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

[PATCH 00/14] ALSA: firewire-motu: code refactoring to obsolete protocol structure
by Takashi Sakamoto 22 May '20

22 May '20

Hi, ALSA firewire-motu driver becomes to support more models since its first commit 2017. As a result, the original structure for generation-dependent protocol is not suitable for differences between models in some cases. This patchset is for refactoring the driver to obsolete the protocol structure. Some helper functions are added with condition statement for protocol version so that the function calls protocol-dependent function. Instead of the protocol structure, specification structure is used to represent model information and to choose model-dependent operation so that it's easy to handle model-dependent quirks. Additionally, this patchset changes the way to calculate the number of chunks in data block. Current implementation calculates by the fixed number of analog input/output ports and flags in the specification structure. Although the flags can represent any physical input/output of the model, it's difficult to debug for supported model and to add support new models, especially for model-dependent quirks. This patchset adds each table for the fixed number of chunks of each model. The calculation is just done for differed part of chunks. Takashi Sakamoto (14): ALSA: firewire-motu: move spec data to v2 protocol file ALSA: firewire-motu: move spec data to v3 protocol file ALSA: firewire-motu: localize protocol data ALSA: firewire-motu: add wrapper functions for protocol-dependent operations ALSA: firewire-motu: drop protocol structure ALSA: firewire-motu: add model-specific table of chunk count ALSA: firewire-motu: add alternative functions to detect packet format for protocol v2 ALSA: firewire-motu: add alternative functions to detect packet format for protocol v3 ALSA: firewire-motu: use table-based calculation of packet formats for proc ALSA: firewire-motu: use table-based calculation of packet formats for stream management ALSA: firewire-motu: remove obsoleted codes ALSA: firewire-motu: refactoring protocol v2 for clock source getter ALSA: firewire-motu: refactoring protocol v3 for clock source getter ALSA: firewire-motu: refactoring protocol v2 for fetching mode switch sound/firewire/motu/amdtp-motu.c | 19 +- sound/firewire/motu/motu-pcm.c | 14 +- sound/firewire/motu/motu-proc.c | 20 +- sound/firewire/motu/motu-protocol-v2.c | 314 +++++++++++++++---------- sound/firewire/motu/motu-protocol-v3.c | 303 ++++++++++++------------ sound/firewire/motu/motu-stream.c | 16 +- sound/firewire/motu/motu.c | 113 +-------- sound/firewire/motu/motu.h | 124 +++++++--- 8 files changed, 463 insertions(+), 460 deletions(-) -- 2.25.1

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[PATCH AUTOSEL 4.19 07/19] ALSA: usb-audio: add mapping for ASRock TRX40 Creator
by Sasha Levin 22 May '20

22 May '20

From: Andrew Oakley <andrew(a)adoakley.name> [ Upstream commit da7a8f1a8fc3e14c6dcc52b4098bddb8f20390be ] This is another TRX40 based motherboard with ALC1220-VB USB-audio that requires a static mapping table. This motherboard also has a PCI device which advertises no codecs. The PCI ID is 1022:1487 and PCI SSID is 1022:d102. As this is using the AMD vendor ID, don't blacklist for now in case other boards have a working audio device with the same ssid. alsa-info.sh report for this board: http://alsa-project.org/db/?f=0a742f89066527497b77ce16bca486daccf8a70c Signed-off-by: Andrew Oakley <andrew(a)adoakley.name> Link: https://lore.kernel.org/r/20200503141639.35519-1-andrew@adoakley.name Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/mixer_maps.c | 5 +++++ sound/usb/quirks-table.h | 1 + 2 files changed, 6 insertions(+) diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index 1689e4f242df..d7a8b23b335b 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -543,6 +543,11 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { .map = trx40_mobo_map, .connector_map = trx40_mobo_connector_map, }, + { /* Asrock TRX40 Creator */ + .id = USB_ID(0x26ce, 0x0a01), + .map = trx40_mobo_map, + .connector_map = trx40_mobo_connector_map, + }, { 0 } /* terminator */ }; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 774aeedde071..4f8a2b98e090 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3412,6 +3412,7 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), ALC1220_VB_DESKTOP(0x0414, 0xa002), /* Gigabyte TRX40 Aorus Pro WiFi */ ALC1220_VB_DESKTOP(0x0db0, 0x0d64), /* MSI TRX40 Creator */ ALC1220_VB_DESKTOP(0x0db0, 0x543d), /* MSI TRX40 */ +ALC1220_VB_DESKTOP(0x26ce, 0x0a01), /* Asrock TRX40 Creator */ #undef ALC1220_VB_DESKTOP #undef USB_DEVICE_VENDOR_SPEC -- 2.25.1

