October 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v1] ALSA: usb-audio: sound: usb: usb true/false for bool return type
by Saurav Girepunje 30 Oct '19

30 Oct '19

Use true/false for bool type return in uac_clock_source_is_valid(). Signed-off-by: Saurav Girepunje <saurav.girepunje(a)gmail.com> --- Changes in v1: - Update the patch subject suggested by Takashi Iwai <tiwai(a)suse.de> - Drop "!!" but keep the ternary operator in last return statement so that function will return always true/false. sound/usb/clock.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 72e9bdf76115..8d5538063598 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -167,21 +167,21 @@ static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, snd_usb_find_clock_source_v3(chip->ctrl_intf, source_id); if (!cs_desc) - return 0; + return false; bmControls = le32_to_cpu(cs_desc->bmControls); } else { /* UAC_VERSION_1/2 */ struct uac_clock_source_descriptor *cs_desc = snd_usb_find_clock_source(chip->ctrl_intf, source_id); if (!cs_desc) - return 0; + return false; bmControls = cs_desc->bmControls; } /* If a clock source can't tell us whether it's valid, we assume it is */ if (!uac_v2v3_control_is_readable(bmControls, UAC2_CS_CONTROL_CLOCK_VALID)) - return 1; + return true; err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR, USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, @@ -193,10 +193,10 @@ static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, dev_warn(&dev->dev, "%s(): cannot get clock validity for id %d\n", __func__, source_id); - return 0; + return false; } - return !!data; + return data ? true : false; } static int __uac_clock_find_source(struct snd_usb_audio *chip, int entity_id, -- 2.20.1

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[alsa-devel] axfer: break transmission of audio data frame when transmission backend queues no PCM event within timeout
by GitHub pull_request - opened 30 Oct '19

30 Oct '19

alsa-project/alsa-utils pull request #31 was opened from takaswie: Axfer internally has the waiter abstraction for polling system call. This is used without explicit timeout for IRQ-based scheduling model. As a result, axfer process is corrupted to be blocked due to a call of polling system call when transmission backend has a bug to queue PCM event in I/O thread or hardware IRQ context. This patchset implements error reporting for such buggy backend. When call of polling system call returns with timeout, axfer reports it as a bug. The timeout is equivalent to the size of buffer. Precisely it should be the size of period, however the larger size is used to stabilize runtime for existent PCM plugins. ``` Takashi Sakamoto (6): axfer: return the number of file descriptors for I/O events from select(2) waiter axfer: fix to return error code when a call of select(2) fails axfer: return ETIMEDOUT when no event occurs after waiter expiration axfer: handle -ETIMEDOUT before handle mmap I/O operation axfer: code refactoring for a helper function to wait for avail buffer space axfer: handle -ETIMEDOUT before non-block I/O operation axfer/waiter-select.c | 4 +-- axfer/xfer-libasound-irq-mmap.c | 19 +++++++++- axfer/xfer-libasound-irq-rw.c | 62 +++++++++++++++++++-------------- axfer/xfer-libasound.c | 21 +++++++---- 4 files changed, 70 insertions(+), 36 deletions(-) ``` Request URL : https://github.com/alsa-project/alsa-utils/pull/31 Patch URL : https://github.com/alsa-project/alsa-utils/pull/31.patch Repository URL: https://github.com/alsa-project/alsa-utils

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[alsa-devel] [PATCH 1/2] ASoC: dt-bindings: fsl_asrc: add compatible string for imx8qm
by Shengjiu Wang 30 Oct '19

30 Oct '19

In order to support the two asrc modules in imx8qm, we need to add compatible string "fsl,imx8qm-asrc0" and "fsl,imx8qm-asrc1" Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Documentation/devicetree/bindings/sound/fsl,asrc.txt | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/fsl,asrc.txt b/Documentation/devicetree/bindings/sound/fsl,asrc.txt index 1d4d9f938689..cd2bd3daa7e1 100644 --- a/Documentation/devicetree/bindings/sound/fsl,asrc.txt +++ b/Documentation/devicetree/bindings/sound/fsl,asrc.txt @@ -8,7 +8,8 @@ three substreams within totally 10 channels. Required properties: - - compatible : Contains "fsl,imx35-asrc" or "fsl,imx53-asrc". + - compatible : Contains "fsl,imx35-asrc", "fsl,imx53-asrc", + "fsl,imx8qm-asrc0" or "fsl,imx8qm-asrc1". - reg : Offset and length of the register set for the device. -- 2.21.0

