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

[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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[alsa-devel] [PATCH] usb: clock.c : usb true/false for bool return type
by Saurav Girepunje 29 Oct '19

29 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> --- 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 6b8c14f9b5d4..8b8ab83fac0d 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -165,21 +165,21 @@ 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, @@ -191,10 +191,10 @@ 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] [PATCH 00/21] Enable BACO for power savings (v2)
by Alex Deucher 29 Oct '19

29 Oct '19

This patch set enables BACO (Bus Active Chip Off) for power savings on VI+ asics. Similar to PowerXpress and Hybrid Graphics (PX/HG) laptops, we can disable GPUs at runtime when they are not in use is they support BACO. The runtime pm code in amdgpu was originally developed for PX/HG laptops, so it was pretty entangled with the vga_switcheroo and ACPI code. Since the GPU contains an audio codec for HDMI/DP, there is some interaction with the hda driver. I am by no means an expert on alsa, so any advice on those patches is much appreciated. I had to enable runtime pm on the hda device to allow the GPU to enter runtime pm because they are linked. These patches depend on the clean patches and CI/VI BACO patches I sent out earlier today. The full tree can be found here: https://cgit.freedesktop.org/~agd5f/linux/log/?h=baco TODO: - Turn off runtime pm when KFD is active - Make sure audio still works - Make sure PX/HG still works v2: - Split whitespace changes from baco callback addition - Fix pci ref counting in px/hg handling in hda driver - Handle CONFIG_ACPI properly - Split adding new hda pci ids from adding runpm flag Alex Deucher (21): drm/amdgpu: add asic callback for BACO support drm/amdgpu/soc15: move struct definition around to align with other soc15 asics drm/amdgpu: add supports_baco callback for soc15 asics. (v2) drm/amdgpu: add supports_baco callback for SI asics. drm/amdgpu: add supports_baco callback for CIK asics. drm/amdgpu: add supports_baco callback for VI asics. drm/amdgpu: add supports_baco callback for NV asics. drm/amdgpu: add a amdgpu_device_supports_baco helper drm/amdgpu: rename amdgpu_device_is_px to amdgpu_device_supports_boco (v2) drm/amdgpu: add additional boco checks to runtime suspend/resume (v2) drm/amdgpu: split swSMU baco_reset into enter and exit drm/amdgpu: add helpers for baco entry and exit drm/amdgpu: add baco support to runtime suspend/resume drm/amdgpu: start to disentangle boco from runtime pm drm/amdgpu: disentangle runtime pm and vga_switcheroo drm/amdgpu: enable runtime pm on BACO capable boards if runpm=1 drm/amdgpu/runpm: enable runpm on baco capable VI+ asics ALSA: hda/hdmi - fix vgaswitcheroo detection for AMD ALSA: hda/hdmi - Add new pci ids for AMD GPU display audio ALSA: hda/hdmi - enable runtime pm for newer AMD display audio ALSA: hda/hdmi - enable automatic runtime pm for AMD HDMI codecs by default drivers/gpu/drm/amd/amdgpu/amdgpu.h | 12 +- drivers/gpu/drm/amd/amdgpu/amdgpu_device.c | 106 ++++++++++++++++-- drivers/gpu/drm/amd/amdgpu/amdgpu_drv.c | 63 ++++++----- drivers/gpu/drm/amd/amdgpu/amdgpu_kms.c | 20 +++- drivers/gpu/drm/amd/amdgpu/cik.c | 18 +++ drivers/gpu/drm/amd/amdgpu/nv.c | 18 ++- drivers/gpu/drm/amd/amdgpu/si.c | 6 + drivers/gpu/drm/amd/amdgpu/soc15.c | 36 +++++- drivers/gpu/drm/amd/amdgpu/vi.c | 22 ++++ drivers/gpu/drm/amd/powerplay/amdgpu_smu.c | 20 +++- drivers/gpu/drm/amd/powerplay/arcturus_ppt.c | 3 +- .../gpu/drm/amd/powerplay/inc/amdgpu_smu.h | 6 +- drivers/gpu/drm/amd/powerplay/inc/smu_v11_0.h | 3 +- drivers/gpu/drm/amd/powerplay/navi10_ppt.c | 3 +- drivers/gpu/drm/amd/powerplay/smu_v11_0.c | 9 +- drivers/gpu/drm/amd/powerplay/vega20_ppt.c | 3 +- sound/pci/hda/hda_intel.c | 84 +++++++++++++- sound/pci/hda/patch_hdmi.c | 1 + 18 files changed, 371 insertions(+), 62 deletions(-) -- 2.23.0

