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

[alsa-devel] [PATCH 0/3] ASoC: SOF: VirtIO: preparatory patches
by Guennadi Liakhovetski 10 Feb '20

10 Feb '20

This is the first set of patches for the SOF virtualisation work. We send these patches first because they touch the ASoC core. 2 of them are mostly cosmetic with no functional changes, but patch 2/3 might cause some discussions. Please review and comment. Thanks Guennadi Guennadi Liakhovetski (3): ASoC: (cosmetic) simplify dpcm_prune_paths() ASoC: add function parameters to enable forced path pruning ASoC: export DPCM runtime update functions include/sound/soc-dpcm.h | 30 +++++++++-- sound/soc/soc-compress.c | 2 +- sound/soc/soc-dapm.c | 8 +-- sound/soc/soc-pcm.c | 126 ++++++++++++++++++++++++++++++----------------- 4 files changed, 112 insertions(+), 54 deletions(-) -- 1.9.3

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[alsa-devel] [RESEND PATCH] ASoC: atmel: fix atmel_ssc_set_audio link failure
by Codrin Ciubotariu 10 Feb '20

10 Feb '20

From: Arnd Bergmann <arnd(a)arndb.de> The ssc audio driver can call into both pdc and dma backends. With the latest rework, the logic to do this in a safe way avoiding link errors was removed, bringing back link errors that were fixed long ago in commit 061981ff8cc8 ("ASoC: atmel: properly select dma driver state") such as sound/soc/atmel/atmel_ssc_dai.o: In function `atmel_ssc_set_audio': atmel_ssc_dai.c:(.text+0xac): undefined reference to `atmel_pcm_pdc_platform_register' Fix it this time using Makefile hacks and a comment to prevent this from accidentally getting removed again rather than Kconfig hacks. Fixes: 18291410557f ("ASoC: atmel: enable SOC_SSC_PDC and SOC_SSC_DMA in Kconfig") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> --- Added my signed-off because I sent the patch, no other changes. sound/soc/atmel/Kconfig | 4 ++-- sound/soc/atmel/Makefile | 10 ++++++++-- 2 files changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index d1dc8e6366dc..71f2d42188c4 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -10,11 +10,11 @@ config SND_ATMEL_SOC if SND_ATMEL_SOC config SND_ATMEL_SOC_PDC - tristate + bool depends on HAS_DMA config SND_ATMEL_SOC_DMA - tristate + bool select SND_SOC_GENERIC_DMAENGINE_PCM config SND_ATMEL_SOC_SSC diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile index 1f6890ed3738..c7d2989791be 100644 --- a/sound/soc/atmel/Makefile +++ b/sound/soc/atmel/Makefile @@ -6,8 +6,14 @@ snd-soc-atmel_ssc_dai-objs := atmel_ssc_dai.o snd-soc-atmel-i2s-objs := atmel-i2s.o snd-soc-mchp-i2s-mcc-objs := mchp-i2s-mcc.o -obj-$(CONFIG_SND_ATMEL_SOC_PDC) += snd-soc-atmel-pcm-pdc.o -obj-$(CONFIG_SND_ATMEL_SOC_DMA) += snd-soc-atmel-pcm-dma.o +# pdc and dma need to both be built-in if any user of +# ssc is built-in. +ifdef CONFIG_SND_ATMEL_SOC_PDC +obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel-pcm-pdc.o +endif +ifdef CONFIG_SND_ATMEL_SOC_DMA +obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel-pcm-dma.o +endif obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o obj-$(CONFIG_SND_ATMEL_SOC_I2S) += snd-soc-atmel-i2s.o obj-$(CONFIG_SND_MCHP_SOC_I2S_MCC) += snd-soc-mchp-i2s-mcc.o -- 2.20.1

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[alsa-devel] [PATCH v2 0/2] Update FE/BE trigger order based on the command
by Ranjani Sridharan 09 Feb '20

09 Feb '20

Currently, the trigger orders SND_SOC_DPCM_TRIGGER_PRE/POST determine the order in which FE DAI and BE DAI are triggered. In the case of SND_SOC_DPCM_TRIGGER_PRE, the FE DAI is triggered before the BE DAI and in the case of SND_SOC_DPCM_TRIGGER_POST, the BE DAI is triggered before the FE DAI. And this order remains the same irrespective of the trigger command. In the case of the SOF driver, during playback, the FW expects the BE DAI to be triggered before the FE DAI during the START trigger. The BE DAI trigger handles the starting of Link DMA and so it must be started before the FE DAI is started to prevent xruns during pause/release. This can be addressed by setting the trigger order for the FE dai link to SND_SOC_DPCM_TRIGGER_POST. But during the STOP trigger, the FW expects the FE DAI to be triggered before the BE DAI. Retaining the same order during the START and STOP commands, results in FW error as the DAI component in the FW is still active. The issue can be fixed by mirroring the trigger order of FE and BE DAI's during the START and STOP trigger. So, with the trigger order set to SND_SOC_DPCM_TRIGGER_PRE, the FE DAI will be trigger first during SNDRV_PCM_TRIGGER_START/STOP/RESUME and the BE DAI will be triggered first during the STOP/SUSPEND/PAUSE commands. Conversely, with the trigger order set to SND_SOC_DPCM_TRIGGER_POST, the BE DAI will be triggered first during the SNDRV_PCM_TRIGGER_START/STOP/RESUME commands and the FE DAI will be triggered first during the SNDRV_PCM_TRIGGER_STOP/SUSPEND/PAUSE commands. The above change has been tested with the Skylake driver on the Kabylake based chromebook and legacy HDA driver on Kabylake based Dell XPS13. It has also been tested on all platforms with SOF CI without any regressions for the past 3 weeks. More details on the issue can be found here: https://github.com/thesofproject/linux/issues/1160 Changes in v2: Fixed comments on function name and consolidating the switch cases. Ranjani Sridharan (2): ASoC: pcm: update FE/BE trigger order based on the command ASoC: SOF: topology: set trigger order for FE DAI link sound/soc/soc-pcm.c | 95 ++++++++++++++++++++++++++++------------ sound/soc/sof/topology.c | 4 ++ 2 files changed, 72 insertions(+), 27 deletions(-) -- 2.17.1

