January 2010 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Error patching file pcm_native.c
by Delusion's Master 20 Nov '24

20 Nov '24

Hi, first of all I apologize in case you receive multiple copies of this message, but I forgot to specify a subject in the original mail. I have a problem compiling the latest alsa driver snapshots. Specifically, it's a problem when patching pcm_native.c: [..] copying file alsa-kernel/core/info.c patching file info.c copying file alsa-kernel/core/pcm.c patching file pcm.c Hunk #3 succeeded at 739 (offset 2 lines). Hunk #5 succeeded at 954 (offset 2 lines). Hunk #7 succeeded at 1045 (offset 2 lines). copying file alsa-kernel/core/pcm_native.c patching file pcm_native.c Hunk #4 succeeded at 1544 (offset -25 lines). Hunk #5 succeeded at 2827 (offset 12 lines). Hunk #6 succeeded at 2878 (offset 10 lines). Hunk #7 FAILED at 2906. Hunk #8 FAILED at 2922. Hunk #9 succeeded at 2915 (offset -23 lines). Hunk #10 succeeded at 3036 (offset 10 lines). Hunk #11 succeeded at 3027 (offset -23 lines). Hunk #12 succeeded at 3074 (offset 10 lines). Hunk #13 succeeded at 3098 (offset -23 lines). Hunk #14 succeeded at 3143 (offset 10 lines). Hunk #15 succeeded at 3124 (offset -23 lines). Hunk #16 succeeded at 3214 (offset 10 lines). Hunk #17 succeeded at 3208 (offset -23 lines). Hunk #18 succeeded at 3271 (offset 10 lines). Hunk #19 succeeded at 3304 (offset -23 lines). Hunk #20 succeeded at 3386 (offset 10 lines). Hunk #21 succeeded at 3411 (offset -23 lines). Hunk #22 succeeded at 3456 (offset 10 lines). Hunk #23 succeeded at 3450 (offset -23 lines). Hunk #24 succeeded at 3525 (offset 10 lines). Hunk #25 succeeded at 3644 (offset -23 lines). 2 out of 25 hunks FAILED -- saving rejects to file pcm_native.c.rej make[2]: *** [pcm_native.c] Error 1 It's some days I get this, but I waited to report this since I thought it could be a temporary problem. My OS is Fedora 11 x86_64. Previous snapshots didn't give me this problem, but I can't say which one broke the compilation, since I usually try updated snapshots only at kernel updates. I searched for this error and the only relevant report I could find is this one: http://mailman.alsa-project.org/pipermail/alsa-devel/2009-April/016420.html a report from last April. What could the problem be? Thanks _________________________________________________________________ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx…

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[alsa-devel] hda-intel missing mixer channels
by Orm Finnendahl 26 Aug '13

26 Aug '13

Hi, I sent this to alsa-user a week ago but didn't get any reply. Does anybody know, what is responsible for the missing faders? Did the driver change or is it some module loading mechanism in the distro which prevents the faders from showing up in alsamixer? This is so strange. Please help! -- Orm -------------------------------------------------------------------- Hi everybody, after the last update of Ubuntu (intrepid to jaunty) on my Lenovo Ideapad my microphone input doesn't work anymore (neither internal nor external). I found out that alsamixer lacks quite a lot of faders in the newer kernel/module compared to the older one. Some of the faders now unavailable were crucial for Mic input to work on the older system so I'd very much like to get them back ;-) Some more infos: Soundcard: Intel 82801G (ICH7 Family) Chip: Realtek ALC269 Module: snd_hda_intel (in both kernels) Older kernel: Ubuntu stock 2.6.27-14-generic Newer kernel: Ubuntu stock 2.6.28-13-rt Is there any way to make the missing controls available or did the driver change and I have to compile the older driver for the new kernel? Yours, Orm

2 3

[alsa-devel] [ASoc] TDM over one SSC channel on AT91 possible?
by Friedrich Lobenstock 24 Jun '11

24 Jun '11

Hello! The background: I'm just designing an ATMEL AT91RM9200 based board which got two TLV320AIC33 on board. It is either possible to hook up the two TLV320AIC33 to a separate SSC (synchronous serial control) channel each or use one SSC channel and configure it for TDM (time division multiplex) mode. There is already a driver for TLV320AIC14k/TLV320AIC2k in the development tree, so adding support for the TLV320AIC33 should be feasible. What I'm not sure about is the fact if ASOC is prepared for using channels, especially SSC, in TDM mode. Is the core ASOC structure prepared to handle that? Would we be better off instantiate one drive twice each time with different parameters for the SSC channel? Thanks for any feedback. -- MfG / Regards Friedrich Lobenstock _________________________________________________________ SCOTTY Group Austria GmbH a SCOTTY Group plc company, http://www.scottygroup.com/ _________________________________________________________

4 3

[alsa-devel] Conexant CX20561 - Microphone problems
by Shawn Starr 19 Apr '11

19 Apr '11

Hello ALSA developers, I am having problems with the microphone on the following hardware this is on a Lenovo W500 laptop. Card: HDA Intel Chip: Conexant CX20561 (Hermosa) Here is the problems im noticing: 1) If I use ALSA 'default' with the microphone I get noise interference on output [high/low/high/low]. If I switch this to use the "ALSA Capture on HDA Intel [CONEXANT Analog] I get interference except it is the standard 60Hz electrical noise [steady noise] 2) Using the internal/external microphones gives an inconsistent volume. It also does not seem to work with the microphone cutting out so much. I need to adjust Digital + another microphone but it still drops capturing. Alsamixer shows: Digital/Docking Microphone/External Microphone/Internal Microphone Here is the module options I give to the sound driver: options snd-intel-hda enable_msi=1 power_save=10 power_save_controller=true model=laptop index=0 Assuming power saving/controller causes the microphone to cut out or not. 3) I'm now aware that this card has no pinouts for PCM capture https://bugzilla.redhat.com/show_bug.cgi?id=502293 Thanks for any help, Shawn.

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[alsa-devel] CA0110 taints kernel
by Vedran Miletić 13 Apr '11

13 Apr '11

Hi, I experience frequent kernel crashes with 2.6.31 and CA0110. I know this card doesn't work, and there is no problem with that; however, it crashing non-stop is not very nice. Also, card doesn't get any controls (and I believe it does get some controls when it's the only card using HDA driver, that is when I disable onboard one). dmesg is here: http://pastebin.com/m7c53562f lspci: 02:00.0 PCI bridge: Creative Labs Device 7006 (prog-if 00 [Normal decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=02, secondary=03, subordinate=03, sec-latency=64 Memory behind bridge: fe900000-fe9fffff Capabilities: [50] Power Management version 3 Capabilities: [60] MSI: Enable- Count=1/16 Maskable- 64bit+ Capabilities: [80] Subsystem: Creative Labs Device 0010 Capabilities: [90] Express PCI/PCI-X Bridge, MSI 00 Capabilities: [100] Advanced Error Reporting Kernel modules: shpchp 03:00.0 Audio device: Creative Labs [SB X-Fi Xtreme Audio] CA0110-IBG Subsystem: Creative Labs Device 0018 Flags: bus master, medium devsel, latency 64, IRQ 16 Memory at fe9fc000 (32-bit, non-prefetchable) [size=16K] Capabilities: [dc] Power Management version 3 Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel Any ideas? -- Vedran Miletić P.S. Please CC me, as I'm not subscribed to mailing list at the moment.

2 2

[alsa-devel] idt STAC927x adat supported (sigmatel HDA codec) ?
by Eric 03 Mar '11

03 Mar '11

Is the ADAT output of the IDT (sigmatel) STAC927x HDA codec supported in alsa ? If not could someone please try to add this incredibly nice feature in the existing alsa driver ? Thank you very much

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[alsa-devel] davinci-i2c,pcm updates
by Troy Kisky 27 Sep '10

27 Sep '10

This patch set is against git://git.alsa-project.org/alsa-kernel.git but the last patch in series won't compile until the sram allocator patch is available.

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[alsa-devel] [PATCH 2.6.32.3] hda_intel: ALSA HD Audio patch for Intel Cougar Point DeviceIDs
by Seth Heasley 13 Sep '10

13 Sep '10

This patch adds the Intel Cougar Point (PCH) HD Audio Controller DeviceIDs. Signed-off-by: Seth Heasley <seth.heasley(a)intel.com> --- linux-2.6.32.3/sound/pci/hda/hda_intel.c.orig 2010-01-06 15:07:45.000000000 -0800 +++ linux-2.6.32.3/sound/pci/hda/hda_intel.c 2010-01-07 13:10:28.000000000 -0800 @@ -116,6 +116,7 @@ "{Intel, ICH9}," "{Intel, ICH10}," "{Intel, PCH}," + "{Intel, CPT}," "{Intel, SCH}," "{ATI, SB450}," "{ATI, SB600}," @@ -2648,6 +2649,8 @@ { PCI_DEVICE(0x8086, 0x3a6e), .driver_data = AZX_DRIVER_ICH }, /* PCH */ { PCI_DEVICE(0x8086, 0x3b56), .driver_data = AZX_DRIVER_ICH }, + /* CPT */ + { PCI_DEVICE(0x8086, 0x1c20), .driver_data = AZX_DRIVER_ICH }, /* SCH */ { PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH }, /* ATI SB 450/600 */

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Re: [alsa-devel] Patchmix-style application for EMU cards
by Camilo Polymeris 30 Aug '10

30 Aug '10

Hello > That is more or less what I had figured out, but it how are playback_1 to > > _16 channels routed to DSP0-15? It seems that it depends on the setting > of > > Front, Surround, LFE, etc. mixer controls. If they are turned up, I > don't > > understand what happens, sound to playback channels appears on various > DSP > > buses, at different levels. > > Really? Well, I didn't much experiment with surround mode. In source code, > I > can see that there are up to playback 8 channels at hw:0,0 with can be > controlled by faders in alsamixer. I tried surround40 and surroun51 on my > emu1616m and it worked as expected. The channles were mapped to first > couple > of DSP ports in a correct order. In fact this should work the same as > audigy > does, except the result is passed to DSP ports instead of physical > outputs. > On my card, playback channels 1 & 2 (hw:0,3) appear on all DSP busses associated with surround. Say you turn up the "Side" fader in alsamixer, channels 1 & 2 will appear at the relevant DSPs (6-7, I think) Whatever, for now I am just keeping all surround controls down (that is, Front, Surround, Center, Side & LFE). > Anyway, I think if there is another layer of routing done in audigy DSP, > it > shouldn't be used in EMU specific application. Or at least not for now. > The > most important thing is to provide all the switches and FPGA routing > scheme > for each EMU model. Or, to be precise, three routing schemes for each EMU > model, because for "double speed" (e.g. 96kHz) and "quadro speed" (e.g. > 192kHz) modes of operation the numebr of both physical and virtual > channels > changes. > I think the switches and enums for the card models are rather trivial, one can handle it with a customized version of qamix (done with a simple XML file), for example. A separate application would not be necessary for only that. > > If I turn the Front/Surround/LFE/etc. faders down, I can mix playback > > channels to DSP using the "Multichannel PCM Send/Routing/Volume" faders, > in > > a more or less intuitive way, besides that routing values do not map > > directly to DSP buses (i.e. 14-15 map to DSP0-1, but 2-3 map to DSP2-3). > Handling of this controls motivated me to write the application in the first place. No sound at all will appear if I don't set the appropiate controls for each playback channel: Multichannel PCM Volume (full), Multichannel PCM Send (an index of 8 controls, the first 4 select the target DSP) & Multichannel PCM Send Volume (again faders 0-3 of 8) By the way, only DSP0-7 work in this way. I have no idea how to make DSP8-31 work. On which card you did tested it? The driver was primary written for the > 1820M/1212M V1, so some things may not work well on other models and need > to > be fixed. I have a 1212m, don't know which version, but I think it's V1-- it does have firewire. I am using driver version 1.0.16. Thanks, Camilo.

