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

presonus StudioLive 24.4.2
by m.eik michalke 18 Jun '20

18 Jun '20

hi, i'm trying to get a presonus StudioLive 24.4.2 up and running: https://www.presonus.com/products/studiolive-24.4.2 so far i'm partly successful -- looks like the board is automatically detected and supported by the firewire dice module. however, i'm only offered 16 of the 24 supported audio channels. in the list of soundcards it is merely shown as "studiolive" without further info on the model. i wasn't able to find further info on this board's status with regards to alsa, so if there's documentation on this somewhere, could someone please point me to it? in case that as of now the studiolive isn't officially supported or the above behavior is due to the fact that so far only the 16.4.2 is known to the module, i'm willing to help as far as i can ;) i hope that already seeing 16 channels is a good point to start and that it's not too hard to make the remaining 8 show up as well... best regards :: m.eik

2 8

[PATCH] ASoC: rockchip: Fix a reference count leak.
by wu000273＠umn.edu 18 Jun '20

18 Jun '20

From: Qiushi Wu <wu000273(a)umn.edu> Calling pm_runtime_get_sync increments the counter even in case of failure, causing incorrect ref count if pm_runtime_put is not called in error handling paths. Call pm_runtime_put if pm_runtime_get_sync fails. Fixes: fc05a5b22253 ("ASoC: rockchip: add support for pdm controller") Signed-off-by: Qiushi Wu <wu000273(a)umn.edu> --- sound/soc/rockchip/rockchip_pdm.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/rockchip/rockchip_pdm.c b/sound/soc/rockchip/rockchip_pdm.c index 7cd42fcfcf38..1707414cfa92 100644 --- a/sound/soc/rockchip/rockchip_pdm.c +++ b/sound/soc/rockchip/rockchip_pdm.c @@ -590,8 +590,10 @@ static int rockchip_pdm_resume(struct device *dev) int ret; ret = pm_runtime_get_sync(dev); - if (ret < 0) + if (ret < 0) { + pm_runtime_put(dev); return ret; + } ret = regcache_sync(pdm->regmap); -- 2.17.1

3 2

[PATCH 13/29] dt: fix broken links due to txt->yaml renames
by Mauro Carvalho Chehab 18 Jun '20

