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[alsa-devel] [sound:for-linus 4/4] sound/oss/pas2_card.c:338:5: warning: too many arguments for format
by kbuild test robot 04 Jan '13

04 Jan '13

tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-linus head: b56458b36c8922d1d57c2320151194059dfb9a31 commit: b56458b36c8922d1d57c2320151194059dfb9a31 [4/4] sound: oss/pas2: Fix possible access out of array config: make ARCH=x86_64 allyesconfig All warnings: sound/oss/pas2_card.c: In function 'attach_pas_card': sound/oss/pas2_card.c:338:5: warning: too many arguments for format [-Wformat-extra-args] vim +338 sound/oss/pas2_card.c ^1da177e Linus Torvalds 2005-04-16 322 return pas_model; ^1da177e Linus Torvalds 2005-04-16 323 } ^1da177e Linus Torvalds 2005-04-16 324 ^1da177e Linus Torvalds 2005-04-16 325 static void __init attach_pas_card(struct address_info *hw_config) ^1da177e Linus Torvalds 2005-04-16 326 { ^1da177e Linus Torvalds 2005-04-16 327 pas_irq = hw_config->irq; ^1da177e Linus Torvalds 2005-04-16 328 ^1da177e Linus Torvalds 2005-04-16 329 if (detect_pas_hw(hw_config)) ^1da177e Linus Torvalds 2005-04-16 330 { ^1da177e Linus Torvalds 2005-04-16 331 ^1da177e Linus Torvalds 2005-04-16 332 if ((pas_model = pas_read(0xFF88))) ^1da177e Linus Torvalds 2005-04-16 333 { ^1da177e Linus Torvalds 2005-04-16 334 char temp[100]; ^1da177e Linus Torvalds 2005-04-16 335 b56458b3 Asim Kadav 2013-01-03 336 if (pas_model < 0 || b56458b3 Asim Kadav 2013-01-03 337 pas_model >= ARRAY_SIZE(pas_model_names)) { b56458b3 Asim Kadav 2013-01-03 @338 printk(KERN_ERR, "pas2 unrecognized model.\n"); b56458b3 Asim Kadav 2013-01-03 339 return; b56458b3 Asim Kadav 2013-01-03 340 } ^1da177e Linus Torvalds 2005-04-16 341 sprintf(temp, ^1da177e Linus Torvalds 2005-04-16 342 "%s rev %d", pas_model_names[(int) pas_model], ^1da177e Linus Torvalds 2005-04-16 343 pas_read(0x2789)); ^1da177e Linus Torvalds 2005-04-16 344 conf_printf(temp, hw_config); ^1da177e Linus Torvalds 2005-04-16 345 } ^1da177e Linus Torvalds 2005-04-16 346 if (config_pas_hw(hw_config)) --- 0-DAY kernel build testing backend Open Source Technology Center Fengguang Wu, Yuanhan Liu Intel Corporation

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[alsa-devel] [Tiny6410/Mini6410] Sound support
by Alexander Nestorov 03 Jan '13

03 Jan '13

I'm working on a Tiny6410 device (S3C6410) on top of a 3.7 kernel and sound isn't working with the default settings. I also wasn't able to make it work enablig the related stuff to AC97 in menuconfig so I'm wondering if there is actually any support for this device? Looking at the code in sound/soc, there is something for the SMDK6410 but I'm not really sure if that's compatible with my device or even if that code is fully functional. Can I get some help with this, please? I'll appreciate any kind of help. Regards PS: I'm including Kukjin Kim and Sangbeom Kim as of recomendation of <broonie> from #alsa-soc at Freenode -- alexandernst

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[alsa-devel] [PATCH v2] ASoC: pxa2xx-ac97: fix oops when codec_driver probe runs
by Mike Dunn 03 Jan '13

03 Jan '13

Change log: v2: pxa2xx_ac97_hw_probe() is called *before* snd_soc_register_dais(), and likewise pxa2xx_ac97_hw_remove() is called after snd_soc_unregister_dais(). Thanks much Mark. This patch fixes a kernel oops that occurs when the snd_soc_codec_driver probe function runs. The exception occurs when the codec calls soc_ac97_ops.reset(). The pxa2xx-ac97's clocks have not been initialized at this point, but the cpu's reset function (pxa_ac97_cold_pxa27x()) expects them to be, resulting in a NULL pointer dereference in the pxa's clk_enable(). The pxa2xx-ac97's clocks are initialized in pxa2xx_ac97_hw_probe(). To ensure that the clocks are initialized when the codec_driver probe runs, the calls to pxa2xx_ac97_hw_{probe,remove}() are moved from the cpu's dai_driver probe/remove functions to those of its platform_driver. This is necessary because snd_soc_instantiate_card() always calls the codec_driver probe before any of the dai_drivers' probes. This change leaves nothing in the cpu's dai_driver probe/remove functions, so those functions are eliminated. Tested on the pxa27x; palmtreo680 machine using the wm9712 codec. Signed-off-by: Mike Dunn <mikedunn(a)newsguy.com> --- sound/soc/pxa/pxa2xx-ac97.c | 27 ++++++++++----------------- 1 files changed, 10 insertions(+), 17 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 4b0a009..ba04f49 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -104,17 +104,6 @@ static int pxa2xx_ac97_resume(struct snd_soc_dai *dai) #define pxa2xx_ac97_resume NULL #endif -static int pxa2xx_ac97_probe(struct snd_soc_dai *dai) -{ - return pxa2xx_ac97_hw_probe(to_platform_device(dai->dev)); -} - -static int pxa2xx_ac97_remove(struct snd_soc_dai *dai) -{ - pxa2xx_ac97_hw_remove(to_platform_device(dai->dev)); - return 0; -} - static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *cpu_dai) @@ -184,8 +173,6 @@ static struct snd_soc_dai_driver pxa_ac97_dai_driver[] = { { .name = "pxa2xx-ac97", .ac97_control = 1, - .probe = pxa2xx_ac97_probe, - .remove = pxa2xx_ac97_remove, .suspend = pxa2xx_ac97_suspend, .resume = pxa2xx_ac97_resume, .playback = { @@ -236,15 +223,20 @@ EXPORT_SYMBOL_GPL(soc_ac97_ops); static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) { + int ret; + if (pdev->id != -1) { dev_err(&pdev->dev, "PXA2xx has only one AC97 port.\n"); return -ENXIO; } - /* Punt most of the init to the SoC probe; we may need the machine - * driver to do interesting things with the clocking to get us up - * and running. - */ + ret = pxa2xx_ac97_hw_probe(pdev); + if (ret < 0) { + dev_err(&pdev->dev, "pxa2xx_ac97_hw_probe failed with %d\n", + ret); + return ret; + } + return snd_soc_register_dais(&pdev->dev, pxa_ac97_dai_driver, ARRAY_SIZE(pxa_ac97_dai_driver)); } @@ -252,6 +244,7 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) static int pxa2xx_ac97_dev_remove(struct platform_device *pdev) { snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(pxa_ac97_dai_driver)); + pxa2xx_ac97_hw_remove(pdev); return 0; } -- 1.7.8.6

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[alsa-devel] [PATCH] ALSA: hda - Switch "On" and "Off" for "Mute-LED Mode" kcontrol
by David Henningsson 03 Jan '13

03 Jan '13

The vmaster hook sends 1 for enabled/unmuted and 0 for disabled/muted, but "Mute-LED Mode" being "On" refers to the LED being on, not the volume being on. Therefore "On" and "Off" should be switched. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 8353c77..b8fb0a5 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2531,7 +2531,7 @@ static int vmaster_mute_mode_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static const char * const texts[] = { - "Off", "On", "Follow Master" + "On", "Off", "Follow Master" }; unsigned int index; -- 1.7.9.5

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Re: [alsa-devel] [PATCH] sound: Add support for Creative BT-D1 via usb sound quirks
by Takashi Iwai 03 Jan '13

03 Jan '13

At Sat, 29 Dec 2012 14:26:25 +0100, Alexander Schremmer wrote: > > From: Alexander Schremmer <alex(a)alexanderweb.de> > > Support the Creative BT-D1 Bluetooth USB audio device. Before this > patch, Linux had trouble finding the correct USB descriptors and bailed > out with these messages: > > no or invalid class specific endpoint descriptor > > Now it still prints these messages on hotplug: > > snd-usb-audio: probe of ...:1.0 failed with error -5 > snd-usb-audio: probe of ...:1.2 failed with error -5 > snd-usb-audio: probe of ...:1.3 failed with error -5 > > But the device works correctly, including the HID support. > > The patch is diff'ed against 3.8-rc1 but should apply to older kernels > as well. > > Signed-off-by: Alexander Schremmer <alex(a)alexanderweb.de> Thanks for the patch. The changes look OK, but could you fix trivial coding issues reported by scripts/checkpatch.pl and resend? Takashi > > --- > diff -uprN linux-3.8-rc1/sound/usb/quirks-table.h > linux-3.8-rc1-mod/sound/usb/quirks-table.h > --- linux-3.8-rc1/sound/usb/quirks-table.h 2012-12-22 > 02:19:00.000000000 +0100 > +++ linux-3.8-rc1-mod/sound/usb/quirks-table.h 2012-12-26 > 11:41:28.000000000 +0100 > @@ -50,6 +50,28 @@ > } > }, > > +{ > + /* Creative BT-D1 */ > + USB_DEVICE(0x041e, 0x0005), > + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { > + .ifnum = 1, > + .type = QUIRK_AUDIO_FIXED_ENDPOINT, > + .data = &(const struct audioformat) { > + .formats = SNDRV_PCM_FMTBIT_S16_LE, > + .channels = 2, > + .iface = 1, > + .altsetting = 1, > + .altset_idx = 1, > + .endpoint = 0x03, > + .ep_attr = USB_ENDPOINT_XFER_ISOC, > + .attributes = 0, > + .rates = SNDRV_PCM_RATE_CONTINUOUS, > + .rate_min = 48000, > + .rate_max = 48000, > + } > + } > +}, > + > /* Creative/Toshiba Multimedia Center SB-0500 */ > { > USB_DEVICE(0x041e, 0x3048), >

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Re: [alsa-devel] [PATCH] pas2: Added a fix where a sound card failure should not result in reading beyond array memory index.
by Takashi Iwai 03 Jan '13

03 Jan '13

At Fri, 28 Dec 2012 12:28:57 -0600, Asim Kadav wrote: > > Signed-off-by: Asim Kadav <kadav(a)cs.wisc.edu> > --- > sound/oss/pas2_card.c | 4 ++++ > 1 file changed, 4 insertions(+) > > diff --git a/sound/oss/pas2_card.c b/sound/oss/pas2_card.c > index dabf8a8..0b8cf64 100644 > --- a/sound/oss/pas2_card.c > +++ b/sound/oss/pas2_card.c > @@ -332,6 +332,10 @@ static void __init attach_pas_card(struct address_info *hw_config) > if ((pas_model = pas_read(0xFF88))) > { > char temp[100]; > + if ((int )pas_model < 0 || (int )pas_model >= 5) { Try to run scripts/checkpatch.pl. You'll get a few errors there. But, you can simply remove the cast to int instead of fixing the coding style issues since the cast is superfluous. Also, use ARRAY_SIZE() instead of a magic number 5. > + printk ("pas2card unrecognized model.\n"); Put a KERN_ERR prefix. thanks, Takashi > + return; > + } > > sprintf(temp, > "%s rev %d", pas_model_names[(int) pas_model], > -- > 1.7.11.3

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[alsa-devel] ASoC: EA3250 V2 machine driver
by Bryan Kaminski 02 Jan '13

02 Jan '13

Hello, I have an Embedded Artists LPC3250 V2 development board, which has the NXP UDA1380 audio codec on-board. I'm wondering if someone could help me get started on making an ASoC driver for it. There already exists a machine driver for a similar Phytec 3250 development board, with the same audio codec. I think it should be possible to modify the existing Phytec driver, but I'm unable to accomplish that. I modified some source and everything compiled, but I can't get the driver to load correctly. After loading the module with 'modprobe', I'm unable to get ALSA to recognize any sound cards. Can anyone give me a jump-start on this? Sorry if this is the wrong list to post on. I would post at the LPCLinux official site but it has been down for multiple weeks now. :| Bryan

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[alsa-devel] [PATCH 0/2] ASoC: palm27x: machine driver fixes and updates
by Mike Dunn 02 Jan '13

02 Jan '13

Hi, A fix and an update to the palm27x machine driver. Happy New Year! Mike Dunn (2): ASoC: palm27x: fix widgets and routes in dai_link init ASoC: palm27x: register card in platform_driver probe sound/soc/pxa/palm27x.c | 36 +++++++++--------------------------- 1 files changed, 9 insertions(+), 27 deletions(-) -- 1.7.8.6

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[alsa-devel] [PATCH 0/8] ASoC: OMAP-twl4030: Add support for more3 boards
by Peter Ujfalusi 02 Jan '13

02 Jan '13

Hello, This series will update the omap-twl4030 machine driver and the twl4030 codec driver to enable support for more boards. With this change we can remove two more machine drivers: zoom2 and sdp3430 since they will be supported by the common omap-twl4030 machine driver. The arch/arm/mach-omap2/ patches will go as a separate series to avoid cross tree issues. Regards, Peter --- Peter Ujfalusi (8): ASoC: twl4030: Correct the support for Voice port ASoC: zoom2: No need to configure the Voice port anymore ASoC: sdp3430: No need to configure the Voice port anymore ASoC: twl4030: Convert MICBIAS to SUPPLY widget ASoC: twl4030: Configure extmute pinmux when the dedicated pin is in use ASoC: sdp3430: No need to configure pin mux for extmute ASoC: omap-twl4030: Add support for routing, voice port and jack detect ASoC: OMAP: Remove obsolete machine drivers for Zoom2 and SDP3430 .../devicetree/bindings/sound/omap-twl4030.txt | 46 ++++ sound/soc/codecs/twl4030.c | 71 ++++-- sound/soc/omap/Kconfig | 19 +- sound/soc/omap/Makefile | 4 - sound/soc/omap/omap-twl4030.c | 204 ++++++++++++++- sound/soc/omap/omap3pandora.c | 8 +- sound/soc/omap/sdp3430.c | 278 --------------------- sound/soc/omap/zoom2.c | 207 --------------- 8 files changed, 304 insertions(+), 533 deletions(-) delete mode 100644 sound/soc/omap/sdp3430.c delete mode 100644 sound/soc/omap/zoom2.c -- 1.8.0.2

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[alsa-devel] [PATCH] ASoC: pxa2xx-ac97: fix oops when codec_driver probe runs
by Mike Dunn 02 Jan '13

02 Jan '13

This patch fixes a kernel oops that occurs when the snd_soc_codec_driver probe function runs. The exception occurs when the codec calls soc_ac97_ops.reset(). The pxa2xx-ac97's clocks have not been initialized at this point, but the cpu's reset function (pxa_ac97_cold_pxa27x()) expects them to be, resulting in a NULL pointer dereference in the pxa's clk_enable(). The pxa2xx-ac97's clocks are initialized in pxa2xx_ac97_hw_probe(). To ensure that the clocks are initialized when the codec_driver probe runs, the calls to pxa2xx_ac97_hw_{probe,remove}() are moved from the cpu's dai_driver probe/remove functions to those of its platform_driver. This is necessary because snd_soc_instantiate_card() always calls the codec_driver probe before any of the dai_drivers' probes. This change leaves nothing in the cpu's dai_driver probe/remove functions, so those functions are eliminated. Tested on the pxa27x; palmtreo680 machine using the wm9712 codec. Signed-off-by: Mike Dunn <mikedunn(a)newsguy.com> --- I'm new to the ASoC code. Experts, please have a look. Thanks! sound/soc/pxa/pxa2xx-ac97.c | 38 +++++++++++++++++++------------------- 1 files changed, 19 insertions(+), 19 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 4b0a009..940e9f4 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -104,17 +104,6 @@ static int pxa2xx_ac97_resume(struct snd_soc_dai *dai) #define pxa2xx_ac97_resume NULL #endif -static int pxa2xx_ac97_probe(struct snd_soc_dai *dai) -{ - return pxa2xx_ac97_hw_probe(to_platform_device(dai->dev)); -} - -static int pxa2xx_ac97_remove(struct snd_soc_dai *dai) -{ - pxa2xx_ac97_hw_remove(to_platform_device(dai->dev)); - return 0; -} - static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *cpu_dai) @@ -184,8 +173,6 @@ static struct snd_soc_dai_driver pxa_ac97_dai_driver[] = { { .name = "pxa2xx-ac97", .ac97_control = 1, - .probe = pxa2xx_ac97_probe, - .remove = pxa2xx_ac97_remove, .suspend = pxa2xx_ac97_suspend, .resume = pxa2xx_ac97_resume, .playback = { @@ -236,21 +223,34 @@ EXPORT_SYMBOL_GPL(soc_ac97_ops); static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) { + int ret; + if (pdev->id != -1) { dev_err(&pdev->dev, "PXA2xx has only one AC97 port.\n"); return -ENXIO; } - /* Punt most of the init to the SoC probe; we may need the machine - * driver to do interesting things with the clocking to get us up - * and running. - */ - return snd_soc_register_dais(&pdev->dev, pxa_ac97_dai_driver, - ARRAY_SIZE(pxa_ac97_dai_driver)); + ret = snd_soc_register_dais(&pdev->dev, pxa_ac97_dai_driver, + ARRAY_SIZE(pxa_ac97_dai_driver)); + if (ret < 0) { + dev_err(&pdev->dev, "snd_soc_register_dais failed with %d\n", + ret); + return ret; + } + + ret = pxa2xx_ac97_hw_probe(pdev); + if (ret < 0) { + dev_err(&pdev->dev, "pxa2xx_ac97_hw_probe failed with %d\n", + ret); + snd_soc_unregister_dais(&pdev->dev, + ARRAY_SIZE(pxa_ac97_dai_driver)); + } + return ret; } static int pxa2xx_ac97_dev_remove(struct platform_device *pdev) { + pxa2xx_ac97_hw_remove(pdev); snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(pxa_ac97_dai_driver)); return 0; } -- 1.7.8.6

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[alsa-devel] Looking for ALSA developer
by Eben Upton 01 Jan '13

01 Jan '13

We're looking for someone to do some contract development on the ALSA driver for the Raspberry Pi. Would anyone here be interested in taking this on? A little background: Our driver is architecturally fairly similar to a USB sound card driver, sitting on top of a stream-like connection to a remote entity implemented in software on an internal DSP. Due to our limited understanding of ALSA internals, there are a number of bugs which lead to underrun conditions in common audio applications, and we'd like to pay someone on and hourly basis to go through the failing cases, root cause them, and propose patches. Eben Upton Executive Director, Raspberry Pi Foundation

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[alsa-devel] [PATCH 0/6] ARM: OMAP3: Audio support update for omap-twl4030
by Peter Ujfalusi 31 Dec '12

31 Dec '12

Hello, omap-twl4030 ASoC machine driver can support more boards with the following series: http://mailman.alsa-project.org/pipermail/alsa-devel/2012-December/058119.h… Rework the omap-twl4030 audio support code in arch/arm/mach-omap2/ to take advantages of this. The same machine driver can support the zoom2 and sdp3430 boards as well. Regards, Peter --- Peter Ujfalusi (6): ARM: OMAP: 3430sdp: Enable extmute functionality for audio ARM: OMAP: zoom: Zoom2 does not have extmute functionality ARM: OMAP2+: twl-common: Add default twl4030 audio configuration ARM: OMAP2+: twl-common: Allow boards to customize the twl4030 audio setup ARM: OMAP: zoom: Audio support via the common omap-twl4030 machine driver ARM: OMAP: sdp3430: Audio support via the common omap-twl4030 machine driver arch/arm/mach-omap2/board-3430sdp.c | 20 +++++++++++++++++++ arch/arm/mach-omap2/board-cm-t35.c | 2 +- arch/arm/mach-omap2/board-devkit8000.c | 2 +- arch/arm/mach-omap2/board-igep0020.c | 2 +- arch/arm/mach-omap2/board-omap3beagle.c | 2 +- arch/arm/mach-omap2/board-omap3evm.c | 2 +- arch/arm/mach-omap2/board-overo.c | 2 +- arch/arm/mach-omap2/board-zoom-peripherals.c | 29 ++++++++++++++++++++-------- arch/arm/mach-omap2/twl-common.c | 17 ++++++++++------ arch/arm/mach-omap2/twl-common.h | 3 ++- 10 files changed, 60 insertions(+), 21 deletions(-) -- 1.8.0.2

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Re: [alsa-devel] Help with adding mic bias to omap-twl4030 audio driver
by Peter Ujfalusi 31 Dec '12

31 Dec '12

Hi Neil, On 12/18/2012 12:18 AM, NeilBrown wrote: > Hi Peter, > I recently discovered your omap-twl4030 driver and thought to use it to > replace the hacked-around driver I'm using for the GTA04 mobile phone > platform. > > It works fine for playing sound out the speaker, but the microphone doesn't > work - I think because there is no mechanism to enable the microphone bias. > > I copied some bits out of the old driver - just enough to make it work. > However this probably makes the code specific to my platform. > > So I'm wondering if you could give me some guidelines on how this "should" > be done. Should these tables be passed in with the platform_data (I haven't > graduated to device-tree yet)? I'm going to send a series soon to update the omap-twl4030 machine driver. This update going to add support for microphones and all other routings as well. With the upcoming series the omap-twl4030 will support zoom2 and sdp3430. I'll CC you with the series so you can take a look if it is going to help with your board. > > My 'hacked-up' driver has a few more entries in this table than just those > few, but I wanted to strip it down to what is needed for the main microphone > to work. > > Thanks for any help you can provide, > NeilBrown > > P.S is there some good reason to use the name > struct omap_tw4030_pdata > rather than > struct omap_twl4030_pdata > > (i.e. with the 'l')?? > > diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/omap/omap-twl4030.c > index 3b97b879..aa4d866 100644 > --- a/sound/soc/omap/omap-twl4030.c > +++ b/sound/soc/omap/omap-twl4030.c > @@ -87,6 +87,32 @@ static struct snd_soc_ops omap_twl4030_ops = { > .hw_params = omap_twl4030_hw_params, > }; > > +static const struct snd_soc_dapm_widget gta04_dapm_widgets[] = { > + SND_SOC_DAPM_MIC("Internal Mic", NULL), > +}; > + > +static const struct snd_soc_dapm_route audio_map[] = { > + {"MAINMIC", NULL, "Mic Bias 1"}, > + {"Mic Bias 1", NULL, "Internal Mic"}, > +}; > + > +static int omap3gta04_init(struct snd_soc_pcm_runtime *runtime) > +{ > + int ret; > + struct snd_soc_codec *codec = runtime->codec; > + struct snd_soc_dapm_context *dapm = &codec->dapm; > + > + ret = snd_soc_dapm_new_controls(dapm, gta04_dapm_widgets, > + ARRAY_SIZE(gta04_dapm_widgets)); > + if (ret < 0) > + return ret; > + > + snd_soc_dapm_add_routes(dapm, audio_map, > + ARRAY_SIZE(audio_map)); > + > + return snd_soc_dapm_sync(dapm); > +} > + > /* Digital audio interface glue - connects codec <--> CPU */ > static struct snd_soc_dai_link omap_twl4030_dai_links[] = { > { > @@ -97,6 +123,7 @@ static struct snd_soc_dai_link omap_twl4030_dai_links[] = { > .platform_name = "omap-pcm-audio", > .codec_name = "twl4030-codec", > .ops = &omap_twl4030_ops, > + .init = &omap3gta04_init, > }, > }; > > -- Péter

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[alsa-devel] [PATCH v2 0/2] ASoC: palm27x: machine driver fixes and updates
by Mike Dunn 29 Dec '12

29 Dec '12

Hi, A fix and an update to the palm27x machine driver. Change log: v2: Patch #2: in palm27x_asoc_remove(), the call to platform_device_unregister() is replaced with snd_soc_unregister_card(). This version v2 is unsolicited; I noticed the omission today. Patch preparation error; tripping over my own patches. No new ones for a while! Thanks for your patience. Mike Dunn (2): ASoC: palm27x: fix widgets and routes in dai_link init ASoC: palm27x: register card in platform_driver probe sound/soc/pxa/palm27x.c | 38 ++++++++++---------------------------- 1 files changed, 10 insertions(+), 28 deletions(-) -- 1.7.8.6

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[alsa-devel] [PATCH] ALSA: usb-audio: Add quirk for Focusrite Scarlett 18i6
by Martin Schwenke 29 Dec '12

29 Dec '12

Probing this device currently fails in snd_usb_audio_probe() because the call to snd_usb_create_mixer() fails. This is due to unknown or non-standard interface descriptor subtypes in parse_audio_unit(): usbaudio: unit 51: unexpected type 0x09 snd-usb-audio: probe of 1-8:1.0 failed with error -5 Some people are working around this by recompiling usb-audio with the call to snd_usb_create_mixer() commented out. It would be nice to avoid that. While the best idea would be to look into the mixer creation failure, a reasonable short-term solution is to use quirks to only probe the trouble-free interfaces. This allows audio and MIDI interfaces to be used without any obvious issues. Interface 0 is the main one to ignore. It contains lots of control-fu, including the unexpected interface descriptor subtypes. Interface 5 is for firmware updates and I'm not sure how to get support for this. Interface 3 is some sort of control interface that I don't understand: Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 9 bInCollection 1 baInterfaceNr( 0) 1 Signed-off-by: Martin Schwenke <martin(a)meltin.net> --- sound/usb/quirks-table.h | 52 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 52 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 88d8ceb..c9dde5f 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2882,6 +2882,58 @@ YAMAHA_DEVICE(0x7010, "UB99"), { /* + * Focusrite Scarlett 18i6 + * + * Avoid mixer creation, which otherwise fails because some of + * the interface descriptor subtypes for interface 0 are + * unknown. That should be fixed or worked-around but this at + * least allows the device to be used successfully with a DAW + * and an external mixer. See comments below about other + * ignored interfaces. + */ + USB_DEVICE(0x1235, 0x8004), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Focusrite", + .product_name = "Scarlett 18i6", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = & (const struct snd_usb_audio_quirk[]) { + { + /* InterfaceSubClass 1 (Control Device) */ + .ifnum = 0, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + /* InterfaceSubClass 1 (Control Device) */ + .ifnum = 3, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = 4, + .type = QUIRK_MIDI_STANDARD_INTERFACE + }, + { + /* InterfaceSubClass 1 (Device Firmware Update) */ + .ifnum = 5, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = -1 + } + } + } +}, + +{ + /* * Some USB MIDI devices don't have an audio control interface, * so we have to grab MIDI streaming interfaces here. */ -- 1.7.10.4

4 5

[alsa-devel] EMU-1212m "no Audiodock - no firmware" patch
by oeai 29 Dec '12

29 Dec '12

Aloha! i've been posting about my problem earlier, but i decided to DIY the problem was that when going up from suspend the firmware for additional card (that i don't have) was trying to load, -but there's no card - sorry, i'm gonna check few more thousand times that caused my cpu to load insanely so in the patch i've changed few more strings for debugging but actually the one (but not the only) bug is here *** 728,739 **** ! if (reg & EMU_HANA_OPTION_DOCK_OFFLINE) { --- 729,740 ---- ! if (reg & !EMU_HANA_OPTION_DOCK_OFFLINE) { the thing is in the "!"EMU_HANA_OPTION_DOCK_OFFLINE -> then the firmware should be loaded. and to debug some errors if something i've put the one more else state so if no audiodock - it could continue to load config *** 777,784 **** /* Unmute all. Default is muted after a firmware load */ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE); } } ! snd_printk(KERN_INFO "emu1010: firmware thread stopping\n"); return 0; } --- 778,790 ---- /* Unmute all. Default is muted after a firmware load */ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE); } + else if ((reg & 0x3f) == 0x15) + { + snd_printk(KERN_INFO "emu1212m: trying to load config emu_hana_fpga_config"); + snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG,0x02); + } } ! snd_printk(KERN_INFO "emu1010: emu dock firmware thread stopping\n"); return 0; } *************** then i've commented the power_down for audiodock - it should be in other place, but as i'm 1212m user - it works for me - you know. *** 841,847 **** outl(0x0005a000, emu->port + HCFG); /* Disable 48Volt power to Audio Dock */ ! snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, 0); /* ID, should read & 0x7f = 0x55. (Bit 7 is the IRQ bit) */ snd_emu1010_fpga_read(emu, EMU_HANA_ID, &reg); --- 847,853 ---- outl(0x0005a000, emu->port + HCFG); /* Disable 48Volt power to Audio Dock */ ! /*snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, 0);*/ /* ID, should read & 0x7f = 0x55. (Bit 7 is the IRQ bit) */ snd_emu1010_fpga_read(emu, EMU_HANA_ID, &reg); *************** and here i've changed that the emu_model was firstly enabled and then ecard - since i don't know what's the ecard i thought that it is a second module of my emu card. *** 2079,2088 **** { if (emu->card_capabilities->ca_cardbus_chip) snd_emu10k1_cardbus_init(emu); ! if (emu->card_capabilities->ecard) ! snd_emu10k1_ecard_init(emu); ! else if (emu->card_capabilities->emu_model) snd_emu10k1_emu1010_init(emu); else snd_emu10k1_ptr_write(emu, AC97SLOT, 0, AC97SLOT_CNTR|AC97SLOT_LFE); snd_emu10k1_init(emu, emu->enable_ir, 1); --- 2098,2111 ---- { if (emu->card_capabilities->ca_cardbus_chip) snd_emu10k1_cardbus_init(emu); ! ! if (emu->card_capabilities->emu_model) ! { snd_emu10k1_emu1010_init(emu); + + } + else if (emu->card_capabilities->ecard) + snd_emu10k1_ecard_init(emu); else snd_emu10k1_ptr_write(emu, AC97SLOT, 0, AC97SLOT_CNTR|AC97SLOT_LFE); snd_emu10k1_init(emu, emu->enable_ir, 1); Now i still have the problem of waking up i cannot request_firmware() properly on resume and it is trying to load nothing and other changes in patch - that was trying to debug the situation so i can at least alsa-reload at anytime. also i'd like to ask if i can use the ctxfi modules - CA0110-IBG i got xfi card (it is second of 5) that actually uses as i believe the same logic - maybe a lil modified, but not so much imho. -- RA OEAI

3 8

[alsa-devel] Fwd: Request for HW auto-recognition: HP Pavilion dv7 with STAC92xx
by Nicolas Maître 27 Dec '12

27 Dec '12

Hi, At Wed, 12 Dec 2012 18:08:25 +0100, Tomas Pospisek wrote: > > Hello, > > I have a HP Pavilion dv7 laptop (LS066EA#UUZ), with: > > # cat /proc/asound/card0/codec* | grep Codec > Codec: IDT 92HD81B1X5 > Codec: Intel CougarPoint HDMI > > when play sound without any tweaks, I get a very weak sound output. If I > want to have "beats audio", that is subwoofer and all speakers, I need to > set: > > # cat /etc/modprobe.d/alsa-base.conf > options snd-hda-intel model=hp-dv7-4000 > > Could the snd_hda_intel alsa driver be updated to recognize and setup the > driver accordingly? > > alsa-info.sh output here: > > > http://www.alsa-project.org/db/?f=54ea874f43f12c1572aff3ee3113dc4b0ad0b398 I'm not quite sure that I have the exact same hardware, but I think so. I have 5 speakers (rear left and right, front left and right + subwoofer) Here is my alsa-info: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1056669/+attachm… In my case, model=hp-dv7-4000 doesn't fully solve the problem. The subwoofer works but it is not mapped as such. After some tests, it seems that the alsa's speakers rear left and rear right are mapped to the subwoofer. The front ones are alright. With the model=hp-dv5 that I'm using, the subwoofer isnt recognized at all, but the rear speakers are correctly mapped. Also, with hp-dv7-4000, the internal microphone is still used when a micro jack is plugged in. That's not the case with hp-dv5. If you think there's a configuration issue on my side, please let me know, but I think hp-dv5 is actually a better default choice if I have to choose between both. Kind regards, -- nimai

4 7

Re: [alsa-devel] [PATCH 001/001] ASoC: Replace max98090 Device Driver
by Mark Brown 27 Dec '12

27 Dec '12

On Thu, Dec 13, 2012 at 11:19:29AM -0800, Jerry Wong wrote: For legibility please submit future revisions as a two stage patch series, one removing the old driver and another adding the new one. > +/* codec platform data */ > +struct max98090_pdata { > + > + int irq; This should be passed in using the normal mechanism provided by the bus, not custom implemented. > + int power_over_performance; This doesn't look like platform data to me, it looks like something that should be configured dynamically at runtime. > + /* Equalizers for DAC */ > + struct max98090_eq_cfg *eq_cfg; > + unsigned int eq_cfgcnt; Coefficients should be configured using SND_SOC_BYTES(). Some older drivers use this method but it's deprecated. > + { 0xE0, 0x00 }, /* E0 */ > + { 0xE1, 0x00 }, /* E1 */ > + { 0xE2, 0x00 }, /* E2 */ Do these number-only registers actually exist in the published register map or are they undefined? If they're undefined and not used by software there's no need to include them here. > + switch (reg) { > + case M98090_REG_01_IRQ_STATUS: > + case M98090_REG_02_JACK_STATUS: > + case M98090_REG_FF_REV_ID: Including the register number in the constants seems to rather defeat the point of having symbolic names for the registers; it means people still have to remember the register numbers. > +static int max98090_get_enab_tlv(struct snd_kcontrol *kcontrol, > + struct snd_ctl_elem_value *ucontrol) What's the relevance of _tlv here? It looks like these are normal get and sets. > +static const char * max98090_dsts_text[] = > + { "None", "Left", "Right", "Left and Right" }; > + > +static const struct soc_enum max98090_dsts_enum = > + SOC_ENUM_SINGLE(M98090_REG_1A_LVL_SIDETONE, M98090_DSTS_SHIFT, > + ARRAY_SIZE(max98090_dsts_text), max98090_dsts_text); > +static const struct snd_kcontrol_new max98090_snd_controls[] = { > + SOC_SINGLE("DMIC MIC Left Enable", M98090_REG_13_MIC_CFG1, > + M98090_DIGMICL_SHIFT, M98090_DIGMICL_NUM - 1, 0), > + SOC_SINGLE("DMIC MIC Right Enable", M98090_REG_13_MIC_CFG1, > + M98090_DIGMICR_SHIFT, M98090_DIGMICR_NUM - 1, 0), This looks like it should be DAPM. > + SOC_ENUM_EXT("External MIC Mux", max98090_extmic_mux_enum, > + max98090_extmic_mux_get, max98090_extmic_mux_set), Likewise. > + SOC_ENUM("Digital Sidetone Source", max98090_dsts_enum), DAPM. > + SOC_SINGLE_EXT_TLV("Digital Sidetone Level Volume", > + M98090_REG_1A_LVL_SIDETONE, M98090_DVST_SHIFT, > + M98090_DVST_NUM - 1, 1, max98090_get_enab_tlv, > + max98090_put_enab_tlv, max98090_micboost_tlv), Level Volume? > + SOC_SINGLE_TLV("Digital Gain Volume", M98090_REG_27_DAI_LVL, > + M98090_DAI_LVL_DVG_SHIFT, M98090_DAI_LVL_DVG_NUM - 1, 0, > + max98090_dvg_tlv), Gain Volume? > + SOC_SINGLE("Digital EQ 3 Band Enable", M98090_REG_41_DSP_EQ_EN, > + M98090_EQ3BANDEN_SHIFT, M98090_EQ3BANDEN_NUM - 1, 0), Switch. > + SOC_SINGLE("DAC HP Playback Performance Mode", M98090_REG_43_DAC_CFG, > + M98090_DAC_PERFMODE_SHIFT, M98090_DAC_PERFMODE_NUM - 1, 0), > + SOC_SINGLE("DAC High Performance Mode", M98090_REG_43_DAC_CFG, > + M98090_DACHP_SHIFT, M98090_DACHP_NUM - 1, 0), These should either be enumerations with real names of Switches if it's just an on/off (looks like the former). > + SOC_SINGLE_TLV("Headphone Left Volume", M98090_REG_2C_LVL_HP_LEFT, > + M98090_HPVOLL_SHIFT, M98090_HPVOLL_NUM - 1, 0, > + max98090_hp_tlv), > + SOC_SINGLE_TLV("Headphone Right Volume", M98090_REG_2D_LVL_HP_RIGHT, > + M98090_HPVOLR_SHIFT, M98090_HPVOLR_NUM - 1, 0, > + max98090_hp_tlv), Should be a single stereo control (and similarly for the other output controls). > + SOC_SINGLE("Quick Setup System Clock", M98090_REG_04_QCFG_SYS_CLK, > + M98090_CLK_ALL_SHIFT, M98090_CLK_ALL_NUM - 1, 0), What do these controls do? > + if (val >= 1) { > + if (w->reg == M98090_REG_10_LVL_MIC1) > + max98090->mic1pre = val - 1; /* Update for volatile */ > + else > + max98090->mic2pre = val - 1; /* Update for volatile */ > + } Use braces consistently within an if statemeent - if you use them for the outer one use them for the inner one too. > +static const struct snd_soc_dapm_route max98090_dapm_routes[] = { > + > + {"MIC1 Input", NULL, "MIC1"}, > + {"MIC2 Input", NULL, "MIC2"}, > + {"MIC1 Input", NULL, "MICBIAS"}, > + {"MIC2 Input", NULL, "MICBIAS"}, Unless this chip is *very* unusual I expect the bias is something that's connected externally and therefore these connections should be being made by the machine driver. > + > + /* Check for user calculated MI and NI ratios */ > + for (i = 0; i < ARRAY_SIZE(user_pclk_rates); i++) { > + if ((user_pclk_rates[i] == max98090->sysclk) && > + (user_lrclk_rates[i] == max98090->lrclk)) { > + dev_info(codec->dev, > + "Found user supported PCLK to LRCLK rates\n"); > + dev_info(codec->dev, "i %d ni %lld mi %lld\n", > + i, ni_value[i], mi_value[i]); dev_dbg() would be fine but dev_info() is too loud. > +/* > + * This accommodates an inverted logic in the MAX98090 chip > + * for Bit Clock Invert (BCI). The inverted logic is only seen > + * for the case of TDM mode. The remaining cases have normal logic. > + */ > + if (max98090->tdm_slots > 1) { > + regval ^= M98090_DAI_BCI_MASK; > + } Coding style, the indentation is inconsistent here and no braces for single line if () statements. > + if (max98090->jack_state == M98090_JACK_STATE_HEADSET) { > + /* > + * Set to normal bias level. > + */ > + snd_soc_update_bits(codec, M98090_REG_12_MIC_BIAS, > + M98090_MBVSEL_MASK, M98090_MBVSEL_2V4); > + > + snd_soc_update_bits(codec, M98090_REG_3E_PWR_EN_IN, > + M98090_PWR_MBEN_MASK, M98090_PWR_MBEN_MASK); > + } This looks suspicous, why does the headset state have any bearing here? > +static int max98090_dai_digital_mute(struct snd_soc_dai *codec_dai, int mute) > +{ > + struct snd_soc_codec *codec = codec_dai->codec; > + int regval; > + > + dev_info(codec->dev, "max98090_dai_digital_mute\n"); Drop this, it's not conveying any useful information. > +static void max98090_dmic_switch(struct snd_soc_codec *codec, int state) > +{ I have no idea what this function is supposed to do... > +static void max98090_headset_button_event(struct snd_soc_codec *codec) > +{ > + dev_info(codec->dev, "max98090_headset_button_event\n"); > +} Delete this, it doesn't do anything except log. > + switch (reg & (M98090_LSNS_MASK | M98090_JKSNS_MASK)) { > + case M98090_LSNS_MASK | M98090_JKSNS_MASK: > + { > + dev_info(codec->dev, "No Headset Detected\n"); > + > + max98090->jack_state = M98090_JACK_STATE_NO_HEADSET; > + > + max98090_set_bias_level(codec, SND_SOC_BIAS_STANDBY); > + > + snd_soc_dapm_disable_pin(&codec->dapm, "HPL"); > + snd_soc_dapm_disable_pin(&codec->dapm, "HPR"); > + snd_soc_dapm_force_enable_pin(&codec->dapm, "SPKL"); > + snd_soc_dapm_force_enable_pin(&codec->dapm, "SPKR"); > + snd_soc_dapm_disable_pin(&codec->dapm, "MIC1"); > + snd_soc_dapm_disable_pin(&codec->dapm, "MIC2"); > + snd_soc_dapm_force_enable_pin(&codec->dapm, "DMIC1"); > + snd_soc_dapm_force_enable_pin(&codec->dapm, "DMIC2"); > + max98090_dmic_switch(codec, 1); > + > + break; > + } This doesn't look obviously sensible... calling set_bias_level() outside of DAPM is going to get overridden in pretty short order, none of the DAPM pin setting looks like a good idea (especially not forcing on the speakers whenever there's no headset) and there's no DAPM sync to make any of this take effect. I'm a bit confused about the intended effect. I also can't see any interaction with soc-jack... What is this code actually intended to do? > + dev_info(codec->dev, "***** max98090_interrupt *****\n"); > + There's lots of these noisy, uninformative prints at info level throughout the driver. > + /* Send work to be scheduled */ > + if (active & M98090_IRQ_CLD_MASK) { > + dev_info(codec->dev, "M98090_IRQ_CLD_MASK\n"); > + } This doesn't actually schedule anything... > + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); > + if (ret != 0) { > + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); > + return ret; > + } SND_SOC_REGMAP. > + if ( (request_threaded_irq(pdata->irq, NULL, > + max98090_interrupt, IRQF_TRIGGER_FALLING, > + "max98090_interrupt", codec)) < 0) { > + dev_info(codec->dev, "request_irq failed\n"); > + } > + /* > + * Clear any old interrupts. Coding style and it's not terribly helpful to hide the return code from users. > + /* Turn on VCM bandgap reference */ > + snd_soc_write(codec, M98090_REG_42_BIAS_CNTL, > + M98090_VCM_MODE_MASK); This shouldn't be managed at runtime? > +/* > + * Driver revision > + */ > +#define MAX98090_REVISION "0.01.00" The kernel already has perfectly good versioning.

1 0

[alsa-devel] [PATCH 1/2] ASoC: wm8985: Refactor set_pll code to avoid gcc warnings
by Fabio Estevam 27 Dec '12

27 Dec '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> Refactor set_pll code to avoid the following warnings: sound/soc/codecs/wm8985.c:852:50: warning: 'pll_div.k' may be used uninitialized in this function sound/soc/codecs/wm8985.c:849:9: warning: 'pll_div.n' may be used uninitialized in this function sound/soc/codecs/wm8985.c:848:23: warning: 'pll_div.div2' may be used uninitialized in this function Do the same as in commit 86ce6c9a (ASoC: WM8804: Refactor set_pll code to avoid GCC warnings). Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/wm8985.c | 43 ++++++++++++++++++++----------------------- 1 file changed, 20 insertions(+), 23 deletions(-) diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c index ab37826..dd6ce3b 100644 --- a/sound/soc/codecs/wm8985.c +++ b/sound/soc/codecs/wm8985.c @@ -830,33 +830,30 @@ static int wm8985_set_pll(struct snd_soc_dai *dai, int pll_id, struct pll_div pll_div; codec = dai->codec; - if (freq_in && freq_out) { + if (!freq_in || !freq_out) { + /* disable the PLL */ + snd_soc_update_bits(codec, WM8985_POWER_MANAGEMENT_1, + WM8985_PLLEN_MASK, 0); + } else { ret = pll_factors(&pll_div, freq_out * 4 * 2, freq_in); if (ret) return ret; - } - /* disable the PLL before reprogramming it */ - snd_soc_update_bits(codec, WM8985_POWER_MANAGEMENT_1, - WM8985_PLLEN_MASK, 0); - - if (!freq_in || !freq_out) - return 0; - - /* set PLLN and PRESCALE */ - snd_soc_write(codec, WM8985_PLL_N, - (pll_div.div2 << WM8985_PLL_PRESCALE_SHIFT) - | pll_div.n); - /* set PLLK */ - snd_soc_write(codec, WM8985_PLL_K_3, pll_div.k & 0x1ff); - snd_soc_write(codec, WM8985_PLL_K_2, (pll_div.k >> 9) & 0x1ff); - snd_soc_write(codec, WM8985_PLL_K_1, (pll_div.k >> 18)); - /* set the source of the clock to be the PLL */ - snd_soc_update_bits(codec, WM8985_CLOCK_GEN_CONTROL, - WM8985_CLKSEL_MASK, WM8985_CLKSEL); - /* enable the PLL */ - snd_soc_update_bits(codec, WM8985_POWER_MANAGEMENT_1, - WM8985_PLLEN_MASK, WM8985_PLLEN); + /* set PLLN and PRESCALE */ + snd_soc_write(codec, WM8985_PLL_N, + (pll_div.div2 << WM8985_PLL_PRESCALE_SHIFT) + | pll_div.n); + /* set PLLK */ + snd_soc_write(codec, WM8985_PLL_K_3, pll_div.k & 0x1ff); + snd_soc_write(codec, WM8985_PLL_K_2, (pll_div.k >> 9) & 0x1ff); + snd_soc_write(codec, WM8985_PLL_K_1, (pll_div.k >> 18)); + /* set the source of the clock to be the PLL */ + snd_soc_update_bits(codec, WM8985_CLOCK_GEN_CONTROL, + WM8985_CLKSEL_MASK, WM8985_CLKSEL); + /* enable the PLL */ + snd_soc_update_bits(codec, WM8985_POWER_MANAGEMENT_1, + WM8985_PLLEN_MASK, WM8985_PLLEN); + } return 0; } -- 1.7.9.5

2 3

Re: [alsa-devel] [PATCH V5 2/2] ASoC: SAMSUNG: Add DT support for i2s
by Padma Venkat 27 Dec '12

27 Dec '12

Hi, On Sat, Dec 22, 2012 at 12:32 AM, Kukjin Kim <kgene.kim(a)samsung.com> wrote: > Padma Venkat wrote: >> >> Hi, >> >> On Wed, Dec 19, 2012 at 10:39 PM, Mark Brown >> <broonie(a)opensource.wolfsonmicro.com> wrote: >> > On Wed, Dec 19, 2012 at 01:24:14PM +0000, Grant Likely wrote: >> >> On Thu, 13 Dec 2012 16:12:53 +0530, Padmavathi Venna >> <padma.v(a)samsung.com> wrote: >> > >> >> > +- compatible : "samsung,samsung-i2s" >> > >> >> Isn't that kind of redundant? :-) >> > >> >> The format of the compatible strings should be "<vendor>,<part- >> number>-i2s". >> >> Please be specific about the part number that you're doing the binding >> >> for. For example; use "samsung,exynos4210-i2s" instead of >> "samsung,exynos-i2s". >> > >> > There are actually versioned IPs here (where the versions are used >> > publically in a few places) but it's not clearly documented which is >> > which. It would be reasonable to use the IP versions here I think. >> >> Samsung has three i2s drivers one for s3c24xx, one for s3c2412 and one >> for rest of the platforms. The above mentioned other platforms has >> Version 3/4/5 of i2s controllers. This dt binding is for for the i2s > > Where is the version defined such as 3, 4, 5? So, what is the > "sound/soc/Samsung/s3c-i2s-v2.[ch]"? Versions 3, 4, 5 are defined in dev-audio.c file of corresponding platforms. s3c-i2s-v2 is used in s3c2412 platform. > >> driver that has support for Version 3/4/5 of i2s controller. So >> "samsung,i2s-v5" is okay as compatible name? Please suggest me. >> > I agree with using version here but we need some consensus about that. > > - Kukjin > > -- > To unsubscribe from this list: send the line "unsubscribe linux-samsung-soc" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html Thanks Padma

2 2

[alsa-devel] [PATCH] ASoC: cs42l52: Convert to devm_input_allocate_device()
by Axel Lin 27 Dec '12

27 Dec '12

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs42l52.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 99bb1c6..73a8049 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -1040,7 +1040,7 @@ static void cs42l52_init_beep(struct snd_soc_codec *codec) struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); int ret; - cs42l52->beep = input_allocate_device(); + cs42l52->beep = devm_input_allocate_device(codec->dev); if (!cs42l52->beep) { dev_err(codec->dev, "Failed to allocate beep device\n"); return; @@ -1061,7 +1061,6 @@ static void cs42l52_init_beep(struct snd_soc_codec *codec) ret = input_register_device(cs42l52->beep); if (ret != 0) { - input_free_device(cs42l52->beep); cs42l52->beep = NULL; dev_err(codec->dev, "Failed to register beep device\n"); } @@ -1078,7 +1077,6 @@ static void cs42l52_free_beep(struct snd_soc_codec *codec) struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); device_remove_file(codec->dev, &dev_attr_beep); - input_unregister_device(cs42l52->beep); cancel_work_sync(&cs42l52->beep_work); cs42l52->beep = NULL; -- 1.7.9.5

2 1

Re: [alsa-devel] [PATCH 1/2] ASoC: core: fix the memory leak in case of device_add() failure
by Mark Brown 27 Dec '12

27 Dec '12

On Wed, Dec 26, 2012 at 12:56:05AM +0800, Chuansheng Liu wrote: > > After called device_initialize(), even device_add() returns > error, we still need use the put_device() to release the reference > to call rtd_release(), which will do the free() action. Applied both, thanks.

1 0

[alsa-devel] [PATCH] ASoC: cs42l52: Catch no-match case in cs42l52_get_clk
by Axel Lin 24 Dec '12

24 Dec '12

In the case of no-match, return -EINVAL instead of 0. Since we assign i to ret in the for loop, ret always less than ARRAY_SIZE(clk_map_table). Thus remove the boundary checking for ret. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs42l52.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 99bb1c6..9811a54 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -737,7 +737,7 @@ static const struct cs42l52_clk_para clk_map_table[] = { static int cs42l52_get_clk(int mclk, int rate) { - int i, ret = 0; + int i, ret = -EINVAL; u_int mclk1, mclk2 = 0; for (i = 0; i < ARRAY_SIZE(clk_map_table); i++) { @@ -749,8 +749,6 @@ static int cs42l52_get_clk(int mclk, int rate) } } } - if (ret > ARRAY_SIZE(clk_map_table)) - return -EINVAL; return ret; } -- 1.7.9.5

3 2

Re: [alsa-devel] [PATCH 3/7] ASoC: tegra: setup DAP3<->DAC3 connection by default
by Mark Brown 24 Dec '12

24 Dec '12

On Thu, Dec 20, 2012 at 12:17:32AM +0100, Lucas Stach wrote: > This connection is used by the AC97 controller. Applied, thanks.

1 0

Re: [alsa-devel] [PATCH 4/7] ASoC: tegra: add function to set ac97 rate
by Mark Brown 24 Dec '12

24 Dec '12

On Thu, Dec 20, 2012 at 12:17:33AM +0100, Lucas Stach wrote: > AC97 uses a fixed rate, unrelated to the sample rate. Add a function to > make the setup more trivial. Applied, though the above isn't strictly true - a number of later AC'97 CODECs had PLLs which allowed them to use non-standard AC'97 MCLKs.

1 0

[alsa-devel] [PATCH v2] ASoC: wm8993: Refactor set_pll code to avoid GCC warnings
by Fabio Estevam 24 Dec '12

24 Dec '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> Refactor set_pll code to avoid the following warnings: sound/soc/codecs/wm8983.c:873:40: warning: 'pll_div.k' may be used uninitialized in this function [-Wuninitialized] sound/soc/codecs/wm8983.c:870:9: warning: 'pll_div.n' may be used uninitialized in this function [-Wuninitialized] sound/soc/codecs/wm8983.c:869:23: warning: 'pll_div.div2' may be used uninitialized in this function [-Wuninitialized] Do the same as in commit 86ce6c9a (ASoC: WM8804: Refactor set_pll code to avoid GCC warnings). Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - Fix the comment sound/soc/codecs/wm8983.c | 41 ++++++++++++++++++++++------------------- 1 file changed, 22 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c index 9fe1e04..e7c73ff 100644 --- a/sound/soc/codecs/wm8983.c +++ b/sound/soc/codecs/wm8983.c @@ -851,30 +851,33 @@ static int wm8983_set_pll(struct snd_soc_dai *dai, int pll_id, struct pll_div pll_div; codec = dai->codec; - if (freq_in && freq_out) { + if (!freq_in || !freq_out) { + /* disable the PLL */ + snd_soc_update_bits(codec, WM8983_POWER_MANAGEMENT_1, + WM8983_PLLEN_MASK, 0); + return 0; + } else { ret = pll_factors(&pll_div, freq_out * 4 * 2, freq_in); if (ret) return ret; - } - - /* disable the PLL before re-programming it */ - snd_soc_update_bits(codec, WM8983_POWER_MANAGEMENT_1, - WM8983_PLLEN_MASK, 0); - if (!freq_in || !freq_out) - return 0; + /* disable the PLL before re-programming it */ + snd_soc_update_bits(codec, WM8983_POWER_MANAGEMENT_1, + WM8983_PLLEN_MASK, 0); + + /* set PLLN and PRESCALE */ + snd_soc_write(codec, WM8983_PLL_N, + (pll_div.div2 << WM8983_PLL_PRESCALE_SHIFT) + | pll_div.n); + /* set PLLK */ + snd_soc_write(codec, WM8983_PLL_K_3, pll_div.k & 0x1ff); + snd_soc_write(codec, WM8983_PLL_K_2, (pll_div.k >> 9) & 0x1ff); + snd_soc_write(codec, WM8983_PLL_K_1, (pll_div.k >> 18)); + /* enable the PLL */ + snd_soc_update_bits(codec, WM8983_POWER_MANAGEMENT_1, + WM8983_PLLEN_MASK, WM8983_PLLEN); + } - /* set PLLN and PRESCALE */ - snd_soc_write(codec, WM8983_PLL_N, - (pll_div.div2 << WM8983_PLL_PRESCALE_SHIFT) - | pll_div.n); - /* set PLLK */ - snd_soc_write(codec, WM8983_PLL_K_3, pll_div.k & 0x1ff); - snd_soc_write(codec, WM8983_PLL_K_2, (pll_div.k >> 9) & 0x1ff); - snd_soc_write(codec, WM8983_PLL_K_1, (pll_div.k >> 18)); - /* enable the PLL */ - snd_soc_update_bits(codec, WM8983_POWER_MANAGEMENT_1, - WM8983_PLLEN_MASK, WM8983_PLLEN); return 0; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: wm8804: Remove redundant check
by Fabio Estevam 24 Dec '12

24 Dec '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> The condition "if (!freq_in || !freq_out)" has already been tested previously, so no need to do it again. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/wm8804.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index d321a87..1704b1e 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -395,9 +395,6 @@ static int wm8804_set_pll(struct snd_soc_dai *dai, int pll_id, /* power down the PLL before reprogramming it */ snd_soc_update_bits(codec, WM8804_PWRDN, 0x1, 0x1); - if (!freq_in || !freq_out) - return 0; - /* set PLLN and PRESCALE */ snd_soc_update_bits(codec, WM8804_PLL4, 0xf | 0x10, pll_div.n | (pll_div.prescale << 4)); -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ALSA: ASoC: cs4271: add optional soft reset workaround
by Daniel Mack 24 Dec '12

24 Dec '12

The CS4271 requires its LRCLK and MCLK to be stable before its RESET line is de-asserted. That also means that clocks cannot be changed without putting the chip back into hardware reset, which also requires a complete re-initialization of all registers. One (undocumented) workaround is to assert and de-assert the PDN bit in the MODE2 register. This patch adds a new flag to both the DT bindings as well as to the platform data to enable that workaround. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- Documentation/devicetree/bindings/sound/cs4271.txt | 12 ++++++++ include/sound/cs4271.h | 15 ++++++++++ sound/soc/codecs/cs4271.c | 34 ++++++++++++++++++++++ 3 files changed, 61 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/cs4271.txt b/Documentation/devicetree/bindings/sound/cs4271.txt index a850fb9..e2cd1d7 100644 --- a/Documentation/devicetree/bindings/sound/cs4271.txt +++ b/Documentation/devicetree/bindings/sound/cs4271.txt @@ -20,6 +20,18 @@ Optional properties: !RESET pin - cirrus,amuteb-eq-bmutec: When given, the Codec's AMUTEB=BMUTEC flag is enabled. + - cirrus,enable-soft-reset: + The CS4271 requires its LRCLK and MCLK to be stable before its RESET + line is de-asserted. That also means that clocks cannot be changed + without putting the chip back into hardware reset, which also requires + a complete re-initialization of all registers. + + One (undocumented) workaround is to assert and de-assert the PDN bit + in the MODE2 register. This workaround can be enabled with this DT + property. + + Note that this is not needed in case the clocks are stable + throughout the entire runtime of the codec. Examples: diff --git a/include/sound/cs4271.h b/include/sound/cs4271.h index dd8c48d..70f4535 100644 --- a/include/sound/cs4271.h +++ b/include/sound/cs4271.h @@ -20,6 +20,21 @@ struct cs4271_platform_data { int gpio_nreset; /* GPIO driving Reset pin, if any */ bool amutec_eq_bmutec; /* flag to enable AMUTEC=BMUTEC */ + + /* + * The CS4271 requires its LRCLK and MCLK to be stable before its RESET + * line is de-asserted. That also means that clocks cannot be changed + * without putting the chip back into hardware reset, which also requires + * a complete re-initialization of all registers. + * + * One (undocumented) workaround is to assert and de-assert the PDN bit + * in the MODE2 register. This workaround can be enabled with the + * following flag. + * + * Note that this is not needed in case the clocks are stable + * throughout the entire runtime of the codec. + */ + bool enable_soft_reset; }; #endif /* __CS4271_H */ diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index ac99238..ffa72dd 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -167,6 +167,8 @@ struct cs4271_private { int gpio_nreset; /* GPIO that disable serial bus, if any */ int gpio_disable; + /* enable soft reset workaround */ + bool enable_soft_reset; }; /* @@ -325,6 +327,33 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, int i, ret; unsigned int ratio, val; + if (cs4271->enable_soft_reset) { + /* + * Put the codec in soft reset and back again in case it's not + * currently streaming data. This way of bringing the codec in + * sync to the current clocks is not explicitly documented in + * the data sheet, but it seems to work fine, and in contrast + * to a read hardware reset, we don't have to sync back all + * registers every time. + */ + + if ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK && + !dai->capture_active) || + (substream->stream == SNDRV_PCM_STREAM_CAPTURE && + !dai->playback_active)) { + ret = snd_soc_update_bits(codec, CS4271_MODE2, + CS4271_MODE2_PDN, + CS4271_MODE2_PDN); + if (ret < 0) + return ret; + + ret = snd_soc_update_bits(codec, CS4271_MODE2, + CS4271_MODE2_PDN, 0); + if (ret < 0) + return ret; + } + } + cs4271->rate = params_rate(params); /* Configure DAC */ @@ -484,6 +513,10 @@ static int cs4271_probe(struct snd_soc_codec *codec) if (of_get_property(codec->dev->of_node, "cirrus,amutec-eq-bmutec", NULL)) amutec_eq_bmutec = true; + + if (of_get_property(codec->dev->of_node, + "cirrus,enable-soft-reset", NULL)) + cs4271->enable_soft_reset = true; } #endif @@ -492,6 +525,7 @@ static int cs4271_probe(struct snd_soc_codec *codec) gpio_nreset = cs4271plat->gpio_nreset; amutec_eq_bmutec = cs4271plat->amutec_eq_bmutec; + cs4271->enable_soft_reset = cs4271plat->enable_soft_reset; } if (gpio_nreset >= 0) -- 1.7.11.7

3 2

[alsa-devel] [PATCH] ASoC: soc-core: Remove unused 'ret' variable
by Fabio Estevam 24 Dec '12

24 Dec '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> commit 9bde4f0b1c (ASoC: core: Fix SOC_DOUBLE_RANGE() macros) introduced the following build warning: sound/soc/soc-core.c:2999:6: warning: unused variable 'ret' [-Wunused-variable] Remove the unused 'ret' variable. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/soc-core.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e0d4630..29046e7 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2996,7 +2996,6 @@ int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol, int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; - int ret; ucontrol->value.integer.value[0] = (snd_soc_read(codec, reg) >> shift) & mask; -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: atmel-soc: make it buildable on other architectures
by Joachim Eastwood 24 Dec '12

24 Dec '12

Not very useful on non AT91/AVR32 platforms but it provides more build coverage and prepares for ARM multiplatform. Also fixes a couple of format type warnings. Signed-off-by: Joachim Eastwood <manabian(a)gmail.com> --- Hi, Build tested on RM9200 and x86_64. ARCH_AT91 depend is pushed down to the machine drivers since they still relies on mach includes. regards Joachim Eastwood sound/soc/atmel/Kconfig | 6 +++--- sound/soc/atmel/atmel-pcm-pdc.c | 4 ++-- sound/soc/atmel/atmel-pcm.c | 2 +- sound/soc/atmel/atmel_ssc_dai.c | 2 -- 4 files changed, 6 insertions(+), 8 deletions(-) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index d1b691b..3fdd87f 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -1,6 +1,6 @@ config SND_ATMEL_SOC tristate "SoC Audio for the Atmel System-on-Chip" - depends on ARCH_AT91 + depends on HAS_IOMEM help Say Y or M if you want to add support for codecs attached to the ATMEL SSC interface. You will also need @@ -24,7 +24,7 @@ config SND_ATMEL_SOC_SSC config SND_AT91_SOC_SAM9G20_WM8731 tristate "SoC Audio support for WM8731-based At91sam9g20 evaluation board" - depends on ATMEL_SSC && SND_ATMEL_SOC && AT91_PROGRAMMABLE_CLOCKS + depends on ARCH_AT91 && ATMEL_SSC && SND_ATMEL_SOC && AT91_PROGRAMMABLE_CLOCKS select SND_ATMEL_SOC_PDC select SND_ATMEL_SOC_SSC select SND_SOC_WM8731 @@ -34,7 +34,7 @@ config SND_AT91_SOC_SAM9G20_WM8731 config SND_AT91_SOC_AFEB9260 tristate "SoC Audio support for AFEB9260 board" - depends on ATMEL_SSC && ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC + depends on ARCH_AT91 && ATMEL_SSC && ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC select SND_ATMEL_SOC_PDC select SND_ATMEL_SOC_SSC select SND_SOC_TLV320AIC23 diff --git a/sound/soc/atmel/atmel-pcm-pdc.c b/sound/soc/atmel/atmel-pcm-pdc.c index 6a293c7..054ea4d 100644 --- a/sound/soc/atmel/atmel-pcm-pdc.c +++ b/sound/soc/atmel/atmel-pcm-pdc.c @@ -159,7 +159,7 @@ static int atmel_pcm_hw_params(struct snd_pcm_substream *substream, pr_debug("atmel-pcm: " "hw_params: DMA for %s initialized " - "(dma_bytes=%u, period_size=%u)\n", + "(dma_bytes=%zu, period_size=%zu)\n", prtd->params->name, runtime->dma_bytes, prtd->period_size); @@ -201,7 +201,7 @@ static int atmel_pcm_trigger(struct snd_pcm_substream *substream, int ret = 0; pr_debug("atmel-pcm:buffer_size = %ld," - "dma_area = %p, dma_bytes = %u\n", + "dma_area = %p, dma_bytes = %zu\n", rtd->buffer_size, rtd->dma_area, rtd->dma_bytes); switch (cmd) { diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c index e99f181..3109db7 100644 --- a/sound/soc/atmel/atmel-pcm.c +++ b/sound/soc/atmel/atmel-pcm.c @@ -49,7 +49,7 @@ static int atmel_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, buf->private_data = NULL; buf->area = dma_alloc_coherent(pcm->card->dev, size, &buf->addr, GFP_KERNEL); - pr_debug("atmel-pcm: alloc dma buffer: area=%p, addr=%p, size=%d\n", + pr_debug("atmel-pcm: alloc dma buffer: area=%p, addr=%p, size=%zu\n", (void *)buf->area, (void *)buf->addr, size); if (!buf->area) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 1c76634..2755750 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -42,8 +42,6 @@ #include <sound/initval.h> #include <sound/soc.h> -#include <mach/hardware.h> - #include "atmel-pcm.h" #include "atmel_ssc_dai.h" -- 1.8.0

4 4

[alsa-devel] [PATCH] ASoC: atmel-ssc: make it buildable on other architectures
by Joachim Eastwood 24 Dec '12

24 Dec '12

Not very useful on non AT91/AVR32 platforms but it provides more build coverage and prepares for ARM multiplatform. Also fixes a truncated warning that would come when building on a 64-bit arch. Signed-off-by: Joachim Eastwood <manabian(a)gmail.com> --- Hi Mark, Sending it to you since seem like you're the last one who commited changes to the driver and the driver is mostly used on ASoC. Tested on RM9200 and build tested on x86_64. regards Joachim Eastwood drivers/misc/Kconfig | 2 +- drivers/misc/atmel-ssc.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/misc/Kconfig b/drivers/misc/Kconfig index b151b7c..8f59d88 100644 --- a/drivers/misc/Kconfig +++ b/drivers/misc/Kconfig @@ -192,7 +192,7 @@ config ICS932S401 config ATMEL_SSC tristate "Device driver for Atmel SSC peripheral" - depends on AVR32 || ARCH_AT91 + depends on HAS_IOMEM ---help--- This option enables device driver support for Atmel Synchronized Serial Communication peripheral (SSC). diff --git a/drivers/misc/atmel-ssc.c b/drivers/misc/atmel-ssc.c index 158da5a..0cee274 100644 --- a/drivers/misc/atmel-ssc.c +++ b/drivers/misc/atmel-ssc.c @@ -167,7 +167,7 @@ static int ssc_probe(struct platform_device *pdev) /* disable all interrupts */ clk_enable(ssc->clk); - ssc_writel(ssc->regs, IDR, ~0UL); + ssc_writel(ssc->regs, IDR, -1); ssc_readl(ssc->regs, SR); clk_disable(ssc->clk); -- 1.8.0

4 3

[alsa-devel] [PATCH v2] ASoC: lm49453: Update lm49453_reg_defs values as per LM49453 HW revision-B
by MR.Swami.Reddy＠ti.com 24 Dec '12

24 Dec '12

Update lm49453_reg_defs values as per LM49453 HW revision-B Signed-off-by: M R Swami Reddy <mr.swami.reddy(a)ti.com> --- sound/soc/codecs/lm49453.c | 61 +++++++++++--------------------------------- 1 files changed, 15 insertions(+), 46 deletions(-) diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index 99b0a9d..808e50a 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -111,9 +111,9 @@ static struct reg_default lm49453_reg_defs[] = { { 101, 0x00 }, { 102, 0x00 }, { 103, 0x01 }, - { 105, 0x01 }, - { 106, 0x00 }, - { 107, 0x01 }, + { 104, 0x01 }, + { 105, 0x00 }, + { 106, 0x01 }, { 107, 0x00 }, { 108, 0x00 }, { 109, 0x00 }, @@ -163,56 +163,25 @@ static struct reg_default lm49453_reg_defs[] = { { 184, 0x00 }, { 185, 0x00 }, { 186, 0x00 }, - { 189, 0x00 }, + { 187, 0x00 }, { 188, 0x00 }, - { 194, 0x00 }, - { 195, 0x00 }, - { 196, 0x00 }, - { 197, 0x00 }, - { 200, 0x00 }, - { 201, 0x00 }, - { 202, 0x00 }, - { 203, 0x00 }, - { 204, 0x00 }, - { 205, 0x00 }, - { 208, 0x00 }, + { 189, 0x00 }, + { 208, 0x06 }, { 209, 0x00 }, - { 210, 0x00 }, - { 211, 0x00 }, - { 213, 0x00 }, - { 214, 0x00 }, - { 215, 0x00 }, - { 216, 0x00 }, - { 217, 0x00 }, - { 218, 0x00 }, - { 219, 0x00 }, + { 210, 0x08 }, + { 211, 0x54 }, + { 212, 0x14 }, + { 213, 0x0d }, + { 214, 0x0d }, + { 215, 0x14 }, + { 216, 0x60 }, { 221, 0x00 }, { 222, 0x00 }, + { 223, 0x00 }, { 224, 0x00 }, - { 225, 0x00 }, - { 226, 0x00 }, - { 227, 0x00 }, - { 228, 0x00 }, - { 229, 0x00 }, - { 230, 0x13 }, - { 231, 0x00 }, - { 232, 0x80 }, - { 233, 0x0C }, - { 234, 0xDD }, - { 235, 0x00 }, - { 236, 0x04 }, - { 237, 0x00 }, - { 238, 0x00 }, - { 239, 0x00 }, - { 240, 0x00 }, - { 241, 0x00 }, - { 242, 0x00 }, - { 243, 0x00 }, - { 244, 0x00 }, - { 245, 0x00 }, { 248, 0x00 }, { 249, 0x00 }, - { 254, 0x00 }, + { 250, 0x00 }, { 255, 0x00 }, };

2 1

[alsa-devel] [PATCH v2] ASoC: lm49453: Fix adc, mic and sidetone volume ranges
by MR.Swami.Reddy＠ti.com 24 Dec '12

24 Dec '12

Add adc, mic, sidetone volume ranges and appropriately added the controls. Fix the DAC HP/EP/LS/LO/HA maximum gain values. Signed-off-by: MR Swami Reddy <mr.swami.reddy(a)ti.com> -- sound/soc/codecs/lm49453.c | 43 ++++++++++++++++++++++++------------------- 1 files changed, 24 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index 808e50a..379fcdf 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -494,36 +494,41 @@ SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_PORT2_TX2_REG, 7, 1, 0), }; /* TLV Declarations */ -static const DECLARE_TLV_DB_SCALE(digital_tlv, -7650, 150, 1); -static const DECLARE_TLV_DB_SCALE(port_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(adc_dac_tlv, -7650, 150, 1); +static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 200, 1); +static const DECLARE_TLV_DB_SCALE(port_tlv, -1800, 600, 0); +static const DECLARE_TLV_DB_SCALE(stn_tlv, -7200, 150, 0); static const struct snd_kcontrol_new lm49453_sidetone_mixer_controls[] = { /* Sidetone supports mono only */ SOC_DAPM_SINGLE_TLV("Sidetone ADCL Volume", LM49453_P0_STN_VOL_ADCL_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), SOC_DAPM_SINGLE_TLV("Sidetone ADCR Volume", LM49453_P0_STN_VOL_ADCR_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), SOC_DAPM_SINGLE_TLV("Sidetone DMIC1L Volume", LM49453_P0_STN_VOL_DMIC1L_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), SOC_DAPM_SINGLE_TLV("Sidetone DMIC1R Volume", LM49453_P0_STN_VOL_DMIC1R_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), SOC_DAPM_SINGLE_TLV("Sidetone DMIC2L Volume", LM49453_P0_STN_VOL_DMIC2L_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), SOC_DAPM_SINGLE_TLV("Sidetone DMIC2R Volume", LM49453_P0_STN_VOL_DMIC2R_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), }; static const struct snd_kcontrol_new lm49453_snd_controls[] = { /* mic1 and mic2 supports mono only */ - SOC_SINGLE_TLV("Mic1 Volume", LM49453_P0_ADC_LEVELL_REG, 0, 6, - 0, digital_tlv), - SOC_SINGLE_TLV("Mic2 Volume", LM49453_P0_ADC_LEVELR_REG, 0, 6, - 0, digital_tlv), + SOC_SINGLE_TLV("Mic1 Volume", LM49453_P0_MICL_REG, 0, 15, 0, mic_tlv), + SOC_SINGLE_TLV("Mic2 Volume", LM49453_P0_MICR_REG, 0, 15, 0, mic_tlv), + + SOC_SINGLE_TLV("ADCL Volume", LM49453_P0_ADC_LEVELL_REG, 0, 63, + 0, adc_dac_tlv), + SOC_SINGLE_TLV("ADCR Volume", LM49453_P0_ADC_LEVELR_REG, 0, 63, + 0, adc_dac_tlv), SOC_DOUBLE_R_TLV("DMIC1 Volume", LM49453_P0_DMIC1_LEVELL_REG, - LM49453_P0_DMIC1_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DMIC1_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_DOUBLE_R_TLV("DMIC2 Volume", LM49453_P0_DMIC2_LEVELL_REG, - LM49453_P0_DMIC2_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DMIC2_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_DAPM_ENUM("Mic2Mode", lm49453_mic2mode_enum), SOC_DAPM_ENUM("DMIC12 SRC", lm49453_dmic12_cfg_enum), @@ -538,16 +543,16 @@ static const struct snd_kcontrol_new lm49453_snd_controls[] = { 2, 1, 0), SOC_DOUBLE_R_TLV("DAC HP Volume", LM49453_P0_DAC_HP_LEVELL_REG, - LM49453_P0_DAC_HP_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DAC_HP_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_DOUBLE_R_TLV("DAC LO Volume", LM49453_P0_DAC_LO_LEVELL_REG, - LM49453_P0_DAC_LO_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DAC_LO_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_DOUBLE_R_TLV("DAC LS Volume", LM49453_P0_DAC_LS_LEVELL_REG, - LM49453_P0_DAC_LS_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DAC_LS_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_DOUBLE_R_TLV("DAC HA Volume", LM49453_P0_DAC_HA_LEVELL_REG, - LM49453_P0_DAC_HA_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DAC_HA_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_SINGLE_TLV("EP Volume", LM49453_P0_DAC_LS_LEVELL_REG, - 0, 6, 0, digital_tlv), + 0, 63, 0, adc_dac_tlv), SOC_SINGLE_TLV("PORT1_1_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG, 0, 3, 0, port_tlv),

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[alsa-devel] [PATCH] ASoC: wm8962: Add device tree support
by Fabio Estevam 24 Dec '12

24 Dec '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> Add device tree support. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Documentation/devicetree/bindings/sound/wm8962.txt | 16 ++++++++++++++++ sound/soc/codecs/wm8962.c | 7 +++++++ 2 files changed, 23 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/wm8962.txt diff --git a/Documentation/devicetree/bindings/sound/wm8962.txt b/Documentation/devicetree/bindings/sound/wm8962.txt new file mode 100644 index 0000000..dceb3b1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8962.txt @@ -0,0 +1,16 @@ +WM8962 audio CODEC + +This device supports I2C only. + +Required properties: + + - compatible : "wlf,wm8962" + + - reg : the I2C address of the device. + +Example: + +codec: wm8962@1a { + compatible = "wlf,wm8962"; + reg = <0x1a>; +}; diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 8fd38cb..1ebeff4 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3758,10 +3758,17 @@ static const struct i2c_device_id wm8962_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, wm8962_i2c_id); +static const struct of_device_id wm8962_of_match[] = { + { .compatible = "wlf,wm8962", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8962_of_match); + static struct i2c_driver wm8962_i2c_driver = { .driver = { .name = "wm8962", .owner = THIS_MODULE, + .of_match_table = wm8962_of_match, .pm = &wm8962_pm, }, .probe = wm8962_i2c_probe, -- 1.7.9.5

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Re: [alsa-devel] [PATCH] ASoC: core: giving WARN when device starting from non-off bias with idle_bias_off
by Mark Brown 24 Dec '12

24 Dec '12

On Fri, Dec 21, 2012 at 06:17:12PM +0800, Chuansheng Liu wrote: > > Just found some cases that some codec drivers set the bias to _STANDBY and > set idle_bias_off to 1 during probing. > It will cause unpaired runtime_get_sync/put() issue. Also as Mark suggested, > there is no reason to start from _STANDBY bias with idle_bias_off == 1. Applied, thanks.

