May 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: codec: realtek: Make the node name generic
by Fabio Estevam 22 May '18

22 May '18

From: Fabio Estevam <fabio.estevam(a)nxp.com> "The name of a node should be somewhat generic, reflecting the function of the device and not its precise programming model." Do as suggested in the bindings examples. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- Documentation/devicetree/bindings/sound/rt274.txt | 2 +- Documentation/devicetree/bindings/sound/rt5514.txt | 2 +- Documentation/devicetree/bindings/sound/rt5616.txt | 2 +- Documentation/devicetree/bindings/sound/rt5645.txt | 2 +- Documentation/devicetree/bindings/sound/rt5651.txt | 2 +- Documentation/devicetree/bindings/sound/rt5663.txt | 2 +- 6 files changed, 6 insertions(+), 6 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/rt274.txt b/Documentation/devicetree/bindings/sound/rt274.txt index e9a6178..791a1bd 100644 --- a/Documentation/devicetree/bindings/sound/rt274.txt +++ b/Documentation/devicetree/bindings/sound/rt274.txt @@ -26,7 +26,7 @@ Pins on the device (for linking into audio routes) for RT274: Example: -codec: rt274@1c { +rt274: codec@1c { compatible = "realtek,rt274"; reg = <0x1c>; interrupts = <7 IRQ_TYPE_EDGE_FALLING>; diff --git a/Documentation/devicetree/bindings/sound/rt5514.txt b/Documentation/devicetree/bindings/sound/rt5514.txt index 4f33b0d..b25ed08 100644 --- a/Documentation/devicetree/bindings/sound/rt5514.txt +++ b/Documentation/devicetree/bindings/sound/rt5514.txt @@ -32,7 +32,7 @@ Pins on the device (for linking into audio routes) for I2C: Example: -codec: rt5514@57 { +rt5514: codec@57 { compatible = "realtek,rt5514"; reg = <0x57>; }; diff --git a/Documentation/devicetree/bindings/sound/rt5616.txt b/Documentation/devicetree/bindings/sound/rt5616.txt index e410858..540a4bf 100644 --- a/Documentation/devicetree/bindings/sound/rt5616.txt +++ b/Documentation/devicetree/bindings/sound/rt5616.txt @@ -26,7 +26,7 @@ Pins on the device (for linking into audio routes) for RT5616: Example: -codec: rt5616@1b { +rt5616: codec@1b { compatible = "realtek,rt5616"; reg = <0x1b>; }; diff --git a/Documentation/devicetree/bindings/sound/rt5645.txt b/Documentation/devicetree/bindings/sound/rt5645.txt index 7cee1f51..a03f9a8 100644 --- a/Documentation/devicetree/bindings/sound/rt5645.txt +++ b/Documentation/devicetree/bindings/sound/rt5645.txt @@ -69,4 +69,4 @@ codec: rt5650@1a { realtek,dmic-en = "true"; realtek,en-jd-func = "true"; realtek,jd-mode = <3>; -}; \ No newline at end of file +}; diff --git a/Documentation/devicetree/bindings/sound/rt5651.txt b/Documentation/devicetree/bindings/sound/rt5651.txt index b852218..a41199a 100644 --- a/Documentation/devicetree/bindings/sound/rt5651.txt +++ b/Documentation/devicetree/bindings/sound/rt5651.txt @@ -50,7 +50,7 @@ Pins on the device (for linking into audio routes) for RT5651: Example: -codec: rt5651@1a { +rt5651: codec@1a { compatible = "realtek,rt5651"; reg = <0x1a>; realtek,dmic-en = "true"; diff --git a/Documentation/devicetree/bindings/sound/rt5663.txt b/Documentation/devicetree/bindings/sound/rt5663.txt index 497bcfc..2338644 100644 --- a/Documentation/devicetree/bindings/sound/rt5663.txt +++ b/Documentation/devicetree/bindings/sound/rt5663.txt @@ -47,7 +47,7 @@ Pins on the device (for linking into audio routes) for RT5663: Example: -codec: rt5663@12 { +rt5663: codec@12 { compatible = "realtek,rt5663"; reg = <0x12>; interrupts = <7 IRQ_TYPE_EDGE_FALLING>; -- 2.7.4

