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[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)>] >

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[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 >

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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

4 4

[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

2 1

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

1 0

[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

2 1

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?

1 0

[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

3 3

[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

1 0

[alsa-devel] [PATCH] ALSA: hda - Headset mic support for two more machines
by David Henningsson 10 Jun '13

10 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 | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 02e22b4..cbd443d 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 +3495,7 @@ 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, 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 0

[alsa-devel] [PATCH 0/8] [RFC] new driver for Echo Audio's Fireworks based devices
by o-takashi＠sakamocchi.jp 08 Jun '13

08 Jun '13

From: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> This driver module is originally developed by Clemens Ladish 2010. But it includes duplicate codes which current snd-firewire-lib has and lack of automatical sampling rate/the number of channels arrangement. http://git.alsa-project.org/?p=alsa-kprivate.git;a=shortlog;h=refs/heads/fi… This driver module is based on snd-firewire-lib with my previous patches. [PATCH v2 0/3] snd-firewire-lib: add handling CMP output connection http://mailman.alsa-project.org/pipermail/alsa-devel/2013-April/061607.html [PATCH v2 0/4] snd-firewire-lib: add handling AMDTP receive stream http://mailman.alsa-project.org/pipermail/alsa-devel/2013-April/061611.html [PATCH 0/2] snd-firewire-lib: add MIDI stream support http://mailman.alsa-project.org/pipermail/alsa-devel/2013-June/062610.html Currently this driver module support just PCM/MIDI kernel streaming via ALSA interfaces. So there are some issues about this driver. 1.whether adding each PCM devices for analog and digital interface 2.where the codes for device control like volume, routing and etc are 3.card definition files for ALSA applications like PulseAudio I hope this driver module promote developers to discuss about how to implement Firewire devices which has many channels and options. Takashi Sakamoto (8): add main file of driver module add device specific command add control interfaces add handling AMDTP stream add MIDI interface add PCM interface add proc interface modify Makefile and Kconfig to build this module sound/firewire/Kconfig | 16 + sound/firewire/Makefile | 2 + sound/firewire/fireworks/Makefile | 2 + sound/firewire/fireworks/fireworks.c | 488 +++++++++++++++++++++++ sound/firewire/fireworks/fireworks.h | 241 ++++++++++++ sound/firewire/fireworks/fireworks_command.c | 535 +++++++++++++++++++++++++ sound/firewire/fireworks/fireworks_control.c | 537 ++++++++++++++++++++++++++ sound/firewire/fireworks/fireworks_midi.c | 236 +++++++++++ sound/firewire/fireworks/fireworks_pcm.c | 530 +++++++++++++++++++++++++ sound/firewire/fireworks/fireworks_proc.c | 183 +++++++++ sound/firewire/fireworks/fireworks_stream.c | 107 +++++ 11 files changed, 2877 insertions(+) create mode 100644 sound/firewire/fireworks/Makefile create mode 100644 sound/firewire/fireworks/fireworks.c create mode 100644 sound/firewire/fireworks/fireworks.h create mode 100644 sound/firewire/fireworks/fireworks_command.c create mode 100644 sound/firewire/fireworks/fireworks_control.c create mode 100644 sound/firewire/fireworks/fireworks_midi.c create mode 100644 sound/firewire/fireworks/fireworks_pcm.c create mode 100644 sound/firewire/fireworks/fireworks_proc.c create mode 100644 sound/firewire/fireworks/fireworks_stream.c -- 1.7.10.4

3 18

[alsa-devel] [PATCH] ASoC: sgtl5000: Make device cache only when powered off
by Mark Brown 08 Jun '13

08 Jun '13

When the regulators have been disabled mark the device as cache only so that we don't try to interact with the hardware. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/sgtl5000.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 1c3b20f..85a32e6 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -911,10 +911,25 @@ static int sgtl5000_set_bias_level(struct snd_soc_codec *codec, if (ret) return ret; udelay(10); + + regcache_cache_only(sgtl5000->regmap, false); + + ret = regcache_sync(sgtl5000->regmap); + if (ret != 0) { + dev_err(codec->dev, + "Failed to restore cache: %d\n", ret); + + regcache_cache_only(sgtl5000->regmap, true); + regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + + return ret; + } } break; case SND_SOC_BIAS_OFF: + regcache_cache_only(sgtl5000->regmap, true); regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); break; -- 1.7.10.4

3 4

[alsa-devel] [PATCH] ASoC: codecs: tlv320aic3x: Remove deadlock from snd_soc_dapm_put_volsw_aic3x()
by Andreas Irestal 07 Jun '13

07 Jun '13

From: Andreas Irestål <Andreas.Irestal(a)axis.com> When calling snd_soc_dapm_sync(), it eventually tries to lock the same mutex already locked in snd_soc_dapm_put_volsw_aic3x() and a deadlock occurs. By moving the mutex unlock to just before snd_soc_dapm_sync(), this deadlock is prevented. This problem was introduced in Linux 3.5 Signed-off-by: Andreas Irestål <Andreas.Irestal(a)axis.com> --- sound/soc/codecs/tlv320aic3x.c | 10 +++++----- 1 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 65d09d6..1514bf8 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -187,14 +187,14 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, break; } - - if (found) - snd_soc_dapm_sync(widget->dapm); } - ret = snd_soc_update_bits(widget->codec, reg, val_mask, val); - mutex_unlock(&widget->codec->mutex); + + if (found) + snd_soc_dapm_sync(widget->dapm); + + ret = snd_soc_update_bits_locked(widget->codec, reg, val_mask, val); return ret; }

2 1

[alsa-devel] [PATCH] ASoC: WM8962: Set CLKREG_OVD if GP5_FN is not its reset value
by Nicolin Chen 07 Jun '13

07 Jun '13

According to WM8962 datasheet, the GP5_FN bit of R516 (GPIO 5) depends on the CLKREG_OVD bit of R8 (Clocking2): "Note that GPIO5 functions are only supported when CLKREG_OVD=1. When CLKREG_OVD=0, the contents of Register R516 must not be changed from the default value." So set CLKREG_OVD bit before regcache_sync() updates the GP5_FN bit to a non-reset value. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- sound/soc/codecs/wm8962.c | 19 +++++++++++++++++++ 1 files changed, 19 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 26219ea..9c8fd11 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3739,6 +3739,23 @@ static int wm8962_i2c_remove(struct i2c_client *client) } #ifdef CONFIG_PM_RUNTIME +static void wm8962_check_gp5_fn(struct snd_soc_codec *codec) +{ + int changed; + + /* + * If GP5_FN is not going to be the reset value 0x0, + * need to set CLKREG_OVD bit of Register CLOCKING2 + * before regcache_sync() updates GP5_FN bit to a + * non-reset value. The reset value should be 0x0. + */ + changed = snd_soc_test_bits(codec, WM8962_GPIO_5, + WM8962_GP5_FN_MASK, 0x0); + if (changed) + snd_soc_update_bits(codec, WM8962_CLOCKING2, + WM8962_CLKREG_OVD_MASK, WM8962_CLKREG_OVD); +} + static int wm8962_runtime_resume(struct device *dev) { struct wm8962_priv *wm8962 = dev_get_drvdata(dev); @@ -3756,6 +3773,8 @@ static int wm8962_runtime_resume(struct device *dev) wm8962_reset(wm8962); + wm8962_check_gp5_fn(wm8962->codec); + regcache_sync(wm8962->regmap); return 0; -- 1.7.1

3 5

[alsa-devel] [PATCH] ALSA: get rid of CONFIG_SND_VERBOSE_PRINTK
by Alan Stern 07 Jun '13

07 Jun '13

According to Takashi Iwai, CONFIG_SND_VERBOSE_PRINTK is "useless". All it does is add filenames and line numbers to certain log messages. This patch gets rid of it. Signed-off-by: Alan Stern <stern(a)rowland.harvard.edu> CC: Joe Perches <joe(a)perches.com> --- Updating all the snd_printk() and snd_printd() calls will be a huge job. Getting rid of the verbose logging is a lot easier, so that's where I'm starting. Alan Stern [as1688] Documentation/DocBook/writing-an-alsa-driver.tmpl | 7 +--- arch/arm/configs/eseries_pxa_defconfig | 1 arch/arm/configs/omap2plus_defconfig | 1 arch/arm/configs/s3c2410_defconfig | 1 arch/arm/configs/trizeps4_defconfig | 1 arch/ia64/configs/generic_defconfig | 1 arch/ia64/configs/gensparse_defconfig | 1 arch/mips/configs/pnx8335_stb225_defconfig | 1 arch/powerpc/configs/ppc6xx_defconfig | 1 arch/sh/configs/edosk7760_defconfig | 1 arch/sh/configs/sh7785lcr_32bit_defconfig | 1 include/sound/core.h | 4 +- sound/core/Kconfig | 9 ------ sound/core/misc.c | 33 ---------------------- sound/pci/hda/hda_intel.c | 4 -- 15 files changed, 5 insertions(+), 62 deletions(-) Index: usb-3.10/Documentation/DocBook/writing-an-alsa-driver.tmpl =================================================================== --- usb-3.10.orig/Documentation/DocBook/writing-an-alsa-driver.tmpl +++ usb-3.10/Documentation/DocBook/writing-an-alsa-driver.tmpl @@ -6087,11 +6087,8 @@ struct _snd_pcm_runtime { <title><function>snd_printk()</function> and friends</title> <para> ALSA provides a verbose version of the - <function>printk()</function> function. If a kernel config - <constant>CONFIG_SND_VERBOSE_PRINTK</constant> is set, this - function prints the given message together with the file name - and the line of the caller. The <constant>KERN_XXX</constant> - prefix is processed as + <function>printk()</function> function. + The <constant>KERN_XXX</constant> prefix is processed as well as the original <function>printk()</function> does, so it's recommended to add this prefix, e.g. Index: usb-3.10/arch/arm/configs/eseries_pxa_defconfig =================================================================== --- usb-3.10.orig/arch/arm/configs/eseries_pxa_defconfig +++ usb-3.10/arch/arm/configs/eseries_pxa_defconfig @@ -90,7 +90,6 @@ CONFIG_SND=m CONFIG_SND_MIXER_OSS=m CONFIG_SND_PCM_OSS=m CONFIG_SND_DYNAMIC_MINORS=y -CONFIG_SND_VERBOSE_PRINTK=y # CONFIG_SND_PCMCIA is not set CONFIG_SND_SOC=m CONFIG_SND_PXA2XX_SOC=m Index: usb-3.10/arch/arm/configs/omap2plus_defconfig =================================================================== --- usb-3.10.orig/arch/arm/configs/omap2plus_defconfig +++ usb-3.10/arch/arm/configs/omap2plus_defconfig @@ -192,7 +192,6 @@ CONFIG_SOUND=m CONFIG_SND=m CONFIG_SND_MIXER_OSS=m CONFIG_SND_PCM_OSS=m -CONFIG_SND_VERBOSE_PRINTK=y CONFIG_SND_DEBUG=y CONFIG_SND_USB_AUDIO=m CONFIG_SND_SOC=m Index: usb-3.10/arch/arm/configs/s3c2410_defconfig =================================================================== --- usb-3.10.orig/arch/arm/configs/s3c2410_defconfig +++ usb-3.10/arch/arm/configs/s3c2410_defconfig @@ -315,7 +315,6 @@ CONFIG_SND_SEQUENCER=m CONFIG_SND_MIXER_OSS=m CONFIG_SND_PCM_OSS=m CONFIG_SND_SEQUENCER_OSS=y -CONFIG_SND_VERBOSE_PRINTK=y # CONFIG_SND_DRIVERS is not set # CONFIG_SND_ARM is not set # CONFIG_SND_SPI is not set Index: usb-3.10/arch/arm/configs/trizeps4_defconfig =================================================================== --- usb-3.10.orig/arch/arm/configs/trizeps4_defconfig +++ usb-3.10/arch/arm/configs/trizeps4_defconfig @@ -159,7 +159,6 @@ CONFIG_SND=y CONFIG_SND_SEQUENCER=y CONFIG_SND_MIXER_OSS=y CONFIG_SND_PCM_OSS=y -CONFIG_SND_VERBOSE_PRINTK=y CONFIG_SND_DEBUG=y CONFIG_SND_PXA2XX_AC97=y CONFIG_SND_USB_AUDIO=m Index: usb-3.10/arch/ia64/configs/generic_defconfig =================================================================== --- usb-3.10.orig/arch/ia64/configs/generic_defconfig +++ usb-3.10/arch/ia64/configs/generic_defconfig @@ -127,7 +127,6 @@ CONFIG_SND_SEQ_DUMMY=m CONFIG_SND_MIXER_OSS=m CONFIG_SND_PCM_OSS=m CONFIG_SND_SEQUENCER_OSS=y -CONFIG_SND_VERBOSE_PRINTK=y CONFIG_SND_DUMMY=m CONFIG_SND_VIRMIDI=m CONFIG_SND_MTPAV=m Index: usb-3.10/arch/ia64/configs/gensparse_defconfig =================================================================== --- usb-3.10.orig/arch/ia64/configs/gensparse_defconfig +++ usb-3.10/arch/ia64/configs/gensparse_defconfig @@ -115,7 +115,6 @@ CONFIG_SND_SEQ_DUMMY=m CONFIG_SND_MIXER_OSS=m CONFIG_SND_PCM_OSS=m CONFIG_SND_SEQUENCER_OSS=y -CONFIG_SND_VERBOSE_PRINTK=y CONFIG_SND_DUMMY=m CONFIG_SND_VIRMIDI=m CONFIG_SND_MTPAV=m Index: usb-3.10/arch/mips/configs/pnx8335_stb225_defconfig =================================================================== --- usb-3.10.orig/arch/mips/configs/pnx8335_stb225_defconfig +++ usb-3.10/arch/mips/configs/pnx8335_stb225_defconfig @@ -73,7 +73,6 @@ CONFIG_SND_SEQUENCER=m CONFIG_SND_MIXER_OSS=m CONFIG_SND_PCM_OSS=m CONFIG_SND_SEQUENCER_OSS=y -CONFIG_SND_VERBOSE_PRINTK=y CONFIG_SND_DEBUG=y CONFIG_EXT2_FS=m # CONFIG_DNOTIFY is not set Index: usb-3.10/arch/powerpc/configs/ppc6xx_defconfig =================================================================== --- usb-3.10.orig/arch/powerpc/configs/ppc6xx_defconfig +++ usb-3.10/arch/powerpc/configs/ppc6xx_defconfig @@ -870,7 +870,6 @@ CONFIG_SND_PCM_OSS=m CONFIG_SND_SEQUENCER_OSS=y CONFIG_SND_DYNAMIC_MINORS=y # CONFIG_SND_SUPPORT_OLD_API is not set -CONFIG_SND_VERBOSE_PRINTK=y CONFIG_SND_DEBUG=y CONFIG_SND_DEBUG_VERBOSE=y CONFIG_SND_PCM_XRUN_DEBUG=y Index: usb-3.10/arch/sh/configs/edosk7760_defconfig =================================================================== --- usb-3.10.orig/arch/sh/configs/edosk7760_defconfig +++ usb-3.10/arch/sh/configs/edosk7760_defconfig @@ -90,7 +90,6 @@ CONFIG_SOUND=y CONFIG_SND=y # CONFIG_SND_SUPPORT_OLD_API is not set # CONFIG_SND_VERBOSE_PROCFS is not set -CONFIG_SND_VERBOSE_PRINTK=y CONFIG_SND_SOC=y # CONFIG_HID_SUPPORT is not set # CONFIG_USB_SUPPORT is not set Index: usb-3.10/arch/sh/configs/sh7785lcr_32bit_defconfig =================================================================== --- usb-3.10.orig/arch/sh/configs/sh7785lcr_32bit_defconfig +++ usb-3.10/arch/sh/configs/sh7785lcr_32bit_defconfig @@ -105,7 +105,6 @@ CONFIG_SND_HRTIMER=y CONFIG_SND_DYNAMIC_MINORS=y # CONFIG_SND_SUPPORT_OLD_API is not set # CONFIG_SND_VERBOSE_PROCFS is not set -CONFIG_SND_VERBOSE_PRINTK=y CONFIG_SND_DEBUG=y CONFIG_SND_DEBUG_VERBOSE=y # CONFIG_SND_DRIVERS is not set Index: usb-3.10/include/sound/core.h =================================================================== --- usb-3.10.orig/include/sound/core.h +++ usb-3.10/include/sound/core.h @@ -335,7 +335,7 @@ enum { SND_PR_VERBOSE, }; -#if defined(CONFIG_SND_DEBUG) || defined(CONFIG_SND_VERBOSE_PRINTK) +#if defined(CONFIG_SND_DEBUG) __printf(4, 5) void __snd_printk(unsigned int level, const char *file, int line, const char *format, ...); @@ -349,7 +349,7 @@ void __snd_printk(unsigned int level, co * @fmt: format string * * Works like printk() but prints the file and the line of the caller - * when configured with CONFIG_SND_VERBOSE_PRINTK. + * when configured with CONFIG_SND_DEBUG. */ #define snd_printk(fmt, args...) \ __snd_printk(0, __FILE__, __LINE__, fmt, ##args) Index: usb-3.10/sound/core/Kconfig =================================================================== --- usb-3.10.orig/sound/core/Kconfig +++ usb-3.10/sound/core/Kconfig @@ -173,15 +173,6 @@ config SND_VERBOSE_PROCFS useful information to developers when a problem occurs). On the other side, it makes the ALSA subsystem larger. -config SND_VERBOSE_PRINTK - bool "Verbose printk" - help - Say Y here to enable verbose log messages. These messages - will help to identify source file and position containing - printed messages. - - You don't need this unless you're debugging ALSA. - config SND_DEBUG bool "Debug" help Index: usb-3.10/sound/core/misc.c =================================================================== --- usb-3.10.orig/sound/core/misc.c +++ usb-3.10/sound/core/misc.c @@ -51,27 +51,11 @@ void release_and_free_resource(struct re EXPORT_SYMBOL(release_and_free_resource); -#ifdef CONFIG_SND_VERBOSE_PRINTK -/* strip the leading path if the given path is absolute */ -static const char *sanity_file_name(const char *path) -{ - if (*path == '/') - return strrchr(path, '/') + 1; - else - return path; -} -#endif - -#if defined(CONFIG_SND_DEBUG) || defined(CONFIG_SND_VERBOSE_PRINTK) +#if defined(CONFIG_SND_DEBUG) void __snd_printk(unsigned int level, const char *path, int line, const char *format, ...) { va_list args; -#ifdef CONFIG_SND_VERBOSE_PRINTK - int kern_level; - struct va_format vaf; - char verbose_fmt[] = KERN_DEFAULT "ALSA %s:%d %pV"; -#endif #ifdef CONFIG_SND_DEBUG if (debug < level) @@ -79,22 +63,7 @@ void __snd_printk(unsigned int level, co #endif va_start(args, format); -#ifdef CONFIG_SND_VERBOSE_PRINTK - vaf.fmt = format; - vaf.va = &args; - - kern_level = printk_get_level(format); - if (kern_level) { - const char *end_of_header = printk_skip_level(format); - memcpy(verbose_fmt, format, end_of_header - format); - vaf.fmt = end_of_header; - } else if (level) - memcpy(verbose_fmt, KERN_DEBUG, sizeof(KERN_DEBUG) - 1); - printk(verbose_fmt, sanity_file_name(path), line, &vaf); - -#else vprintk(format, args); -#endif va_end(args); } EXPORT_SYMBOL_GPL(__snd_printk); Index: usb-3.10/sound/pci/hda/hda_intel.c =================================================================== --- usb-3.10.orig/sound/pci/hda/hda_intel.c +++ usb-3.10/sound/pci/hda/hda_intel.c @@ -189,11 +189,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6}, "{ULI, M5461}}"); MODULE_DESCRIPTION("Intel HDA driver"); -#ifdef CONFIG_SND_VERBOSE_PRINTK -#define SFX /* nop */ -#else #define SFX "hda-intel " -#endif #if defined(CONFIG_PM) && defined(CONFIG_VGA_SWITCHEROO) #ifdef CONFIG_SND_HDA_CODEC_HDMI

4 37

[alsa-devel] [ANNOUNCE] tinycompress: release 0.2.0
by Vinod Koul 07 Jun '13

07 Jun '13

Hello folks, Time to cut another release for the library i am working on. tinycompress library - the user mode library for using the ALSA compressed APIs has received following new features and now is v0.2.0 - support for non blocking I/O - support for recording compressed audio content - no poll when writing last file buffer - get_alsa_rate helper for rate conversion - bunch of other fixes - now all releases will be tagged accordingly For details & contributions, pls see the below changelog. Lakshmi N Vinnakota (1): fix error reporting in tinycompress Richard Fitzgerald (6): tinycompress: cplay: Correct a couple of small message typos compress: Add support for capture streams compress: fix hpointer error when no sample rate compress: copy final version of config struct compress: do not poll if enough space to write remaining data compress: Add non-blocking I/O Vinod Koul (4): compress: check for config first add utils file with rate conversion helper cplay: use get_alsa_rate to convert sample rate Tinycompress version 0.2.0 Android.mk | 2 +- compress.c | 171 +++++++++++++++++++++++++++-------- cplay.c | 52 ++--------- include/tinycompress/tinycompress.h | 27 +++++- include/tinycompress/version.h | 4 +- makefile.linux | 2 +- utils.c | 96 ++++++++++++++++++++ 7 files changed, 263 insertions(+), 91 deletions(-) create mode 100644 utils.c Thanks ~Vinod --

1 0

[alsa-devel] [PATCH V4] ASoC: fsl: add imx-wm8962 machine driver
by Nicolin Chen 07 Jun '13

07 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. .../devicetree/bindings/sound/imx-audio-wm8962.txt | 55 ++++ sound/soc/fsl/Kconfig | 12 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-wm8962.c | 336 ++++++++++++++++++++ 4 files changed, 405 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..0b2eb65 --- /dev/null +++ b/sound/soc/fsl/imx-wm8962.c @@ -0,0 +1,336 @@ +/* + * 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, "could not get codec clk: %d\n", ret); + goto fail; + } + + data->clk_frequency = clk_get_rate(data->codec_clk); + clk_prepare_enable(data->codec_clk); + + 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 4

[alsa-devel] Question on wait_for_avail function
by gsantosh＠codeaurora.org 07 Jun '13

07 Jun '13

Hi Liam, Consider an use case 1) Write is blocked as there is no space to write the data, after the return from snd_pcm_playback_avail()function in wait_for_avail 2) HAL issues pause in parallel to the write I see that the following function snd_pcm_post_pause unblock the write wake_up(&runtime->sleep); wake_up(&runtime->tsleep); But i do not see any case handling for the pause state in the wait_for_avail function, it is not coming out of the for loop as there will not be any change in the buffer states. Please explain what is the intention behind the wake in pause callback. Want to understand why there is no handling of pause in wait_for_avail function. and also is there any restriction from ALSA framework that we should not issue pause while write is blocked or going on. Regards, Santosh M G.

4 5

[alsa-devel] [PATCH V3] ASoC: fsl: add imx-wm8962 machine driver
by Nicolin Chen 07 Jun '13

07 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; .../devicetree/bindings/sound/imx-audio-wm8962.txt | 52 +++ sound/soc/fsl/Kconfig | 12 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-wm8962.c | 336 ++++++++++++++++++++ 4 files changed, 402 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..bb76097 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt @@ -0,0 +1,52 @@ +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 + +Optional properties: +- hp-det-gpios : The gpio port of Headphone jack detection. +- mic-det-gpios: The gpio port of Micphone jack detection. + +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>; + 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..0b2eb65 --- /dev/null +++ b/sound/soc/fsl/imx-wm8962.c @@ -0,0 +1,336 @@ +/* + * 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, "could not get codec clk: %d\n", ret); + goto fail; + } + + data->clk_frequency = clk_get_rate(data->codec_clk); + clk_prepare_enable(data->codec_clk); + + 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

2 1

[alsa-devel] [PATCH V3] ASoC: WM8962: Add device tree binding
by Nicolin Chen 07 Jun '13

07 Jun '13

Document the device tree binding for the WM8962 codec, and modify the driver to extract platform data from the device tree, if present. Based on work of WM8903 by Stephen Warren <swarren(a)nvidia.com> Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- ChangeLog: v1->v2: * Dropped out-of-band value error msg by regarding it as default value. v2->v3: * Revised binding doc to warn users if property absent then use default value. Documentation/devicetree/bindings/sound/wm8962.txt | 23 +++++++++++++ sound/soc/codecs/wm8962.c | 35 +++++++++++++++++++- 2 files changed, 57 insertions(+), 1 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/wm8962.txt b/Documentation/devicetree/bindings/sound/wm8962.txt index dceb3b1..7f82b59 100644 --- a/Documentation/devicetree/bindings/sound/wm8962.txt +++ b/Documentation/devicetree/bindings/sound/wm8962.txt @@ -8,9 +8,32 @@ Required properties: - reg : the I2C address of the device. +Optional properties: + - spk-mono: This is a boolean property. If present, the SPK_MONO bit + of R51 (Class D Control 2) gets set, indicating that the speaker is + in mono mode. + + - mic-cfg : Default register value for R48 (Additional Control 4). + If absent, the default should be the register default. + + - gpio-cfg : A list of GPIO configuration register values. The list must + be 6 entries long. If absent, no configuration of these registers is + performed. And note that only the value within [0x0, 0xffff] is valid. + Any other value is regarded as setting the GPIO register by its reset + value 0x0. + Example: codec: wm8962@1a { compatible = "wlf,wm8962"; reg = <0x1a>; + + gpio-cfg = < + 0x0000 /* 0:Default */ + 0x0000 /* 1:Default */ + 0x0013 /* 2:FN_DMICCLK */ + 0x0000 /* 3:Default */ + 0x8014 /* 4:FN_DMICCDAT */ + 0x0000 /* 5:Default */ + >; }; diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index d56dd86..26219ea 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3584,6 +3584,34 @@ static const struct regmap_config wm8962_regmap = { .cache_type = REGCACHE_RBTREE, }; +static int wm8962_set_pdata_from_of(struct i2c_client *i2c, + struct wm8962_pdata *pdata) +{ + const struct device_node *np = i2c->dev.of_node; + u32 val32; + int i; + + if (of_property_read_bool(np, "spk-mono")) + pdata->spk_mono = true; + + if (of_property_read_u32(np, "mic-cfg", &val32) >= 0) + pdata->mic_cfg = val32; + + if (of_property_read_u32_array(np, "gpio-cfg", pdata->gpio_init, + ARRAY_SIZE(pdata->gpio_init)) >= 0) + for (i = 0; i < ARRAY_SIZE(pdata->gpio_init); i++) { + /* + * The range of GPIO register value is [0x0, 0xffff] + * While the default value of each register is 0x0 + * Any other value will be regarded as default value + */ + if (pdata->gpio_init[i] > 0xffff) + pdata->gpio_init[i] = 0x0; + } + + return 0; +} + static int wm8962_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -3604,8 +3632,13 @@ static int wm8962_i2c_probe(struct i2c_client *i2c, wm8962->irq = i2c->irq; /* If platform data was supplied, update the default data in priv */ - if (pdata) + if (pdata) { memcpy(&wm8962->pdata, pdata, sizeof(struct wm8962_pdata)); + } else if (i2c->dev.of_node) { + ret = wm8962_set_pdata_from_of(i2c, &wm8962->pdata); + if (ret != 0) + return ret; + } for (i = 0; i < ARRAY_SIZE(wm8962->supplies); i++) wm8962->supplies[i].supply = wm8962_supply_names[i]; -- 1.7.1

2 1

[alsa-devel] [tinycompress]how should I call compress_open for aac ltp file playback?
by Keyon.Jey 07 Jun '13

07 Jun '13

I am trying to use tinycompress for hwdecoder. everything works fine, but after compress_open (with profile = SND_AUDIOPROFILE_AAC level=0 ch_mode=0 rate_control=0), the playback of my aac-ltp-16bit-48KHz-Stereo file is faster than it should be. Am I wrong? struct snd_codec { __u32 id; __u32 ch_in; __u32 ch_out; __u32 sample_rate; __u32 bit_rate; __u32 rate_control; __u32 profile; __u32 level; __u32 ch_mode; __u32 format; __u32 align; union snd_codec_options options; __u32 reserved[3]; }; Keyon

3 4

[alsa-devel] [PATCH V2] ASoC: fsl: add imx-wm8962 machine driver
by Nicolin Chen 07 Jun '13

07 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; --- .../devicetree/bindings/sound/imx-audio-wm8962.txt | 61 +++ sound/soc/fsl/Kconfig | 12 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-wm8962.c | 387 ++++++++++++++++++++ 4 files changed, 462 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..cb188a6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt @@ -0,0 +1,61 @@ +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 + + WM8962 pins: + * HPOUTL + * HPOUTR + * SPKOUTL + * SPKOUTR + * MICBIAS + * IN3R + * DMIC + * DMICDAT + + 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 + +- hp-det-gpios : The gpio port of Headphone detection. +- mic-det-gpios: The gpio port of Micphone detection. + +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>; + 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..cb54da4 --- /dev/null +++ b/sound/soc/fsl/imx-wm8962.c @@ -0,0 +1,387 @@ +/* + * 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/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 check_hw_params(struct snd_pcm_substream *substream, + snd_pcm_format_t sample_format, unsigned int sample_rate) +{ + struct imx_priv *priv = &card_priv; + struct device *dev = substream->pcm->card->dev; + + substream->runtime->sample_bits = + snd_pcm_format_physical_width(sample_format); + substream->runtime->rate = sample_rate; + substream->runtime->format = sample_format; + + if (!priv->first_stream) { + priv->first_stream = substream; + } else { + priv->second_stream = substream; + + if (priv->first_stream->runtime->rate != + priv->second_stream->runtime->rate) { + dev_err(dev, "KEEP THE SAME SAMPLE RATE: %d!\n", + priv->first_stream->runtime->rate); + return -EINVAL; + } + if (priv->first_stream->runtime->sample_bits != + priv->second_stream->runtime->sample_bits) { + snd_pcm_format_t first_format = + priv->first_stream->runtime->format; + + dev_err(dev, "KEEP THE SAME FORMAT: %s!\n", + snd_pcm_format_name(first_format)); + return -EINVAL; + } + } + + return 0; +} + +static int imx_hifi_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 device *dev = substream->pcm->card->dev; + struct imx_priv *priv = &card_priv; + struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + unsigned int sample_rate = params_rate(params); + unsigned int sample_format = params_format(params); + int ret = 0; + unsigned int pll_out; + + /* + * WM8962 doesn't support two substreams in different sample rates or + * sample formats. So check the two parameters of two substreams' when + * there're two substreams of playback and of capture in the same time. + */ + ret = check_hw_params(substream, sample_format, sample_rate); + if (ret < 0) { + dev_err(dev, "Failed to match hw params: %d\n", ret); + return ret; + } + + 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; + } + + return 0; +} + +static int imx_hifi_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct device *dev = substream->pcm->card->dev; + struct imx_priv *priv = &card_priv; + struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + int ret; + + /* We don't need to handle anything if there's no substream running */ + if (!priv->first_stream) + return 0; + + if (priv->first_stream == substream) + priv->first_stream = priv->second_stream; + priv->second_stream = NULL; + + if (!priv->first_stream) { + /* + * SYSCLK of wm8962's not disabled yet. If we wanna close FLL, + * we need to keep SYSCLK alive by changing its source to MCLK. + */ + 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: %d\n", ret); + return ret; + } + + /* Disable FLL and let codec do pm_runtime_put() */ + ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, 0, 0, 0); + if (ret < 0) { + dev_err(dev, "Failed to disable FLL: %d\n", ret); + return ret; + } + } + + return 0; +} + +static struct snd_soc_ops imx_hifi_ops = { + .hw_params = imx_hifi_hw_params, + .hw_free = imx_hifi_hw_free, +}; + +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 = clk_get(&codec_dev->dev, NULL); + if (IS_ERR(data->codec_clk)) { + /* assuming clock enabled by default */ + data->codec_clk = NULL; + ret = of_property_read_u32(codec_np, "clock-frequency", + &data->clk_frequency); + if (ret) { + dev_err(&codec_dev->dev, + "clock-frequency missing or invalid\n"); + goto fail; + } + } else { + data->clk_frequency = clk_get_rate(data->codec_clk); + clk_prepare_enable(data->codec_clk); + } + + 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, + + 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 (data->codec_clk) { + clk_disable_unprepare(data->codec_clk); + clk_put(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 (data->codec_clk) { + clk_disable_unprepare(data->codec_clk); + clk_put(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

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[alsa-devel] [tinycompress][PATCH 0/2] Add support for non-blocking I/O
by Richard Fitzgerald 07 Jun '13

07 Jun '13

(Applies on top of the two patches I just uploaded to fix the write logic) Adds support for non-blocking write and read. The API for this is based on the non-blocking API used in the standard PCM alsalib.

