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

[PATCH v1 1/2] dt-bindings: sound: convert rockchip i2s bindings to yaml
by Johan Jonker 24 Mar '20

24 Mar '20

Current dts files with 'i2s' nodes are manually verified. In order to automate this process rockchip-i2s.txt has to be converted to yaml. Signed-off-by: Johan Jonker <jbx6244(a)gmail.com> --- .../devicetree/bindings/sound/rockchip-i2s.txt | 49 ---------- .../devicetree/bindings/sound/rockchip-i2s.yaml | 106 +++++++++++++++++++++ 2 files changed, 106 insertions(+), 49 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/rockchip-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip-i2s.yaml diff --git a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt deleted file mode 100644 index 54aefab71..000000000 --- a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt +++ /dev/null @@ -1,49 +0,0 @@ -* Rockchip I2S controller - -The I2S bus (Inter-IC sound bus) is a serial link for digital -audio data transfer between devices in the system. - -Required properties: - -- compatible: should be one of the following: - - "rockchip,rk3066-i2s": for rk3066 - - "rockchip,px30-i2s", "rockchip,rk3066-i2s": for px30 - - "rockchip,rk3036-i2s", "rockchip,rk3066-i2s": for rk3036 - - "rockchip,rk3188-i2s", "rockchip,rk3066-i2s": for rk3188 - - "rockchip,rk3228-i2s", "rockchip,rk3066-i2s": for rk3228 - - "rockchip,rk3288-i2s", "rockchip,rk3066-i2s": for rk3288 - - "rockchip,rk3328-i2s", "rockchip,rk3066-i2s": for rk3328 - - "rockchip,rk3366-i2s", "rockchip,rk3066-i2s": for rk3366 - - "rockchip,rk3368-i2s", "rockchip,rk3066-i2s": for rk3368 - - "rockchip,rk3399-i2s", "rockchip,rk3066-i2s": for rk3399 -- reg: physical base address of the controller and length of memory mapped - region. -- interrupts: should contain the I2S interrupt. -- dmas: DMA specifiers for tx and rx dma. See the DMA client binding, - Documentation/devicetree/bindings/dma/dma.txt -- dma-names: should include "tx" and "rx". -- clocks: a list of phandle + clock-specifer pairs, one for each entry in clock-names. -- clock-names: should contain the following: - - "i2s_hclk": clock for I2S BUS - - "i2s_clk" : clock for I2S controller -- rockchip,playback-channels: max playback channels, if not set, 8 channels default. -- rockchip,capture-channels: max capture channels, if not set, 2 channels default. - -Required properties for controller which support multi channels -playback/capture: - -- rockchip,grf: the phandle of the syscon node for GRF register. - -Example for rk3288 I2S controller: - -i2s@ff890000 { - compatible = "rockchip,rk3288-i2s", "rockchip,rk3066-i2s"; - reg = <0xff890000 0x10000>; - interrupts = <GIC_SPI 85 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&pdma1 0>, <&pdma1 1>; - dma-names = "tx", "rx"; - clock-names = "i2s_hclk", "i2s_clk"; - clocks = <&cru HCLK_I2S0>, <&cru SCLK_I2S0>; - rockchip,playback-channels = <8>; - rockchip,capture-channels = <2>; -}; diff --git a/Documentation/devicetree/bindings/sound/rockchip-i2s.yaml b/Documentation/devicetree/bindings/sound/rockchip-i2s.yaml new file mode 100644 index 000000000..eff06b4b5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip-i2s.yaml @@ -0,0 +1,106 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rockchip-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Rockchip I2S controller + +description: + The I2S bus (Inter-IC sound bus) is a serial link for digital + audio data transfer between devices in the system. + +maintainers: + - Heiko Stuebner <heiko(a)sntech.de> + +properties: + compatible: + oneOf: + - const: rockchip,rk3066-i2s + - items: + - enum: + - rockchip,px30-i2s + - rockchip,rk3036-i2s + - rockchip,rk3188-i2s + - rockchip,rk3228-i2s + - rockchip,rk3288-i2s + - rockchip,rk3328-i2s + - rockchip,rk3366-i2s + - rockchip,rk3368-i2s + - rockchip,rk3399-i2s + - const: rockchip,rk3066-i2s + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: clock for I2S controller + - description: clock for I2S BUS + + clock-names: + items: + - const: i2s_clk + - const: i2s_hclk + + dmas: + items: + - description: TX DMA Channel + - description: RX DMA Channel + + dma-names: + items: + - const: tx + - const: rx + + rockchip,capture-channels: + allOf: + - $ref: /schemas/types.yaml#/definitions/uint32 + default: 2 + description: + Max capture channels, if not set, 2 channels default. + + rockchip,playback-channels: + allOf: + - $ref: /schemas/types.yaml#/definitions/uint32 + default: 8 + description: + Max playback channels, if not set, 8 channels default. + + rockchip,grf: + $ref: /schemas/types.yaml#/definitions/phandle + description: + The phandle of the syscon node for the GRF register. + Required property for controllers which support multi channel + playback/capture. + +required: + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/rk3288-cru.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + i2s@ff890000 { + compatible = "rockchip,rk3288-i2s", "rockchip,rk3066-i2s"; + reg = <0xff890000 0x10000>; + interrupts = <GIC_SPI 85 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&cru SCLK_I2S0>, <&cru HCLK_I2S0>; + clock-names = "i2s_clk", "i2s_hclk"; + dmas = <&pdma1 0>, <&pdma1 1>; + dma-names = "tx", "rx"; + rockchip,capture-channels = <2>; + rockchip,playback-channels = <8>; + }; -- 2.11.0

