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

[alsa-devel] [PATCH] ALSA: pcm: Fix memory leak at closing a stream without hw_free
by Takashi Iwai 13 Feb '20

13 Feb '20

ALSA PCM core recently introduced a new managed PCM buffer allocation mode that does allocate / free automatically at hw_params and hw_free. However, it overlooked the code path directly calling hw_free PCM ops at releasing the PCM substream, and it may result in a memory leak as spotted by syzkaller when no buffer preallocation is used (e.g. vmalloc buffer). This patch papers over it with a slight refactoring. The hw_free ops call and relevant tasks are unified in a new helper function, and call it from both places. Fixes: 0dba808eae26 ("ALSA: pcm: Introduce managed buffer allocation mode") Reported-by: syzbot+30edd0f34bfcdc548ac4(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_native.c | 24 +++++++++++++++--------- 1 file changed, 15 insertions(+), 9 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index bb23f5066654..4ac42ee1238c 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -786,10 +786,22 @@ static int snd_pcm_hw_params_user(struct snd_pcm_substream *substream, return err; } +static int do_hw_free(struct snd_pcm_substream *substream) +{ + int result = 0; + + snd_pcm_sync_stop(substream); + if (substream->ops->hw_free) + result = substream->ops->hw_free(substream); + if (substream->managed_buffer_alloc) + snd_pcm_lib_free_pages(substream); + return result; +} + static int snd_pcm_hw_free(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime; - int result = 0; + int result; if (PCM_RUNTIME_CHECK(substream)) return -ENXIO; @@ -806,11 +818,7 @@ static int snd_pcm_hw_free(struct snd_pcm_substream *substream) snd_pcm_stream_unlock_irq(substream); if (atomic_read(&substream->mmap_count)) return -EBADFD; - snd_pcm_sync_stop(substream); - if (substream->ops->hw_free) - result = substream->ops->hw_free(substream); - if (substream->managed_buffer_alloc) - snd_pcm_lib_free_pages(substream); + result = do_hw_free(substream); snd_pcm_set_state(substream, SNDRV_PCM_STATE_OPEN); pm_qos_remove_request(&substream->latency_pm_qos_req); return result; @@ -2529,9 +2537,7 @@ void snd_pcm_release_substream(struct snd_pcm_substream *substream) snd_pcm_drop(substream); if (substream->hw_opened) { - if (substream->ops->hw_free && - substream->runtime->status->state != SNDRV_PCM_STATE_OPEN) - substream->ops->hw_free(substream); + do_hw_free(substream); substream->ops->close(substream); substream->hw_opened = 0; } -- 2.16.4

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[alsa-devel] [PATCH v2 0/5] soundwire: intel: add DAI callbacks
by Pierre-Louis Bossart 13 Feb '20

13 Feb '20

The existing mainline code is missing most of the DAI callbacks needed for a functional implementation, and the existing ones need to be modified to provide the relevant information to SOF drivers. As suggested by Vinod, these patches are shared first - with the risk that they are separated from the actual DAI enablement, so reviewers might wonder why they are needed in the first place. For reference, the complete set of 90+ patches required for SoundWire on Intel platforms is available here: https://github.com/thesofproject/linux/pull/1692 changes since v1: Fix string allocation (only feedback from Vinod) Pierre-Louis Bossart (2): soundwire: intel: rename res field as link_res soundwire: intel: free all resources on hw_free() Rander Wang (3): soundwire: intel: add prepare support in sdw dai driver soundwire: intel: add trigger support in sdw dai driver soundwire: intel: add sdw_stream_setup helper for .startup callback drivers/soundwire/intel.c | 197 ++++++++++++++++++++++++++++++++++---- 1 file changed, 177 insertions(+), 20 deletions(-) -- 2.20.1

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[alsa-devel] [PATCH v2 0/5] soundwire: stream: fix state machines and transitions
by Pierre-Louis Bossart 13 Feb '20

13 Feb '20

The existing stream support works fine with simple cases, but does not map well with ALSA transitions for underflows/resume where prepare() can be called multiple times. Concurrency with multiple devices per links or multiple streams enabled on the same link also needs to be fixed. These patches are the result of hours of validation on the Intel side and should benefit other implementations since there is nothing hardware-specific. The Intel-specific changes being reviewed do depend on those stream changes though to be functional. Changes since v1: Removed spurious code block change flagged by Vinod No change (replies provided in v1 thread) Github link issue is public, no reason to remove it Bandwidth computation on ALSA prepare/start (for resume cases) handled internally in stream layer. Kept emacs comment formatting. No additional code/test for concurrent streams (not supported due to locking) Bard Liao (1): soundwire: stream: only prepare stream when it is configured. Pierre-Louis Bossart (2): soundwire: stream: update state machine and add state checks soundwire: stream: do not update parameters during DISABLED-PREPARED transition Rander Wang (2): soundwire: stream: fix support for multiple Slaves on the same link soundwire: stream: don't program ports when a stream that has not been prepared Documentation/driver-api/soundwire/stream.rst | 61 +++++++++---- drivers/soundwire/stream.c | 90 ++++++++++++++++--- 2 files changed, 124 insertions(+), 27 deletions(-) -- 2.20.1