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[alsa-devel] axfer: call of polling system call never return when driver fails to queue no period-wakeup event due to the lack of hw IRQ context
by GitHub issues - edited 29 Oct '19

29 Oct '19

alsa-project/alsa-utils issue #30 was edited from takaswie: Axfer calls polling system call(i.e. poll(2), epoll(7), select(7)) without explicit timeout when running for IRQ-based scheduling model. In a case that driver achieves to control sound hardware for periodical hardware IRQ, the call can return due to any event queued by the driver in hardware IRQ context. However, in a case that driver fails to control it, the call cannot return. Especially, this situation easily occurs when developers work to support hardware newly. It's better to care for the situation. How to reproduce: 1. prepare sound hardware which driver fail to control for periodical hardware IRQ. 2. execute axfer for the driver/hardware with options to: * use IRQ-based scheduling model: `--sched-model=irq` * use polling system call: without `--test-nowait` 3. the running process blocks as a call of polling system call (defaultly poll(2) by snd_pcm_wait()) Issue URL : https://github.com/alsa-project/alsa-utils/issues/30 Repository URL: https://github.com/alsa-project/alsa-utils

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[alsa-devel] axfer: call of polling system call never return when driver fails to queue any events due to the lack of hw IRQ context
by GitHub issues - opened 29 Oct '19

29 Oct '19

alsa-project/alsa-utils issue #30 was opened from takaswie: Axfer calls polling system call(i.e. poll(2), epoll(7), select(7)) without explicit timeout when running for IRQ-based scheduling model. In a case that driver achieves to control sound hardware for periodical hardware IRQ, the call can return due to any event queued by the driver in hardware IRQ context. However, in a case that driver fails to control it, the call cannot return. Especially, this situation easily occurs when developers work to support hardware newly. It's better to care for the situation. How to reproduce: 1. prepare sound hardware which driver fail to control for periodical hardware IRQ. 2. execute axfer for the driver/hardware with options to: * use IRQ-based scheduling model: `--sched-model=irq` * use polling system call: without `--test-nowait` 3. the running process blocks as a call of polling system call (defaultly poll(2) by snd_pcm_wait()) Issue URL : https://github.com/alsa-project/alsa-utils/issues/30 Repository URL: https://github.com/alsa-project/alsa-utils

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[alsa-devel] [PATCH v8 0/6] Add HDMI jack support on RK3288
by Cheng-Yi Chiang 29 Oct '19

29 Oct '19

This patch series supports HDMI jack reporting on RK3288, which uses DRM dw-hdmi driver and hdmi-codec codec driver. The previous discussion about reporting jack status using hdmi-notifier and drm_audio_component is at https://lore.kernel.org/patchwork/patch/1083027/ The new approach is to use a callback mechanism that is specific to hdmi-codec. The dependent change on hdmi-codec.c 6fa5963c37a2 ASoC: hdmi-codec: Add an op to set callback function for plug event has been merged to upstream. Changes from v7 to v8: 1. rockchip_max98090: Allow three different use cases: max98090-only: For backward compatibility where DTS does not specify HDMI node. HDMI-only: For HDMI-only board like veyron-mickey. max98090 + HDMI: For other veyron boards. Pass different compatible string to specify the use case. 2. Add more maintainers to cc-list for new device property reviewing. Cheng-Yi Chiang (6): drm: bridge: dw-hdmi: Report connector status using callback ASoC: rockchip-max98090: Support usage with and without HDMI ASoC: rockchip_max98090: Optionally support HDMI use case ASoC: rockchip_max98090: Add HDMI jack support ARM: dts: rockchip: Add HDMI support to rk3288-veyron-analog-audio ARM: dts: rockchip: Add HDMI audio support to rk3288-veyron-mickey.dts .../bindings/sound/rockchip-max98090.txt | 38 +- .../boot/dts/rk3288-veyron-analog-audio.dtsi | 3 +- arch/arm/boot/dts/rk3288-veyron-mickey.dts | 7 + .../drm/bridge/synopsys/dw-hdmi-i2s-audio.c | 11 + drivers/gpu/drm/bridge/synopsys/dw-hdmi.c | 41 +- include/drm/bridge/dw_hdmi.h | 4 + sound/soc/rockchip/Kconfig | 3 +- sound/soc/rockchip/rockchip_max98090.c | 392 +++++++++++++++--- 8 files changed, 425 insertions(+), 74 deletions(-) -- 2.24.0.rc0.303.g954a862665-goog