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[PATCH 1/2] ALSA: core: Add snd_device_get_state() helper
by Takashi Iwai 23 Mar '20

23 Mar '20

A new small helper to get the current state of the device registration for the given object. It'll be used for USB-audio driver to check the delayed device registrations. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/core.h | 1 + sound/core/device.c | 21 +++++++++++++++++++++ 2 files changed, 22 insertions(+) diff --git a/include/sound/core.h b/include/sound/core.h index ac8b692b69b4..381a010a1bd4 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -266,6 +266,7 @@ void snd_device_disconnect(struct snd_card *card, void *device_data); void snd_device_disconnect_all(struct snd_card *card); void snd_device_free(struct snd_card *card, void *device_data); void snd_device_free_all(struct snd_card *card); +int snd_device_get_state(struct snd_card *card, void *device_data); /* isadma.c */ diff --git a/sound/core/device.c b/sound/core/device.c index cdc5af526739..bf0b04a7ee79 100644 --- a/sound/core/device.c +++ b/sound/core/device.c @@ -237,3 +237,24 @@ void snd_device_free_all(struct snd_card *card) list_for_each_entry_safe_reverse(dev, next, &card->devices, list) __snd_device_free(dev); } + +/** + * snd_device_get_state - Get the current state of the given device + * @card: the card instance + * @device_data: the data pointer to release + * + * Returns the current state of the given device object. For the valid + * device, either @SNDRV_DEV_BUILD, @SNDRV_DEV_REGISTERED or + * @SNDRV_DEV_DISCONNECTED is returned. + * Or for a non-existing device, -1 is returned as an error. + */ +int snd_device_get_state(struct snd_card *card, void *device_data) +{ + struct snd_device *dev; + + dev = look_for_dev(card, device_data); + if (dev) + return dev->state; + return -1; +} +EXPORT_SYMBOL_GPL(snd_device_get_state); -- 2.16.4

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Re: UCM ConflictingDevice/Priority concepts
by Pierre-Louis Bossart 23 Mar '20

