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

[alsa-devel] [PATCH] ALSA: hda/realtek - Add shutup hint
by Takashi Iwai 12 Jun '18

12 Jun '18

The pin shutup callback seems working well on some devices while it does harm on some other devices; e.g. Lenovo laptops show often the noises at (runtime) PM with the pin shutup enabled. Currently, the only way to disable the pin shutup is to hard-code spec->shutup = alc_no_shutup; in the fixup, and this makes the debugging harder for normal users. For allowing users to test the similar effect without recompiling the kernel, this patch adds a new hint string "shutup". It's a boolean value, and by passing false to this, user can turn off the pin shutup call. For example, to turn off the shutup on Lenovo P50, create a "firmware patch" file (e.g. /lib/firmware/alsa/lenovo-p50) containing the following lines: [codec] 0x10ec0298 0x17aa222e 0 [hint] shutup = no and pass the file via patch option of snd-hda-intel module (e.g. patch=alsa/lenovo-p50). Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b1eb6bd4fbd8..e5a002900292 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -793,6 +793,9 @@ static inline void alc_shutup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; + if (!snd_hda_get_bool_hint(codec, "shutup")) + return; /* disabled explicitly by hints */ + if (spec && spec->shutup) spec->shutup(codec); else -- 2.17.1

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[alsa-devel] [PATCH 0/8] ASoC: sh: convert SPDX identifiers
by Kuninori Morimoto 12 Jun '18

12 Jun '18

Hi Mark These patches convert license to SPDX identifiers Kuninori Morimoto (8): ASoC: fsi: convert to SPDX identifiers ASoC: hac: convert to SPDX identifiers ASoC: ssi: convert to SPDX identifiers ASoC: siu: convert to SPDX identifiers ASoC: rsnd: convert to SPDX identifiers ASoC: migor: convert to SPDX identifiers ASoC: dma-sh7760: convert to SPDX identifiers ASoC: sh7760-ac97: convert to SPDX identifiers sound/soc/sh/dma-sh7760.c | 5 ++--- sound/soc/sh/fsi.c | 5 +---- sound/soc/sh/hac.c | 5 ++--- sound/soc/sh/migor.c | 5 +---- sound/soc/sh/rcar/Makefile | 1 + sound/soc/sh/rcar/adg.c | 5 +---- sound/soc/sh/rcar/cmd.c | 5 +---- sound/soc/sh/rcar/core.c | 7 ++----- sound/soc/sh/rcar/ctu.c | 5 +---- sound/soc/sh/rcar/dma.c | 5 +---- sound/soc/sh/rcar/dvc.c | 5 +---- sound/soc/sh/rcar/gen.c | 5 +---- sound/soc/sh/rcar/mix.c | 5 +---- sound/soc/sh/rcar/rsnd.h | 5 +---- sound/soc/sh/rcar/src.c | 5 +---- sound/soc/sh/rcar/ssi.c | 5 +---- sound/soc/sh/rcar/ssiu.c | 5 +---- sound/soc/sh/sh7760-ac97.c | 5 ++--- sound/soc/sh/siu.h | 15 +-------------- sound/soc/sh/siu_dai.c | 15 +-------------- sound/soc/sh/siu_pcm.c | 15 +-------------- sound/soc/sh/ssi.c | 6 ++---- 22 files changed, 27 insertions(+), 112 deletions(-) -- 2.17.1

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[alsa-devel] [PATCH][alsa-tools] ld10k1: define _GNU_SOURCE and use sighandler_t
by Ross Burton 12 Jun '18

12 Jun '18

__sighandler_t is a glibc internal type which doesn't exist in musl. By using AC_USE_SYSTEM_EXTENSIONS to get _GNU_SOURCE defined, both glibc and musl then expose sighandler_t. Signed-off-by: Ross Burton <ross.burton(a)intel.com> --- ld10k1/configure.ac | 1 + ld10k1/src/ld10k1_fnc1.c | 2 +- 2 files changed, 2 insertions(+), 1 deletion(-) diff --git a/ld10k1/configure.ac b/ld10k1/configure.ac index 276b24e..f48ca31 100644 --- a/ld10k1/configure.ac +++ b/ld10k1/configure.ac @@ -4,6 +4,7 @@ AC_CONFIG_MACRO_DIR([m4]) AM_MAINTAINER_MODE([enable]) AC_CONFIG_HEADERS(config.h) +AC_USE_SYSTEM_EXTENSIONS AC_PROG_CC AC_PROG_LD AC_PROG_INSTALL diff --git a/ld10k1/src/ld10k1_fnc1.c b/ld10k1/src/ld10k1_fnc1.c index 343584c..ddc43c6 100644 --- a/ld10k1/src/ld10k1_fnc1.c +++ b/ld10k1/src/ld10k1_fnc1.c @@ -179,7 +179,7 @@ int main_loop(comm_param *param, int audigy, const char *card_id, int tram_size, { fd_set active_fd_set/*, read_fd_set*/; int i, j, res = 0; - __sighandler_t old_sig_pipe; + sighandler_t old_sig_pipe; int main_sock = 0; int data_sock = 0; -- 2.11.0

