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

"aplay -D oss -l" still reads sound cards through ALSA, not OSS
by GitHub issues - edited 17 Sep '20

17 Sep '20

alsa-project/alsa-utils issue #56 was edited from yurivict: I don't have any ALSA devices, only OSS devices. When the OSS PCM is selected ```aplay``` still tries to read ALSA devices: ``` $ /usr/bin/aplay -D oss -l aplay: device_list:272: no soundcards found... ``` ``` $ /usr/bin/aplay -L null Discard all samples (playback) or generate zero samples (capture) oss Open Sound System ``` But ```/usr/bin/aplay x.wav``` plays it through the OSS card. aplay should list OSS cards when -D is selected. The ALSA interface should also list OSS cards when no ALSA cards are present but it doesn't. alsa-lib-1.1.8 alsa-plugins-oss-1.1.6 alsa-utils-1.1.8 Issue URL : https://github.com/alsa-project/alsa-utils/issues/56 Repository URL: https://github.com/alsa-project/alsa-utils

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"aplay -D oss -l" still reads sound cards through ALSA, not OSS
by GitHub issues - opened 17 Sep '20

17 Sep '20

alsa-project/alsa-utils issue #56 was opened from yurivict: I don't have any ALSA devices, only OSS devices. When the OSS PCM is selected ```aplay``` still tries to read ALSA devices: ``` $ /usr/bin/aplay -D oss -l aplay: device_list:272: no soundcards found... ``` ``` $ /usr/bin/aplay -L null Discard all samples (playback) or generate zero samples (capture) oss Open Sound System ``` alsa-lib-1.1.8 alsa-plugins-oss-1.1.6 alsa-utils-1.1.8 Issue URL : https://github.com/alsa-project/alsa-utils/issues/56 Repository URL: https://github.com/alsa-project/alsa-utils

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[PATCH 0/3] ASoC: fsl_sai: update the register list
by Shengjiu Wang 17 Sep '20

17 Sep '20

As sai ip is upgraded, so update sai register list. Shengjiu Wang (3): ASoC: fsl_sai: Add new added registers and new bit definition ASoC: fsl_sai: Add fsl_sai_check_version function ASoC: fsl_sai: Set MCLK input or output direction sound/soc/fsl/fsl_sai.c | 77 ++++++++++++++++++++++++++++++++++++ sound/soc/fsl/fsl_sai.h | 87 +++++++++++++++++++++++++++++++++++++++++ 2 files changed, 164 insertions(+) -- 2.27.0

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Re: [PATCH] ASoC: fsl_audmix: make clock and output src write only
by Nicolin Chen 17 Sep '20

17 Sep '20

On Mon, Sep 14, 2020 at 08:24:34PM +0300, Viorel Suman (OSS) wrote: > From: Viorel Suman <viorel.suman(a)nxp.com> > > "alsactl -f state.conf store/restore" sequence fails because setting > "mixing clock source" and "output source" requires active TDM clock > being started for configuration propagation. Make these two controls > write only so that their values are not stored at "alsactl store". > > Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> > --- > sound/soc/fsl/fsl_audmix.c | 16 ++++++++++++---- > 1 file changed, 12 insertions(+), 4 deletions(-) > > diff --git a/sound/soc/fsl/fsl_audmix.c b/sound/soc/fsl/fsl_audmix.c > index a447baf..7ad5925 100644 > --- a/sound/soc/fsl/fsl_audmix.c > +++ b/sound/soc/fsl/fsl_audmix.c > @@ -199,10 +199,18 @@ static int fsl_audmix_put_out_src(struct snd_kcontrol *kcontrol, > > static const struct snd_kcontrol_new fsl_audmix_snd_controls[] = { > /* FSL_AUDMIX_CTR controls */ > - SOC_ENUM_EXT("Mixing Clock Source", fsl_audmix_enum[0], > - snd_soc_get_enum_double, fsl_audmix_put_mix_clk_src), > - SOC_ENUM_EXT("Output Source", fsl_audmix_enum[1], > - snd_soc_get_enum_double, fsl_audmix_put_out_src), > + { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, > + .name = "Mixing Clock Source", > + .info = snd_soc_info_enum_double, > + .access = SNDRV_CTL_ELEM_ACCESS_WRITE, > + .put = fsl_audmix_put_mix_clk_src, > + .private_value = (unsigned long)&fsl_audmix_enum[0] }, > + { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, > + .name = "Output Source", > + .info = snd_soc_info_enum_double, > + .access = SNDRV_CTL_ELEM_ACCESS_WRITE, > + .put = fsl_audmix_put_out_src, > + .private_value = (unsigned long)&fsl_audmix_enum[1] }, > SOC_ENUM("Output Width", fsl_audmix_enum[2]), > SOC_ENUM("Frame Rate Diff Error", fsl_audmix_enum[3]), > SOC_ENUM("Clock Freq Diff Error", fsl_audmix_enum[4]), > -- > 2.7.4 >