23 Mar '20

[fixing alsa-devel email and rejoining threads] On 3/19/20 4:06 AM, Jaroslav Kysela wrote: > Dne 18. 03. 20 v 22:46 Pierre-Louis Bossart napsal(a): >> Hi, >> >> Traditionally on most PC or mobile platforms, we have one audio output >> that can be routed to either speakers or headphone, and likewise we can >> record from either internal mics or headset mic. We signal with UCM that >> the headphone/speakers and internal mic/headset conflict so hopefully >> PulseAudio/CRAS switch auto-magically. >> >> For SoundWire-based platforms, we typically have a headphone/headset >> codec on one link, and one or more amplifiers on the other. Functionally >> it's supported to capture from local mics and headset mic at the same >> time, or play different streams on speakers and headphones. Recent >> Intel-based Chromebooks have in theory the same capabilities at the >> hardware level even with I2S/TDM + DMIC connections. >> >> So for UCM, should we use the notion of 'ConflictingDevice' to fall-back >> to a more traditional single-endpoint user experience, or is this >> concept only indented to model hardware restrictions? I just checked >> that Chrome/adhd does not seem to use this concept at all, while it's >> prevalent in alsa-ucm-conf >> >> Or should we instead only use the concept of Playback/CapturePriority, >> which is also used in a lot of alsa-ucm-conf files, but again not at all >> in Chrome/adhd? >> >> I did find some UCM files relying both on the concept of >> ConflictingDevices and PlaybackPriorities, which seems rather >> odd/overkill to me. > > ConflictingDevices/SupportedDevices should be used only if there's a > hardware restriction which prevents the simultaneous usage of devices. > The application can decide how to use those devices. > > The priority describes the preference. Usually, headphones has higher > priority than build-in speakers etc. I may be thick on this one, but how would an application use both types of information? Does it e.g. a) revisit the list all devices currently available when an event occurs (uevent card creation, jack detection, etc) b) pick the device with the highest priority for the 'default' stream c) allow for simultaneous use of devices not marked at 'Conflicting', e.g. use the internal microphone for assistant while using the headset mic for a call as suggested by Dylan. In other words the priority is the first key, and additional devices are filtered with the ConflictingDevice information. Did I get this right? > > In my opinion, it's not part of UCM if the application will use one or > multiple devices. The application must decide. It's another upper > usage / abstraction layer. I tend to agree, but I wanted to make sure the use of 'ConflictingDevices' was not expected outside of true hardware limitations. > > Also, we need to consider this to have the whole picture: > > Tanu (the pulseaudio maintainer) has also good question how to ensure, > that the stream can be re-used for the multiple devices. Actually, PA > does not re-open PCM device when the PCM device name and parameters > are similar for the switched devices. I also think that this is also > missing in the UCM specification to resolve this requirement. Usually, > the stream transfer mechanism is separate from the routing control. > But I can assume, that we may have the hardware which will need extra > setup for the streaming (not routing) when the devices are switched. > > I think that adding something like "PlaybackStream" to "PlaybackPCM" > for the stream identification might be sufficient to cover those > cases. So, keep "PlaybackPCM" usage and if "PlaybackStream" exists, > use this value to determine the stream identification. Similar > situation is for the capture direction, of course. I am not sure I understand the notion of stream and stream transfer. Is there a pointer to this so that I could understand the problem statement? Thanks, -Pierre

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Re: [patch 08/22] ACPI: Convert to new X86 CPU match macros
by Andy Shevchenko 23 Mar '20

23 Mar '20

On Fri, Mar 20, 2020 at 3:19 PM Thomas Gleixner <tglx(a)linutronix.de> wrote: > > The new macro set has a consistent namespace and uses C99 initializers > instead of the grufty C89 ones. > > Rename the local macro wrapper to X86_MATCH for consistency. It stays for > readability sake. > + X86_MATCH_INTEL_FAM6_MODEL(ATOM_SILVERMONT, NULL), > + X86_MATCH_INTEL_FAM6_MODEL(ATOM_AIRMONT, NULL), > -#define ICPU(model) { X86_VENDOR_INTEL, 6, model, X86_FEATURE_ANY, } > +#define X86_MATCH(model) X86_MATCH_INTEL_FAM6_MODEL(model, NULL) Maybe we can do a generic macro to avoid all these ', NULL' repetitions? -- With Best Regards, Andy Shevchenko

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[PATCH v1 0/2] Support built-in Mic on Tegra boards that use WM8903
by Dmitry Osipenko 23 Mar '20

23 Mar '20

Hello, This small series adds audio route for built-in microphone on NVIDIA Tegra boards that use WM8903 CODEC. In particular this is needed in order to unmute internal microphone on Acer A500 tablet device. I'm planning to send out the device tree for the A500 for 5.8, so will be nice to get the microphone sorted out. Please review and apply, thanks in advance. Dmitry Osipenko (2): dt-bindings: sound: tegra-wm8903: Document built-in microphone audio source ASoC: tegra: tegra_wm8903: Support DAPM events for built-in microphone .../sound/nvidia,tegra-audio-wm8903.txt | 1 + sound/soc/tegra/tegra_wm8903.c | 18 ++++++++++++++++++ 2 files changed, 19 insertions(+) -- 2.25.1

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[PATCH] ASoC: SOF: fix uninitialised "work" with VirtIO
by Guennadi Liakhovetski 23 Mar '20

