */ + if(pchannel->use) { + snd_pcm_period_elapsed(pchannel->epcm->substream); + //printk(KERN_INFO "interrupt [%d] used\n", i); + } + } + //printk(KERN_INFO "channel=%p\n",pchannel); + //printk(KERN_INFO "interrupt stat76[%d] = %08x, use=%d, channel=%d\n", i, stat76, pchannel->use, pchannel->number); + mask <<= 1; + } + + snd_ca0106_ptr_write(chip, EXTENDED_INT, 0, stat76); + + if (chip->midi.dev_id && + (status & (chip->midi.ipr_tx|chip->midi.ipr_rx))) { + if (chip->midi.interrupt) + chip->midi.interrupt(&chip->midi, status); + else + chip->midi.interrupt_disable(&chip->midi, chip->midi.tx_enable | chip->midi.rx_enable); + } + + // acknowledge the interrupt if necessary + outl(status, chip->port+IPR); + + return IRQ_HANDLED; +} + +static int __devinit snd_ca0106_pcm(struct snd_ca0106 *emu, int device, struct snd_pcm **rpcm) +{ + struct snd_pcm *pcm; + struct snd_pcm_substream *substream; + int err; + + if (rpcm) + *rpcm = NULL; + if ((err = snd_pcm_new(emu->card, "ca0106", device, 1, 1, &pcm)) < 0) + return err; + + pcm->private_data = emu; + + switch (device) { + case 0: + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_ca0106_playback_front_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_ca0106_capture_0_ops); + break; + case 1: + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_ca0106_playback_rear_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_ca0106_capture_1_ops); + break; + case 2: + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_ca0106_playback_center_lfe_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_ca0106_capture_2_ops); + break; + case 3: + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_ca0106_playback_unknown_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_ca0106_capture_3_ops); + break; + } + + pcm->info_flags = 0; + pcm->dev_subclass = SNDRV_PCM_SUBCLASS_GENERIC_MIX; + strcpy(pcm->name, "CA0106"); + emu->pcm = pcm; + + for(substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; + substream; + substream = substream->next) { + if ((err = snd_pcm_lib_preallocate_pages(substream, + SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(emu->pci), + 64*1024, 64*1024)) < 0) /* FIXME: 32*1024 for sound buffer, between 32and64 for Periods table. */ + return err; + } + + for (substream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; + substream; + substream = substream->next) { + if ((err = snd_pcm_lib_preallocate_pages(substream, + SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(emu->pci), + 64*1024, 64*1024)) < 0) + return err; + } + + if (rpcm) + *rpcm = pcm; + + return 0; +} + +#define SPI_REG(reg, value) (((reg) << SPI_REG_SHIFT) | (value)) +static unsigned int spi_dac_init[] = { + SPI_REG(SPI_LDA1_REG, SPI_DA_BIT_0dB), /* 0dB dig. attenuation */ + SPI_REG(SPI_RDA1_REG, SPI_DA_BIT_0dB), + SPI_REG(SPI_PL_REG, SPI_PL_BIT_L_L | SPI_PL_BIT_R_R | SPI_IZD_BIT), + SPI_REG(SPI_FMT_REG, SPI_FMT_BIT_I2S | SPI_IWL_BIT_24), + SPI_REG(SPI_LDA2_REG, SPI_DA_BIT_0dB), + SPI_REG(SPI_RDA2_REG, SPI_DA_BIT_0dB), + SPI_REG(SPI_LDA3_REG, SPI_DA_BIT_0dB), + SPI_REG(SPI_RDA3_REG, SPI_DA_BIT_0dB), + SPI_REG(SPI_MASTDA_REG, SPI_DA_BIT_0dB), + SPI_REG(9, 0x00), + SPI_REG(SPI_MS_REG, SPI_DACD0_BIT | SPI_DACD1_BIT | SPI_DACD2_BIT), + SPI_REG(12, 0x00), + SPI_REG(SPI_LDA4_REG, SPI_DA_BIT_0dB), + SPI_REG(SPI_RDA4_REG, SPI_DA_BIT_0dB | SPI_DA_BIT_UPDATE), + SPI_REG(SPI_DACD4_REG, 0x00), +}; + +static unsigned int i2c_adc_init[][2] = { + { 0x17, 0x00 }, /* Reset */ + { 0x07, 0x00 }, /* Timeout */ + { 0x0b, 0x22 }, /* Interface control */ + { 0x0c, 0x22 }, /* Master mode control */ + { 0x0d, 0x08 }, /* Powerdown control */ + { 0x0e, 0xcf }, /* Attenuation Left 0x01 = -103dB, 0xff = 24dB */ + { 0x0f, 0xcf }, /* Attenuation Right 0.5dB steps */ + { 0x10, 0x7b }, /* ALC Control 1 */ + { 0x11, 0x00 }, /* ALC Control 2 */ + { 0x12, 0x32 }, /* ALC Control 3 */ + { 0x13, 0x00 }, /* Noise gate control */ + { 0x14, 0xa6 }, /* Limiter control */ + { 0x15, ADC_MUX_LINEIN }, /* ADC Mixer control */ +}; + +static int __devinit snd_ca0106_create(int dev, struct snd_card *card, + struct pci_dev *pci, + struct snd_ca0106 **rchip) +{ + struct snd_ca0106 *chip; + struct snd_ca0106_details *c; + int err; + int ch; + static struct snd_device_ops ops = { + .dev_free = snd_ca0106_dev_free, + }; + + *rchip = NULL; + + if ((err = pci_enable_device(pci)) < 0) + return err; + if (pci_set_dma_mask(pci, DMA_32BIT_MASK) < 0 || + pci_set_consistent_dma_mask(pci, DMA_32BIT_MASK) < 0) { + printk(KERN_ERR "error to set 32bit mask DMA\n"); + pci_disable_device(pci); + return -ENXIO; + } + + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + if (chip == NULL) { + pci_disable_device(pci); + return -ENOMEM; + } + + chip->card = card; + chip->pci = pci; + chip->irq = -1; + + spin_lock_init(&chip->emu_lock); + + chip->port = pci_resource_start(pci, 0); + if ((chip->res_port = request_region(chip->port, 0x20, + "snd_ca0106")) == NULL) { + snd_ca0106_free(chip); + printk(KERN_ERR "cannot allocate the port\n"); + return -EBUSY; + } + + if (request_irq(pci->irq, snd_ca0106_interrupt, + IRQF_SHARED, "snd_ca0106", chip)) { + snd_ca0106_free(chip); + printk(KERN_ERR "cannot grab irq\n"); + return -EBUSY; + } + chip->irq = pci->irq; + + /* This stores the periods table. */ + if(snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), 1024, &chip->buffer) < 0) { + snd_ca0106_free(chip); + return -ENOMEM; + } + + pci_set_master(pci); + /* read serial */ + pci_read_config_dword(pci, PCI_SUBSYSTEM_VENDOR_ID, &chip->serial); + pci_read_config_word(pci, PCI_SUBSYSTEM_ID, &chip->model); +#if 1 + printk(KERN_INFO "snd-ca0106: Model %04x Rev %08x Serial %08x\n", chip->model, + pci->revision, chip->serial); +#endif + strcpy(card->driver, "CA0106"); + strcpy(card->shortname, "CA0106"); + + for (c = ca0106_chip_details; c->serial; c++) { + if (subsystem[dev]) { + if (c->serial == subsystem[dev]) + break; + } else if (c->serial == chip->serial) + break; + } + chip->details = c; + if (subsystem[dev]) { + printk(KERN_INFO "snd-ca0106: Sound card name=%s, subsystem=0x%x. Forced to subsystem=0x%x\n", + c->name, chip->serial, subsystem[dev]); + } + + sprintf(card->longname, "%s at 0x%lx irq %i", + c->name, chip->port, chip->irq); + + outl(0, chip->port + INTE); + + /* + * Init to 0x02109204 : + * Clock accuracy = 0 (1000ppm) + * Sample Rate = 2 (48kHz) + * Audio Channel = 1 (Left of 2) + * Source Number = 0 (Unspecified) + * Generation Status = 1 (Original for Cat Code 12) + * Cat Code = 12 (Digital Signal Mixer) + * Mode = 0 (Mode 0) + * Emphasis = 0 (None) + * CP = 1 (Copyright unasserted) + * AN = 0 (Audio data) + * P = 0 (Consumer) + */ + snd_ca0106_ptr_write(chip, SPCS0, 0, + chip->spdif_bits[0] = + SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 | + SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC | + SPCS_GENERATIONSTATUS | 0x00001200 | + 0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT); + /* Only SPCS1 has been tested */ + snd_ca0106_ptr_write(chip, SPCS1, 0, + chip->spdif_bits[1] = + SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 | + SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC | + SPCS_GENERATIONSTATUS | 0x00001200 | + 0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT); + snd_ca0106_ptr_write(chip, SPCS2, 0, + chip->spdif_bits[2] = + SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 | + SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC | + SPCS_GENERATIONSTATUS | 0x00001200 | + 0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT); + snd_ca0106_ptr_write(chip, SPCS3, 0, + chip->spdif_bits[3] = + SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 | + SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC | + SPCS_GENERATIONSTATUS | 0x00001200 | + 0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT); + + snd_ca0106_ptr_write(chip, PLAYBACK_MUTE, 0, 0x00fc0000); + snd_ca0106_ptr_write(chip, CAPTURE_MUTE, 0, 0x00fc0000); + + /* Write 0x8000 to AC97_REC_GAIN to mute it. */ + outb(AC97_REC_GAIN, chip->port + AC97ADDRESS); + outw(0x8000, chip->port + AC97DATA); +#if 0 + snd_ca0106_ptr_write(chip, SPCS0, 0, 0x2108006); + snd_ca0106_ptr_write(chip, 0x42, 0, 0x2108006); + snd_ca0106_ptr_write(chip, 0x43, 0, 0x2108006); + snd_ca0106_ptr_write(chip, 0x44, 0, 0x2108006); +#endif + + //snd_ca0106_ptr_write(chip, SPDIF_SELECT2, 0, 0xf0f003f); /* OSS drivers set this. */ + /* Analog or Digital output */ + snd_ca0106_ptr_write(chip, SPDIF_SELECT1, 0, 0xf); + snd_ca0106_ptr_write(chip, SPDIF_SELECT2, 0, 0x000f0000); /* 0x0b000000 for digital, 0x000b0000 for analog, from win2000 drivers. Use 0x000f0000 for surround71 */ + chip->spdif_enable = 0; /* Set digital SPDIF output off */ + //snd_ca0106_ptr_write(chip, 0x45, 0, 0); /* Analogue out */ + //snd_ca0106_ptr_write(chip, 0x45, 0, 0xf00); /* Digital out */ + + snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 0, 0x40c81000); /* goes to 0x40c80000 when doing SPDIF IN/OUT */ + snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 1, 0xffffffff); /* (Mute) CAPTURE feedback into PLAYBACK volume. Only lower 16 bits matter. */ + snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 2, 0x30300000); /* SPDIF IN Volume */ + snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 3, 0x00700000); /* SPDIF IN Volume, 0x70 = (vol & 0x3f) | 0x40 */ + snd_ca0106_ptr_write(chip, PLAYBACK_ROUTING1, 0, 0x32765410); + snd_ca0106_ptr_write(chip, PLAYBACK_ROUTING2, 0, 0x76767676); + snd_ca0106_ptr_write(chip, CAPTURE_ROUTING1, 0, 0x32765410); + snd_ca0106_ptr_write(chip, CAPTURE_ROUTING2, 0, 0x76767676); + for(ch = 0; ch < 4; ch++) { + snd_ca0106_ptr_write(chip, CAPTURE_VOLUME1, ch, 0x30303030); /* Only high 16 bits matter */ + snd_ca0106_ptr_write(chip, CAPTURE_VOLUME2, ch, 0x30303030); + //snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME1, ch, 0x40404040); /* Mute */ + //snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME2, ch, 0x40404040); /* Mute */ + snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME1, ch, 0xffffffff); /* Mute */ + snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME2, ch, 0xffffffff); /* Mute */ + } + if (chip->details->i2c_adc == 1) { + /* Select MIC, Line in, TAD in, AUX in */ + snd_ca0106_ptr_write(chip, CAPTURE_SOURCE, 0x0, 0x333300e4); + /* Default to CAPTURE_SOURCE to i2s in */ + chip->capture_source = 3; + } else if (chip->details->ac97 == 1) { + /* Default to AC97 in */ + snd_ca0106_ptr_write(chip, CAPTURE_SOURCE, 0x0, 0x444400e4); + /* Default to CAPTURE_SOURCE to AC97 in */ + chip->capture_source = 4; + } else { + /* Select MIC, Line in, TAD in, AUX in */ + snd_ca0106_ptr_write(chip, CAPTURE_SOURCE, 0x0, 0x333300e4); + /* Default to Set CAPTURE_SOURCE to i2s in */ + chip->capture_source = 3; + } + + if (chip->details->gpio_type == 2) { /* The SB0438 use GPIO differently. */ + /* FIXME: Still need to find out what the other GPIO bits do. E.g. For digital spdif out. */ + outl(0x0, chip->port+GPIO); + //outl(0x00f0e000, chip->port+GPIO); /* Analog */ + outl(0x005f5301, chip->port+GPIO); /* Analog */ + } else if (chip->details->gpio_type == 1) { /* The SB0410 and SB0413 use GPIO differently. */ + /* FIXME: Still need to find out what the other GPIO bits do. E.g. For digital spdif out. */ + outl(0x0, chip->port+GPIO); + //outl(0x00f0e000, chip->port+GPIO); /* Analog */ + outl(0x005f5301, chip->port+GPIO); /* Analog */ + } else { + outl(0x0, chip->port+GPIO); + outl(0x005f03a3, chip->port+GPIO); /* Analog */ + //outl(0x005f02a2, chip->port+GPIO); /* SPDIF */ + } + snd_ca0106_intr_enable(chip, 0x105); /* Win2000 uses 0x1e0 */ + + //outl(HCFG_LOCKSOUNDCACHE|HCFG_AUDIOENABLE, chip->port+HCFG); + //outl(0x00001409, chip->port+HCFG); /* 0x1000 causes AC3 to fails. Maybe it effects 24 bit output. */ + //outl(0x00000009, chip->port+HCFG); + outl(HCFG_AC97 | HCFG_AUDIOENABLE, chip->port+HCFG); /* AC97 2.0, Enable outputs. */ + + if (chip->details->i2c_adc == 1) { /* The SB0410 and SB0413 use I2C to control ADC. */ + int size, n; + + size = ARRAY_SIZE(i2c_adc_init); + //snd_printk("I2C:array size=0x%x\n", size); + for (n=0; n < size; n++) { + snd_ca0106_i2c_write(chip, i2c_adc_init[n][0], i2c_adc_init[n][1]); + } + for (n=0; n < 4; n++) { + chip->i2c_capture_volume[n][0]= 0xcf; + chip->i2c_capture_volume[n][1]= 0xcf; + } + chip->i2c_capture_source=2; /* Line in */ + //snd_ca0106_i2c_write(chip, ADC_MUX, ADC_MUX_LINEIN); /* Enable Line-in capture. MIC in currently untested. */ + } + if (chip->details->spi_dac == 1) { /* The SB0570 use SPI to control DAC. */ + int size, n; + + size = ARRAY_SIZE(spi_dac_init); + for (n = 0; n < size; n++) { + int reg = spi_dac_init[n] >> SPI_REG_SHIFT; + + snd_ca0106_spi_write(chip, spi_dac_init[n]); + if (reg < ARRAY_SIZE(chip->spi_dac_reg)) + chip->spi_dac_reg[reg] = spi_dac_init[n]; + } + } + + if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, + chip, &ops)) < 0) { + snd_ca0106_free(chip); + return err; + } + *rchip = chip; + return 0; +} + + +static void ca0106_midi_interrupt_enable(struct snd_ca_midi *midi, int intr) +{ + snd_ca0106_intr_enable((struct snd_ca0106 *)(midi->dev_id), intr); +} + +static void ca0106_midi_interrupt_disable(struct snd_ca_midi *midi, int intr) +{ + snd_ca0106_intr_disable((struct snd_ca0106 *)(midi->dev_id), intr); +} + +static unsigned char ca0106_midi_read(struct snd_ca_midi *midi, int idx) +{ + return (unsigned char)snd_ca0106_ptr_read((struct snd_ca0106 *)(midi->dev_id), + midi->port + idx, 0); +} + +static void ca0106_midi_write(struct snd_ca_midi *midi, int data, int idx) +{ + snd_ca0106_ptr_write((struct snd_ca0106 *)(midi->dev_id), midi->port + idx, 0, data); +} + +static struct snd_card *ca0106_dev_id_card(void *dev_id) +{ + return ((struct snd_ca0106 *)dev_id)->card; +} + +static int ca0106_dev_id_port(void *dev_id) +{ + return ((struct snd_ca0106 *)dev_id)->port; +} + +static int __devinit snd_ca0106_midi(struct snd_ca0106 *chip, unsigned int channel) +{ + struct snd_ca_midi *midi; + char *name; + int err; + + if (channel == CA0106_MIDI_CHAN_B) { + name = "CA0106 MPU-401 (UART) B"; + midi = &chip->midi2; + midi->tx_enable = INTE_MIDI_TX_B; + midi->rx_enable = INTE_MIDI_RX_B; + midi->ipr_tx = IPR_MIDI_TX_B; + midi->ipr_rx = IPR_MIDI_RX_B; + midi->port = MIDI_UART_B_DATA; + } else { + name = "CA0106 MPU-401 (UART)"; + midi = &chip->midi; + midi->tx_enable = INTE_MIDI_TX_A; + midi->rx_enable = INTE_MIDI_TX_B; + midi->ipr_tx = IPR_MIDI_TX_A; + midi->ipr_rx = IPR_MIDI_RX_A; + midi->port = MIDI_UART_A_DATA; + } + + midi->reset = CA0106_MPU401_RESET; + midi->enter_uart = CA0106_MPU401_ENTER_UART; + midi->ack = CA0106_MPU401_ACK; + + midi->input_avail = CA0106_MIDI_INPUT_AVAIL; + midi->output_ready = CA0106_MIDI_OUTPUT_READY; + + midi->channel = channel; + + midi->interrupt_enable = ca0106_midi_interrupt_enable; + midi->interrupt_disable = ca0106_midi_interrupt_disable; + + midi->read = ca0106_midi_read; + midi->write = ca0106_midi_write; + + midi->get_dev_id_card = ca0106_dev_id_card; + midi->get_dev_id_port = ca0106_dev_id_port; + + midi->dev_id = chip; + + if ((err = ca_midi_init(chip, midi, 0, name)) < 0) + return err; + + return 0; +} + + +static int __devinit snd_ca0106_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) +{ + static int dev; + struct snd_card *card; + struct snd_ca0106 *chip; + int err; + + if (dev >= SNDRV_CARDS) + return -ENODEV; + if (!enable[dev]) { + dev++; + return -ENOENT; + } + + card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); + if (card == NULL) + return -ENOMEM; + + if ((err = snd_ca0106_create(dev, card, pci, &chip)) < 0) { + snd_card_free(card); + return err; + } + + if ((err = snd_ca0106_pcm(chip, 0, NULL)) < 0) { + snd_card_free(card); + return err; + } + if ((err = snd_ca0106_pcm(chip, 1, NULL)) < 0) { + snd_card_free(card); + return err; + } + if ((err = snd_ca0106_pcm(chip, 2, NULL)) < 0) { + snd_card_free(card); + return err; + } + if ((err = snd_ca0106_pcm(chip, 3, NULL)) < 0) { + snd_card_free(card); + return err; + } + if (chip->details->ac97 == 1) { /* The SB0410 and SB0413 do not have an AC97 chip. */ + if ((err = snd_ca0106_ac97(chip)) < 0) { + snd_card_free(card); + return err; + } + } + if ((err = snd_ca0106_mixer(chip)) < 0) { + snd_card_free(card); + return err; + } + + snd_printdd("ca0106: probe for MIDI channel A ..."); + if ((err = snd_ca0106_midi(chip,CA0106_MIDI_CHAN_A)) < 0) { + snd_card_free(card); + snd_printdd(" failed, err=0x%x\n",err); + return err; + } + snd_printdd(" done.\n"); + +#ifdef CONFIG_PROC_FS + snd_ca0106_proc_init(chip); +#endif + + snd_card_set_dev(card, &pci->dev); + + if ((err = snd_card_register(card)) < 0) { + snd_card_free(card); + return err; + } + + pci_set_drvdata(pci, card); + dev++; + return 0; +} + +static void __devexit snd_ca0106_remove(struct pci_dev *pci) +{ + snd_card_free(pci_get_drvdata(pci)); + pci_set_drvdata(pci, NULL); +} + +// PCI IDs +static struct pci_device_id snd_ca0106_ids[] = { + { 0x1102, 0x0007, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* Audigy LS or Live 24bit */ + { 0, } +}; +MODULE_DEVICE_TABLE(pci, snd_ca0106_ids); + +// pci_driver definition +static struct pci_driver driver = { + .name = "CA0106", + .id_table = snd_ca0106_ids, + .probe = snd_ca0106_probe, + .remove = __devexit_p(snd_ca0106_remove), +}; + +// initialization of the module +static int __init alsa_card_ca0106_init(void) +{ + return pci_register_driver(&driver); +} + +// clean up the module +static void __exit alsa_card_ca0106_exit(void) +{ + pci_unregister_driver(&driver); +} + +module_init(alsa_card_ca0106_init) +module_exit(alsa_card_ca0106_exit) diff -ruN linux-2.6.22/sound/pci/ca0106/ca0106_mixer.c linux-2.6.22-alsa/sound/pci/ca0106/ca0106_mixer.c --- linux-2.6.22/sound/pci/ca0106/ca0106_mixer.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/pci/ca0106/ca0106_mixer.c 2007-09-01 20:55:23.000000000 +0200 @@ -1,7 +1,7 @@ /* * Copyright (c) 2004 James Courtier-Dutton * Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit - * Version: 0.0.17 + * Version: 0.0.18 * * FEATURES currently supported: * See ca0106_main.c for features. @@ -39,6 +39,8 @@ * Modified Copyright message. * 0.0.17 * Implement Mic and Line in Capture. + * 0.0.18 + * Add support for mute control on SB Live 24bit (cards w/ SPI DAC) * * This code was initally based on code from ALSA's emu10k1x.c which is: * Copyright (c) by Francisco Moraes @@ -77,15 +79,7 @@ static const DECLARE_TLV_DB_SCALE(snd_ca0106_db_scale1, -5175, 25, 1); static const DECLARE_TLV_DB_SCALE(snd_ca0106_db_scale2, -10350, 50, 1); -static int snd_ca0106_shared_spdif_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_ca0106_shared_spdif_info snd_ctl_boolean_mono_info static int snd_ca0106_shared_spdif_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -470,6 +464,42 @@ return change; } +#define spi_mute_info snd_ctl_boolean_mono_info + +static int spi_mute_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol); + unsigned int reg = kcontrol->private_value >> SPI_REG_SHIFT; + unsigned int bit = kcontrol->private_value & SPI_REG_MASK; + + ucontrol->value.integer.value[0] = !(emu->spi_dac_reg[reg] & bit); + return 0; +} + +static int spi_mute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol); + unsigned int reg = kcontrol->private_value >> SPI_REG_SHIFT; + unsigned int bit = kcontrol->private_value & SPI_REG_MASK; + int ret; + + ret = emu->spi_dac_reg[reg] & bit; + if (ucontrol->value.integer.value[0]) { + if (!ret) /* bit already cleared, do nothing */ + return 0; + emu->spi_dac_reg[reg] &= ~bit; + } else { + if (ret) /* bit already set, do nothing */ + return 0; + emu->spi_dac_reg[reg] |= bit; + } + + ret = snd_ca0106_spi_write(emu, emu->spi_dac_reg[reg]); + return ret ? -1 : 1; +} + #define CA_VOLUME(xname,chid,reg) \ { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ @@ -562,6 +592,28 @@ I2C_VOLUME("Aux Capture Volume", 3), }; +#define SPI_SWITCH(xname,reg,bit) \ +{ \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .info = spi_mute_info, \ + .get = spi_mute_get, \ + .put = spi_mute_put, \ + .private_value = (reg<card; char **c; static char *ca0106_remove_ctls[] = { @@ -640,17 +702,9 @@ rename_ctl(card, c[0], c[1]); #endif - for (i = 0; i < ARRAY_SIZE(snd_ca0106_volume_ctls); i++) { - err = snd_ctl_add(card, snd_ctl_new1(&snd_ca0106_volume_ctls[i], emu)); - if (err < 0) - return err; - } + ADD_CTLS(emu, snd_ca0106_volume_ctls); if (emu->details->i2c_adc == 1) { - for (i = 0; i < ARRAY_SIZE(snd_ca0106_volume_i2c_adc_ctls); i++) { - err = snd_ctl_add(card, snd_ctl_new1(&snd_ca0106_volume_i2c_adc_ctls[i], emu)); - if (err < 0) - return err; - } + ADD_CTLS(emu, snd_ca0106_volume_i2c_adc_ctls); if (emu->details->gpio_type == 1) err = snd_ctl_add(card, snd_ctl_new1(&snd_ca0106_capture_mic_line_in, emu)); else /* gpio_type == 2 */ @@ -658,6 +712,8 @@ if (err < 0) return err; } + if (emu->details->spi_dac == 1) + ADD_CTLS(emu, snd_ca0106_volume_spi_dac_ctls); return 0; } diff -ruN linux-2.6.22/sound/pci/ca0106/ca_midi.h linux-2.6.22-alsa/sound/pci/ca0106/ca_midi.h --- linux-2.6.22/sound/pci/ca0106/ca_midi.h 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/pci/ca0106/ca_midi.h 2007-09-01 20:55:23.000000000 +0200 @@ -22,9 +22,9 @@ * */ -#include -#include -#include +#include +#include +#include #define CA_MIDI_MODE_INPUT MPU401_MODE_INPUT #define CA_MIDI_MODE_OUTPUT MPU401_MODE_OUTPUT diff -ruN linux-2.6.22/sound/pci/cmipci.c linux-2.6.22-alsa/sound/pci/cmipci.c --- linux-2.6.22/sound/pci/cmipci.