June 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] sound: pci: ice1712: wm8766.c: Cleaning up uninitialized variables
by Rickard Strandqvist 01 Jun '14

01 Jun '14

There is a risk that the variable will be used without being initialized. This was largely found by using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/pci/ice1712/wm8766.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/ice1712/wm8766.c b/sound/pci/ice1712/wm8766.c index 21b373b..788a31c 100644 --- a/sound/pci/ice1712/wm8766.c +++ b/sound/pci/ice1712/wm8766.c @@ -237,7 +237,7 @@ static int snd_wm8766_ctl_get(struct snd_kcontrol *kcontrol, { struct snd_wm8766 *wm = snd_kcontrol_chip(kcontrol); int n = kcontrol->private_value; - u16 val1, val2; + u16 val1 = 0, val2 = 0; if (wm->ctl[n].get) wm->ctl[n].get(wm, &val1, &val2); -- 1.7.10.4

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[alsa-devel] [PATCH] bebob: Use ARRAY_SIZE() instead of sizeof() for return condition
by Takashi Sakamoto 01 Jun '14

01 Jun '14

The parser for stream format information should return when detecting wrong value for sampling rate. But it doesn't. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/bebob/bebob_stream.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c index 514c7c9..bc4f827 100644 --- a/sound/firewire/bebob/bebob_stream.c +++ b/sound/firewire/bebob/bebob_stream.c @@ -749,7 +749,7 @@ parse_stream_formation(u8 *buf, unsigned int len, if (buf[2] == bridgeco_freq_table[i]) break; } - if (i == sizeof(bridgeco_freq_table)) + if (i == ARRAY_SIZE(bridgeco_freq_table)) return -ENOSYS; /* Avoid double count by different entries for the same rate. */ -- 1.8.3.2

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[alsa-devel] [RFC] [PATCH] firewire-lib: Use IEC 61883-6 compliant labels for Raw Audio/
by Takashi Sakamoto 01 Jun '14

01 Jun '14

Hi Clemens, Would I request you to comment this patch? I don't touch this label issue with my patchset because there is a bit possibility to cause a regression. But now, I have a confidence that this modification is reasonable. In detail, please see an attached patch. Regards Takashi Sakamoto o-takashi(a)sakamocchi.jp

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[alsa-devel] [PATCH] sound: pci: ice1712: wm8776.c: Cleaning up uninitialized variables
by Rickard Strandqvist 01 Jun '14

01 Jun '14

There is a risk that the variable will be used without being initialized. This was largely found by using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/pci/ice1712/wm8776.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/ice1712/wm8776.c b/sound/pci/ice1712/wm8776.c index e66c0da..52da0ff 100644 --- a/sound/pci/ice1712/wm8776.c +++ b/sound/pci/ice1712/wm8776.c @@ -510,7 +510,7 @@ static int snd_wm8776_ctl_get(struct snd_kcontrol *kcontrol, { struct snd_wm8776 *wm = snd_kcontrol_chip(kcontrol); int n = kcontrol->private_value; - u16 val1, val2; + u16 val1 = 0, val2 = 0; if (wm->ctl[n].get) wm->ctl[n].get(wm, &val1, &val2); -- 1.7.10.4

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[alsa-devel] [PATCH] ALSA: ASoC: wm9713: correct mono out PGA sources
by Matt Reimer 01 Jun '14

01 Jun '14

The mono output PGA input only has four possible sources, so omit the rest. Signed-off-by: Matt Reimer <mreimer(a)sdgsystems.com> --- sound/soc/codecs/wm9713.c | 3 +-- 1 files changed, 1 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 5f69b52..caf0269 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -74,8 +74,7 @@ static const char *wm9713_rec_src[] = "Mono Out", "Zh"}; static const char *wm9713_rec_gain[] = {"+1.5dB Steps", "+0.75dB Steps"}; static const char *wm9713_alc_select[] = {"None", "Left", "Right", "Stereo"}; -static const char *wm9713_mono_pga[] = {"Vmid", "Zh", "Mono", "Inv", - "Mono Vmid", "Inv Vmid"}; +static const char *wm9713_mono_pga[] = {"Vmid", "Zh", "Mono", "Inv"}; static const char *wm9713_spk_pga[] = {"Vmid", "Zh", "Headphone", "Speaker", "Inv", "Headphone Vmid", "Speaker Vmid", "Inv Vmid"}; -- 1.7.5.4