2 3

[alsa-devel] [tinycompress][PATCH 0/2] Fixes for write loop logic
by Richard Fitzgerald 07 Jun '13

07 Jun '13

Two small patches which correct the logic of compress_write() so that it will work correctly if asked to write a buffer shorter than the fragment length. The first patch is needed to ensure that if the client requests auto-configuration of fragment size the internal copy of the config contains the configured fragment size. The second patch contains the actual fix to the write loop logic. Richard Fitzgerald (2): compress: copy final version of config struct compress: do not poll if enough space to write remaining data compress.c | 13 ++++++------- 1 files changed, 6 insertions(+), 7 deletions(-) -- 1.7.2.5

2 3

[alsa-devel] Sound issues with ALC269VB-based hardware
by Anisse Astier 06 Jun '13

06 Jun '13

Hi, So I have this hardware that has a few issues (all tested with tiwai/sound for-next) Audio out/speakers: - Audio does not work by default. It needs plugging a jack in either headphone or mic. - disabling "Enable Automute" does make it work. But then it doesn't mute when jack is plugged (obviously). Mic: - needs model=laptop-dmic so that working "Internal Mic 1" appears - "Internal Mic" is useless. - no jack sense, need to select manually "Internal Mic 1" or "Mic" (external) as Input Source Please find alsa-info.sh with and without model=laptop-dmic in attachement. Regards, Anisse

3 17

[alsa-devel] Generic clock divider indices
by Daniel Mack 06 Jun '13

06 Jun '13

Hi Mark, Liam, we're using generic ASoC platform code for some audio devices, which is agnostic of the actual Codecs that it deals with, as the information is provided by DT nodes only. That works very well so far, but I'm now facing a case where I need to pass a special clock divider information down to the codec driver. The fact that the IDs of the dividers can be arbitrarily defined by the drivers leads to the unfortunate situation that I now have to make the platform code aware of the codec drivers again. Would it be an idea to define some commonly used dividers in the core? If they start at 0x80000000, they won't collide with existing ones, and we could simply add them as alternative case statements to existing drivers. Some commonly used ones I can think of are MCLK / BCLK MCLK / LRCLK BCLK / LRCLK That way, platform code could just pass the values dows to the codec drivers, whether or not they know what to do with it. What do you think? Daniel

2 5

Re: [alsa-devel] "https://bugtrack.alsa-project.org/alsa-bug" Tascam US122L
by Takashi Iwai 06 Jun '13

06 Jun '13

Please don't drop Cc to ML. At Thu, 6 Jun 2013 15:03:56 +0100, James Miller wrote: > I'm using us122l. I've read this : > https://bugzilla.kernel.org/show_bug.cgi?id=53241 OK, then try to stace and see what gives error. Also, build your driver with CONFIG_SND_DEBUG_VERBOSE=y and see the kernel messages, too. The problem looks like the mmap. The strange thing is that the mmap code hasn't been changed so much and it's relatively simple in us122l.c. > and the bug is being addressed but there seems like no change so far. When > you say the drivers are pretty hackish - do you think that I might be able > to address them myself? I am about to begin some driver programming and > have a little previous development experience plus I'd be really stoked for > a decent us122l driver. If you know of any resources that might help with > the module driver code then I'd be grateful. Thanks. Nothing special I know of, the standard kernel coding book like "Linux Device Drivers" should be a good starting point to learn. Takashi > On Thu, Jun 6, 2013 at 2:33 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > > > At Thu, 6 Jun 2013 13:56:32 +0100, > > James Miller wrote: > > > > > > Hi, I can't visit this bugtracker page (above) and am also wondering if > > > there is any progress on the Tascam US122L. I understand someone from > > the > > > kxaudio team has tried a kernel bisect but that the problem lies in the > > > driver with a vm switch. > > > > > > http://www.remastersys.com/forums/index.php?topic=3013.0 > > > > > > somewhere I read that this was an alsa problem but I found this kernel > > > patch (referred to) and it has been suggested that it is this that is the > > > cause. I am not a prof. developer but might have a go at recompiling my > > > kernel after changing the module so that I can get my soundcard working > > > with kernel 3.8.13-rt(patch). Do you guys know anything about this that > > > might help me? > > > > > > https://patchwork.kernel.org/patch/1258951/ > > > > Which driver are you using? Is snd-usb-us122l or snd-usb-usx2y? > > > > If the patch above is really the culprit of breakage, the driver must > > have been already broken before that and it just had worked with > > luck. Both drivers are pretty hackish, and need real fixes... > > > > > > Takashi > > > > > > -- > James Stewart Miller Bsc(hons) Psych. > [2 <text/html; ISO-8859-1 (quoted-printable)>] >

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[alsa-devel] "https://bugtrack.alsa-project.org/alsa-bug" Tascam US122L
by James Miller 06 Jun '13

06 Jun '13

Hi, I can't visit this bugtracker page (above) and am also wondering if there is any progress on the Tascam US122L. I understand someone from the kxaudio team has tried a kernel bisect but that the problem lies in the driver with a vm switch. http://www.remastersys.com/forums/index.php?topic=3013.0 somewhere I read that this was an alsa problem but I found this kernel patch (referred to) and it has been suggested that it is this that is the cause. I am not a prof. developer but might have a go at recompiling my kernel after changing the module so that I can get my soundcard working with kernel 3.8.13-rt(patch). Do you guys know anything about this that might help me? https://patchwork.kernel.org/patch/1258951/ thanks James ubuntu studio on mac pro 2006 3.8.13-rt9 #1 SMP PREEMPT RT Thu May 30 18:54:25 BST 2013 x86_64 x86_64 x86_64 GNU/Linux -- James Stewart Miller Bsc(hons) Psych.

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[alsa-devel] [PATCH V2 0/2] Add device tree binding support to WM8962 codec driver
by Nicolin Chen 06 Jun '13

06 Jun '13

*** These two patch added devicetree binding support to WM8962 codec driver *** so that it can fetch pdata not only from old platform_data but also from *** newer dts binding. --- ChangeLog: v1->v2: * PATCH 1/2: *** Embedded a copy of struct wm8962_pdata instead of its pointer *** Removed all 'if (pdata)' checking code as it is always valid. * PATCH 2/2: *** Dropped out-of-band value error msg by regarding it as default value. Nicolin Chen (2): ASoC: WM8962: Create default platform data structure ASoC: WM8962: Add device tree binding Documentation/devicetree/bindings/sound/wm8962.txt | 20 ++++ sound/soc/codecs/wm8962.c | 99 +++++++++++++------ 2 files changed, 88 insertions(+), 31 deletions(-)

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[alsa-devel] [PATCH 1/2] ALSA: hda - Introduce bit flags to snd_hda_codec_read/write()
by Takashi Iwai 06 Jun '13

06 Jun '13

snd_hda_codec_read(), snd_hda_codec_write() & co take the argument "direct" that indicates whether the given NID is a direct reference or an indirect reference. However, the indirect reference is practically unimplemented and never exists. And moreover, we don't need the indication of indirect reference at this high level, as NID can be represented in 16bit values at this point. Meanwhile, there are some cases where it'd be nice to give some operational options to these functions. So, we can reuse this argument as a new bit flag! Pretty frugal, eh? All callers so far pass zero to this argument, thus there is no behavior change by this replacement. The real usage of this new bit option will be added in the following patches. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 41 ++++++++++++++++++++--------------------- sound/pci/hda/hda_codec.h | 8 ++++---- 2 files changed, 24 insertions(+), 25 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 679fba4..503869a 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -185,20 +185,19 @@ EXPORT_SYMBOL_HDA(snd_hda_get_jack_type); * Compose a 32bit command word to be sent to the HD-audio controller */ static inline unsigned int -make_codec_cmd(struct hda_codec *codec, hda_nid_t nid, int direct, +make_codec_cmd(struct hda_codec *codec, hda_nid_t nid, int flags, unsigned int verb, unsigned int parm) { u32 val; - if ((codec->addr & ~0xf) || (direct & ~1) || (nid & ~0x7f) || + if ((codec->addr & ~0xf) || (nid & ~0x7f) || (verb & ~0xfff) || (parm & ~0xffff)) { - printk(KERN_ERR "hda-codec: out of range cmd %x:%x:%x:%x:%x\n", - codec->addr, direct, nid, verb, parm); + printk(KERN_ERR "hda-codec: out of range cmd %x:%x:%x:%x\n", + codec->addr, nid, verb, parm); return ~0; } val = (u32)codec->addr << 28; - val |= (u32)direct << 27; val |= (u32)nid << 20; val |= verb << 8; val |= parm; @@ -209,7 +208,7 @@ make_codec_cmd(struct hda_codec *codec, hda_nid_t nid, int direct, * Send and receive a verb */ static int codec_exec_verb(struct hda_codec *codec, unsigned int cmd, - unsigned int *res) + int flags, unsigned int *res) { struct hda_bus *bus = codec->bus; int err; @@ -255,7 +254,7 @@ static int codec_exec_verb(struct hda_codec *codec, unsigned int cmd, * snd_hda_codec_read - send a command and get the response * @codec: the HDA codec * @nid: NID to send the command - * @direct: direct flag + * @flags: optional bit flags * @verb: the verb to send * @parm: the parameter for the verb * @@ -264,12 +263,12 @@ static int codec_exec_verb(struct hda_codec *codec, unsigned int cmd, * Returns the obtained response value, or -1 for an error. */ unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid, - int direct, + int flags, unsigned int verb, unsigned int parm) { - unsigned cmd = make_codec_cmd(codec, nid, direct, verb, parm); + unsigned cmd = make_codec_cmd(codec, nid, flags, verb, parm); unsigned int res; - if (codec_exec_verb(codec, cmd, &res)) + if (codec_exec_verb(codec, cmd, flags, &res)) return -1; return res; } @@ -279,7 +278,7 @@ EXPORT_SYMBOL_HDA(snd_hda_codec_read); * snd_hda_codec_write - send a single command without waiting for response * @codec: the HDA codec * @nid: NID to send the command - * @direct: direct flag + * @flags: optional bit flags * @verb: the verb to send * @parm: the parameter for the verb * @@ -287,12 +286,12 @@ EXPORT_SYMBOL_HDA(snd_hda_codec_read); * * Returns 0 if successful, or a negative error code. */ -int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int direct, - unsigned int verb, unsigned int parm) +int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int flags, + unsigned int verb, unsigned int parm) { - unsigned int cmd = make_codec_cmd(codec, nid, direct, verb, parm); + unsigned int cmd = make_codec_cmd(codec, nid, flags, verb, parm); unsigned int res; - return codec_exec_verb(codec, cmd, + return codec_exec_verb(codec, cmd, flags, codec->bus->sync_write ? &res : NULL); } EXPORT_SYMBOL_HDA(snd_hda_codec_write); @@ -3582,7 +3581,7 @@ EXPORT_SYMBOL_HDA(snd_hda_create_spdif_in_ctls); * snd_hda_codec_write_cache - send a single command with caching * @codec: the HDA codec * @nid: NID to send the command - * @direct: direct flag + * @flags: optional bit flags * @verb: the verb to send * @parm: the parameter for the verb * @@ -3591,7 +3590,7 @@ EXPORT_SYMBOL_HDA(snd_hda_create_spdif_in_ctls); * Returns 0 if successful, or a negative error code. */ int snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid, - int direct, unsigned int verb, unsigned int parm) + int flags, unsigned int verb, unsigned int parm) { int err; struct hda_cache_head *c; @@ -3600,7 +3599,7 @@ int snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid, cache_only = codec->cached_write; if (!cache_only) { - err = snd_hda_codec_write(codec, nid, direct, verb, parm); + err = snd_hda_codec_write(codec, nid, flags, verb, parm); if (err < 0) return err; } @@ -3624,7 +3623,7 @@ EXPORT_SYMBOL_HDA(snd_hda_codec_write_cache); * snd_hda_codec_update_cache - check cache and write the cmd only when needed * @codec: the HDA codec * @nid: NID to send the command - * @direct: direct flag + * @flags: optional bit flags * @verb: the verb to send * @parm: the parameter for the verb * @@ -3635,7 +3634,7 @@ EXPORT_SYMBOL_HDA(snd_hda_codec_write_cache); * Returns 0 if successful, or a negative error code. */ int snd_hda_codec_update_cache(struct hda_codec *codec, hda_nid_t nid, - int direct, unsigned int verb, unsigned int parm) + int flags, unsigned int verb, unsigned int parm) { struct hda_cache_head *c; u32 key; @@ -3651,7 +3650,7 @@ int snd_hda_codec_update_cache(struct hda_codec *codec, hda_nid_t nid, return 0; } mutex_unlock(&codec->bus->cmd_mutex); - return snd_hda_codec_write_cache(codec, nid, direct, verb, parm); + return snd_hda_codec_write_cache(codec, nid, flags, verb, parm); } EXPORT_SYMBOL_HDA(snd_hda_codec_update_cache); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index c93f902..39a658e 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -945,9 +945,9 @@ int snd_hda_codec_update_widgets(struct hda_codec *codec); * low level functions */ unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid, - int direct, + int flags, unsigned int verb, unsigned int parm); -int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int direct, +int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int flags, unsigned int verb, unsigned int parm); #define snd_hda_param_read(codec, nid, param) \ snd_hda_codec_read(codec, nid, 0, AC_VERB_PARAMETERS, param) @@ -986,11 +986,11 @@ int snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex); /* cached write */ int snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid, - int direct, unsigned int verb, unsigned int parm); + int flags, unsigned int verb, unsigned int parm); void snd_hda_sequence_write_cache(struct hda_codec *codec, const struct hda_verb *seq); int snd_hda_codec_update_cache(struct hda_codec *codec, hda_nid_t nid, - int direct, unsigned int verb, unsigned int parm); + int flags, unsigned int verb, unsigned int parm); void snd_hda_codec_resume_cache(struct hda_codec *codec); /* both for cmd & amp caches */ void snd_hda_codec_flush_cache(struct hda_codec *codec); -- 1.8.3

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[alsa-devel] [PATCH] ALSA: hda - Drop hard dependency on CONFIG_SND_DYNAMIC_MINORS
by Takashi Iwai 06 Jun '13

06 Jun '13

Currently HDMI codec driver sets the hard dependency (reverse selection) on CONFIG_SND_DYNAMIC_MINORS because the recent codecs may support more than two PCMs. But, this doesn't mean that we need always this option, since there can be a single PCM stream even with the modern codecs. This patch drops the hard dependency again but give more sensible error message when no enough PCMs are available due to the lack of this option. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/Kconfig | 1 - sound/pci/hda/hda_codec.c | 19 +++++++++++++++---- 2 files changed, 15 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 80a7d44..0c5371a 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -140,7 +140,6 @@ config SND_HDA_CODEC_VIA config SND_HDA_CODEC_HDMI bool "Build HDMI/DisplayPort HD-audio codec support" - select SND_DYNAMIC_MINORS default y help Say Y here to include HDMI and DisplayPort HD-audio codec diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 55108b5..679fba4 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4461,12 +4461,13 @@ const char *snd_hda_pcm_type_name[HDA_PCM_NTYPES] = { /* * get the empty PCM device number to assign - * - * note the max device number is limited by HDA_MAX_PCMS, currently 10 */ -static int get_empty_pcm_device(struct hda_bus *bus, int type) +static int get_empty_pcm_device(struct hda_bus *bus, unsigned int type) { /* audio device indices; not linear to keep compatibility */ + /* assigned to static slots up to dev#10; if more needed, assign + * the later slot dynamically (when CONFIG_SND_DYNAMIC_MINORS=y) + */ static int audio_idx[HDA_PCM_NTYPES][5] = { [HDA_PCM_TYPE_AUDIO] = { 0, 2, 4, 5, -1 }, [HDA_PCM_TYPE_SPDIF] = { 1, -1 }, @@ -4480,18 +4481,28 @@ static int get_empty_pcm_device(struct hda_bus *bus, int type) return -EINVAL; } - for (i = 0; audio_idx[type][i] >= 0 ; i++) + for (i = 0; audio_idx[type][i] >= 0; i++) { +#ifndef CONFIG_SND_DYNAMIC_MINORS + if (audio_idx[type][i] >= 8) + break; +#endif if (!test_and_set_bit(audio_idx[type][i], bus->pcm_dev_bits)) return audio_idx[type][i]; + } +#ifdef CONFIG_SND_DYNAMIC_MINORS /* non-fixed slots starting from 10 */ for (i = 10; i < 32; i++) { if (!test_and_set_bit(i, bus->pcm_dev_bits)) return i; } +#endif snd_printk(KERN_WARNING "Too many %s devices\n", snd_hda_pcm_type_name[type]); +#ifndef CONFIG_SND_DYNAMIC_MINORS + snd_printk(KERN_WARNING "Consider building the kernel with CONFIG_SND_DYNAMIC_MINORS=y\n"); +#endif return -EAGAIN; } -- 1.8.3

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[alsa-devel] hda question: 'Headphone Playback Switch' cannot unmute the pin's output amplifier?
by Lin, Mengdong 06 Jun '13

06 Jun '13

Hi, I'm using a Realtek HD-A codec ALC282. But the headphone output amplifier is always muted and I cannot unmute it by amixer. I tried to unmute the pin via control 'Headphone Playback Switch', but snd_hda_mixer_amp_switch_put() gets a ZERO value from ucontrol->value.integer.value, thus the pin will be muted. Could somebody give me some tips to further check this issue? Many thanks! Here are the headphone pin info: Node 0x21 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Headphone Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000001c: OUT HP Detect Pin Default 0x0321401f: [Jack] HP Out at Ext Left Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0xc0: OUT HP Unsolicited: tag=01, enabled=1 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 2 0x0c 0x0d* Here is the pin's controls listed by amixer: numid=2,iface=MIXER,name='Headphone Playback Switch' Then I use this command to unmute the pin's amplifier: amixer -c 1 cset numid=2,iface=MIXER,name='Headphone Playback Switch' on My debug message show that snd_hda_mixer_amp_switch_put() gets zero value , indicating to 'mute': [ 346.263519] snd_hda_mixer_amp_switch_put: nid 0x0021, chs 3, control value [0x00000000, 0x00000000] [ 346.263527] snd_hda_codec_amp_update: nid 0x0021 [ 346.263532] codec_amp_update: nid 0x0021, ch 0, direction 1 (OUT 1), index 0, val 0x00000080 [ 346.263536] codec_amp_update: amp val not change [ 346.263538] snd_hda_codec_amp_update: nid 0x0021 [ 346.263542] codec_amp_update: nid 0x0021, ch 1, direction 1 (OUT 1), index 0, val 0x00000080 [ 346.263544] codec_amp_update: amp val not change Thanks Mengdong

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[alsa-devel] [PATCH 0/2] Add device tree binding support to WM8962 codec driver
by Nicolin Chen 06 Jun '13

06 Jun '13

*** These two patch added devicetree binding support to WM8962 codec driver *** so that it can fetch pdata not only from old platform_data but also from *** newer dts binding. Nicolin Chen (2): ASoC: WM8962: Create default platform data structure ASoC: WM8962: Add device tree binding Documentation/devicetree/bindings/sound/wm8962.txt | 20 +++++++ sound/soc/codecs/wm8962.c | 56 ++++++++++++++++++-- 2 files changed, 72 insertions(+), 4 deletions(-)

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Re: [alsa-devel] [PATCH V5 4/5] ARM: dts: add clock provider information for i2s controllers in Exynos5250
by Padma Venkat 06 Jun '13

06 Jun '13

Hi Doug, On Tue, Jun 4, 2013 at 10:49 PM, Doug Anderson <dianders(a)chromium.org> wrote: > Padmavathi, > > On Tue, Jun 4, 2013 at 5:28 AM, Padmavathi Venna <padma.v(a)samsung.com> wrote: >> @@ -471,6 +477,8 @@ >> dmas = <&pdma1 12 >> &pdma1 11>; >> dma-names = "tx", "rx"; >> + clocks = <&clock 307>; >> + clock-names = "iis"; > > ...actually, glancing at the driver I'm a little surprised that you > don't need to list "i2s_opclk0". Did you test i2s1? I didn't test i2s1 but i2s_opclk0 is also required. I will resubmit the patch. > >> pinctrl-names = "default"; >> pinctrl-0 = <&i2s1_bus>; >> }; >> @@ -481,6 +489,8 @@ >> dmas = <&pdma0 12 >> &pdma0 11>; >> dma-names = "tx", "rx"; >> + clocks = <&clock 308>; >> + clock-names = "iis"; > > ...and here. > > > -Doug Thanks Padma

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[alsa-devel] [PATCH] ASoC: dapm: Treat DAI widgets like AIF widgets for power
by Mark Brown 06 Jun '13

06 Jun '13

Even though they are virtual widgets DAI widgets still get counted for the DAPM context power management so we can't just use the active state to check if they should be powered as they may not be part of a complete path. Instead split them into input and output widgets and do the same power checks as we perform on AIFs. Reported-by: Stephen Warren <swarren(a)nvidia.com> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- Compile tested only thus far. include/sound/soc-dapm.h | 3 ++- sound/soc/soc-dapm.c | 49 ++++++++++++++++++++++++---------------------- sound/soc/soc-pcm.c | 7 ++++++- 3 files changed, 34 insertions(+), 25 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index f4f85dd..69e8e07 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -452,7 +452,8 @@ enum snd_soc_dapm_type { snd_soc_dapm_aif_in, /* audio interface input */ snd_soc_dapm_aif_out, /* audio interface output */ snd_soc_dapm_siggen, /* signal generator */ - snd_soc_dapm_dai, /* link to DAI structure */ + snd_soc_dapm_dai_in, /* link to DAI structure */ + snd_soc_dapm_dai_out, snd_soc_dapm_dai_link, /* link between two DAI structures */ }; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 3507346..fdb14e4 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -55,7 +55,8 @@ static int dapm_up_seq[] = { [snd_soc_dapm_clock_supply] = 1, [snd_soc_dapm_micbias] = 2, [snd_soc_dapm_dai_link] = 2, - [snd_soc_dapm_dai] = 3, + [snd_soc_dapm_dai_in] = 3, + [snd_soc_dapm_dai_out] = 3, [snd_soc_dapm_aif_in] = 3, [snd_soc_dapm_aif_out] = 3, [snd_soc_dapm_mic] = 4, @@ -92,7 +93,8 @@ static int dapm_down_seq[] = { [snd_soc_dapm_value_mux] = 9, [snd_soc_dapm_aif_in] = 10, [snd_soc_dapm_aif_out] = 10, - [snd_soc_dapm_dai] = 10, + [snd_soc_dapm_dai_in] = 10, + [snd_soc_dapm_dai_out] = 10, [snd_soc_dapm_dai_link] = 11, [snd_soc_dapm_clock_supply] = 12, [snd_soc_dapm_regulator_supply] = 12, @@ -419,7 +421,8 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, case snd_soc_dapm_clock_supply: case snd_soc_dapm_aif_in: case snd_soc_dapm_aif_out: - case snd_soc_dapm_dai: + case snd_soc_dapm_dai_in: + case snd_soc_dapm_dai_out: case snd_soc_dapm_hp: case snd_soc_dapm_mic: case snd_soc_dapm_spk: @@ -813,7 +816,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, switch (widget->id) { case snd_soc_dapm_adc: case snd_soc_dapm_aif_out: - case snd_soc_dapm_dai: + case snd_soc_dapm_dai_out: if (widget->active) { widget->outputs = snd_soc_dapm_suspend_check(widget); return widget->outputs; @@ -909,7 +912,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, switch (widget->id) { case snd_soc_dapm_dac: case snd_soc_dapm_aif_in: - case snd_soc_dapm_dai: + case snd_soc_dapm_dai_in: if (widget->active) { widget->inputs = snd_soc_dapm_suspend_check(widget); return widget->inputs; @@ -1128,16 +1131,6 @@ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w) return out != 0 && in != 0; } -static int dapm_dai_check_power(struct snd_soc_dapm_widget *w) -{ - DAPM_UPDATE_STAT(w, power_checks); - - if (w->active) - return w->active; - - return dapm_generic_check_power(w); -} - /* Check to see if an ADC has power */ static int dapm_adc_check_power(struct snd_soc_dapm_widget *w) { @@ -2327,7 +2320,8 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, case snd_soc_dapm_clock_supply: case snd_soc_dapm_aif_in: case snd_soc_dapm_aif_out: - case snd_soc_dapm_dai: + case snd_soc_dapm_dai_in: + case snd_soc_dapm_dai_out: case snd_soc_dapm_dai_link: list_add(&path->list, &dapm->card->paths); list_add(&path->list_sink, &wsink->sources); @@ -3134,10 +3128,12 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, break; case snd_soc_dapm_adc: case snd_soc_dapm_aif_out: + case snd_soc_dapm_dai_out: w->power_check = dapm_adc_check_power; break; case snd_soc_dapm_dac: case snd_soc_dapm_aif_in: + case snd_soc_dapm_dai_in: w->power_check = dapm_dac_check_power; break; case snd_soc_dapm_pga: @@ -3157,9 +3153,6 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, case snd_soc_dapm_clock_supply: w->power_check = dapm_supply_check_power; break; - case snd_soc_dapm_dai: - w->power_check = dapm_dai_check_power; - break; default: w->power_check = dapm_always_on_check_power; break; @@ -3380,7 +3373,7 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, template.reg = SND_SOC_NOPM; if (dai->driver->playback.stream_name) { - template.id = snd_soc_dapm_dai; + template.id = snd_soc_dapm_dai_in; template.name = dai->driver->playback.stream_name; template.sname = dai->driver->playback.stream_name; @@ -3398,7 +3391,7 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, } if (dai->driver->capture.stream_name) { - template.id = snd_soc_dapm_dai; + template.id = snd_soc_dapm_dai_out; template.name = dai->driver->capture.stream_name; template.sname = dai->driver->capture.stream_name; @@ -3428,8 +3421,13 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) /* For each DAI widget... */ list_for_each_entry(dai_w, &card->widgets, list) { - if (dai_w->id != snd_soc_dapm_dai) + switch (w->id) { + case snd_soc_dapm_dai_in: + case snd_soc_dapm_dai_out: + break; + default: continue; + } dai = dai_w->priv; @@ -3438,8 +3436,13 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) if (w->dapm != dai_w->dapm) continue; - if (w->id == snd_soc_dapm_dai) + switch (w->id) { + case snd_soc_dapm_dai_in: + case snd_soc_dapm_dai_out: continue; + default: + break; + } if (!w->sname) continue; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index a14335e..b6c6403 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -931,8 +931,13 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream, /* Create any new FE <--> BE connections */ for (i = 0; i < list->num_widgets; i++) { - if (list->widgets[i]->id != snd_soc_dapm_dai) + switch (list->widgets[i]->id) { + case snd_soc_dapm_dai_in: + case snd_soc_dapm_dai_out: + break; + default: continue; + } /* is there a valid BE rtd for this widget */ be = dpcm_get_be(card, list->widgets[i], stream); -- 1.7.10.4

3 2

[alsa-devel] [PATCH] ASoC: max98090: Guard runtime PM callbacks
by Mark Brown 05 Jun '13

05 Jun '13

Otherwise the functions will be defined but unreferenced when runtime PM is disabled, generating warnings. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/max98090.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 1bf96bd..ad5313f 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2350,6 +2350,7 @@ static int max98090_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_PM_RUNTIME static int max98090_runtime_resume(struct device *dev) { struct max98090_priv *max98090 = dev_get_drvdata(dev); @@ -2369,6 +2370,7 @@ static int max98090_runtime_suspend(struct device *dev) return 0; } +#endif static const struct dev_pm_ops max98090_pm = { SET_RUNTIME_PM_OPS(max98090_runtime_suspend, -- 1.7.10.4

1 0

[alsa-devel] [PATCH (alsa-lib)] chmap: Always succeed setting the map to what it already is
by David Henningsson 05 Jun '13

05 Jun '13

If we try to set the channel map to what it already is, this should always succeed. E g, speaker-test can do this sometimes. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- src/pcm/pcm.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 7ec1f0e..ca4d416 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -7361,6 +7361,13 @@ OBSOLETE1(snd_pcm_sw_params_get_silence_size, ALSA_0.9, ALSA_0.9.0rc4); #endif /* DOC_HIDDEN */ +static int chmap_equal(const snd_pcm_chmap_t *a, const snd_pcm_chmap_t *b) +{ + if (a->channels != b->channels) + return 0; + return !memcmp(a->pos, b->pos, a->channels * sizeof(a->pos[0])); +} + /** * \!brief Query the available channel maps * \param pcm PCM handle to query @@ -7415,6 +7422,10 @@ snd_pcm_chmap_t *snd_pcm_get_chmap(snd_pcm_t *pcm) */ int snd_pcm_set_chmap(snd_pcm_t *pcm, const snd_pcm_chmap_t *map) { + const snd_pcm_chmap_t *oldmap = snd_pcm_get_chmap(pcm); + if (oldmap && chmap_equal(oldmap, map)) + return 0; + if (!pcm->ops->set_chmap) return -ENXIO; return pcm->ops->set_chmap(pcm, map); -- 1.7.9.5

2 1

Re: [alsa-devel] CONFIG_SND_DEBUG
by David Henningsson 05 Jun '13

05 Jun '13

On 06/05/2013 12:53 PM, Andy Whitcroft wrote: > On Wed, Jun 05, 2013 at 09:47:30AM +0200, David Henningsson wrote: >>>>> On Wed, 2013-06-05 at 08:54 +0200, Takashi Iwai wrote: >>>>>> At Tue, 04 Jun 2013 23:52:01 -0700, Joe Perches wrote: >>>>>>> On Wed, 2013-06-05 at 08:32 +0200, Takashi Iwai wrote: >>>>>>>> Yes. These are snd_printd() just to be conditionally built in. >>>>>>>> But in most cases it's rather useful to print them (as most distros >>>>>>>> set CONFIG_SND_DEBUG=y). >>>>>>> Ubuntu doesn't, I believe Fedora doesn't. >>>>>> Then they should have done so :) >>>>> But they don't, so what distros do? >>>> >>>> RedHat (including Fedora) and SUSE do at least. >>> >>> Mandriva does too. (still looking around for others) >>> >>> We can ask Ubuntu to enable CONFIG_SND_DEBUG. >>> (cc'd Andy Whitcroft, Leann Ogasawara and David Henningsson) >>> >>> Maybe there are others Canonical folk that >>> should be cc'd? >> >> Adding kernel team mailing list to CC. >> >> Andy/Leann - apparently CONFIG_SND_DEBUG is on by default from >> upstream, and we explicitly disable it. Is there any reason why we >> do that? > > config SND_DEBUG > bool "Debug" > help > Say Y here to enable ALSA debug code. > > It is off by default in upstream, and the really helpful description > would cirtainly tend to lead to it being disabled. But if it is helpful > to your debugging efforts David then I suspect we can enable it in Saucy > and see what happens. Okay, so then the ball is back in Takashi's area - if we're recommended to turn CONFIG_SND_DEBUG on, why is it off by default in the upstream Linux kernel? -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

