February 2020 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] [PATCH 2/2] ASoC: rt5682: Enable PLL2 function
by Mark Brown 11 Feb '20

11 Feb '20

On Thu, Jan 30, 2020 at 10:51:46AM +0000, Derek [方德義] wrote: > Enable RT5682 PLL2 function to implement the more complex > frequency conversion. This doesn't apply against current code, please check and resend.

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Re: [alsa-devel] [PATCH 1/2] ASoC: rl6231: Add new supports on rl6231
by Mark Brown 11 Feb '20

11 Feb '20

On Thu, Jan 30, 2020 at 10:35:07AM +0000, Derek [方德義] wrote: > 1. Increases the max limit of PLL input frequency > on RL6231 shared support. > 2. Add a new pll preset map. > --- You're missing a Signed-off-by here, please see submitting-patches.rst.

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Re: [alsa-devel] [PATCH] ASoC: rt1015: fix typo for bypass boost control
by Mark Brown 11 Feb '20

11 Feb '20

On Thu, Jan 30, 2020 at 10:18:11AM +0000, Jack Yu wrote: > Fix typo for "Bypass Boost" control. > > Signed-off-by: Jack Yu <jack.yu(a)realtek.com> This doesn't apply against current code, please check and resend.

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[alsa-devel] [PATCH] ASoC: max98357a: add speaker switch
by Tzung-Bi Shih 11 Feb '20

11 Feb '20

Some machine may share the same I2S lines for multiple codecs. For example, mediatek/mt8183/mt8183-da7219-max98357 shares the same lines between max98357a and da7219. When writing audio data through the I2S lines, all codecs on the lines would try to generate sound if they accepts DO line. As a result, multiple codecs generate sound at a time. Adds a separate switch to max98357a. Userspace program has choices to turn on or off the switch. Note that, userspace program should change the switch before opening the stream. The switch won't take effects if the stream is already there. Default value of the switch is on to not break existing driver usages (who are unlikely aware of existence of the switch). Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- sound/soc/codecs/max98357a.c | 39 +++++++++++++++++++++++++++++++++++- 1 file changed, 38 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index 16313b973eaa..a2c3be69a0ee 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -22,6 +22,7 @@ struct max98357a_priv { struct gpio_desc *sdmode; unsigned int sdmode_delay; + int spk_switch; }; static int max98357a_daiops_trigger(struct snd_pcm_substream *substream, @@ -29,7 +30,7 @@ static int max98357a_daiops_trigger(struct snd_pcm_substream *substream, { struct max98357a_priv *max98357a = snd_soc_dai_get_drvdata(dai); - if (!max98357a->sdmode) + if (!max98357a->sdmode || !max98357a->spk_switch) return 0; switch (cmd) { @@ -49,6 +50,37 @@ static int max98357a_daiops_trigger(struct snd_pcm_substream *substream, return 0; } +static int max98357a_get_spk_switch(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct max98357a_priv *max98357a = + snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = max98357a->spk_switch; + return 0; +} + +static int max98357a_put_spk_switch(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct max98357a_priv *max98357a = + snd_soc_component_get_drvdata(component); + + max98357a->spk_switch = ucontrol->value.integer.value[0]; + dev_info(component->dev, + "put speaker switch: %d\n", max98357a->spk_switch); + return 0; +} + +static const struct snd_kcontrol_new max98357a_snd_controls[] = { + SOC_SINGLE_BOOL_EXT("Speaker Switch", 0, + max98357a_get_spk_switch, max98357a_put_spk_switch), +}; + static const struct snd_soc_dapm_widget max98357a_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("Speaker"), }; @@ -58,6 +90,8 @@ static const struct snd_soc_dapm_route max98357a_dapm_routes[] = { }; static const struct snd_soc_component_driver max98357a_component_driver = { + .controls = max98357a_snd_controls, + .num_controls = ARRAY_SIZE(max98357a_snd_controls), .dapm_widgets = max98357a_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(max98357a_dapm_widgets), .dapm_routes = max98357a_dapm_routes, @@ -117,6 +151,9 @@ static int max98357a_platform_probe(struct platform_device *pdev) "default: no delay\n"); } + /* For drivers who are not aware of the switch, default set to on. */ + max98357a->spk_switch = 1; + dev_set_drvdata(&pdev->dev, max98357a); return devm_snd_soc_register_component(&pdev->dev, -- 2.25.0.341.g760bfbb309-goog