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[alsa-devel] [PATCH RFT] ASoC: lm49453: Fix mask for setting mode bit in lm49453_set_dai_fmt()
by Axel Lin 24 Dec '12

24 Dec '12

The mode variable is either 0 or 1. To update mode setting, the mask should be BIT(0) rather than BIT(1). Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- Hi M R Swami, I don't have the datasheet. Can you check if this patch is correct? Thanks, Axel sound/soc/codecs/lm49453.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index d75257d..c0d203b 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -1218,7 +1218,7 @@ static int lm49453_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) } snd_soc_update_bits(codec, LM49453_P0_AUDIO_PORT1_BASIC_REG, - LM49453_AUDIO_PORT1_BASIC_FMT_MASK|BIT(1)|BIT(5), + LM49453_AUDIO_PORT1_BASIC_FMT_MASK|BIT(0)|BIT(5), (aif_val | mode | clk_phase)); snd_soc_write(codec, LM49453_P0_AUDIO_PORT1_RX_MSB_REG, clk_shift); -- 1.7.9.5

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[alsa-devel] [PATCH v2] ASoC: sgtl5000: Fix maximum value for microphone gain
by Fabio Estevam 24 Dec '12

24 Dec '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> sgtl5000 microphone gain only has 2 bits of resolution, so maximum value is 3. >From Eric Nelson: "We also found that for the microphones we have here (commodity PC boom mics) a default value of 2 for the gain gives the best results." So change the default microphone gain as well. Signed-off-by: Eric Nelson <eric.nelson(a)boundarydevices.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - Add sgtl5000 into the Subject sound/soc/codecs/sgtl5000.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index cb1675c..92bbfec 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -401,7 +401,7 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { 5, 1, 0), SOC_SINGLE_TLV("Mic Volume", SGTL5000_CHIP_MIC_CTRL, - 0, 4, 0, mic_gain_tlv), + 0, 3, 0, mic_gain_tlv), }; /* mute the codec used by alsa core */ @@ -1344,7 +1344,7 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) SGTL5000_HP_ZCD_EN | SGTL5000_ADC_ZCD_EN); - snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 0); + snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 2); /* * disable DAP -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: Fix maximum value for microphone gain
by Fabio Estevam 24 Dec '12

24 Dec '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> sgtl5000 microphone gain only has 2 bits of resolution, so maximum value is 3. >From Eric Nelson: "We also found that for the microphones we have here (commodity PC boom mics) a default value of 2 for the gain gives the best results." So change the default microphone gain as well. Signed-off-by: Eric Nelson <eric.nelson(a)boundarydevices.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/sgtl5000.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index cb1675c..92bbfec 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -401,7 +401,7 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { 5, 1, 0), SOC_SINGLE_TLV("Mic Volume", SGTL5000_CHIP_MIC_CTRL, - 0, 4, 0, mic_gain_tlv), + 0, 3, 0, mic_gain_tlv), }; /* mute the codec used by alsa core */ @@ -1344,7 +1344,7 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) SGTL5000_HP_ZCD_EN | SGTL5000_ADC_ZCD_EN); - snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 0); + snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 2); /* * disable DAP -- 1.7.9.5

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[alsa-devel] Does it will effect the precision of DMA position?
by Zhang wei 24 Dec '12

24 Dec '12

Hello guys. In the function snd_pcm_update_hw_ptr0(): pos -= pos % runtime->min_align; after running it, the 'pos' maybe smaller than before.because there is need to aligned.so that means lost or drop some Audio data for ever ,is that true? The data playbacked is a bit of different from we send to? Best Regards zw

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[alsa-devel] [alsa-lib, dmix] Support for S24_3BE
by Vincent Vinel 24 Dec '12

24 Dec '12

Hello, The bugtracker seems to be down, so I'm posting here instead. I was trying to use software mixing with my Native Instruments USB soundcard (snd-usb-caiaq), but all attemps to use the dmix pcm ended up with the rather unhelpful error : ALSA lib pcm_dmix.c:1311:(_snd_pcm_dmix_open) Unsupported format When using the hw pcm, however, aplay was kind enough to display the list of available formats, which in my case was limited to S24_3BE. A quick check confirmed that dmix doesn't support that format, which happens to be the only format available for NI USB soundcards (as defined in snd-usb-caiaq). Is it worth writing a patch for 24bit big-endian support in dmix ? External soundcards like these ones are usually used for realtime effects processing and low-latency audio, things for which dmix proably isn't suited anyway. -- Vincent Vinel

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Re: [alsa-devel] [PATCH V5 2/2] ASoC: SAMSUNG: Add DT support for i2s
by Grant Likely 24 Dec '12

24 Dec '12

On Thu, 13 Dec 2012 16:12:53 +0530, Padmavathi Venna <padma.v(a)samsung.com> wrote: > Add support for device based discovery. > > Signed-off-by: Padmavathi Venna <padma.v(a)samsung.com> > --- > .../devicetree/bindings/sound/samsung-i2s.txt | 75 +++++++ > sound/soc/samsung/dma.c | 1 + > sound/soc/samsung/dma.h | 1 + > sound/soc/samsung/i2s.c | 233 ++++++++++++++++---- > 4 files changed, 269 insertions(+), 41 deletions(-) > create mode 100644 Documentation/devicetree/bindings/sound/samsung-i2s.txt > > diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.txt b/Documentation/devicetree/bindings/sound/samsung-i2s.txt > new file mode 100644 > index 0000000..bff47b4 > --- /dev/null > +++ b/Documentation/devicetree/bindings/sound/samsung-i2s.txt > @@ -0,0 +1,75 @@ > +* Samsung I2S controller > + > +Required SoC Specific Properties: > + > +- compatible : "samsung,samsung-i2s" Isn't that kind of redundant? :-) The format of the compatible strings should be "<vendor>,<part-number>-i2s". Please be specific about the part number that you're doing the binding for. For example; use "samsung,exynos4210-i2s" instead of "samsung,exynos-i2s". > +- reg: physical base address of the controller and length of memory mapped > + region. > + > +[PRELIMINARY: the dma channel allocation will change once there are > +official DMA bindings] > + > +- tx-dma-channel-secondary: The dma channel specifier for secondary tx > + operations. The format of the dma specifier depends on the dma > + controller. > +- tx-dma-channel: The dma channel specifier for tx operations. The format of > + the dma specifier depends on the dma controller. > +- rx-dma-channel: The dma channel specifier for rx operations. The format of > + the dma specifier depends on the dma controller. > + > +Optional SoC Specific Properties: > + > +- samsung,supports-6ch: If the I2S Primary sound source has 5.1 Channel > + support, this flag is enabled. > +- samsung,supports-rstclr: This flag should be set if I2S software reset bit > + control is required. When this flag is set I2S software reset bit will be > + enabled or disabled based on need. > +- samsung,supports-secdai:If I2S block has a secondary FIFO and internal DMA, > + then this flag is enabled. > +- samsung,idma-addr: Internal DMA register base address of the audio > + sub system(used in secondary sound source). > + > +Required Board Specific Properties: > + > +- gpios: The gpio specifier for data out,data in, LRCLK, CDCLK and SCLK > + interface lines. The format of the gpio specifier depends on the gpio > + controller. > + The syntax of samsung gpio specifier is > + <[phandle of the gpio controller node] > + [pin number within the gpio controller] > + [mux function] > + [flags and pull up/down] > + [drive strength]> > + > +Aliases: > + > +- All the I2S controller nodes are represented in the aliases node using > + the following format 'i2s{n}' where n is a unique number for the alias. Are the aliases actually being used now? If not, then drop this section. I don't see any references in this patch. Otherwise the binding looks good to me. g.

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[alsa-devel] sound: usb-audio: full capture/playback/spdif support for Digidesign Mbox 2
by Damien Zammit 23 Dec '12

23 Dec '12

Hi folks, I have put together full support for Digidesign Mbox 2, but it needs someone to help me merge it into alsa. I haven't seen the latest snd-usb-audio code since last year, it seems to have branched into multiple files. I have worked out how the Digidesign Mbox 2 needs to be configured, it supports SPDIF i/o simultaneously with analogue but can be toggled on and off by using different altsettings. Also the sample clock rate must be configured via a special method. Therefore I have provided for this in a device_setup parameter with 4 settings: 48Khz Spdif+analog = 0 44.1Khz Spdif+analog = 1 48Khz Analog only = 2 44.1Khz Analog only = 3 ###### audio /* Digidesign Mbox 2 workaround: * supports S24_3BE BIG ENDIAN in an unusual way */ if (chip->usb_id == USB_ID(0x0dba, 0x3000)) pcm_format = SNDRV_PCM_FORMAT_S24_3BE; static int mbox2_skip_setting_quirk(struct snd_usb_audio *chip, int iface, int altno); static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) { ..... ..... /* Digidesign Mbox 2: skip altsets incompatible with device_setup */ if (chip->usb_id == USB_ID(0x0dba, 0x3000) && mbox2_skip_setting_quirk(chip, iface_no, altno)) continue; ..... } #define MBOX2_SET_48K_SPDIF 0 #define MBOX2_SET_44K_SPDIF 1 #define MBOX2_SET_48K_ANALOG 2 #define MBOX2_SET_44K_ANALOG 3 static int mbox2_skip_setting_quirk(struct snd_usb_audio *chip, int iface, int altno) { u8 srate[3]; /* Reset ifaces 2-5 to 0 altsetting. */ usb_set_interface(chip->dev, iface, 0); if (device_setup[chip->index] == MBOX2_SET_48K_SPDIF || device_setup[chip->index] == MBOX2_SET_48K_ANALOG) { //setup 48k srate[0]=0x80; srate[1]=0xbb; srate[2]=0x00; /* Send the magic! */ snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), 0x01, 0x22, 0x0100, 0x0085, &srate, 0x0003, 1000); snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), 0x81, 0xa2, 0x0100, 0x0085, &srate, 0x0003, 1000); snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), 0x81, 0xa2, 0x0100, 0x0086, &srate, 0x0003, 1000); snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), 0x81, 0xa2, 0x0100, 0x0003, &srate, 0x0003, 1000); snd_printk(KERN_INFO "Digidesign Mbox 2: 24 Bit 48000Hz"); if (device_setup[chip->index] == MBOX2_SET_48K_SPDIF) { //setup spdif snd_printk(KERN_INFO ", SPDIF\n"); if (altno != 2) return 1; } else { //setup analog snd_printk(KERN_INFO ", ANALOGUE\n"); if (altno != 3) return 1; } } else { //setup 44k srate[0]=0x44; srate[1]=0xac; srate[2]=0x00; /* Send the magic! */ snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), 0x01, 0x22, 0x0100, 0x0085, &srate, 0x0003, 1000); snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), 0x81, 0xa2, 0x0100, 0x0085, &srate, 0x0003, 1000); snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), 0x81, 0xa2, 0x0100, 0x0086, &srate, 0x0003, 1000); snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), 0x81, 0xa2, 0x0100, 0x0003, &srate, 0x0003, 1000); snd_printk(KERN_INFO "Digidesign Mbox 2: 24 Bit 44100Hz"); if (device_setup[chip->index] == MBOX2_SET_44K_SPDIF || device_setup[chip->index] == MBOX2_SET_44K_ANALOG) { //setup spdif snd_printk(KERN_INFO ", SPDIF\n"); if (altno != 2) return 1; } else { //setup analog snd_printk(KERN_INFO ", ANALOGUE\n"); if (altno != 3) return 1; } } return 0; } #define MBOX2_FIRMWARE_SIZE 646 #define MBOX2_BOOT_LOADING 0x01 /* Hard coded into the device */ #define MBOX2_BOOT_READY 0x02 /* Hard coded into the device */ static int snd_usb_mbox2_boot_quirk(struct usb_device *dev) { struct usb_host_config *config = dev->actconfig; int err; u8 enablemagic[3]; u8 bootresponse; int fwsize; if ((fwsize = le16_to_cpu(get_cfg_desc(config)->wTotalLength)) == MBOX2_FIRMWARE_SIZE) { snd_printdd("Sending Digidesign Mbox 2 boot sequence...\n"); /* From USB Snoop, * * SetupPacket = RT RQ VHVL INDX SIZE * RT = 0xRT Request Type * RQ = 0xRQ Request * VHVL = 0xVLVH Value in reverse byte order * INDX = 0xDXIN Index in reverse byte order * SIZE = 0xZESI Size in reverse byte order * * Magic boot code setup packet: c0 85 01 00 00 00 12 00 * RQ RT VLVH DXIN ZESI * becomes 0x85, 0xc0, 0x0001, 0x0000, &RETURNDATA, 0x0012, TIMEOUT * for snd_usb_ctl_msg() */ mbox2_reboot: snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), 0x85, 0xc0, 0x0001, 0x0000, &bootresponse, 0x0012, 1000); if (bootresponse == MBOX2_BOOT_LOADING) { snd_printdd("device not ready, resending boot sequence...\n"); goto mbox2_reboot; } if (bootresponse == MBOX2_BOOT_READY) { snd_printdd("device initialised!\n"); err = usb_get_descriptor(dev, USB_DT_DEVICE, 0, &dev->descriptor, sizeof(dev->descriptor)); config = dev->actconfig; if (err < 0) snd_printdd("error usb_get_descriptor: %d\n", err); err = usb_reset_configuration(dev); if (err < 0) snd_printdd("error usb_reset_configuration: %d\n", err); snd_printdd("mbox2_boot: new boot length = %d\n", le16_to_cpu(get_cfg_desc(config)->wTotalLength)); /* Successful boot, (should reset all altsettings to 0 like windows does, but works without it) * but now we will try sending capture/playback enable magic * * 80 bb 00 = 24bit mode - S24_3BE * 44 ac 00 = 16bit mode? * */ /* We want 48khz mode for now */ enablemagic[0]=0x80; enablemagic[1]=0xbb; enablemagic[2]=0x00; /* Send the magic! */ snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 0x01, 0x22, 0x0100, 0x0085, &enablemagic, 0x0003, 1000); snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 0x81, 0xa2, 0x0100, 0x0085, &enablemagic, 0x0003, 1000); snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 0x81, 0xa2, 0x0100, 0x0086, &enablemagic, 0x0003, 1000); snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 0x81, 0xa2, 0x0100, 0x0003, &enablemagic, 0x0003, 1000); return 0; /* Succesful boot */ } snd_printdd("unknown bootresponse, ignoring device: %d\n",bootresponse); return -ENODEV; } snd_printk("Invalid firmware size: %d\n",fwsize); return -ENODEV; } ####### quirk: { USB_DEVICE(0x0dba, 0x3000), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .vendor_name = "Digidesign", .product_name = "Mbox 2", .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 0, .type = QUIRK_IGNORE_INTERFACE }, { .ifnum = 1, .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = 2, .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = 3, .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = 4, .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = 5, .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = 6, .type = QUIRK_MIDI_MBOX2, .data = & (const struct snd_usb_midi_endpoint_info) { .out_ep = 0x02, .out_cables = 0x0001, .in_ep = 0x81, .in_interval = 0x01, .in_cables = 0x0001 } }, { .ifnum = -1 } } } }, ##### midi case QUIRK_MIDI_MBOX2: /* Digidesign Mbox 2 uses MIDIMAN MIDI protocol */ umidi->usb_protocol_ops = &snd_usbmidi_midiman_ops; /* * We have to make sure that the USB core looks * again at interface 6 by calling usb_set_interface() on it. */ usb_set_interface(umidi->chip->dev, 6, 0); memcpy(&endpoints[0], quirk->data, sizeof(struct snd_usb_midi_endpoint_info)); err = 0; break; Damien

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[alsa-devel] [PATCH] ALSA: usb-audio: support delay calculation on capture streams
by Pierre-Louis Bossart 21 Dec '12

21 Dec '12

Enable delay report on capture path. The delay is reset when an URB is retired and increment at each call to .pointer based on frame counter changes. The precision of the delay information is limited to 1ms as in the playback case. This reverts commit 3f94fad09538ec988919ec3f371841182df71d04. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/usb/pcm.c | 23 ++++++++++++++++++++--- 1 file changed, 20 insertions(+), 3 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index c659310..3a38447 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -59,7 +59,12 @@ snd_pcm_uframes_t snd_usb_pcm_delay(struct snd_usb_substream *subs, /* Approximation based on number of samples per USB frame (ms), some truncation for 44.1 but the estimate is good enough */ - est_delay = subs->last_delay - (frame_diff * rate / 1000); + est_delay = frame_diff * rate / 1000; + if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) + est_delay = subs->last_delay - est_delay; + else + est_delay = subs->last_delay + est_delay; + if (est_delay < 0) est_delay = 0; return est_delay; @@ -78,8 +83,7 @@ static snd_pcm_uframes_t snd_usb_pcm_pointer(struct snd_pcm_substream *substream return SNDRV_PCM_POS_XRUN; spin_lock(&subs->lock); hwptr_done = subs->hwptr_done; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - substream->runtime->delay = snd_usb_pcm_delay(subs, + substream->runtime->delay = snd_usb_pcm_delay(subs, substream->runtime->rate); spin_unlock(&subs->lock); return hwptr_done / (substream->runtime->frame_bits >> 3); @@ -1147,6 +1151,10 @@ static void retire_capture_urb(struct snd_usb_substream *subs, int i, period_elapsed = 0; unsigned long flags; unsigned char *cp; + int current_frame_number; + + /* read frame number here, update pointer in critical section */ + current_frame_number = usb_get_current_frame_number(subs->dev); stride = runtime->frame_bits >> 3; @@ -1180,6 +1188,15 @@ static void retire_capture_urb(struct snd_usb_substream *subs, subs->transfer_done -= runtime->period_size; period_elapsed = 1; } + /* capture delay is by construction limited to one URB, + * reset delays here + */ + runtime->delay = subs->last_delay = 0; + + /* realign last_frame_number */ + subs->last_frame_number = current_frame_number; + subs->last_frame_number &= 0xFF; /* keep 8 LSBs */ + spin_unlock_irqrestore(&subs->lock, flags); /* copy a data chunk */ if (oldptr + bytes > runtime->buffer_size * stride) { -- 1.7.9.5

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[alsa-devel] [PATCH RFT] ASoC: wm2200: Fix setting dai format in wm2200_set_fmt
by Axel Lin 21 Dec '12

21 Dec '12

According to the defines in wm2200.h: /* * R1284 (0x504) - Audio IF 1_5 */ #define WM2200_AIF1_FMT_MASK 0x0007 /* AIF1_FMT - [2:0] */ #define WM2200_AIF1_FMT_SHIFT 0 /* AIF1_FMT - [2:0] */ #define WM2200_AIF1_FMT_WIDTH 3 /* AIF1_FMT - [2:0] */ We should not left shift 1 bit for fmt_val when setting dai format. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- Hi Mark, wm2200.h defines WM2200_AIF1_FMT_SHIFT to be 0, so current code looks wrong to me. I don't have the datasheet, so can you check if this patch is correct? Axel sound/soc/codecs/wm2200.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index afcf31d..a12fc2f 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1626,7 +1626,7 @@ static int wm2200_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) WM2200_AIF1TX_LRCLK_MSTR | WM2200_AIF1TX_LRCLK_INV, lrclk); snd_soc_update_bits(codec, WM2200_AUDIO_IF_1_5, - WM2200_AIF1_FMT_MASK << 1, fmt_val << 1); + WM2200_AIF1_FMT_MASK, fmt_val); return 0; } -- 1.7.9.5

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[alsa-devel] [PATCH 0/3] ASoC: fsi: cleanup FSI driver
by Kuninori Morimoto 21 Dec '12

21 Dec '12

Hi Mark Cc Simon These patches changed platform setting to use FSI new style, and cleanup FSI driver. Kuninori Morimoto (3): ASoC: fsi: remove platform depended .set_rate() callback support ASoC: fsi: remove SH_FSI_xxx_INV flags ASoC: fsi: cleanup sh_fsi.h arch/arm/mach-shmobile/board-ap4evb.c | 11 +-- arch/arm/mach-shmobile/board-mackerel.c | 11 +-- arch/sh/boards/mach-ecovec24/setup.c | 12 +-- arch/sh/boards/mach-se/7724/setup.c | 12 +-- include/sound/sh_fsi.h | 70 +------------- sound/soc/sh/fsi.c | 156 +++---------------------------- 6 files changed, 27 insertions(+), 245 deletions(-) >> Simon #2 patch modifies both FSI driver and platform. So, it needs your Acked-by Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCHv2 0/9] ALSA: CA0132: DSP Update
by Ian Minett 21 Dec '12

21 Dec '12

From: Ian Minett <ian_minett(a)creativelabs.com> Thanks for the comments. This resubmit should help to correct many of the points raised. For example, use of C99-style init for structures, the previous large patch is now split into several subpatches, and there are more descriptions to the code and individual patchfiles. Patch Summary: 1: Add new definitions and structures that will be used by the DSP routines 2: Init chip registers, DSP parameters and mixer settings 3: Add the new DSP mixer kcontrols and helpers. 4: Update and add PCM callbacks and enhancements. 5: Add unsol handler for DSP response and jack detection reporting 6: Add tuning controls 7/8: Code shuffle and tidy, split into two for a cleaner diff. 9: Updates to the mic sample rate and module description, and fix potential errors on init. Thanks, - Ian Signed-off-by: Ian Minett <ian_minett(a)creativelabs.com>

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Re: [alsa-devel] [PATCH 5/7] ASoC: tegra: add ac97 host driver
by Stephen Warren 20 Dec '12

20 Dec '12

On 12/19/2012 04:17 PM, Lucas Stach wrote: > This adds the driver for the Tegra 2x AC97 host controller. The AC'97 DT binding file should really be added as part of this patch. I'm not at all familiar with AC'97, but this mostly looks fine. I have a bunch of comments below though, mostly for my own understanding. > diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig > +config SND_SOC_TEGRA20_AC97 > + tristate > + depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC This should also select SND_SOC_TEGRA20_DAS, just like the I2S driver. > diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c > +static struct tegra20_ac97 *workdata; Why not put that into the device's drvdata? Oh, is there no way to stash driver-private data into the struct snd_ac97? > +static void tegra20_ac97_codec_reset(struct snd_ac97 *ac97) > +{ > + u32 readback; > + unsigned long timeout; > + > + /* reset line is not driven by DAC pad group, have to toggle GPIO */ > + gpio_set_value(workdata->reset_gpio, 0); > + udelay(2); I'm not quite sure what that implies. Might some (Tegra) HW designs use a dedicated signal from the AC'97 HW to reset the CODEC and others not? That would make the GPIO optional. Or, does the comment mean that Tegra doesn't ever have a dedicated CODEC reset signal, so we always have to use a GPIO? If so, this comment might be more appropriate inside probe() where I asked my related questions. > +static void tegra20_ac97_codec_warm_reset(struct snd_ac97 *ac97) > +{ > + u32 readback; > + unsigned long timeout; > + > + /* > + * although sync line is driven by the DAC pad group warm reset using > + * the controller cmd is not working, have to toggle sync line > + * manually. > + */ > + gpio_request(workdata->sync_gpio, "codec-sync"); Hmm. There's an AC'97 command to reset the CODEC and we don't implement it? Uggh. > +static const struct snd_soc_dai_ops tegra20_ac97_dai_ops = { > + .trigger = tegra20_ac97_trigger, > +}; No .set_fmt or .hw_params? Does the AC'97 ASoC core code handle that somehow? I guess that's what soc_ac97_ops is for? > +static struct snd_soc_dai_driver tegra20_ac97_dai = { > + .name = "tegra-ac97-pcm", > + .ac97_control = 1, > + .probe = tegra20_ac97_probe, > + .playback = { > + .stream_name = "PCM Playback", Out of curiosity, why "PCM Playback" not just "Playback"? > +static bool tegra20_ac97_volatile_reg(struct device *dev, unsigned int reg) > +{ > + switch (reg) { > + case TEGRA20_AC97_CMD: ... > + return true; CMD is volatile; did we put status bits in there? > +static int tegra20_ac97_platform_probe(struct platform_device *pdev) > + ac97->reset_gpio = of_get_named_gpio(pdev->dev.of_node, > + "nvidia,codec-reset-gpio", 0); > + if (gpio_is_valid(ac97->reset_gpio)) { > + ret = devm_gpio_request_one(&pdev->dev, ac97->reset_gpio, > + GPIOF_OUT_INIT_HIGH, "codec-reset"); Shouldn't this get the flags from the GPIO specifier, and deal with active-high/active-low resets, or is the polarity set by the AC'97 spec? > + if (ret) { > + dev_err(&pdev->dev, "could not get codec-reset GPIO\n"); > + goto err_clk_put; > + } > + } else { > + dev_err(&pdev->dev, "no codec-reset GPIO supplied\n"); > + goto err_clk_put; > + } Is a reset GPIO necessarily required? What if a board arranged to reset the CODEC during power-on-reset, and there was no software control over the reset? > + ac97->sync_gpio = of_get_named_gpio(pdev->dev.of_node, > + "nvidia,codec-sync-gpio", 0); > + if (!gpio_is_valid(ac97->sync_gpio)) { > + dev_err(&pdev->dev, "no codec-sync GPIO supplied\n"); > + goto err_clk_put; > + } I don't know what this is, so I'll ask if it's strictly required too. > + ret = snd_soc_register_dais(&pdev->dev, &tegra20_ac97_dai, 1); snd_soc_register_dai() would be marginally simpler. > + ret = clk_prepare_enable(ac97->clk_ac97); > + if (ret) { > + dev_err(&pdev->dev, "clk_enable failed: %d\n", ret); > + goto err_asoc_utils_fini; > + } Can you do runtime PM instead? See the I2S driver for an example. > +MODULE_LICENSE("GPL"); "GPL v2"

2 1

Re: [alsa-devel] [PATCH 5/7] ASoC: tegra: add ac97 host driver
by Stephen Warren 20 Dec '12

20 Dec '12

On 12/20/2012 01:10 PM, Lucas Stach wrote: > Am Donnerstag, den 20.12.2012, 12:44 -0700 schrieb Stephen Warren: >> On 12/19/2012 04:17 PM, Lucas Stach wrote: >>> This adds the driver for the Tegra 2x AC97 host controller. >>> +static void tegra20_ac97_codec_warm_reset(struct snd_ac97 *ac97) >>> +{ >>> + u32 readback; >>> + unsigned long timeout; >>> + >>> + /* >>> + * although sync line is driven by the DAC pad group warm reset using >>> + * the controller cmd is not working, have to toggle sync line >>> + * manually. >>> + */ >>> + gpio_request(workdata->sync_gpio, "codec-sync"); >> >> Hmm. There's an AC'97 command to reset the CODEC and we don't implement >> it? Uggh. > > As far as I could figure from the downstream register doc there is a > command implemented in the host controller that's supposed to do the > warm reset, but in my testing it didn't work. Sadly I have not received > any documentation about the host controller from NVIDIA up until now, so > I don't now if we could make this work somehow. Bug Submission ID#: > 196075 What system was that bug submitted into? It's not a regular NVIDIA internal bug number. I'll see who I can ask about this feature... >>> + ac97->sync_gpio = of_get_named_gpio(pdev->dev.of_node, >>> + "nvidia,codec-sync-gpio", 0); >>> + if (!gpio_is_valid(ac97->sync_gpio)) { >>> + dev_err(&pdev->dev, "no codec-sync GPIO supplied\n"); >>> + goto err_clk_put; >>> + } >> >> I don't know what this is, so I'll ask if it's strictly required too. > > Until we manage to get the host controller warm reset command working > this is fixed to be the _FS line of the DAP connected to the codec, so > strictly required. I'd like to investigate whether the warm reset command can be made to work before actually committing to a DT binding that requires this GPIO to be specified; the DT binding is supposed to be basically static.

1 0

Re: [alsa-devel] [PATCH 7/7] ASoC: tegra: add machine driver using wm9712 codec
by Stephen Warren 20 Dec '12

20 Dec '12

On 12/19/2012 04:17 PM, Lucas Stach wrote: > This adds a simple machine driver using the Wolfson wm9712 AC97 > codec. If the AC'97 code is going to be reworked to probe the CODEC rather than manually specify it in DT, then will this machine driver become generic for any AC'97 system, and hence you could rename the DT binding to nvidia,tegra-audio-ac97? > diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.txt > +- nvidia,audio-codec : The phandle of the WM9712 audio codec I wonder how you'll get that if you dynamically probe the AC'97 bus... > diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig > +config SND_SOC_TEGRA_WM9712 > + tristate "SoC Audio support for Tegra boards using a WM9712 codec" > + depends on SND_SOC_TEGRA I'd suggest making this depend on the Tegra AC97 module, or ARCH_TEGRA_2x_SOC instead, since it can only run on Tegra20. The AC'97 module has been removed on Tegra30 and replaced with an HD Audio controller instead. > + select SND_SOC_TEGRA20_AC97 if ARCH_TEGRA_2x_SOC That select (or a dependency as mentioned above) is always required. > + select SND_SOC_AC97_BUS Shouldn'y the AC'97 driver select that? > diff --git a/sound/soc/tegra/tegra_wm9712.c b/sound/soc/tegra/tegra_wm9712.c > +static int tegra_wm9712_init(struct snd_soc_pcm_runtime *rtd) > + snd_soc_dapm_force_enable_pin(dapm, "Mic Bias"); Hmm. I guess that potentially ties this driver to whichever CODECs actually expose a widget with that exact name... > +static int tegra_wm9712_remove(struct snd_soc_card *card) > +{ > + return 0; > +} Do you need to implement this at all if it does nothing? > +MODULE_LICENSE("GPL"); "GPL v2"

2 1

Re: [alsa-devel] [PATCH 6/7] ASoC: tegra: add ac97 host controller to device tree
by Stephen Warren 20 Dec '12

20 Dec '12

On 12/19/2012 04:17 PM, Lucas Stach wrote: > This adds the DT bindings and a default entry for the Tegra 2x > AC97 host controller. Everything here except the .dtsi file should be part of the previous patch. > diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.txt > +- nvidia,codec-sync-gpio : The Tegra GPIO controller's phandle and the number > + of the GPIO corresponding with the AC97 DAP _FS line Hmm, for I2S at least, FS (frame sync I assume) definitely is something that the audio controller itself drives. Are you absolutely sure that this should be represented as a GPIO here? > diff --git a/arch/arm/boot/dts/tegra20.dtsi b/arch/arm/boot/dts/tegra20.dtsi > + tegra_ac97: ac97@70002000 { There's only 1 node named ac97, so you can leave out the unit address since it's not required to generate a unique node name. In other words just "ac97".

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Re: [alsa-devel] [PATCH 1/7] ARM: tegra: add ac97 clock
by Stephen Warren 20 Dec '12

20 Dec '12

On 12/19/2012 04:17 PM, Lucas Stach wrote: > Add clock used by the AC97 host controller. This is going to conflict with Prashant's clock driver rework. Can you rebase it on top of that once it's checked in. That might be a couple weeks out though.

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Re: [alsa-devel] [PATCH 4/7] ASoC: tegra: add function to set ac97 rate
by Stephen Warren 20 Dec '12

20 Dec '12

On 12/19/2012 04:17 PM, Lucas Stach wrote: > AC97 uses a fixed rate, unrelated to the sample rate. Add a function to > make the setup more trivial. Acked-by: Stephen Warren <swarren(a)nvidia.com>

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Re: [alsa-devel] [PATCH 3/7] ASoC: tegra: setup DAP3<->DAC3 connection by default
by Stephen Warren 20 Dec '12

20 Dec '12

On 12/19/2012 04:17 PM, Lucas Stach wrote: > This connection is used by the AC97 controller. Acked-by: Stephen Warren <swarren(a)nvidia.com>

1 0

[alsa-devel] Fwd: Low volume with Realtek ALC 883.
by prasannatsmkumar 20 Dec '12

20 Dec '12

On Thu, Dec 20, 2012 at 11:07 PM, prasannatsmkumar <prasannatsmkumar(a)gmail.com> wrote: > Hi ALSA people, > > I am having a Realtek ALC 883 chip. The sound volume is very low > compared to what I hear in windows - I maintain same speaker volume > and play same audio. If I keep the volume as 10% in windows I have to > keep a volume of at least 25+% in Linux. Is there any way to improve > it? I have been seeing such a situation for past 4 or 5 years (from > Fedora 9 period till today). I have tried several distributions > (Fedora, Vector Linux, PC Linux OS, Ubuntu, Mint) and all have the > same behaviour. Currently I am running Ubuntu 11.10 with 3.5 kernel. > > $cat /proc/asound/version > Advanced Linux Sound Architecture Driver Version 1.0.25. > > Should I file a bug? If so in which component (kernel or pulseaudio)? > > Thanks in advance, > PrasannaKumar Resending this as this mail was bounced by alsa devel list. Please feel free to ask for information necessary to solve this issue. I have copied output of some commands that may be useful for understanding or digging this issue. dmesg | grep hda output: [ 30.648663] snd_hda_intel 0000:00:1b.0: >irq 43 for MSI/MSI-X [ 30.716741] hda_codec: ALC883: SKU not ready 0x411111f0 lspci -vvv output (only info related audio device is in mail all others are stripped): 00:1b.0 Audio device: Intel Corporation N10/ICH 7 Family High Definition Audio Controller (rev 01) Subsystem: Intel Corporation Device d605 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 43 Region 0: Memory at 501c0000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Capabilities: [60] MSI: Enable+ Count=1/1 Maskable- 64bit+ Address: 00000000fee0300c Data: 4189 Capabilities: [70] Express (v1) Root Complex Integrated Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s <64ns, L1 <1us ExtTag- RBE- FLReset- DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr+ TransPend- LnkCap: Port #0, Speed unknown, Width x0, ASPM unknown, Latency L0 <64ns, L1 <1us ClockPM- Surprise- LLActRep- BwNot- LnkCtl: ASPM Disabled; Disabled- Retrain- CommClk- ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed unknown, Width x0, TrErr- Train- SlotClk- DLActive- BWMgmt- ABWMgmt- Capabilities: [100 v1] Virtual Channel Caps: LPEVC=0 RefClk=100ns PATEntryBits=1 Arb: Fixed- WRR32- WRR64- WRR128- Ctrl: ArbSelect=Fixed Status: InProgress- VC0: Caps: PATOffset=00 MaxTimeSlots=1 RejSnoopTrans- Arb: Fixed- WRR32- WRR64- WRR128- TWRR128- WRR256- Ctrl: Enable+ ID=0 ArbSelect=Fixed TC/VC=01 Status: NegoPending- InProgress- VC1: Caps: PATOffset=00 MaxTimeSlots=1 RejSnoopTrans- Arb: Fixed- WRR32- WRR64- WRR128- TWRR128- WRR256- Ctrl: Enable+ ID=1 ArbSelect=Fixed TC/VC=80 Status: NegoPending- InProgress- Capabilities: [130 v1] Root Complex Link Desc: PortNumber=0f ComponentID=02 EltType=Config Link0: Desc: TargetPort=00 TargetComponent=02 AssocRCRB- LinkType=MemMapped LinkValid+ Addr: 00000000fed1c000 Kernel driver in use: snd_hda_intel Kernel modules: snd-hda-intel amixer output: Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum Playback channels: Mono Limits: Playback 0 - 31 Mono: Playback 13 [42%] [-27.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [0.00dB] [on] Front Right: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'PCM',0 Capabilities: pvolume penum Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [0.00dB] Front Right: Playback 255 [100%] [0.00dB] Simple mixer control 'Front',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [0.00dB] [on] Front Right: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'Front Mic',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [12.00dB] [off] Front Right: Playback 31 [100%] [12.00dB] [off] Simple mixer control 'Front Mic Boost',0 Capabilities: volume penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 3 [100%] [30.00dB] Front Right: 3 [100%] [30.00dB] Simple mixer control 'Surround',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [0.00dB] [on] Front Right: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'Center',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum Playback channels: Mono Limits: Playback 0 - 31 Mono: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'LFE',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum Playback channels: Mono Limits: Playback 0 - 31 Mono: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'Line',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [12.00dB] [off] Front Right: Playback 31 [100%] [12.00dB] [off] Simple mixer control 'Beep',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch penum Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 8 [26%] [0.00dB] [on] Front Right: Capture 8 [26%] [0.00dB] [on] Simple mixer control 'Capture',1 Capabilities: cvolume cswitch penum Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 0 [0%] [-12.00dB] [off] Front Right: Capture 0 [0%] [-12.00dB] [off] Simple mixer control '2ch-Upmix Master',0 Capabilities: volume penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 255 Front Left: 255 [100%] Front Right: 255 [100%] Simple mixer control 'Channel Mode',0 Capabilities: enum Items: '2ch' '4ch' '6ch' Item0: '2ch' Simple mixer control 'Input Source',0 Capabilities: cenum Items: 'Rear Mic' 'Front Mic' 'Line' Item0: 'Front Mic' Simple mixer control 'Input Source',1 Capabilities: cenum Items: 'Rear Mic' 'Front Mic' 'Line' Item0: 'Rear Mic' Simple mixer control 'Rear Mic',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Rear Mic Boost',0 Capabilities: volume penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 2 [67%] [20.00dB] Front Right: 2 [67%] [20.00dB] Thanks and Regards, PrasannaKumar

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[alsa-devel] Fwd: Low volume with Realtek ALC 883.
by prasannatsmkumar 20 Dec '12

20 Dec '12

Hi ALSA / pulseaudio people, Resending this as this mail was bounced by alsa devel list. I am having a Realtek ALC 883 chip. The sound volume is very low compared to what I hear in windows - I maintain same speaker volume and play same audio. If I keep the volume as 10% in windows I have to keep a volume of at least 25+% in Linux. Is there any way to improve it? I have been seeing such a situation for past 4 or 5 years (from Fedora 9 period till today). I have tried several distributions (Fedora, Vector Linux, PC Linux OS, Ubuntu, Mint) and all have the same behaviour. Currently I am running Ubuntu 11.10 with 3.5 kernel. $cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.25. Should I file a bug? If so in which component (kernel or pulseaudio)? Thanks in advance, PrasannaKumar

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[alsa-devel] [PATCH RFT] ASoC: sta529: Fix update register bits in sta529_set_dai_fmt
by Axel Lin 20 Dec '12

20 Dec '12

Both the mask and mode settings are wrong in current code. According to the datasheet: S2PCFG0 (0x0A) BIT[3:1] DATA_FORMAT serial interface protocol format: 000: left Justified 001: I2S (default) 010: right justified 100: PCM no delay 101: PCM delay 111: DSP Thus fixes the defines for LEFT_J_DATA_FORMAT, I2S_DATA_FORMAT, and RIGHT_J_DATA_FORMAT. Also adds define for DATA_FORMAT_MSK. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- Hi Rajeev, I don't have this hardware, can you test this patch? Thanks, Axel sound/soc/codecs/sta529.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c index ab355c4..40c07be 100644 --- a/sound/soc/codecs/sta529.c +++ b/sound/soc/codecs/sta529.c @@ -74,9 +74,10 @@ SNDRV_PCM_FMTBIT_S32_LE) #define S2PC_VALUE 0x98 #define CLOCK_OUT 0x60 -#define LEFT_J_DATA_FORMAT 0x10 -#define I2S_DATA_FORMAT 0x12 -#define RIGHT_J_DATA_FORMAT 0x14 +#define DATA_FORMAT_MSK 0x0E +#define LEFT_J_DATA_FORMAT 0x00 +#define I2S_DATA_FORMAT 0x02 +#define RIGHT_J_DATA_FORMAT 0x04 #define CODEC_MUTE_VAL 0x80 #define POWER_CNTLMSAK 0x40 @@ -289,7 +290,7 @@ static int sta529_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) return -EINVAL; } - snd_soc_update_bits(codec, STA529_S2PCFG0, 0x0D, mode); + snd_soc_update_bits(codec, STA529_S2PCFG0, DATA_FORMAT_MSK, mode); return 0; } -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: pcm: allow backend hardware to be freed in pause state
by Patrick Lai 20 Dec '12

20 Dec '12

When front-end PCM session is in paused state, back-end PCM session will be put in paused state as well if given front-end PCM session is the only client of given back-end. Then, application closes front-end PCM session, DPCM framework will not allow back-end enters HW_FREE state so back-end will never get shutdown completely. Signed-off-by: Patrick Lai <plai(a)codeaurora.org> --- sound/soc/soc-pcm.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index ef22d0b..d340644 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1240,6 +1240,7 @@ static int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) continue; -- 1.7.8.3 Sent by an employee of the Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum.

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[alsa-devel] [PATCH RFT] ASoC: arizona: Do proper shift for setting aif1 rate
by Axel Lin 20 Dec '12

20 Dec '12

ARIZONA_AIF1_RATE_MASK is 0x7800 /* AIF1_RATE - [14:11] */ Thus we need left shift ARIZONA_AIF1_RATE_SHIFT when setting aif1 rate. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/arizona.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index adf397b..38248a7 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -714,7 +714,8 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, snd_soc_update_bits(codec, ARIZONA_ASYNC_SAMPLE_RATE_1, ARIZONA_ASYNC_SAMPLE_RATE_MASK, sr_val); snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL, - ARIZONA_AIF1_RATE_MASK, 8); + ARIZONA_AIF1_RATE_MASK, + 8 << ARIZONA_AIF1_RATE_SHIFT); break; default: arizona_aif_err(dai, "Invalid clock %d\n", dai_priv->clk); -- 1.7.9.5

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Re: [alsa-devel] [PATCH] ASoC: dapm: Fix the unpaired runtime_get/put cases
by Mark Brown 20 Dec '12

20 Dec '12

On Thu, Dec 20, 2012 at 06:37:26AM +0000, Liu, Chuansheng wrote: > Meanwhile, is it useful to add one warning there for that case? > After all, in probing, set the bias to _STANDBY even idle_bias_off == 1, and calling get_runtime_sync(), it > will let the code more obscure. So giving a warning there to indicate the driver: > it is not suggested that in probing, set the bias to _STANDBY even idle_bias_off == 1. Probably, send a patch please. Like I say it is possible to start off in _STANDBY providing the driver grabs the runtime PM reference too but I can't think of any reason for doing that so the warning seems sensible.

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Re: [alsa-devel] [PATCH 2/7] ASoC: allow wolfson wm9712 codec to be instantiated using device tree
by Mark Brown 20 Dec '12

20 Dec '12

On Thu, Dec 20, 2012 at 12:17:31AM +0100, Lucas Stach wrote: > static struct platform_driver wm9712_codec_driver = { > .driver = { > .name = "wm9712-codec", > .owner = THIS_MODULE, > + .of_match_table = wm9712_of_match, > }, No, this is really not a sane way of handling AC'97 at all. They're not platform devices, there is no sane reason why they should appear in the device tree as such. Really we ought to be just hotplugging them, AC'97 is perfectly enumerable.

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[alsa-devel] Adding wm8962 support to mx6qsabresd
by Fabio Estevam 20 Dec '12

20 Dec '12

Hi, I am working on adding audio support to mx6qsabresd board, which contains a wm8962 codec. Here is the log with debug information turned on that I am getting so far: wm8962 0-001a: customer id 0 revision D wm8962 0-001a: read 25 => 8 wm8962 0-001a: read 25 => 8 wm8962 0-001a: read 25 => 8 wm8962 0-001a: read 25 => 8 wm8962 0-001a: read 26 => 8 wm8962 0-001a: read 26 => 8 wm8962 0-001a: read 26 => 8 wm8962 0-001a: read 26 => 8 wm8962 0-001a: read 22 => 9 wm8962 0-001a: read 22 => 9 wm8962 0-001a: read 22 => 9 wm8962 0-001a: read 22 => 9 wm8962 0-001a: read 22 => 9 wm8962 0-001a: read 22 => 9 wm8962 0-001a: read 3a => 0 wm8962 0-001a: read 3a => 0 wm8962 0-001a: read 39 => 0 wm8962 0-001a: read 39 => 0 wm8962 0-001a: read 64 => 0 wm8962 0-001a: read 64 => 0 wm8962 0-001a: read 64 => 0 wm8962 0-001a: read 64 => 0 wm8962 0-001a: read 64 => 0 wm8962 0-001a: read 64 => 0 wm8962 0-001a: read 65 => 0 wm8962 0-001a: read 65 => 0 wm8962 0-001a: read 65 => 0 wm8962 0-001a: read 65 => 0 wm8962 0-001a: read 65 => 0 wm8962 0-001a: read 65 => 0 wm8962 0-001a: read 64 => 0 wm8962 0-001a: read 64 => 0 wm8962 0-001a: read 65 => 0 wm8962 0-001a: read 65 => 0 wm8962 0-001a: read 69 => 0 wm8962 0-001a: read 69 => 0 wm8962 0-001a: read 69 => 0 wm8962 0-001a: read 69 => 0 wm8962 0-001a: read 69 => 0 wm8962 0-001a: read 69 => 0 wm8962 0-001a: read 6a => 0 wm8962 0-001a: read 6a => 0 wm8962 0-001a: read 6a => 0 wm8962 0-001a: read 6a => 0 wm8962 0-001a: read 6a => 0 wm8962 0-001a: read 6a => 0 wm8962 0-001a: read 69 => 0 wm8962 0-001a: read 69 => 0 wm8962 0-001a: read 6a => 0 wm8962 0-001a: read 6a => 0 wm8962 0-001a: read 200 => ffffffff wm8962 0-001a: read 201 => 0 wm8962 0-001a: read 202 => 0 wm8962 0-001a: read 203 => ffffffff wm8962 0-001a: read 204 => 8100 wm8962 0-001a: read 205 => 8100 wm8962 0-001a: DMIC not in use, disabling input: WM8962 Beep Generator as /devices/soc.0/2100000.aips-bus/21a0000.i2c/i2c-0/0-001a/input/input0 wm8962 0-001a: read 31 => ffffffff wm8962 0-001a: read 31 => ffffffff wm8962 0-001a: read 1a => 0 wm8962 0-001a: read 1a => 0 wm8962 0-001a: read 63 => 0 wm8962 0-001a: read 63 => 0 wm8962 0-001a: read 1a => 0 wm8962 0-001a: read 1a => 0 wm8962 0-001a: read 63 => 0 wm8962 0-001a: read 63 => 0 wm8962 0-001a: read 1a => 0 wm8962 0-001a: read 1a => 0 wm8962 0-001a: read 19 => 0 wm8962 0-001a: read 19 => 0 wm8962 0-001a: read 19 => 0 wm8962 0-001a: read 19 => 0 wm8962 0-001a: read 19 => 0 wm8962 0-001a: read 26 => 8 wm8962 0-001a: read 25 => 8 wm8962 0-001a: read 30 => ffffffff wm8962 0-001a: read 30 => ffffffff wm8962 0-001a: read 300 => 1c00 wm8962 0-001a: read 17 => 160 wm8962 0-001a: read 48 => 0 wm8962 0-001a: read 8 => ffffffff wm8962 0-001a: read 52 => 4 wm8962 0-001a: read 19 => 0 wm8962 0-001a: write a0 = 149 wm8962 0-001a: write a1 = 271 wm8962 0-001a: write a2 = 7 wm8962 0-001a: FLL configured for 24000000Hz->11289600Hz wm8962 0-001a: No audio clocks configured ... imx-wm8962 sound.14: wm8962 <-> 2028000.ssi mapping ok root@freescale /home$ ./aplay Front_Left.wav wm8962 0-001a: hw_params set BCLK 1411200Hz LRCLK 44100Hz wm8962 0-001a: wm8962->sysclk_rate = 11289600 wm8962 0-001a: wm8962->lrclk = 44100 wm8962 0-001a: Selected sysclk ratio 256 wm8962 0-001a: read 4 => ffffffff wm8962 0-001a: Failed to read DSPCLK: -1 wm8962 0-001a: read 3 => 100 wm8962 0-001a: write 3 = 100 wm8962 0-001a: ASoC: POST_PMU: HPOUTR PGA event failed: -5 wm8962 0-001a: read 29 => 100 wm8962 0-001a: write 29 = 100 wm8962 0-001a: ASoC: POST_PMU: SPKOUTR PGA event failed: -5 wm8962 0-001a: read 2 => 100 wm8962 0-001a: write 2 = 100 wm8962 0-001a: ASoC: POST_PMU: HPOUTL PGA event failed: -5 wm8962 0-001a: read 28 => 100 wm8962 0-001a: write 28 = 100 wm8962 0-001a: ASoC: POST_PMU: SPKOUTL PGA event failed: -5 wm8962 0-001a: read 42 => ffffffff wm8962 0-001a: Failed to read DCS status: -1 wm8962 0-001a: DC servo complete after 1ms Playing WAVE 'Front_Left.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono Any idea of why register 4 and 42 read fails? wm8962 0-001a: read 4 => ffffffff wm8962 0-001a: Failed to read DSPCLK: -1 wm8962 0-001a: read 42 => ffffffff wm8962 0-001a: Failed to read DCS status: -1 Any suggestion is appreciated. Thanks, Fabio Estevam

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[alsa-devel] how to check the CODEC registers vaule
by Zhang wei 20 Dec '12

20 Dec '12

Hello everyone,I am confused again : ( I read the wm8985.pdf and found the description: If the address is not recognised or the R/W bit is '1' when operating in write only mode,the WM8985 returns to the idle conditon and waits for a new start condition and valid address. Does it mean the wm8985 registers write only, can not be read? I found other kinds of codec datasheet same as it. Is it necessary to read back the value from the register ? Thanks a lot. Best Regards zw

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[alsa-devel] [PATCH 1/2] Revert "ALSA: usb-audio: ignore delay calculation for capture stream"
by Pierre-Louis Bossart 19 Dec '12

19 Dec '12

This reverts commit 3f94fad09538ec988919ec3f371841182df71d04. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/usb/pcm.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index c659310..b09b895 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -78,8 +78,7 @@ static snd_pcm_uframes_t snd_usb_pcm_pointer(struct snd_pcm_substream *substream return SNDRV_PCM_POS_XRUN; spin_lock(&subs->lock); hwptr_done = subs->hwptr_done; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - substream->runtime->delay = snd_usb_pcm_delay(subs, + substream->runtime->delay = snd_usb_pcm_delay(subs, substream->runtime->rate); spin_unlock(&subs->lock); return hwptr_done / (substream->runtime->frame_bits >> 3); -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: core: Fix SOC_DOUBLE_RANGE() macros
by Mark Brown 19 Dec '12

19 Dec '12

Although we've had macros defining double _RANGE controls for a while now they've not actually been backed up properly by the implementation, it's treated everything as mono. Fix that by implementing the handling in the stereo controls, ensuring that the mono controls don't mistakenly get treated as stereo. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 10 ++++++---- sound/soc/soc-core.c | 32 ++++++++++++++++++++++++++++++-- 2 files changed, 36 insertions(+), 6 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 4a27822..1e4ea78 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -58,8 +58,9 @@ .info = snd_soc_info_volsw_range, .get = snd_soc_get_volsw_range, \ .put = snd_soc_put_volsw_range, \ .private_value = (unsigned long)&(struct soc_mixer_control) \ - {.reg = xreg, .shift = xshift, .min = xmin,\ - .max = xmax, .platform_max = xmax, .invert = xinvert} } + {.reg = xreg, .rreg = xreg, .shift = xshift, \ + .rshift = xshift, .min = xmin, .max = xmax, \ + .platform_max = xmax, .invert = xinvert} } #define SOC_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ @@ -88,8 +89,9 @@ .info = snd_soc_info_volsw_range, \ .get = snd_soc_get_volsw_range, .put = snd_soc_put_volsw_range, \ .private_value = (unsigned long)&(struct soc_mixer_control) \ - {.reg = xreg, .shift = xshift, .min = xmin,\ - .max = xmax, .platform_max = xmax, .invert = xinvert} } + {.reg = xreg, .rreg = xreg, .shift = xshift, \ + .rshift = xshift, .min = xmin, .max = xmax, \ + .platform_max = xmax, .invert = xinvert} } #define SOC_DOUBLE(xname, reg, shift_left, shift_right, max, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b1738d5..d4732ac 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2922,7 +2922,7 @@ int snd_soc_info_volsw_range(struct snd_kcontrol *kcontrol, platform_max = mc->platform_max; uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 1; + uinfo->count = snd_soc_volsw_is_stereo(mc) ? 2 : 1; uinfo->value.integer.min = 0; uinfo->value.integer.max = platform_max - min; @@ -2946,12 +2946,14 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); unsigned int reg = mc->reg; + unsigned int rreg = mc->rreg; unsigned int shift = mc->shift; int min = mc->min; int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; unsigned int val, val_mask; + int ret; val = ((ucontrol->value.integer.value[0] + min) & mask); if (invert) @@ -2959,7 +2961,21 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, val_mask = mask << shift; val = val << shift; - return snd_soc_update_bits_locked(codec, reg, val_mask, val); + ret = snd_soc_update_bits_locked(codec, reg, val_mask, val); + if (ret != 0) + return ret; + + if (snd_soc_volsw_is_stereo(mc)) { + val = ((ucontrol->value.integer.value[1] + min) & mask); + if (invert) + val = max - val; + val_mask = mask << shift; + val = val << shift; + + ret = snd_soc_update_bits_locked(codec, rreg, val_mask, val); + } + + return ret; } EXPORT_SYMBOL_GPL(snd_soc_put_volsw_range); @@ -2979,11 +2995,13 @@ int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); unsigned int reg = mc->reg; + unsigned int rreg = mc->rreg; unsigned int shift = mc->shift; int min = mc->min; int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; + int ret; ucontrol->value.integer.value[0] = (snd_soc_read(codec, reg) >> shift) & mask; @@ -2993,6 +3011,16 @@ int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol, ucontrol->value.integer.value[0] = ucontrol->value.integer.value[0] - min; + if (snd_soc_volsw_is_stereo(mc)) { + ucontrol->value.integer.value[1] = + (snd_soc_read(codec, rreg) >> shift) & mask; + if (invert) + ucontrol->value.integer.value[1] = + max - ucontrol->value.integer.value[1]; + ucontrol->value.integer.value[1] = + ucontrol->value.integer.value[1] - min; + } + return 0; } EXPORT_SYMBOL_GPL(snd_soc_get_volsw_range); -- 1.7.10.4

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[alsa-devel] Newbie help with WM8962 and tinyalsa
by Randi Eckstein 19 Dec '12

19 Dec '12

If anyone has an explanation on how to use tinymix to play on a wm8962, I'd really appreciate it. Thanks, Randi Eckstein

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Re: [alsa-devel] [PATCH] ASoC: dapm: Fix the unpaired runtime_get/put cases
by Mark Brown 19 Dec '12

19 Dec '12

On Wed, Dec 19, 2012 at 06:36:37PM +0800, Chuansheng Liu wrote: > But some devices has been set to STANDY bias directly during device probing, > such as cs42l73_probe(): > cs42l73_set_bias_level(codec, SND_SOC_BIAS_STANDBY); > Then it will cause runtime_get() not be called but laterly runtime_put() will > be called. Also found some other uppaired cases. This is just a bug in the driver, if it's idle_bias_off then it really should be starting in _OFF or at the very least starting actually in _STANDBY (including taking the runtime reference) rather than partially in _STANDBY. > So here add new flag get_runtime, and the logic will be: > 1/ when device is from off to non-off bias, runtime_get() will be called if not yet; > 2/ When device is off bias, runtime_put() will be called if runtime_get() has > been called; This is really not a good idea at all, it's just adding new special cases and making the code more obscure.

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[alsa-devel] Speaker not working; muted?
by Martin Michlmayr 19 Dec '12

19 Dec '12

I've a problem with audio on my HP 2560p laptop (using snd_hda_intel): the speaker no longer works. However, I successfully get sound on a headphone. I've attached the output of alsa-info. It would be great if someone could take a look at let me know if this is a bug in the driver or wrong settings. Background: I used ALSA successfully in the past to play audio with the in-built speaker. At some point, I used aumix (which requires OSS modules) to turn the volume to zero. I believe I turned the volume back up again. I also rebooted my laptop at some point. I don't know if it's related to using aumix or something else, but at some point the in-built speaker stopped working... I can still listen to audio using a headphone. Another observation is that my laptop has a speaker mute button. When I reboot my laptop, and the BIOS or grub is running, I see that the mute button is activated. However, pressing the button doesn't change anything... when the Linux kernel loads, the mute button is still on, but when ALSA starts the button changes colour: it seems to me that ALSA thinks that mute is off while in reality it is on. -- Martin Michlmayr http://www.cyrius.com/

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[alsa-devel] [PATCH 1/2] Revert "ALSA: usb-audio: ignore delay calculation for capture stream"
by Pierre-Louis Bossart 19 Dec '12

19 Dec '12

This reverts commit 3f94fad09538ec988919ec3f371841182df71d04. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/usb/pcm.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index c659310..b09b895 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -78,8 +78,7 @@ static snd_pcm_uframes_t snd_usb_pcm_pointer(struct snd_pcm_substream *substream return SNDRV_PCM_POS_XRUN; spin_lock(&subs->lock); hwptr_done = subs->hwptr_done; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - substream->runtime->delay = snd_usb_pcm_delay(subs, + substream->runtime->delay = snd_usb_pcm_delay(subs, substream->runtime->rate); spin_unlock(&subs->lock); return hwptr_done / (substream->runtime->frame_bits >> 3); -- 1.7.9.5

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[alsa-devel] [PATCH 1/3] ASoC: arizona: Add noise gate hold time enumeration
by Mark Brown 18 Dec '12

18 Dec '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 9 +++++++++ sound/soc/codecs/arizona.h | 2 ++ 2 files changed, 11 insertions(+) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index a85a68f..89c0433 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -323,6 +323,15 @@ const struct soc_enum arizona_lhpf4_mode = arizona_lhpf_mode_text); EXPORT_SYMBOL_GPL(arizona_lhpf4_mode); +static const char *arizona_ng_hold_text[] = { + "30ms", "120ms", "250ms", "500ms", +}; + +const struct soc_enum arizona_ng_hold = + SOC_ENUM_SINGLE(ARIZONA_NOISE_GATE_CONTROL, ARIZONA_NGATE_HOLD_SHIFT, + 4, arizona_ng_hold_text); +EXPORT_SYMBOL_GPL(arizona_ng_hold); + int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index db0c8dd..68dfae5 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -176,6 +176,8 @@ extern const struct soc_enum arizona_lhpf2_mode; extern const struct soc_enum arizona_lhpf3_mode; extern const struct soc_enum arizona_lhpf4_mode; +extern const struct soc_enum arizona_ng_hold; + extern int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); -- 1.7.10.4

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[alsa-devel] [PATCH] ASoC: arizona: Correct FLL source definitions
by Mark Brown 18 Dec '12

18 Dec '12

The FLL source constants were numbered as a simple enumeration but were being used in the code as direct values to be written to the registers. Renumber the constants to reflect the usage. Reported-by: Ryo Tsutsui <Ryo.Tsutsui(a)wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.h | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 7f22b4f..68dfae5 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -34,15 +34,15 @@ #define ARIZONA_FLL_SRC_MCLK1 0 #define ARIZONA_FLL_SRC_MCLK2 1 -#define ARIZONA_FLL_SRC_SLIMCLK 2 -#define ARIZONA_FLL_SRC_FLL1 3 -#define ARIZONA_FLL_SRC_FLL2 4 -#define ARIZONA_FLL_SRC_AIF1BCLK 5 -#define ARIZONA_FLL_SRC_AIF2BCLK 6 -#define ARIZONA_FLL_SRC_AIF3BCLK 7 -#define ARIZONA_FLL_SRC_AIF1LRCLK 8 -#define ARIZONA_FLL_SRC_AIF2LRCLK 9 -#define ARIZONA_FLL_SRC_AIF3LRCLK 10 +#define ARIZONA_FLL_SRC_SLIMCLK 3 +#define ARIZONA_FLL_SRC_FLL1 4 +#define ARIZONA_FLL_SRC_FLL2 5 +#define ARIZONA_FLL_SRC_AIF1BCLK 8 +#define ARIZONA_FLL_SRC_AIF2BCLK 9 +#define ARIZONA_FLL_SRC_AIF3BCLK 10 +#define ARIZONA_FLL_SRC_AIF1LRCLK 12 +#define ARIZONA_FLL_SRC_AIF2LRCLK 13 +#define ARIZONA_FLL_SRC_AIF3LRCLK 14 #define ARIZONA_MIXER_VOL_MASK 0x00FE #define ARIZONA_MIXER_VOL_SHIFT 1 -- 1.7.10.4

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[alsa-devel] [PATCH 1/1] ALSA: hda - bug fix for invalid connection list of more Haswell HDMI codec pins
by mengdong.lin＠intel.com 18 Dec '12

18 Dec '12

From: Mengdong Lin <mengdong.lin(a)intel.com> Haswell HDMI codec may report invalid connection list entries on all three pins. But our previous patch only fixes the HDMI pin, not the DP pin, so audio playbak via DP can still fail. This patch adds fixup for Haswell to set proper connection list entries for the other 2 pins including the DP pin. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 42fdf14..73d298a 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1710,8 +1710,14 @@ static void intel_haswell_fixup_connect_list(struct hda_codec *codec, if (vendor_param == -1 || !(vendor_param & 0x02)) return; + snd_hda_override_conn_list(codec, 0x05, 3, list); + hdmi_read_pin_conn(codec, 1); + snd_hda_override_conn_list(codec, 0x06, 3, list); hdmi_read_pin_conn(codec, 1); + + snd_hda_override_conn_list(codec, 0x07, 3, list); + hdmi_read_pin_conn(codec, 1); } -- 1.7.9.5

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Re: [alsa-devel] [pulseaudio-discuss] Front mic not working properly
by Raymond Yau 18 Dec '12

18 Dec '12

>> > >> > I'm totally stuck with a sound problem I fail to debug on my own. >> > >> > My setup: >> > Asus P5Q Motherboard with Realtec ALC1200 soundchip. Stereo speakers plugged in to the rear output and headphone speaker/mic combo plugged in to the front jacks. System: Ubuntu 12.10 >> > Desired behaviour: Sound is played back on headphones AND rear speaker. Front mic should "just-work" Asus P5Q specification Realtek ALC1200, 8-channel High Definition Audio CODEC - Support Jack-Detection, Multi-streaming, and Front Panel Jack-Retasking Alt_playback is still not enabled to support multi streaming playback on realtec codecs >> > Problem 2: >> > This is the biggest problem. >> > As soon as I plug in the front mic, Inside gnome-alsamixer the volume for master, surround,center,lfe start to jump between on and off. Why is the Front mic jack controlling these values? Do you mean BIOS did not setup pin default correctly after asus-p5q model was removed? http://git.kernel.org/?p=linux/kernel/git/tiwai/sound.git;a=commitdiff;h=6f… > How can I debug this behaviour? Post the output of alsa-info.sh > I just want them to stay the same, if the mic is plugged in or not. >> > Inside the gnome volume settings, the output port starts to jump between "Analog Output" and "Analog Headphones" >> >> This look like problem in Jack detection of the headphone and Mic Jack of front audio panel >> >> alsamixer -c0 >> >> Does the playback switch change rapidly on and off in alsamixer ? >> >> Do your computer chassis have a hda front audio panel and you connect them to the front audio connector of the motherboard according to the user manual ? >> >> Does it help if you disabled the automute switch ? >> > Thanks a lot. Your input helped me to at least work around the problem. My chassis has a HDA and an AC97 connector. The HDA connector was plugged in and correctly configured in BIOS. > I now use the AC97 connector (and changed the BIOS settings) and everything is working properly. Checking the front audio panel specification of your computer chassis Setting front audio panel to AC97 in BIOS of Asus motherboard change the miscellaneous bit of pin default of front HP and Mic Jack （i.e. misc = NO_PRESENCE) The alsa driver won't enable the unsolicited event on these nodes > > Regarding your questions: Disabling automute didn't help. And yes, the playback switch toggled rapidly. http://git.kernel.org/?p=linux/kernel/git/torvalds/linux-2.6.git;a=commitdi… Try disable trigger sense by codec->no_trigger_sense = 1; > > One thing I didn't test: I mistakenly plugged the rear speaker into the front channel jack (plugging the green connector into a green jack seemed so obvious). I don't know is this is the same channel as the headphone connector on the front panel... > > In AC97 mode it doesn't matter if I use the front channel jack or the rear channel jack (both at the rear of the computer) to connect my main speakers.

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[alsa-devel] [PATCH 0/2] A few fixes for Reatek codecs
by Takashi Iwai 17 Dec '12

17 Dec '12

Hi, here is a patch series for fixing some obvious errors I've overseen for Realtek codecs. It'll be queued for 3.8-rc1. Takashi

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[alsa-devel] [sound:topic/hda-ca0132-dsp 4/11] sound/pci/hda/patch_ca0132.c:714 dspio_send() warn: always true condition '(res >= 0) => (0-u32max >= 0)'
by Dan Carpenter 17 Dec '12

17 Dec '12

Hi Ian, FYI, there are new smatch warnings show up in tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/hda-ca0132-dsp head: cdc83c59e59d57b250be46c4b9d31e3b2b5ae589 commit: 01ef7dbffb411d9d78d1150b268d9c757f9f2f93 [4/11] ALSA: hda - Update CA0132 codec to load DSP firmware binary + sound/pci/hda/patch_ca0132.c:714 dspio_send() warn: always true condition '(res >= 0) => (0-u32max >= 0)' sound/pci/hda/patch_ca0132.c:1781 dspxfr_one_seg() info: why not propagate 'status' from dsp_dma_stop() instead of (-5)? sound/pci/hda/patch_ca0132.c:1950 dspxfr_image() error: potential null dereference 'dma_engine'. (kzalloc returns null) sound/pci/hda/patch_ca0132.c:1950 dspxfr_image() error: we previously assumed 'dma_engine' could be null (see line 1857) git remote add sound git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git git remote update sound git checkout 01ef7dbffb411d9d78d1150b268d9c757f9f2f93 vim +714 sound/pci/hda/patch_ca0132.c 01ef7dbf Ian Minett 2012-09-20 698 VENDOR_CHIPIO_PLL_PMU_WRITE, 0xff); 01ef7dbf Ian Minett 2012-09-20 699 mutex_unlock(&spec->chipio_mutex); 01ef7dbf Ian Minett 2012-09-20 700 } 01ef7dbf Ian Minett 2012-09-20 701 01ef7dbf Ian Minett 2012-09-20 702 /* 01ef7dbf Ian Minett 2012-09-20 703 * CA0132 DSP IO stuffs 01ef7dbf Ian Minett 2012-09-20 704 */ 01ef7dbf Ian Minett 2012-09-20 705 static int dspio_send(struct hda_codec *codec, unsigned int reg, 01ef7dbf Ian Minett 2012-09-20 706 unsigned int data) 01ef7dbf Ian Minett 2012-09-20 707 { 01ef7dbf Ian Minett 2012-09-20 708 unsigned int res; 01ef7dbf Ian Minett 2012-09-20 709 int retry = 50; 01ef7dbf Ian Minett 2012-09-20 710 01ef7dbf Ian Minett 2012-09-20 711 /* send bits of data specified by reg to dsp */ 01ef7dbf Ian Minett 2012-09-20 712 do { 01ef7dbf Ian Minett 2012-09-20 713 res = snd_hda_codec_read(codec, WIDGET_DSP_CTRL, 0, reg, data); 01ef7dbf Ian Minett 2012-09-20 @714 if ((res >= 0) && (res != VENDOR_STATUS_DSPIO_BUSY)) 01ef7dbf Ian Minett 2012-09-20 715 return res; 01ef7dbf Ian Minett 2012-09-20 716 } while (--retry); 01ef7dbf Ian Minett 2012-09-20 717 01ef7dbf Ian Minett 2012-09-20 718 return -EIO; 01ef7dbf Ian Minett 2012-09-20 719 } 01ef7dbf Ian Minett 2012-09-20 720 01ef7dbf Ian Minett 2012-09-20 721 static void dspio_write_wait(struct hda_codec *codec) 01ef7dbf Ian Minett 2012-09-20 722 { --- 0-DAY kernel build testing backend Open Source Technology Center Fengguang Wu, Yuanhan Liu Intel Corporation

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[alsa-devel] The new alsa-driver repository
by Jaroslav Kysela 17 Dec '12

17 Dec '12

Hello all, the ALSA server has new alsa-driver repository which contains several branches for now: master - just empty one with README build - the previous master branch (but all was moved to the alsa/ subtree) mirror - mirrored code from upstream (ALSA parts only) release - merging branch for mirror/build trees to create a relationship between them All branches are ophan. It means that they have no common parent. I put some handy configuration hints to the ALSA WIKI: http://www.alsa-project.org/main/index.php/GIT_Server#Syncing_the_branch_re… The reason to move the old master branch to the alsa/ subtree is quite simple - you can merge/pull the branch to the full linux-2.6 kernel tree now and do your development there. Also, cherry picks between mirror and the linux-2.6 tree should work. Another handy extension - in the current repo - might be to create the linux-2.6 branch with the whole upstream and create another branch for merging the build branch with the whole tree like 'release26' or so. In this way, we can manage another releases of the out-of-kernel build code for the ALSA tree in stable linux kernel versions. Comments, other ideas are welcome. Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project; Red Hat, Inc.

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[alsa-devel] [PATCH] ALSA: hda - Fix clicking noise on Thinkpad T61/R61i
by David Henningsson 17 Dec '12

17 Dec '12

The bug reporter reports that setting the speaker pin to D3 before turning off its pinctl fixes the clicking noise on powersave for Thinkpad T61. Thanks to c4pp4 for doing most of the work with this bug, including reporting and testing, and writing preliminary patches. BugLink: https://bugs.launchpad.net/bugs/886975 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 36 ++++++++++++++++++++++++++++++++---- 1 file changed, 32 insertions(+), 4 deletions(-) Note: c4pp4 pointed out that in his case, only the speaker needed to go to D3 before pinctl, but noticed no regressions from the below implementation, so I made the fixup more generic (easier to reuse later). diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index a3a2263..27e6ca7 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -144,6 +144,7 @@ struct conexant_spec { unsigned int asus:1; unsigned int pin_eapd_ctrls:1; unsigned int fixup_stereo_dmic:1; + unsigned int d3_before_shutup:1; /* Needs speaker to go to D3 before pinctl */ unsigned int adc_switching:1; @@ -558,13 +559,22 @@ static int conexant_build_controls(struct hda_codec *codec) return 0; } -#ifdef CONFIG_PM static int conexant_suspend(struct hda_codec *codec) { + struct conexant_spec *spec = codec->spec; + + if (spec->d3_before_shutup) + snd_hda_codec_set_power_to_all(codec, codec->afg, AC_PWRST_D3, + false); snd_hda_shutup_pins(codec); return 0; } -#endif + +static void conexant_reboot_notify(struct hda_codec *codec) +{ + conexant_suspend(codec); +} + static const struct hda_codec_ops conexant_patch_ops = { .build_controls = conexant_build_controls, @@ -575,7 +585,7 @@ static const struct hda_codec_ops conexant_patch_ops = { #ifdef CONFIG_PM .suspend = conexant_suspend, #endif - .reboot_notify = snd_hda_shutup_pins, + .reboot_notify = conexant_reboot_notify, }; #ifdef CONFIG_SND_HDA_INPUT_BEEP @@ -4408,7 +4418,7 @@ static const struct hda_codec_ops cx_auto_patch_ops = { #ifdef CONFIG_PM .suspend = conexant_suspend, #endif - .reboot_notify = snd_hda_shutup_pins, + .reboot_notify = conexant_reboot_notify, }; /* @@ -4421,6 +4431,7 @@ enum { CXT_PINCFG_LEMOTE_A1205, CXT_FIXUP_STEREO_DMIC, CXT_FIXUP_INC_MIC_BOOST, + CXT_FIXUP_D3_BEFORE_SHUTUP, }; static void cxt_fixup_stereo_dmic(struct hda_codec *codec, @@ -4430,6 +4441,13 @@ static void cxt_fixup_stereo_dmic(struct hda_codec *codec, spec->fixup_stereo_dmic = 1; } +static void cxt_fixup_d3_before_shutup(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct conexant_spec *spec = codec->spec; + spec->d3_before_shutup = 1; +} + static void cxt5066_increase_mic_boost(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -4499,6 +4517,15 @@ static const struct hda_fixup cxt_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = cxt5066_increase_mic_boost, }, + [CXT_FIXUP_D3_BEFORE_SHUTUP] = { + .type = HDA_FIXUP_FUNC, + .v.func = cxt_fixup_d3_before_shutup, + }, +}; + +static const struct snd_pci_quirk cxt5045_fixups[] = { + SND_PCI_QUIRK(0x17aa, 0x20ac, "Lenovo T61/R61i", CXT_FIXUP_D3_BEFORE_SHUTUP), + {} }; static const struct snd_pci_quirk cxt5051_fixups[] = { @@ -4553,6 +4580,7 @@ static int patch_conexant_auto(struct hda_codec *codec) switch (codec->vendor_id) { case 0x14f15045: codec->single_adc_amp = 1; + snd_hda_pick_fixup(codec, NULL, cxt5045_fixups, cxt_fixups); break; case 0x14f15051: add_cx5051_fake_mutes(codec); -- 1.7.9.5

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[alsa-devel] ASoC updates for v3.8
by Mark Brown 17 Dec '12

17 Dec '12

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[alsa-devel] [PATCH 0/3] at91/ssc: fixes on ASoC tree for 3.8
by Nicolas Ferre 17 Dec '12

17 Dec '12

Hi Mark, This series goes on top of current Linus' git tree and fixes an error that we have while compiling DTBs for AT91: ERROR (phandle_references): Reference to non-existent node or label "pinctrl_ssc0_tx" ERROR: Input tree has errors, aborting (use -f to force output) make[3]: *** [arch/arm/boot/dts/at91sam9g20ek.dtb] Error 2 I have included pending material that is dealing with SSC and pinctrl. The pinctrl part can be merged more easily now that the big pinctrl update for AT91 has been merged upstream. I have also included a typo patch that you have already taken as 315656b (ASoC: atmel-ssc: change disable to disable in dts node) in your topic/atmel branch but that was not described properly. If you find it easier, I can provide you a git tree for this material: just tell me... Thanks for your help, best regards, Bo Shen (3): ARM: at91/dts: fix typo in SSC status property ARM: at91/dts: add pinctrl support for SSC peripheral ASoC: atmel-ssc: add pinctrl selection to driver arch/arm/boot/dts/at91sam9260.dtsi | 18 +++++++++++++++++ arch/arm/boot/dts/at91sam9263.dtsi | 40 ++++++++++++++++++++++++++++++++++++-- arch/arm/boot/dts/at91sam9g45.dtsi | 40 ++++++++++++++++++++++++++++++++++++-- arch/arm/boot/dts/at91sam9n12.dtsi | 26 +++++++++++++++++++++++++ arch/arm/boot/dts/at91sam9x5.dtsi | 32 +++++++++++++++++++++++------- drivers/misc/atmel-ssc.c | 8 ++++++++ 6 files changed, 153 insertions(+), 11 deletions(-) -- 1.8.0

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Re: [alsa-devel] [Resend PATCH v6 1/4] pinctrl: atmel-ssc: add pinctrl support
by Nicolas Ferre 17 Dec '12

17 Dec '12

On 12/17/2012 10:39 AM, Jean-Christophe PLAGNIOL-VILLARD : > On 11:24 Wed 21 Nov , Mark Brown wrote: >> On Wed, Nov 21, 2012 at 10:17:09AM +0800, Bo Shen wrote: >>> Add pinctrl support for atmel ssc peripheral >>> >>> Signed-off-by: Bo Shen <voice.shen(a)atmel.com> >> >> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> > > you handle this one with scc Hold on Jean-Christophe, I made a little experiment this morning, and this patch is not applying correctly on: - linus' tree - linux-next - neither Mark's for-next branch So, *I* will manage this patch and try to make sure that Mark is able to seamlessly include it in his tree on top of a known source for a future ASoC "fixes" pull-request. Mark, what about rebasing this one on top of current Linus' tree (toguether with the one that you have just applied on the topic/atmel branch: (ASoC: atmel-ssc: change disable to disable in dts node)? I hope that once reworked, they will be able to reach Linus' tree soon because the lack of this patch is causing DTB compilation errors on all AT91... Best regards, -- Nicolas Ferre

2 1

[alsa-devel] [PATCH] ASoC: atmel-ssc: change disable to disable in dts node
by Bo Shen 17 Dec '12

17 Dec '12

Change the value of status to disabled to keep the consistent Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- arch/arm/boot/dts/at91sam9260.dtsi | 2 +- arch/arm/boot/dts/at91sam9263.dtsi | 4 ++-- arch/arm/boot/dts/at91sam9g45.dtsi | 4 ++-- arch/arm/boot/dts/at91sam9x5.dtsi | 2 +- 4 files changed, 6 insertions(+), 6 deletions(-) diff --git a/arch/arm/boot/dts/at91sam9260.dtsi b/arch/arm/boot/dts/at91sam9260.dtsi index aaa42d8..03c56cb 100644 --- a/arch/arm/boot/dts/at91sam9260.dtsi +++ b/arch/arm/boot/dts/at91sam9260.dtsi @@ -217,7 +217,7 @@ compatible = "atmel,at91rm9200-ssc"; reg = <0xfffbc000 0x4000>; interrupts = <14 4 5>; - status = "disable"; + status = "disabled"; }; adc0: adc@fffe0000 { diff --git a/arch/arm/boot/dts/at91sam9263.dtsi b/arch/arm/boot/dts/at91sam9263.dtsi index 3b721ee..15f12dd 100644 --- a/arch/arm/boot/dts/at91sam9263.dtsi +++ b/arch/arm/boot/dts/at91sam9263.dtsi @@ -179,14 +179,14 @@ compatible = "atmel,at91rm9200-ssc"; reg = <0xfff98000 0x4000>; interrupts = <16 4 5>; - status = "disable"; + status = "disabled"; }; ssc1: ssc@fff9c000 { compatible = "atmel,at91rm9200-ssc"; reg = <0xfff9c000 0x4000>; interrupts = <17 4 5>; - status = "disable"; + status = "disabled"; }; macb0: ethernet@fffbc000 { diff --git a/arch/arm/boot/dts/at91sam9g45.dtsi b/arch/arm/boot/dts/at91sam9g45.dtsi index acfa207..44a38d0 100644 --- a/arch/arm/boot/dts/at91sam9g45.dtsi +++ b/arch/arm/boot/dts/at91sam9g45.dtsi @@ -232,14 +232,14 @@ compatible = "atmel,at91sam9g45-ssc"; reg = <0xfff9c000 0x4000>; interrupts = <16 4 5>; - status = "disable"; + status = "disabled"; }; ssc1: ssc@fffa0000 { compatible = "atmel,at91sam9g45-ssc"; reg = <0xfffa0000 0x4000>; interrupts = <17 4 5>; - status = "disable"; + status = "disabled"; }; adc0: adc@fffb0000 { diff --git a/arch/arm/boot/dts/at91sam9x5.dtsi b/arch/arm/boot/dts/at91sam9x5.dtsi index 69667d0..0beff72 100644 --- a/arch/arm/boot/dts/at91sam9x5.dtsi +++ b/arch/arm/boot/dts/at91sam9x5.dtsi @@ -92,7 +92,7 @@ compatible = "atmel,at91sam9g45-ssc"; reg = <0xf0010000 0x4000>; interrupts = <28 4 5>; - status = "disable"; + status = "disabled"; }; tcb0: timer@f8008000 { -- 1.7.9.5

3 2

[alsa-devel] "hda-intel: azx_get_response timeout"
by Mantas M. 17 Dec '12

17 Dec '12

After upgrading to Linux 3.7 (on Arch Linux, x86_64), I've been getting the following messages in dmesg: > [Dec11 21:52] hda-intel: azx_get_response timeout, switching to polling mode: last cmd=0x001f0500 > [ +1.006709] hda-intel: No response from codec, disabling MSI: last cmd=0x001f0500 > [ +1.006773] hda_intel: azx_get_response timeout, switching to single_cmd mode: last cmd=0x001f0500 This seems to happen when resuming from suspend-to-memory, and is new to v3.7 – I haven't seen any such messages on any kernels I've used earlier (between v3.0 and v3.6.9). The device is: > # lspci -v > 00:1b.0 Audio device: Intel Corporation 5 Series/3400 Series Chipset High Definition Audio (rev 06) > Subsystem: ASUSTeK Computer Inc. Device 1643 > Flags: bus master, fast devsel, latency 0, IRQ 22 > Memory at d5200000 (64-bit, non-prefetchable) [size=16K] > Capabilities: [50] Power Management version 2 > Capabilities: [60] MSI: Enable- Count=1/1 Maskable- 64bit+ > Capabilities: [70] Express Root Complex Integrated Endpoint, MSI 00 > Capabilities: [100] Virtual Channel > Capabilities: [130] Root Complex Link > Kernel driver in use: snd_hda_intel Is there anything I can/should do about this? -- Mantas Mikulėnas <grawity(a)gmail.com>

2 1

Re: [alsa-devel] Could you help me about the ALSA?
by Zhang wei 17 Dec '12

17 Dec '12

> Yes,when buffer_size wraped the 'pos' is smallar than last one.If we > want to confirm the hw_base need to adjust there must be consider > another confidition: > > hdelta = jiffies - runtime->hw_ptr_jiffies; > > if (hdelta > runtime->hw_ptr_buffer_jiffies/2) > > how to connect this code with checking unnormal interrupt(lost or delayed)? This condition is for the double interrupt acknowledge problems (see comment). It makes sure, that the DMA buffer position was wrapped using another external timing source. Indeed, it's mostly a lowlevel driver fault to call this pointer update routine with the in_interrupt flag when the period was not really elapsed. -------------------------------------- I am sorry. I forget signed my info and this mail sent by me. Yes,I saw the comment and known the target of the pices of code and known alsa use jiffies as another timing source to detect the wrap situation,but I do not understand how to detect or check that case.In other word,why does alsa use the value runtime->hw_ptr_buffer_jiffies/2 not runtime->hw_ptr_buffer_jiffies/3 or runtime->hw_ptr_buffer_jiffies/4 ...? Best Regards zw

1 0

Re: [alsa-devel] Could you help me about the ALSA?
by Jaroslav Kysela 17 Dec '12

17 Dec '12

Date 14.12.2012 09:07, Zhang wei wrote: > Good afternoon . > > Excuse me if I disturb you, I found your contact information from the > alsa-devel > http://mailman.alsa-project.org/pipermail/alsa-devel/2010-August/030605.html > . > > Recently I am studying the alsa-driver and the code blocked me which > from the function snd_pcm_update_hw_ptr0inthe pcm_lib.c. > > =====================start==============================è > > if (in_interrupt) { > > /* we know that one period was processed */ > > /* delta = "expected next hw_ptr" for in_interrupt != > 0 */ > > delta = runtime->hw_ptr_interrupt + runtime->period_size; > > if (delta > new_hw_ptr) { > > /* check for double acknowledged interrupts */ > > hdelta = jiffies - runtime->hw_ptr_jiffies; > > if (hdelta > runtime->hw_ptr_buffer_jiffies/2) { > > hw_base += runtime->buffer_size; > > if (hw_base >= runtime->boundary) > > hw_base = 0; > > new_hw_ptr = hw_base + pos; > > goto __delta; > > } > > } > > } > > ç================end================================= > > I’ve already read your mail but I still do not understand your intend. I > know the target of the pices of code is prevent the variable hw_ptr and > hw_base delayed update from some kind of case .so my question is > > (1) how could happen on delta>new_hw_ptr? And why? > > (2) Why we can conclue the hw_ptr crossed the boundary of > buffer_size when hdelta>runitme->hw_ptr_jiffies/2? What principle we > based on? The interrupts can be lost or delayed in some cases (small period sizes etc.). So you need to check for these situations. The driver returns only position in the ring buffer without count of "cross" points, so we need to trace these situations using another clock source (like system clock). Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project; Red Hat, Inc.

