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[alsa-devel] Wrong sample rate user space -> kernel space?
by Daniel Mack 07 Jul '09

07 Jul '09

On a PXA embedded board, the sample rate propagation from user space to kernel space seems to be wrong. When I start aplay (v1.0.19) from userspace like this ... # aplay -f dat /dev/urandom Playing raw data '/dev/urandom' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo ... params_rate(params) in my snd_soc_ops .hw_params callback returns 88200, not 48000. Can anyone reproduce this? I guess this is unrelated to PXA and ASoC. Thanks, Daniel

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[alsa-devel] Third time: Unlink PCM resource ?
by Werner Van Belle 07 Jul '09

07 Jul '09

Hello, This is the third time I tried to get information on the same topic but there has been no response. Since it is crucial for me to get this to work, I really would appreciate your input. If necssary I can provide further documentation. The problem is as follows: if a process which opened a DSP device forks of a new process, then this child will have the same handle towards the DSP. What I now want to do is to release the handle in the child without closing the device for the parent as well. So essentially the sequence below illustrates the problem if one is unable to 'unlink' a dsp device. a- parent process creates and open dsp device b- parent process forks of a child 1. Child 1 has access to the same dsp device c- parent process closes the DSP device d- client process can no longer write to the DSP device (these are already odd semantics, but okay, can live with it) e- parent process forks of a second child. Child 2 has no dsp device f- Child 2 tries to open the DSP device. This fails because the device, although closed, is not released yet by Child 1. Is this normal behavior ? Is there some rigid documentation on the closing and sharing semantics of pcm devices ? Is there a way to unlink a device without closing it ? If this is the wrong mailinglist, then who should I contact regarding these erratic semantics ? Wkr, -- Dr. Werner Van Belle http://werner.yellowcouch.org/

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[alsa-devel] [PATCH] [RFC 1/13] Intel SST drivers documentation
by Vinod Koul 07 Jul '09

07 Jul '09

This is the first patch of the series. This patch contains the documentation describing the new audio drivers for upcoming Intel MID platform using Atom Processor which uses DSP core for audio processing named Intel smart sound Technology (SST). There are two drivers involved. One driver is SST driver which governs the DSP engine and the other is MAD (MID Audio driver) driver that governs three different sound cards. The intel SST driver enables offload architecture where by the encoded files like mp3, aac and wma can be directly decoded, post processed and rendered by the DSP engine. More details to follow in the following patches The interfaces exposed are documented here as well Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> new file: Documentation/sound/alsa/intel_sst.txt --- Documentation/sound/alsa/intel_sst.txt | 426 ++++++++++++++++++++++++++++++++ 1 files changed, 426 insertions(+), 0 deletions(-) create mode 100644 Documentation/sound/alsa/intel_sst.txt diff --git a/Documentation/sound/alsa/intel_sst.txt b/Documentation/sound/alsa/intel_sst.txt new file mode 100644 index 0000000..e5d6346 --- /dev/null +++ b/Documentation/sound/alsa/intel_sst.txt @@ -0,0 +1,426 @@ + Intel Smart Sound Technology API documentation + ============================================== + + Vinod Koul <vinod.koul(a)intel.com> + Harsha Priya <priya.harsha(a)intel.com> + +General +------- + +Intel SST is a new hardware component that is capable of decoding and +rendering. This hardware takes in encoded data, decodes and renders the same. +The activity on the CPU is much reduced thus reducing the C0 residency time. +This enables low power playback and increases the battery life. This concept +is made use of in Intel Moblin Internet devices. Intel SST driver is the one +that enables this hardware for use. The firmare file is required to be placed +in /lib/firmware for the driver to download the binary to hardware. + +There are 3 sound cards that Intel SST hardware supports and enables. +Intel SST is the only hardware in the platform that can render to sound card. +These sound cards are exposed and mapped through Intel MID driver which is an +ALSA driver. This driver uses Intel SST driver to send data and controls to the +hardware. + +To make use of the low power playback feature, the application or framework +needs to open a Intel SST driver handle and send the encoded stream +like MP3, AAC or WMA through this driver to play out. If its a PCM stream +the application or framework should use ALSA framework. The PCM stream will +be handled by Intel MID driver which will inturn make use of Intel SST driver +to send out pcm data to sound card. + +To enable the applications or framework to send encoded data, the Intel SST +driver exposes few IOCTLS. The following documentation provides how to use the +Intel SST driver interface. + +This documentation provides the interfaces for rendering audio streams and +tones from the operating system and applications to an audio codec, and +ultimately to the speaker or headphones and capturing/recording +audio streams from input devices + +Interfaces supported: +--------------------- +Intel SST driver supports the following set of file operations listed below for +device /dev/intel_sst: + * open + * close + * mmap + * munmap + * read + * write + * readv + * writev + * ioctl + +Intel SST driver supports the following set of file operations for +device /dev/intel_sst_ctl: + * open + * close + * ioctl +The description of each the above file operations are provided in the following +sections. + +The /dev/intel_sst device can be used by framework/application to send encoded +data + +The /dev/intel_sst_ctl device needs to be opened only by a privilaged +application that can control the policy decisions of the system. (AKA Audio +manager). + +open: +~~~~~ +Application can open the character device through /dev/intel_sst or +/dev/intel_sst_ctl. Open call through /dev/intel_sst will succeed only when a +stream can be allocated for the application. But the stream allocation happens +when the application invokes ioctl system call with SNDRV_SST_STREAM_SET_PARAMS +IO Control to set the stream parameters. Each handle is specific to a stream +only and one stream cannot have multiple handles. +For multiple streams, multiple handles are required to be opened. +/dev/intel_sst_ctl is expected to be used only by Audio Manager. +Audio Manager is also assumed to be running as a privileged process with its +own UID. There is no stream associated when opening with /dev/intel_sst_ctl. +Multiple handles for /dev/intel_sst are not allowed, and only ONE handle is +allowed for Audio Manager. + Error Codes returned: + ~~~~~~~~~~~~~~~~~~~~~ + - EACCES Max handles reached + - ENODEV Device is not working + - EBUSY Firmware load fail + - EINVAL Invalid params, Major or minor number +close: +~~~~~~ +Appication can close the character device through standard linux close call +when it completes the playing or recoding The driver will clean up the stream +related data structures. The driver will free/close the streams allocated. +In case the stream consists of a mmapped buffer then that will be unmapped. + Error Codes returned: + ~~~~~~~~~~~~~~~~~~~~~ + - EBADF Invalid/Corrupted handle + - ENODEV Device is not working + - EBUSY Device is busy, should retry after delay + +mmap: +~~~~~ +Applicatiion can call this standard linux call to get mmaped buffers. This call +needs to be called only after stream alloation, else the call would fail +The optimal size of buffer which can be requested using this call is provided +by SNDRV_SST_DRIVER_INFO ioctl. SST Driver will allocate and return the memory +which needs to be used in SNDRV_SST_MMAP_PLAY/ SNDRV_SST_MMAP_CAPTURE. +This call is not required to be called in case the application wants to send +user buffers for playback/capture.Please note that a stream is required to be +mmapped once in lifetime of stream and unmapped once before close. + Error Codes returned: + ~~~~~~~~~~~~~~~~~~~~~ + - EBADF Invalid handle + - EINVAL Invalid arguments + - ENOMEM Memory allocation failed, + retry with smaller size + - EBUSY Device is busy, should retry after delay +munmap: +~~~~~~~ +Application should call this the standard Linux munmap system call to free any + Error Codes returned: + ~~~~~~~~~~~~~~~~~~~~~ + - EBADF Invalid handle + - EINVAL Invalid arguments, address ranges + - EBUSY Device is busy, should retry after delay +write/writev: +~~~~~~~~~~~~ +The application needs to use this system call when it needs to send buffer(s) +containing frames to play audio/speech to the output devices such as +speakers. It is mandatory to invoke SNSDRV_SST_STREAM_SET_PARAMS +ioctl to set up the parameters for the streams before invoking this +system call. This system call will return error if it +is called for a mmapped stream as its expected that mmapped stream uses +SNDRV_SST_MMAP_PLAY ioctl. The buffer doesnt have any max +frames/byte/word length constraints but needs a minimal of one frame size. +For optimization it is recommended that buffer size be a multiple of 20KB along +the frame boundaries. +This call will return when +1) SNDRV_SST_STREAM_DROP is invoked using another application thread +2) The SST firmware has the watermark level for the PCM samples reached +3) The stream encounters an error + Possible Return Values: + ~~~~~~~~~~~~~~~~~~~~~~~ + +ve Bytes written successfully + - EBADF Invalid handle + - EINVAL Invalid arguments + - EBADRQC Invalid request for the stream + - EIO FW error in write +read/readv: +~~~~~~~~~~~ +The application can call this system call when it needs to capture buffer(s) of +recorded audio/speech from the microphone or any other capture device. It is +mandatory to invoke SNDRV_SST_SET_PARAMS ioctl to set up the parameters for the +streams before invoking this system call. This system call will return error if +it is called for a mmapped stream as its expected that mmapped stream used +SNDRV_SST_MMAP_CAPTURE ioctl. The buffer doesnt have any max frames/byte/word +length constraints but needs minimal of one frame size. For optimization its +recommended that buffer size be multiple of 20KB. +The read() will return when +1) SNDRV_SST_STREAM_DROP is invoked using another application thread +2) The SST firmware identifies the watermark level for the PCM samples is reached +3) The stream encounters an error + Possible Return Values: + ~~~~~~~~~~~~~~~~~~~~~~~ + +ve Bytes read successfully. + - EBADF Invalid handle + - EINVAL Invalid arguments + - EBADRQC Invalid request for the stream + - EIO FW error in read +ioctl: +~~~~~~ +SST driver supports a set of IO controls for each device, /dev/intel_sst and +/dev/intel_sst_ctl. All the IOCTLS are blocked till the corresponding operation +is complete. The IO controls are; + +ioctls for /dev/intel_sst: +~~~~~~~~~~~~~~~~~~~~~~~~~~ +SNDRV_SST_STREAM_SET_PARAMS - Set the stream parameter. If sent first + time, the stream will be allocated and be kept + in initialized state. It is required to send + START IO control to start playing. +SNDRV_SST_STREAM_DROP - Stop the stream dropping all the frames. + ALSA IOCTL reused. +SNDRV_SST_STREAM_PAUSE - Pause the stream, no samples are lost + ALSA IOCTL reused. +SNDRV_SST_STREAM_RESUME - Resume from pause, no samples are lost + ALSA IOCTL reused. +SNDRV_SST_STREAM_GET_TSTAMP - Get the time stamp +SNDRV_SST_STREAM_START - Start the stream playback or capture + ALSA IOCTL reused. +SNDRV_SST_MMAP_PLAY - Send the mmap buffer to driver for playback +SNDRV_SST_MMAP_CAPTURE - Send the mmap buffer to driver for capture + +ioctls for /dev/intel_sst_ctrl: (yet to be implemented in the code) +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +SNDRV_SST_FW_INFO - To get or set SST firmware info. +SNDRV_SST_SET_TARGET_DEVICE - To select the target device + (BT/headphones/speakers/etc) + +Common ioctls for /dev/intel_sst and /dev/intel_sst_ctrl: +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +SNDRV_SST_DRIVER_INFO - To obtain version & capabilities of driver + + +SNDRV_SST_STREAM_SET_PARAMS: +~~~~~~~~~~~~~~~~~~~~~~~~~~~ +This IO Control is to allocate a new stream with the firmware for playback or +capture. The parameters will be used to set up the stream for encoding or +decoding. This operation should be invoked to allocate a stream successfully +before using mmap or write or read system call for playback or capture or any +other ioctl. Upon successful completion of this IO Control, the stream id is +filled with a valid number. If the stream is already allocated then driver will +treat this IOCTL as change of stream parameters. Please note that the +parameters can be only changed when SNDRV_SST_STREAM_DROP has been called. + +Note: When the application or middleware has stopped playing or recording and +the stream is idle, it is recommended to close the device and reopen when it +starts again. It will allow other applications to use the stream for playback +or recording as the platform allows only limited number of parallel streams. + +The set of parameters provided in the structure are: + +Codec Parameters which specify the codec specific parameters. +Post/Pre Processing Parameters specify the post/pre processing enabled +for each of the stream. + + Data Structures: + ~~~~~~~~~~~~~~~~ + struct snd_sst_params { + unsigned long result; + unsigned int stream_id; + enum sst_codec_type codec; + enum snd_sst_stream_op ops; + unsigned int stream_type; + struct snd_sst_stream_params sparams; + }; + + /*result: This field contains the completion status of the + requested operation. The value will vary according the stream + control operations. + + Result Codes Description + 0 Success + 1 Stream is not available + 2 Codec is not available + 3 Codec is not supported + 4 Invalid Stream Parameters + 5 Parameter Modified + 6 Invalid stream identifier*/ + + /*stream_id: This is to identify a particular stream + 0 Request to allocate new stream. + Non Zero Valid stream id*/ + /*codec: This field specifies which codec is required for this stream + Please refer to the code for more info*/ + /*ops: This field specifies if this stream is for playback or capture. + + Operation Code Description + STREAM_OPS_PLAYBACK 0x00 For Playback + STREAM_OPS_CAPTURE 0x01 For Recording + STREAM_OPS_PLAYBACK_DRM 0x02 For Protected content playback + STREAM_OPS_PLAYBACK_ALERT 0x03 Alert playback is treated as same as + STREAM_OPS_PLAYBACK. + STREAM_OPS_CAPTURE_VOICE_CALL 0x04 For Voice conversation recording*/ + + /*stream_type: This field will be used in conjunction with stream_op to + determine the specific operation of the stream + + Stream Type Code Description + STREAM_TYPE_MUSIC 0x01 This stream is for Music capture or playback + STREAM_TYPE_VOICE 0x02 This stream is for Voice capture or playback*/ + + /*params: This field contains the stream parameters. Plese refer the code for + the structure*/ + + +SNDRV_SST_STREAM_DROP: +~~~~~~~~~~~~~~~~~~~~~~ +This IO control is to stop the current stream immediately by dropping all +the frames. If the stream is for playback and it is playing currently, +the SST driver shall discard the remaining frames before entering into the +stopped state. If the stream is for capture (recording) and it is capturing +currently, the driver shall return the current captured data and enter into +the stopped state Any data threads which are blocked and waiting will be +unblocked and returned. + + Error Codes returned: + ~~~~~~~~~~~~~~~~~~~~~ + -EBADF - Invalid File Descriptor + -ENODEV - Device FW/codec not available or Sound card not available + -EINVAL - Invalid argument + -EBADRQC - Bad/Invalid request + -EBUSY - FW didn't respond within timeout + +SNDRV_SST_STREAM_PAUSE: +~~~~~~~~~~~~~~~~~~~~~~~ +This IO Control pauses the running stream. If the stream is already paused, +the operation will return success. The SST driver and firmware shall maintain +the current pointers so that it can resume the stream when requested. +The previously called write/read/writev/readv thread if blocked will continue +to be blocked. + Error Codes returned: + ~~~~~~~~~~~~~~~~~~~~~ + Same as given for SNDRV_SST_STREAM_DROP +SNDRV_SST_STREAM_RESUME: +~~~~~~~~~~~~~~~~~~~~~~~~ +This IO Control is to resume the paused stream. If the stream is not paused already +or it is in running state, the operation will return success + Error Codes returned: + ~~~~~~~~~~~~~~~~~~~~~ + Same as given for SNDRV_SST_STREAM_DROP + +SNDRV_SST_STREAM_GET_TSTAMP: +~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +This IO Control returns the rendered stream time in milliseconds + Data Structures: + ~~~~~~~~~~~~~~~~ + unsigned long long time; /*Elapsed time in milli seconds*/ + +The timestamp will be constantly incremented and can be read anytime. In case +of SNDRV_SST_STREAM_PAUSE, the timestamp will also get PAUSED and return the +same value, as long as resume is not called. In the case of +SNDRV_SST_STREAM_DROP, the timestamp will not be incremented or reset. +It will continue to increment from previous value if playback or capture +is invoked again. When a stream is freed, then timestamp will be reset. +If the stream is muted with SNDRV_SST_MUTE the stream is still getting +rendered at zero volume the timestamp will be updated normally + +SNDRV_SST_STREAM_START: +~~~~~~~~~~~~~~~~~~~~~~~ +This IO Control is to start the stream after stream allocation. This call is required +after any read or write, before which the playback or capture will not start. If the +SNDRV_SST_STREAM_DROP is called, then application needs to call SNDRV_SST_STREAM_START +for starting again. + Error Codes returned: + ~~~~~~~~~~~~~~~~~~~~~ + Same as given for SNDRV_SST_STREAM_DROP + +SNDRV_SST_MMAP_PLAY: +~~~~~~~~~~~~~~~~~~~~ +The application can use this ioctl to send mmapped buffer containing frames to +play audio/speech to the output devices such as speakers. It is mandatory to +invoke SNDRV_SST_STREAM_SET_PARAMS ioctl to set up the parameters for the +streams and mmap() to get mmap buffer before invoking this system call. +Driver will only accept mmap buffer for this ioctl. Application should fill +the size and offset fields properly. The buffer doesnt have any max +frames/byte/word length constraints but needs to be minimal of one frame size. +To optimize this is recommended to be a multiple of 20KB along the frame +boundaries. +This ioctl() will return when +1) SNDRV_SST_STREAM_DROP is invoked using another application thread +2) The SST firmware identifies watermark level is reached +3) The stream encounters an error + Data Structures: + ~~~~~~~~~~~~~~~~ + struct snd_sst_buff_entry { + union { + void *user; + /*user buffer pointer, not valid for this ioctl + Used for write() call*/ + unsigned int offset; + /*mmap offset in mmap memory. Valid for this call*/ + } buffer; + unsigned int size; + /*size of data valid i.e. the size of mmap buffer sent*/ + }; + struct snd_sst_buffs { + unsigned int entries; + /*Number of buffer entries sent, minimum to be one*/ + enum snd_sst_buff_type type; + /*Buffer type sent, required to SST_BUF_MMAP for this ioctl*/ + struct snd_sst_buff_entry *buff; + /*Pointer to the list of buffers*/ + }; + + Possible Return Values: + ~~~~~~~~~~~~~~~~~~~~~~~ + +ve - Number of bytes written successfully + -EBADF - Invalid File Descriptor + -ENODEV - Device is not working + -EINVAL - Invalid argument + -EBADRQC- Bad request + +SNDRV_SST_MMAP_CAPTURE: +~~~~~~~~~~~~~~~~~~~~~~~ +The application can use this ioctl to send mmapped buffer to capture +audio/speech from the input devices such as mic. It is mandatory to +invoke SNDRV_SST_STREAM_SET_PARAMS ioctl to set up the parameters for +the streams and mmap() to get mmap buffer before invoking this system call. +Driver will only accept mmap buffer for this ioctl.Application should fill +the size and offset fields properly. The buffer doesnt have any max +frames/byte/word length constraints but needs to be minimal of one frame size. +To optimize this is recommended to be a multiple of 20KB. +This ioctl() will return when +1) SNDRV_SST_STREAM_DROP is invoked using another application thread +2) The SST firmware identifies the watermark level for the PCM samples is reached +3) The stream encounters an error + Data Structures: + ~~~~~~~~~~~~~~~~ + Same as given in SNDRV_SST_MMAP_PLAY ioctl. + + Possible Return Values: + ~~~~~~~~~~~~~~~~~~~~~~~ + Same as given in SNDRV_SST_MMAP_PLAY ioctl. + +SNDRV_SST_DRIVER_INFO: +~~~~~~~~~~~~~~~~~~~~~~ +This IO Control is to obtain the SST driver static and dynamic information. + Data Structures: + ~~~~~~~~~~~~~~~~ + struct snd_sst_driver_info { + unsigned int version; + /*bit0 - bit7 Minor Version + bit8 - bit15 Major Version + bit16 - bit31 Reserved*/ + unsigned int active_pcm_streams; + /*total number of PCM streams active at the point*/ + unsigned int active_enc_streams; + /*total number of encoded streams active at the point*/ + unsigned int max_pcm_streams; + /*number of PCM streams supported by platform */ + unsigned int max_enc_streams; + /*number of encoded streams supported by platform*/ + unsigned int buf_per_stream; + /*optimal size buffer per stream for mmap call*/ + }; -- 1.5.4.5

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Re: [alsa-devel] Intel HDA using VIA vt1708s
by Takashi Iwai 07 Jul '09

07 Jul '09

At Tue, 7 Jul 2009 15:23:45 +0200, Chrissi k wrote: > > Here's what I do after a fresh reboot: > Change "input source" from "mic" to "front mic" > Turn up all recording mixers to 80%, including "mic boost" and "front mic > boost" > > % arecord -fdat -vv foo.wav > > shouting at the mic...after 10 seconds I do ^C > > % aplay foo.wav > > *silence* OK. Just to be sure, could you try % arecord -fdat -c1 -vv foo.wav ?? Also, attach the alsa-info output during recording with the internal mic. > Is it possible, that the mic is not really connected with the vt1708s, but > rather linked to the internal webcam (which is not working at all btw)? Hm, not impossible, but I don't guess so. The webcam is usually a usb device, and the vendor doesn't do a complicated wiring. > That would explain, why I still get a peak from the external mic-in, even when > I set the input-source to "front mic". > > Thanks for your great help with the unstable-snapshot. Did you actually write > the fix, or was it available all the time? (I thought about using the > unstable-snapshot before but didn't) I wrote the fix shortly. You can see two commits in http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git;a=shortlog;h=… BTW, the patches are now in the normal sound tree, so you can use alsa-driver snapshot (no unstable version) from now on again :) thanks, Takashi

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Re: [alsa-devel] Intel HDA using VIA vt1708s
by Takashi Iwai 07 Jul '09

07 Jul '09

At Tue, 7 Jul 2009 14:57:13 +0200, Chrissi k wrote: > > You were so right, arecord + aplay are working fine. Seems to be a problem > with Audacity. > Only thing that's still not showing any sign of life is the internal mic. Is > that part of alsa or should i consider asking the asus-laptop-module guys? > I tried unchecking any "mute" flags using hda-analyzer, but i don't get a peak > using the internal mic. Did you change "Input Source" mixer element to "Front Mic" (or "Mic") properly? The vt170x support code has still no automatic switch via mic jack detection, thus you have to change the recording source manually. Also, there are "Mic Boost" and "Front Mic Boost" volumes that change the input level, too. Takashi

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Re: [alsa-devel] Intel HDA using VIA vt1708s
by Takashi Iwai 07 Jul '09

07 Jul '09

At Tue, 7 Jul 2009 14:38:44 +0200, Chrissi k wrote: > > Thanks for the advice, i tried > ./configure && make && make install && halt > > After a reboot, sound is instantly working, even playing >2 streams at the > same time (mplayer+youtube+totem) works now, too =) OK, I'll merge to the upstream. > But then i started recording something with Audacity, using the external > mic-in. There is the usual peak, so i saved the file and tried re-playing it, > but now the sound is dead again. EAPD is indeed still enabled, but no more > sound from any source. After a reboot, everything works again. Make sure that it's no application problem or mixer setup problem. Try to reproduce with more simpler command sequence, such as, % arecord -fdat -vv foo.wav % aplay foo.wav thanks, Takashi

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[alsa-devel] Intel HDA using VIA vt1708s
by Chrissi k 07 Jul '09

07 Jul '09

Hi, i'd like to report a bug in alsa, up to 1.0.20, in combination with the via vt1708s codec used for example in the asus X5DIJ laptop per default, patch_via.c enables EAPD during initialization. That's not the case especially for the 1708S, so theres no sound at all from internal laptop speakers and line-out. Enabling EAPD using hda_analyzer fixes this issue. I believe there should be some easy fix by initializing the codec with EAPD. I myself can't commit a patch, because my C -kills suck ;) i'd be really glad if someone could take a look at patch_via.c regards didi2002

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Re: [alsa-devel] [LAD] implementing audio device driver on embedded platforms
by Clemens Ladisch 07 Jul '09

07 Jul '09

Jugalkishore Darak, Pankaj wrote: > We are porting Linux 2.6.25.4 to a platform which has ARM11 as its core > processor. My job is to implement audio device driver. Then you'd better ask on alsa-devel (CC'd). > Most of the docs are meant for PC audio which has sound cards on PCI. > And the source code resides in sound folder under the kernel root. > > Since mine is an embedded platform, where will my architecture specific > code resides? Any platform-specific code will be below your arch/ directory. The DMA controller and the codecs are likely to be used on other platforms too, so they'd go into sound/. > Which all folders under sound folder will be used for embedded devices mostly soc/ > and which APIs to look into? linux/Documentation/sound/alsa/soc/*.txt http://alsa-project.org/main/index.php/ASoC http://opensource.wolfsonmicro.com/content/asoc-audio-embedded-linux-systems HTH Clemens

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[alsa-devel] Fwd: Re: No sound with nVidia Corporation MCP79 High Definition Audio (rev b1) (Macbook Pro 5, 5)
by Andreas Nüßlein 07 Jul '09

07 Jul '09

On Tuesday 07 July 2009 08:00:19 you wrote: > At Mon, 6 Jul 2009 22:14:56 +0200, > > Andreas Nüßlein wrote: > > On Monday 06 July 2009 21:26:18 you wrote: > > > At Mon, 06 Jul 2009 11:16:35 -0600, > > > > > > Sean Burke wrote: > > > > Scríobh Takashi Iwai: > > > > > At Mon, 6 Jul 2009 17:53:46 +0200, > > > > > > > > > > Andreas Nüßlein wrote: > > > > >>> The missing pin configuration initialization was already fixed by > > > > >>> the driver overriding it after checking PCI SSID (which is > > > > >>> different from the codec SSID). So, this should be no problem. > > > > >>> > > > > >>> However, the reason why the analog output doesn't work might be > > > > >>> different from that. There might be something else missing, but > > > > >>> I don't know. > > > > >>> > > > > >>> > > > > >>> Takashi > > > > >> > > > > >> oh =( > > > > >> > > > > >> > > > > >> hmm.. anything i can do? would it help if i tried changing values > > > > >> randomly with hda-analyzer.py? > > > > > > > > > > Well, did the driver without my change work more or less with > > > > > the analog audio, or have you never gotten the analog output? > > > > > You can use the generic parser (i.e. the state without cirrus > > > > > patch) by passing model=generic option to snd-hda-intel. > > > > > > > > For my part, nothing worked with the generic driver. I can't confirm > > > > digital out, but I can confirm that the kfree error is gone. What > > > > options are open for figuring out what remains? > > > > > > Easy things to test are GPIO bits. Run hda-verb like > > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_MASK 0x0f > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x0f > > > or > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIR 0x0f > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x0f > > > > > > etc. CS4206 seems to have 4 GPIO lines, and each bit (0-3) > > > corresponds to each GPIO. In many case, GPIO0 or GPIO1 corresponds to > > > the amplifier (EAPD) bit. > > > Define the GPIO direction of each GPIO bit by SET_GPIO_DIR, and > > > turn on/off the GPIO bits by SET_GPIO_DATA. Running > > > hda-verb /dev/snd/hwC0D0 0x01 GET_GPIO_DATA 0 > > > will show the current GPIO data bits. Or you can check it in codec#* > > > proc file. > > > > > > > > > Takashi > > > > w000000000000000000000000000000000t! =) > > > > takashi, thank you _so_ much! > > > > after running all 4 of those: > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_MASK 0x0f > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x0f > > > or > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIR 0x0f > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x0f > > > > i suddendly had sound!! :D :D :D :D :D :D :D > > only via speakers though - there is no sound via headphones right now. > > > > mixer channels: > > - Master (with Mutebutton), PCM and Front (also with Mute) all work =) > > - i don't know what surround would do (or it's extra switch) > > - headphones-volumes and mute button don't affect the speakers, which is > > good =) > > > > > > is there a way to reset what i did with hda-verb, so that i can figure > > out which combination it was exactly? > > You can just change the value 0x0f to a different value. > At least, you can try commands like > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x01 > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x02 > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x04 > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x08 > and check the speaker output at each time. > Also, check the GPIO direction, > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIR 0x01 > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x01 > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIR 0x02 > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x02 > ... > > Regarding the headphone: is the speaker muted when you plug in the > headphone? If not, it's likely an issue of the jack detection. If > the speaker is muted but no headphone output, it's a missing > initialization (or wrong GPIO setup). > > > Takashi so here are my findings so far: hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x08 is sufficient to turn on the right functionality for the speakers; meaning: - 'front speaker'-channel actually controls the two front speakers within the macbookpro - even controlling left and right channel seperately works =) - 'surround' mute-switch and volume-control (why are those not within on thing btw?) also work and control a third speaker in the mac, which really seems responsibly for surround :) headphones do not yet work, however i made a diff of "/proc/asound/card0/codec#0" with and without something plugged in: $ diff -u without_headphones.txt with_headphones.txt --- without_headphones.txt 2009-07-07 11:41:36.558833270 +0200 +++ with_headphones.txt 2009-07-07 11:41:44.286832992 +0200 @@ -113,7 +113,7 @@ Conn = Unknown, Color = Unknown DefAssociation = 0x2, Sequence = 0x1 Misc = NO_PRESENCE - Pin-ctls: 0x40: OUT + Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Delay: 1 samples @@ -125,7 +125,7 @@ Conn = Unknown, Color = Unknown DefAssociation = 0x2, Sequence = 0x0 Misc = NO_PRESENCE - Pin-ctls: 0x40: OUT + Pin-ctls: 0x00: Delay: 1 samples Connection: 1 0x04 thanks =) nutz

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Re: [alsa-devel] No sound with nVidia Corporation MCP79 High Definition Audio (rev b1) (Macbook Pro 5, 5)
by Takashi Iwai 07 Jul '09

07 Jul '09

At Tue, 07 Jul 2009 00:24:51 -0600, Sean Burke wrote: > > Scríobh Takashi Iwai: > > At Mon, 6 Jul 2009 22:14:56 +0200, > > Andreas Nüßlein wrote: > > > >> On Monday 06 July 2009 21:26:18 you wrote: > >> > >>> At Mon, 06 Jul 2009 11:16:35 -0600, > >>> > >>> Easy things to test are GPIO bits. Run hda-verb like > >>> > >>> hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_MASK 0x0f > >>> hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x0f > >>> or > >>> hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIR 0x0f > >>> hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x0f > >>> > >>> etc. CS4206 seems to have 4 GPIO lines, and each bit (0-3) > >>> corresponds to each GPIO. In many case, GPIO0 or GPIO1 corresponds to > >>> the amplifier (EAPD) bit. > >>> Define the GPIO direction of each GPIO bit by SET_GPIO_DIR, and > >>> turn on/off the GPIO bits by SET_GPIO_DATA. Running > >>> hda-verb /dev/snd/hwC0D0 0x01 GET_GPIO_DATA 0 > >>> will show the current GPIO data bits. Or you can check it in codec#* > >>> proc file. > >>> > >>> > >>> Takashi > >>> > >> w000000000000000000000000000000000t! =) > >> > >> takashi, thank you _so_ much! > >> > >> after running all 4 of those: > >> > >>> hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_MASK 0x0f > >>> hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x0f > >>> or > >>> hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIR 0x0f > >>> hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x0f > >>> > >>> > >> i suddendly had sound!! :D :D :D :D :D :D :D > >> only via speakers though - there is no sound via headphones right now. > >> > >> mixer channels: > >> - Master (with Mutebutton), PCM and Front (also with Mute) all work =) > >> - i don't know what surround would do (or it's extra switch) > >> - headphones-volumes and mute button don't affect the speakers, which is good > >> =) > >> > >> > >> is there a way to reset what i did with hda-verb, so that i can figure out > >> which combination it was exactly? > >> > > > > You can just change the value 0x0f to a different value. > > At least, you can try commands like > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x01 > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x02 > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x04 > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x08 > > and check the speaker output at each time. > > Also, check the GPIO direction, > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIR 0x01 > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x01 > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIR 0x02 > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x02 > > ... > > > > Regarding the headphone: is the speaker muted when you plug in the > > headphone? If not, it's likely an issue of the jack detection. If > > the speaker is muted but no headphone output, it's a missing > > initialization (or wrong GPIO setup). > Setting both _DIR and _DATA to 0x08 worked, as did 0x0f. OK, then GPIO2 is EAPD. > Plugging in > headphones does not mute the speaker. I'll check this later. thanks, Takashi

1 1

[alsa-devel] Model quirks and duplicate device IDs
by Ozan Çağlayan 07 Jul '09

07 Jul '09

Hi, A user reports that toshiba model is working fine with its laptop(Subsystem: 152d:0763). I checked the patch_realtek file and saw that the exact ID was mapped against ALC268_ACER 2 years ago for ALSA issue #3343. Both laptops are not-so-popular local brands and checking against 152d points to: 152d QUANTA Computer Inc according to the pci.ids database. So how should we act on those situations seen that we can't match against DMI data(which would possibly be crap on those laptops)? Thanks, Ozan Caglayan

2 2

[alsa-devel] [PATCH] [RFC 3/13] Intel SST driver include headers
by Vinod Koul 07 Jul '09

07 Jul '09

This patch adds the intel_sst.h header file that is places in the include folder for MAD driver (ALSA sound card driver for the platform given in later patches) to use. This file contains the definitions of interfaces exposed by the SST drivers along with the definitions of all the controls for the sound card to be used by MAD driver. intel_sst_ioctl.h - this file exposes the ioctls definitions for player/middleware to use. Many of the definitions are reuse from ALSA Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> new file: include/sound/intel_sst.h new file: include/sound/intel_sst_ioctl.h --- include/sound/intel_sst.h | 259 +++++++++++++++++++++++++++++ include/sound/intel_sst_ioctl.h | 351 +++++++++++++++++++++++++++++++++++++++ 2 files changed, 610 insertions(+), 0 deletions(-) create mode 100644 include/sound/intel_sst.h create mode 100644 include/sound/intel_sst_ioctl.h diff --git a/include/sound/intel_sst.h b/include/sound/intel_sst.h new file mode 100644 index 0000000..c709390 --- /dev/null +++ b/include/sound/intel_sst.h @@ -0,0 +1,259 @@ +#ifndef __INTEL_SST_H__ +#define __INTEL_SST_H__ +/* + * intel_sst.h - Intel SST Driver for audio engine + * + * Copyright (C) 2008-09 Intel Corporation + * Authors: Vinod Koul <vinod.koul(a)intel.com> + * Harsha Priya <priya.harsha(a)intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * 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; version 2 of the License. + * + * 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. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This driver exposes the audio engine functionalities to the ALSA + * and middleware. + * This file is shared between the SST and MAD drivers + */ + +#ifndef CONFIG_SST_IPC_NOT_INCLUDED +#include <asm/ipc_defs.h> +#endif + +#define SST_CARD_NAMES "intel_mid_card" + +/* +control list Pmic & Lpe +*/ +/*Input controls*/ + +enum sst_controls { + SST_SND_ALLOC = 0x1000, + SST_SND_PAUSE = 0x1001, + SST_SND_RESUME = 0x1002, + SST_SND_DROP = 0x1003, + SST_SND_FREE = 0x1004, + SST_SND_BUFFER_POINTER = 0x1005, + SST_SND_STREAM_INIT = 0x1006, + SST_MAX_CONTROLS = 0x1006, + SST_CONTROL_BASE = 0x1000, +}; + +struct pcm_stream_info { + int str_id; + void *mad_substream; + void (*period_elapsed) (void *mad_substream); + unsigned long buffer_ptr; +}; + +struct sst_control_list { + int control_element; + char *value; + struct sst_control_list *next_elem; +}; + +struct stream_buffer { + unsigned long addr; + int length; +}; + + +/* +SST debug print statement definition +*/ + +static inline long sst_get_time(void) +{ + struct timeval t; + do_gettimeofday(&t); + return t.tv_usec; +} + +#define _sst_printk_(level, format, arg...) \ + printk(level "Time:%8ld::%8ld >> SST: %s %d " format, \ + get_seconds(), sst_get_time(), __func__ , __LINE__, ## arg) + +#define sst_err(format, arg...) _sst_printk_(KERN_ERR, "ERR: " format, ## arg) +#define sst_info(format, arg...) \ + _sst_printk_(KERN_INFO, "INFO: " format, ## arg) + +#ifdef CONFIG_SST_DBG_PRINT +#define sst_dbg(format, arg...) _sst_printk_(KERN_DEBUG, "DBG: " format, ## arg) +#else +#define sst_dbg(format, arg...) do {} while (0); +#endif +struct snd_pmic_ops { + int (*set_input_dev) (int value); + int (*get_input_dev) (int *value); + + int (*set_output_dev) (int value); + int (*get_output_dev) (int *value); + + int (*set_mute) (int dev_id, int value); + int (*get_mute) (int dev_id, int *value); + + int (*set_vol) (int dev_id, int value); + int (*get_vol) (int dev_id, int *value); + + int(*init_card) (void); + void (*set_pcm_params) (int sfreq, int word_size); + + int (*init_card_capture) (void); + +}; + +struct intel_sst_card_ops { + char *module_name; + int vendor_id; + int (*control_set) (int control_element, void *value); + int (*send_buffer) (int str_id, struct stream_buffer *mad_buf); + struct snd_pmic_ops *scard_ops; +}; + +int register_sst_card(struct intel_sst_card_ops *card); +void unregister_sst_card(struct intel_sst_card_ops *card); + +/*periphral interrupt interface*/ +enum lpe_periphral { + LPE_DMA = 1, + LPE_SSP0, + LPE_SSP1 +}; + +/*modified for generic access*/ +struct sc_reg_access { + u16 reg_addr; + u8 value; + u8 mask; +}; +enum sc_reg_access_type { + PMIC_READ = 0, + PMIC_WRITE, + PMIC_READ_MODIFY, +}; + +static inline int sst_sc_reg_access(struct sc_reg_access *sc_access, + int type, int num_val) +{ +#ifndef CONFIG_SST_IPC_NOT_INCLUDED + int i, retval = 0, j = 0, k = 0, count = 0; + struct ipc_pmic_reg_data reg_data; + struct ipc_pmic_mod_reg_data pmic_mod_reg = {0}; + + reg_data.ioc = TRUE; + if (type == PMIC_WRITE) { + do { + int max_retries = 0; + + if (num_val <= 4) + count = num_val; + else + count = 4; +retry_write: + for (i = 0; i < count; i++, j++) { + reg_data.pmic_reg_data[i]. + register_address = sc_access[j].reg_addr; + + reg_data.pmic_reg_data[i].value = + sc_access[j].value; + } + reg_data.num_entries = count; + retval = ipc_pmic_register_write(&reg_data, 0); + if (E_NO_INTERRUPT_ON_IOC == retval && + max_retries < 10) { + sst_err("write communcation needs retry\n"); + max_retries++; + goto retry_write; + } + if (0 != retval) { + /*pmic communication fails*/ + sst_err("pmic write failed \n"); + return retval; + } + num_val -= count; + } while (num_val > 0); + } else if (type == PMIC_READ) { + do { + int max_retries = 0; + if (num_val <= 4) + count = num_val; + else + count = 4; +retry_read: + for (i = 0; i < count; i++, j++) + reg_data.pmic_reg_data[i].register_address + = sc_access[j].reg_addr; + reg_data.num_entries = count; + retval = ipc_pmic_register_read(&reg_data); + if (E_NO_INTERRUPT_ON_IOC == retval && + max_retries < 10) { + sst_err("read communcation needs retry\n"); + max_retries++; + goto retry_read; + } + if (0 != retval) { + /*pmic communication fails*/ + sst_err("pmic read failed \n"); + return retval; + } + + for (i = 0; i < count; i++, k++) + sc_access[k].value = + reg_data.pmic_reg_data[i].value; + num_val -= count; + } while (num_val > 0); + } else { + pmic_mod_reg.ioc = TRUE; + do { + int max_retries = 0; + if (num_val <= 4) + count = num_val; + else + count = 4; +retry_readmod: + for (i = 0; i < count; i++, j++) { + pmic_mod_reg.pmic_mod_reg_data[i]. + register_address = sc_access[j].reg_addr; + pmic_mod_reg.pmic_mod_reg_data[i].value = + sc_access[j].value; + pmic_mod_reg.pmic_mod_reg_data[i].bit_map = + sc_access[j].mask; + } + pmic_mod_reg.num_entries = count; + retval = ipc_pmic_register_read_modify(&pmic_mod_reg); + if (E_NO_INTERRUPT_ON_IOC == retval && + max_retries < 10) { + sst_err("read/modify retry\n"); + max_retries++; + goto retry_readmod; + } + if (0 != retval) { + /*pmic communication fails*/ + sst_err("pmic read_modify failed \n"); + return retval; + } + num_val -= count; + } while (num_val > 0); + } + return retval; +#else + return 0; +#endif +} + +int lpe_mask_periphral_intr(enum lpe_periphral device); +int lpe_unmask_periphral_intr(enum lpe_periphral device); +int lpe_periphral_intr_status(enum lpe_periphral device, int *status); +#endif /*__INTEL_SST_H__*/ diff --git a/include/sound/intel_sst_ioctl.h b/include/sound/intel_sst_ioctl.h new file mode 100644 index 0000000..848ea26 --- /dev/null +++ b/include/sound/intel_sst_ioctl.h @@ -0,0 +1,351 @@ +#ifndef __INTEL_SST_IOCTL_H__ +#define __INTEL_SST_IOCTL_H__ +/* + * intel_sst_ipc.h - Intel SST Driver for audio engine + * + * Copyright (C) 2008-09 Intel Corporation + * Authors: Vinod Koul <vinod.koul(a)intel.com> + * Harsha Priya <priya.harsha(a)intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * 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; version 2 of the License. + * + * 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. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This file defines all sst ioctls + */ + +/*codec and post/pre processing related info*/ + +enum sst_codec_types { +/* AUDIO/MUSIC CODEC Type Definitions */ + SST_CODEC_TYPE_UNKNOWN = 0, + SST_CODEC_TYPE_PCM, /* Pass through Audio codec */ + SST_CODEC_TYPE_MP3, + SST_CODEC_TYPE_MP24, + SST_CODEC_TYPE_AAC, + SST_CODEC_TYPE_AACP, + SST_CODEC_TYPE_eAACP, + SST_CODEC_TYPE_WMA9, + SST_CODEC_TYPE_WMA10, + SST_CODEC_TYPE_WMA10P, + SST_CODEC_TYPE_RA, + SST_CODEC_TYPE_DDAC3, + SST_CODEC_TYPE_STEREO_TRUE_HD, + SST_CODEC_TYPE_STEREO_HD_PLUS, + + /* VOICE CODEC Type Definitions */ + SST_CODEC_TYPE_VOICE_PCM = 0x21, /* Pass through voice codec */ + SRC = 0x64, + MIXER = 0x65, + DOWN_MIXER = 0x66, + VOLUME_CONTROL = 0x67, + OEM1 = 0xC8, + OEM2 = 0xC9, +}; + +enum snd_sst_stream_ops { + STREAM_OPS_PLAYBACK = 0, /* Decode */ + STREAM_OPS_CAPTURE, /* Encode */ + STREAM_OPS_PLAYBACK_DRM, /* Play Audio/Voice */ + STREAM_OPS_PLAYBACK_ALERT, /* Play Audio/Voice */ + STREAM_OPS_CAPTURE_VOICE_CALL, /* CSV Voice recording */ +}; + +enum stream_type { + STREAM_TYPE_MUSIC = 1, + STREAM_TYPE_VOICE +}; + +/* Firmware Version info */ +struct snd_sst_fw_version { + __u8 build; /* build number*/ + __u8 minor; /* minor number*/ + __u8 major; /* major number*/ + __u8 type; /* build type*/ +}; + +/* Port info structure */ +struct snd_sst_port_info { + __u16 port_type; + __u16 reserved; +}; + +/* Mixer info structure */ +struct snd_sst_mix_info { + __u16 max_streams; + __u16 reserved; +}; + +/* PCM Parameters */ +struct snd_pcm_params { + __u16 codec; /* codec type */ + __u8 num_chan; /* 1=Mono, 2=Stereo */ + __u8 pcm_wd_sz; /* 16/24 - bit*/ + __u32 brate; /* Bitrate in bits per second */ + __u32 sfreq; /* Sampling rate in Hz */ + __u16 frame_size; + __u16 samples_per_frame; /* Frame size num samples per frame */ + __u32 period_count; /* period elapsed time count, in samples,*/ +}; + +/* MP3 Music Parameters Message */ +struct snd_mp3_params { + __u16 codec; + __u8 num_chan; /* 1=Mono, 2=Stereo */ + __u8 pcm_wd_sz; /* 16/24 - bit*/ + __u32 brate; /* Use the hard coded value. */ + __u32 sfreq; /* Sampling freq eg. 8000, 441000, 48000 */ + __u8 crc_check; /* crc_check - disable (0) or enable (1) */ + __u8 op_align; /* op align 0- 16 bit, 1- MSB, 2 LSB*/ + __u16 reserved; /* Unused */ +}; + +#define AAC_BIT_STREAM_ADTS 0 +#define AAC_BIT_STREAM_ADIF 1 +#define AAC_BIT_STREAM_RAW 2 + +/* AAC Music Parameters Message */ +struct snd_aac_params { + __u16 codec; + __u8 num_chan; /* 1=Mono, 2=Stereo*/ + __u8 pcm_wd_sz; /* 16/24 - bit*/ + __u32 brate; + __u32 sfreq; /* Sampling freq eg. 8000, 441000, 48000 */ + __u32 aac_srate; /* Plain AAC decoder operating sample rate */ + __u8 mpg_id; /* 0=MPEG-2, 1=MPEG-4 */ + __u8 bs_format; /* input bit stream format adts=0, adif=1, raw=2 */ + __u8 aac_profile; /* 0=Main Profile, 1=LC profile, 3=SSR profile */ + __u8 ext_chl; /* No.of external channels */ + __u8 aot; /* Audio object type. 1=Main , 2=LC , 3=SSR, 4=SBR*/ + __u8 op_align; /* output alignment 0=16 bit , 1=MSB, 2= LSB align */ + __u8 brate_type; /* 0=CBR, 1=VBR */ + __u8 crc_check; /* crc check 0= disable, 1=enable */ + __s8 bit_stream_format[8]; /* input bit stream format adts/adif/raw */ +}; + +/* WMA Music Parameters Message */ +struct snd_wma_params { + __u16 codec; + __u8 num_chan; /* 1=Mono, 2=Stereo */ + __u8 pcm_wd_sz; /* 16/24 - bit*/ + __u32 brate; /* Use the hard coded value. */ + __u32 sfreq; /* Sampling freq eg. 8000, 441000, 48000 */ + __u32 channel_mask; /* Channel Mask */ + __u16 format_tag; /* Format Tag */ + __u16 block_align; /* packet size */ + __u16 wma_encode_opt;/* Encoder option */ + __u8 op_align; /* op align 0- 16 bit, 1- MSB, 2 LSB*/ + __u8 reserved; /* reserved */ +}; + +/* Pre processing param structure */ +struct snd_prp_params { + __u32 reserved; /* No pre-processing defined yet */ +}; + +/* Post processing param structure */ +struct snd_pop_params { + s16 volume; /* Voulme in db */ + u16 ramp_duration; /* Ramp duration in ms */ + u16 mute; /* 0=Un-mute, 1=Mute*/ + u16 reserved; /* */ +}; + +/* Post processing Capability info structure */ +struct snd_sst_postproc_info { + __u32 src_min; /* Supported SRC Min sampling freq */ + __u32 src_max; /* Supported SRC Max sampling freq */ + __u8 src; /* 0=Not supported, 1=Supported */ + __u8 bass_boost; /* 0=Not Supported, 1=Supported */ + __u8 stereo_widening; /* 0=Not Supported, 1=Supported */ + __u8 volume_control; /* 0=Not Supported, 1=Supported */ + __s16 min_vol; /* Minimum value of Volume in dB */ + __s16 max_vol; /* Maximum value of Volume in dB */ +}; + +/* pre processing Capability info structure */ +struct snd_sst_prp_info { + __s16 min_vol; /* Minimum value of Volume in dB */ + __s16 max_vol; /* Maximum value of Volume in dB */ + __u8 volume_control; /* 0=Not Supported, 1=Supported */ + __u8 reserved1; /* for 32 bit alignment */ + __u16 reserved2; /* for 32 bit alignment */ +} __attribute__ ((packed)); + +/* Firmware capabilities info */ +struct snd_sst_fw_info { + struct snd_sst_fw_version fw_version; /* Firmware version */ + __u8 audio_codecs_supported[8]; /* Codecs supported by FW */ + __u32 recommend_min_duration; /* Min duration for Low power Playback*/ + __u8 max_pcm_streams_supported; /*Max number of PCM streams supported */ + __u8 max_enc_streams_supported; /*Max number of Encoded streams */ + __u16 reserved; /* 32 bit alignment*/ + struct snd_sst_postproc_info pop_info; /* Post processing capability*/ + struct snd_sst_prp_info prp_info; /* pre_processing mod cap info */ + struct snd_sst_port_info port_info[2]; /* Port info */ + struct snd_sst_mix_info mix_info; /* Mixer info */ + __u32 min_input_buf; /*minmum i/p buffer for decode*/ +}; + +/* Add the codec parameter structures for new codecs to be supported */ +#define CODEC_PARAM_STRUCTURES \ + struct snd_pcm_params pcm_params; \ + struct snd_mp3_params mp3_params; \ + struct snd_aac_params aac_params; \ + struct snd_wma_params wma_params; + +/* Pre and Post Processing param structures */ +#define PPP_PARAM_STRUCTURES \ + struct snd_prp_params prp_params;\ + struct snd_pop_params pop_params; + +/* Codec params struture */ +union snd_sst_codec_params { + CODEC_PARAM_STRUCTURES; +}; + +/* Pre-processing params struture */ +union snd_sst_ppp_params{ + PPP_PARAM_STRUCTURES; +}; + +struct snd_sst_stream_params { + union snd_sst_codec_params uc; + union snd_sst_ppp_params up; +} __attribute__ ((packed)); + +struct snd_sst_params { + unsigned long result; + unsigned int stream_id; + enum sst_codec_types codec; + enum snd_sst_stream_ops ops; + unsigned int stream_type; + struct snd_sst_stream_params sparams; +}; + +/*ioctl related stuff here*/ +struct snd_sst_pmic_config { + __u32 sfreq; /* Sampling rate in Hz */ + __u16 num_chan; /* Mono =1 or Stereo =2 */ + __u16 pcm_wd_sz; /* Number of bits per sample */ +} __attribute__ ((packed)); + +struct snd_sst_get_stream_params { + struct snd_sst_params codec_params; + struct snd_sst_pmic_config pcm_params; +}; + +enum snd_sst_target_type { + SND_SST_TARGET_PMIC = 1, + SND_SST_TARGET_OTHER, +}; + +enum snd_sst_port_action { + SND_SST_PORT_PREPARE = 1, + SND_SST_PORT_ACTIVATE, +}; + +/* Target selection per device structure */ +struct snd_sst_slot_info { + __u8 mix_enable; /* Mixer enable or disable */ + __u8 device_type; + __u8 device_instance; /* 0, 1, 2 */ + __u16 slot[2]; + struct snd_sst_pmic_config pcm_params; + enum snd_sst_target_type type; + __s8 master; + enum snd_sst_port_action action; +} __attribute__ ((packed)); + +/* Target device list structure */ +struct snd_sst_target_device { + __u32 device_route; + struct snd_sst_slot_info devices[2]; +} __attribute__ ((packed)); + +struct snd_sst_driver_info { + unsigned int version; /* Version of the driver */ + unsigned int active_pcm_streams; + unsigned int active_enc_streams; + unsigned int max_pcm_streams; + unsigned int max_enc_streams; + unsigned int buf_per_stream; +}; + +struct snd_sst_vol { + unsigned int stream_id; + int volume; + unsigned long ramp_duration; + __u32 ramp_type; /* Ramp type, default=0 */ +}; + +struct snd_sst_mute { + unsigned int stream_id; + int mute; +}; + +enum snd_sst_buff_type { + SST_BUF_USER = 1, + SST_BUF_MMAP, + SST_BUF_RAR, +}; + +struct snd_sst_buff_entry { + union { + void *user; + unsigned int offset; + } buffer; + unsigned int size; +}; + +struct snd_sst_buffs { + unsigned int entries; + enum snd_sst_buff_type type; + struct snd_sst_buff_entry *buff; +}; + +struct snd_sst_dbufs { + struct snd_sst_buff_entry *ibufs; + struct snd_sst_buff_entry *obufs; + __s8 force; +}; + +/*IOCTL defined here*/ +/*SST MMF IOCTLS only*/ +#define SNDRV_SST_STREAM_SET_PARAMS _IOR('L', 0x00, \ + struct snd_sst_stream_params *) +#define SNDRV_SST_STREAM_GET_PARAMS _IOWR('L', 0x01, \ + struct snd_sst_get_stream_params *) +#define SNDRV_SST_STREAM_GET_TSTAMP _IOWR('L', 0x02, unsigned long long *) +#define SNDRV_SST_STREAM_DECODE _IOWR('L', 0x03, struct snd_sst_dbufs *) +#define SNDRV_SST_STREAM_START _IO('A', 0x42) +#define SNDRV_SST_STREAM_DROP _IO('A', 0x43) +#define SNDRV_SST_STREAM_DRAIN _IO('A', 0x44) +#define SNDRV_SST_STREAM_PAUSE _IOW('A', 0x45, int) +#define SNDRV_SST_STREAM_RESUME _IO('A', 0x47) +#define SNDRV_SST_MMAP_PLAY _IOW('L', 0x05, struct snd_sst_buffs *) +#define SNDRV_SST_MMAP_CAPTURE _IOW('L', 0x06, struct snd_sst_buffs *) +/*SST common ioctls */ +#define SNDRV_SST_DRIVER_INFO _IOR('L', 0x10, struct snd_sst_driver_info *) +#define SNDRV_SST_SET_VOL _IOW('L', 0x11, struct snd_sst_vol *) +#define SNDRV_SST_GET_VOL _IOW('L', 0x12, struct snd_sst_vol *) +#define SNDRV_SST_MUTE _IOW('L', 0x13, struct snd_sst_mute *) + +/*AM Ioctly only*/ +#define SNDRV_SST_FW_INFO _IOR('L', 0x20, struct snd_sst_fw_info *) +#define SNDRV_SST_SET_TARGET_DEVICE _IOW('L', 0x21, \ + struct snd_sst_target_device *) + +#endif /*__INTEL_SST_IOCTL_H__*/ -- 1.5.4.5

4 9

[alsa-devel] [PATCH] [RFC 5/13] Intel SST driver interface module
by Vinod Koul 07 Jul '09

07 Jul '09

This patch adds the SST driver interface modules. Interface module is the one which talks to other layers of SST drivers. intel_sst_interface.c - This file implements the MAD driver registration and de registration functions. SST driver is also a character driver that allows players/middleware to communicate with SST driver. All char driver routines are implemented here. The ioctls used by middleware to open/close, control and configure stream and transfer the data are implemented here Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> Signed-off-by: R Dharageswari <dharageswari.r(a)intel.com> new file: sound/pci/sst/intel_sst_interface.c --- sound/pci/sst/intel_sst_interface.c | 1359 +++++++++++++++++++++++++++++++++++ 1 files changed, 1359 insertions(+), 0 deletions(-) create mode 100644 sound/pci/sst/intel_sst_interface.c diff --git a/sound/pci/sst/intel_sst_interface.c b/sound/pci/sst/intel_sst_interface.c new file mode 100644 index 0000000..f8cefa5 --- /dev/null +++ b/sound/pci/sst/intel_sst_interface.c @@ -0,0 +1,1359 @@ +/* + * intel_sst_interface.c - Intel SST Driver for audio engine + * + * Copyright (C) 2008-09 Intel Corporation + * Authors: Vinod Koul <vinod.koul(a)intel.com> + * Harsha Priya <priya.harsha(a)intel.com> + * R Dharageswari <dharageswari.r(a)intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * 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; version 2 of the License. + * + * 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. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * This driver exposes the audio engine functionalities to the ALSA + * and middleware. + * Upper layer interfaces (MAD driver, MMF) to SST driver + */ + +#include <linux/cdev.h> +#include <linux/pci.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/module.h> +#include <linux/syscalls.h> +#include <linux/fs.h> +#include <linux/file.h> +#include <linux/fcntl.h> +#include <linux/uaccess.h> +#include <linux/interrupt.h> +#include <linux/list.h> +#include <linux/slab.h> +#include <linux/mm.h> +#include <linux/workqueue.h> +#include <linux/pci.h> +#include <linux/firmware.h> +#include <asm/div64.h> +#include <linux/ioctl.h> +#include <sound/intel_sst.h> +#include <sound/intel_sst_ioctl.h> +#include "intel_sst_fw_ipc.h" +#include "intel_sst_common.h" +#include "intel_sst_pvt.h" +#ifdef CONFIG_SST_OSPM_SUPPORT +#include <linux/intel_mid.h> +#endif + + +int sst_download_fw(void) +{ + int retval = 0; + const struct firmware *fw_sst; + + sst_dbg("SST Downloading FW now...\n"); + retval = request_firmware(&fw_sst, + SST_FW_STD_FILENAME, + &sst_ops->pci->dev); + if (0 != retval) { + sst_err("fw load failed %d \n", retval); + return retval; + } + sst_ops->alloc_block[0].sst_id = 0xFF; + sst_load_fw(fw_sst, NULL); + retval = sst_wait_timeout(sst_ops, &sst_ops->alloc_block[0]); + release_firmware(fw_sst); + sst_ops->alloc_block[0].sst_id = BLOCK_UNINIT; + return retval; +} + +/** +* intel_sst_open - opens a handle to driver +* @i_node: inode structure +* @file_ptr:pointer to file +* +* This function is called by OS when a user space component +* tries to get a driver handle. Only one handle at a time +* will be allowed +*/ +int intel_sst_open(struct inode *i_node, struct file *file_ptr) +{ + dev_t device = i_node->i_rdev; + unsigned int retval = 0; + struct ioctl_pvt_data *data = NULL; + + if (sst_ops->pmic_state != PMIC_SND_INIT_DONE) { + sst_err("Sound card not availble \n"); + return -EIO; + } + + if (SST_UN_INIT == sst_ops->sst_state) { + /*FW is not downloaded*/ + retval = sst_download_fw(); + if (retval != 0) { + sst_err("FW download failed...abort\n"); + return -ENODEV; + } + } + if (device == MKDEV(INTEL_SST_MAJOR, 0)) { + /*app open*/ + if (sst_ops->active_cnt < MAX_ENC_STREAM) { + data = kzalloc(sizeof(*data), GFP_KERNEL); + if (NULL == data) + return -ENOMEM; + sst_ops->active_cnt++; + sst_ops->stream_cnt++; + data->pvt_id = sst_assign_pvt_id(sst_ops); + data->str_id = 0; + file_ptr->private_data = (void *)data; + sst_dbg("sst id allocated = %d!\n", data->pvt_id); + } else + retval = -EACCES; + } else if (device == MKDEV(INTEL_SST_MAJOR, 1)) { + /*audio manager open*/ + if (sst_ops->am_cnt < MAX_AM_HANDLES) { + sst_ops->am_cnt++; + sst_dbg("AM handle opened...\n"); + } else + retval = -EACCES; + } else + retval = -EINVAL; + return retval; +} + +/** +* intel_sst_release - releases a handle to driver +* @i_node: inode structure +* @file_ptr: pointer to file +* +* This function is called by OS when a user space component +* tries to release a driver handle. +*/ +int intel_sst_release(struct inode *i_node, struct file *file_ptr) +{ + dev_t device = i_node->i_rdev; + struct ioctl_pvt_data *data = + (struct ioctl_pvt_data *)file_ptr->private_data; + + sst_dbg("Release called \n"); + if (device == MKDEV(INTEL_SST_MAJOR, 0)) { + /*app close*/ + sst_dbg("Closing app handle \n"); + sst_ops->active_cnt--; + sst_ops->stream_cnt--; + if (0 != sst_free_stream(data->str_id)) { + if (sst_validate_strid(data->str_id) == 0) + sst_clean_stream( + &sst_ops->streams[data->str_id]); + } + kfree(data); + } else if (device == MKDEV(INTEL_SST_MAJOR, 1)) + /*audio manager close*/ + sst_dbg("AM handle closed \n"); + sst_ops->am_cnt--; + return 0; +} + +int intel_sst_mmap(struct file *file_ptr, struct vm_area_struct *vma) +{ + int retval = 0, length = 0; + struct ioctl_pvt_data *data = + (struct ioctl_pvt_data *)file_ptr->private_data; + int str_id = data->str_id; + void *mem_area = NULL; + + retval = sst_validate_strid(str_id); + if (retval != 0) + return -EINVAL; + + length = vma->vm_end - vma->vm_start; + sst_dbg("called for stream %d length 0x%x\n", str_id, length); + + if (length > sst_ops->mmap_len) + return -ENOMEM; + if (sst_ops->mmap_mem == NULL) + return -EIO; + + /* round it up to the page bondary */ + mem_area = (void *)((((unsigned long)sst_ops->mmap_mem) + PAGE_SIZE - 1) + & PAGE_MASK); + + /* map the whole physically contiguous area in one piece */ + retval = remap_pfn_range(vma, + vma->vm_start, + virt_to_phys((void *)mem_area) >> PAGE_SHIFT, + length, + vma->vm_page_prot); + if (retval != 0) { + sst_ops->streams[str_id].mmapped = false; + sst_err("mapping failed %d", retval); + } else + sst_ops->streams[str_id].mmapped = true; + + sst_dbg("mmap ret 0x%x \n", retval); + return retval; +} +/** +* intel_sst_write- send data to play out +*/ +int intel_sst_mmap_play_capture(u32 str_id, struct snd_sst_buffs *mmap_buf) +{ + struct sst_stream_bufs *bufs = NULL; + int retval = 0, i; + struct stream_info *stream = NULL; + struct snd_sst_buff_entry *buf_entry = NULL; + + sst_dbg("called for str_id %d \n", str_id); + retval = sst_validate_strid(str_id); + if (retval != 0) { + sst_err("val failed %d", retval); + return -EINVAL; + } + BUG_ON(!mmap_buf); + + stream = &sst_ops->streams[str_id]; + if (stream->mmapped != true) { + sst_err("stream not mapped!!!"); + return -EIO; + } + + if (stream->status != STREAM_RUNNING && + stream->status != STREAM_INIT) { + sst_err("BAD REQUEST!, stream state is %d\n", stream->status); + return -EBADRQC; + } + + sst_dbg("new buffers count %d status %d\n", + mmap_buf->entries, stream->status); + buf_entry = mmap_buf->buff; + for (i = 0; i < mmap_buf->entries; i++) { + BUG_ON(!buf_entry); + bufs = kzalloc(sizeof(*bufs), GFP_KERNEL); + if (bufs == NULL) + return -ENOMEM; + bufs->size = buf_entry->size; + bufs->offset = buf_entry->buffer.offset; + bufs->addr = sst_ops->mmap_mem; + bufs->in_use = false; + buf_entry++; + /*locking here*/ + mutex_lock(&stream->lock); + list_add_tail(&bufs->node, &stream->bufs); + mutex_unlock(&stream->lock); + } + + mutex_lock(&stream->lock); + stream->data_blk.condition = false; + stream->data_blk.ret_code = 0; + if (stream->status == STREAM_INIT && stream->prev != STREAM_UN_INIT) { + stream->prev = stream->status; + stream->status = STREAM_RUNNING; + if (stream->ops == STREAM_OPS_PLAYBACK) { + sst_dbg("play frames...\n"); + if (sst_play_frame(str_id) < 0) { + sst_err("play frames failed \n"); + mutex_unlock(&stream->lock); + return -EIO; + } + } else if (stream->ops == STREAM_OPS_CAPTURE) { + sst_dbg("capture frames...\n"); + if (sst_capture_frame(str_id) < 0) { + sst_err("capture frames failed \n"); + mutex_unlock(&stream->lock); + return -EIO; + } + } + } + mutex_unlock(&stream->lock); + /*Block the call for reply*/ + if (0 == list_empty(&stream->bufs)) { + sst_dbg("ioctl waiting...\n"); + stream->data_blk.on = true; + retval = sst_wait_interruptible(sst_ops, &stream->data_blk); + } + + sst_dbg("end of play/rec ioctl!!\n"); + return retval; +} + +int sst_play_capture(struct stream_info *stream, int str_id) +{ + int retval; + + stream->data_blk.condition = false; + stream->data_blk.ret_code = 0; + mutex_lock(&stream->lock); + if (stream->status == STREAM_INIT && stream->prev != STREAM_UN_INIT) { + /*stream is started*/ + stream->prev = stream->status; + stream->status = STREAM_RUNNING; + } + + if (stream->status == STREAM_INIT && stream->prev == STREAM_UN_INIT) { + /*stream is not started yet*/ + sst_err("Stream isnt started yet state %d, prev %d \n", + stream->status, stream->prev); + } else if (stream->status == STREAM_RUNNING) { + /*stream is started*/ + if (stream->ops == STREAM_OPS_PLAYBACK) { + if (sst_play_frame(str_id) < 0) { + sst_err("play frames failed \n"); + mutex_unlock(&stream->lock); + return -EIO; + } + } else if (stream->ops == STREAM_OPS_CAPTURE) { + if (sst_capture_frame(str_id) < 0) { + sst_err("capture frames failed \n"); + mutex_unlock(&stream->lock); + return -EIO; + } + } + } else + sst_err("Stream state %d invalid will still block, prev %d\n", + stream->status, stream->prev); + mutex_unlock(&stream->lock); + /*Block the call for reply*/ + sst_dbg("write waiting...\n"); + stream->data_blk.on = true; + retval = sst_wait_interruptible(sst_ops, &stream->data_blk); + if (retval != 0) { + stream->status = STREAM_INIT; + sst_dbg("wait returned error...\n"); + } + return retval; +} + +int snd_sst_fill_kernel_list(struct stream_info *stream, + const struct iovec *iovec, unsigned long nr_segs, + struct list_head *copy_to_list) +{ + struct sst_stream_bufs *stream_bufs = NULL; + unsigned long index, data_not_copied, mmap_len; + unsigned char *bufp; + unsigned long size, copied_size; + int retval = 0, add_to_list = 0; + static int sent_offset; + static unsigned long sent_index; + + stream_bufs = kzalloc(sizeof(*stream_bufs), GFP_KERNEL); + if (stream_bufs == NULL) { + sst_err("memory allocation failed \n"); + return -ENOMEM; + } + + mmap_len = sst_ops->mmap_len; + stream_bufs->addr = sst_ops->mmap_mem; + bufp = stream->cur_ptr; + + sst_dbg("mmap_len - %lx\n", mmap_len); + copied_size = 0; + + if (stream->sg_index == 0) + sent_index = sent_offset = 0; + + for (index = stream->sg_index; index < nr_segs; index++) { + stream->sg_index = index; + sst_dbg("index - %lx, cur_ptr - %p\n", index, stream->cur_ptr); + sst_dbg("base - %p, size - 0x%x\n", iovec[index].iov_base, + iovec[index].iov_len); + sst_dbg("bufp - %p\n", bufp); + if (stream->cur_ptr == NULL) + bufp = iovec[index].iov_base; + + size = ((unsigned long)iovec[index].iov_base + + iovec[index].iov_len) - (unsigned long) bufp; + + sst_dbg("size - %lx\n", size); + if ((copied_size + size) > mmap_len) + size = mmap_len - copied_size; + + sst_dbg("size - %lx\n", size); + + if (stream->ops == STREAM_OPS_PLAYBACK) { + sst_dbg("Playback stream copying now....\n"); + data_not_copied = copy_from_user( + (void *)(stream_bufs->addr + copied_size), + bufp, size); + if (data_not_copied > 0) { + /*Clean up the list and return error code */ + sst_err("copy from user not copied -%ld\n", + data_not_copied); + retval = -EIO; + break; + } + } else { + struct snd_sst_user_cap_list *entry = + kzalloc(sizeof(*entry), GFP_KERNEL); + + if (entry == NULL) { + sst_err("mem alloacation failed \n"); + return -ENOMEM; + } + entry->iov_index = index; + entry->iov_offset = (unsigned long) bufp - + (unsigned long)iovec[index].iov_base; + entry->offset = copied_size; + entry->size = size; + sst_dbg("ENTRY:ioindx %d,iooff %ld,koff %ld,ksz %ld \n", + entry->iov_index, entry->iov_offset, + entry->offset, entry->size); + list_add_tail(&entry->node, copy_to_list); + } + + sst_dbg("cur_ptr - %lx\n", (unsigned long) stream->cur_ptr); + stream->cur_ptr = bufp + size; + + if (((unsigned long)iovec[index].iov_base + + iovec[index].iov_len) == + (unsigned long)stream->cur_ptr) { + stream->cur_ptr = NULL; + stream->sg_index++; + } + + copied_size += size; + sst_dbg("copied_size - %lx\n", copied_size); + if ((copied_size >= mmap_len) || + (stream->sg_index == nr_segs)) { + add_to_list = 1; + } + + if (add_to_list) { + stream_bufs->in_use = false; + stream_bufs->size = copied_size; + /*locking here*/ + mutex_lock(&stream->lock); + list_add_tail(&stream_bufs->node, &stream->bufs); + mutex_unlock(&stream->lock); + break; + } + } + return retval; +} + +int snd_sst_copy_userbuf_capture(struct stream_info *stream, + const struct iovec *iovec, + struct list_head *copy_to_list) +{ + struct snd_sst_user_cap_list *entry, *_entry; + struct sst_stream_bufs *kbufs = NULL, *_kbufs; + int retval = 0; + unsigned long data_not_copied; + + /*copy sent buffers*/ + sst_dbg("capture stream copying to user now...\n"); + list_for_each_entry_safe(kbufs, _kbufs, &stream->bufs, node) { + if (kbufs->in_use == true) { + /*copy to user*/ + list_for_each_entry_safe(entry, _entry, + copy_to_list, node) { + sst_dbg("filling now... \n"); + sst_dbg("iindx %d,ioff %ld,koff %ld,ksz %ld \n", + entry->iov_index, entry->iov_offset, + entry->offset, entry->size); + sst_dbg("Copying at %p size %lx\n", + iovec[entry->iov_index].iov_base + + entry->iov_offset, + entry->size); + data_not_copied = copy_to_user((void *) + iovec[entry->iov_index].iov_base + + entry->iov_offset, + kbufs->addr + entry->offset, + entry->size); + if (data_not_copied > 0) { + /*Clean up the list and return error*/ + sst_err("copy to user err -%ld\n", + data_not_copied); + retval = -EIO; + break; + } + list_del(&entry->node); + kfree(entry); + } + sst_dbg("coming out of loop\n"); + } + } + sst_dbg("end of cap copy\n"); + return retval; +} +/* + * snd_sst_userbufs_play_cap - constructs the list from user buffers + * @iovec: pointer to iovec structure + * @nr_segs: number entries in the iovec structure + * @str_id: stream id + * @stream: pointer to stream_info structure + * This function will traverse the user list and copy the data to the kernel + * space buffers. + */ +int snd_sst_userbufs_play_cap(const struct iovec *iovec, + unsigned long nr_segs, unsigned int str_id, + struct stream_info *stream) +{ + int retval = 0; + LIST_HEAD(copy_to_list); + + + retval = snd_sst_fill_kernel_list(stream, iovec, nr_segs, + &copy_to_list); + + retval = sst_play_capture(stream, str_id); + if (retval < 0) + return retval; + + if (stream->ops == STREAM_OPS_CAPTURE) { + retval = snd_sst_copy_userbuf_capture(stream, iovec, + &copy_to_list); + } + return retval; +} + +int intel_sst_read_write(unsigned int str_id, char __user *buf, size_t count) +{ + int retval = 0; + struct stream_info *stream = NULL; + struct iovec iovec; + unsigned long nr_segs; + + retval = sst_validate_strid(str_id); + if (retval != 0) + return -EINVAL; + stream = &sst_ops->streams[str_id]; + if (stream->mmapped == true) { + sst_err("user write & stream is mapped!!!"); + return -EIO; + } + if (count == 0) { + sst_err("args invalid %d", retval); + return -EINVAL; + } + /*copy user buf details*/ + if (stream->status != STREAM_RUNNING && stream->status != STREAM_INIT) { + sst_err("BAD REQUEST!\n"); + return -EBADRQC; + } + sst_dbg("new buffers %p, copy size %d, status %d\n" , + buf, (int) count, (int) stream->status); + + stream->buf_type = SST_BUF_USER_STATIC; + iovec.iov_base = (void *)buf; + iovec.iov_len = count; + nr_segs = 1; + + do { + retval = snd_sst_userbufs_play_cap(&iovec, nr_segs, + str_id, stream); + + if (retval < 0) + break; + + } while (stream->sg_index < nr_segs); + + stream->sg_index = 0; + stream->cur_ptr = NULL; + return retval; +} + +int intel_sst_write(struct file *file_ptr, const char __user *buf, + size_t count, loff_t *offset) +{ + struct ioctl_pvt_data *data = + (struct ioctl_pvt_data *)file_ptr->private_data; + int str_id = data->str_id; + + sst_dbg("called for %d\n", str_id); + return intel_sst_read_write(str_id, (char __user *)buf, count); +} + +/* + * + * intel_sst_aio_write- send data to play out + * + */ +ssize_t intel_sst_aio_write(struct kiocb *kiocb, const struct iovec *iov, + unsigned long nr_segs, loff_t offset) +{ + int retval = 0; + struct ioctl_pvt_data *data = + (struct ioctl_pvt_data *)kiocb->ki_filp->private_data; + int str_id = data->str_id; + struct stream_info *stream = NULL; + + sst_dbg("entry - %ld\n", nr_segs); + + if (is_sync_kiocb(kiocb) == false) { + sst_dbg("aio_read from user space is not allowed\n"); + return -EINVAL; + } + + sst_dbg("called for str_id %d \n", str_id); + retval = sst_validate_strid(str_id); + if (retval != 0) + return -EINVAL; + stream = &sst_ops->streams[str_id]; + if (stream->mmapped == true) { + sst_err("user write & stream is mapped!!!"); + return -EIO; + } + if (stream->status != STREAM_RUNNING && stream->status != STREAM_INIT) { + sst_err("BAD REQUEST!\n"); + return -EBADRQC; + } + sst_dbg("new segs %ld, offset %d, status %d\n" , + nr_segs, (int) offset, (int) stream->status); + stream->buf_type = SST_BUF_USER_STATIC; + do { + retval = snd_sst_userbufs_play_cap(iov, nr_segs, + str_id, stream); + if (retval < 0) + break; + + } while (stream->sg_index < nr_segs); + + stream->sg_index = 0; + stream->cur_ptr = NULL; + + return retval; +} + +/* +* intel_sst_read- get captured data +*/ +int intel_sst_read(struct file *file_ptr, char __user *buf, + size_t count, loff_t *offset) +{ + struct ioctl_pvt_data *data = + (struct ioctl_pvt_data *)file_ptr->private_data; + int str_id = data->str_id; + + sst_dbg("called for %d\n", str_id); + return intel_sst_read_write(str_id, buf, count); +} + +ssize_t intel_sst_aio_read(struct kiocb *kiocb, const struct iovec *iov, + unsigned long nr_segs, loff_t offset) +{ + int retval = 0; + struct ioctl_pvt_data *data = + (struct ioctl_pvt_data *)kiocb->ki_filp->private_data; + int str_id = data->str_id; + struct stream_info *stream = NULL; + + sst_dbg("entry - %ld\n", nr_segs); + + if (is_sync_kiocb(kiocb) == false) { + sst_dbg("aio_read from user space is not allowed\n"); + return -EINVAL; + } + + sst_dbg("called for str_id %d \n", str_id); + retval = sst_validate_strid(str_id); + if (retval != 0) + return -EINVAL; + stream = &sst_ops->streams[str_id]; + if (stream->mmapped == true) { + sst_err("user write & stream is mapped!!!"); + return -EIO; + } + if (stream->status != STREAM_RUNNING && stream->status != STREAM_INIT) { + sst_err("BAD REQUEST!\n"); + return -EBADRQC; + } + sst_dbg("new segs %ld, offset %d, status %d\n" , + nr_segs, (int) offset, (int) stream->status); + stream->buf_type = SST_BUF_USER_STATIC; + do { + retval = snd_sst_userbufs_play_cap(iov, nr_segs, + str_id, stream); + if (retval < 0) + break; + + } while (stream->sg_index < nr_segs); + + stream->sg_index = 0; + stream->cur_ptr = NULL; + + return retval; +} + +int sst_get_stream_allocated(struct snd_sst_params *str_param, + u32 block, u32 pvt_id) +{ + int retval = 0; + + retval = sst_alloc_stream((char *) &str_param->sparams, str_param->ops, + str_param->codec, pvt_id); + if (0 != retval) { + sst_err("sst_alloc_stream failed %d", retval); + goto err; + } + /*Block the call for reply*/ + retval = sst_wait_timeout(sst_ops, &sst_ops->alloc_block[block]); +err: return retval; +} + +void set_port_params(struct snd_sst_params *str_param, + enum snd_sst_stream_ops ops) +{ + int sfreq = str_param->sparams.uc.pcm_params.sfreq; + int word_size = str_param->sparams.uc.pcm_params.pcm_wd_sz; + + sst_dbg("sampling frequency = %d wd_size = %d \n", sfreq, word_size); + + if (ops == STREAM_OPS_PLAYBACK) { + sst_dbg("Setting playback path and port settings...\n"); + sst_ops->scard_ops.init_card(); + sst_ops->scard_ops.set_pcm_params(sfreq, word_size); + } else if (ops == STREAM_OPS_CAPTURE) { + sst_dbg("Setting capture path...\n"); + if (sst_ops->scard_ops.init_card_capture != NULL) + sst_ops->scard_ops.init_card_capture(); + } + return; +} + +int sst_get_stream(struct snd_sst_params *str_param, u32 pvt_id) +{ + int i = 0, retval = 0; + + /*stream is not allocated, we are allocating*/ + i = sst_get_block_stream(sst_ops); + sst_dbg("alloc block allocated = %d\n", i); + if (i < 0) + return -ENOMEM; + sst_ops->alloc_block[i].sst_id = pvt_id; + + retval = sst_get_stream_allocated(str_param, i, pvt_id); + if (retval == -(SST_LIB_ERR_LIB_DNLD_REQUIRED)) { + /*codec download is required*/ + struct snd_sst_alloc_response *response = + sst_ops->alloc_block[i].ops_block.data; + sst_dbg("Codec is required.... trying that\n"); + retval = sst_load_library(&response->lib_dnld, + str_param->ops, pvt_id); + kfree(response); + + if (retval == 0) { + sst_dbg("codec was downloaded sucesfully \n"); + sst_dbg("try alloac again\n"); + sst_ops->alloc_block[i].ops_block.condition = false; + + retval = sst_get_stream_allocated(str_param, i, pvt_id); + + if (retval <= 0) + goto err; + set_port_params(str_param, str_param->ops); + + sst_dbg("Allocation done stream id %d \n", retval); + } else { + sst_dbg("codec download failed \n"); + retval = -EIO; + goto err; + } + } else if (retval <= 0) + goto err; + else + set_port_params(str_param, str_param->ops); + +err: + sst_ops->alloc_block[i].sst_id = BLOCK_UNINIT; + return retval; +} + +/** +* intel_sst_ioctl - recieves the device ioctl's +* @i_node: inode structure +* @file_ptr: pointer to file +* @cmd: Ioctl cmd +* @arg: data +* +* This function is called by OS when a user space component +* sends an Ioctl to SST driver +*/ +int intel_sst_ioctl(struct inode *i_node, struct file *file_ptr, + unsigned int cmd, unsigned long arg) +{ + int retval = 0, i = 0; + struct ioctl_pvt_data *data = (struct ioctl_pvt_data *) + file_ptr->private_data; + int str_id = data->str_id, minor; + + dev_t device = i_node->i_rdev; + + if (device == MKDEV(INTEL_SST_MAJOR, 0)) + minor = 0; + else if (device == MKDEV(INTEL_SST_MAJOR, 1)) + minor = 1; + else + return -EINVAL; + + if (SST_FW_RUNNING != sst_ops->sst_state) { + sst_err("SST Not runng %d\n", sst_ops->sst_state); + return -EBUSY; + } + switch (_IOC_NR(cmd)) { + case _IOC_NR(SNDRV_SST_STREAM_PAUSE): + sst_dbg("SNDRV_SST_IOCTL_PAUSE recieved for %d!\n", str_id); + if (minor != 0) { + retval = -EINVAL; + break; + } + retval = sst_pause_stream(str_id); + break; + + case _IOC_NR(SNDRV_SST_STREAM_RESUME): + sst_dbg("SNDRV_SST_IOCTL_RESUME recieved!\n"); + if (minor != 0) { + retval = -EINVAL; + break; + } + retval = sst_resume_stream(str_id); + break; + + case _IOC_NR(SNDRV_SST_STREAM_SET_PARAMS): { + struct snd_sst_params *str_param = (struct snd_sst_params *)arg; + + sst_dbg("IOCTL_SET_PARAMS recieved!\n"); + if (minor != 0) { + retval = -EINVAL; + break; + } + sst_print_params(&str_param->sparams); + + if (str_id == 0) { + retval = sst_get_stream(str_param, data->pvt_id); + if (retval > 0) { + data->str_id = retval; + retval = 0; + } else { + if (retval == -SST_ERR_INVALID_PARAMS) + retval = -EINVAL; + } + } else { + sst_dbg("SET_STREAM_PARAMS recieved!\n"); + /*allocated set params only*/ + retval = sst_set_stream_param(str_id, str_param); + /*Block the call for reply*/ + if (retval == 0) { + int sfreq = 0, word_size = 0; + sfreq = str_param->sparams.uc.pcm_params.sfreq; + word_size = str_param->sparams. + uc.pcm_params.pcm_wd_sz; + retval = sst_wait_timeout(sst_ops, + &sst_ops->alloc_block[i]); + sst_dbg("SST sampling frequency= %d\n" , sfreq); + sst_ops->scard_ops.set_pcm_params(sfreq, + word_size); + } + } + break; + } + + case _IOC_NR(SNDRV_SST_MMAP_PLAY): + case _IOC_NR(SNDRV_SST_MMAP_CAPTURE): + sst_dbg("SNDRV_SST_MMAP_PLAY/CAPTURE recieved!\n"); + if (minor != 0) { + retval = -EINVAL; + break; + } + retval = intel_sst_mmap_play_capture(str_id, + (struct snd_sst_buffs *)arg); + break; + + case _IOC_NR(SNDRV_SST_STREAM_DROP): + sst_dbg("SNDRV_SST_IOCTL_DROP recieved!\n"); + if (minor != 0) { + retval = -EINVAL; + break; + } + retval = sst_drop_stream(str_id); + break; + + case _IOC_NR(SNDRV_SST_STREAM_GET_TSTAMP): { + unsigned long long *ms = (unsigned long long *)arg; + struct snd_sst_tstamp tstamp = {0}; + unsigned long long time = 0, freq = 0; + unsigned long mod = 0; + + sst_dbg("SNDRV_SST_STREAM_GET_TSTAMP recieved!\n"); + if (minor != 0) { + retval = -EINVAL; + break; + } + memcpy_fromio(&tstamp, + ((void *)(sst_ops->mailbox + SST_TIME_STAMP) + +(str_id * sizeof(tstamp))), + sizeof(tstamp)); + time = tstamp.samples_rendered; + sst_dbg("samples rendered! = 0x%llx\n", time); + freq = (unsigned long long) tstamp.sampling_frequency; + sst_dbg("freq = %llx\n", freq); + mod = do_div(time, freq); + sst_dbg("mod = 0x%lx\n", mod); + sst_dbg("sec = 0x%llx\n", time); + time = time * 1000; + sst_dbg("msec = 0x%llx\n", time); + retval = copy_to_user(ms, &time, sizeof(*ms)); + if (retval != 0) + sst_err("copy failed = %d\n", retval); + break; + } + + case _IOC_NR(SNDRV_SST_STREAM_START):{ + struct stream_info *stream = NULL; + + sst_dbg("SNDRV_SST_STREAM_START recieved!\n"); + if (minor != 0) { + retval = -EINVAL; + break; + } + retval = sst_validate_strid(str_id); + if (retval != 0) + break; + stream = &sst_ops->streams[str_id]; + mutex_lock(&stream->lock); + if (stream->status == STREAM_INIT) { + sst_dbg("calling play frames...\n"); + stream->prev = stream->status; + stream->status = STREAM_RUNNING; + if (stream->ops == STREAM_OPS_PLAYBACK) { + retval = sst_play_frame(str_id); +#ifdef CONFIG_SST_OSPM_SUPPORT + sst_ospm_send_event( + OSPM_EVENT_SUBSYS_START_PLAY); +#endif + } else if (stream->ops == STREAM_OPS_CAPTURE) + retval = sst_capture_frame(str_id); + else { + sst_err("Invalid ops 0x%x\n", stream->ops); + retval = -EINVAL; + mutex_unlock(&sst_ops->streams[str_id].lock); + break; + } + if (retval < 0) { + sst_err("play/capture frames failed \n"); + stream->status = STREAM_INIT; + mutex_unlock(&sst_ops->streams[str_id].lock); + break; + } + } else { + sst_err("Invalid start for stream %d state 0x%x\n", + str_id, stream->status); + retval = -EINVAL; + } + mutex_unlock(&sst_ops->streams[str_id].lock); + break; + } + case _IOC_NR(SNDRV_SST_SET_TARGET_DEVICE): { + struct snd_sst_target_device *target_device = NULL; + + sst_info("SNDRV_SST_SET_TARGET_PLAYBACK DEVICE recieved!\n"); + target_device = (struct snd_sst_target_device *)arg; + BUG_ON(!target_device); + if (minor != 1) { + sst_err("called for non AM handle minor %d\n", minor); + retval = -EINVAL; + break; + } + retval = sst_target_device_select(target_device); + if (0 != retval) + return retval; + break; + + } + + case _IOC_NR(SNDRV_SST_DRIVER_INFO): { + struct snd_sst_driver_info *info = + (struct snd_sst_driver_info *)arg; + sst_dbg("SNDRV_SST_DRIVER_INFO recived \n"); + info->version = SST_VERSION_NUM; + /*hard coding, shud get sumhow later*/ + info->active_pcm_streams = sst_ops->stream_cnt - + sst_ops->active_cnt; + info->active_enc_streams = sst_ops->active_cnt; + info->max_pcm_streams = MAX_ACTIVE_STREAM - MAX_ENC_STREAM; + info->max_enc_streams = MAX_ENC_STREAM; + info->buf_per_stream = sst_ops->mmap_len; + break; + } + + default: + sst_dbg("IOCTL not supported yet !\n"); + retval = -ENOTTY; + } + sst_dbg("...complete ret code = %d\n", retval); + + return retval; +} + +/* + MAD Routines +*/ + +/** +* sst_control_set - Set Control params +* @control_list: list of controls to be set +* +* This function is called by MID sound card driver to set +* SST/Sound card controls. This is registered with MID driver +*/ +int sst_control_set(int control_element, void *value) +{ + int retval = 0, str_id = 0; + + if (SST_UN_INIT == sst_ops->sst_state) { + /*FW is not downloaded*/ + sst_dbg("DSP Downloading FW now...\n"); + retval = sst_download_fw(); + if (retval != 0) { + sst_err("FW download failed = 0x%x, abort\n", retval); + return retval; + } + } + + switch (control_element) { + case SST_SND_ALLOC: { + struct snd_sst_params *str_param = NULL; + int pcm_id = sst_assign_pvt_id(sst_ops); + + str_param = (struct snd_sst_params *)value; + BUG_ON(!str_param); + sst_print_params(&str_param->sparams); + retval = sst_get_stream(str_param, pcm_id); + if (retval >= 0) + sst_ops->stream_cnt++; +#ifdef CONFIG_SST_OSPM_SUPPORT + if (str_param->ops == STREAM_OPS_PLAYBACK) + sst_ospm_send_event(OSPM_EVENT_SUBSYS_START_PLAY); +#endif + + break; + } + + case SST_SND_PAUSE: + retval = sst_pause_stream(*(int *)value); + break; + + case SST_SND_RESUME: + retval = sst_resume_stream(*(int *)value); + break; + + case SST_SND_DROP: + retval = sst_drop_stream(*(int *)value); + break; + + case SST_SND_FREE: { + struct stream_info *stream = NULL; + + str_id = *(int *)value; + stream = &sst_ops->streams[str_id]; + retval = sst_free_stream(str_id); + if (retval == 0) { + stream->pcm_substream = NULL; + stream->period_elapsed = NULL; + sst_ops->stream_cnt--; + } + break; + } + + case SST_SND_STREAM_INIT: { + struct pcm_stream_info *str_info = NULL; + struct stream_info *stream = NULL; + + sst_dbg("stream init called\n"); + str_info = (struct pcm_stream_info *)value; + str_id = str_info->str_id; + retval = sst_validate_strid(str_id); + if (retval != 0) + break; + + stream = &sst_ops->streams[str_id]; + sst_dbg("setting the period ptrs\n"); + stream->pcm_substream = str_info->mad_substream; + stream->period_elapsed = str_info->period_elapsed; + break; + } + + case SST_SND_BUFFER_POINTER: { + struct pcm_stream_info *buf_ptr; + struct snd_sst_tstamp fw_tstamp = {0,}; + struct stream_info *stream = NULL; + + sst_dbg("buffer pointer query\n"); + + buf_ptr = (struct pcm_stream_info *)value; + str_id = buf_ptr->str_id; + retval = sst_validate_strid(str_id); + if (retval != 0) + break; + stream = &sst_ops->streams[str_id]; + + if (stream->pcm_substream == NULL) + break; + memcpy_fromio(&fw_tstamp, + ((void *)(sst_ops->mailbox + SST_TIME_STAMP) + +(str_id * sizeof(fw_tstamp))), + sizeof(fw_tstamp)); + + sst_dbg("strid = %d\n", str_id); + + if (stream->ops == STREAM_OPS_PLAYBACK) + buf_ptr->buffer_ptr = fw_tstamp.samples_rendered; + else + buf_ptr->buffer_ptr = fw_tstamp.samples_processed; + sst_dbg("samples played = %ld\n", + buf_ptr->buffer_ptr); + break; + } + + default: + /*Illegal case*/ + sst_err("illegal req\n"); + return -EINVAL; + } + sst_dbg("...complete ret code = %d\n", retval); + + return retval; +} + + +/** +* sst_send_data_to_HW - send data buffers +* @buffer_data: user buffer +* +* This function is called by MID sound card driver to send buffer +* to HW. This is registered with MID driver +*/ +int sst_send_buffer_to_HW(int str_id, struct stream_buffer *mad_buf) +{ + /*recvd a buffer map it to stream*/ + /*this is a PCM stream and playback*/ + int retval = 0; + bool flag_add = false; + struct sst_stream_bufs *sst_buf = NULL; + struct stream_info *stream = NULL; + + if (NULL == mad_buf || + 0 == mad_buf->addr || + 0 == mad_buf->length) { + sst_err("Null Ptr or buf size = 0\n"); + return -EINVAL; + } + + if (SST_FW_RUNNING != sst_ops->sst_state) { + sst_err("SST Not runng: %d\n", sst_ops->sst_state); + return -EBUSY; + } + + retval = sst_validate_strid(str_id); + if (retval < 0) + return -EINVAL; + + stream = &sst_ops->streams[str_id]; + sst_dbg("stream status = %d strid=%d\n", stream->status, str_id); + sst_dbg("stream codec = %d, prevstate=%d\n", + stream->codec, stream->prev); + if (stream->status == STREAM_UN_INIT) { + sst_err("BAD REQUEST!\n"); + return -EBADRQC; + } + sst_dbg("received addr=0x%x size = 0x%x\n", + (unsigned int)mad_buf->addr, mad_buf->length); + /*list is not empty*/ + list_for_each_entry(sst_buf, &stream->bufs, node) { + if (sst_buf->in_use == true) + continue; + else if ((int) mad_buf->addr != + (int)sst_buf->addr + sst_buf->size) + continue; + else { + sst_buf->size += mad_buf->length; + flag_add = true; + sst_dbg("inc addr = 0x%p, base = 0x%x inc_val = 0x%x\n", + sst_buf->addr, sst_buf->size, mad_buf->length); + break; + } + } + + if (flag_add == false) { + sst_buf = kzalloc(sizeof(*sst_buf), GFP_ATOMIC); + if (sst_buf == NULL) + return -ENOMEM; + sst_buf->size = mad_buf->length; + sst_buf->addr = (void *)mad_buf->addr; + sst_buf->offset = 0; + sst_buf->in_use = false; + list_add_tail(&sst_buf->node, &stream->bufs); + flag_add = true; + sst_dbg("entry added addr = 0x%x size = 0x%x\n", + (unsigned int)mad_buf->addr, mad_buf->length); + } + + if (stream->status == STREAM_INIT) { + sst_dbg("play/capt frames...\n"); + stream->prev = stream->status; + stream->status = STREAM_RUNNING; + stream->data_blk.on = false; + if (stream->ops == STREAM_OPS_PLAYBACK) + retval = sst_play_frame(str_id); + else if (stream->ops == STREAM_OPS_CAPTURE) + retval = sst_capture_frame(str_id); + else + return -EINVAL; + if (retval < 0) { + sst_err("play/capture frames failed \n"); + return -EIO; + } + } + + return retval; +} + +struct intel_sst_card_ops sst_pmic_ops = { + .control_set = sst_control_set, + .send_buffer = sst_send_buffer_to_HW, +}; + +/** +* register_sst_card- function for sound card to register +*@card: pointer to structure of operations +* This function is called card driver loads and is ready for registration +*/ +int register_sst_card(struct intel_sst_card_ops *card) +{ + + if (NULL == card || + NULL == card->module_name) { + sst_err(" Null Pointer Passed\n"); + return -EINVAL; + } + + if (PMIC_SND_UN_INIT == sst_ops->pmic_state) { + /*register this driver*/ + if (0 == (strncmp(SST_CARD_NAMES, card->module_name, + strlen(SST_CARD_NAMES)))) { + sst_ops->pmic_vendor = card->vendor_id; + sst_ops->scard_ops = *card->scard_ops; + sst_pmic_ops.module_name = card->module_name; + sst_ops->pmic_state = PMIC_SND_INIT_DONE; + card->control_set = sst_pmic_ops.control_set; + card->send_buffer = sst_pmic_ops.send_buffer; + return 0; + } else { + sst_err("strcmp failed %s \n", card->module_name); + return -EINVAL; + } + + } else { + /*already registered a driver*/ + sst_err("Repeat for register..denied\n"); + return -EBADRQC; + } + return 0; +} +EXPORT_SYMBOL_GPL(register_sst_card); + +/** +* unregister_sst_card- function for sound card to un-register +*@card: pointer to structure of operations +* This function is called when card driver unloads +*/ +void unregister_sst_card(struct intel_sst_card_ops *card) +{ + if (sst_pmic_ops.module_name == card->module_name) { + /*unreg*/ + sst_pmic_ops.module_name = ""; + sst_ops->pmic_state = PMIC_SND_UN_INIT; + sst_dbg("Unregistered %s\n", card->module_name); + } + return; +} +EXPORT_SYMBOL_GPL(unregister_sst_card); + +int lpe_mask_periphral_intr(enum lpe_periphral device) +{ + union sst_pimr_reg pimr = {{0},}; + if (sst_ops == NULL) + return -EIO; + + pimr.full = readl(sst_ops->shim + SST_PIMR); + + switch (device) { + case LPE_DMA: + pimr.part.dmac_sc = 1; + writel(pimr.full, sst_ops->shim + SST_PIMR); + break; + + case LPE_SSP0: + break; + + case LPE_SSP1: + break; + + default: + break; + } + return 0; +} +EXPORT_SYMBOL_GPL(lpe_mask_periphral_intr); + +int lpe_unmask_periphral_intr(enum lpe_periphral device) +{ + union sst_pimr_reg pimr = {{0},}; + if (sst_ops == NULL) + return -EIO; + + pimr.full = readl(sst_ops->shim + SST_PIMR); + + switch (device) { + case LPE_DMA: + pimr.part.dmac_sc = 0; + writel(pimr.full, sst_ops->shim + SST_PIMR); + break; + + case LPE_SSP0: + break; + + case LPE_SSP1: + break; + + default: + break; + } + return 0; + +} +EXPORT_SYMBOL_GPL(lpe_unmask_periphral_intr); + +int lpe_periphral_intr_status(enum lpe_periphral device, int *status) +{ + union sst_pisr_reg pisr = {{0},}; + if (sst_ops == NULL) + return -EIO; + + pisr.full = readl(sst_ops->shim + SST_PISR); + + switch (device) { + case LPE_DMA: + *status = pisr.part.dmac; + break; + + case LPE_SSP0: + break; + + case LPE_SSP1: + break; + + default: + break; + } + return 0; +} +EXPORT_SYMBOL_GPL(lpe_periphral_intr_status); -- 1.5.4.5

3 5

[alsa-devel] [PATCH] [RFC 0/13] Intel SST drivers
by Koul, Vinod 07 Jul '09

07 Jul '09

Following emails are a of series 13 patches for Intel SST drivers (including documentation). These drivers are for the audio engine/sound card in upcoming Intel based Mobile Internet Devices (MID) platform There are 2 drivers involved. One is for the audio DSP engine that can decode and do audio processing and one is for the ALSA sound card driver with support for three different sound cards that are supported on the platform. The dependant drivers for the platform are not yet available in the upstream. So please consider these patches only for RFC and provide your comments/suggestions. The audio firmware binary associated with this platform that is downloaded by the driver at runtime will be shared in a public location and made available in firmware git when we release the drivers for up streaming. The patches are compiled with 0 warnings and checkpatch-ed. The first patch in the series [RFC 1/13] Intel SST driver's documentation gives the documentation about the drivers and the interface for the audio DSP driver. The rest of the patches that follow have a brief description about the contents. Thanks -Vinod Koul & Harsha Priya

