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Re: [alsa-devel] [PATCH] ASoC: ak4554: add DT support
by Lars-Peter Clausen 08 Jul '13

08 Jul '13

On 07/06/2013 08:58 PM, Sylwester Nawrocki wrote: > On 07/05/2013 07:13 PM, Stephen Warren wrote: >> On 07/05/2013 11:04 AM, Lars-Peter Clausen wrote: >>> On 07/05/2013 06:55 PM, Stephen Warren wrote: >>>> On 07/05/2013 03:37 AM, Lars-Peter Clausen wrote: >>>>>> + - compatible : "asahi-kasei,ak4554" >>>>> >>>>> The vendor prefix for Asahi Kasei is documented as "ak" in >>>>> Documentation/devicetree/bindings/vendor-prefixes.txt. But it doesn;t seem >>>>> to be used in any upstream drivers, Stephen add that prefix maybe he knows >>>>> if it is used. If it is unused you should probably send a patch updating the >>>>> Documentation. >>>> >>>> It's used at: >>>> >>>> arch/arm/boot/dts/tegra20-seaboard.dts:499: compatible = "ak,ak8975"; >>>> >>>> ... although I see that somebody has incorrectly modified >>>> ./drivers/iio/magnetometer/ak8975.c to support the "asahi-kasei," vendor >>>> prefix:-( >>> >>> It looks as if the ak prefix was never in an upstream kernel and the commit >>> date that added the dt table predates the commit date for your devicetree. >> >> The usage I quoted above is in the mainline kernel, along with the entry >> in vendor-prefixes.txt. >> >> When I fixed vendor-prefixes.txt to define a prefix for AK, and updated >> the Tegra DT to use them[1], I guess I didn't even look at the driver, >> since all I2C device matching uses the i2c_device_id table rather than >> the of_device_id, and the i2c_device_id table doesn't have a concept of >> a vendor prefix. Hence, I didn't notice the conflict. >> >> So, even though the asahi-kasei prefix is in the of_device_id table in >> the driver, I'd say the ak prefix trumps it, since it's actually in-use >> in device trees, and is registered in vendor-prefixes.txt. > > I'm going to update this [2] patch to use "ak,ak8975" compatible string > then. Still there are some audio codec drivers that use "asahi-kasei" > vendor prefix: > > Documentation/devicetree/bindings/sound/ak4104.txt: - compatible : > "asahi-kasei,ak4104" > Documentation/devicetree/bindings/sound/ak4104.txt: compatible = > "asahi-kasei,ak4104"; > Documentation/devicetree/bindings/sound/ak4642.txt: - compatible : > "asahi-kasei,ak4642" or "asahi-kasei,ak4643" or "asahi-kasei,ak4648" > Documentation/devicetree/bindings/sound/ak4642.txt: compatible = > "asahi-kasei,ak4642"; > Documentation/devicetree/bindings/sound/ak5386.txt: - compatible : > "asahi-kasei,ak5386" > Documentation/devicetree/bindings/sound/ak5386.txt: compatible = > "asahi-kasei,ak5386"; > ... > sound/soc/codecs/ak4104.c: { .compatible = "asahi-kasei,ak4104", }, > sound/soc/codecs/ak4642.c: { .compatible = "asahi-kasei,ak4642", .data = > &soc_codec_dev_ak4642}, > sound/soc/codecs/ak4642.c: { .compatible = "asahi-kasei,ak4643", .data = > &soc_codec_dev_ak4642}, > sound/soc/codecs/ak4642.c: { .compatible = "asahi-kasei,ak4648", .data = > &soc_codec_dev_ak4648}, > sound/soc/codecs/ak5386.c: { .compatible = "asahi-kasei,ak5386", }, > > It's not in any dts file though, I'm not sure if that makes any > difference. > Well strictly speaking this is all public ABI, so I guess the best is to add a ak prefixed compatible string besides the asahi-kasei one and stick to ak only fore future drivers. - Lars

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Re: [alsa-devel] No sound with Sony VAIO VPCZ1 (ALC889)
by Raymond Yau 08 Jul '13

08 Jul '13

>>> >>> Actually... there should be three mic channels: >>> The built-in (internal) microphone >>> + >>> One mic jack beside the headphone jack >>> + >>> One other hidden within the headphone jack: >>> (a) the Notebook came with Noise-cancelling headsets, but they are small in-ear plugs so there is no place for noise-cancelling logic in the plugs >>> (b) I think I remember to have read somewhere that this thingy actually needs the windows driver installed to be noise-cancelling! >>> (c) the PLUG of these headphones have another channel (visible through another ring on the jack) >>> >>> This explains the 2nd mic channel :) >> >> >> Try hda-jack-sense-test to find out which node is used by the headset mic jack (TRRS) and mic jack. > > > Thanks for the hint. > > Found and installed: snd-hda-tools_0.20111207_amd64.deb > But it gave me: > > /usr/src % hda-jack-sense-test $1 #28540 me@seven pts/4 > Traceback (most recent call last): > File "/usr/bin/hda-jack-sense-test", line 44, in <module> > codec = HDACodec(cardindex, codecindex) > File "/usr/share/snd-hda-tools/hda_codec.py", line 908, in __init__ > self.fd = os.open("/dev/snd/hwC%sD%s" % (card, device), os.O_RDWR) > OSError: [Errno 13] Permission denied: '/dev/snd/hwC0D0' You need root privilege and need 0.20130207 hda-tools > > /usr/src % sudo hda-jack-sense-test $1 #28541 me@seven pts/4 > Traceback (most recent call last): > File "/usr/bin/hda-jack-sense-test", line 44, in <module> > codec = HDACodec(cardindex, codecindex) > File "/usr/share/snd-hda-tools/hda_codec.py", line 921, in __init__ > self.parse_proc() > File "/usr/share/snd-hda-tools/hda_codec.py", line 1003, in parse_proc > self.proc_codec = HDACodecProc(self.card, self.device, file) > File "/usr/share/snd-hda-tools/hda_proc.py", line 474, in __init__ > self.parse(proc_file) > File "/usr/share/snd-hda-tools/hda_proc.py", line 647, in parse > line, nid = self.decodeintw(line, 'Node ') > File "/usr/share/snd-hda-tools/hda_proc.py", line 140, in decodeintw > self.wrongfile('integer decode %s (%s)' % (repr(str), repr(prefix))) > File "/usr/share/snd-hda-tools/hda_proc.py", line 143, in wrongfile > raise ValueError, "wrong proc file format (%s)" % msg > ValueError: wrong proc file format (integer decode 'State of AFG node 0x01:' ('Node ')) > http://git.alsa-project.org/?p=alsa.git;a=commitdiff;h=7859cb050f2cbeba54ce…

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[alsa-devel] [PATCH] ASoC: rt5640: add headset detection function
by bardliao＠realtek.com 08 Jul '13

08 Jul '13

From: Bard Liao <bardliao(a)realtek.com> Add headset detection function to rt5640 codec driver. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5640.c | 60 +++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5640.h | 2 ++ 2 files changed, 62 insertions(+) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index ce585e3..657efb7 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1911,6 +1911,66 @@ static int rt5640_resume(struct snd_soc_codec *codec) #define rt5640_resume NULL #endif +/** + * rt5640_headset_detect - Detect headset. + * @codec: SoC audio codec device. + * @jack_insert: Jack insert or not. + * + * Detect whether is headset or not when jack inserted. + * + * Returns detect status. + */ +int rt5640_headset_detect(struct snd_soc_codec *codec, int jack_insert) +{ + int jack_type; + int sclk_src = 0; + int reg63, reg64; + + if (jack_insert) { + reg63 = snd_soc_read(codec, RT5640_PWR_ANLG1); + reg64 = snd_soc_read(codec, RT5640_PWR_ANLG2); + if (SND_SOC_BIAS_OFF == codec->dapm.bias_level) { + snd_soc_write(codec, RT5640_PWR_ANLG1, 0xa814); + sclk_src = snd_soc_read(codec, RT5640_GLB_CLK) & + RT5640_SCLK_SRC_MASK; + snd_soc_update_bits(codec, RT5640_GLB_CLK, + RT5640_SCLK_SRC_MASK, + 0x2 << RT5640_SCLK_SRC_SFT); + } + snd_soc_update_bits(codec, RT5640_PWR_ANLG1, + RT5640_PWR_LDO2, RT5640_PWR_LDO2); + snd_soc_update_bits(codec, RT5640_PWR_ANLG2, + RT5640_PWR_MB1, RT5640_PWR_MB1); + snd_soc_update_bits(codec, RT5640_MICBIAS, + RT5640_MIC1_OVCD_MASK | RT5640_MIC1_OVTH_MASK | + RT5640_PWR_CLK25M_MASK | RT5640_PWR_MB_MASK, + RT5640_MIC1_OVCD_EN | RT5640_MIC1_OVTH_600UA | + RT5640_PWR_MB_PU | RT5640_PWR_CLK25M_PU); + snd_soc_update_bits(codec, RT5640_DUMMY1, 0x1, 0x1); + msleep(100); + if (snd_soc_read(codec, RT5640_IRQ_CTRL2) & 0x8) + jack_type = RT5640_HEADPHO_DET; + else + jack_type = RT5640_HEADSET_DET; + snd_soc_update_bits(codec, RT5640_IRQ_CTRL2, + RT5640_MB1_OC_CLR, 0); + if (SND_SOC_BIAS_OFF == codec->dapm.bias_level) + snd_soc_update_bits(codec, RT5640_GLB_CLK, + RT5640_SCLK_SRC_MASK, sclk_src); + snd_soc_write(codec, RT5640_PWR_ANLG1, reg63); + snd_soc_write(codec, RT5640_PWR_ANLG2, reg64); + } else { + snd_soc_update_bits(codec, RT5640_MICBIAS, + RT5640_MIC1_OVCD_MASK, + RT5640_MIC1_OVCD_DIS); + + jack_type = RT5640_NO_JACK; + } + + return jack_type; +} +EXPORT_SYMBOL_GPL(rt5640_headset_detect); + #define RT5640_STEREO_RATES SNDRV_PCM_RATE_8000_96000 #define RT5640_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index c48286d..c3856c8 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -2089,4 +2089,6 @@ struct rt5640_priv { int dmic_en; }; +int rt5640_headset_detect(struct snd_soc_codec *codec, int jack_insert); + #endif -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH - oss plugin update revised 1/2] oss: Add support of 32-bit and other formats
by alex.d.wiggins＠gmail.com 08 Jul '13

08 Jul '13

From: Alex Wiggins <alex.d.wiggins(a)gmail.com> Added support of 32-bit and other audio formats via 3 changes: - Additional audio format definitions at the top of the file (if required); - Additional audio formats in oss_hw_params (to allow as input to pass to OSS); - Additional audio formats in oss_hw_constraint (to allow to report to ALSA). Signed-off-by: Alex Wiggins <alex.d.wiggins(a)gmail.com> diff --git a/oss/pcm_oss.c b/oss/pcm_oss.c index d43b44c..d5b5bc4 100644 --- a/oss/pcm_oss.c +++ b/oss/pcm_oss.c @@ -24,6 +24,32 @@ #include <alsa/pcm_external.h> #include <linux/soundcard.h> +/* definitions for additional OSS formats if required */ +#ifndef AFMT_S32_LE +#define AFMT_S32_LE 0x00001000 +#endif +#ifndef AFMT_S32_BE +#define AFMT_S32_BE 0x00002000 +#endif +#ifndef AFMT_S24_LE +#define AFMT_S24_LE 0x00008000 +#endif +#ifndef AFMT_S24_BE +#define AFMT_S24_BE 0x00010000 +#endif +#ifndef AFMT_S24_PACKED +#define AFMT_S24_PACKED 0x00040000 +#endif +#ifndef AFMT_FLOAT +#define AFMT_FLOAT 0x00004000 +#endif +#ifndef AFMT_SPDIF_RAW +#define AFMT_SPDIF_RAW 0x00020000 +#endif +#ifndef AFMT_VORBIS +#define AFMT_VORBIS 0x00000800 +#endif + typedef struct snd_pcm_oss { snd_pcm_ioplug_t io; char *device; @@ -165,6 +191,145 @@ static int oss_hw_params(snd_pcm_ioplug_t *io, case SND_PCM_FORMAT_S16_BE: oss->format = AFMT_S16_BE; break; + /* allow additional data formats (such as 32 bit) as input + to pass to OSS (set in OSS driver) */ + case SND_PCM_FORMAT_S32_LE: + oss->format = AFMT_S32_LE; + break; + case SND_PCM_FORMAT_S32_BE: + oss->format = AFMT_S32_BE; + break; + case SND_PCM_FORMAT_S8: + oss->format = AFMT_S8; + break; + case SND_PCM_FORMAT_U16_LE: + oss->format = AFMT_U16_LE; + break; + case SND_PCM_FORMAT_U16_BE: + oss->format = AFMT_U16_BE; + break; + case SND_PCM_FORMAT_S24_LE: + oss->format = AFMT_S24_LE; + break; + case SND_PCM_FORMAT_S24_BE: + oss->format = AFMT_S24_BE; + break; + case SND_PCM_FORMAT_MU_LAW: + oss->format = AFMT_MU_LAW; + break; + case SND_PCM_FORMAT_A_LAW: + oss->format = AFMT_A_LAW; + break; + case SND_PCM_FORMAT_IMA_ADPCM: + oss->format = AFMT_IMA_ADPCM; + break; + case SND_PCM_FORMAT_MPEG: + oss->format = AFMT_MPEG; + break; + case SND_PCM_FORMAT_S24_3LE: + oss->format = AFMT_S24_PACKED; + break; + case SND_PCM_FORMAT_FLOAT: + oss->format = AFMT_FLOAT; + break; + case SND_PCM_FORMAT_IEC958_SUBFRAME: + oss->format = AFMT_SPDIF_RAW; + break; + /* following formats need defining checking native endian: */ + #if 0 + case SND_PCM_FORMAT_S16: + oss->format = AFMT_S16_NE; + break; + case SND_PCM_FORMAT_U16: + oss->format = AFMT_U16_NE; + break; + case SND_PCM_FORMAT_S24: + oss->format = AFMT_S24_NE; + break; + case SND_PCM_FORMAT_S32: + oss->format = AFMT_S32_NE; + #endif + /* following formats found in ALSA but not in OSS: */ + #if 0 + case SND_PCM_FORMAT_U24_LE: + oss->format = AFMT_U24_LE; + break; + case SND_PCM_FORMAT_U24_BE: + oss->format = AFMT_U24_BE; + break; + case SND_PCM_FORMAT_U32_LE: + oss->format = AFMT_U32_LE; + break; + case SND_PCM_FORMAT_U32_BE: + oss->format = AFMT_U32_BE; + break; + case SND_PCM_FORMAT_FLOAT_LE: + oss->format = AFMT_FLOAT_LE; + break; + case SND_PCM_FORMAT_FLOAT_BE: + oss->format = AFMT_FLOAT_BE; + break; + case SND_PCM_FORMAT_FLOAT64_LE: + oss->format = AFMT_FLOAT64_LE; + break; + case SND_PCM_FORMAT_FLOAT64_BE: + oss->format = AFMT_FLOAT64_BE; + break; + case SND_PCM_FORMAT_IEC958_SUBFRAME_LE: + oss->format = AFMT_SPDIF_RAW_LE; + break; + case SND_PCM_FORMAT_IEC958_SUBFRAME_BE: + oss->format = AFMT_SPDIF_RAW_BE; + break; + case SND_PCM_FORMAT_GSM: + oss->format = AFMT_GSM; + break; + case SND_PCM_FORMAT_SPECIAL: + oss->format = AFMT_SPECIAL; + break; + case SND_PCM_FORMAT_S24_3BE: + oss->format = AFMT_S24_PACKED_BE; + break; + case SND_PCM_FORMAT_U24_3LE: + oss->format = AFMT_U24_PACKED; + break; + case SND_PCM_FORMAT_U24_3BE: + oss->format = AFMT_U24_PACKED_BE; + break; + case SND_PCM_FORMAT_S20_3LE: + oss->format = ; + break; + case SND_PCM_FORMAT_S20_3BE: + oss->format = ; + break; + case SND_PCM_FORMAT_U20_3LE: + oss->format = ; + break; + case SND_PCM_FORMAT_U20_3BE: + oss->format = ; + break; + case SND_PCM_FORMAT_S18_3LE: + oss->format = ; + break; + case SND_PCM_FORMAT_S18_3BE: + oss->format = ; + break; + case SND_PCM_FORMAT_U18_3LE: + oss->format = ; + break; + case SND_PCM_FORMAT_U18_3BE: + oss->format = ; + break; + case SND_PCM_FORMAT_U24: + oss->format = AFMT_U24_NE; + break; + case SND_PCM_FORMAT_U32: + oss->format = AFMT_U32_NE; + break; + case SND_PCM_FORMAT_FLOAT64: + oss->format = AFMT_FLOAT64; + break; + #endif default: fprintf(stderr, "*** OSS: unsupported format %s\n", snd_pcm_format_name(io->format)); return -EINVAL; @@ -239,7 +404,7 @@ static int oss_hw_constraint(snd_pcm_oss_t *oss) SND_PCM_ACCESS_MMAP_INTERLEAVED }; unsigned int nformats; - unsigned int format[5]; + unsigned int format[20]; unsigned int nchannels; unsigned int channel[6]; /* period and buffer bytes must be power of two */ @@ -273,6 +438,40 @@ static int oss_hw_constraint(snd_pcm_oss_t *oss) format[nformats++] = SND_PCM_FORMAT_S16_BE; if (tmp & AFMT_MU_LAW) format[nformats++] = SND_PCM_FORMAT_MU_LAW; + /* allow additional data formats (such as 32 bit) + to be reported from OSS to ALSA below */ + if (tmp & AFMT_S32_LE) + format[nformats++] = SND_PCM_FORMAT_S32_LE; + if (tmp & AFMT_S32_BE) + format[nformats++] = SND_PCM_FORMAT_S32_BE; + if (tmp & AFMT_A_LAW) + format[nformats++] = SND_PCM_FORMAT_A_LAW; + if (tmp & AFMT_IMA_ADPCM) + format[nformats++] = SND_PCM_FORMAT_IMA_ADPCM; + if (tmp & AFMT_S8) + format[nformats++] = SND_PCM_FORMAT_S8; + if (tmp & AFMT_U16_LE) + format[nformats++] = SND_PCM_FORMAT_U16_LE; + if (tmp & AFMT_U16_BE) + format[nformats++] = SND_PCM_FORMAT_U16_BE ; + if (tmp & AFMT_MPEG) + format[nformats++] = SND_PCM_FORMAT_MPEG; + if (tmp & AFMT_FLOAT) + format[nformats++] = SND_PCM_FORMAT_FLOAT; + if (tmp & AFMT_S24_LE) + format[nformats++] = SND_PCM_FORMAT_S24_LE; + if (tmp & AFMT_S24_BE) + format[nformats++] = SND_PCM_FORMAT_S24_BE; + if (tmp & AFMT_SPDIF_RAW) + format[nformats++] = SND_PCM_FORMAT_IEC958_SUBFRAME; + if (tmp & AFMT_S24_PACKED) + format[nformats++] = SND_PCM_FORMAT_S24_3LE; + /* following formats found in OSS but not in ALSA */ + if (tmp & AFMT_AC3) + format[nformats++] = SND_PCM_FORMAT_UNKNOWN; + if (tmp & AFMT_VORBIS) + format[nformats++] = SND_PCM_FORMAT_UNKNOWN; + /* assume signed 16 bit (native CPU endian) if not found above */ if (! nformats) format[nformats++] = SND_PCM_FORMAT_S16; if ((err = snd_pcm_ioplug_set_param_list(io, SND_PCM_IOPLUG_HW_FORMAT, -- 1.8.3.1

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[alsa-devel] ASoC: fsl-ssi: ac97-slave support
by Markus Pargmann 08 Jul '13

08 Jul '13

Hi, This series adds DT support for phycore-ac97. Beside ac97 support, the series adds imx-pcm-fiq and generic DMA bindings to fsl-ssi. Version 9 contains some style fixes. Regards, Markus Changes in v9: - Style and compile fixes. - Removed "ASoC: fsl: Kconfig: Use fsl-ssi for phycore-ac97". This patch is not necessary with the seperated phycore-ac97-dt driver. Changes in v8: - Fix build issues in phycore-ac97. Changes in v7: - Only one soc_ac97_ops struct for both ssi drivers (fsl-ssi and imx-ssi) to make it possible to compile a kernel with both drivers. - Other small bug/style fixes described in the changelogs of the patches. Changes in v6: - Rebased onto 3.10-rc3 with Shawn's pcm cleanup patches - Setup of reset functions moved to phycore-ac97.c Changes in v5: - Pass of_node via parent device to imx-pcm-dma. Changes in v4: - phycore-ac97 uses the iomux functions based on audmux phandle now. - fsl-ssi imx-pcm-fiq got some new comments about problems with imx-ssi and incompatible codecs Changes in v3: - Rebased onto pcm dma cleanup patches - A lot of cleanups for the fsl-ssi ac97 integration - Some small changes which are listed in the notes of the patches. Markus Pargmann (10): ASoC: imx-pcm-dma: DT support ASoC: imx-pcm-fiq: Introduce pcm-fiq-params ASoC: fsl: Move soc_ac97_ops from imx-ssi to fsl_ssi ASoC: fsl-ssi: Add support for imx-pcm-fiq ASoC: fsl-ssi: Use generic DMA bindings if possible ASoC: fsl-ssi: imx ac97 support ARM: imx: Export ac97 reset functions ASoC: Add phycore-ac97-dt driver ASoC: fsl: Move fsl-ssi binding doc to sound/ ASoC: fsl: Update fsl-ssi binding doc .../{powerpc/fsl/ssi.txt => sound/fsl,ssi.txt} | 10 + .../bindings/sound/phytec,phycore-ac97.txt | 14 + arch/arm/mach-imx/mach-pca100.c | 7 +- arch/arm/mach-imx/mach-pcm043.c | 7 +- sound/soc/fsl/Kconfig | 17 +- sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_ssi.c | 451 +++++++++++++++++---- sound/soc/fsl/fsl_ssi.h | 7 + sound/soc/fsl/imx-pcm-dma.c | 1 - sound/soc/fsl/imx-pcm-fiq.c | 18 +- sound/soc/fsl/imx-pcm.h | 15 +- sound/soc/fsl/imx-ssi.c | 14 +- sound/soc/fsl/imx-ssi.h | 1 + sound/soc/fsl/phycore-ac97-dt.c | 242 +++++++++++ sound/soc/fsl/phycore-ac97.c | 13 +- 15 files changed, 704 insertions(+), 115 deletions(-) rename Documentation/devicetree/bindings/{powerpc/fsl/ssi.txt => sound/fsl,ssi.txt} (88%) create mode 100644 Documentation/devicetree/bindings/sound/phytec,phycore-ac97.txt create mode 100644 sound/soc/fsl/phycore-ac97-dt.c

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[alsa-devel] Digital Mute DAI control
by Vishal Thanki 08 Jul '13

08 Jul '13

Hi All, I am new to ALSA development. I want to understand the what is the purpose of ".digital_mute" DAI control. Also, why during ".prepare" routine, the ".digital_mute" is invoked? Thanks in advance, Vishal

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[alsa-devel] ROG Xonar Phoebus support planned?
by Andreas Allacher 06 Jul '13

06 Jul '13

Hello, are there any plans to fully (and officially) support the ROG Xonar Phoebus - with good audio quality (last time there was a discussion regarding this it apparantly was possible to use the card but sound quality was bad)

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[alsa-devel] [usb-audio] VIA/Audinst HUD-mx2: One of two PCM device is available for playback
by Takashi Sakamoto 06 Jul '13

06 Jul '13

Hi, This sound card has two PCM devices. If the sound card is realized "card 0", there are "hw:0,0" and "hw:0,1". But former seems not to generate sound and latter seems to generate. This issue is reported to Ubuntu-jp and LinuxMint forum. The reporter said this happens in Fedora18 so I think this does not depends on these desktop invironment. https://forums.ubuntulinux.jp/viewtopic.php?pid=98940#p98940 http://forums.linuxmint.com/viewtopic.php?f=49&t=128478 Please see attached lsusb.log to know usb information and alsainfo.sh in detail. Audinst HUD-mx2 applies VIA Technologies chip internally. http://www.audinst.com/en/bbs/board.php?bo_table=HUDmx2&wr_id=4 The sound card has a lamp which turn on when data comes. When "hw:0,0" is used, the lamp turns on but no sound. When "hw:0,1" is used, the lamp turns on and sound. My question is: 1. does hw:0,0 seem to be available when some control are changed? 2. are there some ways to use hw:0,1 as a default? Regards Takashi Sakamoto o-takashi(a)sakamocchi.jp

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[alsa-devel] [PATCH] ASoC: Allow imx-pcm-{dma, fiq}.c to be modules
by Arnd Bergmann 05 Jul '13

05 Jul '13

This is required so we can build the imx sound support when ALSA itself is a loadable module. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Cc: alsa-devel(a)alsa-project.org Cc: Mark Brown <broonie(a)kernel.org> Cc: Fabio Estevam <fabio.estevam(a)freescale.com> diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index aa43854..805b83e 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -109,11 +109,11 @@ config SND_SOC_IMX_SSI tristate config SND_SOC_IMX_PCM_FIQ - bool + tristate select FIQ config SND_SOC_IMX_PCM_DMA - bool + tristate select SND_SOC_GENERIC_DMAENGINE_PCM config SND_SOC_IMX_AUDMUX diff --git a/sound/soc/fsl/imx-pcm.h b/sound/soc/fsl/imx-pcm.h index 67f656c..0e3619e 100644 --- a/sound/soc/fsl/imx-pcm.h +++ b/sound/soc/fsl/imx-pcm.h @@ -32,7 +32,7 @@ imx_pcm_dma_params_init_data(struct imx_dma_data *dma_data, dma_data->peripheral_type = IMX_DMATYPE_SSI; } -#ifdef CONFIG_SND_SOC_IMX_PCM_DMA +#if IS_ENABLED(CONFIG_SND_SOC_IMX_PCM_DMA) int imx_pcm_dma_init(struct platform_device *pdev); void imx_pcm_dma_exit(struct platform_device *pdev); #else @@ -46,7 +46,7 @@ static inline void imx_pcm_dma_exit(struct platform_device *pdev) } #endif -#ifdef CONFIG_SND_SOC_IMX_PCM_FIQ +#if IS_ENABLED(CONFIG_SND_SOC_IMX_PCM_FIQ) int imx_pcm_fiq_init(struct platform_device *pdev); void imx_pcm_fiq_exit(struct platform_device *pdev); #else

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[alsa-devel] [PATCH] ASoC: ak4554: add DT support
by Kuninori Morimoto 05 Jul '13

05 Jul '13

Support for loading the ak4554 codec module via devicetree. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- This is for mark/topic/ak4554 branch Documentation/devicetree/bindings/sound/ak4554.c | 11 +++++++++++ sound/soc/codecs/ak4554.c | 7 +++++++ 2 files changed, 18 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ak4554.c diff --git a/Documentation/devicetree/bindings/sound/ak4554.c b/Documentation/devicetree/bindings/sound/ak4554.c new file mode 100644 index 0000000..934fa02 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ak4554.c @@ -0,0 +1,11 @@ +AK4554 ADC/DAC + +Required properties: + + - compatible : "asahi-kasei,ak4554" + +Example: + +ak4554-adc-dac { + compatible = "asahi-kasei,ak4554"; +}; diff --git a/sound/soc/codecs/ak4554.c b/sound/soc/codecs/ak4554.c index c1a1733..6aed9c4 100644 --- a/sound/soc/codecs/ak4554.c +++ b/sound/soc/codecs/ak4554.c @@ -64,10 +64,17 @@ static int ak4554_soc_remove(struct platform_device *pdev) return 0; } +static struct of_device_id ak4554_of_match[] = { + { .compatible = "asahi-kasei,ak4554" }, + {}, +}; +MODULE_DEVICE_TABLE(of, ak4554_of_match); + static struct platform_driver ak4554_driver = { .driver = { .name = "ak4554-adc-dac", .owner = THIS_MODULE, + .of_match_table = ak4554_of_match, }, .probe = ak4554_soc_probe, .remove = ak4554_soc_remove, -- 1.7.9.5

4 5

[alsa-devel] ASoC updates for v3.11
by Mark Brown 05 Jul '13

05 Jul '13

The following changes since commit 8bb495e3f02401ee6f76d1b1d77f3ac9f079e376: Linux 3.10 (2013-06-30 15:13:29 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.11-4 for you to fetch changes up to 70a9f8797ce298544cfd07328ff59f00488bc491: Merge remote-tracking branch 'asoc/fix/wm8962' into asoc-linus (2013-07-04 18:30:03 +0100) ---------------------------------------------------------------- ASoC: Updates for v3.11 A few small fixes, all driver specific. The removal of the GPIO based pinmuxing is a bug fix, since the obsolete nodes had been removed from the DT it stopped the driver loading. ---------------------------------------------------------------- Arnaud Patard (1): ASoC: imx-sgtl5000: return E_PROBE_DEFER if ssi/codec not found Mark Brown (5): ASoC: samsung: Remove obsolete GPIO based DT pinmuxing Merge remote-tracking branch 'asoc/fix/fsl' into asoc-linus Merge remote-tracking branch 'asoc/fix/mxs' into asoc-linus Merge remote-tracking branch 'asoc/fix/samsung' into asoc-linus Merge remote-tracking branch 'asoc/fix/wm8962' into asoc-linus Sachin Kamat (1): ASoC: Samsung: Remove redundant comment Shawn Guo (2): ASoC: mxs: register saif mclk to clock framework ASoC: wm8962: fix NULL pdata pointer Documentation/devicetree/bindings/mfd/arizona.txt | 62 + .../devicetree/bindings/sound/adi,adau1701.txt | 23 + .../devicetree/bindings/sound/imx-audio-wm8962.txt | 46 + .../devicetree/bindings/sound/mxs-saif.txt | 17 +- .../bindings/sound/nvidia,tegra-audio-rt5640.txt | 71 + Documentation/devicetree/bindings/sound/rt5640.txt | 30 + .../devicetree/bindings/sound/sgtl5000.txt | 3 + .../devicetree/bindings/sound/spdif-receiver.txt | 10 + .../bindings/sound/spdif-transmitter.txt | 10 + .../devicetree/bindings/sound/ssm2518.txt | 20 + Documentation/devicetree/bindings/sound/wm8962.txt | 23 + arch/blackfin/mach-bf527/boards/ad7160eval.c | 12 - arch/blackfin/mach-bf527/boards/ezkit.c | 32 +- arch/blackfin/mach-bf533/boards/ezkit.c | 12 - arch/blackfin/mach-bf533/boards/stamp.c | 35 +- arch/blackfin/mach-bf537/boards/stamp.c | 30 +- arch/blackfin/mach-bf548/boards/ezkit.c | 28 +- arch/blackfin/mach-bf561/boards/ezkit.c | 14 +- arch/blackfin/mach-bf609/boards/ezkit.c | 2 +- drivers/mfd/arizona-core.c | 234 ++- drivers/mfd/arizona-i2c.c | 10 +- drivers/mfd/arizona-spi.c | 10 +- drivers/mfd/arizona.h | 12 + drivers/mfd/wm5102-tables.c | 9 +- drivers/mfd/wm5110-tables.c | 8 + drivers/misc/atmel-ssc.c | 8 +- include/linux/mfd/arizona/core.h | 2 + include/linux/mfd/arizona/pdata.h | 2 +- include/linux/mfd/arizona/registers.h | 47 + include/linux/mfd/wm8994/pdata.h | 5 + include/linux/mfd/wm8994/registers.h | 8 + include/linux/platform_data/ssm2518.h | 22 + include/sound/rt5640.h | 22 + include/sound/soc-dapm.h | 3 +- sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/atmel/sam9g20_wm8731.c | 9 - sound/soc/blackfin/Kconfig | 47 +- sound/soc/blackfin/Makefile | 4 - sound/soc/blackfin/bf5xx-ac97-pcm.c | 1 - sound/soc/blackfin/bf5xx-ac97-pcm.h | 26 - sound/soc/blackfin/bf5xx-ac97.c | 8 +- sound/soc/blackfin/bf5xx-ad1836.c | 19 +- sound/soc/blackfin/bf5xx-ad193x.c | 40 +- sound/soc/blackfin/bf5xx-ad1980.c | 1 - sound/soc/blackfin/bf5xx-ad73311.c | 1 - sound/soc/blackfin/bf5xx-i2s-pcm.c | 183 +- sound/soc/blackfin/bf5xx-i2s-pcm.h | 21 +- sound/soc/blackfin/bf5xx-i2s.c | 129 +- sound/soc/blackfin/bf5xx-sport.c | 10 +- sound/soc/blackfin/bf5xx-sport.h | 2 +- sound/soc/blackfin/bf5xx-ssm2602.c | 1 - sound/soc/blackfin/bf5xx-tdm-pcm.c | 345 ---- sound/soc/blackfin/bf5xx-tdm-pcm.h | 18 - sound/soc/blackfin/bf5xx-tdm.c | 328 --- sound/soc/blackfin/bf5xx-tdm.h | 23 - sound/soc/cirrus/Kconfig | 2 +- sound/soc/cirrus/ep93xx-ac97.c | 16 +- sound/soc/cirrus/ep93xx-i2s.c | 16 +- sound/soc/cirrus/ep93xx-pcm.c | 138 +- sound/soc/codecs/88pm860x-codec.c | 9 +- sound/soc/codecs/Kconfig | 18 +- sound/soc/codecs/Makefile | 14 +- sound/soc/codecs/ab8500-codec.c | 85 +- sound/soc/codecs/ab8500-codec.h | 42 +- sound/soc/codecs/adau1701.c | 69 +- sound/soc/codecs/arizona.c | 7 + sound/soc/codecs/arizona.h | 3 +- sound/soc/codecs/{dfbmcs320.c => bt-sco.c} | 37 +- sound/soc/codecs/{omap-hdmi.c => hdmi.c} | 26 +- sound/soc/codecs/jz4740.c | 2 - sound/soc/codecs/max98090.c | 24 +- sound/soc/codecs/rt5640.c | 2128 ++++++++++++++++++++ sound/soc/codecs/rt5640.h | 2092 +++++++++++++++++++ sound/soc/codecs/sgtl5000.c | 267 ++- sound/soc/codecs/sgtl5000.h | 2 +- sound/soc/codecs/sn95031.c | 2 +- sound/soc/codecs/spdif_receiver.c | 10 + .../{spdif_transciever.c => spdif_transmitter.c} | 10 + sound/soc/codecs/ssm2518.c | 856 ++++++++ sound/soc/codecs/ssm2518.h | 20 + sound/soc/codecs/wm0010.c | 10 + sound/soc/codecs/wm5102.c | 205 +- sound/soc/codecs/wm5110.c | 192 ++ sound/soc/codecs/wm8962.c | 145 +- sound/soc/codecs/wm8994.c | 188 +- sound/soc/codecs/wm8994.h | 3 + sound/soc/codecs/wm_adsp.c | 461 ++++- sound/soc/codecs/wm_adsp.h | 3 + sound/soc/davinci/Kconfig | 10 - sound/soc/davinci/Makefile | 2 - sound/soc/davinci/davinci-mcasp.c | 4 +- sound/soc/davinci/davinci-sffsdr.c | 181 -- sound/soc/dwc/designware_i2s.c | 6 +- sound/soc/fsl/Kconfig | 17 +- sound/soc/fsl/Makefile | 13 +- sound/soc/fsl/eukrea-tlv320.c | 2 +- sound/soc/fsl/fsl_ssi.c | 13 +- sound/soc/fsl/imx-audmux.c | 8 - sound/soc/fsl/imx-mc13783.c | 2 +- sound/soc/fsl/imx-pcm-dma.c | 2 + sound/soc/fsl/imx-pcm-fiq.c | 92 + sound/soc/fsl/imx-pcm.c | 145 -- sound/soc/fsl/imx-pcm.h | 10 +- sound/soc/fsl/imx-sgtl5000.c | 43 +- sound/soc/fsl/imx-ssi.c | 44 +- sound/soc/fsl/imx-ssi.h | 3 - sound/soc/fsl/imx-wm8962.c | 323 +++ sound/soc/fsl/mx27vis-aic32x4.c | 2 +- sound/soc/fsl/phycore-ac97.c | 2 +- sound/soc/fsl/wm1133-ev1.c | 2 +- sound/soc/jz4740/jz4740-i2s.c | 17 +- sound/soc/kirkwood/kirkwood-dma.c | 2 +- sound/soc/mid-x86/mfld_machine.c | 1 - sound/soc/mxs/mxs-pcm.c | 18 - sound/soc/mxs/mxs-pcm.h | 7 - sound/soc/mxs/mxs-saif.c | 72 +- sound/soc/mxs/mxs-saif.h | 1 - sound/soc/mxs/mxs-sgtl5000.c | 10 +- sound/soc/omap/Makefile | 1 - sound/soc/omap/omap-hdmi-card.c | 2 +- sound/soc/omap/omap-mcbsp.c | 2 - sound/soc/pxa/Kconfig | 20 - sound/soc/pxa/Makefile | 4 - sound/soc/pxa/mmp-pcm.c | 6 +- sound/soc/pxa/mmp-sspa.c | 2 +- sound/soc/pxa/saarb.c | 190 -- sound/soc/pxa/tavorevb3.c | 189 -- sound/soc/pxa/zylonite.c | 1 - sound/soc/samsung/Kconfig | 2 +- sound/soc/samsung/bells.c | 14 + sound/soc/samsung/i2s.c | 66 +- sound/soc/samsung/neo1973_wm8753.c | 2 +- sound/soc/samsung/s3c-i2s-v2.c | 4 +- sound/soc/samsung/smdk_wm8580pcm.c | 1 - sound/soc/samsung/smdk_wm8994pcm.c | 1 - sound/soc/sh/fsi.c | 10 +- sound/soc/soc-core.c | 106 +- sound/soc/soc-dapm.c | 110 +- sound/soc/soc-pcm.c | 91 +- sound/soc/soc-utils.c | 13 +- sound/soc/spear/Kconfig | 9 + sound/soc/spear/Makefile | 8 + sound/soc/spear/spdif_in.c | 31 +- sound/soc/spear/spdif_out.c | 43 +- sound/soc/spear/spear_pcm.c | 152 +- sound/soc/tegra/Kconfig | 10 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra30_ahub.c | 25 + sound/soc/tegra/tegra30_i2s.c | 26 + sound/soc/tegra/tegra_asoc_utils.c | 23 +- sound/soc/tegra/tegra_rt5640.c | 257 +++ sound/soc/ux500/mop500.c | 2 +- sound/soc/ux500/mop500_ab8500.c | 62 +- sound/soc/ux500/ux500_msp_dai.c | 11 +- sound/soc/ux500/ux500_msp_dai.h | 4 - sound/soc/ux500/ux500_msp_i2s.c | 88 +- sound/soc/ux500/ux500_msp_i2s.h | 74 +- sound/soc/ux500/ux500_pcm.c | 30 + 159 files changed, 8829 insertions(+), 3121 deletions(-) create mode 100644 Documentation/devicetree/bindings/mfd/arizona.txt create mode 100644 Documentation/devicetree/bindings/sound/adi,adau1701.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt create mode 100644 Documentation/devicetree/bindings/sound/rt5640.txt create mode 100644 Documentation/devicetree/bindings/sound/spdif-receiver.txt create mode 100644 Documentation/devicetree/bindings/sound/spdif-transmitter.txt create mode 100644 Documentation/devicetree/bindings/sound/ssm2518.txt create mode 100644 include/linux/platform_data/ssm2518.h create mode 100644 include/sound/rt5640.h delete mode 100644 sound/soc/blackfin/bf5xx-ac97-pcm.h delete mode 100644 sound/soc/blackfin/bf5xx-tdm-pcm.c delete mode 100644 sound/soc/blackfin/bf5xx-tdm-pcm.h delete mode 100644 sound/soc/blackfin/bf5xx-tdm.c delete mode 100644 sound/soc/blackfin/bf5xx-tdm.h rename sound/soc/codecs/{dfbmcs320.c => bt-sco.c} (53%) rename sound/soc/codecs/{omap-hdmi.c => hdmi.c} (69%) create mode 100644 sound/soc/codecs/rt5640.c create mode 100644 sound/soc/codecs/rt5640.h rename sound/soc/codecs/{spdif_transciever.c => spdif_transmitter.c} (88%) create mode 100644 sound/soc/codecs/ssm2518.c create mode 100644 sound/soc/codecs/ssm2518.h delete mode 100644 sound/soc/davinci/davinci-sffsdr.c delete mode 100644 sound/soc/fsl/imx-pcm.c create mode 100644 sound/soc/fsl/imx-wm8962.c delete mode 100644 sound/soc/pxa/saarb.c delete mode 100644 sound/soc/pxa/tavorevb3.c create mode 100644 sound/soc/spear/Kconfig create mode 100644 sound/soc/spear/Makefile create mode 100644 sound/soc/tegra/tegra_rt5640.c

2 1

[alsa-devel] [PATCHv2 00/31] HDSPM: A lot
by Adrian Knoth 05 Jul '13

05 Jul '13

Hi! Here's a revised series of the two patchsets I've sent earlier this week. Features by card: AIO: - Support Analog Expansion Boards - Support all card settings like gain, coax/optical switching etc. - Fixed S/PDIF Sync status and frequency reporting RayDAT: - Controls for S/PDIF and WordClock AES(32): - Support for TCO module (originally by Martin Dausel) All cards: - Code cleanups - Documentation Takashi: As agreed, I've split the big "TCO patch" into smaller chunks to ease reviewing. Adrian Knoth (30): ALSA: hdspm - Add missing defines for RME AIO and RayDAT ALSA: hdspm - Introduce hdspm_is_raydat_or_aio() ALSA: hdspm - Augment HDSPM_TOGGLE_SETTING for AIO/RayDAT ALSA: hdspm - Drop duplicate code in hdspm_set_system_clock_mode() ALSA: hdspm - Add S/PDIF and WCK48 controls for RME RayDAT ALSA: hdspm - Add S/PDIF, XLR, WCK48 and ADAT-in controls for RME AIO cards ALSA: hdspm - Refactor ENUMERATED_CTL_INFO into function ALSA: hdspm - Introduce generic AIO tristate control ALSA: hdspm - Enable AD/DA/PH gains and S/PDIF-Input select on AIO ALSA: hdspm - Add support for AEBs on RME AIO ALSA: hdspm - Fix S/PDIF Sync status and frequency on RME AIO ALSA: hdspm - Create TCO readout function ALSA: hdspm - AES32: Fix TCO sync check reporting ALSA: hdspm - Cosmetics, no real change ALSA: hdspm - AIO: Drop superfluous HDSPM_AUTOSYNC_REF ALSA: hdspm - AES32: Add TCO and Sync-In text entries ALSA: hdspm - Introduce hdspm_get_aes_sample_rate() ALSA: hdspm - Add prototype declarations ALSA: hdspm - Enable AES32 in hdspm_get_wc_sample_rate ALSA: hdspm - Enable AES32 in hdspm_get_tco_sample_rate ALSA: hdspm - AES32: Ignore float/int format bit ALSA: hdspm - AES32: Enable TCO input in hdspm_external_sample_rate() ALSA: hdspm - AES32: Enable TCO/Sync-In in snd_hdspm_put_sync_ref() ALSA: hdspm - AES32: Include TCO and Sync-In in proc output ALSA: hdspm - Introduce hdspm_external_rate_to_enum() helper function ALSA: hdspm - AES32: Report external sample rate to userspace ALSA: hdspm - AES32: Enable TCO support ALSA: hdspm - Use snd_ctl_enum_info for most text arrays ALSA: hdspm - Use snd_ctl_enum_info() for texts_autosync ALSA: hdspm - Use snd_ctl_enum_info() in snd_hdspm_info_autosync_ref Martin Dausel (1): ALSA: hdspm - Added some comments and control register documentation sound/pci/rme9652/hdspm.c | 765 ++++++++++++++++++++++++++++++++-------------- 1 file changed, 538 insertions(+), 227 deletions(-) -- 1.8.3.1

2 32

[alsa-devel] [PATCH 1/3] ASoC: sglt5000: Provide the reg_stride field
by Fabio Estevam 05 Jul '13

05 Jul '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> sgtl5000 has 16-bit registers, so we need to tell regmap to use a reg_stride of two bytes. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/sgtl5000.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index d441559..7c99f3c 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1470,6 +1470,7 @@ static struct snd_soc_codec_driver sgtl5000_driver = { static const struct regmap_config sgtl5000_regmap = { .reg_bits = 16, .val_bits = 16, + .reg_stride = 2, .max_register = SGTL5000_MAX_REG_OFFSET, .volatile_reg = sgtl5000_volatile, -- 1.8.1.2

2 3

[alsa-devel] [PATCH] drm/i915: switch disable_power_well default value to 1
by Paulo Zanoni 04 Jul '13

04 Jul '13

From: Paulo Zanoni <paulo.r.zanoni(a)intel.com> Now that the audio driver is using our power well API, everything should be working correctly, so let's give it a try. Signed-off-by: Paulo Zanoni <paulo.r.zanoni(a)intel.com> --- drivers/gpu/drm/i915/i915_drv.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/gpu/drm/i915/i915_drv.c b/drivers/gpu/drm/i915/i915_drv.c index 33cb973..e6dc81c 100644 --- a/drivers/gpu/drm/i915/i915_drv.c +++ b/drivers/gpu/drm/i915/i915_drv.c @@ -123,10 +123,10 @@ module_param_named(preliminary_hw_support, i915_preliminary_hw_support, int, 060 MODULE_PARM_DESC(preliminary_hw_support, "Enable preliminary hardware support. (default: false)"); -int i915_disable_power_well __read_mostly = 0; +int i915_disable_power_well __read_mostly = 1; module_param_named(disable_power_well, i915_disable_power_well, int, 0600); MODULE_PARM_DESC(disable_power_well, - "Disable the power well when possible (default: false)"); + "Disable the power well when possible (default: true)"); int i915_enable_ips __read_mostly = 1; module_param_named(enable_ips, i915_enable_ips, int, 0600); -- 1.8.1.2

2 1

[alsa-devel] [PATCH] ASoC: arizona: Add default case to silence build warning
by Charles Keepax 04 Jul '13

04 Jul '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index eb5c80d..bcf739d 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -233,6 +233,9 @@ int arizona_init_gpio(struct snd_soc_codec *codec) switch (arizona->type) { case WM5110: snd_soc_dapm_disable_pin(&codec->dapm, "DRC2 Signal Activity"); + break; + default: + break; } snd_soc_dapm_disable_pin(&codec->dapm, "DRC1 Signal Activity"); -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ASoC: wm8962: fix NULL pdata pointer
by Shawn Guo 04 Jul '13

04 Jul '13

There is an error in merge commit 384b834 on conflict resolution which causes the following NULL pdata pointer bug. wm8962 0-001a: customer id 0 revision D Unable to handle kernel NULL pointer dereference at virtual address 00000004 pgd = 80004000 [00000004] *pgd=00000000 Internal error: Oops: 5 [#1] SMP ARM Modules linked in: CPU: 0 PID: 1 Comm: swapper/0 Not tainted 3.10.0+ #1 task: bf870000 ti: bf874000 task.ti: bf874000 PC is at wm8962_probe+0x134/0x6c8 LR is at regmap_unlock_mutex+0x10/0x14 pc : [<80452100>] lr : [<80304cf4>] psr: a0000113 sp : bf875c98 ip : 00000000 fp : bf875cd4 r10: 00000000 r9 : bfb1830c r8 : 80779bc4 r7 : 00000000 r6 : 00000001 r5 : bfbac010 r4 : bfb33e00 r3 : 80304ce4 r2 : 00000000 r1 : 00000001 r0 : fffffffb Flags: NzCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment kernel Control: 10c53c7d Table: 1000404a DAC: 00000017 Process swapper/0 (pid: 1, stack limit = 0xbf874238) Stack: (0xbf875c98 to 0xbf876000) ... Fix the error by assigning pdata a correct pointer. Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- sound/soc/codecs/wm8962.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index b1dc7d4..e2de9ec 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3377,7 +3377,7 @@ static int wm8962_probe(struct snd_soc_codec *codec) { int ret; struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); - struct wm8962_pdata *pdata = dev_get_platdata(codec->dev); + struct wm8962_pdata *pdata = &wm8962->pdata; int i, trigger, irq_pol; bool dmicclk, dmicdat; -- 1.7.9.5

2 1

[alsa-devel] [PATCH 00/11] HDSPM: RME AIO/RayDAT extensions and cleanups
by Adrian Knoth 04 Jul '13

04 Jul '13

Hi! Here's a series of patches that most importantly add support for AIO's Analog Expansion Boards (AEBs) and all the missing controls to adjust card settings (output levels, optical/coaxial etc.). I also refactored a couple of functions to clean up the code and fixed the S/PDIF sync/frequency reporting. Cheers Adrian Knoth (11): ALSA: hdspm - Add missing defines for RME AIO and RayDAT ALSA: hdspm - Introduce hdspm_is_raydat_or_aio() ALSA: hdspm - Augment HDSPM_TOGGLE_SETTING for AIO/RayDAT ALSA: hdspm - Drop duplicate code in hdspm_set_system_clock_mode() ALSA: hdspm - Add S/PDIF and WCK48 controls for RME RayDAT ALSA: hdspm - Add S/PDIF, XLR, WCK48 and ADAT-in controls for RME AIO cards ALSA: hdspm - Refactor ENUMERATED_CTL_INFO into function ALSA: hdspm - Introduce generic AIO tristate control ALSA: hdspm - Enable AD/DA/PH gains and S/PDIF-Input select on AIO ALSA: hdspm - Add support for AEBs on RME AIO ALSA: hdspm - Fix S/PDIF Sync status and frequency on RME AIO sound/pci/rme9652/hdspm.c | 254 ++++++++++++++++++++++++++++++++++++---------- 1 file changed, 199 insertions(+), 55 deletions(-) -- 1.8.3.1

2 13

[alsa-devel] [RFC] [patch 1/1] imx-sgtl5000: return E_PROBE_DEFER if ssi/codec not found
by Arnaud Patard 04 Jul '13

04 Jul '13

If the ssi or codec drivers are not loaded (for instance, because spi or i2c bus drivers are not loaded), returning -EINVAL will for people to unload and then reload the module to get sound working. Returning E_PROBE_DEFER will mitigate this. Signed-off-by: Arnaud Patard <arnaud.patard(a)rtp-net.org> Index: linux/sound/soc/fsl/imx-sgtl5000.c =================================================================== --- linux.orig/sound/soc/fsl/imx-sgtl5000.c 2013-05-16 09:33:01.000000000 +0200 +++ linux/sound/soc/fsl/imx-sgtl5000.c 2013-05-16 09:35:14.000000000 +0200 @@ -113,13 +113,13 @@ static int imx_sgtl5000_probe(struct pla ssi_pdev = of_find_device_by_node(ssi_np); if (!ssi_pdev) { dev_err(&pdev->dev, "failed to find SSI platform device\n"); - ret = -EINVAL; + ret = -EPROBE_DEFER; goto fail; } codec_dev = of_find_i2c_device_by_node(codec_np); if (!codec_dev) { dev_err(&pdev->dev, "failed to find codec platform device\n"); - return -EINVAL; + return -EPROBE_DEFER; } data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL);

3 4

[alsa-devel] [PATCH 10/11] ALSA: hdspm - Add support for AEBs on RME AIO
by Adrian Knoth 04 Jul '13

04 Jul '13

AIO cards allow to use AEB (Analogue Expansion Boards) to add four input and/or output channels. This patch adds the necessary code to detect and enable the additional I/O channels. Signed-off-by: Adrian Knoth <adi(a)drcomp.erfurt.thur.de> diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 111713c..585397b 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -648,7 +648,8 @@ static char *texts_ports_aio_in_ss[] = { "AES.L", "AES.R", "SPDIF.L", "SPDIF.R", "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4", "ADAT.5", "ADAT.6", - "ADAT.7", "ADAT.8" + "ADAT.7", "ADAT.8", + "AEB.1", "AEB.2", "AEB.3", "AEB.4" }; static char *texts_ports_aio_out_ss[] = { @@ -657,14 +658,16 @@ static char *texts_ports_aio_out_ss[] = { "SPDIF.L", "SPDIF.R", "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4", "ADAT.5", "ADAT.6", "ADAT.7", "ADAT.8", - "Phone.L", "Phone.R" + "Phone.L", "Phone.R", + "AEB.1", "AEB.2", "AEB.3", "AEB.4" }; static char *texts_ports_aio_in_ds[] = { "Analogue.L", "Analogue.R", "AES.L", "AES.R", "SPDIF.L", "SPDIF.R", - "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4" + "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4", + "AEB.1", "AEB.2", "AEB.3", "AEB.4" }; static char *texts_ports_aio_out_ds[] = { @@ -672,14 +675,16 @@ static char *texts_ports_aio_out_ds[] = { "AES.L", "AES.R", "SPDIF.L", "SPDIF.R", "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4", - "Phone.L", "Phone.R" + "Phone.L", "Phone.R", + "AEB.1", "AEB.2", "AEB.3", "AEB.4" }; static char *texts_ports_aio_in_qs[] = { "Analogue.L", "Analogue.R", "AES.L", "AES.R", "SPDIF.L", "SPDIF.R", - "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4" + "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4", + "AEB.1", "AEB.2", "AEB.3", "AEB.4" }; static char *texts_ports_aio_out_qs[] = { @@ -687,7 +692,8 @@ static char *texts_ports_aio_out_qs[] = { "AES.L", "AES.R", "SPDIF.L", "SPDIF.R", "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4", - "Phone.L", "Phone.R" + "Phone.L", "Phone.R", + "AEB.1", "AEB.2", "AEB.3", "AEB.4" }; static char *texts_ports_aes32[] = { @@ -764,8 +770,8 @@ static char channel_map_aio_in_ss[HDSPM_MAX_CHANNELS] = { 8, 9, /* aes in, */ 10, 11, /* spdif in */ 12, 13, 14, 15, 16, 17, 18, 19, /* ADAT in */ - -1, -1, - -1, -1, -1, -1, -1, -1, -1, -1, + 2, 3, 4, 5, /* AEB */ + -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, @@ -779,7 +785,8 @@ static char channel_map_aio_out_ss[HDSPM_MAX_CHANNELS] = { 10, 11, /* spdif out */ 12, 13, 14, 15, 16, 17, 18, 19, /* ADAT out */ 6, 7, /* phone out */ - -1, -1, -1, -1, -1, -1, -1, -1, + 2, 3, 4, 5, /* AEB */ + -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, @@ -792,7 +799,8 @@ static char channel_map_aio_in_ds[HDSPM_MAX_CHANNELS] = { 8, 9, /* aes in */ 10, 11, /* spdif in */ 12, 14, 16, 18, /* adat in */ - -1, -1, -1, -1, -1, -1, + 2, 3, 4, 5, /* AEB */ + -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, @@ -807,7 +815,7 @@ static char channel_map_aio_out_ds[HDSPM_MAX_CHANNELS] = { 10, 11, /* spdif out */ 12, 14, 16, 18, /* adat out */ 6, 7, /* phone out */ - -1, -1, -1, -1, + 2, 3, 4, 5, /* AEB */ -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, @@ -821,7 +829,8 @@ static char channel_map_aio_in_qs[HDSPM_MAX_CHANNELS] = { 8, 9, /* aes in */ 10, 11, /* spdif in */ 12, 16, /* adat in */ - -1, -1, -1, -1, -1, -1, -1, -1, + 2, 3, 4, 5, /* AEB */ + -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, @@ -836,7 +845,8 @@ static char channel_map_aio_out_qs[HDSPM_MAX_CHANNELS] = { 10, 11, /* spdif out */ 12, 16, /* adat out */ 6, 7, /* phone out */ - -1, -1, -1, -1, -1, -1, + 2, 3, 4, 5, /* AEB */ + -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, @@ -6602,10 +6612,6 @@ static int snd_hdspm_create(struct snd_card *card, break; case AIO: - if (0 == (hdspm_read(hdspm, HDSPM_statusRegister2) & HDSPM_s2_AEBI_D)) { - snd_printk(KERN_INFO "HDSPM: AEB input board found, but not supported\n"); - } - hdspm->ss_in_channels = AIO_IN_SS_CHANNELS; hdspm->ds_in_channels = AIO_IN_DS_CHANNELS; hdspm->qs_in_channels = AIO_IN_QS_CHANNELS; @@ -6613,6 +6619,20 @@ static int snd_hdspm_create(struct snd_card *card, hdspm->ds_out_channels = AIO_OUT_DS_CHANNELS; hdspm->qs_out_channels = AIO_OUT_QS_CHANNELS; + if (0 == (hdspm_read(hdspm, HDSPM_statusRegister2) & HDSPM_s2_AEBI_D)) { + snd_printk(KERN_INFO "HDSPM: AEB input board found\n"); + hdspm->ss_in_channels += 4; + hdspm->ds_in_channels += 4; + hdspm->qs_in_channels += 4; + } + + if (0 == (hdspm_read(hdspm, HDSPM_statusRegister2) & HDSPM_s2_AEBO_D)) { + snd_printk(KERN_INFO "HDSPM: AEB output board found\n"); + hdspm->ss_out_channels += 4; + hdspm->ds_out_channels += 4; + hdspm->qs_out_channels += 4; + } + hdspm->channel_map_out_ss = channel_map_aio_out_ss; hdspm->channel_map_out_ds = channel_map_aio_out_ds; hdspm->channel_map_out_qs = channel_map_aio_out_qs; -- 1.8.3.1

1 1

[alsa-devel] [PATCH 1/1] ASoC: Samsung: Remove redundant comment
by Sachin Kamat 04 Jul '13

04 Jul '13

There is a typo in the filename (i2c mentioned instead of i2s). However, this is a redundant piece of information. Delete it altogether. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/samsung/s3c-i2s-v2.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c index 20e98d1..e5e81b1 100644 --- a/sound/soc/samsung/s3c-i2s-v2.c +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -1,6 +1,4 @@ -/* sound/soc/samsung/s3c-i2c-v2.c - * - * ALSA Soc Audio Layer - I2S core for newer Samsung SoCs. +/* ALSA Soc Audio Layer - I2S core for newer Samsung SoCs. * * Copyright (c) 2006 Wolfson Microelectronics PLC. * Graeme Gregory graeme.gregory(a)wolfsonmicro.com -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: add ak4554 driver
by Kuninori Morimoto 04 Jul '13

04 Jul '13

ak4554 is very simple DA/AD converter which has no setting register. But, playback format is SND_SOC_DAIFMT_RIGHT_J, and, capture format is SND_SOC_DAIFMT_LEFT_J on same bit clock, LR clock. Because of that, snd_soc_dai_driver consists from "playback only" and "capture only" here, and it has software base symmetric_rates check. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4554.c | 171 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 176 insertions(+) create mode 100644 sound/soc/codecs/ak4554.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2f45f00..b4ec4ea 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -196,6 +196,9 @@ config SND_SOC_AK4104 config SND_SOC_AK4535 tristate +config SND_SOC_AK4554 + tristate + config SND_SOC_AK4641 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index b9e41c9..e601a47 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -11,6 +11,7 @@ snd-soc-adav80x-objs := adav80x.o snd-soc-ads117x-objs := ads117x.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o +snd-soc-ak4554-objs := ak4554.o snd-soc-ak4641-objs := ak4641.o snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o @@ -136,6 +137,7 @@ obj-$(CONFIG_SND_SOC_ADAV80X) += snd-soc-adav80x.o obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o +obj-$(CONFIG_SND_SOC_AK4554) += snd-soc-ak4554.o obj-$(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 diff --git a/sound/soc/codecs/ak4554.c b/sound/soc/codecs/ak4554.c new file mode 100644 index 0000000..277f216 --- /dev/null +++ b/sound/soc/codecs/ak4554.c @@ -0,0 +1,171 @@ +/* + * ak4554.c + * + * Copyright (C) 2013 Renesas Solutions Corp. + * Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/module.h> +#include <sound/soc.h> + +/* + * ak4554 is very simple DA/AD converter which has no setting register. + * But, playback format is SND_SOC_DAIFMT_RIGHT_J, + * and, capture format is SND_SOC_DAIFMT_LEFT_J + * on same bit clock, LR clock. + * Because of that, snd_soc_dai_driver consists from + * "playback only" and "capture only" here. + */ + +struct ak4554_priv { + int rate; + int usrcnt; +}; + +static int ak4554_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + int dai_is_play = (dai->id == 0); + int fmt_is_play = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBS_CFM: + case SND_SOC_DAIFMT_CBM_CFS: + dev_err(dai->dev, "can't be clock master\n"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + fmt_is_play = 1; + break; + case SND_SOC_DAIFMT_LEFT_J: + fmt_is_play = 0; + break; + } + + if (dai_is_play != fmt_is_play) { + dev_err(dai->dev, "format error\n"); + return -EINVAL; + } + + return 0; +} + +static int ak4554_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct ak4554_priv *priv = dev_get_drvdata(dai->dev); + + if ((priv->usrcnt > 0) && + (priv->rate != params_rate(params))) { + dev_err(dai->dev, "asymmetric rate\n"); + return -EIO; + } + + priv->usrcnt++; + priv->rate = params_rate(params); + + return 0; +} + +static void ak4554_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct ak4554_priv *priv = dev_get_drvdata(dai->dev); + + priv->usrcnt--; + + if (priv->usrcnt <= 0) + priv->rate = 0; +} + +static const struct snd_soc_dai_ops ak4554_dai_ops = { + .set_fmt = ak4554_dai_set_fmt, + .hw_params = ak4554_dai_hw_params, + .shutdown = ak4554_dai_shutdown, +}; + +struct snd_soc_dai_driver ak4554_dai[] = { + { + .name = "ak4554-hifi-playback", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &ak4554_dai_ops, + }, { + .name = "ak4554-hifi-capture", + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &ak4554_dai_ops, + } +}; +EXPORT_SYMBOL_GPL(ak4554_dai); + +static int ak4554_probe(struct snd_soc_codec *codec) +{ + dev_info(codec->dev, "probed\n"); + return 0; +} + +static int ak4554_remove(struct snd_soc_codec *codec) +{ + dev_info(codec->dev, "removed\n"); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_ak4554 = { + .probe = ak4554_probe, + .remove = ak4554_remove, +}; + +static int ak4554_soc_probe(struct platform_device *pdev) +{ + struct ak4554_priv *priv; + + priv = devm_kzalloc(&pdev->dev, sizeof(struct ak4554_priv), + GFP_KERNEL); + if (!priv) + return -ENOMEM; + + dev_set_drvdata(&pdev->dev, priv); + + return snd_soc_register_codec(&pdev->dev, + &soc_codec_dev_ak4554, + ak4554_dai, ARRAY_SIZE(ak4554_dai)); +} + +static int ak4554_soc_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +static struct platform_driver ak4554_driver = { + .driver = { + .name = "ak4554-adc-dac", + .owner = THIS_MODULE, + }, + .probe = ak4554_soc_probe, + .remove = ak4554_soc_remove, +}; +module_platform_driver(ak4554_driver); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("SoC AK4554 driver"); +MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>"); -- 1.7.9.5

3 8

[alsa-devel] [PATCH 0/2] HDSPM - Documentation and TCO on AES(32)
by Adrian Knoth 04 Jul '13

04 Jul '13

Hi! Here are two patches that I've received from Martin. The first only adds comments to the driver to document the various register settings. Might be useful for other developers. The second patch adds TCO support on AES(32) cards. I took the liberty to polish Martin's work to reflect the kernel coding style. It still contains plenty of long-line warnings from checkpatch.pl, but I could live with this. There are also a few (five?) whitespace fixes and a typo fix which are technically not related to TCO support. I could manually separate them into two commits, but maybe it's acceptable as it is? Cheers Adrian Knoth (1): ALSA: hdspm - Add TCO support for RME AES(32) Martin Dausel (1): ALSA: hdspm - Added some comments and control register documentation sound/pci/rme9652/hdspm.c | 438 +++++++++++++++++++++++++++++++++------------- 1 file changed, 317 insertions(+), 121 deletions(-) -- 1.8.3.1

2 3

[alsa-devel] [PATCH v3 1/2] mfd: arizona: Add GPIO control register bit definitions
by Charles Keepax 04 Jul '13

04 Jul '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- include/linux/mfd/arizona/gpio.h | 96 ++++++++++++++++++++++++++++++++++++++ 1 files changed, 96 insertions(+), 0 deletions(-) create mode 100644 include/linux/mfd/arizona/gpio.h diff --git a/include/linux/mfd/arizona/gpio.h b/include/linux/mfd/arizona/gpio.h new file mode 100644 index 0000000..d2146bb --- /dev/null +++ b/include/linux/mfd/arizona/gpio.h @@ -0,0 +1,96 @@ +/* + * GPIO configuration for Arizona devices + * + * Copyright 2013 Wolfson Microelectronics. PLC. + * + * Author: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _ARIZONA_GPIO_H +#define _ARIZONA_GPIO_H + +#define ARIZONA_GP_FN_TXLRCLK 0x00 +#define ARIZONA_GP_FN_GPIO 0x01 +#define ARIZONA_GP_FN_IRQ1 0x02 +#define ARIZONA_GP_FN_IRQ2 0x03 +#define ARIZONA_GP_FN_OPCLK 0x04 +#define ARIZONA_GP_FN_FLL1_OUT 0x05 +#define ARIZONA_GP_FN_FLL2_OUT 0x06 +#define ARIZONA_GP_FN_PWM1 0x08 +#define ARIZONA_GP_FN_PWM2 0x09 +#define ARIZONA_GP_FN_SYSCLK_UNDERCLOCKED 0x0A +#define ARIZONA_GP_FN_ASYNCCLK_UNDERCLOCKED 0x0B +#define ARIZONA_GP_FN_FLL1_LOCK 0x0C +#define ARIZONA_GP_FN_FLL2_LOCK 0x0D +#define ARIZONA_GP_FN_FLL1_CLOCK_OK 0x0F +#define ARIZONA_GP_FN_FLL2_CLOCK_OK 0x10 +#define ARIZONA_GP_FN_HEADPHONE_DET 0x12 +#define ARIZONA_GP_FN_MIC_DET 0x13 +#define ARIZONA_GP_FN_WSEQ_STATUS 0x15 +#define ARIZONA_GP_FN_CIF_ADDRESS_ERROR 0x16 +#define ARIZONA_GP_FN_ASRC1_LOCK 0x1A +#define ARIZONA_GP_FN_ASRC2_LOCK 0x1B +#define ARIZONA_GP_FN_ASRC_CONFIG_ERROR 0x1C +#define ARIZONA_GP_FN_DRC1_SIGNAL_DETECT 0x1D +#define ARIZONA_GP_FN_DRC1_ANTICLIP 0x1E +#define ARIZONA_GP_FN_DRC1_DECAY 0x1F +#define ARIZONA_GP_FN_DRC1_NOISE 0x20 +#define ARIZONA_GP_FN_DRC1_QUICK_RELEASE 0x21 +#define ARIZONA_GP_FN_DRC2_SIGNAL_DETECT 0x22 +#define ARIZONA_GP_FN_DRC2_ANTICLIP 0x23 +#define ARIZONA_GP_FN_DRC2_DECAY 0x24 +#define ARIZONA_GP_FN_DRC2_NOISE 0x25 +#define ARIZONA_GP_FN_DRC2_QUICK_RELEASE 0x26 +#define ARIZONA_GP_FN_MIXER_DROPPED_SAMPLE 0x27 +#define ARIZONA_GP_FN_AIF1_CONFIG_ERROR 0x28 +#define ARIZONA_GP_FN_AIF2_CONFIG_ERROR 0x29 +#define ARIZONA_GP_FN_AIF3_CONFIG_ERROR 0x2A +#define ARIZONA_GP_FN_SPK_TEMP_SHUTDOWN 0x2B +#define ARIZONA_GP_FN_SPK_TEMP_WARNING 0x2C +#define ARIZONA_GP_FN_UNDERCLOCKED 0x2D +#define ARIZONA_GP_FN_OVERCLOCKED 0x2E +#define ARIZONA_GP_FN_DSP_IRQ1 0x35 +#define ARIZONA_GP_FN_DSP_IRQ2 0x36 +#define ARIZONA_GP_FN_ASYNC_OPCLK 0x3D +#define ARIZONA_GP_FN_BOOT_DONE 0x44 +#define ARIZONA_GP_FN_DSP1_RAM_READY 0x45 +#define ARIZONA_GP_FN_SYSCLK_ENA_STATUS 0x4B +#define ARIZONA_GP_FN_ASYNCCLK_ENA_STATUS 0x4C + +#define ARIZONA_GPN_DIR 0x8000 /* GPN_DIR */ +#define ARIZONA_GPN_DIR_MASK 0x8000 /* GPN_DIR */ +#define ARIZONA_GPN_DIR_SHIFT 15 /* GPN_DIR */ +#define ARIZONA_GPN_DIR_WIDTH 1 /* GPN_DIR */ +#define ARIZONA_GPN_PU 0x4000 /* GPN_PU */ +#define ARIZONA_GPN_PU_MASK 0x4000 /* GPN_PU */ +#define ARIZONA_GPN_PU_SHIFT 14 /* GPN_PU */ +#define ARIZONA_GPN_PU_WIDTH 1 /* GPN_PU */ +#define ARIZONA_GPN_PD 0x2000 /* GPN_PD */ +#define ARIZONA_GPN_PD_MASK 0x2000 /* GPN_PD */ +#define ARIZONA_GPN_PD_SHIFT 13 /* GPN_PD */ +#define ARIZONA_GPN_PD_WIDTH 1 /* GPN_PD */ +#define ARIZONA_GPN_LVL 0x0800 /* GPN_LVL */ +#define ARIZONA_GPN_LVL_MASK 0x0800 /* GPN_LVL */ +#define ARIZONA_GPN_LVL_SHIFT 11 /* GPN_LVL */ +#define ARIZONA_GPN_LVL_WIDTH 1 /* GPN_LVL */ +#define ARIZONA_GPN_POL 0x0400 /* GPN_POL */ +#define ARIZONA_GPN_POL_MASK 0x0400 /* GPN_POL */ +#define ARIZONA_GPN_POL_SHIFT 10 /* GPN_POL */ +#define ARIZONA_GPN_POL_WIDTH 1 /* GPN_POL */ +#define ARIZONA_GPN_OP_CFG 0x0200 /* GPN_OP_CFG */ +#define ARIZONA_GPN_OP_CFG_MASK 0x0200 /* GPN_OP_CFG */ +#define ARIZONA_GPN_OP_CFG_SHIFT 9 /* GPN_OP_CFG */ +#define ARIZONA_GPN_OP_CFG_WIDTH 1 /* GPN_OP_CFG */ +#define ARIZONA_GPN_DB 0x0100 /* GPN_DB */ +#define ARIZONA_GPN_DB_MASK 0x0100 /* GPN_DB */ +#define ARIZONA_GPN_DB_SHIFT 8 /* GPN_DB */ +#define ARIZONA_GPN_DB_WIDTH 1 /* GPN_DB */ +#define ARIZONA_GPN_FN_MASK 0x007F /* GPN_DB */ +#define ARIZONA_GPN_FN_SHIFT 0 /* GPN_DB */ +#define ARIZONA_GPN_FN_WIDTH 7 /* GPN_DB */ + +#endif -- 1.7.2.5

2 6

[alsa-devel] ASoC: wm8962: NULL pdata pointer
by Shawn Guo 04 Jul '13

04 Jul '13

Mark, I'm running next-20130703 and seeing the following NULL pointer bug. Unable to handle kernel NULL pointer dereference at virtual address 00000004 pgd = 80004000 [00000004] *pgd=00000000 Internal error: Oops: 5 [#1] SMP ARM Modules linked in: CPU: 0 PID: 1 Comm: swapper/0 Not tainted 3.10.0-next-20130703 #6 task: bf870000 ti: bf874000 task.ti: bf874000 PC is at wm8962_probe+0x140/0x6b4 ... It seems to me that it's caused by a merge error in 384b834. And the change below can remove the problem. Shawn ---8<---- diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index b1dc7d4..e2de9ec 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3377,7 +3377,7 @@ static int wm8962_probe(struct snd_soc_codec *codec) { int ret; struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); - struct wm8962_pdata *pdata = dev_get_platdata(codec->dev); + struct wm8962_pdata *pdata = &wm8962->pdata; int i, trigger, irq_pol; bool dmicclk, dmicdat;

3 2

[alsa-devel] [PATCH] Add -lm to test programs to fix build failure.
by Jordi Mallach 04 Jul '13

04 Jul '13

Add -lm to test programs that use sin(), to fix a build failure with recent binutils. Signed-off-by: Jordi Mallach <jordi(a)debian.org> --- test/Makefile.am | 2 ++ 1 file changed, 2 insertions(+) diff --git a/test/Makefile.am b/test/Makefile.am index 8705402..c1896f8 100644 --- a/test/Makefile.am +++ b/test/Makefile.am @@ -7,8 +7,10 @@ check_PROGRAMS=control pcm pcm_min latency seq \ control_LDADD=../src/libasound.la pcm_LDADD=../src/libasound.la +pcm_LDFLAGS= -lm pcm_min_LDADD=../src/libasound.la latency_LDADD=../src/libasound.la +latency_LDFLAGS= -lm seq_LDADD=../src/libasound.la playmidi1_LDADD=../src/libasound.la timer_LDADD=../src/libasound.la -- 1.8.3.1

1 1

[alsa-devel] [PATCH v2 1/2] mfd: arizona: Add GPIO control register bit definitions
by Charles Keepax 03 Jul '13

03 Jul '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- include/linux/mfd/arizona/gpio.h | 96 ++++++++++++++++++++++++++++++++++++++ 1 files changed, 96 insertions(+), 0 deletions(-) create mode 100644 include/linux/mfd/arizona/gpio.h diff --git a/include/linux/mfd/arizona/gpio.h b/include/linux/mfd/arizona/gpio.h new file mode 100644 index 0000000..d2146bb --- /dev/null +++ b/include/linux/mfd/arizona/gpio.h @@ -0,0 +1,96 @@ +/* + * GPIO configuration for Arizona devices + * + * Copyright 2013 Wolfson Microelectronics. PLC. + * + * Author: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _ARIZONA_GPIO_H +#define _ARIZONA_GPIO_H + +#define ARIZONA_GP_FN_TXLRCLK 0x00 +#define ARIZONA_GP_FN_GPIO 0x01 +#define ARIZONA_GP_FN_IRQ1 0x02 +#define ARIZONA_GP_FN_IRQ2 0x03 +#define ARIZONA_GP_FN_OPCLK 0x04 +#define ARIZONA_GP_FN_FLL1_OUT 0x05 +#define ARIZONA_GP_FN_FLL2_OUT 0x06 +#define ARIZONA_GP_FN_PWM1 0x08 +#define ARIZONA_GP_FN_PWM2 0x09 +#define ARIZONA_GP_FN_SYSCLK_UNDERCLOCKED 0x0A +#define ARIZONA_GP_FN_ASYNCCLK_UNDERCLOCKED 0x0B +#define ARIZONA_GP_FN_FLL1_LOCK 0x0C +#define ARIZONA_GP_FN_FLL2_LOCK 0x0D +#define ARIZONA_GP_FN_FLL1_CLOCK_OK 0x0F +#define ARIZONA_GP_FN_FLL2_CLOCK_OK 0x10 +#define ARIZONA_GP_FN_HEADPHONE_DET 0x12 +#define ARIZONA_GP_FN_MIC_DET 0x13 +#define ARIZONA_GP_FN_WSEQ_STATUS 0x15 +#define ARIZONA_GP_FN_CIF_ADDRESS_ERROR 0x16 +#define ARIZONA_GP_FN_ASRC1_LOCK 0x1A +#define ARIZONA_GP_FN_ASRC2_LOCK 0x1B +#define ARIZONA_GP_FN_ASRC_CONFIG_ERROR 0x1C +#define ARIZONA_GP_FN_DRC1_SIGNAL_DETECT 0x1D +#define ARIZONA_GP_FN_DRC1_ANTICLIP 0x1E +#define ARIZONA_GP_FN_DRC1_DECAY 0x1F +#define ARIZONA_GP_FN_DRC1_NOISE 0x20 +#define ARIZONA_GP_FN_DRC1_QUICK_RELEASE 0x21 +#define ARIZONA_GP_FN_DRC2_SIGNAL_DETECT 0x22 +#define ARIZONA_GP_FN_DRC2_ANTICLIP 0x23 +#define ARIZONA_GP_FN_DRC2_DECAY 0x24 +#define ARIZONA_GP_FN_DRC2_NOISE 0x25 +#define ARIZONA_GP_FN_DRC2_QUICK_RELEASE 0x26 +#define ARIZONA_GP_FN_MIXER_DROPPED_SAMPLE 0x27 +#define ARIZONA_GP_FN_AIF1_CONFIG_ERROR 0x28 +#define ARIZONA_GP_FN_AIF2_CONFIG_ERROR 0x29 +#define ARIZONA_GP_FN_AIF3_CONFIG_ERROR 0x2A +#define ARIZONA_GP_FN_SPK_TEMP_SHUTDOWN 0x2B +#define ARIZONA_GP_FN_SPK_TEMP_WARNING 0x2C +#define ARIZONA_GP_FN_UNDERCLOCKED 0x2D +#define ARIZONA_GP_FN_OVERCLOCKED 0x2E +#define ARIZONA_GP_FN_DSP_IRQ1 0x35 +#define ARIZONA_GP_FN_DSP_IRQ2 0x36 +#define ARIZONA_GP_FN_ASYNC_OPCLK 0x3D +#define ARIZONA_GP_FN_BOOT_DONE 0x44 +#define ARIZONA_GP_FN_DSP1_RAM_READY 0x45 +#define ARIZONA_GP_FN_SYSCLK_ENA_STATUS 0x4B +#define ARIZONA_GP_FN_ASYNCCLK_ENA_STATUS 0x4C + +#define ARIZONA_GPN_DIR 0x8000 /* GPN_DIR */ +#define ARIZONA_GPN_DIR_MASK 0x8000 /* GPN_DIR */ +#define ARIZONA_GPN_DIR_SHIFT 15 /* GPN_DIR */ +#define ARIZONA_GPN_DIR_WIDTH 1 /* GPN_DIR */ +#define ARIZONA_GPN_PU 0x4000 /* GPN_PU */ +#define ARIZONA_GPN_PU_MASK 0x4000 /* GPN_PU */ +#define ARIZONA_GPN_PU_SHIFT 14 /* GPN_PU */ +#define ARIZONA_GPN_PU_WIDTH 1 /* GPN_PU */ +#define ARIZONA_GPN_PD 0x2000 /* GPN_PD */ +#define ARIZONA_GPN_PD_MASK 0x2000 /* GPN_PD */ +#define ARIZONA_GPN_PD_SHIFT 13 /* GPN_PD */ +#define ARIZONA_GPN_PD_WIDTH 1 /* GPN_PD */ +#define ARIZONA_GPN_LVL 0x0800 /* GPN_LVL */ +#define ARIZONA_GPN_LVL_MASK 0x0800 /* GPN_LVL */ +#define ARIZONA_GPN_LVL_SHIFT 11 /* GPN_LVL */ +#define ARIZONA_GPN_LVL_WIDTH 1 /* GPN_LVL */ +#define ARIZONA_GPN_POL 0x0400 /* GPN_POL */ +#define ARIZONA_GPN_POL_MASK 0x0400 /* GPN_POL */ +#define ARIZONA_GPN_POL_SHIFT 10 /* GPN_POL */ +#define ARIZONA_GPN_POL_WIDTH 1 /* GPN_POL */ +#define ARIZONA_GPN_OP_CFG 0x0200 /* GPN_OP_CFG */ +#define ARIZONA_GPN_OP_CFG_MASK 0x0200 /* GPN_OP_CFG */ +#define ARIZONA_GPN_OP_CFG_SHIFT 9 /* GPN_OP_CFG */ +#define ARIZONA_GPN_OP_CFG_WIDTH 1 /* GPN_OP_CFG */ +#define ARIZONA_GPN_DB 0x0100 /* GPN_DB */ +#define ARIZONA_GPN_DB_MASK 0x0100 /* GPN_DB */ +#define ARIZONA_GPN_DB_SHIFT 8 /* GPN_DB */ +#define ARIZONA_GPN_DB_WIDTH 1 /* GPN_DB */ +#define ARIZONA_GPN_FN_MASK 0x007F /* GPN_DB */ +#define ARIZONA_GPN_FN_SHIFT 0 /* GPN_DB */ +#define ARIZONA_GPN_FN_WIDTH 7 /* GPN_DB */ + +#endif -- 1.7.2.5