3 6

[alsa-devel] [PATCH] ASoC: cs42l52: Fix off-by-one boundary checking for cs42l52_get_clk()
by Axel Lin 16 Dec '12

16 Dec '12

The caller uses the return value of cs42l52_get_clk as array index, the valid array index is 0 .. ARRAY_SIZE(clk_map_table) - 1. Thus we need to ensure ret is less than ARRAY_SIZE(clk_map_table). Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs42l52.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 99bb1c6..66817f6 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -749,7 +749,7 @@ static int cs42l52_get_clk(int mclk, int rate) } } } - if (ret > ARRAY_SIZE(clk_map_table)) + if (ret >= ARRAY_SIZE(clk_map_table)) return -EINVAL; return ret; } -- 1.7.9.5

1 1

[alsa-devel] [PATCH] ASoC: pcm: allow backend hardware to be freed in pause state
by Patrick Lai 15 Dec '12

15 Dec '12

When front-end PCM session is in paused state, back-end PCM session will be put in paused state as well if given front-end PCM session is the only client of given back-end. Then, application closes front-end PCM session, DPCM framework will not allow back-end enters HW_FREE state so back-end will never get shutdown completely. Signed-off-by: Patrick Lai <plai(a)codeaurora.org> --- sound/soc/soc-pcm.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index ef22d0b..d340644 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1240,7 +1240,7 @@ static int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) continue; -- The Qualcomm Innovation Center, Inc. is a member of Code Aurora Forum, hosted by The Linux Foundation 1.7.8.3

2 1

[alsa-devel] [PATCH] ASoC: Prevent pop_wait overwrite
by Misael Lopez Cruz 15 Dec '12

15 Dec '12

pop_wait is used to determine if a deferred playback close needs to be cancelled when the a PCM is open or if after the power-down delay expires it needs to run. pop_wait is associated with the CODEC DAI, so the CODEC DAI must be unique. This holds true for most CODECs, except for the dummy CODEC and its DAI. In DAI links with non-unique dummy CODECs (e.g. front-ends), pop_wait can be overwritten by another DAI link using also a dummy CODEC. Failure to cancel a deferred close can cause mute due to the DAPM STOP event sent in the deferred work. One scenario where pop_wait is overwritten and causing mute is below (where hw:0,0 and hw:0,1 are two front-ends with default pmdown_time = 5 secs): aplay /dev/urandom -D hw:0,0 -c 2 -r 48000 -f S16_LE -d 1 sleep 1 aplay /dev/urandom -D hw:0,1 -c 2 -r 48000 -f S16_LE -d 3 & aplay /dev/urandom -D hw:0,0 -c 2 -r 48000 -f S16_LE Since CODECs may not be unique, pop_wait is moved to the PCM runtime structure. Creating separate dummy CODECs for each DAI link can also solve the problem, but at this point it's only pop_wait variable in the CODEC DAI that has negative effects by not being unique. Signed-off-by: Misael Lopez Cruz <misael.lopez(a)ti.com> --- include/sound/soc-dai.h | 1 - include/sound/soc.h | 1 + sound/soc/soc-compress.c | 2 +- sound/soc/soc-pcm.c | 12 ++++++------ 4 files changed, 8 insertions(+), 8 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 628db7b..3953cea 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -242,7 +242,6 @@ struct snd_soc_dai { unsigned int symmetric_rates:1; struct snd_pcm_runtime *runtime; unsigned int active; - unsigned char pop_wait:1; unsigned char probed:1; struct snd_soc_dapm_widget *playback_widget; diff --git a/include/sound/soc.h b/include/sound/soc.h index 91244a0..769e27c 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1039,6 +1039,7 @@ struct snd_soc_pcm_runtime { struct snd_soc_dpcm_runtime dpcm[2]; long pmdown_time; + unsigned char pop_wait:1; /* runtime devices */ struct snd_pcm *pcm; diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 967d0e1..5fbfb06 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -113,7 +113,7 @@ static int soc_compr_free(struct snd_compr_stream *cstream) SNDRV_PCM_STREAM_PLAYBACK, SND_SOC_DAPM_STREAM_STOP); } else - codec_dai->pop_wait = 1; + rtd->pop_wait = 1; schedule_delayed_work(&rtd->delayed_work, msecs_to_jiffies(rtd->pmdown_time)); } else { diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 5c3ca2a..d7711fc 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -334,11 +334,11 @@ static void close_delayed_work(struct work_struct *work) dev_dbg(rtd->dev, "ASoC: pop wq checking: %s status: %s waiting: %s\n", codec_dai->driver->playback.stream_name, codec_dai->playback_active ? "active" : "inactive", - codec_dai->pop_wait ? "yes" : "no"); + rtd->pop_wait ? "yes" : "no"); /* are we waiting on this codec DAI stream */ - if (codec_dai->pop_wait == 1) { - codec_dai->pop_wait = 0; + if (rtd->pop_wait == 1) { + rtd->pop_wait = 0; snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, SND_SOC_DAPM_STREAM_STOP); } @@ -408,7 +408,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) SND_SOC_DAPM_STREAM_STOP); } else { /* start delayed pop wq here for playback streams */ - codec_dai->pop_wait = 1; + rtd->pop_wait = 1; schedule_delayed_work(&rtd->delayed_work, msecs_to_jiffies(rtd->pmdown_time)); } @@ -480,8 +480,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) /* cancel any delayed stream shutdown that is pending */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && - codec_dai->pop_wait) { - codec_dai->pop_wait = 0; + rtd->pop_wait) { + rtd->pop_wait = 0; cancel_delayed_work(&rtd->delayed_work); } -- 1.7.9.5

2 1

[alsa-devel] [PATCH] configure: do not detect incorrect cross-compiler
by Olivier Blin 15 Dec '12

15 Dec '12

On Ubuntu 11.04, configuring with --build=x86_64-unknown-linux-gnu --host=x86_64-linux-gnu finds a wrong cross-compiler: checking for cross-compiler... x86_64-x86_64-pc-linux-gnu-gcc This happens because of a dash vs underscore inconsistency in configure.in: host=x86_64-pc-linux-gnu host_cpu=x86_64 host_os=linux-gnu which ${host_cpu}-${host_os}-gcc >/dev/null 2>&1 && echo ${host_cpu}-${host-os}-gcc This bug has been introduced in the initial --with-host support from 2002, commit eb267ade29c9a49c07b1c33dc9bf7a6790217400 This configure command is about "cross-compiling for i586", where the system compiler is used, which just -m32 additional options. The --build value comes from config.guess. --- configure.in | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/configure.in b/configure.in index e397dad..4af917b 100644 --- a/configure.in +++ b/configure.in @@ -31,7 +31,7 @@ then which ${program_prefix}gcc >/dev/null 2>&1 && CC=${program_prefix}gcc which ${host_cpu}-${host_os}-gcc >/dev/null 2>&1 \ - && CC=${host_cpu}-${host-os}-gcc + && CC=${host_cpu}-${host_os}-gcc which ${host_cpu}-${host_vendor}-${host_os}-gcc >/dev/null 2>&1 \ && CC=${host_cpu}-${host_vendor}-${host_os}-gcc -- 1.8.0.2

2 1

[alsa-devel] [PATCH] usb-audio: ignore-quirk for HP Wireless Audio
by Eldad Zack 15 Dec '12

15 Dec '12

As Joe Cooper <swelljoe(a)gmail.com> reported, "On most HP Envy laptops the snd-usb-audio module causes the system to become unresponsive and Gnome Shell 3 to crash.". See also: http://mailman.alsa-project.org/pipermail/alsa-devel/2012-December/057729.h… Add a quirk to ignore this device (for now) to solve the instability issue and allow other USB audio devices to be used. Reported-by: Joe Cooper <swelljoe(a)gmail.com> Tested-by: Isaac Smith <hunternet93(a)gmail.com> Signed-off-by: Eldad Zack <eldad(a)fogrefinery.com> --- sound/usb/quirks-table.h | 36 ++++++++++++++++++++++++++++++++++++ 1 files changed, 36 insertions(+), 0 deletions(-) Applies against mainline 3.7.0 and sound-3.8. diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 49f9af9..e3c59f0 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -99,6 +99,42 @@ }, /* + * HP Wireless Audio + * When not ignored, causes instability issues for some users, forcing them to + * blacklist the entire module. + */ +{ + USB_DEVICE(0x0424, 0xb832), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Standard Microsystems Corp.", + .product_name = "HP Wireless Audio", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + /* Mixer */ + { + .ifnum = 0, + .type = QUIRK_IGNORE_INTERFACE, + }, + /* Playback */ + { + .ifnum = 1, + .type = QUIRK_IGNORE_INTERFACE, + }, + /* Capture */ + { + .ifnum = 2, + .type = QUIRK_IGNORE_INTERFACE, + }, + /* HID Device, .ifnum = 3 */ + { + .ifnum = -1, + } + } + } +}, + +/* * Logitech QuickCam: bDeviceClass is vendor-specific, so generic interface * class matches do not take effect without an explicit ID match. */ -- 1.7.8.6

2 1

[alsa-devel] [PATCH v2 1/1] ALSA: hda - bug fix for invalid connection list of Haswell HDMI codec pin
by mengdong.lin＠intel.com 14 Dec '12

14 Dec '12

From: Mengdong Lin <mengdong.lin(a)intel.com> Haswell HDMI codec pins may report invalid connection list entries, which will cause failure to play audio via HDMI. So this patch - allow generic hdmi patch to pick up and apply fixups for different codecs. - add fixup for Haswell to workaround this hardware issue: enable DP1.2 mode and override the connection list entries with proper value. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> Signed-off-by: Xingchao Wang <xingchao.wang(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 71555cc..42fdf14 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -38,6 +38,7 @@ #include <sound/tlv.h> #include "hda_codec.h" #include "hda_local.h" +#include "hda_auto_parser.h" #include "hda_jack.h" static bool static_hdmi_pcm; @@ -79,6 +80,7 @@ struct hdmi_spec_per_pin { }; struct hdmi_spec { + struct hda_gen_spec gen; int num_cvts; struct hdmi_spec_per_cvt cvts[MAX_HDMI_CVTS]; @@ -1687,6 +1689,54 @@ static const struct hda_codec_ops generic_hdmi_patch_ops = { .unsol_event = hdmi_unsol_event, }; +static void intel_haswell_fixup_connect_list(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + unsigned int vendor_param; + hda_nid_t list[3] = {0x2, 0x3, 0x4}; + + if (action != HDA_FIXUP_ACT_PRE_PROBE) + return; + + vendor_param = snd_hda_codec_read(codec, 0x08, 0, 0xf81, 0); + if (vendor_param == -1 || vendor_param & 0x02) + return; + + /* enable DP1.2 mode */ + vendor_param |= 0x02; + snd_hda_codec_read(codec, 0x08, 0, 0x781, vendor_param); + + vendor_param = snd_hda_codec_read(codec, 0x08, 0, 0xf81, 0); + if (vendor_param == -1 || !(vendor_param & 0x02)) + return; + + snd_hda_override_conn_list(codec, 0x06, 3, list); + hdmi_read_pin_conn(codec, 1); +} + + +/* available models for fixup */ +enum { + INTEL_HASWELL, +}; + +static const struct hda_model_fixup hdmi_models[] = { + {.id = INTEL_HASWELL, .name = "Haswell"}, + {} +}; + +static const struct snd_pci_quirk hdmi_fixup_tbl[] = { + SND_PCI_QUIRK(0x8086, 0x2010, "Haswell", INTEL_HASWELL), + {} /* terminator */ +}; + +static const struct hda_fixup hdmi_fixups[] = { + [INTEL_HASWELL] = { + .type = HDA_FIXUP_FUNC, + .v.func = intel_haswell_fixup_connect_list, + }, +}; + static int patch_generic_hdmi(struct hda_codec *codec) { struct hdmi_spec *spec; @@ -1696,6 +1746,11 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; + snd_hda_gen_init(&spec->gen); + + snd_hda_pick_fixup(codec, hdmi_models, hdmi_fixup_tbl, hdmi_fixups); + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); + if (hdmi_parse_codec(codec) < 0) { codec->spec = NULL; kfree(spec); -- 1.7.9.5

2 2

[alsa-devel] [PATCH 1/1] ALSA: hda - bug fix for invalid connection list of Haswell HDMI codec pin
by mengdong.lin＠intel.com 14 Dec '12

14 Dec '12

From: Mengdong Lin <mengdong.lin(a)intel.com> Haswell HDMI codec pins may report invalid connection list entries, which will cause failure to play audio via HDMI. So this patch - allow generic hdmi patch to pick up and apply fixups for different codecs. - add fixup for Haswell to workaround this hardware issue: enable DP1.2 mode and override invalid connection list entries reported by hardware. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> Signed-off-by: Xingchao Wang <xingchao.wang(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 71555cc..2bf1758 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -38,6 +38,7 @@ #include <sound/tlv.h> #include "hda_codec.h" #include "hda_local.h" +#include "hda_auto_parser.h" #include "hda_jack.h" static bool static_hdmi_pcm; @@ -79,6 +80,7 @@ struct hdmi_spec_per_pin { }; struct hdmi_spec { + struct hda_gen_spec gen; int num_cvts; struct hdmi_spec_per_cvt cvts[MAX_HDMI_CVTS]; @@ -1687,6 +1689,69 @@ static const struct hda_codec_ops generic_hdmi_patch_ops = { .unsol_event = hdmi_unsol_event, }; +static void intel_haswell_fixup_enable_dp12_mode(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + unsigned int vendor_param; + + if (action != HDA_FIXUP_ACT_PRE_PROBE) + return; + + vendor_param = snd_hda_codec_read(codec, 0x08, 0, 0xf81, 0); + if (vendor_param == -1 || vendor_param & 0x02) + return; + + vendor_param |= 0x02; + snd_hda_codec_read(codec, 0x08, 0, 0x781, vendor_param); +} + +static void intel_haswell_fixup_connect_list(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + hda_nid_t list[3] = {0x2, 0x3, 0x4}; + unsigned int vendor_param; + + if (action != HDA_FIXUP_ACT_PROBE) + return; + + vendor_param = snd_hda_codec_read(codec, 0x08, 0, 0xf81, 0); + if (vendor_param == -1 || !(vendor_param & 0x02)) + return; + + snd_hda_override_conn_list(codec, 0x06, 3, list); + hdmi_read_pin_conn(codec, 1); +} + + +/* available models for fixup */ +enum { + INTEL_HASWELL, + INTEL_HASWELL_CONNECT_LIST, +}; + +static const struct hda_model_fixup hdmi_models[] = { + {.id = INTEL_HASWELL, .name = "Haswell"}, + {} +}; + +static const struct snd_pci_quirk hdmi_fixup_tbl[] = { + SND_PCI_QUIRK(0x8086, 0x2010, "Haswell", INTEL_HASWELL), + {} /* terminator */ +}; + +static const struct hda_fixup hdmi_fixups[] = { + [INTEL_HASWELL] = { + .type = HDA_FIXUP_FUNC, + .v.func = intel_haswell_fixup_enable_dp12_mode, + .chained = true, + .chain_id = INTEL_HASWELL_CONNECT_LIST, + }, + [INTEL_HASWELL_CONNECT_LIST] = { + .type = HDA_FIXUP_FUNC, + .v.func = intel_haswell_fixup_connect_list, + }, +}; + static int patch_generic_hdmi(struct hda_codec *codec) { struct hdmi_spec *spec; @@ -1696,11 +1761,19 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; + snd_hda_gen_init(&spec->gen); + + snd_hda_pick_fixup(codec, hdmi_models, hdmi_fixup_tbl, hdmi_fixups); + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); + if (hdmi_parse_codec(codec) < 0) { codec->spec = NULL; kfree(spec); return -EINVAL; } + + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); + codec->patch_ops = generic_hdmi_patch_ops; generic_hdmi_init_per_pins(codec); -- 1.7.9.5

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Re: [alsa-devel] ALSA:aloop -- inconsistent {in-softirq-W} -> {softirq-on-W} usage
by Takashi Iwai 14 Dec '12

14 Dec '12

At Thu, 13 Dec 2012 21:36:05 +0100, Andreas Fenkart wrote: > > > Hi, > > Seen this error when stopping playback to loopback. > > # aplay -Dhw:0 -fcd /dev/urandom > Playing raw data '/dev/urandom' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo > ^CAborted by signa[ 65.562992] > [ 65.565493] ================================= > [ 65.570044] [ INFO: inconsistent lock state ] > [ 65.574601] 3.7.0-rc8+ #303 Not tainted > [ 65.578605] --------------------------------- > [ 65.583156] inconsistent {IN-SOFTIRQ-W} -> {SOFTIRQ-ON-W} usage. > [ 65.589436] aplay/156 [HC0[0]:SC0[0]:HE1:SE1] takes: > [ 65.594620] (&(&dpcm->timer_lock)->rlock){+.?...}, at: [<c02e90cd>] loopback_close+0x29/0x98 > [ 65.603604] {IN-SOFTIRQ-W} state was registered at: > [ 65.608699] [<c005d8ef>] mark_irqflags+0xa3/0x11c > [ 65.613820] [<c005dca1>] __lock_acquire+0x339/0x77c > [ 65.619112] [<c005e575>] lock_acquire+0x65/0xac > [ 65.624038] [<c03e0c17>] _raw_spin_lock+0x1f/0x2c > [ 65.629155] [<c02e8cdd>] loopback_timer_start+0x3d/0x108 > [ 65.634899] [<c02e9455>] loopback_timer_function+0x21/0x3c > [ 65.640821] [<c0035f3f>] call_timer_fn+0x63/0xfc > [ 65.645849] [<c0036127>] run_timer_softirq+0x14f/0x190 > [ 65.651410] [<c00311d1>] __do_softirq+0xb1/0x174 > [ 65.656426] [<c0031529>] irq_exit+0x6d/0x70 > [ 65.660989] [<c000dde9>] handle_IRQ+0x2d/0x60 > [ 65.665743] [<c000848d>] omap3_intc_handle_irq+0x59/0x60 > [ 65.671484] [<c000cfff>] __irq_svc+0x3f/0x64 > [ 65.676138] [<c000de86>] default_idle+0x22/0x24 > [ 65.681063] [<c000df89>] cpu_idle+0x89/0x94 > [ 65.685626] [<c05cf5bb>] start_kernel+0x25b/0x264 > [ 65.690743] irq event stamp: 74155 > [ 65.694296] hardirqs last enabled at (74155): [<c03df9b5>] __mutex_unlock_slowpath+0x85/0xec > [ 65.703215] hardirqs last disabled at (74154): [<c03df96b>] __mutex_unlock_slowpath+0x3b/0xec > [ 65.712132] softirqs last enabled at (72724): [<c0031215>] __do_softirq+0xf5/0x174 > [ 65.720141] softirqs last disabled at (72717): [<c0031529>] irq_exit+0x6d/0x70 > [ 65.727697] > [ 65.727697] other info that might help us debug this: > [ 65.734516] Possible unsafe locking scenario: > [ 65.734516] > [ 65.740697] CPU0 > [ 65.743250] ---- > [ 65.745802] lock(&(&dpcm->timer_lock)->rlock); > [ 65.750639] <Interrupt> > [ 65.753373] lock(&(&dpcm->timer_lock)->rlock); > [ 65.758392] > [ 65.758392] *** DEADLOCK *** > [ 65.758392] > [ 65.764581] 1 lock held by aplay/156: > [ 65.768404] #0: (&pcm->open_mutex){+.+.+.}, at: [<c02e0c49>] snd_pcm_release+0x1d/0x7c > [ 65.776907] > [ 65.776907] stack backtrace: > [ 65.781487] [<c0011a15>] (unwind_backtrace+0x1/0xa0) from [<c005d2c9>] (print_usage_bug+0x149/0x184) > [ 65.791039] [<c005d2c9>] (print_usage_bug+0x149/0x184) from [<c005d4c5>] (mark_lock_irq+0x1c1/0x1e0) > [ 65.800590] [<c005d4c5>] (mark_lock_irq+0x1c1/0x1e0) from [<c005d5f7>] (mark_lock+0x113/0x368) > [ 65.809597] [<c005d5f7>] (mark_lock+0x113/0x368) from [<c005d915>] (mark_irqflags+0xc9/0x11c) > [ 65.818514] [<c005d915>] (mark_irqflags+0xc9/0x11c) from [<c005dca1>] (__lock_acquire+0x339/0x77c) > [ 65.827884] [<c005dca1>] (__lock_acquire+0x339/0x77c) from [<c005e575>] (lock_acquire+0x65/0xac) > [ 65.846083] [<c03e0c17>] (_raw_spin_lock+0x1f/0x2c) from [<c02e90cd>] (loopback_close+0x29/0x98) > [ 65.855273] [<c02e90cd>] (loopback_close+0x29/0x98) from [<c02e0bf1>] (snd_pcm_release_substream+0x39/0x74) > [ 65.865458] [<c02e0bf1>] (snd_pcm_release_substream+0x39/0x74) from [<c02e0c4f>] (snd_pcm_release+0x23/0x7c) > [ 65.875748] [<c02e0c4f>] (snd_pcm_release+0x23/0x7c) from [<c00b4dad>] (__fput+0x61/0x170) > [ 65.884409] [<c00b4dad>] (__fput+0x61/0x170) from [<c0040485>] (task_work_run+0x5d/0x8c) > [ 65.892879] [<c0040485>] (task_work_run+0x5d/0x8c) from [<c001017f>] (do_work_pending+0x67/0x7c) > [ 65.902069] [<c001017f>] (do_work_pending+0x67/0x7c) from [<c000d569>] (work_pending+0x9/0x18) > > This seems to be a real deadlock, since loopback_timer_stop is > triggered from ioctl, while loopback_timer_start can be called > from softirq. Could you check the latest Linus tree? The spinlocks of aloop have been already fixed there. Takashi

1 0

[alsa-devel] Polling issues with the PCM multi plugin
by Devin Anderson 14 Dec '12

14 Dec '12

Hi all, On the jack-devel list, we've been having a discussion about the ALSA PCM multi plugin: http://thread.gmane.org/gmane.comp.audio.jackit/26571/focus=26571 I've been having some issues with setting up my 2 Echo Layla 3Gs as a PCM multi. In particular, I've found that there are times when polling the file descriptors of the multi when poll() returns, but there isn't actually a period of data available for capture/playback, even though the poll descriptors indicate that the data should be available. After some debugging and learning about the PCM multi driver: 1.) The PCM multi driver only returns poll descriptors for one of the devices that makes up the multi 2.) When this happens, the device that's actually being polled *does* have data available 3.) When this happens, the device that's not being polled does not yet have a period's worth of data available After making sure that I had word clock enabled and that my word clock cable was working properly, I started delving into the problem some more. From what I can gather, the PCM multi device uses the function `snd_pcm_link()` to sync its devices, and, for start, stop, drain, pause, suspend, resume, reset, and prepare operations, this appears to work fine; however, I can't find any evidence that a PCM device checks to make sure that all of its linked slave devices have their poll descriptors ready with events before indicating that events are available. Questions: 1.) Should `snd_pcm_link` make I/O availability indication dependent on I/O availability of its linked slave devices? 2.) If not, how should this operation be handled? I'm sure it *can* be handled in userspace by bypassing the multi and polling the file descriptors of all sound cards directly, but duplicating this code for each application that wants to poll multiple sound cards seems wrong. Thanks, -- Devin Anderson surfacepatterns (at) gmail (dot) com blog - http://surfacepatterns.blogspot.com/ psinsights - http://psinsights.googlecode.com/ synthclone - http://synthclone.googlecode.com/

2 5

[alsa-devel] [PATCH] ASoC: wm5110: Split input PGA controls
by Mark Brown 14 Dec '12

14 Dec '12

Though the controls are named as stereo controls in the part the main use case for the analogue inputs to the WM5102 is mono. Reflect this in the controls exposed to userspace, providing a series of mono controls rather than stereo ones. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5110.c | 77 +++++++++++++++++++++++++++------------------ 1 file changed, 46 insertions(+), 31 deletions(-) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 794f050..55a1348 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -55,37 +55,52 @@ SOC_SINGLE("IN3 High Performance Switch", ARIZONA_IN3L_CONTROL, SOC_SINGLE("IN4 High Performance Switch", ARIZONA_IN4L_CONTROL, ARIZONA_IN4_OSR_SHIFT, 1, 0), -SOC_DOUBLE_R_RANGE_TLV("IN1 Volume", ARIZONA_IN1L_CONTROL, - ARIZONA_IN1R_CONTROL, - ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), -SOC_DOUBLE_R_RANGE_TLV("IN2 Volume", ARIZONA_IN2L_CONTROL, - ARIZONA_IN2R_CONTROL, - ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), -SOC_DOUBLE_R_RANGE_TLV("IN3 Volume", ARIZONA_IN3L_CONTROL, - ARIZONA_IN3R_CONTROL, - ARIZONA_IN3L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), - -SOC_DOUBLE_R("IN1 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_1L, - ARIZONA_ADC_DIGITAL_VOLUME_1R, ARIZONA_IN1L_MUTE_SHIFT, 1, 1), -SOC_DOUBLE_R("IN2 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_2L, - ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_IN2L_MUTE_SHIFT, 1, 1), -SOC_DOUBLE_R("IN3 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_3L, - ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_MUTE_SHIFT, 1, 1), -SOC_DOUBLE_R("IN4 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_4L, - ARIZONA_ADC_DIGITAL_VOLUME_4R, ARIZONA_IN4L_MUTE_SHIFT, 1, 1), - -SOC_DOUBLE_R_TLV("IN1 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1L, - ARIZONA_ADC_DIGITAL_VOLUME_1R, ARIZONA_IN1L_DIG_VOL_SHIFT, - 0xbf, 0, digital_tlv), -SOC_DOUBLE_R_TLV("IN2 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2L, - ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_IN2L_DIG_VOL_SHIFT, - 0xbf, 0, digital_tlv), -SOC_DOUBLE_R_TLV("IN3 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_3L, - ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_DIG_VOL_SHIFT, - 0xbf, 0, digital_tlv), -SOC_DOUBLE_R_TLV("IN4 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_4L, - ARIZONA_ADC_DIGITAL_VOLUME_4R, ARIZONA_IN4L_DIG_VOL_SHIFT, - 0xbf, 0, digital_tlv), +SOC_SINGLE_RANGE_TLV("IN1L Volume", ARIZONA_IN1L_CONTROL, + ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN1R Volume", ARIZONA_IN1R_CONTROL, + ARIZONA_IN1R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN2L Volume", ARIZONA_IN2L_CONTROL, + ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN2R Volume", ARIZONA_IN2R_CONTROL, + ARIZONA_IN2R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN3L Volume", ARIZONA_IN3L_CONTROL, + ARIZONA_IN3L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN3R Volume", ARIZONA_IN3R_CONTROL, + ARIZONA_IN3R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), + +SOC_SINGLE("IN1L Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_1L, + ARIZONA_IN1L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN1R Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_1R, + ARIZONA_IN1R_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN2L Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_2L, + ARIZONA_IN2L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN2R Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_2R, + ARIZONA_IN2R_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN3L Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_3L, + ARIZONA_IN3L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN3R Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_3R, + ARIZONA_IN3R_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN4L Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_4L, + ARIZONA_IN4L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN4R Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_4R, + ARIZONA_IN4R_MUTE_SHIFT, 1, 1), + +SOC_SINGLE_TLV("IN1L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1L, + ARIZONA_IN1L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN1R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1R, + ARIZONA_IN1R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN2L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2L, + ARIZONA_IN2L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN2R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2R, + ARIZONA_IN2R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN3L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_3L, + ARIZONA_IN3L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN3R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_3R, + ARIZONA_IN3R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN4L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_4L, + ARIZONA_IN4L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN4R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_4R, + ARIZONA_IN4R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), SOC_ENUM("Input Ramp Up", arizona_in_vi_ramp), SOC_ENUM("Input Ramp Down", arizona_in_vd_ramp), -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ASoC: wm2000: Use clock API integration to configure MCLK divisor
by Mark Brown 14 Dec '12

14 Dec '12

Since we are now using the clock API integration to manage MCLK we can now use clk_get_rate() to determine if we need to divide MCLK without relying on platform data. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/wm2000.h | 3 --- sound/soc/codecs/wm2000.c | 6 +++--- 2 files changed, 3 insertions(+), 6 deletions(-) diff --git a/include/sound/wm2000.h b/include/sound/wm2000.h index aa388ca..4de81f4 100644 --- a/include/sound/wm2000.h +++ b/include/sound/wm2000.h @@ -15,9 +15,6 @@ struct wm2000_platform_data { /** Filename for system-specific image to download to device. */ const char *download_file; - /** Divide MCLK by 2 for system clock? */ - unsigned int mclkdiv2:1; - /** Disable speech clarity enhancement, for use when an * external algorithm is used. */ unsigned int speech_enh_disable:1; diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 0aba8ce..85550dc 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -73,7 +73,6 @@ struct wm2000_priv { unsigned int anc_eng_ena:1; unsigned int spk_ena:1; - unsigned int mclk_div:1; unsigned int speech_clarity:1; int anc_download_size; @@ -133,6 +132,7 @@ static int wm2000_poll_bit(struct i2c_client *i2c, static int wm2000_power_up(struct i2c_client *i2c, int analogue) { struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); + unsigned long rate; int ret; BUG_ON(wm2000->anc_mode != ANC_OFF); @@ -145,7 +145,8 @@ static int wm2000_power_up(struct i2c_client *i2c, int analogue) return ret; } - if (!wm2000->mclk_div) { + rate = clk_get_rate(wm2000->mclk); + if (rate <= 13500000) { dev_dbg(&i2c->dev, "Disabling MCLK divider\n"); wm2000_write(i2c, WM2000_REG_SYS_CTL2, WM2000_MCLK_DIV2_ENA_CLR); @@ -847,7 +848,6 @@ static int wm2000_i2c_probe(struct i2c_client *i2c, filename = "wm2000_anc.bin"; pdata = dev_get_platdata(&i2c->dev); if (pdata) { - wm2000->mclk_div = pdata->mclkdiv2; wm2000->speech_clarity = !pdata->speech_enh_disable; if (pdata->download_file) -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ASoC: wm5102: Split input PGA controls
by Mark Brown 14 Dec '12

14 Dec '12

Though the controls are named as stereo controls in the part the main use case for the analogue inputs to the WM5102 is mono. Reflect this in the controls exposed to userspace, providing a series of mono controls rather than stereo ones. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5102.c | 64 +++++++++++++++++++++++++++------------------ 1 file changed, 38 insertions(+), 26 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index b20feb2..3831645 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -611,32 +611,44 @@ SOC_SINGLE("IN2 High Performance Switch", ARIZONA_IN2L_CONTROL, SOC_SINGLE("IN3 High Performance Switch", ARIZONA_IN3L_CONTROL, ARIZONA_IN3_OSR_SHIFT, 1, 0), -SOC_DOUBLE_R_RANGE_TLV("IN1 Volume", ARIZONA_IN1L_CONTROL, - ARIZONA_IN1R_CONTROL, - ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), -SOC_DOUBLE_R_RANGE_TLV("IN2 Volume", ARIZONA_IN2L_CONTROL, - ARIZONA_IN2R_CONTROL, - ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), -SOC_DOUBLE_R_RANGE_TLV("IN3 Volume", ARIZONA_IN3L_CONTROL, - ARIZONA_IN3R_CONTROL, - ARIZONA_IN3L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), - -SOC_DOUBLE_R("IN1 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_1L, - ARIZONA_ADC_DIGITAL_VOLUME_1R, ARIZONA_IN1L_MUTE_SHIFT, 1, 1), -SOC_DOUBLE_R("IN2 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_2L, - ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_IN2L_MUTE_SHIFT, 1, 1), -SOC_DOUBLE_R("IN3 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_3L, - ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_MUTE_SHIFT, 1, 1), - -SOC_DOUBLE_R_TLV("IN1 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1L, - ARIZONA_ADC_DIGITAL_VOLUME_1R, ARIZONA_IN1L_DIG_VOL_SHIFT, - 0xbf, 0, digital_tlv), -SOC_DOUBLE_R_TLV("IN2 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2L, - ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_IN2L_DIG_VOL_SHIFT, - 0xbf, 0, digital_tlv), -SOC_DOUBLE_R_TLV("IN3 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_3L, - ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_DIG_VOL_SHIFT, - 0xbf, 0, digital_tlv), +SOC_SINGLE_RANGE_TLV("IN1L Volume", ARIZONA_IN1L_CONTROL, + ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN1R Volume", ARIZONA_IN1R_CONTROL, + ARIZONA_IN1R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN2L Volume", ARIZONA_IN2L_CONTROL, + ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN2R Volume", ARIZONA_IN2R_CONTROL, + ARIZONA_IN2R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN3L Volume", ARIZONA_IN3L_CONTROL, + ARIZONA_IN3L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN3R Volume", ARIZONA_IN3R_CONTROL, + ARIZONA_IN3R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), + +SOC_SINGLE("IN1L Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_1L, + ARIZONA_IN1L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN1R Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_1R, + ARIZONA_IN1R_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN2L Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_2L, + ARIZONA_IN2L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN2R Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_2R, + ARIZONA_IN2R_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN3L Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_3L, + ARIZONA_IN3L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN3R Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_3R, + ARIZONA_IN3R_MUTE_SHIFT, 1, 1), + +SOC_SINGLE_TLV("IN1L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1L, + ARIZONA_IN1L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN1R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1R, + ARIZONA_IN1R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN2L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2L, + ARIZONA_IN2L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN2R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2R, + ARIZONA_IN2R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN3L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_3L, + ARIZONA_IN3L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN3R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_3R, + ARIZONA_IN3R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), SOC_ENUM("Input Ramp Up", arizona_in_vi_ramp), SOC_ENUM("Input Ramp Down", arizona_in_vd_ramp), -- 1.7.10.4

1 0

[alsa-devel] Presonus 44VSL AudioBox drivers
by Nicholas Hadaway 14 Dec '12

14 Dec '12

I own a Presonus AudioBox 44VSL and it does not work with alsa currently. I am an active linux user that runs git kernels and keeps an active development environment. I would love to get this hardware supported. what do I need to do to help you guys with this? Thanks! -Nick

1 0

[alsa-devel] Request for HW auto-recognition: HP Pavilion dv7 with STAC92xx
by Tomas Pospisek 13 Dec '12

13 Dec '12

Hello, I have a HP Pavilion dv7 laptop (LS066EA#UUZ), with: # cat /proc/asound/card0/codec* | grep Codec Codec: IDT 92HD81B1X5 Codec: Intel CougarPoint HDMI when play sound without any tweaks, I get a very weak sound output. If I want to have "beats audio", that is subwoofer and all speakers, I need to set: # cat /etc/modprobe.d/alsa-base.conf options snd-hda-intel model=hp-dv7-4000 Could the snd_hda_intel alsa driver be updated to recognize and setup the driver accordingly? alsa-info.sh output here: http://www.alsa-project.org/db/?f=54ea874f43f12c1572aff3ee3113dc4b0ad0b398 Thanks, *t

2 3

[alsa-devel] [PATCH v2 1/1] ALSA: hda - use usleep_range in link reset and change timeout check
by mengdong.lin＠intel.com 13 Dec '12

13 Dec '12

From: Mengdong Lin <mengdong.lin(a)intel.com> Reducing the time on HDA link reset can help to reduce the driver loading time. So we replace msleep with usleep_range to get more accurate time control and change the value to a smaller one. And a 100ms timeout is set for both entering and exiting the link reset. Signed-off-by: Xingchao Wang <xingchao.wang(a)intel.com> Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 72b085a..ed3a290 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1041,7 +1041,7 @@ static void azx_power_notify(struct hda_bus *bus, bool power_up); /* reset codec link */ static int azx_reset(struct azx *chip, int full_reset) { - int count; + unsigned long timeout; if (!full_reset) goto __skip; @@ -1052,24 +1052,26 @@ static int azx_reset(struct azx *chip, int full_reset) /* reset controller */ azx_writel(chip, GCTL, azx_readl(chip, GCTL) & ~ICH6_GCTL_RESET); - count = 50; - while (azx_readb(chip, GCTL) && --count) - msleep(1); + timeout = jiffies + msecs_to_jiffies(100); + while (azx_readb(chip, GCTL) && + time_before(jiffies, timeout)) + usleep_range(500, 1000); /* delay for >= 100us for codec PLL to settle per spec * Rev 0.9 section 5.5.1 */ - msleep(1); + usleep_range(500, 1000); /* Bring controller out of reset */ azx_writeb(chip, GCTL, azx_readb(chip, GCTL) | ICH6_GCTL_RESET); - count = 50; - while (!azx_readb(chip, GCTL) && --count) - msleep(1); + timeout = jiffies + msecs_to_jiffies(100); + while (!azx_readb(chip, GCTL) && + time_before(jiffies, timeout)) + usleep_range(500, 1000); /* Brent Chartrand said to wait >= 540us for codecs to initialize */ - msleep(1); + usleep_range(1000, 1200); __skip: /* check to see if controller is ready */ -- 1.7.9.5

5 12

[alsa-devel] [PATCH 0/2] Make ASUS Xonar U3 S/PDIF channel accessible
by Denis Washington 13 Dec '12

13 Dec '12

Hello, This is my first ALSA patch ever (in fact, also my first kernel patch and my first e-mail to this mailing list); as such, I'm sorry when I did something wrong and/or didn't follow the procedure properly. I hope you can forgive me. I recently bought an ASUS Xonar U3 USB audio card. It general it works well with Linux, but unfortunately, it's PCM output channel is unusable for headphones because it allows virtually no volume control (maximum loudness is already reach at a volume level of 2 or 3). The card does also have a digital (S/PDIF) output channel, which however isn't switched on by the Linux USB audio driver: http://www.mail-archive.com/alsa-user@lists.sourceforge.net/msg28970.html When I boot into Windows (where the ASUS driver seemingly enables the channel) and then reboot into Linux, I can use this channel and control volume as expected. It's just this turning on of S/PDIF which is missing, as suggested by the alsa-user thread linked to above. The patchset that follows enables S/PDIF on the Xonar U3 by using the (seemingly compatible) mixer quirks code as for the U1, just as done and tested by the initiator of the alsa-user thread, together with a very minor cleanup (renaming internal xonar_u1_* identifiers to just xonar_u_*). The change seems to work well (tested with the 3.7.0 Ubuntu kernel as found in today' daily version). Regard, Denis Washington

2 4

Re: [alsa-devel] [PATCH 1/2] ASoC: tlv320aic326x: Support for TI TLV320AIC3262 audio driver
by Mark Brown 13 Dec '12

13 Dec '12

On Wed, Dec 12, 2012 at 04:23:48PM +0530, Mehar Bajwa wrote: > TI TLV320AIC3262 is a flexible, low-power, low-voltage stereo audio > codec with digital microphone inputs and programmable outputs. It includes > powertune capabilities, enhanced fully-programmable miniDSP, fixed predefined > and parameterizable signal processing blocks, integrated PLL and flexible digital > interfaces. Is this in some way different to the patch set you sent yesterday?

2 1

Re: [alsa-devel] [PATCH] ASoC: max98090: Resubmit device driver
by Mark Brown 13 Dec '12

13 Dec '12

On Wed, Dec 12, 2012 at 04:36:10PM -0800, Jerry Wong wrote: > I have made the changes you suggested below, except: I've not read the patch yet but I see that once more I need to ask you to look at and follow SubmittingPatches - you're getting closer but the at least the changelog is nothing like it should be. Please look at both SubmittingPatches and other people's patches and try to follow the standard form.

1 0

[alsa-devel] [PATCH 1/1] ASoC: tpa6130a2: Use devm_* APIs
by Sachin Kamat 13 Dec '12

13 Dec '12

Converted to use devm_gpio_request and devm_regulator_get APIs. These are device managed and make error handling and cleanup a bit simpler. Cc: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- Compile tested on for-next branch of following tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git --- sound/soc/codecs/tpa6130a2.c | 23 ++++++----------------- 1 files changed, 6 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 565ff39..ec78073 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -398,7 +398,8 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client, TPA6130A2_MUTE_L; if (data->power_gpio >= 0) { - ret = gpio_request(data->power_gpio, "tpa6130a2 enable"); + ret = devm_gpio_request(dev, data->power_gpio, + "tpa6130a2 enable"); if (ret < 0) { dev_err(dev, "Failed to request power GPIO (%d)\n", data->power_gpio); @@ -419,16 +420,16 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client, break; } - data->supply = regulator_get(dev, regulator); + data->supply = devm_regulator_get(dev, regulator); if (IS_ERR(data->supply)) { ret = PTR_ERR(data->supply); dev_err(dev, "Failed to request supply: %d\n", ret); - goto err_regulator; + goto err_gpio; } ret = tpa6130a2_power(1); if (ret != 0) - goto err_power; + goto err_gpio; /* Read version */ @@ -440,15 +441,10 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client, /* Disable the chip */ ret = tpa6130a2_power(0); if (ret != 0) - goto err_power; + goto err_gpio; return 0; -err_power: - regulator_put(data->supply); -err_regulator: - if (data->power_gpio >= 0) - gpio_free(data->power_gpio); err_gpio: tpa6130a2_client = NULL; @@ -457,14 +453,7 @@ err_gpio: static int __devexit tpa6130a2_remove(struct i2c_client *client) { - struct tpa6130a2_data *data = i2c_get_clientdata(client); - tpa6130a2_power(0); - - if (data->power_gpio >= 0) - gpio_free(data->power_gpio); - - regulator_put(data->supply); tpa6130a2_client = NULL; return 0; -- 1.7.4.1

2 3

[alsa-devel] [Resend PATCH v6 1/4] pinctrl: atmel-ssc: add pinctrl support
by Bo Shen 13 Dec '12

13 Dec '12

Add pinctrl support for atmel ssc peripheral Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- Resend - based on linux-next-20121115 - commit id: 3805711eb31d3b980c4a55a6994bbc8b22a9ec65 Change since v5 - correct the pull setting, without pull up Change since v4 - This is initial version --- arch/arm/boot/dts/at91sam9260.dtsi | 18 ++++++++++++++++++ arch/arm/boot/dts/at91sam9263.dtsi | 36 ++++++++++++++++++++++++++++++++++++ arch/arm/boot/dts/at91sam9g45.dtsi | 36 ++++++++++++++++++++++++++++++++++++ arch/arm/boot/dts/at91sam9n12.dtsi | 26 ++++++++++++++++++++++++++ arch/arm/boot/dts/at91sam9x5.dtsi | 32 +++++++++++++++++++++++++------- drivers/misc/atmel-ssc.c | 8 ++++++++ 6 files changed, 149 insertions(+), 7 deletions(-) diff --git a/arch/arm/boot/dts/at91sam9260.dtsi b/arch/arm/boot/dts/at91sam9260.dtsi index 3164dcd..f9dc7db 100644 --- a/arch/arm/boot/dts/at91sam9260.dtsi +++ b/arch/arm/boot/dts/at91sam9260.dtsi @@ -217,6 +217,22 @@ }; }; + ssc0 { + pinctrl_ssc0_tx: ssc0_tx-0 { + atmel,pins = + <1 16 0x1 0x0 /* PB16 periph A */ + 1 17 0x1 0x0 /* PB17 periph A */ + 1 18 0x1 0x0>; /* PB18 periph A */ + }; + + pinctrl_ssc0_rx: ssc0_rx-0 { + atmel,pins = + <1 19 0x1 0x0 /* PB19 periph A */ + 1 20 0x1 0x0 /* PB20 periph A */ + 1 21 0x1 0x0>; /* PB21 periph A */ + }; + }; + pioA: gpio@fffff400 { compatible = "atmel,at91rm9200-gpio"; reg = <0xfffff400 0x200>; @@ -350,6 +366,8 @@ compatible = "atmel,at91rm9200-ssc"; reg = <0xfffbc000 0x4000>; interrupts = <14 4 5>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_ssc0_tx &pinctrl_ssc0_rx>; status = "disable"; }; diff --git a/arch/arm/boot/dts/at91sam9263.dtsi b/arch/arm/boot/dts/at91sam9263.dtsi index 96e0bc9..6e9a4ff 100644 --- a/arch/arm/boot/dts/at91sam9263.dtsi +++ b/arch/arm/boot/dts/at91sam9263.dtsi @@ -165,6 +165,38 @@ }; }; + ssc0 { + pinctrl_ssc0_tx: ssc0_tx-0 { + atmel,pins = + <1 0 0x2 0x0 /* PB0 periph B */ + 1 1 0x2 0x0 /* PB1 periph B */ + 1 2 0x2 0x0>; /* PB2 periph B */ + }; + + pinctrl_ssc0_rx: ssc0_rx-0 { + atmel,pins = + <1 3 0x2 0x0 /* PB3 periph B */ + 1 4 0x2 0x0 /* PB4 periph B */ + 1 5 0x2 0x0>; /* PB5 periph B */ + }; + }; + + ssc1 { + pinctrl_ssc1_tx: ssc1_tx-0 { + atmel,pins = + <1 6 0x1 0x0 /* PB6 periph A */ + 1 7 0x1 0x0 /* PB7 periph A */ + 1 8 0x1 0x0>; /* PB8 periph A */ + }; + + pinctrl_ssc1_rx: ssc1_rx-0 { + atmel,pins = + <1 9 0x1 0x0 /* PB9 periph A */ + 1 10 0x1 0x0 /* PB10 periph A */ + 1 11 0x1 0x0>; /* PB11 periph A */ + }; + }; + pioA: gpio@fffff200 { compatible = "atmel,at91rm9200-gpio"; reg = <0xfffff200 0x200>; @@ -262,6 +294,8 @@ compatible = "atmel,at91rm9200-ssc"; reg = <0xfff98000 0x4000>; interrupts = <16 4 5>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_ssc0_tx &pinctrl_ssc0_rx>; status = "disable"; }; @@ -269,6 +303,8 @@ compatible = "atmel,at91rm9200-ssc"; reg = <0xfff9c000 0x4000>; interrupts = <17 4 5>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_ssc1_tx &pinctrl_ssc1_rx>; status = "disable"; }; diff --git a/arch/arm/boot/dts/at91sam9g45.dtsi b/arch/arm/boot/dts/at91sam9g45.dtsi index c40c3eb..63cf206 100644 --- a/arch/arm/boot/dts/at91sam9g45.dtsi +++ b/arch/arm/boot/dts/at91sam9g45.dtsi @@ -198,6 +198,38 @@ }; }; + ssc0 { + pinctrl_ssc0_tx: ssc0_tx-0 { + atmel,pins = + <3 0 0x1 0x0 /* PD0 periph A */ + 3 1 0x1 0x0 /* PD1 periph A */ + 3 2 0x1 0x0>; /* PD2 periph A */ + }; + + pinctrl_ssc0_rx: ssc0_rx-0 { + atmel,pins = + <3 3 0x1 0x0 /* PD3 periph A */ + 3 4 0x1 0x0 /* PD4 periph A */ + 3 5 0x1 0x0>; /* PD5 periph A */ + }; + }; + + ssc1 { + pinctrl_ssc1_tx: ssc1_tx-0 { + atmel,pins = + <3 10 0x1 0x0 /* PD10 periph A */ + 3 11 0x1 0x0 /* PD11 periph A */ + 3 12 0x1 0x0>; /* PD12 periph A */ + }; + + pinctrl_ssc1_rx: ssc1_rx-0 { + atmel,pins = + <3 13 0x1 0x0 /* PD13 periph A */ + 3 14 0x1 0x0 /* PD14 periph A */ + 3 15 0x1 0x0>; /* PD15 periph A */ + }; + }; + pioA: gpio@fffff200 { compatible = "atmel,at91rm9200-gpio"; reg = <0xfffff200 0x200>; @@ -331,6 +363,8 @@ compatible = "atmel,at91sam9g45-ssc"; reg = <0xfff9c000 0x4000>; interrupts = <16 4 5>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_ssc0_tx &pinctrl_ssc0_rx>; status = "disable"; }; @@ -338,6 +372,8 @@ compatible = "atmel,at91sam9g45-ssc"; reg = <0xfffa0000 0x4000>; interrupts = <17 4 5>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_ssc1_tx &pinctrl_ssc1_rx>; status = "disable"; }; diff --git a/arch/arm/boot/dts/at91sam9n12.dtsi b/arch/arm/boot/dts/at91sam9n12.dtsi index 7b644c5..b211da6 100644 --- a/arch/arm/boot/dts/at91sam9n12.dtsi +++ b/arch/arm/boot/dts/at91sam9n12.dtsi @@ -28,6 +28,7 @@ tcb1 = &tcb1; i2c0 = &i2c0; i2c1 = &i2c1; + ssc0 = &ssc0; }; cpus { cpu@0 { @@ -199,6 +200,22 @@ }; }; + ssc0 { + pinctrl_ssc0_tx: ssc0_tx-0 { + atmel,pins = + <0 24 0x2 0x0 /* PA24 periph B */ + 0 25 0x2 0x0 /* PA25 periph B */ + 0 26 0x2 0x0>; /* PA26 periph B */ + }; + + pinctrl_ssc0_rx: ssc0_rx-0 { + atmel,pins = + <0 27 0x2 0x0 /* PA27 periph B */ + 0 28 0x2 0x0 /* PA28 periph B */ + 0 29 0x2 0x0>; /* PA29 periph B */ + }; + }; + pioA: gpio@fffff400 { compatible = "atmel,at91sam9x5-gpio", "atmel,at91rm9200-gpio"; reg = <0xfffff400 0x200>; @@ -249,6 +266,15 @@ status = "disabled"; }; + ssc0: ssc@f0010000 { + compatible = "atmel,at91sam9g45-ssc"; + reg = <0xf0010000 0x4000>; + interrupts = <28 4 5>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_ssc0_tx &pinctrl_ssc0_rx>; + status = "disable"; + }; + usart0: serial@f801c000 { compatible = "atmel,at91sam9260-usart"; reg = <0xf801c000 0x4000>; diff --git a/arch/arm/boot/dts/at91sam9x5.dtsi b/arch/arm/boot/dts/at91sam9x5.dtsi index 1cb26c28..df84f39 100644 --- a/arch/arm/boot/dts/at91sam9x5.dtsi +++ b/arch/arm/boot/dts/at91sam9x5.dtsi @@ -88,13 +88,6 @@ interrupts = <1 4 7>; }; - ssc0: ssc@f0010000 { - compatible = "atmel,at91sam9g45-ssc"; - reg = <0xf0010000 0x4000>; - interrupts = <28 4 5>; - status = "disable"; - }; - tcb0: timer@f8008000 { compatible = "atmel,at91sam9x5-tcb"; reg = <0xf8008000 0x100>; @@ -214,6 +207,22 @@ }; }; + ssc0 { + pinctrl_ssc0_tx: ssc0_tx-0 { + atmel,pins = + <0 24 0x2 0x0 /* PA24 periph B */ + 0 25 0x2 0x0 /* PA25 periph B */ + 0 26 0x2 0x0>; /* PA26 periph B */ + }; + + pinctrl_ssc0_rx: ssc0_rx-0 { + atmel,pins = + <0 27 0x2 0x0 /* PA27 periph B */ + 0 28 0x2 0x0 /* PA28 periph B */ + 0 29 0x2 0x0>; /* PA29 periph B */ + }; + }; + pioA: gpio@fffff400 { compatible = "atmel,at91sam9x5-gpio", "atmel,at91rm9200-gpio"; reg = <0xfffff400 0x200>; @@ -266,6 +275,15 @@ status = "disabled"; }; + ssc0: ssc@f0010000 { + compatible = "atmel,at91sam9g45-ssc"; + reg = <0xf0010000 0x4000>; + interrupts = <28 4 5>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_ssc0_tx &pinctrl_ssc0_rx>; + status = "disable"; + }; + usart0: serial@f801c000 { compatible = "atmel,at91sam9260-usart"; reg = <0xf801c000 0x200>; diff --git a/drivers/misc/atmel-ssc.c b/drivers/misc/atmel-ssc.c index a769719..3b3d0e0 100644 --- a/drivers/misc/atmel-ssc.c +++ b/drivers/misc/atmel-ssc.c @@ -19,6 +19,7 @@ #include <linux/module.h> #include <linux/of.h> +#include <linux/pinctrl/consumer.h> /* Serialize access to ssc_list and user count */ static DEFINE_SPINLOCK(user_lock); @@ -131,6 +132,13 @@ static int ssc_probe(struct platform_device *pdev) struct resource *regs; struct ssc_device *ssc; const struct atmel_ssc_platform_data *plat_dat; + struct pinctrl *pinctrl; + + pinctrl = devm_pinctrl_get_select_default(&pdev->dev); + if (IS_ERR(pinctrl)) { + dev_err(&pdev->dev, "Failed to request pinctrl\n"); + return PTR_ERR(pinctrl); + } ssc = devm_kzalloc(&pdev->dev, sizeof(struct ssc_device), GFP_KERNEL); if (!ssc) { -- 1.7.9.5

2 3

[alsa-devel] "disable" or "disabled" in DT?
by Takashi Iwai 13 Dec '12

13 Dec '12

Hi, in the recent commit for atmel-ssc to add DT support, you added the lines status = "disable"; Now looking through all commits for merging 3.8-rc1, I noticed that other codes have status = "disabled"; I think this should be kept consistent. Takashi

2 1

[alsa-devel] ALSA sound/usb/mixer.c:415 cannot get current value for control 2 ch 1: err = -22
by isaac smith 12 Dec '12

12 Dec '12

Success! I applied the patch against kernel 3.6.9 (AMD's Catalyst driver doesn't support 3.7 yet) and I can now use the snd-usb-audio driver without it hanging up on the wireless audio. I tested a USB sound card, it worked perfectly. There's no sign of the HP Wireless card in ALSA. Thanks a lot! Now I can finally use my slightly decent speakers and my USB MIDI controller. On Tue, Dec 11, 2012 at 12:59 PM, Eldad Zack <eldad(a)fogrefinery.com> wrote: > > On Mon, 10 Dec 2012, isaac smith wrote: > >> I have the same issue with my HP Envy 15. The sound card in question >> is an HP Wireless Audio card that only works with HP-branded wireless >> speakers, so playing to the card produces no output whatsoever, but >> does hang whatever process tries to access it. I doubt that many >> people ever use this feature, so unless someone invests the time to >> get it working I'd like a way to just ignore it completely. I've >> blacklisted the snd-usb-audio module for now, but I have a USB sound >> card and a USB MIDI controller that I'd like to be able to use. > > Hi Isaac, > > Try this patch - applies against 3.7. > > Let me know if that works for you and you can successfully use other USB > audio devices. > > Cheers, > Eldad > > diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h > index 88d8ceb..be68c66 100644 > --- a/sound/usb/quirks-table.h > +++ b/sound/usb/quirks-table.h > @@ -99,6 +99,42 @@ > }, > > /* > + * Standard Microsystems Corp. > + * HP Wireless Audio > + */ > +{ > + USB_DEVICE(0x0424, 0xb832), > + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { > + .vendor_name = "Standard Microsystems Corp.", > + .product_name = "HP Wireless Audio", > + .ifnum = QUIRK_ANY_INTERFACE, > + .type = QUIRK_COMPOSITE, > + .data = (const struct snd_usb_audio_quirk[]) { > + /* Mixer */ > + { > + .ifnum = 0, > + .type = QUIRK_IGNORE_INTERFACE, > + }, > + /* Playback */ > + { > + .ifnum = 1, > + .type = QUIRK_IGNORE_INTERFACE, > + }, > + /* Capture */ > + { > + .ifnum = 2, > + .type = QUIRK_IGNORE_INTERFACE, > + }, > + /* HID Device, maybe standard compliant? */ > + { > + /* .ifnum = 3 */ > + .ifnum = -1, > + } > + } > + } > +}, > + > +/* > * Logitech QuickCam: bDeviceClass is vendor-specific, so generic interface > * class matches do not take effect without an explicit ID match. > */ > >

1 0

[alsa-devel] Bug in aplay?
by Damien Zammit 12 Dec '12

12 Dec '12

Hi, I think i found a small bug in alsa utility "aplay". If you have a sound device that supports the S24_3BE format natively, and you have a wav file of the same format, aplay will not play it because it detects the wrong format and assumes it is default format. (ie 8000Hz Mono 1 channel). I compiled my own version of aplay and it works because I hacked the default format to match this format, but it is not the ideal solution. pi@raspberrypi ~ $ aplay --version aplay: version 1.0.25 by Jaroslav Kysela <perex(a)perex.cz> pi@raspberrypi ~ $ aplay -D hw:1 48000-S24_3BE.wav Playing raw data '48000-S24_3BE.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono aplay: set_params:1081: Sample format non available Available formats: - S24_3BE My version: pi@raspberrypi ~ $ ./aplay -D hw:1 48000-S24_3BE.wav Playing raw data '48000-S24_3BE.wav' : Signed 24 bit Big Endian in 3bytes, Rate 48000 Hz, Stereo aplay: set_params:1145: Unable to install hw params: ACCESS: RW_INTERLEAVED FORMAT: S24_3BE SUBFORMAT: STD SAMPLE_BITS: 24 FRAME_BITS: 48 CHANNELS: 2 RATE: 48000 PERIOD_TIME: 125000 PERIOD_SIZE: 6000 PERIOD_BYTES: 36000 PERIODS: 4 BUFFER_TIME: 500000 BUFFER_SIZE: 24000 BUFFER_BYTES: 144000 TICK_TIME: 0 pi@raspberrypi ~ $ This is making it tricky for me to use my sound card. For some reason it works better with jackd. The error above is also a mystery to me, but it seems to be detecting the format better. Regards, Damien

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[alsa-devel] [PATCH 0/2] Fix usb-audio delay calculation
by Takashi Iwai 12 Dec '12

12 Dec '12

Hi, the following two patches are for fixing the wrong PCM delay reported from usb-audio driver. The first one is trivial, just to fix / optimize for the capture direction. The second one is a deeper fix for corner cases like playback pause. Takashi

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[alsa-devel] [PATCH 2/2] ASoC: aic: Support for AIC family DSPs
by Mehar Bajwa 12 Dec '12

