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

alsa-gobject: misc fixes and documentation for system calls
by GitHub pull_request - opened 13 Jun '20

13 Jun '20

alsa-project/alsa-gobject pull request #34 was opened from takaswie: This patchset includes misc fixes and documentation for system calls for each methods. Request URL : https://github.com/alsa-project/alsa-gobject/pull/34 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/34.patch Repository URL: https://github.com/alsa-project/alsa-gobject

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[PATCH] ASoC: fsl_ssi: Fix bclk calculation for mono channel
by Shengjiu Wang 12 Jun '20

12 Jun '20

For mono channel, ssi will switch to normal mode. In normal mode, the Word Length Control bits control the word length divider in clock generator, which is different with I2S master mode, the word length is fixed to 32bit. So we refine the famula for mono channel, otherwise there will be sound issue for S24_LE. Fixes: b0a7043d5c2c ("ASoC: fsl_ssi: Caculate bit clock rate using slot number and width") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_ssi.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index bad89b0d129e..e347776590f7 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -695,6 +695,11 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, /* Generate bit clock based on the slot number and slot width */ freq = slots * slot_width * params_rate(hw_params); + /* The slot_width is not fixed to 32 for normal mode */ + if (params_channels(hw_params) == 1) + freq = (slots <= 1 ? 2 : slots) * params_width(hw_params) * + params_rate(hw_params); + /* Don't apply it to any non-baudclk circumstance */ if (IS_ERR(ssi->baudclk)) return -EINVAL; -- 2.21.0

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[alsa-devel] [PATCH 0/2 v4] ALSA: pcm: anonymous dup implementation
by Jaroslav Kysela 12 Jun '20

12 Jun '20

This patchset contains the anonymous dup implementation with permissions checking for the ALSA's PCM interface in kernel to enable the restricted DMA sound buffer sharing for the restricted tasks. The code was tested through qemu and it seems to be pretty stable. The initial tinyalsa implementation can be found here: https://github.com/perexg/tinyalsa/commits/anondup v2 of the patches: - change clone parameter to subdevice number for the pcm attach - change SNDRV_PCM_PERM_MAX to SNDRV_PCM_PERM_MASK - the tinyalsa implementation was a little updated (restructured) v3 of the patches: - group integer declarations in snd_pcm_anonymous_dup() - replaced substream->pcm with pcm in snd_pcm_anonymous_dup() - added SNDRV_PCM_PERM_RW check for read/write/readv/writev syscalls v4 of the patches: - more simple restriction control (only two modes - full/buffer) - the tinyalsa implementation follows this change Cc: Phil Burk <philburk(a)google.com> Cc: Zach Riggle <riggle(a)google.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Leo Yan <leo.yan(a)linaro.org> Cc: Baolin Wang <baolin.wang(a)linaro.org> Jaroslav Kysela (2): ALSA: pcm: implement the anonymous dup (inode file descriptor) ALSA: pcm: implement the mmap buffer mode for the anonymous dup include/sound/pcm.h | 10 +++-- include/uapi/sound/asound.h | 6 ++- sound/core/oss/pcm_oss.c | 2 +- sound/core/pcm.c | 13 +++--- sound/core/pcm_compat.c | 1 + sound/core/pcm_native.c | 97 ++++++++++++++++++++++++++++++++++++++++++--- 6 files changed, 110 insertions(+), 19 deletions(-) -- 2.13.6

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[PATCH v3 1/2] dt-bindings: tas2562: Convert the tas2562 binding to yaml
by Dan Murphy 12 Jun '20

