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

[PATCH 0/3] Properly fix headphone noise on the XPS 13 and other ALC256 devices
by Thomas Hebb 31 Mar '20

31 Mar '20

The root cause of various pervasive audio problems on the XPS 13 9350/9360, mostly relating to the headphone jack, turns out to be an undocumented feature of the ALC256 and similar codecs that routes audio along paths not exposed in the HDA node graph. The best we've had so far to configure this feature is magic numbers provided by Realtek, none of which have fully fixed all issues. This series documents the "PC Beep Hidden Register", which controls the feature and which I've reverse engineered using black box techniques, and uses my findings to hopefully fix the headphone issues on my XPS 13 once and for all. Thomas Hebb (3): ALSA: doc: Document PC Beep Hidden Register on Realtek ALC256 ALSA: hda/realtek - Set principled PC Beep configuration for ALC256 ALSA: hda/realtek - Remove now-unnecessary XPS 13 headphone noise fixups Documentation/sound/hd-audio/index.rst | 1 + Documentation/sound/hd-audio/models.rst | 2 - .../sound/hd-audio/realtek-pc-beep.rst | 129 ++++++++++++++++++ sound/pci/hda/patch_realtek.c | 49 ++----- 4 files changed, 139 insertions(+), 42 deletions(-) create mode 100644 Documentation/sound/hd-audio/realtek-pc-beep.rst -- 2.25.2

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[PATCH v2 0/3] Properly fix headphone noise on the XPS 13 and other ALC256 devices
by Thomas Hebb 31 Mar '20

31 Mar '20

The root cause of various pervasive audio problems on the XPS 13 9350/9360, mostly relating to the headphone jack, turns out to be an undocumented feature of the ALC256 and similar codecs that routes audio along paths not exposed in the HDA node graph. The best we've had so far to configure this feature is magic numbers provided by Realtek, none of which have fully fixed all issues. This series documents the "PC Beep Hidden Register", which controls the feature and which I've reverse engineered using black box techniques, and uses my findings to hopefully fix the headphone issues on my XPS 13 once and for all. Changes in v2: - Change fixed value from 0x4727 to 0x5757, which should behave identically, on advice from Kailang. Thomas Hebb (3): ALSA: doc: Document PC Beep Hidden Register on Realtek ALC256 ALSA: hda/realtek - Set principled PC Beep configuration for ALC256 ALSA: hda/realtek - Remove now-unnecessary XPS 13 headphone noise fixups Documentation/sound/hd-audio/index.rst | 1 + Documentation/sound/hd-audio/models.rst | 2 - .../sound/hd-audio/realtek-pc-beep.rst | 129 ++++++++++++++++++ sound/pci/hda/patch_realtek.c | 49 ++----- 4 files changed, 139 insertions(+), 42 deletions(-) create mode 100644 Documentation/sound/hd-audio/realtek-pc-beep.rst -- 2.25.2

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[PATCH 0/5] Add support for SOF on i.MX8M
by Daniel Baluta 31 Mar '20

31 Mar '20

From: Daniel Baluta <daniel.baluta(a)nxp.com> This patch series adds support for SOF on i.MX8M family. First board from this family that has a DSP is i.MX8MP. First 2 patches are trying to fix some compilation issues, the next two are adding the imx8m support and the last one adds the devicetree binding. Daniel Baluta (4): ASoC: SOF: imx: fix undefined reference issue ASoC: SOF: imx: Add i.MX8M HW support ASoC: SOF: Add i.MX8MP device descriptor dt-bindings: dsp: fsl: Add fsl,imx8mp-dsp entry YueHaibing (1): ASoC: SOF: imx8: Fix randbuild error .../devicetree/bindings/dsp/fsl,dsp.yaml | 2 + sound/soc/sof/imx/Kconfig | 32 +- sound/soc/sof/imx/Makefile | 2 + sound/soc/sof/imx/imx8m.c | 279 ++++++++++++++++++ sound/soc/sof/sof-of-dev.c | 14 + 5 files changed, 325 insertions(+), 4 deletions(-) create mode 100644 sound/soc/sof/imx/imx8m.c -- 2.17.1

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PATCH: ALSA USB MIDI: Fix case when USB MIDI interface has more than one extra endpoint descriptor
by Andreas Steinmetz 31 Mar '20

31 Mar '20

The Miditech MIDIFACE 16x16 (USB ID 1290:1749) has more than one extra endpoint descriptor. The first extra descriptor is: 0x06 0x30 0x00 0x00 0x00 0x00 As the code in snd_usbmidi_get_ms_info() looks only at the first extra descriptor to find USB_DT_CS_ENDPOINT the device as such is recognized but there is neither input nor output configured. The patch iterates through the extra descriptors to find the proper one. With this patch the device is correctly configured. Signed-off-by: Andreas Steinmetz <ast(a)domdv.de> --- a/sourd/usb/midi.c 2020-03-26 17:34:40.073929497 +0100 +++ b/sound/usb/midi.c 2020-03-26 17:43:39.123252612 +0100 @@ -1839,7 +1839,8 @@ struct usb_host_endpoint *hostep; struct usb_endpoint_descriptor *ep; struct usb_ms_endpoint_descriptor *ms_ep; - int i, epidx; + int i, epidx, extralen; + unsigned char *extra; intf = umidi->iface; if (!intf) @@ -1863,12 +1864,21 @@ ep = get_ep_desc(hostep); if (!usb_endpoint_xfer_bulk(ep) && !usb_endpoint_xfer_int(ep)) continue; - ms_ep = (struct usb_ms_endpoint_descriptor *)hostep->extra; - if (hostep->extralen < 4 || + extralen = hostep->extralen; + extra = hostep->extra; +repeat: ms_ep = (struct usb_ms_endpoint_descriptor *)extra; + if (extralen < 1) + continue; + if (extralen < 4 || ms_ep->bLength < 4 || ms_ep->bDescriptorType != USB_DT_CS_ENDPOINT || - ms_ep->bDescriptorSubtype != UAC_MS_GENERAL) - continue; + ms_ep->bDescriptorSubtype != UAC_MS_GENERAL) { + if (!extra[0]) + continue; + extralen -= extra[0]; + extra += extra[0]; + goto repeat; + } if (usb_endpoint_dir_out(ep)) { if (endpoints[epidx].out_ep) { if (++epidx >= MIDI_MAX_ENDPOINTS) {

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[PATCH] ASoC: snd_soc_dai_set_fmt add substream independence.
by Matt Flax 31 Mar '20

31 Mar '20

This patch is aims to start a stronger discussion on allowing both CPU and Codec dais to set formats independently based on direction. Currently it is very difficult to change stream formats because there is no way to specify it independently. A previous discussion w.r.t. a codec's stream dependent format setting required a codec's dai to be split. See here : https://mailman.alsa-project.org/pipermail/alsa-devel/2020-March/164492.html A previous discussion w.r.t. changing format based on the stream direction w.r.t. a CPU's dai showed the difficulty in setting fmt based on direection in thw hw_params function. See here : https://mailman.alsa-project.org/pipermail/alsa-devel/2020-March/165101.html It seems that it may be necessary to allow the stream direction as an input argument to snd_soc_dai_set_fmt. This patch is not complete, as there are many codecs and cpus which would require this change. However this patch is aimed as a discussion point. One example of a sound card which requires independent stream formats is an isolated sound card, such as the Audio Injector Isolated sound card. The magnetic isolation chips on the sound card require stream fomats to be different because of digital latency variations on the I2S lines. Signed-off-by: Matt Flax <flatmax(a)flatmax.org> --- include/sound/soc-dai.h | 4 ++-- sound/soc/soc-core.c | 5 +++-- sound/soc/soc-dai.c | 4 ++-- 3 files changed, 7 insertions(+), 6 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index eaaeb00e9e84..5071e006389b 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -123,7 +123,7 @@ int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio); /* Digital Audio interface formatting */ -int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); +int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt, int stream); int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width); @@ -186,7 +186,7 @@ struct snd_soc_dai_ops { * DAI format configuration * Called by soc_card drivers, normally in their hw_params. */ - int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt); + int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt, int stream); int (*xlate_tdm_slot_mask)(unsigned int slots, unsigned int *tx_mask, unsigned int *rx_mask); int (*set_tdm_slot)(struct snd_soc_dai *dai, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 068d809c349a..27a6f01dc2d3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1474,10 +1474,11 @@ int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai *codec_dai; unsigned int i; - int ret; + int ret, stream; for_each_rtd_codec_dai(rtd, i, codec_dai) { - ret = snd_soc_dai_set_fmt(codec_dai, dai_fmt); + int stream = 0; + ret = snd_soc_dai_set_fmt(codec_dai, dai_fmt, stream); if (ret != 0 && ret != -ENOTSUPP) { dev_warn(codec_dai->dev, "ASoC: Failed to set DAI format: %d\n", ret); diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 51031e330179..1f8ee7875656 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -94,11 +94,11 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_bclk_ratio); * * Configures the DAI hardware format and clocking. */ -int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt, int stream) { if (dai->driver->ops->set_fmt == NULL) return -ENOTSUPP; - return dai->driver->ops->set_fmt(dai, fmt); + return dai->driver->ops->set_fmt(dai, fmt, stream); } EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt); -- 2.20.1

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[PATCH 3/3] ASoC: fix regwmask
by 이경택 31 Mar '20

31 Mar '20

If regwshift is 32 and the selected architecture compiles '<<' operator for signed int literal into rotating shift, '1<<regwshift' became 1 and it makes regwmask to 0x0. The literal is set to unsigned long to get intended regwmask. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> --- sound/soc/soc-ops.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 652657dc6809..55ffb34be95e 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -825,7 +825,7 @@ int snd_soc_get_xr_sx(struct snd_kcontrol *kcontrol, unsigned int regbase = mc->regbase; unsigned int regcount = mc->regcount; unsigned int regwshift = component->val_bytes * BITS_PER_BYTE; - unsigned int regwmask = (1<<regwshift)-1; + unsigned int regwmask = (1UL<<regwshift)-1; unsigned int invert = mc->invert; unsigned long mask = (1UL<<mc->nbits)-1; long min = mc->min; @@ -874,7 +874,7 @@ int snd_soc_put_xr_sx(struct snd_kcontrol *kcontrol, unsigned int regbase = mc->regbase; unsigned int regcount = mc->regcount; unsigned int regwshift = component->val_bytes * BITS_PER_BYTE; - unsigned int regwmask = (1<<regwshift)-1; + unsigned int regwmask = (1UL<<regwshift)-1; unsigned int invert = mc->invert; unsigned long mask = (1UL<<mc->nbits)-1; long max = mc->max; -- 2.21.0

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[PATCH v5 0/7] ASoC: Add new module driver for new ASRC
by Shengjiu Wang 31 Mar '20