18 Jun '20

There are some new broken doc links due to yaml renames at DT. Developers should really run: ./scripts/documentation-file-ref-check in order to solve those issues while submitting patches. This tool can even fix most of the issues with: ./scripts/documentation-file-ref-check --fix Signed-off-by: Mauro Carvalho Chehab <mchehab+huawei(a)kernel.org> --- Documentation/devicetree/bindings/display/bridge/sii902x.txt | 2 +- .../devicetree/bindings/display/rockchip/rockchip-drm.yaml | 2 +- Documentation/devicetree/bindings/net/mediatek-bluetooth.txt | 2 +- Documentation/devicetree/bindings/sound/audio-graph-card.txt | 2 +- Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt | 2 +- Documentation/mips/ingenic-tcu.rst | 2 +- MAINTAINERS | 4 ++-- 7 files changed, 8 insertions(+), 8 deletions(-) diff --git a/Documentation/devicetree/bindings/display/bridge/sii902x.txt b/Documentation/devicetree/bindings/display/bridge/sii902x.txt index 6e14e087c0d0..0d1db3f9da84 100644 --- a/Documentation/devicetree/bindings/display/bridge/sii902x.txt +++ b/Documentation/devicetree/bindings/display/bridge/sii902x.txt @@ -37,7 +37,7 @@ Optional properties: simple-card or audio-graph-card binding. See their binding documents on how to describe the way the sii902x device is connected to the rest of the audio system: - Documentation/devicetree/bindings/sound/simple-card.txt + Documentation/devicetree/bindings/sound/simple-card.yaml Documentation/devicetree/bindings/sound/audio-graph-card.txt Note: In case of the audio-graph-card binding the used port index should be 3. diff --git a/Documentation/devicetree/bindings/display/rockchip/rockchip-drm.yaml b/Documentation/devicetree/bindings/display/rockchip/rockchip-drm.yaml index ec8ae742d4da..7204da5eb4c5 100644 --- a/Documentation/devicetree/bindings/display/rockchip/rockchip-drm.yaml +++ b/Documentation/devicetree/bindings/display/rockchip/rockchip-drm.yaml @@ -24,7 +24,7 @@ properties: description: | Should contain a list of phandles pointing to display interface port of vop devices. vop definitions as defined in - Documentation/devicetree/bindings/display/rockchip/rockchip-vop.txt + Documentation/devicetree/bindings/display/rockchip/rockchip-vop.yaml required: - compatible diff --git a/Documentation/devicetree/bindings/net/mediatek-bluetooth.txt b/Documentation/devicetree/bindings/net/mediatek-bluetooth.txt index 219bcbd0d344..9ef5bacda8c1 100644 --- a/Documentation/devicetree/bindings/net/mediatek-bluetooth.txt +++ b/Documentation/devicetree/bindings/net/mediatek-bluetooth.txt @@ -3,7 +3,7 @@ MediaTek SoC built-in Bluetooth Devices This device is a serial attached device to BTIF device and thus it must be a child node of the serial node with BTIF. The dt-bindings details for BTIF -device can be known via Documentation/devicetree/bindings/serial/8250.txt. +device can be known via Documentation/devicetree/bindings/serial/8250.yaml. Required properties: diff --git a/Documentation/devicetree/bindings/sound/audio-graph-card.txt b/Documentation/devicetree/bindings/sound/audio-graph-card.txt index 269682619a70..d5f6919a2d69 100644 --- a/Documentation/devicetree/bindings/sound/audio-graph-card.txt +++ b/Documentation/devicetree/bindings/sound/audio-graph-card.txt @@ -5,7 +5,7 @@ It is based on common bindings for device graphs. see ${LINUX}/Documentation/devicetree/bindings/graph.txt Basically, Audio Graph Card property is same as Simple Card. -see ${LINUX}/Documentation/devicetree/bindings/sound/simple-card.txt +see ${LINUX}/Documentation/devicetree/bindings/sound/simple-card.yaml Below are same as Simple-Card. diff --git a/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt b/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt index 4d51f3f5ea98..a6ffcdec6f6a 100644 --- a/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt +++ b/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt @@ -5,7 +5,7 @@ codec or external codecs. sti sound drivers allows to expose sti SoC audio interface through the generic ASoC simple card. For details about sound card declaration please refer to -Documentation/devicetree/bindings/sound/simple-card.txt. +Documentation/devicetree/bindings/sound/simple-card.yaml. 1) sti-uniperiph-dai: audio dai device. --------------------------------------- diff --git a/Documentation/mips/ingenic-tcu.rst b/Documentation/mips/ingenic-tcu.rst index c5a646b14450..2b75760619b4 100644 --- a/Documentation/mips/ingenic-tcu.rst +++ b/Documentation/mips/ingenic-tcu.rst @@ -68,4 +68,4 @@ and frameworks can be controlled from the same registers, all of these drivers access their registers through the same regmap. For more information regarding the devicetree bindings of the TCU drivers, -have a look at Documentation/devicetree/bindings/timer/ingenic,tcu.txt. +have a look at Documentation/devicetree/bindings/timer/ingenic,tcu.yaml. diff --git a/MAINTAINERS b/MAINTAINERS index 68f21d46614c..0617dd671c2d 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -3946,7 +3946,7 @@ L: linux-crypto(a)vger.kernel.org S: Supported F: drivers/char/hw_random/cctrng.c F: drivers/char/hw_random/cctrng.h -F: Documentation/devicetree/bindings/rng/arm-cctrng.txt +F: Documentation/devicetree/bindings/rng/arm-cctrng.yaml W: https://developer.arm.com/products/system-ip/trustzone-cryptocell/cryptocel… CEC FRAMEWORK @@ -5490,7 +5490,7 @@ F: include/uapi/drm/r128_drm.h DRM DRIVER FOR RAYDIUM RM67191 PANELS M: Robert Chiras <robert.chiras(a)nxp.com> S: Maintained -F: Documentation/devicetree/bindings/display/panel/raydium,rm67191.txt +F: Documentation/devicetree/bindings/display/panel/raydium,rm67191.yaml F: drivers/gpu/drm/panel/panel-raydium-rm67191.c DRM DRIVER FOR ROCKTECH JH057N00900 PANELS -- 2.26.2