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[alsa-devel] [PATCH] sound: synth: emux: soundfont.c: Cleaning up memory leak
by Rickard Strandqvist 01 Jun '14

01 Jun '14

There is a risk for memory leak in when something unexpected happens and the function returns. This was largely found by using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/synth/emux/soundfont.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/synth/emux/soundfont.c b/sound/synth/emux/soundfont.c index 1137b85..09c5eeb 100644 --- a/sound/synth/emux/soundfont.c +++ b/sound/synth/emux/soundfont.c @@ -1021,6 +1021,7 @@ load_guspatch(struct snd_sf_list *sflist, const char __user *data, data, count); if (rc < 0) { sf_sample_delete(sflist, sf, smp); + free(zone); return rc; } /* memory offset is updated after */ -- 1.7.10.4

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[alsa-devel] [PATCH] ASoC: fsl: Remove dependencies of boards for SND_SOC_EUKREA_TLV320
by Alexander Shiyan 01 Jun '14

01 Jun '14

Eukrea-i.MX51 board was converted to use DT, ie we no longer have a MACH_EUKREA_MBIMXSD51_BASEBOARD symbol. Transformation of other boards planned for the near future, so this patch removes all these dependencies and restricts build of this driver to ARCH_MXC. Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> --- sound/soc/fsl/Kconfig | 7 +------ 1 file changed, 1 insertion(+), 6 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index d262ec0..2989319 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -207,12 +207,7 @@ config SND_SOC_PHYCORE_AC97 config SND_SOC_EUKREA_TLV320 tristate "Eukrea TLV320" - depends on MACH_EUKREA_MBIMX27_BASEBOARD \ - || MACH_EUKREA_MBIMXSD25_BASEBOARD \ - || MACH_EUKREA_MBIMXSD35_BASEBOARD \ - || MACH_EUKREA_MBIMXSD51_BASEBOARD \ - || (OF && ARM) - depends on I2C + depends on ARCH_MXC && I2C select SND_SOC_TLV320AIC23_I2C select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI -- 1.8.3.2

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[alsa-devel] [PATCH v5 0/8] ASoC: fsl-ssi: Driver cleanup
by Markus Pargmann 01 Jun '14

01 Jun '14

Hi, These are the remaining patches of this fsl-ssi cleanup series. v5 adds the possibility to set the bitclock frequency using .set_dai_sysclk as suggested. If it is not explicitly set, the bitclock is calculated using the hw_params. I also added the missing signed-off-by tags to the patches. Best Regards, Markus Markus Pargmann (4): ASoC: fsl-ssi: make fsl,mode property optional ASoC: fsl-ssi: Fix baudclock handling ASoC: fsl-ssi: reorder and document fsl_ssi_private ASoC: fsl-ssi: Use regmap Sascha Hauer (4): ASoC: fsl-ssi: introduce SoC specific data ASoC: fsl-ssi: set bitclock in master mode from hw_params ASoC: fsl-ssi: remove unnecessary spinlock ASoC: fsl-ssi: Set framerate divider correctly for i2s master mode sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/fsl_ssi.c | 574 ++++++++++++++++++++++++++++-------------------- sound/soc/fsl/fsl_ssi.h | 50 ++--- 3 files changed, 360 insertions(+), 265 deletions(-) -- 2.0.0.rc2

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[alsa-devel] [PATCH 1/4] ASoC: Intel: Clear stored Baytrail DSP DMA pointer before stream start
by Jarkko Nikula 01 Jun '14

01 Jun '14

Stored DSP DMA pointer must be cleared before starting the stream since PCM pointer callback sst_byt_pcm_pointer() can be called before pointer is updated. In that case last position of previous stream was wronly returned. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- sound/soc/intel/sst-baytrail-pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/sst-baytrail-pcm.c b/sound/soc/intel/sst-baytrail-pcm.c index 3af38576e91e..8eab97368ea7 100644 --- a/sound/soc/intel/sst-baytrail-pcm.c +++ b/sound/soc/intel/sst-baytrail-pcm.c @@ -180,6 +180,7 @@ static int sst_byt_pcm_trigger(struct snd_pcm_substream *substream, int cmd) switch (cmd) { case SNDRV_PCM_TRIGGER_START: + pcm_data->hw_ptr = 0; sst_byt_stream_start(byt, pcm_data->stream, 0); break; case SNDRV_PCM_TRIGGER_RESUME: -- 2.0.0.rc2