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[alsa-devel] USB Audio Interface / Denon MC7000 and MC8000 controller
by Tobias 07 Feb '20

07 Feb '20

Hello dear ALSA developers. I have purchased a MC7000 controller in order to control MIXXX on Ubuntu 16.04. According to the Denon specification the controller should have been class compliant but there is an issue with the Audio interface to work properly giving following message ... $ dmesg ... |[ 74.522831] usb 1-1.3: new high-speed USB device number 6 using xhci_hcd [ 74.623784] usb 1-1.3: New USB device found, idVendor=15e4, idProduct=8004 [ 74.623789] usb 1-1.3: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [ 74.623793] usb 1-1.3: Product: DENON DJ MC7000 [ 74.623796] usb 1-1.3: Manufacturer: DENON DJ [ 74.623798] usb 1-1.3: SerialNumber: 201603 [ 74.625134] usb 1-1.3: clock source 65 is not valid, cannot use "|||clock source 65 is not valid, cannot use" is repeated uncountable times then |I found a discussion here for the MC8000 Audio interface which was not working and showing the same error message: https://alsa-user.narkive.com/2tDAO87f/troubleshooting-new-usb-audio-device… The solution was to change sound/usb/clock.c and recompile the kernel. This is not practical for daily use case so I would like to ask if the ALSA team could possibly find a permanent fix for ordinary users. To get my outputs for aplay -l aplay -L |||aplay -D plughw:CARD=MC7000,DEV=0 Grimmaldika-MakeBelieve.wav |lsusb -v mixxx --controllerDebug|| JackServer error message| you may have a look at the Mixxx Community Forums here(just on page, you don't need to find and read on multiple pages) https://www.mixxx.org/forums/viewtopic.php?f=7&t=12962&start=10 If you need any more information then please let me know. Thanks a lot for your support on this matter. Cheers! OsZ ||

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[alsa-devel] [PATCH] ASoC: dt-bindings: stm32: convert i2s to json-schema
by Olivier Moysan 05 Feb '20

05 Feb '20

Convert the STM32 I2S bindings to DT schema format using json-schema. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- The items for dma-names and clock-names properties are all mandatory, but may be provided in any order. The syntax used for these properties allows to avoid order constraint. --- .../bindings/sound/st,stm32-i2s.txt | 62 ------------- .../bindings/sound/st,stm32-i2s.yaml | 91 +++++++++++++++++++ 2 files changed, 91 insertions(+), 62 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/st,stm32-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml diff --git a/Documentation/devicetree/bindings/sound/st,stm32-i2s.txt b/Documentation/devicetree/bindings/sound/st,stm32-i2s.txt deleted file mode 100644 index cbf24bcd1b8d..000000000000 --- a/Documentation/devicetree/bindings/sound/st,stm32-i2s.txt +++ /dev/null @@ -1,62 +0,0 @@ -STMicroelectronics STM32 SPI/I2S Controller - -The SPI/I2S block supports I2S/PCM protocols when configured on I2S mode. -Only some SPI instances support I2S. - -Required properties: - - compatible: Must be "st,stm32h7-i2s" - - reg: Offset and length of the device's register set. - - interrupts: Must contain the interrupt line id. - - clocks: Must contain phandle and clock specifier pairs for each entry - in clock-names. - - clock-names: Must contain "i2sclk", "pclk", "x8k" and "x11k". - "i2sclk": clock which feeds the internal clock generator - "pclk": clock which feeds the peripheral bus interface - "x8k": I2S parent clock for sampling rates multiple of 8kHz. - "x11k": I2S parent clock for sampling rates multiple of 11.025kHz. - - dmas: DMA specifiers for tx and rx dma. - See Documentation/devicetree/bindings/dma/stm32-dma.txt. - - dma-names: Identifier for each DMA request line. Must be "tx" and "rx". - - pinctrl-names: should contain only value "default" - - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/st,stm32-pinctrl.yaml - -Optional properties: - - resets: Reference to a reset controller asserting the reset controller - -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 = <&i2s2_port>; -}; - -i2s2: audio-controller@40003800 { - compatible = "st,stm32h7-i2s"; - reg = <0x40003800 0x400>; - interrupts = <36>; - clocks = <&rcc PCLK1>, <&rcc SPI2_CK>, <&rcc PLL1_Q>, <&rcc PLL2_P>; - clock-names = "pclk", "i2sclk", "x8k", "x11k"; - dmas = <&dmamux2 2 39 0x400 0x1>, - <&dmamux2 3 40 0x400 0x1>; - dma-names = "rx", "tx"; - pinctrl-names = "default"; - pinctrl-0 = <&pinctrl_i2s2>; - - i2s2_port: port@0 { - cpu_endpoint: endpoint { - remote-endpoint = <&codec_endpoint>; - format = "i2s"; - }; - }; -}; - -audio-codec { - codec_port: port@0 { - codec_endpoint: endpoint { - remote-endpoint = <&cpu_endpoint>; - }; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml b/Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml new file mode 100644 index 000000000000..cdfb375c7a14 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml @@ -0,0 +1,91 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/st,stm32-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: STMicroelectronics STM32 SPI/I2S Controller + +maintainers: + - Olivier Moysan <olivier.moysan(a)st.com> + +description: + The SPI/I2S block supports I2S/PCM protocols when configured on I2S mode. + Only some SPI instances support I2S. + +properties: + compatible: + enum: + - st,stm32h7-i2s + + "#sound-dai-cells": + const: 0 + + reg: + maxItems: 1 + + clocks: + items: + - description: clock feeding the peripheral bus interface. + - description: clock feeding the internal clock generator. + - description: I2S parent clock for sampling rates multiple of 8kHz. + - description: I2S parent clock for sampling rates multiple of 11.025kHz. + + clock-names: + items: + - enum: [ pclk, i2sclk, x8k, x11k ] + - enum: [ pclk, i2sclk, x8k, x11k ] + - enum: [ pclk, i2sclk, x8k, x11k ] + - enum: [ pclk, i2sclk, x8k, x11k ] + allOf: + - $ref: /schemas/types.yaml#/definitions/string-array + + interrupts: + maxItems: 1 + + dmas: + items: + - description: audio capture DMA. + - description: audio playback DMA. + + dma-names: + items: + - enum: [ rx, tx ] + - enum: [ rx, tx ] + allOf: + - $ref: /schemas/types.yaml#/definitions/string-array + + resets: + maxItems: 1 + +required: + - compatible + - "#sound-dai-cells" + - reg + - clocks + - clock-names + - interrupts + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/stm32mp1-clks.h> + i2s2: audio-controller@4000b000 { + compatible = "st,stm32h7-i2s"; + #sound-dai-cells = <0>; + reg = <0x4000b000 0x400>; + clocks = <&rcc SPI2>, <&rcc SPI2_K>, <&rcc PLL3_Q>, <&rcc PLL3_R>; + clock-names = "pclk", "i2sclk", "x8k", "x11k"; + interrupts = <GIC_SPI 35 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&dmamux1 39 0x400 0x01>, + <&dmamux1 40 0x400 0x01>; + dma-names = "rx", "tx"; + pinctrl-names = "default"; + pinctrl-0 = <&i2s2_pins_a>; + }; + +... -- 2.17.1