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[alsa-devel] Adding support for Android's Wired Audio Headset Specification v1.1
by Mads Lønsethagen 30 Oct '19

30 Oct '19

Posting this to alsa-devel without being actually sure it belongs here, so please do tell if it belongs elsewhere. But here goes: If you have e.g. a Bose QC25 headset (wired, not bluetooth) in Linux, you can't get the volume buttons nor the play/pause button on the headphone cable itself to work. As far as I know, these buttons works in both Android and Windows 10, so it would be nice if they would work with plain Linux too. This StackExchange thread discusses how it works: https://electronics.stackexchange.com/questions/242883/how-do-media-buttons… As one poster on StackExchange mentions, Android has a defined standard on these kind of buttons: https://source.android.com/devices/accessories/headset/plug-headset-spec Since these kind of button presses must be decoded from impedance changes in the 3.5mm jack connector, does it belong in alsa somewhere? Bose QC25 buttons works on Intel HDA cards on new Dell XPS laptops, so I'm guessing it should be possible within snd-hda-intel... - Mads

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[alsa-devel] [PATCH v8 0/9] adapt SOF to use snd-hda-codec-hdmi
by Kai Vehmanen 29 Oct '19

29 Oct '19

Hey, here's the 8th round for this series that adapts SOF to use snd-hda-codec-hdmi (patch_hdmi.c) codec driver instead of hdac_hdmi (soc/codecs/hdac_hdmi.c). The primary goal is to unify the HDMI codec implementation between DSP and non-DSP HDA configurations, offer same interface to user-space and reduce maintenance load for all. Intended to go in via Mark's ASoC tree. v8 changes: - rebased on top of Mark's for-next (50484b6a523a) - addressed Takashi's review comment to patch 5 - addressed two sparse warnings in patch 3 reported by kbuild test robot <lkp(a)intel.com> - fixed itnull.cocci warning in patch 3 reported by Julia Lawall <julia.lawall(a)lip6.fr> and kbuild test robot <lkp(a)intel.com> Feature and testing info: - Tested on multiple Intel platforms supported by SOF. - Tested with ALSA console tools as well as with Pulseaudio. - requires Pulseaudio 12.x or newer, see https://lists.freedesktop.org/archives/pulseaudio-discuss/2019-August/03135… - HDMI, DP, DP-MST with multi-monitor use-scenarios work ok. - New feature for SOF: ELD /proc fs works just like in DSP-less mode. - New feature for SOF: jack detection works out-of-the-box with Pulseaudio (no need for card specific UCM for HDMI) Kai Vehmanen (9): ALSA: hda/hdmi - implement mst_no_extra_pcms flag ASoC: hdac_hda: add support for HDMI/DP as a HDA codec ASoC: Intel: skl-hda-dsp-generic: use snd-hda-codec-hdmi ASoC: Intel: skl-hda-dsp-generic: fix include guard name ASoC: SOF: Intel: add support for snd-hda-codec-hdmi ASoC: Intel: bxt-da7219-max98357a: common hdmi codec support ASoC: Intel: glk_rt5682_max98357a: common hdmi codec support ASoC: intel: sof_rt5682: common hdmi codec support ASoC: Intel: bxt_rt298: common hdmi codec support include/sound/hda_codec.h | 1 + include/sound/soc-acpi.h | 2 + sound/pci/hda/patch_hdmi.c | 19 ++- sound/soc/codecs/hdac_hda.c | 114 ++++++++++++++++-- sound/soc/codecs/hdac_hda.h | 13 +- sound/soc/intel/boards/Makefile | 10 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 11 ++ sound/soc/intel/boards/bxt_rt298.c | 11 ++ sound/soc/intel/boards/glk_rt5682_max98357a.c | 11 ++ sound/soc/intel/boards/hda_dsp_common.c | 85 +++++++++++++ sound/soc/intel/boards/hda_dsp_common.h | 32 +++++ sound/soc/intel/boards/skl_hda_dsp_common.c | 6 + sound/soc/intel/boards/skl_hda_dsp_common.h | 27 ++++- sound/soc/intel/boards/skl_hda_dsp_generic.c | 1 + sound/soc/intel/boards/sof_rt5682.c | 11 ++ sound/soc/sof/intel/Kconfig | 10 ++ sound/soc/sof/intel/hda-codec.c | 22 +++- sound/soc/sof/intel/hda.c | 6 + sound/soc/sof/intel/hda.h | 6 +- 19 files changed, 367 insertions(+), 31 deletions(-) create mode 100644 sound/soc/intel/boards/hda_dsp_common.c create mode 100644 sound/soc/intel/boards/hda_dsp_common.h -- 2.17.1

