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

[PATCH 0/2] soundwire: sysfs: expose device number and status
by Pierre-Louis Bossart 17 Sep '20

17 Sep '20

This patchset combines three contributions: Srinivas Kandagalta suggested creating a device when it's detected on the bus but not described in platform firmware. I suggested adding the device number and status to show the difference with 'ghost' devices, described in firmware but not physically present. Vinod suggested a simpler way to report the status. I did not keep Vinod's patch separate since it was using the same group attribute as the other properties, which prevents status and device number from being reported if there is no firmware and no driver. These patches provide a nice addition for integrators/tests and were tested on Qualcomm and Intel platforms. Thanks Srinivas for the initial idea! Pierre-Louis Bossart (1): soundwire: sysfs: add slave status and device number before probe Srinivas Kandagatla (1): soundwire: bus: add enumerated Slave device to device list .../ABI/testing/sysfs-bus-soundwire-slave | 18 ++++++ drivers/soundwire/bus.c | 9 +++ drivers/soundwire/bus.h | 2 + drivers/soundwire/bus_type.c | 9 +++ drivers/soundwire/slave.c | 6 +- drivers/soundwire/sysfs_local.h | 4 ++ drivers/soundwire/sysfs_slave.c | 64 ++++++++++++++++++- 7 files changed, 109 insertions(+), 3 deletions(-) -- 2.25.1

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[PATCH v2 0/2] soundwire: use {u32|u16}p_replace_bits
by Vinod Koul 17 Sep '20

17 Sep '20

This is v2 of the series to fix usage of FIELD_PREP and use {u32|u16}p_replace_bits. v1 used u32_replace_bits but looks like it might be removed so use the {u32|u16}p_replace_bits() instead. Also this helped to find that one instance in intel driver should have used u16p_replace_bits() so that is fixed too Vinod Koul (2): soundwire: cadence: use u32p_replace_bits soundwire: intel: use {u32|u16}p_replace_bits drivers/soundwire/cadence_master.c | 21 ++++++++++----------- drivers/soundwire/intel.c | 14 +++++++------- 2 files changed, 17 insertions(+), 18 deletions(-) -- 2.26.2

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Re: [oss-drivers] [trivial PATCH] treewide: Convert switch/case fallthrough; to break;
by Simon Horman 17 Sep '20

17 Sep '20

On Wed, Sep 09, 2020 at 01:06:39PM -0700, Joe Perches wrote: > fallthrough to a separate case/default label break; isn't very readable. > > Convert pseudo-keyword fallthrough; statements to a simple break; when > the next label is case or default and the only statement in the next > label block is break; > > Found using: > > $ grep-2.5.4 -rP --include=*.[ch] -n "fallthrough;(\s*(case\s+\w+|default)\s*:\s*){1,7}break;" * > > Miscellanea: > > o Move or coalesce a couple label blocks above a default: block. > > Signed-off-by: Joe Perches <joe(a)perches.com> ... > diff --git a/drivers/net/ethernet/netronome/nfp/nfpcore/nfp6000_pcie.c b/drivers/net/ethernet/netronome/nfp/nfpcore/nfp6000_pcie.c > index 252fe06f58aa..1d5b87079104 100644 > --- a/drivers/net/ethernet/netronome/nfp/nfpcore/nfp6000_pcie.c > +++ b/drivers/net/ethernet/netronome/nfp/nfpcore/nfp6000_pcie.c > @@ -345,7 +345,7 @@ static int matching_bar(struct nfp_bar *bar, u32 tgt, u32 act, u32 tok, > baract = NFP_CPP_ACTION_RW; > if (act == 0) > act = NFP_CPP_ACTION_RW; > - fallthrough; > + break; > case NFP_PCIE_BAR_PCIE2CPP_MapType_FIXED: > break; > default: This is a cascading fall-through handling all map types. I don't think this change improves readability. ...

