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

[alsa-devel] [PATCH v2 1/4] ASoC: wm_adsp: Use usleep_range for short delay
by Charles Keepax 26 Sep '16

26 Sep '16

Replace the 1ms msleep in wm_adsp2_ena with a usleep_range, as per normal guidance on delay functions. Also tighten up the delay a little as 1ms was quite generous. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- New since the first series of this patch chain. sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 24485ec..4188c37 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -2237,7 +2237,7 @@ static int wm_adsp2_ena(struct wm_adsp *dsp) if (val & ADSP2_RAM_RDY) break; - msleep(1); + usleep_range(250, 500); } if (!(val & ADSP2_RAM_RDY)) { -- 2.1.4

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[alsa-devel] [PATCH v2] ASoC: simple-card: add support for aux devices
by Nikita Yushchenko 26 Sep '16

26 Sep '16

This patch makes it possible to use simple-card in setups where separate amplifier chip is connected to codec's output. Changes from v1: - moved example usage from commit message to Documentation/, as suggested by Kuninori Morimoto, - fixed typo in example usage. Signed-off-by: Nikita Yushchenko <nikita.yoush(a)cogentembedded.com> --- .../devicetree/bindings/sound/simple-card.txt | 37 ++++++++++++++++++++++ sound/soc/generic/simple-card.c | 34 ++++++++++++++++++++ 2 files changed, 71 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt index 59d8628..c7a9393 100644 --- a/Documentation/devicetree/bindings/sound/simple-card.txt +++ b/Documentation/devicetree/bindings/sound/simple-card.txt @@ -22,6 +22,8 @@ Optional properties: headphones are attached. - simple-audio-card,mic-det-gpio : Reference to GPIO that signals when a microphone is attached. +- simple-audio-card,aux-devs : List of phandles pointing to auxiliary devices, such + as amplifiers, to be added to the sound card. Optional subnodes: @@ -162,3 +164,38 @@ sound { }; }; }; + +Example 3 - route audio from IMX6 SSI2 through TLV320DAC3100 codec +through TPA6130A2 amplifier to headphones: + +&i2c0 { + codec: tlv320dac3100@18 { + compatible = "ti,tlv320dac3100"; + ... + } + + amp: tpa6130a2@60 { + compatible = "ti,tpa6130a2"; + ... + } +} + +sound { + compatible = "simple-audio-card"; + ... + simple-audio-card,widgets = + "Headphone", "Headphone Jack"; + simple-audio-card,routing = + "Headphone Jack", "HPLEFT", + "Headphone Jack", "HPRIGHT", + "LEFTIN", "HPL", + "RIGHTIN", "HPR"; + simple-audio-card,aux-devs = <&amp>; + simple-audio-card,cpu { + sound-dai = <&ssi2>; + }; + simple-audio-card,codec { + sound-dai = <&codec>; + clocks = ... + }; +}; diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 43295f0..f989f34 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -417,6 +417,36 @@ dai_link_of_err: return ret; } +static int asoc_simple_card_parse_aux_devs(struct device_node *node, + struct simple_card_data *priv) +{ + struct device *dev = simple_priv_to_dev(priv); + struct device_node *aux_node; + int i, n, len; + + if (!of_find_property(node, PREFIX "aux-devs", &len)) + return 0; /* Ok to have no aux-devs */ + + n = len / sizeof(__be32); + if (n <= 0) + return -EINVAL; + + priv->snd_card.aux_dev = devm_kzalloc(dev, + n * sizeof(*priv->snd_card.aux_dev), GFP_KERNEL); + if (!priv->snd_card.aux_dev) + return -ENOMEM; + + for (i = 0; i < n; i++) { + aux_node = of_parse_phandle(node, PREFIX "aux-devs", i); + if (!aux_node) + return -EINVAL; + priv->snd_card.aux_dev[i].codec_of_node = aux_node; + } + + priv->snd_card.num_aux_devs = n; + return 0; +} + static int asoc_simple_card_parse_of(struct device_node *node, struct simple_card_data *priv) { @@ -474,6 +504,10 @@ static int asoc_simple_card_parse_of(struct device_node *node, if (ret) return ret; + ret = asoc_simple_card_parse_aux_devs(node, priv); + if (ret) + return ret; + return 0; } -- 2.1.4