1 0

[PATCH AUTOSEL 5.4 08/32] ALSA: usb-audio: add mapping for ASRock TRX40 Creator
by Sasha Levin 22 May '20

22 May '20

From: Andrew Oakley <andrew(a)adoakley.name> [ Upstream commit da7a8f1a8fc3e14c6dcc52b4098bddb8f20390be ] This is another TRX40 based motherboard with ALC1220-VB USB-audio that requires a static mapping table. This motherboard also has a PCI device which advertises no codecs. The PCI ID is 1022:1487 and PCI SSID is 1022:d102. As this is using the AMD vendor ID, don't blacklist for now in case other boards have a working audio device with the same ssid. alsa-info.sh report for this board: http://alsa-project.org/db/?f=0a742f89066527497b77ce16bca486daccf8a70c Signed-off-by: Andrew Oakley <andrew(a)adoakley.name> Link: https://lore.kernel.org/r/20200503141639.35519-1-andrew@adoakley.name Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/mixer_maps.c | 5 +++++ sound/usb/quirks-table.h | 1 + 2 files changed, 6 insertions(+) diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index 39d6c6fa5e33..2255f9abd7a5 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -529,6 +529,11 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { .map = trx40_mobo_map, .connector_map = trx40_mobo_connector_map, }, + { /* Asrock TRX40 Creator */ + .id = USB_ID(0x26ce, 0x0a01), + .map = trx40_mobo_map, + .connector_map = trx40_mobo_connector_map, + }, { 0 } /* terminator */ }; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 8c2f5c23e1b4..aa4c16ce0e57 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3647,6 +3647,7 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), ALC1220_VB_DESKTOP(0x0414, 0xa002), /* Gigabyte TRX40 Aorus Pro WiFi */ ALC1220_VB_DESKTOP(0x0db0, 0x0d64), /* MSI TRX40 Creator */ ALC1220_VB_DESKTOP(0x0db0, 0x543d), /* MSI TRX40 */ +ALC1220_VB_DESKTOP(0x26ce, 0x0a01), /* Asrock TRX40 Creator */ #undef ALC1220_VB_DESKTOP #undef USB_DEVICE_VENDOR_SPEC -- 2.25.1

1 0

[PATCH AUTOSEL 5.6 09/41] ALSA: usb-audio: add mapping for ASRock TRX40 Creator
by Sasha Levin 22 May '20

