December 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [patch] ALSA: cs46xx: add a new line
by Dan Carpenter 07 Dec '16

07 Dec '16

We accidentally deleted a newline so now the "nreallocated++;" statement is hanging out way off to the right of the screen. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/cs46xx/dsp_spos.c b/sound/pci/cs46xx/dsp_spos.c index 4a0cbd2241d8..aa61615288ff 100644 --- a/sound/pci/cs46xx/dsp_spos.c +++ b/sound/pci/cs46xx/dsp_spos.c @@ -107,7 +107,8 @@ static int shadow_and_reallocate_code (struct snd_cs46xx * chip, u32 * data, u32 dev_dbg(chip->card->dev, "handle_wideop:[2] %05x:%05x addr %04x\n", - hival, loval, address); nreallocated++; + hival, loval, address); + nreallocated++; } /* wide_opcodes[j] == wide_op */ } /* for */ } /* mod_type == 0 ... */

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[alsa-devel] [PATCH v7 0/3] Media Device Allocator API
by Shuah Khan 07 Dec '16

07 Dec '16

Media Device Allocator API to allows multiple drivers share a media device. Using this API, drivers can allocate a media device with the shared struct device as the key. Once the media device is allocated by a driver, other drivers can get a reference to it. The media device is released when all the references are released. Patches 0001 and 0002 are rebased to 4.9-rc7. Patch 0003 for snd-usb-audio is a rebase of the patch that was tested with the original Media Device Allocator patch series. snd-usb-audio patch includes the fixes found during 4.7-rc1 time in the original snd-usb-audio patch. Changes to patches in this series: Changes to patch 0001 since v6: - No changes Changes to patch 0002 since v6: - No changes Changes to patch 0003 since v6: - Addressed Takashi's review comments on patch v6 Depends on: https://lkml.org/lkml/2016/11/29/1001 Changes to patch 0001 since v5: (comments from Mauro and Sakari) - Removed struct device from media_device_instance. mdev.dev is used instead. - Added documentation. Changes to patch 0002: - No changes since patch v2, applies cleanly on top of the following: media: Protect enable_source and disable_source handler code paths https://lkml.org/lkml/2016/11/29/1001 Changes to patch 0003: - Changed to hold graph_mutex to check and call enable_source and disable_source handlers - to match au0828 doing the same in: media: Protect enable_source and disable_source handler code paths https://lkml.org/lkml/2016/11/29/1001 Changes to patch 0001 since v4: - Addressed Sakari's review comments with the exception of opting to not introduce media_device_usb_allocate() macro, and to not add a new routine to find media device instance to avoid a one line check. Changes to patch 0001 since v3: - Fixed undefined reference to `__media_device_usb_init compile error when CONFIG_USB is disabled. - Fixed kernel paging error when accessing /dev/mediaX after rmmod of the module that owns the media_device. The fix bumps the reference count for the owner when second driver comes along to share the media_device. If au0828 owns the media_device, then snd_usb_audio will bump the refcount for au0828, so it won't get deleted and vice versa. Changes to patch 0002 since v2: - Updated media_device_delete() to pass in module name. Changes to patch 0003 since the last version in 4.7-rc1: - Included fixes to bugs found during testing. - Updated to use the Media Allocator API. This patch series has been tested with au0828 and snd-usb-audio drivers. Ran bind and unbind loop tests on each driver with mc_nextgen_test and media_device_test app loop tests while checking lsmod and dmesg. Please refer to tools/testing/selftests/media_tests/regression_test.txt for testing done on this series. Shuah Khan (3): media: Media Device Allocator API media: change au0828 to use Media Device Allocator API sound/usb: Use Media Controller API to share media resources Documentation/media/kapi/mc-core.rst | 37 ++++ drivers/media/Makefile | 3 +- drivers/media/media-dev-allocator.c | 133 ++++++++++++++ drivers/media/usb/au0828/au0828-core.c | 12 +- drivers/media/usb/au0828/au0828.h | 1 + include/media/media-dev-allocator.h | 54 ++++++ sound/usb/Kconfig | 4 + sound/usb/Makefile | 2 + sound/usb/card.c | 14 ++ sound/usb/card.h | 3 + sound/usb/media.c | 321 +++++++++++++++++++++++++++++++++ sound/usb/media.h | 73 ++++++++ sound/usb/mixer.h | 3 + sound/usb/pcm.c | 29 ++- sound/usb/quirks-table.h | 1 + sound/usb/stream.c | 2 + sound/usb/usbaudio.h | 6 + 17 files changed, 684 insertions(+), 14 deletions(-) create mode 100644 drivers/media/media-dev-allocator.c create mode 100644 include/media/media-dev-allocator.h create mode 100644 sound/usb/media.c create mode 100644 sound/usb/media.h -- 2.7.4

