December 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH AUTOSEL 5.4 197/350] ASoC: SOF: imx: fix reverse CONFIG_SND_SOC_SOF_OF dependency
by Sasha Levin 19 Dec '19

19 Dec '19

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit f9ad75468453b019b92c5296e6a04bf7c37f49e4 ] updated solution to the problem reported with randconfig: CONFIG_SND_SOC_SOF_IMX depends on CONFIG_SND_SOC_SOF, but is in turn referenced by the sof-of-dev driver. This creates a reverse dependency that manifests in a link error when CONFIG_SND_SOC_SOF_OF is built-in but CONFIG_SND_SOC_SOF_IMX=m: sound/soc/sof/sof-of-dev.o:(.data+0x118): undefined reference to `sof_imx8_ops' use def_trisate to propagate the right settings without select. Fixes: f4df4e4042b0 ("ASoC: SOF: imx8: Fix COMPILE_TEST error") Fixes: 202acc565a1f ("ASoC: SOF: imx: Add i.MX8 HW support") Suggested-by: Arnd Bergmann <arnd(a)arndb.de> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20191101173045.27099-6-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/imx/Kconfig | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/imx/Kconfig b/sound/soc/sof/imx/Kconfig index 5acae75f5750d..71f318bc2c74f 100644 --- a/sound/soc/sof/imx/Kconfig +++ b/sound/soc/sof/imx/Kconfig @@ -11,8 +11,8 @@ config SND_SOC_SOF_IMX_TOPLEVEL if SND_SOC_SOF_IMX_TOPLEVEL -config SND_SOC_SOF_IMX8 - tristate "SOF support for i.MX8" +config SND_SOC_SOF_IMX8_SUPPORT + bool "SOF support for i.MX8" depends on IMX_SCU depends on IMX_DSP help @@ -20,4 +20,8 @@ config SND_SOC_SOF_IMX8 Say Y if you have such a device. If unsure select "N". +config SND_SOC_SOF_IMX8 + def_tristate SND_SOC_SOF_OF + depends on SND_SOC_SOF_IMX8_SUPPORT + endif ## SND_SOC_SOF_IMX_IMX_TOPLEVEL -- 2.20.1

2 2

[alsa-devel] [PATCH v9 0/2] ASoC: codecs: Add WSA881x Smart Speaker amplifier support
by Srinivas Kandagatla 19 Dec '19

19 Dec '19

This patchset adds support to WSA8810/WSA8815 Class-D Smart Speaker Amplifier which is SoundWire interfaced. This patchset along with DB845c machine driver and WCD934x codec driver has been tested on SDM845 SoC based DragonBoard DB845c with two WSA8810 and Lenovo YOGA C630 Laptop based on SDM850 with WSA8815 speaker amplifiers. Most of the code in this driver is rework of Qualcomm downstream drivers used in Andriod. Credits to Banajit Goswami and Patrick Lai's Team. TODO: Add thermal sensor support in WSA881x. Thanks, srini Changes since v8: - Moved soundwire stream handling into this codec as there is strong requirements on sequence of PA and soundwire port enables from hw side. Srinivas Kandagatla (2): dt-bindings: ASoC: Add WSA881x bindings ASoC: codecs: add wsa881x amplifier support .../bindings/sound/qcom,wsa881x.yaml | 68 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wsa881x.c | 1206 +++++++++++++++++ 4 files changed, 1286 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wsa881x.yaml create mode 100644 sound/soc/codecs/wsa881x.c -- 2.21.0

2 4

[alsa-devel] [PATCH v14 7/7] ASoC: amd MMAP_INTERLEAVED Support
by Ravulapati Vishnu vardhan rao 19 Dec '19

19 Dec '19

ACP-I2S device support MMAP_INTERLEAVED. Added support for the same. Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 916f0d8..28a3081 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -22,6 +22,7 @@ static const struct snd_pcm_hardware acp3x_pcm_hardware_playback = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_BATCH | + SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S24_LE | @@ -42,7 +43,8 @@ static const struct snd_pcm_hardware acp3x_pcm_hardware_capture = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_BATCH | - SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, + SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE, -- 2.7.4

3 2

[alsa-devel] [PATCH] ASoC: max98090: Remove empty suspend function
by Mark Brown 19 Dec '19

19 Dec '19

The suspend function is empty so can be removed. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/max98090.c | 7 +------ 1 file changed, 1 insertion(+), 6 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index da23810f958e..68271042f586 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2838,15 +2838,10 @@ static int max98090_resume(struct device *dev) return 0; } - -static int max98090_suspend(struct device *dev) -{ - return 0; -} #endif static const struct dev_pm_ops max98090_pm = { - SET_RUNTIME_PM_OPS(max98090_runtime_suspend, + SET_RUNTIME_PM_OPS(NULL, max98090_runtime_resume, NULL) SET_SYSTEM_SLEEP_PM_OPS(max98090_suspend, max98090_resume) }; -- 2.20.1