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"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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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 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 as we can expected. ### 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 - opened 16 Sep '20

16 Sep '20

alsa-project/alsa-utils issue #55 was opened from maxmbed: ### Description I am encountering an expected 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 as we can expected. ### 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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Re: [PATCH] fsl: imx-audmix: Use devm_kcalloc() in imx_audmix_probe()
by Markus Elfring 16 Sep '20

16 Sep '20

> A multiplication for the size determination of a memory allocation > indicated that an array data structure should be processed. > Thus use the corresponding function "devm_kcalloc". I find the previous patch subject inappropriate. Would you like to choose an alternative? Regards, Markus

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Re: [PATCH v2 3/3] ASoC: tlv320adcx140: Add proper support for master mode
by Dan Murphy 16 Sep '20

16 Sep '20

Camel On 9/16/20 2:52 AM, Camel Guo wrote: > Please forget about this patch since Dan will upload a similar one. > > @Dan, see my comment below. > I have cc'd you on my patchset and as you can see I did not add any of the master mode programming only setting of the master mode bits. So this patch is still viable it's just the master mode bit programming that needs to be looked at. Dan

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[PATCH v3 00/11] USB: new USB control message helper functions
by Greg Kroah-Hartman 16 Sep '20

16 Sep '20

In a recent discussion about a USB networking bug found by syzbot, and fixed by Himadri Pandya, the design of the existing usb_control_msg() call was brought up as not being the "nicest" thing to use by Dmitry Vyukov: https://lore.kernel.org/r/CACT4Y+YbDODLRFn8M5QcY4CazhpeCaunJnP_udXtAs0rYoAS… The function makes it hard to get right, in that it will return the number of bytes sent/received, but almost no one checks to see if a short read/write happens. With a malicious, or broken, USB device, this can cause drivers to act on data that they did not anticipate, and sometimes copy internal kernel data out to userspace. So let's fix this up by creating two new functions, usb_control_msg_send() and usb_control_msg_recv(). These functions either complete the full transation, or they return an error, a short send/recv is now an error. They also accept data off of the stack, saving individual drivers the pain of having to constantly allocate memory on their own for tiny messages, thereby saving overall kernel code space. The api also does not require a raw USB "pipe" to be sent to the function, as we know the direction, so just pass in the endpoint number instead, again making it easier on the USB driver author to use. This series first takes a helper function out of the sound core for verifying USB endpoints to be able to use internally, and then adds the new functions, converts over some internal USB code to use them, and then starts to clean up some drivers using these new functions, as an example of the savings that can happen by using these functions. Thanks to Dmitry and Himadri for the idea on how to do all of this. greg k-h ----- Changes from v2: - add Andy's reviewed-by: to patch 3 - remove unneeded change in usb_enable_link_state() in hub.c in patch 4 Changes from v1: - added acks from Takashi Iwai - dropped changes to one function in patch 04 thanks to review from Alan Stern - typo fix in comment in patch 01 - added new patch 11 to remove some unneeded checks in the sound drivers for endpoint statuses that would always be true. Greg Kroah-Hartman (11): USB: move snd_usb_pipe_sanity_check into the USB core USB: add usb_control_msg_send() and usb_control_msg_recv() USB: core: message.c: use usb_control_msg_send() in a few places USB: core: hub.c: use usb_control_msg_send() in a few places USB: legousbtower: use usb_control_msg_recv() sound: usx2y: move to use usb_control_msg_send() sound: 6fire: move to use usb_control_msg_send() and usb_control_msg_recv() sound: line6: move to use usb_control_msg_send() and usb_control_msg_recv() sound: hiface: move to use usb_control_msg_send() Bluetooth: ath3k: use usb_control_msg_send() and usb_control_msg_recv() ALSA: remove calls to usb_pipe_type_check for control endpoints drivers/bluetooth/ath3k.c | 90 +++++------------ drivers/usb/core/hub.c | 99 ++++++++---------- drivers/usb/core/message.c | 171 ++++++++++++++++++++++++++++---- drivers/usb/core/urb.c | 31 ++++-- drivers/usb/misc/legousbtower.c | 60 ++++------- include/linux/usb.h | 7 ++ sound/usb/6fire/firmware.c | 38 +++---- sound/usb/helper.c | 16 +-- sound/usb/helper.h | 1 - sound/usb/hiface/pcm.c | 14 ++- sound/usb/line6/driver.c | 69 +++++-------- sound/usb/line6/podhd.c | 17 ++-- sound/usb/line6/toneport.c | 8 +- sound/usb/mixer_scarlett_gen2.c | 2 +- sound/usb/quirks.c | 12 +-- sound/usb/usx2y/us122l.c | 42 ++------ 16 files changed, 336 insertions(+), 341 deletions(-) -- 2.28.0

