January 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ALSA: HDA: cxt5066 - Use asus model for Asus U50F, select correct SPDIF output
by Andy Robinson 24 Jan '11

24 Jan '11

The attached patch adds a quirk to use the newly-added asus model for the Asus U50F. It also selects the correct digital outputs based on the pin config instead of always using node 0x21, since the Asus U50F uses node 0x12 for digital out.

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[alsa-devel] ASoC updates for 2.6.38
by Mark Brown 24 Jan '11

24 Jan '11

The following changes since commit f758b53a79ce6076542739563985d5ff0521dc5a: Merge branch 'topic/hda' (2011-01-22 17:29:50 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.38 Usual small, driver-specific updates. Dimitris Papastamos (1): ASoC: WM8995: Fix incorrect use of snd_soc_update_bits() Mark Brown (1): ASoC: Handle low measured DC offsets for wm_hubs devices Qiao Zhou (1): ASoC: WM8994: fix wrong value in tristate function Rajashekhara, Sudhakar (1): ASoC: da8xx/omap-l1xx: match codec_name with i2c ids sound/soc/codecs/wm8994.c | 2 +- sound/soc/codecs/wm8995.c | 2 +- sound/soc/codecs/wm_hubs.c | 15 +++++++-------- sound/soc/davinci/davinci-evm.c | 2 +- 4 files changed, 10 insertions(+), 11 deletions(-)

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[alsa-devel] Support of different channels for ADC driver
by subashchandra.bose＠wipro.com 24 Jan '11

24 Jan '11

Hi All, I am developing General ADC driver with input data for voice channel, High Speed channel and low speed channel. I went into alsa-project website and found alsa pcm interface can be used to configure voice channel. How can I use to configure high speed and low speed channels as well in same driver. Thanks in Advance, -Subash. Please do not print this email unless it is absolutely necessary. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

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[alsa-devel] [PATCH] ALSA: hda - add quirk for Ordissimo EVE using a realtek ALC662
by Anisse Astier 24 Jan '11

24 Jan '11

--- sound/pci/hda/patch_realtek.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7262980..a72c8bd 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -18292,6 +18292,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = { ALC662_3ST_6ch_DIG), SND_PCI_QUIRK_MASK(0x1854, 0xf000, 0x2000, "ASUS H13-200x", ALC663_ASUS_H13), + SND_PCI_QUIRK(0x1991, 0x5628, "Ordissimo EVE", ALC662_LENOVO_101E), {} }; -- 1.7.3.2

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Re: [alsa-devel] Bug in setting channels, related to recent ELD/EDID changes?
by Anssi Hannula 24 Jan '11

24 Jan '11

(restored CC I removed, sorry about that) On 11.01.2011 09:15, Takashi Iwai wrote: > At Mon, 10 Jan 2011 21:36:10 +0100, > Takashi Iwai wrote: >> >> At Mon, 10 Jan 2011 21:58:26 +0200, >> Anssi Hannula wrote: >>> >>> >>> Takashi Iwai kirjoitti: >>>> At Sun, 02 Jan 2011 12:16:44 +0200, >>>> Anssi Hannula wrote: >>>>> >>>>> Hi! >>>>> >>>>> Takashi, it seems the "allow all when not plugged in" behaviour you >>>>> implemented in bbbe3390 is not enough to allow applications to open the >>>>> device before the output device is ready. Switched-off A/V receivers may >>>>> provide the information of the plugged-in television which usually >>>>> restricts the max_channels to 2, causing applications to fail opening >>>>> the device with more channels when a the A/V receiver is switched off. >>>>> >>>>> Also, even if the application can gracefully fallback to stereo, AFAIK >>>>> there is no method for an application to get informed when the >>>>> limitation is lifted, so it can't automatically resume multichannel >>>>> output when the A/V receiver is switched on again. >>>>> >>>>> Based on this info, I guess the restrictions based on ELD should be >>>>> removed :/ >>>>> Unless you have some other ideas to fix the issue, of course. >>>> >>>> What about the patch below? If it's unplugged, the valid flag should >>>> be cleared, so the next open fall backs to the default state (i.e. >>>> allows all). >>> >>> But it is not unplugged. The multichannel-capable receiver forwards EDID >>> of TV which doesn't allow more than 2 channels, when the receiver is >>> swtiched off. >> >> Well, in that case, this is the correct behavior, IMO. At this >> moment, setting channels to 2 is the right thing. Then, how can you >> know that more channels will be available in future at all? >> >> What you want to have a static configuration way by ignoring the ELD >> information. But, the interface isn't pretty easy if it's implemented >> in control API, I suppose. > > ... or, maybe an easier option is to use the module parameter like > the patch below. It can be changed dynamically via sysfs as well. Better than nothing I guess. Note that the condition is backwards in the below patch :) > --- > diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c > index d1b1b57..4115a39 100644 > --- a/sound/pci/hda/patch_hdmi.c > +++ b/sound/pci/hda/patch_hdmi.c > @@ -31,10 +31,14 @@ > #include <linux/init.h> > #include <linux/delay.h> > #include <linux/slab.h> > +#include <linux/moduleparam.h> > #include <sound/core.h> > #include "hda_codec.h" > #include "hda_local.h" > > +static bool static_hdmi_pcm; > +module_param(static_hdmi_pcm, bool, 0644); > + > /* > * The HDMI/DisplayPort configuration can be highly dynamic. A graphics device > * could support two independent pipes, each of them can be connected to one or > @@ -827,7 +831,7 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, > *codec_pars = *hinfo; > > eld = &spec->sink_eld[idx]; > - if (eld->sad_count > 0) { > + if (static_hdmi_pcm && eld->sad_count > 0) { > hdmi_eld_update_pcm_info(eld, hinfo, codec_pars); > if (hinfo->channels_min > hinfo->channels_max || > !hinfo->rates || !hinfo->formats) -- Anssi Hannula