3 4

[alsa-devel] [PATCH 1/2] ASoC: arizona: Correct AEC loopback enable
by Charles Keepax 05 Jun '13

05 Jun '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5102.c | 3 ++- sound/soc/codecs/wm5110.c | 3 ++- 2 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index e895d39..100fdad 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1120,7 +1120,8 @@ SND_SOC_DAPM_AIF_IN("AIF3RX2", NULL, 0, ARIZONA_DSP_WIDGETS(DSP1, "DSP1"), SND_SOC_DAPM_VALUE_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1, - ARIZONA_AEC_LOOPBACK_ENA, 0, &wm5102_aec_loopback_mux), + ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, + &wm5102_aec_loopback_mux), SND_SOC_DAPM_PGA_E("OUT1L", SND_SOC_NOPM, ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev, diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index ba38f06..88ad7db 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -503,7 +503,8 @@ SND_SOC_DAPM_PGA("ASRC2R", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2R_ENA_SHIFT, 0, NULL, 0), SND_SOC_DAPM_VALUE_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1, - ARIZONA_AEC_LOOPBACK_ENA, 0, &wm5110_aec_loopback_mux), + ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, + &wm5110_aec_loopback_mux), SND_SOC_DAPM_AIF_OUT("AIF1TX1", NULL, 0, ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX1_ENA_SHIFT, 0), -- 1.7.2.5

2 3

[alsa-devel] [PATCH 0/2] snd-firewire-lib: add MIDI stream support
by o-takashi＠sakamocchi.jp 05 Jun '13

05 Jun '13

From: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> This series of patch add MIDI stream support to snd-firewire-lib according to IEC 61883-6 and MMA/AMEI RP-027. This series of patch is based on my previous patches for snd-firewire-lib. [PATCH v2 0/3] snd-firewire-lib: add handling CMP output connection http://mailman.alsa-project.org/pipermail/alsa-devel/2013-April/061607.html [PATCH v2 0/4] snd-firewire-lib: add handling AMDTP receive stream http://mailman.alsa-project.org/pipermail/alsa-devel/2013-April/061611.html Takashi Sakamoto (2): Add amdtp_stream_running() and amdtp_stream_pcm_running() Add MIDI stream support with data rate restriction sound/firewire/amdtp.c | 171 +++++++++++++++++++++++++++++++++++++++++++----- sound/firewire/amdtp.h | 37 +++++++++++ 2 files changed, 193 insertions(+), 15 deletions(-) -- 1.7.10.4

3 5

Re: [alsa-devel] [PATCH] ASoC: add RT5640 CODEC driver
by Stephen Warren 05 Jun '13

05 Jun '13

On 05/31/2013 01:04 AM, bardliao(a)realtek.com wrote: > From: Bard Liao <bardliao(a)realtek.com> > > * Use regmap_range_cfg to replace index read/write function. > * Remove I2S3 related code since there is no I2S3 in ALC5640. > * Remove Voice DSP related code since there is no Voice DSP in ALC5640. > * Remove MICBIAS2 since there is no MICBIAS2 in ALC5640. > * Change DMIC1/2 CLK to DMIC1/2 Power since it is for enable/disable DMIC1/2 > * Modify some texts for consistent coding style. > * Merge sto adc l/r mux since it shares the same control bits. > * Other minor changes. > +static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { > + /* Input Lines */ > + SND_SOC_DAPM_INPUT("DMIC1"), > + SND_SOC_DAPM_INPUT("DMIC2"), Can you explain that more? In the spec for this part, I don't see any DMIC1/2 input pins. Instead, I think they're alternative uses for the IN1P/N pins, right? > + SND_SOC_DAPM_MIXER("HPO MIX L", SND_SOC_NOPM, 0, 0, > + rt5640_hpo_mix, ARRAY_SIZE(rt5640_hpo_mix)), > + SND_SOC_DAPM_MIXER("HPO MIX R", SND_SOC_NOPM, 0, 0, > + rt5640_hpo_mix, ARRAY_SIZE(rt5640_hpo_mix)), In the version of the driver I posted, I had replaced this with a single "HPO MIX" DAPM_MIXER, because there is a single bit in HW that controls both the L and R channels. I think you should incorporate that same change. That relies on commit 85762e7 "ASoC: dapm: Implement mixer control sharing". In an earlier review, IIRC, Mark had asked you to implement platform data to configure whether the microphones were single-ended or not. I don't see that in this patch. Finally, I really would prefer if you could implement the support for the LDO1_EN GPIO via platform data. Getting that right requires a bit of knowledge of how set_bias_level() is supposed to work on this CODEC, and I think Realtek have the best information to get that right. I'll certainly send a patch to implement this all by device tree after that. Anyway, I'll go test this patch and see if it works for me on my HW.

2 2

[alsa-devel] [PATCH] ASoC: fsl: add imx-wm8962 machine driver
by Nicolin Chen 05 Jun '13

05 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> --- .../devicetree/bindings/sound/imx-audio-wm8962.txt | 61 +++ sound/soc/fsl/Kconfig | 12 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-wm8962.c | 387 ++++++++++++++++++++ 4 files changed, 462 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..cb188a6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt @@ -0,0 +1,61 @@ +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 + + WM8962 pins: + * HPOUTL + * HPOUTR + * SPKOUTL + * SPKOUTR + * MICBIAS + * IN3R + * DMIC + * DMICDAT + + 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 + +- hp-det-gpios : The gpio port of Headphone detection. +- mic-det-gpios: The gpio port of Micphone detection. + +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>; + 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..7aa42d8 --- /dev/null +++ b/sound/soc/fsl/imx-wm8962.c @@ -0,0 +1,387 @@ +/* + * 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/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 check_hw_params(struct snd_pcm_substream *substream, + snd_pcm_format_t sample_format, unsigned int sample_rate) +{ + struct imx_priv *priv = &card_priv; + struct device *dev = substream->pcm->card->dev; + + substream->runtime->sample_bits = + snd_pcm_format_physical_width(sample_format); + substream->runtime->rate = sample_rate; + substream->runtime->format = sample_format; + + if (!priv->first_stream) { + priv->first_stream = substream; + } else { + priv->second_stream = substream; + + if (priv->first_stream->runtime->rate != + priv->second_stream->runtime->rate) { + dev_err(dev, "KEEP THE SAME SAMPLE RATE: %d!\n", + priv->first_stream->runtime->rate); + return -EINVAL; + } + if (priv->first_stream->runtime->sample_bits != + priv->second_stream->runtime->sample_bits) { + snd_pcm_format_t first_format = + priv->first_stream->runtime->format; + + dev_err(dev, "KEEP THE SAME FORMAT: %s!\n", + snd_pcm_format_name(first_format)); + return -EINVAL; + } + } + + return 0; +} + +static int imx_hifi_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 device *dev = substream->pcm->card->dev; + struct imx_priv *priv = &card_priv; + struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + unsigned int sample_rate = params_rate(params); + unsigned int sample_format = params_format(params); + int ret = 0; + unsigned int pll_out; + + /* + * WM8962 doesn't support two substreams in different sample rates or + * sample formats. So check the two parameters of two substreams' when + * there're two substreams of playback and of capture in the same time. + */ + ret = check_hw_params(substream, sample_format, sample_rate); + if (ret < 0) { + dev_err(dev, "Failed to match hw params: %d\n", ret); + return ret; + } + + 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; + } + + return 0; +} + +static int imx_hifi_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct device *dev = substream->pcm->card->dev; + struct imx_priv *priv = &card_priv; + struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + int ret; + + /* We don't need to handle anything if there's no substream running */ + if (!priv->first_stream) + return 0; + + if (priv->first_stream == substream) + priv->first_stream = priv->second_stream; + priv->second_stream = NULL; + + if (!priv->first_stream) { + /* + * SYSCLK of wm8962's not disabled yet. If we wanna close FLL, + * we need to keep SYSCLK alive by changing its source to MCLK. + */ + 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: %d\n", ret); + return ret; + } + + /* Disable FLL and let codec do pm_runtime_put() */ + ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, 0, 0, 0); + if (ret < 0) { + dev_err(dev, "Failed to disable FLL: %d\n", ret); + return ret; + } + } + + return 0; +} + +static struct snd_soc_ops imx_hifi_ops = { + .hw_params = imx_hifi_hw_params, + .hw_free = imx_hifi_hw_free, +}; + +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 = clk_get(&codec_dev->dev, NULL); + if (IS_ERR(data->codec_clk)) { + /* assuming clock enabled by default */ + data->codec_clk = NULL; + ret = of_property_read_u32(codec_np, "clock-frequency", + &data->clk_frequency); + if (ret) { + dev_err(&codec_dev->dev, + "clock-frequency missing or invalid\n"); + goto fail; + } + } else { + data->clk_frequency = clk_get_rate(data->codec_clk); + clk_prepare_enable(data->codec_clk); + } + + 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, + + 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 (data->codec_clk) { + clk_disable_unprepare(data->codec_clk); + clk_put(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 (data->codec_clk) { + clk_disable_unprepare(data->codec_clk); + clk_put(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

1 0

Re: [alsa-devel] USB audio Gadget - code 10 error
by anish singh 05 Jun '13

05 Jun '13

adding alsa-devel On Tue, Jun 4, 2013 at 10:35 PM, Eamon Hayden <eamon.hayden(a)cloudiumsystems.com> wrote: > I’m trying to use the linux usb audio gadget but I get the error : “This > device cannot start (code 10) when I connect to a windows 7/8 box. I don’t > have this problem when I connect to ubunto 12.0.4 > > > > I’m current using kernel 3.6.9, but have also had the issue with an older > kernel (2.6.39.4). > > > > I’m not sure if this is an appropriate issue for this mail list, but I need > to get a resolution to this... > > > > Thanks for your help. > > Eamon. > > > _______________________________________________ > Kernelnewbies mailing list > Kernelnewbies(a)kernelnewbies.org > http://lists.kernelnewbies.org/mailman/listinfo/kernelnewbies >

2 1

[alsa-devel] [PATCH] ASoC: tegra: add runtime PM to resume functions
by Stephen Warren 04 Jun '13

04 Jun '13

From: Stephen Warren <swarren(a)nvidia.com> Tegra HW needs clocks etc. active when touching registers. Make sure they are active during resume, by calling pm_runtime_get_sync() before touching HW. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/tegra/tegra30_ahub.c | 4 ++++ sound/soc/tegra/tegra30_i2s.c | 9 ++++++++- 2 files changed, 12 insertions(+), 1 deletion(-) diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c index 0f4787c..d554d46 100644 --- a/sound/soc/tegra/tegra30_ahub.c +++ b/sound/soc/tegra/tegra30_ahub.c @@ -640,8 +640,12 @@ static int tegra30_ahub_resume(struct device *dev) { int ret; + ret = pm_runtime_get_sync(dev); + if (ret < 0) + return ret; ret = regcache_sync(ahub->regmap_ahub); ret |= regcache_sync(ahub->regmap_apbif); + pm_runtime_put(dev); return ret; } diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index bd0ebc0..d04146c 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -527,8 +527,15 @@ static int tegra30_i2s_suspend(struct device *dev) static int tegra30_i2s_resume(struct device *dev) { struct tegra30_i2s *i2s = dev_get_drvdata(dev); + int ret; - return regcache_sync(i2s->regmap); + ret = pm_runtime_get_sync(dev); + if (ret < 0) + return ret; + ret = regcache_sync(i2s->regmap); + pm_runtime_put(dev); + + return ret; } #endif -- 1.8.1.5

2 1

[alsa-devel] [PATCH V2] ASoC: sgtl5000: Fix VAG_POWER enabling/disabling order
by Marek Vasut 04 Jun '13

04 Jun '13

The VAG_POWER must be enabled after all other bits in CHIP_ANA_POWER and disabled before any other bit in CHIP_ANA_POWER. See the SGTL5000 datasheet (Table 31, BIT 7, page 42-43). Failing to follow this order will result in ugly loud "POP" noise at the end of playback. To achieve such order, use the _PRE and _POST DAPM widgets to trigger the power_vag_event, where the event type check has to be fixed accordingly as well. Signed-off-by: Marek Vasut <marex(a)denx.de> Cc: Dong Aisheng <dong.aisheng(a)linaro.org> Cc: Eric Nelson <eric.nelson(a)boundarydevices.com> Cc: Fabio Estevam <fabio.estevam(a)freescale.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/sgtl5000.c | 14 +++++--------- 1 file changed, 5 insertions(+), 9 deletions(-) V2: Remove that mdelay/msleep regression diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 92bbfec..1994f01 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -151,12 +151,12 @@ static int power_vag_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { switch (event) { - case SND_SOC_DAPM_PRE_PMU: + case SND_SOC_DAPM_POST_PMU: snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP); break; - case SND_SOC_DAPM_POST_PMD: + case SND_SOC_DAPM_PRE_PMD: snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_VAG_POWERUP, 0); msleep(400); @@ -217,12 +217,11 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = { 0, SGTL5000_CHIP_DIG_POWER, 1, 0), - SND_SOC_DAPM_SUPPLY("VAG_POWER", SGTL5000_CHIP_ANA_POWER, 7, 0, - power_vag_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_ADC("ADC", "Capture", SGTL5000_CHIP_ANA_POWER, 1, 0), SND_SOC_DAPM_DAC("DAC", "Playback", SGTL5000_CHIP_ANA_POWER, 3, 0), + + SND_SOC_DAPM_PRE("VAG_POWER_PRE", power_vag_event), + SND_SOC_DAPM_POST("VAG_POWER_POST", power_vag_event), }; /* routes for sgtl5000 */ @@ -230,16 +229,13 @@ static const struct snd_soc_dapm_route sgtl5000_dapm_routes[] = { {"Capture Mux", "LINE_IN", "LINE_IN"}, /* line_in --> adc_mux */ {"Capture Mux", "MIC_IN", "MIC_IN"}, /* mic_in --> adc_mux */ - {"ADC", NULL, "VAG_POWER"}, {"ADC", NULL, "Capture Mux"}, /* adc_mux --> adc */ {"AIFOUT", NULL, "ADC"}, /* adc --> i2s_out */ - {"DAC", NULL, "VAG_POWER"}, {"DAC", NULL, "AIFIN"}, /* i2s-->dac,skip audio mux */ {"Headphone Mux", "DAC", "DAC"}, /* dac --> hp_mux */ {"LO", NULL, "DAC"}, /* dac --> line_out */ - {"LINE_IN", NULL, "VAG_POWER"}, {"Headphone Mux", "LINE_IN", "LINE_IN"},/* line_in --> hp_mux */ {"HP", NULL, "Headphone Mux"}, /* hp_mux --> hp */ -- 1.7.10.4

2 1

[alsa-devel] [PATCH] ASoC: omap-hdmi-codec: make the driver common for other SoCs
by Barry Song 04 Jun '13

04 Jun '13

to support HDMI on CSR SiRFprimaII and atlasVI, we need one more HDMI pseudo codec, rather than add a new driver, we can make omap HDMI codec common for other SoCs as well. then the omap-hdmi codec becomes a generic HDMI pseudo- codec as HDMI audio features depend on HDMI specification not on SoCs. Cc: Ricardo Neri <ricardo.neri(a)ti.com> Signed-off-by: Barry Song <Baohua.Song(a)csr.com> --- sound/soc/codecs/Kconfig | 4 ++-- sound/soc/codecs/Makefile | 4 ++-- sound/soc/codecs/{omap-hdmi.c => hdmi.c} | 26 +++++++++++++------------- sound/soc/omap/omap-hdmi-card.c | 2 +- 4 files changed, 18 insertions(+), 18 deletions(-) rename sound/soc/codecs/{omap-hdmi.c => hdmi.c} (69%) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2f45f00..d8c4f3d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -53,7 +53,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX9877 if I2C select SND_SOC_MC13783 if MFD_MC13XXX select SND_SOC_ML26124 if I2C - select SND_SOC_OMAP_HDMI_CODEC if OMAP4_DSS_HDMI + select SND_SOC_HDMI_CODEC select SND_SOC_PCM3008 select SND_SOC_RT5631 if I2C select SND_SOC_SGTL5000 if I2C @@ -287,7 +287,7 @@ config SND_SOC_MAX98095 config SND_SOC_MAX9850 tristate -config SND_SOC_OMAP_HDMI_CODEC +config SND_SOC_HDMI_CODEC tristate config SND_SOC_PCM3008 diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index b9e41c9..49ff127 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -41,7 +41,7 @@ snd-soc-max98095-objs := max98095.o snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o -snd-soc-omap-hdmi-codec-objs := omap-hdmi.o +snd-soc-hdmi-codec-objs := hdmi.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-rt5631-objs := rt5631.o snd-soc-sgtl5000-objs := sgtl5000.o @@ -168,7 +168,7 @@ obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o obj-$(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-$(CONFIG_SND_SOC_OMAP_HDMI_CODEC) += snd-soc-omap-hdmi-codec.o +obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o diff --git a/sound/soc/codecs/omap-hdmi.c b/sound/soc/codecs/hdmi.c similarity index 69% rename from sound/soc/codecs/omap-hdmi.c rename to sound/soc/codecs/hdmi.c index 529d064..2bcae2b 100644 --- a/sound/soc/codecs/omap-hdmi.c +++ b/sound/soc/codecs/hdmi.c @@ -1,5 +1,5 @@ /* - * ALSA SoC codec driver for HDMI audio on OMAP processors. + * ALSA SoC codec driver for HDMI audio codecs. * Copyright (C) 2012 Texas Instruments Incorporated - http://www.ti.com/ * Author: Ricardo Neri <ricardo.neri(a)ti.com> * @@ -23,10 +23,10 @@ #define DRV_NAME "hdmi-audio-codec" -static struct snd_soc_codec_driver omap_hdmi_codec; +static struct snd_soc_codec_driver hdmi_codec; -static struct snd_soc_dai_driver omap_hdmi_codec_dai = { - .name = "omap-hdmi-hifi", +static struct snd_soc_dai_driver hdmi_codec_dai = { + .name = "hdmi-hifi", .playback = { .channels_min = 2, .channels_max = 8, @@ -39,31 +39,31 @@ static struct snd_soc_dai_driver omap_hdmi_codec_dai = { }, }; -static int omap_hdmi_codec_probe(struct platform_device *pdev) +static int hdmi_codec_probe(struct platform_device *pdev) { - return snd_soc_register_codec(&pdev->dev, &omap_hdmi_codec, - &omap_hdmi_codec_dai, 1); + return snd_soc_register_codec(&pdev->dev, &hdmi_codec, + &hdmi_codec_dai, 1); } -static int omap_hdmi_codec_remove(struct platform_device *pdev) +static int hdmi_codec_remove(struct platform_device *pdev) { snd_soc_unregister_codec(&pdev->dev); return 0; } -static struct platform_driver omap_hdmi_codec_driver = { +static struct platform_driver hdmi_codec_driver = { .driver = { .name = DRV_NAME, .owner = THIS_MODULE, }, - .probe = omap_hdmi_codec_probe, - .remove = omap_hdmi_codec_remove, + .probe = hdmi_codec_probe, + .remove = hdmi_codec_remove, }; -module_platform_driver(omap_hdmi_codec_driver); +module_platform_driver(hdmi_codec_driver); MODULE_AUTHOR("Ricardo Neri <ricardo.neri(a)ti.com>"); -MODULE_DESCRIPTION("ASoC OMAP HDMI codec driver"); +MODULE_DESCRIPTION("ASoC generic HDMI codec driver"); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/omap/omap-hdmi-card.c b/sound/soc/omap/omap-hdmi-card.c index d4eaa92..7e66e9c 100644 --- a/sound/soc/omap/omap-hdmi-card.c +++ b/sound/soc/omap/omap-hdmi-card.c @@ -35,7 +35,7 @@ static struct snd_soc_dai_link omap_hdmi_dai = { .cpu_dai_name = "omap-hdmi-audio-dai", .platform_name = "omap-pcm-audio", .codec_name = "hdmi-audio-codec", - .codec_dai_name = "omap-hdmi-hifi", + .codec_dai_name = "hdmi-hifi", }; static struct snd_soc_card snd_soc_omap_hdmi = { -- 1.8.2.3

2 1

[alsa-devel] [RFC PATCH] ALSA: hda - Ignore default pin config on Haswell HDMI codecs
by David Henningsson 04 Jun '13

04 Jun '13

According to Daniel Vetter, these pin configs should be ignored, because the graphics driver/userspace could route audio to these pins even if they are disabled by BIOS. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_hdmi.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) It would be good to have this acked by someone at Intel before committing. diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index e12f7a0..8c1bede 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1307,12 +1307,14 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) if (!(caps & (AC_PINCAP_HDMI | AC_PINCAP_DP))) return 0; - config = snd_hda_codec_get_pincfg(codec, pin_nid); - if (get_defcfg_connect(config) == AC_JACK_PORT_NONE) - return 0; - if (codec->vendor_id == 0x80862807) intel_haswell_fixup_connect_list(codec, pin_nid); + else { + /* On Haswell, default pin config should be ignored */ + config = snd_hda_codec_get_pincfg(codec, pin_nid); + if (get_defcfg_connect(config) == AC_JACK_PORT_NONE) + return 0; + } pin_idx = spec->num_pins; per_pin = snd_array_new(&spec->pins); -- 1.7.9.5

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[alsa-devel] dsnoop memory leak
by Иван Больсунов 04 Jun '13

04 Jun '13

Since alsa bug tracker does not work for me for weeks, I have to report bug here. Sorry if it a wrong place. So, to reproduce memory leak (Alsa lib used v.1.0.27), you have to use next asound.conf: pcm_slave.myslave { pcm "hw:0,0" } pcm.mysnoop { type dsnoop ipc_key 1024 slave myslave } Then you simply do: snd_pcm_open(&pcm_handle, "mysnoop", SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK); snd_pcm_close(pcm_handle); snd_config_update_free_global(); Valgrind will show something like this: ==7889== ==7889== HEAP SUMMARY: ==7889== in use at exit: 15,854 bytes in 871 blocks ==7889== total heap usage: 333,212 allocs, 332,341 frees, 11,695,490 bytes allocated ==7889== ==7889== 15,834 (6,264 direct, 9,570 indirect) bytes in 174 blocks are definitely lost in loss record 6 of 6 ==7889== at 0x4027CBF: calloc (vg_replace_malloc.c:566) ==7889== by 0x406B416: _snd_config_make (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x406C69C: snd_config_make (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x406C861: snd_config_make_compound (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x406D00C: _snd_config_copy (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x406F112: snd_config_walk (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x406FE04: snd_config_copy (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x4070370: snd_config_expand (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x407107E: snd_config_search_definition (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x40D2C22: _snd_pcm_direct_get_slave_ipc_offset (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x40D68A2: snd1_pcm_direct_parse_open_conf (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x40D28CA: _snd_pcm_dsnoop_open (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x40909C4: snd_pcm_open_conf (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x4091036: snd_pcm_open_noupdate (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x804873D: main (in /home/blah/al/a.out) ==7889== ==7889== LEAK SUMMARY: ==7889== definitely lost: 6,264 bytes in 174 blocks ==7889== indirectly lost: 9,570 bytes in 696 blocks ==7889== possibly lost: 0 bytes in 0 blocks ==7889== still reachable: 20 bytes in 1 blocks ==7889== suppressed: 0 bytes in 0 blocks ==7889== Reachable blocks (those to which a pointer was found) are not shown. ==7889== To see them, rerun with: --leak-check=full --show-reachable=yes ==7889== ==7889== For counts of detected and suppressed errors, rerun with: -v ==7889== ERROR SUMMARY: 1 errors from 1 contexts (suppressed: 0 from 0)

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[alsa-devel] dsnoop memory leak
by Ivan Bolsunov 04 Jun '13

04 Jun '13

Hi guys, on may 31 I sent message, can you at least confirm what bug exist? Thanks.

1 0

[alsa-devel] [PATCH V5 0/5] clk: Samsung: audss: Register audio subsytem clocks using common clk framework
by Padmavathi Venna 04 Jun '13

04 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 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 (5): 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 .../devicetree/bindings/clock/clk-exynos-audss.txt | 64 ++++++++++ .../devicetree/bindings/sound/samsung-i2s.txt | 40 ++---- 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 ++++++++++++++++++++ include/dt-bindings/clk/exynos-audss-clk.h | 25 ++++ 27 files changed, 279 insertions(+), 54 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 04 Jun '13

04 Jun '13

On Fri, May 31, 2013 at 03:04:59PM +0800, bardliao(a)realtek.com wrote: > From: Bard Liao <bardliao(a)realtek.com> This looks pretty good, a few small things below and probably one actual bug. > > * Use regmap_range_cfg to replace index read/write function. > * Remove I2S3 related code since there is no I2S3 in ALC5640. > * Remove Voice DSP related code since there is no Voice DSP in ALC5640. > * Remove MICBIAS2 since there is no MICBIAS2 in ALC5640. > * Change DMIC1/2 CLK to DMIC1/2 Power since it is for enable/disable DMIC1/2 > * Modify some texts for consistent coding style. > * Merge sto adc l/r mux since it shares the same control bits. > * Other minor changes. Put stuff like this after the --- so it doesn't end up in the commit log. > + switch (event) { > + case SND_SOC_DAPM_POST_PMU: > + snd_soc_update_bits(codec, RT5640_PWR_DIG1, 0x0001, 0x0001); > + regmap_update_bits(rt5640->regmap, 0x1c, 0xf000, 0xf000); > + break; It's a bit weird to be using snd_soc_update_bits() and regmap_update_bits()... > + SND_SOC_DAPM_SUPPLY("DMIC1 Power", SND_SOC_NOPM, 0, 0, > + rt5640_set_dmic1_event, > + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), > + SND_SOC_DAPM_SUPPLY("DMIC2 Power", SND_SOC_NOPM, 0, 0, > + rt5640_set_dmic2_event, > + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), The enable bits could be listed here - this would be a bit more idiomatic, would save the PMD event and would enable the writes to be combined. > + case SND_SOC_BIAS_STANDBY: > + if (SND_SOC_BIAS_OFF == codec->dapm.bias_level) { > + snd_soc_update_bits(codec, RT5640_PWR_ANLG1, > + RT5640_PWR_VREF1 | RT5640_PWR_MB | > + RT5640_PWR_BG | RT5640_PWR_VREF2, > + RT5640_PWR_VREF1 | RT5640_PWR_MB | > + RT5640_PWR_BG | RT5640_PWR_VREF2); > + mdelay(10); > + snd_soc_update_bits(codec, RT5640_PWR_ANLG1, > + RT5640_PWR_FV1 | RT5640_PWR_FV2, > + RT5640_PWR_FV1 | RT5640_PWR_FV2); > + regcache_cache_only(rt5640->regmap, false); > + regcache_sync(rt5640->regmap); This looks wrong - you're writing to the device before you take it out of cache only mode so your delay in the above sequence will have no effect. > + val = snd_soc_read(codec, RT5640_VENDOR_ID2); > + if ((val != RT5640_DEVICE_ID)) { > + dev_err(codec->dev, > + "Device with ID register %x is not rt5640/39\n", val); > + return -ENODEV; > + } > + > + rt5640_reset(codec); Do these in the I2C probe function before you register the CODEC, that way we don't try to start the sound card if we can't talk to the device. > + ret = regmap_register_patch(rt5640->regmap, init_list, > + ARRAY_SIZE(init_list)); > + if (ret != 0) > + dev_warn(codec->dev, "Failed to apply regmap patch: %d\n", > + ret); This should also be in the I2C probe.

2 2

[alsa-devel] regression: from 3.8 to 3.9: headphones output no sound on Intel HDA, codec VIA VT1802
by Alex Riesen 04 Jun '13

04 Jun '13

Hi, I just noticed (use the headphones rarely) that the headphones on this System76 Lemur Ultra (lemu4) stopped working. There is absolutely no output. I tried reverting all changes to patch_via.c since v3.8, and have the sound back, but there wasn't much to revert: Revert "ALSA: hda - Rearrange INPUT_PIN_ATTR_*" Revert "ALSA: hda - Use generic parser for VIA codec driver" Revert "ALSA: hda - Enable parsing the independent HP mode as default for VIA codecs" Revert "ALSA: hda/via - Fix wrong checks of power state bits" Revert "ALSA: hda - Fix phantom jacks on VT1708" Revert "ALSA: hda - detect jacks on VT1708 even when no streams are active" Revert "ALSA: hda - Use generic array for loopback list management" and one change (which I indeed suspect) is large: the use of generic parser. The problem is also present in 3.10-rc1. The output of alsa-info.sh attached, for 3.8.12 and 3.9.2 respectively. Regards, Alex Riesen

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Re: [alsa-devel] [PATCH] ASoC: add RT5640 CODEC driver
by Stephen Warren 04 Jun '13

04 Jun '13

On 05/31/2013 01:15 AM, Bard Liao wrote: > Stephen, > > I remove the DT support from this patch. > Can you add the DT support as a separate patch? Yes, I'll do that. I assume you also want me to add the code to control the LDO1_EN GPIO, since that seems missing from this patch. I thought you said you were going to add it. I assume you tested this on Dalmore by manually setting the GPIO from U-Boot before booting the kernel? I'll try to get to this ASAP; perhaps not today though - we'll see how long the other things I have stacked up take.