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/pci/cmipci.c 2007-09-01 20:55:22.000000000 +0200 @@ -424,7 +424,6 @@ int chip_version; int max_channels; - unsigned int has_dual_dac: 1; unsigned int can_ac3_sw: 1; unsigned int can_ac3_hw: 1; unsigned int can_multi_ch: 1; @@ -2139,15 +2138,7 @@ */ }; -static int snd_cmipci_uswitch_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_cmipci_uswitch_info snd_ctl_boolean_mono_info static int _snd_cmipci_uswitch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol, @@ -2633,46 +2624,40 @@ if (! detect) { /* check reg 08h, bit 24-28 */ detect = snd_cmipci_read(cm, CM_REG_CHFORMAT) & CM_CHIP_MASK1; - if (! detect) { + switch (detect) { + case 0: cm->chip_version = 33; - cm->max_channels = 2; if (cm->do_soft_ac3) cm->can_ac3_sw = 1; else cm->can_ac3_hw = 1; - cm->has_dual_dac = 1; - } else { + break; + case 1: cm->chip_version = 37; - cm->max_channels = 2; cm->can_ac3_hw = 1; - cm->has_dual_dac = 1; + break; + default: + cm->chip_version = 39; + cm->can_ac3_hw = 1; + break; } + cm->max_channels = 2; } else { - /* check reg 0Ch, bit 26 */ - if (detect & CM_CHIP_8768) { - cm->chip_version = 68; - cm->max_channels = 8; - cm->can_ac3_hw = 1; - cm->has_dual_dac = 1; - cm->can_multi_ch = 1; - } else if (detect & CM_CHIP_055) { - cm->chip_version = 55; - cm->max_channels = 6; - cm->can_ac3_hw = 1; - cm->has_dual_dac = 1; - cm->can_multi_ch = 1; - } else if (detect & CM_CHIP_039) { + if (detect & CM_CHIP_039) { cm->chip_version = 39; if (detect & CM_CHIP_039_6CH) /* 4 or 6 channels */ cm->max_channels = 6; else cm->max_channels = 4; - cm->can_ac3_hw = 1; - cm->has_dual_dac = 1; - cm->can_multi_ch = 1; + } else if (detect & CM_CHIP_8768) { + cm->chip_version = 68; + cm->max_channels = 8; } else { - printk(KERN_ERR "chip %x version not supported\n", detect); + cm->chip_version = 55; + cm->max_channels = 6; } + cm->can_ac3_hw = 1; + cm->can_multi_ch = 1; } } @@ -2782,10 +2767,14 @@ if (!fm_port) goto disable_fm; - /* first try FM regs in PCI port range */ - iosynth = cm->iobase + CM_REG_FM_PCI; - err = snd_opl3_create(cm->card, iosynth, iosynth + 2, - OPL3_HW_OPL3, 1, &opl3); + if (cm->chip_version > 33) { + /* first try FM regs in PCI port range */ + iosynth = cm->iobase + CM_REG_FM_PCI; + err = snd_opl3_create(cm->card, iosynth, iosynth + 2, + OPL3_HW_OPL3, 1, &opl3); + } else { + err = -EIO; + } if (err < 0) { /* then try legacy ports */ val = snd_cmipci_read(cm, CM_REG_LEGACY_CTRL) & ~CM_FMSEL_MASK; @@ -2829,9 +2818,9 @@ static struct snd_device_ops ops = { .dev_free = snd_cmipci_dev_free, }; - unsigned int val = 0; + unsigned int val; long iomidi; - int integrated_midi; + int integrated_midi = 0; int pcm_index, pcm_spdif_index; static struct pci_device_id intel_82437vx[] = { { PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_DEVICE_ID_INTEL_82437VX) }, @@ -2931,15 +2920,54 @@ break; } + sprintf(card->shortname, "C-Media %s", card->driver); + if (cm->chip_version < 68) { + val = pci->device < 0x110 ? 8338 : 8738; + sprintf(card->longname, + "C-Media CMI%d (model %d) at 0x%lx, irq %i", + val, cm->chip_version, cm->iobase, cm->irq); + } else { + switch (snd_cmipci_read_b(cm, CM_REG_INT_HLDCLR + 3) & 0x03) { + case 0: + val = 8769; + break; + case 2: + val = 8762; + break; + default: + switch ((pci->subsystem_vendor << 16) | + pci->subsystem_device) { + case 0x13f69761: + case 0x584d3741: + case 0x584d3751: + case 0x584d3761: + case 0x584d3771: + case 0x72848384: + val = 8770; + break; + default: + val = 8768; + break; + } + } + sprintf(card->longname, "C-Media CMI%d at 0x%lx, irq %i", + val, cm->iobase, cm->irq); + } + if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, cm, &ops)) < 0) { snd_cmipci_free(cm); return err; } - integrated_midi = snd_cmipci_read_b(cm, CM_REG_MPU_PCI) != 0xff; - if (integrated_midi && mpu_port[dev] == 1) - iomidi = cm->iobase + CM_REG_MPU_PCI; - else { + val = 0; + if (cm->chip_version > 33 && mpu_port[dev] == 1) { + val = snd_cmipci_read_b(cm, CM_REG_MPU_PCI + 1); + if (val != 0x00 && val != 0xff) { + iomidi = cm->iobase + CM_REG_MPU_PCI; + integrated_midi = 1; + } + } + if (!integrated_midi) { iomidi = mpu_port[dev]; switch (iomidi) { case 0x320: val = CM_VMPU_320; break; @@ -2956,8 +2984,11 @@ } } - if ((err = snd_cmipci_create_fm(cm, fm_port[dev])) < 0) - return err; + if (cm->chip_version < 68) { + err = snd_cmipci_create_fm(cm, fm_port[dev]); + if (err < 0) + return err; + } /* reset mixer */ snd_cmipci_mixer_write(cm, 0, 0); @@ -2969,11 +3000,9 @@ if ((err = snd_cmipci_pcm_new(cm, pcm_index)) < 0) return err; pcm_index++; - if (cm->has_dual_dac) { - if ((err = snd_cmipci_pcm2_new(cm, pcm_index)) < 0) - return err; - pcm_index++; - } + if ((err = snd_cmipci_pcm2_new(cm, pcm_index)) < 0) + return err; + pcm_index++; if (cm->can_ac3_hw || cm->can_ac3_sw) { pcm_spdif_index = pcm_index; if ((err = snd_cmipci_pcm_spdif_new(cm, pcm_index)) < 0) @@ -3057,15 +3086,6 @@ } card->private_data = cm; - sprintf(card->shortname, "C-Media PCI %s", card->driver); - sprintf(card->longname, "%s (model %d) at 0x%lx, irq %i", - card->shortname, - cm->chip_version, - cm->iobase, - cm->irq); - - //snd_printd("%s is detected\n", card->longname); - if ((err = snd_card_register(card)) < 0) { snd_card_free(card); return err; diff -ruN linux-2.6.22/sound/pci/cs4281.c linux-2.6.22-alsa/sound/pci/cs4281.c --- linux-2.6.22/sound/pci/cs4281.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/pci/cs4281.c 2007-09-01 20:55:23.000000000 +0200 @@ -842,12 +842,11 @@ static struct snd_pcm_hardware snd_cs4281_playback = { - .info = (SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_RESUME | - SNDRV_PCM_INFO_SYNC_START), + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME, .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_BE | SNDRV_PCM_FMTBIT_S16_BE | @@ -868,12 +867,11 @@ static struct snd_pcm_hardware snd_cs4281_capture = { - .info = (SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_RESUME | - SNDRV_PCM_INFO_SYNC_START), + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME, .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_BE | SNDRV_PCM_FMTBIT_S16_BE | @@ -904,7 +902,6 @@ dma->right_slot = 1; runtime->private_data = dma; runtime->hw = snd_cs4281_playback; - snd_pcm_set_sync(substream); /* should be detected from the AC'97 layer, but it seems that although CS4297A rev B reports 18-bit ADC resolution, samples are 20-bit */ @@ -924,7 +921,6 @@ dma->right_slot = 11; runtime->private_data = dma; runtime->hw = snd_cs4281_capture; - snd_pcm_set_sync(substream); /* should be detected from the AC'97 layer, but it seems that although CS4297A rev B reports 18-bit ADC resolution, samples are 20-bit */ diff -ruN linux-2.6.22/sound/pci/cs46xx/Makefile linux-2.6.22-alsa/sound/pci/cs46xx/Makefile --- linux-2.6.22/sound/pci/cs46xx/Makefile 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/pci/cs46xx/Makefile 2007-09-01 20:55:23.000000000 +0200 @@ -3,10 +3,8 @@ # Copyright (c) 2001 by Jaroslav Kysela # -snd-cs46xx-objs := cs46xx.o cs46xx_lib.o -ifeq ($(CONFIG_SND_CS46XX_NEW_DSP),y) - snd-cs46xx-objs += dsp_spos.o dsp_spos_scb_lib.o -endif +snd-cs46xx-y := cs46xx.o cs46xx_lib.o +snd-cs46xx-$(CONFIG_SND_CS46XX_NEW_DSP) += dsp_spos.o dsp_spos_scb_lib.o # Toplevel Module Dependency obj-$(CONFIG_SND_CS46XX) += snd-cs46xx.o diff -ruN linux-2.6.22/sound/pci/cs46xx/cs46xx_lib.c linux-2.6.22-alsa/sound/pci/cs46xx/cs46xx_lib.c --- linux-2.6.22/sound/pci/cs46xx/cs46xx_lib.c 2007-07-28 13:40:44.000000000 +0200 +++ linux-2.6.22-alsa/sound/pci/cs46xx/cs46xx_lib.c 2007-09-01 20:55:23.000000000 +0200 @@ -1818,15 +1818,7 @@ } #endif -static int snd_mixer_boolean_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_mixer_boolean_info snd_ctl_boolean_mono_info static int snd_cs46xx_iec958_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -2897,6 +2889,10 @@ } #endif +#ifdef CONFIG_PM + kfree(chip->saved_regs); +#endif + pci_disable_device(chip->pci); kfree(chip); return 0; @@ -3140,6 +3136,23 @@ /* * start and load DSP */ + +static void cs46xx_enable_stream_irqs(struct snd_cs46xx *chip) +{ + unsigned int tmp; + + snd_cs46xx_pokeBA0(chip, BA0_HICR, HICR_IEV | HICR_CHGM); + + tmp = snd_cs46xx_peek(chip, BA1_PFIE); + tmp &= ~0x0000f03f; + snd_cs46xx_poke(chip, BA1_PFIE, tmp); /* playback interrupt enable */ + + tmp = snd_cs46xx_peek(chip, BA1_CIE); + tmp &= ~0x0000003f; + tmp |= 0x00000001; + snd_cs46xx_poke(chip, BA1_CIE, tmp); /* capture interrupt enable */ +} + int __devinit snd_cs46xx_start_dsp(struct snd_cs46xx *chip) { unsigned int tmp; @@ -3214,19 +3227,7 @@ snd_cs46xx_proc_start(chip); - /* - * Enable interrupts on the part. - */ - snd_cs46xx_pokeBA0(chip, BA0_HICR, HICR_IEV | HICR_CHGM); - - tmp = snd_cs46xx_peek(chip, BA1_PFIE); - tmp &= ~0x0000f03f; - snd_cs46xx_poke(chip, BA1_PFIE, tmp); /* playback interrupt enable */ - - tmp = snd_cs46xx_peek(chip, BA1_CIE); - tmp &= ~0x0000003f; - tmp |= 0x00000001; - snd_cs46xx_poke(chip, BA1_CIE, tmp); /* capture interrupt enable */ + cs46xx_enable_stream_irqs(chip); #ifndef CONFIG_SND_CS46XX_NEW_DSP /* set the attenuation to 0dB */ @@ -3665,11 +3666,19 @@ * APM support */ #ifdef CONFIG_PM +static unsigned int saved_regs[] = { + BA0_ACOSV, + BA0_ASER_FADDR, + BA0_ASER_MASTER, + BA1_PVOL, + BA1_CVOL, +}; + int snd_cs46xx_suspend(struct pci_dev *pci, pm_message_t state) { struct snd_card *card = pci_get_drvdata(pci); struct snd_cs46xx *chip = card->private_data; - int amp_saved; + int i, amp_saved; snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); chip->in_suspend = 1; @@ -3680,6 +3689,10 @@ snd_ac97_suspend(chip->ac97[CS46XX_PRIMARY_CODEC_INDEX]); snd_ac97_suspend(chip->ac97[CS46XX_SECONDARY_CODEC_INDEX]); + /* save some registers */ + for (i = 0; i < ARRAY_SIZE(saved_regs); i++) + chip->saved_regs[i] = snd_cs46xx_peekBA0(chip, saved_regs[i]); + amp_saved = chip->amplifier; /* turn off amp */ chip->amplifier_ctrl(chip, -chip->amplifier); @@ -3698,7 +3711,7 @@ { struct snd_card *card = pci_get_drvdata(pci); struct snd_cs46xx *chip = card->private_data; - int amp_saved; + int i, amp_saved; pci_set_power_state(pci, PCI_D0); pci_restore_state(pci); @@ -3716,6 +3729,16 @@ snd_cs46xx_chip_init(chip); + snd_cs46xx_reset(chip); +#ifdef CONFIG_SND_CS46XX_NEW_DSP + cs46xx_dsp_resume(chip); + /* restore some registers */ + for (i = 0; i < ARRAY_SIZE(saved_regs); i++) + snd_cs46xx_pokeBA0(chip, saved_regs[i], chip->saved_regs[i]); +#else + snd_cs46xx_download_image(chip); +#endif + #if 0 snd_cs46xx_codec_write(chip, BA0_AC97_GENERAL_PURPOSE, chip->ac97_general_purpose); @@ -3730,6 +3753,13 @@ snd_ac97_resume(chip->ac97[CS46XX_PRIMARY_CODEC_INDEX]); snd_ac97_resume(chip->ac97[CS46XX_SECONDARY_CODEC_INDEX]); + /* reset playback/capture */ + snd_cs46xx_set_play_sample_rate(chip, 8000); + snd_cs46xx_set_capture_sample_rate(chip, 8000); + snd_cs46xx_proc_start(chip); + + cs46xx_enable_stream_irqs(chip); + if (amp_saved) chip->amplifier_ctrl(chip, 1); /* turn amp on */ else @@ -3896,6 +3926,15 @@ snd_cs46xx_proc_init(card, chip); +#ifdef CONFIG_PM + chip->saved_regs = kmalloc(sizeof(*chip->saved_regs) * + ARRAY_SIZE(saved_regs), GFP_KERNEL); + if (!chip->saved_regs) { + snd_cs46xx_free(chip); + return -ENOMEM; + } +#endif + chip->active_ctrl(chip, -1); /* disable CLKRUN */ snd_card_set_dev(card, &pci->dev); diff -ruN linux-2.6.22/sound/pci/cs46xx/cs46xx_lib.h linux-2.6.22-alsa/sound/pci/cs46xx/cs46xx_lib.h --- linux-2.6.22/sound/pci/cs46xx/cs46xx_lib.h 2007-07-28 13:40:44.000000000 +0200 +++ linux-2.6.22-alsa/sound/pci/cs46xx/cs46xx_lib.h 2007-09-01 20:55:23.000000000 +0200 @@ -86,6 +86,9 @@ struct dsp_spos_instance *cs46xx_dsp_spos_create (struct snd_cs46xx * chip); void cs46xx_dsp_spos_destroy (struct snd_cs46xx * chip); int cs46xx_dsp_load_module (struct snd_cs46xx * chip, struct dsp_module_desc * module); +#ifdef CONFIG_PM +int cs46xx_dsp_resume(struct snd_cs46xx * chip); +#endif struct dsp_symbol_entry *cs46xx_dsp_lookup_symbol (struct snd_cs46xx * chip, char * symbol_name, int symbol_type); #ifdef CONFIG_PROC_FS diff -ruN linux-2.6.22/sound/pci/cs46xx/dsp_spos.c linux-2.6.22-alsa/sound/pci/cs46xx/dsp_spos.c --- linux-2.6.22/sound/pci/cs46xx/dsp_spos.c 2007-07-28 13:40:44.000000000 +0200 +++ linux-2.6.22-alsa/sound/pci/cs46xx/dsp_spos.c 2007-09-01 20:55:23.000000000 +0200 @@ -306,13 +306,59 @@ mutex_unlock(&chip->spos_mutex); } +static int dsp_load_parameter(struct snd_cs46xx *chip, + struct dsp_segment_desc *parameter) +{ + u32 doffset, dsize; + + if (!parameter) { + snd_printdd("dsp_spos: module got no parameter segment\n"); + return 0; + } + + doffset = (parameter->offset * 4 + DSP_PARAMETER_BYTE_OFFSET); + dsize = parameter->size * 4; + + snd_printdd("dsp_spos: " + "downloading parameter data to chip (%08x-%08x)\n", + doffset,doffset + dsize); + if (snd_cs46xx_download (chip, parameter->data, doffset, dsize)) { + snd_printk(KERN_ERR "dsp_spos: " + "failed to download parameter data to DSP\n"); + return -EINVAL; + } + return 0; +} + +static int dsp_load_sample(struct snd_cs46xx *chip, + struct dsp_segment_desc *sample) +{ + u32 doffset, dsize; + + if (!sample) { + snd_printdd("dsp_spos: module got no sample segment\n"); + return 0; + } + + doffset = (sample->offset * 4 + DSP_SAMPLE_BYTE_OFFSET); + dsize = sample->size * 4; + + snd_printdd("dsp_spos: downloading sample data to chip (%08x-%08x)\n", + doffset,doffset + dsize); + + if (snd_cs46xx_download (chip,sample->data,doffset,dsize)) { + snd_printk(KERN_ERR "dsp_spos: failed to sample data to DSP\n"); + return -EINVAL; + } + return 0; +} + int cs46xx_dsp_load_module (struct snd_cs46xx * chip, struct dsp_module_desc * module) { struct dsp_spos_instance * ins = chip->dsp_spos_instance; struct dsp_segment_desc * code = get_segment_desc (module,SEGTYPE_SP_PROGRAM); - struct dsp_segment_desc * parameter = get_segment_desc (module,SEGTYPE_SP_PARAMETER); - struct dsp_segment_desc * sample = get_segment_desc (module,SEGTYPE_SP_SAMPLE); u32 doffset, dsize; + int err; if (ins->nmodules == DSP_MAX_MODULES - 1) { snd_printk(KERN_ERR "dsp_spos: to many modules loaded into DSP\n"); @@ -326,49 +372,20 @@ snd_cs46xx_clear_BA1(chip, DSP_PARAMETER_BYTE_OFFSET, DSP_PARAMETER_BYTE_SIZE); } - if (parameter == NULL) { - snd_printdd("dsp_spos: module got no parameter segment\n"); - } else { - if (ins->nmodules > 0) { - snd_printk(KERN_WARNING "dsp_spos: WARNING current parameter data may be overwriten!\n"); - } - - doffset = (parameter->offset * 4 + DSP_PARAMETER_BYTE_OFFSET); - dsize = parameter->size * 4; - - snd_printdd("dsp_spos: downloading parameter data to chip (%08x-%08x)\n", - doffset,doffset + dsize); - - if (snd_cs46xx_download (chip, parameter->data, doffset, dsize)) { - snd_printk(KERN_ERR "dsp_spos: failed to download parameter data to DSP\n"); - return -EINVAL; - } - } + err = dsp_load_parameter(chip, get_segment_desc(module, + SEGTYPE_SP_PARAMETER)); + if (err < 0) + return err; if (ins->nmodules == 0) { snd_printdd("dsp_spos: clearing sample area\n"); snd_cs46xx_clear_BA1(chip, DSP_SAMPLE_BYTE_OFFSET, DSP_SAMPLE_BYTE_SIZE); } - if (sample == NULL) { - snd_printdd("dsp_spos: module got no sample segment\n"); - } else { - if (ins->nmodules > 0) { - snd_printk(KERN_WARNING "dsp_spos: WARNING current sample data may be overwriten\n"); - } - - doffset = (sample->offset * 4 + DSP_SAMPLE_BYTE_OFFSET); - dsize = sample->size * 4; - - snd_printdd("dsp_spos: downloading sample data to chip (%08x-%08x)\n", - doffset,doffset + dsize); - - if (snd_cs46xx_download (chip,sample->data,doffset,dsize)) { - snd_printk(KERN_ERR "dsp_spos: failed to sample data to DSP\n"); - return -EINVAL; - } - } - + err = dsp_load_sample(chip, get_segment_desc(module, + SEGTYPE_SP_SAMPLE)); + if (err < 0) + return err; if (ins->nmodules == 0) { snd_printdd("dsp_spos: clearing code area\n"); @@ -986,7 +1003,10 @@ return NULL; } - strcpy(ins->tasks[ins->ntask].task_name,name); + if (name) + strcpy(ins->tasks[ins->ntask].task_name, name); + else + strcpy(ins->tasks[ins->ntask].task_name, "(NULL)"); ins->tasks[ins->ntask].address = dest; ins->tasks[ins->ntask].size = size; @@ -995,7 +1015,8 @@ desc = (ins->tasks + ins->ntask); ins->ntask++; - add_symbol (chip,name,dest,SYMBOL_PARAMETER); + if (name) + add_symbol (chip,name,dest,SYMBOL_PARAMETER); return desc; } @@ -1006,6 +1027,7 @@ desc = _map_scb (chip,name,dest); if (desc) { + desc->data = scb_data; _dsp_create_scb(chip,scb_data,dest); } else { snd_printk(KERN_ERR "dsp_spos: failed to map SCB\n"); @@ -1023,6 +1045,7 @@ desc = _map_task_tree (chip,name,dest,size); if (desc) { + desc->data = task_data; _dsp_create_task_tree(chip,task_data,dest,size); } else { snd_printk(KERN_ERR "dsp_spos: failed to map TASK\n"); @@ -1320,8 +1343,10 @@ 0x0000ffff }; - /* dirty hack ... */ - _dsp_create_task_tree (chip,(u32 *)&mix2_ostream_spb,WRITE_BACK_SPB,2); + if (!cs46xx_dsp_create_task_tree(chip, NULL, + (u32 *)&mix2_ostream_spb, + WRITE_BACK_SPB, 2)) + goto _fail_end; } /* input sample converter */ @@ -1622,7 +1647,6 @@ return 0; } - static void cs46xx_dsp_disable_spdif_hw (struct snd_cs46xx *chip) { struct dsp_spos_instance * ins = chip->dsp_spos_instance; @@ -1894,3 +1918,61 @@ return 0; } + +#ifdef CONFIG_PM +int cs46xx_dsp_resume(struct snd_cs46xx * chip) +{ + struct dsp_spos_instance * ins = chip->dsp_spos_instance; + int i, err; + + /* clear parameter, sample and code areas */ + snd_cs46xx_clear_BA1(chip, DSP_PARAMETER_BYTE_OFFSET, + DSP_PARAMETER_BYTE_SIZE); + snd_cs46xx_clear_BA1(chip, DSP_SAMPLE_BYTE_OFFSET, + DSP_SAMPLE_BYTE_SIZE); + snd_cs46xx_clear_BA1(chip, DSP_CODE_BYTE_OFFSET, DSP_CODE_BYTE_SIZE); + + for (i = 0; i < ins->nmodules; i++) { + struct dsp_module_desc *module = &ins->modules[i]; + struct dsp_segment_desc *seg; + u32 doffset, dsize; + + seg = get_segment_desc(module, SEGTYPE_SP_PARAMETER); + err = dsp_load_parameter(chip, seg); + if (err < 0) + return err; + + seg = get_segment_desc(module, SEGTYPE_SP_SAMPLE); + err = dsp_load_sample(chip, seg); + if (err < 0) + return err; + + seg = get_segment_desc(module, SEGTYPE_SP_PROGRAM); + if (!seg) + continue; + + doffset = seg->offset * 4 + module->load_address * 4 + + DSP_CODE_BYTE_OFFSET; + dsize = seg->size * 4; + err = snd_cs46xx_download(chip, + ins->code.data + module->load_address, + doffset, dsize); + if (err < 0) + return err; + } + + for (i = 0; i < ins->ntask; i++) { + struct dsp_task_descriptor *t = &ins->tasks[i]; + _dsp_create_task_tree(chip, t->data, t->address, t->size); + } + + for (i = 0; i < ins->nscb; i++) { + struct dsp_scb_descriptor *s = &ins->scbs[i]; + if (s->deleted) + continue; + _dsp_create_scb(chip, s->data, s->address); + } + + return 0; +} +#endif diff -ruN linux-2.6.22/sound/pci/cs46xx/dsp_spos_scb_lib.c linux-2.6.22-alsa/sound/pci/cs46xx/dsp_spos_scb_lib.c --- linux-2.6.22/sound/pci/cs46xx/dsp_spos_scb_lib.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/pci/cs46xx/dsp_spos_scb_lib.c 2007-09-01 20:55:23.000000000 +0200 @@ -1480,7 +1480,7 @@ if (!pcm_channel->src_scb->ref_count) { cs46xx_dsp_remove_scb(chip,pcm_channel->src_scb); - snd_assert (pcm_channel->src_slot >= 0 && pcm_channel->src_slot <= DSP_MAX_SRC_NR, + snd_assert (pcm_channel->src_slot >= 0 && pcm_channel->src_slot < DSP_MAX_SRC_NR, return ); ins->src_scb_slots[pcm_channel->src_slot] = 0; diff -ruN linux-2.6.22/sound/pci/cs5530.c linux-2.6.22-alsa/sound/pci/cs5530.c --- linux-2.6.22/sound/pci/cs5530.c 1970-01-01 01:00:00.000000000 +0100 +++ linux-2.6.22-alsa/sound/pci/cs5530.c 2007-09-01 20:55:22.000000000 +0200 @@ -0,0 +1,306 @@ +/* + * cs5530.c - Initialisation code for Cyrix/NatSemi VSA1 softaudio + * + * (C) Copyright 2007 Ash Willis + * (C) Copyright 2003 Red Hat Inc + * + * This driver was ported (shamelessly ripped ;) from oss/kahlua.c but I did + * mess with it a bit. The chip seems to have to have trouble with full duplex + * mode. If we're recording in 8bit 8000kHz, say, and we then attempt to + * simultaneously play back audio at 16bit 44100kHz, the device actually plays + * back in the same format in which it is capturing. By forcing the chip to + * always play/capture in 16/44100, we can let alsa-lib convert the samples and + * that way we can hack up some full duplex audio. + * + * XpressAudio(tm) is used on the Cyrix MediaGX (now NatSemi Geode) systems. + * The older version (VSA1) provides fairly good soundblaster emulation + * although there are a couple of bugs: large DMA buffers break record, + * and the MPU event handling seems suspect. VSA2 allows the native driver + * to control the AC97 audio engine directly and requires a different driver. + * + * Thanks to National Semiconductor for providing the needed information + * on the XpressAudio(tm) internals. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2, or (at your option) any + * later version. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * TO DO: + * Investigate whether we can portably support Cognac (5520) in the + * same manner. + */ + +#include +#include +#include +#include +#include +#include +#include + +MODULE_AUTHOR("Ash Willis"); +MODULE_DESCRIPTION("CS5530 Audio"); +MODULE_LICENSE("GPL"); + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; + +struct snd_cs5530 { + struct snd_card *card; + struct pci_dev *pci; + struct snd_sb *sb; + unsigned long pci_base; +}; + +static struct pci_device_id snd_cs5530_ids[] = { + {PCI_VENDOR_ID_CYRIX, PCI_DEVICE_ID_CYRIX_5530_AUDIO, PCI_ANY_ID, + PCI_ANY_ID, 0, 0}, + {0,} +}; + +MODULE_DEVICE_TABLE(pci, snd_cs5530_ids); + +static int snd_cs5530_free(struct snd_cs5530 *chip) +{ + pci_release_regions(chip->pci); + pci_disable_device(chip->pci); + kfree(chip); + return 0; +} + +static int snd_cs5530_dev_free(struct snd_device *device) +{ + struct snd_cs5530 *chip = device->device_data; + return snd_cs5530_free(chip); +} + +static void __devexit snd_cs5530_remove(struct pci_dev *pci) +{ + snd_card_free(pci_get_drvdata(pci)); + pci_set_drvdata(pci, NULL); +} + +static u8 __devinit snd_cs5530_mixer_read(unsigned long io, u8 reg) +{ + outb(reg, io + 4); + udelay(20); + reg = inb(io + 5); + udelay(20); + return reg; +} + +static int __devinit snd_cs5530_create(struct snd_card *card, + struct pci_dev *pci, + struct snd_cs5530 **rchip) +{ + struct snd_cs5530 *chip; + unsigned long sb_base; + u8 irq, dma8, dma16 = 0; + u16 map; + void __iomem *mem; + int err; + + static struct snd_device_ops ops = { + .dev_free = snd_cs5530_dev_free, + }; + *rchip = NULL; + + err = pci_enable_device(pci); + if (err < 0) + return err; + + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + if (chip == NULL) { + pci_disable_device(pci); + return -ENOMEM; + } + + chip->card = card; + chip->pci = pci; + + err = pci_request_regions(pci, "CS5530"); + if (err < 0) { + kfree(chip); + pci_disable_device(pci); + return err; + } + chip->pci_base = pci_resource_start(pci, 0); + + mem = ioremap_nocache(chip->pci_base, pci_resource_len(pci, 0)); + if (mem == NULL) { + kfree(chip); + pci_disable_device(pci); + return -EBUSY; + } + + map = readw(mem + 0x18); + iounmap(mem); + + /* Map bits + 0:1 * 0x20 + 0x200 = sb base + 2 sb enable + 3 adlib enable + 5 MPU enable 0x330 + 6 MPU enable 0x300 + + The other bits may be used internally so must be masked */ + + sb_base = 0x220 + 0x20 * (map & 3); + + if (map & (1<<2)) + printk(KERN_INFO "CS5530: XpressAudio at 0x%lx\n", sb_base); + else { + printk(KERN_ERR "Could not find XpressAudio!