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[alsa-devel] [PATCH] ASoC: Fix type for snd_soc_volatile_register()
by Mark Brown 24 Jan '11

24 Jan '11

We generally refer to registers as unsigned ints (including in the underlying CODEC driver operation). Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 3 ++- sound/soc/soc-core.c | 22 +++++++++++++++++++++- 2 files changed, 23 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index c184f84..1355ef0 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -267,7 +267,8 @@ int snd_soc_register_codec(struct device *dev, const struct snd_soc_codec_driver *codec_drv, struct snd_soc_dai_driver *dai_drv, int num_dai); void snd_soc_unregister_codec(struct device *dev); -int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, int reg); +int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, + unsigned int reg); int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, int addr_bits, int data_bits, enum snd_soc_control_type control); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b0e7689..ba81892 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2029,7 +2029,8 @@ static int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) * * Boolean function indiciating if a CODEC register is volatile. */ -int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, int reg) +int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, + unsigned int reg) { if (codec->volatile_register) return codec->volatile_register(codec, reg); @@ -2039,6 +2040,25 @@ int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, int reg) EXPORT_SYMBOL_GPL(snd_soc_codec_volatile_register); /** + * snd_soc_codec_register_name: Return a plain text register name + * + * @codec: CODEC to query. + * @reg: Register to query for. + * + * Returns a string with a plain text name for the register, or NULL + * if none is available. + */ +const char *snd_soc_codec_register_name(struct snd_soc_codec *codec, + unsigned int reg) +{ + if (codec->volatile_register) + return codec->register_name(codec, reg); + else + return NULL; +} +EXPORT_SYMBOL_GPL(snd_soc_codec_register_name); + +/** * snd_soc_new_ac97_codec - initailise AC97 device * @codec: audio codec * @ops: AC97 bus operations -- 1.7.2.3

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[alsa-devel] [PATCH] ASoC: Fix codec device id format used by some dai_links
by Lars-Peter Clausen 24 Jan '11