4 4

[alsa-devel] [PATCH] [RFC 2/13] Intel SST driver
by Vinod Koul 07 Jul '09

07 Jul '09

This adds the SST driver for the SST DSP engine. This patch adds the main file intel_sst.c which contains the init, exit probe, interrupt routine and PCI suspend/resume implementation. This file also implements the SST Firmware initialization and firmware and codec libraries download steps Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> new file: sound/pci/sst/intel_sst.c --- sound/pci/sst/intel_sst.c | 877 +++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 877 insertions(+), 0 deletions(-) create mode 100644 sound/pci/sst/intel_sst.c diff --git a/sound/pci/sst/intel_sst.c b/sound/pci/sst/intel_sst.c new file mode 100644 index 0000000..bd30833 --- /dev/null +++ b/sound/pci/sst/intel_sst.c @@ -0,0 +1,877 @@ +/* + * intel_sst.c - Intel SST Driver for audio engine + * + * Copyright (C) 2008-09 Intel Corp + * Authors: Vinod Koul <vinod.koul(a)intel.com> + * Harsha Priya <priya.harsha(a)intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * 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; version 2 of the License. + * + * 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. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This driver exposes the audio engine functionalities to the ALSA + * and middleware. + * + * This file contains all init functions + */ + +#include <linux/cdev.h> +#include <linux/pci.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/module.h> +#include <linux/syscalls.h> +#include <linux/fs.h> +#include <linux/file.h> +#include <linux/interrupt.h> +#include <linux/list.h> +#include <linux/workqueue.h> +#include <linux/firmware.h> +#include <linux/mutex.h> +#include <linux/delay.h> +#include <linux/sched.h> +#ifndef CONFIG_SST_IPC_NOT_INCLUDED +#include <asm/ipc_defs.h> +#endif +#include <sound/intel_sst.h> +#include <sound/intel_sst_ioctl.h> +#include "intel_sst_fw_ipc.h" +#include "intel_sst_common.h" +#include "intel_sst_pvt.h" +#ifdef CONFIG_SST_OSPM_SUPPORT +#include <linux/intel_mid.h> +#endif + + +MODULE_AUTHOR("Vinod Koul <vinod.koul(a)intel.com>"); +MODULE_DESCRIPTION("Intel (R) Moorestown Audio Engine Driver"); +MODULE_LICENSE("GPL v2"); +MODULE_VERSION(SST_DRIVER_VERSION); + +struct intel_sst_drv *sst_ops; + +void sst_reset(void) +{ + int retval; + + /*Disable Audio Core clock and Audio Fabric clock*/ + retval = sst_scu_ipc_write(0xff11d83c, 0x80008008); + if (retval != 0) + sst_err("scu ipc write1 failed %d", retval); + /*Reset the Audio subsystem*/ + retval = sst_scu_ipc_write(0xff11d118, 0x7ffffcff); + if (retval != 0) + sst_err("scu ipc write2 failed %d", retval); + /*Bring it out of Audio subsystem reset*/ + retval = sst_scu_ipc_write(0xff11d118, 0x7fffffff); + if (retval != 0) + sst_err("scu ipc write3 failed %d", retval); + /*Enable fabric clock at 50MHz*/ + retval = sst_scu_ipc_write(0xff11d83c, 0x80008301); + if (retval != 0) + sst_err("scu ipc write4 failed %d", retval); + /*Write to the Shim register for ratio 1:1*/ + retval = sst_scu_ipc_write(0xffae8000, 0x382); + if (retval != 0) + sst_err("scu ipc write5 failed %d", retval); + /*Enable the core clock at 1:2*/ + retval = sst_scu_ipc_write(0xff11d83c, 0x80000301); + if (retval != 0) + sst_err("scu ipc write6 failed %d", retval); + return; +} + +void sst_start(void) +{ + union config_status_reg csr; + int retval; + + csr.full = readl(sst_ops->shim + SST_CSR); + csr.part.sst_reset = 0; + csr.part.run_stall = 0; + csr.part.bypass = 0; + /*csr.full = 0x1800;*/ + sst_dbg("Setting SST to execute 0x%x \n", csr.full); + + retval = sst_scu_ipc_write(sst_ops->shim_phy_add + SST_CSR, csr.full); + if (retval != 0) + sst_err("scu ipc write start failed %d", retval); + return; +} +int sst_parse_module(struct fw_module_header *module) +{ + struct dma_block_info *block = NULL; + u32 count; + void __iomem *ram; + + sst_dbg("module sign=%s sz=0x%x blks=0x%x typ=0x%x ep=0x%x sz=0x%x\n", + module->signature, module->mod_size, + module->blocks, module->type, + module->entry_point, sizeof(*module)); + + block = (void *)module + sizeof(*module); + + for (count = 0; count < module->blocks; count++) { + if (0 >= block->size) { + sst_err("block size invalid\n"); + return -EINVAL; + } + switch (block->type) { + case SST_IRAM: + ram = sst_ops->iram; + break; + case SST_DRAM: + ram = sst_ops->dram; + break; + default: + sst_err("wrong ram type 0x%x in block 0x%x-abort\n", + block->type, count); + return -EINVAL; + } + memcpy_toio(ram + block->ram_offset, + (void *)block + sizeof(*block), block->size); + block = (void *)block + sizeof(*block) + block->size; + } + return 0; +} + +/** +* sst_parse_fw_image - FW parse and load +* @fw: Pointer to loaded FW +* +* This function is called to parse and download the FW image +*/ +int sst_parse_fw_image(const struct firmware *sst_fw) +{ + struct fw_header *header = NULL; + u32 count; + int ret_val = 0; + struct fw_module_header *module = NULL; + + BUG_ON(!sst_fw); + + /*Read the header information from the data pointer*/ + header = (struct fw_header *)sst_fw->data; + + /*verify FW*/ + if ((0 != strncmp(header->signature, SST_FW_SIGN, 4)) || + (sst_fw->size != header->file_size + sizeof(*header))) { + /*Invalid FW signature*/ + sst_err("Invalid FW sign/file size mismatch\n"); + return -EINVAL; + } + sst_dbg("header sign=%s size=0x%x modules=0x%x fmt=0x%x size=0x%x\n", + header->signature, header->file_size, header->modules, + header->file_format, sizeof(*header)); + module = (void *)sst_fw->data + sizeof(*header); + for (count = 0; count < header->modules; count++) { + /*module*/ + ret_val = sst_parse_module(module); + if (0 != ret_val) + return ret_val; + module = (void *)module + sizeof(*module) + module->mod_size ; + } + + sst_dbg("done....\n"); + return 0; +} + +/** +* sst_load_fw - function to reset FW +* @fw: Pointer to loaded FW +* This function is called when the FW is loaded +*/ +void sst_load_fw(const struct firmware *fw, void *context) +{ + int ret; + + sst_dbg("called \n"); + BUG_ON(!fw); + + sst_reset(); + /*putting the sst state to init*/ + sst_ops->sst_state = SST_UN_INIT; + + ret = sst_parse_fw_image(fw); + if (0 != ret) { + sst_err("error in FW dma\n"); + return; + } + sst_ops->sst_state = SST_FW_LOADED; + /* 7. ask scu to reset the bypass bits*/ + /* 8.bring sst out of reset */ + sst_start(); + sst_dbg("...successful!!!\n"); + return; +} + +int sst_download_library(const struct firmware *fw_lib, + struct snd_sst_lib_download_info *lib) +{ + /*send IPC message and wait*/ + unsigned int i, pvt_id; + struct ipc_post *msg = NULL; + union config_status_reg csr; + struct snd_sst_str_type str_type = {0}; + int retval = 0; + + if (sst_create_large_msg(&msg) != 0) + return -ENOMEM; + + pvt_id = sst_assign_pvt_id(sst_ops); + i = sst_get_block_stream(sst_ops); + sst_dbg("alloc block allocated = %d, pvt_id %d\n", i, pvt_id); + if (i < 0) { + kfree(msg); + return -ENOMEM; + } + sst_ops->alloc_block[i].sst_id = pvt_id; + sst_fill_header(&msg->header, IPC_IA_PREP_LIB_DNLD, 1, 0); + msg->header.part.data = sizeof(u32) + sizeof(str_type); + str_type.codec_type = lib->dload_lib.lib_info.lib_type; + str_type.pvt_id = pvt_id; + memcpy(msg->mailbox_data, &msg->header, sizeof(u32)); + memcpy(msg->mailbox_data + sizeof(u32), &str_type, sizeof(str_type)); + list_add_tail(&msg->node, &sst_ops->ipc_dispatch_list); + sst_post_message(&sst_ops->ipc_post_msg_wq); + retval = sst_wait_timeout(sst_ops, &sst_ops->alloc_block[i]); + if (retval != 0) { + /*error*/ + sst_ops->alloc_block[i].sst_id = BLOCK_UNINIT; + sst_err("Prep codec downloaded failed %d\n", retval); + return -EIO; + } + sst_dbg("FW responded, ready for download now...\n"); + /*downloading on success*/ + sst_ops->sst_state = SST_FW_LOADED; + csr.full = readl(sst_ops->shim + SST_CSR); + sst_dbg("CSR reg 0x%x \n", csr.full); + csr.part.run_stall = 1; + writel(csr.full, sst_ops->shim + SST_ISRX); + sst_dbg("HALT CSR reg setting to 0x%x \n", csr.full); + retval = sst_scu_ipc_write(sst_ops->shim_phy_add + SST_CSR, csr.full); + if (retval != 0) { + /*error*/ + sst_ops->alloc_block[i].sst_id = BLOCK_UNINIT; + sst_err("IPC failed to Halt SST 0x%x\n", retval); + return -EAGAIN; + } + csr.full = readl(sst_ops->shim + SST_CSR); + csr.part.bypass = 0x7; + writel(csr.full, sst_ops->shim + SST_ISRX); + sst_dbg("Bypass CSR reg setting to 0x%x \n", csr.full); + retval = sst_scu_ipc_write(sst_ops->shim_phy_add + SST_CSR, csr.full); + if (retval != 0) { + /*error*/ + sst_ops->alloc_block[i].sst_id = BLOCK_UNINIT; + sst_err("IPC failed to Bypass SST 0x%x\n", retval); + csr.part.bypass = 0x0; + /*bring LPE out of run stall*/ + csr.part.run_stall = 0x0; + writel(csr.full, sst_ops->shim + SST_ISRX); + sst_dbg("Bypass CSR reg setting to 0x%x \n", csr.full); + retval = sst_scu_ipc_write(sst_ops->shim_phy_add + SST_CSR, + csr.full); + if (retval != 0) + sst_ops->sst_state = SST_UN_INIT; + return -EAGAIN; + } + sst_parse_fw_image(fw_lib); + + /*set the FW to running again*/ + csr.full = readl(sst_ops->shim + SST_CSR); + csr.part.bypass = 0x0; + writel(csr.full, sst_ops->shim + SST_ISRX); + sst_dbg("Bypass CSR reg clearing to 0x%x \n", csr.full); + retval = sst_scu_ipc_write(sst_ops->shim_phy_add + SST_CSR, csr.full); + if (retval != 0) { + sst_ops->alloc_block[i].sst_id = BLOCK_UNINIT; + sst_err("Bypass CSR reg clear failed 0x%x \n", retval); + /*bring LPE out of run stall*/ + csr.part.run_stall = 0x0; + writel(csr.full, sst_ops->shim + SST_ISRX); + sst_dbg("Bypass CSR reg setting to 0x%x \n", csr.full); + retval = sst_scu_ipc_write(sst_ops->shim_phy_add + SST_CSR, + csr.full); + if (retval != 0) + sst_ops->sst_state = SST_UN_INIT; + return -EAGAIN; + } + + csr.full = readl(sst_ops->shim + SST_CSR); + csr.part.run_stall = 0; + writel(csr.full, sst_ops->shim + SST_ISRX); + sst_dbg("Stalll CSR reg clearing to 0x%x \n", csr.full); + retval = sst_scu_ipc_write(sst_ops->shim_phy_add + SST_CSR, csr.full); + if (retval != 0) { + sst_ops->alloc_block[i].sst_id = BLOCK_UNINIT; + sst_err("Stall CSR reg clear failed 0x%x \n", retval); + if (retval != 0) + sst_ops->sst_state = SST_UN_INIT; + return -EAGAIN; + } + /*send download complete and wait*/ + if (sst_create_large_msg(&msg) != 0) { + sst_ops->alloc_block[i].sst_id = BLOCK_UNINIT; + return -ENOMEM; + } + + sst_fill_header(&msg->header, IPC_IA_LIB_DNLD_CMPLT, 1, 0); + msg->header.part.data = sizeof(u32) + sizeof(*lib); + lib->pvt_id = pvt_id; + memcpy(msg->mailbox_data, &msg->header, sizeof(u32)); + memcpy(msg->mailbox_data + sizeof(u32), lib, sizeof(*lib)); + + list_add_tail(&msg->node, &sst_ops->ipc_dispatch_list); + sst_post_message(&sst_ops->ipc_post_msg_wq); + sst_dbg("Waiting for FW to respond on Download complete \n"); + sst_ops->alloc_block[i].ops_block.condition = false; + retval = sst_wait_timeout(sst_ops, &sst_ops->alloc_block[i]); + if (retval != 0) { + /*error*/ + sst_ops->sst_state = SST_FW_RUNNING; + sst_ops->alloc_block[i].sst_id = BLOCK_UNINIT; + return -EIO; + } + + sst_dbg("FW responded sucess on Download complete \n"); + sst_ops->alloc_block[i].sst_id = BLOCK_UNINIT; + sst_ops->sst_state = SST_FW_RUNNING; + return 0; + +} + +int sst_validate_library(const struct firmware *fw_lib, + struct lib_slot_info *slot, + u32 *entry_point) +{ + struct fw_header *header = NULL; + struct fw_module_header *module = NULL; + struct dma_block_info *block = NULL; + unsigned int n_blk, isize = 0, dsize = 0; + int err = 0; + + header = (struct fw_header *)fw_lib->data; + if (header->modules != 1) { + sst_err("Module no mismatch found\n"); + err = -EINVAL; + goto exit; + } + module = (void *)fw_lib->data + sizeof(*header); + *entry_point = module->entry_point; + sst_dbg("Module entry point 0x%x \n", *entry_point); + sst_dbg("Module Sign %s, Size 0x%x, Blocks 0x%x Type 0x%x \n", + module->signature, module->mod_size, + module->blocks, module->type); + + block = (void *)module + sizeof(*module); + for (n_blk = 0; n_blk < module->blocks; n_blk++) { + switch (block->type) { + case SST_IRAM: + isize += block->size; + break; + case SST_DRAM: + dsize += block->size; + break; + default: + sst_err("Invalid block type found for 0x%x\n", n_blk); + err = -EINVAL; + goto exit; + } + block = (void *)block + sizeof(*block) + block->size; + } + if (isize > slot->iram_size || dsize > slot->dram_size) { + sst_err("libarary exceeds size allocated \n"); + err = -EINVAL; + goto exit; + } else + sst_dbg("Library is safe for download...\n"); + + sst_dbg("iram 0x%x, dram 0x%x, allowed iram 0x%x, allowed dram 0x%x\n", + isize, dsize, slot->iram_size, slot->dram_size); +exit: + return err; + +} + +int sst_load_library(struct snd_sst_lib_download *lib, u8 ops, u32 pvt_id) +{ + char buf[20]; + int len = 0, error = 0; + u32 entry_point; + const struct firmware *fw_lib; + struct snd_sst_lib_download_info dload_info = {{{0},},}; + + memset(buf, 0, sizeof(buf)); + + sst_dbg("Lib Type 0x%x, Slot 0x%x, ops 0x%x \n", + lib->lib_info.lib_type, lib->slot_info.slot_num, ops); + sst_dbg("Version 0x%x, name %s, caps 0x%x media type 0x%x \n", + lib->lib_info.lib_version, lib->lib_info.lib_name, + lib->lib_info.lib_caps, lib->lib_info.media_type); + + sst_dbg("IRAM Size 0x%x, offset 0x%x, DRAM Size 0x%x, offset 0x%x \n", + lib->slot_info.iram_size, lib->slot_info.iram_offset, + lib->slot_info.dram_size, lib->slot_info.dram_offset); + + switch (lib->lib_info.lib_type) { + case SST_CODEC_TYPE_MP3: + len += snprintf(buf + len, sizeof(buf) - len, "mp3_"); + break; + case SST_CODEC_TYPE_AAC: + len += snprintf(buf + len, sizeof(buf) - len, "aac_"); + break; + case SST_CODEC_TYPE_WMA9: + len += snprintf(buf + len, sizeof(buf) - len, "wma9_"); + break; + default: + sst_err("Invalid codec type \n"); + error = -EINVAL; + goto wake; + } + + if (ops == STREAM_OPS_CAPTURE) + len += snprintf(buf + len, sizeof(buf) - len, "enc_"); + else + len += snprintf(buf + len, sizeof(buf) - len, "dec_"); + + len += snprintf(buf + len, sizeof(buf) - len, "%d", + lib->slot_info.slot_num); + len += snprintf(buf + len, sizeof(buf) - len, ".bin"); + + sst_dbg("Requesting %s \n", buf); + + error = request_firmware(&fw_lib, buf, &sst_ops->pci->dev); + if (0 != error) { + sst_err("library load failed %d \n", error); + goto wake; + } + error = sst_validate_library(fw_lib, &lib->slot_info, &entry_point); + if (0 != error) { + sst_err("validate lib failed %d \n", error); + goto wake_free; + } + lib->mod_entry_pt = entry_point; + memcpy(&dload_info.dload_lib, lib, sizeof(*lib)); + error = sst_download_library(fw_lib, &dload_info); + if (0 != error) { + sst_err("download lib failed %d \n", error); + goto wake_free; + } + /*lib is downloaded and init send alloc again*/ + sst_dbg("Library is downloaded now... \n"); +wake_free: + /*sst_wake_up_alloc_block(sst_ops, pvt_id, error, NULL);*/ + release_firmware(fw_lib); +wake: + return error; +} + +/*fops Routines*/ +const struct file_operations intel_sst_fops = { + .owner = THIS_MODULE, + .open = intel_sst_open, + .release = intel_sst_release, + .read = intel_sst_read, + .write = intel_sst_write, + .ioctl = intel_sst_ioctl, + .mmap = intel_sst_mmap, + .aio_read = intel_sst_aio_read, + .aio_write = intel_sst_aio_write, +}; + +/* + ISR Routines +*/ +/** +* intel_sst_interrupt - Interrupt service routine for SST +* @irq: irq number of interrupt +* @dev_id: pointer to device structre +* +* This function is called by OS when SST device raises +* an interrupt. This will be result of write in IPC register +* Source can be busy or done interrupt +*/ +static irqreturn_t intel_sst_interrupt(int irq, void *context) +{ + union interrupt_reg isr; + union ipc_header header; + union interrupt_reg imr; + struct intel_sst_drv *drv = (struct intel_sst_drv *) context; + unsigned int size = 0; + + /*Interrupt arrived, check src*/ + isr.full = readl(drv->shim + SST_ISRX); + imr.full = readl(drv->shim + SST_IMRX); + if (1 == isr.part.busy_interrupt) { + header.full = readl(drv->shim + SST_IPCD); + if (1 == header.part.large) + size = header.part.data; + if (0 != (header.part.msg_id & REPLY_MSG)) { + sst_ops->ipc_process_msg.header = header; + memcpy_fromio(sst_ops->ipc_process_msg.mailbox, + drv->mailbox + SST_MAILBOX_RCV, size); + schedule_work(&sst_ops->ipc_process_msg.wq); + } else { + sst_ops->ipc_process_reply.header = header; + memcpy_fromio(sst_ops->ipc_process_reply.mailbox, + drv->mailbox + SST_MAILBOX_RCV, size); + schedule_work(&sst_ops->ipc_process_reply.wq); + } + /*mask busy inetrrupt*/ + imr.part.busy_interrupt = 1; + writel(imr.full, drv->shim + SST_IMRX); + + } else if (1 == isr.part.done_interrupt) { + /*Clear done bit*/ + header.full = readl(drv->shim + SST_IPCX); + header.part.done = 0; + writel(header.full, drv->shim + SST_IPCX); + /* write 1 to clear status register*/; + isr.part.done_interrupt = 1; + writel(isr.full, drv->shim + SST_ISRX); + schedule_work(&sst_ops->ipc_post_msg.wq); + } + return IRQ_HANDLED; +} + + +/* + PCI Routines +*/ +static struct pci_device_id intel_sst_ids[] = { + { 0x8086, 0x080A, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0}, + { 0, } +}; +MODULE_DEVICE_TABLE(pci, intel_sst_ids); + + +/* +* intel_sst_probe - PCI probe function +* @pci: PCI device structure +* @pci_id: PCI device ID structure +* +* This function is called by OS when a device is found +* This enables the device, interrupt etc +*/ +static int __devinit intel_sst_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) +{ + int ret = 0; + u32 bar = 0, size = 0; + + /*Init the device*/ + ret = pci_enable_device(pci); + if (0 != ret) { + sst_err("device cant be enabled\n"); + return ret; + } + sst_ops->pci = pci; + /*map registers*/ + /*SST Shim*/ + bar = pci_resource_start(pci, 1); + size = pci_resource_len(pci, 1); + ret = pci_request_region(pci, 2, SST_DRV_NAME); + if (ret != 0) + goto err_1; + sst_ops->shim_phy_add = bar; + sst_ops->shim = ioremap_nocache(bar, size); + if (sst_ops->shim == NULL) + goto err_2; + sst_dbg("SST Shim 0x%x,Size 0x%x,Ptr %p \n", bar, size, sst_ops->shim); + + /*Shared SRAM*/ + bar = pci_resource_start(pci, 2); + size = pci_resource_len(pci, 2); + ret = pci_request_region(pci, 3, SST_DRV_NAME); + if (ret != 0) + goto err_2; + sst_ops->mailbox = ioremap_nocache(bar, size); + if (NULL == sst_ops->mailbox) + goto err_3; + sst_dbg("SRAM 0x%x, Size 0x%x, Ptr %p \n", bar, size, sst_ops->mailbox); + + /* IRAM*/ + bar = pci_resource_start(pci, 3); + size = pci_resource_len(pci, 3); + ret = pci_request_region(pci, 4, SST_DRV_NAME); + if (ret != 0) + goto err_3; + sst_ops->iram = ioremap_nocache(bar, size); + if (sst_ops->iram == NULL) + goto err_4; + sst_dbg("IRAM 0x%x, Size 0x%x, Ptr %p \n", bar, size, sst_ops->iram); + + /*DRAM*/ + bar = pci_resource_start(pci, 4); + size = pci_resource_len(pci, 4); + ret = pci_request_region(pci, 5, SST_DRV_NAME); + if (ret != 0) + goto err_4; + sst_ops->dram = ioremap_nocache(bar, size); + if (sst_ops->dram == NULL) + goto err_5; + sst_dbg("DRAM 0x%x, Size 0x%x, Ptr %p \n", bar, size, sst_ops->dram); + + /*Register the ISR*/ + ret = request_irq(pci->irq, intel_sst_interrupt, + IRQF_SHARED, SST_DRV_NAME, sst_ops); + if (0 != ret) + goto err_5; + sst_dbg("Registered IRQ 0x%x\n", pci->irq); + sst_ops->sst_state = SST_UN_INIT; + + /* Register with /dev */ + ret = register_chrdev(INTEL_SST_MAJOR, SST_DRV_NAME, &intel_sst_fops); + if (0 != ret) { + sst_err("couldn't register device number\n"); + goto err_6; + } + + sst_dbg("...successfully done!!!\n"); + return 0; + +err_6: free_irq(pci->irq, sst_ops); +err_5: iounmap(sst_ops->dram); +err_4: iounmap(sst_ops->iram); +err_3: iounmap(sst_ops->mailbox); +err_2: iounmap(sst_ops->shim); +err_1: sst_err("Probe failed with 0x%x \n", ret); + return ret; +} + +/** +* intel_sst_remove - PCI remove function +* @pci: PCI device structure +* +* This function is called by OS when a device is unloaded +* This frees the interrupt etc +*/ +static void __devexit intel_sst_remove(struct pci_dev *pci) +{ + sst_ops->sst_state = SST_UN_INIT; + unregister_chrdev(INTEL_SST_MAJOR, SST_DRV_NAME); + free_irq(pci->irq, sst_ops); + iounmap(sst_ops->dram); + iounmap(sst_ops->iram); + iounmap(sst_ops->mailbox); + iounmap(sst_ops->shim); +} + +#ifdef CONFIG_SST_OSPM_SUPPORT +/* + Power Management + +*/ +/** +* intel_sst_suspend - PCI suspend function +* @pci: PCI device structure +* @state: PM message +* +* This function is called by OS when a power event occurs +*/ +static int intel_sst_suspend(struct pci_dev *pci, pm_message_t state) +{ + int i = 0; + sst_dbg("intel_sst_suspend called\n"); + + /*Pause all running streams*/ + for (i = 1; i < MAX_NUM_STREAMS; i++) { + if (sst_ops->streams[i].status == STREAM_RUNNING) { + sst_ops->active_streams[i] = true; + sst_pause_stream(i); + } else + sst_ops->active_streams[i] = false; + } + + pci_set_drvdata(pci, sst_ops); + + /*Disable everything*/ + /*free_irq(pci->irq, sst_ops);*/ + pci_disable_device(pci); + pci_save_state(pci); + pci_set_power_state(pci, pci_choose_state(pci, state)); + return 0; +} + +/** +* intel_sst_resume - PCI resume function +* @pci: PCI device structure +* @state: PM message +* +* This function is called by OS when a power event occurs +*/ +static int intel_sst_resume(struct pci_dev *pci) +{ + int i = 0; + + sst_dbg("\nintel_sst_resume called\n"); + if (pci != NULL) + sst_dbg("pci is not null \n"); + /*Enable*/ + sst_ops = pci_get_drvdata(pci); + if (sst_ops != NULL) + sst_dbg("sst_ops is not null\n"); + if (pci->irq) + sst_dbg("irq %d \n", pci->irq); + + pci_set_power_state(pci, PCI_D0); + pci_restore_state(pci); + + /*ret = request_irq(pci->irq, intel_sst_interrupt, + IRQF_SHARED, "intel_sst_engine", sst_ops); + if (0 != ret) { + sst_err("\tIRQ %d reg error %d\n", pci->irq, ret); + return ret; + }*/ + + /*Start all paused streams*/ + for (i = 1; i < MAX_NUM_STREAMS; i++) { + if (sst_ops->active_streams[i] == true) + sst_resume_stream(i); + } + return 0; +} +/** +* sst_ospm_send_event - Send PM events to OSPM +* @event: OSPM Event +* +* This function is called when stream is started/stopped/paused... +*/ +int sst_ospm_send_event(int event) +{ + return ospm_generate_netlink_event(AUDIO_SUBSYTEM_ID, event); + +} +#endif /* CONFIG_SST_OSPM_SUPPORT */ +static struct pci_driver driver = { + .name = SST_DRV_NAME, + .id_table = intel_sst_ids, + .probe = intel_sst_probe, + .remove = __devexit_p(intel_sst_remove), +#ifdef CONFIG_SST_OSPM_SUPPORT + .suspend = intel_sst_suspend, + .resume = intel_sst_resume, +#endif +}; + +/** +* intel_sst_init - Module init function +* +* Registers with PCI +* Registers with /dev +*Init all data strutures +*/ +/*Module init function*/ +static int __init intel_sst_init(void) +{ + /* + Init all variables, data structure etc.... + */ + int ret = 0, i = 0; + struct stream_info *stream = NULL; + + sst_info("******** SST DRIVER loading.. Ver: %s\n", SST_DRIVER_VERSION); + + sst_ops = kzalloc(sizeof(*sst_ops), GFP_KERNEL); + if (NULL == sst_ops) { + sst_err("intel_sst malloc fail\n"); + return -ENOMEM; + } + + sst_ops->pmic_state = PMIC_SND_UN_INIT; + sst_ops->stream_cnt = 0; + sst_ops->active_cnt = 0; + sst_ops->am_cnt = 0; + sst_ops->unique_id = 0; + + INIT_LIST_HEAD(&sst_ops->ipc_dispatch_list); + INIT_WORK(&sst_ops->ipc_post_msg.wq, sst_post_message); + INIT_WORK(&sst_ops->ipc_process_msg.wq, sst_process_message); + INIT_WORK(&sst_ops->ipc_process_reply.wq, sst_process_reply); + init_waitqueue_head(&sst_ops->wait_queue); + + for (i = 0; i < MAX_ACTIVE_STREAM; i++) { + sst_ops->alloc_block[i].sst_id = BLOCK_UNINIT; + sst_ops->alloc_block[i].ops_block.condition = false; + } + mutex_init(&sst_ops->list_lock); + + for (i = 1; i < MAX_NUM_STREAMS; i++) { + stream = &sst_ops->streams[i]; + INIT_LIST_HEAD(&stream->bufs); + mutex_init(&stream->lock); + } + sst_ops->mmap_mem = NULL; + sst_ops->mmap_len = SST_MMAP_PAGES * PAGE_SIZE; + while (sst_ops->mmap_len > 0) { + sst_ops->mmap_mem = kzalloc(sst_ops->mmap_len, GFP_KERNEL); + if (sst_ops->mmap_mem != NULL) { + sst_dbg("Got memory %p size 0x%x \n", sst_ops->mmap_mem, + sst_ops->mmap_len); + break; + } + sst_ops->mmap_len -= (SST_MMAP_STEP * PAGE_SIZE); + if (sst_ops->mmap_len <= 0) { + sst_err("Couldnt get any mem...abort!!\n"); + goto exit_p; + } + sst_dbg("Failed...trying %d\n", sst_ops->mmap_len); + } + /* Register with PCI */ + ret = pci_register_driver(&driver); + if (0 != ret) { + sst_err("PCI register failed, unregister char drv\n"); + goto exit; + } + + + + sst_dbg("...sucessful\n"); + return 0; + +exit: + pci_unregister_driver(&driver); +exit_p: + kfree(sst_ops->mmap_mem); + kfree(sst_ops); + sst_err("Error in init %d\n", ret); + + return ret; +} + +/*Module exit function*/ +static void __exit intel_sst_exit(void) +{ + int i; + struct stream_info *stream = NULL; + + for (i = 0; i < MAX_NUM_STREAMS; i++) { + stream = &sst_ops->streams[i]; + sst_free_stream(i); + } + sst_ops->pmic_state = PMIC_SND_UN_INIT; + pci_unregister_driver(&driver); + + flush_scheduled_work(); + kfree(sst_ops->mmap_mem); + kfree(sst_ops); + + sst_dbg("...unloaded\n"); + return; +} + +module_init(intel_sst_init); +module_exit(intel_sst_exit); -- 1.5.4.5

3 4

[alsa-devel] [PATCH] [RFC 4/13] Intel SST driver common private headers
by Vinod Koul 07 Jul '09

07 Jul '09

This patch adds header files private to the SST driver and contains the common structures like driver context, DSP register definitions, and inline utility functions used by the SST driver and function prototypes of common functions that are implemented in SST driver Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> new file: sound/pci/sst/intel_sst_common.h new file: sound/pci/sst/intel_sst_pvt.h --- sound/pci/sst/intel_sst_common.h | 292 +++++++++++++++++++++++++++++ sound/pci/sst/intel_sst_pvt.h | 378 ++++++++++++++++++++++++++++++++++++++ 2 files changed, 670 insertions(+), 0 deletions(-) create mode 100644 sound/pci/sst/intel_sst_common.h create mode 100644 sound/pci/sst/intel_sst_pvt.h diff --git a/sound/pci/sst/intel_sst_common.h b/sound/pci/sst/intel_sst_common.h new file mode 100644 index 0000000..e039489 --- /dev/null +++ b/sound/pci/sst/intel_sst_common.h @@ -0,0 +1,292 @@ +#ifndef __INTEL_SST_COMMON_H__ +#define __INTEL_SST_COMMON_H__ +/* + * intel_sst_common.h - Intel SST Driver for audio engine + * + * Copyright (C) 2008-09 Intel Corporation + * Authors: Vinod Koul <vinod.koul(a)intel.com> + * Harsha Priya <priya.harsha(a)intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * 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; version 2 of the License. + * + * 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. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * Common private declarations for SST + */ +#include <linux/time.h> + +/*#define SND_LOOP_TEST*/ + +#define SST_DRIVER_VERSION "0.04.002" +#define SST_VERSION_NUM 0x0402 + +/* +driver names +*/ +#define SST_DRV_NAME "intel_sst_driver" +#define SST_FW_STD_FILENAME "fw_sst.bin" + + +/* +PMIC and SST hardware states +*/ +enum sst_pmic_states { + PMIC_SND_UN_INIT = 0, + PMIC_SND_INIT_DONE, +}; + +enum sst_states { + SST_FW_LOADED = 1, + SST_FW_RUNNING, + SST_UN_INIT, + SST_ERROR, +}; + +#define MAX_ACTIVE_STREAM 3 +#define MAX_ENC_STREAM 1 +#define MAX_AM_HANDLES 1 +#define ALLOC_TIMEOUT 5000 +/* +SST numbers +*/ +#define SST_BLOCK_TIMEOUT 5000 + +#define INTEL_SST_MAJOR 255 +#define BLOCK_UNINIT -1 + +/*SST register map*/ +#define SST_CSR 0x00 +#define SST_PISR 0x08 +#define SST_PIMR 0x10 +#define SST_ISRX 0x18 +#define SST_IMRX 0x28 +#define SST_IPCX 0x38 /*IPC IA-SST*/ +#define SST_IPCD 0x40 /*IPC SST-IA*/ +#define SST_SHIM_SIZE 0X44 + +#define FW_SIGNATURE_SIZE 4 + +/* +stream states +*/ +enum sst_stream_states { + STREAM_UN_INIT = 0, /*Freed/Not used stream*/ + STREAM_RUNNING = 1, /*Running*/ + STREAM_PAUSED = 2, /*Paused stream*/ + STREAM_INIT_WT = 4, /*stream is init and waiting for FW*/ + STREAM_INIT = 5, /*stream init, waiting for data*/ +}; + + +enum sst_ram_type{ + SST_IRAM = 1, + SST_DRAM = 2, +}; +/* +SST shim registers to structure mapping +*/ +union config_status_reg { + struct { + u32 rsvd0:1; + u32 sst_reset:1; + u32 hw_rsvd:3; + u32 sst_clk:2; + u32 bypass:3; + u32 run_stall:1; + u32 rsvd:21; + } part; + u32 full; +}; + +union interrupt_reg { + struct { + u32 done_interrupt:1; + u32 busy_interrupt:1; + u32 rsvd:30; + } part; + u32 full; +}; + +union sst_pisr_reg { + struct { + u32 pssp0:1; + u32 pssp1:1; + u32 rsvd0:3; + u32 dmac:1; + u32 rsvd1:26; + } part; + u32 full; +}; + +union sst_pimr_reg { + struct { + u32 ssp0:1; + u32 ssp1:1; + u32 rsvd0:3; + u32 dmac:1; + u32 rsvd1:10; + u32 ssp0_sc:1; + u32 ssp1_sc:1; + u32 rsvd2:3; + u32 dmac_sc:1; + u32 rsvd3:10; + } part; + u32 full; +}; + + +struct sst_stream_bufs { + struct list_head node; + u32 size; + const char *addr; + u32 data_copied; + bool in_use; + u32 offset; +}; + +struct snd_sst_user_cap_list { + unsigned int iov_index; /*index of iov*/ + unsigned long iov_offset; /*offset in iov*/ + unsigned long offset; /*offset in kmem*/ + unsigned long size; /*size copied*/ + struct list_head node; +}; +/* +This structure is used to block a user/fw data call to another +fw/user call +*/ +struct sst_block { + bool condition; /*condition for blocking check*/ + int ret_code; /*ret code when block is released*/ + void *data; /*data to be appsed for block if any*/ + bool on; +}; + +enum snd_sst_buf_type { + SST_BUF_USER_STATIC = 1, + SST_BUF_USER_DYNAMIC, + SST_BUF_MMAP_STATIC, + SST_BUF_MMAP_DYNAMIC, +}; +/* +structure that holds the stream information +*/ +struct stream_info { + unsigned int status; + unsigned int prev; + unsigned int codec; + unsigned int sst_id; + unsigned int ops; + struct list_head bufs; + struct mutex lock; /*mutex*/ + void *get_parameters; + bool mmapped; + unsigned int sg_index; /* current buf Index */ + unsigned char *cur_ptr; /* Current static bufs */ + struct snd_sst_buf_entry *buf_entry; + struct sst_block data_blk; /*stream ops block*/ + struct sst_block ctrl_blk; /*stream control cmd block*/ + enum snd_sst_buf_type buf_type; + void *pcm_substream; + void (*period_elapsed) (void *pcm_substream); +}; + + + +/* +this structure is used for blocking the user's alloc calls to +fw's response to alloc calls +*/ +struct stream_alloc_block { + int sst_id; /*session id of blocked stream*/ + struct sst_block ops_block; /*ops block struture*/ +}; + +#define SST_FW_SIGN "$SST" +#define SST_FW_LIB_SIGN "$LIB" + +/*FW file headers*/ +struct fw_header { + unsigned char signature[FW_SIGNATURE_SIZE]; /*FW signature*/ + u32 file_size; /*size of fw minus this header*/ + u32 modules; /* # of modules*/ + u32 file_format; /*version of header format*/ + u32 reserved[4]; +}; + +struct fw_module_header { + unsigned char signature[FW_SIGNATURE_SIZE]; /*module signature*/ + u32 mod_size; /*size of module*/ + u32 blocks; /*# of blocks*/ + u32 type; /*codec type, pp lib*/ + u32 entry_point; +}; + +struct dma_block_info { + enum sst_ram_type type; /*IRAM/DRAM*/ + u32 size; /*Bytes*/ + u32 ram_offset; /*Offset in I/DRAM*/ + u32 rsvd; /*Reserved field*/ +}; + +struct ioctl_pvt_data { + int str_id; + int pvt_id; +}; + +struct sst_ipc_msg_wq { + union ipc_header header; + char mailbox[SST_MAILBOX_SIZE]; + struct work_struct wq; +}; +#define SST_MMAP_PAGES (640*1024 / PAGE_SIZE) +#define SST_MMAP_STEP (40*1024 / PAGE_SIZE) + +/*driver ops*/ +struct intel_sst_drv{ + bool pmic_state; + int pmic_vendor; + int sst_state; + void __iomem *shim; + void __iomem *mailbox; + void __iomem *iram; + void __iomem *dram; + unsigned int shim_phy_add; + struct list_head ipc_dispatch_list; + struct work_struct ipc_post_msg_wq; + struct sst_ipc_msg_wq ipc_process_msg; + struct sst_ipc_msg_wq ipc_process_reply; + struct sst_ipc_msg_wq ipc_post_msg; + wait_queue_head_t wait_queue; + + struct stream_info streams[MAX_NUM_STREAMS]; + struct stream_alloc_block alloc_block[MAX_ACTIVE_STREAM]; + + struct mutex list_lock; /*mutex for IPC list locking*/ + struct snd_pmic_ops scard_ops; + struct pci_dev *pci; + int active_streams[MAX_NUM_STREAMS]; + struct sst_block block_status; + + void *mmap_mem; + unsigned int mmap_len; + unsigned int unique_id; + unsigned int stream_cnt; + unsigned int active_cnt; + unsigned int am_cnt; +}; + +extern struct intel_sst_drv *sst_ops; +#endif /*__INTEL_SST_COMMON_H */ diff --git a/sound/pci/sst/intel_sst_pvt.h b/sound/pci/sst/intel_sst_pvt.h new file mode 100644 index 0000000..a1e786e --- /dev/null +++ b/sound/pci/sst/intel_sst_pvt.h @@ -0,0 +1,378 @@ +#ifndef __INTEL_SST_PVT_H__ +#define __INTEL_SST_PVT_H__ +/* + * intel_sst_pvt.h - Intel SST Driver for audio engine + * + * Copyright (C) 2008-09 Intel Corporation + * Authors: Vinod Koul <vinod.koul(a)intel.com> + * Harsha Priya <priya.harsha(a)intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * 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; version 2 of the License. + * + * 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. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This file defines all pvt functions + */ +int sst_alloc_stream(char *params, int stream_ops, int codec, int session_id); +int sst_alloc_stream_response(unsigned int str_id, + struct snd_sst_str_type *type); +int sst_pause_stream(int id); +int sst_resume_stream(int id); +int sst_drop_stream(int id); +int sst_free_stream(int id); +int sst_play_frame(int streamID); +int sst_capture_frame(int streamID); +int sst_set_stream_param(int streamID, struct snd_sst_params *str_param); +int sst_route_control(int streamID); +int sst_target_device_select(struct snd_sst_target_device *target_device); +int sst_set_pmic_config(int streamID, struct snd_sst_pmic_config *pc); + +void sst_post_message(struct work_struct *work); +void sst_process_message(struct work_struct *work); +void sst_process_reply(struct work_struct *work); +void sst_send_loop_test(int x); + +int intel_sst_ioctl(struct inode *i_node, struct file *file_ptr, + unsigned int cmd, unsigned long arg); +int intel_sst_open(struct inode *i_node, struct file *file_ptr); +int intel_sst_release(struct inode *i_node, struct file *file_ptr); +int intel_sst_read(struct file *file_ptr, char __user *buf, + size_t count, loff_t *ppos); +int intel_sst_write(struct file *file_ptr, const char __user *buf, + size_t count, loff_t *ppos); +int intel_sst_mmap(struct file *fp, struct vm_area_struct *vma); +ssize_t intel_sst_aio_write(struct kiocb *kiocb, const struct iovec *iov, + unsigned long nr_segs, loff_t offset); +ssize_t intel_sst_aio_read(struct kiocb *kiocb, const struct iovec *iov, + unsigned long nr_segs, loff_t offset); + +int sst_ospm_send_event(int event); +void sst_load_fw(const struct firmware *fw, void *context); +int sst_load_library(struct snd_sst_lib_download *lib, u8 ops, u32 pvt_id); + +static inline int sst_scu_ipc_write(u32 addr, u32 value) +{ +#ifndef CONFIG_SST_IPC_NOT_INCLUDED + int retval = 0, retry = 3; + struct ipc_reg_data ipc_reg = {0}; + + ipc_reg.address = addr; + ipc_reg.data = value; + ipc_reg.ioc = 1; + + while (retry) { + retval = mrst_ipc_write32(&ipc_reg); + if (retval == 0) + break; + retry--; + /*error*/ + sst_err("IPC write failed %x\n", retval); + } + return retval; +#else + return 0; +#endif +} +static inline void sst_fill_header(union ipc_header *header, + int msg, int large, int strID) +{ + header->part.msg_id = msg; + header->part.str_id = strID; + header->part.large = large; + header->part.done = 0; + header->part.busy = 1; + header->part.data = 0; +} + +static inline int sst_get_stream_id(struct intel_sst_drv *sst_ops, int pvt_id) +{ + int i; + + for (i = 1; i < MAX_NUM_STREAMS; i++) { + if (pvt_id == sst_ops->streams[i].sst_id) + return i; + } + return -EINVAL; +} + +static inline unsigned int sst_assign_pvt_id(struct intel_sst_drv *sst_ops) +{ + sst_ops->unique_id++; + if (sst_ops->unique_id >= MAX_NUM_STREAMS) + sst_ops->unique_id = 1; + return sst_ops->unique_id; +} + +static inline int sst_get_block_stream(struct intel_sst_drv *sst_ops) +{ + int i; + + for (i = 0; i < MAX_ACTIVE_STREAM; i++) { + if (BLOCK_UNINIT == sst_ops->alloc_block[i].sst_id) { + sst_ops->alloc_block[i].ops_block.condition = false; + sst_ops->alloc_block[i].ops_block.ret_code = 0; + sst_ops->alloc_block[i].sst_id = 0; + break; + } + } + if (MAX_ACTIVE_STREAM == i) { + sst_err("max alloc_stream reached \n"); + i = -EBUSY; /*active stream limit reached*/ + } + return i; +} + + + +static inline void sst_print_hex(unsigned char *buf, unsigned int size) +{ + unsigned int i; + + for (i = 0; i < size; i++) { + sst_dbg("%02x ", buf[i]); + if ((i != 0) && ((i % 8) == 0)) + sst_dbg("\n"); + } +} + +static inline void sst_init_stream(struct stream_info *stream, + int codec, int sst_id, int ops) +{ + stream->status = STREAM_INIT; + stream->prev = STREAM_UN_INIT; + stream->codec = codec; + stream->sst_id = sst_id; + stream->ops = ops; + stream->data_blk.on = false; + stream->data_blk.condition = false; + stream->data_blk.ret_code = 0; + stream->data_blk.data = NULL; + stream->ctrl_blk.on = false; + stream->ctrl_blk.condition = false; + stream->ctrl_blk.ret_code = 0; + stream->ctrl_blk.data = NULL; + stream->mmapped = false; +} + +static inline void sst_clean_stream(struct stream_info *stream) +{ + struct sst_stream_bufs *bufs = NULL, *_bufs = NULL; + stream->status = STREAM_UN_INIT; + stream->prev = STREAM_UN_INIT; + list_for_each_entry_safe(bufs, _bufs, &stream->bufs, node) { + list_del(&bufs->node); + kfree(bufs); + } +} + +static inline int sst_wait_interruptible(struct intel_sst_drv *sst_ops, + struct sst_block *block) +{ + int retval = 0; + + if (!wait_event_interruptible(sst_ops->wait_queue, block->condition)) { + /*event wake*/ + if (0 != block->ret_code) { + sst_err("stream failed %d\n", block->ret_code); + retval = -EBUSY; + } else { + sst_dbg("event up\n"); + retval = 0; + } + } else { + sst_err("signal interrupted\n"); + retval = -EINTR; + } + return retval; + +} + +static inline int sst_wait_interruptible_timeout(struct intel_sst_drv *sst_ops, + struct sst_block *block, int timeout) +{ + int retval = 0; + + sst_dbg("waiting....\n"); + if (wait_event_interruptible_timeout(sst_ops->wait_queue, + block->condition, + msecs_to_jiffies(timeout))) { + /*event wake*/ + sst_dbg("Event wake ...\n"); + if (0 != block->ret_code) + sst_err("stream failed %d\n", block->ret_code); + else + sst_dbg("event up\n"); + retval = block->ret_code; + } else { + sst_err("timeout occured...\n"); + retval = -EBUSY; + } + return retval; + +} + +static inline int sst_wait_timeout(struct intel_sst_drv *sst_ops, + struct stream_alloc_block *block) +{ + int retval = 0; + + /*NOTE: + Observed that FW processes the alloc msg and replies even + before the alloc thread has finished execution*/ + sst_dbg("waiting for %x, condition %x \n", block->sst_id, + block->ops_block.condition); + if (wait_event_interruptible_timeout(sst_ops->wait_queue, + block->ops_block.condition, + msecs_to_jiffies(SST_BLOCK_TIMEOUT))) { + /*event wake*/ + sst_dbg("Event wake ... %x \n", block->ops_block.condition); + /*check return*/ + if (0 > block->ops_block.ret_code) + sst_err("blk failed %d\n", block->ops_block.ret_code); + else + sst_dbg("blk ret: %d\n", block->ops_block.ret_code); + retval = block->ops_block.ret_code; + } else { + sst_err("Wait timed-out %x\n", block->ops_block.condition); + retval = -EBUSY; + } + return retval; + +} + + +static inline int sst_create_large_msg(struct ipc_post **arg) +{ + struct ipc_post *msg; + + msg = kzalloc(sizeof(struct ipc_post), GFP_ATOMIC); + if (NULL == msg) { + sst_err("kzalloc msg failed \n"); + return -ENOMEM; + } + + msg->mailbox_data = kzalloc(SST_MAILBOX_SIZE, GFP_ATOMIC); + if (NULL == msg->mailbox_data) { + kfree(msg); + sst_err("kzalloc mailbox_data failed \n"); + return -ENOMEM; + }; + *arg = msg; + return 0; +} + +static inline int sst_create_short_msg(struct ipc_post **arg) +{ + struct ipc_post *msg; + + msg = kzalloc(sizeof(*msg), GFP_ATOMIC); + if (NULL == msg) { + sst_err("kzalloc msg failed \n"); + return -ENOMEM; + } + msg->mailbox_data = NULL; + *arg = msg; + return 0; +} + +static inline void sst_print_params(struct snd_sst_stream_params *sparam) +{ + switch (sparam->uc.pcm_params.codec) { + case SST_CODEC_TYPE_PCM: + sst_dbg("pcm \n"); + sst_dbg("chan=%d, sfreq = %d, wd_sz = %d \ + brate = %d framesize = %d \ + samples_per_frame = %d \ + period_cnt = %d\n", + sparam->uc.pcm_params.num_chan, + sparam->uc.pcm_params.sfreq, + sparam->uc.pcm_params.pcm_wd_sz, + sparam->uc.pcm_params.brate, + sparam->uc.pcm_params.frame_size, + sparam->uc.pcm_params.samples_per_frame, + sparam->uc.pcm_params.period_count); + break; + + case SST_CODEC_TYPE_MP3: + sst_dbg("mp3 \n"); + sst_dbg("chan=%d, brate=%d, sfreq = %d, wd_sz = %d\n", + sparam->uc.mp3_params.num_chan, + sparam->uc.mp3_params.brate, + sparam->uc.mp3_params.sfreq, + sparam->uc.mp3_params.pcm_wd_sz); + break; + + case SST_CODEC_TYPE_AAC: + sst_dbg("aac \n"); + sst_dbg("chan=%d, brate=%d, sfreq = %d, wd_sz = %d,asrate=%d\n", + sparam->uc.aac_params.num_chan, + sparam->uc.aac_params.brate, + sparam->uc.aac_params.sfreq, + sparam->uc.aac_params.pcm_wd_sz, + sparam->uc.aac_params.aac_srate); + sst_dbg("mpgid=%d profile=%d, aot = %d\n", + sparam->uc.aac_params.mpg_id, + sparam->uc.aac_params.aac_profile, + sparam->uc.aac_params.aot); + break; + case SST_CODEC_TYPE_WMA9: + sst_dbg("wma type \n"); + sst_dbg("chan=%d, brate=%d, sfreq = %d, wd_sz = %d, tag=%d\n", + sparam->uc.wma_params.num_chan, + sparam->uc.wma_params.brate, + sparam->uc.wma_params.sfreq, + sparam->uc.wma_params.pcm_wd_sz, + sparam->uc.wma_params.format_tag); + sst_dbg("mask=%d, b align=%d, enc opt =%d, op align =%d\n", + sparam->uc.wma_params.channel_mask, + sparam->uc.wma_params.block_align, + sparam->uc.wma_params.wma_encode_opt, + sparam->uc.wma_params.op_align); + break; + default: + sst_dbg("other codec 0x%x\n", sparam->uc.pcm_params.codec); + } +} +static inline int sst_validate_strid(int str_id) +{ + if (str_id <= 0 || str_id >= MAX_NUM_STREAMS) { + sst_err("invalid stream id \n"); + return -EINVAL; + } else + return 0; +} + +/*NOTE: status will +ve for good cases and -ve for error ones*/ +#define MAX_STREAM_FIELD 255 +static inline void sst_wake_up_alloc_block(struct intel_sst_drv *sst_ops, + u8 sst_id, int status, void *data) +{ + int i; + + /*Unblock with retval code*/ + for (i = 0; i < MAX_STREAM_FIELD; i++) { + if (sst_id == sst_ops->alloc_block[i].sst_id) { + sst_ops->alloc_block[i].ops_block.condition = true; + sst_ops->alloc_block[i].ops_block.ret_code = status; + sst_ops->alloc_block[i].ops_block.data = data; + sst_dbg("wake id %d, sst_id %d condition %x\n", i, + sst_ops->alloc_block[i].sst_id, + sst_ops->alloc_block[i].ops_block.condition); + wake_up(&sst_ops->wait_queue); + break; + } + } +} +#endif /*__INTEL_SST_PVT_H__*/ -- 1.5.4.5

3 2

Re: [alsa-devel] No sound with nVidia Corporation MCP79 High Definition Audio (rev b1) (Macbook Pro 5, 5)
by Takashi Iwai 07 Jul '09

07 Jul '09

At Mon, 6 Jul 2009 22:14:56 +0200, Andreas Nüßlein wrote: > > On Monday 06 July 2009 21:26:18 you wrote: > > At Mon, 06 Jul 2009 11:16:35 -0600, > > > > Sean Burke wrote: > > > Scríobh Takashi Iwai: > > > > At Mon, 6 Jul 2009 17:53:46 +0200, > > > > > > > > Andreas Nüßlein wrote: > > > >>> The missing pin configuration initialization was already fixed by > > > >>> the driver overriding it after checking PCI SSID (which is different > > > >>> from the codec SSID). So, this should be no problem. > > > >>> > > > >>> However, the reason why the analog output doesn't work might be > > > >>> different from that. There might be something else missing, but I > > > >>> don't know. > > > >>> > > > >>> > > > >>> Takashi > > > >> > > > >> oh =( > > > >> > > > >> > > > >> hmm.. anything i can do? would it help if i tried changing values > > > >> randomly with hda-analyzer.py? > > > > > > > > Well, did the driver without my change work more or less with > > > > the analog audio, or have you never gotten the analog output? > > > > You can use the generic parser (i.e. the state without cirrus patch) > > > > by passing model=generic option to snd-hda-intel. > > > > > > For my part, nothing worked with the generic driver. I can't confirm > > > digital out, but I can confirm that the kfree error is gone. What > > > options are open for figuring out what remains? > > > > Easy things to test are GPIO bits. Run hda-verb like > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_MASK 0x0f > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x0f > > or > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIR 0x0f > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x0f > > > > etc. CS4206 seems to have 4 GPIO lines, and each bit (0-3) > > corresponds to each GPIO. In many case, GPIO0 or GPIO1 corresponds to > > the amplifier (EAPD) bit. > > Define the GPIO direction of each GPIO bit by SET_GPIO_DIR, and > > turn on/off the GPIO bits by SET_GPIO_DATA. Running > > hda-verb /dev/snd/hwC0D0 0x01 GET_GPIO_DATA 0 > > will show the current GPIO data bits. Or you can check it in codec#* > > proc file. > > > > > > Takashi > > w000000000000000000000000000000000t! =) > > takashi, thank you _so_ much! > > after running all 4 of those: > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_MASK 0x0f > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x0f > > or > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIR 0x0f > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x0f > > > > i suddendly had sound!! :D :D :D :D :D :D :D > only via speakers though - there is no sound via headphones right now. > > mixer channels: > - Master (with Mutebutton), PCM and Front (also with Mute) all work =) > - i don't know what surround would do (or it's extra switch) > - headphones-volumes and mute button don't affect the speakers, which is good > =) > > > is there a way to reset what i did with hda-verb, so that i can figure out > which combination it was exactly? You can just change the value 0x0f to a different value. At least, you can try commands like hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x01 hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x02 hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x04 hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x08 and check the speaker output at each time. Also, check the GPIO direction, hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIR 0x01 hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x01 hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIR 0x02 hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x02 ... Regarding the headphone: is the speaker muted when you plug in the headphone? If not, it's likely an issue of the jack detection. If the speaker is muted but no headphone output, it's a missing initialization (or wrong GPIO setup). Takashi

1 0

[alsa-devel] ASoC updates for 2.6.31
by Mark Brown 07 Jul '09

07 Jul '09

The following changes since commit da9ff1f796e81976935407251815838bef9868d4: Mark Brown (1): ASoC: Only disable pxa2xx-i2s clocks if we enabled them are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31 Grant Likely (2): ASoC: Fix mpc5200-psc-ac97 to ensure the data ready bit is cleared ASoC: add locking to mpc5200-psc-ac97 driver Lars-Peter Clausen (1): ASoC: Fix wm8753 register cache size and initialization Mark Brown (1): ASoC: Fix register cache initialisation for WM8753 sound/soc/codecs/wm8753.c | 6 +++--- sound/soc/fsl/mpc5200_dma.c | 1 + sound/soc/fsl/mpc5200_dma.h | 1 + sound/soc/fsl/mpc5200_psc_ac97.c | 17 ++++++++++++++++- 4 files changed, 21 insertions(+), 4 deletions(-)

2 1

[alsa-devel] [PATCH 0/2] davinci ASoC interface changes
by Kevin Hilman 07 Jul '09

07 Jul '09

This series updates the DaVinci ASoC support after various DaVinci core interface changes. These core changes are part of the DaVinci core changes submitted for 2.6.32. This compiles on top of the v2.6.31-rc2 based 'davinci-next' branch of the DaVinci git repo here: git://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git Naresh Medisetty (1): ASoC: DaVinci: Change default output volume Sudhakar Rajashekhara (1): ASoC: davinci: update after EDMA interface changes sound/soc/codecs/tlv320aic3x.h | 2 +- sound/soc/davinci/davinci-evm.c | 8 ++++---- sound/soc/davinci/davinci-pcm.c | 6 +++--- 3 files changed, 8 insertions(+), 8 deletions(-)

2 7

[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 07 Jul '09

07 Jul '09

The following changes since commit 2115d2c17369df4b70fd161623a40ce98c261989: Mark Brown (1): Merge branch 'for-2.6.31' into for-2.6.32 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Candelaria Villareal, Jorge (2): ASoC: TWL4030: Add EXTMUTE to reduce pop-noise effect ASoC: Zoom2: Update twl4030_setup_data parameters Grant Likely (2): ASoC: Fix mpc5200-psc-ac97 to ensure the data ready bit is cleared ASoC: add locking to mpc5200-psc-ac97 driver Lars-Peter Clausen (1): ASoC: Fix wm8753 register cache size and initialization Lopez Cruz, Misael (1): ASoC: TWL4030: Add tristate callbacks for HiFi and Voice Mark Brown (7): ASoC: Fix register cache initialisation for WM8753 ASoC: Add CODEC volatile register operation ASoC: Fix leaks in WM8731 probe error handling ASoC: Fix WM8960 leaks on probe failure ASoC: Fix leaks in WM8988 registration error handling ASoC: Remove unused AK4535 hardware read functionality Merge branch 'for-2.6.31' into for-2.6.32 Troy Kisky (8): ASoC: DaVinci: i2s, remove MOD_REG_BIT macro ASoC: DaVinci: i2s toggle clock to complete reset ASoc: DaVinci: i2s, minor cleanup ASoC: DaVinci: i2s cleanup ASoC: DaVinci: i2s, only start sample generator if needed ASoC: DaVinci: i2s, minor cleanup of davinci_i2s_startup ASoC: DaVinci: i2s, fix mcbsp_word_length update ASoC: DaVinci: i2s, add davinci_i2s_prepare and shutdown include/sound/soc.h | 2 + sound/soc/codecs/ak4535.c | 16 --- sound/soc/codecs/twl4030.c | 55 ++++++++++ sound/soc/codecs/twl4030.h | 2 + sound/soc/codecs/wm8731.c | 17 ++- sound/soc/codecs/wm8753.c | 6 +- sound/soc/codecs/wm8960.c | 16 ++- sound/soc/codecs/wm8988.c | 8 +- sound/soc/davinci/davinci-i2s.c | 220 ++++++++++++++++++++++---------------- sound/soc/fsl/mpc5200_dma.c | 1 + sound/soc/fsl/mpc5200_dma.h | 1 + sound/soc/fsl/mpc5200_psc_ac97.c | 17 +++- sound/soc/omap/zoom2.c | 17 +++- sound/soc/soc-core.c | 17 +++ 14 files changed, 268 insertions(+), 127 deletions(-)

1 0

[alsa-devel] [PATCH] cmi8330: find OPL3 port automatically
by Ondrej Zary 06 Jul '09

06 Jul '09

My CMI8329 had OPL3 port specified in SB16 resources. But now I found out that it was my modification of the card's PnP EEPROM a couple of years ago (can be done using C9SETROM.EXE utility). I did it because the OPL3 port was completely missing from PnP data. It seems to be hardwired to 0x388 on CMI8329. Find OPL3 port automatically by searching in WSS and SB16 resources. If not found, assume that it's hardwired to 0x388. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- linux-test/sound/isa/cmi8330.c 2009-07-06 21:18:06.000000000 +0200 +++ linux-2.6.30-pentium/sound/isa/cmi8330.c 2009-07-06 21:45:29.000000000 +0200 @@ -362,7 +362,7 @@ wssport[dev] = pnp_port_start(pdev, 0); wssdma[dev] = pnp_dma(pdev, 0); wssirq[dev] = pnp_irq(pdev, 0); - if (acard->type == CMI8330) + if (pnp_port_start(pdev, 1)) fmport[dev] = pnp_port_start(pdev, 1); /* allocate SB16 resources */ @@ -377,8 +377,13 @@ sbdma8[dev] = pnp_dma(pdev, 0); sbdma16[dev] = pnp_dma(pdev, 1); sbirq[dev] = pnp_irq(pdev, 0); - if (acard->type == CMI8329) - fmport[dev] = pnp_port_start(pdev, 1); + /* On CMI8239, the OPL3 port might be present in SB16 PnP resources */ + if (fmport[dev] == SNDRV_AUTO_PORT) { + if (pnp_port_start(pdev, 1)) + fmport[dev] = pnp_port_start(pdev, 1); + else + fmport[dev] = 0x388; /* Or hardwired */ + } /* allocate MPU-401 resources */ pdev = acard->mpu; -- Ondrej Zary

1 0

Re: [alsa-devel] No sound with nVidia Corporation MCP79 High Definition Audio (rev b1) (Macbook Pro 5, 5)
by Takashi Iwai 06 Jul '09

06 Jul '09

At Mon, 6 Jul 2009 17:53:46 +0200, Andreas Nüßlein wrote: > > > The missing pin configuration initialization was already fixed by > > the driver overriding it after checking PCI SSID (which is different > > from the codec SSID). So, this should be no problem. > > > > However, the reason why the analog output doesn't work might be > > different from that. There might be something else missing, but I > > don't know. > > > > > > Takashi > > oh =( > > > hmm.. anything i can do? would it help if i tried changing values randomly > with hda-analyzer.py? Well, did the driver without my change work more or less with the analog audio, or have you never gotten the analog output? You can use the generic parser (i.e. the state without cirrus patch) by passing model=generic option to snd-hda-intel. Takashi

2 2

Re: [alsa-devel] No sound with nVidia Corporation MCP79 High Definition Audio (rev b1) (Macbook Pro 5, 5)
by Takashi Iwai 06 Jul '09

06 Jul '09

At Mon, 6 Jul 2009 17:23:17 +0200, Andreas Nüßlein wrote: > > On Monday 06 July 2009 17:13:29 you wrote: > > At Mon, 6 Jul 2009 17:03:52 +0200, > > > > Andreas Nüßlein wrote: > > > On Monday 06 July 2009 16:47:48 you wrote: > > > > At Mon, 6 Jul 2009 16:36:09 +0200, > > > > > > > > Andreas Nüßlein wrote: > > > > > On Monday 06 July 2009 16:16:06 you wrote: > > > > > > At Mon, 6 Jul 2009 16:03:04 +0200, > > > > > > > > > > > > Andreas Nüßlein wrote: > > > > > > > On Monday 06 July 2009 15:39:09 you wrote: > > > > > > > > At Mon, 6 Jul 2009 14:59:09 +0200, > > > > > > > > > > > > > > > > Andreas Nüßlein wrote: > > > > > > > > > On Monday 06 July 2009 14:44:56 you wrote: > > > > > > > > > > At Mon, 6 Jul 2009 12:19:11 +0200, > > > > > > > > > > > > > > > > > > > > Andreas Nüßlein wrote: > > > > > > > > > > > On Monday 06 July 2009 12:08:11 you wrote: > > > > > > > > > > > > At Mon, 6 Jul 2009 11:45:07 +0200, > > > > > > > > > > > > > > > > > > > > > > > > Andreas Nüßlein wrote: > > > > > > > > > > > > > [1 <text/plain; iso-8859-1 (quoted-printable)>] > > > > > > > > > > > > > > > > > > > > > > > > > > On Monday 06 July 2009 09:23:07 you wrote: > > > > > > > > > > > > > > At Sun, 5 Jul 2009 12:30:52 +0200, > > > > > > > > > > > > > > > > > > > > > > > > > > > > > however i was able to reproduce the "bad kfree", > > > > > > > > > > > > > > > by changing the volumes in alsamixer. it always > > > > > > > > > > > > > > > gave me the same "called from ID" so i tried > > > > > > > > > > > > > > > "while (true) do cat /proc/kallsyms | grep > > > > > > > > > > > > > > > ffffffffa001b08b; done" and then changed the > > > > > > > > > > > > > > > volume, but the grep never returned anything, > > > > > > > > > > > > > > > dmesg of course got more and more of those > > > > > > > > > > > > > > > kfree-errors. > > > > > > > > > > > > > > > > > > > > > > > > > > > > The address won't match exactly in /proc/kasllsyms. > > > > > > > > > > > > > > But you can guess which function by address. First > > > > > > > > > > > > > > sort /proc/kallsyms then check the position that > > > > > > > > > > > > > > fits with the given address. > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Takashi > > > > > > > > > > > > > > > > > > > > > > > > > > well i'm not really sure _I_ can guess ;) i guess > > > > > > > > > > > > > it's either "t snd_ctl_ioctl [snd]" or "t > > > > > > > > > > > > > snd_ctl_ioctl_compat [snd]". > > > > > > > > > > > > > > > > > > > > > > > > > > here are the greps, "snd: bad kfree (called from > > > > > > > > > > > > > ffffffffa001b08b)" being the error: > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > $ cat /proc/kallsyms | sort | grep ffffffffa001a > > > > > > > > > > > > > ffffffffa001a045 t snd_ctl_elem_info [snd] > > > > > > > > > > > > > ffffffffa001a17a t copy_ctl_value_from_user [snd] > > > > > > > > > > > > > ffffffffa001a34f t snd_ctl_elem_read [snd] > > > > > > > > > > > > > ffffffffa001a413 t snd_ctl_elem_write [snd] > > > > > > > > > > > > > ffffffffa001a515 T snd_ctl_rename_id [snd] > > > > > > > > > > > > > ffffffffa001a5b5 T snd_ctl_activate_id [snd] > > > > > > > > > > > > > ffffffffa001a66e T snd_ctl_remove_id [snd] > > > > > > > > > > > > > ffffffffa001a6ed T snd_ctl_add [snd] > > > > > > > > > > > > > ffffffffa001a6ed u snd_ctl_add [snd_hda_codec] > > > > > > > > > > > > > ffffffffa001a947 t snd_ctl_elem_add [snd] > > > > > > > > > > > > > ffffffffa001abd3 t snd_ctl_elem_add_compat [snd] > > > > > > > > > > > > > ffffffffa001ad37 t snd_ctl_elem_add_user [snd] > > > > > > > > > > > > > ffffffffa001ada6 t snd_ctl_ioctl [snd] > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > $ cat /proc/kallsyms | sort | grep ffffffffa001b > > > > > > > > > > > > > ffffffffa001b4c4 t snd_ctl_ioctl_compat [snd] > > > > > > > > > > > > > ffffffffa001b9ba T snd_ctl_new1 [snd] > > > > > > > > > > > > > ffffffffa001b9ba u snd_ctl_new1 [snd_hda_codec] > > > > > > > > > > > > > ffffffffa001b9ba u snd_ctl_new1 > > > > > > > > > > > > > [snd_hda_codec_cirrus] ffffffffa001bb30 T > > > > > > > > > > > > > snd_pci_quirk_lookup [snd] ffffffffa001bb30 u > > > > > > > > > > > > > snd_pci_quirk_lookup [snd_hda_codec] ffffffffa001bb30 > > > > > > > > > > > > > u snd_pci_quirk_lookup [snd_hda_intel] > > > > > > > > > > > > > ffffffffa001bb93 T snd_verbose_printk [snd] > > > > > > > > > > > > > ffffffffa001bb93 u snd_verbose_printk > > > > > > > > > > > > > [snd_hda_codec] ffffffffa001bb93 u snd_verbose_printk > > > > > > > > > > > > > [snd_hda_intel] ffffffffa001bb93 u > > > > > > > > > > > > > snd_verbose_printk [snd_hwdep] ffffffffa001bb93 u > > > > > > > > > > > > > snd_verbose_printk [snd_pcm] ffffffffa001bb93 u > > > > > > > > > > > > > snd_verbose_printk [snd_timer] ffffffffa001bc7b T > > > > > > > > > > > > > release_and_free_resource [snd] ffffffffa001bcd4 t > > > > > > > > > > > > > snd_device_register_all [snd] ffffffffa001bd76 T > > > > > > > > > > > > > snd_device_register [snd] ffffffffa001bd76 u > > > > > > > > > > > > > snd_device_register [snd_pcm] ffffffffa001be67 t > > > > > > > > > > > > > snd_device_disconnect [snd] ffffffffa001bf5d t > > > > > > > > > > > > > snd_device_disconnect_all [snd] > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > hope that helps =) > > > > > > > > > > > > > > > > > > > > > > > > Maybe the patch below can give a better clue. > > > > > > > > > > > > This will spew stack traces when this error is > > > > > > > > > > > > triggered. Don't be surprised. > > > > > > > > > > > > > > > > > > > > > > oki - new output =) > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > [16306.365597] snd: bad kfree (called from > > > > > > > > > > > ffffffffa001b08f) [16306.365604] Pid: 26533, comm: > > > > > > > > > > > alsamixer Tainted: P W 2.6.30.1 #2 [16306.365610] Call > > > > > > > > > > > Trace: > > > > > > > > > > > [16306.365632] [<ffffffffa001b08f>] ? > > > > > > > > > > > snd_ctl_ioctl+0x2e5/0x71e [snd] [16306.365645] > > > > > > > > > > > [<ffffffff802b8bf9>] ? > > > > > > > > > > > __inc_zone_state+0x20/0x9b [16306.365655] > > > > > > > > > > > [<ffffffff802cfffc>] ? alloc_page_vma+0x197/0x1e1 > > > > > > > > > > > [16306.365666] [<ffffffff802afb63>] ? > > > > > > > > > > > __lru_cache_add+0x93/0xdd [16306.365676] > > > > > > > > > > > [<ffffffff802ea651>] ? vfs_ioctl+0x35/0x97 > > > > > > > > > > > [16306.365686] [<ffffffff802eaae5>] ? > > > > > > > > > > > do_vfs_ioctl+0x432/0x482 [16306.365696] > > > > > > > > > > > [<ffffffff806336bf>] ? > > > > > > > > > > > _spin_lock_irqsave+0x28/0x5c [16306.365705] > > > > > > > > > > > [<ffffffff80247e22>] ? do_page_fault+0x272/0x29e > > > > > > > > > > > [16306.365715] [<ffffffff802eab82>] ? > > > > > > > > > > > sys_ioctl+0x4d/0x81 [16306.365726] [<ffffffff8022af2b>] > > > > > > > > > > > ? > > > > > > > > > > > system_call_fastpath+0x16/0x1b > > > > > > > > > > > > > > > > > > > > Try the very latest alsa-driver-unstable snapshot. > > > > > > > > > > This bug should have been fixed, also possibly with CS420x > > > > > > > > > > parser bug, too. > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Takashi > > > > > > > > > > > > > > > > > > so dmesg knows the following: > > > > > > > > > > > > > > > > > > [18042.848549] HDA Intel 0000:00:08.0: PCI INT A -> > > > > > > > > > Link[LAZA] -> GSI 23 (level, low) -> IRQ 23 [18042.848623] > > > > > > > > > HDA Intel 0000:00:08.0: setting latency timer to 64 > > > > > > > > > [18043.007568] ALSA > > > > > > > > > /home/nutz/tmp/alsa/alsa-driver-unstable/pci/hda/../../alsa-k > > > > > > > > >erne l/pc i/hd a/hda_codec.c:3851: autoconfig: line_outs=1 > > > > > > > > > (0xa/0x0/0x0/0x0/0x0) [18043.007584] ALSA > > > > > > > > > /home/nutz/tmp/alsa/alsa-driver-unstable/pci/hda/../../alsa-k > > > > > > > > >erne l/pc i/hd a/hda_codec.c:3855: speaker_outs=1 > > > > > > > > > (0xb/0x0/0x0/0x0/0x0) [18043.007596] ALSA > > > > > > > > > /home/nutz/tmp/alsa/alsa-driver-unstable/pci/hda/../../alsa-k > > > > > > > > >erne l/pc i/hd a/hda_codec.c:3859: hp_outs=1 > > > > > > > > > (0x9/0x0/0x0/0x0/0x0) [18043.007608] ALSA > > > > > > > > > /home/nutz/tmp/alsa/alsa-driver-unstable/pci/hda/../../alsa-k > > > > > > > > >erne l/pc i/hd a/hda_codec.c:3860: mono: mono_out=0x0 > > > > > > > > > [18043.007618] ALSA > > > > > > > > > /home/nutz/tmp/alsa/alsa-driver-unstable/pci/hda/../../alsa-k > > > > > > > > >erne l/pc i/hd a/hda_codec.c:3863: dig-out=0x10/0x15 > > > > > > > > > [18043.007628] ALSA > > > > > > > > > /home/nutz/tmp/alsa/alsa-driver-unstable/pci/hda/../../alsa-k > > > > > > > > >erne l/pc i/hd a/hda_codec.c:3871: inputs: mic=0xd, > > > > > > > > > fmic=0x0, line=0xc, fline=0x0, cd=0x0, aux=0x0 > > > > > > > > > [18043.007642] ALSA > > > > > > > > > /home/nutz/tmp/alsa/alsa-driver-unstable/pci/hda/../../alsa-k > > > > > > > > >erne l/pc i/hd a/hda_codec.c:3873: dig-in=0x12 > > > > > > > > > [18043.121187] ALSA > > > > > > > > > /home/nutz/tmp/alsa/alsa-driver-unstable/acore/control.c:337: > > > > > > > > > control 2:0:0:IEC958 Capture Switch:0 is already present > > > > > > > > > [18043.121204] hda_codec: cannot build controlsfor #0 (error > > > > > > > > > -16) > > > > > > > > > > > > > > > > > > > > > > > > > > > and i don't have a mixer anymore ;) > > > > > > > > > > > > > > > > It's because your chip was uninitialized by BIOS or whatever, > > > > > > > > the another state you got sometimes. > > > > > > > > > > > > > > > > Now fixed in the driver to force to initialize pins. > > > > > > > > Try the latest tarball again. > > > > > > > > > > > > > > > > > > > > > > > > Takashi > > > > > > > > > > > > > > TAKASHIIIIIIIIIIIII =) > > > > > > > !!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!! > > > > > > > > > > > > > > digital out is working... i tried the new snapshot (eventhough, > > > > > > > as you said my subsystem is now again uninitialized) and i got > > > > > > > back the alsamixer- capabilities; > > > > > > > > > > > > > > i ran mplayer test.wav and started playing around with the > > > > > > > mixer-controls. > > > > > > > > > > > > > > when i reached IEC958 and turned it on, my out-jack began to glow > > > > > > > red =) so that works. i rushed over to my dorm-partners > > > > > > > speakersystem and plugged it in - voila that actually works! (i > > > > > > > tried a DTS video with mplayer -vc hwdts ) > > > > > > > > > > > > > > that is SO cool =) > > > > > > > > > > > > > > however as you might've figured out, analog out is not working, > > > > > > > yet ;( i tried the built-in speakers as well as the headphones > > > > > > > (and of course all the > > > > > > > > > > > > Hm, so you don't get any output although no errors with aplay? > > > > > > > > > > > > Please give alsa-info.sh output with the latest driver to be sure. > > > > > > > > > > > > > > > > > > Takashi > > > > > > > > > > and here is the alsa-info with the "correct" subsystem id. > > > > > > > > > > i also was able to get the wrong subsystem id again: i did > > > > > /etc/init.d/alsasound stop && modprobe snd-hda-intel - that > > > > > changed it. (i'll have a look into init.d/alsasound to see if there > > > > > are any clues) > > > > > > > > Interesting. So, resetting the controller alone can change the > > > > SSID. It doesn't happen on other systems. > > > > > > > > > > > > Could you try to apply the patch below on alsa-kernel directory with > > > > -p2? This will disable the additional COEF setups, which I blindly > > > > copied from older cs4207 patch. > > > > > > > > > > > > thanks, > > > > > > > > Takashi > > > > > > > > --- > > > > diff --git a/sound/pci/hda/patch_cirrus.c > > > > b/sound/pci/hda/patch_cirrus.c index b1fd183..fda83b2 100644 > > > > --- a/sound/pci/hda/patch_cirrus.c > > > > +++ b/sound/pci/hda/patch_cirrus.c > > > > @@ -933,7 +933,7 @@ static void init_input(struct hda_codec *codec) > > > > change_cur_input(codec, spec->cur_input, 1); > > > > if (spec->mic_detect) > > > > cs_automic(codec); > > > > - > > > > +#if 0 > > > > coef = 0x000a; /* ADC1/2 - Digital and Analog Soft Ramp */ > > > > if (is_active_pin(codec, CS_DMIC2_PIN_NID)) > > > > coef |= 0x0500; /* DMIC2 enable 2 channels, disable GPIO1 */ > > > > @@ -943,6 +943,7 @@ static void init_input(struct hda_codec *codec) > > > > * IDX_SPDIF_CTL. > > > > */ > > > > cs_vendor_coef_set(codec, IDX_ADC_CFG, coef); > > > > +#endif > > > > } > > > > > > > > static struct hda_verb cs_coef_init_verbs[] = { > > > > @@ -980,10 +981,10 @@ static int cs_init(struct hda_codec *codec) > > > > { > > > > struct cs_spec *spec = codec->spec; > > > > > > > > - snd_hda_sequence_write(codec, cs_coef_init_verbs); > > > > + /*snd_hda_sequence_write(codec, cs_coef_init_verbs);*/ > > > > init_output(codec); > > > > init_input(codec); > > > > - init_digital(codec); > > > > + /*init_digital(codec);*/ > > > > return 0; > > > > } > > > > > > wow okay... > > > so aplay test.wav didn't give me any sound, but when i tried to stop it > > > (Ctrl-C) i couldn't and i had to kill it from another term, which > > > resulted in this: > > > > When you run like below > > % aplay -Dplughw -vvv somefile.wav > > does it proceed? > > > > If not, it means that the DMA got stuck somewhere. If it goes, > > then the problem is in another place. > > > > > > Takashi > > > $ aplay -Dplughw -vvv media/test.wav > Playing WAVE 'media/test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, > Stereo > Plug PCM: Hardware PCM card 0 'HDA NVidia' device 0 subdevice 0 > Its setup is: > stream : PLAYBACK > access : RW_INTERLEAVED > format : S16_LE > subformat : STD > channels : 2 > rate : 44100 > exact rate : 44100 (44100/1) > msbits : 16 > buffer_size : 16384 > period_size : 4096 > period_time : 92879 > tstamp_mode : NONE > period_step : 1 > avail_min : 4096 > period_event : 0 > start_threshold : 16384 > stop_threshold : 16384 > silence_threshold: 0 > silence_size : 0 > boundary : 4611686018427387904 > appl_ptr : 0 > hw_ptr : 0 > Max peak (8192 samples): 0x0000003f # 0% > Max peak (8192 samples): 0x0000004a # 0% > > ... > > > yeah this works every time. > > so is it okay right now when the subsystem ID is the "uninitialized" one? or > should i restart the machine so i get the "correct one" back? The missing pin configuration initialization was already fixed by the driver overriding it after checking PCI SSID (which is different from the codec SSID). So, this should be no problem. However, the reason why the analog output doesn't work might be different from that. There might be something else missing, but I don't know. Takashi

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Re: [alsa-devel] No sound with nVidia Corporation MCP79 High Definition Audio (rev b1) (Macbook Pro 5, 5)
by Takashi Iwai 06 Jul '09

06 Jul '09

At Mon, 6 Jul 2009 17:03:52 +0200, Andreas Nüßlein wrote: > > On Monday 06 July 2009 16:47:48 you wrote: > > At Mon, 6 Jul 2009 16:36:09 +0200, > > > > Andreas Nüßlein wrote: > > > On Monday 06 July 2009 16:16:06 you wrote: > > > > At Mon, 6 Jul 2009 16:03:04 +0200, > > > > > > > > Andreas Nüßlein wrote: > > > > > On Monday 06 July 2009 15:39:09 you wrote: > > > > > > At Mon, 6 Jul 2009 14:59:09 +0200, > > > > > > > > > > > > Andreas Nüßlein wrote: > > > > > > > On Monday 06 July 2009 14:44:56 you wrote: > > > > > > > > At Mon, 6 Jul 2009 12:19:11 +0200, > > > > > > > > > > > > > > > > Andreas Nüßlein wrote: > > > > > > > > > On Monday 06 July 2009 12:08:11 you wrote: > > > > > > > > > > At Mon, 6 Jul 2009 11:45:07 +0200, > > > > > > > > > > > > > > > > > > > > Andreas Nüßlein wrote: > > > > > > > > > > > [1 <text/plain; iso-8859-1 (quoted-printable)>] > > > > > > > > > > > > > > > > > > > > > > On Monday 06 July 2009 09:23:07 you wrote: > > > > > > > > > > > > At Sun, 5 Jul 2009 12:30:52 +0200, > > > > > > > > > > > > > > > > > > > > > > > > > however i was able to reproduce the "bad kfree", by > > > > > > > > > > > > > changing the volumes in alsamixer. it always gave me > > > > > > > > > > > > > the same "called from ID" so i tried "while (true) do > > > > > > > > > > > > > cat /proc/kallsyms | grep ffffffffa001b08b; done" and > > > > > > > > > > > > > then changed the volume, but the grep never returned > > > > > > > > > > > > > anything, dmesg of course got more and more of those > > > > > > > > > > > > > kfree-errors. > > > > > > > > > > > > > > > > > > > > > > > > The address won't match exactly in /proc/kasllsyms. > > > > > > > > > > > > But you can guess which function by address. First > > > > > > > > > > > > sort /proc/kallsyms then check the position that fits > > > > > > > > > > > > with the given address. > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Takashi > > > > > > > > > > > > > > > > > > > > > > well i'm not really sure _I_ can guess ;) i guess it's > > > > > > > > > > > either "t snd_ctl_ioctl [snd]" or "t > > > > > > > > > > > snd_ctl_ioctl_compat [snd]". > > > > > > > > > > > > > > > > > > > > > > here are the greps, "snd: bad kfree (called from > > > > > > > > > > > ffffffffa001b08b)" being the error: > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > $ cat /proc/kallsyms | sort | grep ffffffffa001a > > > > > > > > > > > ffffffffa001a045 t snd_ctl_elem_info [snd] > > > > > > > > > > > ffffffffa001a17a t copy_ctl_value_from_user [snd] > > > > > > > > > > > ffffffffa001a34f t snd_ctl_elem_read [snd] > > > > > > > > > > > ffffffffa001a413 t snd_ctl_elem_write [snd] > > > > > > > > > > > ffffffffa001a515 T snd_ctl_rename_id [snd] > > > > > > > > > > > ffffffffa001a5b5 T snd_ctl_activate_id [snd] > > > > > > > > > > > ffffffffa001a66e T snd_ctl_remove_id [snd] > > > > > > > > > > > ffffffffa001a6ed T snd_ctl_add [snd] > > > > > > > > > > > ffffffffa001a6ed u snd_ctl_add [snd_hda_codec] > > > > > > > > > > > ffffffffa001a947 t snd_ctl_elem_add [snd] > > > > > > > > > > > ffffffffa001abd3 t snd_ctl_elem_add_compat [snd] > > > > > > > > > > > ffffffffa001ad37 t snd_ctl_elem_add_user [snd] > > > > > > > > > > > ffffffffa001ada6 t snd_ctl_ioctl [snd] > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > $ cat /proc/kallsyms | sort | grep ffffffffa001b > > > > > > > > > > > ffffffffa001b4c4 t snd_ctl_ioctl_compat [snd] > > > > > > > > > > > ffffffffa001b9ba T snd_ctl_new1 [snd] > > > > > > > > > > > ffffffffa001b9ba u snd_ctl_new1 [snd_hda_codec] > > > > > > > > > > > ffffffffa001b9ba u snd_ctl_new1 [snd_hda_codec_cirrus] > > > > > > > > > > > ffffffffa001bb30 T snd_pci_quirk_lookup [snd] > > > > > > > > > > > ffffffffa001bb30 u snd_pci_quirk_lookup [snd_hda_codec] > > > > > > > > > > > ffffffffa001bb30 u snd_pci_quirk_lookup [snd_hda_intel] > > > > > > > > > > > ffffffffa001bb93 T snd_verbose_printk [snd] > > > > > > > > > > > ffffffffa001bb93 u snd_verbose_printk [snd_hda_codec] > > > > > > > > > > > ffffffffa001bb93 u snd_verbose_printk [snd_hda_intel] > > > > > > > > > > > ffffffffa001bb93 u snd_verbose_printk [snd_hwdep] > > > > > > > > > > > ffffffffa001bb93 u snd_verbose_printk [snd_pcm] > > > > > > > > > > > ffffffffa001bb93 u snd_verbose_printk [snd_timer] > > > > > > > > > > > ffffffffa001bc7b T release_and_free_resource [snd] > > > > > > > > > > > ffffffffa001bcd4 t snd_device_register_all [snd] > > > > > > > > > > > ffffffffa001bd76 T snd_device_register [snd] > > > > > > > > > > > ffffffffa001bd76 u snd_device_register [snd_pcm] > > > > > > > > > > > ffffffffa001be67 t snd_device_disconnect [snd] > > > > > > > > > > > ffffffffa001bf5d t snd_device_disconnect_all [snd] > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > hope that helps =) > > > > > > > > > > > > > > > > > > > > Maybe the patch below can give a better clue. > > > > > > > > > > This will spew stack traces when this error is triggered. > > > > > > > > > > Don't be surprised. > > > > > > > > > > > > > > > > > > oki - new output =) > > > > > > > > > > > > > > > > > > > > > > > > > > > [16306.365597] snd: bad kfree (called from ffffffffa001b08f) > > > > > > > > > [16306.365604] Pid: 26533, comm: alsamixer Tainted: P > > > > > > > > > W 2.6.30.1 #2 [16306.365610] Call Trace: > > > > > > > > > [16306.365632] [<ffffffffa001b08f>] ? > > > > > > > > > snd_ctl_ioctl+0x2e5/0x71e [snd] [16306.365645] > > > > > > > > > [<ffffffff802b8bf9>] ? > > > > > > > > > __inc_zone_state+0x20/0x9b [16306.365655] > > > > > > > > > [<ffffffff802cfffc>] ? alloc_page_vma+0x197/0x1e1 > > > > > > > > > [16306.365666] [<ffffffff802afb63>] ? > > > > > > > > > __lru_cache_add+0x93/0xdd [16306.365676] > > > > > > > > > [<ffffffff802ea651>] ? vfs_ioctl+0x35/0x97 > > > > > > > > > [16306.365686] [<ffffffff802eaae5>] ? > > > > > > > > > do_vfs_ioctl+0x432/0x482 [16306.365696] [<ffffffff806336bf>] > > > > > > > > > ? > > > > > > > > > _spin_lock_irqsave+0x28/0x5c [16306.365705] > > > > > > > > > [<ffffffff80247e22>] ? do_page_fault+0x272/0x29e > > > > > > > > > [16306.365715] [<ffffffff802eab82>] ? sys_ioctl+0x4d/0x81 > > > > > > > > > [16306.365726] [<ffffffff8022af2b>] ? > > > > > > > > > system_call_fastpath+0x16/0x1b > > > > > > > > > > > > > > > > Try the very latest alsa-driver-unstable snapshot. > > > > > > > > This bug should have been fixed, also possibly with CS420x > > > > > > > > parser bug, too. > > > > > > > > > > > > > > > > > > > > > > > > Takashi > > > > > > > > > > > > > > so dmesg knows the following: > > > > > > > > > > > > > > [18042.848549] HDA Intel 0000:00:08.0: PCI INT A -> Link[LAZA] -> > > > > > > > GSI 23 (level, low) -> IRQ 23 [18042.848623] HDA Intel > > > > > > > 0000:00:08.0: setting latency timer to 64 [18043.007568] ALSA > > > > > > > /home/nutz/tmp/alsa/alsa-driver-unstable/pci/hda/../../alsa-kerne > > > > > > >l/pc i/hd a/hda_codec.c:3851: autoconfig: line_outs=1 > > > > > > > (0xa/0x0/0x0/0x0/0x0) [18043.007584] ALSA > > > > > > > /home/nutz/tmp/alsa/alsa-driver-unstable/pci/hda/../../alsa-kerne > > > > > > >l/pc i/hd a/hda_codec.c:3855: speaker_outs=1 > > > > > > > (0xb/0x0/0x0/0x0/0x0) [18043.007596] ALSA > > > > > > > /home/nutz/tmp/alsa/alsa-driver-unstable/pci/hda/../../alsa-kerne > > > > > > >l/pc i/hd a/hda_codec.c:3859: hp_outs=1 (0x9/0x0/0x0/0x0/0x0) > > > > > > > [18043.007608] ALSA > > > > > > > /home/nutz/tmp/alsa/alsa-driver-unstable/pci/hda/../../alsa-kerne > > > > > > >l/pc i/hd a/hda_codec.c:3860: mono: mono_out=0x0 [18043.007618] > > > > > > > ALSA > > > > > > > /home/nutz/tmp/alsa/alsa-driver-unstable/pci/hda/../../alsa-kerne > > > > > > >l/pc i/hd a/hda_codec.c:3863: dig-out=0x10/0x15 [18043.007628] > > > > > > > ALSA > > > > > > > /home/nutz/tmp/alsa/alsa-driver-unstable/pci/hda/../../alsa-kerne > > > > > > >l/pc i/hd a/hda_codec.c:3871: inputs: mic=0xd, fmic=0x0, > > > > > > > line=0xc, fline=0x0, cd=0x0, aux=0x0 > > > > > > > [18043.007642] ALSA > > > > > > > /home/nutz/tmp/alsa/alsa-driver-unstable/pci/hda/../../alsa-kerne > > > > > > >l/pc i/hd a/hda_codec.c:3873: dig-in=0x12 [18043.121187] ALSA > > > > > > > /home/nutz/tmp/alsa/alsa-driver-unstable/acore/control.c:337: > > > > > > > control 2:0:0:IEC958 Capture Switch:0 is already present > > > > > > > [18043.121204] hda_codec: cannot build controlsfor #0 (error -16) > > > > > > > > > > > > > > > > > > > > > and i don't have a mixer anymore ;) > > > > > > > > > > > > It's because your chip was uninitialized by BIOS or whatever, the > > > > > > another state you got sometimes. > > > > > > > > > > > > Now fixed in the driver to force to initialize pins. > > > > > > Try the latest tarball again. > > > > > > > > > > > > > > > > > > Takashi > > > > > > > > > > TAKASHIIIIIIIIIIIII =) > > > > > !!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!! > > > > > > > > > > digital out is working... i tried the new snapshot (eventhough, as > > > > > you said my subsystem is now again uninitialized) and i got back the > > > > > alsamixer- capabilities; > > > > > > > > > > i ran mplayer test.wav and started playing around with the > > > > > mixer-controls. > > > > > > > > > > when i reached IEC958 and turned it on, my out-jack began to glow red > > > > > =) so that works. i rushed over to my dorm-partners speakersystem > > > > > and plugged it in - voila that actually works! (i tried a DTS video > > > > > with mplayer -vc hwdts ) > > > > > > > > > > that is SO cool =) > > > > > > > > > > however as you might've figured out, analog out is not working, yet > > > > > ;( i tried the built-in speakers as well as the headphones (and of > > > > > course all the > > > > > > > > Hm, so you don't get any output although no errors with aplay? > > > > > > > > Please give alsa-info.sh output with the latest driver to be sure. > > > > > > > > > > > > Takashi > > > > > > and here is the alsa-info with the "correct" subsystem id. > > > > > > i also was able to get the wrong subsystem id again: i did > > > /etc/init.d/alsasound stop && modprobe snd-hda-intel - that changed > > > it. (i'll have a look into init.d/alsasound to see if there are any > > > clues) > > > > Interesting. So, resetting the controller alone can change the > > SSID. It doesn't happen on other systems. > > > > > > Could you try to apply the patch below on alsa-kernel directory with > > -p2? This will disable the additional COEF setups, which I blindly > > copied from older cs4207 patch. > > > > > > thanks, > > > > Takashi > > > > --- > > diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c > > index b1fd183..fda83b2 100644 > > --- a/sound/pci/hda/patch_cirrus.c > > +++ b/sound/pci/hda/patch_cirrus.c > > @@ -933,7 +933,7 @@ static void init_input(struct hda_codec *codec) > > change_cur_input(codec, spec->cur_input, 1); > > if (spec->mic_detect) > > cs_automic(codec); > > - > > +#if 0 > > coef = 0x000a; /* ADC1/2 - Digital and Analog Soft Ramp */ > > if (is_active_pin(codec, CS_DMIC2_PIN_NID)) > > coef |= 0x0500; /* DMIC2 enable 2 channels, disable GPIO1 */ > > @@ -943,6 +943,7 @@ static void init_input(struct hda_codec *codec) > > * IDX_SPDIF_CTL. > > */ > > cs_vendor_coef_set(codec, IDX_ADC_CFG, coef); > > +#endif > > } > > > > static struct hda_verb cs_coef_init_verbs[] = { > > @@ -980,10 +981,10 @@ static int cs_init(struct hda_codec *codec) > > { > > struct cs_spec *spec = codec->spec; > > > > - snd_hda_sequence_write(codec, cs_coef_init_verbs); > > + /*snd_hda_sequence_write(codec, cs_coef_init_verbs);*/ > > init_output(codec); > > init_input(codec); > > - init_digital(codec); > > + /*init_digital(codec);*/ > > return 0; > > } > > > wow okay... > so aplay test.wav didn't give me any sound, but when i tried to stop it (Ctrl-C) i couldn't and i had to kill it from another term, which resulted in this: When you run like below % aplay -Dplughw -vvv somefile.wav does it proceed? If not, it means that the DMA got stuck somewhere. If it goes, then the problem is in another place. Takashi

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Re: [alsa-devel] No sound with nVidia Corporation MCP79 High Definition Audio (rev b1) (Macbook Pro 5, 5)
by Takashi Iwai 06 Jul '09

06 Jul '09

At Mon, 6 Jul 2009 16:36:09 +0200, Andreas Nüßlein wrote: > > On Monday 06 July 2009 16:16:06 you wrote: > > At Mon, 6 Jul 2009 16:03:04 +0200, > > > > Andreas Nüßlein wrote: > > > On Monday 06 July 2009 15:39:09 you wrote: > > > > At Mon, 6 Jul 2009 14:59:09 +0200, > > > > > > > > Andreas Nüßlein wrote: > > > > > On Monday 06 July 2009 14:44:56 you wrote: > > > > > > At Mon, 6 Jul 2009 12:19:11 +0200, > > > > > > > > > > > > Andreas Nüßlein wrote: > > > > > > > On Monday 06 July 2009 12:08:11 you wrote: > > > > > > > > At Mon, 6 Jul 2009 11:45:07 +0200, > > > > > > > > > > > > > > > > Andreas Nüßlein wrote: > > > > > > > > > [1 <text/plain; iso-8859-1 (quoted-printable)>] > > > > > > > > > > > > > > > > > > On Monday 06 July 2009 09:23:07 you wrote: > > > > > > > > > > At Sun, 5 Jul 2009 12:30:52 +0200, > > > > > > > > > > > > > > > > > > > > > however i was able to reproduce the "bad kfree", by > > > > > > > > > > > changing the volumes in alsamixer. it always gave me the > > > > > > > > > > > same "called from ID" so i tried "while (true) do cat > > > > > > > > > > > /proc/kallsyms | grep ffffffffa001b08b; done" and then > > > > > > > > > > > changed the volume, but the grep never returned anything, > > > > > > > > > > > dmesg of course got more and more of those kfree-errors. > > > > > > > > > > > > > > > > > > > > The address won't match exactly in /proc/kasllsyms. > > > > > > > > > > But you can guess which function by address. First sort > > > > > > > > > > /proc/kallsyms then check the position that fits with the > > > > > > > > > > given address. > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Takashi > > > > > > > > > > > > > > > > > > well i'm not really sure _I_ can guess ;) i guess it's > > > > > > > > > either "t snd_ctl_ioctl [snd]" or "t > > > > > > > > > snd_ctl_ioctl_compat [snd]". > > > > > > > > > > > > > > > > > > here are the greps, "snd: bad kfree (called from > > > > > > > > > ffffffffa001b08b)" being the error: > > > > > > > > > > > > > > > > > > > > > > > > > > > $ cat /proc/kallsyms | sort | grep ffffffffa001a > > > > > > > > > ffffffffa001a045 t snd_ctl_elem_info [snd] > > > > > > > > > ffffffffa001a17a t copy_ctl_value_from_user [snd] > > > > > > > > > ffffffffa001a34f t snd_ctl_elem_read [snd] > > > > > > > > > ffffffffa001a413 t snd_ctl_elem_write [snd] > > > > > > > > > ffffffffa001a515 T snd_ctl_rename_id [snd] > > > > > > > > > ffffffffa001a5b5 T snd_ctl_activate_id [snd] > > > > > > > > > ffffffffa001a66e T snd_ctl_remove_id [snd] > > > > > > > > > ffffffffa001a6ed T snd_ctl_add [snd] > > > > > > > > > ffffffffa001a6ed u snd_ctl_add [snd_hda_codec] > > > > > > > > > ffffffffa001a947 t snd_ctl_elem_add [snd] > > > > > > > > > ffffffffa001abd3 t snd_ctl_elem_add_compat [snd] > > > > > > > > > ffffffffa001ad37 t snd_ctl_elem_add_user [snd] > > > > > > > > > ffffffffa001ada6 t snd_ctl_ioctl [snd] > > > > > > > > > > > > > > > > > > > > > > > > > > > $ cat /proc/kallsyms | sort | grep ffffffffa001b > > > > > > > > > ffffffffa001b4c4 t snd_ctl_ioctl_compat [snd] > > > > > > > > > ffffffffa001b9ba T snd_ctl_new1 [snd] > > > > > > > > > ffffffffa001b9ba u snd_ctl_new1 [snd_hda_codec] > > > > > > > > > ffffffffa001b9ba u snd_ctl_new1 [snd_hda_codec_cirrus] > > > > > > > > > ffffffffa001bb30 T snd_pci_quirk_lookup [snd] > > > > > > > > > ffffffffa001bb30 u snd_pci_quirk_lookup [snd_hda_codec] > > > > > > > > > ffffffffa001bb30 u snd_pci_quirk_lookup [snd_hda_intel] > > > > > > > > > ffffffffa001bb93 T snd_verbose_printk [snd] > > > > > > > > > ffffffffa001bb93 u snd_verbose_printk [snd_hda_codec] > > > > > > > > > ffffffffa001bb93 u snd_verbose_printk [snd_hda_intel] > > > > > > > > > ffffffffa001bb93 u snd_verbose_printk [snd_hwdep] > > > > > > > > > ffffffffa001bb93 u snd_verbose_printk [snd_pcm] > > > > > > > > > ffffffffa001bb93 u snd_verbose_printk [snd_timer] > > > > > > > > > ffffffffa001bc7b T release_and_free_resource [snd] > > > > > > > > > ffffffffa001bcd4 t snd_device_register_all [snd] > > > > > > > > > ffffffffa001bd76 T snd_device_register [snd] > > > > > > > > > ffffffffa001bd76 u snd_device_register [snd_pcm] > > > > > > > > > ffffffffa001be67 t snd_device_disconnect [snd] > > > > > > > > > ffffffffa001bf5d t snd_device_disconnect_all [snd] > > > > > > > > > > > > > > > > > > > > > > > > > > > hope that helps =) > > > > > > > > > > > > > > > > Maybe the patch below can give a better clue. > > > > > > > > This will spew stack traces when this error is triggered. > > > > > > > > Don't be surprised. > > > > > > > > > > > > > > oki - new output =) > > > > > > > > > > > > > > > > > > > > > [16306.365597] snd: bad kfree (called from ffffffffa001b08f) > > > > > > > [16306.365604] Pid: 26533, comm: alsamixer Tainted: P W > > > > > > > 2.6.30.1 #2 [16306.365610] Call Trace: > > > > > > > [16306.365632] [<ffffffffa001b08f>] ? snd_ctl_ioctl+0x2e5/0x71e > > > > > > > [snd] [16306.365645] [<ffffffff802b8bf9>] ? > > > > > > > __inc_zone_state+0x20/0x9b [16306.365655] [<ffffffff802cfffc>] ? > > > > > > > alloc_page_vma+0x197/0x1e1 [16306.365666] [<ffffffff802afb63>] ? > > > > > > > __lru_cache_add+0x93/0xdd [16306.365676] [<ffffffff802ea651>] ? > > > > > > > vfs_ioctl+0x35/0x97 > > > > > > > [16306.365686] [<ffffffff802eaae5>] ? do_vfs_ioctl+0x432/0x482 > > > > > > > [16306.365696] [<ffffffff806336bf>] ? > > > > > > > _spin_lock_irqsave+0x28/0x5c [16306.365705] [<ffffffff80247e22>] > > > > > > > ? do_page_fault+0x272/0x29e [16306.365715] [<ffffffff802eab82>] > > > > > > > ? sys_ioctl+0x4d/0x81 [16306.365726] [<ffffffff8022af2b>] ? > > > > > > > system_call_fastpath+0x16/0x1b > > > > > > > > > > > > Try the very latest alsa-driver-unstable snapshot. > > > > > > This bug should have been fixed, also possibly with CS420x parser > > > > > > bug, too. > > > > > > > > > > > > > > > > > > Takashi > > > > > > > > > > so dmesg knows the following: > > > > > > > > > > [18042.848549] HDA Intel 0000:00:08.0: PCI INT A -> Link[LAZA] -> GSI > > > > > 23 (level, low) -> IRQ 23 [18042.848623] HDA Intel 0000:00:08.0: > > > > > setting latency timer to 64 [18043.007568] ALSA > > > > > /home/nutz/tmp/alsa/alsa-driver-unstable/pci/hda/../../alsa-kernel/pc > > > > >i/hd a/hda_codec.c:3851: autoconfig: line_outs=1 (0xa/0x0/0x0/0x0/0x0) > > > > > [18043.007584] ALSA > > > > > /home/nutz/tmp/alsa/alsa-driver-unstable/pci/hda/../../alsa-kernel/pc > > > > >i/hd a/hda_codec.c:3855: speaker_outs=1 (0xb/0x0/0x0/0x0/0x0) > > > > > [18043.007596] ALSA > > > > > /home/nutz/tmp/alsa/alsa-driver-unstable/pci/hda/../../alsa-kernel/pc > > > > >i/hd a/hda_codec.c:3859: hp_outs=1 (0x9/0x0/0x0/0x0/0x0) > > > > > [18043.007608] ALSA > > > > > /home/nutz/tmp/alsa/alsa-driver-unstable/pci/hda/../../alsa-kernel/pc > > > > >i/hd a/hda_codec.c:3860: mono: mono_out=0x0 [18043.007618] ALSA > > > > > /home/nutz/tmp/alsa/alsa-driver-unstable/pci/hda/../../alsa-kernel/pc > > > > >i/hd a/hda_codec.c:3863: dig-out=0x10/0x15 [18043.007628] ALSA > > > > > /home/nutz/tmp/alsa/alsa-driver-unstable/pci/hda/../../alsa-kernel/pc > > > > >i/hd a/hda_codec.c:3871: inputs: mic=0xd, fmic=0x0, line=0xc, > > > > > fline=0x0, cd=0x0, aux=0x0 > > > > > [18043.007642] ALSA > > > > > /home/nutz/tmp/alsa/alsa-driver-unstable/pci/hda/../../alsa-kernel/pc > > > > >i/hd a/hda_codec.c:3873: dig-in=0x12 [18043.121187] ALSA > > > > > /home/nutz/tmp/alsa/alsa-driver-unstable/acore/control.c:337: control > > > > > 2:0:0:IEC958 Capture Switch:0 is already present [18043.121204] > > > > > hda_codec: cannot build controlsfor #0 (error -16) > > > > > > > > > > > > > > > and i don't have a mixer anymore ;) > > > > > > > > It's because your chip was uninitialized by BIOS or whatever, the > > > > another state you got sometimes. > > > > > > > > Now fixed in the driver to force to initialize pins. > > > > Try the latest tarball again. > > > > > > > > > > > > Takashi > > > > > > TAKASHIIIIIIIIIIIII =) > > > !!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!! > > > > > > digital out is working... i tried the new snapshot (eventhough, as you > > > said my subsystem is now again uninitialized) and i got back the > > > alsamixer- capabilities; > > > > > > i ran mplayer test.wav and started playing around with the > > > mixer-controls. > > > > > > when i reached IEC958 and turned it on, my out-jack began to glow red =) > > > so that works. i rushed over to my dorm-partners speakersystem and > > > plugged it in - voila that actually works! (i tried a DTS video with > > > mplayer -vc hwdts ) > > > > > > that is SO cool =) > > > > > > however as you might've figured out, analog out is not working, yet ;( > > > i tried the built-in speakers as well as the headphones (and of course > > > all the > > > > Hm, so you don't get any output although no errors with aplay? > > > > Please give alsa-info.sh output with the latest driver to be sure. > > > > > > Takashi > > and here is the alsa-info with the "correct" subsystem id. > > i also was able to get the wrong subsystem id again: i did > /etc/init.d/alsasound stop && modprobe snd-hda-intel - that changed it. > (i'll have a look into init.d/alsasound to see if there are any clues) Interesting. So, resetting the controller alone can change the SSID. It doesn't happen on other systems. Could you try to apply the patch below on alsa-kernel directory with -p2? This will disable the additional COEF setups, which I blindly copied from older cs4207 patch. thanks, Takashi --- diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index b1fd183..fda83b2 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -933,7 +933,7 @@ static void init_input(struct hda_codec *codec) change_cur_input(codec, spec->cur_input, 1); if (spec->mic_detect) cs_automic(codec); - +#if 0 coef = 0x000a; /* ADC1/2 - Digital and Analog Soft Ramp */ if (is_active_pin(codec, CS_DMIC2_PIN_NID)) coef |= 0x0500; /* DMIC2 enable 2 channels, disable GPIO1 */ @@ -943,6 +943,7 @@ static void init_input(struct hda_codec *codec) * IDX_SPDIF_CTL. */ cs_vendor_coef_set(codec, IDX_ADC_CFG, coef); +#endif } static struct hda_verb cs_coef_init_verbs[] = { @@ -980,10 +981,10 @@ static int cs_init(struct hda_codec *codec) { struct cs_spec *spec = codec->spec; - snd_hda_sequence_write(codec, cs_coef_init_verbs); + /*snd_hda_sequence_write(codec, cs_coef_init_verbs);*/ init_output(codec); init_input(codec); - init_digital(codec); + /*init_digital(codec);*/ return 0; }

1 0

Re: [alsa-devel] No sound with nVidia Corporation MCP79 High Definition Audio (rev b1) (Macbook Pro 5, 5)
by Andreas Nüßlein 06 Jul '09

06 Jul '09

On Monday 06 July 2009 15:39:09 you wrote: > At Mon, 6 Jul 2009 14:59:09 +0200, > > Andreas Nüßlein wrote: > > On Monday 06 July 2009 14:44:56 you wrote: > > > At Mon, 6 Jul 2009 12:19:11 +0200, > > > > > > Andreas Nüßlein wrote: > > > > On Monday 06 July 2009 12:08:11 you wrote: > > > > > At Mon, 6 Jul 2009 11:45:07 +0200, > > > > > > > > > > Andreas Nüßlein wrote: > > > > > > [1 <text/plain; iso-8859-1 (quoted-printable)>] > > > > > > > > > > > > On Monday 06 July 2009 09:23:07 you wrote: > > > > > > > At Sun, 5 Jul 2009 12:30:52 +0200, > > > > > > > > > > > > > > > however i was able to reproduce the "bad kfree", by changing > > > > > > > > the volumes in alsamixer. it always gave me the same "called > > > > > > > > from ID" so i tried "while (true) do cat /proc/kallsyms | > > > > > > > > grep ffffffffa001b08b; done" and then changed the volume, but > > > > > > > > the grep never returned anything, dmesg of course got more > > > > > > > > and more of those kfree-errors. > > > > > > > > > > > > > > The address won't match exactly in /proc/kasllsyms. > > > > > > > But you can guess which function by address. First sort > > > > > > > /proc/kallsyms then check the position that fits with the given > > > > > > > address. > > > > > > > > > > > > > > > > > > > > > Takashi > > > > > > > > > > > > well i'm not really sure _I_ can guess ;) i guess it's either > > > > > > "t snd_ctl_ioctl [snd]" or "t snd_ctl_ioctl_compat > > > > > > [snd]". > > > > > > > > > > > > here are the greps, "snd: bad kfree (called from > > > > > > ffffffffa001b08b)" being the error: > > > > > > > > > > > > > > > > > > $ cat /proc/kallsyms | sort | grep ffffffffa001a > > > > > > ffffffffa001a045 t snd_ctl_elem_info [snd] > > > > > > ffffffffa001a17a t copy_ctl_value_from_user [snd] > > > > > > ffffffffa001a34f t snd_ctl_elem_read [snd] > > > > > > ffffffffa001a413 t snd_ctl_elem_write [snd] > > > > > > ffffffffa001a515 T snd_ctl_rename_id [snd] > > > > > > ffffffffa001a5b5 T snd_ctl_activate_id [snd] > > > > > > ffffffffa001a66e T snd_ctl_remove_id [snd] > > > > > > ffffffffa001a6ed T snd_ctl_add [snd] > > > > > > ffffffffa001a6ed u snd_ctl_add [snd_hda_codec] > > > > > > ffffffffa001a947 t snd_ctl_elem_add [snd] > > > > > > ffffffffa001abd3 t snd_ctl_elem_add_compat [snd] > > > > > > ffffffffa001ad37 t snd_ctl_elem_add_user [snd] > > > > > > ffffffffa001ada6 t snd_ctl_ioctl [snd] > > > > > > > > > > > > > > > > > > $ cat /proc/kallsyms | sort | grep ffffffffa001b > > > > > > ffffffffa001b4c4 t snd_ctl_ioctl_compat [snd] > > > > > > ffffffffa001b9ba T snd_ctl_new1 [snd] > > > > > > ffffffffa001b9ba u snd_ctl_new1 [snd_hda_codec] > > > > > > ffffffffa001b9ba u snd_ctl_new1 [snd_hda_codec_cirrus] > > > > > > ffffffffa001bb30 T snd_pci_quirk_lookup [snd] > > > > > > ffffffffa001bb30 u snd_pci_quirk_lookup [snd_hda_codec] > > > > > > ffffffffa001bb30 u snd_pci_quirk_lookup [snd_hda_intel] > > > > > > ffffffffa001bb93 T snd_verbose_printk [snd] > > > > > > ffffffffa001bb93 u snd_verbose_printk [snd_hda_codec] > > > > > > ffffffffa001bb93 u snd_verbose_printk [snd_hda_intel] > > > > > > ffffffffa001bb93 u snd_verbose_printk [snd_hwdep] > > > > > > ffffffffa001bb93 u snd_verbose_printk [snd_pcm] > > > > > > ffffffffa001bb93 u snd_verbose_printk [snd_timer] > > > > > > ffffffffa001bc7b T release_and_free_resource [snd] > > > > > > ffffffffa001bcd4 t snd_device_register_all [snd] > > > > > > ffffffffa001bd76 T snd_device_register [snd] > > > > > > ffffffffa001bd76 u snd_device_register [snd_pcm] > > > > > > ffffffffa001be67 t snd_device_disconnect [snd] > > > > > > ffffffffa001bf5d t snd_device_disconnect_all [snd] > > > > > > > > > > > > > > > > > > hope that helps =) > > > > > > > > > > Maybe the patch below can give a better clue. > > > > > This will spew stack traces when this error is triggered. Don't be > > > > > surprised. > > > > > > > > oki - new output =) > > > > > > > > > > > > [16306.365597] snd: bad kfree (called from ffffffffa001b08f) > > > > [16306.365604] Pid: 26533, comm: alsamixer Tainted: P W > > > > 2.6.30.1 #2 [16306.365610] Call Trace: > > > > [16306.365632] [<ffffffffa001b08f>] ? snd_ctl_ioctl+0x2e5/0x71e > > > > [snd] [16306.365645] [<ffffffff802b8bf9>] ? > > > > __inc_zone_state+0x20/0x9b [16306.365655] [<ffffffff802cfffc>] ? > > > > alloc_page_vma+0x197/0x1e1 [16306.365666] [<ffffffff802afb63>] ? > > > > __lru_cache_add+0x93/0xdd [16306.365676] [<ffffffff802ea651>] ? > > > > vfs_ioctl+0x35/0x97 > > > > [16306.365686] [<ffffffff802eaae5>] ? do_vfs_ioctl+0x432/0x482 > > > > [16306.365696] [<ffffffff806336bf>] ? _spin_lock_irqsave+0x28/0x5c > > > > [16306.365705] [<ffffffff80247e22>] ? do_page_fault+0x272/0x29e > > > > [16306.365715] [<ffffffff802eab82>] ? sys_ioctl+0x4d/0x81 > > > > [16306.365726] [<ffffffff8022af2b>] ? system_call_fastpath+0x16/0x1b > > > > > > Try the very latest alsa-driver-unstable snapshot. > > > This bug should have been fixed, also possibly with CS420x parser > > > bug, too. > > > > > > > > > Takashi > > > > so dmesg knows the following: > > > > [18042.848549] HDA Intel 0000:00:08.0: PCI INT A -> Link[LAZA] -> GSI 23 > > (level, low) -> IRQ 23 [18042.848623] HDA Intel 0000:00:08.0: setting > > latency timer to 64 [18043.007568] ALSA > > /home/nutz/tmp/alsa/alsa-driver-unstable/pci/hda/../../alsa-kernel/pci/hd > >a/hda_codec.c:3851: autoconfig: line_outs=1 (0xa/0x0/0x0/0x0/0x0) > > [18043.007584] ALSA > > /home/nutz/tmp/alsa/alsa-driver-unstable/pci/hda/../../alsa-kernel/pci/hd > >a/hda_codec.c:3855: speaker_outs=1 (0xb/0x0/0x0/0x0/0x0) [18043.007596] > > ALSA > > /home/nutz/tmp/alsa/alsa-driver-unstable/pci/hda/../../alsa-kernel/pci/hd > >a/hda_codec.c:3859: hp_outs=1 (0x9/0x0/0x0/0x0/0x0) [18043.007608] ALSA > > /home/nutz/tmp/alsa/alsa-driver-unstable/pci/hda/../../alsa-kernel/pci/hd > >a/hda_codec.c:3860: mono: mono_out=0x0 [18043.007618] ALSA > > /home/nutz/tmp/alsa/alsa-driver-unstable/pci/hda/../../alsa-kernel/pci/hd > >a/hda_codec.c:3863: dig-out=0x10/0x15 [18043.007628] ALSA > > /home/nutz/tmp/alsa/alsa-driver-unstable/pci/hda/../../alsa-kernel/pci/hd > >a/hda_codec.c:3871: inputs: mic=0xd, fmic=0x0, line=0xc, fline=0x0, > > cd=0x0, aux=0x0 > > [18043.007642] ALSA > > /home/nutz/tmp/alsa/alsa-driver-unstable/pci/hda/../../alsa-kernel/pci/hd > >a/hda_codec.c:3873: dig-in=0x12 [18043.121187] ALSA > > /home/nutz/tmp/alsa/alsa-driver-unstable/acore/control.c:337: control > > 2:0:0:IEC958 Capture Switch:0 is already present [18043.121204] > > hda_codec: cannot build controlsfor #0 (error -16) > > > > > > and i don't have a mixer anymore ;) > > It's because your chip was uninitialized by BIOS or whatever, the > another state you got sometimes. > > Now fixed in the driver to force to initialize pins. > Try the latest tarball again. > > > Takashi TAKASHIIIIIIIIIIIII =) !!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!! digital out is working... i tried the new snapshot (eventhough, as you said my subsystem is now again uninitialized) and i got back the alsamixer- capabilities; i ran mplayer test.wav and started playing around with the mixer-controls. when i reached IEC958 and turned it on, my out-jack began to glow red =) so that works. i rushed over to my dorm-partners speakersystem and plugged it in - voila that actually works! (i tried a DTS video with mplayer -vc hwdts ) that is SO cool =) however as you might've figured out, analog out is not working, yet ;( i tried the built-in speakers as well as the headphones (and of course all the devices from aplay -L again) aplay -l shows 2 devices now though: $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: NVidia [HDA NVidia], device 0: Cirrus Analog [Cirrus Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 card 0: NVidia [HDA NVidia], device 1: Cirrus Digital [Cirrus Digital] Subdevices: 1/1 Subdevice #0: subdevice #1 should i try to reboot the machine to get back the other subsystem id? or is that irrelevant now? thanks so much =) andy

2 1

[alsa-devel] snd_pcm_info_alloca ( ptr) allocate an INVALID snd_pcm_info_t
by Guilherme Longo 06 Jul '09

06 Jul '09

Hi all! Mates, I was looking for a reasonable explanation on why the functions *_alloca as snd_pcm_info_alloca allocates an INVALID snd_pcm_info_t?? At my point of view this function reserves memory for the allocation of snd_pcm_info_t and so it is a real space reservation. I don't want get a wrong understanding and as long as there is no information regarding this, I am here asking! Thanks in advanced!

