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

[alsa-devel] [PATCH] ALSA: hda: Request driver probe from an async task
by Paul Menzel 24 Apr '18

24 Apr '18

From: Paul Menzel <pmenzel(a)molgen.mpg.de> Date: Sat, 24 Mar 2018 09:28:43 +0100 On an ASRock E350M1, with Linux 4.17-rc1 according to `initcall_debug` calling `azx_driver_init` takes sometimes more than a few milliseconds, and up to 200 ms. ``` [ 2.892598] calling azx_driver_init+0x0/0xfe4 [snd_hda_intel] @ 218 [ 2.943002] initcall azx_driver_init+0x0/0xfe4 [snd_hda_intel] returned 0 after 49195 usecs ``` Trying to execute the Linux kernel in less than 500 ms, this is quite a hold-up, and therefore request the probe from an async task. With this change, the test shows, that the function returns earlier. ``` [ 3.254800] calling azx_driver_init+0x0/0xfe4 [snd_hda_intel] @ 227 [ 3.254887] initcall azx_driver_init+0x0/0xfe4 [snd_hda_intel] returned 0 after 66 usecs ``` The same behavior is visible on a Dell OptiPlex 7010. The longer times seem to happen, when the module *e1000e* is probed during the same time. Signed-off-by: Paul Menzel <pmenzel(a)molgen.mpg.de> --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 6347256bc017..827834b9c07d 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2671,6 +2671,7 @@ static struct pci_driver azx_driver = { .driver = { .pm = AZX_PM_OPS, }, + .driver.probe_type = PROBE_PREFER_ASYNCHRONOUS, }; module_pci_driver(azx_driver); -- 2.16.2

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[alsa-devel] [PATCH] ALSA: dice: improve support for ancient firmware for DICE
by Takashi Sakamoto 24 Apr '18