\n"); + snd_cs5530_free(chip); + return -ENODEV; + } + + if (map & (1<<5)) + printk(KERN_INFO "CS5530: MPU at 0x300\n"); + else if (map & (1<<6)) + printk(KERN_INFO "CS5530: MPU at 0x330\n"); + + irq = snd_cs5530_mixer_read(sb_base, 0x80) & 0x0F; + dma8 = snd_cs5530_mixer_read(sb_base, 0x81); + + if (dma8 & 0x20) + dma16 = 5; + else if (dma8 & 0x40) + dma16 = 6; + else if (dma8 & 0x80) + dma16 = 7; + else { + printk(KERN_ERR "CS5530: No 16bit DMA enabled\n"); + snd_cs5530_free(chip); + return -ENODEV; + } + + if (dma8 & 0x01) + dma8 = 0; + else if (dma8 & 02) + dma8 = 1; + else if (dma8 & 0x08) + dma8 = 3; + else { + printk(KERN_ERR "CS5530: No 8bit DMA enabled\n"); + snd_cs5530_free(chip); + return -ENODEV; + } + + if (irq & 1) + irq = 9; + else if (irq & 2) + irq = 5; + else if (irq & 4) + irq = 7; + else if (irq & 8) + irq = 10; + else { + printk(KERN_ERR "CS5530: SoundBlaster IRQ not set\n"); + snd_cs5530_free(chip); + return -ENODEV; + } + + printk(KERN_INFO "CS5530: IRQ: %d DMA8: %d DMA16: %d\n", irq, dma8, + dma16); + + err = snd_sbdsp_create(card, sb_base, irq, snd_sb16dsp_interrupt, dma8, + dma16, SB_HW_CS5530, &chip->sb); + if (err < 0) { + printk(KERN_ERR "CS5530: Could not create SoundBlaster\n"); + snd_cs5530_free(chip); + return err; + } + + err = snd_sb16dsp_pcm(chip->sb, 0, &chip->sb->pcm); + if (err < 0) { + printk(KERN_ERR "CS5530: Could not create PCM\n"); + snd_cs5530_free(chip); + return err; + } + + err = snd_sbmixer_new(chip->sb); + if (err < 0) { + printk(KERN_ERR "CS5530: Could not create Mixer\n"); + snd_cs5530_free(chip); + return err; + } + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { + snd_cs5530_free(chip); + return err; + } + + snd_card_set_dev(card, &pci->dev); + *rchip = chip; + return 0; +} + +static int __devinit snd_cs5530_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) +{ + static int dev; + struct snd_card *card; + struct snd_cs5530 *chip = NULL; + int err; + + if (dev >= SNDRV_CARDS) + return -ENODEV; + if (!enable[dev]) { + dev++; + return -ENOENT; + } + + card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); + + if (card == NULL) + return -ENOMEM; + + err = snd_cs5530_create(card, pci, &chip); + if (err < 0) { + snd_card_free(card); + return err; + } + + strcpy(card->driver, "CS5530"); + strcpy(card->shortname, "CS5530 Audio"); + sprintf(card->longname, "%s at 0x%lx", card->shortname, chip->pci_base); + + err = snd_card_register(card); + if (err < 0) { + snd_card_free(card); + return err; + } + pci_set_drvdata(pci, card); + dev++; + return 0; +} + +static struct pci_driver driver = { + .name = "CS5530_Audio", + .id_table = snd_cs5530_ids, + .probe = snd_cs5530_probe, + .remove = __devexit_p(snd_cs5530_remove), +}; + +static int __init alsa_card_cs5530_init(void) +{ + return pci_register_driver(&driver); +} + +static void __exit alsa_card_cs5530_exit(void) +{ + pci_unregister_driver(&driver); +} + +module_init(alsa_card_cs5530_init) +module_exit(alsa_card_cs5530_exit) + diff -ruN linux-2.6.22/sound/pci/cs5535audio/Makefile linux-2.6.22-alsa/sound/pci/cs5535audio/Makefile --- linux-2.6.22/sound/pci/cs5535audio/Makefile 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/pci/cs5535audio/Makefile 2007-09-01 20:55:22.000000000 +0200 @@ -2,11 +2,8 @@ # Makefile for cs5535audio # -snd-cs5535audio-objs := cs5535audio.o cs5535audio_pcm.o - -ifeq ($(CONFIG_PM),y) -snd-cs5535audio-objs += cs5535audio_pm.o -endif +snd-cs5535audio-y := cs5535audio.o cs5535audio_pcm.o +snd-cs5535audio-$(CONFIG_PM) += cs5535audio_pm.o # Toplevel Module Dependency obj-$(CONFIG_SND_CS5535AUDIO) += snd-cs5535audio.o diff -ruN linux-2.6.22/sound/pci/cs5535audio/cs5535audio_pcm.c linux-2.6.22-alsa/sound/pci/cs5535audio/cs5535audio_pcm.c --- linux-2.6.22/sound/pci/cs5535audio/cs5535audio_pcm.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/pci/cs5535audio/cs5535audio_pcm.c 2007-09-01 20:55:22.000000000 +0200 @@ -43,7 +43,6 @@ SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_SYNC_START | SNDRV_PCM_INFO_RESUME ), .formats = ( @@ -71,8 +70,7 @@ SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_SYNC_START + SNDRV_PCM_INFO_MMAP_VALID ), .formats = ( SNDRV_PCM_FMTBIT_S16_LE @@ -102,7 +100,6 @@ runtime->hw = snd_cs5535audio_playback; cs5535au->playback_substream = substream; runtime->private_data = &(cs5535au->dmas[CS5535AUDIO_DMA_PLAYBACK]); - snd_pcm_set_sync(substream); if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) return err; @@ -348,7 +345,6 @@ runtime->hw = snd_cs5535audio_capture; cs5535au->capture_substream = substream; runtime->private_data = &(cs5535au->dmas[CS5535AUDIO_DMA_CAPTURE]); - snd_pcm_set_sync(substream); if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) return err; diff -ruN linux-2.6.22/sound/pci/echoaudio/echoaudio.c linux-2.6.22-alsa/sound/pci/echoaudio/echoaudio.c --- linux-2.6.22/sound/pci/echoaudio/echoaudio.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/pci/echoaudio/echoaudio.c 2007-09-01 20:55:23.000000000 +0200 @@ -1595,15 +1595,7 @@ #ifdef ECHOCARD_HAS_PHANTOM_POWER /******************* Phantom power switch *******************/ -static int snd_echo_phantom_power_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_echo_phantom_power_info snd_ctl_boolean_mono_info static int snd_echo_phantom_power_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1646,15 +1638,7 @@ #ifdef ECHOCARD_HAS_DIGITAL_IN_AUTOMUTE /******************* Digital input automute switch *******************/ -static int snd_echo_automute_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_echo_automute_info snd_ctl_boolean_mono_info static int snd_echo_automute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1695,18 +1679,7 @@ /******************* VU-meters switch *******************/ -static int snd_echo_vumeters_switch_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct echoaudio *chip; - - chip = snd_kcontrol_chip(kcontrol); - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_echo_vumeters_switch_info snd_ctl_boolean_mono_info static int snd_echo_vumeters_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) diff -ruN linux-2.6.22/sound/pci/emu10k1/emu10k1_main.c linux-2.6.22-alsa/sound/pci/emu10k1/emu10k1_main.c --- linux-2.6.22/sound/pci/emu10k1/emu10k1_main.c 2007-07-28 13:40:44.000000000 +0200 +++ linux-2.6.22-alsa/sound/pci/emu10k1/emu10k1_main.c 2007-09-02 13:30:43.000000000 +0200 @@ -31,6 +31,8 @@ * */ +#include +#include #include #include #include @@ -51,9 +53,15 @@ #define HANA_FILENAME "emu/hana.fw" #define DOCK_FILENAME "emu/audio_dock.fw" +#define EMU1010B_FILENAME "emu/emu1010b.fw" +#define MICRO_DOCK_FILENAME "emu/micro_dock.fw" +#define EMU1010_NOTEBOOK_FILENAME "emu/emu1010_notebook.fw" MODULE_FIRMWARE(HANA_FILENAME); MODULE_FIRMWARE(DOCK_FILENAME); +MODULE_FIRMWARE(EMU1010B_FILENAME); +MODULE_FIRMWARE(MICRO_DOCK_FILENAME); +MODULE_FIRMWARE(EMU1010_NOTEBOOK_FILENAME); /************************************************************************* @@ -660,10 +668,12 @@ return err; } snd_printk(KERN_INFO "firmware size=0x%zx\n", fw_entry->size); +#if 0 if (fw_entry->size != 0x133a4) { snd_printk(KERN_ERR "firmware: %s wrong size.\n",filename); return -EINVAL; } +#endif /* The FPGA is a Xilinx Spartan IIE XC2S50E */ /* GPIO7 -> FPGA PGMN @@ -694,6 +704,96 @@ return 0; } +int emu1010_firmware_thread(void *data) { + struct snd_emu10k1 * emu = data; + int tmp,tmp2; + int reg; + int err; + + for (;;) { + /* Delay to allow Audio Dock to settle */ + msleep(1000); + if (kthread_should_stop()) + break; + snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, &tmp ); /* IRQ Status */ + snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, ® ); /* OPTIONS: Which cards are attached to the EMU */ + if (reg & EMU_HANA_OPTION_DOCK_OFFLINE) { + /* Audio Dock attached */ + /* Return to Audio Dock programming mode */ + snd_printk(KERN_INFO "emu1010: Loading Audio Dock Firmware\n"); + snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, EMU_HANA_FPGA_CONFIG_AUDIODOCK ); + if (emu->card_capabilities->emu1010 == 1) { + if ((err = snd_emu1010_load_firmware(emu, DOCK_FILENAME)) != 0) { + return err; + } + } else if (emu->card_capabilities->emu1010 == 2) { + if ((err = snd_emu1010_load_firmware(emu, MICRO_DOCK_FILENAME)) != 0) { + return err; + } + } else if (emu->card_capabilities->emu1010 == 3) { + if ((err = snd_emu1010_load_firmware(emu, MICRO_DOCK_FILENAME)) != 0) { + return err; + } + } + + snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, 0 ); + snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, ® ); + snd_printk(KERN_INFO "emu1010: EMU_HANA+DOCK_IRQ_STATUS=0x%x\n",reg); + /* ID, should read & 0x7f = 0x55 when FPGA programmed. */ + snd_emu1010_fpga_read(emu, EMU_HANA_ID, ® ); + snd_printk(KERN_INFO "emu1010: EMU_HANA+DOCK_ID=0x%x\n",reg); + if ((reg & 0x1f) != 0x15) { + /* FPGA failed to be programmed */ + snd_printk(KERN_INFO "emu1010: Loading Audio Dock Firmware file failed, reg=0x%x\n", reg); + return 0; + return -ENODEV; + } + snd_printk(KERN_INFO "emu1010: Audio Dock Firmware loaded\n"); + snd_emu1010_fpga_read(emu, EMU_DOCK_MAJOR_REV, &tmp ); + snd_emu1010_fpga_read(emu, EMU_DOCK_MINOR_REV, &tmp2 ); + snd_printk("Audio Dock ver:%d.%d\n",tmp ,tmp2); + /* Sync clocking between 1010 and Dock */ + /* Allow DLL to settle */ + msleep(10); + /* Unmute all. Default is muted after a firmware load */ + snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE ); + break; + } + } + return 0; +} + +/* + * EMU-1010 - details found out from this driver, official MS Win drivers, + * testing the card: + * + * Audigy2 (aka Alice2): + * --------------------- + * * communication over PCI + * * conversion of 32-bit data coming over EMU32 links from HANA FPGA + * to 2 x 16-bit, using internal DSP instructions + * * slave mode, clock supplied by HANA + * * linked to HANA using: + * 32 x 32-bit serial EMU32 output channels + * 16 x EMU32 input channels + * (?) x I2S I/O channels (?) + * + * FPGA (aka HANA): + * --------------- + * * provides all (?) physical inputs and outputs of the card + * (ADC, DAC, SPDIF I/O, ADAT I/O, etc.) + * * provides clock signal for the card and Alice2 + * * two crystals - for 44.1kHz and 48kHz multiples + * * provides internal routing of signal sources to signal destinations + * * inputs/outputs to Alice2 - see above + * + * Current status of the driver: + * ---------------------------- + * * only 44.1/48kHz supported (the MS Win driver supports up to 192 kHz) + * * PCM device nb. 2: + * 16 x 16-bit playback - snd_emu10k1_fx8010_playback_ops + * 16 x 32-bit capture - snd_emu10k1_capture_efx_ops + */ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) { unsigned int i; @@ -727,7 +827,7 @@ /* ID, should read & 0x7f = 0x55. (Bit 7 is the IRQ bit) */ snd_emu1010_fpga_read(emu, EMU_HANA_ID, ® ); snd_printdd("reg1=0x%x\n",reg); - if (reg == 0x55) { + if ((reg & 0x3f) == 0x15) { /* FPGA netlist already present so clear it */ /* Return to programming mode */ @@ -735,19 +835,32 @@ } snd_emu1010_fpga_read(emu, EMU_HANA_ID, ® ); snd_printdd("reg2=0x%x\n",reg); - if (reg == 0x55) { + if ((reg & 0x3f) == 0x15) { /* FPGA failed to return to programming mode */ + snd_printk(KERN_INFO "emu1010: FPGA failed to return to programming mode\n"); return -ENODEV; } snd_printk(KERN_INFO "emu1010: EMU_HANA_ID=0x%x\n",reg); - if ((err = snd_emu1010_load_firmware(emu, HANA_FILENAME)) != 0) { - snd_printk(KERN_INFO "emu1010: Loading Hana Firmware file %s failed\n", HANA_FILENAME); - return err; + if (emu->card_capabilities->emu1010 == 1) { + if ((err = snd_emu1010_load_firmware(emu, HANA_FILENAME)) != 0) { + snd_printk(KERN_INFO "emu1010: Loading Hana Firmware file %s failed\n", HANA_FILENAME); + return err; + } + } else if (emu->card_capabilities->emu1010 == 2) { + if ((err = snd_emu1010_load_firmware(emu, EMU1010B_FILENAME)) != 0) { + snd_printk(KERN_INFO "emu1010: Loading Firmware file %s failed\n", EMU1010B_FILENAME); + return err; + } + } else if (emu->card_capabilities->emu1010 == 3) { + if ((err = snd_emu1010_load_firmware(emu, EMU1010_NOTEBOOK_FILENAME)) != 0) { + snd_printk(KERN_INFO "emu1010: Loading Firmware file %s failed\n", EMU1010_NOTEBOOK_FILENAME); + return err; + } } /* ID, should read & 0x7f = 0x55 when FPGA programmed. */ snd_emu1010_fpga_read(emu, EMU_HANA_ID, ® ); - if (reg != 0x55) { + if ((reg & 0x3f) != 0x15) { /* FPGA failed to be programmed */ snd_printk(KERN_INFO "emu1010: Loading Hana Firmware file failed, reg=0x%x\n", reg); return -ENODEV; @@ -765,8 +878,16 @@ snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, ® ); snd_printk(KERN_INFO "emu1010: Card options=0x%x\n",reg); snd_emu1010_fpga_read(emu, EMU_HANA_OPTICAL_TYPE, &tmp ); - /* ADAT input. */ - snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, 0x01 ); + /* Optical -> ADAT I/O */ + /* 0 : SPDIF + * 1 : ADAT + */ + emu->emu1010.optical_in = 1; /* IN_ADAT */ + emu->emu1010.optical_out = 1; /* IN_ADAT */ + tmp = 0; + tmp = (emu->emu1010.optical_in ? EMU_HANA_OPTICAL_IN_ADAT : 0) | + (emu->emu1010.optical_out ? EMU_HANA_OPTICAL_OUT_ADAT : 0); + snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, tmp ); snd_emu1010_fpga_read(emu, EMU_HANA_ADC_PADS, &tmp ); /* Set no attenuation on Audio Dock pads. */ snd_emu1010_fpga_write(emu, EMU_HANA_ADC_PADS, 0x00 ); @@ -850,6 +971,27 @@ EMU_DST_ALICE2_EMU32_6, EMU_SRC_DOCK_ADC2_LEFT1); snd_emu1010_fpga_link_dst_src_write(emu, EMU_DST_ALICE2_EMU32_7, EMU_SRC_DOCK_ADC2_RIGHT1); + /* Pavel Hofman - setting defaults for 8 more capture channels + * Defaults only, users will set their own values anyways, let's + * just copy/paste. + */ + + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_8, EMU_SRC_DOCK_MIC_A1); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_9, EMU_SRC_DOCK_MIC_B1); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_A, EMU_SRC_HAMOA_ADC_LEFT2); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_B, EMU_SRC_HAMOA_ADC_LEFT2); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_C, EMU_SRC_DOCK_ADC1_LEFT1); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_D, EMU_SRC_DOCK_ADC1_RIGHT1); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_E, EMU_SRC_DOCK_ADC2_LEFT1); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_F, EMU_SRC_DOCK_ADC2_RIGHT1); #endif #if 0 /* Original */ @@ -931,38 +1073,12 @@ snd_emu1010_fpga_read(emu, EMU_HANA_SPDIF_MODE, &tmp ); snd_emu1010_fpga_write(emu, EMU_HANA_SPDIF_MODE, 0x10 ); /* SPDIF Format spdif (or 0x11 for aes/ebu) */ - /* Delay to allow Audio Dock to settle */ - msleep(100); - snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, &tmp ); /* IRQ Status */ - snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, ® ); /* OPTIONS: Which cards are attached to the EMU */ - /* FIXME: The loading of this should be able to happen any time, - * as the user can plug/unplug it at any time - */ - if (reg & (EMU_HANA_OPTION_DOCK_ONLINE | EMU_HANA_OPTION_DOCK_OFFLINE) ) { - /* Audio Dock attached */ - /* Return to Audio Dock programming mode */ - snd_printk(KERN_INFO "emu1010: Loading Audio Dock Firmware\n"); - snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, EMU_HANA_FPGA_CONFIG_AUDIODOCK ); - if ((err = snd_emu1010_load_firmware(emu, DOCK_FILENAME)) != 0) { - return err; - } - snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, 0 ); - snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, ® ); - snd_printk(KERN_INFO "emu1010: EMU_HANA+DOCK_IRQ_STATUS=0x%x\n",reg); - /* ID, should read & 0x7f = 0x55 when FPGA programmed. */ - snd_emu1010_fpga_read(emu, EMU_HANA_ID, ® ); - snd_printk(KERN_INFO "emu1010: EMU_HANA+DOCK_ID=0x%x\n",reg); - if (reg != 0x55) { - /* FPGA failed to be programmed */ - snd_printk(KERN_INFO "emu1010: Loading Audio Dock Firmware file failed, reg=0x%x\n", reg); - return 0; - return -ENODEV; - } - snd_printk(KERN_INFO "emu1010: Audio Dock Firmware loaded\n"); - snd_emu1010_fpga_read(emu, EMU_DOCK_MAJOR_REV, &tmp ); - snd_emu1010_fpga_read(emu, EMU_DOCK_MINOR_REV, &tmp2 ); - snd_printk("Audio Dock ver:%d.%d\n",tmp ,tmp2); - } + /* Start Micro/Audio Dock firmware loader thread */ + emu->emu1010.firmware_thread = kthread_create(&emu1010_firmware_thread, + emu, + "emu1010_firmware"); + wake_up_process(emu->emu1010.firmware_thread); + #if 0 snd_emu1010_fpga_link_dst_src_write(emu, EMU_DST_HAMOA_DAC_LEFT1, EMU_SRC_ALICE_EMU32B + 2); /* ALICE2 bus 0xa2 */ @@ -1048,7 +1164,7 @@ emu->emu1010.output_source[23] = 28; /* TEMP: Select SPDIF in/out */ - snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, 0x0); /* Output spdif */ + //snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, 0x0); /* Output spdif */ /* TEMP: Select 48kHz SPDIF out */ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, 0x0); /* Mute all */ @@ -1089,6 +1205,7 @@ if (emu->card_capabilities->emu1010) { /* Disable 48Volt power to Audio Dock */ snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, 0 ); + kthread_stop(emu->emu1010.firmware_thread); } if (emu->memhdr) snd_util_memhdr_free(emu->memhdr); @@ -1227,9 +1344,15 @@ .emu10k2_chip = 1, .ca0108_chip = 1, .ca_cardbus_chip = 1, - .spi_dac = 1, - .i2c_adc = 1, - .spk71 = 1} , + .spk71 = 1 , + .emu1010 = 3} , + {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x40041102, + .driver = "Audigy2", .name = "E-mu 1010b PCI [MAEM????]", + .id = "EMU1010", + .emu10k2_chip = 1, + .ca0108_chip = 1, + .spk71 = 1 , + .emu1010 = 2} , {.vendor = 0x1102, .device = 0x0008, .driver = "Audigy2", .name = "Audigy 2 Value [Unknown]", .id = "Audigy2", @@ -1634,8 +1757,9 @@ goto error; } - emu->page_ptr_table = (void **)vmalloc(emu->max_cache_pages * sizeof(void*)); - emu->page_addr_table = (unsigned long*)vmalloc(emu->max_cache_pages * sizeof(unsigned long)); + emu->page_ptr_table = vmalloc(emu->max_cache_pages * sizeof(void *)); + emu->page_addr_table = vmalloc(emu->max_cache_pages * + sizeof(unsigned long)); if (emu->page_ptr_table == NULL || emu->page_addr_table == NULL) { err = -ENOMEM; goto error; @@ -1665,12 +1789,13 @@ emu->fx8010.extout_mask = extout_mask; emu->enable_ir = enable_ir; + if (emu->card_capabilities->ca_cardbus_chip) { + if ((err = snd_emu10k1_cardbus_init(emu)) < 0) + goto error; + } if (emu->card_capabilities->ecard) { if ((err = snd_emu10k1_ecard_init(emu)) < 0) goto error; - } else if (emu->card_capabilities->ca_cardbus_chip) { - if ((err = snd_emu10k1_cardbus_init(emu)) < 0) - goto error; } else if (emu->card_capabilities->emu1010) { if ((err = snd_emu10k1_emu1010_init(emu)) < 0) { snd_emu10k1_free(emu); @@ -1816,10 +1941,10 @@ void snd_emu10k1_resume_init(struct snd_emu10k1 *emu) { + if (emu->card_capabilities->ca_cardbus_chip) + snd_emu10k1_cardbus_init(emu); if (emu->card_capabilities->ecard) snd_emu10k1_ecard_init(emu); - else if (emu->card_capabilities->ca_cardbus_chip) - snd_emu10k1_cardbus_init(emu); else if (emu->card_capabilities->emu1010) snd_emu10k1_emu1010_init(emu); else diff -ruN linux-2.6.22/sound/pci/emu10k1/emu10k1_main.c.orig linux-2.6.22-alsa/sound/pci/emu10k1/emu10k1_main.c.orig --- linux-2.6.22/sound/pci/emu10k1/emu10k1_main.c.orig 1970-01-01 01:00:00.000000000 +0100 +++ linux-2.6.22-alsa/sound/pci/emu10k1/emu10k1_main.c.orig 2007-09-01 20:55:23.000000000 +0200 @@ -0,0 +1,1976 @@ +/* + * Copyright (c) by Jaroslav Kysela + * Creative Labs, Inc. + * Routines for control of EMU10K1 chips + * + * Copyright (c) by James Courtier-Dutton + * Added support for Audigy 2 Value. + * Added EMU 1010 support. + * General bug fixes and enhancements. + * + * + * BUGS: + * -- + * + * TODO: + * -- + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + + +#include +#include +#include +#include "p16v.h" +#include "tina2.h" +#include "p17v.h" + + +#define HANA_FILENAME "emu/hana.fw" +#define DOCK_FILENAME "emu/audio_dock.fw" +#define EMU1010B_FILENAME "emu/emu1010b.fw" +#define MICRO_DOCK_FILENAME "emu/micro_dock.fw" +#define EMU1010_NOTEBOOK_FILENAME "emu/emu1010_notebook.fw" + +MODULE_FIRMWARE(HANA_FILENAME); +MODULE_FIRMWARE(DOCK_FILENAME); +MODULE_FIRMWARE(EMU1010B_FILENAME); +MODULE_FIRMWARE(MICRO_DOCK_FILENAME); +MODULE_FIRMWARE(EMU1010_NOTEBOOK_FILENAME); + + +/************************************************************************* + * EMU10K1 init / done + *************************************************************************/ + +void snd_emu10k1_voice_init(struct snd_emu10k1 * emu, int ch) +{ + snd_emu10k1_ptr_write(emu, DCYSUSV, ch, 0); + snd_emu10k1_ptr_write(emu, IP, ch, 0); + snd_emu10k1_ptr_write(emu, VTFT, ch, 0xffff); + snd_emu10k1_ptr_write(emu, CVCF, ch, 0xffff); + snd_emu10k1_ptr_write(emu, PTRX, ch, 0); + snd_emu10k1_ptr_write(emu, CPF, ch, 0); + snd_emu10k1_ptr_write(emu, CCR, ch, 0); + + snd_emu10k1_ptr_write(emu, PSST, ch, 0); + snd_emu10k1_ptr_write(emu, DSL, ch, 0x10); + snd_emu10k1_ptr_write(emu, CCCA, ch, 0); + snd_emu10k1_ptr_write(emu, Z1, ch, 0); + snd_emu10k1_ptr_write(emu, Z2, ch, 0); + snd_emu10k1_ptr_write(emu, FXRT, ch, 0x32100000); + + snd_emu10k1_ptr_write(emu, ATKHLDM, ch, 0); + snd_emu10k1_ptr_write(emu, DCYSUSM, ch, 0); + snd_emu10k1_ptr_write(emu, IFATN, ch, 0xffff); + snd_emu10k1_ptr_write(emu, PEFE, ch, 0); + snd_emu10k1_ptr_write(emu, FMMOD, ch, 0); + snd_emu10k1_ptr_write(emu, TREMFRQ, ch, 24); /* 1 Hz */ + snd_emu10k1_ptr_write(emu, FM2FRQ2, ch, 24); /* 1 Hz */ + snd_emu10k1_ptr_write(emu, TEMPENV, ch, 0); + + /*** these are last so OFF prevents writing ***/ + snd_emu10k1_ptr_write(emu, LFOVAL2, ch, 0); + snd_emu10k1_ptr_write(emu, LFOVAL1, ch, 0); + snd_emu10k1_ptr_write(emu, ATKHLDV, ch, 0); + snd_emu10k1_ptr_write(emu, ENVVOL, ch, 0); + snd_emu10k1_ptr_write(emu, ENVVAL, ch, 0); + + /* Audigy extra stuffs */ + if (emu->audigy) { + snd_emu10k1_ptr_write(emu, 0x4c, ch, 0); /* ?? */ + snd_emu10k1_ptr_write(emu, 0x4d, ch, 0); /* ?? */ + snd_emu10k1_ptr_write(emu, 0x4e, ch, 0); /* ?? */ + snd_emu10k1_ptr_write(emu, 0x4f, ch, 0); /* ?? */ + snd_emu10k1_ptr_write(emu, A_FXRT1, ch, 0x03020100); + snd_emu10k1_ptr_write(emu, A_FXRT2, ch, 0x3f3f3f3f); + snd_emu10k1_ptr_write(emu, A_SENDAMOUNTS, ch, 0); + } +} + +static unsigned int spi_dac_init[] = { + 0x00ff, + 0x02ff, + 0x0400, + 0x0520, + 0x0600, + 0x08ff, + 0x0aff, + 0x0cff, + 0x0eff, + 0x10ff, + 0x1200, + 0x1400, + 0x1480, + 0x1800, + 0x1aff, + 0x1cff, + 0x1e00, + 0x0530, + 0x0602, + 0x0622, + 0x1400, +}; + +static unsigned int i2c_adc_init[][2] = { + { 0x17, 0x00 }, /* Reset */ + { 0x07, 0x00 }, /* Timeout */ + { 0x0b, 0x22 }, /* Interface control */ + { 0x0c, 0x22 }, /* Master mode control */ + { 0x0d, 0x08 }, /* Powerdown control */ + { 0x0e, 0xcf }, /* Attenuation Left 0x01 = -103dB, 0xff = 24dB */ + { 0x0f, 0xcf }, /* Attenuation Right 0.5dB steps */ + { 0x10, 0x7b }, /* ALC Control 1 */ + { 0x11, 0x00 }, /* ALC Control 2 */ + { 0x12, 0x32 }, /* ALC Control 3 */ + { 0x13, 0x00 }, /* Noise gate control */ + { 0x14, 0xa6 }, /* Limiter control */ + { 0x15, ADC_MUX_2 }, /* ADC Mixer control. Mic for Audigy 2 ZS Notebook */ +}; + +static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir, int resume) +{ + unsigned int silent_page; + int ch; + u32 tmp; + + /* disable audio and lock cache */ + outl(HCFG_LOCKSOUNDCACHE | HCFG_LOCKTANKCACHE_MASK | HCFG_MUTEBUTTONENABLE, + emu->port + HCFG); + + /* reset recording buffers */ + snd_emu10k1_ptr_write(emu, MICBS, 0, ADCBS_BUFSIZE_NONE); + snd_emu10k1_ptr_write(emu, MICBA, 0, 0); + snd_emu10k1_ptr_write(emu, FXBS, 0, ADCBS_BUFSIZE_NONE); + snd_emu10k1_ptr_write(emu, FXBA, 0, 0); + snd_emu10k1_ptr_write(emu, ADCBS, 0, ADCBS_BUFSIZE_NONE); + snd_emu10k1_ptr_write(emu, ADCBA, 0, 0); + + /* disable channel interrupt */ + outl(0, emu->port + INTE); + snd_emu10k1_ptr_write(emu, CLIEL, 0, 0); + snd_emu10k1_ptr_write(emu, CLIEH, 0, 0); + snd_emu10k1_ptr_write(emu, SOLEL, 0, 0); + snd_emu10k1_ptr_write(emu, SOLEH, 0, 0); + + if (emu->audigy){ + /* set SPDIF bypass mode */ + snd_emu10k1_ptr_write(emu, SPBYPASS, 0, SPBYPASS_FORMAT); + /* enable rear left + rear right AC97 slots */ + snd_emu10k1_ptr_write(emu, AC97SLOT, 0, AC97SLOT_REAR_RIGHT | + AC97SLOT_REAR_LEFT); + } + + /* init envelope engine */ + for (ch = 0; ch < NUM_G; ch++) + snd_emu10k1_voice_init(emu, ch); + + snd_emu10k1_ptr_write(emu, SPCS0, 0, emu->spdif_bits[0]); + snd_emu10k1_ptr_write(emu, SPCS1, 0, emu->spdif_bits[1]); + snd_emu10k1_ptr_write(emu, SPCS2, 0, emu->spdif_bits[2]); + + if (emu->card_capabilities->ca0151_chip) { /* audigy2 */ + /* Hacks for Alice3 to work independent of haP16V driver */ + //Setup SRCMulti_I2S SamplingRate + tmp = snd_emu10k1_ptr_read(emu, A_SPDIF_SAMPLERATE, 0); + tmp &= 0xfffff1ff; + tmp |= (0x2<<9); + snd_emu10k1_ptr_write(emu, A_SPDIF_SAMPLERATE, 0, tmp); + + /* Setup SRCSel (Enable Spdif,I2S SRCMulti) */ + snd_emu10k1_ptr20_write(emu, SRCSel, 0, 0x14); + /* Setup SRCMulti Input Audio Enable */ + /* Use 0xFFFFFFFF to enable P16V sounds. */ + snd_emu10k1_ptr20_write(emu, SRCMULTI_ENABLE, 0, 0xFFFFFFFF); + + /* Enabled Phased (8-channel) P16V playback */ + outl(0x0201, emu->port + HCFG2); + /* Set playback routing. */ + snd_emu10k1_ptr20_write(emu, CAPTURE_P16V_SOURCE, 0, 0x78e4); + } + if (emu->card_capabilities->ca0108_chip) { /* audigy2 Value */ + /* Hacks for Alice3 to work independent of haP16V driver */ + snd_printk(KERN_INFO "Audigy2 value: Special config.