2 2

[alsa-devel] [PATCH] ALSA: hda: Use standard waitqueue for RIRB wakeup
by Takashi Iwai 19 Dec '19

19 Dec '19

The HD-audio CORB/RIRB communication was programmed in a way that was documented in the reference in decades ago, which is essentially a polling in the waiter side. It's working fine but costs CPU cycles on some platforms that support only slow communications. Also, for some platforms that had unreliable communications, we put longer wait time (2 ms), which accumulate quite long time if you execute many verbs in a shot (e.g. at the initialization or resume phase). This patch attempts to improve the situation by introducing the standard waitqueue in the RIRB waiter side instead of polling. The test results on my machine show significant improvements. The time spent for "cat /proc/asound/card*/codec#*" were changed like: * Intel SKL + Realtek codec before the patch: 0.00user 0.04system 0:00.10elapsed 40.0%CPU after the patch: 0.00user 0.01system 0:00.10elapsed 10.0%CPU * Nvidia GP107GL + Nvidia HDMI codec before the patch: 0.00user 0.00system 0:02.76elapsed 0.0%CPU after the patch: 0.00user 0.00system 0:00.01elapsed 17.0%CPU So, for Intel chips, the total time is same, while the total time is greatly reduced (from 2.76 to 0.01s) for Nvidia chips. The only negative data here is the increase of CPU time for Nvidia, but this is the unavoidable cost for faster wakeups, supposedly. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/hdaudio.h | 1 + sound/hda/hdac_bus.c | 1 + sound/hda/hdac_controller.c | 3 +++ sound/pci/hda/hda_controller.c | 29 ++++++++++++++--------------- 4 files changed, 19 insertions(+), 15 deletions(-) diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index e05b95e83d5a..81373a2efd96 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -317,6 +317,7 @@ struct hdac_bus { struct hdac_rb corb; struct hdac_rb rirb; unsigned int last_cmd[HDA_MAX_CODECS]; /* last sent command */ + wait_queue_head_t rirb_wq; /* CORB/RIRB and position buffers */ struct snd_dma_buffer rb; diff --git a/sound/hda/hdac_bus.c b/sound/hda/hdac_bus.c index 8f19876244eb..48b227fff204 100644 --- a/sound/hda/hdac_bus.c +++ b/sound/hda/hdac_bus.c @@ -43,6 +43,7 @@ int snd_hdac_bus_init(struct hdac_bus *bus, struct device *dev, mutex_init(&bus->cmd_mutex); mutex_init(&bus->lock); INIT_LIST_HEAD(&bus->hlink_list); + init_waitqueue_head(&bus->rirb_wq); bus->irq = -1; return 0; } diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 7e7be8e4dcf9..cd1c3b282657 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -216,6 +216,9 @@ void snd_hdac_bus_update_rirb(struct hdac_bus *bus) else if (bus->rirb.cmds[addr]) { bus->rirb.res[addr] = res; bus->rirb.cmds[addr]--; + if (!bus->rirb.cmds[addr] && + waitqueue_active(&bus->rirb_wq)) + wake_up(&bus->rirb_wq); } else { dev_err_ratelimited(bus->dev, "spurious response %#x:%#x, last cmd=%#08x\n", diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 2f3b7a35f2d9..f30a053d981e 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -792,21 +792,25 @@ static int azx_rirb_get_response(struct hdac_bus *bus, unsigned int addr, struct hda_bus *hbus = &chip->bus; unsigned long timeout; unsigned long loopcounter; - int do_poll = 0; + wait_queue_entry_t wait; bool warned = false; + init_wait_entry(&wait, 0); again: timeout = jiffies + msecs_to_jiffies(1000); for (loopcounter = 0;; loopcounter++) { spin_lock_irq(&bus->reg_lock); - if (bus->polling_mode || do_poll) + if (!bus->polling_mode) + prepare_to_wait(&bus->rirb_wq, &wait, + TASK_UNINTERRUPTIBLE); + if (bus->polling_mode) snd_hdac_bus_update_rirb(bus); if (!bus->rirb.cmds[addr]) { - if (!do_poll) - bus->poll_count = 0; if (res) *res = bus->rirb.res[addr]; /* the last value */ + if (!bus->polling_mode) + finish_wait(&bus->rirb_wq, &wait); spin_unlock_irq(&bus->reg_lock); return 0; } @@ -814,7 +818,9 @@ static int azx_rirb_get_response(struct hdac_bus *bus, unsigned int addr, if (time_after(jiffies, timeout)) break; #define LOOP_COUNT_MAX 3000 - if (hbus->needs_damn_long_delay || + if (!bus->polling_mode) { + schedule_timeout(msecs_to_jiffies(2)); + } else if (hbus->needs_damn_long_delay || loopcounter > LOOP_COUNT_MAX) { if (loopcounter > LOOP_COUNT_MAX && !warned) { dev_dbg_ratelimited(chip->card->dev, @@ -829,19 +835,12 @@ static int azx_rirb_get_response(struct hdac_bus *bus, unsigned int addr, } } + if (!bus->polling_mode) + finish_wait(&bus->rirb_wq, &wait); + if (hbus->no_response_fallback) return -EIO; - if (!bus->polling_mode && bus->poll_count < 2) { - dev_dbg(chip->card->dev, - "azx_get_response timeout, polling the codec once: last cmd=0x%08x\n", - bus->last_cmd[addr]); - do_poll = 1; - bus->poll_count++; - goto again; - } - - if (!bus->polling_mode) { dev_warn(chip->card->dev, "azx_get_response timeout, switching to polling mode: last cmd=0x%08x\n", -- 2.16.4

