August 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: wm8741: Simplify wm8741_set_dai_sysclk implementation
by Axel Lin 06 Aug '15

06 Aug '15

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/wm8741.c | 41 ++++++++++++++--------------------------- 1 file changed, 14 insertions(+), 27 deletions(-) diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 430fa7d..9723e44 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -278,51 +278,38 @@ static int wm8741_set_dai_sysclk(struct snd_soc_dai *codec_dai, switch (freq) { case 0: wm8741->sysclk_constraints = NULL; - wm8741->sysclk = freq; - return 0; - + break; case 11289600: wm8741->sysclk_constraints = &constraints_11289; - wm8741->sysclk = freq; - return 0; - + break; case 12288000: wm8741->sysclk_constraints = &constraints_12288; - wm8741->sysclk = freq; - return 0; - + break; case 16384000: wm8741->sysclk_constraints = &constraints_16384; - wm8741->sysclk = freq; - return 0; - + break; case 16934400: wm8741->sysclk_constraints = &constraints_16934; - wm8741->sysclk = freq; - return 0; - + break; case 18432000: wm8741->sysclk_constraints = &constraints_18432; - wm8741->sysclk = freq; - return 0; - + break; case 22579200: case 33868800: wm8741->sysclk_constraints = &constraints_22579; - wm8741->sysclk = freq; - return 0; - + break; case 24576000: wm8741->sysclk_constraints = &constraints_24576; - wm8741->sysclk = freq; - return 0; - + break; case 36864000: wm8741->sysclk_constraints = &constraints_36864; - wm8741->sysclk = freq; - return 0; + break; + default: + return -EINVAL; } - return -EINVAL; + + wm8741->sysclk = freq; + return 0; } static int wm8741_set_dai_fmt(struct snd_soc_dai *codec_dai, -- 2.1.0

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[alsa-devel] query on upstream ASoC driver split across SOUND and DRM subsystems
by Bayyavarapu, Maruthi 06 Aug '15

06 Aug '15

Hi, I want to send AMD ASoC driver for review. This driver has a dependency with AMD's 'DRM' subsystem components, which are not upstreamed yet. Please suggest, whether this ASoC driver can be upstreamed via 'DRM' tree or any alternate approach is preferred. The link to ASoC driver is : http://cgit.freedesktop.org/~agd5f/linux/commit/?h=acp-wip&id=2b67a44150988… ASoC commits and dependencies are currently in 'acp-wip' branch of : git://people.freedesktop.org/~agd5f/linux Thanks, Maruthi

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[alsa-devel] [PATCH v2 0/3] ASoC: topology: UAPI updates for TLV, ops and types
by Liam Girdwood 06 Aug '15

06 Aug '15

This series updates the topology UAPI to allow easier addition of new TLV object types, generic operations and added a private data type for the kernel topology core. Changes since V1 * Split into several patches for UAPI changes. * Fixed {} usage in if statement. * Added __le32 data[SND_SOC_TPLG_TLV_SIZE] back to struct snd_soc_tplg_ctl_tlv Liam Girdwood (1): ASoC: topology: Add private data type and bump ABI version to 3 Mengdong Lin (2): ASoC: topology: Update TLV support so we can support more TLV types ASoC: topology: Add ops support to byte controls UAPI include/uapi/sound/asoc.h | 33 ++++++++++++++++++--------- sound/soc/soc-topology.c | 58 +++++++++++++++++++++++++++++++++-------------- 2 files changed, 63 insertions(+), 28 deletions(-) -- 2.1.4

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[alsa-devel] [PATCH] ASoC: rl6231: Simplify DMIC divider calculation expression
by Anatol Pomozov 06 Aug '15