12 Jun '20

Convert the TAS2562 text file to yaml format. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- .../devicetree/bindings/sound/tas2562.txt | 34 ---------- .../devicetree/bindings/sound/tas2562.yaml | 65 +++++++++++++++++++ 2 files changed, 65 insertions(+), 34 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/tas2562.txt create mode 100644 Documentation/devicetree/bindings/sound/tas2562.yaml diff --git a/Documentation/devicetree/bindings/sound/tas2562.txt b/Documentation/devicetree/bindings/sound/tas2562.txt deleted file mode 100644 index 94796b547184..000000000000 --- a/Documentation/devicetree/bindings/sound/tas2562.txt +++ /dev/null @@ -1,34 +0,0 @@ -Texas Instruments TAS2562 Smart PA - -The TAS2562 is a mono, digital input Class-D audio amplifier optimized for -efficiently driving high peak power into small loudspeakers. -Integrated speaker voltage and current sense provides for -real time monitoring of loudspeaker behavior. - -Required properties: - - #address-cells - Should be <1>. - - #size-cells - Should be <0>. - - compatible: - Should contain "ti,tas2562", "ti,tas2563". - - reg: - The i2c address. Should be 0x4c, 0x4d, 0x4e or 0x4f. - - ti,imon-slot-no:- TDM TX current sense time slot. - -Optional properties: -- interrupt-parent: phandle to the interrupt controller which provides - the interrupt. -- interrupts: (GPIO) interrupt to which the chip is connected. -- shut-down: GPIO used to control the state of the device. - -Examples: -tas2562@4c { - #address-cells = <1>; - #size-cells = <0>; - compatible = "ti,tas2562"; - reg = <0x4c>; - - interrupt-parent = <&gpio1>; - interrupts = <14>; - - shut-down = <&gpio1 15 0>; - ti,imon-slot-no = <0>; -}; - diff --git a/Documentation/devicetree/bindings/sound/tas2562.yaml b/Documentation/devicetree/bindings/sound/tas2562.yaml new file mode 100644 index 000000000000..bb51fd88c41a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2562.yaml @@ -0,0 +1,65 @@ +# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause) +# Copyright (C) 2019 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: "http://devicetree.org/schemas/sound/tas2562.yaml#" +$schema: "http://devicetree.org/meta-schemas/core.yaml#" + +title: Texas Instruments TAS2562 Smart PA + +maintainers: + - Dan Murphy <dmurphy(a)ti.com> + +description: | + The TAS2562 is a mono, digital input Class-D audio amplifier optimized for + efficiently driving high peak power into small loudspeakers. + Integrated speaker voltage and current sense provides for + real time monitoring of loudspeaker behavior. + +properties: + compatible: + enum: + - ti,tas2562 + - ti,tas2563 + + reg: + maxItems: 1 + description: | + I2C address of the device can be one of these 0x4c, 0x4d, 0x4e or 0x4f + + shutdown-gpio: + description: GPIO used to control the state of the device. + + interrupts: + maxItems: 1 + + ti,imon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + description: TDM TX current sense time slot. + + '#sound-dai-cells': + const: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c0 { + #address-cells = <1>; + #size-cells = <0>; + codec: codec@4c { + compatible = "ti,tas2562"; + reg = <0x4c>; + #sound-dai-cells = <1>; + interrupt-parent = <&gpio1>; + interrupts = <14>; + shutdown-gpio = <&gpio1 15 0>; + ti,imon-slot-no = <0>; + }; + }; + -- 2.26.2

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[PATCH] ASoC: rt5682: Register clocks even when mclk is NULL
by Akshu Agrawal 12 Jun '20

12 Jun '20

Fixes kernel crash on platforms which do not have mclk exposed in CCF framework. For these platforms have mclk as NULL and continue to register clocks. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/codecs/rt5682.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index d3245123101d..7c8b8d839db8 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -2829,12 +2829,11 @@ static int rt5682_probe(struct snd_soc_component *component) return ret; } rt5682->mclk = NULL; - } else { - /* Register CCF DAI clock control */ - ret = rt5682_register_dai_clks(component); - if (ret) - return ret; } + /* Register CCF DAI clock control */ + ret = rt5682_register_dai_clks(component); + if (ret) + return ret; /* Initial setup for CCF */ rt5682->lrck[RT5682_AIF1] = CLK_48; #endif -- 2.20.1

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[PATCH v2 0/6] Enable Greybus Audio codec driver
by Vaibhav Agarwal 12 Jun '20

12 Jun '20

The existing GB Audio codec driver is dependent on MSM8994 Audio driver. During the development stage, this dependency was configured due to various changes involved in MSM Audio driver to enable addtional codec card and some of the changes proposed in mainline ASoC framework. However, these are not the real dependencies and some of them can be easily removed. The folowing patch series includes the changes to resolve unnecessary depedencies and make the codec driver functional with the latest kernel. Patch 1,2: Incudes jack framework related changes. Patch 3,4,5: Resolves compilation error observed with the latest kernel and also provides helper APIs required to allow dynamic addition/removal of modules. Patch 6: Finally provides config options and related Makefile changes to enable GB Codec driver. Thanks to Alexandre for raising the headsup [1] and motivating me to provide the necessary changes. [1] https://lore.kernel.org/lkml/20200507212912.599433-1-alexandre.belloni@boot… Changes from v1 - Include the changes for the review comments suggested by Dan - Rebase to latest staging-next Vaibhav Agarwal (6): staging: greybus: audio: Update snd_jack FW usage as per new APIs staging: greybus: audio: Maintain jack list within GB Audio module staging: greybus: audio: Resolve compilation errors for GB codec module staging: greybus: audio: Resolve compilation error in topology parser staging: greybus: audio: Add helper APIs for dynamic audio modules staging: greybus: audio: Enable GB codec, audio module compilation. drivers/staging/greybus/Kconfig | 14 +- drivers/staging/greybus/Makefile | 6 +- drivers/staging/greybus/audio_codec.c | 178 +++++++++++--------- drivers/staging/greybus/audio_codec.h | 12 +- drivers/staging/greybus/audio_helper.c | 197 +++++++++++++++++++++++ drivers/staging/greybus/audio_helper.h | 17 ++ drivers/staging/greybus/audio_module.c | 15 +- drivers/staging/greybus/audio_topology.c | 128 +++++++-------- 8 files changed, 412 insertions(+), 155 deletions(-) create mode 100644 drivers/staging/greybus/audio_helper.c create mode 100644 drivers/staging/greybus/audio_helper.h base-commit: af7b4801030c07637840191c69eb666917e4135d -- 2.26.2