22 May '20

From: Andrew Oakley <andrew(a)adoakley.name> [ Upstream commit da7a8f1a8fc3e14c6dcc52b4098bddb8f20390be ] This is another TRX40 based motherboard with ALC1220-VB USB-audio that requires a static mapping table. This motherboard also has a PCI device which advertises no codecs. The PCI ID is 1022:1487 and PCI SSID is 1022:d102. As this is using the AMD vendor ID, don't blacklist for now in case other boards have a working audio device with the same ssid. alsa-info.sh report for this board: http://alsa-project.org/db/?f=0a742f89066527497b77ce16bca486daccf8a70c Signed-off-by: Andrew Oakley <andrew(a)adoakley.name> Link: https://lore.kernel.org/r/20200503141639.35519-1-andrew@adoakley.name Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/mixer_maps.c | 5 +++++ sound/usb/quirks-table.h | 1 + 2 files changed, 6 insertions(+) diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index 0260c750e156..bfdc6ad52785 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -549,6 +549,11 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = { .map = trx40_mobo_map, .connector_map = trx40_mobo_connector_map, }, + { /* Asrock TRX40 Creator */ + .id = USB_ID(0x26ce, 0x0a01), + .map = trx40_mobo_map, + .connector_map = trx40_mobo_connector_map, + }, { 0 } /* terminator */ }; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 8c2f5c23e1b4..aa4c16ce0e57 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3647,6 +3647,7 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), ALC1220_VB_DESKTOP(0x0414, 0xa002), /* Gigabyte TRX40 Aorus Pro WiFi */ ALC1220_VB_DESKTOP(0x0db0, 0x0d64), /* MSI TRX40 Creator */ ALC1220_VB_DESKTOP(0x0db0, 0x543d), /* MSI TRX40 */ +ALC1220_VB_DESKTOP(0x26ce, 0x0a01), /* Asrock TRX40 Creator */ #undef ALC1220_VB_DESKTOP #undef USB_DEVICE_VENDOR_SPEC -- 2.25.1

1 0

[PATCH for-5.8] ASoC: amd: doesn't print error log if the return value is EPROBE_DEFER
by Hui Wang 22 May '20

22 May '20

The machine driver module and codec driver module don't have dependency, it is possible that the machine driver is loaded ahead of the codec driver, then the register_card() will fail and return EPROBE_DEFER, in this case the driver should not print error log since this is not a real failure. For example, I saw this log from a machine with amd renoir audio: acp_pdm_mach acp_pdm_mach.0: snd_soc_register_card(acp) failed: -517 After applying this patch, there is no error log to confuse users and audio works after the codec driver is loaded. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/soc/amd/acp-da7219-max98357a.c | 3 ++- sound/soc/amd/acp-rt5645.c | 3 ++- sound/soc/amd/acp3x-rt5682-max9836.c | 3 ++- sound/soc/amd/renoir/acp3x-rn.c | 7 ++++--- 4 files changed, 10 insertions(+), 6 deletions(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 9414d7269c4f..a7e755a563e8 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -440,7 +440,8 @@ static int cz_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(card, machine); ret = devm_snd_soc_register_card(&pdev->dev, &cz_card); if (ret) { - dev_err(&pdev->dev, + if (ret != -EPROBE_DEFER) + dev_err(&pdev->dev, "devm_snd_soc_register_card(%s) failed: %d\n", cz_card.name, ret); return ret; diff --git a/sound/soc/amd/acp-rt5645.c b/sound/soc/amd/acp-rt5645.c index 73b31f88a6b5..cffd24eae3a3 100644 --- a/sound/soc/amd/acp-rt5645.c +++ b/sound/soc/amd/acp-rt5645.c @@ -174,7 +174,8 @@ static int cz_probe(struct platform_device *pdev) platform_set_drvdata(pdev, card); ret = devm_snd_soc_register_card(&pdev->dev, &cz_card); if (ret) { - dev_err(&pdev->dev, + if (ret != -EPROBE_DEFER) + dev_err(&pdev->dev, "devm_snd_soc_register_card(%s) failed: %d\n", cz_card.name, ret); return ret; diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index e499c00e0c66..16bcaad9ead2 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -346,7 +346,8 @@ static int acp3x_probe(struct platform_device *pdev) ret = devm_snd_soc_register_card(&pdev->dev, &acp3x_card); if (ret) { - dev_err(&pdev->dev, + if (ret != -EPROBE_DEFER) + dev_err(&pdev->dev, "devm_snd_soc_register_card(%s) failed: %d\n", acp3x_card.name, ret); return ret; diff --git a/sound/soc/amd/renoir/acp3x-rn.c b/sound/soc/amd/renoir/acp3x-rn.c index 306134b89a82..95b616fcad30 100644 --- a/sound/soc/amd/renoir/acp3x-rn.c +++ b/sound/soc/amd/renoir/acp3x-rn.c @@ -54,9 +54,10 @@ static int acp_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(card, machine); ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { - dev_err(&pdev->dev, - "snd_soc_register_card(%s) failed: %d\n", - acp_card.name, ret); + if (ret != -EPROBE_DEFER) + dev_err(&pdev->dev, + "snd_soc_register_card(%s) failed: %d\n", + acp_card.name, ret); return ret; } return 0; -- 2.17.1

