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

[alsa-devel] ALC892 - Line-out not working properly
by Xristos 24 Apr '18

24 Apr '18

Hi everyone, I'm having a problem with my line-out port not being working properly. Specifically it can't play audio when the 'alsamixer' auto-mute mode is enabled, so sound is coming only from the headphones.However, when auto-mute is disabled the sound is coming from both ports simultaneously. I'm using the Asus Z170-A mobo with ALC892 codec, OS: Arch I ran alsa-info.sh on both cases and i saw that the Pin-Complex 0x14 had 'Pin-ctls: OUT' when auto-mute was disabled whereas when auto-mute was enabled it was 'Pin-ctls: 0x00' (as it currenly is). So i ran HDA-analyzer tool and changed that widget setting to 'OUT' and obviously i produced the same behaviour like when i disabled 'auto-mute' (again sound coming from both ports) and also in all cases PulseAudio lists the 'Line-out' port as 'unplugged'. Also 2 more things: 1st: In BIOS when i changed the driver for the Front panel from 'HD-Audio' to 'AC97' and the ports worked correctly, however no 'auto-mute' mode was available so i had to mute the ports manually when i plugged/unplugged the headphones for e.x I tried uninstalling PulseAudio to see if it was the case but the problem persisted so i think it really is some codec problem since i also have a similar problem in Windows OS. 2nd: When i unplugged the Front-panel connection from the mobo the port played correctly. So is it maybe a mixer problem or something related? Finally i tried every 'model' parameter from HD-Audio-Models.txt for ALC892 but got nothing. In any case i'm attaching the current alsa-info.sh. If you would like i could also send alsa-info.sh for all those cases above. I don't think i'm alone in this issue but i would greatly appreciate the help! Thanks in advance

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[alsa-devel] [PATCH] ALSA: dice: fix OUI for TC group
by Takashi Sakamoto 24 Apr '18

24 Apr '18

OUI for TC Electronic is 0x000166, for TC GROUP A/S. 0x001486 is for Echo Digital Audio Corporation. Fixes: 7cafc65b3aa1 ('ALSA: dice: force to add two pcm devices for listed models') Cc: <stable(a)vger.kernel.org> # v4.6+ Reference: http://standards-oui.ieee.org/oui/oui.txt Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/dice/dice.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index 4ddb4cdd054b..96bb01b6b751 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -14,7 +14,7 @@ MODULE_LICENSE("GPL v2"); #define OUI_WEISS 0x001c6a #define OUI_LOUD 0x000ff2 #define OUI_FOCUSRITE 0x00130e -#define OUI_TCELECTRONIC 0x001486 +#define OUI_TCELECTRONIC 0x000166 #define DICE_CATEGORY_ID 0x04 #define WEISS_CATEGORY_ID 0x00 -- 2.14.1

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[alsa-devel] [PATCH] ALSA: usb-audio: Skip broken EU on Dell dock USB-audio
by Takashi Iwai 24 Apr '18

24 Apr '18

The Dell Dock USB-audio device with 0bda:4014 is behaving notoriously bad, and we have already applied some workaround to avoid the firmware hiccup. Yet we still need to skip one thing, the Extension Unit at ID 4, which doesn't react correctly to the mixer ctl access. Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=1090658 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/mixer_maps.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index 9038b2e7df73..eaa03acd4686 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -353,8 +353,11 @@ static struct usbmix_name_map bose_companion5_map[] = { /* * Dell usb dock with ALC4020 codec had a firmware problem where it got * screwed up when zero volume is passed; just skip it as a workaround + * + * Also the extension unit gives an access error, so skip it as well. */ static const struct usbmix_name_map dell_alc4020_map[] = { + { 4, NULL }, /* extension unit */ { 16, NULL }, { 19, NULL }, { 0 } -- 2.16.3

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[alsa-devel] [PATCH] ALSA: usb-audio: Fix missing endian conversion
by Takashi Iwai 24 Apr '18