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[RFC PATCH v3 0/4] Reuse the dma channel if available in Back-End
by Shengjiu Wang 12 Jun '20

12 Jun '20

Reuse the dma channel if available in Back-End Shengjiu Wang (4): ASoC: soc-card: export snd_soc_lookup_component_nolocked ASoC: dmaengine_pcm: export soc_component_to_pcm ASoC: fsl_asrc_dma: Reuse the dma channel if available in Back-End ASoC: fsl_asrc_dma: Fix data copying speed issue with EDMA changes in v3: - update according to Nicolin's comments - split previous 0003 patch to two patches changes in v2: - update according to Mark's comments and split the patch include/sound/dmaengine_pcm.h | 11 +++++++ include/sound/soc.h | 2 ++ sound/soc/fsl/fsl_asrc_common.h | 2 ++ sound/soc/fsl/fsl_asrc_dma.c | 47 +++++++++++++++++++-------- sound/soc/soc-core.c | 3 +- sound/soc/soc-generic-dmaengine-pcm.c | 12 ------- 6 files changed, 50 insertions(+), 27 deletions(-) -- 2.21.0

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alsa-gobject: timer: user_instance: add API to choose the type of event to read
by GitHub pull_request - opened 12 Jun '20

12 Jun '20

alsa-project/alsa-gobject pull request #33 was opened from takaswie: Currently ALSATimer.UserInstance has any attach API to decide the type of event to read. In design of ALSA Timer core, the attach API has two functionality; to attach or to detach with error for absent timer device. Current implementation has inconvenient for the functionality due to execution to choose the type of event in advance. The execution fails when the instance is already attached. This commit adds a new API to choose the type of event and split the call from any attach API. Request URL : https://github.com/alsa-project/alsa-gobject/pull/33 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/33.patch Repository URL: https://github.com/alsa-project/alsa-gobject

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[PATCH] Add implicit feedback quirk for SSL2+.
by Laurence Tratt 12 Jun '20

12 Jun '20

This uses the same quirk as the Motu M2 and M4 to ensure the driver uses the audio interface's clock. Tested on an SSL2+. Signed-off-by: Laurence Tratt <laurie(a)tratt.net> --- sound/usb/pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git sound/usb/pcm.c sound/usb/pcm.c index 8a05dcb13..84c0ae431 100644 --- sound/usb/pcm.c +++ sound/usb/pcm.c @@ -367,6 +367,7 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, ifnum = 0; goto add_sync_ep_from_ifnum; case USB_ID(0x07fd, 0x0008): /* MOTU M Series */ + case USB_ID(0x31e9, 0x0002): /* Solid State Logic SSL2+ */ ep = 0x81; ifnum = 2; goto add_sync_ep_from_ifnum; -- 2.27.0

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Re: [PATCH v2] ASoC: SOF: Intel: hda: unsolicited RIRB response
by Kai Vehmanen 12 Jun '20

12 Jun '20

Hi Brent, On Fri, 12 Jun 2020, Brent Lu wrote: > Port commit 6d011d5057ff ("ALSA: hda: Clear RIRB status before reading > WP") from legacy HDA driver to fix the get response timeout issue. > Current SOF driver does not suffer from this issue because sync write > is enabled in hda_init. The issue will come back if the sync write is > disabled for some reason. better to align the logic, so I'm ok to take this patch to SOF. Can you fix the summary though: - ASoC: SOF: Intel: hda: unsolicited RIRB response + ASoC: SOF: Intel: hda: Clear RIRB status before reading WP Br, Kai

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