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[PATCH v8] ASoC: Intel: kbl_rt5663_rt5514_max98927: Fix kabylake_ssp_fixup function
by Harsha Priya 23 Jul '20

23 Jul '20

kabylake_ssp_fixup function uses snd_soc_dpcm to identify the codecs DAIs. The HW parameters are changed based on the codec DAI of the stream. The earlier approach to get snd_soc_dpcm was using container_of() macro on snd_pcm_hw_params. The structures have been modified over time and snd_soc_dpcm does not have snd_pcm_hw_params as a reference but as a copy. This causes the current driver to crash when used. This patch changes the way snd_soc_dpcm is extracted. snd_soc_pcm_runtime holds 2 dpcm instances (one for playback and one for capture). 2 codecs on the SSP are dmic (capture) and speakers (playback). Based on the stream direction, snd_soc_dpcm is extracted from snd_soc_pcm_runtime. Tested for all use cases of the driver. Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> Signed-off-by: Vamshi Krishna Gopal <vamshi.krishna.gopal(a)intel.com> Tested-by: Lukasz Majczak <lma(a)semihalf.com> --- v1 -> v2: - Extract dmic from SSP0 as every BE should have own fixup function. v2 -> v3: - Restore naming in the dapm route table to not confuse with other drivers - Fixed indentations v3 -> v4: - Updated code and commit description according to solution proposed by Harsha v4 -> v5: - Cosmetic Changes v5 -> v6: - Dmic regression seen with v4 fixed - Using available routines for obtaining dpcm information v6 -> v7: - Updated comments - initilize rtd_dpcm variable - added break statement in the loop v7 -> v8: - Updated commit message - return on rtd_dpcm being NULL - typo fixed --- --- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 41 +++++++++++++++---- 1 file changed, 32 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 584e4f9cedc2..21808fe13481 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -379,22 +379,45 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_interval *chan = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - struct snd_soc_dpcm *dpcm = container_of( - params, struct snd_soc_dpcm, hw_params); - struct snd_soc_dai_link *fe_dai_link = dpcm->fe->dai_link; - struct snd_soc_dai_link *be_dai_link = dpcm->be->dai_link; + struct snd_soc_dpcm *dpcm, *rtd_dpcm = NULL; + + /* + * The following loop will be called only for playback stream + * In this platform, there is only one playback device on every SSP + */ + for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_PLAYBACK, dpcm) { + rtd_dpcm = dpcm; + break; + } + + /* + * This following loop will be called only for capture stream + * In this platform, there is only one capture device on every SSP + */ + for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_CAPTURE, dpcm) { + rtd_dpcm = dpcm; + break; + } + + if (!rtd_dpcm) + return -EINVAL; + + /* + * The above 2 loops are mutually exclusive based on the stream direction, + * thus rtd_dpcm variable will never be overwritten + */ /* * The ADSP will convert the FE rate to 48k, stereo, 24 bit */ - if (!strcmp(fe_dai_link->name, "Kbl Audio Port") || - !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") || - !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) { + if (!strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Port") || + !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Headset Playback") || + !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Capture Port")) { rate->min = rate->max = 48000; chan->min = chan->max = 2; snd_mask_none(fmt); snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); - } else if (!strcmp(fe_dai_link->name, "Kbl Audio DMIC cap")) { + } else if (!strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio DMIC cap")) { if (params_channels(params) == 2 || DMIC_CH(dmic_constraints) == 2) chan->min = chan->max = 2; @@ -405,7 +428,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, * The speaker on the SSP0 supports S16_LE and not S24_LE. * thus changing the mask here */ - if (!strcmp(be_dai_link->name, "SSP0-Codec")) + if (!strcmp(rtd_dpcm->be->dai_link->name, "SSP0-Codec")) snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); return 0; -- 2.17.1

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[PATCH 0/4] ASoC: fsl_asrc: allow selecting arbitrary clocks
by Arnaud Ferraris 23 Jul '20

23 Jul '20

The current ASRC driver hardcodes the input and output clocks used for sample rate conversions. In order to allow greater flexibility and to cover more use cases, it would be preferable to select the clocks using device-tree properties. This series also fix register configuration and clock assignment so conversion can be conducted effectively in both directions with a good quality. Arnaud Ferraris (4): dt-bindings: sound: fsl,asrc: add properties to select in/out clocks ASoC: fsl_asrc: allow using arbitrary input and output clocks ASoC: fsl_asrc: always use ratio for conversion ASoC: fsl_asrc: swap input and output clocks in capture mode Documentation/devicetree/bindings/sound/fsl,asrc.txt | 8 ++++++++ sound/soc/fsl/fsl_asrc.c | 69 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++----- sound/soc/fsl/fsl_asrc_common.h | 3 +++ 3 files changed, 75 insertions(+), 5 deletions(-)

