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

[alsa-devel] [patch] ALSA: hda - if statement not indented
by Dan Carpenter 14 May '14

14 May '14

The "break;" should be indented. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 1613388..589e47c 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3722,7 +3722,7 @@ static void parse_digital(struct hda_codec *codec) } else { spec->multiout.slave_dig_outs = spec->slave_dig_outs; if (nums >= ARRAY_SIZE(spec->slave_dig_outs) - 1) - break; + break; spec->slave_dig_outs[nums - 1] = dig_nid; } nums++;

2 1

[alsa-devel] [patch] ALSA: sb_mixer: missing return statement
by Dan Carpenter 14 May '14

14 May '14

The if condition here was supposed to return on error but the return statement is missing. The effect is that the ->mixername is set to "???" instead of "DT019X". Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/isa/sb/sb_mixer.c b/sound/isa/sb/sb_mixer.c index 6496822..1ff78ec 100644 --- a/sound/isa/sb/sb_mixer.c +++ b/sound/isa/sb/sb_mixer.c @@ -818,12 +818,14 @@ int snd_sbmixer_new(struct snd_sb *chip) return err; break; case SB_HW_DT019X: - if ((err = snd_sbmixer_init(chip, - snd_dt019x_controls, - ARRAY_SIZE(snd_dt019x_controls), - snd_dt019x_init_values, - ARRAY_SIZE(snd_dt019x_init_values), - "DT019X")) < 0) + err = snd_sbmixer_init(chip, + snd_dt019x_controls, + ARRAY_SIZE(snd_dt019x_controls), + snd_dt019x_init_values, + ARRAY_SIZE(snd_dt019x_init_values), + "DT019X"); + if (err < 0) + return err; break; default: strcpy(card->mixername, "???");

2 1

[alsa-devel] [PATCH v3 0/5] ALSA: hda - hdmi: ATI/AMD multi-channel and HBR support
by Anssi Hannula 14 May '14

14 May '14

Hi all! Here's another revision of the ATI/AMD multichannel+HBR patchset. Changes: - added customizable callbacks instead of polluting generic code - fixed wrong codec id in rev3+ check (as previously noted) - cosmetics I've only tested this on NVIDIA/Intel, but I'd guess at least Peter should be able to test this on AMD in a few days or so. The earlier revisions were tested to fully work. Combined patch can be found at: http://onse.fi/files/atihdmi5.patch However, this patchset is on top recent sound-next commits so it does not directly apply on older kernels. Olivier, I hope you will be able to test this at some point as well (same things as last time), even though it will be a bit more work :) Anssi Hannula (5): ALSA: hda - hdmi: Allow HDA patches to customize more operations ALSA: hda - hdmi: Add ATI/AMD multi-channel audio support ALSA: hda - hdmi: Add ELD emulation for ATI/AMD codecs ALSA: hda - hdmi: Add HBR bitstreaming support for ATI/AMD HDMI codecs ALSA: hda - hdmi: Disable ramp-up/down for non-PCM on AMD codecs sound/pci/hda/hda_eld.c | 151 +++++++++++ sound/pci/hda/hda_local.h | 4 + sound/pci/hda/patch_hdmi.c | 643 +++++++++++++++++++++++++++++++++++++-------- 3 files changed, 689 insertions(+), 109 deletions(-) -- Anssi Hannula