2 3

[alsa-devel] [PATCH] mfd: arizona: Move regulator disable to after marking cache only
by Charles Keepax 03 Jul '13

03 Jul '13

If we disable DCVDD before we mark the cache as cache only, we might attempt to write to the chip whilst it is powered down and lose a write. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- drivers/mfd/arizona-core.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/drivers/mfd/arizona-core.c b/drivers/mfd/arizona-core.c index 6ab0304..aa47709 100644 --- a/drivers/mfd/arizona-core.c +++ b/drivers/mfd/arizona-core.c @@ -388,9 +388,9 @@ static int arizona_runtime_suspend(struct device *dev) dev_dbg(arizona->dev, "Entering AoD mode\n"); - regulator_disable(arizona->dcvdd); regcache_cache_only(arizona->regmap, true); regcache_mark_dirty(arizona->regmap); + regulator_disable(arizona->dcvdd); return 0; } -- 1.7.2.5

1 0

[alsa-devel] [v2 0/5] ASoC: atmel-pcm: align with generic dmaengine framwork
by Bo Shen 03 Jul '13

03 Jul '13

This patch set is based on linux next-20130701 mainly implement - move set dma data from hw_params to startup callback to avoid memory leak - align atmel pcm with generic dmaenigne framwork Bo Shen (5): ASoC: atmel_ssc_dai: move set dma data to startup callback ASoC: atmel_ssc_dai: add error mask define ASoC: atmel-pcm-dma: move prepare for dma to dai prepare ARM: atmel-ssc: change phybase type to dma_addr_t ASoC: atmel-pcm: use generic dmaengine framework include/linux/atmel-ssc.h | 2 +- sound/soc/atmel/Kconfig | 1 + sound/soc/atmel/atmel-pcm-dma.c | 118 ++++++--------------------------------- sound/soc/atmel/atmel_ssc_dai.c | 36 ++++++------ 4 files changed, 37 insertions(+), 120 deletions(-) -- 1.7.9.5

3 7

[alsa-devel] [PATCH 0/2] ASoC: atmel-pcm: use generic dmaengine framework
by Bo Shen 03 Jul '13

03 Jul '13

this patch series is based on next-20130701 Bo Shen (2): ASoC: atmel_ssc_dai: move set dma data to startup callback ASoC: atmel-pcm: use generic dmaengine framework sound/soc/atmel/atmel-pcm-dma.c | 118 ++++++--------------------------------- sound/soc/atmel/atmel_ssc_dai.c | 33 ++++++----- 2 files changed, 32 insertions(+), 119 deletions(-) -- 1.7.9.5

3 11

[alsa-devel] no analog playback device for HDA STAC9221 inside VirtualBox 4.2.14
by Raymond Yau 03 Jul '13

03 Jul '13

HDA STAC9221 audio work for (64bit) Fedora 16, Ubuntu 11.10 and Opensuse 11.4 Live CD inside virtualbox 4.2.14 no sound for this emulated HDA sound card since Ubuntu 12.04 Live CD No analog playback device for Saucy or Fedora 19 Live CD

2 7

[alsa-devel] [PATCH] ASoC: mop500: add .owner to struct snd_soc_card
by Wei Yongjun 02 Jul '13

02 Jul '13

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Add missing .owner of struct snd_soc_card. This prevents the module from being removed from underneath its users. Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/ux500/mop500.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c index 8f5cd00..178d1ba 100644 --- a/sound/soc/ux500/mop500.c +++ b/sound/soc/ux500/mop500.c @@ -52,6 +52,7 @@ static struct snd_soc_dai_link mop500_dai_links[] = { static struct snd_soc_card mop500_card = { .name = "MOP500-card", + .owner = THIS_MODULE, .probe = NULL, .dai_link = mop500_dai_links, .num_links = ARRAY_SIZE(mop500_dai_links),

2 1

[alsa-devel] [PATCH] ASoC: db1200: add .owner to struct snd_soc_card
by Wei Yongjun 02 Jul '13

02 Jul '13

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Add missing .owner of struct snd_soc_card. This prevents the module from being removed from underneath its users. Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/au1x/db1200.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c index a497a0c..2d2407d 100644 --- a/sound/soc/au1x/db1200.c +++ b/sound/soc/au1x/db1200.c @@ -73,12 +73,14 @@ static struct snd_soc_dai_link db1300_ac97_dai = { static struct snd_soc_card db1300_ac97_machine = { .name = "DB1300_AC97", + .owner = THIS_MODULE, .dai_link = &db1300_ac97_dai, .num_links = 1, }; static struct snd_soc_card db1550_ac97_machine = { .name = "DB1550_AC97", + .owner = THIS_MODULE, .dai_link = &db1200_ac97_dai, .num_links = 1, }; @@ -145,6 +147,7 @@ static struct snd_soc_dai_link db1300_i2s_dai = { static struct snd_soc_card db1300_i2s_machine = { .name = "DB1300_I2S", + .owner = THIS_MODULE, .dai_link = &db1300_i2s_dai, .num_links = 1, }; @@ -161,6 +164,7 @@ static struct snd_soc_dai_link db1550_i2s_dai = { static struct snd_soc_card db1550_i2s_machine = { .name = "DB1550_I2S", + .owner = THIS_MODULE, .dai_link = &db1550_i2s_dai, .num_links = 1, };

2 1

[alsa-devel] [PATCH] ASoC: imx_mc13783: add .owner to struct snd_soc_card
by Wei Yongjun 02 Jul '13

02 Jul '13

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Add missing .owner of struct snd_soc_card. This prevents the module from being removed from underneath its users. Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/fsl/imx-mc13783.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c index 9df173c..55742a1 100644 --- a/sound/soc/fsl/imx-mc13783.c +++ b/sound/soc/fsl/imx-mc13783.c @@ -90,6 +90,7 @@ static const struct snd_soc_dapm_route imx_mc13783_routes[] = { static struct snd_soc_card imx_mc13783 = { .name = "imx_mc13783", + .owner = THIS_MODULE, .dai_link = imx_mc13783_dai_mc13783, .num_links = ARRAY_SIZE(imx_mc13783_dai_mc13783), .dapm_widgets = imx_mc13783_widget,

2 1

[alsa-devel] [PATCH] ASoC: ttc_dkb: add .owner to struct snd_soc_card
by Wei Yongjun 02 Jul '13

02 Jul '13

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Add missing .owner of struct snd_soc_card. This prevents the module from being removed from underneath its users. Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/pxa/ttc-dkb.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/pxa/ttc-dkb.c b/sound/soc/pxa/ttc-dkb.c index f4ea4f6..13c9ee0 100644 --- a/sound/soc/pxa/ttc-dkb.c +++ b/sound/soc/pxa/ttc-dkb.c @@ -122,6 +122,7 @@ static struct snd_soc_dai_link ttc_pm860x_hifi_dai[] = { /* ttc/td audio machine driver */ static struct snd_soc_card ttc_dkb_card = { .name = "ttc-dkb-hifi", + .owner = THIS_MODULE, .dai_link = ttc_pm860x_hifi_dai, .num_links = ARRAY_SIZE(ttc_pm860x_hifi_dai),

2 1

[alsa-devel] [PATCH] ASoC: brownstone: add .owner to struct snd_soc_card
by Wei Yongjun 02 Jul '13

02 Jul '13

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Add missing .owner of struct snd_soc_card. This prevents the module from being removed from underneath its users. Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/pxa/brownstone.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c index 4ad7609..e582c05 100644 --- a/sound/soc/pxa/brownstone.c +++ b/sound/soc/pxa/brownstone.c @@ -129,6 +129,7 @@ static struct snd_soc_dai_link brownstone_wm8994_dai[] = { /* audio machine driver */ static struct snd_soc_card brownstone = { .name = "brownstone", + .owner = THIS_MODULE, .dai_link = brownstone_wm8994_dai, .num_links = ARRAY_SIZE(brownstone_wm8994_dai),

2 1

[alsa-devel] [PATCH 1/2] mfd: arizona: Add GPIO control register bit definitions
by Charles Keepax 02 Jul '13

02 Jul '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- include/linux/mfd/arizona/registers.h | 35 +++++++++++++++++++++++++++++++++ 1 files changed, 35 insertions(+), 0 deletions(-) diff --git a/include/linux/mfd/arizona/registers.h b/include/linux/mfd/arizona/registers.h index 715b6ba..522798d 100644 --- a/include/linux/mfd/arizona/registers.h +++ b/include/linux/mfd/arizona/registers.h @@ -4131,6 +4131,41 @@ #define ARIZONA_SLIMTX1_PORT_STS_WIDTH 1 /* SLIMTX1_PORT_STS */ /* + * R3072 - R3076 (0xC00 - 0xC04) ARIZONA_GPIO1_CTRL - ARIZONA_GPIO5_CTRL + */ +#define ARIZONA_GPN_DIR 0x8000 /* GPN_DIR */ +#define ARIZONA_GPN_DIR_MASK 0x8000 /* GPN_DIR */ +#define ARIZONA_GPN_DIR_SHIFT 15 /* GPN_DIR */ +#define ARIZONA_GPN_DIR_WIDTH 1 /* GPN_DIR */ +#define ARIZONA_GPN_PU 0x4000 /* GPN_PU */ +#define ARIZONA_GPN_PU_MASK 0x4000 /* GPN_PU */ +#define ARIZONA_GPN_PU_SHIFT 14 /* GPN_PU */ +#define ARIZONA_GPN_PU_WIDTH 1 /* GPN_PU */ +#define ARIZONA_GPN_PD 0x2000 /* GPN_PD */ +#define ARIZONA_GPN_PD_MASK 0x2000 /* GPN_PD */ +#define ARIZONA_GPN_PD_SHIFT 13 /* GPN_PD */ +#define ARIZONA_GPN_PD_WIDTH 1 /* GPN_PD */ +#define ARIZONA_GPN_LVL 0x0800 /* GPN_LVL */ +#define ARIZONA_GPN_LVL_MASK 0x0800 /* GPN_LVL */ +#define ARIZONA_GPN_LVL_SHIFT 11 /* GPN_LVL */ +#define ARIZONA_GPN_LVL_WIDTH 1 /* GPN_LVL */ +#define ARIZONA_GPN_POL 0x0400 /* GPN_POL */ +#define ARIZONA_GPN_POL_MASK 0x0400 /* GPN_POL */ +#define ARIZONA_GPN_POL_SHIFT 10 /* GPN_POL */ +#define ARIZONA_GPN_POL_WIDTH 1 /* GPN_POL */ +#define ARIZONA_GPN_OP_CFG 0x0200 /* GPN_OP_CFG */ +#define ARIZONA_GPN_OP_CFG_MASK 0x0200 /* GPN_OP_CFG */ +#define ARIZONA_GPN_OP_CFG_SHIFT 9 /* GPN_OP_CFG */ +#define ARIZONA_GPN_OP_CFG_WIDTH 1 /* GPN_OP_CFG */ +#define ARIZONA_GPN_DB 0x0100 /* GPN_DB */ +#define ARIZONA_GPN_DB_MASK 0x0100 /* GPN_DB */ +#define ARIZONA_GPN_DB_SHIFT 8 /* GPN_DB */ +#define ARIZONA_GPN_DB_WIDTH 1 /* GPN_DB */ +#define ARIZONA_GPN_FN_MASK 0x007F /* GPN_DB */ +#define ARIZONA_GPN_FN_SHIFT 0 /* GPN_DB */ +#define ARIZONA_GPN_FN_WIDTH 7 /* GPN_DB */ + +/* * R3087 (0xC0F) - IRQ CTRL 1 */ #define ARIZONA_IRQ_POL 0x0400 /* IRQ_POL */ -- 1.7.2.5

2 4

[alsa-devel] [PATCH] ASoC: samsung: Remove obsolete GPIO based DT pinmuxing
by Mark Brown 02 Jul '13

02 Jul '13

From: Mark Brown <broonie(a)linaro.org> Since the Samsung platforms have moved to pinctrl for pin muxing and that is handled in the core the old GPIO based muxing code can just be removed. Something similar had been submitted by Thomas Abraham back in March but a resubmission following review never happened. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/samsung/i2s.c | 66 +++---------------------------------------------- 1 file changed, 4 insertions(+), 62 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 82ebb1a..7a17346 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1016,52 +1016,6 @@ static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec) return i2s; } -#ifdef CONFIG_OF -static int samsung_i2s_parse_dt_gpio(struct i2s_dai *i2s) -{ - struct device *dev = &i2s->pdev->dev; - int index, gpio, ret; - - for (index = 0; index < 7; index++) { - gpio = of_get_gpio(dev->of_node, index); - if (!gpio_is_valid(gpio)) { - dev_err(dev, "invalid gpio[%d]: %d\n", index, gpio); - goto free_gpio; - } - - ret = gpio_request(gpio, dev_name(dev)); - if (ret) { - dev_err(dev, "gpio [%d] request failed\n", gpio); - goto free_gpio; - } - i2s->gpios[index] = gpio; - } - return 0; - -free_gpio: - while (--index >= 0) - gpio_free(i2s->gpios[index]); - return -EINVAL; -} - -static void samsung_i2s_dt_gpio_free(struct i2s_dai *i2s) -{ - unsigned int index; - for (index = 0; index < 7; index++) - gpio_free(i2s->gpios[index]); -} -#else -static int samsung_i2s_parse_dt_gpio(struct i2s_dai *dai) -{ - return -EINVAL; -} - -static void samsung_i2s_dt_gpio_free(struct i2s_dai *dai) -{ -} - -#endif - static const struct of_device_id exynos_i2s_match[]; static inline int samsung_i2s_get_driver_data(struct platform_device *pdev) @@ -1235,18 +1189,10 @@ static int samsung_i2s_probe(struct platform_device *pdev) pri_dai->sec_dai = sec_dai; } - if (np) { - if (samsung_i2s_parse_dt_gpio(pri_dai)) { - dev_err(&pdev->dev, "Unable to configure gpio\n"); - ret = -EINVAL; - goto err; - } - } else { - if (i2s_pdata->cfg_gpio && i2s_pdata->cfg_gpio(pdev)) { - dev_err(&pdev->dev, "Unable to configure gpio\n"); - ret = -EINVAL; - goto err; - } + if (i2s_pdata && i2s_pdata->cfg_gpio && i2s_pdata->cfg_gpio(pdev)) { + dev_err(&pdev->dev, "Unable to configure gpio\n"); + ret = -EINVAL; + goto err; } snd_soc_register_component(&pri_dai->pdev->dev, &samsung_i2s_component, @@ -1267,14 +1213,10 @@ static int samsung_i2s_remove(struct platform_device *pdev) { struct i2s_dai *i2s, *other; struct resource *res; - struct s3c_audio_pdata *i2s_pdata = pdev->dev.platform_data; i2s = dev_get_drvdata(&pdev->dev); other = i2s->pri_dai ? : i2s->sec_dai; - if (!i2s_pdata->cfg_gpio && pdev->dev.of_node) - samsung_i2s_dt_gpio_free(i2s->pri_dai); - if (other) { other->pri_dai = NULL; other->sec_dai = NULL; -- 1.8.3.1

1 0

[alsa-devel] Multiple DA9055 chips
by Mark Brown 02 Jul '13

02 Jul '13

It seems we have two drivers in mainline for da9055 on I2C. We've got one in sound/soc/codecs/da9055.c and one in drivers/mfd/da9055-i2c.c, both registering themselves identically. What's going on here? Is this a combined CODEC and PMIC, has someone decided to release two chips with the same name or is one of the drivers misnamed?

2 5

[alsa-devel] [PATCH] ASoC: adav80x: Add module device table for adav801
by Axel Lin 02 Jul '13

02 Jul '13

This driver can be built as module, thus add module device table for adav801 to support module auto loading. To make the naming consistent, also rename adav80x_id to adav80x_i2c_id. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/adav80x.c | 13 ++++++++++--- 1 file changed, 10 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index 3c839cc..15b012d0 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -868,6 +868,12 @@ static int adav80x_bus_remove(struct device *dev) } #if defined(CONFIG_SPI_MASTER) +static const struct spi_device_id adav80x_spi_id[] = { + { "adav801", 0 }, + { } +}; +MODULE_DEVICE_TABLE(spi, adav80x_spi_id); + static int adav80x_spi_probe(struct spi_device *spi) { return adav80x_bus_probe(&spi->dev, SND_SOC_SPI); @@ -885,15 +891,16 @@ static struct spi_driver adav80x_spi_driver = { }, .probe = adav80x_spi_probe, .remove = adav80x_spi_remove, + .id_table = adav80x_spi_id, }; #endif #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) -static const struct i2c_device_id adav80x_id[] = { +static const struct i2c_device_id adav80x_i2c_id[] = { { "adav803", 0 }, { } }; -MODULE_DEVICE_TABLE(i2c, adav80x_id); +MODULE_DEVICE_TABLE(i2c, adav80x_i2c_id); static int adav80x_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) @@ -913,7 +920,7 @@ static struct i2c_driver adav80x_i2c_driver = { }, .probe = adav80x_i2c_probe, .remove = adav80x_i2c_remove, - .id_table = adav80x_id, + .id_table = adav80x_i2c_id, }; #endif -- 1.8.1.2

3 2

[alsa-devel] [PATCH] ASOC: pcm: support fe dapm widget derived from codec_dai
by Nenghua Cao 02 Jul '13

02 Jul '13

From: nhcao <nhcao(a)marvell.com> In some chips, the codec_dai in cpu is abstracted as one dapm widget. Enhance the framework to use codec_dai as fe widget. Change-Id: I1b545bd1e2f24288f4508a7c6d4efbcfe1e03c5b Signed-off-by: nhcao <nhcao(a)marvell.com> --- sound/soc/soc-pcm.c | 24 ++++++++++++++++++++++-- 1 files changed, 22 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 73bb8ee..6ca23d7 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -856,17 +856,37 @@ static int widget_in_list(struct snd_soc_dapm_widget_list *list, static int dpcm_path_get(struct snd_soc_pcm_runtime *fe, int stream, struct snd_soc_dapm_widget_list **list_) { - struct snd_soc_dai *cpu_dai = fe->cpu_dai; + struct snd_soc_dai *dai = fe->cpu_dai ; struct snd_soc_dapm_widget_list *list; int paths; + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (fe->cpu_dai->playback_widget) { + dai = fe->cpu_dai; + } else if (fe->codec_dai->playback_widget) { + dai = fe->codec_dai; + } else { + dev_err(fe->dev, "ASoC: can't find the correspond playback_widget for the dai\n"); + return -EINVAL; + } + } else if (stream == SNDRV_PCM_STREAM_CAPTURE) { + if (fe->cpu_dai->capture_widget) { + dai = fe->cpu_dai; + } else if (fe->codec_dai->capture_widget) { + dai = fe->codec_dai; + } else { + dev_err(fe->dev, "ASoC: can't find the correspond capture_widget for the dai\n"); + return -EINVAL; + } + } + list = kzalloc(sizeof(struct snd_soc_dapm_widget_list) + sizeof(struct snd_soc_dapm_widget *), GFP_KERNEL); if (list == NULL) return -ENOMEM; /* get number of valid DAI paths and their widgets */ - paths = snd_soc_dapm_dai_get_connected_widgets(cpu_dai, stream, &list); + paths = snd_soc_dapm_dai_get_connected_widgets(dai, stream, &list); dev_dbg(fe->dev, "ASoC: found %d audio %s paths\n", paths, stream ? "capture" : "playback"); -- 1.7.0.4

1 0

[alsa-devel] [PATCH 0/2] ASoC: atmel-pcm: use generic dmaengine framework
by Bo Shen 02 Jul '13

02 Jul '13

this patch series is based on next-20130701 Bo Shen (2): ASoC: atmel_ssc_dai: move set dma data to startup callback ASoC: atmel-pcm: use generic dmaengine framework sound/soc/atmel/atmel-pcm-dma.c | 118 ++++++--------------------------------- sound/soc/atmel/atmel_ssc_dai.c | 33 ++++++----- 2 files changed, 32 insertions(+), 119 deletions(-) -- 1.7.9.5

1 2

[alsa-devel] [PATCH] ASoC: wm8994: Move runtime PM init to platform device init
by Mark Brown 01 Jul '13

01 Jul '13

From: Mark Brown <broonie(a)linaro.org> As well as being better style this allows the device to idle when there is no audio card instantaited which is probably what we want. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/wm8994.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 1d4b1ec..02c320f 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -4014,9 +4014,6 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994->micdet_irq = control->pdata.micdet_irq; - pm_runtime_enable(codec->dev); - pm_runtime_idle(codec->dev); - /* By default use idle_bias_off, will override for WM8994 */ codec->dapm.idle_bias_off = 1; @@ -4389,8 +4386,6 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec) wm8994_set_bias_level(codec, SND_SOC_BIAS_OFF); - pm_runtime_disable(codec->dev); - for (i = 0; i < ARRAY_SIZE(wm8994->fll_locked); i++) wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_FLL1_LOCK + i, &wm8994->fll_locked[i]); @@ -4449,6 +4444,9 @@ static int wm8994_probe(struct platform_device *pdev) wm8994->wm8994 = dev_get_drvdata(pdev->dev.parent); + pm_runtime_enable(&pdev->dev); + pm_runtime_idle(&pdev->dev); + return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm8994, wm8994_dai, ARRAY_SIZE(wm8994_dai)); } @@ -4456,6 +4454,8 @@ static int wm8994_probe(struct platform_device *pdev) static int wm8994_remove(struct platform_device *pdev) { snd_soc_unregister_codec(&pdev->dev); + pm_runtime_disable(&pdev->dev); + return 0; } -- 1.8.3.1

1 0

[alsa-devel] [PATCH 0/4] Fix mxs audio regressions
by Shawn Guo 01 Jul '13

01 Jul '13

The mxs audio got terribly broken during 3.11 development cycle due to the following two commits. af8ee11 ASoC: sgtl5000: Fix driver probe after reset 9e13f34 ASoC: sgtl5000: Let the codec acquire its clock The series fixes the regressions and bring mxs audio back to work. Mark, Please consider to take the whole series as fixes through your tree during 3.11-rc, so that we do not get a regression in the final v3.11 release. Shawn Shawn Guo (4): ASoC: sgtl5000: give it a ramping time before writting ASoC: sgtl5000: defer the probe if clock is not found ASoC: mxs: register saif mclk to clock framework ARM: mxs: saif0 is the clock provider to sgtl5000 arch/arm/boot/dts/imx28-apx4devkit.dts | 2 +- arch/arm/boot/dts/imx28-evk.dts | 2 +- arch/arm/boot/dts/imx28-m28evk.dts | 2 +- arch/arm/boot/dts/imx28.dtsi | 1 + sound/soc/codecs/sgtl5000.c | 10 ++++++++- sound/soc/mxs/mxs-saif.c | 35 ++++++++++++++++++++++++++++++++ 6 files changed, 48 insertions(+), 4 deletions(-) -- 1.7.9.5

7 24

[alsa-devel] [PATCH] ASoC: Add PCM1681 codec driver.
by Marek Belisko 01 Jul '13

01 Jul '13

PCM1681 can be controlled via I2C, SPI or in bootstrap mode. This code add support only for I2C mode. Signed-off-by: Marek Belisko <marek.belisko(a)streamunlimited.com> --- .../devicetree/bindings/sound/ti,pcm1681.txt | 15 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/pcm1681.c | 328 ++++++++++++++++++++ 4 files changed, 349 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,pcm1681.txt create mode 100644 sound/soc/codecs/pcm1681.c diff --git a/Documentation/devicetree/bindings/sound/ti,pcm1681.txt b/Documentation/devicetree/bindings/sound/ti,pcm1681.txt new file mode 100644 index 0000000..4df1718 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,pcm1681.txt @@ -0,0 +1,15 @@ +Texas Instruments PCM1681 8-channel PWM Processor + +Required properties: + + - compatible: Should contain "ti,pcm1681". + - reg: The i2c address. Should contain <0x4c>. + +Examples: + + i2c_bus { + pcm1681@4c { + compatible = "ti,pcm1681"; + reg = <0x4c>; + }; + }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index badb6fb..e2daf82 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -54,6 +54,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MC13783 if MFD_MC13XXX select SND_SOC_ML26124 if I2C select SND_SOC_HDMI_CODEC + select SND_SOC_PCM1681 if I2C select SND_SOC_PCM3008 select SND_SOC_RT5631 if I2C select SND_SOC_RT5640 if I2C @@ -292,6 +293,9 @@ config SND_SOC_MAX9850 config SND_SOC_HDMI_CODEC tristate +config SND_SOC_PCM1681 + tristate + config SND_SOC_PCM3008 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 70fd806..4a068d2 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -42,6 +42,7 @@ snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o snd-soc-hdmi-codec-objs := hdmi.o +snd-soc-pcm1681-objs := pcm1681.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o @@ -171,6 +172,7 @@ obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o +obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c new file mode 100644 index 0000000..6122fd1 --- /dev/null +++ b/sound/soc/codecs/pcm1681.c @@ -0,0 +1,328 @@ +/* + * PCM1681 ASoC codec driver + * + * Copyright (c) StreamUnlimited GmbH 2013 + * Marek Belisko <marek.belisko(a)streamunlimited.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * as published by the Free Software Foundation; either version 2 + * of the License, or (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#include <linux/module.h> +#include <linux/slab.h> +#include <linux/delay.h> +#include <linux/gpio.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/of_device.h> +#include <linux/of_gpio.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#define PCM1681_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +#define PCM1681_PCM_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | \ + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000) + +#define PCM1681_SOFT_MUTE_ALL 0xff +#define PCM1681_DEEMPH_RATE_MASK 0x18 +#define PCM1681_DEEMPH_MASK 0x01 + +#define PCM1681_ATT_CONTROL(X) (X <= 6 ? X : X + 9) /* Attenuation level */ +#define PCM1681_SOFT_MUTE 0x07 /* Soft mute control register */ +#define PCM1681_DAC_CONTROL 0x08 /* DAC operation control */ +#define PCM1681_FMT_CONTROL 0x09 /* Audio interface data format */ +#define PCM1681_DEEMPH_CONTROL 0x0a /* De-emphasis control */ +#define PCM1681_ZERO_DETECT_STATUS 0x0e /* Zero detect status reg */ + +static const struct reg_default pcm1681_reg_defaults[] = { + { 0x01, 0xff }, + { 0x02, 0xff }, + { 0x03, 0xff }, + { 0x04, 0xff }, + { 0x05, 0xff }, + { 0x06, 0xff }, + { 0x07, 0x00 }, + { 0x08, 0x00 }, + { 0x09, 0x06 }, + { 0x0A, 0x00 }, + { 0x0B, 0xff }, + { 0x0C, 0x0f }, + { 0x0D, 0x00 }, + { 0x10, 0xff }, + { 0x11, 0xff }, + { 0x12, 0x00 }, + { 0x13, 0x00 }, +}; + +static bool pcm1681_accessible_reg(struct device *dev, unsigned int reg) +{ + return !((reg == 0x00) || (reg == 0x0f)); +} + +static bool pcm1681_writeable_reg(struct device *dev, unsigned register reg) +{ + return pcm1681_accessible_reg(dev, reg) && + (reg != PCM1681_ZERO_DETECT_STATUS); +} + +struct pcm1681_private { + struct regmap *regmap; + unsigned int format; + /* Current deemphasis status */ + unsigned int deemph; + /* Current rate for deemphasis control */ + unsigned int rate; +}; + +static int pcm1681_deemph[] = { 44100, 48000, 32000 }; + +static int pcm1681_set_deemph(struct snd_soc_codec *codec) +{ + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + int i = 0, val = -1, ret; + + if (priv->deemph) + for (i = 0; i < ARRAY_SIZE(pcm1681_deemph); i++) + if (pcm1681_deemph[i] == priv->rate) + val = i; + + /* enable choosen frequency */ + if (val != -1) + regmap_update_bits(priv->regmap, PCM1681_DEEMPH_CONTROL, + PCM1681_DEEMPH_RATE_MASK, val); + + /* enable/disable deemphasis functionality */ + ret = regmap_update_bits(priv->regmap, PCM1681_DEEMPH_CONTROL, + PCM1681_DEEMPH_MASK, priv->deemph); + + return ret; +} + +static int pcm1681_get_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.enumerated.item[0] = priv->deemph; + + return 0; +} + +static int pcm1681_put_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + + priv->deemph = ucontrol->value.enumerated.item[0]; + + return pcm1681_set_deemph(codec); +} + +static int pcm1681_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + + /* The PCM1681 can only be slave to all clocks */ + if ((format & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) { + dev_err(codec->dev, "Invalid clocking mode\n"); + return -EINVAL; + } + + priv->format = format; + + return 0; +} + +static int pcm1681_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + int ret, val = 0; + + if (mute) + val = PCM1681_SOFT_MUTE_ALL; + + ret = regmap_write(priv->regmap, PCM1681_SOFT_MUTE, val); + if (ret < 0) + return ret; + + return 0; +} + +static int pcm1681_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + int val = 0; + int pcm_format = params_format(params); + + priv->rate = params_rate(params); + + switch (priv->format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + if (pcm_format == SNDRV_PCM_FORMAT_S24_LE) + val = 0x00; + else if (pcm_format == SNDRV_PCM_FORMAT_S16_LE) + val = 0x03; + break; + case SND_SOC_DAIFMT_I2S: + val = 0x04; + break; + case SND_SOC_DAIFMT_LEFT_J: + val = 0x05; + break; + default: + dev_err(codec->dev, "Invalid DAI format\n"); + return -EINVAL; + } + + regmap_update_bits(priv->regmap, PCM1681_FMT_CONTROL, 0x0f, val); + + return 0; +} + +static const struct snd_soc_dai_ops pcm1681_dai_ops = { + .set_fmt = pcm1681_set_dai_fmt, + .hw_params = pcm1681_hw_params, + .digital_mute = pcm1681_digital_mute, +}; + +static const DECLARE_TLV_DB_SCALE(pcm1681_dac_tlv, -6350, 50, 1); + +static const struct snd_kcontrol_new pcm1681_controls[] = { + SOC_DOUBLE_R_TLV("PCM1681 Channel 1/2 Playback Volume", + PCM1681_ATT_CONTROL(1), PCM1681_ATT_CONTROL(2), 0, + 0x7f, 0, pcm1681_dac_tlv), + SOC_DOUBLE_R_TLV("PCM1681 Channel 3/4 Playback Volume", + PCM1681_ATT_CONTROL(3), PCM1681_ATT_CONTROL(4), 0, + 0x7f, 0, pcm1681_dac_tlv), + SOC_DOUBLE_R_TLV("PCM1681 Channel 5/6 Playback Volume", + PCM1681_ATT_CONTROL(5), PCM1681_ATT_CONTROL(6), 0, + 0x7f, 0, pcm1681_dac_tlv), + SOC_DOUBLE_R_TLV("PCM1681 Channel 7/8 Playback Volume", + PCM1681_ATT_CONTROL(7), PCM1681_ATT_CONTROL(8), 0, + 0x7f, 0, pcm1681_dac_tlv), + SOC_SINGLE_BOOL_EXT("PCM1681 De-emphasis Switch", 0, + pcm1681_get_deemph, pcm1681_put_deemph), +}; + +struct snd_soc_dai_driver pcm1681_dai = { + .name = "pcm1681-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 8, + .rates = PCM1681_PCM_RATES, + .formats = PCM1681_PCM_FORMATS, + }, + .ops = &pcm1681_dai_ops, +}; + +#ifdef CONFIG_OF +static const struct of_device_id pcm1681_dt_ids[] = { + { .compatible = "ti,pcm1681", }, + { } +}; +MODULE_DEVICE_TABLE(of, pcm1681_dt_ids); +#endif + +static const struct regmap_config pcm1681_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = ARRAY_SIZE(pcm1681_reg_defaults), + .reg_defaults = pcm1681_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(pcm1681_reg_defaults), + .writeable_reg = pcm1681_writeable_reg, + .readable_reg = pcm1681_accessible_reg, +}; + +static int pcm1681_probe(struct snd_soc_codec *codec) +{ + return 0; +} + +static int pcm1681_remove(struct snd_soc_codec *codec) +{ + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_pcm1681 = { + .probe = pcm1681_probe, + .remove = pcm1681_remove, + .controls = pcm1681_controls, + .num_controls = ARRAY_SIZE(pcm1681_controls), +}; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static const struct i2c_device_id pcm1681_i2c_id[] = { + {"pcm1681", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, pcm1681_i2c_id); + +static int pcm1681_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + int ret; + struct pcm1681_private *priv; + + priv = devm_kzalloc(&client->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->regmap = devm_regmap_init_i2c(client, &pcm1681_regmap); + if (IS_ERR(priv->regmap)) { + ret = PTR_ERR(priv->regmap); + dev_err(&client->dev, "Failed to create regmap: %d\n", ret); + return ret; + } + + i2c_set_clientdata(client, priv); + + return snd_soc_register_codec(&client->dev, &soc_codec_dev_pcm1681, + &pcm1681_dai, 1); +} + +static int pcm1681_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static struct i2c_driver pcm1681_i2c_driver = { + .driver = { + .name = "pcm1681", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(pcm1681_dt_ids), + }, + .id_table = pcm1681_i2c_id, + .probe = pcm1681_i2c_probe, + .remove = pcm1681_i2c_remove, +}; + +module_i2c_driver(pcm1681_i2c_driver); + +#endif /* defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) */ + +MODULE_DESCRIPTION("Texas Instruments PCM1681 ALSA SoC Codec Driver"); +MODULE_AUTHOR("Marek Belisko <marek.belisko(a)streamunlimited.com>"); +MODULE_LICENSE("GPL"); -- 1.7.9.5

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[alsa-devel] [GIT PULL] ASoC updates for v3.11
by Mark Brown 01 Jul '13

01 Jul '13

The following changes since commit a975873a9acc0788c1aee5ca183deb420b5c00e5: ASoC: tas5086: fix Mid-Z implementation (2013-06-28 12:15:30 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.11-3 for you to fetch changes up to 370887f105025ed52acc9cbb0a15b6567d1550ff: Merge remote-tracking branch 'asoc/topic/x86' into asoc-next (2013-06-30 12:42:25 +0100) ---------------------------------------------------------------- ASoC: Final updates for v3.11 A few final updates: - A couple of additional bug fixes for the AC'97 refactoring. - Some fixes for the ADAU1701 driver. ---------------------------------------------------------------- Daniel Mack (2): ASoC: adau1701: more direct regmap usage ASoC: adau1701: remove control_data assignment Kevin Hilman (2): ASoC: pxa2xx: fixup multi-platform AC'97 build failures ASoC: ac97: fixup multi-platform AC'97 module build failure Mark Brown (6): Merge remote-tracking branch 'asoc/topic/adau1701' into asoc-next Merge remote-tracking branch 'asoc/topic/adsp' into asoc-next Merge remote-tracking branch 'asoc/topic/ext' into asoc-next Merge remote-tracking branch 'asoc/topic/tas5086' into asoc-next Merge remote-tracking branch 'asoc/topic/twl6040' into asoc-next Merge remote-tracking branch 'asoc/topic/x86' into asoc-next Thierry Reding (2): ASoC: tegra20-ac97: Remove duplicate error message ASoC: tegra20-ac97: Remove unused variable .../devicetree/bindings/sound/adi,adau1701.txt | 35 ++ include/linux/mfd/twl6040.h | 7 + include/sound/soc.h | 4 +- sound/soc/au1x/ac97c.c | 21 +- sound/soc/au1x/psc-ac97.c | 33 +- sound/soc/blackfin/bf5xx-ac97.c | 29 +- sound/soc/cirrus/ep93xx-ac97.c | 12 +- sound/soc/codecs/88pm860x-codec.c | 6 +- sound/soc/codecs/ac97.c | 7 +- sound/soc/codecs/ad1980.c | 12 +- sound/soc/codecs/adau1701.c | 321 +++++++++++--- sound/soc/codecs/stac9766.c | 26 +- sound/soc/codecs/tlv320aic3x.c | 6 +- sound/soc/codecs/twl6040.c | 109 ++++- sound/soc/codecs/wm8400.c | 9 +- sound/soc/codecs/wm8903.c | 6 +- sound/soc/codecs/wm8904.c | 9 +- sound/soc/codecs/wm8990.c | 11 +- sound/soc/codecs/wm8991.h | 9 +- sound/soc/codecs/wm8994.c | 12 +- sound/soc/codecs/wm8995.h | 7 +- sound/soc/codecs/wm9705.c | 16 +- sound/soc/codecs/wm9712.c | 18 +- sound/soc/codecs/wm9713.c | 18 +- sound/soc/codecs/wm_adsp.c | 459 ++++++++++++++++++++- sound/soc/codecs/wm_adsp.h | 13 +- sound/soc/codecs/wm_hubs.c | 6 +- sound/soc/fsl/imx-ssi.c | 11 +- sound/soc/fsl/mpc5200_psc_ac97.c | 10 +- sound/soc/mid-x86/mfld_machine.c | 32 +- sound/soc/nuc900/nuc900-ac97.c | 60 +-- sound/soc/pxa/pxa2xx-ac97.c | 10 +- sound/soc/pxa/pxa2xx-ac97.h | 3 - sound/soc/samsung/ac97.c | 42 +- sound/soc/sh/hac.c | 8 +- sound/soc/soc-core.c | 17 + sound/soc/tegra/tegra20_ac97.c | 67 ++- sound/soc/txx9/txx9aclc-ac97.c | 17 +- 38 files changed, 1124 insertions(+), 374 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/adi,adau1701.txt

2 1

[alsa-devel] [PATCH 1/2] ASoC: adau1701: more direct regmap usage
by Daniel Mack 30 Jun '13

30 Jun '13

Replace calls to snd_soc_update_bits() with regmap_update_bits(). Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/soc/codecs/adau1701.c | 19 ++++++++++++------- 1 file changed, 12 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index ac8b2c4..ce56065 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -334,7 +334,7 @@ static int adau1701_set_capture_pcm_format(struct snd_soc_codec *codec, mask |= ADAU1701_SEROCTL_MSB_DEALY_MASK; } - snd_soc_update_bits(codec, ADAU1701_SEROCTL, mask, val); + regmap_update_bits(adau1701->regmap, ADAU1701_SEROCTL, mask, val); return 0; } @@ -362,7 +362,7 @@ static int adau1701_set_playback_pcm_format(struct snd_soc_codec *codec, return -EINVAL; } - snd_soc_update_bits(codec, ADAU1701_SERICTL, + regmap_update_bits(adau1701->regmap, ADAU1701_SERICTL, ADAU1701_SERICTL_MODE_MASK, val); return 0; @@ -403,7 +403,7 @@ static int adau1701_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - snd_soc_update_bits(codec, ADAU1701_DSPCTRL, + regmap_update_bits(adau1701->regmap, ADAU1701_DSPCTRL, ADAU1701_DSPCTRL_SR_MASK, val); format = params_format(params); @@ -489,6 +489,7 @@ static int adau1701_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { unsigned int mask = ADAU1701_AUXNPOW_VBPD | ADAU1701_AUXNPOW_VRPD; + struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); switch (level) { case SND_SOC_BIAS_ON: @@ -497,11 +498,13 @@ static int adau1701_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: /* Enable VREF and VREF buffer */ - snd_soc_update_bits(codec, ADAU1701_AUXNPOW, mask, 0x00); + regmap_update_bits(adau1701->regmap, + ADAU1701_AUXNPOW, mask, 0x00); break; case SND_SOC_BIAS_OFF: /* Disable VREF and VREF buffer */ - snd_soc_update_bits(codec, ADAU1701_AUXNPOW, mask, mask); + regmap_update_bits(adau1701->regmap, + ADAU1701_AUXNPOW, mask, mask); break; } @@ -513,6 +516,7 @@ static int adau1701_digital_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; unsigned int mask = ADAU1701_DSPCTRL_DAM; + struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); unsigned int val; if (mute) @@ -520,7 +524,7 @@ static int adau1701_digital_mute(struct snd_soc_dai *dai, int mute) else val = mask; - snd_soc_update_bits(codec, ADAU1701_DSPCTRL, mask, val); + regmap_update_bits(adau1701->regmap, ADAU1701_DSPCTRL, mask, val); return 0; } @@ -542,7 +546,8 @@ static int adau1701_set_sysclk(struct snd_soc_codec *codec, int clk_id, return -EINVAL; } - snd_soc_update_bits(codec, ADAU1701_OSCIPOW, ADAU1701_OSCIPOW_OPD, val); + regmap_update_bits(adau1701->regmap, ADAU1701_OSCIPOW, + ADAU1701_OSCIPOW_OPD, val); adau1701->sysclk = freq; return 0; -- 1.8.1.4

3 4

[alsa-devel] [PATCH - oss plugin update revised 2/2] oss: Add user variables to set period and buffer config
by alex.d.wiggins＠gmail.com 30 Jun '13

30 Jun '13

From: Alex Wiggins <alex.d.wiggins(a)gmail.com> Added 3 user variables to control the OSS period and buffer config: - period_size (integer): if not set then queries OSS for period size (by using SNDCTL_DSP_GETBLKSIZE), if positive then sets the OSS period size (SND_PCM_IOPLUG_HW_PERIOD_BYTES), if negative then allows any power of 2 period size. - periods (integer): sets the number of OSS periods (SND_PCM_IOPLUG_HW_PERIODS); - buffer_size (integer): sets the OSS buffer size (SND_PCM_IOPLUG_HW_BUFFER_BYTES). Note that ALSA performs sanity checks to prevent conflicts. Signed-off-by: Alex Wiggins <alex.d.wiggins(a)gmail.com> diff --git a/oss/pcm_oss.c b/oss/pcm_oss.c index d5b5bc4..a92c3e9 100644 --- a/oss/pcm_oss.c +++ b/oss/pcm_oss.c @@ -60,6 +60,12 @@ typedef struct snd_pcm_oss { unsigned int period_shift; unsigned int periods; unsigned int frame_bytes; + /* user variables set using ALSA config files */ + /* included here as this snd_pcm_oss_t struct + is stored in io->private_data */ + long int user_period_size; + long int user_periods; + long int user_buffer_size; } snd_pcm_oss_t; static snd_pcm_sframes_t oss_write(snd_pcm_ioplug_t *io, @@ -408,6 +414,7 @@ static int oss_hw_constraint(snd_pcm_oss_t *oss) unsigned int nchannels; unsigned int channel[6]; /* period and buffer bytes must be power of two */ + /* minimum: 2^8 = 256; maximum: 2^23 = 8388608 */ static const unsigned int bytes_list[] = { 1U<<8, 1U<<9, 1U<<10, 1U<<11, 1U<<12, 1U<<13, 1U<<14, 1U<<15, 1U<<16, 1U<<17, 1U<<18, 1U<<19, 1U<<20, 1U<<21, 1U<<22, 1U<<23 @@ -500,20 +507,97 @@ static int oss_hw_constraint(snd_pcm_oss_t *oss) if (err < 0) return err; - /* period size (in power of two) */ - err = snd_pcm_ioplug_set_param_list(io, SND_PCM_IOPLUG_HW_PERIOD_BYTES, - ARRAY_SIZE(bytes_list), bytes_list); - if (err < 0) - return err; - /* periods */ - err = snd_pcm_ioplug_set_param_minmax(io, SND_PCM_IOPLUG_HW_PERIODS, 2, 1024); - if (err < 0) - return err; - /* buffer size (in power of two) */ - err = snd_pcm_ioplug_set_param_list(io, SND_PCM_IOPLUG_HW_BUFFER_BYTES, - ARRAY_SIZE(bytes_list), bytes_list); - if (err < 0) - return err; + /* period and buffer sizes: default behaviour below is to + query period size from OSS */ + + /* consider user set paramaters + (period_size; periods; buffer_size) */ + + /* note: ALSA performs sanity checks + to prevent conflicting parameters */ + + /* period_size */ + if (oss->user_period_size < 0) { + /* period_size set to negative value: + allow any power of 2 period size */ + err = snd_pcm_ioplug_set_param_list(io, + SND_PCM_IOPLUG_HW_PERIOD_BYTES, + ARRAY_SIZE(bytes_list), bytes_list); + if (err < 0) + return err; + } else { + if (oss->user_period_size > 0) { + /* period_size set to positive value: + use user set period size */ + const unsigned int user_period_size + = oss->user_period_size; + err = snd_pcm_ioplug_set_param_list(io, + SND_PCM_IOPLUG_HW_PERIOD_BYTES, 1, + &user_period_size); + } else { + /* query OSS for period size (called fragment size) */ + unsigned int fragsize; + if (ioctl(oss->fd, SNDCTL_DSP_GETBLKSIZE, &fragsize) + < 0) + fprintf(stderr, +"*** alsa-plugins-oss: error querying SNDCTL_DSP_GETBLKSIZE.\n"); + + /* note: second query of SNDCTL_DSP_GETBLKSIZE + can give different result + (i.e. 4096 instead of 2048) */ + /* in oss_audio_core.c getblksize function, + 4096 is the default return value: + so second time probably not reliable */ + + /* period size (in power of two) */ + unsigned int x = fragsize; + /* check if fragsize is a power of two */ + if ((x != 0) && ((x & (~x + 1)) == x)) + err = snd_pcm_ioplug_set_param_list(io, + SND_PCM_IOPLUG_HW_PERIOD_BYTES, + 1, &fragsize); + else + err = snd_pcm_ioplug_set_param_list(io, + SND_PCM_IOPLUG_HW_PERIOD_BYTES, + ARRAY_SIZE(bytes_list), bytes_list); + if (err < 0) + return err; + } + } + + /* periods (number of periods per buffer) */ + if (oss->user_periods > 0) { + /* force number of periods to user set periods */ + const unsigned int user_periods = oss->user_periods; + err = snd_pcm_ioplug_set_param_list(io, + SND_PCM_IOPLUG_HW_PERIODS, 1, &user_periods); + if (err < 0) + return err; + } else { + /* allow any number of periods between 2 and 1024 inclusive + (ALSA can choose) */ + err = snd_pcm_ioplug_set_param_minmax(io, + SND_PCM_IOPLUG_HW_PERIODS, 2, 1024); + if (err < 0) + return err; + } + + /* buffer_size */ + if (oss->user_buffer_size > 0) { + /* force buffer size to user set buffer_size */ + const unsigned int user_buffer_size = oss->user_buffer_size; + err = snd_pcm_ioplug_set_param_list(io, + SND_PCM_IOPLUG_HW_BUFFER_BYTES, 1, &user_buffer_size); + if (err < 0) + return err; + } else { + /* buffer size (in power of two) */ + err = snd_pcm_ioplug_set_param_list(io, + SND_PCM_IOPLUG_HW_BUFFER_BYTES, + ARRAY_SIZE(bytes_list), bytes_list); + if (err < 0) + return err; + } return 0; } @@ -558,6 +642,10 @@ SND_PCM_PLUGIN_DEFINE_FUNC(oss) const char *device = "/dev/dsp"; int err; snd_pcm_oss_t *oss; + /* default values for user variables*/ + long int user_period_size = 0; + long int user_periods = 0; + long int user_buffer_size = 0; snd_config_for_each(i, next, conf) { snd_config_t *n = snd_config_iterator_entry(i); @@ -573,6 +661,30 @@ SND_PCM_PLUGIN_DEFINE_FUNC(oss) } continue; } + + /* user variables */ + if (strcmp(id, "period_size") == 0) { + if (snd_config_get_integer(n, &user_period_size) < 0) { + SNDERR("Invalid type for %s", id); + return -EINVAL; + } + continue; + } + if (strcmp(id, "periods") == 0) { + if (snd_config_get_integer(n, &user_periods) < 0) { + SNDERR("Invalid type for %s", id); + return -EINVAL; + } + continue; + } + if (strcmp(id, "buffer_size") == 0) { + if (snd_config_get_integer(n, &user_buffer_size) < 0) { + SNDERR("Invalid type for %s", id); + return -EINVAL; + } + continue; + } + SNDERR("Unknown field %s", id); return -EINVAL; } @@ -597,6 +709,11 @@ SND_PCM_PLUGIN_DEFINE_FUNC(oss) goto error; } + /* save user variables from ALSA config files */ + oss->user_period_size = user_period_size; + oss->user_periods = user_periods; + oss->user_buffer_size = user_buffer_size; + oss->io.version = SND_PCM_IOPLUG_VERSION; oss->io.name = "ALSA <-> OSS PCM I/O Plugin"; oss->io.poll_fd = oss->fd; -- 1.8.3.1

1 0

Re: [alsa-devel] [PATCH 0/2] ASoc: ac97: fix multi-platform build failures
by Mark Brown 29 Jun '13

29 Jun '13

On Fri, Jun 28, 2013 at 11:17:47AM -0700, Kevin Hilman wrote: > Hi Mark, > > When building next-20130628, I noticed a few build errors for > cm_x300_defconfig that were introduced by the new multi-platform AC'97 > support. Applied both, thanks.

1 0

Re: [alsa-devel] [PATCH 1/2] ASoC: tegra20-ac97: Remove duplicate error message
by Mark Brown 29 Jun '13

29 Jun '13

On Fri, Jun 28, 2013 at 09:53:24PM +0200, Thierry Reding wrote: > devm_ioremap_resource() already outputs an error message when any of the > operations it performs fails, so the duplicate in the caller can be > removed. Applied both, thanks.

1 0

[alsa-devel] ASoC updates for v3.11
by Mark Brown 28 Jun '13

28 Jun '13

The following changes since commit 9e895ace5d82df8929b16f58e9f515f6d54ab82d: Linux 3.10-rc7 (2013-06-22 09:47:31 -1000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.11-2 for you to fetch changes up to 27516080b21cbcb936440d2a3171867860b9a881: Merge remote-tracking branch 'asoc/topic/x86' into asoc-next (2013-06-28 12:17:07 +0100) The diffstat is a bit odd I think due to the synthetic tree I generated in an effort to cut out the fixes from v3.10-rc7 from it :/ ---------------------------------------------------------------- ASoC: More updates for v3.11 Some more fixes and enhancements, and also a bunch of refectoring for AC'97 support which enables more than one AC'97 controller driver to be built in. ---------------------------------------------------------------- Daniel Mack (10): ASoC: tas5086: open-code I2C transfer routines ASoC: tas5086: add more register defines ASoC: tas5086: add DAPM mux controls ASoC: tas5086: add support for pwm start mode config ASoC: adau1701: move firmware download to adau1701_reset() ASoC: adau1701: allow configuration of PLL mode pins ASoC: adau1701: switch to direct regmap API usage ASoC: adau1701: add support for pin muxing ASoC: tas5086: fix TAS5086_CLOCK_CONTROL register size ASoC: tas5086: fix Mid-Z implementation Lars-Peter Clausen (11): ASoC: tlv320aix3x: Use SOC_SINGLE_EXT() instead of open-coding it ASoC: wm8400: Use SOC_SINGLE_EXT_TLV() instead of open-coding it ASoC: wm8903: Use SOC_SINGLE_EXT() instead of open-coding it ASoC: wm8904: Use SOC_SINGLE_EXT() instead of open-coding it ASoC: wm8990: Use SOC_SINGLE_EXT_TLV() instead of open-coding it ASoC: wm8991: Use SOC_SINGLE_EXT_TLV() instead of open-coding it ASoC: wm8994: Use SOC_SINGLE_EXT() instead of open-coding it ASoC: wm8995: Use SOC_SINGLE_EXT() instead of open-coding it ASoC: wm_hubs: Use SOC_SINGLE_EXT() instead of open-coding it ASoC: wm_adsp: Use SND_SOC_DAPM_PGA_E() instead of open-coding it ASoC: 88pm860x: Use SND_SOC_DAPM_PGA_E() instead of open-coding it Mark Brown (25): ASoC: stac9766: Remove version number ASoC: wm9705: Remove noisy print on boot ASoC: samsung-ac97: Use devm_clk_get() ASoC: samsung-ac97: Convert to devm_ioremap_resource() ASoC: psc-ac97: Use devm_ioremap_resource() ASoC: psc-ac97: Convert to module_platform_driver() ASoC: ac97c: Use module_platform_driver() ASoC: bf5xx-ac97: Convert to devm_gpio_request_one() ASoC: ep93xx: Remove redundant dev_set_drvdata() calls ASoC: nuc900-ac97: Convert to use devm_ APIs ASoC: tegra20-ac97: Convert to devm_clk_get() ASoC: tegra20-ac97: Convert to devm_ioremap_resource() ASoC: tegra-ac97: Do common and clock init prior to component registration ASoC: txx9aclc_ac97: Convert to devm_ioremap_resource() ASoC: wm9705: Remove noisy print on boot Merge branch 'topic/stac9766' of git://git.kernel.org/.../broonie/sound into asoc-ac97 Merge branch 'topic/wm9705' of git://git.kernel.org/.../broonie/sound into asoc-ac97 ASoC: ac97: Support multi-platform AC'97 ASoC: mfld: Remove unused variable Merge remote-tracking branch 'asoc/topic/adau1701' into asoc-next Merge remote-tracking branch 'asoc/topic/adsp' into asoc-next Merge remote-tracking branch 'asoc/topic/ext' into asoc-next Merge remote-tracking branch 'asoc/topic/tas5086' into asoc-next Merge remote-tracking branch 'asoc/topic/twl6040' into asoc-next Merge remote-tracking branch 'asoc/topic/x86' into asoc-next Peter Ujfalusi (4): ASoC: twl6040: Drop using devm_request_threaded_irq() mfd: twl6040: Update register bit definitions ASoC: twl6040: Assign id for each DAI ASoC: twl6040: Add digital mute support Wei Yongjun (1): ASoC: mid-x86: Convert to use devm_* APIs .../devicetree/bindings/sound/adi,adau1701.txt | 35 ++ .../devicetree/bindings/sound/ti,tas5086.txt | 11 + include/linux/mfd/twl6040.h | 7 + include/sound/soc.h | 4 +- sound/soc/au1x/ac97c.c | 21 +- sound/soc/au1x/psc-ac97.c | 33 +- sound/soc/blackfin/bf5xx-ac97.c | 29 +- sound/soc/cirrus/ep93xx-ac97.c | 12 +- sound/soc/codecs/88pm860x-codec.c | 6 +- sound/soc/codecs/ac97.c | 7 +- sound/soc/codecs/ad1980.c | 12 +- sound/soc/codecs/adau1701.c | 302 ++++++++++++-- sound/soc/codecs/stac9766.c | 26 +- sound/soc/codecs/tas5086.c | 330 ++++++++++++++- sound/soc/codecs/tlv320aic3x.c | 6 +- sound/soc/codecs/twl6040.c | 109 ++++- sound/soc/codecs/wm8400.c | 9 +- sound/soc/codecs/wm8903.c | 6 +- sound/soc/codecs/wm8904.c | 9 +- sound/soc/codecs/wm8990.c | 11 +- sound/soc/codecs/wm8991.h | 9 +- sound/soc/codecs/wm8994.c | 12 +- sound/soc/codecs/wm8995.h | 7 +- sound/soc/codecs/wm9705.c | 16 +- sound/soc/codecs/wm9712.c | 18 +- sound/soc/codecs/wm9713.c | 18 +- sound/soc/codecs/wm_adsp.c | 459 ++++++++++++++++++++- sound/soc/codecs/wm_adsp.h | 13 +- sound/soc/codecs/wm_hubs.c | 6 +- sound/soc/fsl/imx-ssi.c | 11 +- sound/soc/fsl/mpc5200_psc_ac97.c | 10 +- sound/soc/mid-x86/mfld_machine.c | 32 +- sound/soc/nuc900/nuc900-ac97.c | 60 +-- sound/soc/pxa/pxa2xx-ac97.c | 8 +- sound/soc/samsung/ac97.c | 42 +- sound/soc/sh/hac.c | 8 +- sound/soc/soc-core.c | 16 + sound/soc/tegra/tegra20_ac97.c | 66 ++- sound/soc/txx9/txx9aclc-ac97.c | 17 +- 39 files changed, 1447 insertions(+), 366 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/adi,adau1701.txt

2 1

[alsa-devel] [PATCH 1/2] ASoC: tas5086: fix TAS5086_CLOCK_CONTROL register size
by Daniel Mack 28 Jun '13

28 Jun '13

The TAS5086_CLOCK_CONTROL also has a size of 1 byte. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/soc/codecs/tas5086.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index fec9a28..84c3f27 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -130,7 +130,7 @@ static const struct reg_default tas5086_reg_defaults[] = { static int tas5086_register_size(struct device *dev, unsigned int reg) { switch (reg) { - case TAS5086_DEV_ID ... TAS5086_BKNDERR: + case TAS5086_CLOCK_CONTROL ... TAS5086_BKNDERR: return 1; case TAS5086_INPUT_MUX: case TAS5086_PWM_OUTPUT_MUX: -- 1.8.1.4

2 2

[alsa-devel] [PATCH] ASoC: mfld: Remove unused variable
by Mark Brown 28 Jun '13

28 Jun '13

From: Mark Brown <broonie(a)linaro.org> Reported-by: Fengguang Wu <fengguang.wu(a)intel.com> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/mid-x86/mfld_machine.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c index aec29a8..ee36384 100644 --- a/sound/soc/mid-x86/mfld_machine.c +++ b/sound/soc/mid-x86/mfld_machine.c @@ -412,8 +412,6 @@ static int snd_mfld_mc_probe(struct platform_device *pdev) static int snd_mfld_mc_remove(struct platform_device *pdev) { - struct mfld_mc_private *mc_drv_ctx = platform_get_drvdata(pdev); - pr_debug("snd_mfld_mc_remove called\n"); snd_soc_unregister_card(&snd_soc_card_mfld); return 0; -- 1.8.3.1

1 0

[alsa-devel] [PATCH 1/2] ALSA: hda - Fix the max length of control name in generic parser
by Takashi Iwai 28 Jun '13

28 Jun '13

add_control_with_pfx() in hda_generic.c assumes a shorter name string for the control element, and this resulted in the truncation of the long but valid string like "Headphone Surround Switch" in the middle. This patch aligns the max size to the actual limit of snd_ctl_elem_id, 44. Cc: <stable(a)vger.kernel.org> [v3.9+] Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 1485d87..6460fc5 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -854,7 +854,7 @@ static int add_control_with_pfx(struct hda_gen_spec *spec, int type, const char *pfx, const char *dir, const char *sfx, int cidx, unsigned long val) { - char name[32]; + char name[44]; snprintf(name, sizeof(name), "%s %s %s", pfx, dir, sfx); if (!add_control(spec, type, name, cidx, val)) return -ENOMEM; -- 1.8.3.1

1 1

[alsa-devel] [PATCH] ALSA: hda - Guess what, it's two more Dell headset mic quirks
by David Henningsson 28 Jun '13

28 Jun '13

Add two more machines that need quirks for headset mics to work. Tested-by: Shawn Wang <shawn.wang(a)canonical.com> BugLink: https://bugs.launchpad.net/bugs/1195636 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6d7041d..86c2406 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3506,6 +3506,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x05f5, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05f6, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05f8, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05f9, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05fb, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0606, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0608, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0609, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ALSA: hda - Yet another Dell headset mic quirk
by David Henningsson 28 Jun '13

28 Jun '13

This quirk is needed for the headset mic to work on this Dell machine. BugLink: https://bugs.launchpad.net/bugs/1195597 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ae12111..6d7041d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3509,6 +3509,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0606, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0608, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0609, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0613, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), SND_PCI_QUIRK(0x103c, 0x18e6, "HP", ALC269_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x1973, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1), -- 1.7.9.5

2 1

[alsa-devel] C-Media 6632A unsupported (asus xonar u7)
by Julius von kohout 28 Jun '13

28 Jun '13

Hi ive got a new usb soundcard with the brand new C-Media 6632A chip and want to help with debugging. xonar u7 in usb 2.0 mode C-Media 6632A Ubuntu 13.04 julius@R850:~$ alsactl init 1 Found hardware: "USB-Audio" "USB Mixer" "USB1043:857c" "" "" Hardware is initialized using a generic method julius@R850:~$ alsamixer -c 1 cannot load mixer controls: Broken pipe alsa hardware volume (numid=2,iface=MIXER,name='Speaker Playback Volume') level on start is way to low and can only be adjusted via QasHctl. The volume knob on soundcard works for pulseaudio volume. spdif can only be used via direct alsa access without pulseaudio. alsa-info script results http://pastebin.ca/2409660 julius@R850:~$ amixer -c 1 controls numid=7,iface=MIXER,name='PCM Capture Source' numid=5,iface=MIXER,name='PCM Capture Switch' numid=6,iface=MIXER,name='PCM Capture Switch',index=1 numid=4,iface=MIXER,name='IEC958 In Playback Switch' numid=3,iface=MIXER,name='Input Gain Pad Control' numid=1,iface=MIXER,name='Speaker Playback Switch' numid=2,iface=MIXER,name='Speaker Playback Volume' [ 8219.399637] usb 2-1.2.3: new high-speed USB device number 22 using ehci-pci [ 8219.513315] usb 2-1.2.3: New USB device found, idVendor=1043, idProduct=857c [ 8219.513327] usb 2-1.2.3: New USB device strings: Mfr=1, Product=2, SerialNumber=0 [ 8219.513333] usb 2-1.2.3: Product: Xonar U7 [ 8219.513338] usb 2-1.2.3: Manufacturer: ASUS [ 8219.609328] input: ASUS Xonar U7 as /devices/pci0000:00/0000:00:1d.0/usb2/2-1/2-1.2/2-1.2.3/2-1.2.3:1.4/input/input29 [ 8219.609743] hid-generic 0003:1043:857C.0016: input,hidraw4: USB HID v1.00 Device [ASUS Xonar U7] on usb-0000:00:1d.0-1.2.3/input4 [ 8219.676164] cannot get ctl value: req = 0x81, wValue = 0x0, wIndex = 0x1400, type = 3 [ 8219.697280] cannot get ctl value: req = 0x81, wValue = 0x0, wIndex = 0x1400, type = 3 [ 8219.718770] cannot get ctl value: req = 0x81, wValue = 0x0, wIndex = 0x1400, type = 3 [ 8219.738130] cannot get ctl value: req = 0x81, wValue = 0x0, wIndex = 0x1400, type = 3 [ 8219.757999] cannot get ctl value: req = 0x81, wValue = 0x0, wIndex = 0x1400, type = 3 [ 8219.779861] cannot get ctl value: req = 0x81, wValue = 0x0, wIndex = 0x1400, type = 3 [ 8219.800222] cannot get ctl value: req = 0x81, wValue = 0x0, wIndex = 0x1400, type = 3 [ 8219.823964] cannot get ctl value: req = 0x81, wValue = 0x0, wIndex = 0x1400, type = 3 [ 8520.766440] cannot get ctl value: req = 0x81, wValue = 0x0, wIndex = 0x1400, type = 3 [ 8539.332352] cannot get ctl value: req = 0x81, wValue = 0x0, wIndex = 0x1400, type = 3 julius@R850:~$ cat /proc/asound/version Advanced Linux Sound Architecture Driver Version k3.8.0-25-generic. julius@R850:~$ cat /proc/asound/cards 0 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xc4820000 irq 50 1 [U7 ]: USB-Audio - Xonar U7 ASUS Xonar U7 at usb-0000:00:1d.0-1.2.3, high speed julius@R850:~$ lsmod | grep snd snd_hda_codec_hdmi 36913 1 snd_hda_codec_realtek 78399 1 snd_usb_audio 140685 0 snd_usbmidi_lib 24938 1 snd_usb_audio snd_hda_intel 39619 3 snd_hda_codec 136453 3 snd_hda_codec_realtek,snd_hda_codec_hdmi,snd_hda_intel snd_hwdep 13602 2 snd_usb_audio,snd_hda_codec snd_pcm 97451 4 snd_usb_audio,snd_hda_codec_hdmi,snd_hda_codec,snd_hda_intel snd_page_alloc 18710 2 snd_pcm,snd_hda_intel snd_seq_midi 13324 0 snd_seq_midi_event 14899 1 snd_seq_midi snd_rawmidi 30180 2 snd_usbmidi_lib,snd_seq_midi snd_seq 61554 2 snd_seq_midi_event,snd_seq_midi snd_seq_device 14497 3 snd_seq,snd_rawmidi,snd_seq_midi snd_timer 29425 2 snd_pcm,snd_seq snd 68876 18 snd_hda_codec_realtek,snd_usb_audio,snd_hwdep,snd_timer,snd_hda_codec_hdmi,snd_pcm,snd_seq,snd_rawmidi,snd_usbmidi_lib,snd_hda_codec,snd_hda_intel,snd_seq_device soundcore 12680 1 snd julius@R850:~$ lsusb Bus 001 Device 002: ID 8087:0024 Intel Corp. Integrated Rate Matching Hub Bus 002 Device 002: ID 8087:0024 Intel Corp. Integrated Rate Matching Hub Bus 001 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 002 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 003 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 004 Device 001: ID 1d6b:0003 Linux Foundation 3.0 root hub Bus 002 Device 021: ID 18d1:4ee1 Google Inc. Bus 002 Device 015: ID 0424:2514 Standard Microsystems Corp. USB 2.0 Hub Bus 002 Device 016: ID 046d:c245 Logitech, Inc. Bus 002 Device 017: ID 05a4:9881 Ortek Technology, Inc. Bus 002 Device 022: ID 1043:857c iCreate Technologies Corp. Bus 002 Device 020: ID 046a:0180 Cherry GmbH julius@R850:~$ dpkg -l | grep alsa ii alsa-base 1.0.25+dfsg-0ubuntu4 all ALSA driver configuration files ii alsa-utils 1.0.25-4ubuntu2 amd64 Utilities for configuring and using ALSA ii bluez-alsa:amd64 4.101-0ubuntu8b1 amd64 Bluetooth ALSA support ii gstreamer0.10-alsa:amd64 0.10.36-1.1ubuntu1 amd64 GStreamer plugin for ALSA ii gstreamer1.0-alsa:amd64 1.0.6-1 amd64 GStreamer plugin for ALSA

2 1

[alsa-devel] How to clear the ALSA audio settings remembered by system?
by Lin, Mengdong 27 Jun '13

27 Jun '13

Hi, Could anyone tell me how to clear the ALSA audio settings remembered by system? I change some audio codec settings by amixer. And it seems the ALSA has remembered these settings and apply on next boot. And these will change my test result. How can I let system forgot these changes? Many thanks Mengdong

2 2

Re: [alsa-devel] [PATCH 3/6] sound/pci:trivial: replace numeric with standard PM state macros
by Takashi Iwai 27 Jun '13

27 Jun '13

At Thu, 27 Jun 2013 20:55:11 +0800, Yijing Wang wrote: > > Use standard PM state macros PCI_Dx instead of numeric 0/1/2.. > > Signed-off-by: Yijing Wang <wangyijing(a)huawei.com> > Cc: Jaroslav Kysela <perex(a)perex.cz> > Cc: Takashi Iwai <tiwai(a)suse.de> > Cc: Jiri Kosina <trivial(a)kernel.org> > Cc: Bill Pemberton <wfp5p(a)virginia.edu> > Cc: alsa-devel(a)alsa-project.org > Cc: linux-kernel(a)vger.kernel.org Thanks, applied. Takashi > --- > sound/pci/cs4281.c | 2 +- > sound/pci/ens1370.c | 2 +- > sound/pci/ymfpci/ymfpci_main.c | 2 +- > 3 files changed, 3 insertions(+), 3 deletions(-) > > diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c > index 6a86950..2f6a4b6 100644 > --- a/sound/pci/cs4281.c > +++ b/sound/pci/cs4281.c > @@ -1312,7 +1312,7 @@ static int snd_cs4281_free(struct cs4281 *chip) > /* Sound System Power Management - Turn Everything OFF */ > snd_cs4281_pokeBA0(chip, BA0_SSPM, 0); > /* PCI interface - D3 state */ > - pci_set_power_state(chip->pci, 3); > + pci_set_power_state(chip->pci, PCI_D3hot); > > if (chip->irq >= 0) > free_irq(chip->irq, chip); > diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c > index db2dc83..b30afd1 100644 > --- a/sound/pci/ens1370.c > +++ b/sound/pci/ens1370.c > @@ -1939,7 +1939,7 @@ static int snd_ensoniq_free(struct ensoniq *ensoniq) > #endif > if (ensoniq->irq >= 0) > synchronize_irq(ensoniq->irq); > - pci_set_power_state(ensoniq->pci, 3); > + pci_set_power_state(ensoniq->pci, PCI_D3hot); > __hw_end: > #ifdef CHIP1370 > if (ensoniq->dma_bug.area) > diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c > index 22056c5..d591c15 100644 > --- a/sound/pci/ymfpci/ymfpci_main.c > +++ b/sound/pci/ymfpci/ymfpci_main.c > @@ -2258,7 +2258,7 @@ static int snd_ymfpci_free(struct snd_ymfpci *chip) > /* FIXME: temporarily disabled, otherwise we cannot fire up > * the chip again unless reboot. ACPI bug? > */ > - pci_set_power_state(chip->pci, 3); > + pci_set_power_state(chip->pci, PCI_D3hot); > #endif > > #ifdef CONFIG_PM_SLEEP > -- > 1.7.1 > >

1 0

Re: [alsa-devel] [PATCH 2/2] ASoC: tegra20-ac97: Convert to devm_ioremap_resource()
by Mark Brown 27 Jun '13

27 Jun '13

On Thu, Jun 27, 2013 at 12:22:12PM +0200, Thierry Reding wrote: > Can you please remove this error message as well, please? A message to > the same effect is already output by devm_ioremap_resource(). Other than > that, looks good: Too late, send an incremental patch please.

