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[PATCH] ASoC: sh: rz-ssi: Improve error handling in rz_ssi_dma_request function
by Biju Das 15 Aug '21

15 Aug '21

The rz_ssi_dma_request function only checks the NULL condition for the value returned by the dma_request_chan function, but this function can also return an error. If it happens, the subsequent function call to rz_ssi_dma_slave_config can lead to a kernel crash. This patch fixes the issue by checking both error and NULL condition returned by dma_request_chan. Signed-off-by: Biju Das <biju.das.jz(a)bp.renesas.com> --- sound/soc/sh/rz-ssi.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rz-ssi.c b/sound/soc/sh/rz-ssi.c index ea8d33ede5d2..3867e2efd3e0 100644 --- a/sound/soc/sh/rz-ssi.c +++ b/sound/soc/sh/rz-ssi.c @@ -676,11 +676,19 @@ static void rz_ssi_release_dma_channels(struct rz_ssi_priv *ssi) static int rz_ssi_dma_request(struct rz_ssi_priv *ssi, struct device *dev) { ssi->playback.dma_ch = dma_request_chan(dev, "tx"); + if (IS_ERR_OR_NULL(ssi->playback.dma_ch)) + ssi->playback.dma_ch = NULL; + ssi->capture.dma_ch = dma_request_chan(dev, "rx"); + if (IS_ERR_OR_NULL(ssi->capture.dma_ch)) + ssi->capture.dma_ch = NULL; + if (!ssi->playback.dma_ch && !ssi->capture.dma_ch) { ssi->playback.dma_ch = dma_request_chan(dev, "rt"); - if (!ssi->playback.dma_ch) + if (IS_ERR_OR_NULL(ssi->playback.dma_ch)) { + ssi->playback.dma_ch = NULL; goto no_dma; + } ssi->dma_rt = true; } -- 2.17.1

1 1

[PATCH v2] ASoC: sh: rz-ssi: Improve error handling in rz_ssi_dma_request function
by Biju Das 14 Aug '21

14 Aug '21

The pointer returned by the dma_request_chan function is an error pointer for the error cases. But on the rz_ssi_dma_request function, it checks for NULL pointer instead. This patch fixes the issue by checking the error pointer instead of NULL pointer. Signed-off-by: Biju Das <biju.das.jz(a)bp.renesas.com> --- v1->v2: * Replaced IS_ERR_OR_NULL with IS_ERR for error pointer check, as the dma_request_chan function never returns NULL pointer for error case. --- sound/soc/sh/rz-ssi.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rz-ssi.c b/sound/soc/sh/rz-ssi.c index ea8d33ede5d2..2f9345415a15 100644 --- a/sound/soc/sh/rz-ssi.c +++ b/sound/soc/sh/rz-ssi.c @@ -676,11 +676,19 @@ static void rz_ssi_release_dma_channels(struct rz_ssi_priv *ssi) static int rz_ssi_dma_request(struct rz_ssi_priv *ssi, struct device *dev) { ssi->playback.dma_ch = dma_request_chan(dev, "tx"); + if (IS_ERR(ssi->playback.dma_ch)) + ssi->playback.dma_ch = NULL; + ssi->capture.dma_ch = dma_request_chan(dev, "rx"); + if (IS_ERR(ssi->capture.dma_ch)) + ssi->capture.dma_ch = NULL; + if (!ssi->playback.dma_ch && !ssi->capture.dma_ch) { ssi->playback.dma_ch = dma_request_chan(dev, "rt"); - if (!ssi->playback.dma_ch) + if (IS_ERR(ssi->playback.dma_ch)) { + ssi->playback.dma_ch = NULL; goto no_dma; + } ssi->dma_rt = true; } -- 2.17.1

1 0

unable to play audio in a application running as root.
by GitHub issues - opened 14 Aug '21

14 Aug '21

alsa-project/alsa-lib issue #170 was opened from PeaceNIRA: ALSA lib pcm_dmix.c:1075:(snd_pcm_dmix_open) unable to open slave what ive tried: unistall and reinstalling the app adding root to the audio group application: war thunder Issue URL : https://github.com/alsa-project/alsa-lib/issues/170 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH v4 0/3] Add RZ/G2L Sound support
by Biju Das 14 Aug '21