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[alsa-devel] handling voice calls in ALSA soc (on Droid 4)
by Pavel Machek 12 Jun '18

12 Jun '18

Hi! During voice call, I need audio components to be active "as if" aplay or arecord was running, because modem needs to comunicate with speaker and microphone. I hacked something up, but... I believe I'll need help here. Look at "enable_call" for "interesting" stuff I had to do. ...and also. What is right interface for this? Mixer component that says if voice call is active or not? Any ideas? Thanks, Pavel (edited). +++ b/sound/soc/codecs/cpcap.c @@ -330,6 +330,10 @@ static const char * const cpcap_in_left_mux_texts[] = { "Off", "Mic 2", "Ext Left" }; +static const char * const cpcap_mode_texts[] = { + "Normal", "Handsfree", "Call", +}; + /* * input muxes use unusual register layout, so that we need to use custom * getter/setter methods @@ -354,6 +358,8 @@ static SOC_ENUM_SINGLE_DECL(cpcap_hs_l_mux_enum, 0, 6, cpcap_out_mux_texts); static SOC_ENUM_SINGLE_DECL(cpcap_emu_l_mux_enum, 0, 7, cpcap_out_mux_texts); static SOC_ENUM_SINGLE_DECL(cpcap_emu_r_mux_enum, 0, 8, cpcap_out_mux_texts); +static SOC_ENUM_SINGLE_DECL(cpcap_mode_enum, 0, 9, cpcap_mode_texts); + static int cpcap_output_mux_get_enum(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -442,6 +448,211 @@ static int cpcap_output_mux_put_enum(struct snd_kcontrol *kcontrol, return 0; } +static int mode; + +static int cpcap_mode_get_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.enumerated.item[0] = mode; + + return 0; +} + +static struct snd_soc_dai *voice_codec_dai_hack; + +static int enable_call(struct snd_soc_component *component, int on) +{ + struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component); + struct snd_soc_pcm_runtime *rt; + + rt = snd_soc_get_pcm_runtime(component->card, "40126000.mcbsp-cpcap-voice"); + printk("num_dai: %d, got runtime %lx\n", component->num_dai, rt); + + if (rt) { + snd_soc_dapm_stream_event(rt, SNDRV_PCM_STREAM_PLAYBACK, SND_SOC_DAPM_STREAM_START); + snd_soc_dapm_stream_event(rt, SNDRV_PCM_STREAM_CAPTURE, SND_SOC_DAPM_STREAM_START); + } + + cpcap_set_sysclk(cpcap, CPCAP_DAI_VOICE, 1, 19200000); + cpcap_set_samprate(cpcap, CPCAP_DAI_VOICE, 8000); + + cpcap_voice_set_dai_fmt(voice_codec_dai_hack, + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM ); + + return 0; +} + +static int cpcap_mode_put_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_dapm_kcontrol_component(kcontrol); + struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component); + unsigned int muxval = ucontrol->value.enumerated.item[0]; + + printk("Requested mode %d\n", muxval); + + mode = muxval; + + switch (muxval) { + case 1: + enable_call(component, 1); + break; + case 2: + enable_call(component, 1); + + regmap_assert(cpcap, CPCAP_REG_TXI, 0xffff, 0x0cc6); ... + break; + + default: + break; + } + + return 0; +} + static int cpcap_input_right_mux_get_enum(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -630,6 +841,10 @@ static const struct snd_kcontrol_new cpcap_earpiece_mux = SOC_DAPM_ENUM_EXT("Earpiece", cpcap_earpiece_mux_enum, cpcap_output_mux_get_enum, cpcap_output_mux_put_enum); +static const struct snd_kcontrol_new cpcap_mode = + SOC_DAPM_ENUM_EXT("Mode", cpcap_mode_enum, + cpcap_mode_get_enum, cpcap_mode_put_enum); + static const struct snd_kcontrol_new cpcap_hifi_mono_mixer_controls[] = { SOC_DAPM_SINGLE("HiFi Mono Playback Switch", CPCAP_REG_RXSDOA, CPCAP_BIT_MONO_DAC1, 1, 0), @@ -771,6 +986,9 @@ static const struct snd_soc_dapm_widget cpcap_dapm_widgets[] = { SND_SOC_DAPM_MUX("EMU Left Playback Route", SND_SOC_NOPM, 0, 0, &cpcap_emu_left_mux), + SND_SOC_DAPM_MUX("Mode", SND_SOC_NOPM, 0, 0, + &cpcap_mode), + /* Output Amplifier */ SND_SOC_DAPM_PGA("Earpiece PGA", CPCAP_REG_RXOA, CPCAP_BIT_A1_EAR_EN, 0, NULL, 0), @@ -791,7 +1009,7 @@ static const struct snd_soc_dapm_widget cpcap_dapm_widgets[] = { SND_SOC_DAPM_PGA("EMU Left PGA", CPCAP_REG_RXOA, CPCAP_BIT_EMU_SPKR_L_EN, 0, NULL, 0), - /* Headet Charge Pump */ + /* Headset Charge Pump */ SND_SOC_DAPM_SUPPLY("Headset Charge Pump", CPCAP_REG_RXOA, CPCAP_BIT_ST_HS_CP_EN, 0, NULL, 0), @@ -1304,6 +1525,7 @@ static int cpcap_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, u16 val = 0x0000; int err; + voice_codec_dai_hack = codec_dai; dev_dbg(component->dev, "Voice setup dai format (%08x)", fmt); /* @@ -1343,10 +1565,7 @@ static int cpcap_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, break; } - if (val & BIT(CPCAP_BIT_CLK_INV)) - val &= ~BIT(CPCAP_BIT_CLK_INV); - else - val |= BIT(CPCAP_BIT_CLK_INV); + val ^= BIT(CPCAP_BIT_CLK_INV); switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: -- (english) http://www.livejournal.com/~pavelmachek (cesky, pictures) http://atrey.karlin.mff.cuni.cz/~pavel/picture/horses/blog.html