4 5

[PATCH AUTOSEL 5.7 055/388] ASoC: SOF: Do nothing when DSP PM callbacks are not set
by Sasha Levin 18 Jun '20

18 Jun '20

From: Daniel Baluta <daniel.baluta(a)nxp.com> [ Upstream commit c26fde3b15ed41f5f452f1da727795f787833287 ] This provides a better separation between runtime and PM sleep callbacks. Only do nothing if given runtime flag is set and calback is not set. With the current implementation, if PM sleep callback is set but runtime callback is not set then at runtime resume we reload the firmware even if we do not support runtime resume callback. Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Link: https://lore.kernel.org/r/20200515135958.17511-2-kai.vehmanen@linux.intel.c… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/pm.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c index c410822d9920..01d83ddc16ba 100644 --- a/sound/soc/sof/pm.c +++ b/sound/soc/sof/pm.c @@ -90,7 +90,10 @@ static int sof_resume(struct device *dev, bool runtime_resume) int ret; /* do nothing if dsp resume callbacks are not set */ - if (!sof_ops(sdev)->resume || !sof_ops(sdev)->runtime_resume) + if (!runtime_resume && !sof_ops(sdev)->resume) + return 0; + + if (runtime_resume && !sof_ops(sdev)->runtime_resume) return 0; /* DSP was never successfully started, nothing to resume */ @@ -175,7 +178,10 @@ static int sof_suspend(struct device *dev, bool runtime_suspend) int ret; /* do nothing if dsp suspend callback is not set */ - if (!sof_ops(sdev)->suspend) + if (!runtime_suspend && !sof_ops(sdev)->suspend) + return 0; + + if (runtime_suspend && !sof_ops(sdev)->runtime_suspend) return 0; if (sdev->fw_state != SOF_FW_BOOT_COMPLETE) -- 2.25.1

4 4

[PATCH RESEND 0/3] ASoC: Intel: simplify driver/card names for SOF/UCM integration
by Pierre-Louis Bossart 18 Jun '20

18 Jun '20

As suggested by Jaroslav, this patchset simplifies legacy cards compiled with SOF: they now expose an 'SOF' driver name and an 'sof-bytcht <codec>' card name. UCM uses this driver name and additionally checks for the card name to load a configuration shared with the SST driver. This patchset is just a rename with no functionality change. There is no modification when SOF is not used, and Kconfig for SOF are disabled when SST is enabled so no risk of interference. This helps make 5.9 the first kernel version where SOF 'just works'(tm) on legacy platforms with no additional configuration needed (as is already the case with more recent HDaudio+DMIC platforms). Thanks to Jaroslav for all the hard work on the alsa-lib/UCM side (the majority of the configs are already merged in alsa-ucm-conf, only minor updates are in-flight for max98090 and broadwell). Patches sent initially on May 29, probably missed with the merge window. No changes except for the addition of Jaroslav's Reviewed-by tag. Pierre-Louis Bossart (3): ASoC: Intel: byt*: simplify card names for SOF uses ASoC: Intel: cht*: simplify card names for SOF uses ASoC: Intel: broadwell: simplify card names for SOF uses sound/soc/intel/boards/bdw-rt5650.c | 12 +++++++++++- sound/soc/intel/boards/bdw-rt5677.c | 12 +++++++++++- sound/soc/intel/boards/broadwell.c | 12 +++++++++++- sound/soc/intel/boards/bytcht_cx2072x.c | 12 +++++++++++- sound/soc/intel/boards/bytcht_da7213.c | 12 +++++++++++- sound/soc/intel/boards/bytcht_es8316.c | 12 +++++++++++- sound/soc/intel/boards/bytcr_rt5640.c | 12 +++++++++++- sound/soc/intel/boards/bytcr_rt5651.c | 12 +++++++++++- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 12 +++++++++++- sound/soc/intel/boards/cht_bsw_nau8824.c | 12 +++++++++++- sound/soc/intel/boards/cht_bsw_rt5645.c | 17 +++++++++++++++-- sound/soc/intel/boards/cht_bsw_rt5672.c | 12 +++++++++++- 12 files changed, 136 insertions(+), 13 deletions(-) -- 2.20.1