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[alsa-devel] USB Type-C monitor flashes once when play a video file after unplug and re-plug the monitor
by AJ_Cheng＠compal.com 13 Feb '20

13 Feb '20

Hi Sirs, Here is chromebook SW team from Compal. As the mail title, we hit issue that the external monitor will flash once when play video after hot pluging. We can reproduce not only on chromebook but also ubuntu 16.04. There has higher failure rate with Dell Solomon dock and Dell S2718D monitor. We found adding the delay in "sound/pci/hda/patch_hdmi.c " can fix this issue.(as the attachment) May need your help to review and advice. Thanks. Here is the issue number in gitlab for more detail. https://gitlab.freedesktop.org/drm/intel/issues/318 AJ Cheng NID/NID1 e-mail: AJ_Cheng(a)compal.com<mailto:AJ_Cheng@compal.com> Tel: +886-2-8797-8599 ext. 17561 Mobile : +886-932827829 COMPAL Electronics, Inc.

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[alsa-devel] [PATCH] dt-bindings: Convert the binding file google, cros-ec-codec.txt to yaml format.
by Dafna Hirschfeld 13 Feb '20

13 Feb '20

This was tested and verified with: make dt_binding_check DT_SCHEMA_FILES=Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml Signed-off-by: Dafna Hirschfeld <dafna.hirschfeld(a)collabora.com> --- .../bindings/sound/google,cros-ec-codec.txt | 44 ------------- .../bindings/sound/google,cros-ec-codec.yaml | 62 +++++++++++++++++++ 2 files changed, 62 insertions(+), 44 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/google,cros-ec-codec.txt create mode 100644 Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml diff --git a/Documentation/devicetree/bindings/sound/google,cros-ec-codec.txt b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.txt deleted file mode 100644 index 8ca52dcc5572..000000000000 --- a/Documentation/devicetree/bindings/sound/google,cros-ec-codec.txt +++ /dev/null @@ -1,44 +0,0 @@ -Audio codec controlled by ChromeOS EC - -Google's ChromeOS EC codec is a digital mic codec provided by the -Embedded Controller (EC) and is controlled via a host-command interface. - -An EC codec node should only be found as a sub-node of the EC node (see -Documentation/devicetree/bindings/mfd/cros-ec.txt). - -Required properties: -- compatible: Must contain "google,cros-ec-codec" -- #sound-dai-cells: Should be 1. The cell specifies number of DAIs. - -Optional properties: -- reg: Pysical base address and length of shared memory region from EC. - It contains 3 unsigned 32-bit integer. The first 2 integers - combine to become an unsigned 64-bit physical address. The last - one integer is length of the shared memory. -- memory-region: Shared memory region to EC. A "shared-dma-pool". See - ../reserved-memory/reserved-memory.txt for details. - -Example: - -{ - ... - - reserved_mem: reserved_mem { - compatible = "shared-dma-pool"; - reg = <0 0x52800000 0 0x100000>; - no-map; - }; -} - -cros-ec@0 { - compatible = "google,cros-ec-spi"; - - ... - - cros_ec_codec: ec-codec { - compatible = "google,cros-ec-codec"; - #sound-dai-cells = <1>; - reg = <0x0 0x10500000 0x80000>; - memory-region = <&reserved_mem>; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml new file mode 100644 index 000000000000..94a85d0cbf43 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml @@ -0,0 +1,62 @@ +# SPDX-License-Identifier: GPL-2.0-only +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/google,cros-ec-codec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Audio codec controlled by ChromeOS EC + +maintainers: + - Cheng-Yi Chiang <cychiang(a)chromium.org> + +description: | + Google's ChromeOS EC codec is a digital mic codec provided by the + Embedded Controller (EC) and is controlled via a host-command interface. + An EC codec node should only be found as a sub-node of the EC node (see + Documentation/devicetree/bindings/mfd/cros-ec.txt). + +properties: + compatible: + const: google,cros-ec-codec + + "#sound-dai-cells": + const: 1 + + reg: + items: + - description: | + Physical base address and length of shared memory region from EC. + It contains 3 unsigned 32-bit integer. The first 2 integers + combine to become an unsigned 64-bit physical address. + The last one integer is the length of the shared memory. + + memory-region: + $ref: '/schemas/types.yaml#/definitions/phandle' + description: | + Shared memory region to EC. A "shared-dma-pool". + See ../reserved-memory/reserved-memory.txt for details. + +required: + - compatible + - '#sound-dai-cells' + +additionalProperties: false + +examples: + - | + reserved_mem: reserved_mem { + compatible = "shared-dma-pool"; + reg = <0 0x52800000 0 0x100000>; + no-map; + }; + cros-ec@0 { + compatible = "google,cros-ec-spi"; + #address-cells = <2>; + #size-cells = <1>; + cros_ec_codec: ec-codec { + compatible = "google,cros-ec-codec"; + #sound-dai-cells = <1>; + reg = <0x0 0x10500000 0x80000>; + memory-region = <&reserved_mem>; + }; + }; -- 2.17.1