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[alsa-devel] [PATCH] pcm: dmix: Fix hwptr updates at status call
by Takashi Iwai 12 Jun '18

12 Jun '18

In the commit 38a2d2eda880 ("pcm: dmix: Do not discard slave reported delay in status result"), the PCM dmix hwptr update code was rewritten to follow the slave PCM hwptr update. This is based on the similar change in PCM dshare, the commit faf53c197cab. There was a bug in the commit 38a2d2eda880 regarding the PCM state change, and it was addressed in commit 3752e6b8733d ("pcm: dmix: Fix the inconsistent PCM state"). However, we've hit yet another bug in this commit. Namely, the hwptr update was forgotten in the snd_pcm_dmix_sync_ptr0() function. So the hwptr value passed from snd_pcm_dmix_status() isn't properly stored, and it screws up at some long run occasionally. This patch covers the bug by replacing with the right value. Fixes: 38a2d2eda880 ("pcm: dmix: Do not discard slave reported delay in status result") Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=200013 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- src/pcm/pcm_dmix.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c index c65259beb1d6..3a951f533655 100644 --- a/src/pcm/pcm_dmix.c +++ b/src/pcm/pcm_dmix.c @@ -398,7 +398,7 @@ static int snd_pcm_dmix_sync_ptr0(snd_pcm_t *pcm, snd_pcm_uframes_t slave_hw_ptr snd_pcm_sframes_t diff; old_slave_hw_ptr = dmix->slave_hw_ptr; - dmix->slave_hw_ptr = *dmix->spcm->hw.ptr; + dmix->slave_hw_ptr = slave_hw_ptr; diff = slave_hw_ptr - old_slave_hw_ptr; if (diff == 0) /* fast path */ return 0; -- 2.17.1

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[alsa-devel] [PATCH v2 2/2] ALSA: hda: add dock and led support for HP ProBook 640 G4
by Dennis Wassenberg 12 Jun '18