14 Aug '21

This patch series aims to add ASoC support on RZ/G2L SoC's. It is based on the work done by Chris Brandt for RZ/A ASoC driver. v3->v4: * Updated the subject line as per style for the subsystem. * Removed select SND_SIMPLE_CARD from Kconfig * Added C++ comments for copyright and driver description. * Moved validation of channels in hw_params * removed asm issue reported by bot as well as Mark * replaced master/slave macros with provider/consumer macros * Improved locking and added more null pointer checks. v2->v3: * Fixed the dependency on KCONFIG * Merged the binding patch with dma feature added * Updated dt binding example with encoded #dma-cells value. * Improved Error handling in probe function * Removed the passing legacy channel configuration parameters from dmaengine_slave_config function * started using dma_request_chan instead of deprecated dma_request_slave_channel * Removed SoC dtsi and config patches from this series. Will send it later. v1->v2: * Rebased to latest rc kernel Biju Das (3): ASoC: sh: Add RZ/G2L SSIF-2 driver ASoC: dt-bindings: sound: renesas,rz-ssi: Update slave dma channel configuration parameters ASoC: sh: rz-ssi: Add SSI DMAC support .../bindings/sound/renesas,rz-ssi.yaml | 4 +- sound/soc/sh/Kconfig | 6 + sound/soc/sh/Makefile | 4 + sound/soc/sh/rz-ssi.c | 1069 +++++++++++++++++ 4 files changed, 1081 insertions(+), 2 deletions(-) create mode 100644 sound/soc/sh/rz-ssi.c -- 2.17.1

4 10

[PATCH] ALSA: pcm: Add SNDRV_PCM_INFO_EXPLICIT_SYNC flag
by Takashi Iwai 14 Aug '21

14 Aug '21

ALSA PCM core has an optimized way to communicate with user-space for its control and status data via mmap on the supported architectures like x86. Depending on the situation, however, we'd rather want to enforce user-space notifying the applptr or hwptr change explicitly via ioctl. For example, the upcoming non-contig and non-coherent buffer handling would need an explicit sync, and this needs to catch the applptr and hwptr changes. Also, ASoC SOF driver will have the SPIB support that has the similar requirement for the explicit control of the applptr and hwptr. This patch adds the new PCM hardware info flag, SNDRV_PCM_INFO_EXPLICIT_SYNC. When this flag is set, PCM core disables both the control and the status mmap, which enforces user-space to update via SYNC_PTR ioctl. In that way, drivers can catch the applptr and hwptr update and apply the sync operation if needed. Link: https://lore.kernel.org/r/20210812113818.6479-1-tiwai@suse.de Link: https://lore.kernel.org/r/20210610205326.1176400-1-pierre-louis.bossart@lin… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/uapi/sound/asound.h | 1 + sound/core/pcm_native.c | 9 +++++++++ 2 files changed, 10 insertions(+) diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index d17c061950df..1d84ec9db93b 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -299,6 +299,7 @@ typedef int __bitwise snd_pcm_subformat_t; #define SNDRV_PCM_INFO_HAS_LINK_ABSOLUTE_ATIME 0x02000000 /* report absolute hardware link audio time, not reset on startup */ #define SNDRV_PCM_INFO_HAS_LINK_ESTIMATED_ATIME 0x04000000 /* report estimated link audio time */ #define SNDRV_PCM_INFO_HAS_LINK_SYNCHRONIZED_ATIME 0x08000000 /* report synchronized audio/system time */ +#define SNDRV_PCM_INFO_EXPLICIT_SYNC 0x10000000 /* needs explicit sync of pointers and data */ #define SNDRV_PCM_INFO_DRAIN_TRIGGER 0x40000000 /* internal kernel flag - trigger in drain */ #define SNDRV_PCM_INFO_FIFO_IN_FRAMES 0x80000000 /* internal kernel flag - FIFO size is in frames */ diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index dc9fa312fadd..d233cb3b41d8 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3621,6 +3621,12 @@ static int snd_pcm_mmap_control(struct snd_pcm_substream *substream, struct file static bool pcm_status_mmap_allowed(struct snd_pcm_file *pcm_file) { + /* If drivers require the explicit sync (typically for non-coherent + * pages), we have to disable the mmap of status and control data + * to enforce the control via SYNC_PTR ioctl. + */ + if (pcm_file->substream->runtime->hw.info & SNDRV_PCM_INFO_EXPLICIT_SYNC) + return false; /* See pcm_control_mmap_allowed() below. * Since older alsa-lib requires both status and control mmaps to be * coupled, we have to disable the status mmap for old alsa-lib, too. @@ -3635,6 +3641,9 @@ static bool pcm_control_mmap_allowed(struct snd_pcm_file *pcm_file) { if (pcm_file->no_compat_mmap) return false; + /* see above */ + if (pcm_file->substream->runtime->hw.info & SNDRV_PCM_INFO_EXPLICIT_SYNC) + return false; /* Disallow the control mmap when SYNC_APPLPTR flag is set; * it enforces the user-space to fall back to snd_pcm_sync_ptr(), * thus it effectively assures the manual update of appl_ptr. -- 2.26.2