1 0

[alsa-devel] [RFC PATCH v2 00/12] ARM: s3c64xx: Let amba-pl08x driver handle DMA
by Tomasz Figa 27 Jun '13

27 Jun '13

This is another version of my patches extending support of amba-pl08x DMA engine driver to PL080S DMA engine (PL080 modified by Samsung) found in Samsung S3C64xx SoCs. Changes since RFC v1: - Returned to original way of storing quirks as booleans, as suggested by Russell, Linus and Arnd. - Added reg_config field to pl08x_phy_chan struct, which stores variant-specific address of channel config register, as suggested by Russell. - Simplified handling of extended maximum transfer size of PL080S (no more conditional passing of 0 as length to pl08x_cctl_bits()). - Reworked LLI handling in the driver to stop casting arbitrary memory to a struct and allow different word count of LLI entry, as suggested by Linus. - Removed AMBA ID override from S3C64xx PL080 initialization code. - Fixed brokenness of Samsung DMA wrapper API, which caused cyclic buffers to be queued multiple times when DMA engine is used. - Included patch adding clock aliases for DMA engines (depends on Common Clock Framework driver for S3C64xx). - Fixed several minor stylistic issues. For reference, here is the original description of the series: One of the biggest roadblocks on the way of S3C64xx to DeviceTree support is its DMA driver, which is completely platform-specific and provides private API (s3c-dma), not even saying that its design is completely against multiplatform-awareness. The DMA controller present on this SoC series is a custom variant of ARM PrimeCell PL080 modified by Samsung to add some extra features. It is mostly compatible with original PL080, except: - CH_CONTROL2 register is added between CH_CONTROL and CH_CONFIG, - offset of CH_CONFIG register is different, - transfer size field is moved from CH_CONTROL to CH_CONTROL2, - transfer size field is extended to 24 bits, allowing much bigger single transfer, - LLI consists of one more word, to account for CH_CONTROL2 register. Since all the rest is fully compatible with standard PL080 there is no point in having separate driver just for this single variant, so I decided to look into adding support for it to the amba-pl08x driver. There was already some attempt to achieve this before, but this was before Russel's big rework of the driver to use virtual channels, making the old patches being not much of use. This RFC series is a proof of concept that I managed to make during last days of hacking. Except one patch adding clkdev lookup to clock driver (which is being replaced with a CCF-compliant driver ATM), this is enough to get memcpy and slave transfers to work on S3C64xx. I have tested this on Mini6410 and SMDK6410 boards using dmatest for memcpy and Samsung I2S with madplay/aplay for slave transfers. Unfortunately I do not have access to other platforms with PL08x so I could not test for any regressions introduced on them. Credits for two patches go to Alban Bedel, who made a series fixing this driver to make it usable with audio drivers. I rebased his patches on top of mine and corrected coding style a bit. OK, that's all. Any comments are welcome. Feel free to start throwing eggs and tomatoes if you find this awful, but I won't be upset if I get some Tested-by or Acked-by as well. ;) Alban Bedel (2): dmaengine: PL08x: Fix reading the byte count in cctl dmaengine: PL08x: Add cyclic transfer support Tomasz Figa (10): dmaengine: PL08x: Refactor pl08x_getbytes_chan() to lower indentation dmaengine: PL08x: Add support for different offset of CONFIG register dmaengine: PL08x: Rework LLI handling to be less fragile dmaengine: PL08x: Add support for PL080S variant dmaengine: PL08x: Add support for different maximum transfer size ASoC: Samsung: Do not queue cyclic buffers multiple times clk: samsung: s3c64xx: Add aliases for DMA clocks spi: s3c64xx: Do not require legacy DMA API in case of S3C64XX ASoC: Samsung: Do not require legacy DMA API in case of S3C64XX ARM: s3c64xx: Add support for DMA using generic amba-pl08x driver arch/arm/Kconfig | 1 + arch/arm/mach-s3c64xx/Kconfig | 8 +- arch/arm/mach-s3c64xx/Makefile | 1 + arch/arm/mach-s3c64xx/common.h | 5 + arch/arm/mach-s3c64xx/include/mach/dma.h | 65 +++++ arch/arm/mach-s3c64xx/pl080.c | 244 ++++++++++++++++++ arch/arm/plat-samsung/s3c-dma-ops.c | 13 +- drivers/clk/samsung/clk-s3c64xx.c | 2 + drivers/dma/amba-pl08x.c | 426 ++++++++++++++++++++++--------- drivers/spi/Kconfig | 2 +- include/linux/amba/pl080.h | 1 + sound/soc/samsung/Kconfig | 2 +- sound/soc/samsung/dma.c | 7 + 13 files changed, 651 insertions(+), 126 deletions(-) create mode 100644 arch/arm/mach-s3c64xx/pl080.c -- 1.8.2.1

5 29

[alsa-devel] [PATCH 0/4] ASoC: twl6040: Fixes and updates
by Peter Ujfalusi 27 Jun '13

27 Jun '13

Hi, Updates for the twl6040 codec: - Fix soc probe when the machine driver is removed and loaded again without removing the codec driver (was failing since the IRQ has been already requested) - Playback pop noise reduction on stream start and stop with implementing the digital_mute callback and related tracking of the mute for different paths. Regards, Peter --- Peter Ujfalusi (4): ASoC: twl6040: Drop using devm_request_threaded_irq() MFD: twl6040: Update register bit definitions ASoC: twl6040: Assign id for each DAI ASoC: twl6040: Add digital mute support include/linux/mfd/twl6040.h | 7 +++ sound/soc/codecs/twl6040.c | 109 +++++++++++++++++++++++++++++++++++++++++++- 2 files changed, 114 insertions(+), 2 deletions(-) -- 1.8.2.1

3 8

[alsa-devel] [PATCH] ASoC: ac97: Support multi-platform AC'97
by Mark Brown 26 Jun '13

26 Jun '13

From: Mark Brown <broonie(a)linaro.org> Currently we can only have a single platform built in with AC'97 support due to the use of a global variable to provide the bus operations. Fix this by making that variable a pointer and having the bus drivers set the operations prior to registering. This is not a particularly good or nice approach but it avoids blocking multiplatform and a real fix involves fixing the fairly deep problems with AC'97 support - we should be converting it to a real bus. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- include/sound/soc.h | 4 +++- sound/soc/au1x/ac97c.c | 7 +++++-- sound/soc/au1x/psc-ac97.c | 7 +++++-- sound/soc/blackfin/bf5xx-ac97.c | 11 +++++++++-- sound/soc/cirrus/ep93xx-ac97.c | 10 ++++++++-- sound/soc/codecs/ac97.c | 7 ++++--- sound/soc/codecs/ad1980.c | 12 ++++++------ sound/soc/codecs/stac9766.c | 22 +++++++++++----------- sound/soc/codecs/wm9705.c | 14 +++++++------- sound/soc/codecs/wm9712.c | 18 +++++++++--------- sound/soc/codecs/wm9713.c | 18 +++++++++--------- sound/soc/fsl/imx-ssi.c | 11 +++++++++-- sound/soc/fsl/mpc5200_psc_ac97.c | 10 ++++++++-- sound/soc/nuc900/nuc900-ac97.c | 11 ++++++++--- sound/soc/pxa/pxa2xx-ac97.c | 8 ++++++-- sound/soc/samsung/ac97.c | 12 +++++++++--- sound/soc/sh/hac.c | 8 ++++++-- sound/soc/soc-core.c | 16 ++++++++++++++++ sound/soc/tegra/tegra20_ac97.c | 12 ++++++++++-- sound/soc/txx9/txx9aclc-ac97.c | 8 ++++++-- 20 files changed, 154 insertions(+), 72 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 85c1522..6eabee7 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -340,7 +340,7 @@ struct snd_soc_jack_gpio; typedef int (*hw_write_t)(void *,const char* ,int); -extern struct snd_ac97_bus_ops soc_ac97_ops; +extern struct snd_ac97_bus_ops *soc_ac97_ops; enum snd_soc_control_type { SND_SOC_I2C = 1, @@ -467,6 +467,8 @@ int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, struct snd_ac97_bus_ops *ops, int num); void snd_soc_free_ac97_codec(struct snd_soc_codec *codec); +int snd_soc_set_ac97_ops(struct snd_ac97_bus_ops *ops); + /* *Controls */ diff --git a/sound/soc/au1x/ac97c.c b/sound/soc/au1x/ac97c.c index a51dabe..d6f7694 100644 --- a/sound/soc/au1x/ac97c.c +++ b/sound/soc/au1x/ac97c.c @@ -179,13 +179,12 @@ static void au1xac97c_ac97_cold_reset(struct snd_ac97 *ac97) } /* AC97 controller operations */ -struct snd_ac97_bus_ops soc_ac97_ops = { +static struct snd_ac97_bus_ops ac97c_bus_ops = { .read = au1xac97c_ac97_read, .write = au1xac97c_ac97_write, .reset = au1xac97c_ac97_cold_reset, .warm_reset = au1xac97c_ac97_warm_reset, }; -EXPORT_SYMBOL_GPL(soc_ac97_ops); /* globals be gone! */ static int alchemy_ac97c_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) @@ -272,6 +271,10 @@ static int au1xac97c_drvprobe(struct platform_device *pdev) platform_set_drvdata(pdev, ctx); + ret = snd_soc_set_ac97_ops(&ac97c_bus_ops); + if (ret) + return ret; + ret = snd_soc_register_component(&pdev->dev, &au1xac97c_component, &au1xac97c_dai_driver, 1); if (ret) diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index a97ba13..a822ab8 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -201,13 +201,12 @@ static void au1xpsc_ac97_cold_reset(struct snd_ac97 *ac97) } /* AC97 controller operations */ -struct snd_ac97_bus_ops soc_ac97_ops = { +static struct snd_ac97_bus_ops psc_ac97_ops = { .read = au1xpsc_ac97_read, .write = au1xpsc_ac97_write, .reset = au1xpsc_ac97_cold_reset, .warm_reset = au1xpsc_ac97_warm_reset, }; -EXPORT_SYMBOL_GPL(soc_ac97_ops); static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, @@ -417,6 +416,10 @@ static int au1xpsc_ac97_drvprobe(struct platform_device *pdev) platform_set_drvdata(pdev, wd); + ret = snd_soc_set_ac97_ops(&psc_ac97_ops); + if (ret) + return ret; + ret = snd_soc_register_component(&pdev->dev, &au1xpsc_ac97_component, &wd->dai_drv, 1); if (ret) diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index 8010f32..efb1dae 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -198,13 +198,12 @@ static void bf5xx_ac97_cold_reset(struct snd_ac97 *ac97) #endif } -struct snd_ac97_bus_ops soc_ac97_ops = { +static struct snd_ac97_bus_ops bf5xx_ac97_ops = { .read = bf5xx_ac97_read, .write = bf5xx_ac97_write, .warm_reset = bf5xx_ac97_warm_reset, .reset = bf5xx_ac97_cold_reset, }; -EXPORT_SYMBOL_GPL(soc_ac97_ops); #ifdef CONFIG_PM static int bf5xx_ac97_suspend(struct snd_soc_dai *dai) @@ -336,6 +335,12 @@ static int asoc_bfin_ac97_probe(struct platform_device *pdev) goto sport_config_err; } + ret = snd_soc_set_ac97_ops(&bf5xx_ac97_ops); + if (ret != 0) { + dev_err(&pdev->dev, "Failed to set AC'97 ops: %d\n", ret); + goto sport_config_err; + } + ret = snd_soc_register_component(&pdev->dev, &bfin_ac97_component, &bfin_ac97_dai, 1); if (ret) { @@ -350,6 +355,7 @@ static int asoc_bfin_ac97_probe(struct platform_device *pdev) sport_config_err: sport_done(sport_handle); sport_err: + snd_soc_set_ac97_ops(NULL); return ret; } @@ -360,6 +366,7 @@ static int asoc_bfin_ac97_remove(struct platform_device *pdev) snd_soc_unregister_component(&pdev->dev); sport_done(sport_handle); + snd_soc_set_ac97_ops(NULL); return 0; } diff --git a/sound/soc/cirrus/ep93xx-ac97.c b/sound/soc/cirrus/ep93xx-ac97.c index f193701..ac73c60 100644 --- a/sound/soc/cirrus/ep93xx-ac97.c +++ b/sound/soc/cirrus/ep93xx-ac97.c @@ -237,13 +237,12 @@ static irqreturn_t ep93xx_ac97_interrupt(int irq, void *dev_id) return IRQ_HANDLED; } -struct snd_ac97_bus_ops soc_ac97_ops = { +static struct snd_ac97_bus_ops ep93xx_ac97_ops = { .read = ep93xx_ac97_read, .write = ep93xx_ac97_write, .reset = ep93xx_ac97_cold_reset, .warm_reset = ep93xx_ac97_warm_reset, }; -EXPORT_SYMBOL_GPL(soc_ac97_ops); static int ep93xx_ac97_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) @@ -389,6 +388,10 @@ static int ep93xx_ac97_probe(struct platform_device *pdev) ep93xx_ac97_info = info; platform_set_drvdata(pdev, info); + ret = snd_soc_set_ac97_ops(&ep93xx_ac97_ops); + if (ret) + goto fail; + ret = snd_soc_register_component(&pdev->dev, &ep93xx_ac97_component, &ep93xx_ac97_dai, 1); if (ret) @@ -398,6 +401,7 @@ static int ep93xx_ac97_probe(struct platform_device *pdev) fail: ep93xx_ac97_info = NULL; + snd_soc_set_ac97_ops(NULL); return ret; } @@ -412,6 +416,8 @@ static int ep93xx_ac97_remove(struct platform_device *pdev) ep93xx_ac97_info = NULL; + snd_soc_set_ac97_ops(NULL); + return 0; } diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index ef2ae32..ec73518 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -62,13 +62,13 @@ static struct snd_soc_dai_driver ac97_dai = { static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg) { - return soc_ac97_ops.read(codec->ac97, reg); + return soc_ac97_ops->read(codec->ac97, reg); } static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int val) { - soc_ac97_ops.write(codec->ac97, reg, val); + soc_ac97_ops->write(codec->ac97, reg, val); return 0; } @@ -79,7 +79,8 @@ static int ac97_soc_probe(struct snd_soc_codec *codec) int ret; /* add codec as bus device for standard ac97 */ - ret = snd_ac97_bus(codec->card->snd_card, 0, &soc_ac97_ops, NULL, &ac97_bus); + ret = snd_ac97_bus(codec->card->snd_card, 0, soc_ac97_ops, NULL, + &ac97_bus); if (ret < 0) return ret; diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index f385342..89fcf7d 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -108,7 +108,7 @@ static unsigned int ac97_read(struct snd_soc_codec *codec, case AC97_EXTENDED_STATUS: case AC97_VENDOR_ID1: case AC97_VENDOR_ID2: - return soc_ac97_ops.read(codec->ac97, reg); + return soc_ac97_ops->read(codec->ac97, reg); default: reg = reg >> 1; @@ -124,7 +124,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, { u16 *cache = codec->reg_cache; - soc_ac97_ops.write(codec->ac97, reg, val); + soc_ac97_ops->write(codec->ac97, reg, val); reg = reg >> 1; if (reg < ARRAY_SIZE(ad1980_reg)) cache[reg] = val; @@ -154,13 +154,13 @@ static int ad1980_reset(struct snd_soc_codec *codec, int try_warm) u16 retry_cnt = 0; retry: - if (try_warm && soc_ac97_ops.warm_reset) { - soc_ac97_ops.warm_reset(codec->ac97); + if (try_warm && soc_ac97_ops->warm_reset) { + soc_ac97_ops->warm_reset(codec->ac97); if (ac97_read(codec, AC97_RESET) == 0x0090) return 1; } - soc_ac97_ops.reset(codec->ac97); + soc_ac97_ops->reset(codec->ac97); /* Set bit 16slot in register 74h, then every slot will has only 16 * bits. This command is sent out in 20bit mode, in which case the * first nibble of data is eaten by the addr. (Tag is always 16 bit)*/ @@ -186,7 +186,7 @@ static int ad1980_soc_probe(struct snd_soc_codec *codec) printk(KERN_INFO "AD1980 SoC Audio Codec\n"); - ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); + ret = snd_soc_new_ac97_codec(codec, soc_ac97_ops, 0); if (ret < 0) { printk(KERN_ERR "ad1980: failed to register AC97 codec\n"); return ret; diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index cbc7ae3..a5455c1 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -143,14 +143,14 @@ static int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg, if (reg > AC97_STAC_PAGE0) { stac9766_ac97_write(codec, AC97_INT_PAGING, 0); - soc_ac97_ops.write(codec->ac97, reg, val); + soc_ac97_ops->write(codec->ac97, reg, val); stac9766_ac97_write(codec, AC97_INT_PAGING, 1); return 0; } if (reg / 2 >= ARRAY_SIZE(stac9766_reg)) return -EIO; - soc_ac97_ops.write(codec->ac97, reg, val); + soc_ac97_ops->write(codec->ac97, reg, val); cache[reg / 2] = val; return 0; } @@ -162,7 +162,7 @@ static unsigned int stac9766_ac97_read(struct snd_soc_codec *codec, if (reg > AC97_STAC_PAGE0) { stac9766_ac97_write(codec, AC97_INT_PAGING, 0); - val = soc_ac97_ops.read(codec->ac97, reg - AC97_STAC_PAGE0); + val = soc_ac97_ops->read(codec->ac97, reg - AC97_STAC_PAGE0); stac9766_ac97_write(codec, AC97_INT_PAGING, 1); return val; } @@ -173,7 +173,7 @@ static unsigned int stac9766_ac97_read(struct snd_soc_codec *codec, reg == AC97_INT_PAGING || reg == AC97_VENDOR_ID1 || reg == AC97_VENDOR_ID2) { - val = soc_ac97_ops.read(codec->ac97, reg); + val = soc_ac97_ops->read(codec->ac97, reg); return val; } return cache[reg / 2]; @@ -240,15 +240,15 @@ static int stac9766_set_bias_level(struct snd_soc_codec *codec, static int stac9766_reset(struct snd_soc_codec *codec, int try_warm) { - if (try_warm && soc_ac97_ops.warm_reset) { - soc_ac97_ops.warm_reset(codec->ac97); + if (try_warm && soc_ac97_ops->warm_reset) { + soc_ac97_ops->warm_reset(codec->ac97); if (stac9766_ac97_read(codec, 0) == stac9766_reg[0]) return 1; } - soc_ac97_ops.reset(codec->ac97); - if (soc_ac97_ops.warm_reset) - soc_ac97_ops.warm_reset(codec->ac97); + soc_ac97_ops->reset(codec->ac97); + if (soc_ac97_ops->warm_reset) + soc_ac97_ops->warm_reset(codec->ac97); if (stac9766_ac97_read(codec, 0) != stac9766_reg[0]) return -EIO; return 0; @@ -272,7 +272,7 @@ reset: return -EIO; } codec->ac97->bus->ops->warm_reset(codec->ac97); - id = soc_ac97_ops.read(codec->ac97, AC97_VENDOR_ID2); + id = soc_ac97_ops->read(codec->ac97, AC97_VENDOR_ID2); if (id != 0x4c13) { stac9766_reset(codec, 0); reset++; @@ -336,7 +336,7 @@ static int stac9766_codec_probe(struct snd_soc_codec *codec) { int ret = 0; - ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); + ret = snd_soc_new_ac97_codec(codec, soc_ac97_ops, 0); if (ret < 0) goto codec_err; diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index a5fc61d..70ce6793 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -209,7 +209,7 @@ static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg) case AC97_RESET: case AC97_VENDOR_ID1: case AC97_VENDOR_ID2: - return soc_ac97_ops.read(codec->ac97, reg); + return soc_ac97_ops->read(codec->ac97, reg); default: reg = reg >> 1; @@ -225,7 +225,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, { u16 *cache = codec->reg_cache; - soc_ac97_ops.write(codec->ac97, reg, val); + soc_ac97_ops->write(codec->ac97, reg, val); reg = reg >> 1; if (reg < (ARRAY_SIZE(wm9705_reg))) cache[reg] = val; @@ -294,8 +294,8 @@ static struct snd_soc_dai_driver wm9705_dai[] = { static int wm9705_reset(struct snd_soc_codec *codec) { - if (soc_ac97_ops.reset) { - soc_ac97_ops.reset(codec->ac97); + if (soc_ac97_ops->reset) { + soc_ac97_ops->reset(codec->ac97); if (ac97_read(codec, 0) == wm9705_reg[0]) return 0; /* Success */ } @@ -306,7 +306,7 @@ static int wm9705_reset(struct snd_soc_codec *codec) #ifdef CONFIG_PM static int wm9705_soc_suspend(struct snd_soc_codec *codec) { - soc_ac97_ops.write(codec->ac97, AC97_POWERDOWN, 0xffff); + soc_ac97_ops->write(codec->ac97, AC97_POWERDOWN, 0xffff); return 0; } @@ -323,7 +323,7 @@ static int wm9705_soc_resume(struct snd_soc_codec *codec) } for (i = 2; i < ARRAY_SIZE(wm9705_reg) << 1; i += 2) { - soc_ac97_ops.write(codec->ac97, i, cache[i>>1]); + soc_ac97_ops->write(codec->ac97, i, cache[i>>1]); } return 0; @@ -337,7 +337,7 @@ static int wm9705_soc_probe(struct snd_soc_codec *codec) { int ret = 0; - ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); + ret = snd_soc_new_ac97_codec(codec, soc_ac97_ops, 0); if (ret < 0) { printk(KERN_ERR "wm9705: failed to register AC97 codec\n"); return ret; diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 8e9a6a3..c5eb746 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -455,7 +455,7 @@ static unsigned int ac97_read(struct snd_soc_codec *codec, if (reg == AC97_RESET || reg == AC97_GPIO_STATUS || reg == AC97_VENDOR_ID1 || reg == AC97_VENDOR_ID2 || reg == AC97_REC_GAIN) - return soc_ac97_ops.read(codec->ac97, reg); + return soc_ac97_ops->read(codec->ac97, reg); else { reg = reg >> 1; @@ -472,7 +472,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, u16 *cache = codec->reg_cache; if (reg < 0x7c) - soc_ac97_ops.write(codec->ac97, reg, val); + soc_ac97_ops->write(codec->ac97, reg, val); reg = reg >> 1; if (reg < (ARRAY_SIZE(wm9712_reg))) cache[reg] = val; @@ -581,15 +581,15 @@ static int wm9712_set_bias_level(struct snd_soc_codec *codec, static int wm9712_reset(struct snd_soc_codec *codec, int try_warm) { - if (try_warm && soc_ac97_ops.warm_reset) { - soc_ac97_ops.warm_reset(codec->ac97); + if (try_warm && soc_ac97_ops->warm_reset) { + soc_ac97_ops->warm_reset(codec->ac97); if (ac97_read(codec, 0) == wm9712_reg[0]) return 1; } - soc_ac97_ops.reset(codec->ac97); - if (soc_ac97_ops.warm_reset) - soc_ac97_ops.warm_reset(codec->ac97); + soc_ac97_ops->reset(codec->ac97); + if (soc_ac97_ops->warm_reset) + soc_ac97_ops->warm_reset(codec->ac97); if (ac97_read(codec, 0) != wm9712_reg[0]) goto err; return 0; @@ -624,7 +624,7 @@ static int wm9712_soc_resume(struct snd_soc_codec *codec) if (i == AC97_INT_PAGING || i == AC97_POWERDOWN || (i > 0x58 && i != 0x5c)) continue; - soc_ac97_ops.write(codec->ac97, i, cache[i>>1]); + soc_ac97_ops->write(codec->ac97, i, cache[i>>1]); } } @@ -635,7 +635,7 @@ static int wm9712_soc_probe(struct snd_soc_codec *codec) { int ret = 0; - ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); + ret = snd_soc_new_ac97_codec(codec, soc_ac97_ops, 0); if (ret < 0) { printk(KERN_ERR "wm9712: failed to register AC97 codec\n"); return ret; diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index f7afa68..a53e175 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -652,7 +652,7 @@ static unsigned int ac97_read(struct snd_soc_codec *codec, if (reg == AC97_RESET || reg == AC97_GPIO_STATUS || reg == AC97_VENDOR_ID1 || reg == AC97_VENDOR_ID2 || reg == AC97_CD) - return soc_ac97_ops.read(codec->ac97, reg); + return soc_ac97_ops->read(codec->ac97, reg); else { reg = reg >> 1; @@ -668,7 +668,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, { u16 *cache = codec->reg_cache; if (reg < 0x7c) - soc_ac97_ops.write(codec->ac97, reg, val); + soc_ac97_ops->write(codec->ac97, reg, val); reg = reg >> 1; if (reg < (ARRAY_SIZE(wm9713_reg))) cache[reg] = val; @@ -1095,15 +1095,15 @@ static struct snd_soc_dai_driver wm9713_dai[] = { int wm9713_reset(struct snd_soc_codec *codec, int try_warm) { - if (try_warm && soc_ac97_ops.warm_reset) { - soc_ac97_ops.warm_reset(codec->ac97); + if (try_warm && soc_ac97_ops->warm_reset) { + soc_ac97_ops->warm_reset(codec->ac97); if (ac97_read(codec, 0) == wm9713_reg[0]) return 1; } - soc_ac97_ops.reset(codec->ac97); - if (soc_ac97_ops.warm_reset) - soc_ac97_ops.warm_reset(codec->ac97); + soc_ac97_ops->reset(codec->ac97); + if (soc_ac97_ops->warm_reset) + soc_ac97_ops->warm_reset(codec->ac97); if (ac97_read(codec, 0) != wm9713_reg[0]) return -EIO; return 0; @@ -1180,7 +1180,7 @@ static int wm9713_soc_resume(struct snd_soc_codec *codec) if (i == AC97_POWERDOWN || i == AC97_EXTENDED_MID || i == AC97_EXTENDED_MSTATUS || i > 0x66) continue; - soc_ac97_ops.write(codec->ac97, i, cache[i>>1]); + soc_ac97_ops->write(codec->ac97, i, cache[i>>1]); } } @@ -1197,7 +1197,7 @@ static int wm9713_soc_probe(struct snd_soc_codec *codec) return -ENOMEM; snd_soc_codec_set_drvdata(codec, wm9713); - ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); + ret = snd_soc_new_ac97_codec(codec, soc_ac97_ops, 0); if (ret < 0) goto codec_err; diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index a8362be..51be377 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -501,13 +501,12 @@ static void imx_ssi_ac97_warm_reset(struct snd_ac97 *ac97) imx_ssi_ac97_read(ac97, 0); } -struct snd_ac97_bus_ops soc_ac97_ops = { +static struct snd_ac97_bus_ops imx_ssi_ac97_ops = { .read = imx_ssi_ac97_read, .write = imx_ssi_ac97_write, .reset = imx_ssi_ac97_reset, .warm_reset = imx_ssi_ac97_warm_reset }; -EXPORT_SYMBOL_GPL(soc_ac97_ops); static int imx_ssi_probe(struct platform_device *pdev) { @@ -583,6 +582,12 @@ static int imx_ssi_probe(struct platform_device *pdev) platform_set_drvdata(pdev, ssi); + ret = snd_soc_set_ac97_ops(&imx_ssi_ac97_ops); + if (ret != 0) { + dev_err(&pdev->dev, "Failed to set AC'97 ops: %d\n", ret); + goto failed_register; + } + ret = snd_soc_register_component(&pdev->dev, &imx_component, dai, 1); if (ret) { @@ -608,6 +613,7 @@ failed_register: release_mem_region(res->start, resource_size(res)); clk_disable_unprepare(ssi->clk); failed_clk: + snd_soc_set_ac97_ops(NULL); return ret; } @@ -627,6 +633,7 @@ static int imx_ssi_remove(struct platform_device *pdev) release_mem_region(res->start, resource_size(res)); clk_disable_unprepare(ssi->clk); + snd_soc_set_ac97_ops(NULL); return 0; } diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c index 4141b35..3ef7a0c 100644 --- a/sound/soc/fsl/mpc5200_psc_ac97.c +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -131,13 +131,12 @@ static void psc_ac97_cold_reset(struct snd_ac97 *ac97) psc_ac97_warm_reset(ac97); } -struct snd_ac97_bus_ops soc_ac97_ops = { +static struct snd_ac97_bus_ops psc_ac97_ops = { .read = psc_ac97_read, .write = psc_ac97_write, .reset = psc_ac97_cold_reset, .warm_reset = psc_ac97_warm_reset, }; -EXPORT_SYMBOL_GPL(soc_ac97_ops); static int psc_ac97_hw_analog_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, @@ -290,6 +289,12 @@ static int psc_ac97_of_probe(struct platform_device *op) if (rc != 0) return rc; + rc = snd_soc_set_ac97_ops(&psc_ac97_ops); + if (rc != 0) { + dev_err(&op->dev, "Failed to set AC'97 ops: %d\n", ret); + return rc; + } + rc = snd_soc_register_component(&op->dev, &psc_ac97_component, psc_ac97_dai, ARRAY_SIZE(psc_ac97_dai)); if (rc != 0) { @@ -318,6 +323,7 @@ static int psc_ac97_of_remove(struct platform_device *op) { mpc5200_audio_dma_destroy(op); snd_soc_unregister_component(&op->dev); + snd_soc_set_ac97_ops(NULL); return 0; } diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c index 8dea4c1..f4c2417 100644 --- a/sound/soc/nuc900/nuc900-ac97.c +++ b/sound/soc/nuc900/nuc900-ac97.c @@ -197,13 +197,12 @@ static void nuc900_ac97_cold_reset(struct snd_ac97 *ac97) } /* AC97 controller operations */ -struct snd_ac97_bus_ops soc_ac97_ops = { +static struct snd_ac97_bus_ops nuc900_ac97_ops = { .read = nuc900_ac97_read, .write = nuc900_ac97_write, .reset = nuc900_ac97_cold_reset, .warm_reset = nuc900_ac97_warm_reset, -} -EXPORT_SYMBOL_GPL(soc_ac97_ops); +}; static int nuc900_ac97_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) @@ -356,6 +355,10 @@ static int nuc900_ac97_drvprobe(struct platform_device *pdev) nuc900_ac97_data = nuc900_audio; + ret = snd_soc_set_ac97_ops(&nuc900_ac97_ops); + if (ret) + goto out; + ret = snd_soc_register_component(&pdev->dev, &nuc900_ac97_component, &nuc900_ac97_dai, 1); if (ret) @@ -367,6 +370,7 @@ static int nuc900_ac97_drvprobe(struct platform_device *pdev) return 0; out: + snd_soc_set_ac97_ops(NULL); return ret; } @@ -375,6 +379,7 @@ static int nuc900_ac97_drvremove(struct platform_device *pdev) snd_soc_unregister_component(&pdev->dev); nuc900_ac97_data = NULL; + snd_soc_set_ac97_ops(NULL); return 0; } diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 57ea8e6..a3c22ba 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -41,13 +41,12 @@ static void pxa2xx_ac97_cold_reset(struct snd_ac97 *ac97) pxa2xx_ac97_finish_reset(ac97); } -struct snd_ac97_bus_ops soc_ac97_ops = { +static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { .read = pxa2xx_ac97_read, .write = pxa2xx_ac97_write, .warm_reset = pxa2xx_ac97_warm_reset, .reset = pxa2xx_ac97_cold_reset, }; -EXPORT_SYMBOL_GPL(soc_ac97_ops); static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_stereo_out = { .name = "AC97 PCM Stereo out", @@ -244,6 +243,10 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) return -ENXIO; } + ret = snd_soc_set_ac97_ops(&pxa2xx_ac97_ops); + if (ret != 0) + return ret; + /* 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. @@ -255,6 +258,7 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) static int pxa2xx_ac97_dev_remove(struct platform_device *pdev) { snd_soc_unregister_component(&pdev->dev); + snd_soc_set_ac97_ops(NULL); return 0; } diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index 04d7fd4..2dd623f 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -214,13 +214,12 @@ static irqreturn_t s3c_ac97_irq(int irq, void *dev_id) return IRQ_HANDLED; } -struct snd_ac97_bus_ops soc_ac97_ops = { +static struct snd_ac97_bus_ops s3c_ac97_ops = { .read = s3c_ac97_read, .write = s3c_ac97_write, .warm_reset = s3c_ac97_warm_reset, .reset = s3c_ac97_cold_reset, }; -EXPORT_SYMBOL_GPL(soc_ac97_ops); static int s3c_ac97_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, @@ -452,6 +451,12 @@ static int s3c_ac97_probe(struct platform_device *pdev) goto err4; } + ret = snd_soc_set_ac97_ops(&s3c_ac97_ops); + if (ret != 0) { + dev_err(&pdev->dev, "Failed to set AC'97 ops: %d\n", ret); + goto err4; + } + ret = snd_soc_register_component(&pdev->dev, &s3c_ac97_component, s3c_ac97_dai, ARRAY_SIZE(s3c_ac97_dai)); if (ret) @@ -472,7 +477,7 @@ err4: err3: clk_disable_unprepare(s3c_ac97.ac97_clk); err2: - + snd_soc_set_ac97_ops(NULL); return ret; } @@ -488,6 +493,7 @@ static int s3c_ac97_remove(struct platform_device *pdev) free_irq(irq_res->start, NULL); clk_disable_unprepare(s3c_ac97.ac97_clk); + snd_soc_set_ac97_ops(NULL); return 0; } diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c index af19f77..0af2e4d 100644 --- a/sound/soc/sh/hac.c +++ b/sound/soc/sh/hac.c @@ -227,13 +227,12 @@ static void hac_ac97_coldrst(struct snd_ac97 *ac97) hac_ac97_warmrst(ac97); } -struct snd_ac97_bus_ops soc_ac97_ops = { +static struct snd_ac97_bus_ops hac_ac97_ops = { .read = hac_ac97_read, .write = hac_ac97_write, .reset = hac_ac97_coldrst, .warm_reset = hac_ac97_warmrst, }; -EXPORT_SYMBOL_GPL(soc_ac97_ops); static int hac_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, @@ -316,6 +315,10 @@ static const struct snd_soc_component_driver sh4_hac_component = { static int hac_soc_platform_probe(struct platform_device *pdev) { + ret = snd_soc_set_ac97_ops(&hac_ac97_ops); + if (ret != 0) + return ret; + return snd_soc_register_component(&pdev->dev, &sh4_hac_component, sh4_hac_dai, ARRAY_SIZE(sh4_hac_dai)); } @@ -323,6 +326,7 @@ static int hac_soc_platform_probe(struct platform_device *pdev) static int hac_soc_platform_remove(struct platform_device *pdev) { snd_soc_unregister_component(&pdev->dev); + snd_soc_set_ac97_ops(NULL); return 0; } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4489c5b..309e5c9 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2080,6 +2080,22 @@ int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, } EXPORT_SYMBOL_GPL(snd_soc_new_ac97_codec); +struct snd_ac97_bus_ops *soc_ac97_ops; + +int snd_soc_set_ac97_ops(struct snd_ac97_bus_ops *ops) +{ + if (ops == soc_ac97_ops) + return 0; + + if (soc_ac97_ops && ops) + return -EBUSY; + + soc_ac97_ops = ops; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_set_ac97_ops); + /** * snd_soc_free_ac97_codec - free AC97 codec device * @codec: audio codec diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index 48037f7..f52eab6 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -142,13 +142,12 @@ static void tegra20_ac97_codec_write(struct snd_ac97 *ac97_snd, } while (!time_after(jiffies, timeout)); } -struct snd_ac97_bus_ops soc_ac97_ops = { +static struct snd_ac97_bus_ops tegra20_ac97_ops = { .read = tegra20_ac97_codec_read, .write = tegra20_ac97_codec_write, .reset = tegra20_ac97_codec_reset, .warm_reset = tegra20_ac97_codec_warm_reset, }; -EXPORT_SYMBOL_GPL(soc_ac97_ops); static inline void tegra20_ac97_start_playback(struct tegra20_ac97 *ac97) { @@ -409,6 +408,12 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) goto err_asoc_utils_fini; } + ret = snd_soc_set_ac97_ops(&tegra20_ac97_ops); + if (ret) { + dev_err(&pdev->dev, "Failed to set AC'97 ops: %d\n", ret); + goto err_asoc_utils_fini; + } + ret = snd_soc_register_component(&pdev->dev, &tegra20_ac97_component, &tegra20_ac97_dai, 1); if (ret) { @@ -436,6 +441,7 @@ err_asoc_utils_fini: tegra_asoc_utils_fini(&ac97->util_data); err_clk_put: err: + snd_soc_set_ac97_ops(NULL); return ret; } @@ -450,6 +456,8 @@ static int tegra20_ac97_platform_remove(struct platform_device *pdev) clk_disable_unprepare(ac97->clk_ac97); + snd_soc_set_ac97_ops(NULL); + return 0; } diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c index 8ee8d42..4bcce8a 100644 --- a/sound/soc/txx9/txx9aclc-ac97.c +++ b/sound/soc/txx9/txx9aclc-ac97.c @@ -119,12 +119,11 @@ static void txx9aclc_ac97_cold_reset(struct snd_ac97 *ac97) } /* AC97 controller operations */ -struct snd_ac97_bus_ops soc_ac97_ops = { +static struct snd_ac97_bus_ops txx9aclc_ac97_ops = { .read = txx9aclc_ac97_read, .write = txx9aclc_ac97_write, .reset = txx9aclc_ac97_cold_reset, }; -EXPORT_SYMBOL_GPL(soc_ac97_ops); static irqreturn_t txx9aclc_ac97_irq(int irq, void *dev_id) { @@ -206,6 +205,10 @@ static int txx9aclc_ac97_dev_probe(struct platform_device *pdev) if (err < 0) return err; + err = snd_soc_set_ac97_ops(&txx9aclc_ac97_ops); + if (err < 0) + return err; + return snd_soc_register_component(&pdev->dev, &txx9aclc_ac97_component, &txx9aclc_ac97_dai, 1); } @@ -213,6 +216,7 @@ static int txx9aclc_ac97_dev_probe(struct platform_device *pdev) static int txx9aclc_ac97_dev_remove(struct platform_device *pdev) { snd_soc_unregister_component(&pdev->dev); + snd_soc_set_ac97_ops(NULL); return 0; } -- 1.8.3.1

3 3

[alsa-devel] [PATCH 1/2] ASoC: tegra20-ac97: Convert to devm_clk_get()
by Mark Brown 26 Jun '13

26 Jun '13

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/tegra/tegra20_ac97.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index 2f70ea7..d8c142d 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -327,7 +327,7 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) } dev_set_drvdata(&pdev->dev, ac97); - ac97->clk_ac97 = clk_get(&pdev->dev, NULL); + ac97->clk_ac97 = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(ac97->clk_ac97)) { dev_err(&pdev->dev, "Can't retrieve ac97 clock\n"); ret = PTR_ERR(ac97->clk_ac97); @@ -443,7 +443,6 @@ err_unregister_pcm: err_unregister_component: snd_soc_unregister_component(&pdev->dev); err_clk_put: - clk_put(ac97->clk_ac97); err: return ret; } @@ -458,7 +457,6 @@ static int tegra20_ac97_platform_remove(struct platform_device *pdev) tegra_asoc_utils_fini(&ac97->util_data); clk_disable_unprepare(ac97->clk_ac97); - clk_put(ac97->clk_ac97); return 0; } -- 1.8.3.1

2 2

[alsa-devel] [PATCH] ASoC: tegra-ac97: Do common and clock init prior to component registration
by Mark Brown 26 Jun '13

26 Jun '13

From: Mark Brown <broonie(a)linaro.org> Otherwise we may instantiate and hence have something try to access the device while it is still completing initialisation. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/tegra/tegra20_ac97.c | 34 +++++++++++++++++----------------- 1 file changed, 17 insertions(+), 17 deletions(-) diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index 9043626d..48037f7 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -395,23 +395,9 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) ac97->capture_dma_data.maxburst = 4; ac97->capture_dma_data.slave_id = of_dma[0]; - ret = snd_soc_register_component(&pdev->dev, &tegra20_ac97_component, - &tegra20_ac97_dai, 1); - if (ret) { - dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); - ret = -ENOMEM; - goto err_clk_put; - } - - ret = tegra_pcm_platform_register(&pdev->dev); - if (ret) { - dev_err(&pdev->dev, "Could not register PCM: %d\n", ret); - goto err_unregister_component; - } - ret = tegra_asoc_utils_init(&ac97->util_data, &pdev->dev); if (ret) - goto err_unregister_pcm; + goto err_clk_put; ret = tegra_asoc_utils_set_ac97_rate(&ac97->util_data); if (ret) @@ -423,17 +409,31 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) goto err_asoc_utils_fini; } + ret = snd_soc_register_component(&pdev->dev, &tegra20_ac97_component, + &tegra20_ac97_dai, 1); + if (ret) { + dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); + ret = -ENOMEM; + goto err_asoc_utils_fini; + } + + ret = tegra_pcm_platform_register(&pdev->dev); + if (ret) { + dev_err(&pdev->dev, "Could not register PCM: %d\n", ret); + goto err_unregister_component; + } + /* XXX: crufty ASoC AC97 API - only one AC97 codec allowed */ workdata = ac97; return 0; -err_asoc_utils_fini: - tegra_asoc_utils_fini(&ac97->util_data); err_unregister_pcm: tegra_pcm_platform_unregister(&pdev->dev); err_unregister_component: snd_soc_unregister_component(&pdev->dev); +err_asoc_utils_fini: + tegra_asoc_utils_fini(&ac97->util_data); err_clk_put: err: return ret; -- 1.8.3.1

2 1

Re: [alsa-devel] [PATCH -next] ASoC: mid-x86: Convert to use devm_* APIs
by Mark Brown 26 Jun '13

26 Jun '13

On Tue, Jun 25, 2013 at 10:16:56AM +0800, Wei Yongjun wrote: > From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> > > devm_* APIs are device managed and make code simpler. Applied, thanks.

1 0

Re: [alsa-devel] [PATCH] ASoC: Add PCM1681 codec driver.
by Daniel Mack 26 Jun '13

26 Jun '13

On 26.06.2013 15:46, Marek Belisko wrote: > Hi Daniel, > On 06/26/2013 03:16 PM, Daniel Mack wrote: >> Hi Marek, >> >> On 26.06.2013 15:05, Marek Belisko wrote: >>> PCM1681 can be controlled via I2C, SPI or in bootstrap mode. This code add >>> support only for I2C mode. >>> >>> Signed-off-by: Marek Belisko <marek.belisko(a)streamunlimited.com> >>> --- >>> .../devicetree/bindings/sound/ti,pcm1681.txt | 15 + >>> sound/soc/codecs/Kconfig | 4 + >>> sound/soc/codecs/Makefile | 2 + >>> sound/soc/codecs/pcm1681.c | 328 ++++++++++++++++++++ >>> 4 files changed, 349 insertions(+) >>> create mode 100644 Documentation/devicetree/bindings/sound/ti,pcm1681.txt >>> create mode 100644 sound/soc/codecs/pcm1681.c >>> >>> diff --git a/Documentation/devicetree/bindings/sound/ti,pcm1681.txt b/Documentation/devicetree/bindings/sound/ti,pcm1681.txt >>> new file mode 100644 >>> index 0000000..4df1718 >>> --- /dev/null >>> +++ b/Documentation/devicetree/bindings/sound/ti,pcm1681.txt >>> @@ -0,0 +1,15 @@ >>> +Texas Instruments PCM1681 8-channel PWM Processor >>> + >>> +Required properties: >>> + >>> + - compatible: Should contain "ti,pcm1681". >>> + - reg: The i2c address. Should contain <0x4c>. >>> + >>> +Examples: >>> + >>> + i2c_bus { >>> + pcm1681@4c { >>> + compatible = "ti,pcm1681"; >>> + reg = <0x4c>; >>> + }; >>> + }; >>> diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig >>> index badb6fb..e2daf82 100644 >>> --- a/sound/soc/codecs/Kconfig >>> +++ b/sound/soc/codecs/Kconfig >>> @@ -54,6 +54,7 @@ config SND_SOC_ALL_CODECS >>> select SND_SOC_MC13783 if MFD_MC13XXX >>> select SND_SOC_ML26124 if I2C >>> select SND_SOC_HDMI_CODEC >>> + select SND_SOC_PCM1681 if I2C >>> select SND_SOC_PCM3008 >>> select SND_SOC_RT5631 if I2C >>> select SND_SOC_RT5640 if I2C >>> @@ -292,6 +293,9 @@ config SND_SOC_MAX9850 >>> config SND_SOC_HDMI_CODEC >>> tristate >>> >>> +config SND_SOC_PCM1681 >>> + tristate >>> + >>> config SND_SOC_PCM3008 >>> tristate >>> >>> diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile >>> index 70fd806..4a068d2 100644 >>> --- a/sound/soc/codecs/Makefile >>> +++ b/sound/soc/codecs/Makefile >>> @@ -42,6 +42,7 @@ snd-soc-max9850-objs := max9850.o >>> snd-soc-mc13783-objs := mc13783.o >>> snd-soc-ml26124-objs := ml26124.o >>> snd-soc-hdmi-codec-objs := hdmi.o >>> +snd-soc-pcm1681-objs := pcm1681.o >>> snd-soc-pcm3008-objs := pcm3008.o >>> snd-soc-rt5631-objs := rt5631.o >>> snd-soc-rt5640-objs := rt5640.o >>> @@ -171,6 +172,7 @@ obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o >>> obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o >>> obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o >>> obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o >>> +obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o >>> obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o >>> obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o >>> obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o >>> diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c >>> new file mode 100644 >>> index 0000000..6122fd1 >>> --- /dev/null >>> +++ b/sound/soc/codecs/pcm1681.c >>> @@ -0,0 +1,328 @@ >>> +/* >>> + * PCM1681 ASoC codec driver >>> + * >>> + * Copyright (c) StreamUnlimited GmbH 2013 >>> + * Marek Belisko <marek.belisko(a)streamunlimited.com> >>> + * >>> + * This program is free software; you can redistribute it and/or >>> + * modify it under the terms of the GNU General Public License >>> + * as published by the Free Software Foundation; either version 2 >>> + * of the License, or (at your option) any later version. >>> + * >>> + * This program is distributed in the hope that it will be useful, >>> + * but WITHOUT ANY WARRANTY; without even the implied warranty of >>> + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the >>> + * GNU General Public License for more details. >>> + */ >>> + >>> +#include <linux/module.h> >>> +#include <linux/slab.h> >>> +#include <linux/delay.h> >>> +#include <linux/gpio.h> >>> +#include <linux/i2c.h> >>> +#include <linux/regmap.h> >>> +#include <linux/of_device.h> >>> +#include <linux/of_gpio.h> >>> +#include <sound/pcm.h> >>> +#include <sound/pcm_params.h> >>> +#include <sound/soc.h> >>> +#include <sound/tlv.h> >>> + >>> +#define PCM1681_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ >>> + SNDRV_PCM_FMTBIT_S24_LE) >>> + >>> +#define PCM1681_PCM_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | \ >>> + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ >>> + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | \ >>> + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000) >>> + >>> +#define PCM1681_SOFT_MUTE_ALL 0xff >>> +#define PCM1681_DEEMPH_RATE_MASK 0x18 >>> +#define PCM1681_DEEMPH_MASK 0x01 >>> + >>> +#define PCM1681_ATT_CONTROL(X) (X <= 6 ? X : X + 9) /* Attenuation level */ >>> +#define PCM1681_SOFT_MUTE 0x07 /* Soft mute control register */ >>> +#define PCM1681_DAC_CONTROL 0x08 /* DAC operation control */ >>> +#define PCM1681_FMT_CONTROL 0x09 /* Audio interface data format */ >>> +#define PCM1681_DEEMPH_CONTROL 0x0a /* De-emphasis control */ >>> +#define PCM1681_ZERO_DETECT_STATUS 0x0e /* Zero detect status reg */ >>> + >>> +static const struct reg_default pcm1681_reg_defaults[] = { >>> + { 0x01, 0xff }, >>> + { 0x02, 0xff }, >>> + { 0x03, 0xff }, >>> + { 0x04, 0xff }, >>> + { 0x05, 0xff }, >>> + { 0x06, 0xff }, >>> + { 0x07, 0x00 }, >>> + { 0x08, 0x00 }, >>> + { 0x09, 0x06 }, >>> + { 0x0A, 0x00 }, >>> + { 0x0B, 0xff }, >>> + { 0x0C, 0x0f }, >>> + { 0x0D, 0x00 }, >>> + { 0x10, 0xff }, >>> + { 0x11, 0xff }, >>> + { 0x12, 0x00 }, >>> + { 0x13, 0x00 }, >>> +}; >>> + >>> +static bool pcm1681_accessible_reg(struct device *dev, unsigned int reg) >>> +{ >>> + return !((reg == 0x00) || (reg == 0x0f)); >>> +} >>> + >>> +static bool pcm1681_writeable_reg(struct device *dev, unsigned register reg) >>> +{ >>> + return pcm1681_accessible_reg(dev, reg) && >>> + (reg != PCM1681_ZERO_DETECT_STATUS); >>> +} >>> + >>> +struct pcm1681_private { >>> + struct regmap *regmap; >>> + unsigned int format; >>> + /* Current deemphasis status */ >>> + unsigned int deemph; >>> + /* Current rate for deemphasis control */ >>> + unsigned int rate; >>> +}; >>> + >>> +static int pcm1681_deemph[] = { 44100, 48000, 32000 }; >>> + >>> +static int pcm1681_set_deemph(struct snd_soc_codec *codec) >>> +{ >>> + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); >>> + int i = 0, val = -1, ret; >>> + >>> + if (priv->deemph) >>> + for (i = 0; i < ARRAY_SIZE(pcm1681_deemph); i++) >>> + if (pcm1681_deemph[i] == priv->rate) >>> + val = i; >>> + >>> + /* enable choosen frequency */ >>> + if (val != -1) >>> + regmap_update_bits(priv->regmap, PCM1681_DEEMPH_CONTROL, >>> + PCM1681_DEEMPH_RATE_MASK, val); >>> + >>> + /* enable/disable deemphasis functionality */ >>> + ret = regmap_update_bits(priv->regmap, PCM1681_DEEMPH_CONTROL, >>> + PCM1681_DEEMPH_MASK, priv->deemph); >>> + >>> + return ret; >>> +} >>> + >>> +static int pcm1681_get_deemph(struct snd_kcontrol *kcontrol, >>> + struct snd_ctl_elem_value *ucontrol) >>> +{ >>> + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); >>> + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); >>> + >>> + ucontrol->value.enumerated.item[0] = priv->deemph; >>> + >>> + return 0; >>> +} >>> + >>> +static int pcm1681_put_deemph(struct snd_kcontrol *kcontrol, >>> + struct snd_ctl_elem_value *ucontrol) >>> +{ >>> + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); >>> + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); >>> + >>> + priv->deemph = ucontrol->value.enumerated.item[0]; >>> + >>> + return pcm1681_set_deemph(codec); >>> +} >>> + >>> +static int pcm1681_set_dai_fmt(struct snd_soc_dai *codec_dai, >>> + unsigned int format) >>> +{ >>> + struct snd_soc_codec *codec = codec_dai->codec; >>> + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); >>> + >>> + /* The PCM1681 can only be slave to all clocks */ >>> + if ((format & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) { >>> + dev_err(codec->dev, "Invalid clocking mode\n"); >>> + return -EINVAL; >>> + } >>> + >>> + priv->format = format; >>> + >>> + return 0; >>> +} >>> + >>> +static int pcm1681_digital_mute(struct snd_soc_dai *dai, int mute) >>> +{ >>> + struct snd_soc_codec *codec = dai->codec; >>> + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); >>> + int ret, val = 0; >>> + >>> + if (mute) >>> + val = PCM1681_SOFT_MUTE_ALL; >>> + >>> + ret = regmap_write(priv->regmap, PCM1681_SOFT_MUTE, val); >>> + if (ret < 0) >>> + return ret; >>> + >>> + return 0; >>> +} >>> + >>> +static int pcm1681_hw_params(struct snd_pcm_substream *substream, >>> + struct snd_pcm_hw_params *params, >>> + struct snd_soc_dai *dai) >>> +{ >>> + struct snd_soc_codec *codec = dai->codec; >>> + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); >>> + int val = 0; >>> + int pcm_format = params_format(params); >>> + >>> + priv->rate = params_rate(params); >>> + >>> + switch (priv->format & SND_SOC_DAIFMT_FORMAT_MASK) { >>> + case SND_SOC_DAIFMT_RIGHT_J: >>> + if (pcm_format == SNDRV_PCM_FORMAT_S24_LE) >>> + val = 0x00; >>> + else if (pcm_format == SNDRV_PCM_FORMAT_S16_LE) >>> + val = 0x03; >>> + break; >>> + case SND_SOC_DAIFMT_I2S: >>> + val = 0x04; >>> + break; >>> + case SND_SOC_DAIFMT_LEFT_J: >>> + val = 0x05; >>> + break; >>> + default: >>> + dev_err(codec->dev, "Invalid DAI format\n"); >>> + return -EINVAL; >>> + } >>> + >>> + regmap_update_bits(priv->regmap, PCM1681_FMT_CONTROL, 0x0f, val); >>> + >>> + return 0; >>> +} >>> + >>> +static const struct snd_soc_dai_ops pcm1681_dai_ops = { >>> + .set_fmt = pcm1681_set_dai_fmt, >>> + .hw_params = pcm1681_hw_params, >>> + .digital_mute = pcm1681_digital_mute, >>> +}; >>> + >>> +static const DECLARE_TLV_DB_SCALE(pcm1681_dac_tlv, -6350, 50, 1); >>> + >>> +static const struct snd_kcontrol_new pcm1681_controls[] = { >>> + SOC_DOUBLE_R_TLV("PCM1681 Channel 1/2 Playback Volume", >>> + PCM1681_ATT_CONTROL(1), PCM1681_ATT_CONTROL(2), 0, >>> + 0x7f, 0, pcm1681_dac_tlv), >>> + SOC_DOUBLE_R_TLV("PCM1681 Channel 3/4 Playback Volume", >>> + PCM1681_ATT_CONTROL(3), PCM1681_ATT_CONTROL(4), 0, >>> + 0x7f, 0, pcm1681_dac_tlv), >>> + SOC_DOUBLE_R_TLV("PCM1681 Channel 5/6 Playback Volume", >>> + PCM1681_ATT_CONTROL(5), PCM1681_ATT_CONTROL(6), 0, >>> + 0x7f, 0, pcm1681_dac_tlv), >>> + SOC_DOUBLE_R_TLV("PCM1681 Channel 7/8 Playback Volume", >>> + PCM1681_ATT_CONTROL(7), PCM1681_ATT_CONTROL(8), 0, >>> + 0x7f, 0, pcm1681_dac_tlv), >>> + SOC_SINGLE_BOOL_EXT("PCM1681 De-emphasis Switch", 0, >>> + pcm1681_get_deemph, pcm1681_put_deemph), >>> +}; >>> + >>> +struct snd_soc_dai_driver pcm1681_dai = { >>> + .name = "pcm1681-hifi", >>> + .playback = { >>> + .stream_name = "Playback", >>> + .channels_min = 2, >>> + .channels_max = 8, >>> + .rates = PCM1681_PCM_RATES, >>> + .formats = PCM1681_PCM_FORMATS, >>> + }, >>> + .ops = &pcm1681_dai_ops, >>> +}; >>> + >>> +#ifdef CONFIG_OF >>> +static const struct of_device_id pcm1681_dt_ids[] = { >>> + { .compatible = "ti,pcm1681", }, >>> + { } >>> +}; >>> +MODULE_DEVICE_TABLE(of, pcm1681_dt_ids); >>> +#endif >>> + >>> +static const struct regmap_config pcm1681_regmap = { >>> + .reg_bits = 8, >>> + .val_bits = 8, >>> + .max_register = ARRAY_SIZE(pcm1681_reg_defaults), >>> + .reg_defaults = pcm1681_reg_defaults, >>> + .num_reg_defaults = ARRAY_SIZE(pcm1681_reg_defaults), >>> + .writeable_reg = pcm1681_writeable_reg, >>> + .readable_reg = pcm1681_accessible_reg, >>> +}; >>> + >>> +static int pcm1681_probe(struct snd_soc_codec *codec) >>> +{ >>> + return 0; >> >> Is there really nothing the driver has to do in order to bring the codec >> into full operation mode? > This codec have only POWER-ON-RESET function. It means codec need system > clock (normally MCLK) for some amount of time to get out of reset and > then you can communicate with it. >> >>> +} >>> + >>> +static int pcm1681_remove(struct snd_soc_codec *codec) >>> +{ >>> + return 0; >>> +} >>> + >>> +static struct snd_soc_codec_driver soc_codec_dev_pcm1681 = { >>> + .probe = pcm1681_probe, >>> + .remove = pcm1681_remove, >>> + .controls = pcm1681_controls, >>> + .num_controls = ARRAY_SIZE(pcm1681_controls), >>> +}; >>> + >>> +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) >>> +static const struct i2c_device_id pcm1681_i2c_id[] = { >>> + {"pcm1681", 0}, >>> + {} >>> +}; >>> +MODULE_DEVICE_TABLE(i2c, pcm1681_i2c_id); >>> + >>> +static int pcm1681_i2c_probe(struct i2c_client *client, >>> + const struct i2c_device_id *id) >>> +{ >>> + int ret; >>> + struct pcm1681_private *priv; >>> + >>> + priv = devm_kzalloc(&client->dev, sizeof(*priv), GFP_KERNEL); >>> + if (!priv) >>> + return -ENOMEM; >>> + >>> + priv->regmap = devm_regmap_init_i2c(client, &pcm1681_regmap); >>> + if (IS_ERR(priv->regmap)) { >>> + ret = PTR_ERR(priv->regmap); >>> + dev_err(&client->dev, "Failed to create regmap: %d\n", ret); >>> + return ret; >>> + } >>> + >>> + i2c_set_clientdata(client, priv); >>> + >> >> Is there any way to probe the device is there at all? I'd like to bail >> early in case the bus or i2c address doesn't match for example. > Probing is done here because we assume that codec is out of reset. Yes, I know. I wondered whether there is any register that the driver can read back in order to check the presence of the device. But I quickly checked the datasheet, and there does in fact not seem to be such a register. >> Also, I wonder whether there are any reset lines that might be connected >> to GPIOs of the MCU. If so, it would be good idea to add support for >> that in this driver. But that can be done at some later point as well. > No other pins which can be handled. I see. Well ok then, I have no further remarks :) Daniel

1 0

[alsa-devel] [PATCH] ASoC: txx9aclc_ac97: Convert to devm_ioremap_resource()
by Mark Brown 26 Jun '13

26 Jun '13

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/txx9/txx9aclc-ac97.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c index 8a28403..8ee8d42 100644 --- a/sound/soc/txx9/txx9aclc-ac97.c +++ b/sound/soc/txx9/txx9aclc-ac97.c @@ -188,9 +188,9 @@ static int txx9aclc_ac97_dev_probe(struct platform_device *pdev) if (!r) return -EBUSY; - if (!devm_request_mem_region(&pdev->dev, r->start, resource_size(r), - dev_name(&pdev->dev))) - return -EBUSY; + drvdata->base = devm_ioremap_resource(&pdev->dev, r); + if (IS_ERR(drvdata->base)) + return PTR_ERR(drvdata->base); drvdata = devm_kzalloc(&pdev->dev, sizeof(*drvdata), GFP_KERNEL); if (!drvdata) @@ -201,9 +201,6 @@ static int txx9aclc_ac97_dev_probe(struct platform_device *pdev) r->start >= TXX9_DIRECTMAP_BASE && r->start < TXX9_DIRECTMAP_BASE + 0x400000) drvdata->physbase |= 0xf00000000ull; - drvdata->base = devm_ioremap(&pdev->dev, r->start, resource_size(r)); - if (!drvdata->base) - return -EBUSY; err = devm_request_irq(&pdev->dev, irq, txx9aclc_ac97_irq, 0, dev_name(&pdev->dev), drvdata); if (err < 0) -- 1.8.3.1

1 0

[alsa-devel] [PATCH] ASoC: nuc900-ac97: Convert to use devm_ APIs
by Mark Brown 26 Jun '13

26 Jun '13

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/nuc900/nuc900-ac97.c | 43 ++++++++++-------------------------------- 1 file changed, 10 insertions(+), 33 deletions(-) diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c index fe3285c..08e4c6b 100644 --- a/sound/soc/nuc900/nuc900-ac97.c +++ b/sound/soc/nuc900/nuc900-ac97.c @@ -326,32 +326,23 @@ static int nuc900_ac97_drvprobe(struct platform_device *pdev) if (nuc900_ac97_data) return -EBUSY; - nuc900_audio = kzalloc(sizeof(struct nuc900_audio), GFP_KERNEL); + nuc900_audio = devm_kzalloc(&pdev->dev, sizeof(struct nuc900_audio), + GFP_KERNEL); if (!nuc900_audio) return -ENOMEM; spin_lock_init(&nuc900_audio->lock); nuc900_audio->res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!nuc900_audio->res) { - ret = -ENODEV; - goto out0; - } + if (!nuc900_audio->res) + return ret; - if (!request_mem_region(nuc900_audio->res->start, - resource_size(nuc900_audio->res), pdev->name)) { - ret = -EBUSY; - goto out0; - } + nuc900_audio->mmio = devm_ioremap_resource(&pdev->dev, + nuc900_audio->res); + if (IS_ERR(nuc900_audio->mmio)) + return PTR_ERR(nuc900_audio->mmio); - nuc900_audio->mmio = ioremap(nuc900_audio->res->start, - resource_size(nuc900_audio->res)); - if (!nuc900_audio->mmio) { - ret = -ENOMEM; - goto out1; - } - - nuc900_audio->clk = clk_get(&pdev->dev, NULL); + nuc900_audio->clk = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(nuc900_audio->clk)) { ret = PTR_ERR(nuc900_audio->clk); goto out2; @@ -375,15 +366,7 @@ static int nuc900_ac97_drvprobe(struct platform_device *pdev) return 0; -out3: - clk_put(nuc900_audio->clk); -out2: - iounmap(nuc900_audio->mmio); -out1: - release_mem_region(nuc900_audio->res->start, - resource_size(nuc900_audio->res)); -out0: - kfree(nuc900_audio); +out: return ret; } @@ -391,12 +374,6 @@ static int nuc900_ac97_drvremove(struct platform_device *pdev) { snd_soc_unregister_component(&pdev->dev); - clk_put(nuc900_ac97_data->clk); - iounmap(nuc900_ac97_data->mmio); - release_mem_region(nuc900_ac97_data->res->start, - resource_size(nuc900_ac97_data->res)); - - kfree(nuc900_ac97_data); nuc900_ac97_data = NULL; return 0; -- 1.8.3.1

1 0

[alsa-devel] [PATCH] ASoC: ep93xx: Remove redundant dev_set_drvdata() calls
by Mark Brown 26 Jun '13

26 Jun '13

From: Mark Brown <broonie(a)linaro.org> The driver core does this and it's never legal to rely on the value of drvdata if not set in probe() anyway. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/cirrus/ep93xx-ac97.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/cirrus/ep93xx-ac97.c b/sound/soc/cirrus/ep93xx-ac97.c index 3f4f888..f193701 100644 --- a/sound/soc/cirrus/ep93xx-ac97.c +++ b/sound/soc/cirrus/ep93xx-ac97.c @@ -398,7 +398,6 @@ static int ep93xx_ac97_probe(struct platform_device *pdev) fail: ep93xx_ac97_info = NULL; - dev_set_drvdata(&pdev->dev, NULL); return ret; } @@ -412,7 +411,6 @@ static int ep93xx_ac97_remove(struct platform_device *pdev) ep93xx_ac97_write_reg(info, AC97GCR, 0); ep93xx_ac97_info = NULL; - dev_set_drvdata(&pdev->dev, NULL); return 0; } -- 1.8.3.1

1 0

[alsa-devel] [PATCH] ASoC: bf5xx-ac97: Convert to devm_gpio_request_one()
by Mark Brown 26 Jun '13

26 Jun '13

From: Mark Brown <broonie(a)linaro.org> Also clean up the error reporting from failed requests while we're at it. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/blackfin/bf5xx-ac97.c | 18 ++++++------------ 1 file changed, 6 insertions(+), 12 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index c66bef8..8010f32 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -293,13 +293,14 @@ static int asoc_bfin_ac97_probe(struct platform_device *pdev) #ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET /* Request PB3 as reset pin */ - if (gpio_request(CONFIG_SND_BF5XX_RESET_GPIO_NUM, "SND_AD198x RESET")) { - pr_err("Failed to request GPIO_%d for reset\n", - CONFIG_SND_BF5XX_RESET_GPIO_NUM); - ret = -1; + ret = devm_gpio_request_one(&pdev->dev, + CONFIG_SND_BF5XX_RESET_GPIO_NUM, + GPIOF_OUT_INIT_HIGH, "SND_AD198x RESET") { + dev_err(&pdev->dev, + "Failed to request GPIO_%d for reset: %d\n", + CONFIG_SND_BF5XX_RESET_GPIO_NUM, ret); goto gpio_err; } - gpio_direction_output(CONFIG_SND_BF5XX_RESET_GPIO_NUM, 1); #endif sport_handle = sport_init(pdev, 2, sizeof(struct ac97_frame), @@ -349,10 +350,6 @@ static int asoc_bfin_ac97_probe(struct platform_device *pdev) sport_config_err: sport_done(sport_handle); sport_err: -#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET - gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM); -gpio_err: -#endif return ret; } @@ -363,9 +360,6 @@ static int asoc_bfin_ac97_remove(struct platform_device *pdev) snd_soc_unregister_component(&pdev->dev); sport_done(sport_handle); -#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET - gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM); -#endif return 0; } -- 1.8.3.1

1 0

[alsa-devel] [PATCH 1/3] ASoC: psc-ac97: Use devm_ioremap_resource()
by Mark Brown 26 Jun '13

26 Jun '13

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/au1x/psc-ac97.c | 12 +++--------- 1 file changed, 3 insertions(+), 9 deletions(-) diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index 8f1862a..f5a3921 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -383,15 +383,9 @@ static int au1xpsc_ac97_drvprobe(struct platform_device *pdev) if (!iores) return -ENODEV; - if (!devm_request_mem_region(&pdev->dev, iores->start, - resource_size(iores), - pdev->name)) - return -EBUSY; - - wd->mmio = devm_ioremap(&pdev->dev, iores->start, - resource_size(iores)); - if (!wd->mmio) - return -EBUSY; + wd->mmio = devm_ioremap_resource(&pdev->dev, iores); + if (IS_ERR(wd->mmio)) + return PTR_ERR(wd->mmio); dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (!dmares) -- 1.8.3.1

1 2

[alsa-devel] [PATCH 1/3] ASoC: psc-ac97: Use devm_ioremap_resource()
by Mark Brown 26 Jun '13

26 Jun '13

Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/au1x/psc-ac97.c | 12 +++--------- 1 file changed, 3 insertions(+), 9 deletions(-) diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index 8f1862a..f5a3921 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -383,15 +383,9 @@ static int au1xpsc_ac97_drvprobe(struct platform_device *pdev) if (!iores) return -ENODEV; - if (!devm_request_mem_region(&pdev->dev, iores->start, - resource_size(iores), - pdev->name)) - return -EBUSY; - - wd->mmio = devm_ioremap(&pdev->dev, iores->start, - resource_size(iores)); - if (!wd->mmio) - return -EBUSY; + wd->mmio = devm_ioremap_resource(&pdev->dev, iores); + if (IS_ERR(wd->mmio)) + return PTR_ERR(wd->mmio); dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (!dmares) -- 1.8.3.1

1 0

Re: [alsa-devel] [PATCH 2/2] ASoC: samsung-ac97: Convert to devm_ioremap_resource()
by Mark Brown 26 Jun '13

26 Jun '13

On Wed, Jun 26, 2013 at 12:10:15PM +0200, Tomasz Figa wrote: > By the way, Mark, does this driver work for you? I have tried it on both > mini6410 and smdk6410 (configured to use the onboard AC97 codec), with both > old and new DMA drivers and I couldn't get it to play a single sound. I've no access to any hardware which might use this driver, though IIRC it didn't work last time I did have such access.

1 0

[alsa-devel] [PATCH] ASoC: samsung-ac97: Convert to devm_ioremap_resource()
by Mark Brown 26 Jun '13

26 Jun '13

Reviewed-by: Jingoo Han <jg1.han(a)samsung.com> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/samsung/ac97.c | 26 ++++---------------------- 1 file changed, 4 insertions(+), 22 deletions(-) diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index 1c85999..04d7fd4 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -417,11 +417,9 @@ static int s3c_ac97_probe(struct platform_device *pdev) return -ENXIO; } - if (!request_mem_region(mem_res->start, - resource_size(mem_res), "ac97")) { - dev_err(&pdev->dev, "Unable to request register region\n"); - return -EBUSY; - } + s3c_ac97.regs = devm_ioremap_resource(&pdev->dev, mem_res); + if (IS_ERR(s3c_ac97.regs)) + return PTR_ERR(s3c_ac97.regs); s3c_ac97_pcm_out.channel = dmatx_res->start; s3c_ac97_pcm_out.dma_addr = mem_res->start + S3C_AC97_PCM_DATA; @@ -433,13 +431,6 @@ static int s3c_ac97_probe(struct platform_device *pdev) init_completion(&s3c_ac97.done); mutex_init(&s3c_ac97.lock); - s3c_ac97.regs = ioremap(mem_res->start, resource_size(mem_res)); - if (s3c_ac97.regs == NULL) { - dev_err(&pdev->dev, "Unable to ioremap register region\n"); - ret = -ENXIO; - goto err1; - } - s3c_ac97.ac97_clk = devm_clk_get(&pdev->dev, "ac97"); if (IS_ERR(s3c_ac97.ac97_clk)) { dev_err(&pdev->dev, "ac97 failed to get ac97_clock\n"); @@ -481,16 +472,13 @@ err4: err3: clk_disable_unprepare(s3c_ac97.ac97_clk); err2: - iounmap(s3c_ac97.regs); -err1: - release_mem_region(mem_res->start, resource_size(mem_res)); return ret; } static int s3c_ac97_remove(struct platform_device *pdev) { - struct resource *mem_res, *irq_res; + struct resource *irq_res; asoc_dma_platform_unregister(&pdev->dev); snd_soc_unregister_component(&pdev->dev); @@ -501,12 +489,6 @@ static int s3c_ac97_remove(struct platform_device *pdev) clk_disable_unprepare(s3c_ac97.ac97_clk); - iounmap(s3c_ac97.regs); - - mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (mem_res) - release_mem_region(mem_res->start, resource_size(mem_res)); - return 0; } -- 1.8.3.1

1 0

[alsa-devel] [PATCH 1/2] ASoC: samsung-ac97: Use devm_clk_get()
by Mark Brown 26 Jun '13

26 Jun '13

Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/samsung/ac97.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index cb88ead..1c85999 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -440,7 +440,7 @@ static int s3c_ac97_probe(struct platform_device *pdev) goto err1; } - s3c_ac97.ac97_clk = clk_get(&pdev->dev, "ac97"); + s3c_ac97.ac97_clk = devm_clk_get(&pdev->dev, "ac97"); if (IS_ERR(s3c_ac97.ac97_clk)) { dev_err(&pdev->dev, "ac97 failed to get ac97_clock\n"); ret = -ENODEV; @@ -480,7 +480,6 @@ err5: err4: err3: clk_disable_unprepare(s3c_ac97.ac97_clk); - clk_put(s3c_ac97.ac97_clk); err2: iounmap(s3c_ac97.regs); err1: @@ -501,7 +500,6 @@ static int s3c_ac97_remove(struct platform_device *pdev) free_irq(irq_res->start, NULL); clk_disable_unprepare(s3c_ac97.ac97_clk); - clk_put(s3c_ac97.ac97_clk); iounmap(s3c_ac97.regs); -- 1.8.3.1

1 1

[alsa-devel] [PATCH alsa-lib 1/2] pcm_hw: Remove unused fields in struct
by David Henningsson 26 Jun '13

26 Jun '13

These fields are not used, and their name similarity to other fields are quite confusing when trying to debug alsa-lib. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- src/pcm/pcm_hw.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/src/pcm/pcm_hw.c b/src/pcm/pcm_hw.c index 0f3226d..196393d 100644 --- a/src/pcm/pcm_hw.c +++ b/src/pcm/pcm_hw.c @@ -95,8 +95,6 @@ typedef struct { volatile struct snd_pcm_mmap_status * mmap_status; struct snd_pcm_mmap_control *mmap_control; struct snd_pcm_sync_ptr *sync_ptr; - snd_pcm_uframes_t hw_ptr; - snd_pcm_uframes_t appl_ptr; int period_event; snd_timer_t *period_timer; struct pollfd period_timer_pfd; -- 1.7.9.5

3 4

[alsa-devel] *.alsa-project.org web services down?
by Arun Raghavan 26 Jun '13

26 Jun '13

Hello, I can't seem to get to alsa-project.org or git.alsa-project.org. Is the web server down? -- Arun

2 1

[alsa-devel] [patch] ALSA: ak4xx-adda: info leak in ak4xxx_capture_source_info()
by Dan Carpenter 26 Jun '13

26 Jun '13

"idx" is controled by the user and can be a negative offset into the input_names[] array. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/i2c/other/ak4xxx-adda.c b/sound/i2c/other/ak4xxx-adda.c index cef813d2..ed726d1 100644 --- a/sound/i2c/other/ak4xxx-adda.c +++ b/sound/i2c/other/ak4xxx-adda.c @@ -571,7 +571,7 @@ static int ak4xxx_capture_source_info(struct snd_kcontrol *kcontrol, struct snd_akm4xxx *ak = snd_kcontrol_chip(kcontrol); int mixer_ch = AK_GET_SHIFT(kcontrol->private_value); const char **input_names; - int num_names, idx; + unsigned int num_names, idx; num_names = ak4xxx_capture_num_inputs(ak, mixer_ch); if (!num_names)

2 1

[alsa-devel] PulseAudio and softvol
by Arun Raghavan 26 Jun '13

26 Jun '13

Hello, A number of users have intermittently(?) been hitting a crash in alsa-lib 1.0.27 [1, 2] related to the softvol plugin. I'm not able to reproduce this reliably, so can't find an easy way to debug/fix. However, this raises a tangential question - why do we need softvol to be plugged for 'front' at all? David explained to me that this is to guarantee the existence of a PCM control. Perhaps I don't fully understand this, because I'm unconvinced by the reason. Could someone explain/refute? This is especially bad for us, from PulseAudio's perspective, because we aren't getting a zero-copy path. Cheers, Arun [1] https://bugzilla.redhat.com/show_bug.cgi?id=953352 [2] https://bugs.freedesktop.org/show_bug.cgi?id=64299

5 25

[alsa-devel] [PATCH 1/3] clk: mxs: clk-imx28: Provide a dummy clock
by Fabio Estevam 26 Jun '13

26 Jun '13

Sometimes it is useful to be able to pass a dummy clock for a driver. Add support for a dummy clock. Cc: Mike Turquette <mturquette(a)linaro.org> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Documentation/devicetree/bindings/clock/imx28-clock.txt | 1 + drivers/clk/mxs/clk-imx28.c | 3 ++- 2 files changed, 3 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/clock/imx28-clock.txt b/Documentation/devicetree/bindings/clock/imx28-clock.txt index e6587af..5a153ef 100644 --- a/Documentation/devicetree/bindings/clock/imx28-clock.txt +++ b/Documentation/devicetree/bindings/clock/imx28-clock.txt @@ -76,6 +76,7 @@ clocks and IDs. usb0_phy 62 usb1_phy 63 enet_out 64 + dummy 65 Examples: diff --git a/drivers/clk/mxs/clk-imx28.c b/drivers/clk/mxs/clk-imx28.c index 4faf0af..6b55211 100644 --- a/drivers/clk/mxs/clk-imx28.c +++ b/drivers/clk/mxs/clk-imx28.c @@ -143,7 +143,7 @@ enum imx28_clk { emi_xtal, lcdif_div, etm_div, ptp, saif0_div, saif1_div, clk32k_div, rtc, lradc, spdif_div, clk32k, pwm, uart, ssp0, ssp1, ssp2, ssp3, gpmi, spdif, emi, saif0, saif1, lcdif, etm, - fec, can0, can1, usb0, usb1, usb0_phy, usb1_phy, enet_out, + fec, can0, can1, usb0, usb1, usb0_phy, usb1_phy, enet_out, dummy, clk_max }; @@ -234,6 +234,7 @@ int __init mx28_clocks_init(void) clks[usb0_phy] = clk_register_gate(NULL, "usb0_phy", "pll0", 0, PLL0CTRL0, 18, 0, &mxs_lock); clks[usb1_phy] = clk_register_gate(NULL, "usb1_phy", "pll1", 0, PLL1CTRL0, 18, 0, &mxs_lock); clks[enet_out] = clk_register_gate(NULL, "enet_out", "pll2", 0, ENET, 18, 0, &mxs_lock); + clks[dummy] = mxs_clk_fixed("dummy", 0); for (i = 0; i < ARRAY_SIZE(clks); i++) if (IS_ERR(clks[i])) { -- 1.8.1.2

3 4

Re: [alsa-devel] [ALSA] Enable capture from line-in and CD on Revolution 5.1
by Dan Carpenter 26 Jun '13

26 Jun '13

[ This warning is very old, but I was working on some related code and didn't know how to fix it ]. Hello Jochen Voss, This is a semi-automatic email about new static checker warnings. The patch a58e7cb16dfa: "[ALSA] Enable capture from line-in and CD on Revolution 5.1" from Oct 4, 2006, leads to the following Smatch complaint: sound/i2c/other/ak4xxx-adda.c:808 build_adc_controls() error: we previously assumed 'ak->adc_info' could be null (see line 789) sound/i2c/other/ak4xxx-adda.c 788 if (ak->type == SND_AK5365 && (idx % 2) == 0) { 789 if (! ak->adc_info || ^^^^^^^^^^^^ Old check. 790 ! ak->adc_info[mixer_ch].switch_name) { 791 knew.name = "Capture Switch"; 792 knew.index = mixer_ch + ak->idx_offset * 2; 793 } else 794 knew.name = ak->adc_info[mixer_ch].switch_name; 795 knew.info = ak4xxx_switch_info; 796 knew.get = ak4xxx_switch_get; 797 knew.put = ak4xxx_switch_put; 798 knew.access = 0; 799 /* register 2, bit 0 (SMUTE): 0 = normal operation, 800 1 = mute */ 801 knew.private_value = 802 AK_COMPOSE(idx/2, 2, 0, 0) | AK_INVERT; 803 err = snd_ctl_add(ak->card, snd_ctl_new1(&knew, ak)); 804 if (err < 0) 805 return err; 806 807 memset(&knew, 0, sizeof(knew)); 808 knew.name = ak->adc_info[mixer_ch].selector_name; ^^^^^^^^^^^^^^^^^^^^^ New dereference. 809 if (!knew.name) { 810 knew.name = "Capture Channel"; regards, dan carpenter

1 0

[alsa-devel] [PATCH] ASoC: wm9705: Remove noisy print on boot
by Mark Brown 25 Jun '13

25 Jun '13

There's no content in the announcement. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/wm9705.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index 05b1f34..a5fc61d 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -337,8 +337,6 @@ static int wm9705_soc_probe(struct snd_soc_codec *codec) { int ret = 0; - printk(KERN_INFO "WM9705 SoC Audio Codec\n"); - ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); if (ret < 0) { printk(KERN_ERR "wm9705: failed to register AC97 codec\n"); -- 1.8.3.1

1 0

[alsa-devel] [PATCH] ASoC: stac9766: Remove version number
by Mark Brown 25 Jun '13

25 Jun '13

There is no need to have versioning beyond that for the kernel, especially when the version number never gets updated. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/stac9766.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index 2eda85ba..cbc7ae3 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -28,8 +28,6 @@ #include "stac9766.h" -#define STAC9766_VERSION "0.10" - /* * STAC9766 register cache */ @@ -338,8 +336,6 @@ static int stac9766_codec_probe(struct snd_soc_codec *codec) { int ret = 0; - printk(KERN_INFO "STAC9766 SoC Audio Codec %s\n", STAC9766_VERSION); - ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); if (ret < 0) goto codec_err; -- 1.8.3.1

1 0

Re: [alsa-devel] [RFC PATCH v2 00/12] ARM: s3c64xx: Let amba-pl08x driver handle DMA
by Mark Brown 25 Jun '13

25 Jun '13

On Tue, Jun 25, 2013 at 01:22:20PM +0200, Tomasz Figa wrote: > On Tuesday 25 of June 2013 11:28:18 Mark Brown wrote: > > On my non-DT s3c64xx system I get this when trying to use SPI for the > > first time: > Could you clarify first time? Does it work on further retries? No, the system appears to lock up fairly solidly - I suspect poor error handling in the SPI driver. > > [ 1.739385] s3c64xx-spi s3c6410-spi.0: Failed to get RX DMA channel > > [ 1.743437] spi_master spi0: failed to prepare transfer hardware > > though I haven't yet tried your clock patches (there were some conficts > > that I didn't get round to picking up yet). > This shouldn't really matter, as long as the pl080 drivers probes > correctly. Indeed, I was wondering if it was failing to probe due to not getting its clocks though since that's another area you've been actively working on.

1 0

[alsa-devel] [PATCH 1/5] ASoC: codecs: tas5086: fix documentation
by Daniel Mack 25 Jun '13

25 Jun '13

The property 'ti,charge-period' must be given in nanoseconds, not microseconds. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- Documentation/devicetree/bindings/sound/ti,tas5086.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/ti,tas5086.txt b/Documentation/devicetree/bindings/sound/ti,tas5086.txt index 8ea4f5b..e8310f1 100644 --- a/Documentation/devicetree/bindings/sound/ti,tas5086.txt +++ b/Documentation/devicetree/bindings/sound/ti,tas5086.txt @@ -11,7 +11,7 @@ Optional properties: chip's !RESET pin. If specified, the driver will assert a hardware reset at probe time. - - ti,charge-period: This property should contain the time in microseconds + - ti,charge-period: This property should contain the time in nanoseconds that closely matches the external single-ended split-capacitor charge period. The hardware chip waits for this period of time before starting the -- 1.8.1.4

2 7

[alsa-devel] [PATCH] ASoC: Tegra and i.MX are mutually exclusive
by Arnd Bergmann 25 Jun '13

25 Jun '13

Both Tegra and i.MX provide ac97 operations, which can only exist once in the kernel. They can both be built as loadable modules (only one of them needs to be loaded anyway), but we have to disallow enabling them both as built-in. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 3843a18..5958948 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -99,6 +99,7 @@ endif # SND_POWERPC_SOC menuconfig SND_IMX_SOC tristate "SoC Audio for Freescale i.MX CPUs" depends on ARCH_MXC + depends on SND_SOC_TEGRA=n || (SND_SOC_TEGRA=m && m) help Say Y or M if you want to add support for codecs attached to the i.MX CPUs.