2 4

[PATCH] ANDROID: sound: usb: Add vendor's hooking interface
by JaeHun Jung 18 Jun '20

18 Jun '20

In mobile, a co-processor is used when using USB audio to improve power consumption. hooking is required for sync-up when operating the co-processor. So register call-back function. The main operation of the call-back function is as follows: - Initialize the co-processor by transmitting data when initializing. - Change the co-processor setting value through the interface function. - Configure sampling rate - pcm open/close Bug: 156315379 Change-Id: I32e1dd408e64aaef68ee06c480c4b4d4c95546dc Signed-off-by: JaeHun Jung <jh0801.jung(a)samsung.com> --- sound/usb/card.c | 16 ++++++++++++++++ sound/usb/card.h | 1 + sound/usb/clock.c | 5 +++++ sound/usb/pcm.c | 33 +++++++++++++++++++++++++++++++++ sound/usb/usbaudio.h | 30 ++++++++++++++++++++++++++++++ 5 files changed, 85 insertions(+) diff --git a/sound/usb/card.c b/sound/usb/card.c index fd6fd17..2f3fa14 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -111,6 +111,7 @@ MODULE_PARM_DESC(skip_validation, "Skip unit descriptor validation (default: no) static DEFINE_MUTEX(register_mutex); static struct snd_usb_audio *usb_chip[SNDRV_CARDS]; static struct usb_driver usb_audio_driver; +struct snd_usb_audio_vendor_ops *usb_audio_ops; /* * disconnect streams @@ -210,6 +211,12 @@ static int snd_usb_create_stream(struct snd_usb_audio *chip, int ctrlif, int int return 0; } +void snd_set_vender_interface(struct snd_usb_audio_vendor_ops *vendor_ops) +{ + usb_audio_ops = vendor_ops; +} +EXPORT_SYMBOL_GPL(snd_set_vender_interface); + /* * parse audio control descriptor and create pcm/midi streams */ @@ -598,6 +605,9 @@ static int usb_audio_probe(struct usb_interface *intf, if (err < 0) return err; + if (usb_audio_ops && usb_audio_ops->vendor_conn) + usb_audio_ops->vendor_conn(intf, dev); + /* * found a config. now register to ALSA */ @@ -653,6 +663,9 @@ static int usb_audio_probe(struct usb_interface *intf, } dev_set_drvdata(&dev->dev, chip); + if (usb_audio_ops && usb_audio_ops->vendor_usb_add_ctls) + usb_audio_ops->vendor_usb_add_ctls(chip, 0); + /* * For devices with more than one control interface, we assume the * first contains the audio controls. We might need a more specific @@ -737,6 +750,9 @@ static void usb_audio_disconnect(struct usb_interface *intf) card = chip->card; + if (usb_audio_ops && usb_audio_ops->vendor_disc) + usb_audio_ops->vendor_disc(); + mutex_lock(&register_mutex); if (atomic_inc_return(&chip->shutdown) == 1) { struct snd_usb_stream *as; diff --git a/sound/usb/card.h b/sound/usb/card.h index 395403a..a55bb4c 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -175,5 +175,6 @@ struct snd_usb_stream { struct snd_usb_substream substream[2]; struct list_head list; }; +void snd_set_vender_interface(struct snd_usb_audio_vendor_ops *vendor_ops); #endif /* __USBAUDIO_CARD_H */ diff --git a/sound/usb/clock.c b/sound/usb/clock.c index b118cf9..0ceeccb 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -642,8 +642,13 @@ static int set_sample_rate_v2v3(struct snd_usb_audio *chip, int iface, * interface is active. */ if (rate != prev_rate) { usb_set_interface(dev, iface, 0); + if (usb_audio_ops && usb_audio_ops->vendor_set_intf) + usb_audio_ops->vendor_set_intf(dev, alts, iface, 0); snd_usb_set_interface_quirk(dev); usb_set_interface(dev, iface, fmt->altsetting); + if (usb_audio_ops && usb_audio_ops->vendor_set_intf) + usb_audio_ops->vendor_set_intf(dev, alts, iface, + fmt->altsetting); snd_usb_set_interface_quirk(dev); } diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index a4e4064..6cdacac 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -134,6 +134,9 @@ static struct audioformat *find_format(struct snd_usb_substream *subs) found = fp; cur_attr = attr; } + + if (usb_audio_ops && usb_audio_ops->vendor_pcm_binterval) + usb_audio_ops->vendor_pcm_binterval(fp, found, &cur_attr, &attr); } return found; } @@ -568,6 +571,9 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) } dev_dbg(&dev->dev, "setting usb interface %d:%d\n", fmt->iface, fmt->altsetting); + if (usb_audio_ops && usb_audio_ops->vendor_set_pcm_intf) + usb_audio_ops->vendor_set_pcm_intf(dev, fmt->iface, + fmt->altsetting, subs->direction); snd_usb_set_interface_quirk(dev); } @@ -891,6 +897,15 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) struct usb_interface *iface; int ret; + if (usb_audio_ops && usb_audio_ops->vendor_set_pcmbuf) { + ret = usb_audio_ops->vendor_set_pcmbuf(subs->dev); + + if (ret < 0) { + dev_err(&subs->dev->dev, "pcm buf transfer failed\n"); + return ret; + } + } + if (! subs->cur_audiofmt) { dev_err(&subs->dev->dev, "no format is specified!\n"); return -ENXIO; @@ -924,6 +939,15 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) if (ret < 0) goto unlock; + if (usb_audio_ops && usb_audio_ops->vendor_set_rate) { + subs->need_setup_ep = false; + usb_audio_ops->vendor_set_rate( + subs->cur_audiofmt->iface, + subs->cur_rate, + subs->cur_audiofmt->altsetting); + goto unlock; + } + ret = configure_endpoint(subs); if (ret < 0) goto unlock; @@ -1333,6 +1357,9 @@ static int snd_usb_pcm_open(struct snd_pcm_substream *substream) struct snd_usb_substream *subs = &as->substream[direction]; int ret; + if (usb_audio_ops && usb_audio_ops->vendor_pcm_con) + usb_audio_ops->vendor_pcm_con(true, direction); + subs->interface = -1; subs->altset_idx = 0; runtime->hw = snd_usb_hardware; @@ -1361,12 +1388,18 @@ static int snd_usb_pcm_close(struct snd_pcm_substream *substream) struct snd_usb_substream *subs = &as->substream[direction]; int ret; + if (usb_audio_ops && usb_audio_ops->vendor_pcm_con) + usb_audio_ops->vendor_pcm_con(false, direction); + snd_media_stop_pipeline(subs); if (!as->chip->keep_iface && subs->interface >= 0 && !snd_usb_lock_shutdown(subs->stream->chip)) { usb_set_interface(subs->dev, subs->interface, 0); + if (usb_audio_ops && usb_audio_ops->vendor_set_pcm_intf) + usb_audio_ops->vendor_set_pcm_intf(subs->dev, + subs->interface, 0, direction); subs->interface = -1; ret = snd_usb_pcm_change_state(subs, UAC3_PD_STATE_D1); snd_usb_unlock_shutdown(subs->stream->chip); diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 1c892c7..a2fd8a4 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -124,4 +124,34 @@ void snd_usb_unlock_shutdown(struct snd_usb_audio *chip); extern bool snd_usb_use_vmalloc; extern bool snd_usb_skip_validation; +/* for vender function mapping */ +extern struct snd_usb_audio_vendor_ops *usb_audio_ops; + +/* USB audio interface function for audio core */ +struct snd_usb_audio_vendor_ops { + /* Set descriptors and memory map */ + void (*vendor_conn)(struct usb_interface *intf, + struct usb_device *udev); + /* Set disconnection */ + void (*vendor_disc)(void); + /* Set interface info and setting value */ + int (*vendor_set_intf)(struct usb_device *udev, + struct usb_host_interface *alts, int iface, int alt); + /* Set sample rate */ + int (*vendor_set_rate)(int iface, int rate, int alt); + /* Alloc pcm buffer */ + int (*vendor_set_pcmbuf)(struct usb_device *udev); + /* Set pcm interface */ + int (*vendor_set_pcm_intf)(struct usb_device *udev, + int iface, int alt, int direction); + /* informed whether pcm open/close to vendor */ + void (*vendor_pcm_con)(int onoff, int direction); + /* set datainterval */ + void (*vendor_pcm_binterval)(void *fp, void *found, + int *cur_attr, int *attr); + /* control USB F/W */ + int (*vendor_usb_add_ctls)(struct snd_usb_audio *chip, + unsigned long private_value); +}; + #endif /* __USBAUDIO_H */ -- 2.7.4