31 Mar '20

Add new module driver for new ASRC in i.MX8MN, several commits are added for new property fsl,asrc-format Shengjiu Wang (7): ASoC: dt-bindings: fsl_asrc: Add new property fsl,asrc-format ASoC: fsl-asoc-card: Support new property fsl,asrc-format ASoC: fsl_asrc: Support new property fsl,asrc-format ASoC: fsl_asrc: rename asrc_priv to asrc ASoC: fsl_asrc: Move common definition to fsl_asrc_common ASoC: dt-bindings: fsl_easrc: Add document for EASRC ASoC: fsl_easrc: Add EASRC ASoC CPU DAI drivers changes in v5 - Add new property fsl,asrc-format, rather than change fsl,asrc-width to fsl,asrc-formt. - code change for above change. changes in v4 - Add several commit for changing DT binding asrc-width to asrc-format - rename asrc_priv to asrc changes in v3 - add new commit "ASoC: fsl_asrc: Change asrc_width to asrc_format" - modify binding doc to yaml format - remove fsl_easrc_dma.c, make fsl_asrc_dma.c useable for easrc. changes in v2 - change i.MX815 to i.MX8MN - Add changes in Kconfig and Makefile .../devicetree/bindings/sound/fsl,asrc.txt | 5 + .../devicetree/bindings/sound/fsl,easrc.yaml | 101 + sound/soc/fsl/Kconfig | 11 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl-asoc-card.c | 20 +- sound/soc/fsl/fsl_asrc.c | 303 +-- sound/soc/fsl/fsl_asrc.h | 74 +- sound/soc/fsl/fsl_asrc_common.h | 105 + sound/soc/fsl/fsl_asrc_dma.c | 54 +- sound/soc/fsl/fsl_easrc.c | 2111 +++++++++++++++++ sound/soc/fsl/fsl_easrc.h | 651 +++++ 11 files changed, 3203 insertions(+), 234 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,easrc.yaml create mode 100644 sound/soc/fsl/fsl_asrc_common.h create mode 100644 sound/soc/fsl/fsl_easrc.c create mode 100644 sound/soc/fsl/fsl_easrc.h -- 2.21.0

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[PATCH] Revert "ALSA: uapi: Drop asound.h inclusion from asoc.h"
by Takashi Iwai 31 Mar '20

31 Mar '20

This reverts commit 645c08f17f477915f6d900b767e789852f150054 which was reported to break the build a program using this header. The original issue was addressed in the alsa-lib side recently, so we can make the header more self-contained again. Reported-by: Dmitry V. Levin <ldv(a)altlinux.org> Fixes: 645c08f17f47 ("ALSA: uapi: Drop asound.h inclusion from asoc.h") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/uapi/sound/asoc.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 6048553c119d..a74ca232f1fc 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -17,6 +17,7 @@ #define __LINUX_UAPI_SND_ASOC_H #include <linux/types.h> +#include <sound/asound.h> /* * Maximum number of channels topology kcontrol can represent. -- 2.16.4

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[PATCH 1/3] ASoC: dapm: connect virtual mux with default value
by 이경택 31 Mar '20

31 Mar '20

Since a virtual mixer has no backing registers to decide which path to connect, it will try to match with initial value 0. This is to ensure that the default mixer choice will be correctly powered up during initialization. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> --- sound/soc/soc-dapm.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 9fb54e6fe254..fc55a0534252 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -802,7 +802,16 @@ static void dapm_set_mixer_path_status(struct snd_soc_dapm_path *p, int i, val = max - val; p->connect = !!val; } else { - p->connect = 0; + /* since a virtual mixer has no backing registers to + * decide which path to connect, it will try to match + * with initial value 0. This is to ensure + * that the default mixer choice will be + * correctly powered up during initialization. + */ + val = 0; + if (invert) + val = max - val; + p->connect = !!val; } } -- 2.21.0

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[PATCH 2/3] ASoC: topology: use name_prefix for new kcontrol
by 이경택 31 Mar '20

31 Mar '20

Current topology doesn't add prefix of the component to newly created kcontrol. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> --- sound/soc/soc-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 575da6aba807..a152409e8746 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -362,7 +362,7 @@ static int soc_tplg_add_kcontrol(struct soc_tplg *tplg, struct snd_soc_component *comp = tplg->comp; return soc_tplg_add_dcontrol(comp->card->snd_card, - comp->dev, k, NULL, comp, kcontrol); + comp->dev, k, comp->name_prefix, comp, kcontrol); } /* remove a mixer kcontrol */ -- 2.21.0

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[PATCH 0/3] fixes for topology
by 이경택 31 Mar '20

31 Mar '20

Hi, I'd like to submit couple of patches which will fix few problems in corner case scenarios using topology. Any feedback will be greatly appreciated. Lee Gyeongtaek Lee (3): ASoC: dapm: connect virtual mux with default value ASoC: topology: use name_prefix for new kcontrol ASoC: fix regwmask sound/soc/soc-dapm.c | 11 ++++++++++- sound/soc/soc-ops.c | 4 ++-- sound/soc/soc-topology.c | 2 +- 3 files changed, 13 insertions(+), 4 deletions(-) base-commit: 76ccd234269bd05debdbc12c96eafe62dd9a6180 -- 2.21.0

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[PATCH 2/3] ASoC: topology: use name_prefix for new kcontrol
by 이경택 31 Mar '20

31 Mar '20

Current topology doesn't add prefix of the component to newly created kcontrol. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> --- sound/soc/soc-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 575da6aba807..a152409e8746 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -362,7 +362,7 @@ static int soc_tplg_add_kcontrol(struct soc_tplg *tplg, struct snd_soc_component *comp = tplg->comp; return soc_tplg_add_dcontrol(comp->card->snd_card, - comp->dev, k, NULL, comp, kcontrol); + comp->dev, k, comp->name_prefix, comp, kcontrol); } /* remove a mixer kcontrol */ -- 2.21.0

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[PATCH v3] ASoC: dt-bindings: simple-card: switch to yaml base Documentation
by Kuninori Morimoto 31 Mar '20