2 1

Re: [alsa-devel] [PATCH V4 4/4] ARM: dts: add clock provider information for i2s controllers in Exynos5250
by Padma Venkat 04 Jun '13

04 Jun '13

Hi Doug, On Tue, Jun 4, 2013 at 1:48 AM, Doug Anderson <dianders(a)chromium.org> wrote: > Padmavathi, > > On Sun, Jun 2, 2013 at 10:19 PM, Padmavathi Venna <padma.v(a)samsung.com> wrote: >> + 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"; > > Are there bindings for these clocks? Would be nice to see a > description for what they are supposed to be. Yes. They have bindings. I will add the description. > > I looked up i2s_opclk0 / i2s_opclk1 and they look reasonable at a > quick glance. ...and it does seem right that iis and i2s_opclk0 are > the same. ...but I don't see any place that uses the last two. Is > there an in-flight patch? I don't have any patch for these right now. These are used internally. But as these are i2s0 base clocks I added here. > > -Doug Thanks Padma

1 0

Re: [alsa-devel] [PATCH V4 3/4] ARM: dts: add Exynos audio subsystem clock controller node
by Padma Venkat 04 Jun '13

04 Jun '13

Hi Doug, On Tue, Jun 4, 2013 at 1:43 AM, Doug Anderson <dianders(a)chromium.org> wrote: > Padmavathi, > > On Sun, Jun 2, 2013 at 10:19 PM, Padmavathi Venna <padma.v(a)samsung.com> wrote: >> Audio subsystem introduced in s5pv210 and exynos platforms >> which has a internal clock controller. This patch adds a node >> for the same on exynos5250. >> >> Signed-off-by: Padmavathi Venna <padma.v(a)samsung.com> >> Reviewed-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> >> --- >> arch/arm/boot/dts/exynos5250.dtsi | 6 ++++++ >> 1 files changed, 6 insertions(+), 0 deletions(-) >> >> diff --git a/arch/arm/boot/dts/exynos5250.dtsi b/arch/arm/boot/dts/exynos5250.dtsi >> index bccda67..388983e 100644 >> --- a/arch/arm/boot/dts/exynos5250.dtsi >> +++ b/arch/arm/boot/dts/exynos5250.dtsi >> @@ -72,6 +72,12 @@ >> #clock-cells = <1>; >> }; >> >> + clock_audss: audss-clock-controller@3810000 { I removed this leading 0 as Tomasz Figa suggested. > > Nit: other places in the same file have the leading 0, like > i2s0: i2s@03830000 { This was the patch which got merged earlier. So I didn't modify this. Is it okey if I remove leading 0 for both of the nodes now? > > So you could follow suit and do: > > clock_audss: audss-clock-controller@03810000 { > > -Doug Thanks Padma

1 0

[alsa-devel] cplay error "Can't find sync world"
by Keyon.Jey 04 Jun '13

04 Jun '13

hi, all, when I try use clpay -c 1 -d 2 test.mp3 to use tinycompress hwdec on my phone, "Can't find sync world" always occurred, no matter what mp3 music I try to play. vinod/guys, can you help? regards, Keyon

3 3

[alsa-devel] [PATCH] ASoC: dapm: check suspend state in dapm_dai_check_power()
by Stephen Warren 03 Jun '13

03 Jun '13

From: Stephen Warren <swarren(a)nvidia.com> Currently, when the system enters suspend with audio playing, since the DAI widgets are active, the associated CODEC is considered active, and hence is not transition into a low-power bias state as the system suspends. This likely causes audio failure after resume, since the CODEC power has quite possibly been removed during suspend, and the bias level transitions in the CODEC driver are needed to recover from this. At the very least, this issue likely causes warnings in the kernel log due to the CODEC being left on over the suspend transition. To solve this, modify dapm_dai_check_power() not to consider DAI widgets active if the system is entering suspend. Suggested-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 3507346..bad545a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1133,7 +1133,7 @@ static int dapm_dai_check_power(struct snd_soc_dapm_widget *w) DAPM_UPDATE_STAT(w, power_checks); if (w->active) - return w->active; + return snd_soc_dapm_suspend_check(w); return dapm_generic_check_power(w); } -- 1.7.10.4

3 4

[alsa-devel] [PATCH 1/2] ASoC: tegra: implement suspend/resume for Tegra30 AHUB
by Stephen Warren 03 Jun '13

03 Jun '13

From: Stephen Warren <swarren(a)nvidia.com> Add tegra30_ahub_{suspend,resume}. These use regcache functions to restore all HW registers after power loss during a suspend/resume cycle. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/tegra/tegra30_ahub.c | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c index 23e592f..0f4787c 100644 --- a/sound/soc/tegra/tegra30_ahub.c +++ b/sound/soc/tegra/tegra30_ahub.c @@ -627,9 +627,30 @@ static int tegra30_ahub_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_PM_SLEEP +static int tegra30_ahub_suspend(struct device *dev) +{ + regcache_mark_dirty(ahub->regmap_ahub); + regcache_mark_dirty(ahub->regmap_apbif); + + return 0; +} + +static int tegra30_ahub_resume(struct device *dev) +{ + int ret; + + ret = regcache_sync(ahub->regmap_ahub); + ret |= regcache_sync(ahub->regmap_apbif); + + return ret; +} +#endif + static const struct dev_pm_ops tegra30_ahub_pm_ops = { SET_RUNTIME_PM_OPS(tegra30_ahub_runtime_suspend, tegra30_ahub_runtime_resume, NULL) + SET_SYSTEM_SLEEP_PM_OPS(tegra30_ahub_suspend, tegra30_ahub_resume) }; static struct platform_driver tegra30_ahub_driver = { -- 1.7.10.4

2 2

[alsa-devel] [PATCH 1/4] ASoC: wm0010: Clear IRQ as wake source and include missing header
by Charles Keepax 03 Jun '13

03 Jun '13

Both clear the IRQ as being a wake source when we are finished with it and include a missing header file that is required. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm0010.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index b6df319..f5e8356 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -14,6 +14,7 @@ #include <linux/module.h> #include <linux/moduleparam.h> +#include <linux/interrupt.h> #include <linux/irqreturn.h> #include <linux/init.h> #include <linux/spi/spi.h> @@ -1002,6 +1003,8 @@ static int wm0010_spi_remove(struct spi_device *spi) gpio_set_value_cansleep(wm0010->gpio_reset, wm0010->gpio_reset_value); + irq_set_irq_wake(wm0010->irq, 0); + if (wm0010->irq) free_irq(wm0010->irq, wm0010); -- 1.7.2.5

2 2

[alsa-devel] [PATCH 1/1] ASoC: davinci-mcasp: Remove redundant use of of_match_ptr macro
by Sachin Kamat 03 Jun '13

03 Jun '13

'mcasp_dt_ids' is always compiled in. Hence of_match_ptr is not necessary. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/davinci/davinci-mcasp.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 56ecfc7..94f1e46 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1023,7 +1023,7 @@ static struct snd_platform_data *davinci_mcasp_set_pdata_from_of( struct device_node *np = pdev->dev.of_node; struct snd_platform_data *pdata = NULL; const struct of_device_id *match = - of_match_device(of_match_ptr(mcasp_dt_ids), &pdev->dev); + of_match_device(mcasp_dt_ids, &pdev->dev); const u32 *of_serial_dir32; u8 *of_serial_dir; @@ -1256,7 +1256,7 @@ static struct platform_driver davinci_mcasp_driver = { .driver = { .name = "davinci-mcasp", .owner = THIS_MODULE, - .of_match_table = of_match_ptr(mcasp_dt_ids), + .of_match_table = mcasp_dt_ids, }, }; -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: mxs: mxs-saif: Fix the playback/record capabilities
by Fabio Estevam 03 Jun '13

03 Jun '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> Running "aplay -l" and "arecord -l" results in the following outputs: $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: mxssgtl5000 [mxs_sgtl5000], device 0: Playback sgtl5000-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: mxssgtl5000 [mxs_sgtl5000], device 1: Capture sgtl5000-1 [] Subdevices: 1/1 Subdevice #0: subdevice #0 $ arecord -l **** List of CAPTURE Hardware Devices **** card 0: mxssgtl5000 [mxs_sgtl5000], device 0: Playback sgtl5000-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: mxssgtl5000 [mxs_sgtl5000], device 1: Capture sgtl5000-1 [] Subdevices: 1/1 Subdevice #0: subdevice #0 On mx28 there are two serial audio interface ports (SAIF0 and SAIF1) and each one of them are unidirectional. Create a mxs_saif_dai array containing a playback element and a record element only in order to properly describe the dai links. After this change we can correctly report the capabilities as follows: $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: mxssgtl5000 [mxs_sgtl5000], device 0: HiFi Playback sgtl5000-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 $ arecord -l **** List of CAPTURE Hardware Devices **** card 0: mxssgtl5000 [mxs_sgtl5000], device 1: HiFi Capture sgtl5000-1 [] Subdevices: 1/1 Subdevice #0: subdevice #0 Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- This patch also needs: "[PATCH] ASoC: pcm: Require both CODEC and CPU support when declaring stream caps" from Mark Brown. sound/soc/mxs/mxs-saif.c | 38 ++++++++++++++++++++++---------------- 1 file changed, 22 insertions(+), 16 deletions(-) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 49d8700..7f3e67a 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -607,22 +607,28 @@ static int mxs_saif_dai_probe(struct snd_soc_dai *dai) return 0; } -static struct snd_soc_dai_driver mxs_saif_dai = { - .name = "mxs-saif", - .probe = mxs_saif_dai_probe, - .playback = { - .channels_min = 2, - .channels_max = 2, - .rates = MXS_SAIF_RATES, - .formats = MXS_SAIF_FORMATS, +static struct snd_soc_dai_driver mxs_saif_dai[2] = { + { + .name = "saif0", + .probe = mxs_saif_dai_probe, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = MXS_SAIF_RATES, + .formats = MXS_SAIF_FORMATS, + }, + .ops = &mxs_saif_dai_ops, + }, { + .name = "saif1", + .probe = mxs_saif_dai_probe, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = MXS_SAIF_RATES, + .formats = MXS_SAIF_FORMATS, + }, + .ops = &mxs_saif_dai_ops, }, - .capture = { - .channels_min = 2, - .channels_max = 2, - .rates = MXS_SAIF_RATES, - .formats = MXS_SAIF_FORMATS, - }, - .ops = &mxs_saif_dai_ops, }; static const struct snd_soc_component_driver mxs_saif_component = { @@ -735,7 +741,7 @@ static int mxs_saif_probe(struct platform_device *pdev) platform_set_drvdata(pdev, saif); ret = snd_soc_register_component(&pdev->dev, &mxs_saif_component, - &mxs_saif_dai, 1); + &mxs_saif_dai[saif->id], 1); if (ret) { dev_err(&pdev->dev, "register DAI failed\n"); return ret; -- 1.8.1.2

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[alsa-devel] [PATCH] ASoC: pcm: Require both CODEC and CPU support when declaring stream caps
by Mark Brown 03 Jun '13

03 Jun '13

When declaring playback and capture capabilities check for both CODEC side and CPU side support rather than only checking for CODEC side support. While it is unusual some CPUs do have unidirectional DAIs. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-pcm.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 73bb8ee..a9fddf0 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2011,9 +2011,11 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) if (cpu_dai->driver->capture.channels_min) capture = 1; } else { - if (codec_dai->driver->playback.channels_min) + if (codec_dai->driver->playback.channels_min && + cpu_dai->driver->playback.channels_min) playback = 1; - if (codec_dai->driver->capture.channels_min) + if (codec_dai->driver->capture.channels_min && + cpu_dai->driver->capture.channels_min) capture = 1; } -- 1.7.10.4

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[alsa-devel] No headphone control for ALC892 on Zotac Motherboard D2550ITXS-B-E
by Halim Sahin 03 Jun '13

03 Jun '13

Hi, Attached is output of alsa-info. Updated to Alsa-driver 1.0.25 running with 2.6.34 kernel. Frontpanel of my case has mic and headphone output connectors. front mic works but no headphones. Card works except headphones Can you please help? -- Halim Sahin E-Mail: halim.sahin (at) t-online.de

3 5

[alsa-devel] [PATCH] ALSA: usb-audio: fix Roland/Cakewalk UM-3G support
by Clemens Ladisch 03 Jun '13

03 Jun '13

Commit 927c9423dd5f2d1c0b93d5e694ab84b4a5559713 (ALSA: usb-audio: add Edirol UM-3G support) used a wrong quirk type, which would make the driver refuse to attach with the error message "MIDIStreaming interface descriptor not found". Cc: <stable(a)vger.kernel.org> # 3.3 and later Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> --- sound/usb/quirks-table.h | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1792,7 +1792,11 @@ USB_DEVICE_VENDOR_SPEC(0x0582, 0x0108), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .ifnum = 0, - .type = QUIRK_MIDI_STANDARD_INTERFACE + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0007, + .in_cables = 0x0007 + } } }, {

2 1

[alsa-devel] [PATCH] ASoC: dwc: debug message correction.
by Rajeev Kumar 03 Jun '13

03 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; -- 1.7.6.5

1 0

[alsa-devel] [PATCH V2] ARM: dts: wm8994: Add wm8994 regulator support on smdk5250.
by Padmavathi Venna 03 Jun '13

03 Jun '13

This patch adds the required regulator supplies and properties for wm8994 codec on smdk5250 board. Signed-off-by: Padmavathi Venna <padma.v(a)samsung.com> --- Changes since V1: - Clubbed the same supply regulators as suggested by Mark. arch/arm/boot/dts/exynos5250-smdk5250.dts | 37 +++++++++++++++++++++++++++- 1 files changed, 35 insertions(+), 2 deletions(-) diff --git a/arch/arm/boot/dts/exynos5250-smdk5250.dts b/arch/arm/boot/dts/exynos5250-smdk5250.dts index f2a025e..dc2ec25 100644 --- a/arch/arm/boot/dts/exynos5250-smdk5250.dts +++ b/arch/arm/boot/dts/exynos5250-smdk5250.dts @@ -37,6 +37,30 @@ }; }; + vdd:fixed-regulator@0 { + compatible = "regulator-fixed"; + regulator-name = "vdd-supply"; + regulator-min-microvolt = <1800000>; + regulator-max-microvolt = <1800000>; + regulator-always-on; + }; + + dbvdd:fixed-regulator@1 { + compatible = "regulator-fixed"; + regulator-name = "dbvdd-supply"; + regulator-min-microvolt = <3300000>; + regulator-max-microvolt = <3300000>; + regulator-always-on; + }; + + spkvdd:fixed-regulator@2 { + compatible = "regulator-fixed"; + regulator-name = "spkvdd-supply"; + regulator-min-microvolt = <5000000>; + regulator-max-microvolt = <5000000>; + regulator-always-on; + }; + i2c@12C70000 { samsung,i2c-sda-delay = <100>; samsung,i2c-max-bus-freq = <20000>; @@ -47,8 +71,17 @@ }; wm8994: wm8994@1a { - compatible = "wlf,wm8994"; - reg = <0x1a>; + compatible = "wlf,wm8994"; + reg = <0x1a>; + + gpio-controller; + #gpio-cells = <2>; + + AVDD2-supply = <&vdd>; + CPVDD-supply = <&vdd>; + DBVDD-supply = <&dbvdd>; + SPKVDD1-supply = <&spkvdd>; + SPKVDD2-supply = <&spkvdd>; }; }; -- 1.7.4.4

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[alsa-devel] [PATCH V4 0/4] clk: Samsung: audss: Register audio subsytem clocks using common clk framework
by Padmavathi Venna 03 Jun '13

03 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 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 (4): 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 .../devicetree/bindings/clock/clk-exynos-audss.txt | 64 ++++++++++ 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 | 23 +++- 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 ++++++++++++++++++++ include/dt-bindings/clk/exynos-audss-clk.h | 25 ++++ 26 files changed, 269 insertions(+), 27 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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[alsa-devel] [PATCH] ASoC: dwc: Folder path correction in file header.
by Rajeev Kumar 03 Jun '13

03 Jun '13

From: Rajeev Kumar <rajeevk-dlh.kumar(a)st.com> Folder path correction in file header. Signed-off-by: Rajeev Kumar <rajeevk-dlh.kumar(a)st.com> --- sound/soc/dwc/designware_i2s.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index deb30d5..a11050c 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -1,7 +1,7 @@ /* * ALSA SoC Synopsys I2S Audio Layer * - * sound/soc/spear/designware_i2s.c + * sound/soc/dwc/designware_i2s.c * * Copyright (C) 2010 ST Microelectronics * Rajeev Kumar <rajeev-dlh.kumar(a)st.com> -- 1.7.6.5

1 0

[alsa-devel] how to use the invert of the field of struct snd_kcontrl_new ?
by Zhang wei 03 Jun '13

03 Jun '13

Is there some details documents about it? When need I set it as 1 and 0? Thanks BR zw

1 0

[alsa-devel] aplay lists both Playback and Capture devices
by Fabio Estevam 02 Jun '13

02 Jun '13

Hi, When running 'aplay -l' and 'arecord -l' I see the following output on a mx28evk running a 3.10-rc kernel: /$ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: mxssgtl5000 [mxs_sgtl5000], device 0: Playback sgtl5000-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: mxssgtl5000 [mxs_sgtl5000], device 1: Capture sgtl5000-1 [] Subdevices: 1/1 Subdevice #0: subdevice #0 $ arecord -l **** List of CAPTURE Hardware Devices **** card 0: mxssgtl5000 [mxs_sgtl5000], device 0: Playback sgtl5000-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: mxssgtl5000 [mxs_sgtl5000], device 1: Capture sgtl5000-1 [] Subdevices: 1/1 Subdevice #0: subdevice #0 I would expect to see only the Playback entry in aplay and only the Record entry in arecord. Is there anything I should change in sound/soc/mxs/mxs-sgtl5000.c in order to get the correct outputs? Regards, Fabio Estevam

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[alsa-devel] [PATCH 1/2] dma: Add a PAUSE_RESUME capability
by Lars-Peter Clausen 01 Jun '13

01 Jun '13

Add a capability which allows dmaengine consumers to query whether a dmaengine controller supports the DMA_PAUSE and DMA_RESUME commands. Sometimes a user of the dmaengine API needs to know in advance if a channel will support pause/resume. E.g. in ALSA the information whether it is possible to pause/resume a audio stream or whether it is necessary to abort and restart the stream is communicated to userspace so it can act accordingly. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/linux/dmaengine.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/linux/dmaengine.h b/include/linux/dmaengine.h index 96d3e4a..ad98dfb 100644 --- a/include/linux/dmaengine.h +++ b/include/linux/dmaengine.h @@ -74,6 +74,7 @@ enum dma_transaction_type { DMA_SLAVE, DMA_CYCLIC, DMA_INTERLEAVE, + DMA_PAUSE_RESUME, /* last transaction type for creation of the capabilities mask */ DMA_TX_TYPE_END, }; -- 1.8.0

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[alsa-devel] [PATCH 1/2] ASoC: wm8994: fix the codec state machine
by Vinod Koul 01 Jun '13

01 Jun '13

few places the mic_detecting is not updated correctly when state changes, causing jack detection to go wrong Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/codecs/wm8994.c | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 49b2b00..43f82d2 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3547,6 +3547,7 @@ static void wm8958_mic_id(void *data, u16 status) snd_soc_jack_report(wm8994->micdet[0].jack, 0, wm8994->btn_mask | SND_JACK_HEADSET); + wm8994->mic_detecting = true; return; } @@ -3672,7 +3673,7 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) snd_soc_update_bits(codec, WM1811_JACKDET_CTRL, WM1811_JACKDET_DB, WM1811_JACKDET_DB); - wm8994->mic_detecting = false; + wm8994->mic_detecting = true; wm8994->jack_mic = false; snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, WM8958_MICD_ENA, 0); @@ -3696,6 +3697,7 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) /* Since we only report deltas force an update, ensures we * avoid bootstrapping issues with the core. */ snd_soc_jack_report(wm8994->micdet[0].jack, 0, 0); + wm8994->mic_detecting = true; pm_runtime_put(codec->dev); return IRQ_HANDLED; -- 1.7.0.4

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Re: [alsa-devel] [PATCH V3 0/4] clk: Samsung: audss: Register audio subsytem clocks using common clk framework
by Padma Venkat 01 Jun '13

01 Jun '13

Hi Sylwester, On Sat, Jun 1, 2013 at 1:59 PM, Sylwester Nawrocki <sylvester.nawrocki(a)gmail.com> wrote: > Padmavathi, > > > On 05/28/2013 12:02 PM, Padmavathi Venna wrote: >> >> 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 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 > > > Overall it looks good to me. I'm only a bit uncomfortable with using > SAMSUNG_ > prefix for the clock index definitions. Ideally it should be something more > specific to the SoC family. But I can't think of anything better at the > moment. > This covers the Exynos and S5P SoCs, hence EXYNOS_ would not be appropriate. Actuvally I added samsung prefix when I added s5pv210 compatible name. But I also feel Exynos would be better now. I will resend the patch with Exynos prefix and with your reviewed-by. Thanks Padma > > The patch series: > Reviewed-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> > > Thanks, > Sylwester > > >> Padmavathi Venna (4): >> 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 >> >> .../bindings/clock/clk-samsung-audss.txt | 64 ++++++++++ >> 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 | 23 +++- >> 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-samsung-audss.c | 133 >> ++++++++++++++++++++ >> include/dt-bindings/clk/samsung-audss-clk.h | 25 ++++ >> 26 files changed, 269 insertions(+), 27 deletions(-) >> create mode 100644 >> Documentation/devicetree/bindings/clock/clk-samsung-audss.txt >> create mode 100644 drivers/clk/samsung/clk-samsung-audss.c >> create mode 100644 include/dt-bindings/clk/samsung-audss-clk.h

1 0

[alsa-devel] [PATCH] ASoC: sgtl5000: Do not read registers prior to turning on the supplies
by Fabio Estevam 31 May '13

31 May '13

Commit b871f1ad (ASoC: sgtl5000: Read SGTL5000_CHIP_ID in i2c_probe()) placed the code for reading the codec revision prior to turning on the power supplies. Even though this works on some systems that always have the codec power supplies enabled, this is not correct, so revert this commit. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/sgtl5000.c | 39 +++++++++++++++------------------------ 1 file changed, 15 insertions(+), 24 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 1c3b20f..788af6c 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1275,7 +1275,7 @@ static int sgtl5000_replace_vddd_with_ldo(struct snd_soc_codec *codec) static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) { - int reg; + u16 reg; int ret; int rev; int i; @@ -1303,17 +1303,23 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) /* wait for all power rails bring up */ udelay(10); - /* - * workaround for revision 0x11 and later, - * roll back to use internal LDO - */ - - ret = regmap_read(sgtl5000->regmap, SGTL5000_CHIP_ID, &reg); - if (ret) + /* read chip information */ + reg = snd_soc_read(codec, SGTL5000_CHIP_ID); + if (((reg & SGTL5000_PARTID_MASK) >> SGTL5000_PARTID_SHIFT) != + SGTL5000_PARTID_PART_ID) { + dev_err(codec->dev, + "Device with ID register %x is not a sgtl5000\n", reg); + ret = -ENODEV; goto err_regulator_disable; + } rev = (reg & SGTL5000_REVID_MASK) >> SGTL5000_REVID_SHIFT; + dev_info(codec->dev, "sgtl5000 revision 0x%x\n", rev); + /* + * workaround for revision 0x11 and later, + * roll back to use internal LDO + */ if (external_vddd && rev >= 0x11) { /* disable all regulator first */ regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), @@ -1497,7 +1503,7 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { struct sgtl5000_priv *sgtl5000; - int ret, reg, rev; + int ret; sgtl5000 = devm_kzalloc(&client->dev, sizeof(struct sgtl5000_priv), GFP_KERNEL); @@ -1511,21 +1517,6 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, return ret; } - /* read chip information */ - ret = regmap_read(sgtl5000->regmap, SGTL5000_CHIP_ID, &reg); - if (ret) - return ret; - - if (((reg & SGTL5000_PARTID_MASK) >> SGTL5000_PARTID_SHIFT) != - SGTL5000_PARTID_PART_ID) { - dev_err(&client->dev, - "Device with ID register %x is not a sgtl5000\n", reg); - return -ENODEV; - } - - rev = (reg & SGTL5000_REVID_MASK) >> SGTL5000_REVID_SHIFT; - dev_info(&client->dev, "sgtl5000 revision 0x%x\n", rev); - i2c_set_clientdata(client, sgtl5000); /* Ensure sgtl5000 will start with sane register values */ -- 1.8.1.2

5 8

[alsa-devel] Improving or replacing snd_printk()
by Alan Stern 31 May '13

31 May '13

Takashi: Messages produced by snd_printk() aren't as informative as they could be -- especially when CONFIG_SND_VERBOSE_PRINTK isn't enabled, which is the case on most systems. The routine uses printk() rather then dev_dbg(), dev_info(), or dev_err(). As a result, the messages don't say what driver or device is involved. We just get something like: [15919.883604] cannot submit urb (err = -18) That's not much help to anybody. If you didn't know what an urb was, you wouldn't even realize that this referred to a USB device. And if you had multiple USB audio devices, you wouldn't know which one it referred to. Has there been any thought of improving snd_printk(), or even better, changing the call sites to use the dev_*() routines instead? Alan Stern

4 17

[alsa-devel] Fwd: regression: from 3.8 to 3.9: headphones output no sound on Intel HDA, codec VIA VT1802
by Evan 31 May '13

31 May '13

I am having this same issue (with the same hardware). I found a temporary work around for the problem by disabling "Auto-Mute Mode" with alsamixer and then muting my "Speaker" output. If I don't mute the "Speaker" I get audio through both my headphones and my speakers, but when I mute the speakers I get the audio just through the headphones. The output of alsa-info for 3.9.4 is here[1]. [1] http://www.alsa-project.org/db/?f=6120303f74b8e46930e15a544ba11b859de25c04 -E

2 1

[alsa-devel] [PATCH 1/3] ALSA: hdspm - Add support for 128-192kHz WordClock input
by Adrian Knoth 31 May '13

31 May '13

Allow WordClock input rates of 128, 176.4 and 192kHz. 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 ef3cbc0..e070ea8 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -400,8 +400,8 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}"); #define HDSPM_wc_freq0 (1<<5) /* input freq detected via autosync */ #define HDSPM_wc_freq1 (1<<6) /* 001=32, 010==44.1, 011=48, */ -#define HDSPM_wc_freq2 (1<<7) /* 100=64, 101=88.2, 110=96, */ -/* missing Bit for 111=128, 1000=176.4, 1001=192 */ +#define HDSPM_wc_freq2 (1<<7) /* 100=64, 101=88.2, 110=96, 111=128 */ +#define HDSPM_wc_freq3 0x800 /* 1000=176.4, 1001=192 */ #define HDSPM_SyncRef0 0x10000 /* Sync Reference */ #define HDSPM_SyncRef1 0x20000 @@ -412,13 +412,17 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}"); #define HDSPM_wc_valid (HDSPM_wcLock|HDSPM_wcSync) -#define HDSPM_wcFreqMask (HDSPM_wc_freq0|HDSPM_wc_freq1|HDSPM_wc_freq2) +#define HDSPM_wcFreqMask (HDSPM_wc_freq0|HDSPM_wc_freq1|HDSPM_wc_freq2|\ + HDSPM_wc_freq3) #define HDSPM_wcFreq32 (HDSPM_wc_freq0) #define HDSPM_wcFreq44_1 (HDSPM_wc_freq1) #define HDSPM_wcFreq48 (HDSPM_wc_freq0|HDSPM_wc_freq1) #define HDSPM_wcFreq64 (HDSPM_wc_freq2) #define HDSPM_wcFreq88_2 (HDSPM_wc_freq0|HDSPM_wc_freq2) #define HDSPM_wcFreq96 (HDSPM_wc_freq1|HDSPM_wc_freq2) +#define HDSPM_wcFreq128 (HDSPM_wc_freq0|HDSPM_wc_freq1|HDSPM_wc_freq2) +#define HDSPM_wcFreq176_4 (HDSPM_wc_freq3) +#define HDSPM_wcFreq192 (HDSPM_wc_freq0|HDSPM_wc_freq3) #define HDSPM_status1_F_0 0x0400000 #define HDSPM_status1_F_1 0x0800000 @@ -1181,6 +1185,15 @@ static int hdspm_external_sample_rate(struct hdspm *hdspm) case HDSPM_wcFreq96: rate = 96000; break; + case HDSPM_wcFreq128: + rate = 128000; + break; + case HDSPM_wcFreq176_4: + rate = 176400; + break; + case HDSPM_wcFreq192: + rate = 192000; + break; default: rate = 0; break; -- 1.7.10.4

2 3

[alsa-devel] [PATCH] ALSA: hda - Add headset quirk for two Dell machines
by David Henningsson 31 May '13

31 May '13

This quirk is required for the headset mic to work on these two machines. BugLink: https://bugs.launchpad.net/bugs/1186170 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 59d2e91..aa4bf82 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3493,6 +3493,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x05f4, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), 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, 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), SND_PCI_QUIRK(0x103c, 0x1973, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1), -- 1.7.9.5

2 1

[alsa-devel] [PATCH] [resend] snd-es1968: Add radio support for MediaForte M56VAP
by Ondrej Zary 31 May '13

31 May '13

Add support for TEA5757 tuner on MediaForte M56VAP sound+modem+radio card. The GPIO connection type is automatically detected (like snd-fm801 driver). Also add a safety subsystem vendor check to skip radio detection if vendor differs from ESS (so we don't touch GPIOs on laptops). Tested with SF64-PCE2 and M56VAP cards. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- sound/pci/es1968.c | 73 +++++++++++++++++++++++++++++++++++---------------- 1 files changed, 50 insertions(+), 23 deletions(-) diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index a1f32b5..db08641 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -564,6 +564,7 @@ struct es1968 { #ifdef CONFIG_SND_ES1968_RADIO struct v4l2_device v4l2_dev; struct snd_tea575x tea; + unsigned int tea575x_tuner; #endif }; @@ -2557,37 +2558,47 @@ static int snd_es1968_input_register(struct es1968 *chip) bits 1=unmask write to given bit */ #define IO_DIR 8 /* direction register offset from GPIO_DATA bits 0/1=read/write direction */ -/* mask bits for GPIO lines */ -#define STR_DATA 0x0040 /* GPIO6 */ -#define STR_CLK 0x0080 /* GPIO7 */ -#define STR_WREN 0x0100 /* GPIO8 */ -#define STR_MOST 0x0200 /* GPIO9 */ + +/* GPIO to TEA575x maps */ +struct snd_es1968_tea575x_gpio { + u8 data, clk, wren, most; + char *name; +}; + +static struct snd_es1968_tea575x_gpio snd_es1968_tea575x_gpios[] = { + { .data = 6, .clk = 7, .wren = 8, .most = 9, .name = "SF64-PCE2" }, + { .data = 7, .clk = 8, .wren = 6, .most = 10, .name = "M56VAP" }, +}; + +#define get_tea575x_gpio(chip) \ + (&snd_es1968_tea575x_gpios[(chip)->tea575x_tuner]) + static void snd_es1968_tea575x_set_pins(struct snd_tea575x *tea, u8 pins) { struct es1968 *chip = tea->private_data; - unsigned long io = chip->io_port + GPIO_DATA; + struct snd_es1968_tea575x_gpio gpio = *get_tea575x_gpio(chip); u16 val = 0; - val |= (pins & TEA575X_DATA) ? STR_DATA : 0; - val |= (pins & TEA575X_CLK) ? STR_CLK : 0; - val |= (pins & TEA575X_WREN) ? STR_WREN : 0; + val |= (pins & TEA575X_DATA) ? (1 << gpio.data) : 0; + val |= (pins & TEA575X_CLK) ? (1 << gpio.clk) : 0; + val |= (pins & TEA575X_WREN) ? (1 << gpio.wren) : 0; - outw(val, io); + outw(val, chip->io_port + GPIO_DATA); } static u8 snd_es1968_tea575x_get_pins(struct snd_tea575x *tea) { struct es1968 *chip = tea->private_data; - unsigned long io = chip->io_port + GPIO_DATA; - u16 val = inw(io); - u8 ret; + struct snd_es1968_tea575x_gpio gpio = *get_tea575x_gpio(chip); + u16 val = inw(chip->io_port + GPIO_DATA); + u8 ret = 0; - ret = 0; - if (val & STR_DATA) + if (val & (1 << gpio.data)) ret |= TEA575X_DATA; - if (val & STR_MOST) + if (val & (1 << gpio.most)) ret |= TEA575X_MOST; + return ret; } @@ -2596,13 +2607,18 @@ static void snd_es1968_tea575x_set_direction(struct snd_tea575x *tea, bool outpu struct es1968 *chip = tea->private_data; unsigned long io = chip->io_port + GPIO_DATA; u16 odir = inw(io + IO_DIR); + struct snd_es1968_tea575x_gpio gpio = *get_tea575x_gpio(chip); if (output) { - outw(~(STR_DATA | STR_CLK | STR_WREN), io + IO_MASK); - outw(odir | STR_DATA | STR_CLK | STR_WREN, io + IO_DIR); + outw(~((1 << gpio.data) | (1 << gpio.clk) | (1 << gpio.wren)), + io + IO_MASK); + outw(odir | (1 << gpio.data) | (1 << gpio.clk) | (1 << gpio.wren), + io + IO_DIR); } else { - outw(~(STR_CLK | STR_WREN | STR_DATA | STR_MOST), io + IO_MASK); - outw((odir & ~(STR_DATA | STR_MOST)) | STR_CLK | STR_WREN, io + IO_DIR); + outw(~((1 << gpio.clk) | (1 << gpio.wren) | (1 << gpio.data) | (1 << gpio.most)), + io + IO_MASK); + outw((odir & ~((1 << gpio.data) | (1 << gpio.most))) + | (1 << gpio.clk) | (1 << gpio.wren), io + IO_DIR); } } @@ -2772,6 +2788,9 @@ static int snd_es1968_create(struct snd_card *card, snd_card_set_dev(card, &pci->dev); #ifdef CONFIG_SND_ES1968_RADIO + /* don't play with GPIOs on laptops */ + if (chip->pci->subsystem_vendor != 0x125d) + goto no_radio; err = v4l2_device_register(&pci->dev, &chip->v4l2_dev); if (err < 0) { snd_es1968_free(chip); @@ -2781,10 +2800,18 @@ static int snd_es1968_create(struct snd_card *card, chip->tea.private_data = chip; chip->tea.radio_nr = radio_nr; chip->tea.ops = &snd_es1968_tea_ops; - strlcpy(chip->tea.card, "SF64-PCE2", sizeof(chip->tea.card)); sprintf(chip->tea.bus_info, "PCI:%s", pci_name(pci)); - if (!snd_tea575x_init(&chip->tea, THIS_MODULE)) - printk(KERN_INFO "es1968: detected TEA575x radio\n"); + for (i = 0; i < ARRAY_SIZE(snd_es1968_tea575x_gpios); i++) { + chip->tea575x_tuner = i; + if (!snd_tea575x_init(&chip->tea, THIS_MODULE)) { + snd_printk(KERN_INFO "es1968: detected TEA575x radio type %s\n", + get_tea575x_gpio(chip)->name); + strlcpy(chip->tea.card, get_tea575x_gpio(chip)->name, + sizeof(chip->tea.card)); + break; + } + } +no_radio: #endif *chip_ret = chip; -- Ondrej Zary