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[alsa-devel] [PATCH 0/2] Update adau7118 dt-bindings
by Nuno Sá 29 Oct '19

29 Oct '19

This is a follow-up patch as requested by Rob. The dt-bindings patch for this device was applied before Rob's review. Here is the link to the review: https://lore.kernel.org/alsa-devel/20191010210850.GA20184@bogus/ Nuno Sá (2): dt-bindings: asoc: adau7118: Cleanup ASOC: adau7118: Change regulators id .../bindings/sound/adi,adau7118.yaml | 25 ++++++++----------- sound/soc/codecs/adau7118.c | 4 +-- 2 files changed, 12 insertions(+), 17 deletions(-) -- 2.23.0

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[alsa-devel] [PATCH] Revert "ALSA: hda: Flush interrupts on disabling"
by Takashi Iwai 28 Oct '19

28 Oct '19

This reverts commit caa8422d01e983782548648e125fd617cadcec3f. It turned out that this commit caused a regression at shutdown / reboot, as the synchronize_irq() calls seems blocking the whole shutdown. Also another part of the change about shuffling the call order looks suspicious; the azx_stop_chip() call disables the CORB / RIRB while the others may still need the CORB/RIRB update. Since the original commit itself was a cargo-fix, let's revert the whole patch. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=205333 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/hdac_controller.c | 2 -- sound/pci/hda/hda_intel.c | 2 +- 2 files changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index d3999e7b0705..7e7be8e4dcf9 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -447,8 +447,6 @@ static void azx_int_disable(struct hdac_bus *bus) list_for_each_entry(azx_dev, &bus->stream_list, list) snd_hdac_stream_updateb(azx_dev, SD_CTL, SD_INT_MASK, 0); - synchronize_irq(bus->irq); - /* disable SIE for all streams */ snd_hdac_chip_writeb(bus, INTCTL, 0); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index a815bc811799..cf53fbd872ee 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1348,9 +1348,9 @@ static int azx_free(struct azx *chip) } if (bus->chip_init) { - azx_stop_chip(chip); azx_clear_irq_pending(chip); azx_stop_all_streams(chip); + azx_stop_chip(chip); } if (bus->irq >= 0) -- 2.16.4

3 4

[alsa-devel] [PATCH v7 0/9] adapt SOF to use snd-hda-codec-hdmi
by Kai Vehmanen 28 Oct '19

28 Oct '19

Hey, here's the 7th 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. v7 changes: - Only change since v6 is moving inlined logic from hda_dsp_common.h to hda_dsp_common.c and related changes to machine driver make rules. This is to address Takashi's feedback. - Rebased on top of Mark's tree (agreed with Takashi and Pierre to submit via Mark's ASoC tree). 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 | 10 +- 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, 370 insertions(+), 32 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

3 15

[alsa-devel] [PATCH] ASoC: intel - fix the card names
by Jaroslav Kysela 28 Oct '19

28 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..5261f7bf0b3b 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 = "skl-hda", .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..fc4638b8e03b 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 = "sof-rt5682", .owner = THIS_MODULE, .controls = sof_controls, .num_controls = ARRAY_SIZE(sof_controls), -- 2.20.1

2 2

[alsa-devel] [PATCH] ALSA: usb-audio: Fix memory leak in __snd_usbmidi_create
by Navid Emamdoost 28 Oct '19