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[PATCH 01/10] dt-bindings: arm: samsung: pmu: Use unevaluatedProperties
by Krzysztof Kozlowski 16 Sep '20

16 Sep '20

Additional properties actually might appear (e.g. assigned-clocks) so use unevaluatedProperties to fix dtbs_check warnings like: arch/arm64/boot/dts/exynos/exynos5433-tm2.dt.yaml: system-controller@105c0000: 'assigned-clock-parents', 'assigned-clocks' do not match any of the regexes: 'pinctrl-[0-9]+' Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> --- Documentation/devicetree/bindings/arm/samsung/pmu.yaml | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/arm/samsung/pmu.yaml b/Documentation/devicetree/bindings/arm/samsung/pmu.yaml index 686c13c14e32..30ff2da81416 100644 --- a/Documentation/devicetree/bindings/arm/samsung/pmu.yaml +++ b/Documentation/devicetree/bindings/arm/samsung/pmu.yaml @@ -86,7 +86,7 @@ required: - compatible - reg -additionalProperties: false +unevaluatedProperties: false allOf: - if: -- 2.17.1

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[PATCH v2 1/3] dt-bindings: tlv320adcx140: Add GPIO config and drive config
by Camel Guo 15 Sep '20

15 Sep '20

From: Camel Guo <camelg(a)axis.com> Add properties for configuring the General Purpose Input Outputs (GPIO). There are 2 settings for GPIO, configuration and the output drive type. Signed-off-by: Camel Guo <camelg(a)axis.com> --- .../bindings/sound/tlv320adcx140.yaml | 44 +++++++++++++++++++ 1 file changed, 44 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml index f578f17f3e04..7b0b4554da59 100644 --- a/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml +++ b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml @@ -134,6 +134,49 @@ patternProperties: 4d - Drive weak low and active high 5d - Drive Hi-Z and active high + ti,gpio-config: + description: | + Defines the configuration and output driver for the general purpose + input and output pin (GPIO1). Its value is a pair, the first value is for + the configuration type and the second value is for the output drive + type. The array is defined as <GPIO1_CFG GPIO1_DRV> + + configuration for the GPIO pin can be one of the following: + 0 - disabled + 1 - GPIO1 is configured as a general-purpose output (GPO) + 2 - (default) GPIO1 is configured as a device interrupt output (IRQ) + 3 - GPIO1 is configured as a secondary ASI output (SDOUT2) + 4 - GPIO1 is configured as a PDM clock output (PDMCLK) + 8 - GPIO1 is configured as an input to control when MICBIAS turns on or + off (MICBIAS_EN) + 9 - GPIO1 is configured as a general-purpose input (GPI) + 10 - GPIO1 is configured as a master clock input (MCLK) + 11 - GPIO1 is configured as an ASI input for daisy-chain (SDIN) + 12 - GPIO1 is configured as a PDM data input for channel 1 and channel 2 + (PDMDIN1) + 13 - GPIO1 is configured as a PDM data input for channel 3 and channel 4 + (PDMDIN2) + 14 - GPIO1 is configured as a PDM data input for channel 5 and channel 6 + (PDMDIN3) + 15 - GPIO1 is configured as a PDM data input for channel 7 and channel 8 + (PDMDIN4) + + output drive type for the GPIO pin can be one of the following: + 0 - Hi-Z output + 1 - Drive active low and active high + 2 - (default) Drive active low and weak high + 3 - Drive active low and Hi-Z + 4 - Drive weak low and active high + 5 - Drive Hi-Z and active high + + allOf: + - $ref: /schemas/types.yaml#/definitions/uint32-array + - minItems: 2 + maxItems: 2 + items: + maximum: 15 + default: [2, 2] + required: - compatible - reg @@ -150,6 +193,7 @@ examples: ti,mic-bias-source = <6>; ti,pdm-edge-select = <0 1 0 1>; ti,gpi-config = <4 5 6 7>; + ti,gpio-config = <10 2>; ti,gpo-config-1 = <0 0>; ti,gpo-config-2 = <0 0>; reset-gpios = <&gpio0 14 GPIO_ACTIVE_HIGH>; -- 2.20.1