4 3

[PATCH 2/2] ASoC: amd: closing specific instance.
by Ravulapati Vishnu vardhan rao 18 Jun '20

18 Jun '20

The steps to reproduce: Record from the internal mic : (arecord -D hw:1,2 -f dat /dev/null -V stereos) Record from the headphone mic: (arecord -D hw:1,0 -f dat /dev/null -V stereos) Kill the recording from internal mic. We can see the recording from the headphone mic is broken. This patch rectifies the issue reported. Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 30 ++++++++++++++++++++++------- 1 file changed, 23 insertions(+), 7 deletions(-) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index d8f554f369a8..e6386de20ac7 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -342,11 +342,34 @@ static int acp3x_dma_close(struct snd_soc_component *component, { struct snd_soc_pcm_runtime *prtd; struct i2s_dev_data *adata; + struct i2s_stream_instance *ins; prtd = substream->private_data; component = snd_soc_rtdcom_lookup(prtd, DRV_NAME); adata = dev_get_drvdata(component->dev); + ins = substream->runtime->private_data; + if (!ins) + return -EINVAL; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + switch (ins->i2s_instance) { + case I2S_BT_INSTANCE: + adata->play_stream = NULL; + break; + case I2S_SP_INSTANCE: + default: + adata->i2ssp_play_stream = NULL; + } + } else { + switch (ins->i2s_instance) { + case I2S_BT_INSTANCE: + adata->capture_stream = NULL; + break; + case I2S_SP_INSTANCE: + default: + adata->i2ssp_capture_stream = NULL; + } + } /* Disable ACP irq, when the current stream is being closed and * another stream is also not active. @@ -354,13 +377,6 @@ static int acp3x_dma_close(struct snd_soc_component *component, if (!adata->play_stream && !adata->capture_stream && !adata->i2ssp_play_stream && !adata->i2ssp_capture_stream) rv_writel(0, adata->acp3x_base + mmACP_EXTERNAL_INTR_ENB); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - adata->play_stream = NULL; - adata->i2ssp_play_stream = NULL; - } else { - adata->capture_stream = NULL; - adata->i2ssp_capture_stream = NULL; - } return 0; } -- 2.17.1

2 1

[PATCH 00/13] ALSA: hda: Use LED classdev for mute controls consistently
by Takashi Iwai 18 Jun '20

18 Jun '20

Hi, this is a patch set inspired by the recent patch Kai-Heng posted about the HD-audio mic-mute LED control. Currently HD-audio driver deals with the mute and mic-mute LED in several different ways: primarily with the direct callback of vmaster hook and capture sync hook, while another with the LED class device binding. The latter has been used for binding with the platform device LEDs like Thinkpad, Dell, Huawei. And, yet, recently we added our own LED classdev for the mic-mute LED on some HP systems although they are controlled directly with the callback; it's exposed, however, for the DMIC that is governed by a different ASoC driver. This patch set is an attempt to sort out and make them consistent: namely, * All LEDs are now controlled via LED class device * The generic driver provides helper functions to easily build up the LED class dev and the relevant mixer controls * Conversion of the existing framework and clean ups The patches are lightly tested in my side with a couple of machines and also through hda-emu tests. Some devices receive new kcontrols for the mute LED behavior (that have been missing so far), but anything else look good though my tests. thanks, Takashi === Takashi Iwai (13): ALSA: hda: generic: Always call led-trigger for mic mute LED ALSA: hda: generic: Add a helper for mic-mute LED with LED classdev ALSA: hda/realtek: Convert to cdev-variant of mic-mute LED controls ALSA: hda/conexant: Convert to cdev-variant of mic-mute LED controls ALSA: hda/sigmatel: Convert to cdev-variant of mic-mute LED controls ALSA: hda: generic: Drop unused snd_hda_gen_fixup_micmute_led() ALSA: hda: generic: Drop the old mic-mute LED hook ALSA: hda: generic: Add vmaster mute LED helper ALSA: hda/realtek: Use the new vmaster mute LED helper ALSA: hda/conexant: Use the new vmaster mute LED helper ALSA: hda/sigmatel: Use the new vmaster mute LED helper ALSA: hda/realtek: Unify LED helper code ALSA: hda: Let LED cdev handling suspend/resume sound/pci/hda/Kconfig | 9 ++ sound/pci/hda/hda_generic.c | 134 ++++++++++++++++-------- sound/pci/hda/hda_generic.h | 13 ++- sound/pci/hda/patch_conexant.c | 47 ++++----- sound/pci/hda/patch_realtek.c | 220 +++++++++++++++++++--------------------- sound/pci/hda/patch_sigmatel.c | 22 ++-- sound/pci/hda/thinkpad_helper.c | 19 +--- 7 files changed, 248 insertions(+), 216 deletions(-) -- 2.16.4