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[alsa-devel] [patch] dbri: move dereference after check for NULL
by Dan Carpenter 06 Dec '16

06 Dec '16

We accidentally introduced a dereference before the NULL check in xmit_descs() as part of silencing a GCC warning. Fixes: 16f46050e709 ("dbri: Fix compiler warning") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- Goes into the sparc tree. diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index 3fe4468..52063b2 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -1702,7 +1702,7 @@ interrupts are disabled. static void xmit_descs(struct snd_dbri *dbri) { struct dbri_streaminfo *info; - u32 dvma_addr = (u32)dbri->dma_dvma; + u32 dvma_addr; s32 *cmd; unsigned long flags; int first_td; @@ -1710,6 +1710,7 @@ static void xmit_descs(struct snd_dbri *dbri) if (dbri == NULL) return; /* Disabled */ + dvma_addr = (u32)dbri->dma_dvma; info = &dbri->stream_info[DBRI_REC]; spin_lock_irqsave(&dbri->lock, flags);

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[alsa-devel] Applied "ASoC: Intel: update bxt_da7219_max98357a to support quad ch dmic capture" to the asoc tree
by Mark Brown 06 Dec '16

06 Dec '16

The patch ASoC: Intel: update bxt_da7219_max98357a to support quad ch dmic capture has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From c1aee1d828caa86a663c57a56eead7fd5f22d80f Mon Sep 17 00:00:00 2001 From: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> Date: Tue, 6 Dec 2016 07:08:36 -0800 Subject: [PATCH] ASoC: Intel: update bxt_da7219_max98357a to support quad ch dmic capture This patch updates FE channel constraints & BE fixup to support quad channel DMIC capture. DMIC pin's BE fixup is configured based on channel input, i.e. either stereo or quad. Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> Acked-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/bxt_da7219_max98357a.c | 20 +++++++++++++++++--- 1 file changed, 17 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index bff80b467262..3a8c24a59020 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -30,6 +30,7 @@ #define BXT_DIALOG_CODEC_DAI "da7219-hifi" #define BXT_MAXIM_CODEC_DAI "HiFi" #define DUAL_CHANNEL 2 +#define QUAD_CHANNEL 4 static struct snd_soc_jack broxton_headset; @@ -182,6 +183,16 @@ static struct snd_pcm_hw_constraint_list constraints_channels = { .mask = 0, }; +static unsigned int channels_quad[] = { + QUAD_CHANNEL, +}; + +static struct snd_pcm_hw_constraint_list constraints_channels_quad = { + .count = ARRAY_SIZE(channels_quad), + .list = channels_quad, + .mask = 0, +}; + static int bxt_fe_startup(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -258,7 +269,10 @@ static int broxton_dmic_fixup(struct snd_soc_pcm_runtime *rtd, { struct snd_interval *channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); - channels->min = channels->max = DUAL_CHANNEL; + if (params_channels(params) == 2) + channels->min = channels->max = 2; + else + channels->min = channels->max = 4; return 0; } @@ -267,9 +281,9 @@ static int broxton_dmic_startup(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; - runtime->hw.channels_max = DUAL_CHANNEL; + runtime->hw.channels_min = runtime->hw.channels_max = QUAD_CHANNEL; snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - &constraints_channels); + &constraints_channels_quad); return snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); -- 2.10.2

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[alsa-devel] [PATCH] ASoC: Intel: update bxt_da7219_max98357a to support quad ch dmic capture
by sathyanarayana.nujella＠intel.com 06 Dec '16