23 Mar '20

In the VirtIO case the sof_pcm_open() function isn't called on the host during guest streaming, which then leaves "work" structures uninitialised. However it is then used to handle position update messages from the DSP. Move their initialisation to immediately after allocation of the containing structure. Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> --- This is a re-send of "[PATCH 08/14] ASoC: SOF: fix uninitialised "work" with VirtIO" as suggested by Mark, also taking into account a comment from Ranjani - thanks. Note: I haven't sent patches before from mutt, hope this will work, if not - will have to re-send. sound/soc/sof/pcm.c | 4 +--- sound/soc/sof/sof-audio.h | 3 +++ sound/soc/sof/topology.c | 17 ++++++++++++----- 3 files changed, 16 insertions(+), 8 deletions(-) diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c index f4769e1..47cd741 100644 --- a/sound/soc/sof/pcm.c +++ b/sound/soc/sof/pcm.c @@ -57,7 +57,7 @@ static int sof_pcm_dsp_params(struct snd_sof_pcm *spcm, struct snd_pcm_substream /* * sof pcm period elapse work */ -static void sof_pcm_period_elapsed_work(struct work_struct *work) +void snd_sof_pcm_period_elapsed_work(struct work_struct *work) { struct snd_sof_pcm_stream *sps = container_of(work, struct snd_sof_pcm_stream, @@ -475,8 +475,6 @@ static int sof_pcm_open(struct snd_soc_component *component, dev_dbg(component->dev, "pcm: open stream %d dir %d\n", spcm->pcm.pcm_id, substream->stream); - INIT_WORK(&spcm->stream[substream->stream].period_elapsed_work, - sof_pcm_period_elapsed_work); caps = &spcm->pcm.caps[substream->stream]; diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h index eacd10e..bf65f31a 100644 --- a/sound/soc/sof/sof-audio.h +++ b/sound/soc/sof/sof-audio.h @@ -11,6 +11,8 @@ #ifndef __SOUND_SOC_SOF_AUDIO_H #define __SOUND_SOC_SOF_AUDIO_H +#include <linux/workqueue.h> + #include <sound/soc.h> #include <sound/control.h> #include <sound/sof/stream.h> /* needs to be included before control.h */ @@ -189,6 +191,7 @@ struct snd_sof_pcm *snd_sof_find_spcm_comp(struct snd_soc_component *scomp, struct snd_sof_pcm *snd_sof_find_spcm_pcm_id(struct snd_soc_component *scomp, unsigned int pcm_id); void snd_sof_pcm_period_elapsed(struct snd_pcm_substream *substream); +void snd_sof_pcm_period_elapsed_work(struct work_struct *work); /* * Mixer IPC diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 058de94..fe8ba3e 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -9,6 +9,7 @@ // #include <linux/firmware.h> +#include <linux/workqueue.h> #include <sound/tlv.h> #include <sound/pcm_params.h> #include <uapi/sound/sof/tokens.h> @@ -2448,7 +2449,7 @@ static int sof_dai_load(struct snd_soc_component *scomp, int index, struct snd_soc_tplg_stream_caps *caps; struct snd_soc_tplg_private *private = &pcm->priv; struct snd_sof_pcm *spcm; - int stream = SNDRV_PCM_STREAM_PLAYBACK; + int stream; int ret = 0; /* nothing to do for BEs atm */ @@ -2460,8 +2461,12 @@ static int sof_dai_load(struct snd_soc_component *scomp, int index, return -ENOMEM; spcm->scomp = scomp; - spcm->stream[SNDRV_PCM_STREAM_PLAYBACK].comp_id = COMP_ID_UNASSIGNED; - spcm->stream[SNDRV_PCM_STREAM_CAPTURE].comp_id = COMP_ID_UNASSIGNED; + + for_each_pcm_streams(stream) { + spcm->stream[stream].comp_id = COMP_ID_UNASSIGNED; + INIT_WORK(&spcm->stream[stream].period_elapsed_work, + snd_sof_pcm_period_elapsed_work); + } spcm->pcm = *pcm; dev_dbg(scomp->dev, "tplg: load pcm %s\n", pcm->dai_name); @@ -2482,8 +2487,10 @@ static int sof_dai_load(struct snd_soc_component *scomp, int index, if (!spcm->pcm.playback) goto capture; + stream = SNDRV_PCM_STREAM_PLAYBACK; + dev_vdbg(scomp->dev, "tplg: pcm %s stream tokens: playback d0i3:%d\n", - spcm->pcm.pcm_name, spcm->stream[0].d0i3_compatible); + spcm->pcm.pcm_name, spcm->stream[stream].d0i3_compatible); caps = &spcm->pcm.caps[stream]; @@ -2513,7 +2520,7 @@ static int sof_dai_load(struct snd_soc_component *scomp, int index, return ret; dev_vdbg(scomp->dev, "tplg: pcm %s stream tokens: capture d0i3:%d\n", - spcm->pcm.pcm_name, spcm->stream[1].d0i3_compatible); + spcm->pcm.pcm_name, spcm->stream[stream].d0i3_compatible); caps = &spcm->pcm.caps[stream]; -- 1.9.3