2 3

[PATCH RESEND 0/4] Minor WM8994 MFD/codec fixes
by Marek Szyprowski 22 May '20

22 May '20

Hi! This is a resend of the minor WM8994 MFD/codec driver fixes posted in last days of the February 2020: https://lore.kernel.org/patchwork/project/lkml/list/?series=431296 https://lore.kernel.org/patchwork/project/lkml/list/?series=431721 I hope this time the patches will find their way to mainline. Best regards, Marek Szyprowski Patch summary: Marek Szyprowski (4): mfd: wm8994: Fix driver operation if loaded as modules mfd: wm8994: Fix unbalanced calls to regulator_bulk_disable() mfd: wm8994: Silence warning about supplies during deferred probe ASoC: wm8994: Silence warnings during deferred probe drivers/mfd/wm8994-core.c | 8 +++++++- sound/soc/codecs/wm8994.c | 3 ++- 2 files changed, 9 insertions(+), 2 deletions(-) -- 2.17.1

4 17

[GIT PULL] soundwire updates for v5.8-rc1
by Vinod Koul 22 May '20

22 May '20

Hi Greg, Please pull to receive updates for soundwire. The big news the sdw_master_device getting completed along with sysfs attributes for master and slave devices. That closes the long cleanup work done by Pierre and Bard for the devices. The following changes since commit 8f3d9f354286745c751374f5f1fcafee6b3f3136: Linux 5.7-rc1 (2020-04-12 12:35:55 -0700) are available in the Git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/soundwire.git tags/soundwire-5.8-rc1 for you to fetch changes up to fcbcf1f7b56855ffe1cec768cb1013c5a4c854c3: soundwire: intel: use a single module (2020-05-20 19:24:55 +0530) ---------------------------------------------------------------- soundwire updates for v5.8-rc1 This contains sdw_master_device patches and other updates done by Intel folks. Details: - sdw_master_device to represent the master instances. - sysfs properties for sdw_master_device and sdw_slave. - Documentation update for TDM modes. - some code cleanup patches and odd updates. ---------------------------------------------------------------- Bard Liao (2): soundwire: bus: add unique bus id soundwire: master: add runtime pm support Guennadi Liakhovetski (2): soundwire: (cosmetic) remove multiple superfluous "else" statements soundwire: intel: (cosmetic) remove multiple superfluous "else" statements Kuninori Morimoto (1): soundwire: intel: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer Pierre-Louis Bossart (11): soundwire: qcom: fix error handling in probe Documentation: SoundWire: clarify TDM mode support soundwire: slave: don't init debugfs on device registration error soundwire: debugfs: clarify SDPX license with GPL-2.0-only soundwire: bus: reduce verbosity on enumeration soundwire: bus: rename sdw_bus_master_add/delete, add arguments soundwire: bus_type: introduce sdw_slave_type and sdw_master_type soundwire: bus_type: add sdw_master_device support soundwire: disco: s/ch/channels/ soundwire: master: add sysfs support soundwire: add Slave sysfs support Rander Wang (1): soundwire: intel: use a single module Samuel Zou (1): soundwire: qcom: Use IRQF_ONESHOT Vinod Koul (2): soundwire: fix trailing line in sysfs_slave.c soundwire: fix spelling mistake .../ABI/testing/sysfs-bus-soundwire-master | 23 ++ .../ABI/testing/sysfs-bus-soundwire-slave | 91 +++++++ Documentation/driver-api/soundwire/stream.rst | 89 +++++- Documentation/driver-api/soundwire/summary.rst | 7 +- drivers/soundwire/Makefile | 8 +- drivers/soundwire/bus.c | 71 +++-- drivers/soundwire/bus.h | 4 + drivers/soundwire/bus_type.c | 22 +- drivers/soundwire/cadence_master.c | 8 +- drivers/soundwire/debugfs.c | 2 +- drivers/soundwire/intel.c | 13 +- drivers/soundwire/intel_init.c | 4 +- drivers/soundwire/master.c | 172 ++++++++++++ drivers/soundwire/mipi_disco.c | 11 +- drivers/soundwire/qcom.c | 34 ++- drivers/soundwire/slave.c | 10 +- drivers/soundwire/sysfs_local.h | 14 + drivers/soundwire/sysfs_slave.c | 214 +++++++++++++++ drivers/soundwire/sysfs_slave_dpn.c | 300 +++++++++++++++++++++ include/linux/soundwire/sdw.h | 32 ++- include/linux/soundwire/sdw_type.h | 9 +- 21 files changed, 1066 insertions(+), 72 deletions(-) create mode 100644 Documentation/ABI/testing/sysfs-bus-soundwire-master create mode 100644 Documentation/ABI/testing/sysfs-bus-soundwire-slave create mode 100644 drivers/soundwire/master.c create mode 100644 drivers/soundwire/sysfs_local.h create mode 100644 drivers/soundwire/sysfs_slave.c create mode 100644 drivers/soundwire/sysfs_slave_dpn.c -- ~Vinod