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Re: [alsa-devel] [FFADO-user] Toneweal / Trigaudio FW66 device
by Takashi Sakamoto 04 Feb '20

04 Feb '20

Hi, I'm an author of ALSA bebob driver (snd-bebob). On Sun, Jan 19, 2020 at 03:01:00PM +0100, Daniel Jozsef wrote: > It's a Toneweal FW66, a 6x6 in/out 24bit 96kHz audio interface with MIDI. I > kinda like it for its build quality and the time I spent with it. :D The > largest chip in the device is labeled Trigaudio MNP-TA110. I searched for > information on this device, and found very little - it's a Taiwanese > company. There seems evidence that a "Trigaudio FW66" also exists, so the > device may have been sold under different brands in different markets. In my opinion, FW66 is an application of ArchWave BeBoB solution for audio and music units on IEEE 1394 bus (ArchWave is formerly known as BridgeCo) since I can find file structure for the solution in driver package shipped by the vendor. I guess that you can see large ASIC labelled with 'BridgeCo' (or 'ArchWave') DM1000/1100/1500 inner the device. If so, the device is possibly handled by implementation of FFADO. But as a quick glance to your log, the implementation of FFADO fails to detect it: > daniel@gibbonmoon:~$ sudo ffado-test Discover > ... > 02878614308: Warning (bebob_avdevice.cpp)[ 228] discover: Using generic BeBoB support for unsupported device 'ToneWeal FW66' > 02878620133: Debug (bebob_avdevice_subunit.cpp)[ 83] discover: Discovering BeBoB::AudioSubunit... > 02878620175: Debug (avc_audiosubunit.cpp)[ 56] discover: Discovering BeBoB::AudioSubunit... > 02878620196: Debug (avc_subunit.cpp)[ 108] discoverPlugs: Discovering plugs... > 02878778208: Error (bebob_avplug.cpp)[ 237] discoverPlugType: Plug does not implement extended plug info plug type info command > 02878778236: Error (bebob_avplug.cpp)[ 120] discover: discover: Could not discover plug type (1,1,0,0,1) > 02878778272: Error (avc_subunit.cpp)[ 189] discoverPlugs: plug discover failed > 02878778283: Error (avc_subunit.cpp)[ 131] discoverPlugs: destination plug discovering failed > 02878778301: Error (avc_subunit.cpp)[ 99] discover: plug discovery failed > 02878778312: Error (avc_unit.cpp)[ 283] enumerateSubUnits: enumerateSubUnits: Could not discover subunit_id = 0, subunit_type = 1 (Audio) > 02878778335: Error (avc_unit.cpp)[ 177] discover: Could not enumerate sub units > 02878778347: Error (bebob_avdevice.cpp)[ 232] discover: Could not discover unit > 02878778376: Error (devicemanager.cpp)[ 628] discover: could not discover device > 02878778419: Debug (devicemanager.cpp)[ 661] discover: Discovery finished... ... The reason is the unit returns NOT_IMPLEMENTED response against vendor specific AV/C command (Extended Plug Information command defined by BridgeCo.) for Audio subunit. I guess that the device has no Audio subunit but the implementation performs to use it without checking available subunits. For my information, would you please clone linux-firewire-utils[1] into your system and build it, then run below command to dump device information: $ ./firewire-request /dev/fw1 read 0xffffc8020000 60 result: 000: 62 72 69 64 67 65 43 6f 03 00 00 00 00 00 00 00 bridgeCo........ result: 010: 00 96 14 00 22 03 00 00 19 00 00 00 00 00 00 00 ...."........... result: 020: 32 30 30 39 30 36 30 39 31 36 32 39 34 30 20 20 20090609162940 result: 030: 00 00 00 19 07 1f 00 00 80 00 0c 40 70 35 14 00 ...........@p5.. result: 040: 32 30 30 39 30 36 30 39 31 36 32 32 31 39 20 20 20090609162219 result: 050: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ................ The above is a sample of Phonic Firefly 202. I expect the first line includes bytes represent 'bridgeCo'. [1] https://github.com/cladisch/linux-firewire-utils Regards Takashi Sakamoto