24 Jan '11

The id part of a I2C device name is created with the "%d-%04x" format string. So for example for a I2C device, which is connected to the adapter with the id 0 and has its address set to 0x1a the id part of the devices name would be "0-001a". Currently some sound board drivers have the id part the codec_name field of their dai_link structures set as if it had been created by a "%d-0x%x" format string. For example "0-0x1a" instead of "0-001a". As a result there is no match between the codec device and the dai_link and no sound card is instantiated. This patch fixes it. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Cc: stable(a)kernel.org (for 2.6.37) --- This problem has been noted with the neo1993_gta02 sound board driver and has only been runtime tested for the neo1973 and neo1973_gta02 sound board drivers, but in theory the problem should be the same for all of the drivers changed by this patch. --- sound/soc/atmel/playpaq_wm8510.c | 2 +- sound/soc/atmel/snd-soc-afeb9260.c | 2 +- sound/soc/blackfin/bf5xx-ssm2602.c | 2 +- sound/soc/imx/wm1133-ev1.c | 2 +- sound/soc/samsung/goni_wm8994.c | 4 ++-- sound/soc/samsung/jive_wm8750.c | 2 +- sound/soc/samsung/neo1973_gta02_wm8753.c | 4 ++-- sound/soc/samsung/neo1973_wm8753.c | 4 ++-- sound/soc/samsung/s3c24xx_simtec_hermes.c | 2 +- sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c | 2 +- sound/soc/samsung/smartq_wm8987.c | 2 +- 11 files changed, 14 insertions(+), 14 deletions(-) diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c index 1aac2f4..e2b6595 100644 --- a/sound/soc/atmel/playpaq_wm8510.c +++ b/sound/soc/atmel/playpaq_wm8510.c @@ -356,7 +356,7 @@ static struct snd_soc_dai_link playpaq_wm8510_dai = { .stream_name = "WM8510 PCM", .cpu_dai_name= "atmel-ssc-dai.0", .platform_name = "atmel-pcm-audio", - .codec_name = "wm8510-codec.0-0x1a", + .codec_name = "wm8510-codec.0-001a", .codec_dai_name = "wm8510-hifi", .init = playpaq_wm8510_init, .ops = &playpaq_wm8510_ops, diff --git a/sound/soc/atmel/snd-soc-afeb9260.c b/sound/soc/atmel/snd-soc-afeb9260.c index da2208e..5e4d499 100644 --- a/sound/soc/atmel/snd-soc-afeb9260.c +++ b/sound/soc/atmel/snd-soc-afeb9260.c @@ -129,7 +129,7 @@ static struct snd_soc_dai_link afeb9260_dai = { .cpu_dai_name = "atmel-ssc-dai.0", .codec_dai_name = "tlv320aic23-hifi", .platform_name = "atmel_pcm-audio", - .codec_name = "tlv320aic23-codec.0-0x1a", + .codec_name = "tlv320aic23-codec.0-001a", .init = afeb9260_tlv320aic23_init, .ops = &afeb9260_ops, }; diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c index e902b24..ad28663 100644 --- a/sound/soc/blackfin/bf5xx-ssm2602.c +++ b/sound/soc/blackfin/bf5xx-ssm2602.c @@ -119,7 +119,7 @@ static struct snd_soc_dai_link bf5xx_ssm2602_dai = { .cpu_dai_name = "bf5xx-i2s", .codec_dai_name = "ssm2602-hifi", .platform_name = "bf5xx-pcm-audio", - .codec_name = "ssm2602-codec.0-0x1b", + .codec_name = "ssm2602-codec.0-001b", .ops = &bf5xx_ssm2602_ops, }; diff --git a/sound/soc/imx/wm1133-ev1.c b/sound/soc/imx/wm1133-ev1.c index 75b4c72..f507e57 100644 --- a/sound/soc/imx/wm1133-ev1.c +++ b/sound/soc/imx/wm1133-ev1.c @@ -246,7 +246,7 @@ static struct snd_soc_dai_link wm1133_ev1_dai = { .cpu_dai_name = "imx-ssi.0", .codec_dai_name = "wm8350-hifi", .platform_name = "imx-fiq-pcm-audio.0", - .codec_name = "wm8350-codec.0-0x1a", + .codec_name = "wm8350-codec.0-001a", .init = wm1133_ev1_init, .ops = &wm1133_ev1_ops, .symmetric_rates = 1, diff --git a/sound/soc/samsung/goni_wm8994.c b/sound/soc/samsung/goni_wm8994.c index f6b3a3c..3a7861c 100644 --- a/sound/soc/samsung/goni_wm8994.c +++ b/sound/soc/samsung/goni_wm8994.c @@ -238,7 +238,7 @@ static struct snd_soc_dai_link goni_dai[] = { .cpu_dai_name = "samsung-i2s.0", .codec_dai_name = "wm8994-hifi", .platform_name = "samsung-audio", - .codec_name = "wm8994-codec.0-0x1a", + .codec_name = "wm8994-codec.0-001a", .init = goni_wm8994_init, .ops = &goni_hifi_ops, }, { @@ -247,7 +247,7 @@ static struct snd_soc_dai_link goni_dai[] = { .cpu_dai_name = "goni-voice-dai", .codec_dai_name = "wm8994-voice", .platform_name = "samsung-audio", - .codec_name = "wm8994-codec.0-0x1a", + .codec_name = "wm8994-codec.0-001a", .ops = &goni_voice_ops, }, }; diff --git a/sound/soc/samsung/jive_wm8750.c b/sound/soc/samsung/jive_wm8750.c index 3b53ad5..14eb6ea 100644 --- a/sound/soc/samsung/jive_wm8750.c +++ b/sound/soc/samsung/jive_wm8750.c @@ -131,7 +131,7 @@ static struct snd_soc_dai_link jive_dai = { .cpu_dai_name = "s3c2412-i2s", .codec_dai_name = "wm8750-hifi", .platform_name = "samsung-audio", - .codec_name = "wm8750-codec.0-0x1a", + .codec_name = "wm8750-codec.0-001a", .init = jive_wm8750_init, .ops = &jive_ops, }; diff --git a/sound/soc/samsung/neo1973_gta02_wm8753.c b/sound/soc/samsung/neo1973_gta02_wm8753.c index 69e08fd..5e77cd1 100644 --- a/sound/soc/samsung/neo1973_gta02_wm8753.c +++ b/sound/soc/samsung/neo1973_gta02_wm8753.c @@ -391,7 +391,7 @@ static struct snd_soc_dai_link neo1973_gta02_dai[] = { .codec_dai_name = "wm8753-hifi", .init = neo1973_gta02_wm8753_init, .platform_name = "samsung-audio", - .codec_name = "wm8753-codec.0-0x1a", + .codec_name = "wm8753-codec.0-001a", .ops = &neo1973_gta02_hifi_ops, }, { /* Voice via BT */ @@ -400,7 +400,7 @@ static struct snd_soc_dai_link neo1973_gta02_dai[] = { .cpu_dai_name = "bluetooth-dai", .codec_dai_name = "wm8753-voice", .ops = &neo1973_gta02_voice_ops, - .codec_name = "wm8753-codec.0-0x1a", + .codec_name = "wm8753-codec.0-001a", .platform_name = "samsung-audio", }, }; diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index c7a2451..cf69e14 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -561,7 +561,7 @@ static struct snd_soc_dai_link neo1973_dai[] = { .platform_name = "samsung-audio", .cpu_dai_name = "s3c24xx-i2s", .codec_dai_name = "wm8753-hifi", - .codec_name = "wm8753-codec.0-0x1a", + .codec_name = "wm8753-codec.0-001a", .init = neo1973_wm8753_init, .ops = &neo1973_hifi_ops, }, @@ -571,7 +571,7 @@ static struct snd_soc_dai_link neo1973_dai[] = { .platform_name = "samsung-audio", .cpu_dai_name = "bluetooth-dai", .codec_dai_name = "wm8753-voice", - .codec_name = "wm8753-codec.0-0x1a", + .codec_name = "wm8753-codec.0-001a", .ops = &neo1973_voice_ops, }, }; diff --git a/sound/soc/samsung/s3c24xx_simtec_hermes.c b/sound/soc/samsung/s3c24xx_simtec_hermes.c index d7b3e6e..2fa6d1b 100644 --- a/sound/soc/samsung/s3c24xx_simtec_hermes.c +++ b/sound/soc/samsung/s3c24xx_simtec_hermes.c @@ -84,7 +84,7 @@ static int simtec_hermes_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link simtec_dai_aic33 = { .name = "tlv320aic33", .stream_name = "TLV320AIC33", - .codec_name = "tlv320aic3x-codec.0-0x1a", + .codec_name = "tlv320aic3x-codec.0-001a", .cpu_dai_name = "s3c24xx-i2s", .codec_dai_name = "tlv320aic3x-hifi", .platform_name = "samsung-audio", diff --git a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c index ff6168f..d219223 100644 --- a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c +++ b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c @@ -73,7 +73,7 @@ static int simtec_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link simtec_dai_aic23 = { .name = "tlv320aic23", .stream_name = "TLV320AIC23", - .codec_name = "tlv320aic3x-codec.0-0x1a", + .codec_name = "tlv320aic3x-codec.0-001a", .cpu_dai_name = "s3c24xx-i2s", .codec_dai_name = "tlv320aic3x-hifi", .platform_name = "samsung-audio", diff --git a/sound/soc/samsung/smartq_wm8987.c b/sound/soc/samsung/smartq_wm8987.c index 0a2c4f2..d14f939 100644 --- a/sound/soc/samsung/smartq_wm8987.c +++ b/sound/soc/samsung/smartq_wm8987.c @@ -207,7 +207,7 @@ static struct snd_soc_dai_link smartq_dai[] = { .cpu_dai_name = "samsung-i2s.0", .codec_dai_name = "wm8750-hifi", .platform_name = "samsung-audio", - .codec_name = "wm8750-codec.0-0x1a", + .codec_name = "wm8750-codec.0-001a", .init = smartq_wm8987_init, .ops = &smartq_hifi_ops, }, -- 1.7.2.3