12 Dec '12

AIC family of audio CODECs from TI features a programmable miniDSP for performing signal processing operations. Due to commonality of functions across the CODECs a common library will be used to provide support for them. Signed-off-by: Mehar Bajwa <mehar.bajwa(a)ti.com> --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/aic3xxx_cfw.h | 529 +++++++++++++++++++ sound/soc/codecs/aic3xxx_cfw_ops.c | 989 ++++++++++++++++++++++++++++++++++++ sound/soc/codecs/aic3xxx_cfw_ops.h | 98 ++++ 5 files changed, 1623 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/aic3xxx_cfw.h create mode 100644 sound/soc/codecs/aic3xxx_cfw_ops.c create mode 100644 sound/soc/codecs/aic3xxx_cfw_ops.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 73d8cea..52b3601 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -188,6 +188,11 @@ config SND_SOC_ADAV80X config SND_SOC_ADS117X tristate +config SND_SOC_AIC_CFW + tristate + default y if SND_SOC_TLV320AIC3262=y + default m if SND_SOC_TLV320AIC3262=m + config SND_SOC_AK4104 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index a3f8f4f..ffda11b 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -9,6 +9,7 @@ snd-soc-adau1701-objs := adau1701.o snd-soc-adau1373-objs := adau1373.o snd-soc-adav80x-objs := adav80x.o snd-soc-ads117x-objs := ads117x.o +snd-soc-aic3xxx-cfw-ops-objs := aic3xxx_cfw_ops.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o snd-soc-ak4641-objs := ak4641.o @@ -132,6 +133,7 @@ obj-$(CONFIG_SND_SOC_ADAU1373) += snd-soc-adau1373.o obj-$(CONFIG_SND_SOC_ADAU1701) += snd-soc-adau1701.o obj-$(CONFIG_SND_SOC_ADAV80X) += snd-soc-adav80x.o obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o +obj-$(CONFIG_SND_SOC_AIC_CFW) += snd-soc-aic3xxx-cfw-ops.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_AK4641) += snd-soc-ak4641.o diff --git a/sound/soc/codecs/aic3xxx_cfw.h b/sound/soc/codecs/aic3xxx_cfw.h new file mode 100644 index 0000000..b88485d --- /dev/null +++ b/sound/soc/codecs/aic3xxx_cfw.h @@ -0,0 +1,529 @@ +/* + * aic3xxx_cfw.h -- SoC audio for TI OMAP44XX SDP + * Codec Firmware Declarations + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef AIC_FIRMWARE_H_ +#define AIC_FIRMWARE_H_ + + +#define AIC_FW_MAGIC 0xC0D1F1ED + + +/** \defgroup pd Arbitrary Limitations */ +/* @{ */ +#ifndef AIC_MAX_ID +# define AIC_MAX_ID (64) /**<Max length of string identifies*/ +# define AIC_MAX_VARS (256) /**<Number of "variables" alive at the*/ + /**<same time in an acx file*/ +#endif + +/* @} */ + + + +/** \defgroup st Enums, Flags, Macros and Supporting Types */ +/* @{ */ + + +/** + * Device Family Identifier + * + */ +enum __attribute__ ((__packed__)) aic_dfamily { + AIC_DFM_TYPE_A, + AIC_DFM_TYPE_B, + AIC_DFM_TYPE_C +}; + +/** + * Device Identifier + * + */ +enum __attribute__ ((__packed__)) aic_device { + AIC_DEV_DAC3120, + AIC_DEV_DAC3100, + AIC_DEV_AIC3120, + AIC_DEV_AIC3100, + AIC_DEV_AIC3110, + AIC_DEV_AIC3111, + AIC_DEV_AIC36, + AIC_DEV_AIC3206, + AIC_DEV_AIC3204, + AIC_DEV_AIC3254, + AIC_DEV_AIC3256, + AIC_DEV_AIC3253, + AIC_DEV_AIC3212, + AIC_DEV_AIC3262, + AIC_DEV_AIC3017, + AIC_DEV_AIC3008, + + AIC_DEV_AIC3266, + AIC_DEV_AIC3285, +}; + +/** + * Transition Sequence Identifier + * + */ +enum aic_transition_t { + AIC_TRN_INIT, + AIC_TRN_RESUME, + AIC_TRN_NEUTRAL, + AIC_TRN_A_MUTE, + AIC_TRN_D_MUTE, + AIC_TRN_AD_MUTE, + AIC_TRN_A_UNMUTE, + AIC_TRN_D_UNMUTE, + AIC_TRN_AD_UNMUTE, + AIC_TRN_SUSPEND, + AIC_TRN_EXIT, + AIC_TRN_N +}; + +#ifndef __cplusplus +static const char *const aic_transition_id[] = { + [AIC_TRN_INIT] "INIT", + [AIC_TRN_RESUME] "RESUME", + [AIC_TRN_NEUTRAL] "NEUTRAL", + [AIC_TRN_A_MUTE] "A_MUTE", + [AIC_TRN_D_MUTE] "D_MUTE", + [AIC_TRN_AD_MUTE] "AD_MUTE", + [AIC_TRN_A_UNMUTE] "A_UNMUTE", + [AIC_TRN_D_UNMUTE] "D_UNMUTE", + [AIC_TRN_AD_UNMUTE]"AD_UNMUTE", + [AIC_TRN_SUSPEND] "SUSPEND", + [AIC_TRN_EXIT] "EXIT", +}; +#endif + +/* @} */ + +/** \defgroup ds Data Structures */ +/* @{ */ + + +/** +* AIC Command +* These commands do not appear in the register +* set of the device. +*/ +enum __attribute__ ((__packed__)) aic_cmd_id { + AIC_CMD_NOP = 0x80, + AIC_CMD_DELAY, + AIC_CMD_UPDTBITS, + AIC_CMD_WAITBITS, + AIC_CMD_LOCK, + AIC_CMD_BURST, + AIC_CMD_RBURST, + AIC_CMD_LOAD_VAR_IM, + AIC_CMD_LOAD_VAR_ID, + AIC_CMD_STORE_VAR, + AIC_CMD_COND, + AIC_CMD_BRANCH, + AIC_CMD_BRANCH_IM, + AIC_CMD_BRANCH_ID, + AIC_CMD_PRINT, + AIC_CMD_OP_ADD = 0xC0, + AIC_CMD_OP_SUB, + AIC_CMD_OP_MUL, + AIC_CMD_OP_DIV, + AIC_CMD_OP_AND, + AIC_CMD_OP_OR, + AIC_CMD_OP_SHL, + AIC_CMD_OP_SHR, + AIC_CMD_OP_RR, + AIC_CMD_OP_XOR, + AIC_CMD_OP_NOT, + AIC_CMD_OP_LNOT, +}; + +/** +* AIC Delay +* Used for the cmd command delay +* Has one parameter of delay time in ms +*/ +struct aic_cmd_delay { + u16 delay; + enum aic_cmd_id cid; + u8 delay_fine; +}; + +/** +* AIC Lock +* Take codec mutex to avoid clashing with DAPM operations +*/ +struct aic_cmd_lock { + u16 lock; + enum aic_cmd_id cid; + u8 unused; +}; + + +/** + * AIC UPDTBITS, WAITBITS, CHKBITS + * Both these cmd commands have same arguments + * cid will be used to specify which command it is + * has parameters of book, page, offset and mask + */ +struct aic_cmd_bitop { + u16 unused1; + enum aic_cmd_id cid; + u8 mask; +}; + +/** + * AIC CMD Burst header + * Burst writes inside command array + * Followed by burst address, first byte + */ +struct aic_cmd_bhdr { + u16 len; + enum aic_cmd_id cid; + u8 unused; +}; + +/** + * AIC CMD Burst + * Burst writes inside command array + * Followed by data to the extent indicated in previous len + * Can be safely cast to aic_burst + */ +struct aic_cmd_burst { + u8 book; + u8 page; + u8 offset; + u8 data[1]; +}; +#define AIC_CMD_BURST_LEN(n) (2 + ((n) - 1 + 3)/4) + +/** + * AIC CMD Scratch register + * For load + * if (svar != dvar) + * dvar = setbits(svar, mask) // Ignore reg + * else + * dvar = setbits(reg, mask) + * For store + * if (svar != dvar) + * reg = setbits(svar, dvar) + * else + * reg = setbits(svar, mask) + * + */ +struct aic_cmd_ldst { + u8 dvar; + u8 svar; + enum aic_cmd_id cid; + u8 mask; +}; + +/** + * AIC CMD Conditional + * May only precede branch. Followed by nmatch+1 jump + * instructions + * cond = svar&mask + * At each of the following nmatch+1 branch command + * if (cond == match) + * take the branch + */ +struct aic_cmd_cond { + u8 svar; + u8 nmatch; + enum aic_cmd_id cid; + u8 mask; +}; +#define AIC_CMD_COND_LEN(nm) (1 + ((nm)+1)) + +/** + * AIC CMD Goto + * For branch, break, continue and stop + */ +struct aic_cmd_branch { + u16 address; + enum aic_cmd_id cid; + u8 match; +}; + +/** + * AIC Debug print + * For diagnostics + */ +struct aic_cmd_print { + u8 fmtlen; + u8 nargs; + enum aic_cmd_id cid; + char fmt[1]; +}; + +#define AIC_CMD_PRINT_LEN(p) (1 + ((p).fmtlen/4) + (((p).nargs + 3)/4)) +#define AIC_CMD_PRINT_ARG(p) (1 + ((p).fmtlen/4)) + +/** + * AIC Arithmetic and logical operations + * Bit 5 indicates if op1 is indirect + * Bit 6 indicates if op2 is indirect + */ +struct aic_cmd_op { + u8 op1; + u8 op2; + enum aic_cmd_id cid; + u8 dst; +}; +#define AIC_CMD_OP1_ID (1u<<5) +#define AIC_CMD_OP2_ID (1u<<4) + +#define AIC_CMD_OP_START AIC_CMD_OP_ADD +#define AIC_CMD_OP_END (AIC_CMD_OP_LNOT|AIC_CMD_OP1_ID|AIC_CMD_OP2_ID) +#define AIC_CMD_OP_IS_UNARY(x) \ + (((x) == AIC_CMD_OP_NOT) || ((x) == AIC_CMD_OP_LNOT)) + + +/** + * AIC Register + * + * A single reg write + * + */ +union aic_register { + struct { + u8 book; + u8 page; + u8 offset; + u8 data; + }; + u32 bpod; +}; + + + +/** + * AIC Command + * + * Can be a either a + * -# single register write, or + * -# command + * + */ +union aic_cmd { + struct { + u16 unused1; + enum aic_cmd_id cid; + u8 unused2; + }; + union aic_register reg; + struct aic_cmd_delay delay; + struct aic_cmd_lock lock; + struct aic_cmd_bitop bitop; + struct aic_cmd_bhdr bhdr; + struct aic_cmd_burst burst; + struct aic_cmd_ldst ldst; + struct aic_cmd_cond cond; + struct aic_cmd_branch branch; + struct aic_cmd_print print; + u8 print_arg[4]; + struct aic_cmd_op op; +}; + +#define AIC_REG_IS_CMD(x) ((x).cid >= AIC_CMD_DELAY) + +/** + * AIC Block Type + * + * Block identifier + * + */ +enum __attribute__ ((__packed__)) aic_block_t { + AIC_BLOCK_SYSTEM_PRE, + AIC_BLOCK_A_INST, + AIC_BLOCK_A_A_COEF, + AIC_BLOCK_A_B_COEF, + AIC_BLOCK_A_F_COEF, + AIC_BLOCK_D_INST, + AIC_BLOCK_D_A1_COEF, + AIC_BLOCK_D_B1_COEF, + AIC_BLOCK_D_A2_COEF, + AIC_BLOCK_D_B2_COEF, + AIC_BLOCK_D_F_COEF, + AIC_BLOCK_SYSTEM_POST, + AIC_BLOCK_N, + AIC_BLOCK_INVALID, +}; +#define AIC_BLOCK_D_A_COEF AIC_BLOCK_D_A1_COEF +#define AIC_BLOCK_D_B_COEF AIC_BLOCK_D_B1_COEF + +/** + * AIC Block + * + * A block of logically grouped sequences/commands/cmd-commands + * + */ +struct aic_block { + enum aic_block_t type; + int ncmds; + union aic_cmd cmd[]; +}; +#define AIC_BLOCK_SIZE(ncmds) (sizeof(struct aic_block) + \ + ((ncmds)*sizeof(union aic_cmd))) + +/** + * AIC Image + * + * A downloadable image + */ +struct aic_image { + char name[AIC_MAX_ID]; /**< Name of the pfw/overlay/configuration*/ + char *desc; /**< User string*/ + int mute_flags; + struct aic_block *block[AIC_BLOCK_N]; +}; + + + +/** + * AIC PLL + * + * PLL configuration sequence and match critirea + */ +struct aic_pll { + char name[AIC_MAX_ID]; /**< Name of the PLL sequence*/ + char *desc; /**< User string*/ + struct aic_block *seq; +}; + +/** + * AIC Control + * + * Run-time control for a process flow + */ +struct aic_control { + char name[AIC_MAX_ID]; /**< Control identifier*/ + char *desc; /**< User string*/ + int mute_flags; + + int min; /**< Min value of control (*100)*/ + int max; /**< Max value of control (*100)*/ + int step; /**< Control step size (*100)*/ + + int imax; /**< Max index into controls array*/ + int ireset; /**< Reset control to defaults*/ + int icur; /**< Last value set*/ + struct aic_block **output; /**< Array of sequences to send*/ +}; + +/** + * Process flow + * + * Complete description of a process flow + */ +struct aic_pfw { + char name[AIC_MAX_ID]; /**< Name of the process flow*/ + char *desc; /**< User string*/ + u32 version; + u8 prb_a; + u8 prb_d; + int novly; /**< Number of overlays (1 or more)*/ + int ncfg; /**< Number of configurations (0 or more)*/ + int nctrl; /**< Number of run-time controls*/ + struct aic_image *base; /**< Base sequence*/ + struct aic_image **ovly_cfg; /**< Overlay and cfg*/ + /**< patches (if any)*/ + struct aic_control **ctrl; /**< Array of run-time controls*/ +}; + +#define AIC_OCFG_NDX(p, o, c) (((o)*(p)->ncfg)+(c)) +/** + * Process transition + * + * Sequence for specific state transisitions within the driver + * + */ +struct aic_transition { + char name[AIC_MAX_ID]; /**< Name of the transition*/ + char *desc; /**< User string*/ + struct aic_block *block; +}; + +/** + * Device audio mode + * + * Link operating modes to process flows, + * configurations and sequences + * + */ +struct aic_mode { + char name[AIC_MAX_ID]; + char *desc; /**< User string*/ + u32 flags; + u8 pfw; + u8 ovly; + u8 cfg; + u8 pll; + struct aic_block *entry; + struct aic_block *exit; +}; + +struct aic_asoc_toc_entry { + char etext[AIC_MAX_ID]; + int mode; + int cfg; +}; + +struct aic_asoc_toc { + int nentries; + struct aic_asoc_toc_entry entry[]; +}; + +/** + * AIC Project + * + * Top level structure describing the AIC project + */ +struct aic_project { + u32 magic; /**< magic number for identifying F/W file*/ + u32 if_id; /**< Interface match code */ + u32 size; /**< Total size of the firmware (including this header)*/ + u32 cksum; /**< CRC32 of the pickled firmware */ + u32 version; /**< Firmware version (from CFD file)*/ + u32 tstamp; /**< Time stamp of firmware build (epoch seconds)*/ + char name[AIC_MAX_ID]; /**< Project name*/ + char *desc; /**< User string*/ + enum aic_dfamily dfamily; /**< Device family*/ + enum aic_device device; /**< Device identifier*/ + u32 flags; /**< AIC flags*/ + + struct aic_transition **transition; /**< Transition sequences*/ + + u16 npll; /**< Number of PLL settings*/ + struct aic_pll **pll; /**< PLL settings*/ + + u16 npfw; /**< Number of process flows*/ + struct aic_pfw **pfw; /**< Process flows*/ + + u16 nmode; /**< Number of operating modes*/ + struct aic_mode **mode; /**< Modes*/ + + struct aic_asoc_toc *asoc_toc; /**< list of amixer controls*/ +}; + + +/* @} */ + +/* **AIC_INTERFACE_ID=0x3FA6D547** */ + +#endif /* AIC_FIRMWARE_H_ */ diff --git a/sound/soc/codecs/aic3xxx_cfw_ops.c b/sound/soc/codecs/aic3xxx_cfw_ops.c new file mode 100644 index 0000000..79b975b --- /dev/null +++ b/sound/soc/codecs/aic3xxx_cfw_ops.c @@ -0,0 +1,989 @@ +/* + * linux/sound/soc/codecs/aic3xxx/aic3xxx_cfw_ops.c + * + * Copyright (C) 2011 Texas Instruments Inc., + * + * This package is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * THIS PACKAGE IS PROVIDED ``AS IS'' AND WITHOUT ANY EXPRESS OR + * IMPLIED WARRANTIES, INCLUDING, WITHOUT LIMITATION, THE IMPLIED + * WARRANTIES OF MERCHANTIBILITY AND FITNESS FOR A PARTICULAR PURPOSE. + * + */ + +#include <linux/module.h> +#include <linux/delay.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <linux/slab.h> +#include <sound/tlv.h> + +#include "aic3xxx_cfw.h" +#include "aic3xxx_cfw_ops.h" + +#define aic_warn(_cfw, fmt, ...)\ + dev_warn(_cfw->dev, "AIC: " fmt, ##__VA_ARGS__) +#define aic_error(_cfw, fmt, ...)\ + dev_err(_cfw->dev, "AIC: " fmt, ##__VA_ARGS__) +#define aic_dbg(_cfw, fmt, ...)\ + dev_dbg(_cfw->dev, "AIC: " fmt, ##__VA_ARGS__) + +#if defined(CONFIG_MFD_AIC3111) || defined(CONFIG_AIC3111_CORE) + +# define AIC3XXX_AIC_DEVICE "aic3111_cfw" +#elif defined(CONFIG_MFD_AIC3256) || defined(CONFIG_AIC3256_CORE) + +# define AIC3XXX_AIC_DEVICE "aic3256_cfw" +#elif defined(CONFIG_MFD_AIC262) || defined(CONFIG_AIC3262_CORE) +# define AIC3XXX_AIC_DEVICE "aic3262_cfw" +#else +# define AIC3XXX_AIC_DEVICE "aic3xxx_cfw" +#endif + +/* **Code beyond this point is compilable on host** */ + +/* + * Firmware version numbers are used to make sure that the + * host and target code stay in sync. It is _not_ recommended + * to provide this number from the outside (E.g., from a makefile) + * Instead, a set of automated tools are relied upon to keep the numbers + * in sync at the time of host testing. + */ +#undef AIC_FW_IF_ID +#define AIC_FW_IF_ID 0x3FA6D547 +static int aic_dlimage(struct aic_state *ps, struct aic_image *pim); +static int aic_dlcfg(struct aic_state *ps, struct aic_image *pim); +static int aic_dlctl(struct aic_state *ps, struct aic_block *pb, + u32 mute_flags); + +static void aic_dlcmds(struct aic_state *ps, struct aic_block *pb); +static int aic_set_mode_id(struct aic_state *ps); +static int aic_mute(struct aic_state *ps, int mute, u32 flags); +static int aic_setmode_cfg_u(struct aic_state *ps, int mode, int cfg); +static int aic_setcfg_u(struct aic_state *ps, int cfg); +static int aic_transition_u(struct aic_state *ps, char *ttype); +static int aic_set_pll_u(struct aic_state *ps, int asi); +static int aic_control_u(struct aic_state *ps, char *cname, int param); +static struct aic_project *aic_unpickle(void *pcfw, int n); + +static void aic_wait(struct aic_state *ps, unsigned int reg, u8 mask, + u8 data); +static void aic_set_bits(u8 *data, u8 mask, u8 val); + +int aic_init(struct aic_state *ps, const struct aic3xxx_codec_ops *ops, + struct snd_soc_codec *codec) +{ + ps->ops = ops; + ps->codec = codec; + ps->pjt = NULL; + mutex_init(&ps->mutex); + + return 0; +} +EXPORT_SYMBOL_GPL(aic_init); + +int aic_lock(struct aic_state *ps, int lock) +{ + if (lock) + mutex_lock(&ps->mutex); + else + mutex_unlock(&ps->mutex); + return 0; +} +EXPORT_SYMBOL_GPL(aic_lock); + +int aic_reload(struct aic_state *ps, void *pcfw, int n) +{ + ps->pjt = aic_unpickle(pcfw, n); + ps->cur_mode_id = + ps->cur_mode = ps->cur_pll = ps->cur_pfw = + ps->cur_ovly = ps->cur_cfg = -1; + if (ps->pjt == NULL) { + aic_error(ps, "version mismatch\n"); + return -EINVAL; + } + return 0; +} +EXPORT_SYMBOL_GPL(aic_reload); + +int aic_setmode(struct aic_state *ps, int mode) +{ + struct aic_project *pjt; + int ret; + + aic_lock(ps, 1); + pjt = ps->pjt; + if (pjt == NULL) { + aic_lock(ps, 0); + return -EINVAL; + } + ret = aic_setmode_cfg_u(ps, mode, pjt->mode[mode]->cfg); + aic_lock(ps, 0); + return ret; +} + +int aic_setcfg(struct aic_state *ps, int cfg) +{ + int ret; + + aic_lock(ps, 1); + ret = aic_setcfg_u(ps, cfg); + aic_lock(ps, 0); + return ret; +} + +static int aic_setcfg_u(struct aic_state *ps, int cfg) +{ + struct aic_project *pjt = ps->pjt; + struct aic_pfw *pfw; + struct aic_image *patch; + + if (pjt == NULL) + return -EINVAL; + if (ps->cur_pfw < 0 || ps->cur_pfw >= pjt->npfw) + return -EINVAL; /* Non miniDSP */ + if (ps->cur_cfg == cfg) + return 0; + pfw = pjt->pfw[ps->cur_pfw]; + if (pfw->ncfg == 0 && cfg != 0) + return -EINVAL; + if (cfg > 0 && cfg >= pfw->ncfg) + return -EINVAL; + ps->cur_cfg = cfg; + aic_set_mode_id(ps); + patch = + pfw->ovly_cfg[AIC_OCFG_NDX(pfw, ps->cur_ovly, ps->cur_cfg)]; + if (pfw->ncfg != 0) + return aic_dlcfg(ps, patch); + return 0; +} + +int aic_setmode_cfg(struct aic_state *ps, int mode, int cfg) +{ + int ret; + + aic_lock(ps, 1); + ret = aic_setmode_cfg_u(ps, mode, cfg); + aic_lock(ps, 0); + return ret; +} +EXPORT_SYMBOL_GPL(aic_setmode_cfg); + +static int aic_setmode_cfg_u(struct aic_state *ps, int mode, int cfg) +{ + struct aic_project *pjt = ps->pjt; + struct aic_mode *pmode; + int which = 0, ocndx; + + if (pjt == NULL) + goto err; + if ((mode < 0) || (mode >= pjt->nmode)) + goto err; + if (cfg < 0) + goto err; + if (mode == ps->cur_mode) + return aic_setcfg_u(ps, cfg); + + /* Apply exit sequence for previous mode if present */ + if (ps->cur_mode >= 0) + aic_dlcmds(ps, pjt->mode[ps->cur_mode]->exit); + pmode = pjt->mode[mode]; + if (pjt->mode[mode]->pfw < pjt->npfw) { /* New mode uses miniDSP */ + struct aic_image *im; + struct aic_pfw *pfw = pjt->pfw[pmode->pfw]; + + /* Make sure cfg is valid and supported in this mode */ + if (pfw->ncfg == 0 && cfg != 0) + goto err; + if (cfg > 0 && cfg >= pfw->ncfg) + goto err; + + /* + * Decisions about which miniDSP to stop/restart are taken + * on the basis of sections present in the _base_ image + * This allows for correct sync mode operation even in cases + * where the base PFW uses both miniDSPs where a particular + * overlay applies only to one + */ + im = pfw->base; + if (im->block[AIC_BLOCK_A_INST]) + which |= AIC3XXX_COPS_MDSP_A; + if (im->block[AIC_BLOCK_D_INST]) + which |= AIC3XXX_COPS_MDSP_D; + + if (pmode->pfw != ps->cur_pfw) { + + /* New mode requires different PFW */ + ps->cur_pfw = pmode->pfw; + ps->cur_ovly = 0; + ps->cur_cfg = 0; + + which = ps->ops->stop(ps->codec, which); + aic_dlimage(ps, im); + if (pmode->ovly && pmode->ovly < pfw->novly) { + + /* New mode uses ovly */ + ocndx = AIC_OCFG_NDX(pfw, pmode->ovly, cfg); + aic_dlimage(ps, + pfw->ovly_cfg[ocndx]); + } else if (pfw->ncfg > 0) { + + /* new mode needs only a cfg change */ + ocndx = AIC_OCFG_NDX(pfw, 0, cfg); + aic_dlimage(ps, + pfw->ovly_cfg[ocndx]); + } + ps->ops->restore(ps->codec, which); + + } else if (pmode->ovly != ps->cur_ovly) { + + /* New mode requires only an ovly change */ + ocndx = AIC_OCFG_NDX(pfw, pmode->ovly, cfg); + which = ps->ops->stop(ps->codec, which); + aic_dlimage(ps, pfw->ovly_cfg[ocndx]); + ps->ops->restore(ps->codec, which); + } else if (pfw->ncfg > 0 && cfg != ps->cur_cfg) { + + /* New mode requires only a cfg change */ + ocndx = AIC_OCFG_NDX(pfw, pmode->ovly, cfg); + aic_dlcfg(ps, pfw->ovly_cfg[ocndx]); + } + ps->cur_ovly = pmode->ovly; + ps->cur_cfg = cfg; + + ps->cur_mode = mode; + aic_set_pll_u(ps, 0); + + } else if (pjt->mode[mode]->pfw != 0xFF) { + + /* Not bypass mode */ + aic_warn(ps, "Bad pfw setting detected (%d). Max pfw=%d", + pmode->pfw, pjt->npfw); + } + ps->cur_mode = mode; + aic_set_mode_id(ps); + + /* Transition to netural mode */ + aic_transition_u(ps, "NEUTRAL"); + + /* Apply entry sequence if present */ + aic_dlcmds(ps, pmode->entry); + + aic_dbg(ps, "setmode_cfg: DONE (mode=%d pfw=%d ovly=%d cfg=%d)", + ps->cur_mode, ps->cur_pfw, ps->cur_ovly, ps->cur_cfg); + return 0; + +err: + aic_dbg(ps, "Failed to set firmware mode"); + return -EINVAL; +} + +int aic_transition(struct aic_state *ps, char *ttype) +{ + int ret; + + aic_lock(ps, 1); + ret = aic_transition_u(ps, ttype); + aic_lock(ps, 0); + return ret; +} +EXPORT_SYMBOL_GPL(aic_transition); + +static int aic_transition_u(struct aic_state *ps, char *ttype) +{ + int i; + + if (ps->pjt == NULL) + return -EINVAL; + for (i = 0; i < AIC_TRN_N; ++i) { + if (!strcasecmp(ttype, aic_transition_id[i])) { + struct aic_transition *pt = ps->pjt->transition[i]; + aic_dbg(ps, "Sending transition %s[%d]", ttype, i); + if (pt) + aic_dlcmds(ps, pt->block); + return 0; + } + } + aic_warn(ps, "Transition %s not present or invalid", ttype); + return 0; +} + +int aic_set_pll(struct aic_state *ps, int asi) +{ + int ret; + + aic_lock(ps, 1); + ret = aic_set_pll_u(ps, asi); + aic_lock(ps, 0); + return ret; +} +EXPORT_SYMBOL_GPL(aic_set_pll); + +static int aic_set_pll_u(struct aic_state *ps, int asi) +{ + struct aic_project *pjt = ps->pjt; + int pll_id; + + if (pjt == NULL) + return -EINVAL; + if (ps->cur_mode < 0) + return -EINVAL; + pll_id = pjt->mode[ps->cur_mode]->pll; + if (ps->cur_pll != pll_id) { + aic_dbg(ps, "Re-configuring PLL: %s==>%d", + pjt->pll[pll_id]->name, + pll_id); + aic_dlcmds(ps, pjt->pll[pll_id]->seq); + ps->cur_pll = pll_id; + } + return 0; +} + +int aic_control(struct aic_state *ps, char *cname, int param) +{ + int ret; + + aic_lock(ps, 1); + ret = aic_control_u(ps, cname, param); + aic_lock(ps, 0); + return ret; +} + +static int aic_control_u(struct aic_state *ps, char *cname, int param) +{ + struct aic_pfw *pfw; + int i; + + if (ps->cur_pfw < 0 || ps->cur_pfw >= ps->pjt->npfw) { + aic_warn(ps, "Not in MiniDSP mode"); + return 0; + } + pfw = ps->pjt->pfw[ps->cur_pfw]; + for (i = 0; i < pfw->nctrl; ++i) { + struct aic_control *pc = pfw->ctrl[i]; + if (strcasecmp(cname, pfw->ctrl[i]->name)) + continue; + if (param < 0 || param > pc->imax) { + aic_warn(ps, "Parameter out of range\n"); + return -EINVAL; + } + aic_dbg(ps, "Sending control %s[%d]", cname, param); + pc->icur = param; + aic_dlctl(ps, pc->output[param], pc->mute_flags); + return 0; + } + aic_warn(ps, "Control named %s not found in pfw %s", cname, pfw->name); + + return -EINVAL; +} + +static void aic_op(struct aic_state *ps, unsigned char *var, + struct aic_cmd_op cmd) +{ + u32 op1, op2; + u32 cid = cmd.cid; + + op1 = cmd.op1; + op2 = cmd.op2; + if (cid & AIC_CMD_OP1_ID) + op1 = var[op1]; + if (cid & AIC_CMD_OP2_ID) + op2 = var[op2]; + cid &= ~(AIC_CMD_OP1_ID | AIC_CMD_OP2_ID); + + switch (cid) { + case AIC_CMD_OP_ADD: + var[cmd.dst] = op1 + op2; + break; + case AIC_CMD_OP_SUB: + var[cmd.dst] = op1 - op2; + break; + case AIC_CMD_OP_MUL: + var[cmd.dst] = op1 * op2; + break; + case AIC_CMD_OP_DIV: + var[cmd.dst] = op1 / op2; + break; + case AIC_CMD_OP_AND: + var[cmd.dst] = op1 & op2; + break; + case AIC_CMD_OP_OR: + var[cmd.dst] = op1 | op2; + break; + case AIC_CMD_OP_SHL: + var[cmd.dst] = (op1 << op2); + break; + case AIC_CMD_OP_SHR: + var[cmd.dst] = (op1 >> op2); + break; + case AIC_CMD_OP_RR: + while (op2--) + var[cmd.dst] = (op1 >> 1) | ((op1 & 1) << 7); + break; + case AIC_CMD_OP_XOR: + var[cmd.dst] = op1 ^ op2; + break; + case AIC_CMD_OP_NOT: + var[cmd.dst] = ~op1; + break; + case AIC_CMD_OP_LNOT: + var[cmd.dst] = !op1; + break; + default: + break; + } +} + +static void aic_dlcmds(struct aic_state *ps, struct aic_block *pb) +{ + int pc = 0, cond = 0; + unsigned char var[256]; + + if (!pb) + return; + while (pc < pb->ncmds) { + union aic_cmd *c = &(pb->cmd[pc]); + if (c->cid != AIC_CMD_BRANCH_IM && + c->cid != AIC_CMD_BRANCH_ID && c->cid != AIC_CMD_NOP) + cond = 0; + switch (c->cid) { + case 0 ... (AIC_CMD_NOP - 1): + ps->ops->reg_write(ps->codec, c->reg.bpod, + c->reg.data); + pc += 1; + break; + case AIC_CMD_NOP: + pc += 1; + break; + case AIC_CMD_DELAY: + mdelay(c->delay.delay); + pc += 1; + break; + case AIC_CMD_UPDTBITS: + ps->ops->set_bits(ps->codec, c[1].reg.bpod, + c->bitop.mask, c[1].reg.data); + pc += 2; + break; + case AIC_CMD_WAITBITS: + aic_wait(ps, c[1].reg.bpod, c->bitop.mask, + c[1].reg.data); + pc += 2; + break; + case AIC_CMD_LOCK: + if (c->delay.delay) + ps->ops->lock(ps->codec); + else + ps->ops->unlock(ps->codec); + pc += 1; + break; + case AIC_CMD_BURST: + ps->ops->bulk_write(ps->codec, c[1].reg.bpod, + c->bhdr.len, c[1].burst.data); + pc += AIC_CMD_BURST_LEN(c->bhdr.len); + break; + case AIC_CMD_RBURST: + ps->ops->bulk_read(ps->codec, c[1].reg.bpod, + c->bhdr.len, c[1].burst.data); + pc += AIC_CMD_BURST_LEN(c->bhdr.len); + break; + case AIC_CMD_LOAD_VAR_IM: + aic_set_bits(&var[c->ldst.dvar], + c->ldst.mask, c->ldst.svar); + pc += 1; + break; + case AIC_CMD_LOAD_VAR_ID: + if (c->ldst.svar != c->ldst.dvar) { + aic_set_bits(&var[c->ldst.dvar], + c->ldst.mask, + var[c->ldst.svar]); + pc += 1; + } else { + u8 data; + data = ps->ops->reg_read(ps->codec, + c[1].reg.bpod); + aic_set_bits(&var[c->ldst.dvar], + c->ldst.mask, data); + pc += 2; + } + break; + case AIC_CMD_STORE_VAR: + if (c->ldst.svar != c->ldst.dvar) + ps->ops->set_bits(ps->codec, + c[1].reg.bpod, + var[c->ldst.dvar], + var[c->ldst.svar]); + else + ps->ops->set_bits(ps->codec, + c[1].reg.bpod, + c->ldst.mask, + var[c->ldst.svar]); + pc += 2; + break; + case AIC_CMD_COND: + cond = var[c->cond.svar] & c->cond.mask; + pc += 1; + break; + case AIC_CMD_BRANCH: + pc = c->branch.address; + break; + case AIC_CMD_BRANCH_IM: + if (c->branch.match == cond) + pc = c->branch.address; + else + pc += 1; + break; + case AIC_CMD_BRANCH_ID: + if (var[c->branch.match] == cond) + pc = c->branch.address; + else + pc += 1; + break; + case AIC_CMD_PRINT: + { + union aic_cmd *parglist = + c + AIC_CMD_PRINT_ARG(c->print); + printk(c->print.fmt, + var[parglist->print_arg[0]], + var[parglist->print_arg[1]], + var[parglist->print_arg[2]], + var[parglist->print_arg[3]]); + pc += AIC_CMD_PRINT_LEN(c->print); + } + break; + case AIC_CMD_OP_START ... AIC_CMD_OP_END: + aic_op(ps, var, c->op); + pc += 1; + break; + default: + aic_warn(ps, "Unknown cmd command %x. Skipped", c->cid); + pc += 1; + break; + } + } +} + +static void aic_wait(struct aic_state *ps, unsigned int reg, u8 mask, + u8 data) +{ + while ((ps->ops->reg_read(ps->codec, reg) & mask) != data) + usleep_range(500, 1000); +} + +static inline void aic_set_bits(u8 *data, u8 mask, u8 val) +{ + *data = (*data & (~mask)) | (val & mask); +} + +static const struct { + u32 mdsp; + int buf_a, buf_b; + u32 swap; +} csecs[] = { + { + .mdsp = AIC3XXX_COPS_MDSP_A, + .swap = AIC3XXX_ABUF_MDSP_A, + .buf_a = AIC_BLOCK_A_A_COEF, + .buf_b = AIC_BLOCK_A_B_COEF + }, + { + .mdsp = AIC3XXX_COPS_MDSP_D, + .swap = AIC3XXX_ABUF_MDSP_D1, + .buf_a = AIC_BLOCK_D_A1_COEF, + .buf_b = AIC_BLOCK_D_B1_COEF + }, + { + .mdsp = AIC3XXX_COPS_MDSP_D, + .swap = AIC3XXX_ABUF_MDSP_D2, + .buf_a = AIC_BLOCK_D_A2_COEF, + .buf_b = AIC_BLOCK_D_B2_COEF + }, +}; +static int aic_dlctl(struct aic_state *ps, struct aic_block *pb, + u32 mute_flags) +{ + int i, btype = pb->type; + int run_state = ps->ops->lock(ps->codec); + + aic_dbg(ps, "Download CTL"); + for (i = 0; i < sizeof(csecs) / sizeof(csecs[0]); ++i) { + if (csecs[i].buf_a != btype && csecs[i].buf_b != btype) + continue; + aic_dbg(ps, "\tDownload once to %d\n", btype); + aic_dlcmds(ps, pb); + if (run_state & csecs[i].mdsp) { + aic_dbg(ps, "Download again to make sure it reaches B"); + aic_mute(ps, 1, run_state & mute_flags); + ps->ops->bswap(ps->codec, csecs[i].swap); + aic_mute(ps, 0, run_state & mute_flags); + aic_dlcmds(ps, pb); + } + break; + } + ps->ops->unlock(ps->codec); + return 0; +} + +static int aic_dlcfg(struct aic_state *ps, struct aic_image *pim) +{ + int i, run_state, swap; + + aic_dbg(ps, "Download CFG %s", pim->name); + run_state = ps->ops->lock(ps->codec); + swap = 0; + for (i = 0; i < sizeof(csecs) / sizeof(csecs[0]); ++i) { + if (!pim->block[csecs[i].buf_a]) + continue; + aic_dlcmds(ps, pim->block[csecs[i].buf_a]); + aic_dlcmds(ps, pim->block[csecs[i].buf_b]); + if (run_state & csecs[i].mdsp) + swap |= csecs[i].swap; + } + if (swap) { + aic_mute(ps, 1, run_state & pim->mute_flags); + ps->ops->bswap(ps->codec, swap); + aic_mute(ps, 0, run_state & pim->mute_flags); + for (i = 0; i < sizeof(csecs) / sizeof(csecs[0]); ++i) { + if (!pim->block[csecs[i].buf_a]) + continue; + if (!(run_state & csecs[i].mdsp)) + continue; + aic_dlcmds(ps, pim->block[csecs[i].buf_a]); + aic_dlcmds(ps, pim->block[csecs[i].buf_b]); + } + } + ps->ops->unlock(ps->codec); + return 0; +} + +static int aic_dlimage(struct aic_state *ps, struct aic_image *pim) +{ + int i; + + if (!pim) + return 0; + aic_dbg(ps, "Download IMAGE %s", pim->name); + for (i = 0; i < AIC_BLOCK_N; ++i) + aic_dlcmds(ps, pim->block[i]); + return 0; +} + +static int aic_mute(struct aic_state *ps, int mute, u32 flags) +{ + if ((flags & AIC3XXX_COPS_MDSP_D) && (flags & AIC3XXX_COPS_MDSP_A)) + aic_transition_u(ps, + mute ? "AD_MUTE" : "AD_UNMUTE"); + else if (flags & AIC3XXX_COPS_MDSP_D) + aic_transition_u(ps, mute ? "D_MUTE" : "D_UNMUTE"); + else if (flags & AIC3XXX_COPS_MDSP_A) + aic_transition_u(ps, mute ? "A_MUTE" : "A_UNMUTE"); + return 0; +} + +static inline void *aic_ndx2ptr(void *p, u8 *base) +{ + return &base[(int)p]; +} +static inline char *aic_desc(void *p, u8 *base) +{ + if (p) + return aic_ndx2ptr(p, base); + return NULL; +} + +static void aic_unpickle_image(struct aic_image *im, void *p) +{ + int i; + + im->desc = aic_desc(im->desc, p); + for (i = 0; i < AIC_BLOCK_N; ++i) + if (im->block[i]) + im->block[i] = aic_ndx2ptr(im->block[i], p); +} + +static void aic_unpickle_control(struct aic_control *ct, void *p) +{ + int i; + + ct->output = aic_ndx2ptr(ct->output, p); + ct->desc = aic_desc(ct->desc, p); + for (i = 0; i <= ct->imax; ++i) + ct->output[i] = aic_ndx2ptr(ct->output[i], p); +} + +static struct aic_project *aic_unpickle(void *p, int n) +{ + struct aic_project *pjt = p; + int i, j; + + if (pjt->magic != AIC_FW_MAGIC || pjt->size != n || + pjt->if_id != AIC_FW_IF_ID) { + return NULL; + } + + pjt->desc = aic_desc(pjt->desc, p); + pjt->transition = aic_ndx2ptr(pjt->transition, p); + for (i = 0; i < AIC_TRN_N; i++) { + if (!pjt->transition[i]) + continue; + pjt->transition[i] = aic_ndx2ptr(pjt->transition[i], p); + pjt->transition[i]->desc = aic_desc( + pjt->transition[i]->desc, p); + pjt->transition[i]->block = aic_ndx2ptr( + pjt->transition[i]->block, p); + } + pjt->pll = aic_ndx2ptr(pjt->pll, p); + for (i = 0; i < pjt->npll; i++) { + pjt->pll[i] = aic_ndx2ptr(pjt->pll[i], p); + pjt->pll[i]->desc = aic_desc(pjt->pll[i]->desc, p); + pjt->pll[i]->seq = aic_ndx2ptr(pjt->pll[i]->seq, p); + } + + pjt->pfw = aic_ndx2ptr(pjt->pfw, p); + for (i = 0; i < pjt->npfw; i++) { + pjt->pfw[i] = aic_ndx2ptr(pjt->pfw[i], p); + pjt->pfw[i]->desc = aic_desc(pjt->pfw[i]->desc, p); + if (pjt->pfw[i]->base) { + pjt->pfw[i]->base = aic_ndx2ptr( + pjt->pfw[i]->base, p); + aic_unpickle_image(pjt->pfw[i]->base, p); + } + pjt->pfw[i]->ovly_cfg = aic_ndx2ptr( + pjt->pfw[i]->ovly_cfg, p); + for (j = 0; j < pjt->pfw[i]->novly * pjt->pfw[i]->ncfg; ++j) { + pjt->pfw[i]->ovly_cfg[j] = aic_ndx2ptr( + pjt->pfw[i]->ovly_cfg[j], p); + aic_unpickle_image(pjt->pfw[i]->ovly_cfg[j], p); + } + if (pjt->pfw[i]->nctrl) + pjt->pfw[i]->ctrl = aic_ndx2ptr( + pjt->pfw[i]->ctrl, p); + for (j = 0; j < pjt->pfw[i]->nctrl; ++j) { + pjt->pfw[i]->ctrl[j] = aic_ndx2ptr( + pjt->pfw[i]->ctrl[j], p); + aic_unpickle_control(pjt->pfw[i]->ctrl[j], p); + } + } + + pjt->mode = aic_ndx2ptr(pjt->mode, p); + for (i = 0; i < pjt->nmode; i++) { + pjt->mode[i] = aic_ndx2ptr(pjt->mode[i], p); + pjt->mode[i]->desc = aic_desc(pjt->mode[i]->desc, p); + if (pjt->mode[i]->entry) + pjt->mode[i]->entry = aic_ndx2ptr( + pjt->mode[i]->entry, p); + if (pjt->mode[i]->exit) + pjt->mode[i]->exit = aic_ndx2ptr( + pjt->mode[i]->exit, p); + } + if (pjt->asoc_toc) + pjt->asoc_toc = aic_ndx2ptr(pjt->asoc_toc, p); + else + return NULL; + + return pjt; +} +static int aic_set_mode_id(struct aic_state *ps) +{ + struct aic_asoc_toc *toc = ps->pjt->asoc_toc; + int i; + + for (i = 0; i < toc->nentries; ++i) { + if (toc->entry[i].cfg == ps->cur_cfg && + toc->entry[i].mode == ps->cur_mode) { + ps->cur_mode_id = i; + return 0; + } + } + aic_dbg(ps, "Unknown mode,cfg combination [%d,%d]", ps->cur_mode, + ps->cur_cfg); + return -EINVAL; +} + +/* **Code beyond this point is not compilable on host** */ + +static int aic3xxx_get_control(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct aic_state *ps = (struct aic_state *)kcontrol->private_value; + struct aic_pfw *pfw; + int i; + + if (ps->cur_pfw >= ps->pjt->npfw) { + aic_dbg(ps, "Not in MiniDSP mode"); + return 0; + } + pfw = ps->pjt->pfw[ps->cur_pfw]; + for (i = 0; i < pfw->nctrl; ++i) { + if (!strcasecmp(kcontrol->id.name, pfw->ctrl[i]->name)) { + struct aic_control *pc = pfw->ctrl[i]; + ucontrol->value.integer.value[0] = pc->icur; + return 0; + } + } + return 0; +} + +static int aic3xxx_put_control(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct aic_state *ps = (struct aic_state *)kcontrol->private_value; + + aic_control(ps, kcontrol->id.name, + ucontrol->value.integer.value[0]); + return 0; +} + +static int aic3xxx_info_control(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *ucontrol) +{ + struct aic_state *ps = (struct aic_state *)kcontrol->private_value; + struct aic_pfw *pfw; + int i; + + if (ps->cur_pfw >= ps->pjt->npfw) { + aic_dbg(ps, "Not in MiniDSP mode"); + return 0; + } + pfw = ps->pjt->pfw[ps->cur_pfw]; + for (i = 0; i < pfw->nctrl; ++i) { + if (!strcasecmp(kcontrol->id.name, pfw->ctrl[i]->name)) { + struct aic_control *pc = pfw->ctrl[i]; + ucontrol->value.integer.min = 0; + ucontrol->value.integer.max = pc->imax; + if (pc->imax == 1) + ucontrol->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + else + ucontrol->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + } + } + + ucontrol->count = 1; + return 0; +} +int aic_add_controls(struct snd_soc_codec *codec, struct aic_state *ps) +{ + int i, j; + struct aic_pfw *pfw; + + for (j = 0; j < ps->pjt->npfw; ++j) { + pfw = ps->pjt->pfw[j]; + + for (i = 0; i < pfw->nctrl; ++i) { + struct aic_control *pc = pfw->ctrl[i]; + struct snd_kcontrol_new *generic_control = + devm_kzalloc(codec->dev, + sizeof(struct snd_kcontrol_new), + GFP_KERNEL); + unsigned int *tlv_array = + devm_kzalloc(codec->dev, + 4 * sizeof(unsigned int), GFP_KERNEL); + + if (generic_control == NULL) + return -ENOMEM; + generic_control->access = + SNDRV_CTL_ELEM_ACCESS_TLV_READ | + SNDRV_CTL_ELEM_ACCESS_READWRITE; + tlv_array[0] = SNDRV_CTL_TLVT_DB_SCALE; + tlv_array[1] = 2 * sizeof(unsigned int); + tlv_array[2] = pc->min; + tlv_array[3] = ((pc->step) & TLV_DB_SCALE_MASK); + if (pc->step > 0) + generic_control->tlv.p = tlv_array; + generic_control->name = pc->name; + generic_control->private_value = (unsigned long) ps; + generic_control->get = aic3xxx_get_control; + generic_control->put = aic3xxx_put_control; + generic_control->info = aic3xxx_info_control; + generic_control->iface = SNDRV_CTL_ELEM_IFACE_MIXER; + snd_soc_add_codec_controls(codec, generic_control, 1); + aic_dbg(ps, "Added control %s", pc->name); + } + } + return 0; + +} +EXPORT_SYMBOL_GPL(aic_add_controls); + +static int aic3xxx_get_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_enum *e = (struct soc_enum *) kcontrol->private_value; + struct aic_state *ps = (struct aic_state *) e->mask; + + ucontrol->value.enumerated.item[0] = ps->cur_mode_id; + + return 0; +} + +static int aic3xxx_put_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_enum *e = (struct soc_enum *) kcontrol->private_value; + struct aic_state *ps = (struct aic_state *) e->mask; + struct aic_asoc_toc *toc; + int index, ret; + + aic_lock(ps, 1); + toc = ps->pjt->asoc_toc; + + index = ucontrol->value.enumerated.item[0]; + if (index < 0 || index >= toc->nentries) { + aic_lock(ps, 0); + return -EINVAL; + } + ret = aic_setmode_cfg_u(ps, toc->entry[index].mode, + toc->entry[index].cfg); + aic_lock(ps, 0); + return ret; +} +EXPORT_SYMBOL_GPL(aic3xxx_put_mode); + +int aic_add_modes(struct snd_soc_codec *codec, struct aic_state *ps) +{ + int j; + struct aic_asoc_toc *toc = ps->pjt->asoc_toc; + struct soc_enum *mode_cfg_enum = + devm_kzalloc(codec->dev, sizeof(struct soc_enum), GFP_KERNEL); + struct snd_kcontrol_new *mode_cfg_control = + devm_kzalloc(codec->dev, sizeof(struct snd_kcontrol_new), + GFP_KERNEL); + + if (mode_cfg_enum == NULL) + goto mem_err; + if (mode_cfg_control == NULL) + goto mem_err; + + mode_cfg_enum->dtexts = devm_kzalloc(codec->dev, + toc->nentries * sizeof(char *), + GFP_KERNEL); + if (mode_cfg_enum->dtexts == NULL) + goto mem_err; + + for (j = 0; j < toc->nentries; j++) + mode_cfg_enum->dtexts[j] = toc->entry[j].etext; + + mode_cfg_enum->reg = j; + mode_cfg_enum->max = toc->nentries; + mode_cfg_enum->mask = (unsigned int) ps; + mode_cfg_control->name = "Codec Firmware Setmode"; + mode_cfg_control->get = aic3xxx_get_mode; + mode_cfg_control->put = aic3xxx_put_mode; + mode_cfg_control->info = snd_soc_info_enum_ext; + mode_cfg_control->private_value = (unsigned long) mode_cfg_enum; + mode_cfg_control->iface = SNDRV_CTL_ELEM_IFACE_MIXER; + snd_soc_add_codec_controls(codec, mode_cfg_control, 1); + return 0; +mem_err: + kfree(mode_cfg_control); + kfree(mode_cfg_enum); + kfree(mode_cfg_enum->texts); + return -ENOMEM; + +} +EXPORT_SYMBOL_GPL(aic_add_modes); + +MODULE_AUTHOR("Hari Rajagopala <harik(a)ti.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/aic3xxx_cfw_ops.h b/sound/soc/codecs/aic3xxx_cfw_ops.h new file mode 100644 index 0000000..6a600fb --- /dev/null +++ b/sound/soc/codecs/aic3xxx_cfw_ops.h @@ -0,0 +1,98 @@ +/* + * aic3xxx_cfw_ops.h -- SoC audio for TI OMAP44XX SDP + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef AIC3XXX_CFW_OPS_H_ +#define AIC3XXX_CFW_OPS_H_ + +#include <linux/mutex.h> +#include <sound/soc.h> +#include <linux/cdev.h> + +struct aic_project; +struct aic3xxx_codec_ops; + +struct aic_state { + const char *dev_name; + int version; + struct aic_project *pjt; + const struct aic3xxx_codec_ops *ops; + struct snd_soc_codec *codec; + struct device *dev; + struct mutex mutex; + int cur_mode_id; + int cur_pll; + int cur_mode; + int cur_pfw; + int cur_ovly; + int cur_cfg; + struct cdev cdev; + int is_open; +}; + +int aic_init(struct aic_state *ps, const struct aic3xxx_codec_ops *ops, + struct snd_soc_codec *codec); +int aic_lock(struct aic_state *ps, int lock); +int aic_reload(struct aic_state *ps, void *pcfw, int n); +int aic_setmode(struct aic_state *ps, int mode); +int aic_setmode_cfg(struct aic_state *ps, int mode, int cfg); +int aic_setcfg(struct aic_state *ps, int cfg); +int aic_transition(struct aic_state *ps, char *ttype); +int aic_set_pll(struct aic_state *ps, int asi); +int aic_control(struct aic_state *ps, char *cname, int param); +int aic_add_controls(struct snd_soc_codec *codec, struct aic_state *ps); +int aic_add_modes(struct snd_soc_codec *codec, struct aic_state *ps); + + +#define AIC3XXX_COPS_MDSP_D_L (0x00000002u) +#define AIC3XXX_COPS_MDSP_D_R (0x00000001u) +#define AIC3XXX_COPS_MDSP_D (AIC3XXX_COPS_MDSP_D_L|AIC3XXX_COPS_MDSP_D_R) + +#define AIC3XXX_COPS_MDSP_A_L (0x00000020u) +#define AIC3XXX_COPS_MDSP_A_R (0x00000010u) +#define AIC3XXX_COPS_MDSP_A (AIC3XXX_COPS_MDSP_A_L|AIC3XXX_COPS_MDSP_A_R) + +#define AIC3XXX_COPS_MDSP_ALL (AIC3XXX_COPS_MDSP_D|AIC3XXX_COPS_MDSP_A) + +#define AIC3XXX_ABUF_MDSP_D1 (0x00000001u) +#define AIC3XXX_ABUF_MDSP_D2 (0x00000002u) +#define AIC3XXX_ABUF_MDSP_A (0x00000010u) +#define AIC3XXX_ABUF_MDSP_ALL \ + (AIC3XXX_ABUF_MDSP_D1|AIC3XXX_ABUF_MDSP_D2|AIC3XXX_ABUF_MDSP_A) + + +struct aic3xxx_codec_ops { + int (*reg_read) (struct snd_soc_codec *codec, unsigned int reg); + int (*reg_write) (struct snd_soc_codec *codec, unsigned int reg, + unsigned char val); + int (*set_bits) (struct snd_soc_codec *codec, unsigned int reg, + unsigned char mask, unsigned char val); + int (*bulk_read) (struct snd_soc_codec *codec, unsigned int reg, + int count, u8 *buf); + int (*bulk_write) (struct snd_soc_codec *codec, unsigned int reg, + int count, const u8 *buf); + int (*lock) (struct snd_soc_codec *codec); + int (*unlock) (struct snd_soc_codec *codec); + int (*stop) (struct snd_soc_codec *codec, int mask); + int (*restore) (struct snd_soc_codec *codec, int runstate); + int (*bswap) (struct snd_soc_codec *codec, int mask); +}; + +MODULE_LICENSE("GPL"); + +#endif -- 1.7.0.4

1 0

[alsa-devel] how to update the hw_ptr in the interrupt?
by Zhang wei 12 Dec '12

12 Dec '12

Hello every one.i am in a trouble and could anyone help me figure out ? I have spent couple days trying to understand ,but I do not know the pricinple clearly which decides how to adjust the hw_ptr in the interrupt routine. Below is the reference sourcecode in the snd_pcm_update_hw_ptr0(). When the DMA interrupt ocurrs the process will enter the snd_pcm_update_hw_ptr0() with the in_interrupt=1. ========================================================== if (in_interrupt) { /* we know that one period was processed */ /* delta = "expected next hw_ptr" for in_interrupt != 0 */ delta = runtime->hw_ptr_interrupt + runtime->period_size; if (delta > new_hw_ptr) { /* check for double acknowledged interrupts */ hdelta = jiffies - runtime->hw_ptr_jiffies; if (hdelta > runtime->hw_ptr_buffer_jiffies/2) { hw_base += runtime->buffer_size; if (hw_base >= runtime->boundary) hw_base = 0; new_hw_ptr = hw_base + pos; goto __delta; } } } ========================================================== The question is: what can we get from the result of the compare between delta and new_hw_ptr.? Why we can conclue it? The delta represent the expected location in the range of *boundary* , and it producted based the hw_ptr_interrupt(*not fixed*) and perod_size(fixed)? The new_hw_ptr is producted by runtime->hw_ptr_base plus pos. What Is the relationship between delta and new_hw_ptr? Why we can compare them with each other ? Any tips will be ok. Thanks a lot. Best Regards zw

1 0

[alsa-devel] [PATCH] ASoC: arizona: Implement tristate support
by Mark Brown 12 Dec '12

12 Dec '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 231fae0..a85a68f 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -859,11 +859,27 @@ static int arizona_dai_set_sysclk(struct snd_soc_dai *dai, return snd_soc_dapm_sync(&codec->dapm); } +static int arizona_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct snd_soc_codec *codec = dai->codec; + int base = dai->driver->base; + unsigned int reg; + + if (tristate) + reg = ARIZONA_AIF1_TRI; + else + reg = 0; + + return snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL, + ARIZONA_AIF1_TRI, reg); +} + const struct snd_soc_dai_ops arizona_dai_ops = { .startup = arizona_startup, .set_fmt = arizona_set_fmt, .hw_params = arizona_hw_params, .set_sysclk = arizona_dai_set_sysclk, + .set_tristate = arizona_set_tristate, }; EXPORT_SYMBOL_GPL(arizona_dai_ops); -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ASoC: wm8993: Refactor set_pll code to avoid GCC warnings
by Fabio Estevam 12 Dec '12

12 Dec '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> Refactor set_pll code to avoid the following warnings: sound/soc/codecs/wm8983.c:873:40: warning: 'pll_div.k' may be used uninitialized in this function [-Wuninitialized] sound/soc/codecs/wm8983.c:870:9: warning: 'pll_div.n' may be used uninitialized in this function [-Wuninitialized] sound/soc/codecs/wm8983.c:869:23: warning: 'pll_div.div2' may be used uninitialized in this function [-Wuninitialized] Do the same as in commit 86ce6c9a (ASoC: WM8804: Refactor set_pll code to avoid GCC warnings). Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/wm8983.c | 41 ++++++++++++++++++++++------------------- 1 file changed, 22 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c index 9fe1e04..e7c73ff 100644 --- a/sound/soc/codecs/wm8983.c +++ b/sound/soc/codecs/wm8983.c @@ -851,30 +851,33 @@ static int wm8983_set_pll(struct snd_soc_dai *dai, int pll_id, struct pll_div pll_div; codec = dai->codec; - if (freq_in && freq_out) { + if (!freq_in || !freq_out) { + /* disable the PLL before re-programming it */ + snd_soc_update_bits(codec, WM8983_POWER_MANAGEMENT_1, + WM8983_PLLEN_MASK, 0); + return 0; + } else { ret = pll_factors(&pll_div, freq_out * 4 * 2, freq_in); if (ret) return ret; - } - - /* disable the PLL before re-programming it */ - snd_soc_update_bits(codec, WM8983_POWER_MANAGEMENT_1, - WM8983_PLLEN_MASK, 0); - if (!freq_in || !freq_out) - return 0; + /* disable the PLL before re-programming it */ + snd_soc_update_bits(codec, WM8983_POWER_MANAGEMENT_1, + WM8983_PLLEN_MASK, 0); + + /* set PLLN and PRESCALE */ + snd_soc_write(codec, WM8983_PLL_N, + (pll_div.div2 << WM8983_PLL_PRESCALE_SHIFT) + | pll_div.n); + /* set PLLK */ + snd_soc_write(codec, WM8983_PLL_K_3, pll_div.k & 0x1ff); + snd_soc_write(codec, WM8983_PLL_K_2, (pll_div.k >> 9) & 0x1ff); + snd_soc_write(codec, WM8983_PLL_K_1, (pll_div.k >> 18)); + /* enable the PLL */ + snd_soc_update_bits(codec, WM8983_POWER_MANAGEMENT_1, + WM8983_PLLEN_MASK, WM8983_PLLEN); + } - /* set PLLN and PRESCALE */ - snd_soc_write(codec, WM8983_PLL_N, - (pll_div.div2 << WM8983_PLL_PRESCALE_SHIFT) - | pll_div.n); - /* set PLLK */ - snd_soc_write(codec, WM8983_PLL_K_3, pll_div.k & 0x1ff); - snd_soc_write(codec, WM8983_PLL_K_2, (pll_div.k >> 9) & 0x1ff); - snd_soc_write(codec, WM8983_PLL_K_1, (pll_div.k >> 18)); - /* enable the PLL */ - snd_soc_update_bits(codec, WM8983_POWER_MANAGEMENT_1, - WM8983_PLLEN_MASK, WM8983_PLLEN); return 0; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: wm8962: Unconditionally wait for the FLL to lock
by Mark Brown 12 Dec '12

12 Dec '12

If the FLL is being shut down we will exit early so there is no need to check here and in fact we're checking the wrong thing anyway. Reported-by: Graeme Gregory <gg(a)slimlogic.co.uk> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 26 ++++++++++++-------------- 1 file changed, 12 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 09816e8..50b0983 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2875,22 +2875,20 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source, ret = 0; - if (fll1 & WM8962_FLL_ENA) { - /* This should be a massive overestimate but go even - * higher if we'll error out - */ - if (wm8962->irq) - timeout = msecs_to_jiffies(5); - else - timeout = msecs_to_jiffies(1); + /* This should be a massive overestimate but go even + * higher if we'll error out + */ + if (wm8962->irq) + timeout = msecs_to_jiffies(5); + else + timeout = msecs_to_jiffies(1); - timeout = wait_for_completion_timeout(&wm8962->fll_lock, - timeout); + timeout = wait_for_completion_timeout(&wm8962->fll_lock, + timeout); - if (timeout == 0 && wm8962->irq) { - dev_err(codec->dev, "FLL lock timed out"); - ret = -ETIMEDOUT; - } + if (timeout == 0 && wm8962->irq) { + dev_err(codec->dev, "FLL lock timed out"); + ret = -ETIMEDOUT; } wm8962->fll_fref = Fref; -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ALSA - Replace msleep() with usleep_range() to save module loading time
by Wang Xingchao 12 Dec '12

12 Dec '12

msleep(1) is not so accurate and may cause almost 15ms delay in azx_reset(). usleep_range() can reduce ~30ms loading time for Intel Haswell which has two HDA controllers. Signed-off-by: Wang Xingchao <wangxingchao2011(a)gmail.com> --- sound/pci/hda/hda_intel.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index eb48109..865df90 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1075,22 +1075,22 @@ static int azx_reset(struct azx *chip, int full_reset) count = 50; while (azx_readb(chip, GCTL) && --count) - msleep(1); + usleep_range(1000, 1500); /* delay for >= 100us for codec PLL to settle per spec * Rev 0.9 section 5.5.1 */ - msleep(1); + usleep_range(1000, 1500); /* Bring controller out of reset */ azx_writeb(chip, GCTL, azx_readb(chip, GCTL) | ICH6_GCTL_RESET); count = 50; while (!azx_readb(chip, GCTL) && --count) - msleep(1); + usleep_range(1000, 1500); /* Brent Chartrand said to wait >= 540us for codecs to initialize */ - msleep(1); + usleep_range(1000, 1500); __skip: /* check to see if controller is ready */ -- 1.7.9.5

3 2

[alsa-devel] ALSA sound/usb/mixer.c:415 cannot get current value for control 2 ch 1: err = -22
by isaac smith 11 Dec '12

11 Dec '12

Hello, I have the same issue with my HP Envy 15. The sound card in question is an HP Wireless Audio card that only works with HP-branded wireless speakers, so playing to the card produces no output whatsoever, but does hang whatever process tries to access it. I doubt that many people ever use this feature, so unless someone invests the time to get it working I'd like a way to just ignore it completely. I've blacklisted the snd-usb-audio module for now, but I have a USB sound card and a USB MIDI controller that I'd like to be able to use. Here's the output of "lsusb -vd 0424:b832" (apologies for length): Bus 004 Device 004: ID 0424:b832 Standard Microsystems Corp. Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 64 idVendor 0x0424 Standard Microsystems Corp. idProduct 0xb832 bcdDevice 75.28 iManufacturer 1 iProduct 2 iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 225 bNumInterfaces 4 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x80 (Bus Powered) MaxPower 500mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 3 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 72 bInCollection 2 baInterfaceNr( 0) 1 baInterfaceNr( 1) 2 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 2 wTerminalType 0x0402 Headset bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 6 wTerminalType 0x0301 Speaker bAssocTerminal 0 bSourceID 11 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 7 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 12 iTerminal 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 11 bSourceID 1 bControlSize 1 bmaControls( 0) 0x01 Mute Control bmaControls( 1) 0x02 Volume Control bmaControls( 2) 0x02 Volume Control iFeature 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 12 bSourceID 2 bControlSize 1 bmaControls( 0) 0x01 Mute Control bmaControls( 1) 0x02 Volume Control bmaControls( 2) 0x02 Volume Control iFeature 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 1 bDelay 0 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 2 Discrete tSamFreq[ 0] 44100 tSamFreq[ 1] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 13 Transfer Type Isochronous Synch Type Synchronous Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 4 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 1 Milliseconds wLockDelay 0 Milliseconds Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 7 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 2 Discrete tSamFreq[ 0] 48000 tSamFreq[ 1] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 13 Transfer Type Isochronous Synch Type Synchronous Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 4 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bLockDelayUnits 1 Milliseconds wLockDelay 0 Milliseconds Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 3 Human Interface Device bInterfaceSubClass 0 No Subclass bInterfaceProtocol 0 None iInterface 0 HID Device Descriptor: bLength 9 bDescriptorType 33 bcdHID 1.10 bCountryCode 0 Not supported bNumDescriptors 1 bDescriptorType 34 Report wDescriptorLength 74 Report Descriptors: ** UNAVAILABLE ** Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x8a EP 10 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0008 1x 8 bytes bInterval 10 Device Status: 0x060a (Bus Powered) Remote Wakeup Enabled > > On Thu, 6 Dec 2012, Joe Cooper wrote: > > On most HP Envy laptops the snd-usb-audio module causes the system to > > become unresponsive and Gnome Shell 3 to crash. This effects every kernel > > I've tried from 3.3.4 on up to 3.7rc8. It also produces these errors on the > > console: > > > > ALSA sound/usb/mixer.c:415 cannot get current value for control 2 ch 1: err > > = -22 > > > > ALSA sound/usb/clock.c:228 3:1:1: cannot set freq 44100 to ep 0x1 > > > > Blacklisting this module allows the system to function, but with most of > > its speakers not working. It also prevents use of other USB audio devices, > > making multitrack recording impossible. > > > > I have a ticket about it in the Fedora bug tracker, but was referred to > > alsa-devel. That ticket links to other discussions of this bug, as well. > > Here's that ticket: https://bugzilla.redhat.com/show_bug.cgi?id=884368 > > > > alsa-info is here: https://bugzilla.redhat.com/attachment.cgi?id=658989 > > > > Please let me know if you need further details, or if there's anything else > > I can do to help troubleshoot this. > > Hi Joe, > > I can try to help. > > Do you get any sound output at all (even if it stutters or sounds > wrong)? > > Can you please send the output of "lsusb -v -d 0424:b832"? > I think I can work out a patch for you to skip this device, so > you can use other audio usb devices until this gets proper support. > > Cheers, > Eldad

2 1

[alsa-devel] Denon DN-X 1600 Support
by Florian Beginn 11 Dec '12

11 Dec '12

Hello, i just made my first steps fiddling with the alsa-driver and got my DN-X 1600 working. It's a dj-mixer connected via usb, offering 8 channel input/output and midi. Samplerate can be set to either 44.1, 48 or 96khz. It seems to not answer properly on samplerate requests from snd_usb_ctl_msg() so i hardcoded it to 96khz. Regrettably, this is a pretty ugly workaround and somehow broke midi capability, although audio is running fine. I'd be happy if one of you could help me work out how to do this right/better. Find attached the patches and all information i could gather. P.S. I'm not used to git (well, any VCS) and hope the diff format is ok for you. --- a/sound/usb/clock.c 2012-01-25 08:56:49.000000000 +0100 +++ b/sound/usb/clock.c 2012-12-11 17:33:47.685196544 +0100 @@ -255,6 +255,14 @@ int err, crate; int clock = snd_usb_clock_find_source(chip, fmt->clock); + /* + * Denon DN-X 1600 won't respond when asked for samplerate. + * Currently hardcoded to only work with 96khz + */ + + if (chip->usb_id == USB_ID(0x154e, 0x500e) && rate == 96000) + return 0; + if (clock < 0) return clock; --- a/sound/usb/format.c 2012-12-07 17:11:00.000000000 +0100 +++ b/sound/usb/format.c 2012-12-11 17:42:14.069181583 +0100 @@ -281,6 +281,21 @@ int nr_triplets, data_size, ret = 0; int clock = snd_usb_clock_find_source(chip, fp->clock); + /* + * Denon DN-X 1600 won't respond when asked for samplerate. + * Currently hardcoded to only work with 96khz + */ + if (chip->usb_id == USB_ID(0x154e, 0x500e)) + { + fp->nr_rates = 1; + fp->rate_min = 96000; + fp->rate_max = 96000; + fp->rates = SNDRV_PCM_RATE_96000; + fp->rate_table = kmalloc(sizeof(int), GFP_KERNEL); + fp->rate_table[0] = 96000; + return 0; + } + if (clock < 0) { snd_printk(KERN_ERR "%s(): unable to find clock source (clock %d)\n", __func__, clock); --- a/sound/usb/pcm.c 2012-01-25 08:56:49.000000000 +0100 +++ b/sound/usb/pcm.c 2012-12-11 17:57:11.579301294 +0100 @@ -198,6 +198,13 @@ if (!(fmt->attributes & UAC_EP_CS_ATTR_PITCH_CONTROL)) return 0; + /* + * Denon DN-X 1600 doesn't properly handle this -> bail out, too + */ + + if (chip->usb_id == USB_ID(0x154e, 0x500e)) + return 0; + switch (altsd->bInterfaceProtocol) { case UAC_VERSION_1: default: cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.25. lsusb (...) Bus 002 Device 003: ID 154e:500e D&M Holdings, Inc. (Denon/Marantz) lsusb -t (...) /: Bus 02.Port 1: Dev 1, Class=root_hub, Driver=ehci_hcd/2p, 480M |__ Port 1: Dev 2, If 0, Class=hub, Driver=hub/6p, 480M |__ Port 2: Dev 3, If 0, Class=audio, Driver=snd-usb-audio, 480M |__ Port 2: Dev 3, If 1, Class=audio, Driver=snd-usb-audio, 480M |__ Port 2: Dev 3, If 2, Class=audio, Driver=snd-usb-audio, 480M |__ Port 2: Dev 3, If 3, Class=audio, Driver=snd-usb-audio, 480M |__ Port 2: Dev 3, If 4, Class=audio, Driver=snd-usb-audio, 480M |__ Port 2: Dev 3, If 5, Class=HID, Driver=usbhid, 480M (...) lsusb -v (...) Bus 002 Device 003: ID 154e:500e D&M Holdings, Inc. (Denon/Marantz) Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 239 Miscellaneous Device bDeviceSubClass 2 ? bDeviceProtocol 1 Interface Association bMaxPacketSize0 64 idVendor 0x154e D&M Holdings, Inc. (Denon/Marantz) idProduct 0x500e bcdDevice 0.01 iManufacturer 1 iProduct 2 iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 385 bNumInterfaces 6 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xc0 Self Powered MaxPower 0mA Interface Association: bLength 8 bDescriptorType 11 bFirstInterface 0 bInterfaceCount 3 bFunctionClass 255 Vendor Specific Class bFunctionSubClass 0 bFunctionProtocol 32 iFunction 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 32 iInterface 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 2.00 bCategory 0 wTotalLength 92 bmControl 0x01 Latency control Control (read-only) AudioControl Interface Descriptor: bLength 17 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bCSourceID 7 bNrChannels 8 bmChannelConfig 0x000000ff Front Left (FL) Front Right (FR) Front Center (FC) Low Frequency Effects (LFE) Back Left (BL) Back Right (BR) Front Left of Center (FLC) Front Right of Center (FRC) bmControls 0x0555 Copy Protect Control (read-only) Connector Control (read-only) Overload Control (read-only) Cluster Control (read-only) Underflow Control (read-only) Overflow Control (read-only) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 2 wTerminalType 0x0301 Speaker bAssocTerminal 0 bSourceID 1 bCSourceID 7 bmControls 0x0155 Copy Protect Control (read-only) Connector Control (read-only) Overload Control (read-only) Underflow Control (read-only) Overflow Control (read-only) iTerminal 0 AudioControl Interface Descriptor: bLength 17 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 3 wTerminalType 0x0201 Microphone bAssocTerminal 0 bCSourceID 7 bNrChannels 8 bmChannelConfig 0x000000ff Front Left (FL) Front Right (FR) Front Center (FC) Low Frequency Effects (LFE) Back Left (BL) Back Right (BR) Front Left of Center (FLC) Front Right of Center (FRC) bmControls 0x0555 Copy Protect Control (read-only) Connector Control (read-only) Overload Control (read-only) Cluster Control (read-only) Underflow Control (read-only) Overflow Control (read-only) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 4 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 3 bCSourceID 7 bmControls 0x0155 Copy Protect Control (read-only) Connector Control (read-only) Overload Control (read-only) Underflow Control (read-only) Overflow Control (read-only) iTerminal 0 AudioControl Interface Descriptor: bLength 8 bDescriptorType 36 bDescriptorSubtype 10 (CLOCK_SOURCE) bClockID 5 bmAttributes 0x02 Internal variable Clock bmControls 0x03 Clock Frequency Control (read/write) bAssocTerminal 0 iClockSource 0 AudioControl Interface Descriptor: bLength 8 bDescriptorType 36 bDescriptorSubtype 10 (CLOCK_SOURCE) bClockID 6 bmAttributes 0x02 Internal variable Clock bmControls 0x03 Clock Frequency Control (read/write) bAssocTerminal 0 iClockSource 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 11 (CLOCK_SELECTOR) bUnitID 7 bNrInPins 2 baCSourceID( 0) 5 baCSourceID( 1) 6 bmControls 0x01 Clock Selector Control (read-only) iClockSelector 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 0 AudioStreaming Interface Descriptor: bLength 16 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 1 bmControls 0x05 Active Alternate Setting Control (read-only) Valid Alternate Setting Control (read-only) bFormatType 1 bmFormats 1 PCM bNrChannels 8 bmChannelConfig 0x000000ff Front Left (FL) Front Right (FR) Front Center (FC) Low Frequency Effects (LFE) Back Left (BL) Back Right (BR) Front Left of Center (FLC) Front Right of Center (FRC) iChannelNames 0 AudioStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bSubslotSize 3 bBitResolution 24 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0138 1x 312 bytes bInterval 1 AudioControl Endpoint Descriptor: bLength 8 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bmControls 0x15 Pitch Control (read-only) Data Overrun Control (read-only) Data Underrun Control (read-only) bLockDelayUnits 1 Milliseconds wLockDelay 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 0 AudioStreaming Interface Descriptor: bLength 16 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 4 bmControls 0x05 Active Alternate Setting Control (read-only) Valid Alternate Setting Control (read-only) bFormatType 1 bmFormats 1 PCM bNrChannels 8 bmChannelConfig 0x000000ff Front Left (FL) Front Right (FR) Front Center (FC) Low Frequency Effects (LFE) Back Left (BL) Back Right (BR) Front Left of Center (FLC) Front Right of Center (FRC) iChannelNames 0 AudioStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bSubslotSize 3 bBitResolution 24 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x85 EP 5 IN bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0138 1x 312 bytes bInterval 1 AudioControl Endpoint Descriptor: bLength 8 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bmControls 0x15 Pitch Control (read-only) Data Overrun Control (read-only) Data Underrun Control (read-only) bLockDelayUnits 1 Milliseconds wLockDelay 0 Interface Association: bLength 8 bDescriptorType 11 bFirstInterface 3 bInterfaceCount 2 bFunctionClass 1 Audio bFunctionSubClass 0 bFunctionProtocol 0 iFunction 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 9 bInCollection 1 baInterfaceNr( 0) 4 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 4 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 1 Audio bInterfaceSubClass 3 MIDI Streaming bInterfaceProtocol 0 iInterface 0 MIDIStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 97 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 1 Embedded bJackID 1 iJack 0 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 1 Embedded bJackID 17 iJack 0 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 2 External bJackID 2 iJack 0 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 2 External bJackID 18 iJack 0 MIDIStreaming Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) bJackType 1 Embedded bJackID 3 bNrInputPins 1 baSourceID( 0) 2 BaSourcePin( 0) 1 iJack 0 MIDIStreaming Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) bJackType 1 Embedded bJackID 19 bNrInputPins 1 baSourceID( 0) 18 BaSourcePin( 0) 1 iJack 0 MIDIStreaming Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) bJackType 2 External bJackID 4 bNrInputPins 1 baSourceID( 0) 1 BaSourcePin( 0) 1 iJack 0 MIDIStreaming Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) bJackType 2 External bJackID 20 bNrInputPins 1 baSourceID( 0) 17 BaSourcePin( 0) 1 iJack 0 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0040 1x 64 bytes bInterval 0 bRefresh 0 bSynchAddress 0 MIDIStreaming Endpoint Descriptor: bLength 6 bDescriptorType 37 bDescriptorSubtype 1 (GENERAL) bNumEmbMIDIJack 2 baAssocJackID( 0) 1 baAssocJackID( 1) 17 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0040 1x 64 bytes bInterval 0 bRefresh 0 bSynchAddress 0 MIDIStreaming Endpoint Descriptor: bLength 6 bDescriptorType 37 bDescriptorSubtype 1 (GENERAL) bNumEmbMIDIJack 2 baAssocJackID( 0) 3 baAssocJackID( 1) 19 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 5 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 3 Human Interface Device bInterfaceSubClass 0 No Subclass bInterfaceProtocol 0 None iInterface 0 HID Device Descriptor: bLength 9 bDescriptorType 33 bcdHID 1.11 bCountryCode 0 Not supported bNumDescriptors 1 bDescriptorType 34 Report wDescriptorLength 34 Report Descriptors: ** UNAVAILABLE ** Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x84 EP 4 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0020 1x 32 bytes bInterval 1

1 0

[alsa-devel] [PATCH 2/2] Rename internal xonar_u1_* names to just xonar_u_*
by Denis Washington 11 Dec '12

11 Dec '12

Now that the code applies both to the U1 and the U3, xonar_u1 is not a good name anymore. Signed-off-by: Denis Washington <denisw(a)online.de> --- sound/usb/mixer.h | 2 +- sound/usb/mixer_quirks.c | 24 ++++++++++++------------ 2 files changed, 13 insertions(+), 13 deletions(-) diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index a7f3d45..9e7ec0e 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -22,7 +22,7 @@ struct usb_mixer_interface { u8 rc_buffer[6]; u8 audigy2nx_leds[3]; - u8 xonar_u1_status; + u8 xonar_u_status; }; #define MAX_CHANNELS 16 /* max logical channels */ diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 2223089..de6db70 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -416,23 +416,23 @@ static void snd_audigy2nx_proc_read(struct snd_info_entry *entry, } } -static int snd_xonar_u1_switch_get(struct snd_kcontrol *kcontrol, +static int snd_xonar_u_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); - ucontrol->value.integer.value[0] = !!(mixer->xonar_u1_status & 0x02); + ucontrol->value.integer.value[0] = !!(mixer->xonar_u_status & 0x02); return 0; } -static int snd_xonar_u1_switch_put(struct snd_kcontrol *kcontrol, +static int snd_xonar_u_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); u8 old_status, new_status; int err, changed; - old_status = mixer->xonar_u1_status; + old_status = mixer->xonar_u_status; if (ucontrol->value.integer.value[0]) new_status = old_status | 0x02; else @@ -449,27 +449,27 @@ static int snd_xonar_u1_switch_put(struct snd_kcontrol *kcontrol, up_read(&mixer->chip->shutdown_rwsem); if (err < 0) return err; - mixer->xonar_u1_status = new_status; + mixer->xonar_u_status = new_status; return changed; } -static struct snd_kcontrol_new snd_xonar_u1_output_switch = { +static struct snd_kcontrol_new snd_xonar_u_output_switch = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Digital Playback Switch", .info = snd_ctl_boolean_mono_info, - .get = snd_xonar_u1_switch_get, - .put = snd_xonar_u1_switch_put, + .get = snd_xonar_u_switch_get, + .put = snd_xonar_u_switch_put, }; -static int snd_xonar_u1_controls_create(struct usb_mixer_interface *mixer) +static int snd_xonar_u_controls_create(struct usb_mixer_interface *mixer) { int err; err = snd_ctl_add(mixer->chip->card, - snd_ctl_new1(&snd_xonar_u1_output_switch, mixer)); + snd_ctl_new1(&snd_xonar_u_output_switch, mixer)); if (err < 0) return err; - mixer->xonar_u1_status = 0x05; + mixer->xonar_u_status = 0x05; return 0; } @@ -1110,7 +1110,7 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x0b05, 0x1739): /* ASUS Xonar U1 */ case USB_ID(0x0b05, 0x1743): /* ASUS Xonar U1 (2) */ case USB_ID(0x0b05, 0x17a0): /* ASUS Xonar U3 */ - err = snd_xonar_u1_controls_create(mixer); + err = snd_xonar_u_controls_create(mixer); break; case USB_ID(0x17cc, 0x1011): /* Traktor Audio 6 */ -- 1.8.0

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[alsa-devel] [PATCH 1/2] Enable S/PDIF on the ASUS Xonar U3
by Denis Washington 11 Dec '12

11 Dec '12

The only required change is to extend the existing Xonar U1 mixer quirks to the U3, which seems to be controlled the same way. Signed-off-by: Denis Washington <denisw(a)online.de> --- sound/usb/mixer_quirks.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index ae2b714..2223089 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -1107,8 +1107,9 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) err = snd_ftu_create_mixer(mixer); break; - case USB_ID(0x0b05, 0x1739): - case USB_ID(0x0b05, 0x1743): + case USB_ID(0x0b05, 0x1739): /* ASUS Xonar U1 */ + case USB_ID(0x0b05, 0x1743): /* ASUS Xonar U1 (2) */ + case USB_ID(0x0b05, 0x17a0): /* ASUS Xonar U3 */ err = snd_xonar_u1_controls_create(mixer); break; -- 1.8.0

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Re: [alsa-devel] is it a memory bug?
by l9jj＠163.com 11 Dec '12

11 Dec '12

thanks a lot. 在 2012-12-10 19:41:58, Clemens Ladisch <clemens(a)ladisch.de>写道: l9jj(a)163.com wrote: > just call alloca() without free()? <http://linux.die.net/man/3/alloca> Regards, Clemens _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel l9jj(a)163.com ——网易邮箱手机客户端——

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[alsa-devel] how this will happen about hw_ptr_buffer_jiffies?
by l9jj 11 Dec '12

11 Dec '12

recently,i am tracing the process into the snd_pcm_update_hw_ptr0() and find the statement. i do not understand when the if (hdelta > runtime->hw_ptr_buffer_jiffies/2) (1)will happen? (2)and the reason? (3)the other question is if the dma interrupt will can enter the snd_pcm_update_hw_ptr0() every time when it ocurrs? thanks for any suggestion. Best Regards L9jj =====================code start==============================> if (in_interrupt) { /* we know that one period was processed */ /* delta = "expected next hw_ptr" for in_interrupt != 0 */ delta = runtime->hw_ptr_interrupt + runtime->period_size; if (delta > new_hw_ptr) { /* check for double acknowledged interrupts */ hdelta = jiffies - runtime->hw_ptr_jiffies; if (hdelta > runtime->hw_ptr_buffer_jiffies/2) { hw_base += runtime->buffer_size; if (hw_base >= runtime->boundary) hw_base = 0; new_hw_ptr = hw_base + pos; goto __delta; } } <=====================code end============================ -- ------------------------------------ l9jj Regards

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[alsa-devel] [PATCH 2/2] Rename internal xonar_u1_* names to just xonar_u_*
by Denis Washington 11 Dec '12

11 Dec '12

Now that the code applies both to the U1 and the U3, xonar_u1 is not a good name anymore. --- sound/usb/mixer.h | 2 +- sound/usb/mixer_quirks.c | 24 ++++++++++++------------ 2 files changed, 13 insertions(+), 13 deletions(-) diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index a7f3d45..9e7ec0e 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -22,7 +22,7 @@ struct usb_mixer_interface { u8 rc_buffer[6]; u8 audigy2nx_leds[3]; - u8 xonar_u1_status; + u8 xonar_u_status; }; #define MAX_CHANNELS 16 /* max logical channels */ diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 2223089..de6db70 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -416,23 +416,23 @@ static void snd_audigy2nx_proc_read(struct snd_info_entry *entry, } } -static int snd_xonar_u1_switch_get(struct snd_kcontrol *kcontrol, +static int snd_xonar_u_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); - ucontrol->value.integer.value[0] = !!(mixer->xonar_u1_status & 0x02); + ucontrol->value.integer.value[0] = !!(mixer->xonar_u_status & 0x02); return 0; } -static int snd_xonar_u1_switch_put(struct snd_kcontrol *kcontrol, +static int snd_xonar_u_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); u8 old_status, new_status; int err, changed; - old_status = mixer->xonar_u1_status; + old_status = mixer->xonar_u_status; if (ucontrol->value.integer.value[0]) new_status = old_status | 0x02; else @@ -449,27 +449,27 @@ static int snd_xonar_u1_switch_put(struct snd_kcontrol *kcontrol, up_read(&mixer->chip->shutdown_rwsem); if (err < 0) return err; - mixer->xonar_u1_status = new_status; + mixer->xonar_u_status = new_status; return changed; } -static struct snd_kcontrol_new snd_xonar_u1_output_switch = { +static struct snd_kcontrol_new snd_xonar_u_output_switch = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Digital Playback Switch", .info = snd_ctl_boolean_mono_info, - .get = snd_xonar_u1_switch_get, - .put = snd_xonar_u1_switch_put, + .get = snd_xonar_u_switch_get, + .put = snd_xonar_u_switch_put, }; -static int snd_xonar_u1_controls_create(struct usb_mixer_interface *mixer) +static int snd_xonar_u_controls_create(struct usb_mixer_interface *mixer) { int err; err = snd_ctl_add(mixer->chip->card, - snd_ctl_new1(&snd_xonar_u1_output_switch, mixer)); + snd_ctl_new1(&snd_xonar_u_output_switch, mixer)); if (err < 0) return err; - mixer->xonar_u1_status = 0x05; + mixer->xonar_u_status = 0x05; return 0; } @@ -1110,7 +1110,7 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x0b05, 0x1739): /* ASUS Xonar U1 */ case USB_ID(0x0b05, 0x1743): /* ASUS Xonar U1 (2) */ case USB_ID(0x0b05, 0x17a0): /* ASUS Xonar U3 */ - err = snd_xonar_u1_controls_create(mixer); + err = snd_xonar_u_controls_create(mixer); break; case USB_ID(0x17cc, 0x1011): /* Traktor Audio 6 */ -- 1.8.0

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[alsa-devel] [PATCH 1/2] Enable S/PDIF on the ASUS Xonar U3
by Denis Washington 11 Dec '12

11 Dec '12

The only required change is to extend the existing Xonar U1 mixer quirks to the U3, which seems to be controlled the same way. --- sound/usb/mixer_quirks.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index ae2b714..2223089 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -1107,8 +1107,9 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) err = snd_ftu_create_mixer(mixer); break; - case USB_ID(0x0b05, 0x1739): - case USB_ID(0x0b05, 0x1743): + case USB_ID(0x0b05, 0x1739): /* ASUS Xonar U1 */ + case USB_ID(0x0b05, 0x1743): /* ASUS Xonar U1 (2) */ + case USB_ID(0x0b05, 0x17a0): /* ASUS Xonar U3 */ err = snd_xonar_u1_controls_create(mixer); break; -- 1.8.0

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Re: [alsa-devel] [PATCH V4 0/2] Add DT support for i2s
by Padma Venkat 10 Dec '12

10 Dec '12

Hi Mark, On Wed, Nov 28, 2012 at 4:17 PM, Padmavathi Venna <padma.v(a)samsung.com> wrote: > V4 patches are based on Mark Brown's for-next branch of > "git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git" > > Changes since V3: > - Added syntex of gpio specifier as suggested by Mark Brown in > DT document file > - Added the requirement for having I2S alias ids > - Explained about gpio lines > - Made a seperate patch to get the I2S rclk src clk from alias ID > > Changes since V2: > - Rebased on 3.7-rc3 > - Custom DT bindings are prefixed with samsung > - As generic device tree DMA helpers not yet mainlined > I am still using custom dma bindings. So added a > priliminary statement regarding the same. I will rework > on my patch once generic DMA helpers are mainlined. > > Chnages since V1: > - Rebased on 3.6-rc6 > > Padmavathi Venna (2): > ASoC: Samsung: Get I2S src_clk from clock alias id. > ASoC: SAMSUNG: Add DT support for i2s Any comments on DT support for i2s patch? If not when this patch can be pulled? > > .../devicetree/bindings/sound/samsung-i2s.txt | 78 +++++++ > include/linux/platform_data/asoc-s3c.h | 6 - > sound/soc/samsung/dma.c | 1 + > sound/soc/samsung/dma.h | 1 + > sound/soc/samsung/i2s.c | 238 +++++++++++++++---- > 5 files changed, 268 insertions(+), 56 deletions(-) > create mode 100644 Documentation/devicetree/bindings/sound/samsung-i2s.txt > > -- > 1.7.4.4 > Thanks Padma

3 6

[alsa-devel] is it a memory bug?
by l9jj＠163.com 10 Dec '12

10 Dec '12

project: alsa-lib-1.0.25 file: dlmisc.c function: snd_dlsym_verify() description: the pointer vname hold a memory until exit the function without release.just call alloca() without free()？ l9jj(a)163.com ——网易邮箱手机客户端——

2 1

Re: [alsa-devel] [PATCH 2/2]: ALSA: VIA HDA: Add support for codec VT180
by Takashi Iwai 10 Dec '12

10 Dec '12

At Mon, 10 Dec 2012 19:03:26 +0800, Lydia wrote: > > From: Lydia Wang <lydiawang(a)viatech.com.cn> > Subject: ALSA: VIA HDA: Add support for codec VT1808. > > Add support for new codec VT1808, which is similiar with VT1705CF. > > Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Thanks, applied both patches now. Takashi > Index: sound/sound/pci/hda/patch_via.c > =================================================================== > --- sound.orig/sound/pci/hda/patch_via.c 2012-12-10 15:09:52.000000000 +0800 > +++ sound/sound/pci/hda/patch_via.c 2012-12-10 15:12:50.000000000 +0800 > @@ -77,6 +77,7 @@ > VT1812, > VT1802, > VT1705CF, > + VT1808, > CODEC_TYPES, > }; > > @@ -300,6 +301,8 @@ > codec_type = VT1802; > else if (dev_id == 0x4760) > codec_type = VT1705CF; > + else if (dev_id == 0x4761 || dev_id == 0x4762) > + codec_type = VT1808; > else > codec_type = UNKNOWN; > return codec_type; > @@ -1112,6 +1115,7 @@ > parm = enable ? 0x00 : 0xe0; /* 0x00: 4/40x, 0xe0: 1x */ > break; > case VT1705CF: > + case VT1808: > verb = 0xf82; > parm = enable ? 0x00 : 0xe0; /* 0x00: 4/40x, 0xe0: 1x */ > break; > @@ -4061,6 +4065,10 @@ > .patch = patch_vt2002P}, > { .id = 0x11064760, .name = "VT1705CF", > .patch = patch_vt3476}, > + { .id = 0x11064761, .name = "VT1708SCE", > + .patch = patch_vt3476}, > + { .id = 0x11064762, .name = "VT1808", > + .patch = patch_vt3476}, > {} /* terminator */ > }; > > > >

1 0

[alsa-devel] [PATCH 0/7] ALSA: CA0132: DSP Update
by Ian Minett 10 Dec '12

10 Dec '12

From: Ian Minett <ian_minett(a)creativelabs.com> Hi Takashi, thanks for the info about the endianness. If the controller natively swaps each element as it gets read/written to the DSP then it sounds like it should work as is... So with that in mind, we'd like to carry on with the DSP updates, if that's ok. The first batch is included in this patch series. 1: Add new definitions that will be used by the DSP routines 2: Add the new DSP mixer, switches and controls. This is quite large, but it's a little difficult to break it down further - hope it's ok. 3: Add code for handling data returned from the DSP. 4: Add support for tuning controls and data 5/6: Code tidy, split into two for a cleaner diff. 7: Updates to the mic sample rate, module description and memalloc error handling. Thanks very much, please let us know if we need to alter anything. - Ian Signed-off-by: Ian Minett <ian_minett(a)creativelabs.com>

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Re: [alsa-devel] what is period_size, buffer_size and boundary size, i confuse them
by l9jj＠163.com 10 Dec '12

10 Dec '12

thank you very much for your help. i got another question in the interface snd_pcm_update_hw_ptr0(). The reference code is below. (1)i know that the result of representation (hdelta * HZ) / runtime->rate means the delta jiffies which spend on moving the data from the new_hw_ptr to the old_hw_ptr in the dma buffer. (2)and the jdelta represents the interval jiffies between now and last time enter here. (3)the question is why can we think there is more than one interrupt has lost across comparing the two variables size? (4)and if the compare idear setup that means one interrupt will take jdelta jiffies in time,so if someone is bigger it we can know that.is it true? (5)if the (4) is correct, why do not use the representation (runtime->period_size * HZ) / runtime->rate) as the interrupt interval, because it is fixed and the jdelta is not. (6)why plus the HZ/100 after jdelta and (runtime->period_size * HZ) / runtime->rate) ? for ensuring have enought time? HZ ==> 1 s HZ/1000 ===> 1 ms HZ/100 ===> 10 ms ? =================================reference code start ====================================> /* Skip the jiffies check for hardwares with BATCH flag. * Such hardware usually just increases the position at each IRQ, * thus it can't give any strange position. */if (runtime->hw.info & SNDRV_PCM_INFO_BATCH) goto no_jiffies_check; hdelta = delta; if (hdelta < runtime->delay) goto no_jiffies_check; hdelta -= runtime->delay; jdelta = jiffies - runtime->hw_ptr_jiffies; if (((hdelta * HZ) / runtime->rate) > jdelta + HZ/100) { delta = jdelta /(((runtime->period_size * HZ) / runtime->rate) + HZ/100); /* move new_hw_ptr according jiffies not pos variable */new_hw_ptr = old_hw_ptr; hw_base = delta; /* use loop to avoid checks for delta overflows *//* the delta value is small or zero in most cases */while (delta > 0) { new_hw_ptr += runtime->period_size; if (new_hw_ptr >= runtime->boundary) new_hw_ptr -= runtime->boundary; delta--; } /* align hw_base to buffer_size */hw_ptr_error(substream, "hw_ptr skipping! %s" "(pos=%ld, delta=%ld, period=%ld, " "jdelta=%lu/%lu/%lu, hw_ptr=%ld/%ld)\n", in_interrupt ? "[Q] " : "", (long)pos, (long)hdelta, (long)runtime->period_size, jdelta, ((hdelta * HZ) / runtime->rate), hw_base, (unsigned long)old_hw_ptr, (unsigned long)new_hw_ptr); /* reset values to proper state */delta = 0; hw_base = new_hw_ptr - (new_hw_ptr % runtime->buffer_size); } <======================================end =================================================== Best Regards. 在 2012-12-10 16:24:59, Clemens Ladisch <clemens(a)ladisch.de>写道: l9jj(a)163.com wrote: > does the variable buffer_size equal to DMA size? Yes. > (2)how the pos>buffer_size will happen in the interface snd_pcm_update_hw_ptr0()? It happens when your driver has a bug. The .pointer callback always must return value that is inside the DMA buffer. Regards, Clemens _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel l9jj(a)163.com ——网易邮箱手机客户端——

1 0

[alsa-devel] [PATCH 0/2]: ALSA: VIA HDA: patch series
by lydiawang 10 Dec '12

10 Dec '12

Following patch series mainly includes: -Support codec VT1705CF. -Support codec VT1808.