28 Oct '19

In the implementation of __snd_usbmidi_create() there is a memory leak caused by incorrect goto destination. Go to free_midi if snd_usbmidi_create_endpoints_midiman() or snd_usbmidi_create_endpoints() fail. Fixes: 731209cc0417 ("ALSA: usb-midi: Use common error handling code in __snd_usbmidi_create()") Signed-off-by: Navid Emamdoost <navid.emamdoost(a)gmail.com> --- sound/usb/midi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index b737f0ec77d0..22db37fbfbbd 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -2476,7 +2476,7 @@ int __snd_usbmidi_create(struct snd_card *card, else err = snd_usbmidi_create_endpoints(umidi, endpoints); if (err < 0) - goto exit; + goto free_midi; usb_autopm_get_interface_no_resume(umidi->iface); -- 2.17.1

3 5

[alsa-devel] [PATCH] ASoC: SOF: ipc: Fix memory leak in sof_set_get_large_ctrl_data
by Navid Emamdoost 28 Oct '19

28 Oct '19

In the implementation of sof_set_get_large_ctrl_data() there is a memory leak in case an error. Release partdata if sof_get_ctrl_copy_params() fails. Fixes: 54d198d5019d ("ASoC: SOF: Propagate sof_get_ctrl_copy_params() error properly") Signed-off-by: Navid Emamdoost <navid.emamdoost(a)gmail.com> --- sound/soc/sof/ipc.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/ipc.c b/sound/soc/sof/ipc.c index b2f359d2f7e5..086eeeab8679 100644 --- a/sound/soc/sof/ipc.c +++ b/sound/soc/sof/ipc.c @@ -572,8 +572,10 @@ static int sof_set_get_large_ctrl_data(struct snd_sof_dev *sdev, else err = sof_get_ctrl_copy_params(cdata->type, partdata, cdata, sparams); - if (err < 0) + if (err < 0) { + kfree(partdata); return err; + } msg_bytes = sparams->msg_bytes; pl_size = sparams->pl_size; -- 2.17.1

3 3

[alsa-devel] [PATCH] ASoC: SOF: Fix memory leak in sof_dfsentry_write
by Navid Emamdoost 28 Oct '19

28 Oct '19

In the implementation of sof_dfsentry_write() memory allocated for string is leaked in case of an error. Go to error handling path if the d_name.name is not valid. Fixes: 091c12e1f50c ("ASoC: SOF: debug: add new debugfs entries for IPC flood test") Signed-off-by: Navid Emamdoost <navid.emamdoost(a)gmail.com> --- sound/soc/sof/debug.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/debug.c b/sound/soc/sof/debug.c index 54cd431faab7..5529e8eeca46 100644 --- a/sound/soc/sof/debug.c +++ b/sound/soc/sof/debug.c @@ -152,8 +152,10 @@ static ssize_t sof_dfsentry_write(struct file *file, const char __user *buffer, */ dentry = file->f_path.dentry; if (strcmp(dentry->d_name.name, "ipc_flood_count") && - strcmp(dentry->d_name.name, "ipc_flood_duration_ms")) - return -EINVAL; + strcmp(dentry->d_name.name, "ipc_flood_duration_ms")) { + ret = -EINVAL; + goto out; + } if (!strcmp(dentry->d_name.name, "ipc_flood_duration_ms")) flood_duration_test = true; -- 2.17.1

3 2

[alsa-devel] [PATCH] ASoC: SOF: Intel: hda-stream: fix the CONFIG_ prefix missing
by Pierre-Louis Bossart 28 Oct '19