\n"); + //Setup SRCMulti_I2S SamplingRate + tmp = snd_emu10k1_ptr_read(emu, A_SPDIF_SAMPLERATE, 0); + tmp &= 0xfffff1ff; + tmp |= (0x2<<9); + snd_emu10k1_ptr_write(emu, A_SPDIF_SAMPLERATE, 0, tmp); + + /* Setup SRCSel (Enable Spdif,I2S SRCMulti) */ + outl(0x600000, emu->port + 0x20); + outl(0x14, emu->port + 0x24); + + /* Setup SRCMulti Input Audio Enable */ + outl(0x7b0000, emu->port + 0x20); + outl(0xFF000000, emu->port + 0x24); + + /* Setup SPDIF Out Audio Enable */ + /* The Audigy 2 Value has a separate SPDIF out, + * so no need for a mixer switch + */ + outl(0x7a0000, emu->port + 0x20); + outl(0xFF000000, emu->port + 0x24); + tmp = inl(emu->port + A_IOCFG) & ~0x8; /* Clear bit 3 */ + outl(tmp, emu->port + A_IOCFG); + } + if (emu->card_capabilities->spi_dac) { /* Audigy 2 ZS Notebook with DAC Wolfson WM8768/WM8568 */ + int size, n; + + size = ARRAY_SIZE(spi_dac_init); + for (n = 0; n < size; n++) + snd_emu10k1_spi_write(emu, spi_dac_init[n]); + + snd_emu10k1_ptr20_write(emu, 0x60, 0, 0x10); + /* Enable GPIOs + * GPIO0: Unknown + * GPIO1: Speakers-enabled. + * GPIO2: Unknown + * GPIO3: Unknown + * GPIO4: IEC958 Output on. + * GPIO5: Unknown + * GPIO6: Unknown + * GPIO7: Unknown + */ + outl(0x76, emu->port + A_IOCFG); /* Windows uses 0x3f76 */ + + } + if (emu->card_capabilities->i2c_adc) { /* Audigy 2 ZS Notebook with ADC Wolfson WM8775 */ + int size, n; + + snd_emu10k1_ptr20_write(emu, P17V_I2S_SRC_SEL, 0, 0x2020205f); + tmp = inl(emu->port + A_IOCFG); + outl(tmp | 0x4, emu->port + A_IOCFG); /* Set bit 2 for mic input */ + tmp = inl(emu->port + A_IOCFG); + size = ARRAY_SIZE(i2c_adc_init); + for (n = 0; n < size; n++) + snd_emu10k1_i2c_write(emu, i2c_adc_init[n][0], i2c_adc_init[n][1]); + for (n=0; n < 4; n++) { + emu->i2c_capture_volume[n][0]= 0xcf; + emu->i2c_capture_volume[n][1]= 0xcf; + } + + } + + + snd_emu10k1_ptr_write(emu, PTB, 0, emu->ptb_pages.addr); + snd_emu10k1_ptr_write(emu, TCB, 0, 0); /* taken from original driver */ + snd_emu10k1_ptr_write(emu, TCBS, 0, 4); /* taken from original driver */ + + silent_page = (emu->silent_page.addr << 1) | MAP_PTI_MASK; + for (ch = 0; ch < NUM_G; ch++) { + snd_emu10k1_ptr_write(emu, MAPA, ch, silent_page); + snd_emu10k1_ptr_write(emu, MAPB, ch, silent_page); + } + + if (emu->card_capabilities->emu1010) { + outl(HCFG_AUTOMUTE_ASYNC | + HCFG_EMU32_SLAVE | + HCFG_AUDIOENABLE, emu->port + HCFG); + /* + * Hokay, setup HCFG + * Mute Disable Audio = 0 + * Lock Tank Memory = 1 + * Lock Sound Memory = 0 + * Auto Mute = 1 + */ + } else if (emu->audigy) { + if (emu->revision == 4) /* audigy2 */ + outl(HCFG_AUDIOENABLE | + HCFG_AC3ENABLE_CDSPDIF | + HCFG_AC3ENABLE_GPSPDIF | + HCFG_AUTOMUTE | HCFG_JOYENABLE, emu->port + HCFG); + else + outl(HCFG_AUTOMUTE | HCFG_JOYENABLE, emu->port + HCFG); + /* FIXME: Remove all these emu->model and replace it with a card recognition parameter, + * e.g. card_capabilities->joystick */ + } else if (emu->model == 0x20 || + emu->model == 0xc400 || + (emu->model == 0x21 && emu->revision < 6)) + outl(HCFG_LOCKTANKCACHE_MASK | HCFG_AUTOMUTE, emu->port + HCFG); + else + // With on-chip joystick + outl(HCFG_LOCKTANKCACHE_MASK | HCFG_AUTOMUTE | HCFG_JOYENABLE, emu->port + HCFG); + + if (enable_ir) { /* enable IR for SB Live */ + if (emu->card_capabilities->emu1010) { + ; /* Disable all access to A_IOCFG for the emu1010 */ + } else if (emu->card_capabilities->i2c_adc) { + ; /* Disable A_IOCFG for Audigy 2 ZS Notebook */ + } else if (emu->audigy) { + unsigned int reg = inl(emu->port + A_IOCFG); + outl(reg | A_IOCFG_GPOUT2, emu->port + A_IOCFG); + udelay(500); + outl(reg | A_IOCFG_GPOUT1 | A_IOCFG_GPOUT2, emu->port + A_IOCFG); + udelay(100); + outl(reg, emu->port + A_IOCFG); + } else { + unsigned int reg = inl(emu->port + HCFG); + outl(reg | HCFG_GPOUT2, emu->port + HCFG); + udelay(500); + outl(reg | HCFG_GPOUT1 | HCFG_GPOUT2, emu->port + HCFG); + udelay(100); + outl(reg, emu->port + HCFG); + } + } + + if (emu->card_capabilities->emu1010) { + ; /* Disable all access to A_IOCFG for the emu1010 */ + } else if (emu->card_capabilities->i2c_adc) { + ; /* Disable A_IOCFG for Audigy 2 ZS Notebook */ + } else if (emu->audigy) { /* enable analog output */ + unsigned int reg = inl(emu->port + A_IOCFG); + outl(reg | A_IOCFG_GPOUT0, emu->port + A_IOCFG); + } + + return 0; +} + +static void snd_emu10k1_audio_enable(struct snd_emu10k1 *emu) +{ + /* + * Enable the audio bit + */ + outl(inl(emu->port + HCFG) | HCFG_AUDIOENABLE, emu->port + HCFG); + + /* Enable analog/digital outs on audigy */ + if (emu->card_capabilities->emu1010) { + ; /* Disable all access to A_IOCFG for the emu1010 */ + } else if (emu->card_capabilities->i2c_adc) { + ; /* Disable A_IOCFG for Audigy 2 ZS Notebook */ + } else if (emu->audigy) { + outl(inl(emu->port + A_IOCFG) & ~0x44, emu->port + A_IOCFG); + + if (emu->card_capabilities->ca0151_chip) { /* audigy2 */ + /* Unmute Analog now. Set GPO6 to 1 for Apollo. + * This has to be done after init ALice3 I2SOut beyond 48KHz. + * So, sequence is important. */ + outl(inl(emu->port + A_IOCFG) | 0x0040, emu->port + A_IOCFG); + } else if (emu->card_capabilities->ca0108_chip) { /* audigy2 value */ + /* Unmute Analog now. */ + outl(inl(emu->port + A_IOCFG) | 0x0060, emu->port + A_IOCFG); + } else { + /* Disable routing from AC97 line out to Front speakers */ + outl(inl(emu->port + A_IOCFG) | 0x0080, emu->port + A_IOCFG); + } + } + +#if 0 + { + unsigned int tmp; + /* FIXME: the following routine disables LiveDrive-II !! */ + // TOSLink detection + emu->tos_link = 0; + tmp = inl(emu->port + HCFG); + if (tmp & (HCFG_GPINPUT0 | HCFG_GPINPUT1)) { + outl(tmp|0x800, emu->port + HCFG); + udelay(50); + if (tmp != (inl(emu->port + HCFG) & ~0x800)) { + emu->tos_link = 1; + outl(tmp, emu->port + HCFG); + } + } + } +#endif + + snd_emu10k1_intr_enable(emu, INTE_PCIERRORENABLE); +} + +int snd_emu10k1_done(struct snd_emu10k1 * emu) +{ + int ch; + + outl(0, emu->port + INTE); + + /* + * Shutdown the chip + */ + for (ch = 0; ch < NUM_G; ch++) + snd_emu10k1_ptr_write(emu, DCYSUSV, ch, 0); + for (ch = 0; ch < NUM_G; ch++) { + snd_emu10k1_ptr_write(emu, VTFT, ch, 0); + snd_emu10k1_ptr_write(emu, CVCF, ch, 0); + snd_emu10k1_ptr_write(emu, PTRX, ch, 0); + snd_emu10k1_ptr_write(emu, CPF, ch, 0); + } + + /* reset recording buffers */ + snd_emu10k1_ptr_write(emu, MICBS, 0, 0); + snd_emu10k1_ptr_write(emu, MICBA, 0, 0); + snd_emu10k1_ptr_write(emu, FXBS, 0, 0); + snd_emu10k1_ptr_write(emu, FXBA, 0, 0); + snd_emu10k1_ptr_write(emu, FXWC, 0, 0); + snd_emu10k1_ptr_write(emu, ADCBS, 0, ADCBS_BUFSIZE_NONE); + snd_emu10k1_ptr_write(emu, ADCBA, 0, 0); + snd_emu10k1_ptr_write(emu, TCBS, 0, TCBS_BUFFSIZE_16K); + snd_emu10k1_ptr_write(emu, TCB, 0, 0); + if (emu->audigy) + snd_emu10k1_ptr_write(emu, A_DBG, 0, A_DBG_SINGLE_STEP); + else + snd_emu10k1_ptr_write(emu, DBG, 0, EMU10K1_DBG_SINGLE_STEP); + + /* disable channel interrupt */ + snd_emu10k1_ptr_write(emu, CLIEL, 0, 0); + snd_emu10k1_ptr_write(emu, CLIEH, 0, 0); + snd_emu10k1_ptr_write(emu, SOLEL, 0, 0); + snd_emu10k1_ptr_write(emu, SOLEH, 0, 0); + + /* disable audio and lock cache */ + outl(HCFG_LOCKSOUNDCACHE | HCFG_LOCKTANKCACHE_MASK | HCFG_MUTEBUTTONENABLE, emu->port + HCFG); + snd_emu10k1_ptr_write(emu, PTB, 0, 0); + + return 0; +} + +/************************************************************************* + * ECARD functional implementation + *************************************************************************/ + +/* In A1 Silicon, these bits are in the HC register */ +#define HOOKN_BIT (1L << 12) +#define HANDN_BIT (1L << 11) +#define PULSEN_BIT (1L << 10) + +#define EC_GDI1 (1 << 13) +#define EC_GDI0 (1 << 14) + +#define EC_NUM_CONTROL_BITS 20 + +#define EC_AC3_DATA_SELN 0x0001L +#define EC_EE_DATA_SEL 0x0002L +#define EC_EE_CNTRL_SELN 0x0004L +#define EC_EECLK 0x0008L +#define EC_EECS 0x0010L +#define EC_EESDO 0x0020L +#define EC_TRIM_CSN 0x0040L +#define EC_TRIM_SCLK 0x0080L +#define EC_TRIM_SDATA 0x0100L +#define EC_TRIM_MUTEN 0x0200L +#define EC_ADCCAL 0x0400L +#define EC_ADCRSTN 0x0800L +#define EC_DACCAL 0x1000L +#define EC_DACMUTEN 0x2000L +#define EC_LEDN 0x4000L + +#define EC_SPDIF0_SEL_SHIFT 15 +#define EC_SPDIF1_SEL_SHIFT 17 +#define EC_SPDIF0_SEL_MASK (0x3L << EC_SPDIF0_SEL_SHIFT) +#define EC_SPDIF1_SEL_MASK (0x7L << EC_SPDIF1_SEL_SHIFT) +#define EC_SPDIF0_SELECT(_x) (((_x) << EC_SPDIF0_SEL_SHIFT) & EC_SPDIF0_SEL_MASK) +#define EC_SPDIF1_SELECT(_x) (((_x) << EC_SPDIF1_SEL_SHIFT) & EC_SPDIF1_SEL_MASK) +#define EC_CURRENT_PROM_VERSION 0x01 /* Self-explanatory. This should + * be incremented any time the EEPROM's + * format is changed. */ + +#define EC_EEPROM_SIZE 0x40 /* ECARD EEPROM has 64 16-bit words */ + +/* Addresses for special values stored in to EEPROM */ +#define EC_PROM_VERSION_ADDR 0x20 /* Address of the current prom version */ +#define EC_BOARDREV0_ADDR 0x21 /* LSW of board rev */ +#define EC_BOARDREV1_ADDR 0x22 /* MSW of board rev */ + +#define EC_LAST_PROMFILE_ADDR 0x2f + +#define EC_SERIALNUM_ADDR 0x30 /* First word of serial number. The + * can be up to 30 characters in length + * and is stored as a NULL-terminated + * ASCII string. Any unused bytes must be + * filled with zeros */ +#define EC_CHECKSUM_ADDR 0x3f /* Location at which checksum is stored */ + + +/* Most of this stuff is pretty self-evident. According to the hardware + * dudes, we need to leave the ADCCAL bit low in order to avoid a DC + * offset problem. Weird. + */ +#define EC_RAW_RUN_MODE (EC_DACMUTEN | EC_ADCRSTN | EC_TRIM_MUTEN | \ + EC_TRIM_CSN) + + +#define EC_DEFAULT_ADC_GAIN 0xC4C4 +#define EC_DEFAULT_SPDIF0_SEL 0x0 +#define EC_DEFAULT_SPDIF1_SEL 0x4 + +/************************************************************************** + * @func Clock bits into the Ecard's control latch. The Ecard uses a + * control latch will is loaded bit-serially by toggling the Modem control + * lines from function 2 on the E8010. This function hides these details + * and presents the illusion that we are actually writing to a distinct + * register. + */ + +static void snd_emu10k1_ecard_write(struct snd_emu10k1 * emu, unsigned int value) +{ + unsigned short count; + unsigned int data; + unsigned long hc_port; + unsigned int hc_value; + + hc_port = emu->port + HCFG; + hc_value = inl(hc_port) & ~(HOOKN_BIT | HANDN_BIT | PULSEN_BIT); + outl(hc_value, hc_port); + + for (count = 0; count < EC_NUM_CONTROL_BITS; count++) { + + /* Set up the value */ + data = ((value & 0x1) ? PULSEN_BIT : 0); + value >>= 1; + + outl(hc_value | data, hc_port); + + /* Clock the shift register */ + outl(hc_value | data | HANDN_BIT, hc_port); + outl(hc_value | data, hc_port); + } + + /* Latch the bits */ + outl(hc_value | HOOKN_BIT, hc_port); + outl(hc_value, hc_port); +} + +/************************************************************************** + * @func Set the gain of the ECARD's CS3310 Trim/gain controller. The + * trim value consists of a 16bit value which is composed of two + * 8 bit gain/trim values, one for the left channel and one for the + * right channel. The following table maps from the Gain/Attenuation + * value in decibels into the corresponding bit pattern for a single + * channel. + */ + +static void snd_emu10k1_ecard_setadcgain(struct snd_emu10k1 * emu, + unsigned short gain) +{ + unsigned int bit; + + /* Enable writing to the TRIM registers */ + snd_emu10k1_ecard_write(emu, emu->ecard_ctrl & ~EC_TRIM_CSN); + + /* Do it again to insure that we meet hold time requirements */ + snd_emu10k1_ecard_write(emu, emu->ecard_ctrl & ~EC_TRIM_CSN); + + for (bit = (1 << 15); bit; bit >>= 1) { + unsigned int value; + + value = emu->ecard_ctrl & ~(EC_TRIM_CSN | EC_TRIM_SDATA); + + if (gain & bit) + value |= EC_TRIM_SDATA; + + /* Clock the bit */ + snd_emu10k1_ecard_write(emu, value); + snd_emu10k1_ecard_write(emu, value | EC_TRIM_SCLK); + snd_emu10k1_ecard_write(emu, value); + } + + snd_emu10k1_ecard_write(emu, emu->ecard_ctrl); +} + +static int snd_emu10k1_ecard_init(struct snd_emu10k1 * emu) +{ + unsigned int hc_value; + + /* Set up the initial settings */ + emu->ecard_ctrl = EC_RAW_RUN_MODE | + EC_SPDIF0_SELECT(EC_DEFAULT_SPDIF0_SEL) | + EC_SPDIF1_SELECT(EC_DEFAULT_SPDIF1_SEL); + + /* Step 0: Set the codec type in the hardware control register + * and enable audio output */ + hc_value = inl(emu->port + HCFG); + outl(hc_value | HCFG_AUDIOENABLE | HCFG_CODECFORMAT_I2S, emu->port + HCFG); + inl(emu->port + HCFG); + + /* Step 1: Turn off the led and deassert TRIM_CS */ + snd_emu10k1_ecard_write(emu, EC_ADCCAL | EC_LEDN | EC_TRIM_CSN); + + /* Step 2: Calibrate the ADC and DAC */ + snd_emu10k1_ecard_write(emu, EC_DACCAL | EC_LEDN | EC_TRIM_CSN); + + /* Step 3: Wait for awhile; XXX We can't get away with this + * under a real operating system; we'll need to block and wait that + * way. */ + snd_emu10k1_wait(emu, 48000); + + /* Step 4: Switch off the DAC and ADC calibration. Note + * That ADC_CAL is actually an inverted signal, so we assert + * it here to stop calibration. */ + snd_emu10k1_ecard_write(emu, EC_ADCCAL | EC_LEDN | EC_TRIM_CSN); + + /* Step 4: Switch into run mode */ + snd_emu10k1_ecard_write(emu, emu->ecard_ctrl); + + /* Step 5: Set the analog input gain */ + snd_emu10k1_ecard_setadcgain(emu, EC_DEFAULT_ADC_GAIN); + + return 0; +} + +static int snd_emu10k1_cardbus_init(struct snd_emu10k1 * emu) +{ + unsigned long special_port; + unsigned int value; + + /* Special initialisation routine + * before the rest of the IO-Ports become active. + */ + special_port = emu->port + 0x38; + value = inl(special_port); + outl(0x00d00000, special_port); + value = inl(special_port); + outl(0x00d00001, special_port); + value = inl(special_port); + outl(0x00d0005f, special_port); + value = inl(special_port); + outl(0x00d0007f, special_port); + value = inl(special_port); + outl(0x0090007f, special_port); + value = inl(special_port); + + snd_emu10k1_ptr20_write(emu, TINA2_VOLUME, 0, 0xfefefefe); /* Defaults to 0x30303030 */ + return 0; +} + +static int snd_emu1010_load_firmware(struct snd_emu10k1 * emu, const char * filename) +{ + int err; + int n, i; + int reg; + int value; + const struct firmware *fw_entry; + + if ((err = request_firmware(&fw_entry, filename, &emu->pci->dev)) != 0) { + snd_printk(KERN_ERR "firmware: %s not found. Err=%d\n",filename, err); + return err; + } + snd_printk(KERN_INFO "firmware size=0x%zx\n", fw_entry->size); +#if 0 + if (fw_entry->size != 0x133a4) { + snd_printk(KERN_ERR "firmware: %s wrong size.\n",filename); + return -EINVAL; + } +#endif + + /* The FPGA is a Xilinx Spartan IIE XC2S50E */ + /* GPIO7 -> FPGA PGMN + * GPIO6 -> FPGA CCLK + * GPIO5 -> FPGA DIN + * FPGA CONFIG OFF -> FPGA PGMN + */ + outl(0x00, emu->port + A_IOCFG); /* Set PGMN low for 1uS. */ + udelay(1); + outl(0x80, emu->port + A_IOCFG); /* Leave bit 7 set during netlist setup. */ + udelay(100); /* Allow FPGA memory to clean */ + for(n = 0; n < fw_entry->size; n++) { + value=fw_entry->data[n]; + for(i = 0; i < 8; i++) { + reg = 0x80; + if (value & 0x1) + reg = reg | 0x20; + value = value >> 1; + outl(reg, emu->port + A_IOCFG); + outl(reg | 0x40, emu->port + A_IOCFG); + } + } + /* After programming, set GPIO bit 4 high again. */ + outl(0x10, emu->port + A_IOCFG); + + + release_firmware(fw_entry); + return 0; +} + +int emu1010_firmware_thread(void *data) { + struct snd_emu10k1 * emu = data; + int tmp,tmp2; + int reg; + int err; + + for (;;) { + /* Delay to allow Audio Dock to settle */ + msleep(1000); + if (kthread_should_stop()) + break; + snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, &tmp ); /* IRQ Status */ + snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, ® ); /* OPTIONS: Which cards are attached to the EMU */ + if (reg & EMU_HANA_OPTION_DOCK_OFFLINE) { + /* Audio Dock attached */ + /* Return to Audio Dock programming mode */ + snd_printk(KERN_INFO "emu1010: Loading Audio Dock Firmware\n"); + snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, EMU_HANA_FPGA_CONFIG_AUDIODOCK ); + if (emu->card_capabilities->emu1010 == 1) { + if ((err = snd_emu1010_load_firmware(emu, DOCK_FILENAME)) != 0) { + return err; + } + } else if (emu->card_capabilities->emu1010 == 2) { + if ((err = snd_emu1010_load_firmware(emu, MICRO_DOCK_FILENAME)) != 0) { + return err; + } + } else if (emu->card_capabilities->emu1010 == 3) { + if ((err = snd_emu1010_load_firmware(emu, MICRO_DOCK_FILENAME)) != 0) { + return err; + } + } + + snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, 0 ); + snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, ® ); + snd_printk(KERN_INFO "emu1010: EMU_HANA+DOCK_IRQ_STATUS=0x%x\n",reg); + /* ID, should read & 0x7f = 0x55 when FPGA programmed. */ + snd_emu1010_fpga_read(emu, EMU_HANA_ID, ® ); + snd_printk(KERN_INFO "emu1010: EMU_HANA+DOCK_ID=0x%x\n",reg); + if ((reg & 0x1f) != 0x15) { + /* FPGA failed to be programmed */ + snd_printk(KERN_INFO "emu1010: Loading Audio Dock Firmware file failed, reg=0x%x\n", reg); + return 0; + return -ENODEV; + } + snd_printk(KERN_INFO "emu1010: Audio Dock Firmware loaded\n"); + snd_emu1010_fpga_read(emu, EMU_DOCK_MAJOR_REV, &tmp ); + snd_emu1010_fpga_read(emu, EMU_DOCK_MINOR_REV, &tmp2 ); + snd_printk("Audio Dock ver:%d.%d\n",tmp ,tmp2); + /* Sync clocking between 1010 and Dock */ + /* Allow DLL to settle */ + msleep(10); + /* Unmute all. Default is muted after a firmware load */ + snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE ); + break; + } + } + return 0; +} + +/* + * EMU-1010 - details found out from this driver, official MS Win drivers, + * testing the card: + * + * Audigy2 (aka Alice2): + * --------------------- + * * communication over PCI + * * conversion of 32-bit data coming over EMU32 links from HANA FPGA + * to 2 x 16-bit, using internal DSP instructions + * * slave mode, clock supplied by HANA + * * linked to HANA using: + * 32 x 32-bit serial EMU32 output channels + * 16 x EMU32 input channels + * (?) x I2S I/O channels (?) + * + * FPGA (aka HANA): + * --------------- + * * provides all (?) physical inputs and outputs of the card + * (ADC, DAC, SPDIF I/O, ADAT I/O, etc.) + * * provides clock signal for the card and Alice2 + * * two crystals - for 44.1kHz and 48kHz multiples + * * provides internal routing of signal sources to signal destinations + * * inputs/outputs to Alice2 - see above + * + * Current status of the driver: + * ---------------------------- + * * only 44.1/48kHz supported (the MS Win driver supports up to 192 kHz) + * * PCM device nb. 2: + * 16 x 16-bit playback - snd_emu10k1_fx8010_playback_ops + * 16 x 32-bit capture - snd_emu10k1_capture_efx_ops + */ +static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) +{ + unsigned int i; + int tmp,tmp2; + int reg; + int err; + + snd_printk(KERN_INFO "emu1010: Special config.\n"); + /* AC97 2.1, Any 16Meg of 4Gig address, Auto-Mute, EMU32 Slave, + * Lock Sound Memory Cache, Lock Tank Memory Cache, + * Mute all codecs. + */ + outl(0x0005a00c, emu->port + HCFG); + /* AC97 2.1, Any 16Meg of 4Gig address, Auto-Mute, EMU32 Slave, + * Lock Tank Memory Cache, + * Mute all codecs. + */ + outl(0x0005a004, emu->port + HCFG); + /* AC97 2.1, Any 16Meg of 4Gig address, Auto-Mute, EMU32 Slave, + * Mute all codecs. + */ + outl(0x0005a000, emu->port + HCFG); + /* AC97 2.1, Any 16Meg of 4Gig address, Auto-Mute, EMU32 Slave, + * Mute all codecs. + */ + outl(0x0005a000, emu->port + HCFG); + + /* Disable 48Volt power to Audio Dock */ + snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, 0 ); + + /* ID, should read & 0x7f = 0x55. (Bit 7 is the IRQ bit) */ + snd_emu1010_fpga_read(emu, EMU_HANA_ID, ® ); + snd_printdd("reg1=0x%x\n",reg); + if ((reg & 0x3f) == 0x15) { + /* FPGA netlist already present so clear it */ + /* Return to programming mode */ + + snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, 0x02 ); + } + snd_emu1010_fpga_read(emu, EMU_HANA_ID, ® ); + snd_printdd("reg2=0x%x\n",reg); + if ((reg & 0x3f) == 0x15) { + /* FPGA failed to return to programming mode */ + snd_printk(KERN_INFO "emu1010: FPGA failed to return to programming mode\n"); + return -ENODEV; + } + snd_printk(KERN_INFO "emu1010: EMU_HANA_ID=0x%x\n",reg); + if (emu->card_capabilities->emu1010 == 1) { + if ((err = snd_emu1010_load_firmware(emu, HANA_FILENAME)) != 0) { + snd_printk(KERN_INFO "emu1010: Loading Hana Firmware file %s failed\n", HANA_FILENAME); + return err; + } + } else if (emu->card_capabilities->emu1010 == 2) { + if ((err = snd_emu1010_load_firmware(emu, EMU1010B_FILENAME)) != 0) { + snd_printk(KERN_INFO "emu1010: Loading Firmware file %s failed\n", EMU1010B_FILENAME); + return err; + } + } else if (emu->card_capabilities->emu1010 == 3) { + if ((err = snd_emu1010_load_firmware(emu, EMU1010_NOTEBOOK_FILENAME)) != 0) { + snd_printk(KERN_INFO "emu1010: Loading Firmware file %s failed\n", EMU1010_NOTEBOOK_FILENAME); + return err; + } + } + + /* ID, should read & 0x7f = 0x55 when FPGA programmed. */ + snd_emu1010_fpga_read(emu, EMU_HANA_ID, ® ); + if ((reg & 0x3f) != 0x15) { + /* FPGA failed to be programmed */ + snd_printk(KERN_INFO "emu1010: Loading Hana Firmware file failed, reg=0x%x\n", reg); + return -ENODEV; + } + + snd_printk(KERN_INFO "emu1010: Hana Firmware loaded\n"); + snd_emu1010_fpga_read(emu, EMU_HANA_MAJOR_REV, &tmp ); + snd_emu1010_fpga_read(emu, EMU_HANA_MINOR_REV, &tmp2 ); + snd_printk("Hana ver:%d.