1 1

[PATCH v2] usb-audio: Input source control - digidesign mbox
by Damien Zammit 14 Aug '21

14 Aug '21

This adds a second mixer control to Digidesign Mbox to select between Analog/SPDIF capture. Users will note that selecting the SPDIF input source automatically switches the clock mode to sync to SPDIF, which is a feature of the hardware. (You can change the clock source back to internal if you want to capture from spdif but not sync to its clock although this mode is probably not recommended). Unfortunately, starting the stream resets both modes to Internally clocked and Analog input mode. Signed-off-by: Damien Zammit <damien(a)zamaudio.com> Tested-by: Damien Zammit <damien(a)zamaudio.com> --- sound/usb/mixer_quirks.c | 265 ++++++++++++++++++++++++++++++++++++----------- 1 file changed, 203 insertions(+), 62 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 0a3cb8fd7..a66ce0375 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -594,85 +594,208 @@ static int snd_xonar_u1_controls_create(struct usb_mixer_interface *mixer) &snd_xonar_u1_output_switch, NULL); } +/* Digidesign Mbox 1 helper functions */ + +static int snd_mbox1_is_spdif_synced(struct snd_usb_audio *chip) +{ + unsigned char buff[3]; + int err; + int is_spdif_synced; + + /* Read clock source */ + err = snd_usb_ctl_msg(chip->dev, + usb_rcvctrlpipe(chip->dev, 0), 0x81, + USB_DIR_IN | + USB_TYPE_CLASS | + USB_RECIP_ENDPOINT, 0x100, 0x81, buff, 3); + if (err < 0) + return err; + + /* spdif sync: buff is all zeroes */ + is_spdif_synced = !(buff[0] | buff[1] | buff[2]); + return is_spdif_synced; +} + +static int snd_mbox1_set_clk_source(struct snd_usb_audio *chip, int rate_or_zero) +{ + /* 2 possibilities: Internal -> expects sample rate + * S/PDIF sync -> expects rate = 0 + */ + unsigned char buff[3]; + + buff[0] = (rate_or_zero >> 0) & 0xff; + buff[1] = (rate_or_zero >> 8) & 0xff; + buff[2] = (rate_or_zero >> 16) & 0xff; + + /* Set clock source */ + return snd_usb_ctl_msg(chip->dev, + usb_sndctrlpipe(chip->dev, 0), 0x1, + USB_TYPE_CLASS | + USB_RECIP_ENDPOINT, 0x100, 0x81, buff, 3); +} + +static int snd_mbox1_is_spdif_input(struct snd_usb_audio *chip) +{ + /* Hardware gives 2 possibilities: ANALOG Source -> 0x01 + * S/PDIF Source -> 0x02 + */ + int err; + unsigned char source[1]; + + /* Read input source */ + err = snd_usb_ctl_msg(chip->dev, + usb_rcvctrlpipe(chip->dev, 0), 0x81, + USB_DIR_IN | + USB_TYPE_CLASS | + USB_RECIP_INTERFACE, 0x00, 0x500, source, 1); + if (err < 0) + return err; + + return (source[0] == 2); +} + +static int snd_mbox1_set_input_source(struct snd_usb_audio *chip, int is_spdif) +{ + /* NB: Setting the input source to S/PDIF resets the clock source to S/PDIF + * Hardware expects 2 possibilities: ANALOG Source -> 0x01 + * S/PDIF Source -> 0x02 + */ + unsigned char buff[1]; + + buff[0] = (is_spdif & 1) + 1; + + /* Set input source */ + return snd_usb_ctl_msg(chip->dev, + usb_sndctrlpipe(chip->dev, 0), 0x1, + USB_TYPE_CLASS | + USB_RECIP_INTERFACE, 0x00, 0x500, buff, 1); +} + /* Digidesign Mbox 1 clock source switch (internal/spdif) */ -static int snd_mbox1_switch_get(struct snd_kcontrol *kctl, - struct snd_ctl_elem_value *ucontrol) +static int snd_mbox1_clk_switch_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) { + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kctl); + struct snd_usb_audio *chip = list->mixer->chip; + int err; + + err = snd_usb_lock_shutdown(chip); + if (err < 0) + goto err; + + err = snd_mbox1_is_spdif_synced(chip); + if (err < 0) + goto err; + + kctl->private_value = err; + err = 0; ucontrol->value.enumerated.item[0] = kctl->private_value; - return 0; +err: + snd_usb_unlock_shutdown(chip); + return err; } -static int snd_mbox1_switch_update(struct usb_mixer_interface *mixer, int val) +static int snd_mbox1_clk_switch_update(struct usb_mixer_interface *mixer, int is_spdif_sync) { struct snd_usb_audio *chip = mixer->chip; int err; - unsigned char buff[3]; err = snd_usb_lock_shutdown(chip); if (err < 0) return err; - /* Prepare for magic command to toggle clock source */ - err = snd_usb_ctl_msg(chip->dev, - usb_rcvctrlpipe(chip->dev, 0), 0x81, - USB_DIR_IN | - USB_TYPE_CLASS | - USB_RECIP_INTERFACE, 0x00, 0x500, buff, 1); + err = snd_mbox1_is_spdif_input(chip); if (err < 0) goto err; - err = snd_usb_ctl_msg(chip->dev, - usb_rcvctrlpipe(chip->dev, 0), 0x81, - USB_DIR_IN | - USB_TYPE_CLASS | - USB_RECIP_ENDPOINT, 0x100, 0x81, buff, 3); + + err = snd_mbox1_is_spdif_synced(chip); if (err < 0) goto err; - /* 2 possibilities: Internal -> send sample rate - * S/PDIF sync -> send zeroes - * NB: Sample rate locked to 48kHz on purpose to - * prevent user from resetting the sample rate - * while S/PDIF sync is enabled and confusing - * this configuration. - */ - if (val == 0) { - buff[0] = 0x80; - buff[1] = 0xbb; - buff[2] = 0x00; - } else { - buff[0] = buff[1] = buff[2] = 0x00; - } + /* FIXME: hardcoded sample rate */ + err = snd_mbox1_set_clk_source(chip, is_spdif_sync ? 