2 1

[alsa-devel] [PATCH 1/1] ALSA: hda - add dock support for Thinkpad T431s
by Ebben Aries 31 May '13

31 May '13

Add a model/fixup string "lenovo-dock", for Thinkpad T431s, to allow sound in docking station. Tested on Lenovo T431s with ThinkPad Mini Dock Plus Series 3 Signed-off-by: Ebben Aries <earies(a)dscp.org> --- 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 59d2e91..9658faf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3530,6 +3530,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21fa, "Thinkpad X230", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x21f3, "Thinkpad T430", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK), + SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), -- 1.7.12.1

2 1

[alsa-devel] [PATCH] ALSA: sis7019: fix error return code in sis_chip_create()
by Wei Yongjun 31 May '13

31 May '13

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Fix to return a negative error code in the pci_set_dma_mask() error handling case instead of 0, as done elsewhere in this function. Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/pci/sis7019.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c index f2639a0..e413b4e 100644 --- a/sound/pci/sis7019.c +++ b/sound/pci/sis7019.c @@ -1341,7 +1341,8 @@ static int sis_chip_create(struct snd_card *card, if (rc) goto error_out; - if (pci_set_dma_mask(pci, DMA_BIT_MASK(30)) < 0) { + rc = pci_set_dma_mask(pci, DMA_BIT_MASK(30)); + if (rc < 0) { dev_err(&pci->dev, "architecture does not support 30-bit PCI busmaster DMA"); goto error_out_enabled; }

2 1

Re: [alsa-devel] [PATCH] ALSA: hda - Enable runtime pm for Haswell
by Wang, Xingchao 31 May '13

31 May '13

Hi Rafael, > -----Original Message----- > From: Wysocki, Rafael J > Sent: Wednesday, May 22, 2013 7:09 PM > To: Takashi Iwai > Cc: Wang, Xingchao; Girdwood, Liam R; Lin, Mengdong; Li, Jocelyn; > alsa-devel(a)alsa-project.org; Wang Xingchao > Subject: Re: [PATCH] ALSA: hda - Enable runtime pm for Haswell > > On 5/22/2013 7:51 AM, Takashi Iwai wrote: > > At Wed, 22 May 2013 03:56:00 +0000, > > Wang, Xingchao wrote: > >> Hi, > >> > >> Add Rafael in loop. > >> > >>> -----Original Message----- > >>> From: Takashi Iwai [mailto:tiwai@suse.de] > >>> Sent: Thursday, May 16, 2013 4:49 PM > >>> To: Wang Xingchao > >>> Cc: Girdwood, Liam R; Lin, Mengdong; Li, Jocelyn; > >>> alsa-devel(a)alsa-project.org; Wang, Xingchao > >>> Subject: Re: [PATCH] ALSA: hda - Enable runtime pm for Haswell > >>> > >>> At Thu, 16 May 2013 16:29:05 +0800, > >>> Wang Xingchao wrote: > >>>> Haswell doesnot support runtime pm by default. > >>>> This patch let haswell Display HD-A controller enter runtime > >>>> suspend, and bring more power saving whith power-well. > >>>> > >>>> Signed-off-by: Wang Xingchao <xingchao.wang(a)linux.intel.com> > >>> I don't think it's good to fiddle such a thing in the driver side. > >>> If Haswell can support runtime PM really, it should be set commonly. > >> I wonder whether it's HD-A driver's policy to only support runtime PM if the > device can support signal wakup? > >> According to Rafael, the device can support runtime PM regardless, no > matter it supports PME or not. > >> If so, we should remove the "if" condition check here. > > Well, if the decision is purely a driver issue, then we can get rid of > > PME check. > > Yes, it is. > > > But in that case, it should be simply like: > > > > azx_probe() { > > ... > > if (chip->driver_caps & AZX_DCAPS_PM_RUNTIME) > > pm_runtime_put_noidle(&pci->dev); > > ... > > } > > > > azx_remove() { > > ... > > if (chip->driver_caps & AZX_DCAPS_PM_RUNTIME) > > pm_runtime_get_noresume(&pci->dev); > > ... > > } > > > > > > AFAIU, calling pm_runtime_allow() enables the runtime PM *forcibly*. > > Usually this isn't a good thing. > > > > That's correct. Moreover, calling both pm_runtime_allow() *and* > pm_runtime_put_noidle() together would be a bug IMO. It's interesting something maybe wrong in my test: 1. withtout calling pm_runtime_allow(), azx_runteim_idle/suspend() will not be called. 2. another trick is on my Haswell ULT C stepping board, the runtime PM only work after exit from resume: echo mem > /sys/power/state if you donot let system enter suspend manually, the runtime pm will not be triggered. Is there any dependency between runtime pm suspend and normal suspend? Thanks --xingchao > > Thanks, > Rafael > > > > > >>>> --- > >>>> sound/pci/hda/hda_intel.c | 6 ++++++ > >>>> 1 file changed, 6 insertions(+) > >>>> > >>>> diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c > >>>> index bf27693..eb25888 100644 > >>>> --- a/sound/pci/hda/hda_intel.c > >>>> +++ b/sound/pci/hda/hda_intel.c > >>>> @@ -3755,6 +3755,12 @@ static int azx_probe(struct pci_dev *pci, > >>>> > >>>> if (pci_dev_run_wake(pci)) > >>>> pm_runtime_put_noidle(&pci->dev); > >>>> + else if (chip->driver_caps > >>>> + & AZX_DCAPS_I915_POWERWELL) { > >>>> + /* Haswell doesnot support runtime pm by default */ > >>>> + pm_runtime_put_noidle(&pci->dev); > >>>> + pm_runtime_allow(&pci->dev); > >>>> + } > >>>> > >>>> dev++; > >>>> complete_all(&chip->probe_wait); > >>>> -- > >>>> 1.7.9.5 > >>>>

2 2

[alsa-devel] [PATCH v7 00/11] ASoC: fsl-ssi: ac97-slave support
by Markus Pargmann 31 May '13

31 May '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. Regards, Markus 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: phycore-ac97: Add DT support 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 | 4 +- 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.c | 236 +++++++++-- 13 files changed, 656 insertions(+), 128 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

2 12

[alsa-devel] [PATCH] alsa-utils 1.0.27.1 still ships 90-alsa-restore.rules
by Elimar Riesebieter 30 May '13

30 May '13

Hi all, [0] tells: alsactl: remove 90-alsa-restore.rules from the tar ball but it still ships 90-alsa-restore.rules.in and installs the file as 90-alsa-restore.rules [0] http://www.alsa-project.org/main/index.php/Changes_v1.0.27_v1.0.27.1 Thanks Signed-off-by: Elimar Riesebieter <riesebie(a)lxtec.de> --- alsactl/90-alsa-restore.rules.in | 8 -------- alsactl/Makefile.am | 5 ----- 2 files changed, 13 deletions(-) diff --git a/alsactl/90-alsa-restore.rules.in b/alsactl/90-alsa-restore.rules.in deleted file mode 100644 index 88e12e0..0000000 --- a/alsactl/90-alsa-restore.rules.in +++ /dev/null @@ -1,8 +0,0 @@ -ACTION=="add", SUBSYSTEM=="sound", KERNEL=="controlC*", KERNELS!="card*", GOTO="alsa_restore_go" -GOTO="alsa_restore_end" - -LABEL="alsa_restore_go" -TEST!="@daemonswitch@", RUN+="@sbindir@/alsactl restore $attr{number}" -TEST=="@daemonswitch@", RUN+="@sbindir@/alsactl nrestore $attr{number}" - -LABEL="alsa_restore_end" diff --git a/alsactl/Makefile.am b/alsactl/Makefile.am index bac84eb..3bf8eca 100644 --- a/alsactl/Makefile.am +++ b/alsactl/Makefile.am @@ -11,8 +11,6 @@ alsactl_SOURCES=alsactl.c state.c lock.c utils.c init_parse.c daemon.c alsactl_CFLAGS=$(AM_CFLAGS) -D__USE_GNU -DSYS_ASOUNDRC=\"$(ASOUND_STATE_DIR)/asound.state\" -DSYS_PIDFILE=\"$(ALSACTL_PIDFILE_DIR)/alsactl.pid\" noinst_HEADERS=alsactl.h list.h init_sysdeps.c init_utils_string.c init_utils_run.c init_sysfs.c -udevrules_DATA = \ - 90-alsa-restore.rules if HAVE_SYSTEMD @@ -49,20 +47,17 @@ alsa-restore.service: alsa-restore.service.in alsa-store.service: alsa-store.service.in $(edit) -90-alsa-restore.rules: 90-alsa-restore.rules.in $(edit) EXTRA_DIST += \ alsa-state.service.in \ alsa-restore.service.in \ alsa-store.service.in \ - 90-alsa-restore.rules.in CLEANFILES = \ alsa-state.service \ alsa-restore.service \ alsa-store.service \ - 90-alsa-restore.rules %.7: %.xml xmlto man $?

2 5

[alsa-devel] [PATCH 1/5] ASoC: wm8994: Add digital loopback paths
by Mark Brown 30 May '13

30 May '13

There is loopback control within the audio interfaces, provide control of this as there are some obscure scenarios where this could be used in production. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 31 +++++++++++++++++++++++++++++-- 1 file changed, 29 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 561c2f9..2081fa1 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1499,6 +1499,24 @@ static const char *aif1dac_text[] = { "AIF1DACDAT", "AIF3DACDAT", }; +static const char *loopback_text[] = { + "None", "ADCDAT", +}; + +static const struct soc_enum aif1_loopback_enum = + SOC_ENUM_SINGLE(WM8994_AIF1_CONTROL_2, WM8994_AIF1_LOOPBACK_SHIFT, 2, + loopback_text); + +static const struct snd_kcontrol_new aif1_loopback = + SOC_DAPM_ENUM("AIF1 Loopback", aif1_loopback_enum); + +static const struct soc_enum aif2_loopback_enum = + SOC_ENUM_SINGLE(WM8994_AIF2_CONTROL_2, WM8994_AIF2_LOOPBACK_SHIFT, 2, + loopback_text); + +static const struct snd_kcontrol_new aif2_loopback = + SOC_DAPM_ENUM("AIF2 Loopback", aif2_loopback_enum); + static const struct soc_enum aif1dac_enum = SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 0, 2, aif1dac_text); @@ -1745,6 +1763,9 @@ SND_SOC_DAPM_ADC("DMIC1R", NULL, WM8994_POWER_MANAGEMENT_4, 2, 0), SND_SOC_DAPM_ADC("ADCL", NULL, SND_SOC_NOPM, 1, 0), SND_SOC_DAPM_ADC("ADCR", NULL, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_MUX("AIF1 Loopback", SND_SOC_NOPM, 0, 0, &aif1_loopback), +SND_SOC_DAPM_MUX("AIF2 Loopback", SND_SOC_NOPM, 0, 0, &aif2_loopback), + SND_SOC_DAPM_POST("Debug log", post_ev), }; @@ -1876,9 +1897,9 @@ static const struct snd_soc_dapm_route intercon[] = { { "AIF1DAC2L", NULL, "AIF1DAC Mux" }, { "AIF1DAC2R", NULL, "AIF1DAC Mux" }, - { "AIF1DAC Mux", "AIF1DACDAT", "AIF1DACDAT" }, + { "AIF1DAC Mux", "AIF1DACDAT", "AIF1 Loopback" }, { "AIF1DAC Mux", "AIF3DACDAT", "AIF3DACDAT" }, - { "AIF2DAC Mux", "AIF2DACDAT", "AIF2DACDAT" }, + { "AIF2DAC Mux", "AIF2DACDAT", "AIF2 Loopback" }, { "AIF2DAC Mux", "AIF3DACDAT", "AIF3DACDAT" }, { "AIF2ADC Mux", "AIF2ADCDAT", "AIF2ADCL" }, { "AIF2ADC Mux", "AIF2ADCDAT", "AIF2ADCR" }, @@ -1929,6 +1950,12 @@ static const struct snd_soc_dapm_route intercon[] = { { "AIF3ADCDAT", "AIF2DACDAT", "AIF2DACL" }, { "AIF3ADCDAT", "AIF2DACDAT", "AIF2DACR" }, + /* Loopback */ + { "AIF1 Loopback", "ADCDAT", "AIF1ADCDAT" }, + { "AIF1 Loopback", "None", "AIF1DACDAT" }, + { "AIF2 Loopback", "ADCDAT", "AIF2ADCDAT" }, + { "AIF2 Loopback", "None", "AIF2DACDAT" }, + /* Sidetone */ { "Left Sidetone", "ADC/DMIC1", "ADCL Mux" }, { "Left Sidetone", "DMIC2", "DMIC2L" }, -- 1.7.10.4

2 5

[alsa-devel] [PATCH] hda_analyzer: don't fail to display audio selector if caps variables aren't available
by Anisse Astier 30 May '13

30 May '13

In particular, when an audio selector node has Amp-In caps: N/A, it allows to still interact with the selector, and not have TypeError traceback throwns because of NoneType variables. Signed-off-by: Anisse Astier <anisse(a)astier.eu> --- Hi Jared, I'm not sure if this is the proper way to fix this, so please consider this a formal report for the error thrown while parsing this node: Node 0x22 [Audio Selector] wcaps 0x30010b: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Connection: 7 0x18* 0x19 0x1a 0x1b 0x1d 0x0b 0x12 Regards, Anisse -- hda-analyzer/hda_guilib.py | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/hda-analyzer/hda_guilib.py b/hda-analyzer/hda_guilib.py index 93c85e1..62eddab 100644 --- a/hda-analyzer/hda_guilib.py +++ b/hda-analyzer/hda_guilib.py @@ -285,9 +285,9 @@ class NodeGui(gtk.ScrolledWindow): frame.set_border_width(4) vbox = gtk.VBox(False, 0) if caps: - str = 'Offset: %d\n' % caps.ofs - str += 'Number of steps: %d\n' % caps.nsteps - str += 'Step size: %d\n' % caps.stepsize + str = 'Offset: %s\n' % (caps.ofs if caps.ofs else "N/A") + str += 'Number of steps: %s\n' % (caps.nsteps if caps.nsteps else "N/A") + str += 'Step size: %s\n' % (caps.stepsize if caps.stepsize else "N/A") str += 'Mute: %s\n' % (caps.mute and "True" or "False") vbox.pack_start(self.__new_text_view(text=str), True, True, 0) idx = 0 -- 1.8.3.rc1

2 2

[alsa-devel] [PATCH 01/10] ASoC: blackfin: Remove unused bf5xx-{i2s, tdm, ac97}-pcm.h
by Lars-Peter Clausen 30 May '13

30 May '13

The structs defined in these files are completely unused, so remove both the structs and the files. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/blackfin/bf5xx-ac97-pcm.c | 1 - sound/soc/blackfin/bf5xx-ac97-pcm.h | 26 -------------------------- sound/soc/blackfin/bf5xx-ad1836.c | 1 - sound/soc/blackfin/bf5xx-ad193x.c | 1 - sound/soc/blackfin/bf5xx-ad1980.c | 1 - sound/soc/blackfin/bf5xx-ad73311.c | 1 - sound/soc/blackfin/bf5xx-i2s-pcm.c | 1 - sound/soc/blackfin/bf5xx-i2s-pcm.h | 26 -------------------------- sound/soc/blackfin/bf5xx-ssm2602.c | 1 - sound/soc/blackfin/bf5xx-tdm-pcm.c | 1 - sound/soc/blackfin/bf5xx-tdm-pcm.h | 18 ------------------ 11 files changed, 78 deletions(-) delete mode 100644 sound/soc/blackfin/bf5xx-ac97-pcm.h delete mode 100644 sound/soc/blackfin/bf5xx-i2s-pcm.h delete mode 100644 sound/soc/blackfin/bf5xx-tdm-pcm.h diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c index 7e2f360..53f8408 100644 --- a/sound/soc/blackfin/bf5xx-ac97-pcm.c +++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c @@ -39,7 +39,6 @@ #include <asm/dma.h> -#include "bf5xx-ac97-pcm.h" #include "bf5xx-ac97.h" #include "bf5xx-sport.h" diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.h b/sound/soc/blackfin/bf5xx-ac97-pcm.h deleted file mode 100644 index d324d58..0000000 --- a/sound/soc/blackfin/bf5xx-ac97-pcm.h +++ /dev/null @@ -1,26 +0,0 @@ -/* - * linux/sound/arm/bf5xx-ac97-pcm.h -- ALSA PCM interface for the Blackfin - * - * Copyright 2007 Analog Device Inc. - * - * 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 _BF5XX_AC97_PCM_H -#define _BF5XX_AC97_PCM_H - -struct bf5xx_pcm_dma_params { - char *name; /* stream identifier */ -}; - -struct bf5xx_gpio { - u32 sys; - u32 rx; - u32 tx; - u32 clk; - u32 frm; -}; - -#endif diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c index d23f4b0..1528176 100644 --- a/sound/soc/blackfin/bf5xx-ad1836.c +++ b/sound/soc/blackfin/bf5xx-ad1836.c @@ -30,7 +30,6 @@ #include "../codecs/ad1836.h" -#include "bf5xx-tdm-pcm.h" #include "bf5xx-tdm.h" static struct snd_soc_card bf5xx_ad1836; diff --git a/sound/soc/blackfin/bf5xx-ad193x.c b/sound/soc/blackfin/bf5xx-ad193x.c index 0e55e9f..ce773b3 100644 --- a/sound/soc/blackfin/bf5xx-ad193x.c +++ b/sound/soc/blackfin/bf5xx-ad193x.c @@ -39,7 +39,6 @@ #include "../codecs/ad193x.h" -#include "bf5xx-tdm-pcm.h" #include "bf5xx-tdm.h" static struct snd_soc_card bf5xx_ad193x; diff --git a/sound/soc/blackfin/bf5xx-ad1980.c b/sound/soc/blackfin/bf5xx-ad1980.c index b30f88b..3450e8f 100644 --- a/sound/soc/blackfin/bf5xx-ad1980.c +++ b/sound/soc/blackfin/bf5xx-ad1980.c @@ -48,7 +48,6 @@ #include "../codecs/ad1980.h" -#include "bf5xx-ac97-pcm.h" #include "bf5xx-ac97.h" static struct snd_soc_card bf5xx_board; diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c index 61cc91d..786bbdd 100644 --- a/sound/soc/blackfin/bf5xx-ad73311.c +++ b/sound/soc/blackfin/bf5xx-ad73311.c @@ -45,7 +45,6 @@ #include "../codecs/ad73311.h" #include "bf5xx-sport.h" -#include "bf5xx-i2s-pcm.h" #if CONFIG_SND_BF5XX_SPORT_NUM == 0 #define bfin_write_SPORT_TCR1 bfin_write_SPORT0_TCR1 diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c index 262c1de..8726c3a 100644 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.c +++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c @@ -39,7 +39,6 @@ #include <asm/dma.h> -#include "bf5xx-i2s-pcm.h" #include "bf5xx-sport.h" static void bf5xx_dma_irq(void *data) diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.h b/sound/soc/blackfin/bf5xx-i2s-pcm.h deleted file mode 100644 index 0c2c5a6..0000000 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.h +++ /dev/null @@ -1,26 +0,0 @@ -/* - * linux/sound/arm/bf5xx-i2s-pcm.h -- ALSA PCM interface for the Blackfin - * - * Copyright 2007 Analog Device Inc. - * - * 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 _BF5XX_I2S_PCM_H -#define _BF5XX_I2S_PCM_H - -struct bf5xx_pcm_dma_params { - char *name; /* stream identifier */ -}; - -struct bf5xx_gpio { - u32 sys; - u32 rx; - u32 tx; - u32 clk; - u32 frm; -}; - -#endif diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c index 7dbeef1..9c19ccc 100644 --- a/sound/soc/blackfin/bf5xx-ssm2602.c +++ b/sound/soc/blackfin/bf5xx-ssm2602.c @@ -40,7 +40,6 @@ #include <linux/gpio.h> #include "../codecs/ssm2602.h" #include "bf5xx-sport.h" -#include "bf5xx-i2s-pcm.h" static struct snd_soc_card bf5xx_ssm2602; diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.c b/sound/soc/blackfin/bf5xx-tdm-pcm.c index 19e855d..a6b5457 100644 --- a/sound/soc/blackfin/bf5xx-tdm-pcm.c +++ b/sound/soc/blackfin/bf5xx-tdm-pcm.c @@ -39,7 +39,6 @@ #include <asm/dma.h> -#include "bf5xx-tdm-pcm.h" #include "bf5xx-tdm.h" #include "bf5xx-sport.h" diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.h b/sound/soc/blackfin/bf5xx-tdm-pcm.h deleted file mode 100644 index 7f8cc01..0000000 --- a/sound/soc/blackfin/bf5xx-tdm-pcm.h +++ /dev/null @@ -1,18 +0,0 @@ -/* - * sound/soc/blackfin/bf5xx-tdm-pcm.h -- ALSA PCM interface for the Blackfin - * - * Copyright 2009 Analog Device Inc. - * - * 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 _BF5XX_TDM_PCM_H -#define _BF5XX_TDM_PCM_H - -struct bf5xx_pcm_dma_params { - char *name; /* stream identifier */ -}; - -#endif -- 1.8.0

2 14

[alsa-devel] [PATCH 0/4 V6] Power-well API implementation for Haswell
by Wang Xingchao 30 May '13

30 May '13

Hi all, This is V6 and here're some changes notes: change from V5-->V6: - Remove duplication code in new introduced probe work - move duplication code in azx_probe_continue - remove unused #ifdef - replace request_module with symbol_request - replace spin_lock_irq with spin_lock_irqsave in gfx side - other typo fixes (review by Takashi) change from V4-->V5: - fix reference count bug - new patch on general runtime pm support for audio pci device - new patch to avoid request_module() deadlock change between V3-->V4: - add new structure i915_power_well - initialize drm_device pointer at module init time - change function name change between V2-->V3: - make SND_HDA_I915 selectable - use snd_printdd to output message - add return error code check - use symbol_request to replace symbol_get - release power_well at azx_free - some typo fixes changes between V1-->V2: - use reference count to track power-well usage - remove external module, compiled into snd-hda-intel instead - manage symbols and module loading properly - remove IS_HSW macro, use flag instead - remove audio callback for gfx driver to avoid dependency - split whole patch into two pieces for easy review - more typo fixes Takashi Iwai (1): ALSA: hda - Move azx_first_init() into azx_probe_continue() Wang Xingchao (3): ALSA: hda - Fix runtime PM check ALSA: hda - Add power-welll support for haswell HDA i915/drm: Add private api for power well usage drivers/gpu/drm/i915/i915_dma.c | 6 +++ drivers/gpu/drm/i915/i915_drv.h | 12 +++++ drivers/gpu/drm/i915/intel_drv.h | 4 ++ drivers/gpu/drm/i915/intel_pm.c | 92 +++++++++++++++++++++++++++++++++++--- include/drm/i915_powerwell.h | 36 +++++++++++++++ sound/pci/hda/Kconfig | 10 +++++ sound/pci/hda/Makefile | 2 + sound/pci/hda/hda_i915.c | 75 +++++++++++++++++++++++++++++++ sound/pci/hda/hda_i915.h | 35 +++++++++++++++ sound/pci/hda/hda_intel.c | 87 ++++++++++++++++++++++++++++------- 10 files changed, 335 insertions(+), 24 deletions(-) create mode 100644 include/drm/i915_powerwell.h create mode 100644 sound/pci/hda/hda_i915.c create mode 100644 sound/pci/hda/hda_i915.h -- 1.7.9.5

3 6

[alsa-devel] [PATCH] ASoC: OMAP: Remove obsolete Makefile line
by Paul Bolle 30 May '13

30 May '13

Support for omap2evm was removed in v3.0. But only one of its two lines in this Makefile was removed. Remove the second line too. Signed-off-by: Paul Bolle <pebolle(a)tiscali.nl> --- Eyeball tested only. sound/soc/omap/Makefile | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 2b22594..a725905 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -26,7 +26,6 @@ obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(CONFIG_SND_OMAP_SOC_RX51) += snd-soc-rx51.o obj-$(CONFIG_SND_OMAP_SOC_AMS_DELTA) += snd-soc-ams-delta.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o -obj-$(CONFIG_SND_OMAP_SOC_OMAP2EVM) += snd-soc-omap2evm.o obj-$(CONFIG_SND_OMAP_SOC_AM3517EVM) += snd-soc-am3517evm.o obj-$(CONFIG_SND_OMAP_SOC_OMAP_ABE_TWL6040) += snd-soc-omap-abe-twl6040.o obj-$(CONFIG_SND_OMAP_SOC_OMAP_TWL4030) += snd-soc-omap-twl4030.o -- 1.7.11.7

3 2

[alsa-devel] [PATCH] ASoC: wm0010: Set IRQ as a wake source
by Charles Keepax 30 May '13

30 May '13

The DSPs IRQ should be a wake source as several of the possible algorithms may run whilst the AP is asleepi and require to wake the AP to push or pull more data, such as compressed playback. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm0010.c | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 370af0c..b6df319 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -972,6 +972,13 @@ static int wm0010_spi_probe(struct spi_device *spi) } wm0010->irq = irq; + ret = irq_set_irq_wake(irq, 1); + if (ret) { + dev_err(wm0010->dev, "Failed to set IRQ %d as wake source: %d\n", + irq, ret); + return ret; + } + if (spi->max_speed_hz) wm0010->board_max_spi_speed = spi->max_speed_hz; else -- 1.7.2.5

2 1

[alsa-devel] [PATCH v6 00/10] ASoC: fsl-ssi: ac97-slave support
by Markus Pargmann 30 May '13

30 May '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. Regards, Markus 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: phycore-ac97: Add DT support ASoC: imx-pcm-dma: DT support ASoC: imx-pcm-fiq: Introduce pcm-fiq-params ASoC: fsl-ssi: Add support for imx-pcm-fiq ASoC: fsl-ssi: Use generic DMA bindings if possible ARM: imx: Export ac97 reset functions ASoC: fsl-ssi: imx ac97 support 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 | 2 +- sound/soc/fsl/fsl_ssi.c | 441 +++++++++++++++++---- 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 | 12 +- sound/soc/fsl/imx-ssi.c | 7 +- sound/soc/fsl/imx-ssi.h | 1 + sound/soc/fsl/phycore-ac97.c | 241 +++++++++-- 13 files changed, 643 insertions(+), 125 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

4 20

[alsa-devel] [PATCH 0/2] ASoC: cs42l52: HP gain enum and PCM Mixer dB scale fixes.
by Nicolas Schichan 29 May '13

29 May '13

Hi, The two patches in the serie contains the following fixes: - Use the correct shift value in hp_gain_enum, effectively allowing the higher gain values to be effective and the correct gain value to be reported to userland wrt what is written to the register. - Fix the PCM Mixer dB scale so that userland can report the correct dB gain values. Note that in its current form, the PCM Mixer control will only allow the range [-0.5 dB (0x7f), +12 dB (0x18)] to be programmed to the hardware. I don't know if it is intentional or not. Regards, Nicolas Schichan (2): ASoC: cs42l52: use correct PCM mixer TLV dB scale to match datasheet. ASoC: cs42l52: fix hp_gain_enum shift value. sound/soc/codecs/cs42l52.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) -- 1.7.10.4

3 6

[alsa-devel] [PATCH] ASoC: wm8994: Avoid leaking pm_runtime reference on removed jack race
by Mark Brown 29 May '13

29 May '13

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 1ddb067..4aaa8a0 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3886,7 +3886,7 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) ret); } else if (!(ret & WM1811_JACKDET_LVL)) { dev_dbg(codec->dev, "Ignoring removed jack\n"); - return IRQ_HANDLED; + goto out; } } else if (!(reg & WM8958_MICD_STS)) { snd_soc_jack_report(wm8994->micdet[0].jack, 0, -- 1.7.10.4

1 0

[alsa-devel] UCM {Supported,Conflicting}Device confusion
by Juho Hämäläinen 29 May '13

29 May '13

Hello, Firstly as a sidenote, Supported/Conflicting is checked only one-way, if say devices A and B are enabled and say B has ConflictingDevice "C", and device C doesn't list anything, enabling of C is possible (and after that disabling C is not possible before B is disabled first). Now, looking at the way SupportedDevice and ConflictingDevice are implemented in ucm and I came to following conclusion: If no SupportedDevice or ConflictingDevice: All devices and combinations are compatible, so implicit "SupportedDevice ALL". If SupportedDevice is listed: Device can be enabled as long as at least one enabled device is in SupportedDevice list. If ConflictingDevice is listed: Device can be enabled if none of active devices are in ConflictingDevice list (noting aforementioned limit in checking this) This is at least how I figured all combinations of multiple devices can be defined with only zero or one either SupportedDevice or ConflictingDevice entry per device. Did I understand this right? -- juho

2 1

[alsa-devel] [PATCH 1/3] lx6464es: remove useles pci_set_drvdata(pdev, NULL)
by Andy Shevchenko 29 May '13

29 May '13

The driver core handles this. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/pci/lx6464es/lx6464es.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c index 298bc9b..3230e57 100644 --- a/sound/pci/lx6464es/lx6464es.c +++ b/sound/pci/lx6464es/lx6464es.c @@ -1139,7 +1139,6 @@ out_free: static void snd_lx6464es_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } -- 1.8.2.rc0.22.gb3600c3