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[alsa-devel] [PATCH 1/4] ASoC: DaVinci: remove requirement that dma_params is 1st in structure
by Troy Kisky 13 Jul '10

13 Jul '10

Remove requirement that dma_params is 1st in the structures davinci_audio_dev and davinci_mcbsp_dev. Signed-off-by: Troy Kisky <troy.kisky(a)boundarydevices.com> --- sound/soc/davinci/davinci-i2s.c | 6 +----- sound/soc/davinci/davinci-mcasp.c | 1 + sound/soc/davinci/davinci-mcasp.h | 5 ----- sound/soc/davinci/davinci-pcm.c | 7 ++++--- 4 files changed, 6 insertions(+), 13 deletions(-) diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 4ae7070..9e69a4e 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -98,11 +98,6 @@ enum { }; struct davinci_mcbsp_dev { - /* - * dma_params must be first because rtd->dai->cpu_dai->private_data - * is cast to a pointer of an array of struct davinci_pcm_dma_params in - * davinci_pcm_open. - */ struct davinci_pcm_dma_params dma_params[2]; void __iomem *base; #define MOD_DSP_A 0 @@ -547,6 +542,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = res->start; davinci_i2s_dai.private_data = dev; + davinci_i2s_dai.dma_data = dev->dma_params; ret = snd_soc_register_dai(&davinci_i2s_dai); if (ret != 0) goto err_free_mem; diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 5d1f98a..5023915 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -907,6 +907,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dma_data->channel = res->start; davinci_mcasp_dai[pdata->op_mode].private_data = dev; + davinci_mcasp_dai[pdata->op_mode].dma_data = dev->dma_params; davinci_mcasp_dai[pdata->op_mode].dev = &pdev->dev; ret = snd_soc_register_dai(&davinci_mcasp_dai[pdata->op_mode]); diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 9d179cc..582c924 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -39,11 +39,6 @@ enum { }; struct davinci_audio_dev { - /* - * dma_params must be first because rtd->dai->cpu_dai->private_data - * is cast to a pointer of an array of struct davinci_pcm_dma_params in - * davinci_pcm_open. - */ struct davinci_pcm_dma_params dma_params[2]; void __iomem *base; int sample_rate; diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index c73a915..c2c2e85 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -238,10 +238,11 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream) struct davinci_runtime_data *prtd; int ret = 0; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct davinci_pcm_dma_params *pa = rtd->dai->cpu_dai->private_data; - struct davinci_pcm_dma_params *params = &pa[substream->stream]; - if (!params) + struct davinci_pcm_dma_params *pa = rtd->dai->cpu_dai->dma_data; + struct davinci_pcm_dma_params *params; + if (!pa) return -ENODEV; + params = &pa[substream->stream]; snd_soc_set_runtime_hwparams(substream, &davinci_pcm_hardware); /* ensure that buffer size is a multiple of period size */ -- 1.5.6.3

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[alsa-devel] snd-hda-intel: no data when recording from mic, "IRQ timing workaround is activated for card #0. Suggest a bigger bdl_pos_adj" logged in kern.log
by Aleksander Adamowski 17 Mar '10

17 Mar '10

Hi! I think I have some interesting problems to analyse for Takashi. Maybe my sound card model is a non typical one, but I have a continuous stream of capture-related problems with it. It's an integrated sound card on my Gigabyte GA-MA74GM-S2H Rev. 1.1 motherboard. "grep ^Codec /proc/asound/card0/codec#0" shows: Codec: Realtek ALC888 I'm attaching the full contents of /proc/asound/card0/codec#0 for identification. But first, some background. With this particular card, I've been continuously hit by the "azx_get_response timeout, switching to single_cmd mode" problem, described here: http://www.kernel.org/pub/linux/kernel/people/tiwai/docs/HD-Audio.html#_cod… However, in my case setting probe_mask=1 (experimentally determined) didn't make the problem go away, although it seemed to reduce its frequency (might be just a coincidence). Whenever the dreaded "azx_get_response timeout, switching to single_cmd mode" message appeared in the kernel log, recording stopped working. The card got into the bad state where it would not record sound, and arecord would be giving a stream of "overrun" error messages: arecord -f cd output.wav Recording WAVE 'a.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo overrun!!! (at least 27.876 ms long) overrun!!! (at least 26.965 ms long) overrun!!! (at least 16.169 ms long) overrun!!! (at least 11.749 ms long) overrun!!! (at least 11.298 ms long) overrun!!! (at least 11.814 ms long) overrun!!! (at least 11.190 ms long) overrun!!! (at least 11.190 ms long) overrun!!! (at least 11.747 ms long) overrun!!! (at least 11.819 ms long) overrun!!! (at least 11.149 ms long) overrun!!! (at least 11.257 ms long) overrun!!! (at least 11.315 ms long) overrun!!! (at least 11.651 ms long) overrun!!! (at least 11.745 ms long) overrun!!! (at least 11.235 ms long) overrun!!! (at least 11.169 ms long) overrun!!! (at least 11.776 ms long) overrun!!! (at least 11.391 ms long) overrun!!! (at least 11.669 ms long) overrun!!! (at least 11.714 ms long) overrun!!! (at least 11.671 ms long) overrun!!! (at least 11.190 ms long) What's notable, sound playback would continue to work OK. After being fed up with interrupted Skype conversations (where I weren't immediately aware that the other side stopped hearing me, since I was hearing them), I've decided to upgrade do kernel 2.6.29 (citing the http://www.kernel.org/pub/linux/kernel/people/tiwai/docs/HD-Audio.html#_cod… document: "Since 2.6.29 kernel, the driver has a more robust probing method, so this error might happen rarely, though"). After upgrading to 2.6.29 I've removed any extra snd-hda-intel options from modprobe config files. Now the azx_get_response timeout seems to be gone, but I'm experiencing another problem: after some powered up time, recording starts giving empty streams - there's no error, but audio data has zero lengh. e.g. when I arecord, I get a .wav file that is only 44 bytes long (which seems to be the length of the WAV header). When I make a Skype test call to echo123, and it starts playing back my message, it's zero length too - the playback end signal can be heard immediately after the playback start signal. This problem seems to concide with the following error message in the kernel log: hda-intel: IRQ timing workaround is activated for card #0. Suggest a bigger bdl_pos_adj. Do you have any idea what could be the cause and how to debug this issue further? -- Best Regards, Aleksander Adamowski http://olo.org.pl

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[alsa-devel] [PATCH 3/5] ASoC: DaVinci: CQ93VC Voice Codec
by miguel.aguilar＠ridgerun.com 08 Mar '10

08 Mar '10

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> Currently the DM365 is the only SoC that includes this Voice Codec. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cq93vc.c | 288 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cq93vc.h | 29 +++++ 4 files changed, 323 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/cq93vc.c create mode 100644 sound/soc/codecs/cq93vc.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 52b005f..0daca22 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -21,6 +21,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4535 if I2C select SND_SOC_AK4642 if I2C select SND_SOC_AK4671 if I2C + select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS4270 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 @@ -108,6 +109,9 @@ config SND_SOC_AK4642 config SND_SOC_AK4671 tristate +config SND_SOC_CQ0093VC + tristate + # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index dbaecb1..dff91fb 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -8,6 +8,7 @@ snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o +snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs4270-objs := cs4270.o snd-soc-cx20442-objs := cx20442.o snd-soc-l3-objs := l3.o @@ -64,6 +65,7 @@ obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o +obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c new file mode 100644 index 0000000..37f6ba7 --- /dev/null +++ b/sound/soc/codecs/cq93vc.c @@ -0,0 +1,288 @@ +/* + * ALSA SoC CQ0093 Voice Codec Driver for DaVinci platforms + * + * Copyright (C) 2010 Texas Instruments, Inc + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.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; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/io.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/platform_device.h> +#include <linux/device.h> +#include <linux/clk.h> +#include <linux/mfd/davinci_voicecodec.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include <mach/dm365.h> + +#include "cq93vc.h" + +static inline unsigned int cq93vc_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + struct davinci_vc *davinci_vc = codec->control_data; + + return readl(davinci_vc->base + reg); +} + +static inline int cq93vc_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + struct davinci_vc *davinci_vc = codec->control_data; + + writel(value, davinci_vc->base + reg); + + return 0; +} + +static const struct snd_kcontrol_new cq93vc_snd_controls[] = { + SOC_SINGLE("PGA Capture Volume", DAVINCI_VC_REG05, 0, 0x03, 0), + SOC_SINGLE("Mono DAC Playback Volume", DAVINCI_VC_REG09, 0, 0x3f, 0), +}; + +static int cq93vc_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u8 reg = cq93vc_read(codec, DAVINCI_VC_REG09) & ~DAVINCI_VC_REG09_MUTE; + + if (mute) + cq93vc_write(codec, DAVINCI_VC_REG09, + reg | DAVINCI_VC_REG09_MUTE); + else + cq93vc_write(codec, DAVINCI_VC_REG09, reg); + + return 0; +} + +static int cq93vc_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct davinci_vc *davinci_vc = codec->control_data; + + switch (freq) { + case 22579200: + case 27000000: + case 33868800: + davinci_vc->cq93vc.sysclk = freq; + return 0; + } + + return -EINVAL; +} + +static int cq93vc_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + cq93vc_write(codec, DAVINCI_VC_REG12, + DAVINCI_VC_REG12_POWER_ALL_ON); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + cq93vc_write(codec, DAVINCI_VC_REG12, + DAVINCI_VC_REG12_POWER_ALL_OFF); + break; + case SND_SOC_BIAS_OFF: + /* force all power off */ + cq93vc_write(codec, DAVINCI_VC_REG12, + DAVINCI_VC_REG12_POWER_ALL_OFF); + break; + } + codec->bias_level = level; + + return 0; +} + +#define CQ93VC_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000) +#define CQ93VC_FORMATS (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE) + +static struct snd_soc_dai_ops cq93vc_dai_ops = { + .digital_mute = cq93vc_mute, + .set_sysclk = cq93vc_set_dai_sysclk, +}; + +struct snd_soc_dai cq93vc_dai = { + .name = "CQ93VC", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CQ93VC_RATES, + .formats = CQ93VC_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CQ93VC_RATES, + .formats = CQ93VC_FORMATS,}, + .ops = &cq93vc_dai_ops, +}; +EXPORT_SYMBOL_GPL(cq93vc_dai); + +static int cq93vc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + cq93vc_set_bias_level(codec, codec->suspend_bias_level); + + return 0; +} + +static struct snd_soc_codec *cq93vc_codec; + +static int cq93vc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct device *dev = &pdev->dev; + struct snd_soc_codec *codec; + int ret; + + socdev->card->codec = cq93vc_codec; + codec = socdev->card->codec; + + /* Register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(dev, "%s: failed to create pcms\n", pdev->name); + return ret; + } + + /* Set controls */ + snd_soc_add_controls(codec, cq93vc_snd_controls, + ARRAY_SIZE(cq93vc_snd_controls)); + + /* Off, with power on */ + cq93vc_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +static int cq93vc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_cq93vc = { + .probe = cq93vc_probe, + .remove = cq93vc_remove, + .resume = cq93vc_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_cq93vc); + +static __init int cq93vc_codec_probe(struct platform_device *pdev) +{ + struct davinci_vc *davinci_vc = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret; + + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) { + dev_dbg(davinci_vc->dev, + "could not allocate memory for codec data\n"); + return -ENOMEM; + } + + davinci_vc->cq93vc.codec = codec; + + cq93vc_dai.dev = &pdev->dev; + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + codec->dev = &pdev->dev; + codec->name = "CQ93VC"; + codec->owner = THIS_MODULE; + codec->read = cq93vc_read; + codec->write = cq93vc_write; + codec->set_bias_level = cq93vc_set_bias_level; + codec->dai = &cq93vc_dai; + codec->num_dai = 1; + codec->control_data = davinci_vc; + + cq93vc_codec = codec; + + ret = snd_soc_register_dai(&cq93vc_dai); + if (ret != 0) { + dev_err(davinci_vc->dev, "could register dai\n"); + goto fail1; + } + return 0; + +fail1: + kfree(codec); + cq93vc_codec = NULL; + + return ret; +} + +static int __devexit cq93vc_codec_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + snd_soc_unregister_dai(&cq93vc_dai); + + kfree(codec); + cq93vc_codec = NULL; + + return 0; +} + +static struct platform_driver cq93vc_codec_driver = { + .driver = { + .name = "cq93vc", + .owner = THIS_MODULE, + }, + .probe = cq93vc_codec_probe, + .remove = __devexit_p(cq93vc_codec_remove), +}; + +static __init int cq93vc_init(void) +{ + return platform_driver_probe(&cq93vc_codec_driver, cq93vc_codec_probe); +} +module_init(cq93vc_init); + +static __exit void cq93vc_exit(void) +{ + platform_driver_unregister(&cq93vc_codec_driver); +} +module_exit(cq93vc_exit); + +MODULE_DESCRIPTION("Texas Instruments DaVinci ASoC CQ0093 Voice Codec Driver"); +MODULE_AUTHOR("Miguel Aguilar"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cq93vc.h b/sound/soc/codecs/cq93vc.h new file mode 100644 index 0000000..845b196 --- /dev/null +++ b/sound/soc/codecs/cq93vc.h @@ -0,0 +1,29 @@ +/* + * ALSA SoC CQ0093 Voice Codec Driver for DaVinci platforms + * + * Copyright (C) 2010 Texas Instruments, Inc + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.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; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef _CQ93VC_H +#define _CQ93VC_H + +extern struct snd_soc_dai cq93vc_dai; +extern struct snd_soc_codec_device soc_codec_dev_cq93vc; + +#endif -- 1.6.0.4