0 : 48000); + if (err < 0) + goto err; - /* Send the magic command to toggle the clock source */ - err = snd_usb_ctl_msg(chip->dev, - usb_sndctrlpipe(chip->dev, 0), 0x1, - USB_TYPE_CLASS | - USB_RECIP_ENDPOINT, 0x100, 0x81, buff, 3); + err = snd_mbox1_is_spdif_synced(chip); +err: + snd_usb_unlock_shutdown(chip); + return err; +} + +static int snd_mbox1_clk_switch_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kctl); + struct usb_mixer_interface *mixer = list->mixer; + int err; + bool cur_val, new_val; + + cur_val = kctl->private_value; + new_val = ucontrol->value.enumerated.item[0]; + if (cur_val == new_val) + return 0; + + kctl->private_value = new_val; + err = snd_mbox1_clk_switch_update(mixer, new_val); + return err < 0 ? err : 1; +} + +static int snd_mbox1_clk_switch_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const texts[2] = { + "Internal", + "S/PDIF" + }; + + return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(texts), texts); +} + +static int snd_mbox1_clk_switch_resume(struct usb_mixer_elem_list *list) +{ + return snd_mbox1_clk_switch_update(list->mixer, list->kctl->private_value); +} + +/* Digidesign Mbox 1 input source switch (analog/spdif) */ + +static int snd_mbox1_src_switch_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.enumerated.item[0] = kctl->private_value; + return 0; +} + +static int snd_mbox1_src_switch_update(struct usb_mixer_interface *mixer, int is_spdif_input) +{ + struct snd_usb_audio *chip = mixer->chip; + int err; + + err = snd_usb_lock_shutdown(chip); + if (err < 0) + return err; + + err = snd_mbox1_is_spdif_input(chip); if (err < 0) goto err; - err = snd_usb_ctl_msg(chip->dev, - usb_rcvctrlpipe(chip->dev, 0), 0x81, - USB_DIR_IN | - USB_TYPE_CLASS | - USB_RECIP_ENDPOINT, 0x100, 0x81, buff, 3); + + err = snd_mbox1_set_input_source(chip, is_spdif_input); if (err < 0) goto err; - err = snd_usb_ctl_msg(chip->dev, - usb_rcvctrlpipe(chip->dev, 0), 0x81, - USB_DIR_IN | - USB_TYPE_CLASS | - USB_RECIP_ENDPOINT, 0x100, 0x2, buff, 3); + + err = snd_mbox1_is_spdif_input(chip); if (err < 0) goto err; + err = snd_mbox1_is_spdif_synced(chip); err: snd_usb_unlock_shutdown(chip); return err; } -static int snd_mbox1_switch_put(struct snd_kcontrol *kctl, - struct snd_ctl_elem_value *ucontrol) +static int snd_mbox1_src_switch_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_elem_list *list = snd_kcontrol_chip(kctl); struct usb_mixer_interface *mixer = list->mixer; @@ -685,42 +808,60 @@ static int snd_mbox1_switch_put(struct snd_kcontrol *kctl, return 0; kctl->private_value = new_val; - err = snd_mbox1_switch_update(mixer, new_val); + err = snd_mbox1_src_switch_update(mixer, new_val); return err < 0 ? err : 1; } -static int snd_mbox1_switch_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) +static int snd_mbox1_src_switch_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) { static const char *const texts[2] = { - "Internal", + "Analog", "S/PDIF" }; return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(texts), texts); } -static int snd_mbox1_switch_resume(struct usb_mixer_elem_list *list) +static int snd_mbox1_src_switch_resume(struct usb_mixer_elem_list *list) { - return snd_mbox1_switch_update(list->mixer, list->kctl->private_value); + return snd_mbox1_src_switch_update(list->mixer, list->kctl->private_value); } -static const struct snd_kcontrol_new snd_mbox1_switch = { +static const struct snd_kcontrol_new snd_mbox1_clk_switch = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Clock Source", .index = 0, .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, - .info = snd_mbox1_switch_info, - .get = snd_mbox1_switch_get, - .put = snd_mbox1_switch_put, + .info = snd_mbox1_clk_switch_info, + .get = snd_mbox1_clk_switch_get, + .put = snd_mbox1_clk_switch_put, + .private_value = 0 +}; + +static const struct snd_kcontrol_new snd_mbox1_src_switch = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Input Source", + .index = 1, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_mbox1_src_switch_info, + .get = snd_mbox1_src_switch_get, + .put = snd_mbox1_src_switch_put, .private_value = 0 }; -static int snd_mbox1_create_sync_switch(struct usb_mixer_interface *mixer) +static int snd_mbox1_controls_create(struct usb_mixer_interface *mixer) { - return add_single_ctl_with_resume(mixer, 0, - snd_mbox1_switch_resume, - &snd_mbox1_switch, NULL); + int err; + err = add_single_ctl_with_resume(mixer, 0, + snd_mbox1_clk_switch_resume, + &snd_mbox1_clk_switch, NULL); + if (err < 0) + return err; + + return add_single_ctl_with_resume(mixer, 1, + snd_mbox1_src_switch_resume, + &snd_mbox1_src_switch, NULL); } /* Native Instruments device quirks */ @@ -3029,7 +3170,7 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) break; case USB_ID(0x0dba, 0x1000): /* Digidesign Mbox 1 */ - err = snd_mbox1_create_sync_switch(mixer); + err = snd_mbox1_controls_create(mixer); break; case USB_ID(0x17cc, 0x1011): /* Traktor Audio 6 */ -- 2.13.1