12 Jun '18

This patch adds missing initialisation for HP 2013 UltraSlim Dock Line-In/Out PINs and activates keyboard mute/micmute leds for HP ProBook 640 G4 Signed-off-by: Dennis Wassenberg <dennis.wassenberg(a)secunet.com> --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index bca71f35b34f..ff0c52e98977 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -959,6 +959,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x103c, 0x807C, "HP EliteBook 820 G3", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x80FD, "HP ProBook 640 G2", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x83b3, "HP EliteBook 830 G5", CXT_FIXUP_HP_DOCK), + SND_PCI_QUIRK(0x103c, 0x83d3, "HP ProBook 640 G4", CXT_FIXUP_HP_DOCK), 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(0x103c, 0x814f, "HP ZBook 15u G3", CXT_FIXUP_MUTE_LED_GPIO), -- 2.17.1

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[alsa-devel] [PATCH 1/2] ALSA: hda: add dock and led support for HP EliteBook 830 G5
by Dennis Wassenberg 12 Jun '18

12 Jun '18

This patch adds missing initialisation for HP 2013 UltraSlim Dock Line-In/Out PINs and activates keyboard mute/micmute leds for HP EliteBook 830 G5 --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 5b4dbcec6de8..bca71f35b34f 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -959,6 +959,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x103c, 0x8079, "HP EliteBook 840 G3", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x807C, "HP EliteBook 820 G3", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x80FD, "HP ProBook 640 G2", CXT_FIXUP_HP_DOCK), + SND_PCI_QUIRK(0x103c, 0x83b3, "HP EliteBook 830 G5", CXT_FIXUP_HP_DOCK), 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(0x103c, 0x814f, "HP ZBook 15u G3", CXT_FIXUP_MUTE_LED_GPIO), -- 2.17.1

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[alsa-devel] [PATCH] ALSA: usb-audio: Disable the quirk for Nura headset
by Takashi Iwai 12 Jun '18

12 Jun '18

The commit 33193dca671c ("ALSA: usb-audio: Add a quirk for Nura's first gen headset") added a quirk for Nura headset with USB ID 0a12:1243, with a hope that it doesn't conflict with others. Unfortunately, other devices (e.g. Philips Wecall) with the very same ID got broken by this change, spewing an error like: usb 2-1.8.2: 2:1: cannot set freq 48000 to ep 0x3 Until we find a proper solution, fix the regression at first by disabling the added quirk entry. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=199905 Fixes: 33193dca671c ("ALSA: usb-audio: Add a quirk for Nura's first gen headset") Reviewed-by: Martin Peres <martin.peres(a)free.fr> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/quirks-table.h | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 0e37e358ca97..8aac48f9c322 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3277,6 +3277,10 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } }, +/* disabled due to regression for other devices; + * see https://bugzilla.kernel.org/show_bug.cgi?id=199905 + */ +#if 0 { /* * Nura's first gen headphones use Cambridge Silicon Radio's vendor @@ -3324,6 +3328,7 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } } }, +#endif /* disabled */ { /* -- 2.17.0

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[alsa-devel] Order of device overrides in config files
by Pavel Hofman 11 Jun '18

11 Jun '18

Hello, Please in what order are devices specified in alsa conf overriden? The pcm.!default device is defined in /usr/share/alsa/pulse-alsa.conf. Also, I redefine this device in my .asoundrc. Strace dump shows my .asoundrc is loaded after /usr/share/alsa/pulse-alsa.conf. Yet the default device remains the pulse plugin. Also, what is the exact meaning of the exclamation mark in the device name? I very much appreciate any help. Best regards, Pavel.

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[alsa-devel] [PATCH] ALSA: emu10k1: add error handling for snd_ctl_add
by Zhouyang Jia 11 Jun '18

11 Jun '18

When snd_ctl_add fails, the lack of error-handling code may cause unexpected results. This patch adds error-handling code after calling snd_ctl_add. Signed-off-by: Zhouyang Jia <jiazhouyang09(a)gmail.com> --- sound/pci/emu10k1/emupcm.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index d39458a..69f9b10 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -1858,7 +1858,9 @@ int snd_emu10k1_pcm_efx(struct snd_emu10k1 *emu, int device) if (!kctl) return -ENOMEM; kctl->id.device = device; - snd_ctl_add(emu->card, kctl); + err = snd_ctl_add(emu->card, kctl); + if (err < 0) + return err; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(emu->pci), 64*1024, 64*1024); -- 2.7.4

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