2 10

[alsa-devel] ASoC updates for v3.10
by Mark Brown 29 May '13

29 May '13

The following changes since commit e6135fe960bba1d1a3a6b6818839e6d8d3f21293: ALSA: usb-audio: proc: use found syncmaxsize to determine feedback format (2013-05-17 08:05:34 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.10-rc3 for you to fetch changes up to 1ab9ecc24819a8cf8ee982aaf6fb83298f094b0d: Merge remote-tracking branch 'asoc/fix/wm8994' into asoc-linus (2013-05-28 15:41:37 +0100) The diffstat and so on being rather obscured by the fact that I've got a newer -rc merged in here than you do in your branch, sorry. ---------------------------------------------------------------- ASoC: Updates for v3.10 A series of driver specific updates, none particularly critical, plus one fix to the compressed API code to handle capture streams properly which is very safe for mainline as there's no current users. ---------------------------------------------------------------- Alan Stern (6): USB: xHCI: override bogus bulk wMaxPacketSize values USB: fix Kconfig logic for USB_UHCI_HCD USB: UHCI: fix for suspend of virtual HP controller USB: fix latency in uhci-hcd and ohci-hcd USB: OHCI: fix logic for scheduling isochronous URBs USB: remove remaining instances of USB_SUSPEND Alasdair G Kergon (1): dm thin: fix metadata dev resize detection Albert Pool (1): rtlwifi: rtl8192cu: Add new USB ID Alex Deucher (10): drm/radeon/dce2: use 10khz units for audio dto calculation drm/radeon: add chip family for Hainan drm/radeon: fill in GPU init for Hainan (v2) drm/radeon: don't touch DCE or VGA regs on Hainan (v3) drm/radeon: fill in ucode loading support for Hainan drm/radeon: radeon-asic updates for Hainan drm/radeon: track which asics have UVD drm/radeon: sun/hainan chips do not have UVD (v2) drm/radeon: add golden register settings for Hainan (v2) drm/radeon: add Hainan pci ids Alex Elder (33): rbd: fix leak of format 2 snapshot context rbd: revalidate only for mapping size changes rbd: update capacity in rbd_dev_refresh() rbd: kill rbd_update_mapping_size() rbd: don't print warning if not mapping a parent rbd: don't look up snapshot id in rbd_dev_mapping_set() rbd: kill rbd_dev_clear_mapping() rbd: always set read-only flag in rbd_add() rbd: set the mapping size and features later rbd: zero format 1 header structure earlier rbd: refactor rbd_header_from_disk() rbd: update in-core header directly rbd: simplify rbd_dev_v1_probe() rbd: get rid of trivial v1 header wrappers rbd: define rbd_dev_v2_header_info() rbd: fix an incorrect assertion condition rbd: support reading parent page data rbd: set mapping read-only flag in rbd_add() rbd: only set up watch for mapped images rbd: kill rbd_img_request_get() libceph: init sent and completed when starting rbd: fix parent request size assumption rbd: support reading parent page data for writes rbd: ignore zero-overlap parent rbd: get parent info on refresh rbd: don't release write request until necessary rbd: define rbd_dev_unparent() rbd: define parent image request routines rbd: reference count parent requests rbd: detect when clone image is flattened rbd: re-submit read request for flattened clone rbd: re-submit write request for flattened clone rbd: re-submit flattened write request (part 2) Alex Hung (1): ACPI video: ignore BIOS initial backlight value for HP 1000 Alexander Bondar (1): iwlwifi: mvm: Prevent setting assoc flag in MAC_CONTEXT_CMD Alexander Gordeev (1): powerpc: Fix irq_set_affinity() return values Alexander Shiyan (2): s390: remove non existent reference to GENERIC_KERNEL_THREAD cpufreq: Fix incorrect dependecies for ARM SA11xx drivers Alexander Sverdlin (1): i2c: suppress lockdep warning on delete_device Alexandre Bounine (4): rapidio: make enumeration/discovery configurable rapidio: add enumeration/discovery start from user space rapidio: documentation update for enumeration changes rapidio/tsi721: fix bug in MSI interrupt handling Alexandre Courbot (1): ARM: tegra: defconfig fixes Alexandre Oliva (1): btrfs: do away with non-whole_page extent I/O Alexandru Gheorghiu (1): drivers: char: ipmi: Replaced kmalloc and strcpy with kstrdup Alistair Popple (1): powerpc: Add an in memory udbg console Amerigo Wang (2): virtio_net: use default napi weight by default ipv6,gre: do not leak info to user-space Amitkumar Karwar (1): mwifiex: fix memory leak issue when driver unload Andreas Larsson (2): spi: Initialize cs_gpio and cs_gpios with -ENOENT spi: Return error from of_spi_register_master on bad "cs-gpios" property Andreas Philipp (1): Correct allowed raid levels on balance. Andrew Bresticker (1): ASoC: max98090: request IRQF_ONESHOT interrupt Andrew Lunn (2): ARM: Orion: Remove redundant init_dma_coherent_pool_size() mv643xx_eth: fix NAPI weight being > 64 Andrew Morton (1): revert "selftest: add simple test for soft-dirty bit" Andy Adamson (1): NFSv4.1 Fix a pNFS session draining deadlock Andy Grover (3): target: Remove unused struct members in se_dev_entry target: Use FD_MAX_SECTORS/FD_BLOCKSIZE for blockdevs using fileio target: Have dev/enable show if TCM device is configured Andy Shevchenko (1): dma: acpi-dma: parse CSRT to extract additional resources Aneesh Kumar K.V (3): powerpc/mm: Use the correct mask value when looking at pgtable address powerpc: Fix build errors STRICT_MM_TYPECHECKS mm/THP: use pmd_populate() to update the pmd with pgtable_t pointer Anton Blanchard (1): powerpc/kexec: Fix kexec when using VMX optimised memcpy Antonio Quartulli (2): batman-adv: make DAT drop ARP requests targeting local clients batman-adv: reorder clean up routine in order to avoid race conditions Ard Biesheuvel (1): ARM: 7723/1: crypto: sha1-armv4-large.S: fix SP handling Arend van Spriel (1): brcmfmac: announce P2P_DEVICE support in wiphy structure Arnd Bergmann (4): staging/drm: imx: add missing dependencies staging/solo6x10: depend on CONFIG_FONTS mfd: ab8500: Debugfs code depends on gpadc USB: EHCI: remove bogus #error Aron Xu (1): MIPS: N64: Wire getdents64(2) Axel Lin (7): wm831x_backup: Fix wrong kfree call for devdata->backup.name pm2301_charger: Fix module alias prefix hwmon: (iio_hwmon) Fix missing iio_channel_release_all call if devm_kzalloc fail hwmon: (iio_hwmon) Fix null pointer dereference gpio: mcp23s08: Fix build error when CONFIG_SPI_MASTER=y && CONFIG_I2C=m gpio: Don't override the error code in probe error handling iio: dac: Fix build error when CONFIG_SPI_MASTER=y && CONFIG_I2C=m Bastian Triller (1): ACPI / video: Add "Asus UL30A" to ACPI video detect blacklist Ben Hutchings (7): sfc: Delete EFX_PAGE_IP_ALIGN, equivalent to NET_IP_ALIGN sfc: Reduce RX scatter buffer size, and reduce alignment if appropriate perf: net_dropmonitor: Fix trace parameter order perf: net_dropmonitor: Fix symbol-relative addresses perf: net_dropmonitor: Do not assume ordering of dictionaries perf: net_dropmonitor: Use bisection in symbol lookup perf: net_dropmonitor: Remove progress indicator Ben Skeggs (9): drm/nouveau/bios: fix thinko in ZM_MASK_ADD opcode drm/nvc0/ce: disable ce1 on a number of chipsets drm/nvc0/ltcg: fix handling of disabled partitions drm/nve0/ltcg: poke the partition count into yet another register drm/nve0/fifo: prevent races between clients updating playlists drm/nvc0/fifo: prevent races between clients updating playlists drm/nvc0/fifo: prevent CHAN_TABLE_ERROR:CHANNEL_PENDING on fifo fini drm/nv50/fifo: prevent races between clients updating playlists drm/nouveau: ensure channels are stopped before saving fences for suspend Ben Widawsky (1): Revert "drm/i915: Calculate correct stolen size for GEN7+" Benjamin Herrenschmidt (10): powerpc/powernv: Properly drop characters if console is closed powerpc/powernv: Improve kexec reliability powerpc/powernv: Detect OPAL v3 API version powerpc/powernv: Fix starting of secondary CPUs on OPALv2 and v3 powerpc: Set show_unhandled_signals to 1 by default powerpc: Fix TLB cleanup at boot on POWER8 powerpc/pci: Fix bogus message at boot about empty memory resources powerpc/powernv: Fix condition for when to invalidate the TCE cache powerpc: Make radeon 32-bit MSI quirk work on powernv powerpc/powernv: Build a zImage.epapr Benjamin LaHaise (2): ipmi: ipmi_devintf: compat_ioctl method fails to take ipmi_mutex aio: fix kioctx not being freed after cancellation at exit time Bing Zhao (1): mwifiex: clear is_suspended flag when interrupt is received early Bjorn Helgaas (1): Revert "x86/pci/mrst: Use configuration mechanism 1 for 00:00.0, 00:02.0, 00:03.0" Bob Peterson (2): GFS2: two minor quota fixups GFS2: Use single-block reservations for directories Boris BREZILLON (1): ARM: at91/dt: fix macb pinctrl_macb_rmii_mii_alt definition Borislav Petkov (3): cpufreq, ondemand: Remove leftover debug line PM / hibernate: Correct documentation amd64_edac: Fix bogus sysfs file permissions Brian Behlendorf (1): drivers/block/brd.c: fix brd_lookup_page() race Brian King (1): powerpc/pseries: Make 32-bit MSI quirk work on systems lacking firmware support Catalin Marinas (3): arm64: Invoke the of_platform_populate() at arch_initcall() level arm64: Do not source kernel/time/Kconfig explicitly ACPI / scan: Fix memory leak on acpi_scan_init_hotplug() error path Charles Keepax (2): ASoC: soc-compress: Send correct stream event for capture start ASoC: wm5110: Correct DSP4R Mixer control name Chen Gang (3): arm64: Fix duplicate definition of early_console drivers/char/ipmi: memcpy, need additional 2 bytes to avoid memory overflow net: irda: using kzalloc() instead of kmalloc() to avoid strncpy() issue. Chris Cummins (1): drm: Use names of ioctls in debug traces Chris Mason (2): Btrfs: use a btrfs bioset instead of abusing bio internals Merge branch 'for-chris' of git://git.kernel.org/.../josef/btrfs-next Chris Wilson (1): drm/i915: Always normalize return timeout for wait_timeout_ioctl Christian Borntraeger (4): s390: disable pfmf for clear page instruction s390: fix gmap_ipte_notifier vs. software dirty pages s390/xpram: mark xpram as non-rotational s390/pgtable: fix ipte notify bit Christian Gmeiner (1): drivers/leds/leds-ot200.c: fix error caused by shifted mask Christian Ruppert (1): ARC: [TB10x] Remove redundant abilis,simple-pinctrl mechanism Christopher Harvey (4): drm/mgag200: Don't change unrelated registers during modeset drm/mgag200: Fix writes into MGA1064_PIX_CLK_CTL register drm/mgag200: Convert counter delays to jiffies drm/mgag200: Fix framebuffer base address programming Chuck Lever (1): NFS: Fix SETCLIENTID fallback if GSS is not available Cliff Wickman (1): mm/pagewalk.c: walk_page_range should avoid VM_PFNMAP areas Cong Wang (1): xfrm6: release dev before returning error Corey Minyard (1): ipmi: Improve error messages on failed irq enable Damien Lespiau (2): drm: Add missing break in the command line mode parsing code drm: Don't prune modes loudly when a connector is disconnected Dan Carpenter (6): xen/privcmd: fix condition in privcmd_close() ntb: off by one sanity checks NTB: fix pointer math issues ipg: fix an unsigned widening cast of '~' truncation issue USB: cxacru: potential underflow in cxacru_cm_get_array() isdn/kcapi: fix a small underflow Dan Magenheimer (1): staging: ramster: add how-to document Dan Williams (1): USB: option: add device IDs for Dell 5804 (Novatel E371) WWAN card Daniel Drake (1): mwifiex: fix setting of multicast filter Daniel Mack (1): ASoC: davinci: fix sample rotation Daniel Tang (1): ARM: 7712/1: Remove trailing whitespace in arch/arm/Makefile Daniel Vetter (3): drm/mm: fix dump table BUG drm: don't check modeset locks in panic handler Revert "drm/i915: revert eDP bpp clamping code changes" Darrick J. Wong (1): ttyprintk: Fix NULL pointer deref by setting tty_port ops after initializing port Dave Airlie (11): Merge branch 'for-linux-next' of git://people.freedesktop.org/~danvet/drm-intel into drm-next drm: remove unused wrapper macros drm/qxl: fix ioport interactions for kernel submitted commands. qxl: drop unused variable. drm/qxl: drop active_user_framebuffer as its unneeded qxl: fix bug with object eviction and update area drm/radeon: restore nomodeset operation (v2) drm/nouveau: fix build with nv50->nvc0 Merge branch 'drm-nouveau-fixes-3.10' of git://anongit.freedesktop.org/git/nouveau/linux-2.6 into drm-next Merge branch 'drm-fixes-3.10' of git://people.freedesktop.org/~agd5f/linux into drm-next Merge branch 'drm-fixes-3.10-sun' of git://people.freedesktop.org/~agd5f/linux into drm-next Dave Chinner (7): xfs: fix sub-page blocksize data integrity writes xfs: fix rounding in xfs_free_file_space xfs: Don't reference the EFI after it is freed xfs: fix missing KM_NOFS tags to keep lockdep happy xfs: xfs_da3_node_read_verify() doesn't handle XFS_ATTR3_LEAF_MAGIC xfs: xfs_attr_shortform_allfit() does not handle attr3 format. xfs: remote attribute lookups require the value length Dave Jones (1): Add missing module license tag to vring helpers. David Cohen (1): gpio-langwell: fix irq conflicts when DT is not used David Daney (3): Revert "MIPS: microMIPS: Support dynamic ASID sizing." Revert "MIPS: Allow ASID size to be determined at boot time." MIPS: Make virt_to_phys() work for all unmapped addresses. David Rientjes (2): mfd: tps65912: Select MFD_CORE usb, chipidea: fix link error when USB_EHCI_HCD is a module David S. Miller (9): Merge branch 'wireless' Merge branch 'qlcnic' Merge tag 'batman-adv-fix-for-davem' of git://git.open-mesh.org/linux-merge Merge branch 'master' of git://git.kernel.org/.../pablo/nf Merge tag 'batman-adv-fix-for-davem' of git://git.open-mesh.org/linux-merge Merge branch 'for-davem' of git://git.kernel.org/.../linville/wireless Merge branch 'perf_net_dropmonitor' Merge tag 'batman-adv-fix-for-davem' of git://git.open-mesh.org/linux-merge net: Revert unused variable changes. David Sterba (1): btrfs: annotate quota tree for lockdep David Woodhouse (1): powerpc: Provide __bswapdi2 Denis CIOCCA (1): iio:common:st: added disable function after read info raw data Denis Efremov (4): ipv4: ip_output: remove inline marking of EXPORT_SYMBOL functions staging: nvec: remove inline marking of EXPORT_SYMBOL functions iwlegacy: remove inline marking of EXPORT_SYMBOL functions MIPS: MSP71xx: remove inline marking of EXPORT_SYMBOL functions Dirk Behme (1): ARM i.MX6: correct MLB clock configuration Dirk Brandewie (6): cpufreq / intel_pstate: remove idle time and duration from sample and calculations cpufreq / intel_pstate: use lowest requested max performance cpufreq / intel_pstate: fix ffmpeg regression cpufreq / intel_pstate: Remove idle mode PID cpufreq / intel_pstate: remove #ifdef MODULE compile fence cpufreq / intel_pstate: Add additional supported CPU ID Emilio López (4): ARM: ux500: select REGULATOR net: ethernet: apple: drop unused variable net: ethernet: korina: drop unused variable net: ethernet: sun: drop unused variable Emmanuel Grumbach (2): iwlwifi: mvm: tell firmware to let multicast frames in iwlwifi: mvm: fix aggregation drain flow Eric Dumazet (8): ipv6: do not clear pinet6 field tcp: fix tcp_md5_hash_skb_data() net/802/mrp: fix lockdep splat tcp: gso: do not generate out of order packets bonding: allow TSO being set on bonding master ipv6: fix possible crashes in ip6_cork_release() ip_gre: fix a possible crash in ipgre_err() tcp: xps: fix reordering issues EunBong Song (1): MIPS: Fix build error for crash_dump.c in 3.10-rc1 Ezequiel Garcia (3): ARM: mvebu: Add a ranges entry to translate devbus childs ARM: mvebu: Fix ranges entry on XP GP board serial: 8250_dw: Add valid clk pointer check Fabio Baltieri (5): mfd: abx500-core: Fix sparse warning mfd: ab8500-sysctrl: Fix sparse warning mfd: ab8500-sysctrl: Set sysctrl_dev during probe mfd: ab8500-sysctrl: Always enable pm_power_off handler mfd: ab8500-sysctrl: Let sysctrl driver work without pdata Fabio Estevam (3): fec: Fix inconsistent lock state fec: Invert the order of function calls in fec_restart() staging: imx-drm: imx-tve: Check the return value of 'regulator_enable()' Felix Fietkau (5): ath9k: fix key allocation error handling for powersave keys mac80211: fix spurious RCU warning and update documentation ath9k: fix aggregation stop/flush handling ath9k: fix rate handling/reporting ath9k: fix draining aggregation tid buffers Florian Fainelli (1): broadcom: add include guards to include/linux/brcmphy.h Gabriel de Perthuis (1): btrfs: don't stop searching after encountering the wrong item Geert Uytterhoeven (14): iscsi-target: Make buf param of iscsit_do_crypto_hash_buf() const void * net/ethernet: NET_CALXEDA_XGMAC should depend on HAS_DMA net/ethernet: STMMAC_ETH should depend on HAS_DMA net/wireless: ATH9K should depend on HAS_DMA net/ethernet: ARM_AT91_ETHER should depend on HAS_DMA net/ethernet: MACB should depend on HAS_DMA caif: CAIF_VIRTIO should depend on HAS_DMA mfd: si476x: Do not use binary constants mfd: cros_ec_spi: Use %z to format pointer differences mfd: si476x: Use get_unaligned_be16() for unaligned be16 loads staging: zcache: Fix incorrect module_param_array types net/wireless: ATH9K should depend on HAS_DMA uio: UIO_DMEM_GENIRQ should depend on HAS_DMA fbdev: FB_GOLDFISH should depend on HAS_DMA Gernot Vormayr (1): drivers/block/xsysace.c: fix id with missing port-number Gomella, Andrew (NIH/NHLBI) [F] (1): USB: ftdi_sio: Add support for Newport CONEX motor drivers Grant Likely (1): Merge branch 'for-next' of git://sources.calxeda.com/kernel/linux into HEAD Greg Kroah-Hartman (3): Merge tag 'fixes-for-v3.10-rc2' of git://git.kernel.org/.../balbi/usb into usb-linus driver core: export subsys_virtual_register Merge tag 'iio-fixes-for-3.10a' of git://git.kernel.org/.../jic23/iio into staging-linus Gregory CLEMENT (3): ARM: mvebu: select ARCH_REQUIRE_GPIOLIB for mvebu platform ARM: plat-orion: Fix num_resources and id for ge10 and ge11 ARM: dts: mvebu: Fix wrong the address reg value for the L2-cache node Guenter Roeck (2): hwmon: (nct6775) Do not create non-existing attributes MAINTAINERS: Add myself as maintainer for LM95234 and TMP401 drivers H Hartley Sweeten (1): staging: comedi: ni_labpc: fix build when VIRT_TO_BUS is not defined Haiyang Zhang (1): MAINTAINERS: update Hyper-V file list Hans Schillstrom (1): netfilter: log: netns NULL ptr bug when calling from conntrack Hans-Christian Egtvedt (2): avr32: move NODES_SHIFT into Kconfig and delete numnodes.h avr32: fix relocation check for signed 18-bit offset Hans-Christoph Schemmel (1): qmi_wwan: Added support for Cinterion's PLxx WWAN Interface Heikki Krogerus (1): serial: 8250_dw: add ACPI ID for Intel BayTrail Heiko Abraham (1): Input: egalax_ts - ABS_MT_POSITION_Y not reported well Heiko Carstens (2): s390/disassembler: prevent endless loop in print_fn_code() s390/ftrace: fix mcount adjustment Helge Deller (8): parisc: implement irq stacks - part 2 (v2) parisc: make default cross compiler search more robust (v3) MPILIB: disable usage of floating point registers on parisc parisc: add rp5470 entry to machine database parisc: use PAGE_SHIFT instead of hardcoded value 12 in pacache.S parisc: add additional parisc git tree to MAINTAINERS file parisc: show number of FPE and unaligned access handler calls in /proc/interrupts parisc: fix irq stack on UP and SMP Hema Prathaban (1): staging: vt6656: use free_netdev instead of kfree Himanshu Madhani (2): qlcnic: Fix missing bracket in module parameter. qlcnic: Fix ethtool supported port status for 83xx Ian Abbott (4): staging: comedi: allow buffer resize if previous resize failed staging: comedi: work without HAS_DMA staging: comedi: ni_mio_common: only do counter commands for ni_pcimio staging: comedi: complain if dma buffer allocation not supported Ilan Peer (2): nl80211: Add wdev identifier to some nl80211 notifications iwlwifi: mvm: Always use SCAN_TYPE_FORCED Imre Deak (2): drm/i915: hsw: fix link training for eDP on port-A wait: fix false timeouts when using wait_event_timeout() Ingo Molnar (1): Merge branch 'rcu/doc' of git://git.kernel.org/.../paulmck/linux-rcu into core/urgent Jaccon Bastiaansen (1): ARM: 7720/1: ARM v6/v7 cmpxchg64 shouldn't clear upper 32 bits of the old/new value Jan Kara (1): jbd,jbd2: fix oops in jbd2_journal_put_journal_head() Jan-Simon Möller (1): ACPI: Fix section to __init. Align with usage in acpixf.h Jani Nikula (1): drm/i915: clear the stolen fb before resuming Jarod Wilson (1): drivers/char/random.c: fix priming of last_data Jason Wang (1): virtio_net: enable napi for all possible queues during open Jean Delvare (3): staging: Swap zram and zsmalloc in Kconfig i2c: i801: Document feature bits in modinfo hwmon: (tmp401) Drop redundant safety on cache lifetime Jean-Christophe PLAGNIOL-VILLARD (3): ARM: at91/trivial: fix model name for SAM9X25-EK ARM: at91: rm9200 fix time support ARM: at91: at91sam9n12: move external irq declatation to DT Jeff Layton (1): cifs: only set ops for inodes in I_NEW state Jeff Mahoney (1): hfs: avoid crash in hfs_bnode_create Jeff Moyer (1): aio: fix io_getevents documentation Jiang Liu (1): avr32: fix building warnings caused by redefinitions of HZ Jingoo Han (1): usb: gadget: s3c-hsotg: pass 'struct usb_request *' to usb_gadget_unmap_request() Jiri Kosina (1): random: fix accounting race condition with lockless irq entropy_count update Jiri Olsa (2): perf: Fix EXIT event notification perf: Factor out auxiliary events notification Jiri Pirko (1): macvlan: fix passthru mode race between dev removal and rx path Jiri Slaby (1): TTY: rocket, fix more no-PCI warnings Joachim Eastwood (1): spi/atmel: fix speed_hz check in atmel_spi_transfer() Joern Engel (2): target: removed unused transport_state flag target: close target_put_sess_cmd() vs. core_tmr_abort_task() race Johan Hovold (7): USB: serial: add wait_until_sent operation USB: serial: add generic wait_until_sent implementation USB: ftdi_sio: clean up get_modem_status USB: ftdi_sio: fix chars_in_buffer overhead USB: io_ti: fix chars_in_buffer overhead USB: ti_usb_3410_5052: fix chars_in_buffer overhead USB: serial: clean up chars_in_buffer Johannes Berg (9): cfg80211: fix wiphy_register error path cfg80211: fix WoWLAN wakeup tracing mac80211: fix IEEE80211_SDATA_DISCONNECT_RESUME mac80211: fix HT beacon-based channel switch handling mac80211: use just spin_lock() in ieee80211_get_tkip_p2k() mac80211: fix AP-mode frame matching cfg80211: fix interface down/disconnect state handling mac80211: report deauth to cfg80211 for local state change cfg80211: fix sending WoWLAN TCP wakeup settings Johannes Weiner (1): mm: memcg: remove incorrect VM_BUG_ON for swap cache pages in uncharge John David Anglin (2): parisc: fix SMP races when updating PTE and TLB entries in entry.S parisc: make interrupt and interruption stack allocation reentrant John Stultz (1): time: Revert ALWAYS_USE_PERSISTENT_CLOCK compile time optimizaitons John W. Linville (3): Merge branch 'master' of git://git.kernel.org/.../linville/wireless into for-davem Merge branch 'for-john' of git://git.kernel.org/.../iwlwifi/iwlwifi-fixes Merge branch 'master' of git://git.kernel.org/.../linville/wireless into for-davem Jon Hunter (1): gpio/omap: ensure gpio context is initialised Jon Mason (8): NTB: variable dereferenced before check NTB: Handle 64bit BAR sizes NTB: Link toggle memory leak NTB: reset tx_index on link toggle NTB: Correctly handle receive buffers of the minimal size NTB: memcpy lockup workaround ntb_netdev: remove from list on exit NTB: Multiple NTB client fix Jonathan Corbet (1): dummy-irq: require the user to specify an IRQ number Joonsoo Kim (1): workqueue: correct handling of the pool spin_lock Josef Ahmad (1): i2c: designware: fix RX FIFO overrun Josef Bacik (4): Btrfs: don't null pointer deref on abort Btrfs: remove warn on in free space cache writeout Btrfs: handle running extent ops with skinny metadata Btrfs: make sure roots are assigned before freeing their nodes Joseph Qi (2): ocfs2: unlock rw lock if inode lock failed ocfs2: goto out_unlock if ocfs2_get_clusters_nocache() failed in ocfs2_fiemap() Josh Wu (1): ARM: at91/sama5: fix incorrect PMC pcr div definition K. Y. Srinivasan (1): Drivers: hv: Fix a bug in get_vp_index() Kees Cook (2): drm: refactor call to request_module drm/radeon: check incoming cliprects pointer Kent Yoder (1): drivers/crypto/nx: Fixes for multiple races and issues Kevin Hilman (1): idle: Fix hlt/nohlt command-line handling in new generic idle Konrad Rzeszutek Wilk (11): x86/microcode: Add local mutex to fix physical CPU hot-add deadlock Merge tag 'v3.10-rc1' into stable/for-linus-3.10 xen/tmem: Cleanup. Remove the parts that say temporary. xen/tmem: Move all of the boot and module parameters to the top of the file. xen/tmem: Split out the different module/boot options. xen/tmem: Fix compile warning. xen/tmem: s/disable_// and change the logic. xen/tmem: Remove the boot options and fold them in the tmem.X parameters. xen/tmem: Remove the usage of 'noselfshrink' and use 'tmem.selfshrink' bool instead. xen/tmem: Remove the usage of '[no|]selfballoon' and use 'tmem.selfballooning' bool instead. xen/tmem: Don't use self[ballooning|shrinking] if frontswap is off. Kyle McMartin (1): score: remove redundant kcore_list entries Lachlan McIlroy (1): ext4: limit group search loop for non-extent files Lan Tianyu (2): ACPI / EC: Restart transaction even when the IBF flag set ACPI / AC: Add sleep quirk for Thinkpad e530 Larry Finger (1): rtlwifi: rtl8188ee: Fix warning when building on big-endian systems Lars-Peter Clausen (2): tty: nwpserial: Pass correct pointer to free_irq() drivers/rtc/rtc-pl031.c: pass correct pointer to free_irq() Laurent Pinchart (2): pinctrl: generic: Fix typos and clarify comments pinctrl/lantiq: Free mapping configs for both pin and groups Lee Jones (12): ARM: 7714/1: mmc: mmci: Ensure return value of regulator_enable() is checked ARM: ux500: Remove duplicated assignment of ab8500_platdata ARM: ux500: Fix incorrect DEBUG UART virtual addresses ARM: ux500: Provide device enumeration number suffix for SMSC911x pinctrl: abx500: Specify failed sub-driver by ID instead of driver_data pinctrl: abx500: Rejiggle platform data and DT initialisation staging: ste_rmi4: Suppress 'ignoring return value of ‘regulator_enable()' warning mfd: ab8500-gpadc: Suppress 'ignoring regulator_enable() return value' warning mfd: ab8500-core: Use the correct driver name when enabling gpio/pinctrl mfd: db8500-prcmu: Supply the pdata_size attribute for db8500-thermal mfd: ab8500-core: Pass GPADC compatible string to MFD core serial: pl011: protect attribute read from NULL platform data struct Leonid Yegoshin (1): mm compaction: fix of improper cache flush in migration code Lespiau, Damien (2): drm: Make the HPD status updates debug logs more readable drm: Only print a debug message when the polled connector has changed Li Zhong (6): powerpc: Fix MAX_STACK_TRACE_ENTRIES too low warning again powerpc: Syscall hooks for context tracking subsystem powerpc: Exception hooks for context tracking subsystem powerpc: Exit user context on notify resume powerpc: Use the new schedule_user API on userspace preemption powerpc: select HAVE_CONTEXT_TRACKING for pSeries Libo Chen (4): usb: ehci-s5p: fix memleak when fallback to pdata usb: isp1760-if: fix memleak when platform_get_resource fail usb: ohci: fix goto wrong tag in err case MIPS: Xway: Fix clk leak Linus Lüssing (1): batman-adv: Fix rcu_barrier() miss due to double call_rcu() in TT code Linus Torvalds (72): Merge tag 'fixes-for-linus' of git://git.kernel.org/.../rusty/linux Merge branch 'drm-next' of git://people.freedesktop.org/~airlied/linux Merge tag 'spi-v3.10-rc1' of git://git.kernel.org/.../broonie/spi Merge git://git.kernel.org/.../davem/net Merge branch 'parisc-for-3.10' of git://git.kernel.org/.../deller/parisc-linux Merge tag 'fixes-for-3.10-rc2-tag' of git://git.kernel.org/.../sstabellini/xen Merge branch 'merge' of git://git.kernel.org/.../benh/powerpc Merge branch 'for-3.10-fixes' of git://git.kernel.org/.../tj/wq Merge tag 'ext4_for_linus_stable' of git://git.kernel.org/.../tytso/ext4 Merge branch 'for-linus' of git://git.kernel.org/.../sage/ceph-client Merge branch 'fixes' of git://git.linaro.org/people/rmk/linux-arm Merge branch 'core-urgent-for-linus' of git://git.kernel.org/.../tip/tip Merge branch 'timers-urgent-for-linus' of git://git.kernel.org/.../tip/tip Merge branch 'perf-urgent-for-linus' of git://git.kernel.org/.../tip/tip Merge branch 'x86-urgent-for-linus' of git://git.kernel.org/.../tip/tip Merge tag 'trace-fixes-v3.10-rc1' of git://git.kernel.org/.../rostedt/linux-trace Merge branch 'queue' of git://git.kernel.org/.../nab/target-pending Merge branch 'rcu/urgent' of git://git.kernel.org/.../paulmck/linux-rcu Merge branch 'for-3.10-fixes' of git://git.kernel.org/.../tj/wq Merge branch 'ipmi' (minor ipmi fixes from Corey) Merge tag 'ntb-bugfixes-3.10' of git://github.com/jonmason/ntb Merge tag 'pm+acpi-3.10-rc2' of git://git.kernel.org/.../rafael/linux-pm Merge branch 'drm-next' of git://people.freedesktop.org/~airlied/linux Merge tag 'sound-3.10' of git://git.kernel.org/.../tiwai/sound Merge tag 'arm64-fixes' of git://git.kernel.org/.../cmarinas/linux-aarch64 Merge tag 'kmemleak-fixes' of git://git.kernel.org/.../cmarinas/linux-aarch64 Merge branch 'upstream' of git://git.linux-mips.org/pub/scm/ralf/upstream-linus Merge tag 'devicetree-for-linus' of git://git.secretlab.ca/git/linux Merge branch 'devm_no_resource_check' of git://git.kernel.org/.../wsa/linux Merge branch 'for-linus' of git://git.kernel.org/.../mason/linux-btrfs Merge tag 'dm-3.10-fixes' of git://git.kernel.org/.../agk/linux-dm Merge branch 'rc-fixes' of git://git.kernel.org/.../mmarek/kbuild Merge tag 'pinctrl-fixes-v3.10-2' of git://git.kernel.org/.../linusw/linux-pinctrl x86: Fix bit corruption at CPU resume time Merge tag 'hwmon-for-linus' of git://git.kernel.org/.../groeck/linux-staging Merge tag 'stable/for-linus-3.10-rc1-tag' of git://git.kernel.org/.../konrad/xen Linux 3.10-rc2 Merge branch 'drm-fixes' of git://people.freedesktop.org/~airlied/linux Merge branch 'drm-radeon-sun-hainan' of git://people.freedesktop.org/~airlied/linux Merge branch 'for-linus' of git://git.kernel.org/.../s390/linux Merge branch 'i2c/for-current' of git://git.kernel.org/.../wsa/linux Merge branch 'leds-fixes-3.10' of git://git.kernel.org/.../cooloney/linux-leds Merge tag 'fixes-for-linus' of git://git.kernel.org/.../rusty/linux Merge tag 'mfd-fixes-3.10-1' of git://git.kernel.org/.../sameo/mfd-fixes Merge branch 'upstream' of git://git.linux-mips.org/pub/scm/ralf/upstream-linus Merge branch 'for-linus' of git://git.kernel.org/.../egtvedt/linux-avr32 Merge git://git.kernel.org/pub/scm/virt/kvm/kvm Merge tag 'usb-3.10-rc2' of git://git.kernel.org/.../gregkh/usb Merge tag 'char-misc-3.10-rc2' of git://git.kernel.org/.../gregkh/char-misc Merge tag 'driver-core-3.10-rc2' of git://git.kernel.org/.../gregkh/driver-core Merge tag 'staging-3.10-rc2' of git://git.kernel.org/.../gregkh/staging Merge tag 'tty-3.10-rc2' of git://git.kernel.org/.../gregkh/tty Merge branch 'for-linus' of git://git.kernel.org/.../dtor/input Merge tag 'pci-v3.10-fixes-2' of git://git.kernel.org/.../helgaas/pci Merge tag 'gpio-fixes-v3.10-1' of git://git.kernel.org/.../linusw/linux-gpio Merge git://git.kernel.org/.../davem/net Merge branch 'merge' of git://git.kernel.org/.../benh/powerpc Merge git://git.kernel.org/.../steve/gfs2-3.0-fixes Merge branch 'for-next' of git://git.samba.org/sfrench/cifs-2.6 Merge tag 'trace-fixes-v3.10-rc2' of git://git.kernel.org/.../rostedt/linux-trace Merge tag 'fixes-for-linus' of git://git.kernel.org/.../arm/arm-soc Merge branch 'akpm' (incoming from Andrew Morton) Merge branch 'fixes' of git://git.linaro.org/people/rmk/linux-arm Merge tag 'arc-v3.10-fixes' of git://git.kernel.org/.../vgupta/arc Merge branch 'fixes' of git://git.infradead.org/users/vkoul/slave-dma Merge tag 'pm+acpi-3.10-rc3' of git://git.kernel.org/.../rafael/linux-pm Merge tag 'for-v3.10-fixes' of git://git.infradead.org/battery-2.6 Merge tag 'for-linus-v3.10-rc3' of git://oss.sgi.com/xfs/xfs Merge branch 'parisc-for-3.10' of git://git.kernel.org/.../deller/parisc-linux Merge tag 'edac_fixes_for_3.10' of git://git.kernel.org/.../bp/bp Merge tag 'nfs-for-3.10-3' of git://git.linux-nfs.org/projects/trondmy/linux-nfs Linux 3.10-rc3 Linus Walleij (2): mfd: ab8500: Pass AB8500 IRQ to debugfs code by resource ARM: nomadik: fix IRQ assignment for SMC ethernet Lisa Nguyen (3): xen/xenbus: Fixed indentation error in switch case xen/xenbus: Fixed over 80 character limit issue xen: Fixed assignment error in if statement Liu Bo (2): Btrfs: fix off-by-one in fiemap Btrfs: return errno if possible when we fail to allocate memory Lucas Stach (2): clk: tegra: remove USB from clk init table clk: tegra: add ac97 controller clock Ludovic Desroches (1): ARM: at91: dts: request only spi cs-gpios used on sama5d3x cpu module Luiz Capitulino (1): package: Makefile: unbreak binrpm-pkg target Malcolm Priestley (1): staging: vt6656: [bug] Fix missing spin lock in iwctl_siwpower. Manfred Spraul (1): ipc/sem.c: Fix missing wakeups in do_smart_update_queue() Manish Chopra (2): qlcnic: Fix setting MAC address qlcnic: Fix bug in diagnostics test reset recovery path Manjunathappa, Prakash (1): spi/davinci: fix module build error Marc Dietrich (3): staging: nvec: add missing module aliases staging: nvec: implement unregistering of notifiers staging: nvec: cleanup childs on remove Marc Dionne (1): workqueue: Make schedule_work() available again to non GPL modules Marek Lindner (2): batman-adv: check proto length before accessing proto string buffer batman-adv: check return value of pskb_trim_rcsum() Marek Vasut (1): staging: video: imx: Select VIDEOMODE_HELPERS for parallel display Mark Brown (11): Merge branch 'spi/merge' of git://git.secretlab.ca/git/linux into spi-fix-grant Merge remote-tracking branch 'spi/fix/atmel' into spi-linus Merge remote-tracking branch 'spi/fix/grant' into spi-linus ASoC: wm5110: Add missing speaker initialisation ASoC: wm8994: Fix reporting of accessory removal on WM8958 Merge remote-tracking branch 'asoc/fix/arizona' into asoc-linus Merge remote-tracking branch 'asoc/fix/compress' into asoc-linus Merge remote-tracking branch 'asoc/fix/cs42l52' into asoc-linus Merge remote-tracking branch 'asoc/fix/davinci' into asoc-linus Merge remote-tracking branch 'asoc/fix/max98090' into asoc-linus Merge remote-tracking branch 'asoc/fix/wm8994' into asoc-linus Martin Hundebøll (1): batman-adv: Avoid double freeing of bat_counters Martin Michlmayr (1): Kirkwood: Enable PCIe port 1 on QNAP TS-11x/TS-21x Martin Schwidefsky (1): s390/smp: fix cpu re-scan vs. cpu state Masami Hiramatsu (5): tracing/kprobes: Use rcu_dereference_raw for tp->files tracing/kprobes: Fix a sparse warning for incorrect type in assignment tracing/kprobes: Make print_*probe_event static tracing: Return -EBUSY when event_enable_func() fails to get module kprobes/mips: Fix to check double free of insn slot Matthijs Kooijman (2): MIPS: ralink: use the dwc2 driver for the rt305x USB controller staging: dwc2: Fix dma-enabled platform devices using a default dma_mask Matwey V. Kornilov (2): tty: mxser: fix usage of opmode_ioaddr tty: mxser: Fix build warning introduced by dfc7b837c7f9 (Re: linux-next: build warning after merge of the tty.current tree) Maxime Ripard (3): ARM: sunxi: Fix Mini X-plus device tree build ARM: sunxi: select ARCH_REQUIRE_GPIOLIB gpio: mxs: Use set and clear capabilities of the gpio controller Miao Xie (10): Btrfs: don't abort the current transaction if there is no enough space for inode cache Btrfs: don't use global block reservation for inode cache truncation Btrfs: optimize the error handle of use_block_rsv() Btrfs: don't steal the reserved space from the global reserve if their space type is different Btrfs: update the global reserve if it is empty Btrfs: fix accessing a freed tree root Btrfs: fix unprotected root node of the subvolume's inode rb-tree Btrfs: pause the space balance when remounting to R/O Btrfs: remove BUG_ON() in btrfs_read_fs_tree_no_radix() Btrfs: don't invoke btrfs_invalidate_inodes() in the spin lock context Michael Chan (1): tg3: Fix data corruption on 5725 with TSO Michael Ellerman (2): powerpc: Make CONFIG_RTAS_PROC depend on CONFIG_PROC_FS powerpc: Context switch more PMU related SPRs Michael Grzeschik (1): usb: otg: mxs-phy: add missing type to usb_phy Michael Neuling (3): powerpc/perf: Move BHRB code into CONFIG_PPC64 region powerpc/pmu: Fix order of interpreting BHRB target entries powerpc/perf: Fix setting of "to" addresses for BHRB Michael S. Tsirkin (1): virtio_console: fix uapi header Michał Mirosław (1): staging/iio/mxs-lradc: fix preenable for multiple buffers Mika Westerberg (2): i2c: designware: always clear interrupts before enabling them i2c: designware: add Intel BayTrail ACPI ID Nandita Dukkipati (1): tcp: bug fix in proportional rate reduction. Naveen Krishna Chatradhi (1): iio: exynos_adc: fix wrong structure extration in suspend and resume Nicholas Bellinger (5): iscsi-target: Fix NULL pointer dereference in iscsit_send_reject target/rd: Add ramdisk bit for NULLIO operation iscsi-target: Fix typos in RDMAEXTENSIONS macro usage MAINTAINERS: Update target git tree URL target/iblock: Fix WCE=1 + DPOFUA=1 backend WRITE regression Nicolas Ferre (2): ARM: at91/trivial: typo in GEM compatible string net/macb: fix ISR clear-on-write behavior only for some SoC Nicolas Pitre (1): ARM: 7715/1: MCPM: adapt to GIC changes after upstream merge Nicolas Schichan (4): ASoC: cs42l52: microphone bias is controlled by IFACE_CTL2 register. ASoC: cs42l52: fix bogus shifts in "Speaker Volume" and "PCM Mixer Volume" controls. ASoC: cs42l52: fix master playback mute mask. ASoC: cs42l52: fix default