2 1

[PATCH v2 0/7] ASoC: add soc-link
by Kuninori Morimoto 22 May '20

22 May '20

Hi Mark Current ALSA SoC is handling dai_link related operation, but it is implmemented directly without using function/macro, and at random place. This v2 patch-set creates new snd_soc_link_xxx() functions which handles dai_link related operation, and implmement these at new soc-link.c. v1 -> v2 - #include <sound/soc-link.h> is added on each c source file instead of soc.h - not have extra error message after snd_soc_link_xxx(), because it already indicate it via snc_link_ret() - snd_soc_link_compr_xxx() doesn't have rtd parameter, because it can be created from cstream Link: https://lore.kernel.org/r/877dx868op.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (7): ASoC: add soc-link.c ASoC: soc-link: move soc_rtd_xxx() ASoC: soc-link: remove unneeded parameter from snd_soc_link_xxx() ASoC: soc-link: add snd_soc_link_be_hw_params_fixup() ASoC: soc-link: add snd_soc_link_compr_startup() ASoC: soc-link: add snd_soc_link_compr_shutdown() ASoC: soc-link: add snd_soc_link_compr_set_params() include/sound/soc-link.h | 27 +++++++ sound/soc/Makefile | 2 +- sound/soc/soc-compress.c | 46 ++++-------- sound/soc/soc-core.c | 18 ++--- sound/soc/soc-dai.c | 9 ++- sound/soc/soc-link.c | 150 +++++++++++++++++++++++++++++++++++++++ sound/soc/soc-pcm.c | 86 ++++------------------ 7 files changed, 219 insertions(+), 119 deletions(-) create mode 100644 include/sound/soc-link.h create mode 100644 sound/soc/soc-link.c -- 2.17.1 Thank you for your help !! Best regards --- Kuninori Morimoto

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Difference between PlaybackMixer and PlaybackMixerElem
by Curtis Malainey 22 May '20

22 May '20

Hello Jaroslav, Just a quick question, what is the difference between PlaybackMixer and PlaybackMixerElem? I can see that one can be parsed using snd_use_case_parse _selem_id? Is that the only difference? Thanks Curtis

1 0

[PATCH] ASoC: Intel: kbl_rt5663_rt5514_max98927: Split be_hw_params_fixup function
by Lukasz Majczak 21 May '20