06 Aug '15

Existing implementation checks all divider values and tracks 'red' proximity value for the frequency. But as divider array is monotonically increasing the first divider that gives DMIC rate in 3MHz range is the best one we should use. No need for 'red' zone tracking. Additionally make sure that DMIC frequency is higher 1MHz. Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- sound/soc/codecs/rl6231.c | 30 +++++++++++++++--------------- 1 file changed, 15 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/rl6231.c b/sound/soc/codecs/rl6231.c index 57e51c1..aca479f 100644 --- a/sound/soc/codecs/rl6231.c +++ b/sound/soc/codecs/rl6231.c @@ -62,31 +62,31 @@ int rl6231_get_pre_div(struct regmap *map, unsigned int reg, int sft) EXPORT_SYMBOL_GPL(rl6231_get_pre_div); /** - * rl6231_calc_dmic_clk - Calculate the parameter of dmic. + * rl6231_calc_dmic_clk - Calculate the frequency divider parameter of dmic. * * @rate: base clock rate. * - * Choose dmic clock between 1MHz and 3MHz. - * It is better for clock to approximate 3MHz. + * Choose divider parameter that gives the highest possible DMIC frequency in + * 1MHz - 3MHz range. */ int rl6231_calc_dmic_clk(int rate) { - int div[] = {2, 3, 4, 6, 8, 12}, idx = -EINVAL; - int i, red, bound, temp; + int div[] = {2, 3, 4, 6, 8, 12}; + int i; + + if (rate < 1000000 * div[0]) { + pr_warn("Base clock rate %d is too low\n", rate); + return -EINVAL; + } - red = 3000000 * 12; for (i = 0; i < ARRAY_SIZE(div); i++) { - bound = div[i] * 3000000; - if (rate > bound) - continue; - temp = bound - rate; - if (temp < red) { - red = temp; - idx = i; - } + /* find divider that gives DMIC frequency below 3MHz */ + if (3000000 * div[i] >= rate) + return i; } - return idx; + pr_warn("Base clock rate %d is too high\n", rate); + return -EINVAL; } EXPORT_SYMBOL_GPL(rl6231_calc_dmic_clk); -- 2.5.0.rc2.392.g76e840b

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[alsa-devel] [PATCH v2] ASoC: topology: Use kcalloc instead of kzalloc for array allocation
by Axel Lin 06 Aug '15

06 Aug '15

Also remove unnecessary memset. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- v2: One more kcalloc conversion in soc_tplg_dapm_widget_dmixer_create sound/soc/soc-topology.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 4dadb5e..e4b7fc6 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1069,7 +1069,7 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dmixer_create( struct snd_soc_tplg_mixer_control *mc; int i, err; - kc = kzalloc(sizeof(*kc) * num_kcontrols, GFP_KERNEL); + kc = kcalloc(num_kcontrols, sizeof(*kc), GFP_KERNEL); if (kc == NULL) return NULL; @@ -1250,7 +1250,7 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dbytes_create( struct snd_kcontrol_new *kc; int i, err; - kc = kzalloc(sizeof(*kc) * count, GFP_KERNEL); + kc = kcalloc(count, sizeof(*kc), GFP_KERNEL); if (!kc) return NULL; @@ -1273,7 +1273,6 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dbytes_create( "ASoC: adding bytes kcontrol %s with access 0x%x\n", be->hdr.name, be->hdr.access); - memset(kc, 0, sizeof(*kc)); kc[i].name = be->hdr.name; kc[i].private_value = (long)sbe; kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; -- 2.1.0

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[alsa-devel] Verb for 0x17 pin config
by Kailang 06 Aug '15

06 Aug '15

Hi Takashi, commit http://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/?id=977e… ALSA: hda/realtek - Support Dell headset mode for ALC298 Dell had use verb to do speaker preset. So, The same SSID maybe 0x17 verb was different. 0x17 0x90170110 0x17 0x90170140 0x17 0x90170150 Upper verb will show on this case. Could I change it to quirk for SSID? BR, Kailang

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Re: [alsa-devel] [ALC668]: Asus N751JK - mic selection not working
by Raymond Yau 06 Aug '15