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[alsa-devel] [PATCH] ASoC: tpa6130a2: fix volume setting when no stream is running
by Nikita Yushchenko 26 Sep '16

26 Sep '16

After moving tpa6130a2 power management to DAPM, if chip can be physically powered off (either reset_gpio is defined, or regulator indeed removes power), then volume change no longer works unless chip is on due to a running stream. Fix that by entering regcache cache_only mode while chip is off. Move regcache calls to tpa6130a2_power() to get them at driver init time as well. Signed-off-by: Nikita Yushchenko <nikita.yoush(a)cogentembedded.com> --- sound/soc/codecs/tpa6130a2.c | 49 +++++++++++++++++++++++--------------------- 1 file changed, 26 insertions(+), 23 deletions(-) diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index f1ea052..3b6faed 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -52,7 +52,7 @@ struct tpa6130a2_data { static int tpa6130a2_power(struct tpa6130a2_data *data, bool enable) { - int ret; + int ret = 0, ret2; if (enable) { ret = regulator_enable(data->supply); @@ -64,20 +64,34 @@ static int tpa6130a2_power(struct tpa6130a2_data *data, bool enable) /* Power on */ if (data->power_gpio >= 0) gpio_set_value(data->power_gpio, 1); + + /* Sync registers */ + regcache_cache_only(data->regmap, false); + ret = regcache_sync(data->regmap); + if (ret != 0) { + dev_err(data->dev, + "Failed to sync registers: %d\n", ret); + goto regcache_sync_failed; + } } else { + /* Powered off device does not retain registers. While device + * is off, any register updates (i.e. volume changes) should + * happen in cache only. + */ + regcache_mark_dirty(data->regmap); +regcache_sync_failed: + regcache_cache_only(data->regmap, true); + /* Power off */ if (data->power_gpio >= 0) gpio_set_value(data->power_gpio, 0); - ret = regulator_disable(data->supply); - if (ret != 0) { + ret2 = regulator_disable(data->supply); + if (ret2 != 0) { dev_err(data->dev, - "Failed to disable supply: %d\n", ret); - return ret; + "Failed to disable supply: %d\n", ret2); + return ret ? ret : ret2; } - - /* device regs does not match the cache state anymore */ - regcache_mark_dirty(data->regmap); } return ret; @@ -88,25 +102,14 @@ static int tpa6130a2_power_event(struct snd_soc_dapm_widget *w, { struct snd_soc_component *c = snd_soc_dapm_to_component(w->dapm); struct tpa6130a2_data *data = snd_soc_component_get_drvdata(c); - int ret; - /* before widget power up */ if (SND_SOC_DAPM_EVENT_ON(event)) { - /* Turn on the chip */ - tpa6130a2_power(data, true); - /* Sync the registers */ - ret = regcache_sync(data->regmap); - if (ret < 0) { - dev_err(c->dev, "Failed to initialize chip\n"); - tpa6130a2_power(data, false); - return ret; - } - /* after widget power down */ + /* Before widget power up: turn chip on, sync registers */ + return tpa6130a2_power(data, true); } else { - tpa6130a2_power(data, false); + /* After widget power down: turn chip off */ + return tpa6130a2_power(data, false); } - - return 0; } /* -- 2.1.4

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[alsa-devel] [asoc:topic/intel 53/55] sound/soc/codecs/hdac_hdmi.c:1028:5: warning: passing argument 4 of 'snd_hdac_acomp_get_eld' from incompatible pointer type
by kbuild test robot 26 Sep '16