value for MASTERA_VOL. Niels Ole Salscheider (2): drm/radeon: Remove superfluous variable drm/radeon: Fix VRAM size calculation for VRAM >= 4GB Nishanth Menon (1): cpufreq: cpufreq-cpu0: defer probe when regulator is not ready Nithin Sujir (3): tg3: Skip powering down function 0 on certain serdes devices tg3: Ensure boot code has completed initialization before accessing hardware tg3: Update version to 3.132 OGAWA Hirofumi (1): fat: fix possible overflow for fat_clusters Oleg Nesterov (1): usermodehelper: check subprocess_info->path != NULL Oliver Neukum (1): USB: reset resume quirk needed by a hub Olof Johansson (10): Merge tag 'fixes-3.10-1' of git://git.infradead.org/users/jcooper/linux into fixes Merge tag 'at91-fixes' of git://github.com/at91linux/linux-at91 into fixes Merge tag 'omap-for-v3.10-rc1/fixes' of git://git.kernel.org/.../tmlind/linux-omap into fixes Merge tag 'fixes-3.10-2' of git://git.infradead.org/users/jcooper/linux into fixes Merge tag 'imx-fixes-3.10' of git://git.linaro.org/people/shawnguo/linux-2.6 into fixes Merge tag 'ux500-arm-soc-v3.10-fixes' of git://git.kernel.org/.../linusw/linux-stericsson into fixes Merge tag 'omap-fixes-a-for-3.10-rc' of git://git.kernel.org/.../pjw/omap-pending into fixes Merge tag 'renesas-boards-marzen-fixes-for-v3.10' of git://git.kernel.org/.../horms/renesas into fixes Merge tag 'at91-fixes' of git://github.com/at91linux/linux-at91 into fixes Merge tag 'sunxi-fixes-for-3.10' of git://github.com/mripard/linux into fixes Or Gerlitz (1): net/mlx4_core: Add missing report on VST and spoof-checking dev caps Pablo Neira Ayuso (3): netfilter: nf_{log,queue}: fix compilation without CONFIG_PROC_FS netfilter: update MAINTAINERS file netfilter: xt_TCPOPTSTRIP: fix possible mangling beyond packet boundary Paolo Bonzini (1): KVM: take over co-maintainership from Marcelo, fix MAINTAINERS entry Paul Bolle (1): usb: phy: remove CONFIG_USB_OTG_UTILS once more Paul E. McKenney (2): kthread: Document ways of reducing OS jitter due to per-CPU kthreads rcu: Fix comparison sense in rcu_needs_cpu() Paul Moore (1): netlabel: improve domain mapping validation Paul Walmsley (1): ARM: OMAP2+: Add missing CONFIG_ARCH_MULTI_V6=y to omap2plus_defconfig Paul Zimmerman (1): staging: dwc2: remove compile warning for USB_DWC2_TRACK_MISSED_SOFS Peter Huewe (1): parisc/superio: Use module_pci_driver to register driver Peter Hurley (1): tty/vt: Fix vc_deallocate() lock order Peter Oberparleiter (1): s390/cio: add channel ID sysfs attribute Petko Manolov (1): drivers: net: usb: rtl8150: concurrent URB bugfix Petri Gynther (1): emac: Fix EMAC soft reset on 460EX/GT Philipp Zabel (1): ARM i.MX6q: Fix periph_clk2_sel and periph2_clk2_sel clocks Ping Cheng (3): Input: wacom - add three new display tablets Input: wacom - add a few new styli for Cintiq series Input: wacom - add an eraser to DTH2242/DTK2241 Rafael J. Wysocki (10): ACPI / PM: Move processor suspend/resume to syscore_ops ACPI / LPSS: register clock device for Lynxpoint DMA properly Merge branch 'pm-cpufreq' Merge branch 'acpi-fixes' Merge branch 'pm-fixes' Merge branch 'pm-cpufreq' Merge branch 'acpi-fixes' ACPI / PM: Allow device power states to be used for CONFIG_PM unset Merge branch 'acpi-fixes' Merge branch 'pm-cpufreq' Rafał Bilski (1): cpufreq / e_powersaver: Fix linker error when ACPI processor is a module Rafał Miłecki (1): bcma: add more core IDs Rahul Sharma (1): of/documentation: move video device bindings to a common place Rajendra Nayak (2): ARM: OMAP2+: hwmod: Fix sidle programming in _enable_sysc()/_idle_sysc() ARM: OMAP2+: hwmod: Add a new flag to handle SIDLE in SWSUP only in active Rajesh Borundia (2): qlcnic: Fix mailbox response handling. qlcnic: Fix validation of link event command. Ralf Baechle (13): MIPS: Rewrite pfn_valid to work in modules, too. Add include dependencies to <linux/printk.h>. MIPS: clock.h: Remove declaration of cpu_wait. MIPS: Consolidate idle loop / WAIT instruction support in a single file. MIPS: Idle: cleaup SMTC idle hook as per Linux coding style. MIPS: Idle: Consistently reformat inline assembler. MIPS: Idle: Make call of function pointer readable. MIPS: Idle: Re-enable irqs at the end of r3081, au1k and loongson2 cpu_wait. MIPS: Idle: Don't call local_irq_disable() in cpu_wait() implementations. MIPS: Idle: Consolidate all declarations in <asm/idle.h>. MIPS: Idle: Do address fiddlery in helper functions. MIPS: Idle: Break r4k_wait into two functions and fix it. mm: Fix virt_to_page() warning Randy Dunlap (7): lp8788-charger: Fix kconfig dependency staging: sep: fix driver build and kconfig lib: make iovec obj instead of lib GFS2: fix DLM depends to fix build errors linux/kernel.h: fix kernel-doc warning auditfilter.c: fix kernel-doc warnings mm/memory_hotplug.c: fix printk format warnings Richard Genoud (1): spi/spi-atmel: BUG: fix doesn' support 16 bits transfers using PIO Rob Herring (1): Merge tag 'tegra-for-3.10-dtc-cpp-chroot-std-headers' of git://git.kernel.org/.../swarren/linux-tegra into for-next Robert Jarzmik (1): usb: phy: Fix NULL pointer exception during usb_get_phy Robert Jennings (1): powerpc: Bring all threads online prior to migration/hibernation Robert P. J. Day (2): Correct typo "supperspeed" to "superspeed". Add a couple kernel-doc lines to prevent warnings. Rony Efraim (1): net/mlx4: Strengthen VLAN tags/priorities enforcement in VST mode Russell King (3): ARM: finally enable IRQ time accounting config NET: mv643xx_eth: avoid lockdep dump on interface down i2c: mv64xxx: work around signals causing I2C transactions to be aborted Rusty Russell (2): lguest: clear cached last cpu when guest_set_pgd() called. Hoist memcpy_fromiovec/memcpy_toiovec into lib/ Ryusuke Konishi (1): nilfs2: fix issue of nilfs_set_page_dirty() for page at EOF boundary Sachin Kamat (12): usb: gadget: atmel_usba_udc: Remove redundant platform_set_drvdata() usb: gadget: bcm63xx_udc: Remove redundant platform_set_drvdata() usb: gadget: dummy_hcd: Remove redundant platform_set_drvdata() usb: gadget: f_uac2: Remove redundant platform_set_drvdata() usb: gadget: imx_udc: Remove redundant platform_set_drvdata() usb: gadget: pxa25x_udc: Remove redundant platform_set_drvdata() usb: gadget: s3c2410_udc: Remove redundant platform_set_drvdata() usb: phy: ab8500-usb: Remove redundant platform_set_drvdata() usb: phy: gpio-vbus-usb: Remove redundant platform_set_drvdata() usb: phy: mv-usb: Remove redundant platform_set_drvdata() usb: phy: mxs-usb: Remove redundant platform_set_drvdata() usb: phy: nop: Remove redundant platform_set_drvdata() Samuel Ortiz (1): mei: bus: Reset event_cb when disabling a device Sanjay Lal (2): KVM/MIPS32: Move include/asm/kvm.h => include/uapi/asm/kvm.h since it is a user visible API. KVM/MIPS32: Wrap calls to gfn_to_pfn() with srcu_read_lock/unlock() Santosh Shilimkar (4): ARM: OMAP2+: hwmod-data: UART IP needs software control to manage sidle modes ARM: OMAP2+: serial: Remove the un-used slave idle hooks SERIAL: OMAP: Remove the slave idle handling from the driver ARM: OMAP2+: hwmod: Remove sysc slave idle and auto idle apis Sarveshwar Bandi (1): be2net: Avoid double insertion of vlan tags. Sasha Levin (2): rcu: Don't allocate bootmem from rcu_init() mac80211_hwsim: correctly register the platform driver Schemmel Hans-Christoph (1): USB: Blacklisted Cinterion's PLxx WWAN Interface Scott Wood (2): powerpc: hard_irq_disable(): Call trace_hardirqs_off after disabling powerpc/booke64: Fix kernel hangs at kernel_dbg_exc Sebastian Hesselbarth (2): ARM: 7711/1: dove: fix Dove cpu type from V7 to PJ4 ARM: plat-orion: add missing ehci include to common.h Sergei Shtylyov (2): <linux/of_platform.h>: fix compilation warnings with DT disabled 3c59x: fix PCI resource management Shahed Shaikh (2): qlcnic: Fix ethtool strings qlcnic: Fix updating netdev->features Shawn Guo (4): net: fec: enable hardware checksum only on imx6q-fec ARM: imx: replicate the diagnostic register of boot cpu into secondary cores ARM: imx: fix typo in gpu3d_shader_sels net: fec: use a more proper compatible string for MVF type device Shlomo Pongratz (1): iscsi-target: Fix processing of OOO commands Shuah Khan (1): PM: Avoid calling kfree() under spinlock in dev_pm_put_subsys_data() Simon Horman (1): ARM: shmobile: marzen: Use error values in usb_power_* Sony Chacko (3): qlcnic: Fix reset recovery after transmit timeout qlcnic: Return proper error codes from probe failure paths qlcnic: remove netdev->trans_start updates within the driver Sridhar Samudrala (1): vxlan: Update vxlan fdb 'used' field after each usage Srinivas Kandagatla (1): ARM: vt8500: Add missing NULL terminator in dt_compat Srivatsa S. Bhat (2): rcu/idle: Wrap cpu-idle poll mode within rcu_idle_enter/exit cpufreq: Preserve sysfs files across suspend/resume Stanislaw Gruszka (2): ath5k: do not reschedule tx_complete_work on stop iwl4965: workaround connection regression on passive channel Stefan Behrens (3): Btrfs: don't allow device replace on RAID5/RAID6 Btrfs: fix possible memory leak in replace_path() Btrfs: explicitly use global_block_rsv for quota_tree Stefano Stabellini (3): xen/arm: initialize pm functions later xen/arm: do not handle VCPUOP_register_vcpu_info failures xen/arm: rename xen_secondary_init and run it on every online cpu Stephen Boyd (1): ARM: smp: Drop RCU_NONIDLE usage in cpu_die() Stephen Warren (2): USB: set device dma_mask without reference to global data drivers/video: implement a simple framebuffer driver Steven Capper (1): ARM: 7727/1: remove the .vm_mm value from gate_vma Steven Rostedt (2): kmemleak: Scan all allocated, writeable and not executable module sections kmemleak: No need for scanning specific module sections Steven Rostedt (Red Hat) (1): tracing: Fix leaks of filter preds Steven Whitehouse (1): GFS2: Fix typo in gfs2_log_end_write loop Sujith Manoharan (4): ath9k: Fix beacon reconfiguration ath9k: Update initvals for AR9565 ath9k: Fix crash on module unload ath9k_hw: Enable manual peak calibration for AR9485 Sukanto Ghosh (1): arm64: mm: Fix operands of clz in __flush_dcache_all Tang Yuantian (1): of/base: release the node correctly in of_parse_phandle_with_args() Tejun Heo (2): workqueue: workqueue_congested() shouldn't translate WORK_CPU_UNBOUND into node number workqueue: don't perform NUMA-aware allocations on offline nodes in wq_numa_init() Teppo Kotilainen (1): usb: option: Add Telewell TW-LTE 4G Thadeu Lima de Souza Cascardo (1): qlge: fix dma map leak when the last chunk is not allocated Theodore Ts'o (1): ext4: revert "ext4: use io_end for multiple bios" Thomas Gleixner (5): Merge branch 'linus' into core/urgent tick: Use zalloc_cpumask_var for allocating offstack cpumasks x86: Fix idle consolidation fallout tick: Cleanup NOHZ per cpu data on cpu down tick: Don't invoke tick_nohz_stop_sched_tick() if the cpu is offline Thomas Petazzoni (2): ARM: mvebu: do not duplicate the mpic alias arm: mvebu: fix the 'ranges' property to handle PCIe Thommy Jakobsson (1): B43: Handle DMA RX descriptor underrun Timo Teräs (2): leds: leds-gpio: reserve gpio before using it xfrm: properly handle invalid states as an error Tirupathi Reddy (1): timer: Don't reinitialize the cpu base lock during CPU_UP_PREPARE Tomas Winkler (1): mei: fix out of array access to me clients array Tomasz Figa (1): drivers/rtc/rtc-max8998.c: check for pdata presence before dereferencing Tony Lindgren (1): ARM: dts: Fix musb interrupt for device tree booting Tony Prisk (1): pinctrl: vt8500: Fix incorrect data in WM8750 pinctrl table Tony Wu (2): MIPS: Fix sibling call handling in get_frame_info MIPS: Extract schedule_mfi info from __schedule Trond Myklebust (4): SUNRPC: Fix a bug in gss_create_upcall SUNRPC: Faster detection if gssd is actually running SUNRPC: Convert auth_gss pipe detection to work in namespaces SUNRPC: Prevent an rpc_task wakeup race Ulf Hansson (1): mfd: db8500-prcmu: Update stored DSI PLL divider value Uwe Kleine-König (1): Documentation/devicetree: make semantic of initrd-end more explicit Vaibhav Hiremath (1): ARM: AM33XX: Add missing .clkdm_name to clkdiv32k_ick clock Vasant Hegde (1): powerpc/rtas_flash: Fix validate_flash buffer overflow issue Ville Syrjälä (3): drm/i915: Fix pipe enabled mask for pipe C in WM calculations drm: Add kernel-doc for drm_fb_helper_funcs->initial_config drm: Remove pointless '-' characters from drm_fb_helper documentation Vineet Gupta (5): ARC: [mm] Prevent stray dcache lines after__sync_icache_dcach() ARC: copy_(to|from)_user() to honor usermode-access permissions ARC: Brown paper bag bug in macro for checking cache color ARC: Use enough bits for determining page's cache color ARC: lazy dcache flush broke gdb in non-aliasing configs Vinod Koul (2): ASoC: wm8994: use the correct pointer to get the control value ASoC: wm8994: check for array index returned Viresh Kumar (12): cpufreq: ARM big LITTLE: Select PM_OPP cpufreq: ARM big LITTLE DT: Return correct transition latency cpufreq: ARM big LITTLE DT: Return CPUFREQ_ETERNAL if clock-latency isn't found cpufreq: ARM big LITTLE: Move cpu_to_cluster() to arm_big_little.h cpufreq: ARM big LITTLE: Improve print message cpufreq: governors: Fix CPUFREQ_GOV_POLICY_{INIT|EXIT} notifiers cpufreq: Issue CPUFREQ_GOV_POLICY_EXIT notifier before dropping policy refcount cpufreq: cpufreq-cpu0: Free parent node for error cases cpufreq: ARM big LITTLE: Fix Kconfig entries cpufreq: Drop rwsem lock around CPUFREQ_GOV_POLICY_EXIT cpufreq: arm_big_little_dt: Register driver only if DT has valid data cpufreq: arm_big_little_dt: Instantiate as platform_driver Vivek Gautam (1): usb: dwc3: Fix compilation break when building with USB_DWC3_DUAL_ROLE=y Wang Shilong (1): Btrfs: fix possible memory leak in the find_parent_nodes() Wang YanQing (1): TTY: Fix tty miss restart after we turn off flow-control Wei WANG (1): USB: usb-stor: realtek_cr: Fix compile error Wei Yang (1): be2net: bug fix on returning an invalid nic descriptor Wei Yongjun (20): hwmon: fix error return code in abituguru_probe() bna: add missing iounmap() on error in bnad_init() cpufreq / intel_pstate: use vzalloc() instead of vmalloc()/memset(0) pinctrl: single: fix error return code in pcs_parse_one_pinctrl_entry() usb: gadget: zero: fix error return code in zero_bind() usb: musb: omap2430: add missing platform_device_put() on error in omap2430_probe() usb: musb: dsps: fix error return code in dsps_create_musb_pdev() usb: gadget: s3c2410_udc: fix error return code in s3c2410_udc_probe() usb: gadget: r8a66597-udc: fix error return code in r8a66597_probe() usb: gadget: m66592-udc: fix error return code in m66592_probe() usb: gadget: fusb300_udc: fix error return code in fusb300_probe() usb: gadget: dummy_hcd: fix error return code in init() usb: gadget: fix error return code in configfs_composite_bind() ARM: OMAP: fix error return code in omap1_system_dma_init() gianfar: add missing iounmap() on error in gianfar_ptp_probe() serial: samsung: add missing platform_driver_unregister() when module exit tty: serial: mpc5xxx: fix error handing in mpc52xx_uart_init() serial: mcf: missing uart_unregister_driver() on error in mcf_init() TTY: ehv_bytechan: add missing platform_driver_unregister() when module exit staging:iio:light:tsl2x7x: fix the error handling in tsl2x7x_probe() Will Deacon (3): arm64: debug: clear mdscr_el1 instead of taking the OS lock arm64: debug: fix mdscr.ss check when enabling debug exceptions ARM: 7729/1: vfp: ensure VFP_arch is non-zero when VFP is not supported Wolfram Sang (36): cpufreq / kirkwood: don't check resource with devm_ioremap_resource drivers/ata: don't check resource with devm_ioremap_resource drivers/char/hw_random: don't check resource with devm_ioremap_resource usb: gadget: don't check resource with devm_ioremap_resource usb: phy: don't check resource with devm_ioremap_resource drivers/cpufreq: don't check resource with devm_ioremap_resource drivers/dma: don't check resource with devm_ioremap_resource drivers/gpio: don't check resource with devm_ioremap_resource drivers/gpu/drm/exynos: don't check resource with devm_ioremap_resource drivers/gpu/host1x/drm: don't check resource with devm_ioremap_resource drivers/i2c/busses: don't check resource with devm_ioremap_resource drivers/memory: don't check resource with devm_ioremap_resource drivers/mfd: don't check resource with devm_ioremap_resource drivers/misc: don't check resource with devm_ioremap_resource drivers/mtd/nand: don't check resource with devm_ioremap_resource drivers/net/ethernet/renesas: don't check resource with devm_ioremap_resource drivers/pinctrl: don't check resource with devm_ioremap_resource drivers/pwm: don't check resource with devm_ioremap_resource drivers/rtc: don't check resource with devm_ioremap_resource drivers/spi: don't check resource with devm_ioremap_resource drivers/staging/dwc2: don't check resource with devm_ioremap_resource drivers/staging/nvec: don't check resource with devm_ioremap_resource drivers/thermal: don't check resource with devm_ioremap_resource drivers/usb/chipidea: don't check resource with devm_ioremap_resource drivers/usb/gadget: don't check resource with devm_ioremap_resource drivers/usb/host: don't check resource with devm_ioremap_resource drivers/usb/phy: don't check resource with devm_ioremap_resource drivers/video/omap2: don't check resource with devm_ioremap_resource drivers/video/omap2/dss: don't check resource with devm_ioremap_resource drivers/w1/masters: don't check resource with devm_ioremap_resource drivers/watchdog: don't check resource with devm_ioremap_resource arch/arm/mach-tegra: don't check resource with devm_ioremap_resource arch/arm/plat-samsung: don't check resource with devm_ioremap_resource arch/mips/lantiq/xway: don't check resource with devm_ioremap_resource sound/soc/fsl: don't check resource with devm_ioremap_resource sound/soc/kirkwood: don't check resource with devm_ioremap_resource Xiao Guangrong (1): mm: mmu_notifier: re-fix freed page still mapped in secondary MMU Xiong Zhou (2): bq27x00: Fix I2C dependency in KConfig staging: android: logger: use kuid_t instead of uid_t Yan, Zheng (1): ext4: fix fio regression Yinghai Lu (1): PCI: acpiphp: Re-enumerate devices when host bridge receives Bus Check Yuval Mintz (1): bnx2x: Fix GSO for 57710/57711 chips Zhang Rui (2): PM: Documentation update for freeze state PM / Documentation: remove inaccurate suspend/hibernate transition lantency statement Zhang Yanfei (1): x86/mm: Add missing comments for initial kernel direct mapping dyoung(a)redhat.com (1): driver core: print sysfs attribute name when warning about bogus permissions françois romieu (1): r8169: fix offloaded tx checksum for small packets. jean-philippe francois (1): ARM: OMAP: fix __init section mismatch for _enable_preprogram nikolay(a)redhat.com (4): bonding: fix set mode race conditions bonding: replace %x with %pI4 for IPv4 addresses bonding: arp_ip_count and arp_targets can be wrong bonding: fix multiple 3ad mode sysfs race conditions salina(a)us.ibm.com (1): Char: lp, protect LPGETSTATUS with port_mutex stephen hemminger (1): 8139cp: reset BQL when ring tx ring cleared wang, biao (1): klist: del waiter from klist_remove_waiters before wakeup waitting process Documentation/devicetree/bindings/net/macb.txt | 2 +- .../{drm/exynos/hdmi.txt => video/exynos_hdmi.txt} | 0 .../hdmiddc.txt => video/exynos_hdmiddc.txt} | 0 .../hdmiphy.txt => video/exynos_hdmiphy.txt} | 0 .../exynos/mixer.txt => video/exynos_mixer.txt} | 0 .../bindings/video/simple-framebuffer.txt | 25 + Documentation/devicetree/usage-model.txt | 8 +- Documentation/kernel-parameters.txt | 21 + Documentation/kernel-per-CPU-kthreads.txt | 202 +++++ Documentation/power/devices.txt | 15 +- Documentation/power/interface.txt | 4 +- Documentation/power/notifiers.txt | 6 +- Documentation/power/states.txt | 30 +- Documentation/rapidio/rapidio.txt | 128 ++- Documentation/rapidio/sysfs.txt | 17 + MAINTAINERS | 43 +- Makefile | 2 +- arch/Kconfig | 3 + arch/arc/boot/dts/abilis_tb100_dvk.dts | 2 +- arch/arc/boot/dts/abilis_tb101_dvk.dts | 2 +- arch/arc/boot/dts/abilis_tb10x.dtsi | 6 +- arch/arc/include/asm/cacheflush.h | 6 +- arch/arc/include/asm/page.h | 9 - arch/arc/include/asm/pgtable.h | 26 +- arch/arc/include/asm/tlb.h | 2 +- arch/arc/mm/cache_arc700.c | 23 +- arch/arc/mm/tlb.c | 3 +- arch/arc/mm/tlbex.S | 6 +- arch/arc/plat-tb10x/tb10x.c | 26 - arch/arm/Kconfig | 3 +- arch/arm/Makefile | 2 +- arch/arm/boot/dts/Makefile | 4 +- arch/arm/boot/dts/armada-370-xp.dtsi | 3 +- arch/arm/boot/dts/armada-370.dtsi | 7 +- arch/arm/boot/dts/armada-xp-gp.dts | 3 + arch/arm/boot/dts/armada-xp-openblocks-ax3-4.dts | 3 + arch/arm/boot/dts/armada-xp.dtsi | 2 +- arch/arm/boot/dts/at91sam9260.dtsi | 2 +- arch/arm/boot/dts/at91sam9n12.dtsi | 1 + arch/arm/boot/dts/at91sam9x25ek.dts | 2 +- arch/arm/boot/dts/omap3.dtsi | 2 +- arch/arm/boot/dts/sama5d3.dtsi | 12 +- arch/arm/boot/dts/sama5d3xcm.dtsi | 4 + arch/arm/boot/dts/ste-nomadik-s8815.dts | 12 +- arch/arm/boot/dts/sun4i-a10-mini-xplus.dts | 4 +- arch/arm/common/mcpm_platsmp.c | 3 - arch/arm/configs/omap1_defconfig | 1 - arch/arm/configs/omap2plus_defconfig | 2 +- arch/arm/configs/tegra_defconfig | 3 +- arch/arm/crypto/sha1-armv4-large.S | 2 +- arch/arm/include/asm/cmpxchg.h | 8 +- arch/arm/include/debug/ux500.S | 6 +- arch/arm/kernel/process.c | 1 - arch/arm/kernel/smp.c | 2 +- arch/arm/mach-at91/at91rm9200_time.c | 7 +- arch/arm/mach-at91/at91sam9n12.c | 6 - arch/arm/mach-at91/include/mach/at91_pmc.h | 6 +- arch/arm/mach-imx/clk-imx6q.c | 11 +- arch/arm/mach-imx/headsmp.S | 12 + arch/arm/mach-imx/platsmp.c | 14 + arch/arm/mach-kirkwood/common.c | 6 - arch/arm/mach-kirkwood/ts219-setup.c | 2 +- arch/arm/mach-mvebu/Kconfig | 1 + arch/arm/mach-mvebu/armada-370-xp.c | 7 - arch/arm/mach-omap1/dma.c | 1 + arch/arm/mach-omap2/cclock33xx_data.c | 26 +- arch/arm/mach-omap2/omap_hwmod.c | 113 +-- arch/arm/mach-omap2/omap_hwmod.h | 7 +- arch/arm/mach-omap2/omap_hwmod_2xxx_ipblock_data.c | 3 + arch/arm/mach-omap2/omap_hwmod_33xx_data.c | 6 + arch/arm/mach-omap2/omap_hwmod_3xxx_data.c | 4 + arch/arm/mach-omap2/omap_hwmod_44xx_data.c | 6 +- arch/arm/mach-omap2/serial.c | 31 - arch/arm/mach-orion5x/common.c | 7 - arch/arm/mach-shmobile/board-marzen.c | 6 +- arch/arm/mach-sunxi/Kconfig | 1 + arch/arm/mach-tegra/tegra2_emc.c | 5 - arch/arm/mach-ux500/Kconfig | 1 + arch/arm/mach-ux500/board-mop500.c | 6 +- arch/arm/mach-ux500/cpu-db8500.c | 6 +- arch/arm/mach-ux500/setup.h | 2 +- arch/arm/mach-vt8500/vt8500.c | 1 + arch/arm/plat-orion/common.c | 12 +- arch/arm/plat-orion/include/plat/common.h | 1 + arch/arm/plat-samsung/adc.c | 5 - arch/arm/vfp/entry.S | 2 +- arch/arm/xen/enlighten.c | 33 +- arch/arm64/Kconfig | 2 - arch/arm64/include/asm/assembler.h | 2 +- arch/arm64/kernel/debug-monitors.c | 2 - arch/arm64/kernel/early_printk.c | 5 +- arch/arm64/kernel/setup.c | 12 +- arch/arm64/mm/cache.S | 2 +- arch/arm64/mm/proc.S | 3 +- arch/avr32/Kconfig | 5 + arch/avr32/include/asm/Kbuild | 1 + arch/avr32/include/asm/numnodes.h | 7 - arch/avr32/include/asm/param.h | 9 - arch/avr32/include/uapi/asm/Kbuild | 1 + arch/avr32/include/uapi/asm/param.h | 18 - arch/avr32/kernel/module.c | 2 +- arch/mips/alchemy/board-gpr.c | 1 + arch/mips/alchemy/common/time.c | 1 + arch/mips/ath79/setup.c | 1 + arch/mips/cobalt/reset.c | 1 + arch/mips/configs/db1000_defconfig | 1 - arch/mips/configs/db1235_defconfig | 1 - arch/mips/configs/lemote2f_defconfig | 1 - arch/mips/include/asm/clock.h | 2 - arch/mips/include/asm/idle.h | 23 + arch/mips/include/asm/io.h | 2 +- arch/mips/include/asm/kvm_host.h | 2 +- arch/mips/include/asm/mmu_context.h | 95 +- arch/mips/include/asm/page.h | 17 +- arch/mips/include/asm/processor.h | 1 - arch/mips/include/{ => uapi}/asm/kvm.h | 0 arch/mips/include/uapi/asm/unistd.h | 5 +- arch/mips/kernel/Makefile | 2 +- arch/mips/kernel/cpu-probe.c | 198 ----- arch/mips/kernel/crash_dump.c | 1 + arch/mips/kernel/genex.S | 8 +- arch/mips/kernel/idle.c | 244 +++++ arch/mips/kernel/kprobes.c | 5 +- arch/mips/kernel/proc.c | 1 + arch/mips/kernel/process.c | 53 +- arch/mips/kernel/scall64-64.S | 1 + arch/mips/kernel/smp.c | 1 + arch/mips/kernel/smtc.c | 15 +- arch/mips/kernel/traps.c | 15 +- arch/mips/kvm/kvm_mips_emul.c | 29 +- arch/mips/kvm/kvm_tlb.c | 61 +- arch/mips/lantiq/xway/gptu.c | 6 +- arch/mips/lib/dump_tlb.c | 5 +- arch/mips/lib/r3k_dump_tlb.c | 7 +- arch/mips/loongson/common/reset.c | 1 + arch/mips/loongson1/common/reset.c | 1 + arch/mips/mm/tlb-r3k.c | 20 +- arch/mips/mm/tlb-r4k.c | 2 +- arch/mips/mm/tlb-r8k.c | 2 +- arch/mips/mm/tlbex.c | 79 -- arch/mips/netlogic/xlp/setup.c | 1 + arch/mips/netlogic/xlr/setup.c | 1 + arch/mips/pmcs-msp71xx/msp_prom.c | 2 +- arch/mips/pmcs-msp71xx/msp_setup.c | 1 + arch/mips/ralink/dts/rt3050.dtsi | 10 + arch/mips/ralink/dts/rt3052_eval.dts | 4 + arch/mips/txx9/generic/setup.c | 1 + arch/mips/vr41xx/common/pmu.c | 1 + arch/mips/wrppmc/reset.c | 1 + arch/parisc/Kconfig | 2 +- arch/parisc/Makefile | 21 +- arch/parisc/include/asm/assembly.h | 1 - arch/parisc/include/asm/hardirq.h | 10 +- arch/parisc/include/asm/processor.h | 18 - arch/parisc/kernel/entry.S | 174 ++-- arch/parisc/kernel/hardware.c | 1 + arch/parisc/kernel/irq.c | 122 ++- arch/parisc/kernel/pacache.S | 12 +- arch/parisc/kernel/traps.c | 1 + arch/parisc/kernel/unaligned.c | 3 + arch/parisc/mm/init.c | 4 +- arch/powerpc/Kconfig.debug | 23 + arch/powerpc/configs/ps3_defconfig | 1 - arch/powerpc/include/asm/context_tracking.h | 10 + arch/powerpc/include/asm/firmware.h | 4 +- arch/powerpc/include/asm/hw_irq.h | 5 +- arch/powerpc/include/asm/opal.h | 5 +- arch/powerpc/include/asm/pci-bridge.h | 2 + arch/powerpc/include/asm/pgalloc-64.h | 2 +- arch/powerpc/include/asm/processor.h | 6 + arch/powerpc/include/asm/pte-hash64-64k.h | 2 +- arch/powerpc/include/asm/rtas.h | 2 + arch/powerpc/include/asm/thread_info.h | 7 +- arch/powerpc/include/asm/udbg.h | 1 + arch/powerpc/kernel/asm-offsets.c | 6 + arch/powerpc/kernel/cpu_setup_power.S | 8 +- arch/powerpc/kernel/entry_32.S | 2 - arch/powerpc/kernel/entry_64.S | 33 +- arch/powerpc/kernel/exceptions-64e.S | 8 +- arch/powerpc/kernel/machine_kexec_64.c | 4 + arch/powerpc/kernel/misc_32.S | 11 + arch/powerpc/kernel/misc_64.S | 11 + arch/powerpc/kernel/pci-common.c | 12 +- arch/powerpc/kernel/pci_64.c | 10 + arch/powerpc/kernel/pci_dn.c | 8 + arch/powerpc/kernel/ppc_ksyms.c | 3 +- arch/powerpc/kernel/process.c | 8 + arch/powerpc/kernel/ptrace.c | 5 + arch/powerpc/kernel/rtas.c | 113 +++ arch/powerpc/kernel/rtas_flash.c | 10 +- arch/powerpc/kernel/signal.c | 7 +- arch/powerpc/kernel/traps.c | 80 +- arch/powerpc/kernel/udbg.c | 3 + arch/powerpc/mm/fault.c | 41 +- arch/powerpc/mm/hash_utils_64.c | 36 +- arch/powerpc/mm/init_64.c | 3 +- arch/powerpc/perf/core-book3s.c | 280 +++--- arch/powerpc/platforms/Kconfig | 2 +- arch/powerpc/platforms/powernv/Kconfig | 1 + arch/powerpc/platforms/powernv/opal.c | 30 +- arch/powerpc/platforms/powernv/pci-ioda.c | 36 +- arch/powerpc/platforms/powernv/pci.c | 18 +- arch/powerpc/platforms/powernv/pci.h | 2 + arch/powerpc/platforms/powernv/powernv.h | 2 + arch/powerpc/platforms/powernv/setup.c | 16 +- arch/powerpc/platforms/powernv/smp.c | 62 +- arch/powerpc/platforms/pseries/Kconfig | 1 + arch/powerpc/platforms/pseries/msi.c | 75 +- arch/powerpc/platforms/pseries/suspend.c | 22 + arch/powerpc/platforms/wsp/ics.c | 2 +- arch/powerpc/sysdev/Makefile | 2 + arch/powerpc/sysdev/ehv_pic.c | 2 +- arch/powerpc/sysdev/mpic.c | 2 +- arch/powerpc/sysdev/udbg_memcons.c | 105 +++ arch/powerpc/sysdev/xics/ics-opal.c | 2 +- arch/s390/Kconfig | 1 - arch/s390/include/asm/ftrace.h | 12 +- arch/s390/include/asm/page.h | 20 +- arch/s390/include/asm/pgtable.h | 4 +- arch/s390/kernel/dis.c | 2 + arch/s390/kernel/ftrace.c | 9 +- arch/s390/kernel/mcount.S | 2 + arch/s390/kernel/mcount64.S | 2 + arch/s390/kernel/smp.c | 2 +- arch/s390/mm/pgtable.c | 3 +- arch/score/mm/init.c | 2 - arch/x86/Kconfig | 1 - arch/x86/kernel/head64.c | 2 +- arch/x86/kernel/microcode_intel_early.c | 5 +- arch/x86/kernel/process.c | 5 +- arch/x86/mm/init.c | 19 +- arch/x86/pci/mrst.c | 10 +- drivers/acpi/Makefile | 3 +- drivers/acpi/ac.c | 33 + drivers/acpi/acpi_lpss.c | 26 +- drivers/acpi/csrt.c | 159 ---- drivers/acpi/device_pm.c | 126 +-- drivers/acpi/ec.c | 4 +- drivers/acpi/internal.h | 1 - drivers/acpi/pci_root.c | 4 +- drivers/acpi/processor_driver.c | 8 +- drivers/acpi/processor_idle.c | 29 +- drivers/acpi/scan.c | 4 +- drivers/acpi/video.c | 8 + drivers/acpi/video_detect.c | 8 + drivers/ata/pata_ep93xx.c | 5 - drivers/base/bus.c | 1 + drivers/base/core.c | 6 +- drivers/base/power/common.c | 12 +- drivers/bcma/scan.c | 2 + drivers/block/brd.c | 4 +- drivers/block/rbd.c | 935 ++++++++++++-------- drivers/block/xsysace.c | 3 +- drivers/char/hw_random/mxc-rnga.c | 6 - drivers/char/hw_random/omap-rng.c | 5 - drivers/char/ipmi/ipmi_bt_sm.c | 4 +- drivers/char/ipmi/ipmi_devintf.c | 14 +- drivers/char/ipmi/ipmi_msghandler.c | 3 +- drivers/char/ipmi/ipmi_si_intf.c | 16 +- drivers/char/lp.c | 3 + drivers/char/random.c | 54 +- drivers/char/ttyprintk.c | 2 +- drivers/clk/tegra/clk-tegra20.c | 11 +- drivers/clk/x86/clk-lpt.c | 15 +- drivers/cpufreq/Kconfig | 2 +- drivers/cpufreq/Kconfig.arm | 15 +- drivers/cpufreq/Kconfig.x86 | 2 +- drivers/cpufreq/arm_big_little.c | 7 +- drivers/cpufreq/arm_big_little.h | 5 + drivers/cpufreq/arm_big_little_dt.c | 94 +- drivers/cpufreq/cpufreq-cpu0.c | 27 +- drivers/cpufreq/cpufreq.c | 19 +- drivers/cpufreq/cpufreq_governor.c | 11 +- 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drivers/usb/musb/omap2430.c | 3 +- drivers/usb/phy/Kconfig | 5 +- drivers/usb/phy/phy-ab8500-usb.c | 2 - drivers/usb/phy/phy-fsl-usb.c | 1 + drivers/usb/phy/phy-gpio-vbus-usb.c | 3 +- drivers/usb/phy/phy-isp1301.c | 1 + drivers/usb/phy/phy-mv-u3d-usb.c | 5 - drivers/usb/phy/phy-mv-usb.c | 3 - drivers/usb/phy/phy-mxs-usb.c | 8 +- drivers/usb/phy/phy-nop.c | 2 - drivers/usb/phy/phy-samsung-usb2.c | 5 - drivers/usb/phy/phy-samsung-usb3.c | 5 - drivers/usb/serial/ftdi_sio.c | 30 +- drivers/usb/serial/ftdi_sio_ids.h | 2 + drivers/usb/serial/generic.c | 31 + drivers/usb/serial/io_ti.c | 22 +- drivers/usb/serial/option.c | 13 +- drivers/usb/serial/ti_usb_3410_5052.c | 23 +- drivers/usb/serial/usb-serial.c | 30 +- drivers/usb/storage/realtek_cr.c | 8 +- drivers/vhost/vringh.c | 3 + drivers/video/Kconfig | 19 +- drivers/video/Makefile | 1 + drivers/video/console/Makefile | 2 + drivers/video/omap2/dss/hdmi.c | 4 - drivers/video/omap2/vrfb.c | 5 - drivers/video/simplefb.c | 234 +++++ drivers/w1/masters/omap_hdq.c | 5 - drivers/watchdog/ath79_wdt.c | 5 - drivers/watchdog/davinci_wdt.c | 5 - drivers/watchdog/imx2_wdt.c | 5 - drivers/xen/Kconfig | 7 +- drivers/xen/balloon.c | 3 +- drivers/xen/privcmd.c | 2 +- drivers/xen/tmem.c | 87 +- drivers/xen/xen-selfballoon.c | 47 +- drivers/xen/xenbus/xenbus_dev_backend.c | 21 +- fs/aio.c | 7 +- fs/btrfs/backref.c | 3 +- fs/btrfs/check-integrity.c | 2 +- fs/btrfs/ctree.c | 4 +- fs/btrfs/ctree.h | 8 +- fs/btrfs/delayed-ref.h | 1 + fs/btrfs/dev-replace.c | 5 + fs/btrfs/disk-io.c | 52 +- fs/btrfs/extent-tree.c | 94 +- fs/btrfs/extent_io.c | 138 +-- fs/btrfs/extent_io.h | 2 + fs/btrfs/free-space-cache.c | 43 +- fs/btrfs/free-space-cache.h | 2 + fs/btrfs/inode-map.c | 8 +- fs/btrfs/inode.c | 81 +- fs/btrfs/ioctl.c | 10 +- fs/btrfs/raid56.c | 2 +- fs/btrfs/relocation.c | 7 +- fs/btrfs/scrub.c | 10 +- fs/btrfs/super.c | 1 + fs/btrfs/volumes.c | 54 +- fs/btrfs/volumes.h | 20 + fs/cifs/inode.c | 3 +- fs/ext4/ext4.h | 8 +- fs/ext4/extents.c | 9 +- fs/ext4/extents_status.c | 17 +- fs/ext4/extents_status.h | 3 +- fs/ext4/file.c | 4 +- fs/ext4/inode.c | 85 +- fs/ext4/mballoc.c | 6 +- fs/ext4/page-io.c | 121 +-- fs/fat/inode.c | 15 +- fs/gfs2/Kconfig | 2 +- fs/gfs2/lops.c | 2 +- fs/gfs2/quota.c | 4 +- fs/gfs2/rgrp.c | 9 +- fs/hfs/bnode.c | 6 +- fs/nfs/callback_proc.c | 2 +- fs/nfs/callback_xdr.c | 2 +- fs/nfs/nfs4client.c | 2 +- fs/nfs/nfs4proc.c | 2 +- fs/nfs/nfs4session.c | 4 +- fs/nfs/nfs4session.h | 13 +- fs/nfs/nfs4state.c | 15 +- fs/nilfs2/inode.c | 27 +- fs/ocfs2/extent_map.c | 2 +- fs/ocfs2/file.c | 2 +- fs/xfs/xfs_aops.c | 19 + fs/xfs/xfs_attr_leaf.c | 27 +- fs/xfs/xfs_buf.c | 2 +- fs/xfs/xfs_da_btree.c | 7 +- fs/xfs/xfs_dir2_leaf.c | 2 +- fs/xfs/xfs_extfree_item.c | 5 +- fs/xfs/xfs_log_cil.c | 2 +- fs/xfs/xfs_vnodeops.c | 4 +- include/acpi/acpi_bus.h | 40 +- include/acpi/acpiosxf.h | 2 +- include/acpi/processor.h | 10 +- include/drm/drmP.h | 3 +- include/drm/drm_fb_helper.h | 15 +- include/drm/drm_os_linux.h | 9 - include/drm/drm_pciids.h | 6 + include/linux/acpi_dma.h | 4 + include/linux/bcma/bcma.h | 5 +- include/linux/brcmphy.h | 5 + include/linux/journal-head.h | 8 +- include/linux/kernel.h | 5 +- include/linux/kref.h | 33 + include/linux/mfd/abx500/ab8500.h | 2 - include/linux/mlx4/qp.h | 29 +- include/linux/netdevice.h | 11 + include/linux/of_platform.h | 5 +- include/linux/pci-acpi.h | 2 + include/linux/pinctrl/pinconf-generic.h | 12 +- include/linux/platform_data/clk-lpss.h | 5 + include/linux/platform_data/serial-omap.h | 2 - include/linux/printk.h | 1 + include/linux/rio.h | 18 +- include/linux/rio_drv.h | 1 + include/linux/socket.h | 2 - include/linux/spi/spi.h | 4 +- include/linux/time.h | 4 - include/linux/uio.h | 3 + include/linux/usb/gadget.h | 5 +- include/linux/usb/serial.h | 4 + include/linux/vt_kern.h | 2 +- include/linux/wait.h | 16 +- include/net/mac80211.h | 12 +- include/net/netfilter/nf_log.h | 3 +- include/net/netfilter/nfnetlink_log.h | 3 +- include/net/sock.h | 12 + include/target/target_core_base.h | 5 - include/trace/events/ext4.h | 4 +- include/uapi/linux/virtio_console.h | 2 +- ipc/sem.c | 27 +- kernel/auditfilter.c | 3 - kernel/cpu/idle.c | 2 + kernel/events/core.c | 240 ++--- kernel/kmod.c | 5 + kernel/module.c | 21 +- kernel/rcutree_plugin.h | 4 +- kernel/time/Kconfig | 5 - kernel/time/tick-broadcast.c | 10 +- kernel/time/tick-sched.c | 3 +- kernel/timer.c | 2 +- kernel/trace/trace_events.c | 4 +- kernel/trace/trace_events_filter.c | 4 + kernel/trace/trace_kprobe.c | 53 +- kernel/workqueue.c | 19 +- lib/Makefile | 2 +- lib/iovec.c | 53 ++ lib/klist.c | 2 +- lib/mpi/longlong.h | 5 +- mm/huge_memory.c | 7 +- mm/memcontrol.c | 14 +- mm/memory_hotplug.c | 9 +- mm/migrate.c | 2 +- mm/mmu_notifier.c | 79 +- mm/page_alloc.c | 2 +- mm/pagewalk.c | 70 +- net/802/mrp.c | 4 +- net/batman-adv/distributed-arp-table.c | 13 + net/batman-adv/main.c | 19 +- net/batman-adv/network-coding.c | 8 +- net/batman-adv/originator.c | 16 + net/batman-adv/originator.h | 1 + net/batman-adv/soft-interface.c | 1 + net/batman-adv/translation-table.c | 7 +- net/bridge/netfilter/ebt_log.c | 11 +- net/bridge/netfilter/ebt_ulog.c | 18 +- net/ceph/osd_client.c | 5 +- net/core/iovec.c | 50 -- net/core/sock.c | 12 - net/ipv4/ip_gre.c | 3 +- net/ipv4/ip_output.c | 2 +- net/ipv4/netfilter/ipt_ULOG.c | 13 +- net/ipv4/tcp.c | 29 +- net/ipv4/tcp_input.c | 23 +- net/ipv4/tcp_output.c | 10 +- net/ipv6/ip6_gre.c | 2 + net/ipv6/ip6_output.c | 2 +- net/ipv6/tcp_ipv6.c | 12 + net/ipv6/udp.c | 13 +- net/ipv6/udp_impl.h | 2 + net/ipv6/udplite.c | 2 +- net/ipv6/xfrm6_policy.c | 4 +- net/irda/irlap_frame.c | 2 +- net/mac80211/ieee80211_i.h | 1 + net/mac80211/mlme.c | 61 +- net/mac80211/rate.c | 9 +- net/mac80211/rx.c | 3 + net/mac80211/tkip.c | 4 +- net/mac80211/util.c | 7 + net/netfilter/nf_log.c | 7 +- net/netfilter/nfnetlink_log.c | 6 +- net/netfilter/nfnetlink_queue_core.c | 2 + net/netfilter/xt_LOG.c | 13 +- net/netfilter/xt_NFLOG.c | 3 +- net/netfilter/xt_TCPOPTSTRIP.c | 17 +- net/netlabel/netlabel_domainhash.c | 69 ++ net/sunrpc/auth_gss/auth_gss.c | 62 +- net/sunrpc/netns.h | 4 + net/sunrpc/rpc_pipe.c | 5 + net/sunrpc/sched.c | 8 +- net/wireless/core.c | 17 +- net/wireless/nl80211.c | 4 + net/wireless/sme.c | 3 +- net/wireless/trace.h | 23 +- net/xfrm/xfrm_output.c | 1 + scripts/package/Makefile | 2 +- sound/soc/codecs/cs42l52.c | 8 +- sound/soc/codecs/cs42l52.h | 2 +- sound/soc/codecs/max98090.c | 2 +- sound/soc/codecs/wm5110.c | 4 +- sound/soc/codecs/wm8994.c | 12 +- sound/soc/davinci/davinci-mcasp.c | 7 +- sound/soc/fsl/imx-ssi.c | 6 - sound/soc/kirkwood/kirkwood-i2s.c | 5 - sound/soc/soc-compress.c | 8 +- tools/perf/scripts/python/net_dropmonitor.py | 39 +- tools/testing/selftests/Makefile | 1 - tools/testing/selftests/soft-dirty/Makefile | 10 - tools/testing/selftests/soft-dirty/soft-dirty.c | 114 --- 841 files changed, 9100 insertions(+), 5717 deletions(-) rename Documentation/devicetree/bindings/{drm/exynos/hdmi.txt => video/exynos_hdmi.txt} (100%) rename Documentation/devicetree/bindings/{drm/exynos/hdmiddc.txt => video/exynos_hdmiddc.txt} (100%) rename Documentation/devicetree/bindings/{drm/exynos/hdmiphy.txt => video/exynos_hdmiphy.txt} (100%) rename Documentation/devicetree/bindings/{drm/exynos/mixer.txt => video/exynos_mixer.txt} (100%) create mode 100644 Documentation/devicetree/bindings/video/simple-framebuffer.txt create mode 100644 Documentation/kernel-per-CPU-kthreads.txt delete mode 100644 arch/avr32/include/asm/numnodes.h delete mode 100644 arch/avr32/include/asm/param.h delete mode 100644 arch/avr32/include/uapi/asm/param.h create mode 100644 arch/mips/include/asm/idle.h rename arch/mips/include/{ => uapi}/asm/kvm.h (100%) create mode 100644 arch/mips/kernel/idle.c create mode 100644 arch/powerpc/include/asm/context_tracking.h create mode 100644 arch/powerpc/sysdev/udbg_memcons.c delete mode 100644 drivers/acpi/csrt.c create mode 100644 drivers/staging/zcache/ramster/ramster-howto.txt create mode 100644 drivers/video/simplefb.c create mode 100644 lib/iovec.c delete mode 100644 tools/testing/selftests/soft-dirty/Makefile delete mode 100644 tools/testing/selftests/soft-dirty/soft-dirty.c