28 Oct '19

From: Keyon Jie <yang.jie(a)linux.intel.com> We are missing the 'CONFIG_' prefix when using the kernel configure item SND_SOC_SOF_HDA_ALWAYS_ENABLE_DMI_L1, here correct them. Fixes: 43b2ab9009b13b ('ASoC: SOF: Intel: hda: Disable DMI L1 entry during capture') Signed-off-by: Keyon Jie <yang.jie(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/intel/hda-stream.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index 450f9c55785f..29ab43281670 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -190,7 +190,7 @@ hda_dsp_stream_get(struct snd_sof_dev *sdev, int direction) * Workaround to address a known issue with host DMA that results * in xruns during pause/release in capture scenarios. */ - if (!IS_ENABLED(SND_SOC_SOF_HDA_ALWAYS_ENABLE_DMI_L1)) + if (!IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_ALWAYS_ENABLE_DMI_L1)) if (stream && direction == SNDRV_PCM_STREAM_CAPTURE) snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, HDA_VS_INTEL_EM2, @@ -228,7 +228,7 @@ int hda_dsp_stream_put(struct snd_sof_dev *sdev, int direction, int stream_tag) spin_unlock_irq(&bus->reg_lock); /* Enable DMI L1 entry if there are no capture streams open */ - if (!IS_ENABLED(SND_SOC_SOF_HDA_ALWAYS_ENABLE_DMI_L1)) + if (!IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_ALWAYS_ENABLE_DMI_L1)) if (!active_capture_stream) snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, HDA_VS_INTEL_EM2, -- 2.20.1

2 1

[alsa-devel] [PATCH] ASoC: ti: sdma-pcm: Add back the flags parameter for non standard dma names
by Peter Ujfalusi 28 Oct '19

28 Oct '19

When non standard names are used it is possible that one of the directions are not provided, thus the flags needs to be present to tell the core that we have half duplex setup. Fixes: 642aafea8889 ("ASoC: ti: remove compat dma probing") Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/ti/sdma-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/ti/sdma-pcm.c b/sound/soc/ti/sdma-pcm.c index a236350beb10..2b0bc234e1b6 100644 --- a/sound/soc/ti/sdma-pcm.c +++ b/sound/soc/ti/sdma-pcm.c @@ -62,7 +62,7 @@ int sdma_pcm_platform_register(struct device *dev, config->chan_names[0] = txdmachan; config->chan_names[1] = rxdmachan; - return devm_snd_dmaengine_pcm_register(dev, config, 0); + return devm_snd_dmaengine_pcm_register(dev, config, flags); } EXPORT_SYMBOL_GPL(sdma_pcm_platform_register); -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

2 1

[alsa-devel] [PATCH] ASoC: rockchip: rockchip_max98090: Enable SHDN to fix headset detection
by Cheng-Yi Chiang 28 Oct '19

28 Oct '19

max98090 spec states that chip needs to be in turned-on state to supply mic bias. Enable SHDN dapm widget along with MICBIAS widget to actually turn on mic bias for proper headset button detection. This is similar to cht_ti_jack_event in sound/soc/intel/boards/cht_bsw_max98090_ti.c. Note that due to ts3a227e reports the jack event right away before the notifier is registered, if headset is plugged on boot, headset button will not get detected until headset is unplugged and plugged. This is still an issue to be fixed. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- sound/soc/rockchip/rockchip_max98090.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/rockchip/rockchip_max98090.c b/sound/soc/rockchip/rockchip_max98090.c index 0097df1fae66..e80b09143b63 100644 --- a/sound/soc/rockchip/rockchip_max98090.c +++ b/sound/soc/rockchip/rockchip_max98090.c @@ -66,10 +66,13 @@ static int rk_jack_event(struct notifier_block *nb, unsigned long event, struct snd_soc_jack *jack = (struct snd_soc_jack *)data; struct snd_soc_dapm_context *dapm = &jack->card->dapm; - if (event & SND_JACK_MICROPHONE) + if (event & SND_JACK_MICROPHONE) { snd_soc_dapm_force_enable_pin(dapm, "MICBIAS"); - else + snd_soc_dapm_force_enable_pin(dapm, "SHDN"); + } else { snd_soc_dapm_disable_pin(dapm, "MICBIAS"); + snd_soc_dapm_disable_pin(dapm, "SHDN"); + } snd_soc_dapm_sync(dapm); -- 2.24.0.rc0.303.g954a862665-goog

2 1

[alsa-devel] [PATCH] ASoC: fsl: fsl_dma: fix build failure
by Michael Ellerman 28 Oct '19