9 70

[alsa-devel] [GIT PULL] ASoC fixes for Intel for v3.15
by Mark Brown 14 May '14

14 May '14

The following changes since commit 95e9ee92e20162681e9f65c25962e0606db9d149: ASoC: Intel: Fix audio crash due to negative address offset (2014-04-22 19:22:53 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git asoc-v3.15-rc5-intel for you to fetch changes up to cffd6665f57ed18f4be9185c4330c8c98c22e201: ASoC: Intel: Fix Baytrail SST DSP firmware loading (2014-05-13 18:21:02 +0100) ---------------------------------------------------------------- ASoC: Intel fixes for v3.15 This is a relatively large batch of fixes for the newly added Haswell/Baytrail drivers from Intel. It's a bit larger than is good for this point in the cycle but it's all for a newly added driver so not so worrying as it might otherwise be. Some of it's integration problems, some of it's the sort of problem usually turned up in stress tests. ---------------------------------------------------------------- Jarkko Nikula (2): ASoC: Intel: Cancel hsw_notification_work before freeing the stream ASoC: Intel: Fix Baytrail SST DSP firmware loading Liam Girdwood (8): ASoC: Intel: Fix block allocation so we only allocate blocks once. ASoC: Intel: Fix allocated block list usage when adding blocks. ASoC: Intel: Fix Haswell/Broadwell DSP page table creation. ASoC: Intel: Fix Audio DSP usage when IOMMU is enabled. ASoC: Intel: Fix allow hw_params to be called more than once. ASoC: Intel: Fix stream position pointer. ASoC: Intel: Fix check for pdata usage before dereference. ASoC: Intel: Fix block offset calculations. Wenkai Du (1): ASoC: Intel: Fix audio crash due to race condition in stream deletion sound/soc/intel/sst-acpi.c | 1 + sound/soc/intel/sst-baytrail-dsp.c | 2 +- sound/soc/intel/sst-baytrail-ipc.c | 8 +++ sound/soc/intel/sst-dsp-priv.h | 1 + sound/soc/intel/sst-dsp.c | 1 + sound/soc/intel/sst-dsp.h | 1 + sound/soc/intel/sst-firmware.c | 25 +++------ sound/soc/intel/sst-haswell-dsp.c | 4 +- sound/soc/intel/sst-haswell-ipc.c | 31 ++++++++++- sound/soc/intel/sst-haswell-ipc.h | 4 +- sound/soc/intel/sst-haswell-pcm.c | 107 ++++++++++++++++++++++++------------- 11 files changed, 126 insertions(+), 59 deletions(-)

2 1

[alsa-devel] [GIT PULL] ASoC driver fixes for v3.15
by Mark Brown 14 May '14

14 May '14

The following changes since commit 4e17d2d33abe4b523f79fec037ef66ba93643f65: ASoC: cs42l52: Convert to use devm_gpio_request_one (2014-04-18 15:14:39 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git asoc-v3.15-rc5-drivers for you to fetch changes up to f9a405961eec4ee9a71a433a473b244ae2d74c3c: Merge remote-tracking branches 'asoc/fix/audmux', 'asoc/fix/cs42l52', 'asoc/fix/fsl-esai', 'asoc/fix/fsl-spdif', 'asoc/fix/rcar', 'asoc/fix/tlv320aic31xx' and 'asoc/fix/wm8962' into asoc-linus (2014-05-14 12:49:10 +0100) ---------------------------------------------------------------- ASoC: Driver fixes for v3.15 A small set of driver fixes, nothing remarkable in itself or of any relevance outside of the driver. ---------------------------------------------------------------- Charles Keepax (1): ASoC: wm8962: Update register CLASS_D_CONTROL_1 to be non-volatile Joe Perches (1): ASoC: tlv320aic31xx: Convert /n to \n Kuninori Morimoto (1): ASoC: rsnd: call rsnd_dai_pointer_update() from outside of lock Lars-Peter Clausen (1): ASoC: imx-audmux: Fix section mismatch Mark Brown (1): Merge remote-tracking branches 'asoc/fix/audmux', 'asoc/fix/cs42l52', 'asoc/fix/fsl-esai', 'asoc/fix/fsl-spdif', 'asoc/fix/rcar', 'asoc/fix/tlv320aic31xx' and 'asoc/fix/wm8962' into asoc-linus Nicolin Chen (3): ASoC: fsl_esai: Fix incorrect condition within ratio range check for FP ASoC: fsl_esai: Only bypass sck_div for EXTAL source ASoC: fsl_esai: Set PCRC and PRRC registers at the end of hw_params() .../devicetree/bindings/sound/tlv320aic31xx.txt | 6 +++--- sound/soc/codecs/tlv320aic31xx.c | 2 +- sound/soc/codecs/wm8962.c | 15 ++++++++++++--- sound/soc/codecs/wm8962.h | 4 ++++ sound/soc/fsl/fsl_esai.c | 22 ++++++++++++++-------- sound/soc/fsl/fsl_spdif.h | 4 ++-- sound/soc/fsl/imx-audmux.c | 2 +- sound/soc/sh/rcar/core.c | 5 ++--- sound/soc/sh/rcar/src.c | 4 ++-- sound/soc/sh/rcar/ssi.c | 4 ++-- 10 files changed, 43 insertions(+), 25 deletions(-)