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[alsa-devel] [PATCH v2 0/9] add ASoC components for AHUB
by Sameer Pujar 11 Feb '20

11 Feb '20

Overview ======== The Audio Hub (AHUB) is part of the Audio Processing Engine (APE) which comprises a collection of hardware accelerators for audio pre-processing and post-processing. It also includes a programmable full crossbar for routing audio data across these accelerators. This series exposes some of these below mentioned HW devices as ASoC components for Tegra platforms from Tegra210 onwards. * ADMAIF : The interface between ADMA and AHUB * XBAR : Crossbar for routing audio samples across various modules * I2S : Inter-IC Sound Controller * DMIC : Digital Microphone * DSPK : Digital Speaker Following is the summary of current series. 1. Add YAML DT binding documentation for above mentioned modules. 2. Helper function for ACIF programming is exposed for Tegra210 and later. 3. Add ASoC driver components for each of the above modules. 4. Add DT entries for above components for Tegra210, Tegra186 and Tegra194. 5. Enable these components for Jetson-Tx1, Jetson-Tx2 and Jetson-Xavier. Machine driver series will be sent separately. Changelog ========= v1 -> v2 -------- * [1/9] "dt-bindings: sound: tegra: add DT binding for AHUB" - no changes * [2/9] "ASoC: tegra: add support for CIF programming" - removed CIF programming changes for legacy chips. - this patch now exposes helper function for CIF programming, which can be used on Tegra210 later. - later tegra_cif.c can be extended for legacy chips as well. - updated commit message accordingly * [3/9] "ASoC: tegra: add Tegra210 based DMIC driver" - removed unnecessary initialization of 'ret' in probe() * [4/9] "ASoC: tegra: add Tegra210 based I2S driver" - removed unnecessary initialization of 'ret' in probe() - fixed indentation - added consistent bracing for if-else clauses - updated 'rx_fifo_th' type to 'unsigned int' - used BIT() macro for defines like '1 << {x}' in tegra210_i2s.h * [5/9] "ASoC: tegra: add Tegra210 based AHUB driver" - used of_device_get_match_data() to get 'soc_data' and removed explicit of_match_device() - used devm_platform_ioremap_resource() and removed explicit platform_get_resource() - fixed indentation for devm_snd_soc_register_component() - updated commit message - updated commit message to reflect compatible binding for Tegra186 and Tegra194. * [6/9] "ASoC: tegra: add Tegra186 based DSPK driver" - removed unnecessary initialization of 'ret' in probe() - updated 'max_th' to 'unsigned int' - shortened lengthy macro names to avoid wrapping in tegra186_dspk_wr_reg() and to be consistent * [7/9] "ASoC: tegra: add Tegra210 based ADMAIF driver" - used of_device_get_match_data() and removed explicit of_match_device() - used BIT() macro for defines like '1 << {x}' in tegra210_admaif.h - updated commit message to reflect compatible binding for Tegra186 and Tegra194. * [8/9] "arm64: tegra: add AHUB components for few Tegra chips" - no change * [9/9] "arm64: tegra: enable AHUB