2 1

[alsa-devel] [RFC] Alsa mixer over serial port
by Janusz Krzysztofik 05 Jul '09

05 Jul '09

Hi, I'm currently working on ASoC driver for a voice modem codec chip (Conexant CX20442) available on ARM OMAP amsdelta board. The driver should provide some controls for switching audio path from handset to speakerphone and back, changing volume etc. The only way of controlling some aspects of the codec that I know of is over an acompanying voice modem chip (CX81801) with AT voice commands. The modem can be accessed over a platform serial device. I can see two ways of getting this working: 1) add a new line discipline that would take control over the serial line and convert it to an alsa mixer, 2) create a low level driver that would replace standard serial stuff completely (like snd-serial-u16550 does, for example). Please advise which one is better. Thanks, Janusz

2 2

[alsa-devel] [PATCH] ASoC: Remove use of hw_read from TLV320AIC3x driver
by Mark Brown 05 Jul '09

05 Jul '09

The TLV320AIC3x driver is currently the only user of the CODEC hw_read operation and is jumping through some hoops in order to do so. In order to support future refactoring to make the hw_read operation more usable unwrap the usage in this driver to avoid its use. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- I don't have any hardware I can test this on so it'd be good if someone with suitable hardware could give it a spin. Daniel, you wrote the code that does the readbacks so I guess you're most likely to have a good test setup for this? sound/soc/codecs/tlv320aic3x.c | 11 ++--------- 1 files changed, 2 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index ab099f4..0cf401f 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -145,8 +145,8 @@ static int aic3x_read(struct snd_soc_codec *codec, unsigned int reg, u8 *value) { *value = reg & 0xff; - if (codec->hw_read(codec->control_data, value, 1) != 1) - return -EIO; + + value[0] = i2c_smbus_read_byte_data(codec->control_data, value[0]); aic3x_write_reg_cache(codec, reg, *value); return 0; @@ -1316,12 +1316,6 @@ static struct i2c_driver aic3x_i2c_driver = { .id_table = aic3x_i2c_id, }; -static int aic3x_i2c_read(struct i2c_client *client, u8 *value, int len) -{ - value[0] = i2c_smbus_read_byte_data(client, value[0]); - return (len == 1); -} - static int aic3x_add_i2c_device(struct platform_device *pdev, const struct aic3x_setup_data *setup) { @@ -1394,7 +1388,6 @@ static int aic3x_probe(struct platform_device *pdev) #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) if (setup->i2c_address) { codec->hw_write = (hw_write_t) i2c_master_send; - codec->hw_read = (hw_read_t) aic3x_i2c_read; ret = aic3x_add_i2c_device(pdev, setup); } #else -- 1.6.3.3

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[alsa-devel] pulseaudio complains of bad pcm delay values
by Karl Kleinpaste 05 Jul '09

05 Jul '09

I'm sending this somewhat blindly, due to messages from the nightly log summary. I'm just cut/pasting what is there. It says "report this to alsa developers," but it would have been kinda cool if somebody would have included a definite address to which such a report should be sent. (First try, I sent this to alsa-devel@sourceforge. Got a bounce about the list having moved. Resending as indicated. No, I'm not subscribing to the list. Sorry, but I'm offering a bug report per request, not trying to engage in general devel conversation on this topic.) Hardware is a Toshiba Satellite P205 series, with this sound hardware: 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03) Subsystem: Toshiba America Info Systems Device ff00 Flags: bus master, fast devsel, latency 0, IRQ 22 Memory at f8500000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] MSI: Mask- 64bit+ Count=1/1 Enable- Capabilities: [70] Express Root Complex Integrated Endpoint, MSI 00 Capabilities: [100] Virtual Channel <?> Capabilities: [130] Root Complex Link <?> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel This is Fedora11, 2.6.29.5-191.fc11.i686.PAE (32-bit, obviously). Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: snd_pcm_delay() returned a value that is exceptionally large: 1480877280 bytes (8394995 ms). Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: snd_pcm_dump(): Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: Soft volume PCM Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: Control: PCM Playback Volume Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: min_dB: -51 Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: max_dB: 0 Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: resolution: 256 Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: Its setup is: Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: stream : CAPTURE Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: access : MMAP_INTERLEAVED Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: format : S16_LE Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: subformat : STD Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: channels : 2 Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: rate : 44100 Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: exact rate : 44100 (44100/1) Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: msbits : 16 Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: buffer_size : 88192 Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: period_size : 44096 Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: period_time : 999909 Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: tstamp_mode : ENABLE Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: period_step : 1 Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: avail_min : 87310 Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: period_event : 0 Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: start_threshold : -1 Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: stop_threshold : 1444937728 Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: silence_threshold: 0 Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: silence_size : 0 Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: boundary : 1444937728 Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: Slave: Hardware PCM card 0 'HDA Intel' device 0 subdevice 0 Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: Its setup is: Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: stream : CAPTURE Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: access : MMAP_INTERLEAVED Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: format : S16_LE Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: subformat : STD Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: channels : 2 Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: rate : 44100 Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: exact rate : 44100 (44100/1) Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: msbits : 16 Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: buffer_size : 88192 Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: period_size : 44096 Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: period_time : 999909 Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: tstamp_mode : ENABLE Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: period_step : 1 Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: avail_min : 87310 Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: period_event : 0 Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: start_threshold : -1 Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: stop_threshold : 1444937728 Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: silence_threshold: 0 Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: silence_size : 0 Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: boundary : 1444937728 Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: appl_ptr : 932488 Jul 4 09:20:35 awol pulseaudio[2835]: alsa-util.c: hw_ptr : 1444893632 Jul 4 09:20:40 awol pulseaudio[2835]: ratelimit.c: 1106 events suppressed Jul 4 09:20:45 awol pulseaudio[2835]: cpulimit.c: Received request to terminate due to CPU overload. Jul 4 09:20:47 awol pulseaudio[2835]: memblock.c: Memory pool destroyed but not all memory blocks freed! 236 remain. Jul 4 09:26:13 awol pulseaudio[17025]: alsa-sink.c: Increasing minimal latency to 1.00 ms Jul 4 09:26:13 awol pulseaudio[17025]: alsa-sink.c: Increasing minimal latency to 2.00 ms Jul 4 09:36:02 awol pulseaudio[17025]: alsa-source.c: Increasing minimal latency to 1.00 ms Jul 4 09:36:02 awol pulseaudio[17025]: alsa-source.c: Increasing minimal latency to 2.00 ms Jul 4 09:36:03 awol pulseaudio[17025]: alsa-source.c: Increasing minimal latency to 4.00 ms Jul 4 09:36:08 awol pulseaudio[17025]: alsa-source.c: Increasing minimal latency to 8.00 ms Jul 4 09:36:09 awol pulseaudio[17025]: alsa-source.c: Increasing minimal latency to 16.00 ms Jul 4 09:36:11 awol pulseaudio[17025]: alsa-source.c: Increasing minimal latency to 26.00 ms Jul 4 09:36:12 awol pulseaudio[17025]: alsa-source.c: Increasing wakeup watermark to 15.99 ms Jul 4 09:36:14 awol pulseaudio[17025]: alsa-source.c: Increasing minimal latency to 36.00 ms Jul 4 09:36:26 awol pulseaudio[17025]: alsa-source.c: Increasing wakeup watermark to 25.99 ms Jul 4 09:36:33 awol pulseaudio[17025]: alsa-source.c: Increasing minimal latency to 46.00 ms Jul 4 09:36:44 awol pulseaudio[17025]: alsa-source.c: Increasing wakeup watermark to 35.99 ms Jul 4 09:38:48 awol pulseaudio[17025]: alsa-source.c: Increasing minimal latency to 56.00 ms

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[alsa-devel] VIA 82C686B AC'97 controller initialization
by Gerhard Pircher 05 Jul '09

05 Jul '09

Hi, I'm trying to get AC'97 sound working on my AmigaOne PowerPC machine with a VIA 82C686B southbridge and a SigmaTel STAC9752 codec. The firmware doesn't initialize the AC'97 controller at all. Thus I had to implement some PCI quirks, which first setup the I/O BARs of the AC'97 PCI device and secondly power up the AC'97 controller by triggering a soft resume event through the southbridge's ACPI controller. This seems to work fine so far, but playback works in slow motion when I first power up the machine. After a warm reset audio playback sounds normal. Does somebody know how to fix this problem or were the problem could be? BTW can the snd-via82xx and snd-via82xx-modem drivers be used at the same time? Thanks Gerhard p.s.: Please put me on CC:, as I'm not subscribed to this mailing list. -- Neu: GMX Doppel-FLAT mit Internet-Flatrate + Telefon-Flatrate für nur 19,99 Euro/mtl.!* http://portal.gmx.net/de/go/dsl02

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[alsa-devel] [RFC] cmi8330: Add basic CMI8329 support
by Ondrej Zary 05 Jul '09

05 Jul '09

Add basic support for CMI8329 cards. It seems to be very similar to CMI8330, but the PnP devices are different: - AD1848 (WSS) and SB16 device IDs are swapped - the OPL3 port is contained in SB16 device, not AD1848 (WSS) - there is A@@0001 device (looks like some control interface - it has only one 8-bit I/O address) Is my detection code correct, shouldn't the struct pnp_dev returned pnp_request_card_device() be released somehow? Also the mixer seems to be a bit different - master volume does not work currently. My card includes wavetable MIDI - it works in Windows when MPU-401 is selected. It does not work with this driver - pmidi plays but no sound comes out. OPL3 works. My card has also an IDE interface but it's not visible as PnP device and it does not work even in Windows (and I'm unable to find any driver for it for any OS). Here's a scan of the card: http://www.rainbow-software.org/images/hardware/cmi8329a_.jpg The main chip is labelled AD Chips CMI8329A. --- linux-test/sound/isa/cmi8330.c 2009-06-27 16:32:13.000000000 +0200 +++ linux-2.6.30-pentium/sound/isa/cmi8330.c 2009-06-27 16:33:25.000000000 +0200 @@ -156,6 +156,11 @@ typedef int (*snd_pcm_open_callback_t)(struct snd_pcm_substream *); +enum card_type { + CMI8329, + CMI8330 +}; + struct snd_cmi8330 { #ifdef CONFIG_PNP struct pnp_dev *cap; @@ -172,6 +177,8 @@ snd_pcm_open_callback_t open; void *private_data; /* sb or wss */ } streams[2]; + + enum card_type type; }; #ifdef CONFIG_PNP @@ -328,12 +335,26 @@ { struct pnp_dev *pdev; int err; + const char *sb16_id, *wss_id; + + /* CMI8329 has a device with ID A@@0001, CMI8330 does not */ + pdev = pnp_request_card_device(card, "A@@0001", NULL); + if (pdev) { + acard->type = CMI8329; + pnp_release_card_device(pdev); + sb16_id = id->devs[0].id; + wss_id = id->devs[1].id; + } else { + acard->type = CMI8330; + sb16_id = id->devs[1].id; + wss_id = id->devs[0].id; + } - acard->cap = pnp_request_card_device(card, id->devs[0].id, NULL); + acard->cap = pnp_request_card_device(card, wss_id, NULL); if (acard->cap == NULL) return -EBUSY; - acard->play = pnp_request_card_device(card, id->devs[1].id, NULL); + acard->play = pnp_request_card_device(card, sb16_id, NULL); if (acard->play == NULL) return -EBUSY; @@ -351,7 +372,8 @@ wssport[dev] = pnp_port_start(pdev, 0); wssdma[dev] = pnp_dma(pdev, 0); wssirq[dev] = pnp_irq(pdev, 0); - fmport[dev] = pnp_port_start(pdev, 1); + if (acard->type == CMI8330) + fmport[dev] = pnp_port_start(pdev, 1); /* allocate SB16 resources */ pdev = acard->play; @@ -365,6 +387,8 @@ sbdma8[dev] = pnp_dma(pdev, 0); sbdma16[dev] = pnp_dma(pdev, 1); sbirq[dev] = pnp_irq(pdev, 0); + if (acard->type == CMI8329) + fmport[dev] = pnp_port_start(pdev, 1); /* allocate MPU-401 resources */ pdev = acard->mpu; -- Ondrej Zary

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Re: [alsa-devel] Laptop speakers nonfunctional, jacks functional
by Takashi Iwai 05 Jul '09

05 Jul '09

At Sat, 4 Jul 2009 22:16:18 +0200, Ludwig Maes wrote: > > but then, how come I have audio on line out and headphones? No idea. But this implies that you have very strange setup. Takashi > > thanks, > Ludwig > > 2009/7/4 Takashi Iwai <tiwai(a)suse.de> > > At Sat, 4 Jul 2009 19:15:23 +0200, > Ludwig Maes wrote: > > > > Thanks for your attention, > > I dont have /dev/snd/hwC0D0 > > but I do have > > /dev/snd/seq > > /dev/snd/timer > > /dev/sndstat > > I guess I should try it on seq, but I'll await your advice > > No, don't, that's wrong! > > It's an odd situation. I guess you didn't install the driver properly. > Otherwise you must have /dev/snd/control*, PCM, else. > > How to install the alsa-driver externally depends on the distro. > I'd advise you to ask on a distro forum about it at first. > The rest debugging starts after you can install the sound drivers > properly. > > thanks, > > Takashi > > > friendly greetings > > Ludwig > > > > 2009/6/30 Takashi Iwai <tiwai(a)suse.de> > > > > At Mon, 29 Jun 2009 20:06:47 +0200, > > Ludwig Maes wrote: > > > > > > Tried the snapshot, > > > still the same, alsa-info.txt attached > > > > Hm, just through a quick look at the output, there is no obvious > > problem. It's only the problem from speaker, not about the > headphone, > > right? > > > > If so, the problem is likely the missing amplifier setup, which is > > usually powered up via COEF verbs or GPIO verbs (ALC888/1200 have no > > EAPD verbs). > > > > Try to install hda-verb program and check the command below. > > > > # hda-verb /dev/snd/hwC0D0 0x20 SET_COEF_INDEX 7 > > # hda-verb /dev/snd/hwC0D0 0x20 GET_PROC_COEF 0 > > > > Which value do you get? What happens if you set > > > > # hda-verb /dev/snd/hwC0D0 0x20 SET_COEF_INDEX 7 > > # hda-verb /dev/snd/hwC0D0 0x20 SET_PROC_COEF 0x3060 > > or > > # hda-verb /dev/snd/hwC0D0 0x20 SET_COEF_INDEX 7 > > # hda-verb /dev/snd/hwC0D0 0x20 SET_PROC_COEF 0x3070 > > or > > # hda-verb /dev/snd/hwC0D0 0x20 SET_COEF_INDEX 7 > > # hda-verb /dev/snd/hwC0D0 0x20 SET_PROC_COEF 0x0830 > > or so? > > > > Takashi > > > > > >

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Re: [alsa-devel] Laptop speakers nonfunctional, jacks functional
by Takashi Iwai 04 Jul '09

04 Jul '09

At Sat, 4 Jul 2009 19:15:23 +0200, Ludwig Maes wrote: > > Thanks for your attention, > I dont have /dev/snd/hwC0D0 > but I do have > /dev/snd/seq > /dev/snd/timer > /dev/sndstat > I guess I should try it on seq, but I'll await your advice No, don't, that's wrong! It's an odd situation. I guess you didn't install the driver properly. Otherwise you must have /dev/snd/control*, PCM, else. How to install the alsa-driver externally depends on the distro. I'd advise you to ask on a distro forum about it at first. The rest debugging starts after you can install the sound drivers properly. thanks, Takashi > friendly greetings > Ludwig > > 2009/6/30 Takashi Iwai <tiwai(a)suse.de> > > At Mon, 29 Jun 2009 20:06:47 +0200, > Ludwig Maes wrote: > > > > Tried the snapshot, > > still the same, alsa-info.txt attached > > Hm, just through a quick look at the output, there is no obvious > problem. It's only the problem from speaker, not about the headphone, > right? > > If so, the problem is likely the missing amplifier setup, which is > usually powered up via COEF verbs or GPIO verbs (ALC888/1200 have no > EAPD verbs). > > Try to install hda-verb program and check the command below. > > # hda-verb /dev/snd/hwC0D0 0x20 SET_COEF_INDEX 7 > # hda-verb /dev/snd/hwC0D0 0x20 GET_PROC_COEF 0 > > Which value do you get? What happens if you set > > # hda-verb /dev/snd/hwC0D0 0x20 SET_COEF_INDEX 7 > # hda-verb /dev/snd/hwC0D0 0x20 SET_PROC_COEF 0x3060 > or > # hda-verb /dev/snd/hwC0D0 0x20 SET_COEF_INDEX 7 > # hda-verb /dev/snd/hwC0D0 0x20 SET_PROC_COEF 0x3070 > or > # hda-verb /dev/snd/hwC0D0 0x20 SET_COEF_INDEX 7 > # hda-verb /dev/snd/hwC0D0 0x20 SET_PROC_COEF 0x0830 > or so? > > Takashi > >

1 0

[alsa-devel] Help required with debugging
by Harsha, Priya 04 Jul '09

04 Jul '09

Hi, I see the following behavior and have a couple of questions that would help me debug my new driver/hardware. snd_pcm_lib_read1() - > snd_pcm_update_hw_ptr() ->ops->ack() This enables the driver to send data to hardware for capture snd_pcm_period_elapsed() -> hw_ptr_interrupt This updates the runtime->hw_ptr_jiffies. The jdelta calculated is proper and I find no issues with this. snd_pcm_lib_read1() is returned back Alsa calls driver's ack inbetween here snd_pcm_period_elapsed() -> snd_pcm_update_ hw_ptr_interrupt This updates the runtime->hw_ptr_jiffies. The jdelta calculated is proper and I find no issues with this. snd_pcm_lib_read1() - > snd_pcm_update_ hw_ptr() Here the runtime->hw_ptr_jiffies is updated and this causes my next jdelta to be very less in snd_pcm_period_elapsed() -> snd_pcm_update_hw_ptr_interrupt. Why is runtime->hw_ptr_jiffies updated in hw_ptr()? This should be only updated in the interrupt context right? Without this statement, my driver/hardware works fine. Or is there any constraint I can set somehow such that the hw_ptr_jiffies is only updated in the hw_ptr_interrupt function? Thanks, Harsha

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[alsa-devel] ESI Maya44 Driver
by Dany Svirsky 04 Jul '09

04 Jul '09

Hi All, This subject was disscused before, but it has been very long time without any updates. It been more then half a year since the patch is out. Was someone able to contact Rainer? If not what can be done to finally merge the patch? Is it possible for someone to take over the development of the driver? Thanks, Dany

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[alsa-devel] [PATCH] Allow opt-out from speex dependency
by Ingmar Vanhassel 04 Jul '09

04 Jul '09

--- configure.in | 8 +++++--- 1 files changed, 5 insertions(+), 3 deletions(-) This patch makes the --with-speex configure option work as intended, allowing to explicitly disable speex linking. diff --git a/configure.in b/configure.in index 36740e9..36d0d2a 100644 --- a/configure.in +++ b/configure.in @@ -106,14 +106,16 @@ AC_SUBST(AVCODEC_CFLAGS) AC_SUBST(AVCODEC_LIBS) AC_SUBST(AVCODEC_HEADER) -PKG_CHECK_MODULES(speexdsp, [speexdsp >= 1.2], [HAVE_SPEEXDSP="yes"], [HAVE_SPEEXDSP=""]) -AM_CONDITIONAL(HAVE_SPEEXDSP, test "$HAVE_SPEEXDSP" = "yes") - AC_ARG_WITH([speex], AS_HELP_STRING([--with-speex={builtin|lib|no}], [build speex resampler (built-in code, link with external lib, or no build)]), [PPH=$withval], [PPH="lib"]) +if test "x$with_speex" != "xno"; then + PKG_CHECK_MODULES(speexdsp, [speexdsp >= 1.2], [HAVE_SPEEXDSP="yes"], [HAVE_SPEEXDSP=""]) +fi +AM_CONDITIONAL(HAVE_SPEEXDSP, test "$HAVE_SPEEXDSP" = "yes") + USE_LIBSPEEX="" if test "$PPH" = "lib"; then if test "$HAVE_SPEEXDSP" = "yes"; then -- 1.6.3.3

2 1

[alsa-devel] [PATCH] ALSA: hda - move 8086:fb30 quirk (stac9205) to the proper section
by Herton Ronaldo Krzesinski 04 Jul '09

04 Jul '09

Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com.br> --- sound/pci/hda/patch_sigmatel.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 14f3c3e..826b1a0 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1590,8 +1590,6 @@ static struct snd_pci_quirk stac9200_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_REF), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0xfb30, - "SigmaTel",STAC_9205_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, "DFI LanParty", STAC_REF), /* Dell laptops have BIOS problem */ @@ -2344,6 +2342,8 @@ static struct snd_pci_quirk stac9205_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_9205_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0xfb30, + "SigmaTel", STAC_9205_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, "DFI LanParty", STAC_9205_REF), /* Dell */

2 1

[alsa-devel] soc_pcm_trigger
by ccku 04 Jul '09

04 Jul '09

Hi, in soc-core.c ,soc_pcm_trigger(), The function only calls three sub-functions codec_dai->ops->trigger platform->pcm_ops->trigger cpu_dai->ops->trigger why not give machine a chance to run its trigger?? like machine->ops->trigger. Thanks static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_dai_link *machine = rtd->dai; struct snd_soc_card *card = socdev->card; struct snd_soc_platform *platform = card->platform; struct snd_soc_dai *cpu_dai = machine->cpu_dai; struct snd_soc_dai *codec_dai = machine->codec_dai; struct snd_soc_codec *codec = card->codec; mutex_lock(&pcm_mutex); /* apply codec digital mute */ if (!codec->active) snd_soc_dai_digital_mute(codec_dai, 1); /* free any machine hw params */ if (machine->ops && machine->ops->hw_free) machine->ops->hw_free(substream); /* free any DMA resources */ if (platform->pcm_ops->hw_free) platform->pcm_ops->hw_free(substream); /* now free hw params for the DAI's */ if (codec_dai->ops->hw_free) codec_dai->ops->hw_free(substream, codec_dai); if (cpu_dai->ops->hw_free) cpu_dai->ops->hw_free(substream, cpu_dai); mutex_unlock(&pcm_mutex); return 0; }

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[alsa-devel] [PATCH v2] Sample generation on big endian platforms was broken.
by Kenneth Johansson 03 Jul '09

03 Jul '09

Has not worked since commit 3d1fa924906996463ac33cba5b5143f762d913cf Signed-off-by: Kenneth Johansson <kenneth(a)southpole.se> --- Fixes regression for 3 byte format in version 1. tested on both little and big endian platform test/pcm.c | 24 +++++++++++++----------- 1 files changed, 13 insertions(+), 11 deletions(-) diff --git a/test/pcm.c b/test/pcm.c index ee27422..0476c5f 100644 --- a/test/pcm.c +++ b/test/pcm.c @@ -34,14 +34,10 @@ static void generate_sine(const snd_pcm_channel_area_t *areas, static double max_phase = 2. * M_PI; double phase = *_phase; double step = max_phase*freq/(double)rate; - double res; - unsigned char *samples[channels], *tmp; + unsigned char *samples[channels]; int steps[channels]; - unsigned int chn, byte; - union { - int i; - unsigned char c[4]; - } ires; + unsigned int chn; + unsigned int maxval = (1 << (snd_pcm_format_width(format) - 1)) - 1; int bps = snd_pcm_format_width(format) / 8; /* bytes per sample */ @@ -61,12 +57,18 @@ static void generate_sine(const snd_pcm_channel_area_t *areas, } /* fill the channel areas */ while (count-- > 0) { + int res,i; res = sin(phase) * maxval; - ires.i = res; - tmp = ires.c; for (chn = 0; chn < channels; chn++) { - for (byte = 0; byte < (unsigned int)bps; byte++) - *(samples[chn] + byte) = tmp[byte]; + /* Generate data in native endian format */ + for ( i=0 ; i < bps ; i++ ){ +#if (__BYTE_ORDER == __BIG_ENDIAN) + *(samples[chn] + bps-1-i) = (res >> i*8) & 0xff; +#else + + *(samples[chn] + i) = (res >> i*8) & 0xff; +#endif + } samples[chn] += steps[chn]; } phase += step; -- 1.6.1.GIT

2 1

[alsa-devel] Conexant CX20582 HDA codec support
by Daniel Drake 03 Jul '09

03 Jul '09

Hi, I'm working on some sound issues with the new OLPC XO-1.5 alpha board. This laptop is now based on a VIA VX855 chipset and our board includes a CX20582-10Q HDA codec. We are running kernel v2.6.30. We have a few issues, which are: 1. Volume range is ridiculous, you can go all the way down to -74dB gain. Beyond -42dB you can barely hear anything from the speakers, and I note that all my other systems only go down as far as -46.5dB. 2. No microphone automute. The internal microphone continues recording sound while there is an external one plugged in. 3. No speaker automute. The internal speakers continue playing sound while there are headphones plugged in. 4. Mixer arrangement is odd. There are 3 microphones: Mic, Mic 1, and Mic 2. Mic (the default) records nothing. Mic 1 and Mic 2 both seem to behave the same, and record from both internal and external microphones. I looked into adding some quirks into patch_conexant.c to hopefully solve some of the above. However I note that there is no support for the CX20582, hence the generic HDA parser is being used. How has support for the other CX chips been developed -- are datasheets always required? (I'll try and get hold of one anyway) Here is some more info. codec dump http://dev.laptop.org/~dsd/20090629/codec0.txt codec graph http://dev.laptop.org/~dsd/20090629/codecgraph.svg lspci entry http://dev.laptop.org/~dsd/20090629/lspci-entry.txt verbose dmesg from probing http://dev.laptop.org/~dsd/20090629/dmesg-probe.txt Any thoughts/input appreciated! Thanks, Daniel

4 8

[alsa-devel] [PATCH] Sample generation on big endian platforms was broken.
by Kenneth Johansson 03 Jul '09

03 Jul '09

Has not worked since commit 3d1fa924906996463ac33cba5b5143f762d913cf Signed-off-by: Kenneth Johansson <kenneth(a)southpole.se> --- test/pcm.c | 24 ++++++++++++++---------- 1 files changed, 14 insertions(+), 10 deletions(-) diff --git a/test/pcm.c b/test/pcm.c index ee27422..9b8a923 100644 --- a/test/pcm.c +++ b/test/pcm.c @@ -34,14 +34,11 @@ static void generate_sine(const snd_pcm_channel_area_t *areas, static double max_phase = 2. * M_PI; double phase = *_phase; double step = max_phase*freq/(double)rate; - double res; + int res; unsigned char *samples[channels], *tmp; int steps[channels]; - unsigned int chn, byte; - union { - int i; - unsigned char c[4]; - } ires; + unsigned int chn; + unsigned int maxval = (1 << (snd_pcm_format_width(format) - 1)) - 1; int bps = snd_pcm_format_width(format) / 8; /* bytes per sample */ @@ -62,11 +59,18 @@ static void generate_sine(const snd_pcm_channel_area_t *areas, /* fill the channel areas */ while (count-- > 0) { res = sin(phase) * maxval; - ires.i = res; - tmp = ires.c; for (chn = 0; chn < channels; chn++) { - for (byte = 0; byte < (unsigned int)bps; byte++) - *(samples[chn] + byte) = tmp[byte]; + /* Generate data in native endian format */ + switch(bps){ + case 1: + *(samples[chn]) = res; + break; + case 2: + *(short*)(samples[chn]) = res; + break; + case 4: + *(int*)(samples[chn]) = res; + } samples[chn] += steps[chn]; } phase += step; -- 1.6.1.GIT

2 10

Re: [alsa-devel] MCP7A HDMI passthrough audio Linux/ALSA
by Stephane BERTHELOT 03 Jul '09

03 Jul '09

Takashi Iwai a écrit : > At Tue, 30 Jun 2009 15:04:05 +0200, > Stéphane BERTHELOT wrote: > >> Hi Takashi, >> >> Thanks for you answer. >> I have the exact same problem and configuration than reported here >> https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4406 >> apart : >> - I'm using a ION platform (Acer Revo R3600 but still MCP7A : 0x10de0007) >> - I never get the Dolby Digital or DTS indicator illuminated on my Amp. >> - My TV is a Sharp LC-52X20E >> >> I thought this was a known problem and than someone at Nvidia could >> provide the missing HDA Verbs, if you prefer I can post to the >> alsa-devel mailing list to investigate further. >> I can get a alsa-info output tonight, though it's very very similar to >> the one in the bug report. (and Raymond was saying that we should get >> "PCM AC3" on digital outputs) >> > > Try ac3dec for a primary testing instead of mplayer. It's included in > alsa-tools. > > % ac3dec -C -H foo.ac3 > > Use the latest one so that it accespt -H option. > > > Takashi > > > Thank you Takashi for taking time to review my problem. I cc'ed alsa-devel since I have more info now and maybe some other people have the same problem or already a workaround. I'm attaching the output of alsa-info.sh with my mixer settings and after having run ac3dec -H -C file.ac3 It works with -R instead of -C but does not work with -C or with -P (consumer or pro) I also tried modifying settings using iecset (audio off / audio on and consumer/pro) and no combination seems to work. When I play stereo LPCM my TV set plays sound and so does my Amp in PCM mode. When I try to play in AC3 mode (-C) my TV set says "incompatible format received" and both the TV and Amp stay silent. I may try again on XP/Vista tomorrow to be really sure it works on them. When changing play mode AES0 changes from 0x00 to 0x02 and AES1 to AES3 stay the same (0x82,0x00,0x02 if I remember well) Is there any other test you want me to do, I'll have time tomorrow evening (CEST). Thanks a lot, Stephane. (http://www.alsa-project.org/db/?f=cf3bc4202b4fcbf93ba68db55e492d51a3e46177) !!################################ !!ALSA Information Script v 0.4.56 !!################################ !!Script ran on: Tue Jun 30 20:22:23 UTC 2009 !!Linux Distribution !!------------------ Slackware 13.0.0.0.0 !!Kernel Information !!------------------ Kernel release: 2.6.29.5 Operating System: GNU/Linux Architecture: x86_64 Processor: Intel(R) Atom(TM) CPU 230 @ 1.60GHz SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.20 Library version: 1.0.18 Utilities version: 1.0.18 !!Loaded ALSA modules !!------------------- snd_hda_intel !!Sound Servers on this system !!---------------------------- ESound Daemon: Installed - Yes (/usr/bin/esd) Running - No !!Soundcards recognised by ALSA !!----------------------------- 0 [NVidia ]: HDA-Intel - HDA NVidia HDA NVidia at 0xfae78000 irq 22 !!PCI Soundcards installed in the system !!-------------------------------------- 00:08.0 Audio device: nVidia Corporation MCP79 High Definition Audio (rev b1) !!Advanced information - PCI Vendor/Device/Susbsystem ID's !!-------------------------------------------------------- 00:08.0 0403: 10de:0ac0 (rev b1) Subsystem: 1025:0222 !!Loaded sound module options !!-------------------------- !!Module: snd_hda_intel bdl_pos_adj : 32,-1,-1,-1,-1,-1,-1,-1 enable : Y,Y,Y,Y,Y,Y,Y,Y enable_msi : 0 id : <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> index : -1,-1,-1,-1,-1,-1,-1,-1 model : <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> position_fix : 0,0,0,0,0,0,0,0 power_save : 0 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1 probe_only : N,N,N,N,N,N,N,N single_cmd : N !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: Realtek ALC662 rev1 Address: 0 Function Id: 0x1 Vendor Id: 0x10ec0662 Subsystem Id: 0x10250222 Revision Id: 0x100101 No Modem Function Group found Default PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=2, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0x1d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x00 0x00] Converter: stream=0, channel=0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x1d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x00 0x00] Converter: stream=0, channel=0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Audio Output] wcaps 0x1d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x00 0x00] Converter: stream=0, channel=0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x05 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=0x09, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x89 0x89] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=0x09, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x00 0x00] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x22 Node 0x0a [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 9 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 Node 0x0c [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Connection: 2 0x02 0x0b Node 0x0d [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Connection: 2 0x03 0x0b Node 0x0e [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Connection: 2 0x04 0x0b Node 0x0f [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x10 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x12 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0001003c: IN OUT HP EAPD Detect EAPD 0x0: Pin Default 0x01214010: [Jack] HP Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0xc0: OUT HP Unsolicited: tag=00, enabled=0 Connection: 1 0x0c Node 0x15 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00010034: IN OUT EAPD Detect EAPD 0x0: Pin Default 0x99130120: [Fixed] Speaker at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0x2, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x0d Node 0x16 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00000034: IN OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Connection: 1 0x0e Node 0x17 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00001734: IN OUT Detect Vref caps: HIZ 50 GRD 80 Pin Default 0x01a19830: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x3, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Connection: 1 0x0e Node 0x19 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000173c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 2 0x0c* 0x0e Node 0x1a [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00000034: IN OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Connection: 1 0x0d Node 0x1b [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000173c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 2 0x0c* 0x0e Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0x40020601: [N/A] Line Out at Ext N/A Conn = 1/4, Color = Unknown DefAssociation = 0x0, Sequence = 0x1 Pin-ctls: 0x20: IN Node 0x1e [Pin Complex] wcaps 0x400300: Mono Digital Pincap 0x00000010: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x06 Node 0x1f [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=12 Processing Coefficient: 0x00 Coefficient Index: 0x06 Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 10 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x0b Node 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 10 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x0b Node 0x24 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x25 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x26 [Vendor Defined Widget] wcaps 0xf00000: Mono Codec: Nvidia MCP7A HDMI Address: 3 Function Id: 0x1 Vendor Id: 0x10de0007 Subsystem Id: 0x10de0101 Revision Id: 0x100100 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 Node 0x04 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Enabled Non-Audio Pro Digital category: 0x0 PCM: rates [0xc0]: 48000 88200 bits [0xf]: 8 16 20 24 formats [0x1]: PCM Node 0x05 [Pin Complex] wcaps 0x400381: Stereo Digital Pincap 0x00000014: OUT Detect Pin Default 0x18560110: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x04 Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0xc0]: 48000 88200 bits [0xf]: 8 16 20 24 formats [0x1]: PCM Node 0x07 [Pin Complex] wcaps 0x400381: Stereo Digital Pincap 0x00000014: OUT Detect Pin Default 0x58560121: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x2, Sequence = 0x1 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 1 0x06 Node 0x08 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0xc0]: 48000 88200 bits [0xf]: 8 16 20 24 formats [0x1]: PCM Node 0x09 [Pin Complex] wcaps 0x400381: Stereo Digital Pincap 0x00000014: OUT Detect Pin Default 0x58560122: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x2, Sequence = 0x2 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 1 0x08 Node 0x0a [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0xc0]: 48000 88200 bits [0xf]: 8 16 20 24 formats [0x1]: PCM Node 0x0b [Pin Complex] wcaps 0x400381: Stereo Digital Pincap 0x00000014: OUT Detect Pin Default 0x58560123: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x2, Sequence = 0x3 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 1 0x0a Node 0x0c [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0xc0]: 48000 88200 bits [0xf]: 8 16 20 24 formats [0x1]: PCM Node 0x0d [Pin Complex] wcaps 0x400381: Stereo Digital Pincap 0x00000014: OUT Detect Pin Default 0x58560124: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x2, Sequence = 0x4 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 1 0x0c --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw---- 1 root audio 116, 0 Jul 1 2009 /dev/snd/controlC0 crw-rw---- 1 root audio 116, 4 Jul 1 2009 /dev/snd/hwC0D0 crw-rw---- 1 root audio 116, 7 Jul 1 2009 /dev/snd/hwC0D3 crw-rw---- 1 root audio 116, 24 Jul 1 2009 /dev/snd/pcmC0D0c crw-rw---- 1 root audio 116, 16 Jun 30 22:11 /dev/snd/pcmC0D0p crw-rw---- 1 root audio 116, 19 Jun 30 22:21 /dev/snd/pcmC0D3p crw-rw---- 1 root audio 116, 1 Jun 30 22:10 /dev/snd/seq crw-rw---- 1 root audio 116, 33 Jul 1 2009 /dev/snd/timer !!ALSA configuration files !!------------------------ !!System wide config file (/etc/asound.conf) #pcm.!default { # type plug # slave.pcm "hdmi" #} pcm.hdmiconv { type plug slave { pcm "hw:0,3" rate 48000 } } pcm.headset { type bluetooth } ctl.headset { type bluetooth } !!Aplay/Arecord output !!------------ APLAY **** List of PLAYBACK Hardware Devices **** card 0: NVidia [HDA NVidia], device 0: ALC662 Analog [ALC662 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: NVidia [HDA NVidia], device 3: NVIDIA HDMI [NVIDIA HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: NVidia [HDA NVidia], device 0: ALC662 Analog [ALC662 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 0 [NVidia] Card hw:0 'NVidia'/'HDA NVidia at 0xfae78000 irq 22' Mixer name : 'Nvidia MCP7A HDMI' Components : 'HDA:10ec0662,10250222,00100101 HDA:10de0007,10de0101,00100100' Controls : 18 Simple ctrls : 9 Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 64 Mono: Playback 52 [81%] [-12.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 0 [0%] [-64.00dB] [on] Front Right: Playback 0 [0%] [-64.00dB] [on] Simple mixer control 'PCM',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 213 [84%] [-8.40dB] Front Right: Playback 213 [84%] [-8.40dB] Simple mixer control 'Front',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 52 [81%] [-12.00dB] [on] Front Right: Playback 52 [81%] [-12.00dB] [on] Simple mixer control 'Mic',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] Front Right: 0 [0%] Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 0 [0%] [-13.50dB] [on] Front Right: Capture 0 [0%] [-13.50dB] [on] Simple mixer control 'Beep',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [on] Front Right: Playback 0 [0%] [-34.50dB] [on] !!Alsactl output !!------------- --startcollapse-- state.NVidia { control.1 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 64' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'Front Playback Volume' value.0 52 value.1 52 } control.2 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Front Playback Switch' value.0 true value.1 true } control.3 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 64' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'Headphone Playback Volume' value.0 0 value.1 0 } control.4 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Headphone Playback Switch' value.0 true value.1 true } control.5 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'Mic Playback Volume' value.0 0 value.1 0 } control.6 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Mic Playback Switch' value.0 false value.1 false } control.7 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 3' comment.dbmin 0 comment.dbmax 3000 iface MIXER name 'Mic Boost' value.0 0 value.1 0 } control.8 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Capture Switch' value.0 true value.1 true } control.9 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -1350 comment.dbmax 3300 iface MIXER name 'Capture Volume' value.0 0 value.1 0 } control.10 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'Beep Playback Volume' value.0 0 value.1 0 } control.11 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Beep Playback Switch' value.0 true value.1 true } control.12 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 64' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'Master Playback Volume' value 52 } control.13 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Master Playback Switch' value true } control.14 { comment.access read comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Con Mask' value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.15 { comment.access read comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Pro Mask' value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.16 { comment.access 'read write' comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Default' value '0782000200000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.17 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'IEC958 Playback Switch' value true } control.18 { comment.access 'read write user' comment.type INTEGER comment.count 2 comment.range '0 - 255' comment.tlv '0000000100000008ffffec1400000014' comment.dbmin -5100 comment.dbmax 0 iface MIXER name 'PCM Playback Volume' value.0 213 value.1 213 } } --endcollapse-- !!All Loaded Modules !!------------------ Module snd_seq_dummy snd_seq_oss snd_seq_midi_event snd_seq snd_seq_device snd_pcm_oss snd_mixer_oss ipv6 ppdev lp parport_pc parport fuse snd_hda_codec_nvhdmi nvidia snd_hda_codec_realtek ath5k mac80211 rfkill led_class snd_hda_intel cfg80211 snd_hda_codec psmouse serio_raw shpchp snd_hwdep snd_pcm snd_timer sg snd soundcore forcedeth snd_page_alloc rtc_cmos rtc_core thermal wmi rtc_lib processor button thermal_sys hwmon evdev

2 5

[alsa-devel] [PATCH] [RFC 13/13] Intel SST Driver Config changes & makefile
by Vinod Koul 03 Jul '09

03 Jul '09

This patch adds the makefiles and Kconfig changes for both SST and MAD drivers Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> modified: sound/pci/Kconfig modified: sound/pci/Makefile new file: sound/pci/sst/Makefile --- sound/pci/Kconfig | 45 +++++++++++++++++++++++++++++++++++++++++++++ sound/pci/Makefile | 4 +++- sound/pci/sst/Makefile | 9 +++++++++ 3 files changed, 57 insertions(+), 1 deletions(-) create mode 100644 sound/pci/sst/Makefile diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 748f6b7..6901393 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -9,6 +9,51 @@ menuconfig SND_PCI if SND_PCI +config SND_INTEL_SST + bool "MRST SST DRIVER" + depends on X86 && LNW_IPC + default y + help + Say Y here to include support + +config SST_OSPM_SUPPORT + bool "MRST SST DRIVER SUPPORT for OSPM" + depends on SND_INTEL_SST && MSTWN_POWER_MGMT + default n + help + Say Y here to include support in SST driver for OSPM + +config SST_IPC_NOT_INCLUDED + bool "IPC NOT INCLUDED" + depends on SND_INTEL_SST + default y + help + Say Y here to include workaround in LPE driver for removing + IPC Driver calls + +config SST_DBG_PRINT + bool "MRST SST DRIVER DEBUG PRINTS" + depends on SND_INTEL_SST + default n + help + Say Y here to include debug prints for SST driver + +config SND_INTELMID + bool "MRST SOUND CARD DRIVER" + depends on SND_INTEL_SST && SPI_MRST + default y + select SND_MIXER_OSS + select SND_PCM_OSS + help + Say Y here to include support + +config MID_DBG_PRINT + bool "MRST MID DRIVER DEBUG PRINTS" + depends on SND_INTELMID + default n + help + Say Y here to include debug prints for MAD driver + config SND_AD1889 tristate "Analog Devices AD1889" select SND_AC97_CODEC diff --git a/sound/pci/Makefile b/sound/pci/Makefile index ecfc609..0aea1bc 100644 --- a/sound/pci/Makefile +++ b/sound/pci/Makefile @@ -77,4 +77,6 @@ obj-$(CONFIG_SND) += \ rme9652/ \ trident/ \ ymfpci/ \ - vx222/ + vx222/ \ + sst/ + diff --git a/sound/pci/sst/Makefile b/sound/pci/sst/Makefile new file mode 100644 index 0000000..e87dae5 --- /dev/null +++ b/sound/pci/sst/Makefile @@ -0,0 +1,9 @@ +# +# Makefile for Intel MID Audio drivers +# +EXTRA_CFLAGS=-g +snd-intel-sst-objs := intel_sst.o intel_sst_ipc.o intel_sst_stream.o intel_sst_interface.o +snd-intelmid-objs := intelmid.o intelmid_v0_control.o intelmid_v1_control.o intelmid_v2_control.o +# Toplevel Module Dependency +obj-$(CONFIG_SND_INTEL_SST) += snd-intel-sst.o +obj-$(CONFIG_SND_INTELMID) += snd-intelmid.o -- 1.5.4.5

2 1

Re: [alsa-devel] [PATCH] Enable loudspeakers in ALC889 for Acers and others
by ruben＠ugr.es 03 Jul '09

03 Jul '09

> At Thu, 2 Jul 2009 22:12:02 +0200 (CEST), > ruben(a)ugr.es wrote: >> >> > At Wed, 1 Jul 2009 20:04:36 +0200 (CEST), >> > ruben(a)ugr.es wrote: >> >> >> >> Hi all >> >> >> >> I managed to get sound working on my acer 6935, which has an ALC889 >> >> driven >> >> by the intel_hda driver >> >> The lack of sound seems to appear in other laptops as well. >> >> ... >> > Does this happen even with the latest alsa-driver snapshot? >> > ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver-sna… >> > >> > >> > Takashi >> > >> >> I installed the snapshot, and it has some good things and some bad >> things. >> --Good new things >> -The loudspeakers work correctly on boot. >> -The headphones jack sense works, and mutes the loudspeakers when I >> insert >> headphones. >> >> --Bad new things >> -The headphones don't provide any sound (regression with respect to >> 1.0.20), >> I put all the bars to maximum, but no luck. >> >> Do you think I can activate the headphones with hda-verb? > > Better run alsa-info.sh (with --no-upload option) on both 1.0.20 and > the latest snapshot, and attach the generated files. > > The regression is likely because of my latest change (merging alc882 and > alc883 parsers). Earlier is easier to fix :) > > > thanks, > > Takashi > Attached alsa info from yesterday's snapshot, todays, and 1.0.20 (vanilla, without my patch). The volume levels are not exactly the same; I can redo them if you want. Headphones are still muted in today's snapshot.

2 4

Re: [alsa-devel] No sound with nVidia Corporation MCP79 High Definition Audio (rev b1) (Macbook Pro 5, 5)
by Sean Burke 03 Jul '09

03 Jul '09

A quick google shows that the vendor ID is Cirrus Logic. The Boot Camp driver set does include installers for Cirrus audio cards. I don't have Windows installed on here, so I can't check to see if it is indeed the Cirrus drivers that get installed. What can be done to facilitate the implementation of this codec? Sean

3 10

[alsa-devel] [PATCH] ASoC: Fix wm8753 register cache size and initialization
by Lars-Peter Clausen 03 Jul '09

03 Jul '09

The register cache size is of by one. There are 63 registers in use but the register cache size has only space for 62. Furthermore the codec's reg_cache_size is of by another one. Since the wm8753 register cache uses one-based indexing we have to add one to its size. Register cache initialization only copied the first sizeof(void*) elements leaving the others uninitialized. Fix it by using the size of the reg cache template. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/wm8753.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index d28eeac..18a3ef9 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -79,7 +79,7 @@ static const u16 wm8753_reg[] = { 0x0097, 0x0097, 0x0000, 0x0004, 0x0000, 0x0083, 0x0024, 0x01ba, 0x0000, 0x0083, 0x0024, 0x01ba, - 0x0000, 0x0000 + 0x0000, 0x0000, 0x0000 }; /* codec private data */ @@ -1660,11 +1660,11 @@ static int wm8753_register(struct wm8753_priv *wm8753) codec->set_bias_level = wm8753_set_bias_level; codec->dai = wm8753_dai; codec->num_dai = 2; - codec->reg_cache_size = ARRAY_SIZE(wm8753->reg_cache); + codec->reg_cache_size = ARRAY_SIZE(wm8753->reg_cache) + 1; codec->reg_cache = &wm8753->reg_cache; codec->private_data = wm8753; - memcpy(codec->reg_cache, wm8753_reg, sizeof(codec->reg_cache)); + memcpy(codec->reg_cache, wm8753_reg, sizeof(wm8753_reg)); INIT_DELAYED_WORK(&codec->delayed_work, wm8753_work); ret = wm8753_reset(codec); -- 1.5.6.5

2 3

[alsa-devel] [PATCH] ASoC: TWL4030: Add tristate callbacks for HiFi and Voice
by Lopez Cruz, Misael 03 Jul '09

03 Jul '09

Add "set_tristate" callbacks for HiFi and Voice DAIs. Machine drivers can enable and disable tristate for each DAI with "snd_soc_dai_set_tristate" function. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl4030.c | 28 ++++++++++++++++++++++++++++ 1 files changed, 28 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 6da94ca..818fb37 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1849,6 +1849,19 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } +static int twl4030_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct snd_soc_codec *codec = dai->codec; + u8 reg = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF); + + if (tristate) + reg |= TWL4030_AIF_TRI_EN; + else + reg &= ~TWL4030_AIF_TRI_EN; + + return twl4030_write(codec, TWL4030_REG_AUDIO_IF, reg); +} + /* In case of voice mode, the RX1 L(VRX) for downlink and the TX2 L/R * (VTXL, VTXR) for uplink has to be enabled/disabled. */ static void twl4030_voice_enable(struct snd_soc_codec *codec, int direction, @@ -2023,6 +2036,19 @@ static int twl4030_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } +static int twl4030_voice_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct snd_soc_codec *codec = dai->codec; + u8 reg = twl4030_read_reg_cache(codec, TWL4030_REG_VOICE_IF); + + if (tristate) + reg |= TWL4030_VIF_TRI_EN; + else + reg &= ~TWL4030_VIF_TRI_EN; + + return twl4030_write(codec, TWL4030_REG_VOICE_IF, reg); +} + #define TWL4030_RATES (SNDRV_PCM_RATE_8000_48000) #define TWL4030_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE) @@ -2032,6 +2058,7 @@ static struct snd_soc_dai_ops twl4030_dai_ops = { .hw_params = twl4030_hw_params, .set_sysclk = twl4030_set_dai_sysclk, .set_fmt = twl4030_set_dai_fmt, + .set_tristate = twl4030_set_tristate, }; static struct snd_soc_dai_ops twl4030_dai_voice_ops = { @@ -2040,6 +2067,7 @@ static struct snd_soc_dai_ops twl4030_dai_voice_ops = { .hw_params = twl4030_voice_hw_params, .set_sysclk = twl4030_voice_set_dai_sysclk, .set_fmt = twl4030_voice_set_dai_fmt, + .set_tristate = twl4030_voice_set_tristate, }; struct snd_soc_dai twl4030_dai[] = { -- 1.5.4.3

3 2

[alsa-devel] [PATCH] ASoC: Zoom2: Update twl4030_setup_data parameters
by Candelaria Villareal, Jorge 03 Jul '09

03 Jul '09

Add support for EXTMUTE in Zoom2 machine driver. This is necessary to further reduce pop noise problem. Signal EXTMUTE is connected to signal GPIO 153 in Zoom2 board. In addition, change ramp delay value to 3 (218/161/109 ms). With previous ramp delay value, pop noise was louder. With a longer value the beep tone can be observed. Signed-off-by: Jorge Eduardo Candelaria <x0107209(a)ti.com> --- sound/soc/omap/zoom2.c | 17 +++++++++++++++-- 1 files changed, 15 insertions(+), 2 deletions(-) diff --git a/sound/soc/omap/zoom2.c b/sound/soc/omap/zoom2.c index be2e307..ec549a4 100644 --- a/sound/soc/omap/zoom2.c +++ b/sound/soc/omap/zoom2.c @@ -35,7 +35,8 @@ #include "omap-pcm.h" #include "../codecs/twl4030.h" -#define ZOOM2_HEADSET_MUX_GPIO (OMAP_MAX_GPIO_LINES + 15) +#define ZOOM2_HEADSET_MUX_GPIO (OMAP_MAX_GPIO_LINES + 15) +#define ZOOM2_HEADSET_EXTMUTE_GPIO 153 static int zoom2_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) @@ -234,10 +235,18 @@ static struct snd_soc_card snd_soc_zoom2 = { .num_links = ARRAY_SIZE(zoom2_dai), }; +/* EXTMUTE callback function */ +void zoom2_set_hs_extmute(int mute) +{ + gpio_set_value(ZOOM2_HEADSET_EXTMUTE_GPIO, mute); +} + /* twl4030 setup */ static struct twl4030_setup_data twl4030_setup = { - .ramp_delay_value = 2, /* 81 ms */ + .ramp_delay_value = 3, /* 161 ms */ .sysclk = 26000, + .hs_extmute = 1, + .set_hs_extmute = zoom2_set_hs_extmute, }; /* Audio subsystem */ @@ -277,6 +286,9 @@ static int __init zoom2_soc_init(void) BUG_ON(gpio_request(ZOOM2_HEADSET_MUX_GPIO, "hs_mux") < 0); gpio_direction_output(ZOOM2_HEADSET_MUX_GPIO, 0); + BUG_ON(gpio_request(ZOOM2_HEADSET_EXTMUTE_GPIO, "ext_mute") < 0); + gpio_direction_output(ZOOM2_HEADSET_EXTMUTE_GPIO, 0); + return 0; err1: @@ -290,6 +302,7 @@ module_init(zoom2_soc_init); static void __exit zoom2_soc_exit(void) { gpio_free(ZOOM2_HEADSET_MUX_GPIO); + gpio_free(ZOOM2_HEADSET_EXTMUTE_GPIO); platform_device_unregister(zoom2_snd_device); } -- 1.6.0.4

3 2

[alsa-devel] [PATCH 1/2] ASoC: Fix mpc5200-psc-ac97 to ensure the data ready bit is cleared
by Grant Likely 03 Jul '09

03 Jul '09

From: Grant Likely <grant.likely(a)secretlab.ca> When doing register reads, it is possible for there to be a stale data ready bit set which will cause subsequent reads to return prematurely with incorrect data. This patch fixes the issues by ensuring stale data is cleared before starting another transaction. Signed-off-by: Grant Likely <grant.likely(a)secretlab.ca> --- sound/soc/fsl/mpc5200_psc_ac97.c | 4 ++++ 1 files changed, 4 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c index 794a247..9b8503f 100644 --- a/sound/soc/fsl/mpc5200_psc_ac97.c +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -41,6 +41,10 @@ static unsigned short psc_ac97_read(struct snd_ac97 *ac97, unsigned short reg) pr_err("timeout on ac97 bus (rdy)\n"); return -ENODEV; } + + /* Force clear the data valid bit */ + in_be32(&psc_dma->psc_regs->ac97_data); + /* Send the read */ out_be32(&psc_dma->psc_regs->ac97_cmd, (1<<31) | ((reg & 0x7f) << 24));

4 6

[alsa-devel] [PATCH] ASoC: TWL4030: Add tristate callbacks for HiFi and Voice
by Lopez Cruz, Misael 03 Jul '09

03 Jul '09

Add "set_tristate" callbacks for HiFi and Voice DAIs. Machine drivers can enable and disable tristate for each DAI with "snd_soc_dai_set_tristate" function provided by SoC core. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl4030.c | 28 ++++++++++++++++++++++++++++ 1 files changed, 28 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 6da94ca..ffeb53f 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1849,6 +1849,19 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } +static int twl4030_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct snd_soc_codec *codec = dai->codec; + u8 reg = twl4030_read_reg_cache(codec, TWL4030_REG_VOICE_IF); + + if (tristate) + reg |= TWL4030_AIF_TRI_EN; + else + reg &= ~TWL4030_AIF_TRI_EN; + + return twl4030_write(codec, TWL4030_REG_AUDIO_IF, reg); +} + /* In case of voice mode, the RX1 L(VRX) for downlink and the TX2 L/R * (VTXL, VTXR) for uplink has to be enabled/disabled. */ static void twl4030_voice_enable(struct snd_soc_codec *codec, int direction, @@ -2023,6 +2036,19 @@ static int twl4030_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } +static int twl4030_voice_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct snd_soc_codec *codec = dai->codec; + u8 reg = twl4030_read_reg_cache(codec, TWL4030_REG_VOICE_IF); + + if (tristate) + reg |= TWL4030_VIF_TRI_EN; + else + reg &= ~TWL4030_VIF_TRI_EN; + + return twl4030_write(codec, TWL4030_REG_VOICE_IF, reg); +} + #define TWL4030_RATES (SNDRV_PCM_RATE_8000_48000) #define TWL4030_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE) @@ -2032,6 +2058,7 @@ static struct snd_soc_dai_ops twl4030_dai_ops = { .hw_params = twl4030_hw_params, .set_sysclk = twl4030_set_dai_sysclk, .set_fmt = twl4030_set_dai_fmt, + .set_tristate = twl4030_set_tristate, }; static struct snd_soc_dai_ops twl4030_dai_voice_ops = { @@ -2040,6 +2067,7 @@ static struct snd_soc_dai_ops twl4030_dai_voice_ops = { .hw_params = twl4030_voice_hw_params, .set_sysclk = twl4030_voice_set_dai_sysclk, .set_fmt = twl4030_voice_set_dai_fmt, + .set_tristate = twl4030_voice_set_tristate, }; struct snd_soc_dai twl4030_dai[] = { -- 1.5.4.3

2 2

[alsa-devel] [PATCH] [RFC 7/13] Intel SST driver stream module
by Vinod Koul 03 Jul '09

03 Jul '09

This adds the stream module which contains the function for stream operations & control. For a stream the control and data are two major parts. This module implements the control (play/pause/resume/stop/free/alloc) for a stream. It also implements data play/capture frames where buffers are sent/received to Firmware. The objective of SST driver is to achieve Low power playback by utilizing DSP as much as possible. So SST gets large music buffers from player/middleware and sends them to firmware in a scatter gather list. The firmware decodes and renders them, while IA can goto low power states Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> Signed-off-by: R Dharageswari <dharageswari.r(a)intel.com> new file: sound/pci/sst/intel_sst_stream.c --- sound/pci/sst/intel_sst_stream.c | 711 ++++++++++++++++++++++++++++++++++++++ 1 files changed, 711 insertions(+), 0 deletions(-) create mode 100644 sound/pci/sst/intel_sst_stream.c diff --git a/sound/pci/sst/intel_sst_stream.c b/sound/pci/sst/intel_sst_stream.c new file mode 100644 index 0000000..aabf057 --- /dev/null +++ b/sound/pci/sst/intel_sst_stream.c @@ -0,0 +1,711 @@ +/* + * intel_sst_stream.c - Intel SST Driver for audio engine + * + * Copyright (C) 2008-09 Intel Corp + * Authors: Vinod Koul <vinod.koul(a)intel.com> + * Harsha Priya <priya.harsha(a)intel.com> + * R Dharageswari <dharageswari.r(a)intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * 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; version 2 of the License. + * + * 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. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This file contains the stream operations of SST driver + */ + +#include <linux/cdev.h> +#include <linux/pci.h> +#include <linux/kernel.h> +#include <linux/syscalls.h> +#include <linux/file.h> +#include <linux/interrupt.h> +#include <linux/list.h> +#include <linux/uaccess.h> +#include <linux/firmware.h> +#include <sound/intel_sst_ioctl.h> +#include "intel_sst_fw_ipc.h" +#include <sound/intel_sst.h> +#include "intel_sst_common.h" +#include "intel_sst_pvt.h" +#ifdef CONFIG_SST_OSPM_SUPPORT +#include <linux/intel_mid.h> +#endif + +/** +* sst_alloc_stream - Send msg for a new stream ID +* @params: stream params +* @stream_ops: operation of stream PB/capture +* @codec: codec for stream +* @session_id: pvt_id passed by MMF to distinguish stream +* +* This function is called by any function which wants to start +* a new stream. This also check if a stream exists which is idle +* it initializes idle stream id to this request +*/ +int sst_alloc_stream(char *params, int stream_ops, int codec, int session_id) +{ + struct ipc_post *msg = NULL; + struct snd_sst_alloc_params alloc_param = {{0,},}; + + sst_dbg("entering sst_alloc_stream \n"); + sst_dbg("%d %d %d\n", stream_ops, codec, session_id); + + BUG_ON(!params); + + /*send msg to FW to allocate a stream*/ + if (sst_create_large_msg(&msg) != 0) + return -ENOMEM; + + sst_fill_header(&msg->header, IPC_IA_ALLOC_STREAM, 1, 0); + msg->header.part.data = sizeof(alloc_param) + sizeof(u32); + alloc_param.str_type.codec_type = codec; + alloc_param.str_type.str_type = 2; /*music*/ + alloc_param.str_type.operation = stream_ops; + alloc_param.str_type.protected_str = 0; /*non drm*/ + alloc_param.str_type.pvt_id = session_id; + memcpy(&alloc_param.stream_params, params, + sizeof(struct snd_sst_stream_params)); + + memcpy(msg->mailbox_data, &msg->header, sizeof(u32)); + memcpy(msg->mailbox_data + sizeof(u32), &alloc_param, + sizeof(alloc_param)); + + list_add_tail(&msg->node, &sst_ops->ipc_dispatch_list); + sst_post_message(&sst_ops->ipc_post_msg_wq); + sst_dbg("alloc stream done\n"); + return 0; +} + +/** +* sst_alloc_stream_response - process alloc reply +* @alloc_header: respose header +* +* This function is called by process_reply when reply from FW comes +* based on response has to process further +*/ +int sst_alloc_stream_response(unsigned int str_id, + struct snd_sst_str_type *type) +{ + int retval = 0, i, valid_str = 0; + struct ipc_post *msg = NULL; + + /*allocation succesfull*/ + sst_dbg("stream number given = %d \n", str_id); + + for (i = 0; i < MAX_STREAM_FIELD; i++) { + if (type->pvt_id == sst_ops->alloc_block[i].sst_id) { + valid_str = 1; + break; + } + } + if (valid_str == 0) { + /*this is not valid stream*/ + sst_err("Invalid stream allocation detetcted... freeing\n"); + if (sst_create_short_msg(&msg) != 0) + return -ENOMEM; + sst_fill_header(&msg->header, IPC_IA_FREE_STREAM, 0, str_id); + list_add_tail(&msg->node, &sst_ops->ipc_dispatch_list); + sst_post_message(&sst_ops->ipc_post_msg_wq); + return 0; + } + + sst_init_stream(&sst_ops->streams[str_id], type->codec_type, + type->pvt_id, type->operation); + + sst_dbg("stream pvt id = %d \n", type->pvt_id); + + /*Unblock with retval code*/ + sst_wake_up_alloc_block(sst_ops, type->pvt_id, str_id, NULL); + return retval; +} + +/** +* sst_pause_stream - Send msg for a pausing stream +* @id: stream ID +* +* This function is called by any function which wants to pause +* an already running stream. +*/ +int sst_pause_stream(int str_id) +{ + int retval = 0; + struct ipc_post *msg = NULL; + struct stream_info *str_info = NULL; + + sst_dbg("sst_pause_stream for %d\n", str_id); + retval = sst_validate_strid(str_id); + if (retval != 0) + return retval; + str_info = &sst_ops->streams[str_id]; + if (STREAM_PAUSED == str_info->status) + return 0; + + if (STREAM_RUNNING == str_info->status || + STREAM_INIT == str_info->status) { + + if (str_info->prev == STREAM_UN_INIT) + return -EBADRQC; + + mutex_lock(&str_info->lock); + if (sst_create_short_msg(&msg) != 0) { + mutex_unlock(&str_info->lock); + return -ENOMEM; + } + sst_fill_header(&msg->header, IPC_IA_PAUSE_STREAM, 0, str_id); + list_add_tail(&msg->node, &sst_ops->ipc_dispatch_list); + mutex_unlock(&str_info->lock); + sst_post_message(&sst_ops->ipc_post_msg_wq); + str_info->ctrl_blk.condition = false; + str_info->ctrl_blk.ret_code = 0; + str_info->ctrl_blk.on = true; + retval = sst_wait_interruptible_timeout(sst_ops, + &str_info->ctrl_blk, SST_BLOCK_TIMEOUT); + if (retval == 0) { + str_info->prev = str_info->status; + str_info->status = STREAM_PAUSED; +#ifdef CONFIG_SST_OSPM_SUPPORT + sst_ospm_send_event(OSPM_EVENT_SUBSYS_STOP_PLAY); +#endif + } else if (retval == SST_ERR_INVALID_STREAM_ID) { + retval = EINVAL; + sst_clean_stream(str_info); + } + } else { + retval = -EBADRQC; + sst_err("BADQRC for stream\n"); + } + + return retval; +} + +/** +* sst_resume_stream - Send msg for resuming stream +* @id: stream ID +* +* This function is called by any function which wants to resume +* an already paused stream. +*/ +int sst_resume_stream(int str_id) +{ + int retval = 0; + struct ipc_post *msg = NULL; + struct stream_info *str_info = NULL; + + sst_dbg("sst_resume_stream for %d\n", str_id); + retval = sst_validate_strid(str_id); + if (retval != 0) + return retval; + str_info = &sst_ops->streams[str_id]; + if (STREAM_RUNNING == str_info->status) + return 0; + if (STREAM_PAUSED == str_info->status) { + mutex_lock(&str_info->lock); + if (sst_create_short_msg(&msg) != 0) { + mutex_unlock(&str_info->lock); + return -ENOMEM; + } + sst_fill_header(&msg->header, IPC_IA_RESUME_STREAM, 0, str_id); + list_add_tail(&msg->node, &sst_ops->ipc_dispatch_list); + mutex_unlock(&str_info->lock); + sst_post_message(&sst_ops->ipc_post_msg_wq); + str_info->ctrl_blk.condition = false; + str_info->ctrl_blk.ret_code = 0; + str_info->ctrl_blk.on = true; + retval = sst_wait_interruptible_timeout(sst_ops, + &str_info->ctrl_blk, SST_BLOCK_TIMEOUT); + if (retval == 0) { + if (STREAM_RUNNING == str_info->prev) + str_info->status = STREAM_RUNNING; + else + str_info->status = STREAM_INIT; + str_info->prev = STREAM_PAUSED; +#ifdef CONFIG_SST_OSPM_SUPPORT + sst_ospm_send_event(OSPM_EVENT_SUBSYS_START_PLAY); +#endif + } else if (retval == SST_ERR_INVALID_STREAM_ID) { + retval = EINVAL; + sst_clean_stream(str_info); + } + } else { + retval = -EBADRQC; + sst_err("BADQRC for stream\n"); + } + + return retval; +} + + +/** +* sst_drop_stream - Send msg for stopping stream +* @id: stream ID +* +* This function is called by any function which wants to stop +* a stream. +*/ +int sst_drop_stream(int str_id) +{ + int retval = 0; + struct ipc_post *msg = NULL; + struct sst_stream_bufs *bufs = NULL, *_bufs = NULL; + struct stream_info *str_info = NULL; + + sst_dbg("sst_drop_stream for %d\n", str_id); + retval = sst_validate_strid(str_id); + if (retval != 0) + return retval; + str_info = &sst_ops->streams[str_id]; + if (STREAM_UN_INIT != str_info->status) { + + mutex_lock(&str_info->lock); + + if (sst_create_short_msg(&msg) != 0) { + mutex_unlock(&str_info->lock); + return -ENOMEM; + } + sst_fill_header(&msg->header, IPC_IA_DROP_STREAM, 0, str_id); + list_add_tail(&msg->node, &sst_ops->ipc_dispatch_list); + mutex_unlock(&sst_ops->streams[str_id].lock); + sst_post_message(&sst_ops->ipc_post_msg_wq); + str_info->ctrl_blk.condition = false; + str_info->ctrl_blk.ret_code = 0; + str_info->ctrl_blk.on = true; + retval = sst_wait_interruptible_timeout(sst_ops, + &str_info->ctrl_blk, SST_BLOCK_TIMEOUT); + if (retval == 0) { + str_info->prev = str_info->status; + str_info->status = STREAM_INIT; + if (str_info->data_blk.on == true) { + /*moved codec specfic check to generic check*/ + str_info->data_blk.condition = true; + str_info->data_blk.ret_code = 0; + wake_up(&sst_ops->wait_queue); + } + list_for_each_entry_safe(bufs, _bufs, + &str_info->bufs, node) { + list_del(&bufs->node); + kfree(bufs); + } +#ifdef CONFIG_SST_OSPM_SUPPORT + sst_ospm_send_event(OSPM_EVENT_SUBSYS_STOP_PLAY); +#endif + } else if (retval == SST_ERR_INVALID_STREAM_ID) { + retval = EINVAL; + sst_clean_stream(str_info); + } + } else { + retval = -EBADRQC; + sst_err("BADQRC for stream\n"); + } + + return retval; +} + +/** +* sst_free_stream - Frees a stream +* @id: stream ID +* +* This function is called by any function which wants to free +* a stream. +*/ +int sst_free_stream(int str_id) +{ + int retval = 0; + struct ipc_post *msg = NULL; + struct stream_info *str_info = NULL; + + sst_dbg("sst_free_stream for %d\n", str_id); + + retval = sst_validate_strid(str_id); + if (0 != retval) + return retval; + str_info = &sst_ops->streams[str_id]; + + if (STREAM_UN_INIT != str_info->status) { + mutex_lock(&str_info->lock); + if (sst_create_short_msg(&msg) != 0) { + mutex_unlock(&str_info->lock); + return -ENOMEM; + } + sst_fill_header(&msg->header, IPC_IA_FREE_STREAM, 0, str_id); + list_add_tail(&msg->node, &sst_ops->ipc_dispatch_list); + mutex_unlock(&str_info->lock); + sst_post_message(&sst_ops->ipc_post_msg_wq); + str_info->prev = str_info->status; + str_info->status = STREAM_UN_INIT; + if (str_info->data_blk.on == true) { + str_info->data_blk.condition = true; + str_info->data_blk.ret_code = 0; + wake_up(&sst_ops->wait_queue); + } + sst_clean_stream(str_info); + sst_dbg("Stream freed\n"); + +#ifdef CONFIG_SST_OSPM_SUPPORT + sst_ospm_send_event(OSPM_EVENT_SUBSYS_STOP_PLAY); +#endif + } else { + retval = -EBADRQC; + sst_dbg("BADQRC for stream\n"); + } + + return retval; +} + + +/** +* sst_set_stream_param - Send msg for setting stream +* @id: stream id +* @params: stream params +* +* This function sets stream params during runtime +*/ +int sst_set_stream_param(int str_id, struct snd_sst_params *str_param) +{ + int retval = 0; + struct ipc_post *msg = NULL; + struct stream_info *str_info = NULL; + + BUG_ON(!str_param); + retval = sst_validate_strid(str_id); + if (retval != 0) + return retval; + sst_dbg("set_stream for %d\n", str_id); + if (STREAM_INIT == sst_ops->streams[str_id].status) { + mutex_lock(&sst_ops->streams[str_id].lock); + + if (sst_create_large_msg(&msg) != 0) { + mutex_unlock(&sst_ops->streams[str_id].lock); + return -ENOMEM; + } + + sst_fill_header(&msg->header, + IPC_IA_SET_STREAM_PARAMS, 1, str_id); + msg->header.part.data = sizeof(u32) + + sizeof(str_param->sparams); + memcpy(msg->mailbox_data, &msg->header, sizeof(u32)); + memcpy(msg->mailbox_data + sizeof(u32), &str_param->sparams, + sizeof(str_param->sparams)); + list_add_tail(&msg->node, &sst_ops->ipc_dispatch_list); + sst_post_message(&sst_ops->ipc_post_msg_wq); + mutex_unlock(&sst_ops->streams[str_id].lock); + str_info->ctrl_blk.condition = false; + str_info->ctrl_blk.ret_code = 0; + str_info->ctrl_blk.on = true; + retval = sst_wait_interruptible_timeout(sst_ops, + &str_info->ctrl_blk, SST_BLOCK_TIMEOUT); + if (retval != 0) { + retval = EINVAL; + sst_free_stream(str_id); + } else if (retval == SST_ERR_INVALID_STREAM_ID) { + retval = EINVAL; + sst_clean_stream(str_info); + } + } else { + retval = -EBADRQC; + sst_err("BADQRC for stream\n"); + } + return retval; +} + +/** +* sst_route_control - To start or drop the routing from once device to another. +* @id: stream id +* +* This function gets stream params during runtime +*/ +/* +int sst_route_control(int str_id) +{ + int retval = 0; + struct ipc_post *msg = NULL; + struct snd_sst_control_routing *route = {0,}; + struct stream_info *str_info = NULL; + sst_dbg("sst_route_control for %d\n", str_id); + retval = sst_validate_strid(str_id); + route->control = 0; + if (0 != retval) + return retval; + str_info = &sst_ops->streams[str_id]; + mutex_lock(&str_info->lock); + if (sst_create_large_msg(&msg) != 0) { + mutex_unlock(&str_info->lock); + return -ENOMEM; + } + sst_fill_header(&msg->header, IPC_IA_CONTROL_ROUTING, 1, str_id); + msg->header.part.data = sizeof(u32) + sizeof(*route); + memcpy(msg->mailbox_data, &msg->header, sizeof(u32)); + memcpy(msg->mailbox_data + sizeof(u32), route, + sizeof(*route)); + + + list_add_tail(&msg->node, &sst_ops->ipc_dispatch_list); + mutex_unlock(&str_info->lock); + sst_post_message(&sst_ops->ipc_post_msg_wq); + + str_info->ctrl_blk.condition = false; + str_info->ctrl_blk.ret_code = 0; + str_info->ctrl_blk.on = true; + retval = sst_wait_interruptible_timeout(sst_ops, + &str_info->ctrl_blk, SST_BLOCK_TIMEOUT); + if (retval == 0) + return retval; + else if (retval == SST_ERR_INVALID_STREAM_ID) { + retval = EINVAL; + sst_clean_stream(str_info); + } + return retval; + +}*/ + +/** +* sst_target_device_select - This fuction allows to select the target device +* for the playback or capture +* @id: stream id +* +* This function gets stream params and the target device during runtime +*/ + +int sst_target_device_select(struct snd_sst_target_device *target_device) +{ +/* int retval = 0; + struct ipc_post *msg = NULL; + struct snd_sst_slot_info *slot_info = NULL; + struct stream_info *str_info = NULL; +*/ sst_dbg("stream_control "); + return 0; +#if 0 + if (sst_create_large_msg(&msg) != 0) { + mutex_unlock(&str_info->lock); + return -ENOMEM; + } + sst_fill_header(&msg->header, IPC_IA_TARGET_DEV_SELECT, + 1, str_id); + msg->header.part.data = sizeof(u32) + sizeof(*target_device); + memcpy(msg->mailbox_data, &msg->header, sizeof(u32)); + memcpy(msg->mailbox_data + sizeof(u32), target_device, + sizeof(*target_device)); + list_add_tail(&msg->node, &sst_ops->ipc_dispatch_list); + mutex_unlock(&str_info->lock); + sst_post_message(&sst_ops->ipc_post_msg_wq); + + str_info->ctrl_blk.condition = false; + str_info->ctrl_blk.ret_code = 0; + str_info->ctrl_blk.on = true; + retval = sst_wait_interruptible_timeout(sst_ops, + &str_info->ctrl_blk, SST_BLOCK_TIMEOUT); + if (retval == SST_ERR_INVALID_STREAM_ID) { + retval = -EINVAL; + sst_clean_stream(str_info); + } + if (target_device->device_route < 0 || + target_device->device_route > 2) + return -EINVAL; + slot_info = &target_device->devices[0]; + + if (slot_info->type == SND_SST_TARGET_PMIC) + retval = sst_set_pmic_config(str_id, &slot_info->pcm_params); + if (target_device->device_route == 2) { + slot_info = &target_device->devices[1]; + if (target_device->devices[1].type == SND_SST_TARGET_PMIC) + retval = sst_set_pmic_config(str_id, + &slot_info->pcm_params); + } + if (retval >= 0) + retval = sst_route_control(str_id); + return retval; +#endif +} + + +int sst_set_pmic_config(int str_id, struct snd_sst_pmic_config *pc) +{ + int retval = 0; + sst_dbg("stream_control for %d\n", str_id); + retval = sst_validate_strid(str_id); + if (0 != retval) + return retval; + sst_ops->scard_ops.set_pcm_params(pc->sfreq, pc->pcm_wd_sz); + return retval; +} + + +int sst_create_sg_list(struct stream_info *stream, + struct sst_frame_info *sg_list) +{ + struct sst_stream_bufs *kbufs = NULL; + int i = 0; + + list_for_each_entry(kbufs, &stream->bufs, node) { + if (kbufs->in_use == false) { + sg_list->addr[i].addr = + virt_to_phys((void *) + kbufs->addr + kbufs->offset); + sst_dbg("phy addr[%d] 0x%x Size 0x%x\n", i, + sg_list->addr[i].addr, kbufs->size); + sg_list->addr[i].size = kbufs->size; + kbufs->in_use = true; + i++; + } + if (i >= MAX_NUM_SCATTER_BUFFERS) + break; + } + + sg_list->num_entries = i; + sst_dbg("sg list entries = %d \n", sg_list->num_entries); + return i; +} + +/** +* sst_play_frame - Send msg for sending stream frames +* @streamID: ID of stream +* +* This function is called when codec set is sucessfull +* it send the initial frames to SST +*/ +int sst_play_frame(int str_id) +{ + int i = 0, retval = 0; + struct ipc_post *msg = NULL; + struct sst_frame_info sg_list = {0}; + struct sst_stream_bufs *kbufs = NULL, *_kbufs; + struct stream_info *stream = NULL; + + sst_dbg("play frame for %d\n", str_id); + retval = sst_validate_strid(str_id); + if (retval != 0) + return retval; + + stream = &sst_ops->streams[str_id]; + if (stream->status == STREAM_UN_INIT || + stream->status == STREAM_PAUSED) { + sst_err("stream in UNIT state\n"); + return -EINVAL; + } + /*clear prev sent buffers*/ + list_for_each_entry_safe(kbufs, _kbufs, &stream->bufs, node) { + if (kbufs->in_use == true) { + list_del(&kbufs->node); + kfree(kbufs); + sst_dbg("del node \n"); + } + } + if (list_empty(&stream->bufs)) { + /*no user buffer available*/ + sst_dbg("Null buffer!!!!stream status = %d \n", stream->status); + stream->prev = stream->status; + stream->status = STREAM_INIT; + sst_dbg("new stream status = %d \n", stream->status); + if (stream->data_blk.on == true) { + sst_dbg("user list is empty.. wake \n"); + /*unblock*/ + stream->data_blk.ret_code = 0; + stream->data_blk.condition = true; + stream->data_blk.on = false; + wake_up(&sst_ops->wait_queue); + } +#ifdef CONFIG_SST_OSPM_SUPPORT + sst_ospm_send_event(OSPM_EVENT_AUDIO_BUF_EMPTY); +#endif + return -EINVAL; + } + + /*create list*/ + i = sst_create_sg_list(stream, &sg_list); + + /*post msg*/ + if (0 != sst_create_large_msg(&msg)) + return -ENOMEM; + + sst_fill_header(&msg->header, IPC_IA_PLAY_FRAMES, 1, str_id); + msg->header.part.data = sizeof(u32) + sizeof(sg_list); + memcpy(msg->mailbox_data, &msg->header, sizeof(u32)); + memcpy(msg->mailbox_data + sizeof(u32), &sg_list, sizeof(sg_list)); + list_add_tail(&msg->node, &sst_ops->ipc_dispatch_list); + sst_post_message(&sst_ops->ipc_post_msg_wq); + return 0; + +} + +/** +* sst_capture_frame - switches based on ecoded/pcm capture for sending stream frames +* @streamID: ID of stream +* +* This function is called when frames to SST are sent for capture +*/ +int sst_capture_frame(int str_id) +{ + int i = 0, retval = 0; + struct ipc_post *msg = NULL; + struct sst_frame_info sg_list = {0}; + struct sst_stream_bufs *kbufs = NULL, *_kbufs; + struct stream_info *stream = NULL; + + + sst_dbg("capture frame for %d\n", str_id); + retval = sst_validate_strid(str_id); + if (retval != 0) + return retval; + + stream = &sst_ops->streams[str_id]; + if (stream->status == STREAM_UN_INIT || + stream->status == STREAM_PAUSED) { + sst_err("stream in UNIT state\n"); + return -EINVAL; + } + + /*clear prev sent buffers*/ + list_for_each_entry_safe(kbufs, _kbufs, &stream->bufs, node) { + if (kbufs->in_use == true) { + list_del(&kbufs->node); + kfree(kbufs); + sst_dbg("del node \n"); + } + } + + if (list_empty(&stream->bufs)) { + /*no user buffer available*/ + sst_dbg("Null buffer!!!!stream status = %d \n", stream->status); + stream->prev = stream->status; + stream->status = STREAM_INIT; + sst_dbg("new stream status = %d \n", stream->status); + if (stream->data_blk.on == true) { + sst_dbg("user list is empty.. wake \n"); + /*unblock*/ + stream->data_blk.ret_code = 0; + stream->data_blk.condition = true; + stream->data_blk.on = false; + wake_up(&sst_ops->wait_queue); + } +#ifdef CONFIG_SST_OSPM_SUPPORT + sst_ospm_send_event(OSPM_EVENT_AUDIO_BUF_FULL); +#endif + return -EINVAL; + } + + /*create new sg list*/ + i = sst_create_sg_list(stream, &sg_list); + + /*post msg*/ + if (0 != sst_create_large_msg(&msg)) + return -ENOMEM; + + sst_fill_header(&msg->header, IPC_IA_CAPT_FRAMES, 1, str_id); + msg->header.part.data = sizeof(u32) + sizeof(sg_list); + memcpy(msg->mailbox_data, &msg->header, sizeof(u32)); + memcpy(msg->mailbox_data + sizeof(u32), &sg_list, sizeof(sg_list)); + list_add_tail(&msg->node, &sst_ops->ipc_dispatch_list); + sst_post_message(&sst_ops->ipc_post_msg_wq); + return 0; +} -- 1.5.4.5

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[alsa-devel] [PATCH] [RFC 6/13] Intel SST driver dsp engine interface module
by Vinod Koul 03 Jul '09

03 Jul '09

This adds the IPC module which uses Inter process mechanism to communicate between driver & SST engine (DSP processor). To communicate between IA processor and DSP, IPC doorbell registers are used. A write to these registers triggers an interrupt to other side. The format of messages and "mailbox" for message payload is defined in intel_lpe_fw_ipc.h Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> new file: sound/pci/sst/intel_sst_fw_ipc.h new file: sound/pci/sst/intel_sst_ipc.c --- sound/pci/sst/intel_sst_fw_ipc.h | 368 ++++++++++++++++++++++++++ sound/pci/sst/intel_sst_ipc.c | 536 ++++++++++++++++++++++++++++++++++++++ 2 files changed, 904 insertions(+), 0 deletions(-) create mode 100644 sound/pci/sst/intel_sst_fw_ipc.h create mode 100644 sound/pci/sst/intel_sst_ipc.c diff --git a/sound/pci/sst/intel_sst_fw_ipc.h b/sound/pci/sst/intel_sst_fw_ipc.h new file mode 100644 index 0000000..6520e4e --- /dev/null +++ b/sound/pci/sst/intel_sst_fw_ipc.h @@ -0,0 +1,368 @@ +#ifndef __INTEL_SST_FW_IPC_H__ +#define __INTEL_SST_FW_IPC_H__ +/* +* intel_sst_fw_ipc.h - Intel SST Driver for audio engine +* +* Copyright (C) 2008-09 Intel Corporation +* Author: Vinod Koul <vinod.koul(a)intel.com> +* Harsha Priya <priya.harsha(a)intel.com> +* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +* +* 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; version 2 of the License. +* +* 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. +* +* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +* +* This driver exposes the audio engine functionalities to the ALSA +* and middleware. +* This file has definitions shared between the firmware and driver +*/ + +#define MAX_NUM_STREAMS 4 +#define MAX_DBG_RW_BYTES 80 +#define MAX_NUM_SCATTER_BUFFERS 8 +#define MAX_LOOP_BACK_DWORDS 8 +/* IPC base address and mailbox, timestamp offsets */ +#define SST_MAILBOX_SIZE 0x0400 +#define SST_MAILBOX_SEND 0x0000 +#define SST_MAILBOX_RCV 0x0804 +#define SST_TIME_STAMP 0x1800 +#define SST_RESERVED_OFFSET 0x1840 +#define SST_CHEKPOINT_OFFSET 0x1C00 +#define REPLY_MSG 0x80 + +/* Message ID's for IPC messages */ +/* Bits B7: SST or IA/SC ; B6-B4: Msg Category; B3-B0: Msg Type */ + +/* I2L Firmware/Codec Download msgs */ +#define IPC_IA_PREP_LIB_DNLD 0x01 +#define IPC_IA_LIB_DNLD_CMPLT 0x02 + +#define IPC_IA_SET_PMIC_TYPE 0x03 +#define IPC_IA_GET_FW_VERSION 0x04 +#define IPC_IA_GET_FW_BUILD_INF 0x05 +#define IPC_IA_GET_FW_INFO 0x06 + +/* I2L Codec Config/control msgs */ +#define IPC_IA_SET_CODEC_PARAMS 0x10 +#define IPC_IA_GET_CODEC_PARAMS 0x11 +#define IPC_IA_SET_PPP_PARAMS 0x12 +#define IPC_IA_GET_PPP_PARAMS 0x13 +#define IPC_IA_PLAY_FRAMES 0x14 +#define IPC_IA_CAPT_FRAMES 0x15 +#define IPC_IA_PLAY_VOICE 0x16 +#define IPC_IA_CAPT_VOICE 0x17 +#define IPC_IA_DECODE_FRAMES 0x18 + +/* I2L Stream config/control msgs */ +#define IPC_IA_ALLOC_STREAM 0x20 /* Allocate a stream ID */ +#define IPC_IA_FREE_STREAM 0x21 /* Free the stream ID */ +#define IPC_IA_SET_STREAM_PARAMS 0x22 +#define IPC_IA_GET_STREAM_PARAMS 0x23 +#define IPC_IA_PAUSE_STREAM 0x24 +#define IPC_IA_RESUME_STREAM 0x25 +#define IPC_IA_DROP_STREAM 0x26 +#define IPC_IA_DRAIN_STREAM 0x27 /* Short msg with str_id */ +#define IPC_IA_TARGET_DEV_SELECT 0x28 +#define IPC_IA_CONTROL_ROUTING 0x29 + +/* Debug msgs */ +#define IPC_IA_DBG_MEM_READ 0x40 +#define IPC_IA_DBG_MEM_WRITE 0x41 +#define IPC_IA_DBG_LOOP_BACK 0x42 + +/* L2I Firmware/Codec Download msgs */ +#define IPC_IA_FW_INIT_CMPLT 0x81 + +/* L2I Codec Config/control msgs */ +#define IPC_SST_GET_PLAY_FRAMES 0x90 /* Request IA more data */ +#define IPC_SST_GET_CAPT_FRAMES 0x91 /* Request IA more data */ +#define IPC_SST_BUF_UNDER_RUN 0x92 /* PB Under run and stopped */ +#define IPC_SST_BUF_OVER_RUN 0x93 /* CAP Under run and stopped */ +#define IPC_SST_DRAIN_END 0x94 /* PB Drain complete and stopped */ +#define IPC_SST_CHNGE_SSP_PARAMS 0x95 /* PB SSP parameters changed */ +#define IPC_SST_STREAM_PROCESS_FATAL_ERR 0x96/*error in processing a stream*/ +#define IPC_SST_PERIOD_ELAPSED 0x97 /*period elapsed*/ +#define IPC_IA_TARGET_DEV_CHNGD 0x98 /* error in processing a stream */ + +/* L2S messages */ +#define IPC_SC_DDR_LINK_UP 0xC0 +#define IPC_SC_DDR_LINK_DOWN 0xC1 + +/* L2I Error reporting msgs */ +#define IPC_IA_MEM_ALLOC_FAIL 0xE0 +#define IPC_IA_PROC_ERR 0xE1 /* error in processing a + stream can be used by playback and + capture modules */ + +/* L2I Debug msgs*/ +#define IPC_IA_PRINT_STRING 0xF0 + +/* Command Response or Acknowledge message to any IPC message will have + * same message ID and stream ID information which is sent. + * There is no specific Ack message ID. The data field is used as response + * meaning. + */ +enum ackData { + IPC_ACK_SUCCESS = 0, + IPC_ACK_FAILURE +}; + + +enum sst_error_codes { + /* Error code,response to msgId: Description */ + /* Common error codes */ + SST_SUCCESS = 0, /*Success*/ + SST_ERR_INVALID_STREAM_ID, /*Invalid stream ID */ + SST_ERR_INVALID_MSG_ID, /*Invalid message ID */ + SST_ERR_INVALID_STREAM_OP, /*Invalid stream operation request */ + SST_ERR_INVALID_PARAMS, /*Invalid params */ + SST_ERR_INVALID_CODEC, /*Invalid codec type */ + SST_ERR_INVALID_MEDIA_TYPE, /*Invalid media type */ + SST_ERR_STREAM_ERR, /*ANY: Stream control or config or + processing error */ + + /* IPC specific error codes */ + SST_IPC_ERR_CALL_BACK_NOT_REGD, /*Call back for msg not regd*/ + SST_IPC_ERR_STREAM_NOT_ALLOCATED, /*Stream is not allocated */ + SST_IPC_ERR_STREAM_ALLOC_FAILED, /*ALLOC:Stream alloc failed*/ + SST_IPC_ERR_GET_STREAM_FAILED, /*ALLOC:Get stream id failed*/ + SST_ERR_MOD_NOT_AVAIL, /* SET/GET: Mod(AEC/AGC/ALC) not available */ + SST_ERR_MOD_DNLD_RQD, /* SET/GET: Mod(AEC/AGC/ALC) download required*/ + SST_ERR_STREAM_STOPPED, /* ANY: Stream is in stopped state */ + SST_ERR_STREAM_IN_USE, /* ANY: Stream is already in use */ + + /* Capture specific error codes*/ + SST_CAP_ERR_INCMPLTE_CAPTURE_MSG,/*ANY:Incomplete message */ + SST_CAP_ERR_CAPTURE_FAIL, /*ANY:Capture op failed */ + SST_CAP_ERR_GET_DDR_NEW_SGLIST, /*AR: Meena */ + SST_CAP_ERR_UNDER_RUN, /* lack of input data */ + SST_CAP_ERR_OVERFLOW, /* lack of output space */ + + /* Playback specific error codes*/ + SST_PB_ERR_INCMPLTE_PLAY_MSG, /* ANY: Incomplete message */ + SST_PB_ERR_PLAY_FAIL, /* ANY: Playback operation failed */ + SST_PB_ERR_GET_DDR_NEW_SGLIST, /* AR: Ashok */ + + /* Codec manager specific error codes*/ + SST_LIB_ERR_LIB_DNLD_REQUIRED, /* ALLOC: Codec download required */ + SST_LIB_ERR_LIB_NOT_SUPPORTED, /* Library is not supported */ + + /* Library manager specific error codes*/ + SST_SCC_ERR_PREP_DNLD_FAILED, /* Failed to prepare for codec download */ + SST_SCC_ERR_LIB_DNLD_RES_FAILED, /* Lib download resume failed */ + /* Scheduler specific error codes*/ + SST_SCH_ERR_FAIL, /* REPORT: */ + + /* DMA specific error codes */ + SST_DMA_ERR_NO_CHNL_AVAILABLE, /* DMA Ch not available*/ + SST_DMA_ERR_INVALID_INPUT_PARAMS, /* Invalid input params */ + SST_DMA_ERR_CHNL_ALREADY_SUSPENDED, /* Ch is suspended */ + SST_DMA_ERR_CHNL_ALREADY_STARTED, /* Ch already started */ + SST_DMA_ERR_CHNL_NOT_ENABLED, /* Ch not enabled */ + SST_DMA_ERR_TRANSFER_FAILED, /* Transfer failed */ + SST_SSP_ERR_ALREADY_ENABLED, /* REPORT: SSP already enabled*/ + SST_SSP_ERR_ALREADY_DISABLED, /* REPORT: SSP already disabled*/ + SST_SSP_ERR_NOT_INITIALIZED, + + /* Other error codes */ + SST_ERR_MOD_INIT_FAIL, /* Firmware Module init failed */ + + /* FW init error codes */ + SST_RDR_ERR_IO_DEV_SEL_NOT_ALLOWED, + SST_RDR_ERR_ROUTE_ALREADY_STARTED, + SST_RDR_PREP_CODEC_DNLD_FAILED, + + /* Memory debug error codes */ + SST_ERR_DBG_MEM_READ_FAIL, + SST_ERR_DBG_MEM_WRITE_FAIL, + + +}; + +enum dbg_mem_data_type { + /* Data type of debug read/write */ + DATA_TYPE_U32, + DATA_TYPE_U16, + DATA_TYPE_U8, +}; + +/* CAUTION NOTE: All IPC message body must be multiple of 32 bits.*/ + +/* IPC Header */ +union ipc_header { + struct { + u32 msg_id:8; /* Message ID - Max 256 Message Types */ + u32 str_id:3; /* Undefined for SC communication */ + u32 large:1; /* Large Message if large = 1 */ + u32 reserved:4;/* Reserved for future use */ + u32 data:14; /* Ack/Info for msg, size of msg in Mailbox */ + u32 done:1; /* bit 30 */ + u32 busy:1; /* bit 31 */ + } part; + u32 full; +} __attribute__ ((packed)); + +struct ipc_header_fw_init { + struct snd_sst_fw_version fw_version;/* Firmware version details*/ + u16 result; /* Fw init result */ + u8 module_id; /* Module ID in case of error */ + u8 debug_info; /* Debug info from Module ID in case of fail */ +} __attribute__ ((packed)); + +/* Firmware build info */ +struct sst_fw_build_info { + unsigned char date[16]; /* Firmware build date */ + unsigned char time[16]; /* Firmware build time */ +} __attribute__ ((packed)); + +/* Address and size info of a frame buffer in DDR */ +struct sst_address_info { + u32 addr; /* Address at IA */ + u32 size; /* Size of the buffer */ +} __attribute__ ((packed)); + +/* Time stamp */ +struct snd_sst_tstamp { + u64 samples_processed; + u64 samples_rendered; + u32 sampling_frequency; +}; + +/* Frame info to play or capture */ +struct sst_frame_info { + u16 num_entries; /* number of entries to follow */ + u16 rsrvd; + struct sst_address_info addr[MAX_NUM_SCATTER_BUFFERS]; +} __attribute__ ((packed)); + +/* Frames info for decode */ +struct snd_sst_decode_info { + struct snd_sst_dbufs frames_in; + struct snd_sst_dbufs frames_out; +} __attribute__ ((packed)); +/* SST to IA print debug message*/ +struct ipc_sst_ia_print_params { + u32 string_size; /* Max value is 160 */ + u8 prt_string[160]; /* Null terminated Char string */ +} __attribute__ ((packed)); +/* Voice data message */ +struct snd_sst_voice_data { + u16 num_bytes; /* Number of valid voice data bytes */ + u8 pcm_wd_size; /* 0=8 bit, 1=16 bit 2=32 bit*/ + u8 reserved; /* Reserved */ + u8 voice_data_buf[0]; /* Voice data buffer in bytes, little endian */ +} __attribute__ ((packed)); + +/* SST to IA memory read debug message */ +struct ipc_sst_ia_dbg_mem_rw { + u16 num_bytes; /* Maximum of MAX_DBG_RW_BYTES */ + u16 data_type; /* enum: dbg_mem_data_type */ + u32 address; /* Memory address of data memory of data_type */ + u8 rw_bytes[MAX_DBG_RW_BYTES];/* Maximum of 64 bytes can be RW */ +} __attribute__ ((packed)); + +struct ipc_sst_ia_dbg_loop_back { + u16 num_dwords; /* Maximum of MAX_DBG_RW_BYTES */ + u16 increment_val;/* Increments dwords by this value, 0- no increment */ + u32 lpbk_dwords[MAX_LOOP_BACK_DWORDS];/* Maximum of 8 dwords loopback */ +} __attribute__ ((packed)); + +/* Stream type params struture for Alloc stream */ +struct snd_sst_str_type { + u8 codec_type; /* Codec type */ + u8 str_type; /* 1 = voice 2 = music */ + u8 operation; /* Playback or Capture */ + u8 protected_str; /* 0=Non DRM, 1=DRM */ + u8 pvt_id; /* Driver Private ID */ + u8 reserved; /* Reserved */ + u16 result; /* Result used for acknowledgment */ +} __attribute__ ((packed)); + +/* Library info structure */ +struct module_info { + u32 lib_version; + u32 lib_type; + u32 media_type; + u8 lib_name[12]; + u32 lib_caps; + unsigned char b_date[16]; /* Lib build date */ + unsigned char b_time[16]; /* Lib build time */ +} __attribute__ ((packed)); + +/* Library slot info */ +struct lib_slot_info { + u8 slot_num; /* 1 or 2*/ + u8 reserved1; + u16 reserved2; + u32 iram_size; /* slot size in IRAM */ + u32 dram_size; /* slot size in DRAM */ + u32 iram_offset; /* starting offset of slot in IRAM */ + u32 dram_offset; /* starting offset of slot in DRAM */ +} __attribute__ ((packed)); + +struct snd_sst_lib_download { + struct module_info lib_info; /* library info type, capabilities etc*/ + struct lib_slot_info slot_info; /* slot info to be downloaded */ + u32 mod_entry_pt; +}; + +struct snd_sst_lib_download_info { + struct snd_sst_lib_download dload_lib; + u16 result; /* Result used for acknowledgment */ + u8 pvt_id; /*Private ID*/ + u8 reserved; /*for alignment*/ +}; + +/* Alloc stream params structure */ +struct snd_sst_alloc_params { + struct snd_sst_str_type str_type; + struct snd_sst_stream_params stream_params; +}; + +struct snd_sst_fw_get_stream_params { + struct snd_sst_str_type str_type; + struct snd_sst_params codec_params; + struct snd_sst_pmic_config pcm_params; +}; + +/* Alloc stream response message */ +struct snd_sst_alloc_response { + struct snd_sst_str_type str_type; /* Stream type for allocation*/ + struct snd_sst_lib_download lib_dnld; /*Valid only for codec dnld*/ +}; + +/*struct ipc_msg_body { + union { + CODEC_PARAM_STRUCTURES; + PPP_PARAM_STRUCTURES; + struct snd_sst_alloc_params alloc_params; + struct snd_sst_alloc_response alloc_response; + struct snd_sst_stream_params stream_params; + struct sst_frame_info frames_info; + struct ipc_sst_ia_print_params print_params; + struct ipc_sst_ia_dbg_mem_rw dbg_mem_rw; + struct ipc_sst_ia_dbg_loop_back loop_back; + struct pmic_pcm_params ssp_params; + } u; +};*/ + + + +struct ipc_post { + struct list_head node; + union ipc_header header; /* driver specific */ + char *mailbox_data; +}; + +#endif /*__INTEL_SST_FW_IPC_H__*/ diff --git a/sound/pci/sst/intel_sst_ipc.c b/sound/pci/sst/intel_sst_ipc.c new file mode 100644 index 0000000..98b4031 --- /dev/null +++ b/sound/pci/sst/intel_sst_ipc.c @@ -0,0 +1,536 @@ +/* + * intel_sst_ipc.c - Intel SST Driver for audio engine + * + * Copyright (C) 2008-09 Intel Corporation + * Authors: Vinod Koul <vinod.koul(a)intel.com> + * Harsha Priya <priya.harsha(a)intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * 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; version 2 of the License. + * + * 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. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This file defines all ipc functions + */ + +#include <linux/cdev.h> +#include <linux/pci.h> +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/syscalls.h> +#include <linux/file.h> +#include <linux/interrupt.h> +#include <linux/list.h> +#include <linux/workqueue.h> +#include <linux/mutex.h> +#include <linux/firmware.h> +#include <sound/intel_sst.h> +#include <sound/intel_sst_ioctl.h> +#include "intel_sst_fw_ipc.h" +#include "intel_sst_common.h" +#include "intel_sst_pvt.h" + +void sst_send_loop_test(int loop_no) +{ + struct ipc_post *msg = NULL; + struct ipc_sst_ia_dbg_loop_back loop_msg; + static int large_num; + + sst_dbg("Loop testing %d \n", loop_no); + + if (large_num >= 4) { + sst_dbg("Loop testing complete.....\n"); + return; + } + if (loop_no >= 4) { + /*large loop*/ + large_num++; + sst_dbg("Large msg \n"); + if (0 != sst_create_large_msg(&msg)) + return; + + loop_msg.increment_val = 1; + loop_msg.lpbk_dwords[0] = 0x11111111; + loop_msg.lpbk_dwords[1] = 0x22222222; + loop_msg.lpbk_dwords[2] = 0x33333333; + loop_msg.lpbk_dwords[3] = 0x44444444; + loop_msg.num_dwords = 4; + sst_fill_header(&msg->header, IPC_IA_DBG_LOOP_BACK, 1, loop_no); + msg->header.part.data = sizeof(u32) + sizeof(loop_msg); + memcpy(msg->mailbox_data, &msg->header, sizeof(u32)); + memcpy(msg->mailbox_data + sizeof(u32), + &loop_msg, sizeof(loop_msg)); + } else { + /*short loop*/ + sst_dbg("Loop Short msg \n"); + if (0 != sst_create_short_msg(&msg)) + return; + sst_fill_header(&msg->header, IPC_IA_DBG_LOOP_BACK, 0, loop_no); + } + list_add_tail(&msg->node, &sst_ops->ipc_dispatch_list); + sst_post_message(&sst_ops->ipc_post_msg_wq); + return; +} + +void sst_send_pmic_type(void) +{ + struct ipc_post *msg = NULL; + + sst_dbg("...called\n"); + + if (0 != sst_create_short_msg(&msg)) + return; + + sst_fill_header(&msg->header, IPC_IA_SET_PMIC_TYPE, 0, 0); + msg->header.part.data = sst_ops->pmic_vendor; + list_add_tail(&msg->node, &sst_ops->ipc_dispatch_list); + sst_post_message(&sst_ops->ipc_post_msg_wq); + return; +} + +void sst_get_fw_version(void) +{ + struct ipc_post *msg = NULL; + + sst_dbg("...called\n"); + + if (0 != sst_create_short_msg(&msg)) + return; + + sst_fill_header(&msg->header, IPC_IA_GET_FW_VERSION, 0, 0); + list_add_tail(&msg->node, &sst_ops->ipc_dispatch_list); + sst_post_message(&sst_ops->ipc_post_msg_wq); + if (0 != sst_create_short_msg(&msg)) + return; + + sst_fill_header(&msg->header, IPC_IA_GET_FW_BUILD_INF, 0, 0); + list_add_tail(&msg->node, &sst_ops->ipc_dispatch_list); + sst_post_message(&sst_ops->ipc_post_msg_wq); + return; +} + +/** +* sst_post_message - Posts message to SST +* @work: Pointer to work structure +* +* This function is called by any component in driver which +* wants to send an IPC message. This will post message only if +* busy bit is free +*/ +void sst_post_message(struct work_struct *work) +{ + struct ipc_post *msg; + union ipc_header header; + union interrupt_reg imr; + imr.full = 0; + + sst_dbg("..called \n"); + mutex_lock(&sst_ops->list_lock); + /*check list*/ + if (0 != list_empty(&sst_ops->ipc_dispatch_list)) { + /*list is empty, mask imr*/ + sst_dbg(" Empty msg queue... masking \n"); + imr.full = readl(sst_ops->shim + SST_IMRX); + imr.part.done_interrupt = 1; + writel(imr.full, sst_ops->shim + SST_IMRX); + mutex_unlock(&sst_ops->list_lock); + return; + } + + /*check busy bit*/ + header.full = readl(sst_ops->shim + SST_IPCX); + if (1 == header.part.busy) { + /*busy, unmask*/ + sst_dbg("Busy not free... unmasking\n"); + imr.full = readl(sst_ops->shim + SST_IMRX); + imr.part.done_interrupt = 0; + writel(imr.full, sst_ops->shim + SST_IMRX); + mutex_unlock(&sst_ops->list_lock); + return; + } + /*copy msg from list*/ + msg = list_entry(sst_ops->ipc_dispatch_list.next, + struct ipc_post, node); + list_del(&msg->node); + sst_dbg("Post message: header = %x\n", msg->header.full); + sst_dbg("size: = %x\n", msg->header.part.data); + if (1 == msg->header.part.large) + memcpy_toio(sst_ops->mailbox + SST_MAILBOX_SEND, + msg->mailbox_data, msg->header.part.data); + writel(msg->header.full, sst_ops->shim + SST_IPCX); + mutex_unlock(&sst_ops->list_lock); + + kfree(msg->mailbox_data); + kfree(msg); + sst_dbg("...done\n"); + return; +} + +void sst_clear_interrupt(void) +{ + union interrupt_reg isr; + union interrupt_reg imr; + union ipc_header clear_ipc; + + sst_dbg("sst clearing interrupt \n"); + imr.full = readl(sst_ops->shim + SST_IMRX); + isr.full = readl(sst_ops->shim + SST_ISRX); + /* write 1 to clear */; + isr.part.busy_interrupt = 1; + writel(isr.full, sst_ops->shim + SST_ISRX); + /*Set IA done bit*/ + clear_ipc.full = readl(sst_ops->shim + SST_IPCD); + clear_ipc.part.busy = 0; + clear_ipc.part.done = 1; + clear_ipc.part.data = IPC_ACK_SUCCESS; + writel(clear_ipc.full, sst_ops->shim + SST_IPCD); + /*un mask busy interrupt*/ + imr.part.busy_interrupt = 0; + writel(imr.full, sst_ops->shim + SST_IMRX); +} + +/** +* sst_process_message - Processes message from SST +* @work: Pointer to work structure +* +* This function is scheduled by ISR +* It take a msg from process_queue and does action based on msg +*/ +void sst_process_message(struct work_struct *work) +{ + struct sst_ipc_msg_wq *msg = + container_of(work, struct sst_ipc_msg_wq, wq); + int str_id = msg->header.part.str_id; + + sst_dbg("called \n"); + + /*based on msg in list call respective handler*/ + switch (msg->header.part.msg_id) { + case IPC_SST_BUF_UNDER_RUN: + case IPC_SST_BUF_OVER_RUN: + if (sst_validate_strid(str_id) != 0) { + sst_err("stream id %d invalid\n", str_id); + break; + } + sst_err("Buffer under/overrun for %d\n", + msg->header.part.str_id); + sst_dbg("Got Underrun & not to send data...ignore\n"); + break; + + case IPC_SST_GET_PLAY_FRAMES: + { + struct stream_info *stream ; + + if (sst_validate_strid(str_id) != 0) { + sst_err("stream id %d invalid\n", str_id); + break; + } + /*call sst_play_frame*/ + stream = &sst_ops->streams[str_id]; + sst_dbg("sst_play_frames for %d\n", msg->header.part.str_id); + mutex_lock(&sst_ops->streams[str_id].lock); + sst_play_frame(msg->header.part.str_id); + mutex_unlock(&sst_ops->streams[str_id].lock); + break; + } + + case IPC_SST_PERIOD_ELAPSED: + { + struct snd_sst_tstamp fw_tstamp = {0,}; + struct stream_info *stream ; + + if (sst_validate_strid(str_id) != 0) { + sst_err("stream id %d invalid\n", str_id); + break; + } + stream = &sst_ops->streams[str_id]; + + sst_dbg("Period elapsed \n"); + memcpy_fromio(&fw_tstamp, + ((void *)(sst_ops->mailbox + SST_TIME_STAMP) + + (str_id * sizeof(fw_tstamp))), + sizeof(fw_tstamp)); + sst_dbg("samples played = %lld\n", fw_tstamp.samples_processed); + sst_dbg("diff in mesg = %d\n", msg->header.part.data); + sst_clear_interrupt(); + if (stream->period_elapsed) + stream->period_elapsed(stream->pcm_substream); + return; + } + + case IPC_SST_GET_CAPT_FRAMES: + /*call sst_capture_frame*/ + if (sst_validate_strid(str_id) != 0) { + sst_err("stream id %d invalid\n", str_id); + break; + } + sst_dbg("sst_capture_frames for %d\n", msg->header.part.str_id); + mutex_lock(&sst_ops->streams[str_id].lock); + if (sst_ops->streams[str_id].ops == STREAM_OPS_CAPTURE && + sst_ops->streams[str_id].mmapped == false && + sst_ops->streams[str_id].codec != + SST_CODEC_TYPE_PCM) { + sst_dbg("waking up block for copy...\n"); + sst_ops->streams[str_id].data_blk.ret_code = 0; + sst_ops->streams[str_id].data_blk.condition = true; + sst_ops->streams[str_id].data_blk.on = false; + wake_up(&sst_ops->wait_queue); + } else + sst_capture_frame(msg->header.part.str_id); + mutex_unlock(&sst_ops->streams[str_id].lock); + break; + + case IPC_IA_PRINT_STRING: + sst_dbg("been asked to print something by fw\n"); + /*TBD*/ + break; + + case IPC_IA_FW_INIT_CMPLT: { + /*send next data to FW*/ + struct ipc_header_fw_init *init = + (struct ipc_header_fw_init *)msg->mailbox; + + sst_dbg("*** FW Init msg came*** \n"); + if (0 == init->result) { + sst_ops->sst_state = SST_FW_RUNNING; + sst_info("FW Version %x.%x \n", + init->fw_version.major, init->fw_version.minor); + sst_info("Build No %x Type %x \n", + init->fw_version.build, init->fw_version.type); +#ifdef SND_LOOP_TEST + sst_send_loop_test(0); +#endif + sst_send_pmic_type(); + sst_get_fw_version(); + } else { + sst_ops->sst_state = SST_ERROR; + sst_dbg("FW Init failed, Error %x\n", init->result); + sst_dbg("FW Init failed, Module %x, Debug Info %x \n", + init->module_id, init->debug_info); + } + sst_dbg("Waking up... open\n"); + sst_wake_up_alloc_block(sst_ops, 0xFF, 0, NULL); + break; + } + + case IPC_SST_STREAM_PROCESS_FATAL_ERR: + if (sst_validate_strid(str_id) != 0) { + sst_err("stream id %d invalid\n", str_id); + break; + } + sst_err(" codec fatal error %x for stream %d... \n", + msg->header.full, msg->header.part.str_id); + sst_err("Dropping the stream \n"); + sst_drop_stream(msg->header.part.str_id); + break; + + default: + /*Illegal case*/ + sst_err("Unhandled case msg_id %x message %x\n", + msg->header.part.msg_id, msg->header.full); + } + sst_clear_interrupt(); + return; +} + +/** +* sst_process_reply - Processes reply message from SST +* @work: Pointer to work structure +* +* This function is scheduled by ISR +* It take a reply msg from response_queue and +* does action based on msg +*/ +void sst_process_reply(struct work_struct *work) +{ + struct sst_ipc_msg_wq *msg = + container_of(work, struct sst_ipc_msg_wq, wq); + int str_id = msg->header.part.str_id; + struct stream_info *str_info = NULL; + + sst_dbg("called for %x \n", msg->header.full); + + switch (msg->header.part.msg_id) { + case IPC_IA_DROP_STREAM: + case IPC_IA_PAUSE_STREAM: + case IPC_IA_RESUME_STREAM: + case IPC_IA_SET_STREAM_PARAMS: + if (0 == msg->header.part.data) { + sst_dbg("Msg succedded %x \n", msg->header.part.msg_id); + } else { + sst_err("Msg %x reply error %x \n", + msg->header.part.msg_id, + msg->header.part.data); + } + if (sst_validate_strid(str_id) != 0) { + sst_err("stream id %d invalid\n", str_id); + break; + } + + str_info = &sst_ops->streams[str_id]; + str_info->ctrl_blk.ret_code = msg->header.part.data; + if (str_info->ctrl_blk.on == true) { + str_info->ctrl_blk.condition = true; + wake_up(&sst_ops->wait_queue); + } + break; + + case IPC_IA_FREE_STREAM: + if (0 == msg->header.part.data) { + sst_dbg("Stream %d freed\n", str_id); + } else { + sst_err("Free for %d returned error %x\n", + str_id, msg->header.part.data); + } + break; + case IPC_IA_GET_STREAM_PARAMS: { + if (sst_validate_strid(str_id) != 0) { + sst_err("stream id %d invalid\n", str_id); + break; + } + if (0 == msg->header.part.data) { + struct snd_sst_stream_params *params = NULL; + params = kzalloc(sizeof(*params), GFP_KERNEL); + if (NULL == params) { + sst_err("mem allocation failed\n"); + break; + } + memcpy(params, &msg->mailbox, sizeof(*params)); + str_info = &sst_ops->streams[str_id]; + str_info->get_parameters = params; + } else { + sst_err("Msg %x reply error %x \n", + msg->header.part.msg_id, msg->header.part.data); + } + break; + } + + case IPC_IA_ALLOC_STREAM: { + /*map to stream, call play*/ + struct snd_sst_alloc_response *resp = + (struct snd_sst_alloc_response *)msg->mailbox; + if (0 != resp->str_type.result) { + /*error case*/ + struct snd_sst_alloc_response *lib = NULL; + sst_err("error alloc stream = %x \n", + resp->str_type.result); + if (resp->str_type.result == + SST_LIB_ERR_LIB_DNLD_REQUIRED) { + lib = kzalloc(sizeof(*lib), GFP_ATOMIC); + if (NULL == lib) { + sst_err("mem allocation failed \n"); + break; + } + memcpy(lib, msg->mailbox, sizeof(*lib)); + /*library needs to be downloaded*/ + sst_dbg("Codec Download required \n"); + } + sst_wake_up_alloc_block(sst_ops, resp->str_type.pvt_id, + (-resp->str_type.result), lib); + break; + } + sst_alloc_stream_response(str_id, &resp->str_type); + break; + } + + case IPC_IA_DBG_LOOP_BACK: + /*Debug loop back msg*/ + sst_dbg("Loop back came \n"); + if (0 != msg->header.part.data) + sst_dbg("Possible error if not large \n"); + sst_dbg("Loop ID: %d\n", str_id); + if (msg->header.part.large) { + struct ipc_sst_ia_dbg_loop_back *loop_msg = + (struct ipc_sst_ia_dbg_loop_back *)msg->mailbox; + int i; + sst_dbg("Got large loop back: Words %d\n", + loop_msg->num_dwords); + for (i = 0; i < loop_msg->num_dwords; i++) { + sst_dbg("Loop Word %d = %d \n", i, + loop_msg->lpbk_dwords[i]); + } + } + sst_send_loop_test((str_id + 1)); + break; + + case IPC_IA_PLAY_FRAMES: + case IPC_IA_CAPT_FRAMES: + if (sst_validate_strid(str_id) != 0) { + sst_err("stream id %d invalid\n", str_id); + break; + } + sst_dbg("Ack for play/capt frames recived \n"); + break; + + case IPC_IA_PREP_LIB_DNLD: { + struct snd_sst_str_type *str_type = + (struct snd_sst_str_type *)msg->mailbox; + sst_dbg("Prep Lib download %x\n", msg->header.part.msg_id); + if (0 != str_type->result) + sst_err("Error in prep lib download 0x%x\n", + str_type->result); + else + sst_dbg("Need to download codec now...\n"); + str_type->result = 0; + sst_wake_up_alloc_block(sst_ops, str_type->pvt_id, + str_type->result, NULL); + break; + } + + case IPC_IA_LIB_DNLD_CMPLT: { + struct snd_sst_lib_download_info *resp = + (struct snd_sst_lib_download_info *)msg->mailbox; + int retval = resp->result; + + sst_dbg("Lib download cmplt %x\n", msg->header.part.msg_id); + if (0 != retval) + sst_err("Error in lib dload %x\n", resp->result); + else { + sst_dbg("Codec download complete...\n"); + sst_info("Dloaded codec Type %d Ver %d Built %s: %s\n", + resp->dload_lib.lib_info.lib_type, + resp->dload_lib.lib_info.lib_version, + resp->dload_lib.lib_info.b_date, + resp->dload_lib.lib_info.b_time); + } + sst_wake_up_alloc_block(sst_ops, resp->pvt_id, retval, NULL); + break; + } + + case IPC_IA_GET_FW_VERSION: { + struct snd_sst_fw_version *version = + (struct snd_sst_fw_version *)msg->mailbox; + sst_info("***LOADED SST FW VERSION*** = %2d.%02d\n", + version->major, version->minor); + sst_info("Build No %x Type %x \n", + version->build, version->type); + break; + } + case IPC_IA_GET_FW_BUILD_INF: { + struct sst_fw_build_info *build = + (struct sst_fw_build_info *)msg->mailbox; + sst_info("Build date %s Time %s\n", + build->date, build->time); + break; + } + + default: + /*Illegal case*/ + sst_err("process reply :default case = %x\n", + msg->header.full); + + } + sst_clear_interrupt(); + return; +} -- 1.5.4.5