24 Apr '18

The UAC2 jack detection support introduced the bmControls checks in a couple of places, but they forgot the endian conversion; the bmControls of UAC2 terminal descriptor is __le16, not a byte like in UAC1. Fixes: 5a222e849452 ("ALSA: usb-audio: UAC2 jack detection") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/mixer.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 301ad61ed426..f78b500b3c0e 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1859,7 +1859,7 @@ static int parse_audio_input_terminal(struct mixer_build *state, int unitid, check_input_term(state, d->bTerminalID, &iterm); if (state->mixer->protocol == UAC_VERSION_2) { /* Check for jack detection. */ - if (uac_v2v3_control_is_readable(d->bmControls, + if (uac_v2v3_control_is_readable(le16_to_cpu(d->bmControls), UAC2_TE_CONNECTOR)) { build_connector_control(state, &iterm, true); } @@ -2561,7 +2561,7 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) if (err < 0 && err != -EINVAL) return err; - if (uac_v2v3_control_is_readable(desc->bmControls, + if (uac_v2v3_control_is_readable(le16_to_cpu(desc->bmControls), UAC2_TE_CONNECTOR)) { build_connector_control(&state, &state.oterm, false); -- 2.16.3

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[alsa-devel] [PATCH] ASoC: Intel: board: add BE DAI link for WoV
by mac.chiang＠intel.com 24 Apr '18

24 Apr '18

From: Mac Chiang <mac.chiang(a)intel.com> create dai link in kbl_da7219_max98357a driver for wake on voice functionality. changes picked from broonie's tree commit 0c7941a63a0f ("ASoC: Intel: Skylake: Use refcap device for mono recording") commit 2154be362c90 ("ASoc: Intel: boards: Add WOV as sink for nau88l25_ssm4567 machine") Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> --- sound/soc/intel/boards/kbl_da7219_max98357a.c | 35 +++++++++++++++++++++++++++ 1 file changed, 35 insertions(+) diff --git a/sound/soc/intel/boards/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c index e84baaf..c332a32 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98357a.c +++ b/sound/soc/intel/boards/kbl_da7219_max98357a.c @@ -45,6 +45,7 @@ struct kbl_codec_private { enum { KBL_DPCM_AUDIO_PB = 0, KBL_DPCM_AUDIO_CP, + KBL_DPCM_AUDIO_REF_CP, KBL_DPCM_AUDIO_DMIC_CP, KBL_DPCM_AUDIO_HDMI1_PB, KBL_DPCM_AUDIO_HDMI2_PB, @@ -349,6 +350,27 @@ static const unsigned int ch_mono[] = { 1, }; +static const struct snd_pcm_hw_constraint_list constraints_refcap = { + .count = ARRAY_SIZE(ch_mono), + .list = ch_mono, +}; + +static int kabylake_refcap_startup(struct snd_pcm_substream *substream) +{ + substream->runtime->hw.channels_max = 1; + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_refcap); + + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_16000); +} + +static struct snd_soc_ops skylaye_refcap_ops = { + .startup = kabylake_refcap_startup, +}; + /* kabylake digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link kabylake_dais[] = { /* Front End DAI links */ @@ -380,6 +402,19 @@ static struct snd_soc_dai_link kabylake_dais[] = { SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, .dpcm_capture = 1, }, + [KBL_DPCM_AUDIO_REF_CP] = { + .name = "Kbl Audio Reference cap", + .stream_name = "Wake on Voice", + .cpu_dai_name = "Reference Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .init = NULL, + .dpcm_capture = 1, + .nonatomic = 1, + .dynamic = 1, + .ops = &skylaye_refcap_ops, + }, [KBL_DPCM_AUDIO_DMIC_CP] = { .name = "Kbl Audio DMIC cap", .stream_name = "dmiccap", -- 2.7.4

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[alsa-devel] [PATCH 0/3] ASoc: TAS6424: Add support for mute, standby, and faster power on
by Jean-Jacques Hiblot 24 Apr '18