3 3

[alsa-devel] 2.6.30-rc8 skipping regression also affects cmipci
by Frédéric Brière 07 Mar '10

07 Mar '10

Hi everybody, I was only recently able to upgrade from 2.6.26 to 2.6.32 (long story). In so doing, I noticed infrequent occurrences of audio skipping, which I managed to bisect down to c87d973. It would appear that this commit also affects cmipci (CMI8738), albeit not as badly as it did for intel8x0. (I guess I'm the only person left on Earth still using a stand-alone PCI sound card, which is why this was never reported. <g>) Here's a sample log output after turning xrun debugging on: [ 171.555860] PCM: hw_ptr skipping! (pos=6341, delta=16383, period=1024, jdelta=0/85) [ 181.114680] PCM: hw_ptr skipping! (pos=12529, delta=16370, period=1024, jdelta=0/85) [ 181.114770] PCM: hw_ptr skipping! (pos=12534, delta=16375, period=1024, jdelta=0/85) [ 1713.774316] PCM: hw_ptr skipping! (pos=9456, delta=16369, period=1024, jdelta=0/85) [ 1713.774346] PCM: hw_ptr skipping! (pos=9458, delta=16371, period=1024, jdelta=0/85) [ 1713.774409] PCM: hw_ptr skipping! (pos=9461, delta=16374, period=1024, jdelta=0/85) [ 1713.774613] PCM: hw_ptr skipping! (pos=9470, delta=16383, period=1024, jdelta=0/85) [ 1831.092000] PCM: hw_ptr skipping! (pos=1712, delta=16373, period=1024, jdelta=0/85) [ 1953.347453] PCM: hw_ptr skipping! (pos=3723, delta=16316, period=1024, jdelta=0/84) [ 1953.347465] PCM: hw_ptr skipping! (pos=3724, delta=16317, period=1024, jdelta=0/84) [ 1953.347485] PCM: hw_ptr skipping! (pos=3725, delta=16318, period=1024, jdelta=0/84) You'll notice that this occurs much less frequently than it did for intel8x0. However, I left MPlayer on pause for the night, and at some point the kernel started spewing several syslog messages every couple of seconds: [25332.211286] PCM: hw_ptr skipping! [Q] (pos=4, delta=-1073740800, period=1024, jdelta=5/44744/0) [25332.232637] PCM: hw_ptr skipping! [Q] (pos=1029, delta=-1073739771, period=1024, jdelta=6/44749/0) [25332.253957] PCM: hw_ptr skipping! [Q] (pos=2053, delta=-1073738747, period=1024, jdelta=5/44755/0) [25332.275292] PCM: hw_ptr skipping! [Q] (pos=3077, delta=-1073737723, period=1024, jdelta=5/44760/0) [25332.296616] PCM: hw_ptr skipping! [Q] (pos=4101, delta=-1073736699, period=1024, jdelta=6/44765/0) [25332.317932] PCM: hw_ptr skipping! [Q] (pos=5125, delta=-1073735675, period=1024, jdelta=5/44771/0) [25332.339269] PCM: hw_ptr skipping! [Q] (pos=6149, delta=-1073734651, period=1024, jdelta=5/44776/0) [25332.360600] PCM: hw_ptr skipping! [Q] (pos=7173, delta=-1073733627, period=1024, jdelta=6/44781/0) [25332.381942] PCM: hw_ptr skipping! [Q] (pos=8197, delta=-1073732603, period=1024, jdelta=5/44787/0) [25332.403261] PCM: hw_ptr skipping! [Q] (pos=9221, delta=-1073731579, period=1024, jdelta=5/44792/0) [25337.244901] PCM: hw_ptr skipping! [Q] (pos=12293, delta=-1073728507, period=1024, jdelta=6/44808/0) [25337.266223] PCM: hw_ptr skipping! [Q] (pos=13317, delta=-1073727483, period=1024, jdelta=5/44813/0) This kept going on until I quit MPlayer (which, as I said, was not actually playing anything all this time). Not that skipped silence is much of an issue, but I thought this might add useful information. Let me know if I can be of any help in resolving this. -- If you want to travel around the world and be invited to speak at a lot of different places, just write a Unix operating system. -- Linus Torvalds

2 3

Re: [alsa-devel] HD-Audio device unsuppored (Nvidia G2xx-series GPU implemented)
by Stefan Ringel 02 Mar '10

02 Mar '10

I forgot this attachment 1.patch and alsa-info.txt.

4 20

Re: [alsa-devel] [alsa-cvslog] alsa-kernel: Official ALSA project GIT repository for Linux 2.6 branch, master now at v2.6.31-rc8-407-gfc9e4a3
by Takashi Iwai 26 Feb '10

26 Feb '10

At Tue, 6 Oct 2009 08:36:04 +0200 (CEST), noreply-git(a)alsa-project.org wrote: > commit fc9e4a3811567f8d1d4c263a89e27b52c90d6098 > Author: Clemens Ladisch <clemens(a)ladisch.de> > AuthorDate: Tue Oct 6 08:21:04 2009 +0200 > Commit: Clemens Ladisch <clemens(a)ladisch.de> > CommitDate: Tue Oct 6 08:21:04 2009 +0200 > > sound: via82xx: move DXS volume controls to PCM interface > > The "VIA DXS" controls are actually volume controls that apply to the > four PCM substreams, so we better indicate this connection by moving the > controls to the PCM interface. > > Commit b452e08e73c0e3dbb0be82130217be4b7084299e in 2.6.30 broke the > restoring of these volumes by "amixer restore" that most distributions > use; the renaming in this patch cures that regression by preventing > amixer from applying the old, wrong volume levels to the new controls. > http://bugzilla.kernel.org/show_bug.cgi?id=14151 > http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=532613 > > Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> > Cc: <stable(a)kernel.org> Well, this control was once removed, and then revived due to complains by users who have used it in mixer. Moving this to PCM iface does almost the same thing, disabling an easy way to access it. I'm afraid that this patch might be regarded as another "regression". For such an old driver, the compatibility is the most important item. If any, we must provide a way to back to the old behavior. thanks, Takashi > > ----------------------------------------------------------------------- > > Diffstat: > sound/pci/via82xx.c | 27 ++++++++++++++++++--------- > 1 files changed, 18 insertions(+), 9 deletions(-) > > > Patch -------------- (max 1500 lines) > diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c > index acfa476..91683a3 100644 > --- a/sound/pci/via82xx.c > +++ b/sound/pci/via82xx.c > @@ -1626,7 +1626,7 @@ static int snd_via8233_dxs_volume_get(struct snd_kcontrol *kcontrol, > struct snd_ctl_elem_value *ucontrol) > { > struct via82xx *chip = snd_kcontrol_chip(kcontrol); > - unsigned int idx = snd_ctl_get_ioff(kcontrol, &ucontrol->id); > + unsigned int idx = kcontrol->id.subdevice; > > ucontrol->value.integer.value[0] = VIA_DXS_MAX_VOLUME - chip->playback_volume[idx][0]; > ucontrol->value.integer.value[1] = VIA_DXS_MAX_VOLUME - chip->playback_volume[idx][1]; > @@ -1646,7 +1646,7 @@ static int snd_via8233_dxs_volume_put(struct snd_kcontrol *kcontrol, > struct snd_ctl_elem_value *ucontrol) > { > struct via82xx *chip = snd_kcontrol_chip(kcontrol); > - unsigned int idx = snd_ctl_get_ioff(kcontrol, &ucontrol->id); > + unsigned int idx = kcontrol->id.subdevice; > unsigned long port = chip->port + 0x10 * idx; > unsigned char val; > int i, change = 0; > @@ -1705,11 +1705,12 @@ static struct snd_kcontrol_new snd_via8233_pcmdxs_volume_control __devinitdata = > }; > > static struct snd_kcontrol_new snd_via8233_dxs_volume_control __devinitdata = { > - .name = "VIA DXS Playback Volume", > - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, > + .iface = SNDRV_CTL_ELEM_IFACE_PCM, > + .device = 0, > + /* .subdevice set later */ > + .name = "PCM Playback Volume", > .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | > SNDRV_CTL_ELEM_ACCESS_TLV_READ), > - .count = 4, > .info = snd_via8233_dxs_volume_info, > .get = snd_via8233_dxs_volume_get, > .put = snd_via8233_dxs_volume_put, > @@ -1936,10 +1937,18 @@ static int __devinit snd_via8233_init_misc(struct via82xx *chip) > } > else /* Using DXS when PCM emulation is enabled is really weird */ > { > - /* Standalone DXS controls */ > - err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_via8233_dxs_volume_control, chip)); > - if (err < 0) > - return err; > + for (i = 0; i < 4; ++i) { > + struct snd_kcontrol *kctl; > + > + kctl = snd_ctl_new1( > + &snd_via8233_dxs_volume_control, chip); > + if (!kctl) > + return -ENOMEM; > + kctl->id.subdevice = i; > + err = snd_ctl_add(chip->card, kctl); > + if (err < 0) > + return err; > + } > } > } > /* select spdif data slot 10/11 */ > ------- End of patch > > hooks/update > --- > Git Source Code Management System > hooks/update refs/heads/master \ > 62f08ef9739c5cb5fff5b5bdec3bc02d522b9a12 \ > fc9e4a3811567f8d1d4c263a89e27b52c90d6098 > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ > Alsa-cvslog mailing list > Alsa-cvslog(a)lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/alsa-cvslog >

6 22

[alsa-devel] Getting pcm_usb_stream plugin to know its limits.
by michael＠cadilhac.name 22 Feb '10