21 May '20

Split be_hw_params_fixup function for different codecs as current common function, leads to crash while trying to get snd_soc_dpcm with container_of() macro in kabylake_ssp_fixup(). The crash call path looks as below: soc_pcm_hw_params() snd_soc_dai_hw_params(codec_dai, substream, &codec_params); rtd->dai_link->be_hw_params_fixup(rtd, params) kabylake_ssp_fixup() In this case, codec_params is just a copy of an internal structure and is not embedded into struct snd_soc_dpcm thus we cannot use container_of() on it. Signed-off-by: Lukasz Majczak <lma(a)semihalf.com> --- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 130 ++++++++++++------ 1 file changed, 85 insertions(+), 45 deletions(-) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 1b1f8d7a4ea3..12a9983979e0 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -171,8 +171,8 @@ static const struct snd_soc_dapm_route kabylake_map[] = { { "hs_in", NULL, "ssp1 Rx" }, { "ssp1 Rx", NULL, "AIF Capture" }, - { "codec1_in", NULL, "ssp0 Rx" }, - { "ssp0 Rx", NULL, "AIF1 Capture" }, + { "codec1_in", NULL, "DMIC01 Rx" }, + { "DMIC01 Rx", NULL, "AIF1 Capture" }, /* IV feedback path */ { "codec0_fb_in", NULL, "ssp0 Rx"}, @@ -328,42 +328,52 @@ static const struct snd_soc_ops kabylake_rt5663_fe_ops = { .startup = kbl_fe_startup, }; -static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) +static void kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params, snd_pcm_format_t pcm_fmt) { struct snd_interval *rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *chan = hw_param_interval(params, + struct snd_interval *channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - struct snd_soc_dpcm *dpcm = container_of( - params, struct snd_soc_dpcm, hw_params); - struct snd_soc_dai_link *fe_dai_link = dpcm->fe->dai_link; - struct snd_soc_dai_link *be_dai_link = dpcm->be->dai_link; /* * The ADSP will convert the FE rate to 48k, stereo, 24 bit */ - if (!strcmp(fe_dai_link->name, "Kbl Audio Port") || - !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") || - !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) { - rate->min = rate->max = 48000; - chan->min = chan->max = 2; - snd_mask_none(fmt); - snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); - } else if (!strcmp(fe_dai_link->name, "Kbl Audio DMIC cap")) { + + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + snd_mask_none(fmt); + snd_mask_set_format(fmt, pcm_fmt); +} + +static int kabylake_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + kabylake_ssp_fixup(rtd, params, SNDRV_PCM_FORMAT_S16_LE); + return 0; +} + +static int kabylake_ssp1_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + + kabylake_ssp_fixup(rtd, params, SNDRV_PCM_FORMAT_S24_LE); + return 0; +} + +static int kabylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + if (params_channels(params) == 2 || DMIC_CH(dmic_constraints) == 2) - chan->min = chan->max = 2; + channels->min = channels->max = 2; else - chan->min = chan->max = 4; - } - /* - * The speaker on the SSP0 supports S16_LE and not S24_LE. - * thus changing the mask here - */ - if (!strcmp(be_dai_link->name, "SSP0-Codec")) - snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); + channels->min = channels->max = 4; return 0; } @@ -400,20 +410,6 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, int ret = 