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[PATCH v3 0/3] soundwire: qcom: fix IP version v1.5.1 support
by Srinivas Kandagatla 16 Sep '20

16 Sep '20

While testing Qualcomm soundwire controller version 1.5.1, found two issue, Firstly the frame shape information configured vs the bus parameters are out of sync. secondly some ports on this ip version require block packing mode support. With this patchset I was able to test 2 WSA speakers! Also I found a regression due to move to REG_FIELD, which patch 1 fixes it! thanks, srini Changes since v2: - updated to use u32_replace_bits(). - wrapped up in single lines as suggested by vkoul. Srinivas Kandagatla (3): soundwire: qcom: clear BIT FIELDs before value set. soundwire: qcom: add support to block packing mode soundwire: qcom: get max rows and cols info from compatible drivers/soundwire/qcom.c | 70 +++++++++++++++++++++++++++++----------- 1 file changed, 51 insertions(+), 19 deletions(-) -- 2.21.0

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[PATCH v2 0/3] soundwire: qcom: fix IP version v1.5.1 support
by Srinivas Kandagatla 16 Sep '20

16 Sep '20

While testing Qualcomm soundwire controller version 1.5.1, found two issue, Firstly the frame shape information configured vs the bus parameters are out of sync. secondly some ports on this ip version require block packing mode support. With this patchset I was able to test 2 WSA speakers! Also I found a regression due to move to REG_FIELD, which patch 1 fixes it! thanks, srini Changes since v1: - rebased on top of REG_FILED patch or soundwire-next branch - udated qcom_swrm_data to use u32 instead of int as suggested by VKoul Srinivas Kandagatla (3): soundwire: qcom: clear BIT FIELDs before value set. soundwire: qcom: add support to block packing mode soundwire: qcom: get max rows and cols info from compatible drivers/soundwire/qcom.c | 76 ++++++++++++++++++++++++++++++---------- 1 file changed, 58 insertions(+), 18 deletions(-) -- 2.21.0

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[PATCH 1/2] soundwire: cadence: use u32_replace_bits
by Vinod Koul 16 Sep '20