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[alsa-devel] [PATCH v3] ASoC: cs42l56: Fix new value argument in snd_soc_update_bits calls
by Axel Lin 01 Jun '14

01 Jun '14

The new value argument needs proper shift to match the mask bit fields. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- v3: Define CS42L56_MUTE_ALL as 0xff, so all the mask bits will be set. v2: Address Brian's comment in v1. sound/soc/codecs/cs42l56.c | 70 ++++++++++++++++++++-------------------------- sound/soc/codecs/cs42l56.h | 10 ++++--- 2 files changed, 37 insertions(+), 43 deletions(-) diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index 5bb134b..fdc4bd2 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -763,14 +763,14 @@ static int cs42l56_set_sysclk(struct snd_soc_dai *codec_dai, case CS42L56_MCLK_11P2896MHZ: case CS42L56_MCLK_12MHZ: case CS42L56_MCLK_12P288MHZ: - cs42l56->mclk_div2 = 1; + cs42l56->mclk_div2 = CS42L56_MCLK_DIV2; cs42l56->mclk_prediv = 0; break; case CS42L56_MCLK_22P5792MHZ: case CS42L56_MCLK_24MHZ: case CS42L56_MCLK_24P576MHZ: - cs42l56->mclk_div2 = 1; - cs42l56->mclk_prediv = 1; + cs42l56->mclk_div2 = CS42L56_MCLK_DIV2; + cs42l56->mclk_prediv = CS42L56_MCLK_PREDIV; break; default: return -EINVAL; @@ -844,57 +844,49 @@ static int cs42l56_digital_mute(struct snd_soc_dai *dai, int mute) /* Hit the DSP Mixer first */ snd_soc_update_bits(codec, CS42L56_DSP_MUTE_CTL, CS42L56_ADCAMIX_MUTE_MASK | - CS42L56_ADCBMIX_MUTE_MASK | - CS42L56_PCMAMIX_MUTE_MASK | - CS42L56_PCMBMIX_MUTE_MASK | - CS42L56_MSTB_MUTE_MASK | - CS42L56_MSTA_MUTE_MASK, - CS42L56_MUTE); + CS42L56_ADCBMIX_MUTE_MASK | + CS42L56_PCMAMIX_MUTE_MASK | + CS42L56_PCMBMIX_MUTE_MASK | + CS42L56_MSTB_MUTE_MASK | + CS42L56_MSTA_MUTE_MASK, + CS42L56_MUTE_ALL); /* Mute ADC's */ snd_soc_update_bits(codec, CS42L56_MISC_ADC_CTL, - CS42L56_ADCA_MUTE_MASK | - CS42L56_ADCB_MUTE_MASK, - CS42L56_MUTE); + CS42L56_ADCA_MUTE_MASK | + CS42L56_ADCB_MUTE_MASK, + CS42L56_MUTE_ALL); /* HP And LO */ snd_soc_update_bits(codec, CS42L56_HPA_VOLUME, - CS42L56_HP_MUTE_MASK, - CS42L56_MUTE); + CS42L56_HP_MUTE_MASK, CS42L56_MUTE_ALL); snd_soc_update_bits(codec, CS42L56_HPB_VOLUME, - CS42L56_HP_MUTE_MASK, - CS42L56_MUTE); + CS42L56_HP_MUTE_MASK, CS42L56_MUTE_ALL); snd_soc_update_bits(codec, CS42L56_LOA_VOLUME, - CS42L56_LO_MUTE_MASK, - CS42L56_MUTE); + CS42L56_LO_MUTE_MASK, CS42L56_MUTE_ALL); snd_soc_update_bits(codec, CS42L56_LOB_VOLUME, - CS42L56_LO_MUTE_MASK, - CS42L56_MUTE); - - + CS42L56_LO_MUTE_MASK, CS42L56_MUTE_ALL); } else { snd_soc_update_bits(codec, CS42L56_DSP_MUTE_CTL, CS42L56_ADCAMIX_MUTE_MASK | - CS42L56_ADCBMIX_MUTE_MASK | - CS42L56_PCMAMIX_MUTE_MASK | - CS42L56_PCMBMIX_MUTE_MASK | - CS42L56_MSTB_MUTE_MASK | - CS42L56_MSTA_MUTE_MASK, - CS42L56_UNMUTE); + CS42L56_ADCBMIX_MUTE_MASK | + CS42L56_PCMAMIX_MUTE_MASK | + CS42L56_PCMBMIX_MUTE_MASK | + CS42L56_MSTB_MUTE_MASK | + CS42L56_MSTA_MUTE_MASK, + CS42L56_UNMUTE); + snd_soc_update_bits(codec, CS42L56_MISC_ADC_CTL, - CS42L56_ADCA_MUTE_MASK | - CS42L56_ADCB_MUTE_MASK, - CS42L56_UNMUTE); + CS42L56_ADCA_MUTE_MASK | + CS42L56_ADCB_MUTE_MASK, + CS42L56_UNMUTE); + snd_soc_update_bits(codec, CS42L56_HPA_VOLUME, - CS42L56_HP_MUTE_MASK, - CS42L56_UNMUTE); + CS42L56_HP_MUTE_MASK, CS42L56_UNMUTE); snd_soc_update_bits(codec, CS42L56_HPB_VOLUME, - CS42L56_HP_MUTE_MASK, - CS42L56_UNMUTE); + CS42L56_HP_MUTE_MASK, CS42L56_UNMUTE); snd_soc_update_bits(codec, CS42L56_LOA_VOLUME, - CS42L56_LO_MUTE_MASK, - CS42L56_UNMUTE); + CS42L56_LO_MUTE_MASK, CS42L56_UNMUTE); snd_soc_update_bits(codec, CS42L56_LOB_VOLUME, - CS42L56_LO_MUTE_MASK, - CS42L56_UNMUTE); + CS42L56_LO_MUTE_MASK, CS42L56_UNMUTE); } return 0; } diff --git a/sound/soc/codecs/cs42l56.h b/sound/soc/codecs/cs42l56.h index ad2b50a..5025ec9 100644 --- a/sound/soc/codecs/cs42l56.h +++ b/sound/soc/codecs/cs42l56.h @@ -80,19 +80,21 @@ #define CS42L56_PDN_HPB_MASK 0xc0 /* serial port and clk masks */ -#define CS42L56_MASTER_MODE 1 +#define CS42L56_MASTER_MODE 0x40 #define CS42L56_SLAVE_MODE 0 #define CS42L56_MS_MODE_MASK 0x40 -#define CS42L56_SCLK_INV 1 +#define CS42L56_SCLK_INV 0x20 #define CS42L56_SCLK_INV_MASK 0x20 #define CS42L56_SCLK_MCLK_MASK 0x18 +#define CS42L56_MCLK_PREDIV 0x04 #define CS42L56_MCLK_PREDIV_MASK 0x04 +#define CS42L56_MCLK_DIV2 0x02 #define CS42L56_MCLK_DIV2_MASK 0x02 #define CS42L56_MCLK_DIS_MASK 0x01 #define CS42L56_CLK_AUTO_MASK 0x20 #define CS42L56_CLK_RATIO_MASK 0x1f #define CS42L56_DIG_FMT_I2S 0 -#define CS42L56_DIG_FMT_LEFT_J 1 +#define CS42L56_DIG_FMT_LEFT_J 0x08 #define CS42L56_DIG_FMT_MASK 0x08 /* Class H and misc ctl masks */ @@ -116,7 +118,7 @@ #define CS42L56_DEEMPH_MASK 0x40 #define CS42L56_PLYBCK_GANG_MASK 0x10 #define CS42L56_PCM_INV_MASK 0x0c -#define CS42L56_MUTE 1 +#define CS42L56_MUTE_ALL 0xff #define CS42L56_UNMUTE 0 #define CS42L56_ADCAMIX_MUTE_MASK 0x40 #define CS42L56_ADCBMIX_MUTE_MASK 0x80 -- 1.8.3.2

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