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[alsa-devel] commit 109fef9edcc100952eec980acbc2e1295627fbab ALSA: timer: automatically load the high-resolution timer causing MIDI playback tempo problems
by Arthur Marsh 23 Jan '11

23 Jan '11

Hi, on an old machine I have with Intel 440BX chipset and PII-266 CPU, I found recent kernels causing MIDI sound output from a Soundblaster Audigy 2 ZS to be varying widely in tempo. dmesg reports with recent kernels (both those that allow correct playback of MIDI files and those that give wide variation in playback tempo): Clocksource tsc unstable (delta = 132040172 ns) Switching to clocksource pit After git-bisection I found: git bisect bad 109fef9edcc100952eec980acbc2e1295627fbab is the first bad commit commit 109fef9edcc100952eec980acbc2e1295627fbab Author: Clemens Ladisch <clemens(a)ladisch.de> Date: Thu Nov 18 09:53:54 2010 +0100 ALSA: timer: automatically load the high-resolution timer Increase the default timer limit so that snd-hrtimer.ko can be automatically loaded when needed, e.g., when used as the default sequencer timer. This replaces the check for the obsolete CONFIG_SND_HPET. Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> :040000 040000 c7b3b046ea99e563228f36ea4a00f9b763bf5813 741f137a201b31bdf5eaaa58c3031b4b2a68c6d6 M sound diff --git a/sound/core/timer.c b/sound/core/timer.c index 13afb60..b3aaa60 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -34,8 +34,8 @@ #include <sound/initval.h> #include <linux/kmod.h> -#if defined(CONFIG_SND_HPET) || defined(CONFIG_SND_HPET_MODULE) -#define DEFAULT_TIMER_LIMIT 3 +#if defined(CONFIG_SND_HRTIMER) || defined(CONFIG_SND_HRTIMER_MODULE) +#define DEFAULT_TIMER_LIMIT 4 #elif defined(CONFIG_SND_RTCTIMER) || defined(CONFIG_SND_RTCTIMER_MODULE) #define DEFAULT_TIMER_LIMIT 2 #else I am currently rebuilding kernel 2.6.38-rc2 with this patch reverted to verify that this commit triggers the problem with this machine. Arthur.

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[alsa-devel] [PATCH] pcm: fix typo in snd_pcm_recover() error message
by cosoleto＠gmail.com 23 Jan '11

23 Jan '11

From: Francesco Cosoleto <cosoleto(a)gmail.com> Signed-off-by: Francesco Cosoleto <cosoleto(a)gmail.com> --- src/pcm/pcm.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 906b502..02dea0d 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -7313,7 +7313,7 @@ int snd_pcm_recover(snd_pcm_t *pcm, int err, int silent) else s = "overrun"; if (!silent) - SNDERR("%s occured", s); + SNDERR("%s occurred", s); err = snd_pcm_prepare(pcm); if (err < 0) { SNDERR("cannot recovery from %s, prepare failed: %s", s, snd_strerror(err)); -- 1.7.1

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[alsa-devel] [PATCH] ASoC: EDB93xx machine sound driver with CS4271
by Alexander Sverdlin 23 Jan '11