1 0

[alsa-devel] [PATCH] [RFC 12/13] Intel SST sound card headers
by Vinod Koul 03 Jul '09

03 Jul '09

This adds header files of sound card driver, it defines some utility functions and common declarations for vendor sound cards as well Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> new file: sound/pci/sst/intelmid.h new file: sound/pci/sst/intelmid_pvt.h new file: sound/pci/sst/intelmid_snd_control.h --- sound/pci/sst/intelmid.h | 274 ++++++++++++++++++++++++++++++++++ sound/pci/sst/intelmid_pvt.h | 198 ++++++++++++++++++++++++ sound/pci/sst/intelmid_snd_control.h | 153 +++++++++++++++++++ 3 files changed, 625 insertions(+), 0 deletions(-) create mode 100644 sound/pci/sst/intelmid.h create mode 100644 sound/pci/sst/intelmid_pvt.h create mode 100644 sound/pci/sst/intelmid_snd_control.h diff --git a/sound/pci/sst/intelmid.h b/sound/pci/sst/intelmid.h new file mode 100644 index 0000000..938ad52 --- /dev/null +++ b/sound/pci/sst/intelmid.h @@ -0,0 +1,274 @@ +/* + * intelmid.c - Intel Sound card driver for MID + * + * Copyright (C) 2008-09 Intel Corp + * Authors: Vinod Koul <vinod.koul(a)intel.com> + * Harsha Priya <priya.harsha(a)intel.com> + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * 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 version 2 of the License. + * + * 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. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * ALSA driver header for Intel MAD chipset + */ +#ifndef __INTELMID_H +#define __INTELMID_H + +#include <linux/time.h> +#define DRIVER_NAME "pmic_audio" +#define PMIC_SOUND_IRQ_TYPE_MASK (1 << 15) +#define AUDINT_BASE (0xFFFFEFF8 + (6 * sizeof(u8))) +/*#define FULL_CTRL*/ +/*#define REG_IRQ*/ +/*values #defined */ +/* will differ for different hw - to be taken from config */ +#define MAX_DEVICES 1 +#define MIN_RATE 8000 +#define MAX_RATE 48000 +#define MAX_BUFFER (320*1024) +#define MIN_BUFFER (320*1024) +#define MAX_PERIODS (1024*2) +#define MIN_PERIODS 1 +#define MAX_PERIOD_BYTES MAX_BUFFER +#define MIN_PERIOD_BYTES 32 +#define MAX_MUTE 1 +#define MIN_MUTE 0 +#define MONO_CNTL 1 +#define STEREO_CNTL 2 +#define MIN_CHANNEL 1 +#define MAX_CHANNEL 2 +#define FIFO_SIZE 0 /*fifo not being used*/ +#define INTEL_MAD "Intel MAD" +#define MUTE_ALL_DEV 9 +/*TODO +6 db*/ +#define MAX_VOL 64 +/*TODO -57 db*/ +#define MIN_VOL 0 +#define PLAYBACK_COUNT 1 +#define CAPTURE_COUNT 1 + +struct mad_device { + unsigned int mic_plugged:1; + unsigned int headphones_plugged:1; + unsigned int line_in_plugged:1; +}; + +struct mad_wq { + struct snd_pcm_substream *substream; + struct work_struct wq; + struct mutex wq_lock; +}; + +struct mad_capture_wq { + struct snd_pcm_substream *substream; + struct stream_buffer buffer_to_sst; + struct work_struct wq; +}; +struct snd_intelmad { + struct snd_card *card; /*ptr to the card details*/ + int card_index;/* card index */ + char *card_id; /*card id*/ + struct mad_device pin_sense; /*current pin sense */ + struct intel_sst_card_ops *sstdrv_ops;/*ptr to sst driver ops*/ + struct spi_device *spi; + int irq; + int pmic_status; + void __iomem *int_base; +}; + +struct snd_control_val { + int mic1_vol_max; + int mic1_vol_min; + int mic2_vol_max; + int mic2_vol_min; + int dmic_vol_max; + int dmic_vol_min; + int linein_vol_max; + int linein_vol_min; + int hp_vol_max; + int hp_vol_min; + int speaker_vol_max; + int speaker_vol_min; + int earpiece_vol_max; + int earpiece_vol_min; +}; + +struct mad_stream_pvt { + int stream_id; + int stream_status; + int stream_ops; + unsigned int stream_type; /*0:Audio 1:Voice Default:Audio*/ + struct snd_pcm_substream *substream; + unsigned int pcm_size; + unsigned int pcm_jiffie; + unsigned int pcm_buf_pos; + struct snd_pcm_indirect pcm_indirect; + struct pcm_stream_info buf_ptr; + unsigned long period_elapsed_ptr; + struct mad_wq mad_msg_wq; + struct mad_capture_wq mad_capture_wq; +}; + +enum mad_drv_status { + INIT = 1, + STARTED, + RUNNING, + PAUSED, + DROPPED, +}; + +enum mad_pmic_status { + PMIC_UNINIT = 1, + PMIC_INIT, +}; +#ifdef FULL_CTRL +enum _widget_ctrl { + LINE_IN_VOL = 1, + LINE_IN_MUTE, + DMIC_VOL, + DMIC_MUTE, + MIC1_VOL, + MIC1_MUTE, + MIC2_VOL, + MIC2_MUTE, + INPUT_SEL, + HEADPHONES_VOL , + HEADPHONES_MUTE, + INTERNAL_SPEAKERS_VOL, + INTERNAL_SPEAKERS_MUTE, + EARPIECE_VOL, + EARPIECE_MUTE, + OUTPUT_SEL, + MASTER_MUTE +}; +#else +enum _widget_ctrl { + HEADPHONES_VOL = 1 , + HEADPHONES_MUTE, +}; +#endif + +enum { + AUDIO_GENL_ATTR_UNSPEC = 0, + AUDIO_GENL_ATTR_EVENT, /* Audio event info needed by user space */ + AUDIO_GENL_ATTR_MAX, +}; +enum { + AUDIO_GENL_CMD_UNSPEC, + AUDIO_GENL_CMD_EVENT, /* kernel->user notifications for OSPM events */ + AUDIO_GENL_CMD_MAX, +}; + +enum eaudio_events { + AUDIO_EVENT_HP_DETECT, + AUDIO_EVENT_HS_DETECT, + AUDIO_EVENT_SHORT_PRESS, + AUDIO_EVENT_LONG_PRESS, + AUDIO_EVENT_COUNT, +}; + +struct audio_genl_event { + u32 orig; + enum eaudio_events event; +}; + +struct snd_pmic_ops v[3] = { + { + .set_input_dev = fs_set_selected_input_dev, + .set_output_dev = fs_set_selected_output_dev, + .set_mute = fs_set_mute, + .get_mute = fs_get_mute, + .set_vol = fs_set_vol, + .get_vol = fs_get_vol, + .init_card = fs_init_card, + .set_pcm_params = fs_set_pcm_params, + .init_card_capture = fs_init_capture_card, + }, + { + .set_input_dev = mx_set_selected_input_dev, + .set_output_dev = mx_set_selected_output_dev, + .set_mute = mx_set_mute, + .get_mute = mx_get_mute, + .set_vol = mx_set_vol, + .get_vol = mx_get_vol, + .init_card = mx_init_card, + .set_pcm_params = mx_set_pcm_params, + .init_card_capture = mx_init_capture_card, + }, + { + .set_input_dev = nc_set_selected_input_dev, + .set_output_dev = nc_set_selected_output_dev, + .set_mute = nc_set_mute, + .get_mute = nc_get_mute, + .set_vol = nc_set_vol, + .get_vol = nc_get_vol, + .init_card = nc_init_card, + .set_pcm_params = nc_set_pcm_params, + .init_card_capture = nc_init_capture_card, + }, +}; + +struct snd_control_val ctrl_val[3] = { + { + .mic1_vol_max = MAX_VOL, + .mic1_vol_min = MIN_VOL, + .mic2_vol_max = MAX_VOL, + .mic2_vol_min = MIN_VOL, + .dmic_vol_max = MAX_VOL, + .dmic_vol_min = MIN_VOL, + .linein_vol_max = MAX_VOL, + .linein_vol_min = MIN_VOL, + .hp_vol_max = MAX_VOL, + .hp_vol_min = MIN_VOL, + .speaker_vol_max = MAX_VOL, + .speaker_vol_min = MIN_VOL, + .earpiece_vol_max = MAX_VOL, + .earpiece_vol_min = MIN_VOL, + }, + { + .mic1_vol_max = MAX_VOL, + .mic1_vol_min = MIN_VOL, + .mic2_vol_max = MAX_VOL, + .mic2_vol_min = MIN_VOL, + .dmic_vol_max = MAX_VOL, + .dmic_vol_min = MIN_VOL, + .linein_vol_max = MAX_VOL, + .linein_vol_min = MIN_VOL, + .hp_vol_max = MAX_VOL, + .hp_vol_min = MIN_VOL, + .speaker_vol_max = MAX_VOL, + .speaker_vol_min = MIN_VOL, + .earpiece_vol_max = MAX_VOL, + .earpiece_vol_min = MIN_VOL, + }, + { + .mic1_vol_max = 30, + .mic1_vol_min = 10, + .mic2_vol_max = 30, + .mic2_vol_min = 10, + .dmic_vol_max = 0, + .dmic_vol_min = -63, + .linein_vol_max = MAX_VOL, + .linein_vol_min = MIN_VOL, + .hp_vol_max = 6, + .hp_vol_min = -25, + .speaker_vol_max = 6, + .speaker_vol_min = -25, + .earpiece_vol_max = 6, + .earpiece_vol_min = -25, + }, +}; + +#endif /*__INTELMID_H*/ diff --git a/sound/pci/sst/intelmid_pvt.h b/sound/pci/sst/intelmid_pvt.h new file mode 100644 index 0000000..73cb9b5 --- /dev/null +++ b/sound/pci/sst/intelmid_pvt.h @@ -0,0 +1,198 @@ +#ifndef __INTEL_MID_PVT_H__ +#define __INTEL_MID_PVT_H__ +/* + * intelmid_pvt.h - Intel Sound card driver for MID + * + * Copyright (C) 2008-09 Intel Corp + * Authors: Vinod Koul <vinod.koul(a)intel.com> + * Harsha Priya <priya.harsha(a)intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * 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; version 2 of the License. + * + * 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. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * ALSA driver for Intel MID sound card chipset - holding private functions + */ + +static int snd_intelmad_volume_info(struct snd_ctl_elem_info *uinfo, + int control_type, int max, int min) +{ + WARN_ON(!uinfo); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = control_type; + uinfo->value.integer.min = min; + uinfo->value.integer.max = max; + return 0; +} + +#ifdef FULL_CTRL +/** +* mute_all - mutes all the DAC and ADC controls +* @kcontrol: pointer to the control of master mute switch +* @uval: pointer to the structure where the control's info need +* to be filled +* This function is internally called from snd_intelmad_mute_set for master_mute +*/ +static int mute_all(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *uval) +{ + int i = 0, value = 0, ret_val = 0; + struct snd_intelmad *intelmaddata = NULL; + struct snd_pmic_ops *scard_ops = NULL; + int cntl_list[MUTE_ALL_DEV] = + {PMIC_SND_LEFT_HP_MUTE, + PMIC_SND_RIGHT_HP_MUTE, + PMIC_SND_RIGHT_SPEAKER_MUTE, + PMIC_SND_LEFT_SPEAKER_MUTE, + PMIC_SND_MONO_EARPIECE_MUTE, + PMIC_SND_INPUT_MUTE_LINE_IN, + PMIC_SND_INPUT_MUTE_MIC1, + PMIC_SND_INPUT_MUTE_MIC2, + PMIC_SND_INPUT_MUTE_DMIC + }; + + sst_dbg("called\n"); + + WARN_ON(!uval); + WARN_ON(!kcontrol); + + value = uval->value.integer.value[0]; + if (value == 0) + value = 1; + else + value = 0; + + intelmaddata = kcontrol->private_data; + + WARN_ON(!intelmaddata->sstdrv_ops); + + if (PMIC_UNINIT == intelmaddata->pmic_status) { + ret_val = intelmaddata->sstdrv_ops->scard_ops->init_card(); + intelmaddata->pmic_status = PMIC_INIT; + } + + if (0 != ret_val) + return ret_val; + + scard_ops = intelmaddata->sstdrv_ops->scard_ops; + + WARN_ON(!scard_ops); + sst_dbg("************uval::%ld\n", uval->value.integer.value[0]); + sst_dbg("************value::%d\n", value); + for (i = 0; i < MUTE_ALL_DEV; i++) { + ret_val = scard_ops->set_mute(cntl_list[i], value); + if (0 != ret_val) + return ret_val; + } + kcontrol->private_value = 0; + return ret_val; +} +#endif + +void period_elapsed(void *mad_substream) +{ + struct snd_pcm_substream *substream = mad_substream; + struct mad_stream_pvt *stream = substream->runtime->private_data; + + sst_dbg("called\n"); + if (stream->stream_status != RUNNING) + return; + sst_dbg("calling period elapsed\n"); + snd_pcm_period_elapsed(substream); + return; +} + + +int snd_intelmad_alloc_stream(struct snd_pcm_substream *substream) +{ + struct snd_intelmad *intelmaddata = NULL; + struct mad_stream_pvt *stream = substream->runtime->private_data; + struct snd_sst_stream_params param = {{{0,},},}; + struct snd_sst_params str_params = {0}; + int ret_val = 0, stream_type = 0; + unsigned int bits_per_sec = 0; + + intelmaddata = snd_pcm_substream_chip(substream); + + if (SNDRV_PCM_FMTBIT_S8 == substream->runtime->format || + SNDRV_PCM_FMTBIT_U8 == substream->runtime->format) + stream->stream_type = STREAM_TYPE_VOICE; + else + stream->stream_type = STREAM_TYPE_MUSIC; + + stream_type = substream->stream; + sst_dbg("stream type = %d \n", stream_type); + + bits_per_sec = (substream->runtime->sample_bits/8) + * (substream->runtime->channels) + * (substream->runtime->rate); + /*set codec params and inform SST driver the same*/ + + param.uc.pcm_params.codec = SST_CODEC_TYPE_PCM; + param.uc.pcm_params.brate = bits_per_sec; + param.uc.pcm_params.num_chan = substream->runtime->channels; + param.uc.pcm_params.sfreq = substream->runtime->rate; + param.uc.pcm_params.pcm_wd_sz = substream->runtime->sample_bits; + param.uc.pcm_params.frame_size = 0; + param.uc.pcm_params.samples_per_frame = 250; + param.uc.pcm_params.period_count = substream->runtime->period_size; + sst_dbg("period_count = %d\n", param.uc.pcm_params.period_count); + sst_dbg("sfreq= %d, wd_sz = %d\n", param.uc.pcm_params.sfreq, + param.uc.pcm_params.pcm_wd_sz); + + str_params.sparams = param; + str_params.codec = SST_CODEC_TYPE_PCM; + + if (SNDRV_PCM_STREAM_PLAYBACK == stream_type) + str_params.ops = STREAM_OPS_PLAYBACK; + else + str_params.ops = STREAM_OPS_CAPTURE; + ret_val = intelmaddata->sstdrv_ops->control_set(SST_SND_ALLOC, + &str_params); + sst_dbg("SST_SND_PLAY/CAPTURE ret_val = %x\n", + ret_val); + if (ret_val < 0) + return ret_val; + + stream->stream_id = ret_val; + stream->stream_status = INIT; + stream->period_elapsed_ptr = 0; + stream->buf_ptr.buffer_ptr = 0; + sst_dbg("str id : %d\n", stream->stream_id); + + return ret_val; +} + +int snd_intelmad_init_stream(struct snd_pcm_substream *substream) +{ + struct mad_stream_pvt *stream = substream->runtime->private_data; + struct snd_intelmad *intelmaddata = NULL; + int ret_val = 0; + + sst_dbg("setting buffer ptr param\n"); + intelmaddata = snd_pcm_substream_chip(substream); + stream->buf_ptr.str_id = stream->stream_id; + stream->buf_ptr.period_elapsed = period_elapsed; + stream->buf_ptr.mad_substream = substream; + stream->buf_ptr.buffer_ptr = 0; + ret_val = intelmaddata->sstdrv_ops->control_set(SST_SND_STREAM_INIT, + &stream->buf_ptr); + if (0 != ret_val) + sst_err("error code = %d \n", ret_val); + return ret_val; + +} + +#endif diff --git a/sound/pci/sst/intelmid_snd_control.h b/sound/pci/sst/intelmid_snd_control.h new file mode 100644 index 0000000..49fdf9c --- /dev/null +++ b/sound/pci/sst/intelmid_snd_control.h @@ -0,0 +1,153 @@ +#ifndef __INTELMID_SND_CTRL_H__ +#define __INTELMID_SND_CTRL_H__ +/* + * intelmid_snd_control.h - Intel Sound card driver for MID + * + * Copyright (C) 2008-09 Intel Corporation + * Authors: Vinod Koul <vinod.koul(a)intel.com> + * Harsha Priya <priya.harsha(a)intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * 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; version 2 of the License. + * + * 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. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This file defines all snd control functions + */ + +/* +Mask bits +*/ +#define MASK0 0x01 /*0000 0001*/ +#define MASK1 0x02 /*0000 0010*/ +#define MASK2 0x04 /*0000 0100*/ +#define MASK3 0x08 /*0000 1000*/ +#define MASK4 0x10 /*0001 0000*/ +#define MASK5 0x20 /*0010 0000*/ +#define MASK6 0x40 /*0100 0000*/ +#define MASK7 0x80 /*1000 0000*/ +/* +value bits +*/ +#define VALUE0 0x01 /*0000 0001*/ +#define VALUE1 0x02 /*0000 0010*/ +#define VALUE2 0x04 /*0000 0100*/ +#define VALUE3 0x08 /*0000 1000*/ +#define VALUE4 0x10 /*0001 0000*/ +#define VALUE5 0x20 /*0010 0000*/ +#define VALUE6 0x40 /*0100 0000*/ +#define VALUE7 0x80 /*1000 0000*/ + +#define MUTE 0 /* ALSA Passes 0 for mute */ +#define UNMUTE 1 /* ALSA Passes 1 for unmute */ + +#define MAX_VOL_PMIC_VENDOR0 6 /* max volume in dB for stereo & voice DAC */ +#define MIN_VOL_PMIC_VENDOR0 (-57) /* min vol in dB for stereo & voice DAC*/ +/* Head phone volume control */ +#define MAX_HP_VOL_PMIC_VENDOR1 6 /* max volume in dB for HP */ +#define MIN_HP_VOL_PMIC_VENDOR1 (-84) /* min volume in dB for HP */ +#define MAX_HP_VOL_INDX_PMIC_VENDOR1 40 /* Number of HP volume control values */ + +/* Mono Earpiece Volume control */ +#define MAX_EP_VOL_PMIC_VENDOR1 0 /* max volume in dB for EP */ +#define MIN_EP_VOL_PMIC_VENDOR1 (-75) /* min volume in dB for EP */ +#define MAX_EP_VOL_INDX_PMIC_VENDOR1 32 /* Number of EP volume control values */ + +int nc_set_selected_input_dev(int value); +int nc_get_selected_input_dev(int *value); +int nc_set_selected_output_dev(int value); +int nc_get_selected_output_dev(int *value); +int nc_set_mute(int dev_id, int value); +int nc_get_mute(int dev_id, int *value); +int nc_set_vol(int dev_id, int value); +int nc_get_vol(int dev_id, int *value); +int nc_init_card(void); +void nc_set_pcm_params(int sfreq, int word_size); +int nc_init_capture_card(void); + +int fs_set_selected_input_dev(int value); +int fs_get_selected_input_dev(int *value); +int fs_set_selected_output_dev(int value); +int fs_get_selected_output_dev(int *value); +int fs_set_mute(int dev_id, int value); +int fs_get_mute(int dev_id, int *value); +int fs_set_vol(int dev_id, int value); +int fs_get_vol(int dev_id, int *value); +int fs_init_card(void); +void fs_set_pcm_params(int sfreq, int word_size); +int fs_init_capture_card(void); + +int mx_get_selected_input_dev(int *value); +int mx_set_selected_input_dev(int value); +int mx_set_selected_output_dev(int value); +int mx_get_selected_output_dev(int *value); +int mx_set_mute(int dev_id, int value); +int mx_get_mute(int dev_id, int *value); +int mx_set_vol(int dev_id, int value); +int mx_get_vol(int dev_id, int *value); +int mx_init_card(void); +void mx_set_pcm_params(int sfreq, int word_size); +int mx_init_capture_card(void); + + +/*device*/ +enum SND_INPUT_DEVICE { + LINE_IN = 0, + MIC1, + MIC2, + DMIC, + IN_UNDEFINED +}; + +enum SND_OUTPUT_DEVICE { + MONO_HS = 0, + STEREO_HEADPHONE, + INTERNAL_SPKR, + OUT_UNDEFINED +}; + +enum SND_CARDS { + SND_FS = 0, + SND_MX, + SND_NC, +}; + +enum pmic_controls { + PMIC_SND_INPUT_MUTE_LINE_IN = 0x0001, + PMIC_SND_INPUT_MUTE_MIC1 = 0x0002, + PMIC_SND_INPUT_MUTE_MIC2 = 0x0003, + PMIC_SND_INPUT_MUTE_DMIC = 0x0004, + PMIC_SND_INPUT_VOL_MIC1 = 0x0005, + PMIC_SND_INPUT_VOL_MIC2 = 0x0006, + PMIC_SND_INPUT_VOL_DMIC = 0x0007, + PMIC_SND_INPUT_VOL_RIGHT_LINE_IN = 0x0008, + PMIC_SND_INPUT_VOL_LEFT_LINE_IN = 0x0009, +/*Output controls*/ + PMIC_SND_LEFT_HP_VOL = 0x0010, + PMIC_SND_RIGHT_HP_VOL = 0x0011, + PMIC_SND_LEFT_HP_MUTE = 0x0012, + PMIC_SND_RIGHT_HP_MUTE = 0x0013, + PMIC_SND_LEFT_SPEAKER_VOL = 0x0014, + PMIC_SND_RIGHT_SPEAKER_VOL = 0x0015, + PMIC_SND_LEFT_SPEAKER_MUTE = 0x0016, + PMIC_SND_RIGHT_SPEAKER_MUTE = 0x0017, + PMIC_SND_MONO_EARPIECE_VOL = 0x0018, + PMIC_SND_MONO_EARPIECE_MUTE = 0x0019, +/*Other controls*/ + PMIC_MAX_CONTROLS = 0x0019, +}; + +#endif /*__INTELMID_SND_CTRL_H__*/ + + -- 1.5.4.5

1 0

[alsa-devel] [PATCH] [RFC 11/13] Intel SST sound card vendor2 module
by Vinod Koul 03 Jul '09

03 Jul '09

This adds the support for vendor PMICs for widget control and configurations. The MID platform supports three different vendor implementation of PMIC. The sound card, ie all analog components like DAC, ADC, mixer settings are different for three vendors. This patch implements these settings for one of the vendor Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> Signed-off-by: R Dharageswari <dharageswari.r(a)intel.com> new file: sound/pci/sst/intelmid_v2_control.c --- sound/pci/sst/intelmid_v2_control.c | 629 +++++++++++++++++++++++++++++++++++ 1 files changed, 629 insertions(+), 0 deletions(-) create mode 100644 sound/pci/sst/intelmid_v2_control.c diff --git a/sound/pci/sst/intelmid_v2_control.c b/sound/pci/sst/intelmid_v2_control.c new file mode 100644 index 0000000..21e3cdb --- /dev/null +++ b/sound/pci/sst/intelmid_v2_control.c @@ -0,0 +1,629 @@ +/* + * intelmid_v2_control.c - Intel Sound card driver for MID + * + * Copyright (C) 2008-09 Intel Corp + * Authors: Vinod Koul <vinod.koul(a)intel.com> + * Harsha Priya <priya.harsha(a)intel.com> + * R Dharageswari <dharageswari.r(a)intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * 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; version 2 of the License. + * + * 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. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This file contains the control operations of vendor 3 + */ + +#include <linux/cdev.h> +#include <linux/pci.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/module.h> +#include <linux/syscalls.h> +#include <linux/file.h> +#include <linux/fcntl.h> +#include <linux/uaccess.h> +#include <linux/interrupt.h> +#include <linux/list.h> +#include <linux/slab.h> +#include <linux/mm.h> +#include <linux/workqueue.h> +#include <linux/pci.h> +#include <linux/firmware.h> +#include <sound/intel_sst.h> +#include <sound/intel_sst_ioctl.h> +#include "intel_sst_fw_ipc.h" +#include "intel_sst_common.h" +#include "intel_sst_pvt.h" +#include "intelmid_snd_control.h" + +#include <asm/ipc_defs.h> +enum reg_v3 { + VAUDIOCNT = 0x51, + VOICEPORT1 = 0x100, + VOICEPORT2 = 0x101, + AUDIOPORT1 = 0x102, + AUDIOPORT2 = 0x103, + ADCSAMPLERATE = 0x104, + DMICCTRL1 = 0x105, + DMICCTRL2 = 0x106, + MICCTRL = 0x107, + MICSELVOL = 0x108, + LILSEL = 0x109, + LIRSEL = 0x10a, + VOICEVOL = 0x10b, + AUDIOLVOL = 0x10c, + AUDIORVOL = 0x10d, + LMUTE = 0x10e, + RMUTE = 0x10f, + POWERCTRL1 = 0x110, + POWERCTRL2 = 0x111, + DRVPOWERCTRL = 0x112, + VREFPLL = 0x113, + PCMBUFCTRL = 0x114, + SOFTMUTE = 0x115, + DTMFPATH = 0x116, + DTMFVOL = 0x117, + DTMFFREQ = 0x118, + DTMFHFREQ = 0x119, + DTMFLFREQ = 0x11a, + DTMFCTRL = 0x11b, + DTMFASON = 0x11c, + DTMFASOFF = 0x11d, + DTMFASINUM = 0x11e, + CLASSDVOL = 0x11f, + VOICEDACAVOL = 0x120, + AUDDACAVOL = 0x121, + LOMUTEVOL = 0x122, + HPLVOL = 0x123, + HPRVOL = 0x124, + MONOVOL = 0x125, + LINEOUTMIXVOL = 0x126, + EPMIXVOL = 0x127, + LINEOUTLSEL = 0x128, + LINEOUTRSEL = 0x129, + EPMIXOUTSEL = 0x12a, + HPLMIXSEL = 0x12b, + HPRMIXSEL = 0x12c, + LOANTIPOP = 0x12d, +}; + + +int nc_init_card(void) +{ + struct sc_reg_access sc_access[] = { + + {VAUDIOCNT, 0x27, 0}, + {VOICEPORT1, 0xd5, 0}, + {VOICEPORT2, 0x08, 0}, + {AUDIOPORT1, 0x98, 0}, + {AUDIOPORT2, 0x09, 0}, + {AUDIOLVOL, 0x0e, 0}, + {AUDIORVOL, 0x0e, 0}, + {LMUTE, 0x03, 0}, + {RMUTE, 0x03, 0}, + {POWERCTRL1, 0xff, 0}, + {POWERCTRL2, 0xff, 0}, + {DRVPOWERCTRL, 0xff, 0}, + {VREFPLL, 0xff , 0}, + {HPLMIXSEL, 0xef, 0}, + {HPRMIXSEL, 0xf7, 0}, + {VAUDIOCNT, 0x27, 0}, + {PCMBUFCTRL, 0x13, 0}, + {DTMFPATH, 0x02, 0}, + {DTMFCTRL, 0x00, 0}, + {VOICEPORT1, 0xd5, 0}, + {VOICEPORT2, 0x08, 0}, + {VREFPLL, 0x3f, 0}, + {POWERCTRL1, 0xff, 0}, + {POWERCTRL2, 0xff, 0}, + {DRVPOWERCTRL, 0xff, 0}, + {HPLMIXSEL, 0xde, 0}, + {HPRMIXSEL, 0xde, 0}, + {VOICEVOL, 0x0e, 0}, + {LMUTE, 0x01, 0}, + {AUDIOPORT1, 0xa8, 0}, + {AUDIOPORT2, 0x17, 0}, + {AUDIOLVOL, 0x0e, 0}, + {AUDIORVOL, 0x0e, 0}, + {LMUTE, 0x03, 0}, + {RMUTE, 0x03, 0}, + {HPLMIXSEL, 0xef, 0}, + {HPRMIXSEL, 0xf7, 0}, + }; + sst_sc_reg_access(sc_access, PMIC_WRITE, 37); + sst_dbg("complete!!\n"); + return 0; +} + +int nc_init_capture_card(void) +{ + struct sc_reg_access sc_access_capture[] = { + /*voice port enabling workaorund*/ + {0x03A, 0x05, 0x00}, + {0x051, 0x27, 0x00}, + {0x113, 0x3A, 0x00}, + {0x100, 0x90, 0x00}, + {0x101, 0x07, 0x00}, + {0x114, 0x33, 0x00}, + {0x107, 0x10, 0x00}, + {0x104, 0x83, 0x00}, + {0x12B, 0xDE, 0x00}, + {0x12C, 0xDE, 0x00}, + {0x108, 0x53, 0x00}, + {0x10A, 0x0C, 0x00}, + {0x10B, 0x0E, 0x00}, + {0x123, 0x0C, 0x00}, + {0x124, 0x0C, 0x00}, + {0x10E, 0x03, 0x00}, + {0x10F, 0x03, 0x00}, + {0x110, 0xD4, 0x00}, + {0x111, 0x0C, 0x00}, + {0x112, 0x06, 0x00}, + + /*audio port*/ + {0x051, 0x27, 0x00}, + {0x113, 0x35, 0x00}, + {0x102, 0x90, 0x00}, + {0x103, 0x00, 0x00}, + {0x114, 0x33, 0x00}, + {0x107, 0x11, 0x00}, + {0x104, 0x83, 0x00}, + {0x12B, 0xE6, 0x00}, + {0x12C, 0xE6, 0x00}, + {0x108, 0x53, 0x00}, + {0x10A, 0x0C, 0x00}, + {0x10B, 0x0E, 0x00}, + {0x123, 0x0C, 0x00}, + {0x124, 0x0C, 0x00}, + {0x10E, 0x03, 0x00}, + {0x10F, 0x03, 0x00}, + {0x110, 0xFF, 0x00}, + {0x111, 0x0C, 0x00}, + {0x112, 0x06, 0x00}, + }; + sst_dbg("calling capture init\n"); + sst_sc_reg_access(sc_access_capture, PMIC_WRITE, 39); + sst_dbg("complete!!\n"); + return 0; +} + +void nc_set_pcm_params(int sfreq, int word_size) +{ + int config2 = 0; + struct sc_reg_access sc_access; + + switch (word_size) { + case 16: + switch (sfreq) { + case 8000: + config2 = 0x00; + break; + case 11025: + config2 = 0x01; + break; + case 12000: + config2 = 0x02; + break; + case 16000: + config2 = 0x03; + break; + case 22050: + config2 = 0x04; + break; + case 24000: + config2 = 0x05; + break; + case 32000: + config2 = 0x07; + break; + case 44100: + config2 = 0x08; + break; + case 48000: + config2 = 0x09; + break; + } + sst_dbg("word_size = %d\n", word_size); + sc_access.reg_addr = AUDIOPORT1; + sc_access.value = 0x98; + sst_sc_reg_access(&sc_access, PMIC_WRITE, 1); + sc_access.reg_addr = AUDIOPORT2; + sc_access.value = config2; + sst_sc_reg_access(&sc_access, PMIC_WRITE, 1); + break; + case 24: + switch (sfreq) { + case 8000: + config2 = 0x10; + break; + case 11025: + config2 = 0x11; + break; + case 12000: + config2 = 0x12; + break; + case 16000: + config2 = 0x13; + break; + case 22050: + config2 = 0x14; + break; + case 24000: + config2 = 0x15; + break; + case 32000: + config2 = 0x17; + break; + case 44100: + config2 = 0x18; + break; + case 48000: + config2 = 0x19; + break; + } + sst_dbg("word_size = %d\n", word_size); + sc_access.reg_addr = AUDIOPORT1; + sc_access.value = 0xAB; + sst_sc_reg_access(&sc_access, PMIC_WRITE, 1); + sc_access.reg_addr = AUDIOPORT2; + sc_access.value = config2; + sst_sc_reg_access(&sc_access, PMIC_WRITE, 1); + break; + + } + sst_dbg("config2 = 0x%x\n", config2); + return; + +} +int nc_set_selected_output_dev(int value) +{ + int retval = 0; + struct sc_reg_access sc_access_MHS[] = { + {RMUTE, 0X02, 0}, + {POWERCTRL2, 0Xff, 0}, + {DRVPOWERCTRL, 0xff, 0}, + {VREFPLL, 0xff, 0}, + {LMUTE, 0x02, 0}, + {HPLMIXSEL, 0x00, 0}, + {HPRMIXSEL, 0x00, 0}, + {POWERCTRL1, 0xFF, 0}, + {AUDIOLVOL, 0x00, 0}, + {AUDIORVOL, 0x00, 0}, + {LOMUTEVOL, 0x00, 0} + }; + + struct sc_reg_access sc_access_SH[] = { + {LMUTE, 0x03, 0x00}, + {RMUTE, 0x02, 0x00} + }; + + struct sc_reg_access sc_access_IS[] = { + {LMUTE, 0x05, 0x00}, + {RMUTE, 0x04, 0x00} + }; + + + sst_dbg("nc set selected output:%d", value); + + switch (value) { + case STEREO_HEADPHONE: + retval = sst_sc_reg_access(sc_access_SH, PMIC_WRITE, 2); + break; + + case INTERNAL_SPKR: + retval = sst_sc_reg_access(sc_access_IS, PMIC_WRITE, 2); + break; + + case MONO_HS: + retval = sst_sc_reg_access(sc_access_MHS, PMIC_WRITE, 11); + break; + default: + sst_err("rcvd illegal request:"); + return -EINVAL; + } + return retval; +} + +int nc_set_mute(int dev_id, int value) +{ + int retval = 0; + struct sc_reg_access sc_access; + + sst_dbg("set device id::%d\n", dev_id); + sst_dbg("set device value::%d\n", value); + switch (dev_id) { + case PMIC_SND_INPUT_MUTE_MIC1: + sst_dbg("PMIC_SND_INPUT_MUTE_MIC1: value::%d", value); + if (value == UNMUTE) { + /* unmute the system, set the 6th bit to one */ + sc_access.value = 0x00; + } else { + /* mute the system, reset the 6th bit to zero */ + sc_access.value = 0x40; + } + sc_access.reg_addr = LILSEL; + sc_access.mask = MASK6; + break; + case PMIC_SND_INPUT_MUTE_MIC2: + sst_dbg("PMIC_SND_INPUT_MUTE_MIC2: value::%d", value); + if (value == UNMUTE) { + /* unmute the system, set the 6th bit to one */ + sc_access.value = 0x00; + } else { + /* mute the system, reset the 6th bit to zero */ + sc_access.value = 0x40; + } + sc_access.reg_addr = LIRSEL; + sc_access.mask = MASK6; + break; + case PMIC_SND_INPUT_MUTE_DMIC: + sst_dbg("PMIC_SND_INPUT_MUTE_DMIC: value::%d", value); + if (value == UNMUTE) { + /* unmute the system, set the 6th bit to one */ + sc_access.value = 0x00; + } else { + /* mute the system, reset the 6th bit to zero */ + sc_access.value = 0x40; + } + sc_access.reg_addr = DMICCTRL1; + sc_access.mask = MASK6; + break; + case PMIC_SND_LEFT_HP_MUTE: + case PMIC_SND_RIGHT_HP_MUTE: + if (value == UNMUTE) + sc_access.value = 0x0; + else + sc_access.value = 0x04; + if (dev_id == PMIC_SND_LEFT_HP_MUTE) { + sc_access.reg_addr = LMUTE; + sst_dbg("PMIC_SND_LEFT_HP_MUTE:: value::%d\n", + sc_access.value); + } else { + sc_access.reg_addr = RMUTE; + sst_dbg("PMIC_SND_RIGHT_HP_MUTE:: value::%d\n", + sc_access.value); + } + sc_access.mask = MASK2; + break; + case PMIC_SND_MONO_EARPIECE_MUTE: + if (value == UNMUTE) + sc_access.value = 0x00; + else + sc_access.value = 0x02; + sc_access.reg_addr = LMUTE; + sst_dbg("PMIC_SND_MONO_EARPIECE_MUTE::%d\n", sc_access.value); + sc_access.mask = MASK1; + break; + default: + sst_dbg("Invalid Device_id\n"); + return -EINVAL; + } + retval = sst_sc_reg_access(&sc_access, PMIC_READ_MODIFY, 1); + return retval; + +} + +int nc_set_vol(int dev_id, int value) +{ + int retval = 0, MAX_VOL = 6; + struct sc_reg_access sc_access; + + sst_dbg("set volume:%d", dev_id); + switch (dev_id) { + case PMIC_SND_INPUT_VOL_MIC1: + sst_dbg("PMIC_SND_INPUT_VOL_MIC1:: value::%d", value); + sc_access.value = abs(value); + sc_access.reg_addr = LILSEL; + sc_access.mask = MASK0|MASK1|MASK2|MASK3|MASK4|MASK5; + break; + case PMIC_SND_INPUT_VOL_MIC2: + sst_dbg("PMIC_SND_INPUT_VOL_MIC2:: value::%d", value); + sc_access.value = abs(value); + sc_access.reg_addr = LIRSEL; + sc_access.mask = MASK0|MASK1|MASK2|MASK3|MASK4|MASK5; + break; + case PMIC_SND_INPUT_VOL_DMIC: + sst_dbg("PMIC_SND_INPUT_VOL_DMIC:: value::%d", value); + sc_access.value = abs(value); + sc_access.reg_addr = DMICCTRL1; + sc_access.mask = MASK0|MASK1|MASK3|MASK4|MASK5|MASK6; + break; + case PMIC_SND_INPUT_VOL_RIGHT_LINE_IN: + break; + case PMIC_SND_INPUT_VOL_LEFT_LINE_IN: + break; + case PMIC_SND_LEFT_HP_VOL: + sc_access.value = MAX_VOL - value; + sst_dbg("PMIC_SND_LEFT_HP_VOL:%d", value); + sc_access.reg_addr = HPLVOL; + sc_access.mask = (MASK0|MASK1|MASK2|MASK3|MASK4); + break; + + case PMIC_SND_RIGHT_HP_VOL: + sc_access.value = MAX_VOL-value; + sst_dbg("PMIC_SND_RIGHT_HP_VOL: value::%d\n", value); + sc_access.reg_addr = HPRVOL; + sc_access.mask = (MASK0|MASK1|MASK2|MASK3|MASK4); + break; + + case PMIC_SND_MONO_EARPIECE_VOL: + sc_access.value = MAX_VOL-value; + sst_dbg("PMIC_MONO_EARPIECE_VOL: value::%d", value); + sc_access.reg_addr = MONOVOL; + sc_access.mask = (MASK0|MASK1|MASK2|MASK3|MASK4); + break; + default: + sst_dbg("Invalid Device_id\n"); + return -EINVAL;; + + } + /*sst_sc_read_modify(&reg_adrs, &difference, 1);*/ + retval = sst_sc_reg_access(&sc_access, PMIC_READ_MODIFY, 1); + return retval; +} + +int nc_set_selected_input_dev(int value) +{ + struct sc_reg_access sc_access[3]; + int retval = 0; + u8 num_val; + + sst_dbg("nc set selected input:%d", value); + + switch (value) { + case MIC1: + sst_dbg("MIC1\n"); + sc_access[0].reg_addr = LILSEL; + sc_access[0].mask = MASK7; + sc_access[0].value = 0x00; + num_val = 1; + break; + + case MIC2: + sst_dbg("MIC2\n"); + sc_access[0].reg_addr = LIRSEL; + sc_access[0].value = 0x06; + sc_access[0].mask = MASK0|MASK1|MASK2|MASK3|MASK4| + MASK5|MASK6|MASK7; + + sc_access[1].reg_addr = MICSELVOL; + sc_access[1].value = 0x13; + sc_access[1].mask = MASK0|MASK1|MASK4 ; + + sc_access[2].reg_addr = MICCTRL; + sc_access[2].value = 0x10; + sc_access[2].mask = MASK4; + num_val = 3; + + break; + + case DMIC: + sst_dbg("DMIC\n"); + sc_access[0].reg_addr = DMICCTRL1; + sc_access[0].value = 0x40; + sc_access[0].mask = MASK6; + num_val = 1; + + break; + default: + sst_err("rcvd illegal request:"); + return -EINVAL; + } + retval = sst_sc_reg_access(sc_access, PMIC_READ_MODIFY, num_val); + return retval; +} + +int nc_get_selected_output_dev(int *value) +{ + return 0; +} + +int nc_get_mute(int dev_id, int *value) +{ + int retval = 0, mask = 0; + struct sc_reg_access sc_access = {0,}; + + sst_dbg("get mute::%d\n", dev_id); + + switch (dev_id) { + case PMIC_SND_INPUT_MUTE_MIC1: + sst_dbg("PMIC_SND_INPUT_MUTE_MIC1\n"); + sc_access.reg_addr = LILSEL; + mask = MASK6; + break; + case PMIC_SND_INPUT_MUTE_MIC2: + sst_dbg("PMIC_SND_INPUT_MUTE_MIC2\n"); + sc_access.reg_addr = LIRSEL; + mask = MASK6; + break; + case PMIC_SND_LEFT_HP_MUTE: + case PMIC_SND_RIGHT_HP_MUTE: + mask = MASK2; + sst_dbg("PMIC_SN_LEFT/RIGHT_HP_MUTE\n"); + if (dev_id == PMIC_SND_RIGHT_HP_MUTE) + sc_access.reg_addr = RMUTE; + else + sc_access.reg_addr = LMUTE; + break; + + case PMIC_SND_MONO_EARPIECE_MUTE: + sst_dbg("PMIC_MONO_EARPIECE_MUTE\n"); + sc_access.reg_addr = RMUTE; + mask = MASK1; + break; + default: + sst_dbg("Invalid Device_id\n"); + return -EINVAL; + + } + retval = sst_sc_reg_access(&sc_access, PMIC_READ_MODIFY, 1); + sst_dbg("reg value = %d\n", sc_access.value); + if (retval != 0) + return retval; + *value = (sc_access.value) & mask; + if (*value) + *value = 0; + else + *value = 1; + sst_dbg("value returned = 0x%x\n", *value); + return retval; +} + +int nc_get_vol(int dev_id, int *value) +{ + int retval = 0, mask = 0, MAX_VOL = 6; + struct sc_reg_access sc_access = {0,}; + + switch (dev_id) { + case PMIC_SND_INPUT_VOL_MIC1: + sst_dbg("PMIC_SND_INPUT_VOL_MIC1\n"); + sc_access.reg_addr = LILSEL; + mask = (MASK0|MASK1|MASK2|MASK3|MASK4|MASK5); + break; + case PMIC_SND_INPUT_VOL_MIC2: + sst_dbg("PMIC_SND_INPUT_VOL_MIC2\n"); + sc_access.reg_addr = LIRSEL; + mask = (MASK0|MASK1|MASK2|MASK3|MASK4|MASK5); + break; + case PMIC_SND_INPUT_VOL_DMIC: + sst_dbg("PMIC_SND_INPUT_VOL_DMIC\n"); + break; + case PMIC_SND_LEFT_HP_VOL: + sst_dbg("GET_VOLUME_PMIC_LEFT_HP_VOL\n"); + sc_access.reg_addr = HPLVOL; + mask = (MASK0|MASK1|MASK2|MASK3|MASK4); + break; + case PMIC_SND_RIGHT_HP_VOL: + sst_dbg("GET_VOLUME_PMIC_RIGHT_HP_VOL\n"); + sc_access.reg_addr = HPRVOL; + mask = (MASK0|MASK1|MASK2|MASK3|MASK4); + break; + case PMIC_SND_MONO_EARPIECE_VOL: + sst_dbg("GET_VOLUME_MONOEARPIECE_VOL\n"); + sc_access.reg_addr = MONOVOL; + mask = (MASK0|MASK1|MASK2|MASK3|MASK4); + break; + default: + sst_dbg("Invalid Device_id = %d\n", dev_id); + return -EINVAL; + + } + retval = sst_sc_reg_access(&sc_access, PMIC_READ, 1); + *value = (sc_access.value) & mask; + *value = MAX_VOL - *value; + sst_dbg("value returned = 0x%x\n", *value); + return retval; +} + -- 1.5.4.5

1 0

[alsa-devel] [PATCH] [RFC 10/13] Intel SST sound card vendor2 module
by Vinod Koul 03 Jul '09

03 Jul '09

This adds the support for vendor PMICs for widget control and configurations. The MID platform supports three different vendor implementation of PMIC. The sound card, ie all analog components like DAC, ADC, mixer settings are different for three vendors. This patch implements these settings for one of the vendor. Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> Signed-off-by: R Dharageswari <dharageswari.r(a)intel.com> new file: sound/pci/sst/intelmid_v1_control.c --- sound/pci/sst/intelmid_v1_control.c | 637 +++++++++++++++++++++++++++++++++++ 1 files changed, 637 insertions(+), 0 deletions(-) create mode 100644 sound/pci/sst/intelmid_v1_control.c diff --git a/sound/pci/sst/intelmid_v1_control.c b/sound/pci/sst/intelmid_v1_control.c new file mode 100644 index 0000000..32c2c51 --- /dev/null +++ b/sound/pci/sst/intelmid_v1_control.c @@ -0,0 +1,637 @@ +/* + * intelmid_v1_control.c - Intel SST Driver for audio engine + * + * Copyright (C) 2008-09 Intel Corp + * Authors: Vinod Koul <vinod.koul(a)intel.com> + * Harsha Priya <priya.harsha(a)intel.com> + * R Dharageswari <dharageswari.r(a)intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * 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; version 2 of the License. + * + * 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. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This file contains the control operations of vendor 3 + */ + +#include <linux/cdev.h> +#include <linux/pci.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/module.h> +#include <linux/syscalls.h> +#include <linux/file.h> +#include <linux/fcntl.h> +#include <linux/uaccess.h> +#include <linux/interrupt.h> +#include <linux/list.h> +#include <linux/slab.h> +#include <linux/mm.h> +#include <linux/workqueue.h> +#include <linux/pci.h> +#include <linux/firmware.h> +#include <sound/intel_sst.h> +#include <sound/intel_sst_ioctl.h> +#include "intel_sst_fw_ipc.h" +#include "intel_sst_common.h" +#include "intel_sst_pvt.h" +#include "intelmid_snd_control.h" + +#include <asm/ipc_defs.h> +enum _reg_v2 { + + MASTER_CLOCK_PRESCALAR = 0x205, + SET_MASTER_AND_LR_CLK1 = 0x20b, + SET_MASTER_AND_LR_CLK2 = 0x20c, + MASTER_MODE_AND_DATA_DELAY = 0x20d, + DIGITAL_INTERFACE_TO_DAI2 = 0x20e, + CLK_AND_FS1 = 0x208, + CLK_AND_FS2 = 0x209, + DAI2_TO_DAC_HP = 0x210, + HP_OP_SINGLE_ENDED = 0x224, + ENABLE_OPDEV_CTRL = 0x226, + ENABLE_DEV_AND_USE_XTAL = 0x227, + + /* Max audio subsystem (PQ49) MAX 8921 */ + AS_IP_MODE_CTL = 0xF9, + AS_LEFT_SPKR_VOL_CTL = 0xFA, /* Mono Earpiece volume control */ + AS_RIGHT_SPKR_VOL_CTL = 0xFB, + AS_LEFT_HP_VOL_CTL = 0xFC, + AS_RIGHT_HP_VOL_CTL = 0xFD, + AS_OP_MIX_CTL = 0xFE, + AS_CONFIG = 0xFF, + + /* Headphone volume control & mute registers */ + VOL_CTRL_LT = 0x21c, + VOL_CTRL_RT = 0x21d, + +}; + +/* Headphone volume control values in dB */ +int mx_hp_vol[MAX_HP_VOL_INDX_PMIC_VENDOR1] = +{ + (+6), + (+6), /* +5.5, commented as this is not an integer */ + (+5), + (+5), /* +4.5, commented as this is not an integer */ + (+4), + (+4), /* +3.5, commented as this is not an integer */ + (+3), + (+2), + (+1), + 0, + (-1), + (-2), + (-3), + (-4), + (-5), + (-6), + (-8), + (-10), + (-12), + (-14), + (-16), + (-18), + (-20), + (-22), + (-26), + (-30), + (-34), + (-38), + (-42), + (-46), + (-50), + (-54), + (-58), + (-62), + (-66), + (-70), + (-74), + (-78), + (-82), + (-84) +}; /* mx_hp_vol[] */ + +/* Mono Earpiece volume control values in dB */ +int mx_ep_vol[MAX_EP_VOL_INDX_PMIC_VENDOR1] = +{ + (-75), + (-71), + (-67), + (-63), + (-59), + (-55), + (-51), + (-47), + (-44), + (-41), + (-38), + (-35), + (-32), + (-29), + (-26), + (-23), + (-21), + (-19), + (-17), + (-15), + (-13), + (-11), + (-9), + (-7), + (-6), + (-5), + (-4), + (-3), + (-2), + (-1), + (0) +}; /* mx_ep_vol[] */ + +int mx_init_card(void) +{ + + struct sc_reg_access sc_access[17] = {{0,},}; + int retval = 0; + sst_dbg("mx initializing card...\n"); + + sc_access[0].reg_addr = MASTER_CLOCK_PRESCALAR; + sc_access[0].value = 0x10; + sc_access[1].reg_addr = SET_MASTER_AND_LR_CLK1; + sc_access[1].value = 0x60; + sc_access[2].reg_addr = SET_MASTER_AND_LR_CLK2; + sc_access[2].value = 0x00; + sc_access[3].reg_addr = MASTER_MODE_AND_DATA_DELAY ; + sc_access[3].value = 0x90; + sc_access[4].reg_addr = DIGITAL_INTERFACE_TO_DAI2; + sc_access[4].value = 0x51; + sc_access[5].reg_addr = CLK_AND_FS1; + sc_access[5].value = 0x90; + sc_access[6].reg_addr = CLK_AND_FS2; + sc_access[6].value = 0x51;/*Mx fix for 50Mhz*/ + sc_access[7].reg_addr = DAI2_TO_DAC_HP; + sc_access[7].value = 0x21; + sc_access[8].reg_addr = VOL_CTRL_LT; + sc_access[8].value = 0x09; + sc_access[9].reg_addr = VOL_CTRL_RT; + sc_access[9].value = 0x00; + sc_access[10].reg_addr = HP_OP_SINGLE_ENDED; + sc_access[10].value = 0x04; + sc_access[11].reg_addr = ENABLE_OPDEV_CTRL; + sc_access[11].value = 0x0c; + sc_access[12].reg_addr = ENABLE_DEV_AND_USE_XTAL ; + sc_access[12].value = 0x88; + /* mono earpiece - default register configuration */ + /* Input mode control - zero crossing detection + * enabled */ + sc_access[13].reg_addr = AS_IP_MODE_CTL; + sc_access[13].value = 0x40; + + /* Left speaker volume (Mono EP) (-6) dB */ + sc_access[14].reg_addr = AS_LEFT_SPKR_VOL_CTL; + sc_access[14].value = 0x19; + + /* output mixer -INA1 + INA2 enabled */ + sc_access[15].reg_addr = AS_OP_MIX_CTL; + sc_access[15].value = 0x60; + /* configuration - Disable the audio subsytem (Max 8921) */ + sc_access[16].reg_addr = AS_CONFIG; + sc_access[16].value = 0x00; + + retval = sst_sc_reg_access(sc_access, PMIC_WRITE, 17); + if (0 != retval) { + sst_dbg("pmic communication failed \n"); + return retval; + } + + sst_dbg("mx Initilaisation complete!!\n"); + return 0; +} + +int mx_init_capture_card(void) +{ + return 0; +} + +void mx_set_pcm_params(int sfreq, int word_size) +{ + int config1 = 0, config2 = 0; + int retval = 0; + struct sc_reg_access sc_access[4] = {{0,},}; + + switch (sfreq) { + case 8000: + config1 = 0x10; + config2 = 0x00; + break; + case 11025: + config1 = 0x16; + config2 = 0x0d; + break; + case 12000: + config1 = 0x18; + config2 = 0x00; + break; + case 16000: + config1 = 0x20; + config2 = 0x00; + break; + case 22050: + config1 = 0x2c; + config2 = 0x1a; + break; + case 24000: + config1 = 0x30; + config2 = 0x00; + break; + case 32000: + config1 = 0x40; + config2 = 0x00; + break; + case 44100: + config1 = 0x58; + config2 = 0x33; + break; + case 48000: + config1 = 0x60; + config2 = 0x00; + break; + } + + sc_access[0].reg_addr = SET_MASTER_AND_LR_CLK1; + sc_access[0].value = config1; + sc_access[1].reg_addr = SET_MASTER_AND_LR_CLK2; + sc_access[1].value = config2; + switch (word_size) { + case 16: + config1 = 0x51; + config2 = 0x31; + break; + case 24: + config1 = 0x52; + config2 = 0x32; + break; + } + + sst_dbg("config1 = %x\n", config1); + sst_dbg("config2 = %x\n", config2); + sc_access[2].reg_addr = DIGITAL_INTERFACE_TO_DAI2; + sc_access[2].value = config1; + sc_access[3].reg_addr = CLK_AND_FS2; + sc_access[3].value = config2; + retval = sst_sc_reg_access(sc_access, PMIC_WRITE, 4); + if (0 != retval) + sst_dbg("pmic communication failed \n"); + return; +} /* mx_set_pcm_params */ + +int mx_set_selected_output_dev(int dev_id) +{ + struct sc_reg_access sc_access[4] = {{0,},}; + int retval = 0; + + sst_dbg("mx_set_selected_input_dev dev_id:0x%x\n", dev_id); + switch (dev_id) { + case INTERNAL_SPKR: + break; + + case STEREO_HEADPHONE: + /* Enable stereo headphones - Begin */ + /* Volume Left Un-Mute */ + sc_access[0].reg_addr = VOL_CTRL_LT; + sc_access[0].value = 0x00; + sc_access[0].mask = MASK6; + /* Volume right Un-Mute */ + sc_access[1].reg_addr = VOL_CTRL_RT; + sc_access[1].value = 0x00; + sc_access[1].mask = MASK6; + /* Enable stereo headphones - End */ + + /* Disable Mono earpiece - Begin */ + /* configuration - AS Disable */ + sc_access[2].reg_addr = AS_CONFIG; + sc_access[2].value = 0x00; + sc_access[2].mask = MASK7; + + /* Line output (L&R) Disable */ + sc_access[3].reg_addr = ENABLE_OPDEV_CTRL; + sc_access[3].value = 0x00; + sc_access[3].mask = MASK4|MASK5; + retval = sst_sc_reg_access(sc_access, PMIC_READ_MODIFY, 4); + if (0 != retval) + return retval; + /* Disable Mono earpiece - End */ + break; + + case MONO_HS: + /* select mono earpiece - Begin */ + /* configuration - AS enable, left speaker enable */ + sc_access[0].reg_addr = AS_CONFIG; + sc_access[0].value = 0xA0; + sc_access[0].mask = (MASK5|MASK7); + /* Line output (L&R), DAC output (L&R) enable */ + sc_access[1].reg_addr = ENABLE_OPDEV_CTRL; + sc_access[1].value = 0x3C; + sc_access[1].mask = (MASK2|MASK3|MASK4|MASK5); + retval = sst_sc_reg_access(sc_access, PMIC_READ_MODIFY, 2); + if (0 != retval) + return retval; + /* select mono earpiece -end */ + break; + } /* switch (dev_id) */ + + return retval; +} /* mx_set_selected_output_dev */ + +int mx_set_selected_input_dev(int value) +{ + return 0; +} /* mx_set_selected_input_dev */ + +int mx_set_mute(int dev_id, int value) +{ + struct sc_reg_access sc_access[1] = {{0,},}; + int retval = 0; + + sst_dbg("mx_set_mute dev_id:0x%x , value:%d \n", dev_id, value); + + /* MAD passes value:0 for mute, value:1 for unmute */ + switch (dev_id) { + case PMIC_SND_INPUT_MUTE_LINE_IN: + break; + + case PMIC_SND_INPUT_MUTE_MIC1: + break; + + case PMIC_SND_INPUT_MUTE_MIC2: + break; + + case PMIC_SND_INPUT_MUTE_DMIC: + break; + + case PMIC_SND_LEFT_SPEAKER_MUTE: + break; + + case PMIC_SND_RIGHT_SPEAKER_MUTE: + break; + + case PMIC_SND_LEFT_HP_MUTE: + sc_access[0].reg_addr = VOL_CTRL_LT; + if (value == MUTE) + sc_access[0].value = 0x40; + else + sc_access[0].value = 0x00; + sc_access[0].mask = MASK6; + retval = sst_sc_reg_access(sc_access, PMIC_READ_MODIFY, 1); + break; + + case PMIC_SND_RIGHT_HP_MUTE: + sc_access[0].reg_addr = VOL_CTRL_RT; + if (value == MUTE) + sc_access[0].value = 0x40; + else + sc_access[0].value = 0x00; + sc_access[0].mask = MASK6; + retval = sst_sc_reg_access(sc_access, PMIC_READ_MODIFY, 1); + break; + + case PMIC_SND_MONO_EARPIECE_MUTE: + sc_access[0].reg_addr = AS_LEFT_SPKR_VOL_CTL; + if (value == MUTE) + sc_access[0].value = 0x00; + else + sc_access[0].value = 0x19; + sc_access[0].mask = (MASK0|MASK1|MASK2|MASK3|MASK4); + retval = sst_sc_reg_access(sc_access, PMIC_READ_MODIFY, 1); + break; + } /* switch (dev_id) */ + + return retval; +} /* mx_set_mute */ + +int mx_set_vol(int dev_id, int value) +{ + struct sc_reg_access sc_access[1] = {{0,},}; + int retval = 0, i; + + sst_dbg("mx_set_vol dev_id:0x%x , value:%d \n", dev_id, value); + switch (dev_id) { + case PMIC_SND_INPUT_VOL_MIC1: + case PMIC_SND_INPUT_VOL_MIC2: + break; + case PMIC_SND_INPUT_VOL_DMIC: + break; + case PMIC_SND_LEFT_SPEAKER_VOL: + break; + case PMIC_SND_RIGHT_SPEAKER_VOL: + break; + case PMIC_SND_LEFT_HP_VOL: + for (i = 0; i < MAX_HP_VOL_INDX_PMIC_VENDOR1; i++) { + if (value >= mx_hp_vol[i]) { + sc_access[0].value = i; + break; /* exit out of the for loop */ + } /* if(value >= mx_hp_vol[i]) */ + + } /* for(i=0; i< MAX_HP_VOL_INDX_PMIC_VENDOR1; i++) */ + sc_access[0].reg_addr = VOL_CTRL_LT; + sc_access[0].mask = (MASK0|MASK1|MASK2|MASK3|MASK4|MASK5); + break; + case PMIC_SND_RIGHT_HP_VOL: + for (i = 0; i < MAX_HP_VOL_INDX_PMIC_VENDOR1; i++) { + if (value >= mx_hp_vol[i]) { + sc_access[0].value = i; + break; /* exit out of the for loop */ + } /* if(value >= mx_hp_vol[i]) */ + } /* for(i=0; i< MAX_HP_VOL_INDX_PMIC_VENDOR1; i++) */ + sc_access[0].reg_addr = VOL_CTRL_RT; + sc_access[0].mask = (MASK0|MASK1|MASK2|MASK3|MASK4|MASK5); + break; + case PMIC_SND_MONO_EARPIECE_VOL: + for (i = 0; i < MAX_EP_VOL_INDX_PMIC_VENDOR1; i++) { + if (value <= mx_ep_vol[i]) { + sc_access[0].value = i; + break; /* exit out of the for loop */ + } /* if(value >= mx_ep_vol[i]) */ + } /* for(i=0; i< MAX_EP_VOL_INDX_PMIC_VENDOR1; i++) */ + sc_access[0].reg_addr = AS_LEFT_SPKR_VOL_CTL; + sc_access[0].mask = (MASK0|MASK1|MASK2|MASK3|MASK4); + break; + } /* switch (dev_id) */ + retval = sst_sc_reg_access(sc_access, PMIC_READ_MODIFY, 1); + return retval; +} /* mx_set_vol */ + + +int mx_get_selected_input_dev(int *value) +{ + return 0; +} /* mx_get_selected_input_dev */ + +int mx_get_selected_output_dev(int *value) +{ + struct sc_reg_access sc_access[1] = {{0,},}; + int retval = 0, temp_value; + + sc_access[0].reg_addr = AS_CONFIG; + retval = sst_sc_reg_access(sc_access, PMIC_READ, 1); + if (0 != retval) { + /*pmic communication fails*/ + sst_err("pmic commn failed \n"); + return retval; + } + temp_value = sc_access[0].value; + if (0 == temp_value) + /* Max 8921 is disabled hence Mono earpiece is disabled */ + *value = STEREO_HEADPHONE; + else + *value = MONO_HS; + return 0; +} /* mx_get_selected_output_dev */ + +int mx_get_mute(int dev_id, int *value) +{ + struct sc_reg_access sc_access[1] = {{0,},}; + int retval = 0, temp_value; + + switch (dev_id) { + case PMIC_SND_INPUT_MUTE_LINE_IN: + break; + + case PMIC_SND_INPUT_MUTE_MIC1: + break; + + case PMIC_SND_INPUT_MUTE_MIC2: + break; + + case PMIC_SND_INPUT_MUTE_DMIC: + break; + + case PMIC_SND_LEFT_SPEAKER_MUTE: + break; + + case PMIC_SND_RIGHT_SPEAKER_MUTE: + break; + + case PMIC_SND_LEFT_HP_MUTE: + sc_access[0].reg_addr = VOL_CTRL_LT; + retval = sst_sc_reg_access(sc_access, PMIC_READ, 1); + if (0 != retval) { + /*pmic communication fails*/ + sst_err("pmic commn failed \n"); + return retval; + } + temp_value = sc_access[0].value & (MASK6); + if (0 == temp_value) + *value = UNMUTE; + else + *value = MUTE; + break; + case PMIC_SND_RIGHT_HP_MUTE: + sc_access[0].reg_addr = VOL_CTRL_RT; + retval = sst_sc_reg_access(sc_access, PMIC_READ, 1); + if (0 != retval) { + /*pmic communication fails*/ + sst_err("pmic commn failed \n"); + return retval; + } + temp_value = sc_access[0].value & + (MASK6); + if (0 == temp_value) + *value = UNMUTE; + else + *value = MUTE; + break; + case PMIC_SND_MONO_EARPIECE_MUTE: + sc_access[0].reg_addr = AS_LEFT_SPKR_VOL_CTL; + retval = sst_sc_reg_access(sc_access, PMIC_READ, 1); + if (0 != retval) { + /*pmic communication fails*/ + sst_err("pmic commn failed \n"); + return retval; + } + temp_value = sc_access[0].value & + (MASK0|MASK1|MASK2|MASK3|MASK4); + + if (0 == temp_value) + *value = UNMUTE; + else + *value = MUTE; + break; + } /* switch (dev_id) */ + + sst_dbg("dev id:0x%2x, :0x%2x \n", dev_id, *value); + return retval; +} /* mx_get_mute */ + +int mx_get_vol(int dev_id, int *value) +{ + struct sc_reg_access sc_access[1] = {{0,},}; + int retval = 0, temp_value = 0; + + switch (dev_id) { + case PMIC_SND_INPUT_VOL_MIC1: + case PMIC_SND_INPUT_VOL_MIC2: + break; + case PMIC_SND_INPUT_VOL_DMIC: + break; + case PMIC_SND_LEFT_SPEAKER_VOL: + break; + case PMIC_SND_RIGHT_SPEAKER_VOL: + break; + case PMIC_SND_LEFT_HP_VOL: + sc_access[0].reg_addr = VOL_CTRL_LT; + retval = sst_sc_reg_access(sc_access, PMIC_READ, 1); + if (0 != retval) { + /*pmic communication fails*/ + sst_err("pmic commn failed \n"); + return retval; + } + temp_value = sc_access[0].value + & (MASK0|MASK1|MASK2|MASK3|MASK4|MASK5); + *value = mx_hp_vol[temp_value]; + break; + case PMIC_SND_RIGHT_HP_VOL: + sc_access[0].reg_addr = VOL_CTRL_RT; + retval = sst_sc_reg_access(sc_access, PMIC_READ, 1); + if (0 != retval) { + /*pmic communication fails*/ + sst_err("pmic commn failed \n"); + return retval; + } + temp_value = sc_access[0].value + & (MASK0|MASK1|MASK2|MASK3|MASK4|MASK5); + *value = mx_hp_vol[temp_value]; + break; + case PMIC_SND_MONO_EARPIECE_VOL: + sc_access[0].reg_addr = AS_LEFT_SPKR_VOL_CTL; + retval = sst_sc_reg_access(sc_access, PMIC_READ, 1); + if (0 != retval) { + /*pmic communication fails*/ + sst_err("pmic commn failed \n"); + return retval; + } + temp_value = sc_access[0].value + & (MASK0|MASK1|MASK2|MASK3|MASK4); + *value = mx_ep_vol[temp_value]; + break; + } /* switch (dev_id) */ + + sst_dbg("value read from the register:0x%2x\n", temp_value); + sst_dbg("dev_id:0x%2x , volume:0x%2x in dB\n", dev_id, *value); + + return retval; +} /* mx_get_vol() */ + + + -- 1.5.4.5

1 0

[alsa-devel] [PATCH] [RFC 9/13] Intel SST sound card vendor0 module
by Vinod Koul 03 Jul '09

03 Jul '09

This adds the support for vendor PMICs for widget control and configurations. The MID platform supports three different vendor implementation of PMIC. The sound card, ie all analog components like DAC, ADC, mixer settings are different for three vendors. This patch implements these settings for one of the vendor. Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> Signed-off-by: R Dharageswari <dharageswari.r(a)intel.com> new file: sound/pci/sst/intelmid_v0_control.c --- sound/pci/sst/intelmid_v0_control.c | 529 +++++++++++++++++++++++++++++++++++ 1 files changed, 529 insertions(+), 0 deletions(-) create mode 100644 sound/pci/sst/intelmid_v0_control.c diff --git a/sound/pci/sst/intelmid_v0_control.c b/sound/pci/sst/intelmid_v0_control.c new file mode 100644 index 0000000..cbab23a --- /dev/null +++ b/sound/pci/sst/intelmid_v0_control.c @@ -0,0 +1,529 @@ +/* + * intelmid_v0_control.c - Intel SST Driver for audio engine + * + * Copyright (C) 2008-09 Intel Corporation + * Authors: Vinod Koul <vinod.koul(a)intel.com> + * Harsha Priya <priya.harsha(a)intel.com> + * R Dharageswari <dharageswari.r(a)intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * 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; version 2 of the License. + * + * 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. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This file contains the control operations of vendor 1 + */ + +#include <linux/cdev.h> +#include <linux/pci.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/module.h> +#include <linux/syscalls.h> +#include <linux/file.h> +#include <linux/fcntl.h> +#include <linux/uaccess.h> +#include <linux/interrupt.h> +#include <linux/list.h> +#include <linux/slab.h> +#include <linux/mm.h> +#include <linux/workqueue.h> +#include <linux/pci.h> +#include <linux/firmware.h> +#include <sound/intel_sst.h> +#include <sound/intel_sst_ioctl.h> +#include "intel_sst_fw_ipc.h" +#include "intel_sst_common.h" +#include "intel_sst_pvt.h" +#include "intelmid_snd_control.h" +#include <asm/ipc_defs.h> + +enum _reg_v1 { + VOICEPORT1 = 0x180, + VOICEPORT2 = 0x181, + AUDIOPORT1 = 0x182, + AUDIOPORT2 = 0x183, + MISCVOICECTRL = 0x184, + MISCAUDCTRL = 0x185, + DMICCTRL1 = 0x186, + DMICCTRL2 = 0x187, + MICCTRL = 0x188, + MICLICTRL1 = 0x189, + MICLICTRL2 = 0x18A, + MICLICTRL3 = 0x18B, + VOICEDACCTRL1 = 0x18C, + STEREOADCCTRL = 0x18D, + AUD15 = 0x18E, + AUD16 = 0x18F, + AUD17 = 0x190, + AUD18 = 0x191, + RMIXOUTSEL = 0x192, + ANALOGLBR = 0x193, + ANALOGLBL = 0x194, + AUD22 = 0x195, + POWERCTRL2 = 0x196, + HEADSETDETECTINT = 0x197, + HEADSETDETECTINTMASK = 0x198, + TRIMENABLE = 0x199, +}; + +int rev_id = 0x20; + +int fs_init_card(void) +{ + + struct sc_reg_access sc_access[17] = {{0,},}; + int retval = 0; + + sst_dbg("FS-B0 initializing card...\n"); + + /*send message to SCU to get the PMIC vendor-*/ + sc_access[0].reg_addr = 0; + retval = sst_sc_reg_access(sc_access, PMIC_READ, 1); + if (0 != retval) { + /*pmic communication fails*/ + sst_err("pmic commn failed \n"); + return retval; + } + rev_id = sc_access[0].value; + sst_dbg("value of reg 0:0x%02x \n", rev_id); + + + sc_access[0].reg_addr = 0x184; + sc_access[0].value = 0x1c; + sc_access[1].reg_addr = 0x185; + if (rev_id == 0x20) + sc_access[1].value = 0x4A;/*FS-B0*/ + else + sc_access[1].value = 0x48; + + sc_access[2].reg_addr = 0x187 ; + sc_access[2].value = 0x39; + sc_access[3].reg_addr = 0x196 ; + sc_access[3].value = 0x7F; + sc_access[4].reg_addr = 0x180; + sc_access[4].value = 0xF5; + sc_access[5].reg_addr = 0x181; + sc_access[5].value = 0x26; + sc_access[6].reg_addr = 0x182; + if (rev_id == 0x20) + sc_access[6].value = 0xE8;/*FS-B0*/ + else + sc_access[6].value = 0xD8; + sc_access[7].reg_addr = 0x183; + sc_access[7].value = 0x19; + sc_access[8].reg_addr = 0x186; + sc_access[8].value = 0x79; + sc_access[9].reg_addr = 0x189; + sc_access[9].value = 0xB9; + sc_access[10].reg_addr = 0x18a; + sc_access[10].value = 0xB9; + sc_access[11].reg_addr = 0x18e; + sc_access[11].value = 0x39; + sc_access[12].reg_addr = 0x18f; + sc_access[12].value = 0x39; + sc_access[13].reg_addr = 0x190; + sc_access[13].value = 0x39; + sc_access[14].reg_addr = 0x191; + sc_access[14].value = 0x11; + sc_access[15].reg_addr = 0x192; + sc_access[15].value = 0x0E; + sc_access[16].reg_addr = 0x195; + sc_access[16].value = 0x06; + + retval = sst_sc_reg_access(sc_access, PMIC_WRITE, 17); + if (0 != retval) { + /*pmic communication fails*/ + sst_err("pmic commn failed \n"); + return retval; + } + + sst_dbg("Initialisation complete!!\n"); + return 0; +} + + +int fs_init_capture_card(void) +{ + return 0; +} + +void fs_set_pcm_params(int sfreq, int word_size) +{ + int config1 = 0, retval = 0; + struct sc_reg_access sc_access[2]; + + switch (sfreq) { + case 8000: + config1 = 0x10; + break; + case 11025: + config1 = 0x11; + break; + case 12000: + config1 = 0x12; + break; + case 16000: + config1 = 0x13; + break; + case 22050: + config1 = 0x14; + break; + case 24000: + config1 = 0x15; + break; + case 26000: + config1 = 0x16; + break; + case 32000: + config1 = 0x17; + break; + case 44100: + config1 = 0x18; + break; + case 48000: + config1 = 0x19; + break; + } + + sst_dbg("sfreq:%d, Register value = %x\n", sfreq, config1); + + sc_access[0].reg_addr = AUDIOPORT2; + sc_access[0].value = config1; + + sst_dbg("Setting word size ...\n"); + switch (word_size) { + case 16: + if (rev_id == 0x20) + config1 = 0xE8; /* FS B0 */ + else + config1 = 0xD8; /* FS A0 */ + break; + case 24: + if (rev_id == 0x20) + config1 = 0xEB; /* FS B0 */ + else + config1 = 0xDB; /* FS A0 */ + break; + } /* switch (word_size) */ + + sst_dbg("rev_id:0x%2x, pcm_wd_sz:%d, Register_value= %x\n", + rev_id, word_size, config1); + + sc_access[1].reg_addr = AUDIOPORT1; + sc_access[1].value = config1; + retval = sst_sc_reg_access(sc_access, PMIC_WRITE, 2); + if (0 != retval) + sst_err("pmic commn failed \n"); + return; +} + +int fs_set_selected_input_dev(int value) +{ + + return 0; +} + +int fs_set_selected_output_dev(int value) +{ + struct sc_reg_access sc_access[2]; + int retval = 0; + + sc_access[0].reg_addr = AUD22; + sc_access[0].mask = MASK6|MASK7|MASK4|MASK3|MASK2|MASK1|MASK0; + + switch (value) { + case STEREO_HEADPHONE: + sc_access[0].value = VALUE2|VALUE1; + sc_access[0].mask = MASK2|MASK1; + sc_access[1].reg_addr = 0x192; + sc_access[1].mask = MASK2; + sc_access[1].value = 1; + break; + + case INTERNAL_SPKR: + break; + + case MONO_HS: + sc_access[0].value = VALUE7; + sc_access[0].mask = MASK7; + sc_access[1].reg_addr = 0x192; + sc_access[1].mask = MASK2; + sc_access[1].value = 0; + break; + } + sst_dbg("dev_id:0x%2x reg:0x%2x value:0x%2x\n", + value, sc_access[0].reg_addr, sc_access[0].value); + retval = sst_sc_reg_access(sc_access, PMIC_READ_MODIFY, 2); + if (0 != retval) + sst_err("pmic commn failed \n"); + + return retval; +} + +int fs_set_mute(int dev_id, int value) +{ + struct sc_reg_access sc_access[1] = {{0,},}; + int retval = 0; + + sst_dbg("dev_id:0x%2x value:0x%2x", dev_id, value); + switch (dev_id) { + case PMIC_SND_INPUT_MUTE_LINE_IN: + break; + + case PMIC_SND_INPUT_MUTE_MIC1: + break; + + case PMIC_SND_INPUT_MUTE_MIC2: + break; + + case PMIC_SND_INPUT_MUTE_DMIC: + break; + + case PMIC_SND_LEFT_HP_MUTE: + sc_access[0].reg_addr = AUD18; + sc_access[0].mask = MASK2; + if (value == MUTE) + sc_access[0].value = VALUE2; + else + sc_access[0].value = 0x00; + break; + + case PMIC_SND_LEFT_SPEAKER_MUTE: + break; + + case PMIC_SND_RIGHT_SPEAKER_MUTE: + break; + + case PMIC_SND_RIGHT_HP_MUTE: + sc_access[0].reg_addr = AUD18; + sc_access[0].mask = MASK1; + if (value == MUTE) + sc_access[0].value = VALUE1; + else + sc_access[0].value = 0x00; + break; + + + case PMIC_SND_MONO_EARPIECE_MUTE: + sc_access[0].reg_addr = AUD18; + sc_access[0].mask = MASK0; + if (value == MUTE) + sc_access[0].value = VALUE0; + else + sc_access[0].value = 0x00; + break; + default: + sst_err("rcvd illegal request"); + /*Illegal case*/ + return -EINVAL; + } + + sst_dbg("dev_id:0x%02x arg_value:%d reg:0x%02x value:0x%02x\n", dev_id, + value, sc_access[0].reg_addr, sc_access[0].value); + + retval = sst_sc_reg_access(sc_access, PMIC_READ_MODIFY, 1); + if (0 != retval) + sst_err("pmic commn failed \n"); + + return retval; +} + +int fs_set_vol(int dev_id, int value) +{ + struct sc_reg_access sc_access[2] = {{0,},}; + int retval = 0; + + switch (dev_id) { + case PMIC_SND_INPUT_VOL_MIC1: + break; + + case PMIC_SND_INPUT_VOL_MIC2: + break; + + case PMIC_SND_INPUT_VOL_DMIC: + break; + + case PMIC_SND_INPUT_VOL_RIGHT_LINE_IN: + break; + + case PMIC_SND_INPUT_VOL_LEFT_LINE_IN: + break; + + case PMIC_SND_LEFT_SPEAKER_VOL: + break; + + case PMIC_SND_RIGHT_SPEAKER_VOL: + break; + + case PMIC_SND_MONO_EARPIECE_VOL: + sc_access[0].reg_addr = AUD16; + sc_access[0].mask = MASK5|MASK4|MASK3|MASK2|MASK1|MASK0; + sc_access[0].value = (value - MIN_VOL_PMIC_VENDOR0); + sc_access[1].reg_addr = AUD17; + sc_access[1].mask = MASK5|MASK4|MASK3|MASK2|MASK1|MASK0; + sc_access[1].value = (value - MIN_VOL_PMIC_VENDOR0); + + sst_dbg("dev_id:0x%02x arg_value:%d reg:0x%02x value:0x%02x\n", + dev_id, value, sc_access[0].reg_addr, + sc_access[0].value); + sst_dbg("reg2:0x%02x value2:0x%02x\n", + sc_access[1].reg_addr, sc_access[1].value); + retval = sst_sc_reg_access(sc_access, PMIC_READ_MODIFY, 2); + if (0 != retval) + sst_err("pmic commn failed \n"); + return retval; + + case PMIC_SND_LEFT_HP_VOL: + sc_access[0].reg_addr = AUD16; + sc_access[0].mask = MASK5|MASK4|MASK3|MASK2|MASK1|MASK0; + sc_access[0].value = (value - MIN_VOL_PMIC_VENDOR0); + break; + + case PMIC_SND_RIGHT_HP_VOL: + sc_access[0].reg_addr = AUD17; + sc_access[0].mask = MASK5|MASK4|MASK3|MASK2|MASK1|MASK0; + sc_access[0].value = (value - MIN_VOL_PMIC_VENDOR0); + break; + + default: + sst_err("rcvd illegal request"); + /*Illegal case*/ + return -EINVAL; + } + + sst_dbg("dev_id:0x%2x arg_value:%d reg:0x%2x value:0x%2x\n", dev_id, + value, sc_access[0].reg_addr, sc_access[0].value); + + retval = sst_sc_reg_access(sc_access, PMIC_READ_MODIFY, 1); + if (0 != retval) + sst_err("pmic commn failed \n"); + return retval; +} + +int fs_get_selected_input_dev(int *value) +{ + return 0; +} + +int fs_get_selected_output_dev(int *value) +{ + return 0; +} + +int fs_get_mute(int dev_id, int *value) +{ + struct sc_reg_access sc_access[1] = {{0,},}; + + int retval = 0, temp_value = 0, mask = 0; + + switch (dev_id) { + case PMIC_SND_INPUT_MUTE_LINE_IN: + break; + case PMIC_SND_INPUT_MUTE_MIC1: + break; + case PMIC_SND_INPUT_MUTE_MIC2: + break; + case PMIC_SND_INPUT_MUTE_DMIC: + break; + case PMIC_SND_LEFT_SPEAKER_MUTE: + break; + case PMIC_SND_RIGHT_SPEAKER_MUTE: + break; + case PMIC_SND_LEFT_HP_MUTE: + mask = MASK2; + break; + case PMIC_SND_RIGHT_HP_MUTE: + mask = MASK1; + break; + case PMIC_SND_MONO_EARPIECE_MUTE: + mask = MASK0; + break; + /*line out mute not supported*/ + default: + sst_err("rcvd illegal request"); + /*Illegal case*/ + return -EINVAL; + } + sc_access[0].reg_addr = AUD18; + + retval = sst_sc_reg_access(sc_access, PMIC_READ, 1); + if (0 != retval) { + sst_err("pmic commn failed \n"); + return retval; + } + temp_value = sc_access[0].value & mask; + if (0 == temp_value) + *value = UNMUTE; + else + *value = MUTE; + return retval; +} + +int fs_get_vol(int dev_id, int *value) +{ + struct sc_reg_access sc_access[2] = {{0,},}; + int retval = 0, mask = 0, temp_value = 0; + + switch (dev_id) { + case PMIC_SND_INPUT_VOL_MIC1: + break; + + case PMIC_SND_INPUT_VOL_MIC2: + break; + + case PMIC_SND_INPUT_VOL_DMIC: + break; + + case PMIC_SND_INPUT_VOL_RIGHT_LINE_IN: + break; + + case PMIC_SND_INPUT_VOL_LEFT_LINE_IN: + break; + + case PMIC_SND_LEFT_SPEAKER_VOL: + break; + + case PMIC_SND_RIGHT_SPEAKER_VOL: + break; + + case PMIC_SND_LEFT_HP_VOL: + sc_access[0].reg_addr = AUD16; + mask = (MASK5|MASK4|MASK3|MASK2|MASK1|MASK0); + break; + case PMIC_SND_RIGHT_HP_VOL: + sc_access[0].reg_addr = AUD17; + mask = (MASK5|MASK4|MASK3|MASK2|MASK1|MASK0); + break; + case PMIC_SND_MONO_EARPIECE_VOL: + sc_access[0].reg_addr = AUD17; + mask = (MASK5|MASK4|MASK3|MASK2|MASK1|MASK0); + break; + default: + sst_err("rcvd illegal request"); + /*Illegal case*/ + return -EINVAL; + } + + retval = sst_sc_reg_access(sc_access, PMIC_READ, 1); + if (0 != retval) { + sst_err("pmic commn failed \n"); + return retval; + } + + temp_value = sc_access[0].value & mask; + *value = temp_value + MIN_VOL_PMIC_VENDOR0; + + return retval; +} /* fs_get_vol */ -- 1.5.4.5

1 0

[alsa-devel] RFC: Jack-detection control elements on HD-audio
by Takashi Iwai 03 Jul '09

03 Jul '09

Hi, yet another thing I've worked recently on is a bit more generalized framework for jack-sense reporting for HD-audio. This supersedes the existing jack-sense reporting via the input layer. In addition, it gives the corresponding control elements. For example, you'll have "Jack HP-Out" (or a bit more verbose like "Jack HP-Out at Ext Rear") control via control API as well as other mixer controls. This can notify when a jack is plugged/unplugged. But, another big win is that you can read the current pin sense just by reading this element. I think this is a sort of feature Lennart wanted for PA. Lennart, does it make sense? The test codes are already in topic/hda-jack branch on sound-unstable GIT tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable-2.6.git topic/hda-jack Any comments appreciated. This is also intended for 2.6.32, not for 2.6.31 merge, of course :) thanks, Takashi

5 10

[alsa-devel] [PATCH] ASoC: TWL4030: Add EXTMUTE to reduce pop-noise effect
by Candelaria Villareal, Jorge 02 Jul '09

02 Jul '09

According to TRM, an external FET controlled by a 1.8V output signal can be used to reduce the pop-noise heard when the audio amplifier is switched on. It is suggested that GPIO6 of TWL4030 be used, but any other gpio can be used instead. This is indicated in machine driver with the following twl4030_setup_data members: -hs_extmute. Set to 1 if board has support for EXTMUTE. -set_hs_extmute. Set to a callback funcion to control an external gpio line. Set to NULL if MUTE[GPIO6] pin is used. Codec driver takes care of enabling and disabling this output during the headset pop attenuation sequence. Also add a delay to let VMID settle in ramp up sequence. Signed-off-by: Jorge Eduardo Candelaria <x0107209(a)ti.com> --- sound/soc/codecs/twl4030.c | 27 +++++++++++++++++++++++++++ sound/soc/codecs/twl4030.h | 2 ++ 2 files changed, 29 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 06a95f2..4e9e0e1 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -620,6 +620,9 @@ static int handsfreerpga_event(struct snd_soc_dapm_widget *w, static void headset_ramp(struct snd_soc_codec *codec, int ramp) { + struct snd_soc_device *socdev = codec->socdev; + struct twl4030_setup_data *setup = socdev->codec_data; + unsigned char hs_gain, hs_pop; struct twl4030_priv *twl4030 = codec->private_data; /* Base values for ramp delay calculation: 2^19 - 2^26 */ @@ -629,6 +632,17 @@ static void headset_ramp(struct snd_soc_codec *codec, int ramp) hs_gain = twl4030_read_reg_cache(codec, TWL4030_REG_HS_GAIN_SET); hs_pop = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET); + /* Enable external mute control, this dramatically reduces + * the pop-noise */ + if (setup && setup->hs_extmute) { + if (setup->set_hs_extmute) { + setup->set_hs_extmute(1); + } else { + hs_pop |= TWL4030_EXTMUTE; + twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop); + } + } + if (ramp) { /* Headset ramp-up according to the TRM */ hs_pop |= TWL4030_VMID_EN; @@ -636,6 +650,9 @@ static void headset_ramp(struct snd_soc_codec *codec, int ramp) twl4030_write(codec, TWL4030_REG_HS_GAIN_SET, hs_gain); hs_pop |= TWL4030_RAMP_EN; twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop); + /* Wait ramp delay time + 1, so the VMID can settle */ + mdelay((ramp_base[(hs_pop & TWL4030_RAMP_DELAY) >> 2] / + twl4030->sysclk) + 1); } else { /* Headset ramp-down _not_ according to * the TRM, but in a way that it is working */ @@ -652,6 +669,16 @@ static void headset_ramp(struct snd_soc_codec *codec, int ramp) hs_pop &= ~TWL4030_VMID_EN; twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop); } + + /* Disable external mute */ + if (setup && setup->hs_extmute) { + if (setup->set_hs_extmute) { + setup->set_hs_extmute(0); + } else { + hs_pop &= ~TWL4030_EXTMUTE; + twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop); + } + } } static int headsetlpga_event(struct snd_soc_dapm_widget *w, diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index fe5f395..2b4bfa2 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -274,6 +274,8 @@ extern struct snd_soc_codec_device soc_codec_dev_twl4030; struct twl4030_setup_data { unsigned int ramp_delay_value; unsigned int sysclk; + unsigned int hs_extmute:1; + void (*set_hs_extmute)(int mute); }; #endif /* End of __TWL4030_AUDIO_H__ */ -- 1.6.0.4

5 4

[alsa-devel] [PATCH] ALSA: snd_usb_caiaq: reparent sound device
by Daniel Mack 02 Jul '09

02 Jul '09

The sound device instance needs to be a child of the USB interface, not the USB device. Newer udev versions pay attention to that. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Reported-by: Lennart Poettering <lennart(a)poettering.net> --- sound/usb/caiaq/device.c | 10 ++++++---- 1 files changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index 0e5db71..de38108 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -35,7 +35,7 @@ #include "input.h" MODULE_AUTHOR("Daniel Mack <daniel(a)caiaq.de>"); -MODULE_DESCRIPTION("caiaq USB audio, version 1.3.17"); +MODULE_DESCRIPTION("caiaq USB audio, version 1.3.18"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, RigKontrol3}," @@ -349,7 +349,9 @@ static void __devinit setup_card(struct snd_usb_caiaqdev *dev) log("Unable to set up control system (ret=%d)\n", ret); } -static int create_card(struct usb_device* usb_dev, struct snd_card **cardp) +static int create_card(struct usb_device *usb_dev, + struct usb_interface *intf, + struct snd_card **cardp) { int devnum; int err; @@ -374,7 +376,7 @@ static int create_card(struct usb_device* usb_dev, struct snd_card **cardp) dev->chip.usb_id = USB_ID(le16_to_cpu(usb_dev->descriptor.idVendor), le16_to_cpu(usb_dev->descriptor.idProduct)); spin_lock_init(&dev->spinlock); - snd_card_set_dev(card, &usb_dev->dev); + snd_card_set_dev(card, &intf->dev); *cardp = card; return 0; @@ -461,7 +463,7 @@ static int __devinit snd_probe(struct usb_interface *intf, struct snd_card *card; struct usb_device *device = interface_to_usbdev(intf); - ret = create_card(device, &card); + ret = create_card(device, intf, &card); if (ret < 0) return ret; -- 1.6.3.1

2 3

[alsa-devel] [PATCH] Enable loudspeakers in ALC889 for Acers and others
by ruben＠ugr.es 02 Jul '09

02 Jul '09

Hi all I managed to get sound working on my acer 6935, which has an ALC889 driven by the intel_hda driver The lack of sound seems to appear in other laptops as well. The bug report is at https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4261 I know Takashi Iwai is familiar with the problems, since I used his workaround with hda-verb to get it working initially, so if he is free and wants to get this fixed, I'd be grateful (if anybody else can have a look, that'd be great as well). The patch is very easy (you can download it at the url above). I added snd_hda_codec_write(codec, 0x15, 0, AC_VERB_SET_EAPD_BTLENABLE, 2); at the end of patch_alc883 before the return. Please check whether there is a problem with other models; in that case a new patch_alc889 which calls patch_alc883 plus the line can be added, or more detailed checks on the pci ids can be done. Thanks for your time. Ruben

2 1

[alsa-devel] ASoC updates for 2.6.31
by Mark Brown 02 Jul '09

02 Jul '09

The following changes since commit 10121a12e2380fb34b6e646a8e367add06d036c3: Takashi Iwai (1): Merge branch 'for-2.6.31' of git://git.kernel.org/.../broonie/sound-2.6 into fix/asoc are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31 Grant Likely (2): ASoC: Fix typo in MPC5200 PSC AC97 driver Kconfig ASoC: remove BROKEN from Efika and pcm030 fabric drivers Janusz Krzysztofik (1): ASoC: OMAP: fix OMAP1510 broken PCM pointer callback Mark Brown (1): ASoC: Only disable pxa2xx-i2s clocks if we enabled them sound/soc/fsl/Kconfig | 6 +++--- sound/soc/omap/omap-pcm.c | 11 +++++++---- sound/soc/pxa/pxa2xx-i2s.c | 7 ++++++- 3 files changed, 16 insertions(+), 8 deletions(-)

2 3

[alsa-devel] Problems with snd-hda-intel on hp 6930p (2.6.30)
by Rafael J. Wysocki 02 Jul '09

02 Jul '09

Hi Takashi, I have a problem with snd-hda-intel from 2.6.30 on an hp EliteBook 6930p, which is that the sound is not audible, although everything appears to work just fine. Except maybe that when the box is rebooted and KDE (3.5) is started, the master control is "muted" and the led indicator next to the keyboard shows "no sound". Then, changing the "muted" status of the master control doesn't change the state of the led indicator and vice versa (the two things appear to be totally independent of each other). Can you please tell me what I can do to debug this issue? Best, Rafael

2 8

[alsa-devel] laptop multichannel ethernet OR PCMCIA sound cards
by Matt Flax 02 Jul '09

02 Jul '09

We are after an ultra low latency connection to a multichannel in/out sound card for a laptop. We want a portable system and don't want to network to a different computer to achieve this goal. As we are after ultra low latency, we are interested in both PCMCIA and ethernet (gigabit) solutions. Is anyone aware of any such ultra low latency hardware ? Can anyone suggest any other solution for a laptop which is portable and not based on ethernet or PCMCIA ? I think that USB and firewire are good, but want even lower latency to the hardware if possible. thanks -- Matt Flax, BE, BSc, PhD Research Engineer National Acoustic Laboratories A Division of Australian Hearing 126 Greville Street Chatswood NSW 2067 Australia P +61 2 9412 6985 F +61 2 9411 8273 Please consider the environment before printing this email

6 5

[alsa-devel] [PATCH] Line In for Acer Inspire 6530G model
by Tony Vroon 01 Jul '09

01 Jul '09

The Line In connector is set up as PIN_IN by default, using VREF_HIZ. It is connected to both ADCs, so add it to both input selectors. Also add the ability to use the input mix (on a SoundBlaster one would call this "What You Hear"). Signed-off-by: Tony Vroon <tony(a)linx.net> diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3345331..67dd883 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1560,18 +1560,22 @@ static struct hda_input_mux alc888_2_capture_sources[2] = { static struct hda_input_mux alc888_acer_aspire_6530_sources[2] = { /* Interal mic only available on one ADC */ { - .num_items = 3, + .num_items = 5, .items = { { "Ext Mic", 0x0 }, + { "Line In", 0x2 }, { "CD", 0x4 }, + { "Input Mix", 0xa }, { "Int Mic", 0xb }, }, }, { - .num_items = 2, + .num_items = 4, .items = { { "Ext Mic", 0x0 }, + { "Line In", 0x2 }, { "CD", 0x4 }, + { "Input Mix", 0xa }, }, } }; @@ -8189,6 +8193,8 @@ static struct snd_kcontrol_new alc888_acer_aspire_6530_mixer[] = { HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("LFE Playback Volume", 0x0f, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("LFE Playback Switch", 0x0f, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),

6 8

[alsa-devel] [PATCH] alsa/soc: add locking to mpc5200-psc-ac97 driver
by Grant Likely 01 Jul '09

01 Jul '09

From: Grant Likely <grant.likely(a)secretlab.ca> AC97 bus register read/write hooks need to provide locking, but the mpc5200-psc-ac97 driver does not. This patch adds a mutex around the register access routines. Signed-off-by: Grant Likely <grant.likely(a)secretlab.ca> --- sound/soc/fsl/mpc5200_dma.c | 1 + sound/soc/fsl/mpc5200_dma.h | 1 + sound/soc/fsl/mpc5200_psc_ac97.c | 17 ++++++++++++++++- 3 files changed, 18 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index efec33a..f0a2d40 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -456,6 +456,7 @@ int mpc5200_audio_dma_create(struct of_device *op) return -ENODEV; spin_lock_init(&psc_dma->lock); + mutex_init(&psc_dma->mutex); psc_dma->id = be32_to_cpu(*prop); psc_dma->irq = irq; psc_dma->psc_regs = regs; diff --git a/sound/soc/fsl/mpc5200_dma.h b/sound/soc/fsl/mpc5200_dma.h index 2000803..8d396bb 100644 --- a/sound/soc/fsl/mpc5200_dma.h +++ b/sound/soc/fsl/mpc5200_dma.h @@ -55,6 +55,7 @@ struct psc_dma { unsigned int irq; struct device *dev; spinlock_t lock; + struct mutex mutex; u32 sicr; uint sysclk; int imr; diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c index 794a247..7eb5499 100644 --- a/sound/soc/fsl/mpc5200_psc_ac97.c +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -34,13 +34,20 @@ static unsigned short psc_ac97_read(struct snd_ac97 *ac97, unsigned short reg) int status; unsigned int val; + mutex_lock(&psc_dma->mutex); + /* Wait for command send status zero = ready */ status = spin_event_timeout(!(in_be16(&psc_dma->psc_regs->sr_csr.status) & MPC52xx_PSC_SR_CMDSEND), 100, 0); if (status == 0) { pr_err("timeout on ac97 bus (rdy)\n"); + mutex_unlock(&psc_dma->mutex); return -ENODEV; } + + /* Force clear the data valid bit */ + in_be32(&psc_dma->psc_regs->ac97_data); + /* Send the read */ out_be32(&psc_dma->psc_regs->ac97_cmd, (1<<31) | ((reg & 0x7f) << 24)); @@ -50,16 +57,19 @@ static unsigned short psc_ac97_read(struct snd_ac97 *ac97, unsigned short reg) if (status == 0) { pr_err("timeout on ac97 read (val) %x\n", in_be16(&psc_dma->psc_regs->sr_csr.status)); + mutex_unlock(&psc_dma->mutex); return -ENODEV; } /* Get the data */ val = in_be32(&psc_dma->psc_regs->ac97_data); if (((val >> 24) & 0x7f) != reg) { pr_err("reg echo error on ac97 read\n"); + mutex_unlock(&psc_dma->mutex); return -ENODEV; } val = (val >> 8) & 0xffff; + mutex_unlock(&psc_dma->mutex); return (unsigned short) val; } @@ -68,16 +78,21 @@ static void psc_ac97_write(struct snd_ac97 *ac97, { int status; + mutex_lock(&psc_dma->mutex); + /* Wait for command status zero = ready */ status = spin_event_timeout(!(in_be16(&psc_dma->psc_regs->sr_csr.status) & MPC52xx_PSC_SR_CMDSEND), 100, 0); if (status == 0) { pr_err("timeout on ac97 bus (write)\n"); - return; + goto out; } /* Write data */ out_be32(&psc_dma->psc_regs->ac97_cmd, ((reg & 0x7f) << 24) | (val << 8)); + + out: + mutex_unlock(&psc_dma->mutex); } static void psc_ac97_warm_reset(struct snd_ac97 *ac97)

4 13

[alsa-devel] [PATCH 1/4] ASoC: pxa-ssp: enhance I2S and add Left_J support
by Eric Miao 01 Jul '09

01 Jul '09

Make the pxa I2S configuration generic, add support for Left_J, add support for variable frame width like 32fs, 48fs, 64fs and 96fs Signed-off-by: Paul Shen <bshen9(a)marvell.com> Signed-off-by: Eric Miao <eric.miao(a)marvell.com> Cc: Daniel Mack <daniel(a)caiaq.de> --- arch/arm/mach-pxa/include/mach/regs-ssp.h | 14 +++--- sound/soc/pxa/pxa-ssp.c | 62 ++++++++++++++-------------- sound/soc/pxa/pxa-ssp.h | 9 ++++ 3 files changed, 47 insertions(+), 38 deletions(-) diff --git a/arch/arm/mach-pxa/include/mach/regs-ssp.h b/arch/arm/mach-pxa/include/mach/regs-ssp.h index 6a2ed35..27f0cd4 100644 --- a/arch/arm/mach-pxa/include/mach/regs-ssp.h +++ b/arch/arm/mach-pxa/include/mach/regs-ssp.h @@ -108,21 +108,21 @@ #define SSSR_TINT (1 << 19) /* Receiver Time-out Interrupt */ #define SSSR_PINT (1 << 18) /* Peripheral Trailing Byte Interrupt */ -#if defined(CONFIG_PXA3xx) -#define SSPSP_EDMYSTOP(x) ((x) << 28) /* Extended Dummy Stop */ -#define SSPSP_EDMYSTRT(x) ((x) << 26) /* Extended Dummy Start */ -#endif - #define SSPSP_FSRT (1 << 25) /* Frame Sync Relative Timing */ -#define SSPSP_DMYSTOP(x) ((x) << 23) /* Dummy Stop */ #define SSPSP_SFRMWDTH(x) ((x) << 16) /* Serial Frame Width */ #define SSPSP_SFRMDLY(x) ((x) << 9) /* Serial Frame Delay */ -#define SSPSP_DMYSTRT(x) ((x) << 7) /* Dummy Start */ #define SSPSP_STRTDLY(x) ((x) << 4) /* Start Delay */ #define SSPSP_ETDS (1 << 3) /* End of Transfer data State */ #define SSPSP_SFRMP (1 << 2) /* Serial Frame Polarity */ #define SSPSP_SCMODE(x) ((x) << 0) /* Serial Bit Rate Clock Mode */ +/* NOTE: PXA3xx extends the bit number of dummy start and stop, the macros + * below are compatible with PXA25x/27x as long as the parameter is within + * the correct limits, driver code has to take care of this. + */ +#define SSPSP_DMYSTRT(x) ((((x) & 3) << 7) | ((((x) >> 2) & 3) << 26)) +#define SSPSP_DMYSTOP(x) ((((x) & 3) << 23) | ((((x) >> 2) & 7) << 28)) + #define SSACD_SCDB (1 << 3) /* SSPSYSCLK Divider Bypass */ #define SSACD_ACPS(x) ((x) << 4) /* Audio clock PLL select */ #define SSACD_ACDS(x) ((x) << 0) /* Audio clock divider select */ diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 6fc7876..2831c16 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -463,7 +463,8 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - sscr0 |= SSCR0_PSP; + case SND_SOC_DAIFMT_LEFT_J: + sscr0 |= SSCR0_PSP | SSCR0_MOD; sscr1 |= SSCR1_RWOT | SSCR1_TRAIL; /* See hw_params() */ @@ -541,6 +542,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, int chn = params_channels(params); u32 sscr0; u32 sspsp; + int frame_width; int width = snd_pcm_format_physical_width(params_format(params)); int ttsa = ssp_read_reg(ssp, SSTSA) & 0xf; @@ -585,40 +587,38 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - sspsp = ssp_read_reg(ssp, SSPSP); - - if ((ssp_get_scr(ssp) == 4) && (width == 16)) { - /* This is a special case where the bitclk is 64fs - * and we're not dealing with 2*32 bits of audio - * samples. - * - * The SSP values used for that are all found out by - * trying and failing a lot; some of the registers - * needed for that mode are only available on PXA3xx. - */ + sspsp = ssp_read_reg(ssp, SSPSP); + frame_width = PXA_SSP_FRM_WIDTH(priv->dai_fmt); -#ifdef CONFIG_PXA3xx - if (!cpu_is_pxa3xx()) - return -EINVAL; - - sspsp |= SSPSP_SFRMWDTH(width * 2); - sspsp |= SSPSP_SFRMDLY(width * 4); - sspsp |= SSPSP_EDMYSTOP(3); - sspsp |= SSPSP_DMYSTOP(3); - sspsp |= SSPSP_DMYSTRT(1); -#else + if (frame_width < width * 2) return -EINVAL; -#endif - } else { - /* The frame width is the width the LRCLK is - * asserted for; the delay is expressed in - * half cycle units. We need the extra cycle - * because the data starts clocking out one BCLK - * after LRCLK changes polarity. + + if (frame_width == width * 2) + /* frame width is exactly double of data sample width, + * use FSRT instead */ - sspsp |= SSPSP_SFRMWDTH(width + 1); - sspsp |= SSPSP_SFRMDLY((width + 1) * 2); + sspsp |= SSPSP_FSRT | SSPSP_SFRMWDTH(width); + else { sspsp |= SSPSP_DMYSTRT(1); + sspsp |= SSPSP_DMYSTOP((frame_width / 2 - width - 1)); + sspsp |= SSPSP_SFRMWDTH(frame_width / 2); + } + + ssp_write_reg(ssp, SSPSP, sspsp); + break; + + case SND_SOC_DAIFMT_LEFT_J: + sspsp = ssp_read_reg(ssp, SSPSP); + frame_width = PXA_SSP_FRM_WIDTH(priv->dai_fmt); + + if (frame_width < width * 2) + return -EINVAL; + + if (frame_width == width * 2) + sspsp |= SSPSP_SFRMWDTH(width); + else { + sspsp |= SSPSP_DMYSTOP((frame_width / 2 - width)); + sspsp |= SSPSP_SFRMWDTH(frame_width / 2); } ssp_write_reg(ssp, SSPSP, sspsp); diff --git a/sound/soc/pxa/pxa-ssp.h b/sound/soc/pxa/pxa-ssp.h index 91deadd..fda51d0 100644 --- a/sound/soc/pxa/pxa-ssp.h +++ b/sound/soc/pxa/pxa-ssp.h @@ -40,6 +40,15 @@ #define PXA_SSP_CLK_SCDB_1 1 #define PXA_SSP_CLK_SCDB_8 2 +/* frame size definitions for I2S and Left_J formats - default is + * 32fs, other possibilities are 48fs, 64fs and 96fs + */ +#define PXA_SSP_FRM_32FS (0 << 16) +#define PXA_SSP_FRM_48FS (1 << 16) +#define PXA_SSP_FRM_64FS (2 << 16) +#define PXA_SSP_FRM_96FS (3 << 16) +#define PXA_SSP_FRM_WIDTH(x) (((((x) >> 16) & 0x3) + 2) << 4) + #define PXA_SSP_PLL_OUT 0 extern struct snd_soc_dai pxa_ssp_dai[4]; -- 1.6.0.4

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Re: [alsa-devel] ALSA patch Nvidia 8 channel hdmi audio
by Pau Rodriguez-Estivill 01 Jul '09

01 Jul '09

I attach a patch to support 24bits and 32kHz, 44.1kHz, 48kHz, 96kHz, 192kHz, all rates that appears in the screenshot submitted by Alan Alan. I created also an asound.conf to correct the speakers positions. Since current driver only support 2, 4, 6 and 8 channels I specified it in the asound.conf but I'm not sure if this slave.channels should be fixed. The main problem is that by using asound.conf I didn't found any way to route to surround51 (or surround50 or surround40, ...) depending of the number of channels offered by the sink player. And when I play directly to hdmi, the AMP reports automatically the correct number of speakers involved. Of course it would be nice to have this routes hardcoded in the driver...