modules for few Tegra chips" - no change * common changes for patch [3/9] to [7/9] - sorted headers in alphabetical order - moved MODULE_DEVICE_TABLE() right below *_of_match table - removed macro DRV_NAME - removed explicit 'owner' field from platform_driver structure - added 'const' to snd_soc_dai_ops structure ================= Sameer Pujar (9): dt-bindings: sound: tegra: add DT binding for AHUB ASoC: tegra: add support for CIF programming ASoC: tegra: add Tegra210 based DMIC driver ASoC: tegra: add Tegra210 based I2S driver ASoC: tegra: add Tegra210 based AHUB driver ASoC: tegra: add Tegra186 based DSPK driver ASoC: tegra: add Tegra210 based ADMAIF driver arm64: tegra: add AHUB components for few Tegra chips arm64: tegra: enable AHUB modules for few Tegra chips .../bindings/sound/nvidia,tegra186-dspk.yaml | 105 +++ .../bindings/sound/nvidia,tegra210-admaif.yaml | 165 ++++ .../bindings/sound/nvidia,tegra210-ahub.yaml | 130 +++ .../bindings/sound/nvidia,tegra210-dmic.yaml | 105 +++ .../bindings/sound/nvidia,tegra210-i2s.yaml | 112 +++ arch/arm64/boot/dts/nvidia/tegra186-p2771-0000.dts | 48 ++ arch/arm64/boot/dts/nvidia/tegra186.dtsi | 231 ++++- arch/arm64/boot/dts/nvidia/tegra194-p2972-0000.dts | 36 + arch/arm64/boot/dts/nvidia/tegra194.dtsi | 239 +++++- arch/arm64/boot/dts/nvidia/tegra210-p2371-2180.dts | 40 + arch/arm64/boot/dts/nvidia/tegra210.dtsi | 145 ++++ sound/soc/tegra/Kconfig | 56 ++ sound/soc/tegra/Makefile | 12 + sound/soc/tegra/tegra186_dspk.c | 512 +++++++++++ sound/soc/tegra/tegra186_dspk.h | 73 ++ sound/soc/tegra/tegra210_admaif.c | 886 +++++++++++++++++++ sound/soc/tegra/tegra210_admaif.h | 164 ++++ sound/soc/tegra/tegra210_ahub.c | 653 ++++++++++++++ sound/soc/tegra/tegra210_ahub.h | 125 +++ sound/soc/tegra/tegra210_dmic.c | 517 ++++++++++++ sound/soc/tegra/tegra210_dmic.h | 85 ++ sound/soc/tegra/tegra210_i2s.c | 939 +++++++++++++++++++++ sound/soc/tegra/tegra210_i2s.h | 132 +++ sound/soc/tegra/tegra_cif.c | 34 + sound/soc/tegra/tegra_cif.h | 47 ++ sound/soc/tegra/tegra_pcm.c | 222 ++++- sound/soc/tegra/tegra_pcm.h | 23 +- 27 files changed, 5832 insertions(+), 4 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra186-dspk.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-admaif.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-ahub.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-dmic.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-i2s.yaml create mode 100644 sound/soc/tegra/tegra186_dspk.c create mode 100644 sound/soc/tegra/tegra186_dspk.h create mode 100644 sound/soc/tegra/tegra210_admaif.c create mode 100644 sound/soc/tegra/tegra210_admaif.h create mode 100644 sound/soc/tegra/tegra210_ahub.c create mode 100644 sound/soc/tegra/tegra210_ahub.h create mode 100644 sound/soc/tegra/tegra210_dmic.c create mode 100644 sound/soc/tegra/tegra210_dmic.h create mode 100644 sound/soc/tegra/tegra210_i2s.c create mode 100644 sound/soc/tegra/tegra210_i2s.h create mode 100644 sound/soc/tegra/tegra_cif.c create mode 100644 sound/soc/tegra/tegra_cif.h -- 2.7.4

