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[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.

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[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

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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(-)

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[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

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[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(-)

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[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

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[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.

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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

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[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

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[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

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