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Re: [PATCH v4 2/5] ASoC: dt-bindings: Add dt binding for lpass hdmi
by Rob Herring 15 Sep '20

15 Sep '20

On Tue, Sep 08, 2020 at 05:44:39PM +0000, Srinivasa Rao Mandadapu wrote: > From: V Sujith Kumar Reddy <vsujithk(a)codeaurora.org> > > Adds bindings for lpass hdmi interface > which can support audio path over dp. > > Signed-off-by: Srinivasa Rao <srivasam(a)codeaurora.org> > Signed-off-by: V Sujith Kumar Reddy <vsujithk(a)codeaurora.org> > --- > .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 51 ++++++++++++++++++++-- > 1 file changed, 47 insertions(+), 4 deletions(-) > > diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml > index 09c9bd2..7c2ac0c 100644 > --- a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml > +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml > @@ -22,6 +22,7 @@ properties: > - qcom,lpass-cpu > - qcom,apq8016-lpass-cpu > - qcom,sc7180-lpass-cpu > + - qcom,sc7180-lpass-hdmi > > reg: > maxItems: 1 > @@ -60,10 +61,12 @@ properties: > const: 0 > > patternProperties: > - "(^mi2s-[0-9a-f]$|mi2s)": > + "^dai@[0-9a-f]$": > type: object > - description: Required properties for each DAI > - > + description: | > + LPASS CPU dai node for each I2S device. Bindings of each node > + depends on the specific driver providing the functionality and > + properties. > properties: > reg: > maxItems: 1 > @@ -145,6 +148,22 @@ allOf: > - iommus > - power-domains > > + - if: > + properties: > + compatible: > + contains: > + const: qcom,sc7180-lpass-hdmi > + then: > + properties: > + clock-names: > + items: > + - const: pcnoc-sway-clk > + - const: audio-core > + - const: pcnoc-mport-clk > + required: > + - iommus > + - power-domains > + > examples: > - | > #include <dt-bindings/sound/sc7180-lpass.h> > @@ -178,12 +197,36 @@ examples: > #address-cells = <1>; > #size-cells = <0>; > /* Optional to set different MI2S SD lines */ > - mi2s-primary@0 { > + dai@mi2s-primary { The unit address should be a number. As this is not failing checks, then you are missing an 'additionalProperties: false'. > reg = <MI2S_PRIMARY>; > qcom,playback-sd-lines = <1>; > qcom,capture-sd-lines = <0>; > }; > }; > + > + lpassh@62d87000 { > + compatible = "qcom,sc7180-lpass-hdmi"; > + > + reg = <0 0x62d87000 0 0x68000>; > + > + iommus = <&apps_smmu 0x1032 0>; > + > + power-domains = <&lpass_hm 0>; > + > + clocks = <&gcc 131>, > + <&lpasscc 6>, > + <&lpasscc 10>; > + > + clock-names = "pcnoc-sway-clk", "audio-core", > + "pcnoc-mport-clk"; > + > + #sound-dai-cells = <1>; > + > + #address-cells = <1>; > + #size-cells = <0>; > + > + interrupts = <0 268 1>; > + }; > }; > > ... > -- > Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., > is a member of Code Aurora Forum, a Linux Foundation Collaborative Project. >