28 Oct '19

Commit 4ac85de9977e ("ASoC: fsl: fsl_dma: remove snd_pcm_ops") removed fsl_dma_ops but left a usage, leading to a build error for some configs, eg. mpc85xx_defconfig: sound/soc/fsl/fsl_dma.c: In function ‘fsl_soc_dma_probe’: sound/soc/fsl/fsl_dma.c:905:18: error: ‘fsl_dma_ops’ undeclared (first use in this function) dma->dai.ops = &fsl_dma_ops; ^~~~~~~~~~~ Remove the usage to fix the build. Fixes: 4ac85de9977e ("ASoC: fsl: fsl_dma: remove snd_pcm_ops") Signed-off-by: Michael Ellerman <mpe(a)ellerman.id.au> --- sound/soc/fsl/fsl_dma.c | 1 - 1 file changed, 1 deletion(-) This breakage is only in linux-next. diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index a092726510d4..2868c4f97cb2 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -901,7 +901,6 @@ static int fsl_soc_dma_probe(struct platform_device *pdev) } dma->dai.name = DRV_NAME; - dma->dai.ops = &fsl_dma_ops; dma->dai.open = fsl_dma_open; dma->dai.close = fsl_dma_close; dma->dai.ioctl = snd_soc_pcm_lib_ioctl; -- 2.21.0

2 1

[alsa-devel] [PATCH -next] ASoC: sunxi: sun4i-codec: remove unneeded semicolon
by YueHaibing 28 Oct '19

28 Oct '19

remove unneeded semicolon. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/sunxi/sun4i-codec.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index ee448d5..34f3e0b 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -1442,7 +1442,7 @@ static struct snd_soc_card *sun8i_a23_codec_create_card(struct device *dev) if (!aux_dev.dlc.of_node) { dev_err(dev, "Can't find analog controls for codec.\n"); return ERR_PTR(-EINVAL); - }; + } card->dai_link = sun4i_codec_create_link(dev, &card->num_links); if (!card->dai_link) @@ -1480,7 +1480,7 @@ static struct snd_soc_card *sun8i_h3_codec_create_card(struct device *dev) if (!aux_dev.dlc.of_node) { dev_err(dev, "Can't find analog controls for codec.\n"); return ERR_PTR(-EINVAL); - }; + } card->dai_link = sun4i_codec_create_link(dev, &card->num_links); if (!card->dai_link) @@ -1518,7 +1518,7 @@ static struct snd_soc_card *sun8i_v3s_codec_create_card(struct device *dev) if (!aux_dev.dlc.of_node) { dev_err(dev, "Can't find analog controls for codec.\n"); return ERR_PTR(-EINVAL); - }; + } card->dai_link = sun4i_codec_create_link(dev, &card->num_links); if (!card->dai_link) -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: eve: implement set_bias_level function for rt5514
by Brent Lu 28 Oct '19

28 Oct '19

The first DMIC capture always fail (zero sequence data from PCM port) after using DSP hotwording function (i.e. Google assistant). This rt5514 codec requires to control mclk directly in the set_bias_level function. Implement this function in machine driver to control the ssp1_mclk clock explicitly could fix this issue. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 50 ++++++++++++++++++++++ 1 file changed, 50 insertions(+) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index dc09a85..b546de8 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -653,6 +653,55 @@ static struct snd_soc_dai_link kabylake_dais[] = { }, }; +static int kabylake_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) +{ + struct snd_soc_component *component = dapm->component; + struct kbl_codec_private *priv = snd_soc_card_get_drvdata(card); + int ret = 0; + + if (!component || strcmp(component->name, RT5514_DEV_NAME)) + return 0; + + if (IS_ERR(priv->mclk)) + return 0; + + /* + * It's required to control mclk directly in the set_bias_level + * function for rt5514 codec or the recording function could + * break. + */ + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (dapm->bias_level == SND_SOC_BIAS_ON) { + dev_dbg(card->dev, "Disable mclk"); + clk_disable_unprepare(priv->mclk); + } else { + dev_dbg(card->dev, "Enable mclk"); + ret = clk_set_rate(priv->mclk, 24000000); + if (ret) { + dev_err(card->dev, "Can't set rate for mclk, err: %d\n", + ret); + return ret; + } + + ret = clk_prepare_enable(priv->mclk); + if (ret) { + dev_err(card->dev, "Can't enable mclk, err: %d\n", + ret); + + /* mclk is already enabled in FW */ + ret = 0; + } + } + break; + default: + break; + } + + return ret; +} + static int kabylake_card_late_probe(struct snd_soc_card *card) { struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(card); @@ -692,6 +741,7 @@ static struct snd_soc_card kabylake_audio_card = { .owner = THIS_MODULE, .dai_link = kabylake_dais, .num_links = ARRAY_SIZE(kabylake_dais), + .set_bias_level = kabylake_set_bias_level, .controls = kabylake_controls, .num_controls = ARRAY_SIZE(kabylake_controls), .dapm_widgets = kabylake_widgets, -- 2.7.4