24 Apr '18

In early stage of firmware SDK, DICE seems to lose its backward compatibility due to some registers on global address section. I found this with Alesis Multimix 12 FireWire with ancient firmware (approx. shipped version). According to retrieved log from the unit, global section has 96 byte space. On the other hand, current version of ALSA dice driver assumes that all of supported unit has at least 100 byte space. $ ./firewire-request /dev/fw1 read 0xffffe0000000 28 result: 000: 00 00 00 0a 00 00 00 18 00 00 00 22 00 00 00 8a result: 010: 00 00 00 ac 00 00 01 12 00 00 00 00 00 00 00 00 result: 020: 00 00 00 00 00 00 00 00 This commit adds support for the ancient firmware. Check of global section is loosened to accept the smaller space. The lack of information is already compensated by hard-coded parameters. I experienced that the latest version of Windows driver for this model can't handle this unit, too. This means that TCAT releases firmware SDK without backward compatibility for the ancient firmware. Below list is a early history of driver/firmware package released by Alesis. I investigated on wayback machine on Internet Archive: * Unknown: PAL v1.0.41.2, firmware v1.0.3 * Mar 2006: PAL v1.54.0, firmware v1.0.4 * Dec 2006: PAL v2.0.0.2, firmware v2.0 * Jun 2007: PAL v3.0.41.5, firmware v2.0 * Jul 2007: PAL v3.0.56.2. firmware v2.0 * Jan 2008: PAL v3.0.81.1080, firmware v2.0 If I can assume that firmware version is the same as DICE version, DICE version for the issued firmware may be v1.0.3. According to code base of userspace driver project (FFADO), I can read DICE v1.0.4 supports global space larger than 100 byte. I guess the smaller space of global section is a feature of DICE v1.0.3 but no way to clear it. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/dice/dice-interface.h | 9 +++++-- sound/firewire/dice/dice-proc.c | 10 +++---- sound/firewire/dice/dice-transaction.c | 49 +++++++++++++++++++--------------- 3 files changed, 40 insertions(+), 28 deletions(-) diff --git a/sound/firewire/dice/dice-interface.h b/sound/firewire/dice/dice-interface.h index 15a484b05298..9cad3d608229 100644 --- a/sound/firewire/dice/dice-interface.h +++ b/sound/firewire/dice/dice-interface.h @@ -174,14 +174,19 @@ */ #define GLOBAL_SAMPLE_RATE 0x05c +/* + * Some old firmware versions do not have the following global registers. + * Windows drivers produced by TCAT lost backward compatibility in its + * early release because they can handle firmware only which supports the + * following registers. + */ + /* * The version of the DICE driver specification that this device conforms to; * read-only. */ #define GLOBAL_VERSION 0x060 -/* Some old firmware versions do not have the following global registers: */ - /* * Supported sample rates and clock sources; read-only. */ diff --git a/sound/firewire/dice/dice-proc.c b/sound/firewire/dice/dice-proc.c index f5c1d1bced59..cc079323ed30 100644 --- a/sound/firewire/dice/dice-proc.c +++ b/sound/firewire/dice/dice-proc.c @@ -148,12 +148,12 @@ static void dice_proc_read(struct snd_info_entry *entry, >> CLOCK_RATE_SHIFT)); snd_iprintf(buffer, " ext status: %08x\n", buf.global.extended_status); snd_iprintf(buffer, " sample rate: %u\n", buf.global.sample_rate); - snd_iprintf(buffer, " version: %u.%u.%u.%u\n", - (buf.global.version >> 24) & 0xff, - (buf.global.version >> 16) & 0xff, - (buf.global.version >> 8) & 0xff, - (buf.global.version >> 0) & 0xff); if (quadlets >= 90) { + snd_iprintf(buffer, " version: %u.%u.%u.%u\n", + (buf.global.version >> 24) & 0xff, + (buf.global.version >> 16) & 0xff, + (buf.global.version >> 8) & 0xff, + (buf.global.version >> 0) & 0xff); snd_iprintf(buffer, " clock caps:"); for (i = 0; i <= 6; ++i) if (buf.global.clock_caps & (1 << i)) diff --git a/sound/firewire/dice/dice-transaction.c b/sound/firewire/dice/dice-transaction.c index 0f0350320ae8..2410c082bf8e 100644 --- a/sound/firewire/dice/dice-transaction.c +++ b/sound/firewire/dice/dice-transaction.c @@ -265,7 +265,7 @@ int snd_dice_transaction_reinit(struct snd_dice *dice) static int get_subaddrs(struct snd_dice *dice) { static const int min_values[10] = { - 10, 0x64 / 4, + 10, 0x60 / 4, 10, 0x18 / 4, 10, 0x18 / 4, 0, 0, @@ -301,33 +301,40 @@ static int get_subaddrs(struct snd_dice *dice) } } - /* - * Check that the implemented DICE driver specification major version - * number matches. - */ - err = snd_fw_transaction(dice->unit, TCODE_READ_QUADLET_REQUEST, - DICE_PRIVATE_SPACE + - be32_to_cpu(pointers[0]) * 4 + GLOBAL_VERSION, - &version, sizeof(version), 0); - if (err < 0) - goto end; + if (be32_to_cpu(pointers[1]) > 0x18) { + /* + * Check that the implemented DICE driver specification major + * version number matches. + */ + err = snd_fw_transaction(dice->unit, TCODE_READ_QUADLET_REQUEST, + DICE_PRIVATE_SPACE + + be32_to_cpu(pointers[0]) * 4 + GLOBAL_VERSION, + &version, sizeof(version), 0); + if (err < 0) + goto end; - if ((version & cpu_to_be32(0xff000000)) != cpu_to_be32(0x01000000)) { - dev_err(&dice->unit->device, - "unknown DICE version: 0x%08x\n", be32_to_cpu(version)); - err = -ENODEV; - goto end; + if ((version & cpu_to_be32(0xff000000)) != + cpu_to_be32(0x01000000)) { + dev_err(&dice->unit->device, + "unknown DICE version: 0x%08x\n", + be32_to_cpu(version)); + err = -ENODEV; + goto end; + } + + /* Set up later. */ + dice->clock_caps = 1; } dice->global_offset = be32_to_cpu(pointers[0]) * 4; dice->tx_offset = be32_to_cpu(pointers[2]) * 4; dice->rx_offset = be32_to_cpu(pointers[4]) * 4; - dice->sync_offset = be32_to_cpu(pointers[6]) * 4; - dice->rsrv_offset = be32_to_cpu(pointers[8]) * 4; - /* Set up later. */ - if (be32_to_cpu(pointers[1]) * 4 >= GLOBAL_CLOCK_CAPABILITIES + 4) - dice->clock_caps = 1; + /* Old firmware doesn't support these fields. */ + if (pointers[7] > 0) + dice->sync_offset = be32_to_cpu(pointers[6]) * 4; + if (pointers[9] > 0) + dice->rsrv_offset = be32_to_cpu(pointers[8]) * 4; end: kfree(pointers); return err; -- 2.14.1

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[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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