16 Sep '20

FIELD_PREP() does not replace the bits so it is not apt in case where we modify a register. Use u32_replace_bits() instead. Fixes: 3cf25d63b1b9 ("soundwire: cadence: use FIELD_{GET|PREP}") Signed-off-by: Vinod Koul <vkoul(a)kernel.org> --- drivers/soundwire/cadence_master.c | 23 ++++++++++++----------- 1 file changed, 12 insertions(+), 11 deletions(-) diff --git a/drivers/soundwire/cadence_master.c b/drivers/soundwire/cadence_master.c index f1efe5beead6..bb293e7010f3 100644 --- a/drivers/soundwire/cadence_master.c +++ b/drivers/soundwire/cadence_master.c @@ -1278,9 +1278,9 @@ static int cdns_port_params(struct sdw_bus *bus, dpn_config = cdns_readl(cdns, dpn_config_off); - dpn_config |= FIELD_PREP(CDNS_DPN_CONFIG_WL, (p_params->bps - 1)); - dpn_config |= FIELD_PREP(CDNS_DPN_CONFIG_PORT_FLOW, p_params->flow_mode); - dpn_config |= FIELD_PREP(CDNS_DPN_CONFIG_PORT_DAT, p_params->data_mode); + dpn_config = u32_replace_bits(dpn_config, (p_params->bps - 1), CDNS_DPN_CONFIG_WL); + dpn_config = u32_replace_bits(dpn_config, p_params->flow_mode, CDNS_DPN_CONFIG_PORT_FLOW); + dpn_config = u32_replace_bits(dpn_config, p_params->data_mode, CDNS_DPN_CONFIG_PORT_DAT); cdns_writel(cdns, dpn_config_off, dpn_config); @@ -1316,18 +1316,19 @@ static int cdns_transport_params(struct sdw_bus *bus, } dpn_config = cdns_readl(cdns, dpn_config_off); - - dpn_config |= FIELD_PREP(CDNS_DPN_CONFIG_BGC, t_params->blk_grp_ctrl); - dpn_config |= FIELD_PREP(CDNS_DPN_CONFIG_BPM, t_params->blk_pkg_mode); + dpn_config = u32_replace_bits(dpn_config, t_params->blk_grp_ctrl, CDNS_DPN_CONFIG_BGC); + dpn_config = u32_replace_bits(dpn_config, t_params->blk_pkg_mode, CDNS_DPN_CONFIG_BPM); cdns_writel(cdns, dpn_config_off, dpn_config); - dpn_offsetctrl |= FIELD_PREP(CDNS_DPN_OFFSET_CTRL_1, t_params->offset1); - dpn_offsetctrl |= FIELD_PREP(CDNS_DPN_OFFSET_CTRL_2, t_params->offset2); + dpn_offsetctrl = u32_replace_bits(dpn_offsetctrl, + t_params->offset1, CDNS_DPN_OFFSET_CTRL_1); + dpn_offsetctrl = u32_replace_bits(dpn_offsetctrl, + t_params->offset2, CDNS_DPN_OFFSET_CTRL_2); cdns_writel(cdns, dpn_offsetctrl_off, dpn_offsetctrl); - dpn_hctrl |= FIELD_PREP(CDNS_DPN_HCTRL_HSTART, t_params->hstart); - dpn_hctrl |= FIELD_PREP(CDNS_DPN_HCTRL_HSTOP, t_params->hstop); - dpn_hctrl |= FIELD_PREP(CDNS_DPN_HCTRL_LCTRL, t_params->lane_ctrl); + dpn_hctrl = u32_replace_bits(dpn_hctrl, t_params->hstart, CDNS_DPN_HCTRL_HSTART); + dpn_hctrl = u32_replace_bits(dpn_hctrl, t_params->hstop, CDNS_DPN_HCTRL_HSTOP); + dpn_hctrl = u32_replace_bits(dpn_hctrl, t_params->lane_ctrl, CDNS_DPN_HCTRL_LCTRL); cdns_writel(cdns, dpn_hctrl_off, dpn_hctrl); cdns_writel(cdns, dpn_samplectrl_off, (t_params->sample_interval - 1)); -- 2.26.2

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aplay: playing mono audio file alteres pulseaudio default Master left or right volume level
by GitHub issues - edited 16 Sep '20