1 0

Re: [alsa-devel] PC speaker beeps in 2.6.30?
by Michael Tokarev 30 Jun '09

30 Jun '09

[Adding some more Cc's...] Am I the only concerned about this? My 2.6.30 is still silent w.r.t. old good PC speaker beeps, and I wasn't able to make it to produce any sound. Yes, as pointed out by others there is a control now, in alsa, and I can hear ugly and scary beeps from my stereo speakers (when they're turned on and when the control is un-muted). But that's.. not a solution/answer to the original question... ;) Thanks! Michael Tokarev wrote at Mon, 15 Jun 2009 18:41:42 +0400: > Anyone know what happened with PC speaker in 2.6.30? > Basically, it does not work anymore here, with the > config which is very similar to the one used for > previous 2.6.29 kernel (where PC speaker worked). > At least I don't see any relevant differences and > the (seemengly) relevant symbols -- which are > INPUT_PCSPKR and PCSPKR_PLATFORM -- are both set > to 'y' for both 2.6.29 and 2.6.30. > > Help? :) > > Thanks. > > /mjt -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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Re: [alsa-devel] Laptop speakers nonfunctional, jacks functional
by Takashi Iwai 30 Jun '09

30 Jun '09

At Mon, 29 Jun 2009 20:06:47 +0200, Ludwig Maes wrote: > > Tried the snapshot, > still the same, alsa-info.txt attached Hm, just through a quick look at the output, there is no obvious problem. It's only the problem from speaker, not about the headphone, right? If so, the problem is likely the missing amplifier setup, which is usually powered up via COEF verbs or GPIO verbs (ALC888/1200 have no EAPD verbs). Try to install hda-verb program and check the command below. # hda-verb /dev/snd/hwC0D0 0x20 SET_COEF_INDEX 7 # hda-verb /dev/snd/hwC0D0 0x20 GET_PROC_COEF 0 Which value do you get? What happens if you set # hda-verb /dev/snd/hwC0D0 0x20 SET_COEF_INDEX 7 # hda-verb /dev/snd/hwC0D0 0x20 SET_PROC_COEF 0x3060 or # hda-verb /dev/snd/hwC0D0 0x20 SET_COEF_INDEX 7 # hda-verb /dev/snd/hwC0D0 0x20 SET_PROC_COEF 0x3070 or # hda-verb /dev/snd/hwC0D0 0x20 SET_COEF_INDEX 7 # hda-verb /dev/snd/hwC0D0 0x20 SET_PROC_COEF 0x0830 or so? Takashi

1 0

[alsa-devel] Understanding _snd_pcm_channel_area Struct (PCM Interface)!
by Guilherme 30 Jun '09

30 Jun '09

Hi all. I am a bit confused on what I get with this function. I finished my first project but I did not understand what exactly the _snd_pcm_channel_area at http://www.alsa-project.org/alsa-doc/alsa-lib/struct__snd__pcm__channel__ar… does. There are 3 fields in the documentation. addr, first and step. It seems that addr is the memory address of the channel samples... so far so good. The "step" is the distance between 2 sample (but I am not sure) and the least, "first", I could not figure out what does it means. Could I get some help from here? Thanks in advanced! Tks! ------------------- Guilherme Longo Dept. Eng. da Computação Unaerp Linux User - #484927 *Before Asking http://www.istf.com.br/?page=perguntas !- I'd rather die on my feet than live on my knees -!

2 7

[alsa-devel] Merged patch_alc882() and patch_alc883()
by Takashi Iwai 30 Jun '09

30 Jun '09

Hi, I hacked around patch_realtek.c to clean it up a bit. As a result, now patch_alc883() gets merged to patch_alc882(). This cuts off lots of lines. I think this action is mostly OK as long as I tested with hda-emu. But, if you find any problems with this, please let me know. At best, get alsa-info.sh output before updating. It's always recommended to keep alsa-info.sh output once on a working system. This will make easier to track down the problem. thanks, Takashi

1 0

[alsa-devel] External PCM hook SO is dlclosed after calling install, making it hard to have hooks in it
by Paul Fertser 30 Jun '09

30 Jun '09

Hi, I've tried to implement an external shared object to be able to execute arbitrary functions on PCM device opening and closing. After quite some source code reading i had an almost working but segfaulting example. This segfault is because snd_dlclose is called right after calling a hook install function and therefore i can't snd_pcm_hook_add functions from the same shared object. To me it looks like a bug but probably i just don't see how this functionality is intended to be used. Takashi, git-blame showed that it's you who stuffed this dlclose call that bothers me ;) TIA Background information: We're working on audio subsystem for OpenMoko Freerunner. It's an idea of Joerg Reisenweber to have virtual alsa devices for every reasonable use-case (like dedicated device for stereout, dedicated device for recording from gsm etc), for that we need a hook on every device opening and closing mostly to set up routing inside WM8753 codec. ctl_elems is not enough because we need a more complex locking scheme (with priorities, some kind of callbacks to the apps using particular devices etc) and other flexibility. -- Be free, use free (http://www.gnu.org/philosophy/free-sw.html) software! mailto:fercerpav@gmail.com

2 7

[alsa-devel] [snd-hdsp] module mapping missing
by u2＠whoplays.asia 30 Jun '09

30 Jun '09

Module-hdsp maintainers Hi, Thanks for your great sound system, but I found a missing or invalid map of the soundcard and module for "RME HDSPe AIO". [RME HDSPe AIO] http://www.rme-audio.de/en_products_hdspe_aio.php However the soundcard shoud be recognized as the product "HDSPe ExpressCard" as in soundcard matrix I think, but was recognized as the product "Hammerfall DSP Madi". [Matrix:Vendor-RME] http://www.alsa-project.org/main/index.php/Matrix:Vendor-RME [RME HDSP MADI] http://www.rme-audio.de/en_products_hdsp_madi.php Because product-id looks categorized as "HDSP MADI", the module snd-hdspm for "HDSP MADI" was installed for "HDSPe AIO". $ uname -srvmo Linux 2.6.26-2-amd64 #1 SMP Sun Jun 21 04:47:08 UTC 2009 x86_64 GNU/Linux $ lspci -nn | grep RME 02:00.0 Multimedia audio controller [0401]: Xilinx Corporation RME Hammerfall DSP MADI [10ee:3fc6] (rev d4) $ cat /proc/asound/modules 0 snd_hdspm $ pcimodules | grep hdsp snd-hdspm $ cat /lib/modules/2.6.26-2-amd64/modules.pcimap | grep hdsp snd-hdsp 0x000010ee 0x00003fc5 0xffffffff 0xffffffff 0x00000000 0x00000000 0x0 snd-hdspm 0x000010ee 0x00003fc6 0xffffffff 0xffffffff 0x00000000 0x00000000 0x0 $ setpci -d 10ee:3fc6 vendor_id device_id subsystem_vendor_id subsystem_id 10ee 3fc6 00000000 0000 How can I help for it ? ---- Toshiya Miyagi

2 2

[PATCH] pcm_hooks: cosmetic removal of unused variables
by Paul Fertser 30 Jun '09

30 Jun '09

Signed-off-by: Paul Fertser <fercerpav(a)gmail.com> --- src/pcm/pcm_hooks.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/src/pcm/pcm_hooks.c b/src/pcm/pcm_hooks.c index 2fbee73..3a99d55 100644 --- a/src/pcm/pcm_hooks.c +++ b/src/pcm/pcm_hooks.c @@ -99,7 +99,6 @@ static int snd_pcm_hooks_close(snd_pcm_t *pcm) } } while (!list_empty(&h->dllist)) { - struct snd_pcm_hook_dllist *dl; pos = h->dllist.next; hook_remove_dlobj(list_entry(pos, struct snd_pcm_hook_dllist, list)); } @@ -345,7 +344,6 @@ static int snd_pcm_hook_add_conf(snd_pcm_t *pcm, snd_config_t *root, snd_config_ snd_config_iterator_t i, next; int (*install_func)(snd_pcm_t *pcm, snd_config_t *args) = NULL; void *h = NULL; - struct snd_pcm_hook_dllist *dl; if (snd_config_get_type(conf) != SND_CONFIG_TYPE_COMPOUND) { SNDERR("Invalid hook definition"); -- 1.6.1.3 --IMjqdzrDRly81ofr--

1 0

[alsa-devel] [PATCH] [RFC] ASoC: OMAP: fix OMAP1510 broken PCM pointer callback
by Janusz Krzysztofik 30 Jun '09

30 Jun '09

This patch tries to work around the problem of broken OMAP1510 PCM playback pointer calculation by replacing DMA function call that incorrectly tries to read the value form DMA hardware with a value computed locally from an already maintained variable omap_runtime_data.period_index. Tested on OMAP5910 based Amstrad Delta (E3) using work in progress ASoC driver. Based on linux-2.6-asoc.git v2.6.31-rc1. Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- It seems that on OMAP1510, DMA Channel Progress Counter registers (DMA_CPC_CH[0-8]) always contain values derived from DMA channels destination port address, even if constant, and there are no DMA registers available that would follow DMA channels source port address. Because of this limitation, current implementation of omap_get_dma_src_pos() for OMAP1510 is broken and doesn't seem to be easy correctable. --- linux-2.6.31-rc1/sound/soc/omap/omap-pcm.c.orig 2009-06-27 20:20:16.000000000 +0200 +++ linux-2.6.31-rc1/sound/soc/omap/omap-pcm.c 2009-06-27 23:21:42.000000000 +0200 @@ -216,12 +216,15 @@ static snd_pcm_uframes_t omap_pcm_pointe dma_addr_t ptr; snd_pcm_uframes_t offset; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - ptr = omap_get_dma_src_pos(prtd->dma_ch); - else + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { ptr = omap_get_dma_dst_pos(prtd->dma_ch); + offset = bytes_to_frames(runtime, ptr - runtime->dma_addr); + } else if (!(cpu_is_omap1510())) { + ptr = omap_get_dma_src_pos(prtd->dma_ch); + offset = bytes_to_frames(runtime, ptr - runtime->dma_addr); + } else + offset = prtd->period_index * runtime->period_size; - offset = bytes_to_frames(runtime, ptr - runtime->dma_addr); if (offset >= runtime->buffer_size) offset = 0;

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[alsa-devel] Can a phone hook switch follow alsa jack model?
by Janusz Krzysztofik 29 Jun '09

29 Jun '09

Hi, I am wondering if it is a good idea to create support for a phone hook switch, or a handset pick up switch, like that found on Amstrad E3 (Delta) videophone, using alsa jack framework. After my initial attempt to add support for the switch using gpio-keys driver, I am no longer sure if it is a good idea to follow the keyboard model, that the driver has been designed after, for driving a switch that has nothing to do with keyboards and may required a different approach. OTOH, the switch is closely related to a handset, and handsets can be seen as sound devices, can't they? So maybe alsa jack would fit better than keyboard? However, I am not sure if the switch in question matches the alsa jack model closely enough. I see the switch usage not as simple as turning handset microphone/speaker on or off. It can be used for other purposes as well, like accepting a phone call or actually dialing a number that has been just typed in. Furthermore, it can be used to turn off a speakerphone function, while, in turn, the related handset microphone/speaker pair can be turned off not only with this switch, but with a handsfree button as well, for example. All that extra functionality looks like belonging to userspace rather then kernel, not like other alsa jack implementations that seem to do all the job of switching media paths inside the kernel. That is why I am not sure if the jack model is suitable for the purpose. Thanks, Janusz -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

2 6

[alsa-devel] IMX31 Audio Driver issue
by Harish Kumar 29 Jun '09

29 Jun '09

Hi, I am using a imx31 based board from Garz & fricke. I need to configure the audio driver from the linux 2.6.23 BSP available. I configured the audio driver by selecting the SSI, ALSA & PMIC Audio options in the kernel configuration. The PMIC Audio card is listing /proc/asound/cards and all the necessary devices are created using the udev available on the rootfs. I used the check_audio.out application available from IMX PDk BSP test programs. We are getting the following error. I am using the /unit_tests/check_audio.out 0 /unit_tests/audio8k16S.wav Hi... playing on /dev/sound/adsp (SSI1) Configuration : 16 8000 2 audio driver write error: Input/output error audio driver write error: Input/output error a The program is not exiting. also tried with SSI2 as below /unit_tests/check_audio.out 1 /unit_tests/audio8k16S.wav Hi... playing on /dev/sound/dsp (SSI2) Configuration : 16 8000 2 audio driver write error: Input/output error audio driver write error: Input/output error But the same hardware is working with WINCE without any issue. I feel there might be issues with drivers that i am using or may some configuration mismatch. Request your help in resolving the same. Request you to provide the troubleshooting steps so that issue can be resolved. The applications used and the capture of the command execution is as attached. Thanks & Regards Harish Kumar.B

4 5

[alsa-devel] warning "hda-intel: IRQ timing workaround is activated for card #0. Suggest a bigger bdl_pos_adj."
by Simos Xenitellis 29 Jun '09

29 Jun '09

Hi All, I am getting the above warning when trying to play audio files. Contrary to other similar cases with this warning, in my case the playback of the media files fails. For every single media file, the playback fails after a random amount of time (a few seconds to a few minutes). I am running Ubuntu 9.04 (kernel 2.6.28-13-generic, x86_64, upgraded to Alsa 1.0.20 using AlsaUpgrade, http://ubuntuforums.org/showthread.php?p=6589810). I increased bdl_pos_adj to 32 and there is no effect (I added "options snd-hda-intel bdl_pos_adj=32" to /etc/modprobe.d/alsa-base.conf, then "/sbin/alsa force-reload"). lspci -vvv says: 00:1b.0 Audio device: Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 03) Subsystem: Dell Device 02bb Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 2293 Region 0: Memory at f6500000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable+ Address: 00000000fee0300c Data: 41d1 Capabilities: [70] Express (v1) Root Complex Integrated Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s <64ns, L1 <1us ExtTag- RBE- FLReset+ DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr+ TransPend- LnkCap: Port #0, Speed unknown, Width x0, ASPM unknown, Latency L0 <64ns, L1 <1us ClockPM- Suprise- LLActRep- BwNot- LnkCtl: ASPM Disabled; Disabled- Retrain- CommClk- ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed unknown, Width x0, TrErr- Train- SlotClk- DLActive- BWMgmt- ABWMgmt- Capabilities: [100] Virtual Channel <?> Capabilities: [130] Root Complex Link <?> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel lshw says: *-multimedia description: Audio device product: 82801I (ICH9 Family) HD Audio Controller vendor: Intel Corporation physical id: 1b bus info: pci@0000:00:1b.0 version: 03 width: 64 bits clock: 33MHz capabilities: pm msi pciexpress bus_master cap_list configuration: driver=HDA Intel latency=0 module=snd_hda_intel Simos

2 3

[alsa-devel] Bug #4560: USB webcam Philips SPC 530NC micro not working
by Nicolas George 29 Jun '09

29 Jun '09

Hi. Some time ago, I submitted a bug about the integrated micro of the USB webcam Philips SPC 530NC: record: pcm_read:1617: read error: Input/output error This is bug #4560: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4560 I would really appreciate if someone familiar with the USB audio system could give it some attention. It is probably very similar to bug #3927. Raymond seems to have been able to help somewhat, but the command that is said to work for the reporter does not work for me. Regards, -- Nicolas George

1 0

[alsa-devel] [PATCH 1/2] alsa/soc: Fix typo in MPC5200 PSC AC97 driver Kconfig
by Grant Likely 29 Jun '09

29 Jun '09

From: Grant Likely <grant.likely(a)secretlab.ca> ALSA SoC drivers should be specify SND_SOC_AC97_BUS instead, not AC97_BUS. Without SND_SOC_AC97_BUS defined, an AC97 device will not get correctly registered on the AC97 bus, which prevents thinks like the WM9712 touchscreen driver from getting probed. Tested against 2.6.31-rc1. Signed-off-by: Grant Likely <grant.likely(a)secretlab.ca> --- sound/soc/fsl/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 5dbebf8..5661876 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -33,7 +33,7 @@ config SND_SOC_MPC5200_I2S config SND_SOC_MPC5200_AC97 tristate "Freescale MPC5200 PSC in AC97 mode driver" depends on PPC_MPC52xx && PPC_BESTCOMM - select AC97_BUS + select SND_SOC_AC97_BUS select SND_MPC52xx_DMA select PPC_BESTCOMM_GEN_BD help

3 4

Re: [alsa-devel] [PATCH 62/62] sound: Remove unnecessary semicolons
by Mark Brown 29 Jun '09

29 Jun '09

On Sun, Jun 28, 2009 at 09:27:07AM -0700, Joe Perches wrote: > Signed-off-by: Joe Perches <joe(a)perches.com> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com>

1 0

[alsa-devel] No sound with nVidia Corporation MCP79 High Definition Audio (rev b1) (Macbook Pro 5, 5)
by Andreas Nuesslein 29 Jun '09

29 Jun '09

I just got a macbookpro (mid2009 version) with a new nvidia chipset. there is no sound at all. neither via the built-in speakers nor via out-jack. i tried aplay -D myfile.wav with everything from aplay -L as well. i also tried a recent snapshot (alsa-driver-20090626.tar.bz2) as well as gentoo ebuilds alsa-headers-9999 and alsa-driver-9999 (which uses git to pull recent versions). so here's a little information i can gather: $ lspci -s 00:08.0 -v 00:08.0 Audio device: nVidia Corporation MCP79 High Definition Audio (rev b1) Subsystem: nVidia Corporation Device cb79 Flags: bus master, 66MHz, fast devsel, latency 0, IRQ 23 Memory at d3480000 (32-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel alsa-info: http://www.alsa-project.org/db/?f=80e89b82fec184b68a88b9ed4128544a276775fc modinfo snd-hda-intel: http://www.nopaste.com/p/a3B7miINP i also found a thread about that problem (w/o solution obviously): http://www.uluga.ubuntuforums.org/showthread.php?t=1188849&page=2 hope that's all information necessary. if i can i'd love to help in some way. thanks

2 1

[alsa-devel] Laptop speakers nonfunctional, jacks functional
by Ludwig Maes 29 Jun '09

29 Jun '09

---------- Forwarded message ---------- From: Ludwig Maes <ludwig.maes(a)gmail.com> Date: 2009/6/26 Subject: Laptop speakers nonfunctional, jacks functional To: alsa-user(a)lists.sourceforge.net Realtek ALC888S (according to preinstalled Realtek HD audio manager on windows) visible jacks working, builtin laptop speakers and subwoofer silent. Running Ubuntu 9.04 compiled and installed alsa 1.0.20. http://www.alsa-project.org/db/?f=fb64ba9a8ab28d8c16532ece21e3adba7219e4b9 Notice it thinks it is ALC1200. I searched this and found it in patch_realtek.c, I changed the equality test so we would end up in ALC888 section... result was I got an extra volume control on mixer for BEEP. previously did not hear beeps when hitting down in terminal. BEEP is in headphones or line out. still not on speakers. aplay -l listed ALC888 too but codec#0 still showed ALC1200... I dont know the flow of information changes in ALSA to change this codec... I undid the equality test change and now its back to original realtek patch, recompiled reinstalled... added model=generic, to asound.state, this gave me more controls IIRC.. downloaded codecgraph utility, result: http://img20.imageshack.us/img20/8692/codec0.png I dont know how these codecgraphs work, but 4 jacks are right! the internal speakers appear twice?? how can one rewire these, or enable assuming dotted is disabled, does every amplifier triangle correspond to a volume control? is there a direct way to make changes? dmesg |grep hda: [ 16.172695] hda_codec: Unknown model for ALC883, trying auto-probe from BIOS... [ 18.037866] ALSA /usr/src/alsa/alsa-driver-1.0.20/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:636: hda_intel: azx_get_response timeout, switching to polling mode: last cmd=0x00bb0000 Realtek datasheet for ALC888S, reference design, if helpful: http://www.realtek.com.tw/products/productsView.aspx?Langid=1&PFid=28&Level… I am very willing to test proposals... friendly greetings, Ludwig

2 1

[alsa-devel] Samsung P50 Heaphone Jack-Sense not working with latest ALSA
by Martin Egge 29 Jun '09

29 Jun '09

Hi, last year Takashi did help me to get the digital output and the internal/external mic up and running on my Samsung P50 notebook with an integrated Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller. Some time ago I switched from Ubuntu 7.10 to 8.10 and a new problem with the soundcard occurred. The Headphone/Line-Out Jack-Sense is not working. When I plug in my Headphones the internal speakers of the notebook aren't switched off. I've ignored that problem for a long time but now I'm willing to change this. I've already upgraded my system to ALSA 1.0.20 with a script provided in the Ubuntu Forums [2]. I've used the build-in support for the latest snapshots provided at [3] of the script. But the problem still exists. Can anyone (perhaps Takashi again) help me? You can find all ALSA related information about my system at [4]. Regards, Martin [1] http://tinyurl.com/mkw8mz [2] http://ubuntuforums.org/showthread.php?p=6589810#post6589810 [3] http://tinyurl.com/5ux2nf [4] http://pastebin.ca/1471032

2 7

[alsa-devel] cmi8330: revert comments about AD1848 back
by Ondrej Zary 29 Jun '09

29 Jun '09

In ALSA 1.0.20, the comments were changed to say CMI8330 instead of AD1848. The CMI8330 chip includes two codecs - AD1848 and SB16, so the comments were correct and are misleading now. Revert them back. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- linux-2.6.30-orig/sound/isa/cmi8330.c 2009-06-10 05:05:27.000000000 +0200 +++ linux-test/sound/isa/cmi8330.c 2009-06-27 16:20:51.000000000 +0200 @@ -35,7 +35,7 @@ * * This card has two mixers and two PCM devices. I've cheesed it such * that recording and playback can be done through the same device. - * The driver "magically" routes the capturing to the CMI8330 codec, + * The driver "magically" routes the capturing to the AD1848 codec, * and playback to the SB16 codec. This allows for full-duplex mode * to some extent. * The utilities in alsa-utils are aware of both devices, so passing @@ -345,7 +345,7 @@ err = pnp_activate_dev(pdev); if (err < 0) { - snd_printk(KERN_ERR "CMI8330/C3D PnP configure failure\n"); + snd_printk(KERN_ERR "CMI8330/C3D (AD1848) PnP configure failure\n"); return -EBUSY; } wssport[dev] = pnp_port_start(pdev, 0); @@ -527,11 +527,11 @@ wssdma[dev], -1, WSS_HW_DETECT, 0, &acard->wss); if (err < 0) { - snd_printk(KERN_ERR PFX "(CMI8330) device busy??\n"); + snd_printk(KERN_ERR PFX "(AD1848) device busy??\n"); return err; } if (acard->wss->hardware != WSS_HW_CMI8330) { - snd_printk(KERN_ERR PFX "(CMI8330) not found during probe\n"); + snd_printk(KERN_ERR PFX "(AD1848) not found during probe\n"); return -ENODEV; } -- Ondrej Zary

2 1

[alsa-devel] cmi8330: fix MPU-401 PnP init copy&paste bug
by Ondrej Zary 29 Jun '09

29 Jun '09

Fix copy&paste bug in PnP MPU-401 initialization. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- linux-2.6.30-orig/sound/isa/cmi8330.c 2009-06-10 05:05:27.000000000 +0200 +++ linux-test/sound/isa/cmi8330.c 2009-06-27 13:37:56.000000000 +0200 @@ -338,7 +338,7 @@ return -EBUSY; acard->mpu = pnp_request_card_device(card, id->devs[2].id, NULL); - if (acard->play == NULL) + if (acard->mpu == NULL) return -EBUSY; pdev = acard->cap; -- Ondrej Zary

2 1

Re: [alsa-devel] Testers wanted: New SB X-Fi driver
by Takashi Iwai 28 Jun '09

28 Jun '09

At Sun, 07 Jun 2009 14:00:26 +0200, MonsieurWeller wrote: > > Thanks for reply. Hope Creative engineers read this :D. > Two other little things :D: > - The sound seems really really to loud compared to my audigy2 zs ( for > the same level: pcm/master/etc... 74 in alsamixer for the audigy vs 26 > for the xfi) Yes, it's a known issue. 100% doesn't correspond to 0dB but overdriving. 0dB is about 70% or so, and the volume scale doesn't look logarithmic. > - There's no cursor for bass and treble in alsamixer: not supported? Not supported. Some DSP code would be required, which isn't open yet. thanks, Takashi

6 23

Re: [alsa-devel] ice1712 error
by Takashi Iwai 28 Jun '09

28 Jun '09

At Sun, 28 Jun 2009 10:43:04 +0200, Udo van den Heuvel wrote: > > On 2009-06-28 10:39, Takashi Iwai wrote: > > At Sat, 27 Jun 2009 13:33:42 +0200, > > Udo van den Heuvel wrote: > >> Hello Takashi, > >> > >> WHat to do when I have 3 cards total, two cards of these 3 using the > >> same driver: hda-intel. > >> Hwo can I fix this one specific card to a snd slot? The computer somehow > >> puts the card in the wrong order. > >> (i.e.: C-B-A instead of A-B-C where A and C use hda-intel) > >> Can I specify PCI-id's similar to the USB vid/pid that can be passed? > > > > You can specify index option, e.g. index=2,1,0. > > Yes, OK, but hen index 0 and 2 are both hda-intel, how do I tie a > *specific* hda-intel card to each slot? There is no generic way. The assignment order is usually static and unchanged for such PCI devices. (Or, you can change the slot later via sysfs, if any.) In general, you shouldn't stick with the assignment order but rather use the id string to each device. Takashi

1 0

Re: [alsa-devel] ice1712 error
by Takashi Iwai 28 Jun '09

28 Jun '09

At Sat, 27 Jun 2009 13:33:42 +0200, Udo van den Heuvel wrote: > > Hello Takashi, > > WHat to do when I have 3 cards total, two cards of these 3 using the > same driver: hda-intel. > Hwo can I fix this one specific card to a snd slot? The computer somehow > puts the card in the wrong order. > (i.e.: C-B-A instead of A-B-C where A and C use hda-intel) > Can I specify PCI-id's similar to the USB vid/pid that can be passed? You can specify index option, e.g. index=2,1,0. Takashi

1 0

Re: [alsa-devel] snd_pcm_lib_read1 timeout
by Harsha, Priya 27 Jun '09

27 Jun '09

I found another clue to this but can anyone please explain what could be going wrong? The execution enters into the following function >static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream >*substream) >{ > .... > new_hw_ptr = hw_base + pos; this gets updated correctly > ... > hdelta = new_hw_ptr - old_hw_ptr; > jdelta = jiffies - runtime->hw_ptr_jiffies; as far as I don't get into the following loop, am good...the moment I get into this loop my new_hw_ptr gets modified and it does not progress in capture. Any clues? > if (((hdelta * HZ) / runtime->rate) > jdelta + HZ/100) { > delta = jdelta / > (((runtime->period_size * HZ) / runtime->rate) > + HZ/100); > hw_ptr_error(substream, > "hw_ptr skipping! [Q] " > "(pos=%ld, delta=%ld, period=%ld, " > "jdelta=%lu/%lu/%lu)\n", > (long)pos, (long)hdelta, > (long)runtime->period_size, jdelta, > ((hdelta * HZ) / runtime->rate), delta); > hw_ptr_interrupt = runtime->hw_ptr_interrupt + > runtime->period_size * delta; > if (hw_ptr_interrupt >= runtime->boundary) > hw_ptr_interrupt -= runtime->boundary; > /* rebase to interrupt position */ > hw_base = new_hw_ptr = hw_ptr_interrupt; > /* align hw_base to buffer_size */ > hw_base -= hw_base % runtime->buffer_size; > delta = 0; > } > ... >} Thanks, Harsha ______________________________________ From: Harsha, Priya Sent: Saturday, June 27, 2009 9:18 AM To: alsa-devel(a)alsa-project.org Subject: snd_pcm_lib_read1 timeout Hi, I have a basic question; There is a schedule_timeout() in snd_pcm_lib_read1 -> wait_for_avail_min. When and where does this timeout get signaled? What would be the reason that this timeout happens? Thanks, Harsha

1 0

Re: [alsa-devel] Should my Creative X-fi card be loading the snd-hda-intel module
by Susan Cragin 27 Jun '09

27 Jun '09

>> Shouldn't it be loading snd-hda-codec-ca0110 instead? > >snd-hda-codec-ca0110 is a slave module loaded by snd-hda-intel. > >Takashi The problem is, my Creative X-fi isn't being recognized any more, even though the modules are there and have been loaded. My onboard card is an hda-intel. And my utilities and programs don't find the Creative card. They only find the onboard card. I have added asoundconf to my alsa-utils, and asoundconf list shows only: Intel system (my usb card) winecfg shows the same. My goal is to get the card running again under wine. Susan

2 1

[alsa-devel] snd_pcm_lib_read1 timeout
by Harsha, Priya 27 Jun '09

27 Jun '09

Hi, I have a basic question; There is a schedule_timeout() in snd_pcm_lib_read1 -> wait_for_avail_min. When and where does this timeout get signaled? What would be the reason that this timeout happens? Thanks, Harsha

1 0

[alsa-devel] [PATCH] ASoC: TWL4030: Add EXT_MUTE gpio to reduce pop-noise effect
by Jorge Eduardo Candelaria 27 Jun '09

27 Jun '09

According to TRM, an external FET controlled by a 1.8V output signal can be used to reduce the pop-noise heard when the audio amplifier is switched on. The signal is controlled by a gpio pin and should be defined in the machine driver. However, the codec driver takes care of enabling and disabling this output during the headset pop attenuation sequence. If the board does not have an EXT_MUTE, then the machine driver should set the value of ext_mute_gpio to -EINVAL through ramp delay setup data, to bypass execution of the code that manages the gpio line. Also add a delay to let VMID settle in ramp up sequence. Signed-off-by: Jorge Eduardo Candelaria <x0107209(a)ti.com> --- sound/soc/codecs/twl4030.c | 23 +++++++++++++++++++++++ sound/soc/codecs/twl4030.h | 1 + 2 files changed, 24 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index df42fa2..e850b30 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -27,6 +27,7 @@ #include <linux/i2c.h> #include <linux/platform_device.h> #include <linux/i2c/twl4030.h> +#include <linux/gpio.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -620,6 +621,9 @@ static int handsfreerpga_event(struct snd_soc_dapm_widget *w, static void headset_ramp(struct snd_soc_codec *codec, int ramp) { + struct snd_soc_device *socdev = codec->socdev; + struct twl4030_setup_data *setup = socdev->codec_data; + unsigned char hs_gain, hs_pop; struct twl4030_priv *twl4030 = codec->private_data; /* Base values for ramp delay calculation: 2^19 - 2^26 */ @@ -629,6 +633,11 @@ static void headset_ramp(struct snd_soc_codec *codec, int ramp) hs_gain = twl4030_read_reg_cache(codec, TWL4030_REG_HS_GAIN_SET); hs_pop = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET); + /* Enable external mute, this dramatically reduces + * the pop-noise */ + if (setup && gpio_is_valid(setup->ext_mute_gpio)) + gpio_set_value(setup->ext_mute_gpio, 1); + if (ramp) { /* Headset ramp-up according to the TRM */ hs_pop |= TWL4030_VMID_EN; @@ -636,6 +645,9 @@ static void headset_ramp(struct snd_soc_codec *codec, int ramp) twl4030_write(codec, TWL4030_REG_HS_GAIN_SET, hs_gain); hs_pop |= TWL4030_RAMP_EN; twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop); + /* Wait ramp delay time + 1, so the VMID can settle */ + mdelay((ramp_base[(hs_pop & TWL4030_RAMP_DELAY) >> 2] / + twl4030->sysclk) + 1); } else { /* Headset ramp-down _not_ according to * the TRM, but in a way that it is working */ @@ -652,6 +664,10 @@ static void headset_ramp(struct snd_soc_codec *codec, int ramp) hs_pop &= ~TWL4030_VMID_EN; twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop); } + + /* Disable external mute */ + if (setup && gpio_is_valid(setup->ext_mute_gpio)) + gpio_set_value(setup->ext_mute_gpio, 0); } static int headsetlpga_event(struct snd_soc_dapm_widget *w, @@ -2101,6 +2117,7 @@ static int twl4030_init(struct snd_soc_device *socdev) /* Configuration for headset ramp delay from setup data */ if (setup) { unsigned char hs_pop; + unsigned int ext_mute; if (setup->sysclk) twl4030->sysclk = setup->sysclk; @@ -2111,6 +2128,12 @@ static int twl4030_init(struct snd_soc_device *socdev) hs_pop &= ~TWL4030_RAMP_DELAY; hs_pop |= (setup->ramp_delay_value << 2); twl4030_write_reg_cache(codec, TWL4030_REG_HS_POPN_SET, hs_pop); + + if (gpio_is_valid(setup->ext_mute_gpio)) { + ext_mute = setup->ext_mute_gpio; + BUG_ON(gpio_request(ext_mute, "ext_mute") < 0); + gpio_direction_output(ext_mute, 0); + } } else { twl4030->sysclk = 26000; } diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index fe5f395..23a75e2 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -274,6 +274,7 @@ extern struct snd_soc_codec_device soc_codec_dev_twl4030; struct twl4030_setup_data { unsigned int ramp_delay_value; unsigned int sysclk; + unsigned int ext_mute_gpio; }; #endif /* End of __TWL4030_AUDIO_H__ */ -- 1.6.0.4

4 4

[alsa-devel] routing audio stream to multiple devices
by Baek Chang 26 Jun '09

26 Jun '09

I was wondering if it was possible to route a single audio stream to multiple alsa devices or plugins? For example, I am playing a song and want it to stream to both my sound card at hw:0 and a bluetooth device, an alsa plugin. Is this possible? It seems that configuring asound.conf only lets you route streams serially to a single device at a time. Any info would be greatly appreciated. Thanks in advance Baek -- Baek Chang baeksan(a)ccrma.stanford.edu 407-267-6701

2 4

[alsa-devel] [PATCH] ASoC: Remove word "Switch" from Handsfree switch name
by Lopez Cruz, Misael 26 Jun '09

26 Jun '09

SoC dapm adds the suffix "Switch" to SND_SOC_DAPM_SWITCH controls, removing word "Switch" from HandsfreeL/HandsfreeR widget name for avoiding to duplicate it. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl4030.c | 12 ++++++------ 1 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index df42fa2..49ceb62 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1258,14 +1258,14 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { /* HandsfreeL/R */ SND_SOC_DAPM_MUX("HandsfreeL Mux", SND_SOC_NOPM, 0, 0, &twl4030_dapm_handsfreel_control), - SND_SOC_DAPM_SWITCH("HandsfreeL Switch", SND_SOC_NOPM, 0, 0, + SND_SOC_DAPM_SWITCH("HandsfreeL", SND_SOC_NOPM, 0, 0, &twl4030_dapm_handsfreelmute_control), SND_SOC_DAPM_PGA_E("HandsfreeL PGA", SND_SOC_NOPM, 0, 0, NULL, 0, handsfreelpga_event, SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_MUX("HandsfreeR Mux", SND_SOC_NOPM, 5, 0, &twl4030_dapm_handsfreer_control), - SND_SOC_DAPM_SWITCH("HandsfreeR Switch", SND_SOC_NOPM, 0, 0, + SND_SOC_DAPM_SWITCH("HandsfreeR", SND_SOC_NOPM, 0, 0, &twl4030_dapm_handsfreermute_control), SND_SOC_DAPM_PGA_E("HandsfreeR PGA", SND_SOC_NOPM, 0, 0, NULL, 0, handsfreerpga_event, @@ -1377,15 +1377,15 @@ static const struct snd_soc_dapm_route intercon[] = { {"HandsfreeL Mux", "AudioL1", "Analog L1 Playback Mixer"}, {"HandsfreeL Mux", "AudioL2", "Analog L2 Playback Mixer"}, {"HandsfreeL Mux", "AudioR2", "Analog R2 Playback Mixer"}, - {"HandsfreeL Switch", "Switch", "HandsfreeL Mux"}, - {"HandsfreeL PGA", NULL, "HandsfreeL Switch"}, + {"HandsfreeL", "Switch", "HandsfreeL Mux"}, + {"HandsfreeL PGA", NULL, "HandsfreeL"}, /* HandsfreeR */ {"HandsfreeR Mux", "Voice", "Analog Voice Playback Mixer"}, {"HandsfreeR Mux", "AudioR1", "Analog R1 Playback Mixer"}, {"HandsfreeR Mux", "AudioR2", "Analog R2 Playback Mixer"}, {"HandsfreeR Mux", "AudioL2", "Analog L2 Playback Mixer"}, - {"HandsfreeR Switch", "Switch", "HandsfreeR Mux"}, - {"HandsfreeR PGA", NULL, "HandsfreeR Switch"}, + {"HandsfreeR", "Switch", "HandsfreeR Mux"}, + {"HandsfreeR PGA", NULL, "HandsfreeR"}, /* Vibra */ {"Vibra Mux", "AudioL1", "DAC Left1"}, {"Vibra Mux", "AudioR1", "DAC Right1"}, -- 1.5.4.3

3 3

[alsa-devel] RFC: ASoC driver for S3C24XX Simtec boards
by Ben Dooks 26 Jun '09

26 Jun '09

I'd like some comments on the following driver which is currently being developed to get ASoC working on the TLV320AIC23 based Simtec boards. My question is: 1) I'm not seeing a control to enable/disable the 'Speaker Amp', do I need to change the controls or how this is exported? 2) There are two boards with an GPIO AMP which has two gain controls. This is a TPA2001D1 which provides a gain of 6,12,18 or 23.5dB depending on the GPIO settings. [ http://focus.ti.com/docs/prod/folders/print/tpa2001d1.html ] This driver has yet to be tested on all our boards. /* sound/soc/s3c24xx/s3c24xx_simtec.c * * Copyright 2009 Simtec Electronics * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as * published by the Free Software Foundation. */ #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/platform_device.h> #include <linux/gpio.h> #include <linux/clk.h> #include <linux/i2c.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> #include <sound/soc-dapm.h> #include <plat/audio-simtec.h> #include "s3c24xx-pcm.h" #include "s3c24xx-i2s.h" #include "../codecs/tlv320aic23.h" /* supported machines: * * Machine Connections AMP * ------- ----------- --- * BAST MIC, HPOUT, LOUT, LIN TPA2001D1 (HPOUTL,R) (gain hardwired) * VR1000 HPOUT, LIN None * VR2000 LIN, LOUT, MIC, HP LM4871 (HPOUTL,R) * DePicture LIN, LOUT, MIC, HP LM4871 (HPOUTL,R) * Anubis LIN, LOUT, MIC, HP TPA2001D1 (HPOUTL,R) */ static struct s3c24xx_audio_simtec_pdata *pdata; static struct clk *xtal_clk; /** * simtec_spk_event - handle power management event for speaker * @w: The widget for this event * @k: The sound control the event is for. * @event; The event being processed. * * Handle power management events to deal with the state of the * GPIO AMP attached to some boards. */ static int simtec_spk_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { int value = SND_SOC_DAPM_EVENT_ON(event) ? 0 : 1; gpio_set_value(pdata->amp_gpio, value); return 0; } static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_LINE("Line In", NULL), SND_SOC_DAPM_LINE("Line Out", NULL), SND_SOC_DAPM_MIC("Mic Jack", NULL), SND_SOC_DAPM_PGA_E("Speaker Amp", SND_SOC_NOPM, 0, 0, NULL, 0, simtec_spk_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), }; static const struct snd_soc_dapm_route audio_map[] = { { "Headphone Jack", NULL, "LHPOUT"}, { "Headphone Jack", NULL, "RHPOUT"}, { "Line Out", NULL, "LLOUT" }, { "Line Out", NULL, "RLOUT" }, { "LLINEIN", NULL, "Line In"}, { "RLINEIN", NULL, "Line In"}, { "MICIN", NULL, "Mic Jack"}, }; static const struct snd_soc_dapm_route amp_map[] = { { "Speaker Amp", NULL, "RHPOUT" }, { "Speaker Amp", NULL, "LHPOUT" }, }; static int spk_gain; /** * speaker_gain_get - read the speaker gain setting. * @kcontrol: The control for the speaker gain. * @ucontrol: The value that needs to be updated. * * Read the value for the AMP gain control. */ static int speaker_gain_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { ucontrol->value.integer.value[0] = spk_gain; return 0; } /** * speaker_gain_put - set the speaker gain setting. * @kcontrol: The control for the speaker gain. * @ucontrol: The value that needs to be set. * * Set the value of the speaker gain from the specified * @ucontrol setting. */ static int speaker_gain_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { int value = ucontrol->value.integer.value[0]; spk_gain = value; gpio_set_value_cansleep(pdata->amp_gain[0], value & 1); gpio_set_value_cansleep(pdata->amp_gain[1], value >> 1); return 0; } static const struct snd_kcontrol_new amp_controls[] = { SOC_SINGLE_EXT("Speaker Gain", 0, 0, 3, 0, speaker_gain_get, speaker_gain_put), }; /** * simtec_tlv320aic23_init - initialise and add controls * @codec; The codec instance to attach to. * * Attach our controls and configure the necessary codec * mappings for our sound card instance. */ static int simtec_tlv320aic23_init(struct snd_soc_codec *codec) { snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets, ARRAY_SIZE(tlv320aic23_dapm_widgets)); snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); if (pdata->amp_gpio > 0) { printk(KERN_DEBUG "%s: adding amp routes\n", __func__); snd_soc_dapm_add_routes(codec, amp_map, ARRAY_SIZE(amp_map)); snd_soc_dapm_enable_pin(codec, "Speaker Amp"); } if (pdata->amp_gain[0] > 0) { printk(KERN_DEBUG "%s: adding amp controls\n", __func__); snd_soc_add_controls(codec, amp_controls, ARRAY_SIZE(amp_controls)); } snd_soc_dapm_enable_pin(codec, "Headphone Jack"); snd_soc_dapm_enable_pin(codec, "Line In"); snd_soc_dapm_enable_pin(codec, "Line Out"); snd_soc_dapm_enable_pin(codec, "Mic Jack"); snd_soc_dapm_sync(codec); return 0; } #define CODEC_CLOCK 12000000 /** * simtec_hw_params - update hardware parameters * @substream: The audio substream instance. * @params: The parameters requested. * * Update the codec data routing and configuration settings * from the supplied data. */ static int simtec_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; int ret; /* Set the CODEC as the bus clock master, I2S */ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret) { printk(KERN_ERR "%s: failed set cpu dai format\n", __func__); return ret; } /* Set the CODEC as the bus clock master */ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret) { printk(KERN_ERR "%s: failed set codec dai format\n", __func__); return ret; } ret = snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN); if (ret) { printk(KERN_ERR "%s: failed setting codec sysclk\n", __func__); return ret; } if (pdata->use_mpllin) { ret = snd_soc_dai_set_sysclk(cpu_dai, S3C24XX_CLKSRC_MPLL, 0, SND_SOC_CLOCK_OUT); if (ret) { printk(KERN_ERR "%s: failed to set MPLLin as clksrc\n", __func__); return ret; } } if (pdata->output_cdclk) { int cdclk_scale; cdclk_scale = clk_get_rate(xtal_clk) / CODEC_CLOCK; cdclk_scale--; ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, cdclk_scale); } return 0; } static int simtec_hw_startup(struct snd_pcm_substream *substream) { /* call any board supplied startup code, this currently only * covers the bast/vr1000 which have a CPLD in the way of the * LRCLK */ if (pdata->startup) pdata->startup(); return 0; } static struct snd_soc_ops simtec_ops = { .startup = simtec_hw_startup, .hw_params = simtec_hw_params, }; static struct snd_soc_dai_link simtec_dai_aic23 = { .name = "tlv320aic23", .stream_name = "TLV320AIC23", .cpu_dai = &s3c24xx_i2s_dai, .codec_dai = &tlv320aic23_dai, .init = simtec_tlv320aic23_init, .ops = &simtec_ops, }; /* simtec audio machine driver */ static struct snd_soc_card snd_soc_machine_simtec_aic23 = { .name = "Simtec", .platform = &s3c24xx_soc_platform, .dai_link = &simtec_dai_aic23, .num_links = 1, }; /* simtec audio subsystem */ static struct snd_soc_device simtec_snd_devdata_aic23 = { .card = &snd_soc_machine_simtec_aic23, .codec_dev = &soc_codec_dev_tlv320aic23, }; /** * attach_gpio_amp - get and configure the necessary gpios * @dev: The device we're probing. * @pd: The platform data supplied by the board. * * If there is a GPIO based amplifier attached to the board, claim * the necessary GPIO lines for it, and set default values. */ static int attach_gpio_amp(struct device *dev, struct s3c24xx_audio_simtec_pdata *pd) { int ret; /* attach gpio amp gain (if any) */ if (pdata->amp_gain[0] > 0) { ret = gpio_request(pd->amp_gain[0], "gpio-amp-gain"); if (ret) { dev_err(dev, "cannot get amp gpio gain0\n"); return ret; } ret = gpio_request(pd->amp_gain[1], "gpio-amp-gain"); if (ret) { dev_err(dev, "cannot get amp gpio gain0\n"); gpio_free(pdata->amp_gain[0]); return ret; } gpio_direction_output(pd->amp_gain[0], 0); gpio_direction_output(pd->amp_gain[1], 0); } /* note, curently we assume GPA0 isn't valid amp */ if (pdata->amp_gpio > 0) { ret = gpio_request(pd->amp_gpio, "gpio-amp"); if (ret) { dev_err(dev, "cannot get amp gpio %d (%d)\n", pd->amp_gpio, ret); goto err_amp; } /* start with amp off */ gpio_direction_output(pd->amp_gpio, 1); } return 0; err_amp: if (pd->amp_gain[0] > 0) { gpio_free(pd->amp_gain[0]); gpio_free(pd->amp_gain[1]); } return ret; } static void detach_gpio_amp(struct s3c24xx_audio_simtec_pdata *pd) { if (pd->amp_gain[0] > 0) { gpio_free(pd->amp_gain[0]); gpio_free(pd->amp_gain[1]); } if (pd->amp_gpio > 0) gpio_free(pd->amp_gpio); } static int __devinit simtec_audio_probe(struct platform_device *pdev) { struct platform_device *snd_dev; int ret; dev_info(&pdev->dev, "probe called\n"); pdata = pdev->dev.platform_data; if (!pdata) { dev_err(&pdev->dev, "no platform data supplied\n"); return -EINVAL; } xtal_clk = clk_get(&pdev->dev, "xtal"); if (IS_ERR(xtal_clk)) { dev_err(&pdev->dev, "could not get clkout0\n"); return -EINVAL; } dev_info(&pdev->dev, "xtal rate is %ld\n", clk_get_rate(xtal_clk)); ret = attach_gpio_amp(&pdev->dev, pdata); if (ret) goto err_clk; snd_dev = platform_device_alloc("soc-audio", -1); if (!snd_dev) { dev_err(&pdev->dev, "failed to alloc soc-audio devicec\n"); ret = -ENOMEM; goto err_gpio; } platform_set_drvdata(snd_dev, &simtec_snd_devdata_aic23); simtec_snd_devdata_aic23.dev = &snd_dev->dev; ret = platform_device_add(snd_dev); if (ret) { dev_err(&pdev->dev, "failed to add soc-audio dev\n"); platform_device_put(snd_dev); goto err_clk; } platform_set_drvdata(pdev, snd_dev); return 0; err_gpio: detach_gpio_amp(pdata); err_clk: clk_put(xtal_clk); return ret; } static int __devexit simtec_audio_remove(struct platform_device *pdev) { struct platform_device *snd_dev = platform_get_drvdata(pdev); platform_device_unregister(snd_dev); detach_gpio_amp(pdata); clk_put(xtal_clk); return 0; } static struct platform_driver simtec_audio_platdrv = { .driver = { .owner = THIS_MODULE, .name = "s3c24xx-tlv320aic23", }, .probe = simtec_audio_probe, .remove = __devexit_p(simtec_audio_remove), }; MODULE_ALIAS("platform:s3c24xx-tlv320aic23"); static int __init simtec_init(void) { return platform_driver_register(&simtec_audio_platdrv); } static void __exit simtec_exit(void) { platform_driver_unregister(&simtec_audio_platdrv); } module_init(simtec_init); module_exit(simtec_exit); MODULE_AUTHOR("Ben Dooks <ben(a)simtec.co.uk>"); MODULE_DESCRIPTION("ALSA SoC Simtec Audio support"); MODULE_LICENSE("GPL");

3 3

Re: [alsa-devel] Regarding Sound on Acer Aspire 6935G
by Emilio López 26 Jun '09

26 Jun '09

Well, it seems my last email was too heavy with the 4 files attached. I have uploaded them to drop.io, here is the link: http://drop.io/alsaonaceraspire6930 Thanks! 2009/5/23 Emilio López <buhitoescolar(a)gmail.com> > Here you are, I ran alsa-info.sh in the old alsa (with model=acer) and > saved the output of the script when side was enabled and disabled, with > volumes 0 and 100. Hope you can work out the problem with this, do not > hesitate to ask for more debug info if you need. > > Thanks for all! > > Emilio > > PS: On my previous email (cited below) I wrote "Mic sensing" incorrectly, I > wanted to write "headphones sensing". > > 2009/5/23 Takashi Iwai <tiwai(a)suse.de> > > At Sat, 23 May 2009 12:19:12 -0300, >> Emilio López wrote: >> > >> > I reinstalled 1.0.20 today and almost everything started working when I >> set >> > model=acer-aspire-4930g. Mic sensing works, my mic now records too. But >> my >> > bass does not, and I cannot control it with "Side" anymore. >> > >> > Any idea on how to fix it? The audio setup for this laptop will be fully >> > working when I get the bass to play once again. >> >> Could you check with the older version and what "Side" control changes >> in hd-audio codec registers? You can just get alsa-info.sh output >> before and after changing the "Side" control value. >> >> >> thanks, >> >> Takashi >> > >

5 40

Re: [alsa-devel] Proposal for more reliable audio DMA.
by Mark Brown 26 Jun '09

26 Jun '09

On 24 Jun 2009, at 17:07, Jon Smirl <jonsmirl(a)gmail.com> wrote: > Trying to directly expose the ring buffer to the app seems like a good > > way to avoid a copy, but it isn't achieving that. pulse is not > decoding audio straight into the ring buffer, it decodes first and > then copies into the buffer. Pulse is using the timers to estimate the > destination for the copy. Move this copy down into the drivers, the > drivers know the correct destination for the copy. Pulse isn't just doing straight playback, a large part of what it's there for is to do software mixing. When you have multiple sources active pulse is going to be forced to do the copy as part of the mixing process so putting that bit of the buffer management in kernel won't help in the way you think it does. Part of what's going on here is that the kernel code is trying to give userspace access to the data for as long as possible.

4 7

[alsa-devel] [PATCH] ASoC: txx9aclc: dynamically allocate dmaengine devname
by Atsushi Nemoto 25 Jun '09

25 Jun '09

Use kasprintf to allocate temporary devname string instead of a fixed size string. This fixes "FIXME" introduced on removal of BUS_ID_SIZE. Signed-off-by: Atsushi Nemoto <anemo(a)mba.ocn.ne.jp> --- sound/soc/txx9/txx9aclc.c | 10 ++++++---- 1 files changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c index 938a58a..efed64b 100644 --- a/sound/soc/txx9/txx9aclc.c +++ b/sound/soc/txx9/txx9aclc.c @@ -297,15 +297,17 @@ static int txx9aclc_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, static bool filter(struct dma_chan *chan, void *param) { struct txx9aclc_dmadata *dmadata = param; - char devname[20 + 2]; /* FIXME: old BUS_ID_SIZE + 2 */ + char *devname; + bool found = false; - snprintf(devname, sizeof(devname), "%s.%d", dmadata->dma_res->name, + devname = kasprintf(GFP_KERNEL, "%s.%d", dmadata->dma_res->name, (int)dmadata->dma_res->start); if (strcmp(dev_name(chan->device->dev), devname) == 0) { chan->private = &dmadata->dma_slave; - return true; + found = true; } - return false; + kfree(devname); + return found; } static int txx9aclc_dma_init(struct txx9aclc_soc_device *dev, -- 1.5.6.5

2 1

[alsa-devel] Reenable support for Auzentech X-Fi Prelude in snd-ctxfi module
by Kamil Kaminski 25 Jun '09

25 Jun '09

Hi guys, I noticed that latest alsa-driver-unstable does not detect my card as if PCI IDs were discarded or purposely disabled to only work on Creative cards, the earlier snapshots used to work fine. This is what I get: ctatc: Card not recognised ctxfi: Something wrong!!! SB-XFi: probe of 0000:06:01.0 failed with error -2 This is the card: 06:01.0 Multimedia audio controller: Creative Labs SB X-Fi Subsystem: Auzentech, Inc. Device 0034 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64 (1000ns min, 1250ns max) Interrupt: pin A routed to IRQ 22 Region 0: I/O ports at 8f00 [size=32] Region 1: Memory at dfc00000 (64-bit, non-prefetchable) [size=2M] Region 3: Memory at d8000000 (64-bit, non-prefetchable) [size=64M] Capabilities: [40] Power Management version 2 Flags: PMEClk- DSI+ D1+ D2+ AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [50] MSI: Mask- 64bit+ Count=1/1 Enable- Address: 0000000000000000 Data: 0000 Kernel driver in use: CTALSA Kernel modules: ctxfi I'm currently using v1.0 ctxfi alsa driver from Creative, It would be nice to have Auzentech X-Fi Prelude working in 2.6.31 kernel ;) Thanks.

4 3

[alsa-devel] Proposal for more reliable audio DMA.
by Jon Smirl 25 Jun '09

25 Jun '09

This algorithm is fairly similar to what currently exists in ALSA with a few modifications. I'm trying to come up with guaranteed way to play glitch free audio. Does this algorithm work or could it be modified to work? The main change is to track a high resolution timer as the means of estimating where in the buffer to insert new, low latency data. This tracking is done in a timer tick interrupt. Jiffies are too coarse and not all hardware allows you to ask the current DMA position. The other change is to keep all of the buffers filled with silence if there isn't any pending data. (not doing this caused problem in my mpc5200 AC97 driver) Three buffers are used as a way to deterministically bound the DMA pointer without needing interrupts. --- Use three chained buffers. Buffer size is samples/tick (or maybe 1.5 samples/tick) Initialize buffers to silence. Set end of buffer three to automatically terminate DMA Set FIFO to minimum bus allows Fill buffer one with samples available, start playing. On timer tick call into driver.. If buffer one has finished playing move it to third position. New last buffer is set to automatically stop DMA. Return swap status to ALSA If swapped, fill buffer three with silence or pending data It is important to fill this buffer with silence if there is no pending data The size of the buffers needs to be large enough to cover worst case timer tick latency. Buffer two has to be large enough to ensure that the tick routine will notice buffer one has been played before buffer three starts. If two buffers have been completed when the tick runs, let the third buffer finish without swapping. Since you don't know where you are in the third buffer, it is unsafe to swap at this point. Playing the last buffer causes the end of buffer interrupt to happen. Callback into ALSA to alert it of the underrun and need to restart DMA. If this callback happens, make the buffers larger. Long term observation of buffer completion status in tick handler will allow accurate computation of samples/HPET unit. Record high accuracy timer (HPET) on tick. New low latency data that arrives can be inserted into the buffers dynamically. Use the high resolution timer source to estimate where to place it. Minimum FIFO ensures low latency play. Design a minimum power mode. Allows a huge FIFO to be loaded (like 128KB). Call into driver to reset to minimum latency, small FIFO mode. Advantages: 1) If ALSA goes away, hardware will stop on it's own with no noise. 2) No need to know the current position of DMA hardware. 3) Both low and high latency modes. 4) Audio interrupts only generated as error condition 5) Behavior is deterministic. Nothing is left to guess work. -- Jon Smirl jonsmirl(a)gmail.com

4 9

Re: [alsa-devel] ASoC updates for 2.6.31
by Mark Brown 25 Jun '09

25 Jun '09

The following changes since commit 28d0325ce6e0a52f53d8af687e6427fee59004d3: Linus Torvalds (1): Linux 2.6.31-rc1 are available in the git repository at: ssh://master.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31 Peter Ujfalusi (1): MAINTAINERS: Add entry for twl4030 series soc codec driver MAINTAINERS | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-)

2 1

[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 25 Jun '09

25 Jun '09

The following changes since commit 28d0325ce6e0a52f53d8af687e6427fee59004d3: Linus Torvalds (1): Linux 2.6.31-rc1 are available in the git repository at: ssh://master.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Barry Song (1): ASoC: Blackfin: convert internal names from bf52x to bf5xx Daniel Ribeiro (2): ASoC: Allow 32 bit registers for DAPM ASoC: remove duplicated code on pxa-ssp.c Lopez Cruz, Misael (4): ASoC: Zoom2: Add machine driver for Zoom2 board ASoC: TWL4030: Fix voice interface clock masters ASoC: TWL4030: Add AVADC Clock Priority ASoC: TWL4030: Correct bypass event for voice sidetone Mark Brown (21): ASoC: Sort DAPM power sequences while building lists ASoC: Factor out DAPM sequence execution ASoC: Coalesce register writes for DAPM sequences ASoC: Sort specialised mixers and muxes together ASoC: Coalesce power updates for DAPM widgets with events ASoC: Coalesce power updates for PGAs ASoC: Make DAPM power sequence lists local variables Merge branch 'dapm' into for-2.6.32 ASoC: Add WM8961 driver ASoC: Automatically control WM8903 sloping stopband filter ASoC: Automatically manage WM8900 sloping stopband filter ASoC: Automatically manage WM8350 sloping stopband filter ASoC: Add stub suspend and resume calls for ASoC subdevices ASoC: Add suspend and resume callbacks to Wolfson CODEC drivers ASoC: Regulator support for WM8580 ASoC: Fix shadowed variables in twl4030 ASoC: Staticise put_twl4030_opmode_enum_double() Merge branch 'wm8961' into for-2.6.32 ASoC: Add a shutdown callback ASoC: Add core suspend and resume callbacks to WM8961 Merge branch 'for-2.6.31' into for-2.6.32 Peter Ujfalusi (1): MAINTAINERS: Add entry for twl4030 series soc codec driver Philipp Zabel (1): ASoC: UDA1380: refactor device registration MAINTAINERS | 7 + include/sound/soc-dapm.h | 1 + include/sound/soc.h | 11 +- include/sound/uda1380.h | 22 + sound/soc/blackfin/bf5xx-ad73311.c | 16 +- sound/soc/blackfin/bf5xx-ssm2602.c | 16 +- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/twl4030.c | 38 +- sound/soc/codecs/uda1380.c | 313 +++++---- sound/soc/codecs/uda1380.h | 8 - sound/soc/codecs/wm8350.c | 40 +- sound/soc/codecs/wm8400.c | 17 + sound/soc/codecs/wm8580.c | 55 ++- sound/soc/codecs/wm8731.c | 39 ++ sound/soc/codecs/wm8753.c | 35 + sound/soc/codecs/wm8900.c | 30 +- sound/soc/codecs/wm8903.c | 27 +- sound/soc/codecs/wm8940.c | 17 + sound/soc/codecs/wm8960.c | 17 + sound/soc/codecs/wm8961.c | 1326 ++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8961.h | 866 +++++++++++++++++++++++ sound/soc/codecs/wm8988.c | 34 + sound/soc/codecs/wm9081.c | 17 + sound/soc/omap/Kconfig | 7 + sound/soc/omap/Makefile | 2 + sound/soc/omap/sdp3430.c | 2 +- sound/soc/omap/zoom2.c | 301 ++++++++ sound/soc/pxa/magician.c | 54 +- sound/soc/pxa/pxa-ssp.c | 50 +- sound/soc/soc-core.c | 83 ++- sound/soc/soc-dapm.c | 347 +++++++--- 32 files changed, 3452 insertions(+), 352 deletions(-) create mode 100644 include/sound/uda1380.h create mode 100644 sound/soc/codecs/wm8961.c create mode 100644 sound/soc/codecs/wm8961.h create mode 100644 sound/soc/omap/zoom2.c

1 0

[alsa-devel] [PATCH 0/1] Add maintainer for the twl4030 codec
by Peter Ujfalusi 25 Jun '09

25 Jun '09

Hello, Mark Brown kindly asked me to send the following patch to add myself to the MAINTAINERS file for the twl4030 series soc codec driver. I have thought about it and here is the patch. Thank you Mark for the suggestion, I would not have sent this patch otherwise. Thanks for Steve Sakoman to write the initial twl4030 codec driver. I'll try my best with the maintainer responsibility! --- Peter Ujfalusi (1): MAINTAINERS: Add entry for twl4030 series soc codec driver MAINTAINERS | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-)

3 3

[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 25 Jun '09

25 Jun '09

The following changes since commit 28d0325ce6e0a52f53d8af687e6427fee59004d3: Linus Torvalds (1): Linux 2.6.31-rc1 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Barry Song (1): ASoC: Blackfin: convert internal names from bf52x to bf5xx Daniel Ribeiro (2): ASoC: Allow 32 bit registers for DAPM ASoC: remove duplicated code on pxa-ssp.c Lopez Cruz, Misael (4): ASoC: Zoom2: Add machine driver for Zoom2 board ASoC: TWL4030: Fix voice interface clock masters ASoC: TWL4030: Add AVADC Clock Priority ASoC: TWL4030: Correct bypass event for voice sidetone Mark Brown (20): ASoC: Sort DAPM power sequences while building lists ASoC: Factor out DAPM sequence execution ASoC: Coalesce register writes for DAPM sequences ASoC: Sort specialised mixers and muxes together ASoC: Coalesce power updates for DAPM widgets with events ASoC: Coalesce power updates for PGAs ASoC: Make DAPM power sequence lists local variables Merge branch 'dapm' into for-2.6.32 ASoC: Add WM8961 driver ASoC: Automatically control WM8903 sloping stopband filter ASoC: Automatically manage WM8900 sloping stopband filter ASoC: Automatically manage WM8350 sloping stopband filter ASoC: Add stub suspend and resume calls for ASoC subdevices ASoC: Add suspend and resume callbacks to Wolfson CODEC drivers ASoC: Regulator support for WM8580 ASoC: Fix shadowed variables in twl4030 ASoC: Staticise put_twl4030_opmode_enum_double() Merge branch 'wm8961' into for-2.6.32 ASoC: Add a shutdown callback Merge branch 'topic/asoc' of git://git.kernel.org/.../tiwai/sound-2.6 into for-2.6.32 Philipp Zabel (1): ASoC: UDA1380: refactor device registration include/sound/soc-dapm.h | 1 + include/sound/soc.h | 11 +- include/sound/uda1380.h | 22 + sound/soc/blackfin/bf5xx-ad73311.c | 16 +- sound/soc/blackfin/bf5xx-ssm2602.c | 16 +- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/twl4030.c | 38 +- sound/soc/codecs/uda1380.c | 313 +++++---- sound/soc/codecs/uda1380.h | 8 - sound/soc/codecs/wm8350.c | 40 +- sound/soc/codecs/wm8400.c | 17 + sound/soc/codecs/wm8580.c | 55 ++- sound/soc/codecs/wm8731.c | 39 ++ sound/soc/codecs/wm8753.c | 35 + sound/soc/codecs/wm8900.c | 30 +- sound/soc/codecs/wm8903.c | 27 +- sound/soc/codecs/wm8940.c | 17 + sound/soc/codecs/wm8960.c | 17 + sound/soc/codecs/wm8961.c | 1309 ++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8961.h | 866 ++++++++++++++++++++++++ sound/soc/codecs/wm8988.c | 34 + sound/soc/codecs/wm9081.c | 17 + sound/soc/omap/Kconfig | 7 + sound/soc/omap/Makefile | 2 + sound/soc/omap/sdp3430.c | 2 +- sound/soc/omap/zoom2.c | 301 +++++++++ sound/soc/pxa/magician.c | 54 +- sound/soc/pxa/pxa-ssp.c | 50 +- sound/soc/soc-core.c | 83 ++- sound/soc/soc-dapm.c | 347 ++++++++--- 31 files changed, 3428 insertions(+), 352 deletions(-) create mode 100644 include/sound/uda1380.h create mode 100644 sound/soc/codecs/wm8961.c create mode 100644 sound/soc/codecs/wm8961.h create mode 100644 sound/soc/omap/zoom2.c

2 5

[alsa-devel] Support for MSM7 in ASoC
by Didier "Ptitjes" 25 Jun '09

25 Jun '09

Hello all, I'm working together with FreeSmartphone.org and SHR. We would like to support the MSM7-based phones. We already have the modem working, but audio is a show-stopper for us, as actual implementation of the audio subsystem in Android is in binary form, and does not us Alsa !!! Can someone look at the specs for the MSM7200A specs and tell whether it could be difficult to support its audio subsystem ? Thanks for your help, Didier.

3 3

Re: [alsa-devel] Reg: Implementation of GSM digital audio path
by Peter Ujfalusi 25 Jun '09

25 Jun '09

On Thursday 25 June 2009 14:11:04 ext shariff wrote: > Hi Peter, > > Thanks for quick response. I understood some what from your mail. > > I will explain, how my hardware is connected. > > > USB GSM chip -> McBSP3 > > McBSP2 -> TWL4030 > > Here, playback functionality will not be there in the GSM path. I > need to take audio data from GSM. and redirect this to TWL4030 codec. I see. Sorry I have confused things a bit, In this case you need to do what Mark has suggested: dummy dai for the codec, connect it to McBSP3 as it were a normal codec. Than you will have two PCM on the sound card: 0.0: TWL digital audio 0.1: Your codec arecord -Dplughw:0.1 | aplay -Dplughw:0.0 or arecord -Dhw:0.1 | aplay -Dhw:0.0 > > The thing is, How and where to implement this GSM path without > playback? as I am new to this ASoC framework. > > What I understood from ur mail is. Audio is being redirected from > the application space i.e arecord will record and store in buffer and this > buffer is > > being played using aplay to other hardware (twl4030). > > Can we handle the same thing in kernel space ( i.e in driver itself). You anyway have to copy from one buffer to another to be in the safe side (in theory you could use the same buffer for recording and playback, but it should be avoided). So you have to copy the samples from the capture buffer to the playback buffer. It really does not matter if it is done inside of the kernel or in user space. This kind of 'routing' clearly belongs to the user space. > > Regards, > Shariff -- Péter

2 1

[alsa-devel] Reg: Implementation of GSM digital audio path
by shariff 25 Jun '09

25 Jun '09

Hi all, Iam working on OMAP3 based custom board. On our board, GSM digital audio path is connected to McBSP3 and TWL4030 digital audio path is connected to McBSP2. I want to implement a module that transfers GSM audio data and store audio data into some buffer and redirect it to TWL4030 audio codec through McBSP2. I question is how can I implement this in ASoC? I studied ASoC framework, but I am unable to understand where to implement my stuff. Regards, Shariff

3 4

[alsa-devel] [PATCH 3/3] ASoC: TWL4030: Fix voice interface clock masters
by Lopez Cruz, Misael 25 Jun '09

25 Jun '09

Voice interface of twl4030 codec supports: CBM_CFM and CBS_CFS. It doesn't support CBS_CFM. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl4030.c | 2 +- sound/soc/omap/sdp3430.c | 2 +- sound/soc/omap/zoom2.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index fd5b30f..a6fb1d6 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1969,7 +1969,7 @@ static int twl4030_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFM: + case SND_SOC_DAIFMT_CBM_CFM: format &= ~(TWL4030_VIF_SLAVE_EN); break; case SND_SOC_DAIFMT_CBS_CFS: diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index b719e5d..c51594d 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -96,7 +96,7 @@ static int sdp3430_hw_voice_params(struct snd_pcm_substream *substream, ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF | - SND_SOC_DAIFMT_CBS_CFM); + SND_SOC_DAIFMT_CBM_CFM); if (ret) { printk(KERN_ERR "can't set codec DAI configuration\n"); return ret; diff --git a/sound/soc/omap/zoom2.c b/sound/soc/omap/zoom2.c index 17ae5c2..c84bc1c 100644 --- a/sound/soc/omap/zoom2.c +++ b/sound/soc/omap/zoom2.c @@ -89,7 +89,7 @@ static int zoom2_hw_voice_params(struct snd_pcm_substream *substream, ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF | - SND_SOC_DAIFMT_CBS_CFM); + SND_SOC_DAIFMT_CBM_CFM); if (ret) { printk(KERN_ERR "can't set codec DAI configuration\n"); return ret; -- 1.5.4.3

5 7

[alsa-devel] [PATCH] alsa: lx6464es - configure ethersound io channels
by Tim Blechmann 25 Jun '09

25 Jun '09

as long as the io channel number is not set by the driver, the card is not visible from the ethersound network Signed-off-by: Tim Blechmann <tim(a)klingt.org> --- sound/pci/lx6464es/lx6464es.c | 7 +++---- 1 files changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c index ccf1b38..c92fba5 100644 --- a/sound/pci/lx6464es/lx6464es.c +++ b/sound/pci/lx6464es/lx6464es.c @@ -654,13 +654,12 @@ static int __devinit lx_init_ethersound_config(struct lx6464es *chip) int i; u32 orig_conf_es = lx_dsp_reg_read(chip, eReg_CONFES); - u32 default_conf_es = (64 << IOCR_OUTPUTS_OFFSET) | + /* configure 64 io channels */ + u32 conf_es = (orig_conf_es & CONFES_READ_PART_MASK) | (64 << IOCR_INPUTS_OFFSET) | + (64 << IOCR_OUTPUTS_OFFSET) | (FREQ_RATIO_SINGLE_MODE << FREQ_RATIO_OFFSET); - u32 conf_es = (orig_conf_es & CONFES_READ_PART_MASK) - | (default_conf_es & CONFES_WRITE_PART_MASK); - snd_printdd("->lx_init_ethersound\n"); chip->freq_ratio = FREQ_RATIO_SINGLE_MODE; -- 1.6.2.1

2 1

[alsa-devel] Patch for suspend/resume in ICE1724 for Audiotrak Prodigy HD2
by Igor Chernyshev 25 Jun '09

25 Jun '09

Hi Alsa Developers! I've built a small HTPC and had to add suspend/resume support in ice1724 driver. There seem to be 3 existing bugs related to that: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4413 https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3748 https://bugtrack.alsa-project.org/alsa-bug/view.php?id=2314 Due to hardware (un)availability, I only enabled the fix for Audiotrak Prodigy HD2 card, which is installed in my HTPC. However, most of my code should be reusable in the future on other ice1724-based cards as well (as long as people add card-specific peices of code). The fix is currently based on ALSA 1.0.20 and works on my MythBuntu 9.04 HTPC (using 2.6.28-11 kernel). The following files have been modified: alsa-kernel/pci/ice1712/ice1712.h alsa-kernel/pci/ice1712/ice1724.c alsa-kernel/pci/ice1712/prodigy_hifi.c Now, I'm a complete newbie in Linux development and not sure I can safely rebuild latest Linux Kernel under MythBuntu, etc. Are there any resources for people like me to read? I want to provide the patch without destroying my computer a hunder times, but also make it more convenient and safer for ALSA maintainers. Thanks, Igor

2 9

[alsa-devel] [PATCH 19/19] sound: Use PCI_VDEVICE
by Joe Perches 25 Jun '09

25 Jun '09

Signed-off-by: Joe Perches <joe(a)perches.com> --- sound/oss/kahlua.c | 2 +- sound/pci/atiixp.c | 8 +++--- sound/pci/atiixp_modem.c | 4 +- sound/pci/au88x0/au8810.c | 3 +- sound/pci/au88x0/au8820.c | 3 +- sound/pci/au88x0/au8830.c | 3 +- sound/pci/cmipci.c | 10 ++++---- sound/pci/cs4281.c | 2 +- sound/pci/cs46xx/cs46xx.c | 6 ++-- sound/pci/ens1370.c | 6 ++-- sound/pci/es1938.c | 2 +- sound/pci/ice1712/ice1712.c | 2 +- sound/pci/ice1712/ice1724.c | 2 +- sound/pci/intel8x0.c | 46 +++++++++++++++++++++--------------------- sound/pci/intel8x0m.c | 34 +++++++++++++++--------------- sound/pci/mixart/mixart.c | 2 +- sound/pci/nm256/nm256.c | 6 ++-- sound/pci/rme32.c | 9 ++----- sound/pci/rme96.c | 12 +++------- sound/pci/sonicvibes.c | 2 +- sound/pci/via82xx.c | 4 +- sound/pci/via82xx_modem.c | 2 +- sound/pci/ymfpci/ymfpci.c | 12 +++++----- 23 files changed, 86 insertions(+), 96 deletions(-) diff --git a/sound/oss/kahlua.c b/sound/oss/kahlua.c index c180598..89466b0 100644 --- a/sound/oss/kahlua.c +++ b/sound/oss/kahlua.c @@ -199,7 +199,7 @@ MODULE_LICENSE("GPL"); */ static struct pci_device_id id_tbl[] = { - { PCI_VENDOR_ID_CYRIX, PCI_DEVICE_ID_CYRIX_5530_AUDIO, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, + { PCI_VDEVICE(CYRIX, PCI_DEVICE_ID_CYRIX_5530_AUDIO), 0 }, { } }; diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index 71515dd..d6752df 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -287,10 +287,10 @@ struct atiixp { /* */ static struct pci_device_id snd_atiixp_ids[] = { - { 0x1002, 0x4341, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* SB200 */ - { 0x1002, 0x4361, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* SB300 */ - { 0x1002, 0x4370, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* SB400 */ - { 0x1002, 0x4382, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* SB600 */ + { PCI_VDEVICE(ATI, 0x4341), 0 }, /* SB200 */ + { PCI_VDEVICE(ATI, 0x4361), 0 }, /* SB300 */ + { PCI_VDEVICE(ATI, 0x4370), 0 }, /* SB400 */ + { PCI_VDEVICE(ATI, 0x4382), 0 }, /* SB600 */ { 0, } }; diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c index c3136cc..e7e147b 100644 --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -262,8 +262,8 @@ struct atiixp_modem { /* */ static struct pci_device_id snd_atiixp_ids[] = { - { 0x1002, 0x434d, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* SB200 */ - { 0x1002, 0x4378, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* SB400 */ + { PCI_VDEVICE(ATI, 0x434d), 0 }, /* SB200 */ + { PCI_VDEVICE(ATI, 0x4378), 0 }, /* SB400 */ { 0, } }; diff --git a/sound/pci/au88x0/au8810.c b/sound/pci/au88x0/au8810.c index fce22c7..c0e8c6b 100644 --- a/sound/pci/au88x0/au8810.c +++ b/sound/pci/au88x0/au8810.c @@ -1,8 +1,7 @@ #include "au8810.h" #include "au88x0.h" static struct pci_device_id snd_vortex_ids[] = { - {PCI_VENDOR_ID_AUREAL, PCI_DEVICE_ID_AUREAL_ADVANTAGE, - PCI_ANY_ID, PCI_ANY_ID, 0, 0, 1,}, + {PCI_VDEVICE(AUREAL, PCI_DEVICE_ID_AUREAL_ADVANTAGE), 1,}, {0,} }; diff --git a/sound/pci/au88x0/au8820.c b/sound/pci/au88x0/au8820.c index d1fbcce..a652733 100644 --- a/sound/pci/au88x0/au8820.c +++ b/sound/pci/au88x0/au8820.c @@ -1,8 +1,7 @@ #include "au8820.h" #include "au88x0.h" static struct pci_device_id snd_vortex_ids[] = { - {PCI_VENDOR_ID_AUREAL, PCI_DEVICE_ID_AUREAL_VORTEX_1, - PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0,}, + {PCI_VDEVICE(AUREAL, PCI_DEVICE_ID_AUREAL_VORTEX_1), 0,}, {0,} }; diff --git a/sound/pci/au88x0/au8830.c b/sound/pci/au88x0/au8830.c index d4f2717..6c702ad 100644 --- a/sound/pci/au88x0/au8830.c +++ b/sound/pci/au88x0/au8830.c @@ -1,8 +1,7 @@ #include "au8830.h" #include "au88x0.h" static struct pci_device_id snd_vortex_ids[] = { - {PCI_VENDOR_ID_AUREAL, PCI_DEVICE_ID_AUREAL_VORTEX_2, - PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0,}, + {PCI_VDEVICE(AUREAL, PCI_DEVICE_ID_AUREAL_VORTEX_2), 0,}, {0,} }; diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index 449fe02..ddcd4a9 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -2797,11 +2797,11 @@ static inline void snd_cmipci_proc_init(struct cmipci *cm) {} static struct pci_device_id snd_cmipci_ids[] = { - {PCI_VENDOR_ID_CMEDIA, PCI_DEVICE_ID_CMEDIA_CM8338A, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0}, - {PCI_VENDOR_ID_CMEDIA, PCI_DEVICE_ID_CMEDIA_CM8338B, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0}, - {PCI_VENDOR_ID_CMEDIA, PCI_DEVICE_ID_CMEDIA_CM8738, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0}, - {PCI_VENDOR_ID_CMEDIA, PCI_DEVICE_ID_CMEDIA_CM8738B, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0}, - {PCI_VENDOR_ID_AL, PCI_DEVICE_ID_CMEDIA_CM8738, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0}, + {PCI_VDEVICE(CMEDIA, PCI_DEVICE_ID_CMEDIA_CM8338A), 0}, + {PCI_VDEVICE(CMEDIA, PCI_DEVICE_ID_CMEDIA_CM8338B), 0}, + {PCI_VDEVICE(CMEDIA, PCI_DEVICE_ID_CMEDIA_CM8738), 0}, + {PCI_VDEVICE(CMEDIA, PCI_DEVICE_ID_CMEDIA_CM8738B), 0}, + {PCI_VDEVICE(AL, PCI_DEVICE_ID_CMEDIA_CM8738), 0}, {0,}, }; diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c index f6286f8..e2e0359 100644 --- a/sound/pci/cs4281.c +++ b/sound/pci/cs4281.c @@ -495,7 +495,7 @@ struct cs4281 { static irqreturn_t snd_cs4281_interrupt(int irq, void *dev_id); static struct pci_device_id snd_cs4281_ids[] = { - { 0x1013, 0x6005, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* CS4281 */ + { PCI_VDEVICE(CIRRUS, 0x6005), 0, }, /* CS4281 */ { 0, } }; diff --git a/sound/pci/cs46xx/cs46xx.c b/sound/pci/cs46xx/cs46xx.c index c9b3e3d..004a59a 100644 --- a/sound/pci/cs46xx/cs46xx.c +++ b/sound/pci/cs46xx/cs46xx.c @@ -65,9 +65,9 @@ module_param_array(mmap_valid, bool, NULL, 0444); MODULE_PARM_DESC(mmap_valid, "Support OSS mmap."); static struct pci_device_id snd_cs46xx_ids[] = { - { 0x1013, 0x6001, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* CS4280 */ - { 0x1013, 0x6003, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* CS4612 */ - { 0x1013, 0x6004, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* CS4615 */ + { PCI_VDEVICE(CIRRUS, 0x6001), 0, }, /* CS4280 */ + { PCI_VDEVICE(CIRRUS, 0x6003), 0, }, /* CS4612 */ + { PCI_VDEVICE(CIRRUS, 0x6004), 0, }, /* CS4615 */ { 0, } }; diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index 18f4d1e..d589bbc 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -445,11 +445,11 @@ 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 */ + { PCI_VDEVICE(ENSONIQ, 0x5000), 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 */ + { PCI_VDEVICE(ENSONIQ, 0x1371), 0, }, /* ES1371 */ + { PCI_VDEVICE(ENSONIQ, 0x5880), 0, }, /* ES1373 - CT5880 */ { 0x1102, 0x8938, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* Ectiva EV1938 */ #endif { 0, } diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c index fbd2ac0..b47232c 100644 --- a/sound/pci/es1938.c +++ b/sound/pci/es1938.c @@ -244,7 +244,7 @@ struct es1938 { static irqreturn_t snd_es1938_interrupt(int irq, void *dev_id); static struct pci_device_id snd_es1938_ids[] = { - { 0x125d, 0x1969, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* Solo-1 */ + { PCI_VDEVICE(ESS, 0x1969), 0, }, /* Solo-1 */ { 0, } }; diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 0d0cdbd..cecf1ff 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -107,7 +107,7 @@ MODULE_PARM_DESC(dxr_enable, "Enable DXR support for Terratec DMX6FIRE."); static const struct pci_device_id snd_ice1712_ids[] = { - { PCI_VENDOR_ID_ICE, PCI_DEVICE_ID_ICE_1712, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* ICE1712 */ + { PCI_VDEVICE(ICE, PCI_DEVICE_ID_ICE_1712), 0 }, /* ICE1712 */ { 0, } }; diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 36ade77..cc84a83 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -93,7 +93,7 @@ MODULE_PARM_DESC(model, "Use the given board model."); /* Both VT1720 and VT1724 have the same PCI IDs */ static const struct pci_device_id snd_vt1724_ids[] = { - { PCI_VENDOR_ID_ICE, PCI_DEVICE_ID_VT1724, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, + { PCI_VDEVICE(ICE, PCI_DEVICE_ID_VT1724), 0 }, { 0, } }; diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 8aa5687..171ada5 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -421,29 +421,29 @@ struct intel8x0 { }; static struct pci_device_id snd_intel8x0_ids[] = { - { 0x8086, 0x2415, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* 82801AA */ - { 0x8086, 0x2425, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* 82901AB */ - { 0x8086, 0x2445, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* 82801BA */ - { 0x8086, 0x2485, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* ICH3 */ - { 0x8086, 0x24c5, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL_ICH4 }, /* ICH4 */ - { 0x8086, 0x24d5, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL_ICH4 }, /* ICH5 */ - { 0x8086, 0x25a6, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL_ICH4 }, /* ESB */ - { 0x8086, 0x266e, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL_ICH4 }, /* ICH6 */ - { 0x8086, 0x27de, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL_ICH4 }, /* ICH7 */ - { 0x8086, 0x2698, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL_ICH4 }, /* ESB2 */ - { 0x8086, 0x7195, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* 440MX */ - { 0x1039, 0x7012, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_SIS }, /* SI7012 */ - { 0x10de, 0x01b1, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_NFORCE }, /* NFORCE */ - { 0x10de, 0x003a, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_NFORCE }, /* MCP04 */ - { 0x10de, 0x006a, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_NFORCE }, /* NFORCE2 */ - { 0x10de, 0x0059, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_NFORCE }, /* CK804 */ - { 0x10de, 0x008a, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_NFORCE }, /* CK8 */ - { 0x10de, 0x00da, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_NFORCE }, /* NFORCE3 */ - { 0x10de, 0x00ea, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_NFORCE }, /* CK8S */ - { 0x10de, 0x026b, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_NFORCE }, /* MCP51 */ - { 0x1022, 0x746d, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* AMD8111 */ - { 0x1022, 0x7445, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* AMD768 */ - { 0x10b9, 0x5455, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_ALI }, /* Ali5455 */ + { PCI_VDEVICE(INTEL, 0x2415), DEVICE_INTEL }, /* 82801AA */ + { PCI_VDEVICE(INTEL, 0x2425), DEVICE_INTEL }, /* 82901AB */ + { PCI_VDEVICE(INTEL, 0x2445), DEVICE_INTEL }, /* 82801BA */ + { PCI_VDEVICE(INTEL, 0x2485), DEVICE_INTEL }, /* ICH3 */ + { PCI_VDEVICE(INTEL, 0x24c5), DEVICE_INTEL_ICH4 }, /* ICH4 */ + { PCI_VDEVICE(INTEL, 0x24d5), DEVICE_INTEL_ICH4 }, /* ICH5 */ + { PCI_VDEVICE(INTEL, 0x25a6), DEVICE_INTEL_ICH4 }, /* ESB */ + { PCI_VDEVICE(INTEL, 0x266e), DEVICE_INTEL_ICH4 }, /* ICH6 */ + { PCI_VDEVICE(INTEL, 0x27de), DEVICE_INTEL_ICH4 }, /* ICH7 */ + { PCI_VDEVICE(INTEL, 0x2698), DEVICE_INTEL_ICH4 }, /* ESB2 */ + { PCI_VDEVICE(INTEL, 0x7195), DEVICE_INTEL }, /* 440MX */ + { PCI_VDEVICE(SI, 0x7012), DEVICE_SIS }, /* SI7012 */ + { PCI_VDEVICE(NVIDIA, 0x01b1), DEVICE_NFORCE }, /* NFORCE */ + { PCI_VDEVICE(NVIDIA, 0x003a), DEVICE_NFORCE }, /* MCP04 */ + { PCI_VDEVICE(NVIDIA, 0x006a), DEVICE_NFORCE }, /* NFORCE2 */ + { PCI_VDEVICE(NVIDIA, 0x0059), DEVICE_NFORCE }, /* CK804 */ + { PCI_VDEVICE(NVIDIA, 0x008a), DEVICE_NFORCE }, /* CK8 */ + { PCI_VDEVICE(NVIDIA, 0x00da), DEVICE_NFORCE }, /* NFORCE3 */ + { PCI_VDEVICE(NVIDIA, 0x00ea), DEVICE_NFORCE }, /* CK8S */ + { PCI_VDEVICE(NVIDIA, 0x026b), DEVICE_NFORCE }, /* MCP51 */ + { PCI_VDEVICE(AMD, 0x746d), DEVICE_INTEL }, /* AMD8111 */ + { PCI_VDEVICE(AMD, 0x7445), DEVICE_INTEL }, /* AMD768 */ + { PCI_VDEVICE(AL, 0x5455), DEVICE_ALI }, /* Ali5455 */ { 0, } }; diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index 6ec0fc5..9e7d12e 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -220,24 +220,24 @@ struct intel8x0m { }; static struct pci_device_id snd_intel8x0m_ids[] = { - { 0x8086, 0x2416, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* 82801AA */ - { 0x8086, 0x2426, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* 82901AB */ - { 0x8086, 0x2446, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* 82801BA */ - { 0x8086, 0x2486, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* ICH3 */ - { 0x8086, 0x24c6, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* ICH4 */ - { 0x8086, 0x24d6, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* ICH5 */ - { 0x8086, 0x266d, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* ICH6 */ - { 0x8086, 0x27dd, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* ICH7 */ - { 0x8086, 0x7196, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* 440MX */ - { 0x1022, 0x7446, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* AMD768 */ - { 0x1039, 0x7013, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_SIS }, /* SI7013 */ - { 0x10de, 0x01c1, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_NFORCE }, /* NFORCE */ - { 0x10de, 0x0069, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_NFORCE }, /* NFORCE2 */ - { 0x10de, 0x0089, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_NFORCE }, /* NFORCE2s */ - { 0x10de, 0x00d9, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_NFORCE }, /* NFORCE3 */ + { PCI_VDEVICE(INTEL, 0x2416), DEVICE_INTEL }, /* 82801AA */ + { PCI_VDEVICE(INTEL, 0x2426), DEVICE_INTEL }, /* 82901AB */ + { PCI_VDEVICE(INTEL, 0x2446), DEVICE_INTEL }, /* 82801BA */ + { PCI_VDEVICE(INTEL, 0x2486), DEVICE_INTEL }, /* ICH3 */ + { PCI_VDEVICE(INTEL, 0x24c6), DEVICE_INTEL }, /* ICH4 */ + { PCI_VDEVICE(INTEL, 0x24d6), DEVICE_INTEL }, /* ICH5 */ + { PCI_VDEVICE(INTEL, 0x266d), DEVICE_INTEL }, /* ICH6 */ + { PCI_VDEVICE(INTEL, 0x27dd), DEVICE_INTEL }, /* ICH7 */ + { PCI_VDEVICE(INTEL, 0x7196), DEVICE_INTEL }, /* 440MX */ + { PCI_VDEVICE(AMD, 0x7446), DEVICE_INTEL }, /* AMD768 */ + { PCI_VDEVICE(SI, 0x7013), DEVICE_SIS }, /* SI7013 */ + { PCI_VDEVICE(NVIDIA, 0x01c1), DEVICE_NFORCE }, /* NFORCE */ + { PCI_VDEVICE(NVIDIA, 0x0069), DEVICE_NFORCE }, /* NFORCE2 */ + { PCI_VDEVICE(NVIDIA, 0x0089), DEVICE_NFORCE }, /* NFORCE2s */ + { PCI_VDEVICE(NVIDIA, 0x00d9), DEVICE_NFORCE }, /* NFORCE3 */ #if 0 - { 0x1022, 0x746d, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* AMD8111 */ - { 0x10b9, 0x5455, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_ALI }, /* Ali5455 */ + { PCI_VDEVICE(AMD, 0x746d), DEVICE_INTEL }, /* AMD8111 */ + { PCI_VDEVICE(AL, 0x5455), DEVICE_ALI }, /* Ali5455 */ #endif { 0, } }; diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index 82bc5b9..a83d196 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -61,7 +61,7 @@ MODULE_PARM_DESC(enable, "Enable Digigram " CARD_NAME " soundcard."); */ static struct pci_device_id snd_mixart_ids[] = { - { 0x1057, 0x0003, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* MC8240 */ + { PCI_VDEVICE(MOTOROLA, 0x0003), 0, }, /* MC8240 */ { 0, } }; diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c index 522a040..97a0731 100644 --- a/sound/pci/nm256/nm256.c +++ b/sound/pci/nm256/nm256.c @@ -263,9 +263,9 @@ struct nm256 { * PCI ids */ static struct pci_device_id snd_nm256_ids[] = { - {PCI_VENDOR_ID_NEOMAGIC, PCI_DEVICE_ID_NEOMAGIC_NM256AV_AUDIO, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0}, - {PCI_VENDOR_ID_NEOMAGIC, PCI_DEVICE_ID_NEOMAGIC_NM256ZX_AUDIO, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0}, - {PCI_VENDOR_ID_NEOMAGIC, PCI_DEVICE_ID_NEOMAGIC_NM256XL_PLUS_AUDIO, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0}, + {PCI_VDEVICE(NEOMAGIC, PCI_DEVICE_ID_NEOMAGIC_NM256AV_AUDIO), 0}, + {PCI_VDEVICE(NEOMAGIC, PCI_DEVICE_ID_NEOMAGIC_NM256ZX_AUDIO), 0}, + {PCI_VDEVICE(NEOMAGIC, PCI_DEVICE_ID_NEOMAGIC_NM256XL_PLUS_AUDIO), 0}, {0,}, }; diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c index d7b966e..f977dba 100644 --- a/sound/pci/rme32.c +++ b/sound/pci/rme32.c @@ -227,12 +227,9 @@ struct rme32 { }; static struct pci_device_id snd_rme32_ids[] = { - {PCI_VENDOR_ID_XILINX_RME, PCI_DEVICE_ID_RME_DIGI32, - PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0,}, - {PCI_VENDOR_ID_XILINX_RME, PCI_DEVICE_ID_RME_DIGI32_8, - PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0,}, - {PCI_VENDOR_ID_XILINX_RME, PCI_DEVICE_ID_RME_DIGI32_PRO, - PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0,}, + {PCI_VDEVICE(XILINX_RME, PCI_DEVICE_ID_RME_DIGI32), 0,}, + {PCI_VDEVICE(XILINX_RME, PCI_DEVICE_ID_RME_DIGI32_8), 0,}, + {PCI_VDEVICE(XILINX_RME, PCI_DEVICE_ID_RME_DIGI32_PRO), 0,}, {0,} }; diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index 55fb1c1..c01296c 100644 --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -232,14 +232,10 @@ struct rme96 { }; static struct pci_device_id snd_rme96_ids[] = { - { PCI_VENDOR_ID_XILINX, PCI_DEVICE_ID_RME_DIGI96, - PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, - { PCI_VENDOR_ID_XILINX, PCI_DEVICE_ID_RME_DIGI96_8, - PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, - { PCI_VENDOR_ID_XILINX, PCI_DEVICE_ID_RME_DIGI96_8_PRO, - PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, - { PCI_VENDOR_ID_XILINX, PCI_DEVICE_ID_RME_DIGI96_8_PAD_OR_PST, - PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, + { PCI_VDEVICE(XILINX, PCI_DEVICE_ID_RME_DIGI96), 0, }, + { PCI_VDEVICE(XILINX, PCI_DEVICE_ID_RME_DIGI96_8), 0, }, + { PCI_VDEVICE(XILINX, PCI_DEVICE_ID_RME_DIGI96_8_PRO), 0, }, + { PCI_VDEVICE(XILINX, PCI_DEVICE_ID_RME_DIGI96_8_PAD_OR_PST), 0, }, { 0, } }; diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c index 7dc60ad..1f6406c 100644 --- a/sound/pci/sonicvibes.c +++ b/sound/pci/sonicvibes.c @@ -243,7 +243,7 @@ struct sonicvibes { }; static struct pci_device_id snd_sonic_ids[] = { - { 0x5333, 0xca00, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, + { PCI_VDEVICE(S3, 0xca00), 0, }, { 0, } }; diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 949fcaf..acfa476 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -402,9 +402,9 @@ struct via82xx { static struct pci_device_id snd_via82xx_ids[] = { /* 0x1106, 0x3058 */ - { PCI_VENDOR_ID_VIA, PCI_DEVICE_ID_VIA_82C686_5, PCI_ANY_ID, PCI_ANY_ID, 0, 0, TYPE_CARD_VIA686, }, /* 686A */ + { PCI_VDEVICE(VIA, PCI_DEVICE_ID_VIA_82C686_5), TYPE_CARD_VIA686, }, /* 686A */ /* 0x1106, 0x3059 */ - { PCI_VENDOR_ID_VIA, PCI_DEVICE_ID_VIA_8233_5, PCI_ANY_ID, PCI_ANY_ID, 0, 0, TYPE_CARD_VIA8233, }, /* VT8233 */ + { PCI_VDEVICE(VIA, PCI_DEVICE_ID_VIA_8233_5), TYPE_CARD_VIA8233, }, /* VT8233 */ { 0, } }; diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c index 0d54e35..47eb615 100644 --- a/sound/pci/via82xx_modem.c +++ b/sound/pci/via82xx_modem.c @@ -261,7 +261,7 @@ struct via82xx_modem { }; static struct pci_device_id snd_via82xx_modem_ids[] = { - { 0x1106, 0x3068, PCI_ANY_ID, PCI_ANY_ID, 0, 0, TYPE_CARD_VIA82XX_MODEM, }, + { PCI_VDEVICE(VIA, 0x3068), TYPE_CARD_VIA82XX_MODEM, }, { 0, } }; diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c index 4af6666..822b60c 100644 --- a/sound/pci/ymfpci/ymfpci.c +++ b/sound/pci/ymfpci/ymfpci.c @@ -67,12 +67,12 @@ module_param_array(rear_switch, bool, NULL, 0444); MODULE_PARM_DESC(rear_switch, "Enable shared rear/line-in switch"); static struct pci_device_id snd_ymfpci_ids[] = { - { 0x1073, 0x0004, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* YMF724 */ - { 0x1073, 0x000d, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* YMF724F */ - { 0x1073, 0x000a, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* YMF740 */ - { 0x1073, 0x000c, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* YMF740C */ - { 0x1073, 0x0010, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* YMF744 */ - { 0x1073, 0x0012, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* YMF754 */ + { PCI_VDEVICE(YAMAHA, 0x0004), 0, }, /* YMF724 */ + { PCI_VDEVICE(YAMAHA, 0x000d), 0, }, /* YMF724F */ + { PCI_VDEVICE(YAMAHA, 0x000a), 0, }, /* YMF740 */ + { PCI_VDEVICE(YAMAHA, 0x000c), 0, }, /* YMF740C */ + { PCI_VDEVICE(YAMAHA, 0x0010), 0, }, /* YMF744 */ + { PCI_VDEVICE(YAMAHA, 0x0012), 0, }, /* YMF754 */ { 0, } }; -- 1.6.3.1.10.g659a0.dirty