31 Mar '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This patch switches from .txt base to .yaml base Document. This patch is assuming that "sound-dai" is already defined. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v2 -> v3 - uses maxItems for many place which were already defined. - assuming that "sound-dai" is already defined - (@.*) -> (@[0-9a-f]+) .../devicetree/bindings/sound/simple-card.txt | 351 ------------- .../bindings/sound/simple-card.yaml | 487 ++++++++++++++++++ 2 files changed, 487 insertions(+), 351 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/simple-card.txt create mode 100644 Documentation/devicetree/bindings/sound/simple-card.yaml diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt deleted file mode 100644 index 79954cd6e37b..000000000000 --- a/Documentation/devicetree/bindings/sound/simple-card.txt +++ /dev/null @@ -1,351 +0,0 @@ -Simple-Card: - -Simple-Card specifies audio DAI connections of SoC <-> codec. - -Required properties: - -- compatible : "simple-audio-card" - -Optional properties: - -- simple-audio-card,name : User specified audio sound card name, one string - property. -- simple-audio-card,widgets : Please refer to widgets.txt. -- simple-audio-card,routing : A list of the connections between audio components. - Each entry is a pair of strings, the first being the - connection's sink, the second being the connection's - source. -- simple-audio-card,mclk-fs : Multiplication factor between stream rate and codec - mclk. When defined, mclk-fs property defined in - dai-link sub nodes are ignored. -- simple-audio-card,hp-det-gpio : Reference to GPIO that signals when - headphones are attached. -- simple-audio-card,mic-det-gpio : Reference to GPIO that signals when - a microphone is attached. -- simple-audio-card,aux-devs : List of phandles pointing to auxiliary devices, such - as amplifiers, to be added to the sound card. -- simple-audio-card,pin-switches : List of strings containing the widget names for - which pin switches must be created. - -Optional subnodes: - -- simple-audio-card,dai-link : Container for dai-link level - properties and the CPU and CODEC - sub-nodes. This container may be - omitted when the card has only one - DAI link. See the examples and the - section below. - -Dai-link subnode properties and subnodes: - -If dai-link subnode is omitted and the subnode properties are directly -under "sound"-node the subnode property and subnode names have to be -prefixed with "simple-audio-card,"-prefix. - -Required dai-link subnodes: - -- cpu : CPU sub-node -- codec : CODEC sub-node - -Optional dai-link subnode properties: - -- format : CPU/CODEC common audio format. - "i2s", "right_j", "left_j" , "dsp_a" - "dsp_b", "ac97", "pdm", "msb", "lsb" -- frame-master : Indicates dai-link frame master. - phandle to a cpu or codec subnode. -- bitclock-master : Indicates dai-link bit clock master. - phandle to a cpu or codec subnode. -- bitclock-inversion : bool property. Add this if the - dai-link uses bit clock inversion. -- frame-inversion : bool property. Add this if the - dai-link uses frame clock inversion. -- mclk-fs : Multiplication factor between stream - rate and codec mclk, applied only for - the dai-link. - -For backward compatibility the frame-master and bitclock-master -properties can be used as booleans in codec subnode to indicate if the -codec is the dai-link frame or bit clock master. In this case there -should be no dai-link node, the same properties should not be present -at sound-node level, and the bitclock-inversion and frame-inversion -properties should also be placed in the codec node if needed. - -Required CPU/CODEC subnodes properties: - -- sound-dai : phandle and port of CPU/CODEC - -Optional CPU/CODEC subnodes properties: - -- dai-tdm-slot-num : Please refer to tdm-slot.txt. -- dai-tdm-slot-width : Please refer to tdm-slot.txt. -- clocks / system-clock-frequency : specify subnode's clock if needed. - it can be specified via "clocks" if system has - clock node (= common clock), or "system-clock-frequency" - (if system doens't support common clock) - If a clock is specified, it is - enabled with clk_prepare_enable() - in dai startup() and disabled with - clk_disable_unprepare() in dai - shutdown(). - If a clock is specified and a - multiplication factor is given with - mclk-fs, the clock will be set to the - calculated mclk frequency when the - stream starts. -- system-clock-direction-out : specifies clock direction as 'out' on - initialization. It is useful for some aCPUs with - fixed clocks. - -------------------------------------------- -Example 1 - single DAI link: -------------------------------------------- - -sound { - compatible = "simple-audio-card"; - simple-audio-card,name = "VF610-Tower-Sound-Card"; - simple-audio-card,format = "left_j"; - simple-audio-card,bitclock-master = <&dailink0_master>; - simple-audio-card,frame-master = <&dailink0_master>; - simple-audio-card,widgets = - "Microphone", "Microphone Jack", - "Headphone", "Headphone Jack", - "Speaker", "External Speaker"; - simple-audio-card,routing = - "MIC_IN", "Microphone Jack", - "Headphone Jack", "HP_OUT", - "External Speaker", "LINE_OUT"; - - simple-audio-card,cpu { - sound-dai = <&sh_fsi2 0>; - }; - - dailink0_master: simple-audio-card,codec { - sound-dai = <&ak4648>; - clocks = <&osc>; - }; -}; - -&i2c0 { - ak4648: ak4648@12 { - #sound-dai-cells = <0>; - compatible = "asahi-kasei,ak4648"; - reg = <0x12>; - }; -}; - -sh_fsi2: sh_fsi2@ec230000 { - #sound-dai-cells = <1>; - compatible = "renesas,sh_fsi2"; - reg = <0xec230000 0x400>; - interrupt-parent = <&gic>; - interrupts = <0 146 0x4>; -}; - -------------------------------------------- -Example 2 - many DAI links: -------------------------------------------- - -sound { - compatible = "simple-audio-card"; - simple-audio-card,name = "Cubox Audio"; - - simple-audio-card,dai-link@0 { /* I2S - HDMI */ - reg = <0>; - format = "i2s"; - cpu { - sound-dai = <&audio1 0>; - }; - codec { - sound-dai = <&tda998x 0>; - }; - }; - - simple-audio-card,dai-link@1 { /* S/PDIF - HDMI */ - reg = <1>; - cpu { - sound-dai = <&audio1 1>; - }; - codec { - sound-dai = <&tda998x 1>; - }; - }; - - simple-audio-card,dai-link@2 { /* S/PDIF - S/PDIF */ - reg = <2>; - cpu { - sound-dai = <&audio1 1>; - }; - codec { - sound-dai = <&spdif_codec>; - }; - }; -}; - -------------------------------------------- -Example 3 - route audio from IMX6 SSI2 through TLV320DAC3100 codec -through TPA6130A2 amplifier to headphones: -------------------------------------------- - -&i2c0 { - codec: tlv320dac3100@18 { - compatible = "ti,tlv320dac3100"; - ... - } - - amp: tpa6130a2@60 { - compatible = "ti,tpa6130a2"; - ... - } -} - -sound { - compatible = "simple-audio-card"; - ... - simple-audio-card,widgets = - "Headphone", "Headphone Jack"; - simple-audio-card,routing = - "Headphone Jack", "HPLEFT", - "Headphone Jack", "HPRIGHT", - "LEFTIN", "HPL", - "RIGHTIN", "HPR"; - simple-audio-card,aux-devs = <&amp>; - simple-audio-card,cpu { - sound-dai = <&ssi2>; - }; - simple-audio-card,codec { - sound-dai = <&codec>; - clocks = ... - }; -}; - -------------------------------------------- -Example 4. Sampling Rate Conversion -------------------------------------------- - -sound { - compatible = "simple-audio-card"; - - simple-audio-card,name = "rsnd-ak4643"; - simple-audio-card,format = "left_j"; - simple-audio-card,bitclock-master = <&sndcodec>; - simple-audio-card,frame-master = <&sndcodec>; - - simple-audio-card,convert-rate = <48000>; - - simple-audio-card,prefix = "ak4642"; - simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", - "DAI0 Capture", "ak4642 Capture"; - - sndcpu: simple-audio-card,cpu { - sound-dai = <&rcar_sound>; - }; - - sndcodec: simple-audio-card,codec { - sound-dai = <&ak4643>; - system-clock-frequency = <11289600>; - }; -}; - -------------------------------------------- -Example 5. 2 CPU 1 Codec (Mixing) -------------------------------------------- -sound { - compatible = "simple-audio-card"; - - simple-audio-card,name = "rsnd-ak4643"; - simple-audio-card,format = "left_j"; - simple-audio-card,bitclock-master = <&dpcmcpu>; - simple-audio-card,frame-master = <&dpcmcpu>; - - simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", - "ak4642 Playback", "DAI1 Playback"; - - dpcmcpu: cpu@0 { - sound-dai = <&rcar_sound 0>; - }; - - cpu@1 { - sound-dai = <&rcar_sound 1>; - }; - - codec { - prefix = "ak4642"; - sound-dai = <&ak4643>; - clocks = <&audio_clock>; - }; -}; - -------------------------------------------- -Example 6 - many DAI links with DPCM: -------------------------------------------- - -CPU0 ------ ak4613 -CPU1 ------ PCM3168A-p /* DPCM 1ch/2ch */ -CPU2 --/ /* DPCM 3ch/4ch */ -CPU3 --/ /* DPCM 5ch/6ch */ -CPU4 --/ /* DPCM 7ch/8ch */ -CPU5 ------ PCM3168A-c - -sound { - compatible = "simple-audio-card"; - - simple-audio-card,routing = - "pcm3168a Playback", "DAI1 Playback", - "pcm3168a Playback", "DAI2 Playback", - "pcm3168a Playback", "DAI3 Playback", - "pcm3168a Playback", "DAI4 Playback"; - - simple-audio-card,dai-link@0 { - format = "left_j"; - bitclock-master = <&sndcpu0>; - frame-master = <&sndcpu0>; - - sndcpu0: cpu { - sound-dai = <&rcar_sound 0>; - }; - codec { - sound-dai = <&ak4613>; - }; - }; - simple-audio-card,dai-link@1 { - format = "i2s"; - bitclock-master = <&sndcpu1>; - frame-master = <&sndcpu1>; - - convert-channels = <8>; /* TDM Split */ - - sndcpu1: cpu@0 { - sound-dai = <&rcar_sound 1>; - }; - cpu@1 { - sound-dai = <&rcar_sound 2>; - }; - cpu@2 { - sound-dai = <&rcar_sound 3>; - }; - cpu@3 { - sound-dai = <&rcar_sound 4>; - }; - codec { - mclk-fs = <512>; - prefix = "pcm3168a"; - dai-tdm-slot-num = <8>; - sound-dai = <&pcm3168a 0>; - }; - }; - simple-audio-card,dai-link@2 { - format = "i2s"; - bitclock-master = <&sndcpu2>; - frame-master = <&sndcpu2>; - - sndcpu2: cpu { - sound-dai = <&rcar_sound 5>; - }; - codec { - mclk-fs = <512>; - prefix = "pcm3168a"; - sound-dai = <&pcm3168a 1>; - }; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/simple-card.yaml b/Documentation/devicetree/bindings/sound/simple-card.yaml new file mode 100644 index 000000000000..cd5ea50daa22 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/simple-card.yaml @@ -0,0 +1,487 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/simple-card.