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[alsa-devel] [PATCH] ASoC: dt-bindings: stm32: convert sai to json-schema
by Olivier Moysan 11 Feb '20

11 Feb '20

Convert the STM32 SAI bindings to DT schema format using json-schema. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- .../bindings/sound/st,stm32-sai.txt | 107 ---------- .../bindings/sound/st,stm32-sai.yaml | 193 ++++++++++++++++++ 2 files changed, 193 insertions(+), 107 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/st,stm32-sai.txt create mode 100644 Documentation/devicetree/bindings/sound/st,stm32-sai.yaml diff --git a/Documentation/devicetree/bindings/sound/st,stm32-sai.txt b/Documentation/devicetree/bindings/sound/st,stm32-sai.txt deleted file mode 100644 index 944743dd9212..000000000000 --- a/Documentation/devicetree/bindings/sound/st,stm32-sai.txt +++ /dev/null @@ -1,107 +0,0 @@ -STMicroelectronics STM32 Serial Audio Interface (SAI). - -The SAI interface (Serial Audio Interface) offers a wide set of audio protocols -as I2S standards, LSB or MSB-justified, PCM/DSP, TDM, and AC'97. -The SAI contains two independent audio sub-blocks. Each sub-block has -its own clock generator and I/O lines controller. - -Required properties: - - compatible: Should be "st,stm32f4-sai" or "st,stm32h7-sai" - - reg: Base address and size of SAI common register set. - - clocks: Must contain phandle and clock specifier pairs for each entry - in clock-names. - - clock-names: Must contain "pclk" "x8k" and "x11k" - "pclk": Clock which feeds the peripheral bus interface. - Mandatory for "st,stm32h7-sai" compatible. - Not used for "st,stm32f4-sai" compatible. - "x8k": SAI parent clock for sampling rates multiple of 8kHz. - "x11k": SAI parent clock for sampling rates multiple of 11.025kHz. - - interrupts: cpu DAI interrupt line shared by SAI sub-blocks - -Optional properties: - - resets: Reference to a reset controller asserting the SAI - -SAI subnodes: -Two subnodes corresponding to SAI sub-block instances A et B can be defined. -Subnode can be omitted for unsused sub-block. - -SAI subnodes required properties: - - compatible: Should be "st,stm32-sai-sub-a" or "st,stm32-sai-sub-b" - for SAI sub-block A or B respectively. - - reg: Base address and size of SAI sub-block register set. - - clocks: Must contain one phandle and clock specifier pair - for sai_ck which feeds the internal clock generator. - If the SAI shares a master clock, with another SAI set as MCLK - clock provider, SAI provider phandle must be specified here. - - clock-names: Must contain "sai_ck". - Must also contain "MCLK", if SAI shares a master clock, - with a SAI set as MCLK clock provider. - - dmas: see Documentation/devicetree/bindings/dma/stm32-dma.txt - - dma-names: identifier string for each DMA request line - "tx": if sai sub-block is configured as playback DAI - "rx": if sai sub-block is configured as capture DAI - - pinctrl-names: should contain only value "default" - - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/st,stm32-pinctrl.yaml - -SAI subnodes Optional properties: - - st,sync: specify synchronization mode. - By default SAI sub-block is in asynchronous mode. - This property sets SAI sub-block as slave of another SAI sub-block. - Must contain the phandle and index of the sai sub-block providing - the synchronization. - - st,iec60958: support S/PDIF IEC6958 protocol for playback - IEC60958 protocol is not available for capture. - By default, custom protocol is assumed, meaning that protocol is - configured according to protocol defined in related DAI link node, - such as i2s, left justified, right justified, dsp and pdm protocols. - Note: ac97 protocol is not supported by SAI driver - - #clock-cells: should be 0. This property must be present if the SAI device - is a master clock provider, according to clocks bindings, described in - Documentation/devicetree/bindings/clock/clock-bindings.txt. - -The device node should contain one 'port' child node with one child 'endpoint' -node, according to the bindings defined in Documentation/devicetree/bindings/ -graph.txt. - -Example: -sound_card { - compatible = "audio-graph-card"; - dais = <&sai1b_port>; -}; - -sai1: sai1@40015800 { - compatible = "st,stm32h7-sai"; - #address-cells = <1>; - #size-cells = <1>; - ranges = <0 0x40015800 0x400>; - reg = <0x40015800 0x4>; - clocks = <&rcc SAI1_CK>, <&rcc PLL1_Q>, <&rcc PLL2_P>; - clock-names = "pclk", "x8k", "x11k"; - interrupts = <87>; - - sai1a: audio-controller@40015804 { - compatible = "st,stm32-sai-sub-a"; - reg = <0x4 0x1C>; - clocks = <&rcc SAI1_CK>; - clock-names = "sai_ck"; - dmas = <&dmamux1 1 87 0x400 0x0>; - dma-names = "tx"; - pinctrl-names = "default"; - pinctrl-0 = <&pinctrl_sai1a>; - - sai1b_port: port { - cpu_endpoint: endpoint { - remote-endpoint = <&codec_endpoint>; - format = "i2s"; - }; - }; - }; -}; - -audio-codec { - codec_port: port { - codec_endpoint: endpoint { - remote-endpoint = <&cpu_endpoint>; - }; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml b/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml new file mode 100644 index 000000000000..33dca007fc86 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml @@ -0,0 +1,193 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/st,stm32-sai.