2 1

Re: [alsa-devel] what is period_size, buffer_size and boundary size, i confuse them
by l9jj＠163.com 10 Dec '12

10 Dec '12

thank you very much. but, （1）what is the relationship between DMA buffer size and the boundary size？does the variable buffer_size equal to DMA size？ (2)how the pos>buffer_size will happen in the interface snd_pcm_update_hw_ptr0()？ the variable pos is in the range of DMA_size，it can not bigger than the buffer_size.it blocked me. thanks a lot. 在 2012-12-10 00:23:55, Clemens Ladisch <clemens(a)ladisch.de>写道: l9jj wrote: > can anyone help me to figure out the concepts between period_size ,buffer_size and boundary size? > n * period_size = buffer_size? Yes. On many devices, n is not required to be an integer. > m * buffer_size = boundary_size? Yes, where m is very big. In applications, the actual value of the boundary size does not matter (except as a special value for some sw params). > and how the dma buffer mapped to the alsa buffer? The DMA buffer _is_ the ALSA buffer. Regards, Clemens _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel l9jj(a)163.com ——网易邮箱手机客户端——

2 1

[alsa-devel] [PATCH 2/2]: ALSA: VIA HDA: Add support for codec VT1808
by lydiawang 10 Dec '12

10 Dec '12

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Add support for codec VT1808. Add support for new codec VT1808, which is similiar with VT1705CF. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound/sound/pci/hda/patch_via.c =================================================================== --- sound.orig/sound/pci/hda/patch_via.c 2012-12-10 15:09:52.000000000 +0800 +++ sound/sound/pci/hda/patch_via.c 2012-12-10 15:12:50.000000000 +0800 @@ -77,6 +77,7 @@ VT1812, VT1802, VT1705CF, + VT1808, CODEC_TYPES, }; @@ -300,6 +301,8 @@ codec_type = VT1802; else if (dev_id == 0x4760) codec_type = VT1705CF; + else if (dev_id == 0x4761 || dev_id == 0x4762) + codec_type = VT1808; else codec_type = UNKNOWN; return codec_type; @@ -1112,6 +1115,7 @@ parm = enable ? 0x00 : 0xe0; /* 0x00: 4/40x, 0xe0: 1x */ break; case VT1705CF: + case VT1808: verb = 0xf82; parm = enable ? 0x00 : 0xe0; /* 0x00: 4/40x, 0xe0: 1x */ break; @@ -4061,6 +4065,10 @@ .patch = patch_vt2002P}, { .id = 0x11064760, .name = "VT1705CF", .patch = patch_vt3476}, + { .id = 0x11064761, .name = "VT1708SCE", + .patch = patch_vt3476}, + { .id = 0x11064762, .name = "VT1808", + .patch = patch_vt3476}, {} /* terminator */ };

1 0

[alsa-devel] [PATCH 1/2]: ALSA: VIA HDA: Add support for codec VT1705CF
by lydiawang 10 Dec '12

10 Dec '12

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Add support for codec VT1705CF. Add support for new codec VT1705CF. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound/sound/pci/hda/patch_via.c =================================================================== --- sound.orig/sound/pci/hda/patch_via.c 2012-12-10 14:22:56.000000000 +0800 +++ sound/sound/pci/hda/patch_via.c 2012-12-10 15:02:02.000000000 +0800 @@ -76,6 +76,7 @@ VT2002P, VT1812, VT1802, + VT1705CF, CODEC_TYPES, }; @@ -220,6 +221,7 @@ int vt1708_hp_present; void (*set_widgets_power_state)(struct hda_codec *codec); + unsigned int dac_stream_tag[4]; struct hda_loopback_check loopback; int num_loopbacks; @@ -296,6 +298,8 @@ codec_type = VT1708S; else if ((dev_id & 0xfff) == 0x446) codec_type = VT1802; + else if (dev_id == 0x4760) + codec_type = VT1705CF; else codec_type = UNKNOWN; return codec_type; @@ -711,6 +715,28 @@ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, parm); } +static void update_conv_power_state(struct hda_codec *codec, hda_nid_t nid, + unsigned int parm, unsigned int index) +{ + struct via_spec *spec = codec->spec; + unsigned int format; + if (snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_POWER_STATE, 0) == parm) + return; + format = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0); + if (format && (spec->dac_stream_tag[index] != format)) + spec->dac_stream_tag[index] = format; + + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, parm); + if (parm == AC_PWRST_D0) { + format = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0); + if (!format && (spec->dac_stream_tag[index] != format)) + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_CHANNEL_STREAMID, + spec->dac_stream_tag[index]); + } +} + static void set_pin_power_state(struct hda_codec *codec, hda_nid_t nid, unsigned int *affected_parm) { @@ -1085,6 +1111,10 @@ verb = 0xf93; parm = enable ? 0x00 : 0xe0; /* 0x00: 4/40x, 0xe0: 1x */ break; + case VT1705CF: + verb = 0xf82; + parm = enable ? 0x00 : 0xe0; /* 0x00: 4/40x, 0xe0: 1x */ + break; default: return; /* other codecs are not supported */ } @@ -3817,6 +3847,125 @@ return 0; } +/* patch for vt3476 */ + +static const struct hda_verb vt3476_init_verbs[] = { + /* Enable DMic 8/16/32K */ + {0x1, 0xF7B, 0x30}, + /* Enable Boost Volume backdoor */ + {0x1, 0xFB9, 0x20}, + /* Enable AOW-MW9 path */ + {0x1, 0xFB8, 0x10}, + { } +}; + +static void set_widgets_power_state_vt3476(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int imux_is_smixer; + unsigned int parm, parm2; + /* MUX10 (1eh) = stereo mixer */ + imux_is_smixer = + snd_hda_codec_read(codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 4; + /* inputs */ + /* PW 5/6/7 (29h/2ah/2bh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x29, &parm); + set_pin_power_state(codec, 0x2a, &parm); + set_pin_power_state(codec, 0x2b, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; + /* MUX10/11 (1eh/1fh), AIW 0/1 (10h/11h) */ + update_power_state(codec, 0x1e, parm); + update_power_state(codec, 0x1f, parm); + update_power_state(codec, 0x10, parm); + update_power_state(codec, 0x11, parm); + + /* outputs */ + /* PW3 (27h), MW3(37h), AOW3 (bh) */ + if (spec->codec_type == VT1705CF) { + parm = AC_PWRST_D3; + update_power_state(codec, 0x27, parm); + update_power_state(codec, 0x37, parm); + } else { + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x27, &parm); + update_power_state(codec, 0x37, parm); + } + + /* PW2 (26h), MW2(36h), AOW2 (ah) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x26, &parm); + update_power_state(codec, 0x36, parm); + if (spec->smart51_enabled) { + /* PW7(2bh), MW7(3bh), MUX7(1Bh) */ + set_pin_power_state(codec, 0x2b, &parm); + update_power_state(codec, 0x3b, parm); + update_power_state(codec, 0x1b, parm); + } + update_conv_power_state(codec, 0xa, parm, 2); + + /* PW1 (25h), MW1(35h), AOW1 (9h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x25, &parm); + update_power_state(codec, 0x35, parm); + if (spec->smart51_enabled) { + /* PW6(2ah), MW6(3ah), MUX6(1ah) */ + set_pin_power_state(codec, 0x2a, &parm); + update_power_state(codec, 0x3a, parm); + update_power_state(codec, 0x1a, parm); + } + update_conv_power_state(codec, 0x9, parm, 1); + + /* PW4 (28h), MW4 (38h), MUX4(18h), AOW3(bh)/AOW0(8h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x28, &parm); + update_power_state(codec, 0x38, parm); + update_power_state(codec, 0x18, parm); + if (spec->hp_independent_mode) + update_conv_power_state(codec, 0xb, parm, 3); + parm2 = parm; /* for pin 0x0b */ + + /* PW0 (24h), MW0(34h), MW9(3fh), AOW0 (8h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x24, &parm); + update_power_state(codec, 0x34, parm); + if (!spec->hp_independent_mode && parm2 != AC_PWRST_D3) + parm = parm2; + update_conv_power_state(codec, 0x8, parm, 0); + /* MW9 (21h), Mw2 (1ah), AOW0 (8h) */ + update_power_state(codec, 0x3f, imux_is_smixer ? AC_PWRST_D0 : parm); +} + +static int patch_vt3476(struct hda_codec *codec) +{ + struct via_spec *spec; + int err; + + /* create a codec specific record */ + spec = via_new_spec(codec); + if (spec == NULL) + return -ENOMEM; + + spec->aa_mix_nid = 0x3f; + add_secret_dac_path(codec); + + /* automatic parse from the BIOS config */ + err = via_parse_auto_config(codec); + if (err < 0) { + via_free(codec); + return err; + } + + spec->init_verbs[spec->num_iverbs++] = vt3476_init_verbs; + + codec->patch_ops = via_patch_ops; + + spec->set_widgets_power_state = set_widgets_power_state_vt3476; + + return 0; +} + /* * patch entries */ @@ -3910,6 +4059,8 @@ .patch = patch_vt2002P}, { .id = 0x11068446, .name = "VT1802", .patch = patch_vt2002P}, + { .id = 0x11064760, .name = "VT1705CF", + .patch = patch_vt3476}, {} /* terminator */ };

1 0

[alsa-devel] [PATCH 3/3] sound/soc/soc-core.c: drop kfree of devm_kzalloc's data
by Julia Lawall 10 Dec '12

10 Dec '12

From: Julia Lawall <Julia.Lawall(a)lip6.fr> Using kfree to free data allocated with devm_kzalloc causes double frees. The semantic patch that fixes this problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> @@ expression x; @@ x = devm_kzalloc(...) ... ?-kfree(x); // </smpl> Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> --- The kfrees were introduced in b761c0ca. I sent this a few months ago, and I still think it should be applied... sound/soc/soc-core.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f81c597..86d7a36 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4136,7 +4136,6 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, dev_err(card->dev, "Property '%s' index %d could not be read: %d\n", propname, 2 * i, ret); - kfree(routes); return -EINVAL; } ret = of_property_read_string_index(np, propname, @@ -4145,7 +4144,6 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, dev_err(card->dev, "Property '%s' index %d could not be read: %d\n", propname, (2 * i) + 1, ret); - kfree(routes); return -EINVAL; } }

2 4

[alsa-devel] [PATCH] ASoC: adsp: Correct version number masks
by Mark Brown 10 Dec '12

10 Dec '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 3f0e69b..00caaa5 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -394,7 +394,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) algs = be32_to_cpu(adsp1_id.algs); adsp_info(dsp, "Firmware: %x v%d.%d.%d, %zu algorithms\n", be32_to_cpu(adsp1_id.fw.id), - (be32_to_cpu(adsp1_id.fw.ver) & 0xff000) >> 16, + (be32_to_cpu(adsp1_id.fw.ver) & 0xff0000) >> 16, (be32_to_cpu(adsp1_id.fw.ver) & 0xff00) >> 8, be32_to_cpu(adsp1_id.fw.ver) & 0xff, algs); @@ -415,7 +415,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) algs = be32_to_cpu(adsp2_id.algs); adsp_info(dsp, "Firmware: %x v%d.%d.%d, %zu algorithms\n", be32_to_cpu(adsp2_id.fw.id), - (be32_to_cpu(adsp2_id.fw.ver) & 0xff000) >> 16, + (be32_to_cpu(adsp2_id.fw.ver) & 0xff0000) >> 16, (be32_to_cpu(adsp2_id.fw.ver) & 0xff00) >> 8, be32_to_cpu(adsp2_id.fw.ver) & 0xff, algs); @@ -465,7 +465,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) case WMFW_ADSP1: adsp_info(dsp, "%d: ID %x v%d.%d.%d\n", i, be32_to_cpu(adsp1_alg[i].alg.id), - (be32_to_cpu(adsp1_alg[i].alg.ver) & 0xff000) >> 16, + (be32_to_cpu(adsp1_alg[i].alg.ver) & 0xff0000) >> 16, (be32_to_cpu(adsp1_alg[i].alg.ver) & 0xff00) >> 8, be32_to_cpu(adsp1_alg[i].alg.ver) & 0xff); break; @@ -473,7 +473,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) case WMFW_ADSP2: adsp_info(dsp, "%d: ID %x v%d.%d.%d\n", i, be32_to_cpu(adsp2_alg[i].alg.id), - (be32_to_cpu(adsp2_alg[i].alg.ver) & 0xff000) >> 16, + (be32_to_cpu(adsp2_alg[i].alg.ver) & 0xff0000) >> 16, (be32_to_cpu(adsp2_alg[i].alg.ver) & 0xff00) >> 8, be32_to_cpu(adsp2_alg[i].alg.ver) & 0xff); break; -- 1.7.10.4

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[alsa-devel] [PATCH 1/3] ASoC: cs42l73: Add DMIC's as DAPM inputs.
by Paul Handrigan 10 Dec '12

10 Dec '12

Signed-off-by: Paul Handrigan <Paul.Handrigan(a)cirrus.com> --- sound/soc/codecs/cs42l73.c | 4 ++++ 1 files changed, 4 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 2c08c4c..4766795 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -589,6 +589,8 @@ static const struct snd_kcontrol_new cs42l73_snd_controls[] = { }; static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("DMICA"), + SND_SOC_DAPM_INPUT("DMICB"), SND_SOC_DAPM_INPUT("LINEINA"), SND_SOC_DAPM_INPUT("LINEINB"), SND_SOC_DAPM_INPUT("MIC1"), @@ -795,6 +797,8 @@ static const struct snd_soc_dapm_route cs42l73_audio_map[] = { {"ADC Left", NULL, "PGA Left"}, {"ADC Right", NULL, "PGA Right"}, + {"DMIC Left", NULL, "DMICA"}, + {"DMIC Right", NULL, "DMICB"}, {"Input Left Capture", "ADC Left Input", "ADC Left"}, {"Input Right Capture", "ADC Right Input", "ADC Right"}, -- 1.7.0.4

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[alsa-devel] [PATCH] ASoC: sigmadsp: Fix endianness conversion issue
by Lars-Peter Clausen 10 Dec '12

10 Dec '12

The 'addr' field of the sigma_action struct is stored as big endian in the firmware file. Cc: stable(a)vger.kernel.org Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/sigmadsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/sigmadsp.c b/sound/soc/codecs/sigmadsp.c index 5be42bf..4068f24 100644 --- a/sound/soc/codecs/sigmadsp.c +++ b/sound/soc/codecs/sigmadsp.c @@ -225,7 +225,7 @@ EXPORT_SYMBOL(process_sigma_firmware); static int sigma_action_write_regmap(void *control_data, const struct sigma_action *sa, size_t len) { - return regmap_raw_write(control_data, le16_to_cpu(sa->addr), + return regmap_raw_write(control_data, be16_to_cpu(sa->addr), sa->payload, len - 2); } -- 1.8.0

2 1

[alsa-devel] [PATCH] ASoC: pcm: allow backend hardware to be freed in pause state
by Patrick Lai 10 Dec '12

10 Dec '12

When front-end PCM session is in paused state, back-end PCM session will be put in paused state as well if given front-end PCM session is the only client of given back-end. Then, application closes front-end PCM session, DPCM framework will not allow back-end enters HW_FREE state so back-end will never get shutdown completely. Signed-off-by: Patrick Lai <plai(a)codeaurora.org> --- sound/soc/soc-pcm.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index ef22d0b..d340644 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1240,7 +1240,7 @@ static int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) continue; -- The Qualcomm Innovation Center, Inc. is a member of Code Aurora Forum, hosted by The Linux Foundation 1.7.8.3

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[alsa-devel] what is period_size, buffer_size and boundary size, i confuse them
by l9jj 09 Dec '12

09 Dec '12

can anyone help me to figure out the concepts between period_size ,buffer_size and boundary size? n * period_size = buffer_size? m * buffer_size = boundary_size? and how the dma buffer mapped to the alsa buffer? thank you . br

2 1

Re: [alsa-devel] [PATCH 00/25] ASoC: remove __dev* attributes for retirement of HOTPLUG
by Mark Brown 09 Dec '12

09 Dec '12

On Fri, Dec 07, 2012 at 09:26:12AM -0500, Bill Pemberton wrote: > Remove use of __devinit, __devexit_p, __devinitdata, __devinitconst, > and __devexit so that HOTPLUG can be retired. Applied all, thanks.

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[alsa-devel] ALSA sound/usb/mixer.c:415 cannot get current value for control 2 ch 1: err = -22
by Joe Cooper 09 Dec '12

09 Dec '12

On most HP Envy laptops the snd-usb-audio module causes the system to become unresponsive and Gnome Shell 3 to crash. This effects every kernel I've tried from 3.3.4 on up to 3.7rc8. It also produces these errors on the console: ALSA sound/usb/mixer.c:415 cannot get current value for control 2 ch 1: err = -22 ALSA sound/usb/clock.c:228 3:1:1: cannot set freq 44100 to ep 0x1 Blacklisting this module allows the system to function, but with most of its speakers not working. It also prevents use of other USB audio devices, making multitrack recording impossible. I have a ticket about it in the Fedora bug tracker, but was referred to alsa-devel. That ticket links to other discussions of this bug, as well. Here's that ticket: https://bugzilla.redhat.com/show_bug.cgi?id=884368 alsa-info is here: https://bugzilla.redhat.com/attachment.cgi?id=658989 Please let me know if you need further details, or if there's anything else I can do to help troubleshoot this.

2 1

[alsa-devel] [FT C400,PATCH RFC,v4 00/10] M-Audio Fast Track C400
by Eldad Zack 09 Dec '12

09 Dec '12

Hi, The following patches adds support for the M-Audio Fast Track C400. This is version 4 of the patch series. * Incoroprated the feedback from Clemens Ladisch <clemens(a)ladisch.de>: Moved the quirks structure to the right place (also in other places, though not strictly required) and removed the "set quirks" log messages. Thanks, Clemens! * Added a patch to address the regression introduced by commit 947d299686aa9cc8aecf749d54e8475c6e498956 , "ALSA: snd-usb: properly initialize the sync endpoint". I believe there are no other devices that affected by this issue. Opening only the playback stream without first opening the capture stream now works with the C400. v3: http://mailman.alsa-project.org/pipermail/alsa-devel/2012-November/057477.h… v2: http://mailman.alsa-project.org/pipermail/alsa-devel/2012-November/056750.h… This series was tested with the latest mainline tree, 3.7-rc7 (HEAD e23739b4ade80a3a7f87198f008f6c44a7cbc9fd ) Also applies against current sound-unstable (not tested). (HEAD 900b0690ee7b2a227c5db5f1642ceb67d90c49fc ) * Everything works: mixer controls, effects control, playback and capture (with the correct sampling frequency set). Note that the device doesn't really have a master control or left/right pan. It's just "emulated" in the software. * To deal with the mixer channel IDs, limited by cmask bit length, I introduced an offset value. This allows to get rid of the shorehorning of the IDs into cmask and device-referencing in the low-level functions. Is this a good approach? * Audio in/out seems to work good in this series, mainly because the playback is an implicit feedback endpoint, and it is now configured with the capture endpoint as its sync-buddy. Also needed to correct the calculation, because of the different channel numbers between the two endpoints on the C400 (6 playback - implicit feedback, 4 capture). * The clock sources are still named "Unit 129" (0x81, internal) and "Unit 130" (0x82, SPDIF). I'll fix this later. I also plan to fix the channel names (i.e., instead of "AIn3", use "SPDIF In L"). * I've touched one FTU mixer creation function. I'd appreciate if someone can test this with the FTU to make sure I didn't break something. (I haven't gotten any feedback on that one yet) Cheers, Eldad Signed-off-by: Eldad Zack <eldad(a)fogrefinery.com> Eldad Zack (10): usb-audio: replace hardcoded value with const usb-audio: correct sync endpoint initialization usb-audio: use sender stride for implicit feedback usb-audio: add control index offset usb-audio: skip UAC2 EFFECT_UNIT usb-audio: parameterize FTU effect unit control usb-audio: M-Audio Fast Track C400 quirks table usb-audio: Fast Track C400 mixer ranges usb-audio: Fast Track C400 mixer controls usb-audio: FT C400 sync playback EP to capture EP sound/usb/endpoint.c | 9 ++- sound/usb/mixer.c | 44 +++++++++- sound/usb/mixer.h | 1 + sound/usb/mixer_quirks.c | 216 ++++++++++++++++++++++++++++++++++++++++++++-- sound/usb/pcm.c | 94 ++++++++++++++++++-- sound/usb/quirks-table.h | 71 +++++++++++++++ 6 files changed, 414 insertions(+), 21 deletions(-) -- 1.7.8.6

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[alsa-devel] [asoc:topic/arizona 34/34] ERROR: "arizona_out_vi_ramp" [sound/soc/codecs/snd-soc-wm5110.ko] undefined!
by kbuild test robot 09 Dec '12

09 Dec '12

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/arizona head: 8c3e248639a0a1b15b3b245f1e2ec05150f544d5 commit: 8c3e248639a0a1b15b3b245f1e2ec05150f544d5 [34/34] ASoC: wm5110: Enable volume ramp control config: x86_64-randconfig-h424 (attached as .config) All error/warnings: ERROR: "arizona_out_vi_ramp" [sound/soc/codecs/snd-soc-wm5110.ko] undefined! ERROR: "arizona_in_vi_ramp" [sound/soc/codecs/snd-soc-wm5110.ko] undefined! ERROR: "arizona_out_vd_ramp" [sound/soc/codecs/snd-soc-wm5110.ko] undefined! ERROR: "arizona_in_vd_ramp" [sound/soc/codecs/snd-soc-wm5110.ko] undefined! --- 0-DAY kernel build testing backend Open Source Technology Center Fengguang Wu, Yuanhan Liu Intel Corporation

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[alsa-devel] [PATCH 1/3] ASoC: arizona: Add volume ramp controls
by Mark Brown 09 Dec '12

09 Dec '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 25 +++++++++++++++++++++++++ sound/soc/codecs/arizona.h | 6 ++++++ 2 files changed, 31 insertions(+) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index d5385dc..a377d1f 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -274,6 +274,31 @@ EXPORT_SYMBOL_GPL(arizona_mixer_values); const DECLARE_TLV_DB_SCALE(arizona_mixer_tlv, -3200, 100, 0); EXPORT_SYMBOL_GPL(arizona_mixer_tlv); +static const char *arizona_vol_ramp_text[] = { + "0ms/6dB", "0.5ms/6dB", "1ms/6dB", "2ms/6dB", "4ms/6dB", "8ms/6dB", + "15ms/6dB", "30ms/6dB", +}; + +const struct soc_enum arizona_in_vd_ramp = + SOC_ENUM_SINGLE(ARIZONA_INPUT_VOLUME_RAMP, + ARIZONA_IN_VD_RAMP_SHIFT, 7, arizona_vol_ramp_text); +EXPORT_SYMBOL_GPL(arizona_in_vd_ramp); + +const struct soc_enum arizona_in_vi_ramp = + SOC_ENUM_SINGLE(ARIZONA_INPUT_VOLUME_RAMP, + ARIZONA_IN_VI_RAMP_SHIFT, 7, arizona_vol_ramp_text); +EXPORT_SYMBOL_GPL(arizona_in_vi_ramp); + +const struct soc_enum arizona_out_vd_ramp = + SOC_ENUM_SINGLE(ARIZONA_OUTPUT_VOLUME_RAMP, + ARIZONA_OUT_VD_RAMP_SHIFT, 7, arizona_vol_ramp_text); +EXPORT_SYMBOL_GPL(arizona_out_vd_ramp); + +const struct soc_enum arizona_out_vi_ramp = + SOC_ENUM_SINGLE(ARIZONA_OUTPUT_VOLUME_RAMP, + ARIZONA_OUT_VI_RAMP_SHIFT, 7, arizona_vol_ramp_text); +EXPORT_SYMBOL_GPL(arizona_out_vi_ramp); + static const char *arizona_lhpf_mode_text[] = { "Low-pass", "High-pass" }; diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 24b9692..4f09bb5 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -165,6 +165,12 @@ extern int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS]; ARIZONA_MIXER_ROUTES(name, name "L"), \ ARIZONA_MIXER_ROUTES(name, name "R") +extern const struct soc_enum arizona_in_vi_ramp; +extern const struct soc_enum arizona_in_vd_ramp; + +extern const struct soc_enum arizona_out_vi_ramp; +extern const struct soc_enum arizona_out_vd_ramp; + extern const struct soc_enum arizona_lhpf1_mode; extern const struct soc_enum arizona_lhpf2_mode; extern const struct soc_enum arizona_lhpf3_mode; -- 1.7.10.4

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Re: [alsa-devel] [PATCH] ASoC: adsp: Set DSP clock less than SYSCLK at initialisation
by Mark Brown 09 Dec '12

09 Dec '12

On Fri, Dec 07, 2012 at 01:55:22PM +0000, Chris Rattray wrote: > In some configurations it is necessary to set the ADSP core clock > to less than the SYSCLK rate in order for the ADSP RAM to initialise > successfully. Once the firmware image is downloaded, the ADSP core > clock is then brought back up to the SYSCLK rate. As followed up offline this is a system-specific issue you were running into that this happened to work around. > Signed-off-by: Chris Rattray <crattray(a)opensource.wolfsonmicro.com> > Change-Id: I14412a301dcf1c0a6a728fa8c12d23715e944cd4 Don't include noise like Change-id that's specific to internal systems in upstream submissions.

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Re: [alsa-devel] [PATCH 03/20] ALSA: pxa2xx: remove __dev* attributes
by Haojian Zhuang 08 Dec '12

08 Dec '12

On Fri, Dec 7, 2012 at 1:35 AM, Bill Pemberton <wfp5p(a)virginia.edu> wrote: > CONFIG_HOTPLUG is going away as an option. As result the __dev* > markings will be going away. > > Remove use of __devinit, __devexit_p, __devinitdata, __devinitconst, > and __devexit. > > Signed-off-by: Bill Pemberton <wfp5p(a)virginia.edu> > Cc: Eric Miao <eric.y.miao(a)gmail.com> > Cc: Russell King <linux(a)arm.linux.org.uk> > Cc: Haojian Zhuang <haojian.zhuang(a)gmail.com> > Cc: linux-arm-kernel(a)lists.infradead.org > --- > sound/arm/pxa2xx-ac97-lib.c | 2 +- > sound/arm/pxa2xx-ac97.c | 6 +++--- > 2 files changed, 4 insertions(+), 4 deletions(-) > > diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c > index 48d7c0a..6fc0ae9 100644 > --- a/sound/arm/pxa2xx-ac97-lib.c > +++ b/sound/arm/pxa2xx-ac97-lib.c > @@ -314,7 +314,7 @@ int pxa2xx_ac97_hw_resume(void) > EXPORT_SYMBOL_GPL(pxa2xx_ac97_hw_resume); > #endif > > -int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev) > +int pxa2xx_ac97_hw_probe(struct platform_device *dev) > { > int ret; > pxa2xx_audio_ops_t *pdata = dev->dev.platform_data; > diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c > index 4e1fda7..ec54be4 100644 > --- a/sound/arm/pxa2xx-ac97.c > +++ b/sound/arm/pxa2xx-ac97.c > @@ -163,7 +163,7 @@ static int pxa2xx_ac97_resume(struct device *dev) > static SIMPLE_DEV_PM_OPS(pxa2xx_ac97_pm_ops, pxa2xx_ac97_suspend, pxa2xx_ac97_resume); > #endif > > -static int __devinit pxa2xx_ac97_probe(struct platform_device *dev) > +static int pxa2xx_ac97_probe(struct platform_device *dev) > { > struct snd_card *card; > struct snd_ac97_bus *ac97_bus; > @@ -224,7 +224,7 @@ err_dev: > return ret; > } > > -static int __devexit pxa2xx_ac97_remove(struct platform_device *dev) > +static int pxa2xx_ac97_remove(struct platform_device *dev) > { > struct snd_card *card = platform_get_drvdata(dev); > > @@ -239,7 +239,7 @@ static int __devexit pxa2xx_ac97_remove(struct platform_device *dev) > > static struct platform_driver pxa2xx_ac97_driver = { > .probe = pxa2xx_ac97_probe, > - .remove = __devexit_p(pxa2xx_ac97_remove), > + .remove = pxa2xx_ac97_remove, > .driver = { > .name = "pxa2xx-ac97", > .owner = THIS_MODULE, > -- > 1.8.0.1 > Acked-by: Haojian Zhuang <haojian.zhuang(a)gmail.com>

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Re: [alsa-devel] [PATCH 05/25] ASoC: pxa: remove __dev* attributes
by Haojian Zhuang 08 Dec '12

08 Dec '12

On Fri, Dec 7, 2012 at 10:26 PM, Bill Pemberton <wfp5p(a)virginia.edu> wrote: > CONFIG_HOTPLUG is going away as an option. As result the __dev* > markings will be going away. > > Remove use of __devinit, __devexit_p, __devinitdata, __devinitconst, > and __devexit. > > Signed-off-by: Bill Pemberton <wfp5p(a)virginia.edu> > Cc: Eric Miao <eric.y.miao(a)gmail.com> > Cc: Russell King <linux(a)arm.linux.org.uk> > Cc: Haojian Zhuang <haojian.zhuang(a)gmail.com> > Cc: linux-arm-kernel(a)lists.infradead.org > --- > sound/soc/pxa/brownstone.c | 6 +++--- > sound/soc/pxa/corgi.c | 6 +++--- > sound/soc/pxa/e740_wm9705.c | 6 +++--- > sound/soc/pxa/e750_wm9705.c | 6 +++--- > sound/soc/pxa/e800_wm9712.c | 6 +++--- > sound/soc/pxa/imote2.c | 6 +++--- > sound/soc/pxa/mioa701_wm9713.c | 6 +++--- > sound/soc/pxa/mmp-pcm.c | 6 +++--- > sound/soc/pxa/mmp-sspa.c | 6 +++--- > sound/soc/pxa/palm27x.c | 4 ++-- > sound/soc/pxa/poodle.c | 6 +++--- > sound/soc/pxa/pxa-ssp.c | 6 +++--- > sound/soc/pxa/pxa2xx-ac97.c | 8 ++++---- > sound/soc/pxa/pxa2xx-i2s.c | 4 ++-- > sound/soc/pxa/pxa2xx-pcm.c | 6 +++--- > sound/soc/pxa/tosa.c | 6 +++--- > sound/soc/pxa/ttc-dkb.c | 6 +++--- > 17 files changed, 50 insertions(+), 50 deletions(-) > Acked-by: Haojian Zhuang <haojian.zhuang(a)gmail.com>

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[alsa-devel] [patch] usb6fire: prevent driver panic state when stopping
by Jurgen Kramer 07 Dec '12

07 Dec '12

Hi, The patch below prevents the 6fire usb driver going into panic state when stopping playing. On some systems the urb in handler (usb6fire_pcm_in_urb_handler) is being called while urbs are being killed off, this causes the driver to set panic state and can result in the kernel warning 'URB %p submitted while active'. Signed-off-by: Jurgen Kramer <gtmkramer(a)xs4all.nl> diff -uNrp linux/sound/usb/6fire/pcm.c linux.new/sound/usb/6fire/pcm.c --- linux/sound/usb/6fire/pcm.c 2012-11-23 17:01:41.754673597 +0100 +++ linux.new/sound/usb/6fire/pcm.c 2012-12-07 10:09:11.386666744 +0100 @@ -135,6 +135,9 @@ static void usb6fire_pcm_stream_stop(str struct control_runtime *ctrl_rt = rt->chip->control; if (rt->stream_state != STREAM_DISABLED) { + + rt->stream_state = STREAM_STOPPING; + for (i = 0; i < PCM_N_URBS; i++) { usb_kill_urb(&rt->in_urbs[i].instance); usb_kill_urb(&rt->out_urbs[i].instance); Jurgen

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[alsa-devel] M-Audio Audiophile 24/96 (snd-ice1712) lacks ADCs
by Pils Bert 07 Dec '12

07 Dec '12

Hi guys, I have a problem with my M-Audio Audiophile 24/96 soundcard: There are no ADC volume sliders present neither in alsamixer nor in envy24control. But instead there are also DAC sliders on the capture page in alsamixer?! But changing them doesn't have an effect on the H/W input gain. I read on website that some other users had the same problem (but it seams to be a rare problem). First of all my card has been correctly detected as Audiophile 24/96. But in /sys/module/snd_ice1712/parameters/model are only null values. I loaded the module with the model=audiophile parameter, but no ADCs appear. Second I skimmed over the module source code and found the lines case ICE1712_SUBDEVICE_AUDIOPHILE: ice->num_total_dacs = 2; ice->num_total_adcs = 2; break; in __devinit snd_ice1712_delta_init(struct snd_ice1712 *ice) in delta.c So the module says there have to be 2 ADCs... I also disabled my onboard soundcard (HDA Intel), but no ADCs appear. Maybe its some hardware configuration problem? Moreover, I had to edit my .asoundrc file in order to play and capture audio anyway (so no 'out of the box' experience here...): pcm.audiophile { type hw card 1 device 0 } pcm.analog { type plug slave.pcm audiophile ttable.0.0 1 # H/W 1 ttable.1.1 1 # H/W 2 } I'm using plain ALSA 1.0.25 (without pulseaudio) on Debian Testing (Linux Rowdy 3.2.0-4-amd64 #1 SMP Debian 3.2.32-1 x86_64 GNU/Linux) Maybe some of you guys have an good idea ;) Best regards, Jannik

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Re: [alsa-devel] [PATCH RESEND V2 1/2] ASoC: Samsung: Do not register samsung audio dma device as pdev
by Mark Brown 07 Dec '12

07 Dec '12

On Fri, Dec 07, 2012 at 01:59:21PM +0530, Padmavathi Venna wrote: > Previously, the ASoC 'platform' (PCM/DMA) object was instantiated via a > platform_device. This didn't represent the hardware well, since there > was no separate hardware associated with this platform_device; it was a > virtual device with sole purpose to call snd_soc_register_platform(). Applied, thanks.

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Re: [alsa-devel] [PATCH 00/20] ALSA: remove __dev* attributes for retirement of HOTPLUG
by Takashi Iwai 07 Dec '12

07 Dec '12

At Thu, 06 Dec 2012 12:45:56 -0500, Bill Pemberton wrote: > > Takashi Iwai writes: > > > > At Thu, 6 Dec 2012 12:35:09 -0500, > > Bill Pemberton wrote: > > > > > > Remove use of __devinit, __devexit_p, __devinitdata, __devinitconst, > > > and __devexit so that HOTPLUG can be retired. > > > > > > Bill Pemberton (20): > > > ALSA: pci: remove __dev* attributes > > > ALSA: AACI: remove __dev* attributes > > > ALSA: pxa2xx: remove __dev* attributes > > > ALSA: atmel: remove __dev* attributes > > > ALSA: firewire-speakers: remove __dev* attributes > > > ALSA: mips: remove __dev* attributes > > > ALSA: parisc/harmony: remove __dev* attributes > > > ALSA: oss: remove __dev* attributes > > > ALSA: ad1889: remove __dev* attributes > > > ALSA: bt87X: remove __dev* attributes > > > ALSA: oxygen: remove __dev* attributes > > > > Any reason the three above to be separate patches? > > > > They are separately listed in MAINTAINERS, so I broke them out for the > appropriate Cc:'s. Ah, understood. In general, split isn't needed for a case like this. Most of such trivial cleanup patches are applied at once to the relevant codes in the whole subsystem. In anyway, I applied all patches now. thanks, Takashi

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[alsa-devel] Compilation error for static build of alsa-lib
by Pankaj DEV 07 Dec '12

07 Dec '12

When building alsa (using buildroot) for static build, compilation errors occur. Don't know how to report bug, since the "bug tracker" link on web-page doesn't open The following patch solves the issue : --- a/modules/mixer/simple/sbasedl.c +++ b/modules/mixer/simple/sbasedl.c @@ -27,7 +27,11 @@ #include <fcntl.h> #include <sys/ioctl.h> #include <math.h> +#ifdef HAVE_LIBDL #include <dlfcn.h> +#else +#define RTLD_NOW 0 +#endif #include "config.h" #include "asoundlib.h" #include "mixer_abst.h" diff --git a/src/mixer/simple_abst.c b/src/mixer/simple_abst.c index 9e9aaf5..03070e0 100644 --- a/src/mixer/simple_abst.c +++ b/src/mixer/simple_abst.c @@ -34,7 +34,12 @@ #include <fcntl.h> #include <sys/ioctl.h> #include <math.h> +#ifdef HAVE_LIBDL #include <dlfcn.h> +#else +#define RTLD_NOW 0 +#define RTLD_GLOBAL 8 +#endif #include "config.h" #include "asoundlib.h" #include "mixer_simple.h"

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Re: [alsa-devel] [PATCH V2 1/2] ASoC: Samsung: Do not register samsung audio dma device as pdev
by Mark Brown 07 Dec '12

07 Dec '12

On Thu, Dec 06, 2012 at 09:20:36AM +0530, Padmavathi Venna wrote: > Previously, the ASoC 'platform' (PCM/DMA) object was instantiated via a > platform_device. This didn't represent the hardware well, since there > was no separate hardware associated with this platform_device; it was a > virtual device with sole purpose to call snd_soc_register_platform(). Looks good however this doesn't apply against my topic/samsung branch - can you please check if I need anything from Kukjin's tree? Though I guess the easiest thing is just to wait until the merge window is over, we're near to the next release.

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Re: [alsa-devel] Compilation error for static build of alsa-lib
by Pankaj DEV 07 Dec '12

07 Dec '12

Forgot to mention the version 1.0.26 From: Pankaj DEV Sent: Friday, December 07, 2012 12:01 PM To: alsa-devel(a)alsa-project.org Subject: Compilation error for static build of alsa-lib When building alsa (using buildroot) for static build, compilation errors occur. Don't know how to report bug, since the "bug tracker" link on web-page doesn't open The following patch solves the issue : --- a/modules/mixer/simple/sbasedl.c +++ b/modules/mixer/simple/sbasedl.c @@ -27,7 +27,11 @@ #include <fcntl.h> #include <sys/ioctl.h> #include <math.h> +#ifdef HAVE_LIBDL #include <dlfcn.h> +#else +#define RTLD_NOW 0 +#endif #include "config.h" #include "asoundlib.h" #include "mixer_abst.h" diff --git a/src/mixer/simple_abst.c b/src/mixer/simple_abst.c index 9e9aaf5..03070e0 100644 --- a/src/mixer/simple_abst.c +++ b/src/mixer/simple_abst.c @@ -34,7 +34,12 @@ #include <fcntl.h> #include <sys/ioctl.h> #include <math.h> +#ifdef HAVE_LIBDL #include <dlfcn.h> +#else +#define RTLD_NOW 0 +#define RTLD_GLOBAL 8 +#endif #include "config.h" #include "asoundlib.h" #include "mixer_simple.h"

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[alsa-devel] Compilation error for static build of alsa-utils 1.0.26
by Pankaj DEV 07 Dec '12

07 Dec '12

When building alsa (using buildroot) for static build, compilation errors occur. Don't know how to report bug, since the "bug tracker" link on web-page doesn't open The following patch solves the issue : --- a/configure +++ b/configure @@ -6247,7 +6247,7 @@ if test "$alsa_prefix" != "" ; then LDFLAGS="$LDFLAGS $ALSA_LIBS" fi -ALSA_LIBS="$ALSA_LIBS -lasound -lm -ldl -lpthread" +ALSA_LIBS="$ALSA_LIBS -lasound -lm -lpthread" LIBS="$ALSA_LIBS $LIBS" { $as_echo "$as_me:${as_lineno-$LINENO}: result: $ALSA_LIBS" >&5 $as_echo "$ALSA_LIBS" >&6; }

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[alsa-devel] [PATCH v2 1/3] ALSA: ASoC: McASP: remove unused variables
by Daniel Mack 07 Dec '12

07 Dec '12

codec_fmt and sample_rate variables are unused in both snd_platform_data and davinci_audio_dev, so drop them. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- include/linux/platform_data/davinci_asp.h | 1 - sound/soc/davinci/davinci-mcasp.c | 1 - sound/soc/davinci/davinci-mcasp.h | 2 -- 3 files changed, 4 deletions(-) diff --git a/include/linux/platform_data/davinci_asp.h b/include/linux/platform_data/davinci_asp.h index f3d6e4f..8db5ae0 100644 --- a/include/linux/platform_data/davinci_asp.h +++ b/include/linux/platform_data/davinci_asp.h @@ -23,7 +23,6 @@ struct snd_platform_data { u32 rx_dma_offset; int asp_chan_q; /* event queue number for ASP channel */ int ram_chan_q; /* event queue number for RAM channel */ - unsigned int codec_fmt; /* * Allowing this is more efficient and eliminates left and right swaps * caused by underruns, but will swap the left and right channels diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 46350b8..023a70d 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1174,7 +1174,6 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dev->tdm_slots = pdata->tdm_slots; dev->num_serializer = pdata->num_serializer; dev->serial_dir = pdata->serial_dir; - dev->codec_fmt = pdata->codec_fmt; dev->version = pdata->version; dev->txnumevt = pdata->txnumevt; dev->rxnumevt = pdata->rxnumevt; diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 156f15f..a42b5d9 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -40,9 +40,7 @@ enum { struct davinci_audio_dev { struct davinci_pcm_dma_params dma_params[2]; void __iomem *base; - int sample_rate; struct device *dev; - unsigned int codec_fmt; /* McASP specific data */ int tdm_slots; -- 1.7.11.7

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[alsa-devel] [PATCH] ASoC: omap-twl4030: Update the header file to support more boards
by Peter Ujfalusi 07 Dec '12

07 Dec '12

The common machine driver will be able to support new boards where the voice port is also in use. At the same time add possibility to fine tune the connections from twl4030 on the board: jack detection GPIO, input/output selection Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi Mark, Liam Tony, I just finalized a series which I intend to send for 3.9: It is going to update the omap-twl4030 ASoC machine driver to support more boards, thus allowing me to remove the driver for zoom2 and sdp3440. The issue is that in order to do it I need to make modification in arch/arm/mach-omap2/ and sound/soc/omap/ as well. One patch which changes the header file is the only compile time requirement for the changes. As a reference this is how my tree looks like regarding to this change (need to sort them still), git log --oneline : 1702213 ASoC: OMAP: Remove obsolete machine drivers for Zoom2 and SDP3403 41ff1a6 ARM: OMAP: sdp3430: Audio support via the common omap-twl4030 machine driver f51363ed ARM: OMAP: zoom: Audio support via the common omap-twl4030 machine driver 159e443 ASoC: omap-twl4030: Add support for routing, voice port and jack detect 77bda45 ARM: OMAP2+: twl-common: Allow boards to customize the twl4030 audio setup 0db92ac ARM: OMAP2+: twl-common: Add default twl4030 audio configuration 9005d5b ARM: OMAP: zoom: Zoom2 does not have extmute functionality c7ffd18 ASoC: sdp3430: No need to configure pin mux for extmute 43dc9d9 ASoC: twl4030: Configure extmute pinmux when the dedicated pin is in use db9207c ARM: OMAP: 3430sdp: Enable extmute functionality for audio e6aea6f ASoC: twl4030: Convert MICBIAS to SUPPLY widget 8eadb74 ASoC: sdp3430: No need to configure the Voice port anymore f11c1be ASoC: zoom2: No need to configure the Voice port anymore 63da45d ASoC: twl4030: Correct the support for Voice port a98e934 ASoC: omap-twl4030: Update the header file to support more boards I know it is quite late already, but Mark would you be able the squeeze in this patch for 3.8? If this is in, I can separate the arch/arm/mach-omap2/ and sound/soc/omap/ patches and queue them via the appropriate tree without creating cross tree merge issues (yes, again cross tree issue). All of these will come together for 3.9 and in linux-next. This header change does not affect the current driver since the new parameters are ignored by the current driver Mark, would you be able to do this? Thank you, Peter include/linux/platform_data/omap-twl4030.h | 26 ++++++++++++++++++++++++++ 1 file changed, 26 insertions(+) diff --git a/include/linux/platform_data/omap-twl4030.h b/include/linux/platform_data/omap-twl4030.h index c7bef78..ee60ef7 100644 --- a/include/linux/platform_data/omap-twl4030.h +++ b/include/linux/platform_data/omap-twl4030.h @@ -25,8 +25,34 @@ #ifndef _OMAP_TWL4030_H_ #define _OMAP_TWL4030_H_ +/* To select if only one channel is connected in a stereo port */ +#define OMAP_TWL4030_LEFT (1 << 0) +#define OMAP_TWL4030_RIGHT (1 << 1) + struct omap_tw4030_pdata { const char *card_name; + /* Voice port is connected to McBSP3 */ + bool voice_connected; + + /* The driver will parse the connection flags if this flag is set */ + bool custom_routing; + /* Flags to indicate connected audio ports. */ + u8 has_hs; + u8 has_hf; + u8 has_predriv; + u8 has_carkit; + bool has_ear; + + bool has_mainmic; + bool has_submic; + bool has_hsmic; + bool has_carkitmic; + bool has_digimic0; + bool has_digimic1; + u8 has_linein; + + /* Jack detect GPIO or <= 0 if it is not implemented */ + int jack_detect; }; #endif /* _OMAP_TWL4030_H_ */ -- 1.8.0

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[alsa-devel] [PATCH] ASoC: arizona: Log the clock we're setting the DAI to use
by Mark Brown 07 Dec '12

07 Dec '12

Useful for diagnostics. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index a7eb553..d5385dc 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -811,6 +811,9 @@ static int arizona_dai_set_sysclk(struct snd_soc_dai *dai, return -EBUSY; } + dev_dbg(codec->dev, "Setting AIF%d to %s\n", dai->id + 1, + arizona_dai_clk_str(clk_id)); + memset(&routes, 0, sizeof(routes)); routes[0].sink = dai->driver->capture.stream_name; routes[1].sink = dai->driver->playback.stream_name; -- 1.7.10.4

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Re: [alsa-devel] I would like to resubmit an ALSA device driver for the Maxim 98090 Codec...
by Mark Brown 07 Dec '12

07 Dec '12

On Thu, Dec 06, 2012 at 05:14:50PM -0800, Jerry Wong wrote: > Here is the resubmission of the sound/soc/max98090.c driver with the changes you suggested including regmap I/O and some stylistic updates. Please fix your mailer to word wrap within paragraphs. Please also follow the submission procedure in SubmittingPatches, as previously mentioned this is essential especially the Signed-off-by bit. I really can't apply or even properly review something that doesn't follow that. Looking at the code you want to check the register defaults - defaults for volatile registers don't make a huge amount of sense. For the ranges based on bases use ... or if (else) rather than listing the individual registers by hand. For the custom get and sets I can't see why you're not able to use generic controls, if you aren't there's enough of these that you should add a generic type. Gains should always be volume controls with TLV, enable/disable stuff should be Switch and controls called Gain Volume don't make a lot of sense (just Volume). Prints that just log the build time aren't helpful, you'll notice no other mainline drivers do that... there's already a kernel level build time. Similarly other stuff like interrupt numbers that are getting logged are a bit noisy. Build time options for things like performance need to be made runtime or platform data based. Are you sure you want the bandgap reference running the entire time the system is powered up?

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Re: [alsa-devel] [PATCH 05/20] ALSA: firewire-speakers: remove __dev* attributes
by Clemens Ladisch 06 Dec '12

06 Dec '12

Bill Pemberton wrote: > CONFIG_HOTPLUG is going away as an option. As result the __dev* > markings will be going away. > > Remove use of __devinit, __devexit_p, __devinitdata, __devinitconst, > and __devexit. > > Signed-off-by: Bill Pemberton <wfp5p(a)virginia.edu> > Cc: Clemens Ladisch <clemens(a)ladisch.de> Acked-by: Clemens Ladisch <clemens(a)ladisch.de> for all these (5,10,11): > sound/firewire/speakers.c | 8 ++++---- > sound/pci/bt87x.c | 22 +++++++++++----------- > sound/pci/oxygen/oxygen.c | 10 +++++----- > sound/pci/oxygen/virtuoso.c | 10 +++++----- > sound/pci/oxygen/xonar_cs43xx.c | 4 ++-- > sound/pci/oxygen/xonar_pcm179x.c | 4 ++-- > sound/pci/oxygen/xonar_wm87x6.c | 4 ++--

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[alsa-devel] How to mix two PCM channels with ALSA
by itxEpsilon 06 Dec '12

06 Dec '12

Dear Alsa developers: I'm trying to write a simple application for mixing two PCM sound channels in C++. Could anyone provide me a quick example about how to implement this operation? Thank you in advance. Itx

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Re: [alsa-devel] [PATCH 00/20] ALSA: remove __dev* attributes for retirement of HOTPLUG
by Takashi Iwai 06 Dec '12

06 Dec '12

At Thu, 6 Dec 2012 12:35:09 -0500, Bill Pemberton wrote: > > Remove use of __devinit, __devexit_p, __devinitdata, __devinitconst, > and __devexit so that HOTPLUG can be retired. > > Bill Pemberton (20): > ALSA: pci: remove __dev* attributes > ALSA: AACI: remove __dev* attributes > ALSA: pxa2xx: remove __dev* attributes > ALSA: atmel: remove __dev* attributes > ALSA: firewire-speakers: remove __dev* attributes > ALSA: mips: remove __dev* attributes > ALSA: parisc/harmony: remove __dev* attributes > ALSA: oss: remove __dev* attributes > ALSA: ad1889: remove __dev* attributes > ALSA: bt87X: remove __dev* attributes > ALSA: oxygen: remove __dev* attributes Any reason the three above to be separate patches? > ALSA: isa: remove __dev* attributes > ALSA: sound/ps3: remove __dev* attributes > ALSA: ppc: remove __dev* attributes > ALSA: sh: remove __dev* attributes > ALSA: sparc: remove __dev* attributes > ALSA: at73c213: remove __dev* attributes > ALSA: drivers: remove __dev* attributes > ALSA: snd-usb-6fire: remove __dev* attributes > ALSA: snd-usb-caiaq: remove __dev* attributes Also, these can be a single clean up for sound/usb/*... It's no big matter, but I just wonder since I don't see a logic... Takashi > sound/arm/aaci.c | 18 +++--- > sound/arm/pxa2xx-ac97-lib.c | 2 +- > sound/arm/pxa2xx-ac97.c | 6 +- > sound/atmel/abdac.c | 8 +-- > sound/atmel/ac97c.c | 10 ++-- > sound/drivers/aloop.c | 16 +++--- > sound/drivers/dummy.c | 14 ++--- > sound/drivers/ml403-ac97cr.c | 10 ++-- > sound/drivers/mpu401/mpu401.c | 18 +++--- > sound/drivers/mtpav.c | 14 ++--- > sound/drivers/mts64.c | 40 +++++++------- > sound/drivers/pcsp/pcsp.c | 14 ++--- > sound/drivers/pcsp/pcsp_input.c | 2 +- > sound/drivers/pcsp/pcsp_input.h | 2 +- > sound/drivers/pcsp/pcsp_lib.c | 2 +- > sound/drivers/pcsp/pcsp_mixer.c | 10 ++-- > sound/drivers/portman2x4.c | 18 +++--- > sound/drivers/serial-u16550.c | 32 +++++------ > sound/drivers/virmidi.c | 6 +- > sound/firewire/speakers.c | 8 +-- > sound/isa/ad1816a/ad1816a.c | 18 +++--- > sound/isa/ad1816a/ad1816a_lib.c | 19 ++++--- > sound/isa/ad1848/ad1848.c | 8 +-- > sound/isa/adlib.c | 8 +-- > sound/isa/als100.c | 22 ++++---- > sound/isa/azt2320.c | 26 ++++----- > sound/isa/cmi8328.c | 8 +-- > sound/isa/cmi8330.c | 42 +++++++------- > sound/isa/cs423x/cs4231.c | 8 +-- > sound/isa/cs423x/cs4236.c | 50 ++++++++--------- > sound/isa/es1688/es1688.c | 28 +++++----- > sound/isa/es18xx.c | 80 +++++++++++++-------------- > sound/isa/galaxy/galaxy.c | 26 ++++----- > sound/isa/gus/gusclassic.c | 15 ++--- > sound/isa/gus/gusextreme.c | 24 ++++---- > sound/isa/gus/gusmax.c | 16 +++--- > sound/isa/gus/interwave.c | 54 +++++++++--------- > sound/isa/msnd/msnd.h | 2 +- > sound/isa/msnd/msnd_pinnacle.c | 44 +++++++-------- > sound/isa/msnd/msnd_pinnacle_mixer.c | 2 +- > sound/isa/opl3sa2.c | 40 +++++++------- > sound/isa/opti9xx/miro.c | 68 +++++++++++------------ > sound/isa/opti9xx/opti92x-ad1848.c | 42 +++++++------- > sound/isa/sb/emu8000.c | 22 ++++---- > sound/isa/sb/jazz16.c | 18 +++--- > sound/isa/sb/sb16.c | 26 ++++----- > sound/isa/sb/sb8.c | 8 +-- > sound/isa/sc6000.c | 38 ++++++------- > sound/isa/sscape.c | 32 +++++------ > sound/isa/wavefront/wavefront.c | 53 +++++++++--------- > sound/isa/wavefront/wavefront_fx.c | 2 +- > sound/isa/wavefront/wavefront_midi.c | 2 +- > sound/isa/wavefront/wavefront_synth.c | 14 ++--- > sound/mips/au1x00.c | 4 +- > sound/mips/hal2.c | 14 ++--- > sound/mips/sgio2audio.c | 28 +++++----- > sound/oss/ad1848.c | 2 +- > sound/oss/kahlua.c | 10 ++-- > sound/parisc/harmony.c | 12 ++-- > sound/pci/ad1889.c | 18 +++--- > sound/pci/ak4531_codec.c | 10 ++-- > sound/pci/ali5451/ali5451.c | 32 +++++------ > sound/pci/als300.c | 14 ++--- > sound/pci/als4000.c | 12 ++-- > sound/pci/asihpi/asihpi.c | 55 +++++++++---------- > sound/pci/asihpi/hpioctl.c | 6 +- > sound/pci/asihpi/hpioctl.h | 6 +- > sound/pci/atiixp.c | 32 +++++------ > sound/pci/atiixp_modem.c | 20 +++---- > sound/pci/au88x0/au88x0.c | 10 ++-- > sound/pci/au88x0/au88x0_a3d.c | 6 +- > sound/pci/au88x0/au88x0_core.c | 2 +- > sound/pci/au88x0/au88x0_eq.c | 10 ++-- > sound/pci/au88x0/au88x0_game.c | 2 +- > sound/pci/au88x0/au88x0_mixer.c | 2 +- > sound/pci/au88x0/au88x0_mpu401.c | 2 +- > sound/pci/au88x0/au88x0_pcm.c | 6 +- > sound/pci/aw2/aw2-alsa.c | 28 +++++----- > sound/pci/azt3328.c | 22 ++++---- > sound/pci/bt87x.c | 22 ++++---- > sound/pci/ca0106/ca0106_main.c | 12 ++-- > sound/pci/ca0106/ca0106_mixer.c | 26 ++++----- > sound/pci/ca0106/ca0106_proc.c | 2 +- > sound/pci/ca0106/ca_midi.c | 2 +- > sound/pci/cmipci.c | 48 ++++++++-------- > sound/pci/cs4281.c | 30 +++++----- > sound/pci/cs46xx/cs46xx.c | 8 +-- > sound/pci/cs46xx/cs46xx_lib.c | 35 ++++++------ > sound/pci/cs5530.c | 16 +++--- > sound/pci/cs5535audio/cs5535audio.c | 18 +++--- > sound/pci/cs5535audio/cs5535audio.h | 10 ++-- > sound/pci/cs5535audio/cs5535audio_olpc.c | 10 ++-- > sound/pci/cs5535audio/cs5535audio_pcm.c | 2 +- > sound/pci/ctxfi/ctatc.c | 20 +++---- > sound/pci/ctxfi/ctatc.h | 8 +-- > sound/pci/ctxfi/cthardware.c | 4 +- > sound/pci/ctxfi/cthw20k1.c | 4 +- > sound/pci/ctxfi/cthw20k2.c | 4 +- > sound/pci/ctxfi/xfi.c | 6 +- > sound/pci/echoaudio/echoaudio.c | 46 ++++++++-------- > sound/pci/echoaudio/echoaudio.h | 4 +- > sound/pci/echoaudio/midi.c | 4 +- > sound/pci/emu10k1/emu10k1.c | 8 +-- > sound/pci/emu10k1/emu10k1_main.c | 4 +- > sound/pci/emu10k1/emu10k1x.c | 29 +++++----- > sound/pci/emu10k1/emufx.c | 25 +++++---- > sound/pci/emu10k1/emumixer.c | 22 ++++---- > sound/pci/emu10k1/emumpu401.c | 6 +- > sound/pci/emu10k1/emupcm.c | 11 ++-- > sound/pci/emu10k1/emuproc.c | 2 +- > sound/pci/emu10k1/p16v.c | 8 +-- > sound/pci/emu10k1/timer.c | 2 +- > sound/pci/ens1370.c | 52 +++++++++--------- > sound/pci/es1938.c | 20 +++---- > sound/pci/es1968.c | 42 +++++++------- > sound/pci/fm801.c | 26 ++++----- > sound/pci/hda/hda_intel.c | 48 ++++++++-------- > sound/pci/ice1712/amp.c | 6 +- > sound/pci/ice1712/aureon.c | 24 ++++---- > sound/pci/ice1712/delta.c | 42 +++++++------- > sound/pci/ice1712/ews.c | 28 +++++----- > sound/pci/ice1712/hoontech.c | 26 ++++----- > sound/pci/ice1712/ice1712.c | 94 ++++++++++++++++---------------- > sound/pci/ice1712/ice1724.c | 64 +++++++++++----------- > sound/pci/ice1712/juli.c | 26 ++++----- > sound/pci/ice1712/maya44.c | 16 +++--- > sound/pci/ice1712/phase.c | 22 ++++---- > sound/pci/ice1712/pontis.c | 10 ++-- > sound/pci/ice1712/prodigy192.c | 12 ++-- > sound/pci/ice1712/prodigy_hifi.c | 20 +++---- > sound/pci/ice1712/psc724.c | 8 +-- > sound/pci/ice1712/quartet.c | 24 ++++---- > sound/pci/ice1712/revo.c | 28 +++++----- > sound/pci/ice1712/se.c | 20 +++---- > sound/pci/ice1712/vt1720_mobo.c | 10 ++-- > sound/pci/ice1712/wtm.c | 10 ++-- > sound/pci/intel8x0.c | 56 +++++++++---------- > sound/pci/intel8x0m.c | 30 +++++----- > sound/pci/korg1212/korg1212.c | 12 ++-- > sound/pci/lola/lola.c | 14 ++--- > sound/pci/lola/lola_clock.c | 2 +- > sound/pci/lola/lola_mixer.c | 32 +++++------ > sound/pci/lola/lola_pcm.c | 4 +- > sound/pci/lola/lola_proc.c | 2 +- > sound/pci/lx6464es/lx6464es.c | 30 +++++----- > sound/pci/lx6464es/lx_core.c | 2 +- > sound/pci/lx6464es/lx_core.h | 2 +- > sound/pci/maestro3.c | 24 ++++---- > sound/pci/mixart/mixart.c | 12 ++-- > sound/pci/nm256/nm256.c | 18 +++--- > sound/pci/oxygen/oxygen.c | 10 ++-- > sound/pci/oxygen/virtuoso.c | 10 ++-- > sound/pci/oxygen/xonar_cs43xx.c | 4 +- > sound/pci/oxygen/xonar_pcm179x.c | 4 +- > sound/pci/oxygen/xonar_wm87x6.c | 4 +- > sound/pci/pcxhr/pcxhr.c | 14 ++--- > sound/pci/riptide/riptide.c | 20 +++---- > sound/pci/rme32.c | 10 ++-- > sound/pci/rme96.c | 14 ++--- > sound/pci/rme9652/hdsp.c | 14 ++--- > sound/pci/rme9652/hdspm.c | 41 +++++++------- > sound/pci/rme9652/rme9652.c | 22 ++++---- > sound/pci/sis7019.c | 16 +++--- > sound/pci/sonicvibes.c | 37 +++++++------ > sound/pci/trident/trident.c | 8 +-- > sound/pci/trident/trident_main.c | 46 ++++++++-------- > sound/pci/via82xx.c | 50 ++++++++--------- > sound/pci/via82xx_modem.c | 26 ++++----- > sound/pci/vx222/vx222.c | 14 ++--- > sound/pci/ymfpci/ymfpci.c | 12 ++-- > sound/pci/ymfpci/ymfpci_main.c | 44 ++++++++------- > sound/ppc/awacs.c | 54 +++++++++--------- > sound/ppc/beep.c | 2 +- > sound/ppc/burgundy.c | 22 ++++---- > sound/ppc/daca.c | 2 +- > sound/ppc/keywest.c | 4 +- > sound/ppc/pmac.c | 12 ++-- > sound/ppc/powermac.c | 6 +- > sound/ppc/snd_ps3.c | 12 ++-- > sound/ppc/tumbler.c | 16 +++--- > sound/sh/aica.c | 13 ++--- > sound/sh/sh_dac_audio.c | 10 ++-- > sound/sparc/amd7930.c | 16 +++--- > sound/sparc/cs4231.c | 38 ++++++------- > sound/sparc/dbri.c | 28 +++++----- > sound/spi/at73c213.c | 20 +++---- > sound/usb/6fire/chip.c | 4 +- > sound/usb/6fire/comm.c | 2 +- > sound/usb/6fire/comm.h | 2 +- > sound/usb/6fire/control.c | 8 +-- > sound/usb/6fire/control.h | 2 +- > sound/usb/6fire/firmware.h | 2 +- > sound/usb/6fire/midi.c | 2 +- > sound/usb/6fire/midi.h | 2 +- > sound/usb/6fire/pcm.c | 8 +-- > sound/usb/6fire/pcm.h | 2 +- > sound/usb/caiaq/control.c | 8 +-- > sound/usb/caiaq/device.c | 6 +- > 198 files changed, 1794 insertions(+), 1785 deletions(-) > > -- > 1.8.0.1 >

1 0

Re: [alsa-devel] [PATCH 20/20] ALSA: snd-usb-caiaq: remove __dev* attributes
by Daniel Mack 06 Dec '12

06 Dec '12

On 06.12.2012 18:35, Bill Pemberton wrote: > CONFIG_HOTPLUG is going away as an option. As result the __dev* > markings will be going away. > > Remove use of __devinit, __devexit_p, __devinitdata, __devinitconst, > and __devexit. > > Signed-off-by: Bill Pemberton <wfp5p(a)virginia.edu> > Cc: Daniel Mack <zonque(a)gmail.com> Acked-by: Daniel Mack <zonque(a)gmail.com> Thanks! Daniel > --- > sound/usb/caiaq/control.c | 8 ++++---- > sound/usb/caiaq/device.c | 6 +++--- > 2 files changed, 7 insertions(+), 7 deletions(-) > > diff --git a/sound/usb/caiaq/control.c b/sound/usb/caiaq/control.c > index 00e5d0a..adb8d03 100644 > --- a/sound/usb/caiaq/control.c > +++ b/sound/usb/caiaq/control.c > @@ -137,7 +137,7 @@ static int control_put(struct snd_kcontrol *kcontrol, > return 1; > } > > -static struct snd_kcontrol_new kcontrol_template __devinitdata = { > +static struct snd_kcontrol_new kcontrol_template = { > .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, > .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, > .index = 0, > @@ -489,8 +489,8 @@ static struct caiaq_controller kontrols4_controller[] = { > { "LED: FX2: Mode", 133 | CNT_INTVAL }, > }; > > -static int __devinit add_controls(struct caiaq_controller *c, int num, > - struct snd_usb_caiaqdev *dev) > +static int add_controls(struct caiaq_controller *c, int num, > + struct snd_usb_caiaqdev *dev) > { > int i, ret; > struct snd_kcontrol *kc; > @@ -507,7 +507,7 @@ static int __devinit add_controls(struct caiaq_controller *c, int num, > return 0; > } > > -int __devinit snd_usb_caiaq_control_init(struct snd_usb_caiaqdev *dev) > +int snd_usb_caiaq_control_init(struct snd_usb_caiaqdev *dev) > { > int ret = 0; > > diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c > index 7da0d0a..c828f81 100644 > --- a/sound/usb/caiaq/device.c > +++ b/sound/usb/caiaq/device.c > @@ -289,7 +289,7 @@ int snd_usb_caiaq_set_auto_msg(struct snd_usb_caiaqdev *dev, > tmp, sizeof(tmp)); > } > > -static void __devinit setup_card(struct snd_usb_caiaqdev *dev) > +static void setup_card(struct snd_usb_caiaqdev *dev) > { > int ret; > char val[4]; > @@ -407,7 +407,7 @@ static int create_card(struct usb_device *usb_dev, > return 0; > } > > -static int __devinit init_card(struct snd_usb_caiaqdev *dev) > +static int init_card(struct snd_usb_caiaqdev *dev) > { > char *c, usbpath[32]; > struct usb_device *usb_dev = dev->chip.dev; > @@ -481,7 +481,7 @@ static int __devinit init_card(struct snd_usb_caiaqdev *dev) > return 0; > } > > -static int __devinit snd_probe(struct usb_interface *intf, > +static int snd_probe(struct usb_interface *intf, > const struct usb_device_id *id) > { > int ret; >

1 0

[alsa-devel] [PATCH] ASoC: lm49453: Fix analog and digital volume ranges
by MR.Swami.Reddy＠ti.com 06 Dec '12

06 Dec '12

From: MR Swami Reddy <MR.Swami.Reddy(a)ti.com> [PATCH] ASoC: lm49453: Fix analog and digital volume ranges Signed-off-by: M R Swami Reddy <mr.swami.reddy(a)ti.com> -- sound/soc/codecs/lm49453.c | 7 ++++--- 1 files changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index 808e50a..afbefa2 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -495,7 +495,8 @@ SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_PORT2_TX2_REG, 7, 1, 0), /* TLV Declarations */ static const DECLARE_TLV_DB_SCALE(digital_tlv, -7650, 150, 1); -static const DECLARE_TLV_DB_SCALE(port_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(analog_tlv, 0, 200, 1); +static const DECLARE_TLV_DB_SCALE(port_tlv, -1800, 600, 0); static const struct snd_kcontrol_new lm49453_sidetone_mixer_controls[] = { /* Sidetone supports mono only */ @@ -516,9 +517,9 @@ SOC_DAPM_SINGLE_TLV("Sidetone DMIC2R Volume", LM49453_P0_STN_VOL_DMIC2R_REG, static const struct snd_kcontrol_new lm49453_snd_controls[] = { /* mic1 and mic2 supports mono only */ SOC_SINGLE_TLV("Mic1 Volume", LM49453_P0_ADC_LEVELL_REG, 0, 6, - 0, digital_tlv), + 0, analog_tlv), SOC_SINGLE_TLV("Mic2 Volume", LM49453_P0_ADC_LEVELR_REG, 0, 6, - 0, digital_tlv), + 0, analog_tlv), SOC_DOUBLE_R_TLV("DMIC1 Volume", LM49453_P0_DMIC1_LEVELL_REG, LM49453_P0_DMIC1_LEVELR_REG, 0, 6, 0, digital_tlv),

4 3

[alsa-devel] [PATCH 1/3] ALSA: ASoC: McASP: remove unused variables
by Daniel Mack 06 Dec '12

06 Dec '12

codec_fmt and sample_rate variables are unused in both snd_platform_data and davinci_audio_dev, so drop them. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- include/linux/platform_data/davinci_asp.h | 1 - sound/soc/davinci/davinci-mcasp.c | 1 - sound/soc/davinci/davinci-mcasp.h | 2 -- 3 files changed, 4 deletions(-) diff --git a/include/linux/platform_data/davinci_asp.h b/include/linux/platform_data/davinci_asp.h index f3d6e4f..8db5ae0 100644 --- a/include/linux/platform_data/davinci_asp.h +++ b/include/linux/platform_data/davinci_asp.h @@ -23,7 +23,6 @@ struct snd_platform_data { u32 rx_dma_offset; int asp_chan_q; /* event queue number for ASP channel */ int ram_chan_q; /* event queue number for RAM channel */ - unsigned int codec_fmt; /* * Allowing this is more efficient and eliminates left and right swaps * caused by underruns, but will swap the left and right channels diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 46350b8..023a70d 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1174,7 +1174,6 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dev->tdm_slots = pdata->tdm_slots; dev->num_serializer = pdata->num_serializer; dev->serial_dir = pdata->serial_dir; - dev->codec_fmt = pdata->codec_fmt; dev->version = pdata->version; dev->txnumevt = pdata->txnumevt; dev->rxnumevt = pdata->rxnumevt; diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 156f15f..a42b5d9 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -40,9 +40,7 @@ enum { struct davinci_audio_dev { struct davinci_pcm_dma_params dma_params[2]; void __iomem *base; - int sample_rate; struct device *dev; - unsigned int codec_fmt; /* McASP specific data */ int tdm_slots; -- 1.7.11.7

2 5

[alsa-devel] [PATCH] ASoC: lm49453: Fix adc, mic and sidetone volume ranges
by MR.Swami.Reddy＠ti.com 06 Dec '12

06 Dec '12

From: MR Swami Reddy <MR.Swami.Reddy(a)ti.com> [PATCH] ASoC: lm49453: Fix adc, mic and sidetone volume ranges Signed-off-by: MR Swami Reddy <mr.swami.reddy(a)ti.com> -- sound/soc/codecs/lm49453.c | 43 ++++++++++++++++++++++++------------------- 1 files changed, 24 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index 808e50a..379fcdf 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -494,36 +494,41 @@ SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_PORT2_TX2_REG, 7, 1, 0), }; /* TLV Declarations */ -static const DECLARE_TLV_DB_SCALE(digital_tlv, -7650, 150, 1); -static const DECLARE_TLV_DB_SCALE(port_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(adc_dac_tlv, -7650, 150, 1); +static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 200, 1); +static const DECLARE_TLV_DB_SCALE(port_tlv, -1800, 600, 0); +static const DECLARE_TLV_DB_SCALE(stn_tlv, -7200, 150, 0); static const struct snd_kcontrol_new lm49453_sidetone_mixer_controls[] = { /* Sidetone supports mono only */ SOC_DAPM_SINGLE_TLV("Sidetone ADCL Volume", LM49453_P0_STN_VOL_ADCL_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), SOC_DAPM_SINGLE_TLV("Sidetone ADCR Volume", LM49453_P0_STN_VOL_ADCR_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), SOC_DAPM_SINGLE_TLV("Sidetone DMIC1L Volume", LM49453_P0_STN_VOL_DMIC1L_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), SOC_DAPM_SINGLE_TLV("Sidetone DMIC1R Volume", LM49453_P0_STN_VOL_DMIC1R_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), SOC_DAPM_SINGLE_TLV("Sidetone DMIC2L Volume", LM49453_P0_STN_VOL_DMIC2L_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), SOC_DAPM_SINGLE_TLV("Sidetone DMIC2R Volume", LM49453_P0_STN_VOL_DMIC2R_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), }; static const struct snd_kcontrol_new lm49453_snd_controls[] = { /* mic1 and mic2 supports mono only */ - SOC_SINGLE_TLV("Mic1 Volume", LM49453_P0_ADC_LEVELL_REG, 0, 6, - 0, digital_tlv), - SOC_SINGLE_TLV("Mic2 Volume", LM49453_P0_ADC_LEVELR_REG, 0, 6, - 0, digital_tlv), + SOC_SINGLE_TLV("Mic1 Volume", LM49453_P0_MICL_REG, 0, 15, 0, mic_tlv), + SOC_SINGLE_TLV("Mic2 Volume", LM49453_P0_MICR_REG, 0, 15, 0, mic_tlv), + + SOC_SINGLE_TLV("ADCL Volume", LM49453_P0_ADC_LEVELL_REG, 0, 63, + 0, adc_dac_tlv), + SOC_SINGLE_TLV("ADCR Volume", LM49453_P0_ADC_LEVELR_REG, 0, 63, + 0, adc_dac_tlv), SOC_DOUBLE_R_TLV("DMIC1 Volume", LM49453_P0_DMIC1_LEVELL_REG, - LM49453_P0_DMIC1_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DMIC1_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_DOUBLE_R_TLV("DMIC2 Volume", LM49453_P0_DMIC2_LEVELL_REG, - LM49453_P0_DMIC2_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DMIC2_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_DAPM_ENUM("Mic2Mode", lm49453_mic2mode_enum), SOC_DAPM_ENUM("DMIC12 SRC", lm49453_dmic12_cfg_enum), @@ -538,16 +543,16 @@ static const struct snd_kcontrol_new lm49453_snd_controls[] = { 2, 1, 0), SOC_DOUBLE_R_TLV("DAC HP Volume", LM49453_P0_DAC_HP_LEVELL_REG, - LM49453_P0_DAC_HP_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DAC_HP_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_DOUBLE_R_TLV("DAC LO Volume", LM49453_P0_DAC_LO_LEVELL_REG, - LM49453_P0_DAC_LO_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DAC_LO_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_DOUBLE_R_TLV("DAC LS Volume", LM49453_P0_DAC_LS_LEVELL_REG, - LM49453_P0_DAC_LS_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DAC_LS_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_DOUBLE_R_TLV("DAC HA Volume", LM49453_P0_DAC_HA_LEVELL_REG, - LM49453_P0_DAC_HA_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DAC_HA_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_SINGLE_TLV("EP Volume", LM49453_P0_DAC_LS_LEVELL_REG, - 0, 6, 0, digital_tlv), + 0, 63, 0, adc_dac_tlv), SOC_SINGLE_TLV("PORT1_1_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG, 0, 3, 0, port_tlv),

1 0

[alsa-devel] [PATCH] ASoC: arizona: Store the DAI clock ID when setting
by Mark Brown 06 Dec '12