%d\n",tmp ,tmp2); + /* Enable 48Volt power to Audio Dock */ + snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, EMU_HANA_DOCK_PWR_ON ); + + snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, ® ); + snd_printk(KERN_INFO "emu1010: Card options=0x%x\n",reg); + snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, ® ); + snd_printk(KERN_INFO "emu1010: Card options=0x%x\n",reg); + snd_emu1010_fpga_read(emu, EMU_HANA_OPTICAL_TYPE, &tmp ); + /* Optical -> ADAT I/O */ + /* 0 : SPDIF + * 1 : ADAT + */ + emu->emu1010.optical_in = 1; /* IN_ADAT */ + emu->emu1010.optical_out = 1; /* IN_ADAT */ + tmp = 0; + tmp = (emu->emu1010.optical_in ? EMU_HANA_OPTICAL_IN_ADAT : 0) | + (emu->emu1010.optical_out ? EMU_HANA_OPTICAL_OUT_ADAT : 0); + snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, tmp ); + snd_emu1010_fpga_read(emu, EMU_HANA_ADC_PADS, &tmp ); + /* Set no attenuation on Audio Dock pads. */ + snd_emu1010_fpga_write(emu, EMU_HANA_ADC_PADS, 0x00 ); + emu->emu1010.adc_pads = 0x00; + snd_emu1010_fpga_read(emu, EMU_HANA_DOCK_MISC, &tmp ); + /* Unmute Audio dock DACs, Headphone source DAC-4. */ + snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_MISC, 0x30 ); + snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, 0x12 ); + snd_emu1010_fpga_read(emu, EMU_HANA_DAC_PADS, &tmp ); + /* DAC PADs. */ + snd_emu1010_fpga_write(emu, EMU_HANA_DAC_PADS, 0x0f ); + emu->emu1010.dac_pads = 0x0f; + snd_emu1010_fpga_read(emu, EMU_HANA_DOCK_MISC, &tmp ); + snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_MISC, 0x30 ); + snd_emu1010_fpga_read(emu, EMU_HANA_SPDIF_MODE, &tmp ); + /* SPDIF Format. Set Consumer mode, 24bit, copy enable */ + snd_emu1010_fpga_write(emu, EMU_HANA_SPDIF_MODE, 0x10 ); + /* MIDI routing */ + snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_IN, 0x19 ); + /* Unknown. */ + snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_OUT, 0x0c ); + /* snd_emu1010_fpga_write(emu, 0x09, 0x0f ); // IRQ Enable: All on */ + /* IRQ Enable: All off */ + snd_emu1010_fpga_write(emu, EMU_HANA_IRQ_ENABLE, 0x00 ); + + snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, ® ); + snd_printk(KERN_INFO "emu1010: Card options3=0x%x\n",reg); + /* Default WCLK set to 48kHz. */ + snd_emu1010_fpga_write(emu, EMU_HANA_DEFCLOCK, 0x00 ); + /* Word Clock source, Internal 48kHz x1 */ + snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, EMU_HANA_WCLOCK_INT_48K ); + //snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, EMU_HANA_WCLOCK_INT_48K | EMU_HANA_WCLOCK_4X ); + /* Audio Dock LEDs. */ + snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, 0x12 ); + +#if 0 + /* For 96kHz */ + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_0, EMU_SRC_HAMOA_ADC_LEFT1); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_1, EMU_SRC_HAMOA_ADC_RIGHT1); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_4, EMU_SRC_HAMOA_ADC_LEFT2); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_5, EMU_SRC_HAMOA_ADC_RIGHT2); +#endif +#if 0 + /* For 192kHz */ + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_0, EMU_SRC_HAMOA_ADC_LEFT1); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_1, EMU_SRC_HAMOA_ADC_RIGHT1); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_2, EMU_SRC_HAMOA_ADC_LEFT2); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_3, EMU_SRC_HAMOA_ADC_RIGHT2); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_4, EMU_SRC_HAMOA_ADC_LEFT3); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_5, EMU_SRC_HAMOA_ADC_RIGHT3); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_6, EMU_SRC_HAMOA_ADC_LEFT4); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_7, EMU_SRC_HAMOA_ADC_RIGHT4); +#endif +#if 1 + /* For 48kHz */ + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_0, EMU_SRC_DOCK_MIC_A1); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_1, EMU_SRC_DOCK_MIC_B1); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_2, EMU_SRC_HAMOA_ADC_LEFT2); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_3, EMU_SRC_HAMOA_ADC_LEFT2); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_4, EMU_SRC_DOCK_ADC1_LEFT1); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_5, EMU_SRC_DOCK_ADC1_RIGHT1); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_6, EMU_SRC_DOCK_ADC2_LEFT1); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_7, EMU_SRC_DOCK_ADC2_RIGHT1); + /* Pavel Hofman - setting defaults for 8 more capture channels + * Defaults only, users will set their own values anyways, let's + * just copy/paste. + */ + + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_8, EMU_SRC_DOCK_MIC_A1); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_9, EMU_SRC_DOCK_MIC_B1); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_A, EMU_SRC_HAMOA_ADC_LEFT2); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_B, EMU_SRC_HAMOA_ADC_LEFT2); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_C, EMU_SRC_DOCK_ADC1_LEFT1); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_D, EMU_SRC_DOCK_ADC1_RIGHT1); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_E, EMU_SRC_DOCK_ADC2_LEFT1); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_F, EMU_SRC_DOCK_ADC2_RIGHT1); +#endif +#if 0 + /* Original */ + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_4, EMU_SRC_HANA_ADAT); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_5, EMU_SRC_HANA_ADAT + 1); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_6, EMU_SRC_HANA_ADAT + 2); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_7, EMU_SRC_HANA_ADAT + 3); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_8, EMU_SRC_HANA_ADAT + 4); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_9, EMU_SRC_HANA_ADAT + 5); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_A, EMU_SRC_HANA_ADAT + 6); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_B, EMU_SRC_HANA_ADAT + 7); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_C, EMU_SRC_DOCK_MIC_A1); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_D, EMU_SRC_DOCK_MIC_B1); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_E, EMU_SRC_HAMOA_ADC_LEFT2); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_F, EMU_SRC_HAMOA_ADC_LEFT2); +#endif + for (i = 0;i < 0x20; i++ ) { + /* AudioDock Elink <- Silence */ + snd_emu1010_fpga_link_dst_src_write(emu, 0x0100+i, EMU_SRC_SILENCE); + } + for (i = 0;i < 4; i++) { + /* Hana SPDIF Out <- Silence */ + snd_emu1010_fpga_link_dst_src_write(emu, 0x0200+i, EMU_SRC_SILENCE); + } + for (i = 0;i < 7; i++) { + /* Hamoa DAC <- Silence */ + snd_emu1010_fpga_link_dst_src_write(emu, 0x0300+i, EMU_SRC_SILENCE); + } + for (i = 0;i < 7; i++) { + /* Hana ADAT Out <- Silence */ + snd_emu1010_fpga_link_dst_src_write(emu, EMU_DST_HANA_ADAT + i, EMU_SRC_SILENCE); + } + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE_I2S0_LEFT, EMU_SRC_DOCK_ADC1_LEFT1); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE_I2S0_RIGHT, EMU_SRC_DOCK_ADC1_RIGHT1); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE_I2S1_LEFT, EMU_SRC_DOCK_ADC2_LEFT1); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE_I2S1_RIGHT, EMU_SRC_DOCK_ADC2_RIGHT1); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE_I2S2_LEFT, EMU_SRC_DOCK_ADC3_LEFT1); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE_I2S2_RIGHT, EMU_SRC_DOCK_ADC3_RIGHT1); + snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, 0x01 ); // Unmute all + + snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &tmp ); + + /* AC97 1.03, Any 32Meg of 2Gig address, Auto-Mute, EMU32 Slave, + * Lock Sound Memory Cache, Lock Tank Memory Cache, + * Mute all codecs. + */ + outl(0x0000a000, emu->port + HCFG); + /* AC97 1.03, Any 32Meg of 2Gig address, Auto-Mute, EMU32 Slave, + * Lock Sound Memory Cache, Lock Tank Memory Cache, + * Un-Mute all codecs. + */ + outl(0x0000a001, emu->port + HCFG); + + /* Initial boot complete. Now patches */ + + snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &tmp ); + snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_IN, 0x19 ); /* MIDI Route */ + snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_OUT, 0x0c ); /* Unknown */ + snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_IN, 0x19 ); /* MIDI Route */ + snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_OUT, 0x0c ); /* Unknown */ + snd_emu1010_fpga_read(emu, EMU_HANA_SPDIF_MODE, &tmp ); + snd_emu1010_fpga_write(emu, EMU_HANA_SPDIF_MODE, 0x10 ); /* SPDIF Format spdif (or 0x11 for aes/ebu) */ + + /* Start Micro/Audio Dock firmware loader thread */ + emu->emu1010.firmware_thread = kthread_create(&emu1010_firmware_thread, + emu, + "emu1010_firmware"); + wake_up_process(emu->emu1010.firmware_thread); + +#if 0 + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_HAMOA_DAC_LEFT1, EMU_SRC_ALICE_EMU32B + 2); /* ALICE2 bus 0xa2 */ + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_HAMOA_DAC_RIGHT1, EMU_SRC_ALICE_EMU32B + 3); /* ALICE2 bus 0xa3 */ + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_HANA_SPDIF_LEFT1, EMU_SRC_ALICE_EMU32A + 2); /* ALICE2 bus 0xb2 */ + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_HANA_SPDIF_RIGHT1, EMU_SRC_ALICE_EMU32A + 3); /* ALICE2 bus 0xb3 */ +#endif + /* Default outputs */ + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_DOCK_DAC1_LEFT1, EMU_SRC_ALICE_EMU32A + 0); /* ALICE2 bus 0xa0 */ + emu->emu1010.output_source[0] = 21; + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_DOCK_DAC1_RIGHT1, EMU_SRC_ALICE_EMU32A + 1); + emu->emu1010.output_source[1] = 22; + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_DOCK_DAC2_LEFT1, EMU_SRC_ALICE_EMU32A + 2); + emu->emu1010.output_source[2] = 23; + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_DOCK_DAC2_RIGHT1, EMU_SRC_ALICE_EMU32A + 3); + emu->emu1010.output_source[3] = 24; + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_DOCK_DAC3_LEFT1, EMU_SRC_ALICE_EMU32A + 4); + emu->emu1010.output_source[4] = 25; + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_DOCK_DAC3_RIGHT1, EMU_SRC_ALICE_EMU32A + 5); + emu->emu1010.output_source[5] = 26; + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_DOCK_DAC4_LEFT1, EMU_SRC_ALICE_EMU32A + 6); + emu->emu1010.output_source[6] = 27; + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_DOCK_DAC4_RIGHT1, EMU_SRC_ALICE_EMU32A + 7); + emu->emu1010.output_source[7] = 28; + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_DOCK_PHONES_LEFT1, EMU_SRC_ALICE_EMU32A + 0); /* ALICE2 bus 0xa0 */ + emu->emu1010.output_source[8] = 21; + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_DOCK_PHONES_RIGHT1, EMU_SRC_ALICE_EMU32A + 1); + emu->emu1010.output_source[9] = 22; + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_DOCK_SPDIF_LEFT1, EMU_SRC_ALICE_EMU32A + 0); /* ALICE2 bus 0xa0 */ + emu->emu1010.output_source[10] = 21; + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_DOCK_SPDIF_RIGHT1, EMU_SRC_ALICE_EMU32A + 1); + emu->emu1010.output_source[11] = 22; + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_HANA_SPDIF_LEFT1, EMU_SRC_ALICE_EMU32A + 0); /* ALICE2 bus 0xa0 */ + emu->emu1010.output_source[12] = 21; + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_HANA_SPDIF_RIGHT1, EMU_SRC_ALICE_EMU32A + 1); + emu->emu1010.output_source[13] = 22; + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_HAMOA_DAC_LEFT1, EMU_SRC_ALICE_EMU32A + 0); /* ALICE2 bus 0xa0 */ + emu->emu1010.output_source[14] = 21; + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_HAMOA_DAC_RIGHT1, EMU_SRC_ALICE_EMU32A + 1); + emu->emu1010.output_source[15] = 22; + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_HANA_ADAT, EMU_SRC_ALICE_EMU32A + 0); /* ALICE2 bus 0xa0 */ + emu->emu1010.output_source[16] = 21; + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_HANA_ADAT + 1, EMU_SRC_ALICE_EMU32A + 1); + emu->emu1010.output_source[17] = 22; + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_HANA_ADAT + 2, EMU_SRC_ALICE_EMU32A + 2); + emu->emu1010.output_source[18] = 23; + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_HANA_ADAT + 3, EMU_SRC_ALICE_EMU32A + 3); + emu->emu1010.output_source[19] = 24; + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_HANA_ADAT + 4, EMU_SRC_ALICE_EMU32A + 4); + emu->emu1010.output_source[20] = 25; + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_HANA_ADAT + 5, EMU_SRC_ALICE_EMU32A + 5); + emu->emu1010.output_source[21] = 26; + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_HANA_ADAT + 6, EMU_SRC_ALICE_EMU32A + 6); + emu->emu1010.output_source[22] = 27; + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_HANA_ADAT + 7, EMU_SRC_ALICE_EMU32A + 7); + emu->emu1010.output_source[23] = 28; + + /* TEMP: Select SPDIF in/out */ + //snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, 0x0); /* Output spdif */ + + /* TEMP: Select 48kHz SPDIF out */ + snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, 0x0); /* Mute all */ + snd_emu1010_fpga_write(emu, EMU_HANA_DEFCLOCK, 0x0); /* Default fallback clock 48kHz */ + /* Word Clock source, Internal 48kHz x1 */ + snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, EMU_HANA_WCLOCK_INT_48K ); + //snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, EMU_HANA_WCLOCK_INT_48K | EMU_HANA_WCLOCK_4X ); + emu->emu1010.internal_clock = 1; /* 48000 */ + snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, 0x12);/* Set LEDs on Audio Dock */ + snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, 0x1); /* Unmute all */ + //snd_emu1010_fpga_write(emu, 0x7, 0x0); /* Mute all */ + //snd_emu1010_fpga_write(emu, 0x7, 0x1); /* Unmute all */ + //snd_emu1010_fpga_write(emu, 0xe, 0x12); /* Set LEDs on Audio Dock */ + + return 0; +} +/* + * Create the EMU10K1 instance + */ + +#ifdef CONFIG_PM +static int alloc_pm_buffer(struct snd_emu10k1 *emu); +static void free_pm_buffer(struct snd_emu10k1 *emu); +#endif + +static int snd_emu10k1_free(struct snd_emu10k1 *emu) +{ + if (emu->port) { /* avoid access to already used hardware */ + snd_emu10k1_fx8010_tram_setup(emu, 0); + snd_emu10k1_done(emu); + /* remove reserved page */ + if (emu->reserved_page) { + snd_emu10k1_synth_free(emu, (struct snd_util_memblk *)emu->reserved_page); + emu->reserved_page = NULL; + } + snd_emu10k1_free_efx(emu); + } + if (emu->card_capabilities->emu1010) { + /* Disable 48Volt power to Audio Dock */ + snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, 0 ); + kthread_stop(emu->emu1010.firmware_thread); + } + if (emu->memhdr) + snd_util_memhdr_free(emu->memhdr); + if (emu->silent_page.area) + snd_dma_free_pages(&emu->silent_page); + if (emu->ptb_pages.area) + snd_dma_free_pages(&emu->ptb_pages); + vfree(emu->page_ptr_table); + vfree(emu->page_addr_table); +#ifdef CONFIG_PM + free_pm_buffer(emu); +#endif + if (emu->irq >= 0) + free_irq(emu->irq, emu); + if (emu->port) + pci_release_regions(emu->pci); + if (emu->card_capabilities->ca0151_chip) /* P16V */ + snd_p16v_free(emu); + pci_disable_device(emu->pci); + kfree(emu); + return 0; +} + +static int snd_emu10k1_dev_free(struct snd_device *device) +{ + struct snd_emu10k1 *emu = device->device_data; + return snd_emu10k1_free(emu); +} + +static struct snd_emu_chip_details emu_chip_details[] = { + /* Audigy 2 Value AC3 out does not work yet. Need to find out how to turn off interpolators.*/ + /* Tested by James@superbug.co.uk 3rd July 2005 */ + /* DSP: CA0108-IAT + * DAC: CS4382-KQ + * ADC: Philips 1361T + * AC97: STAC9750 + * CA0151: None + */ + {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x10011102, + .driver = "Audigy2", .name = "Audigy 2 Value [SB0400]", + .id = "Audigy2", + .emu10k2_chip = 1, + .ca0108_chip = 1, + .spk71 = 1, + .ac97_chip = 1} , + /* Audigy4 (Not PRO) SB0610 */ + /* Tested by James@superbug.co.uk 4th April 2006 */ + /* A_IOCFG bits + * Output + * 0: ? + * 1: ? + * 2: ? + * 3: 0 - Digital Out, 1 - Line in + * 4: ? + * 5: ? + * 6: ? + * 7: ? + * Input + * 8: ? + * 9: ? + * A: Green jack sense (Front) + * B: ? + * C: Black jack sense (Rear/Side Right) + * D: Yellow jack sense (Center/LFE/Side Left) + * E: ? + * F: ? + * + * Digital Out/Line in switch using A_IOCFG bit 3 (0x08) + * 0 - Digital Out + * 1 - Line in + */ + /* Mic input not tested. + * Analog CD input not tested + * Digital Out not tested. + * Line in working. + * Audio output 5.1 working. Side outputs not working. + */ + /* DSP: CA10300-IAT LF + * DAC: Cirrus Logic CS4382-KQZ + * ADC: Philips 1361T + * AC97: Sigmatel STAC9750 + * CA0151: None + */ + {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x10211102, + .driver = "Audigy2", .name = "Audigy 4 [SB0610]", + .id = "Audigy2", + .emu10k2_chip = 1, + .ca0108_chip = 1, + .spk71 = 1, + .adc_1361t = 1, /* 24 bit capture instead of 16bit */ + .ac97_chip = 1} , + /* Audigy 2 ZS Notebook Cardbus card.*/ + /* Tested by James@superbug.co.uk 6th November 2006 */ + /* Audio output 7.1/Headphones working. + * Digital output working. (AC3 not checked, only PCM) + * Audio Mic/Line inputs working. + * Digital input not tested. + */ + /* DSP: Tina2 + * DAC: Wolfson WM8768/WM8568 + * ADC: Wolfson WM8775 + * AC97: None + * CA0151: None + */ + /* Tested by James@superbug.co.uk 4th April 2006 */ + /* A_IOCFG bits + * Output + * 0: Not Used + * 1: 0 = Mute all the 7.1 channel out. 1 = unmute. + * 2: Analog input 0 = line in, 1 = mic in + * 3: Not Used + * 4: Digital output 0 = off, 1 = on. + * 5: Not Used + * 6: Not Used + * 7: Not Used + * Input + * All bits 1 (0x3fxx) means nothing plugged in. + * 8-9: 0 = Line in/Mic, 2 = Optical in, 3 = Nothing. + * A-B: 0 = Headphones, 2 = Optical out, 3 = Nothing. + * C-D: 2 = Front/Rear/etc, 3 = nothing. + * E-F: Always 0 + * + */ + {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x20011102, + .driver = "Audigy2", .name = "Audigy 2 ZS Notebook [SB0530]", + .id = "Audigy2", + .emu10k2_chip = 1, + .ca0108_chip = 1, + .ca_cardbus_chip = 1, + .spi_dac = 1, + .i2c_adc = 1, + .spk71 = 1} , + {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x42011102, + .driver = "Audigy2", .name = "E-mu 1010 Notebook [MAEM8950]", + .id = "EMU1010", + .emu10k2_chip = 1, + .ca0108_chip = 1, + .ca_cardbus_chip = 1, + .spk71 = 1 , + .emu1010 = 3} , + {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x40041102, + .driver = "Audigy2", .name = "E-mu 1010b PCI [MAEM????]", + .id = "EMU1010", + .emu10k2_chip = 1, + .ca0108_chip = 1, + .spk71 = 1 , + .emu1010 = 2} , + {.vendor = 0x1102, .device = 0x0008, + .driver = "Audigy2", .name = "Audigy 2 Value [Unknown]", + .id = "Audigy2", + .emu10k2_chip = 1, + .ca0108_chip = 1, + .ac97_chip = 1} , + /* Tested by James@superbug.co.uk 8th July 2005. No sound available yet. */ + {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x40011102, + .driver = "Audigy2", .name = "E-mu 1010 [4001]", + .id = "EMU1010", + .emu10k2_chip = 1, + .ca0102_chip = 1, + .spk71 = 1, + .emu1010 = 1} , + /* Tested by James@superbug.co.uk 3rd July 2005 */ + {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x20071102, + .driver = "Audigy2", .name = "Audigy 4 PRO [SB0380]", + .id = "Audigy2", + .emu10k2_chip = 1, + .ca0102_chip = 1, + .ca0151_chip = 1, + .spk71 = 1, + .spdif_bug = 1, + .ac97_chip = 1} , + /* Tested by shane-alsa@cm.nu 5th Nov 2005 */ + /* The 0x20061102 does have SB0350 written on it + * Just like 0x20021102 + */ + {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x20061102, + .driver = "Audigy2", .name = "Audigy 2 [SB0350b]", + .id = "Audigy2", + .emu10k2_chip = 1, + .ca0102_chip = 1, + .ca0151_chip = 1, + .spk71 = 1, + .spdif_bug = 1, + .ac97_chip = 1} , + {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x20021102, + .driver = "Audigy2", .name = "Audigy 2 ZS [SB0350]", + .id = "Audigy2", + .emu10k2_chip = 1, + .ca0102_chip = 1, + .ca0151_chip = 1, + .spk71 = 1, + .spdif_bug = 1, + .ac97_chip = 1} , + {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x20011102, + .driver = "Audigy2", .name = "Audigy 2 ZS [2001]", + .id = "Audigy2", + .emu10k2_chip = 1, + .ca0102_chip = 1, + .ca0151_chip = 1, + .spk71 = 1, + .spdif_bug = 1, + .ac97_chip = 1} , + /* Audigy 2 */ + /* Tested by James@superbug.co.uk 3rd July 2005 */ + /* DSP: CA0102-IAT + * DAC: CS4382-KQ + * ADC: Philips 1361T + * AC97: STAC9721 + * CA0151: Yes + */ + {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x10071102, + .driver = "Audigy2", .name = "Audigy 2 [SB0240]", + .id = "Audigy2", + .emu10k2_chip = 1, + .ca0102_chip = 1, + .ca0151_chip = 1, + .spk71 = 1, + .spdif_bug = 1, + .adc_1361t = 1, /* 24 bit capture instead of 16bit */ + .ac97_chip = 1} , + {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x10051102, + .driver = "Audigy2", .name = "Audigy 2 EX [1005]", + .id = "Audigy2", + .emu10k2_chip = 1, + .ca0102_chip = 1, + .ca0151_chip = 1, + .spk71 = 1, + .spdif_bug = 1} , + /* Dell OEM/Creative Labs Audigy 2 ZS */ + /* See ALSA bug#1365 */ + {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x10031102, + .driver = "Audigy2", .name = "Audigy 2 ZS [SB0353]", + .id = "Audigy2", + .emu10k2_chip = 1, + .ca0102_chip = 1, + .ca0151_chip = 1, + .spk71 = 1, + .spdif_bug = 1, + .ac97_chip = 1} , + {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x10021102, + .driver = "Audigy2", .name = "Audigy 2 Platinum [SB0240P]", + .id = "Audigy2", + .emu10k2_chip = 1, + .ca0102_chip = 1, + .ca0151_chip = 1, + .spk71 = 1, + .spdif_bug = 1, + .adc_1361t = 1, /* 24 bit capture instead of 16bit. Fixes ALSA bug#324 */ + .ac97_chip = 1} , + {.vendor = 0x1102, .device = 0x0004, .revision = 0x04, + .driver = "Audigy2", .name = "Audigy 2 [Unknown]", + .id = "Audigy2", + .emu10k2_chip = 1, + .ca0102_chip = 1, + .ca0151_chip = 1, + .spdif_bug = 1, + .ac97_chip = 1} , + {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x00531102, + .driver = "Audigy", .name = "Audigy 1 [SB0090]", + .id = "Audigy", + .emu10k2_chip = 1, + .ca0102_chip = 1, + .ac97_chip = 1} , + {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x00521102, + .driver = "Audigy", .name = "Audigy 1 ES [SB0160]", + .id = "Audigy", + .emu10k2_chip = 1, + .ca0102_chip = 1, + .spdif_bug = 1, + .ac97_chip = 1} , + {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x00511102, + .driver = "Audigy", .name = "Audigy 1 [SB0090]", + .id = "Audigy", + .emu10k2_chip = 1, + .ca0102_chip = 1, + .ac97_chip = 1} , + {.vendor = 0x1102, .device = 0x0004, + .driver = "Audigy", .name = "Audigy 1 [Unknown]", + .id = "Audigy", + .emu10k2_chip = 1, + .ca0102_chip = 1, + .ac97_chip = 1} , + {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x806B1102, + .driver = "EMU10K1", .name = "SBLive! [SB0105]", + .id = "Live", + .emu10k1_chip = 1, + .ac97_chip = 