2 1

[PATCH v2 0/4] ASoC: SOF: Intel: DMI L1 power optimization for HDaudio platforms
by Pierre-Louis Bossart 13 Aug '21

13 Aug '21

This patchset provides an optimization that result in significant power savings on Intel HDAudio platforms using SOF (Sound Open Firmware). We previously prevented the Intel DSP from enabling the DMI_L1 capability to work-around issues with pause on capture streams. It turns out that this also prevented the platform from entering high C states in full-duplex usages such as videoconferencing - a rather basic use case since the start of the pandemic. The support for pause_push/release was already a bit controversial for Intel platforms, in theory platforms should only enable PAUSE if they can resume on the same sample, which is not the case on any Intel platform. With this patchset, when the user enables DMI L1 via a kernel parameter, the PAUSE support is disabled for capture streams. A kernel parameter is far from ideal but it's a placeholder until we have an API to negotiate capabilities between applications and driver, and it's far less confusing than a Kconfig option. Changes since v1: Removal of SPIB support since it may conflict with Takashi's memalloc changes. These SPIB changes will be provided after rebase. Addition of one cleanup for cppcheck warning Move all changes to intel/ directory, no changes in shared code Flipped the logic: the selection of DMI L1 disables PAUSE Pierre-Louis Bossart (4): ASoC: SOF: Intel: Kconfig: clarify DMI L1 option description ASoC: SOF: Intel: hda-stream: remove always true condition ASoC: SOF: Intel: simplify logic for DMI_L1 handling ASoC: SOF: Intel: make DMI L1 selection more robust sound/soc/sof/intel/Kconfig | 10 ---------- sound/soc/sof/intel/hda-pcm.c | 16 ++++++++++++++-- sound/soc/sof/intel/hda-stream.c | 11 +++++------ 3 files changed, 19 insertions(+), 18 deletions(-) -- 2.25.1