16 Sep '20

alsa-project/alsa-utils issue #55 was edited from maxmbed: ### Description I am encountering an unexpected behaviour with pulse audio master volume in my application. If I set for example left channel to 100% and right channel 0%, and play mono audio file (i.e. audio file with 1 channel only), then I would expect to hear only the audio file on left channel. The actual behaviour in the example of above is when playing the audio file, the Master volume right channel will be automatically adjusted to 50%. If play again it increases to 75% and so on. This issue happens also with other R/L volume level, the case of above is just an example. **Important to note the case does not happen with stereo audio file** where audio sounds on left only channel without altering the volume level of right channel as we can expect. ### Steps to repeat issue: (a) Set mixer Lch to 100%, Rch to 0% ``` $ amixer -D default sset Master 65535,0 Simple mixer control 'Master',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 65536 Mono: Front Left: Playback 65535 [100%] [on] Front Right: Playback 0 [0%] [on] ``` (b) Play mono audio file ``` $ aplay -D default ringtone.wav Playing WAVE 'ringtone.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono ``` ### Request (1) Is it by design expected ? (2) Is there a way to prevent this phenomenon (command option) ? Issue URL : https://github.com/alsa-project/alsa-utils/issues/55 Repository URL: https://github.com/alsa-project/alsa-utils

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aplay: playing mono audio file alteres pulseaudio default Master left or right volume level
by GitHub issues - edited 16 Sep '20

16 Sep '20

alsa-project/alsa-utils issue #55 was edited from maxmbed: ### Description I am encountering an unexpected behaviour with pulse audio master volume in my application. If I set for example left channel to 100% and right channel 0%, and play mono audio file (i.e. audio file with 1 channel only), then I would expect to hear only the audio file on left channel. The actual behaviour in the example of above is when playing the audio file, the Master volume right channel will be automatically adjusted to 50%. If play again it increases to 75% and so on. This issue happens also with other R/L volume level, the case of above is just an example. **Important to note the case does not happen with stereo audio file** where audio sounds on left only channel without altering the volume level as we can expect. ### Steps to repeat issue: (a) Set mixer Lch to 100%, Rch to 0% ``` $ amixer -D default sset Master 65535,0 Simple mixer control 'Master',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 65536 Mono: Front Left: Playback 65535 [100%] [on] Front Right: Playback 0 [0%] [on] ``` (b) Play mono audio file ``` $ aplay -D default ringtone.wav Playing WAVE 'ringtone.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono ``` ### Request (1) Is it by design expected ? (2) Is there a way to prevent this phenomenon (command option) ? Issue URL : https://github.com/alsa-project/alsa-utils/issues/55 Repository URL: https://github.com/alsa-project/alsa-utils

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[PATCH v2 0/3] regmap: add SoundWire 1.2 MBQ support
by Pierre-Louis Bossart 16 Sep '20

16 Sep '20

In preparation of the upstream contribution of SDCA (SoundWire Device Class for Audio) ASoC codec drivers [1] [2], add regmap support SoundWire 1.2 MBQ support. The MBQ (Multi-Byte Quantity) registers need to be handled in a different way from regular 8-bit SoundWire registers, their main application is going to be for volume/gain controls. The second patch was initially suggested for inclusion in the SoundWire tree, and was modified to add more background information on SDCA in the commit message as requested by Vinod Koul. Changes since v1: Rebased on regmap tree (conflict with SPI stuff). Removed mod_devicetable.h header Removed -EOPNOTSUPP error codes, use -ENOTSUPP Added long description of SDCA Used FIELD_PREP/GET as suggested by Vinod Koul Added Bard Liao's Acked-by tag. Pierre-Louis Bossart (3): regmap: sdw: add required header files soundwire: SDCA: add helper macro to access controls regmap: sdw: add support for SoundWire 1.2 MBQ drivers/base/regmap/Kconfig | 6 +- drivers/base/regmap/Makefile | 1 + drivers/base/regmap/regmap-sdw-mbq.c | 101 ++++++++++++++++++++++++ drivers/base/regmap/regmap-sdw.c | 2 + include/linux/regmap.h | 21 +++++ include/linux/soundwire/sdw_registers.h | 33 ++++++++ 6 files changed, 163 insertions(+), 1 deletion(-) create mode 100644 drivers/base/regmap/regmap-sdw-mbq.c base-commit: d17343b87da422a59d99a3ed130573dbeb96c582 -- 2.25.1

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