1, + .sblive51 = 1} , + {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x806A1102, + .driver = "EMU10K1", .name = "SBLive! Value [SB0103]", + .id = "Live", + .emu10k1_chip = 1, + .ac97_chip = 1, + .sblive51 = 1} , + {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80691102, + .driver = "EMU10K1", .name = "SBLive! Value [SB0101]", + .id = "Live", + .emu10k1_chip = 1, + .ac97_chip = 1, + .sblive51 = 1} , + /* Tested by ALSA bug#1680 26th December 2005 */ + /* note: It really has SB0220 written on the card. */ + {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80661102, + .driver = "EMU10K1", .name = "SB Live 5.1 Dell OEM [SB0220]", + .id = "Live", + .emu10k1_chip = 1, + .ac97_chip = 1, + .sblive51 = 1} , + /* Tested by Thomas Zehetbauer 27th Aug 2005 */ + {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80651102, + .driver = "EMU10K1", .name = "SB Live 5.1 [SB0220]", + .id = "Live", + .emu10k1_chip = 1, + .ac97_chip = 1, + .sblive51 = 1} , + {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x100a1102, + .driver = "EMU10K1", .name = "SB Live 5.1 [SB0220]", + .id = "Live", + .emu10k1_chip = 1, + .ac97_chip = 1, + .sblive51 = 1} , + {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80641102, + .driver = "EMU10K1", .name = "SB Live 5.1", + .id = "Live", + .emu10k1_chip = 1, + .ac97_chip = 1, + .sblive51 = 1} , + /* Tested by alsa bugtrack user "hus" bug #1297 12th Aug 2005 */ + {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80611102, + .driver = "EMU10K1", .name = "SBLive 5.1 [SB0060]", + .id = "Live", + .emu10k1_chip = 1, + .ac97_chip = 2, /* ac97 is optional; both SBLive 5.1 and platinum + * share the same IDs! + */ + .sblive51 = 1} , + {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80511102, + .driver = "EMU10K1", .name = "SBLive! Value [CT4850]", + .id = "Live", + .emu10k1_chip = 1, + .ac97_chip = 1, + .sblive51 = 1} , + {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80401102, + .driver = "EMU10K1", .name = "SBLive! Platinum [CT4760P]", + .id = "Live", + .emu10k1_chip = 1, + .ac97_chip = 1} , + {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80321102, + .driver = "EMU10K1", .name = "SBLive! Value [CT4871]", + .id = "Live", + .emu10k1_chip = 1, + .ac97_chip = 1, + .sblive51 = 1} , + {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80311102, + .driver = "EMU10K1", .name = "SBLive! Value [CT4831]", + .id = "Live", + .emu10k1_chip = 1, + .ac97_chip = 1, + .sblive51 = 1} , + {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80281102, + .driver = "EMU10K1", .name = "SBLive! Value [CT4870]", + .id = "Live", + .emu10k1_chip = 1, + .ac97_chip = 1, + .sblive51 = 1} , + /* Tested by James@superbug.co.uk 3rd July 2005 */ + {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80271102, + .driver = "EMU10K1", .name = "SBLive! Value [CT4832]", + .id = "Live", + .emu10k1_chip = 1, + .ac97_chip = 1, + .sblive51 = 1} , + {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80261102, + .driver = "EMU10K1", .name = "SBLive! Value [CT4830]", + .id = "Live", + .emu10k1_chip = 1, + .ac97_chip = 1, + .sblive51 = 1} , + {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80231102, + .driver = "EMU10K1", .name = "SB PCI512 [CT4790]", + .id = "Live", + .emu10k1_chip = 1, + .ac97_chip = 1, + .sblive51 = 1} , + {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80221102, + .driver = "EMU10K1", .name = "SBLive! Value [CT4780]", + .id = "Live", + .emu10k1_chip = 1, + .ac97_chip = 1, + .sblive51 = 1} , + {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x40011102, + .driver = "EMU10K1", .name = "E-mu APS [4001]", + .id = "APS", + .emu10k1_chip = 1, + .ecard = 1} , + {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x00211102, + .driver = "EMU10K1", .name = "SBLive! [CT4620]", + .id = "Live", + .emu10k1_chip = 1, + .ac97_chip = 1, + .sblive51 = 1} , + {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x00201102, + .driver = "EMU10K1", .name = "SBLive! Value [CT4670]", + .id = "Live", + .emu10k1_chip = 1, + .ac97_chip = 1, + .sblive51 = 1} , + {.vendor = 0x1102, .device = 0x0002, + .driver = "EMU10K1", .name = "SB Live [Unknown]", + .id = "Live", + .emu10k1_chip = 1, + .ac97_chip = 1, + .sblive51 = 1} , + { } /* terminator */ +}; + +int __devinit snd_emu10k1_create(struct snd_card *card, + struct pci_dev * pci, + unsigned short extin_mask, + unsigned short extout_mask, + long max_cache_bytes, + int enable_ir, + uint subsystem, + struct snd_emu10k1 ** remu) +{ + struct snd_emu10k1 *emu; + int idx, err; + int is_audigy; + unsigned int silent_page; + const struct snd_emu_chip_details *c; + static struct snd_device_ops ops = { + .dev_free = snd_emu10k1_dev_free, + }; + + *remu = NULL; + + /* enable PCI device */ + if ((err = pci_enable_device(pci)) < 0) + return err; + + emu = kzalloc(sizeof(*emu), GFP_KERNEL); + if (emu == NULL) { + pci_disable_device(pci); + return -ENOMEM; + } + emu->card = card; + spin_lock_init(&emu->reg_lock); + spin_lock_init(&emu->emu_lock); + spin_lock_init(&emu->voice_lock); + spin_lock_init(&emu->synth_lock); + spin_lock_init(&emu->memblk_lock); + mutex_init(&emu->fx8010.lock); + INIT_LIST_HEAD(&emu->mapped_link_head); + INIT_LIST_HEAD(&emu->mapped_order_link_head); + emu->pci = pci; + emu->irq = -1; + emu->synth = NULL; + emu->get_synth_voice = NULL; + /* read revision & serial */ + emu->revision = pci->revision; + pci_read_config_dword(pci, PCI_SUBSYSTEM_VENDOR_ID, &emu->serial); + pci_read_config_word(pci, PCI_SUBSYSTEM_ID, &emu->model); + snd_printdd("vendor=0x%x, device=0x%x, subsystem_vendor_id=0x%x, subsystem_id=0x%x\n",pci->vendor, pci->device, emu->serial, emu->model); + + for (c = emu_chip_details; c->vendor; c++) { + if (c->vendor == pci->vendor && c->device == pci->device) { + if (subsystem) { + if (c->subsystem && (c->subsystem == subsystem) ) { + break; + } else continue; + } else { + if (c->subsystem && (c->subsystem != emu->serial) ) + continue; + if (c->revision && c->revision != emu->revision) + continue; + } + break; + } + } + if (c->vendor == 0) { + snd_printk(KERN_ERR "emu10k1: Card not recognised\n"); + kfree(emu); + pci_disable_device(pci); + return -ENOENT; + } + emu->card_capabilities = c; + if (c->subsystem && !subsystem) + snd_printdd("Sound card name=%s\n", c->name); + else if (subsystem) + snd_printdd("Sound card name=%s, vendor=0x%x, device=0x%x, subsystem=0x%x. Forced to subsytem=0x%x\n", + c->name, pci->vendor, pci->device, emu->serial, c->subsystem); + else + snd_printdd("Sound card name=%s, vendor=0x%x, device=0x%x, subsystem=0x%x.\n", + c->name, pci->vendor, pci->device, emu->serial); + + if (!*card->id && c->id) { + int i, n = 0; + strlcpy(card->id, c->id, sizeof(card->id)); + for (;;) { + for (i = 0; i < snd_ecards_limit; i++) { + if (snd_cards[i] && !strcmp(snd_cards[i]->id, card->id)) + break; + } + if (i >= snd_ecards_limit) + break; + n++; + if (n >= SNDRV_CARDS) + break; + snprintf(card->id, sizeof(card->id), "%s_%d", c->id, n); + } + } + + is_audigy = emu->audigy = c->emu10k2_chip; + + /* set the DMA transfer mask */ + emu->dma_mask = is_audigy ? AUDIGY_DMA_MASK : EMU10K1_DMA_MASK; + if (pci_set_dma_mask(pci, emu->dma_mask) < 0 || + pci_set_consistent_dma_mask(pci, emu->dma_mask) < 0) { + snd_printk(KERN_ERR "architecture does not support PCI busmaster DMA with mask 0x%lx\n", emu->dma_mask); + kfree(emu); + pci_disable_device(pci); + return -ENXIO; + } + if (is_audigy) + emu->gpr_base = A_FXGPREGBASE; + else + emu->gpr_base = FXGPREGBASE; + + if ((err = pci_request_regions(pci, "EMU10K1")) < 0) { + kfree(emu); + pci_disable_device(pci); + return err; + } + emu->port = pci_resource_start(pci, 0); + + if (request_irq(pci->irq, snd_emu10k1_interrupt, IRQF_SHARED, + "EMU10K1", emu)) { + err = -EBUSY; + goto error; + } + emu->irq = pci->irq; + + emu->max_cache_pages = max_cache_bytes >> PAGE_SHIFT; + if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), + 32 * 1024, &emu->ptb_pages) < 0) { + err = -ENOMEM; + goto error; + } + + emu->page_ptr_table = vmalloc(emu->max_cache_pages * sizeof(void *)); + emu->page_addr_table = vmalloc(emu->max_cache_pages * + sizeof(unsigned long)); + if (emu->page_ptr_table == NULL || emu->page_addr_table == NULL) { + err = -ENOMEM; + goto error; + } + + if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), + EMUPAGESIZE, &emu->silent_page) < 0) { + err = -ENOMEM; + goto error; + } + emu->memhdr = snd_util_memhdr_new(emu->max_cache_pages * PAGE_SIZE); + if (emu->memhdr == NULL) { + err = -ENOMEM; + goto error; + } + emu->memhdr->block_extra_size = sizeof(struct snd_emu10k1_memblk) - + sizeof(struct snd_util_memblk); + + pci_set_master(pci); + + emu->fx8010.fxbus_mask = 0x303f; + if (extin_mask == 0) + extin_mask = 0x3fcf; + if (extout_mask == 0) + extout_mask = 0x7fff; + emu->fx8010.extin_mask = extin_mask; + emu->fx8010.extout_mask = extout_mask; + emu->enable_ir = enable_ir; + + if (emu->card_capabilities->ca_cardbus_chip) { + if ((err = snd_emu10k1_cardbus_init(emu)) < 0) + goto error; + } + if (emu->card_capabilities->ecard) { + if ((err = snd_emu10k1_ecard_init(emu)) < 0) + goto error; + } else if (emu->card_capabilities->emu1010) { + if ((err = snd_emu10k1_emu1010_init(emu)) < 0) { + snd_emu10k1_free(emu); + return err; + } + } else { + /* 5.1: Enable the additional AC97 Slots. If the emu10k1 version + does not support this, it shouldn't do any harm */ + snd_emu10k1_ptr_write(emu, AC97SLOT, 0, AC97SLOT_CNTR|AC97SLOT_LFE); + } + + /* initialize TRAM setup */ + emu->fx8010.itram_size = (16 * 1024)/2; + emu->fx8010.etram_pages.area = NULL; + emu->fx8010.etram_pages.bytes = 0; + + /* + * Init to 0x02109204 : + * Clock accuracy = 0 (1000ppm) + * Sample Rate = 2 (48kHz) + * Audio Channel = 1 (Left of 2) + * Source Number = 0 (Unspecified) + * Generation Status = 1 (Original for Cat Code 12) + * Cat Code = 12 (Digital Signal Mixer) + * Mode = 0 (Mode 0) + * Emphasis = 0 (None) + * CP = 1 (Copyright unasserted) + * AN = 0 (Audio data) + * P = 0 (Consumer) + */ + emu->spdif_bits[0] = emu->spdif_bits[1] = + emu->spdif_bits[2] = SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 | + SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC | + SPCS_GENERATIONSTATUS | 0x00001200 | + 0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT; + + emu->reserved_page = (struct snd_emu10k1_memblk *) + snd_emu10k1_synth_alloc(emu, 4096); + if (emu->reserved_page) + emu->reserved_page->map_locked = 1; + + /* Clear silent pages and set up pointers */ + memset(emu->silent_page.area, 0, PAGE_SIZE); + silent_page = emu->silent_page.addr << 1; + for (idx = 0; idx < MAXPAGES; idx++) + ((u32 *)emu->ptb_pages.area)[idx] = cpu_to_le32(silent_page | idx); + + /* set up voice indices */ + for (idx = 0; idx < NUM_G; idx++) { + emu->voices[idx].emu = emu; + emu->voices[idx].number = idx; + } + + if ((err = snd_emu10k1_init(emu, enable_ir, 0)) < 0) + goto error; +#ifdef CONFIG_PM + if ((err = alloc_pm_buffer(emu)) < 0) + goto error; +#endif + + /* Initialize the effect engine */ + if ((err = snd_emu10k1_init_efx(emu)) < 0) + goto error; + snd_emu10k1_audio_enable(emu); + + if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, emu, &ops)) < 0) + goto error; + +#ifdef CONFIG_PROC_FS + snd_emu10k1_proc_init(emu); +#endif + + snd_card_set_dev(card, &pci->dev); + *remu = emu; + return 0; + + error: + snd_emu10k1_free(emu); + return err; +} + +#ifdef CONFIG_PM +static unsigned char saved_regs[] = { + CPF, PTRX, CVCF, VTFT, Z1, Z2, PSST, DSL, CCCA, CCR, CLP, + FXRT, MAPA, MAPB, ENVVOL, ATKHLDV, DCYSUSV, LFOVAL1, ENVVAL, + ATKHLDM, DCYSUSM, LFOVAL2, IP, IFATN, PEFE, FMMOD, TREMFRQ, FM2FRQ2, + TEMPENV, ADCCR, FXWC, MICBA, ADCBA, FXBA, + MICBS, ADCBS, FXBS, CDCS, GPSCS, SPCS0, SPCS1, SPCS2, + SPBYPASS, AC97SLOT, CDSRCS, GPSRCS, ZVSRCS, MICIDX, ADCIDX, FXIDX, + 0xff /* end */ +}; +static unsigned char saved_regs_audigy[] = { + A_ADCIDX, A_MICIDX, A_FXWC1, A_FXWC2, A_SAMPLE_RATE, + A_FXRT2, A_SENDAMOUNTS, A_FXRT1, + 0xff /* end */ +}; + +static int __devinit alloc_pm_buffer(struct snd_emu10k1 *emu) +{ + int size; + + size = ARRAY_SIZE(saved_regs); + if (emu->audigy) + size += ARRAY_SIZE(saved_regs_audigy); + emu->saved_ptr = vmalloc(4 * NUM_G * size); + if (! emu->saved_ptr) + return -ENOMEM; + if (snd_emu10k1_efx_alloc_pm_buffer(emu) < 0) + return -ENOMEM; + if (emu->card_capabilities->ca0151_chip && + snd_p16v_alloc_pm_buffer(emu) < 0) + return -ENOMEM; + return 0; +} + +static void free_pm_buffer(struct snd_emu10k1 *emu) +{ + vfree(emu->saved_ptr); + snd_emu10k1_efx_free_pm_buffer(emu); + if (emu->card_capabilities->ca0151_chip) + snd_p16v_free_pm_buffer(emu); +} + +void snd_emu10k1_suspend_regs(struct snd_emu10k1 *emu) +{ + int i; + unsigned char *reg; + unsigned int *val; + + val = emu->saved_ptr; + for (reg = saved_regs; *reg != 0xff; reg++) + for (i = 0; i < NUM_G; i++, val++) + *val = snd_emu10k1_ptr_read(emu, *reg, i); + if (emu->audigy) { + for (reg = saved_regs_audigy; *reg != 0xff; reg++) + for (i = 0; i < NUM_G; i++, val++) + *val = snd_emu10k1_ptr_read(emu, *reg, i); + } + if (emu->audigy) + emu->saved_a_iocfg = inl(emu->port + A_IOCFG); + emu->saved_hcfg = inl(emu->port + HCFG); +} + +void snd_emu10k1_resume_init(struct snd_emu10k1 *emu) +{ + if (emu->card_capabilities->ca_cardbus_chip) + snd_emu10k1_cardbus_init(emu); + if (emu->card_capabilities->ecard) + snd_emu10k1_ecard_init(emu); + else if (emu->card_capabilities->emu1010) + snd_emu10k1_emu1010_init(emu); + else + snd_emu10k1_ptr_write(emu, AC97SLOT, 0, AC97SLOT_CNTR|AC97SLOT_LFE); + snd_emu10k1_init(emu, emu->enable_ir, 1); +} + +void snd_emu10k1_resume_regs(struct snd_emu10k1 *emu) +{ + int i; + unsigned char *reg; + unsigned int *val; + + snd_emu10k1_audio_enable(emu); + + /* resore for spdif */ + if (emu->audigy) + outl(emu->saved_a_iocfg, emu->port + A_IOCFG); + outl(emu->saved_hcfg, emu->port + HCFG); + + val = emu->saved_ptr; + for (reg = saved_regs; *reg != 0xff; reg++) + for (i = 0; i < NUM_G; i++, val++) + snd_emu10k1_ptr_write(emu, *reg, i, *val); + if (emu->audigy) { + for (reg = saved_regs_audigy; *reg != 0xff; reg++) + for (i = 0; i < NUM_G; i++, val++) + snd_emu10k1_ptr_write(emu, *reg, i, *val); + } +} +#endif diff -ruN linux-2.6.22/sound/pci/emu10k1/emu10k1x.c linux-2.6.22-alsa/sound/pci/emu10k1/emu10k1x.c --- linux-2.6.22/sound/pci/emu10k1/emu10k1x.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/pci/emu10k1/emu10k1x.c 2007-09-02 13:30:43.000000000 +0200 @@ -1062,14 +1062,7 @@ return 0; } -static int snd_emu10k1x_shared_spdif_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_emu10k1x_shared_spdif_info snd_ctl_boolean_mono_info static int snd_emu10k1x_shared_spdif_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) diff -ruN linux-2.6.22/sound/pci/emu10k1/emufx.c linux-2.6.22-alsa/sound/pci/emu10k1/emufx.c --- linux-2.6.22/sound/pci/emu10k1/emufx.c 2007-07-28 13:40:44.000000000 +0200 +++ linux-2.6.22-alsa/sound/pci/emu10k1/emufx.c 2007-09-01 20:55:23.000000000 +0200 @@ -1123,6 +1123,11 @@ ctl->translation = EMU10K1_GPR_TRANSLATION_ONOFF; } +/* + * Used for emu1010 - conversion from 32-bit capture inputs from HANA + * to 2 x 16-bit registers in audigy - their values are read via DMA. + * Conversion is performed by Audigy DSP instructions of FX8010. + */ static int snd_emu10k1_audigy_dsp_convert_32_to_2x16( struct snd_emu10k1_fx8010_code *icode, u32 *ptr, int tmp, int bit_shifter16, @@ -1193,12 +1198,16 @@ snd_emu10k1_ptr_write(emu, A_DBG, 0, (emu->fx8010.dbg = 0) | A_DBG_SINGLE_STEP); #if 1 - /* PCM front Playback Volume (independent from stereo mix) */ + /* PCM front Playback Volume (independent from stereo mix) + * playback = 0 + ( gpr * FXBUS_PCM_LEFT_FRONT >> 31) + * where gpr contains attenuation from corresponding mixer control + * (snd_emu10k1_init_stereo_control) + */ A_OP(icode, &ptr, iMAC0, A_GPR(playback), A_C_00000000, A_GPR(gpr), A_FXBUS(FXBUS_PCM_LEFT_FRONT)); A_OP(icode, &ptr, iMAC0, A_GPR(playback+1), A_C_00000000, A_GPR(gpr+1), A_FXBUS(FXBUS_PCM_RIGHT_FRONT)); snd_emu10k1_init_stereo_control(&controls[nctl++], "PCM Front Playback Volume", gpr, 100); gpr += 2; - + /* PCM Surround Playback (independent from stereo mix) */ A_OP(icode, &ptr, iMAC0, A_GPR(playback+2), A_C_00000000, A_GPR(gpr), A_FXBUS(FXBUS_PCM_LEFT_REAR)); A_OP(icode, &ptr, iMAC0, A_GPR(playback+3), A_C_00000000, A_GPR(gpr+1), A_FXBUS(FXBUS_PCM_RIGHT_REAR)); @@ -1258,8 +1267,16 @@ /* emu1212 DSP 0 and DSP 1 Capture */ if (emu->card_capabilities->emu1010) { - A_OP(icode, &ptr, iMAC0, A_GPR(capture+0), A_GPR(capture+0), A_GPR(gpr), A_P16VIN(0x0)); - A_OP(icode, &ptr, iMAC0, A_GPR(capture+1), A_GPR(capture+1), A_GPR(gpr+1), A_P16VIN(0x1)); + if (emu->card_capabilities->ca0108_chip) { + /* Note:JCD:No longer bit shift lower 16bits to upper 16bits of 32bit value. */ + A_OP(icode, &ptr, iMACINT0, A_GPR(tmp), A_C_00000000, A3_EMU32IN(0x0), A_C_00000001); + A_OP(icode, &ptr, iMAC0, A_GPR(capture+0), A_GPR(capture+0), A_GPR(gpr), A_GPR(tmp)); + A_OP(icode, &ptr, iMACINT0, A_GPR(tmp), A_C_00000000, A3_EMU32IN(0x1), A_C_00000001); + A_OP(icode, &ptr, iMAC0, A_GPR(capture+1), A_GPR(capture+1), A_GPR(gpr), A_GPR(tmp)); + } else { + A_OP(icode, &ptr, iMAC0, A_GPR(capture+0), A_GPR(capture+0), A_GPR(gpr), A_P16VIN(0x0)); + A_OP(icode, &ptr, iMAC0, A_GPR(capture+1), A_GPR(capture+1), A_GPR(gpr+1), A_P16VIN(0x1)); + } snd_emu10k1_init_stereo_control(&controls[nctl++], "EMU Capture Volume", gpr, 0); gpr += 2; } @@ -1507,7 +1524,11 @@ /* EMU1010 Outputs from PCM Front, Rear, Center, LFE, Side */ snd_printk("EMU outputs on\n"); for (z = 0; z < 8; z++) { - A_OP(icode, &ptr, iACC3, A_EMU32OUTL(z), A_GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + z), A_C_00000000, A_C_00000000); + if (emu->card_capabilities->ca0108_chip) { + A_OP(icode, &ptr, iACC3, A3_EMU32OUT(z), A_GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + z), A_C_00000000, A_C_00000000); + } else { + A_OP(icode, &ptr, iACC3, A_EMU32OUTL(z), A_GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + z), A_C_00000000, A_C_00000000); + } } } @@ -1548,41 +1569,116 @@ #endif if (emu->card_capabilities->emu1010) { - snd_printk("EMU inputs on\n"); - /* Capture 8 channels of S32_LE sound */ - - /* printk("emufx.c: gpr=0x%x, tmp=0x%x\n",gpr, tmp); */ - /* For the EMU1010: How to get 32bit values from the DSP. High 16bits into L, low 16bits into R. */ - /* A_P16VIN(0) is delayed by one sample, - * so all other A_P16VIN channels will need to also be delayed - */ - /* Left ADC in. 1 of 2 */ - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_P16VIN(0x0), A_FXBUS2(0) ); - /* Right ADC in 1 of 2 */ - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(2) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x1), A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(4) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x2), A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(6) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x3), A_C_00000000, A_C_00000000); - /* For 96kHz mode */ - /* Left ADC in. 2 of 2 */ - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0x8) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x4), A_C_00000000, A_C_00000000); - /* Right ADC in 2 of 2 */ - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xa) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x5), A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xc) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x6), A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xe) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x7), A_C_00000000, A_C_00000000); + if (emu->card_capabilities->ca0108_chip) { + snd_printk("EMU2 inputs on\n"); + for (z = 0; z < 0x10; z++) { + snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, + bit_shifter16, + A3_EMU32IN(z), + A_FXBUS2(z*2) ); + } + } else { + snd_printk("EMU inputs on\n"); + /* Capture 16 (originally 8) channels of S32_LE sound */ + + /* printk("emufx.c: gpr=0x%x, tmp=0x%x\n",gpr, tmp); */ + /* For the EMU1010: How to get 32bit values from the DSP. High 16bits into L, low 16bits into R. */ + /* A_P16VIN(0) is delayed by one sample, + * so all other A_P16VIN channels will need to also be delayed + */ + /* Left ADC in. 1 of 2 */ + snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_P16VIN(0x0), A_FXBUS2(0) ); + /* Right ADC in 1 of 2 */ + gpr_map[gpr++] = 0x00000000; + /* Delaying by one sample: instead of copying the input + * value A_P16VIN to output A_FXBUS2 as in the first channel, + * we use an auxiliary register, delaying the value by one + * sample + */ + snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(2) ); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x1), A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(4) ); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x2), A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(6) ); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x3), A_C_00000000, A_C_00000000); + /* For 96kHz mode */ + /* Left ADC in. 2 of 2 */ + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0x8) ); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x4), A_C_00000000, A_C_00000000); + /* Right ADC in 2 of 2 */ + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xa) ); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x5), A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xc) ); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x6), A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xe) ); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x7), A_C_00000000, A_C_00000000); + /* Pavel Hofman - we still have voices, A_FXBUS2s, and + * A_P16VINs available - + * let's add 8 more capture channels - total of 16 + */ + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, + bit_shifter16, + A_GPR(gpr - 1), + A_FXBUS2(0x10)); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x8), + A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, + bit_shifter16, + A_GPR(gpr - 1), + A_FXBUS2(0x12)); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x9), + A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, + bit_shifter16, + A_GPR(gpr - 1), + A_FXBUS2(0x14)); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xa), + A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, + bit_shifter16, + A_GPR(gpr - 1), + A_FXBUS2(0x16)); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xb), + A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, + bit_shifter16, + A_GPR(gpr - 1), + A_FXBUS2(0x18)); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xc), + A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, + bit_shifter16, + A_GPR(gpr - 1), + A_FXBUS2(0x1a)); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xd), + A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, + bit_shifter16, + A_GPR(gpr - 