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Simple Audio Card Driver Device Tree Bindings + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> + +definitions: + + frame-master: + description: Indicates dai-link frame master. + allOf: + - $ref: /schemas/types.yaml#/definitions/phandle-array + - maxItems: 1 + + bitclock-master: + description: Indicates dai-link bit clock master + allOf: + - $ref: /schemas/types.yaml#/definitions/phandle-array + - maxItems: 1 + + frame-inversion: + description: dai-link uses frame clock inversion + $ref: /schemas/types.yaml#/definitions/flag + + bitclock-inversion: + description: dai-link uses bit clock inversion + $ref: /schemas/types.yaml#/definitions/flag + + dai-tdm-slot-num: + description: see tdm-slot.txt. + $ref: /schemas/types.yaml#/definitions/uint32 + + dai-tdm-slot-width: + description: see tdm-slot.txt. + $ref: /schemas/types.yaml#/definitions/uint32 + + clocks: + maxItems: 1 + + system-clock-frequency: + description: | + If a clock is specified and a multiplication factor is given with + mclk-fs, the clock will be set to the calculated mclk frequency + when the stream starts. + $ref: /schemas/types.yaml#/definitions/uint32 + + system-clock-direction-out: + description: | + specifies clock direction as 'out' on initialization. + It is useful for some aCPUs with fixed clocks. + $ref: /schemas/types.yaml#/definitions/flag + + mclk-fs: + description: | + Multiplication factor between stream rate and codec mclk. + When defined, mclk-fs property defined in dai-link sub nodes are ignored. + $ref: /schemas/types.yaml#/definitions/uint32 + + aux-devs: + description: | + List of phandles pointing to auxiliary devices, such + as amplifiers, to be added to the sound card. + $ref: /schemas/types.yaml#/definitions/phandle-array + + convert-rate: + description: CPU to Codec rate convert. + $ref: /schemas/types.yaml#/definitions/uint32 + + convert-channels: + description: CPU to Codec rate channels. + $ref: /schemas/types.yaml#/definitions/uint32 + + prefix: + description: "device name prefix" + $ref: /schemas/types.yaml#/definitions/string + + label: + maxItems: 1 + + routing: + description: | + A list of the connections between audio components. + Each entry is a pair of strings, the first being the + connection's sink, the second being the connection's source. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + + widgets: + description: User specified audio sound widgets. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + + pin-switches: + description: the widget names for which pin switches must be created. + $ref: /schemas/types.yaml#/definitions/string-array + + format: + description: audio format. + items: + enum: + - i2s + - right_j + - left_j + - dsp_a + - dsp_b + - ac97 + - pdm + - msb + - lsb + + dai: + type: object + properties: + sound-dai: + maxItems: 1 + + # common properties + mclk-fs: + $ref: "#/definitions/mclk-fs" + prefix: + $ref: "#/definitions/prefix" + frame-inversion: + $ref: "#/definitions/frame-inversion" + bitclock-inversion: + $ref: "#/definitions/bitclock-inversion" + frame-master: + $ref: /schemas/types.yaml#/definitions/flag + bitclock-master: + $ref: /schemas/types.yaml#/definitions/flag + + dai-tdm-slot-num: + $ref: "#/definitions/dai-tdm-slot-num" + dai-tdm-slot-width: + $ref: "#/definitions/dai-tdm-slot-width" + clocks: + $ref: "#/definitions/clocks" + system-clock-frequency: + $ref: "#/definitions/system-clock-frequency" + system-clock-direction-out: + $ref: "#/definitions/system-clock-direction-out" + required: + - sound-dai + +properties: + compatible: + contains: + enum: + - simple-audio-card + - simple-scu-audio-card + + "#address-cells": + const: 1 + "#size-cells": + const: 0 + + label: + $ref: "#/definitions/label" + + simple-audio-card,name: + description: User specified audio sound card name. + $ref: /schemas/types.yaml#/definitions/string + +# use patternProperties to use "#definitions/xxx" +patternProperties: + "^simple-audio-card,widgets$": + $ref: "#/definitions/widgets" + "^simple-audio-card,routing$": + $ref: "#/definitions/routing" + "^simple-audio-card,cpu(@[0-9a-f]+)?": + $ref: "#/definitions/dai" + "^simple-audio-card,codec(@[0-9a-f]+)?": + $ref: "#/definitions/dai" + + # common properties + "^simple-audio-card,frame-master$": + $ref: "#/definitions/frame-master" + "^simple-audio-card,bitclock-master$": + $ref: "#/definitions/bitclock-master" + "^simple-audio-card,frame-inversion$": + $ref: "#/definitions/frame-inversion" + "^simple-audio-card,bitclock-inversion$": + $ref: "#/definitions/bitclock-inversion" + "^simple-audio-card,format$": + $ref: "#/definitions/format" + "^simple-audio-card,mclk-fs$": + $ref: "#/definitions/mclk-fs" + "^simple-audio-card,aux-devs$": + $ref: "#/definitions/aux-devs" + "^simple-audio-card,convert-rate$": + $ref: "#/definitions/convert-rate" + "^simple-audio-card,convert-channels$": + $ref: "#/definitions/convert-channels" + "^simple-audio-card,prefix$": + $ref: "#/definitions/prefix" + "^simple-audio-card,pin-switches$": + $ref: "#/definitions/pin-switches" + "^simple-audio-card,hp-det-gpio$": + maxItems: 1 + "^simple-audio-card,mic-det-gpio$": + maxItems: 1 + + "^simple-audio-card,dai-link(@[0-9a-f]+)?$": + description: | + Container for dai-link level properties and the CPU and CODEC sub-nodes. + This container may be omitted when the card has only one DAI link. + type: object + properties: + reg: + maxItems: 1 + + # common properties + frame-master: + $ref: "#/definitions/frame-master" + bitclock-master: + $ref: "#/definitions/bitclock-master" + frame-inversion: + $ref: "#/definitions/frame-inversion" + bitclock-inversion: + $ref: "#/definitions/bitclock-inversion" + format: + $ref: "#/definitions/format" + mclk-fs: + $ref: "#/definitions/mclk-fs" + aux-devs: + $ref: "#/definitions/aux-devs" + convert-rate: + $ref: "#/definitions/convert-rate" + convert-channels: + $ref: "#/definitions/convert-channels" + prefix: + $ref: "#/definitions/prefix" + pin-switches: + $ref: "#/definitions/pin-switches" + hp-det-gpio: + maxItems: 1 + mic-det-gpio: + maxItems: 1 + + patternProperties: + "^cpu(@[0-9a-f]+)?": + $ref: "#/definitions/dai" + "^codec(@[0-9a-f]+)?": + $ref: "#/definitions/dai" + additionalProperties: false + +required: + - compatible + +additionalProperties: false + +examples: +#-------------------- +# single DAI link +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + simple-audio-card,name = "VF610-Tower-Sound-Card"; + simple-audio-card,format = "left_j"; + simple-audio-card,bitclock-master = <&dailink0_master>; + simple-audio-card,frame-master = <&dailink0_master>; + simple-audio-card,widgets = + "Microphone", "Microphone Jack", + "Headphone", "Headphone Jack", + "Speaker", "External Speaker"; + simple-audio-card,routing = + "MIC_IN", "Microphone Jack", + "Headphone Jack", "HP_OUT", + "External Speaker", "LINE_OUT"; + + simple-audio-card,cpu { + sound-dai = <&sh_fsi2 0>; + }; + + dailink0_master: simple-audio-card,codec { + sound-dai = <&ak4648>; + clocks = <&osc>; + }; + }; + +#-------------------- +# Multi DAI links +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + simple-audio-card,name = "Cubox Audio"; + + #address-cells = <1>; + #size-cells = <0>; + + simple-audio-card,dai-link@0 { /* I2S - HDMI */ + reg = <0>; + format = "i2s"; + cpu { + sound-dai = <&audio0>; + }; + codec { + sound-dai = <&tda998x0>; + }; + }; + + simple-audio-card,dai-link@1 { /* S/PDIF - HDMI */ + reg = <1>; + cpu { + sound-dai = <&audio1>; + }; + codec { + sound-dai = <&tda998x1>; + }; + }; + + simple-audio-card,dai-link@2 { /* S/PDIF - S/PDIF */ + reg = <2>; + cpu { + sound-dai = <&audio2>; + }; + codec { + sound-dai = <&spdif_codec>; + }; + }; + }; + +#-------------------- +# route audio from IMX6 SSI2 through TLV320DAC3100 codec +# through TPA6130A2 amplifier to headphones: +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + + simple-audio-card,widgets = + "Headphone", "Headphone Jack"; + simple-audio-card,routing = + "Headphone Jack", "HPLEFT", + "Headphone Jack", "HPRIGHT", + "LEFTIN", "HPL", + "RIGHTIN", "HPR"; + simple-audio-card,aux-devs = <&amp>; + simple-audio-card,cpu { + sound-dai = <&ssi2>; + }; + simple-audio-card,codec { + sound-dai = <&codec>; + clocks = <&clocks>; + }; + }; + +#-------------------- +# Sampling Rate Conversion +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + + simple-audio-card,name = "rsnd-ak4643"; + simple-audio-card,format = "left_j"; + simple-audio-card,bitclock-master = <&sndcodec>; + simple-audio-card,frame-master = <&sndcodec>; + + simple-audio-card,convert-rate = <48000>; + + simple-audio-card,prefix = "ak4642"; + simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", + "DAI0 Capture", "ak4642 Capture"; + + sndcpu: simple-audio-card,cpu { + sound-dai = <&rcar_sound>; + }; + + sndcodec: simple-audio-card,codec { + sound-dai = <&ak4643>; + system-clock-frequency = <11289600>; + }; + }; + +#-------------------- +# 2 CPU 1 Codec (Mixing) +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + + simple-audio-card,name = "rsnd-ak4643"; + simple-audio-card,format = "left_j"; + simple-audio-card,bitclock-master = <&dpcmcpu>; + simple-audio-card,frame-master = <&dpcmcpu>; + + simple-audio-card,convert-rate = <48000>; + simple-audio-card,convert-channels = <2>; + + simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", + "ak4642 Playback", "DAI1 Playback"; + + dpcmcpu: simple-audio-card,cpu@0 { + sound-dai = <&rcar_sound 0>; + }; + + simple-audio-card,cpu@1 { + sound-dai = <&rcar_sound 1>; + }; + + simple-audio-card,codec { + prefix = "ak4642"; + sound-dai = <&ak4643>; + clocks = <&audio_clock>; + }; + }; + +#-------------------- +# Multi DAI links with DPCM: +# +# CPU0 ------ ak4613 +# CPU1 ------ PCM3168A-p /* DPCM 1ch/2ch */ +# CPU2 --/ /* DPCM 3ch/4ch */ +# CPU3 --/ /* DPCM 5ch/6ch */ +# CPU4 --/ /* DPCM 7ch/8ch */ +# CPU5 ------ PCM3168A-c +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + + simple-audio-card,routing = + "pcm3168a Playback", "DAI1 Playback", + "pcm3168a Playback", "DAI2 Playback", + "pcm3168a Playback", "DAI3 Playback", + "pcm3168a Playback", "DAI4 Playback"; + + simple-audio-card,dai-link@0 { + format = "left_j"; + bitclock-master = <&sndcpu0>; + frame-master = <&sndcpu0>; + + sndcpu0: cpu { + sound-dai = <&rcar_sound 0>; + }; + codec { + sound-dai = <&ak4613>; + }; + }; + + simple-audio-card,dai-link@1 { + format = "i2s"; + bitclock-master = <&sndcpu1>; + frame-master = <&sndcpu1>; + + convert-channels = <8>; /* TDM Split */ + + sndcpu1: cpu@0 { + sound-dai = <&rcar_sound 1>; + }; + cpu@1 { + sound-dai = <&rcar_sound 2>; + }; + cpu@2 { + sound-dai = <&rcar_sound 3>; + }; + cpu@3 { + sound-dai = <&rcar_sound 4>; + }; + codec { + mclk-fs = <512>; + prefix = "pcm3168a"; + dai-tdm-slot-num = <8>; + sound-dai = <&pcm3168a 0>; + }; + }; + + simple-audio-card,dai-link@2 { + format = "i2s"; + bitclock-master = <&sndcpu2>; + frame-master = <&sndcpu2>; + + sndcpu2: cpu { + sound-dai = <&rcar_sound 5>; + }; + codec { + mclk-fs = <512>; + prefix = "pcm3168a"; + sound-dai = <&pcm3168a 1>; + }; + }; + }; -- 2.17.1