4 6

[alsa-devel] [PATCH V3] ASoC: fsl_esai: Add spin lock to protect reset, stop and start
by Shengjiu Wang 28 Oct '19

28 Oct '19

xrun may happen at the end of stream, the trigger->fsl_esai_trigger_stop maybe called in the middle of fsl_esai_hw_reset, this may cause esai in wrong state after stop, and there may be endless xrun interrupt. This issue may also happen with trigger->fsl_esai_trigger_start. So Add spin lock to lock those functions. Fixes: 7ccafa2b3879 ("ASoC: fsl_esai: recover the channel swap after xrun") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- Change in v3 -refine comments Change in v2 -add lock for fsl_esai_trigger_start. sound/soc/fsl/fsl_esai.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 37b14c48b537..453392a9cd68 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -33,6 +33,7 @@ * @fsysclk: system clock source to derive HCK, SCK and FS * @spbaclk: SPBA clock (optional, depending on SoC design) * @task: tasklet to handle the reset operation + * @lock: spin lock between hw_reset() and trigger() * @fifo_depth: depth of tx/rx FIFO * @slot_width: width of each DAI slot * @slots: number of slots @@ -56,6 +57,7 @@ struct fsl_esai { struct clk *fsysclk; struct clk *spbaclk; struct tasklet_struct task; + spinlock_t lock; /* Protect hw_reset and trigger */ u32 fifo_depth; u32 slot_width; u32 slots; @@ -676,8 +678,10 @@ static void fsl_esai_hw_reset(unsigned long arg) { struct fsl_esai *esai_priv = (struct fsl_esai *)arg; bool tx = true, rx = false, enabled[2]; + unsigned long lock_flags; u32 tfcr, rfcr; + spin_lock_irqsave(&esai_priv->lock, lock_flags); /* Save the registers */ regmap_read(esai_priv->regmap, REG_ESAI_TFCR, &tfcr); regmap_read(esai_priv->regmap, REG_ESAI_RFCR, &rfcr); @@ -715,6 +719,8 @@ static void fsl_esai_hw_reset(unsigned long arg) fsl_esai_trigger_start(esai_priv, tx); if (enabled[rx]) fsl_esai_trigger_start(esai_priv, rx); + + spin_unlock_irqrestore(&esai_priv->lock, lock_flags); } static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, @@ -722,6 +728,7 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, { struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + unsigned long lock_flags; esai_priv->channels[tx] = substream->runtime->channels; @@ -729,12 +736,16 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + spin_lock_irqsave(&esai_priv->lock, lock_flags); fsl_esai_trigger_start(esai_priv, tx); + spin_unlock_irqrestore(&esai_priv->lock, lock_flags); break; case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + spin_lock_irqsave(&esai_priv->lock, lock_flags); fsl_esai_trigger_stop(esai_priv, tx); + spin_unlock_irqrestore(&esai_priv->lock, lock_flags); break; default: return -EINVAL; @@ -1002,6 +1013,7 @@ static int fsl_esai_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, esai_priv); + spin_lock_init(&esai_priv->lock); ret = fsl_esai_hw_init(esai_priv); if (ret) return ret; -- 2.21.0

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[alsa-devel] [PATCH V3] ASoC: fsl_asrc: refine the setting of internal clock divider
by Shengjiu Wang 28 Oct '19