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[alsa-devel] [PATCH] ALSA: usb-audio: Apply 48kHz fixed rate playback for Jabra Evolve 65 headset
by Takashi Iwai 11 Feb '20

11 Feb '20

Jabra Evolve 65 headset appears as if supporting lower rates than 48kHz, but it actually doesn't work but with 48kHz for playback. This patch applies a workaround to enforce the 48kHz like LINE6 devices already did. The workaround is put in a unified helper function, set_fixed_rate(), to be called from both places now. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206149 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/format.c | 33 ++++++++++++++++++++++----------- 1 file changed, 22 insertions(+), 11 deletions(-) diff --git a/sound/usb/format.c b/sound/usb/format.c index 9260136e4c9b..50cb183958bf 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -151,6 +151,19 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip, return pcm_formats; } +static int set_fixed_rate(struct audioformat *fp, int rate, int rate_bits) +{ + kfree(fp->rate_table); + fp->rate_table = kmalloc(sizeof(int), GFP_KERNEL); + if (!fp->rate_table) + return -ENOMEM; + fp->nr_rates = 1; + fp->rate_min = rate; + fp->rate_max = rate; + fp->rates = rate_bits; + fp->rate_table[0] = rate; + return 0; +} /* * parse the format descriptor and stores the possible sample rates @@ -223,6 +236,14 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof fp->rate_min = combine_triple(&fmt[offset + 1]); fp->rate_max = combine_triple(&fmt[offset + 4]); } + + /* Jabra Evolve 65 headset */ + if (chip->usb_id == USB_ID(0x0b0e, 0x030b)) { + /* only 48kHz for playback while keeping 16kHz for capture */ + if (fp->nr_rates != 1) + return set_fixed_rate(fp, 48000, SNDRV_PCM_RATE_48000); + } + return 0; } @@ -299,17 +320,7 @@ static int line6_parse_audio_format_rates_quirk(struct snd_usb_audio *chip, case USB_ID(0x0e41, 0x4248): /* Line6 Helix >= fw 2.82 */ case USB_ID(0x0e41, 0x4249): /* Line6 Helix Rack >= fw 2.82 */ case USB_ID(0x0e41, 0x424a): /* Line6 Helix LT >= fw 2.82 */ - /* supported rates: 48Khz */ - kfree(fp->rate_table); - fp->rate_table = kmalloc(sizeof(int), GFP_KERNEL); - if (!fp->rate_table) - return -ENOMEM; - fp->nr_rates = 1; - fp->rate_min = 48000; - fp->rate_max = 48000; - fp->rates = SNDRV_PCM_RATE_48000; - fp->rate_table[0] = 48000; - return 0; + return set_fixed_rate(fp, 48000, SNDRV_PCM_RATE_48000); } return -ENODEV; -- 2.16.4

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[alsa-devel] [PATCH] do not set close-on-exec flag on descriptor if it was already set
by Rolf Eike Beer 11 Feb '20

11 Feb '20

There is no need to set this again if O_CLOEXEC is supported. Signed-off-by: Rolf Eike Beer <eb(a)emlix.com> --- include/local.h | 2 ++ 1 file changed, 2 insertions(+) diff --git a/include/local.h b/include/local.h index ea0ec32d..ed6ba936 100644 --- a/include/local.h +++ b/include/local.h @@ -320,8 +320,10 @@ static inline int snd_open_device(const char *filename, int fmode) fd = rsm_open_device(filename, fmode); } #endif +#ifndef O_CLOEXEC if (fd >= 0) fcntl(fd, F_SETFD, FD_CLOEXEC); +#endif return fd; } -- 2.25.0 -- Rolf Eike Beer, emlix GmbH, http://www.emlix.com Fon +49 551 30664-0, Fax +49 551 30664-11 Gothaer Platz 3, 37083 Göttingen, Germany Sitz der Gesellschaft: Göttingen, Amtsgericht Göttingen HR B 3160 Geschäftsführung: Heike Jordan, Dr. Uwe Kracke – Ust-IdNr.: DE 205 198 055 emlix - smart embedded open source Besuchen Sie uns auf der Embedded World 2020 in Nürnberg! -> Halle 4, Stand 368

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[alsa-devel] alsa-gobject: GSource optimization
by GitHub pull_request - opened 11 Feb '20