2 2

[alsa-devel] [PATCH v3 0/7] ASoC: soc-pcm cleanup step2
by Kuninori Morimoto 31 Mar '20

31 Mar '20

Hi Mark These are v3 of soc-pcm cleanup. Original [5/7] [6/7] patch call open / start function for all DAI / component. And, it calls close / stop function for all if some of them got error. But in such case, error occured DAI / component don't need to call close / stop. This issue can be solved if it had flag, and my local tree has such patch. But it was planed to post to ML a little later. This time, I merged original patch and this new flag patch on [5/7] [6/7]. These mention that "it is prepare for soc-pcm-open() cleanup". But it will be happen later. These are based on v5.6-rc1 Kuninori Morimoto (8): 1) ASoC: soc-pcm: add snd_soc_runtime_action() 2) ASoC: soc-pcm: adjustment for DAI member 0 reset 3) ASoC: soc-pcm: add for_each_dapm_widgets() macro 4) ASoC: soc-pcm: don't use bit-OR'ed error 5) ASoC: soc-pcm: call snd_soc_dai_startup()/shutdown() once 6) ASoC: soc-pcm: call snd_soc_component_open/close() once 7) ASoC: soc-pcm: move soc_pcm_close() next to soc_pcm_open() 8) ASoC: soc-pcm: tidyup soc_pcm_open() order include/sound/soc-component.h | 7 +- include/sound/soc-dai.h | 5 +- include/sound/soc-dapm.h | 5 + sound/soc/soc-component.c | 35 ++++-- sound/soc/soc-dai.c | 11 +- sound/soc/soc-dapm.c | 8 +- sound/soc/soc-pcm.c | 246 +++++++++++++++++++----------------------- 7 files changed, 161 insertions(+), 156 deletions(-) -- 2.7.4

6 32

[alsa-devel] [PATCH 0/5] ALSA: Fix uapi headers
by Takashi Iwai 30 Mar '20

30 Mar '20

Hi, here is a series of fixes for include/uapi/sound/*.h files and the relevant code changes, so that they can be used for building alsa-tools programs again. The corresponding alsa-lib update will be followed. Takashi === Takashi Iwai (5): ALSA: emu10k1: Make uapi/emu10k1.h compilable again ALSA: hdsp: Make uapi/hdsp.h compilable again ALSA: hdspm: Drop linux/types.h inclusion in uapi header ALSA: uapi: Fix typos and header inclusion in asound.h ALSA: uapi: Drop asound.h inclusion from asoc.h include/uapi/sound/asoc.h | 1 - include/uapi/sound/asound.h | 6 +++--- include/uapi/sound/emu10k1.h | 38 ++++++++++++++++++++++++++------------ include/uapi/sound/hdsp.h | 4 +--- include/uapi/sound/hdspm.h | 2 -- sound/pci/emu10k1/emufx.c | 26 ++++++++++++++------------ sound/pci/rme9652/hdsp.c | 2 +- 7 files changed, 45 insertions(+), 34 deletions(-) -- 2.16.4

2 7

[PATCH] ASoC: soc-dai: fix DAI startup/shutdown sequence
by Pierre-Louis Bossart 30 Mar '20