06 Dec '12

So the code to suppress duplicate changes is effective. Reported-by: Kyung Kwee Ryu <Kyung-Kwee.Ryu@wolfsonmicro.comyu@wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 22 ++++++++++++++-------- 1 file changed, 14 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index ef68f7a..dfaeca9 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1,3 +1,4 @@ +#define DEBUG /* * arizona.c - Wolfson Arizona class device shared support * @@ -679,6 +680,9 @@ static int arizona_hw_params_rate(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona_dai_priv *dai_priv = &priv->dai[dai->id - 1]; + int base = dai->driver->base; int i, sr_val; /* @@ -699,14 +703,16 @@ static int arizona_hw_params_rate(struct snd_pcm_substream *substream, case ARIZONA_CLK_SYSCLK: snd_soc_update_bits(codec, ARIZONA_SAMPLE_RATE_1, ARIZONA_SAMPLE_RATE_1_MASK, sr_val); - snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL, - ARIZONA_AIF1_RATE_MASK, 0); + if (base) + snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL, + ARIZONA_AIF1_RATE_MASK, 0); break; case ARIZONA_CLK_ASYNCCLK: snd_soc_update_bits(codec, ARIZONA_ASYNC_SAMPLE_RATE_1, ARIZONA_ASYNC_SAMPLE_RATE_MASK, sr_val); - snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL, - ARIZONA_AIF1_RATE_MASK, 8); + if (base) + snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL, + ARIZONA_AIF1_RATE_MASK, 8); break; default: arizona_aif_err(dai, "Invalid clock %d\n", dai_priv->clk); @@ -721,12 +727,10 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; - struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); - struct arizona_dai_priv *dai_priv = &priv->dai[dai->id - 1]; int base = dai->driver->base; const int *rates; int i, ret; - int bclk, lrclk, wl, frame, sr_val; + int bclk, lrclk, wl, frame; if (params_rate(params) % 8000) rates = &arizona_44k1_bclk_rates[0]; @@ -754,7 +758,7 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, wl = snd_pcm_format_width(params_format(params)); frame = wl << ARIZONA_AIF1TX_WL_SHIFT | wl; - ret = arizona_hw_params_rate(codec, params, dai); + ret = arizona_hw_params_rate(substream, params, dai); if (ret != 0) return ret; @@ -823,6 +827,8 @@ static int arizona_dai_set_sysclk(struct snd_soc_dai *dai, routes[1].source = arizona_dai_clk_str(clk_id); snd_soc_dapm_add_routes(&codec->dapm, routes, ARRAY_SIZE(routes)); + dai_priv->clk = clk_id; + return snd_soc_dapm_sync(&codec->dapm); } -- 1.7.10.4

1 0

[alsa-devel] [RFC][PATCH 0/2] ASoC: add DT support on simple-card
by Kuninori Morimoto 06 Dec '12

06 Dec '12

Hi Mark These patches add DT support for soc-simple-card driver. Kuninori Morimoto (2): ASoC: add snd_soc_of_parse_daifmt() ASoC: simple-card: add DT support include/sound/soc.h | 2 + sound/soc/generic/simple-card.c | 90 +++++++++++++++++++++++++++++++++++++-- sound/soc/soc-core.c | 72 +++++++++++++++++++++++++++++++ 3 files changed, 161 insertions(+), 3 deletions(-) 1st patch adds snd_soc_of_parse_daifmt() to select SND_SOC_DAIFMT_xxx, but I'm not sure whether this is good idea. So these are [RFC] patches. I thought that it will be trouble if... 1) platform/driver DT used snd_fmt = <0xYYYY> style for SND_SOC_DAIFMT_xxx 2) someone might update SND_SOC_DAIFMT_xxx value for some reasons. 3) bootloader should update DT value according to kernel version if 2) happened I put snd_soc_of_parse_daifmt() on soc-core.c because I thought that other driver can reuse it. I need your comment for it. Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH] ASoC: arizona: Make FLL lock timeout very high
by Mark Brown 06 Dec '12

06 Dec '12

Provide robustness against low quality FLL sync clocks by increasing the timeout for lock to an absurdly high point; we should never get anywhere near hitting the timeout in a real system unless it is failing. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 5fc11bc..ef68f7a 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1083,7 +1083,7 @@ int arizona_set_fll(struct arizona_fll *fll, int source, ARIZONA_FLL1_SYNC_ENA); ret = wait_for_completion_timeout(&fll->ok, - msecs_to_jiffies(25)); + msecs_to_jiffies(250)); if (ret == 0) arizona_fll_warn(fll, "Timed out waiting for lock\n"); -- 1.7.10.4

1 0

[alsa-devel] [PATCH 00/14] ASoC: Use devm_* APIs
by Sachin Kamat 06 Dec '12

06 Dec '12

This series is build tested against the for-next branch of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tree. Sachin Kamat (14): ASoC: wm8350: Use devm_regulator_bulk_get ASoC: wm8962: Use devm_regulator_bulk_get ASoC: wm8400: Use devm_regulator_bulk_get ASoC: wm8993: Use devm_regulator_bulk_get ASoC: ak4535: Use devm_regmap_init_i2c() ASoC: da7210: Use devm_regmap_init_i2c() ASoC: lm49453: Use devm_regmap_init_i2c() ASoC: max9768: Use devm_regmap_init_i2c() ASoC: wm8955: Use devm_regmap_init_i2c() ASoC: wm8960: Use devm_regmap_init_i2c() ASoC: wm8962: Use devm_regmap_init_i2c() ASoC: wm8978: Use devm_regmap_init_i2c() ASoC: wm8993: Use devm_regmap_init_i2c() ASoC: wm9081: Use devm_regmap_init_i2c() sound/soc/codecs/ak4535.c | 7 +------ sound/soc/codecs/da7210.c | 13 ++----------- sound/soc/codecs/lm49453.c | 10 ++-------- sound/soc/codecs/max9768.c | 7 ++----- sound/soc/codecs/wm8350.c | 4 +--- sound/soc/codecs/wm8400.c | 14 +++----------- sound/soc/codecs/wm8955.c | 11 +---------- sound/soc/codecs/wm8960.c | 2 +- sound/soc/codecs/wm8962.c | 24 ++++++++---------------- sound/soc/codecs/wm8978.c | 13 +++---------- sound/soc/codecs/wm8993.c | 14 ++++---------- sound/soc/codecs/wm9081.c | 22 ++++++---------------- 12 files changed, 34 insertions(+), 107 deletions(-) -- 1.7.4.1

2 30

[alsa-devel] [PATCH v3] HDA: Add PCI device prefix for clarity
by Daniel J Blueman 05 Dec '12

05 Dec '12

When printing, use a prefix of the PCI domain, bus, device and function as in other drivers, to differentiate multiple devices. Important for reporting and debugging. A future step is to tidy this up with dev_printk et al. v2: Move conversion specifier into call site, preventing build issues v3: Refactor for Takashi's for-next branch Signed-off-by: Daniel J Blueman <daniel(a)quora.org> --- sound/pci/hda/hda_intel.c | 125 +++++++++++++++++++++++---------------------- 1 file changed, 64 insertions(+), 61 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 22ecadc..eb48109 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -192,7 +192,7 @@ MODULE_DESCRIPTION("Intel HDA driver"); #ifdef CONFIG_SND_VERBOSE_PRINTK #define SFX /* nop */ #else -#define SFX "hda-intel: " +#define SFX "hda-intel " #endif #if defined(CONFIG_PM) && defined(CONFIG_VGA_SWITCHEROO) @@ -717,7 +717,7 @@ static int azx_alloc_cmd_io(struct azx *chip) snd_dma_pci_data(chip->pci), PAGE_SIZE, &chip->rb); if (err < 0) { - snd_printk(KERN_ERR SFX "cannot allocate CORB/RIRB\n"); + snd_printk(KERN_ERR SFX "%s: cannot allocate CORB/RIRB\n", pci_name(chip->pci)); return err; } mark_pages_wc(chip, &chip->rb, true); @@ -894,9 +894,9 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus, } if (!chip->polling_mode && chip->poll_count < 2) { - snd_printdd(SFX "azx_get_response timeout, " + snd_printdd(SFX "%s: azx_get_response timeout, " "polling the codec once: last cmd=0x%08x\n", - chip->last_cmd[addr]); + pci_name(chip->pci), chip->last_cmd[addr]); do_poll = 1; chip->poll_count++; goto again; @@ -904,17 +904,17 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus, if (!chip->polling_mode) { - snd_printk(KERN_WARNING SFX "azx_get_response timeout, " + snd_printk(KERN_WARNING SFX "%s: azx_get_response timeout, " "switching to polling mode: last cmd=0x%08x\n", - chip->last_cmd[addr]); + pci_name(chip->pci), chip->last_cmd[addr]); chip->polling_mode = 1; goto again; } if (chip->msi) { - snd_printk(KERN_WARNING SFX "No response from codec, " + snd_printk(KERN_WARNING SFX "%s: No response from codec, " "disabling MSI: last cmd=0x%08x\n", - chip->last_cmd[addr]); + pci_name(chip->pci), chip->last_cmd[addr]); free_irq(chip->irq, chip); chip->irq = -1; pci_disable_msi(chip->pci); @@ -980,8 +980,8 @@ static int azx_single_wait_for_response(struct azx *chip, unsigned int addr) udelay(1); } if (printk_ratelimit()) - snd_printd(SFX "get_response timeout: IRS=0x%x\n", - azx_readw(chip, IRS)); + snd_printd(SFX "%s: get_response timeout: IRS=0x%x\n", + pci_name(chip->pci), azx_readw(chip, IRS)); chip->rirb.res[addr] = -1; return -EIO; } @@ -1008,8 +1008,8 @@ static int azx_single_send_cmd(struct hda_bus *bus, u32 val) udelay(1); } if (printk_ratelimit()) - snd_printd(SFX "send_cmd timeout: IRS=0x%x, val=0x%x\n", - azx_readw(chip, IRS), val); + snd_printd(SFX "%s: send_cmd timeout: IRS=0x%x, val=0x%x\n", + pci_name(chip->pci), azx_readw(chip, IRS), val); return -EIO; } @@ -1095,7 +1095,7 @@ static int azx_reset(struct azx *chip, int full_reset) __skip: /* check to see if controller is ready */ if (!azx_readb(chip, GCTL)) { - snd_printd(SFX "azx_reset: controller not ready!\n"); + snd_printd(SFX "%s: azx_reset: controller not ready!\n", pci_name(chip->pci)); return -EBUSY; } @@ -1107,7 +1107,7 @@ static int azx_reset(struct azx *chip, int full_reset) /* detect codecs */ if (!chip->codec_mask) { chip->codec_mask = azx_readw(chip, STATESTS); - snd_printdd(SFX "codec_mask = 0x%x\n", chip->codec_mask); + snd_printdd(SFX "%s: codec_mask = 0x%x\n", pci_name(chip->pci), chip->codec_mask); } return 0; @@ -1251,7 +1251,7 @@ static void azx_init_pci(struct azx *chip) * The PCI register TCSEL is defined in the Intel manuals. */ if (!(chip->driver_caps & AZX_DCAPS_NO_TCSEL)) { - snd_printdd(SFX "Clearing TCSEL\n"); + snd_printdd(SFX "%s: Clearing TCSEL\n", pci_name(chip->pci)); update_pci_byte(chip->pci, ICH6_PCIREG_TCSEL, 0x07, 0); } @@ -1259,7 +1259,7 @@ static void azx_init_pci(struct azx *chip) * we need to enable snoop. */ if (chip->driver_caps & AZX_DCAPS_ATI_SNOOP) { - snd_printdd(SFX "Setting ATI snoop: %d\n", azx_snoop(chip)); + snd_printdd(SFX "%s: Setting ATI snoop: %d\n", pci_name(chip->pci), azx_snoop(chip)); update_pci_byte(chip->pci, ATI_SB450_HDAUDIO_MISC_CNTR2_ADDR, 0x07, azx_snoop(chip) ? ATI_SB450_HDAUDIO_ENABLE_SNOOP : 0); @@ -1267,7 +1267,7 @@ static void azx_init_pci(struct azx *chip) /* For NVIDIA HDA, enable snoop */ if (chip->driver_caps & AZX_DCAPS_NVIDIA_SNOOP) { - snd_printdd(SFX "Setting Nvidia snoop: %d\n", azx_snoop(chip)); + snd_printdd(SFX "%s: Setting Nvidia snoop: %d\n", pci_name(chip->pci), azx_snoop(chip)); update_pci_byte(chip->pci, NVIDIA_HDA_TRANSREG_ADDR, 0x0f, NVIDIA_HDA_ENABLE_COHBITS); @@ -1292,8 +1292,8 @@ static void azx_init_pci(struct azx *chip) pci_read_config_word(chip->pci, INTEL_SCH_HDA_DEVC, &snoop); } - snd_printdd(SFX "SCH snoop: %s\n", - (snoop & INTEL_SCH_HDA_DEVC_NOSNOOP) + snd_printdd(SFX "%s: SCH snoop: %s\n", + pci_name(chip->pci), (snoop & INTEL_SCH_HDA_DEVC_NOSNOOP) ? "Disabled" : "Enabled"); } } @@ -1453,8 +1453,8 @@ static int azx_setup_periods(struct azx *chip, pos_align; pos_adj = frames_to_bytes(runtime, pos_adj); if (pos_adj >= period_bytes) { - snd_printk(KERN_WARNING SFX "Too big adjustment %d\n", - bdl_pos_adj[chip->dev_index]); + snd_printk(KERN_WARNING SFX "%s: Too big adjustment %d\n", + pci_name(chip->pci), bdl_pos_adj[chip->dev_index]); pos_adj = 0; } else { ofs = setup_bdle(chip, substream, azx_dev, @@ -1478,8 +1478,8 @@ static int azx_setup_periods(struct azx *chip, return 0; error: - snd_printk(KERN_ERR SFX "Too many BDL entries: buffer=%d, period=%d\n", - azx_dev->bufsize, period_bytes); + snd_printk(KERN_ERR SFX "%s: Too many BDL entries: buffer=%d, period=%d\n", + pci_name(chip->pci), azx_dev->bufsize, period_bytes); return -EINVAL; } @@ -1576,7 +1576,7 @@ static int probe_codec(struct azx *chip, int addr) mutex_unlock(&chip->bus->cmd_mutex); if (res == -1) return -EIO; - snd_printdd(SFX "codec #%d probed OK\n", addr); + snd_printdd(SFX "%s: codec #%d probed OK\n", pci_name(chip->pci), addr); return 0; } @@ -1653,7 +1653,7 @@ static int DELAYED_INIT_MARK azx_codec_create(struct azx *chip, const char *mode return err; if (chip->driver_caps & AZX_DCAPS_RIRB_DELAY) { - snd_printd(SFX "Enable delay in RIRB handling\n"); + snd_printd(SFX "%s: Enable delay in RIRB handling\n", pci_name(chip->pci)); chip->bus->needs_damn_long_delay = 1; } @@ -1670,8 +1670,8 @@ static int DELAYED_INIT_MARK azx_codec_create(struct azx *chip, const char *mode * that don't exist */ snd_printk(KERN_WARNING SFX - "Codec #%d probe error; " - "disabling it...\n", c); + "%s: Codec #%d probe error; " + "disabling it...\n", pci_name(chip->pci), c); chip->codec_mask &= ~(1 << c); /* More badly, accessing to a non-existing * codec often screws up the controller chip, @@ -1691,7 +1691,8 @@ static int DELAYED_INIT_MARK azx_codec_create(struct azx *chip, const char *mode * access works around the stall. Grrr... */ if (chip->driver_caps & AZX_DCAPS_SYNC_WRITE) { - snd_printd(SFX "Enable sync_write for stable communication\n"); + snd_printd(SFX "%s: Enable sync_write for stable communication\n", + pci_name(chip->pci)); chip->bus->sync_write = 1; chip->bus->allow_bus_reset = 1; } @@ -1709,7 +1710,7 @@ static int DELAYED_INIT_MARK azx_codec_create(struct azx *chip, const char *mode } } if (!codecs) { - snd_printk(KERN_ERR SFX "no codecs initialized\n"); + snd_printk(KERN_ERR SFX "%s: no codecs initialized\n", pci_name(chip->pci)); return -ENXIO; } return 0; @@ -2019,16 +2020,16 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) ctls); if (!format_val) { snd_printk(KERN_ERR SFX - "invalid format_val, rate=%d, ch=%d, format=%d\n", - runtime->rate, runtime->channels, runtime->format); + "%s: invalid format_val, rate=%d, ch=%d, format=%d\n", + pci_name(chip->pci), runtime->rate, runtime->channels, runtime->format); return -EINVAL; } bufsize = snd_pcm_lib_buffer_bytes(substream); period_bytes = snd_pcm_lib_period_bytes(substream); - snd_printdd(SFX "azx_pcm_prepare: bufsize=0x%x, format=0x%x\n", - bufsize, format_val); + snd_printdd(SFX "%s: azx_pcm_prepare: bufsize=0x%x, format=0x%x\n", + pci_name(chip->pci), bufsize, format_val); if (bufsize != azx_dev->bufsize || period_bytes != azx_dev->period_bytes || @@ -2287,9 +2288,9 @@ static unsigned int azx_get_position(struct azx *chip, delay += azx_dev->bufsize; if (delay >= azx_dev->period_bytes) { snd_printk(KERN_WARNING SFX - "Unstable LPIB (%d >= %d); " + "%s: Unstable LPIB (%d >= %d); " "disabling LPIB delay counting\n", - delay, azx_dev->period_bytes); + pci_name(chip->pci), delay, azx_dev->period_bytes); delay = 0; chip->driver_caps &= ~AZX_DCAPS_COUNT_LPIB_DELAY; } @@ -2446,7 +2447,8 @@ azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec, list_for_each_entry(apcm, &chip->pcm_list, list) { if (apcm->pcm->device == pcm_dev) { - snd_printk(KERN_ERR SFX "PCM %d already exists\n", pcm_dev); + snd_printk(KERN_ERR SFX "%s: PCM %d already exists\n", + pci_name(chip->pci), pcm_dev); return -EBUSY; } } @@ -2771,15 +2773,14 @@ static void azx_vs_set_state(struct pci_dev *pci, } } else { snd_printk(KERN_INFO SFX - "%s %s via VGA-switcheroo\n", - disabled ? "Disabling" : "Enabling", - pci_name(chip->pci)); + "%s: %s via VGA-switcheroo\n", pci_name(chip->pci), + disabled ? "Disabling" : "Enabling"); if (disabled) { azx_suspend(&pci->dev); chip->disabled = true; if (snd_hda_lock_devices(chip->bus)) - snd_printk(KERN_WARNING SFX - "Cannot lock devices!\n"); + snd_printk(KERN_WARNING SFX "%s: Cannot lock devices!\n", + pci_name(chip->pci)); } else { snd_hda_unlock_devices(chip->bus); chip->disabled = false; @@ -2998,11 +2999,11 @@ static int __devinit check_position_fix(struct azx *chip, int fix) /* Check VIA/ATI HD Audio Controller exist */ if (chip->driver_caps & AZX_DCAPS_POSFIX_VIA) { - snd_printd(SFX "Using VIACOMBO position fix\n"); + snd_printd(SFX "%s: Using VIACOMBO position fix\n", pci_name(chip->pci)); return POS_FIX_VIACOMBO; } if (chip->driver_caps & AZX_DCAPS_POSFIX_LPIB) { - snd_printd(SFX "Using LPIB position fix\n"); + snd_printd(SFX "%s: Using LPIB position fix\n", pci_name(chip->pci)); return POS_FIX_LPIB; } return POS_FIX_AUTO; @@ -3118,8 +3119,8 @@ static void __devinit azx_check_snoop_available(struct azx *chip) } if (snoop != chip->snoop) { - snd_printk(KERN_INFO SFX "Force to %s mode\n", - snoop ? "snoop" : "non-snoop"); + snd_printk(KERN_INFO SFX "%s: Force to %s mode\n", + pci_name(chip->pci), snoop ? "snoop" : "non-snoop"); chip->snoop = snoop; } } @@ -3145,7 +3146,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, chip = kzalloc(sizeof(*chip), GFP_KERNEL); if (!chip) { - snd_printk(KERN_ERR SFX "cannot allocate chip\n"); + snd_printk(KERN_ERR SFX "%s: Cannot allocate chip\n", pci_name(pci)); pci_disable_device(pci); return -ENOMEM; } @@ -3193,7 +3194,8 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); if (err < 0) { - snd_printk(KERN_ERR SFX "Error creating device [card]!\n"); + snd_printk(KERN_ERR SFX "%s: Error creating device [card]!\n", + pci_name(chip->pci)); azx_free(chip); return err; } @@ -3228,7 +3230,7 @@ static int DELAYED_INIT_MARK azx_first_init(struct azx *chip) chip->addr = pci_resource_start(pci, 0); chip->remap_addr = pci_ioremap_bar(pci, 0); if (chip->remap_addr == NULL) { - snd_printk(KERN_ERR SFX "ioremap error\n"); + snd_printk(KERN_ERR SFX "%s: ioremap error\n", pci_name(chip->pci)); return -ENXIO; } @@ -3243,7 +3245,7 @@ static int DELAYED_INIT_MARK azx_first_init(struct azx *chip) synchronize_irq(chip->irq); gcap = azx_readw(chip, GCAP); - snd_printdd(SFX "chipset global capabilities = 0x%x\n", gcap); + snd_printdd(SFX "%s: chipset global capabilities = 0x%x\n", pci_name(chip->pci), gcap); /* disable SB600 64bit support for safety */ if (chip->pci->vendor == PCI_VENDOR_ID_ATI) { @@ -3260,7 +3262,7 @@ static int DELAYED_INIT_MARK azx_first_init(struct azx *chip) /* disable 64bit DMA address on some devices */ if (chip->driver_caps & AZX_DCAPS_NO_64BIT) { - snd_printd(SFX "Disabling 64bit DMA\n"); + snd_printd(SFX "%s: Disabling 64bit DMA\n", pci_name(chip->pci)); gcap &= ~ICH6_GCAP_64OK; } @@ -3315,7 +3317,7 @@ static int DELAYED_INIT_MARK azx_first_init(struct azx *chip) chip->azx_dev = kcalloc(chip->num_streams, sizeof(*chip->azx_dev), GFP_KERNEL); if (!chip->azx_dev) { - snd_printk(KERN_ERR SFX "cannot malloc azx_dev\n"); + snd_printk(KERN_ERR SFX "%s: cannot malloc azx_dev\n", pci_name(chip->pci)); return -ENOMEM; } @@ -3325,7 +3327,7 @@ static int DELAYED_INIT_MARK azx_first_init(struct azx *chip) snd_dma_pci_data(chip->pci), BDL_SIZE, &chip->azx_dev[i].bdl); if (err < 0) { - snd_printk(KERN_ERR SFX "cannot allocate BDL\n"); + snd_printk(KERN_ERR SFX "%s: cannot allocate BDL\n", pci_name(chip->pci)); return -ENOMEM; } mark_pages_wc(chip, &chip->azx_dev[i].bdl, true); @@ -3335,7 +3337,7 @@ static int DELAYED_INIT_MARK azx_first_init(struct azx *chip) snd_dma_pci_data(chip->pci), chip->num_streams * 8, &chip->posbuf); if (err < 0) { - snd_printk(KERN_ERR SFX "cannot allocate posbuf\n"); + snd_printk(KERN_ERR SFX "%s: cannot allocate posbuf\n", pci_name(chip->pci)); return -ENOMEM; } mark_pages_wc(chip, &chip->posbuf, true); @@ -3353,7 +3355,7 @@ static int DELAYED_INIT_MARK azx_first_init(struct azx *chip) /* codec detection */ if (!chip->codec_mask) { - snd_printk(KERN_ERR SFX "no codecs found!\n"); + snd_printk(KERN_ERR SFX "%s: no codecs found!\n", pci_name(chip->pci)); return -ENODEV; } @@ -3389,7 +3391,8 @@ static void azx_firmware_cb(const struct firmware *fw, void *context) struct pci_dev *pci = chip->pci; if (!fw) { - snd_printk(KERN_ERR SFX "Cannot load firmware, aborting\n"); + snd_printk(KERN_ERR SFX "%s: Cannot load firmware, aborting\n", + pci_name(chip->pci)); goto error; } @@ -3425,7 +3428,7 @@ static int __devinit azx_probe(struct pci_dev *pci, err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); if (err < 0) { - snd_printk(KERN_ERR SFX "Error creating card!\n"); + snd_printk(KERN_ERR "hda-intel: Error creating card!\n"); return err; } @@ -3441,14 +3444,14 @@ static int __devinit azx_probe(struct pci_dev *pci, err = register_vga_switcheroo(chip); if (err < 0) { snd_printk(KERN_ERR SFX - "Error registering VGA-switcheroo client\n"); + "%s: Error registering VGA-switcheroo client\n", pci_name(pci)); goto out_free; } if (check_hdmi_disabled(pci)) { - snd_printk(KERN_INFO SFX "VGA controller for %s is disabled\n", + snd_printk(KERN_INFO SFX "%s: VGA controller is disabled\n", pci_name(pci)); - snd_printk(KERN_INFO SFX "Delaying initialization\n"); + snd_printk(KERN_INFO SFX "%s: Delaying initialization\n", pci_name(pci)); chip->disabled = true; } @@ -3461,8 +3464,8 @@ static int __devinit azx_probe(struct pci_dev *pci, #ifdef CONFIG_SND_HDA_PATCH_LOADER if (patch[dev] && *patch[dev]) { - snd_printk(KERN_ERR SFX "Applying patch firmware '%s'\n", - patch[dev]); + snd_printk(KERN_ERR SFX "%s: Applying patch firmware '%s'\n", + pci_name(pci), patch[dev]); err = request_firmware_nowait(THIS_MODULE, true, patch[dev], &pci->dev, GFP_KERNEL, card, azx_firmware_cb); -- 1.7.10.4

2 1

[alsa-devel] [PATCH 1/3] ALSA: ASoC: cs4271: fix sparse warning
by Daniel Mack 05 Dec '12

05 Dec '12

Make the flag in the pdata of type bool to fix a sparse warning. Signed-off-by: Daniel Mack <zonque(a)gmail.com> Reported-by: Fengguang Wu <fengguang.wu(a)intel.com> --- include/sound/cs4271.h | 2 +- sound/soc/codecs/cs4271.c | 4 ++-- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/include/sound/cs4271.h b/include/sound/cs4271.h index 6d9e15e..dd8c48d 100644 --- a/include/sound/cs4271.h +++ b/include/sound/cs4271.h @@ -19,7 +19,7 @@ struct cs4271_platform_data { int gpio_nreset; /* GPIO driving Reset pin, if any */ - int amutec_eq_bmutec:1; /* flag to enable AMUTEC=BMUTEC */ + bool amutec_eq_bmutec; /* flag to enable AMUTEC=BMUTEC */ }; #endif /* __CS4271_H */ diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 6ad3878..70de900 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -474,7 +474,7 @@ static int cs4271_probe(struct snd_soc_codec *codec) struct cs4271_platform_data *cs4271plat = codec->dev->platform_data; int ret; int gpio_nreset = -EINVAL; - int amutec_eq_bmutec = 0; + bool amutec_eq_bmutec = false; #ifdef CONFIG_OF if (of_match_device(cs4271_dt_ids, codec->dev)) { @@ -483,7 +483,7 @@ static int cs4271_probe(struct snd_soc_codec *codec) if (!of_get_property(codec->dev->of_node, "cirrus,amutec-eq-bmutec", NULL)) - amutec_eq_bmutec = 1; + amutec_eq_bmutec = true; } #endif -- 1.7.11.7

2 6

[alsa-devel] [PATCH] ASoC:lm49453: Update lm49453_reg_defs values as per LM49453 HW version-RevB
by MR.Swami.Reddy＠ti.com 05 Dec '12

05 Dec '12

From: MR Swami Reddy <MR.Swami.Reddy(a)ti.com> [PATCH] ASoC:lm49453: Update lm49453_reg_defs values as per LM49453 HW version-RevB Signed-off-by: M R Swami Reddy <mr.swami.reddy(a)ti.com> --- sound/soc/codecs/lm49453.c | 61 +++++++++++--------------------------------- 1 files changed, 15 insertions(+), 46 deletions(-) diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index 99b0a9d..808e50a 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -111,9 +111,9 @@ static struct reg_default lm49453_reg_defs[] = { { 101, 0x00 }, { 102, 0x00 }, { 103, 0x01 }, - { 105, 0x01 }, - { 106, 0x00 }, - { 107, 0x01 }, + { 104, 0x01 }, + { 105, 0x00 }, + { 106, 0x01 }, { 107, 0x00 }, { 108, 0x00 }, { 109, 0x00 }, @@ -163,56 +163,25 @@ static struct reg_default lm49453_reg_defs[] = { { 184, 0x00 }, { 185, 0x00 }, { 186, 0x00 }, - { 189, 0x00 }, + { 187, 0x00 }, { 188, 0x00 }, - { 194, 0x00 }, - { 195, 0x00 }, - { 196, 0x00 }, - { 197, 0x00 }, - { 200, 0x00 }, - { 201, 0x00 }, - { 202, 0x00 }, - { 203, 0x00 }, - { 204, 0x00 }, - { 205, 0x00 }, - { 208, 0x00 }, + { 189, 0x00 }, + { 208, 0x06 }, { 209, 0x00 }, - { 210, 0x00 }, - { 211, 0x00 }, - { 213, 0x00 }, - { 214, 0x00 }, - { 215, 0x00 }, - { 216, 0x00 }, - { 217, 0x00 }, - { 218, 0x00 }, - { 219, 0x00 }, + { 210, 0x08 }, + { 211, 0x54 }, + { 212, 0x14 }, + { 213, 0x0d }, + { 214, 0x0d }, + { 215, 0x14 }, + { 216, 0x60 }, { 221, 0x00 }, { 222, 0x00 }, + { 223, 0x00 }, { 224, 0x00 }, - { 225, 0x00 }, - { 226, 0x00 }, - { 227, 0x00 }, - { 228, 0x00 }, - { 229, 0x00 }, - { 230, 0x13 }, - { 231, 0x00 }, - { 232, 0x80 }, - { 233, 0x0C }, - { 234, 0xDD }, - { 235, 0x00 }, - { 236, 0x04 }, - { 237, 0x00 }, - { 238, 0x00 }, - { 239, 0x00 }, - { 240, 0x00 }, - { 241, 0x00 }, - { 242, 0x00 }, - { 243, 0x00 }, - { 244, 0x00 }, - { 245, 0x00 }, { 248, 0x00 }, { 249, 0x00 }, - { 254, 0x00 }, + { 250, 0x00 }, { 255, 0x00 }, };

1 0

[alsa-devel] [query] codec suspend not called when bias level is SND_SOC_BIAS_ON
by Shiraz Hashim 05 Dec '12

05 Dec '12

Hi Mark, I was going through the suspend procedure for soc-core and I could see that codec suspend is called only when its bias level is SND_SOC_BIAS_OFF. if (!codec->suspended && codec->driver->suspend) { switch (codec->dapm.bias_level) { case SND_SOC_BIAS_STANDBY: /* * If the CODEC is capable of idle * bias off then being in STANDBY * means it's doing something, * otherwise fall through. */ if (codec->dapm.idle_bias_off) { dev_dbg(codec->dev, "idle_bias_off CODEC on over suspend\n"); break; } case SND_SOC_BIAS_OFF: codec->driver->suspend(codec); codec->suspended = 1; codec->cache_sync = 1; break; default: dev_dbg(codec->dev, "CODEC is on over suspend\n"); break; } } My question is, during on-going audio play when system is put into suspend, codec->driver->suspend would not be called as its bias level is still SND_SOC_BIAS_ON, thus not giving a chance to codec to save its state. Is my understanding correct ? How to deal with this ? -- regards Shiraz

2 1

[alsa-devel] [PATCH 0/5] Realtek auto-mic enhancements / cleanups
by Takashi Iwai 05 Dec '12

05 Dec '12

Hi, this is a series of patches to enhance and clean up the automatic input selection via mic jack on HD-audio Realtek drivers. [PATCH 1/5] ALSA: hda - Rearrange INPUT_PIN_ATTR_* [PATCH 2/5] ALSA: hda - More generic auto-mic switching for Realtek [PATCH 3/5] ALSA: hda - Use standard sort function in [PATCH 4/5] ALSA: hda - Implement shared front mic/hp jack for HP [PATCH 5/5] ALSA: hda - Refactor alc_kcontrol_new() usages The first motivation was to generalize the code a bit better. Then, I got a machine that should be able to switch front mic and the secondary headphone. The machine is suitable for the new auto-mic test, so I included this hack in 4th path as well. Takashi

3 36

[alsa-devel] [PATCH] ASoC: lm49453: Fix analog and digital volume ranges
by MR.Swami.Reddy＠ti.com 05 Dec '12

05 Dec '12

From: MR Swami Reddy <MR.Swami.Reddy(a)ti.com> [PATCH] ASoC: lm49453: Fix analog and digital volume ranges" Signed-off-by: M R Swami Reddy <mr.swami.reddy(a)ti.com> -- sound/soc/codecs/lm49453.c | 7 +++++---- 1 files changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index 808e50a..ba1aec4 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -494,8 +494,9 @@ SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_PORT2_TX2_REG, 7, 1, 0), }; /* TLV Declarations */ -static const DECLARE_TLV_DB_SCALE(port_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(analog_tlv, 0, 200, 1); +static const DECLARE_TLV_DB_SCALE(port_tlv, -1800, 600, 0); static const struct snd_kcontrol_new lm49453_sidetone_mixer_controls[] = { /* Sidetone supports mono only */ @@ -516,9 +517,9 @@ SOC_DAPM_SINGLE_TLV("Sidetone DMIC2R Volume", LM49453_P0_STN_VOL_DMIC2R_REG, static const struct snd_kcontrol_new lm49453_snd_controls[] = { /* mic1 and mic2 supports mono only */ SOC_SINGLE_TLV("Mic1 Volume", LM49453_P0_ADC_LEVELL_REG, 0, 6, - 0, digital_tlv), + 0, analog_tlv), SOC_SINGLE_TLV("Mic2 Volume", LM49453_P0_ADC_LEVELR_REG, 0, 6, - 0, digital_tlv), + 0, analog_tlv), SOC_DOUBLE_R_TLV("DMIC1 Volume", LM49453_P0_DMIC1_LEVELL_REG, LM49453_P0_DMIC1_LEVELR_REG, 0, 6, 0, digital_tlv),

2 1

[alsa-devel] [PATCH 2/2] ASoC: wm5110: Add LHPF coefficient configuration
by Mark Brown 05 Dec '12

05 Dec '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5110.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index fe54596..ed093ee 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -160,6 +160,11 @@ ARIZONA_MIXER_CONTROLS("LHPF2", ARIZONA_HPLP2MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("LHPF3", ARIZONA_HPLP3MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("LHPF4", ARIZONA_HPLP4MIX_INPUT_1_SOURCE), +SND_SOC_BYTES("LHPF1 Coefficients", ARIZONA_HPLPF1_2, 1), +SND_SOC_BYTES("LHPF2 Coefficients", ARIZONA_HPLPF2_2, 1), +SND_SOC_BYTES("LHPF3 Coefficients", ARIZONA_HPLPF3_2, 1), +SND_SOC_BYTES("LHPF4 Coefficients", ARIZONA_HPLPF4_2, 1), + SOC_ENUM("LHPF1 Mode", arizona_lhpf1_mode), SOC_ENUM("LHPF2 Mode", arizona_lhpf2_mode), SOC_ENUM("LHPF3 Mode", arizona_lhpf3_mode), -- 1.7.10.4

1 0

[alsa-devel] changes in hda-emu kernel code management
by Takashi Iwai 04 Dec '12

04 Dec '12

Hi, right now I changed the way to manage the kernel code in hda-emu slightly, and pushed out the commits. Formerly, each kernel file (hda_*.[ch] and patch_*.c) is symlinked to kernel/ subdirectory in configure script, and it's built there. Now, instead of creating each symlink, a master symlink is created as a subdirectory hda, and the build is done in kernel/*.c which includes "../hda/xxx.c". This makes easier to switch between repos, also little messy. If you find any regression by this change, let me know. Takashi

1 0

[alsa-devel] [PATCH v2] HDA: Add PCI device prefix for clarity
by Daniel J Blueman 04 Dec '12

04 Dec '12

When printing, use a prefix of the PCI domain, bus, device and function as in other drivers, to differentiate multiple devices. Important for reporting and debugging. A future step is to tidy this up with dev_printk et al. v2: Move conversion specifier into call site, preventing build issues Signed-off-by: Daniel J Blueman <daniel(a)quora.org> --- sound/pci/hda/hda_intel.c | 134 +++++++++++++++++++++++---------------------- 1 file changed, 69 insertions(+), 65 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index f9d870e..eb92ab4 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -185,7 +185,7 @@ MODULE_DESCRIPTION("Intel HDA driver"); #ifdef CONFIG_SND_VERBOSE_PRINTK #define SFX /* nop */ #else -#define SFX "hda-intel: " +#define SFX "hda-intel " #endif #if defined(CONFIG_PM) && defined(CONFIG_VGA_SWITCHEROO) @@ -703,7 +703,7 @@ static int azx_alloc_cmd_io(struct azx *chip) snd_dma_pci_data(chip->pci), PAGE_SIZE, &chip->rb); if (err < 0) { - snd_printk(KERN_ERR SFX "cannot allocate CORB/RIRB\n"); + snd_printk(KERN_ERR SFX "%s: cannot allocate CORB/RIRB\n", pci_name(chip->pci)); return err; } mark_pages_wc(chip, &chip->rb, true); @@ -835,8 +835,8 @@ static void azx_update_rirb(struct azx *chip) smp_wmb(); chip->rirb.cmds[addr]--; } else - snd_printk(KERN_ERR SFX "spurious response %#x:%#x, " - "last cmd=%#08x\n", + snd_printk(KERN_ERR SFX "%s: spurious response %#x:%#x, " + "last cmd=%#08x\n", pci_name(chip->pci), res, res_ex, chip->last_cmd[addr]); } @@ -879,9 +879,9 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus, } if (!chip->polling_mode && chip->poll_count < 2) { - snd_printdd(SFX "azx_get_response timeout, " + snd_printdd(SFX "%s: azx_get_response timeout, " "polling the codec once: last cmd=0x%08x\n", - chip->last_cmd[addr]); + pci_name(chip->pci), chip->last_cmd[addr]); do_poll = 1; chip->poll_count++; goto again; @@ -889,17 +889,17 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus, if (!chip->polling_mode) { - snd_printk(KERN_WARNING SFX "azx_get_response timeout, " + snd_printk(KERN_WARNING SFX "%s: azx_get_response timeout, " "switching to polling mode: last cmd=0x%08x\n", - chip->last_cmd[addr]); + pci_name(chip->pci), chip->last_cmd[addr]); chip->polling_mode = 1; goto again; } if (chip->msi) { - snd_printk(KERN_WARNING SFX "No response from codec, " + snd_printk(KERN_WARNING SFX "%s: No response from codec, " "disabling MSI: last cmd=0x%08x\n", - chip->last_cmd[addr]); + pci_name(chip->pci), chip->last_cmd[addr]); free_irq(chip->irq, chip); chip->irq = -1; pci_disable_msi(chip->pci); @@ -965,8 +965,8 @@ static int azx_single_wait_for_response(struct azx *chip, unsigned int addr) udelay(1); } if (printk_ratelimit()) - snd_printd(SFX "get_response timeout: IRS=0x%x\n", - azx_readw(chip, IRS)); + snd_printd(SFX "%s: get_response timeout: IRS=0x%x\n", + pci_name(chip->pci), azx_readw(chip, IRS)); chip->rirb.res[addr] = -1; return -EIO; } @@ -993,8 +993,8 @@ static int azx_single_send_cmd(struct hda_bus *bus, u32 val) udelay(1); } if (printk_ratelimit()) - snd_printd(SFX "send_cmd timeout: IRS=0x%x, val=0x%x\n", - azx_readw(chip, IRS), val); + snd_printd(SFX "%s: send_cmd timeout: IRS=0x%x, val=0x%x\n", + pci_name(chip->pci), azx_readw(chip, IRS), val); return -EIO; } @@ -1080,7 +1080,7 @@ static int azx_reset(struct azx *chip, int full_reset) __skip: /* check to see if controller is ready */ if (!azx_readb(chip, GCTL)) { - snd_printd(SFX "azx_reset: controller not ready!\n"); + snd_printd(SFX "%s: azx_reset: controller not ready!\n", pci_name(chip->pci)); return -EBUSY; } @@ -1092,7 +1092,7 @@ static int azx_reset(struct azx *chip, int full_reset) /* detect codecs */ if (!chip->codec_mask) { chip->codec_mask = azx_readw(chip, STATESTS); - snd_printdd(SFX "codec_mask = 0x%x\n", chip->codec_mask); + snd_printdd(SFX "%s: codec_mask = 0x%x\n", pci_name(chip->pci), chip->codec_mask); } return 0; @@ -1236,7 +1236,7 @@ static void azx_init_pci(struct azx *chip) * The PCI register TCSEL is defined in the Intel manuals. */ if (!(chip->driver_caps & AZX_DCAPS_NO_TCSEL)) { - snd_printdd(SFX "Clearing TCSEL\n"); + snd_printdd(SFX "%s: Clearing TCSEL\n", pci_name(chip->pci)); update_pci_byte(chip->pci, ICH6_PCIREG_TCSEL, 0x07, 0); } @@ -1244,7 +1244,7 @@ static void azx_init_pci(struct azx *chip) * we need to enable snoop. */ if (chip->driver_caps & AZX_DCAPS_ATI_SNOOP) { - snd_printdd(SFX "Setting ATI snoop: %d\n", azx_snoop(chip)); + snd_printdd(SFX "%s: Setting ATI snoop: %d\n", pci_name(chip->pci), azx_snoop(chip)); update_pci_byte(chip->pci, ATI_SB450_HDAUDIO_MISC_CNTR2_ADDR, 0x07, azx_snoop(chip) ? ATI_SB450_HDAUDIO_ENABLE_SNOOP : 0); @@ -1252,7 +1252,7 @@ static void azx_init_pci(struct azx *chip) /* For NVIDIA HDA, enable snoop */ if (chip->driver_caps & AZX_DCAPS_NVIDIA_SNOOP) { - snd_printdd(SFX "Setting Nvidia snoop: %d\n", azx_snoop(chip)); + snd_printdd(SFX "%s: Setting Nvidia snoop: %d\n", pci_name(chip->pci), azx_snoop(chip)); update_pci_byte(chip->pci, NVIDIA_HDA_TRANSREG_ADDR, 0x0f, NVIDIA_HDA_ENABLE_COHBITS); @@ -1277,8 +1277,8 @@ static void azx_init_pci(struct azx *chip) pci_read_config_word(chip->pci, INTEL_SCH_HDA_DEVC, &snoop); } - snd_printdd(SFX "SCH snoop: %s\n", - (snoop & INTEL_SCH_HDA_DEVC_NOSNOOP) + snd_printdd(SFX "%s: SCH snoop: %s\n", + pci_name(chip->pci), (snoop & INTEL_SCH_HDA_DEVC_NOSNOOP) ? "Disabled" : "Enabled"); } } @@ -1438,8 +1438,8 @@ static int azx_setup_periods(struct azx *chip, pos_align; pos_adj = frames_to_bytes(runtime, pos_adj); if (pos_adj >= period_bytes) { - snd_printk(KERN_WARNING SFX "Too big adjustment %d\n", - bdl_pos_adj[chip->dev_index]); + snd_printk(KERN_WARNING SFX "%s: Too big adjustment %d\n", + pci_name(chip->pci), bdl_pos_adj[chip->dev_index]); pos_adj = 0; } else { ofs = setup_bdle(chip, substream, azx_dev, @@ -1463,8 +1463,8 @@ static int azx_setup_periods(struct azx *chip, return 0; error: - snd_printk(KERN_ERR SFX "Too many BDL entries: buffer=%d, period=%d\n", - azx_dev->bufsize, period_bytes); + snd_printk(KERN_ERR SFX "%s: Too many BDL entries: buffer=%d, period=%d\n", + pci_name(chip->pci), azx_dev->bufsize, period_bytes); return -EINVAL; } @@ -1561,7 +1561,7 @@ static int probe_codec(struct azx *chip, int addr) mutex_unlock(&chip->bus->cmd_mutex); if (res == -1) return -EIO; - snd_printdd(SFX "codec #%d probed OK\n", addr); + snd_printdd(SFX "%s: codec #%d probed OK\n", pci_name(chip->pci), addr); return 0; } @@ -1622,7 +1622,7 @@ static int DELAYED_INIT_MARK azx_codec_create(struct azx *chip, const char *mode return err; if (chip->driver_caps & AZX_DCAPS_RIRB_DELAY) { - snd_printd(SFX "Enable delay in RIRB handling\n"); + snd_printd(SFX "%s: Enable delay in RIRB handling\n", pci_name(chip->pci)); chip->bus->needs_damn_long_delay = 1; } @@ -1639,8 +1639,8 @@ static int DELAYED_INIT_MARK azx_codec_create(struct azx *chip, const char *mode * that don't exist */ snd_printk(KERN_WARNING SFX - "Codec #%d probe error; " - "disabling it...\n", c); + "%s: Codec #%d probe error; " + "disabling it...\n", pci_name(chip->pci), c); chip->codec_mask &= ~(1 << c); /* More badly, accessing to a non-existing * codec often screws up the controller chip, @@ -1660,7 +1660,8 @@ static int DELAYED_INIT_MARK azx_codec_create(struct azx *chip, const char *mode * access works around the stall. Grrr... */ if (chip->driver_caps & AZX_DCAPS_SYNC_WRITE) { - snd_printd(SFX "Enable sync_write for stable communication\n"); + snd_printd(SFX "%s: Enable sync_write for stable communication\n", + pci_name(chip->pci)); chip->bus->sync_write = 1; chip->bus->allow_bus_reset = 1; } @@ -1677,7 +1678,7 @@ static int DELAYED_INIT_MARK azx_codec_create(struct azx *chip, const char *mode } } if (!codecs) { - snd_printk(KERN_ERR SFX "no codecs initialized\n"); + snd_printk(KERN_ERR SFX "%s: no codecs initialized\n", pci_name(chip->pci)); return -ENXIO; } return 0; @@ -1916,16 +1917,16 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) ctls); if (!format_val) { snd_printk(KERN_ERR SFX - "invalid format_val, rate=%d, ch=%d, format=%d\n", - runtime->rate, runtime->channels, runtime->format); + "%s: invalid format_val, rate=%d, ch=%d, format=%d\n", + pci_name(chip->pci), runtime->rate, runtime->channels, runtime->format); return -EINVAL; } bufsize = snd_pcm_lib_buffer_bytes(substream); period_bytes = snd_pcm_lib_period_bytes(substream); - snd_printdd(SFX "azx_pcm_prepare: bufsize=0x%x, format=0x%x\n", - bufsize, format_val); + snd_printdd(SFX "%s: azx_pcm_prepare: bufsize=0x%x, format=0x%x\n", + pci_name(chip->pci), bufsize, format_val); if (bufsize != azx_dev->bufsize || period_bytes != azx_dev->period_bytes || @@ -2165,9 +2166,9 @@ static unsigned int azx_get_position(struct azx *chip, delay += azx_dev->bufsize; if (delay >= azx_dev->period_bytes) { snd_printk(KERN_WARNING SFX - "Unstable LPIB (%d >= %d); " + "%s: Unstable LPIB (%d >= %d); " "disabling LPIB delay counting\n", - delay, azx_dev->period_bytes); + pci_name(chip->pci), delay, azx_dev->period_bytes); delay = 0; chip->driver_caps &= ~AZX_DCAPS_COUNT_LPIB_DELAY; } @@ -2324,7 +2325,8 @@ azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec, list_for_each_entry(apcm, &chip->pcm_list, list) { if (apcm->pcm->device == pcm_dev) { - snd_printk(KERN_ERR SFX "PCM %d already exists\n", pcm_dev); + snd_printk(KERN_ERR SFX "%s: PCM %d already exists\n", + pci_name(chip->pci), pcm_dev); return -EBUSY; } } @@ -2648,15 +2650,14 @@ static void azx_vs_set_state(struct pci_dev *pci, } } else { snd_printk(KERN_INFO SFX - "%s %s via VGA-switcheroo\n", - disabled ? "Disabling" : "Enabling", - pci_name(chip->pci)); + "%s: %s via VGA-switcheroo\n", pci_name(chip->pci), + disabled ? "Disabling" : "Enabling"); if (disabled) { azx_suspend(&pci->dev); chip->disabled = true; if (snd_hda_lock_devices(chip->bus)) - snd_printk(KERN_WARNING SFX - "Cannot lock devices!\n"); + snd_printk(KERN_WARNING SFX "%s: Cannot lock devices!\n", + pci_name(chip->pci)); } else { snd_hda_unlock_devices(chip->bus); chip->disabled = false; @@ -2871,11 +2872,11 @@ static int __devinit check_position_fix(struct azx *chip, int fix) /* Check VIA/ATI HD Audio Controller exist */ if (chip->driver_caps & AZX_DCAPS_POSFIX_VIA) { - snd_printd(SFX "Using VIACOMBO position fix\n"); + snd_printd(SFX "%s: Using VIACOMBO position fix\n", pci_name(chip->pci)); return POS_FIX_VIACOMBO; } if (chip->driver_caps & AZX_DCAPS_POSFIX_LPIB) { - snd_printd(SFX "Using LPIB position fix\n"); + snd_printd(SFX "%s: Using LPIB position fix\n", pci_name(chip->pci)); return POS_FIX_LPIB; } return POS_FIX_AUTO; @@ -2991,8 +2992,8 @@ static void __devinit azx_check_snoop_available(struct azx *chip) } if (snoop != chip->snoop) { - snd_printk(KERN_INFO SFX "Force to %s mode\n", - snoop ? "snoop" : "non-snoop"); + snd_printk(KERN_INFO SFX "%s: Force to %s mode\n", + pci_name(chip->pci), snoop ? "snoop" : "non-snoop"); chip->snoop = snoop; } } @@ -3018,7 +3019,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, chip = kzalloc(sizeof(*chip), GFP_KERNEL); if (!chip) { - snd_printk(KERN_ERR SFX "cannot allocate chip\n"); + snd_printk(KERN_ERR SFX "%s: Cannot allocate chip\n", pci_name(pci)); pci_disable_device(pci); return -ENOMEM; } @@ -3064,10 +3065,11 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, } if (check_hdmi_disabled(pci)) { - snd_printk(KERN_INFO SFX "VGA controller for %s is disabled\n", - pci_name(pci)); + snd_printk(KERN_INFO SFX "%s: VGA controller is disabled\n", + pci_name(chip->pci)); if (use_vga_switcheroo(chip)) { - snd_printk(KERN_INFO SFX "Delaying initialization\n"); + snd_printk(KERN_INFO SFX "%s: Delaying initialization\n", + pci_name(chip->pci)); chip->disabled = true; goto ok; } @@ -3085,7 +3087,8 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, ok: err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); if (err < 0) { - snd_printk(KERN_ERR SFX "Error creating device [card]!\n"); + snd_printk(KERN_ERR SFX "%s: Error creating device [card]!\n", + pci_name(chip->pci)); azx_free(chip); return err; } @@ -3120,7 +3123,7 @@ static int DELAYED_INIT_MARK azx_first_init(struct azx *chip) chip->addr = pci_resource_start(pci, 0); chip->remap_addr = pci_ioremap_bar(pci, 0); if (chip->remap_addr == NULL) { - snd_printk(KERN_ERR SFX "ioremap error\n"); + snd_printk(KERN_ERR SFX "%s: ioremap error\n", pci_name(chip->pci)); return -ENXIO; } @@ -3135,7 +3138,7 @@ static int DELAYED_INIT_MARK azx_first_init(struct azx *chip) synchronize_irq(chip->irq); gcap = azx_readw(chip, GCAP); - snd_printdd(SFX "chipset global capabilities = 0x%x\n", gcap); + snd_printdd(SFX "%s: chipset global capabilities = 0x%x\n", pci_name(chip->pci), gcap); /* disable SB600 64bit support for safety */ if (chip->pci->vendor == PCI_VENDOR_ID_ATI) { @@ -3152,7 +3155,7 @@ static int DELAYED_INIT_MARK azx_first_init(struct azx *chip) /* disable 64bit DMA address on some devices */ if (chip->driver_caps & AZX_DCAPS_NO_64BIT) { - snd_printd(SFX "Disabling 64bit DMA\n"); + snd_printd(SFX "%s: Disabling 64bit DMA\n", pci_name(chip->pci)); gcap &= ~ICH6_GCAP_64OK; } @@ -3207,7 +3210,7 @@ static int DELAYED_INIT_MARK azx_first_init(struct azx *chip) chip->azx_dev = kcalloc(chip->num_streams, sizeof(*chip->azx_dev), GFP_KERNEL); if (!chip->azx_dev) { - snd_printk(KERN_ERR SFX "cannot malloc azx_dev\n"); + snd_printk(KERN_ERR SFX "%s: cannot malloc azx_dev\n", pci_name(chip->pci)); return -ENOMEM; } @@ -3217,7 +3220,7 @@ static int DELAYED_INIT_MARK azx_first_init(struct azx *chip) snd_dma_pci_data(chip->pci), BDL_SIZE, &chip->azx_dev[i].bdl); if (err < 0) { - snd_printk(KERN_ERR SFX "cannot allocate BDL\n"); + snd_printk(KERN_ERR SFX "%s: cannot allocate BDL\n", pci_name(chip->pci)); return -ENOMEM; } mark_pages_wc(chip, &chip->azx_dev[i].bdl, true); @@ -3227,7 +3230,7 @@ static int DELAYED_INIT_MARK azx_first_init(struct azx *chip) snd_dma_pci_data(chip->pci), chip->num_streams * 8, &chip->posbuf); if (err < 0) { - snd_printk(KERN_ERR SFX "cannot allocate posbuf\n"); + snd_printk(KERN_ERR SFX "%s: cannot allocate posbuf\n", pci_name(chip->pci)); return -ENOMEM; } mark_pages_wc(chip, &chip->posbuf, true); @@ -3245,7 +3248,7 @@ static int DELAYED_INIT_MARK azx_first_init(struct azx *chip) /* codec detection */ if (!chip->codec_mask) { - snd_printk(KERN_ERR SFX "no codecs found!\n"); + snd_printk(KERN_ERR SFX "%s: no codecs found!\n", pci_name(chip->pci)); return -ENODEV; } @@ -3281,7 +3284,8 @@ static void azx_firmware_cb(const struct firmware *fw, void *context) struct pci_dev *pci = chip->pci; if (!fw) { - snd_printk(KERN_ERR SFX "Cannot load firmware, aborting\n"); + snd_printk(KERN_ERR SFX "%s: Cannot load firmware, aborting\n", + pci_name(chip->pci)); goto error; } @@ -3317,7 +3321,7 @@ static int __devinit azx_probe(struct pci_dev *pci, err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); if (err < 0) { - snd_printk(KERN_ERR SFX "Error creating card!\n"); + snd_printk(KERN_ERR "hda-intel: Error creating card!\n"); return err; } @@ -3331,8 +3335,8 @@ static int __devinit azx_probe(struct pci_dev *pci, #ifdef CONFIG_SND_HDA_PATCH_LOADER if (patch[dev] && *patch[dev]) { - snd_printk(KERN_ERR SFX "Applying patch firmware '%s'\n", - patch[dev]); + snd_printk(KERN_ERR SFX "%s: Applying patch firmware '%s'\n", + pci_name(pci), patch[dev]); err = request_firmware_nowait(THIS_MODULE, true, patch[dev], &pci->dev, GFP_KERNEL, card, azx_firmware_cb); @@ -3355,8 +3359,8 @@ static int __devinit azx_probe(struct pci_dev *pci, err = register_vga_switcheroo(chip); if (err < 0) { - snd_printk(KERN_ERR SFX - "Error registering VGA-switcheroo client\n"); + snd_printk(KERN_ERR SFX "%s: Error registering VGA-switcheroo client\n", + pci_name(chip->pci)); goto out_free; } -- 1.7.10.4

2 1

[alsa-devel] [PATCH] HDA: Add PCI device prefix for clarity
by Daniel J Blueman 04 Dec '12

04 Dec '12

When printing, use a prefix of the PCI domain, bus, device and function as in other drivers, to differentiate multiple devices. Important for reporting and debugging. Signed-off-by: Daniel J Blueman <daniel(a)quora.org> --- sound/pci/hda/hda_intel.c | 110 +++++++++++++++++++++++---------------------- 1 file changed, 57 insertions(+), 53 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index f9d870e..cdfebbd 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -185,7 +185,7 @@ MODULE_DESCRIPTION("Intel HDA driver"); #ifdef CONFIG_SND_VERBOSE_PRINTK #define SFX /* nop */ #else -#define SFX "hda-intel: " +#define SFX "hda-intel %s: " #endif #if defined(CONFIG_PM) && defined(CONFIG_VGA_SWITCHEROO) @@ -703,7 +703,7 @@ static int azx_alloc_cmd_io(struct azx *chip) snd_dma_pci_data(chip->pci), PAGE_SIZE, &chip->rb); if (err < 0) { - snd_printk(KERN_ERR SFX "cannot allocate CORB/RIRB\n"); + snd_printk(KERN_ERR SFX "cannot allocate CORB/RIRB\n", pci_name(chip->pci)); return err; } mark_pages_wc(chip, &chip->rb, true); @@ -836,7 +836,7 @@ static void azx_update_rirb(struct azx *chip) chip->rirb.cmds[addr]--; } else snd_printk(KERN_ERR SFX "spurious response %#x:%#x, " - "last cmd=%#08x\n", + "last cmd=%#08x\n", pci_name(chip->pci), res, res_ex, chip->last_cmd[addr]); } @@ -881,7 +881,7 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus, if (!chip->polling_mode && chip->poll_count < 2) { snd_printdd(SFX "azx_get_response timeout, " "polling the codec once: last cmd=0x%08x\n", - chip->last_cmd[addr]); + pci_name(chip->pci), chip->last_cmd[addr]); do_poll = 1; chip->poll_count++; goto again; @@ -891,7 +891,7 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus, if (!chip->polling_mode) { snd_printk(KERN_WARNING SFX "azx_get_response timeout, " "switching to polling mode: last cmd=0x%08x\n", - chip->last_cmd[addr]); + pci_name(chip->pci), chip->last_cmd[addr]); chip->polling_mode = 1; goto again; } @@ -899,7 +899,7 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus, if (chip->msi) { snd_printk(KERN_WARNING SFX "No response from codec, " "disabling MSI: last cmd=0x%08x\n", - chip->last_cmd[addr]); + pci_name(chip->pci), chip->last_cmd[addr]); free_irq(chip->irq, chip); chip->irq = -1; pci_disable_msi(chip->pci); @@ -966,7 +966,7 @@ static int azx_single_wait_for_response(struct azx *chip, unsigned int addr) } if (printk_ratelimit()) snd_printd(SFX "get_response timeout: IRS=0x%x\n", - azx_readw(chip, IRS)); + pci_name(chip->pci), azx_readw(chip, IRS)); chip->rirb.res[addr] = -1; return -EIO; } @@ -994,7 +994,7 @@ static int azx_single_send_cmd(struct hda_bus *bus, u32 val) } if (printk_ratelimit()) snd_printd(SFX "send_cmd timeout: IRS=0x%x, val=0x%x\n", - azx_readw(chip, IRS), val); + pci_name(chip->pci), azx_readw(chip, IRS), val); return -EIO; } @@ -1080,7 +1080,7 @@ static int azx_reset(struct azx *chip, int full_reset) __skip: /* check to see if controller is ready */ if (!azx_readb(chip, GCTL)) { - snd_printd(SFX "azx_reset: controller not ready!\n"); + snd_printd(SFX "azx_reset: controller not ready!\n", pci_name(chip->pci)); return -EBUSY; } @@ -1092,7 +1092,7 @@ static int azx_reset(struct azx *chip, int full_reset) /* detect codecs */ if (!chip->codec_mask) { chip->codec_mask = azx_readw(chip, STATESTS); - snd_printdd(SFX "codec_mask = 0x%x\n", chip->codec_mask); + snd_printdd(SFX "codec_mask = 0x%x\n", pci_name(chip->pci), chip->codec_mask); } return 0; @@ -1236,7 +1236,7 @@ static void azx_init_pci(struct azx *chip) * The PCI register TCSEL is defined in the Intel manuals. */ if (!(chip->driver_caps & AZX_DCAPS_NO_TCSEL)) { - snd_printdd(SFX "Clearing TCSEL\n"); + snd_printdd(SFX "Clearing TCSEL\n", pci_name(chip->pci)); update_pci_byte(chip->pci, ICH6_PCIREG_TCSEL, 0x07, 0); } @@ -1244,7 +1244,7 @@ static void azx_init_pci(struct azx *chip) * we need to enable snoop. */ if (chip->driver_caps & AZX_DCAPS_ATI_SNOOP) { - snd_printdd(SFX "Setting ATI snoop: %d\n", azx_snoop(chip)); + snd_printdd(SFX "Setting ATI snoop: %d\n", pci_name(chip->pci), azx_snoop(chip)); update_pci_byte(chip->pci, ATI_SB450_HDAUDIO_MISC_CNTR2_ADDR, 0x07, azx_snoop(chip) ? ATI_SB450_HDAUDIO_ENABLE_SNOOP : 0); @@ -1252,7 +1252,7 @@ static void azx_init_pci(struct azx *chip) /* For NVIDIA HDA, enable snoop */ if (chip->driver_caps & AZX_DCAPS_NVIDIA_SNOOP) { - snd_printdd(SFX "Setting Nvidia snoop: %d\n", azx_snoop(chip)); + snd_printdd(SFX "Setting Nvidia snoop: %d\n", pci_name(chip->pci), azx_snoop(chip)); update_pci_byte(chip->pci, NVIDIA_HDA_TRANSREG_ADDR, 0x0f, NVIDIA_HDA_ENABLE_COHBITS); @@ -1278,7 +1278,7 @@ static void azx_init_pci(struct azx *chip) INTEL_SCH_HDA_DEVC, &snoop); } snd_printdd(SFX "SCH snoop: %s\n", - (snoop & INTEL_SCH_HDA_DEVC_NOSNOOP) + pci_name(chip->pci), (snoop & INTEL_SCH_HDA_DEVC_NOSNOOP) ? "Disabled" : "Enabled"); } } @@ -1439,7 +1439,7 @@ static int azx_setup_periods(struct azx *chip, pos_adj = frames_to_bytes(runtime, pos_adj); if (pos_adj >= period_bytes) { snd_printk(KERN_WARNING SFX "Too big adjustment %d\n", - bdl_pos_adj[chip->dev_index]); + pci_name(chip->pci), bdl_pos_adj[chip->dev_index]); pos_adj = 0; } else { ofs = setup_bdle(chip, substream, azx_dev, @@ -1464,7 +1464,7 @@ static int azx_setup_periods(struct azx *chip, error: snd_printk(KERN_ERR SFX "Too many BDL entries: buffer=%d, period=%d\n", - azx_dev->bufsize, period_bytes); + pci_name(chip->pci), azx_dev->bufsize, period_bytes); return -EINVAL; } @@ -1561,7 +1561,7 @@ static int probe_codec(struct azx *chip, int addr) mutex_unlock(&chip->bus->cmd_mutex); if (res == -1) return -EIO; - snd_printdd(SFX "codec #%d probed OK\n", addr); + snd_printdd(SFX "codec #%d probed OK\n", pci_name(chip->pci), addr); return 0; } @@ -1622,7 +1622,7 @@ static int DELAYED_INIT_MARK azx_codec_create(struct azx *chip, const char *mode return err; if (chip->driver_caps & AZX_DCAPS_RIRB_DELAY) { - snd_printd(SFX "Enable delay in RIRB handling\n"); + snd_printd(SFX "Enable delay in RIRB handling\n", pci_name(chip->pci)); chip->bus->needs_damn_long_delay = 1; } @@ -1640,7 +1640,7 @@ static int DELAYED_INIT_MARK azx_codec_create(struct azx *chip, const char *mode */ snd_printk(KERN_WARNING SFX "Codec #%d probe error; " - "disabling it...\n", c); + "disabling it...\n", pci_name(chip->pci), c); chip->codec_mask &= ~(1 << c); /* More badly, accessing to a non-existing * codec often screws up the controller chip, @@ -1660,7 +1660,8 @@ static int DELAYED_INIT_MARK azx_codec_create(struct azx *chip, const char *mode * access works around the stall. Grrr... */ if (chip->driver_caps & AZX_DCAPS_SYNC_WRITE) { - snd_printd(SFX "Enable sync_write for stable communication\n"); + snd_printd(SFX "Enable sync_write for stable communication\n", + pci_name(chip->pci)); chip->bus->sync_write = 1; chip->bus->allow_bus_reset = 1; } @@ -1677,7 +1678,7 @@ static int DELAYED_INIT_MARK azx_codec_create(struct azx *chip, const char *mode } } if (!codecs) { - snd_printk(KERN_ERR SFX "no codecs initialized\n"); + snd_printk(KERN_ERR SFX "no codecs initialized\n", pci_name(chip->pci)); return -ENXIO; } return 0; @@ -1917,7 +1918,7 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) if (!format_val) { snd_printk(KERN_ERR SFX "invalid format_val, rate=%d, ch=%d, format=%d\n", - runtime->rate, runtime->channels, runtime->format); + pci_name(chip->pci), runtime->rate, runtime->channels, runtime->format); return -EINVAL; } @@ -1925,7 +1926,7 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) period_bytes = snd_pcm_lib_period_bytes(substream); snd_printdd(SFX "azx_pcm_prepare: bufsize=0x%x, format=0x%x\n", - bufsize, format_val); + pci_name(chip->pci), bufsize, format_val); if (bufsize != azx_dev->bufsize || period_bytes != azx_dev->period_bytes || @@ -2167,7 +2168,7 @@ static unsigned int azx_get_position(struct azx *chip, snd_printk(KERN_WARNING SFX "Unstable LPIB (%d >= %d); " "disabling LPIB delay counting\n", - delay, azx_dev->period_bytes); + pci_name(chip->pci), delay, azx_dev->period_bytes); delay = 0; chip->driver_caps &= ~AZX_DCAPS_COUNT_LPIB_DELAY; } @@ -2324,7 +2325,8 @@ azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec, list_for_each_entry(apcm, &chip->pcm_list, list) { if (apcm->pcm->device == pcm_dev) { - snd_printk(KERN_ERR SFX "PCM %d already exists\n", pcm_dev); + snd_printk(KERN_ERR SFX "PCM %d already exists\n", + pci_name(chip->pci), pcm_dev); return -EBUSY; } } @@ -2636,27 +2638,26 @@ static void azx_vs_set_state(struct pci_dev *pci, chip->disabled = disabled; if (!disabled) { snd_printk(KERN_INFO SFX - "%s: Start delayed initialization\n", + "Start delayed initialization\n", pci_name(chip->pci)); if (azx_first_init(chip) < 0 || azx_probe_continue(chip) < 0) { snd_printk(KERN_ERR SFX - "%s: initialization error\n", + "initialization error\n", pci_name(chip->pci)); chip->init_failed = true; } } } else { snd_printk(KERN_INFO SFX - "%s %s via VGA-switcheroo\n", - disabled ? "Disabling" : "Enabling", - pci_name(chip->pci)); + "%s via VGA-switcheroo\n", pci_name(chip->pci), + disabled ? "Disabling" : "Enabling"); if (disabled) { azx_suspend(&pci->dev); chip->disabled = true; if (snd_hda_lock_devices(chip->bus)) - snd_printk(KERN_WARNING SFX - "Cannot lock devices!\n"); + snd_printk(KERN_WARNING SFX "Cannot lock devices!\n", + pci_name(chip->pci)); } else { snd_hda_unlock_devices(chip->bus); chip->disabled = false; @@ -2685,7 +2686,7 @@ static void __devinit init_vga_switcheroo(struct azx *chip) struct pci_dev *p = get_bound_vga(chip->pci); if (p) { snd_printk(KERN_INFO SFX - "%s: Handle VGA-switcheroo audio client\n", + "Handle VGA-switcheroo audio client\n", pci_name(chip->pci)); chip->use_vga_switcheroo = 1; pci_dev_put(p); @@ -2871,11 +2872,11 @@ static int __devinit check_position_fix(struct azx *chip, int fix) /* Check VIA/ATI HD Audio Controller exist */ if (chip->driver_caps & AZX_DCAPS_POSFIX_VIA) { - snd_printd(SFX "Using VIACOMBO position fix\n"); + snd_printd(SFX "Using VIACOMBO position fix\n", pci_name(chip->pci)); return POS_FIX_VIACOMBO; } if (chip->driver_caps & AZX_DCAPS_POSFIX_LPIB) { - snd_printd(SFX "Using LPIB position fix\n"); + snd_printd(SFX "Using LPIB position fix\n", pci_name(chip->pci)); return POS_FIX_LPIB; } return POS_FIX_AUTO; @@ -2992,7 +2993,7 @@ static void __devinit azx_check_snoop_available(struct azx *chip) if (snoop != chip->snoop) { snd_printk(KERN_INFO SFX "Force to %s mode\n", - snoop ? "snoop" : "non-snoop"); + pci_name(chip->pci), snoop ? "snoop" : "non-snoop"); chip->snoop = snoop; } } @@ -3018,7 +3019,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, chip = kzalloc(sizeof(*chip), GFP_KERNEL); if (!chip) { - snd_printk(KERN_ERR SFX "cannot allocate chip\n"); + snd_printk(KERN_ERR SFX "Cannot allocate chip\n", pci_name(pci)); pci_disable_device(pci); return -ENOMEM; } @@ -3064,10 +3065,11 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, } if (check_hdmi_disabled(pci)) { - snd_printk(KERN_INFO SFX "VGA controller for %s is disabled\n", - pci_name(pci)); + snd_printk(KERN_INFO SFX "VGA controller is disabled\n", + pci_name(chip->pci)); if (use_vga_switcheroo(chip)) { - snd_printk(KERN_INFO SFX "Delaying initialization\n"); + snd_printk(KERN_INFO SFX "Delaying initialization\n", + pci_name(chip->pci)); chip->disabled = true; goto ok; } @@ -3085,7 +3087,8 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, ok: err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); if (err < 0) { - snd_printk(KERN_ERR SFX "Error creating device [card]!\n"); + snd_printk(KERN_ERR SFX "Error creating device [card]!\n", + pci_name(chip->pci)); azx_free(chip); return err; } @@ -3120,7 +3123,7 @@ static int DELAYED_INIT_MARK azx_first_init(struct azx *chip) chip->addr = pci_resource_start(pci, 0); chip->remap_addr = pci_ioremap_bar(pci, 0); if (chip->remap_addr == NULL) { - snd_printk(KERN_ERR SFX "ioremap error\n"); + snd_printk(KERN_ERR SFX "ioremap error\n", pci_name(chip->pci)); return -ENXIO; } @@ -3135,7 +3138,7 @@ static int DELAYED_INIT_MARK azx_first_init(struct azx *chip) synchronize_irq(chip->irq); gcap = azx_readw(chip, GCAP); - snd_printdd(SFX "chipset global capabilities = 0x%x\n", gcap); + snd_printdd(SFX "chipset global capabilities = 0x%x\n", pci_name(chip->pci), gcap); /* disable SB600 64bit support for safety */ if (chip->pci->vendor == PCI_VENDOR_ID_ATI) { @@ -3152,7 +3155,7 @@ static int DELAYED_INIT_MARK azx_first_init(struct azx *chip) /* disable 64bit DMA address on some devices */ if (chip->driver_caps & AZX_DCAPS_NO_64BIT) { - snd_printd(SFX "Disabling 64bit DMA\n"); + snd_printd(SFX "Disabling 64bit DMA\n", pci_name(chip->pci)); gcap &= ~ICH6_GCAP_64OK; } @@ -3207,7 +3210,7 @@ static int DELAYED_INIT_MARK azx_first_init(struct azx *chip) chip->azx_dev = kcalloc(chip->num_streams, sizeof(*chip->azx_dev), GFP_KERNEL); if (!chip->azx_dev) { - snd_printk(KERN_ERR SFX "cannot malloc azx_dev\n"); + snd_printk(KERN_ERR SFX "cannot malloc azx_dev\n", pci_name(chip->pci)); return -ENOMEM; } @@ -3217,7 +3220,7 @@ static int DELAYED_INIT_MARK azx_first_init(struct azx *chip) snd_dma_pci_data(chip->pci), BDL_SIZE, &chip->azx_dev[i].bdl); if (err < 0) { - snd_printk(KERN_ERR SFX "cannot allocate BDL\n"); + snd_printk(KERN_ERR SFX "cannot allocate BDL\n", pci_name(chip->pci)); return -ENOMEM; } mark_pages_wc(chip, &chip->azx_dev[i].bdl, true); @@ -3227,7 +3230,7 @@ static int DELAYED_INIT_MARK azx_first_init(struct azx *chip) snd_dma_pci_data(chip->pci), chip->num_streams * 8, &chip->posbuf); if (err < 0) { - snd_printk(KERN_ERR SFX "cannot allocate posbuf\n"); + snd_printk(KERN_ERR SFX "cannot allocate posbuf\n", pci_name(chip->pci)); return -ENOMEM; } mark_pages_wc(chip, &chip->posbuf, true); @@ -3245,7 +3248,7 @@ static int DELAYED_INIT_MARK azx_first_init(struct azx *chip) /* codec detection */ if (!chip->codec_mask) { - snd_printk(KERN_ERR SFX "no codecs found!\n"); + snd_printk(KERN_ERR SFX "no codecs found!\n", pci_name(chip->pci)); return -ENODEV; } @@ -3281,7 +3284,8 @@ static void azx_firmware_cb(const struct firmware *fw, void *context) struct pci_dev *pci = chip->pci; if (!fw) { - snd_printk(KERN_ERR SFX "Cannot load firmware, aborting\n"); + snd_printk(KERN_ERR SFX "Cannot load firmware, aborting\n", + pci_name(chip->pci)); goto error; } @@ -3317,7 +3321,7 @@ static int __devinit azx_probe(struct pci_dev *pci, err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); if (err < 0) { - snd_printk(KERN_ERR SFX "Error creating card!\n"); + snd_printk(KERN_ERR "hda-intel: Error creating card!\n"); return err; } @@ -3332,7 +3336,7 @@ static int __devinit azx_probe(struct pci_dev *pci, #ifdef CONFIG_SND_HDA_PATCH_LOADER if (patch[dev] && *patch[dev]) { snd_printk(KERN_ERR SFX "Applying patch firmware '%s'\n", - patch[dev]); + pci_name(pci), patch[dev]); err = request_firmware_nowait(THIS_MODULE, true, patch[dev], &pci->dev, GFP_KERNEL, card, azx_firmware_cb); @@ -3355,8 +3359,8 @@ static int __devinit azx_probe(struct pci_dev *pci, err = register_vga_switcheroo(chip); if (err < 0) { - snd_printk(KERN_ERR SFX - "Error registering VGA-switcheroo client\n"); + snd_printk(KERN_ERR SFX "Error registering VGA-switcheroo client\n", + pci_name(chip->pci)); goto out_free; } -- 1.7.10.4

2 1

[alsa-devel] Support request for codec WM8731 porting
by Cyril GERMAINE 04 Dec '12

04 Dec '12

Hello, I work on a board based from DM365EVM baseboard, but with the WM8731 codec instead of the TLV320AIC3x. I tried with several linux kernels (one is the 3.6.6 release). Now the playback works but not the capture, and I'm not able to debug this issue. I have many "0" frames in the captured data, although data and rate on I2S bus is OK. 22 null frames, then 2 non null frames. I tried to capture a sinusoidal wave. Version of kernel : 3.6.6-dirty Version of alsa lib : 1.0.25 Here are the attached files: - dmesg.txt : dmesg log after kernel init - .config - all the files of the kernel I modified/added for my board - capture_alsalib_traces.txt : some traces I put in the alsa lib during capture - dmesg_capture.txt : some traces I put in the kernel during capture - capture_test.wav.hexdump : the result of the capture (arecord -f S16_LE) I modified the wm8731.c file because I wasn't able to activate the required functions of the codec with alsamixer, and I don't undestand how dapm works... Maybe I missed something in the board or platform dependant files? About the DMA? I don't undestand why the playback works but not the capture. I also worked with the kernel provided with the DM365EVM board (from Montavista, linux_2.6.18_pro500, DaVinciLSP_02_10_00_08), the results are better, I "only" have one null frame for 2 frames, even with a stereo input signal (on the left and right input line). Thanks for your help.