2 1

[alsa-devel] [GIT PULL] ASoC core fixes for v3.15
by Mark Brown 14 May '14

14 May '14

The following changes since commit 2697e4fb9209dfe1d1b24c92d254158f63d4bc8e: ASoC: dapm: Fix widget double free with auto-disable DAPM kcontrol (2014-04-21 11:49:53 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git asoc-v3.15-rc5-core for you to fetch changes up to cf86197ec5655ed1f54f1f7ba34a01d576e3737d: Merge remote-tracking branch 'asoc/fix/pcm' into asoc-linus (2014-05-14 12:52:41 +0100) ---------------------------------------------------------------- ASoC: Core fixes for v3.15 A few things here: - Fix the creation of spurious CODEC<->CODEC links which caused DAPM to have audio paths which shouldn't be present causing spurious powerups and potential audible issues for users. - Ensure the suspend->off transition doesn't have spurious transitions to prepare added to the sequence. - Fix incorrect skipping of PCM suspension for active audio streams. - Remove Timur Tabi from the CS4270 maintainers, Cirrus are now doing this and Timur no longer has the boards that he was using. ---------------------------------------------------------------- Lars-Peter Clausen (2): ASoC: dapm: Skip CODEC<->CODEC links in connect_dai_link_widgets() ASoC: dapm: Fix SUSPEND -> OFF bias sequence Mark Brown (2): Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branch 'asoc/fix/pcm' into asoc-linus Nicolin Chen (1): ASoC: pcm: Fix incorrect condition check for case SNDRV_PCM_TRIGGER_SUSPEND Paul Handrigan (1): ASoC: Update Cirrus Logic CODEC maintainers. MAINTAINERS | 6 ------ sound/soc/soc-dapm.c | 14 ++++++++++---- sound/soc/soc-pcm.c | 2 +- 3 files changed, 11 insertions(+), 11 deletions(-)

2 1

[alsa-devel] [PATCH] ASoC: samsung: Add missing pm ops for Snow sound card driver
by Tushar Behera 14 May '14

14 May '14

Adding missing pm ops so that audio playback works across suspend and resume cycle. Signed-off-by: Tushar Behera <tushar.behera(a)linaro.org> --- sound/soc/samsung/snow.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/samsung/snow.c b/sound/soc/samsung/snow.c index 0fa89a4..014c177 100644 --- a/sound/soc/samsung/snow.c +++ b/sound/soc/samsung/snow.c @@ -111,6 +111,7 @@ static struct platform_driver snow_driver = { .driver = { .name = "snow-audio", .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, .of_match_table = snow_of_match, }, .probe = snow_probe, -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/2] ASoC: Intel: Fix Baytrail SST DSP firmware loading
by Jarkko Nikula 14 May '14

14 May '14

Commit 10df350977b1 ("ASoC: Intel: Fix Audio DSP usage when IOMMU is enabled.") caused following regression in Baytrail SST: baytrail-pcm-audio baytrail-pcm-audio: error: DMA alloc failed baytrail-pcm-audio baytrail-pcm-audio: error: failed to load firmware Fix this by calling dma_coerce_mask_and_coherent() in sst_byt_init() with the same dma_dev device what is now used in sst_fw_new() when allocating the DMA buffer. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- sound/soc/intel/sst-baytrail-dsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/sst-baytrail-dsp.c b/sound/soc/intel/sst-baytrail-dsp.c index 4a5d489e520b..fc588764ffa3 100644 --- a/sound/soc/intel/sst-baytrail-dsp.c +++ b/sound/soc/intel/sst-baytrail-dsp.c @@ -324,7 +324,7 @@ static int sst_byt_init(struct sst_dsp *sst, struct sst_pdata *pdata) return ret; } - ret = dma_coerce_mask_and_coherent(dev, DMA_BIT_MASK(32)); + ret = dma_coerce_mask_and_coherent(sst->dma_dev, DMA_BIT_MASK(32)); if (ret) return ret; -- 2.0.0.rc0

2 11

[alsa-devel] [PATCH 0/5] ASoC: rsnd: add DVC support
by Kuninori Morimoto 13 May '14

13 May '14

Hi Mark These patches add DVC support to Renesas sound driver. DVC is one of feature of CMD unit, it can control sound Volume. Kuninori Morimoto (5): ASoC: rsnd: add rsnd_path_parse() macro ASoC: rsnd: add rsnd_get_adinr() ASoC: rsnd: remove duplicate parameter from rsnd_mod_ops ASoC: rsnd: enable to use multi parameter on rsnd_dai_call/rsnd_mod_call ASoC: rsnd: add DVC support include/sound/rcar_snd.h | 10 ++ sound/soc/sh/rcar/Makefile | 2 +- sound/soc/sh/rcar/adg.c | 18 +++ sound/soc/sh/rcar/core.c | 117 +++++++++++++------ sound/soc/sh/rcar/dvc.c | 273 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sh/rcar/gen.c | 11 ++ sound/soc/sh/rcar/rsnd.h | 65 ++++++++--- sound/soc/sh/rcar/src.c | 121 +++++++------------- sound/soc/sh/rcar/ssi.c | 34 +++--- 9 files changed, 502 insertions(+), 149 deletions(-)