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[RFC] dpcm_fe_dai_do_trigger: implicit removal of _DRAIN cmd since v5.4
by Cezary Rojewski 15 Sep '20

15 Sep '20

Hello, Till v5.4 dpcm_fe_dai_do_trigger was supporting all pcm cmds as each case: SND_SOC_DPCM_TRIGGER_PRE/ _POST/ _BESPOKE was simply passing cmd's value to appropriate variant of _trigger(). Since the addition of: acbf27746ecfa96b290b54cc7f05273482ea128a ASoC: pcm: update FE/BE trigger order based on the command additional filters have been introduced for _PRE and _POST cases: switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: ret = dpcm_dai_trigger_fe_be(substream, cmd, true); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: ret = dpcm_dai_trigger_fe_be(substream, cmd, false); break; default: ret = -EINVAL; break; } effectively removing support for _DRAIN command - all requests will yield -EINVAL. _BESPOKE was left alone so support remained there. Now, is DPCM no longer supporting _DRAIN and that's how things should be -or- DPCM still intends to support _DRAIN and mentioned change is unintended regression? Czarek

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Re: [PATCH v4 1/5] ASoC: Add sc7180-lpass binding header hdmi define
by Rob Herring 15 Sep '20

15 Sep '20

On Tue, 08 Sep 2020 17:44:33 +0000, Srinivasa Rao Mandadapu wrote: > From: V Sujith Kumar Reddy <vsujithk(a)codeaurora.org> > > Add header defining hdmi dai-id for SC7180 lpass soc > in dt bindings. > > Signed-off-by: Srinivasa Rao <srivasam(a)codeaurora.org> > Signed-off-by: V Sujith Kumar Reddy <vsujithk(a)codeaurora.org> > --- > include/dt-bindings/sound/sc7180-lpass.h | 1 + > 1 file changed, 1 insertion(+) > Acked-by: Rob Herring <robh(a)kernel.org>

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a saner API for allocating DMA addressable pages v2
by Christoph Hellwig 15 Sep '20

15 Sep '20

Hi all, this series replaced the DMA_ATTR_NON_CONSISTENT flag to dma_alloc_attrs with a separate new dma_alloc_pages API, which is available on all platforms. In addition to cleaning up the convoluted code path, this ensures that other drivers that have asked for better support for non-coherent DMA to pages with incurring bounce buffering over can finally be properly supported. I'm still a little unsure about the API naming, as alloc_pages sort of implies a struct page return value, but we return a kernel virtual address. The other alternative would be to name the API dma_alloc_noncoherent, but the whole non-coherent naming seems to put people off. As a follow up I plan to move the implementation of the DMA_ATTR_NO_KERNEL_MAPPING flag over to this framework as well, given that is also is a fundamentally non coherent allocation. The replacement for that flag would then return a struct page, as it is allowed to actually return pages without a kernel mapping as the name suggested (although most of the time they will actually have a kernel mapping..) In addition to the conversions of the existing non-coherent DMA users, I've also added a patch to convert the firewire ohci driver to use the new dma_alloc_pages API. Note that I haven't carried over any Tested-by: tags for the noncoherent allocation conversions as there was a bit of a patch reshuffle, but the result should be the same. The first patch is queued up for 5.9 in the media tree, but included here for completeness. A git tree is available here: git://git.infradead.org/users/hch/misc.git dma_alloc_pages Gitweb: http://git.infradead.org/users/hch/misc.git/shortlog/refs/heads/dma_alloc_p… Changes since v1: - rebased on the latests dma-mapping tree, which merged many of the cleanups - fix an argument passing typo in 53c700, caught by sparse - rename a few macro arguments in 53c700 - pass the right device to the DMA API in the lib82596 drivers - fix memory ownershiptransfers in sgiseeq - better document what a page in the direct kernel mapping means - split into dma_alloc_pages that returns a struct page and is in the direct mapping vs dma_alloc_noncoherent that can be vmapped - conver the firewire ohci driver to dma_alloc_pages Diffstat:

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