2 5

[alsa-devel] [PATCH] ASoC: Zoom2: Update twl4030_setup_data parameters
by Jorge Eduardo Candelaria 25 Jun '09

25 Jun '09

Add EXT_MUTE support in Zoom2 machine driver. Member ext_mute_gpio is passed to TWL4030 as an external mute control for headset power on/off sequence. This is necessary to further reduce pop noise problem. Gpio 153 is connected to signal EXT_MUTE in Zoom2 board. In addition, change ramp delay value to 3 (218/161/109 ms). With previous ramp delay value, pop noise was louder, and with any longer value beep tone can be observed. Signed-off-by: Jorge Eduardo Candelaria <x0107209(a)ti.com> --- sound/soc/omap/zoom2.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/omap/zoom2.c b/sound/soc/omap/zoom2.c index 3de6d2b..8941f45 100644 --- a/sound/soc/omap/zoom2.c +++ b/sound/soc/omap/zoom2.c @@ -236,8 +236,9 @@ static struct snd_soc_card snd_soc_zoom2 = { /* twl4030 setup */ static struct twl4030_setup_data twl4030_setup = { - .ramp_delay_value = 2, /* 81 ms */ + .ramp_delay_value = 3, /* 161 ms */ .sysclk = 26000, + .ext_mute_gpio = 153, }; /* Audio subsystem */ -- 1.6.0.4

1 0

[alsa-devel] Should my Creative X-fi card be loading the snd-hda-intel module
by Susan Cragin 25 Jun '09

25 Jun '09

Shouldn't it be loading snd-hda-codec-ca0110 instead? I am using the 6/24 stable snapshot. lspci -vv gives me this. 05:00.0 Audio device: Creative Labs [SB X-Fi Xtreme Audio] CA0110-IBG Subsystem: Creative Labs Device 0018 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64 (500ns min, 5000ns max) Interrupt: pin A routed to IRQ 18 Region 0: Memory at fdffc000 (32-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel

2 1

[alsa-devel] RFC: PCM extra attributes
by Takashi Iwai 25 Jun '09

25 Jun '09

Hi, this is yet another topic I'm (currently) working on -- the addition of PCM ioctls to get/set some extra attributes. Basically, it adds two simple ioctls for getting/setting extra attributes to the PCM substream. The attribute has a sort of TLV form, /* PCM extra attributes */ struct snd_pcm_attr { unsigned int type; /* SNDRC_PCM_TYPE_ATTR_XXX */ unsigned int len; /* GET R: the max elements in value array * W: the actually written # elements * SET R/W: # elements to store */ unsigned int value[0]; /* value(s) to read / write */ }; And corresponding two ioctls #define SNDRV_PCM_IOCTL_GET_ATTR _IOWR('A', 0x14, struct snd_pcm_attr) #define SNDRV_PCM_IOCTL_SET_ATTR _IOWR('A', 0x15, struct snd_pcm_attr) The caller sets type and len fields. For getting attributes, the len points the max buffer length, and in return, the actual size is stored there. For setting, it's the size to set. Now, for getting the control ids associated to a PCM substream, it'll be like unsigned int buf[32]; buf[0] = SNDRV_PCM_ATTR_ASSOC_CTLS; buf[1] = ARRAY_SIZE(buf) - 2; /* minus header size */ ioctl(fd, SNDRV_PCM_IOCTL_GET_ATTR, buf); The kernel side implementation (a core part) is attached below. In this prototype, each driver is supposed to allocate and set substream->assoc_ctls and substream->num_assoc_ctls appropriately. I chose it since the control ids are statically associated to the PCM substreams at the driver loading time. It could be, of course, a liked list with kmalloc, but I thought it's overkill. Well, that's a pretty subtle thing and can be changed at any time. But, we need to define ABI properly from the beginning. So, if you have any comments on ABI or on the whole design, please speak up. An important note is that I'm planning to use this framework for getting/setting the PCM (surround) channel mapping. It's straightforward for this ABI. The internal implementation may vary, but the interface is the question now. For example, one could imagine that the channel mapping belongs to hw_params. But, it looks difficult because of the nature of channel mapping; it doesn't match with a simple integer parameter as hw_parmas has. The PCM channel mapping feature will be certainly useful for PA (and even for normal apps with a new alsa-lib plugin to convert the correct the channel mapping properly on demand). So, this is rather more interesting stuff to me, too... thanks, Takashi --- diff --git a/include/sound/asound.h b/include/sound/asound.h index 82aed3f..159826b 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -445,6 +445,21 @@ struct snd_xfern { snd_pcm_uframes_t frames; }; +/* PCM extra attributes */ +struct snd_pcm_attr { + unsigned int type; /* SNDRC_PCM_TYPE_ATTR_XXX */ + unsigned int len; /* GET R: the max elements in value array + * W: the actually written # elements + * SET R/W: # elements to store + */ + unsigned int value[0]; /* value(s) to read / write */ +}; + +/* PCM attribute types */ +enum { + SNDRV_PCM_ATTR_ASSOC_CTLS, /* array of associated ctl ids */ +}; + enum { SNDRV_PCM_TSTAMP_TYPE_GETTIMEOFDAY = 0, /* gettimeofday equivalent */ SNDRV_PCM_TSTAMP_TYPE_MONOTONIC, /* posix_clock_monotonic equivalent */ @@ -459,6 +474,8 @@ enum { #define SNDRV_PCM_IOCTL_HW_PARAMS _IOWR('A', 0x11, struct snd_pcm_hw_params) #define SNDRV_PCM_IOCTL_HW_FREE _IO('A', 0x12) #define SNDRV_PCM_IOCTL_SW_PARAMS _IOWR('A', 0x13, struct snd_pcm_sw_params) +#define SNDRV_PCM_IOCTL_GET_ATTR _IOWR('A', 0x14, struct snd_pcm_attr) +#define SNDRV_PCM_IOCTL_SET_ATTR _IOWR('A', 0x15, struct snd_pcm_attr) #define SNDRV_PCM_IOCTL_STATUS _IOR('A', 0x20, struct snd_pcm_status) #define SNDRV_PCM_IOCTL_DELAY _IOR('A', 0x21, snd_pcm_sframes_t) #define SNDRV_PCM_IOCTL_HWSYNC _IO('A', 0x22) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 2389352..98416b0 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -392,6 +392,10 @@ struct snd_pcm_substream { struct snd_info_entry *proc_prealloc_entry; struct snd_info_entry *proc_prealloc_max_entry; #endif + /* associated controls */ + unsigned int num_assoc_ctls; + unsigned int *assoc_ctls; + /* misc flags */ unsigned int hw_opened: 1; }; diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 145931a..7a46674 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -748,6 +748,7 @@ static void snd_pcm_free_stream(struct snd_pcm_str * pstr) substream_next = substream->next; snd_pcm_timer_done(substream); snd_pcm_substream_proc_done(substream); + kfree(substream->assoc_ctls); kfree(substream); substream = substream_next; } diff --git a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c index 08bfed5..07512b3 100644 --- a/sound/core/pcm_compat.c +++ b/sound/core/pcm_compat.c @@ -485,6 +485,8 @@ static long snd_pcm_ioctl_compat(struct file *file, unsigned int cmd, unsigned l case SNDRV_PCM_IOCTL_TTSTAMP: case SNDRV_PCM_IOCTL_HWSYNC: case SNDRV_PCM_IOCTL_PREPARE: + case SNDRV_PCM_IOCTL_GET_ATTR: + case SNDRV_PCM_IOCTL_SET_ATTR: case SNDRV_PCM_IOCTL_RESET: case SNDRV_PCM_IOCTL_START: case SNDRV_PCM_IOCTL_DROP: diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 84da3ba..46129c7 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2492,6 +2492,43 @@ static int snd_pcm_tstamp(struct snd_pcm_substream *substream, int __user *_arg) return 0; } +/* store back the associated ctl ids to the given PCM substream */ +static int snd_pcm_get_attr_assoc_ctls(struct snd_pcm_substream *substream, + unsigned int __user *arg) +{ + unsigned int len, i; + + arg++; + if (get_user(len, arg)) + return -EFAULT; + for (i = 0; i < len && i < substream->num_assoc_ctls; i++) + put_user(substream->assoc_ctls[i], arg + i + 1); + put_user(i, arg); + return 0; +} + +/* GET_ATTR ioctl */ +static int snd_pcm_get_attr(struct snd_pcm_substream *substream, + unsigned int __user *arg) +{ + unsigned int type; + + if (get_user(type, arg)) + return -EFAULT; + switch (type) { + case SNDRV_PCM_ATTR_ASSOC_CTLS: + return snd_pcm_get_attr_assoc_ctls(substream, arg); + } + return -ENOENT; +} + +/* SET_ATTR ioctl */ +static int snd_pcm_set_attr(struct snd_pcm_substream *substream, + struct snd_pcm_attr __user *arg) +{ + return -ENOENT; +} + static int snd_pcm_common_ioctl1(struct file *file, struct snd_pcm_substream *substream, unsigned int cmd, void __user *arg) @@ -2519,6 +2556,10 @@ static int snd_pcm_common_ioctl1(struct file *file, return snd_pcm_channel_info_user(substream, arg); case SNDRV_PCM_IOCTL_PREPARE: return snd_pcm_prepare(substream, file); + case SNDRV_PCM_IOCTL_GET_ATTR: + return snd_pcm_get_attr(substream, arg); + case SNDRV_PCM_IOCTL_SET_ATTR: + return snd_pcm_set_attr(substream, arg); case SNDRV_PCM_IOCTL_RESET: return snd_pcm_reset(substream); case SNDRV_PCM_IOCTL_START:

3 17

Re: [alsa-devel] Should my Creative X-fi card be loading the snd-hda-intel module
by Susan Cragin 24 Jun '09

24 Jun '09

Bill, My creative x-fi card used to work great with the daily build, and the module I mentioned. I've been away for it for a couple of months because I need sound for wine programs and there were issues with wine sound, and I just got back. I was surprised to see such a big step backward. When I was last using a daily build, both unstable and stable worked for me. Susan Susan, I also had a CA110-IBG chip on my Creative X-Fi. Never could get it to work, not under Ubuntu or Fedora, not with the Creative-supplied driver and also not with ctxfi. So we dropped that card and went to using an Asus PCI-e sound card instead. As for what module should load, here is what Takashi said on June 11: Are you sure that your board has really emu20k1 or emu20k2 chip? X-Fi isn't always with these chips. Some are HD-audio only (thus snd-hda-intel is the right driver) and some are with CA0106 chip. -------- Original Message -------- Shouldn't it be loading snd-hda-codec-ca0110 instead? I am using the 6/24 stable snapshot. lspci -vv gives me this. 05:00.0 Audio device: Creative Labs [SB X-Fi Xtreme Audio] CA0110-IBG Subsystem: Creative Labs Device 0018 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64 (500ns min, 5000ns max) Interrupt: pin A routed to IRQ 18 Region 0: Memory at fdffc000 (32-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel

1 0

[alsa-devel] Acer Aspire 7730G support
by Denis Kup 24 Jun '09

24 Jun '09

Hey, when i first installed alsa to my laptop, there no sound from subwoofer, jacks were map very strange (2 surrounds for ex.). I try all models from the list, but all don't support my subwoofer. After a weak of hard work i made a patch that enables ACER_ASPIRE_7730G model for my laptop. It have ALC888S, 3 jacks, 6 channels, stereo internal frontal speakers and internal subwoofer.You can see acer.com for more info about hardware and jacks. It's the changes i made to /alsa-kernel/pci/hda/patch_realtek.c: > enum { > ................... > ALC888_ACER_ASPIRE_7730G, > ................... > }; > ////////////////////////////////////////////////////////////// > static struct hda_verb alc888_acer_aspire_7730G_verbs[] = { > {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /*CHANGE DIRECT FRONT TO BLACK JACK*/ > {0x17, AC_VERB_SET_CONNECT_SEL, 0x02}, /* CHANGE: ENABLES SUBWOOFER ON 7730*/ > {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, //CHANGE FOR AUTOMUTE > { } /* end */ > }; > > /*CHANGE ENABLE AUTOMUTING ON ACER 7730G > ////////////////////////////////////////////////////// > //////////////////////////////////////////////////////*/ > > static void alc888_acer_aspire_7730G_front_automute(struct hda_codec *codec) > { > unsigned int present; > > present = snd_hda_codec_read(codec, 0x15, 0, > AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; > snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, > HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); > snd_hda_codec_amp_stereo(codec, 0x17, HDA_OUTPUT, 0, > HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); > } > > static void alc888_acer_aspire_7730G_unsol_event(struct hda_codec *codec, > unsigned int res) > { > if ((res >> 26) == ALC880_HP_EVENT) > alc888_acer_aspire_7730G_front_automute(codec); > } > > /*//////////////////////////////////////////////////// > ////////////////////////////////////////////////// > //////////////////////////////////////////////////////*/ > static struct alc_config_preset alc883_presets[] = { > > [ALC888_ACER_ASPIRE_7730G] = { > .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer }, > .init_verbs = { alc883_init_verbs, alc888_acer_aspire_7730G_verbs }, > .num_dacs = ARRAY_SIZE(alc883_dac_nids), > .dac_nids = alc883_dac_nids, > .dig_out_nid = ALC883_DIGOUT_NID, > .dig_in_nid = ALC883_DIGIN_NID, > .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes), > .channel_mode = alc883_3ST_6ch_modes, > .need_dac_fix = 1, > .input_mux = &alc883_capture_source, > .unsol_event = alc888_acer_aspire_7730G_unsol_event, /* CHANGE AUTOMUTE DECLARATION*/ > .init_hook = alc888_acer_aspire_7730G_front_automute, > }, > > /////////////////////////////////////////////////////////////// > > static const char *alc883_models[ALC883_MODEL_LAST] = { > ................................. > [ALC888_ACER_ASPIRE_7730G] = "acer-aspire-7730g", > ......................... > > //////////////////////////////// > static struct snd_pci_quirk alc883_cfg_tbl[] = { > ...................................... > SND_PCI_QUIRK(0x1025, 0x0142, "Acer Aspire 7730G", ALC888_ACER_ASPIRE_7730G), > .......................................... After changing file i install driver, and it automatically detect its model (maybe there are old settings from previous install, so i dont 100% sure that it was detected). With this settings subwoofer works perfect, internal stereo and subwoofer mute when headphone are plugged all works, except 1 thing: subwoofer uses center channel instead LFE. To fix it i use such asound.conf file: > pcm.upmix_20to51 { > type plug > slave.pcm lowpass_21to21 > slave.channels 3 > ttable { > 0.0 1 # left channel > 1.1 1 # right channel > 0.2 1 # mix left and right ... > 1.2 1 # ... channel for subwoofer > } > } > pcm.lowpass_21to21 { > type ladspa > slave.pcm upmix_21to51 > path "/usr/lib/ladspa" > channels 3 > plugins { > 0 { > id 1098 # Identity (Audio) (1098/identity_audio) > policy duplicate > input.bindings.0 "Input"; > output.bindings.0 "Output"; > } > 1 { > id 1671 # 4 Pole Low-Pass Filter with Resonance (FCRCIA) (1672/lp4pole_fcrcia_oa) > policy none > input.bindings.2 "Input"; > output.bindings.2 "Output"; > input { > controls [ 1000 2 ] > } > } > } > } > pcm.upmix_21to51 { > type plug > slave.pcm surround51 > slave.channels 6 > ttable { > 0.0 1 # front left > 1.1 1 # front right > 0.2 1 # rear left > 1.3 1 # rear right > 0.5 0.5 # swap channels > 1.5 0.5 > 2.4 1 > } > } > > pcm.swapCLFE_51 { > type plug > slave.pcm surround51 > slave.channels 6 > ttable { > 0.0 1 # front left > 1.1 1 # front right > 2.2 1 # rear left > 3.3 1 # rear right > 4.5 1 # swap channels > 5.4 1 > } > } I use upmix_21to51 and swapCLFE_51 devices in Amarok settings. Is there any chance to swap Center and LFE in patch_realtek.c? And can my 7730g config be used in future version of alsa? And how can I detect params for snd_pci_quirk alc883_cfg_tbl ? And some config settings for my laptop, if needed: > linux-wpg9:/home/dener # cat /proc/asound/card0/codec* > Codec: Realtek ALC888 > Address: 0 > Function Id: 0x1 > Vendor Id: 0x10ec0888 > Subsystem Id: 0x10250142 > Revision Id: 0x100202 > No Modem Function Group found > Default PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Default Amp-In caps: N/A > Default Amp-Out caps: N/A > GPIO: io=2, o=0, i=0, unsolicited=1, wake=1 > IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > Node 0x02 [Audio Output] wcaps 0x411: Stereo > Converter: stream=5, channel=0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Node 0x03 [Audio Output] wcaps 0x411: Stereo > Converter: stream=5, channel=0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Node 0x04 [Audio Output] wcaps 0x411: Stereo > Converter: stream=5, channel=0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Node 0x05 [Audio Output] wcaps 0x411: Stereo > Converter: stream=0, channel=0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Node 0x06 [Audio Output] wcaps 0x611: Stereo Digital > Converter: stream=0, channel=0 > Digital: Enabled GenLevel > Digital category: 0x2 > PCM: > rates [0x5e0]: 44100 48000 88200 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x08 [Audio Input] wcaps 0x10051b: Stereo Amp-In > Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 > Amp-In vals: [0x16 0x16] > Converter: stream=0, channel=0 > SDI-Select: 0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Connection: 1 > 0x23 > Node 0x09 [Audio Input] wcaps 0x10051b: Stereo Amp-In > Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 > Amp-In vals: [0x9f 0x9f] > Converter: stream=0, channel=0 > SDI-Select: 0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Connection: 1 > 0x22 > Node 0x0a [Audio Input] wcaps 0x100711: Stereo Digital > Converter: stream=0, channel=0 > SDI-Select: 0 > Digital: > Digital category: 0x0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Connection: 1 > 0x1f > Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 > Amp-In vals: [0x80 0x80] [0x1f 0x1f] [0x00 0x00] [0x80 0x80] [0x17 0x17] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] > Connection: 10 > 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 > Node 0x0c [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x00 0x00] [0x00 0x00] > Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 > Amp-Out vals: [0x12 0x12] > Connection: 2 > 0x02 0x0b > Node 0x0d [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x00 0x00] [0x00 0x00] > Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 > Amp-Out vals: [0x15 0x15] > Connection: 2 > 0x03 0x0b > Node 0x0e [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x00 0x00] [0x00 0x00] > Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 > Amp-Out vals: [0x15 0x15] > Connection: 2 > 0x04 0x0b > Node 0x0f [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x80 0x80] [0x80 0x80] > Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 > Amp-Out vals: [0x00 0x00] > Connection: 2 > 0x05 0x0b > Node 0x10 [Audio Output] wcaps 0x611: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > PCM: > rates [0x5e0]: 44100 48000 88200 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Node 0x11 [Pin Complex] wcaps 0x400780: Mono Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 1 > 0x10 > Node 0x12 [Pin Complex] wcaps 0x400401: Stereo > Pincap 0x00000020: IN > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Power: setting=D0, actual=D0 > Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x14 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x0001003e: IN OUT HP EAPD Detect Trigger > EAPD 0x0: > Pin Default 0x99130110: [Fixed] Speaker at Int ATAPI > Conn = ATAPI, Color = Unknown > DefAssociation = 0x1, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c* 0x0d 0x0e 0x0f 0x26 > Node 0x15 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x0001003e: IN OUT HP EAPD Detect Trigger > EAPD 0x0: > Pin Default 0x0221101f: [Jack] HP Out at Ext Front > Conn = 1/8, Color = Black > DefAssociation = 0x1, Sequence = 0xf > Pin-ctls: 0x40: OUT > Unsolicited: tag=04, enabled=1 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c* 0x0d 0x0e 0x0f 0x26 > Node 0x16 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x00000036: IN OUT Detect Trigger > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c 0x0d 0x0e* 0x0f 0x26 > Node 0x17 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x00000036: IN OUT Detect Trigger > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c 0x0d 0x0e* 0x0f 0x26 > Node 0x18 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x0000373e: IN OUT HP Detect Trigger > Vref caps: HIZ 50 GRD 80 100 > Pin Default 0x02a19c30: [Jack] Mic at Ext Front > Conn = 1/8, Color = Pink > DefAssociation = 0x3, Sequence = 0x0 > Pin-ctls: 0x40: OUT VREF_HIZ > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c 0x0d 0x0e* 0x0f 0x26 > Node 0x19 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x80 0x80] > Pincap 0x0000373e: IN OUT HP Detect Trigger > Vref caps: HIZ 50 GRD 80 100 > Pin Default 0x99a30931: [Fixed] Mic at Int ATAPI > Conn = ATAPI, Color = Unknown > DefAssociation = 0x3, Sequence = 0x1 > Misc = NO_PRESENCE > Pin-ctls: 0x24: IN VREF_80 > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c* 0x0d 0x0e 0x0f 0x26 > Node 0x1a [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x0000373e: IN OUT HP Detect Trigger > Vref caps: HIZ 50 GRD 80 100 > Pin Default 0x0281343f: [Jack] Line In at Ext Front > Conn = 1/8, Color = Blue > DefAssociation = 0x3, Sequence = 0xf > Pin-ctls: 0x40: OUT VREF_HIZ > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c 0x0d* 0x0e 0x0f 0x26 > Node 0x1b [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x0000373e: IN OUT HP Detect Trigger > Vref caps: HIZ 50 GRD 80 100 > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0xc0: OUT HP VREF_HIZ > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c* 0x0d 0x0e 0x0f 0x26 > Node 0x1c [Pin Complex] wcaps 0x400481: Stereo > Pincap 0x00000024: IN Detect > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x20: IN > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Node 0x1d [Pin Complex] wcaps 0x400400: Mono > Pincap 0x00000020: IN > Pin Default 0x4016852d: [N/A] Speaker at Ext N/A > Conn = Digital, Color = Purple > DefAssociation = 0x2, Sequence = 0xd > Misc = NO_PRESENCE > Pin-ctls: 0x20: IN > Power: setting=D0, actual=D0 > Node 0x1e [Pin Complex] wcaps 0x400780: Mono Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x02451120: [Jack] SPDIF Out at Ext Front > Conn = Optical, Color = Black > DefAssociation = 0x2, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 1 > 0x06 > Node 0x1f [Pin Complex] wcaps 0x400680: Mono Digital > Pincap 0x00000024: IN Detect > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x20: IN > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono > Processing caps: benign=0, ncoeff=25 > Processing Coefficient: 0x00 > Coefficient Index: 0x09 > Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x80 0x80] [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] > Connection: 12 > 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b 0x12 > Node 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] > Connection: 11 > 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b > Node 0x24 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x25 [Audio Output] wcaps 0x411: Stereo > Converter: stream=0, channel=0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Node 0x26 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x00 0x00] [0x80 0x80] > Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 > Amp-Out vals: [0x00 0x00] > Connection: 2 > 0x25 0x0b > Codec: LSI ID 1040 > Address: 1 > Function Id: 0x2 > Vendor Id: 0x11c11040 > Subsystem Id: 0x1025013d > Revision Id: 0x100200 > Modem Function Group: 0x1 > Codec: Nvidia MCP78 HDMI > Address: 3 > Function Id: 0x1 > Vendor Id: 0x10de0006 > Subsystem Id: 0x10de0101 > Revision Id: 0x100000 > No Modem Function Group found > Default PCM: > rates [0x0]: > bits [0x0]: > formats [0x0]: > Default Amp-In caps: N/A > Default Amp-Out caps: N/A > GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 > Node 0x04 [Audio Output] wcaps 0x211: Stereo Digital > Converter: stream=0, channel=0 > Digital: Enabled > Digital category: 0x0 > PCM: > rates [0xc0]: 48000 88200 > bits [0xf]: 8 16 20 24 > formats [0x1]: PCM > Node 0x05 [Pin Complex] wcaps 0x400381: Stereo Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x18560110: [Jack] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0x1, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Connection: 1 > 0x04 > Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > PCM: > rates [0xc0]: 48000 88200 > bits [0xf]: 8 16 20 24 > formats [0x1]: PCM > Node 0x07 [Pin Complex] wcaps 0x400381: Stereo Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x58560121: [N/A] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0x2, Sequence = 0x1 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Unsolicited: tag=00, enabled=0 > Connection: 1 > 0x06 > Node 0x08 [Audio Output] wcaps 0x211: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > PCM: > rates [0xc0]: 48000 88200 > bits [0xf]: 8 16 20 24 > formats [0x1]: PCM > Node 0x09 [Pin Complex] wcaps 0x400381: Stereo Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x58560122: [N/A] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0x2, Sequence = 0x2 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Unsolicited: tag=00, enabled=0 > Connection: 1 > 0x08 > Node 0x0a [Audio Output] wcaps 0x211: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > PCM: > rates [0xc0]: 48000 88200 > bits [0xf]: 8 16 20 24 > formats [0x1]: PCM > Node 0x0b [Pin Complex] wcaps 0x400381: Stereo Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x58560123: [N/A] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0x2, Sequence = 0x3 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Unsolicited: tag=00, enabled=0 > Connection: 1 > 0x0a > Node 0x0c [Audio Output] wcaps 0x211: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > PCM: > rates [0xc0]: 48000 88200 > bits [0xf]: 8 16 20 24 > formats [0x1]: PCM > Node 0x0d [Pin Complex] wcaps 0x400381: Stereo Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x58560124: [N/A] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0x2, Sequence = 0x4 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Unsolicited: tag=00, enabled=0 > Connection: 1 > 0x0c > > linux-wpg9:/home/dener # cat /proc/asound/version > Advanced Linux Sound Architecture Driver Version 1.0.20. > Compiled on Jun 14 2009 for kernel 2.6.27.23-0.1-pae (SMP). > > cat /proc/asound/devices > 0: [ 0] : control > 1: : sequencer > 4: [ 0- 0]: hardware dependent > 5: [ 0- 1]: hardware dependent > 7: [ 0- 3]: hardware dependent > 16: [ 0- 0]: digital audio playback > 17: [ 0- 1]: digital audio playback > 19: [ 0- 3]: digital audio playback > 24: [ 0- 0]: digital audio capture > 25: [ 0- 1]: digital audio capture > 26: [ 0- 2]: digital audio capture > 33: : timer > > cat /proc/asound/pcm > 00-00: ALC888 Analog : ALC888 Analog : playback 1 : capture 1 > 00-01: ALC888 Digital : ALC888 Digital : playback 1 : capture 1 > 00-02: ALC888 Analog : ALC888 Analog : capture 1 > 00-03: NVIDIA HDMI : NVIDIA HDMI : playback 1 > > lspci | grep -i audio > 00:1b.0 Audio device: Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 03) > I have 1.0.20 alsa version and OpenSUSE 11.1 distributive. I hope my materials are usefull PS I am from Russia, sorry for my bad English :)

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[alsa-devel] Need help on mmap
by Christopher Tsui 24 Jun '09

24 Jun '09

Hi I am in the process of add mmap support in my ALSA driver. So this is what i have done so far In snd_pcm_hardware structure .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID In snd_pcm_ops structure .mmap = my_pcm_mmap So to test my code, I use aplay -M -Dhw:0 rock.wav and this is what I get ALSA lib pcm_mmap.c:378:(snd_pcm_mmap) mmap failed: Invalid argument aplay: set_params:1022: Unable to install hw params: ACCESS: MMAP_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 32 CHANNELS: 2 RATE: 44100 PERIOD_TIME: (27210 27211) PERIOD_SIZE: 1200 PERIOD_BYTES: 4800 PERIODS: 4 BUFFER_TIME: (108843 108844) BUFFER_SIZE: 4800 BUFFER_BYTES: 19200 TICK_TIME: 10000 What am I doing wrong?? Thanks in advance

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[alsa-devel] Getting board to work on 2.6.30
by Tony Hoyle 24 Jun '09

24 Jun '09

Hi, I'm trying to get sound working on a board I'm working with. The sound works correctly on 2.6.24.7 but not 2.6.30. I has an intel HDA (Poulson) chipset and a sigmatel 9202 codec. Physically there's only one output - the speakers, and no input, so from a testing point of view it either works or it doesn't. Kernel output from both looks similar: 2.6.24.7: [ 19.942029] ALSA sound/pci/hda/hda_intel.c:916: HDA snoop disabled, try to enable ... OK [ 20.157458] ALSA sound/pci/hda/hda_codec.c:2760: autoconfig: line_outs=1 (0xd/0x0/0x0/0x0/0x0) [ 20.157467] ALSA sound/pci/hda/hda_codec.c:2764: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 20.157474] ALSA sound/pci/hda/hda_codec.c:2768: hp_outs=1 (0xa/0x0/0x0/0x0/0x0) [ 20.157481] ALSA sound/pci/hda/hda_codec.c:2776: inputs: mic=0x0, fmic=0xb, line=0xc, fline=0x0, cd=0x11, aux=0x0 2.6.30 (running alsa snapshot from last night): 19.379811] ALSA sound/pci/hda/hda_codec.c:3795: autoconfig: line_outs=1 (0x10/0x0/0x0/0x0/0x0) [ 19.379824] ALSA sound/pci/hda/hda_codec.c:3799: speaker_outs=2 (0xa/0x7/0x0/0x0/0x0) [ 19.379833] ALSA sound/pci/hda/hda_codec.c:3803: hp_outs=1 (0xd/0x0/0x0/0x0/0x0) [ 19.379842] ALSA sound/pci/hda/hda_codec.c:3804: mono: mono_out=0x10 [ 19.379849] ALSA sound/pci/hda/hda_codec.c:3807: dig-out=0x8/0x0 [ 19.379856] ALSA sound/pci/hda/hda_codec.c:3815: inputs: mic=0xb, fmic=0x0, line=0xc, fline=0x0, cd=0x11, aux=0x0 [ 19.390010] input: HDA Intel MID CD at Ext Rear Jack as /class/input/input3 [ 19.401109] input: HDA Intel MID Line In at Ext Rear Jack as /class/input/input4 [ 19.414726] input: HDA Intel MID Mic at Ext Rear Jack as /class/input/input5 [ 19.430427] input: HDA Intel MID Line Out at Ext Rear Jack as /class/input/input6 [ 19.447561] input: HDA Intel MID HP Out at Ext Front Jack as /class/input/input7 It seems to be creating the outputs as inputs? This seems wrong. I've tried every model setting I could find and reckon I need to create a new one.. but don't actually know what to create to get things working - there are lots of differences in the alsa between the two kernels and it isn't something that looks easy to pin down short of going nuts with git bisect. Can someone give me a pointer where to start looking at this? Tony Codec: SigmaTel STAC9202 Address: 0 Function Id: 0x1 Vendor Id: 0x83847632 Subsystem Id: 0x00000100 Revision Id: 0x100101 No Modem Function Group found Default PCM: rates [0x7e0]: 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 GPIO: io=4, o=0, i=0, unsolicited=1, wake=1 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0xd0401: Stereo Converter: stream=0, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x03 [Audio Input] wcaps 0x1d0541: Stereo Converter: stream=0, channel=0 SDI-Select: 0 Power: setting=D0, actual=D0 Delay: 13 samples Connection: 1 0x14 Processing caps: benign=0, ncoeff=0 Node 0x04 [Audio Input] wcaps 0x140311: Stereo Digital Converter: stream=0, channel=0 SDI-Select: 0 Digital: Digital category: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Delay: 4 samples Connection: 1 0x07 Node 0x05 [Audio Output] wcaps 0x40211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x1e0]: 44100 48000 88200 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Delay: 4 samples Node 0x06 [Audio Selector] wcaps 0x300901: Stereo R/L Connection: 3 0x02* 0x07 0x14 Node 0x07 [Pin Complex] wcaps 0x430681: Stereo Digital Pincap 0x00010024: IN EAPD Detect EAPD 0x0: Pin Default 0x01110014: [Jack] Speaker at Ext Rear Conn = 1/8, Color = Unknown DefAssociation = 0x1, Sequence = 0x4 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Delay: 3 samples Node 0x08 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x01410013: [Jack] SPDIF Out at Ext Rear Conn = 1/8, Color = Unknown DefAssociation = 0x1, Sequence = 0x3 Pin-ctls: 0x40: OUT Connection: 2 0x05* 0x14 Node 0x09 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=0 Amp-Out vals: [0x07 0x07] Connection: 1 0x0f Node 0x0a [Pin Complex] wcaps 0x400181: Stereo Pincap 0x0000003f: IN OUT HP Detect Trigger ImpSense Pin Default 0x02111011: [Jack] Speaker at Ext Front Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0x1 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x0e Node 0x0b [Pin Complex] wcaps 0x400181: Stereo Pincap 0x00001737: IN OUT Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x01a10021: [Jack] Mic at Ext Rear Conn = 1/8, Color = Unknown DefAssociation = 0x2, Sequence = 0x1 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=02, enabled=1 Connection: 1 0x0e Node 0x0c [Pin Complex] wcaps 0x400181: Stereo Pincap 0x00001737: IN OUT Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x01810022: [Jack] Line In at Ext Rear Conn = 1/8, Color = Unknown DefAssociation = 0x2, Sequence = 0x2 Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=03, enabled=1 Connection: 1 0x0e Node 0x0d [Pin Complex] wcaps 0x400181: Stereo Pincap 0x0000003f: IN OUT HP Detect Trigger ImpSense Pin Default 0x02214012: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x2 Pin-ctls: 0x00: Unsolicited: tag=01, enabled=1 Connection: 1 0x0e Node 0x0e [Audio Selector] wcaps 0x300105: Stereo Amp-Out Amp-Out caps: N/A Amp-Out vals: [0x17 0x17] Connection: 1 0x06 Node 0x0f [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-Out vals: [0x00 0x00] Connection: 5 0x0b* 0x0c 0x0d 0x0a 0x11 Node 0x10 [Pin Complex] wcaps 0x400104: Mono Amp-Out Amp-Out caps: N/A Amp-Out vals: [0x00] Pincap 0x00000010: OUT Pin Default 0x01010031: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Unknown DefAssociation = 0x3, Sequence = 0x1 Pin-ctls: 0x40: OUT Connection: 1 0x12 Node 0x11 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x01310023: [Jack] CD at Ext Rear Conn = 1/8, Color = Unknown DefAssociation = 0x2, Sequence = 0x3 Pin-ctls: 0x20: IN Node 0x12 [Audio Mixer] wcaps 0x200100: Mono Connection: 1 0x06 Node 0x13 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Amp-Out caps: ofs=0x03, nsteps=0x03, stepsize=0x17, mute=1 Amp-Out vals: [0x00] Node 0x14 [Audio Selector] wcaps 0x30090d: Stereo Amp-Out R/L Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Connection: 2 0x09* 0x15 Node 0x15 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x01d10024: [Jack] Digitial In at Ext Rear Conn = 1/8, Color = Unknown DefAssociation = 0x2, Sequence = 0x4 Pin-ctls: 0x20: IN

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[alsa-devel] [PATCH 1/3] ASoC: TWL4030: Enable voice filters for voice sidetone
by Lopez Cruz, Misael 24 Jun '09

24 Jun '09

Digital voice loopback (sidetone) requires voice filters to be enabled: VTXL, VTXR, VRX. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl4030.c | 67 +++++++++++++++++++++++++------------------- 1 files changed, 38 insertions(+), 29 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 4dbb853..780217a 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -332,6 +332,28 @@ static void twl4030_power_down(struct snd_soc_codec *codec) twl4030_codec_enable(codec, 0); } +/* In case of voice mode, the RX1 L(VRX) for downlink and the TX2 L/R + * (VTXL, VTXR) for uplink has to be enabled/disabled. */ +static void twl4030_voice_enable(struct snd_soc_codec *codec, int direction, + int enable) +{ + u8 reg, mask; + + reg = twl4030_read_reg_cache(codec, TWL4030_REG_OPTION); + + if (direction == SNDRV_PCM_STREAM_PLAYBACK) + mask = TWL4030_ARXL1_VRX_EN; + else + mask = TWL4030_ATXL2_VTXL_EN | TWL4030_ATXR2_VTXR_EN; + + if (enable) + reg |= mask; + else + reg &= ~mask; + + twl4030_write(codec, TWL4030_REG_OPTION, reg); +} + /* Earpiece */ static const struct snd_kcontrol_new twl4030_dapm_earpiece_controls[] = { SOC_DAPM_SINGLE("Voice", TWL4030_REG_EAR_CTL, 0, 1, 0), @@ -712,7 +734,22 @@ static int bypass_event(struct snd_soc_dapm_widget *w, reg = twl4030_read_reg_cache(w->codec, m->reg); - if (m->reg <= TWL4030_REG_ARXR2_APGA_CTL) { + if (m->reg == TWL4030_REG_VSTPGA) { + /* Voice digital bypass */ + if (reg) { + twl4030->bypass_state |= (1 << 5); + twl4030_voice_enable(w->codec, + SNDRV_PCM_STREAM_PLAYBACK, 1); + twl4030_voice_enable(w->codec, + SNDRV_PCM_STREAM_CAPTURE, 1); + } else { + twl4030->bypass_state &= ~(1 << 5); + twl4030_voice_enable(w->codec, + SNDRV_PCM_STREAM_PLAYBACK, 0); + twl4030_voice_enable(w->codec, + SNDRV_PCM_STREAM_CAPTURE, 0); + } + } else if (m->reg <= TWL4030_REG_ARXR2_APGA_CTL) { /* Analog bypass */ if (reg & (1 << m->shift)) twl4030->bypass_state |= @@ -726,12 +763,6 @@ static int bypass_event(struct snd_soc_dapm_widget *w, twl4030->bypass_state |= (1 << 4); else twl4030->bypass_state &= ~(1 << 4); - } else if (m->reg == TWL4030_REG_VSTPGA) { - /* Voice digital bypass */ - if (reg) - twl4030->bypass_state |= (1 << 5); - else - twl4030->bypass_state &= ~(1 << 5); } else { /* Digital bypass */ if (reg & (0x7 << m->shift)) @@ -1806,28 +1837,6 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } -/* In case of voice mode, the RX1 L(VRX) for downlink and the TX2 L/R - * (VTXL, VTXR) for uplink has to be enabled/disabled. */ -static void twl4030_voice_enable(struct snd_soc_codec *codec, int direction, - int enable) -{ - u8 reg, mask; - - reg = twl4030_read_reg_cache(codec, TWL4030_REG_OPTION); - - if (direction == SNDRV_PCM_STREAM_PLAYBACK) - mask = TWL4030_ARXL1_VRX_EN; - else - mask = TWL4030_ATXL2_VTXL_EN | TWL4030_ATXR2_VTXR_EN; - - if (enable) - reg |= mask; - else - reg &= ~mask; - - twl4030_write(codec, TWL4030_REG_OPTION, reg); -} - static int twl4030_voice_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { -- 1.5.4.3

3 10

[alsa-devel] USB microphone not working. No sound, although device visible.
by Shapeshifter 23 Jun '09

23 Jun '09

Hi there. Following problem, details below: PROBLEM: I can't record anything using the microphone which is integrated into my USB webcam. The cam shows up as a device, and I can adjust the only available mixer which is for <Mic> in [Capture]. It's not muted and it's maxed out. DETAILS and PASTES: Alsa-info output: http://www.alsa-project.org/db/?f=f12fd357bc269d27ebdb4b5d562227bbe22a1086 You might notice the device comes up as [default]. If I run... $ arecord -Dplughw:1,0 -f cd -vv /dev/null or $ arecord -Ddefault -f cd -vv /dev/null then it out puts a lot of stuff (both the same) which I pasted in the below pastebin, but at the bottom it doesn't show any #'s and %'s to indicate the volume. The blue light on the webcam also turns on. If I save to a file instead of /dev/null and aplay it, it's just silence. Here's the output from the above command: http://pastie.org/521697 Although if I run $ arecord -Dhw:1,0 -f cd -vv /dev/null then it fails with: -> Recording WAVE '/dev/null' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo -> arecord: set_params:984: Channels count non available" I made a screenshot illustrating that everything is turned up and unmuted: http://omploader.org/vMXZldg ALREADY TRIED: - Switching the two devices so that 0 becomes the webcam and 1 the intel-hda through modprobe.conf but that had no other effect other than the switching. - Playing a lot with the device, muting, triplechecking volumes etc. etc. - Trying recording something in audacity. First time, the light goes on but it doesn't record, on subsequent tries, audacity spits an error stating "error opening sound device". - Trying it in linphone, light turns on, but other party can't hear anything. Any ideas how to fix hits? Btw, it used to work in ubuntu 9.04 for some reason. Tried it on two Archlinux boxes, it's exactly the same on both. But everything works really, everything is very up to date (I'm sure ubuntu has older versions of stuff). Well, all the info is in the alsa-info link above. Hope I can work this out! Thanks for any input! Regards.

1 0

[alsa-devel] ASoC CODEC power management on reboot
by Ben Dooks 23 Jun '09

23 Jun '09

I've run into a problem where when the system reboots the audio starts playing random data. I suspect that this is due to the CODEC being left powered and when the during/after the reset process the GPIO lines which form the I2S bus are left floating. Should the ASoC (possibly DAPM) layer sort out powering down the CODEC before reboot, or should it be down to the codec or the mapping driver to do this? -- Ben Q: What's a light-year? A: One-third less calories than a regular year.

2 3

[alsa-devel] Please help in adding ams-delta support to ASoC
by Janusz Krzysztofik 23 Jun '09

23 Jun '09

Hi, I am trying to add sound support for ams-delta omap machine, yet without much success. Three years ago, Mark Underwood created an omap-alsa compatible driver that basically worked[1]. It was derieved from similiar driver for aic23 codec found on omap osk machine. It looks like Mark has never managed to finish his work. It's not clear for me if he has found that getting a working fullduplex sound is technically impossible on a voice modem codec, shared among modem and cpu, controlable only from modem side afaik, but I have decided to give it a try. Since Mark's initial work, omap-alsa framework has been depreciated in favour of soc-omap. API changes are so significant that Mark's code is rather not useable directly any more. However, I am trying to use it as a starting point, by comparing it against it's prototype osk/aic23 code. Following Mark, I am trying to derieve the new ams-delta sound driver from current asoc driver for omap osk9512. For codec part, I decided to base my work on much more simple ad73311 rather that tlv320aic23. Comparing Mark's code agaist it's osk/aic23 prototype, I can see the folowing significant changes: 1. rate tables/bitmaps found in hw_constraint_rates, snd_omap_alsa_playback and snd_omap_alsa_capture structures limited to 8kHz, 2. hardware related code found in codec_configure_dev(), codec_clock_on() and codec_clock_off() callback functions replaced with ams-delta hardware specific code that switches the codec DAI pins from modem chip to mcbsp cpu interface and back, 3. codec_set_samplerate() and all mixer related functions replaced with stubs. 4. the following McBSP register settings changes: - .xcr2 = XPHASE | XFRLEN2(OMAP_MCBSP_WORD_8) | - XWDLEN2(OMAP_MCBSP_WORD_16) | XDATDLY(0) | XFIG, + .xcr2 = XPHASE | XWDLEN2(OMAP_MCBSP_WORD_16) | XFRLEN2(0), - .srgr1 = FWID(DEFAULT_BITPERSAMPLE - 1), + .srgr1 = CLKGDV(0), - .srgr2 = GSYNC | CLKSP | FSGM | FPER(DEFAULT_BITPERSAMPLE * 2 - 1), + .srgr2 = GSYNC, - .pcr0 = CLKXP | CLKRP, /* mcbsp: slave */ I have found points 1. to 3. rather trivial to implement in new framework. Regarding point 4., it looks like detailed register settings are now done inside omap-mcbsp.c, based on machine specified format, so I have to find out what format should be specified for ams-delta. Please correct me if I am missing something. There was one more modification in Mark's code, addressing dma chaining problem on omap15xx hardware, but as far as I could see, the problem was already solved in the asoc omap framework. Initially, I based my work on ompenembedded provided linux-omap.git revision 90e758af52ba803cba233fabee81176d99589f09. The results were rather poor - total system hangup after first device access, with no single message. So I have switched to linux-2.6.30-rc5 and now I can safely access the device, however it does not work as expected. aplay and arecord wait forever, cat to/from /dev/dsp breaks with hardware error messgae. DMA interrput counters stay at 0. However, codec switching that I do from machine->ops->startup/shutdown seems working, as modem stops producing any sounds while the alsa device is in use and gets back thereafter. First of all, I'd like to make sure if my problem is related to my code only. As I am new in these areas, I would like to ask you if the omap asoc framework is stable enough to relay on. If yes, could you please look at my dirty code (attached) an give me some hints? I can provide you with more information if necessary. Regards, Janusz [1] http://www.earth.li/pipermail/e3-hacking/2006-April/000481.html

6 52

[alsa-devel] Indirect Callback with ack is called with no data
by Singaravelan Nallasellan 23 Jun '09

23 Jun '09

I am using indirect callback mechanism inside ack. I see that ack does not call my indirect callback pointer. Going through the code, I suspect that the appl_ptr is not being incremented and there is no data for the indirect callback to be called. After few acks, I get the indirect callback called. With aplay, I am able to play the wav file. But with flash, there is not data being sent to the driver. Please help me with any clues. Thanks in advance. Sing _________________________________________________________________ Insert movie times and more without leaving Hotmail®. http://windowslive.com/Tutorial/Hotmail/QuickAdd?ocid=TXT_TAGLM_WL_HM_Tutor…

3 10

Re: [alsa-devel] alc889
by Takashi Iwai 23 Jun '09

23 Jun '09

At Tue, 23 Jun 2009 00:05:42 +0200, Stefan Krastanov wrote: > > Hello, > > Thank you for your fast response, I appreciate it, knowing you don't have all > that free time. > > As I have said my hardware configuration is: laptop with alc889 soundcard > - two front > speakers > - one > sub-woofer > - one > internal microphone > - jacks > with jacksensing (tested on windows) - mic, line-in, headphones > - digital > IO > - > potentiometer for volume control > > I installed the latest snapshot using ubuntu's automated script (I manually > verified that it's the latest) and used model=auto. The results were the same > when I tried it few months ago: > - "Front" controls the front speakers OK > - "Front" also controls headphones problem > - "Headphones" slider does not control anything problem > - No control slider for the build-in sub-woofer problem > - microphones working (external and internal) OK > - potentiometer for volume control not working problem > - Line-in not tested, digital IO not tested not tested > - jack sensing on the headphones is not working problem > - jack sensing on microphone and line in not tested not tested > > Attached to the message you can find the output from alsa-info. Hm, how about model=fujitsu-pi2515 or model=fujitsu-xa3530 option? Just a wild guess from the PCI SSID, though. Takashi

2 2

[alsa-devel] Noisy USB audio stream using MPlayer
by Eran Ben-Avi 23 Jun '09

23 Jun '09

Hi, I encountered some issue with my USB audio device and couldn't find any similar issue in previous mailing lists, any help will be appreciated. When playing an audio stream using MPlayer Ver1.0rc2-4.1.2 (tried different sample rates 44K, 48K and different files WAV , MP3) using my Creative SB0271 USB device on ARM based system(Marvell Kirkwood SoC) and kernel 2.6.22.18, I hear an interfering constant noise additionally to the original audio stream. At the beginning I heard this noise also when I used aplay(for WAV) but after I reinstalled the ALSA-LIB and ALSA-UTILS(version 1.0.14rc4), the aplay suddenly started to work smooth. I tested different USB audio device(AKM) but got same noisy results. Another piece of information I can provide is that increasing the CPU utilization eliminates the noise completely - maybe a timing issue ??? Any ideas ? Thanks, Eran.

2 3

[alsa-devel] [PATCH] ASoC: TWL4030: Correct bypass event for voice sidetone
by Lopez Cruz, Misael 23 Jun '09

23 Jun '09

Event for voice sidetone was being interpreted as an analog HiFi bypass event because VSTPGA register offset is less than ARXR2_APGA_CTL offset. Reordering the register checks allows to handle voice digital bypass event properly. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl4030.c | 20 +++++++++++++------- 1 files changed, 13 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index f916a9a..df42fa2 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -712,7 +712,19 @@ static int bypass_event(struct snd_soc_dapm_widget *w, reg = twl4030_read_reg_cache(w->codec, m->reg); - if (m->reg <= TWL4030_REG_ARXR2_APGA_CTL) { + /* + * bypass_state[0:3] - analog HiFi bypass + * bypass_state[4] - analog voice bypass + * bypass_state[5] - digital voice bypass + * bypass_state[6:7] - digital HiFi bypass + */ + if (m->reg == TWL4030_REG_VSTPGA) { + /* Voice digital bypass */ + if (reg) + twl4030->bypass_state |= (1 << 5); + else + twl4030->bypass_state &= ~(1 << 5); + } else if (m->reg <= TWL4030_REG_ARXR2_APGA_CTL) { /* Analog bypass */ if (reg & (1 << m->shift)) twl4030->bypass_state |= @@ -726,12 +738,6 @@ static int bypass_event(struct snd_soc_dapm_widget *w, twl4030->bypass_state |= (1 << 4); else twl4030->bypass_state &= ~(1 << 4); - } else if (m->reg == TWL4030_REG_VSTPGA) { - /* Voice digital bypass */ - if (reg) - twl4030->bypass_state |= (1 << 5); - else - twl4030->bypass_state &= ~(1 << 5); } else { /* Digital bypass */ if (reg & (0x7 << m->shift)) -- 1.5.4.3

3 2

[alsa-devel] [PATCH] ASoC: TWL4030: Add AVADC Clock Priority
by Lopez Cruz, Misael 23 Jun '09

23 Jun '09

AVDAC clk priority allows to determine the path ADC must be connected when the codec is in option2 and both HiFi and Voice paths are enabled. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl4030.c | 12 ++++++++++++ 1 files changed, 12 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 780217a..42cdcce 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1036,6 +1036,16 @@ static DECLARE_TLV_DB_SCALE(digital_capture_tlv, 0, 100, 0); */ static DECLARE_TLV_DB_SCALE(input_gain_tlv, 0, 600, 0); +/* AVADC clock priority */ +static const char *twl4030_avadc_clk_priority_texts[] = { + "Voice high priority", "HiFi high priority" +}; + +static const struct soc_enum twl4030_avadc_clk_priority_enum = + SOC_ENUM_SINGLE(TWL4030_REG_AVADC_CTL, 2, + ARRAY_SIZE(twl4030_avadc_clk_priority_texts), + twl4030_avadc_clk_priority_texts); + static const char *twl4030_rampdelay_texts[] = { "27/20/14 ms", "55/40/27 ms", "109/81/55 ms", "218/161/109 ms", "437/323/218 ms", "874/645/437 ms", "1748/1291/874 ms", @@ -1137,6 +1147,8 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = { SOC_DOUBLE_TLV("Analog Capture Volume", TWL4030_REG_ANAMIC_GAIN, 0, 3, 5, 0, input_gain_tlv), + SOC_ENUM("AVADC Clock Priority", twl4030_avadc_clk_priority_enum), + SOC_ENUM("HS ramp delay", twl4030_rampdelay_enum), SOC_ENUM("Vibra H-bridge mode", twl4030_vibradirmode_enum), -- 1.5.4.3

3 2

[alsa-devel] Nmode issues
by Dmitri Seletski 23 Jun '09

23 Jun '09

Hello Guys. I got upgraded my kernel to uname -a Linux dimko 2.6.30 #3 SMP PREEMPT Mon Jun 22 23:58:16 IST 2009 x86_64 AMD Athlon(tm) 64 X2 Dual Core Processor 6000+ AuthenticAMD GNU/Linux i was able to make files now!(previously i couldnt) make make install modprobe ath9k [ 941.365887] f71882fg: Not a Fintek device [ 941.365932] f71882fg: Found f71882fg chip at 0xa00, revision 32 [ 941.366116] f71882fg f71882fg.2560: Fan: 1 is in duty-cycle mode [ 941.366118] f71882fg f71882fg.2560: Fan: 2 is in duty-cycle mode [ 941.366119] f71882fg f71882fg.2560: Fan: 3 is in duty-cycle mode [ 941.366121] f71882fg f71882fg.2560: Fan: 4 is in duty-cycle mode [ 941.377478] k8temp 0000:00:18.3: Temperature readouts might be wrong - check erratum #141 [ 956.642074] IRQ 19/nvidia: IRQF_DISABLED is not guaranteed on shared IRQs [ 957.433234] IRQ 19/nvidia: IRQF_DISABLED is not guaranteed on shared IRQs [ 1415.180230] AV200 0000:09:04.0: PCI INT A -> Link[LN4A] -> GSI 16 (level, low) -> IRQ 16 [ 1509.674361] cfg80211: Calling CRDA to update world regulatory domain [ 1509.674399] general protection fault: 0000 [#1] PREEMPT SMP [ 1509.674411] last sysfs file: /sys/devices/pci0000:00/0000:00:04.1/usb2/2-1/2-1:1.0/host8/target8:0:0/8:0:0:0/block/sdd/sdd1/stat [ 1509.674418] CPU 1 [ 1509.674422] Modules linked in: cfg80211 rfkill_backport snd_seq snd_virtuoso snd_oxygen_lib snd_pcm snd_timer snd_page_alloc snd_mpu401_uart snd_rawmidi snd_seq_device snd soundcore k8temp f71882fg hwmon i2c_dev nvidia(P) i2c_core [ 1509.674464] Pid: 29571, comm: modprobe Tainted: P 2.6.30 #3 MS-7375 [ 1509.674470] RIP: 0010:[<ffffffff810710f1>] [<ffffffff810710f1>] module_put+0x31/0x80 [ 1509.674488] RSP: 0018:ffff880119cebf38 EFLAGS: 00010287 [ 1509.674493] RAX: 0000000000000001 RBX: ffffffffa0886d00 RCX: 0000000000000000 [ 1509.674498] RDX: ffff10013a92b6c0 RSI: 0000000000000000 RDI: 0000000000000001 [ 1509.674503] RBP: ffff880119cebf48 R08: 0000000000000000 R09: 0000000000000000 [ 1509.674508] R10: 0000000000000000 R11: 0000000000000000 R12: 0000000000000000 [ 1509.674513] R13: 00007f4dae114010 R14: 00000000006da7f0 R15: 00000000006da808 [ 1509.674519] FS: 00007f4dae1766f0(0000) GS:ffff88002803b000(0000) knlGS:0000000000000000 [ 1509.674524] CS: 0010 DS: 0000 ES: 0000 CR0: 000000008005003b [ 1509.674529] CR2: 00007fb3a126d098 CR3: 000000010f527000 CR4: 00000000000006e0 [ 1509.674534] DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 [ 1509.674539] DR3: 0000000000000000 DR6: 00000000ffff0ff0 DR7: 0000000000000400 [ 1509.674545] Process modprobe (pid: 29571, threadinfo ffff880119cea000, task ffff880119d0c380) [ 1509.674549] Stack: [ 1509.674552] ffffffffa0886d00 ffffffffa0886d00 ffff880119cebf78 ffffffff810730ce [ 1509.674560] 0000000000000000 00000000006da680 0000000000000000 00000000006da6e0 [ 1509.674569] 00000000006da680 ffffffff8100beeb 0000000000000246 00000000006da6c0 [ 1509.674578] Call Trace: [ 1509.674583] [<ffffffff810730ce>] sys_init_module+0x10e/0x200 [ 1509.674592] [<ffffffff8100beeb>] system_call_fastpath+0x16/0x1b [ 1509.674604] Code: 89 fb 48 83 ec 08 48 85 ff 74 44 bf 01 00 00 00 e8 25 d5 fc ff e8 d0 48 15 00 48 98 48 8b 93 f0 01 00 00 48 03 14 c5 20 70 67 81 <48> ff 0a 83 3b 02 74 23 bf 01 00 00 00 e8 6d d4 fc ff 65 48 8b [ 1509.674680] RIP [<ffffffff810710f1>] module_put+0x31/0x80 [ 1509.674689] RSP <ffff880119cebf38> [ 1509.674695] ---[ end trace 6aaa73f49bc73faf ]--- [ 1509.674701] note: modprobe[29571] exited with preempt_count 1 after i tried to unload drivers(dont remember nvidia ones or ath9k ones) machine became unresponsive, i had to raise elephants. after which new kernel did not reboot properly. recompiled, it from working kernel, relaoded machine, everything seesm to be working now however, i still get same issue. pwd /home/dimko/compat-wireless-2009-06-20 tcp dump -i wlan0 [start] ip link set wlan0 start iwconfig wlan0 essid dlink dmesg 761.433090] wlan0: authenticated [ 761.433094] wlan0: associate with AP 00:1e:58:b4:f6:83 [ 761.633264] wlan0: associate with AP 00:1e:58:b4:f6:83 [ 761.640078] wlan0: deauthenticated (Reason: 9) [ 762.640264] wlan0: direct probe to AP 00:1e:58:b4:f6:83 try 1 [ 762.643782] wlan0 direct probe responded [ 762.643787] wlan0: authenticate with AP 00:1e:58:b4:f6:83 [ 762.648350] wlan0: authenticated [ 762.648354] wlan0: associate with AP 00:1e:58:b4:f6:83 [ 762.848263] wlan0: associate with AP 00:1e:58:b4:f6:83 [ 762.855024] wlan0: deauthenticated (Reason: 9) i tcpdum [stop] tcpdump -i wlan0 tcpdump: WARNING: wlan0: no IPv4 address assigned tcpdump: verbose output suppressed, use -v or -vv for full protocol decode listening on wlan0, link-type EN10MB (Ethernet), capture size 96 bytes ^C 0 packets captured 0 packets received by filter 0 packets dropped by kernel Same problem persist... pstree -p init(1)-+-acpid(4077) |-agetty(4579) |-agetty(4580) |-agetty(4581) |-agetty(4582) |-bonobo-activati(5073)---{bonobo-activati}(5074) |-console-kit-dae(4607)-+-{console-kit-dae}(4611) | |-{console-kit-dae}(4615) | |-{console-kit-dae}(4616) | |-{console-kit-dae}(4617) | |-{console-kit-dae}(4618) | |-{console-kit-dae}(4620) | |-{console-kit-dae}(4621) | |-{console-kit-dae}(4622) | |-{console-kit-dae}(4623) | |-{console-kit-dae}(4624) | |-{console-kit-dae}(4625) | |-{console-kit-dae}(4626) | |-{console-kit-dae}(4627) | |-{console-kit-dae}(4628) | |-{console-kit-dae}(4629) | |-{console-kit-dae}(4630) | |-{console-kit-dae}(4631) | |-{console-kit-dae}(4632) | |-{console-kit-dae}(4633) | |-{console-kit-dae}(4634) | |-{console-kit-dae}(4635) | |-{console-kit-dae}(4636) | |-{console-kit-dae}(4637) | |-{console-kit-dae}(4638) | |-{console-kit-dae}(4639) | |-{console-kit-dae}(4640) | |-{console-kit-dae}(4641) | |-{console-kit-dae}(4642) | |-{console-kit-dae}(4643) | |-{console-kit-dae}(4644) | |-{console-kit-dae}(4645) | |-{console-kit-dae}(4646) | |-{console-kit-dae}(4647) | |-{console-kit-dae}(4648) | |-{console-kit-dae}(4649) | |-{console-kit-dae}(4650) | |-{console-kit-dae}(4651) | |-{console-kit-dae}(4652) | |-{console-kit-dae}(4653) | |-{console-kit-dae}(4654) | |-{console-kit-dae}(4655) | |-{console-kit-dae}(4656) | |-{console-kit-dae}(4657) | |-{console-kit-dae}(4658) | |-{console-kit-dae}(4659) | |-{console-kit-dae}(4660) | |-{console-kit-dae}(4661) | |-{console-kit-dae}(4662) | |-{console-kit-dae}(4663) | |-{console-kit-dae}(4664) | |-{console-kit-dae}(4665) | |-{console-kit-dae}(4666) | |-{console-kit-dae}(4667) | |-{console-kit-dae}(4668) | |-{console-kit-dae}(4669) | |-{console-kit-dae}(4670) | |-{console-kit-dae}(4671) | |-{console-kit-dae}(4672) | |-{console-kit-dae}(4673) | |-{console-kit-dae}(4877) | |-{console-kit-dae}(4926) | `-{console-kit-dae}(4983) |-dbus-daemon(4233) |-dbus-daemon(5021) |-dbus-launch(5020) |-dhcpcd(3192) |-exaile(5149) |-gconfd-2(5027) |-gdm(4979)---gdm(4980)-+-X(4982) | `-gnome-session(5005)-+-gnome-panel(5070) | |-metacity(5066) | |-nautilus(5077) | |-seahorse-agent(5035) | |-ssh-agent(5025) | |-vino-server(5084) | `-{gnome-session}(5047) |-gnome-dictionar(5098) |-gnome-keyring-d(5044) |-gnome-power-man(5103) |-gnome-screensav(5065) |-gnome-settings-(5048)---{gnome-settings-}(5052) |-gnome-terminal(5163)-+-bash(5166)---su(5172)---bash(5175) | |-bash(5218)---su(5223)---bash(5226)---pstree(52+ | |-gnome-pty-helpe(5165) | `-{gnome-terminal}(5168) |-gnome-volume-ma(5104) |-gvfs-hal-volume(5102) |-gvfsd(5037) |-gvfsd-burn(5119) |-gvfsd-trash(5107) |-hald(4292)---hald-runner(4293)-+-hald-addon-acpi(4307) | |-hald-addon-cpuf(4306) | |-hald-addon-inpu(4298) | `-hald-addon-stor(4317) |-login(4577)---bash(4878) |-login(4578)---bash(4921) |-mdadm(2738) |-mixer_applet2(5090)---{mixer_applet2}(5117) |-mozilla-launche(5121)---thunderbird-bin(5131)-+-{thunderbird-bin}(513+ | |-{thunderbird-bin}(513+ | |-{thunderbird-bin}(513+ | |-{thunderbird-bin}(513+ | |-{thunderbird-bin}(513+ | |-{thunderbird-bin}(514+ | |-{thunderbird-bin}(514+ | `-{thunderbird-bin}(514+ |-multiload-apple(5080) |-smartd(4507) |-syslog-ng(4018) `-udevd(932) lsmod Module Size Used by ath9k 245620 0 mac80211 188280 1 ath9k ath 9512 1 ath9k cfg80211 98080 3 ath9k,mac80211,ath rfkill_backport 21520 1 cfg80211 nvidia 8121160 36 f71882fg 28632 0 snd_seq 62016 0 output 3432 0 ext4 241640 1 jbd2 63000 1 ext4 crc16 1960 1 ext4 snd_virtuoso 14564 1 snd_oxygen_lib 34344 1 snd_virtuoso snd_pcm 88816 1 snd_oxygen_lib snd_timer 24536 2 snd_seq,snd_pcm snd_page_alloc 9976 1 snd_pcm snd_mpu401_uart 8424 1 snd_oxygen_lib snd_rawmidi 23336 1 snd_mpu401_uart snd_seq_device 7772 2 snd_seq,snd_rawmidi snd 60968 10 snd_seq,snd_virtuoso,snd_oxygen_lib,snd_pcm,snd_timer,snd_mpu401_uart,snd_rawmidi,snd_seq_device r8169 37100 0 k8temp 5608 0 hwmon 2920 2 f71882fg,k8temp soundcore 1648 1 snd i2c_nforce2 8336 0 i2c_core 27328 2 nvidia,i2c_nforce2

2 1

[alsa-devel] alc889
by Stefan Krastanov 22 Jun '09

22 Jun '09

Hi I would like to help (or at least help myself) with the creation of a model config for alc889. I'm not a sound developer. I'm using a laptop with alc889 based sound card. The reported pins are really messed up (it's hard not to use more profane expression). My laptop is Fujitsu-Siemens Amilo xi but the problems are the same as here: https://answers.launchpad.net/ubuntu/+source/alsa-driver/+question/71205 or here: http://mailman.alsa-project.org/pipermail/alsa-devel/2008-October/011295.ht… namely: 1.- no jack sensing (extremely annoying - for two years allready) 2.- misnamed mixers in volume control for gnome or alsamixer (example: front instead of headphone) 3.- the volume-control potentiometer on the side of the laptop is not working 4.- I need to mute two different channel at the same time (center and LFE) to mute one physical output (build-in sub-woofer) 5.- A great deal of other small annoyances destroying the user experience I'm using model 3stack-dig and the optical output is working (at least is lighted up - I do not have the hardware to check it) My questions are: - What I need to do, to create new model option based on 3stack-dig with different pin mapping? - Is it possible that my present configuration (3stack-dig) is using pins that should not be used? I'm asking in search of an explanation of problem 4. - What I need to do to rename the controls in alsamixer? - What to do to repair jack sensing? - that's the most important of all - What to do to repair the potentiometer for volume control? I have seen that hda-analyser can create diff files, but where should they be used. (And why is hda-analyser not working - I was forced to use http://ubuntuforums.org/showpost.php?p=7227626&postcount=33. Is it that much to expect working tools on the official website. I understand that small bug will always exist, but that one is there for at least 10 months) I won't be surprised if many of my questions are answered in the documentation, but I was unable to find anything useful. It would be enough to point me a place to start reading. I know you guys are working hard and you have done great things, but it's really hard to find something in the documentation(which is occasionally quite old) Best regards and thank you for the great work Stefan Krastanov

2 1

[alsa-devel] [PATCH] sound: remove redundant tests on unsigned
by Roel Kluin 22 Jun '09

22 Jun '09

reg and channel_id are unsigned. When negative they are wrapped and caught by the other test. Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- sound/pci/ca0106/ca0106_proc.c | 2 +- sound/pci/emu10k1/emu10k1x.c | 3 +-- sound/pci/emu10k1/emuproc.c | 2 +- sound/soc/codecs/tlv320aic23.c | 2 +- 4 files changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/pci/ca0106/ca0106_proc.c b/sound/pci/ca0106/ca0106_proc.c index c62b7d1..2deac54 100644 --- a/sound/pci/ca0106/ca0106_proc.c +++ b/sound/pci/ca0106/ca0106_proc.c @@ -304,7 +304,7 @@ static void snd_ca0106_proc_reg_write32(struct snd_info_entry *entry, while (!snd_info_get_line(buffer, line, sizeof(line))) { if (sscanf(line, "%x %x", &reg, &val) != 2) continue; - if ((reg < 0x40) && (reg >=0) && (val <= 0xffffffff) ) { + if ((reg < 0x40) && (val <= 0xffffffff)) { spin_lock_irqsave(&emu->emu_lock, flags); outl(val, emu->port + (reg & 0xfffffffc)); spin_unlock_irqrestore(&emu->emu_lock, flags); diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index 4d3ad79..957b130 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -1040,8 +1040,7 @@ static void snd_emu10k1x_proc_reg_write(struct snd_info_entry *entry, if (sscanf(line, "%x %x %x", &reg, &channel_id, &val) != 3) continue; - if ((reg < 0x49) && (reg >= 0) && (val <= 0xffffffff) - && (channel_id >= 0) && (channel_id <= 2) ) + if ((reg < 0x49) && (val <= 0xffffffff) && (channel_id <= 2)) snd_emu10k1x_ptr_write(emu, reg, channel_id, val); } } diff --git a/sound/pci/emu10k1/emuproc.c b/sound/pci/emu10k1/emuproc.c index 216f974..6ad3825 100644 --- a/sound/pci/emu10k1/emuproc.c +++ b/sound/pci/emu10k1/emuproc.c @@ -527,7 +527,7 @@ static void snd_emu_proc_ptr_reg_write(struct snd_info_entry *entry, while (!snd_info_get_line(buffer, line, sizeof(line))) { if (sscanf(line, "%x %x %x", &reg, &channel_id, &val) != 3) continue; - if ((reg < 0xa0) && (reg >= 0) && (val <= 0xffffffff) && (channel_id >= 0) && (channel_id <= 3) ) + if ((reg < 0xa0) && (val <= 0xffffffff) && (channel_id <= 3)) snd_ptr_write(emu, iobase, reg, channel_id, val); } } diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 0b8dcb5..6f8cd8d 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -85,7 +85,7 @@ static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg, * of data into val */ - if ((reg < 0 || reg > 9) && (reg != 15)) { + if ((reg > 9) && (reg != 15)) { printk(KERN_WARNING "%s Invalid register R%u\n", __func__, reg); return -1; }

2 1

Re: [alsa-devel] [PATCH] New ASoC Drivers for ADI AD1938 codec
by Mark Brown 22 Jun '09

22 Jun '09

On Fri, Jun 19, 2009 at 05:28:15PM +0800, Barry Song wrote: > 1. add AD1938 codec driver (codec) > 2. add blackfin SPORT-TDM DAI and PCM driver (platform) > 3. add bf5xx board with AD1938 driver (machine) As Liam said you really need to submit this as a patch series rather than as a single big patch - as your commit log here indicates you've got several different things going on here. > +++ b/include/sound/soc-dai.h > @@ -30,6 +30,7 @@ struct snd_pcm_substream; > #define SND_SOC_DAIFMT_DSP_A 3 /* L data msb after FRM LRC */ > #define SND_SOC_DAIFMT_DSP_B 4 /* L data msb during FRM LRC */ > #define SND_SOC_DAIFMT_AC97 5 /* AC97 */ > +#define SND_SOC_DAIFMT_SPORT_TDM 6 /* SPORT TDM for ADI parts */ If you're going to add a new DAI format that really needs more explanation than this explaining what the DAI format is. It'd be very surprising to see hardware needing a new format. Looking at the datasheet for the ad1938 it appears that the actual format here is just normal I2S with TDM. This does not need a new DAI format or new CPU DAI, you just need to add suport for TDM to the Blackfin I2S driver. The format is fairly standard and implemented by a number of other devices. See set_tdm_slot() for setting up the higher channel counts - there's some ongoing revisions to that API so you'll want to also ensure that the code is set up so that it can cope with specification of the sample width for each slot in set_tdm_slot(). Given this I've only looked at the CODEC driver below. > diff --git a/sound/soc/codecs/ad1938.c b/sound/soc/codecs/ad1938.c > new file mode 100644 > index 0000000..9aa78e1 > --- /dev/null > +++ b/sound/soc/codecs/ad1938.c > + * > + * Revision history > + * 4 June 2009 Initial version. Don't include this, git provides code history for us. > +struct snd_soc_device *ad1938_socdev; > + > +/* dac de-emphasis enum control */ > +static const char *ad1938_deemp[] = {"flat", "48kHz", "44.1kHz", "32kHz"}; For consistency with other drivers "flat" should be "None". > +/* AD1938 volume/mute/de-emphasis etc. controls */ > +static const struct snd_kcontrol_new ad1938_snd_controls[] = { > + /* DAC volume control */ > + SOC_SINGLE("DAC L1 Volume", AD1938_DAC_L1_VOL, 0, 0xFF, 1), > + SOC_SINGLE("DAC R1 Volume", AD1938_DAC_R1_VOL, 0, 0xFF, 1), These (and the other stereo pairs below) should be SOC_DOUBLE_R(). This allows ALSA to represent them as stereo controls to applications rather than as two separate controls. You should also provide TLV information so actually SOC_DOUBLE_R_TLV() if possible. > + /* DAC mute control */ > + SOC_SINGLE("DAC L1 Switch", AD1938_DAC_CHNL_MUTE, 0, 1, 1), > + SOC_SINGLE("DAC R1 Switch", AD1938_DAC_CHNL_MUTE, 1, 1, 1), These should be stereo controls too - SOC_DOUBLE() since they're in the same register. > + /* ADC mute control */ > + SOC_SINGLE("ADC L1 Switch", AD1938_ADC_CTRL0, ADC0_MUTE, 1, 1), > + SOC_SINGLE("ADC R1 Switch", AD1938_ADC_CTRL0, ADC1_MUTE, 1, 1), These too. > + /* DAC de-emphasis */ > + SOC_ENUM("Playback Deemphasis", ad1938_enum[0]), Don't put your enums in an array, use named variables for them. This makes drivers easier to maintian when you get a lot of enums. > +static int ad1938_add_controls(struct snd_soc_codec *codec) > +{ > + int err, i; > + > + for (i = 0; i < ARRAY_SIZE(ad1938_snd_controls); i++) { > + err = snd_ctl_add(codec->card, > + snd_soc_cnew(&ad1938_snd_controls[i], codec, NULL)); Use snd_soc_add_controls() here - you can replace the entire function with a call to that. > +/* dac/adc/pll poweron/off functions */ > +static int ad1938_dac_powerctrl(struct snd_soc_codec *codec, int cmd) > +{ > + int reg; > + > + reg = codec->read(codec, AD1938_DAC_CTRL0); > + if (cmd) > + reg &= ~DAC_POWERDOWN; > + else > + reg |= DAC_POWERDOWN; > + codec->write(codec, AD1938_DAC_CTRL0, reg); This should be handled by DAPM - either have a single DAC widget representing all the channels (since you don't appear to have independant control anyway) or have a bunch of dummy DAC widgets and a supply widget representing the actual power control. The same thing applies to the ADCs. > +static int ad1938_set_pll(struct snd_soc_dai *codec_dai, > + int pll_id, unsigned int freq_in, unsigned int freq_out) > +{ > + struct snd_soc_codec *codec = codec_dai->codec; > + > + if (freq_out) > + ad1938_pll_powerctrl(codec, 1); > + else { > + /* playing while recording, framework will poweroff-poweron pll redundantly */ > + if ((!codec_dai->capture.active) && (!codec_dai->playback.active)) > + ad1938_pll_powerctrl(codec, 0); > + } Hrm. This appears to completely ignore the frequencies supplied for the PLL and just provide power control. I suspect that you can just handle the PLL as a SND_SOC_DAPM_SUPPLY(), there seems to be no need to expose the set_pll() operation and make machine drivers call it given that there isn't any frequency configuration going on. > +static int ad1938_mute(struct snd_soc_dai *dai, int mute) > +{ > + struct snd_soc_codec *codec = dai->codec; > + > + if (!mute) > + codec->write(codec, AD1938_DAC_CHNL_MUTE, 0); > + else > + codec->write(codec, AD1938_DAC_CHNL_MUTE, 0xff); > + > + return 0; > +} This isn't going to play well with the explicit mute controls you've got above - it's writing to the same register bits without any coordination. One or the other set of controls ought to be removed. > +static int ad1938_tdm_set(struct snd_soc_codec *codec) > +{ > + codec->write(codec, AD1938_DAC_CTRL0, (codec->read(codec, AD1938_DAC_CTRL0) & > + (~DAC_SERFMT_MASK)) | DAC_SERFMT_TDM); > + codec->write(codec, AD1938_DAC_CTRL1, 0x84); /* invert bclk, 256bclk/frame, latch in mid */ > + codec->write(codec, AD1938_ADC_CTRL1, 0x43); /* sata delay=1, adc aux mode */ > + codec->write(codec, AD1938_ADC_CTRL2, 0x6F); /* left high, driver on rising edge */ > + > + return 0; > +} If you use set_tdm_slot() then the BCLK/frame ratio will be set by that. Inversion of BCLK (and any other clocks) should be handled by the set_dai_fmt() operation based on the machine driver request rather than done unconditionally. > + /* bit size */ > + switch (params_format(params)) { > + case SNDRV_PCM_FORMAT_S16_LE: > + word_len = 3; > + break; Once you implement set_tdm_slot() you should allow the word length to be configured there if it's called or otherwise keep this code here - see Daniel Ribeiro's patche "change set_tdm_slot api to allow slot_width override" posted to the ALSA list this week. > +static int __devinit ad1938_spi_probe(struct spi_device *spi) > +{ > + spi->dev.power.power_state = PMSG_ON; > + ad1938_socdev->card->codec->control_data = spi; > + > + return 0; > +} > + > +static int __devexit ad1938_spi_remove(struct spi_device *spi) > +{ > + return 0; > +} Your device probing should all be restructured so that the SPI device for the CODEC is registered as any other SPI device rather than being set up as part of probing the ASoC device. See the wm8731 driver for an example of doing this for a SPI device. This will require that the arch code for any systems with the ad1938 do the setup of the device. > + .name = "AD1938", > + .playback = { > + .stream_name = "Playback", > + .channels_min = 2, > + .channels_max = 8, > + .rates = SNDRV_PCM_RATE_48000, > + .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, }, Please keep your lines to under 80 columns. > +#define AD1938_PLL_CLK_CTRL0 0 > +#define PLL_POWERDOWN 0x01 > +#define AD1938_PLL_CLK_CTRL1 1 > +#define AD1938_DAC_CTRL0 2 > +#define DAC_POWERDOWN 0x01 > +#define DAC_SERFMT_MASK 0xC0 > +#define DAC_SERFMT_STEREO (0 << 6) > +#define DAC_SERFMT_TDM (1 << 6) These defines need namespacing if they're going to appear in the headers - everything should have the AD1938_ prefix.

4 10

[PATCH] ASoC: Add a shutdown callback
by Mark Brown 22 Jun '09

22 Jun '09

Ensure that the audio subsystem is powered down cleanly when the system shuts down by providing a shutdown operation. This ensures that all the components have been returned to an off state cleanly which should avoid audio issues from partially charged capacitors or noise on digital inputs if the system is restarted quickly. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 1 + sound/soc/soc-core.c | 16 ++++++++++++++++ sound/soc/soc-dapm.c | 29 +++++++++++++++++++++++++++++ 3 files changed, 46 insertions(+), 0 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index ec8a45f..35814ce 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -279,6 +279,7 @@ int snd_soc_dapm_add_routes(struct snd_soc_codec *codec, /* dapm events */ int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, char *stream, int event); +void snd_soc_dapm_shutdown(struct snd_soc_device *socdev); /* dapm sys fs - used by the core */ int snd_soc_dapm_sys_add(struct device *dev); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4414117..55d45c4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1020,6 +1020,21 @@ static int soc_remove(struct platform_device *pdev) return 0; } +static void soc_shutdown(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_card *card = socdev->card; + + if (!card->instantiated) + return; + + /* Flush out pmdown_time work - we actually do want to run it + * now, we're shutting down so no imminent restart. */ + run_delayed_work(&card->delayed_work); + + snd_soc_dapm_shutdown(socdev); +} + /* ASoC platform driver */ static struct platform_driver soc_driver = { .driver = { @@ -1030,6 +1045,7 @@ static struct platform_driver soc_driver = { .remove = soc_remove, .suspend = soc_suspend, .resume = soc_resume, + .shutdown = soc_shutdown, }; /* create a new pcm */ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 6534359..b9129ef 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2032,6 +2032,35 @@ void snd_soc_dapm_free(struct snd_soc_device *socdev) } EXPORT_SYMBOL_GPL(snd_soc_dapm_free); +/* + * snd_soc_dapm_shutdown - callback for system shutdown + */ +void snd_soc_dapm_shutdown(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->card->codec; + struct snd_soc_dapm_widget *w; + LIST_HEAD(down_list); + int powerdown = 0; + + list_for_each_entry(w, &codec->dapm_widgets, list) { + if (w->power) { + dapm_seq_insert(w, &down_list, dapm_down_seq); + powerdown = 1; + } + } + + /* If there were no widgets to power down we're already in + * standby. + */ + if (powerdown) { + snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_PREPARE); + dapm_seq_run(codec, &down_list, 0, dapm_down_seq); + snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_STANDBY); + } + + snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_OFF); +} + /* Module information */ MODULE_AUTHOR("Liam Girdwood, lrg(a)slimlogic.co.uk"); MODULE_DESCRIPTION("Dynamic Audio Power Management core for ALSA SoC"); -- 1.6.3.1

1 0

[alsa-devel] [PATCH 2/3] ASoC: TWL4030: Add AVADC Clock Priority
by Lopez Cruz, Misael 22 Jun '09

22 Jun '09

AVDAC clk priority allows to determine the path ADC must be connected when the codec is in option2 and both audio and voice paths are enabled. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl4030.c | 12 ++++++++++++ 1 files changed, 12 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 780217a..fd5b30f 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1036,6 +1036,16 @@ static DECLARE_TLV_DB_SCALE(digital_capture_tlv, 0, 100, 0); */ static DECLARE_TLV_DB_SCALE(input_gain_tlv, 0, 600, 0); +/* AVADC clock priority */ +static const char *twl4030_avadc_clk_priority_texts[] = { + "Voice high priority", "Audio high priority" +}; + +static const struct soc_enum twl4030_avadc_clk_priority_enum = + SOC_ENUM_SINGLE(TWL4030_REG_AVADC_CTL, 2, + ARRAY_SIZE(twl4030_avadc_clk_priority_texts), + twl4030_avadc_clk_priority_texts); + static const char *twl4030_rampdelay_texts[] = { "27/20/14 ms", "55/40/27 ms", "109/81/55 ms", "218/161/109 ms", "437/323/218 ms", "874/645/437 ms", "1748/1291/874 ms", @@ -1137,6 +1147,8 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = { SOC_DOUBLE_TLV("Analog Capture Volume", TWL4030_REG_ANAMIC_GAIN, 0, 3, 5, 0, input_gain_tlv), + SOC_ENUM("AVADC Clock Priority", twl4030_avadc_clk_priority_enum), + SOC_ENUM("HS ramp delay", twl4030_rampdelay_enum), SOC_ENUM("Vibra H-bridge mode", twl4030_vibradirmode_enum), -- 1.5.4.3

3 3

[alsa-devel] ASoC updates for 2.6.31
by Mark Brown 22 Jun '09

22 Jun '09

The following changes since commit ef39412622b6e8f09c383de9565b07e93553fc27: Takashi Iwai (1): ASoC: Kill BUS_ID_SIZE are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31 Barry Song (1): ASoC: Blackfin: update the bf5xx_i2s_resume parameters Cliff Cai (1): ASoC: Blackfin: keep better track of SPORT configuration state sound/soc/blackfin/bf5xx-i2s.c | 10 +++++++--- 1 files changed, 7 insertions(+), 3 deletions(-)

2 1

[alsa-devel] [PATCH] via82xx: add option to disable 500ms delay in snd_via82xx_codec_wait
by Simon Arlott 22 Jun '09

22 Jun '09

There's a large 500ms delay in snd_via82xx_codec_wait() that, at least on my hardware, appears to be unnecessary. The rest of the init of the card works without logging any warnings or errors and both audio and mixer settings work. This adds an "nodelay" parameter to disable this (undocumented in the code) large delay improving bootup time by 489-500ms. [ 1.034217] initcall alsa_card_via82xx_init+0x0/0x16 returned 0 after 505757 usecs vs. [ 0.533136] initcall alsa_card_via82xx_init+0x0/0x16 returned 0 after 15915 usecs Signed-off-by: Simon Arlott <simon(a)fire.lp0.eu> --- sound/pci/via82xx.c | 6 +++++- 1 files changed, 5 insertions(+), 1 deletions(-) diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 1ef58c5..949fcaf 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -85,6 +85,7 @@ static int joystick; static int ac97_clock = 48000; static char *ac97_quirk; static int dxs_support; +static int nodelay; module_param(index, int, 0444); MODULE_PARM_DESC(index, "Index value for VIA 82xx bridge."); @@ -102,6 +103,8 @@ module_param(ac97_quirk, charp, 0444); MODULE_PARM_DESC(ac97_quirk, "AC'97 workaround for strange hardware."); module_param(dxs_support, int, 0444); MODULE_PARM_DESC(dxs_support, "Support for DXS channels (0 = auto, 1 = enable, 2 = disable, 3 = 48k only, 4 = no VRA, 5 = enable any sample rate)"); +module_param(nodelay, int, 0444); +MODULE_PARM_DESC(nodelay, "Disable 500ms init delay"); /* just for backward compatibility */ static int enable; @@ -549,7 +552,8 @@ static void snd_via82xx_codec_wait(struct snd_ac97 *ac97) int err; err = snd_via82xx_codec_ready(chip, ac97->num); /* here we need to wait fairly for long time.. */ - msleep(500); + if (!nodelay) + msleep(500); } static void snd_via82xx_codec_write(struct snd_ac97 *ac97, -- 1.6.3.1 -- Simon Arlott

2 1

[alsa-devel] Need help on MMAP
by Christopher Tsui 22 Jun '09

22 Jun '09

Hi I am in the process of add mmap support in my ALSA driver. So this is what i have done so far In snd_pcm_hardware structure .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID In snd_pcm_ops structure .mmap = my_pcm_mmap So to test my code, I use aplay -M -Dhw:0 rock.wav and this is what I get ALSA lib pcm_mmap.c:378:(snd_pcm_mmap) mmap failed: Invalid argument aplay: set_params:1022: Unable to install hw params: ACCESS: MMAP_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 32 CHANNELS: 2 RATE: 44100 PERIOD_TIME: (27210 27211) PERIOD_SIZE: 1200 PERIOD_BYTES: 4800 PERIODS: 4 BUFFER_TIME: (108843 108844) BUFFER_SIZE: 4800 BUFFER_BYTES: 19200 TICK_TIME: 10000 What am I doing wrong?? Thanks in advance

1 0

[alsa-devel] Acer 7730G support for ALSA.
by Denis Kup 21 Jun '09

21 Jun '09

Hey, when i first installed alsa to my laptop, there no sound from subwoofer, jacks were map very strange (2 surrounds for ex.). I try all models from the list, but all don't support my subwoofer. After a weak of hard work i made a patch that enables ACER_ASPIRE_7730G model for my laptop. It have ALC888S, 3 jacks, 6 channels, stereo internal frontal speakers and internal subwoofer.You can see acer.com for more info about hardware and jacks. It's the changes i made to /alsa-kernel/pci/hda/patch_realtek.c: > enum { > ................... > ALC888_ACER_ASPIRE_7730G, > ................... > }; > ////////////////////////////////////////////////////////////// > static struct hda_verb alc888_acer_aspire_7730G_verbs[] = { > {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /*CHANGE DIRECT FRONT TO BLACK JACK*/ > {0x17, AC_VERB_SET_CONNECT_SEL, 0x02}, /* CHANGE: ENABLES SUBWOOFER ON 7730*/ > {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, //CHANGE FOR AUTOMUTE > { } /* end */ > }; > > /*CHANGE ENABLE AUTOMUTING ON ACER 7730G > ////////////////////////////////////////////////////// > //////////////////////////////////////////////////////*/ > > static void alc888_acer_aspire_7730G_front_automute(struct hda_codec *codec) > { > unsigned int present; > > present = snd_hda_codec_read(codec, 0x15, 0, > AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; > snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, > HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); > snd_hda_codec_amp_stereo(codec, 0x17, HDA_OUTPUT, 0, > HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); > } > > static void alc888_acer_aspire_7730G_unsol_event(struct hda_codec *codec, > unsigned int res) > { > if ((res >> 26) == ALC880_HP_EVENT) > alc888_acer_aspire_7730G_front_automute(codec); > } > > /*//////////////////////////////////////////////////// > ////////////////////////////////////////////////// > //////////////////////////////////////////////////////*/ > static struct alc_config_preset alc883_presets[] = { > > [ALC888_ACER_ASPIRE_7730G] = { > .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer }, > .init_verbs = { alc883_init_verbs, alc888_acer_aspire_7730G_verbs }, > .num_dacs = ARRAY_SIZE(alc883_dac_nids), > .dac_nids = alc883_dac_nids, > .dig_out_nid = ALC883_DIGOUT_NID, > .dig_in_nid = ALC883_DIGIN_NID, > .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes), > .channel_mode = alc883_3ST_6ch_modes, > .need_dac_fix = 1, > .input_mux = &alc883_capture_source, > .unsol_event = alc888_acer_aspire_7730G_unsol_event, /* CHANGE AUTOMUTE DECLARATION*/ > .init_hook = alc888_acer_aspire_7730G_front_automute, > }, > > /////////////////////////////////////////////////////////////// > > static const char *alc883_models[ALC883_MODEL_LAST] = { > ................................. > [ALC888_ACER_ASPIRE_7730G] = "acer-aspire-7730g", > ......................... > > //////////////////////////////// > static struct snd_pci_quirk alc883_cfg_tbl[] = { > ...................................... > SND_PCI_QUIRK(0x1025, 0x0142, "Acer Aspire 7730G", ALC888_ACER_ASPIRE_7730G), > .......................................... After changing file i install driver, and it automatically detect its model (maybe there are old settings from previous install, so i dont 100% sure that it was detected). With this settings subwoofer works perfect, internal stereo and subwoofer mute when headphone are plugged all works, except 1 thing: subwoofer uses center channel instead LFE. To fix it i use such asound.conf file: > pcm.upmix_20to51 { > type plug > slave.pcm lowpass_21to21 > slave.channels 3 > ttable { > 0.0 1 # left channel > 1.1 1 # right channel > 0.2 1 # mix left and right ... > 1.2 1 # ... channel for subwoofer > } > } > pcm.lowpass_21to21 { > type ladspa > slave.pcm upmix_21to51 > path "/usr/lib/ladspa" > channels 3 > plugins { > 0 { > id 1098 # Identity (Audio) (1098/identity_audio) > policy duplicate > input.bindings.0 "Input"; > output.bindings.0 "Output"; > } > 1 { > id 1671 # 4 Pole Low-Pass Filter with Resonance (FCRCIA) (1672/lp4pole_fcrcia_oa) > policy none > input.bindings.2 "Input"; > output.bindings.2 "Output"; > input { > controls [ 1000 2 ] > } > } > } > } > pcm.upmix_21to51 { > type plug > slave.pcm surround51 > slave.channels 6 > ttable { > 0.0 1 # front left > 1.1 1 # front right > 0.2 1 # rear left > 1.3 1 # rear right > 0.5 0.5 # swap channels > 1.5 0.5 > 2.4 1 > } > } > > pcm.swapCLFE_51 { > type plug > slave.pcm surround51 > slave.channels 6 > ttable { > 0.0 1 # front left > 1.1 1 # front right > 2.2 1 # rear left > 3.3 1 # rear right > 4.5 1 # swap channels > 5.4 1 > } > } I use upmix_21to51 and swapCLFE_51 devices in Amarok settings. Is there any chance to swap Center and LFE in patch_realtek.c? And can my 7730g config be used in future version of alsa? And how can I detect params for snd_pci_quirk alc883_cfg_tbl ? And some config settings for my laptop, if needed: > linux-wpg9:/home/dener # cat /proc/asound/card0/codec* > Codec: Realtek ALC888 > Address: 0 > Function Id: 0x1 > Vendor Id: 0x10ec0888 > Subsystem Id: 0x10250142 > Revision Id: 0x100202 > No Modem Function Group found > Default PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Default Amp-In caps: N/A > Default Amp-Out caps: N/A > GPIO: io=2, o=0, i=0, unsolicited=1, wake=1 > IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > Node 0x02 [Audio Output] wcaps 0x411: Stereo > Converter: stream=5, channel=0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Node 0x03 [Audio Output] wcaps 0x411: Stereo > Converter: stream=5, channel=0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Node 0x04 [Audio Output] wcaps 0x411: Stereo > Converter: stream=5, channel=0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Node 0x05 [Audio Output] wcaps 0x411: Stereo > Converter: stream=0, channel=0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Node 0x06 [Audio Output] wcaps 0x611: Stereo Digital > Converter: stream=0, channel=0 > Digital: Enabled GenLevel > Digital category: 0x2 > PCM: > rates [0x5e0]: 44100 48000 88200 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x08 [Audio Input] wcaps 0x10051b: Stereo Amp-In > Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 > Amp-In vals: [0x16 0x16] > Converter: stream=0, channel=0 > SDI-Select: 0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Connection: 1 > 0x23 > Node 0x09 [Audio Input] wcaps 0x10051b: Stereo Amp-In > Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 > Amp-In vals: [0x9f 0x9f] > Converter: stream=0, channel=0 > SDI-Select: 0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Connection: 1 > 0x22 > Node 0x0a [Audio Input] wcaps 0x100711: Stereo Digital > Converter: stream=0, channel=0 > SDI-Select: 0 > Digital: > Digital category: 0x0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Connection: 1 > 0x1f > Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 > Amp-In vals: [0x80 0x80] [0x1f 0x1f] [0x00 0x00] [0x80 0x80] [0x17 0x17] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] > Connection: 10 > 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 > Node 0x0c [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x00 0x00] [0x00 0x00] > Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 > Amp-Out vals: [0x12 0x12] > Connection: 2 > 0x02 0x0b > Node 0x0d [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x00 0x00] [0x00 0x00] > Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 > Amp-Out vals: [0x15 0x15] > Connection: 2 > 0x03 0x0b > Node 0x0e [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x00 0x00] [0x00 0x00] > Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 > Amp-Out vals: [0x15 0x15] > Connection: 2 > 0x04 0x0b > Node 0x0f [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x80 0x80] [0x80 0x80] > Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 > Amp-Out vals: [0x00 0x00] > Connection: 2 > 0x05 0x0b > Node 0x10 [Audio Output] wcaps 0x611: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > PCM: > rates [0x5e0]: 44100 48000 88200 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Node 0x11 [Pin Complex] wcaps 0x400780: Mono Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 1 > 0x10 > Node 0x12 [Pin Complex] wcaps 0x400401: Stereo > Pincap 0x00000020: IN > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Power: setting=D0, actual=D0 > Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x14 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x0001003e: IN OUT HP EAPD Detect Trigger > EAPD 0x0: > Pin Default 0x99130110: [Fixed] Speaker at Int ATAPI > Conn = ATAPI, Color = Unknown > DefAssociation = 0x1, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c* 0x0d 0x0e 0x0f 0x26 > Node 0x15 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x0001003e: IN OUT HP EAPD Detect Trigger > EAPD 0x0: > Pin Default 0x0221101f: [Jack] HP Out at Ext Front > Conn = 1/8, Color = Black > DefAssociation = 0x1, Sequence = 0xf > Pin-ctls: 0x40: OUT > Unsolicited: tag=04, enabled=1 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c* 0x0d 0x0e 0x0f 0x26 > Node 0x16 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x00000036: IN OUT Detect Trigger > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c 0x0d 0x0e* 0x0f 0x26 > Node 0x17 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x00000036: IN OUT Detect Trigger > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c 0x0d 0x0e* 0x0f 0x26 > Node 0x18 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x0000373e: IN OUT HP Detect Trigger > Vref caps: HIZ 50 GRD 80 100 > Pin Default 0x02a19c30: [Jack] Mic at Ext Front > Conn = 1/8, Color = Pink > DefAssociation = 0x3, Sequence = 0x0 > Pin-ctls: 0x40: OUT VREF_HIZ > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c 0x0d 0x0e* 0x0f 0x26 > Node 0x19 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x80 0x80] > Pincap 0x0000373e: IN OUT HP Detect Trigger > Vref caps: HIZ 50 GRD 80 100 > Pin Default 0x99a30931: [Fixed] Mic at Int ATAPI > Conn = ATAPI, Color = Unknown > DefAssociation = 0x3, Sequence = 0x1 > Misc = NO_PRESENCE > Pin-ctls: 0x24: IN VREF_80 > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c* 0x0d 0x0e 0x0f 0x26 > Node 0x1a [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x0000373e: IN OUT HP Detect Trigger > Vref caps: HIZ 50 GRD 80 100 > Pin Default 0x0281343f: [Jack] Line In at Ext Front > Conn = 1/8, Color = Blue > DefAssociation = 0x3, Sequence = 0xf > Pin-ctls: 0x40: OUT VREF_HIZ > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c 0x0d* 0x0e 0x0f 0x26 > Node 0x1b [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x0000373e: IN OUT HP Detect Trigger > Vref caps: HIZ 50 GRD 80 100 > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0xc0: OUT HP VREF_HIZ > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c* 0x0d 0x0e 0x0f 0x26 > Node 0x1c [Pin Complex] wcaps 0x400481: Stereo > Pincap 0x00000024: IN Detect > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x20: IN > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Node 0x1d [Pin Complex] wcaps 0x400400: Mono > Pincap 0x00000020: IN > Pin Default 0x4016852d: [N/A] Speaker at Ext N/A > Conn = Digital, Color = Purple > DefAssociation = 0x2, Sequence = 0xd > Misc = NO_PRESENCE > Pin-ctls: 0x20: IN > Power: setting=D0, actual=D0 > Node 0x1e [Pin Complex] wcaps 0x400780: Mono Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x02451120: [Jack] SPDIF Out at Ext Front > Conn = Optical, Color = Black > DefAssociation = 0x2, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 1 > 0x06 > Node 0x1f [Pin Complex] wcaps 0x400680: Mono Digital > Pincap 0x00000024: IN Detect > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x20: IN > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono > Processing caps: benign=0, ncoeff=25 > Processing Coefficient: 0x00 > Coefficient Index: 0x09 > Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x80 0x80] [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] > Connection: 12 > 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b 0x12 > Node 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] > Connection: 11 > 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b > Node 0x24 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x25 [Audio Output] wcaps 0x411: Stereo > Converter: stream=0, channel=0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Node 0x26 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x00 0x00] [0x80 0x80] > Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 > Amp-Out vals: [0x00 0x00] > Connection: 2 > 0x25 0x0b > Codec: LSI ID 1040 > Address: 1 > Function Id: 0x2 > Vendor Id: 0x11c11040 > Subsystem Id: 0x1025013d > Revision Id: 0x100200 > Modem Function Group: 0x1 > Codec: Nvidia MCP78 HDMI > Address: 3 > Function Id: 0x1 > Vendor Id: 0x10de0006 > Subsystem Id: 0x10de0101 > Revision Id: 0x100000 > No Modem Function Group found > Default PCM: > rates [0x0]: > bits [0x0]: > formats [0x0]: > Default Amp-In caps: N/A > Default Amp-Out caps: N/A > GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 > Node 0x04 [Audio Output] wcaps 0x211: Stereo Digital > Converter: stream=0, channel=0 > Digital: Enabled > Digital category: 0x0 > PCM: > rates [0xc0]: 48000 88200 > bits [0xf]: 8 16 20 24 > formats [0x1]: PCM > Node 0x05 [Pin Complex] wcaps 0x400381: Stereo Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x18560110: [Jack] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0x1, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Connection: 1 > 0x04 > Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > PCM: > rates [0xc0]: 48000 88200 > bits [0xf]: 8 16 20 24 > formats [0x1]: PCM > Node 0x07 [Pin Complex] wcaps 0x400381: Stereo Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x58560121: [N/A] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0x2, Sequence = 0x1 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Unsolicited: tag=00, enabled=0 > Connection: 1 > 0x06 > Node 0x08 [Audio Output] wcaps 0x211: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > PCM: > rates [0xc0]: 48000 88200 > bits [0xf]: 8 16 20 24 > formats [0x1]: PCM > Node 0x09 [Pin Complex] wcaps 0x400381: Stereo Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x58560122: [N/A] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0x2, Sequence = 0x2 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Unsolicited: tag=00, enabled=0 > Connection: 1 > 0x08 > Node 0x0a [Audio Output] wcaps 0x211: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > PCM: > rates [0xc0]: 48000 88200 > bits [0xf]: 8 16 20 24 > formats [0x1]: PCM > Node 0x0b [Pin Complex] wcaps 0x400381: Stereo Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x58560123: [N/A] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0x2, Sequence = 0x3 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Unsolicited: tag=00, enabled=0 > Connection: 1 > 0x0a > Node 0x0c [Audio Output] wcaps 0x211: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > PCM: > rates [0xc0]: 48000 88200 > bits [0xf]: 8 16 20 24 > formats [0x1]: PCM > Node 0x0d [Pin Complex] wcaps 0x400381: Stereo Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x58560124: [N/A] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0x2, Sequence = 0x4 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Unsolicited: tag=00, enabled=0 > Connection: 1 > 0x0c > > linux-wpg9:/home/dener # cat /proc/asound/version > Advanced Linux Sound Architecture Driver Version 1.0.20. > Compiled on Jun 14 2009 for kernel 2.6.27.23-0.1-pae (SMP). > > cat /proc/asound/devices > 0: [ 0] : control > 1: : sequencer > 4: [ 0- 0]: hardware dependent > 5: [ 0- 1]: hardware dependent > 7: [ 0- 3]: hardware dependent > 16: [ 0- 0]: digital audio playback > 17: [ 0- 1]: digital audio playback > 19: [ 0- 3]: digital audio playback > 24: [ 0- 0]: digital audio capture > 25: [ 0- 1]: digital audio capture > 26: [ 0- 2]: digital audio capture > 33: : timer > > cat /proc/asound/pcm > 00-00: ALC888 Analog : ALC888 Analog : playback 1 : capture 1 > 00-01: ALC888 Digital : ALC888 Digital : playback 1 : capture 1 > 00-02: ALC888 Analog : ALC888 Analog : capture 1 > 00-03: NVIDIA HDMI : NVIDIA HDMI : playback 1 > > lspci | grep -i audio > 00:1b.0 Audio device: Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 03) > I have 1.0.20 alsa version and OpenSUSE 11.1 distributive. I hope my materials are usefull PS I am from Russia, sorry for my bad English :)

1 0

[alsa-devel] [PATCH] Acer Inspire 6530G model for Realtek ALC888
by Tony Vroon 21 Jun '09

21 Jun '09

The selected 4930G model seemed to keep the subwoofer 'tuba' function from operating correctly. Removing the existing PCI ID match made this work again, but it was mapped to 'Side' instead of to LFE as one would expect. This attempts to enable all functionality and keep the amount of available mixer sliders low. Any slider that had no audible effect on the output audio has been removed, and as such EAPD is not currently enabled. Signed-off-by: Tony Vroon <tony(a)linx.net> --- linux-2.6.30-git14/sound/pci/hda/patch_realtek.c.orig 2009-06-20 21:55:53.000000000 +0100 +++ linux-2.6.30-git14/sound/pci/hda/patch_realtek.c 2009-06-21 00:19:53.000000000 +0100 @@ -224,6 +224,7 @@ ALC883_ACER, ALC883_ACER_ASPIRE, ALC888_ACER_ASPIRE_4930G, + ALC888_ACER_ASPIRE_6530G, ALC888_ACER_ASPIRE_8930G, ALC883_MEDION, ALC883_MEDION_MD2, @@ -1471,6 +1472,25 @@ }; /* + * ALC888 Acer Aspire 6530G model + */ + +static struct hda_verb alc888_acer_aspire_6530g_verbs[] = { +/* Bias voltage on for external mic port */ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN | PIN_VREF80}, +/* Enable unsolicited event for HP jack */ + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, +/* Enable speaker output */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, +/* Enable headphone output */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | PIN_HP}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + { } +}; + +/* * ALC889 Acer Aspire 8930G model */ @@ -1544,6 +1564,25 @@ } }; +static struct hda_input_mux alc888_acer_aspire_6530_sources[2] = { + /* Interal mic only available on one ADC */ + { + .num_items = 3, + .items = { + { "Ext Mic", 0x0 }, + { "CD", 0x4 }, + { "Int Mic", 0xb }, + }, + }, + { + .num_items = 2, + .items = { + { "Ext Mic", 0x0 }, + { "CD", 0x4 }, + }, + } +}; + static struct hda_input_mux alc889_capture_sources[3] = { /* Digital mic only available on first "ADC" */ { @@ -8153,6 +8192,19 @@ { } /* end */ }; +static struct snd_kcontrol_new alc888_acer_aspire_6530_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("LFE Playback Volume", 0x0f, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("LFE Playback Switch", 0x0f, 2, HDA_INPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + { } /* end */ +}; + static struct snd_kcontrol_new alc888_lenovo_sky_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -9021,7 +9073,7 @@ SND_PCI_QUIRK(0x1025, 0x015e, "Acer Aspire 6930G", ALC888_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0x0166, "Acer Aspire 6530G", - ALC888_ACER_ASPIRE_4930G), + ALC888_ACER_ASPIRE_6530G), /* default Acer -- disabled as it causes more problems. * model=auto should work fine now */ @@ -9255,6 +9307,24 @@ .unsol_event = alc_automute_amp_unsol_event, .init_hook = alc888_acer_aspire_4930g_init_hook, }, + [ALC888_ACER_ASPIRE_6530G] = { + .mixers = { alc888_acer_aspire_6530_mixer }, + .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs, + alc888_acer_aspire_6530g_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev), + .adc_nids = alc883_adc_nids_rev, + .capsrc_nids = alc883_capsrc_nids_rev, + .dig_out_nid = ALC883_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), + .channel_mode = alc883_3ST_2ch_modes, + .num_mux_defs = + ARRAY_SIZE(alc888_2_capture_sources), + .input_mux = alc888_acer_aspire_6530_sources, + .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc888_acer_aspire_4930g_init_hook, + }, [ALC888_ACER_ASPIRE_8930G] = { .mixers = { alc888_base_mixer, alc883_chmode_mixer },

2 1

[alsa-devel] [PATCH] ASoC: Blackfin: convert internal names from bf52x to bf5xx
by Mike Frysinger 20 Jun '09

20 Jun '09

From: Barry Song <barry.song(a)analog.com> These drivers aren't BF52x specific, so don't use bf52x in the names. Signed-off-by: Barry Song <barry.song(a)analog.com> Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- sound/soc/blackfin/bf5xx-ad73311.c | 16 ++++++++-------- sound/soc/blackfin/bf5xx-ssm2602.c | 16 ++++++++-------- 2 files changed, 16 insertions(+), 16 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c index edfbdc0..9825b71 100644 --- a/sound/soc/blackfin/bf5xx-ad73311.c +++ b/sound/soc/blackfin/bf5xx-ad73311.c @@ -203,23 +203,23 @@ static struct snd_soc_device bf5xx_ad73311_snd_devdata = { .codec_dev = &soc_codec_dev_ad73311, }; -static struct platform_device *bf52x_ad73311_snd_device; +static struct platform_device *bf5xx_ad73311_snd_device; static int __init bf5xx_ad73311_init(void) { int ret; pr_debug("%s enter\n", __func__); - bf52x_ad73311_snd_device = platform_device_alloc("soc-audio", -1); - if (!bf52x_ad73311_snd_device) + bf5xx_ad73311_snd_device = platform_device_alloc("soc-audio", -1); + if (!bf5xx_ad73311_snd_device) return -ENOMEM; - platform_set_drvdata(bf52x_ad73311_snd_device, &bf5xx_ad73311_snd_devdata); - bf5xx_ad73311_snd_devdata.dev = &bf52x_ad73311_snd_device->dev; - ret = platform_device_add(bf52x_ad73311_snd_device); + platform_set_drvdata(bf5xx_ad73311_snd_device, &bf5xx_ad73311_snd_devdata); + bf5xx_ad73311_snd_devdata.dev = &bf5xx_ad73311_snd_device->dev; + ret = platform_device_add(bf5xx_ad73311_snd_device); if (ret) - platform_device_put(bf52x_ad73311_snd_device); + platform_device_put(bf5xx_ad73311_snd_device); return ret; } @@ -227,7 +227,7 @@ static int __init bf5xx_ad73311_init(void) static void __exit bf5xx_ad73311_exit(void) { pr_debug("%s enter\n", __func__); - platform_device_unregister(bf52x_ad73311_snd_device); + platform_device_unregister(bf5xx_ad73311_snd_device); } module_init(bf5xx_ad73311_init); diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c index bc0cdde..3a00fa4 100644 --- a/sound/soc/blackfin/bf5xx-ssm2602.c +++ b/sound/soc/blackfin/bf5xx-ssm2602.c @@ -148,24 +148,24 @@ static struct snd_soc_device bf5xx_ssm2602_snd_devdata = { .codec_data = &bf5xx_ssm2602_setup, }; -static struct platform_device *bf52x_ssm2602_snd_device; +static struct platform_device *bf5xx_ssm2602_snd_device; static int __init bf5xx_ssm2602_init(void) { int ret; pr_debug("%s enter\n", __func__); - bf52x_ssm2602_snd_device = platform_device_alloc("soc-audio", -1); - if (!bf52x_ssm2602_snd_device) + bf5xx_ssm2602_snd_device = platform_device_alloc("soc-audio", -1); + if (!bf5xx_ssm2602_snd_device) return -ENOMEM; - platform_set_drvdata(bf52x_ssm2602_snd_device, + platform_set_drvdata(bf5xx_ssm2602_snd_device, &bf5xx_ssm2602_snd_devdata); - bf5xx_ssm2602_snd_devdata.dev = &bf52x_ssm2602_snd_device->dev; - ret = platform_device_add(bf52x_ssm2602_snd_device); + bf5xx_ssm2602_snd_devdata.dev = &bf5xx_ssm2602_snd_device->dev; + ret = platform_device_add(bf5xx_ssm2602_snd_device); if (ret) - platform_device_put(bf52x_ssm2602_snd_device); + platform_device_put(bf5xx_ssm2602_snd_device); return ret; } @@ -173,7 +173,7 @@ static int __init bf5xx_ssm2602_init(void) static void __exit bf5xx_ssm2602_exit(void) { pr_debug("%s enter\n", __func__); - platform_device_unregister(bf52x_ssm2602_snd_device); + platform_device_unregister(bf5xx_ssm2602_snd_device); } module_init(bf5xx_ssm2602_init); -- 1.6.3.1

3 4

[alsa-devel] [PATCH 1/5] ASoC codec: SSM2602: remove unsupported sample rates
by Mike Frysinger 20 Jun '09

20 Jun '09

From: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- sound/soc/codecs/ssm2602.c | 8 +++----- 1 files changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 87f606c..d6af069 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -497,11 +497,9 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec, return 0; } -#define SSM2602_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ - SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ - SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\ - SNDRV_PCM_RATE_96000) +#define SSM2602_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_32000 |\ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) #define SSM2602_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -- 1.6.3.1

7 40

[alsa-devel] The new mixer logic in PA
by Lennart Poettering 20 Jun '09

20 Jun '09

Heya! During the last weeks I have been reworking the ALSA mixer and device enumeration handling in PA to be more flexible and support more features of usual mixers/cards then it previously did. (Previously we just picked one simple mixer element and used it for volume and muting; 'Master' for output; 'Capture' for input; and the ALSA device name list we used was hardcoded). I just landed all of this in PA, and would like to pass this at least roughly by you folks, hoping for comments. So here's a rough overview of what's going on: The current ALSA mixer APIs have been criticized in the past (including by me) for their complexity and stacking of abstraction layers. Some folks (including me) suggested reworking the ALSA mixer APIs to simplify the APIs. After doing my research I am not so sure anymore that this really would be of much benefit -- at least for software like PA. The reason for that is simply that PA has very specific needs on the mixer interfacing: we are only interested in a subset of the potential functionality: a) we want to do hw volume control when supported for a specific snd_pcm_t with good granularity and range, b) we want mute control, and c) we want to select the output/input port for a snd_pcm_t (i.e. line-in vs. mic/speaker vs. headset). And that's about it. We generally don't want to support any fancier setups even if the hardware would provide it: we don't want simulatenous output to multiple distinct outports, we want no hw mixing, we don't want to support legacy input sources such as CD/Phone/PCSpk..., no "3D" features, not even input feedback. OTOH PA can do stuff other clients cannot do, e.g. extending in softwre the capabilities of the hw if possible and sensible. e.g. we can extend the volume control range/granularity in software beyond the discrete steps the hw provides. It has been suggested to make queryable how mixer elements relate to each other via some mixer API, possibly using the HDA codec graph data. While that would be nice to have I believe that this information would be much more than PA could ever need, and also certainly not help simplicity of an API. I am hence not asking for that, what I do ask for however is that a couple of assumptions I make stay valid. (see below) Given that what PA needs and doesn't need is very different from the needs of almost all other applications needed further abstraction can happen in PA, and doesn't need to be done in ALSA. Which is basically the reason why I decided to do all this mixer abstraction work inside of PA, instead of trying to get that into ALSA. To be able to control more than just one element, PA needs to have some minimal clue how elements are related to each other. ALSA (as mentioned) currently doesn't provide that info, so PA must guess. Hardcoding assumptions about how mixer elements relate to each other of course sucks, which is why I chose to define the list of elements PA should control plus minimal information about how they work together in config files that can be updated more easily and that during runtime. I tried to encode only minimal and the most obvious assumptions. These config files simply list a series of (simple) mixer controls and the way how to initialize them/expose them in PA. Switches can either be bound to always off or on, or be used to follow PA's mute status, or be exposed as distinct options in the UI (which however almost no switch should do). Volumes can either be bound to the minimal values, or to 0dB or be integrated into the volume slider PA exposes. Enumeration values can be exposed in the UI, too, but only if that makes sense. The most interesting part of that is probably the volume handling: for that we go from top to botton in the list of elements in those config files and try to set the volume on the first element marked for it, then divide the actual volume set from our target volume and find the next volume element and apply the remaining volume there, and so on. What's left we then do in attenuate software. All this mixer handling assumes that PA is the only one in charge of volume control. Also, I assume that while I now control a substantial part of the available mixer controls that those I don't control will be initialized properly by "alsactl init". Here's an example how that looks like, for implementing a Mic path (which is the most complex path, due to the input selection stuff) http://git.0pointer.de/?p=pulseaudio.git;a=blob;f=src/modules/alsa/mixer/pa… http://git.0pointer.de/?p=pulseaudio.git;a=blob;f=src/modules/alsa/mixer/pa… http://git.0pointer.de/?p=pulseaudio.git;a=blob;f=src/modules/alsa/mixer/pa… To simplifiy these rules a bit I'd like to see some cleanups in the names used for controls: - Some drivers have a "Input Source Select" enumeration, others a "Input Source" and even others "Capture Source". These could be named the same, can't they? Or is there any difference in their behaviour from an apps PoV? (There's also a "Mic Select" enum, which however probably makes sense to keep seperate given that it the name suggests it is only useful if "Mic" is activated in a capture group) - The options of said enums are really chaotically named. That could really be cleaned up, and would probably be a trivial fix. - Since the various "boost" switches effectively control the volume I'd prefer if they'd be exposed as volume controls with dB info attached. Some Boosts encode the dB info in the name, which sucks. - Some cards have an element "TV Tuner", others "Video". I think that could be merged. I'll prepare a patch for at least some of these suggestions. Of course renaming controls on a big scale will break alsactl restore. Is that a problem? There's one assumption I am currently encoding in the files I am not really sure about. Which is that I assume that the volume level of the surround outputs is influenced by the following volume elements if that they exist: PCM -> Master -> Front, Rear, Surround, Side, Center, LFE while if switched to headphones I assume: PCM -> Master -> Headphone May I assume that? Or is sometimes the routing PCM -> Headphones, and 'Master' does not have influence? There's something else I was wondering, in contrast to the usual code that makes use of the ALSA mixer's dB data I use them to do actual calculations while previously they were just used for "enriching" the information shown on the screen. I wonder if this is going to backfire? How reliable is the dB data actually? [1] For more information on how all this really works you might find this mail I posted to the PA ML interesting: https://tango.0pointer.de/pipermail/pulseaudio-discuss/2009-June/004229.html It goes more into details in things which are relevant for PA folks, probably not so much for the ALSA folks. I'd be very interested in opinions on all of this! Thanks, Lennart [1] To verify and measure the dB data I actually wrote a little tool that can measure the data, which seemed to yield quite correct information on my HDA chip: http://git.0pointer.de/?p=dbmeasure.git;a=tree http://git.0pointer.de/?p=dbmeasure.git;a=blob;f=README -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net http://0pointer.net/lennart/ GnuPG 0x1A015CC4

2 1

[alsa-devel] Why analog loopback of sigmatel devices got removed?
by Maxim Levitsky 20 Jun '09

20 Jun '09

Hi, Due to lack of time, I didn't update kernel on my desktop system. Now I notice that analog loopback go removed. Why? Searching around I found this: http://article.gmane.org/gmane.linux.alsa.devel/59878 * Removed analog loopback mixer items, not mentioned in sigmatel datasheets and reported to break sound support. First they are mention. It took me lot of time to find them. I only have added the loopback items for devices that have this feature in datasheet. Then how it can break sound support? could you explain? I need this feature since I use it with my TV card, and I know more users that use that feature. I don't know about other devices, but I attach the my patch that was included in kernel for this feature. Best regards, Maxim Levitsky

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[alsa-devel] [Bug] ICE1712 analog outputs are unusable
by Peter Lawler 20 Jun '09

20 Jun '09

http://mailman.alsa-project.org/pipermail/alsa-devel/2009-January/014108.ht… https://bugzilla.redhat.com/show_bug.cgi?id=499435 http://mailman.alsa-project.org/pipermail/alsa-devel/2009-June/018616.html The original patch to fix this was declined and thus the issue has been left to fester. I'm available to help test out any new patch proposals etc. Please don't hesitate to contact me etc. Pete.