4 4

[alsa-devel] [PATCH] snd-firewire-lib: remove unused header inclusion
by Takashi Sakamoto 25 Jun '13

25 Jun '13

spinlock is not used in amdtp.h. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp.h | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/firewire/amdtp.h b/sound/firewire/amdtp.h index b680c5e..f6103d6 100644 --- a/sound/firewire/amdtp.h +++ b/sound/firewire/amdtp.h @@ -3,7 +3,6 @@ #include <linux/interrupt.h> #include <linux/mutex.h> -#include <linux/spinlock.h> #include "packets-buffer.h" /** -- 1.7.10.4

2 1

[alsa-devel] [PATCH] ALSA: hda - clean up code to reset hda link
by mengdong.lin＠intel.com 25 Jun '13

25 Jun '13

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch is a cleanup to the previous patch "reset hda link during system/ runtime suspend". In this patch - azx_enter_link_reset() and azx_exit_link_reset() are defined for entering and exiting the link reset respectively. azx_link_reset() is no longer used and replaced by azx_enter_link_reset(). - azx_reset() reuses the above two new functions for a link reset cycle 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 9f110c7..f39de90 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1121,7 +1121,7 @@ static void azx_load_dsp_cleanup(struct hda_bus *bus, #endif /* enter link reset */ -static void azx_reset_link(struct azx *chip) +static void azx_enter_link_reset(struct azx *chip) { unsigned long timeout; @@ -1134,11 +1134,22 @@ static void azx_reset_link(struct azx *chip) usleep_range(500, 1000); } -/* reset codec link */ -static int azx_reset(struct azx *chip, int full_reset) +/* exit link reset */ +static void azx_exit_link_reset(struct azx *chip) { unsigned long timeout; + azx_writeb(chip, GCTL, azx_readb(chip, GCTL) | ICH6_GCTL_RESET); + + timeout = jiffies + msecs_to_jiffies(100); + while (!azx_readb(chip, GCTL) && + time_before(jiffies, timeout)) + usleep_range(500, 1000); +} + +/* reset codec link */ +static int azx_reset(struct azx *chip, int full_reset) +{ if (!full_reset) goto __skip; @@ -1146,12 +1157,7 @@ static int azx_reset(struct azx *chip, int full_reset) azx_writeb(chip, STATESTS, STATESTS_INT_MASK); /* reset controller */ - azx_writel(chip, GCTL, azx_readl(chip, GCTL) & ~ICH6_GCTL_RESET); - - timeout = jiffies + msecs_to_jiffies(100); - while (azx_readb(chip, GCTL) && - time_before(jiffies, timeout)) - usleep_range(500, 1000); + azx_enter_link_reset(chip); /* delay for >= 100us for codec PLL to settle per spec * Rev 0.9 section 5.5.1 @@ -1159,12 +1165,7 @@ static int azx_reset(struct azx *chip, int full_reset) usleep_range(500, 1000); /* Bring controller out of reset */ - azx_writeb(chip, GCTL, azx_readb(chip, GCTL) | ICH6_GCTL_RESET); - - timeout = jiffies + msecs_to_jiffies(100); - while (!azx_readb(chip, GCTL) && - time_before(jiffies, timeout)) - usleep_range(500, 1000); + azx_exit_link_reset(chip); /* Brent Chartrand said to wait >= 540us for codecs to initialize */ usleep_range(1000, 1200); @@ -2908,7 +2909,7 @@ static int azx_suspend(struct device *dev) if (chip->initialized) snd_hda_suspend(chip->bus); azx_stop_chip(chip); - azx_reset_link(chip); + azx_enter_link_reset(chip); if (chip->irq >= 0) { free_irq(chip->irq, chip); chip->irq = -1; @@ -2961,7 +2962,7 @@ static int azx_runtime_suspend(struct device *dev) struct azx *chip = card->private_data; azx_stop_chip(chip); - azx_reset_link(chip); + azx_enter_link_reset(chip); azx_clear_irq_pending(chip); return 0; } -- 1.7.10.4

2 1

[alsa-devel] [PATCH] ALSA: hda - Add In-driver connection info
by Wang Xingchao 25 Jun '13

25 Jun '13

Pin's connection list may change dynamically with hot-plug event on Intel Haswell chip. Users would see connections be "0" in codec#. when play audio on this pin, software driver choose converter from cache connections. So add "In-driver connection" info to avoid confuse when raw connections are different with cache connection. Signed-off-by: Wang Xingchao <xingchao.wang(a)linux.intel.com> --- sound/pci/hda/hda_proc.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 0fee8fa..9760f00 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -504,6 +504,8 @@ static void print_conn_list(struct snd_info_buffer *buffer, int conn_len) { int c, curr = -1; + const hda_nid_t *list; + int cache_len; if (conn_len > 1 && wid_type != AC_WID_AUD_MIX && @@ -521,6 +523,19 @@ static void print_conn_list(struct snd_info_buffer *buffer, } snd_iprintf(buffer, "\n"); } + + /* Get Cache connections info */ + cache_len = snd_hda_get_conn_list(codec, nid, &list); + if (cache_len != conn_len + || memcmp(list, conn, conn_len)) { + snd_iprintf(buffer, " In-driver Connection: %d\n", cache_len); + if (cache_len > 0) { + snd_iprintf(buffer, " "); + for (c = 0; c < cache_len; c++) + snd_iprintf(buffer, " 0x%02x", list[c]); + snd_iprintf(buffer, "\n"); + } + } } static void print_gpio(struct snd_info_buffer *buffer, -- 1.8.1.2

2 1

[alsa-devel] [PATCH v3 0/3] ASoC: codecs: some more improvements for adau1701
by Daniel Mack 25 Jun '13

25 Jun '13

v3: - fixed documentation for adi,pll-clkdiv - droped implicit I2C_M_STOP - moved regcache_sync() to adau1761_reset() - added regcache_mark_dirty() - made gpio_pll_mode and array - dropped ADAU1701_CLKDIV_MCLK_LRCLK Thanks for the review, Lars! Daniel Mack (3): ASoC: codecs: adau1701: allow configuration of PLL mode pins ASoC: codecs: adau1701: switch to direct regmap API usage ASoC: codecs: adau1701: add support for pin muxing .../devicetree/bindings/sound/adi,adau1701.txt | 21 ++ sound/soc/codecs/adau1701.c | 248 ++++++++++++++++----- 2 files changed, 214 insertions(+), 55 deletions(-) -- 1.8.1.4

5 11

Re: [alsa-devel] [PATCH - oss plugin update 1/2] oss: Add support of 32-bit and other formats
by Takashi Iwai 25 Jun '13

25 Jun '13

At Tue, 25 Jun 2013 03:38:11 +0100, Alex Wiggins wrote: > > Thank you for the prompt review Takashi. > > By #if/#endif in this case, do you mean #ifdef to see if the formats are > defined or something else? Not necessarily to be so fine grained. The point is that commenting out multiple lines of codes is a bad style. For disabling multiple lines of codes, use #if 0 or #ifndef XXX or whatever. Takashi > Good spot on the formats array - I guess I didn't have any soundcards with > more than 5 formats but you're right it needs to be increased now. > > I'll have a look at fixing these in the weekend and resubmitting. > > Best Regards, > > Alex > > > On 24 June 2013 08:54, Takashi Iwai <tiwai(a)suse.de> wrote: > > > At Sun, 23 Jun 2013 01:25:18 +0100, > > alex.d.wiggins(a)gmail.com wrote: > > > > > > From: Alex Wiggins <alex.d.wiggins(a)gmail.com> > > > > > > Added support of 32-bit and other audio formats via 3 changes: > > > > > > - Additional audio format definitions > > > at the top of the file (if required); > > > > > > - Additional audio formats in oss_hw_params > > > (to allow as input to pass to OSS); > > > > > > - Additional audio formats in oss_hw_constraint > > > (to allow to report to ALSA). > > > > > > Signed-off-by: Alex Wiggins <alex.d.wiggins(a)gmail.com> > > > > > > diff --git a/oss/pcm_oss.c b/oss/pcm_oss.c > > > index d43b44c..6d8dea4 100644 > > > --- a/oss/pcm_oss.c > > > +++ b/oss/pcm_oss.c > > > @@ -24,6 +24,32 @@ > > > #include <alsa/pcm_external.h> > > > #include <linux/soundcard.h> > > > > > > +/* definitions for additional OSS formats if required */ > > > +#ifndef AFMT_S32_LE > > > +#define AFMT_S32_LE 0x00001000 > > > +#endif > > > +#ifndef AFMT_S32_BE > > > +#define AFMT_S32_BE 0x00002000 > > > +#endif > > > +#ifndef AFMT_S24_LE > > > +#define AFMT_S24_LE 0x00008000 > > > +#endif > > > +#ifndef AFMT_S24_BE > > > +#define AFMT_S24_BE 0x00010000 > > > +#endif > > > +#ifndef AFMT_S24_PACKED > > > +#define AFMT_S24_PACKED 0x00040000 > > > +#endif > > > +#ifndef AFMT_FLOAT > > > +#define AFMT_FLOAT 0x00004000 > > > +#endif > > > +#ifndef AFMT_SPDIF_RAW > > > +#define AFMT_SPDIF_RAW 0x00020000 > > > +#endif > > > +#ifndef AFMT_VORBIS > > > +#define AFMT_VORBIS 0x00000800 > > > +#endif > > > + > > > typedef struct snd_pcm_oss { > > > snd_pcm_ioplug_t io; > > > char *device; > > > @@ -165,6 +191,143 @@ static int oss_hw_params(snd_pcm_ioplug_t *io, > > > case SND_PCM_FORMAT_S16_BE: > > > oss->format = AFMT_S16_BE; > > > break; > > > + /* allow additional data formats (such as 32 bit) as input > > > + to pass to OSS (set in OSS driver) */ > > > + case SND_PCM_FORMAT_S32_LE: > > > + oss->format = AFMT_S32_LE; > > > + break; > > > + case SND_PCM_FORMAT_S32_BE: > > > + oss->format = AFMT_S32_BE; > > > + break; > > > + case SND_PCM_FORMAT_S8: > > > + oss->format = AFMT_S8; > > > + break; > > > + case SND_PCM_FORMAT_U16_LE: > > > + oss->format = AFMT_U16_LE; > > > + break; > > > + case SND_PCM_FORMAT_U16_BE: > > > + oss->format = AFMT_U16_BE; > > > + break; > > > + case SND_PCM_FORMAT_S24_LE: > > > + oss->format = AFMT_S24_LE; > > > + break; > > > + case SND_PCM_FORMAT_S24_BE: > > > + oss->format = AFMT_S24_BE; > > > + break; > > > + case SND_PCM_FORMAT_MU_LAW: > > > + oss->format = AFMT_MU_LAW; > > > + break; > > > + case SND_PCM_FORMAT_A_LAW: > > > + oss->format = AFMT_A_LAW; > > > + break; > > > + case SND_PCM_FORMAT_IMA_ADPCM: > > > + oss->format = AFMT_IMA_ADPCM; > > > + break; > > > + case SND_PCM_FORMAT_MPEG: > > > + oss->format = AFMT_MPEG; > > > + break; > > > + case SND_PCM_FORMAT_S24_3LE: > > > + oss->format = AFMT_S24_PACKED; > > > + break; > > > + case SND_PCM_FORMAT_FLOAT: > > > + oss->format = AFMT_FLOAT; > > > + break; > > > + case SND_PCM_FORMAT_IEC958_SUBFRAME: > > > + oss->format = AFMT_SPDIF_RAW; > > > + break; > > > + /* following formats need defining checking native endian: > > > + case SND_PCM_FORMAT_S16: > > > + oss->format = AFMT_S16_NE; > > > + break; > > > + case SND_PCM_FORMAT_U16: > > > + oss->format = AFMT_U16_NE; > > > + break; > > > + case SND_PCM_FORMAT_S24: > > > + oss->format = AFMT_S24_NE; > > > + break; > > > + case SND_PCM_FORMAT_S32: > > > + oss->format = AFMT_S32_NE; > > > + */ > > > > Use #if/#endif instead of multi-line comments. > > > > > + /* following formats found in ALSA but not in OSS: > > > + case SND_PCM_FORMAT_U24_LE: > > > + oss->format = AFMT_U24_LE; > > > + break; > > > + case SND_PCM_FORMAT_U24_BE: > > > + oss->format = AFMT_U24_BE; > > > + break; > > > + case SND_PCM_FORMAT_U32_LE: > > > + oss->format = AFMT_U32_LE; > > > + break; > > > + case SND_PCM_FORMAT_U32_BE: > > > + oss->format = AFMT_U32_BE; > > > + break; > > > + case SND_PCM_FORMAT_FLOAT_LE: > > > + oss->format = AFMT_FLOAT_LE; > > > + break; > > > + case SND_PCM_FORMAT_FLOAT_BE: > > > + oss->format = AFMT_FLOAT_BE; > > > + break; > > > + case SND_PCM_FORMAT_FLOAT64_LE: > > > + oss->format = AFMT_FLOAT64_LE; > > > + break; > > > + case SND_PCM_FORMAT_FLOAT64_BE: > > > + oss->format = AFMT_FLOAT64_BE; > > > + break; > > > + case SND_PCM_FORMAT_IEC958_SUBFRAME_LE: > > > + oss->format = AFMT_SPDIF_RAW_LE; > > > + break; > > > + case SND_PCM_FORMAT_IEC958_SUBFRAME_BE: > > > + oss->format = AFMT_SPDIF_RAW_BE; > > > + break; > > > + case SND_PCM_FORMAT_GSM: > > > + oss->format = AFMT_GSM; > > > + break; > > > + case SND_PCM_FORMAT_SPECIAL: > > > + oss->format = AFMT_SPECIAL; > > > + break; > > > + case SND_PCM_FORMAT_S24_3BE: > > > + oss->format = AFMT_S24_PACKED_BE; > > > + break; > > > + case SND_PCM_FORMAT_U24_3LE: > > > + oss->format = AFMT_U24_PACKED; > > > + break; > > > + case SND_PCM_FORMAT_U24_3BE: > > > + oss->format = AFMT_U24_PACKED_BE; > > > + break; > > > + case SND_PCM_FORMAT_S20_3LE: > > > + oss->format = ; > > > + break; > > > + case SND_PCM_FORMAT_S20_3BE: > > > + oss->format = ; > > > + break; > > > + case SND_PCM_FORMAT_U20_3LE: > > > + oss->format = ; > > > + break; > > > + case SND_PCM_FORMAT_U20_3BE: > > > + oss->format = ; > > > + break; > > > + case SND_PCM_FORMAT_S18_3LE: > > > + oss->format = ; > > > + break; > > > + case SND_PCM_FORMAT_S18_3BE: > > > + oss->format = ; > > > + break; > > > + case SND_PCM_FORMAT_U18_3LE: > > > + oss->format = ; > > > + break; > > > + case SND_PCM_FORMAT_U18_3BE: > > > + oss->format = ; > > > + break; > > > + case SND_PCM_FORMAT_U24: > > > + oss->format = AFMT_U24_NE; > > > + break; > > > + case SND_PCM_FORMAT_U32: > > > + oss->format = AFMT_U32_NE; > > > + break; > > > + case SND_PCM_FORMAT_FLOAT64: > > > + oss->format = AFMT_FLOAT64; > > > + break; > > > + */ > > > > Ditto. > > > > > > > default: > > > fprintf(stderr, "*** OSS: unsupported format %s\n", > > snd_pcm_format_name(io->format)); > > > return -EINVAL; > > > @@ -273,6 +436,40 @@ static int oss_hw_constraint(snd_pcm_oss_t *oss) > > > format[nformats++] = SND_PCM_FORMAT_S16_BE; > > > if (tmp & AFMT_MU_LAW) > > > format[nformats++] = SND_PCM_FORMAT_MU_LAW; > > > + /* allow additional data formats (such as 32 bit) > > > + to be reported from OSS to ALSA below */ > > > + if (tmp & AFMT_S32_LE) > > > + format[nformats++] = SND_PCM_FORMAT_S32_LE; > > > + if (tmp & AFMT_S32_BE) > > > + format[nformats++] = SND_PCM_FORMAT_S32_BE; > > > + if (tmp & AFMT_A_LAW) > > > + format[nformats++] = SND_PCM_FORMAT_A_LAW; > > > + if (tmp & AFMT_IMA_ADPCM) > > > + format[nformats++] = SND_PCM_FORMAT_IMA_ADPCM; > > > + if (tmp & AFMT_S8) > > > + format[nformats++] = SND_PCM_FORMAT_S8; > > > + if (tmp & AFMT_U16_LE) > > > + format[nformats++] = SND_PCM_FORMAT_U16_LE; > > > + if (tmp & AFMT_U16_BE) > > > + format[nformats++] = SND_PCM_FORMAT_U16_BE ; > > > + if (tmp & AFMT_MPEG) > > > + format[nformats++] = SND_PCM_FORMAT_MPEG; > > > + if (tmp & AFMT_FLOAT) > > > + format[nformats++] = SND_PCM_FORMAT_FLOAT; > > > + if (tmp & AFMT_S24_LE) > > > + format[nformats++] = SND_PCM_FORMAT_S24_LE; > > > + if (tmp & AFMT_S24_BE) > > > + format[nformats++] = SND_PCM_FORMAT_S24_BE; > > > + if (tmp & AFMT_SPDIF_RAW) > > > + format[nformats++] = SND_PCM_FORMAT_IEC958_SUBFRAME; > > > + if (tmp & AFMT_S24_PACKED) > > > + format[nformats++] = SND_PCM_FORMAT_S24_3LE; > > > + /* following formats found in OSS but not in ALSA */ > > > + if (tmp & AFMT_AC3) > > > + format[nformats++] = SND_PCM_FORMAT_UNKNOWN; > > > + if (tmp & AFMT_VORBIS) > > > + format[nformats++] = SND_PCM_FORMAT_UNKNOWN; > > > + /* assume signed 16 bit (native CPU endian) if not found above */ > > > if (! nformats) > > > format[nformats++] = SND_PCM_FORMAT_S16; > > > if ((err = snd_pcm_ioplug_set_param_list(io, > > SND_PCM_IOPLUG_HW_FORMAT, > > > > You need to adjust format[] array size if you add more possible > > formats. > > > > > > thanks, > > > > Takashi > > > [2 <text/html; ISO-8859-1 (quoted-printable)>] >

1 0

[alsa-devel] [PATCH 1/2] ALSA: hda - Remove unused variable
by Wang Xingchao 25 Jun '13

25 Jun '13

Signed-off-by: Wang Xingchao <xingchao.wang(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 7803ddd..0f7a380 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1891,7 +1891,6 @@ static void intel_haswell_fixup_connect_list(struct hda_codec *codec, /* override pins connection list */ snd_printdd("hdmi: haswell: override pin connection 0x%x\n", nid); - nconns = max(spec->num_cvts, 4); snd_hda_override_conn_list(codec, nid, spec->num_cvts, spec->cvt_nids); } -- 1.8.1.2

3 5

[alsa-devel] [PATCH v4 0/4] ASoC: codecs: some more improvements for adau1701
by Daniel Mack 24 Jun '13

24 Jun '13

v4: - added one more patch to move the firmware download, for more clarity. - postpone the firmware download to the first call of hw_params(), so we don't need to pass a default pll div in DT. - drop adi,pll-clkdiv entirely v3: - fixed documentation for adi,pll-clkdiv - droped implicit I2C_M_STOP - moved regcache_sync() to adau1761_reset() - added regcache_mark_dirty() - made gpio_pll_mode and array - dropped ADAU1701_CLKDIV_MCLK_LRCLK Daniel Mack (4): ASoC: codecs: adau1701: move firmware download to adau1701_reset() ASoC: codecs: adau1701: allow configuration of PLL mode pins ASoC: codecs: adau1701: switch to direct regmap API usage ASoC: codecs: adau1701: add support for pin muxing .../devicetree/bindings/sound/adi,adau1701.txt | 12 + sound/soc/codecs/adau1701.c | 277 ++++++++++++++++----- 2 files changed, 232 insertions(+), 57 deletions(-) -- 1.8.1.4

2 6

[alsa-devel] [PATCH 0/3] Support auto-mute via amp in HD-audio generic driver
by Takashi Iwai 24 Jun '13

24 Jun '13

Hi, this is a patch set to add the support of the headphone/line-out auto- mute via amp mute instead of pin-control values. The motivation of this is that we found that some machines don't honor the pin-control values but just ignore and keep speaker on. So, instead of muting via the pin control value, we need to mute via the amp value of the pin. The mechanism to add an extra mute check in software is relatively easy: just add the new mute flag and check it in the mixer put callback. The only tricky part is to update all relevant controls by jack plug/unplug or the auto-mute switch. For that, the new helper function, snd_ctl_vmaster_sync(), is added, so that the codec driver just updates the bit mask and call the sync at the last. There should be no behavior change unless the new flag is set. [PATCH 1/3] ALSA: vmaster: Add snd_ctl_sync_vmaster() helper function [PATCH 2/3] ALSA: hda - Add auto_mute_via_amp flag to generic parser [PATCH 3/3] ALSA: hda - Use auto_mute_via_amp=1 for VT1708 Takashi

1 3

[alsa-devel] [GIT PULL] ASoC updates for v3.10
by Mark Brown 24 Jun '13

24 Jun '13

The following changes since commit 7d132055814ef17a6c7b69f342244c410a5e000f: Linux 3.10-rc6 (2013-06-15 11:51:07 -1000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.10-rc7 for you to fetch changes up to b12a530682ec725ff7a95871d8003d1753e315b8: Merge remote-tracking branch 'asoc/fix/wm8962' into asoc-linus (2013-06-19 19:49:23 +0100) ---------------------------------------------------------------- ASoC: Updates for v3.10 A few small updates, one small core fix and several simple driver ones. The wm8962 changes are a little large but are pretty important. ---------------------------------------------------------------- Axel Lin (1): ASoC: adav80x: Select SND_SOC_ADAV80X if SND_SOC_I2C_AND_SPI Charles Keepax (1): ASoC: wm5110: Correct rate control for DSP4 Lars-Peter Clausen (1): ASoC: dapm: Add snd_soc_dapm_switch to the power up/down sequence table Mark Brown (4): Merge remote-tracking branch 'asoc/fix/adav80x' into asoc-linus Merge remote-tracking branch 'asoc/fix/adsp' into asoc-linus Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branch 'asoc/fix/wm8962' into asoc-linus Nicolin Chen (2): ASoC: wm8962: Remove remaining direct register cache accesses ASoC: wm8962: Enable start-up and normal bias after reset in runtime resume sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/wm8962.c | 35 ++++++++++++++++++++++++----------- sound/soc/codecs/wm_adsp.c | 2 +- sound/soc/soc-dapm.c | 2 ++ 4 files changed, 28 insertions(+), 13 deletions(-)

2 2

[alsa-devel] [PATCH 1/2] ALSA: hda - Remove unused variable
by Wang Xingchao 24 Jun '13

24 Jun '13

Signed-off-by: Wang Xingchao <xingchao.wang(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 7803ddd..0f7a380 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1891,7 +1891,6 @@ static void intel_haswell_fixup_connect_list(struct hda_codec *codec, /* override pins connection list */ snd_printdd("hdmi: haswell: override pin connection 0x%x\n", nid); - nconns = max(spec->num_cvts, 4); snd_hda_override_conn_list(codec, nid, spec->num_cvts, spec->cvt_nids); } -- 1.8.1.2

3 4

[alsa-devel] [RFC PATCH] ALSA: hda - reset hda link during system/runtime suspend
by mengdong.lin＠intel.com 24 Jun '13

24 Jun '13

From: Mengdong Lin <mengdong.lin(a)intel.com> If all the codecs report ClkStopOK (OK to stop bus clock) after being put to D3, this patch will reset the HDA link before the controller is put to D3. So the link will be in reset during system or runtime suspend, the bus clock stops and the codecs are in D3(ClkStop) state. This may help to reduce power consumption by dozens of mW on some peripheral hda codecs. 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 f089fa0..9f110c7 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1120,6 +1120,20 @@ static void azx_load_dsp_cleanup(struct hda_bus *bus, struct snd_dma_buffer *dmab); #endif +/* enter link reset */ +static void azx_reset_link(struct azx *chip) +{ + unsigned long timeout; + + /* reset controller */ + azx_writel(chip, GCTL, azx_readl(chip, GCTL) & ~ICH6_GCTL_RESET); + + timeout = jiffies + msecs_to_jiffies(100); + while ((azx_readb(chip, GCTL) & ICH6_GCTL_RESET) && + time_before(jiffies, timeout)) + usleep_range(500, 1000); +} + /* reset codec link */ static int azx_reset(struct azx *chip, int full_reset) { @@ -2894,6 +2908,7 @@ static int azx_suspend(struct device *dev) if (chip->initialized) snd_hda_suspend(chip->bus); azx_stop_chip(chip); + azx_reset_link(chip); if (chip->irq >= 0) { free_irq(chip->irq, chip); chip->irq = -1; @@ -2946,6 +2961,7 @@ static int azx_runtime_suspend(struct device *dev) struct azx *chip = card->private_data; azx_stop_chip(chip); + azx_reset_link(chip); azx_clear_irq_pending(chip); return 0; } -- 1.7.10.4

3 2

Re: [alsa-devel] [PATCH - oss plugin update 2/2] oss: Add user variables to set period and buffer config
by Takashi Iwai 24 Jun '13

24 Jun '13

At Sun, 23 Jun 2013 01:25:19 +0100, alex.d.wiggins(a)gmail.com wrote: > > From: Alex Wiggins <alex.d.wiggins(a)gmail.com> > > Added 4 user variables to control the OSS period and buffer config: > > - period_size_allowany (boolean): > if not set then queries OSS for period size > (by using SNDCTL_DSP_GETBLKSIZE), > if set to true then allows any power of 2 period size; > I'd suggest to put an underscore between "allow" and "any". Or, instead of adding a flag... > - period_size (integer): > sets the OSS period size (SND_PCM_IOPLUG_HW_PERIOD_BYTES); ... handle allowany when period_size < 0, for example. Takashi > - periods (integer): > sets the number of OSS periods (SND_PCM_IOPLUG_HW_PERIODS); > > - buffer_size (integer): > sets the OSS buffer size (SND_PCM_IOPLUG_HW_BUFFER_BYTES). > > Note that ALSA performs sanity checks to prevent conflicts. > > Signed-off-by: Alex Wiggins <alex.d.wiggins(a)gmail.com> > > diff --git a/oss/pcm_oss.c b/oss/pcm_oss.c > index 6d8dea4..7131e54 100644 > --- a/oss/pcm_oss.c > +++ b/oss/pcm_oss.c > @@ -60,6 +60,13 @@ typedef struct snd_pcm_oss { > unsigned int period_shift; > unsigned int periods; > unsigned int frame_bytes; > + /* user variables set using ALSA config files */ > + /* included here as this snd_pcm_oss_t struct > + is stored in io->private_data */ > + int user_period_size_allowany; > + long int user_period_size; > + long int user_periods; > + long int user_buffer_size; > } snd_pcm_oss_t; > > static snd_pcm_sframes_t oss_write(snd_pcm_ioplug_t *io, > @@ -406,6 +413,7 @@ static int oss_hw_constraint(snd_pcm_oss_t *oss) > unsigned int nchannels; > unsigned int channel[6]; > /* period and buffer bytes must be power of two */ > + /* minimum: 2^8 = 256; maximum: 2^23 = 8388608 */ > static const unsigned int bytes_list[] = { > 1U<<8, 1U<<9, 1U<<10, 1U<<11, 1U<<12, 1U<<13, 1U<<14, 1U<<15, > 1U<<16, 1U<<17, 1U<<18, 1U<<19, 1U<<20, 1U<<21, 1U<<22, 1U<<23 > @@ -498,20 +506,96 @@ static int oss_hw_constraint(snd_pcm_oss_t *oss) > if (err < 0) > return err; > > - /* period size (in power of two) */ > - err = snd_pcm_ioplug_set_param_list(io, SND_PCM_IOPLUG_HW_PERIOD_BYTES, > - ARRAY_SIZE(bytes_list), bytes_list); > - if (err < 0) > - return err; > - /* periods */ > - err = snd_pcm_ioplug_set_param_minmax(io, SND_PCM_IOPLUG_HW_PERIODS, 2, 1024); > - if (err < 0) > - return err; > - /* buffer size (in power of two) */ > - err = snd_pcm_ioplug_set_param_list(io, SND_PCM_IOPLUG_HW_BUFFER_BYTES, > - ARRAY_SIZE(bytes_list), bytes_list); > - if (err < 0) > - return err; > + /* period and buffer sizes: default behaviour below is to > + query period size from OSS */ > + > + /* consider user set paramaters > + (period_size_allowany; period_size; periods; buffer_size) */ > + /* note: ALSA performs sanity checks > + to prevent conflicting parameters */ > + > + /* period_size_allowany and period_size */ > + if (oss->user_period_size_allowany) { > + /* period_size_allowany flag set: > + allow any power of 2 period size */ > + err = snd_pcm_ioplug_set_param_list(io, > + SND_PCM_IOPLUG_HW_PERIOD_BYTES, > + ARRAY_SIZE(bytes_list), bytes_list); > + if (err < 0) > + return err; > + } else { > + if (oss->user_period_size > 0) { > + /* period_size set to positive value: > + use user set period size */ > + const unsigned int user_period_size > + = oss->user_period_size; > + err = snd_pcm_ioplug_set_param_list(io, > + SND_PCM_IOPLUG_HW_PERIOD_BYTES, 1, > + &user_period_size); > + } else { > + /* query OSS for period size (called fragment size) */ > + unsigned int fragsize; > + if (ioctl(oss->fd, SNDCTL_DSP_GETBLKSIZE, &fragsize) > + < 0) > + fprintf(stderr, > +"*** alsa-plugins-oss: error querying SNDCTL_DSP_GETBLKSIZE.\n"); > + > + /* note: second query of SNDCTL_DSP_GETBLKSIZE > + can give different result > + (i.e. 4096 instead of 2048) */ > + /* in oss_audio_core.c getblksize function, > + 4096 is the default return value: > + so second time probably not reliable */ > + > + /* period size (in power of two) */ > + unsigned int x = fragsize; > + /* check if fragsize is a power of two */ > + if ((x != 0) && ((x & (~x + 1)) == x)) > + err = snd_pcm_ioplug_set_param_list(io, > + SND_PCM_IOPLUG_HW_PERIOD_BYTES, > + 1, &fragsize); > + else > + err = snd_pcm_ioplug_set_param_list(io, > + SND_PCM_IOPLUG_HW_PERIOD_BYTES, > + ARRAY_SIZE(bytes_list), bytes_list); > + if (err < 0) > + return err; > + } > + } > + > + /* periods (number of periods per buffer) */ > + if (oss->user_periods > 0) { > + /* force number of periods to user set periods */ > + const unsigned int user_periods = oss->user_periods; > + err = snd_pcm_ioplug_set_param_list(io, > + SND_PCM_IOPLUG_HW_PERIODS, 1, &user_periods); > + if (err < 0) > + return err; > + } else { > + /* allow any number of periods between 2 and 1024 inclusive > + (ALSA can choose) */ > + err = snd_pcm_ioplug_set_param_minmax(io, > + SND_PCM_IOPLUG_HW_PERIODS, 2, 1024); > + if (err < 0) > + return err; > + } > + > + /* buffer_size */ > + if (oss->user_buffer_size > 0) { > + /* force buffer size to user set buffer_size */ > + const unsigned int user_buffer_size = oss->user_buffer_size; > + err = snd_pcm_ioplug_set_param_list(io, > + SND_PCM_IOPLUG_HW_BUFFER_BYTES, 1, &user_buffer_size); > + if (err < 0) > + return err; > + } else { > + /* buffer size (in power of two) */ > + err = snd_pcm_ioplug_set_param_list(io, > + SND_PCM_IOPLUG_HW_BUFFER_BYTES, > + ARRAY_SIZE(bytes_list), bytes_list); > + if (err < 0) > + return err; > + } > > return 0; > } > @@ -556,6 +640,11 @@ SND_PCM_PLUGIN_DEFINE_FUNC(oss) > const char *device = "/dev/dsp"; > int err; > snd_pcm_oss_t *oss; > + /* default values for user variables*/ > + int user_period_size_allowany = 0; > + long int user_period_size = 0; > + long int user_periods = 0; > + long int user_buffer_size = 0; > > snd_config_for_each(i, next, conf) { > snd_config_t *n = snd_config_iterator_entry(i); > @@ -571,6 +660,39 @@ SND_PCM_PLUGIN_DEFINE_FUNC(oss) > } > continue; > } > + > + /* user variables */ > + if (strcmp(id, "period_size_allowany") == 0) { > + if (snd_config_get_bool(n) < 0) { > + SNDERR("Invalid type for %s", id); > + return -EINVAL; > + } else > + user_period_size_allowany = > + snd_config_get_bool(n); > + continue; > + } > + if (strcmp(id, "period_size") == 0) { > + if (snd_config_get_integer(n, &user_period_size) < 0) { > + SNDERR("Invalid type for %s", id); > + return -EINVAL; > + } > + continue; > + } > + if (strcmp(id, "periods") == 0) { > + if (snd_config_get_integer(n, &user_periods) < 0) { > + SNDERR("Invalid type for %s", id); > + return -EINVAL; > + } > + continue; > + } > + if (strcmp(id, "buffer_size") == 0) { > + if (snd_config_get_integer(n, &user_buffer_size) < 0) { > + SNDERR("Invalid type for %s", id); > + return -EINVAL; > + } > + continue; > + } > + > SNDERR("Unknown field %s", id); > return -EINVAL; > } > @@ -595,6 +717,12 @@ SND_PCM_PLUGIN_DEFINE_FUNC(oss) > goto error; > } > > + /* save user variables from ALSA config files */ > + oss->user_period_size_allowany = user_period_size_allowany; > + oss->user_period_size = user_period_size; > + oss->user_periods = user_periods; > + oss->user_buffer_size = user_buffer_size; > + > oss->io.version = SND_PCM_IOPLUG_VERSION; > oss->io.name = "ALSA <-> OSS PCM I/O Plugin"; > oss->io.poll_fd = oss->fd; > -- > 1.8.3.1 >

1 0

Re: [alsa-devel] [PATCH - oss plugin update 1/2] oss: Add support of 32-bit and other formats
by Takashi Iwai 24 Jun '13

24 Jun '13

At Sun, 23 Jun 2013 01:25:18 +0100, alex.d.wiggins(a)gmail.com wrote: > > From: Alex Wiggins <alex.d.wiggins(a)gmail.com> > > Added support of 32-bit and other audio formats via 3 changes: > > - Additional audio format definitions > at the top of the file (if required); > > - Additional audio formats in oss_hw_params > (to allow as input to pass to OSS); > > - Additional audio formats in oss_hw_constraint > (to allow to report to ALSA). > > Signed-off-by: Alex Wiggins <alex.d.wiggins(a)gmail.com> > > diff --git a/oss/pcm_oss.c b/oss/pcm_oss.c > index d43b44c..6d8dea4 100644 > --- a/oss/pcm_oss.c > +++ b/oss/pcm_oss.c > @@ -24,6 +24,32 @@ > #include <alsa/pcm_external.h> > #include <linux/soundcard.h> > > +/* definitions for additional OSS formats if required */ > +#ifndef AFMT_S32_LE > +#define AFMT_S32_LE 0x00001000 > +#endif > +#ifndef AFMT_S32_BE > +#define AFMT_S32_BE 0x00002000 > +#endif > +#ifndef AFMT_S24_LE > +#define AFMT_S24_LE 0x00008000 > +#endif > +#ifndef AFMT_S24_BE > +#define AFMT_S24_BE 0x00010000 > +#endif > +#ifndef AFMT_S24_PACKED > +#define AFMT_S24_PACKED 0x00040000 > +#endif > +#ifndef AFMT_FLOAT > +#define AFMT_FLOAT 0x00004000 > +#endif > +#ifndef AFMT_SPDIF_RAW > +#define AFMT_SPDIF_RAW 0x00020000 > +#endif > +#ifndef AFMT_VORBIS > +#define AFMT_VORBIS 0x00000800 > +#endif > + > typedef struct snd_pcm_oss { > snd_pcm_ioplug_t io; > char *device; > @@ -165,6 +191,143 @@ static int oss_hw_params(snd_pcm_ioplug_t *io, > case SND_PCM_FORMAT_S16_BE: > oss->format = AFMT_S16_BE; > break; > + /* allow additional data formats (such as 32 bit) as input > + to pass to OSS (set in OSS driver) */ > + case SND_PCM_FORMAT_S32_LE: > + oss->format = AFMT_S32_LE; > + break; > + case SND_PCM_FORMAT_S32_BE: > + oss->format = AFMT_S32_BE; > + break; > + case SND_PCM_FORMAT_S8: > + oss->format = AFMT_S8; > + break; > + case SND_PCM_FORMAT_U16_LE: > + oss->format = AFMT_U16_LE; > + break; > + case SND_PCM_FORMAT_U16_BE: > + oss->format = AFMT_U16_BE; > + break; > + case SND_PCM_FORMAT_S24_LE: > + oss->format = AFMT_S24_LE; > + break; > + case SND_PCM_FORMAT_S24_BE: > + oss->format = AFMT_S24_BE; > + break; > + case SND_PCM_FORMAT_MU_LAW: > + oss->format = AFMT_MU_LAW; > + break; > + case SND_PCM_FORMAT_A_LAW: > + oss->format = AFMT_A_LAW; > + break; > + case SND_PCM_FORMAT_IMA_ADPCM: > + oss->format = AFMT_IMA_ADPCM; > + break; > + case SND_PCM_FORMAT_MPEG: > + oss->format = AFMT_MPEG; > + break; > + case SND_PCM_FORMAT_S24_3LE: > + oss->format = AFMT_S24_PACKED; > + break; > + case SND_PCM_FORMAT_FLOAT: > + oss->format = AFMT_FLOAT; > + break; > + case SND_PCM_FORMAT_IEC958_SUBFRAME: > + oss->format = AFMT_SPDIF_RAW; > + break; > + /* following formats need defining checking native endian: > + case SND_PCM_FORMAT_S16: > + oss->format = AFMT_S16_NE; > + break; > + case SND_PCM_FORMAT_U16: > + oss->format = AFMT_U16_NE; > + break; > + case SND_PCM_FORMAT_S24: > + oss->format = AFMT_S24_NE; > + break; > + case SND_PCM_FORMAT_S32: > + oss->format = AFMT_S32_NE; > + */ Use #if/#endif instead of multi-line comments. > + /* following formats found in ALSA but not in OSS: > + case SND_PCM_FORMAT_U24_LE: > + oss->format = AFMT_U24_LE; > + break; > + case SND_PCM_FORMAT_U24_BE: > + oss->format = AFMT_U24_BE; > + break; > + case SND_PCM_FORMAT_U32_LE: > + oss->format = AFMT_U32_LE; > + break; > + case SND_PCM_FORMAT_U32_BE: > + oss->format = AFMT_U32_BE; > + break; > + case SND_PCM_FORMAT_FLOAT_LE: > + oss->format = AFMT_FLOAT_LE; > + break; > + case SND_PCM_FORMAT_FLOAT_BE: > + oss->format = AFMT_FLOAT_BE; > + break; > + case SND_PCM_FORMAT_FLOAT64_LE: > + oss->format = AFMT_FLOAT64_LE; > + break; > + case SND_PCM_FORMAT_FLOAT64_BE: > + oss->format = AFMT_FLOAT64_BE; > + break; > + case SND_PCM_FORMAT_IEC958_SUBFRAME_LE: > + oss->format = AFMT_SPDIF_RAW_LE; > + break; > + case SND_PCM_FORMAT_IEC958_SUBFRAME_BE: > + oss->format = AFMT_SPDIF_RAW_BE; > + break; > + case SND_PCM_FORMAT_GSM: > + oss->format = AFMT_GSM; > + break; > + case SND_PCM_FORMAT_SPECIAL: > + oss->format = AFMT_SPECIAL; > + break; > + case SND_PCM_FORMAT_S24_3BE: > + oss->format = AFMT_S24_PACKED_BE; > + break; > + case SND_PCM_FORMAT_U24_3LE: > + oss->format = AFMT_U24_PACKED; > + break; > + case SND_PCM_FORMAT_U24_3BE: > + oss->format = AFMT_U24_PACKED_BE; > + break; > + case SND_PCM_FORMAT_S20_3LE: > + oss->format = ; > + break; > + case SND_PCM_FORMAT_S20_3BE: > + oss->format = ; > + break; > + case SND_PCM_FORMAT_U20_3LE: > + oss->format = ; > + break; > + case SND_PCM_FORMAT_U20_3BE: > + oss->format = ; > + break; > + case SND_PCM_FORMAT_S18_3LE: > + oss->format = ; > + break; > + case SND_PCM_FORMAT_S18_3BE: > + oss->format = ; > + break; > + case SND_PCM_FORMAT_U18_3LE: > + oss->format = ; > + break; > + case SND_PCM_FORMAT_U18_3BE: > + oss->format = ; > + break; > + case SND_PCM_FORMAT_U24: > + oss->format = AFMT_U24_NE; > + break; > + case SND_PCM_FORMAT_U32: > + oss->format = AFMT_U32_NE; > + break; > + case SND_PCM_FORMAT_FLOAT64: > + oss->format = AFMT_FLOAT64; > + break; > + */ Ditto. > default: > fprintf(stderr, "*** OSS: unsupported format %s\n", snd_pcm_format_name(io->format)); > return -EINVAL; > @@ -273,6 +436,40 @@ static int oss_hw_constraint(snd_pcm_oss_t *oss) > format[nformats++] = SND_PCM_FORMAT_S16_BE; > if (tmp & AFMT_MU_LAW) > format[nformats++] = SND_PCM_FORMAT_MU_LAW; > + /* allow additional data formats (such as 32 bit) > + to be reported from OSS to ALSA below */ > + if (tmp & AFMT_S32_LE) > + format[nformats++] = SND_PCM_FORMAT_S32_LE; > + if (tmp & AFMT_S32_BE) > + format[nformats++] = SND_PCM_FORMAT_S32_BE; > + if (tmp & AFMT_A_LAW) > + format[nformats++] = SND_PCM_FORMAT_A_LAW; > + if (tmp & AFMT_IMA_ADPCM) > + format[nformats++] = SND_PCM_FORMAT_IMA_ADPCM; > + if (tmp & AFMT_S8) > + format[nformats++] = SND_PCM_FORMAT_S8; > + if (tmp & AFMT_U16_LE) > + format[nformats++] = SND_PCM_FORMAT_U16_LE; > + if (tmp & AFMT_U16_BE) > + format[nformats++] = SND_PCM_FORMAT_U16_BE ; > + if (tmp & AFMT_MPEG) > + format[nformats++] = SND_PCM_FORMAT_MPEG; > + if (tmp & AFMT_FLOAT) > + format[nformats++] = SND_PCM_FORMAT_FLOAT; > + if (tmp & AFMT_S24_LE) > + format[nformats++] = SND_PCM_FORMAT_S24_LE; > + if (tmp & AFMT_S24_BE) > + format[nformats++] = SND_PCM_FORMAT_S24_BE; > + if (tmp & AFMT_SPDIF_RAW) > + format[nformats++] = SND_PCM_FORMAT_IEC958_SUBFRAME; > + if (tmp & AFMT_S24_PACKED) > + format[nformats++] = SND_PCM_FORMAT_S24_3LE; > + /* following formats found in OSS but not in ALSA */ > + if (tmp & AFMT_AC3) > + format[nformats++] = SND_PCM_FORMAT_UNKNOWN; > + if (tmp & AFMT_VORBIS) > + format[nformats++] = SND_PCM_FORMAT_UNKNOWN; > + /* assume signed 16 bit (native CPU endian) if not found above */ > if (! nformats) > format[nformats++] = SND_PCM_FORMAT_S16; > if ((err = snd_pcm_ioplug_set_param_list(io, SND_PCM_IOPLUG_HW_FORMAT, You need to adjust format[] array size if you add more possible formats. thanks, Takashi

1 0

[alsa-devel] [PATCH] Add M2Tech hiFace USB-SPDIF driver
by Antonio Ospite 24 Jun '13

24 Jun '13

Add driver for M2Tech hiFace USB-SPDIF interface and compatible devices. The supported products are: * M2Tech Young * M2Tech hiFace * M2Tech North Star * M2Tech W4S Young * M2Tech Corrson * M2Tech AUDIA * M2Tech SL Audio * M2Tech Empirical * M2Tech Rockna * M2Tech Pathos * M2Tech Metronome * M2Tech CAD * M2Tech Audio Esclusive * M2Tech Rotel * M2Tech Eeaudio * The Chord Company CHORD * AVA Group A/S Vitus Signed-off-by: Antonio Ospite <ao2(a)amarulasolutions.com> --- Hi, this is my first ALSA driver I still have some doubts and I marked them in the comments below with a XXX prefix, could someone please take a look at them? I plan on submitting a version 2 of the driver after I get some feedback, especially on those doubts. I also noticed that many drivers under sound/usb/ have text in the Kconfig entry stating "Say Y here to include support for ..." but I only see "[N/m/?]" as options when I run "make oldconfig" maybe because of some dependency, I am not sure, should such text be removed in order to avoid confusion? Thanks, Antonio sound/usb/Kconfig | 33 +++ sound/usb/Makefile | 2 +- sound/usb/hiface/Makefile | 2 + sound/usb/hiface/chip.c | 331 ++++++++++++++++++++++ sound/usb/hiface/chip.h | 34 +++ sound/usb/hiface/pcm.c | 666 +++++++++++++++++++++++++++++++++++++++++++++ sound/usb/hiface/pcm.h | 24 ++ 7 files changed, 1091 insertions(+), 1 deletion(-) create mode 100644 sound/usb/hiface/Makefile create mode 100644 sound/usb/hiface/chip.c create mode 100644 sound/usb/hiface/chip.h create mode 100644 sound/usb/hiface/pcm.c create mode 100644 sound/usb/hiface/pcm.h diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index 225dfd7..2b206e8 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -115,5 +115,38 @@ config SND_USB_6FIRE and further help can be found at http://sixfireusb.sourceforge.net +config SND_USB_HIFACE + tristate "M2Tech hiFace USB-SPDIF driver" + select SND_PCM + help + Say Y here to include support for M2Tech hiFace USB-SPDIF interface. + + M2Tech hiFace and compatible devices offer a Hi-End S/PDIF Output + Interface, see http://www.m2tech.biz/hiface.html + + This driver supports the original M2Tech hiFace and some other + compatible devices. The supported products are: + + * M2Tech Young + * M2Tech hiFace + * M2Tech North Star + * M2Tech W4S Young + * M2Tech Corrson + * M2Tech AUDIA + * M2Tech SL Audio + * M2Tech Empirical + * M2Tech Rockna + * M2Tech Pathos + * M2Tech Metronome + * M2Tech CAD + * M2Tech Audio Esclusive + * M2Tech Rotel + * M2Tech Eeaudio + * The Chord Company CHORD + * AVA Group A/S Vitus + + To compile this driver as a module, choose M here: the module + will be called snd-usb-hiface. + endif # SND_USB diff --git a/sound/usb/Makefile b/sound/usb/Makefile index ac256dc..abe668f 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -23,4 +23,4 @@ obj-$(CONFIG_SND_USB_UA101) += snd-usbmidi-lib.o obj-$(CONFIG_SND_USB_USX2Y) += snd-usbmidi-lib.o obj-$(CONFIG_SND_USB_US122L) += snd-usbmidi-lib.o -obj-$(CONFIG_SND) += misc/ usx2y/ caiaq/ 6fire/ +obj-$(CONFIG_SND) += misc/ usx2y/ caiaq/ 6fire/ hiface/ diff --git a/sound/usb/hiface/Makefile b/sound/usb/hiface/Makefile new file mode 100644 index 0000000..463b136 --- /dev/null +++ b/sound/usb/hiface/Makefile @@ -0,0 +1,2 @@ +snd-usb-hiface-objs := chip.o pcm.o +obj-$(CONFIG_SND_USB_HIFACE) += snd-usb-hiface.o diff --git a/sound/usb/hiface/chip.c b/sound/usb/hiface/chip.c new file mode 100644 index 0000000..dce74e9 --- /dev/null +++ b/sound/usb/hiface/chip.c @@ -0,0 +1,331 @@ +/* + * Linux driver for M2Tech HiFace compatible devices + * + * Copyright 2012-2013 (C) M2TECH S.r.l and Amarula Solutions B.V. + * + * Authors: Michael Trimarchi <michael(a)amarulasolutions.com> + * Antonio Ospite <ao2(a)amarulasolutions.com> + * + * The driver is based on the work done in TerraTec DMX 6Fire USB + * + * 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. + */ + +#include <linux/module.h> +#include <linux/slab.h> +#include <sound/initval.h> + +#include "chip.h" +#include "pcm.h" + +MODULE_AUTHOR("Michael Trimarchi <michael(a)amarulasolutions.com>"); +MODULE_AUTHOR("Antonio Ospite <ao2(a)amarulasolutions.com>"); +MODULE_DESCRIPTION("M2Tech HiFace USB audio driver"); +MODULE_LICENSE("GPL v2"); +MODULE_SUPPORTED_DEVICE("{{HiFace, Evo}}"); + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-max */ +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* Id for card */ +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ + +#define DRIVER_NAME "snd-usb-hiface" +#define CARD_NAME "HiFace" + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for " CARD_NAME " soundcard."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard."); + +static struct hiface_chip *chips[SNDRV_CARDS] = SNDRV_DEFAULT_PTR; + +static DEFINE_MUTEX(register_mutex); + +struct hiface_vendor_quirk { + const char *device_name; + u8 extra_freq; +}; + +static int hiface_dev_free(struct snd_device *device) +{ + struct hiface_chip *chip = device->device_data; + kfree(chip); + return 0; +} + +static int hiface_chip_create(struct usb_device *device, int idx, + const struct hiface_vendor_quirk *quirk, + struct hiface_chip **rchip) +{ + struct snd_card *card = NULL; + struct hiface_chip *chip; + int ret; + static struct snd_device_ops ops = { + .dev_free = hiface_dev_free, + }; + + *rchip = NULL; + + /* if we are here, card can be registered in alsa. */ + ret = snd_card_create(index[idx], id[idx], THIS_MODULE, 0, &card); + if (ret < 0) { + snd_printk(KERN_ERR "cannot create alsa card.\n"); + return ret; + } + + strcpy(card->driver, DRIVER_NAME); + + if (quirk && quirk->device_name) + strcpy(card->shortname, quirk->device_name); + else + strcpy(card->shortname, "M2Tech generic audio"); + + sprintf(card->longname, "%s at %d:%d", card->shortname, + device->bus->busnum, device->devnum); + + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + if (!chip) { + snd_card_free(card); + return -ENOMEM; + } + + chip->dev = device; + chip->index = idx; + chip->card = card; + + ret = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (ret < 0) { + kfree(chip); + snd_card_free(card); + return ret; + } + + *rchip = chip; + return 0; +} + +static int hiface_chip_probe(struct usb_interface *intf, + const struct usb_device_id *usb_id) +{ + const struct hiface_vendor_quirk *quirk = (struct hiface_vendor_quirk *)usb_id->driver_info; + int ret; + int i; + struct hiface_chip *chip; + struct usb_device *device = interface_to_usbdev(intf); + + pr_info("Probe " DRIVER_NAME " driver.\n"); + + ret = usb_set_interface(device, 0, 0); + if (ret != 0) { + snd_printk(KERN_ERR "can't set first interface for " CARD_NAME " device.\n"); + return -EIO; + } + + /* check whether the card is already registered */ + chip = NULL; + mutex_lock(&register_mutex); + for (i = 0; i < SNDRV_CARDS; i++) { + if (chips[i] && chips[i]->dev == device) { + if (chips[i]->shutdown) { + snd_printk(KERN_ERR CARD_NAME " device is in the shutdown state, cannot create a card instance\n"); + ret = -ENODEV; + goto err; + } + chip = chips[i]; + break; + } + } + if (!chip) { + /* it's a fresh one. + * now look for an empty slot and create a new card instance + */ + for (i = 0; i < SNDRV_CARDS; i++) + if (enable[i] && !chips[i]) { + ret = hiface_chip_create(device, i, quirk, + &chip); + if (ret < 0) + goto err; + + snd_card_set_dev(chip->card, &intf->dev); + break; + } + if (!chip) { + snd_printk(KERN_ERR "no available " CARD_NAME " audio device\n"); + ret = -ENODEV; + goto err; + } + } + + ret = hiface_pcm_init(chip, quirk ? quirk->extra_freq : 0); + if (ret < 0) + goto err_chip_destroy; + + ret = snd_card_register(chip->card); + if (ret < 0) { + snd_printk(KERN_ERR "cannot register " CARD_NAME " card\n"); + goto err_chip_destroy; + } + + chips[chip->index] = chip; + chip->intf_count++; + + mutex_unlock(&register_mutex); + + usb_set_intfdata(intf, chip); + return 0; + +err_chip_destroy: + snd_card_free(chip->card); +err: + mutex_unlock(&register_mutex); + return ret; +} + +static void hiface_chip_disconnect(struct usb_interface *intf) +{ + struct hiface_chip *chip; + struct snd_card *card; + + pr_debug("%s: called.\n", __func__); + + chip = usb_get_intfdata(intf); + if (!chip) + return; + + card = chip->card; + chip->intf_count--; + if (chip->intf_count <= 0) { + /* Make sure that the userspace cannot create new request */ + snd_card_disconnect(card); + + mutex_lock(&register_mutex); + chips[chip->index] = NULL; + mutex_unlock(&register_mutex); + + chip->shutdown = true; + hiface_pcm_abort(chip); + snd_card_free_when_closed(card); + } +} + +static struct usb_device_id device_table[] = { + { + USB_DEVICE(0x04b4, 0x0384), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Young", + .extra_freq = 1, + } + }, + { + USB_DEVICE(0x04b4, 0x930b), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "hiFace", + } + }, + { + USB_DEVICE(0x04b4, 0x931b), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "North Star", + } + }, + { + USB_DEVICE(0x04b4, 0x931c), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "W4S Young", + } + }, + { + USB_DEVICE(0x04b4, 0x931d), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Corrson", + } + }, + { + USB_DEVICE(0x04b4, 0x931e), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "AUDIA", + } + }, + { + USB_DEVICE(0x04b4, 0x931f), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "SL Audio", + } + }, + { + USB_DEVICE(0x04b4, 0x9320), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Empirical", + } + }, + { + USB_DEVICE(0x04b4, 0x9321), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Rockna", + } + }, + { + USB_DEVICE(0x249c, 0x9001), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Pathos", + } + }, + { + USB_DEVICE(0x249c, 0x9002), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Metronome", + } + }, + { + USB_DEVICE(0x249c, 0x9006), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "CAD", + } + }, + { + USB_DEVICE(0x249c, 0x9008), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Audio Esclusive", + } + }, + { + USB_DEVICE(0x249c, 0x931c), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Rotel", + } + }, + { + USB_DEVICE(0x249c, 0x932c), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Eeaudio", + } + }, + { + USB_DEVICE(0x245f, 0x931c), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "CHORD", + } + }, + { + USB_DEVICE(0x25c6, 0x9002), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Vitus", + } + }, + {} +}; + +MODULE_DEVICE_TABLE(usb, device_table); + +static struct usb_driver hiface_usb_driver = { + .name = DRIVER_NAME, + .probe = hiface_chip_probe, + .disconnect = hiface_chip_disconnect, + .id_table = device_table, +}; + +module_usb_driver(hiface_usb_driver); diff --git a/sound/usb/hiface/chip.h b/sound/usb/hiface/chip.h new file mode 100644 index 0000000..c2e9fda --- /dev/null +++ b/sound/usb/hiface/chip.h @@ -0,0 +1,34 @@ +/* + * Linux driver for M2Tech HiFace compatible devices + * + * Copyright 2012-2013 (C) M2TECH S.r.l and Amarula Solutions B.V. + * + * Authors: Michael Trimarchi <michael(a)amarulasolutions.com> + * Antonio Ospite <ao2(a)amarulasolutions.com> + * + * The driver is based on the work done in TerraTec DMX 6Fire USB + * + * 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. + */ + +#ifndef HIFACE_CHIP_H +#define HIFACE_CHIP_H + +#include <linux/usb.h> +#include <sound/core.h> + +struct pcm_runtime; + +struct hiface_chip { + struct usb_device *dev; + struct snd_card *card; + int intf_count; /* number of registered interfaces */ + int index; /* index in module parameter arrays */ + bool shutdown; + + struct pcm_runtime *pcm; +}; +#endif /* HIFACE_CHIP_H */ diff --git a/sound/usb/hiface/pcm.c b/sound/usb/hiface/pcm.c new file mode 100644 index 0000000..34fe0ed --- /dev/null +++ b/sound/usb/hiface/pcm.c @@ -0,0 +1,666 @@ +/* + * Linux driver for M2Tech HiFace compatible devices + * + * Copyright 2012-2013 (C) M2TECH S.r.l and Amarula Solutions B.V. + * + * Authors: Michael Trimarchi <michael(a)amarulasolutions.com> + * Antonio Ospite <ao2(a)amarulasolutions.com> + * + * The driver is based on the work done in TerraTec DMX 6Fire USB + * + * 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. + */ + +#include <linux/slab.h> +#include <sound/pcm.h> + +#include "pcm.h" +#include "chip.h" + +#define OUT_EP 0x2 +#define PCM_N_URBS 8 +#define PCM_MAX_PACKET_SIZE 4096 +#define MAX_BUFSIZE (2 * PCM_N_URBS * PCM_MAX_PACKET_SIZE) + +struct pcm_urb { + struct hiface_chip *chip; + + struct urb instance; + struct usb_anchor submitted; + u8 *buffer; +}; + +struct pcm_substream { + spinlock_t lock; + struct snd_pcm_substream *instance; + + bool active; + snd_pcm_uframes_t dma_off; /* current position in alsa dma_area */ + snd_pcm_uframes_t period_off; /* current position in current period */ +}; + +enum { /* pcm streaming states */ + STREAM_DISABLED, /* no pcm streaming */ + STREAM_STARTING, /* pcm streaming requested, waiting to become ready */ + STREAM_RUNNING, /* pcm streaming running */ + STREAM_STOPPING +}; + +struct pcm_runtime { + struct hiface_chip *chip; + struct snd_pcm *instance; + + struct pcm_substream playback; + bool panic; /* if set driver won't do anymore pcm on device */ + + struct pcm_urb out_urbs[PCM_N_URBS]; + + struct mutex stream_mutex; + u8 stream_state; /* one of STREAM_XXX */ + u8 rate; /* one of PCM_RATE_XXX */ + u8 extra_freq; + wait_queue_head_t stream_wait_queue; + bool stream_wait_cond; +}; + +static const unsigned int rates[] = { 44100, 48000, 88200, 96000, 176400, 192000, + 352800, 384000 }; +static struct snd_pcm_hw_constraint_list constraints_rates = { + .count = ARRAY_SIZE(rates) - 2, /* by default rates up to 192000 are supported */ + .list = rates, + .mask = 0, +}; + +static const int rates_alsaid[] = { + SNDRV_PCM_RATE_44100, SNDRV_PCM_RATE_48000, + SNDRV_PCM_RATE_88200, SNDRV_PCM_RATE_96000, + SNDRV_PCM_RATE_176400, SNDRV_PCM_RATE_192000, + SNDRV_PCM_RATE_KNOT, SNDRV_PCM_RATE_KNOT }; + +static const struct snd_pcm_hardware pcm_hw = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BATCH, + + .formats = SNDRV_PCM_FMTBIT_S32_LE, + + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_176400 | + SNDRV_PCM_RATE_192000 | + SNDRV_PCM_RATE_KNOT, + + .rate_min = 44100, + .rate_max = 192000, /* changes in hiface_pcm_open to support extra rates */ + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = MAX_BUFSIZE, + .period_bytes_min = PCM_MAX_PACKET_SIZE, + .period_bytes_max = MAX_BUFSIZE, + .periods_min = 2, + .periods_max = 1024 +}; + +static int hiface_pcm_set_rate(struct pcm_runtime *rt, unsigned int rate) +{ + u8 rate_value[] = { 0x43, 0x4b, 0x42, 0x4a, 0x40, 0x48, 0x58, 0x68 }; + int ret; + struct usb_device *device = rt->chip->dev; + + /* We are already sure that the rate is supported here thanks to the + * contraints set in hiface_pcm_open(), but we have to retrieve the + * index to access rate_value. + */ + for (rt->rate = 0; rt->rate < ARRAY_SIZE(rates); rt->rate++) + if (rate == rates[rt->rate]) + break; + + if (unlikely(rt->rate == ARRAY_SIZE(rates))) { + pr_err("Unsupported rate %d\n", rate); + return -EINVAL; + } + + /* + * USBIO: Vendor 0xb0(wValue=0x0043, wIndex=0x0000) + * 43 b0 43 00 00 00 00 00 + * USBIO: Vendor 0xb0(wValue=0x004b, wIndex=0x0000) + * 43 b0 4b 00 00 00 00 00 + * This control message doesn't have any ack from the + * other side + */ + ret = usb_control_msg(device, usb_sndctrlpipe(device, 0), + 0xb0, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, + rate_value[rt->rate], 0, NULL, 0, 100); + if (ret < 0) { + snd_printk(KERN_ERR "Error setting samplerate %d.\n", + rates[rt->rate]); + return ret; + } + + return 0; +} + +static struct pcm_substream *hiface_pcm_get_substream( + struct snd_pcm_substream *alsa_sub) +{ + struct pcm_runtime *rt = snd_pcm_substream_chip(alsa_sub); + + if (alsa_sub->stream == SNDRV_PCM_STREAM_PLAYBACK) + return &rt->playback; + + pr_debug("Error getting pcm substream slot.\n"); + return NULL; +} + +/* call with stream_mutex locked */ +static void hiface_pcm_stream_stop(struct pcm_runtime *rt) +{ + int i, time; + + if (rt->stream_state != STREAM_DISABLED) { + rt->stream_state = STREAM_STOPPING; + + for (i = 0; i < PCM_N_URBS; i++) { + time = usb_wait_anchor_empty_timeout( + &rt->out_urbs[i].submitted, 100); + if (!time) + usb_kill_anchored_urbs( + &rt->out_urbs[i].submitted); + usb_kill_urb(&rt->out_urbs[i].instance); + } + + rt->stream_state = STREAM_DISABLED; + } +} + +/* call with stream_mutex locked */ +static int hiface_pcm_stream_start(struct pcm_runtime *rt) +{ + int ret = 0, i; + + if (rt->stream_state == STREAM_DISABLED) { + /* submit our out urbs zero init */ + rt->stream_state = STREAM_STARTING; + for (i = 0; i < PCM_N_URBS; i++) { + memset(rt->out_urbs[i].buffer, 0, PCM_MAX_PACKET_SIZE); + usb_anchor_urb(&rt->out_urbs[i].instance, + &rt->out_urbs[i].submitted); + ret = usb_submit_urb(&rt->out_urbs[i].instance, + GFP_ATOMIC); + if (ret) { + hiface_pcm_stream_stop(rt); + return ret; + } + } + + /* wait for first out urb to return (sent in in urb handler) */ + wait_event_timeout(rt->stream_wait_queue, rt->stream_wait_cond, + HZ); + if (rt->stream_wait_cond) { + pr_debug("%s: Stream is running wakeup event\n", + __func__); + rt->stream_state = STREAM_RUNNING; + } else { + hiface_pcm_stream_stop(rt); + return -EIO; + } + } + return ret; +} + + +/* The hardware wants word-swapped 32-bit values */ +void memcpy_swahw32(u8 *dest, u8 *src, unsigned int n) +{ + unsigned int i; + + for (i = 0; i < n / 4; i++) + ((u32 *)dest)[i] = swahw32(((u32 *)src)[i]); +} + +/* call with substream locked */ +static int hiface_pcm_playback(struct pcm_substream *sub, + struct pcm_urb *urb) +{ + struct snd_pcm_runtime *alsa_rt = sub->instance->runtime; + u8 *source; + unsigned int pcm_buffer_size; + + /* XXX Can an invalid format make it to here? + * Isn't ALSA checking pcm_hw.formats ? + */ + if (alsa_rt->format != SNDRV_PCM_FORMAT_S32_LE) { + pr_err("Unsupported sample format\n"); + return -EINVAL; + } + + pcm_buffer_size = snd_pcm_lib_buffer_bytes(sub->instance); + + if (sub->dma_off + PCM_MAX_PACKET_SIZE <= pcm_buffer_size) { + pr_debug("%s: (1) buffer_size %#x dma_offset %#x\n", __func__, + (unsigned int) pcm_buffer_size, + (unsigned int) sub->dma_off); + + source = alsa_rt->dma_area + sub->dma_off; + memcpy_swahw32(urb->buffer, source, PCM_MAX_PACKET_SIZE); + } else { + /* wrap around at end of ring buffer */ + unsigned int len; + + pr_debug("%s: (2) buffer_size %#x dma_offset %#x\n", __func__, + (unsigned int) pcm_buffer_size, + (unsigned int) sub->dma_off); + + len = pcm_buffer_size - sub->dma_off; + + source = alsa_rt->dma_area + sub->dma_off; + memcpy_swahw32(urb->buffer, source, len); + + source = alsa_rt->dma_area; + memcpy_swahw32(urb->buffer + len, source, + PCM_MAX_PACKET_SIZE - len); + } + sub->dma_off += PCM_MAX_PACKET_SIZE; + if (sub->dma_off >= pcm_buffer_size) + sub->dma_off -= pcm_buffer_size; + + sub->period_off += PCM_MAX_PACKET_SIZE; + + return 0; +} + +static void hiface_pcm_out_urb_handler(struct urb *usb_urb) +{ + struct pcm_urb *out_urb = usb_urb->context; + struct pcm_runtime *rt = out_urb->chip->pcm; + struct pcm_substream *sub; + unsigned long flags; + + pr_debug("%s: called.\n", __func__); + + if (usb_urb->status || rt->panic || rt->stream_state == STREAM_STOPPING) + return; + + if (rt->stream_state == STREAM_STARTING) { + rt->stream_wait_cond = true; + wake_up(&rt->stream_wait_queue); + } + + /* now send our playback data (if a free out urb was found) */ + sub = &rt->playback; + spin_lock_irqsave(&sub->lock, flags); + if (sub->active) { + int ret; + + ret = hiface_pcm_playback(sub, out_urb); + if (ret < 0) { + spin_unlock_irqrestore(&sub->lock, flags); + goto out_fail; + } + if (sub->period_off >= sub->instance->runtime->period_size) { + sub->period_off %= sub->instance->runtime->period_size; + spin_unlock_irqrestore(&sub->lock, flags); + snd_pcm_period_elapsed(sub->instance); + } else { + spin_unlock_irqrestore(&sub->lock, flags); + } + } else { + memset(out_urb->buffer, 0, PCM_MAX_PACKET_SIZE); + spin_unlock_irqrestore(&sub->lock, flags); + } +out_fail: + usb_submit_urb(&out_urb->instance, GFP_ATOMIC); +} + +static int hiface_pcm_open(struct snd_pcm_substream *alsa_sub) +{ + struct pcm_runtime *rt = snd_pcm_substream_chip(alsa_sub); + struct pcm_substream *sub = NULL; + struct snd_pcm_runtime *alsa_rt = alsa_sub->runtime; + int ret; + + pr_debug("%s: called.\n", __func__); + + if (rt->panic) + return -EPIPE; + + mutex_lock(&rt->stream_mutex); + alsa_rt->hw = pcm_hw; + + if (alsa_sub->stream == SNDRV_PCM_STREAM_PLAYBACK) { + + /* XXX can we avoid setting hw.rates below? + * + * FWIU setting alsa_rt->hw.rates in the open callback can be + * useful in two cases: + * + * - different hardware revisions have different + * capabilities + * + * - different stream types must have the same capabilities + * + * but in our case it should not be needed as we only have one + * playback stream and we define the capabilities via + * constraints already to avoid invalid rates to be used. + * + * Moreover, following the code stolen from 6fire it seems + * impossible to get to setting alsa_rt->hw.rates here: + * + * hiface_pcm_init: + * rt->rate = ARRAY_SIZE(rates); + * + * hiface_pcm_open: + * if (rt->rate < ARRAY_SIZE(rates)) + * alsa_rt->hw.rates = rates_alsaid[rt->rate]; + * // So it does not get here on the first open + * + * hiface_pcm_prepare: + * hiface_pcm_set_rate: + * for (rt->rate = 0; rt->rate < ARRAY_SIZE(rates); rt->rate++) + * if (rate == rates[rt->rate]) + * break; + * + * hiface_pcm_close: + * rt->rate = ARRAY_SIZE(rates); + * // So it won't set alsa_rt->hw.rates at the next open either + * + * So the only way to set alsa_rt->hw.rates is to have two + * calls to the open callback, is that even possible? ALSA + * does not allow sharing PCM streams AFAIK and we use only + * one playback stream. + * + * Maybe the 6fire driver was setting alsa_rt->hw.rates in + * order to have the playback stream and the capture stream + * use the same rate? I.e. the first open,prepare,set_rate() + * sequence on one type of stream will decide the rate and the + * second open on the other type would restrict the supported + * rate to the one already in use, but I am not sure. + * + * If the rationale makes sense then we don't need to set + * alsa_rt->hw.rates and we can remove this and related code. + */ + if (rt->rate < ARRAY_SIZE(rates)) + alsa_rt->hw.rates = rates_alsaid[rt->rate]; + + sub = &rt->playback; + } + + if (!sub) { + mutex_unlock(&rt->stream_mutex); + pr_err("Invalid stream type\n"); + return -EINVAL; + } + + if (rt->extra_freq) { + alsa_rt->hw.rate_max = 384000; + constraints_rates.count = ARRAY_SIZE(rates); + } + + ret = snd_pcm_hw_constraint_list(alsa_sub->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + if (ret < 0) { + mutex_unlock(&rt->stream_mutex); + return ret; + } + + sub->instance = alsa_sub; + sub->active = false; + mutex_unlock(&rt->stream_mutex); + return 0; +} + +static int hiface_pcm_close(struct snd_pcm_substream *alsa_sub) +{ + struct pcm_runtime *rt = snd_pcm_substream_chip(alsa_sub); + struct pcm_substream *sub = hiface_pcm_get_substream(alsa_sub); + unsigned long flags; + + if (rt->panic) + return 0; + + pr_debug("%s: called.\n", __func__); + + mutex_lock(&rt->stream_mutex); + if (sub) { + /* deactivate substream */ + spin_lock_irqsave(&sub->lock, flags); + sub->instance = NULL; + sub->active = false; + spin_unlock_irqrestore(&sub->lock, flags); + + /* all substreams closed? if so, stop streaming */ + if (!rt->playback.instance) { + hiface_pcm_stream_stop(rt); + rt->rate = ARRAY_SIZE(rates); + } + } + mutex_unlock(&rt->stream_mutex); + return 0; +} + +static int hiface_pcm_hw_params(struct snd_pcm_substream *alsa_sub, + struct snd_pcm_hw_params *hw_params) +{ + pr_debug("%s: called.\n", __func__); + return snd_pcm_lib_malloc_pages(alsa_sub, + params_buffer_bytes(hw_params)); +} + +static int hiface_pcm_hw_free(struct snd_pcm_substream *alsa_sub) +{ + pr_debug("%s: called.\n", __func__); + return snd_pcm_lib_free_pages(alsa_sub); +} + +static int hiface_pcm_prepare(struct snd_pcm_substream *alsa_sub) +{ + struct pcm_runtime *rt = snd_pcm_substream_chip(alsa_sub); + struct pcm_substream *sub = hiface_pcm_get_substream(alsa_sub); + struct snd_pcm_runtime *alsa_rt = alsa_sub->runtime; + int ret; + + pr_debug("%s: called.\n", __func__); + + if (rt->panic) + return -EPIPE; + if (!sub) + return -ENODEV; + + mutex_lock(&rt->stream_mutex); + + sub->dma_off = 0; + sub->period_off = 0; + + if (rt->stream_state == STREAM_DISABLED) { + + ret = hiface_pcm_set_rate(rt, alsa_rt->rate); + if (ret) { + mutex_unlock(&rt->stream_mutex); + return ret; + } + ret = hiface_pcm_stream_start(rt); + if (ret) { + mutex_unlock(&rt->stream_mutex); + return ret; + } + } + mutex_unlock(&rt->stream_mutex); + return 0; +} + +static int hiface_pcm_trigger(struct snd_pcm_substream *alsa_sub, int cmd) +{ + struct pcm_substream *sub = hiface_pcm_get_substream(alsa_sub); + struct pcm_runtime *rt = snd_pcm_substream_chip(alsa_sub); + unsigned long flags; + + pr_debug("%s: called.\n", __func__); + + if (rt->panic) + return -EPIPE; + if (!sub) + return -ENODEV; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + spin_lock_irqsave(&sub->lock, flags); + sub->active = true; + spin_unlock_irqrestore(&sub->lock, flags); + return 0; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + spin_lock_irqsave(&sub->lock, flags); + sub->active = false; + spin_unlock_irqrestore(&sub->lock, flags); + return 0; + + default: + return -EINVAL; + } +} + +static snd_pcm_uframes_t hiface_pcm_pointer(struct snd_pcm_substream *alsa_sub) +{ + struct pcm_substream *sub = hiface_pcm_get_substream(alsa_sub); + struct pcm_runtime *rt = snd_pcm_substream_chip(alsa_sub); + unsigned long flags; + snd_pcm_uframes_t dma_offset; + + if (rt->panic || !sub) + return SNDRV_PCM_STATE_XRUN; + + spin_lock_irqsave(&sub->lock, flags); + dma_offset = sub->dma_off; + spin_unlock_irqrestore(&sub->lock, flags); + return bytes_to_frames(alsa_sub->runtime, dma_offset); +} + +static struct snd_pcm_ops pcm_ops = { + .open = hiface_pcm_open, + .close = hiface_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = hiface_pcm_hw_params, + .hw_free = hiface_pcm_hw_free, + .prepare = hiface_pcm_prepare, + .trigger = hiface_pcm_trigger, + .pointer = hiface_pcm_pointer, +}; + +static int hiface_pcm_init_urb(struct pcm_urb *urb, + struct hiface_chip *chip, + unsigned int ep, + void (*handler)(struct urb *)) +{ + urb->chip = chip; + usb_init_urb(&urb->instance); + + urb->buffer = kzalloc(PCM_MAX_PACKET_SIZE, GFP_KERNEL); + if (!urb->buffer) + return -ENOMEM; + + usb_fill_bulk_urb(&urb->instance, chip->dev, + usb_sndbulkpipe(chip->dev, ep), (void *)urb->buffer, + PCM_MAX_PACKET_SIZE, handler, urb); + init_usb_anchor(&urb->submitted); + + return 0; +} + +void hiface_pcm_abort(struct hiface_chip *chip) +{ + struct pcm_runtime *rt = chip->pcm; + + if (rt) { + rt->panic = true; + + if (rt->playback.instance) { + snd_pcm_stop(rt->playback.instance, + SNDRV_PCM_STATE_XRUN); + } + mutex_lock(&rt->stream_mutex); + hiface_pcm_stream_stop(rt); + mutex_unlock(&rt->stream_mutex); + } +} + +static void hiface_pcm_destroy(struct hiface_chip *chip) +{ + struct pcm_runtime *rt = chip->pcm; + int i; + + for (i = 0; i < PCM_N_URBS; i++) + kfree(rt->out_urbs[i].buffer); + + kfree(chip->pcm); + chip->pcm = NULL; +} + +static void hiface_pcm_free(struct snd_pcm *pcm) +{ + struct pcm_runtime *rt = pcm->private_data; + + pr_debug("%s: called.\n", __func__); + + if (rt) + hiface_pcm_destroy(rt->chip); +} + +int hiface_pcm_init(struct hiface_chip *chip, u8 extra_freq) +{ + int i; + int ret; + struct snd_pcm *pcm; + struct pcm_runtime *rt; + + rt = kzalloc(sizeof(*rt), GFP_KERNEL); + if (!rt) + return -ENOMEM; + + rt->chip = chip; + rt->stream_state = STREAM_DISABLED; + rt->rate = ARRAY_SIZE(rates); + if (extra_freq) + rt->extra_freq = 1; + + init_waitqueue_head(&rt->stream_wait_queue); + mutex_init(&rt->stream_mutex); + spin_lock_init(&rt->playback.lock); + + for (i = 0; i < PCM_N_URBS; i++) + hiface_pcm_init_urb(&rt->out_urbs[i], chip, OUT_EP, + hiface_pcm_out_urb_handler); + + ret = snd_pcm_new(chip->card, "USB-SPDIF Audio", 0, 1, 0, &pcm); + if (ret < 0) { + kfree(rt); + pr_err("Cannot create pcm instance\n"); + return ret; + } + + pcm->private_data = rt; + pcm->private_free = hiface_pcm_free; + + strcpy(pcm->name, "USB-SPDIF Audio"); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &pcm_ops); + + snd_pcm_lib_preallocate_pages_for_all(pcm, + SNDRV_DMA_TYPE_CONTINUOUS, + snd_dma_continuous_data(GFP_KERNEL), + MAX_BUFSIZE, MAX_BUFSIZE); + rt->instance = pcm; + + chip->pcm = rt; + return 0; +} diff --git a/sound/usb/hiface/pcm.h b/sound/usb/hiface/pcm.h new file mode 100644 index 0000000..ddc9fe7 --- /dev/null +++ b/sound/usb/hiface/pcm.h @@ -0,0 +1,24 @@ +/* + * Linux driver for M2Tech HiFace compatible devices + * + * Copyright 2012-2013 (C) M2TECH S.r.l and Amarula Solutions B.V. + * + * Authors: Michael Trimarchi <michael(a)amarulasolutions.com> + * Antonio Ospite <ao2(a)amarulasolutions.com> + * + * The driver is based on the work done in TerraTec DMX 6Fire USB + * + * 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. + */ + +#ifndef HIFACE_PCM_H +#define HIFACE_PCM_H + +struct hiface_chip; + +int hiface_pcm_init(struct hiface_chip *chip, u8 extra_freq); +void hiface_pcm_abort(struct hiface_chip *chip); +#endif /* HIFACE_PCM_H */ -- 1.7.10.4

6 22

Re: [alsa-devel] [kbuild] sound/soc/soc-fw.c:945:9-16: ERROR: se is NULL but dereferenced.
by Fengguang Wu 23 Jun '13