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Re: linux-next: Tree for Jul 22 (sound/soc/tegra/snd-soc-tegra210-dmic)
by Randy Dunlap 22 Jul '20

22 Jul '20

On 7/22/20 6:16 AM, Stephen Rothwell wrote: > Hi all, > > Changes since 20200721: > on i386: ERROR: modpost: "__udivdi3" [sound/soc/tegra/snd-soc-tegra210-dmic.ko] undefined! ERROR: modpost: "__divdi3" [sound/soc/tegra/snd-soc-tegra210-dmic.ko] undefined! -- ~Randy Reported-by: Randy Dunlap <rdunlap(a)infradead.org>

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[PATCH] dt-bindings: sound: convert ROHM BD28623 amplifier binding to yaml
by Katsuhiro Suzuki 22 Jul '20

22 Jul '20

This patch converts ROHM BD28623UMV class D speaker amplifier binding to DT schema. Signed-off-by: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> --- .../bindings/sound/rohm,bd28623.txt | 29 --------- .../bindings/sound/rohm,bd28623.yaml | 65 +++++++++++++++++++ 2 files changed, 65 insertions(+), 29 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/rohm,bd28623.txt create mode 100644 Documentation/devicetree/bindings/sound/rohm,bd28623.yaml diff --git a/Documentation/devicetree/bindings/sound/rohm,bd28623.txt b/Documentation/devicetree/bindings/sound/rohm,bd28623.txt deleted file mode 100644 index d84557c2686e..000000000000 --- a/Documentation/devicetree/bindings/sound/rohm,bd28623.txt +++ /dev/null @@ -1,29 +0,0 @@ -ROHM BD28623MUV Class D speaker amplifier for digital input - -This codec does not have any control buses such as I2C, it detect format and -rate of I2S signal automatically. It has two signals that can be connected -to GPIOs: reset and mute. - -Required properties: -- compatible : should be "rohm,bd28623" -- #sound-dai-cells: should be 0. -- VCCA-supply : regulator phandle for the VCCA supply -- VCCP1-supply : regulator phandle for the VCCP1 supply -- VCCP2-supply : regulator phandle for the VCCP2 supply - -Optional properties: -- reset-gpios : GPIO specifier for the active low reset line -- mute-gpios : GPIO specifier for the active low mute line - -Example: - - codec { - compatible = "rohm,bd28623"; - #sound-dai-cells = <0>; - - VCCA-supply = <&vcc_reg>; - VCCP1-supply = <&vcc_reg>; - VCCP2-supply = <&vcc_reg>; - reset-gpios = <&gpio 0 GPIO_ACTIVE_LOW>; - mute-gpios = <&gpio 1 GPIO_ACTIVE_LOW>; - }; diff --git a/Documentation/devicetree/bindings/sound/rohm,bd28623.yaml b/Documentation/devicetree/bindings/sound/rohm,bd28623.yaml new file mode 100644 index 000000000000..acd8609252b4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rohm,bd28623.yaml @@ -0,0 +1,65 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rohm,bd28623.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: ROHM BD28623MUV Class D speaker amplifier for digital input + +description: + This codec does not have any control buses such as I2C, it detect + format and rate of I2S signal automatically. It has two signals + that can be connected to GPIOs reset and mute. + +maintainers: + - Katsuhiro Suzuki <katsuhiro(a)katsuster.net> + +properties: + compatible: + const: rohm,bd28623 + + "#sound-dai-cells": + const: 0 + + VCCA-supply: + description: + regulator phandle for the VCCA (for analog) power supply + + VCCP1-supply: + description: + regulator phandle for the VCCP1 (for ch1) power supply + + VCCP2-supply: + description: + regulator phandle for the VCCP2 (for ch2) power supply + + reset-gpios: + maxItems: 1 + description: + GPIO specifier for the active low reset line + + mute-gpios: + maxItems: 1 + description: + GPIO specifier for the active low mute line + +required: + - compatible + - VCCA-supply + - VCCP1-supply + - VCCP2-supply + - "#sound-dai-cells" + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + codec { + compatible = "rohm,bd28623"; + #sound-dai-cells = <0>; + + VCCA-supply = <&vcc_reg>; + VCCP1-supply = <&vcc_reg>; + VCCP2-supply = <&vcc_reg>; + reset-gpios = <&gpio 0 GPIO_ACTIVE_LOW>; + mute-gpios = <&gpio 1 GPIO_ACTIVE_LOW>; + }; -- 2.27.0

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Several issues with HP OMEN dc-0019-ur
by GitHub issues - edited 22 Jul '20