1 0

[alsa-devel] IDT quirks...where?
by Adam Williamson 20 Jun '09

20 Jun '09

Hi, guys. Following up a Fedora forum quirk report: http://forums.fedoraforum.org/showpost.php?p=1229209&postcount=16 elsewhere in the thread, the reporter is instructed to try the hp-dv5 model parameter, and reports that this works. The alsa-info output shows the subsystem ID as 103c:3621, and the codec as IDT 92HD71B7X. This is one I haven't seen before. I wanted to write up a bug report with a patch to add a quirk for this chipset, but I can't see which of the quirk files it should go into - none of the patch_*.c files seems to cover IDT codecs. What's the story with these? Where do their quirks go? Does current ALSA have one already? (I did try searching through each of the patch_*.c files for 3621 and found nothing, but I may have missed something). Thanks! -- Adam Williamson Fedora QA Community Monkey IRC: adamw | Fedora Talk: adamwill AT fedoraproject DOT org http://www.happyassassin.net

2 2

[alsa-devel] LSB inclusion of ALSA
by Robert Schweikert 19 Jun '09

19 Jun '09

Hi, Currently ALSA is a trial use module for LSB (http://www.linuxfoundation.org/collaborate/workgroups/lsb) 4.0 and it is the intention to make ALSA mandatory with the LSB 4.1 release targeted for Q1 2010. One of the requirements of a mandatory module is the existence of a test suite for the module that covers a good chunk of the interfaces provided by the LSB spec. An initial exploratory look at the existing test cases showed that tests appear to be tied to hardware drivers and are interactive. Is this initial impression correct? _From an LSB testing perspective, interface testing is very important; i.e. I call the interface with the arguments required and I get the expected result back. Further the tests need to run in an automated fashion and produce a certain output (automated processing). The output can be created by retrofitting existing tests or by specifically writing tests within the LSB test framework. IMHO it would be best to have tests that are useful to both, the project and the LSB and in a best case scenario we wouldn't have to start over as the existing tests already cover a good part of the interface. With this as background there are a few questions that have been raised while discussing this during LSB workgroup sessions. 1.) Is there interest from the community side to participate in this effort and accept patches? 2.) Is it reasonable to expect that the existing tests can be used in some way by using a dummy sound device or a sound loop device to verify output? One requirement for LSB testing is that a test has only two states, pass or fail, and a recovery or try again mode is not supported. Further a test must pass or fail. 3.) Would the community be more comfortable if this is an effort that creates new tests separate from the existing HW focused tests? 4.) Is anyone interested in helping with this effort, writing tests, answering questions about ALSA for those not familiar with the interface and or sound in general? Comments, thoughts, etc. are much appreciated. Thanks, Robert -- Robert Schweikert MAY THE SOURCE BE WITH YOU Software Engineer Consultant LINUX rschweikert(a)novell.com 781-464-8147 Novell Making IT Work As One

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Re: [alsa-devel] Laptop (Samsung Q45) speakers unmute when master channel is unmuted even with headphones connected
by Takashi Iwai 19 Jun '09

19 Jun '09

At Fri, 19 Jun 2009 11:56:29 +0100, Sam Morris wrote: > > On Thu, 2009-05-07 at 17:11 +0200, Takashi Iwai wrote: > > At Tue, 05 May 2009 11:58:39 +0100, > > Sam Morris wrote: > > > > > > On Tue, 2009-05-05 at 12:47 +0200, Takashi Iwai wrote: > > > > At Sat, 2 May 2009 13:14:42 +0100, > > > > Sam Morris wrote: > > > > > > > > > > Since Linux 2.6.28 and through to 2.6.30-rc4, the hda-intel driver has had a > > > > > bug on my laptop (a Samsung Q45). > > > > > > > > > > If I connect my headphones, the laptop's internal speakers are muted--fine. But > > > > > if I then mute and unmute the master mixer channel, the speakers come back to > > > > > life, even though the headphones are still connected. > > > > > > > > Could you run alsa-info.sh with --no-upload option, and attach the > > > > generated file? > > > > > > Here you go! > > > > Thanks. > > > > What if you use model=auto option? > > This is fixed in 2.6.30, but only if I use the model=auto option. :) What about with the later version, e.g. Linus git tree? Or you can use sound git tree git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git I think this was already fixed there. Takashi

1 0

Re: [alsa-devel] [PATCH] New ASoC Drivers for ADI AD1938 codec
by Liam Girdwood 19 Jun '09

19 Jun '09

On Fri, 2009-06-19 at 17:28 +0800, Barry Song wrote: > 1. add AD1938 codec driver (codec) > 2. add blackfin SPORT-TDM DAI and PCM driver (platform) > 3. add bf5xx board with AD1938 driver (machine) > Signed-off-by: Barry Song <21cnbao(a)gmail.com> > --- > include/sound/soc-dai.h | 1 + > sound/soc/blackfin/Kconfig | 46 +++- > sound/soc/blackfin/Makefile | 6 + > sound/soc/blackfin/bf5xx-ad1938.c | 178 ++++++++++++ > sound/soc/blackfin/bf5xx-tdm-pcm.c | 330 ++++++++++++++++++++++ > sound/soc/blackfin/bf5xx-tdm-pcm.h | 21 ++ > sound/soc/blackfin/bf5xx-tdm.c | 295 +++++++++++++++++++ > sound/soc/blackfin/bf5xx-tdm.h | 14 + > sound/soc/codecs/Kconfig | 4 + > sound/soc/codecs/Makefile | 2 + > sound/soc/codecs/ad1938.c | 548 ++++++++++++++++++++++++++++++++++++ > sound/soc/codecs/ad1938.h | 68 +++++ > 12 files changed, 1512 insertions(+), 1 deletions(-) > create mode 100644 sound/soc/blackfin/bf5xx-ad1938.c > create mode 100644 sound/soc/blackfin/bf5xx-tdm-pcm.c > create mode 100644 sound/soc/blackfin/bf5xx-tdm-pcm.h > create mode 100644 sound/soc/blackfin/bf5xx-tdm.c > create mode 100644 sound/soc/blackfin/bf5xx-tdm.h > create mode 100644 sound/soc/codecs/ad1938.c > create mode 100644 sound/soc/codecs/ad1938.h > Could you break this patch into smaller chunks. This should make it easier to review and apply. Thanks Liam

1 1

[alsa-devel] RFC: new TLV types for dB range with min/max
by Takashi Iwai 19 Jun '09

19 Jun '09

Hi, while I was checking bug reports regarding USB mixer dB ranges, I found that my old code might not work always properly. In usbmixer.c, /* USB descriptions contain the dB scale in 1/256 dB unit * while ALSA TLV contains in 1/100 dB unit */ scale[2] = (convert_signed_value(cval, cval->min) * 100) / 256; scale[3] = (convert_signed_value(cval, cval->res) * 100) / 256; Since the step size may have a rounding error, this can accumulate to a significant amount at the max value. A simple way to avoid this is to just give min and max values instead of min and step values, and interpolate the middle. For that, I create two new TLV types. Why two? Since the mute bit can't be packed into an integer (the dB range can be both positive and negative values) unlike the step size, I provided here both with and without mute, thus two. The kernel patch is attached below. This and a patch to usb-audio can be found in topic/tlv-minmax branch of sound-unstable GIT tree. git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable-2.6.git topic/tlv-minmax I already wrote and test it with the modified alsa-lib, and the result looks good. If anyone has suggestions or objections to this, please let me know. If this change is OK, I'm going to merge this later for the next, 2.6.32 kernel. thanks, Takashi --- From 085f30654175a91c28d2b66b9ea6cceab627fed0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai <tiwai(a)suse.de> Date: Tue, 16 Jun 2009 13:57:07 +0200 Subject: [PATCH] ALSA: Add new TLV types for dBwith min/max Add new types for TLV dB scale specified with min/max values instead of min/step since the resolution can't match always with the one a device provides. For example, usb audio devices give 1/256 dB resolution while ALSA TLV is based on 1/100 dB resolution. The new min/max types have less problems because the possible rounding error happens only at min/max. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/tlv.h | 14 ++++++++++++++ sound/core/vmaster.c | 8 ++++++-- 2 files changed, 20 insertions(+), 2 deletions(-) diff --git a/include/sound/tlv.h b/include/sound/tlv.h index d136ea2..9fd5b19 100644 --- a/include/sound/tlv.h +++ b/include/sound/tlv.h @@ -35,6 +35,8 @@ #define SNDRV_CTL_TLVT_DB_SCALE 1 /* dB scale */ #define SNDRV_CTL_TLVT_DB_LINEAR 2 /* linear volume */ #define SNDRV_CTL_TLVT_DB_RANGE 3 /* dB range container */ +#define SNDRV_CTL_TLVT_DB_MINMAX 4 /* dB scale with min/max */ +#define SNDRV_CTL_TLVT_DB_MINMAX_MUTE 5 /* dB scale with min/max with mute */ #define TLV_DB_SCALE_ITEM(min, step, mute) \ SNDRV_CTL_TLVT_DB_SCALE, 2 * sizeof(unsigned int), \ @@ -42,6 +44,18 @@ #define DECLARE_TLV_DB_SCALE(name, min, step, mute) \ unsigned int name[] = { TLV_DB_SCALE_ITEM(min, step, mute) } +/* dB scale specified with min/max values instead of step */ +#define TLV_DB_MINMAX_ITEM(min_dB, max_dB) \ + SNDRV_CTL_TLVT_DB_MINMAX, 2 * sizeof(unsigned int), \ + (min_dB), (max_dB) +#define TLV_DB_MINMAX_MUTE_ITEM(min_dB, max_dB) \ + SNDRV_CTL_TLVT_DB_MINMAX_MUTE, 2 * sizeof(unsigned int), \ + (min_dB), (max_dB) +#define DECLARE_TLV_DB_MINMAX(name, min_dB, max_dB) \ + unsigned int name[] = { TLV_DB_MINMAX_ITEM(min_dB, max_dB) } +#define DECLARE_TLV_DB_MINMAX_MUTE(name, min_dB, max_dB) \ + unsigned int name[] = { TLV_DB_MINMAX_MUTE_ITEM(min_dB, max_dB) } + /* linear volume between min_dB and max_dB (.01dB unit) */ #define TLV_DB_LINEAR_ITEM(min_dB, max_dB) \ SNDRV_CTL_TLVT_DB_LINEAR, 2 * sizeof(unsigned int), \ diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c index 257624b..3b9b550 100644 --- a/sound/core/vmaster.c +++ b/sound/core/vmaster.c @@ -353,7 +353,8 @@ static void master_free(struct snd_kcontrol *kcontrol) * * The optional argument @tlv can be used to specify the TLV information * for dB scale of the master control. It should be a single element - * with #SNDRV_CTL_TLVT_DB_SCALE type, and should be the max 0dB. + * with #SNDRV_CTL_TLVT_DB_SCALE, #SNDRV_CTL_TLV_DB_MINMAX or + * #SNDRV_CTL_TLVT_DB_MINMAX_MUTE type, and should be the max 0dB. */ struct snd_kcontrol *snd_ctl_make_virtual_master(char *name, const unsigned int *tlv) @@ -384,7 +385,10 @@ struct snd_kcontrol *snd_ctl_make_virtual_master(char *name, kctl->private_free = master_free; /* additional (constant) TLV read */ - if (tlv && tlv[0] == SNDRV_CTL_TLVT_DB_SCALE) { + if (tlv && + (tlv[0] == SNDRV_CTL_TLVT_DB_SCALE || + tlv[0] == SNDRV_CTL_TLVT_DB_MINMAX || + tlv[0] == SNDRV_CTL_TLVT_DB_MINMAX_MUTE)) { kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ; memcpy(master->tlv, tlv, sizeof(master->tlv)); kctl->tlv.p = master->tlv; -- 1.6.3.2

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Re: [alsa-devel] RFC: PCM extra attributes
by Takashi Iwai 19 Jun '09

19 Jun '09

At Fri, 19 Jun 2009 10:54:15 +0200, Florian Faber wrote: > > Takashi, > > > An important note is that I'm planning to use this framework for > > getting/setting the PCM (surround) channel mapping. > > I don't understand what this means. Who maps what on what and why? It's a mapping of surround channels. Right now, ALSA assumes (usually) the order of channels like front-left, front-rear, surround-left, surround-rear, center, LFE, ... But this really depends on the device. Some (or many) need FL/FR/C/LFE/SR/SR order, for example. And, some devices can allow you even to reassign the channel map. This is likely an additional information, which not all drivers have to provide. Takashi

1 0

[alsa-devel] RFC: "patch" loading for HD-audio driver
by Takashi Iwai 19 Jun '09

19 Jun '09

Hi, another changes I've worked recently on is a "patch"-loading mechanism for HD-audio. Here, a "patch" refers to a sort of firmware file, corresponding "*.INI" file on Windows. As many of you know, HD-audio hardwares have many problems, partly because of crappy broken BIOS and partly because of lack of information that the driver can detect automatically. Until now, these are provided statically in the driver. In the recent version of HD-audio driver, you can override the setup and reconfigure the driver via sysfs. For example, changing the default pin-configurations can be done via sysfs and then reconfigure. However, this procedure can be done only after the driver is set up, not before the driver initialization. So an idea of "patching" the setup before the driver initialization came up. A "patch" file is a text file looks like: ================================================================ [codec] 0x12345678 0xabcd1234 2 [model] auto [pincfg] 0x12 0x411111f0 [verb] 0x20 0x500 0x03 0x20 0x400 0xff [hint] hp_detect = yes ================================================================ This file can be provided in the standard firmware path, typically /lib/firmware, through a module parameter "patch", e.g. options snd-hda-intel patch=my-hda-patch where /lib/firmware/my-hda-patch is the patch file path. The whole patch (I mean program patch) series can be found in topic/hda-patch branch of sound-unstable GIT tree. git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable-2.6.git topic/hda-patch If you are interested, please take a look. Any comments appreciated. This work is intended for 2.6.32 merge as well as other hacks there. thanks, Takashi

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Re: [alsa-devel] kerneloops.org report for the week of June 14 2009
by Arjan van de Ven 19 Jun '09

19 Jun '09

On Thu, 18 Jun 2009 13:08:39 -0400 Bob Copeland <me(a)bobcopeland.com> wrote: > On Sun, Jun 14, 2009 at 8:33 PM, Arjan van de > Ven<arjan(a)infradead.org> wrote: > > Rank 9: minstrel_get_rate (warning) > > Reported 64 times (384 total reports) > > Issue with the ath5k driver > > This warning was last seen in version 2.6.30-rc3, and first > > seen in 2.6.27.12. More info: > > http://www.kerneloops.org/searchweek.php?search=minstrel_get_rate > > I believe this is fixed by http://lkml.org/lkml/2009/6/5/269, now > in pre-2.6.31. Hmm, odd though, we should be seeing reports from > other wireless drivers since it's a general memory corruption > unrelated to ath5k (but not all use the mac80211 rate controllers). > well.. out of the 534 reports or so that I have by now... 525 have ath5k loaded..... it's not 100% but it'd pretty close. -- Arjan van de Ven Intel Open Source Technology Centre For development, discussion and tips for power savings, visit http://www.lesswatts.org

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Re: [alsa-devel] [PATCH v2] alsa-plugins/pulse: Implement 'pause'.
by Takashi Iwai 18 Jun '09

18 Jun '09

At Thu, 18 Jun 2009 14:55:21 +0100 (BST), Troy Moure wrote: > > Just cork or uncork the stream to pause or unpause it. > > Signed-off-by: Troy Moure <twmoure(a)szypr.net> Applied now. Thanks! Takashi > --- > On Wed, 17 Jun 2009, Lennart Poettering wrote: > > > Looks mostly good to me, but: > > > ... > > > > pa_stream_cork() will return a pa_operation object on success. The > > least you need to do is call pa_operation_unref() on it, which you can > > safely do right-away. Otherwise this will leak memory. > > > > Lennart > > Thanks, Lennart, for looking at the first verson I sent. I've added a > call to pa_operation_unref() as you suggested. > > pulse/pcm_pulse.c | 25 +++++++++++++++++++++++++ > 1 files changed, 25 insertions(+), 0 deletions(-) > > diff --git a/pulse/pcm_pulse.c b/pulse/pcm_pulse.c > index db8d1e1..c276839 100644 > --- a/pulse/pcm_pulse.c > +++ b/pulse/pcm_pulse.c > @@ -739,6 +739,30 @@ static int pulse_close(snd_pcm_ioplug_t * io) > return 0; > } > > +static int pulse_pause(snd_pcm_ioplug_t * io, int enable) > +{ > + snd_pcm_pulse_t *pcm = io->private_data; > + int err = 0; > + > + assert (pcm); > + assert (pcm->p); > + > + pa_threaded_mainloop_lock(pcm->p->mainloop); > + > + if (pcm->stream) { > + pa_operation *o; > + o = pa_stream_cork(pcm->stream, enable, NULL, NULL); > + if (o) > + pa_operation_unref(o); > + else > + err = -EIO; > + } > + > + pa_threaded_mainloop_unlock(pcm->p->mainloop); > + > + return err; > +} > + > static const snd_pcm_ioplug_callback_t pulse_playback_callback = { > .start = pulse_start, > .stop = pulse_stop, > @@ -750,6 +774,7 @@ static const snd_pcm_ioplug_callback_t pulse_playback_callback = { > .prepare = pulse_prepare, > .hw_params = pulse_hw_params, > .close = pulse_close, > + .pause = pulse_pause > }; > > > -- > 1.6.2.2 > > > > > > > > > -- > > Lennart Poettering Red Hat, Inc. > > lennart [at] poettering [dot] net > > http://0pointer.net/lennart/ GnuPG 0x1A015CC4 > > >

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Re: [alsa-devel] [PATCH v2] alsa-plugins/pulse: Implement 'pause'.
by Lennart Poettering 18 Jun '09

18 Jun '09

On Thu, 18.06.09 14:55, Troy Moure (twmoure(a)szypr.net) wrote: > > pa_stream_cork() will return a pa_operation object on success. The > > least you need to do is call pa_operation_unref() on it, which you can > > safely do right-away. Otherwise this will leak memory. > > > > Lennart > > Thanks, Lennart, for looking at the first verson I sent. I've added a > call to pa_operation_unref() as you suggested. Looks good to me now. > > pulse/pcm_pulse.c | 25 +++++++++++++++++++++++++ > 1 files changed, 25 insertions(+), 0 deletions(-) > > diff --git a/pulse/pcm_pulse.c b/pulse/pcm_pulse.c > index db8d1e1..c276839 100644 > --- a/pulse/pcm_pulse.c > +++ b/pulse/pcm_pulse.c > @@ -739,6 +739,30 @@ static int pulse_close(snd_pcm_ioplug_t * io) > return 0; > } > > +static int pulse_pause(snd_pcm_ioplug_t * io, int enable) > +{ > + snd_pcm_pulse_t *pcm = io->private_data; > + int err = 0; > + > + assert (pcm); > + assert (pcm->p); > + > + pa_threaded_mainloop_lock(pcm->p->mainloop); > + > + if (pcm->stream) { > + pa_operation *o; > + o = pa_stream_cork(pcm->stream, enable, NULL, NULL); > + if (o) > + pa_operation_unref(o); > + else > + err = -EIO; > + } > + > + pa_threaded_mainloop_unlock(pcm->p->mainloop); > + > + return err; > +} > + > static const snd_pcm_ioplug_callback_t pulse_playback_callback = { > .start = pulse_start, > .stop = pulse_stop, > @@ -750,6 +774,7 @@ static const snd_pcm_ioplug_callback_t pulse_playback_callback = { > .prepare = pulse_prepare, > .hw_params = pulse_hw_params, > .close = pulse_close, > + .pause = pulse_pause > }; > > Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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Re: [alsa-devel] Regarding internal mic on Aspire 6935G
by Karthik Ramgopal 18 Jun '09

18 Jun '09

Hi, How to check if it works in stereo or mono mode? I remember reading hector's entry on his blog that on account of some wierd sum/difference problem he had duplicated the + signal across both outputs so it was actually mono? Regards, Karthik

2 1

[alsa-devel] preventing SNDRV_PCM_STATE_XRUN
by Sergey Lapin 18 Jun '09

18 Jun '09

Hi, all! I have uncontrollable data stream which I need to play through kernel via ALSA snd_pcm_lib_write call. Everything is fine, but once a few seconds I get SNDRV_PCM_STATE_XRUN on stream and thus, -EPIPE. Sound glitches are minor, but annoying. Is it possible to somehow prevent this from happenning? Is it possible to distinguish if my next chunk of data will fit? Thanks a lot, S.

1 0

[alsa-devel] Regarding internal mic on Aspire 6935G
by Karthik Ramgopal 18 Jun '09

18 Jun '09

Hi, Sorry for the confusion. My internal mic does work, but wierdly not in Gnome Sound recorder. Skype detects it just fine and I am able to talk without making any changes to the code. So, I guess 8930G and 6935G are same after all. Thanks and Regards, Karthik PS: Please also add the PCI ID's of 6920G (0x1025 0146) and 8920G (0x1025 0145) under the same quirk. They also use ALC889 chipset with the same pinconfig.

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Re: [alsa-devel] trace PCM data from usb bus to alsa lib
by Takashi Iwai 18 Jun '09

18 Jun '09

At Wed, 17 Jun 2009 16:10:26 +0800, Zero Luo wrote: > > Dear Takashi, > > Data from usb dongle is 16K*16 bit PCM data. And what is the format and channels you are using in alsa-lib? alsa-lib itself can convert formats and channels automatically if needed. Takashi > 2009/6/17, Takashi Iwai <tiwai(a)suse.de>: > > At Wed, 17 Jun 2009 07:25:25 +0000 (UTC), > ZeroCluo wrote: > > > > Hi > > > > I am trying to trace the Audio data from usb bus. > > in kernel: > > dump data in retire_capture_urb() ,found data format is like: > > AABB 0000 CCDD 0000 ; > > > > but in alsa library , > > dump data in snd_pcm_mmap_begin(),the "0000" is cut,only "AABB CCDD", > > > > Do you know when and where those "0000" is cut? > > Are you using the native format that the hardware runs? > > Takashi > >

2 3

Re: [alsa-devel] snd-ctxfi oops
by The Source 18 Jun '09

18 Jun '09

On 15.06.2009 17:01, Takashi Iwai wrote: > At Mon, 15 Jun 2009 15:00:03 +0200, > I wrote: > >> At Mon, 15 Jun 2009 16:59:53 +0400, >> The Source wrote: >> >>> On 15.06.2009 16:56, Takashi Iwai wrote: >>> >>>> At Sat, 13 Jun 2009 10:07:49 +0200, >>>> I wrote: >>>> >>>> >>>>> >>>>> >>>>>> Tested the patch more. Really no oopses. Instead of them I get rare >>>>>> system lockups. >>>>>> >>>>>> >>>>> In which situation, exactly? >>>>> Could you check whether it happens with use_system_timer=1 module >>>>> option, too? >>>>> >>>>> >>>> Or try the very latest alsa-driver-snapshot (not daily tarball). >>>> I fixed a deadlock issue in cttimer.c now. >>>> >>>> >>>> thanks, >>>> >>>> Takashi >>>> >>>> >>>> >>> Should I apply that patch to this snapshot? >>> >> Which patch do you mean? >> Just grab the latest alsa-driver-snapshot.tar.gz from >> ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/ >> > Also, check alsa-driver/alsa-kernel/HEAD whether the commit > 8dca419721d188bfee5f19fad45275856c619a5c > ALSA: ctxfi - Fix deadlock with xfi-timer > is included. > > > Takashi > > Tested the driver and it's fine so far. No craches with any application including Doom 3, which caused system lockups often with earlier versions of the driver.

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[alsa-devel] [PATCH] alsa-plugins/pulse: Implement 'pause'.
by Troy Moure 18 Jun '09

18 Jun '09

Just cork or uncork the stream to pause or unpause it. Signed-off-by: Troy Moure <twmoure(a)szypr.net> --- I'm not very familiar with alsa or pulseaudio, so this might not be right. If it is, feel free to apply it, though. It *does* fix my problem, which was that pause and unpause didn't work in 'cmus' after I upgraded to Fedora 11. Since the pulse plug-in didn't support snd_pcm_pause, pausing in cmus would just cause cmus to stop writing to the buffer, leading to buffer underruns in pulseaudio. With this patch, pausing in cmus works perfectly for me. If this should be fixed another way, please let me know what it is! Thanks. pulse/pcm_pulse.c | 20 ++++++++++++++++++++ 1 files changed, 20 insertions(+), 0 deletions(-) diff --git a/pulse/pcm_pulse.c b/pulse/pcm_pulse.c index db8d1e1..c322884 100644 --- a/pulse/pcm_pulse.c +++ b/pulse/pcm_pulse.c @@ -739,6 +739,25 @@ static int pulse_close(snd_pcm_ioplug_t * io) return 0; } +static int pulse_pause(snd_pcm_ioplug_t * io, int enable) +{ + snd_pcm_pulse_t *pcm = io->private_data; + int err = 0; + + assert (pcm); + assert (pcm->p); + + pa_threaded_mainloop_lock(pcm->p->mainloop); + + if (pcm->stream) + if (!pa_stream_cork(pcm->stream, enable, NULL, NULL)) + err = -EIO; + + pa_threaded_mainloop_unlock(pcm->p->mainloop); + + return err; +} + static const snd_pcm_ioplug_callback_t pulse_playback_callback = { .start = pulse_start, .stop = pulse_stop, @@ -750,6 +769,7 @@ static const snd_pcm_ioplug_callback_t pulse_playback_callback = { .prepare = pulse_prepare, .hw_params = pulse_hw_params, .close = pulse_close, + .pause = pulse_pause }; -- 1.6.2.2

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[alsa-devel] ICE1712 softdevices fixes
by Stefan Kost 17 Jun '09

17 Jun '09

hi, > The following patches fixes the default device as they appear for > ICE1712-based cards. Without this cards, 'aplay -L' will show a series of > devices like surround40 and surround71 that are not meaningful for this card, > and a front device that would not work. > > Diego E. 'Flameeyes' Pettenò (2): Remove definition for surround sound > devices on ICE1712. Make the front:idx device work for ICE1712. thanks for making those. this was exactly what I needed to get pulseaudio working with my card again. I also agree that even though that my card (Hoontech DSP 24) has 8/10 physical in/outs, those are essentially mono channels. The surround mapping seem to be more harm that use to me. If a musician indeed wires 6 3.5 jack outs somehow to his 5.1 amp, then its probably okay to uncomment such mappings. So what about making diegos patches the default? Stefan

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[alsa-devel] [PATCH] ALSA: snd_usb_caiaq: fix legacy input streaming
by Daniel Mack 17 Jun '09

17 Jun '09

Seems that nobody recently tried the input on the very first supported sound card model, RK2. This patch fixes the byte offset to make it running again. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- sound/usb/caiaq/audio.c | 5 +++-- sound/usb/caiaq/device.c | 2 +- 2 files changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index b144513..8f9b60c 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -199,8 +199,9 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) dev->period_out_count[index] = BYTES_PER_SAMPLE + 1; dev->audio_out_buf_pos[index] = BYTES_PER_SAMPLE + 1; } else { - dev->period_in_count[index] = BYTES_PER_SAMPLE; - dev->audio_in_buf_pos[index] = BYTES_PER_SAMPLE; + int in_pos = (dev->spec.data_alignment == 2) ? 0 : 2; + dev->period_in_count[index] = BYTES_PER_SAMPLE + in_pos; + dev->audio_in_buf_pos[index] = BYTES_PER_SAMPLE + in_pos; } if (dev->streaming) diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index 2240624..0e5db71 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -35,7 +35,7 @@ #include "input.h" MODULE_AUTHOR("Daniel Mack <daniel(a)caiaq.de>"); -MODULE_DESCRIPTION("caiaq USB audio, version 1.3.16"); +MODULE_DESCRIPTION("caiaq USB audio, version 1.3.17"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, RigKontrol3}," -- 1.6.3.1

2 1

[alsa-devel] sound device is buzy to run pcm.c
by Guilherme Raymo Longo 17 Jun '09

17 Jun '09

I am getting this problem when running the pcm.c demo program at http://alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html This is the execution of the program: guilherme@lap:/home/guilherme/Experimentos# ./PCM Playback device is plughw:0,0 Stream parameters are 44100Hz, S16_LE, 1 channels sine wave rate is 0 0.0000Hz Using transfer method: write Playback open error: Device or resource busy I tried to check if the device is really buzy: root@lap:/home/guilherme/Experimentos# lsof |grep snd lsof: WARNING: can't stat() fuse.gvfs-fuse-daemon file system */ home/guilherme/*.gvfs Output information may be incomplete. esd 5433 guilherme mem CHR 116,16 6947 /dev/snd/pcmC0D0p esd 5433 guilherme 5r CHR 116,33 6787 /dev/snd/timer esd 5433 guilherme 6w CHR 116,16 6947 /dev/snd/pcmC0D0p esd 5433 guilherme 7r CHR 116,0 6987 /dev/snd/controlC0 and it seems to be ok. That's my card root@lap:/home/guilherme/Experimentos# aplay -l **** List of PLAYBACK Hardware Devices **** card 0: SB [HDA ATI SB], device 0: ALC883 Analog [ALC883 Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 Is there a way to free the device or smth is wrong with my code?? I am using slackware 12.2 kernel 2.6 Great Regads;

3 2

[alsa-devel] OSS support
by The Source 17 Jun '09

17 Jun '09

Is oss emulation still supported in latest alsa-driver snapshot? I installed that driver in Fedora 11 on both my static machine (X-Fi platinum fatality champion) and laptop (HDA Intel) and I don't have /dev/dsp or /dev/mixer. Should I compile the driver with some specific flag?

3 3

[alsa-devel] [PATCH 3/3] ASoC: pxa-ssp.c handle the I2S and LEFT_J cases.
by Daniel Ribeiro 17 Jun '09

17 Jun '09

Handles I2S and LEFT_J cases, if the user does _not_ call set_tdm_slot we assume that the codec is I2S compliant, and that we can use frame_width = sample_width * channels. If the user calls set_tdm_slot then we assume that the wire is _not_ networked, and that the user wants to set a non-standard frame format instead, in this case we set the frame duration (the LRCLK falling edge) to be half of the wire frame width. eg. * standard 2*[16|24|32|..] bits I2S/LEFT_J doesn't need to call set_tdm_slot * to use 2*16bits I2S/LEFTJ on 64bits frames, with the LRCLK falling edge at the 32nd bitclk (+1 for I2S) needs to set_tdm_slot(5, 5, 4, 16) (4 slots of 16 bits, first and third RX/TX slots active). This should work even if pxa is slave of SFRM. Signed-off-by: Daniel Ribeiro <drwyrm(a)gmail.com> --- arch/arm/mach-pxa/include/mach/regs-ssp.h | 14 ++-- sound/soc/pxa/pxa-ssp.c | 121 +++++++++++++--------------- 2 files changed, 63 insertions(+), 72 deletions(-) diff --git a/arch/arm/mach-pxa/include/mach/regs-ssp.h b/arch/arm/mach-pxa/include/mach/regs-ssp.h index 6a2ed35..060e23b 100644 --- a/arch/arm/mach-pxa/include/mach/regs-ssp.h +++ b/arch/arm/mach-pxa/include/mach/regs-ssp.h @@ -108,21 +108,21 @@ #define SSSR_TINT (1 << 19) /* Receiver Time-out Interrupt */ #define SSSR_PINT (1 << 18) /* Peripheral Trailing Byte Interrupt */ -#if defined(CONFIG_PXA3xx) -#define SSPSP_EDMYSTOP(x) ((x) << 28) /* Extended Dummy Stop */ -#define SSPSP_EDMYSTRT(x) ((x) << 26) /* Extended Dummy Start */ -#endif - #define SSPSP_FSRT (1 << 25) /* Frame Sync Relative Timing */ -#define SSPSP_DMYSTOP(x) ((x) << 23) /* Dummy Stop */ #define SSPSP_SFRMWDTH(x) ((x) << 16) /* Serial Frame Width */ #define SSPSP_SFRMDLY(x) ((x) << 9) /* Serial Frame Delay */ -#define SSPSP_DMYSTRT(x) ((x) << 7) /* Dummy Start */ #define SSPSP_STRTDLY(x) ((x) << 4) /* Start Delay */ #define SSPSP_ETDS (1 << 3) /* End of Transfer data State */ #define SSPSP_SFRMP (1 << 2) /* Serial Frame Polarity */ #define SSPSP_SCMODE(x) ((x) << 0) /* Serial Bit Rate Clock Mode */ +/* NOTE: PXA3xx extends the bit number of dummy start and stop, the macros + * below are compatible with PXA25x/27x as long as the parameter is within + * the correct limits, driver code has to take care of this. + */ +#define SSPSP_DMYSTRT(x) ((((x) & 3) << 7) | ((((x) >> 2) & 3) << 26)) +#define SSPSP_DMYSTOP(x) ((((x) & 3) << 23) | ((((x) >> 2) & 7) << 28)) + #define SSACD_SCDB (1 << 3) /* SSPSYSCLK Divider Bypass */ #define SSACD_ACPS(x) ((x) << 4) /* Audio clock PLL select */ #define SSACD_ACDS(x) ((x) << 0) /* Audio clock divider select */ diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index d60492e..aac85fe 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -179,21 +179,6 @@ static void ssp_set_scr(struct ssp_device *ssp, u32 div) ssp_write_reg(ssp, SSCR0, sscr0); } -/** - * ssp_get_clkdiv - get SSP clock divider - */ -static u32 ssp_get_scr(struct ssp_device *ssp) -{ - u32 sscr0 = ssp_read_reg(ssp, SSCR0); - u32 div; - - if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) - div = ((sscr0 >> 8) & 0xff) * 2 + 2; - else - div = ((sscr0 >> 8) & 0xfff) + 1; - return div; -} - /* * Set the SSP ports SYSCLK. */ @@ -487,17 +472,14 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, } switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_I2S: - sscr0 |= SSCR0_PSP; - sscr1 |= SSCR1_RWOT | SSCR1_TRAIL; - /* See hw_params() */ - break; - case SND_SOC_DAIFMT_DSP_A: sspsp |= SSPSP_FSRT; case SND_SOC_DAIFMT_DSP_B: + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_I2S: sscr0 |= SSCR0_PSP; sscr1 |= SSCR1_TRAIL | SSCR1_RWOT; + /* See hw_params() for I2S and LEFT_J */ break; default: @@ -561,6 +543,59 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, sscr0 |= SSCR0_FPCKE; #endif + sspsp = ssp_read_reg(ssp, SSPSP); + switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + /* + * I2S and LEFT_J are stereo only, we have to send data for + * both channels. + */ + if (chn == 1) + frame_width *= 2; + + /* + * If the user did not use network mode, we assume the codec + * is I2S compliant. + */ + if (frame_width > 0) { + sspsp |= SSPSP_SFRMWDTH(frame_width / 2); + sspsp |= SSPSP_FSRT; + } else { + /* + * Otherwise we assume that it is a single TDM slot, and + * the user is abusing set_tdm_slot to support an + * out of spec codec. + */ + int slots = ((sscr0 & SSCR0_SlotsPerFrm(8)) >> 24) + 1; + + /* PXA2XX doesn't support DMYSTOP > 3 */ + if (slot_width != (width * 2) && !cpu_is_pxa3xx()) + return -EINVAL; + + sspsp |= SSPSP_DMYSTRT(1); + sspsp |= SSPSP_DMYSTOP( + (slot_width * slots) / 2 - width - 1); + sspsp |= SSPSP_SFRMWDTH((slot_width * slots) / 2); + } + break; + + case SND_SOC_DAIFMT_LEFT_J: + if (chn == 1) + frame_width *= 2; + + if (frame_width > 0) { + sspsp |= SSPSP_SFRMWDTH(frame_width / 2); + } else { + int slots = ((sscr0 & SSCR0_SlotsPerFrm(8)) >> 24) + 1; + + sspsp |= SSPSP_SFRMWDTH((slot_width * slots) / 2); + } + break; + default: + break; + } + ssp_write_reg(ssp, SSPSP, sspsp); + if (frame_width > 0) { /* Not using network mode */ if (frame_width > 16) @@ -598,50 +633,6 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, ssp_write_reg(ssp, SSCR0, sscr0); - switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_I2S: - sspsp = ssp_read_reg(ssp, SSPSP); - - if ((ssp_get_scr(ssp) == 4) && (width == 16)) { - /* This is a special case where the bitclk is 64fs - * and we're not dealing with 2*32 bits of audio - * samples. - * - * The SSP values used for that are all found out by - * trying and failing a lot; some of the registers - * needed for that mode are only available on PXA3xx. - */ - -#ifdef CONFIG_PXA3xx - if (!cpu_is_pxa3xx()) - return -EINVAL; - - sspsp |= SSPSP_SFRMWDTH(width * 2); - sspsp |= SSPSP_SFRMDLY(width * 4); - sspsp |= SSPSP_EDMYSTOP(3); - sspsp |= SSPSP_DMYSTOP(3); - sspsp |= SSPSP_DMYSTRT(1); -#else - return -EINVAL; -#endif - } else { - /* The frame width is the width the LRCLK is - * asserted for; the delay is expressed in - * half cycle units. We need the extra cycle - * because the data starts clocking out one BCLK - * after LRCLK changes polarity. - */ - sspsp |= SSPSP_SFRMWDTH(width + 1); - sspsp |= SSPSP_SFRMDLY((width + 1) * 2); - sspsp |= SSPSP_DMYSTRT(1); - } - - ssp_write_reg(ssp, SSPSP, sspsp); - break; - default: - break; - } - dump_registers(ssp); return 0; -- tg: (119677b..) asoc/leftj-and-i2s (depends on: asoc/ssp-internals) -- Daniel Ribeiro

3 3

[alsa-devel] trace PCM data from usb bus to alsa lib
by ZeroCluo 17 Jun '09

17 Jun '09

Hi I am trying to trace the Audio data from usb bus. in kernel: dump data in retire_capture_urb() ,found data format is like: AABB 0000 CCDD 0000 ; but in alsa library , dump data in snd_pcm_mmap_begin(),the "0000" is cut,only "AABB CCDD", Do you know when and where those "0000" is cut? btw,use mmap method map dma_area to user space; and use snd_pcm_readi to get audio data; Thank you very very much. Best Regards, Zero

2 1

Re: [alsa-devel] No jack sense on Intel 82801I / IDT 92HD71B7X in a HP dv4
by Wael Nasreddine (a.k.a eMxyzptlk) 17 Jun '09

17 Jun '09

All the information you need are here -> http://www.alsa-project.org/db/?f=6b588ea53af5cf2794fb8b4239311722809b3c34 2009/6/15 Jean-Pierre André <jean-pierre.andre(a)wanadoo.fr>: > Hi Wael, > > Wael Nasreddine (a.k.a eMxyzptlk) wrote: >> >> I don't know if this concerns my situation ( >> http://mailman.alsa-project.org/pipermail/alsa-devel/2009-June/018262.html >> ) or not.. >> >> headphones work for me, front speaker work but LFE doesn't and the >> Irq22 stays on 0 too >> >> # cat /proc/interrupts | grep '22:' >> 22: 0 0 IO-APIC-fasteoi uhci_hcd:usb6, HDA Intel >> >> Takashi told me to play with hda-analyser and hda-verb though I don't >> what to do with hda-verb and I don't if what hda-analyser showing me >> is the current situation or the desired one.. >> >> Thanks > > So it looks like this a general situation on this hardware. > > If you get sound, at least from the front speakers, > without the IRQ22 being triggered, what are your > kernel and alsa versions, your boot parameters and > alsa (modprobe) parameters ? > > I have no knowledge about the hda-verb, etc. > so I am not to help you much in this area. > > Regards > > Jean-Pierre > > > > -- Wael Nasreddine Blog : http://wael.nasreddine.com E-mail : wael.nasreddine(a)gmail.com gTalk : wael.nasreddine(a)gmail.com Tel : +33.6.32.94.70.13 Skype : eMxyzptlk Twitter : @eMxyzptlk Sabayon Linux Chief Development Officer - http://www.sabayonlinux.org PGP: 1024D/C8DD18A2 06F6 1622 4BC8 4CEB D724 DE12 5565 3945 C8DD 18A2 .: An infinite number of monkeys typing into GNU emacs, would never make a good program. (L. Torvalds 1995) :.

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[alsa-devel] [PATCH 1/1] ALC889A chip identification
by Torben Schulz 17 Jun '09

17 Jun '09

With this patch the ALC889A chip will be identified correctly. The functions patch_alc882() and patch_alc883() have been altered to set the stream names correctly. Without this patch 'aplay -l' would identify an ALC889A chip as ALC885 for example. Apart from that a kernel info message will be printed if ALC885 or ALC889A chips are handled by the wrong patch function. Signed-off by: Torben Schulz <public(a)letorbi.de> --- alsa-driver/alsa-kernel/pci/hda/patch_realtek.c 2009-05-14 18:23:17.000000000 +0200 +++ alsa-driver-mod/alsa-kernel/pci/hda/patch_realtek.c 2009-05-15 01:08:16.000000000 +0200 @@ -7338,8 +7338,17 @@ setup_preset(spec, &alc882_presets[board_config]); if (codec->vendor_id == 0x10ec0885) { - spec->stream_name_analog = "ALC885 Analog"; - spec->stream_name_digital = "ALC885 Digital"; + if ((codec->revision_id == 0x100101) || + (codec->revision_id == 0x100103)) { + spec->stream_name_analog = "ALC889A Analog"; + spec->stream_name_digital = "ALC889A Digital"; + printk(KERN_INFO + "hda_codec: ALC889A handled by patch_alc882()," + " should be handled by patch_alc883()\n"); + } else { + spec->stream_name_analog = "ALC885 Analog"; + spec->stream_name_digital = "ALC885 Digital"; + } } else { spec->stream_name_analog = "ALC882 Analog"; spec->stream_name_digital = "ALC882 Digital"; @@ -9231,6 +9240,27 @@ setup_preset(spec, &alc883_presets[board_config]); switch (codec->vendor_id) { + case 0x10ec0885: + if ((codec->revision_id == 0x100101) || + (codec->revision_id == 0x100103)) { + spec->stream_name_analog = "ALC889A Analog"; + spec->stream_name_digital = "ALC889A Digital"; + if (!spec->num_adc_nids) { + spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids); + spec->adc_nids = alc883_adc_nids; + } + if (!spec->capsrc_nids) + spec->capsrc_nids = alc883_capsrc_nids; + spec->capture_style = CAPT_MIX; /* matrix-style capture */ + spec->init_amp = ALC_INIT_DEFAULT; /* always initialize */ + } else { + spec->stream_name_analog = "ALC885 Analog"; + spec->stream_name_digital = "ALC885 Digital"; + printk(KERN_INFO + "hda_codec: ALC885 handled by patch_alc883()," + " should be handled by patch_alc882()\n"); + } + break; case 0x10ec0888: if (codec->revision_id == 0x100101) { spec->stream_name_analog = "ALC1200 Analog";

3 5

[alsa-devel] Playback open error: Device or resource busy
by Guilherme 17 Jun '09

17 Jun '09

Mates... I am running a simple app that I found called pcm.c and I can't get through with that. It creates and send a simple sine wave to the speakers! When I run the app I get the following error: guilherme@lap:/home/guilherme/Experimentos# ./PCM Playback device is plughw:0,0 Stream parameters are 44100Hz, S16_LE, 1 channels sine wave rate is 0 0.0000Hz Using transfer method: write Playback open error: Device or resource busy Every application I start playbacks the sound without a problem. Is there any special conf I should do to free the Device? Here is the code: #include <stdio.h> #include <stdlib.h> #include <string.h> #include <sched.h> #include <errno.h> #include <getopt.h> #include <alsa/asoundlib.h> #include <sys/time.h> #include <math.h> static char *device = "plughw:0,0"; //playback device aplay -l static snd_pcm_format_t format = SND_PCM_FORMAT_S16; //sample format unsigned 16 bit endian static unsigned int rate = 44100; //stream rate static unsigned int channels = 1; //contagem dos canais static unsigned int buffer_time = 500000; //ring buffer length on us static unsigned int period_time = 100000; //period time in us static double freq = 400; // frequencia da senóide em Hz static int verbose = 0; //verbose flag static int resample = 1; //abilitar alsa-lib resampling static int period_event = 0; // produz um event pool a cada periodo static snd_pcm_sframes_t buffer_size; //quantidade de buffers frames static snd_pcm_sframes_t period_size; //quantidade de periods frames static snd_output_t *output = NULL; //ALSA usa esse ponteiro para lidar com output objects static void generate_sine(const snd_pcm_channel_area_t *areas, snd_pcm_uframes_t offset, int count, double *_phase) { static double max_phase = 2. * M_PI; double phase = *_phase; double step = max_phase*freq / (double)rate; double res; unsigned char *samples[channels], *tmp; int steps[channels]; unsigned int chn, byte; union { int i; unsigned char c[4]; } ires; unsigned int maxval = (1 << (snd_pcm_format_width(format) - 1)) - 1; int bps = snd_pcm_format_width(format) / 8; //bits per sample /*verificar e preparar o conteúdo das areas*/ for (chn = 0; chn < channels; chn++) { if ((areas[chn].first % 8) != 0) { printf("areas[%i].first == %i, abortando...\n", chn, areas[chn].first); exit(EXIT_FAILURE); } //pega endereço e offset da area samples[chn] = /*(signed short *)*/(((unsigned char *)areas[chn].addr) + (areas[chn].first / 8)); if ((areas[chn].step % 16) != 0) { printf("areas[%i].step == %i, aborting...\n", chn, areas[chn].step); exit(EXIT_FAILURE); } steps[chn] = areas[chn].step / 8; samples[chn] += offset * steps[chn]; } //completa os channels da area while(count-- > 0) { res = sin(phase) * maxval; ires.i = res; tmp = ires.c; for (chn = 0; chn < channels; chn++) { for (byte = 0; byte < (unsigned int)bps; byte++) *(samples[chn] + byte) = tmp[byte]; samples[chn] += steps[chn]; } phase += step; if (phase >= max_phase) phase -= max_phase; } *_phase = phase; } static int set_hwparams(snd_pcm_t *handle, snd_pcm_hw_params_t *params, snd_pcm_access_t access) { unsigned int rrate; snd_pcm_uframes_t size; int err, dir; /*escolha todos os parametros*/ err = snd_pcm_hw_params_any(handle, params); if (err < 0) { printf("Broken configuration for playback: no configuration available: %s\n", snd_strerror(err)); return err; } /*set hardware resampling*/ err = snd_pcm_hw_params_set_rate_resample(handle, params, resample); if (err < 0) { printf("Resampling setup failed for playback: %s\n", snd_strerror(err)); return err; } /* set the interleaved read/write format */ err = snd_pcm_hw_params_set_access(handle, params, access); if (err < 0) { printf("Access type not available for playback: %s\n", snd_strerror(err)); return err; } /* set the sample format SND_PCM_FORMAT_S16 */ err = snd_pcm_hw_params_set_format(handle, params, format); if (err < 0) { printf("Sample format not available for playback: %s\n", snd_strerror(err)); return err; } /* set the count of channels 1 */ err = snd_pcm_hw_params_set_channels(handle, params, channels); if (err < 0) { printf("Channels count (%i) not available for playbacks: %s\n", channels, snd_strerror(err)); return err; } /* set the stream rate */ rrate = rate; err = snd_pcm_hw_params_set_rate_near(handle, params, &rrate, 0); if (err < 0) { printf("Rate %iHz not available for playback: %s\n", rate, snd_strerror(err)); return err; } if (rrate != rate) { printf("Rate doesn't match (requested %iHz, get %iHz)\n", rate, err); return -EINVAL; } /* set the buffer time */ err = snd_pcm_hw_params_set_buffer_time_near(handle, params, &buffer_time, &dir); if (err < 0) { printf("Unable to set buffer time %i for playback: %s\n", buffer_time, snd_strerror(err)); return err; } err = snd_pcm_hw_params_get_buffer_size(params, &size); if (err < 0) { printf("Unable to get buffer size for playback: %s\n", snd_strerror(err)); return err; } buffer_size = size; /* set the period time */ err = snd_pcm_hw_params_set_period_time_near(handle, params, &period_time, &dir); if (err < 0) { printf("Unable to set period time %i for playback: %s\n", period_time, snd_strerror(err)); return err; } err = snd_pcm_hw_params_get_period_size(params, &size, &dir); if (err < 0) { printf("Unable to get period size for playback: %s\n", snd_strerror(err)); return err; } period_size = size; /* write the parameters to device */ err = snd_pcm_hw_params(handle, params); if (err < 0) { printf("Unable to set hw params for playback: %s\n", snd_strerror(err)); return err; } return 0; } static int set_swparams(snd_pcm_t *handle, snd_pcm_sw_params_t *swparams) { int err; /*pegar os parametros atuais swparams*/ err = snd_pcm_sw_params_current(handle, swparams); if (err < 0) { printf("Unable to determine current swparams for playback: %s\n", snd_strerror(err)); return err; } /*Começa a transferência quando o buffer está quase cheio*/ /*(buffer_size / avail_min) * avail_min */ err = snd_pcm_sw_params_set_avail_min(handle, swparams, (buffer_size / period_size) * period_size); if (err < 0) { printf("Unable to set start threshold mode for playback: %s\n", snd_strerror(err)); return err; } /* allow the transfer when at least period_size samples can be processed */ /* or disable this mechanism when period event is enabled (aka interrupt like style processing) */ err = snd_pcm_sw_params_set_avail_min(handle, swparams, period_event ? buffer_size : period_size); if (err < 0) { printf("Unable to set avail min for playback: %s\n", snd_strerror(err)); return err; } /* enable period events when requested */ if (period_event) { err = snd_pcm_sw_params_set_period_event(handle, swparams, 1); if (err < 0) { printf("Unable to set period event: %s\n", snd_strerror(err)); return err; } } /* write the parameters to the playback device */ err = snd_pcm_sw_params(handle, swparams); if (err < 0) { printf("Unable to set sw params for playback: %s\n", snd_strerror(err)); return err; } return 0; } /* Tenta recuperar estados de underrun e suspend */ static int xrun_recovery(snd_pcm_t *handle, int err) { if (verbose) printf("stream recovery\n"); if (err == -EPIPE) { //under-run*/ err = snd_pcm_prepare(handle); if (err < 0) printf("Can't recovery from underrun, prepare failed: %s\n", snd_strerror(err)); return 0; } else if (err == -ESTRPIPE) { while ((err = snd_pcm_resume(handle)) == -EAGAIN) sleep(1); /*espera pela release flag*/ if (err < 0) { err = snd_pcm_prepare(handle); if (err < 0) printf("Can't recovery from suspend, prepare failed: %s\n", snd_strerror(err)); } return 0; } return err; } /* Metodo de transferência *WRITE-ONLY* */ static int write_loop(snd_pcm_t *handle, signed short *samples, snd_pcm_channel_area_t *areas) { double phase = 0; signed short *ptr; int err, cptr; while(1) { generate_sine(areas, 0, period_size, &phase); ptr = samples; cptr = period_size; while (cptr > 0) { err = snd_pcm_writei(handle, ptr, cptr); if (err == -EAGAIN) continue; if (err < 0) { if (xrun_recovery(handle, err) < 0) { printf("Write error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); }//if ptr += err * channels; cptr -= err; }//if }//if }//while } /* * Método de transferência - write and wait por espaço no buffer */ static int wait_for_poll(snd_pcm_t *handle, struct pollfd *ufds, unsigned int count) { unsigned short revents; while (1) { poll(ufds, count, -1); snd_pcm_poll_descriptors_revents(handle, ufds, count, &revents); if (revents & POLLERR) return -EIO; if (revents & POLLOUT) return 0; } } static int write_and_poll_loop(snd_pcm_t *handle, signed short *samples, snd_pcm_channel_area_t *areas) { struct pollfd *ufds; double phase = 0; signed short *ptr; int err, count, cptr, init; count = snd_pcm_poll_descriptors_count (handle); if (count <= 0) { printf("Invalid poll descriptors count\n"); return count; } ufds = malloc(sizeof(struct pollfd) * count); if (ufds == NULL) { printf("No enough memory\n"); return -ENOMEM; } if ((err = snd_pcm_poll_descriptors(handle, ufds, count)) < 0) { printf("Unable to obtain poll descriptors for playback: %s\n", snd_strerror(err)); return err; } init = 1; while (1) { if (!init) { err = wait_for_poll(handle, ufds, count); if (err < 0) { if (snd_pcm_state(handle) == SND_PCM_STATE_XRUN || snd_pcm_state(handle) == SND_PCM_STATE_SUSPENDED) { err = snd_pcm_state(handle) == SND_PCM_STATE_XRUN ? -EPIPE : -ESTRPIPE; if (xrun_recovery(handle, err) < 0) { printf("Write error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } init = 1; } else { printf("Wait for poll failed\n"); return err; } } } generate_sine(areas, 0, period_size, &phase); ptr = samples; cptr = period_size; while (cptr > 0) { err = snd_pcm_writei(handle, ptr, cptr); if (err < 0) { if (xrun_recovery(handle, err) < 0) { printf("Write error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } init = 1; break; /* skip one period */ } if (snd_pcm_state(handle) == SND_PCM_STATE_RUNNING) init = 0; ptr += err * channels; cptr -= err; if (cptr == 0) break; /* it is possible, that the initial buffer cannot store */ /* all data from the last period, so wait awhile */ err = wait_for_poll(handle, ufds, count); if (err < 0) { if (snd_pcm_state(handle) == SND_PCM_STATE_XRUN || snd_pcm_state(handle) == SND_PCM_STATE_SUSPENDED) { err = snd_pcm_state(handle) == SND_PCM_STATE_XRUN ? -EPIPE : -ESTRPIPE; if (xrun_recovery(handle, err) < 0) { printf("Write error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } init = 1; } else { printf("Wait for poll failed\n"); return err; } } } } } /* * Metodo de transferencia - Notificação assincrona */ struct async_private_data { signed short *samples; snd_pcm_channel_area_t *areas; double phase; }; static void async_callback(snd_async_handler_t *ahandler) { snd_pcm_t *handle = snd_async_handler_get_pcm(ahandler); struct async_private_data *data = snd_async_handler_get_callback_private(ahandler); signed short *samples = data->samples; snd_pcm_channel_area_t *areas = data->areas; snd_pcm_sframes_t avail; int err; avail = snd_pcm_avail_update(handle); while (avail >= period_size) { generate_sine(areas, 0, period_size, &data->phase); err = snd_pcm_writei(handle, samples, period_size); if (err < 0) { printf("Write error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } if (err != period_size) { printf("Write error: written %i expected %li\n", err, period_size); exit(EXIT_FAILURE); } avail = snd_pcm_avail_update(handle); } } static int async_loop(snd_pcm_t *handle, signed short *samples, snd_pcm_channel_area_t *areas) { struct async_private_data data; snd_async_handler_t *ahandler; int err, count; data.samples = samples; data.areas = areas; data.phase = 0; err = snd_async_add_pcm_handler(&ahandler, handle, async_callback, &data); if (err < 0) { printf("Unable to register async handler\n"); exit(EXIT_FAILURE); } for (count = 0; count < 2; count++) { generate_sine(areas, 0, period_size, &data.phase); err = snd_pcm_writei(handle, samples, period_size); if (err < 0) { printf("Initial write error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } if (err != period_size) { printf("Initial write error: written %i expected %li\n", err, period_size); exit(EXIT_FAILURE); } } if (snd_pcm_state(handle) == SND_PCM_STATE_PREPARED) { err = snd_pcm_start(handle); if (err < 0) { printf("Start error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } } /* because all other work is done in the signal handler, suspend the process */ while (1) { sleep(1); } } /* * Transfer method - asynchronous notification + direct write */ static void async_direct_callback(snd_async_handler_t *ahandler) { snd_pcm_t *handle = snd_async_handler_get_pcm(ahandler); struct async_private_data *data = snd_async_handler_get_callback_private(ahandler); const snd_pcm_channel_area_t *my_areas; snd_pcm_uframes_t offset, frames, size; snd_pcm_sframes_t avail, commitres; snd_pcm_state_t state; int first = 0, err; while (1) { state = snd_pcm_state(handle); if (state == SND_PCM_STATE_XRUN) { err = xrun_recovery(handle, -EPIPE); if (err < 0) { printf("XRUN recovery failed: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } first = 1; } else if (state == SND_PCM_STATE_SUSPENDED) { err = xrun_recovery(handle, -ESTRPIPE); if (err < 0) { printf("SUSPEND recovery failed: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } } avail = snd_pcm_avail_update(handle); if (avail < 0) { err = xrun_recovery(handle, avail); if (err < 0) { printf("avail update failed: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } first = 1; continue; } if (avail < period_size) { if (first) { first = 0; err = snd_pcm_start(handle); if (err < 0) { printf("Start error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } } else { break; } continue; } size = period_size; while (size > 0) { frames = size; err = snd_pcm_mmap_begin(handle, &my_areas, &offset, &frames); if (err < 0) { if ((err = xrun_recovery(handle, err)) < 0) { printf("MMAP begin avail error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } first = 1; } generate_sine(my_areas, offset, frames, &data->phase); commitres = snd_pcm_mmap_commit(handle, offset, frames); if (commitres < 0 || (snd_pcm_uframes_t)commitres != frames) { if ((err = xrun_recovery(handle, commitres >= 0 ? -EPIPE : commitres)) < 0) { printf("MMAP commit error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } first = 1; } size -= frames; } } } static int async_direct_loop(snd_pcm_t *handle, signed short *samples ATTRIBUTE_UNUSED, snd_pcm_channel_area_t *areas ATTRIBUTE_UNUSED) { struct async_private_data data; snd_async_handler_t *ahandler; const snd_pcm_channel_area_t *my_areas; snd_pcm_uframes_t offset, frames, size; snd_pcm_sframes_t commitres; int err, count; data.samples = NULL; /* we do not require the global sample area for direct write */ data.areas = NULL; /* we do not require the global areas for direct write */ data.phase = 0; err = snd_async_add_pcm_handler(&ahandler, handle, async_direct_callback, &data); if (err < 0) { printf("Unable to register async handler\n"); exit(EXIT_FAILURE); } for (count = 0; count < 2; count++) { size = period_size; while (size > 0) { frames = size; err = snd_pcm_mmap_begin(handle, &my_areas, &offset, &frames); if (err < 0) { if ((err = xrun_recovery(handle, err)) < 0) { printf("MMAP begin avail error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } } generate_sine(my_areas, offset, frames, &data.phase); commitres = snd_pcm_mmap_commit(handle, offset, frames); if (commitres < 0 || (snd_pcm_uframes_t)commitres != frames) { if ((err = xrun_recovery(handle, commitres >= 0 ? -EPIPE : commitres)) < 0) { printf("MMAP commit error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } } size -= frames; } } err = snd_pcm_start(handle); if (err < 0) { printf("Start error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } /* because all other work is done in the signal handler, suspend the process */ while (1) { sleep(1); } } /* * Transfer method - direct write only */ static int direct_loop(snd_pcm_t *handle, signed short *samples ATTRIBUTE_UNUSED, snd_pcm_channel_area_t *areas ATTRIBUTE_UNUSED) { double phase = 0; const snd_pcm_channel_area_t *my_areas; snd_pcm_uframes_t offset, frames, size; snd_pcm_sframes_t avail, commitres; snd_pcm_state_t state; int err, first = 1; while (1) { state = snd_pcm_state(handle); if (state == SND_PCM_STATE_XRUN) { err = xrun_recovery(handle, -EPIPE); if (err < 0) { printf("XRUN recovery failed: %s\n", snd_strerror(err)); return err; } first = 1; } else if (state == SND_PCM_STATE_SUSPENDED) { err = xrun_recovery(handle, -ESTRPIPE); if (err < 0) { printf("SUSPEND recovery failed: %s\n", snd_strerror(err)); return err; } } avail = snd_pcm_avail_update(handle); if (avail < 0) { err = xrun_recovery(handle, avail); if (err < 0) { printf("avail update failed: %s\n", snd_strerror(err)); return err; } first = 1; continue; } if (avail < period_size) { if (first) { first = 0; err = snd_pcm_start(handle); if (err < 0) { printf("Start error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } } else { err = snd_pcm_wait(handle, -1); if (err < 0) { if ((err = xrun_recovery(handle, err)) < 0) { printf("snd_pcm_wait error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } first = 1; } } continue; } size = period_size; while (size > 0) { frames = size; err = snd_pcm_mmap_begin(handle, &my_areas, &offset, &frames); if (err < 0) { if ((err = xrun_recovery(handle, err)) < 0) { printf("MMAP begin avail error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } first = 1; } generate_sine(my_areas, offset, frames, &phase); commitres = snd_pcm_mmap_commit(handle, offset, frames); if (commitres < 0 || (snd_pcm_uframes_t)commitres != frames) { if ((err = xrun_recovery(handle, commitres >= 0 ? -EPIPE : commitres)) < 0) { printf("MMAP commit error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } first = 1; } size -= frames; } } } /* * Transfer method - direct write only using mmap_write functions */ static int direct_write_loop(snd_pcm_t *handle, signed short *samples, snd_pcm_channel_area_t *areas) { double phase = 0; signed short *ptr; int err, cptr; while (1) { generate_sine(areas, 0, period_size, &phase); ptr = samples; cptr = period_size; while (cptr > 0) { err = snd_pcm_mmap_writei(handle, ptr, cptr); if (err == -EAGAIN) continue; if (err < 0) { if (xrun_recovery(handle, err) < 0) { printf("Write error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } break; /* skip one period */ } ptr += err * channels; cptr -= err; } } } struct transfer_method { const char *name; snd_pcm_access_t access; int (*transfer_loop)(snd_pcm_t *handle, signed short *samples, snd_pcm_channel_area_t *areas); }; static struct transfer_method transfer_methods[] = { { "write", SND_PCM_ACCESS_RW_INTERLEAVED, write_loop }, { "write_and_poll", SND_PCM_ACCESS_RW_INTERLEAVED, write_and_poll_loop }, { "async", SND_PCM_ACCESS_RW_INTERLEAVED, async_loop }, { "async_direct", SND_PCM_ACCESS_MMAP_INTERLEAVED, async_direct_loop }, { "direct_interleaved", SND_PCM_ACCESS_MMAP_INTERLEAVED, direct_loop }, { "direct_noninterleaved", SND_PCM_ACCESS_MMAP_NONINTERLEAVED, direct_loop }, { "direct_write", SND_PCM_ACCESS_MMAP_INTERLEAVED, direct_write_loop }, { NULL, SND_PCM_ACCESS_RW_INTERLEAVED, NULL } }; static void help(void) { int k; printf( "Usage: pcm [OPTION]... [FILE]...\n" "-h,--help help\n" "-D,--device playback device\n" "-r,--rate stream rate in Hz\n" "-c,--channels count of channels in stream\n" "-f,--frequency sine wave frequency in Hz\n" "-b,--buffer ring buffer size in us\n" "-p,--period period size in us\n" "-m,--method transfer method\n" "-o,--format sample format\n" "-v,--verbose show the PCM setup parameters\n" "-n,--noresample do not resample\n" "-e,--pevent enable poll event after each period\n" "\n"); printf("Recognized sample formats are:"); for (k = 0; k < SND_PCM_FORMAT_LAST; ++k) { const char *s = snd_pcm_format_name(k); if (s) printf(" %s", s); } printf("\n"); printf("Recognized transfer methods are:"); for (k = 0; transfer_methods[k].name; k++) printf(" %s", transfer_methods[k].name); printf("\n"); } int main (int argc, char *argv[]) { struct option long_option[] = { {"help", 0, NULL, 'h'}, {"device", 1, NULL, 'd'}, {"rate", 1, NULL, 'r'}, {"channels", 1, NULL, 'c'}, {"frequency", 1, NULL, 'f'}, {"buffer", 1, NULL, 'b'}, {"period", 1, NULL, 'p'}, {"method", 1, NULL, 'm'}, {"format", 1, NULL, 'o'}, {"verbose", 1, NULL, 'v'}, {"noresample", 1, NULL, 'n'}, {"pevent", 1, NULL, 'e'}, {NULL, 0, NULL, 0}, }; snd_pcm_t *handle; //ponteiro para pcm handle int err, morehelp; snd_pcm_hw_params_t *hwparams; //parametros de hardware snd_pcm_sw_params_t *swparams; //parametros de software int method = 0; signed short *samples; unsigned int chn; snd_pcm_channel_area_t *areas; //snd_pcm_hw_params_alloca(&hwparams); //snd_pcm_sw_params_alloca(&swparams); morehelp = 0; while (1) { int c; if ((c = getopt_long(argc, argv, "hd:r:c:f:b:p:m:o:vne", long_option, NULL)) < 0) break; switch(c) { case 'h': morehelp++; break; case 'd': device = strdup(optarg); break; case 'r': rate = atoi(optarg); rate = rate < 4000 ? 4000 : rate; rate = rate > 196000 ? 19600 : rate; break; case 'c': channels = atoi(optarg); channels = channels < 1 ? 1 : channels; channels = channels > 1024 ? 1024 : channels; break; case 'f': freq = atoi(optarg); freq = freq < 50 ? 50 : freq; freq = freq > 5000 ? 5000 : freq; break; case 'b': buffer_time = atoi(optarg); buffer_time = buffer_time < 1000 ? 1000 : buffer_time; buffer_time = buffer_time > 1000000 ? 1000000 : buffer_time; break; case 'p': period_time = atoi(optarg); period_time = period_time < 1000 ? 1000 : period_time; period_time = period_time > 1000000 ? 1000000 : period_time; break; case 'm': for (method = 0; transfer_methods[method].name; method++) if (!strcasecmp(transfer_methods[method].name, optarg)) break; if (transfer_methods[method].name == NULL) method = 0; break; case 'o': for (format = 0; format < SND_PCM_FORMAT_LAST; format++) { const char *format_name = snd_pcm_format_name(format); if (format_name) if (!strcasecmp(format_name, optarg)) break; } if (format == SND_PCM_FORMAT_LAST) format = SND_PCM_FORMAT_S16; break; case 'v': verbose = 1; break; case 'n': resample = 0; break; case 'e': period_event = 1; break; }//switch }//while if(morehelp) { help(); return 0; } //cria um novo output objeto err = snd_output_stdio_attach(&output, &stdout, 0); if (err < 0) { printf("Output failed: %s\n", snd_strerror(err)); return 0; } printf("Playback device is %s\n", device); printf("Stream parameters are %iHz, %s, %i channels\n", rate, snd_pcm_format_name(format), channels); printf("sine wave rate is %i %.4fHz\n", freq); printf("Using transfer method: %s\n", transfer_methods[method].name); if ((err = snd_pcm_open(&handle, device, SND_PCM_STREAM_PLAYBACK, 0)) < 0) { printf("Playback open error: %s\n", snd_strerror(err)); return 0; } if ((err = set_hwparams(handle, hwparams, transfer_methods[method].access)) < 0) { printf("Setting of hwparams failed: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } if ((err = set_swparams(handle, swparams)) < 0) { printf("Setting of swparams failed: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } if (verbose > 0) snd_pcm_dump(handle, output); samples = malloc((period_size * channels * snd_pcm_format_physical_width(format)) / 8); if (samples == NULL) { printf("No enough memory\n"); exit(EXIT_FAILURE); } areas = calloc(channels, sizeof(snd_pcm_channel_area_t)); if (areas == NULL) { printf("No enough memory\n"); exit(EXIT_FAILURE); } for (chn = 0; chn < channels; chn++) { areas[chn].addr = samples; areas[chn].first = chn * snd_pcm_format_physical_width(format); areas[chn].step = channels * snd_pcm_format_physical_width(format); } err = transfer_methods[method].transfer_loop(handle, samples, areas); if (err < 0) printf("Transfer failed: %s\n", snd_strerror(err)); free(areas); free(samples); snd_pcm_close(handle); return 0; } Thanks in advance; -- Tks! ------------------- Guilherme Longo Dept. Eng. da Computação Unaerp Linux User - #484927 *Before Asking http://www.istf.com.br/?page=perguntas !- I'd rather die on my feet than live on my knees -!

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[alsa-devel] kerneloops.org report for the week of June 14 2009
by Arjan van de Ven 16 Jun '09

16 Jun '09

Hi, First, a word of thanks to the kernel.org team for helping to host the kerneloops.org database on a real machine rather than a small virtualized host. Performance and continuity will be a lot better in the new setup! (as a result of the transition and hosting issues on the original server, there was a downtime of two days, meaning that the total number of reports is lower than typical) New sightings this report include the XFI driver, the i915 gem code and the DMAR code. This week, a total of 4026 oopses and warnings have been reported for kernels 2.6.29 and later. Reports for kernels earlier than 2.6.29 have been omitted from this report. Per file statistics 830 drivers/gpu/drm/i915/i915_gem_tiling.c 470 drivers/net/wireless/iwlwifi/iwl-sta.c 303 kernel/time.c 276 drivers/parport/procfs.c 269 include/linux/slob_def.h 187 include/linux/hrtimer.h 138 arch/x86/kernel/cpu/mtrr/generic.c 120 drivers/pci/dmar.c 65 net/mac80211/rc80211_minstrel.c 62 drivers/net/r8169.c 37 net/ipv4/tcp.c 34 fs/cifs/cifssmb.c Rank 1: i915_gem_set_tiling (warning) Reported 830 times (3700 total reports) [gem] Failure in the tiling code This warning was last seen in version 2.6.30, and first seen in 2.6.29-rc2. More info: http://www.kerneloops.org/searchweek.php?search=i915_gem_set_tiling Rank 2: iwl_set_dynamic_key (warning) Reported 470 times (13682 total reports) "no space for new kew" message. This warning was last seen in version 2.6.30, and first seen in 2.6.27. More info: http://www.kerneloops.org/searchweek.php?search=iwl_set_dynamic_key Rank 3: getnstimeofday (warning) Reported 309 times (2446 total reports) [suspend resume] getnstimeofday() is called before timekeeping is resumed This warning was last seen in version 2.6.29.4, and first seen in 2.6.24. More info: http://www.kerneloops.org/searchweek.php?search=getnstimeofday Rank 4: ct_vm_map (warning) - sleeping in atomic context Reported 285 times (960 total reports) Bug in the creative XFI driver (backported to Fedora) This warning was last seen in version 2.6.29.4, and first seen in 2.6.29-rc4-git1. More info: http://www.kerneloops.org/searchweek.php?search=ct_vm_map Rank 5: parport_device_proc_unregister (warning) Reported 278 times (1810 total reports) Alan has a fix for this on LKML This warning was last seen in version 2.6.29.4, and first seen in 2.6.27. More info: http://www.kerneloops.org/searchweek.php?search=parport_device_proc_unregis… Rank 6: hres_timers_resume (warning) Reported 188 times (1024 total reports) [suspend resume] hres_timers_resume() is incorrectly called with interrupts on This warning was last seen in version 2.6.30-rc6, and first seen in 2.6.24.7. More info: http://www.kerneloops.org/searchweek.php?search=hres_timers_resume Rank 7: generic_get_mtrr (warning) Reported 139 times (1009 total reports) BIOS bug where the MTRRs are not set up correctly This warning was last seen in version 2.6.30, and first seen in 2.6.25.3. More info: http://www.kerneloops.org/searchweek.php?search=generic_get_mtrr Rank 8: dmar_table_init (warning) Reported 121 times (507 total reports) BIOS bug exposed via a WARN_ON This warning was last seen in version 2.6.30-rc8, and first seen in 2.6.29. More info: http://www.kerneloops.org/searchweek.php?search=dmar_table_init Rank 9: minstrel_get_rate (warning) Reported 64 times (384 total reports) Issue with the ath5k driver This warning was last seen in version 2.6.30-rc3, and first seen in 2.6.27.12. More info: http://www.kerneloops.org/searchweek.php?search=minstrel_get_rate Rank 10: dev_watchdog(r8169) (warning) Reported 62 times (9116 total reports) This warning was last seen in version 2.6.30-rc7, and first seen in 2.6.26.6. More info: http://www.kerneloops.org/searchweek.php?search=dev_watchdog(r8169) Rank 11: nv_tx_done_optimized (warning) Reported 61 times (2974 total reports) the NV ethernet driver is getting caught with the new DMA API checks This warning was last seen in version 2.6.30, and first seen in 2.6.29-rc3-git7. More info: http://www.kerneloops.org/searchweek.php?search=nv_tx_done_optimized Rank 12: tcp_recvmsg (warning) Reported 37 times (3797 total reports) fixed in 2.6.30 (commit 775273131810caa41dfc7f9e552ea5d8508caf40) This warning was last seen in version 2.6.30, and first seen in 2.6.25. More info: http://www.kerneloops.org/searchweek.php?search=tcp_recvmsg -- Arjan van de Ven Intel Open Source Technology Centre For development, discussion and tips for power savings, visit http://www.lesswatts.org

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Re: [alsa-devel] ALSA 1.0.20 Speex PCM Plugin
by Takashi Iwai 16 Jun '09

16 Jun '09

At Mon, 8 Jun 2009 15:56:38 -0400, Robert Krakora wrote: > > Takashi: > > I applied your Speex echo cancellation patch. I had to modify it > since the 'buf_out' buffer was not being malloc'd. Attached are the > files affected by your patch. The echo cancellation seems to work > pretty well. However, turning on 'denoise' seems to cause audio to > drop in and out at a very periodic rate. Do you have any example of > how to set up all of the parameters? A tutorial or something? Thanks > again. No idea, I just wrote it up blindly :) Can the preprocessor be dropped, possibly, like the patch below? Takashi --- diff --git a/doc/speexdsp.txt b/doc/speexdsp.txt index 875fc19..5b5e5a0 100644 --- a/doc/speexdsp.txt +++ b/doc/speexdsp.txt @@ -12,7 +12,7 @@ using libspeex DSP API. You can use the plugin with the plugin type Then record like - % arecord -fdat -c1 -Dplug:speex foo.wav + % arecord -fdat -c1 -Dplug:my_pcm foo.wav so that you'll get 48kHz mono stream with the denoising effect. @@ -44,6 +44,15 @@ The following parameters can be set optionally: A boolean value to enable/disable dereverb function. Default is no. +* echo + + A boolean value to enable/disable echo-cancellation function. + Default is no. + +* filter_length + + Number of samples of echo to cancel. As default it's 256. + For example, you can enable agc like pcm.my_pcm { diff --git a/speex/pcm_speex.c b/speex/pcm_speex.c index 7bb9213..757a400 100644 --- a/speex/pcm_speex.c +++ b/speex/pcm_speex.c @@ -1,5 +1,5 @@ /* - * Speex preprocess plugin + * Speex DSP plugin * * Copyright (c) 2009 by Takashi Iwai <tiwai(a)suse.de> * @@ -21,12 +21,15 @@ #include <alsa/asoundlib.h> #include <alsa/pcm_external.h> #include <speex/speex_preprocess.h> +#include <speex/speex_echo.h> -/* preprocessing parameters */ +/* DSP parameters */ struct spx_parms { int frames; int denoise; int agc; + int echo; + int filter_length; float agc_level; int dereverb; float dereverb_decay; @@ -38,7 +41,9 @@ typedef struct { struct spx_parms parms; /* instance and intermedate buffer */ SpeexPreprocessState *state; + SpeexEchoState *echo_state; short *buf; + short *outbuf; /* running states */ unsigned int filled; unsigned int processed; @@ -64,6 +69,18 @@ spx_transfer(snd_pcm_extplug_t *ext, short *src = area_addr(src_areas, src_offset); short *dst = area_addr(dst_areas, dst_offset); unsigned int count = size; + short *databuf; + + if (!spx->state && !spx->echo_state) { + /* no DSP processing */ + memcpy(dst, src, count * 2); + return size; + } + + if (spx->echo_state) + databuf = spx->outbuf; + else + databuf = spx->buf; while (count > 0) { unsigned int chunk; @@ -72,14 +89,20 @@ spx_transfer(snd_pcm_extplug_t *ext, else chunk = count; if (spx->processed) - memcpy(dst, spx->buf + spx->filled, chunk * 2); + memcpy(dst, databuf + spx->filled, chunk * 2); else memset(dst, 0, chunk * 2); dst += chunk; memcpy(spx->buf + spx->filled, src, chunk * 2); spx->filled += chunk; if (spx->filled == spx->parms.frames) { - speex_preprocess_run(spx->state, spx->buf); + if (spx->echo_state) + speex_echo_capture(spx->echo_state, spx->buf, + spx->outbuf); + if (spx->state) + speex_preprocess_run(spx->state, databuf); + if (spx->echo_state) + speex_echo_playback(spx->echo_state, databuf); spx->processed = 1; spx->filled = 0; } @@ -94,6 +117,9 @@ static int spx_init(snd_pcm_extplug_t *ext) { snd_pcm_speex_t *spx = (snd_pcm_speex_t *)ext; + spx->filled = 0; + spx->processed = 0; + if (!spx->buf) { spx->buf = malloc(spx->parms.frames * 2); if (!spx->buf) @@ -101,12 +127,43 @@ static int spx_init(snd_pcm_extplug_t *ext) } memset(spx->buf, 0, spx->parms.frames * 2); - if (spx->state) + if (!spx->outbuf) { + spx->outbuf = malloc(spx->parms.frames * 2); + if (!spx->outbuf) + return -ENOMEM; + } + memset(spx->outbuf, 0, spx->parms.frames * 2); + + if (spx->state) { speex_preprocess_state_destroy(spx->state); + spx->state = NULL; + } + if (spx->echo_state) { + speex_echo_state_destroy(spx->echo_state); + spx->echo_state = NULL; + } + + if (spx->parms.echo) { + spx->echo_state = speex_echo_state_init(spx->parms.frames, + spx->parms.filter_length); + if (!spx->echo_state) + return -EIO; + speex_echo_ctl(spx->echo_state, SPEEX_ECHO_SET_SAMPLING_RATE, + &spx->ext.rate); + } + + /* no preprocessor? */ + if (!spx->parms.denoise && !spx->parms.agc && !spx->parms.dereverb) + return 0; + spx->state = speex_preprocess_state_init(spx->parms.frames, spx->ext.rate); if (!spx->state) return -EIO; + if (spx->echo_state) + speex_preprocess_ctl(spx->state, + SPEEX_PREPROCESS_SET_ECHO_STATE, + spx->echo_state); speex_preprocess_ctl(spx->state, SPEEX_PREPROCESS_SET_DENOISE, &spx->parms.denoise); @@ -120,18 +177,18 @@ static int spx_init(snd_pcm_extplug_t *ext) &spx->parms.dereverb_decay); speex_preprocess_ctl(spx->state, SPEEX_PREPROCESS_SET_DEREVERB_LEVEL, &spx->parms.dereverb_level); - - spx->filled = 0; - spx->processed = 0; return 0; } static int spx_close(snd_pcm_extplug_t *ext) { snd_pcm_speex_t *spx = (snd_pcm_speex_t *)ext; + free(spx->outbuf); free(spx->buf); if (spx->state) speex_preprocess_state_destroy(spx->state); + if (spx->echo_state) + speex_echo_state_destroy(spx->echo_state); return 0; } @@ -205,6 +262,8 @@ SND_PCM_PLUGIN_DEFINE_FUNC(speex) .dereverb = 0, .dereverb_decay = 0, .dereverb_level = 0, + .echo = 0, + .filter_length = 256, }; snd_config_for_each(i, next, conf) { @@ -242,6 +301,13 @@ SND_PCM_PLUGIN_DEFINE_FUNC(speex) &parms.dereverb_level); if (err) goto ok; + err = get_bool_parm(n, id, "echo", &parms.echo); + if (err) + goto ok; + err = get_int_parm(n, id, "filter_length", + &parms.filter_length); + if (err) + goto ok; SNDERR("Unknown field %s", id); err = -EINVAL; ok: @@ -259,7 +325,7 @@ SND_PCM_PLUGIN_DEFINE_FUNC(speex) return -ENOMEM; spx->ext.version = SND_PCM_EXTPLUG_VERSION; - spx->ext.name = "Speex Denoise Plugin"; + spx->ext.name = "Speex DSP Plugin"; spx->ext.callback = &speex_callback; spx->ext.private_data = spx; spx->parms = parms;

2 2

[alsa-devel] [PATCH 2/3] ASoC: pxa-ssp.c, Automatically set TDM when needed
by Daniel Ribeiro 16 Jun '09

16 Jun '09

* Automatically sets TDM mode for frame_width larger than 32 bits, if the user doesn't setup the TDM slots with set_tdm_slot(). * Reset SSCR0_EDSS and SSCR0_DSS on pxa_ssp_set_dai_fmt. * Makes SSCR0_MOD optional. * Clears SSCR1_RWOT case SSCR0_MOD is set. Signed-off-by: Daniel Ribeiro <drwyrm(a)gmail.com> --- sound/soc/pxa/pxa-ssp.c | 92 +++++++++++++++++++++++++++-------------------- 1 files changed, 53 insertions(+), 39 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 5b9ed64..d60492e 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -452,8 +452,8 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, } /* reset port settings */ - sscr0 = ssp_read_reg(ssp, SSCR0) & - (SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); + sscr0 = ssp_read_reg(ssp, SSCR0) & (SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | + SSCR0_ACS | SSCR0_DSS | SSCR0_EDSS); sscr1 = SSCR1_RxTresh(8) | SSCR1_TxTresh(7); sspsp = 0; @@ -496,7 +496,7 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, case SND_SOC_DAIFMT_DSP_A: sspsp |= SSPSP_FSRT; case SND_SOC_DAIFMT_DSP_B: - sscr0 |= SSCR0_MOD | SSCR0_PSP; + sscr0 |= SSCR0_PSP; sscr1 |= SSCR1_TRAIL | SSCR1_RWOT; break; @@ -532,48 +532,70 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, struct ssp_priv *priv = cpu_dai->private_data; struct ssp_device *ssp = priv->dev.ssp; int chn = params_channels(params); - u32 sscr0; - u32 sspsp; + u32 sscr0, sscr1, sspsp; int width = snd_pcm_format_physical_width(params_format(params)); - int ttsa = ssp_read_reg(ssp, SSTSA) & 0xf; + int slot_width, frame_width = 0; + + /* check if the user explicitly set a slot_width */ + sscr0 = ssp_read_reg(ssp, SSCR0); + + if (sscr0 & (SSCR0_EDSS | SSCR0_DSS)) + slot_width = (sscr0 & SSCR0_DSS) + + (sscr0 & SSCR0_EDSS ? 17 : 1); + else + frame_width = slot_width = width * chn; /* generate correct DMA params */ if (cpu_dai->dma_data) kfree(cpu_dai->dma_data); - /* Network mode with one active slot (ttsa == 1) can be used - * to force 16-bit frame width on the wire (for S16_LE), even - * with two channels. Use 16-bit DMA transfers for this case. - */ - cpu_dai->dma_data = ssp_get_dma_params(ssp, - ((chn == 2) && (ttsa != 1)) || (width == 32), + cpu_dai->dma_data = ssp_get_dma_params(ssp, slot_width > 16, substream->stream == SNDRV_PCM_STREAM_PLAYBACK); /* we can only change the settings if the port is not in use */ - if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) + if (sscr0 & SSCR0_SSE) return 0; - /* clear selected SSP bits */ - sscr0 = ssp_read_reg(ssp, SSCR0) & ~(SSCR0_DSS | SSCR0_EDSS); - ssp_write_reg(ssp, SSCR0, sscr0); - - /* bit size */ - sscr0 = ssp_read_reg(ssp, SSCR0); - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: #ifdef CONFIG_PXA3xx - if (cpu_is_pxa3xx()) - sscr0 |= SSCR0_FPCKE; + if (slot_width == 16 && cpu_is_pxa3xx()) + sscr0 |= SSCR0_FPCKE; #endif - sscr0 |= SSCR0_DataSize(16); - break; - case SNDRV_PCM_FORMAT_S24_LE: - sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(8)); - break; - case SNDRV_PCM_FORMAT_S32_LE: - sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(16)); - break; + + if (frame_width > 0) { + /* Not using network mode */ + if (frame_width > 16) + sscr0 |= SSCR0_EDSS | SSCR0_DataSize(frame_width - 16); + else + sscr0 |= SSCR0_DataSize(frame_width); + + if (frame_width > 32) { + /* + * Network mode is needed to support this frame width. + * We assume the wire is not networked and setup a + * fake network mode here. Use as many slots as needed + * each with 32 bits. + */ + int slots = frame_width / 32; + + sscr0 |= SSCR0_MOD; + sscr0 |= SSCR0_SlotsPerFrm(slots); + + /* + * Set active slots. Only set an active TX slot + * if we are going to use it. + */ + ssp_write_reg(ssp, SSRSA, slots - 1); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + ssp_write_reg(ssp, SSTSA, slots - 1); + } } + + /* If SSCR0_MOD is set we can't use SSCR1_RWOT */ + if (sscr0 & SSCR0_MOD) { + sscr1 = ssp_read_reg(ssp, SSCR1); + ssp_write_reg(ssp, SSCR1, sscr1 & ~SSCR1_RWOT); + } + ssp_write_reg(ssp, SSCR0, sscr0); switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { @@ -620,14 +642,6 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, break; } - /* When we use a network mode, we always require TDM slots - * - complain loudly and fail if they've not been set up yet. - */ - if ((sscr0 & SSCR0_MOD) && !ttsa) { - dev_err(&ssp->pdev->dev, "No TDM timeslot configured\n"); - return -EINVAL; - } - dump_registers(ssp); return 0; -- tg: (b977873..) asoc/ssp-internals (depends on: asoc/set_tdm_slot) -- Daniel Ribeiro

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[alsa-devel] [PATCH 1/3] ASoC: magician: fix PXA SSP clock polarity
by Philipp Zabel 16 Jun '09

16 Jun '09

Follow-up fix needed since "ASoC: pxa-ssp.c fix clock/frame invert". Signed-off-by: Philipp Zabel <philipp.zabel(a)gmail.com> --- sound/soc/pxa/magician.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index a04ddc3..fa0e7e6 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -183,7 +183,7 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, /* set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | - SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBS_CFS); + SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; -- 1.6.3.1

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[alsa-devel] PROBLEM WITH EMAIL
by Guilherme 15 Jun '09

15 Jun '09

MATES... My emails are not getting through...! How can I send e-mail to the list? -- Tks! ------------------- Guilherme Longo Dept. Eng. da Computação Unaerp Linux User - #484927 *Before Asking http://www.istf.com.br/?page=perguntas !- I'd rather die on my feet than live on my knees -!

2 1

[alsa-devel] Timing of linked devices
by louis＠museresearch.com 15 Jun '09

15 Jun '09

I am developing an application which opens both inputs and outputs for digital and s/pdif (so, 4 devices). The sync IDs all match. However, under certain conditions, I am receiving partial buffers from the input. So, I have a few questions: Does linking input and output devices, from a programmer's perspective, mean that input is ready at precisely the same time that the device is ready to output a buffer? Or, does it only mean that the ins and outs will be synchronized for each given buffer? More to the point, will blocking on a write also accurately tell the program when the device is ready for a read (including under non-ideal circumstances like after a bunch of xruns)? And, in the case that I am doing a non-blocking read (pcm_read) from a buffer which is not yet ready, should I receive an outdated buffer, a blank buffer, or one which has both partially new and partially old data? I am using ALSA 1.0.15 and the ICE1724 driver. Thanks in advance for your help! - Louis

2 1

Re: [alsa-devel] snd-ctxfi oops
by The Source 15 Jun '09

15 Jun '09

On 12.06.2009 22:22, The Source wrote: > On 12.06.2009 11:20, Takashi Iwai wrote: >> At Fri, 12 Jun 2009 07:52:22 +0400, >> The Source wrote: >>> Hello. I use snd-ctxfi from 8 June snapshot. I hadn't oopses since I >>> installed this driver but recently I got one again: >>> >>> Kernel failure message 1: >>> BUG: sleeping function called from invalid context at mm/slub.c:1599 >> Could you try the patch below? >> >> >> thanks, >> >> Takashi >> >> --- >> diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c >> index 80fb2ba..b0adc80 100644 >> --- a/sound/pci/ctxfi/ctatc.c >> +++ b/sound/pci/ctxfi/ctatc.c >> @@ -259,7 +259,6 @@ static int atc_pcm_playback_prepare(struct ct_atc >> *atc, struct ct_atc_pcm *apcm) >> int n_amixer = apcm->substream->runtime->channels, i = 0; >> int device = apcm->substream->pcm->device; >> unsigned int pitch; >> - unsigned long flags; >> >> if (NULL != apcm->src) { >> /* Prepared pcm playback */ >> @@ -311,10 +310,10 @@ static int atc_pcm_playback_prepare(struct >> ct_atc *atc, struct ct_atc_pcm *apcm) >> src = apcm->src; >> for (i = 0; i< n_amixer; i++) { >> amixer = apcm->amixers[i]; >> - spin_lock_irqsave(&atc->atc_lock, flags); >> + mutex_lock(&atc->atc_mutex); >> amixer->ops->setup(amixer,&src->rsc, >> INIT_VOL, atc->pcm[i+device*2]); >> - spin_unlock_irqrestore(&atc->atc_lock, flags); >> + mutex_unlock(&atc->atc_mutex); >> src = src->ops->next_interleave(src); >> if (NULL == src) >> src = apcm->src; >> @@ -865,7 +864,6 @@ static int >> spdif_passthru_playback_setup(struct ct_atc *atc, struct ct_atc_pcm >> *apcm) >> { >> struct dao *dao = container_of(atc->daios[SPDIFOO], struct dao, >> daio); >> - unsigned long flags; >> unsigned int rate = apcm->substream->runtime->rate; >> unsigned int status; >> int err; >> @@ -885,7 +883,7 @@ spdif_passthru_playback_setup(struct ct_atc *atc, >> struct ct_atc_pcm *apcm) >> return -ENOENT; >> } >> >> - spin_lock_irqsave(&atc->atc_lock, flags); >> + mutex_lock(&atc->atc_mutex); >> dao->ops->get_spos(dao,&status); >> if (((status>> 24)& IEC958_AES3_CON_FS) != iec958_con_fs) { >> status&= ((~IEC958_AES3_CON_FS)<< 24); >> @@ -895,7 +893,7 @@ spdif_passthru_playback_setup(struct ct_atc *atc, >> struct ct_atc_pcm *apcm) >> } >> if ((rate != atc->pll_rate)&& (32000 != rate)) >> err = atc_pll_init(atc, rate); >> - spin_unlock_irqrestore(&atc->atc_lock, flags); >> + mutex_unlock(&atc->atc_mutex); >> >> return err; >> } >> @@ -908,7 +906,6 @@ spdif_passthru_playback_prepare(struct ct_atc >> *atc, struct ct_atc_pcm *apcm) >> struct dao *dao; >> int err; >> int i; >> - unsigned long flags; >> >> if (NULL != apcm->src) >> return 0; >> @@ -934,13 +931,13 @@ spdif_passthru_playback_prepare(struct ct_atc >> *atc, struct ct_atc_pcm *apcm) >> src = apcm->src; >> } >> /* Connect to SPDIFOO */ >> - spin_lock_irqsave(&atc->atc_lock, flags); >> + mutex_lock(&atc->atc_mutex); >> dao = container_of(atc->daios[SPDIFOO], struct dao, daio); >> amixer = apcm->amixers[0]; >> dao->ops->set_left_input(dao,&amixer->rsc); >> amixer = apcm->amixers[1]; >> dao->ops->set_right_input(dao,&amixer->rsc); >> - spin_unlock_irqrestore(&atc->atc_lock, flags); >> + mutex_unlock(&atc->atc_mutex); >> >> ct_timer_prepare(apcm->timer); >> >> @@ -1088,7 +1085,6 @@ static int atc_spdif_out_set_status(struct >> ct_atc *atc, unsigned int status) >> >> static int atc_spdif_out_passthru(struct ct_atc *atc, unsigned char >> state) >> { >> - unsigned long flags; >> struct dao_desc da_dsc = {0}; >> struct dao *dao; >> int err; >> @@ -1096,7 +1092,7 @@ static int atc_spdif_out_passthru(struct ct_atc >> *atc, unsigned char state) >> struct rsc *rscs[2] = {NULL}; >> unsigned int spos = 0; >> >> - spin_lock_irqsave(&atc->atc_lock, flags); >> + mutex_lock(&atc->atc_mutex); >> dao = container_of(atc->daios[SPDIFOO], struct dao, daio); >> da_dsc.msr = state ? 1 : atc->msr; >> da_dsc.passthru = state ? 1 : 0; >> @@ -1114,7 +1110,7 @@ static int atc_spdif_out_passthru(struct ct_atc >> *atc, unsigned char state) >> } >> dao->ops->set_spos(dao, spos); >> dao->ops->commit_write(dao); >> - spin_unlock_irqrestore(&atc->atc_lock, flags); >> + mutex_unlock(&atc->atc_mutex); >> >> return err; >> } >> @@ -1572,7 +1568,7 @@ int __devinit ct_atc_create(struct snd_card >> *card, struct pci_dev *pci, >> atc->msr = msr; >> atc->chip_type = chip_type; >> >> - spin_lock_init(&atc->atc_lock); >> + mutex_init(&atc->atc_mutex); >> >> /* Find card model */ >> err = atc_identify_card(atc); >> diff --git a/sound/pci/ctxfi/ctatc.h b/sound/pci/ctxfi/ctatc.h >> index a033472..9fe620e 100644 >> --- a/sound/pci/ctxfi/ctatc.h >> +++ b/sound/pci/ctxfi/ctatc.h >> @@ -19,7 +19,7 @@ >> #define CTATC_H >> >> #include<linux/types.h> >> -#include<linux/spinlock_types.h> >> +#include<linux/mutex.h> >> #include<linux/pci.h> >> #include<linux/timer.h> >> #include<sound/core.h> >> @@ -90,7 +90,7 @@ struct ct_atc { >> void (*unmap_audio_buffer)(struct ct_atc *atc, struct >> ct_atc_pcm *apcm); >> unsigned long (*get_ptp_phys)(struct ct_atc *atc, int index); >> >> - spinlock_t atc_lock; >> + struct mutex atc_mutex; >> >> int (*pcm_playback_prepare)(struct ct_atc *atc, >> struct ct_atc_pcm *apcm); >> > No oopses with this patch so far, thank you. Tested the patch more. Really no oopses. Instead of them I get rare system lockups.

2 2

[alsa-devel] ASoC: About amp driver
by Joonyoung Shim 15 Jun '09

15 Jun '09

Hi. I have one question. Currently, the amp driver of ASoC is included in machine driver. (ex. neo1973_wm8753 machine driver) I want to use the same amp driver in more than one machine driver, but then the same amp driver code exists double in each machine driver. So, how about spliting between the amp driver and the machine driver? and Is it possible?

2 3

[alsa-devel] [PATCH 0/4] ASoC updates for 2.6.31
by Mark Brown 15 Jun '09

15 Jun '09

The following changes since commit 236e6723bedb483b2ebf73ada01e2c853c5cac01: Mark Brown (1): ASoC: Fix lm4857 control are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31 Mark Brown (3): ASoC: Remove odd bit clock ratios for WM8903 ASoC: Revert duplicated code in SSM2602 driver ASoC: Instantiate any forgotten DAPM widgets Philipp Zabel (1): ASoC: magician: fix PXA SSP clock polarity sound/soc/codecs/ssm2602.c | 4 ---- sound/soc/codecs/wm8903.c | 4 ---- sound/soc/pxa/magician.c | 2 +- sound/soc/soc-core.c | 3 +++ 4 files changed, 4 insertions(+), 9 deletions(-)

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Re: [alsa-devel] [Alsa-user] Problems with Soundblaster X-Fi
by WILMES Andreas 15 Jun '09

15 Jun '09

Daniel Herrmann wrote: > Same error than before... > > Didn't change anything, but thanks Are you sure that you're using the same gcc-compiler on the alsa modules as same as on the kernel? The compiler-versions might being different.