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[alsa-devel] [PATCH 11/17] ASoC: cros_ec_codec: Use cros_ec_send_cmd_msg()
by Prashant Malani 03 Feb '20

03 Feb '20

Replace send_ec_host_command() with cros_ec_send_cmd_msg() which does the same thing, but is defined in a common location in platform/chrome and exposed for other modules to use. This allows us to remove send_ec_host_command() entirely. Signed-off-by: Prashant Malani <pmalani(a)chromium.org> --- sound/soc/codecs/cros_ec_codec.c | 71 ++++++++++---------------------- 1 file changed, 21 insertions(+), 50 deletions(-) diff --git a/sound/soc/codecs/cros_ec_codec.c b/sound/soc/codecs/cros_ec_codec.c index 6a24f570c5e86f..49adb07d766963 100644 --- a/sound/soc/codecs/cros_ec_codec.c +++ b/sound/soc/codecs/cros_ec_codec.c @@ -69,38 +69,6 @@ static int ec_codec_capable(struct cros_ec_codec_priv *priv, uint8_t cap) return priv->ec_capabilities & BIT(cap); } -static int send_ec_host_command(struct cros_ec_device *ec_dev, uint32_t cmd, - uint8_t *out, size_t outsize, - uint8_t *in, size_t insize) -{ - int ret; - struct cros_ec_command *msg; - - msg = kmalloc(sizeof(*msg) + max(outsize, insize), GFP_KERNEL); - if (!msg) - return -ENOMEM; - - msg->version = 0; - msg->command = cmd; - msg->outsize = outsize; - msg->insize = insize; - - if (outsize) - memcpy(msg->data, out, outsize); - - ret = cros_ec_cmd_xfer_status(ec_dev, msg); - if (ret < 0) - goto error; - - if (insize) - memcpy(in, msg->data, insize); - - ret = 0; -error: - kfree(msg); - return ret; -} - static int calculate_sha256(struct cros_ec_codec_priv *priv, uint8_t *buf, uint32_t size, uint8_t *digest) { @@ -149,7 +117,7 @@ static int dmic_get_gain(struct snd_kcontrol *kcontrol, p.cmd = EC_CODEC_DMIC_GET_GAIN_IDX; p.get_gain_idx_param.channel = EC_CODEC_DMIC_CHANNEL_0; - ret = send_ec_host_command(priv->ec_device, EC_CMD_EC_CODEC_DMIC, + ret = cros_ec_send_cmd_msg(priv->ec_device, 0, EC_CMD_EC_CODEC_DMIC, (uint8_t *)&p, sizeof(p), (uint8_t *)&r, sizeof(r)); if (ret < 0) @@ -158,7 +126,7 @@ static int dmic_get_gain(struct snd_kcontrol *kcontrol, p.cmd = EC_CODEC_DMIC_GET_GAIN_IDX; p.get_gain_idx_param.channel = EC_CODEC_DMIC_CHANNEL_1; - ret = send_ec_host_command(priv->ec_device, EC_CMD_EC_CODEC_DMIC, + ret = cros_ec_send_cmd_msg(priv->ec_device, 0, EC_CMD_EC_CODEC_DMIC, (uint8_t *)&p, sizeof(p), (uint8_t *)&r, sizeof(r)); if (ret < 0) @@ -191,7 +159,7 @@ static int dmic_put_gain(struct snd_kcontrol *kcontrol, p.cmd = EC_CODEC_DMIC_SET_GAIN_IDX; p.set_gain_idx_param.channel = EC_CODEC_DMIC_CHANNEL_0; p.set_gain_idx_param.gain = left; - ret = send_ec_host_command(priv->ec_device, EC_CMD_EC_CODEC_DMIC, + ret = cros_ec_send_cmd_msg(priv->ec_device, 0, EC_CMD_EC_CODEC_DMIC, (uint8_t *)&p, sizeof(p), NULL, 0); if (ret < 0) return ret; @@ -199,7 +167,7 @@ static int dmic_put_gain(struct snd_kcontrol *kcontrol, p.cmd = EC_CODEC_DMIC_SET_GAIN_IDX; p.set_gain_idx_param.channel = EC_CODEC_DMIC_CHANNEL_1; p.set_gain_idx_param.gain = right; - return send_ec_host_command(priv->ec_device, EC_CMD_EC_CODEC_DMIC, + return cros_ec_send_cmd_msg(priv->ec_device, 0, EC_CMD_EC_CODEC_DMIC, (uint8_t *)&p, sizeof(p), NULL, 0); } @@ -231,7 +199,7 @@ static int dmic_probe(struct snd_soc_component *component) p.cmd = EC_CODEC_DMIC_GET_MAX_GAIN; - ret = send_ec_host_command(priv->ec_device, EC_CMD_EC_CODEC_DMIC, + ret = cros_ec_send_cmd_msg(priv->ec_device, 0, EC_CMD_EC_CODEC_DMIC, (uint8_t *)&p, sizeof(p), (uint8_t *)&r, sizeof(r)); if (ret < 0) { @@ -279,7 +247,7 @@ static int i2s_rx_hw_params(struct snd_pcm_substream *substream, p.cmd = EC_CODEC_I2S_RX_SET_SAMPLE_DEPTH; p.set_sample_depth_param.depth = depth; - ret = send_ec_host_command(priv->ec_device, EC_CMD_EC_CODEC_I2S_RX, + ret = cros_ec_send_cmd_msg(priv->ec_device, 0, EC_CMD_EC_CODEC_I2S_RX, (uint8_t *)&p, sizeof(p), NULL, 0); if (ret < 0) return ret; @@ -289,7 +257,7 @@ static int i2s_rx_hw_params(struct snd_pcm_substream *substream, p.cmd = EC_CODEC_I2S_RX_SET_BCLK; p.set_bclk_param.bclk = snd_soc_params_to_bclk(params); - return send_ec_host_command(priv->ec_device, EC_CMD_EC_CODEC_I2S_RX, + return