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[alsa-devel] [PATCH 0/6] Convert JZ4740 to dmaengine
by Lars-Peter Clausen 29 May '13

29 May '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. - 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 | 616 +++++++++++++++++++++++++++ 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, 675 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

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[alsa-devel] [PATCH] ASoC: sgtl5000: Fix VAG_POWER enabling/disabling order
by Marek Vasut 28 May '13

28 May '13

The VAG_POWER must be enabled after all other bits in CHIP_ANA_POWER and disabled before any other bit in CHIP_ANA_POWER. See the SGTL5000 datasheet (Table 31, BIT 7, page 42-43). Failing to follow this order will result in ugly loud "POP" noise at the end of playback. To achieve such order, use the _PRE and _POST DAPM widgets to trigger the power_vag_event, where the event type check has to be fixed accordingly as well. Signed-off-by: Marek Vasut <marex(a)denx.de> Cc: Dong Aisheng <dong.aisheng(a)linaro.org> Cc: Eric Nelson <eric.nelson(a)boundarydevices.com> Cc: Fabio Estevam <fabio.estevam(a)freescale.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/sgtl5000.c | 16 ++++++---------- 1 file changed, 6 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 92bbfec..2068c36 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -151,15 +151,15 @@ static int power_vag_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { switch (event) { - case SND_SOC_DAPM_PRE_PMU: + case SND_SOC_DAPM_POST_PMU: snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP); break; - case SND_SOC_DAPM_POST_PMD: + case SND_SOC_DAPM_PRE_PMD: snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_VAG_POWERUP, 0); - msleep(400); + mdelay(400); break; default: break; @@ -217,12 +217,11 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = { 0, SGTL5000_CHIP_DIG_POWER, 1, 0), - SND_SOC_DAPM_SUPPLY("VAG_POWER", SGTL5000_CHIP_ANA_POWER, 7, 0, - power_vag_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_ADC("ADC", "Capture", SGTL5000_CHIP_ANA_POWER, 1, 0), SND_SOC_DAPM_DAC("DAC", "Playback", SGTL5000_CHIP_ANA_POWER, 3, 0), + + SND_SOC_DAPM_PRE("VAG_POWER_PRE", power_vag_event), + SND_SOC_DAPM_POST("VAG_POWER_POST", power_vag_event), }; /* routes for sgtl5000 */ @@ -230,16 +229,13 @@ static const struct snd_soc_dapm_route sgtl5000_dapm_routes[] = { {"Capture Mux", "LINE_IN", "LINE_IN"}, /* line_in --> adc_mux */ {"Capture Mux", "MIC_IN", "MIC_IN"}, /* mic_in --> adc_mux */ - {"ADC", NULL, "VAG_POWER"}, {"ADC", NULL, "Capture Mux"}, /* adc_mux --> adc */ {"AIFOUT", NULL, "ADC"}, /* adc --> i2s_out */ - {"DAC", NULL, "VAG_POWER"}, {"DAC", NULL, "AIFIN"}, /* i2s-->dac,skip audio mux */ {"Headphone Mux", "DAC", "DAC"}, /* dac --> hp_mux */ {"LO", NULL, "DAC"}, /* dac --> line_out */ - {"LINE_IN", NULL, "VAG_POWER"}, {"Headphone Mux", "LINE_IN", "LINE_IN"},/* line_in --> hp_mux */ {"HP", NULL, "Headphone Mux"}, /* hp_mux --> hp */ -- 1.7.10.4

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