24 Apr '18

mute and standby pins are available on the codec. If they are connected, they should be managed by the driver, instead of relying on gpio hogs or on the initial state of the GPIOs. This series also includes a patch to improve the start-up time of the channels by disabling built-in DC diagnostics. Those diagnosdtics basically serve to detect : - wires shorted together - wire shorted to ground or vbat - wire disconnected This is not useful for all platforms and the addition to the startup time is quite noticeable (230ms). Jean-Jacques Hiblot (3): ASoC: tas6424: Allow disabling auto diagnostics for faster power-on ASoC: tas6424: Add support for the standby pin ASoC: tas6424: Add support for the mute pin .../devicetree/bindings/sound/ti,tas6424.txt | 4 ++ sound/soc/codecs/tas6424.c | 74 +++++++++++++++++++++- sound/soc/codecs/tas6424.h | 3 + 3 files changed, 79 insertions(+), 2 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH] sound, isapnp: allow building more drivers with COMPILE_TEST
by Mauro Carvalho Chehab 24 Apr '18

24 Apr '18

Drivers that depend on ISAPNP currently can't be built with COMPILE_TEST. However, looking at isapnp.h, there are already stubs there to allow drivers to include it even when isa PNP is not supported. So, remove such dependencies when COMPILE_TEST. Signed-off-by: Mauro Carvalho Chehab <mchehab(a)s-opensource.com> --- drivers/pnp/isapnp/Kconfig | 2 +- sound/isa/Kconfig | 6 +++--- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/drivers/pnp/isapnp/Kconfig b/drivers/pnp/isapnp/Kconfig index f1ef36673ad4..a1af146d2d90 100644 --- a/drivers/pnp/isapnp/Kconfig +++ b/drivers/pnp/isapnp/Kconfig @@ -3,7 +3,7 @@ # config ISAPNP bool "ISA Plug and Play support" - depends on ISA + depends on ISA || COMPILE_TEST help Say Y here if you would like support for ISA Plug and Play devices. Some information is in <file:Documentation/isapnp.txt>. diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index d2a6cdd0395c..43b35a873d78 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -39,7 +39,7 @@ config SND_ADLIB config SND_AD1816A tristate "Analog Devices SoundPort AD1816A" - depends on PNP && ISA + depends on PNP select ISAPNP select SND_OPL3_LIB select SND_MPU401_UART @@ -67,7 +67,7 @@ config SND_AD1848 config SND_ALS100 tristate "Diamond Tech. DT-019x and Avance Logic ALSxxx" - depends on PNP && ISA + depends on PNP select ISAPNP select SND_OPL3_LIB select SND_MPU401_UART @@ -108,7 +108,7 @@ config SND_AZT2316 config SND_AZT2320 tristate "Aztech Systems AZT2320" - depends on PNP && ISA + depends on PNP select ISAPNP select SND_OPL3_LIB select SND_MPU401_UART -- 2.14.3

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[alsa-devel] [PATCH 0/8] ALSA: Hardening for potential Spectre v1
by Takashi Iwai 24 Apr '18

24 Apr '18

Hi, this is a patchset to cover the potential issues wrt Spectre v1 that were spotted out recently by Smatch: https://marc.info/?l=linux-kernel&m=152411496503418&w=2 Although all reports for sound/* seem like no serious issues, it's better to cover potential usages than a too late sorry. The first patch is basically irrelevant for Spectre but a preliminary fix for the following fix. Most of the rest patches are straightforward, but the second one, the fix for OSS emulation, became a fair amount of code refactoring in the end. An open question is whether we should put Cc-to-stable to these fixes. Maybe yes, although we've never proven that the problem would actually exist, the issue is still "hot", so people may want to see them more-or-less covered... thanks, Takashi === Takashi Iwai (8): ALSA: seq: oss: Fix unbalanced use lock for synth MIDI device ALSA: seq: oss: Hardening for potential Spectre v1 ALSA: control: Hardening for potential Spectre v1 ALSA: hda: Hardening for potential Spectre v1 ALSA: opl3: Hardening for potential Spectre v1 ALSA: asihpi: Hardening for potential Spectre v1 ALSA: hdspm: Hardening for potential Spectre v1 ALSA: rme9652: Hardening for potential Spectre v1 include/sound/control.h | 7 +++- sound/core/seq/oss/seq_oss_event.c | 15 ++++--- sound/core/seq/oss/seq_oss_midi.c | 2 + sound/core/seq/oss/seq_oss_synth.c | 85 ++++++++++++++++++++++---------------- sound/core/seq/oss/seq_oss_synth.h | 3 +- sound/drivers/opl3/opl3_synth.c | 7 +++- sound/pci/asihpi/hpimsginit.c | 13 ++++-- sound/pci/asihpi/hpioctl.c | 4 +- sound/pci/hda/hda_hwdep.c | 12 +++++- sound/pci/rme9652/hdspm.c | 24 ++++++----- sound/pci/rme9652/rme9652.c | 6 ++- 11 files changed, 113 insertions(+), 65 deletions(-) -- 2.16.3