cros_ec_send_cmd_msg(priv->ec_device, 0, EC_CMD_EC_CODEC_I2S_RX, (uint8_t *)&p, sizeof(p), NULL, 0); } @@ -333,7 +301,7 @@ static int i2s_rx_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) p.cmd = EC_CODEC_I2S_RX_SET_DAIFMT; p.set_daifmt_param.daifmt = daifmt; - return send_ec_host_command(priv->ec_device, EC_CMD_EC_CODEC_I2S_RX, + return cros_ec_send_cmd_msg(priv->ec_device, 0, EC_CMD_EC_CODEC_I2S_RX, (uint8_t *)&p, sizeof(p), NULL, 0); } @@ -364,7 +332,7 @@ static int i2s_rx_event(struct snd_soc_dapm_widget *w, return 0; } - return send_ec_host_command(priv->ec_device, EC_CMD_EC_CODEC_I2S_RX, + return cros_ec_send_cmd_msg(priv->ec_device, 0, EC_CMD_EC_CODEC_I2S_RX, (uint8_t *)&p, sizeof(p), NULL, 0); } @@ -415,7 +383,7 @@ static void *wov_map_shm(struct cros_ec_codec_priv *priv, p.cmd = EC_CODEC_GET_SHM_ADDR; p.get_shm_addr_param.shm_id = shm_id; - if (send_ec_host_command(priv->ec_device, EC_CMD_EC_CODEC, + if (cros_ec_send_cmd_msg(priv->ec_device, 0, EC_CMD_EC_CODEC, (uint8_t *)&p, sizeof(p), (uint8_t *)&r, sizeof(r)) < 0) { dev_err(priv->dev, "failed to EC_CODEC_GET_SHM_ADDR\n"); @@ -453,7 +421,7 @@ static void *wov_map_shm(struct cros_ec_codec_priv *priv, p.set_shm_addr_param.phys_addr = priv->ap_shm_last_alloc; p.set_shm_addr_param.len = req; p.set_shm_addr_param.shm_id = shm_id; - if (send_ec_host_command(priv->ec_device, EC_CMD_EC_CODEC, + if (cros_ec_send_cmd_msg(priv->ec_device, 0, EC_CMD_EC_CODEC, (uint8_t *)&p, sizeof(p), NULL, 0) < 0) { dev_err(priv->dev, "failed to EC_CODEC_SET_SHM_ADDR\n"); @@ -571,7 +539,7 @@ static int wov_read_audio_shm(struct cros_ec_codec_priv *priv) int ret; p.cmd = EC_CODEC_WOV_READ_AUDIO_SHM; - ret = send_ec_host_command(priv->ec_device, EC_CMD_EC_CODEC_WOV, + ret = cros_ec_send_cmd_msg(priv->ec_device, 0, EC_CMD_EC_CODEC_WOV, (uint8_t *)&p, sizeof(p), (uint8_t *)&r, sizeof(r)); if (ret) { @@ -596,7 +564,8 @@ static int wov_read_audio(struct cros_ec_codec_priv *priv) while (remain >= 0) { p.cmd = EC_CODEC_WOV_READ_AUDIO; - ret = send_ec_host_command(priv->ec_device, EC_CMD_EC_CODEC_WOV, + ret = cros_ec_send_cmd_msg(priv->ec_device, 0, + EC_CMD_EC_CODEC_WOV, (uint8_t *)&p, sizeof(p), (uint8_t *)&r, sizeof(r)); if (ret) { @@ -669,7 +638,8 @@ static int wov_enable_put(struct snd_kcontrol *kcontrol, else p.cmd = EC_CODEC_WOV_DISABLE; - ret = send_ec_host_command(priv->ec_device, EC_CMD_EC_CODEC_WOV, + ret = cros_ec_send_cmd_msg(priv->ec_device, 0, + EC_CMD_EC_CODEC_WOV, (uint8_t *)&p, sizeof(p), NULL, 0); if (ret) { dev_err(priv->dev, "failed to %s wov\n", @@ -716,7 +686,7 @@ static int wov_set_lang_shm(struct cros_ec_codec_priv *priv, p.cmd = EC_CODEC_WOV_SET_LANG_SHM; memcpy(pp->hash, digest, SHA256_DIGEST_SIZE); pp->total_len = size; - ret = send_ec_host_command(priv->ec_device, EC_CMD_EC_CODEC_WOV, + ret = cros_ec_send_cmd_msg(priv->ec_device, 0, EC_CMD_EC_CODEC_WOV, (uint8_t *)&p, sizeof(p), NULL, 0); if (ret) { dev_err(priv->dev, "failed to EC_CODEC_WOV_SET_LANG_SHM\n"); @@ -743,7 +713,8 @@ static int wov_set_lang(struct cros_ec_codec_priv *priv, pp->offset = i; memcpy(pp->buf, buf + i, req); pp->len = req; - ret = send_ec_host_command(priv->ec_device, EC_CMD_EC_CODEC_WOV, + ret = cros_ec_send_cmd_msg(priv->ec_device, 0, + EC_CMD_EC_CODEC_WOV, (uint8_t *)&p, sizeof(p), NULL, 0); if (ret) { dev_err(priv->dev, "failed to EC_CODEC_WOV_SET_LANG\n"); @@ -782,7 +753,7 @@ static int wov_hotword_model_put(struct snd_kcontrol *kcontrol, goto leave; p.cmd = EC_CODEC_WOV_GET_LANG; - ret = send_ec_host_command(priv->ec_device, EC_CMD_EC_CODEC_WOV, + ret = cros_ec_send_cmd_msg(priv->ec_device, 0, EC_CMD_EC_CODEC_WOV, (uint8_t *)&p, sizeof(p), (uint8_t *)&r, sizeof(r)); if (ret) @@ -1020,7 +991,7 @@ static int cros_ec_codec_platform_probe(struct platform_device *pdev) atomic_set(&priv->dmic_probed, 0); p.cmd = EC_CODEC_GET_CAPABILITIES; - ret = send_ec_host_command(priv->ec_device, EC_CMD_EC_CODEC, + ret = cros_ec_send_cmd_msg(priv->ec_device, 0, EC_CMD_EC_CODEC, (uint8_t *)&p, sizeof(p), (uint8_t *)&r, sizeof(r)); if (ret) { -- 2.25.0.341.g760bfbb309-goog