30 Mar '20

The addition of a single flag to track the DAI status prevents the DAI startup sequence from being called on capture if the DAI is already used for playback. Fix by extending the existing code with one flag per direction. Reported-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Tested-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Fixes: b56be800f1292 ("ASoC: soc-pcm: call snd_soc_dai_startup()/shutdown() once") Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- include/sound/soc-dai.h | 2 +- sound/soc/soc-dai.c | 8 ++++---- 2 files changed, 5 insertions(+), 5 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 78bac995db15..d4825b82c7a3 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -351,7 +351,7 @@ struct snd_soc_dai { /* bit field */ unsigned int probed:1; - unsigned int started:1; + unsigned int started[SNDRV_PCM_STREAM_LAST + 1]; }; static inline struct snd_soc_pcm_stream * diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 19142f6e533c..8f3cad8db89a 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -295,12 +295,12 @@ int snd_soc_dai_startup(struct snd_soc_dai *dai, { int ret = 0; - if (!dai->started && + if (!dai->started[substream->stream] && dai->driver->ops->startup) ret = dai->driver->ops->startup(substream, dai); if (ret == 0) - dai->started = 1; + dai->started[substream->stream] = 1; return ret; } @@ -308,11 +308,11 @@ int snd_soc_dai_startup(struct snd_soc_dai *dai, void snd_soc_dai_shutdown(struct snd_soc_dai *dai, struct snd_pcm_substream *substream) { - if (dai->started && + if (dai->started[substream->stream] && dai->driver->ops->shutdown) dai->driver->ops->shutdown(substream, dai); - dai->started = 0; + dai->started[substream->stream] = 0; } int snd_soc_dai_prepare(struct snd_soc_dai *dai, -- 2.20.1

2 1

snd_hda_intel/sst-acpi sound breakage on suspend/resume since 5.6-rc1
by Dominik Brodowski 30 Mar '20

30 Mar '20

Hi! While 5.5.x works fine, mainline as of ac309e7744be (v5.6-rc6+) causes me some sound-related trouble: after boot, the sound works fine -- but once I suspend and resume my broadwell-based XPS13, I need to switch to headphone and back to speaker to hear something. But what I hear isn't music but garbled output. A few dmesg snippets from v5.6-rc6-9-gac309e7744be which might be of interest. I've highlighted the lines differing from v.5.5.x which might be of special interest: ... snd_hda_intel 0000:00:03.0: enabling device (0000 -> 0002) usbcore: registered new interface driver snd-usb-audio snd_hda_intel 0000:00:03.0: bound 0000:00:02.0 (ops i915_audio_component_bind_ops) input: HDA Intel HDMI HDMI/DP,pcm=3 as /devices/pci0000:00/0000:00:03.0/sound/card0/input13 input: HDA Intel HDMI HDMI/DP,pcm=7 as /devices/pci0000:00/0000:00:03.0/sound/card0/input14 input: HDA Intel HDMI HDMI/DP,pcm=8 as /devices/pci0000:00/0000:00:03.0/sound/card0/input15 input: HDA Intel HDMI HDMI/DP,pcm=9 as /devices/pci0000:00/0000:00:03.0/sound/card0/input16 input: HDA Intel HDMI HDMI/DP,pcm=10 as /devices/pci0000:00/0000:00:03.0/sound/card0/input17 Console: switching to colour frame buffer device 240x67 !!! sst-acpi INT3438:00: WARN: Device release is not defined so it is not safe to unbind this driver while in use i915 0000:00:02.0: fb0: i915drmfb frame buffer device sst-acpi INT3438:00: DesignWare DMA Controller, 8 channels psmouse serio1: synaptics: Unable to query device: -5 haswell-pcm-audio haswell-pcm-audio: Direct firmware load for intel/IntcPP01.bin failed with error -2 haswell-pcm-audio haswell-pcm-audio: fw image intel/IntcPP01.bin not available(-2) haswell-pcm-audio haswell-pcm-audio: FW loaded, mailbox readback FW info: type 01, - version: 00.00, build 77, source commit id: 876ac6906f31a43b6772b23c7c983ce9dcb18a19 rt286 i2c-INT343A:00: ASoC: sink widget DMIC1 overwritten rt286 i2c-INT343A:00: ASoC: source widget DMIC1 overwritten broadwell-audio broadwell-audio: snd-soc-dummy-dai <-> System Pin mapping ok broadwell-audio broadwell-audio: snd-soc-dummy-dai <-> Offload0 Pin mapping ok broadwell-audio broadwell-audio: snd-soc-dummy-dai <-> Offload1 Pin mapping ok broadwell-audio broadwell-audio: snd-soc-dummy-dai <-> Loopback Pin mapping ok broadwell-audio broadwell-audio: rt286-aif1 <-> snd-soc-dummy-dai mapping ok input: broadwell-rt286 Headset as /devices/pci0000:00/INT3438:00/broadwell-audio/sound/card1/input18 ... ALSA device list: #0: HDA Intel HDMI at 0xf7218000 irq 48 #1: DellInc.-XPS139343--0TM99H ... !!! haswell-pcm-audio haswell-pcm-audio: warning: stream is NULL, no stream to reset, ignore it. !!! haswell-pcm-audio haswell-pcm-audio: warning: stream is NULL, no stream to free, ignore it. (these last two messages already are printed a couple of time after boot, and then again during a suspend/resume cycle. On v.5.5.y, there are similar messages "no context buffer need to restore!"). Everything is built-in, no modules are loaded. Unfortunately, I cannot bisect this issue easily -- i915 was broken for quite some time on this system[*], prohibiting boot... Thanks for taking a look at this issue! Dominik [*] https://gitlab.freedesktop.org/drm/intel/issues/1151

8 38

[PATCH] dt-bindings: sound: cs42l51: Remove unneeded I2C unit name
by Fabio Estevam 30 Mar '20

30 Mar '20

The following warning is seen with 'make dt_binding_check': Documentation/devicetree/bindings/sound/cirrus,cs42l51.example.dts:18.15-34.11: Warning (unit_address_vs_reg): /example-0/i2c@0: node has a unit name, but no reg or ranges property Fix it by removing the unneeded i2c unit name. Signed-off-by: Fabio Estevam <festevam(a)gmail.com> --- Documentation/devicetree/bindings/sound/cirrus,cs42l51.yaml | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/cirrus,cs42l51.yaml b/Documentation/devicetree/bindings/sound/cirrus,cs42l51.yaml index efce847a3408..83f44f07ac3f 100644 --- a/Documentation/devicetree/bindings/sound/cirrus,cs42l51.yaml +++ b/Documentation/devicetree/bindings/sound/cirrus,cs42l51.yaml @@ -49,7 +49,7 @@ required: examples: - | #include <dt-bindings/gpio/gpio.h> - i2c@0 { + i2c { #address-cells = <1>; #size-cells = <0>; -- 2.17.1

3 2

[PATCH] ASoC: bcm: Fix pointer cast warning
by Takashi Iwai 30 Mar '20

30 Mar '20

The NULL check can be done gracefully without cast. It fixes a compile warning like: sound/soc/bcm/bcm63xx-pcm-whistler.c:184:6: warning: cast to pointer from integer of different size [-Wint-to-pointer-cast] Fixes: 88eb404ccc3e ("ASoC: brcm: Add DSL/PON SoC audio driver") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/bcm/bcm63xx-pcm-whistler.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/bcm/bcm63xx-pcm-whistler.c b/sound/soc/bcm/bcm63xx-pcm-whistler.c index e46c390683e7..b7a1efc7406e 100644 --- a/sound/soc/bcm/bcm63xx-pcm-whistler.c +++ b/sound/soc/bcm/bcm63xx-pcm-whistler.c @@ -181,7 +181,7 @@ bcm63xx_pcm_pointer(struct snd_soc_component *component, snd_pcm_uframes_t x; struct bcm63xx_runtime_data *prtd = substream->runtime->private_data; - if ((void *)prtd->dma_addr_next == NULL) + if (!prtd->dma_addr_next) prtd->dma_addr_next = substream->runtime->dma_addr; x = bytes_to_frames(substream->runtime, -- 2.16.4

2 1

[PATCH] ASoC: AMD: Clear format bits before setting them
by Akshu Agrawal 30 Mar '20

30 Mar '20

This avoids residual bit form previous format when the format is changed. Hence, the resultant format is not an invalid one. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> --- sound/soc/amd/raven/acp3x-i2s.c | 1 + sound/soc/amd/raven/acp3x.h | 2 ++ 2 files changed, 3 insertions(+) diff --git a/sound/soc/amd/raven/acp3x-i2s.c b/sound/soc/amd/raven/acp3x-i2s.c index 3a3c47e820ab..f160d35a6832 100644 --- a/sound/soc/amd/raven/acp3x-i2s.c +++ b/sound/soc/amd/raven/acp3x-i2s.c @@ -139,6 +139,7 @@ static int acp3x_i2s_hwparams(struct snd_pcm_substream *substream, rv_writel(adata->tdm_fmt, rtd->acp3x_base + frmt_reg); } val = rv_readl(rtd->acp3x_base + reg_val); + val &= ~ACP3x_ITER_IRER_SAMP_LEN_MASK; val = val | (rtd->xfer_resolution << 3); rv_writel(val, rtd->acp3x_base + reg_val); return 0; diff --git a/sound/soc/amd/raven/acp3x.h b/sound/soc/amd/raven/acp3x.h index 21e7ac017f2b..03fe93913e12 100644 --- a/sound/soc/amd/raven/acp3x.h +++ b/sound/soc/amd/raven/acp3x.h @@ -76,6 +76,8 @@ #define ACP_POWERED_OFF 0x02 #define ACP_POWER_OFF_IN_PROGRESS 0x03 +#define ACP3x_ITER_IRER_SAMP_LEN_MASK 0x38 + struct acp3x_platform_info { u16 play_i2s_instance; u16 cap_i2s_instance; -- 2.20.1

2 1

[PATCH 0/4] ucm2: Add support to SDM845 based platforms
by Srinivas Kandagatla 30 Mar '20

30 Mar '20

This patchset adds UCM support for WSA881x, WCD934x codec and two platforms, DB845c DragonBoard and Lenovo YOGA C360 Laptop that use these codecs. Tested on Ubuntu 20.04 with pulseaudio 13 along with with this fix https://github.com/pulseaudio/pulseaudio/commit/cabd387e26b5fe03661f4b89482… Thanks, srini Srinivas Kandagatla (4): ucm2: codecs: wcd934x: Add Codec sequences. ucm2: codecs: wsa881x: add codec sequences ucm2: DB845c: Add ucm for DB845c board ucm2: Add ucm for Lenovo-YOGA-C630-13Q50 laptop ucm2/DB845c/DB845c.conf | 11 +++ ucm2/DB845c/HDMI.conf | 33 +++++++ ucm2/DB845c/HiFi.conf | 46 +++++++++ ucm2/Lenovo-YOGA-C630-13Q50/HiFi.conf | 98 +++++++++++++++++++ .../Lenovo-YOGA-C630-13Q50.conf | 6 ++ ucm2/codecs/wcd934x/DefaultDisableSeq.conf | 5 + ucm2/codecs/wcd934x/DefaultEnableSeq.conf | 14 +++ ucm2/codecs/wcd934x/HeadphoneDisableSeq.conf | 6 ++ ucm2/codecs/wcd934x/HeadphoneEnableSeq.conf | 6 ++ .../wcd934x/HeadphoneMicDisableSeq.conf | 2 + .../codecs/wcd934x/HeadphoneMicEnableSeq.conf | 3 + ucm2/codecs/wcd934x/SpeakerDisableSeq.conf | 4 + ucm2/codecs/wcd934x/SpeakerEnableSeq.conf | 4 + ucm2/codecs/wsa881x/DefaultEnableSeq.conf | 8 ++ ucm2/codecs/wsa881x/SpeakerDisableSeq.conf | 8 ++ ucm2/codecs/wsa881x/SpeakerEnableSeq.conf | 8 ++ 16 files changed, 262 insertions(+) create mode 100644 ucm2/DB845c/DB845c.conf create mode 100644 ucm2/DB845c/HDMI.conf create mode 100644 ucm2/DB845c/HiFi.conf create mode 100644 ucm2/Lenovo-YOGA-C630-13Q50/HiFi.conf create mode 100644 ucm2/Lenovo-YOGA-C630-13Q50/Lenovo-YOGA-C630-13Q50.conf create mode 100644 ucm2/codecs/wcd934x/DefaultDisableSeq.conf create mode 100644 ucm2/codecs/wcd934x/DefaultEnableSeq.conf create mode 100644 ucm2/codecs/wcd934x/HeadphoneDisableSeq.conf create mode 100644 ucm2/codecs/wcd934x/HeadphoneEnableSeq.conf create mode 100644 ucm2/codecs/wcd934x/HeadphoneMicDisableSeq.conf create mode 100644 ucm2/codecs/wcd934x/HeadphoneMicEnableSeq.conf create mode 100644 ucm2/codecs/wcd934x/SpeakerDisableSeq.conf create mode 100644 ucm2/codecs/wcd934x/SpeakerEnableSeq.conf create mode 100644 ucm2/codecs/wsa881x/DefaultEnableSeq.conf create mode 100644 ucm2/codecs/wsa881x/SpeakerDisableSeq.conf create mode 100644 ucm2/codecs/wsa881x/SpeakerEnableSeq.conf -- 2.21.0