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: STMicroelectronics STM32 Serial Audio Interface (SAI) + +maintainers: + - Olivier Moysan <olivier.moysan(a)st.com> + +description: + The SAI interface (Serial Audio Interface) offers a wide set of audio + protocols as I2S standards, LSB or MSB-justified, PCM/DSP, TDM, and AC'97. + The SAI contains two independent audio sub-blocks. Each sub-block has + its own clock generator and I/O lines controller. + +properties: + compatible: + enum: + - st,stm32f4-sai + - st,stm32h7-sai + + reg: + items: + - description: Base address and size of SAI common register set. + - description: Base address and size of SAI identification register set. + minItems: 1 + maxItems: 2 + + ranges: + maxItems: 1 + + interrupts: + maxItems: 1 + + resets: + maxItems: 1 + + "#address-cells": + const: 1 + + "#size-cells": + const: 1 + + clocks: + items: + - description: pclk feeds the peripheral bus interface. + - description: x8k, SAI parent clock for sampling rates multiple of 8kHz. + - description: x11k, SAI parent clock for sampling rates multiple of 11.025kHz. + + clock-names: + items: + enum: [ pclk, x8k, x11k ] + minItems: 3 + maxItems: 3 + +required: + - compatible + - reg + - ranges + - "#address-cells" + - "#size-cells" + - clocks + - clock-names + +patternProperties: + "^audio-controller@[0-9a-f]+$": + type: object + description: + Two subnodes corresponding to SAI sub-block instances A et B + can be defined. Subnode can be omitted for unsused sub-block. + + properties: + compatible: + description: Compatible for SAI sub-block A or B. + enum: + - st,stm32-sai-sub-a + - st,stm32-sai-sub-b + + "#sound-dai-cells": + const: 0 + + reg: + maxItems: 1 + + clocks: + items: + - description: sai_ck clock feeding the internal clock generator. + - description: MCLK clock from a SAI set as master clock provider. + minItems: 1 + maxItems: 2 + + clock-names: + items: + - const: sai_ck + - const: MCLK + minItems: 1 + maxItems: 2 + + dmas: + items: + - description: SAI sub-block is configured as a capture DAI. + - description: SAI sub-block is configured as a playback DAI. + minItems: 1 + maxItems: 1 + + dma-names: + items: + - enum: [ rx, tx ] + + st,sync: + description: + Configure the SAI sub-block as slave of another SAI sub-block. + By default SAI sub-block is in asynchronous mode. + Must contain the phandle and index of the SAI sub-block providing + the synchronization. + allOf: + - $ref: /schemas/types.yaml#definitions/phandle-array + - maxItems: 1 + + st,iec60958: + description: + If set, support S/PDIF IEC6958 protocol for playback. + IEC60958 protocol is not available for capture. + By default, custom protocol is assumed, meaning that protocol is + configured according to protocol defined in related DAI link node, + such as i2s, left justified, right justified, dsp and pdm protocols. + allOf: + - $ref: /schemas/types.yaml#definitions/flag + + "#clock-cells": + description: Configure the SAI device as master clock provider. + const: 0 + + required: + - compatible + - "#sound-dai-cells" + - reg + - clocks + - clock-names + - dmas + - dma-names + +allOf: + - if: + properties: + compatible: + contains: + const: st,stm32f4-sai + + - then: + properties: + clocks: + minItems: 2 + maxItems: 2 + + clock-names: + items: + enum: [ x8k, x11k ] + minItems: 2 + maxItems: 2 + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/stm32mp1-clks.h> + #include <dt-bindings/reset/stm32mp1-resets.h> + sai1: sai@4400a000 { + compatible = "st,stm32h7-sai"; + #address-cells = <1>; + #size-cells = <1>; + ranges = <0 0x4400a000 0x400>; + reg = <0x4400a000 0x4>, <0x4400a3f0 0x10>; + interrupts = <GIC_SPI 87 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&rcc SAI1>, <&rcc PLL1_Q>, <&rcc PLL2_P>; + clock-names = "pclk", "x8k", "x11k"; + resets = <&rcc SAI1_R>; + + sai1a: audio-controller@4400a004 { + compatible = "st,stm32-sai-sub-a"; + #sound-dai-cells = <0>; + reg = <0x4 0x1c>; + clocks = <&rcc SAI1_K>; + clock-names = "sai_ck"; + dmas = <&dmamux1 87 0x400 0x01>; + dma-names = "tx"; + }; + }; + +... -- 2.17.1