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[alsa-devel] [PATCH v5 00/17] soundwire: intel: implement new ASoC interfaces
by Pierre-Louis Bossart 03 Feb '20

03 Feb '20

This patchset applies on top of soundwire/next, now that the interface definitions are merged. The changes are essentially a removal of the platform devices, replaced by explicit device(s)/driver for the SoundWire Master(s), and the implementation of the new interfaces required to scan the ACPI tables, probe the links and start them. The missing prepare, trigger and setup ASoC callbacks are also implemented. The hw_params and free callbacks use the new interfaces as well. While there are quite a few lines of code changed, this is mostly about interface changes. The next series will contain more functional changes and deal with race conditions on probe, enumeration and suspend/resume issues. Thanks to Vinod Kould and GregKH for the reviews on previous versions, much appreciated. Changes since v4: (feedback from GregKH except last point flagged by Intel validation) Clarified error handling with uevents Added better commit messages and rationale behind exposing a Master Device. Used 'master_device' instead of 'md' when possible (kept the shortcut with really long function names) Added namespace check support for Intel code (the Cadence and core stuff should be done in a separate patchset). Fixed GPLv2 license/EXPORT_SYMBOL_GPL error for the Master Device code. Fixed missing error handling in sdw_md_add Added kerneldoc comments for sdw_md_driver structure (with explanation on what 'md' stands for) Fixed NULL pointer assignment leading to issues with driver_unregister Changes since v3: One line change to re-add EXPORT_SYMBOL Add missing driver_registration Changes since v2: moved uevent handling to slave_type (Vinod) Changes since v1: fix typo (Vinod) removed uevent open for Master (Vinod) clarified commit messages (Cezary) no functionality change Bard Liao (1): soundwire: register master device driver Pierre-Louis Bossart (13): soundwire: renames to prepare support for master drivers/devices soundwire: rename dev_to_sdw_dev macro soundwire: rename drv_to_sdw_slave_driver macro soundwire: bus_type: rename sdw_drv_ to sdw_slave_drv soundwire: intel: rename res field as link_res soundwire: add support for sdw_slave_type soundwire: slave: move uevent handling to slave device level soundwire: add initial definitions for sdw_master_device soundwire: intel: remove platform devices and use 'Master Devices' instead soundwire: intel: free all resources on hw_free() soundwire: intel_init: add implementation of sdw_intel_enable_irq() soundwire: intel_init: use EXPORT_SYMBOL_NS soundwire: intel: use EXPORT_SYMBOL_NS 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/base/regmap/regmap-sdw.c | 4 +- drivers/soundwire/Makefile | 2 +- drivers/soundwire/bus.c | 2 +- drivers/soundwire/bus.h | 2 + drivers/soundwire/bus_type.c | 70 ++++--- drivers/soundwire/intel.c | 281 +++++++++++++++++++++----- drivers/soundwire/intel.h | 8 +- drivers/soundwire/intel_init.c | 312 ++++++++++++++++++++++------- drivers/soundwire/master.c | 64 ++++++ drivers/soundwire/slave.c | 10 +- include/linux/soundwire/sdw.h | 42 +++- include/linux/soundwire/sdw_type.h | 34 +++- 12 files changed, 668 insertions(+), 163 deletions(-) create mode 100644 drivers/soundwire/master.c -- 2.20.1