22 Feb '10

Hi there everyone, I've been fighting for a few hours with a USB sound device (namely, Tascam US-144), to get it to work with Audacity (this not being the relevant part). At first, Audacity died in a painful segfault, without any explanation, when associated with the usb_stream plugin[1]. Audacity uses PortAudio, which checks for a few possible parameters of audio rate to find the list of accepted ones. It tries to set a parameter (snd_pcm_hw_params), and if it fails, closes the device, and continues its way. Suppose the `prepare` act of the usb_stream is aborted because of wrong parameters. Then, PortAudio tries to close the device, by calling snd_pcm_close. This in turn calls snd_pcm_drop, which calls the fast_op drop (that of ioplug), which is snd_pcm_ioplug_drop, which then calls the *stop* callback of the usb_stream (snd_pcm_us_stop). The stream not being properly initialized, us->uus->write_area is nil, and the memset fails. This is the modification I did for that purpose : Next, Audacity stopped segfaulting but still didn't work: it couldn't manage to configure the plugin properly. Here's how PortAudio searches for an acceptable audio rate. For each possible sample rate, it gets a full hw_params for the device: snd_pcm_hw_params_any( pcm, hwParams ); then set the sample rate to the hwParams. It then tries to set those params to the device: snd_pcm_hw_params( pcm, hwParams ); The period_size parameter is left untouched, and by default, the pcm inherits of a high value (maybe the highest). Now the problem is the following. pcm_usb_stream says that the maximum accepted period_size is 64*4096: (err = snd_pcm_ioplug_set_param_minmax(&us->io, SND_PCM_IOPLUG_HW_PERIOD_BYTES, 128, 64*4096) BUT the us122l driver[2] (from 2.6.32) explicitly prohibits period_size greater than 0x3000: if ((cfg->sample_rate != 44100 && cfg->sample_rate != 48000 && (!high_speed || (cfg->sample_rate != 88200 && cfg->sample_rate != 96000))) || cfg->frame_size != 6 || cfg->period_frames > 0x3000) err = -EINVAL; This is now what happens in PortAudio: PortAudio tries each and every sound rate with the "default" period_size, this being greater than 0x3000. It follows that no sound rate will be accepted, because of the period_size, and the device fails to initialize. I'm not sure whose fault is this, and for the time being, I just did the following: Well, that's all folks. Happy new year to everyone. Footnotes: [1] Configured as pcm.!usb_stream { type usb_stream card "1" } which works with a fine-tuned aplay. [2] linux/sound/usb/usx2y/us122l.c -- Michaël `Micha' Cadilhac (LITQ, U. de Montréal) -- http://michael.cadilhac.name || La culture, c'est comme l'amour. || Faut y aller à petits coups au début pour bien en jouir plus tard. || -- Pierre Desproges

2 9

[alsa-devel] [PATCH 0/5] Echoaudio - add suspend/resume support
by Giuliano Pochini 15 Feb '10

15 Feb '10

The following patch set adds suspend/resume support to all echoaudio drivers. It has been tested on an IndigoIOx and a Gina3G. Rawmidi device has been tested partially. PCM and controls interfaces have been tested more and I think it's reasonably safe. I also have a Gina24 and a Gina3G, but I didn't test suspend/resume with them because I couldn't manage to make PM test system work on my pmac. Echoaudio soundcards do not support power management and Dx PCI states. Anyway, the card can enter a "comatose" state by sending a command to the DSP that is supposed to turn parts of the hardware off and reduce power consumption. Doing that causes the loss of the card's internal state, thus it is impossible, upon resume, to make the card continue from the exact point where it was suspended. The driver restores all controls and PCM settings and the applications have to call prepare() to restart the substreams. Rawmidi is restarted automatically, but some data in the i/o buffers may be lost. -- Giuliano.

2 4

Re: [alsa-devel] HP 6930p mute button/led out of sync; fixed by suspend/resume cycle
by drago01 12 Feb '10

12 Feb '10

On Sun, Jan 10, 2010 at 3:31 PM, drago01 <drago01(a)gmail.com> wrote: > Hi, > > I experience an odd issue with my HP6930p laptop. > > After boot pressing the mute button/led causes the audio to be muted > but the sate is not reflected in software (i.e the mixer does not show > the control as muted). > However pressing after a suspend/resume cycle it works as expected. > The hardware mute button/led and the software control are in sync > (pressing the hw button toggles the mixer to mute/unmute; muting the > software mixer results into the led turning red). > > This was with 2.6.31.9 and 2.6.32.1; also installing alsa-drivers from > source does not change this. > > It seems something is not initialized correctly during boot up but is > done on resume, any pointers on how to debug this? > (Resending to correct list)

2 5

[alsa-devel] Sound distortion with Realtek ALC889A
by Damien Brossard 11 Feb '10

11 Feb '10

Hi. I recently bought a Gigabyte GA-MA78GM-S2H motherboard. This board uses the (in)famous (from what i've read for now) alc889a chip. While playing 5.1 content, I can hear noise or distortion coming from the center channel. I can not test my configuration with another OS, so I can't fully dismiss a HW problem. However, i made several tests with 5.1 and 2.0 speakers which points me to soundcard problem. I've made some tests with speaker-test (speaker-test -D surround51 -c 6) : 1) Each time a sound is played either on front left channel or rear left channel, a noise can be heared on the center channel (and subwoofer, this is confirmed pluging a stereo headphone on the center/lfe jack). No noise when a sound is played on right channels (front or rear). 2) In alsamixer, if I set the volume to 0 for the front or rear left channels, i can hear the noise on center/lfe when a sound is "played" on this channels 3) In alsamixer, if I mute the front or rear left channels, i can hear no more noise on center/lfe Complementary tests have been made with alsaplayer and using a simple mp3 file. In this configuration (stereo output), center and rear channels play the same sound as front channel. 1) If all channels are unmuted, I hear no noise. 2) If either the front channel or the rear channel are muted, I can hear the noise on center channel. I tried this with alsa-driver 1.0.20 and 1.0.21 with the same result, I recently went back to 1.0.20 due to kernel upgrade. Things that seem strange to me (maybe not so strange to alsa dev :)) : 1) alsamixer shows volumes in percentage, however when I set a channel volume to 50% and the master volume to 50%, I hope to find a resulting volume of 25%, not 0. 2) lspci reports my soundcard subsystem id as 1458:a022, but in /proc/asound/card0/codec#0 the subsystem id is 1458:a002 Googling around made me find few references (in alsa-devel list) to a similar problem : http://mailman.alsa-project.org/pipermail/alsa-devel/2008-July/008942.html http://mailman.alsa-project.org/pipermail/alsa-devel/2008-August/009754.html The distortion / noise described here is exactly what i can hear with my system. Apparently the user solved the issue buying another soundcard. I'd like to be sure it is the only way to go. Attached is the file produced by alsa-info.sh Thanks in advance. Best regards Damien Brossard

2 3

[alsa-devel] [PATCH] ALSA: ASoC: cs4270: enable regulators at probe time
by Daniel Mack 09 Feb '10

09 Feb '10

Enable the bulk regulators at probe time so we can safely disable them again when going to suspend without confusing the reference counter. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Timur Tabi <timur(a)freescale.com> Cc: Liam Girdwood <lrg(a)slimlogic.co.uk> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/cs4270.c | 10 ++++++++++ 1 files changed, 10 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 8b54575..70c5edb 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -617,8 +617,17 @@ static int cs4270_probe(struct platform_device *pdev) if (ret < 0) goto error_free_pcms; + ret = regulator_bulk_enable(ARRAY_SIZE(cs4270->supplies), + cs4270->supplies); + if (ret < 0) + goto error_free_regulators; + return 0; +error_free_regulators: + regulator_bulk_free(ARRAY_SIZE(cs4270->supplies), + cs4270->supplies); + error_free_pcms: snd_soc_free_pcms(socdev); @@ -638,6 +647,7 @@ static int cs4270_remove(struct platform_device *pdev) struct cs4270_private *cs4270 = codec->private_data; snd_soc_free_pcms(socdev); + regulator_bulk_disable(ARRAY_SIZE(cs4270->supplies), cs4270->supplies); regulator_bulk_free(ARRAY_SIZE(cs4270->supplies), cs4270->supplies); return 0; -- 1.6.3.3

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[alsa-devel] [PATCH 0/4] ASOC: Add AM3517 EVM support in ASOC
by Anuj Aggarwal 05 Feb '10

05 Feb '10

This patch series add support for OMAP3517 / AM3517 EVM in ASOC. It also enables the required drivers - I2C and McBSP, along with Alsa SoC subsystem, in the default configuration for the EVM. Anuj Aggarwal (4): Audio: Adding OMAP3517 / AM3517 EVM support in ASOC Audio: Modifying Kconfig/Makefile for AM3517 EVM Audio: Modifying board-evm file for audio codec Audio: Modifying the defconfig for OMAP3517 / AM3517 EVM arch/arm/configs/am3517_evm_defconfig | 78 ++++++++++++- arch/arm/mach-omap2/board-am3517evm.c | 21 ++++ sound/soc/omap/Kconfig | 9 ++ sound/soc/omap/Makefile | 2 + sound/soc/omap/am3517evm.c | 202 +++++++++++++++++++++++++++++++++ 5 files changed, 309 insertions(+), 3 deletions(-) create mode 100644 sound/soc/omap/am3517evm.c

4 5

[alsa-devel] [PATCH 1/4 v4] ASoC: add a WM8978 codec driver
by Guennadi Liakhovetski 03 Feb '10