1), + A_FXBUS2(0x1c)); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xe), + A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, + bit_shifter16, + A_GPR(gpr - 1), + A_FXBUS2(0x1e)); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xf), + A_C_00000000, A_C_00000000); + } #if 0 for (z = 4; z < 8; z++) { @@ -2344,14 +2440,13 @@ strcpy(dst, src); } -static int snd_emu10k1_fx8010_info(struct snd_emu10k1 *emu, +static void snd_emu10k1_fx8010_info(struct snd_emu10k1 *emu, struct snd_emu10k1_fx8010_info *info) { char **fxbus, **extin, **extout; unsigned short fxbus_mask, extin_mask, extout_mask; int res; - memset(info, 0, sizeof(info)); info->internal_tram_size = emu->fx8010.itram_size; info->external_tram_size = emu->fx8010.etram_pages.bytes / 2; fxbus = fxbuses; @@ -2368,7 +2463,6 @@ for (res = 16; res < 32; res++, extout++) copy_string(info->extout_names[res], extout_mask & (1 << res) ? *extout : NULL, "Unused", res); info->gpr_controls = emu->fx8010.gpr_count; - return 0; } static int snd_emu10k1_fx8010_ioctl(struct snd_hwdep * hw, struct file *file, unsigned int cmd, unsigned long arg) @@ -2389,10 +2483,7 @@ info = kmalloc(sizeof(*info), GFP_KERNEL); if (!info) return -ENOMEM; - if ((res = snd_emu10k1_fx8010_info(emu, info)) < 0) { - kfree(info); - return res; - } + snd_emu10k1_fx8010_info(emu, info); if (copy_to_user(argp, info, sizeof(*info))) { kfree(info); return -EFAULT; diff -ruN linux-2.6.22/sound/pci/emu10k1/emumixer.c linux-2.6.22-alsa/sound/pci/emu10k1/emumixer.c --- linux-2.6.22/sound/pci/emu10k1/emumixer.c 2007-07-28 13:40:44.000000000 +0200 +++ linux-2.6.22-alsa/sound/pci/emu10k1/emumixer.c 2007-09-01 20:55:23.000000000 +0200 @@ -77,6 +77,10 @@ return 0; } +/* + * Items labels in enum mixer controls assigning source data to + * each destination + */ static char *emu1010_src_texts[] = { "Silence", "Dock Mic A", @@ -133,6 +137,9 @@ "DSP 31", }; +/* + * List of data sources available for each destination + */ static unsigned int emu1010_src_regs[] = { EMU_SRC_SILENCE,/* 0 */ EMU_SRC_DOCK_MIC_A1, /* 1 */ @@ -189,6 +196,10 @@ EMU_SRC_ALICE_EMU32B+0xf, /* 52 */ }; +/* + * Data destinations - physical EMU outputs. + * Each destination has an enum mixer control to choose a data source + */ static unsigned int emu1010_output_dst[] = { EMU_DST_DOCK_DAC1_LEFT1, /* 0 */ EMU_DST_DOCK_DAC1_RIGHT1, /* 1 */ @@ -216,6 +227,11 @@ EMU_DST_HANA_ADAT+7, /* 23 */ }; +/* + * Data destinations - HANA outputs going to Alice2 (audigy) for + * capture (EMU32 + I2S links) + * Each destination has an enum mixer control to choose a data source + */ static unsigned int emu1010_input_dst[] = { EMU_DST_ALICE2_EMU32_0, EMU_DST_ALICE2_EMU32_1, @@ -384,15 +400,7 @@ - -static int snd_emu1010_adc_pads_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_emu1010_adc_pads_info snd_ctl_boolean_mono_info static int snd_emu1010_adc_pads_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -440,14 +448,7 @@ EMU1010_ADC_PADS("ADC1 14dB PAD 0202 Capture Switch", EMU_HANA_0202_ADC_PAD1), }; -static int snd_emu1010_dac_pads_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_emu1010_dac_pads_info snd_ctl_boolean_mono_info static int snd_emu1010_dac_pads_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -500,17 +501,19 @@ static int snd_emu1010_internal_clock_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[2] = { - "44100", "48000" + static char *texts[4] = { + "44100", "48000", "SPDIF", "ADAT" }; - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; - uinfo->value.enumerated.items = 2; - if (uinfo->value.enumerated.item > 1) - uinfo->value.enumerated.item = 1; + uinfo->value.enumerated.items = 4; + if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) + uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); return 0; + + } static int snd_emu1010_internal_clock_get(struct snd_kcontrol *kcontrol, @@ -568,6 +571,44 @@ /* Unmute all */ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE ); break; + + case 2: /* Take clock from S/PDIF IN */ + /* Mute all */ + snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_MUTE ); + /* Default fallback clock 48kHz */ + snd_emu1010_fpga_write(emu, EMU_HANA_DEFCLOCK, EMU_HANA_DEFCLOCK_48K ); + /* Word Clock source, sync to S/PDIF input */ + snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, + EMU_HANA_WCLOCK_HANA_SPDIF_IN | EMU_HANA_WCLOCK_1X ); + /* Set LEDs on Audio Dock */ + snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, + EMU_HANA_DOCK_LEDS_2_EXT | EMU_HANA_DOCK_LEDS_2_LOCK ); + /* FIXME: We should set EMU_HANA_DOCK_LEDS_2_LOCK only when clock signal is present and valid */ + /* Allow DLL to settle */ + msleep(10); + /* Unmute all */ + snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE ); + break; + + case 3: + /* Take clock from ADAT IN */ + /* Mute all */ + snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_MUTE ); + /* Default fallback clock 48kHz */ + snd_emu1010_fpga_write(emu, EMU_HANA_DEFCLOCK, EMU_HANA_DEFCLOCK_48K ); + /* Word Clock source, sync to ADAT input */ + snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, + EMU_HANA_WCLOCK_HANA_ADAT_IN | EMU_HANA_WCLOCK_1X ); + /* Set LEDs on Audio Dock */ + snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, EMU_HANA_DOCK_LEDS_2_EXT | EMU_HANA_DOCK_LEDS_2_LOCK ); + /* FIXME: We should set EMU_HANA_DOCK_LEDS_2_LOCK only when clock signal is present and valid */ + /* Allow DLL to settle */ + msleep(10); + /* Unmute all */ + snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE ); + + + break; } } return change; @@ -855,7 +896,7 @@ .access = SNDRV_CTL_ELEM_ACCESS_READ, .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,MASK), - .count = 4, + .count = 3, .info = snd_emu10k1_spdif_info, .get = snd_emu10k1_spdif_get_mask }; @@ -864,7 +905,7 @@ { .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT), - .count = 4, + .count = 3, .info = snd_emu10k1_spdif_info, .get = snd_emu10k1_spdif_get, .put = snd_emu10k1_spdif_put @@ -1310,14 +1351,7 @@ .put = snd_emu10k1_efx_attn_put }; -static int snd_emu10k1_shared_spdif_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_emu10k1_shared_spdif_info snd_ctl_boolean_mono_info static int snd_emu10k1_shared_spdif_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) diff -ruN linux-2.6.22/sound/pci/emu10k1/emupcm.c linux-2.6.22-alsa/sound/pci/emu10k1/emupcm.c --- linux-2.6.22/sound/pci/emu10k1/emupcm.c 2007-07-28 13:40:44.000000000 +0200 +++ linux-2.6.22-alsa/sound/pci/emu10k1/emupcm.c 2007-09-01 20:55:23.000000000 +0200 @@ -1233,24 +1233,26 @@ runtime->hw.rate_min = runtime->hw.rate_max = 48000; spin_lock_irq(&emu->reg_lock); if (emu->card_capabilities->emu1010) { - /* TODO + /* Nb. of channels has been increased to 16 */ + /* TODO * SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE * SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | * SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | * SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000 * rate_min = 44100, * rate_max = 192000, - * channels_min = 8, - * channels_max = 8, + * channels_min = 16, + * channels_max = 16, * Need to add mixer control to fix sample rate * - * There are 16 mono channels of 16bits each. + * There are 32 mono channels of 16bits each. * 24bit Audio uses 2x channels over 16bit * 96kHz uses 2x channels over 48kHz * 192kHz uses 4x channels over 48kHz - * So, for 48kHz 24bit, one has 8 channels - * for 96kHz 24bit, one has 4 channels - * for 192kHz 24bit, one has 2 channels + * So, for 48kHz 24bit, one has 16 channels + * for 96kHz 24bit, one has 8 channels + * for 192kHz 24bit, one has 4 channels + * */ #if 1 switch (emu->emu1010.internal_clock) { @@ -1258,13 +1260,15 @@ /* For 44.1kHz */ runtime->hw.rates = SNDRV_PCM_RATE_44100; runtime->hw.rate_min = runtime->hw.rate_max = 44100; - runtime->hw.channels_min = runtime->hw.channels_max = 8; + runtime->hw.channels_min = + runtime->hw.channels_max = 16; break; case 1: /* For 48kHz */ runtime->hw.rates = SNDRV_PCM_RATE_48000; runtime->hw.rate_min = runtime->hw.rate_max = 48000; - runtime->hw.channels_min = runtime->hw.channels_max = 8; + runtime->hw.channels_min = + runtime->hw.channels_max = 16; break; }; #endif @@ -1282,7 +1286,7 @@ #endif runtime->hw.formats = SNDRV_PCM_FMTBIT_S32_LE; /* efx_voices_mask[0] is expected to be zero - * efx_voices_mask[1] is expected to have 16bits set + * efx_voices_mask[1] is expected to have 32bits set */ } else { runtime->hw.channels_min = runtime->hw.channels_max = 0; @@ -1787,11 +1791,24 @@ /* emu->efx_voices_mask[0] = FXWC_DEFAULTROUTE_C | FXWC_DEFAULTROUTE_A; */ if (emu->audigy) { emu->efx_voices_mask[0] = 0; - emu->efx_voices_mask[1] = 0xffff; + if (emu->card_capabilities->emu1010) + /* Pavel Hofman - 32 voices will be used for + * capture (write mode) - + * each bit = corresponding voice + */ + emu->efx_voices_mask[1] = 0xffffffff; + else + emu->efx_voices_mask[1] = 0xffff; } else { emu->efx_voices_mask[0] = 0xffff0000; emu->efx_voices_mask[1] = 0; } + /* For emu1010, the control has to set 32 upper bits (voices) + * out of the 64 bits (voices) to true for the 16-channels capture + * to work correctly. Correct A_FXWC2 initial value (0xffffffff) + * is already defined but the snd_emu10k1_pcm_efx_voices_mask + * control can override this register's value. + */ kctl = snd_ctl_new1(&snd_emu10k1_pcm_efx_voices_mask, emu); if (!kctl) return -ENOMEM; diff -ruN linux-2.6.22/sound/pci/emu10k1/emuproc.c linux-2.6.22-alsa/sound/pci/emu10k1/emuproc.c --- linux-2.6.22/sound/pci/emu10k1/emuproc.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/pci/emu10k1/emuproc.c 2007-09-01 20:55:23.000000000 +0200 @@ -240,8 +240,42 @@ struct snd_info_buffer *buffer) { struct snd_emu10k1 *emu = entry->private_data; - snd_emu10k1_proc_spdif_status(emu, buffer, "CD-ROM S/PDIF In", CDCS, CDSRCS); - snd_emu10k1_proc_spdif_status(emu, buffer, "Optical or Coax S/PDIF In", GPSCS, GPSRCS); + u32 value; + u32 value2; + unsigned long flags; + u32 rate; + + if (emu->card_capabilities->emu1010) { + spin_lock_irqsave(&emu->emu_lock, flags); + snd_emu1010_fpga_read(emu, 0x38, &value); + spin_unlock_irqrestore(&emu->emu_lock, flags); + if ((value & 0x1) == 0) { + spin_lock_irqsave(&emu->emu_lock, flags); + snd_emu1010_fpga_read(emu, 0x2a, &value); + snd_emu1010_fpga_read(emu, 0x2b, &value2); + spin_unlock_irqrestore(&emu->emu_lock, flags); + rate = 0x1770000 / (((value << 5) | value2)+1); + snd_iprintf(buffer, "ADAT Locked : %u\n", rate); + } else { + snd_iprintf(buffer, "ADAT Unlocked\n"); + } + spin_lock_irqsave(&emu->emu_lock, flags); + snd_emu1010_fpga_read(emu, 0x20, &value); + spin_unlock_irqrestore(&emu->emu_lock, flags); + if ((value & 0x4) == 0) { + spin_lock_irqsave(&emu->emu_lock, flags); + snd_emu1010_fpga_read(emu, 0x28, &value); + snd_emu1010_fpga_read(emu, 0x29, &value2); + spin_unlock_irqrestore(&emu->emu_lock, flags); + rate = 0x1770000 / (((value << 5) | value2)+1); + snd_iprintf(buffer, "SPDIF Locked : %d\n", rate); + } else { + snd_iprintf(buffer, "SPDIF Unlocked\n"); + } + } else { + snd_emu10k1_proc_spdif_status(emu, buffer, "CD-ROM S/PDIF In", CDCS, CDSRCS); + snd_emu10k1_proc_spdif_status(emu, buffer, "Optical or Coax S/PDIF In", GPSCS, GPSRCS); + } #if 0 val = snd_emu10k1_ptr_read(emu, ZVSRCS, 0); snd_iprintf(buffer, "\nZoomed Video\n"); @@ -379,20 +413,16 @@ struct snd_info_buffer *buffer) { struct snd_emu10k1 *emu = entry->private_data; - unsigned long value; + int value; unsigned long flags; - unsigned long regs; int i; snd_iprintf(buffer, "EMU1010 Registers:\n\n"); - for(i = 0; i < 0x30; i+=1) { + for(i = 0; i < 0x40; i+=1) { spin_lock_irqsave(&emu->emu_lock, flags); - regs=i+0x40; /* 0x40 upwards are registers. */ - outl(regs, emu->port + A_IOCFG); - outl(regs | 0x80, emu->port + A_IOCFG); /* High bit clocks the value into the fpga. */ - value = inl(emu->port + A_IOCFG); + snd_emu1010_fpga_read(emu, i, &value); spin_unlock_irqrestore(&emu->emu_lock, flags); - snd_iprintf(buffer, "%02X: %08lX, %02lX\n", i, value, (value >> 8) & 0x7f); + snd_iprintf(buffer, "%02X: %08X, %02X\n", i, value, (value >> 8) & 0x7f); } } @@ -555,9 +585,9 @@ { struct snd_info_entry *entry; #ifdef CONFIG_SND_DEBUG - if ((emu->card_capabilities->emu1010) && - snd_card_proc_new(emu->card, "emu1010_regs", &entry)) { - snd_info_set_text_ops(entry, emu, snd_emu_proc_emu1010_reg_read); + if (emu->card_capabilities->emu1010) { + if (! snd_card_proc_new(emu->card, "emu1010_regs", &entry)) + snd_info_set_text_ops(entry, emu, snd_emu_proc_emu1010_reg_read); } if (! snd_card_proc_new(emu->card, "io_regs", &entry)) { snd_info_set_text_ops(entry, emu, snd_emu_proc_io_reg_read); diff -ruN linux-2.6.22/sound/pci/emu10k1/io.c linux-2.6.22-alsa/sound/pci/emu10k1/io.c --- linux-2.6.22/sound/pci/emu10k1/io.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/pci/emu10k1/io.c 2007-09-01 20:55:23.000000000 +0200 @@ -226,9 +226,9 @@ return 0; } -int snd_emu1010_fpga_write(struct snd_emu10k1 * emu, int reg, int value) +int snd_emu1010_fpga_write(struct snd_emu10k1 * emu, u32 reg, u32 value) { - if (reg < 0 || reg > 0x3f) + if (reg > 0x3f) return 1; reg += 0x40; /* 0x40 upwards are registers. */ if (value < 0 || value > 0x3f) /* 0 to 0x3f are values */ @@ -244,9 +244,9 @@ return 0; } -int snd_emu1010_fpga_read(struct snd_emu10k1 * emu, int reg, int *value) +int snd_emu1010_fpga_read(struct snd_emu10k1 * emu, u32 reg, u32 *value) { - if (reg < 0 || reg > 0x3f) + if (reg > 0x3f) return 1; reg += 0x40; /* 0x40 upwards are registers. */ outl(reg, emu->port + A_IOCFG); @@ -261,7 +261,7 @@ /* Each Destination has one and only one Source, * but one Source can feed any number of Destinations simultaneously. */ -int snd_emu1010_fpga_link_dst_src_write(struct snd_emu10k1 * emu, int dst, int src) +int snd_emu1010_fpga_link_dst_src_write(struct snd_emu10k1 * emu, u32 dst, u32 src) { snd_emu1010_fpga_write(emu, 0x00, ((dst >> 8) & 0x3f) ); snd_emu1010_fpga_write(emu, 0x01, (dst & 0x3f) ); diff -ruN linux-2.6.22/sound/pci/emu10k1/p16v.c linux-2.6.22-alsa/sound/pci/emu10k1/p16v.c --- linux-2.6.22/sound/pci/emu10k1/p16v.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22-alsa/sound/pci/emu10k1/p16v.c 2007-09-01 20:55:23.000000000 +0200 @@ -124,11 +124,12 @@ /* hardware definition */ static struct snd_pcm_hardware snd_p16v_playback_hw = { - .info = (SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_RESUME | - SNDRV_PCM_INFO_MMAP_VALID), + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_SYNC_START, .formats = SNDRV_PCM_FMTBIT_S32_LE, /* Only supports 24-bit samples padded to 32 bits. */ .rates = SNDRV_PCM_RATE_192000 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_44100, .rate_min = 44100, @@ -207,6 +208,11 @@ if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) return err; + runtime->sync.id32[0] = substream->pcm->card->number; + runtime->sync.id32[1] = 'P'; + runtime->sync.id32[2] = 16; + runtime->sync.id32[3] = 'V'; + return 0; } /* open_capture callback */ @@ -448,6 +454,9 @@ break; } snd_pcm_group_for_each_entry(s, substream) { + if (snd_pcm_substream_chip(s) != emu || + s->stream != SNDRV_PCM_STREAM_PLAYBACK) + continue; runtime = s->runtime; epcm = runtime->private_data; channel = substream->pcm->device-emu->p16v_device_offset; diff -ruN linux-2.6.22/sound/pci/ens1370.c linux-2.6.22-alsa/sound/pci/ens1370.c --- linux-2.6.22/sound/pci/ens1370.c 2007-07-28 13:40:44.000000000 +0200 +++ linux-2.6.22-alsa/sound/pci/ens1370.c 2007-09-02 13:30:43.000000000 +0200 @@ -1419,15 +1419,7 @@ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .info = snd_es1371_spdif_info, \ .get = snd_es1371_spdif_get, .put = snd_es1371_spdif_put } -static int snd_es1371_spdif_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_es1371_spdif_info snd_ctl_boolean_mono_info static int snd_es1371_spdif_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1489,15 +1481,7 @@ }; -static int snd_es1373_rear_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_es1373_rear_info snd_ctl_boolean_mono_info static int snd_es1373_rear_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1542,15 +1526,7 @@ .put = snd_es1373_rear_put, }; -static int snd_es1373_line_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_es1373_line_info snd_ctl_boolean_mono_info static int snd_es1373_line_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1607,8 +1583,8 @@ unsigned char rev; /* revision */ }; -static int __devinit es1371_quirk_lookup(struct ensoniq *ensoniq, - struct es1371_quirk *list) +static int es1371_quirk_lookup(struct ensoniq *ensoniq, + struct es1371_quirk *list) { while (list->vid != (unsigned short)PCI_ANY_ID) { if (ensoniq->pci->vendor == list->vid && @@ -1707,15 +1683,7 @@ .get = snd_ensoniq_control_get, .put = snd_ensoniq_control_put, \ .private_value = mask } -static int snd_ensoniq_control_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define snd_ensoniq_control_info snd_ctl_boolean_mono_info static int snd_ensoniq_control_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) diff -ruN linux-2.6.22/sound/pci/ens1370.c.orig linux-2.6.22-alsa/sound/pci/ens1370.c.orig --- linux-2.6.22/sound/pci/ens1370.c.orig 1970-01-01 01:00:00.000000000 +0100 +++ linux-2.6.22-alsa/sound/pci/ens1370.c.orig 2007-09-01 20:55:23.000000000 +0200 @@ -0,0 +1,2496 @@ +/* + * Driver for Ensoniq ES1370/ES1371 AudioPCI soundcard + * Copyright (c) by Jaroslav Kysela , + * Thomas Sailer + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +/* Power-Management-Code ( CONFIG_PM ) + * for ens1371 only ( FIXME ) + * derived from cs4281.c, atiixp.c and via82xx.c + * using http://www.alsa-project.org/~iwai/writing-an-alsa-driver/c1540.htm + * by Kurt J. Bosch + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#ifdef CHIP1371 +#include +#else +#include +#endif +#include +#include + +#ifndef CHIP1371 +#undef CHIP1370 +#define CHIP1370 +#endif + +#ifdef CHIP1370 +#define DRIVER_NAME "ENS1370" +#else +#define DRIVER_NAME "ENS1371" +#endif + + +MODULE_AUTHOR("Jaroslav Kysela , Thomas Sailer "); +MODULE_LICENSE("GPL"); +#ifdef CHIP1370 +MODULE_DESCRIPTION("Ensoniq AudioPCI ES1370"); +MODULE_SUPPORTED_DEVICE("{{Ensoniq,AudioPCI-97 ES1370}," + "{Creative Labs,SB PCI64/128 (ES1370)}}"); +#endif +#ifdef CHIP1371 +MODULE_DESCRIPTION("Ensoniq/Creative AudioPCI ES1371+"); +MODULE_SUPPORTED_DEVICE("{{Ensoniq,AudioPCI ES1371/73}," + "{Ensoniq,AudioPCI ES1373}," + "{Creative Labs,Ectiva EV1938}," + "{Creative Labs,SB PCI64/128 (ES1371/73)}," + "{Creative Labs,Vibra PCI128}," + "{Ectiva,EV1938}}"); +#endif + +#if defined(CONFIG_GAMEPORT) || (defined(MODULE) && defined(CONFIG_GAMEPORT_MODULE)) +#define SUPPORT_JOYSTICK +#endif + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable switches */ +#ifdef SUPPORT_JOYSTICK +#ifdef CHIP1371 +static int joystick_port[SNDRV_CARDS]; +#else +static int joystick[SNDRV_CARDS]; +#endif +#endif +#ifdef CHIP1371 +static int spdif[SNDRV_CARDS]; +static int lineio[SNDRV_CARDS]; +#endif + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for Ensoniq AudioPCI soundcard."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for Ensoniq AudioPCI soundcard."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable Ensoniq AudioPCI soundcard."); +#ifdef SUPPORT_JOYSTICK +#ifdef CHIP1371 +module_param_array(joystick_port, int, NULL, 0444); +MODULE_PARM_DESC(joystick_port, "Joystick port address."); +#else +module_param_array(joystick, bool, NULL, 0444); +MODULE_PARM_DESC(joystick, "Enable joystick."); +#endif +#endif /* SUPPORT_JOYSTICK */ +#ifdef CHIP1371 +module_param_array(spdif, int, NULL, 0444); +MODULE_PARM_DESC(spdif, "S/PDIF output (-1 = none, 0 = auto, 1 = force)."); +module_param_array(lineio, int, NULL, 0444); +MODULE_PARM_DESC(lineio, "Line In to Rear Out (0 = auto, 1 = force)."); +#endif + +/* ES1371 chip ID */ +/* This is a little confusing because all ES1371 compatible chips have the + same DEVICE_ID, the only thing differentiating them is the REV_ID field. + This is only significant if you want to enable features on the later parts. + Yes, I know it's stupid and why didn't we use the sub IDs? +*/ +#define ES1371REV_ES1373_A 0x04 +#define ES1371REV_ES1373_B 0x06 +#define ES1371REV_CT5880_A 0x07 +#define CT5880REV_CT5880_C 0x02 +#define CT5880REV_CT5880_D 0x03 /* ??? -jk */ +#define CT5880REV_CT5880_E 0x04 /* mw */ +#define ES1371REV_ES1371_B 0x09 +#define EV1938REV_EV1938_A 0x00 +#define ES1371REV_ES1373_8 0x08 + +/* + * Direct registers + */ + +#define ES_REG(ensoniq, x) ((ensoniq)->port + ES_REG_##x) + +#define ES_REG_CONTROL 0x00 /* R/W: Interrupt/Chip select control register */ +#define ES_1370_ADC_STOP (1<<31) /* disable capture buffer transfers */ +#define ES_1370_XCTL1 (1<<30) /* general purpose output bit */ +#define ES_1373_BYPASS_P1 (1<<31) /* bypass SRC for PB1 */ +#define ES_1373_BYPASS_P2 (1<<30) /* bypass SRC for PB2 */ +#define ES_1373_BYPASS_R (1<<29) /* bypass SRC for REC */ +#define ES_1373_TEST_BIT (1<<28) /* should be set to 0 for normal operation */ +#define ES_1373_RECEN_B (1<<27) /* mix record with playback for I2S/SPDIF out */ +#define ES_1373_SPDIF_THRU (1<<26) /* 0 = SPDIF thru mode, 1 = SPDIF == dig out */ +#define ES_1371_JOY_ASEL(o) (((o)&0x03)<<24)/* joystick port mapping */ +#define ES_1371_JOY_ASELM (0x03<<24) /* mask for above */ +#define ES_1371_JOY_ASELI(i) (((i)>>24)&0x03) +#define ES_1371_GPIO_IN(i) (((i)>>20)&0x0f)/* GPIO in [3:0] pins - R/O */ +#define ES_1370_PCLKDIVO(o) (((o)&0x1fff)<<16)/* clock divide ratio for DAC2 */ +#define ES_1370_PCLKDIVM ((0x1fff)<<16) /* mask for above */ +#define ES_1370_PCLKDIVI(i) (((i)>>16)&0x1fff)/* clock divide ratio for DAC2 */ +#define ES_1371_GPIO_OUT(o) (((o)&0x0f)<<16)/* GPIO out [3:0] pins - W/R */ +#define ES_1371_GPIO_OUTM (0x0f<<16) /* mask for above */ +#define ES_MSFMTSEL (1<<15) /* MPEG serial data format; 0 = SONY, 1 = I2S */ +#define ES_1370_M_SBB (1<<14) /* clock source for DAC - 0 = clock generator; 1 = MPEG clocks */ +#define ES_1371_SYNC_RES (1<<14) /* Warm AC97 reset */ +#define ES_1370_WTSRSEL(o) (((o)&0x03)<<12)/* fixed frequency clock for DAC1 */ +#define ES_1370_WTSRSELM (0x03<<12) /* mask for above */ +#define ES_1371_ADC_STOP (1<<13) /* disable CCB transfer capture information */ +#define ES_1371_PWR_INTRM (1<<12) /* power level change interrupts enable */ +#define ES_1370_DAC_SYNC (1<<11) /* DAC's are synchronous */ +#define ES_1371_M_CB (1<<11) /* capture clock source; 0 = AC'97 ADC; 1 = I2S */ +#define ES_CCB_INTRM (1<<10) /* CCB voice interrupts enable */ +#define ES_1370_M_CB (1<<9) /* capture clock source; 0 = ADC; 1 = MPEG */ +#define ES_1370_XCTL0 (1<<8) /* generap purpose output bit */ +#define ES_1371_PDLEV(o) (((o)&0x03)<<8) /* current power down level */ +#define ES_1371_PDLEVM (0x03<<8) /* mask for above */ +#define ES_BREQ (1<<7) /* memory bus