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[alsa-devel] [PATCH v3 0/4] ALSA: hda - Add DP-MST support for NVIDIA codecs
by Nikhil Mahale 03 Feb '20

03 Feb '20

Today, DisplayPort multi-stream transport (DP-MST) audio is not supported on codec drivers which don't use the audio component (acomp) binding. The commit ade49db337a9 (ALSA: hda/hdmi - Allow audio component for AMD/ATI and Nvidia HDMI) added the DRM audio component binding for the NVIDIA codec drivers, but the traditional HD-audio unsolicited event code path is still in use by the proprietary NVIDIA graphics driver. This patch set adds DP-MST audio support for non-acomp codec drivers which use the traditional HD-audio unsolicited event code path. This patch set adds DP-MST support for NVIDIA codecs. The patch set has been tested for HDMI/DP-SST/DP-MST(4 dp-mst audio streams) configurations on NVIDIA Kepler and Maxwell GPUs, using both the nouveau driver and the proprietary NVIDIA graphics driver. Nikhil Mahale (4): ALSA: hda - Rename snd_hda_pin_sense to snd_hda_jack_pin_sense ALSA: hda - Add DP-MST jack support ALSA: hda - Add DP-MST support for non-acomp codecs ALSA: hda - Add DP-MST support for NVIDIA codecs sound/pci/hda/hda_jack.c | 151 ++++++++++++++++++-------- sound/pci/hda/hda_jack.h | 107 ++++++++++++++++-- sound/pci/hda/patch_hdmi.c | 263 +++++++++++++++++++++++++++++++-------------- 3 files changed, 385 insertions(+), 136 deletions(-) -- 2.16.4

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[alsa-devel] KASAN: use-after-free Read in snd_timer_resolution
by syzbot 02 Feb '20