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[alsa-devel] [PATCH 29/33] ALSA: oxfw: use match_string() helper
by Yisheng Xie 22 May '18

22 May '18

match_string() returns the index of an array for a matching string, which can be used intead of open coded variant. Cc: Clemens Ladisch <clemens(a)ladisch.de> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Yisheng Xie <xieyisheng1(a)huawei.com> --- sound/firewire/oxfw/oxfw.c | 8 +------- 1 file changed, 1 insertion(+), 7 deletions(-) diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index 413ab63..1e5b2c8 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -49,7 +49,6 @@ static bool detect_loud_models(struct fw_unit *unit) "Tapco LINK.firewire 4x6", "U.420"}; char model[32]; - unsigned int i; int err; err = fw_csr_string(unit->directory, CSR_MODEL, @@ -57,12 +56,7 @@ static bool detect_loud_models(struct fw_unit *unit) if (err < 0) return false; - for (i = 0; i < ARRAY_SIZE(models); i++) { - if (strcmp(models[i], model) == 0) - break; - } - - return (i < ARRAY_SIZE(models)); + return match_string(models, ARRAY_SIZE(models), model) >= 0; } static int name_card(struct snd_oxfw *oxfw) -- 1.7.12.4

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[alsa-devel] [asoc:for-4.18 50/211] sound//soc/soc-topology.c:1776:27: error: 'snd_soc_new_compress' undeclared
by kbuild test robot 22 May '18

22 May '18

Hi Liam, FYI, the error/warning still remains. tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-4.18 head: a6f933f63f2ffdb211dbf203dd9750449af000d3 commit: 5db6aab6f36f7560dc95f7ca340d5632b7a3be6a [50/211] ASoC: topology: Add support for compressed PCMs config: x86_64-randconfig-s2-05221211 (attached as .config) compiler: gcc-6 (Debian 6.4.0-9) 6.4.0 20171026 reproduce: git checkout 5db6aab6f36f7560dc95f7ca340d5632b7a3be6a # save the attached .config to linux build tree make ARCH=x86_64 All errors (new ones prefixed by >>): sound//soc/soc-topology.c: In function 'soc_tplg_dai_create': >> sound//soc/soc-topology.c:1776:27: error: 'snd_soc_new_compress' undeclared (first use in this function) dai_drv->compress_new = snd_soc_new_compress; ^~~~~~~~~~~~~~~~~~~~ sound//soc/soc-topology.c:1776:27: note: each undeclared identifier is reported only once for each function it appears in vim +/snd_soc_new_compress +1776 sound//soc/soc-topology.c 1746 1747 static int soc_tplg_dai_create(struct soc_tplg *tplg, 1748 struct snd_soc_tplg_pcm *pcm) 1749 { 1750 struct snd_soc_dai_driver *dai_drv; 1751 struct snd_soc_pcm_stream *stream; 1752 struct snd_soc_tplg_stream_caps *caps; 1753 int ret; 1754 1755 dai_drv = kzalloc(sizeof(struct snd_soc_dai_driver), GFP_KERNEL); 1756 if (dai_drv == NULL) 1757 return -ENOMEM; 1758 1759 if (strlen(pcm->dai_name)) 1760 dai_drv->name = kstrdup(pcm->dai_name, GFP_KERNEL); 1761 dai_drv->id = pcm->dai_id; 1762 1763 if (pcm->playback) { 1764 stream = &dai_drv->playback; 1765 caps = &pcm->caps[SND_SOC_TPLG_STREAM_PLAYBACK]; 1766 set_stream_info(stream, caps); 1767 } 1768 1769 if (pcm->capture) { 1770 stream = &dai_drv->capture; 1771 caps = &pcm->caps[SND_SOC_TPLG_STREAM_CAPTURE]; 1772 set_stream_info(stream, caps); 1773 } 1774 1775 if (pcm->compress) > 1776 dai_drv->compress_new = snd_soc_new_compress; 1777 1778 /* pass control to component driver for optional further init */ 1779 ret = soc_tplg_dai_load(tplg, dai_drv, pcm, NULL); 1780 if (ret < 0) { 1781 dev_err(tplg->comp->dev, "ASoC: DAI loading failed\n"); 1782 kfree(dai_drv); 1783 return ret; 1784 } 1785 1786 dai_drv->dobj.index = tplg->index; 1787 dai_drv->dobj.ops = tplg->ops; 1788 dai_drv->dobj.type = SND_SOC_DOBJ_PCM; 1789 list_add(&dai_drv->dobj.list, &tplg->comp->dobj_list); 1790 1791 /* register the DAI to the component */ 1792 return snd_soc_register_dai(tplg->comp, dai_drv); 1793 } 1794 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH 28/33] ALSA: dice use match_string() helper
by Yisheng Xie 22 May '18