23 Jun '13

On Sat, Jun 22, 2013 at 12:35:44PM -0300, Raphael S Carvalho wrote: > I'm in a Linux event now, and I can't clone the tree (probably the port for > that purpose isn't allowed) > Then I sending a simplified patch that shows which lines must be changed. Patch looks good to me. Thanks! Reviewed-by: Fengguang Wu <fengguang.wu(a)intel.com> > Signed-off: Raphael S. Carvalho <raphael.scarv(a)gmail.com> > > --- soc-fw_old.c 2013-06-22 12:13:48.000000000 -0300 > +++ soc-fw_new.c 2013-06-22 12:19:41.000000000 -0300 > @@ -893,7 +893,7 @@ > if (err < 0) { > dev_err(sfw->dev, "ASoC: could not create" > " values for %s\n", ec->hdr.name); > - goto err_se; > + goto err; > } > /* fall through to create texts */ > case SOC_CONTROL_TYPE_ENUM: > @@ -905,13 +905,13 @@ > if (err < 0) { > dev_err(sfw->dev, "ASoC: could not create" > " texts for %s\n", ec->hdr.name); > - goto err_se; > + goto err; > } > break; > default: > dev_err(sfw->dev, "ASoC: invalid enum control type %d for %s\n", > ec->hdr.index, ec->hdr.name); > - goto err_se; > + goto err; > } > > /* map standard io handlers and check for external handlers */ > @@ -928,21 +928,19 @@ > SOC_CONTROL_GET_ID_GET(ec->hdr.index), > SOC_CONTROL_GET_ID_PUT(ec->hdr.index), > SOC_CONTROL_GET_ID_INFO(ec->hdr.index)); > - goto err_se; > + goto err; > } > > err = soc_fw_init_kcontrol(sfw, kc); > if (err < 0) { > dev_err(sfw->dev, "ASoC: failed to init %s\n", > ec->hdr.name); > - goto err_se; > + goto err; > } > } > return kc; > > -err_se: > - kfree(kc); > - > +err: > /* free values and texts */ > if (se->dvalues) > kfree(se->dvalues); > @@ -950,7 +948,8 @@ > kfree(se->dtexts[i]); > > kfree(se); > - > +err_se: > + kfree(kc); > return NULL; > } > > > > On Sat, Jun 22, 2013 at 11:48 AM, Fengguang Wu <fengguang.wu(a)intel.com>wrote: > > > TO: Liam Girdwood <lrg(a)ti.com> > > CC: Peter Ujfalusi <peter.ujfalusi(a)ti.com> > > > > > > Hi Liam, > > > > FYI, there are coccinelle warnings in > > > > tree: git://gitorious.org/omap-audio/linux-audiopeter/topic/ti-audio-next-bnw-wip > > head: ac1981249f0c606cede98d6dab091ecb1ae480c9 > > commit: 6d98dbbcfb5da2176b3e60faaf86d2279790fe67 ASoC: firmware: Add > > support for dynamic components > > date: 5 weeks ago > > :::::: branch date: 5 days ago > > :::::: commit date: 5 weeks ago > > > > >> sound/soc/soc-fw.c:945:9-16: ERROR: se is NULL but dereferenced. > > -- > > >> sound/soc/soc-fw.c:1523:8-9: ERROR: reference preceded by free on line > > 1523 > > >> sound/soc/soc-fw.c:1443:8-9: ERROR: reference preceded by free on line > > 1443 > > >> sound/soc/soc-fw.c:1483:8-9: ERROR: reference preceded by free on line > > 1483 > > -- > > >> sound/soc/soc-fw.c:1528:6-8: ERROR: invalid reference to the index > > variable of the iterator on line 1513 > > >> sound/soc/soc-fw.c:1448:6-8: ERROR: invalid reference to the index > > variable of the iterator on line 1433 > > >> sound/soc/soc-fw.c:1488:6-8: ERROR: invalid reference to the index > > variable of the iterator on line 1473 > > > > git remote add omap-audio git://gitorious.org/omap-audio/linux-audio > > git remote update omap-audio > > git checkout 6d98dbbcfb5da2176b3e60faaf86d2279790fe67 > > vim +945 sound/soc/soc-fw.c > > > > 6d98dbbc Liam Girdwood 2012-12-05 929 goto > > err_se; > > 6d98dbbc Liam Girdwood 2012-12-05 930 } > > 6d98dbbc Liam Girdwood 2012-12-05 931 > > 6d98dbbc Liam Girdwood 2012-12-05 932 err = > > soc_fw_init_kcontrol(sfw, kc); > > 6d98dbbc Liam Girdwood 2012-12-05 933 if (err < 0) { > > 6d98dbbc Liam Girdwood 2012-12-05 934 > > dev_err(sfw->dev, "ASoC: failed to init %s\n", > > 6d98dbbc Liam Girdwood 2012-12-05 935 > > ec->hdr.name); > > 6d98dbbc Liam Girdwood 2012-12-05 936 goto > > err_se; > > 6d98dbbc Liam Girdwood 2012-12-05 937 } > > 6d98dbbc Liam Girdwood 2012-12-05 938 } > > 6d98dbbc Liam Girdwood 2012-12-05 939 return kc; > > 6d98dbbc Liam Girdwood 2012-12-05 940 > > 6d98dbbc Liam Girdwood 2012-12-05 941 err_se: > > 6d98dbbc Liam Girdwood 2012-12-05 942 kfree(kc); > > 6d98dbbc Liam Girdwood 2012-12-05 943 > > 6d98dbbc Liam Girdwood 2012-12-05 944 /* free values and texts */ > > 6d98dbbc Liam Girdwood 2012-12-05 @945 if (se->dvalues) > > 6d98dbbc Liam Girdwood 2012-12-05 946 kfree(se->dvalues); > > 6d98dbbc Liam Girdwood 2012-12-05 947 for (i = 0; i < ec->max; > > i++) > > 6d98dbbc Liam Girdwood 2012-12-05 948 > > kfree(se->dtexts[i]); > > 6d98dbbc Liam Girdwood 2012-12-05 949 > > 6d98dbbc Liam Girdwood 2012-12-05 950 kfree(se); > > 6d98dbbc Liam Girdwood 2012-12-05 951 > > 6d98dbbc Liam Girdwood 2012-12-05 952 return NULL; > > 6d98dbbc Liam Girdwood 2012-12-05 953 } > > > > --- > > 0-DAY kernel build testing backend Open Source Technology > > Center > > http://lists.01.org/mailman/listinfo/kbuild Intel > > Corporation > > _______________________________________________ > > kbuild mailing list > > kbuild(a)lists.01.org > > https://lists.01.org/mailman/listinfo/kbuild > >

1 0

[alsa-devel] [GIT PULL] ASoC updates for v3.11
by Mark Brown 21 Jun '13

21 Jun '13

The following changes since commit 7d132055814ef17a6c7b69f342244c410a5e000f: Linux 3.10-rc6 (2013-06-15 11:51:07 -1000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.11 for you to fetch changes up to 84bbc4aa2de97143db0f6ef0c3fa0bd84f73b207: Merge remote-tracking branch 'asoc/topic/x86' into asoc-next (2013-06-17 17:20:33 +0100) ---------------------------------------------------------------- ASoC: Updates for v3.11 Not a big release subsystem wise, the main changes have been some nice improvements on the driver side: - Lots of cleanups and fixes for Blackfin, SGTL5000 and UX500. - Generalisation of the Bluetooth and HDMI stub drivers. - New CODEC drivers for SSM2518 and RT5640. - New machine driver for Tegra CPUs with RT5640. ---------------------------------------------------------------- Andrew Bresticker (1): ASoC: max98090: add digital mic mux to record path Axel Lin (2): ASoC: ssm2518: Fix trivial typo in checking tx_mask and rx_mask values ASoC: adav80x: Select SND_SOC_ADAV80X if SND_SOC_I2C_AND_SPI Bard Liao (1): ASoC: add RT5640 CODEC driver Barry Song (2): ASoC: dfbmcs320: make the driver common for other BT modules ASoC: omap-hdmi-codec: make the driver common for other SoCs Boris BREZILLON (1): ASoC: atmel-ssc: prepare clk before calling enable Charles Keepax (10): ASoC: wm5102: Correct OSR control name for EPOUT mfd: wm5102: Manually apply register patch mfd: arizona: Change fast_start pdata name to better reflect functionality ASoC: bells: Hookup DMICs for Bells ASoC: wm0010: Set IRQ as a wake source ASoC: wm0010: Clear IRQ as wake source and include missing header ASoC: arizona: Hookup SYSCLK to inputs and noise generators mfd: wm5102: Expose DRE control registers ASoC: wm5102: Expose controls for DRE ASoC: wm5110: Correct rate control for DSP4 Dan Carpenter (2): ASoC: wm_adsp: memory leak in wm_adsp_create_control() ASoC: wm8994: missing break in wm8994_get_fll_config() Daniel Mack (3): ASoC: adau1701: refactor firmware loading function ASoC: codecs: adau1701: add DT bindings ASoC: codecs: adau1701: factor out firmware reset Dimitris Papastamos (2): ASoC: wm_adsp: Expose coefficient blocks as ALSA binary controls ASoC: wm_adsp: Ensure set controls are synced on each boot Emil Goode (1): ASoC: Fix double assignment of .owner in struct snd_soc_card Fabio Baltieri (15): ASoC: ux500: register controls to card instead of codec ASoC: ux500: drop clock gating widgets from machine driver ASoC: ab8500-codec: Add missing ad_to_slot definitions ASoC: ab8500-codec: Set tx dai slots from tx_mask ASoC: ab8500-codec: Set rx dai slots from rx_mask ASoC: ux500: Drop pinctrl sleep support ASoC: ab8500-codec: Move codec ops on a separate structure ASoC: ux500: Drop dangling struct i2s_controller ASoC: ux500: Drop unused code from msp headers ASoC: ux500: Add missing mop500_ab8500.h include ASoC: ux500: Drop redundant msp id enumerations ASoC: ux500: Ensure consistent configuration between DAIs ASoC: ux500: Move DMA parameters into ux500_msp ASoC: ux500: Set DMA address during device init ASoC: ux500: Add DMA slave config prepare routine Fabio Estevam (15): ASoC: imx-sgtl5000: Do not enter the error path on success ASoC: mxs-sgtl5000: Remove unneeded 'ret' variable ASoC: mxs-sgtl5000: Remove unneeded fields from snd_soc_dai_link ASoC: fsl: imx-audmux: Let device core handle pinctrl ASoC: mxs: mxs-saif: Let device core handle pinctrl ASoC: sgtl5000: Fix comment about register addresses ASoC: sgtl5000: Convert to use regmap directly ASoC: mxs-sgtl5000: Remove unneeded 'ret' variable ASoC: mxs-sgtl5000: Remove unneeded fields from snd_soc_dai_link ASoC: sgtl5000: Read SGTL5000_CHIP_ID in i2c_probe() ASoC: sgtl5000: Fix driver probe after reset ASoC: sgtl5000: Let the codec acquire its clock ASoC: sgtl5000: Use i2c_get_clientdata() ASoC: sgtl5000: Add 'clocks' entry as a required propery ASoC: imx-sgtl5000: Use devm_clk_get() Geert Uytterhoeven (1): blackfin: bf533-stamp: Remove bogus "||" Kuninori Morimoto (1): ASoC: fsi: fixup sparse errors Lars-Peter Clausen (36): ASoC: spear: Setup dma data in DAI probe ASoC: spear: Use generic dmaengine PCM ASoC: jz4740-i2s: Use clk_prepare_enable/clk_disable_unprepare ASoC: ep93xx: Setup dma data in DAI probe ASoC: ep93xx: Use generic dmaengine PCM ASoC: core: Move snd_soc_set_runtime_hwparams() to soc-pcm.c ASoC: core: Add helper function to initialize the runtime pcm ASoC: core: Use kasprintf instead of opencoding it ASoC: ep93xx-i2s: Staticize non exported struct ASoC: kirkwood-dma: Staticize non exported struct ASoC: spear: spdif_in: Staticize non exported struct ASoC: spear: spdif_out: Staticize unexported function ASoC: mmp-pcm: Staticize non exported structs and functions ASoC: mmp-sspa: Staticize non exported struct ASoC: bf5xx-tdm-pcm: Staticize non exported struct ASoC: mop500_ab8500: Staticize non exported functions ASoC: mop500: Staticize non exported struct ASoC: sn95031: Staticize non exported struct ASoC: davinci-sffsdr: Staticize non exported struct ASoC: Fix early event callback list iteration ASoC: Add ssm2518 support ASoC: blackfin: Remove unused bf5xx-{i2s, tdm, ac97}-pcm.h ASoC: blackfin: bf5xx-i2s: Use dev_{err, dbg} instead of pr_{error, debug} ASoC: blackfin: bf5xx-sport: Allow setting rx and tx mask independently ASoC: blackfin: bf5xx-i2s: Allocate buffer only as large as requested ASoC: blackfin: bf5xx-i2s-pcm: Use snd_pcm_lib_preallocate_pages_for_all() ASoC: blackfin: bf5xx-i2s: Add support for TDM mode ASoC: blackfin: Switch bf5xx-ad193x from bf5xx-tdm to bf5xx-i2s ASoC: blackfin: Switch bf5xx-ad1836 from bf5xx-tdm to bf5xx-i2s ASoC: blackfin: Remove bf5xx-tdm driver blackfin: Remove references to the bf5x_tdm driver ASoC: dapm: Setup private_free callback for dapm kcontrols ASoC: dapm: Remove unnecessary loop ASoC: dapm: Add a helper function to free a DAPM path ASoC: dapm: Remove unused long_name field from snd_soc_dapm_path struct ASoC: dapm: Add snd_soc_dapm_switch to the power up/down sequence table Marek Belisko (3): ASoC: spdif_transceiver: Change driver filename to spdif_transmitter.c. ASoC: spdif_transmitter: Add DT support. ASoC: spdif_receiver: Add DT support. Marek Vasut (1): ASoC: sgtl5000: Fix VAG_POWER enabling/disabling order Mark Brown (68): ASoC: sam9g20ek: Let device core handle pinctrl ASoC: dapm: Provide early event callbacks for power up and down ASoC: wm8994: Support EFS mode for FLL mfd: arizona: Read the device identification information after boot mfd: arizona: Add missing cleanup on remove mfd: arizona: Disable interrupts during suspend mfd: Add device tree bindings for Arizona class devices mfd: arizona: Support use of external DCVDD ASoC: wm8994: Handle LRCLK inversion for WM8958 and WM1811A ASoC: arizona: Provide simple DAI ops for autoconfiguring interfaces ASoC: wm5102: Stub hookup for Slimbus interface ASoC: wm5110: Stub hookup for Slimbus interface Merge tag 'v3.10-rc2' into asoc-ux500 mfd: wm5110: Make DSPn_STATUS_3 readable Merge tag 'mfd-arizona-v3.10-rc2' of git://git.kernel.org/.../broonie/misc into asoc-arizona ASoC: wm8994: Remove restore of DAC enable state ASoC: wm8994: Ensure lambda is zeroed for WM8994 ASoC: wm8994: Defer declaration of open circuit microphones ASoC: wm8994: Add digital loopback paths ASoC: wm8994: Remove unused variable ASoC: wm8994: Allow debounce before MICDET identification ASoC: sgtl5000: Make device cache only when powered off ASoC: max98090: Guard runtime PM callbacks Merge branch 'topic/fsl' of git://git.kernel.org/.../broonie/sound into asoc-sgtl5000 ASoC: wm8962: Restore device state after reset in runtime resume Merge branch 'fix/wm8962' of git://git.kernel.org/.../broonie/sound into asoc-wm8962 ASoC: SPEAr: Add dependency on dmaengine helpers ASoC: SPEAr: Hide component drivers in Kconfig ASoC: spear: Remove nonexistant EVM options ASoC: spear: Normalise module names Merge remote-tracking branch 'asoc/fix/adav80x' into asoc-linus Merge remote-tracking branch 'asoc/fix/adsp' into asoc-linus Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branch 'asoc/fix/wm8962' into asoc-linus Merge remote-tracking branch 'asoc/topic/88pm860x' into asoc-next Merge remote-tracking branch 'asoc/topic/adau1701' into asoc-next Merge remote-tracking branch 'asoc/topic/adsp' 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/blackfin' into asoc-next Merge remote-tracking branch 'asoc/topic/core' into asoc-next Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/davinci' into asoc-next Merge remote-tracking branch 'asoc/topic/dfmbcs320' into asoc-next Merge remote-tracking branch 'asoc/topic/dwc' into asoc-next Merge remote-tracking branch 'asoc/topic/ep93xx' 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/hdmi' 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/max98090' into asoc-next Merge remote-tracking branch 'asoc/topic/mxs' into asoc-next Merge remote-tracking branch 'asoc/topic/omap' into asoc-next Merge remote-tracking branch 'asoc/topic/pxa' into asoc-next Merge remote-tracking branch 'asoc/topic/rt5640' into asoc-next Merge remote-tracking branch 'asoc/topic/samsung' into asoc-next Merge remote-tracking branch 'asoc/topic/sgtl5000' into asoc-next Merge remote-tracking branch 'asoc/topic/sn95031' into asoc-next Merge remote-tracking branch 'asoc/topic/spdif' into asoc-next Merge remote-tracking branch 'asoc/topic/spear' into asoc-next Merge remote-tracking branch 'asoc/topic/ssm2518' into asoc-next Merge remote-tracking branch 'asoc/topic/tegra' 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/wm8962' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8994' into asoc-next Merge remote-tracking branch 'asoc/topic/x86' into asoc-next Michał Mirosław (1): ASoC: fix kernel message grepability Nicolin Chen (5): ASoC: WM8962: Create default platform data structure ASoC: wm8962: Add device tree binding ASoC: fsl: add imx-wm8962 machine driver ASoC: wm8962: Remove remaining direct register cache accesses ASoC: wm8962: Enable start-up and normal bias after reset in runtime resume Paul Bolle (2): ASoC: remove saarb and tavorevb3 machine drivers ASoC: OMAP: Remove obsolete Makefile line Rajeev Kumar (3): ASoC: dwc: debug message correction. ASoC: dwc: Folder path correction in file header. ASoC: Add Kconfig and Makefile to support SPEAr audio driver Sachin Kamat (7): ASoC: 88pm860x: Remove redundant platform_set_drvdata() ASoC: ep93xx: Remove redundant platform_set_drvdata() ASoC: jz4740: Remove redundant platform_set_drvdata() ASoC: omap-mcbsp: Remove redundant platform_set_drvdata() ASoC: Samsung: Remove redundant platform_set_drvdata() ASoC: mid-x86: Remove redundant platform_set_drvdata() ASoC: davinci: Remove redundant use of of_match_ptr macro Sekhar Nori (1): ASoC: davinci: remove sffsdr machine support Shawn Guo (6): ASoC: mxs: move to use generic DMA helper ASoC: fsl: remove use of imx-pcm-audio from fsl_ssi ASoC: fsl: remove use of imx-pcm-audio from imx-ssi ASoC: fsl: create function imx_pcm_fiq_exit() ASoC: fsl: remove use of imx-fiq-pcm-audio from imx-ssi ASoC: fsl: remove imx-pcm driver Stephen Warren (7): ASoC: tegra: always use clk_get() in utility code ASoC: tegra: implement suspend/resume for Tegra30 AHUB ASoC: tegra: implement suspend/resume for Tegra30 I2S ASoC: tegra: add runtime PM to resume functions ASoC: rt5640: add device tree support ASoC: tegra: add tegra+RT5640 machine driver ASoC: rt5640: fix sparse warnings Tushar Behera (1): ASoC: spear: Convert to use devm_ioremap_resource Uwe Kleine-König (3): ASoC: SPEAr spdif_{in,out}: use devm for clk and a few more cleanups ASoC: simplify registration of snd-soc-dummy device ASoC: SPEAr spdif_{in,out}: fix fallout of previous cleanup Documentation/devicetree/bindings/mfd/arizona.txt | 62 + .../devicetree/bindings/sound/adi,adau1701.txt | 23 + .../devicetree/bindings/sound/imx-audio-wm8962.txt | 46 + .../devicetree/bindings/sound/mxs-saif.txt | 17 +- .../bindings/sound/nvidia,tegra-audio-rt5640.txt | 71 + Documentation/devicetree/bindings/sound/rt5640.txt | 30 + .../devicetree/bindings/sound/sgtl5000.txt | 3 + .../devicetree/bindings/sound/spdif-receiver.txt | 10 + .../bindings/sound/spdif-transmitter.txt | 10 + .../devicetree/bindings/sound/ssm2518.txt | 20 + Documentation/devicetree/bindings/sound/wm8962.txt | 23 + arch/blackfin/mach-bf527/boards/ad7160eval.c | 12 - arch/blackfin/mach-bf527/boards/ezkit.c | 32 +- arch/blackfin/mach-bf533/boards/ezkit.c | 12 - arch/blackfin/mach-bf533/boards/stamp.c | 35 +- arch/blackfin/mach-bf537/boards/stamp.c | 30 +- arch/blackfin/mach-bf548/boards/ezkit.c | 28 +- arch/blackfin/mach-bf561/boards/ezkit.c | 14 +- arch/blackfin/mach-bf609/boards/ezkit.c | 2 +- drivers/mfd/arizona-core.c | 234 ++- drivers/mfd/arizona-i2c.c | 10 +- drivers/mfd/arizona-spi.c | 10 +- drivers/mfd/arizona.h | 12 + drivers/mfd/wm5102-tables.c | 9 +- drivers/mfd/wm5110-tables.c | 8 + drivers/misc/atmel-ssc.c | 8 +- include/linux/mfd/arizona/core.h | 2 + include/linux/mfd/arizona/pdata.h | 2 +- include/linux/mfd/arizona/registers.h | 47 + include/linux/mfd/wm8994/pdata.h | 5 + include/linux/mfd/wm8994/registers.h | 8 + include/linux/platform_data/ssm2518.h | 22 + include/sound/rt5640.h | 22 + include/sound/soc-dapm.h | 3 +- sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/atmel/sam9g20_wm8731.c | 9 - sound/soc/blackfin/Kconfig | 47 +- sound/soc/blackfin/Makefile | 4 - sound/soc/blackfin/bf5xx-ac97-pcm.c | 1 - sound/soc/blackfin/bf5xx-ac97-pcm.h | 26 - sound/soc/blackfin/bf5xx-ac97.c | 8 +- sound/soc/blackfin/bf5xx-ad1836.c | 19 +- sound/soc/blackfin/bf5xx-ad193x.c | 40 +- sound/soc/blackfin/bf5xx-ad1980.c | 1 - sound/soc/blackfin/bf5xx-ad73311.c | 1 - sound/soc/blackfin/bf5xx-i2s-pcm.c | 183 +- sound/soc/blackfin/bf5xx-i2s-pcm.h | 21 +- sound/soc/blackfin/bf5xx-i2s.c | 129 +- sound/soc/blackfin/bf5xx-sport.c | 10 +- sound/soc/blackfin/bf5xx-sport.h | 2 +- sound/soc/blackfin/bf5xx-ssm2602.c | 1 - sound/soc/blackfin/bf5xx-tdm-pcm.c | 345 ---- sound/soc/blackfin/bf5xx-tdm-pcm.h | 18 - sound/soc/blackfin/bf5xx-tdm.c | 328 --- sound/soc/blackfin/bf5xx-tdm.h | 23 - sound/soc/cirrus/Kconfig | 2 +- sound/soc/cirrus/ep93xx-ac97.c | 16 +- sound/soc/cirrus/ep93xx-i2s.c | 16 +- sound/soc/cirrus/ep93xx-pcm.c | 138 +- sound/soc/codecs/88pm860x-codec.c | 9 +- sound/soc/codecs/Kconfig | 18 +- sound/soc/codecs/Makefile | 14 +- sound/soc/codecs/ab8500-codec.c | 85 +- sound/soc/codecs/ab8500-codec.h | 42 +- sound/soc/codecs/adau1701.c | 69 +- sound/soc/codecs/arizona.c | 7 + sound/soc/codecs/arizona.h | 3 +- sound/soc/codecs/{dfbmcs320.c => bt-sco.c} | 37 +- sound/soc/codecs/{omap-hdmi.c => hdmi.c} | 26 +- sound/soc/codecs/jz4740.c | 2 - sound/soc/codecs/max98090.c | 24 +- sound/soc/codecs/rt5640.c | 2128 ++++++++++++++++++++ sound/soc/codecs/rt5640.h | 2092 +++++++++++++++++++ sound/soc/codecs/sgtl5000.c | 267 ++- sound/soc/codecs/sgtl5000.h | 2 +- sound/soc/codecs/sn95031.c | 2 +- sound/soc/codecs/spdif_receiver.c | 10 + .../{spdif_transciever.c => spdif_transmitter.c} | 10 + sound/soc/codecs/ssm2518.c | 856 ++++++++ sound/soc/codecs/ssm2518.h | 20 + sound/soc/codecs/wm0010.c | 10 + sound/soc/codecs/wm5102.c | 205 +- sound/soc/codecs/wm5110.c | 192 ++ sound/soc/codecs/wm8962.c | 143 +- sound/soc/codecs/wm8994.c | 188 +- sound/soc/codecs/wm8994.h | 3 + sound/soc/codecs/wm_adsp.c | 461 ++++- sound/soc/codecs/wm_adsp.h | 3 + sound/soc/davinci/Kconfig | 10 - sound/soc/davinci/Makefile | 2 - sound/soc/davinci/davinci-mcasp.c | 4 +- sound/soc/davinci/davinci-sffsdr.c | 181 -- sound/soc/dwc/designware_i2s.c | 6 +- sound/soc/fsl/Kconfig | 17 +- sound/soc/fsl/Makefile | 13 +- sound/soc/fsl/eukrea-tlv320.c | 2 +- sound/soc/fsl/fsl_ssi.c | 13 +- sound/soc/fsl/imx-audmux.c | 8 - sound/soc/fsl/imx-mc13783.c | 2 +- sound/soc/fsl/imx-pcm-dma.c | 2 + sound/soc/fsl/imx-pcm-fiq.c | 92 + sound/soc/fsl/imx-pcm.c | 145 -- sound/soc/fsl/imx-pcm.h | 10 +- sound/soc/fsl/imx-sgtl5000.c | 39 +- sound/soc/fsl/imx-ssi.c | 44 +- sound/soc/fsl/imx-ssi.h | 3 - sound/soc/fsl/imx-wm8962.c | 323 +++ sound/soc/fsl/mx27vis-aic32x4.c | 2 +- sound/soc/fsl/phycore-ac97.c | 2 +- sound/soc/fsl/wm1133-ev1.c | 2 +- sound/soc/jz4740/jz4740-i2s.c | 17 +- sound/soc/kirkwood/kirkwood-dma.c | 2 +- sound/soc/mid-x86/mfld_machine.c | 1 - sound/soc/mxs/mxs-pcm.c | 18 - sound/soc/mxs/mxs-pcm.h | 7 - sound/soc/mxs/mxs-saif.c | 37 +- sound/soc/mxs/mxs-saif.h | 1 - sound/soc/mxs/mxs-sgtl5000.c | 10 +- sound/soc/omap/Makefile | 1 - sound/soc/omap/omap-hdmi-card.c | 2 +- sound/soc/omap/omap-mcbsp.c | 2 - sound/soc/pxa/Kconfig | 20 - sound/soc/pxa/Makefile | 4 - sound/soc/pxa/mmp-pcm.c | 6 +- sound/soc/pxa/mmp-sspa.c | 2 +- sound/soc/pxa/saarb.c | 190 -- sound/soc/pxa/tavorevb3.c | 189 -- sound/soc/pxa/zylonite.c | 1 - sound/soc/samsung/Kconfig | 2 +- sound/soc/samsung/bells.c | 14 + sound/soc/samsung/neo1973_wm8753.c | 2 +- sound/soc/samsung/smdk_wm8580pcm.c | 1 - sound/soc/samsung/smdk_wm8994pcm.c | 1 - sound/soc/sh/fsi.c | 10 +- sound/soc/soc-core.c | 106 +- sound/soc/soc-dapm.c | 110 +- sound/soc/soc-pcm.c | 91 +- sound/soc/soc-utils.c | 13 +- sound/soc/spear/Kconfig | 9 + sound/soc/spear/Makefile | 8 + sound/soc/spear/spdif_in.c | 31 +- sound/soc/spear/spdif_out.c | 43 +- sound/soc/spear/spear_pcm.c | 152 +- sound/soc/tegra/Kconfig | 10 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra30_ahub.c | 25 + sound/soc/tegra/tegra30_i2s.c | 26 + sound/soc/tegra/tegra_asoc_utils.c | 23 +- sound/soc/tegra/tegra_rt5640.c | 257 +++ sound/soc/ux500/mop500.c | 2 +- sound/soc/ux500/mop500_ab8500.c | 62 +- sound/soc/ux500/ux500_msp_dai.c | 11 +- sound/soc/ux500/ux500_msp_dai.h | 4 - sound/soc/ux500/ux500_msp_i2s.c | 88 +- sound/soc/ux500/ux500_msp_i2s.h | 74 +- sound/soc/ux500/ux500_pcm.c | 30 + 157 files changed, 8786 insertions(+), 3053 deletions(-) create mode 100644 Documentation/devicetree/bindings/mfd/arizona.txt create mode 100644 Documentation/devicetree/bindings/sound/adi,adau1701.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt create mode 100644 Documentation/devicetree/bindings/sound/rt5640.txt create mode 100644 Documentation/devicetree/bindings/sound/spdif-receiver.txt create mode 100644 Documentation/devicetree/bindings/sound/spdif-transmitter.txt create mode 100644 Documentation/devicetree/bindings/sound/ssm2518.txt create mode 100644 include/linux/platform_data/ssm2518.h create mode 100644 include/sound/rt5640.h delete mode 100644 sound/soc/blackfin/bf5xx-ac97-pcm.h delete mode 100644 sound/soc/blackfin/bf5xx-tdm-pcm.c delete mode 100644 sound/soc/blackfin/bf5xx-tdm-pcm.h delete mode 100644 sound/soc/blackfin/bf5xx-tdm.c delete mode 100644 sound/soc/blackfin/bf5xx-tdm.h rename sound/soc/codecs/{dfbmcs320.c => bt-sco.c} (53%) rename sound/soc/codecs/{omap-hdmi.c => hdmi.c} (69%) create mode 100644 sound/soc/codecs/rt5640.c create mode 100644 sound/soc/codecs/rt5640.h rename sound/soc/codecs/{spdif_transciever.c => spdif_transmitter.c} (88%) create mode 100644 sound/soc/codecs/ssm2518.c create mode 100644 sound/soc/codecs/ssm2518.h delete mode 100644 sound/soc/davinci/davinci-sffsdr.c delete mode 100644 sound/soc/fsl/imx-pcm.c create mode 100644 sound/soc/fsl/imx-wm8962.c delete mode 100644 sound/soc/pxa/saarb.c delete mode 100644 sound/soc/pxa/tavorevb3.c create mode 100644 sound/soc/spear/Kconfig create mode 100644 sound/soc/spear/Makefile create mode 100644 sound/soc/tegra/tegra_rt5640.c

2 1

[alsa-devel] [PATCH 0/6] Convert JZ4740 to dmaengine
by Lars-Peter Clausen 21 Jun '13

21 Jun '13

Hi, This series replaces the custom JZ4740 DMA API with a dmaengine driver. This is done in 3 steps: 1) Add a dmaengine driver which wraps the custom JZ4740 DMA API 2) Update all users of the JZ4740 DMA API to use dmaengine instead 3) Remove the custom API and move all direct hardware access to the dmaengine driver. The first two patches in the series also make sure that the clock of the DMA core is enabled. Since the patches in this series depend on each other I'd prefer if they could all go through the DMA tree. There are a few minor changes to the patch 3 in v2 of this series, all other patches are identical to v1. - Lars Lars-Peter Clausen (4): dma: Add a jz4740 dmaengine driver MIPS: jz4740: Register jz4740 DMA device ASoC: jz4740: Use the generic dmaengine PCM driver MIPS: jz4740: Remove custom DMA API Maarten ter Huurne (2): MIPS: jz4740: Correct clock gate bit for DMA controller MIPS: jz4740: Acquire and enable DMA controller clock arch/mips/include/asm/mach-jz4740/dma.h | 56 --- arch/mips/include/asm/mach-jz4740/platform.h | 1 + arch/mips/jz4740/Makefile | 2 +- arch/mips/jz4740/board-qi_lb60.c | 1 + arch/mips/jz4740/clock.c | 2 +- arch/mips/jz4740/dma.c | 287 ------------- arch/mips/jz4740/platform.c | 21 + drivers/dma/Kconfig | 6 + drivers/dma/Makefile | 1 + drivers/dma/dma-jz4740.c | 617 +++++++++++++++++++++++++++ sound/soc/jz4740/Kconfig | 1 + sound/soc/jz4740/jz4740-i2s.c | 48 +-- sound/soc/jz4740/jz4740-pcm.c | 310 +------------- sound/soc/jz4740/jz4740-pcm.h | 20 - 14 files changed, 676 insertions(+), 697 deletions(-) delete mode 100644 arch/mips/jz4740/dma.c create mode 100644 drivers/dma/dma-jz4740.c delete mode 100644 sound/soc/jz4740/jz4740-pcm.h -- 1.8.2.1

3 17

[alsa-devel] [PATCH 0/4] ALSA: usb: misc consistency fixes
by Antonio Ospite 21 Jun '13

21 Jun '13

Hi, when working on the hiFace driver (which I plan on re-sending later today BTW) I've been reading the code of other drivers and I've stumbled on a couple of minor inconsistencies which were also spotted during the review rounds for the hiFace driver. I am sending some patches to address these issues, so that new drivers authors won't do the same mistakes I did when copying code over. Please take a closer look at the ones about vmalloc buffers, do we need an actual Tested-by for these? Patches are against 3.10-rc6. Thanks, Antonio Antonio Ospite (4): ALSA: snd-usb-caiaq: remove the unused snd_card_used variable ALSA: snd-usb-caiaq: use vmalloc buffers ALSA: snd-usb-6fire: use vmalloc buffers ALSA: usb: uniform style used in MODULE_SUPPORTED_DEVICE() sound/usb/6fire/chip.c | 2 +- sound/usb/6fire/pcm.c | 12 +++++------- sound/usb/caiaq/audio.c | 12 +++++------- sound/usb/caiaq/device.c | 31 +++++++++++++++---------------- sound/usb/usx2y/usbusx2y.c | 2 +- 5 files changed, 27 insertions(+), 32 deletions(-) -- Antonio Ospite http://ao2.it A: Because it messes up the order in which people normally read text. See http://en.wikipedia.org/wiki/Posting_style Q: Why is top-posting such a bad thing?

2 5

[alsa-devel] [patch] ALSA: vx_core: off by one in vx_read_status()
by Dan Carpenter 21 Jun '13

21 Jun '13

This code is older than git, and I haven't tested it, but if size == SIZE_MAX_STATUS then we would write one space past the end of the rmh->Stat[] array. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/drivers/vx/vx_core.c b/sound/drivers/vx/vx_core.c index c39961c..8359689 100644 --- a/sound/drivers/vx/vx_core.c +++ b/sound/drivers/vx/vx_core.c @@ -205,7 +205,7 @@ static int vx_read_status(struct vx_core *chip, struct vx_rmh *rmh) if (size < 1) return 0; - if (snd_BUG_ON(size > SIZE_MAX_STATUS)) + if (snd_BUG_ON(size >= SIZE_MAX_STATUS)) return -EINVAL; for (i = 1; i <= size; i++) {

2 1

[alsa-devel] WM8741 DAC with AT91SAM9G20
by Michael Lathion 21 Jun '13

21 Jun '13

Hi all, I'm trying to use a WM8742 dac instead of WM8731 on a custom board based on AT91SAM9G20EK (developpment board from Atmel for AT91SAM9G20 soc) with Linux Kernel 2.6.38. I have the driver working in alsa (driver compiled, mixer up in alsa, ...) but I can't have the ssc interface working. When trying to play sound with aplay I get only the masterclock and bitclock but no frame clock (LRCLK), see output below. What I've done is modifying the /sound/soc/atmel/sam9g20_wm8731.c file but it seems something is not configured as it should (probably related to WM8742 being slave and the WM8731 used on original dev board was master for clock distribution). My modified sam9g20_wm8741.c is enclosed to this mail, i suspect something is wrongly setup in CPU DAI configuration but can't find what. I'm trying to set this up from some weeks now but can't find the right config so if someone have any idea on what can has to be done it could be great! Thanks for help Regards Michael Linux kernel 2.6.38 AT91SAM9G20 processor WM8741 DAC Kernel boot output : ASoC: Mclk set to 22579200 wm8741-codec 0-001b: Successful registration FOX_Board_wm8741 : PB1003A: wm8741__init() called wm8741-codec 0-001b: wm8741_set_dai_sysclk info: freq=22579200 asoc: wm8741 <-> atmel-ssc-dai.0 mapping ok ALSA device list: #0: DAISY16 When trying play sound kernel driver print: wm8741-codec 0-001b: wm8741_set_dai_fmt: Format=0, Clock Inv=0 wm8741-codec 0-001b: wm8741_hw_params: bit size param = 2

2 2

[alsa-devel] [PATCH 1/2] ALSA: hda - Remove unused variable
by Wang Xingchao 21 Jun '13

21 Jun '13

Signed-off-by: Wang Xingchao <xingchao.wang(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 7803ddd..0f7a380 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1891,7 +1891,6 @@ static void intel_haswell_fixup_connect_list(struct hda_codec *codec, /* override pins connection list */ snd_printdd("hdmi: haswell: override pin connection 0x%x\n", nid); - nconns = max(spec->num_cvts, 4); snd_hda_override_conn_list(codec, nid, spec->num_cvts, spec->cvt_nids); } -- 1.8.1.2

1 1

Re: [alsa-devel] [RFC PATCH 00/11] ARM: s3c64xx: Let amba-pl08x driver handle DMA
by Mark Brown 21 Jun '13

21 Jun '13

On Wed, Jun 19, 2013 at 08:26:12PM +0200, Tomasz Figa wrote: > On Wednesday 19 of June 2013 18:40:47 Mark Brown wrote: > > - ret = pd->get_signal(plchan->cd); > > + ret = (pd->get_signal)(plchan->cd); > Hmm, that's strange. The former is a completely valid piece of code... I know, hence... > > to get it to build which makes me suspect the compiler a bit as well... ...my comment about suspecting the compiler. > > I was applying this to -next, are there any other dependencies I need or > > anything? > Hmm, I've been testing this on top of my common clock framework and device > tree patches, but I don't think this had any effect. Did you add necessary > clkdev lookups to the clock driver? No, I didn't - that's most likely it, I didn't really investigate. I didn't test the watchdog stuff as the clocks didn't get sent to me. > In Samsung CCF alias notation it looks like this: > + ALIAS(HCLK_DMA1, "dma-pl080s.1", "apb_pclk"), > + ALIAS(HCLK_DMA0, "dma-pl080s.0", "apb_pclk"), > Not sure how hard it will be to add such lookups to the old clock driver, > though. It's pretty much the same providing you know which clock needs to be used. > I will test this applied directly on top of current linux-next when I find > some time, but for now you might check out my v3.11-devel branch on my > github: > https://github.com/tom3q/linux.git Will try to get round to it.

2 4

[alsa-devel] [PATCH v2 0/3] ASoC: codecs: some more improvements for adau1701
by Daniel Mack 21 Jun '13

21 Jun '13

Sorry Lars, it's been a while since the last round. I hope I addressed all the details you mentioned. Thanks for having another look, Daniel Daniel Mack (3): ASoC: codecs: adau1701: allow configuration of PLL mode pins ASoC: codecs: adau1701: switch to direct regmap API usage ASoC: codecs: adau1701: add support for pin muxing .../devicetree/bindings/sound/adi,adau1701.txt | 18 ++ sound/soc/codecs/adau1701.c | 250 ++++++++++++++++----- sound/soc/codecs/adau1701.h | 4 + 3 files changed, 216 insertions(+), 56 deletions(-) -- 1.8.1.4

3 17

Re: [alsa-devel] [PATCH V6 6/6] clk: exynos5250: Add enum entries for divider clock of i2s1 and i2s2
by Padma Venkat 21 Jun '13

21 Jun '13

Hi Mike, On Wed, Jun 12, 2013 at 10:15 PM, Mike Turquette <mturquette(a)linaro.org> wrote: > Quoting Padmavathi Venna (2013-06-12 01:07:43) >> This patch adds enum entries for div_i2s1 and div_i2s2 which are >> required for i2s1 and i2s2 controllers. >> >> Signed-off-by: Padmavathi Venna <padma.v(a)samsung.com> > > Looks good. Did you want me to take the clk patches or just gathering > Acks? Please take the patches :). Thanks Padma > > Regards, > Mike > >> --- >> drivers/clk/samsung/clk-exynos5250.c | 5 +++-- >> 1 files changed, 3 insertions(+), 2 deletions(-) >> >> diff --git a/drivers/clk/samsung/clk-exynos5250.c b/drivers/clk/samsung/clk-exynos5250.c >> index 5c97e75..7c68850 100644 >> --- a/drivers/clk/samsung/clk-exynos5250.c >> +++ b/drivers/clk/samsung/clk-exynos5250.c >> @@ -87,6 +87,7 @@ enum exynos5250_clks { >> sclk_mmc0, sclk_mmc1, sclk_mmc2, sclk_mmc3, sclk_sata, sclk_usb3, >> sclk_jpeg, sclk_uart0, sclk_uart1, sclk_uart2, sclk_uart3, sclk_pwm, >> sclk_audio1, sclk_audio2, sclk_spdif, sclk_spi0, sclk_spi1, sclk_spi2, >> + div_i2s1, div_i2s2, >> >> /* gate clocks */ >> gscl0 = 256, gscl1, gscl2, gscl3, gscl_wa, gscl_wb, smmu_gscl0, >> @@ -291,8 +292,8 @@ struct samsung_div_clock exynos5250_div_clks[] __initdata = { >> DIV(none, "div_pcm1", "sclk_audio1", DIV_PERIC4, 4, 8), >> DIV(none, "div_audio2", "mout_audio2", DIV_PERIC4, 16, 4), >> DIV(none, "div_pcm2", "sclk_audio2", DIV_PERIC4, 20, 8), >> - DIV(none, "div_i2s1", "sclk_audio1", DIV_PERIC5, 0, 6), >> - DIV(none, "div_i2s2", "sclk_audio2", DIV_PERIC5, 8, 6), >> + DIV(div_i2s1, "div_i2s1", "sclk_audio1", DIV_PERIC5, 0, 6), >> + DIV(div_i2s2, "div_i2s2", "sclk_audio2", DIV_PERIC5, 8, 6), >> DIV(sclk_pixel, "div_hdmi_pixel", "sclk_vpll", DIV_DISP1_0, 28, 4), >> DIV_A(none, "armclk", "div_arm", DIV_CPU0, 28, 3, "armclk"), >> DIV_F(none, "div_mipi1_pre", "div_mipi1", >> -- >> 1.7.4.4

3 3

[alsa-devel] mint 13 mate + ATI IXP SB4x0 High Definition Audio Controller (rev 01)
by Pavel 20 Jun '13

20 Jun '13

Hello! i have a big problem - no sound after install linux mint 13 mate. well, sometimes if i reboot - i have it, but after nex reboot - no sound no more, seems like pulseaudio don't see my sound card. Laptop: Manufacturer: FUJITSU SIEMENS Product Name: AMILO Pa 1510 Product Version: Not Applicable Firmware Version: 1.07C soundcard: 00:14.2 Audio device: Advanced Micro Devices [AMD] nee ATI IXP SB4x0 High Definition Audio Controller (rev 01) kernel 3.2.0-48-generic dmesg: http://pastebin.com/KF8ABwnY alsa-info: http://pastebin.com/QYBYNJi9 i read this topic: https://bugzilla.kernel.org/show_bug.cgi?id=18512 and tried to add "options snd-hda-intel model=6stack-dig single_cmd=1" to alsa-base.conf - it helped, after reboot i has sound, till i reboot - now no sound no more.. i tried everything i found in the google, even alsa-update - http://www.stchman.com/alsa_update.html nothing can help me, but i understand that's the problem is in the driver.. can someone pls help me to fix the problem? thank you.

1 0

[alsa-devel] mint 13 mate + ATI IXP SB4x0 High Definition Audio Controller (rev 01)
by Pavel 20 Jun '13

20 Jun '13

Hello! i have a big problem - no sound after install linux mint 13 mate. well, sometimes if i reboot - i have it, but after nex reboot - no sound no more, seems like pulseaudio don't see my sound card. Laptop: Manufacturer: FUJITSU SIEMENS Product Name: AMILO Pa 1510 Product Version: Not Applicable Firmware Version: 1.07C soundcard: 00:14.2 Audio device: Advanced Micro Devices [AMD] nee ATI IXP SB4x0 High Definition Audio Controller (rev 01) kernel 3.2.0-48-generic dmesg: http://pastebin.com/KF8ABwnY alsa-info: http://pastebin.com/QYBYNJi9 i read this topic: https://bugzilla.kernel.org/show_bug.cgi?id=18512 and tried to add "options snd-hda-intel model=6stack-dig single_cmd=1" to alsa-base.conf - it helped, after reboot i has sound, till i reboot - now no sound no more.. i tried everything i found in the google, even alsa-update - http://www.stchman.com/alsa_update.html nothing can help me, but i understand that's the problem is in the driver.. can someone pls help me to fix the problem? thank you.

1 0

[alsa-devel] cs42l52 alsa driver broken beep volume control
by Lars-Peter Clausen 20 Jun '13

20 Jun '13

Hi, I stumbled upon this while making some changes to the SOC_SINGLE_VALUE macro. For the Beep Volume control the cs42l52 driver sets the invert field to 7, which doesn't make much sense and is causing a build warning with my changes. I assume the intention was to actually use SOC_SINGLE_SX_TLV instead of SOC_SINGLE_TLV. But I'm not sure if setting min to 0x1f and max 0x7 is right either. Also according to the datasheet the tlv dB info seems to be wrong. The Beep Volume uses 2dB steps while the info uses 0.5dB steps. - Lars

2 2

[alsa-devel] [PATCH v8 00/11] ASoC: fsl-ssi: ac97-slave support
by Markus Pargmann 20 Jun '13

20 Jun '13

Hi, This series adds DT support for phycore-ac97. Beside ac97 support, the series adds imx-pcm-fiq and generic DMA bindings to fsl-ssi. In version 8 I fixed the build issues by seperating the DT capable ac97 phycore driver from the old driver. Regards, Markus Changes in v8: - Fix build issues in phycore-ac97. Changes in v7: - Only one soc_ac97_ops struct for both ssi drivers (fsl-ssi and imx-ssi) to make it possible to compile a kernel with both drivers. - Other small bug/style fixes described in the changelogs of the patches. Changes in v6: - Rebased onto 3.10-rc3 with Shawn's pcm cleanup patches - Setup of reset functions moved to phycore-ac97.c Changes in v5: - Pass of_node via parent device to imx-pcm-dma. Changes in v4: - phycore-ac97 uses the iomux functions based on audmux phandle now. - fsl-ssi imx-pcm-fiq got some new comments about problems with imx-ssi and incompatible codecs Changes in v3: - Rebased onto pcm dma cleanup patches - A lot of cleanups for the fsl-ssi ac97 integration - Some small changes which are listed in the notes of the patches. Markus Pargmann (11): ASoC: imx-pcm-dma: DT support ASoC: imx-pcm-fiq: Introduce pcm-fiq-params ASoC: fsl: Move soc_ac97_ops from imx-ssi to fsl_ssi ASoC: fsl-ssi: Add support for imx-pcm-fiq ASoC: fsl-ssi: Use generic DMA bindings if possible ASoC: fsl-ssi: imx ac97 support ARM: imx: Export ac97 reset functions ASoC: Add phycore-ac97-dt driver ASoC: fsl: Kconfig: Use fsl-ssi for phycore-ac97 ASoC: fsl: Move fsl-ssi binding doc to sound/ ASoC: fsl: Update fsl-ssi binding doc .../{powerpc/fsl/ssi.txt => sound/fsl,ssi.txt} | 10 + .../bindings/sound/phytec,phycore-ac97.txt | 14 + arch/arm/mach-imx/mach-pca100.c | 7 +- arch/arm/mach-imx/mach-pcm043.c | 7 +- sound/soc/fsl/Kconfig | 20 +- sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_ssi.c | 450 +++++++++++++++++---- sound/soc/fsl/fsl_ssi.h | 7 + sound/soc/fsl/imx-pcm-dma.c | 1 - sound/soc/fsl/imx-pcm-fiq.c | 18 +- sound/soc/fsl/imx-pcm.h | 15 +- sound/soc/fsl/imx-ssi.c | 14 +- sound/soc/fsl/imx-ssi.h | 1 + sound/soc/fsl/phycore-ac97-dt.c | 243 +++++++++++ sound/soc/fsl/phycore-ac97.c | 13 +- 15 files changed, 706 insertions(+), 116 deletions(-) rename Documentation/devicetree/bindings/{powerpc/fsl/ssi.txt => sound/fsl,ssi.txt} (88%) create mode 100644 Documentation/devicetree/bindings/sound/phytec,phycore-ac97.txt create mode 100644 sound/soc/fsl/phycore-ac97-dt.c

3 16

[alsa-devel] [PATCH] ASoC: SPEAr: Add device tree support + SPEAr machine driver
by Rajeev Kumar 20 Jun '13

20 Jun '13

The patch adds device tree support + SPEAr machine driver Signed-off-by: Rajeev Kumar <rajeev-dlh.kumar(a)st.com> --- .../devicetree/bindings/sound/spear-evb.txt | 20 +++ sound/soc/spear/spear_evb.c | 135 ++++++++++++++++++++ 2 files changed, 155 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/spear-evb.txt create mode 100644 sound/soc/spear/spear_evb.c diff --git a/Documentation/devicetree/bindings/sound/spear-evb.txt b/Documentation/devicetree/bindings/sound/spear-evb.txt new file mode 100644 index 0000000..cb011b4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/spear-evb.txt @@ -0,0 +1,20 @@ +* SPEAr audio sta529 codec + +Required properties: +- compatible: "spear,spear-evb" +- audio-controllers: The phandle list of synopsys audio i2s controller +- audio-codec: The phandle of the sta529 audio codec +- codec_dai_name: DAI name for codec. +- stream_name: stream name. + +Example: + +sound { + compatible = "spear,spear-evb"; + audio-controllers = <&i2s0 &i2s1>; + audio-codecs = <&sta529 &sta529>; + codec_dai_name = "sta529-audio", "sta529-audio"; + stream_name = "i2s-play", "i2s-cap"; + dai_name = "i2s0-pcm", "i2s1-pcm"; +}; + diff --git a/sound/soc/spear/spear_evb.c b/sound/soc/spear/spear_evb.c new file mode 100644 index 0000000..6da1732 --- /dev/null +++ b/sound/soc/spear/spear_evb.c @@ -0,0 +1,135 @@ +/* + * ASoC machine driver for SPEAr evaluation boards + * + * sound/soc/spear/spear_evb.c + * + * Copyright (C) 2010 ST Microelectronics + * Rajeev Kumar<rajeev-dlh.kumar(a)st.com> + * + * This file is licensed under the terms of the GNU General Public + * License version 2. This program is licensed "as is" without any + * warranty of any kind, whether express or implied. + */ + +#include <linux/io.h> +#include <linux/of.h> +#include <linux/platform_device.h> +#include <linux/module.h> + +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/spear_dma.h> + +struct snd_soc_card spear_soc_card; + +static int spear_evb_probe(struct platform_device *pdev) +{ + struct snd_soc_dai_link *evb_dai; + struct device_node *np = pdev->dev.of_node; + struct device_node *spear_audio_np, *codec_np; + int i, nr_controllers, ret = 0; + + if (!np) + return -EINVAL; /* no device tree */ + + of_property_read_u32(np, "nr_controllers", &nr_controllers); + + evb_dai = devm_kzalloc(&pdev->dev, + (nr_controllers * sizeof(*evb_dai)), GFP_KERNEL); + if (!evb_dai) { + dev_warn(&pdev->dev, "kzalloc fail for evb_dai\n"); + return -ENOMEM; + } + + for (i = 0; i < nr_controllers ; i++) { + spear_audio_np = of_parse_phandle(np, "audio-controllers", i); + codec_np = of_parse_phandle(np, "audio-codecs", i); + if (!spear_audio_np || !codec_np) { + dev_err(&pdev->dev, "phandle missing or invalid\n"); + return -EINVAL; + } + + ret = of_property_read_string_index(np, "dai_names", i, + &evb_dai[i].name); + if (ret < 0) { + dev_err(&pdev->dev, "Cannot parse names: %d\n", ret); + goto err; + } + + ret = of_property_read_string_index(np, "codec_dai_names", i, + &evb_dai[i].codec_dai_name); + if (ret < 0) { + dev_err(&pdev->dev, "Can't parse codec-dai-names: %d\n", + ret); + goto err; + } + ret = of_property_read_string_index(np, "stream_names", i, + &evb_dai[i].stream_name); + if (ret < 0) { + dev_err(&pdev->dev, "Cannot parse stream names: %d\n", + ret); + goto err; + } + + evb_dai[i].cpu_dai_name = NULL; + evb_dai[i].cpu_of_node = spear_audio_np; + evb_dai[i].platform_of_node = np; + evb_dai[i].codec_name = NULL; + evb_dai[i].codec_of_node = codec_np; + if (!(strcmp(evb_dai[i].codec_dai_name, "sta529-audio"))) + evb_dai[i].dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_CBS_CFM; + of_node_put(spear_audio_np); + of_node_put(codec_np); + } + + spear_soc_card.name = "spear-evb"; + spear_soc_card.dai_link = evb_dai; + spear_soc_card.num_links = nr_controllers; + + spear_soc_card.dev = &pdev->dev; + platform_set_drvdata(pdev, &spear_soc_card); + + ret = snd_soc_register_card(&spear_soc_card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed: %d\n", ret); + goto err; + } + + return 0; +err: + return ret; +} + +static int spear_evb_remove(struct platform_device *pdev) +{ + struct snd_soc_card *spear_soc_card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(spear_soc_card); + + return 0; +} + +static const struct of_device_id spear_evb_dt_ids[] = { + { .compatible = "spear,spear-evb", }, + { } +}; +MODULE_DEVICE_TABLE(of, spear_evb_dt_ids); + +static struct platform_driver spear_evb_driver = { + .driver = { + .name = "spear-evb", + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + .of_match_table = spear_evb_dt_ids, + }, + .probe = spear_evb_probe, + .remove = spear_evb_remove, +}; + +module_platform_driver(spear_evb_driver); + +MODULE_AUTHOR("Rajeev Kumar <rajeev-dlh.kumar(a)st.com>"); +MODULE_DESCRIPTION("ST SPEAr EVB ASoC driver"); +MODULE_LICENSE("GPL"); -- 1.6.0.2

2 1

[alsa-devel] [PATCH] sound: using strlcpy instead of strncpy
by Zhao Hongjiang 20 Jun '13

20 Jun '13

for NUL terminated string, need alway set '\0' in the end. Signed-off-by: Zhao Hongjiang <zhaohongjiang(a)huawei.com> --- sound/oss/soundcard.c | 2 +- sound/soc/codecs/88pm860x-codec.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/oss/soundcard.c b/sound/oss/soundcard.c index e778034..53ac987 100644 --- a/sound/oss/soundcard.c +++ b/sound/oss/soundcard.c @@ -99,7 +99,7 @@ int *load_mixer_volumes(char *name, int *levels, int present) } n = num_mixer_volumes++; - strncpy(mixer_vols[n].name, name, 32); + strlcpy(mixer_vols[n].name, name, sizeof(mixer_vols[n].name)); if (present) mixer_vols[n].num = n; diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 60159c0..6b2e6b3 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -1447,7 +1447,7 @@ static int pm860x_codec_probe(struct platform_device *pdev) goto out; } pm860x->irq[i] = res->start + chip->irq_base; - strncpy(pm860x->name[i], res->name, MAX_NAME_LEN); + strlcpy(pm860x->name[i], res->name, sizeof(pm860x->name[i])); } ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_pm860x, -- 1.7.1

2 1

[alsa-devel] [PATCH] ALSA: hda - Make Thinkpad X220-tablet use generic parser
by David Henningsson 20 Jun '13

20 Jun '13

Like the X220, this quirk was added to support docking station, so enable the fixup instead. According to Jan, the generic parser works equal or better than the current parser. This was tested under a 3.9 kernel. Reported-by: Jan Alexander Steffens <jan.steffens(a)gmail.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index b314d3e..de00ce1 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -2947,7 +2947,6 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS), - SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo U350", CXT5066_ASUS), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS), {} @@ -3318,6 +3317,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x21ce, "Lenovo T420", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT_PINCFG_LENOVO_TP410), + SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC), -- 1.7.9.5

2 1

[alsa-devel] [alsa-ucm] segfault at __list_del function in list.h
by sajeesh sidharthan 20 Jun '13

20 Jun '13

Hello. I am testing pulseaudio ucm. I have faced a crash when active ucm device list empty. Please let me know if following fix can be checked in alsa-lib-1.0.25/include/list.h old code: static __inline__ void list_del(struct list_head *entry) { __list_del(entry->prev, entry->next); } new code: static __inline__ void list_del(struct list_head *entry) { if(entry->prev && entry->next) __list_del(entry->prev, entry->next); } Sajeesh

1 0

Re: [alsa-devel] [RFC PATCH 00/11] ARM: s3c64xx: Let amba-pl08x driver handle DMA
by Mark Brown 20 Jun '13

20 Jun '13

On Wed, Jun 19, 2013 at 09:32:44PM +0200, Tomasz Figa wrote: > On Wednesday 19 of June 2013 20:22:11 Mark Brown wrote: > > No, I didn't - that's most likely it, I didn't really investigate. I > > didn't test the watchdog stuff as the clocks didn't get sent to me. > I always try to keep you on Cc of my patches for s3c64xx, as you are the > most active user of this platform (if not the only one other than me) and > this was the case for clock patches as well, just checked that. Yes, I had the clock patches but they depended on something or other I think. I did test a previous verion IIRC. There are other users I'm aware of (you can probably take a wild guess where from the board) though they tend not to be terribly active upstream.

1 0

Re: [alsa-devel] [RFC PATCH 11/11] ARM: s3c64xx: Add support for DMA using generic amba-pl08x driver
by Linus Walleij 19 Jun '13

19 Jun '13

On Wed, Jun 19, 2013 at 8:23 PM, Tomasz Figa <tomasz.figa(a)gmail.com> wrote: > On Monday 17 of June 2013 16:04:47 Linus Walleij wrote: >> On Sun, Jun 16, 2013 at 10:54 PM, Tomasz Figa <tomasz.figa(a)gmail.com> > wrote: >> > +static AMBA_AHB_DEVICE(s3c64xx_dma1, "dma-pl080s.1", 0x0a141080, >> > + 0x75100000, {IRQ_DMA1}, >> > &s3c64xx_dma1_plat_data); >> Ah, does this mean the device does not have any AMBA PrimeCell ID >> contents? > > Hmm, not really. The ID here is what I copied from PrimeCell ID registers > and the method of registration is taken from S5PV210. If I remove the > override, it will read the same value from registers. OK then don't override. Overriding is for hardware which is all zeroes in these registers. >> Then use the ID 0x00053080 instead of 0x0a141080 >> >> 0x53 is "S" for Samsung. Variant 0. > > Well, that's kind of solution to make sure that it won't conflict with any > future variant manufactured by ARM, I guess. Yeah, but as stated, if the hardware has proper IDs (these are probably inspired by the PCI vendor/device/revision fields) then rely on those. Yours, Linus Walleij

1 0

Re: [alsa-devel] [RFC PATCH 00/11] ARM: s3c64xx: Let amba-pl08x driver handle DMA
by Arnd Bergmann 19 Jun '13

19 Jun '13

On Wednesday 19 June 2013, Tomasz Figa wrote: > On Wednesday 19 of June 2013 18:40:47 Mark Brown wrote: > > On Sun, Jun 16, 2013 at 10:54:07PM +0200, Tomasz Figa wrote: > > > One of the biggest roadblocks on the way of S3C64xx to DeviceTree > > > support is its DMA driver, which is completely platform-specific and > > > provides private API (s3c-dma), not even saying that its design is > > > completely against multiplatform-awareness. > > > > I tried to test this on my s3c64xx based system but it gave me a kernel > > that didn't boot far enough to give console output (there's some early > > init stuff that uses SPI...). That said, I needed: > > > > diff --git a/drivers/dma/amba-pl08x.c b/drivers/dma/amba-pl08x.c > > index 210a893..0f49707 100644 > > --- a/drivers/dma/amba-pl08x.c > > +++ b/drivers/dma/amba-pl08x.c > > @@ -313,7 +313,7 @@ static int pl08x_request_mux(struct pl08x_dma_chan > > *plchan) int ret; > > > > if (plchan->mux_use++ == 0 && pd->get_signal) { > > - ret = pd->get_signal(plchan->cd); > > + ret = (pd->get_signal)(plchan->cd); > > Hmm, that's strange. The former is a completely valid piece of code... get_signal is a macro defined in include/linux/signal.h. If that header gets included, neither of the two is valid. Arnd

2 1

[alsa-devel] [PATCH 01/11] ASoC: tlv320aix3x: Use SOC_SINGLE_EXT() instead of open-coding it
by Lars-Peter Clausen 19 Jun '13

19 Jun '13

Cc: Jarkko Nikula <jhnikula(a)gmail.com> Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/tlv320aic3x.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 1514bf8..e5b9268 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -128,10 +128,8 @@ static const u8 aic3x_reg[AIC3X_CACHEREGNUM] = { }; #define SOC_DAPM_SINGLE_AIC3X(xname, reg, shift, mask, invert) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ - .info = snd_soc_info_volsw, \ - .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw_aic3x, \ - .private_value = SOC_SINGLE_VALUE(reg, shift, mask, invert) } + SOC_SINGLE_EXT(xname, reg, shift, mask, invert, \ + snd_soc_dapm_get_volsw, snd_soc_dapm_put_volsw_aic3x) /* * All input lines are connected when !0xf and disconnected with 0xf bit field, -- 1.8.0

2 11

[alsa-devel] [RFC PATCH 00/11] ARM: s3c64xx: Let amba-pl08x driver handle DMA
by Tomasz Figa 19 Jun '13

19 Jun '13

One of the biggest roadblocks on the way of S3C64xx to DeviceTree support is its DMA driver, which is completely platform-specific and provides private API (s3c-dma), not even saying that its design is completely against multiplatform-awareness. The DMA controller present on this SoC series is a custom variant of ARM PrimeCell PL080 modified by Samsung to add some extra features. It is mostly compatible with original PL080, except: - CH_CONTROL2 register is added between CH_CONTROL and CH_CONFIG, - offset of CH_CONFIG register is different, - transfer size field is moved from CH_CONTROL to CH_CONTROL2, - transfer size field is extended to 24 bits, allowing much bigger single transfer, - LLI consists of one more word, to account for CH_CONTROL2 register. Since all the rest is fully compatible with standard PL080 there is no point in having separate driver just for this single variant, so I decided to look into adding support for it to the amba-pl08x driver. There was already some attempt to achieve this before, but this was before Russel's big rework of the driver to use virtual channels, making the old patches being not much of use. This RFC series is a proof of concept that I managed to make during last days of hacking. Except one patch adding clkdev lookup to clock driver (which is being replaced with a CCF-compliant driver ATM), this is enough to get memcpy and slave transfers to work on S3C64xx. I have tested this on Mini6410 and SMDK6410 boards using dmatest for memcpy and Samsung I2S with madplay/aplay for slave transfers. Unfortunately I do not have access to other platforms with PL08x so I could not test for any regressions introduced on them. Credits for two patches go to Alban Bedel, who made a series fixing this driver to make it usable with audio drivers. I rebased his patches on top of mine and corrected coding style a bit. OK, that's all. Any comments are welcome. Feel free to start throwing eggs and tomatoes if you find this awful, but I won't be upset if I get some Tested-by or Acked-by as well. ;) Alban Bedel (2): dmaengine: PL08x: Fix reading the byte count in cctl dmaengine: PL08x: Add cyclic transfer support Tomasz Figa (9): dma: amba-pl08x: Use bitmap to pass variant specific quirks dma: amba-pl08x: Refactor pl08x_getbytes_chan() to lower indentation dma: amba-pl08x: Add support for different offset of CONFIG register dma: amba-pl08x: Add support for PL080S variant dma: amba-pl08x: Add support for different maximum transfer size dma: amba-pl08x: Keep LLIs aligned to 4-word boundary spi: s3c64xx: Do not require legacy DMA API in case of S3C64XX ASoC: samsung: Do not require legacy DMA API in case of S3C64XX ARM: s3c64xx: Add support for DMA using generic amba-pl08x driver arch/arm/Kconfig | 1 + arch/arm/mach-s3c64xx/Kconfig | 8 +- arch/arm/mach-s3c64xx/Makefile | 1 + arch/arm/mach-s3c64xx/common.h | 5 + arch/arm/mach-s3c64xx/include/mach/dma.h | 65 +++++ arch/arm/mach-s3c64xx/pl080.c | 244 +++++++++++++++++++ drivers/dma/amba-pl08x.c | 399 ++++++++++++++++++++++--------- drivers/spi/Kconfig | 2 +- include/linux/amba/pl080.h | 1 + sound/soc/samsung/Kconfig | 2 +- 10 files changed, 613 insertions(+), 115 deletions(-) create mode 100644 arch/arm/mach-s3c64xx/pl080.c -- 1.8.2.1

5 41

Re: [alsa-devel] [PATCH v3] sound/sgtl5000: fix codec register initial values and handling
by Mark Brown 19 Jun '13

19 Jun '13

On Wed, Jun 19, 2013 at 01:16:06PM +0000, Oskar Schirmer wrote: This is a set of three unrelated changes, they should be sent as three separate patches. > Further, mask computation in declarative part is obviously wrong: > Fix FRAC DIVISOR to provide an 11 bit mask correctly. What makes you say that this is obviously wrong? It's not entirely clear. > -#define SGTL5000_PLL_FRAC_DIV_MASK 0x0700 > +#define SGTL5000_PLL_FRAC_DIV_MASK 0x07ff Based on your changelog I was expecting a mask with only two bits set here?

1 0

Re: [alsa-devel] [PATCH] ALSA: firewire: fix error return code in scs_probe()
by Takashi Iwai 18 Jun '13

18 Jun '13

At Tue, 18 Jun 2013 21:09:42 +0800, Wei Yongjun wrote: > > From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> > > Fix to return -ENOMEM in the kmalloc() error handling > case instead of 0, as done elsewhere in this function. > > Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Thanks, applied with ack by Clemens. Takashi > --- > sound/firewire/scs1x.c | 4 +++- > 1 file changed, 3 insertions(+), 1 deletion(-) > > diff --git a/sound/firewire/scs1x.c b/sound/firewire/scs1x.c > index 844a555..b252c21 100644 > --- a/sound/firewire/scs1x.c > +++ b/sound/firewire/scs1x.c > @@ -405,8 +405,10 @@ static int scs_probe(struct device *unit_dev) > scs->output_idle = true; > > scs->buffer = kmalloc(HSS1394_MAX_PACKET_SIZE, GFP_KERNEL); > - if (!scs->buffer) > + if (!scs->buffer) { > + err = -ENOMEM; > goto err_card; > + } > > scs->hss_handler.length = HSS1394_MAX_PACKET_SIZE; > scs->hss_handler.address_callback = handle_hss; >

1 0

[alsa-devel] [PATCH V1] ALSA: hda - Avoid choose same converter for unused pins
by Wang Xingchao 18 Jun '13

18 Jun '13

For Intel Haswell HDMI codecs, the pins choose converter 0 by default. This would cause conflict when playing audio on unused pins,the pin with physical device connected would get audio data too. i.e. Pin 0/1/2 default choose converter 0, pin 1 has HDMI monitor connected. when play audio on Pin 0 or pin 2, pin 1 could get audio data too. This patch configure unused pins to choose different converter. Signed-off-by: Wang Xingchao <xingchao.wang(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 91 ++++++++++++++++++++++++++++++++++++-------- 1 file changed, 76 insertions(+), 15 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 8983747..43f8b76 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1110,26 +1110,15 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, return 0; } -/* - * HDA PCM callbacks - */ -static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) +static int hdmi_choose_cvt(struct hda_codec *codec, + int pin_idx, int *cvt_id, int *mux_id) { struct hdmi_spec *spec = codec->spec; - struct snd_pcm_runtime *runtime = substream->runtime; - int pin_idx, cvt_idx, mux_idx = 0; struct hdmi_spec_per_pin *per_pin; - struct hdmi_eld *eld; struct hdmi_spec_per_cvt *per_cvt = NULL; + int cvt_idx, mux_idx = 0; - /* Validate hinfo */ - pin_idx = hinfo_to_pin_index(spec, hinfo); - if (snd_BUG_ON(pin_idx < 0)) - return -EINVAL; per_pin = get_pin(spec, pin_idx); - eld = &per_pin->sink_eld; /* Dynamically assign converter to stream */ for (cvt_idx = 0; cvt_idx < spec->num_cvts; cvt_idx++) { @@ -1147,17 +1136,89 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, continue; break; } + /* No free converters */ if (cvt_idx == spec->num_cvts) return -ENODEV; + if (cvt_id) + *cvt_id = cvt_idx; + if (mux_id) + *mux_id = mux_idx; + + return 0; +} + +static void haswell_config_cvts(struct hda_codec *codec, + int pin_id, int mux_id) +{ + struct hdmi_spec *spec = codec->spec; + struct hdmi_spec_per_pin *per_pin; + int pin_idx, mux_idx; + int curr; + int err; + + for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { + per_pin = get_pin(spec, pin_idx); + + if (pin_idx == pin_id) + continue; + + curr = snd_hda_codec_read(codec, per_pin->pin_nid, 0, + AC_VERB_GET_CONNECT_SEL, 0); + + /* Choose another unused converter */ + if (curr == mux_id) { + err = hdmi_choose_cvt(codec, pin_idx, NULL, &mux_idx); + if (err < 0) + return; + snd_printdd("HDMI: choose converter %d for pin %d\n", mux_idx, pin_idx); + snd_hda_codec_write_cache(codec, per_pin->pin_nid, 0, + AC_VERB_SET_CONNECT_SEL, + mux_idx); + } + } +} + +/* + * HDA PCM callbacks + */ +static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct hdmi_spec *spec = codec->spec; + struct snd_pcm_runtime *runtime = substream->runtime; + int pin_idx, cvt_idx, mux_idx = 0; + struct hdmi_spec_per_pin *per_pin; + struct hdmi_eld *eld; + struct hdmi_spec_per_cvt *per_cvt = NULL; + int err; + + /* Validate hinfo */ + pin_idx = hinfo_to_pin_index(spec, hinfo); + if (snd_BUG_ON(pin_idx < 0)) + return -EINVAL; + per_pin = get_pin(spec, pin_idx); + eld = &per_pin->sink_eld; + + err = hdmi_choose_cvt(codec, pin_idx, &cvt_idx, &mux_idx); + if (err < 0) + return err; + + per_cvt = get_cvt(spec, cvt_idx); /* Claim converter */ per_cvt->assigned = 1; hinfo->nid = per_cvt->cvt_nid; - snd_hda_codec_write(codec, per_pin->pin_nid, 0, + snd_hda_codec_write_cache(codec, per_pin->pin_nid, 0, AC_VERB_SET_CONNECT_SEL, mux_idx); + + /* configure unused pins to choose other converters */ + if (codec->vendor_id == 0x80862807) + haswell_config_cvts(codec, pin_idx, mux_idx); + snd_hda_spdif_ctls_assign(codec, pin_idx, per_cvt->cvt_nid); /* Initially set the converter's capabilities */ -- 1.7.9.5

3 2

Re: [alsa-devel] [PATCH] ALSA: firewire: fix error return code in scs_probe()
by Clemens Ladisch 18 Jun '13

18 Jun '13

Wei Yongjun wrote: > Fix to return -ENOMEM in the kmalloc() error handling > case instead of 0, as done elsewhere in this function. > > Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Acked-by: Clemens Ladisch <clemens(a)ladisch.de> > scs->buffer = kmalloc(HSS1394_MAX_PACKET_SIZE, GFP_KERNEL); > - if (!scs->buffer) > + if (!scs->buffer) { > + err = -ENOMEM; > goto err_card; > + }

1 0

[alsa-devel] [PATCH] ALSA: hda - Avoid choose same converter for unused pins
by Wang Xingchao 18 Jun '13

18 Jun '13

For Intel Haswell HDMI codecs, the pins choose converter 0 by default. This would cause conflict when playing audio on unused pins,the pin with physical device connected would get audio data too. i.e. Pin 0/1/2 default choose converter 0, pin 1 has HDMI monitor connected. when play audio on Pin 0 or pin 2, pin 1 could get audio data too. This patch configure unused pins to choose different converter. Signed-off-by: Wang Xingchao <xingchao.wang(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 89 +++++++++++++++++++++++++++++++++++++------- 1 file changed, 75 insertions(+), 14 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 8983747..e887584 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1110,26 +1110,15 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, return 0; } -/* - * HDA PCM callbacks - */ -static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) +static int hdmi_choose_cvt(struct hda_codec *codec, + int pin_idx, int *cvt_id, int *mux_id) { struct hdmi_spec *spec = codec->spec; - struct snd_pcm_runtime *runtime = substream->runtime; - int pin_idx, cvt_idx, mux_idx = 0; struct hdmi_spec_per_pin *per_pin; - struct hdmi_eld *eld; struct hdmi_spec_per_cvt *per_cvt = NULL; + int cvt_idx, mux_idx = 0; - /* Validate hinfo */ - pin_idx = hinfo_to_pin_index(spec, hinfo); - if (snd_BUG_ON(pin_idx < 0)) - return -EINVAL; per_pin = get_pin(spec, pin_idx); - eld = &per_pin->sink_eld; /* Dynamically assign converter to stream */ for (cvt_idx = 0; cvt_idx < spec->num_cvts; cvt_idx++) { @@ -1147,10 +1136,77 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, continue; break; } + /* No free converters */ if (cvt_idx == spec->num_cvts) return -ENODEV; + if (cvt_id) + *cvt_id = cvt_idx; + if (mux_id) + *mux_id = mux_idx; + + return 0; +} + +static void haswell_config_cvts(struct hda_codec *codec, + int pin_id, int mux_id) +{ + struct hdmi_spec *spec = codec->spec; + struct hdmi_spec_per_pin *per_pin; + int pin_idx, mux_idx; + int curr = -1; + int err; + + for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { + per_pin = get_pin(spec, pin_idx); + + if (pin_idx == pin_id) + continue; + + curr = snd_hda_codec_read(codec, per_pin->pin_nid, 0, + AC_VERB_GET_CONNECT_SEL, 0); + + /* Choose another unused converter */ + if (curr == mux_id) { + err = hdmi_choose_cvt(codec, pin_idx, NULL, &mux_idx); + if (err < 0) + return; + snd_printk("HDMI: choose converter %d for pin %d\n", mux_idx, pin_idx); + snd_hda_codec_write(codec, per_pin->pin_nid, 0, + AC_VERB_SET_CONNECT_SEL, + mux_idx); + } + } +} + +/* + * HDA PCM callbacks + */ +static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct hdmi_spec *spec = codec->spec; + struct snd_pcm_runtime *runtime = substream->runtime; + int pin_idx, cvt_idx, mux_idx = 0; + struct hdmi_spec_per_pin *per_pin; + struct hdmi_eld *eld; + struct hdmi_spec_per_cvt *per_cvt = NULL; + int err; + + /* Validate hinfo */ + pin_idx = hinfo_to_pin_index(spec, hinfo); + if (snd_BUG_ON(pin_idx < 0)) + return -EINVAL; + per_pin = get_pin(spec, pin_idx); + eld = &per_pin->sink_eld; + + err = hdmi_choose_cvt(codec, pin_idx, &cvt_idx, &mux_idx); + if (err < 0) + return err; + + per_cvt = get_cvt(spec, cvt_idx); /* Claim converter */ per_cvt->assigned = 1; hinfo->nid = per_cvt->cvt_nid; @@ -1158,6 +1214,11 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, snd_hda_codec_write(codec, per_pin->pin_nid, 0, AC_VERB_SET_CONNECT_SEL, mux_idx); + + /* configure unused pins to choose other converters */ + if (codec->vendor_id == 0x80862807) + haswell_config_cvts(codec, pin_idx, mux_idx); + snd_hda_spdif_ctls_assign(codec, pin_idx, per_cvt->cvt_nid); /* Initially set the converter's capabilities */ -- 1.7.9.5

4 4

[alsa-devel] [PATCH 0/4] Haswell Display audio routing bug fix
by Wang Xingchao 18 Jun '13

18 Jun '13

This patchset used to fix some display audio issues on Haswell platform. I tested this patch on Haswell ult board, C0 stepping, with eDP pannel, HDMI monitor, DP monitor. The fixed issues: 1) when only HDMI or DP monitor connected, hear sound on ALL three HDMI devices. 2) when HDMI + DP monitors connected, hear sound on HDMI and DP at the same time no matter you're playing audio on DP or HDMI device. 3) When DP + HDMI + eDP connected, no sound could be heard. 4) After bootup, play audio on HDMI device 7(use second pin), no sound. If play audio on device 3 at first(use first pin), it has sound again. There's quite different hardware changes for haswell display audio, compared with previous Ivybridge/Sandybridge. The common hd-a driver and hdmi driver need make improvement to support. Haswell chip supports Dp1.2 feature(see ref 1). This is very cool feture windows had supported. We get confirm linux gfx team has plan to support Dp1.2 in future too. We're ready to support dp1.2 feature in audio side, as the converter selection dependency is fixed in this patchset. Also we need some improvement in HDA driver side because Haswell added new Verb to support DP1.2. here’s a video show about DP1.2 feature http://www.club-3d.com/index.php/products/reader.en/product/mst-hub-1-3.html Please apply the patches based on Daniel's drm-intel-next-queued branch: http://cgit.freedesktop.org/~danvet/drm-intel/log/?h=drm-intel-next-queued last commit 80e83831a64b9a5d49e844691037b2d4be0f14f9 Please feel free to let me know the issues you meet during test. Wang Xingchao (4): ALSA: hda - Haswell converter power state D0 verify ALSA: hda - Return error when open empty hdmi device drm/i915: Add display audio routing APIs for ALSA ALSA: hda - Add display audio routing API for haswell drivers/gpu/drm/i915/i915_drv.h | 18 +++++ drivers/gpu/drm/i915/intel_ddi.c | 131 +++++++++++++++++++++++++++++++++-- drivers/gpu/drm/i915/intel_display.c | 7 +- drivers/gpu/drm/i915/intel_drv.h | 1 + include/drm/i915_powerwell.h | 5 ++ sound/pci/hda/hda_i915.c | 83 ++++++++++++++++++++++ sound/pci/hda/hda_i915.h | 4 ++ sound/pci/hda/patch_hdmi.c | 37 ++++++++++ 8 files changed, 281 insertions(+), 5 deletions(-) -- 1.8.1.2

6 17

[alsa-devel] [PATCH] ALSA: hda - Haswell converter power state D0 verify
by Wang Xingchao 18 Jun '13

18 Jun '13

Haswell converters maybe in wrong power state before usage. i.e. only converter 0 is in D0, converter 1/2 are in D3. When pin choose converter 1/2, there's no audio output, this cause dependency when playing differnt stream on pins. AUD_PWRST ConvertorA_Widget_Power_State_Current D0 AUD_PWRST ConvertorA_Widget_Power_State_Requsted D0 AUD_PWRST ConvertorB_Widget_Power_State_Current D3 AUD_PWRST ConvertorB_Widget_Power_State_Requested D3 AUD_PWRST ConvC_Widget_PwrSt_Curr D3 AUD_PWRST ConvC_Widget_PwrSt_Req D3 This patch check converter's power state and set D0 if it's in D3 mode. Signed-off-by: Wang Xingchao <xingchao.wang(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index e12f7a0..760eeb1 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1018,10 +1018,19 @@ static void hdmi_unsol_event(struct hda_codec *codec, unsigned int res) hdmi_non_intrinsic_event(codec, res); } -static void haswell_verify_pin_D0(struct hda_codec *codec, hda_nid_t nid) +static void haswell_verify_pin_D0(struct hda_codec *codec, + hda_nid_t cvt_nid, hda_nid_t nid) { int pwr, lamp, ramp; + /* For Haswell, the converter 1/2 may keep in D3 state after bootup, + * thus pins could only choose converter 0 for use. Make sure the + * converters are in correct power state */ + pwr = snd_hda_codec_read(codec, cvt_nid, 0, AC_VERB_GET_POWER_STATE, 0); + pwr = (pwr & AC_PWRST_ACTUAL) >> AC_PWRST_ACTUAL_SHIFT; + if (pwr != AC_PWRST_D0) + snd_hda_codec_write(codec, cvt_nid, 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + pwr = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_POWER_STATE, 0); pwr = (pwr & AC_PWRST_ACTUAL) >> AC_PWRST_ACTUAL_SHIFT; if (pwr != AC_PWRST_D0) { @@ -1068,7 +1077,7 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, int new_pinctl = 0; if (codec->vendor_id == 0x80862807) - haswell_verify_pin_D0(codec, pin_nid); + haswell_verify_pin_D0(codec, cvt_nid, pin_nid); if (snd_hda_query_pin_caps(codec, pin_nid) & AC_PINCAP_HBR) { pinctl = snd_hda_codec_read(codec, pin_nid, 0, -- 1.8.1.2

2 1

[alsa-devel] [PATCH v10 0/2] ARM: davinci: move EDMA API to arm/common
by Sekhar Nori 18 Jun '13

18 Jun '13

This is a repost of first two patches of Matt Porter's EDMA series below: http://www.mail-archive.com/davinci-linux-open-source@linux.davincidsp.com/… The only change that has been made is to drop changes related to DaVinci SFFSDR machine support for sound since that code is now queued to be removed. Thanks, Sekhar Matt Porter (2): ARM: davinci: move private EDMA API to arm/common ARM: edma: remove unused transfer controller handlers arch/arm/Kconfig | 1 + arch/arm/common/Kconfig | 3 + arch/arm/common/Makefile | 1 + arch/arm/{mach-davinci/dma.c => common/edma.c} | 39 +-------- arch/arm/mach-davinci/Makefile | 2 +- arch/arm/mach-davinci/board-tnetv107x-evm.c | 2 +- arch/arm/mach-davinci/davinci.h | 2 +- arch/arm/mach-davinci/devices-tnetv107x.c | 2 +- arch/arm/mach-davinci/devices.c | 6 +- arch/arm/mach-davinci/dm355.c | 2 +- arch/arm/mach-davinci/dm365.c | 2 +- arch/arm/mach-davinci/dm644x.c | 2 +- arch/arm/mach-davinci/dm646x.c | 2 +- arch/arm/mach-davinci/include/mach/da8xx.h | 2 +- drivers/dma/edma.c | 2 +- drivers/mmc/host/davinci_mmc.c | 1 + include/linux/mfd/davinci_voicecodec.h | 3 +- .../mach => include/linux/platform_data}/edma.h | 89 +------------------- include/linux/platform_data/spi-davinci.h | 2 +- sound/soc/davinci/davinci-evm.c | 1 + sound/soc/davinci/davinci-pcm.c | 1 + sound/soc/davinci/davinci-pcm.h | 2 +- 22 files changed, 29 insertions(+), 140 deletions(-) rename arch/arm/{mach-davinci/dma.c => common/edma.c} (97%) rename {arch/arm/mach-davinci/include/mach => include/linux/platform_data}/edma.h (59%) -- 1.7.10.1

4 7

[alsa-devel] [PATCH 1/2] ASoC: spear: Remove nonexistant EVM options
by Mark Brown 18 Jun '13

18 Jun '13

The source wasn't added. Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/spear/Kconfig | 21 --------------------- sound/soc/spear/Makefile | 3 --- 2 files changed, 24 deletions(-) diff --git a/sound/soc/spear/Kconfig b/sound/soc/spear/Kconfig index e3851d6..3567d73 100644 --- a/sound/soc/spear/Kconfig +++ b/sound/soc/spear/Kconfig @@ -1,24 +1,3 @@ -config SND_SPEAR_EVM - tristate "SoC Audio support for SPEAr EVM" - select SND_DESIGNWARE_I2S - select SND_SOC_STA529 - select SND_SPEAR_SOC - help - Say Y if you want to add support for SoC audio on SPEAr - platform - -config SND_SPEAR1340_EVM - tristate "SoC Audio support for SPEAr1340 EVM" - select SND_DESIGNWARE_I2S - select SND_SOC_STA529 - select SND_SPEAR_SPDIF_OUT - select SND_SPEAR_SPDIF_IN - select SND_SOC_SPDIF - select SND_SPEAR_SOC - help - Say Y if you want to add support for SoC audio on SPEAr1340 - platform - config SND_SPEAR_SOC tristate select SND_SOC_DMAENGINE_PCM diff --git a/sound/soc/spear/Makefile b/sound/soc/spear/Makefile index b365126..9f47bb4 100644 --- a/sound/soc/spear/Makefile +++ b/sound/soc/spear/Makefile @@ -2,6 +2,3 @@ obj-$(CONFIG_SND_SPEAR_SOC) += spear_pcm.o obj-$(CONFIG_SND_SPEAR_SPDIF_IN) += spdif_in.o obj-$(CONFIG_SND_SPEAR_SPDIF_OUT) += spdif_out.o - -# SPEAR Machine Support -obj-$(CONFIG_SND_SPEAR_EVM) += spear_evb.o -- 1.7.10.4

2 3

[alsa-devel] VIA VT1708S mic booast problem
by kbd_server＠rbcmail.ru 18 Jun '13

18 Jun '13

Dears, Please, check bug report - https://bugzilla.kernel.org/show_bug.cgi?id=59861 In 3.9.5-030905-generic version - Rear (Front) Mic Boost Volume option is absent for VIA VT1708S codec. 3.8.0-19-generic - everything OK. -- Почтовый ящик предоставлен сервисом qip.ru Заведи бесплатную почту в домене rbcmail.ru и ты!