2 7

[GIT PULL] ASoC updates for v5.7
by Mark Brown 30 Mar '20

30 Mar '20

The following changes since commit 16fbf79b0f83bc752cee8589279f1ebfe57b3b6e: Linux 5.6-rc7 (2020-03-22 18:31:56 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v5.7 for you to fetch changes up to 1c521d7e62262793789845989edca57dea24eb7d: Merge branch 'asoc-5.7' into asoc-next (2020-03-27 17:29:20 +0000) ---------------------------------------------------------------- ASoC: Updates for v5.7 This is a very big update for the core since Morimoto-san has been rather busy continuing his refactorings to clean up a lot of the cruft that we have accumilated over the years. We've also gained several new drivers, including initial (but still not complete) parts of the Intel SoundWire support. - Lots of refactorings to modernize the code from Morimoto-san. - Conversion of SND_SOC_ALL_CODECS to use imply from Geert Uytterhoeven. - Continued refactoring and fixing of the Intel support. - Soundwire and more advanced clocking support for Realtek RT5682. - Support for amlogic GX, Meson 8, Meson 8B and T9015 DAC, Broadcom DSL/PON, Ingenic JZ4760 and JZ4770, Realtek RL6231, and TI TAS2563 and TLV320ADCX140. ---------------------------------------------------------------- Akshu Agrawal (4): ASoC: amd: Add machine driver for Raven based platform ASoC: amd: Allow I2S wake event after ACP is powerd On ASoc: amd: Add DMIC switch capability to machine driver ASoC: amd: Fix compile warning of argument type Amadeusz Sławiński (1): ASoC: topology: Perform component check upfront Amery Song (2): ASoC: SOF: Intel: hda: remove unnecessary ROM IPC filter function ASoC: SOF: Intel: remove unnecessary waitq before loading firmware Baolin Wang (1): ASoC: sprd: Allow the MCDT driver to build into modules Bard Liao (5): ASoC: Return error if the function does not support multi-cpu ASoC: pcm: check if cpu-dai supports a given stream ASoC: SOF: Intel: hda-dai: add stream capability ASoC: SOF: topology: connect dai widget to all cpu-dais ASoC: SOF: Intel: hda: merge IPC, stream and SoundWire interrupt handlers Benjamin Gaignard (1): ASoC: Convert cirrus,cs42l51 to json-schema Cezary Rojewski (20): ALSA: core: Expand DMA buffer information ALSA: core: Implement compress page allocation and free routines ASoC: SOF: Intel: Account for compress streams when servicing IRQs ASoC: SOF: Implement Probe IPC API ASoC: SOF: Generic probe compress operations ASoC: SOF: Intel: Expose SDnFMT helpers ASoC: SOF: Intel: Probe compress operations ASoC: SOF: Provide probe debugfs support ASoC: SOF: Intel: Add Probe compress CPU DAIs ASoC: SOF: Fix probe point getter ASoC: Intel: Skylake: Remove superfluous chip initialization ASoC: Intel: Skylake: Select hda configuration permissively ASoC: Intel: Skylake: Enable codec wakeup during chip init ASoC: Intel: Skylake: Shield against no-NHLT configurations ASoC: Intel: Allow for ROM init retry on CNL platforms ASoC: Intel: Skylake: Await purge request ack on CNL ASoC: SOF: Intel: Fix stream cleanup on hw free ASoC: Intel: broadwell: Revert back SSP0 link to use dummy components ASoC: Intel: bdw-rt5677: Revert SSP0 link to use dummy components ASoC: Intel: bdw-rt5650: Revert SSP0 link to use dummy components Charles Keepax (1): ASoC: samsung: Update dependencies for Arizona machine drivers Colin Ian King (6): ASoC: Intel: mrfld: return error codes when an error occurs ASoC: rt5659: remove redundant assignment to variable idx ASoC: ti: davinci-mcasp: remove redundant assignment to variable ret ASoC: Intel: mrfld: fix incorrect check on p->sink ASoC: mchp-i2s-mcc: make signed 1 bit bitfields unsigned ASoC: amd: acp3x-pcm-dma: clean up two indentation issues Curtis Malainey (1): ASoC: Intel: Make glk+rt5682 echo ref dynamic Dafna Hirschfeld (1): dt-bindings: Convert the binding file google, cros-ec-codec.txt to yaml format. Dan Murphy (10): ASoC: tas2562: Add support for ISENSE and VSENSE dt-bindings: sound: Add TLV320ADCx140 dt bindings ASoC: tlv320adcx140: Add the tlv320adcx140 codec driver family ASoC: tas2562: Add support for digital volume control ASoC: tlv320adcx140: Add DRE and AGC support ASoC: tlv320adcx140: Add decimation filter support ASoC: dt-bindings: Add TAS2563 compatible to the TAS2562 binding ASoC: tas2562: Add entries for the TAS2563 audio amplifier ASoC: tlv320adcx140: Fix mic_bias and vref device tree verification ASoC: tlv320adcx140: Remove undocumented property Derek Fang (3): ASoC: rl6231: Add new supports on rl6231 ASoC: rt5682: Add CCF usage for providing I2S clks ASoC: rt5682: Add DAI clock binding info for WCLK/BCLK CCF usage Dmitry Osipenko (2): ASoC: tegra: tegra_wm8903: Support DAPM events for built-in microphone ASoC: tegra-wm8903: Document built-in microphone audio source Enric Balletbo i Serra (1): ASoC: amd: AMD RV RT5682 should depends on CROS_EC Geert Uytterhoeven (7): spi: pxa2xx: Enable support for compile-testing ASoC: Use imply for SND_SOC_ALL_CODECS ASoC: sh: fsi: Restore devm_ioremap() alignment ASoC: Fix SND_SOC_ALL_CODECS imply SPI fallout ASoC: Fix SND_SOC_ALL_CODECS imply I2C fallout ASoC: Fix SND_SOC_ALL_CODECS imply misc fallout ASoC: Fix SND_SOC_ALL_CODECS imply ac97 fallout Guennadi Liakhovetski (7): ASoC: (cosmetic) simplify dpcm_prune_paths() ASoC: export DPCM runtime update functions ASoC: codecs: hdac_hdmi: (cosmetic) remove redundant variable initialisations ASoC: Intel: skylake: (cosmetic) remove redundant variable initialisations ASoC: Intel: (cosmetic) simplify structure member access ASoC: SOF: (cosmetic) use for_each_pcm_streams() in sof_dai_load() ASoC: SOF: fix uninitialised "work" with VirtIO Gustavo A. R. Silva (1): ASoC: wm0010: Replace zero-length array with flexible-array member Jack Yu (2): ASoC: rt1015: fix typo for bypass boost control ASoC: rt1015: modify some structure to be static. James Schulman (1): MAINTAINERS: Update Cirrus Logic codec driver maintainers Jaska Uimonen (1): ASoC: SOF: ipc: check ipc return value before data copy Jeff Chang (1): ASoC: MT6660 update to 1.0.8_G Jerome Brunet (23): ASoC: core: allow a dt node to provide several components ASoC: meson: g12a: extract codec-to-codec utils ASoC: meson: aiu: add audio output dt-bindings ASoC: meson: aiu: add i2s and spdif support ASoC: meson: aiu: add hdmi codec control support ASoC: meson: aiu: add internal dac codec control support ASoC: meson: axg: extract sound card utils ASoC: meson: gx: add sound card dt-binding documentation ASoC: meson: gx: add sound card support ASoC: core: ensure component names are unique ASoC: meson: aiu: remove unused encoder structure ASoC: meson: aiu: fix clk bulk size allocation ASoC: meson: aiu: fix irq registration ASoC: meson: aiu: fix acodec dai input name init ASoC: meson: codec-glue: fix pcm format cast warning ASoC: meson: aiu: simplify component addition ASoC: fix card registration regression. ASoC: meson: add t9015 internal codec binding documentation ASoC: meson: add t9015 internal DAC driver ASoC: dpcm: remove confusing trace in dpcm_get_be() ASoC: meson: g12a: add toacodec dt-binding documentation ASoC: meson: g12a: add internal DAC glue driver ASoC: meson: axg-card: add toacodec support Johan Jonker (2): dt-bindings: sound: convert rockchip i2s bindings to yaml dt-bindings: sound: rockchip-i2s: add #sound-dai-cells property Jonghwan Choi (1): ASoC: tas2562: Fixed incorrect amp_level setting. Kai Vehmanen (10): ASoC: intel/skl/hda - add no-HDMI cases to generic HDA driver ASoC: SOF: Intel: hda: allow operation without i915 gfx ASoC: soc-pcm: fix state tracking error in snd_soc_component_open/close() MAINTAINERS: add entry for Sound Open Firmware drivers ASoC: Intel: boards: drop reverse deps for SND_HDA_CODEC_HDMI ASoC: Intel: sof_pcm512x: drop reverse deps for SND_HDA_CODEC_HDMI ASoC: Intel: sof_pcm512x: make HDMI optional for all platforms ASoC: SOF: Intel: hda: remove SND_SOC_SOF_HDA_COMMON_HDMI_CODEC ASoC: SOF: Intel: hda: do not leave clock gating off upon error ASoC: SOF: Intel: hda: call codec wake at chip init Karol Trzcinski (2): ASoC: SOF: Make sof_ipc_ext_data enum more rigid ASoC: SOF: Remove SOF_IPC_EXT_DMA_BUFFER Kevin Li (2): ASoC: brcm: Add DSL/PON SoC audio driver ASoC: brcm: DSL/PON SoC device tree bindings of audio driver Keyon Jie (2): ASoC: SOF: pcm: skip DMA buffer pre-allocation ASoC: SOF: Intel: hda-loader: clear the IPC ack bit after FW_PURGE done Kuninori Morimoto (95): ASoC: soc-pcm: add snd_soc_runtime_action() ASoC: soc-pcm: adjustment for DAI member 0 reset ASoC: soc-pcm: add for_each_dapm_widgets() macro ASoC: soc-pcm: don't use bit-OR'ed error ASoC: soc-pcm: call snd_soc_dai_startup()/shutdown() once ASoC: soc-pcm: call snd_soc_component_open/close() once ASoC: soc-pcm: move soc_pcm_close() next to soc_pcm_open() ASoC: soc-pcm: tidyup soc_pcm_open() order ASoC: soc-pcm: move dai_get_widget() ASoC: soc-pcm: use dai_get_widget() at dpcm_get_be() ASoC: soc-pcm: use dai_get_widget() at dpcm_end_walk_at_be() ASoC: soc-pcm: use dpcm_get_be() at dpcm_end_walk_at_be() ASoC: soc-pcm: remove soc_dpcm_be_digital_mute() ASoC: soc-pcm: remove snd_soc_dpcm_be_get/set_state() ASoC: soc-pcm: add snd_soc_dpcm_can_be() and remove duplicate code ASoC: soc-pcm: use goto and remove multi return ASoC: soc-pcm: merge playback/cature_active into stream_active ALSA: pcm.h: add for_each_pcm_streams() ASoC: soc-core: use for_each_pcm_streams() macro ASoC: soc-pcm: use for_each_pcm_streams() macro ASoC: soc-generic-dmaengine-pcm: use for_each_pcm_streams() macro ASoC: dwc: dwc-i2s: use for_each_pcm_streams() macro ASoC: fsl: fsl_asrc_dma: use for_each_pcm_streams() macro ASoC: qcom: lpass-platform: use for_each_pcm_streams() macro ASoC: sof: sof-audio: use for_each_pcm_streams() macro ALSA: usx2y: use for_each_pcm_streams() macro ASoC: soundwaire: qcom: use for_each_rtd_codec_dai() macro ASoC: qcom: sdm845: use for_each_rtd_codec_dai() macro ASoC: qcom: apq8016_sbc: use for_each_rtd_codec_dai() macro ASoC: intel: cml_rt1011_rt5682: use for_each_rtd_codec_dai() macro ASoC: intel: kbl_da7219_max98927: use for_each_rtd_codec_dai() macro ASoC: mediatek: mt8183-da7219-max98357: use for_each_rtd_codec_dai() macro ASoC: soc-pcm: add snd_soc_dai_get_pcm_stream() ASoC: soc-pcm: cleanup soc_pcm_apply_msb() ASoC: soc-pcm: add snd_soc_dai_get_widget() ASoC: soc-pcm: merge dpcm_run_new/old_update() into dpcm_fe_runtime_update() ASoC: soc-pcm: move dpcm_path_put() to soc-pcm.c ASoC: soc-pcm: move CONFIG_DEBUG_FS functions to top side ASoC: soc-pcm: add dpcm_create/remove_debugfs_state() ASoC: soc-dapm: don't use rtd->cpu_dai on for_each_rtd_cpu_dai() ASoC: soc-pcm: use defined stream ASoC: soc-pcm: remove duplicate be check from dpcm_add_paths() ASoC: soc-pcm: move dpcm_fe_dai_close() ASoC: soc-pcm: add dpcm_fe_dai_cleanup() ASoC: soc-pcm: use snd_soc_dai_get_pcm_stream() at dpcm_set_fe_runtime() ASoC: soc-pcm: tidyup dulicate handing at dpcm_fe_dai_startup() ASoC: soc-pcm: check DAI's activity more simply ASoC: soc-pcm: Do Digital Mute for both CPU/Codec in same timing. ASoC: simple-card-utils: use for_each_pcm_streams() ASoC: soc.h: add for_each_rtd_codecs/cpus_dai() macro ASoC: Intel: use for_each_rtd_codecs/cpus_dai() macro ASoC: mediatek: use for_each_rtd_codecs/cpus_dai() macro ASoC: meson: use for_each_rtd_codecs/cpus_dai() macro ASoC: qcom: use for_each_rtd_codecs/cpus_dai() macro ASoC: soc: use for_each_rtd_codecs/cpus_dai() macro ASoC: soc.h: remove non plural form for_each_xxx macro ASoC: soc-dapm: add for_each_card_dapms() macro ASoC: soc-dapm: add for_each_card_widgets() macro ASoC: soc-core: Merge CPU/Codec DAIs ASoC: soc-core: Merge for_each_rtd_cpu/codec_dais() ASoC: soc-dapm: Merge for_each_rtd_cpu/codec_dais() ASoC: soc-pcm: Merge for_each_rtd_cpu/codec_dais() ASoC: soc-core: Merge CPU/Codec for soc_dai_pcm_new() ASoC: soc-core: add asoc_rtd_to_cpu/codec() macro ASoC: amd: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: atmel: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: au1x: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: bcm: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: cirrus: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: dwc: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: fsl: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: generic: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: img: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: intel: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: kirkwood: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: mediatek: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: meson: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: mxs: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: pxa: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: qcom: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: rockchip: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: samsung: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: sh: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: sof: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: sprd: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: stm: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: sunxi: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: tegra: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: ti: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: txx9: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: uniphier: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: ux500: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: xtensa: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: arm: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: codecs: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer Lukas Bulwahn (1): ASoC: MT6660: make spdxcheck.py happy Marek Szyprowski (1): ASoC: samsung: Silence warnings during deferred probe Mark Brown (37): Merge branch 'for-5.6' of https://git.kernel.org/.../broonie/sound into asoc-5.7 ASoC: soc-pcm cleanup step4 Merge