22 May '18

match_string() returns the index of an array for a matching string, which can be used intead of open coded variant. Cc: Clemens Ladisch <clemens(a)ladisch.de> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Yisheng Xie <xieyisheng1(a)huawei.com> --- sound/firewire/dice/dice.c | 8 +------- 1 file changed, 1 insertion(+), 7 deletions(-) diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index 96bb01b..0639074 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -35,19 +35,13 @@ static bool force_two_pcm_support(struct fw_unit *unit) "SAFFIRE_PRO_40_1", }; char model[32]; - unsigned int i; int err; err = fw_csr_string(unit->directory, CSR_MODEL, model, sizeof(model)); if (err < 0) return false; - for (i = 0; i < ARRAY_SIZE(models); i++) { - if (strcmp(models[i], model) == 0) - break; - } - - return i < ARRAY_SIZE(models); + return match_string(models, ARRAY_SIZE(models), model) >= 0; } static int check_dice_category(struct fw_unit *unit) -- 1.7.12.4

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[alsa-devel] [PATCH 32/33] ASoC: max98095: use match_string() helper
by Yisheng Xie 22 May '18

22 May '18

match_string() returns the index of an array for a matching string, which can be used intead of open coded variant. Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Yisheng Xie <xieyisheng1(a)huawei.com> --- sound/soc/codecs/max98095.c | 11 ++++------- 1 file changed, 4 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 6bf2d0b..0ef42dc 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -1634,15 +1634,12 @@ static void max98095_handle_eq_pdata(struct snd_soc_component *component) static int max98095_get_bq_channel(struct snd_soc_component *component, const char *name) { - int i; - - for (i = 0; i < ARRAY_SIZE(bq_mode_name); i++) - if (strcmp(name, bq_mode_name[i]) == 0) - return i; + int ret = match_string(bq_mode_name, ARRAY_SIZE(bq_mode_name), name); /* Shouldn't happen */ - dev_err(component->dev, "Bad biquad channel name '%s'\n", name); - return -EINVAL; + if (ret < 0) + dev_err(component->dev, "Bad biquad channel name '%s'\n", name); + return ret; } static int max98095_put_bq_enum(struct snd_kcontrol *kcontrol, -- 1.7.12.4

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[alsa-devel] [PATCH 30/33] ALSA: oxygen: use match_string() helper
by Yisheng Xie 22 May '18