2 5

[PATCH v5 0/3] Add RZ/G2L Sound support
by Biju Das 13 Aug '21

13 Aug '21

This patch series aims to add ASoC support on RZ/G2L SoC's. It is based on the work done by Chris Brandt for RZ/A ASoC driver. v4->v5 * Moved validation of sample bits in hw_params * Removed validation of frame bits as it is redundant * split the rz_ssi_start_stop function into rz_ssi_start and rz_ssi_stop. * remove the spin_lock around rz_ssi_stream_init. * Updated dmas description and removed fixes as it is an enhancement now. * updated ssi_start functions with setting fifo thresholds and ssi_stop function with cancel all dma txn. v3->v4: * Updated the subject line as per style for the subsystem. * Removed select SND_SIMPLE_CARD from Kconfig * Added C++ comments for copyright and driver description. * Moved validation of channels in hw_params * removed asm issue reported by bot as well as Mark * replaced master/slave macros with provider/consumer macros * Improved locking and added more null pointer checks. v2->v3: * Fixed the dependency on KCONFIG * Merged the binding patch with dma feature added * Updated dt binding example with encoded #dma-cells value. * Improved Error handling in probe function * Removed the passing legacy channel configuration parameters from dmaengine_slave_config function * started using dma_request_chan instead of deprecated dma_request_slave_channel * Removed SoC dtsi and config patches from this series. Will send it later. v1->v2: * Rebased to latest rc kernel Biju Das (3): ASoC: sh: Add RZ/G2L SSIF-2 driver ASoC: dt-bindings: renesas,rz-ssi: Update slave dma channel configuration parameters ASoC: sh: rz-ssi: Add SSI DMAC support .../bindings/sound/renesas,rz-ssi.yaml | 22 +- sound/soc/sh/Kconfig | 6 + sound/soc/sh/Makefile | 4 + sound/soc/sh/rz-ssi.c | 1063 +++++++++++++++++ 4 files changed, 1093 insertions(+), 2 deletions(-) create mode 100644 sound/soc/sh/rz-ssi.c -- 2.17.1

2 4

[PATCH] ASoC: rt5682: Remove unused variable in rt5682_i2c_remove()
by Douglas Anderson 13 Aug '21

13 Aug '21

In commit 772d44526e20 ("ASoC: rt5682: Properly turn off regulators if wrong device ID") I deleted code but forgot to delete a variable that's now unused. Delete it. Fixes: 772d44526e20 ("ASoC: rt5682: Properly turn off regulators if wrong device ID") Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Douglas Anderson <dianders(a)chromium.org> --- sound/soc/codecs/rt5682-i2c.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/rt5682-i2c.c b/sound/soc/codecs/rt5682-i2c.c index e559b965a0a6..b9d5d7a0975b 100644 --- a/sound/soc/codecs/rt5682-i2c.c +++ b/sound/soc/codecs/rt5682-i2c.c @@ -294,8 +294,6 @@ static void rt5682_i2c_shutdown(struct i2c_client *client) static int rt5682_i2c_remove(struct i2c_client *client) { - struct rt5682_priv *rt5682 = i2c_get_clientdata(client); - rt5682_i2c_shutdown(client); return 0; -- 2.33.0.rc1.237.g0d66db33f3-goog

3 2

USB-Audio: Device or resource busy (strace log)
by Muni Sekhar 13 Aug '21

13 Aug '21

Hi All, $ cat /proc/asound/cards 0 [USB ]: USB-Audio - Plantronics .Audio 628 USB Plantronics Plantronics .Audio 628 USB at usb-0000:00:14.0-2, full speed I am using a Plantronics USB Audio headset. $ arecord --device hw:0,0 --channels 2 --format S16_LE --rate 44100Hz x.wav arecord: main:722: audio open error: Device or resource busy 'arecord' command always fails the first time after system boot in my system. But subsequent execution of the 'arecord' command runs fine. I've attached the strace log for the "audio open error: Device or resource busy" failure. Is there any fix available for this issue? -- Thanks, Sekhar

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