11 Feb '20

alsa-project/alsa-gobject pull request #8 was opened from takaswie: As long as using the created Gsource with GMainContext, the call of poll(2) system call with infinite timeout surely returns when quit() method is called for the context. All of GMainContext implements GWakeup with eventfd and the call of quit() emits event via the file descriptor of eventfd. This brings wakeup from blocking when poll(2) is called with inifinite timeout. In current implementation of alsactl/alsatimer, open(2) is called with O_NONBLOCK and GSource includes explicit timeout to avoid the infinite blocking. However, they are not necessarily required for GMainContext. This patchset obsoletes these points and adds options as the alternatives. Request URL : https://github.com/alsa-project/alsa-gobject/pull/8 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/8.patch Repository URL: https://github.com/alsa-project/alsa-gobject

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[alsa-devel] [PATCH] ALSA: hda - fixup for the bass speaker on Lenovo Carbon X1 7th gen
by Jaroslav Kysela 11 Feb '20

11 Feb '20

The auto-parser assigns the bass speaker to DAC3 (NID 0x06) which is without the volume control. I do not see a reason to use DAC2, because the shared output to all speakers produces the sufficient and well balanced sound. The stereo support is enough for this purpose (laptop). Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/pci/hda/patch_realtek.c | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d2bf70a1d2fd..a596790d4245 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5544,6 +5544,16 @@ static void alc295_fixup_disable_dac3(struct hda_codec *codec, } } +/* force NID 0x17 (Bass Speaker) to DAC1 to share it with the main speaker */ +static void alc285_fixup_speaker2_to_dac1(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + hda_nid_t conn[1] = { 0x02 }; + snd_hda_override_conn_list(codec, 0x17, 1, conn); + } +} + /* Hook to update amp GPIO4 for automute */ static void alc280_hp_gpio4_automute_hook(struct hda_codec *codec, struct hda_jack_callback *jack) @@ -5846,6 +5856,7 @@ enum { ALC225_FIXUP_DISABLE_MIC_VREF, ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, ALC295_FIXUP_DISABLE_DAC3, + ALC285_FIXUP_SPEAKER2_TO_DAC1, ALC280_FIXUP_HP_HEADSET_MIC, ALC221_FIXUP_HP_FRONT_MIC, ALC292_FIXUP_TPT460, @@ -6646,6 +6657,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc295_fixup_disable_dac3, }, + [ALC285_FIXUP_SPEAKER2_TO_DAC1] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_speaker2_to_dac1, + }, [ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -7221,6 +7236,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x224c, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x224d, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x225d, "Thinkpad T480", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x2293, "Thinkpad X1 Carbon 7th", ALC285_FIXUP_SPEAKER2_TO_DAC1), SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x310c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), @@ -7405,6 +7421,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC255_FIXUP_DELL_SPK_NOISE, .name = "dell-spk-noise"}, {.id = ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "alc225-dell1"}, {.id = ALC295_FIXUP_DISABLE_DAC3, .name = "alc295-disable-dac3"}, + {.id = ALC285_FIXUP_SPEAKER2_TO_DAC1, .name = "alc285-speaker2-to-dac1"}, {.id = ALC280_FIXUP_HP_HEADSET_MIC, .name = "alc280-hp-headset"}, {.id = ALC221_FIXUP_HP_FRONT_MIC, .name = "alc221-hp-mic"}, {.id = ALC298_FIXUP_SPK_VOLUME, .name = "alc298-spk-volume"}, -- 2.20.1

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[alsa-devel] broadwell-rt286: more conformance and other fixes
by GitHub pull_request - edited 11 Feb '20

11 Feb '20

alsa-project/alsa-ucm-conf pull request #12 was edited from dpward: Remove JackHWMute, because the hardware does not forcibly shut off the internal speakers or microphone when the jack is in use. The CaptureMixerElem and CaptureMasterElem devices were swapped. Do not manually set the "ADC0 Capture Switch" when disabling a capture device; it will be set automatically instead. Correct "Handset" to "Headset". Apply maximum gain to the headset microphone ("AMIC Volume") when in use, based on testing. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/12 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/12.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[alsa-devel] broadwell-rt286: more conformance and other fixes
by GitHub pull_request - edited 11 Feb '20

11 Feb '20

alsa-project/alsa-ucm-conf pull request #12 was edited from dpward: Remove JackHWMute, because the hardware does not forcibly shut off the internal speakers or microphone when the jack is in use. Leave the "ADC0 Capture Switch" enabled after a capture devices is disabled. The CaptureMixerElem and CaptureMasterElem were swapped. Correct "Handset" to "Headset". Apply maximum gain to the headset microphone ("AMIC Volume") when in use, based on testing. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/12 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/12.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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