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[alsa-devel] [PATCH 00/26] ASoC: SOF: enable S0ix support for Intel platforms
by Pierre-Louis Bossart 29 Oct '19

29 Oct '19

In S0ix (aka S0 Idle), the audio DSP can enter the D0i3 substate and keep some of the pipelines and processing active for e.g. Hotwording and other acoustic event detection while optimizing power consumption. This patchset mainly adds flags and state machines to track which streams may remain active (tagged as such in the topology file), and keep them active when the ALSA core tries to suspend everything. The Intel specific parts are limited to the changes in the IPC, where instead of using an SRAM window, the IPC commands needed to enter and exit D0i3 are sent in a compressed form using IPC registers themselves. This patchset is made of 26 patches, which is rather large, but each change is rather small and contained. The code was initially reviewed on GitHub, with 165+ comments provided and no regressions found [1]. The S0ix mode was tested on a CometLake platform. Future improvements will consist in an additional idle detection so that the D0i3 mode can be entered even on devices where S0ix is not enabled. [1] https://github.com/thesofproject/linux/pull/1337 Keyon Jie (26): ASoC: SOF: add a field to store the current D0 substate of DSP ASoC: SOF: reset default d0_substate at probe() and resume() ASoC: SOF: add set_power_state() to dsp_ops for power state update ASoC: SOF: Intel: hda-dsp: Add helper for setting DSP D0ix substate ASoC: SOF: Intel: CNL: add set_power_state() ops ASoC: SOF: Intel: APL: add set_power_state() ops ASoC: SOF: add flag to snd_sof_pcm_stream for D0i3 compatible stream ASoC: SOF: token: add tokens for PCM compatible with D0i3 substate ASoC: SOF: topology: parse and store d0i3_compatible flag ASoC: SOF: Intel: hda-dsp: align the comments for D0I3C update ASoC: SOF: Intel: HDA: use macro for register polling retry count ASoC: SOF: PM: rename sof_send_pm_ipc to sof_send_pm_ctx_ipc ASoC: SOF: ipc: introduce message for DSP power gating ASoC: SOF: Intel: hda-ipc: Don't read mailbox for PM_GATE reply ASoC: SOF: Intel: HDA: add cAVS specific compact IPC header file ASoC: SOF: configure D0ix IPC flags in set_power_state ASoC: SOF: PM: add helpers for setting D0 substate for ADSP ASoC: SOF: Intel: CNL: add support for sending compact IPC ASoC: SOF: add a flag to indicate the system suspend target ASoC: SOF: add a flag suspend_ignored for sof stream ASoC: SOF: PM: implement prepare/complete callbacks ASoC: SOF: ignore suspend/resume for D0ix compatible streams ASoC: SOF: Intel: hda-dsp: implement suspend/resume for S0ix<->S0 transition ASoC: SOF: return -ENOTSUPP if D0I3 is not supported ASoC: SOF: PM: Add support for DSP D0i3 state when entering S0ix ASoC: SOF: pci: Add prepare/complete PM callbacks include/sound/sof/header.h | 1 + include/sound/sof/pm.h | 8 +++ include/uapi/sound/sof/tokens.h | 4 ++ sound/soc/sof/core.c | 3 ++ sound/soc/sof/intel/apl.c | 1 + sound/soc/sof/intel/cnl.c | 43 +++++++++++++-- sound/soc/sof/intel/hda-dsp.c | 91 +++++++++++++++++++++++++++++++ sound/soc/sof/intel/hda-ipc.c | 6 ++- sound/soc/sof/intel/hda-ipc.h | 51 ++++++++++++++++++ sound/soc/sof/intel/hda.h | 12 +++++ sound/soc/sof/ops.h | 10 ++++ sound/soc/sof/pcm.c | 31 +++++++++++ sound/soc/sof/pm.c | 95 +++++++++++++++++++++++++++++++-- sound/soc/sof/sof-pci-dev.c | 2 + sound/soc/sof/sof-priv.h | 23 ++++++++ sound/soc/sof/topology.c | 43 ++++++++++++++- 16 files changed, 413 insertions(+), 11 deletions(-) create mode 100644 sound/soc/sof/intel/hda-ipc.h -- 2.20.1