2 6

[alsa-devel] [PATCH] ASoC: wm8962: Update register CLASS_D_CONTROL_1 to be non-volatile
by Charles Keepax 13 May '14

13 May '14

The register CLASS_D_CONTROL_1 is marked as volatile because it contains a bit, DAC_MUTE, which is also mirrored in the ADC_DAC_CONTROL_1 register. This causes problems for the "Speaker Switch" control, which will report an error if the CODEC is suspended because it relies on a volatile register. To resolve this issue mark CLASS_D_CONTROL_1 as non-volatile and manually keep the register cache in sync by updating both bits when changing the mute status. Reported-by: Shawn Guo <shawn.guo(a)linaro.org> Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Hi Shawn, I am afraid I don't have access to hardware for 8962 at the moment, so have only build tested this, but I believe it should fix your issue. Please let me know how you get on. Thanks, Charles sound/soc/codecs/wm8962.c | 15 ++++++++++++--- sound/soc/codecs/wm8962.h | 4 ++++ 2 files changed, 16 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 5522d25..ecd26dd 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -154,6 +154,7 @@ static struct reg_default wm8962_reg[] = { { 40, 0x0000 }, /* R40 - SPKOUTL volume */ { 41, 0x0000 }, /* R41 - SPKOUTR volume */ + { 49, 0x0010 }, /* R49 - Class D Control 1 */ { 51, 0x0003 }, /* R51 - Class D Control 2 */ { 56, 0x0506 }, /* R56 - Clocking 4 */ @@ -795,7 +796,6 @@ static bool wm8962_volatile_register(struct device *dev, unsigned int reg) case WM8962_ALC2: case WM8962_THERMAL_SHUTDOWN_STATUS: case WM8962_ADDITIONAL_CONTROL_4: - case WM8962_CLASS_D_CONTROL_1: case WM8962_DC_SERVO_6: case WM8962_INTERRUPT_STATUS_1: case WM8962_INTERRUPT_STATUS_2: @@ -2929,13 +2929,22 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source, static int wm8962_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - int val; + int val, ret; if (mute) - val = WM8962_DAC_MUTE; + val = WM8962_DAC_MUTE | WM8962_DAC_MUTE_ALT; else val = 0; + /** + * The DAC mute bit is mirrored in two registers, update both to keep + * the register cache consistent. + */ + ret = snd_soc_update_bits(codec, WM8962_CLASS_D_CONTROL_1, + WM8962_DAC_MUTE_ALT, val); + if (ret < 0) + return ret; + return snd_soc_update_bits(codec, WM8962_ADC_DAC_CONTROL_1, WM8962_DAC_MUTE, val); } diff --git a/sound/soc/codecs/wm8962.h b/sound/soc/codecs/wm8962.h index a1a5d52..910aafd 100644 --- a/sound/soc/codecs/wm8962.h +++ b/sound/soc/codecs/wm8962.h @@ -1954,6 +1954,10 @@ #define WM8962_SPKOUTL_ENA_MASK 0x0040 /* SPKOUTL_ENA */ #define WM8962_SPKOUTL_ENA_SHIFT 6 /* SPKOUTL_ENA */ #define WM8962_SPKOUTL_ENA_WIDTH 1 /* SPKOUTL_ENA */ +#define WM8962_DAC_MUTE_ALT 0x0010 /* DAC_MUTE */ +#define WM8962_DAC_MUTE_ALT_MASK 0x0010 /* DAC_MUTE */ +#define WM8962_DAC_MUTE_ALT_SHIFT 4 /* DAC_MUTE */ +#define WM8962_DAC_MUTE_ALT_WIDTH 1 /* DAC_MUTE */ #define WM8962_SPKOUTL_PGA_MUTE 0x0002 /* SPKOUTL_PGA_MUTE */ #define WM8962_SPKOUTL_PGA_MUTE_MASK 0x0002 /* SPKOUTL_PGA_MUTE */ #define WM8962_SPKOUTL_PGA_MUTE_SHIFT 1 /* SPKOUTL_PGA_MUTE */ -- 1.7.2.5

3 2