3 9

[alsa-devel] [PATCH] slimbus: qcom-ngd-ctrl: Use dma_request_chan() instead dma_request_slave_channel()
by Peter Ujfalusi 19 Dec '19

19 Dec '19

dma_request_slave_channel() is a wrapper on top of dma_request_chan() eating up the error code. By using dma_request_chan() directly the driver can support deferred probing against DMA. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- drivers/slimbus/qcom-ngd-ctrl.c | 20 ++++++++++++-------- 1 file changed, 12 insertions(+), 8 deletions(-) diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index 29fbab55c3b3..e3f5ebc0c05e 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -666,10 +666,12 @@ static int qcom_slim_ngd_init_rx_msgq(struct qcom_slim_ngd_ctrl *ctrl) struct device *dev = ctrl->dev; int ret, size; - ctrl->dma_rx_channel = dma_request_slave_channel(dev, "rx"); - if (!ctrl->dma_rx_channel) { - dev_err(dev, "Failed to request dma channels"); - return -EINVAL; + ctrl->dma_rx_channel = dma_request_chan(dev, "rx"); + if (IS_ERR(ctrl->dma_rx_channel)) { + dev_err(dev, "Failed to request RX dma channel"); + ret = PTR_ERR(ctrl->dma_rx_channel); + ctrl->dma_rx_channel = NULL; + return ret; } size = QCOM_SLIM_NGD_DESC_NUM * SLIM_MSGQ_BUF_LEN; @@ -703,10 +705,12 @@ static int qcom_slim_ngd_init_tx_msgq(struct qcom_slim_ngd_ctrl *ctrl) int ret = 0; int size; - ctrl->dma_tx_channel = dma_request_slave_channel(dev, "tx"); - if (!ctrl->dma_tx_channel) { - dev_err(dev, "Failed to request dma channels"); - return -EINVAL; + ctrl->dma_tx_channel = dma_request_chan(dev, "tx"); + if (IS_ERR(ctrl->dma_tx_channel)) { + dev_err(dev, "Failed to request TX dma channel"); + ret = PTR_ERR(ctrl->dma_tx_channel); + ctrl->dma_tx_channel = NULL; + return ret; } size = ((QCOM_SLIM_NGD_DESC_NUM + 1) * SLIM_MSGQ_BUF_LEN); -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: set the interface format after resume on Dell WD19
by Hui Wang 19 Dec '19