28 Oct '19

The output divider should align with the output sample rate, if use ideal sample rate, there will be a lot of overload, which would cause underrun. The maximum divider of asrc clock is 1024, but there is no judgement for this limitation in driver, which may cause the divider setting not correct. For non-ideal ratio mode, the clock rate should divide the sample rate with no remainder, and the quotient should be less than 1024. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- Change in v3 - refine comments Change in v2 - remove p2p/m2m word - use use_ideal_rate sound/soc/fsl/fsl_asrc.c | 45 ++++++++++++++++++++++++++++++---------- 1 file changed, 34 insertions(+), 11 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index 0bf91a6f54b9..a3cfceea7d2f 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -259,8 +259,15 @@ static int fsl_asrc_set_ideal_ratio(struct fsl_asrc_pair *pair, * It configures those ASRC registers according to a configuration instance * of struct asrc_config which includes in/output sample rate, width, channel * and clock settings. + * + * Note: + * The ideal ratio configuration can work with a flexible clock rate setting. + * Using IDEAL_RATIO_RATE gives a faster converting speed but overloads ASRC. + * For a regular audio playback, the clock rate should not be slower than an + * clock rate aligning with the output sample rate; For a use case requiring + * faster conversion, set use_ideal_rate to have the faster speed. */ -static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) +static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair, bool use_ideal_rate) { struct asrc_config *config = pair->config; struct fsl_asrc *asrc_priv = pair->asrc_priv; @@ -268,7 +275,8 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) enum asrc_word_width input_word_width; enum asrc_word_width output_word_width; u32 inrate, outrate, indiv, outdiv; - u32 clk_index[2], div[2]; + u32 clk_index[2], div[2], rem[2]; + u64 clk_rate; int in, out, channels; int pre_proc, post_proc; struct clk *clk; @@ -351,27 +359,42 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) /* We only have output clock for ideal ratio mode */ clk = asrc_priv->asrck_clk[clk_index[ideal ? OUT : IN]]; - div[IN] = clk_get_rate(clk) / inrate; - if (div[IN] == 0) { + clk_rate = clk_get_rate(clk); + rem[IN] = do_div(clk_rate, inrate); + div[IN] = (u32)clk_rate; + + /* + * The divider range is [1, 1024], defined by the hardware. For non- + * ideal ratio configuration, clock rate has to be strictly aligned + * with the sample rate. For ideal ratio configuration, clock rates + * only result in different converting speeds. So remainder does not + * matter, as long as we keep the divider within its valid range. + */ + if (div[IN] == 0 || (!ideal && (div[IN] > 1024 || rem[IN] != 0))) { pair_err("failed to support input sample rate %dHz by asrck_%x\n", inrate, clk_index[ideal ? OUT : IN]); return -EINVAL; } + div[IN] = min_t(u32, 1024, div[IN]); + clk = asrc_priv->asrck_clk[clk_index[OUT]]; - - /* Use fixed output rate for Ideal Ratio mode (INCLK_NONE) */ - if (ideal) - div[OUT] = clk_get_rate(clk) / IDEAL_RATIO_RATE; + clk_rate = clk_get_rate(clk); + if (ideal && use_ideal_rate) + rem[OUT] = do_div(clk_rate, IDEAL_RATIO_RATE); else - div[OUT] = clk_get_rate(clk) / outrate; + rem[OUT] = do_div(clk_rate, outrate); + div[OUT] = clk_rate; - if (div[OUT] == 0) { + /* Output divider has the same limitation as the input one */ + if (div[OUT] == 0 || (!ideal && (div[OUT] > 1024 || rem[OUT] != 0))) { pair_err("failed to support output sample rate %dHz by asrck_%x\n", outrate, clk_index[OUT]); return -EINVAL; } + div[OUT] = min_t(u32, 1024, div[OUT]); + /* Set the channel number */ channels = config->channel_num; @@ -560,7 +583,7 @@ static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, config.output_sample_rate = rate; } - ret = fsl_asrc_config_pair(pair); + ret = fsl_asrc_config_pair(pair, false); if (ret) { dev_err(dai->dev, "fail to config asrc pair\n"); return ret; -- 2.21.0

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