0, j; for_each_rtd_codec_dais(rtd, j, codec_dai) { - if (!strcmp(codec_dai->component->name, RT5514_DEV_NAME)) { - ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0, 8, 16); - if (ret < 0) { - dev_err(rtd->dev, "set TDM slot err:%d\n", ret); - return ret; - } - - ret = snd_soc_dai_set_sysclk(codec_dai, - RT5514_SCLK_S_MCLK, 24576000, SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(rtd->dev, "set sysclk err: %d\n", ret); - return ret; - } - } if (!strcmp(codec_dai->component->name, MAXIM_DEV0_NAME)) { ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x30, 3, 8, 16); if (ret < 0) { @@ -433,10 +429,37 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, return ret; } +static int kabylake_dmic01_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int ret = 0; + + ret = snd_soc_dai_set_tdm_slot(rtd->codec_dai, 0xF, 0, 8, 16); + if (ret < 0) { + dev_err(rtd->dev, "set TDM slot err:%d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(rtd->codec_dai, + RT5514_SCLK_S_MCLK, 24576000, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(rtd->dev, "set sysclk err: %d\n", ret); + return ret; + } + + return ret; +} + static struct snd_soc_ops kabylake_ssp0_ops = { .hw_params = kabylake_ssp0_hw_params, }; +static struct snd_soc_ops kabylake_dmic01_ops = { + .hw_params = kabylake_dmic01_hw_params, +}; + + static const unsigned int channels_dmic[] = { 4, }; @@ -507,14 +530,19 @@ SND_SOC_DAILINK_DEF(ssp0_pin, SND_SOC_DAILINK_DEF(ssp0_codec, DAILINK_COMP_ARRAY( /* Left */ COMP_CODEC(MAXIM_DEV0_NAME, KBL_MAXIM_CODEC_DAI), - /* Right */COMP_CODEC(MAXIM_DEV1_NAME, KBL_MAXIM_CODEC_DAI), - /* dmic */ COMP_CODEC(RT5514_DEV_NAME, KBL_REALTEK_DMIC_CODEC_DAI))); + /* Right */COMP_CODEC(MAXIM_DEV1_NAME, KBL_MAXIM_CODEC_DAI))); SND_SOC_DAILINK_DEF(ssp1_pin, DAILINK_COMP_ARRAY(COMP_CPU("SSP1 Pin"))); SND_SOC_DAILINK_DEF(ssp1_codec, DAILINK_COMP_ARRAY(COMP_CODEC(RT5663_DEV_NAME, KBL_REALTEK_CODEC_DAI))); +SND_SOC_DAILINK_DEF(dmic01_pin, + DAILINK_COMP_ARRAY(COMP_CPU("DMIC01 Pin"))); +SND_SOC_DAILINK_DEF(dmic01_codec, + DAILINK_COMP_ARRAY( + COMP_CODEC(RT5514_DEV_NAME, KBL_REALTEK_DMIC_CODEC_DAI))); + SND_SOC_DAILINK_DEF(idisp1_pin, DAILINK_COMP_ARRAY(COMP_CPU("iDisp1 Pin"))); SND_SOC_DAILINK_DEF(idisp1_codec, @@ -618,9 +646,8 @@ static struct snd_soc_dai_link kabylake_dais[] = { SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, .ignore_pmdown_time = 1, - .be_hw_params_fixup = kabylake_ssp_fixup, + .be_hw_params_fixup = kabylake_ssp0_fixup, .dpcm_playback = 1, - .dpcm_capture = 1, .ops = &kabylake_ssp0_ops, SND_SOC_DAILINK_REG(ssp0_pin, ssp0_codec, platform), }, @@ -632,12 +659,25 @@ static struct snd_soc_dai_link kabylake_dais[] = { .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, .ignore_pmdown_time = 1, - .be_hw_params_fixup = kabylake_ssp_fixup, + .be_hw_params_fixup = kabylake_ssp1_fixup, .ops = &kabylake_rt5663_ops, .dpcm_playback = 1, .dpcm_capture = 1, SND_SOC_DAILINK_REG(ssp1_pin, ssp1_codec, platform), }, + { + .name = "dmic01", + .id = 2, + .no_pcm = 1, + .dai_fmt = SND_SOC_DAIFMT_DSP_B | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = kabylake_dmic_fixup, + .dpcm_capture = 1, + .ops = &kabylake_dmic01_ops, + SND_SOC_DAILINK_REG(dmic01_pin, dmic01_codec, platform), + }, { .name = "iDisp1", .id = 3, base-commit: a4f6fc98cd2fa1774bcaeb248c67156ef9402a56 -- 2.25.1

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