03 Feb '10

The WM8978 codec from Wolfson Microelectronics is very similar to wm8974, but is stereo and also has some differences in pin configuration and internal signal routing. This driver is based on wm8974 and takes the differences into account. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> --- v3 -> v4: 1. Removed "Mute" from two controls 2. Removed left-over "freq" parameter description 3. Now also suitably selecting f_PLLOUT, if OPCLK is not used 4. Switching PLL off for configuration 5. Switching PLL off and on and putting codec to sleep and waking up in suspend() / resume() diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 62ff26a..0aad72f 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -57,6 +57,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8961 if I2C select SND_SOC_WM8971 if I2C select SND_SOC_WM8974 if I2C + select SND_SOC_WM8978 if I2C select SND_SOC_WM8988 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8990 if I2C select SND_SOC_WM8993 if I2C @@ -230,6 +231,9 @@ config SND_SOC_WM8971 config SND_SOC_WM8974 tristate +config SND_SOC_WM8978 + tristate + config SND_SOC_WM8988 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index ea98354..fbd290e 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -44,6 +44,7 @@ snd-soc-wm8960-objs := wm8960.o snd-soc-wm8961-objs := wm8961.o snd-soc-wm8971-objs := wm8971.o snd-soc-wm8974-objs := wm8974.o +snd-soc-wm8978-objs := wm8978.o snd-soc-wm8988-objs := wm8988.o snd-soc-wm8990-objs := wm8990.o snd-soc-wm8993-objs := wm8993.o @@ -103,6 +104,7 @@ obj-$(CONFIG_SND_SOC_WM8960) += snd-soc-wm8960.o obj-$(CONFIG_SND_SOC_WM8961) += snd-soc-wm8961.o obj-$(CONFIG_SND_SOC_WM8971) += snd-soc-wm8971.o obj-$(CONFIG_SND_SOC_WM8974) += snd-soc-wm8974.o +obj-$(CONFIG_SND_SOC_WM8978) += snd-soc-wm8978.o obj-$(CONFIG_SND_SOC_WM8988) += snd-soc-wm8988.o obj-$(CONFIG_SND_SOC_WM8990) += snd-soc-wm8990.o obj-$(CONFIG_SND_SOC_WM8993) += snd-soc-wm8993.o diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c new file mode 100644 index 0000000..b86ef47 --- /dev/null +++ b/sound/soc/codecs/wm8978.c @@ -0,0 +1,1124 @@ +/* + * wm8978.c -- WM8978 ALSA SoC Audio Codec driver + * + * Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> + * Copyright (C) 2007 Carlos Munoz <carlos(a)kenati.com> + * Copyright 2006-2009 Wolfson Microelectronics PLC. + * Based on wm8974 and wm8990 by Liam Girdwood <lrg(a)slimlogic.co.uk> + * + * 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/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <asm/div64.h> + +#include "wm8978.h" + +static struct snd_soc_codec *wm8978_codec; + +/* wm8978 register cache. Note that register 0 is not included in the cache. */ +static const u16 wm8978_reg[WM8978_CACHEREGNUM] = { + 0x0000, 0x0000, 0x0000, 0x0000, /* 0x00...0x03 */ + 0x0050, 0x0000, 0x0140, 0x0000, /* 0x04...0x07 */ + 0x0000, 0x0000, 0x0000, 0x00ff, /* 0x08...0x0b */ + 0x00ff, 0x0000, 0x0100, 0x00ff, /* 0x0c...0x0f */ + 0x00ff, 0x0000, 0x012c, 0x002c, /* 0x10...0x13 */ + 0x002c, 0x002c, 0x002c, 0x0000, /* 0x14...0x17 */ + 0x0032, 0x0000, 0x0000, 0x0000, /* 0x18...0x1b */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 0x1c...0x1f */ + 0x0038, 0x000b, 0x0032, 0x0000, /* 0x20...0x23 */ + 0x0008, 0x000c, 0x0093, 0x00e9, /* 0x24...0x27 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 0x28...0x2b */ + 0x0033, 0x0010, 0x0010, 0x0100, /* 0x2c...0x2f */ + 0x0100, 0x0002, 0x0001, 0x0001, /* 0x30...0x33 */ + 0x0039, 0x0039, 0x0039, 0x0039, /* 0x34...0x37 */ + 0x0001, 0x0001, /* 0x38...0x3b */ +}; + +/* codec private data */ +struct wm8978_priv { + struct snd_soc_codec codec; + unsigned int f_pllout; + unsigned int f_mclk; + unsigned int f_256fs; + unsigned int f_opclk; + enum wm8978_sysclk_src sysclk; + u16 reg_cache[WM8978_CACHEREGNUM]; +}; + +static const char *wm8978_companding[] = {"Off", "NC", "u-law", "A-law"}; +static const char *wm8978_eqmode[] = {"Capture", "Playback"}; +static const char *wm8978_bw[] = {"Narrow", "Wide"}; +static const char *wm8978_eq1[] = {"80Hz", "105Hz", "135Hz", "175Hz"}; +static const char *wm8978_eq2[] = {"230Hz", "300Hz", "385Hz", "500Hz"}; +static const char *wm8978_eq3[] = {"650Hz", "850Hz", "1.1kHz", "1.4kHz"}; +static const char *wm8978_eq4[] = {"1.8kHz", "2.4kHz", "3.2kHz", "4.1kHz"}; +static const char *wm8978_eq5[] = {"5.3kHz", "6.9kHz", "9kHz", "11.7kHz"}; +static const char *wm8978_alc3[] = {"ALC", "Limiter"}; +static const char *wm8978_alc1[] = {"Off", "Right", "Left", "Both"}; + +static const SOC_ENUM_SINGLE_DECL(adc_compand, WM8978_COMPANDING_CONTROL, 1, + wm8978_companding); +static const SOC_ENUM_SINGLE_DECL(dac_compand, WM8978_COMPANDING_CONTROL, 3, + wm8978_companding); +static const SOC_ENUM_SINGLE_DECL(eqmode, WM8978_EQ1, 8, wm8978_eqmode); +static const SOC_ENUM_SINGLE_DECL(eq1, WM8978_EQ1, 5, wm8978_eq1); +static const SOC_ENUM_SINGLE_DECL(eq2bw, WM8978_EQ2, 8, wm8978_bw); +static const SOC_ENUM_SINGLE_DECL(eq2, WM8978_EQ2, 5, wm8978_eq2); +static const SOC_ENUM_SINGLE_DECL(eq3bw, WM8978_EQ3, 8, wm8978_bw); +static const SOC_ENUM_SINGLE_DECL(eq3, WM8978_EQ3, 5, wm8978_eq3); +static const SOC_ENUM_SINGLE_DECL(eq4bw, WM8978_EQ4, 8, wm8978_bw); +static const SOC_ENUM_SINGLE_DECL(eq4, WM8978_EQ4, 5, wm8978_eq4); +static const SOC_ENUM_SINGLE_DECL(eq5, WM8978_EQ5, 5, wm8978_eq5); +static const SOC_ENUM_SINGLE_DECL(alc3, WM8978_ALC_CONTROL_3, 8, wm8978_alc3); +static const SOC_ENUM_SINGLE_DECL(alc1, WM8978_ALC_CONTROL_1, 7, wm8978_alc1); + +static const DECLARE_TLV_DB_SCALE(digital_tlv, -12750, 50, 1); +static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); +static const DECLARE_TLV_DB_SCALE(inpga_tlv, -1200, 75, 0); +static const DECLARE_TLV_DB_SCALE(spk_tlv, -5700, 100, 0); +static const DECLARE_TLV_DB_SCALE(boost_tlv, -1500, 300, 1); + +static const struct snd_kcontrol_new wm8978_snd_controls[] = { + + SOC_SINGLE("Digital Loopback Switch", + WM8978_COMPANDING_CONTROL, 0, 1, 0), + + SOC_ENUM("ADC Companding", adc_compand), + SOC_ENUM("DAC Companding", dac_compand), + + SOC_DOUBLE("DAC Inversion Switch", WM8978_DAC_CONTROL, 0, 1, 1, 0), + + SOC_DOUBLE_R_TLV("PCM Volume", + WM8978_LEFT_DAC_DIGITAL_VOLUME, WM8978_RIGHT_DAC_DIGITAL_VOLUME, + 0, 255, 0, digital_tlv), + + SOC_SINGLE("High Pass Filter Switch", WM8978_ADC_CONTROL, 8, 1, 0), + SOC_SINGLE("High Pass Cut Off", WM8978_ADC_CONTROL, 4, 7, 0), + SOC_DOUBLE("ADC Inversion Switch", WM8978_ADC_CONTROL, 0, 1, 1, 0), + + SOC_DOUBLE_R_TLV("ADC Volume", + WM8978_LEFT_ADC_DIGITAL_VOLUME, WM8978_RIGHT_ADC_DIGITAL_VOLUME, + 0, 255, 0, digital_tlv), + + SOC_ENUM("Equaliser Function", eqmode), + SOC_ENUM("EQ1 Cut Off", eq1), + SOC_SINGLE_TLV("EQ1 Volume", WM8978_EQ1, 0, 24, 1, eq_tlv), + + SOC_ENUM("Equaliser EQ2 Bandwith", eq2bw), + SOC_ENUM("EQ2 Cut Off", eq2), + SOC_SINGLE_TLV("EQ2 Volume", WM8978_EQ2, 0, 24, 1, eq_tlv), + + SOC_ENUM("Equaliser EQ3 Bandwith", eq3bw), + SOC_ENUM("EQ3 Cut Off", eq3), + SOC_SINGLE_TLV("EQ3 Volume", WM8978_EQ3, 0, 24, 1, eq_tlv), + + SOC_ENUM("Equaliser EQ4 Bandwith", eq4bw), + SOC_ENUM("EQ4 Cut Off", eq4), + SOC_SINGLE_TLV("EQ4 Volume", WM8978_EQ4, 0, 24, 1, eq_tlv), + + SOC_ENUM("EQ5 Cut Off", eq5), + SOC_SINGLE_TLV("EQ5 Volume", WM8978_EQ5, 0, 24, 1, eq_tlv), + + SOC_SINGLE("DAC Playback Limiter Switch", + WM8978_DAC_LIMITER_1, 8, 1, 0), + SOC_SINGLE("DAC Playback Limiter Decay", + WM8978_DAC_LIMITER_1, 4, 15, 0), + SOC_SINGLE("DAC Playback Limiter Attack", + WM8978_DAC_LIMITER_1, 0, 15, 0), + + SOC_SINGLE("DAC Playback Limiter Threshold", + WM8978_DAC_LIMITER_2, 4, 7, 0), + SOC_SINGLE("DAC Playback Limiter Boost", + WM8978_DAC_LIMITER_2, 0, 15, 0), + + SOC_ENUM("ALC Enable Switch", alc1), + SOC_SINGLE("ALC Capture Min Gain", WM8978_ALC_CONTROL_1, 0, 7, 0), + SOC_SINGLE("ALC Capture Max Gain", WM8978_ALC_CONTROL_1, 3, 7, 0), + + SOC_SINGLE("ALC Capture Hold", WM8978_ALC_CONTROL_2, 4, 7, 0), + SOC_SINGLE("ALC Capture Target", WM8978_ALC_CONTROL_2, 0, 15, 0), + + SOC_ENUM("ALC Capture Mode", alc3), + SOC_SINGLE("ALC Capture Decay", WM8978_ALC_CONTROL_3, 4, 15, 0), + SOC_SINGLE("ALC Capture Attack", WM8978_ALC_CONTROL_3, 0, 15, 0), + + SOC_SINGLE("ALC Capture Noise Gate Switch", WM8978_NOISE_GATE, 3, 1, 0), + SOC_SINGLE("ALC Capture Noise Gate Threshold", + WM8978_NOISE_GATE, 0, 7, 0), + + SOC_DOUBLE_R("Capture PGA ZC Switch", + WM8978_LEFT_INP_PGA_CONTROL, WM8978_RIGHT_INP_PGA_CONTROL, + 7, 1, 0), + + /* OUT1 - Headphones */ + SOC_DOUBLE_R("Headphone Playback ZC Switch", + WM8978_LOUT1_HP_CONTROL, WM8978_ROUT1_HP_CONTROL, 7, 1, 0), + + SOC_DOUBLE_R_TLV("Headphone Playback Volume", + WM8978_LOUT1_HP_CONTROL, WM8978_ROUT1_HP_CONTROL, + 0, 63, 0, spk_tlv), + + /* OUT2 - Speakers */ + SOC_DOUBLE_R("Speaker Playback ZC Switch", + WM8978_LOUT2_SPK_CONTROL, WM8978_ROUT2_SPK_CONTROL, 7, 1, 0), + + SOC_DOUBLE_R_TLV("Speaker Playback Volume", + WM8978_LOUT2_SPK_CONTROL, WM8978_ROUT2_SPK_CONTROL, + 0, 63, 0, spk_tlv), + + /* OUT3/4 - Line Output */ + SOC_DOUBLE_R("Line Playback Switch", + WM8978_OUT3_MIXER_CONTROL, WM8978_OUT4_MIXER_CONTROL, 6, 1, 1), + + /* Mixer #3: Boost (Input) mixer */ + SOC_DOUBLE_R("PGA Boost (+20dB)", + WM8978_LEFT_ADC_BOOST_CONTROL, WM8978_RIGHT_ADC_BOOST_CONTROL, + 8, 1, 0), + SOC_DOUBLE_R_TLV("L2/R2 Boost Volume", + WM8978_LEFT_ADC_BOOST_CONTROL, WM8978_RIGHT_ADC_BOOST_CONTROL, + 4, 7, 0, boost_tlv), + SOC_DOUBLE_R_TLV("Aux Boost Volume", + WM8978_LEFT_ADC_BOOST_CONTROL, WM8978_RIGHT_ADC_BOOST_CONTROL, + 0, 7, 0, boost_tlv), + + /* Input PGA volume */ + SOC_DOUBLE_R_TLV("Input PGA Volume", + WM8978_LEFT_INP_PGA_CONTROL, WM8978_RIGHT_INP_PGA_CONTROL, + 0, 63, 0, inpga_tlv), + + /* Headphone */ + SOC_DOUBLE_R("Headphone Switch", + WM8978_LOUT1_HP_CONTROL, WM8978_ROUT1_HP_CONTROL, 6, 1, 1), + + /* Speaker */ + SOC_DOUBLE_R("Speaker Switch", + WM8978_LOUT2_SPK_CONTROL, WM8978_ROUT2_SPK_CONTROL, 6, 1, 1), +}; + +/* Mixer #1: Output (OUT1, OUT2) Mixer: mix AUX, Input mixer output and DAC */ +static const struct snd_kcontrol_new wm8978_left_out_mixer[] = { + SOC_DAPM_SINGLE("Line Bypass Switch", WM8978_LEFT_MIXER_CONTROL, 1, 1, 0), + SOC_DAPM_SINGLE("Aux Playback Switch", WM8978_LEFT_MIXER_CONTROL, 5, 1, 0), + SOC_DAPM_SINGLE("PCM Playback Switch", WM8978_LEFT_MIXER_CONTROL, 0, 1, 0), +}; + +static const struct snd_kcontrol_new wm8978_right_out_mixer[] = { + SOC_DAPM_SINGLE("Line Bypass Switch", WM8978_RIGHT_MIXER_CONTROL, 1, 1, 0), + SOC_DAPM_SINGLE("Aux Playback Switch", WM8978_RIGHT_MIXER_CONTROL, 5, 1, 0), + SOC_DAPM_SINGLE("PCM Playback Switch", WM8978_RIGHT_MIXER_CONTROL, 0, 1, 0), +}; + +/* OUT3/OUT4 Mixer not implemented */ + +/* Mixer #2: Input PGA Mute */ +static const struct snd_kcontrol_new wm8978_left_input_mixer[] = { + SOC_DAPM_SINGLE("L2 Switch", WM8978_INPUT_CONTROL, 2, 1, 0), + SOC_DAPM_SINGLE("MicN Switch", WM8978_INPUT_CONTROL, 1, 1, 0), + SOC_DAPM_SINGLE("MicP Switch", WM8978_INPUT_CONTROL, 0, 1, 0), +}; +static const struct snd_kcontrol_new wm8978_right_input_mixer[] = { + SOC_DAPM_SINGLE("R2 Switch", WM8978_INPUT_CONTROL, 6, 1, 0), + SOC_DAPM_SINGLE("MicN Switch", WM8978_INPUT_CONTROL, 5, 1, 0), + SOC_DAPM_SINGLE("MicP Switch", WM8978_INPUT_CONTROL, 4, 1, 0), +}; + +static const struct snd_soc_dapm_widget wm8978_dapm_widgets[] = { + SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", + WM8978_POWER_MANAGEMENT_3, 0, 0), + SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", + WM8978_POWER_MANAGEMENT_3, 1, 0), + SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture", + WM8978_POWER_MANAGEMENT_2, 0, 0), + SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture", + WM8978_POWER_MANAGEMENT_2, 1, 0), + + /* Mixer #1: OUT1,2 */ + SOC_MIXER_ARRAY("Left Output Mixer", WM8978_POWER_MANAGEMENT_3, + 2, 0, wm8978_left_out_mixer), + SOC_MIXER_ARRAY("Right Output Mixer", WM8978_POWER_MANAGEMENT_3, + 