22 May '18

match_string() returns the index of an array for a matching string, which can be used intead of open coded variant. Cc: Clemens Ladisch <clemens(a)ladisch.de> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Yisheng Xie <xieyisheng1(a)huawei.com> --- sound/pci/oxygen/oxygen_mixer.c | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index 4ca1266..fbd8f5e 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -1052,10 +1052,9 @@ static int add_controls(struct oxygen *chip, [CONTROL_CD_CAPTURE_SWITCH] = "CD Capture Switch", [CONTROL_AUX_CAPTURE_SWITCH] = "Aux Capture Switch", }; - unsigned int i, j; struct snd_kcontrol_new template; struct snd_kcontrol *ctl; - int err; + int i, j, err; for (i = 0; i < count; ++i) { template = controls[i]; @@ -1086,11 +1085,11 @@ static int add_controls(struct oxygen *chip, err = snd_ctl_add(chip->card, ctl); if (err < 0) return err; - for (j = 0; j < CONTROL_COUNT; ++j) - if (!strcmp(ctl->id.name, known_ctl_names[j])) { - chip->controls[j] = ctl; - ctl->private_free = oxygen_any_ctl_free; - } + j = match_string(known_ctl_names, CONTROL_COUNT, ctl->id.name); + if (j >= 0) { + chip->controls[j] = ctl; + ctl->private_free = oxygen_any_ctl_free; + } } return 0; } -- 1.7.12.4

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[alsa-devel] [PATCH] ASoC: intel: skylake: fix spelling mistake: "Homogenous" -> "Homogeneous"
by Colin King 21 May '18

21 May '18

From: Colin Ian King <colin.king(a)canonical.com> Trivial fix to spelling mistake in snprintf literal string Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/intel/skylake/skl-debug.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/skylake/skl-debug.c b/sound/soc/intel/skylake/skl-debug.c index a016455a6ddb..302b40e63e6c 100644 --- a/sound/soc/intel/skylake/skl-debug.c +++ b/sound/soc/intel/skylake/skl-debug.c @@ -142,8 +142,8 @@ static ssize_t module_read(struct file *file, char __user *user_buf, mconfig->max_out_queue, ret, false); ret += snprintf(buf + ret, MOD_BUF - ret, - "Other:\n\tDomain %d\n\tHomogenous Input %s\n\t" - "Homogenous Output %s\n\tIn Queue Mask %d\n\t" + "Other:\n\tDomain %d\n\tHomogeneous Input %s\n\t" + "Homogeneous Output %s\n\tIn Queue Mask %d\n\t" "Out Queue Mask %d\n\tDMA ID %d\n\tMem Pages %d\n\t" "Module Type %d\n\tModule State %d\n", mconfig->domain, -- 2.17.0

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[alsa-devel] [PATCH] ASoC: fsl: Mark 'big-endian' property as optional
by Fabio Estevam 21 May '18