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Re: [alsa-devel] [Alsa-user] Problems with Soundblaster X-Fi
by WILMES Andreas 15 Jun '09

15 Jun '09

Daniel Herrmann wrote: > WILMES Andreas schrieb: >> Daniel Herrmann wrote: >>> WILMES Andreas schrieb: >>>> Daniel Herrmann wrote: >>>> >>>>> Hey all, >>>>> >>>>> i tried out the new devel-snapshot including alsa-support for my >>>>> Soundblaster x-Fi card. >>>>> >>>>> Installed it via ./configure; make; make install and rebooted. >>>>> >>>>> The driver didnt load at all. >>>>> >>>>> Trying to load the driver manually brings up the following message: >>>>> >>>>> >>>>>> modprobe snd-ctxfi >>>>>> >>>>> FATAL: Error inserting snd_ctxfi >>>>> (/lib/modules/2.6.28-11-generic/kernel/sound/pci/ctxfi/snd-ctxfi.ko): >>>>> Unknown symbol in module, or unknown parameter >>>>> >>>>> >>>>>> dmesg shows the following: >>>>>> >>>>> [ 2146.563073] snd_ctxfi: disagrees about version of symbol >>>>> snd_ctl_add >>>>> [ 2146.563081] snd_ctxfi: Unknown symbol snd_ctl_add >>>>> [ 2146.563291] snd_ctxfi: disagrees about version of symbol >>>>> snd_pcm_new >>>>> [ 2146.563295] snd_ctxfi: Unknown symbol snd_pcm_new >>>>> [ 2146.563482] snd_ctxfi: disagrees about version of symbol >>>>> snd_card_register >>>>> >>>>> [...] and so on, about 60 lines >>>>> >>>>> [ 3198.031060] snd_ctxfi: disagrees about version of symbol >>>>> snd_pcm_period_elapsed >>>>> [ 3198.031063] snd_ctxfi: Unknown symbol snd_pcm_period_elapsed >>>>> >>>>> >>>>>> lspci | grep -i audio >>>>>> >>>>> 00:0a.0 Multimedia audio controller: Creative Labs SB X-Fi >>>>> >>>>> Currently using Ubuntu 9.04 Jaunty with the current >>>>> 2.6.28-11-generic kernel >>>>> >>>>> Maybe anyone has an idea, why i cant install the driver? >>>>> It worked for me before, but unfortunatly i had to reinstall >>>>> ubuntu after getting a weird dependeny problem in apt. >>>>> Not shure why it doesn't do so now... >>>>> >>>>> Thanks in advance for your help and thanks for the great work you do! >>>>> >>>>> >>>> I had the same error. Then I rebooted and the error disappeared. >>>> Hope that will help. >>>> >>> Hm, i rebooted more than once - same error. >>> >>> Daniel >>> >> I deleted /usr/include/asm and I symlinked it to >> /usr/src/linux-headers-2.6.28-11/include/asm-generic and I did a >> recompile. After rebooting then the error disappeared. >> > HM... You deleted it and made a symlink from it? How does this work? > How can i link something deleted?! > > Can't follow you ;) Oh I'm Sorry. /usr/include/asm was a static folder containing a lot of files before. I linked /usr/src/linux-headers-2.6.28-11/include/asm-generic to /usr/include/asm then.

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[alsa-devel] [PATCH v4 3/4] ARM: DaVinci: ASoC: Add mcasp support for DM646x
by Chaithrika U S 15 Jun '09

15 Jun '09

Adds driver support for the two instances of McASP on TI's DM646x. The multichannel audio serial port (McASP) functions as a general-purpose audio serial port optimized for the needs of multichannel audio application. (http://www.ti.com/litv/pdf/spruer1b). There are two instances of McASP on DM646x. The McASP0 module includes up to 4 serializers that can be individually enabled to either transmit or receive in different modes. The McASP1 module is limited with only 1 pinned-out serializer that can be enabled to only transmit in DIT mode (neither receiving in any mode nor transmitting in either Burst or TDM mode is supported). McASP0 consists of transmit and receive sections that may operate synchronized, or completely independently with separate master clocks, bit clocks, and frame syncs, and using different transmit modes with different bit-stream formats. Signed-off-by: Steve Chen <schen(a)mvista.com> Signed-off-by: Pavel Kiryukhin <pkiryukhin(a)ru.mvista.com> Signed-off-by: Naresh Medisetty <naresh(a)ti.com> Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- This patch applies against ALSA mainline - the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git sound/soc/davinci/Kconfig | 13 + sound/soc/davinci/Makefile | 3 + sound/soc/davinci/davinci-mcasp.c | 874 +++++++++++++++++++++++++++++++++++++ sound/soc/davinci/davinci-mcasp.h | 55 +++ sound/soc/davinci/davinci-pcm.h | 18 +- 5 files changed, 955 insertions(+), 8 deletions(-) create mode 100644 sound/soc/davinci/davinci-mcasp.c create mode 100644 sound/soc/davinci/davinci-mcasp.h diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 411a710..6802dd5 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -9,6 +9,9 @@ config SND_DAVINCI_SOC config SND_DAVINCI_SOC_I2S tristate +config SND_DAVINCI_SOC_MCASP + tristate + config SND_DAVINCI_SOC_EVM tristate "SoC Audio support for DaVinci DM6446 or DM355 EVM" depends on SND_DAVINCI_SOC @@ -19,6 +22,16 @@ config SND_DAVINCI_SOC_EVM Say Y if you want to add support for SoC audio on TI DaVinci DM6446 or DM355 EVM platforms. +config SND_DM6467_SOC_EVM + tristate "SoC Audio support for DaVinci DM6467 EVM" + depends on SND_DAVINCI_SOC && MACH_DAVINCI_DM6467_EVM + select SND_DAVINCI_SOC_MCASP + select SND_SOC_TLV320AIC3X + select SND_SOC_SPDIF + + help + Say Y if you want to add support for SoC audio on TI + config SND_DAVINCI_SOC_SFFSDR tristate "SoC Audio support for SFFSDR" depends on SND_DAVINCI_SOC && MACH_SFFSDR diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index ca8bae1..67be54f 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -1,13 +1,16 @@ # DAVINCI Platform Support snd-soc-davinci-objs := davinci-pcm.o snd-soc-davinci-i2s-objs := davinci-i2s.o +snd-soc-davinci-mcasp-objs:= davinci-mcasp.o obj-$(CONFIG_SND_DAVINCI_SOC) += snd-soc-davinci.o obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o +obj-$(CONFIG_SND_DAVINCI_SOC_MCASP) += snd-soc-davinci-mcasp.o # DAVINCI Machine Support snd-soc-evm-objs := davinci-evm.o snd-soc-sffsdr-objs := davinci-sffsdr.o obj-$(CONFIG_SND_DAVINCI_SOC_EVM) += snd-soc-evm.o +obj-$(CONFIG_SND_DM6467_SOC_EVM) += snd-soc-evm.o obj-$(CONFIG_SND_DAVINCI_SOC_SFFSDR) += snd-soc-sffsdr.o diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c new file mode 100644 index 0000000..b27aab6 --- /dev/null +++ b/sound/soc/davinci/davinci-mcasp.c @@ -0,0 +1,874 @@ +/* + * ALSA SoC McASP Audio Layer for TI DAVINCI processor + * + * Multi-channel Audio Serial Port Driver + * + * Author: Nirmal Pandey <n-pandey(a)ti.com>, + * Suresh Rajashekara <suresh.r(a)ti.com> + * Steve Chen <schen@.mvista.com> + * + * Copyright: (C) 2009 MontaVista Software, Inc., <source(a)mvista.com> + * Copyright: (C) 2009 Texas Instruments, India + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/clk.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include "davinci-pcm.h" +#include "davinci-mcasp.h" + +/* + * McASP register definitions + */ +#define DAVINCI_MCASP_PID_REG 0x00 +#define DAVINCI_MCASP_PWREMUMGT_REG 0x04 + +#define DAVINCI_MCASP_PFUNC_REG 0x10 +#define DAVINCI_MCASP_PDIR_REG 0x14 +#define DAVINCI_MCASP_PDOUT_REG 0x18 +#define DAVINCI_MCASP_PDSET_REG 0x1c + +#define DAVINCI_MCASP_PDCLR_REG 0x20 + +#define DAVINCI_MCASP_TLGC_REG 0x30 +#define DAVINCI_MCASP_TLMR_REG 0x34 + +#define DAVINCI_MCASP_GBLCTL_REG 0x44 +#define DAVINCI_MCASP_AMUTE_REG 0x48 +#define DAVINCI_MCASP_LBCTL_REG 0x4c + +#define DAVINCI_MCASP_TXDITCTL_REG 0x50 + +#define DAVINCI_MCASP_GBLCTLR_REG 0x60 +#define DAVINCI_MCASP_RXMASK_REG 0x64 +#define DAVINCI_MCASP_RXFMT_REG 0x68 +#define DAVINCI_MCASP_RXFMCTL_REG 0x6c + +#define DAVINCI_MCASP_ACLKRCTL_REG 0x70 +#define DAVINCI_MCASP_AHCLKRCTL_REG 0x74 +#define DAVINCI_MCASP_RXTDM_REG 0x78 +#define DAVINCI_MCASP_EVTCTLR_REG 0x7c + +#define DAVINCI_MCASP_RXSTAT_REG 0x80 +#define DAVINCI_MCASP_RXTDMSLOT_REG 0x84 +#define DAVINCI_MCASP_RXCLKCHK_REG 0x88 +#define DAVINCI_MCASP_REVTCTL_REG 0x8c + +#define DAVINCI_MCASP_GBLCTLX_REG 0xa0 +#define DAVINCI_MCASP_TXMASK_REG 0xa4 +#define DAVINCI_MCASP_TXFMT_REG 0xa8 +#define DAVINCI_MCASP_TXFMCTL_REG 0xac + +#define DAVINCI_MCASP_ACLKXCTL_REG 0xb0 +#define DAVINCI_MCASP_AHCLKXCTL_REG 0xb4 +#define DAVINCI_MCASP_TXTDM_REG 0xb8 +#define DAVINCI_MCASP_EVTCTLX_REG 0xbc + +#define DAVINCI_MCASP_TXSTAT_REG 0xc0 +#define DAVINCI_MCASP_TXTDMSLOT_REG 0xc4 +#define DAVINCI_MCASP_TXCLKCHK_REG 0xc8 +#define DAVINCI_MCASP_XEVTCTL_REG 0xcc + +/* Left(even TDM Slot) Channel Status Register File */ +#define DAVINCI_MCASP_DITCSRA_REG 0x100 +/* Right(odd TDM slot) Channel Status Register File */ +#define DAVINCI_MCASP_DITCSRB_REG 0x118 +/* Left(even TDM slot) User Data Register File */ +#define DAVINCI_MCASP_DITUDRA_REG 0x130 +/* Right(odd TDM Slot) User Data Register File */ +#define DAVINCI_MCASP_DITUDRB_REG 0x148 + +/* Serializer n Control Register */ +#define DAVINCI_MCASP_XRSRCTL_BASE_REG 0x180 +#define DAVINCI_MCASP_XRSRCTL_REG(n) (DAVINCI_MCASP_XRSRCTL_BASE_REG + \ + (n << 2)) + +/* Transmit Buffer for Serializer n */ +#define DAVINCI_MCASP_TXBUF_REG 0x200 +/* Receive Buffer for Serializer n */ +#define DAVINCI_MCASP_RXBUF_REG 0x280 + + +/* + * DAVINCI_MCASP_PWREMUMGT_REG - Power Down and Emulation Management + * Register Bits + */ +#define MCASP_FREE BIT(0) +#define MCASP_SOFT BIT(1) + +/* + * DAVINCI_MCASP_PFUNC_REG - Pin Function / GPIO Enable Register Bits + */ +#define AXR(n) (1<<n) +#define PFUNC_AMUTE BIT(25) +#define ACLKX BIT(26) +#define AHCLKX BIT(27) +#define AFSX BIT(28) +#define ACLKR BIT(29) +#define AHCLKR BIT(30) +#define AFSR BIT(31) + +/* + * DAVINCI_MCASP_PDIR_REG - Pin Direction Register Bits + */ +#define AXR(n) (1<<n) +#define PDIR_AMUTE BIT(25) +#define ACLKX BIT(26) +#define AHCLKX BIT(27) +#define AFSX BIT(28) +#define ACLKR BIT(29) +#define AHCLKR BIT(30) +#define AFSR BIT(31) + +/* + * DAVINCI_MCASP_TXDITCTL_REG - Transmit DIT Control Register Bits + */ +#define DITEN BIT(0) /* Transmit DIT mode enable/disable */ +#define VA BIT(2) +#define VB BIT(3) + +/* + * DAVINCI_MCASP_TXFMT_REG - Transmit Bitstream Format Register Bits + */ +#define TXROT(val) (val) +#define TXSEL BIT(3) +#define TXSSZ(val) (val<<4) +#define TXPBIT(val) (val<<8) +#define TXPAD(val) (val<<13) +#define TXORD BIT(15) +#define FSXDLY(val) (val<<16) + +/* + * DAVINCI_MCASP_RXFMT_REG - Receive Bitstream Format Register Bits + */ +#define RXROT(val) (val) +#define RXSEL BIT(3) +#define RXSSZ(val) (val<<4) +#define RXPBIT(val) (val<<8) +#define RXPAD(val) (val<<13) +#define RXORD BIT(15) +#define FSRDLY(val) (val<<16) + +/* + * DAVINCI_MCASP_TXFMCTL_REG - Transmit Frame Control Register Bits + */ +#define FSXPOL BIT(0) +#define AFSXE BIT(1) +#define FSXDUR BIT(4) +#define FSXMOD(val) (val<<7) + +/* + * DAVINCI_MCASP_RXFMCTL_REG - Receive Frame Control Register Bits + */ +#define FSRPOL BIT(0) +#define AFSRE BIT(1) +#define FSRDUR BIT(4) +#define FSRMOD(val) (val<<7) + +/* + * DAVINCI_MCASP_ACLKXCTL_REG - Transmit Clock Control Register Bits + */ +#define ACLKXDIV(val) (val) +#define ACLKXE BIT(5) +#define TX_ASYNC BIT(6) +#define ACLKXPOL BIT(7) + +/* + * DAVINCI_MCASP_ACLKRCTL_REG Receive Clock Control Register Bits + */ +#define ACLKRDIV(val) (val) +#define ACLKRE BIT(5) +#define RX_ASYNC BIT(6) +#define ACLKRPOL BIT(7) + +/* + * DAVINCI_MCASP_AHCLKXCTL_REG - High Frequency Transmit Clock Control + * Register Bits + */ +#define AHCLKXDIV(val) (val) +#define AHCLKXPOL BIT(14) +#define AHCLKXE BIT(15) + +/* + * DAVINCI_MCASP_AHCLKRCTL_REG - High Frequency Receive Clock Control + * Register Bits + */ +#define AHCLKRDIV(val) (val) +#define AHCLKRPOL BIT(14) +#define AHCLKRE BIT(15) + +/* + * DAVINCI_MCASP_XRSRCTL_BASE_REG - Serializer Control Register Bits + */ +#define MODE(val) (val) +#define DISMOD (val)(val<<2) +#define TXSTATE BIT(4) +#define RXSTATE BIT(5) + +/* + * DAVINCI_MCASP_LBCTL_REG - Loop Back Control Register Bits + */ +#define LBEN BIT(0) +#define LBORD BIT(1) +#define LBGENMODE(val) (val<<2) + +/* + * DAVINCI_MCASP_TXTDMSLOT_REG - Transmit TDM Slot Register configuration + */ +#define TXTDMS(n) (1<<n) + +/* + * DAVINCI_MCASP_RXTDMSLOT_REG - Receive TDM Slot Register configuration + */ +#define RXTDMS(n) (1<<n) + +/* + * DAVINCI_MCASP_GBLCTL_REG - Global Control Register Bits + */ +#define RXCLKRST BIT(0) /* Receiver Clock Divider Reset */ +#define RXHCLKRST BIT(1) /* Receiver High Frequency Clock Divider */ +#define RXSERCLR BIT(2) /* Receiver Serializer Clear */ +#define RXSMRST BIT(3) /* Receiver State Machine Reset */ +#define RXFSRST BIT(4) /* Frame Sync Generator Reset */ +#define TXCLKRST BIT(8) /* Transmitter Clock Divider Reset */ +#define TXHCLKRST BIT(9) /* Transmitter High Frequency Clock Divider*/ +#define TXSERCLR BIT(10) /* Transmit Serializer Clear */ +#define TXSMRST BIT(11) /* Transmitter State Machine Reset */ +#define TXFSRST BIT(12) /* Frame Sync Generator Reset */ + +/* + * DAVINCI_MCASP_AMUTE_REG - Mute Control Register Bits + */ +#define MUTENA(val) (val) +#define MUTEINPOL BIT(2) +#define MUTEINENA BIT(3) +#define MUTEIN BIT(4) +#define MUTER BIT(5) +#define MUTEX BIT(6) +#define MUTEFSR BIT(7) +#define MUTEFSX BIT(8) +#define MUTEBADCLKR BIT(9) +#define MUTEBADCLKX BIT(10) +#define MUTERXDMAERR BIT(11) +#define MUTETXDMAERR BIT(12) + +/* + * DAVINCI_MCASP_REVTCTL_REG - Receiver DMA Event Control Register bits + */ +#define RXDATADMADIS BIT(0) + +/* + * DAVINCI_MCASP_XEVTCTL_REG - Transmitter DMA Event Control Register bits + */ +#define TXDATADMADIS BIT(0) + +#define DAVINCI_MCASP_NUM_SERIALIZER 16 + +static inline void mcasp_set_bits(void __iomem *reg, u32 val) +{ + __raw_writel(__raw_readl(reg) | val, reg); +} + +static inline void mcasp_clr_bits(void __iomem *reg, u32 val) +{ + __raw_writel((__raw_readl(reg) & ~(val)), reg); +} + +static inline void mcasp_mod_bits(void __iomem *reg, u32 val, u32 mask) +{ + __raw_writel((__raw_readl(reg) & ~mask) | val, reg); +} + +static inline void mcasp_set_reg(void __iomem *reg, u32 val) +{ + __raw_writel(val, reg); +} + +static inline u32 mcasp_get_reg(void __iomem *reg) +{ + return (unsigned int)__raw_readl(reg); +} + +static inline void mcasp_set_ctl_reg(void __iomem *regs, u32 val) +{ + int i = 0; + + mcasp_set_bits(regs, val); + + /* programming GBLCTL needs to read back from GBLCTL and verfiy */ + /* loop count is to avoid the lock-up */ + for (i = 0; i < 1000; i++) { + if ((mcasp_get_reg(regs) & val) == val) + break; + } + + if (i == 1000 && ((mcasp_get_reg(regs) & val) != val)) + printk(KERN_ERR "GBLCTL write error\n"); +} + +static int davinci_mcasp_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct davinci_audio_dev *dev = cpu_dai->private_data; + cpu_dai->dma_data = dev->dma_params[substream->stream]; + return 0; +} + +static void mcasp_start_rx(struct davinci_audio_dev *dev) +{ + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXHCLKRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXCLKRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSERCLR); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXBUF_REG, 0); + + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSMRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXFSRST); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXBUF_REG, 0); + + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSMRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXFSRST); +} + +static void mcasp_start_tx(struct davinci_audio_dev *dev) +{ + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXHCLKRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXCLKRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXSERCLR); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); + + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXSMRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXFSRST); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); +} + +static void davinci_mcasp_start(struct davinci_audio_dev *dev, int stream) +{ + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + mcasp_start_tx(dev); + else + mcasp_start_rx(dev); +} + +static void mcasp_stop_rx(struct davinci_audio_dev *dev) +{ + mcasp_set_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, 0); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF); +} + +static void mcasp_stop_tx(struct davinci_audio_dev *dev) +{ + mcasp_set_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, 0); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF); +} + +static void davinci_mcasp_stop(struct davinci_audio_dev *dev, int stream) +{ + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + mcasp_stop_tx(dev); + else + mcasp_stop_rx(dev); +} + +static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct davinci_audio_dev *dev = cpu_dai->private_data; + void __iomem *base = dev->base; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + /* codec is clock and frame slave */ + mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE); + + mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); + mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE); + + mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, (0x7 << 26)); + break; + + case SND_SOC_DAIFMT_CBM_CFM: + /* codec is clock and frame master */ + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE); + + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); + mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE); + + mcasp_clr_bits(base + DAVINCI_MCASP_PDIR_REG, (0x3f << 26)); + break; + + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_NF: + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + + mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + break; + + case SND_SOC_DAIFMT_NB_IF: + mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + break; + + case SND_SOC_DAIFMT_IB_IF: + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + + mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + break; + + case SND_SOC_DAIFMT_NB_NF: + mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + break; + + default: + return -EINVAL; + } + + return 0; +} + +static int davinci_config_channel_size(struct davinci_audio_dev *dev, + int channel_size) +{ + u32 fmt = 0; + + switch (channel_size) { + case DAVINCI_AUDIO_WORD_8: + fmt = 0x03; + break; + + case DAVINCI_AUDIO_WORD_12: + fmt = 0x05; + break; + + case DAVINCI_AUDIO_WORD_16: + fmt = 0x07; + break; + + case DAVINCI_AUDIO_WORD_20: + fmt = 0x09; + break; + + case DAVINCI_AUDIO_WORD_24: + fmt = 0x0B; + break; + + case DAVINCI_AUDIO_WORD_28: + fmt = 0x0D; + break; + + case DAVINCI_AUDIO_WORD_32: + fmt = 0x0F; + break; + + default: + return -EINVAL; + } + + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, + RXSSZ(fmt), RXSSZ(0x0F)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, + TXSSZ(fmt), TXSSZ(0x0F)); + return 0; +} + +static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) +{ + int i; + + /* Default configuration */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT); + + /* All PINS as McASP */ + mcasp_set_reg(dev->base + DAVINCI_MCASP_PFUNC_REG, 0x00000000); + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_XEVTCTL_REG, + TXDATADMADIS); + } else { + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_REVTCTL_REG, + RXDATADMADIS); + } + + for (i = 0; i < dev->num_serializer; i++) { + mcasp_set_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), + dev->serial_dir[i]); + if (dev->serial_dir[i] == TX_MODE) + mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, + AXR(i)); + else if (dev->serial_dir[i] == RX_MODE) + mcasp_clr_bits(dev->base + DAVINCI_MCASP_PDIR_REG, + AXR(i)); + } +} + +static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) +{ + int i, active_slots; + u32 mask = 0; + + active_slots = (dev->tdm_slots > 31) ? 32 : dev->tdm_slots; + for (i = 0; i < active_slots; i++) + mask |= (1 << i); + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* bit stream is MSB first with no delay */ + /* DSP_B mode */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG, + AHCLKXE); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXTDM_REG, mask); + mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXORD); + + if ((dev->tdm_slots >= 2) || (dev->tdm_slots <= 32)) + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, + FSXMOD(dev->tdm_slots), FSXMOD(0x1FF)); + else + printk(KERN_ERR "playback tdm slot %d not supported\n", + dev->tdm_slots); + + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0xFFFFFFFF); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR); + } else { + /* bit stream is MSB first with no delay */ + /* DSP_B mode */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, RXORD); + mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKRCTL_REG, + AHCLKRE); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXTDM_REG, mask); + + if ((dev->tdm_slots >= 2) || (dev->tdm_slots <= 32)) + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, + FSRMOD(dev->tdm_slots), FSRMOD(0x1FF)); + else + printk(KERN_ERR "capture tdm slot %d not supported\n", + dev->tdm_slots); + + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, 0xFFFFFFFF); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR); + } +} + +/* S/PDIF */ +static void davinci_hw_dit_param(struct davinci_audio_dev *dev) +{ + /* Set the PDIR for Serialiser as output */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AFSX); + + /* TXMASK for 24 bits */ + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0x00FFFFFF); + + /* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0 + and LSB first */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, + TXROT(6) | TXSSZ(15)); + + /* Set TX frame synch : DIT Mode, 1 bit width, internal, rising edge */ + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXFMCTL_REG, + AFSXE | FSXMOD(0x180)); + + /* Set the TX tdm : for all the slots */ + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXTDM_REG, 0xFFFFFFFF); + + /* Set the TX clock controls : div = 1 and internal */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_ACLKXCTL_REG, + ACLKXE | TX_ASYNC); + + mcasp_clr_bits(dev->base + DAVINCI_MCASP_XEVTCTL_REG, TXDATADMADIS); + + /* Only 44100 and 48000 are valid, both have the same setting */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXDIV(3)); + + /* Enable the DIT */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_TXDITCTL_REG, DITEN); +} + +static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + struct davinci_audio_dev *dev = cpu_dai->private_data; + struct davinci_pcm_dma_params *dma_params = + dev->dma_params[substream->stream]; + int word_length; + + davinci_hw_common_param(dev, substream->stream); + + if (dev->op_mode == DAVINCI_MCASP_DIT_MODE) + davinci_hw_dit_param(dev); + else + davinci_hw_param(dev, substream->stream); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + dma_params->data_type = 1; + word_length = DAVINCI_AUDIO_WORD_8; + break; + + case SNDRV_PCM_FORMAT_S16_LE: + dma_params->data_type = 2; + word_length = DAVINCI_AUDIO_WORD_16; + break; + + case SNDRV_PCM_FORMAT_S32_LE: + dma_params->data_type = 4; + word_length = DAVINCI_AUDIO_WORD_32; + break; + + default: + printk(KERN_WARNING "davinci-mcasp: unsupported PCM format"); + return -EINVAL; + } + davinci_config_channel_size(dev, word_length); + + return 0; +} + +static int davinci_mcasp_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *cpu_dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + davinci_mcasp_start(dev, substream->stream); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + davinci_mcasp_stop(dev, substream->stream); + break; + + default: + ret = -EINVAL; + } + + return ret; +} + +static struct snd_soc_dai_ops davinci_mcasp_dai_ops = { + .startup = davinci_mcasp_startup, + .trigger = davinci_mcasp_trigger, + .hw_params = davinci_mcasp_hw_params, + .set_fmt = davinci_mcasp_set_dai_fmt, + +}; + +struct snd_soc_dai davinci_mcasp_dai[] = { + { + .name = "davinci-i2s", + .id = 0, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = DAVINCI_MCASP_RATES, + .formats = SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = DAVINCI_MCASP_RATES, + .formats = SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &davinci_mcasp_dai_ops, + + }, + { + .name = "davinci-dit", + .id = 1, + .playback = { + .channels_min = 1, + .channels_max = 384, + .rates = DAVINCI_MCASP_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &davinci_mcasp_dai_ops, + }, + +}; +EXPORT_SYMBOL_GPL(davinci_mcasp_dai); + +static int davinci_mcasp_probe(struct platform_device *pdev) +{ + struct davinci_pcm_dma_params *dma_data; + struct resource *mem, *ioarea, *res; + struct snd_platform_data *pdata; + struct davinci_audio_dev *dev; + int count = 0; + int ret = 0; + + dev = kzalloc(sizeof(struct davinci_audio_dev), GFP_KERNEL); + if (!dev) + return -ENOMEM; + + dma_data = kzalloc(sizeof(struct davinci_pcm_dma_params) * 2, + GFP_KERNEL); + if (!dma_data) { + ret = -ENOMEM; + goto err_release_dev; + } + + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!mem) { + dev_err(&pdev->dev, "no mem resource?\n"); + ret = -ENODEV; + goto err_release_data; + } + + ioarea = request_mem_region(mem->start, + (mem->end - mem->start) + 1, pdev->name); + if (!ioarea) { + dev_err(&pdev->dev, "Audio region already claimed\n"); + ret = -EBUSY; + goto err_release_data; + } + + pdata = pdev->dev.platform_data; + dev->clk = clk_get(&pdev->dev, pdata->clk_name); + if (IS_ERR(dev->clk)) { + ret = -ENODEV; + goto err_release_region; + } + + clk_enable(dev->clk); + + dev->base = (void __iomem *)IO_ADDRESS(mem->start); + dev->op_mode = pdata->op_mode; + dev->tdm_slots = pdata->tdm_slots; + dev->num_serializer = pdata->num_serializer; + dev->serial_dir = pdata->serial_dir; + dev->codec_fmt = pdata->codec_fmt; + + dma_data[count].name = "I2S PCM Stereo out"; + dma_data[count].eventq_no = pdata->eventq_no; + dma_data[count].dma_addr = (dma_addr_t) (pdata->tx_dma_offset + + io_v2p(dev->base)); + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK] = &dma_data[count]; + + /* first TX, then RX */ + res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!res) { + dev_err(&pdev->dev, "no DMA resource\n"); + goto err_release_region; + } + + dma_data[count].channel = res->start; + count++; + dma_data[count].name = "I2S PCM Stereo in"; + dma_data[count].eventq_no = pdata->eventq_no; + dma_data[count].dma_addr = (dma_addr_t)(pdata->rx_dma_offset + + io_v2p(dev->base)); + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE] = &dma_data[count]; + + res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!res) { + dev_err(&pdev->dev, "no DMA resource\n"); + goto err_release_region; + } + + dma_data[count].channel = res->start; + davinci_mcasp_dai[pdev->id].private_data = dev; + davinci_mcasp_dai[pdev->id].dev = &pdev->dev; + ret = snd_soc_register_dai(&davinci_mcasp_dai[pdev->id]); + + if (ret != 0) + goto err_release_region; + return 0; + +err_release_region: + release_mem_region(mem->start, (mem->end - mem->start) + 1); +err_release_data: + kfree(dma_data); +err_release_dev: + kfree(dev); + + return ret; +} + +static int davinci_mcasp_remove(struct platform_device *pdev) +{ + struct davinci_pcm_dma_params *dma_data; + struct davinci_audio_dev *dev; + struct resource *mem; + + snd_soc_unregister_dai(&davinci_mcasp_dai[pdev->id]); + dev = davinci_mcasp_dai[pdev->id].private_data; + clk_disable(dev->clk); + clk_put(dev->clk); + dev->clk = NULL; + + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + release_mem_region(mem->start, (mem->end - mem->start) + 1); + + dma_data = dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]; + kfree(dma_data); + kfree(dev); + + return 0; +} + +static struct platform_driver davinci_mcasp_driver = { + .probe = davinci_mcasp_probe, + .remove = davinci_mcasp_remove, + .driver = { + .name = "davinci-mcasp", + .owner = THIS_MODULE, + }, +}; + +static int __init davinci_mcasp_init(void) +{ + return platform_driver_register(&davinci_mcasp_driver); +} +module_init(davinci_mcasp_init); + +static void __exit davinci_mcasp_exit(void) +{ + platform_driver_unregister(&davinci_mcasp_driver); +} +module_exit(davinci_mcasp_exit); + +MODULE_AUTHOR("Steve Chen"); +MODULE_DESCRIPTION("TI DAVINCI McASP SoC Interface"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h new file mode 100644 index 0000000..36b7104 --- /dev/null +++ b/sound/soc/davinci/davinci-mcasp.h @@ -0,0 +1,55 @@ +/* + * ALSA SoC McASP Audio Layer for TI DAVINCI processor + * + * MCASP related definitions + * + * Author: Nirmal Pandey <n-pandey(a)ti.com>, + * Suresh Rajashekara <suresh.r(a)ti.com> + * Steve Chen <schen@.mvista.com> + * + * Copyright: (C) 2009 MontaVista Software, Inc., <source(a)mvista.com> + * Copyright: (C) 2009 Texas Instruments, India + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef DAVINCI_MCASP_H +#define DAVINCI_MCASP_H + +#include <linux/io.h> +#include <mach/asp.h> +#include "davinci-pcm.h" + +extern struct snd_soc_dai davinci_mcasp_dai[]; + +#define DAVINCI_MCASP_RATES SNDRV_PCM_RATE_8000_96000 +#define DAVINCI_MCASP_I2S_DAI 0 +#define DAVINCI_MCASP_DIT_DAI 1 + +enum { + DAVINCI_AUDIO_WORD_8 = 0, + DAVINCI_AUDIO_WORD_12, + DAVINCI_AUDIO_WORD_16, + DAVINCI_AUDIO_WORD_20, + DAVINCI_AUDIO_WORD_24, + DAVINCI_AUDIO_WORD_32, + DAVINCI_AUDIO_WORD_28, /* This is only valid for McASP */ +}; + +struct davinci_audio_dev { + void __iomem *base; + int sample_rate; + struct clk *clk; + struct davinci_pcm_dma_params *dma_params[2]; + unsigned int codec_fmt; + + /* McASP specific data */ + int tdm_slots; + u8 op_mode; + u8 num_serializer; + u8 *serial_dir; +}; + +#endif /* DAVINCI_MCASP_H */ diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index 62cb4eb..eb4287f 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -12,17 +12,19 @@ #ifndef _DAVINCI_PCM_H #define _DAVINCI_PCM_H +#include <mach/edma.h> +#include <mach/asp.h> + + struct davinci_pcm_dma_params { - char *name; /* stream identifier */ - int channel; /* sync dma channel ID */ - dma_addr_t dma_addr; /* device physical address for DMA */ - unsigned int data_type; /* xfer data type */ + char *name; /* stream identifier */ + int channel; /* sync dma channel ID */ + dma_addr_t dma_addr; /* device physical address for DMA */ + enum dma_event_q eventq_no; /* event queue number */ + unsigned char data_type; /* xfer data type */ + unsigned char convert_mono_stereo; }; -struct evm_snd_platform_data { - int tx_dma_ch; - int rx_dma_ch; -}; extern struct snd_soc_platform davinci_soc_platform; -- 1.5.6

3 2

Re: [alsa-devel] [Alsa-user] Problems with Soundblaster X-Fi
by WILMES Andreas 15 Jun '09

15 Jun '09

Daniel Herrmann wrote: > WILMES Andreas schrieb: >> Daniel Herrmann wrote: >> >>> Hey all, >>> >>> i tried out the new devel-snapshot including alsa-support for my >>> Soundblaster x-Fi card. >>> >>> Installed it via ./configure; make; make install and rebooted. >>> >>> The driver didnt load at all. >>> >>> Trying to load the driver manually brings up the following message: >>> >>> >>>> modprobe snd-ctxfi >>>> >>> FATAL: Error inserting snd_ctxfi >>> (/lib/modules/2.6.28-11-generic/kernel/sound/pci/ctxfi/snd-ctxfi.ko): >>> Unknown symbol in module, or unknown parameter >>> >>> >>>> dmesg shows the following: >>>> >>> [ 2146.563073] snd_ctxfi: disagrees about version of symbol snd_ctl_add >>> [ 2146.563081] snd_ctxfi: Unknown symbol snd_ctl_add >>> [ 2146.563291] snd_ctxfi: disagrees about version of symbol snd_pcm_new >>> [ 2146.563295] snd_ctxfi: Unknown symbol snd_pcm_new >>> [ 2146.563482] snd_ctxfi: disagrees about version of symbol >>> snd_card_register >>> >>> [...] and so on, about 60 lines >>> >>> [ 3198.031060] snd_ctxfi: disagrees about version of symbol >>> snd_pcm_period_elapsed >>> [ 3198.031063] snd_ctxfi: Unknown symbol snd_pcm_period_elapsed >>> >>> >>>> lspci | grep -i audio >>>> >>> 00:0a.0 Multimedia audio controller: Creative Labs SB X-Fi >>> >>> Currently using Ubuntu 9.04 Jaunty with the current >>> 2.6.28-11-generic kernel >>> >>> Maybe anyone has an idea, why i cant install the driver? >>> It worked for me before, but unfortunatly i had to reinstall ubuntu >>> after getting a weird dependeny problem in apt. >>> Not shure why it doesn't do so now... >>> >>> Thanks in advance for your help and thanks for the great work you do! >>> >>> >> I had the same error. Then I rebooted and the error disappeared. Hope >> that will help. >> > Hm, i rebooted more than once - same error. > > Daniel > I deleted /usr/include/asm and I symlinked it to /usr/src/linux-headers-2.6.28-11/include/asm-generic and I did a recompile. After rebooting then the error disappeared.

1 0

[alsa-devel] [PATCH v4 1/4] ARM: DaVinci: ASoC: Add the platform devices for ASP
by Chaithrika U S 15 Jun '09

15 Jun '09

1) Registers the platform devices for ASP on dm355, dm644x and dm646x so that the machine driver can probe to get ASP related platform data. 2) Move towards definition of the asp clocks using physical name(for dm355 and dm644x) 3) Add platform data to board specific files. Signed-off-by: Naresh Medisetty <naresh(a)ti.com> Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- This patch applies to the DaVinci git tree available at http://git.kernel.org/?p=linux/kernel/git/khilman/linux-davinci.git arch/arm/mach-davinci/board-dm355-evm.c | 7 +++ arch/arm/mach-davinci/board-dm644x-evm.c | 5 ++ arch/arm/mach-davinci/board-dm646x-evm.c | 33 ++++++++++++ arch/arm/mach-davinci/dm355.c | 43 +++++++++++++++- arch/arm/mach-davinci/dm644x.c | 36 +++++++++++++- arch/arm/mach-davinci/dm646x.c | 74 +++++++++++++++++++++++++++ arch/arm/mach-davinci/include/mach/asp.h | 35 ++++++++++++- arch/arm/mach-davinci/include/mach/dm355.h | 5 ++ arch/arm/mach-davinci/include/mach/dm644x.h | 2 + arch/arm/mach-davinci/include/mach/dm646x.h | 3 + 10 files changed, 238 insertions(+), 5 deletions(-) diff --git a/arch/arm/mach-davinci/board-dm355-evm.c b/arch/arm/mach-davinci/board-dm355-evm.c index 5ac2f56..78a9604 100644 --- a/arch/arm/mach-davinci/board-dm355-evm.c +++ b/arch/arm/mach-davinci/board-dm355-evm.c @@ -118,6 +118,10 @@ static struct davinci_i2c_platform_data i2c_pdata = { .bus_delay = 0 /* usec */, }; +static struct snd_platform_data dm355_evm_snd_data = { + .clk_name = "asp1", +}; + static int dm355evm_mmc_gpios = -EINVAL; static void dm355evm_mmcsd_gpios(unsigned gpio) @@ -280,6 +284,9 @@ static __init void dm355_evm_init(void) dm355_init_spi0(BIT(0), dm355_evm_spi_info, ARRAY_SIZE(dm355_evm_spi_info)); + + /* DM335 EVM uses ASP1; line-out is a stereo mini-jack */ + dm355_init_asp1(ASP1_TX_EVT_EN | ASP1_RX_EVT_EN, &dm355_evm_snd_data); } static __init void dm355_evm_irq_init(void) diff --git a/arch/arm/mach-davinci/board-dm644x-evm.c b/arch/arm/mach-davinci/board-dm644x-evm.c index d9d4045..30b2fd4 100644 --- a/arch/arm/mach-davinci/board-dm644x-evm.c +++ b/arch/arm/mach-davinci/board-dm644x-evm.c @@ -226,6 +226,10 @@ static struct platform_device ide_dev = { }, }; +static struct snd_platform_data dm644x_evm_snd_data = { + .clk_name = "asp0", +}; + /*----------------------------------------------------------------------*/ /* @@ -667,6 +671,7 @@ static __init void davinci_evm_init(void) davinci_setup_mmc(0, &dm6446evm_mmc_config); davinci_serial_init(&uart_config); + dm644x_init_asp(&dm644x_evm_snd_data); soc_info->emac_pdata->phy_mask = DM644X_EVM_PHY_MASK; soc_info->emac_pdata->mdio_max_freq = DM644X_EVM_MDIO_FREQUENCY; diff --git a/arch/arm/mach-davinci/board-dm646x-evm.c b/arch/arm/mach-davinci/board-dm646x-evm.c index e17de63..575c6ca 100644 --- a/arch/arm/mach-davinci/board-dm646x-evm.c +++ b/arch/arm/mach-davinci/board-dm646x-evm.c @@ -207,6 +207,37 @@ static struct at24_platform_data eeprom_info = { .context = (void *)0x7f00, }; +static u8 dm646x_iis_serializer_direction[] = { + TX_MODE, RX_MODE, INACTIVE_MODE, INACTIVE_MODE, +}; + +static u8 dm646x_dit_serializer_direction[] = { + TX_MODE, +}; + +static struct snd_platform_data dm646x_evm_snd_data[] = { + { + .clk_name = "mcasp0", + .tx_dma_offset = 0x400, + .rx_dma_offset = 0x400, + .op_mode = DAVINCI_MCASP_IIS_MODE, + .num_serializer = ARRAY_SIZE(dm646x_iis_serializer_direction), + .tdm_slots = 2, + .serial_dir = dm646x_iis_serializer_direction, + .eventq_no = EVENTQ_0, + }, + { + .clk_name = "mcasp1", + .tx_dma_offset = 0x400, + .rx_dma_offset = 0, + .op_mode = DAVINCI_MCASP_DIT_MODE, + .num_serializer = ARRAY_SIZE(dm646x_dit_serializer_direction), + .tdm_slots = 32, + .serial_dir = dm646x_dit_serializer_direction, + .eventq_no = EVENTQ_0, + }, +}; + static struct i2c_board_info __initdata i2c_info[] = { { I2C_BOARD_INFO("24c256", 0x50), @@ -240,6 +271,8 @@ static __init void evm_init(void) evm_init_i2c(); davinci_serial_init(&uart_config); + dm646x_init_mcasp0(&dm646x_evm_snd_data[0]); + dm646x_init_mcasp1(&dm646x_evm_snd_data[1]); soc_info->emac_pdata->phy_mask = DM646X_EVM_PHY_MASK; soc_info->emac_pdata->mdio_max_freq = DM646X_EVM_MDIO_FREQUENCY; diff --git a/arch/arm/mach-davinci/dm355.c b/arch/arm/mach-davinci/dm355.c index 9baeed3..1298d8d 100644 --- a/arch/arm/mach-davinci/dm355.c +++ b/arch/arm/mach-davinci/dm355.c @@ -30,6 +30,7 @@ #include <mach/time.h> #include <mach/serial.h> #include <mach/common.h> +#include <mach/asp.h> #include "clock.h" #include "mux.h" @@ -360,8 +361,8 @@ static struct davinci_clk dm355_clks[] = { CLK(NULL, "uart1", &uart1_clk), CLK(NULL, "uart2", &uart2_clk), CLK("i2c_davinci.1", NULL, &i2c_clk), - CLK("soc-audio.0", NULL, &asp0_clk), - CLK("soc-audio.1", NULL, &asp1_clk), + CLK(NULL, "asp0", &asp0_clk), + CLK(NULL, "asp1", &asp1_clk), CLK("davinci_mmc.0", NULL, &mmcsd0_clk), CLK("davinci_mmc.1", NULL, &mmcsd1_clk), CLK(NULL, "spi0", &spi0_clk), @@ -623,6 +624,31 @@ static struct platform_device dm355_edma_device = { .resource = edma_resources, }; +static struct resource dm355_asp1_resources[] = { + { + .start = DAVINCI_ASP1_BASE, + .end = DAVINCI_ASP1_BASE + SZ_8K - 1, + .flags = IORESOURCE_MEM, + }, + { + .start = DAVINCI_DMA_ASP1_TX, + .end = DAVINCI_DMA_ASP1_TX, + .flags = IORESOURCE_DMA, + }, + { + .start = DAVINCI_DMA_ASP1_RX, + .end = DAVINCI_DMA_ASP1_RX, + .flags = IORESOURCE_DMA, + }, +}; + +static struct platform_device dm355_asp1_device = { + .name = "davinci-asp", + .id = -1, + .num_resources = ARRAY_SIZE(dm355_asp1_resources), + .resource = dm355_asp1_resources, +}; + /*----------------------------------------------------------------------*/ static struct map_desc dm355_io_desc[] = { @@ -732,6 +758,19 @@ static struct davinci_soc_info davinci_soc_info_dm355 = { .sram_len = SZ_32K, }; +void __init dm355_init_asp1(u32 evt_enable, struct snd_platform_data *pdata) +{ + /* we don't use ASP1 IRQs, or we'd need to mux them ... */ + if (evt_enable & ASP1_TX_EVT_EN) + davinci_cfg_reg(DM355_EVT8_ASP1_TX); + + if (evt_enable & ASP1_RX_EVT_EN) + davinci_cfg_reg(DM355_EVT9_ASP1_RX); + + dm355_asp1_device.dev.platform_data = pdata; + platform_device_register(&dm355_asp1_device); +} + void __init dm355_init(void) { davinci_common_init(&davinci_soc_info_dm355); diff --git a/arch/arm/mach-davinci/dm644x.c b/arch/arm/mach-davinci/dm644x.c index 1b3aec8..351a131 100644 --- a/arch/arm/mach-davinci/dm644x.c +++ b/arch/arm/mach-davinci/dm644x.c @@ -27,6 +27,7 @@ #include <mach/time.h> #include <mach/serial.h> #include <mach/common.h> +#include <mach/asp.h> #include "clock.h" #include "mux.h" @@ -303,7 +304,7 @@ struct davinci_clk dm644x_clks[] = { CLK("davinci_emac.1", NULL, &emac_clk), CLK("i2c_davinci.1", NULL, &i2c_clk), CLK("palm_bk3710", NULL, &ide_clk), - CLK("soc-audio.0", NULL, &asp_clk), + CLK(NULL, "asp0", &asp_clk), CLK("davinci_mmc.0", NULL, &mmcsd_clk), CLK(NULL, "spi", &spi_clk), CLK(NULL, "gpio", &gpio_clk), @@ -549,6 +550,32 @@ static struct platform_device dm644x_edma_device = { .resource = edma_resources, }; +/* DM6446 EVM uses ASP0; line-out is a pair of RCA jacks */ +static struct resource dm644x_asp_resources[] = { + { + .start = DAVINCI_ASP0_BASE, + .end = DAVINCI_ASP0_BASE + SZ_8K - 1, + .flags = IORESOURCE_MEM, + }, + { + .start = DAVINCI_DMA_ASP0_TX, + .end = DAVINCI_DMA_ASP0_TX, + .flags = IORESOURCE_DMA, + }, + { + .start = DAVINCI_DMA_ASP0_RX, + .end = DAVINCI_DMA_ASP0_RX, + .flags = IORESOURCE_DMA, + }, +}; + +static struct platform_device dm644x_asp_device = { + .name = "davinci-asp", + .id = -1, + .num_resources = ARRAY_SIZE(dm644x_asp_resources), + .resource = dm644x_asp_resources, +}; + /*----------------------------------------------------------------------*/ static struct map_desc dm644x_io_desc[] = { @@ -666,6 +693,13 @@ static struct davinci_soc_info davinci_soc_info_dm644x = { .sram_len = SZ_16K, }; +void __init dm644x_init_asp(struct snd_platform_data *pdata) +{ + davinci_cfg_reg(DM644X_MCBSP); + dm644x_asp_device.dev.platform_data = pdata; + platform_device_register(&dm644x_asp_device); +} + void __init dm644x_init(void) { davinci_common_init(&davinci_soc_info_dm644x); diff --git a/arch/arm/mach-davinci/dm646x.c b/arch/arm/mach-davinci/dm646x.c index e4d7d0f..591aecd 100644 --- a/arch/arm/mach-davinci/dm646x.c +++ b/arch/arm/mach-davinci/dm646x.c @@ -27,6 +27,7 @@ #include <mach/time.h> #include <mach/serial.h> #include <mach/common.h> +#include <mach/asp.h> #include "clock.h" #include "mux.h" @@ -587,6 +588,66 @@ static struct platform_device dm646x_edma_device = { .resource = edma_resources, }; +static struct resource dm646x_mcasp0_resources[] = { + { + .name = "mcasp0", + .start = DAVINCI_DM646X_MCASP0_REG_BASE, + .end = DAVINCI_DM646X_MCASP0_REG_BASE + (SZ_1K << 1) - 1, + .flags = IORESOURCE_MEM, + }, + /* first TX, then RX */ + { + .start = DAVINCI_DM646X_DMA_MCASP0_AXEVT0, + .end = DAVINCI_DM646X_DMA_MCASP0_AXEVT0, + .flags = IORESOURCE_DMA, + }, + { + .start = DAVINCI_DM646X_DMA_MCASP0_AREVT0, + .end = DAVINCI_DM646X_DMA_MCASP0_AREVT0, + .flags = IORESOURCE_DMA, + }, +}; + +static struct resource dm646x_mcasp1_resources[] = { + { + .name = "mcasp1", + .start = DAVINCI_DM646X_MCASP1_REG_BASE, + .end = DAVINCI_DM646X_MCASP1_REG_BASE + (SZ_1K << 1) - 1, + .flags = IORESOURCE_MEM, + }, + /* DIT mode, only TX event */ + { + .start = DAVINCI_DM646X_DMA_MCASP1_AXEVT1, + .end = DAVINCI_DM646X_DMA_MCASP1_AXEVT1, + .flags = IORESOURCE_DMA, + }, + /* DIT mode, dummy entry */ + { + .start = -1, + .end = -1, + .flags = IORESOURCE_DMA, + }, +}; + +static struct platform_device dm646x_mcasp0_device = { + .name = "davinci-mcasp", + .id = 0, + .num_resources = ARRAY_SIZE(dm646x_mcasp0_resources), + .resource = dm646x_mcasp0_resources, +}; + +static struct platform_device dm646x_mcasp1_device = { + .name = "davinci-mcasp", + .id = 1, + .num_resources = ARRAY_SIZE(dm646x_mcasp1_resources), + .resource = dm646x_mcasp1_resources, +}; + +static struct platform_device dm646x_dit_device = { + .name = "spdif-dit", + .id = -1, +}; + /*----------------------------------------------------------------------*/ static struct map_desc dm646x_io_desc[] = { @@ -697,6 +758,19 @@ static struct davinci_soc_info davinci_soc_info_dm646x = { .sram_len = SZ_32K, }; +void __init dm646x_init_mcasp0(struct snd_platform_data *pdata) +{ + dm646x_mcasp0_device.dev.platform_data = pdata; + platform_device_register(&dm646x_mcasp0_device); +} + +void __init dm646x_init_mcasp1(struct snd_platform_data *pdata) +{ + dm646x_mcasp1_device.dev.platform_data = pdata; + platform_device_register(&dm646x_mcasp1_device); + platform_device_register(&dm646x_dit_device); +} + void __init dm646x_init(void) { davinci_common_init(&davinci_soc_info_dm646x); diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index e0abc43..038ecb7 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -5,21 +5,52 @@ #define __ASM_ARCH_DAVINCI_ASP_H #include <mach/irqs.h> +#include <mach/edma.h> -/* Bases of register banks */ +/* Bases of dm644x and dm355 register banks */ #define DAVINCI_ASP0_BASE 0x01E02000 #define DAVINCI_ASP1_BASE 0x01E04000 -/* EDMA channels */ +/* Bases of dm646x register banks */ +#define DAVINCI_DM646X_MCASP0_REG_BASE 0x01D01000 +#define DAVINCI_DM646X_MCASP1_REG_BASE 0x01D01800 + +/* EDMA channels of dm644x and dm355 */ #define DAVINCI_DMA_ASP0_TX 2 #define DAVINCI_DMA_ASP0_RX 3 #define DAVINCI_DMA_ASP1_TX 8 #define DAVINCI_DMA_ASP1_RX 9 +/* EDMA channels of dm646x */ +#define DAVINCI_DM646X_DMA_MCASP0_AXEVT0 6 +#define DAVINCI_DM646X_DMA_MCASP0_AREVT0 9 +#define DAVINCI_DM646X_DMA_MCASP1_AXEVT1 12 + /* Interrupts */ #define DAVINCI_ASP0_RX_INT IRQ_MBRINT #define DAVINCI_ASP0_TX_INT IRQ_MBXINT #define DAVINCI_ASP1_RX_INT IRQ_MBRINT #define DAVINCI_ASP1_TX_INT IRQ_MBXINT +struct snd_platform_data { + char *clk_name; + u32 tx_dma_offset; + u32 rx_dma_offset; + enum dma_event_q eventq_no; /* event queue number */ + unsigned int codec_fmt; + + /* McASP specific fields */ + int tdm_slots; + u8 op_mode; + u8 num_serializer; + u8 *serial_dir; +}; + +#define INACTIVE_MODE 0 +#define TX_MODE 1 +#define RX_MODE 2 + +#define DAVINCI_MCASP_IIS_MODE 0 +#define DAVINCI_MCASP_DIT_MODE 1 + #endif /* __ASM_ARCH_DAVINCI_ASP_H */ diff --git a/arch/arm/mach-davinci/include/mach/dm355.h b/arch/arm/mach-davinci/include/mach/dm355.h index 54903b7..03faaef 100644 --- a/arch/arm/mach-davinci/include/mach/dm355.h +++ b/arch/arm/mach-davinci/include/mach/dm355.h @@ -12,11 +12,16 @@ #define __ASM_ARCH_DM355_H #include <mach/hardware.h> +#include <mach/asp.h> + +#define ASP1_TX_EVT_EN 1 +#define ASP1_RX_EVT_EN 2 struct spi_board_info; void __init dm355_init(void); void dm355_init_spi0(unsigned chipselect_mask, struct spi_board_info *info, unsigned len); +void __init dm355_init_asp1(u32 evt_enable, struct snd_platform_data *pdata); #endif /* __ASM_ARCH_DM355_H */ diff --git a/arch/arm/mach-davinci/include/mach/dm644x.h b/arch/arm/mach-davinci/include/mach/dm644x.h index 15d42b9..8b157ce 100644 --- a/arch/arm/mach-davinci/include/mach/dm644x.h +++ b/arch/arm/mach-davinci/include/mach/dm644x.h @@ -25,6 +25,7 @@ #include <linux/platform_device.h> #include <mach/hardware.h> #include <mach/emac.h> +#include <mach/asp.h> #define DM644X_EMAC_BASE (0x01C80000) #define DM644X_EMAC_CNTRL_OFFSET (0x0000) @@ -34,5 +35,6 @@ #define DM644X_EMAC_CNTRL_RAM_SIZE (0x2000) void __init dm644x_init(void); +void __init dm644x_init_asp(struct snd_platform_data *pdata); #endif /* __ASM_ARCH_DM644X_H */ diff --git a/arch/arm/mach-davinci/include/mach/dm646x.h b/arch/arm/mach-davinci/include/mach/dm646x.h index 1fc764c..0585484 100644 --- a/arch/arm/mach-davinci/include/mach/dm646x.h +++ b/arch/arm/mach-davinci/include/mach/dm646x.h @@ -13,6 +13,7 @@ #include <mach/hardware.h> #include <mach/emac.h> +#include <mach/asp.h> #define DM646X_EMAC_BASE (0x01C80000) #define DM646X_EMAC_CNTRL_OFFSET (0x0000) @@ -22,5 +23,7 @@ #define DM646X_EMAC_CNTRL_RAM_SIZE (0x2000) void __init dm646x_init(void); +void __init dm646x_init_mcasp0(struct snd_platform_data *pdata); +void __init dm646x_init_mcasp1(struct snd_platform_data *pdata); #endif /* __ASM_ARCH_DM646X_H */ -- 1.5.6

3 2

[alsa-devel] [PATCH - quirk for USB Aureon cards 0/1] reworked naming
by Andrea Borgia 15 Jun '09

15 Jun '09

This patch changes yet again the ID for the TTA cards, resulting in a more reasonable name: 1 [Aureon51MkII ]: USB-Audio - Aureon5.1MkII TerraTec Aureon5.1MkII at usb-0000:00:03.0-2, full speed Andrea Borgia (1): rework quirk for TerraTec Aureon USB 5.1 MkII sound/usb/usbquirks.h | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-)

2 3

[alsa-devel] [PATCH 26/33] sound/ps3: Fix checkpatch issues
by Geert Uytterhoeven 15 Jun '09

15 Jun '09

Signed-off-by: Geert Uytterhoeven <Geert.Uytterhoeven(a)sonycom.com> Cc: alsa-devel(a)alsa-project.org Cc: Takashi Iwai <tiwai(a)suse.de> --- sound/ppc/snd_ps3.c | 32 +++++++++++++++----------------- 1 files changed, 15 insertions(+), 17 deletions(-) diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c index f361c26..d660b0f 100644 --- a/sound/ppc/snd_ps3.c +++ b/sound/ppc/snd_ps3.c @@ -165,8 +165,7 @@ static const struct snd_pcm_hardware snd_ps3_pcm_hw = { .fifo_size = PS3_AUDIO_FIFO_SIZE }; -static struct snd_pcm_ops snd_ps3_pcm_spdif_ops = -{ +static struct snd_pcm_ops snd_ps3_pcm_spdif_ops = { .open = snd_ps3_pcm_open, .close = snd_ps3_pcm_close, .prepare = snd_ps3_pcm_prepare, @@ -183,7 +182,7 @@ static int snd_ps3_verify_dma_stop(struct snd_ps3_card_info *card, int dma_ch, done, retries, stop_forced = 0; uint32_t status; - for (dma_ch = 0; dma_ch < 8; dma_ch ++) { + for (dma_ch = 0; dma_ch < 8; dma_ch++) { retries = count; do { status = read_reg(PS3_AUDIO_KICK(dma_ch)) & @@ -259,9 +258,7 @@ static void snd_ps3_kick_dma(struct snd_ps3_card_info *card) /* * convert virtual addr to ioif bus addr. */ -static dma_addr_t v_to_bus(struct snd_ps3_card_info *card, - void * paddr, - int ch) +static dma_addr_t v_to_bus(struct snd_ps3_card_info *card, void *paddr, int ch) { return card->dma_start_bus_addr[ch] + (paddr - card->dma_start_vaddr[ch]); @@ -321,7 +318,7 @@ static int snd_ps3_program_dma(struct snd_ps3_card_info *card, spin_lock_irqsave(&card->dma_lock, irqsave); for (ch = 0; ch < 2; ch++) { start_vaddr = card->dma_next_transfer_vaddr[0]; - for (stage = 0; stage < fill_stages; stage ++) { + for (stage = 0; stage < fill_stages; stage++) { dma_ch = stage * 2 + ch; if (silent) dma_addr = card->null_buffer_start_dma_addr; @@ -619,7 +616,7 @@ static int snd_ps3_change_avsetting(struct snd_ps3_card_info *card) PS3_AUDIO_AO_3WMCTRL_ASOEN(2) | PS3_AUDIO_AO_3WMCTRL_ASOEN(3)), 0); - wmb(); /* ensure the hardware sees the change */ + wmb(); /* ensure the hardware sees the change */ /* wait for actually stopped */ retries = 1000; while ((read_reg(PS3_AUDIO_AO_3WMCTRL) & @@ -798,20 +795,20 @@ static struct snd_kcontrol_new spdif_ctls[] = { { .access = SNDRV_CTL_ELEM_ACCESS_READ, .iface = SNDRV_CTL_ELEM_IFACE_PCM, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,CON_MASK), + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, CON_MASK), .info = snd_ps3_spdif_mask_info, .get = snd_ps3_spdif_cmask_get, }, { .access = SNDRV_CTL_ELEM_ACCESS_READ, .iface = SNDRV_CTL_ELEM_IFACE_PCM, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,PRO_MASK), + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, PRO_MASK), .info = snd_ps3_spdif_mask_info, .get = snd_ps3_spdif_pmask_get, }, { .iface = SNDRV_CTL_ELEM_IFACE_PCM, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT), + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, DEFAULT), .info = snd_ps3_spdif_mask_info, .get = snd_ps3_spdif_default_get, .put = snd_ps3_spdif_default_put, @@ -1020,11 +1017,12 @@ static int __init snd_ps3_driver_probe(struct ps3_system_bus_device *dev) * its size should be lager than PS3_AUDIO_FIFO_STAGE_SIZE * 2 * PAGE_SIZE is enogh */ - if (!(the_card.null_buffer_start_vaddr = - dma_alloc_coherent(&the_card.ps3_dev->core, - PAGE_SIZE, - &the_card.null_buffer_start_dma_addr, - GFP_KERNEL))) { + the_card.null_buffer_start_vaddr = + dma_alloc_coherent(&the_card.ps3_dev->core, + PAGE_SIZE, + &the_card.null_buffer_start_dma_addr, + GFP_KERNEL); + if (!the_card.null_buffer_start_vaddr) { pr_info("%s: nullbuffer alloc failed\n", __func__); goto clean_preallocate; } @@ -1148,7 +1146,7 @@ static irqreturn_t snd_ps3_interrupt(int irq, void *dev_id) SND_PS3_DMA_FILLTYPE_SILENT_FIRSTFILL : SND_PS3_DMA_FILLTYPE_SILENT_RUNNING); snd_ps3_kick_dma(card); - card->silent --; + card->silent--; } else { snd_ps3_program_dma(card, (underflow_occured) ? -- 1.6.2.4

3 7

[alsa-devel] Crashing on x86_64
by WILMES Andreas 14 Jun '09

14 Jun '09

Hello guys, at first a big thank for your grateful work! My X-Fi card even sounds better than on my XP installation. I'm using Songbird on Kubuntu 8.04 with the latest snapshop. By increasing or fast pausing of songs my machine freezes. I can't even get any logs for that. Any ideas?

1 1

[alsa-devel] [PATCH, RFC 0/3] magician fix and UDA1380 i2c device registration change
by Philipp Zabel 14 Jun '09

14 Jun '09

Hi Mark, Patch 1/3 is a follow-up fix to commit a820532002e70e3a06f1ea7133e9b02443d07382 "ASoC: pxa-ssp.c fix clock/frame invert". Patch 2/3 removes the I2C board info registration from the UDA1380 driver. Instead, device instantiation will be done via global i2c_board_info from board files (patch 3/3). At the same time, platform specific configuration is moved to platform data. Common power/reset GPIO handling moves into the codec driver, which will allow to completely power down the codec depending on ASoC power state in the future.

1 0

[PATCH - quirk for USB Aureon cards 1/1] rework quirk for TerraTec Aureon USB 5.1 MkII
by Andrea Borgia 14 Jun '09

14 Jun '09

Signed-off-by: Andrea Borgia <andrea(a)borgia.bo.it> diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index f0f7624..f6f201e 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -1989,7 +1989,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), USB_DEVICE(0x0ccd, 0x0028), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .vendor_name = "TerraTec", - .product_name = "Aureon 5.1 MkII", + .product_name = "Aureon5.1MkII", .ifnum = QUIRK_NO_INTERFACE } }, -- 1.6.0.4 --------------000205060801080305030206--

1 0

[alsa-devel] Maybe the build script for Alsa Libs does not properly install for 64-bit systems? (alsa-lib-1.0.20.tar.bz2)
by bill＠tellytopia.com 14 Jun '09

14 Jun '09

Maybe the build script for Alsa Libs does not properly install for 64-bit systems? (alsa-lib-1.0.20.tar.bz2) Built alsa-lib and did make install; built alsa-driver and did make install; tried to build alsa-utils and found Compile errors when building on a 64-bit platform in Fedora Core 9. Notice the old date on /lib64/libasound.so.2.0.0: [root@localhost alsa]# ls -l /lib64/libasound* lrwxrwxrwx 1 root root 18 2009-06-12 14:29 /lib64/libasound.so.2 -> libasound.so.2.0.0 -rwxr-xr-x 1 root root 920640 2008-04-04 01:26 /lib64/libasound.so.2.0.0 [root@localhost alsa]# ls -l /usr/lib/libasound* -rwxr-xr-x 1 root root 813 2009-06-12 17:02 /usr/lib/libasound.la lrwxrwxrwx 1 root root 18 2009-06-12 17:02 /usr/lib/libasound.so -> libasound.so.2.0.0 lrwxrwxrwx 1 root root 18 2009-06-12 17:02 /usr/lib/libasound.so.2 -> libasound.so.2.0.0 -rwxr-xr-x 1 root root 3544926 2009-06-12 17:02 /usr/lib/libasound.so.2.0.0 Here is the system this is built on: [root@localhost alsa-utils-1.0.20]# uname -a Linux localhost.localdomain 2.6.25-14.fc9.x86_64 #1 SMP Thu May 1 06:06:21 EDT 2008 x86_64 x86_64 x86_64 GNU/Linux Here is the error when compiling utils: --------------------------------------- make[1]: Entering directory `/root/Desktop/alsa/alsa-utils-1.0.20/aplay' if gcc -DHAVE_CONFIG_H -I. -I. -I../include -I../include -g -O2 -MT aplay.o -MD -MP -MF ".deps/aplay.Tpo" -c -o aplay.o aplay.c; \ then mv -f ".deps/aplay.Tpo" ".deps/aplay.Po"; else rm -f ".deps/aplay.Tpo"; exit 1; fi gcc -g -O2 -o aplay aplay.o -lrt -lasound -lm -ldl -lpthread aplay.o: In function `set_params': /root/Desktop/alsa/alsa-utils-1.0.20/aplay/aplay.c:1038: undefined reference to `snd_pcm_hw_params_is_monotonic' aplay.o: In function `do_test_position': /root/Desktop/alsa/alsa-utils-1.0.20/aplay/aplay.c:1455: undefined reference to `snd_pcm_avail_delay' collect2: ld returned 1 exit status make[1]: *** [aplay] Error 1 make[1]: Leaving directory `/root/Desktop/alsa/alsa-utils-1.0.20/aplay' make: *** [all-recursive] Error 1 [root@localhost alsa-utils-1.0.20]# So then link the /lib64 libraries to the /usr/lib equivalents and cross fingers: [root@localhost lib64]# ls -l libasound* lrwxrwxrwx 1 root root 27 2009-06-13 01:25 libasound.2.0.0 -> /usr/lib/libasound.so.2.0.0 lrwxrwxrwx 1 root root 27 2009-06-13 01:26 libasound.so.2 -> /usr/lib/libasound.so.2.0.0 -rwxr-xr-x 1 root root 923208 2008-04-04 01:26 libasound.so.2.0.0-original Then compile utils again. No compile errors. So the compile gave no errors when the "old" /lib64/libasound.so.2 was not used, and the /usr/lib/libasound.so.2 was used instead. Does this mean that the build script for alsa-lib-1.0.20 did not put in the correct library? Or what? Thanks for thoughtful responses. Bill

2 1

[alsa-devel] Request for Device ID addition : 6935G
by Karthik Ramgopal 14 Jun '09

14 Jun '09

Hi, Acer Aspire 6935G sound system is confirmed to be functional with model=acer-aspire-8930g. Kindly add device ID 0x1025 0x0146 to the quirks list in patch_realtek.c . Jack detetction and playback is working without any hda-verbing. Just the internal mic seems to be a problem. Can't hear my voice, can just hear static when I playback a recording. Anyways big thanks to Hector and Takashi for getting it all to work. Regards, Karthik

3 3

Re: [alsa-devel] No jack sense on Intel 82801I / IDT 92HD71B7X in a HP dv4
by Jean-Pierre André 14 Jun '09

14 Jun '09

Hi, > At Fri, 12 Jun 2009 15:01:09 -0300, > Gustavo Maciel Dias Vieira wrote: > >> > >> > Em Qui, 2009-06-11 às 21:27 +0200, Takashi Iwai escreveu: >> >>>> > > > Interrupt 22 stays always with 0. Even when correctly playing >>>> >>> > > sounds. I >>> >>>> > > > tested this on another machine with HD-Audio, and the equivalent >>>> > > > interrupt increased when playing. Also, in this other system >>>> >>> > > interrupt 0 >>> >>>> > > > (timer) is hardly issued, but in this laptop it is firing all the >>>> >>> > > time. >>> >>>> > > > >>>> > > > How can the driver be working without interrupts? >>>> >>> > > >>> > > That's weird. >>> > > >>> > > Did you try enable_msi=1 module option, BTW? >>> >> > >> > Just tried now. The interrupts started flowing and, tada, jack sense >> > works! >> > >> > Thanks, Takashi, that solves it for me. Is there a way to automate this >> > in the driver for the other less technically inclined users? >> > > Well, as I mentioned, it's basically a bug in the lower layer like > ACPI or BIOS. So, the best is to fix them. Using MSI is a kind of > workaround, and I'm wondering whether this is specific to your device > or in general with other HP dv4, since I didn't hear of the similar > bug reports. > > > Takashi > Weird indeed. Actually I had already reported this behaviour, see http://article.gmane.org/gmane.linux.alsa.devel/59809 So far I have not been able to get the interrupts at IRQ22. They still occur at IRQ20. However the problem is hidden when using boot options irqpoll or noapic. This is not a hardware or bios problem, as the sound is correct on Windows. Initially I was using kernel 2.6.27 and alsa-driver 1.0.19, currently using kernel 2.6.29 and alsa-driver 1.0.20 with the same behaviour. I have tried numerous modprobe options. Currently I am using : alias snd-card-0 snd-hda-intel options sound slots=snd-hda-intel options snd-hda-intel model=hp-m4 enable_msi=1 bdl_pos_adj=32 probe_mask=0x1ff As Gustavo seems to have succeeded, can he post kernel and alsa-driver versions, boot parameters and modprobe parameters ? Thanks in advance Jean-Pierre

2 1

[alsa-devel] [RFC PATCH] ASoC: au1x: platform driver restructuring
by Manuel Lauss 13 Jun '09

13 Jun '09

Hi Mark! Please have a look at the patch below and verify it does what you suggested earlier, and if you have no objections, add it to your patchqueue. Both the AC97 and I2S modules are converted to full platform drivers; the PCM (DMA) interface is initialized during platform probe. Run-tested on DB1200 board. Thank you! Manuel Lauss --- From: Manuel Lauss <manuel.lauss(a)gmail.com> Subject: [PATCH] ASoC: au1x: platform driver restructuring Make the AC97/I2S modules full platform drivers and update the sample machine code. Signed-off-by: Manuel Lauss <manuel.lauss(a)gmail.com> --- sound/soc/au1x/dbdma2.c | 24 +++++----- sound/soc/au1x/psc-ac97.c | 103 +++++++++++++++++++++++++----------------- sound/soc/au1x/psc-i2s.c | 103 ++++++++++++++++++++++++------------------ sound/soc/au1x/psc.h | 7 ++- sound/soc/au1x/sample-ac97.c | 48 +++++++++++-------- 5 files changed, 165 insertions(+), 120 deletions(-) diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index 594c6c5..10d21ec 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -1,8 +1,8 @@ /* * Au12x0/Au1550 PSC ALSA ASoC audio support. * - * (c) 2007-2008 MSC Vertriebsges.m.b.H., - * Manuel Lauss <mano(a)roarinelk.homelinux.net> + * (c) 2007-2009 MSC Vertriebsges.m.b.H., + * Manuel Lauss <manuel.lauss(a)gmail.com> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as @@ -331,7 +331,7 @@ static int au1xpsc_pcm_new(struct snd_card *card, return 0; } -static int au1xpsc_pcm_probe(struct platform_device *pdev) +int au1xpsc_asoc_dbdma_init(struct platform_device *pdev) { struct resource *r; int ret; @@ -375,8 +375,9 @@ out1: au1xpsc_audio_pcmdma[PCM_TX] = NULL; return ret; } +EXPORT_SYMBOL_GPL(au1xpsc_asoc_dbdma_init); -static int au1xpsc_pcm_remove(struct platform_device *pdev) +void au1xpsc_asoc_dbdma_exit(void) { int i; @@ -387,36 +388,33 @@ static int au1xpsc_pcm_remove(struct platform_device *pdev) au1xpsc_audio_pcmdma[i] = NULL; } } - - return 0; } +EXPORT_SYMBOL_GPL(au1xpsc_asoc_dbdma_exit); /* au1xpsc audio platform */ struct snd_soc_platform au1xpsc_soc_platform = { .name = "au1xpsc-pcm-dbdma", - .probe = au1xpsc_pcm_probe, - .remove = au1xpsc_pcm_remove, .pcm_ops = &au1xpsc_pcm_ops, .pcm_new = au1xpsc_pcm_new, .pcm_free = au1xpsc_pcm_free_dma_buffers, }; EXPORT_SYMBOL_GPL(au1xpsc_soc_platform); -static int __init au1xpsc_audio_dbdma_init(void) +static int __init au1xpsc_audio_dbdma_load(void) { au1xpsc_audio_pcmdma[PCM_TX] = NULL; au1xpsc_audio_pcmdma[PCM_RX] = NULL; return snd_soc_register_platform(&au1xpsc_soc_platform); } -static void __exit au1xpsc_audio_dbdma_exit(void) +static void __exit au1xpsc_audio_dbdma_unload(void) { snd_soc_unregister_platform(&au1xpsc_soc_platform); } -module_init(au1xpsc_audio_dbdma_init); -module_exit(au1xpsc_audio_dbdma_exit); +module_init(au1xpsc_audio_dbdma_load); +module_exit(au1xpsc_audio_dbdma_unload); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au12x0/Au1550 PSC Audio DMA driver"); -MODULE_AUTHOR("Manuel Lauss <mano(a)roarinelk.homelinux.net>"); +MODULE_AUTHOR("Manuel Lauss"); diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index 479d7bd..ee8e642 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -1,8 +1,8 @@ /* * Au12x0/Au1550 PSC ALSA ASoC audio support. * - * (c) 2007-2008 MSC Vertriebsges.m.b.H., - * Manuel Lauss <mano(a)roarinelk.homelinux.net> + * (c) 2007-2009 MSC Vertriebsges.m.b.H., + * Manuel Lauss <manuel.lauss(a)gmail.com> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as @@ -236,8 +236,32 @@ static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream, return ret; } -static int au1xpsc_ac97_probe(struct platform_device *pdev, - struct snd_soc_dai *dai) +static struct snd_soc_dai_ops au1xpsc_ac97_dai_ops = { + .trigger = au1xpsc_ac97_trigger, + .hw_params = au1xpsc_ac97_hw_params, +}; + +struct snd_soc_dai au1xpsc_ac97_dai = { + .name = "au1xpsc_ac97", + .ac97_control = 1, + .playback = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .capture = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .ops = &au1xpsc_ac97_dai_ops, +}; +EXPORT_SYMBOL_GPL(au1xpsc_ac97_dai); + + +static int __devinit au1xpsc_ac97_probe(struct platform_device *pdev) { int ret; struct resource *r; @@ -287,8 +311,18 @@ static int au1xpsc_ac97_probe(struct platform_device *pdev, * there may not be any codec clock yet. */ - return 0; + ret = au1xpsc_asoc_dbdma_init(pdev); + if (ret) + goto out2; + + ret = snd_soc_register_dai(&au1xpsc_ac97_dai); + if (!ret) + return 0; + + au1xpsc_asoc_dbdma_exit(); +out2: + iounmap(au1xpsc_ac97_workdata->mmio); out1: release_resource(au1xpsc_ac97_workdata->ioarea); kfree(au1xpsc_ac97_workdata->ioarea); @@ -298,9 +332,11 @@ out0: return ret; } -static void au1xpsc_ac97_remove(struct platform_device *pdev, - struct snd_soc_dai *dai) +static int __devexit au1xpsc_ac97_remove(struct platform_device *pdev) { + snd_soc_unregister_dai(&au1xpsc_ac97_dai); + au1xpsc_asoc_dbdma_exit(); + /* disable PSC completely */ au_writel(0, AC97_CFG(au1xpsc_ac97_workdata)); au_sync(); @@ -312,9 +348,11 @@ static void au1xpsc_ac97_remove(struct platform_device *pdev, kfree(au1xpsc_ac97_workdata->ioarea); kfree(au1xpsc_ac97_workdata); au1xpsc_ac97_workdata = NULL; + + return 0; } -static int au1xpsc_ac97_suspend(struct snd_soc_dai *dai) +static int au1xpsc_ac97_suspend(struct platform_device *pdev, pm_message_t m) { /* save interesting registers and disable PSC */ au1xpsc_ac97_workdata->pm[0] = @@ -328,7 +366,7 @@ static int au1xpsc_ac97_suspend(struct snd_soc_dai *dai) return 0; } -static int au1xpsc_ac97_resume(struct snd_soc_dai *dai) +static int au1xpsc_ac97_resume(struct platform_device *pdev) { /* restore PSC clock config */ au_writel(au1xpsc_ac97_workdata->pm[0] | PSC_SEL_PS_AC97MODE, @@ -342,48 +380,31 @@ static int au1xpsc_ac97_resume(struct snd_soc_dai *dai) return 0; } -static struct snd_soc_dai_ops au1xpsc_ac97_dai_ops = { - .trigger = au1xpsc_ac97_trigger, - .hw_params = au1xpsc_ac97_hw_params, -}; - -struct snd_soc_dai au1xpsc_ac97_dai = { - .name = "au1xpsc_ac97", - .ac97_control = 1, - .probe = au1xpsc_ac97_probe, - .remove = au1xpsc_ac97_remove, - .suspend = au1xpsc_ac97_suspend, - .resume = au1xpsc_ac97_resume, - .playback = { - .rates = AC97_RATES, - .formats = AC97_FMTS, - .channels_min = 2, - .channels_max = 2, +static struct platform_driver au1xpsc_ac97_driver = { + .driver = { + .name = "au1xpsc_ac97", + .owner = THIS_MODULE, }, - .capture = { - .rates = AC97_RATES, - .formats = AC97_FMTS, - .channels_min = 2, - .channels_max = 2, - }, - .ops = &au1xpsc_ac97_dai_ops, + .probe = au1xpsc_ac97_probe, + .remove = __devexit_p(au1xpsc_ac97_remove), + .suspend = au1xpsc_ac97_suspend, + .resume = au1xpsc_ac97_resume, }; -EXPORT_SYMBOL_GPL(au1xpsc_ac97_dai); -static int __init au1xpsc_ac97_init(void) +static int __init au1xpsc_ac97_load(void) { au1xpsc_ac97_workdata = NULL; - return snd_soc_register_dai(&au1xpsc_ac97_dai); + return platform_driver_register(&au1xpsc_ac97_driver); } -static void __exit au1xpsc_ac97_exit(void) +static void __exit au1xpsc_ac97_unload(void) { - snd_soc_unregister_dai(&au1xpsc_ac97_dai); + platform_driver_unregister(&au1xpsc_ac97_driver); } -module_init(au1xpsc_ac97_init); -module_exit(au1xpsc_ac97_exit); +module_init(au1xpsc_ac97_load); +module_exit(au1xpsc_ac97_unload); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au12x0/Au1550 PSC AC97 ALSA ASoC audio driver"); -MODULE_AUTHOR("Manuel Lauss <mano(a)roarinelk.homelinux.net>"); +MODULE_AUTHOR("Manuel Lauss"); diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index bb58932..f830f5a 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -1,8 +1,8 @@ /* * Au12x0/Au1550 PSC ALSA ASoC audio support. * - * (c) 2007-2008 MSC Vertriebsges.m.b.H., - * Manuel Lauss <mano(a)roarinelk.homelinux.net> + * (c) 2007-2009 MSC Vertriebsges.m.b.H., + * Manuel Lauss <manuel.lauss(a)gmail.com> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as @@ -262,8 +262,31 @@ static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } -static int au1xpsc_i2s_probe(struct platform_device *pdev, - struct snd_soc_dai *dai) +static struct snd_soc_dai_ops au1xpsc_i2s_dai_ops = { + .trigger = au1xpsc_i2s_trigger, + .hw_params = au1xpsc_i2s_hw_params, + .set_fmt = au1xpsc_i2s_set_fmt, +}; + +struct snd_soc_dai au1xpsc_i2s_dai = { + .name = "au1xpsc_i2s", + .playback = { + .rates = AU1XPSC_I2S_RATES, + .formats = AU1XPSC_I2S_FMTS, + .channels_min = 2, + .channels_max = 8, /* 2 without external help */ + }, + .capture = { + .rates = AU1XPSC_I2S_RATES, + .formats = AU1XPSC_I2S_FMTS, + .channels_min = 2, + .channels_max = 8, /* 2 without external help */ + }, + .ops = &au1xpsc_i2s_dai_ops, +}; +EXPORT_SYMBOL(au1xpsc_i2s_dai); + +static int au1xpsc_i2s_probe(struct platform_device *pdev) { struct resource *r; unsigned long sel; @@ -294,9 +317,7 @@ static int au1xpsc_i2s_probe(struct platform_device *pdev, if (!au1xpsc_i2s_workdata->mmio) goto out1; - /* preserve PSC clock source set up by platform (dev.platform_data - * is already occupied by soc layer) - */ + /* preserve PSC clock source set up by platform */ sel = au_readl(PSC_SEL(au1xpsc_i2s_workdata)) & PSC_SEL_CLK_MASK; au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata)); au_sync(); @@ -312,9 +333,16 @@ static int au1xpsc_i2s_probe(struct platform_device *pdev, * be running yet; depending on clock input for PSC a wait might * time out. */ + ret = au1xpsc_asoc_dbdma_init(pdev); + if (ret) + goto out2; - return 0; + ret = snd_soc_register_dai(&au1xpsc_i2s_dai); + if (!ret) + return 0; +out2: + iounmap(au1xpsc_i2s_workdata->mmio); out1: release_resource(au1xpsc_i2s_workdata->ioarea); kfree(au1xpsc_i2s_workdata->ioarea); @@ -324,9 +352,11 @@ out0: return ret; } -static void au1xpsc_i2s_remove(struct platform_device *pdev, - struct snd_soc_dai *dai) +static int __devexit au1xpsc_i2s_remove(struct platform_device *pdev) { + snd_soc_unregister_dai(&au1xpsc_i2s_dai); + au1xpsc_asoc_dbdma_exit(); + au_writel(0, I2S_CFG(au1xpsc_i2s_workdata)); au_sync(); au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata)); @@ -337,9 +367,11 @@ static void au1xpsc_i2s_remove(struct platform_device *pdev, kfree(au1xpsc_i2s_workdata->ioarea); kfree(au1xpsc_i2s_workdata); au1xpsc_i2s_workdata = NULL; + + return 0; } -static int au1xpsc_i2s_suspend(struct snd_soc_dai *cpu_dai) +static int au1xpsc_i2s_suspend(struct platform_device *pdev, pm_message_t m) { /* save interesting register and disable PSC */ au1xpsc_i2s_workdata->pm[0] = @@ -353,7 +385,7 @@ static int au1xpsc_i2s_suspend(struct snd_soc_dai *cpu_dai) return 0; } -static int au1xpsc_i2s_resume(struct snd_soc_dai *cpu_dai) +static int au1xpsc_i2s_resume(struct platform_device *pdev) { /* select I2S mode and PSC clock */ au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata)); @@ -367,48 +399,31 @@ static int au1xpsc_i2s_resume(struct snd_soc_dai *cpu_dai) return 0; } -static struct snd_soc_dai_ops au1xpsc_i2s_dai_ops = { - .trigger = au1xpsc_i2s_trigger, - .hw_params = au1xpsc_i2s_hw_params, - .set_fmt = au1xpsc_i2s_set_fmt, -}; - -struct snd_soc_dai au1xpsc_i2s_dai = { - .name = "au1xpsc_i2s", - .probe = au1xpsc_i2s_probe, - .remove = au1xpsc_i2s_remove, - .suspend = au1xpsc_i2s_suspend, - .resume = au1xpsc_i2s_resume, - .playback = { - .rates = AU1XPSC_I2S_RATES, - .formats = AU1XPSC_I2S_FMTS, - .channels_min = 2, - .channels_max = 8, /* 2 without external help */ +static struct platform_driver au1xpsc_i2s_driver = { + .driver = { + .name = "au1xpsc_i2s", + .owner = THIS_MODULE, }, - .capture = { - .rates = AU1XPSC_I2S_RATES, - .formats = AU1XPSC_I2S_FMTS, - .channels_min = 2, - .channels_max = 8, /* 2 without external help */ - }, - .ops = &au1xpsc_i2s_dai_ops, + .probe = au1xpsc_i2s_probe, + .remove = __devexit_p(au1xpsc_i2s_remove), + .suspend = au1xpsc_i2s_suspend, + .resume = au1xpsc_i2s_resume, }; -EXPORT_SYMBOL(au1xpsc_i2s_dai); -static int __init au1xpsc_i2s_init(void) +static int __init au1xpsc_i2s_load(void) { au1xpsc_i2s_workdata = NULL; - return snd_soc_register_dai(&au1xpsc_i2s_dai); + return platform_driver_register(&au1xpsc_i2s_driver); } -static void __exit au1xpsc_i2s_exit(void) +static void __exit au1xpsc_i2s_unload(void) { - snd_soc_unregister_dai(&au1xpsc_i2s_dai); + platform_driver_unregister(&au1xpsc_i2s_driver); } -module_init(au1xpsc_i2s_init); -module_exit(au1xpsc_i2s_exit); +module_init(au1xpsc_i2s_load); +module_exit(au1xpsc_i2s_unload); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au12x0/Au1550 PSC I2S ALSA ASoC audio driver"); -MODULE_AUTHOR("Manuel Lauss <mano(a)roarinelk.homelinux.net>"); +MODULE_AUTHOR("Manuel Lauss"); diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h index 8fdb1a0..9ee4f59 100644 --- a/sound/soc/au1x/psc.h +++ b/sound/soc/au1x/psc.h @@ -1,8 +1,8 @@ /* * Au12x0/Au1550 PSC ALSA ASoC audio support. * - * (c) 2007-2008 MSC Vertriebsges.m.b.H., - * Manuel Lauss <mano(a)roarinelk.homelinux.net> + * (c) 2007-2009 MSC Vertriebsges.m.b.H., + * Manuel Lauss <manuel.lauss(a)gmail.com> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as @@ -21,6 +21,9 @@ extern struct snd_soc_dai au1xpsc_i2s_dai; extern struct snd_soc_platform au1xpsc_soc_platform; extern struct snd_ac97_bus_ops soc_ac97_ops; +extern int au1xpsc_asoc_dbdma_init(struct platform_device *pdev); +extern void au1xpsc_asoc_dbdma_exit(void); + struct au1xpsc_audio_data { void __iomem *mmio; diff --git a/sound/soc/au1x/sample-ac97.c b/sound/soc/au1x/sample-ac97.c index 27683eb..86380f6 100644 --- a/sound/soc/au1x/sample-ac97.c +++ b/sound/soc/au1x/sample-ac97.c @@ -1,7 +1,7 @@ /* * Sample Au12x0/Au1550 PSC AC97 sound machine. * - * Copyright (c) 2007-2008 Manuel Lauss <mano(a)roarinelk.homelinux.net> + * Copyright (c) 2007-2009 Manuel Lauss <manuel.lauss(a)gmail.com> * * This program is free software; you can redistribute it and/or modify * it under the terms outlined in the file COPYING at the root of this @@ -44,13 +44,13 @@ static struct snd_soc_dai_link au1xpsc_sample_ac97_dai = { static struct snd_soc_card au1xpsc_sample_ac97_machine = { .name = "Au1xxx PSC AC97 Audio", + .platform = &au1xpsc_soc_platform, /* see dbdma2.c */ .dai_link = &au1xpsc_sample_ac97_dai, .num_links = 1, }; static struct snd_soc_device au1xpsc_sample_ac97_devdata = { .card = &au1xpsc_sample_ac97_machine, - .platform = &au1xpsc_soc_platform, /* see dbdma2.c */ .codec_dev = &soc_codec_dev_ac97, }; @@ -82,7 +82,14 @@ static struct resource au1xpsc_psc1_res[] = { }, }; -static struct platform_device *au1xpsc_sample_ac97_dev; +static struct platform_device psc_ac97_dev = { + .name = "au1xpsc_ac97", + .id = 0, + .resource = au1xpsc_psc1_res, + .num_resources = ARRAY_SIZE(au1xpsc_psc1_res), +}; + +static struct platform_device *au1xpsc_sample_asoc_dev; static int __init au1xpsc_sample_ac97_load(void) { @@ -99,41 +106,42 @@ static int __init au1xpsc_sample_ac97_load(void) au_sync(); #endif - ret = -ENOMEM; - /* setup PSC clock source for AC97 part: external clock provided * by codec. The psc-ac97.c driver depends on this setting! */ au_writel(PSC_SEL_CLK_SERCLK, PSC1_BASE_ADDR + PSC_SEL_OFFSET); au_sync(); - au1xpsc_sample_ac97_dev = platform_device_alloc("soc-audio", -1); - if (!au1xpsc_sample_ac97_dev) - goto out; + /* register the PSC-AC97 platform device */ + ret = platform_device_register(&psc_ac97_dev); + if (ret) + return -ENODEV; - au1xpsc_sample_ac97_dev->resource = - kmemdup(au1xpsc_psc1_res, sizeof(struct resource) * - ARRAY_SIZE(au1xpsc_psc1_res), GFP_KERNEL); - au1xpsc_sample_ac97_dev->num_resources = ARRAY_SIZE(au1xpsc_psc1_res); - au1xpsc_sample_ac97_dev->id = 1; + /* register machine fabric with ASoC core */ + ret = -ENOMEM; + au1xpsc_sample_asoc_dev = platform_device_alloc("soc-audio", -1); + if (!au1xpsc_sample_asoc_dev) + goto out; - platform_set_drvdata(au1xpsc_sample_ac97_dev, + platform_set_drvdata(au1xpsc_sample_asoc_dev, &au1xpsc_sample_ac97_devdata); - au1xpsc_sample_ac97_devdata.dev = &au1xpsc_sample_ac97_dev->dev; - ret = platform_device_add(au1xpsc_sample_ac97_dev); + au1xpsc_sample_ac97_devdata.dev = &au1xpsc_sample_asoc_dev->dev; + ret = platform_device_add(au1xpsc_sample_asoc_dev); if (ret) { - platform_device_put(au1xpsc_sample_ac97_dev); - au1xpsc_sample_ac97_dev = NULL; + platform_device_put(au1xpsc_sample_asoc_dev); + au1xpsc_sample_asoc_dev = NULL; } out: + platform_device_unregister(&psc_ac97_dev); return ret; } static void __exit au1xpsc_sample_ac97_exit(void) { - platform_device_unregister(au1xpsc_sample_ac97_dev); + platform_device_unregister(au1xpsc_sample_asoc_dev); + platform_device_unregister(&psc_ac97_dev); } module_init(au1xpsc_sample_ac97_load); @@ -141,4 +149,4 @@ module_exit(au1xpsc_sample_ac97_exit); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au1xxx PSC sample AC97 machine"); -MODULE_AUTHOR("Manuel Lauss <mano(a)roarinelk.homelinux.net>"); +MODULE_AUTHOR("Manuel Lauss"); -- 1.6.3.1

2 3

[PATCH] ASoC: Add stub suspend and resume calls for ASoC subdevices
by Mark Brown 13 Jun '09

13 Jun '09

Now that ASoC subdevices can be regular devices they can have normal suspend and resume calls from their buses. However, suspending them individually is not desirable since this can lead to problems such as pops and clicks from devices being suspended with their signals being amplified or clocks being stopped suddenly. This will be resolved by having the normal device model suspend and resume calls call into ASoC which will suspend the entire card while any of its components are suspended. At present this is not yet implemented but in order to aid the transition of drivers to the standard device model this patch adds API calls for the notifications. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 5 +++++ sound/soc/soc-core.c | 39 +++++++++++++++++++++++++++++++++++++++ 2 files changed, 44 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 5297ba7..e6704c0 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -192,6 +192,11 @@ void snd_soc_unregister_platform(struct snd_soc_platform *platform); int snd_soc_register_codec(struct snd_soc_codec *codec); void snd_soc_unregister_codec(struct snd_soc_codec *codec); +#ifdef CONFIG_PM +int snd_soc_suspend_device(struct device *dev); +int snd_soc_resume_device(struct device *dev); +#endif + /* pcm <-> DAI connect */ void snd_soc_free_pcms(struct snd_soc_device *socdev); int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e1a920c..4414117 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -788,6 +788,45 @@ static int soc_resume(struct platform_device *pdev) return 0; } +/** + * snd_soc_suspend_device: Notify core of device suspend + * + * @dev: Device being suspended. + * + * In order to ensure that the entire audio subsystem is suspended in a + * coordinated fashion ASoC devices should suspend themselves when + * called by ASoC. When the standard kernel suspend process asks the + * device to suspend it should call this function to initiate a suspend + * of the entire ASoC card. + * + * \note Currently this function is stubbed out. + */ +int snd_soc_suspend_device(struct device *dev) +{ + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_suspend_device); + +/** + * snd_soc_resume_device: Notify core of device resume + * + * @dev: Device being resumed. + * + * In order to ensure that the entire audio subsystem is resumed in a + * coordinated fashion ASoC devices should resume themselves when called + * by ASoC. When the standard kernel resume process asks the device + * to resume it should call this function. Once all the components of + * the card have notified that they are ready to be resumed the card + * will be resumed. + * + * \note Currently this function is stubbed out. + */ +int snd_soc_resume_device(struct device *dev) +{ + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_resume_device); + #else #define soc_suspend NULL #define soc_resume NULL -- 1.6.3.1

1 0

Re: [alsa-devel] No jack sense on Intel 82801I / IDT 92HD71B7X in a HP dv4
by Takashi Iwai 13 Jun '09

13 Jun '09

At Sat, 18 Apr 2009 21:36:35 -0300, Gustavo Vieira wrote: > > Em Sáb, 2009-04-18 às 10:57 +0200, Takashi Iwai escreveu: > > > The jack sense is still broken, though. I'm attaching the alsa-info > > with > > > the snapshot loaded. > > > > Could you attach alsa-info.sh output at both HP plugged / unplugged > > states? > > > > Now things got really interesting. :) > > When trying to get the output you requested I discovered that running > alsa-info with the headphones plugged made them work and silenced the > speakers. Then, removing the HP do not make the sound come back to the > speakers, but running alsa-info again does. That is, I can force the > jack sense by running alsa-info. > > I'm attaching the following files: > * alsa-info-snapshot.txt: The output when the system is turned on, > without HP. When I plug the HP and run alsa-info, the HP work > and the out is the same as this file. > * alsa-info-snapshot-hp-1.txt: The output when HP work. > * alsa-info-snapshot-hp-out.txt: The output of the execution of > alsa-info that makes the speakers work when HP are removed. Hm, what happens if you pass model=hp-dv5 option? Takashi

3 13

[alsa-devel] [PATCH] ASoC: Zoom2: Add machine driver for Zoom2 board
by Lopez Cruz, Misael 12 Jun '09

12 Jun '09

Add support for Zoom2 board. Zoom2 machine driver connects both codec DAIs (audio and voice) to omap3 McBSP ports in the following way: HiFi <-> McBSP2 Voice <-> McBSP3 The zoom2 driver has the following DAPM widgets: * Ext Mic: MAINMIC, SUBMIC (with bias) * Ext Spk: HFL, HFR * Headset Stereophone: HSOL, HSOR * Headset Mic: HSMIC (with bias) * Aux In: AUXL, AUXR Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/omap/Kconfig | 7 + sound/soc/omap/Makefile | 2 + sound/soc/omap/zoom2.c | 301 +++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 310 insertions(+), 0 deletions(-) create mode 100644 sound/soc/omap/zoom2.c diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 675732e..397f3d8 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -64,4 +64,11 @@ config SND_OMAP_SOC_OMAP3_BEAGLE help Say Y if you want to add support for SoC audio on the Beagleboard. +config SND_OMAP_SOC_ZOOM2 + tristate "SoC Audio support for Zoom2" + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_ZOOM2 + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for Soc audio on Zoom2 board. diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 0c9e4ac..5016840 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -13,6 +13,7 @@ snd-soc-omap2evm-objs := omap2evm.o snd-soc-sdp3430-objs := sdp3430.o snd-soc-omap3pandora-objs := omap3pandora.o snd-soc-omap3beagle-objs := omap3beagle.o +snd-soc-zoom2-objs := zoom2.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o @@ -21,3 +22,4 @@ obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o +obj-$(CONFIG_SND_OMAP_SOC_ZOOM2) += snd-soc-zoom2.o diff --git a/sound/soc/omap/zoom2.c b/sound/soc/omap/zoom2.c new file mode 100644 index 0000000..be2e307 --- /dev/null +++ b/sound/soc/omap/zoom2.c @@ -0,0 +1,301 @@ +/* + * zoom2.c -- SoC audio for Zoom2 + * + * Author: Misael Lopez Cruz <x0052729(a)ti.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * 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., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/clk.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <mach/hardware.h> +#include <mach/gpio.h> +#include <mach/mcbsp.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/twl4030.h" + +#define ZOOM2_HEADSET_MUX_GPIO (OMAP_MAX_GPIO_LINES + 15) + +static int zoom2_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops zoom2_ops = { + .hw_params = zoom2_hw_params, +}; + +static int zoom2_hw_voice_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBS_CFM); + if (ret) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops zoom2_voice_ops = { + .hw_params = zoom2_hw_voice_params, +}; + +/* Zoom2 machine DAPM */ +static const struct snd_soc_dapm_widget zoom2_twl4030_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Ext Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_HP("Headset Stereophone", NULL), + SND_SOC_DAPM_LINE("Aux In", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* External Mics: MAINMIC, SUBMIC with bias*/ + {"MAINMIC", NULL, "Mic Bias 1"}, + {"SUBMIC", NULL, "Mic Bias 2"}, + {"Mic Bias 1", NULL, "Ext Mic"}, + {"Mic Bias 2", NULL, "Ext Mic"}, + + /* External Speakers: HFL, HFR */ + {"Ext Spk", NULL, "HFL"}, + {"Ext Spk", NULL, "HFR"}, + + /* Headset Stereophone: HSOL, HSOR */ + {"Headset Stereophone", NULL, "HSOL"}, + {"Headset Stereophone", NULL, "HSOR"}, + + /* Headset Mic: HSMIC with bias */ + {"HSMIC", NULL, "Headset Mic Bias"}, + {"Headset Mic Bias", NULL, "Headset Mic"}, + + /* Aux In: AUXL, AUXR */ + {"Aux In", NULL, "AUXL"}, + {"Aux In", NULL, "AUXR"}, +}; + +static int zoom2_twl4030_init(struct snd_soc_codec *codec) +{ + int ret; + + /* Add Zoom2 specific widgets */ + ret = snd_soc_dapm_new_controls(codec, zoom2_twl4030_dapm_widgets, + ARRAY_SIZE(zoom2_twl4030_dapm_widgets)); + if (ret) + return ret; + + /* Set up Zoom2 specific audio path audio_map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + /* Zoom2 connected pins */ + snd_soc_dapm_enable_pin(codec, "Ext Mic"); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + snd_soc_dapm_enable_pin(codec, "Headset Mic"); + snd_soc_dapm_enable_pin(codec, "Headset Stereophone"); + snd_soc_dapm_enable_pin(codec, "Aux In"); + + /* TWL4030 not connected pins */ + snd_soc_dapm_nc_pin(codec, "CARKITMIC"); + snd_soc_dapm_nc_pin(codec, "DIGIMIC0"); + snd_soc_dapm_nc_pin(codec, "DIGIMIC1"); + + snd_soc_dapm_nc_pin(codec, "OUTL"); + snd_soc_dapm_nc_pin(codec, "OUTR"); + snd_soc_dapm_nc_pin(codec, "EARPIECE"); + snd_soc_dapm_nc_pin(codec, "PREDRIVEL"); + snd_soc_dapm_nc_pin(codec, "PREDRIVER"); + snd_soc_dapm_nc_pin(codec, "CARKITL"); + snd_soc_dapm_nc_pin(codec, "CARKITR"); + + ret = snd_soc_dapm_sync(codec); + + return ret; +} + +static int zoom2_twl4030_voice_init(struct snd_soc_codec *codec) +{ + unsigned short reg; + + /* Enable voice interface */ + reg = codec->read(codec, TWL4030_REG_VOICE_IF); + reg |= TWL4030_VIF_DIN_EN | TWL4030_VIF_DOUT_EN | TWL4030_VIF_EN; + codec->write(codec, TWL4030_REG_VOICE_IF, reg); + + return 0; +} + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link zoom2_dai[] = { + { + .name = "TWL4030 I2S", + .stream_name = "TWL4030 Audio", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], + .init = zoom2_twl4030_init, + .ops = &zoom2_ops, + }, + { + .name = "TWL4030 PCM", + .stream_name = "TWL4030 Voice", + .cpu_dai = &omap_mcbsp_dai[1], + .codec_dai = &twl4030_dai[TWL4030_DAI_VOICE], + .init = zoom2_twl4030_voice_init, + .ops = &zoom2_voice_ops, + }, +}; + +/* Audio machine driver */ +static struct snd_soc_card snd_soc_zoom2 = { + .name = "Zoom2", + .platform = &omap_soc_platform, + .dai_link = zoom2_dai, + .num_links = ARRAY_SIZE(zoom2_dai), +}; + +/* twl4030 setup */ +static struct twl4030_setup_data twl4030_setup = { + .ramp_delay_value = 2, /* 81 ms */ + .sysclk = 26000, +}; + +/* Audio subsystem */ +static struct snd_soc_device zoom2_snd_devdata = { + .card = &snd_soc_zoom2, + .codec_dev = &soc_codec_dev_twl4030, + .codec_data = &twl4030_setup, +}; + +static struct platform_device *zoom2_snd_device; + +static int __init zoom2_soc_init(void) +{ + int ret; + + if (!machine_is_omap_zoom2()) { + pr_debug("Not Zoom2!\n"); + return -ENODEV; + } + printk(KERN_INFO "Zoom2 SoC init\n"); + + zoom2_snd_device = platform_device_alloc("soc-audio", -1); + if (!zoom2_snd_device) { + printk(KERN_ERR "Platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(zoom2_snd_device, &zoom2_snd_devdata); + zoom2_snd_devdata.dev = &zoom2_snd_device->dev; + *(unsigned int *)zoom2_dai[0].cpu_dai->private_data = 1; /* McBSP2 */ + *(unsigned int *)zoom2_dai[1].cpu_dai->private_data = 2; /* McBSP3 */ + + ret = platform_device_add(zoom2_snd_device); + if (ret) + goto err1; + + BUG_ON(gpio_request(ZOOM2_HEADSET_MUX_GPIO, "hs_mux") < 0); + gpio_direction_output(ZOOM2_HEADSET_MUX_GPIO, 0); + + return 0; + +err1: + printk(KERN_ERR "Unable to add platform device\n"); + platform_device_put(zoom2_snd_device); + + return ret; +} +module_init(zoom2_soc_init); + +static void __exit zoom2_soc_exit(void) +{ + gpio_free(ZOOM2_HEADSET_MUX_GPIO); + + platform_device_unregister(zoom2_snd_device); +} +module_exit(zoom2_soc_exit); + +MODULE_AUTHOR("Misael Lopez Cruz <x0052729(a)ti.com>"); +MODULE_DESCRIPTION("ALSA SoC Zoom2"); +MODULE_LICENSE("GPL"); + -- 1.5.4.3

2 1

[alsa-devel] snd-ctxfi surround volume
by The Source 12 Jun '09

12 Jun '09

Hello again. I have another minor problem with this driver. When I use hw:0,1 (surround) device for output, the volume even with slider on maximum is much lower than through front (hw:0,0) device. Can you do something about it? Thank you.

2 1

Re: [alsa-devel] SB X-Fi driver fixes (Re: Testers wanted:
by bill＠tellytopia.com 12 Jun '09

12 Jun '09

Looking at the board, it is model SB1040, with CA0110-IBG LF chip. Is that an hda type chip? > -------- Original Message -------- > Subject: Re: [alsa-devel] SB X-Fi driver fixes (Re: Testers wanted: > From: Takashi Iwai <tiwai(a)suse.de> > Date: Thu, June 11, 2009 6:54 pm > To: bill(a)tellytopia.com > Cc: "Gernot Kohlhaas" <gernot.kohlhaas(a)googlemail.com>, > alsa-devel(a)alsa-project.org > At Thu, 11 Jun 2009 14:28:02 -0700, > bill(a)tellytopia.com wrote: > > > > Yes, I have loaded the module snd-ctfxi many times, > > and unloaded snd-hda-intl. > > Also blacklisted snd-hda-intel, and put snd-ctxfi into modprobe.conf. > Are you sure that your board has really emu20k1 or emu20k2 chip? > X-Fi isn't always with these chips. Some are HD-audio only (thus > snd-hda-intel is the right driver) and some are with CA0106 chip. > Takashi

3 2

[alsa-devel] Backported sbxfi driver (UNTESTED!)
by Takashi Iwai 12 Jun '09

12 Jun '09

Hi, $SUBJECT is now on my sound-unstable git tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable-2.6.git together with other experimental patches. If you're using 2.6.27-rc* git tree, pull the master branch of the tree above into yours, and run make oldconfig. That is, % cd /your/git-tree % git pull git://...../sound-unstable-2.6.git master If you are not using 2.6.27-rc*, or not familiar with git, you can try alsa-driver-unstable snapshot tarball available at ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/ Run configure, make and make install as usual ALSA-driver tarball. (I didn't check whether the tarball correctly includes the sbxfi stuff. It should have been generated automatically. If not, wait for a while. If it still doesn't include sbxfi code, please report. I'll fix it tomorrow morning.) The driver is built only for 2.6.26 or later. If you have an older kernel, edit alsa-driver*/kconfig-vers and change the version of CONFIG_SND_SBXFI to 2.6.24 or whatever you want. Then run ./gitcompile, instead of configure in this case to update the configure script. **NOTE** The driver is totally untested. It's just compiled without errors, but not reviewed after a quick writing. So, don't expect it ever runs at the first try. A crash is highly possible. There are some build conditions found in sound/pci/sbxfi/sbxfi.c, starting with XXX_*. You can change it if you want. As default, it's for non-fullduplex but accept different rates. Not sure whether this works at all. Any test- (and better debugging-) reports are appreciated. thanks, Takashi

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Re: [alsa-devel] plug:dmix slows playback
by Jaroslav Kysela 12 Jun '09

12 Jun '09

On Fri, 12 Jun 2009, Steve Langstaff wrote: >> From: alsa-devel-bounces(a)alsa-project.org [mailto:alsa-devel- >> bounces(a)alsa-project.org] On Behalf Of Jaroslav Kysela > >> On Fri, 12 Jun 2009, Steve Langstaff wrote: > >>> Slave: Hardware PCM card 0 'LM4549' device 0 subdevice 0 >> >>> msbits : 16 >> >> If I remember correctly, this value was 32 in your dmix setup. >> Could you try command: >> >> aplay -D"plug:'dmix:FORMAT=S16_LE'" <filename> > > Yes, that works fine - is there something wrong with my configuration do you > think? It seems that 32-bit samples are not handled correctly in driver. What's your hardware? Output from alsa-info.sh script might be helpful for identification. Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

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