06 Dec '16

From: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> This patch updates FE channel constraints & BE fixup to support quad channel DMIC capture. DMIC pin's BE fixup is configured based on channel input, i.e. either stereo or quad. Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> Acked-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/boards/bxt_da7219_max98357a.c | 20 +++++++++++++++++--- 1 file changed, 17 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index bff80b4..3a8c24a 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -30,6 +30,7 @@ #define BXT_DIALOG_CODEC_DAI "da7219-hifi" #define BXT_MAXIM_CODEC_DAI "HiFi" #define DUAL_CHANNEL 2 +#define QUAD_CHANNEL 4 static struct snd_soc_jack broxton_headset; @@ -182,6 +183,16 @@ static struct snd_pcm_hw_constraint_list constraints_channels = { .mask = 0, }; +static unsigned int channels_quad[] = { + QUAD_CHANNEL, +}; + +static struct snd_pcm_hw_constraint_list constraints_channels_quad = { + .count = ARRAY_SIZE(channels_quad), + .list = channels_quad, + .mask = 0, +}; + static int bxt_fe_startup(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -258,7 +269,10 @@ static int broxton_dmic_fixup(struct snd_soc_pcm_runtime *rtd, { struct snd_interval *channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); - channels->min = channels->max = DUAL_CHANNEL; + if (params_channels(params) == 2) + channels->min = channels->max = 2; + else + channels->min = channels->max = 4; return 0; } @@ -267,9 +281,9 @@ static int broxton_dmic_startup(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; - runtime->hw.channels_max = DUAL_CHANNEL; + runtime->hw.channels_min = runtime->hw.channels_max = QUAD_CHANNEL; snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - &constraints_channels); + &constraints_channels_quad); return snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); -- 2.8.4

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[alsa-devel] [PATCH 0/2 v2] usb-audio misc fix
by Jiada Wang 06 Dec '16

06 Dec '16

This patch set contains the following patches Andreas Pape (1): ALSA: usb-audio: more tolerant packetsize Daniel Girnus (1): ALSA: usb-audio: avoid setting of sample rate multiple times on bus sound/usb/endpoint.c | 4 ++-- sound/usb/pcm.c | 21 +++++++++++---------- 2 files changed, 13 insertions(+), 12 deletions(-) -- 2.9.3

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[alsa-devel] [PATCH] ASoC: nau8825: disable sinc filter for high THD of ADC
by John Hsu 06 Dec '16

06 Dec '16

This bit will enable 4th order SINC filter. =1, filter will enable; but it consumes higher power. =0, the sinc filter is disable, and it should always keep 0 value to get high THD. Therefor, disable the filter when codec initiation for better performance when recording. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 3 ++- sound/soc/codecs/nau8825.h | 2 ++ 2 files changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 2b89569..efe3a44 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1846,7 +1846,8 @@ static void nau8825_init_regs(struct nau8825 *nau8825) * (audible hiss). Set it to something better. */ regmap_update_bits(regmap, NAU8825_REG_ADC_RATE, - NAU8825_ADC_SYNC_DOWN_MASK, NAU8825_ADC_SYNC_DOWN_64); + NAU8825_ADC_SYNC_DOWN_MASK | NAU8825_ADC_SINC4_EN, + NAU8825_ADC_SYNC_DOWN_64); regmap_update_bits(regmap, NAU8825_REG_DAC_CTRL1, NAU8825_DAC_OVERSAMPLE_MASK, NAU8825_DAC_OVERSAMPLE_64); /* Disable DACR/L power */ diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index 0672a25..5d1704e 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -269,6 +269,8 @@ #define NAU8825_BIQ_PATH_DAC (1 << NAU8825_BIQ_PATH_SFT) /* ADC_RATE (0x2b) */ +#define NAU8825_ADC_SINC4_SFT 4 +#define NAU8825_ADC_SINC4_EN (1 << NAU8825_ADC_SINC4_SFT) #define NAU8825_ADC_SYNC_DOWN_SFT 0 #define NAU8825_ADC_SYNC_DOWN_MASK 0x3 #define NAU8825_ADC_SYNC_DOWN_32 0 -- 2.6.4

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[alsa-devel] [PATCH] ASoC: cs35l34: Simplify the logic to set CS35L34_MCLK_CTL setting
by Axel Lin 06 Dec '16

06 Dec '16

The logic of "value = ~CS35L34_MCLK_DIV & CS35L34_MCLK_RATE_XXXXXX;" is unnecessary complex. By setting CS35L34_MCLK_DIV | CS35L34_MCLK_RATE_MASK as the mask for regmap_update_bits() call, what the code does is exactly the same as setting value = CS35L34_MCLK_RATE_XXXXXX. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs35l34.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/cs35l34.c b/sound/soc/codecs/cs35l34.c index e0f672a..7c5d151 100644 --- a/sound/soc/codecs/cs35l34.c +++ b/sound/soc/codecs/cs35l34.c @@ -610,15 +610,15 @@ static int cs35l34_dai_set_sysclk(struct snd_soc_dai *dai, switch (freq) { case CS35L34_MCLK_5644: - value = ~CS35L34_MCLK_DIV & CS35L34_MCLK_RATE_5P6448; + value = CS35L34_MCLK_RATE_5P6448; cs35l34->mclk_int = freq; break; case CS35L34_MCLK_6: - value = ~CS35L34_MCLK_DIV & CS35L34_MCLK_RATE_6P0000; + value = CS35L34_MCLK_RATE_6P0000; cs35l34->mclk_int = freq; break; case CS35L34_MCLK_6144: - value = ~CS35L34_MCLK_DIV & CS35L34_MCLK_RATE_6P1440; + value = CS35L34_MCLK_RATE_6P1440; cs35l34->mclk_int = freq; break; case CS35L34_MCLK_11289: -- 2.9.3