21 May '18

From: Fabio Estevam <fabio.estevam(a)nxp.com> Currently the 'big-endian' property is listed as required, which is not correct. i.MX SoCs do not need such property, so move it under 'Optional properties' entry instead. Also, fsl-sai.txt incorrectly referenced 'FTM_PWM registers', so change it to 'SAI registers', which is the intended description. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- Documentation/devicetree/bindings/sound/fsl,asrc.txt | 10 ++++++---- Documentation/devicetree/bindings/sound/fsl,esai.txt | 2 ++ Documentation/devicetree/bindings/sound/fsl,spdif.txt | 2 ++ Documentation/devicetree/bindings/sound/fsl-sai.txt | 8 +++++--- 4 files changed, 15 insertions(+), 7 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl,asrc.txt b/Documentation/devicetree/bindings/sound/fsl,asrc.txt index f5a1411..1d4d9f9 100644 --- a/Documentation/devicetree/bindings/sound/fsl,asrc.txt +++ b/Documentation/devicetree/bindings/sound/fsl,asrc.txt @@ -31,14 +31,16 @@ Required properties: it. This property is optional depending on the SoC design. - - big-endian : If this property is absent, the little endian mode - will be in use as default. Otherwise, the big endian - mode will be in use for all the device registers. - - fsl,asrc-rate : Defines a mutual sample rate used by DPCM Back Ends. - fsl,asrc-width : Defines a mutual sample width used by DPCM Back Ends. +Optional properties: + + - big-endian : If this property is absent, the little endian mode + will be in use as default. Otherwise, the big endian + mode will be in use for all the device registers. + Example: asrc: asrc@2034000 { diff --git a/Documentation/devicetree/bindings/sound/fsl,esai.txt b/Documentation/devicetree/bindings/sound/fsl,esai.txt index cacd18b..5b99143 100644 --- a/Documentation/devicetree/bindings/sound/fsl,esai.txt +++ b/Documentation/devicetree/bindings/sound/fsl,esai.txt @@ -42,6 +42,8 @@ Required properties: means all the settings for Receiving would be duplicated from Transmition related registers. +Optional properties: + - big-endian : If this property is absent, the native endian mode will be in use as default, or the big endian mode will be in use for all the device registers. diff --git a/Documentation/devicetree/bindings/sound/fsl,spdif.txt b/Documentation/devicetree/bindings/sound/fsl,spdif.txt index 38cfa75..8b324f8 100644 --- a/Documentation/devicetree/bindings/sound/fsl,spdif.txt +++ b/Documentation/devicetree/bindings/sound/fsl,spdif.txt @@ -33,6 +33,8 @@ Required properties: it. This property is optional depending on the SoC design. +Optional properties: + - big-endian : If this property is absent, the native endian mode will be in use as default, or the big endian mode will be in use for all the device registers. diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt index 740b467..dd9e597 100644 --- a/Documentation/devicetree/bindings/sound/fsl-sai.txt +++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt @@ -28,9 +28,6 @@ Required properties: pinctrl-names. See ../pinctrl/pinctrl-bindings.txt for details of the property values. - - big-endian : Boolean property, required if all the FTM_PWM - registers are big-endian rather than little-endian. - - lsb-first : Configures whether the LSB or the MSB is transmitted first for the fifo data. If this property is absent, the MSB is transmitted first as default, or the LSB @@ -48,6 +45,11 @@ Required properties: receive data by following their own bit clocks and frame sync clocks separately. +Optional properties: + + - big-endian : Boolean property, required if all the SAI + registers are big-endian rather than little-endian. + Optional properties (for mx6ul): - fsl,sai-mclk-direction-output: This is a boolean property. If present, -- 2.7.4

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[alsa-devel] [PATCH] ASoC: core: fix return code in error message
by Daniel Mack 21 May '18

21 May '18

Log the correct error code in case the .open() call to a component fails. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- sound/soc/soc-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 68d9dc930096..abc1dc7074f5 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -517,7 +517,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) if (__ret < 0) { dev_err(component->dev, "ASoC: can't open component %s: %d\n", - component->name, ret); + component->name, __ret); ret = __ret; } } -- 2.14.3

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[alsa-devel] [PATCH v9 00/15] ASoC: qcom: Add support to QDSP based Audio
by Srinivas Kandagatla 21 May '18