22 Jul '20

alsa-project/alsa-lib issue #73 was edited from UnicodeMan: Hi. I am using HP OMEN dc0019ur laptop with linux 5.7-xanmod kernel and KDE neon (Ubuntu-based) system. Laptop has combined 3.5mm jack and a seperate stereo microphone jack. Tried various quirks by creating /etc/modprobe.d/mute.conf file and adding line "options snd-hda-intel model=XX", where XX is the name of the quirk from documentation here: https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/tree/Doc… With no modprobe changes: mute led does not work, headphones work, but microphone is not detected, and if i plug anything into microphone jack, it switches to microphone in combined jack. With hp-line1-mic1 quirk: mute led works fine, headphones are detected, except microphone, though it works anyway. Seperate microphone jack does not work at all. With headset-mic quirk: mute led does not work, headphones jack works, microphone (from headphones jack) is not detected when present but works when switched to anyway Microphone jack works perfectly (detects a plugged in mic, works in stereo). --- alsa-info.sh logs --- no quirks: http://alsa-project.org/db/?f=a73a83ba0610e3fd88310999b197fdde4372249a hp-line1-mic1: http://alsa-project.org/db/?f=f4d08dec7918c79a391fb1898d50c57a2448d872 headset-mic: https://alsa-project.org/db/?f=4b1eeef821f39eea28071ac2f2c0babcf515ef14 Not sure if it matters, but I want to add that I'm using nvidia render offload with reverse PRIME support, the gpu has one DP and one HDMI port wired directly to it. aaand, I hope I am writing this issue in correct place :D Issue URL : https://github.com/alsa-project/alsa-lib/issues/73 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH v2 0/9] ASoC: qdsp6: add gapless compressed audio support
by Srinivas Kandagatla 22 Jul '20

22 Jul '20

This patchset adds gapless compressed audio support on q6asm. Gapless on q6asm is implemented using 2 streams in a single q6asm session. First few patches such as stream id per each command, gapless flags and silence meta data are for preparedness for adding gapless support. Last patch implements copy callback to allow finer control over buffer offsets, specially in partial drain cases. This patchset is tested on RB3 aka DB845c platform. This patchset as it is will support gapless however QDSP can also support switching decoders on a single stream. Patches to support such feature are send in different patchset which involves adding generic interfaces. Thanks, srini Changes since v1: - Fixed all the comments to use "q6asm" wording correctly. - dropped patches that are already applied Srinivas Kandagatla (9): ASoC: q6asm: rename misleading session id variable ASoC: q6asm: make commands specific to streams ASoC: q6asm: use flags directly from q6asm-dai ASoC: q6asm: add length to write command token ASoC: q6asm: add support to remove intial and trailing silence ASoC: q6asm: add support to gapless flag in q6asm open ASoC: q6asm-dai: add next track metadata support ASoC: qdsp6-dai: add gapless support ASoC: q6asm-dai: add support to copy callback sound/soc/qcom/qdsp6/q6asm-dai.c | 413 +++++++++++++++++++++++-------- sound/soc/qcom/qdsp6/q6asm.c | 169 +++++++++---- sound/soc/qcom/qdsp6/q6asm.h | 48 ++-- 3 files changed, 467 insertions(+), 163 deletions(-) -- 2.21.0

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[PATCH AUTOSEL 4.4 7/9] ALSA: hda/hdmi: fix failures at PCM open on Intel ICL and later
by Sasha Levin 22 Jul '20

22 Jul '20

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 56275036d8185f92eceac7479d48b858ee3dab84 ] When HDMI PCM devices are opened in a specific order, with at least one HDMI/DP receiver connected, ALSA PCM open fails to -EBUSY on the connected monitor, on recent Intel platforms (ICL/JSL and newer). While this is not a typical sequence, at least Pulseaudio does this every time when it is started, to discover the available PCMs. The rootcause is an invalid assumption in hdmi_add_pin(), where the total number of converters is assumed to be known at the time the function is called. On older Intel platforms this held true, but after ICL/JSL, the order how pins and converters are in the subnode list as returned by snd_hda_get_sub_nodes(), was changed. As a result, information for some converters was not stored to per_pin->mux_nids. And this means some pins cannot be connected to all converters, and application instead gets -EBUSY instead at open. The assumption that converters are always before pins in the subnode list, is not really a valid one. Fix the problem in hdmi_parse_codec() by introducing separate loops for discovering converters and pins. BugLink: https://github.com/thesofproject/linux/issues/1978 BugLink: https://github.com/thesofproject/linux/issues/2216 BugLink: https://github.com/thesofproject/linux/issues/2217 Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20200703153818.2808592-1-kai.vehmanen@linux.intel… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_hdmi.c | 36 +++++++++++++++++++++++------------- 1 file changed, 23 insertions(+), 13 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index b249b1b857464..6aa7403ad80ac 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1735,33 +1735,43 @@ static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) static int hdmi_parse_codec(struct hda_codec *codec) { - hda_nid_t nid; + hda_nid_t start_nid; + unsigned int caps; int i, nodes; - nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &nid); - if (!nid || nodes < 0) { + nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &start_nid); + if (!start_nid || nodes < 0) { codec_warn(codec, "HDMI: failed to get afg sub nodes\n"); return -EINVAL; } - for (i = 0; i < nodes; i++, nid++) { - unsigned int caps; - unsigned int type; + /* + * hdmi_add_pin() assumes total amount of converters to + * be known, so first discover all converters + */ + for (i = 0; i < nodes; i++) { + hda_nid_t nid = start_nid + i; caps = get_wcaps(codec, nid); - type = get_wcaps_type(caps); if (!(caps & AC_WCAP_DIGITAL)) continue; - switch (type) { - case AC_WID_AUD_OUT: + if (get_wcaps_type(caps) == AC_WID_AUD_OUT) hdmi_add_cvt(codec, nid); - break; - case AC_WID_PIN: + } + + /* discover audio pins */ + for (i = 0; i < nodes; i++) { + hda_nid_t nid = start_nid + i; + + caps = get_wcaps(codec, nid); + + if (!(caps & AC_WCAP_DIGITAL)) + continue; + + if (get_wcaps_type(caps) == AC_WID_PIN) hdmi_add_pin(codec, nid); - break; - } } return 0; -- 2.25.1