1 0

Re: [alsa-devel] [PATCH 1/3 v2] mfd: arizona: Integrate wm8997 into Arizona mfd
by Samuel Ortiz 18 Jun '13

18 Jun '13

Hi Charles, On Thu, Jun 13, 2013 at 09:43:29AM +0100, Charles Keepax wrote: > The wm8997 is a compact, high-performance audio hub CODEC with SLIMbus > interfacing, for smartphones, tablets and other portable audio devices > based on the Arizona platform. > > This patch integrates the wm8997 into the Arizona mfd. > > Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> > --- > drivers/mfd/Kconfig | 6 + > drivers/mfd/Makefile | 3 + > drivers/mfd/arizona-core.c | 27 + > drivers/mfd/arizona-i2c.c | 6 + > drivers/mfd/arizona-irq.c | 8 + > drivers/mfd/arizona.h | 5 + > drivers/mfd/wm8997-tables.c | 1525 ++++++++++++++++++++++++++++++++++++++ > include/linux/mfd/arizona/core.h | 2 + > 8 files changed, 1582 insertions(+), 0 deletions(-) > create mode 100644 drivers/mfd/wm8997-tables.c Applied to mfd-next, thanks. Cheers, Samuel. -- Intel Open Source Technology Centre http://oss.intel.com/

1 0

[alsa-devel] [PATCH 1/2] mfd: wm5102: Expose DRE control registers
by Charles Keepax 18 Jun '13

18 Jun '13

Certain use cases may require specific DRE settings so expose the necessary registers. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- drivers/mfd/wm5102-tables.c | 9 ++++++- include/linux/mfd/arizona/registers.h | 44 +++++++++++++++++++++++++++++++++ 2 files changed, 52 insertions(+), 1 deletions(-) diff --git a/drivers/mfd/wm5102-tables.c b/drivers/mfd/wm5102-tables.c index 155c4a1..802dd3c 100644 --- a/drivers/mfd/wm5102-tables.c +++ b/drivers/mfd/wm5102-tables.c @@ -65,7 +65,8 @@ static const struct reg_default wm5102_revb_patch[] = { { 0x418, 0xa080 }, { 0x420, 0xa080 }, { 0x428, 0xe000 }, - { 0x443, 0xDC1A }, + { 0x442, 0x3F0A }, + { 0x443, 0xDC1F }, { 0x4B0, 0x0066 }, { 0x458, 0x000b }, { 0x212, 0x0000 }, @@ -424,6 +425,9 @@ static const struct reg_default wm5102_reg_default[] = { { 0x00000435, 0x0180 }, /* R1077 - DAC Digital Volume 5R */ { 0x00000436, 0x0081 }, /* R1078 - DAC Volume Limit 5R */ { 0x00000437, 0x0200 }, /* R1079 - Noise Gate Select 5R */ + { 0x00000440, 0x8FFF }, /* R1088 - DRE Enable */ + { 0x00000442, 0x3F0A }, /* R1090 - DRE Control 2 */ + { 0x00000443, 0xDC1F }, /* R1090 - DRE Control 3 */ { 0x00000450, 0x0000 }, /* R1104 - DAC AEC Control 1 */ { 0x00000458, 0x000B }, /* R1112 - Noise Gate Control */ { 0x00000490, 0x0069 }, /* R1168 - PDM SPK1 CTRL 1 */ @@ -1197,6 +1201,9 @@ static bool wm5102_readable_register(struct device *dev, unsigned int reg) case ARIZONA_DAC_DIGITAL_VOLUME_5R: case ARIZONA_DAC_VOLUME_LIMIT_5R: case ARIZONA_NOISE_GATE_SELECT_5R: + case ARIZONA_DRE_ENABLE: + case ARIZONA_DRE_CONTROL_2: + case ARIZONA_DRE_CONTROL_3: case ARIZONA_DAC_AEC_CONTROL_1: case ARIZONA_NOISE_GATE_CONTROL: case ARIZONA_PDM_SPK1_CTRL_1: diff --git a/include/linux/mfd/arizona/registers.h b/include/linux/mfd/arizona/registers.h index 4730b5c..4706d3d 100644 --- a/include/linux/mfd/arizona/registers.h +++ b/include/linux/mfd/arizona/registers.h @@ -215,6 +215,9 @@ #define ARIZONA_DAC_DIGITAL_VOLUME_6R 0x43D #define ARIZONA_DAC_VOLUME_LIMIT_6R 0x43E #define ARIZONA_NOISE_GATE_SELECT_6R 0x43F +#define ARIZONA_DRE_ENABLE 0x440 +#define ARIZONA_DRE_CONTROL_2 0x442 +#define ARIZONA_DRE_CONTROL_3 0x443 #define ARIZONA_DAC_AEC_CONTROL_1 0x450 #define ARIZONA_NOISE_GATE_CONTROL 0x458 #define ARIZONA_PDM_SPK1_CTRL_1 0x490 @@ -3133,6 +3136,47 @@ #define ARIZONA_OUT6R_NGATE_SRC_WIDTH 12 /* OUT6R_NGATE_SRC - [11:0] */ /* + * R1088 (0x440) - DRE Enable + */ +#define ARIZONA_DRE3L_ENA 0x0010 /* DRE3L_ENA */ +#define ARIZONA_DRE3L_ENA_MASK 0x0010 /* DRE3L_ENA */ +#define ARIZONA_DRE3L_ENA_SHIFT 4 /* DRE3L_ENA */ +#define ARIZONA_DRE3L_ENA_WIDTH 1 /* DRE3L_ENA */ +#define ARIZONA_DRE2R_ENA 0x0008 /* DRE2R_ENA */ +#define ARIZONA_DRE2R_ENA_MASK 0x0008 /* DRE2R_ENA */ +#define ARIZONA_DRE2R_ENA_SHIFT 3 /* DRE2R_ENA */ +#define ARIZONA_DRE2R_ENA_WIDTH 1 /* DRE2R_ENA */ +#define ARIZONA_DRE2L_ENA 0x0004 /* DRE2L_ENA */ +#define ARIZONA_DRE2L_ENA_MASK 0x0004 /* DRE2L_ENA */ +#define ARIZONA_DRE2L_ENA_SHIFT 2 /* DRE2L_ENA */ +#define ARIZONA_DRE2L_ENA_WIDTH 1 /* DRE2L_ENA */ +#define ARIZONA_DRE1R_ENA 0x0002 /* DRE1R_ENA */ +#define ARIZONA_DRE1R_ENA_MASK 0x0002 /* DRE1R_ENA */ +#define ARIZONA_DRE1R_ENA_SHIFT 1 /* DRE1R_ENA */ +#define ARIZONA_DRE1R_ENA_WIDTH 1 /* DRE1R_ENA */ +#define ARIZONA_DRE1L_ENA 0x0001 /* DRE1L_ENA */ +#define ARIZONA_DRE1L_ENA_MASK 0x0001 /* DRE1L_ENA */ +#define ARIZONA_DRE1L_ENA_SHIFT 0 /* DRE1L_ENA */ +#define ARIZONA_DRE1L_ENA_WIDTH 1 /* DRE1L_ENA */ + +/* + * R1090 (0x442) - DRE Control 2 + */ +#define ARIZONA_DRE_T_LOW_MASK 0x3F00 /* DRE_T_LOW - [13:8] */ +#define ARIZONA_DRE_T_LOW_SHIFT 8 /* DRE_T_LOW - [13:8] */ +#define ARIZONA_DRE_T_LOW_WIDTH 6 /* DRE_T_LOW - [13:8] */ + +/* + * R1091 (0x443) - DRE Control 3 + */ +#define ARIZONA_DRE_GAIN_SHIFT_MASK 0xC000 /* DRE_GAIN_SHIFT - [15:14] */ +#define ARIZONA_DRE_GAIN_SHIFT_SHIFT 14 /* DRE_GAIN_SHIFT - [15:14] */ +#define ARIZONA_DRE_GAIN_SHIFT_WIDTH 2 /* DRE_GAIN_SHIFT - [15:14] */ +#define ARIZONA_DRE_LOW_LEVEL_ABS_MASK 0x000F /* LOW_LEVEL_ABS - [3:0] */ +#define ARIZONA_DRE_LOW_LEVEL_ABS_SHIFT 0 /* LOW_LEVEL_ABS - [3:0] */ +#define ARIZONA_DRE_LOW_LEVEL_ABS_WIDTH 4 /* LOW_LEVEL_ABS - [3:0] */ + +/* * R1104 (0x450) - DAC AEC Control 1 */ #define ARIZONA_AEC_LOOPBACK_SRC_MASK 0x003C /* AEC_LOOPBACK_SRC - [5:2] */ -- 1.7.2.5

3 4

[alsa-devel] [PATCH RESEND V2] ARM: dts: Correct the base address of pinctrl_3 on Exynos5250
by Padmavathi Venna 17 Jun '13

17 Jun '13

This patch corrects the base address of pinctrl_3 on Exynos5250 platform. Signed-off-by: Padmavathi Venna <padma.v(a)samsung.com> --- Changes since V1: - Added platform name in the subject line. arch/arm/boot/dts/exynos5250-pinctrl.dtsi | 2 +- arch/arm/boot/dts/exynos5250.dtsi | 4 ++-- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/arch/arm/boot/dts/exynos5250-pinctrl.dtsi b/arch/arm/boot/dts/exynos5250-pinctrl.dtsi index d1650fb..ded558b 100644 --- a/arch/arm/boot/dts/exynos5250-pinctrl.dtsi +++ b/arch/arm/boot/dts/exynos5250-pinctrl.dtsi @@ -763,7 +763,7 @@ }; }; - pinctrl@03680000 { + pinctrl@03860000 { gpz: gpz { gpio-controller; #gpio-cells = <2>; diff --git a/arch/arm/boot/dts/exynos5250.dtsi b/arch/arm/boot/dts/exynos5250.dtsi index 7154e3d..e9bfd13 100644 --- a/arch/arm/boot/dts/exynos5250.dtsi +++ b/arch/arm/boot/dts/exynos5250.dtsi @@ -169,9 +169,9 @@ interrupts = <0 50 0>; }; - pinctrl_3: pinctrl@03680000 { + pinctrl_3: pinctrl@03860000 { compatible = "samsung,exynos5250-pinctrl"; - reg = <0x0368000 0x1000>; + reg = <0x03860000 0x1000>; interrupts = <0 47 0>; }; -- 1.7.4.4

3 5

[alsa-devel] [PATCH V2 12/15] ASoC: spear: Convert to use devm_ioremap_resource
by Tushar Behera 17 Jun '13

17 Jun '13

Commit 75096579c3ac ("lib: devres: Introduce devm_ioremap_resource()") introduced devm_ioremap_resource() and deprecated the use of devm_request_and_ioremap(). devm_request_mem_region is called in devm_ioremap_resource(). Hence that part can also be removed. Since devm_ioremap_resource prints error message on failure, there is no need to print an explicit warning message. Signed-off-by: Tushar Behera <tushar.behera(a)linaro.org> CC: alsa-devel(a)alsa-project.org CC: Liam Girdwood <lgirdwood(a)gmail.com> CC: Mark Brown <broonie(a)kernel.org> --- Changes for V2: * Removed redundant print of error message from devm_ioremap_resource exit path * Call to devm_request_mem_region is removed. sound/soc/spear/spdif_out.c | 20 ++++---------------- 1 file changed, 4 insertions(+), 16 deletions(-) diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c index 4bde512..64db05a 100644 --- a/sound/soc/spear/spdif_out.c +++ b/sound/soc/spear/spdif_out.c @@ -282,28 +282,16 @@ static int spdif_out_probe(struct platform_device *pdev) struct resource *res; int ret; - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) - return -EINVAL; - - if (!devm_request_mem_region(&pdev->dev, res->start, - resource_size(res), pdev->name)) { - dev_warn(&pdev->dev, "Failed to get memory resourse\n"); - return -ENOENT; - } - host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL); if (!host) { dev_warn(&pdev->dev, "kzalloc fail\n"); return -ENOMEM; } - host->io_base = devm_request_and_ioremap(&pdev->dev, res->start, - resource_size(res)); - if (!host->io_base) { - dev_warn(&pdev->dev, "ioremap failed\n"); - return -ENOMEM; - } + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + host->io_base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(host->io_base)) + return PTR_ERR(host->io_base); host->clk = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(host->clk)) -- 1.7.9.5

2 3

[alsa-devel] [PATCH 1/2 (v2)] ALSA: hda - Headset mic support for three more machines
by David Henningsson 17 Jun '13

17 Jun '13

They need these quirks to have headset mic support. BugLink: https://bugs.launchpad.net/bugs/1189363 Tested-by: Shawn Wang <shawn.wang(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 3 +++ 1 file changed, 3 insertions(+) Note that this patch replaces the patch sent yesterday. I added one more ID in this version. diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 02e22b4..af9e71b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3483,6 +3483,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x05ca, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05cb, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05de, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05e0, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05e9, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05ea, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05eb, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), @@ -3494,6 +3495,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x05f5, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05f6, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05f8, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0606, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0608, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0609, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), SND_PCI_QUIRK(0x103c, 0x18e6, "HP", ALC269_FIXUP_HP_GPIO_LED), -- 1.7.9.5

2 7

[alsa-devel] [PATCH] sound/soc/davinci: remove sffsdr machine support
by Sekhar Nori 17 Jun '13

17 Jun '13

sffsdr machine support does not build since at least v2.6.36 (~3 years). There is little hope of it being fixed, so remove the support. Signed-off-by: Sekhar Nori <nsekhar(a)ti.com> --- sound/soc/davinci/Kconfig | 10 -- sound/soc/davinci/Makefile | 2 - sound/soc/davinci/davinci-sffsdr.c | 181 ------------------------------------ 3 files changed, 193 deletions(-) delete mode 100644 sound/soc/davinci/davinci-sffsdr.c diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 9e11a14..c82f89c 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -54,16 +54,6 @@ config SND_DM6467_SOC_EVM help Say Y if you want to add support for SoC audio on TI -config SND_DAVINCI_SOC_SFFSDR - tristate "SoC Audio support for SFFSDR" - depends on SND_DAVINCI_SOC && MACH_SFFSDR - select SND_DAVINCI_SOC_I2S - select SND_SOC_PCM3008 - select SFFSDR_FPGA - help - Say Y if you want to add support for SoC audio on - Lyrtech SFFSDR board. - config SND_DA830_SOC_EVM tristate "SoC Audio support for DA830/OMAP-L137 EVM" depends on SND_DAVINCI_SOC && MACH_DAVINCI_DA830_EVM diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index a93679d..a396ab6 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -11,10 +11,8 @@ obj-$(CONFIG_SND_DAVINCI_SOC_VCIF) += snd-soc-davinci-vcif.o # DAVINCI Machine Support snd-soc-evm-objs := davinci-evm.o -snd-soc-sffsdr-objs := davinci-sffsdr.o obj-$(CONFIG_SND_DAVINCI_SOC_EVM) += snd-soc-evm.o obj-$(CONFIG_SND_DM6467_SOC_EVM) += snd-soc-evm.o obj-$(CONFIG_SND_DA830_SOC_EVM) += snd-soc-evm.o obj-$(CONFIG_SND_DA850_SOC_EVM) += snd-soc-evm.o -obj-$(CONFIG_SND_DAVINCI_SOC_SFFSDR) += snd-soc-sffsdr.o diff --git a/sound/soc/davinci/davinci-sffsdr.c b/sound/soc/davinci/davinci-sffsdr.c deleted file mode 100644 index 5be65aa..0000000 --- a/sound/soc/davinci/davinci-sffsdr.c +++ /dev/null @@ -1,181 +0,0 @@ -/* - * ASoC driver for Lyrtech SFFSDR board. - * - * Author: Hugo Villeneuve - * Copyright (C) 2008 Lyrtech inc - * - * Based on ASoC driver for TI DAVINCI EVM platform, original copyright follow: - * Copyright: (C) 2007 MontaVista Software, Inc., <source(a)mvista.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/timer.h> -#include <linux/interrupt.h> -#include <linux/platform_device.h> -#include <linux/gpio.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> - -#include <asm/dma.h> -#include <asm/mach-types.h> -#ifdef CONFIG_SFFSDR_FPGA -#include <asm/plat-sffsdr/sffsdr-fpga.h> -#endif - -#include <mach/edma.h> - -#include "../codecs/pcm3008.h" -#include "davinci-pcm.h" -#include "davinci-i2s.h" - -/* - * CLKX and CLKR are the inputs for the Sample Rate Generator. - * FSX and FSR are outputs, driven by the sample Rate Generator. - */ -#define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \ - SND_SOC_DAIFMT_CBM_CFS | \ - SND_SOC_DAIFMT_IB_NF) - -static int sffsdr_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 *cpu_dai = rtd->cpu_dai; - int fs; - int ret = 0; - - /* Fsref can be 32000, 44100 or 48000. */ - fs = params_rate(params); - -#ifndef CONFIG_SFFSDR_FPGA - /* Without the FPGA module, the Fs is fixed at 44100 Hz */ - if (fs != 44100) { - pr_debug("warning: only 44.1 kHz is supported without SFFSDR FPGA module\n"); - return -EINVAL; - } -#endif - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, AUDIO_FORMAT); - if (ret < 0) - return ret; - - pr_debug("sffsdr_hw_params: rate = %d Hz\n", fs); - -#ifndef CONFIG_SFFSDR_FPGA - return 0; -#else - return sffsdr_fpga_set_codec_fs(fs); -#endif -} - -static struct snd_soc_ops sffsdr_ops = { - .hw_params = sffsdr_hw_params, -}; - -/* davinci-sffsdr digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link sffsdr_dai = { - .name = "PCM3008", /* Codec name */ - .stream_name = "PCM3008 HiFi", - .cpu_dai_name = "davinci-mcbsp", - .codec_dai_name = "pcm3008-hifi", - .codec_name = "pcm3008-codec", - .platform_name = "davinci-mcbsp", - .ops = &sffsdr_ops, -}; - -/* davinci-sffsdr audio machine driver */ -static struct snd_soc_card snd_soc_sffsdr = { - .name = "DaVinci SFFSDR", - .owner = THIS_MODULE, - .dai_link = &sffsdr_dai, - .num_links = 1, -}; - -/* sffsdr audio private data */ -static struct pcm3008_setup_data sffsdr_pcm3008_setup = { - .dem0_pin = GPIO(45), - .dem1_pin = GPIO(46), - .pdad_pin = GPIO(47), - .pdda_pin = GPIO(38), -}; - -struct platform_device pcm3008_codec = { - .name = "pcm3008-codec", - .id = 0, - .dev = { - .platform_data = &sffsdr_pcm3008_setup, - }, -}; - -static struct resource sffsdr_snd_resources[] = { - { - .start = DAVINCI_MCBSP_BASE, - .end = DAVINCI_MCBSP_BASE + SZ_8K - 1, - .flags = IORESOURCE_MEM, - }, -}; - -static struct evm_snd_platform_data sffsdr_snd_data = { - .tx_dma_ch = DAVINCI_DMA_MCBSP_TX, - .rx_dma_ch = DAVINCI_DMA_MCBSP_RX, -}; - -static struct platform_device *sffsdr_snd_device; - -static int __init sffsdr_init(void) -{ - int ret; - - if (!machine_is_sffsdr()) - return -EINVAL; - - platform_device_register(&pcm3008_codec); - - sffsdr_snd_device = platform_device_alloc("soc-audio", 0); - if (!sffsdr_snd_device) { - printk(KERN_ERR "platform device allocation failed\n"); - return -ENOMEM; - } - - platform_set_drvdata(sffsdr_snd_device, &snd_soc_sffsdr); - platform_device_add_data(sffsdr_snd_device, &sffsdr_snd_data, - sizeof(sffsdr_snd_data)); - - ret = platform_device_add_resources(sffsdr_snd_device, - sffsdr_snd_resources, - ARRAY_SIZE(sffsdr_snd_resources)); - if (ret) { - printk(KERN_ERR "platform device add resources failed\n"); - goto error; - } - - ret = platform_device_add(sffsdr_snd_device); - if (ret) - goto error; - - return ret; - -error: - platform_device_put(sffsdr_snd_device); - return ret; -} - -static void __exit sffsdr_exit(void) -{ - platform_device_unregister(sffsdr_snd_device); - platform_device_unregister(&pcm3008_codec); -} - -module_init(sffsdr_init); -module_exit(sffsdr_exit); - -MODULE_AUTHOR("Hugo Villeneuve"); -MODULE_DESCRIPTION("Lyrtech SFFSDR ASoC driver"); -MODULE_LICENSE("GPL"); -- 1.7.10.1

2 2

[alsa-devel] [PATCH 1/3] ALSA: hda - Headset mic support for three more machines
by David Henningsson 17 Jun '13

17 Jun '13

They need these quirks to have headset mic support. BugLink: https://bugs.launchpad.net/bugs/1189363 Tested-by: Shawn Wang <shawn.wang(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d7a77f1..f5b1e51 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3493,6 +3493,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x05ca, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05cb, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05de, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05e0, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05e9, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05ea, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05eb, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), @@ -3504,6 +3505,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x05f5, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05f6, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05f8, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0606, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0608, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0609, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), SND_PCI_QUIRK(0x103c, 0x18e6, "HP", ALC269_FIXUP_HP_GPIO_LED), -- 1.7.9.5

1 2

[alsa-devel] [PATCH 0/3] Add wm8997 driver
by Charles Keepax 17 Jun '13

17 Jun '13

This series of patches adds support for the Wolfson wm8997 audio hub. Thanks, Charles Charles Keepax (3): mfd: arizona: Integrate wm8997 into Arizona mfd ASoC: wm8997: Initial CODEC driver gpio: arizona: Add wm8997 support to probe drivers/gpio/gpio-arizona.c | 1 + drivers/mfd/Kconfig | 6 + drivers/mfd/Makefile | 3 + drivers/mfd/arizona-core.c | 26 + drivers/mfd/arizona-i2c.c | 6 + drivers/mfd/arizona-irq.c | 8 + drivers/mfd/arizona.h | 5 + drivers/mfd/wm8997-tables.c | 1525 ++++++++++++++++++++++++++++++++++++++ include/linux/mfd/arizona/core.h | 2 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/arizona.c | 13 +- sound/soc/codecs/wm8997.c | 1008 +++++++++++++++++++++++++ sound/soc/codecs/wm8997.h | 23 + 14 files changed, 2631 insertions(+), 3 deletions(-) create mode 100644 drivers/mfd/wm8997-tables.c create mode 100644 sound/soc/codecs/wm8997.c create mode 100644 sound/soc/codecs/wm8997.h -- 1.7.2.5

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[alsa-devel] [patch] ALSA: usx2y: remove some old dead code
by Dan Carpenter 17 Jun '13

17 Jun '13

USB_QUEUE_BULK isn't defined any more. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index b376532..4967fe9 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -695,9 +695,6 @@ static int usX2Y_rate_set(struct usX2Ydev *usX2Y, int rate) ((char*)(usbdata + i))[1] = ra[i].c2; usb_fill_bulk_urb(us->urb[i], usX2Y->dev, usb_sndbulkpipe(usX2Y->dev, 4), usbdata + i, 2, i_usX2Y_04Int, usX2Y); -#ifdef OLD_USB - us->urb[i]->transfer_flags = USB_QUEUE_BULK; -#endif } us->submitted = 0; us->len = NOOF_SETRATE_URBS;

2 1

[alsa-devel] ALSA: usb-audio: work around Anroid accessory firmware bug
by Clemens Ladisch 17 Jun '13

17 Jun '13

When the Android firmware enables the audio interfaces in accessory mode, it always declares in the control interface's baInterfaceNr array that interfaces 0 and 1 belong to the audio function. However, the accessory interface itself, if also enabled, already is at index 0 and shifts the actual audio interface numbers to 1 and 2, which prevents the PCM streaming interface from being seen by the host driver. To get the PCM interface interface to work, detect when the descriptors point to the (for this driver useless) accessory interface, and redirect to the correct one. Reported-by: Jeremy Rosen <jeremy.rosen(a)openwide.fr> Tested-by: Jeremy Rosen <jeremy.rosen(a)openwide.fr> Cc: <stable(a)vger.kernel.org> Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> --- sound/usb/card.c | 22 ++++++++++++++++++++-- 1 file changed, 20 insertions(+), 2 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 563854a..5a5153c 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -148,14 +148,32 @@ static int snd_usb_create_stream(struct snd_usb_audio *chip, int ctrlif, int int return -EINVAL; } + alts = &iface->altsetting[0]; + altsd = get_iface_desc(alts); + + /* + * Android with both accessory and audio interfaces enabled gets the + * interface numbers wrong. + */ + if ((chip->usb_id == USB_ID(0x18d1, 0x2d04) || + chip->usb_id == USB_ID(0x18d1, 0x2d05)) && + interface == 0 && + altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC && + altsd->bInterfaceSubClass == USB_SUBCLASS_VENDOR_SPEC) { + interface = 2; + iface = usb_ifnum_to_if(dev, interface); + if (!iface) + return -EINVAL; + alts = &iface->altsetting[0]; + altsd = get_iface_desc(alts); + } + if (usb_interface_claimed(iface)) { snd_printdd(KERN_INFO "%d:%d:%d: skipping, already claimed\n", dev->devnum, ctrlif, interface); return -EINVAL; } - alts = &iface->altsetting[0]; - altsd = get_iface_desc(alts); if ((altsd->bInterfaceClass == USB_CLASS_AUDIO || altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC) && altsd->bInterfaceSubClass == USB_SUBCLASS_MIDISTREAMING) {

2 1

[alsa-devel] sound/oss/ad1848.c:2867:32: warning: ‘id_table’ defined but not used
by Tommi Rantala 17 Jun '13

17 Jun '13

Hello, I'm seeing the following compiler warning, does it look useful? I'm doing a v3.10-rc5 x86-64 build. /build/linux/sound/oss/ad1848.c:2867:32: warning: ‘id_table’ defined but not used [-Wunused-variable] $ gcc --version gcc (Ubuntu/Linaro 4.7.3-1ubuntu1) 4.7.3 Copyright (C) 2012 Free Software Foundation, Inc. This is free software; see the source for copying conditions. There is NO warranty; not even for MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. config: http://fpaste.org/18443/13711363/ Tommi

2 1

[alsa-devel] [stable] usb-audio validation fixes [3.0, 3.2]
by Ben Hutchings 17 Jun '13

17 Jun '13

It looks these fixes are suitable for inclusion in the 3.0.y and 3.2.y stable branches: commit 4fa0e81b83503900be277e6273a79651b375e288 Author: Xi Wang <xi.wang(a)gmail.com> Date: Sun Jan 8 09:02:52 2012 -0500 ALSA: usb-audio: fix possible hang and overflow in parse_uac2_sample_rate_range() commit 8866f405efd4171f9d9c91901d2dd02f01bacb60 Author: Xi Wang <xi.wang(a)gmail.com> Date: Tue Feb 14 05:18:48 2012 -0500 ALSA: usb-audio: avoid integer overflow in create_fixed_stream_quirk() Do you agree? Ben. -- Ben Hutchings If at first you don't succeed, you're doing about average.

2 3

[alsa-devel] Timestamps using I/O Plugin
by Johan Hellman 17 Jun '13

17 Jun '13

I was excited when I found that it was possible to write user space drivers for ALSA, but I can't get my head around how timestamping works when using a I/O plugin instance. I better start explaining what we want to do. Basically we just want to transfer raw audio between the HOST running ALSA and one or more small(tiny) network connected MCUs with audio codecs. We need to synchronize the different network sources with the local HW sources (on the ALSA HOST). Starting looking into how timestamps is handled by ALSA I could not find how it was possible to set the timestamp, there seems to be some differences compared to using a normal kernel driver!? If I understand it correctly timestamp can be obtained by calling snd_pcm_status() but the ioplugs implementation of the .status() newer fills in the ->tstamp only the ->trigger_tstamp, so how can I set the timestamp of the data my plugin delivers to ALSA? Additionally snd_pcm_htimestamp() only seams to return the current monotonic clock at the point the function was called for an ioplug instance!? I saw that there are plugins for pulseaudio and jack doing what we whant to achieve but because of the footprint of the MCU we can't use PA/Jack on these MCU:s. But how have they solved this issue or isn't it an issue for their usecase? /JohanHellman

2 5

[alsa-devel] [PATCH 1/2] ASoC: SPEAr: Add dependency on dmaengine helpers
by Mark Brown 15 Jun '13

15 Jun '13

I'd be very surprised if anyone has used the Kconfig... Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/spear/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/spear/Kconfig b/sound/soc/spear/Kconfig index 3b7cdad..5b744ed 100644 --- a/sound/soc/spear/Kconfig +++ b/sound/soc/spear/Kconfig @@ -27,6 +27,7 @@ config SND_SPEAR_SOC Say Y or M if you want to add support for any of the audio controllers (I2S/SPDIF). You will also need to select the audio interface codecs to support below. + select SND_SOC_DMAENGINE_PCM config SND_SPEAR_SPDIF_OUT tristate "SPEAr SPDIF Out Device Driver" -- 1.7.10.4

3 4

[alsa-devel] [PATCH] ASoC: wm5110: Correct rate control for DSP4
by Charles Keepax 14 Jun '13

14 Jun '13

Reported-by: Dennis May <dennis.may(a)wolfsonmicro.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 3470b64..0a3c7f7 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -279,7 +279,7 @@ static const struct soc_enum wm_adsp2_rate_enum[] = { ARIZONA_DSP1_RATE_SHIFT, 0xf, ARIZONA_RATE_ENUM_SIZE, arizona_rate_text, arizona_rate_val), - SOC_VALUE_ENUM_SINGLE(ARIZONA_DSP3_CONTROL_1, + SOC_VALUE_ENUM_SINGLE(ARIZONA_DSP4_CONTROL_1, ARIZONA_DSP1_RATE_SHIFT, 0xf, ARIZONA_RATE_ENUM_SIZE, arizona_rate_text, arizona_rate_val), -- 1.7.2.5

2 1

[alsa-devel] [ALSA] Codec Registration after soc_probe ()
by Rajan Vaja 14 Jun '13

14 Jun '13

Hi, My codec is registered after SOC init is done (sdp4430_init()). In soc init, ALSA cannot find codec registered (given in SOC dai) and hence it defers sound card registration. Later on codec is registered successfully but sound card registration is not done again. Is there any way to register codec later? I am using kernel version 3.4.0 with OMAP Panda. Thanks, Rajan

1 0

[alsa-devel] Raspberry Pi + Bluetooth Headset + Broadcom adapter
by Alexander Winnig 14 Jun '13

14 Jun '13

Hello. I want to give you a short summary of what I'm doing and what I'm trying to achieve: I have a Bluetooth headset "Iqua BHS-611" ( http://www.iqua.com/sites/default/files/product/iqua_slim_4.png ) and a Belkin USB adapter( http://www.amazon.co.uk/Belkin-USB-4-0-Bluetooth-Adapter/dp/B009IQB3US ). I can hear A2DP-transmitted audio on my headset. But when it comes to using the microphone I record zero bytes on a regular basis using parecord -r -d bluez_source.<whatever mac address> test.wav No error output, just a zero byte test.wav. Trying to use paplay to play a file(I tried 8000 Hz, 44,100 Hz files I created using another application, which contained audio data) also doesn't work using the HSP. Using the A2DP again, there is no problem. On your " contact us" - page ( http://bluetooth-alsa.sourceforge.net/contact.html ) you wrote > You might be able to get a broadcom adapter to work but it may require > that you boot with a kernel argument (the force_scofix parameter of > the hci_usb driver?) to > force sco mtu values I do have a Broadcom adapter, the belkin one. But I am using the raspberry pi, which only has a cmdline.txt for kernel parameters. I added force_scofix=1 but that didn't change anything. So now I am back to > dwc_otg.lpm_enable=0 console=ttyAMA0,115200 kgdboc=ttyAMA0,115200 > console=tty1 root=/dev/mmcblk0p2 rootfstype=ext4 elevator=deadline > rootwait for cmdline.txt. How can I make sure, that the reason is really the broadcom adapter, so I won't buy a new one in vain? After all A2DP is working. PS: Plugging in the Bluetooth adapter into an USB port of a computer that is running Windows 7 and installing the drivers that were included on the included CD-ROM, I can use all functionality of the headset, I can use the microphone just as a normal microphone in the microphone's list, I can use the stereo audio output, everything works. This is the hcidump for the recording attempt: HCI sniffer - Bluetooth packet analyzer ver 2.4 device: hci0 snap_len: 1028 filter: 0xffffffff > HCI Event: Connect Request (0x04) plen 10 bdaddr 00:1A:7D:11:C0:66 class 0x240404 type ACL < HCI Command: Accept Connection Request (0x01|0x0009) plen 7 bdaddr 00:1A:7D:11:C0:66 role 0x00 Role: Master > HCI Event: Command Status (0x0f) plen 4 Accept Connection Request (0x01|0x0009) status 0x00 ncmd 1 > HCI Event: Role Change (0x12) plen 8 status 0x00 bdaddr 00:1A:7D:11:C0:66 role 0x00 Role: Master > HCI Event: Connect Complete (0x03) plen 11 status 0x00 handle 11 bdaddr 00:1A:7D:11:C0:66 type ACL encrypt 0x00 < HCI Command: Read Remote Supported Features (0x01|0x001b) plen 2 handle 11 > HCI Event: Command Status (0x0f) plen 4 Read Remote Supported Features (0x01|0x001b) status 0x00 ncmd 1 > HCI Event: Max Slots Change (0x1b) plen 3 handle 11 slots 5 > HCI Event: Read Remote Supported Features (0x0b) plen 11 status 0x00 handle 11 Features: 0xff 0xfe 0x0f 0xc6 0x8b 0xef 0x59 0x83 < HCI Command: Read Remote Extended Features (0x01|0x001c) plen 3 handle 11 page 1 > HCI Event: Command Status (0x0f) plen 4 Read Remote Extended Features (0x01|0x001c) status 0x00 ncmd 1 > HCI Event: Read Remote Extended Features (0x23) plen 13 status 0x00 handle 11 page 1 max 1 Features: 0x01 0x00 0x00 0x00 0x00 0x00 0x00 0x00 < HCI Command: Remote Name Request (0x01|0x0019) plen 10 bdaddr 00:1A:7D:11:C0:66 mode 2 clkoffset 0x0000 > ACL data: handle 11 flags 0x02 dlen 12 L2CAP(s): Connect req: psm 1 scid 0x0040 > HCI Event: Command Status (0x0f) plen 4 Remote Name Request (0x01|0x0019) status 0x00 ncmd 1 < ACL data: handle 11 flags 0x00 dlen 16 L2CAP(s): Connect rsp: dcid 0x0040 scid 0x0040 result 1 status 0 Connection pending - No futher information available < ACL data: handle 11 flags 0x00 dlen 10 L2CAP(s): Info req: type 2 > HCI Event: Number of Completed Packets (0x13) plen 5 handle 11 packets 2 > ACL data: handle 11 flags 0x02 dlen 16 L2CAP(s): Info rsp: type 2 result 0 Extended feature mask 0x0000 < ACL data: handle 11 flags 0x00 dlen 16 L2CAP(s): Connect rsp: dcid 0x0040 scid 0x0040 result 0 status 0 Connection successful < ACL data: handle 11 flags 0x00 dlen 12 L2CAP(s): Config req: dcid 0x0040 flags 0x00 clen 0 > ACL data: handle 11 flags 0x02 dlen 16 L2CAP(s): Config req: dcid 0x0040 flags 0x00 clen 4 MTU 48 < ACL data: handle 11 flags 0x00 dlen 18 L2CAP(s): Config rsp: scid 0x0040 flags 0x00 result 0 clen 4 MTU 48 > ACL data: handle 11 flags 0x02 dlen 14 L2CAP(s): Config rsp: scid 0x0040 flags 0x00 result 0 clen 0 Success > ACL data: handle 11 flags 0x02 dlen 24 L2CAP(d): cid 0x0040 len 20 [psm 1] SDP SSA Req: tid 0x1 len 0xf pat uuid-32 0x111f (Handsfree AG) max 38 aid(s) 0x0004 (ProtocolDescList) cont 00 > HCI Event: Remote Name Req Complete (0x07) plen 255 status 0x00 bdaddr 00:1A:7D:11:C0:66 name 'Iqua Slim' > HCI Event: Number of Completed Packets (0x13) plen 5 handle 11 packets 2 < ACL data: handle 11 flags 0x00 dlen 33 L2CAP(d): cid 0x0040 len 29 [psm 1] SDP SSA Rsp: tid 0x1 len 0x18 count 21 record #0 aid 0x0004 (ProtocolDescList) < < uuid-16 0x0100 (L2CAP) > < uuid-16 0x0003 (RFCOMM) uint 0xd > > cont 00 > HCI Event: Number of Completed Packets (0x13) plen 5 handle 11 packets 2 > ACL data: handle 11 flags 0x02 dlen 12 L2CAP(s): Disconn req: dcid 0x0040 scid 0x0040 < ACL data: handle 11 flags 0x00 dlen 12 L2CAP(s): Disconn rsp: dcid 0x0040 scid 0x0040 > HCI Event: Link Key Request (0x17) plen 6 bdaddr 00:1A:7D:11:C0:66 < HCI Command: Link Key Request Reply (0x01|0x000b) plen 22 bdaddr 00:1A:7D:11:C0:66 key 4118887B24F8519BEBF20B626C48413E > HCI Event: Command Complete (0x0e) plen 10 Link Key Request Reply (0x01|0x000b) ncmd 1 status 0x00 bdaddr 00:1A:7D:11:C0:66 > HCI Event: Encrypt Change (0x08) plen 4 status 0x00 handle 11 encrypt 0x01 > ACL data: handle 11 flags 0x02 dlen 12 L2CAP(s): Connect req: psm 3 scid 0x0041 < ACL data: handle 11 flags 0x00 dlen 16 L2CAP(s): Connect rsp: dcid 0x0040 scid 0x0041 result 0 status 0 Connection successful < ACL data: handle 11 flags 0x00 dlen 16 L2CAP(s): Config req: dcid 0x0041 flags 0x00 clen 4 MTU 1013 > HCI Event: Number of Completed Packets (0x13) plen 5 handle 11 packets 2 > ACL data: handle 11 flags 0x02 dlen 16 L2CAP(s): Config req: dcid 0x0040 flags 0x00 clen 4 MTU 895 < ACL data: handle 11 flags 0x00 dlen 18 L2CAP(s): Config rsp: scid 0x0041 flags 0x00 result 0 clen 4 MTU 895 > ACL data: handle 11 flags 0x02 dlen 18 L2CAP(s): Config rsp: scid 0x0040 flags 0x00 result 0 clen 4 MTU 1013 > HCI Event: Number of Completed Packets (0x13) plen 5 handle 11 packets 2 > ACL data: handle 11 flags 0x02 dlen 8 L2CAP(d): cid 0x0040 len 4 [psm 3] RFCOMM(s): SABM: cr 1 dlci 0 pf 1 ilen 0 fcs 0x1c < ACL data: handle 11 flags 0x00 dlen 8 L2CAP(d): cid 0x0041 len 4 [psm 3] RFCOMM(s): UA: cr 1 dlci 0 pf 1 ilen 0 fcs 0xd7 > ACL data: handle 11 flags 0x02 dlen 18 L2CAP(d): cid 0x0040 len 14 [psm 3] RFCOMM(s): PN CMD: cr 1 dlci 0 pf 0 ilen 10 fcs 0x70 mcc_len 8 dlci 26 frame_type 0 credit_flow 15 pri 0 ack_timer 0 frame_size 126 max_retrans 0 credits 0 < ACL data: handle 11 flags 0x00 dlen 18 L2CAP(d): cid 0x0041 len 14 [psm 3] RFCOMM(s): PN RSP: cr 0 dlci 0 pf 0 ilen 10 fcs 0xaa mcc_len 8 dlci 26 frame_type 0 credit_flow 14 pri 0 ack_timer 0 frame_size 126 max_retrans 0 credits 7 > HCI Event: Number of Completed Packets (0x13) plen 5 handle 11 packets 2 > ACL data: handle 11 flags 0x02 dlen 8 L2CAP(d): cid 0x0040 len 4 [psm 3] RFCOMM(s): SABM: cr 1 dlci 26 pf 1 ilen 0 fcs 0xe7 < HCI Command: Authentication Requested (0x01|0x0011) plen 2 handle 11 > HCI Event: Command Status (0x0f) plen 4 Authentication Requested (0x01|0x0011) status 0x00 ncmd 1 > HCI Event: Link Key Request (0x17) plen 6 bdaddr 00:1A:7D:11:C0:66 < HCI Command: Link Key Request Reply (0x01|0x000b) plen 22 bdaddr 00:1A:7D:11:C0:66 key 4118887B24F8519BEBF20B626C48413E > HCI Event: Command Complete (0x0e) plen 10 Link Key Request Reply (0x01|0x000b) ncmd 1 status 0x00 bdaddr 00:1A:7D:11:C0:66 > HCI Event: Auth Complete (0x06) plen 3 status 0x00 handle 11 < HCI Command: Set Connection Encryption (0x01|0x0013) plen 3 handle 11 encrypt 0x01 > HCI Event: Command Status (0x0f) plen 4 Set Connection Encryption (0x01|0x0013) status 0x00 ncmd 1 > HCI Event: Encryption Key Refresh Complete (0x30) plen 3 status 0x00 handle 11 > HCI Event: Encrypt Change (0x08) plen 4 status 0x00 handle 11 encrypt 0x01 < ACL data: handle 11 flags 0x00 dlen 8 L2CAP(d): cid 0x0041 len 4 [psm 3] RFCOMM(s): UA: cr 1 dlci 26 pf 1 ilen 0 fcs 0x2c < ACL data: handle 11 flags 0x00 dlen 12 L2CAP(d): cid 0x0041 len 8 [psm 3] RFCOMM(s): MSC CMD: cr 0 dlci 0 pf 0 ilen 4 fcs 0xaa mcc_len 2 dlci 26 fc 0 rtc 1 rtr 1 ic 0 dv 1 b1 1 b2 0 b3 0 len 6 > HCI Event: Number of Completed Packets (0x13) plen 5 handle 11 packets 2 > ACL data: handle 11 flags 0x02 dlen 12 L2CAP(d): cid 0x0040 len 8 [psm 3] RFCOMM(s): MSC CMD: cr 1 dlci 0 pf 0 ilen 4 fcs 0x70 mcc_len 2 dlci 26 fc 0 rtc 1 rtr 1 ic 0 dv 0 b1 1 b2 0 b3 0 len 6 < ACL data: handle 11 flags 0x00 dlen 12 L2CAP(d): cid 0x0041 len 8 [psm 3] RFCOMM(s): MSC RSP: cr 0 dlci 0 pf 0 ilen 4 fcs 0xaa mcc_len 2 dlci 26 fc 0 rtc 1 rtr 1 ic 0 dv 0 b1 1 b2 0 b3 0 len 6 > ACL data: handle 11 flags 0x02 dlen 12 L2CAP(d): cid 0x0040 len 8 [psm 3] RFCOMM(s): MSC RSP: cr 1 dlci 0 pf 0 ilen 4 fcs 0x70 mcc_len 2 dlci 26 fc 0 rtc 1 rtr 1 ic 0 dv 1 b1 1 b2 0 b3 0 len 6 < ACL data: handle 11 flags 0x00 dlen 9 L2CAP(d): cid 0x0041 len 5 [psm 3] RFCOMM(d): UIH: cr 0 dlci 26 pf 1 ilen 0 fcs 0x22 credits 33 > ACL data: handle 11 flags 0x02 dlen 9 L2CAP(d): cid 0x0040 len 5 [psm 3] RFCOMM(d): UIH: cr 1 dlci 26 pf 1 ilen 0 fcs 0xf8 credits 15 > HCI Event: Number of Completed Packets (0x13) plen 5 handle 11 packets 2 > ACL data: handle 11 flags 0x02 dlen 19 L2CAP(d): cid 0x0040 len 15 [psm 3] RFCOMM(d): UIH: cr 1 dlci 26 pf 0 ilen 11 fcs 0xe4 < ACL data: handle 11 flags 0x00 dlen 22 L2CAP(d): cid 0x0041 len 18 [psm 3] RFCOMM(d): UIH: cr 0 dlci 26 pf 0 ilen 14 fcs 0x3e < ACL data: handle 11 flags 0x00 dlen 14 L2CAP(d): cid 0x0041 len 10 [psm 3] RFCOMM(d): UIH: cr 0 dlci 26 pf 0 ilen 6 fcs 0x3e > HCI Event: Number of Completed Packets (0x13) plen 5 handle 11 packets 2 > ACL data: handle 11 flags 0x02 dlen 19 L2CAP(d): cid 0x0040 len 15 [psm 3] RFCOMM(d): UIH: cr 1 dlci 26 pf 1 ilen 10 fcs 0xf8 credits 2 < ACL data: handle 11 flags 0x00 dlen 134 L2CAP(d): cid 0x0041 len 130 [psm 3] RFCOMM(d): UIH: cr 0 dlci 26 pf 0 ilen 126 fcs 0x3e < ACL data: handle 11 flags 0x00 dlen 14 L2CAP(d): cid 0x0041 len 10 [psm 3] RFCOMM(d): UIH: cr 0 dlci 26 pf 0 ilen 6 fcs 0x3e < ACL data: handle 11 flags 0x00 dlen 14 L2CAP(d): cid 0x0041 len 10 [psm 3] RFCOMM(d): UIH: cr 0 dlci 26 pf 0 ilen 6 fcs 0x3e > HCI Event: Number of Completed Packets (0x13) plen 5 handle 11 packets 2 > ACL data: handle 11 flags 0x02 dlen 18 L2CAP(d): cid 0x0040 len 14 [psm 3] RFCOMM(d): UIH: cr 1 dlci 26 pf 1 ilen 9 fcs 0xf8 credits 3 < ACL data: handle 11 flags 0x00 dlen 32 L2CAP(d): cid 0x0041 len 28 [psm 3] RFCOMM(d): UIH: cr 0 dlci 26 pf 0 ilen 24 fcs 0x3e < ACL data: handle 11 flags 0x00 dlen 14 L2CAP(d): cid 0x0041 len 10 [psm 3] RFCOMM(d): UIH: cr 0 dlci 26 pf 0 ilen 6 fcs 0x3e > HCI Event: Number of Completed Packets (0x13) plen 5 handle 11 packets 2 > ACL data: handle 11 flags 0x02 dlen 28 L2CAP(d): cid 0x0040 len 24 [psm 3] RFCOMM(d): UIH: cr 1 dlci 26 pf 1 ilen 19 fcs 0xf8 credits 2 < ACL data: handle 11 flags 0x00 dlen 14 L2CAP(d): cid 0x0041 len 10 [psm 3] RFCOMM(d): UIH: cr 0 dlci 26 pf 0 ilen 6 fcs 0x3e > HCI Event: Number of Completed Packets (0x13) plen 5 handle 11 packets 2 > ACL data: handle 11 flags 0x02 dlen 19 L2CAP(d): cid 0x0040 len 15 [psm 3] RFCOMM(d): UIH: cr 1 dlci 26 pf 1 ilen 10 fcs 0xf8 credits 1 < ACL data: handle 11 flags 0x00 dlen 14 L2CAP(d): cid 0x0041 len 10 [psm 3] RFCOMM(d): UIH: cr 0 dlci 26 pf 0 ilen 6 fcs 0x3e > ACL data: handle 11 flags 0x02 dlen 12 L2CAP(s): Connect req: psm 1 scid 0x0042 < ACL data: handle 11 flags 0x00 dlen 16 L2CAP(s): Connect rsp: dcid 0x0041 scid 0x0042 result 0 status 0 Connection successful < ACL data: handle 11 flags 0x00 dlen 12 L2CAP(s): Config req: dcid 0x0042 flags 0x00 clen 0 > HCI Event: Number of Completed Packets (0x13) plen 5 handle 11 packets 2 > ACL data: handle 11 flags 0x02 dlen 16 L2CAP(s): Config req: dcid 0x0041 flags 0x00 clen 4 MTU 48 < ACL data: handle 11 flags 0x00 dlen 18 L2CAP(s): Config rsp: scid 0x0042 flags 0x00 result 0 clen 4 MTU 48 > ACL data: handle 11 flags 0x02 dlen 14 L2CAP(s): Config rsp: scid 0x0041 flags 0x00 result 0 clen 0 Success > HCI Event: Number of Completed Packets (0x13) plen 5 handle 11 packets 2 > ACL data: handle 11 flags 0x02 dlen 19 L2CAP(d): cid 0x0040 len 15 [psm 3] RFCOMM(d): UIH: cr 1 dlci 26 pf 1 ilen 10 fcs 0xf8 credits 1 > ACL data: handle 11 flags 0x02 dlen 24 L2CAP(d): cid 0x0041 len 20 [psm 1] SDP SSA Req: tid 0x1 len 0xf pat uuid-32 0x111f (Handsfree AG) max 38 aid(s) 0x0009 (BTProfileDescList) cont 00 < ACL data: handle 11 flags 0x00 dlen 14 L2CAP(d): cid 0x0041 len 10 [psm 3] RFCOMM(d): UIH: cr 0 dlci 26 pf 0 ilen 6 fcs 0x3e < ACL data: handle 11 flags 0x00 dlen 29 L2CAP(d): cid 0x0042 len 25 [psm 1] SDP SSA Rsp: tid 0x1 len 0x14 count 17 record #0 aid 0x0009 (BTProfileDescList) < < uuid-16 0x111e (Handsfree) uint 0x105 > > cont 00 > HCI Event: Number of Completed Packets (0x13) plen 5 handle 11 packets 2 > ACL data: handle 11 flags 0x02 dlen 12 L2CAP(s): Disconn req: dcid 0x0041 scid 0x0042 < ACL data: handle 11 flags 0x00 dlen 12 L2CAP(s): Disconn rsp: dcid 0x0041 scid 0x0042 > ACL data: handle 11 flags 0x02 dlen 30 L2CAP(d): cid 0x0040 len 26 [psm 3] RFCOMM(d): UIH: cr 1 dlci 26 pf 1 ilen 21 fcs 0xf8 credits 1 < ACL data: handle 11 flags 0x00 dlen 17 L2CAP(d): cid 0x0041 len 13 [psm 3] RFCOMM(d): UIH: cr 0 dlci 26 pf 0 ilen 9 fcs 0x3e > HCI Event: Number of Completed Packets (0x13) plen 5 handle 11 packets 2 < ACL data: handle 11 flags 0x00 dlen 12 L2CAP(s): Connect req: psm 25 scid 0x0041 > ACL data: handle 11 flags 0x02 dlen 16 L2CAP(s): Connect rsp: dcid 0x0043 scid 0x0041 result 1 status 2 Connection pending - Authorization pending > ACL data: handle 11 flags 0x02 dlen 16 L2CAP(s): Connect rsp: dcid 0x0043 scid 0x0041 result 0 status 0 Connection successful < ACL data: handle 11 flags 0x00 dlen 12 L2CAP(s): Config req: dcid 0x0043 flags 0x00 clen 0 > ACL data: handle 11 flags 0x02 dlen 16 L2CAP(s): Config req: dcid 0x0041 flags 0x00 clen 4 MTU 895 < ACL data: handle 11 flags 0x00 dlen 18 L2CAP(s): Config rsp: scid 0x0043 flags 0x00 result 0 clen 4 MTU 895 > HCI Event: Number of Completed Packets (0x13) plen 5 handle 11 packets 2 > ACL data: handle 11 flags 0x02 dlen 14 L2CAP(s): Config rsp: scid 0x0041 flags 0x00 result 0 clen 0 Success < ACL data: handle 11 flags 0x00 dlen 12 L2CAP(s): Connect req: psm 23 scid 0x0042 < ACL data: handle 11 flags 0x00 dlen 6 L2CAP(d): cid 0x0043 len 2 [psm 25] AVDTP(s): Discover cmd: transaction 0 nsp 0x00 > HCI Event: Number of Completed Packets (0x13) plen 5 handle 11 packets 2 > ACL data: handle 11 flags 0x02 dlen 16 L2CAP(s): Connect rsp: dcid 0x0044 scid 0x0042 result 1 status 2 Connection pending - Authorization pending > HCI Event: Number of Completed Packets (0x13) plen 5 handle 11 packets 1 > ACL data: handle 11 flags 0x02 dlen 8 L2CAP(d): cid 0x0041 len 4 [psm 25] AVDTP(s): Discover rsp: transaction 0 nsp 0x00 ACP SEID 1 - Audio Sink < ACL data: handle 11 flags 0x00 dlen 7 L2CAP(d): cid 0x0043 len 3 [psm 25] AVDTP(s): Capabilities cmd: transaction 1 nsp 0x00 ACP SEID 1 > ACL data: handle 11 flags 0x02 dlen 20 L2CAP(d): cid 0x0041 len 16 [psm 25] AVDTP(s): Capabilities rsp: transaction 1 nsp 0x00 Media Transport Media Codec - SBC 16kHz 32kHz 44.1kHz 48kHz Mono DualChannel Stereo JointStereo 4 8 12 16 Blocks 4 8 Subbands SNR Loudness Bitpool Range 2-53 Content Protection 02 00 > ACL data: handle 11 flags 0x02 dlen 16 L2CAP(s): Connect rsp: dcid 0x0044 scid 0x0042 result 0 status 0 Connection successful < ACL data: handle 11 flags 0x00 dlen 12 L2CAP(s): Config req: dcid 0x0044 flags 0x00 clen 0 > ACL data: handle 11 flags 0x02 dlen 16 L2CAP(s): Config req: dcid 0x0042 flags 0x00 clen 4 MTU 895 < ACL data: handle 11 flags 0x00 dlen 18 L2CAP(s): Config rsp: scid 0x0044 flags 0x00 result 0 clen 4 MTU 895 > HCI Event: Number of Completed Packets (0x13) plen 5 handle 11 packets 2 > ACL data: handle 11 flags 0x02 dlen 14 L2CAP(s): Config rsp: scid 0x0042 flags 0x00 result 0 clen 0 Success < HCI Command: Write Scan Enable (0x03|0x001a) plen 1 enable 0 > HCI Event: Command Complete (0x0e) plen 4 Write Scan Enable (0x03|0x001a) ncmd 1 status 0x00 < HCI Command: Read Scan Enable (0x03|0x0019) plen 0 > HCI Event: Command Complete (0x0e) plen 5 Read Scan Enable (0x03|0x0019) ncmd 1 status 0x00 enable 0 > HCI Event: Number of Completed Packets (0x13) plen 5 handle 11 packets 1 < ACL data: handle 11 flags 0x00 dlen 12 L2CAP(s): Disconn req: dcid 0x0043 scid 0x0041 < ACL data: handle 11 flags 0x00 dlen 12 L2CAP(s): Disconn req: dcid 0x0044 scid 0x0042 > HCI Event: Number of Completed Packets (0x13) plen 5 handle 11 packets 2 > ACL data: handle 11 flags 0x02 dlen 12 L2CAP(s): Disconn rsp: dcid 0x0043 scid 0x0041 > ACL data: handle 11 flags 0x02 dlen 12 L2CAP(s): Disconn rsp: dcid 0x0044 scid 0x0042 < HCI Command: Write Class of Device (0x03|0x0024) plen 3 class 0x4a0100 > HCI Event: Command Complete (0x0e) plen 4 Write Class of Device (0x03|0x0024) ncmd 1 status 0x00 < HCI Command: Write Extended Inquiry Response (0x03|0x0052) plen 241 fec 0x00 Unknown type 0x45 with 8 bytes data Unknown type 0x2d with 2 bytes data > HCI Event: Command Complete (0x0e) plen 4 Write Extended Inquiry Response (0x03|0x0052) ncmd 1 status 0x00 < HCI Command: Write Class of Device (0x03|0x0024) plen 3 class 0x4e0100 > HCI Event: Command Complete (0x0e) plen 4 Write Class of Device (0x03|0x0024) ncmd 1 status 0x00 < HCI Command: Setup Synchronous Connection (0x01|0x0028) plen 17 handle 11 voice setting 0x0060 ptype 0x003f > HCI Event: Command Status (0x0f) plen 4 Setup Synchronous Connection (0x01|0x0028) status 0x00 ncmd 1 > HCI Event: Synchronous Connect Complete (0x2c) plen 17 status 0x00 handle 6 bdaddr 00:1A:7D:11:C0:66 type eSCO Air mode: CVSD < ACL data: handle 11 flags 0x00 dlen 18 L2CAP(d): cid 0x0041 len 14 [psm 3] RFCOMM(d): UIH: cr 0 dlci 26 pf 0 ilen 10 fcs 0x3e < ACL data: handle 11 flags 0x00 dlen 19 L2CAP(d): cid 0x0041 len 15 [psm 3] RFCOMM(d): UIH: cr 0 dlci 26 pf 0 ilen 11 fcs 0x3e < SCO data: handle 6 flags 0x00 dlen 48 < SCO data: handle 6 flags 0x00 dlen 48 < HCI Command: Write Scan Enable (0x03|0x001a) plen 1 enable 3 > HCI Event: Command Complete (0x0e) plen 4 Write Scan Enable (0x03|0x001a) ncmd 1 status 0x00 < HCI Command: Read Scan Enable (0x03|0x0019) plen 0 > HCI Event: Command Complete (0x0e) plen 5 Read Scan Enable (0x03|0x0019) ncmd 1 status 0x00 enable 3 > HCI Event: Mode Change (0x14) plen 6 status 0x00 handle 11 mode 0x02 interval 160 Mode: Sniff -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

1 0

[alsa-devel] SEGFAULT in ALSA lib 1.0.27.1 pcm_direct parser
by khalasa＠piap.pl 14 Jun '13

14 Jun '13

Hi, Fixes a segfault when getgrnam_r() returns 0 and still sets pgrp to NULL when the group simply doesn't exist. --- alsa-lib/src/pcm/pcm_direct.c 2013-05-21 10:48:28.000000000 +0200 +++ alsa-lib/src/pcm/pcm_direct.c 2013-06-14 13:31:28.981013646 +0200 @@ -1636,7 +1636,7 @@ if (buffer == NULL) return -ENOMEM; int st = getgrnam_r(group, &grp, buffer, len, &pgrp); - if (st != 0) { + if (st != 0 || !pgrp) { SNDERR("The field ipc_gid must be a valid group (create group %s)", group); free(buffer); return -EINVAL; -- Krzysztof Halasa Research Institute for Automation and Measurements PIAP Al. Jerozolimskie 202, 02-486 Warsaw, Poland

2 1

[alsa-devel] [PATCH (alsa-utils)] amixer: Print TLV of channel map controls
by David Henningsson 14 Jun '13

14 Jun '13

Previously these were written as "unk-25..." which wasn't very user friendly. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- amixer/amixer.c | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) diff --git a/amixer/amixer.c b/amixer/amixer.c index fe83b49..7727943 100644 --- a/amixer/amixer.c +++ b/amixer/amixer.c @@ -443,6 +443,7 @@ static void decode_tlv(unsigned int spaces, unsigned int *tlv, unsigned int tlv_ unsigned int type = tlv[0]; unsigned int size; unsigned int idx = 0; + const char *chmap_type = NULL; if (tlv_size < 2 * sizeof(unsigned int)) { printf("TLV size error!\n"); @@ -541,6 +542,27 @@ static void decode_tlv(unsigned int spaces, unsigned int *tlv, unsigned int tlv_ } break; #endif +#ifdef SND_CTL_TLVT_CHMAP_FIXED + case SND_CTL_TLVT_CHMAP_FIXED: + chmap_type = "fixed"; + /* Fall through */ + case SND_CTL_TLVT_CHMAP_VAR: + if (!chmap_type) + chmap_type = "variable"; + /* Fall through */ + case SND_CTL_TLVT_CHMAP_PAIRED: + if (!chmap_type) + chmap_type = "paired"; + printf("chmap-%s=", chmap_type); + + while (size > 0) { + printf("%s", snd_pcm_chmap_name(tlv[idx++])); + size -= sizeof(unsigned int); + if (size > 0) + printf(","); + } + break; +#endif default: printf("unk-%i-", type); while (size > 0) { -- 1.8.3

2 4

[alsa-devel] [PATCH v2] ASoC: WM8962: Enable start-up and normal bias after reset in runtime resume
by Nicolin Chen 14 Jun '13

14 Jun '13

This part of bias settings are essential for WM8962 to power up. Without it "wm8962 0-001a: DC servo timed out" might be prompted due to power-up failure that happens to FLL if being used. The driver's also bringing the bias down in the suspend path so it needs to be powered up in the resume path for symmetry. According to dapm_pre_sequence_async(), DAPM would call pm_runtime_get_sync() to let driver finish the bias settings in pm_runtime_resume() before the bias level being set to STANDBY. So no need to worry about disordered settings for VMID of WM8962. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- Changelog: v1->v2 * Revised the commit comment to make it clear. sound/soc/codecs/wm8962.c | 11 +++++++++++ 1 files changed, 11 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 7a7a056..c63b58e 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3773,6 +3773,17 @@ static int wm8962_runtime_resume(struct device *dev) regcache_sync(wm8962->regmap); + regmap_update_bits(wm8962->regmap, WM8962_ANTI_POP, + WM8962_STARTUP_BIAS_ENA | WM8962_VMID_BUF_ENA, + WM8962_STARTUP_BIAS_ENA | WM8962_VMID_BUF_ENA); + + /* Bias enable at 2*5k (fast start-up) */ + regmap_update_bits(wm8962->regmap, WM8962_PWR_MGMT_1, + WM8962_BIAS_ENA | WM8962_VMID_SEL_MASK, + WM8962_BIAS_ENA | 0x180); + + msleep(5); + return 0; } -- 1.7.1

2 1

[alsa-devel] [PATCH 1/5] ASoC: dapm: Add snd_soc_dapm_switch to the power up/down sequence table
by Lars-Peter Clausen 14 Jun '13

14 Jun '13

The power up/down sequence order for DAPM switch widgets is not explicitly initialized, causing them to be run always as the first widget type for both power up and down. Move it to the same position in the sequence as other mixer widget types. Cc: stable(a)vger.kernel.org Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-dapm.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 2201f22..10290c7 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -64,6 +64,7 @@ static int dapm_up_seq[] = { [snd_soc_dapm_virt_mux] = 5, [snd_soc_dapm_value_mux] = 5, [snd_soc_dapm_dac] = 6, + [snd_soc_dapm_switch] = 7, [snd_soc_dapm_mixer] = 7, [snd_soc_dapm_mixer_named_ctl] = 7, [snd_soc_dapm_pga] = 8, @@ -83,6 +84,7 @@ static int dapm_down_seq[] = { [snd_soc_dapm_line] = 2, [snd_soc_dapm_out_drv] = 2, [snd_soc_dapm_pga] = 4, + [snd_soc_dapm_switch] = 5, [snd_soc_dapm_mixer_named_ctl] = 5, [snd_soc_dapm_mixer] = 5, [snd_soc_dapm_dac] = 6, -- 1.8.0

2 5

[alsa-devel] [PATCH] ASoC: WM8962: Enable start-up and normal bias after reset in runtime resume
by Nicolin Chen 14 Jun '13

14 Jun '13

The analogue circuits in WM8962 require a mid-rail analogue reference voltage, VMID, which generator is enabled by STARTUP_BIAS_ENA bit, while VMID_BUF_ENA bit can enable buffered VMID to avoid audible pops. Besides, these circuits also need a bias current enabled by BIAS_ENA. Then set VMID_SEL to 2x5k divider, the fast start-up mode, to power it up. And let set_bias_level() handle its 2x250k for low power standby and 2x50k for normal operation. This patch might be kinda reverting operation against commit 52c0eee3 but after testing with WM8962 CODEC, these configurations during power up should be essential and soc-core would not do any of them. So the codec drvier need to handle it. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- sound/soc/codecs/wm8962.c | 11 +++++++++++ 1 files changed, 11 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 7a7a056..c63b58e 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3773,6 +3773,17 @@ static int wm8962_runtime_resume(struct device *dev) regcache_sync(wm8962->regmap); + regmap_update_bits(wm8962->regmap, WM8962_ANTI_POP, + WM8962_STARTUP_BIAS_ENA | WM8962_VMID_BUF_ENA, + WM8962_STARTUP_BIAS_ENA | WM8962_VMID_BUF_ENA); + + /* Bias enable at 2*5k (fast start-up) */ + regmap_update_bits(wm8962->regmap, WM8962_PWR_MGMT_1, + WM8962_BIAS_ENA | WM8962_VMID_SEL_MASK, + WM8962_BIAS_ENA | 0x180); + + msleep(5); + return 0; } -- 1.7.1

2 6

[alsa-devel] [PATCH] ASoC: Fix double assignment of .owner in struct snd_soc_card
by Emil Goode 14 Jun '13

14 Jun '13

In struct snd_soc_card zylonite .owner is assigned THIS_MODULE twice, remove one of them. Signed-off-by: Emil Goode <emilgoode(a)gmail.com> --- sound/soc/pxa/zylonite.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index ceb6566..db8aadf 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -256,7 +256,6 @@ static struct snd_soc_card zylonite = { .resume_pre = &zylonite_resume_pre, .dai_link = zylonite_dai, .num_links = ARRAY_SIZE(zylonite_dai), - .owner = THIS_MODULE, }; static struct platform_device *zylonite_snd_ac97_device; -- 1.7.10.4

2 1

[alsa-devel] [PATCH] ASoC: WM8962: Remove remaining direct register cache accesses
by Nicolin Chen 14 Jun '13

14 Jun '13

Also fix return values for headphone switch updates. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- sound/soc/codecs/wm8962.c | 23 ++++++++++++----------- 1 files changed, 12 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 40f0e71..a24f6dd 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -1601,7 +1601,6 @@ static int wm8962_put_hp_sw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - u16 *reg_cache = codec->reg_cache; int ret; /* Apply the update (if any) */ @@ -1610,16 +1609,19 @@ static int wm8962_put_hp_sw(struct snd_kcontrol *kcontrol, return 0; /* If the left PGA is enabled hit that VU bit... */ - if (snd_soc_read(codec, WM8962_PWR_MGMT_2) & WM8962_HPOUTL_PGA_ENA) - return snd_soc_write(codec, WM8962_HPOUTL_VOLUME, - reg_cache[WM8962_HPOUTL_VOLUME]); + ret = snd_soc_read(codec, WM8962_PWR_MGMT_2); + if (ret & WM8962_HPOUTL_PGA_ENA) { + snd_soc_write(codec, WM8962_HPOUTL_VOLUME, + snd_soc_read(codec, WM8962_HPOUTL_VOLUME)); + return 1; + } /* ...otherwise the right. The VU is stereo. */ - if (snd_soc_read(codec, WM8962_PWR_MGMT_2) & WM8962_HPOUTR_PGA_ENA) - return snd_soc_write(codec, WM8962_HPOUTR_VOLUME, - reg_cache[WM8962_HPOUTR_VOLUME]); + if (ret & WM8962_HPOUTR_PGA_ENA) + snd_soc_write(codec, WM8962_HPOUTR_VOLUME, + snd_soc_read(codec, WM8962_HPOUTR_VOLUME)); - return 0; + return 1; } /* The VU bits for the speakers are in a different register to the mute @@ -3376,7 +3378,6 @@ static int wm8962_probe(struct snd_soc_codec *codec) int ret; struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); struct wm8962_pdata *pdata = &wm8962->pdata; - u16 *reg_cache = codec->reg_cache; int i, trigger, irq_pol; bool dmicclk, dmicdat; @@ -3433,8 +3434,8 @@ static int wm8962_probe(struct snd_soc_codec *codec) /* Put the speakers into mono mode? */ if (pdata->spk_mono) - reg_cache[WM8962_CLASS_D_CONTROL_2] - |= WM8962_SPK_MONO; + snd_soc_update_bits(codec, WM8962_CLASS_D_CONTROL_2, + WM8962_SPK_MONO_MASK, WM8962_SPK_MONO); /* Micbias setup, detection enable and detection * threasholds. */ -- 1.7.1

2 2

Re: [alsa-devel] [PATCH] ASoC: dmaengine: add runtime status checking in dmaengine_pcm_dma_complete
by Mark Brown 14 Jun '13

14 Jun '13

On Fri, Jun 07, 2013 at 07:57:29PM +0800, Qiao Zhou wrote: > the dmaengine_pcm_dma_complete callback is usually executed after > the dma interrupt, which uses tasklet_schedule, workqueue, or other > method for quick int handler return. > > in some corner case, where pcm stream is released unexpected, like > media server is killed, the runtime parameter will be freed. if it > happens between the t1 and t2 in below chart, then the callback > will try to access members of paramters which is already freed, > and kernel panics. > > to avoid this issue, add runtime checking before other handling in > dmaengine_pcm_dma_complete. if pcm stream is already released, > just ignore the current handling and return. This doesn't seem like a good or robust way of fixing this, if we're tearing down the resources the DMA is using while the DMA is in progress then in the worst case that might include the memory being DMAed and of course there's races if you just check the pointer - the pointer can be checked at the same time as it's being freed (or between the free and the clear). I think we should be either halting the DMA or waiting for it to finish here.

4 8

[alsa-devel] [PATCH] ASoc: si476x: Do not use binary constants
by James Hogan 14 Jun '13

14 Jun '13

Gcc < 4.3 doesn't understand binary constants (0b*): sound/soc/codecs/si476x.c:172:8: error: invalid suffix "b1111110" on integer constant sound/soc/codecs/si476x.c:219:9: error: invalid suffix "b111" on integer constant sound/soc/codecs/si476x.c:219:56: error: invalid suffix "b111" on integer constant So use hexadecimal constants (0x*) instead. Signed-off-by: James Hogan <james.hogan(a)imgtec.com> Cc: Andrey Smirnov <andrey.smirnov(a)convergeddevices.net> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: alsa-devel(a)alsa-project.org --- sound/soc/codecs/si476x.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c index 721587c..73e205c 100644 --- a/sound/soc/codecs/si476x.c +++ b/sound/soc/codecs/si476x.c @@ -38,9 +38,9 @@ enum si476x_digital_io_output_format { SI476X_DIGITAL_IO_SAMPLE_SIZE_SHIFT = 8, }; -#define SI476X_DIGITAL_IO_OUTPUT_WIDTH_MASK ((0b111 << SI476X_DIGITAL_IO_SLOT_SIZE_SHIFT) | \ - (0b111 << SI476X_DIGITAL_IO_SAMPLE_SIZE_SHIFT)) -#define SI476X_DIGITAL_IO_OUTPUT_FORMAT_MASK (0b1111110) +#define SI476X_DIGITAL_IO_OUTPUT_WIDTH_MASK ((0x7 << SI476X_DIGITAL_IO_SLOT_SIZE_SHIFT) | \ + (0x7 << SI476X_DIGITAL_IO_SAMPLE_SIZE_SHIFT)) +#define SI476X_DIGITAL_IO_OUTPUT_FORMAT_MASK (0x7e) enum si476x_daudio_formats { SI476X_DAUDIO_MODE_I2S = (0x0 << 1), -- 1.8.1.2

1 0

[alsa-devel] tascam us800
by someone deleted 14 Jun '13

14 Jun '13

my tascam us800 works great in 3.8.0-21 except for one issue... no usb-mixer here is output of lsusb -v http://studio213.us/usbif.txt here is output of alsa-info.sh http://www.alsa-project.org/db/?f=336a5faa6233bc80a856089e657d80da1d32e081 alsamixer -c 1 shows amixer: Mixer hw:1 load error: Connection timed out dmesg when I do that shows [115350.348159] cannot get ctl value: req = 0x83, wValue = 0x101, wIndex = 0x1f00, type = 4 [115351.348158] cannot get ctl value: req = 0x81, wValue = 0x101, wIndex = 0x1f00, type = 4 [115732.480250] cannot get ctl value: req = 0x83, wValue = 0x101, wIndex = 0x1f00, type = 4 [115733.480241] cannot get ctl value: req = 0x81, wValue = 0x101, wIndex = 0x1f00, type = 4 plugging in the card shows [ 45.550396] usbcore: registered new interface driver snd-usb-audio in dmesg any info on how I might access this usb-mixer built into this card?