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[alsa-devel] [PATCH] ASoC: Intel: bxt_da7219_max98357a: update dai_link platform name
by mac.chiang＠intel.com 29 Oct '19

29 Oct '19

From: Mac Chiang <mac.chiang(a)intel.com> redefine soc platform name "0000:00:1f.3" if cml core Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> Signed-off-by: Sathya Prakash M R <sathya.prakash.m.r(a)intel.com> --- sound/soc/intel/boards/bxt_da7219_max98357a.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index ac1dea5..38b7dbe 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -40,6 +40,13 @@ struct bxt_card_private { struct list_head hdmi_pcm_list; }; +static struct snd_soc_dai_link_component platform_component[] = { + { + /* name might be overridden during probe */ + .name = "0000:00:1f.3" + } +}; + enum { BXT_DPCM_AUDIO_PB = 0, BXT_DPCM_AUDIO_CP, @@ -708,6 +715,7 @@ static int broxton_audio_probe(struct platform_device *pdev) broxton_dais[i].name = "SSP0-Codec"; broxton_dais[i].cpus->dai_name = "SSP0 Pin"; } + broxton_dais[i].platforms = platform_component; } } -- 2.7.4

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[alsa-devel] [PATCH] ASoC: SOF - remove the dead code (skylake/kabylake)
by Jaroslav Kysela 29 Oct '19

29 Oct '19

Appearently the CONFIG_SND_SOC_SOF_KABYLAKE and CONFIG_SND_SOC_SOF_SKYLAKE options are not present in Kconfig and 'struct snd_sof_dsp_ops sof_skl_ops' is not declared in the code, too. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/intel/hda.h | 1 - sound/soc/sof/sof-pci-dev.c | 44 ------------------------------------- 2 files changed, 45 deletions(-) diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 23e430d3e056..f521c48ec2eb 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -596,7 +596,6 @@ extern struct snd_soc_dai_driver skl_dai[]; */ extern const struct snd_sof_dsp_ops sof_apl_ops; extern const struct snd_sof_dsp_ops sof_cnl_ops; -extern const struct snd_sof_dsp_ops sof_skl_ops; extern const struct sof_intel_dsp_desc apl_chip_info; extern const struct sof_intel_dsp_desc cnl_chip_info; diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index 3a9e0e2a150d..bca3fb791a6b 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -168,42 +168,6 @@ static const struct sof_dev_desc icl_desc = { }; #endif -#if IS_ENABLED(CONFIG_SND_SOC_SOF_SKYLAKE) -static const struct sof_dev_desc skl_desc = { - .machines = snd_soc_acpi_intel_skl_machines, - .resindex_lpe_base = 0, - .resindex_pcicfg_base = -1, - .resindex_imr_base = -1, - .irqindex_host_ipc = -1, - .resindex_dma_base = -1, - .chip_info = &skl_chip_info, - .default_fw_path = "intel/sof", - .default_tplg_path = "intel/sof-tplg", - .nocodec_fw_filename = "sof-skl.ri", - .nocodec_tplg_filename = "sof-skl-nocodec.tplg", - .ops = &sof_skl_ops, - .arch_ops = &sof_xtensa_arch_ops -}; -#endif - -#if IS_ENABLED(CONFIG_SND_SOC_SOF_KABYLAKE) -static const struct sof_dev_desc kbl_desc = { - .machines = snd_soc_acpi_intel_kbl_machines, - .resindex_lpe_base = 0, - .resindex_pcicfg_base = -1, - .resindex_imr_base = -1, - .irqindex_host_ipc = -1, - .resindex_dma_base = -1, - .chip_info = &skl_chip_info, - .default_fw_path = "intel/sof", - .default_tplg_path = "intel/sof-tplg", - .nocodec_fw_filename = "sof-kbl.ri", - .nocodec_tplg_filename = "sof-kbl-nocodec.tplg", - .ops = &sof_skl_ops, - .arch_ops = &sof_xtensa_arch_ops -}; -#endif - #if IS_ENABLED(CONFIG_SND_SOC_SOF_TIGERLAKE) static const struct sof_dev_desc tgl_desc = { .machines = snd_soc_acpi_intel_tgl_machines, @@ -407,14 +371,6 @@ static const struct pci_device_id sof_pci_ids[] = { { PCI_DEVICE(0x8086, 0xa348), .driver_data = (unsigned long)&cfl_desc}, #endif -#if IS_ENABLED(CONFIG_SND_SOC_SOF_KABYLAKE) - { PCI_DEVICE(0x8086, 0x9d71), - .driver_data = (unsigned long)&kbl_desc}, -#endif -#if IS_ENABLED(CONFIG_SND_SOC_SOF_SKYLAKE) - { PCI_DEVICE(0x8086, 0x9d70), - .driver_data = (unsigned long)&skl_desc}, -#endif #if IS_ENABLED(CONFIG_SND_SOC_SOF_ICELAKE) { PCI_DEVICE(0x8086, 0x34C8), .driver_data = (unsigned long)&icl_desc}, -- 2.20.1