06 Aug '15

>> >> Ports: >> >> analog-output-speaker: Speakers (priority: 10000) >> analog-output-headphones: Headphones (priority: 9000, not available) >> Active Port: analog-output-speaker >> >> >> When you select model=dell-headset-multi, the driver create "Headphone Mic Jack" kctl with type 1 which is SND_JACK_HEADPHONE >> >> type 2 is SND_JACK_MICROPHONE >> >> type 3 is SND_JACK_HEADSET but hda jack does not use this type >> >> https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/include/s… >> >> JACK created Headphone Mic, type 1 >> CTRL: add: Headphone Mic Jack:0 >> JACK report Headphone Mic, status 0 >> JACK created Headset Mic Phantom, type 2 >> CTRL: add: Headset Mic Phantom Jack:0 >> JACK report Headset Mic Phantom, status 2 >> CTL Notify: Headset Mic Phantom Jack:0, mask=1 >> JACK created Internal Mic Phantom, type 2 >> CTRL: add: Internal Mic Phantom Jack:0 >> JACK report Internal Mic Phantom, status 2 >> CTL Notify: Internal Mic Phantom Jack:0, mask=1 >> JACK created Speaker Front Phantom, type 4 >> CTRL: add: Speaker Front Phantom Jack:0 >> JACK report Speaker Front Phantom, status 4 >> CTL Notify: Speaker Front Phantom Jack:0, mask=1 >> JACK created External Subwoofer, type 4 >> CTRL: add: External Subwoofer Jack:0 >> >> The problem of hda jack kctls are those retaskable I/O jack controls >> >> 1) multi io jacks in those 3 stack desktop >> 2) headphone/mic jack in asus netbook >> 3) dell-headset-multi combo jack >> >> the role of the jack in 1) change from input to output when "Channel mode" change from "2ch" to "4ch" or "6ch" >> >> the role of the jack in 2) and 3) change from Output to Input when "Capture Source" change from Internal Mic/Headset Mic to Headphone Mic >> >> In your case, Headphone Mic Jack kctl return the presence of HP jack except when the combo jack is set to support conventional mic (i.e. when the driver change pin ctl of hp pin node 0x15 to zero and set pin ctl of mic pin to IN), >> >> The driver already unmute the internal speaker when you change capture source to Headphone Mic >> >> The only point is whether pulseaudio still regard this Headphone Mic Jack as the availability of the headphone port when the capture source is Headphone Mic >> >> http://cgit.freedesktop.org/pulseaudio/pulseaudio/commit/?id=ca4942e89cf462… >> >> > > > I really appreciate your efforts in this matter. Unfortunately I understand parts of our talks and I am unsure what I can do. > I am not a kernel / driver developer. > > Are there any steps I can do knowing my limited skills on driver development? > If so, would you mind to describe them step by step, so this beginner understands? :D Don't forget cc David since he implement dell-headset-multi https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… static void alc_update_headset_mode(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + hda_nid_t mux_pin = spec->gen.imux_pins[spec->gen.cur_mux[0]]; + hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + + int new_headset_mode; if (!snd_hda_jack_detect(codec, hp_pin)) + new_headset_mode = ALC_HEADSET_MODE_UNPLUGGED; + else if (mux_pin == spec->headset_mic_pin) + new_headset_mode = ALC_HEADSET_MODE_HEADSET; + else if (mux_pin == spec->headphone_mic_pin) + new_headset_mode = ALC_HEADSET_MODE_MIC; + else + new_headset_mode = ALC_HEADSET_MODE_HEADPHONE; + + if (new_headset_mode == spec->current_headset_mode) + return; + + switch (new_headset_mode) { + case ALC_HEADSET_MODE_UNPLUGGED: + alc_headset_mode_unplugged(codec); + spec->gen.hp_jack_present = false; + break; + case ALC_HEADSET_MODE_HEADSET: + if (spec->current_headset_type == ALC_HEADSET_TYPE_UNKNOWN) + alc_determine_headset_type(codec); + if (spec->current_headset_type == ALC_HEADSET_TYPE_CTIA) + alc_headset_mode_ctia(codec); + else if (spec->current_headset_type == ALC_HEADSET_TYPE_OMTP) + alc_headset_mode_omtp(codec); + spec->gen.hp_jack_present = true; + break; + case ALC_HEADSET_MODE_MIC: + alc_headset_mode_mic_in(codec, hp_pin, spec->headphone_mic_pin); + spec->gen.hp_jack_present = false; + break; + case ALC_HEADSET_MODE_HEADPHONE: + alc_headset_mode_default(codec); + spec->gen.hp_jack_present = true; + break; + } + if (new_headset_mode != ALC_HEADSET_MODE_MIC) { + snd_hda_set_pin_ctl_cache(codec, hp_pin, + AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN); + if (spec->headphone_mic_pin) + snd_hda_set_pin_ctl_cache(codec, spec->headphone_mic_pin, + PIN_VREFHIZ); + } + spec->current_headset_mode = new_headset_mode; + + snd_hda_gen_update_outputs(codec); +} You have to post the output of pactl list sources when the is no jack plugged, headphone plugged, headset plugged and mic jack plugged The logic is defined in alc_update_headset_mode() Although auto mic selection is disabled when the combo jack can support conventional mic and headset mic The capture source of these kind of laptop is not as same as the capture source of ac97 codec of pci sound cards, it is not really manually selected in sound preference When there is no jack plugged in, the capture source should be internal mic. As the codec cannot differentitate headset, headphone and mic when there is jack plugged in and driver use mux_pin to determine the type of the jack In pulseaudio term, [Jack Headphone Mic] can determine the availability of internal mic is YES when unplugged but unknown when plugged http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mix… [Jack Headphone Mic] The availability of headphone mic is unknown when plugged and NO when unplugged http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mix… [Jack Headphone Mic] The availability of headset mic is unknown when plugged and NO when unplugged http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mix… It is not exactly UNKNOWN but determined by the capture source when plugged