1 0

[alsa-devel] [RFC PATCH 2/2] tea575x: Move from sound to media
by Ondrej Zary 13 Jun '13

13 Jun '13

Move tea575x from sound/i2c/other to drivers/media/radio Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- drivers/media/radio/Kconfig | 5 + drivers/media/radio/Makefile | 1 + drivers/media/radio/tea575x.c | 577 +++++++++++++++++++++++++++++++++++++++ sound/i2c/other/Makefile | 2 - sound/i2c/other/tea575x-tuner.c | 577 --------------------------------------- sound/pci/Kconfig | 5 - 6 files changed, 583 insertions(+), 584 deletions(-) create mode 100644 drivers/media/radio/tea575x.c delete mode 100644 sound/i2c/other/tea575x-tuner.c diff --git a/drivers/media/radio/Kconfig b/drivers/media/radio/Kconfig index c0beee2..0f6341d 100644 --- a/drivers/media/radio/Kconfig +++ b/drivers/media/radio/Kconfig @@ -12,6 +12,11 @@ menuconfig RADIO_ADAPTERS if RADIO_ADAPTERS && VIDEO_V4L2 +config RADIO_TEA575X + tristate + depends on SND_FM801_TEA575X_BOOL || SND_ES1968_RADIO || RADIO_SF16FMR2 || RADIO_MAXIRADIO || RADIO_SHARK + default SND_FM801 || SND_ES1968 || RADIO_SF16FMR2 || RADIO_MAXIRADIO || RADIO_SHARK + config RADIO_SI470X bool "Silicon Labs Si470x FM Radio Receiver support" depends on VIDEO_V4L2 diff --git a/drivers/media/radio/Makefile b/drivers/media/radio/Makefile index 0dcdb32..3b64560 100644 --- a/drivers/media/radio/Makefile +++ b/drivers/media/radio/Makefile @@ -32,6 +32,7 @@ obj-$(CONFIG_RADIO_TEF6862) += tef6862.o obj-$(CONFIG_RADIO_TIMBERDALE) += radio-timb.o obj-$(CONFIG_RADIO_WL1273) += radio-wl1273.o obj-$(CONFIG_RADIO_WL128X) += wl128x/ +obj-$(CONFIG_RADIO_TEA575X) += tea575x.o shark2-objs := radio-shark2.o radio-tea5777.o diff --git a/drivers/media/radio/tea575x.c b/drivers/media/radio/tea575x.c new file mode 100644 index 0000000..a1cb208 --- /dev/null +++ b/drivers/media/radio/tea575x.c @@ -0,0 +1,577 @@ +/* + * ALSA driver for TEA5757/5759 Philips AM/FM radio tuner chips + * + * Copyright (c) 2004 Jaroslav Kysela <perex(a)perex.cz> + * + * + * 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 <asm/io.h> +#include <linux/delay.h> +#include <linux/module.h> +#include <linux/init.h> +#include <linux/slab.h> +#include <linux/sched.h> +#include <media/v4l2-device.h> +#include <media/v4l2-dev.h> +#include <media/v4l2-fh.h> +#include <media/v4l2-ioctl.h> +#include <media/v4l2-event.h> +#include <media/tea575x.h> + +MODULE_AUTHOR("Jaroslav Kysela <perex(a)perex.cz>"); +MODULE_DESCRIPTION("Routines for control of TEA5757/5759 Philips AM/FM radio tuner chips"); +MODULE_LICENSE("GPL"); + +/* + * definitions + */ + +#define TEA575X_BIT_SEARCH (1<<24) /* 1 = search action, 0 = tuned */ +#define TEA575X_BIT_UPDOWN (1<<23) /* 0 = search down, 1 = search up */ +#define TEA575X_BIT_MONO (1<<22) /* 0 = stereo, 1 = mono */ +#define TEA575X_BIT_BAND_MASK (3<<20) +#define TEA575X_BIT_BAND_FM (0<<20) +#define TEA575X_BIT_BAND_MW (1<<20) +#define TEA575X_BIT_BAND_LW (2<<20) +#define TEA575X_BIT_BAND_SW (3<<20) +#define TEA575X_BIT_PORT_0 (1<<19) /* user bit */ +#define TEA575X_BIT_PORT_1 (1<<18) /* user bit */ +#define TEA575X_BIT_SEARCH_MASK (3<<16) /* search level */ +#define TEA575X_BIT_SEARCH_5_28 (0<<16) /* FM >5uV, AM >28uV */ +#define TEA575X_BIT_SEARCH_10_40 (1<<16) /* FM >10uV, AM > 40uV */ +#define TEA575X_BIT_SEARCH_30_63 (2<<16) /* FM >30uV, AM > 63uV */ +#define TEA575X_BIT_SEARCH_150_1000 (3<<16) /* FM > 150uV, AM > 1000uV */ +#define TEA575X_BIT_DUMMY (1<<15) /* buffer */ +#define TEA575X_BIT_FREQ_MASK 0x7fff + +enum { BAND_FM, BAND_FM_JAPAN, BAND_AM }; + +static const struct v4l2_frequency_band bands[] = { + { + .type = V4L2_TUNER_RADIO, + .index = 0, + .capability = V4L2_TUNER_CAP_LOW | V4L2_TUNER_CAP_STEREO | + V4L2_TUNER_CAP_FREQ_BANDS, + .rangelow = 87500 * 16, + .rangehigh = 108000 * 16, + .modulation = V4L2_BAND_MODULATION_FM, + }, + { + .type = V4L2_TUNER_RADIO, + .index = 0, + .capability = V4L2_TUNER_CAP_LOW | V4L2_TUNER_CAP_STEREO | + V4L2_TUNER_CAP_FREQ_BANDS, + .rangelow = 76000 * 16, + .rangehigh = 91000 * 16, + .modulation = V4L2_BAND_MODULATION_FM, + }, + { + .type = V4L2_TUNER_RADIO, + .index = 1, + .capability = V4L2_TUNER_CAP_LOW | V4L2_TUNER_CAP_FREQ_BANDS, + .rangelow = 530 * 16, + .rangehigh = 1710 * 16, + .modulation = V4L2_BAND_MODULATION_AM, + }, +}; + +/* + * lowlevel part + */ + +static void snd_tea575x_write(struct snd_tea575x *tea, unsigned int val) +{ + u16 l; + u8 data; + + if (tea->ops->write_val) + return tea->ops->write_val(tea, val); + + tea->ops->set_direction(tea, 1); + udelay(16); + + for (l = 25; l > 0; l--) { + data = (val >> 24) & TEA575X_DATA; + val <<= 1; /* shift data */ + tea->ops->set_pins(tea, data | TEA575X_WREN); + udelay(2); + tea->ops->set_pins(tea, data | TEA575X_WREN | TEA575X_CLK); + udelay(2); + tea->ops->set_pins(tea, data | TEA575X_WREN); + udelay(2); + } + + if (!tea->mute) + tea->ops->set_pins(tea, 0); +} + +static u32 snd_tea575x_read(struct snd_tea575x *tea) +{ + u16 l, rdata; + u32 data = 0; + + if (tea->ops->read_val) + return tea->ops->read_val(tea); + + tea->ops->set_direction(tea, 0); + tea->ops->set_pins(tea, 0); + udelay(16); + + for (l = 24; l--;) { + tea->ops->set_pins(tea, TEA575X_CLK); + udelay(2); + if (!l) + tea->tuned = tea->ops->get_pins(tea) & TEA575X_MOST ? 0 : 1; + tea->ops->set_pins(tea, 0); + udelay(2); + data <<= 1; /* shift data */ + rdata = tea->ops->get_pins(tea); + if (!l) + tea->stereo = (rdata & TEA575X_MOST) ? 0 : 1; + if (rdata & TEA575X_DATA) + data++; + udelay(2); + } + + if (tea->mute) + tea->ops->set_pins(tea, TEA575X_WREN); + + return data; +} + +static u32 snd_tea575x_val_to_freq(struct snd_tea575x *tea, u32 val) +{ + u32 freq = val & TEA575X_BIT_FREQ_MASK; + + if (freq == 0) + return freq; + + switch (tea->band) { + case BAND_FM: + /* freq *= 12.5 */ + freq *= 125; + freq /= 10; + /* crystal fixup */ + freq -= TEA575X_FMIF; + break; + case BAND_FM_JAPAN: + /* freq *= 12.5 */ + freq *= 125; + freq /= 10; + /* crystal fixup */ + freq += TEA575X_FMIF; + break; + case BAND_AM: + /* crystal fixup */ + freq -= TEA575X_AMIF; + break; + } + + return clamp(freq * 16, bands[tea->band].rangelow, + bands[tea->band].rangehigh); /* from kHz */ +} + +static u32 snd_tea575x_get_freq(struct snd_tea575x *tea) +{ + return snd_tea575x_val_to_freq(tea, snd_tea575x_read(tea)); +} + +void snd_tea575x_set_freq(struct snd_tea575x *tea) +{ + u32 freq = tea->freq / 16; /* to kHz */ + u32 band = 0; + + switch (tea->band) { + case BAND_FM: + band = TEA575X_BIT_BAND_FM; + /* crystal fixup */ + freq += TEA575X_FMIF; + /* freq /= 12.5 */ + freq *= 10; + freq /= 125; + break; + case BAND_FM_JAPAN: + band = TEA575X_BIT_BAND_FM; + /* crystal fixup */ + freq -= TEA575X_FMIF; + /* freq /= 12.5 */ + freq *= 10; + freq /= 125; + break; + case BAND_AM: + band = TEA575X_BIT_BAND_MW; + /* crystal fixup */ + freq += TEA575X_AMIF; + break; + } + + tea->val &= ~(TEA575X_BIT_FREQ_MASK | TEA575X_BIT_BAND_MASK); + tea->val |= band; + tea->val |= freq & TEA575X_BIT_FREQ_MASK; + snd_tea575x_write(tea, tea->val); + tea->freq = snd_tea575x_val_to_freq(tea, tea->val); +} + +/* + * Linux Video interface + */ + +static int vidioc_querycap(struct file *file, void *priv, + struct v4l2_capability *v) +{ + struct snd_tea575x *tea = video_drvdata(file); + + strlcpy(v->driver, tea->v4l2_dev->name, sizeof(v->driver)); + strlcpy(v->card, tea->card, sizeof(v->card)); + strlcat(v->card, tea->tea5759 ? " TEA5759" : " TEA5757", sizeof(v->card)); + strlcpy(v->bus_info, tea->bus_info, sizeof(v->bus_info)); + v->device_caps = V4L2_CAP_TUNER | V4L2_CAP_RADIO; + if (!tea->cannot_read_data) + v->device_caps |= V4L2_CAP_HW_FREQ_SEEK; + v->capabilities = v->device_caps | V4L2_CAP_DEVICE_CAPS; + return 0; +} + +static int vidioc_enum_freq_bands(struct file *file, void *priv, + struct v4l2_frequency_band *band) +{ + struct snd_tea575x *tea = video_drvdata(file); + int index; + + if (band->tuner != 0) + return -EINVAL; + + switch (band->index) { + case 0: + if (tea->tea5759) + index = BAND_FM_JAPAN; + else + index = BAND_FM; + break; + case 1: + if (tea->has_am) { + index = BAND_AM; + break; + } + /* Fall through */ + default: + return -EINVAL; + } + + *band = bands[index]; + if (!tea->cannot_read_data) + band->capability |= V4L2_TUNER_CAP_HWSEEK_BOUNDED; + + return 0; +} + +static int vidioc_g_tuner(struct file *file, void *priv, + struct v4l2_tuner *v) +{ + struct snd_tea575x *tea = video_drvdata(file); + struct v4l2_frequency_band band_fm = { 0, }; + + if (v->index > 0) + return -EINVAL; + + snd_tea575x_read(tea); + vidioc_enum_freq_bands(file, priv, &band_fm); + + memset(v, 0, sizeof(*v)); + strlcpy(v->name, tea->has_am ? "FM/AM" : "FM", sizeof(v->name)); + v->type = V4L2_TUNER_RADIO; + v->capability = band_fm.capability; + v->rangelow = tea->has_am ? bands[BAND_AM].rangelow : band_fm.rangelow; + v->rangehigh = band_fm.rangehigh; + v->rxsubchans = tea->stereo ? V4L2_TUNER_SUB_STEREO : V4L2_TUNER_SUB_MONO; + v->audmode = (tea->val & TEA575X_BIT_MONO) ? + V4L2_TUNER_MODE_MONO : V4L2_TUNER_MODE_STEREO; + v->signal = tea->tuned ? 0xffff : 0; + return 0; +} + +static int vidioc_s_tuner(struct file *file, void *priv, + const struct v4l2_tuner *v) +{ + struct snd_tea575x *tea = video_drvdata(file); + u32 orig_val = tea->val; + + if (v->index) + return -EINVAL; + tea->val &= ~TEA575X_BIT_MONO; + if (v->audmode == V4L2_TUNER_MODE_MONO) + tea->val |= TEA575X_BIT_MONO; + /* Only apply changes if currently tuning FM */ + if (tea->band != BAND_AM && tea->val != orig_val) + snd_tea575x_set_freq(tea); + + return 0; +} + +static int vidioc_g_frequency(struct file *file, void *priv, + struct v4l2_frequency *f) +{ + struct snd_tea575x *tea = video_drvdata(file); + + if (f->tuner != 0) + return -EINVAL; + f->type = V4L2_TUNER_RADIO; + f->frequency = tea->freq; + return 0; +} + +static int vidioc_s_frequency(struct file *file, void *priv, + const struct v4l2_frequency *f) +{ + struct snd_tea575x *tea = video_drvdata(file); + + if (f->tuner != 0 || f->type != V4L2_TUNER_RADIO) + return -EINVAL; + + if (tea->has_am && f->frequency < (20000 * 16)) + tea->band = BAND_AM; + else if (tea->tea5759) + tea->band = BAND_FM_JAPAN; + else + tea->band = BAND_FM; + + tea->freq = clamp_t(u32, f->frequency, bands[tea->band].rangelow, + bands[tea->band].rangehigh); + snd_tea575x_set_freq(tea); + return 0; +} + +static int vidioc_s_hw_freq_seek(struct file *file, void *fh, + const struct v4l2_hw_freq_seek *a) +{ + struct snd_tea575x *tea = video_drvdata(file); + unsigned long timeout; + int i, spacing; + + if (tea->cannot_read_data) + return -ENOTTY; + if (a->tuner || a->wrap_around) + return -EINVAL; + + if (file->f_flags & O_NONBLOCK) + return -EWOULDBLOCK; + + if (a->rangelow || a->rangehigh) { + for (i = 0; i < ARRAY_SIZE(bands); i++) { + if ((i == BAND_FM && tea->tea5759) || + (i == BAND_FM_JAPAN && !tea->tea5759) || + (i == BAND_AM && !tea->has_am)) + continue; + if (bands[i].rangelow == a->rangelow && + bands[i].rangehigh == a->rangehigh) + break; + } + if (i == ARRAY_SIZE(bands)) + return -EINVAL; /* No matching band found */ + if (i != tea->band) { + tea->band = i; + tea->freq = clamp(tea->freq, bands[i].rangelow, + bands[i].rangehigh); + snd_tea575x_set_freq(tea); + } + } + + spacing = (tea->band == BAND_AM) ? 5 : 50; /* kHz */ + + /* clear the frequency, HW will fill it in */ + tea->val &= ~TEA575X_BIT_FREQ_MASK; + tea->val |= TEA575X_BIT_SEARCH; + if (a->seek_upward) + tea->val |= TEA575X_BIT_UPDOWN; + else + tea->val &= ~TEA575X_BIT_UPDOWN; + snd_tea575x_write(tea, tea->val); + timeout = jiffies + msecs_to_jiffies(10000); + for (;;) { + if (time_after(jiffies, timeout)) + break; + if (schedule_timeout_interruptible(msecs_to_jiffies(10))) { + /* some signal arrived, stop search */ + tea->val &= ~TEA575X_BIT_SEARCH; + snd_tea575x_set_freq(tea); + return -ERESTARTSYS; + } + if (!(snd_tea575x_read(tea) & TEA575X_BIT_SEARCH)) { + u32 freq; + + /* Found a frequency, wait until it can be read */ + for (i = 0; i < 100; i++) { + msleep(10); + freq = snd_tea575x_get_freq(tea); + if (freq) /* available */ + break; + } + if (freq == 0) /* shouldn't happen */ + break; + /* + * if we moved by less than the spacing, or in the + * wrong direction, continue seeking + */ + if (abs(tea->freq - freq) < 16 * spacing || + (a->seek_upward && freq < tea->freq) || + (!a->seek_upward && freq > tea->freq)) { + snd_tea575x_write(tea, tea->val); + continue; + } + tea->freq = freq; + tea->val &= ~TEA575X_BIT_SEARCH; + return 0; + } + } + tea->val &= ~TEA575X_BIT_SEARCH; + snd_tea575x_set_freq(tea); + return -ENODATA; +} + +static int tea575x_s_ctrl(struct v4l2_ctrl *ctrl) +{ + struct snd_tea575x *tea = container_of(ctrl->handler, struct snd_tea575x, ctrl_handler); + + switch (ctrl->id) { + case V4L2_CID_AUDIO_MUTE: + tea->mute = ctrl->val; + snd_tea575x_set_freq(tea); + return 0; + } + + return -EINVAL; +} + +static const struct v4l2_file_operations tea575x_fops = { + .unlocked_ioctl = video_ioctl2, + .open = v4l2_fh_open, + .release = v4l2_fh_release, + .poll = v4l2_ctrl_poll, +}; + +static const struct v4l2_ioctl_ops tea575x_ioctl_ops = { + .vidioc_querycap = vidioc_querycap, + .vidioc_g_tuner = vidioc_g_tuner, + .vidioc_s_tuner = vidioc_s_tuner, + .vidioc_g_frequency = vidioc_g_frequency, + .vidioc_s_frequency = vidioc_s_frequency, + .vidioc_s_hw_freq_seek = vidioc_s_hw_freq_seek, + .vidioc_enum_freq_bands = vidioc_enum_freq_bands, + .vidioc_log_status = v4l2_ctrl_log_status, + .vidioc_subscribe_event = v4l2_ctrl_subscribe_event, + .vidioc_unsubscribe_event = v4l2_event_unsubscribe, +}; + +static const struct video_device tea575x_radio = { + .ioctl_ops = &tea575x_ioctl_ops, + .release = video_device_release_empty, +}; + +static const struct v4l2_ctrl_ops tea575x_ctrl_ops = { + .s_ctrl = tea575x_s_ctrl, +}; + +/* + * initialize all the tea575x chips + */ +int snd_tea575x_init(struct snd_tea575x *tea, struct module *owner) +{ + int retval; + + tea->mute = true; + + /* Not all devices can or know how to read the data back. + Such devices can set cannot_read_data to true. */ + if (!tea->cannot_read_data) { + snd_tea575x_write(tea, 0x55AA); + if (snd_tea575x_read(tea) != 0x55AA) + return -ENODEV; + } + + tea->val = TEA575X_BIT_BAND_FM | TEA575X_BIT_SEARCH_5_28; + tea->freq = 90500 * 16; /* 90.5Mhz default */ + snd_tea575x_set_freq(tea); + + tea->vd = tea575x_radio; + video_set_drvdata(&tea->vd, tea); + mutex_init(&tea->mutex); + strlcpy(tea->vd.name, tea->v4l2_dev->name, sizeof(tea->vd.name)); + tea->vd.lock = &tea->mutex; + tea->vd.v4l2_dev = tea->v4l2_dev; + tea->fops = tea575x_fops; + tea->fops.owner = owner; + tea->vd.fops = &tea->fops; + set_bit(V4L2_FL_USE_FH_PRIO, &tea->vd.flags); + /* disable hw_freq_seek if we can't use it */ + if (tea->cannot_read_data) + v4l2_disable_ioctl(&tea->vd, VIDIOC_S_HW_FREQ_SEEK); + + if (!tea->cannot_mute) { + tea->vd.ctrl_handler = &tea->ctrl_handler; + v4l2_ctrl_handler_init(&tea->ctrl_handler, 1); + v4l2_ctrl_new_std(&tea->ctrl_handler, &tea575x_ctrl_ops, + V4L2_CID_AUDIO_MUTE, 0, 1, 1, 1); + retval = tea->ctrl_handler.error; + if (retval) { + v4l2_err(tea->v4l2_dev, "can't initialize controls\n"); + v4l2_ctrl_handler_free(&tea->ctrl_handler); + return retval; + } + + if (tea->ext_init) { + retval = tea->ext_init(tea); + if (retval) { + v4l2_ctrl_handler_free(&tea->ctrl_handler); + return retval; + } + } + + v4l2_ctrl_handler_setup(&tea->ctrl_handler); + } + + retval = video_register_device(&tea->vd, VFL_TYPE_RADIO, tea->radio_nr); + if (retval) { + v4l2_err(tea->v4l2_dev, "can't register video device!\n"); + v4l2_ctrl_handler_free(tea->vd.ctrl_handler); + return retval; + } + + return 0; +} + +void snd_tea575x_exit(struct snd_tea575x *tea) +{ + video_unregister_device(&tea->vd); + v4l2_ctrl_handler_free(tea->vd.ctrl_handler); +} + +static int __init alsa_tea575x_module_init(void) +{ + return 0; +} + +static void __exit alsa_tea575x_module_exit(void) +{ +} + +module_init(alsa_tea575x_module_init) +module_exit(alsa_tea575x_module_exit) + +EXPORT_SYMBOL(snd_tea575x_init); +EXPORT_SYMBOL(snd_tea575x_exit); +EXPORT_SYMBOL(snd_tea575x_set_freq); diff --git a/sound/i2c/other/Makefile b/sound/i2c/other/Makefile index c95d8f1..5526b03 100644 --- a/sound/i2c/other/Makefile +++ b/sound/i2c/other/Makefile @@ -8,10 +8,8 @@ snd-ak4117-objs := ak4117.o snd-ak4113-objs := ak4113.o snd-ak4xxx-adda-objs := ak4xxx-adda.o snd-pt2258-objs := pt2258.o -snd-tea575x-tuner-objs := tea575x-tuner.o # Module Dependency obj-$(CONFIG_SND_PDAUDIOCF) += snd-ak4117.o obj-$(CONFIG_SND_ICE1712) += snd-ak4xxx-adda.o obj-$(CONFIG_SND_ICE1724) += snd-ak4114.o snd-ak4113.o snd-ak4xxx-adda.o snd-pt2258.o -obj-$(CONFIG_SND_TEA575X) += snd-tea575x-tuner.o diff --git a/sound/i2c/other/tea575x-tuner.c b/sound/i2c/other/tea575x-tuner.c deleted file mode 100644 index a1cb208..0000000 --- a/sound/i2c/other/tea575x-tuner.c +++ /dev/null @@ -1,577 +0,0 @@ -/* - * ALSA driver for TEA5757/5759 Philips AM/FM radio tuner chips - * - * Copyright (c) 2004 Jaroslav Kysela <perex(a)perex.cz> - * - * - * 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 <asm/io.h> -#include <linux/delay.h> -#include <linux/module.h> -#include <linux/init.h> -#include <linux/slab.h> -#include <linux/sched.h> -#include <media/v4l2-device.h> -#include <media/v4l2-dev.h> -#include <media/v4l2-fh.h> -#include <media/v4l2-ioctl.h> -#include <media/v4l2-event.h> -#include <media/tea575x.h> - -MODULE_AUTHOR("Jaroslav Kysela <perex(a)perex.cz>"); -MODULE_DESCRIPTION("Routines for control of TEA5757/5759 Philips AM/FM radio tuner chips"); -MODULE_LICENSE("GPL"); - -/* - * definitions - */ - -#define TEA575X_BIT_SEARCH (1<<24) /* 1 = search action, 0 = tuned */ -#define TEA575X_BIT_UPDOWN (1<<23) /* 0 = search down, 1 = search up */ -#define TEA575X_BIT_MONO (1<<22) /* 0 = stereo, 1 = mono */ -#define TEA575X_BIT_BAND_MASK (3<<20) -#define TEA575X_BIT_BAND_FM (0<<20) -#define TEA575X_BIT_BAND_MW (1<<20) -#define TEA575X_BIT_BAND_LW (2<<20) -#define TEA575X_BIT_BAND_SW (3<<20) -#define TEA575X_BIT_PORT_0 (1<<19) /* user bit */ -#define TEA575X_BIT_PORT_1 (1<<18) /* user bit */ -#define TEA575X_BIT_SEARCH_MASK (3<<16) /* search level */ -#define TEA575X_BIT_SEARCH_5_28 (0<<16) /* FM >5uV, AM >28uV */ -#define TEA575X_BIT_SEARCH_10_40 (1<<16) /* FM >10uV, AM > 40uV */ -#define TEA575X_BIT_SEARCH_30_63 (2<<16) /* FM >30uV, AM > 63uV */ -#define TEA575X_BIT_SEARCH_150_1000 (3<<16) /* FM > 150uV, AM > 1000uV */ -#define TEA575X_BIT_DUMMY (1<<15) /* buffer */ -#define TEA575X_BIT_FREQ_MASK 0x7fff - -enum { BAND_FM, BAND_FM_JAPAN, BAND_AM }; - -static const struct v4l2_frequency_band bands[] = { - { - .type = V4L2_TUNER_RADIO, - .index = 0, - .capability = V4L2_TUNER_CAP_LOW | V4L2_TUNER_CAP_STEREO | - V4L2_TUNER_CAP_FREQ_BANDS, - .rangelow = 87500 * 16, - .rangehigh = 108000 * 16, - .modulation = V4L2_BAND_MODULATION_FM, - }, - { - .type = V4L2_TUNER_RADIO, - .index = 0, - .capability = V4L2_TUNER_CAP_LOW | V4L2_TUNER_CAP_STEREO | - V4L2_TUNER_CAP_FREQ_BANDS, - .rangelow = 76000 * 16, - .rangehigh = 91000 * 16, - .modulation = V4L2_BAND_MODULATION_FM, - }, - { - .type = V4L2_TUNER_RADIO, - .index = 1, - .capability = V4L2_TUNER_CAP_LOW | V4L2_TUNER_CAP_FREQ_BANDS, - .rangelow = 530 * 16, - .rangehigh = 1710 * 16, - .modulation = V4L2_BAND_MODULATION_AM, - }, -}; - -/* - * lowlevel part - */ - -static void snd_tea575x_write(struct snd_tea575x *tea, unsigned int val) -{ - u16 l; - u8 data; - - if (tea->ops->write_val) - return tea->ops->write_val(tea, val); - - tea->ops->set_direction(tea, 1); - udelay(16); - - for (l = 25; l > 0; l--) { - data = (val >> 24) & TEA575X_DATA; - val <<= 1; /* shift data */ - tea->ops->set_pins(tea, data | TEA575X_WREN); - udelay(2); - tea->ops->set_pins(tea, data | TEA575X_WREN | TEA575X_CLK); - udelay(2); - tea->ops->set_pins(tea, data | TEA575X_WREN); - udelay(2); - } - - if (!tea->mute) - tea->ops->set_pins(tea, 0); -} - -static u32 snd_tea575x_read(struct snd_tea575x *tea) -{ - u16 l, rdata; - u32 data = 0; - - if (tea->ops->read_val) - return tea->ops->read_val(tea); - - tea->ops->set_direction(tea, 0); - tea->ops->set_pins(tea, 0); - udelay(16); - - for (l = 24; l--;) { - tea->ops->set_pins(tea, TEA575X_CLK); - udelay(2); - if (!l) - tea->tuned = tea->ops->get_pins(tea) & TEA575X_MOST ? 0 : 1; - tea->ops->set_pins(tea, 0); - udelay(2); - data <<= 1; /* shift data */ - rdata = tea->ops->get_pins(tea); - if (!l) - tea->stereo = (rdata & TEA575X_MOST) ? 0 : 1; - if (rdata & TEA575X_DATA) - data++; - udelay(2); - } - - if (tea->mute) - tea->ops->set_pins(tea, TEA575X_WREN); - - return data; -} - -static u32 snd_tea575x_val_to_freq(struct snd_tea575x *tea, u32 val) -{ - u32 freq = val & TEA575X_BIT_FREQ_MASK; - - if (freq == 0) - return freq; - - switch (tea->band) { - case BAND_FM: - /* freq *= 12.5 */ - freq *= 125; - freq /= 10; - /* crystal fixup */ - freq -= TEA575X_FMIF; - break; - case BAND_FM_JAPAN: - /* freq *= 12.5 */ - freq *= 125; - freq /= 10; - /* crystal fixup */ - freq += TEA575X_FMIF; - break; - case BAND_AM: - /* crystal fixup */ - freq -= TEA575X_AMIF; - break; - } - - return clamp(freq * 16, bands[tea->band].rangelow, - bands[tea->band].rangehigh); /* from kHz */ -} - -static u32 snd_tea575x_get_freq(struct snd_tea575x *tea) -{ - return snd_tea575x_val_to_freq(tea, snd_tea575x_read(tea)); -} - -void snd_tea575x_set_freq(struct snd_tea575x *tea) -{ - u32 freq = tea->freq / 16; /* to kHz */ - u32 band = 0; - - switch (tea->band) { - case BAND_FM: - band = TEA575X_BIT_BAND_FM; - /* crystal fixup */ - freq += TEA575X_FMIF; - /* freq /= 12.5 */ - freq *= 10; - freq /= 125; - break; - case BAND_FM_JAPAN: - band = TEA575X_BIT_BAND_FM; - /* crystal fixup */ - freq -= TEA575X_FMIF; - /* freq /= 12.5 */ - freq *= 10; - freq /= 125; - break; - case BAND_AM: - band = TEA575X_BIT_BAND_MW; - /* crystal fixup */ - freq += TEA575X_AMIF; - break; - } - - tea->val &= ~(TEA575X_BIT_FREQ_MASK | TEA575X_BIT_BAND_MASK); - tea->val |= band; - tea->val |= freq & TEA575X_BIT_FREQ_MASK; - snd_tea575x_write(tea, tea->val); - tea->freq = snd_tea575x_val_to_freq(tea, tea->val); -} - -/* - * Linux Video interface - */ - -static int vidioc_querycap(struct file *file, void *priv, - struct v4l2_capability *v) -{ - struct snd_tea575x *tea = video_drvdata(file); - - strlcpy(v->driver, tea->v4l2_dev->name, sizeof(v->driver)); - strlcpy(v->card, tea->card, sizeof(v->card)); - strlcat(v->card, tea->tea5759 ? " TEA5759" : " TEA5757", sizeof(v->card)); - strlcpy(v->bus_info, tea->bus_info, sizeof(v->bus_info)); - v->device_caps = V4L2_CAP_TUNER | V4L2_CAP_RADIO; - if (!tea->cannot_read_data) - v->device_caps |= V4L2_CAP_HW_FREQ_SEEK; - v->capabilities = v->device_caps | V4L2_CAP_DEVICE_CAPS; - return 0; -} - -static int vidioc_enum_freq_bands(struct file *file, void *priv, - struct v4l2_frequency_band *band) -{ - struct snd_tea575x *tea = video_drvdata(file); - int index; - - if (band->tuner != 0) - return -EINVAL; - - switch (band->index) { - case 0: - if (tea->tea5759) - index = BAND_FM_JAPAN; - else - index = BAND_FM; - break; - case 1: - if (tea->has_am) { - index = BAND_AM; - break; - } - /* Fall through */ - default: - return -EINVAL; - } - - *band = bands[index]; - if (!tea->cannot_read_data) - band->capability |= V4L2_TUNER_CAP_HWSEEK_BOUNDED; - - return 0; -} - -static int vidioc_g_tuner(struct file *file, void *priv, - struct v4l2_tuner *v) -{ - struct snd_tea575x *tea = video_drvdata(file); - struct v4l2_frequency_band band_fm = { 0, }; - - if (v->index > 0) - return -EINVAL; - - snd_tea575x_read(tea); - vidioc_enum_freq_bands(file, priv, &band_fm); - - memset(v, 0, sizeof(*v)); - strlcpy(v->name, tea->has_am ? "FM/AM" : "FM", sizeof(v->name)); - v->type = V4L2_TUNER_RADIO; - v->capability = band_fm.capability; - v->rangelow = tea->has_am ? bands[BAND_AM].rangelow : band_fm.rangelow; - v->rangehigh = band_fm.rangehigh; - v->rxsubchans = tea->stereo ? V4L2_TUNER_SUB_STEREO : V4L2_TUNER_SUB_MONO; - v->audmode = (tea->val & TEA575X_BIT_MONO) ? - V4L2_TUNER_MODE_MONO : V4L2_TUNER_MODE_STEREO; - v->signal = tea->tuned ? 0xffff : 0; - return 0; -} - -static int vidioc_s_tuner(struct file *file, void *priv, - const struct v4l2_tuner *v) -{ - struct snd_tea575x *tea = video_drvdata(file); - u32 orig_val = tea->val; - - if (v->index) - return -EINVAL; - tea->val &= ~TEA575X_BIT_MONO; - if (v->audmode == V4L2_TUNER_MODE_MONO) - tea->val |= TEA575X_BIT_MONO; - /* Only apply changes if currently tuning FM */ - if (tea->band != BAND_AM && tea->val != orig_val) - snd_tea575x_set_freq(tea); - - return 0; -} - -static int vidioc_g_frequency(struct file *file, void *priv, - struct v4l2_frequency *f) -{ - struct snd_tea575x *tea = video_drvdata(file); - - if (f->tuner != 0) - return -EINVAL; - f->type = V4L2_TUNER_RADIO; - f->frequency = tea->freq; - return 0; -} - -static int vidioc_s_frequency(struct file *file, void *priv, - const struct v4l2_frequency *f) -{ - struct snd_tea575x *tea = video_drvdata(file); - - if (f->tuner != 0 || f->type != V4L2_TUNER_RADIO) - return -EINVAL; - - if (tea->has_am && f->frequency < (20000 * 16)) - tea->band = BAND_AM; - else if (tea->tea5759) - tea->band = BAND_FM_JAPAN; - else - tea->band = BAND_FM; - - tea->freq = clamp_t(u32, f->frequency, bands[tea->band].rangelow, - bands[tea->band].rangehigh); - snd_tea575x_set_freq(tea); - return 0; -} - -static int vidioc_s_hw_freq_seek(struct file *file, void *fh, - const struct v4l2_hw_freq_seek *a) -{ - struct snd_tea575x *tea = video_drvdata(file); - unsigned long timeout; - int i, spacing; - - if (tea->cannot_read_data) - return -ENOTTY; - if (a->tuner || a->wrap_around) - return -EINVAL; - - if (file->f_flags & O_NONBLOCK) - return -EWOULDBLOCK; - - if (a->rangelow || a->rangehigh) { - for (i = 0; i < ARRAY_SIZE(bands); i++) { - if ((i == BAND_FM && tea->tea5759) || - (i == BAND_FM_JAPAN && !tea->tea5759) || - (i == BAND_AM && !tea->has_am)) - continue; - if (bands[i].rangelow == a->rangelow && - bands[i].rangehigh == a->rangehigh) - break; - } - if (i == ARRAY_SIZE(bands)) - return -EINVAL; /* No matching band found */ - if (i != tea->band) { - tea->band = i; - tea->freq = clamp(tea->freq, bands[i].rangelow, - bands[i].rangehigh); - snd_tea575x_set_freq(tea); - } - } - - spacing = (tea->band == BAND_AM) ? 5 : 50; /* kHz */ - - /* clear the frequency, HW will fill it in */ - tea->val &= ~TEA575X_BIT_FREQ_MASK; - tea->val |= TEA575X_BIT_SEARCH; - if (a->seek_upward) - tea->val |= TEA575X_BIT_UPDOWN; - else - tea->val &= ~TEA575X_BIT_UPDOWN; - snd_tea575x_write(tea, tea->val); - timeout = jiffies + msecs_to_jiffies(10000); - for (;;) { - if (time_after(jiffies, timeout)) - break; - if (schedule_timeout_interruptible(msecs_to_jiffies(10))) { - /* some signal arrived, stop search */ - tea->val &= ~TEA575X_BIT_SEARCH; - snd_tea575x_set_freq(tea); - return -ERESTARTSYS; - } - if (!(snd_tea575x_read(tea) & TEA575X_BIT_SEARCH)) { - u32 freq; - - /* Found a frequency, wait until it can be read */ - for (i = 0; i < 100; i++) { - msleep(10); - freq = snd_tea575x_get_freq(tea); - if (freq) /* available */ - break; - } - if (freq == 0) /* shouldn't happen */ - break; - /* - * if we moved by less than the spacing, or in the - * wrong direction, continue seeking - */ - if (abs(tea->freq - freq) < 16 * spacing || - (a->seek_upward && freq < tea->freq) || - (!a->seek_upward && freq > tea->freq)) { - snd_tea575x_write(tea, tea->val); - continue; - } - tea->freq = freq; - tea->val &= ~TEA575X_BIT_SEARCH; - return 0; - } - } - tea->val &= ~TEA575X_BIT_SEARCH; - snd_tea575x_set_freq(tea); - return -ENODATA; -} - -static int tea575x_s_ctrl(struct v4l2_ctrl *ctrl) -{ - struct snd_tea575x *tea = container_of(ctrl->handler, struct snd_tea575x, ctrl_handler); - - switch (ctrl->id) { - case V4L2_CID_AUDIO_MUTE: - tea->mute = ctrl->val; - snd_tea575x_set_freq(tea); - return 0; - } - - return -EINVAL; -} - -static const struct v4l2_file_operations tea575x_fops = { - .unlocked_ioctl = video_ioctl2, - .open = v4l2_fh_open, - .release = v4l2_fh_release, - .poll = v4l2_ctrl_poll, -}; - -static const struct v4l2_ioctl_ops tea575x_ioctl_ops = { - .vidioc_querycap = vidioc_querycap, - .vidioc_g_tuner = vidioc_g_tuner, - .vidioc_s_tuner = vidioc_s_tuner, - .vidioc_g_frequency = vidioc_g_frequency, - .vidioc_s_frequency = vidioc_s_frequency, - .vidioc_s_hw_freq_seek = vidioc_s_hw_freq_seek, - .vidioc_enum_freq_bands = vidioc_enum_freq_bands, - .vidioc_log_status = v4l2_ctrl_log_status, - .vidioc_subscribe_event = v4l2_ctrl_subscribe_event, - .vidioc_unsubscribe_event = v4l2_event_unsubscribe, -}; - -static const struct video_device tea575x_radio = { - .ioctl_ops = &tea575x_ioctl_ops, - .release = video_device_release_empty, -}; - -static const struct v4l2_ctrl_ops tea575x_ctrl_ops = { - .s_ctrl = tea575x_s_ctrl, -}; - -/* - * initialize all the tea575x chips - */ -int snd_tea575x_init(struct snd_tea575x *tea, struct module *owner) -{ - int retval; - - tea->mute = true; - - /* Not all devices can or know how to read the data back. - Such devices can set cannot_read_data to true. */ - if (!tea->cannot_read_data) { - snd_tea575x_write(tea, 0x55AA); - if (snd_tea575x_read(tea) != 0x55AA) - return -ENODEV; - } - - tea->val = TEA575X_BIT_BAND_FM | TEA575X_BIT_SEARCH_5_28; - tea->freq = 90500 * 16; /* 90.5Mhz default */ - snd_tea575x_set_freq(tea); - - tea->vd = tea575x_radio; - video_set_drvdata(&tea->vd, tea); - mutex_init(&tea->mutex); - strlcpy(tea->vd.name, tea->v4l2_dev->name, sizeof(tea->vd.name)); - tea->vd.lock = &tea->mutex; - tea->vd.v4l2_dev = tea->v4l2_dev; - tea->fops = tea575x_fops; - tea->fops.owner = owner; - tea->vd.fops = &tea->fops; - set_bit(V4L2_FL_USE_FH_PRIO, &tea->vd.flags); - /* disable hw_freq_seek if we can't use it */ - if (tea->cannot_read_data) - v4l2_disable_ioctl(&tea->vd, VIDIOC_S_HW_FREQ_SEEK); - - if (!tea->cannot_mute) { - tea->vd.ctrl_handler = &tea->ctrl_handler; - v4l2_ctrl_handler_init(&tea->ctrl_handler, 1); - v4l2_ctrl_new_std(&tea->ctrl_handler, &tea575x_ctrl_ops, - V4L2_CID_AUDIO_MUTE, 0, 1, 1, 1); - retval = tea->ctrl_handler.error; - if (retval) { - v4l2_err(tea->v4l2_dev, "can't initialize controls\n"); - v4l2_ctrl_handler_free(&tea->ctrl_handler); - return retval; - } - - if (tea->ext_init) { - retval = tea->ext_init(tea); - if (retval) { - v4l2_ctrl_handler_free(&tea->ctrl_handler); - return retval; - } - } - - v4l2_ctrl_handler_setup(&tea->ctrl_handler); - } - - retval = video_register_device(&tea->vd, VFL_TYPE_RADIO, tea->radio_nr); - if (retval) { - v4l2_err(tea->v4l2_dev, "can't register video device!\n"); - v4l2_ctrl_handler_free(tea->vd.ctrl_handler); - return retval; - } - - return 0; -} - -void snd_tea575x_exit(struct snd_tea575x *tea) -{ - video_unregister_device(&tea->vd); - v4l2_ctrl_handler_free(tea->vd.ctrl_handler); -} - -static int __init alsa_tea575x_module_init(void) -{ - return 0; -} - -static void __exit alsa_tea575x_module_exit(void) -{ -} - -module_init(alsa_tea575x_module_init) -module_exit(alsa_tea575x_module_exit) - -EXPORT_SYMBOL(snd_tea575x_init); -EXPORT_SYMBOL(snd_tea575x_exit); -EXPORT_SYMBOL(snd_tea575x_set_freq); diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index fe6fa93..3ef9f63 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -1,10 +1,5 @@ # ALSA PCI drivers -config SND_TEA575X - tristate - depends on SND_FM801_TEA575X_BOOL || SND_ES1968_RADIO || RADIO_SF16FMR2 || RADIO_MAXIRADIO || RADIO_SHARK - default SND_FM801 || SND_ES1968 || RADIO_SF16FMR2 || RADIO_MAXIRADIO || RADIO_SHARK - menuconfig SND_PCI bool "PCI sound devices" depends on PCI -- Ondrej Zary

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[alsa-devel] [RFC PATCH 1/2] tea575x: Move header from sound to media
by Ondrej Zary 13 Jun '13

13 Jun '13

Move include/sound/tea575x-tuner.h to include/media/tea575x.h and update files that include it. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- drivers/media/radio/radio-maxiradio.c | 2 +- drivers/media/radio/radio-sf16fmr2.c | 2 +- drivers/media/radio/radio-shark.c | 2 +- include/media/tea575x.h | 78 +++++++++++++++++++++++++++++++++ include/sound/tea575x-tuner.h | 78 --------------------------------- sound/i2c/other/tea575x-tuner.c | 2 +- sound/pci/es1968.c | 2 +- sound/pci/fm801.c | 2 +- 8 files changed, 84 insertions(+), 84 deletions(-) create mode 100644 include/media/tea575x.h delete mode 100644 include/sound/tea575x-tuner.h diff --git a/drivers/media/radio/radio-maxiradio.c b/drivers/media/radio/radio-maxiradio.c index bd4d3a7..a76712a 100644 --- a/drivers/media/radio/radio-maxiradio.c +++ b/drivers/media/radio/radio-maxiradio.c @@ -42,7 +42,7 @@ #include <linux/videodev2.h> #include <linux/io.h> #include <linux/slab.h> -#include <sound/tea575x-tuner.h> +#include <media/tea575x.h> #include <media/v4l2-device.h> #include <media/v4l2-ioctl.h> #include <media/v4l2-fh.h> diff --git a/drivers/media/radio/radio-sf16fmr2.c b/drivers/media/radio/radio-sf16fmr2.c index 9c09904..f1e3714 100644 --- a/drivers/media/radio/radio-sf16fmr2.c +++ b/drivers/media/radio/radio-sf16fmr2.c @@ -14,7 +14,7 @@ #include <linux/io.h> /* outb, outb_p */ #include <linux/isa.h> #include <linux/pnp.h> -#include <sound/tea575x-tuner.h> +#include <media/tea575x.h> MODULE_AUTHOR("Ondrej Zary"); MODULE_DESCRIPTION("MediaForte SF16-FMR2 and SF16-FMD2 FM radio card driver"); diff --git a/drivers/media/radio/radio-shark.c b/drivers/media/radio/radio-shark.c index 8fa18ab..b914772 100644 --- a/drivers/media/radio/radio-shark.c +++ b/drivers/media/radio/radio-shark.c @@ -33,7 +33,7 @@ #include <linux/usb.h> #include <linux/workqueue.h> #include <media/v4l2-device.h> -#include <sound/tea575x-tuner.h> +#include <media/tea575x.h> #if defined(CONFIG_LEDS_CLASS) || \ (defined(CONFIG_LEDS_CLASS_MODULE) && defined(CONFIG_RADIO_SHARK_MODULE)) diff --git a/include/media/tea575x.h b/include/media/tea575x.h new file mode 100644 index 0000000..098c4de --- /dev/null +++ b/include/media/tea575x.h @@ -0,0 +1,78 @@ +#ifndef __SOUND_TEA575X_TUNER_H +#define __SOUND_TEA575X_TUNER_H + +/* + * ALSA driver for TEA5757/5759 Philips AM/FM tuner chips + * + * Copyright (c) 2004 Jaroslav Kysela <perex(a)perex.cz> + * + * 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/videodev2.h> +#include <media/v4l2-ctrls.h> +#include <media/v4l2-dev.h> +#include <media/v4l2-device.h> + +#define TEA575X_FMIF 10700 +#define TEA575X_AMIF 450 + +#define TEA575X_DATA (1 << 0) +#define TEA575X_CLK (1 << 1) +#define TEA575X_WREN (1 << 2) +#define TEA575X_MOST (1 << 3) + +struct snd_tea575x; + +struct snd_tea575x_ops { + /* Drivers using snd_tea575x must either define read_ and write_val */ + void (*write_val)(struct snd_tea575x *tea, u32 val); + u32 (*read_val)(struct snd_tea575x *tea); + /* Or define the 3 pin functions */ + void (*set_pins)(struct snd_tea575x *tea, u8 pins); + u8 (*get_pins)(struct snd_tea575x *tea); + void (*set_direction)(struct snd_tea575x *tea, bool output); +}; + +struct snd_tea575x { + struct v4l2_device *v4l2_dev; + struct v4l2_file_operations fops; + struct video_device vd; /* video device */ + int radio_nr; /* radio_nr */ + bool tea5759; /* 5759 chip is present */ + bool has_am; /* Device can tune to AM freqs */ + bool cannot_read_data; /* Device cannot read the data pin */ + bool cannot_mute; /* Device cannot mute */ + bool mute; /* Device is muted? */ + bool stereo; /* receiving stereo */ + bool tuned; /* tuned to a station */ + unsigned int val; /* hw value */ + u32 band; /* 0: FM, 1: FM-Japan, 2: AM */ + u32 freq; /* frequency */ + struct mutex mutex; + struct snd_tea575x_ops *ops; + void *private_data; + u8 card[32]; + u8 bus_info[32]; + struct v4l2_ctrl_handler ctrl_handler; + int (*ext_init)(struct snd_tea575x *tea); +}; + +int snd_tea575x_init(struct snd_tea575x *tea, struct module *owner); +void snd_tea575x_exit(struct snd_tea575x *tea); +void snd_tea575x_set_freq(struct snd_tea575x *tea); + +#endif /* __SOUND_TEA575X_TUNER_H */ diff --git a/include/sound/tea575x-tuner.h b/include/sound/tea575x-tuner.h deleted file mode 100644 index 098c4de..0000000 --- a/include/sound/tea575x-tuner.h +++ /dev/null @@ -1,78 +0,0 @@ -#ifndef __SOUND_TEA575X_TUNER_H -#define __SOUND_TEA575X_TUNER_H - -/* - * ALSA driver for TEA5757/5759 Philips AM/FM tuner chips - * - * Copyright (c) 2004 Jaroslav Kysela <perex(a)perex.cz> - * - * 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/videodev2.h> -#include <media/v4l2-ctrls.h> -#include <media/v4l2-dev.h> -#include <media/v4l2-device.h> - -#define TEA575X_FMIF 10700 -#define TEA575X_AMIF 450 - -#define TEA575X_DATA (1 << 0) -#define TEA575X_CLK (1 << 1) -#define TEA575X_WREN (1 << 2) -#define TEA575X_MOST (1 << 3) - -struct snd_tea575x; - -struct snd_tea575x_ops { - /* Drivers using snd_tea575x must either define read_ and write_val */ - void (*write_val)(struct snd_tea575x *tea, u32 val); - u32 (*read_val)(struct snd_tea575x *tea); - /* Or define the 3 pin functions */ - void (*set_pins)(struct snd_tea575x *tea, u8 pins); - u8 (*get_pins)(struct snd_tea575x *tea); - void (*set_direction)(struct snd_tea575x *tea, bool output); -}; - -struct snd_tea575x { - struct v4l2_device *v4l2_dev; - struct v4l2_file_operations fops; - struct video_device vd; /* video device */ - int radio_nr; /* radio_nr */ - bool tea5759; /* 5759 chip is present */ - bool has_am; /* Device can tune to AM freqs */ - bool cannot_read_data; /* Device cannot read the data pin */ - bool cannot_mute; /* Device cannot mute */ - bool mute; /* Device is muted? */ - bool stereo; /* receiving stereo */ - bool tuned; /* tuned to a station */ - unsigned int val; /* hw value */ - u32 band; /* 0: FM, 1: FM-Japan, 2: AM */ - u32 freq; /* frequency */ - struct mutex mutex; - struct snd_tea575x_ops *ops; - void *private_data; - u8 card[32]; - u8 bus_info[32]; - struct v4l2_ctrl_handler ctrl_handler; - int (*ext_init)(struct snd_tea575x *tea); -}; - -int snd_tea575x_init(struct snd_tea575x *tea, struct module *owner); -void snd_tea575x_exit(struct snd_tea575x *tea); -void snd_tea575x_set_freq(struct snd_tea575x *tea); - -#endif /* __SOUND_TEA575X_TUNER_H */ diff --git a/sound/i2c/other/tea575x-tuner.c b/sound/i2c/other/tea575x-tuner.c index 8a36a1d..a1cb208 100644 --- a/sound/i2c/other/tea575x-tuner.c +++ b/sound/i2c/other/tea575x-tuner.c @@ -31,7 +31,7 @@ #include <media/v4l2-fh.h> #include <media/v4l2-ioctl.h> #include <media/v4l2-event.h> -#include <sound/tea575x-tuner.h> +#include <media/tea575x.h> MODULE_AUTHOR("Jaroslav Kysela <perex(a)perex.cz>"); MODULE_DESCRIPTION("Routines for control of TEA5757/5759 Philips AM/FM radio tuner chips"); diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index a1f32b5..c7c23b3 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -113,7 +113,7 @@ #include <sound/initval.h> #ifdef CONFIG_SND_ES1968_RADIO -#include <sound/tea575x-tuner.h> +#include <media/tea575x.h> #endif #define CARD_NAME "ESS Maestro1/2" diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index 4f07fda..b1c8ce2 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -37,7 +37,7 @@ #include <asm/io.h> #ifdef CONFIG_SND_FM801_TEA575X_BOOL -#include <sound/tea575x-tuner.h> +#include <media/tea575x.h> #endif MODULE_AUTHOR("Jaroslav Kysela <perex(a)perex.cz>"); -- Ondrej Zary

1 0

[alsa-devel] [PATCH 1/2] ASoC: rt5640: fix sparse warnings
by Stephen Warren 13 Jun '13

13 Jun '13

From: Stephen Warren <swarren(a)nvidia.com> This fixes: 975:9: sparse: Using plain integer as NULL pointer 1917:24: sparse: symbol 'rt5640_aif_dai_ops' was not declared. Should it be static? 1924:27: sparse: symbol 'rt5640_dai' was not declared. Should it be static? 2079:19: sparse: symbol 'rt5640_i2c_driver' was not declared. Should it be static? Reported-by: Fengguang Wu <fengguang.wu(a)intel.com> Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/codecs/rt5640.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 8761552..ce585e3 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -974,7 +974,7 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("LDO2", RT5640_PWR_ANLG1, RT5640_PWR_LDO2_BIT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5640_PWR_ANLG2, - RT5640_PWR_MB1_BIT, 0, 0, 0), + RT5640_PWR_MB1_BIT, 0, NULL, 0), /* Input Lines */ SND_SOC_DAPM_INPUT("DMIC1"), SND_SOC_DAPM_INPUT("DMIC2"), @@ -1915,14 +1915,14 @@ static int rt5640_resume(struct snd_soc_codec *codec) #define RT5640_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) -struct snd_soc_dai_ops rt5640_aif_dai_ops = { +static const struct snd_soc_dai_ops rt5640_aif_dai_ops = { .hw_params = rt5640_hw_params, .set_fmt = rt5640_set_dai_fmt, .set_sysclk = rt5640_set_dai_sysclk, .set_pll = rt5640_set_dai_pll, }; -struct snd_soc_dai_driver rt5640_dai[] = { +static struct snd_soc_dai_driver rt5640_dai[] = { { .name = "rt5640-aif1", .id = RT5640_AIF1, @@ -2112,7 +2112,7 @@ static int rt5640_i2c_remove(struct i2c_client *i2c) return 0; } -struct i2c_driver rt5640_i2c_driver = { +static struct i2c_driver rt5640_i2c_driver = { .driver = { .name = "rt5640", .owner = THIS_MODULE, -- 1.8.1.5

2 5

[alsa-devel] [PATCH (alsa-tools)] hdajackretask: Fix killing PulseAudio on newer PulseAudio versions
by David Henningsson 13 Jun '13

13 Jun '13

The standard configuration directory for PulseAudio has changed, I think since 3.0. Adjust the directory lookup mechanism to match. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- hdajackretask/apply-changes.c | 15 ++++++++++++++- 1 file changed, 14 insertions(+), 1 deletion(-) diff --git a/hdajackretask/apply-changes.c b/hdajackretask/apply-changes.c index eda404e..77563f7 100644 --- a/hdajackretask/apply-changes.c +++ b/hdajackretask/apply-changes.c @@ -97,7 +97,20 @@ gboolean run_sudo_script(const gchar* script_name, GError** err) static gchar* get_pulseaudio_client_conf() { - return g_strdup_printf("%s/.pulse/client.conf", g_get_home_dir()); + /* Reference: See src/pulsecore/core-util.c in pulseaudio */ + gchar* fname; + gchar* dir = g_strdup_printf("%s/.pulse", g_get_home_dir()); + if (access(dir, F_OK) < 0) { + const gchar* xch = g_getenv("XDG_CONFIG_HOME"); + g_free(dir); + if (xch) + dir = g_strdup_printf("%s/pulse", xch); + else + dir = g_strdup_printf("%s/.config/pulse", g_get_home_dir()); + } + fname = g_strdup_printf("%s/client.conf", dir); + g_free(dir); + return fname; } static gboolean kill_pulseaudio(gboolean* was_killed, int card, GError** err) -- 1.7.9.5

1 0

[alsa-devel] [GIT PULL] ASoC updates for v3.10
by Mark Brown 13 Jun '13

13 Jun '13

The following changes since commit 317ddd256b9c24b0d78fa8018f80f1e495481a10: Linux 3.10-rc5 (2013-06-08 17:41:04 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.10-rc5 for you to fetch changes up to 902a75cdc491066514aac35b40a756aaa9740173: Merge remote-tracking branch 'asoc/fix/wm8994' into asoc-linus (2013-06-10 10:28:51 +0100) Takashi is travelling at the minute and it'd be good to get the MAINTAINERS update in here merged so sending directly. ---------------------------------------------------------------- ASoC: Updates for v3.10 As well as the usual driver specifics we've got a couple of core fixes here, one fixing capabilities for unidirectional streams and the other fixing suspend while audio streams are active. The suspend fix is a little involved but mostly as a result of removing some special casing that was doing the wrong thing. ---------------------------------------------------------------- Andreas Irestål (1): ASoC: tlv320aic3x: Remove deadlock from snd_soc_dapm_put_volsw_aic3x() Charles Keepax (1): ASoC: arizona: Correct AEC loopback enable Mark Brown (11): ASoC: wm8994: Avoid leaking pm_runtime reference on removed jack race ASoC: wm8994: Ensure microphone detection state is reset on removal MAINTAINERS: Remove myself from Wolfson maintainers ASoC: pcm: Require both CODEC and CPU support when declaring stream caps ASoC: dapm: Treat DAI widgets like AIF widgets for power Merge remote-tracking branch 'asoc/fix/arizona' into asoc-linus Merge remote-tracking branch 'asoc/fix/core' into asoc-linus Merge remote-tracking branch 'asoc/fix/cs42l52' into asoc-linus Merge remote-tracking branch 'asoc/fix/maintainers' into asoc-linus Merge remote-tracking branch 'asoc/fix/tlv320aic3x' into asoc-linus Merge remote-tracking branch 'asoc/fix/wm8994' into asoc-linus Nicolas Schichan (2): ASoC: cs42l52: use correct PCM mixer TLV dB scale to match datasheet. ASoC: cs42l52: fix hp_gain_enum shift value. MAINTAINERS | 3 +-- include/sound/soc-dapm.h | 3 ++- sound/soc/codecs/cs42l52.c | 6 +++-- sound/soc/codecs/tlv320aic3x.c | 10 ++++---- sound/soc/codecs/wm5102.c | 3 ++- sound/soc/codecs/wm5110.c | 3 ++- sound/soc/codecs/wm8994.c | 3 ++- sound/soc/soc-dapm.c | 49 +++++++++++++++++++++------------------- sound/soc/soc-pcm.c | 13 ++++++++--- 9 files changed, 54 insertions(+), 39 deletions(-)

1 0

[alsa-devel] [PATCH] ASoC: SPEAr spdif_{in, out}: use devm for clk and a few more cleanups
by Uwe Kleine-König 13 Jun '13

13 Jun '13

Drop dev_set_drvdata as this is handled in the core and use devm_request_and_ioremap instead of devm_ioremap to properly register the address range used Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/soc/spear/spdif_in.c | 17 ++--------------- sound/soc/spear/spdif_out.c | 17 +++-------------- 2 files changed, 5 insertions(+), 29 deletions(-) diff --git a/sound/soc/spear/spdif_in.c b/sound/soc/spear/spdif_in.c index c7c4b20..d40bb34 100644 --- a/sound/soc/spear/spdif_in.c +++ b/sound/soc/spear/spdif_in.c @@ -231,7 +231,7 @@ static int spdif_in_probe(struct platform_device *pdev) if (host->irq < 0) return -EINVAL; - host->clk = clk_get(&pdev->dev, NULL); + host->clk = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(host->clk)) return PTR_ERR(host->clk); @@ -253,33 +253,20 @@ static int spdif_in_probe(struct platform_device *pdev) ret = devm_request_irq(&pdev->dev, host->irq, spdif_in_irq, 0, "spdif-in", host); if (ret) { - clk_put(host->clk); dev_warn(&pdev->dev, "request_irq failed\n"); return ret; } - ret = snd_soc_register_dai(&pdev->dev, &spdif_in_dai); - if (ret != 0) { - clk_put(host->clk); - return ret; - } - - return 0; + return snd_soc_register_dai(&pdev->dev, &spdif_in_dai); } static int spdif_in_remove(struct platform_device *pdev) { - struct spdif_in_dev *host = dev_get_drvdata(&pdev->dev); - snd_soc_unregister_dai(&pdev->dev); - dev_set_drvdata(&pdev->dev, NULL); - - clk_put(host->clk); return 0; } - static struct platform_driver spdif_in_driver = { .probe = spdif_in_probe, .remove = spdif_in_remove, diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c index 5eac4cd..b2488b4 100644 --- a/sound/soc/spear/spdif_out.c +++ b/sound/soc/spear/spdif_out.c @@ -293,14 +293,14 @@ static int spdif_out_probe(struct platform_device *pdev) return -ENOMEM; } - host->io_base = devm_ioremap(&pdev->dev, res->start, + host->io_base = devm_request_and_ioremap(&pdev->dev, res->start, resource_size(res)); if (!host->io_base) { dev_warn(&pdev->dev, "ioremap failed\n"); return -ENOMEM; } - host->clk = clk_get(&pdev->dev, NULL); + host->clk = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(host->clk)) return PTR_ERR(host->clk); @@ -314,23 +314,12 @@ static int spdif_out_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, host); - ret = snd_soc_register_dai(&pdev->dev, &spdif_out_dai); - if (ret != 0) { - clk_put(host->clk); - return ret; - } - - return 0; + return snd_soc_register_dai(&pdev->dev, &spdif_out_dai); } static int spdif_out_remove(struct platform_device *pdev) { - struct spdif_out_dev *host = dev_get_drvdata(&pdev->dev); - snd_soc_unregister_dai(&pdev->dev); - dev_set_drvdata(&pdev->dev, NULL); - - clk_put(host->clk); return 0; } -- 1.8.2.rc2

3 4

[alsa-devel] [PATCH 3/3 v2] gpio: arizona: Add wm8997 support to probe
by Charles Keepax 13 Jun '13

13 Jun '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- drivers/gpio/gpio-arizona.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/drivers/gpio/gpio-arizona.c b/drivers/gpio/gpio-arizona.c index 0ea853f..05deab61 100644 --- a/drivers/gpio/gpio-arizona.c +++ b/drivers/gpio/gpio-arizona.c @@ -113,6 +113,7 @@ static int arizona_gpio_probe(struct platform_device *pdev) switch (arizona->type) { case WM5102: case WM5110: + case WM8997: arizona_gpio->gpio_chip.ngpio = 5; break; default: -- 1.7.2.5

1 0

[alsa-devel] Problem using alsa to implement an Android dock
by Jeremy Rosen 13 Jun '13

13 Jun '13

Hello everybody I am trying to implement an android dock on linux, and there is some cases where alsa seems to not properly recognise the Android phone as a USB sound device... First a little bit of context since Android 4.1 it is possible to put an android phone (connected to a PC via USB) in "dock mode" Basically you send a couple of special USB packet to the phone and the phone will disconnect from the USB bus then reconnect as a USB sound device. You can then use any tool to handle it as an audio source and redirct the sound to your speaker. For those intereset the details can be found at http://source.android.com/tech/accessories/aoap/aoa2.html This works really well, no problem with that... However Android also has something called "accessory mode" which is used to create custom apps that communicate in "raw" mode with the host my problems arrive when both the dock mode and the accessory mode are enabled. in that case Alsa will not recognise the usb sound device exposed by android. It will create an entry for it in /proc/asound but there are no mixer or input stream I attach the output of lsusb -v in both cases and a small C program that I use to put the phone in the proper modes (there is a small section of code to (un)comment to en/disable accessory mode, it uses libusb for usb access and should be trivial to compile with it) At this point I am not sure how to proceed, is this a real bug or something not implemented ? what extra info do I need to provide ? any help would be welcome Regards Jérémy Rosen fight key loggers : write some perl using vim

2 8

[alsa-devel] [GIT PULL] ASoC updates for v3.10
by Mark Brown 12 Jun '13

12 Jun '13

The following changes since commit 317ddd256b9c24b0d78fa8018f80f1e495481a10: Linux 3.10-rc5 (2013-06-08 17:41:04 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.10-rc5 for you to fetch changes up to 902a75cdc491066514aac35b40a756aaa9740173: Merge remote-tracking branch 'asoc/fix/wm8994' into asoc-linus (2013-06-10 10:28:51 +0100) ---------------------------------------------------------------- ASoC: Updates for v3.10 As well as the usual driver specifics we've got a couple of core fixes here, one fixing capabilities for unidirectional streams and the other fixing suspend while audio streams are active. The suspend fix is a little involved but mostly as a result of removing some special casing that was doing the wrong thing. ---------------------------------------------------------------- Andreas Irestål (1): ASoC: tlv320aic3x: Remove deadlock from snd_soc_dapm_put_volsw_aic3x() Charles Keepax (1): ASoC: arizona: Correct AEC loopback enable Mark Brown (11): ASoC: wm8994: Avoid leaking pm_runtime reference on removed jack race ASoC: wm8994: Ensure microphone detection state is reset on removal MAINTAINERS: Remove myself from Wolfson maintainers ASoC: pcm: Require both CODEC and CPU support when declaring stream caps ASoC: dapm: Treat DAI widgets like AIF widgets for power Merge remote-tracking branch 'asoc/fix/arizona' into asoc-linus Merge remote-tracking branch 'asoc/fix/core' into asoc-linus Merge remote-tracking branch 'asoc/fix/cs42l52' into asoc-linus Merge remote-tracking branch 'asoc/fix/maintainers' into asoc-linus Merge remote-tracking branch 'asoc/fix/tlv320aic3x' into asoc-linus Merge remote-tracking branch 'asoc/fix/wm8994' into asoc-linus Nicolas Schichan (2): ASoC: cs42l52: use correct PCM mixer TLV dB scale to match datasheet. ASoC: cs42l52: fix hp_gain_enum shift value. MAINTAINERS | 3 +-- include/sound/soc-dapm.h | 3 ++- sound/soc/codecs/cs42l52.c | 6 +++-- sound/soc/codecs/tlv320aic3x.c | 10 ++++---- sound/soc/codecs/wm5102.c | 3 ++- sound/soc/codecs/wm5110.c | 3 ++- sound/soc/codecs/wm8994.c | 3 ++- sound/soc/soc-dapm.c | 49 +++++++++++++++++++++------------------- sound/soc/soc-pcm.c | 13 ++++++++--- 9 files changed, 54 insertions(+), 39 deletions(-)

2 1

[alsa-devel] [PATCH] ASoC: tegra: add tegra+RT5640 machine driver
by Stephen Warren 12 Jun '13

12 Jun '13

From: Stephen Warren <swarren(a)nvidia.com> Initially, this binding and driver only describe/support playback to headphones and speakers. This driver will support Beaver and Dalmore. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- .../bindings/sound/nvidia,tegra-audio-rt5640.txt | 71 ++++++ sound/soc/tegra/Kconfig | 10 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra_rt5640.c | 257 +++++++++++++++++++++ 4 files changed, 340 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt create mode 100644 sound/soc/tegra/tegra_rt5640.c diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt new file mode 100644 index 0000000..d130818 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt @@ -0,0 +1,71 @@ +NVIDIA Tegra audio complex, with RT5640 CODEC + +Required properties: +- compatible : "nvidia,tegra-audio-rt5640" +- clocks : Must contain an entry for each entry in clock-names. +- clock-names : Must include the following entries: + "pll_a" (The Tegra clock of that name), + "pll_a_out0" (The Tegra clock of that name), + "mclk" (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk) +- nvidia,model : The user-visible name of this sound complex. +- nvidia,audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the RT5640's pins, and the jacks on the board: + + RT5640 pins: + + * DMIC1 + * DMIC2 + * MICBIAS1 + * IN1P + * IN1R + * IN2P + * IN2R + * HPOL + * HPOR + * LOUTL + * LOUTR + * MONOP + * MONON + * SPOLP + * SPOLN + * SPORP + * SPORN + + Board connectors: + + * Headphones + * Speakers + +- nvidia,i2s-controller : The phandle of the Tegra I2S controller that's + connected to the CODEC. +- nvidia,audio-codec : The phandle of the RT5640 audio codec. This binding + assumes that AIF1 on the CODEC is connected to Tegra. + +Optional properties: +- nvidia,hp-det-gpios : The GPIO that detects headphones are plugged in + +Example: + +sound { + compatible = "nvidia,tegra-audio-rt5640-dalmore", + "nvidia,tegra-audio-rt5640"; + nvidia,model = "NVIDIA Tegra Dalmore"; + + nvidia,audio-routing = + "Headphones", "HPOR", + "Headphones", "HPOL", + "Speakers", "SPORP", + "Speakers", "SPORN", + "Speakers", "SPOLP", + "Speakers", "SPOLN"; + + nvidia,i2s-controller = <&tegra_i2s1>; + nvidia,audio-codec = <&rt5640>; + + nvidia,hp-det-gpios = <&gpio 143 0>; /* GPIO PR7 */ + + clocks = <&tegra_car 216>, <&tegra_car 217>, <&tegra_car 120>; + clock-names = "pll_a", "pll_a_out0", "mclk"; +}; diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index b1c9d57..995b120 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -59,6 +59,16 @@ config SND_SOC_TEGRA30_I2S Tegra30 I2S interface. You will also need to select the individual machine drivers to support below. +config SND_SOC_TEGRA_RT5640 + tristate "SoC Audio support for Tegra boards using an RT5640 codec" + depends on SND_SOC_TEGRA && I2C + select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC + select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC + select SND_SOC_RT5640 + help + Say Y or M here if you want to add support for SoC audio on Tegra + boards using the RT5640 codec, such as Dalmore. + config SND_SOC_TEGRA_WM8753 tristate "SoC Audio support for Tegra boards using a WM8753 codec" depends on SND_SOC_TEGRA && I2C diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile index 416a14b..21d2550 100644 --- a/sound/soc/tegra/Makefile +++ b/sound/soc/tegra/Makefile @@ -18,12 +18,14 @@ obj-$(CONFIG_SND_SOC_TEGRA30_AHUB) += snd-soc-tegra30-ahub.o obj-$(CONFIG_SND_SOC_TEGRA30_I2S) += snd-soc-tegra30-i2s.o # Tegra machine Support +snd-soc-tegra-rt5640-objs := tegra_rt5640.o snd-soc-tegra-wm8753-objs := tegra_wm8753.o snd-soc-tegra-wm8903-objs := tegra_wm8903.o snd-soc-tegra-wm9712-objs := tegra_wm9712.o snd-soc-tegra-trimslice-objs := trimslice.o snd-soc-tegra-alc5632-objs := tegra_alc5632.o +obj-$(CONFIG_SND_SOC_TEGRA_RT5640) += snd-soc-tegra-rt5640.o obj-$(CONFIG_SND_SOC_TEGRA_WM8753) += snd-soc-tegra-wm8753.o obj-$(CONFIG_SND_SOC_TEGRA_WM8903) += snd-soc-tegra-wm8903.o obj-$(CONFIG_SND_SOC_TEGRA_WM9712) += snd-soc-tegra-wm9712.o diff --git a/sound/soc/tegra/tegra_rt5640.c b/sound/soc/tegra/tegra_rt5640.c new file mode 100644 index 0000000..08794f9 --- /dev/null +++ b/sound/soc/tegra/tegra_rt5640.c @@ -0,0 +1,257 @@ +/* +* tegra_rt5640.c - Tegra machine ASoC driver for boards using WM8903 codec. + * + * Copyright (c) 2013, NVIDIA CORPORATION. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope 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, see <http://www.gnu.org/licenses/>. + * + * Based on code copyright/by: + * + * Copyright (C) 2010-2012 - NVIDIA, Inc. + * Copyright (C) 2011 The AC100 Kernel Team <ac100(a)lists.lauchpad.net> + * (c) 2009, 2010 Nvidia Graphics Pvt. Ltd. + * Copyright 2007 Wolfson Microelectronics PLC. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> + +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "../codecs/rt5640.h" + +#include "tegra_asoc_utils.h" + +#define DRV_NAME "tegra-snd-rt5640" + +struct tegra_rt5640 { + struct tegra_asoc_utils_data util_data; + int gpio_hp_det; +}; + +static int tegra_rt5640_asoc_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_codec *codec = codec_dai->codec; + struct snd_soc_card *card = codec->card; + struct tegra_rt5640 *machine = snd_soc_card_get_drvdata(card); + int srate, mclk; + int err; + + srate = params_rate(params); + mclk = 256 * srate; + + err = tegra_asoc_utils_set_rate(&machine->util_data, srate, mclk); + if (err < 0) { + dev_err(card->dev, "Can't configure clocks\n"); + return err; + } + + err = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_MCLK, mclk, + SND_SOC_CLOCK_IN); + if (err < 0) { + dev_err(card->dev, "codec_dai clock not set\n"); + return err; + } + + return 0; +} + +static struct snd_soc_ops tegra_rt5640_ops = { + .hw_params = tegra_rt5640_asoc_hw_params, +}; + +static struct snd_soc_jack tegra_rt5640_hp_jack; + +static struct snd_soc_jack_pin tegra_rt5640_hp_jack_pins[] = { + { + .pin = "Headphones", + .mask = SND_JACK_HEADPHONE, + }, +}; + +static struct snd_soc_jack_gpio tegra_rt5640_hp_jack_gpio = { + .name = "Headphone detection", + .report = SND_JACK_HEADPHONE, + .debounce_time = 150, + .invert = 1, +}; + +static const struct snd_soc_dapm_widget tegra_rt5640_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_SPK("Speakers", NULL), +}; + +static const struct snd_kcontrol_new tegra_rt5640_controls[] = { + SOC_DAPM_PIN_SWITCH("Speakers"), +}; + +static int tegra_rt5640_asoc_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = codec_dai->codec; + struct tegra_rt5640 *machine = snd_soc_card_get_drvdata(codec->card); + + snd_soc_jack_new(codec, "Headphones", SND_JACK_HEADPHONE, + &tegra_rt5640_hp_jack); + snd_soc_jack_add_pins(&tegra_rt5640_hp_jack, + ARRAY_SIZE(tegra_rt5640_hp_jack_pins), + tegra_rt5640_hp_jack_pins); + + if (gpio_is_valid(machine->gpio_hp_det)) { + tegra_rt5640_hp_jack_gpio.gpio = machine->gpio_hp_det; + snd_soc_jack_add_gpios(&tegra_rt5640_hp_jack, + 1, + &tegra_rt5640_hp_jack_gpio); + } + + return 0; +} + +static struct snd_soc_dai_link tegra_rt5640_dai = { + .name = "RT5640", + .stream_name = "RT5640 PCM", + .codec_dai_name = "rt5640-aif1", + .init = tegra_rt5640_asoc_init, + .ops = &tegra_rt5640_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, +}; + +static struct snd_soc_card snd_soc_tegra_rt5640 = { + .name = "tegra-rt5640", + .owner = THIS_MODULE, + .dai_link = &tegra_rt5640_dai, + .num_links = 1, + .controls = tegra_rt5640_controls, + .num_controls = ARRAY_SIZE(tegra_rt5640_controls), + .dapm_widgets = tegra_rt5640_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tegra_rt5640_dapm_widgets), + .fully_routed = true, +}; + +static int tegra_rt5640_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct snd_soc_card *card = &snd_soc_tegra_rt5640; + struct tegra_rt5640 *machine; + int ret; + + machine = devm_kzalloc(&pdev->dev, + sizeof(struct tegra_rt5640), GFP_KERNEL); + if (!machine) { + dev_err(&pdev->dev, "Can't allocate tegra_rt5640\n"); + return -ENOMEM; + } + + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + snd_soc_card_set_drvdata(card, machine); + + machine->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0); + if (machine->gpio_hp_det == -EPROBE_DEFER) + return -EPROBE_DEFER; + + ret = snd_soc_of_parse_card_name(card, "nvidia,model"); + if (ret) + goto err; + + ret = snd_soc_of_parse_audio_routing(card, "nvidia,audio-routing"); + if (ret) + goto err; + + tegra_rt5640_dai.codec_of_node = of_parse_phandle(np, + "nvidia,audio-codec", 0); + if (!tegra_rt5640_dai.codec_of_node) { + dev_err(&pdev->dev, + "Property 'nvidia,audio-codec' missing or invalid\n"); + ret = -EINVAL; + goto err; + } + + tegra_rt5640_dai.cpu_of_node = of_parse_phandle(np, + "nvidia,i2s-controller", 0); + if (!tegra_rt5640_dai.cpu_of_node) { + dev_err(&pdev->dev, + "Property 'nvidia,i2s-controller' missing or invalid\n"); + ret = -EINVAL; + goto err; + } + + tegra_rt5640_dai.platform_of_node = tegra_rt5640_dai.cpu_of_node; + + ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev); + if (ret) + goto err; + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", + ret); + goto err_fini_utils; + } + + return 0; + +err_fini_utils: + tegra_asoc_utils_fini(&machine->util_data); +err: + return ret; +} + +static int tegra_rt5640_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct tegra_rt5640 *machine = snd_soc_card_get_drvdata(card); + + snd_soc_jack_free_gpios(&tegra_rt5640_hp_jack, 1, + &tegra_rt5640_hp_jack_gpio); + + snd_soc_unregister_card(card); + + tegra_asoc_utils_fini(&machine->util_data); + + return 0; +} + +static const struct of_device_id tegra_rt5640_of_match[] = { + { .compatible = "nvidia,tegra-audio-rt5640", }, + {}, +}; + +static struct platform_driver tegra_rt5640_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + .of_match_table = tegra_rt5640_of_match, + }, + .probe = tegra_rt5640_probe, + .remove = tegra_rt5640_remove, +}; +module_platform_driver(tegra_rt5640_driver); + +MODULE_AUTHOR("Stephen Warren <swarren(a)nvidia.com>"); +MODULE_DESCRIPTION("Tegra+RT5640 machine ASoC driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, tegra_rt5640_of_match); -- 1.8.1.5

2 1

[alsa-devel] [PATCH V2] ASoC: rt5640: add device tree support
by Stephen Warren 12 Jun '13

12 Jun '13

From: Stephen Warren <swarren(a)nvidia.com> Modify the RT5640 driver to parse platform data from device tree. Write a DT binding document to describe those properties. Slight re-ordering of rt5640_i2c_probe() to better fit the DT parsing. Since ldo1_en is optional, guard usage of it with gpio_is_valid(), rather than open-coding an if (gpio) check. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- v2: Treat ldo1_en==0 in pdata as an invalid (unspecified) GPIO. --- Documentation/devicetree/bindings/sound/rt5640.txt | 30 ++++++++++++++ sound/soc/codecs/rt5640.c | 48 +++++++++++++++++++--- 2 files changed, 72 insertions(+), 6 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/rt5640.txt diff --git a/Documentation/devicetree/bindings/sound/rt5640.txt b/Documentation/devicetree/bindings/sound/rt5640.txt new file mode 100644 index 0000000..005bcb2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt5640.txt @@ -0,0 +1,30 @@ +RT5640 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt5640". + +- reg : The I2C address of the device. + +- interrupts : The CODEC's interrupt output. + +Optional properties: + +- realtek,in1-differential +- realtek,in2-differential + Boolean. Indicate MIC1/2 input are differential, rather than single-ended. + +- realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin. + +Example: + +rt5640 { + compatible = "realtek,rt5640"; + reg = <0x1c>; + interrupt-parent = <&gpio>; + interrupts = <TEGRA_GPIO(W, 3) GPIO_ACTIVE_HIGH>; + realtek,ldo1-en-gpios = + <&gpio TEGRA_GPIO(V, 3) GPIO_ACTIVE_HIGH>; +}; diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 288c17c..8761552 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -18,6 +18,7 @@ #include <linux/gpio.h> #include <linux/i2c.h> #include <linux/regmap.h> +#include <linux/of_gpio.h> #include <linux/platform_device.h> #include <linux/spi/spi.h> #include <sound/core.h> @@ -1998,6 +1999,28 @@ static const struct i2c_device_id rt5640_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, rt5640_i2c_id); +static int rt5640_parse_dt(struct rt5640_priv *rt5640, struct device_node *np) +{ + rt5640->pdata.in1_diff = of_property_read_bool(np, + "realtek,in1-differential"); + rt5640->pdata.in2_diff = of_property_read_bool(np, + "realtek,in2-differential"); + + rt5640->pdata.ldo1_en = of_get_named_gpio(np, + "realtek,ldo1-en-gpios", 0); + /* + * LDO1_EN is optional (it may be statically tied on the board). + * -ENOENT means that the property doesn't exist, i.e. there is no + * GPIO, so is not an error. Any other error code means the property + * exists, but could not be parsed. + */ + if (!gpio_is_valid(rt5640->pdata.ldo1_en) && + (rt5640->pdata.ldo1_en != -ENOENT)) + return rt5640->pdata.ldo1_en; + + return 0; +} + static int rt5640_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -2011,6 +2034,24 @@ static int rt5640_i2c_probe(struct i2c_client *i2c, GFP_KERNEL); if (NULL == rt5640) return -ENOMEM; + i2c_set_clientdata(i2c, rt5640); + + if (pdata) { + rt5640->pdata = *pdata; + /* + * Translate zero'd out (default) pdata value to an invalid + * GPIO ID. This makes the pdata and DT paths consistent in + * terms of the value left in this field when no GPIO is + * specified, but means we can't actually use GPIO 0. + */ + if (!rt5640->pdata.ldo1_en) + rt5640->pdata.ldo1_en = -EINVAL; + } else if (i2c->dev.of_node) { + ret = rt5640_parse_dt(rt5640, i2c->dev.of_node); + if (ret) + return ret; + } else + rt5640->pdata.ldo1_en = -EINVAL; rt5640->regmap = devm_regmap_init_i2c(i2c, &rt5640_regmap); if (IS_ERR(rt5640->regmap)) { @@ -2020,12 +2061,7 @@ static int rt5640_i2c_probe(struct i2c_client *i2c, return ret; } - if (pdata) - rt5640->pdata = *pdata; - - i2c_set_clientdata(i2c, rt5640); - - if (rt5640->pdata.ldo1_en) { + if (gpio_is_valid(rt5640->pdata.ldo1_en)) { ret = devm_gpio_request_one(&i2c->dev, rt5640->pdata.ldo1_en, GPIOF_OUT_INIT_HIGH, "RT5640 LDO1_EN"); -- 1.8.1.5

2 1

Re: [alsa-devel] [PATCH 1/3] mfd: arizona: Integrate wm8997 into Arizona mfd
by Mark Brown 12 Jun '13

12 Jun '13

On Wed, Jun 12, 2013 at 05:29:18PM +0100, Charles Keepax wrote: > The wm8997 is a compact, high-performance audio hub CODEC with SLIMbus > interfacing, for smartphones, tablets and other portable audio devices > based on the Arizona platform. > > This patch integrates the wm8997 into the Arizona mfd. There should be an update to add the OF device ID to the driver in here.