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[PATCH 0/5] soundwire: add sdw_master_device support on Qualcomm platforms
by Pierre-Louis Bossart 23 Mar '20

23 Mar '20

This patchset provides the support for sdw_master_device and sdw_master_driver reviewed earlier and accepted as 'sane' by Greg KH. This patchset only focuses on Qualcomm platforms and addresses all previous comments and objections from Vinod Koul: there is no driver used and no overhead added by this patchset. Many thanks to Srinivas Kandagatla for testing these patches on Qualcomm devices and contributing the missing DT parsing fix. The transition from platform devices to sdw_master_devices on Intel platforms was already provided and will be re-sent separately when this infrastructure change is agreed. Pierre-Louis Bossart (4): soundwire: bus_type: add master_device/driver support soundwire: bus_type: protect cases where no driver name is provided soundwire: qcom: fix error handling in probe soundwire: qcom: add sdw_master_device support Srinivas Kandagatla (1): soundwire: master: use device node pointer from master device drivers/soundwire/Makefile | 2 +- drivers/soundwire/bus_type.c | 143 +++++++++++++++++++++++++++-- drivers/soundwire/master.c | 130 ++++++++++++++++++++++++++ drivers/soundwire/qcom.c | 49 ++++++++-- drivers/soundwire/slave.c | 7 +- include/linux/soundwire/sdw.h | 59 ++++++++++++ include/linux/soundwire/sdw_type.h | 36 +++++++- 7 files changed, 407 insertions(+), 19 deletions(-) create mode 100644 drivers/soundwire/master.c base-commit: 1ce7139436603dda9e155df0c3e275c87a725761 -- 2.20.1

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[PATCH] soundwire: stream: only change state if needed
by Pierre-Louis Bossart 23 Mar '20

23 Mar '20

In a multi-cpu DAI context, the stream routines may be called from multiple DAI callbacks. Make sure the stream state only changes for the first call, and don't return error messages if the target state is already reached. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- drivers/soundwire/stream.c | 19 +++++++++++++++++++ 1 file changed, 19 insertions(+) diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index 1b43d03c79ea..3319121cd706 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -1572,6 +1572,7 @@ int sdw_prepare_stream(struct sdw_stream_runtime *stream) sdw_acquire_bus_lock(stream); if (stream->state == SDW_STREAM_PREPARED) { + /* nothing to do */ ret = 0; goto state_err; } @@ -1661,6 +1662,12 @@ int sdw_enable_stream(struct sdw_stream_runtime *stream) sdw_acquire_bus_lock(stream); + if (stream->state == SDW_STREAM_ENABLED) { + /* nothing to do */ + ret = 0; + goto state_err; + } + if (stream->state != SDW_STREAM_PREPARED && stream->state != SDW_STREAM_DISABLED) { pr_err("%s: %s: inconsistent state state %d\n", @@ -1744,6 +1751,12 @@ int sdw_disable_stream(struct sdw_stream_runtime *stream) sdw_acquire_bus_lock(stream); + if (stream->state == SDW_STREAM_DISABLED) { + /* nothing to do */ + ret = 0; + goto state_err; + } + if (stream->state != SDW_STREAM_ENABLED) { pr_err("%s: %s: inconsistent state state %d\n", __func__, stream->name, stream->state); @@ -1809,6 +1822,12 @@ int sdw_deprepare_stream(struct sdw_stream_runtime *stream) sdw_acquire_bus_lock(stream); + if (stream->state == SDW_STREAM_DEPREPARED) { + /* nothing to do */ + ret = 0; + goto state_err; + } + if (stream->state != SDW_STREAM_PREPARED && stream->state != SDW_STREAM_DISABLED) { pr_err("%s: %s: inconsistent state state %d\n", -- 2.20.1

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[PATCH 00/14] ASoC: SOF DSP virtualisation
by Guennadi Liakhovetski 23 Mar '20