3, 0, wm8978_right_out_mixer), + + SOC_MIXER_ARRAY("Left Input Mixer", WM8978_POWER_MANAGEMENT_2, + 2, 0, wm8978_left_input_mixer), + SOC_MIXER_ARRAY("Right Input Mixer", WM8978_POWER_MANAGEMENT_2, + 3, 0, wm8978_right_input_mixer), + + SND_SOC_DAPM_PGA("Left Boost Mixer", WM8978_POWER_MANAGEMENT_2, + 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Boost Mixer", WM8978_POWER_MANAGEMENT_2, + 5, 0, NULL, 0), + + SND_SOC_DAPM_PGA("Left Capture PGA", WM8978_LEFT_INP_PGA_CONTROL, + 6, 1, NULL, 0), + SND_SOC_DAPM_PGA("Right Capture PGA", WM8978_RIGHT_INP_PGA_CONTROL, + 6, 1, NULL, 0), + + SND_SOC_DAPM_PGA("Left Headphone Out", WM8978_POWER_MANAGEMENT_2, + 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Headphone Out", WM8978_POWER_MANAGEMENT_2, + 8, 0, NULL, 0), + + SND_SOC_DAPM_PGA("Left Speaker Out", WM8978_POWER_MANAGEMENT_3, + 6, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Speaker Out", WM8978_POWER_MANAGEMENT_3, + 5, 0, NULL, 0), + + SND_SOC_DAPM_MIXER("OUT4 VMID", WM8978_POWER_MANAGEMENT_3, + 8, 0, NULL, 0), + + SND_SOC_DAPM_MICBIAS("Mic Bias", WM8978_POWER_MANAGEMENT_1, 4, 0), + + SND_SOC_DAPM_INPUT("LMICN"), + SND_SOC_DAPM_INPUT("LMICP"), + SND_SOC_DAPM_INPUT("RMICN"), + SND_SOC_DAPM_INPUT("RMICP"), + SND_SOC_DAPM_INPUT("LAUX"), + SND_SOC_DAPM_INPUT("RAUX"), + SND_SOC_DAPM_INPUT("L2"), + SND_SOC_DAPM_INPUT("R2"), + SND_SOC_DAPM_OUTPUT("LHP"), + SND_SOC_DAPM_OUTPUT("RHP"), + SND_SOC_DAPM_OUTPUT("LSPK"), + SND_SOC_DAPM_OUTPUT("RSPK"), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* Output mixer */ + {"Right Output Mixer", "PCM Playback Switch", "Right DAC"}, + {"Right Output Mixer", "Aux Playback Switch", "RAUX"}, + {"Right Output Mixer", "Line Bypass Switch", "Right Boost Mixer"}, + + {"Left Output Mixer", "PCM Playback Switch", "Left DAC"}, + {"Left Output Mixer", "Aux Playback Switch", "LAUX"}, + {"Left Output Mixer", "Line Bypass Switch", "Left Boost Mixer"}, + + /* Outputs */ + {"Right Headphone Out", NULL, "Right Output Mixer"}, + {"RHP", NULL, "Right Headphone Out"}, + + {"Left Headphone Out", NULL, "Left Output Mixer"}, + {"LHP", NULL, "Left Headphone Out"}, + + {"Right Speaker Out", NULL, "Right Output Mixer"}, + {"RSPK", NULL, "Right Speaker Out"}, + + {"Left Speaker Out", NULL, "Left Output Mixer"}, + {"LSPK", NULL, "Left Speaker Out"}, + + /* Boost Mixer */ + {"Right ADC", NULL, "Right Boost Mixer"}, + + {"Right Boost Mixer", NULL, "RAUX"}, + {"Right Boost Mixer", NULL, "Right Capture PGA"}, + {"Right Boost Mixer", NULL, "R2"}, + + {"Left ADC", NULL, "Left Boost Mixer"}, + + {"Left Boost Mixer", NULL, "LAUX"}, + {"Left Boost Mixer", NULL, "Left Capture PGA"}, + {"Left Boost Mixer", NULL, "L2"}, + + /* Input PGA */ + {"Right Capture PGA", NULL, "Right Input Mixer"}, + {"Left Capture PGA", NULL, "Left Input Mixer"}, + + {"Right Input Mixer", "R2 Switch", "R2"}, + {"Right Input Mixer", "MicN Switch", "RMICN"}, + {"Right Input Mixer", "MicP Switch", "RMICP"}, + + {"Left Input Mixer", "L2 Switch", "L2"}, + {"Left Input Mixer", "MicN Switch", "LMICN"}, + {"Left Input Mixer", "MicP Switch", "LMICP"}, +}; + +static int wm8978_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, wm8978_dapm_widgets, + ARRAY_SIZE(wm8978_dapm_widgets)); + + /* set up the WM8978 audio map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + return 0; +} + +/* PLL divisors */ +struct wm8978_pll_div { + u32 k; + u8 n; + u8 div2; +}; + +#define FIXED_PLL_SIZE (1 << 24) + +static void pll_factors(struct wm8978_pll_div *pll_div, unsigned int target, + unsigned int source) +{ + u64 k_part; + unsigned int k, n_div, n_mod; + + n_div = target / source; + if (n_div < 6) { + source >>= 1; + pll_div->div2 = 1; + n_div = target / source; + } else { + pll_div->div2 = 0; + } + + if (n_div < 6 || n_div > 12) + dev_warn(wm8978_codec->dev, + "WM8978 N value exceeds recommended range! N = %u\n", + n_div); + + pll_div->n = n_div; + n_mod = target - source * n_div; + k_part = FIXED_PLL_SIZE * (long long)n_mod + source / 2; + + do_div(k_part, source); + + k = k_part & 0xFFFFFFFF; + + pll_div->k = k; +} +/* + * Calculate internal frequencies and dividers, according to Figure 40 + * "PLL and Clock Select Circuit" in WM8978 datasheet Rev. 2.6 + */ +static int wm8978_configure_pll(struct snd_soc_codec *codec) +{ + struct wm8978_priv *wm8978 = codec->private_data; + struct wm8978_pll_div pll_div; + unsigned int f_opclk = wm8978->f_opclk, f_mclk = wm8978->f_mclk, + f_256fs = wm8978->f_256fs; + unsigned int f2, opclk_div; + + if (!f_mclk) + return -EINVAL; + + if (f_opclk) { + /* + * The user needs OPCLK. Choose OPCLKDIV to put + * 6 <= R = f2 / f1 < 13, 1 <= OPCLKDIV <= 4. + * f_opclk = f_mclk * prescale * R / 4 / OPCLKDIV, where + * prescale = 1, or prescale = 2. Prescale is calculated inside + * pll_factors(). We have to select f_PLLOUT, such that + * f_mclk * 3 / 4 <= f_PLLOUT < f_mclk * 13 / 4. Must be + * f_mclk * 3 / 16 <= f_opclk < f_mclk * 13 / 4. + */ + if (16 * f_opclk < 3 * f_mclk || 4 * f_opclk >= 13 * f_mclk) + return -EINVAL; + + if (4 * f_opclk < 3 * f_mclk) + /* Have to use OPCLKDIV */ + opclk_div = (3 * f_mclk / 4 + f_opclk - 1) / f_opclk; + else + opclk_div = 1; + + dev_dbg(codec->dev, "%s: OPCLKDIV=%d\n", __func__, opclk_div); + + snd_soc_update_bits(codec, WM8978_GPIO_CONTROL, 0x30, + (opclk_div - 1) << 4); + + wm8978->f_pllout = f_opclk * opclk_div; + } else if (f_256fs) { + /* + * Not using OPCLK, choose R: + * 6 <= R = f2 / f1 < 13, to put 1 <= MCLKDIV <= 12. + * f_256fs = f_mclk * prescale * R / 4 / MCLKDIV, where + * prescale = 1, or prescale = 2. Prescale is calculated inside + * pll_factors(). We have to select f_PLLOUT, such that + * f_mclk * 3 / 4 <= f_PLLOUT < f_mclk * 13 / 4. Must be + * f_mclk * 3 / 48 <= f_256fs < f_mclk * 13 / 4. This means MCLK + * must be 3.781MHz <= f_MCLK <= 32.768MHz + */ + if (48 * f_256fs < 3 * f_mclk || 4 * f_256fs >= 13 * f_mclk) + return -EINVAL; + + /* + * MCLKDIV will be selected in .hw_params(), just choose a + * suitable f_PLLOUT + */ + if (4 * f_256fs < 3 * f_mclk) + /* Will have to use MCLKDIV */ + wm8978->f_pllout = wm8978->f_mclk * 3 / 4; + else + wm8978->f_pllout = f_256fs; + + /* GPIO1 into default mode as input - before configuring PLL */ + snd_soc_update_bits(codec, WM8978_GPIO_CONTROL, 7, 0); + } else { + return -EINVAL; + } + + f2 = wm8978->f_pllout * 4; + + dev_dbg(codec->dev, "%s: f_MCLK=%uHz, f_PLLOUT=%uHz\n", __func__, + wm8978->f_mclk, wm8978->f_pllout); + + pll_factors(&pll_div, f2, wm8978->f_mclk); + + dev_dbg(codec->dev, "%s: calculated PLL N=0x%x, K=0x%x, div2=%d\n", + __func__, pll_div.n, pll_div.k, pll_div.div2); + + /* Turn PLL off for configuration... */ + snd_soc_update_bits(codec, WM8978_POWER_MANAGEMENT_1, 0x20, 0); + + snd_soc_write(codec, WM8978_PLL_N, (pll_div.div2 << 4) | pll_div.n); + snd_soc_write(codec, WM8978_PLL_K1, pll_div.k >> 18); + snd_soc_write(codec, WM8978_PLL_K2, (pll_div.k >> 9) & 0x1ff); + snd_soc_write(codec, WM8978_PLL_K3, pll_div.k & 0x1ff); + + /* ...and on again */ + snd_soc_update_bits(codec, WM8978_POWER_MANAGEMENT_1, 0x20, 0x20); + + if (f_opclk) + /* Output PLL (OPCLK) to GPIO1 */ + snd_soc_update_bits(codec, WM8978_GPIO_CONTROL, 7, 4); + + return 0; +} + +/* + * Configure WM8978 clock dividers. + */ +static int wm8978_set_dai_clkdiv(struct snd_soc_dai *codec_dai, + int div_id, int div) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8978_priv *wm8978 = codec->private_data; + int ret = 0; + + switch (div_id) { + case WM8978_OPCLKRATE: + wm8978->f_opclk = div; + + if (wm8978->f_mclk) + ret = wm8978_configure_pll(codec); + break; + case WM8978_MCLKDIV: + if (div & ~0xe0) + return -EINVAL; + snd_soc_update_bits(codec, WM8978_CLOCKING, 0xe0, div); + break; + case WM8978_ADCCLK: + if (div & ~8) + return -EINVAL; + snd_soc_update_bits(codec, WM8978_ADC_CONTROL, 8, div); + break; + case WM8978_DACCLK: + if (div & ~8) + return -EINVAL; + snd_soc_update_bits(codec, WM8978_DAC_CONTROL, 8, div); + break; + case WM8978_BCLKDIV: + if (div & ~0x1c) + return -EINVAL; + snd_soc_update_bits(codec, WM8978_CLOCKING, 0x1c, div); + break; + default: + return -EINVAL; + } + + dev_dbg(codec->dev, "%s: ID %d, value %u\n", __func__, div_id, div); + + return ret; +} + +/* + * @freq: when .set_pll() us not used, freq is codec MCLK input frequency + */ +static int wm8978_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8978_priv *wm8978 = codec->private_data; + int ret = 0; + + dev_dbg(codec->dev, "%s: ID %d, freq %u\n", __func__, clk_id, freq); + + if (freq) { + wm8978->f_mclk = freq; + + /* Even if MCLK is used for system clock, might have to drive OPCLK */ + if (wm8978->f_opclk) + ret = wm8978_configure_pll(codec); + + /* Our sysclk is fixed to 256 * fs, will configure in .hw_params() */ + + if (!ret) + wm8978->sysclk = clk_id; + } + + if (wm8978->sysclk == WM8978_PLL && (!freq || clk_id == WM8978_MCLK)) { + /* Clock CODEC directly from MCLK */ + snd_soc_update_bits(codec, WM8978_CLOCKING, 0x100, 0); + + /* GPIO1 into default mode as input - before configuring PLL */ + snd_soc_update_bits(codec, WM8978_GPIO_CONTROL, 7, 0); + + /* Turn off PLL */ + snd_soc_update_bits(codec, WM8978_POWER_MANAGEMENT_1, 0x20, 0); + wm8978->sysclk = WM8978_MCLK; + wm8978->f_pllout = 0; + wm8978->f_opclk = 0; + } + + return ret; +} + +/* + * Set ADC and Voice DAC format. + */ +static int wm8978_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + /* + * BCLK polarity mask = 0x100, LRC clock polarity mask = 0x80, + * Data Format mask = 0x18: all will be calculated anew + */ + u16 iface = snd_soc_read(codec, WM8978_AUDIO_INTERFACE) & ~0x198; + u16 clk = snd_soc_read(codec, WM8978_CLOCKING); + + dev_dbg(codec->dev, "%s\n", __func__); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + clk |= 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + clk &= ~1; + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= 0x10; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= 0x8; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= 0x18; + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= 0x180; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= 0x100; + break; + case SND_SOC_DAIFMT_NB_IF: + iface |= 0x80; + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, WM8978_AUDIO_INTERFACE, iface); + snd_soc_write(codec, WM8978_CLOCKING, clk); + + return 0; +} + +/* MCLK dividers */ +static const int mclk_numerator[] = {1, 3, 2, 3, 4, 6, 8, 12}; +static const int mclk_denominator[] = {1, 2, 1, 1, 1, 1, 1, 1}; + +/* + * Set PCM DAI bit size and sample rate. + */ +static int wm8978_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct wm8978_priv *wm8978 = codec->private_data; + /* Word length mask = 0x60 */ + u16 iface_ctl = snd_soc_read(codec, WM8978_AUDIO_INTERFACE) & ~0x60; + /* Sampling rate mask = 0xe (for filters) */ + u16 add_ctl = snd_soc_read(codec, WM8978_ADDITIONAL_CONTROL) & ~0xe; + u16 clking = snd_soc_read(codec, WM8978_CLOCKING); + enum wm8978_sysclk_src current_clk_id = clking & 0x100 ? + WM8978_PLL : WM8978_MCLK; + unsigned int f_sel, diff, diff_best = INT_MAX; + int i, best = 0; + + if (!wm8978->f_mclk) + return -EINVAL; + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface_ctl |= 0x20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface_ctl |= 0x40; + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface_ctl |= 0x60; + break; + } + + /* filter coefficient */ + switch (params_rate(params)) { + case 8000: + add_ctl |= 0x5 << 1; + break; + case 11025: + add_ctl |= 0x4 << 1; + break; + case 16000: + add_ctl |= 0x3 << 1; + break; + case 22050: + add_ctl |= 0x2 << 1; + break; + case 32000: + add_ctl |= 0x1 << 1; + break; + case 44100: + case 48000: + break; + } + + /* Sampling rate is known now, can configure the MCLK divider */ + wm8978->f_256fs = params_rate(params) * 256; + + if (wm8978->sysclk == WM8978_MCLK) { + f_sel = wm8978->f_mclk; + } else { + if (!wm8978->f_pllout) { + int ret = wm8978_configure_pll(codec); + if (ret < 0) + return ret; + } + f_sel = wm8978->f_pllout; + } + + /* + * In some cases it is possible to reconfigure PLL to a higher frequency + * by raising OPCLKDIV, but normally OPCLK is configured to 256 * fs or + * 512 * fs, so, we should be fine. + */ + if (f_sel < wm8978->f_256fs || f_sel > 12 * wm8978->f_256fs) + return -EINVAL; + + for (i = 0; i < ARRAY_SIZE(mclk_numerator); i++) { + diff = abs(wm8978->f_256fs * 3 - + f_sel * 3 * mclk_denominator[i] / mclk_numerator[i]); + + if (diff < diff_best) { + diff_best = diff; + best = i; + } + + if (!diff) + break; + } + + if (diff) + dev_warn(codec->dev, "Imprecise clock: %u%s\n", + f_sel * mclk_denominator[best] / mclk_numerator[best], + wm8978->sysclk == WM8978_MCLK ? + ", consider using PLL" : ""); + + dev_dbg(codec->dev, "%s: fmt %d, rate %u, MCLK divisor #%d\n", __func__, + params_format(params), params_rate(params), best); + + /* MCLK divisor mask = 0xe0 */ + snd_soc_update_bits(codec, WM8978_CLOCKING, 0xe0, best << 5); + + snd_soc_write(codec, WM8978_AUDIO_INTERFACE, iface_ctl); + snd_soc_write(codec, WM8978_ADDITIONAL_CONTROL, add_ctl); + + if (wm8978->sysclk != current_clk_id) { + if (wm8978->sysclk == WM8978_PLL) + /* Run CODEC from PLL instead of MCLK */ + snd_soc_update_bits(codec, WM8978_CLOCKING, + 0x100, 0x100); + else + /* Clock CODEC directly from MCLK */ + snd_soc_update_bits(codec, WM8978_CLOCKING, 0x100, 0); + } + + return 0; +} + +static int wm8978_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + + dev_dbg(codec->dev, "%s: %d\n", __func__, mute); + + if (mute) + snd_soc_update_bits(codec, WM8978_DAC_CONTROL, 0x40, 0x40); + else + snd_soc_update_bits(codec, WM8978_DAC_CONTROL, 0x40, 0); + + return 0; +} + +static int wm8978_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 power1 = snd_soc_read(codec, WM8978_POWER_MANAGEMENT_1) & ~3; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + power1 |= 1; /* VMID 75k */ + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_1, power1); + break; + case SND_SOC_BIAS_STANDBY: + /* bit 3: enable bias, bit 2: enable I/O tie off buffer */ + power1 |= 0xc; + + if (codec->bias_level == SND_SOC_BIAS_OFF) { + /* Initial cap charge at VMID 5k */ + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_1, + power1 | 0x3); + mdelay(100); + } + + power1 |= 0x2; /* VMID 500k */ + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_1, power1); + break; + case SND_SOC_BIAS_OFF: + /* Preserve PLL - OPCLK may be used by someone */ + snd_soc_update_bits(codec, WM8978_POWER_MANAGEMENT_1, ~0x20, 0); + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_2, 0); + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_3, 0); + break; + } + + dev_dbg(codec->dev, "%s: %d, %x\n", __func__, level, power1); + + codec->bias_level = level; + return 0; +} + +#define WM8978_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops wm8978_dai_ops = { + .hw_params = wm8978_hw_params, + .digital_mute = wm8978_mute, + .set_fmt = wm8978_set_dai_fmt, + .set_clkdiv = wm8978_set_dai_clkdiv, + .set_sysclk = wm8978_set_dai_sysclk, +}; + +/* Also supports 12kHz */ +struct snd_soc_dai wm8978_dai = { + .name = "WM8978 HiFi", + .id = 1, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = WM8978_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = WM8978_FORMATS, + }, + .ops = &wm8978_dai_ops, +}; +EXPORT_SYMBOL_GPL(wm8978_dai); + +static int wm8978_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + wm8978_set_bias_level(codec, SND_SOC_BIAS_OFF); + /* Also switch PLL off */ + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_1, 0); + /* Put to sleep */ + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_2, 0x40); + + return 0; +} + +static int wm8978_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + struct wm8978_priv *wm8978 = codec->private_data; + int i; + u16 *cache = codec->reg_cache; + + /* Wake up the codec */ + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_2, 0); + + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(wm8978_reg); i++) { + if (i == WM8978_RESET) + continue; + if (cache[i] != wm8978_reg[i]) + snd_soc_write(codec, i, cache[i]); + } + + wm8978_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + if (wm8978->f_pllout) + /* Switch PLL on */ + snd_soc_update_bits(codec, WM8978_POWER_MANAGEMENT_1, 0x20, 0x20); + + return 0; +} + +static int wm8978_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (wm8978_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = wm8978_codec; + codec = wm8978_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms: %d\n", ret); + goto pcm_err; + } + + snd_soc_add_controls(codec, wm8978_snd_controls, + ARRAY_SIZE(wm8978_snd_controls)); + wm8978_add_widgets(codec); + +pcm_err: + return ret; +} + +/* power down chip */ +static int wm8978_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8978 = { + .probe = wm8978_probe, + .remove = wm8978_remove, + .suspend = wm8978_suspend, + .resume = wm8978_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8978); + +/* + * These registers contain an "update" bit - bit 8. This means, for example, + * that one can write new DAC digital volume for both channels, but only when + * the update bit is set, will also the volume be updated - simultaneously for + * both channels. + */ +static const int update_reg[] = { + WM8978_LEFT_DAC_DIGITAL_VOLUME, + WM8978_RIGHT_DAC_DIGITAL_VOLUME, + WM8978_LEFT_ADC_DIGITAL_VOLUME, + WM8978_RIGHT_ADC_DIGITAL_VOLUME, + WM8978_LEFT_INP_PGA_CONTROL, + WM8978_RIGHT_INP_PGA_CONTROL, + WM8978_LOUT1_HP_CONTROL, + WM8978_ROUT1_HP_CONTROL, + WM8978_LOUT2_SPK_CONTROL, + WM8978_ROUT2_SPK_CONTROL, +}; + +static __devinit int wm8978_register(struct wm8978_priv *wm8978) +{ + int ret, i; + struct snd_soc_codec *codec = &wm8978->codec; + + if (wm8978_codec) { + dev_err(codec->dev, "Another WM8978 is registered\n"); + return -EINVAL; + } + + /* + * Set default system clock to PLL, it is more precise, this is also the + * default hardware setting + */ + wm8978->sysclk = WM8978_PLL; + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = wm8978; + codec->name = "WM8978"; + codec->owner = THIS_MODULE; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = wm8978_set_bias_level; + codec->dai = &wm8978_dai; + codec->num_dai = 1; + codec->reg_cache_size = WM8978_CACHEREGNUM; + codec->reg_cache = &wm8978->reg_cache; + + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_I2C); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + goto err; + } + + memcpy(codec->reg_cache, wm8978_reg, sizeof(wm8978_reg)); + + /* + * Set the update bit in all registers, that have one. This way all + * writes to those registers will also cause the update bit to be + * written. + */ + for (i = 0; i < ARRAY_SIZE(update_reg); i++) + ((u16 *)codec->reg_cache)[update_reg[i]] |= 0x100; + + /* Reset the codec */ + ret = snd_soc_write(codec, WM8978_RESET, 0); + if (ret < 0) { + dev_err(codec->dev, "Failed to issue reset\n"); + goto err; + } + + wm8978_dai.dev = codec->dev; + + wm8978_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + wm8978_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto err; + } + + ret = snd_soc_register_dai(&wm8978_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + goto err_codec; + } + + return 0; + +err_codec: + snd_soc_unregister_codec(codec); +err: + kfree(wm8978); + return ret; +} + +static __devexit void wm8978_unregister(struct wm8978_priv *wm8978) +{ + wm8978_set_bias_level(&wm8978->codec, SND_SOC_BIAS_OFF); + snd_soc_unregister_dai(&wm8978_dai); + snd_soc_unregister_codec(&wm8978->codec); + kfree(wm8978); + wm8978_codec = NULL; +} + +static __devinit int wm8978_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct wm8978_priv *wm8978; + struct snd_soc_codec *codec; + + wm8978 = kzalloc(sizeof(struct wm8978_priv), GFP_KERNEL); + if (wm8978 == NULL) + return -ENOMEM; + + codec = &wm8978->codec; + codec->hw_write = (hw_write_t)i2c_master_send; + + i2c_set_clientdata(i2c, wm8978); + codec->control_data = i2c; + + codec->dev = &i2c->dev; + + return wm8978_register(wm8978); +} + +static __devexit int wm8978_i2c_remove(struct i2c_client *client) +{ + struct wm8978_priv *wm8978 = i2c_get_clientdata(client); + wm8978_unregister(wm8978); + return 0; +} + +static const struct i2c_device_id wm8978_i2c_id[] = { + { "wm8978", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8978_i2c_id); + +static struct i2c_driver wm8978_i2c_driver = { + .driver = { + .name = "WM8978", + .owner = THIS_MODULE, + }, + .probe = wm8978_i2c_probe, + .remove = __devexit_p(wm8978_i2c_remove), + .id_table = wm8978_i2c_id, +}; + +static int __init wm8978_modinit(void) +{ + return i2c_add_driver(&wm8978_i2c_driver); +} +module_init(wm8978_modinit); + +static void __exit wm8978_exit(void) +{ + i2c_del_driver(&wm8978_i2c_driver); +} +module_exit(wm8978_exit); + +MODULE_DESCRIPTION("ASoC WM8978 codec driver"); +MODULE_AUTHOR("Guennadi Liakhovetski <g.liakhovetski(a)gmx.de>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8978.h b/sound/soc/codecs/wm8978.h new file mode 100644 index 0000000..b58f0bf --- /dev/null +++ b/sound/soc/codecs/wm8978.h @@ -0,0 +1,89 @@ +/* + * wm8978.h -- codec driver for WM8978 + * + * Copyright 2009 Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> + * + * 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 __WM8978_H__ +#define __WM8978_H__ + +/* + * Register values. + */ +#define WM8978_RESET 0x00 +#define WM8978_POWER_MANAGEMENT_1 0x01 +#define WM8978_POWER_MANAGEMENT_2 0x02 +#define WM8978_POWER_MANAGEMENT_3 0x03 +#define WM8978_AUDIO_INTERFACE 0x04 +#define WM8978_COMPANDING_CONTROL 0x05 +#define WM8978_CLOCKING 0x06 +#define WM8978_ADDITIONAL_CONTROL 0x07 +#define WM8978_GPIO_CONTROL 0x08 +#define WM8978_JACK_DETECT_CONTROL_1 0x09 +#define WM8978_DAC_CONTROL 0x0A +#define WM8978_LEFT_DAC_DIGITAL_VOLUME 0x0B +#define WM8978_RIGHT_DAC_DIGITAL_VOLUME 0x0C +#define WM8978_JACK_DETECT_CONTROL_2 0x0D +#define WM8978_ADC_CONTROL 0x0E +#define WM8978_LEFT_ADC_DIGITAL_VOLUME 0x0F +#define WM8978_RIGHT_ADC_DIGITAL_VOLUME 0x10 +#define WM8978_EQ1 0x12 +#define WM8978_EQ2 0x13 +#define WM8978_EQ3 0x14 +#define WM8978_EQ4 0x15 +#define WM8978_EQ5 0x16 +#define WM8978_DAC_LIMITER_1 0x18 +#define WM8978_DAC_LIMITER_2 0x19 +#define WM8978_NOTCH_FILTER_1 0x1b +#define WM8978_NOTCH_FILTER_2 0x1c +#define WM8978_NOTCH_FILTER_3 0x1d +#define WM8978_NOTCH_FILTER_4 0x1e +#define WM8978_ALC_CONTROL_1 0x20 +#define WM8978_ALC_CONTROL_2 0x21 +#define WM8978_ALC_CONTROL_3 0x22 +#define WM8978_NOISE_GATE 0x23 +#define WM8978_PLL_N 0x24 +#define WM8978_PLL_K1 0x25 +#define WM8978_PLL_K2 0x26 +#define WM8978_PLL_K3 0x27 +#define WM8978_3D_CONTROL 0x29 +#define WM8978_BEEP_CONTROL 0x2b +#define WM8978_INPUT_CONTROL 0x2c +#define WM8978_LEFT_INP_PGA_CONTROL 0x2d +#define WM8978_RIGHT_INP_PGA_CONTROL 0x2e +#define WM8978_LEFT_ADC_BOOST_CONTROL 0x2f +#define WM8978_RIGHT_ADC_BOOST_CONTROL 0x30 +#define WM8978_OUTPUT_CONTROL 0x31 +#define WM8978_LEFT_MIXER_CONTROL 0x32 +#define WM8978_RIGHT_MIXER_CONTROL 0x33 +#define WM8978_LOUT1_HP_CONTROL 0x34 +#define WM8978_ROUT1_HP_CONTROL 0x35 +#define WM8978_LOUT2_SPK_CONTROL 0x36 +#define WM8978_ROUT2_SPK_CONTROL 0x37 +#define WM8978_OUT3_MIXER_CONTROL 0x38 +#define WM8978_OUT4_MIXER_CONTROL 0x39 + +#define WM8978_CACHEREGNUM 58 + +/* Clock divider Id's */ +enum wm8978_clk_id { + WM8978_OPCLKRATE, + WM8978_MCLKDIV, + WM8978_ADCCLK, + WM8978_DACCLK, + WM8978_BCLKDIV, +}; + +enum wm8978_sysclk_src { + WM8978_PLL, + WM8978_MCLK +}; + +extern struct snd_soc_dai wm8978_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm8978; + +#endif /* __WM8978_H__ */