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[PATCH 0/3] ASoC: mediatek: mt8183: support machine driver for max98357b
by Tzung-Bi Shih 22 Jul '20

22 Jul '20

The series re-uses mt8183-mt6358-ts3a227-max98357.c to support machine driver with max98357b. The 1st patch enables left justified format from mt8183 audio platform. The 2nd patch adds document for the new proposed compatible string for max98357b. The 3rd patch supports machine driver with max98357b and uses left justified format for it. Tzung-Bi Shih (3): ASoC: mediatek: mt8183: support left justified format for I2S ASoC: dt-bindings: mt8183: add compatible string for using max98357b ASoC: mediatek: mt8183: support machine driver with max98357b .../sound/mt8183-mt6358-ts3a227-max98357.txt | 1 + sound/soc/mediatek/mt8183/mt8183-dai-i2s.c | 59 ++++++++++++++++--- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 22 ++++++- 3 files changed, 73 insertions(+), 9 deletions(-) -- 2.28.0.rc0.105.gf9edc3c819-goog

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[PATCH 0/7] SOF IMX fixes
by Daniel Baluta 22 Jul '20

22 Jul '20

From: Daniel Baluta <daniel.baluta(a)nxp.com> This patchseries contains a couple of SOF IMX fixes found during our first IMX SOF release. Daniel Baluta (7): ASoC: SOF: define INFO_ flags in dsp_ops for imx8 ASoC: SOF: imx: Use ARRAY_SIZE instead of hardcoded value ASoC: SOF: imx8: Fix ESAI DAI driver name for i.MX8/iMX8X ASoC: SOF: imx8m: Fix SAI DAI driver for i.MX8M ASoC: SOF: imx8: Add SAI dai driver for i.MX/i.MX8X ASoC: SOF: topology: Update SAI config bclk/fsync rate ASoC: SOF: pcm: Update rate/channels for SAI/ESAI DAIs sound/soc/sof/imx/imx8.c | 24 +++++++++++++++++++++--- sound/soc/sof/imx/imx8m.c | 4 ++-- sound/soc/sof/pcm.c | 8 ++++++++ sound/soc/sof/topology.c | 2 ++ 4 files changed, 33 insertions(+), 5 deletions(-) -- 2.17.1

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[PATCH 1/3] ASoC: hdac_hda: call patch_ops.free() on probe error
by Kai Vehmanen 22 Jul '20

22 Jul '20

Add error handling for patch_ops in hdac_hda_codec_probe(). Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/codecs/hdac_hda.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/hdac_hda.c b/sound/soc/codecs/hdac_hda.c index 473efe9ef998..72bd779bf942 100644 --- a/sound/soc/codecs/hdac_hda.c +++ b/sound/soc/codecs/hdac_hda.c @@ -467,7 +467,7 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component) ret = snd_hda_codec_parse_pcms(hcodec); if (ret < 0) { dev_err(&hdev->dev, "unable to map pcms to dai %d\n", ret); - goto error_regmap; + goto error_patch; } /* HDMI controls need to be created in machine drivers */ @@ -476,7 +476,7 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component) if (ret < 0) { dev_err(&hdev->dev, "unable to create controls %d\n", ret); - goto error_regmap; + goto error_patch; } } @@ -496,6 +496,9 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component) return 0; +error_patch: + if (hcodec->patch_ops.free) + hcodec->patch_ops.free(hcodec); error_regmap: snd_hdac_regmap_exit(hdev); error_pm: -- 2.27.0

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