23 Jan '11

From: Alexander Sverdlin <subaparts(a)yandex.ru> Added support for EDB93xx sound with CS4271 CODEC. Features: - Playback, Capture - Sample rates from 8kHz to 96kHz tested Limitations: - No DAPM support PS. The same code as submitted on 8 Dec 2010, but now all dependencies are applied. Signed-off-by: Alexander Sverdlin <subaparts(a)yandex.ru> --- arch/arm/mach-ep93xx/edb93xx.c | 54 +++++++++++++++ sound/soc/ep93xx/Kconfig | 9 +++ sound/soc/ep93xx/Makefile | 2 + sound/soc/ep93xx/edb93xx.c | 142 ++++++++++++++++++++++++++++++++++++++++ 4 files changed, 207 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-ep93xx/edb93xx.c b/arch/arm/mach-ep93xx/edb93xx.c index 4b04316..f453e5a 100644 --- a/arch/arm/mach-ep93xx/edb93xx.c +++ b/arch/arm/mach-ep93xx/edb93xx.c @@ -30,12 +30,21 @@ #include <linux/gpio.h> #include <linux/i2c.h> #include <linux/i2c-gpio.h> +#include <linux/spi/spi.h> +#include <mach/ep93xx_spi.h> #include <mach/hardware.h> #include <asm/mach-types.h> #include <asm/mach/arch.h> +#include <sound/cs4271.h> + +#define edb93xx_has_audio() (machine_is_edb9301() || \ + machine_is_edb9302() || \ + machine_is_edb9302a() || \ + machine_is_edb9307a() || \ + machine_is_edb9315a()) static void __init edb93xx_register_flash(void) { @@ -93,6 +102,49 @@ static void __init edb93xx_register_i2c(void) /************************************************************************* + * EDB93xx SPI peripheral handling + *************************************************************************/ +static struct cs4271_platform_data cs4271_edb93xx_gpio = { + .gpio_nreset = EP93XX_GPIO_LINE_EGPIO1, + .gpio_disable = EP93XX_GPIO_LINE_EGPIO6, +}; + +static struct spi_board_info edb93xx_spi_board_info[] = { + { + .modalias = "cs4271-codec", + .max_speed_hz = 6000000, + .bus_num = 0, + .chip_select = 0, + .mode = SPI_MODE_3, + .platform_data = &cs4271_edb93xx_gpio, + }, +}; + +static struct ep93xx_spi_info edb93xx_spi_info = { + .num_chipselect = ARRAY_SIZE(edb93xx_spi_board_info), +}; + +static void __init edb93xx_register_spi(void) +{ + if (edb93xx_has_audio()) { + ep93xx_register_spi(&edb93xx_spi_info, + edb93xx_spi_board_info, + ARRAY_SIZE(edb93xx_spi_board_info)); + } +} + +/************************************************************************* + * EDB93xx I2S + *************************************************************************/ +static void __init edb93xx_register_i2s(void) +{ + if (edb93xx_has_audio()) { + ep93xx_register_i2s(); + } +} + + +/************************************************************************* * EDB93xx pwm *************************************************************************/ static void __init edb93xx_register_pwm(void) @@ -117,6 +169,8 @@ static void __init edb93xx_init_machine(void) edb93xx_register_flash(); ep93xx_register_eth(&edb93xx_eth_data, 1); edb93xx_register_i2c(); + edb93xx_register_spi(); + edb93xx_register_i2s(); edb93xx_register_pwm(); } diff --git a/sound/soc/ep93xx/Kconfig b/sound/soc/ep93xx/Kconfig index 5742904..91a28de 100644 --- a/sound/soc/ep93xx/Kconfig +++ b/sound/soc/ep93xx/Kconfig @@ -30,3 +30,12 @@ config SND_EP93XX_SOC_SIMONE help Say Y or M here if you want to add support for AC97 audio on the Simplemachines Sim.One board. + +config SND_EP93XX_SOC_EDB93XX + tristate "SoC Audio support for Cirrus Logic EDB93xx boards" + depends on SND_EP93XX_SOC && (MACH_EDB9301 || MACH_EDB9302 || MACH_EDB9302A || MACH_EDB9307A || MACH_EDB9315A) + select SND_EP93XX_SOC_I2S + select SND_SOC_CS4271 + help + Say Y or M here if you want to add support for I2S audio on the + Cirrus Logic EDB93xx boards. diff --git a/sound/soc/ep93xx/Makefile b/sound/soc/ep93xx/Makefile index 8e7977f..5514146 100644 --- a/sound/soc/ep93xx/Makefile +++ b/sound/soc/ep93xx/Makefile @@ -10,6 +10,8 @@ obj-$(CONFIG_SND_EP93XX_SOC_AC97) += snd-soc-ep93xx-ac97.o # EP93XX Machine Support snd-soc-snappercl15-objs := snappercl15.o snd-soc-simone-objs := simone.o +snd-soc-edb93xx-objs := edb93xx.o obj-$(CONFIG_SND_EP93XX_SOC_SNAPPERCL15) += snd-soc-snappercl15.o obj-$(CONFIG_SND_EP93XX_SOC_SIMONE) += snd-soc-simone.o +obj-$(CONFIG_SND_EP93XX_SOC_EDB93XX) += snd-soc-edb93xx.o diff --git a/sound/soc/ep93xx/edb93xx.c b/sound/soc/ep93xx/edb93xx.c new file mode 100644 index 0000000..b270085 --- /dev/null +++ b/sound/soc/ep93xx/edb93xx.c @@ -0,0 +1,142 @@ +/* + * SoC audio for EDB93xx + * + * Copyright (c) 2010 Alexander Sverdlin <subaparts(a)yandex.ru> + * + * 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. + * + * This driver support CS4271 codec being master or slave, working + * in control port mode, connected either via SPI or I2C. + * The data format accepted is I2S or left-justified. + * DAPM support not implemented. + */ + +#include <linux/platform_device.h> +#include <linux/gpio.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <asm/mach-types.h> +#include <mach/hardware.h> +#include "ep93xx-pcm.h" + +#define edb93xx_has_audio() (machine_is_edb9301() || \ + machine_is_edb9302() || \ + machine_is_edb9302a() || \ + machine_is_edb9307a() || \ + machine_is_edb9315a()) + +static int edb93xx_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int err; + unsigned int rate = params_rate(params); + /* + * We set LRCLK equal to `rate' and SCLK = LRCLK * 64, + * because our sample size is 32 bit * 2 channels. + * I2S standard permits us to transmit more bits than + * the codec uses. + * MCLK = SCLK * 4 is the best recommended value, + * but we have to fall back to ratio 2 for higher + * sample rates. + */ + unsigned int mclk_rate = rate * 64 * ((rate <= 48000) ? 4 : 2); + + err = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBS_CFS); + if (err) + return err; + + err = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBS_CFS); + if (err) + return err; + + err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk_rate, + SND_SOC_CLOCK_IN); + if (err) + return err; + + return snd_soc_dai_set_sysclk(cpu_dai, 0, mclk_rate, + SND_SOC_CLOCK_OUT); +} + +static struct snd_soc_ops edb93xx_ops = { + .hw_params = edb93xx_hw_params, +}; + +static struct snd_soc_dai_link edb93xx_dai = { + .name = "CS4271", + .stream_name = "CS4271 HiFi", + .platform_name = "ep93xx-pcm-audio", + .cpu_dai_name = "ep93xx-i2s", + .codec_name = "spi0.0", + .codec_dai_name = "cs4271-hifi", + .ops = &edb93xx_ops, +}; + +static struct snd_soc_card snd_soc_edb93xx = { + .name = "EDB93XX", + .dai_link = &edb93xx_dai, + .num_links = 1, +}; + +static struct platform_device *edb93xx_snd_device; + +static int __init edb93xx_init(void) +{ + int ret; + + if (!edb93xx_has_audio()) + return -ENODEV; + + ret = ep93xx_i2s_acquire(EP93XX_SYSCON_DEVCFG_I2SONAC97, + EP93XX_SYSCON_I2SCLKDIV_ORIDE | + EP93XX_SYSCON_I2SCLKDIV_SPOL); + if (ret) + return ret; + + edb93xx_snd_device = platform_device_alloc("soc-audio", -1); + if (!edb93xx_snd_device) { + ret = -ENOMEM; + goto free_i2s; + } + + platform_set_drvdata(edb93xx_snd_device, &snd_soc_edb93xx); + ret = platform_device_add(edb93xx_snd_device); + if (ret) + goto device_put; + + return 0; + +device_put: + platform_device_put(edb93xx_snd_device); +free_i2s: + ep93xx_i2s_release(); + return ret; +} +module_init(edb93xx_init); + +static void __exit edb93xx_exit(void) +{ + platform_device_unregister(edb93xx_snd_device); + ep93xx_i2s_release(); +} +module_exit(edb93xx_exit); + +MODULE_AUTHOR("Alexander Sverdlin <subaparts(a)yandex.ru>"); +MODULE_DESCRIPTION("ALSA SoC EDB93xx"); +MODULE_LICENSE("GPL");

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