2 2

[alsa-devel] sound/usb/midi.c:1088 substream_open() warn: inconsistent returns mutex:&umidi->mutex: locked (1058) unlocked (1048, 1088)
by Dan Carpenter 04 Dec '12

04 Dec '12

Hi Takashi, FYI, there are new smatch warnings show up in tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git master head: d34bb8cb0b61e91a036aa516dc6f9ced5af2adc4 commit: 6e28507b716f16618070517dbfcaafe2ca7e9cbe ALSA: usb-audio: Fix missing autopm for MIDI input date: 6 hours ago + sound/usb/midi.c:1088 substream_open() warn: inconsistent returns mutex:&umidi->mutex: locked (1058) unlocked (1048,1088) sound/usb/midi.c:2166 snd_usbmidi_create() error: we previously assumed 'quirk' could be null (see line 2155) git remote update sound-unstable git checkout 6e28507b716f16618070517dbfcaafe2ca7e9cbe vim +1088 sound/usb/midi.c 96f61d9a sound/usb/usbmidi.c Clemens Ladisch 2009-10-22 1042 struct snd_kcontrol *ctl; 6e28507b sound/usb/midi.c Takashi Iwai 2012-12-03 1043 int err; 96f61d9a sound/usb/usbmidi.c Clemens Ladisch 2009-10-22 1044 cea1c011 sound/usb/midi.c Takashi Iwai 2012-12-03 1045 down_read(&umidi->disc_rwsem); cea1c011 sound/usb/midi.c Takashi Iwai 2012-12-03 1046 if (umidi->disconnected) { cea1c011 sound/usb/midi.c Takashi Iwai 2012-12-03 1047 up_read(&umidi->disc_rwsem); 6e28507b sound/usb/midi.c Takashi Iwai 2012-12-03 @1048 return open ? -ENODEV : 0; cea1c011 sound/usb/midi.c Takashi Iwai 2012-12-03 1049 } cea1c011 sound/usb/midi.c Takashi Iwai 2012-12-03 1050 96f61d9a sound/usb/usbmidi.c Clemens Ladisch 2009-10-22 1051 mutex_lock(&umidi->mutex); 96f61d9a sound/usb/usbmidi.c Clemens Ladisch 2009-10-22 1052 if (open) { 6e28507b sound/usb/midi.c Takashi Iwai 2012-12-03 1053 if (!umidi->opened[0] && !umidi->opened[1]) { 6e28507b sound/usb/midi.c Takashi Iwai 2012-12-03 1054 err = usb_autopm_get_interface(umidi->iface); 6e28507b sound/usb/midi.c Takashi Iwai 2012-12-03 1055 umidi->autopm_reference = err >= 0; 6e28507b sound/usb/midi.c Takashi Iwai 2012-12-03 1056 if (err < 0 && err != -EACCES) { 6e28507b sound/usb/midi.c Takashi Iwai 2012-12-03 1057 up_read(&umidi->disc_rwsem); 6e28507b sound/usb/midi.c Takashi Iwai 2012-12-03 @1058 return -EIO; 6e28507b sound/usb/midi.c Takashi Iwai 2012-12-03 1059 } 6e28507b sound/usb/midi.c Takashi Iwai 2012-12-03 1060 if (umidi->roland_load_ctl) { 6e28507b sound/usb/midi.c Takashi Iwai 2012-12-03 1061 ctl = umidi->roland_load_ctl; 6e28507b sound/usb/midi.c Takashi Iwai 2012-12-03 1062 ctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE; 6e28507b sound/usb/midi.c Takashi Iwai 2012-12-03 1063 snd_ctl_notify(umidi->card, 96f61d9a sound/usb/usbmidi.c Clemens Ladisch 2009-10-22 1064 SNDRV_CTL_EVENT_MASK_INFO, &ctl->id); 6e28507b sound/usb/midi.c Takashi Iwai 2012-12-03 1065 update_roland_altsetting(umidi); 6e28507b sound/usb/midi.c Takashi Iwai 2012-12-03 1066 } 96f61d9a sound/usb/usbmidi.c Clemens Ladisch 2009-10-22 1067 } 6e28507b sound/usb/midi.c Takashi Iwai 2012-12-03 1068 umidi->opened[dir]++; 6e28507b sound/usb/midi.c Takashi Iwai 2012-12-03 1069 if (umidi->opened[1]) 6e28507b sound/usb/midi.c Takashi Iwai 2012-12-03 1070 snd_usbmidi_input_start(&umidi->list); 96f61d9a sound/usb/usbmidi.c Clemens Ladisch 2009-10-22 1071 } else { 6e28507b sound/usb/midi.c Takashi Iwai 2012-12-03 1072 umidi->opened[dir]--; 6e28507b sound/usb/midi.c Takashi Iwai 2012-12-03 1073 if (!umidi->opened[1]) 6e28507b sound/usb/midi.c Takashi Iwai 2012-12-03 1074 snd_usbmidi_input_stop(&umidi->list); 6e28507b sound/usb/midi.c Takashi Iwai 2012-12-03 1075 if (!umidi->opened[0] && !umidi->opened[1]) { 6e28507b sound/usb/midi.c Takashi Iwai 2012-12-03 1076 if (umidi->roland_load_ctl) { 6e28507b sound/usb/midi.c Takashi Iwai 2012-12-03 1077 ctl = umidi->roland_load_ctl; 6e28507b sound/usb/midi.c Takashi Iwai 2012-12-03 1078 ctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE; 6e28507b sound/usb/midi.c Takashi Iwai 2012-12-03 1079 snd_ctl_notify(umidi->card, 96f61d9a sound/usb/usbmidi.c Clemens Ladisch 2009-10-22 1080 SNDRV_CTL_EVENT_MASK_INFO, &ctl->id); 6e28507b sound/usb/midi.c Takashi Iwai 2012-12-03 1081 } 6e28507b sound/usb/midi.c Takashi Iwai 2012-12-03 1082 if (umidi->autopm_reference) 6e28507b sound/usb/midi.c Takashi Iwai 2012-12-03 1083 usb_autopm_put_interface(umidi->iface); 96f61d9a sound/usb/usbmidi.c Clemens Ladisch 2009-10-22 1084 } 96f61d9a sound/usb/usbmidi.c Clemens Ladisch 2009-10-22 1085 } 96f61d9a sound/usb/usbmidi.c Clemens Ladisch 2009-10-22 1086 mutex_unlock(&umidi->mutex); cea1c011 sound/usb/midi.c Takashi Iwai 2012-12-03 1087 up_read(&umidi->disc_rwsem); 6e28507b sound/usb/midi.c Takashi Iwai 2012-12-03 @1088 return 0; 96f61d9a sound/usb/usbmidi.c Clemens Ladisch 2009-10-22 1089 } 96f61d9a sound/usb/usbmidi.c Clemens Ladisch 2009-10-22 1090 86e07d34 sound/usb/usbmidi.c Takashi Iwai 2005-11-17 1091 static int snd_usbmidi_output_open(struct snd_rawmidi_substream *substream) --- 0-DAY kernel build testing backend Open Source Technology Center Fengguang Wu, Yuanhan Liu Intel Corporation

2 1

[alsa-devel] HIFACE family driver
by Michael Trimarchi 04 Dec '12

04 Dec '12

Hi all, I have written in my spare time an alsa driver for HIFACE devices family. Now I'm not an alsa expert and I would like to know if you think that it's useful to include it in mainline https://github.com/panicking/snd-usb-asyncaudio The list of usb devices supported https://github.com/panicking/snd-usb-asyncaudio/blob/master/chip.c The main difference is that I need to use bulk message to talk to the usb dongle and ofcourse there are a lot of improvement that we can do on it. I have already tested on several kernel version and it's already included in some linux distribution. Wait for some feedback ;) Michael

3 4

[alsa-devel] [PATCH 0/2] USB MIDI fixes
by Takashi Iwai 04 Dec '12

04 Dec '12

Hi, this is a series of patches to fix pending issues wrt USB MIDI in the recent kernels. The first patch is to close the disconnection race for autopm and other codes in sound/usb/midi.c. The second patch is a fix for missing autopm handling for USB MIDI input, which has been discussed shortly ago. It's based on the latest 3.7-rc, but should be applicable to the latest stable-queue for 3.6 and older, too. The previous autopm fix in stable tree isn't released yet, so you can't apply it to the released 3.6.y. You'll need pick up the patch from stable-queue tree. Takashi

2 4

[alsa-devel] [PATCH 0/3] Cleaning up the hdspm driver
by Adrian Knoth 03 Dec '12

03 Dec '12

Hi! Here's a series of patches to clean up more or less redundant code in the hdspm driver. To ease the review process, I've made three separate commits. Feel free to merge the patches if you think individual commits are not necessary. Cheers Adrian Knoth (3): ALSA: hdspm - Implement generic function to toggle settings ALSA: hdspm - Use HDSPM_TOGGLE_SETTING to alter settings ALSA: hdspm - Remove obsolete settings functions sound/pci/rme9652/hdspm.c | 398 ++++----------------------------------------- 1 file changed, 31 insertions(+), 367 deletions(-) -- 1.7.10.4

2 4

[alsa-devel] ASoC updates for 3.8
by Mark Brown 03 Dec '12

03 Dec '12

The following changes since commit 9489e9dcae718d5fde988e4a684a0f55b5f94d17: Linux 3.7-rc7 (2012-11-25 17:59:19 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-3.8 for you to fetch changes up to 854ea639bb6b0b7ec433e0a59405f4f199ffae4f: Merge remote-tracking branch 'asoc/topic/wm9090' into asoc-next (2012-12-02 13:35:31 +0900) ---------------------------------------------------------------- ASoC: Updates for v3.8 Very quiet release for ASoC really: - Standardisation of the logging. - DT and dmaengine support for Atmel. - Support for Wolfson ADSP cores. - New drivers for Freescale/iVeia P1022 and Maxim MAX98090. ---------------------------------------------------------------- Andrey Smirnov (1): Add a codec driver for SI476X MFD Ashish Chavan (2): ASoC: da9055: Add support for DSP mode of AIF ASoC: da9055: DAPMise Headphone and Lineout enable controls Bill Pemberton (2): ASoC: remove use of __devinitconst ASoC: max98088: remove use of __devexit_p Bo Shen (11): ASoC: atmel-ssc: use devm_xxx() managed function ASoC: atmel-ssc: use module_platform_driver macro ASoC: sam9g20-wm8731: convert to use snd_soc_register_card() ASoC: sam9g20: using platform device for audio part ARM: at91: atmel-ssc: add platform device id table ARM: at91: atmel-ssc: add device tree support ASoC: atmel-ssc-dai: register dai and pcm directly ASoC: atmel-ssc-dai: match new method of dai and pcm register ASoC: at91sam9g20ek-wm8731: convert to dt support ASoC: atmel-pcm: split into two file ASoC: atmel-pcm: dma support based on pcm dmaengine Charles Keepax (4): ASoC: wm5102: Correct micbias DAPM supplies ASoC: arizona: Add support for multiplexer with no associated mixer ASoC: wm5102: Remove mixer widgets on the ASRC ASoC: wm5110: Remove mixer widgets on the ASRC Daniel Mack (10): ASoC: ak4104: handle more sample rates ASoC: ak4104: add DT bindings ASoC: cs4271: add support for AMUTEB=BMUTEC feature ASoC: enable 192KHz support for McASP ASoC: McASP: add support for clock dividers ASoC: McASP: make AHCLK direction configurable ASoC: McASP: set format parameters in dependence of the DAI format ASoC: McASP: add support for 24 bit samples ASoC: McASP: treat SNDRV_PCM_FORMAT_U24_* as 32 bit words ASoC: ak4104: add reset line property to DT bindings Dimitris Papastamos (3): ASoC: bells: Up to 512fs ASoC: wm5102: Register DSP1 Aux widgets ASoC: wm0010: Up the completion timeout to 20ms instead of 10ms Eric Millbrandt (2): ASoC: fsl/pcm030 add unique dai_link names ASoC: wm8978: Remove remuxing of GPIO1 pin Javier Martin (2): ASoC: tlv320aic32x4: Fix problem with first capture. ASoC: tlv320aic32x4: Add rstn gpio to platform data. Kees Cook (1): ASoC: fsl: remove CONFIG_EXPERIMENTAL Kuninori Morimoto (13): ASoC: fsi: fixup channels_min/max ASoC: fsi: use devm_request_irq() ASoC: fsi: fsi_set_master_clk() was called from fsi_hw_xxx() only ASoC: fsi: care fsi_hw_start/stop() return value ASoC: fsi: add master clock control functions ASoC: fsi: tidyup FSIA/B settings ASoC: fsi: tidyup sh_fsi_platform_info pointer ASoC: fsi: SPDIF format become independent from platform flags ASoC: fsi: master clock selection become independent from platform flags ASoC: fsi: stream mode become independent from platform flags ASoC: fsi: add SND_SOC_DAIFMT_INV_xxx support ASoC: ak4642: cleanup struct ak4642_priv ASoC: Add max98090 codec driver Lars-Peter Clausen (3): ASoC: jz4740-codec: Add dB information ASoC: jz4740-codec: Use devm_request_and_ioremap ASoC: jz4740-codec: Use regmap Lee Jones (1): ASoC: Ux500: Initialise PCM from MSP probe rather than as a device Liam Girdwood (5): ASoC: dapm: Standardise ASoC DAPM messages ASoC: cache: Standardise ASoC cache messages ASoC: Jack: Standardise ASoC Jack messages ASoC: pcm: Standardise ASoC PCM messages ASoC: core: Standardise ASoC messages Lothar Waßmann (6): ASoC: imx: add MODULE_LICENSE("GPL") ASoC: imx-sgtl5000: prevent module from being unloaded when in use ASoC: dmaengine_pcm: set MODULE_LICENSE ASoC: fsl: fix miscompilation of snd-soc-imx-pcm ASoC: mxs-saif: export mxs_saif_put_mclk,mxs_saif_get_mclk ASoC: fsl: fix miscompilation of snd-soc-imx-pcm Mark Brown (117): ASoC: wm2200: Convert to devm APIs ASoC: wm2200: Implement AEC loopback support ASoC: wm2200: Map DSPs into regmap ASoC: wm2200: Fully plumb the DSPs into the routing map ASoC: wm2200: Initial DSP support ASoC: wm2200: Provide initial coefficient loading regmap: Rename n_ranges to num_ranges regmap: When we sanity check during range adds say what errors we find regmap: Allow ranges to be named regmap: Factor out debugfs register read regmap: Provide debugfs read of register ranges regmap: Factor range lookup out of page selection regmap: Make return code checks consistent regmap: Split raw writes that cross window boundaries ASoC: bells: Add WM0010 support ASoC: bells: Correct typo in sub speaker DAI name for WM5110 ASoC: bells: Provide additional parameterisation ASoC: bells: Add WM2200 support ASoC: wm8750: Convert to direct regmap API usage ASoC: wm8770: Convert to direct regmap API usage ASoC: wm8770: Remove unneeded bias level manipulation ASoC: wm8770: Remove regulator allocation to SPI probe ASoC: wm8770: Conver to table based DAPM and control init ASoC: wm8971: Convert to direct regmap API usage ASoC: wm8994: Allow a delay between jack insertion and microphone detect ASoC: wm8994: Tune VMID power up sequence ASoC: wm8994: Tune VMID power down sequence ASoC: wm8958: Load firmwares simultaneously Merge tag 'regmap/range' of git://git.kernel.org/.../broonie/regmap into asoc-wm2200 ASoC: wm2200: Add names for ranges ASoC: wm8994: Lower AIFnCLK divisor when dropping to 32kHz ASoC: adsp: Add ADSP base support Merge branch 'topic/adsp' of git://git.kernel.org/.../broonie/sound into asoc-arizona ASoC: arizona: Define standard hookup for ADSP2 ASoC: wm5102: Hook up DSP1 Merge branch 'topic/adsp' of git://git.kernel.org/.../broonie/sound into asoc-wm2200 ASoC: wm2200: Convert over to wm_adsp for ADSP1 support Merge tag 'v3.7-rc3' into HEAD ASoC: ab8500: Staticise non-exported ab85000_codec_dai Merge tag 'v3.7-rc6' into asoc-arizona Merge tag 'v3.7-rc6' into asoc-wm2200 ASoC: wm5102: Remove output OSR and PGA volume control ASoC: wm8994: Lower AIFnCLK divisor when dropping to 32kHz mfd: wm8994: Store platform data in device ASoC: wm8994: Use pdata cached in MFD driver ASoC: arizona: Support higher clock rates ASoC: wm2000: Get the CODEC pointer from the widget ASoC: wm5102: Add missing routes for ASRC inputs ASoC: arizona: Suppress noop FLL updates ASoC: wm0010: Ignore interrupts in power down mode ASoC: pcm: Report error code when we fail to init platform Merge tag 'v3.7-rc7' into asoc-ux500 Merge tag 'v3.7-rc7' into asoc-wm2200 ASoC: arizona: Record FLL setting when disabling mfd: arizona: Allow the CODEC DAPM context to be accessed elsewhere Input - arizona-haptics: Add driver haptics module on Arizona CODECs ASoC: wm5102: Implement haptics support ASoC: wm5110: Implement haptics support mfd: arizona: Register haptics devices ASoC: wm8994: Use the same DCS codes for all WM1811 variants ASoC: wm_adsp: Enable DVFS for ADSP2 Merge branch 'topic/adsp' of git://git.kernel.org/.../broonie/sound into asoc-arizona ASoC: wm5102: Enable DVFS support for the DSP ASoC: wm_adsp: Set the core enable as well as start bit for ADSP2 ASoC: arizona: Add parentheses ASoC: wm5102: Correct base address for Y region Merge remote-tracking branch 'asoc/topic/ab8500' into asoc-next Merge remote-tracking branch 'asoc/topic/adsp' into asoc-next Merge remote-tracking branch 'asoc/topic/ak4104' into asoc-next Merge remote-tracking branch 'asoc/topic/ak4535' into asoc-next Merge remote-tracking branch 'asoc/topic/ak4642' into asoc-next Merge remote-tracking branch 'asoc/topic/arizona' into asoc-next Merge remote-tracking branch 'asoc/topic/atmel' into asoc-next Merge remote-tracking branch 'asoc/topic/cs4271' into asoc-next Merge remote-tracking branch 'asoc/topic/da7210' into asoc-next Merge remote-tracking branch 'asoc/topic/da9055' into asoc-next Merge remote-tracking branch 'asoc/topic/davinci' into asoc-next Merge remote-tracking branch 'asoc/topic/dmaengine' into asoc-next Merge remote-tracking branch 'asoc/topic/fsi' into asoc-next Merge remote-tracking branch 'asoc/topic/fsl' into asoc-next Merge remote-tracking branch 'asoc/topic/hotplug' into asoc-next Merge remote-tracking branch 'asoc/topic/jack' into asoc-next Merge remote-tracking branch 'asoc/topic/jz4740' into asoc-next Merge remote-tracking branch 'asoc/topic/kirkwood' into asoc-next Merge remote-tracking branch 'asoc/topic/lm49453' into asoc-next Merge remote-tracking branch 'asoc/topic/log' into asoc-next Merge remote-tracking branch 'asoc/topic/max9768' into asoc-next Merge remote-tracking branch 'asoc/topic/max98090' into asoc-next Merge remote-tracking branch 'asoc/topic/omap' into asoc-next Merge remote-tracking branch 'asoc/topic/rt5631' into asoc-next Merge remote-tracking branch 'asoc/topic/samsung' into asoc-next Merge remote-tracking branch 'asoc/topic/si476x' into asoc-next Merge remote-tracking branch 'asoc/topic/tlv320aic32x4' into asoc-next Merge remote-tracking branch 'asoc/topic/ux500' into asoc-next Merge remote-tracking branch 'asoc/topic/wm0010' into asoc-next Merge remote-tracking branch 'asoc/topic/wm2000' into asoc-next Merge remote-tracking branch 'asoc/topic/wm2200' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8350' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8400' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8510' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8741' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8750' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8753' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8770' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8804' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8955' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8960' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8962' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8971' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8978' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8985' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8988' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8993' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8994' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8995' into asoc-next Merge remote-tracking branch 'asoc/topic/wm9081' into asoc-next Merge remote-tracking branch 'asoc/topic/wm9090' into asoc-next Matt Porter (2): ASoC: davinci: replace private sram api with genalloc ARM: davinci: enable SRAM ping ping buffering on DA850 Nicolas Ferre (1): ASoC: atmel-ssc: add phybase in device structure Padmavathi Venna (1): ASoC: Samsung: Get I2S src_clk from clock alias id. Peter Ujfalusi (1): ASoC: omap-abe-twl6040: Use "ti,jack-detection" DT property as boolean Randy Dunlap (1): ASoC: fix wm2200.c printk format warnings Russell King (5): ASoC: kirkwood-i2s: use devm_* APIs ASoC: kirkwood-i2s: better handling of play/record control registers ASoC: kirkwood-dma: remove restriction on sample rates ASoC: kirkwood-i2s: add support for external clock rates ASoC: kirkwood-dma: remove channel restrictions Sachin Kamat (16): ASoC: rt5631: Fix return value ASoC: wm8350: Use devm_regulator_bulk_get ASoC: wm8962: Use devm_regulator_bulk_get ASoC: wm8400: Use devm_regulator_bulk_get ASoC: wm8993: Use devm_regulator_bulk_get ASoC: ak4535: Use devm_regmap_init_i2c() ASoC: da7210: Use devm_regmap_init_i2c() ASoC: lm49453: Use devm_regmap_init_i2c() ASoC: wm2200: Remove empty labels ASoC: wm9090: Use devm_regmap_init_i2c() ASoC: wm9081: Use devm_regmap_init_i2c() ASoC: wm8993: Use devm_regmap_init_i2c() ASoC: wm8962: Use devm_regmap_init_i2c() ASoC: max9768: Use devm_regmap_init_i2c() ASoC: wm8955: Use devm_regmap_init_i2c() ASoC: wm8960: Use devm_regmap_init_i2c() Scott Ling (4): ASoC: wm0010: Remove boot_done variable as no longer required. ASoC: wm0010: Split out the firmware file parsing from the boot ASoC: wm0010: Split out the stage2 load from the boot function ASoC: wm0010: Add checking for .dfw info record version. Thomas Abraham (4): ASoC: SAMSUNG: ac97: use clk_prepare_enable and clk_disable_unprepare ASoC: SAMSUNG: i2s: use clk_prepare_enable and clk_disable_unprepare ASoC: SAMSUNG: pcm: use clk_prepare_enable and clk_disable_unprepare ASoC: SAMSUNG: spdif: use clk_prepare_enable and clk_disable_unprepare Timur Tabi (1): ASoC: add support for the Freescale / iVeia P1022 RDK reference board Tony Lindgren (1): ASoC: OMAP: mcbsp fixes for enabling ARM multiplatform support Tushar Behera (10): ASoC: wm8510: remove invalid free of devm_ allocated data ASoC: wm8753: remove invalid free of devm_ allocated data ASoC: wm8995: Convert to use devm_kzalloc() ASoC: wm8741: Convert to use devm_regmap_init_{spi,i2c}() ASoC: wm8753: Convert to use devm_regmap_init_{spi,i2c}() ASoC: wm8995: Convert to use devm_regmap_init_{spi,i2c}() ASoC: wm8804: Convert to use devm_regmap_init_{spi,i2c}() ASoC: wm8985: Convert to use devm_regmap_init_{spi,i2c}() ASoC: wm8988: Convert to use devm_regmap_init_{spi,i2c}() ASoC: ak4104: Convert to use devm_regmap_init_spi() Ulf Hansson (2): ASoC: Ux500: Fixup use of clocks ASoC: Ux500: Control apb clock Wei Yongjun (1): ASoC: jack: remove unused variable in snd_soc_jack_report() .../devicetree/bindings/misc/atmel-ssc.txt | 15 + Documentation/devicetree/bindings/sound/ak4104.txt | 22 + .../sound/atmel-at91sam9g20ek-wm8731-audio.txt | 26 + Documentation/devicetree/bindings/sound/cs4271.txt | 2 + .../devicetree/bindings/sound/omap-abe-twl6040.txt | 4 +- arch/arm/boot/dts/at91sam9260.dtsi | 8 + arch/arm/boot/dts/at91sam9263.dtsi | 16 + arch/arm/boot/dts/at91sam9g20ek_common.dtsi | 32 +- arch/arm/boot/dts/at91sam9g45.dtsi | 16 + arch/arm/boot/dts/at91sam9x5.dtsi | 8 + arch/arm/mach-at91/at91rm9200.c | 9 +- arch/arm/mach-at91/at91rm9200_devices.c | 6 +- arch/arm/mach-at91/at91sam9260.c | 3 +- arch/arm/mach-at91/at91sam9260_devices.c | 2 +- arch/arm/mach-at91/at91sam9261.c | 9 +- arch/arm/mach-at91/at91sam9261_devices.c | 6 +- arch/arm/mach-at91/at91sam9263.c | 6 +- arch/arm/mach-at91/at91sam9263_devices.c | 4 +- arch/arm/mach-at91/at91sam9g45.c | 6 +- arch/arm/mach-at91/at91sam9g45_devices.c | 4 +- arch/arm/mach-at91/at91sam9rl.c | 6 +- arch/arm/mach-at91/at91sam9rl_devices.c | 4 +- arch/arm/mach-at91/at91sam9x5.c | 1 + arch/arm/mach-at91/board-sam9g20ek.c | 11 + arch/arm/mach-davinci/board-da850-evm.c | 24 +- drivers/base/regmap/internal.h | 2 + drivers/base/regmap/regmap-debugfs.c | 50 +- drivers/base/regmap/regmap.c | 154 +++-- drivers/input/misc/Kconfig | 10 + drivers/input/misc/Makefile | 1 + drivers/input/misc/arizona-haptics.c | 255 ++++++++ drivers/mfd/arizona-core.c | 2 + drivers/mfd/wm8994-core.c | 35 +- drivers/misc/atmel-ssc.c | 135 ++-- include/linux/atmel-ssc.h | 6 + include/linux/mfd/arizona/core.h | 4 + include/linux/mfd/arizona/pdata.h | 6 + include/linux/mfd/wm8994/core.h | 4 + include/linux/mfd/wm8994/pdata.h | 5 + include/linux/platform_data/asoc-s3c.h | 6 - include/linux/platform_data/davinci_asp.h | 3 + include/linux/regmap.h | 6 +- include/sound/cs4271.h | 1 + include/sound/sh_fsi.h | 6 + include/sound/tlv320aic32x4.h | 1 + sound/soc/atmel/Kconfig | 13 +- sound/soc/atmel/Makefile | 4 + sound/soc/atmel/atmel-pcm-dma.c | 240 +++++++ sound/soc/atmel/atmel-pcm-pdc.c | 401 ++++++++++++ sound/soc/atmel/atmel-pcm.c | 401 +----------- sound/soc/atmel/atmel-pcm.h | 34 + sound/soc/atmel/atmel_ssc_dai.c | 168 ++--- sound/soc/atmel/atmel_ssc_dai.h | 3 +- sound/soc/atmel/sam9g20_wm8731.c | 116 +++- sound/soc/codecs/Kconfig | 16 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/ab8500-codec.c | 2 +- sound/soc/codecs/ak4104.c | 65 +- sound/soc/codecs/ak4535.c | 7 +- sound/soc/codecs/ak4642.c | 23 +- sound/soc/codecs/arizona.c | 21 + sound/soc/codecs/arizona.h | 65 +- sound/soc/codecs/cs4271.c | 21 +- sound/soc/codecs/da7210.c | 13 +- sound/soc/codecs/da9055.c | 43 +- sound/soc/codecs/jz4740.c | 142 ++--- sound/soc/codecs/lm49453.c | 10 +- sound/soc/codecs/max9768.c | 7 +- sound/soc/codecs/max98088.c | 14 +- sound/soc/codecs/max98090.c | 577 +++++++++++++++++ sound/soc/codecs/rt5631.c | 2 +- sound/soc/codecs/si476x.c | 255 ++++++++ sound/soc/codecs/tlv320aic32x4.c | 24 + sound/soc/codecs/tlv320aic32x4.h | 3 + sound/soc/codecs/wm0010.c | 419 +++++++----- sound/soc/codecs/wm2000.c | 2 +- sound/soc/codecs/wm2200.c | 269 ++++++-- sound/soc/codecs/wm5100.c | 2 +- sound/soc/codecs/wm5102.c | 120 ++-- sound/soc/codecs/wm5110.c | 49 +- sound/soc/codecs/wm8350.c | 4 +- sound/soc/codecs/wm8400.c | 14 +- sound/soc/codecs/wm8510.c | 3 - sound/soc/codecs/wm8741.c | 4 +- sound/soc/codecs/wm8750.c | 86 ++- sound/soc/codecs/wm8753.c | 33 +- sound/soc/codecs/wm8770.c | 217 ++++--- sound/soc/codecs/wm8804.c | 17 +- sound/soc/codecs/wm8955.c | 11 +- sound/soc/codecs/wm8958-dsp2.c | 79 +-- sound/soc/codecs/wm8960.c | 2 +- sound/soc/codecs/wm8962.c | 24 +- sound/soc/codecs/wm8971.c | 80 ++- sound/soc/codecs/wm8978.c | 3 - sound/soc/codecs/wm8985.c | 30 +- sound/soc/codecs/wm8988.c | 14 +- sound/soc/codecs/wm8993.c | 14 +- sound/soc/codecs/wm8994.c | 184 +++--- sound/soc/codecs/wm8994.h | 1 - sound/soc/codecs/wm8995.c | 40 +- sound/soc/codecs/wm9081.c | 22 +- sound/soc/codecs/wm9090.c | 20 +- sound/soc/codecs/wm_adsp.c | 666 ++++++++++++++++++++ sound/soc/codecs/wm_adsp.h | 59 ++ sound/soc/codecs/wmfw.h | 128 ++++ sound/soc/davinci/davinci-evm.c | 5 + sound/soc/davinci/davinci-mcasp.c | 93 ++- sound/soc/davinci/davinci-mcasp.h | 2 +- sound/soc/davinci/davinci-pcm.c | 53 +- sound/soc/davinci/davinci-pcm.h | 2 + sound/soc/fsl/Kconfig | 20 +- sound/soc/fsl/Makefile | 14 +- sound/soc/fsl/imx-pcm.c | 4 + sound/soc/fsl/imx-sgtl5000.c | 1 + sound/soc/fsl/p1022_rdk.c | 392 ++++++++++++ sound/soc/fsl/pcm030-audio-fabric.c | 4 +- sound/soc/kirkwood/kirkwood-dma.c | 16 +- sound/soc/kirkwood/kirkwood-i2s.c | 235 ++++--- sound/soc/kirkwood/kirkwood.h | 11 +- sound/soc/mxs/mxs-saif.c | 2 + sound/soc/omap/mcbsp.c | 4 +- sound/soc/omap/mcbsp.h | 6 + sound/soc/omap/omap-abe-twl6040.c | 4 +- sound/soc/omap/omap-mcbsp.c | 5 +- sound/soc/samsung/ac97.c | 6 +- sound/soc/samsung/bells.c | 222 +++++-- sound/soc/samsung/i2s.c | 20 +- sound/soc/samsung/pcm.c | 12 +- sound/soc/samsung/spdif.c | 12 +- sound/soc/sh/fsi.c | 550 ++++++++++++++-- sound/soc/soc-cache.c | 10 +- sound/soc/soc-core.c | 243 +++---- sound/soc/soc-dapm.c | 134 ++-- sound/soc/soc-dmaengine-pcm.c | 2 + sound/soc/soc-jack.c | 16 +- sound/soc/soc-pcm.c | 195 +++--- sound/soc/tegra/tegra20_das.c | 2 +- sound/soc/tegra/tegra20_i2s.c | 4 +- sound/soc/tegra/tegra20_spdif.c | 2 +- sound/soc/tegra/tegra30_ahub.c | 6 +- sound/soc/tegra/tegra30_i2s.c | 4 +- sound/soc/tegra/tegra_alc5632.c | 2 +- sound/soc/tegra/tegra_wm8753.c | 2 +- sound/soc/tegra/tegra_wm8903.c | 2 +- sound/soc/tegra/trimslice.c | 2 +- sound/soc/ux500/mop500.c | 4 +- sound/soc/ux500/ux500_msp_dai.c | 55 +- sound/soc/ux500/ux500_msp_dai.h | 1 + sound/soc/ux500/ux500_pcm.c | 19 +- sound/soc/ux500/ux500_pcm.h | 3 + 150 files changed, 6428 insertions(+), 2229 deletions(-) create mode 100644 Documentation/devicetree/bindings/misc/atmel-ssc.txt create mode 100644 Documentation/devicetree/bindings/sound/ak4104.txt create mode 100644 Documentation/devicetree/bindings/sound/atmel-at91sam9g20ek-wm8731-audio.txt create mode 100644 drivers/input/misc/arizona-haptics.c create mode 100644 sound/soc/atmel/atmel-pcm-dma.c create mode 100644 sound/soc/atmel/atmel-pcm-pdc.c create mode 100644 sound/soc/codecs/max98090.c create mode 100644 sound/soc/codecs/si476x.c create mode 100644 sound/soc/codecs/wm_adsp.c create mode 100644 sound/soc/codecs/wm_adsp.h create mode 100644 sound/soc/codecs/wmfw.h create mode 100644 sound/soc/fsl/p1022_rdk.c

2 1

[alsa-devel] [PATCH] ASoC: zoom2: Remove HS mux GPIO handling to avoid kernel crash due to BUG_ON()
by Peter Ujfalusi 03 Dec '12

03 Dec '12

The machine driver try to use GPIO15 of twl4030 for HS MUX which supposed to select between TWL's HSOL/R and tlv320aic3254's HPL/R. The TWL's GPIO allocated dynamically so the (OMAP_MAX_GPIO_LINES + 15) is no longer valid GPIO number causing a kernel crash due to BUG_ON() Also the current machine driver supports only TWL audio currently: there is no need to control the GPIO. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi Liam, Mark, I just got one zoom2 board and this patch is needed to get the board booting when audio is enabled. For 3.9 I will convert the machine driver to platform device/driver, but to get 3.8 kernel booting this patch is needed. Regards, Peter sound/soc/omap/zoom2.c | 7 ------- 1 file changed, 7 deletions(-) diff --git a/sound/soc/omap/zoom2.c b/sound/soc/omap/zoom2.c index 1ff6bb9..771bff2 100644 --- a/sound/soc/omap/zoom2.c +++ b/sound/soc/omap/zoom2.c @@ -37,8 +37,6 @@ #include "omap-mcbsp.h" #include "omap-pcm.h" -#define ZOOM2_HEADSET_MUX_GPIO (OMAP_MAX_GPIO_LINES + 15) - static int zoom2_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -187,9 +185,6 @@ static int __init zoom2_soc_init(void) if (ret) goto err1; - BUG_ON(gpio_request(ZOOM2_HEADSET_MUX_GPIO, "hs_mux") < 0); - gpio_direction_output(ZOOM2_HEADSET_MUX_GPIO, 0); - return 0; err1: @@ -202,8 +197,6 @@ module_init(zoom2_soc_init); static void __exit zoom2_soc_exit(void) { - gpio_free(ZOOM2_HEADSET_MUX_GPIO); - platform_device_unregister(zoom2_snd_device); } module_exit(zoom2_soc_exit); -- 1.8.0

2 1

[alsa-devel] [PATCH 1/1] ASoC: da7210: Remove unnecessary regmap_exit call
by Sachin Kamat 03 Dec '12

03 Dec '12

Use of devm_regmap_init_spi does not require an explicit regmap_exit call. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/codecs/da7210.c | 11 +---------- 1 files changed, 1 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index ab1ee5b..843c1eb 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -1337,24 +1337,15 @@ static int __devinit da7210_spi_probe(struct spi_device *spi) if (ret != 0) dev_warn(&spi->dev, "Failed to apply regmap patch: %d\n", ret); - ret = snd_soc_register_codec(&spi->dev, + ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_da7210, &da7210_dai, 1); - if (ret < 0) - goto err_regmap; - - return ret; - -err_regmap: - regmap_exit(da7210->regmap); return ret; } static int __devexit da7210_spi_remove(struct spi_device *spi) { - struct da7210_priv *da7210 = spi_get_drvdata(spi); snd_soc_unregister_codec(&spi->dev); - regmap_exit(da7210->regmap); return 0; } -- 1.7.4.1

2 1

Re: [alsa-devel] [PATCH 2/2] ASoC: Samsung: Do not register samsung audio dma device as pdev
by Mark Brown 02 Dec '12

02 Dec '12

On Sat, Dec 01, 2012 at 03:37:27PM +0530, Padmavathi Venna wrote: > sound/soc/samsung/dma.c | 24 ++++++------------------ > sound/soc/samsung/dma.h | 3 +++ > sound/soc/samsung/i2s.c | 3 +++ > sound/soc/samsung/smdk_wm8994.c | 7 +++++-- > 4 files changed, 17 insertions(+), 20 deletions(-) This is fine in principle but it's going to break every Samsung based platform except the SMDK with WM8994 as only machine driver updated to reflect the changed DAI name is that one, the others need updating too. This includes pretty much all the platforms I actively work on...

1 0

[alsa-devel] [PATCH 1/2] ASoC: atmel-pcm: split into two file
by Bo Shen 02 Dec '12

02 Dec '12

This patch is split original atmel-pcm.c into new atmel-pcm.c and atmel-pcm-pdc.c two files. The new atmel-pcm.c is the share routine while will be used for pdc or dma transfer. Using SND_ATMEL_SOC_PDC to select using PDC for audio transfer Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- sound/soc/atmel/Kconfig | 5 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/atmel-pcm-pdc.c | 401 +++++++++++++++++++++++++++++++++++++++ sound/soc/atmel/atmel-pcm.c | 390 ++----------------------------------- sound/soc/atmel/atmel-pcm.h | 21 +- sound/soc/atmel/atmel_ssc_dai.c | 4 +- 6 files changed, 441 insertions(+), 382 deletions(-) create mode 100644 sound/soc/atmel/atmel-pcm-pdc.c diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index 397ec75..6051993 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -6,6 +6,10 @@ config SND_ATMEL_SOC the ATMEL SSC interface. You will also need to select the audio interfaces to support below. +config SND_ATMEL_SOC_PDC + tristate + depends on SND_ATMEL_SOC + config SND_ATMEL_SOC_SSC tristate depends on SND_ATMEL_SOC @@ -17,6 +21,7 @@ config SND_ATMEL_SOC_SSC config SND_AT91_SOC_SAM9G20_WM8731 tristate "SoC Audio support for WM8731-based At91sam9g20 evaluation board" depends on ATMEL_SSC && SND_ATMEL_SOC && AT91_PROGRAMMABLE_CLOCKS + select SND_ATMEL_SOC_PDC select SND_ATMEL_SOC_SSC select SND_SOC_WM8731 help diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile index a5c0bf1..2e37864 100644 --- a/sound/soc/atmel/Makefile +++ b/sound/soc/atmel/Makefile @@ -1,8 +1,10 @@ # AT91 Platform Support snd-soc-atmel-pcm-objs := atmel-pcm.o +snd-soc-atmel-pcm-pdc-objs := atmel-pcm-pdc.o snd-soc-atmel_ssc_dai-objs := atmel_ssc_dai.o obj-$(CONFIG_SND_ATMEL_SOC) += snd-soc-atmel-pcm.o +obj-$(CONFIG_SND_ATMEL_SOC_PDC) += snd-soc-atmel-pcm-pdc.o obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o # AT91 Machine Support diff --git a/sound/soc/atmel/atmel-pcm-pdc.c b/sound/soc/atmel/atmel-pcm-pdc.c new file mode 100644 index 0000000..1df64ba --- /dev/null +++ b/sound/soc/atmel/atmel-pcm-pdc.c @@ -0,0 +1,401 @@ +/* + * atmel-pcm.c -- ALSA PCM interface for the Atmel atmel SoC. + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Authors: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> + * + * Based on at91-pcm. by: + * Frank Mandarino <fmandarino(a)endrelia.com> + * Copyright 2006 Endrelia Technologies Inc. + * + * Based on pxa2xx-pcm.c by: + * + * Author: Nicolas Pitre + * Created: Nov 30, 2004 + * Copyright: (C) 2004 MontaVista Software, Inc. + * + * 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/init.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/dma-mapping.h> +#include <linux/atmel_pdc.h> +#include <linux/atmel-ssc.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "atmel-pcm.h" + + +/*--------------------------------------------------------------------------*\ + * Hardware definition +\*--------------------------------------------------------------------------*/ +/* TODO: These values were taken from the AT91 platform driver, check + * them against real values for AT32 + */ +static const struct snd_pcm_hardware atmel_pcm_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_PAUSE, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .period_bytes_min = 32, + .period_bytes_max = 8192, + .periods_min = 2, + .periods_max = 1024, + .buffer_bytes_max = ATMEL_SSC_DMABUF_SIZE, +}; + + +/*--------------------------------------------------------------------------*\ + * Data types +\*--------------------------------------------------------------------------*/ +struct atmel_runtime_data { + struct atmel_pcm_dma_params *params; + dma_addr_t dma_buffer; /* physical address of dma buffer */ + dma_addr_t dma_buffer_end; /* first address beyond DMA buffer */ + size_t period_size; + + dma_addr_t period_ptr; /* physical address of next period */ + + /* PDC register save */ + u32 pdc_xpr_save; + u32 pdc_xcr_save; + u32 pdc_xnpr_save; + u32 pdc_xncr_save; +}; + +/*--------------------------------------------------------------------------*\ + * ISR +\*--------------------------------------------------------------------------*/ +static void atmel_pcm_dma_irq(u32 ssc_sr, + struct snd_pcm_substream *substream) +{ + struct atmel_runtime_data *prtd = substream->runtime->private_data; + struct atmel_pcm_dma_params *params = prtd->params; + static int count; + + count++; + + if (ssc_sr & params->mask->ssc_endbuf) { + pr_warn("atmel-pcm: buffer %s on %s (SSC_SR=%#x, count=%d)\n", + substream->stream == SNDRV_PCM_STREAM_PLAYBACK + ? "underrun" : "overrun", + params->name, ssc_sr, count); + + /* re-start the PDC */ + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + prtd->period_ptr += prtd->period_size; + if (prtd->period_ptr >= prtd->dma_buffer_end) + prtd->period_ptr = prtd->dma_buffer; + + ssc_writex(params->ssc->regs, params->pdc->xpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xcr, + prtd->period_size / params->pdc_xfer_size); + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_enable); + } + + if (ssc_sr & params->mask->ssc_endx) { + /* Load the PDC next pointer and counter registers */ + prtd->period_ptr += prtd->period_size; + if (prtd->period_ptr >= prtd->dma_buffer_end) + prtd->period_ptr = prtd->dma_buffer; + + ssc_writex(params->ssc->regs, params->pdc->xnpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xncr, + prtd->period_size / params->pdc_xfer_size); + } + + snd_pcm_period_elapsed(substream); +} + + +/*--------------------------------------------------------------------------*\ + * PCM operations +\*--------------------------------------------------------------------------*/ +static int atmel_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct atmel_runtime_data *prtd = runtime->private_data; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + /* this may get called several times by oss emulation + * with different params */ + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = params_buffer_bytes(params); + + prtd->params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + prtd->params->dma_intr_handler = atmel_pcm_dma_irq; + + prtd->dma_buffer = runtime->dma_addr; + prtd->dma_buffer_end = runtime->dma_addr + runtime->dma_bytes; + prtd->period_size = params_period_bytes(params); + + pr_debug("atmel-pcm: " + "hw_params: DMA for %s initialized " + "(dma_bytes=%u, period_size=%u)\n", + prtd->params->name, + runtime->dma_bytes, + prtd->period_size); + return 0; +} + +static int atmel_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct atmel_runtime_data *prtd = substream->runtime->private_data; + struct atmel_pcm_dma_params *params = prtd->params; + + if (params != NULL) { + ssc_writex(params->ssc->regs, SSC_PDC_PTCR, + params->mask->pdc_disable); + prtd->params->dma_intr_handler = NULL; + } + + return 0; +} + +static int atmel_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct atmel_runtime_data *prtd = substream->runtime->private_data; + struct atmel_pcm_dma_params *params = prtd->params; + + ssc_writex(params->ssc->regs, SSC_IDR, + params->mask->ssc_endx | params->mask->ssc_endbuf); + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + return 0; +} + +static int atmel_pcm_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + struct snd_pcm_runtime *rtd = substream->runtime; + struct atmel_runtime_data *prtd = rtd->private_data; + struct atmel_pcm_dma_params *params = prtd->params; + int ret = 0; + + pr_debug("atmel-pcm:buffer_size = %ld," + "dma_area = %p, dma_bytes = %u\n", + rtd->buffer_size, rtd->dma_area, rtd->dma_bytes); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + prtd->period_ptr = prtd->dma_buffer; + + ssc_writex(params->ssc->regs, params->pdc->xpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xcr, + prtd->period_size / params->pdc_xfer_size); + + prtd->period_ptr += prtd->period_size; + ssc_writex(params->ssc->regs, params->pdc->xnpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xncr, + prtd->period_size / params->pdc_xfer_size); + + pr_debug("atmel-pcm: trigger: " + "period_ptr=%lx, xpr=%u, " + "xcr=%u, xnpr=%u, xncr=%u\n", + (unsigned long)prtd->period_ptr, + ssc_readx(params->ssc->regs, params->pdc->xpr), + ssc_readx(params->ssc->regs, params->pdc->xcr), + ssc_readx(params->ssc->regs, params->pdc->xnpr), + ssc_readx(params->ssc->regs, params->pdc->xncr)); + + ssc_writex(params->ssc->regs, SSC_IER, + params->mask->ssc_endx | params->mask->ssc_endbuf); + ssc_writex(params->ssc->regs, SSC_PDC_PTCR, + params->mask->pdc_enable); + + pr_debug("sr=%u imr=%u\n", + ssc_readx(params->ssc->regs, SSC_SR), + ssc_readx(params->ssc->regs, SSC_IER)); + break; /* SNDRV_PCM_TRIGGER_START */ + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + break; + + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_enable); + break; + + default: + ret = -EINVAL; + } + + return ret; +} + +static snd_pcm_uframes_t atmel_pcm_pointer( + struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct atmel_runtime_data *prtd = runtime->private_data; + struct atmel_pcm_dma_params *params = prtd->params; + dma_addr_t ptr; + snd_pcm_uframes_t x; + + ptr = (dma_addr_t) ssc_readx(params->ssc->regs, params->pdc->xpr); + x = bytes_to_frames(runtime, ptr - prtd->dma_buffer); + + if (x == runtime->buffer_size) + x = 0; + + return x; +} + +static int atmel_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct atmel_runtime_data *prtd; + int ret = 0; + + snd_soc_set_runtime_hwparams(substream, &atmel_pcm_hardware); + + /* ensure that buffer size is a multiple of period size */ + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + goto out; + + prtd = kzalloc(sizeof(struct atmel_runtime_data), GFP_KERNEL); + if (prtd == NULL) { + ret = -ENOMEM; + goto out; + } + runtime->private_data = prtd; + + out: + return ret; +} + +static int atmel_pcm_close(struct snd_pcm_substream *substream) +{ + struct atmel_runtime_data *prtd = substream->runtime->private_data; + + kfree(prtd); + return 0; +} + +static struct snd_pcm_ops atmel_pcm_ops = { + .open = atmel_pcm_open, + .close = atmel_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = atmel_pcm_hw_params, + .hw_free = atmel_pcm_hw_free, + .prepare = atmel_pcm_prepare, + .trigger = atmel_pcm_trigger, + .pointer = atmel_pcm_pointer, + .mmap = atmel_pcm_mmap, +}; + + +/*--------------------------------------------------------------------------*\ + * ASoC platform driver +\*--------------------------------------------------------------------------*/ +#ifdef CONFIG_PM +static int atmel_pcm_suspend(struct snd_soc_dai *dai) +{ + struct snd_pcm_runtime *runtime = dai->runtime; + struct atmel_runtime_data *prtd; + struct atmel_pcm_dma_params *params; + + if (!runtime) + return 0; + + prtd = runtime->private_data; + params = prtd->params; + + /* disable the PDC and save the PDC registers */ + + ssc_writel(params->ssc->regs, PDC_PTCR, params->mask->pdc_disable); + + prtd->pdc_xpr_save = ssc_readx(params->ssc->regs, params->pdc->xpr); + prtd->pdc_xcr_save = ssc_readx(params->ssc->regs, params->pdc->xcr); + prtd->pdc_xnpr_save = ssc_readx(params->ssc->regs, params->pdc->xnpr); + prtd->pdc_xncr_save = ssc_readx(params->ssc->regs, params->pdc->xncr); + + return 0; +} + +static int atmel_pcm_resume(struct snd_soc_dai *dai) +{ + struct snd_pcm_runtime *runtime = dai->runtime; + struct atmel_runtime_data *prtd; + struct atmel_pcm_dma_params *params; + + if (!runtime) + return 0; + + prtd = runtime->private_data; + params = prtd->params; + + /* restore the PDC registers and enable the PDC */ + ssc_writex(params->ssc->regs, params->pdc->xpr, prtd->pdc_xpr_save); + ssc_writex(params->ssc->regs, params->pdc->xcr, prtd->pdc_xcr_save); + ssc_writex(params->ssc->regs, params->pdc->xnpr, prtd->pdc_xnpr_save); + ssc_writex(params->ssc->regs, params->pdc->xncr, prtd->pdc_xncr_save); + + ssc_writel(params->ssc->regs, PDC_PTCR, params->mask->pdc_enable); + return 0; +} +#else +#define atmel_pcm_suspend NULL +#define atmel_pcm_resume NULL +#endif + +static struct snd_soc_platform_driver atmel_soc_platform = { + .ops = &atmel_pcm_ops, + .pcm_new = atmel_pcm_new, + .pcm_free = atmel_pcm_free, + .suspend = atmel_pcm_suspend, + .resume = atmel_pcm_resume, +}; + +int atmel_pcm_pdc_platform_register(struct device *dev) +{ + return snd_soc_register_platform(dev, &atmel_soc_platform); +} +EXPORT_SYMBOL(atmel_pcm_pdc_platform_register); + +void atmel_pcm_pdc_platform_unregister(struct device *dev) +{ + snd_soc_unregister_platform(dev); +} +EXPORT_SYMBOL(atmel_pcm_pdc_platform_unregister); + +MODULE_AUTHOR("Sedji Gaouaou <sedji.gaouaou(a)atmel.com>"); +MODULE_DESCRIPTION("Atmel PCM module"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c index 40e17d1..e99f181 100644 --- a/sound/soc/atmel/atmel-pcm.c +++ b/sound/soc/atmel/atmel-pcm.c @@ -32,80 +32,25 @@ */ #include <linux/module.h> -#include <linux/init.h> -#include <linux/platform_device.h> -#include <linux/slab.h> #include <linux/dma-mapping.h> -#include <linux/atmel_pdc.h> -#include <linux/atmel-ssc.h> - -#include <sound/core.h> #include <sound/pcm.h> -#include <sound/pcm_params.h> #include <sound/soc.h> - #include "atmel-pcm.h" - -/*--------------------------------------------------------------------------*\ - * Hardware definition -\*--------------------------------------------------------------------------*/ -/* TODO: These values were taken from the AT91 platform driver, check - * them against real values for AT32 - */ -static const struct snd_pcm_hardware atmel_pcm_hardware = { - .info = SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_PAUSE, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - .period_bytes_min = 32, - .period_bytes_max = 8192, - .periods_min = 2, - .periods_max = 1024, - .buffer_bytes_max = 32 * 1024, -}; - - -/*--------------------------------------------------------------------------*\ - * Data types -\*--------------------------------------------------------------------------*/ -struct atmel_runtime_data { - struct atmel_pcm_dma_params *params; - dma_addr_t dma_buffer; /* physical address of dma buffer */ - dma_addr_t dma_buffer_end; /* first address beyond DMA buffer */ - size_t period_size; - - dma_addr_t period_ptr; /* physical address of next period */ - - /* PDC register save */ - u32 pdc_xpr_save; - u32 pdc_xcr_save; - u32 pdc_xnpr_save; - u32 pdc_xncr_save; -}; - - -/*--------------------------------------------------------------------------*\ - * Helper functions -\*--------------------------------------------------------------------------*/ static int atmel_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) { struct snd_pcm_substream *substream = pcm->streams[stream].substream; struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = atmel_pcm_hardware.buffer_bytes_max; + size_t size = ATMEL_SSC_DMABUF_SIZE; buf->dev.type = SNDRV_DMA_TYPE_DEV; buf->dev.dev = pcm->card->dev; buf->private_data = NULL; buf->area = dma_alloc_coherent(pcm->card->dev, size, - &buf->addr, GFP_KERNEL); - pr_debug("atmel-pcm:" - "preallocate_dma_buffer: area=%p, addr=%p, size=%d\n", - (void *) buf->area, - (void *) buf->addr, - size); + &buf->addr, GFP_KERNEL); + pr_debug("atmel-pcm: alloc dma buffer: area=%p, addr=%p, size=%d\n", + (void *)buf->area, (void *)buf->addr, size); if (!buf->area) return -ENOMEM; @@ -113,258 +58,19 @@ static int atmel_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, buf->bytes = size; return 0; } -/*--------------------------------------------------------------------------*\ - * ISR -\*--------------------------------------------------------------------------*/ -static void atmel_pcm_dma_irq(u32 ssc_sr, - struct snd_pcm_substream *substream) -{ - struct atmel_runtime_data *prtd = substream->runtime->private_data; - struct atmel_pcm_dma_params *params = prtd->params; - static int count; - - count++; - - if (ssc_sr & params->mask->ssc_endbuf) { - pr_warning("atmel-pcm: buffer %s on %s" - " (SSC_SR=%#x, count=%d)\n", - substream->stream == SNDRV_PCM_STREAM_PLAYBACK - ? "underrun" : "overrun", - params->name, ssc_sr, count); - - /* re-start the PDC */ - ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, - params->mask->pdc_disable); - prtd->period_ptr += prtd->period_size; - if (prtd->period_ptr >= prtd->dma_buffer_end) - prtd->period_ptr = prtd->dma_buffer; - - ssc_writex(params->ssc->regs, params->pdc->xpr, - prtd->period_ptr); - ssc_writex(params->ssc->regs, params->pdc->xcr, - prtd->period_size / params->pdc_xfer_size); - ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, - params->mask->pdc_enable); - } - - if (ssc_sr & params->mask->ssc_endx) { - /* Load the PDC next pointer and counter registers */ - prtd->period_ptr += prtd->period_size; - if (prtd->period_ptr >= prtd->dma_buffer_end) - prtd->period_ptr = prtd->dma_buffer; - - ssc_writex(params->ssc->regs, params->pdc->xnpr, - prtd->period_ptr); - ssc_writex(params->ssc->regs, params->pdc->xncr, - prtd->period_size / params->pdc_xfer_size); - } - - snd_pcm_period_elapsed(substream); -} - - -/*--------------------------------------------------------------------------*\ - * PCM operations -\*--------------------------------------------------------------------------*/ -static int atmel_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct atmel_runtime_data *prtd = runtime->private_data; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - - /* this may get called several times by oss emulation - * with different params */ - - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - runtime->dma_bytes = params_buffer_bytes(params); - - prtd->params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - prtd->params->dma_intr_handler = atmel_pcm_dma_irq; - - prtd->dma_buffer = runtime->dma_addr; - prtd->dma_buffer_end = runtime->dma_addr + runtime->dma_bytes; - prtd->period_size = params_period_bytes(params); - - pr_debug("atmel-pcm: " - "hw_params: DMA for %s initialized " - "(dma_bytes=%u, period_size=%u)\n", - prtd->params->name, - runtime->dma_bytes, - prtd->period_size); - return 0; -} - -static int atmel_pcm_hw_free(struct snd_pcm_substream *substream) -{ - struct atmel_runtime_data *prtd = substream->runtime->private_data; - struct atmel_pcm_dma_params *params = prtd->params; - - if (params != NULL) { - ssc_writex(params->ssc->regs, SSC_PDC_PTCR, - params->mask->pdc_disable); - prtd->params->dma_intr_handler = NULL; - } - - return 0; -} - -static int atmel_pcm_prepare(struct snd_pcm_substream *substream) -{ - struct atmel_runtime_data *prtd = substream->runtime->private_data; - struct atmel_pcm_dma_params *params = prtd->params; - - ssc_writex(params->ssc->regs, SSC_IDR, - params->mask->ssc_endx | params->mask->ssc_endbuf); - ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, - params->mask->pdc_disable); - return 0; -} - -static int atmel_pcm_trigger(struct snd_pcm_substream *substream, - int cmd) -{ - struct snd_pcm_runtime *rtd = substream->runtime; - struct atmel_runtime_data *prtd = rtd->private_data; - struct atmel_pcm_dma_params *params = prtd->params; - int ret = 0; - - pr_debug("atmel-pcm:buffer_size = %ld," - "dma_area = %p, dma_bytes = %u\n", - rtd->buffer_size, rtd->dma_area, rtd->dma_bytes); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - prtd->period_ptr = prtd->dma_buffer; - - ssc_writex(params->ssc->regs, params->pdc->xpr, - prtd->period_ptr); - ssc_writex(params->ssc->regs, params->pdc->xcr, - prtd->period_size / params->pdc_xfer_size); - - prtd->period_ptr += prtd->period_size; - ssc_writex(params->ssc->regs, params->pdc->xnpr, - prtd->period_ptr); - ssc_writex(params->ssc->regs, params->pdc->xncr, - prtd->period_size / params->pdc_xfer_size); - - pr_debug("atmel-pcm: trigger: " - "period_ptr=%lx, xpr=%u, " - "xcr=%u, xnpr=%u, xncr=%u\n", - (unsigned long)prtd->period_ptr, - ssc_readx(params->ssc->regs, params->pdc->xpr), - ssc_readx(params->ssc->regs, params->pdc->xcr), - ssc_readx(params->ssc->regs, params->pdc->xnpr), - ssc_readx(params->ssc->regs, params->pdc->xncr)); - - ssc_writex(params->ssc->regs, SSC_IER, - params->mask->ssc_endx | params->mask->ssc_endbuf); - ssc_writex(params->ssc->regs, SSC_PDC_PTCR, - params->mask->pdc_enable); - - pr_debug("sr=%u imr=%u\n", - ssc_readx(params->ssc->regs, SSC_SR), - ssc_readx(params->ssc->regs, SSC_IER)); - break; /* SNDRV_PCM_TRIGGER_START */ - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, - params->mask->pdc_disable); - break; - - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, - params->mask->pdc_enable); - break; - default: - ret = -EINVAL; - } - - return ret; -} - -static snd_pcm_uframes_t atmel_pcm_pointer( - struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct atmel_runtime_data *prtd = runtime->private_data; - struct atmel_pcm_dma_params *params = prtd->params; - dma_addr_t ptr; - snd_pcm_uframes_t x; - - ptr = (dma_addr_t) ssc_readx(params->ssc->regs, params->pdc->xpr); - x = bytes_to_frames(runtime, ptr - prtd->dma_buffer); - - if (x == runtime->buffer_size) - x = 0; - - return x; -} - -static int atmel_pcm_open(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct atmel_runtime_data *prtd; - int ret = 0; - - snd_soc_set_runtime_hwparams(substream, &atmel_pcm_hardware); - - /* ensure that buffer size is a multiple of period size */ - ret = snd_pcm_hw_constraint_integer(runtime, - SNDRV_PCM_HW_PARAM_PERIODS); - if (ret < 0) - goto out; - - prtd = kzalloc(sizeof(struct atmel_runtime_data), GFP_KERNEL); - if (prtd == NULL) { - ret = -ENOMEM; - goto out; - } - runtime->private_data = prtd; - - out: - return ret; -} - -static int atmel_pcm_close(struct snd_pcm_substream *substream) -{ - struct atmel_runtime_data *prtd = substream->runtime->private_data; - - kfree(prtd); - return 0; -} - -static int atmel_pcm_mmap(struct snd_pcm_substream *substream, +int atmel_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma) { return remap_pfn_range(vma, vma->vm_start, substream->dma_buffer.addr >> PAGE_SHIFT, vma->vm_end - vma->vm_start, vma->vm_page_prot); } +EXPORT_SYMBOL_GPL(atmel_pcm_mmap); -static struct snd_pcm_ops atmel_pcm_ops = { - .open = atmel_pcm_open, - .close = atmel_pcm_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = atmel_pcm_hw_params, - .hw_free = atmel_pcm_hw_free, - .prepare = atmel_pcm_prepare, - .trigger = atmel_pcm_trigger, - .pointer = atmel_pcm_pointer, - .mmap = atmel_pcm_mmap, -}; - - -/*--------------------------------------------------------------------------*\ - * ASoC platform driver -\*--------------------------------------------------------------------------*/ static u64 atmel_pcm_dmamask = DMA_BIT_MASK(32); -static int atmel_pcm_new(struct snd_soc_pcm_runtime *rtd) +int atmel_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; struct snd_pcm *pcm = rtd->pcm; @@ -376,6 +82,7 @@ static int atmel_pcm_new(struct snd_soc_pcm_runtime *rtd) card->dev->coherent_dma_mask = DMA_BIT_MASK(32); if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { + pr_debug("atmel-pcm: allocating PCM playback DMA buffer\n"); ret = atmel_pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_PLAYBACK); if (ret) @@ -383,8 +90,7 @@ static int atmel_pcm_new(struct snd_soc_pcm_runtime *rtd) } if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { - pr_debug("atmel-pcm:" - "Allocating PCM capture DMA buffer\n"); + pr_debug("atmel-pcm: allocating PCM capture DMA buffer\n"); ret = atmel_pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_CAPTURE); if (ret) @@ -393,8 +99,9 @@ static int atmel_pcm_new(struct snd_soc_pcm_runtime *rtd) out: return ret; } +EXPORT_SYMBOL_GPL(atmel_pcm_new); -static void atmel_pcm_free_dma_buffers(struct snd_pcm *pcm) +void atmel_pcm_free(struct snd_pcm *pcm) { struct snd_pcm_substream *substream; struct snd_dma_buffer *buf; @@ -413,78 +120,5 @@ static void atmel_pcm_free_dma_buffers(struct snd_pcm *pcm) buf->area = NULL; } } +EXPORT_SYMBOL_GPL(atmel_pcm_free); -#ifdef CONFIG_PM -static int atmel_pcm_suspend(struct snd_soc_dai *dai) -{ - struct snd_pcm_runtime *runtime = dai->runtime; - struct atmel_runtime_data *prtd; - struct atmel_pcm_dma_params *params; - - if (!runtime) - return 0; - - prtd = runtime->private_data; - params = prtd->params; - - /* disable the PDC and save the PDC registers */ - - ssc_writel(params->ssc->regs, PDC_PTCR, params->mask->pdc_disable); - - prtd->pdc_xpr_save = ssc_readx(params->ssc->regs, params->pdc->xpr); - prtd->pdc_xcr_save = ssc_readx(params->ssc->regs, params->pdc->xcr); - prtd->pdc_xnpr_save = ssc_readx(params->ssc->regs, params->pdc->xnpr); - prtd->pdc_xncr_save = ssc_readx(params->ssc->regs, params->pdc->xncr); - - return 0; -} - -static int atmel_pcm_resume(struct snd_soc_dai *dai) -{ - struct snd_pcm_runtime *runtime = dai->runtime; - struct atmel_runtime_data *prtd; - struct atmel_pcm_dma_params *params; - - if (!runtime) - return 0; - - prtd = runtime->private_data; - params = prtd->params; - - /* restore the PDC registers and enable the PDC */ - ssc_writex(params->ssc->regs, params->pdc->xpr, prtd->pdc_xpr_save); - ssc_writex(params->ssc->regs, params->pdc->xcr, prtd->pdc_xcr_save); - ssc_writex(params->ssc->regs, params->pdc->xnpr, prtd->pdc_xnpr_save); - ssc_writex(params->ssc->regs, params->pdc->xncr, prtd->pdc_xncr_save); - - ssc_writel(params->ssc->regs, PDC_PTCR, params->mask->pdc_enable); - return 0; -} -#else -#define atmel_pcm_suspend NULL -#define atmel_pcm_resume NULL -#endif - -static struct snd_soc_platform_driver atmel_soc_platform = { - .ops = &atmel_pcm_ops, - .pcm_new = atmel_pcm_new, - .pcm_free = atmel_pcm_free_dma_buffers, - .suspend = atmel_pcm_suspend, - .resume = atmel_pcm_resume, -}; - -int atmel_pcm_platform_register(struct device *dev) -{ - return snd_soc_register_platform(dev, &atmel_soc_platform); -} -EXPORT_SYMBOL(atmel_pcm_platform_register); - -void atmel_pcm_platform_unregister(struct device *dev) -{ - snd_soc_unregister_platform(dev); -} -EXPORT_SYMBOL(atmel_pcm_platform_unregister); - -MODULE_AUTHOR("Sedji Gaouaou <sedji.gaouaou(a)atmel.com>"); -MODULE_DESCRIPTION("Atmel PCM module"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/atmel/atmel-pcm.h b/sound/soc/atmel/atmel-pcm.h index e6d67b3..6bd63b0 100644 --- a/sound/soc/atmel/atmel-pcm.h +++ b/sound/soc/atmel/atmel-pcm.h @@ -36,6 +36,8 @@ #include <linux/atmel-ssc.h> +#define ATMEL_SSC_DMABUF_SIZE (64 * 1024) + /* * Registers and status bits that are required by the PCM driver. */ @@ -80,7 +82,22 @@ struct atmel_pcm_dma_params { #define ssc_readx(base, reg) (__raw_readl((base) + (reg))) #define ssc_writex(base, reg, value) __raw_writel((value), (base) + (reg)) -int atmel_pcm_platform_register(struct device *dev); -void atmel_pcm_platform_unregister(struct device *dev); +int atmel_pcm_new(struct snd_soc_pcm_runtime *rtd); +void atmel_pcm_free(struct snd_pcm *pcm); +int atmel_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma); + +#ifdef CONFIG_SND_ATMEL_SOC_PDC +int atmel_pcm_pdc_platform_register(struct device *dev); +void atmel_pcm_pdc_platform_unregister(struct device *dev); +#else +static inline int atmel_pcm_pdc_platform_register(struct device *dev) +{ + return 0; +} +static inline void atmel_pcm_pdc_platform_unregister(struct device *dev) +{ +} +#endif #endif /* _ATMEL_PCM_H */ diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 7932c05..1705614 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -729,7 +729,7 @@ static int asoc_ssc_init(struct device *dev) goto err; } - ret = atmel_pcm_platform_register(dev); + ret = atmel_pcm_pdc_platform_register(dev); if (ret) { dev_err(dev, "Could not register PCM: %d\n", ret); goto err_unregister_dai; @@ -745,7 +745,7 @@ err: static void asoc_ssc_exit(struct device *dev) { - atmel_pcm_platform_unregister(dev); + atmel_pcm_pdc_platform_unregister(dev); snd_soc_unregister_dai(dev); } -- 1.7.9.5

2 3

[alsa-devel] [PATCH V2 1/1] ASoC: ak5702: add support for ak5702 -- 4ch ADC
by Paolo Doz 02 Dec '12