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[alsa-devel] [PATCH alsa-ucm-conf 0/7] DB410c/msm8916-wcd UCM fixes/additions
by Stephan Gerhold 11 Feb '20

11 Feb '20

This is a collection of fixes/refactoring/additions for the DB410c UCM files. My goal is to also make them usable for other MSM8916 devices, e.g. the three MSM8916 smartphones which can run mainline Linux since https://lore.kernel.org/linux-arm-msm/20190722092211.100586-1-stephan@gerho… For now this is just preparation - this patch series refactors the DB410c UCM configuration into re-usable device fragments that can be included when needed. By the way: it seems like the UCM2 conversion (specifically, commit 14facf88b56 ("DB410c: adapt to ucm2") breaks PulseAudio because the Playback/CaptureChannels were moved from all the "SectionDevice"s to "SectionVerb": I: [pulseaudio] alsa-ucm.c: UCM available for card msm8916 I: [pulseaudio] alsa-ucm.c: Set UCM verb to HiFi W: [pulseaudio] alsa-ucm.c: UCM file does not specify 'PlaybackChannels' or 'CaptureChannels'for device DigitalMic, assuming stereo duplex. W: [pulseaudio] alsa-ucm.c: UCM file does not specify 'PlaybackChannels' or 'CaptureChannels'for device SecondaryMic, assuming stereo duplex. W: [pulseaudio] alsa-ucm.c: UCM file does not specify 'PlaybackChannels' or 'CaptureChannels'for device PrimaryMic, assuming stereo duplex. W: [pulseaudio] alsa-ucm.c: UCM file does not specify 'PlaybackChannels' or 'CaptureChannels'for device Handset, assuming stereo duplex. W: [pulseaudio] alsa-ucm.c: UCM file does not specify 'PlaybackChannels' or 'CaptureChannels'for device Earpiece, assuming stereo duplex. W: [pulseaudio] alsa-ucm.c: UCM file does not specify 'PlaybackChannels' or 'CaptureChannels'for device Headphones, assuming stereo duplex. W: [pulseaudio] alsa-ucm.c: UCM file does not specify 'PlaybackChannels' or 'CaptureChannels'for device Speaker, assuming stereo duplex. I: [pulseaudio] module-alsa-card.c: Found UCM profiles E: [pulseaudio] alsa-ucm.c: No sink and source at HiFi: DigitalMic E: [pulseaudio] alsa-ucm.c: No sink and source at HiFi: SecondaryMic E: [pulseaudio] alsa-ucm.c: No sink and source at HiFi: PrimaryMic E: [pulseaudio] alsa-ucm.c: No sink and source at HiFi: Handset E: [pulseaudio] alsa-ucm.c: No sink and source at HiFi: Earpiece E: [pulseaudio] alsa-ucm.c: No sink and source at HiFi: Headphones E: [pulseaudio] alsa-ucm.c: No sink and source at HiFi: Speaker I: [pulseaudio] alsa-ucm.c: Set ucm verb to HiFi It seems like PulseAudio does not inherit them from "SectionVerb", but I'm not sure if this is a problem in PulseAudio or alsa-lib. Restoring Playback/CaptureChannels for all the "SectionDevice"s fixes the problem. Stephan Gerhold (7): DB410c: Rename Handset to HeadsetMic DB410c: Split devices into re-usable fragments in codecs/msm8916-wcd codecs/msm8916-wcd: Speaker: Reset volume on disable codecs/msm8916-wcd: Headphones: Drop invalid RDAC2 MUX value codecs/msm8916-wcd: Earpiece: Add missing Enable/DisableSequence codecs/msm8916-wcd: SecondaryMic: Select correct input codecs/msm8916-wcd: Add ConflictingDevice for capture devices ucm2/DB410c/HiFi.conf | 128 +--------------------- ucm2/codecs/msm8916-wcd/DigitalMic.conf | 18 +++ ucm2/codecs/msm8916-wcd/Earpiece.conf | 22 ++++ ucm2/codecs/msm8916-wcd/Headphones.conf | 29 +++++ ucm2/codecs/msm8916-wcd/HeadsetMic.conf | 22 ++++ ucm2/codecs/msm8916-wcd/PrimaryMic.conf | 20 ++++ ucm2/codecs/msm8916-wcd/SecondaryMic.conf | 22 ++++ ucm2/codecs/msm8916-wcd/Speaker.conf | 21 ++++ 8 files changed, 160 insertions(+), 122 deletions(-) create mode 100644 ucm2/codecs/msm8916-wcd/DigitalMic.conf create mode 100644 ucm2/codecs/msm8916-wcd/Earpiece.conf create mode 100644 ucm2/codecs/msm8916-wcd/Headphones.conf create mode 100644 ucm2/codecs/msm8916-wcd/HeadsetMic.conf create mode 100644 ucm2/codecs/msm8916-wcd/PrimaryMic.conf create mode 100644 ucm2/codecs/msm8916-wcd/SecondaryMic.conf create mode 100644 ucm2/codecs/msm8916-wcd/Speaker.conf -- 2.24.1

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[alsa-devel] [PATCH 0/9] ASoC: SOF: update S0ix/D0ix support
by Pierre-Louis Bossart 11 Feb '20