26 Sep '16

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/intel head: 2d995e5dc283adbfbf9c1eb81bf35ca7af2d22a6 commit: fe27f4e0545d3fc1b0518fafb4fe0460d757651d [53/55] ASoC: hdac_hdmi: use audio component framework to read ELD config: x86_64-randconfig-n0-09251512 (attached as .config) compiler: gcc-4.8 (Debian 4.8.4-1) 4.8.4 reproduce: git checkout fe27f4e0545d3fc1b0518fafb4fe0460d757651d # save the attached .config to linux build tree make ARCH=x86_64 All warnings (new ones prefixed by >>): sound/soc/codecs/hdac_hdmi.c: In function 'hdac_hdmi_present_sense': sound/soc/codecs/hdac_hdmi.c:1028:5: warning: passing argument 3 of 'snd_hdac_acomp_get_eld' makes pointer from integer without a cast [enabled by default] ELD_MAX_SIZE); ^ In file included from sound/soc/codecs/hdac_hdmi.c:30:0: include/sound/hda_i915.h:14:5: note: expected 'bool *' but argument is of type 'int' int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, ^ >> sound/soc/codecs/hdac_hdmi.c:1028:5: warning: passing argument 4 of 'snd_hdac_acomp_get_eld' from incompatible pointer type [enabled by default] ELD_MAX_SIZE); ^ In file included from sound/soc/codecs/hdac_hdmi.c:30:0: include/sound/hda_i915.h:14:5: note: expected 'char *' but argument is of type 'bool *' int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, ^ sound/soc/codecs/hdac_hdmi.c:1028:5: warning: passing argument 5 of 'snd_hdac_acomp_get_eld' makes integer from pointer without a cast [enabled by default] ELD_MAX_SIZE); ^ In file included from sound/soc/codecs/hdac_hdmi.c:30:0: include/sound/hda_i915.h:14:5: note: expected 'int' but argument is of type 'char *' int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, ^ sound/soc/codecs/hdac_hdmi.c:1028:5: error: too many arguments to function 'snd_hdac_acomp_get_eld' ELD_MAX_SIZE); ^ In file included from sound/soc/codecs/hdac_hdmi.c:30:0: include/sound/hda_i915.h:14:5: note: declared here int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, ^ vim +/snd_hdac_acomp_get_eld +1028 sound/soc/codecs/hdac_hdmi.c 1012 1013 return 0; 1014 } 1015 1016 static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin) 1017 { 1018 struct hdac_ext_device *edev = pin->edev; 1019 struct hdac_hdmi_priv *hdmi = edev->private_data; 1020 struct hdac_hdmi_pcm *pcm; 1021 int size; 1022 1023 mutex_lock(&hdmi->pin_mutex); 1024 pin->eld.monitor_present = false; 1025 1026 size = snd_hdac_acomp_get_eld(&edev->hdac, pin->nid, -1, 1027 &pin->eld.monitor_present, pin->eld.eld_buffer, > 1028 ELD_MAX_SIZE); 1029 1030 if (size > 0) { 1031 size = min(size, ELD_MAX_SIZE); 1032 if (hdac_hdmi_parse_eld(edev, pin) < 0) 1033 size = -EINVAL; 1034 } 1035 1036 if (size > 0) { --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] DAPM: codec to codec link documentation[RFC]
by anish kumar 26 Sep '16