1 0

[alsa-devel] [PATCH] ASoC: rt5640: add device tree support
by Stephen Warren 12 Jun '13

12 Jun '13

From: Stephen Warren <swarren(a)nvidia.com> Modify the RT5640 driver to parse platform data from device tree. Write a DT binding document to describe those properties. Slight re-ordering of rt5640_i2c_probe() to better fit the DT parsing. Since ldo1_en is optional, guard usage of it with gpio_is_valid(), rather than open-coding an if (gpio) check. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- Mark, If the underlying rt5640 driver is reposted, this might need a new version to rebase it, but otherwise I believe should be complete. Bard, FYI you can find this patch and the DT changes for Dalmore at: git://nv-tegra.nvidia.com/user/swarren/linux-2.6 linux-next_common Documentation/devicetree/bindings/sound/rt5640.txt | 30 ++++++++++++++++ sound/soc/codecs/rt5640.c | 40 ++++++++++++++++++---- 2 files changed, 64 insertions(+), 6 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/rt5640.txt diff --git a/Documentation/devicetree/bindings/sound/rt5640.txt b/Documentation/devicetree/bindings/sound/rt5640.txt new file mode 100644 index 0000000..005bcb2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt5640.txt @@ -0,0 +1,30 @@ +RT5640 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt5640". + +- reg : The I2C address of the device. + +- interrupts : The CODEC's interrupt output. + +Optional properties: + +- realtek,in1-differential +- realtek,in2-differential + Boolean. Indicate MIC1/2 input are differential, rather than single-ended. + +- realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin. + +Example: + +rt5640 { + compatible = "realtek,rt5640"; + reg = <0x1c>; + interrupt-parent = <&gpio>; + interrupts = <TEGRA_GPIO(W, 3) GPIO_ACTIVE_HIGH>; + realtek,ldo1-en-gpios = + <&gpio TEGRA_GPIO(V, 3) GPIO_ACTIVE_HIGH>; +}; diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 288c17c..f7e73d2 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -18,6 +18,7 @@ #include <linux/gpio.h> #include <linux/i2c.h> #include <linux/regmap.h> +#include <linux/of_gpio.h> #include <linux/platform_device.h> #include <linux/spi/spi.h> #include <sound/core.h> @@ -1998,6 +1999,28 @@ static const struct i2c_device_id rt5640_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, rt5640_i2c_id); +static int rt5640_parse_dt(struct rt5640_priv *rt5640, struct device_node *np) +{ + rt5640->pdata.in1_diff = of_property_read_bool(np, + "realtek,in1-differential"); + rt5640->pdata.in2_diff = of_property_read_bool(np, + "realtek,in2-differential"); + + rt5640->pdata.ldo1_en = of_get_named_gpio(np, + "realtek,ldo1-en-gpios", 0); + /* + * LDO1_EN is optional (it may be statically tied on the board). + * -ENOENT means that the property doesn't exist, i.e. there is no + * GPIO, so is not an error. Any other error code means the property + * exists, but could not be parsed. + */ + if (!gpio_is_valid(rt5640->pdata.ldo1_en) && + (rt5640->pdata.ldo1_en != -ENOENT)) + return rt5640->pdata.ldo1_en; + + return 0; +} + static int rt5640_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -2011,6 +2034,16 @@ static int rt5640_i2c_probe(struct i2c_client *i2c, GFP_KERNEL); if (NULL == rt5640) return -ENOMEM; + i2c_set_clientdata(i2c, rt5640); + + if (pdata) + rt5640->pdata = *pdata; + else if (i2c->dev.of_node) { + ret = rt5640_parse_dt(rt5640, i2c->dev.of_node); + if (ret) + return ret; + } else + rt5640->pdata.ldo1_en = -1; rt5640->regmap = devm_regmap_init_i2c(i2c, &rt5640_regmap); if (IS_ERR(rt5640->regmap)) { @@ -2020,12 +2053,7 @@ static int rt5640_i2c_probe(struct i2c_client *i2c, return ret; } - if (pdata) - rt5640->pdata = *pdata; - - i2c_set_clientdata(i2c, rt5640); - - if (rt5640->pdata.ldo1_en) { + if (gpio_is_valid(rt5640->pdata.ldo1_en)) { ret = devm_gpio_request_one(&i2c->dev, rt5640->pdata.ldo1_en, GPIOF_OUT_INIT_HIGH, "RT5640 LDO1_EN"); -- 1.8.1.5

2 3

Re: [alsa-devel] [PATCH] ASoC: add RT5640 CODEC driver
by Mark Brown 12 Jun '13

12 Jun '13

On Tue, Jun 11, 2013 at 01:10:16PM +0800, bardliao(a)realtek.com wrote: > From: Bard Liao <bardliao(a)realtek.com> > > This patch adds the ALC5640 codec driver. I'll apply this on the basis that there's going to be some followup patches fixing the outstanding issues.

1 0

Re: [alsa-devel] [PATCH v2] sound/sgtl5000: fix codec register initial values and mask
by Fabio Estevam 12 Jun '13

12 Jun '13

On 06/11/2013 12:22 PM, Oskar Schirmer wrote: > According to documentation bit 3:2 in register SSS_CTRL are > reserved and zero, so initially setting the register to 0x0008 > does not make much sense. Instead, bit 4 should be marked set, > as this is the power up default. > > Further, mask computation in declarative part is obviously wrong: > Fix FRAC DIVISOR to provide an 11 bit mask correctly. > > Signed-off-by: Oskar Schirmer <oskar(a)scara.com> > Tested-by: Fabio Estevam <fabio.estevam(a)freescale.com> Looks good, but please copy Mark Brown. Just added him in Cc now. > --- > sound/soc/codecs/sgtl5000.c | 2 +- > sound/soc/codecs/sgtl5000.h | 2 +- > 2 files changed, 2 insertions(+), 2 deletions(-) > > diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c > index d441559..10092ba 100644 > --- a/sound/soc/codecs/sgtl5000.c > +++ b/sound/soc/codecs/sgtl5000.c > @@ -38,7 +38,7 @@ > static const struct reg_default sgtl5000_reg_defaults[] = { > { SGTL5000_CHIP_CLK_CTRL, 0x0008 }, > { SGTL5000_CHIP_I2S_CTRL, 0x0010 }, > - { SGTL5000_CHIP_SSS_CTRL, 0x0008 }, > + { SGTL5000_CHIP_SSS_CTRL, 0x0010 }, > { SGTL5000_CHIP_DAC_VOL, 0x3c3c }, > { SGTL5000_CHIP_PAD_STRENGTH, 0x015f }, > { SGTL5000_CHIP_ANA_HP_CTRL, 0x1818 }, > diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h > index 4b69229..52bd843 100644 > --- a/sound/soc/codecs/sgtl5000.h > +++ b/sound/soc/codecs/sgtl5000.h > @@ -347,7 +347,7 @@ > #define SGTL5000_PLL_INT_DIV_MASK 0xf800 > #define SGTL5000_PLL_INT_DIV_SHIFT 11 > #define SGTL5000_PLL_INT_DIV_WIDTH 5 > -#define SGTL5000_PLL_FRAC_DIV_MASK 0x0700 > +#define SGTL5000_PLL_FRAC_DIV_MASK 0x07ff > #define SGTL5000_PLL_FRAC_DIV_SHIFT 0 > #define SGTL5000_PLL_FRAC_DIV_WIDTH 11 > >

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[alsa-devel] [PATCH 0/3] DMA fixes for ux500 ASoC driver
by Fabio Baltieri 12 Jun '13

12 Jun '13

Hi Mark, The DMA driver used on ux500 has been recently reworked to make it device-tree capable, but the rework required to drop some "features", such as opportunistic channel allocation and hardcoded addresses, that the ASoC driver was relying upon. This series fixes the audio driver to make it work with the current dma40 implementation, and works fine even when booting with device-tree. Would you consider merging these for asoc/topic/ux500? Thanks, Fabio Fabio Baltieri (3): ASoC: ux500: Move DMA parameters into ux500_msp ASoC: ux500: Set DMA address during device init ASoC: ux500: Add DMA slave config prepare routine sound/soc/ux500/ux500_msp_dai.c | 11 ++++------- sound/soc/ux500/ux500_msp_dai.h | 2 -- sound/soc/ux500/ux500_msp_i2s.c | 13 +++++++++---- sound/soc/ux500/ux500_msp_i2s.h | 15 ++++++++------- sound/soc/ux500/ux500_pcm.c | 30 ++++++++++++++++++++++++++++++ 5 files changed, 51 insertions(+), 20 deletions(-) -- 1.8.2

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Re: [alsa-devel] [PATCH] ASoC: add RT5640 CODEC driver
by Stephen Warren 12 Jun '13

12 Jun '13

On 06/10/2013 11:10 PM, bardliao(a)realtek.com wrote: > From: Bard Liao <bardliao(a)realtek.com> > > This patch adds the ALC5640 codec driver. > diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c > +static int rt5640_i2c_probe(struct i2c_client *i2c, > + rt5640->regmap = devm_regmap_init_i2c(i2c, &rt5640_regmap); > + if (rt5640->pdata.ldo1_en) { > + ret = devm_gpio_request_one(&i2c->dev, rt5640->pdata.ldo1_en, > + GPIOF_OUT_INIT_HIGH, > + "RT5640 LDO1_EN"); > + if (ret < 0) { > + dev_err(&i2c->dev, "Failed to request LDO1_EN %d: %d\n", > + rt5640->pdata.ldo1_en, ret); > + return ret; > + } > + msleep(400); > + } Oh I see this is the only place ldo1_en is touched. I had assumed you were going to add code to turn it off/on based on bias level. That's why I had asked you to add that feature, since you'd know any HW requirements for doing that. Still, as Mark mentioned, that can certainly be added later. One question though: Don't you want to initially enable ldo1_en before you create the regmap? At least some regmap_init() calls end up trying to read from the device to populate the register cache, and that won't work until ldo1_en is active. Anyway, I'll go test this again, and work on the patch to pull the platform data from DT.

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[alsa-devel] [PATCH V6 0/6] clk: Samsung: audss: Register audio subsytem clocks using common clk framework
by Padmavathi Venna 12 Jun '13

12 Jun '13

Samsung S5PV210 and Exynos SoC has a separate subsystem for audio. This subsystem has a internal clock controller which controls i2s0 and pcm0 clocks. This patch series adds the Samsung audio subsytem clock to the common clock framework and provides the I2S controllers clock information in the dtsi file. This patch series is made based on Kukjin Kim for-next branch Changes since V5: - Added i2s_opclk0 entry for i2s1 and i2s2 nodes - Added Acked-by: Mike Turquette <mturquette(a)linaro.org> - Added Reviewed-by: Doug Anderson <dianders(a)chromium.org> Changes since V4: - Reworked on the nits given by Doug. - Removed mout_audss and mout_i2s from i2s nodes as we are not getting these clocks in the i2s driver. - Modified the I2S binding documentation for clocks and pinmux. Changes since V3: - Replaced samsung with exynos in the macro prefixes and function names as this driver supports mainly exynos and s5p family. - Added Reviewed-by:Sylwester Nawrocki <s.nawrocki(a)samsung.com> Changes since V2: - Removed s5pv210 compatible name from driver as it is not yet supported which is different from Exynos series audio subsystem clock conroller. - Removed clkdev lookup support and added alias names in the i2s0 controller node. Changes since V1: - Reworked on all review comments by Sylwester Nawrocki - Added a header file for all clock indexes as requested by Sylwester - Added different compatible names for s5pv210, exynos4 and exynos5 - Registered the pcm clocks with common clock framework Padmavathi Venna (6): ARM: samsung: use #include for all device trees clk: samsung: register audio subsystem clocks using common clock framework ARM: dts: add Exynos audio subsystem clock controller node ARM: dts: add clock provider information for i2s controllers in Exynos5250 ARM: dts: Update Samsung I2S documentation clk: exynos5250: Add enum entries for divider clock of i2s1 and i2s2 .../devicetree/bindings/clock/clk-exynos-audss.txt | 64 ++++++++++ .../devicetree/bindings/sound/samsung-i2s.txt | 46 +++---- arch/arm/boot/dts/exynos4.dtsi | 2 +- arch/arm/boot/dts/exynos4210-origen.dts | 2 +- arch/arm/boot/dts/exynos4210-smdkv310.dts | 2 +- arch/arm/boot/dts/exynos4210-trats.dts | 2 +- arch/arm/boot/dts/exynos4210-universal_c210.dts | 2 +- arch/arm/boot/dts/exynos4210.dtsi | 4 +- arch/arm/boot/dts/exynos4212.dtsi | 2 +- arch/arm/boot/dts/exynos4412-odroidx.dts | 2 +- arch/arm/boot/dts/exynos4412-origen.dts | 2 +- arch/arm/boot/dts/exynos4412-smdk4412.dts | 2 +- arch/arm/boot/dts/exynos4412.dtsi | 2 +- arch/arm/boot/dts/exynos4x12.dtsi | 4 +- arch/arm/boot/dts/exynos5250-arndale.dts | 2 +- arch/arm/boot/dts/exynos5250-smdk5250.dts | 2 +- arch/arm/boot/dts/exynos5250-snow.dts | 4 +- arch/arm/boot/dts/exynos5250.dtsi | 20 +++- arch/arm/boot/dts/exynos5440-sd5v1.dts | 2 +- arch/arm/boot/dts/exynos5440-ssdk5440.dts | 2 +- arch/arm/boot/dts/exynos5440.dtsi | 2 +- arch/arm/boot/dts/s3c2416-smdk2416.dts | 2 +- arch/arm/boot/dts/s3c2416.dtsi | 4 +- arch/arm/boot/dts/s3c24xx.dtsi | 2 +- drivers/clk/samsung/Makefile | 1 + drivers/clk/samsung/clk-exynos-audss.c | 133 ++++++++++++++++++++ drivers/clk/samsung/clk-exynos5250.c | 5 +- include/dt-bindings/clk/exynos-audss-clk.h | 25 ++++ 28 files changed, 288 insertions(+), 56 deletions(-) create mode 100644 Documentation/devicetree/bindings/clock/clk-exynos-audss.txt create mode 100644 drivers/clk/samsung/clk-exynos-audss.c create mode 100644 include/dt-bindings/clk/exynos-audss-clk.h -- 1.7.4.4

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Re: [alsa-devel] [PATCH] ASoC: add RT5640 CODEC driver
by Mark Brown 12 Jun '13

12 Jun '13

On Tue, Jun 11, 2013 at 01:10:16PM +0800, bardliao(a)realtek.com wrote: > Signed-off-by: Bard Liao <bardliao(a)realtek.com> > Signed-off-by: Stephen Warren <swarren(a)nvidia.com> I don't think that signoff for Stephen is right... > +static int rt5640_set_bias_level(struct snd_soc_codec *codec, > + enum snd_soc_bias_level level) > +{ > + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); > + switch (level) { > + case SND_SOC_BIAS_STANDBY: > + if (SND_SOC_BIAS_OFF == codec->dapm.bias_level) { > + regcache_cache_only(rt5640->regmap, false); You mark the device cache only in suspend so for symmetry this should really be in _BIAS_OFF, especially since the device is idle_bias_off so it'll be able to power off without a full suspend. > + if (rt5640->pdata.ldo1_en) { > + ret = devm_gpio_request_one(&i2c->dev, rt5640->pdata.ldo1_en, > + GPIOF_OUT_INIT_HIGH, > + "RT5640 LDO1_EN"); > + if (ret < 0) { > + dev_err(&i2c->dev, "Failed to request LDO1_EN %d: %d\n", > + rt5640->pdata.ldo1_en, ret); > + return ret; > + } > + msleep(400); > + } I'd expect to see some handling fo the LDO over suspend but that can be dealt with in a followup as previously discussed. Ideally we'd be able to power off when in _BIAS_OFF too but if it's going to take 400ms to ramp the LDO (which does seem high) that might be an issue.

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Re: [alsa-devel] [PATCH V5 2/5] clk: samsung: register audio subsystem clocks using common clock framework
by Padma Venkat 12 Jun '13

12 Jun '13

Hi Mike, On Wed, Jun 12, 2013 at 3:43 AM, Mike Turquette <mturquette(a)linaro.org> wrote: > Quoting Padmavathi Venna (2013-06-04 05:28:07) >> Audio subsystem is introduced in s5pv210 and exynos platforms. >> This has seperate clock controller which can control i2s0 and >> pcm0 clocks. This patch registers the audio subsystem clocks >> with the common clock framework on Exynos family. >> >> Signed-off-by: Padmavathi Venna <padma.v(a)samsung.com> >> Reviewed-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> > > This looks good to me. You have my Ack, or I can take this patch > through clk-next. Let me know what works for you. I have to rework for one small change in one of the patch in the set. So I will do that and put your ack and will repost the patches. After that you can take the patches through clk-next. Thanks Padma > > Regards, > Mike > >> --- >> .../devicetree/bindings/clock/clk-exynos-audss.txt | 64 ++++++++++ >> drivers/clk/samsung/Makefile | 1 + >> drivers/clk/samsung/clk-exynos-audss.c | 133 ++++++++++++++++++++ >> include/dt-bindings/clk/exynos-audss-clk.h | 25 ++++ >> 4 files changed, 223 insertions(+), 0 deletions(-) >> create mode 100644 Documentation/devicetree/bindings/clock/clk-exynos-audss.txt >> create mode 100644 drivers/clk/samsung/clk-exynos-audss.c >> create mode 100644 include/dt-bindings/clk/exynos-audss-clk.h >> >> diff --git a/Documentation/devicetree/bindings/clock/clk-exynos-audss.txt b/Documentation/devicetree/bindings/clock/clk-exynos-audss.txt >> new file mode 100644 >> index 0000000..a120180 >> --- /dev/null >> +++ b/Documentation/devicetree/bindings/clock/clk-exynos-audss.txt >> @@ -0,0 +1,64 @@ >> +* Samsung Audio Subsystem Clock Controller >> + >> +The Samsung Audio Subsystem clock controller generates and supplies clocks >> +to Audio Subsystem block available in the S5PV210 and Exynos SoCs. The clock >> +binding described here is applicable to all SoC's in Exynos family. >> + >> +Required Properties: >> + >> +- compatible: should be one of the following: >> + - "samsung,exynos4210-audss-clock" - controller compatible with all Exynos4 SoCs. >> + - "samsung,exynos5250-audss-clock" - controller compatible with all Exynos5 SoCs. >> + >> +- reg: physical base address and length of the controller's register set. >> + >> +- #clock-cells: should be 1. >> + >> +The following is the list of clocks generated by the controller. Each clock is >> +assigned an identifier and client nodes use this identifier to specify the >> +clock which they consume. Some of the clocks are available only on a particular >> +Exynos4 SoC and this is specified where applicable. >> + >> +Provided clocks: >> + >> +Clock ID SoC (if specific) >> +----------------------------------------------- >> + >> +mout_audss 0 >> +mout_i2s 1 >> +dout_srp 2 >> +dout_aud_bus 3 >> +dout_i2s 4 >> +srp_clk 5 >> +i2s_bus 6 >> +sclk_i2s 7 >> +pcm_bus 8 >> +sclk_pcm 9 >> + >> +Example 1: An example of a clock controller node is listed below. >> + >> +clock_audss: audss-clock-controller@3810000 { >> + compatible = "samsung,exynos5250-audss-clock"; >> + reg = <0x03810000 0x0C>; >> + #clock-cells = <1>; >> +}; >> + >> +Example 2: I2S controller node that consumes the clock generated by the clock >> + controller. Refer to the standard clock bindings for information >> + about 'clocks' and 'clock-names' property. >> + >> +i2s0: i2s@03830000 { >> + compatible = "samsung,i2s-v5"; >> + reg = <0x03830000 0x100>; >> + dmas = <&pdma0 10 >> + &pdma0 9 >> + &pdma0 8>; >> + dma-names = "tx", "rx", "tx-sec"; >> + clocks = <&clock_audss EXYNOS_I2S_BUS>, >> + <&clock_audss EXYNOS_I2S_BUS>, >> + <&clock_audss EXYNOS_SCLK_I2S>, >> + <&clock_audss EXYNOS_MOUT_AUDSS>, >> + <&clock_audss EXYNOS_MOUT_I2S>; >> + clock-names = "iis", "i2s_opclk0", "i2s_opclk1", >> + "mout_audss", "mout_i2s"; >> +}; >> diff --git a/drivers/clk/samsung/Makefile b/drivers/clk/samsung/Makefile >> index b7c232e..1876810 100644 >> --- a/drivers/clk/samsung/Makefile >> +++ b/drivers/clk/samsung/Makefile >> @@ -6,3 +6,4 @@ obj-$(CONFIG_COMMON_CLK) += clk.o clk-pll.o >> obj-$(CONFIG_ARCH_EXYNOS4) += clk-exynos4.o >> obj-$(CONFIG_SOC_EXYNOS5250) += clk-exynos5250.o >> obj-$(CONFIG_SOC_EXYNOS5440) += clk-exynos5440.o >> +obj-$(CONFIG_ARCH_EXYNOS) += clk-exynos-audss.o >> diff --git a/drivers/clk/samsung/clk-exynos-audss.c b/drivers/clk/samsung/clk-exynos-audss.c >> new file mode 100644 >> index 0000000..9b1bbd5 >> --- /dev/null >> +++ b/drivers/clk/samsung/clk-exynos-audss.c >> @@ -0,0 +1,133 @@ >> +/* >> + * Copyright (c) 2013 Samsung Electronics Co., Ltd. >> + * Author: Padmavathi Venna <padma.v(a)samsung.com> >> + * >> + * This program is free software; you can redistribute it and/or modify >> + * it under the terms of the GNU General Public License version 2 as >> + * published by the Free Software Foundation. >> + * >> + * Common Clock Framework support for Audio Subsystem Clock Controller. >> +*/ >> + >> +#include <linux/clkdev.h> >> +#include <linux/io.h> >> +#include <linux/clk-provider.h> >> +#include <linux/of_address.h> >> +#include <linux/syscore_ops.h> >> + >> +#include <dt-bindings/clk/exynos-audss-clk.h> >> + >> +static DEFINE_SPINLOCK(lock); >> +static struct clk **clk_table; >> +static void __iomem *reg_base; >> +static struct clk_onecell_data clk_data; >> + >> +#define ASS_CLK_SRC 0x0 >> +#define ASS_CLK_DIV 0x4 >> +#define ASS_CLK_GATE 0x8 >> + >> +static unsigned long reg_save[][2] = { >> + {ASS_CLK_SRC, 0}, >> + {ASS_CLK_DIV, 0}, >> + {ASS_CLK_GATE, 0}, >> +}; >> + >> +/* list of all parent clock list */ >> +static const char *mout_audss_p[] = { "fin_pll", "fout_epll" }; >> +static const char *mout_i2s_p[] = { "mout_audss", "cdclk0", "sclk_audio0" }; >> + >> +#ifdef CONFIG_PM_SLEEP >> +static int exynos_audss_clk_suspend(void) >> +{ >> + int i; >> + >> + for (i = 0; i < ARRAY_SIZE(reg_save); i++) >> + reg_save[i][1] = readl(reg_base + reg_save[i][0]); >> + >> + return 0; >> +} >> + >> +static void exynos_audss_clk_resume(void) >> +{ >> + int i; >> + >> + for (i = 0; i < ARRAY_SIZE(reg_save); i++) >> + writel(reg_save[i][1], reg_base + reg_save[i][0]); >> +} >> + >> +static struct syscore_ops exynos_audss_clk_syscore_ops = { >> + .suspend = exynos_audss_clk_suspend, >> + .resume = exynos_audss_clk_resume, >> +}; >> +#endif /* CONFIG_PM_SLEEP */ >> + >> +/* register exynos_audss clocks */ >> +void __init exynos_audss_clk_init(struct device_node *np) >> +{ >> + reg_base = of_iomap(np, 0); >> + if (!reg_base) { >> + pr_err("%s: failed to map audss registers\n", __func__); >> + return; >> + } >> + >> + clk_table = kzalloc(sizeof(struct clk *) * EXYNOS_AUDSS_MAX_CLKS, >> + GFP_KERNEL); >> + if (!clk_table) { >> + pr_err("%s: could not allocate clk lookup table\n", __func__); >> + return; >> + } >> + >> + clk_data.clks = clk_table; >> + clk_data.clk_num = EXYNOS_AUDSS_MAX_CLKS; >> + of_clk_add_provider(np, of_clk_src_onecell_get, &clk_data); >> + >> + clk_table[EXYNOS_MOUT_AUDSS] = clk_register_mux(NULL, "mout_audss", >> + mout_audss_p, ARRAY_SIZE(mout_audss_p), 0, >> + reg_base + ASS_CLK_SRC, 0, 1, 0, &lock); >> + >> + clk_table[EXYNOS_MOUT_I2S] = clk_register_mux(NULL, "mout_i2s", >> + mout_i2s_p, ARRAY_SIZE(mout_i2s_p), 0, >> + reg_base + ASS_CLK_SRC, 2, 2, 0, &lock); >> + >> + clk_table[EXYNOS_DOUT_SRP] = clk_register_divider(NULL, "dout_srp", >> + "mout_audss", 0, reg_base + ASS_CLK_DIV, 0, 4, >> + 0, &lock); >> + >> + clk_table[EXYNOS_DOUT_AUD_BUS] = clk_register_divider(NULL, >> + "dout_aud_bus", "dout_srp", 0, >> + reg_base + ASS_CLK_DIV, 4, 4, 0, &lock); >> + >> + clk_table[EXYNOS_DOUT_I2S] = clk_register_divider(NULL, "dout_i2s", >> + "mout_i2s", 0, reg_base + ASS_CLK_DIV, 8, 4, 0, >> + &lock); >> + >> + clk_table[EXYNOS_SRP_CLK] = clk_register_gate(NULL, "srp_clk", >> + "dout_srp", CLK_SET_RATE_PARENT, >> + reg_base + ASS_CLK_GATE, 0, 0, &lock); >> + >> + clk_table[EXYNOS_I2S_BUS] = clk_register_gate(NULL, "i2s_bus", >> + "dout_aud_bus", CLK_SET_RATE_PARENT, >> + reg_base + ASS_CLK_GATE, 2, 0, &lock); >> + >> + clk_table[EXYNOS_SCLK_I2S] = clk_register_gate(NULL, "sclk_i2s", >> + "dout_i2s", CLK_SET_RATE_PARENT, >> + reg_base + ASS_CLK_GATE, 3, 0, &lock); >> + >> + clk_table[EXYNOS_PCM_BUS] = clk_register_gate(NULL, "pcm_bus", >> + "sclk_pcm", CLK_SET_RATE_PARENT, >> + reg_base + ASS_CLK_GATE, 4, 0, &lock); >> + >> + clk_table[EXYNOS_SCLK_PCM] = clk_register_gate(NULL, "sclk_pcm", >> + "div_pcm0", CLK_SET_RATE_PARENT, >> + reg_base + ASS_CLK_GATE, 5, 0, &lock); >> + >> +#ifdef CONFIG_PM_SLEEP >> + register_syscore_ops(&exynos_audss_clk_syscore_ops); >> +#endif >> + >> + pr_info("Exynos: Audss: clock setup completed\n"); >> +} >> +CLK_OF_DECLARE(exynos4210_audss_clk, "samsung,exynos4210-audss-clock", >> + exynos_audss_clk_init); >> +CLK_OF_DECLARE(exynos5250_audss_clk, "samsung,exynos5250-audss-clock", >> + exynos_audss_clk_init); >> diff --git a/include/dt-bindings/clk/exynos-audss-clk.h b/include/dt-bindings/clk/exynos-audss-clk.h >> new file mode 100644 >> index 0000000..8279f42 >> --- /dev/null >> +++ b/include/dt-bindings/clk/exynos-audss-clk.h >> @@ -0,0 +1,25 @@ >> +/* >> + * This header provides constants for Samsung audio subsystem >> + * clock controller. >> + * >> + * The constants defined in this header are being used in dts >> + * and exynos audss driver. >> + */ >> + >> +#ifndef _DT_BINDINGS_CLK_EXYNOS_AUDSS_H >> +#define _DT_BINDINGS_CLK_EXYNOS_AUDSS_H >> + >> +#define EXYNOS_MOUT_AUDSS 0 >> +#define EXYNOS_MOUT_I2S 1 >> +#define EXYNOS_DOUT_SRP 2 >> +#define EXYNOS_DOUT_AUD_BUS 3 >> +#define EXYNOS_DOUT_I2S 4 >> +#define EXYNOS_SRP_CLK 5 >> +#define EXYNOS_I2S_BUS 6 >> +#define EXYNOS_SCLK_I2S 7 >> +#define EXYNOS_PCM_BUS 8 >> +#define EXYNOS_SCLK_PCM 9 >> + >> +#define EXYNOS_AUDSS_MAX_CLKS 10 >> + >> +#endif >> -- >> 1.7.4.4

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Re: [alsa-devel] [PATCH] ASoC: add RT5640 CODEC driver
by Stephen Warren 11 Jun '13

11 Jun '13

On 06/10/2013 11:10 PM, bardliao(a)realtek.com wrote: > From: Bard Liao <bardliao(a)realtek.com> > > This patch adds the ALC5640 codec driver. > > Signed-off-by: Bard Liao <bardliao(a)realtek.com> > Signed-off-by: Stephen Warren <swarren(a)nvidia.com> (Oh, I suppose those two lines above should really be swapped) Tested-by: Stephen Warren <swarren(a)nvidia.com> This time, both headphones /and/ speakers worked for me:-) I haven't tested capture yet though.

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Re: [alsa-devel] [PATCH] sound/sgtl5000: fix codec register initial values and mask
by Fabio Estevam 11 Jun '13

11 Jun '13

Hi Oskar, On 06/11/2013 05:12 AM, Oskar Schirmer wrote: > According to documentation bit 3:2 in register SSS_CTRL are > reserved and zero, so initially setting the register to 0x0008 > does not make much sense. Instead, bit 4 should be marked set, > as this is the power up default. > > Further, mask computation in declarative part is obviously wrong: > Fix FRAC DIVISOR to provide an 11 bit mask correctly. > > Signed-off-by: Oskar Schirmer <oskar(a)scara.com> Please rebase your patch against Mark Brown's topic/sgtl5000 branch: https://git.kernel.org/cgit/linux/kernel/git/broonie/sound.git/log/?h=topic… After that, you can add my: Tested-by: Fabio Estevam <fabio.estevam(a)freescale.com>

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[alsa-devel] [PATCH V6] ASoC: fsl: add imx-wm8962 machine driver
by Nicolin Chen 11 Jun '13

11 Jun '13

This is the initial imx-wm8962 device-tree-only machine driver working with fsl_ssi driver. More features can be added on top of it later. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- ChangeLog: v1->v2: * Fixed a typo in probe(): *** data->dai.ops = &imx_hifi_ops, ==> data->dai.ops = &imx_hifi_ops; v2->v3: * Moved Headphone/Microphone jack detection GPIO port to Optional properties. * Use set_bias_level() instead of hw_params/hw_free() to open/close FLL. * Use devm_clk_get() instead of clk_get(). * Dropped clock-frequency grabbing code. * Fixed a typo: *** data->card.late_probe = imx_wm8962_late_probe, *** ==> *** data->card.late_probe = imx_wm8962_late_probe; v3->v4: * Revised binding doc to describe two gpio pins more clearly. v4->v5: * Added return value checking for clk_prepare_enable(). * Revised error message for devm_clk_get() to keep same pattern as above. v5->v6: * Removed un-used members from struct imx_priv. * Removed un-needed head files. * Used dev_err instead of pr_err in set_bias_level(). * Droped jack detection gpio pins in binding doc. .../devicetree/bindings/sound/imx-audio-wm8962.txt | 46 +++ sound/soc/fsl/Kconfig | 12 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-wm8962.c | 323 ++++++++++++++++++++ 4 files changed, 383 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt create mode 100644 sound/soc/fsl/imx-wm8962.c diff --git a/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt b/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt new file mode 100644 index 0000000..f49450a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt @@ -0,0 +1,46 @@ +Freescale i.MX audio complex with WM8962 codec + +Required properties: +- compatible : "fsl,imx-audio-wm8962" +- model : The user-visible name of this sound complex +- ssi-controller : The phandle of the i.MX SSI controller +- audio-codec : The phandle of the WM8962 audio codec +- audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names could be power + supplies, WM8962 pins, and the jacks on the board: + + Power supplies: + * Mic Bias + + Board connectors: + * Mic Jack + * Headphone Jack + * Ext Spk + +- mux-int-port : The internal port of the i.MX audio muxer (AUDMUX) +- mux-ext-port : The external port of the i.MX audio muxer + +Note: The AUDMUX port numbering should start at 1, which is consistent with +hardware manual. + +Example: + +sound { + compatible = "fsl,imx6q-sabresd-wm8962", + "fsl,imx-audio-wm8962"; + model = "wm8962-audio"; + ssi-controller = <&ssi2>; + audio-codec = <&codec>; + audio-routing = + "Headphone Jack", "HPOUTL", + "Headphone Jack", "HPOUTR", + "Ext Spk", "SPKOUTL", + "Ext Spk", "SPKOUTR", + "MICBIAS", "AMIC", + "IN3R", "MICBIAS", + "DMIC", "MICBIAS", + "DMICDAT", "DMIC"; + mux-int-port = <2>; + mux-ext-port = <3>; +}; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 7860cc2..aa43854 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -168,6 +168,18 @@ config SND_SOC_EUKREA_TLV320 Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface +config SND_SOC_IMX_WM8962 + tristate "SoC Audio support for i.MX boards with wm8962" + depends on OF && I2C + select SND_SOC_WM8962 + select SND_SOC_IMX_PCM_DMA + select SND_SOC_IMX_AUDMUX + select SND_SOC_FSL_SSI + select SND_SOC_FSL_UTILS + help + Say Y if you want to add support for SoC audio on an i.MX board with + a wm8962 codec. + config SND_SOC_IMX_SGTL5000 tristate "SoC Audio support for i.MX boards with sgtl5000" depends on OF && I2C diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 91883f8..d4b4aa8 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -42,6 +42,7 @@ snd-soc-phycore-ac97-objs := phycore-ac97.o snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o +snd-soc-imx-wm8962-objs := imx-wm8962.o snd-soc-imx-mc13783-objs := imx-mc13783.o obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o @@ -49,4 +50,5 @@ obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o +obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c new file mode 100644 index 0000000..52a36a9 --- /dev/null +++ b/sound/soc/fsl/imx-wm8962.c @@ -0,0 +1,323 @@ +/* + * Copyright 2013 Freescale Semiconductor, Inc. + * + * Based on imx-sgtl5000.c + * Copyright 2012 Freescale Semiconductor, Inc. + * Copyright 2012 Linaro Ltd. + * + * The code contained herein is licensed under the GNU General Public + * License. You may obtain a copy of the GNU General Public License + * Version 2 or later at the following locations: + * + * http://www.opensource.org/licenses/gpl-license.html + * http://www.gnu.org/copyleft/gpl.html + */ + +#include <linux/module.h> +#include <linux/of_platform.h> +#include <linux/of_i2c.h> +#include <linux/slab.h> +#include <linux/clk.h> +#include <sound/soc.h> +#include <sound/pcm_params.h> +#include <sound/soc-dapm.h> +#include <linux/pinctrl/consumer.h> + +#include "../codecs/wm8962.h" +#include "imx-audmux.h" + +#define DAI_NAME_SIZE 32 + +struct imx_wm8962_data { + struct snd_soc_dai_link dai; + struct snd_soc_card card; + char codec_dai_name[DAI_NAME_SIZE]; + char platform_name[DAI_NAME_SIZE]; + struct clk *codec_clk; + unsigned int clk_frequency; +}; + +struct imx_priv { + struct platform_device *pdev; +}; +static struct imx_priv card_priv; + +static const struct snd_soc_dapm_widget imx_wm8962_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_MIC("AMIC", NULL), + SND_SOC_DAPM_MIC("DMIC", NULL), +}; + +static int sample_rate = 44100; +static snd_pcm_format_t sample_format = SNDRV_PCM_FORMAT_S16_LE; + +static int imx_hifi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + sample_rate = params_rate(params); + sample_format = params_format(params); + + return 0; +} + +static struct snd_soc_ops imx_hifi_ops = { + .hw_params = imx_hifi_hw_params, +}; + +static int imx_wm8962_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct imx_priv *priv = &card_priv; + struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + struct device *dev = &priv->pdev->dev; + unsigned int pll_out; + int ret; + + if (dapm->dev != codec_dai->dev) + return 0; + + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { + if (sample_format == SNDRV_PCM_FORMAT_S24_LE) + pll_out = sample_rate * 384; + else + pll_out = sample_rate * 256; + + ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, + WM8962_FLL_MCLK, data->clk_frequency, + pll_out); + if (ret < 0) { + dev_err(dev, "failed to start FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, + WM8962_SYSCLK_FLL, pll_out, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(dev, "failed to set SYSCLK: %d\n", ret); + return ret; + } + } + break; + + case SND_SOC_BIAS_STANDBY: + if (dapm->bias_level == SND_SOC_BIAS_PREPARE) { + ret = snd_soc_dai_set_sysclk(codec_dai, + WM8962_SYSCLK_MCLK, data->clk_frequency, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(dev, + "failed to switch away from FLL: %d\n", + ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, + 0, 0, 0); + if (ret < 0) { + dev_err(dev, "failed to stop FLL: %d\n", ret); + return ret; + } + } + break; + + default: + break; + } + + dapm->bias_level = level; + + return 0; +} + +static int imx_wm8962_late_probe(struct snd_soc_card *card) +{ + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct imx_priv *priv = &card_priv; + struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + struct device *dev = &priv->pdev->dev; + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, + data->clk_frequency, SND_SOC_CLOCK_IN); + if (ret < 0) + dev_err(dev, "failed to set sysclk in %s\n", __func__); + + return ret; +} + +static int imx_wm8962_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct device_node *ssi_np, *codec_np; + struct platform_device *ssi_pdev; + struct imx_priv *priv = &card_priv; + struct i2c_client *codec_dev; + struct imx_wm8962_data *data; + int int_port, ext_port; + int ret; + + priv->pdev = pdev; + + ret = of_property_read_u32(np, "mux-int-port", &int_port); + if (ret) { + dev_err(&pdev->dev, "mux-int-port missing or invalid\n"); + return ret; + } + ret = of_property_read_u32(np, "mux-ext-port", &ext_port); + if (ret) { + dev_err(&pdev->dev, "mux-ext-port missing or invalid\n"); + return ret; + } + + /* + * The port numbering in the hardware manual starts at 1, while + * the audmux API expects it starts at 0. + */ + int_port--; + ext_port--; + ret = imx_audmux_v2_configure_port(int_port, + IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TFSDIR | + IMX_AUDMUX_V2_PTCR_TCLKDIR, + IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); + if (ret) { + dev_err(&pdev->dev, "audmux internal port setup failed\n"); + return ret; + } + imx_audmux_v2_configure_port(ext_port, + IMX_AUDMUX_V2_PTCR_SYN, + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); + if (ret) { + dev_err(&pdev->dev, "audmux external port setup failed\n"); + return ret; + } + + ssi_np = of_parse_phandle(pdev->dev.of_node, "ssi-controller", 0); + codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0); + if (!ssi_np || !codec_np) { + dev_err(&pdev->dev, "phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + + ssi_pdev = of_find_device_by_node(ssi_np); + if (!ssi_pdev) { + dev_err(&pdev->dev, "failed to find SSI platform device\n"); + ret = -EINVAL; + goto fail; + } + codec_dev = of_find_i2c_device_by_node(codec_np); + if (!codec_dev || !codec_dev->driver) { + dev_err(&pdev->dev, "failed to find codec platform device\n"); + return -EINVAL; + } + + data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); + if (!data) { + ret = -ENOMEM; + goto fail; + } + + data->codec_clk = devm_clk_get(&codec_dev->dev, NULL); + if (IS_ERR(data->codec_clk)) { + ret = PTR_ERR(data->codec_clk); + dev_err(&codec_dev->dev, "failed to get codec clk: %d\n", ret); + goto fail; + } + + data->clk_frequency = clk_get_rate(data->codec_clk); + ret = clk_prepare_enable(data->codec_clk); + if (ret) { + dev_err(&codec_dev->dev, "failed to enable codec clk: %d\n", ret); + goto fail; + } + + data->dai.name = "HiFi"; + data->dai.stream_name = "HiFi"; + data->dai.codec_dai_name = "wm8962"; + data->dai.codec_of_node = codec_np; + data->dai.cpu_dai_name = dev_name(&ssi_pdev->dev); + data->dai.platform_of_node = ssi_np; + data->dai.ops = &imx_hifi_ops; + data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM; + + data->card.dev = &pdev->dev; + ret = snd_soc_of_parse_card_name(&data->card, "model"); + if (ret) + goto clk_fail; + ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); + if (ret) + goto clk_fail; + data->card.num_links = 1; + data->card.dai_link = &data->dai; + data->card.dapm_widgets = imx_wm8962_dapm_widgets; + data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8962_dapm_widgets); + + data->card.late_probe = imx_wm8962_late_probe; + data->card.set_bias_level = imx_wm8962_set_bias_level; + + ret = snd_soc_register_card(&data->card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + goto clk_fail; + } + + platform_set_drvdata(pdev, data); + of_node_put(ssi_np); + of_node_put(codec_np); + + return 0; + +clk_fail: + if (!IS_ERR(data->codec_clk)) + clk_disable_unprepare(data->codec_clk); +fail: + if (ssi_np) + of_node_put(ssi_np); + if (codec_np) + of_node_put(codec_np); + + return ret; +} + +static int imx_wm8962_remove(struct platform_device *pdev) +{ + struct imx_wm8962_data *data = platform_get_drvdata(pdev); + + if (!IS_ERR(data->codec_clk)) + clk_disable_unprepare(data->codec_clk); + snd_soc_unregister_card(&data->card); + + return 0; +} + +static const struct of_device_id imx_wm8962_dt_ids[] = { + { .compatible = "fsl,imx-audio-wm8962", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_wm8962_dt_ids); + +static struct platform_driver imx_wm8962_driver = { + .driver = { + .name = "imx-wm8962", + .owner = THIS_MODULE, + .of_match_table = imx_wm8962_dt_ids, + }, + .probe = imx_wm8962_probe, + .remove = imx_wm8962_remove, +}; +module_platform_driver(imx_wm8962_driver); + +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_DESCRIPTION("Freescale i.MX WM8962 ASoC machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:imx-wm8962"); -- 1.7.0.4

2 2

[alsa-devel] [PATCH] ASoC: dwc: debug message correction.
by Rajeev Kumar 11 Jun '13

11 Jun '13

From: Rajeev Kumar <rajeevk-dlh.kumar(a)st.com> Debug message correction. Signed-off-by: Rajeev Kumar <rajeevk-dlh.kumar(a)st.com> --- sound/soc/dwc/designware_i2s.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index deb30d5..deb975e 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -392,7 +392,7 @@ static int dw_i2s_probe(struct platform_device *pdev) } if (cap & DWC_I2S_PLAY) { - dev_dbg(&pdev->dev, " SPEAr: play supported\n"); + dev_dbg(&pdev->dev, " designware: play supported\n"); dw_i2s_dai->playback.channels_min = MIN_CHANNEL_NUM; dw_i2s_dai->playback.channels_max = pdata->channel; dw_i2s_dai->playback.formats = pdata->snd_fmts; @@ -400,7 +400,7 @@ static int dw_i2s_probe(struct platform_device *pdev) } if (cap & DWC_I2S_RECORD) { - dev_dbg(&pdev->dev, "SPEAr: record supported\n"); + dev_dbg(&pdev->dev, "designware: record supported\n"); dw_i2s_dai->capture.channels_min = MIN_CHANNEL_NUM; dw_i2s_dai->capture.channels_max = pdata->channel; dw_i2s_dai->capture.formats = pdata->snd_fmts; -- 1.6.0.2

2 2

[alsa-devel] [PATCH] ASoC: imx-sgtl5000: Use devm_clk_get()
by Fabio Estevam 11 Jun '13

11 Jun '13

Commit 9e13f345 (ASoC: sgtl5000: Let the codec acquire its clock) removed the clk_put calls. Let's use devm_clk_get() instead, so that we do not need to call them anymore. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/imx-sgtl5000.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 823151b..7a8bc12 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -128,7 +128,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) goto fail; } - data->codec_clk = clk_get(&codec_dev->dev, NULL); + data->codec_clk = devm_clk_get(&codec_dev->dev, NULL); if (IS_ERR(data->codec_clk)) goto fail; -- 1.8.1.2

4 5

[alsa-devel] [PATCH V5] ASoC: fsl: add imx-wm8962 machine driver
by Nicolin Chen 10 Jun '13

10 Jun '13

This is the initial imx-wm8962 device-tree-only machine driver working with fsl_ssi driver. More features can be added on top of it later. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- ChangeLog: v1->v2: * Fixed a typo in probe(): *** data->dai.ops = &imx_hifi_ops, ==> data->dai.ops = &imx_hifi_ops; v2->v3: * Moved Headphone/Microphone jack detection GPIO port to Optional properties. * Use set_bias_level() instead of hw_params/hw_free() to open/close FLL. * Use devm_clk_get() instead of clk_get(). * Dropped clock-frequency grabbing code. * Fixed a typo: *** data->card.late_probe = imx_wm8962_late_probe, *** ==> *** data->card.late_probe = imx_wm8962_late_probe; v3->v4: * Revised binding doc to describe two gpio pins more clearly. v4->v5: * Added return value checking for clk_prepare_enable(). * Revised error message for devm_clk_get() to keep same pattern as above. .../devicetree/bindings/sound/imx-audio-wm8962.txt | 55 ++++ sound/soc/fsl/Kconfig | 12 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-wm8962.c | 340 ++++++++++++++++++++ 4 files changed, 409 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt create mode 100644 sound/soc/fsl/imx-wm8962.c diff --git a/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt b/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt new file mode 100644 index 0000000..a34b94b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt @@ -0,0 +1,55 @@ +Freescale i.MX audio complex with WM8962 codec + +Required properties: +- compatible : "fsl,imx-audio-wm8962" +- model : The user-visible name of this sound complex +- ssi-controller : The phandle of the i.MX SSI controller +- audio-codec : The phandle of the WM8962 audio codec +- audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names could be power + supplies, WM8962 pins, and the jacks on the board: + + Power supplies: + * Mic Bias + + Board connectors: + * Mic Jack + * Headphone Jack + * Ext Spk + +- mux-int-port : The internal port of the i.MX audio muxer (AUDMUX) +- mux-ext-port : The external port of the i.MX audio muxer + +Note: The AUDMUX port numbering should start at 1, which is consistent with +hardware manual. + +Optional properties: +- hp-det-gpios : The gpio pin to detect plug in/out event that happens to + Headphone jack. + +- mic-det-gpios: The gpio pin to detect plug in/out event that happens to + Micphone jack. + +Example: + +sound { + compatible = "fsl,imx6q-sabresd-wm8962", + "fsl,imx-audio-wm8962"; + model = "wm8962-audio"; + ssi-controller = <&ssi2>; + audio-codec = <&codec>; + audio-routing = + "Headphone Jack", "HPOUTL", + "Headphone Jack", "HPOUTR", + "Ext Spk", "SPKOUTL", + "Ext Spk", "SPKOUTR", + "MICBIAS", "AMIC", + "IN3R", "MICBIAS", + "DMIC", "MICBIAS", + "DMICDAT", "DMIC"; + mux-int-port = <2>; + mux-ext-port = <3>; + hp-det-gpios = <&gpio7 8 1>; /* active low */ + mic-det-gpios = <&gpio1 9 1>; /* active low */ +}; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 7860cc2..aa43854 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -168,6 +168,18 @@ config SND_SOC_EUKREA_TLV320 Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface +config SND_SOC_IMX_WM8962 + tristate "SoC Audio support for i.MX boards with wm8962" + depends on OF && I2C + select SND_SOC_WM8962 + select SND_SOC_IMX_PCM_DMA + select SND_SOC_IMX_AUDMUX + select SND_SOC_FSL_SSI + select SND_SOC_FSL_UTILS + help + Say Y if you want to add support for SoC audio on an i.MX board with + a wm8962 codec. + config SND_SOC_IMX_SGTL5000 tristate "SoC Audio support for i.MX boards with sgtl5000" depends on OF && I2C diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 91883f8..d4b4aa8 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -42,6 +42,7 @@ snd-soc-phycore-ac97-objs := phycore-ac97.o snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o +snd-soc-imx-wm8962-objs := imx-wm8962.o snd-soc-imx-mc13783-objs := imx-mc13783.o obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o @@ -49,4 +50,5 @@ obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o +obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c new file mode 100644 index 0000000..3bc5b9d --- /dev/null +++ b/sound/soc/fsl/imx-wm8962.c @@ -0,0 +1,340 @@ +/* + * Copyright 2013 Freescale Semiconductor, Inc. + * + * Based on imx-sgtl5000.c + * Copyright 2012 Freescale Semiconductor, Inc. + * Copyright 2012 Linaro Ltd. + * + * The code contained herein is licensed under the GNU General Public + * License. You may obtain a copy of the GNU General Public License + * Version 2 or later at the following locations: + * + * http://www.opensource.org/licenses/gpl-license.html + * http://www.gnu.org/copyleft/gpl.html + */ + +#include <linux/module.h> +#include <linux/of.h> +#include <linux/of_platform.h> +#include <linux/of_i2c.h> +#include <linux/of_gpio.h> +#include <linux/slab.h> +#include <linux/clk.h> +#include <linux/clk-provider.h> +#include <linux/irq.h> +#include <sound/soc.h> +#include <sound/pcm_params.h> +#include <sound/soc-dapm.h> +#include <linux/pinctrl/consumer.h> + +#include "../codecs/wm8962.h" +#include "imx-audmux.h" + +#define DAI_NAME_SIZE 32 + +struct imx_wm8962_data { + struct snd_soc_dai_link dai; + struct snd_soc_card card; + char codec_dai_name[DAI_NAME_SIZE]; + char platform_name[DAI_NAME_SIZE]; + struct clk *codec_clk; + unsigned int clk_frequency; +}; + +struct imx_priv { + int hp_gpio; + int hp_active_low; + int hp_status; + int amic_gpio; + int amic_active_low; + int amic_status; + struct platform_device *pdev; + struct snd_pcm_substream *first_stream; + struct snd_pcm_substream *second_stream; +}; +static struct imx_priv card_priv; + +static const struct snd_soc_dapm_widget imx_wm8962_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_MIC("AMIC", NULL), + SND_SOC_DAPM_MIC("DMIC", NULL), +}; + +static int sample_rate = 44100; +static snd_pcm_format_t sample_format = SNDRV_PCM_FORMAT_S16_LE; + +static int imx_hifi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + sample_rate = params_rate(params); + sample_format = params_format(params); + + return 0; +} + +static struct snd_soc_ops imx_hifi_ops = { + .hw_params = imx_hifi_hw_params, +}; + +static int imx_wm8962_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct imx_priv *priv = &card_priv; + struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + unsigned int pll_out; + int ret; + + if (dapm->dev != codec_dai->dev) + return 0; + + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { + if (sample_format == SNDRV_PCM_FORMAT_S24_LE) + pll_out = sample_rate * 384; + else + pll_out = sample_rate * 256; + + ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, + WM8962_FLL_MCLK, data->clk_frequency, + pll_out); + if (ret < 0) { + pr_err("Failed to start FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, + WM8962_SYSCLK_FLL, pll_out, + SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err("Failed to set SYSCLK: %d\n", ret); + return ret; + } + } + break; + + case SND_SOC_BIAS_STANDBY: + if (dapm->bias_level == SND_SOC_BIAS_PREPARE) { + ret = snd_soc_dai_set_sysclk(codec_dai, + WM8962_SYSCLK_MCLK, data->clk_frequency, + SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err("Failed to switch away from FLL: %d\n", + ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, + 0, 0, 0); + if (ret < 0) { + pr_err("Failed to stop FLL: %d\n", ret); + return ret; + } + } + break; + + default: + break; + } + + dapm->bias_level = level; + + return 0; +} + +static int imx_wm8962_late_probe(struct snd_soc_card *card) +{ + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct imx_priv *priv = &card_priv; + struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, + data->clk_frequency, SND_SOC_CLOCK_IN); + if (ret < 0) + dev_err(card->dev, "Failed to set sysclk in %s\n", __func__); + + return ret; +} + +static int imx_wm8962_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct device_node *ssi_np, *codec_np; + struct platform_device *ssi_pdev; + struct imx_priv *priv = &card_priv; + struct i2c_client *codec_dev; + struct imx_wm8962_data *data; + int int_port, ext_port; + int ret; + + priv->pdev = pdev; + + ret = of_property_read_u32(np, "mux-int-port", &int_port); + if (ret) { + dev_err(&pdev->dev, "mux-int-port missing or invalid\n"); + return ret; + } + ret = of_property_read_u32(np, "mux-ext-port", &ext_port); + if (ret) { + dev_err(&pdev->dev, "mux-ext-port missing or invalid\n"); + return ret; + } + + /* + * The port numbering in the hardware manual starts at 1, while + * the audmux API expects it starts at 0. + */ + int_port--; + ext_port--; + ret = imx_audmux_v2_configure_port(int_port, + IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TFSDIR | + IMX_AUDMUX_V2_PTCR_TCLKDIR, + IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); + if (ret) { + dev_err(&pdev->dev, "audmux internal port setup failed\n"); + return ret; + } + imx_audmux_v2_configure_port(ext_port, + IMX_AUDMUX_V2_PTCR_SYN, + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); + if (ret) { + dev_err(&pdev->dev, "audmux external port setup failed\n"); + return ret; + } + + ssi_np = of_parse_phandle(pdev->dev.of_node, "ssi-controller", 0); + codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0); + if (!ssi_np || !codec_np) { + dev_err(&pdev->dev, "phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + + ssi_pdev = of_find_device_by_node(ssi_np); + if (!ssi_pdev) { + dev_err(&pdev->dev, "failed to find SSI platform device\n"); + ret = -EINVAL; + goto fail; + } + codec_dev = of_find_i2c_device_by_node(codec_np); + if (!codec_dev || !codec_dev->driver) { + dev_err(&pdev->dev, "failed to find codec platform device\n"); + return -EINVAL; + } + + priv->hp_gpio = of_get_named_gpio_flags(np, "hp-det-gpios", 0, + (enum of_gpio_flags *)&priv->hp_active_low); + priv->amic_gpio = of_get_named_gpio_flags(np, "mic-det-gpios", 0, + (enum of_gpio_flags *)&priv->amic_active_low); + + priv->first_stream = NULL; + priv->second_stream = NULL; + + data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); + if (!data) { + ret = -ENOMEM; + goto fail; + } + + data->codec_clk = devm_clk_get(&codec_dev->dev, NULL); + if (IS_ERR(data->codec_clk)) { + ret = PTR_ERR(data->codec_clk); + dev_err(&codec_dev->dev, "failed to get codec clk: %d\n", ret); + goto fail; + } + + data->clk_frequency = clk_get_rate(data->codec_clk); + ret = clk_prepare_enable(data->codec_clk); + if (ret) { + dev_err(&codec_dev->dev, "failed to enable codec clk: %d\n", ret); + goto fail; + } + + data->dai.name = "HiFi"; + data->dai.stream_name = "HiFi"; + data->dai.codec_dai_name = "wm8962"; + data->dai.codec_of_node = codec_np; + data->dai.cpu_dai_name = dev_name(&ssi_pdev->dev); + data->dai.platform_of_node = ssi_np; + data->dai.ops = &imx_hifi_ops; + data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM; + + data->card.dev = &pdev->dev; + ret = snd_soc_of_parse_card_name(&data->card, "model"); + if (ret) + goto clk_fail; + ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); + if (ret) + goto clk_fail; + data->card.num_links = 1; + data->card.dai_link = &data->dai; + data->card.dapm_widgets = imx_wm8962_dapm_widgets; + data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8962_dapm_widgets); + + data->card.late_probe = imx_wm8962_late_probe; + data->card.set_bias_level = imx_wm8962_set_bias_level; + + ret = snd_soc_register_card(&data->card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + goto clk_fail; + } + + platform_set_drvdata(pdev, data); + of_node_put(ssi_np); + of_node_put(codec_np); + + return 0; + +clk_fail: + if (!IS_ERR(data->codec_clk)) + clk_disable_unprepare(data->codec_clk); +fail: + if (ssi_np) + of_node_put(ssi_np); + if (codec_np) + of_node_put(codec_np); + + return ret; +} + +static int imx_wm8962_remove(struct platform_device *pdev) +{ + struct imx_wm8962_data *data = platform_get_drvdata(pdev); + + if (!IS_ERR(data->codec_clk)) + clk_disable_unprepare(data->codec_clk); + snd_soc_unregister_card(&data->card); + + return 0; +} + +static const struct of_device_id imx_wm8962_dt_ids[] = { + { .compatible = "fsl,imx-audio-wm8962", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_wm8962_dt_ids); + +static struct platform_driver imx_wm8962_driver = { + .driver = { + .name = "imx-wm8962", + .owner = THIS_MODULE, + .of_match_table = imx_wm8962_dt_ids, + }, + .probe = imx_wm8962_probe, + .remove = imx_wm8962_remove, +}; +module_platform_driver(imx_wm8962_driver); + +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_DESCRIPTION("Freescale i.MX WM8962 ASoC machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:imx-wm8962"); -- 1.7.1

3 2

[alsa-devel] [PATCH] ASoC: ssm2518: Fix trivial typo in checking tx_mask and rx_mask values
by Axel Lin 10 Jun '13

10 Jun '13

Otherwise, ssm2518_set_tdm_slot() always returns error if slots != 0. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/ssm2518.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c index a76b341..f4883d9 100644 --- a/sound/soc/codecs/ssm2518.c +++ b/sound/soc/codecs/ssm2518.c @@ -538,7 +538,7 @@ static int ssm2518_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, SSM2518_REG_SAI_CTRL1, SSM2518_SAI_CTRL1_SAI_MASK, SSM2518_SAI_CTRL1_SAI_I2S); - if (tx_mask == 0 || tx_mask != 0) + if (tx_mask == 0 || rx_mask != 0) return -EINVAL; if (slots == 1) { -- 1.8.1.2

3 2

[alsa-devel] firewire-kernel-streaming branch into main kernel tree?
by Rolf Anderegg 10 Jun '13

10 Jun '13

Hello Clemens, it's been a long time since I last inquired about the snd_dice module provided by your firewire-kernel-streaming branch (http://git.alsa-project.org/?p=alsa-kprivate.git;a=shortlog;h=refs/heads/fi…), which is a good sign I guess. In the meantime we have been making heavy use of the snd_dice module for over a year with all of our Weiss DICE based products (obviously playback only), and it has proven to be extremely stable (currently 3.4.18-rt29, patched with firewire-kernel-streaming). Therefore I have been meaning to ask you for some time now whether you have any interest in merging this firewire-kernel-streaming branch into the main kernel tree (at least as an experimental option, mind you), or are there any major objections? Given that you don't have unlimited time at hand to further develop the snd_dice module, let alone provide capture support in the near future, AND the fact that DICE is likely to see its EOL soon (and replaced by a sophisticated successor by TCAT), I realize that this is probably not a high priority. However from our perspective it would be sad to see these ALSA Firewire proceedings be lost and not be appreciated by a larger audience. With respect to this: * what would be the customary approach to requesting such a merge? * what are your reservations, resp. are there any existing shortcomings that you are yet unhappy with? * have you received any other (negative/positive) feedback from snd_dice users? I know of the Vortexbox community, which has been offering an optional patch to support Firewire plackback devices; as far as I know, they're happy with it. * if you need any backing or assistance, we'd be happy to help out Keen to hear your feedback on this suggestion. Best regards, Rolf Anderegg (Weiss Engineering Ltd.)

2 2

[alsa-devel] [PATCH (alsa-lib)] pcm_hw: Fix buffer overflow in chmap
by David Henningsson 10 Jun '13

10 Jun '13

We can't calculate memory storage in bytes, when we're supposed to store ints in it! Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- src/pcm/pcm_hw.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/pcm/pcm_hw.c b/src/pcm/pcm_hw.c index 196393d..9ff83a9 100644 --- a/src/pcm/pcm_hw.c +++ b/src/pcm/pcm_hw.c @@ -1187,7 +1187,7 @@ static snd_pcm_chmap_t *snd_pcm_hw_get_chmap(snd_pcm_t *pcm) snd_pcm_state_name(FAST_PCM_STATE(hw))); return NULL; } - map = malloc(pcm->channels + 1); + map = malloc(pcm->channels * sizeof(map->pos[0]) + sizeof(*map)); if (!map) return NULL; map->channels = pcm->channels; -- 1.8.1.2

1 1

[alsa-devel] [PATCH 1/2] ASoC: sgtl5000: Use i2c_get_clientdata()
by Fabio Estevam 10 Jun '13

10 Jun '13

We should use i2c_get_clientdata() to get the codec private structure. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/sgtl5000.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 2e0227b..d441559 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1571,11 +1571,7 @@ disable_clk: static int sgtl5000_i2c_remove(struct i2c_client *client) { - struct sgtl5000_priv *sgtl5000; - sgtl5000 = devm_kzalloc(&client->dev, sizeof(struct sgtl5000_priv), - GFP_KERNEL); - if (!sgtl5000) - return -ENOMEM; + struct sgtl5000_priv *sgtl5000 = i2c_get_clientdata(client); snd_soc_unregister_codec(&client->dev); clk_disable_unprepare(sgtl5000->mclk); -- 1.8.1.2

2 2

[alsa-devel] [PATCH] ASoC: adav80x: Select SND_SOC_ADAV80X if SND_SOC_I2C_AND_SPI
by Axel Lin 10 Jun '13

10 Jun '13

This driver is useless if both SPI and I2C are not configured. Thus don't build this driver if both SPI and I2C are not configured. This patch silences below build warning if both SPI and I2C are not configured. CC sound/soc/codecs/adav80x.o sound/soc/codecs/adav80x.c:842:12: warning: 'adav80x_bus_probe' defined but not used [-Wunused-function] sound/soc/codecs/adav80x.c:863:12: warning: 'adav80x_bus_remove' defined but not used [-Wunused-function] Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index c90d298..6234900 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -19,7 +19,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AD1980 if SND_SOC_AC97_BUS select SND_SOC_AD73311 select SND_SOC_ADAU1373 if I2C - select SND_SOC_ADAV80X + select SND_SOC_ADAV80X if SND_SOC_I2C_AND_SPI select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C -- 1.8.1.2

3 2

[alsa-devel] ASoC: ssm2518: bug in ssm2518_set_tdm_slot
by Axel Lin 10 Jun '13

10 Jun '13

ASoC: ssm2518: bug in ssm2518_set_tdm_slot The code in ssm2518_set_tdm_slot looks wrong: if (tx_mask == 0 || tx_mask != 0) return -EINVAL; So ssm2518_set_tdm_slot() always returns error.

2 1

[alsa-devel] [PATCH 0/4] ASoC: codecs: some more improvements for adau1701
by Daniel Mack 10 Jun '13

10 Jun '13

Here are some more patches that I worked on while bootstrapping a new audio device featuring an ADAU1701 DSP. Daniel Mack (4): ASoC: codecs: adau1701: factor out firmware reset ASoC: codecs: adau1701: allow configuration of PLL mode pins ASoC: codecs: adau1701: switch to direct regmap API usage ASoC: codecs: adau1701: add support for pin muxing .../devicetree/bindings/sound/adi,adau1701.txt | 17 ++ sound/soc/codecs/adau1701.c | 255 ++++++++++++++++++--- sound/soc/codecs/adau1701.h | 4 + 3 files changed, 238 insertions(+), 38 deletions(-) -- 1.8.1.4

3 9

[alsa-devel] [PATCH] ASoC: wm8962: Restore device state after reset in runtime resume
by Mark Brown 10 Jun '13

10 Jun '13

After the device has been reset we need to repeat the same initialisation we do on probe to make sure that the device is in a known state. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/wm8962.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index d56dd86..32c2c47 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3723,6 +3723,21 @@ static int wm8962_runtime_resume(struct device *dev) wm8962_reset(wm8962); + /* SYSCLK defaults to on; make sure it is off so we can safely + * write to registers if the device is declocked. + */ + regmap_update_bits(wm8962->regmap, WM8962_CLOCKING2, + WM8962_SYSCLK_ENA, 0); + + /* Ensure we have soft control over all registers */ + regmap_update_bits(wm8962->regmap, WM8962_CLOCKING2, + WM8962_CLKREG_OVD, WM8962_CLKREG_OVD); + + /* Ensure that the oscillator and PLLs are disabled */ + regmap_update_bits(wm8962->regmap, WM8962_PLL2, + WM8962_OSC_ENA | WM8962_PLL2_ENA | WM8962_PLL3_ENA, + 0); + regcache_sync(wm8962->regmap); return 0; -- 1.7.10.4

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[alsa-devel] [PATCH] ASoC: Add Kconfig and Makefile to support SPEAr audio driver
by Rajeev Kumar 10 Jun '13

10 Jun '13

This patch adds Kconfig and Makefile to support SPEAr Audio driver. Signed-off-by: Rajeev Kumar <rajeev-dlh.kumar(a)st.com> --- sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/spear/Kconfig | 39 +++++++++++++++++++++++++++++++++++++++ sound/soc/spear/Makefile | 7 +++++++ 4 files changed, 48 insertions(+), 0 deletions(-) create mode 100644 sound/soc/spear/Kconfig create mode 100644 sound/soc/spear/Makefile diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 5da8ca7..c6d98e2 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -47,6 +47,7 @@ source "sound/soc/pxa/Kconfig" source "sound/soc/samsung/Kconfig" source "sound/soc/s6000/Kconfig" source "sound/soc/sh/Kconfig" +source "sound/soc/spear/Kconfig" source "sound/soc/tegra/Kconfig" source "sound/soc/txx9/Kconfig" source "sound/soc/ux500/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 99f32f7..b86bf5a 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -25,6 +25,7 @@ obj-$(CONFIG_SND_SOC) += pxa/ obj-$(CONFIG_SND_SOC) += samsung/ obj-$(CONFIG_SND_SOC) += s6000/ obj-$(CONFIG_SND_SOC) += sh/ +obj-$(CONFIG_SND_SOC) += spear/ obj-$(CONFIG_SND_SOC) += tegra/ obj-$(CONFIG_SND_SOC) += txx9/ obj-$(CONFIG_SND_SOC) += ux500/ diff --git a/sound/soc/spear/Kconfig b/sound/soc/spear/Kconfig new file mode 100644 index 0000000..3b7cdad --- /dev/null +++ b/sound/soc/spear/Kconfig @@ -0,0 +1,39 @@ +config SND_SPEAR_EVM + tristate "SoC Audio support for SPEAr EVM" + select SND_DESIGNWARE_I2S + select SND_SOC_STA529 + select SND_SPEAR_SOC + help + Say Y if you want to add support for SoC audio on SPEAr + platform + +config SND_SPEAR1340_EVM + tristate "SoC Audio support for SPEAr1340 EVM" + select SND_DESIGNWARE_I2S + select SND_SOC_STA529 + select SND_SPEAR_SPDIF_OUT + select SND_SPEAR_SPDIF_IN + select SND_SOC_SPDIF + select SND_SPEAR_SOC + help + Say Y if you want to add support for SoC audio on SPEAr1340 + platform + +config SND_SPEAR_SOC + tristate "SoC Audio for the ST chip" + depends on SND_DESIGNWARE_I2S || SND_SPEAR_SPDIF_OUT || \ + SND_SPEAR_SPDIF_IN + help + Say Y or M if you want to add support for any of the audio + controllers (I2S/SPDIF). You will also need to select + the audio interface codecs to support below. + +config SND_SPEAR_SPDIF_OUT + tristate "SPEAr SPDIF Out Device Driver" + help + Say Y or M if you want to add support for SPDIF OUT driver. + +config SND_SPEAR_SPDIF_IN + tristate "SPEAr SPDIF IN Device Driver" + help + Say Y or M if you want to add support for SPDIF IN driver. diff --git a/sound/soc/spear/Makefile b/sound/soc/spear/Makefile new file mode 100644 index 0000000..b365126 --- /dev/null +++ b/sound/soc/spear/Makefile @@ -0,0 +1,7 @@ +# SPEAR Platform Support +obj-$(CONFIG_SND_SPEAR_SOC) += spear_pcm.o +obj-$(CONFIG_SND_SPEAR_SPDIF_IN) += spdif_in.o +obj-$(CONFIG_SND_SPEAR_SPDIF_OUT) += spdif_out.o + +# SPEAR Machine Support +obj-$(CONFIG_SND_SPEAR_EVM) += spear_evb.o -- 1.6.0.2

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Re: [alsa-devel] [PATCH 12/15] ASoC: spear: Convert to use devm_ioremap_resource
by Sachin Kamat 10 Jun '13

10 Jun '13

On 10 June 2013 17:05, Tushar Behera <tushar.behera(a)linaro.org> wrote: > Commit 75096579c3ac ("lib: devres: Introduce devm_ioremap_resource()") > introduced devm_ioremap_resource() and deprecated the use of > devm_request_and_ioremap(). This patch does a little more than just conversion. May that change could be mentioned here. > > Signed-off-by: Tushar Behera <tushar.behera(a)linaro.org> > CC: alsa-devel(a)alsa-project.org > CC: Liam Girdwood <lgirdwood(a)gmail.com> > CC: Mark Brown <broonie(a)kernel.org> > --- > sound/soc/spear/spdif_out.c | 20 +++++--------------- > 1 file changed, 5 insertions(+), 15 deletions(-) > > diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c > index 4bde512..896ee7f 100644 > --- a/sound/soc/spear/spdif_out.c > +++ b/sound/soc/spear/spdif_out.c > @@ -282,27 +282,17 @@ static int spdif_out_probe(struct platform_device *pdev) > struct resource *res; > int ret; > > - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); > - if (!res) > - return -EINVAL; > - > - if (!devm_request_mem_region(&pdev->dev, res->start, > - resource_size(res), pdev->name)) { > - dev_warn(&pdev->dev, "Failed to get memory resourse\n"); > - return -ENOENT; > - } > - > host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL); > if (!host) { > dev_warn(&pdev->dev, "kzalloc fail\n"); > return -ENOMEM; > } > > - host->io_base = devm_request_and_ioremap(&pdev->dev, res->start, > - resource_size(res)); > - if (!host->io_base) { > - dev_warn(&pdev->dev, "ioremap failed\n"); > - return -ENOMEM; > + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); > + host->io_base = devm_ioremap_resource(&pdev->dev, res); > + if (IS_ERR(host->io_base)) { > + dev_warn(&pdev->dev, "devm_ioremap_resource failed\n"); This print is not needed as devm_ioremap_resource prints its own messages. > + return PTR_ERR(host->io_base); > } > > host->clk = devm_clk_get(&pdev->dev, NULL); > -- > 1.7.9.5 > > -- > To unsubscribe from this list: send the line "unsubscribe linux-kernel" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html > Please read the FAQ at http://www.tux.org/lkml/ -- With warm regards, Sachin

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[alsa-devel] Missing Kconfig and Makefile for SPEAr platform
by Rajeev kumar 10 Jun '13

10 Jun '13

Hello Mark, The Makefile and Kconfig is missing for SPEAr platform. Please find below the link of circulated patch for the same. http://mailman.alsa-project.org/pipermail/alsa-devel/2012-June/052872.html Please update. Thanks and Best Regards Rajeev

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Re: [alsa-devel] [PATCH] ASoC: dmaengine: add runtime status checking in dmaengine_pcm_dma_complete
by Mark Brown 10 Jun '13

10 Jun '13

On Sat, Jun 08, 2013 at 04:34:02PM +0800, Qiao Zhou wrote: > This call back only takes substream as parameter. In case that a > wrong/fake interrupt of other drivers triggers this call back, like > a careless/bad design, then it panics in audio driver. Can we add > such checking and give a warning log rather than to let the system > die directly. Could you consider it? Why not fix the problem in the paths that do the teardown instead?

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[alsa-devel] Codec driver gets probed prior to the machine driver
by Fabio Estevam 10 Jun '13

10 Jun '13

Hi, When running linux-next 20130607 on a mx6qsabrelite I see that the audio drivers fail to probe. Added some printk's to better illustrate the problem: mmcblk0: mmc0:b5a1 SD04G 3.69 GiB **** reading the chip ID sgtl5000: probe of 0-000a failed with error -5 ****** enabling the codec clock imx-sgtl5000 sound.13: ASoC: CODEC (null) not registered imx-sgtl5000 sound.13: snd_soc_register_card failed (-517) platform sound.13: Driver imx-sgtl5000 requests probe defer The problem here is that the codec driver (sgtl5000) is being probed prior to the machine driver (imx-sgtl5000). The reading of the chip ID fails because at this point in time there is no codec clock, which is turned on by imx-sgtl5000. Running 3.10-rc4, the order of the events is as expected: imx-sgtl5000 is called first and sgtl5000 is called later. In this case the probe works as usual. Tried doing a bisect, but this was not succesful due to several bisect breakages (build issues and non-booting kernels) on its way. Would anyone have some suggestion as to why the codec driver is getting probed first? Thanks, Fabio Estevam

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[alsa-devel] Device enumeration
by Robert Billing 10 Jun '13

10 Jun '13

I am currently developing a package to play theatrical sound effects, which has two play channels that I want to be able to attach to two different ALSA devices. To allow the users, who are theatre directors and are *not* technical to do this I am trying to provide a drop-down list of play devices. I can get the raw devices by parsing either the output from aplay -L, or from /proc/asound. The problem comes with devices defined in .asoundrc. I have this in .asoundrc on the development system: pcm.speakers { type pulse device alsa_output.pci-0000_02_06.0.analog-stereo } ctl.speakers { type pulse device alsa_output.pci-0000_02_06.0.analog-stereo } pcm.cans { type pulse device alsa_output.pci-0000_00_1f.5.analog-stereo } ctl.cans { type pulse device alsa_output.pci-0000_00_1f.5.analog-stereo } This works and it is possible to play to the devices "cans" or "speakers" to get output on headphones and speakers respectively, but these names don't appear in the output from aplay. What I need is some clean way of getting a list of the aliases. This package will have to be capable of being installed on any Linux system without needing to edit configuration files manually. However if some tool or some part of the installation process has edited a configuration file I want the aliases to appear in my list of devices. What is the right way to do this? -- Robert Billing, Phone/fax +44 (0) 1344-772849 bob(a)tnglwood.demon.co.uk Tanglewood, 64, Pinehill Road, Crowthorne, Berks, RG45 7JR

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