request enable */ +#define ES_DAC1_EN (1<<6) /* DAC1 playback channel enable */ +#define ES_DAC2_EN (1<<5) /* DAC2 playback channel enable */ +#define ES_ADC_EN (1<<4) /* ADC capture channel enable */ +#define ES_UART_EN (1<<3) /* UART enable */ +#define ES_JYSTK_EN (1<<2) /* Joystick module enable */ +#define ES_1370_CDC_EN (1<<1) /* Codec interface enable */ +#define ES_1371_XTALCKDIS (1<<1) /* Xtal clock disable */ +#define ES_1370_SERR_DISABLE (1<<0) /* PCI serr signal disable */ +#define ES_1371_PCICLKDIS (1<<0) /* PCI clock disable */ +#define ES_REG_STATUS 0x04 /* R/O: Interrupt/Chip select status register */ +#define ES_INTR (1<<31) /* Interrupt is pending */ +#define ES_1371_ST_AC97_RST (1<<29) /* CT5880 AC'97 Reset bit */ +#define ES_1373_REAR_BIT27 (1<<27) /* rear bits: 000 - front, 010 - mirror, 101 - separate */ +#define ES_1373_REAR_BIT26 (1<<26) +#define ES_1373_REAR_BIT24 (1<<24) +#define ES_1373_GPIO_INT_EN(o)(((o)&0x0f)<<20)/* GPIO [3:0] pins - interrupt enable */ +#define ES_1373_SPDIF_EN (1<<18) /* SPDIF enable */ +#define ES_1373_SPDIF_TEST (1<<17) /* SPDIF test */ +#define ES_1371_TEST (1<<16) /* test ASIC */ +#define ES_1373_GPIO_INT(i) (((i)&0x0f)>>12)/* GPIO [3:0] pins - interrupt pending */ +#define ES_1370_CSTAT (1<<10) /* CODEC is busy or register write in progress */ +#define ES_1370_CBUSY (1<<9) /* CODEC is busy */ +#define ES_1370_CWRIP (1<<8) /* CODEC register write in progress */ +#define ES_1371_SYNC_ERR (1<<8) /* CODEC synchronization error occurred */ +#define ES_1371_VC(i) (((i)>>6)&0x03) /* voice code from CCB module */ +#define ES_1370_VC(i) (((i)>>5)&0x03) /* voice code from CCB module */ +#define ES_1371_MPWR (1<<5) /* power level interrupt pending */ +#define ES_MCCB (1<<4) /* CCB interrupt pending */ +#define ES_UART (1<<3) /* UART interrupt pending */ +#define ES_DAC1 (1<<2) /* DAC1 channel interrupt pending */ +#define ES_DAC2 (1<<1) /* DAC2 channel interrupt pending */ +#define ES_ADC (1<<0) /* ADC channel interrupt pending */ +#define ES_REG_UART_DATA 0x08 /* R/W: UART data register */ +#define ES_REG_UART_STATUS 0x09 /* R/O: UART status register */ +#define ES_RXINT (1<<7) /* RX interrupt occurred */ +#define ES_TXINT (1<<2) /* TX interrupt occurred */ +#define ES_TXRDY (1<<1) /* transmitter ready */ +#define ES_RXRDY (1<<0) /* receiver ready */ +#define ES_REG_UART_CONTROL 0x09 /* W/O: UART control register */ +#define ES_RXINTEN (1<<7) /* RX interrupt enable */ +#define ES_TXINTENO(o) (((o)&0x03)<<5) /* TX interrupt enable */ +#define ES_TXINTENM (0x03<<5) /* mask for above */ +#define ES_TXINTENI(i) (((i)>>5)&0x03) +#define ES_CNTRL(o) (((o)&0x03)<<0) /* control */ +#define ES_CNTRLM (0x03<<0) /* mask for above */ +#define ES_REG_UART_RES 0x0a /* R/W: UART reserver register */ +#define ES_TEST_MODE (1<<0) /* test mode enabled */ +#define ES_REG_MEM_PAGE 0x0c /* R/W: Memory page register */ +#define ES_MEM_PAGEO(o) (((o)&0x0f)<<0) /* memory page select - out */ +#define ES_MEM_PAGEM (0x0f<<0) /* mask for above */ +#define ES_MEM_PAGEI(i) (((i)>>0)&0x0f) /* memory page select - in */ +#define ES_REG_1370_CODEC 0x10 /* W/O: Codec write register address */ +#define ES_1370_CODEC_WRITE(a,d) ((((a)&0xff)<<8)|(((d)&0xff)<<0)) +#define ES_REG_1371_CODEC 0x14 /* W/R: Codec Read/Write register address */ +#define ES_1371_CODEC_RDY (1<<31) /* codec ready */ +#define ES_1371_CODEC_WIP (1<<30) /* codec register access in progress */ +#define ES_1371_CODEC_PIRD (1<<23) /* codec read/write select register */ +#define ES_1371_CODEC_WRITE(a,d) ((((a)&0x7f)<<16)|(((d)&0xffff)<<0)) +#define ES_1371_CODEC_READS(a) ((((a)&0x7f)<<16)|ES_1371_CODEC_PIRD) +#define ES_1371_CODEC_READ(i) (((i)>>0)&0xffff) + +#define ES_REG_1371_SMPRATE 0x10 /* W/R: Codec rate converter interface register */ +#define ES_1371_SRC_RAM_ADDRO(o) (((o)&0x7f)<<25)/* address of the sample rate converter */ +#define ES_1371_SRC_RAM_ADDRM (0x7f<<25) /* mask for above */ +#define ES_1371_SRC_RAM_ADDRI(i) (((i)>>25)&0x7f)/* address of the sample rate converter */ +#define ES_1371_SRC_RAM_WE (1<<24) /* R/W: read/write control for sample rate converter */ +#define ES_1371_SRC_RAM_BUSY (1<<23) /* R/O: sample rate memory is busy */ +#define ES_1371_SRC_DISABLE (1<<22) /* sample rate converter disable */ +#define ES_1371_DIS_P1 (1<<21) /* playback channel 1 accumulator update disable */ +#define ES_1371_DIS_P2 (1<<20) /* playback channel 1 accumulator update disable */ +#define ES_1371_DIS_R1 (1<<19) /* capture channel accumulator update disable */ +#define ES_1371_SRC_RAM_DATAO(o) (((o)&0xffff)<<0)/* current value of the sample rate converter */ +#define ES_1371_SRC_RAM_DATAM (0xffff<<0) /* mask for above */ +#define ES_1371_SRC_RAM_DATAI(i) (((i)>>0)&0xffff)/* current value of the sample rate converter */ + +#define ES_REG_1371_LEGACY 0x18 /* W/R: Legacy control/status register */ +#define ES_1371_JFAST (1<<31) /* fast joystick timing */ +#define ES_1371_HIB (1<<30) /* host interrupt blocking enable */ +#define ES_1371_VSB (1<<29) /* SB; 0 = addr 0x220xH, 1 = 0x22FxH */ +#define ES_1371_VMPUO(o) (((o)&0x03)<<27)/* base register address; 0 = 0x320xH; 1 = 0x330xH; 2 = 0x340xH; 3 = 0x350xH */ +#define ES_1371_VMPUM (0x03<<27) /* mask for above */ +#define ES_1371_VMPUI(i) (((i)>>27)&0x03)/* base register address */ +#define ES_1371_VCDCO(o) (((o)&0x03)<<25)/* CODEC; 0 = 0x530xH; 1 = undefined; 2 = 0xe80xH; 3 = 0xF40xH */ +#define ES_1371_VCDCM (0x03<<25) /* mask for above */ +#define ES_1371_VCDCI(i) (((i)>>25)&0x03)/* CODEC address */ +#define ES_1371_FIRQ (1<<24) /* force an interrupt */ +#define ES_1371_SDMACAP (1<<23) /* enable event capture for slave DMA controller */ +#define ES_1371_SPICAP (1<<22) /* enable event capture for slave IRQ controller */ +#define ES_1371_MDMACAP (1<<21) /* enable event capture for master DMA controller */ +#define ES_1371_MPICAP (1<<20) /* enable event capture for master IRQ controller */ +#define ES_1371_ADCAP (1<<19) /* enable event capture for ADLIB register; 0x388xH */ +#define ES_1371_SVCAP (1<<18) /* enable event capture for SB registers */ +#define ES_1371_CDCCAP (1<<17) /* enable event capture for CODEC registers */ +#define ES_1371_BACAP (1<<16) /* enable event capture for SoundScape base address */ +#define ES_1371_EXI(i) (((i)>>8)&0x07) /* event number */ +#define ES_1371_AI(i) (((i)>>3)&0x1f) /* event significant I/O address */ +#define ES_1371_WR (1<<2) /* event capture; 0 = read; 1 = write */ +#define ES_1371_LEGINT (1<<0) /* interrupt for legacy events; 0 = interrupt did occur */ + +#define ES_REG_CHANNEL_STATUS 0x1c /* R/W: first 32-bits from S/PDIF channel status block, es1373 */ + +#define ES_REG_SERIAL 0x20 /* R/W: Serial interface control register */ +#define ES_1371_DAC_TEST (1<<22) /* DAC test mode enable */ +#define ES_P2_END_INCO(o) (((o)&0x07)<<19)/* binary offset value to increment / loop end */ +#define ES_P2_END_INCM (0x07<<19) /* mask for above */ +#define ES_P2_END_INCI(i) (((i)>>16)&0x07)/* binary offset value to increment / loop end */ +#define ES_P2_ST_INCO(o) (((o)&0x07)<<16)/* binary offset value to increment / start */ +#define ES_P2_ST_INCM (0x07<<16) /* mask for above */ +#define ES_P2_ST_INCI(i) (((i)<<16)&0x07)/* binary offset value to increment / start */ +#define ES_R1_LOOP_SEL (1<<15) /* ADC; 0 - loop mode; 1 = stop mode */ +#define ES_P2_LOOP_SEL (1<<14) /* DAC2; 0 - loop mode; 1 = stop mode */ +#define ES_P1_LOOP_SEL (1<<13) /* DAC1; 0 - loop mode; 1 = stop mode */ +#define ES_P2_PAUSE (1<<12) /* DAC2; 0 - play mode; 1 = pause mode */ +#define ES_P1_PAUSE (1<<11) /* DAC1; 0 - play mode; 1 = pause mode */ +#define ES_R1_INT_EN (1<<10) /* ADC interrupt enable */ +#define ES_P2_INT_EN (1<<9) /* DAC2 interrupt enable */ +#define ES_P1_INT_EN (1<<8) /* DAC1 interrupt enable */ +#define ES_P1_SCT_RLD (1<<7) /* force sample counter reload for DAC1 */ +#define ES_P2_DAC_SEN (1<<6) /* when stop mode: 0 - DAC2 play back zeros; 1 = DAC2 play back last sample */ +#define ES_R1_MODEO(o) (((o)&0x03)<<4) /* ADC mode; 0 = 8-bit mono; 1 = 8-bit stereo; 2 = 16-bit mono; 3 = 16-bit stereo */ +#define ES_R1_MODEM (0x03<<4) /* mask for above */ +#define ES_R1_MODEI(i) (((i)>>4)&0x03) +#define ES_P2_MODEO(o) (((o)&0x03)<<2) /* DAC2 mode; -- '' -- */ +#define ES_P2_MODEM (0x03<<2) /* mask for above */ +#define ES_P2_MODEI(i) (((i)>>2)&0x03) +#define ES_P1_MODEO(o) (((o)&0x03)<<0) /* DAC1 mode; -- '' -- */ +#define ES_P1_MODEM (0x03<<0) /* mask for above */ +#define ES_P1_MODEI(i) (((i)>>0)&0x03) + +#define ES_REG_DAC1_COUNT 0x24 /* R/W: DAC1 sample count register */ +#define ES_REG_DAC2_COUNT 0x28 /* R/W: DAC2 sample count register */ +#define ES_REG_ADC_COUNT 0x2c /* R/W: ADC sample count register */ +#define ES_REG_CURR_COUNT(i) (((i)>>16)&0xffff) +#define ES_REG_COUNTO(o) (((o)&0xffff)<<0) +#define ES_REG_COUNTM (0xffff<<0) +#define ES_REG_COUNTI(i) (((i)>>0)&0xffff) + +#define ES_REG_DAC1_FRAME 0x30 /* R/W: PAGE 0x0c; DAC1 frame address */ +#define ES_REG_DAC1_SIZE 0x34 /* R/W: PAGE 0x0c; DAC1 frame size */ +#define ES_REG_DAC2_FRAME 0x38 /* R/W: PAGE 0x0c; DAC2 frame address */ +#define ES_REG_DAC2_SIZE 0x3c /* R/W: PAGE 0x0c; DAC2 frame size */ +#define ES_REG_ADC_FRAME 0x30 /* R/W: PAGE 0x0d; ADC frame address */ +#define ES_REG_ADC_SIZE 0x34 /* R/W: PAGE 0x0d; ADC frame size */ +#define ES_REG_FCURR_COUNTO(o) (((o)&0xffff)<<16) +#define ES_REG_FCURR_COUNTM (0xffff<<16) +#define ES_REG_FCURR_COUNTI(i) (((i)>>14)&0x3fffc) +#define ES_REG_FSIZEO(o) (((o)&0xffff)<<0) +#define ES_REG_FSIZEM (0xffff<<0) +#define ES_REG_FSIZEI(i) (((i)>>0)&0xffff) +#define ES_REG_PHANTOM_FRAME 0x38 /* R/W: PAGE 0x0d: phantom frame address */ +#define ES_REG_PHANTOM_COUNT 0x3c /* R/W: PAGE 0x0d: phantom frame count */ + +#define ES_REG_UART_FIFO 0x30 /* R/W: PAGE 0x0e; UART FIFO register */ +#define ES_REG_UF_VALID (1<<8) +#define ES_REG_UF_BYTEO(o) (((o)&0xff)<<0) +#define ES_REG_UF_BYTEM (0xff<<0) +#define ES_REG_UF_BYTEI(i) (((i)>>0)&0xff) + + +/* + * Pages + */ + +#define ES_PAGE_DAC 0x0c +#define ES_PAGE_ADC 0x0d +#define ES_PAGE_UART 0x0e +#define ES_PAGE_UART1 0x0f + +/* + * Sample rate converter addresses + */ + +#define ES_SMPREG_DAC1 0x70 +#define ES_SMPREG_DAC2 0x74 +#define ES_SMPREG_ADC 0x78 +#define ES_SMPREG_VOL_ADC 0x6c +#define ES_SMPREG_VOL_DAC1 0x7c +#define ES_SMPREG_VOL_DAC2 0x7e +#define ES_SMPREG_TRUNC_N 0x00 +#define ES_SMPREG_INT_REGS 0x01 +#define ES_SMPREG_ACCUM_FRAC 0x02 +#define ES_SMPREG_VFREQ_FRAC 0x03 + +/* + * Some contants + */ + +#define ES_1370_SRCLOCK 1411200 +#define ES_1370_SRTODIV(x) (ES_1370_SRCLOCK/(x)-2) + +/* + * Open modes + */ + +#define ES_MODE_PLAY1 0x0001 +#define ES_MODE_PLAY2 0x0002 +#define ES_MODE_CAPTURE 0x0004 + +#define ES_MODE_OUTPUT 0x0001 /* for MIDI */ +#define ES_MODE_INPUT 0x0002 /* for MIDI */ + +/* + + */ + +struct ensoniq { + spinlock_t reg_lock; + struct mutex src_mutex; + + int irq; + + unsigned long playback1size; + unsigned long playback2size; + unsigned long capture3size; + + unsigned long port; + unsigned int mode; + unsigned int uartm; /* UART mode */ + + unsigned int ctrl; /* control register */ + unsigned int sctrl; /* serial control register */ + unsigned int cssr; /* control status register */ + unsigned int uartc; /* uart control register */ + unsigned int rev; /* chip revision */ + + union { +#ifdef CHIP1371 + struct { + struct snd_ac97 *ac97; + } es1371; +#else + struct { + int pclkdiv_lock; + struct snd_ak4531 *ak4531; + } es1370; +#endif + } u; + + struct pci_dev *pci; + struct snd_card *card; + struct snd_pcm *pcm1; /* DAC1/ADC PCM */ + struct snd_pcm *pcm2; /* DAC2 PCM */ + struct snd_pcm_substream *playback1_substream; + struct snd_pcm_substream *playback2_substream; + struct snd_pcm_substream *capture_substream; + unsigned int p1_dma_size; + unsigned int p2_dma_size; + unsigned int c_dma_size; + unsigned int p1_period_size; + unsigned int p2_period_size; + unsigned int c_period_size; + struct snd_rawmidi *rmidi; + struct snd_rawmidi_substream *midi_input; + struct snd_rawmidi_substream *midi_output; + + unsigned int spdif; + unsigned int spdif_default; + unsigned int spdif_stream; + +#ifdef CHIP1370 + struct snd_dma_buffer dma_bug; +#endif + +#ifdef SUPPORT_JOYSTICK + struct gameport *gameport; +#endif +}; + +static irqreturn_t snd_audiopci_interrupt(int irq, void *dev_id); + +static struct pci_device_id snd_audiopci_ids[] = { +#ifdef CHIP1370 + { 0x1274, 0x5000, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* ES1370 */ +#endif +#ifdef CHIP1371 + { 0x1274, 0x1371, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* ES1371 */ + { 0x1274, 0x5880, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* ES1373 - CT5880 */ + { 0x1102, 0x8938, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* Ectiva EV1938 */ +#endif + { 0, } +}; + +MODULE_DEVICE_TABLE(pci, snd_audiopci_ids); + +/* + * constants + */ + +#define POLL_COUNT 0xa000 + +#ifdef CHIP1370 +static unsigned int snd_es1370_fixed_rates[] = + {5512, 11025, 22050, 44100}; +static struct snd_pcm_hw_constraint_list snd_es1370_hw_constraints_rates = { + .count = 4, + .list = snd_es1370_fixed_rates, + .mask = 0, +}; +static struct snd_ratnum es1370_clock = { + .num = ES_1370_SRCLOCK, + .den_min = 29, + .den_max = 353, + .den_step = 1, +}; +static struct snd_pcm_hw_constraint_ratnums snd_es1370_hw_constraints_clock = { + .nrats = 1, + .rats = &es1370_clock, +}; +#else +static struct snd_ratden es1371_dac_clock = { + .num_min = 3000 * (1 << 15), + .num_max = 48000 * (1 << 15), + .num_step = 3000, + .den = 1 << 15, +}; +static struct snd_pcm_hw_constraint_ratdens snd_es1371_hw_constraints_dac_clock = { + .nrats = 1, + .rats = &es1371_dac_clock, +}; +static struct snd_ratnum es1371_adc_clock = { + .num = 48000 << 15, + .den_min = 32768, + .den_max = 393216, + .den_step = 1, +}; +static struct snd_pcm_hw_constraint_ratnums snd_es1371_hw_constraints_adc_clock = { + .nrats = 1, + .rats = &es1371_adc_clock, +}; +#endif +static const unsigned int snd_ensoniq_sample_shift[] = + {0, 1, 1, 2}; + +/* + * common I/O routines + */ + +#ifdef CHIP1371 + +static unsigned int snd_es1371_wait_src_ready(struct ensoniq * ensoniq) +{ + unsigned int t, r = 0; + + for (t = 0; t < POLL_COUNT; t++) { + r = inl(ES_REG(ensoniq, 1371_SMPRATE)); + if ((r & ES_1371_SRC_RAM_BUSY) == 0) + return r; + cond_resched(); + } + snd_printk(KERN_ERR "wait source ready timeout 0x%lx [0x%x]\n", + ES_REG(ensoniq, 1371_SMPRATE), r); + return 0; +} + +static unsigned int snd_es1371_src_read(struct ensoniq * ensoniq, unsigned short reg) +{ + unsigned int temp, i, orig, r; + + /* wait for ready */ + temp = orig = snd_es1371_wait_src_ready(ensoniq); + + /* expose the SRC state bits */ + r = temp & (ES_1371_SRC_DISABLE | ES_1371_DIS_P1 | + ES_1371_DIS_P2 | ES_1371_DIS_R1); + r |= ES_1371_SRC_RAM_ADDRO(reg) | 0x10000; + outl(r, ES_REG(ensoniq, 1371_SMPRATE)); + + /* now, wait for busy and the correct time to read */ + temp = snd_es1371_wait_src_ready(ensoniq); + + if ((temp & 0x00870000) != 0x00010000) { + /* wait for the right state */ + for (i = 0; i < POLL_COUNT; i++) { + temp = inl(ES_REG(ensoniq, 1371_SMPRATE)); + if ((temp & 0x00870000) == 0x00010000) + break; + } + } + + /* hide the state bits */ + r = orig & (ES_1371_SRC_DISABLE | ES_1371_DIS_P1 | + ES_1371_DIS_P2 | ES_1371_DIS_R1); + r |= ES_1371_SRC_RAM_ADDRO(reg); + outl(r, ES_REG(ensoniq, 1371_SMPRATE)); + + return temp; +} + +static void snd_es1371_src_write(struct ensoniq * ensoniq, + unsigned short reg, unsigned short data) +{ + unsigned int r; + + r = snd_es1371_wait_src_ready(ensoniq) & + (ES_1371_SRC_DISABLE | ES_1371_DIS_P1 | + ES_1371_DIS_P2 | ES_1371_DIS_R1); + r |= ES_1371_SRC_RAM_ADDRO(reg) | ES_1371_SRC_RAM_DATAO(data); + outl(r | ES_1371_SRC_RAM_WE, ES_REG(ensoniq, 1371_SMPRATE)); +} + +#endif /* CHIP1371 */ + +#ifdef CHIP1370 + +static void snd_es1370_codec_write(struct snd_ak4531 *ak4531, + unsigned short reg, unsigned short val) +{ + struct ensoniq *ensoniq = ak4531->private_data; + unsigned long end_time = jiffies + HZ / 10; + +#if 0 + printk("CODEC WRITE: reg = 0x%x, val = 0x%x (0x%x), creg = 0x%x\n", + reg, val, ES_1370_CODEC_WRITE(reg, val), ES_REG(ensoniq, 1370_CODEC)); +#endif + do { + if (!(inl(ES_REG(ensoniq, STATUS)) & ES_1370_CSTAT)) { + outw(ES_1370_CODEC_WRITE(reg, val), ES_REG(ensoniq, 1370_CODEC)); + return; + } + schedule_timeout_uninterruptible(1); + } while (time_after(end_time, jiffies)); + snd_printk(KERN_ERR "codec write timeout, status = 0x%x\n", + inl(ES_REG(ensoniq, STATUS))); +} + +#endif /* CHIP1370 */ + +#ifdef CHIP1371 + +static void snd_es1371_codec_write(struct snd_ac97 *ac97, + unsigned short reg, unsigned short val) +{ + struct ensoniq *ensoniq = ac97->private_data; + unsigned int t, x; + + mutex_lock(&ensoniq->src_mutex); + for (t = 0; t < POLL_COUNT; t++) { + if (!(inl(ES_REG(ensoniq, 1371_CODEC)) & ES_1371_CODEC_WIP)) { + /* save the current state for latter */ + x = snd_es1371_wait_src_ready(ensoniq); + outl((x & (ES_1371_SRC_DISABLE | ES_1371_DIS_P1 | + ES_1371_DIS_P2 | ES_1371_DIS_R1)) | 0x00010000, + ES_REG(ensoniq, 1371_SMPRATE)); + /* wait for not busy (state 0) first to avoid + transition states */ + for (t = 0; t < POLL_COUNT; t++) { + if ((inl(ES_REG(ensoniq, 1371_SMPRATE)) & 0x00870000) == + 0x00000000) + break; + } + /* wait for a SAFE time to write addr/data and then do it, dammit */ + for (t = 0; t < POLL_COUNT; t++) { + if ((inl(ES_REG(ensoniq, 1371_SMPRATE)) & 0x00870000) == + 0x00010000) + break; + } + outl(ES_1371_CODEC_WRITE(reg, val), ES_REG(ensoniq, 1371_CODEC)); + /* restore SRC reg */ + snd_es1371_wait_src_ready(ensoniq); + outl(x, ES_REG(ensoniq, 1371_SMPRATE)); + mutex_unlock(&ensoniq->src_mutex); + return; + } + } + mutex_unlock(&ensoniq->src_mutex); + snd_printk(KERN_ERR "codec write timeout at 0x%lx [0x%x]\n", + ES_REG(ensoniq, 1371_CODEC), inl(ES_REG(ensoniq, 1371_CODEC))); +} + +static unsigned short snd_es1371_codec_read(struct snd_ac97 *ac97, + unsigned short reg) +{ + struct ensoniq *ensoniq = ac97->private_data; + unsigned int t, x, fail = 0; + + __again: + mutex_lock(&ensoniq->src_mutex); + for (t = 0; t < POLL_COUNT; t++) { + if (!(inl(ES_REG(ensoniq, 1371_CODEC)) & ES_1371_CODEC_WIP)) { + /* save the current state for latter */ + x = snd_es1371_wait_src_ready(ensoniq); + outl((x & (ES_1371_SRC_DISABLE | ES_1371_DIS_P1 | + ES_1371_DIS_P2 | ES_1371_DIS_R1)) | 0x00010000, + ES_REG(ensoniq, 1371_SMPRATE)); + /* wait for not busy (state 0) first to avoid + transition states */ + for (t = 0; t < POLL_COUNT; t++) { + if ((inl(ES_REG(ensoniq, 1371_SMPRATE)) & 0x00870000) == + 0x00000000) + break; + } + /* wait for a SAFE time to write addr/data and then do it, dammit */ + for (t = 0; t < POLL_COUNT; t++) { + if ((inl(ES_REG(ensoniq, 1371_SMPRATE)) & 0x00870000) == + 0x00010000) + break; + } + outl(ES_1371_CODEC_READS(reg), ES_REG(ensoniq, 1371_CODEC)); + /* restore SRC reg */ + snd_es1371_wait_src_ready(ensoniq); + outl(x, ES_REG(ensoniq, 1371_SMPRATE)); + /* wait for WIP again */ + for (t = 0; t < POLL_COUNT; t++) { + if (!(inl(ES_REG(ensoniq, 1371_CODEC)) & ES_1371_CODEC_WIP)) + break; + } + /* now wait for the stinkin' data (RDY) */ + for (t = 0; t < POLL_COUNT; t++) { + if ((x = inl(ES_REG(ensoniq, 1371_CODEC))) & ES_1371_CODEC_RDY) { + mutex_unlock(&ensoniq->src_mutex); + return ES_1371_CODEC_READ(x); + } + } + mutex_unlock(&ensoniq->src_mutex); + if (++fail > 10) { + snd_printk(KERN_ERR "codec read timeout (final) " + "at 0x%lx, reg = 0x%x [0x%x]\n", + ES_REG(ensoniq, 1371_CODEC), reg, + inl(ES_REG(ensoniq, 1371_CODEC))); + return 0; + } + goto __again; + } + } + mutex_unlock(&ensoniq->src_mutex); + snd_printk(KERN_ERR "es1371: codec read timeout at 0x%lx [0x%x]\n", + ES_REG(ensoniq, 1371_CODEC), inl(ES_REG(ensoniq, 1371_CODEC))); + return 0; +} + +static void snd_es1371_codec_wait(struct snd_ac97 *ac97) +{ + msleep(750); + snd_es1371_codec_read(ac97, AC97_RESET); + snd_es1371_codec_read(ac97, AC97_VENDOR_ID1); + snd_es1371_codec_read(ac97, AC97_VENDOR_ID2); + msleep(50); +} + +static void snd_es1371_adc_rate(struct ensoniq * ensoniq, unsigned int rate) +{ + unsigned int n, truncm, freq, result; + + mutex_lock(&ensoniq->src_mutex); + n = rate / 3000; + if ((1 << n) & ((1 << 15) | (1 << 13) | (1 << 11) | (1 << 9))) + n--; + truncm = (21 * n - 1) | 1; + freq = ((48000UL << 15) / rate) * n; + result = (48000UL << 15) / (freq / n); + if (rate >= 24000) { + if (truncm > 239) + truncm = 239; + snd_es1371_src_write(ensoniq, ES_SMPREG_ADC + ES_SMPREG_TRUNC_N, + (((239 - truncm) >> 1) << 9) | (n << 4)); + } else { + if (truncm > 119) + truncm = 119; + snd_es1371_src_write(ensoniq, ES_SMPREG_ADC + ES_SMPREG_TRUNC_N, + 0x8000 | (((119 - truncm) >> 1) << 9) | (n << 4)); + } + snd_es1371_src_write(ensoniq, ES_SMPREG_ADC + ES_SMPREG_INT_REGS, + (snd_es1371_src_read(ensoniq, ES_SMPREG_ADC + + ES_SMPREG_INT_REGS) & 0x00ff) | + ((freq >> 5) & 0xfc00)); + snd_es1371_src_write(ensoniq, ES_SMPREG_ADC + ES_SMPREG_VFREQ_FRAC, freq & 0x7fff); + snd_es1371_src_write(ensoniq, ES_SMPREG_VOL_ADC, n << 8); + snd_es1371_src_write(ensoniq, ES_SMPREG_VOL_ADC + 1, n << 8); + mutex_unlock(&ensoniq->src_mutex); +} + +static void snd_es1371_dac1_rate(struct ensoniq * ensoniq, unsigned int rate) +{ + unsigned int freq, r; + + mutex_lock(&ensoniq->src_mutex); + freq = ((rate << 15) + 1500) / 3000; + r = (snd_es1371_wait_src_ready(ensoniq) & (ES_1371_SRC_DISABLE | + ES_1371_DIS_P2 | ES_1371_DIS_R1)) | + ES_1371_DIS_P1; + outl(r, ES_REG(ensoniq, 1371_SMPRATE)); + snd_es1371_src_write(ensoniq, ES_SMPREG_DAC1 + ES_SMPREG_INT_REGS, + (snd_es1371_src_read(ensoniq, ES_SMPREG_DAC1 + + ES_SMPREG_INT_REGS) & 0x00ff) | + ((freq >> 5) & 0xfc00)); + snd_es1371_src_write(ensoniq, ES_SMPREG_DAC1 + ES_SMPREG_VFREQ_FRAC, freq & 0x7fff); + r = (snd_es1371_wait_src_ready(ensoniq) & (ES_1371_SRC_DISABLE | + ES_1371_DIS_P2 | ES_1371_DIS_R1)); + outl(r, ES_REG(ensoniq, 1371_SMPRATE)); + mutex_unlock(&ensoniq->src_mutex); +} + +static void snd_es1371_dac2_rate(struct ensoniq * ensoniq, unsigned int rate) +{ + unsigned int freq, r; + + mutex_lock(&ensoniq->src_mutex); + freq = ((rate << 15) + 1500) / 3000; + r = (snd_es1371_wait_src_ready(ensoniq) & (ES_1371_SRC_DISABLE | + ES_1371_DIS_P1 | ES_1371_DIS_R1)) | + ES_1371_DIS_P2; + outl(r, ES_REG(ensoniq, 1371_SMPRATE)); + snd_es1371_src_write(ensoniq, ES_SMPREG_DAC2 + ES_SMPREG_INT_REGS, + (snd_es1371_src_read(ensoniq, ES_SMPREG_DAC2 + + ES_SMPREG_INT_REGS) & 0x00ff) | + ((freq >> 5) & 0xfc00)); + snd_es1371_src_write(ensoniq, ES_SMPREG_DAC2 + ES_SMPREG_VFREQ_FRAC, + freq & 0x7fff); + r = (snd_es1371_wait_src_ready(ensoniq) & (ES_1371_SRC_DISABLE | + ES_1371_DIS_P1 | ES_1371_DIS_R1)); + outl(r, ES_REG(ensoniq, 1371_SMPRATE)); + mutex_unlock(&ensoniq->src_mutex); +} + +#endif /*