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[alsa-devel] [PATCH 0/4] USB Audio Device Class 3.0 BADD profiles support
by Ruslan Bilovol 24 Apr '18

24 Apr '18

This patchset adds BADD profiles support from the USB Audio Device Class 3.0 spec [1]. BADD profile support is defined as mandatory feature of UAC3-compliant device, it should be implemented as a separate USB configuration. Notable issue with BADD configuration is that it misses class-specific descriptors (and it's mandatory as per spec), so host should guess them from BADD profile number and parameters of endpoints (type, number of endpoints and max packet size) This patchset adds support of all known/existing BADD profiles from the UAC3 specification. It's an alternative implementation comparing to [2], and doesn't build usb descriptors on the host but instead initializes alsa-usb structures with known parameters, so we don't need to keep whole class-specific descriptors in the driver since we anyway need to have BADD-specific logic. I've picked one Jorge's UAC1 patch-improvement and updated it to v4.17 wich contais recently introduced header's sanity checks. Remaining part is to add interrupt endpoint support so we will be able to detect jack insertion in the Headset Adapter profile. This has been tested on ARM and x86-64 machines with custom UAC3 gadget which I'll post later to linux-usb Comments and testing are welcome. [1] http://www.usb.org/developers/docs/devclass_docs/USB_Audio_v3.0.zip [2] https://www.spinics.net/lists/alsa-devel/msg71614.html Jorge Sanjuan (1): ALSA: usb: Only get AudioControl header for UAC1 class. Ruslan Bilovol (3): ALSA: usb: stream: refactor audio interface parsing include: usb: audio-v3: add BADD-specific values ALSA: usb: add UAC3 BADD profiles support include/linux/usb/audio-v3.h | 26 ++ sound/usb/card.c | 53 ++-- sound/usb/clock.c | 9 +- sound/usb/mixer.c | 313 +++++++++++++++++++- sound/usb/mixer_maps.c | 65 +++++ sound/usb/stream.c | 678 +++++++++++++++++++++++++------------------ sound/usb/usbaudio.h | 2 + 7 files changed, 836 insertions(+), 310 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH v4 0/3] ASoC: Add multiple CPU DAI support
by Vinod Koul 24 Apr '18

24 Apr '18

As discussed in [1], DAI link supports multiple codec DAIs however it does not do so for CPU DAIs. So, add support for multiple CPU DAIs on a DAI Link by adding multiple CPU DAI in Card instantiation, suspend and resume functions, PCM ops, stream handling functions and DAPM. [1]: https://www.spinics.net/lists/alsa-devel/msg71369.html changes in v4: - rebase on asoc-next - update changelogs to add more description of changes done Shreyas NC (3): ASoC: Add initial support for multiple CPU DAIs ASoC: Add multiple CPU DAI support for PCM ops ASoC: Add multiple CPU DAI support in DAPM include/sound/soc.h | 6 + sound/soc/soc-core.c | 283 ++++++++++++++++++++---------- sound/soc/soc-dapm.c | 71 +++++--- sound/soc/soc-pcm.c | 483 +++++++++++++++++++++++++++++++++------------------ 4 files changed, 565 insertions(+), 278 deletions(-) -- 2.7.4

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