26 Sep '16

Hello, I had a tough time figuring out how to get codec to codec link to work. So thought of making it easier for other people who wants to use the same. I will appreciate any inputs for below documentation. Excuse my diagrams if it doesn't render on your browser. I am still learning as how to best draw txt diagrams. Mostly the flow of audio is always from CPU to codec so your system will look as below: ---------- ---------- | | dai | | CPU -------> codec | | | | ---------- ---------- In case your system looks as below: ---------- | | codec-2 | | ---------- | dai-2 | ---------- ---------- | | dai-1 | | CPU -------> codec-1 | | | | ---------- ---------- | dai-3 | ---------- | | codec-3 | | ---------- Suppose codec-2 is a bluetooth chip and codec-3 is connected to a speaker and you have a below scenario: codec-2 will receive the audio data and the user wants to play that audio through codec-3 without involving the CPU.This aforementioned case is the ideal case when codec to codec connection should be used. Your dai_link should appear as below in your machine file: static const struct snd_soc_pcm_stream dummy_params = { .formats = SNDRV_PCM_FMTBIT_S24_LE, .rate_min = 48000, .rate_max = 48000, .channels_min = 2, .channels_max = 2, }; { .name = "your_name", .stream_name = "your_stream_name", .cpu_dai_name = "snd-soc-dummy-dai", .codec_name = "codec-2, .codec_dai_name = "codec-2-dai_name", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, .ignore_suspend = 1, .params = &dummy_params, }, { .name = "your_name", .stream_name = "your_stream_name", .cpu_dai_name = "snd-soc-dummy-dai", .codec_name = "codec-3, .codec_dai_name = "codec-3-dai_name", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, .ignore_suspend = 1, .params = &dummy_params, }, Note the "params" callback which lets the dapm know that this dai_link is a codec to codec connection. Also, in above code cpu_dai should be replaced with your actual cpu dai but in case you don't have a actual cpu dai then dummy will do. You can browse the speyside.c for an actual example code in mainline. In dapm core a route is created between cpu_dai playback widget and codec_dai capture widget for playback path and vice-versa is true for capture path. In order for this aforementioned route to get triggered, DAPM needs to find a valid endpoint which could be either a sink or source widget corresponding to playback and capture path respectively. Below is what you can use it to trigger the widgets provided you have stream name ending with "Playback" and "Capture" for cpu and codec dai's. static const struct snd_soc_dapm_widget aif_dapm_widgets[] = { SND_SOC_DAPM_SPK("dummyspk", NULL), SND_SOC_DAPM_MIC("dummymic", NULL), }; static const struct snd_soc_dapm_route audio_i2s_map[] = { {"dummyspk", NULL, "Playback"}, {"Capture", NULL, "dummymic"}, }; Thanks,

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[alsa-devel] [PATCH] ALSA: hda - Adding one more ALC255 pin definition for headset problem
by Hui Wang 26 Sep '16

26 Sep '16

We have two new Dell laptop models, they have the same ALC255 pin definition, but not in the pin quirk table yet, as a result, the headset microphone can't work. After adding the definition in the table, the headset microphone works well. Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1fe8750..b58e8c7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5855,6 +5855,10 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x14, 0x90170120}, {0x21, 0x02211030}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x14, 0x90170110}, + {0x1b, 0x02011020}, + {0x21, 0x0221101f}), + SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x14, 0x90170130}, {0x1b, 0x01014020}, {0x21, 0x0221103f}), -- 1.9.1

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[alsa-devel] [PATCH] ALSA: usb-line6: use the same declaration as definition in header for MIDI manufacturer ID
by Takashi Sakamoto 25 Sep '16

25 Sep '16

Currently, usb-line6 module exports an array of MIDI manufacturer ID and usb-pod module uses it. However, the declaration is not the definition in common header. The difference is explicit length of array. Although compiler calculates it and everything goes well, it's better to use the same representation between definition and declaration. This commit fills the length of array for usb-line6 module. As a small good sub-effect, this commit suppress below warnings from static analysis by sparse v0.5.0. sound/usb/line6/driver.c:274:43: error: cannot size expression sound/usb/line6/driver.c:275:16: error: cannot size expression sound/usb/line6/driver.c:276:16: error: cannot size expression sound/usb/line6/driver.c:277:16: error: cannot size expression Fixes: 705ececd1c60 ("Staging: add line6 usb driver") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/usb/line6/driver.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c index f9224ec..ad7efc0 100644 --- a/sound/usb/line6/driver.c +++ b/sound/usb/line6/driver.c @@ -30,7 +30,7 @@ /* This is Line 6's MIDI manufacturer ID. */ -const unsigned char line6_midi_id[] = { +const unsigned char line6_midi_id[3] = { 0x00, 0x01, 0x0c }; EXPORT_SYMBOL_GPL(line6_midi_id); -- 2.7.4

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[alsa-devel] [PATCH v5 0/3] ASoC: sun4i-codec: Distinguish sun4i from sun7i
by Danny Milosavljevic 25 Sep '16