02 Dec '12

This codec driver adds support for Asahi Kasei AK5702. Signed-off-by: Paolo Doz <paolo.doz(a)gmail.com> --- include/sound/ak5702.h | 27 +++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak5702.c | 507 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ak5702.h | 171 +++++++++++++++ 5 files changed, 711 insertions(+) diff --git a/include/sound/ak5702.h b/include/sound/ak5702.h new file mode 100644 index 0000000..8c0d4e9 --- /dev/null +++ b/include/sound/ak5702.h @@ -0,0 +1,27 @@ +/* + * AK5702 ALSA SoC Codec driver + * + * Author: Paolo Doz <paolo.doz(a)gmail.com> + * Copyright: (C) 2012 Soft In S.r.l. + * + * 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 __AK5702_H +#define __AK5702_H + +/** + * enum ak5702_clock mode + */ + +enum ak5702_clock_mode { + AK5702_CLKPLL_SLAVE, + AK5702_CLKPLL_SLAVE2, + AK5702_CLKPLL_MASTER, + AK5702_CLKEXT_SLAVE, + AK5702_CLKEXT_MASTER, +}; + +#endif /* __AK5702_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3a84782..d8156c7 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -26,6 +26,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4641 if I2C select SND_SOC_AK4642 if I2C select SND_SOC_AK4671 if I2C + select SND_SOC_AK5702 if I2C select SND_SOC_ALC5623 if I2C select SND_SOC_ALC5632 if I2C select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC @@ -202,6 +203,9 @@ config SND_SOC_AK4642 config SND_SOC_AK4671 tristate +config SND_SOC_AK5702 + tristate + config SND_SOC_ALC5623 tristate config SND_SOC_ALC5632 diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f6e8e36..1f88602 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -14,6 +14,7 @@ snd-soc-ak4535-objs := ak4535.o snd-soc-ak4641-objs := ak4641.o snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o +snd-soc-ak5702-objs := ak5702.o snd-soc-arizona-objs := arizona.o snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs42l51-objs := cs42l51.o @@ -136,6 +137,7 @@ obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_AK4641) += snd-soc-ak4641.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o +obj-$(CONFIG_SND_SOC_AK5702) += snd-soc-ak5702.o obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o obj-$(CONFIG_SND_SOC_ALC5632) += snd-soc-alc5632.o obj-$(CONFIG_SND_SOC_ARIZONA) += snd-soc-arizona.o diff --git a/sound/soc/codecs/ak5702.c b/sound/soc/codecs/ak5702.c new file mode 100644 index 0000000..9d35062 --- /dev/null +++ b/sound/soc/codecs/ak5702.c @@ -0,0 +1,507 @@ +/* + * ak5702.c -- AK5702 Soc Audio driver + * + * Author: Paolo Doz <paolo.doz(a)gmail.com> + * Copyright: (C) 2012 Soft In S.r.l. + * 2009 Freescale Semiconductor, Inc. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This code comes from the original Freescale Ak5702 Audio driver + */ + +/* #define DEBUG 1 */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.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 <linux/slab.h> +#include <sound/tlv.h> +#include <linux/regmap.h> + +#include <sound/ak5702.h> +#include "ak5702.h" + +#define AK5702_VERSION "0.5" +#define DRV_NAME "ak5702" + +/* codec private data */ +struct ak5702_priv { + struct regmap *regmap; + enum ak5702_clock_mode clock_mode; +}; + +/* + * ak5702 register cache + */ +static const struct reg_default ak5702_reg_defaults[] = { + { 0, 0x0000}, /* R0 - Power Management */ + { 1, 0x0024}, /* R1 - PLL Control */ + { 2, 0x0000}, /* R2 - Signal Select */ + { 3, 0x0001}, /* R3 - Mic Gain Control */ + { 4, 0x0023}, /* R4 - Audio Format Select */ + { 5, 0x001F}, /* R5 - fs Select */ + { 6, 0x0000}, /* R6 - Clock Output Select */ + { 7, 0x0001}, /* R7 - Volume Control */ + { 8, 0x0091}, /* R8 - Lch Input Volume Control */ + { 9, 0x0091}, /* R9 - Rch Input Volume Control */ + { 10, 0x0000}, /* R10 - Timer Select */ + { 11, 0x00E1}, /* R11 - ALC Mode Control 1 */ + { 12, 0x0000}, /* R12 - ALC Mode Control 2 */ + { 13, 0x00A0}, /* R13 - Mode Control 1 */ + { 14, 0x0000}, /* R14 - Mode Control 2 */ + { 15, 0x0000}, /* R15 - Mode Control 3 */ + { 16, 0x0000}, /* R16 - Power Management */ + { 17, 0x0000}, /* R17 - PLL Control */ + { 18, 0x0000}, /* R18 - Signal Select */ + { 19, 0x0001}, /* R19 - Mic Gain Control */ + { 20, 0x0020}, /* R20 - Audio Format Select */ + { 21, 0x0000}, /* R21 - fs Select */ + { 22, 0x0000}, /* R22 - Clock Output Select */ + { 23, 0x0001}, /* R23 - Volume Control */ + { 24, 0x0091}, /* R24 - Lch Input Volume Control */ + { 25, 0x0091}, /* R25 - Rch Input Volume Control */ + { 26, 0x0000}, /* R26 - Timer Select */ + { 27, 0x00E1}, /* R27 - ALC Mode Control 1 */ + { 28, 0x0000}, /* R28 - ALC Mode Control 2 */ + { 29, 0x0000}, /* R29 - Mode Control 1 */ + { 30, 0x0000}, /* R30 - Mode Control 2 */ +}; + +static bool ak5702_writeable(struct device *dev, unsigned int reg) +{ + return reg < AK5702_MAX_REGISTER; +} + +static const unsigned int mic_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 2, TLV_DB_SCALE_ITEM(0, 1500, 0), + 3, 3, TLV_DB_SCALE_ITEM(3600, 0, 0), +}; + +static const DECLARE_TLV_DB_SCALE(in_tlv, -5400, 37, 1); + +static const char *ak5702_adca_left12_input[] = { "Left1", "Left2" }; +static const char *ak5702_adca_right12_input[] = { "Right1", "Right2" }; +static const char *ak5702_adca_left5_input[] = { "Left1/2", "Left5" }; +static const char *ak5702_adca_right5_input[] = { "Right1/2", "Right5" }; + +static const char *ak5702_adcb_left12_input[] = { "Left3", "Left4" }; +static const char *ak5702_adcb_right12_input[] = { "Right3", "Right4" }; +static const char *ak5702_adcb_left5_input[] = { "Left3-4", "Left5" }; +static const char *ak5702_adcb_right5_input[] = { "Right3-4", "Right5" }; + +static const struct soc_enum ak5702_enum[] = { + SOC_ENUM_SINGLE(AK5702_SIG1, 0, 2, ak5702_adca_left12_input), + SOC_ENUM_SINGLE(AK5702_SIG1, 1, 2, ak5702_adca_right12_input), + SOC_ENUM_SINGLE(AK5702_SIG2, 0, 2, ak5702_adcb_left12_input), + SOC_ENUM_SINGLE(AK5702_SIG2, 1, 2, ak5702_adcb_right12_input), + SOC_ENUM_SINGLE(AK5702_SIG1, 5, 2, ak5702_adca_left5_input), + SOC_ENUM_SINGLE(AK5702_SIG1, 6, 2, ak5702_adca_right5_input), + SOC_ENUM_SINGLE(AK5702_SIG2, 5, 2, ak5702_adcb_left5_input), + SOC_ENUM_SINGLE(AK5702_SIG2, 6, 2, ak5702_adcb_right5_input), +}; + +static const struct snd_kcontrol_new ak5702_snd_controls[] = { + SOC_ENUM("ADCA Left1/2 Capture Source", ak5702_enum[0]), + SOC_ENUM("ADCA Right1/2 Capture Source", ak5702_enum[1]), + SOC_ENUM("ADCB Left1/2 Capture Source", ak5702_enum[2]), + SOC_ENUM("ADCB Right1/2 Capture Source", ak5702_enum[3]), + SOC_ENUM("ADCA Left5 Capture Source", ak5702_enum[4]), + SOC_ENUM("ADCA Right5 Capture Source", ak5702_enum[5]), + SOC_ENUM("ADCB Left5 Capture Source", ak5702_enum[6]), + SOC_ENUM("ADCB Right5 Capture Source", ak5702_enum[7]), + SOC_SINGLE_TLV("Mic A Boost Gain", AK5702_MICG1, 0, 3, 0, mic_tlv), + SOC_SINGLE_TLV("Mic B Boost Gain", AK5702_MICG2, 0, 3, 0, mic_tlv), + SOC_DOUBLE_R_TLV("ADCA Capture Volume", AK5702_LVOL1, AK5702_RVOL1, 0, + 241, 0, in_tlv), + SOC_DOUBLE_R_TLV("ADCB Capture Volume", AK5702_LVOL2, AK5702_RVOL2, 0, + 241, 0, in_tlv), +}; + +/* ak5702 dapm widgets */ +static const struct snd_soc_dapm_widget ak5702_dapm_widgets[] = { + SND_SOC_DAPM_ADC("ADCA Left", "Capture", AK5702_PM1, 0, 0), + SND_SOC_DAPM_ADC("ADCA Right", "Capture", AK5702_PM1, 1, 0), + SND_SOC_DAPM_ADC("ADCB Left", "Capture", AK5702_PM2, 0, 0), + SND_SOC_DAPM_ADC("ADCB Right", "Capture", AK5702_PM2, 1, 0), + + SND_SOC_DAPM_INPUT("ADCA Left Input"), + SND_SOC_DAPM_INPUT("ADCA Right Input"), + SND_SOC_DAPM_INPUT("ADCB Left Input"), + SND_SOC_DAPM_INPUT("ADCB Right Input"), +}; + +static const struct snd_soc_dapm_route ak5702_dapm_routes[] = { + {"ADCA Left", NULL, "ADCA Left Input"}, + {"ADCA Right", NULL, "ADCA Right Input"}, + {"ADCB Left", NULL, "ADCB Left Input"}, + {"ADCB Right", NULL, "ADCB Right Input"}, +}; + +static int ak5702_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct ak5702_priv *ak5702 = snd_soc_codec_get_drvdata(codec); + u8 value; + + switch (level) { + case SND_SOC_BIAS_ON: + /* power on ADCA */ + value = AK5702_PM1_PMADAL | + AK5702_PM1_PMADAR | + AK5702_PM1_PMVCM; + snd_soc_write(codec, AK5702_PM1, value); + /* power up ADCB */ + value = AK5702_PM2_PMADBL | AK5702_PM2_PMADBR; + snd_soc_write(codec, AK5702_PM2, value); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + regcache_sync(ak5702->regmap); + /* power down ADCA */ + value = AK5702_POWERDOWN | AK5702_PM1_PMVCM; + snd_soc_write(codec, AK5702_PM1, value); + /* power down ADCB */ + value = AK5702_POWERDOWN; + snd_soc_write(codec, AK5702_PM2, value); + break; + case SND_SOC_BIAS_OFF: + /* power down */ + value = AK5702_POWERDOWN; + snd_soc_write(codec, AK5702_PM1, value); + value = AK5702_POWERDOWN; + snd_soc_write(codec, AK5702_PM2, value); + regcache_mark_dirty(ak5702->regmap); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static int ak5702_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + u8 value; + struct snd_soc_codec *codec = dai->codec; + struct ak5702_priv *ak5702 = snd_soc_codec_get_drvdata(codec); + + pr_debug("Entered %s\n", __func__); + + switch (params_channels(params)) { + case 2: + snd_soc_update_bits(codec, AK5702_FMT1, AK5702_FMT1_TDM_MASK, + AK5702_FMT1_STEREO); + break; + case 8: + if (params_format(params) == SNDRV_PCM_FORMAT_S16_LE) + value = AK5702_FMT1_TDM128; + else if (params_format(params) == SNDRV_PCM_FORMAT_S32_LE) + value = AK5702_FMT1_TDM256; + else + return -EINVAL; + snd_soc_update_bits(codec, AK5702_FMT1, AK5702_FMT1_TDM_MASK, + value); + break; + default: + return -EINVAL; + } + + switch (params_rate(params)) { + case 48000: + if (ak5702->clock_mode == AK5702_CLKPLL_SLAVE2) + value = AK5702_FS1_BCLK_MODE2; + else if (ak5702->clock_mode == AK5702_CLKEXT_SLAVE) + value = AK5702_FS1_SLAVE_256FS; /* mode 3 256fs*/ + else + return -EINVAL; + break; + default: + return -EINVAL; + } + snd_soc_update_bits(codec, AK5702_FS1, + AK5702_FS1_FS_MASK | AK5702_FS1_BCKO_MASK, value); + return 0; +} + +static int ak5702_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 ak5702_priv *ak5702 = snd_soc_codec_get_drvdata(codec); + u8 value = 0; + + pr_debug("Entered %s\n", __func__); + + switch (clk_id) { + case AK5702_CLKPLL_MASTER: + case AK5702_CLKEXT_MASTER: + if (dir != SND_SOC_CLOCK_OUT) + return -EINVAL; + break; + case AK5702_CLKPLL_SLAVE: + case AK5702_CLKPLL_SLAVE2: + if (dir != SND_SOC_CLOCK_IN) + return -EINVAL; + value = AK5702_PLL1_POWERUP | + AK5702_PLL1_SLAVE | + AK5702_PLL1_MODE2; + break; + case AK5702_CLKEXT_SLAVE: + if (dir != SND_SOC_CLOCK_IN) + return -EINVAL; + value = AK5702_PLL1_POWERDOWN | AK5702_PLL1_SLAVE; + break; + default: + return -EINVAL; + } + snd_soc_write(codec, AK5702_PLL1, value); + ak5702->clock_mode = clk_id; + return 0; +} + +static int ak5702_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 value = 0; + pr_debug("Entered %s", __func__); + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_I2S: + value = AK5702_FMT1_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + value = AK5702_FMT1_MSB; + break; + default: + return -EINVAL; + } + snd_soc_update_bits(codec, AK5702_FMT1, AK5702_FMT1_FMT_MASK, value); + snd_soc_write(codec, AK5702_FMT2, AK5702_FMT2_STEREO); + return 0; +} + +static int ak5702_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 reg = 0; + pr_debug("Entered %s", __func__); + reg = snd_soc_read(codec, AK5702_PLL1); + switch (pll_id) { + case AK5702_PLL_POWERDOWN: + reg &= (~AK5702_PLL1_PM_MASK); + reg |= AK5702_PLL1_POWERDOWN; + break; + case AK5702_PLL_MASTER: + reg &= (~AK5702_PLL1_MODE_MASK); + reg |= AK5702_PLL1_MASTER | AK5702_PLL1_POWERUP; + break; + case AK5702_PLL_SLAVE: + reg &= (~AK5702_PLL1_MODE_MASK); + reg |= AK5702_PLL1_SLAVE | AK5702_PLL1_POWERUP; + break; + default: + return -ENODEV; + } + reg &= (~AK5702_PLL1_PLL_MASK); + switch (freq_in) { + case 11289600: + reg |= AK5702_PLL1_11289600; + break; + case 12000000: + reg |= AK5702_PLL1_12000000; + break; + case 12288000: + reg |= AK5702_PLL1_12288000; + break; + case 19200000: + reg |= AK5702_PLL1_19200000; + break; + default: + return -ENODEV; + } + snd_soc_write(codec, AK5702_PLL1, reg); + return 0; +} + +static int ak5702_set_dai_clkdiv(struct snd_soc_dai *codec_dai, + int div_id, int div) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 value = 0; + pr_debug("Entered %s", __func__); + if (div_id == AK5702_BCLK_CLKDIV) { + switch (div) { + case AK5702_BCLK_DIV_32: + value = AK5702_FS1_BCKO_32FS; + break; + case AK5702_BCLK_DIV_64: + value = AK5702_FS1_BCKO_64FS; + break; + default: + return -EINVAL; + } + } + snd_soc_update_bits(codec, AK5702_FS1, AK5702_FS1_BCKO_MASK, value); + return 0; +} + +#ifdef CONFIG_PM +static int ak5702_suspend(struct snd_soc_codec *codec) +{ + ak5702_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int ak5702_resume(struct snd_soc_codec *codec) +{ + ak5702_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} +#else +#define ak5702_suspend NULL +#define ak5702_resume NULL +#endif + +static const struct snd_soc_dai_ops ak5702_ops = { + .set_sysclk = ak5702_set_dai_sysclk, + .set_pll = ak5702_set_dai_pll, + .set_clkdiv = ak5702_set_dai_clkdiv, + .set_fmt = ak5702_set_dai_fmt, + .hw_params = ak5702_dai_hw_params, +}; + +struct snd_soc_dai_driver ak5702_dai = { + .name = "ak5702-hifi", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &ak5702_ops, +}; + +static int ak5702_probe(struct snd_soc_codec *codec) +{ + struct ak5702_priv *ak5702 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + dev_info(codec->dev, "AK5702 Audio Codec %s", AK5702_VERSION); + + codec->control_data = ak5702->regmap; + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + ak5702_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return ret; +} + +static int ak5702_remove(struct snd_soc_codec *codec) +{ + ak5702_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static const struct regmap_config ak5702_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = AK5702_MAX_REGISTER, + .writeable_reg = ak5702_writeable, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = ak5702_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(ak5702_reg_defaults), +}; + +static __devinit int ak5702_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ak5702_priv *ak5702; + int ret; + pr_debug("Entered %s", __func__); + + ak5702 = devm_kzalloc(&i2c->dev, sizeof(struct ak5702_priv), + GFP_KERNEL); + if (!ak5702) + return -ENOMEM; + + ak5702->regmap = devm_regmap_init_i2c(i2c, &ak5702_regmap); + if (IS_ERR(ak5702->regmap)) { + ret = PTR_ERR(ak5702->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + i2c_set_clientdata(i2c, ak5702); + + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_ak5702, &ak5702_dai, 1); + if (ret != 0) + dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret); + + return ret; +} + +static __devexit int ak5702_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id ak5702_i2c_id[] = { + { "ak5702-codec", 0 }, + { } +}; + +MODULE_DEVICE_TABLE(i2c, ak5702_i2c_id); + +static struct i2c_driver ak5702_i2c_driver = { + .driver = { + .name = "ak5702-codec", + .owner = THIS_MODULE, + }, + .probe = ak5702_i2c_probe, + .remove = __devexit_p(ak5702_i2c_remove), + .id_table = ak5702_i2c_id, +}; + +struct snd_soc_codec_driver soc_codec_dev_ak5702 = { + .probe = ak5702_probe, + .remove = ak5702_remove, + .suspend = ak5702_suspend, + .resume = ak5702_resume, + .set_bias_level = ak5702_set_bias_level, + .controls = ak5702_snd_controls, + .num_controls = ARRAY_SIZE(ak5702_snd_controls), + .dapm_widgets = ak5702_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ak5702_dapm_widgets), + .dapm_routes = ak5702_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(ak5702_dapm_routes), +}; + +module_i2c_driver(ak5702_i2c_driver); + +MODULE_DESCRIPTION("Asahi Kasei AK5702 ALSA SoC driver"); +MODULE_AUTHOR("Paolo Doz <paolo.doz(a)gmail.com"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ak5702.h b/sound/soc/codecs/ak5702.h new file mode 100644 index 0000000..a492a61 --- /dev/null +++ b/sound/soc/codecs/ak5702.h @@ -0,0 +1,171 @@ +/* + * ak5702.h -- AK5702 Soc Audio driver + * + * Author: Paolo Doz <paolo.doz(a)gmail.com> + * Copyright: (C) 2012 Soft In S.r.l. + * 2009 Freescale Semiconductor, Inc. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This code comes from the original Freescale Ak5702 Audio driver + */ + +#ifndef _AK5702_H +#define _AK5702_H + +/* AK5702 register space */ + +#define AK5702_PM1 0x00 +#define AK5702_PLL1 0x01 +#define AK5702_SIG1 0x02 +#define AK5702_MICG1 0x03 +#define AK5702_FMT1 0x04 +#define AK5702_FS1 0x05 +#define AK5702_CLK1 0x06 +#define AK5702_VOLCTRL1 0x07 +#define AK5702_LVOL1 0x08 +#define AK5702_RVOL1 0x09 +#define AK5702_TIMER1 0x0A +#define AK5702_ALC11 0x0B +#define AK5702_ALC12 0x0C +#define AK5702_MODE11 0x0D +#define AK5702_MODE12 0x0E +#define AK5702_MODE13 0x0F + +#define AK5702_PM2 0x10 +#define AK5702_PLL2 0x11 +#define AK5702_SIG2 0x12 +#define AK5702_MICG2 0x13 +#define AK5702_FMT2 0x14 +#define AK5702_FS2 0x15 +#define AK5702_CLK2 0x16 +#define AK5702_VOLCTRL2 0x17 +#define AK5702_LVOL2 0x18 +#define AK5702_RVOL2 0x19 +#define AK5702_TIMER2 0x1A +#define AK5702_ALC21 0x1B +#define AK5702_ALC22 0x1C +#define AK5702_MODE21 0x1D +#define AK5702_MODE22 0x1E + +#define AK5702_MAX_REGISTER 0x1F + +#define AK5702_CACHEREGNUM 0x1F + +#define AK5702_POWERDOWN 0x0 +#define AK5702_PM1_PMADAL 0x01 +#define AK5702_PM1_PMADAR 0x02 +#define AK5702_PM1_PMVCM 0x04 +#define AK5702_PM2_PMADBL 0x01 +#define AK5702_PM2_PMADBR 0x02 + +#define AK5702_PLL1_POWERDOWN 0x0 +#define AK5702_PLL1_POWERUP 0x01 +#define AK5702_PLL1_MASTER 0x02 +#define AK5702_PLL1_SLAVE 0x0 +#define AK5702_PLL1_MODE0 0x0 +#define AK5702_PLL1_MODE2 0x08 +#define AK5702_PLL1_MODE3 0x0C +#define AK5702_PLL1_MODE4 0x10 +#define AK5702_PLL1_MODE5 0x14 +#define AK5702_PLL1_MODE6 0x18 +#define AK5702_PLL1_MODE7 0x1C +#define AK5702_PLL1_MODE8 0x20 +#define AK5702_PLL1_MODE9 0x24 +#define AK5702_PLL1_MODE12 0x30 +#define AK5702_PLL1_MODE13 0x34 +#define AK5702_PLL1_MODE14 0x38 +#define AK5702_PLL1_MODE15 0x3C +#define AK5702_PLL1_11289600 0x10 +#define AK5702_PLL1_12000000 0x24 +#define AK5702_PLL1_12288000 0x14 +#define AK5702_PLL1_19200000 0x20 + +#define AK5702_FS1_MCKI_MODE0 0x0 +#define AK5702_FS1_MCKI_MODE1 0x01 +#define AK5702_FS1_MCKI_MODE2 0x02 +#define AK5702_FS1_MCKI_MODE3 0x03 +#define AK5702_FS1_MCKI_MODE4 0x04 +#define AK5702_FS1_MCKI_MODE5 0x05 +#define AK5702_FS1_MCKI_MODE6 0x06 +#define AK5702_FS1_MCKI_MODE7 0x07 +#define AK5702_FS1_MCKI_MODE10 0x0A +#define AK5702_FS1_MCKI_MODE11 0x0B +#define AK5702_FS1_MCKI_MODE14 0x0E +#define AK5702_FS1_MCKI_MODE15 0x0F + +#define AK5702_FS1_BCLK_MODE0 0x0 +#define AK5702_FS1_BCLK_MODE1 0x04 +#define AK5702_FS1_BCLK_MODE2 0x08 + +#define AK5702_FS1_SLAVE_1024FS 0x01 +#define AK5702_FS1_SLAVE_512FS 0x02 +#define AK5702_FS1_SLAVE_256FS 0x03 + +#define AK5702_SIG1_L_LIN1 0x0 +#define AK5702_SIG1_L_LIN2 0x01 +#define AK5702_SIG1_R_RIN1 0x0 +#define AK5702_SIG1_R_RIN2 0x02 +#define AK5702_SIG1_PMMPA 0x10 +#define AK5702_SIG1_L_LIN5 0x20 +#define AK5702_SIG1_R_RIN5 0x40 +#define AK5702_SIG2_L_LIN3 0x0 +#define AK5702_SIG2_L_LIN4 0x01 +#define AK5702_SIG2_R_RIN3 0x0 +#define AK5702_SIG2_R_RIN4 0x02 +#define AK5702_SIG2_PMMPB 0x10 + +#define AK5702_MICGAIN_0DB 0x0 + +#define AK5702_FMT1_I2S 0x03 +#define AK5702_FMT1_MSB 0x02 +#define AK5702_FMT1_STEREO 0x00 +#define AK5702_FMT1_TDM128 0x60 +#define AK5702_FMT1_TDM256 0xE0 +#define AK5702_FMT2_STEREO 0x20 +#define AK5702_FS1_BCKO_32FS 0x10 +#define AK5702_FS1_BCKO_64FS 0x20 +#define AK5702_CLK1_PS_256FS 0x0 +#define AK5702_CLK1_PS_128FS 0x01 +#define AK5702_CLK1_PS_64FS 0x02 +#define AK5702_CLK1_PS_32FS 0x03 +#define AK5702_VOLCTRL_INDEP 0x0 +#define AK5702_VOLCTRL_DEP 0x01 +#define AK5702_VOL_INIT 0x91 + +#define AK5702_PM1_MASK 0x07 +#define AK5702_PM2_MASK 0x03 +#define AK5702_PLL1_PM_MASK 0x01 +#define AK5702_PLL1_MODE_MASK 0x02 +#define AK5702_PLL1_PLL_MASK 0x3C +#define AK5702_FS1_BCKO_MASK 0x30 +#define AK5702_FS1_FS_MASK 0x0F +#define AK5702_CLK1_PS_MASK 0x03 +#define AK5702_FMT1_FMT_MASK 0x03 +#define AK5702_FMT1_TDM_MASK 0xC0 +#define AK5702_FMT2_OUT_MASK 0x10 +/* clock divider id */ +#define AK5702_BCLK_CLKDIV 0 +#define AK5702_MCLK_CLKDIV 1 + +/* bit clock div values */ +#define AK5702_BCLK_DIV_32 0 +#define AK5702_BCLK_DIV_64 1 + +/* m clock div values */ +#define AK5702_MCLK_DIV_32 0 +#define AK5702_MCLK_DIV_64 1 +#define AK5702_MCLK_DIV_128 2 +#define AK5702_MCLK_DIV_256 3 + +/* PLL master and slave modes */ +#define AK5702_PLL_POWERDOWN 0 +#define AK5702_PLL_MASTER 1 +#define AK5702_PLL_SLAVE 2 + +extern struct snd_soc_dai_driver ak5702_dai; +extern struct snd_soc_codec_driver soc_codec_dev_ak5702; + +#endif

2 1

[alsa-devel] [PATCH 1/6] ALSA: PCM: Remove redundant null check before kfree
by Sachin Kamat 02 Dec '12

02 Dec '12

kfree on a null pointer is a no-op. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- Build tested and based on linux-next 20121115. --- sound/core/pcm.c | 3 +-- 1 files changed, 1 insertions(+), 2 deletions(-) diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 030102c..61798f8 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -981,8 +981,7 @@ void snd_pcm_detach_substream(struct snd_pcm_substream *substream) PAGE_ALIGN(sizeof(struct snd_pcm_mmap_control))); kfree(runtime->hw_constraints.rules); #ifdef CONFIG_SND_PCM_XRUN_DEBUG - if (runtime->hwptr_log) - kfree(runtime->hwptr_log); + kfree(runtime->hwptr_log); #endif kfree(runtime); substream->runtime = NULL; -- 1.7.4.1

3 14

[alsa-devel] [PATCH 1/1] ASoC: wm9090: Use devm_regmap_init_i2c()
by Sachin Kamat 02 Dec '12

02 Dec '12

devm_regmap_init_i2c() is device managed and makes error handling and code cleanup simpler. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/codecs/wm9090.c | 20 ++++++-------------- 1 files changed, 6 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index c7ddc56..bb55a70 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -628,7 +628,7 @@ static int wm9090_i2c_probe(struct i2c_client *i2c, return -ENOMEM; } - wm9090->regmap = regmap_init_i2c(i2c, &wm9090_regmap); + wm9090->regmap = devm_regmap_init_i2c(i2c, &wm9090_regmap); if (IS_ERR(wm9090->regmap)) { ret = PTR_ERR(wm9090->regmap); dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret); @@ -637,16 +637,16 @@ static int wm9090_i2c_probe(struct i2c_client *i2c, ret = regmap_read(wm9090->regmap, WM9090_SOFTWARE_RESET, &reg); if (ret < 0) - goto err; + return ret; + if (reg != 0x9093) { dev_err(&i2c->dev, "Device is not a WM9090, ID=%x\n", reg); - ret = -ENODEV; - goto err; + return -ENODEV; } ret = regmap_write(wm9090->regmap, WM9090_SOFTWARE_RESET, 0); if (ret < 0) - goto err; + return ret; if (i2c->dev.platform_data) memcpy(&wm9090->pdata, i2c->dev.platform_data, @@ -658,23 +658,15 @@ static int wm9090_i2c_probe(struct i2c_client *i2c, &soc_codec_dev_wm9090, NULL, 0); if (ret != 0) { dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret); - goto err; + return ret; } return 0; - -err: - regmap_exit(wm9090->regmap); - return ret; } static int __devexit wm9090_i2c_remove(struct i2c_client *i2c) { - struct wm9090_priv *wm9090 = i2c_get_clientdata(i2c); - snd_soc_unregister_codec(&i2c->dev); - regmap_exit(wm9090->regmap); - return 0; } -- 1.7.4.1

2 1

[alsa-devel] [PATCH - hdsp.c 1/1] HDSP fixes for RPM
by Karl Grill 01 Dec '12

01 Dec '12

- changed iobox recognition and firmware loading to mimick XP driver - added loop to check_for_iobox() to fix timing issues Signed-off-by: Karl Grill <kgrill(a)chello.at> diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 0d6930c..4229b0d 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -155,10 +155,11 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin"); #define HDSP_BIGENDIAN_MODE 0x200 #define HDSP_RD_MULTIPLE 0x400 #define HDSP_9652_ENABLE_MIXER 0x800 +#define HDSP_DUNNO 0x1000 #define HDSP_TDO 0x10000000 -#define HDSP_S_PROGRAM (HDSP_PROGRAM|HDSP_CONFIG_MODE_0) -#define HDSP_S_LOAD (HDSP_PROGRAM|HDSP_CONFIG_MODE_1) +#define HDSP_S_PROGRAM (HDSP_DUNNO|HDSP_PROGRAM|HDSP_CONFIG_MODE_0) +#define HDSP_S_LOAD (HDSP_DUNNO|HDSP_PROGRAM|HDSP_CONFIG_MODE_1) /* Control Register bits */ @@ -676,13 +677,20 @@ static unsigned int hdsp_read(struct hdsp *hdsp, int reg) static int hdsp_check_for_iobox (struct hdsp *hdsp) { + int i; + if (hdsp->io_type == H9652 || hdsp->io_type == H9632) return 0; - if (hdsp_read (hdsp, HDSP_statusRegister) & HDSP_ConfigError) { - snd_printk("Hammerfall-DSP: no IO box connected!\n"); - hdsp->state &= ~HDSP_FirmwareLoaded; - return -EIO; + for(i=0; i<10000; i++) { + if ((hdsp_read (hdsp, HDSP_statusRegister) & HDSP_ConfigError)==0) { + if(i) + snd_printk("Hammerfall-DSP: IO box found after %d ms\n", i); + return 0; + } + msleep(1); } - return 0; + snd_printk("Hammerfall-DSP: no IO box connected!\n"); + hdsp->state &= ~HDSP_FirmwareLoaded; + return -EIO; } static int hdsp_wait_for_iobox(struct hdsp *hdsp, unsigned int loops, @@ -722,6 +730,7 @@ static int snd_hdsp_load_firmware_from_cache(struct hdsp *hdsp) { if (hdsp_fifo_wait (hdsp, 0, HDSP_LONG_WAIT)) { snd_printk ("Hammerfall-DSP: timeout waiting for download preparation\n"); + hdsp_write (hdsp, HDSP_control2Reg, 0x800); return -EIO; } @@ -731,25 +740,25 @@ static int snd_hdsp_load_firmware_from_cache(struct hdsp *hdsp) { hdsp_write(hdsp, HDSP_fifoData, hdsp->firmware_cache[i]); if (hdsp_fifo_wait (hdsp, 127, HDSP_LONG_WAIT)) { snd_printk ("Hammerfall-DSP: timeout during firmware loading\n"); + hdsp_write (hdsp, HDSP_control2Reg, 0x800); return -EIO; } } - ssleep(3); - - if (hdsp_fifo_wait (hdsp, 0, HDSP_LONG_WAIT)) { - snd_printk ("Hammerfall-DSP: timeout at end of firmware loading\n"); - return -EIO; - } + hdsp_fifo_wait (hdsp, 3, HDSP_LONG_WAIT); + hdsp_write (hdsp, HDSP_control2Reg, 0x800); + ssleep(3); #ifdef SNDRV_BIG_ENDIAN hdsp->control2_register = HDSP_BIGENDIAN_MODE; #else hdsp->control2_register = 0; #endif + hdsp_write (hdsp, HDSP_control2Reg, hdsp->control2_register); snd_printk ("Hammerfall-DSP: finished firmware loading\n"); + } if (hdsp->state & HDSP_InitializationComplete) { snd_printk(KERN_INFO "Hammerfall-DSP: firmware loaded from cache, restoring defaults\n"); @@ -767,24 +776,46 @@ static int hdsp_get_iobox_version (struct hdsp *hdsp) { if ((hdsp_read (hdsp, HDSP_statusRegister) & HDSP_DllError) != 0) { - hdsp_write (hdsp, HDSP_control2Reg, HDSP_PROGRAM); + hdsp_write (hdsp, HDSP_control2Reg, 0x10a0); hdsp_write (hdsp, HDSP_fifoData, 0); - if (hdsp_fifo_wait (hdsp, 0, HDSP_SHORT_WAIT) < 0) - return -EIO; - - hdsp_write (hdsp, HDSP_control2Reg, HDSP_S_LOAD); + if (hdsp_fifo_wait (hdsp, 0, HDSP_SHORT_WAIT) < 0) { + hdsp_write(hdsp, HDSP_control2Reg, 0x900); + hdsp_write(hdsp, HDSP_control2Reg, 0x10a0); + } + hdsp_write(hdsp, HDSP_control2Reg, 0x820); hdsp_write (hdsp, HDSP_fifoData, 0); - - if (hdsp_fifo_wait(hdsp, 0, HDSP_SHORT_WAIT)) { - hdsp_write(hdsp, HDSP_control2Reg, HDSP_VERSION_BIT); - hdsp_write(hdsp, HDSP_control2Reg, HDSP_S_LOAD); - if (hdsp_fifo_wait(hdsp, 0, HDSP_SHORT_WAIT)) - hdsp->io_type = RPM; - else - hdsp->io_type = Multiface; - } else { - hdsp->io_type = Digiface; + if (hdsp_fifo_wait (hdsp, 0, HDSP_SHORT_WAIT) < 0) { + hdsp->io_type=Multiface; + snd_printk("Hammerfall-DSP: Multiface found\n"); + return 0; + } + hdsp_write(hdsp, HDSP_control2Reg, 0x10a0); + hdsp_write (hdsp, HDSP_fifoData, 0); + if (hdsp_fifo_wait (hdsp, 0, HDSP_SHORT_WAIT) == 0) { + hdsp->io_type=Digiface; + snd_printk("Hammerfall-DSP: Digiface found\n"); + return 0; + } + hdsp_write(hdsp, HDSP_control2Reg, 0x900); + hdsp_write(hdsp, HDSP_control2Reg, 0x10a0); + hdsp_write (hdsp, HDSP_fifoData, 0); + if (hdsp_fifo_wait (hdsp, 0, HDSP_SHORT_WAIT) == 0) { + hdsp->io_type=Multiface; + snd_printk("Hammerfall-DSP: Multiface found\n"); + return 0; + } + hdsp_write(hdsp, HDSP_control2Reg, 0x900); + hdsp_write(hdsp, HDSP_control2Reg, 0x10a0); + hdsp_write (hdsp, HDSP_fifoData, 0); + if (hdsp_fifo_wait (hdsp, 0, HDSP_SHORT_WAIT) < 0) { + hdsp->io_type=Multiface; + snd_printk("Hammerfall-DSP: Multiface found\n"); + return 0; } + hdsp->io_type=RPM; + snd_printk("Hammerfall-DSP: RPM found\n"); + return 0; + } else { /* firmware was already loaded, get iobox type */ if (hdsp_read(hdsp, HDSP_status2Register) & HDSP_version2) -- 1.7.2.5

2 2

[alsa-devel] Preliminary driver for RME MADI FX
by Adrian Knoth 01 Dec '12

01 Dec '12

Hi! I've been working on a driver for the new 390 channels RME MADI FX. For those who are interested, I've made the source available here: https://github.com/adiknoth/madifx As mentioned there, I only had remote access, so not all possible hardware configurations have been tested. This might change next year, until then, consider everything as "coincidentally working" ;) Most importantly, the userspace API isn't stable, yet, since we still have to develop new tools for metering and mixer adjustments. If we exclude this API (the ioctls) for now, it is moderately safe to go mainline, though we definitely lack decent test coverage. ALSA devs: Feel free to demand any changes that are required for mainline inclusion. The github repo is meant to serve as an incubator from which we'll eventually derive the final kernel patch. Cheers

1 0

Re: [alsa-devel] [PATCH 1/2] ASoC: Samsung: WM8994: Add basic device tree support for machine file
by Mark Brown 01 Dec '12

01 Dec '12

On Sat, Dec 01, 2012 at 03:37:26PM +0530, Padmavathi Venna wrote: > Add the basic device tree based lookup. > > Signed-off-by: Padmavathi Venna <padma.v(a)samsung.com> > --- > sound/soc/samsung/smdk_wm8994.c | 23 +++++++++++++++++++++++ Any new device tree bindings need binding documentation. Your subject line is also very misleading here... it should say that it's talking about the SMDK. There's at least two other WM8994 based Samsung boards in mainline.

1 0

[alsa-devel] [PATCH 1/1 v2] ALSA: CA0132 - Handle endianness of SCP buffer data
by Ian Minett 01 Dec '12

01 Dec '12

From: Ian Minett <ian_minett(a)creativelabs.com> Thanks for the info - we've changed the previous unaligned access calls to use cpu_to_le32() and le32_to_cpu() instead for the swapping operations. - Ian -- Handle endianness of the SCP buffer being sent to or from the DSP chip on big-endian host architectures. The SCP buffer is prepared in dspio_scp(), and bytes are swapped before being transferred by the codec in dspio_send(). Signed-off-by: Ian Minett <ian_minett(a)creativelabs.com> diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 2fd3121..fa19219 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -701,11 +701,15 @@ static int dspio_send(struct hda_codec *codec, unsigned int reg, unsigned int res; int retry = 50; + /* Handling for Big Endian machine */ + unsigned int val = cpu_to_le32(data); + /* send bits of data specified by reg to dsp */ do { - res = snd_hda_codec_read(codec, WIDGET_DSP_CTRL, 0, reg, data); - if ((res >= 0) && (res != VENDOR_STATUS_DSPIO_BUSY)) - return res; + res = snd_hda_codec_read(codec, WIDGET_DSP_CTRL, 0, reg, val); + val = le32_to_cpu(res); + if ((val >= 0) && (val != VENDOR_STATUS_DSPIO_BUSY)) + return val; } while (--retry); return -EIO; @@ -719,11 +723,15 @@ static void dspio_write_wait(struct hda_codec *codec) int status; unsigned long timeout = jiffies + msecs_to_jiffies(1000); + /* Handling for Big Endian machine */ + unsigned int val; + do { status = snd_hda_codec_read(codec, WIDGET_DSP_CTRL, 0, VENDOR_DSPIO_STATUS, 0); - if ((status == VENDOR_STATUS_DSPIO_OK) || - (status == VENDOR_STATUS_DSPIO_SCP_RESPONSE_QUEUE_EMPTY)) + val = le32_to_cpu(status); + if ((val == VENDOR_STATUS_DSPIO_OK) || + (val == VENDOR_STATUS_DSPIO_SCP_RESPONSE_QUEUE_EMPTY)) break; msleep(1); } while (time_before(jiffies, timeout)); @@ -736,6 +744,8 @@ static int dspio_write(struct hda_codec *codec, unsigned int scp_data) { struct ca0132_spec *spec = codec->spec; int status; + /* Handling for Big Endian machine */ + unsigned int val = 0; dspio_write_wait(codec); @@ -751,8 +761,10 @@ static int dspio_write(struct hda_codec *codec, unsigned int scp_data) goto error; /* OK, now check if the write itself has executed*/ - status = snd_hda_codec_read(codec, WIDGET_DSP_CTRL, 0, + val = snd_hda_codec_read(codec, WIDGET_DSP_CTRL, 0, VENDOR_DSPIO_STATUS, 0); + status = le32_to_cpu(val); + error: mutex_unlock(&spec->chipio_mutex); -- 1.7.4.1

2 1

[alsa-devel] ASoC updates for v3.8
by Mark Brown 29 Nov '12

29 Nov '12

The following changes since commit 9489e9dcae718d5fde988e4a684a0f55b5f94d17: Linux 3.7-rc7 (2012-11-25 17:59:19 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next for you to fetch changes up to 536bc8f605a8ea62d1f75d5ec3ed8bc0526a6271: Merge branches 'fix/fsl', 'topic/ab8500', 'topic/adsp', 'topic/ak4104', 'topic/ak4535', 'topic/ak4642', 'topic/atmel', 'topic/cs4271', 'topic/da7210', 'topic/da9055', 'topic/davinci', 'topic/dmaengine', 'topic/fsi', 'topic/fsl', 'topic/hotplug', 'topic/jack', 'topic/jz4740', 'topic/kirkwood', 'topic/lm49453', 'topic/log', 'topic/max98090', 'topic/rt5631', 'topic/samsung', 'topic/si476x', 'topic/tlv320aic32x4', 'topic/ux500', 'topic/wm0010', 'topic/wm2000', 'topic/wm2200', 'topic/wm8350', 'topic/wm8400', 'topic/wm8510', 'topic/wm8741', 'topic/wm8750', 'topic/wm8753', 'topic/wm8770', 'topic/wm8804', 'topic/wm8962', 'topic/wm8971', 'topic/wm8978', 'topic/wm8985', 'topic/wm8988', 'topic/wm8993', 'topic/wm8994' and 'topic/wm8995' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-next (2012-11-29 15:48:59 +0000) ---------------------------------------------------------------- ASoC: Updates for v3.8 Very quiet release for ASoC really: - Standardisation of the logging. - DT support for Atmel. - Support for Wolfson ADSP cores. - New drivers for Freescale/iVeia P1022 and Maxim MAX98090. ---------------------------------------------------------------- Will probably have another update with some bug fixes and trivial cleanups tomorrow. Andrey Smirnov (1): Add a codec driver for SI476X MFD Ashish Chavan (2): ASoC: da9055: Add support for DSP mode of AIF ASoC: da9055: DAPMise Headphone and Lineout enable controls Bill Pemberton (2): ASoC: remove use of __devinitconst ASoC: max98088: remove use of __devexit_p Bo Shen (9): ASoC: atmel-ssc: use devm_xxx() managed function ASoC: atmel-ssc: use module_platform_driver macro ASoC: sam9g20-wm8731: convert to use snd_soc_register_card() ASoC: sam9g20: using platform device for audio part ARM: at91: atmel-ssc: add platform device id table ARM: at91: atmel-ssc: add device tree support ASoC: atmel-ssc-dai: register dai and pcm directly ASoC: atmel-ssc-dai: match new method of dai and pcm register ASoC: at91sam9g20ek-wm8731: convert to dt support Charles Keepax (4): ASoC: wm5102: Correct micbias DAPM supplies ASoC: arizona: Add support for multiplexer with no associated mixer ASoC: wm5102: Remove mixer widgets on the ASRC ASoC: wm5110: Remove mixer widgets on the ASRC Daniel Mack (10): ASoC: ak4104: handle more sample rates ASoC: ak4104: add DT bindings ASoC: cs4271: add support for AMUTEB=BMUTEC feature ASoC: enable 192KHz support for McASP ASoC: McASP: add support for clock dividers ASoC: McASP: make AHCLK direction configurable ASoC: McASP: set format parameters in dependence of the DAI format ASoC: McASP: add support for 24 bit samples ASoC: McASP: treat SNDRV_PCM_FORMAT_U24_* as 32 bit words ASoC: ak4104: add reset line property to DT bindings Dimitris Papastamos (3): ASoC: bells: Up to 512fs ASoC: wm5102: Register DSP1 Aux widgets ASoC: wm0010: Up the completion timeout to 20ms instead of 10ms Eric Millbrandt (2): ASoC: fsl/pcm030 add unique dai_link names ASoC: wm8978: Remove remuxing of GPIO1 pin Javier Martin (2): ASoC: tlv320aic32x4: Fix problem with first capture. ASoC: tlv320aic32x4: Add rstn gpio to platform data. Kees Cook (1): ASoC: fsl: remove CONFIG_EXPERIMENTAL Kuninori Morimoto (13): ASoC: fsi: fixup channels_min/max ASoC: fsi: use devm_request_irq() ASoC: fsi: fsi_set_master_clk() was called from fsi_hw_xxx() only ASoC: fsi: care fsi_hw_start/stop() return value ASoC: fsi: add master clock control functions ASoC: fsi: tidyup FSIA/B settings ASoC: fsi: tidyup sh_fsi_platform_info pointer ASoC: fsi: SPDIF format become independent from platform flags ASoC: fsi: master clock selection become independent from platform flags ASoC: fsi: stream mode become independent from platform flags ASoC: fsi: add SND_SOC_DAIFMT_INV_xxx support ASoC: ak4642: cleanup struct ak4642_priv ASoC: Add max98090 codec driver Lars-Peter Clausen (3): ASoC: jz4740-codec: Add dB information ASoC: jz4740-codec: Use devm_request_and_ioremap ASoC: jz4740-codec: Use regmap Lee Jones (1): ASoC: Ux500: Initialise PCM from MSP probe rather than as a device Liam Girdwood (5): ASoC: dapm: Standardise ASoC DAPM messages ASoC: cache: Standardise ASoC cache messages ASoC: Jack: Standardise ASoC Jack messages ASoC: pcm: Standardise ASoC PCM messages ASoC: core: Standardise ASoC messages Lothar Waßmann (6): ASoC: imx: add MODULE_LICENSE("GPL") ASoC: imx-sgtl5000: prevent module from being unloaded when in use ASoC: dmaengine_pcm: set MODULE_LICENSE ASoC: fsl: fix miscompilation of snd-soc-imx-pcm ASoC: mxs-saif: export mxs_saif_put_mclk,mxs_saif_get_mclk ASoC: fsl: fix miscompilation of snd-soc-imx-pcm Mark Brown (67): ASoC: wm2200: Convert to devm APIs ASoC: wm2200: Implement AEC loopback support ASoC: wm2200: Map DSPs into regmap ASoC: wm2200: Fully plumb the DSPs into the routing map ASoC: wm2200: Initial DSP support ASoC: wm2200: Provide initial coefficient loading regmap: Rename n_ranges to num_ranges regmap: When we sanity check during range adds say what errors we find regmap: Allow ranges to be named regmap: Factor out debugfs register read regmap: Provide debugfs read of register ranges regmap: Factor range lookup out of page selection regmap: Make return code checks consistent regmap: Split raw writes that cross window boundaries ASoC: bells: Add WM0010 support ASoC: bells: Correct typo in sub speaker DAI name for WM5110 ASoC: bells: Provide additional parameterisation ASoC: bells: Add WM2200 support ASoC: wm8750: Convert to direct regmap API usage ASoC: wm8770: Convert to direct regmap API usage ASoC: wm8770: Remove unneeded bias level manipulation ASoC: wm8770: Remove regulator allocation to SPI probe ASoC: wm8770: Conver to table based DAPM and control init ASoC: wm8971: Convert to direct regmap API usage ASoC: wm8994: Allow a delay between jack insertion and microphone detect ASoC: wm8994: Tune VMID power up sequence ASoC: wm8994: Tune VMID power down sequence ASoC: wm8958: Load firmwares simultaneously Merge tag 'regmap/range' of git://git.kernel.org/.../broonie/regmap into asoc-wm2200 ASoC: wm2200: Add names for ranges ASoC: wm8994: Lower AIFnCLK divisor when dropping to 32kHz ASoC: adsp: Add ADSP base support Merge branch 'topic/adsp' of git://git.kernel.org/.../broonie/sound into asoc-arizona ASoC: arizona: Define standard hookup for ADSP2 ASoC: wm5102: Hook up DSP1 Merge branch 'topic/adsp' of git://git.kernel.org/.../broonie/sound into asoc-wm2200 ASoC: wm2200: Convert over to wm_adsp for ADSP1 support Merge tag 'v3.7-rc3' into HEAD ASoC: ab8500: Staticise non-exported ab85000_codec_dai Merge tag 'v3.7-rc6' into asoc-arizona Merge tag 'v3.7-rc6' into asoc-wm2200 ASoC: wm5102: Remove output OSR and PGA volume control ASoC: wm8994: Lower AIFnCLK divisor when dropping to 32kHz mfd: wm8994: Store platform data in device ASoC: wm8994: Use pdata cached in MFD driver ASoC: arizona: Support higher clock rates ASoC: wm2000: Get the CODEC pointer from the widget ASoC: wm5102: Add missing routes for ASRC inputs ASoC: arizona: Suppress noop FLL updates ASoC: wm0010: Ignore interrupts in power down mode ASoC: pcm: Report error code when we fail to init platform Merge tag 'v3.7-rc7' into asoc-ux500 Merge tag 'v3.7-rc7' into asoc-wm2200 ASoC: arizona: Record FLL setting when disabling mfd: arizona: Allow the CODEC DAPM context to be accessed elsewhere Input - arizona-haptics: Add driver haptics module on Arizona CODECs ASoC: wm5102: Implement haptics support ASoC: wm5110: Implement haptics support mfd: arizona: Register haptics devices ASoC: wm8994: Use the same DCS codes for all WM1811 variants ASoC: wm_adsp: Enable DVFS for ADSP2 Merge branch 'topic/adsp' of git://git.kernel.org/.../broonie/sound into asoc-arizona ASoC: wm5102: Enable DVFS support for the DSP ASoC: wm_adsp: Set the core enable as well as start bit for ADSP2 Merge branches 'topic/omap' and 'topic/arizona' of git://git.kernel.org/.../broonie/sound into asoc-next ASoC: arizona: Add parentheses Merge branches 'fix/fsl', 'topic/ab8500', 'topic/adsp', 'topic/ak4104', 'topic/ak4535', 'topic/ak4642', 'topic/atmel', 'topic/cs4271', 'topic/da7210', 'topic/da9055', 'topic/davinci', 'topic/dmaengine', 'topic/fsi', 'topic/fsl', 'topic/hotplug', 'topic/jack', 'topic/jz4740', 'topic/kirkwood', 'topic/lm49453', 'topic/log', 'topic/max98090', 'topic/rt5631', 'topic/samsung', 'topic/si476x', 'topic/tlv320aic32x4', 'topic/ux500', 'topic/wm0010', 'topic/wm2000', 'topic/wm2200', 'topic/wm8350', 'topic/wm8400', 'topic/wm8510', 'topic/wm8741', 'topic/wm8750', 'topic/wm8753', 'topic/wm8770', 'topic/wm8804', 'topic/wm8962', 'topic/wm8971', 'topic/wm8978', 'topic/wm8985', 'topic/wm8988', 'topic/wm8993', 'topic/wm8994' and 'topic/wm8995' of git://git.kernel.org/.../broonie/sound into asoc-next Matt Porter (2): ASoC: davinci: replace private sram api with genalloc ARM: davinci: enable SRAM ping ping buffering on DA850 Nicolas Ferre (1): ASoC: atmel-ssc: add phybase in device structure Padmavathi Venna (1): ASoC: Samsung: Get I2S src_clk from clock alias id. Peter Ujfalusi (1): ASoC: omap-abe-twl6040: Use "ti,jack-detection" DT property as boolean Randy Dunlap (1): ASoC: fix wm2200.c printk format warnings Russell King (5): ASoC: kirkwood-i2s: use devm_* APIs ASoC: kirkwood-i2s: better handling of play/record control registers ASoC: kirkwood-dma: remove restriction on sample rates ASoC: kirkwood-i2s: add support for external clock rates ASoC: kirkwood-dma: remove channel restrictions Sachin Kamat (9): ASoC: rt5631: Fix return value ASoC: wm8350: Use devm_regulator_bulk_get ASoC: wm8962: Use devm_regulator_bulk_get ASoC: wm8400: Use devm_regulator_bulk_get ASoC: wm8993: Use devm_regulator_bulk_get ASoC: ak4535: Use devm_regmap_init_i2c() ASoC: da7210: Use devm_regmap_init_i2c() ASoC: lm49453: Use devm_regmap_init_i2c() ASoC: wm2200: Remove empty labels Scott Ling (4): ASoC: wm0010: Remove boot_done variable as no longer required. ASoC: wm0010: Split out the firmware file parsing from the boot ASoC: wm0010: Split out the stage2 load from the boot function ASoC: wm0010: Add checking for .dfw info record version. Thomas Abraham (4): ASoC: SAMSUNG: ac97: use clk_prepare_enable and clk_disable_unprepare ASoC: SAMSUNG: i2s: use clk_prepare_enable and clk_disable_unprepare ASoC: SAMSUNG: pcm: use clk_prepare_enable and clk_disable_unprepare ASoC: SAMSUNG: spdif: use clk_prepare_enable and clk_disable_unprepare Timur Tabi (1): ASoC: add support for the Freescale / iVeia P1022 RDK reference board Tony Lindgren (1): ASoC: OMAP: mcbsp fixes for enabling ARM multiplatform support Tushar Behera (10): ASoC: wm8510: remove invalid free of devm_ allocated data ASoC: wm8753: remove invalid free of devm_ allocated data ASoC: wm8995: Convert to use devm_kzalloc() ASoC: wm8741: Convert to use devm_regmap_init_{spi,i2c}() ASoC: wm8753: Convert to use devm_regmap_init_{spi,i2c}() ASoC: wm8995: Convert to use devm_regmap_init_{spi,i2c}() ASoC: wm8804: Convert to use devm_regmap_init_{spi,i2c}() ASoC: wm8985: Convert to use devm_regmap_init_{spi,i2c}() ASoC: wm8988: Convert to use devm_regmap_init_{spi,i2c}() ASoC: ak4104: Convert to use devm_regmap_init_spi() Ulf Hansson (2): ASoC: Ux500: Fixup use of clocks ASoC: Ux500: Control apb clock Wei Yongjun (1): ASoC: jack: remove unused variable in snd_soc_jack_report() .../devicetree/bindings/misc/atmel-ssc.txt | 15 + Documentation/devicetree/bindings/sound/ak4104.txt | 22 + .../sound/atmel-at91sam9g20ek-wm8731-audio.txt | 26 + Documentation/devicetree/bindings/sound/cs4271.txt | 2 + .../devicetree/bindings/sound/omap-abe-twl6040.txt | 4 +- arch/arm/boot/dts/at91sam9260.dtsi | 8 + arch/arm/boot/dts/at91sam9263.dtsi | 16 + arch/arm/boot/dts/at91sam9g20ek_common.dtsi | 32 +- arch/arm/boot/dts/at91sam9g45.dtsi | 16 + arch/arm/boot/dts/at91sam9x5.dtsi | 8 + arch/arm/mach-at91/at91rm9200.c | 9 +- arch/arm/mach-at91/at91rm9200_devices.c | 6 +- arch/arm/mach-at91/at91sam9260.c | 3 +- arch/arm/mach-at91/at91sam9260_devices.c | 2 +- arch/arm/mach-at91/at91sam9261.c | 9 +- arch/arm/mach-at91/at91sam9261_devices.c | 6 +- arch/arm/mach-at91/at91sam9263.c | 6 +- arch/arm/mach-at91/at91sam9263_devices.c | 4 +- arch/arm/mach-at91/at91sam9g45.c | 6 +- arch/arm/mach-at91/at91sam9g45_devices.c | 4 +- arch/arm/mach-at91/at91sam9rl.c | 6 +- arch/arm/mach-at91/at91sam9rl_devices.c | 4 +- arch/arm/mach-at91/at91sam9x5.c | 1 + arch/arm/mach-at91/board-sam9g20ek.c | 11 + arch/arm/mach-davinci/board-da850-evm.c | 24 +- drivers/base/regmap/internal.h | 2 + drivers/base/regmap/regmap-debugfs.c | 50 +- drivers/base/regmap/regmap.c | 154 +++-- drivers/input/misc/Kconfig | 10 + drivers/input/misc/Makefile | 1 + drivers/input/misc/arizona-haptics.c | 255 ++++++++ drivers/mfd/arizona-core.c | 2 + drivers/mfd/wm8994-core.c | 35 +- drivers/misc/atmel-ssc.c | 135 ++-- include/linux/atmel-ssc.h | 6 + include/linux/mfd/arizona/core.h | 4 + include/linux/mfd/arizona/pdata.h | 6 + include/linux/mfd/wm8994/core.h | 4 + include/linux/mfd/wm8994/pdata.h | 5 + include/linux/platform_data/asoc-s3c.h | 6 - include/linux/platform_data/davinci_asp.h | 3 + include/linux/regmap.h | 6 +- include/sound/cs4271.h | 1 + include/sound/sh_fsi.h | 6 + include/sound/tlv320aic32x4.h | 1 + sound/soc/atmel/Kconfig | 3 +- sound/soc/atmel/atmel-pcm.c | 23 +- sound/soc/atmel/atmel-pcm.h | 3 + sound/soc/atmel/atmel_ssc_dai.c | 157 ++--- sound/soc/atmel/atmel_ssc_dai.h | 3 +- sound/soc/atmel/sam9g20_wm8731.c | 116 +++- sound/soc/codecs/Kconfig | 16 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/ab8500-codec.c | 2 +- sound/soc/codecs/ak4104.c | 65 +- sound/soc/codecs/ak4535.c | 7 +- sound/soc/codecs/ak4642.c | 23 +- sound/soc/codecs/arizona.c | 21 + sound/soc/codecs/arizona.h | 65 +- sound/soc/codecs/cs4271.c | 21 +- sound/soc/codecs/da7210.c | 13 +- sound/soc/codecs/da9055.c | 43 +- sound/soc/codecs/jz4740.c | 142 ++--- sound/soc/codecs/lm49453.c | 10 +- sound/soc/codecs/max98088.c | 14 +- sound/soc/codecs/max98090.c | 577 +++++++++++++++++ sound/soc/codecs/rt5631.c | 2 +- sound/soc/codecs/si476x.c | 255 ++++++++ sound/soc/codecs/tlv320aic32x4.c | 24 + sound/soc/codecs/tlv320aic32x4.h | 3 + sound/soc/codecs/wm0010.c | 419 +++++++----- sound/soc/codecs/wm2000.c | 2 +- sound/soc/codecs/wm2200.c | 269 ++++++-- sound/soc/codecs/wm5100.c | 2 +- sound/soc/codecs/wm5102.c | 120 ++-- sound/soc/codecs/wm5110.c | 49 +- sound/soc/codecs/wm8350.c | 4 +- sound/soc/codecs/wm8400.c | 14 +- sound/soc/codecs/wm8510.c | 3 - sound/soc/codecs/wm8741.c | 4 +- sound/soc/codecs/wm8750.c | 86 ++- sound/soc/codecs/wm8753.c | 33 +- sound/soc/codecs/wm8770.c | 217 ++++--- sound/soc/codecs/wm8804.c | 17 +- sound/soc/codecs/wm8958-dsp2.c | 79 +-- sound/soc/codecs/wm8962.c | 7 +- sound/soc/codecs/wm8971.c | 80 ++- sound/soc/codecs/wm8978.c | 3 - sound/soc/codecs/wm8985.c | 30 +- sound/soc/codecs/wm8988.c | 14 +- sound/soc/codecs/wm8993.c | 7 +- sound/soc/codecs/wm8994.c | 184 +++--- sound/soc/codecs/wm8994.h | 1 - sound/soc/codecs/wm8995.c | 40 +- sound/soc/codecs/wm_adsp.c | 666 ++++++++++++++++++++ sound/soc/codecs/wm_adsp.h | 59 ++ sound/soc/codecs/wmfw.h | 128 ++++ sound/soc/davinci/davinci-evm.c | 5 + sound/soc/davinci/davinci-mcasp.c | 93 ++- sound/soc/davinci/davinci-mcasp.h | 2 +- sound/soc/davinci/davinci-pcm.c | 53 +- sound/soc/davinci/davinci-pcm.h | 2 + sound/soc/fsl/Kconfig | 20 +- sound/soc/fsl/Makefile | 14 +- sound/soc/fsl/imx-pcm.c | 4 + sound/soc/fsl/imx-sgtl5000.c | 1 + sound/soc/fsl/p1022_rdk.c | 392 ++++++++++++ sound/soc/fsl/pcm030-audio-fabric.c | 4 +- sound/soc/kirkwood/kirkwood-dma.c | 16 +- sound/soc/kirkwood/kirkwood-i2s.c | 235 ++++--- sound/soc/kirkwood/kirkwood.h | 11 +- sound/soc/mxs/mxs-saif.c | 2 + sound/soc/omap/mcbsp.c | 4 +- sound/soc/omap/mcbsp.h | 6 + sound/soc/omap/omap-abe-twl6040.c | 4 +- sound/soc/omap/omap-mcbsp.c | 5 +- sound/soc/samsung/ac97.c | 6 +- sound/soc/samsung/bells.c | 222 +++++-- sound/soc/samsung/i2s.c | 20 +- sound/soc/samsung/pcm.c | 12 +- sound/soc/samsung/spdif.c | 12 +- sound/soc/sh/fsi.c | 550 ++++++++++++++-- sound/soc/soc-cache.c | 10 +- sound/soc/soc-core.c | 243 +++---- sound/soc/soc-dapm.c | 134 ++-- sound/soc/soc-dmaengine-pcm.c | 2 + sound/soc/soc-jack.c | 16 +- sound/soc/soc-pcm.c | 195 +++--- sound/soc/tegra/tegra20_das.c | 2 +- sound/soc/tegra/tegra20_i2s.c | 4 +- sound/soc/tegra/tegra20_spdif.c | 2 +- sound/soc/tegra/tegra30_ahub.c | 6 +- sound/soc/tegra/tegra30_i2s.c | 4 +- sound/soc/tegra/tegra_alc5632.c | 2 +- sound/soc/tegra/tegra_wm8753.c | 2 +- sound/soc/tegra/tegra_wm8903.c | 2 +- sound/soc/tegra/trimslice.c | 2 +- sound/soc/ux500/mop500.c | 4 +- sound/soc/ux500/ux500_msp_dai.c | 55 +- sound/soc/ux500/ux500_msp_dai.h | 1 + sound/soc/ux500/ux500_pcm.c | 19 +- sound/soc/ux500/ux500_pcm.h | 3 + 142 files changed, 5700 insertions(+), 1796 deletions(-) create mode 100644 Documentation/devicetree/bindings/misc/atmel-ssc.txt create mode 100644 Documentation/devicetree/bindings/sound/ak4104.txt create mode 100644 Documentation/devicetree/bindings/sound/atmel-at91sam9g20ek-wm8731-audio.txt create mode 100644 drivers/input/misc/arizona-haptics.c create mode 100644 sound/soc/codecs/max98090.c create mode 100644 sound/soc/codecs/si476x.c create mode 100644 sound/soc/codecs/wm_adsp.c create mode 100644 sound/soc/codecs/wm_adsp.h create mode 100644 sound/soc/codecs/wmfw.h create mode 100644 sound/soc/fsl/p1022_rdk.c

2 6

[alsa-devel] [PATCH 1/4] ASoC: firmware: Add support for FW based kcontrols.
by Liam Girdwood 29 Nov '12

29 Nov '12

This patch adds initial support for firmware based kcontrols by allowing soc.h to be included by any userspace firmware generation tools and assigns IDs to the standard ASoC kcontrol types using the kcontrol_new index and IDs to kcontrol get/put/info functions. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc.h | 135 +++++++++++++++++++++++++++++++++++++++++++-------- 1 file changed, 116 insertions(+), 19 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 91244a0..b0b1703 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -13,6 +13,8 @@ #ifndef __LINUX_SND_SOC_H #define __LINUX_SND_SOC_H +#ifdef __KERNEL__ + #include <linux/platform_device.h> #include <linux/types.h> #include <linux/notifier.h> @@ -27,6 +29,8 @@ #include <sound/control.h> #include <sound/ac97_codec.h> +#endif + /* * Convenience kcontrol builders */ @@ -51,12 +55,12 @@ #define SOC_SINGLE(xname, reg, shift, max, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\ - .put = snd_soc_put_volsw, \ + .put = snd_soc_put_volsw, .index = SOC_CONTROL_IO_VOLSW, \ .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } #define SOC_SINGLE_RANGE(xname, xreg, xshift, xmin, xmax, xinvert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .info = snd_soc_info_volsw_range, .get = snd_soc_get_volsw_range, \ - .put = snd_soc_put_volsw_range, \ + .put = snd_soc_put_volsw_range, .index = SOC_CONTROL_IO_RANGE, \ .private_value = (unsigned long)&(struct soc_mixer_control) \ {.reg = xreg, .shift = xshift, .min = xmin,\ .max = xmax, .platform_max = xmax, .invert = xinvert} } @@ -66,7 +70,7 @@ SNDRV_CTL_ELEM_ACCESS_READWRITE,\ .tlv.p = (tlv_array), \ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\ - .put = snd_soc_put_volsw, \ + .put = snd_soc_put_volsw, .index = SOC_CONTROL_IO_VOLSW, \ .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } #define SOC_SINGLE_SX_TLV(xname, xreg, xshift, xmin, xmax, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ @@ -76,6 +80,7 @@ .info = snd_soc_info_volsw, \ .get = snd_soc_get_volsw_sx,\ .put = snd_soc_put_volsw_sx, \ + .index = SOC_CONTROL_IO_VOLSW_SX \ .private_value = (unsigned long)&(struct soc_mixer_control) \ {.reg = xreg, .rreg = xreg, \ .shift = xshift, .rshift = xshift, \ @@ -85,7 +90,7 @@ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ SNDRV_CTL_ELEM_ACCESS_READWRITE,\ .tlv.p = (tlv_array), \ - .info = snd_soc_info_volsw_range, \ + .info = snd_soc_info_volsw_range, .index = SOC_CONTROL_IO_RANGE, \ .get = snd_soc_get_volsw_range, .put = snd_soc_put_volsw_range, \ .private_value = (unsigned long)&(struct soc_mixer_control) \ {.reg = xreg, .shift = xshift, .min = xmin,\ @@ -93,19 +98,19 @@ #define SOC_DOUBLE(xname, reg, shift_left, shift_right, max, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \ - .put = snd_soc_put_volsw, \ + .put = snd_soc_put_volsw, .index = SOC_CONTROL_IO_VOLSW, \ .private_value = SOC_DOUBLE_VALUE(reg, shift_left, shift_right, \ max, invert) } #define SOC_DOUBLE_R(xname, reg_left, reg_right, xshift, xmax, xinvert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ - .info = snd_soc_info_volsw, \ + .info = snd_soc_info_volsw, .index = SOC_CONTROL_IO_VOLSW, \ .get = snd_soc_get_volsw, .put = snd_soc_put_volsw, \ .private_value = SOC_DOUBLE_R_VALUE(reg_left, reg_right, xshift, \ xmax, xinvert) } #define SOC_DOUBLE_R_RANGE(xname, reg_left, reg_right, xshift, xmin, \ xmax, xinvert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ - .info = snd_soc_info_volsw_range, \ + .info = snd_soc_info_volsw_range, .index = SOC_CONTROL_IO_RANGE, \ .get = snd_soc_get_volsw_range, .put = snd_soc_put_volsw_range, \ .private_value = SOC_DOUBLE_R_RANGE_VALUE(reg_left, reg_right, \ xshift, xmin, xmax, xinvert) } @@ -115,7 +120,7 @@ SNDRV_CTL_ELEM_ACCESS_READWRITE,\ .tlv.p = (tlv_array), \ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \ - .put = snd_soc_put_volsw, \ + .put = snd_soc_put_volsw, .index = SOC_CONTROL_IO_VOLSW, \ .private_value = SOC_DOUBLE_VALUE(reg, shift_left, shift_right, \ max, invert) } #define SOC_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax, xinvert, tlv_array) \ @@ -123,7 +128,7 @@ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ SNDRV_CTL_ELEM_ACCESS_READWRITE,\ .tlv.p = (tlv_array), \ - .info = snd_soc_info_volsw, \ + .info = snd_soc_info_volsw, .index = SOC_CONTROL_IO_VOLSW, \ .get = snd_soc_get_volsw, .put = snd_soc_put_volsw, \ .private_value = SOC_DOUBLE_R_VALUE(reg_left, reg_right, xshift, \ xmax, xinvert) } @@ -133,7 +138,7 @@ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ SNDRV_CTL_ELEM_ACCESS_READWRITE,\ .tlv.p = (tlv_array), \ - .info = snd_soc_info_volsw_range, \ + .info = snd_soc_info_volsw_range, .index = SOC_CONTROL_IO_RANGE, \ .get = snd_soc_get_volsw_range, .put = snd_soc_put_volsw_range, \ .private_value = SOC_DOUBLE_R_RANGE_VALUE(reg_left, reg_right, \ xshift, xmin, xmax, xinvert) } @@ -145,6 +150,7 @@ .info = snd_soc_info_volsw, \ .get = snd_soc_get_volsw_sx, \ .put = snd_soc_put_volsw_sx, \ + .index = SOC_CONTROL_IO_SX, \ .private_value = (unsigned long)&(struct soc_mixer_control) \ {.reg = xreg, .rreg = xrreg, \ .shift = xshift, .rshift = xshift, \ @@ -155,7 +161,7 @@ SNDRV_CTL_ELEM_ACCESS_READWRITE, \ .tlv.p = (tlv_array), \ .info = snd_soc_info_volsw_s8, .get = snd_soc_get_volsw_s8, \ - .put = snd_soc_put_volsw_s8, \ + .put = snd_soc_put_volsw_s8, .index = SOC_CONTROL_IO_VOLSW_S8, \ .private_value = (unsigned long)&(struct soc_mixer_control) \ {.reg = xreg, .min = xmin, .max = xmax, \ .platform_max = xmax} } @@ -174,7 +180,7 @@ SOC_VALUE_ENUM_DOUBLE(xreg, xshift, xshift, xmask, xmax, xtexts, xvalues) #define SOC_ENUM(xname, xenum) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname,\ - .info = snd_soc_info_enum_double, \ + .info = snd_soc_info_enum_double, .index = SOC_CONTROL_IO_ENUM, \ .get = snd_soc_get_enum_double, .put = snd_soc_put_enum_double, \ .private_value = (unsigned long)&xenum } #define SOC_VALUE_ENUM(xname, xenum) \ @@ -182,17 +188,18 @@ .info = snd_soc_info_enum_double, \ .get = snd_soc_get_value_enum_double, \ .put = snd_soc_put_value_enum_double, \ + .index = SOC_CONTROL_IO_ENUM_VALUE, \ .private_value = (unsigned long)&xenum } #define SOC_SINGLE_EXT(xname, xreg, xshift, xmax, xinvert,\ xhandler_get, xhandler_put) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ - .info = snd_soc_info_volsw, \ + .info = snd_soc_info_volsw, .index = SOC_CONTROL_IO_EXT, \ .get = xhandler_get, .put = xhandler_put, \ .private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) } #define SOC_DOUBLE_EXT(xname, reg, shift_left, shift_right, max, invert,\ xhandler_get, xhandler_put) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ - .info = snd_soc_info_volsw, \ + .info = snd_soc_info_volsw, .index = SOC_CONTROL_IO_EXT, \ .get = xhandler_get, .put = xhandler_put, \ .private_value = \ SOC_DOUBLE_VALUE(reg, shift_left, shift_right, max, invert) } @@ -202,7 +209,7 @@ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ SNDRV_CTL_ELEM_ACCESS_READWRITE,\ .tlv.p = (tlv_array), \ - .info = snd_soc_info_volsw, \ + .info = snd_soc_info_volsw, .index = SOC_CONTROL_IO_EXT, \ .get = xhandler_get, .put = xhandler_put, \ .private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) } #define SOC_DOUBLE_EXT_TLV(xname, xreg, shift_left, shift_right, xmax, xinvert,\ @@ -211,7 +218,7 @@ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ SNDRV_CTL_ELEM_ACCESS_READWRITE, \ .tlv.p = (tlv_array), \ - .info = snd_soc_info_volsw, \ + .info = snd_soc_info_volsw, .index = SOC_CONTROL_IO_EXT,\ .get = xhandler_get, .put = xhandler_put, \ .private_value = SOC_DOUBLE_VALUE(xreg, shift_left, shift_right, \ xmax, xinvert) } @@ -221,24 +228,25 @@ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ SNDRV_CTL_ELEM_ACCESS_READWRITE, \ .tlv.p = (tlv_array), \ - .info = snd_soc_info_volsw, \ + .info = snd_soc_info_volsw, .index = SOC_CONTROL_IO_EXT, \ .get = xhandler_get, .put = xhandler_put, \ .private_value = SOC_DOUBLE_R_VALUE(reg_left, reg_right, xshift, \ xmax, xinvert) } #define SOC_SINGLE_BOOL_EXT(xname, xdata, xhandler_get, xhandler_put) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ - .info = snd_soc_info_bool_ext, \ + .info = snd_soc_info_bool_ext, .index = SOC_CONTROL_IO_BOOL_EXT, \ .get = xhandler_get, .put = xhandler_put, \ .private_value = xdata } #define SOC_ENUM_EXT(xname, xenum, xhandler_get, xhandler_put) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ - .info = snd_soc_info_enum_ext, \ + .info = snd_soc_info_enum_ext, .index = SOC_CONTROL_IO_ENUM_EXT, \ .get = xhandler_get, .put = xhandler_put, \ .private_value = (unsigned long)&xenum } #define SND_SOC_BYTES(xname, xbase, xregs) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_bytes_info, .get = snd_soc_bytes_get, \ + .index = SOC_CONTROL_IO_BYTES, \ .put = snd_soc_bytes_put, .private_value = \ ((unsigned long)&(struct soc_bytes) \ {.base = xbase, .num_regs = xregs }) } @@ -247,6 +255,7 @@ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_bytes_info, .get = snd_soc_bytes_get, \ .put = snd_soc_bytes_put, .private_value = \ + .index = SOC_CONTROL_IO_BYTES, \ ((unsigned long)&(struct soc_bytes) \ {.base = xbase, .num_regs = xregs, \ .mask = xmask }) } @@ -256,6 +265,7 @@ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ .info = snd_soc_info_xr_sx, .get = snd_soc_get_xr_sx, \ .put = snd_soc_put_xr_sx, \ + .index = SOC_CONTROL_IO_XR_SX, \ .private_value = (unsigned long)&(struct soc_mreg_control) \ {.regbase = xregbase, .regcount = xregcount, .nbits = xnbits, \ .invert = xinvert, .min = xmin, .max = xmax} } @@ -281,6 +291,92 @@ #define SOC_VALUE_ENUM_SINGLE_DECL(name, xreg, xshift, xmask, xtexts, xvalues) \ SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xmask, xtexts, xvalues) + +/* + * Numeric IDs for stock mixer types that are used to enumerate FW based mixers. + */ + +#define SOC_CONTROL_ID_PUT(p) ((p & 0xff) << 16) +#define SOC_CONTROL_ID_GET(g) ((g & 0xff) << 8) +#define SOC_CONTROL_ID_INFO(i) ((i & 0xff) << 0) +#define SOC_CONTROL_ID(g, p, i) \ + (SOC_CONTROL_ID_PUT(p) | SOC_CONTROL_ID_GET(g) |\ + SOC_CONTROL_ID_INFO(i)) + +#define SOC_CONTROL_GET_ID_PUT(id) ((id & 0xff0000) >> 16) +#define SOC_CONTROL_GET_ID_GET(id) ((id & 0x00ff00) >> 8) +#define SOC_CONTROL_GET_ID_INFO(id) ((id & 0x0000ff) >> 0) + +/* individual kcontrol info types - can be mixed with other types */ +#define SOC_CONTROL_TYPE_EXT 0 /* driver defined */ +#define SOC_CONTROL_TYPE_VOLSW 1 +#define SOC_CONTROL_TYPE_VOLSW_SX 2 +#define SOC_CONTROL_TYPE_VOLSW_S8 3 +#define SOC_CONTROL_TYPE_VOLSW_XR_SX 4 +#define SOC_CONTROL_TYPE_ENUM 6 +#define SOC_CONTROL_TYPE_ENUM_EXT 7 +#define SOC_CONTROL_TYPE_BYTES 8 +#define SOC_CONTROL_TYPE_BOOL_EXT 9 +#define SOC_CONTROL_TYPE_ENUM_VALUE 10 +#define SOC_CONTROL_TYPE_RANGE 11 +#define SOC_CONTROL_TYPE_STROBE 12 + +/* compound control IDs */ +#define SOC_CONTROL_IO_VOLSW \ + SOC_CONTROL_ID(SOC_CONTROL_TYPE_VOLSW, \ + SOC_CONTROL_TYPE_VOLSW, \ + SOC_CONTROL_TYPE_VOLSW) +#define SOC_CONTROL_IO_VOLSW_SX \ + SOC_CONTROL_ID(SOC_CONTROL_TYPE_VOLSW_SX, \ + SOC_CONTROL_TYPE_VOLSW_SX, \ + SOC_CONTROL_TYPE_VOLSW) +#define SOC_CONTROL_IO_VOLSW_S8 \ + SOC_CONTROL_ID(SOC_CONTROL_TYPE_VOLSW_S8, \ + SOC_CONTROL_TYPE_VOLSW_S8, \ + SOC_CONTROL_TYPE_VOLSW_S8) +#define SOC_CONTROL_IO_VOLSW_XR_SX \ + SOC_CONTROL_ID(SOC_CONTROL_TYPE_VOLSW_XR_SX, \ + SOC_CONTROL_TYPE_VOLSW_XR_SX, \ + SOC_CONTROL_TYPE_VOLSW_XR_SX) +#define SOC_CONTROL_IO_EXT \ + SOC_CONTROL_ID(SOC_CONTROL_TYPE_EXT, \ + SOC_CONTROL_TYPE_EXT, \ + SOC_CONTROL_TYPE_VOLSW) +#define SOC_CONTROL_IO_ENUM \ + SOC_CONTROL_ID(SOC_CONTROL_TYPE_ENUM, \ + SOC_CONTROL_TYPE_ENUM, \ + SOC_CONTROL_TYPE_ENUM) +#define SOC_CONTROL_IO_ENUM_EXT \ + SOC_CONTROL_ID(SOC_CONTROL_TYPE_EXT, \ + SOC_CONTROL_TYPE_EXT, \ + SOC_CONTROL_TYPE_ENUM_EXT) +#define SOC_CONTROL_IO_BYTES \ + SOC_CONTROL_ID(SOC_CONTROL_TYPE_BYTES, \ + SOC_CONTROL_TYPE_BYTES, \ + SOC_CONTROL_TYPE_BYTES) +#define SOC_CONTROL_IO_BOOL_EXT \ + SOC_CONTROL_ID(SOC_CONTROL_TYPE_EXT, \ + SOC_CONTROL_TYPE_EXT, \ + SOC_CONTROL_TYPE_BOOL_EXT) +#define SOC_CONTROL_IO_ENUM_VALUE \ + SOC_CONTROL_ID(SOC_CONTROL_TYPE_ENUM_VALUE, \ + SOC_CONTROL_TYPE_ENUM_VALUE, \ + SOC_CONTROL_TYPE_ENUM) +#define SOC_CONTROL_IO_RANGE \ + SOC_CONTROL_ID(SOC_CONTROL_TYPE_RANGE, \ + SOC_CONTROL_TYPE_RANGE, \ + SOC_CONTROL_TYPE_RANGE) +#define SOC_CONTROL_IO_STROBE \ + SOC_CONTROL_ID(SOC_CONTROL_TYPE_STROBE, \ + SOC_CONTROL_TYPE_STROBE, \ + SOC_CONTROL_TYPE_STROBE) + +#ifdef __KERNEL__ + +#define snd_soc_get_enum_text(soc_enum, idx) \ + (soc_enum->texts ? soc_enum->texts[idx] : soc_enum->dtexts[idx]) + + /* * Component probe and remove ordering levels for components with runtime * dependencies. @@ -1178,3 +1274,4 @@ extern struct dentry *snd_soc_debugfs_root; extern const struct dev_pm_ops snd_soc_pm_ops; #endif +#endif -- 1.7.10.4

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[alsa-devel] [PATCH] ALSA: usb - Don't create "Speaker" mixer controls on headphones and headsets
by David Henningsson 29 Nov '12

29 Nov '12

A lot of headsets/headphones have a "Speaker" mixer control. This confuses PulseAudio to think it is a speaker instead of a headphone/headset. Therfore, we rename it to "Headphone". We determine if something is a headphone similar to how udev determines form factor (see 78-sound-card.rules). BugLink: https://bugs.launchpad.net/bugs/1082357 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/usb/mixer.c | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) I tried to find a better way of doing this using the existing quirk or map infrastructure but it didn't seem to fit, so added a new explicit check. diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 298070e..63fd758 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1094,6 +1094,32 @@ static size_t append_ctl_name(struct snd_kcontrol *kctl, const char *str) return strlcat(kctl->id.name, str, sizeof(kctl->id.name)); } +/* A lot of headsets/headphones have a "Speaker" mixer. Make sure we + rename it to "Headphone". We determine if something is a headphone + similar to how udev determines form factor. */ +static void check_no_speaker_on_headset(struct snd_kcontrol *kctl, + struct snd_card *card) +{ + const char *names_to_check[] = { + "Headset", "headset", "Headphone", "headphone", NULL}; + const char **s; + bool found = 0; + + if (strcmp("Speaker", kctl->id.name)) + return; + + for (s = names_to_check; *s; s++) + if (strstr(card->shortname, *s)) { + found = 1; + break; + } + + if (!found) + return; + + strlcpy(kctl->id.name, "Headphone", sizeof(kctl->id.name)); +} + static void build_feature_ctl(struct mixer_build *state, void *raw_desc, unsigned int ctl_mask, int control, struct usb_audio_term *iterm, int unitid, @@ -1180,6 +1206,10 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, len = snprintf(kctl->id.name, sizeof(kctl->id.name), "Feature %d", unitid); } + + if (!mapped_name) + check_no_speaker_on_headset(kctl, state->mixer->chip->card); + /* determine the stream direction: * if the connected output is USB stream, then it's likely a * capture stream. otherwise it should be playback (hopefully :) -- 1.7.9.5

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[alsa-devel] [asoc:topic/adsp 139/139] sound/soc/codecs/wm_adsp.c:556 wm_adsp2_event() warn: add some parenthesis here?
by Dan Carpenter 29 Nov '12

29 Nov '12

Hi Mark, FYI, there are new smatch warnings show up in tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/adsp head: 973838a030d2a91fbd524c95281991510ed0a865 commit: 973838a030d2a91fbd524c95281991510ed0a865 [139/139] ASoC: wm_adsp: Enable DVFS for ADSP2 + sound/soc/codecs/wm_adsp.c:556 wm_adsp2_event() warn: add some parenthesis here? git remote update asoc git checkout 973838a030d2a91fbd524c95281991510ed0a865 vim +556 sound/soc/codecs/wm_adsp.c 2159ad93 Mark Brown 2012-10-11 540 struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec); 2159ad93 Mark Brown 2012-10-11 541 struct wm_adsp *dsp = &dsps[w->shift]; 973838a0 Mark Brown 2012-11-28 542 unsigned int val; 2159ad93 Mark Brown 2012-10-11 543 int ret; 2159ad93 Mark Brown 2012-10-11 544 2159ad93 Mark Brown 2012-10-11 545 switch (event) { 2159ad93 Mark Brown 2012-10-11 546 case SND_SOC_DAPM_POST_PMU: 973838a0 Mark Brown 2012-11-28 547 if (dsp->dvfs) { 973838a0 Mark Brown 2012-11-28 548 ret = regmap_read(dsp->regmap, 973838a0 Mark Brown 2012-11-28 549 dsp->base + ADSP2_CLOCKING, &val); 973838a0 Mark Brown 2012-11-28 550 if (ret != 0) { 973838a0 Mark Brown 2012-11-28 551 dev_err(dsp->dev, 973838a0 Mark Brown 2012-11-28 552 "Failed to read clocking: %d\n", ret); 973838a0 Mark Brown 2012-11-28 553 return ret; 973838a0 Mark Brown 2012-11-28 554 } 973838a0 Mark Brown 2012-11-28 555 973838a0 Mark Brown 2012-11-28 @556 if (val & ADSP2_CLK_SEL_MASK >= 3) { Precedence bug. The intent was: if (val & ADSP2_CLK_SEL_MASK) >= 3) { 973838a0 Mark Brown 2012-11-28 557 ret = regulator_enable(dsp->dvfs); 973838a0 Mark Brown 2012-11-28 558 if (ret != 0) { 973838a0 Mark Brown 2012-11-28 559 dev_err(dsp->dev, 973838a0 Mark Brown 2012-11-28 560 "Failed to enable supply: %d\n", 973838a0 Mark Brown 2012-11-28 561 ret); 973838a0 Mark Brown 2012-11-28 562 return ret; 973838a0 Mark Brown 2012-11-28 563 } 973838a0 Mark Brown 2012-11-28 564 --- 0-DAY kernel build testing backend Open Source Technology Center Fengguang Wu, Yuanhan Liu Intel Corporation

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[alsa-devel] [asoc:topic/arizona 16/21] drivers/input/misc/arizona-haptics.c:164 arizona_haptics_close() warn: variable dereferenced before check 'haptics->arizona->dapm' (see line 158)
by Dan Carpenter 29 Nov '12

29 Nov '12

Hi Mark, FYI, there are new smatch warnings show up in tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/arizona head: cac27cff066cf86c6fefe33d99d935e9d0d9c78c commit: 9dd555e2f4de1af1bb0f75cc84ed0708fcdb5987 [16/21] Input - arizona-haptics: Add driver haptics module on Arizona CODECs + drivers/input/misc/arizona-haptics.c:164 arizona_haptics_close() warn: variable dereferenced before check 'haptics->arizona->dapm' (see line 158) git remote update asoc git checkout 9dd555e2f4de1af1bb0f75cc84ed0708fcdb5987 vim +164 drivers/input/misc/arizona-haptics.c 9dd555e2 Mark Brown 2012-11-26 152 return 0; 9dd555e2 Mark Brown 2012-11-26 153 } 9dd555e2 Mark Brown 2012-11-26 154 9dd555e2 Mark Brown 2012-11-26 155 static void arizona_haptics_close(struct input_dev *input) 9dd555e2 Mark Brown 2012-11-26 156 { 9dd555e2 Mark Brown 2012-11-26 157 struct arizona_haptics *haptics = input_get_drvdata(input); 9dd555e2 Mark Brown 2012-11-26 @158 struct mutex *dapm_mutex = &haptics->arizona->dapm->card->dapm_mutex; 9dd555e2 Mark Brown 2012-11-26 159 9dd555e2 Mark Brown 2012-11-26 160 cancel_work_sync(&haptics->work); 9dd555e2 Mark Brown 2012-11-26 161 9dd555e2 Mark Brown 2012-11-26 162 mutex_lock_nested(dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); 9dd555e2 Mark Brown 2012-11-26 163 9dd555e2 Mark Brown 2012-11-26 @164 if (haptics->arizona->dapm) 9dd555e2 Mark Brown 2012-11-26 165 snd_soc_dapm_disable_pin(haptics->arizona->dapm, "HAPTICS"); 9dd555e2 Mark Brown 2012-11-26 166 9dd555e2 Mark Brown 2012-11-26 167 mutex_unlock(dapm_mutex); --- 0-DAY kernel build testing backend Open Source Technology Center Fengguang Wu, Yuanhan Liu Intel Corporation

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[alsa-devel] aic3104 confusion with input selection
by rohith R 29 Nov '12

29 Nov '12

shouldn't there be a .get function for SOC_DAPM_SINGLE_AIC3X macro? There is a hole in the list of allowable values for this registers which the macro addresses (0-8 and 15 for bank 0 registers 17 and register 18 )

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