19 Dec '19

Recently we found the headset-mic on the Dell Dock WD19 doesn't work anymore after s3 (s2i or deep), this problem could be workarounded by closeing (pcm_close) the app and then reopening (pcm_open) the app, so this bug is not easy to be detected by users. When problem happens, retire_capture_urb() could still be called periodically, but the size of captured data is always 0, it could be a firmware bug on the dock. Anyway I found after resuming, the snd_usb_pcm_prepare() will be called, and if we forcibly run set_format() to set the interface and its endpoint, the capture size will be normal again. This problem and workaound also apply to playback. To fix it in the kernel, add a quirk to let set_format() run forcibly once after resume. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/usb/card.h | 1 + sound/usb/pcm.c | 21 +++++++++++++++++++-- sound/usb/quirks-table.h | 3 ++- sound/usb/quirks.c | 11 +++++++++++ sound/usb/usbaudio.h | 3 ++- 5 files changed, 35 insertions(+), 4 deletions(-) diff --git a/sound/usb/card.h b/sound/usb/card.h index 2991b9986f66..395403a2d33f 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -145,6 +145,7 @@ struct snd_usb_substream { struct snd_usb_endpoint *sync_endpoint; unsigned long flags; bool need_setup_ep; /* (re)configure EP at prepare? */ + bool need_setup_fmt; /* (re)configure fmt after resume? */ unsigned int speed; /* USB_SPEED_XXX */ u64 formats; /* format bitmasks (all or'ed) */ diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 8a52996041e9..4ac571b2e9a6 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -523,11 +523,11 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) if (WARN_ON(altsd->bAlternateSetting != fmt->altsetting)) return -EINVAL; - if (fmt == subs->cur_audiofmt) + if (fmt == subs->cur_audiofmt && !subs->need_setup_fmt) return 0; /* close the old interface */ - if (subs->interface >= 0 && subs->interface != fmt->iface) { + if (subs->interface >= 0 && (subs->interface != fmt->iface || subs->need_setup_fmt)) { if (!subs->stream->chip->keep_iface) { err = usb_set_interface(subs->dev, subs->interface, 0); if (err < 0) { @@ -541,6 +541,9 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) subs->altset_idx = 0; } + if (subs->need_setup_fmt) + subs->need_setup_fmt = false; + /* set interface */ if (iface->cur_altsetting != alts) { err = snd_usb_select_mode_quirk(subs, fmt); @@ -1734,6 +1737,13 @@ static int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substrea subs->data_endpoint->retire_data_urb = retire_playback_urb; subs->running = 0; return 0; + case SNDRV_PCM_TRIGGER_SUSPEND: + if (subs->stream->chip->setup_fmt_after_resume_quirk) { + stop_endpoints(subs, true); + subs->need_setup_fmt = true; + return 0; + } + break; } return -EINVAL; @@ -1766,6 +1776,13 @@ static int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream subs->data_endpoint->retire_data_urb = retire_capture_urb; subs->running = 1; return 0; + case SNDRV_PCM_TRIGGER_SUSPEND: + if (subs->stream->chip->setup_fmt_after_resume_quirk) { + stop_endpoints(subs, true); + subs->need_setup_fmt = true; + return 0; + } + break; } return -EINVAL; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 70c338f3ae24..d187aa6d50db 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3466,7 +3466,8 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), .vendor_name = "Dell", .product_name = "WD19 Dock", .profile_name = "Dell-WD15-Dock", - .ifnum = QUIRK_NO_INTERFACE + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_SETUP_FMT_AFTER_RESUME } }, /* MOTU Microbook II */ diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 349e1e52996d..a81c2066499f 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -508,6 +508,16 @@ static int create_standard_mixer_quirk(struct snd_usb_audio *chip, return snd_usb_create_mixer(chip, quirk->ifnum, 0); } + +static int setup_fmt_after_resume_quirk(struct snd_usb_audio *chip, + struct usb_interface *iface, + struct usb_driver *driver, + const struct snd_usb_audio_quirk *quirk) +{ + chip->setup_fmt_after_resume_quirk = 1; + return 1; /* Continue with creating streams and mixer */ +} + /* * audio-interface quirks * @@ -546,6 +556,7 @@ int snd_usb_create_quirk(struct snd_usb_audio *chip, [QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk, [QUIRK_AUDIO_ALIGN_TRANSFER] = create_align_transfer_quirk, [QUIRK_AUDIO_STANDARD_MIXER] = create_standard_mixer_quirk, + [QUIRK_SETUP_FMT_AFTER_RESUME] = setup_fmt_after_resume_quirk, }; if (quirk->type < QUIRK_TYPE_COUNT) { diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index ff3cbf653de8..6fe3ab582ec6 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -33,7 +33,7 @@ struct snd_usb_audio { wait_queue_head_t shutdown_wait; unsigned int txfr_quirk:1; /* Subframe boundaries on transfers */ unsigned int tx_length_quirk:1; /* Put length specifier in transfers */ - + unsigned int setup_fmt_after_resume_quirk:1; /* setup the format to interface after resume */ int num_interfaces; int num_suspended_intf; int sample_rate_read_error; @@ -98,6 +98,7 @@ enum quirk_type { QUIRK_AUDIO_EDIROL_UAXX, QUIRK_AUDIO_ALIGN_TRANSFER, QUIRK_AUDIO_STANDARD_MIXER, + QUIRK_SETUP_FMT_AFTER_RESUME, QUIRK_TYPE_COUNT }; -- 2.17.1