23 Mar '20

This patch series extends the SOF driver to add support for DSP virtualisation to ASoC. It is built on top of VirtIO, contains a guest driver and a vhost in-kernel guest driver. This version supports a single playback and a single capture interface on the guest. The specific guest audio topology is supplied by the host from a file, specified on the QEMU command line. This solution is designed to use advanced DSP functionality, available on the host. In case no DSP is available on the host, a fall-back should be provided in the future to the pure audio virtualisation standard, currently developed by OASIS. Extensive documentation is available at [1]. Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> [1] https://thesofproject.github.io/latest/developer_guides/virtualization/virt… Guennadi Liakhovetski (14): ASoC: add function parameters to enable forced path pruning ASoC: SOF: extract firmware-related operation into a function ASoC: SOF: VirtIO: make a function global vhost: convert VHOST_VSOCK_SET_RUNNING to a generic ioctl ASoC: SOF: support IPC with immediate response ASoC: SOF: add a power status IPC ASoC: SOF: add two helper lookup functions ASoC: SOF: fix uninitialised "work" with VirtIO ASoC: SOF: add a VirtIO DSP driver ASoC: SOF: add a vhost driver: sound part vhost: add an SOF DSP driver ASoC: SOF: VirtIO: check guest component IDs ASoC: SOF: VirtIO: free guest pipelines upon termination ASoC: SOF: VirtIO: enable simultaneous playback and capture drivers/vhost/Kconfig | 7 + drivers/vhost/Makefile | 5 + drivers/vhost/dsp.c | 731 ++++++++++++++++++++++ include/sound/soc-dpcm.h | 28 +- include/sound/soc-topology.h | 3 + include/sound/sof.h | 4 + include/sound/sof/header.h | 3 + include/sound/sof/topology.h | 1 + include/sound/sof/virtio.h | 206 +++++++ include/uapi/linux/vhost.h | 9 +- include/uapi/linux/vhost_types.h | 7 + include/uapi/linux/virtio_ids.h | 1 + sound/soc/soc-compress.c | 2 +- sound/soc/soc-dapm.c | 8 +- sound/soc/soc-pcm.c | 98 ++- sound/soc/sof/Kconfig | 8 + sound/soc/sof/Makefile | 8 + sound/soc/sof/core.c | 114 ++-- sound/soc/sof/ipc.c | 39 +- sound/soc/sof/loader.c | 4 + sound/soc/sof/ops.h | 10 +- sound/soc/sof/pcm.c | 17 +- sound/soc/sof/pm.c | 4 + sound/soc/sof/sof-audio.c | 33 + sound/soc/sof/sof-audio.h | 21 + sound/soc/sof/sof-priv.h | 54 ++ sound/soc/sof/topology.c | 76 ++- sound/soc/sof/vhost-be.c | 1248 ++++++++++++++++++++++++++++++++++++++ sound/soc/sof/virtio-fe.c | 922 ++++++++++++++++++++++++++++ 29 files changed, 3560 insertions(+), 111 deletions(-) create mode 100644 drivers/vhost/dsp.c create mode 100644 include/sound/sof/virtio.h create mode 100644 sound/soc/sof/vhost-be.c create mode 100644 sound/soc/sof/virtio-fe.c -- 1.9.3

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[PATCH 0/7] SoundWire: intel: fix SHIM programming sequences
by Pierre-Louis Bossart 23 Mar '20

23 Mar '20

The low-level register programming sequences contributed in 71bb8a1b059ecd ('soundwire: intel: Add Intel Master driver') do not follow the internal documentation and recommended flows. It's anyone's guess how the code might have worked. Fix and add all missing helpers for clock-stop and hardware-based synchronization. This patchset needs to be applied on top of "[PATCH 00/16] SoundWire: cadence: add clock stop and fix programming sequences" Reviewers might object that the code is provided without some required initializations for mutexes and shim masks, they will be added as part of the transition to sdw_master_device - still stuck as of 3/11. Pierre-Louis Bossart (6): soundwire: intel: add helpers for link power down and shim wake soundwire: intel: reuse code for wait loops to set/clear bits soundwire: intel: add mutex to prevent concurrent access to SHIM registers soundwire: intel: add definitions for shim_mask soundwire: intel: introduce a helper to arm link synchronization soundwire: intel: introduce helper for link synchronization Rander Wang (1): soundwire: intel: follow documentation sequences for SHIM registers drivers/soundwire/intel.c | 342 ++++++++++++++++++++++------ drivers/soundwire/intel.h | 4 + include/linux/soundwire/sdw_intel.h | 2 + 3 files changed, 277 insertions(+), 71 deletions(-) -- 2.20.1

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