21 May '18

Thankyou everyone for providing feedback and testing v8 patchset. Changes since v8 (https://lkml.org/lkml/2018/5/9/491) - Fixed usecase across afe, adm, asm where user can trigger manual unbind if dsp does not respond/responds very lately. Spotted by Mark B. - rebased patchset on top of(removed patches arleady applied) https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git/?h=for-ne… This patchset aims to provide a basic version of QCOM DSP based audio support which is available in downstream andriod kernels. This patchset support audio playback on HDMI-RX, MI2S, SLIMBus and will add support to other features as we move on. QDSP has both static and dynamic modules. static modules like AFE (Audio FrontEnd), ADM (Audio Device Manager), ASM(Audio Stream Manager) and CORE to provide this audio services. All these services use APR (Asynchronous Packet Router) protocol via smd/glink transport to communicate with Application processor. More details on each module is available in there respective patch. This patchset is tested on DB820c, with HDMI audio playback, MI2S on DB410c on top of mainline, Also tested SLIMBus analog audio using wcd9355 with an additional patches. Patches are also tested on SDM845 by Rohit. Here is my test branch incase someone want to try these patches https://git.linaro.org/people/srinivas.kandagatla/linux.git/log/?h=v4.17-qd… Here is block diagram to give a quick overview of the components +---------+ +---------+ +---------+ | q6asm | |q6routing| | q6afe | | dais | <------> | mixers | <-----> | dais | +---------+ +---------+ +---------+ ^ ^ ^ | | | | +------------------+----------------+ | | | | | | v v v v v +---------+ +---------+ +---------+ | q6ASM | | q6ADM | | q6AFE | +---------+ +---------+ +---------+ ^ ^ ^ ^ | | | CPU Side | ------+---------------------+-------------------+-------- | | | | |APR(smd/glink) | | | | | +------------------+----------------+ | | | | | | +-----+--+-----------------------------------+--+------- | | | | | QDSP Side | v v v v v v +---------+ +---------+ +---------+ | ASM | <------> | ADM | <-----> | AFE | +---------+ +---------+ +---------+ ^ | +-------------------+ | ---------------------------+-------------------------- | Audio I/O | v v +--------------------------------------------------+ | Audio devices | | CODEC | HDMI-TX | PCM | SLIMBUS | I2S |MI2S |...| | | +--------------------------------------------------+ Thanks, srini Srinivas Kandagatla (15): ASoC: qdsp6: q6afe: Add q6afe driver ASoC: qdsp6: qdafe: Add SLIMBus port Support ASoC: qdsp6: q6afe: Add support to MI2S ports ASoC: qdsp6: q6afe: Add support to MI2S sysclks ASoC: qdsp6: q6adm: Add q6adm driver ASoC: qdsp6: q6asm: Add q6asm driver ASoC: qdsp6: q6asm: Add support to memory map and unmap ASoC: qdsp6: q6asm: Add support to audio stream apis ASoC: qdsp6: q6routing: Add q6routing driver ASoC: qdsp6: q6routing: Add support to all SLIMBus Mixers ASoC: qdsp6: q6routing: Add support to MI2S Mixers ASoC: qdsp6: q6afe: Add q6afe dai driver ASoC: qdsp6: q6asm: Add q6asm dai driver ASoC: qdsp6: dt-bindings: Add apq8096 machine bindings ASoC: qcom: apq8096: Add db820c machine driver .../devicetree/bindings/sound/qcom,apq8096.txt | 109 ++ sound/soc/qcom/Kconfig | 33 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/apq8096.c | 255 ++++ sound/soc/qcom/qdsp6/Makefile | 6 + sound/soc/qcom/qdsp6/q6adm.c | 646 +++++++++ sound/soc/qcom/qdsp6/q6adm.h | 27 + sound/soc/qcom/qdsp6/q6afe-dai.c | 748 +++++++++++ sound/soc/qcom/qdsp6/q6afe.c | 1068 +++++++++++++++ sound/soc/qcom/qdsp6/q6afe.h | 193 +++ sound/soc/qcom/qdsp6/q6asm-dai.c | 624 +++++++++ sound/soc/qcom/qdsp6/q6asm.c | 1399 ++++++++++++++++++++ sound/soc/qcom/qdsp6/q6asm.h | 69 + sound/soc/qcom/qdsp6/q6routing.c | 990 ++++++++++++++ sound/soc/qcom/qdsp6/q6routing.h | 9 + 15 files changed, 6178 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,apq8096.txt create mode 100644 sound/soc/qcom/apq8096.c create mode 100644 sound/soc/qcom/qdsp6/q6adm.c create mode 100644 sound/soc/qcom/qdsp6/q6adm.h create mode 100644 sound/soc/qcom/qdsp6/q6afe-dai.c create mode 100644 sound/soc/qcom/qdsp6/q6afe.c create mode 100644 sound/soc/qcom/qdsp6/q6afe.h create mode 100644 sound/soc/qcom/qdsp6/q6asm-dai.c create mode 100644 sound/soc/qcom/qdsp6/q6asm.c create mode 100644 sound/soc/qcom/qdsp6/q6asm.h create mode 100644 sound/soc/qcom/qdsp6/q6routing.c create mode 100644 sound/soc/qcom/qdsp6/q6routing.h -- 2.16.2

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