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[alsa-devel] About HDMI audio driver
by Wei Ni 03 Feb '10

03 Feb '10

Hi, Takashi I'm a nvidia engineer. We plan to develop the new hdmi audio driver for the new chipset. It support the latest HD Audio Specification. I download the latest kernel source, it has patch_intelhdmi.c. And this file have many defines and routines for HDMI operations. If I develop hdmi driver, it also need to define the same functions in patch_nvhdmi.c, it seems not very well. Do you have plan to change these codes to common code? I think these codes are common for HDMI audio. BTW I have a another question. Because the standard ALSA channel mapping is different with the Windows setting. So we could use channel-slot assignment (verb 0x734) to fix. How can I know the audio is using ALSA setting or Windows setting? Does there have any mechanism to support this operation? Thanks Wei. ----------------------------------------------------------------------------------- This email message is for the sole use of the intended recipient(s) and may contain confidential information. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -----------------------------------------------------------------------------------

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[alsa-devel] [PATCH] ALSA: ASoC: fix PXA SSP port resume
by Daniel Mack 02 Feb '10

02 Feb '10

Invalidate the cached hardware format on resume for PXA SSP ports. Otherwise hw_params() will bail out early at the next stream start, leaving the registers in a bogus state. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Eric Miao <eric.y.miao(a)gmail.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Philipp Zabel <philipp.zabel(a)gmail.com> --- sound/soc/pxa/pxa-ssp.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 3bd7712..ebde673 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -146,6 +146,9 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai) { struct ssp_priv *priv = cpu_dai->private_data; + /* the cached format information is invalid now */ + priv->dai_fmt = (unsigned int) -1; + if (!cpu_dai->active) return 0; -- 1.6.3.3

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[alsa-devel] [PATCH 00/10] treewide: Fix format strings that misuse continuations
by Joe Perches 02 Feb '10

02 Feb '10

Format strings that are continued with \ are frequently misused. Change them to use mostly single line formats, some longer than 80 chars. Fix a few miscellaneous typos at the same time. Joe Perches (10): arch/powerpc: Fix continuation line formats arch/blackfin: Fix continuation line formats drivers/ide: Fix continuation line formats drivers/serial/bfin_5xx.c: Fix continuation line formats drivers/scsi/arcmsr: Fix continuation line formats drivers/staging: Fix continuation line formats drivers/net/amd8111e.c: Fix continuation line formats fs/proc/array.c: Fix continuation line formats mm/slab.c: Fix continuation line formats sound/soc/blackfin: Fix continuation line formats arch/blackfin/mach-common/smp.c | 4 +- arch/powerpc/kernel/nvram_64.c | 6 +- arch/powerpc/platforms/pseries/hotplug-cpu.c | 8 ++-- arch/powerpc/platforms/pseries/smp.c | 4 +- drivers/ide/au1xxx-ide.c | 4 +- drivers/ide/pmac.c | 4 +- drivers/net/amd8111e.c | 3 +- drivers/scsi/arcmsr/arcmsr_hba.c | 49 +++++++++---------- drivers/serial/bfin_5xx.c | 6 +-- drivers/staging/dream/qdsp5/audio_mp3.c | 3 +- .../rtl8187se/ieee80211/ieee80211_softmac_wx.c | 3 +- drivers/staging/rtl8187se/r8180_core.c | 3 +- drivers/staging/sep/sep_driver.c | 3 +- fs/proc/array.c | 7 ++- mm/slab.c | 4 +- sound/soc/blackfin/bf5xx-ac97-pcm.c | 8 +-- sound/soc/blackfin/bf5xx-i2s-pcm.c | 3 +- sound/soc/blackfin/bf5xx-tdm-pcm.c | 3 +- 18 files changed, 55 insertions(+), 70 deletions(-)

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