11 Feb '20

The initial support for S0ix had limitations that were identified during testing, and opportunitic transitions such as S0/D0ix was not a case that was handled. Ranjani suggested a major rework reviewed at length on GitHub [1][2]. So far all known issues are solved and no regressions are identified on 5 different platforms, so it's time to share upstream. This is not intended for 5.6 and can wait after the merge window closes. But since the code isn't simple we want to allow plenty of time for further reviews. Thanks Ranjani for this important update! [1] https://github.com/thesofproject/linux/pull/1631 [2] https://github.com/thesofproject/linux/pull/1702 Ranjani Sridharan (9): ASoC: SOF: Do not reset hw_params for streams that ignored suspend ASoC: SOF: pm: Unify suspend/resume routines ASoC: SOF: Add system_suspend_target field to struct snd_sof_dev ASoC: SOF: pm: Introduce DSP power states ASoC: SOF: Move DSP power state transitions to platform-specific ops ASoC: SOF: audio: Add helper to check if only D0i3 streams are active ASoC: SOF: Intel: hda: Amend the DSP state transition diagram ASoC: SOF: Intel: cnl: Implement feature to support DSP D0i3 in S0 ASoC: SOF: Intel: hda: Allow trace DMA in S0 when DSP is in D0I3 for debug sound/soc/sof/core.c | 4 +- sound/soc/sof/intel/cnl.c | 37 ++++- sound/soc/sof/intel/hda-dsp.c | 289 +++++++++++++++++++++++++++++++--- sound/soc/sof/intel/hda.c | 5 + sound/soc/sof/intel/hda.h | 21 ++- sound/soc/sof/ipc.c | 29 +++- sound/soc/sof/ops.h | 16 +- sound/soc/sof/pcm.c | 2 +- sound/soc/sof/pm.c | 176 ++++++++------------- sound/soc/sof/sof-audio.c | 42 ++++- sound/soc/sof/sof-audio.h | 3 +- sound/soc/sof/sof-priv.h | 43 +++-- 12 files changed, 502 insertions(+), 165 deletions(-) -- 2.20.1

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[alsa-devel] [PATCH] ALSA: hda - fixup for the bass speaker on Lenovo Carbon X1 7th gen
by Jaroslav Kysela 11 Feb '20

11 Feb '20

The auto-parser assigns the bass speaker to DAC3 (NID 0x06) which is without the volume control. I do not see a reason to use DAC2, because the shared output to all speakers produces the sufficient and well balanced sound. The stereo support is enough for this purpose (laptop). Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/pci/hda/patch_realtek.c | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d2bf70a1d2fd..a596790d4245 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5544,6 +5544,16 @@ static void alc295_fixup_disable_dac3(struct hda_codec *codec, } } +/* force NID 0x17 (Bass Speaker) to DAC1 to share it with the main speaker */ +static void alc285_fixup_speaker2_to_dac1(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + hda_nid_t conn[1] = { 0x02 }; + snd_hda_override_conn_list(codec, 0x17, 1, conn); + } +} + /* Hook to update amp GPIO4 for automute */ static void alc280_hp_gpio4_automute_hook(struct hda_codec *codec, struct hda_jack_callback *jack) @@ -5846,6 +5856,7 @@ enum { ALC225_FIXUP_DISABLE_MIC_VREF, ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, ALC295_FIXUP_DISABLE_DAC3, + ALC285_FIXUP_SPEAKER2_TO_DAC1, ALC280_FIXUP_HP_HEADSET_MIC, ALC221_FIXUP_HP_FRONT_MIC, ALC292_FIXUP_TPT460, @@ -6646,6 +6657,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc295_fixup_disable_dac3, }, + [ALC285_FIXUP_SPEAKER2_TO_DAC1] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_speaker2_to_dac1, + }, [ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -7221,6 +7236,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x224c, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x224d, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x225d, "Thinkpad T480", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x2293, "Thinkpad X1 Carbon 7th", ALC285_FIXUP_SPEAKER2_TO_DAC1), SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x310c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), @@ -7405,6 +7421,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC255_FIXUP_DELL_SPK_NOISE, .name = "dell-spk-noise"}, {.id = ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "alc225-dell1"}, {.id = ALC295_FIXUP_DISABLE_DAC3, .name = "alc295-disable-dac3"}, + {.id = ALC285_FIXUP_SPEAKER2_TO_DAC1, .name = "alc285-speaker2-to-dac1"}, {.id = ALC280_FIXUP_HP_HEADSET_MIC, .name = "alc280-hp-headset"}, {.id = ALC221_FIXUP_HP_FRONT_MIC, .name = "alc221-hp-mic"}, {.id = ALC298_FIXUP_SPK_VOLUME, .name = "alc298-spk-volume"}, -- 2.20.1

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