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[alsa-devel] [PATCH v2] ASoC: intel - fix the card names
by Jaroslav Kysela 29 Oct '19

29 Oct '19

Those strings are exposed to the user space as the card name thus used in the GUIs. The common standard is to avoid '_' here. The worst case is 'sof-skl_hda_card' string. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 2 +- sound/soc/intel/boards/skl_hda_dsp_generic.c | 2 +- sound/soc/intel/boards/sof_rt5682.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 74dda8784f1a..d14ca327c684 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -623,7 +623,7 @@ static int kabylake_card_late_probe(struct snd_soc_card *card) * kabylake audio machine driver for MAX98927 + RT5514 + RT5663 */ static struct snd_soc_card kabylake_audio_card = { - .name = "kbl_r5514_5663_max", + .name = "kbl-r5514-5663-max", .owner = THIS_MODULE, .dai_link = kabylake_dais, .num_links = ARRAY_SIZE(kabylake_dais), diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index 1778acdc367c..e8d676c192f6 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -90,7 +90,7 @@ skl_hda_add_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *link) } static struct snd_soc_card hda_soc_card = { - .name = "skl_hda_card", + .name = "hda-dsp", .owner = THIS_MODULE, .dai_link = skl_hda_be_dai_links, .dapm_widgets = skl_hda_widgets, diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 4f6e58c3954a..320a9b9273db 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -370,7 +370,7 @@ static int dmic_init(struct snd_soc_pcm_runtime *rtd) /* sof audio machine driver for rt5682 codec */ static struct snd_soc_card sof_audio_card_rt5682 = { - .name = "sof_rt5682", + .name = "rt5682", /* the sof- prefix is added by the core */ .owner = THIS_MODULE, .controls = sof_controls, .num_controls = ARRAY_SIZE(sof_controls), -- 2.20.1

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[alsa-devel] [ANNOUNCE] Draft schedule for 2019 Linux Audio miniconference
by Mark Brown 29 Oct '19

29 Oct '19

Hi, As in previous years we're going to have an audio miniconference so we can get together and talk through issues, especially design decisions, face to face. This year's event will be held on October 31st in Lyon, France, the day after ELC-E. This will be held at the Lyon Convention Center (the ELC-E venue) and will be free of charge to attend, generously sponsored by Intel. The agenda is looking pretty full now, what we've got so far is: - Use case management enhancements (Curtis) - DSP framework integration (Liam) - SOF integration with ACPI and DT (Daniel) - SOF support for non-modular drivers (Daniel) - Gapless compressed playback (Patrick) - Soundwire status (Liam?) - Multi-CPU support - Mixing with HDA/I2S in a single system - PCM issues (Patrick) - PCM device volume control - Attaching timestamps to buffers - Virtualization (Liam) - User experiences (Jerome) - Unified graph (Patrick) - Componentisation status/plans (Me) If you're planning to attend please fill out the form here: https://docs.google.com/forms/d/e/1FAIpQLSd6FV-tWYZ1fHlCmyzQhG98OtYH9W9GX-1… This event will be covered by the same code of conduct as ELC-E. Thanks again to Intel for supporting this event. Thanks, Mark

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