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[alsa-devel] [PATCH 5/6][RESEND] ASoC: fsl_ssi: instantiate AC'97 CODEC
by Maciej S. Szmigiero 05 Aug '15

05 Aug '15

Instantiate AC'97 CODEC in fsl_ssi driver AC'97 mode. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- This is a resend without changes, to keep the whole series together. sound/soc/fsl/fsl_ssi.c | 21 +++++++++++++++++++++ 1 files changed, 21 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 154bcf6..8e5ff5e 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1460,6 +1460,27 @@ done: _fsl_ssi_set_dai_fmt(&pdev->dev, ssi_private, ssi_private->dai_fmt); + if (fsl_ssi_is_ac97(ssi_private)) { + u32 ssi_idx; + + ret = of_property_read_u32(np, "cell-index", &ssi_idx); + if (ret) { + dev_err(&pdev->dev, "cannot get SSI index property\n"); + goto error_sound_card; + } + + ssi_private->pdev = + platform_device_register_data(NULL, + "ac97-codec", ssi_idx, NULL, 0); + if (IS_ERR(ssi_private->pdev)) { + ret = PTR_ERR(ssi_private->pdev); + dev_err(&pdev->dev, + "failed to register AC97 codec platform: %d\n", + ret); + goto error_sound_card; + } + } + return 0; error_sound_card:

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[alsa-devel] [PATCH 4/6][RESEND] ASoC: fsl_ssi: add AC'97 ops setting check and cleanup
by Maciej S. Szmigiero 05 Aug '15

05 Aug '15

Check whether setting AC'97 ops succeeded and clean them on removal so the fsl_ssi driver can be reloaded. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- This is a resend without changes, to keep the whole series together. sound/soc/fsl/fsl_ssi.c | 9 ++++++++- 1 files changed, 8 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 7f4f0b9..154bcf6 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1340,7 +1340,11 @@ static int fsl_ssi_probe(struct platform_device *pdev) fsl_ac97_data = ssi_private; - snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev); + ret = snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev); + if (ret) { + dev_err(&pdev->dev, "could not set AC'97 ops\n"); + return ret; + } } else { /* Initialize this copy of the CPU DAI driver structure */ memcpy(&ssi_private->cpu_dai_drv, &fsl_ssi_dai_template, @@ -1480,6 +1484,9 @@ static int fsl_ssi_remove(struct platform_device *pdev) if (ssi_private->soc->imx) fsl_ssi_imx_clean(pdev, ssi_private); + if (fsl_ssi_is_ac97(ssi_private)) + snd_soc_set_ac97_ops(NULL); + return 0; }

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[alsa-devel] [PATCH 3/6][RESEND] ASoC: fsl_ssi: enable AC'97 asymmetric rates
by Maciej S. Szmigiero 05 Aug '15

05 Aug '15

AC'97 bus can support asymmetric playback/capture rates so enable them in this case in fsl_ssi driver. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- This is a resend without changes, to keep the whole series together. sound/soc/fsl/fsl_ssi.c | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index a83b900..7f4f0b9 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1377,7 +1377,9 @@ static int fsl_ssi_probe(struct platform_device *pdev) /* Are the RX and the TX clocks locked? */ if (!of_find_property(np, "fsl,ssi-asynchronous", NULL)) { - ssi_private->cpu_dai_drv.symmetric_rates = 1; + if (!fsl_ssi_is_ac97(ssi_private)) + ssi_private->cpu_dai_drv.symmetric_rates = 1; + ssi_private->cpu_dai_drv.symmetric_channels = 1; ssi_private->cpu_dai_drv.symmetric_samplebits = 1; }

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