25 Sep '16

ASoC: sun4i-codec: Introduce mechanism to detect sun7i and provide a different regmap different compared to sun4i Allwinner A10. The controls will be extended in a forthcoming patch - it is necessary to distinguish between sun4i and sun7i controls because they have different registers. Renamed SUN4I_CODEC_AC_SYS_VERI to SUN7I_CODEC_AC_DAC_CAL and renamed SUN4I_CODEC_AC_MIC_PHONE_CAL to SUN7I_CODEC_AC_MIC_PHONE_CAL because these are actually not present on Allwinner A10. Handle quirks by regmap config and codec and select the correct quirks automatically. Danny Milosavljevic (3): ASoC: sun4i-codec: rename "sun4i_codec_widgets" to "sun4i_codec_controls" for consistency with the struct field name. ASoC: rename "SUN4I_CODEC_SYS_VERI" to "SUN7I_CODEC_AC_DAC_CAL"; rename "SUN4I_CODEC_AC_MIC_PHONE_CAL" to "SUN7I_CODEC_AC_MIC_PHONE_CAL". ASoC: sun4i-codec: Add custom regmap configs for the A10 and A20 variants. sound/soc/sunxi/sun4i-codec.c | 50 +++++++++++++++++++++++++++++++++++-------- 1 file changed, 41 insertions(+), 9 deletions(-)

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[alsa-devel] [PATCH v2 0/4] ASoC: Intel: Skylake: Add support for generating unique module ids
by Vinod Koul 25 Sep '16

25 Sep '16

Driver needs to send unique module instance id to firmware while creating the module and uses this id to communicate with DSP for setting parameters while audio use case is ongoing. But, we have upper bound of instance ID. The current IDs are coming from topology but it doesn't know the upper bound and can't assign unique id's subject to upper bounds as we can create a big graph but not all parts running at same time. So this series first adds APIs to allocate/free a 128bit unique id which is built on top of ffz(). It also adds a table for finding modules of neighbouring modules. Then driver patches are updated to use this. Dharageswari R (4): ASoC: Intel: Skylake: Add module instance id generation APIs ASoC: Intel: Skylake: Use private instance id of modules in IPC ASoC: Intel: Skylake: Add table for module id for quick ref ASoC: Intel: Skylake: Update to use instance ids generated sound/soc/intel/skylake/skl-messages.c | 24 +++-- sound/soc/intel/skylake/skl-sst-dsp.h | 6 ++ sound/soc/intel/skylake/skl-sst-utils.c | 150 ++++++++++++++++++++++++++- sound/soc/intel/skylake/skl-topology.c | 32 +++++- sound/soc/intel/skylake/skl-topology.h | 11 ++ sound/soc/intel/skylake/skl-tplg-interface.h | 3 +- 6 files changed, 211 insertions(+), 15 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH 0/9] ASoC: Intel: Skylake: Driver updates
by Vinod Koul 25 Sep '16

25 Sep '16

So we had some driver updates collated, time to send them upstream :) - manifest data size was not checked so that is fixed - previous conversion to use core parsing for HDA capabilities was wrong at one place, that is fixed. - DMA config size was wrong, that is also fixed - Add 32bit format - The module instance IDs are manged in driver now. Dharageswari R (5): ASoC: Intel: Skylake: Unload all the loadable modules ASoC: Intel: Skylake: Create dynamic instance ids for DSP modules ASoC: Intel: Skylake: Use private instance id of modules in IPC ASoC: Intel: Skylake: Table for module instance id and private id ASoC: Intel: Skylake: Override the actual instance id's to pvt_id's Jeeja KP (1): ASoC: Intel: Skylake: Fix DMA control config size Samaga Krishna (1): ASoC: Intel: Skylake: Add 32bit support Vinod Koul (2): ASoC: Intel: Skylake: check manifest size ASoC: Intel: Skylake: Fix the inverted logic check sound/soc/intel/skylake/skl-messages.c | 24 +++-- sound/soc/intel/skylake/skl-pcm.c | 5 +- sound/soc/intel/skylake/skl-sst-dsp.h | 6 ++ sound/soc/intel/skylake/skl-sst-utils.c | 133 ++++++++++++++++++++++++++- sound/soc/intel/skylake/skl-topology.c | 44 ++++++++- sound/soc/intel/skylake/skl-topology.h | 13 ++- sound/soc/intel/skylake/skl-tplg-interface.h | 3 +- 7 files changed, 208 insertions(+), 20 deletions(-) -- 1.9.1

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