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[PATCH v3 1/2] ALSA: hda/realtek: Add COEF controlled micmute LED support
by Kai-Heng Feng 18 Jun '20

18 Jun '20

Currently, HDA codec LED class can only be used by GPIO controlled LED. However, there are some new systems that control LED via COEF instead of GPIO. In order to support those systems, create a new helper that can be facilitated by both COEF controlled and GPIO controlled LED. In addition to that, add LED_CORE_SUSPENDRESUME flag since some systems don't restore the LED properly after suspend. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- v3: - Wording. - Properly prefix exported symbol. v2: - Prevent platforms like Dell, Lenovoe and Huawei create double LED class devices. sound/pci/hda/hda_generic.c | 7 ++++--- sound/pci/hda/hda_generic.h | 1 + sound/pci/hda/patch_realtek.c | 29 ++++++++++++++++------------- 3 files changed, 21 insertions(+), 16 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index f4e9d9445e18..e4f534e9a88c 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3897,7 +3897,7 @@ enum { MICMUTE_LED_FOLLOW_MUTE, }; -static void call_micmute_led_update(struct hda_codec *codec) +void snd_hda_gen_call_micmute_led_update(struct hda_codec *codec) { struct hda_gen_spec *spec = codec->spec; unsigned int val; @@ -3924,6 +3924,7 @@ static void call_micmute_led_update(struct hda_codec *codec) if (spec->micmute_led.update) spec->micmute_led.update(codec); } +EXPORT_SYMBOL_GPL(snd_hda_gen_call_micmute_led_update); static void update_micmute_led(struct hda_codec *codec, struct snd_kcontrol *kcontrol, @@ -3945,7 +3946,7 @@ static void update_micmute_led(struct hda_codec *codec, spec->micmute_led.capture |= mask; else spec->micmute_led.capture &= ~mask; - call_micmute_led_update(codec); + snd_hda_gen_call_micmute_led_update(codec); } } @@ -3982,7 +3983,7 @@ static int micmute_led_mode_put(struct snd_kcontrol *kcontrol, if (mode == spec->micmute_led.led_mode) return 0; spec->micmute_led.led_mode = mode; - call_micmute_led_update(codec); + snd_hda_gen_call_micmute_led_update(codec); return 1; } diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index fb9f1a90238b..062be242339a 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -353,6 +353,7 @@ unsigned int snd_hda_gen_path_power_filter(struct hda_codec *codec, void snd_hda_gen_stream_pm(struct hda_codec *codec, hda_nid_t nid, bool on); int snd_hda_gen_fix_pin_power(struct hda_codec *codec, hda_nid_t pin); +void snd_hda_gen_call_micmute_led_update(struct hda_codec *codec); int snd_hda_gen_add_micmute_led(struct hda_codec *codec, void (*hook)(struct hda_codec *)); void snd_hda_gen_fixup_micmute_led(struct hda_codec *codec, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6d73f8beadb6..0587d1e96b19 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4114,10 +4114,10 @@ static int micmute_led_set(struct led_classdev *led_cdev, enum led_brightness brightness) { struct hda_codec *codec = dev_to_hda_codec(led_cdev->dev->parent); - struct alc_spec *spec = codec->spec; + struct hda_gen_spec *spec = codec->spec; - alc_update_gpio_led(codec, spec->gpio_mic_led_mask, - spec->micmute_led_polarity, !!brightness); + spec->micmute_led.led_mode = !brightness; + snd_hda_gen_call_micmute_led_update(codec); return 0; } @@ -4126,7 +4126,17 @@ static struct led_classdev micmute_led_cdev = { .max_brightness = 1, .brightness_set_blocking = micmute_led_set, .default_trigger = "audio-micmute", + .flags = LED_CORE_SUSPENDRESUME, }; + +static void alc_register_micmute_led(struct hda_codec *codec) +{ + micmute_led_cdev.brightness = ledtrig_audio_get(LED_AUDIO_MICMUTE); + if (devm_led_classdev_register(&codec->core.dev, &micmute_led_cdev)) + codec_warn(codec, "failed to register micmute LED\n"); +} +#else +static inline void alc_register_micmute_led(struct hda_codec *codec) {} #endif /* setup mute and mic-mute GPIO bits, add hooks appropriately */ @@ -4136,9 +4146,6 @@ static void alc_fixup_hp_gpio_led(struct hda_codec *codec, unsigned int micmute_mask) { struct alc_spec *spec = codec->spec; -#if IS_REACHABLE(CONFIG_LEDS_TRIGGER_AUDIO) - int err; -#endif alc_fixup_gpio(codec, action, mute_mask | micmute_mask); @@ -4151,13 +4158,7 @@ static void alc_fixup_hp_gpio_led(struct hda_codec *codec, if (micmute_mask) { spec->gpio_mic_led_mask = micmute_mask; snd_hda_gen_add_micmute_led(codec, alc_gpio_micmute_update); - -#if IS_REACHABLE(CONFIG_LEDS_TRIGGER_AUDIO) - micmute_led_cdev.brightness = ledtrig_audio_get(LED_AUDIO_MICMUTE); - err = devm_led_classdev_register(&codec->core.dev, &micmute_led_cdev); - if (err) - codec_warn(codec, "failed to register micmute LED\n"); -#endif + alc_register_micmute_led(codec); } } @@ -4305,6 +4306,7 @@ static void alc285_fixup_hp_coef_micmute_led(struct hda_codec *codec, spec->mic_led_coefbit_on = 1<<13; spec->mic_led_coefbit_off = 0; snd_hda_gen_add_micmute_led(codec, alc_hp_cap_micmute_update); + alc_register_micmute_led(codec); } } @@ -4319,6 +4321,7 @@ static void alc236_fixup_hp_coef_micmute_led(struct hda_codec *codec, spec->mic_led_coefbit_on = 2<<2; spec->mic_led_coefbit_off = 1<<2; snd_hda_gen_add_micmute_led(codec, alc_hp_cap_micmute_update); + alc_register_micmute_led(codec); } } -- 2.17.1