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[alsa-devel] [PATCH] ALSA: hda - Gate the mic jack on HP Z1 Gen3 AiO
by Takashi Iwai 06 Dec '16

06 Dec '16

HP Z1 Gen3 AiO with Conexant codec doesn't give an unsolicited event to the headset mic pin upon the jack plugging, it reports only to the headphone pin. It results in the missing mic switching. Let's fix up by simply gating the jack event. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_conexant.c | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index ed62748a6d55..c15c51bea26d 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -262,6 +262,7 @@ enum { CXT_FIXUP_CAP_MIX_AMP_5047, CXT_FIXUP_MUTE_LED_EAPD, CXT_FIXUP_HP_SPECTRE, + CXT_FIXUP_HP_GATE_MIC, }; /* for hda_fixup_thinkpad_acpi() */ @@ -633,6 +634,17 @@ static void cxt_fixup_cap_mix_amp_5047(struct hda_codec *codec, (1 << AC_AMPCAP_MUTE_SHIFT)); } +static void cxt_fixup_hp_gate_mic_jack(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + /* the mic pin (0x19) doesn't give an unsolicited event; + * probe the mic pin together with the headphone pin (0x16) + */ + if (action == HDA_FIXUP_ACT_PROBE) + snd_hda_jack_set_gating_jack(codec, 0x19, 0x16); +} + /* ThinkPad X200 & co with cxt5051 */ static const struct hda_pintbl cxt_pincfg_lenovo_x200[] = { { 0x16, 0x042140ff }, /* HP (seq# overridden) */ @@ -774,6 +786,10 @@ static const struct hda_fixup cxt_fixups[] = { { } } }, + [CXT_FIXUP_HP_GATE_MIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = cxt_fixup_hp_gate_mic_jack, + }, }; static const struct snd_pci_quirk cxt5045_fixups[] = { @@ -824,6 +840,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT_FIXUP_ASPIRE_DMIC), SND_PCI_QUIRK(0x1025, 0x054f, "Acer Aspire 4830T", CXT_FIXUP_ASPIRE_DMIC), SND_PCI_QUIRK(0x103c, 0x8174, "HP Spectre x360", CXT_FIXUP_HP_SPECTRE), + SND_PCI_QUIRK(0x103c, 0x8115, "HP Z1 Gen3", CXT_FIXUP_HP_GATE_MIC), SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN), SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT_FIXUP_OLPC_XO), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410), -- 2.11.0

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[alsa-devel] [PATCH v2] ALSA: hda: when comparing pin configurations, ignore assoc in addition to seq
by Kai-Heng Feng 06 Dec '16

06 Dec '16

Commit [64047d7f4912 ALSA: hda - ignore the assoc and seq when comparing pin configurations] intented to ignore both seq and assoc at pin comparing, but it only ignored seq. So that commit may still fail to match pins on some machines. Change the bitmask to also ignore assoc. v2: Use macro to do bit masking. Thanks to Hui Wang for the analysis. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/hda_auto_parser.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 4ad29f8..a03cf68 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -884,6 +884,8 @@ void snd_hda_apply_fixup(struct hda_codec *codec, int action) } EXPORT_SYMBOL_GPL(snd_hda_apply_fixup); +#define IGNORE_SEQ_ASSOC (~(AC_DEFCFG_SEQUENCE | AC_DEFCFG_DEF_ASSOC)) + static bool pin_config_match(struct hda_codec *codec, const struct hda_pintbl *pins) { @@ -901,7 +903,7 @@ static bool pin_config_match(struct hda_codec *codec, for (; t_pins->nid; t_pins++) { if (t_pins->nid == nid) { found = 1; - if ((t_pins->val & 0xfffffff0) == (cfg & 0xfffffff0)) + if ((t_pins->val & IGNORE_SEQ_ASSOC) == (cfg & IGNORE_SEQ_ASSOC)) break; else if ((cfg & 0xf0000000) == 0x40000000 && (t_pins->val & 0xf0000000) == 0x40000000) break; -- 2.10.2

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