02 Feb '20

Hello, syzbot found the following crash on: HEAD commit: e033e7d4 Merge branch 'dhowells' (patches from DavidH) git tree: upstream console output: https://syzkaller.appspot.com/x/log.txt?x=12f2bb25e00000 kernel config: https://syzkaller.appspot.com/x/.config?x=d9290aeb7e6cf1c4 dashboard link: https://syzkaller.appspot.com/bug?extid=2b2ef983f973e5c40943 compiler: gcc (GCC) 9.0.0 20181231 (experimental) userspace arch: i386 Unfortunately, I don't have any reproducer for this crash yet. IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+2b2ef983f973e5c40943(a)syzkaller.appspotmail.com ================================================================== BUG: KASAN: use-after-free in snd_timer_resolution+0xf1/0x110 sound/core/timer.c:441 Read of size 8 at addr ffff888094155800 by task syz-executor.0/18632 CPU: 1 PID: 18632 Comm: syz-executor.0 Not tainted 5.5.0-rc6-syzkaller #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0x197/0x210 lib/dump_stack.c:118 print_address_description.constprop.0.cold+0xd4/0x30b mm/kasan/report.c:374 __kasan_report.cold+0x1b/0x41 mm/kasan/report.c:506 kasan_report+0x12/0x20 mm/kasan/common.c:639 __asan_report_load8_noabort+0x14/0x20 mm/kasan/generic_report.c:135 snd_timer_resolution+0xf1/0x110 sound/core/timer.c:441 snd_seq_info_timer_read+0x95/0x2f1 sound/core/seq/seq_timer.c:480 snd_info_seq_show+0xcb/0x120 sound/core/info.c:363 seq_read+0x4ca/0x1170 fs/seq_file.c:229 proc_reg_read+0x1fc/0x2c0 fs/proc/inode.c:223 do_loop_readv_writev fs/read_write.c:714 [inline] do_loop_readv_writev fs/read_write.c:701 [inline] do_iter_read+0x4a4/0x660 fs/read_write.c:935 compat_readv+0x187/0x1f0 fs/read_write.c:1186 do_compat_preadv64+0x190/0x1c0 fs/read_write.c:1235 __do_compat_sys_preadv fs/read_write.c:1255 [inline] __se_compat_sys_preadv fs/read_write.c:1249 [inline] __ia32_compat_sys_preadv+0xc7/0x140 fs/read_write.c:1249 do_syscall_32_irqs_on arch/x86/entry/common.c:337 [inline] do_fast_syscall_32+0x27b/0xe16 arch/x86/entry/common.c:408 entry_SYSENTER_compat+0x70/0x7f arch/x86/entry/entry_64_compat.S:139 RIP: 0023:0xf7f6da39 Code: 00 00 00 89 d3 5b 5e 5f 5d c3 b8 80 96 98 00 eb c4 8b 04 24 c3 8b 1c 24 c3 8b 34 24 c3 8b 3c 24 c3 51 52 55 89 e5 0f 34 cd 80 <5d> 5a 59 c3 90 90 90 90 eb 0d 90 90 90 90 90 90 90 90 90 90 90 90 RSP: 002b:00000000f5d690cc EFLAGS: 00000296 ORIG_RAX: 000000000000014d RAX: ffffffffffffffda RBX: 0000000000000004 RCX: 00000000200017c0 RDX: 00000000000003da RSI: 0000000000000000 RDI: 0000000000000000 RBP: 0000000000000000 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000000 R12: 0000000000000000 R13: 0000000000000000 R14: 0000000000000000 R15: 0000000000000000 Allocated by task 18631: save_stack+0x23/0x90 mm/kasan/common.c:72 set_track mm/kasan/common.c:80 [inline] __kasan_kmalloc mm/kasan/common.c:513 [inline] __kasan_kmalloc.constprop.0+0xcf/0xe0 mm/kasan/common.c:486 kasan_kmalloc+0x9/0x10 mm/kasan/common.c:527 kmem_cache_alloc_trace+0x158/0x790 mm/slab.c:3551 kmalloc include/linux/slab.h:556 [inline] kzalloc include/linux/slab.h:670 [inline] snd_timer_instance_new+0x4a/0x300 sound/core/timer.c:96 snd_seq_timer_open+0x1c0/0x590 sound/core/seq/seq_timer.c:275 queue_use+0xf1/0x270 sound/core/seq/seq_queue.c:489 snd_seq_queue_alloc+0x2c5/0x4d0 sound/core/seq/seq_queue.c:176 snd_seq_ioctl_create_queue+0xb0/0x330 sound/core/seq/seq_clientmgr.c:1548 snd_seq_kernel_client_ctl+0xf8/0x140 sound/core/seq/seq_clientmgr.c:2353 alloc_seq_queue.isra.0+0xdc/0x180 sound/core/seq/oss/seq_oss_init.c:357 snd_seq_oss_open+0x2ff/0x960 sound/core/seq/oss/seq_oss_init.c:215 odev_open+0x70/0x90 sound/core/seq/oss/seq_oss.c:125 soundcore_open+0x453/0x610 sound/sound_core.c:593 chrdev_open+0x245/0x6b0 fs/char_dev.c:414 do_dentry_open+0x4e6/0x1380 fs/open.c:797 vfs_open+0xa0/0xd0 fs/open.c:914 do_last fs/namei.c:3420 [inline] path_openat+0x10df/0x4500 fs/namei.c:3537 do_filp_open+0x1a1/0x280 fs/namei.c:3567 do_sys_open+0x3fe/0x5d0 fs/open.c:1097 __do_compat_sys_openat fs/open.c:1143 [inline] __se_compat_sys_openat fs/open.c:1141 [inline] __ia32_compat_sys_openat+0x98/0xf0 fs/open.c:1141 do_syscall_32_irqs_on arch/x86/entry/common.c:337 [inline] do_fast_syscall_32+0x27b/0xe16 arch/x86/entry/common.c:408 entry_SYSENTER_compat+0x70/0x7f arch/x86/entry/entry_64_compat.S:139 Freed by task 18630: save_stack+0x23/0x90 mm/kasan/common.c:72 set_track mm/kasan/common.c:80 [inline] kasan_set_free_info mm/kasan/common.c:335 [inline] __kasan_slab_free+0x102/0x150 mm/kasan/common.c:474 kasan_slab_free+0xe/0x10 mm/kasan/common.c:483 __cache_free mm/slab.c:3426 [inline] kfree+0x10a/0x2c0 mm/slab.c:3757 snd_timer_instance_free sound/core/timer.c:120 [inline] snd_timer_instance_free+0x7c/0xa0 sound/core/timer.c:114 snd_seq_timer_close+0x99/0xe0 sound/core/seq/seq_timer.c:319 queue_delete+0x52/0xb0 sound/core/seq/seq_queue.c:134 snd_seq_queue_delete+0x4e/0x70 sound/core/seq/seq_queue.c:196 snd_seq_ioctl_delete_queue+0x6a/0x90 sound/core/seq/seq_clientmgr.c:1570 snd_seq_kernel_client_ctl+0xf8/0x140 sound/core/seq/seq_clientmgr.c:2353 delete_seq_queue.part.0+0xb6/0x120 sound/core/seq/oss/seq_oss_init.c:376 delete_seq_queue sound/core/seq/oss/seq_oss_init.c:372 [inline] snd_seq_oss_release+0x116/0x150 sound/core/seq/oss/seq_oss_init.c:421 odev_release+0x54/0x80 sound/core/seq/oss/seq_oss.c:140 __fput+0x2ff/0x890 fs/file_table.c:280 ____fput+0x16/0x20 fs/file_table.c:313 task_work_run+0x145/0x1c0 kernel/task_work.c:113 tracehook_notify_resume include/linux/tracehook.h:188 [inline] exit_to_usermode_loop+0x316/0x380 arch/x86/entry/common.c:164 prepare_exit_to_usermode arch/x86/entry/common.c:195 [inline] syscall_return_slowpath arch/x86/entry/common.c:278 [inline] do_syscall_32_irqs_on arch/x86/entry/common.c:352 [inline] do_fast_syscall_32+0xbbd/0xe16 arch/x86/entry/common.c:408 entry_SYSENTER_compat+0x70/0x7f arch/x86/entry/entry_64_compat.S:139 The buggy address belongs to the object at ffff888094155800 which belongs to the cache kmalloc-256 of size 256 The buggy address is located 0 bytes inside of 256-byte region [ffff888094155800, ffff888094155900) The buggy address belongs to the page: page:ffffea0002505540 refcount:1 mapcount:0 mapping:ffff8880aa4008c0 index:0x0 raw: 00fffe0000000200 ffffea00024df348 ffffea00024af508 ffff8880aa4008c0 raw: 0000000000000000 ffff888094155000 0000000100000008 0000000000000000 page dumped because: kasan: bad access detected Memory state around the buggy address: ffff888094155700: fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc ffff888094155780: fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc > ffff888094155800: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ^ ffff888094155880: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ffff888094155900: fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc ================================================================== --- This bug is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot.

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