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[RFC PATCH 0/2] TAS2563 DSP Firmware Loader
by Dan Murphy 18 Jun '20

18 Jun '20

Hello The TAS2563 amplifier has a DSP that can run programs and configurations to produce alternate audio experiences. The DSP firmware is not a typical firmware as the binary may contain various programs and configurations that are selectable during run time. These programs and configurations are selectable via files under the I2C dev node. There may be a better way to select this through ALSA controls but I was unable to find a good example of this. This is why this is an RFC patchset. Dan Dan Murphy (2): dt-bindings: tas2562: Add firmware support for tas2563 ASoc: tas2563: DSP Firmware loading support .../devicetree/bindings/sound/tas2562.yaml | 4 + sound/soc/codecs/Makefile | 2 +- sound/soc/codecs/tas2562.c | 48 ++- sound/soc/codecs/tas2562.h | 25 ++ sound/soc/codecs/tas25xx_dsp_loader.c | 377 ++++++++++++++++++ sound/soc/codecs/tas25xx_dsp_loader.h | 93 +++++ 6 files changed, 530 insertions(+), 19 deletions(-) create mode 100644 sound/soc/codecs/tas25xx_dsp_loader.c create mode 100644 sound/soc/codecs/tas25xx_dsp_loader.h -- 2.26.2

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