3 6

[alsa-devel] [PATCH v3 0/8] ASoC: soc-core cleanup step8
by Kuninori Morimoto 19 Dec '19

19 Dec '19

Hi Mark # did I lost email ?? These are v3 of soc-core cleanup step8. The diff from v2 are - add Reported-by, Reviewed-By for each patches - [2/8] is added - rebase to latest for-v5.6 branch Kuninori Morimoto (8): ASoC: soc-core: remove snd_soc_rtdcom_list ASoC: soc-core: rename snd_soc_rtdcom_add() to snd_soc_rtd_add_component() ASoC: soc-core: care .ignore_suspend for Component suspend ASoC: soc-core: remove duplicate pinctrl operation when suspend ASoC: soc-core: do pinctrl_pm_select_xxx() as component ASoC: soc-core: add snd_soc_close_delayed_work() ASoC: soc-dapm: add snd_soc_dapm_stream_stop() ASoC: use snd_soc_dapm_stream_stop() for SND_SOC_DAPM_STREAM_STOP include/sound/soc-dapm.h | 1 + include/sound/soc.h | 19 +++--- sound/soc/soc-component.c | 33 +++++---- sound/soc/soc-compress.c | 124 ++++++++++------------------------ sound/soc/soc-core.c | 167 +++++++++++++++++++++++----------------------- sound/soc/soc-dapm.c | 23 +++++++ sound/soc/soc-pcm.c | 115 ++++++++----------------------- 7 files changed, 191 insertions(+), 291 deletions(-) -- 2.7.4

1 8

[alsa-devel] [PATCH v3 0/8] ASoC: soc-core cleanup step8
by Kuninori Morimoto 19 Dec '19

19 Dec '19

Hi Mark These are v3 of soc-core cleanup step8. The difference from v2 are... - add Reviewed-By, Reported-by for each patches - [2/8] is added - rebase to latest for-5.6 branch Kuninori Morimoto (8): 1) ASoC: soc-core: remove snd_soc_rtdcom_list 2) ASoC: soc-core: rename snd_soc_rtdcom_add() to snd_soc_rtd_add_component() 3) ASoC: soc-core: care .ignore_suspend for Component suspend 4) ASoC: soc-core: remove duplicate pinctrl operation when suspend 5) ASoC: soc-core: do pinctrl_pm_select_xxx() as component 6) ASoC: soc-core: add snd_soc_close_delayed_work() 7) ASoC: soc-dapm: add snd_soc_dapm_stream_stop() 8) ASoC: use snd_soc_dapm_stream_stop() for SND_SOC_DAPM_STREAM_STOP include/sound/soc-dapm.h | 1 + include/sound/soc.h | 19 +++--- sound/soc/soc-component.c | 33 +++++---- sound/soc/soc-compress.c | 124 ++++++++++------------------------ sound/soc/soc-core.c | 167 +++++++++++++++++++++++----------------------- sound/soc/soc-dapm.c | 23 +++++++ sound/soc/soc-pcm.c | 115 ++++++++----------------------- 7 files changed, 191 insertions(+), 291 deletions(-) -- 2.7.4

1 0

[alsa-devel] [PATCH 0/7 v2] ASoC: soc-core cleanup step8
by Kuninori Morimoto 19 Dec '19

19 Dec '19

Hi Mark These are v2 of soc-core cleanup step8. There is no relationship between each patches. - [6/7] renamed function name - [7/7] new patch which cares SND_SOC_DAPM_STREAM_STOP Kuninori Morimoto (7): 1) ASoC: soc-core: remove snd_soc_rtdcom_list 2) ASoC: soc-core: care .ignore_suspend for Component suspend 3) ASoC: soc-core: remove duplicate pinctrl operation when suspend 4) ASoC: soc-core: do pinctrl_pm_select_xxx() as component 5) ASoC: soc-core: add snd_soc_close_delayed_work() 6) ASoC: soc-dapm: add snd_soc_dapm_stream_stop() 7) ASoC: use snd_soc_dapm_stream_stop() for SND_SOC_DAPM_STREAM_STOP include/sound/soc-dapm.h | 1 + include/sound/soc.h | 19 +++--- sound/soc/soc-component.c | 33 +++++----- sound/soc/soc-compress.c | 112 +++++++++------------------------ sound/soc/soc-core.c | 157 +++++++++++++++++++++++----------------------- sound/soc/soc-dapm.c | 23 +++++++ sound/soc/soc-pcm.c | 115 ++++++++------------------------- 7 files changed, 181 insertions(+), 279 deletions(-) -- 2.7.4

5 17