branch 'for-5.6' of https://git.kernel.org/.../broonie/sound into asoc-5.7 SoundWire: ASoC interfaces for multi-cpu dais and DisCo helpers ASoC: Add Multi CPU DAI support Merge series "ASoC: SOF: updates for 5.7" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge branch 'for-5.6' of https://git.kernel.org/.../broonie/sound into asoc-5.7 Merge branch 'for-5.6' of https://git.kernel.org/.../broonie/sound into asoc-5.7 Merge branch 'for-5.6' of https://git.kernel.org/.../broonie/sound into asoc-5.7 Merge series "ASoC: wcd934x: minor fixes" from Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org>: Merge series "simple-audio-card codec2codec support" from Samuel Holland <samuel(a)sholland.org>: Merge series "ASoC: soc-pcm cleanup step5" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "ASoC: soc-pcm cleanup step6" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "ASoC: rt5682: fix Kconfig/compilation issues when I2C is not selected" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge tag 'v5.6-rc5' into asoc-5.7 Merge series "ASoC: qdsp6: fix default FE dais and routings." from Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org>: Merge series "ASoC: SOF: preparatory patches" from Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com>: Merge series "ASoC: brcm: add dsl and pon chip audio driver" from Kevin Li <kevin-ke.li(a)broadcom.com>: Merge series "ASoC: Intel: machine driver updates for 5.7" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: SOF: multi-cpu dais, IPC and Intel improvements for 5.7" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge branch 'for-5.6' of https://git.kernel.org/.../broonie/sound into asoc-5.7 Merge series "ALSA: compress: Add wma, alac and ape support" from Vinod Koul <vkoul(a)kernel.org>: Merge series "ASoC: sdm845: fix soundwire stream handling" from Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org>: Merge series "ASoC: stm32: manage rebind issue" from Olivier Moysan <olivier.moysan(a)st.com>: Merge branch 'topic/ro_wordlength' of git://git.kernel.org/.../vkoul/soundwire into asoc-5.7 Merge series "Support built-in Mic on Tegra boards that use WM8903" from Dmitry Osipenko <digetx(a)gmail.com>: ASoC: pxa: Select regmap from AC'97 machines ASoC: pxa: Enable AC'97 bus support for PXA machines Merge series "ASoC: SOF: cleanups and improvements" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: Intel: machine driver updates" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: rt1308-sdw: configure amplifier with set_tdm_slot()" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: SOF: Intel: add SoundWire support" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: remove rtd->cpu/codec_dai{s}" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge branch 'for-5.6' of https://git.kernel.org/.../broonie/sound into asoc-5.7 Merge series "ASoC: Intel: add SoundWire machine driver" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge branch 'asoc-5.6' into asoc-linus Merge branch 'asoc-5.7' into asoc-next Martin Blumenstingl (3): ASoC: meson: aiu: Document Meson8 and Meson8b support in the dt-bindings ASoC: meson: aiu: introduce a struct for platform specific information ASoC: meson: aiu: add support for the Meson8 and Meson8b SoC families Nathan Chancellor (1): ASoC: wcd934x: Remove some unnecessary NULL checks Naveen Manohar (2): ASoC: Intel: common: add match table for TGL RT5682 SoundWire driver ASoC: Intel: sof_sdw: Add Volteer support with RT5682 SNDW helper function Oder Chiou (8): ASoC: rt5682: Add the field "is_sdw" of private data ASoC: rt5682: Add the soundwire support ASoC: rt5682: Revise the function name ASoC: rt5682: Revise the DAC1 volume setting ASoC: rt5682: Fine tune the HP performance in soundwire mode ASoC: rt5682: Add a property for DMIC clock rate ASoC: rt5682: Add a property for DMIC delay ASoC: rt5682: Add the descriptions for the DMIC clock rate and delay settings Olivier Moysan (11): ASoC: stm32: sai: manage error when getting reset controller ASoC: stm32: spdifrx: manage error when getting reset controller ASoC: stm32: i2s: manage error when getting reset controller ASoC: stm32: sai: improve error management on probe deferral ASoC: stm32: spdifrx: improve error management on probe deferral ASoC: stm32: i2s: improve error management on probe deferral ASoC: dt-bindings: stm32: convert i2s to json-schema ASoC: dt-bindings: stm32: convert spdfirx to json-schema ASoC: stm32: spdifrx: fix regmap status check ASoC: stm32: spdifrx: manage rebind issue ASoC: stm32: i2s: manage rebind issue Paul Cercueil (6): ASoC: jz4740-i2s: Fix divider written at incorrect offset in register ASoC: jz4740-i2s: Add local dev variable in probe function ASoC: jz4740-i2s: Avoid passing enum as match data ASoC: jz4740-i2s: Add support for the JZ4760 ASoC: jz4740-i2s: Add support for the JZ4770 ASoC: Convert jz4740-i2s doc to YAML Paul Olaru (3): ASoC: SOF: Rename i.MX8 platform to i.MX8X ASoC: SOF: imx8: Add ops for i.MX8QM ASoC: SOF: Add i.MX8QM device descriptor Peter Ujfalusi (5): ASoC: ti: Add udma-pcm platform driver for UDMA ASoC: ti: davinci-mcasp: Add support for platforms using UDMA ALSA: dmaengine_pcm: Consider DMA cache caused delay in pointer callback ALSA: dmaengine_pcm: Consider DMA cache caused delay in pointer callback ALSA: dmaengine_pcm: No need to take runtime reference twice in pcm_pointer Pierre-Louis Bossart (20): ASoC: Intel: sof_pcm512x: add support for SOF platforms with pcm512x ASoC: Intel: BXT: switch pcm512x based boards to sof_pcm512x ASoC: Intel: CHT: add support for pcm512x boards ASoC: soc-dai: add get_sdw_stream() callback ASoC: soc-core: disable route checks for legacy devices ASoC: Intel: skl_nau88l25_ssm4567: disable route checks ASoC: rt5682: fix unmet dependencies ASoC: rt5682-sdw: fix 'defined but not used' pm functions ASoC: Intel: don't use GFP_ATOMIC for machine driver contexts ASoC: rt1308-sdw: add set_tdm_slot() support ASoC: rt1308-sdw: use slot and rx_mask to configure stream ASoC: soc-acpi: expand description of _ADR-based devices ASoC: SOF: Intel: add SoundWire configuration interface ASoC: SOF: IPC: dai-intel: move ALH declarations in header file ASoC: SOF: Intel: hda: add SoundWire stream config/free callbacks ASoC: SOF: Intel: hda: initial SoundWire machine driver autodetect ASoC: SOF: Intel: hda: disable SoundWire interrupts on suspend ASoC: SOF: Intel: hda: add parameter to control SoundWire clock stop quirks ASoC: SOF: Intel: hda-ctrl: add reset cycle before parsing capabilities ASoC: Intel: boards: add sof_sdw machine driver Rander Wang (3): ASoC: SOF: Intel: hda: add WAKEEN interrupt support for SoundWire Asoc: SOF: Intel: hda: check SoundWire wakeen interrupt in irq thread ASoC: Intel: soc-acpi: update topology and driver name for SoundWire platforms Ranjani Sridharan (11): ASoC: SOF: Do not reset hw_params for streams that ignored suspend ASoC: SOF: pm: Unify suspend/resume routines ASoC: SOF: Add system_suspend_target field to struct snd_sof_dev ASoC: SOF: pm: Introduce DSP power states ASoC: SOF: Move DSP power state transitions to platform-specific ops ASoC: SOF: audio: Add helper to check if only D0i3 streams are active ASoC: SOF: Intel: hda: Amend the DSP state transition diagram ASoC: SOF: Intel: cnl: Implement feature to support DSP D0i3 in S0 ASoC: SOF: Intel: hda: Allow trace DMA in S0 when DSP is in D0I3 for debug ASoC: SOF: Intel: hda: use snd_sof_dsp_set_power_state() op ASoC: SOF: Intel: hda: Improve DSP state logging Ravulapati Vishnu vardhan rao (1): ASoC: amd: Adding TDM support in hw_params. Rob Herring (1): ASoC: dt-bindings: google, cros-ec-codec: Fix dtc warnings in example Robin Murphy (2): ASoC: dt-bindings: Make RK3328 codec GPIO explicit ASoC: rockchip: Make RK3328 GPIO_MUTE control explicit Samuel Holland (4): ASoC: sun8i-codec: Remove unused dev from codec struct ALSA: pcm: Add a standalone version of snd_pcm_limit_hw_rates ASoC: pcm: Export parameter intersection logic ASoC: simple-card: Add support for codec2codec DAI links Sathyanarayana Nujella (2): ASoC: Intel: sof_rt5682: Add support for tgl-max98373-rt5682 ASoC: Intel: common: Add mach table for tgl-max98373-rt5682 Shreyas NC (4): ASoC: Add initial support for multiple CPU DAIs ASoC: Add multiple CPU DAI support for PCM ops ASoC: Add dapm_add_valid_dai_widget helper ASoC: Add multiple CPU DAI support in DAPM Shuming Fan (3): ASoC: rt5682: fix the random recording noise of headset ASoC: rt5682: remove noisy debug messages ASoC: rt5682: move DAI clock registry to I2S mode Srinivas Kandagatla (10): ASoC: wcd934x: fix High Accuracy Buck enable ASoC: wcd934x: remove unused headers ASoC: wcd9335: fix address map representation ASoC: qdsp6: q6asm-dai: only enable dais from device tree ASoC: qdsp6: q6routing: remove default routing ASoC: codecs: wsa881x: request gpio direction before setting ASoC: soc-dai: return proper error for get_sdw_stream() ASoC: codecs: wsa881x: remove soundwire stream handling ASoC: qcom: sdm845: handle soundwire stream soundwire: stream: Add read_only_wordlength flag to port properties Stephan Gerhold (1): ASoC: soc-pcm: fix regression in soc_new_pcm() Takashi Iwai (2): ASoC: pcm: Fix (again) possible buffer overflow in dpcm state sysfs output ASoC: wm_adsp: Use scnprintf() for the limited buffer output Tomasz Lauda (1): ASoC: SOF: add core id to sof_ipc_comp Tzung-Bi Shih (12): ASoC: mediatek: mt8173-rt5650: support HDMI jack reporting drm/mediatek: exit earlier if failed to register audio driver drm/mediatek: support HDMI jack status reporting ASoC: max98357a: move control of SD_MODE to DAPM ASoC: mediatek: mt8183-da7219: add speaker switch ASoC: mediatek: mt8183-da7219: change supported formats of DL2 and UL1 ASoC: mediatek: mt8183-da7219: pull TDM GPIO pins down when probed ASoC: mediatek: mt8183-da7219: support TDM out and 8ch I2S out ASoC: mediatek: mt8183-da7219: apply some refactors drm/mediatek: fix race condition for HDMI jack status reporting ASoC: dapm: select sleep_state when initializing PINCTRL widget ASoC: mediatek: mt8183-da7219: use SND_SOC_DAPM_PINCTRL in TDM out Vinod Koul (9): ALSA: compress: add wma codec profiles ALSA: compress: Add wma decoder params ASoC: qcom: q6asm: pass codec profile to q6asm_open_write ASoC: qcom: q6asm: add support to wma config ASoC: qcom: q6asm-dai: add support to wma decoder ALSA: compress: add alac & ape decoder params ASoC: qcom: q6asm: add support for alac and ape configs ASoC: qcom: q6asm-dai: add support for ALAC and APE decoders ALSA: compress: bump the version Wolfram Sang (1): ASoC: pxa: magician: convert to use i2c_new_client_device() Yong Zhi (4): ASoC: Intel: sof_da7219_max98373: Add support for max98360a speaker amp ASoC: Intel: sof_rt5682: Add rt1015 speaker amp support ASoC: max98357a: Add ACPI HID MAX98360A ASoC: intel: sof_da7219_max98373: Add speaker switch Yu-Hsuan Hsu (1): ASoC: cros_ec_codec: Support setting bclk ratio YueHaibing (6): ASoC: Remove unused including <linux/version.h> ASoC: wcd934x: Remove set but not unused variable 'hph_comp_ctrl7' ASoC: rt5682: Make rt5682_clock_config static ASoC: rt1015: set snd_soc_dai_ops in rt1015_dai driver ASoc: amd: acp3x: Add missing include <linux/io.h> ASoC: wm8974: remove unused variables derek.fang (1): ASoC: rt5682: Enable PLL2 function kbuild test robot (1): ASoC: meson: aiu: fix semicolon.cocci warnings tangbin (2): ASoC: zte: zx-spdif: remove redundant dev_err message ASoC: zte: zx-tdm: remove redundant variables dev .../devicetree/bindings/sound/amlogic,aiu.yaml | 113 ++ .../bindings/sound/amlogic,g12a-toacodec.yaml | 51 + .../bindings/sound/amlogic,gx-sound-card.yaml | 113 ++ .../devicetree/bindings/sound/amlogic,t9015.yaml | 58 + .../bindings/sound/brcm,bcm63xx-audio.txt | 29 + .../devicetree/bindings/sound/cirrus,cs42l51.yaml | 69 + .../devicetree/bindings/sound/cs42l51.txt | 33 - .../bindings/sound/google,cros-ec-codec.txt | 44 - .../bindings/sound/google,cros-ec-codec.yaml | 67 + .../devicetree/bindings/sound/ingenic,aic.yaml | 92 ++ .../bindings/sound/ingenic,jz4740-i2s.txt | 23 - .../bindings/sound/nvidia,tegra-audio-wm8903.txt | 1 + .../bindings/sound/rockchip,rk3328-codec.txt | 7 +- .../devicetree/bindings/sound/rockchip-i2s.txt | 49 - .../devicetree/bindings/sound/rockchip-i2s.yaml | 111 ++ Documentation/devicetree/bindings/sound/rt5682.txt | 18 + .../devicetree/bindings/sound/st,stm32-i2s.txt | 62 - .../devicetree/bindings/sound/st,stm32-i2s.yaml | 87 ++ .../devicetree/bindings/sound/st,stm32-spdifrx.txt | 56 - .../bindings/sound/st,stm32-spdifrx.yaml | 80 + .../devicetree/bindings/sound/tas2562.txt | 2 +- .../devicetree/bindings/sound/tlv320adcx140.yaml | 82 + Documentation/sound/soc/codec-to-codec.rst | 9 +- MAINTAINERS | 15 +- drivers/gpu/drm/mediatek/mtk_hdmi.c | 54 +- drivers/soundwire/qcom.c | 7 +- drivers/soundwire/stream.c | 16 +- drivers/spi/Kconfig | 2 +- include/dt-bindings/sound/meson-aiu.h | 18 + include/dt-bindings/sound/meson-g12a-toacodec.h | 10 + include/linux/soundwire/sdw.h | 2 + include/sound/compress_driver.h | 40 +- include/sound/hdaudio.h | 2 + include/sound/pcm.h | 14 +- include/sound/rt5682.h | 10 + include/sound/soc-acpi.h | 39 +- include/sound/soc-dai.h | 44 +- include/sound/soc-dapm.h | 6 + include/sound/soc-dpcm.h | 20 +- include/sound/soc.h | 44 +- include/sound/sof/dai-intel.h | 18 +- include/sound/sof/header.h | 11 + include/sound/sof/info.h | 22 +- include/sound/sof/topology.h | 3 +- include/uapi/sound/compress_offload.h | 2 +- include/uapi/sound/compress_params.h | 37 +- include/uapi/sound/sof/abi.h | 2 +- sound/arm/pxa2xx-pcm-lib.c | 8 +- sound/core/compress_offload.c | 42 + sound/core/pcm_dmaengine.c | 6 +- sound/core/pcm_misc.c | 18 +- sound/soc/amd/Kconfig | 10 + sound/soc/amd/Makefile | 2 + sound/soc/amd/acp-da7219-max98357a.c | 2 +- sound/soc/amd/acp-rt5645.c | 4 +- sound/soc/amd/acp3x-rt5682-max9836.c | 376 +++++ sound/soc/amd/raven/acp3x-i2s.c | 44 +- sound/soc/amd/raven/acp3x-pcm-dma.c | 6 +- sound/soc/amd/raven/pci-acp3x.c | 7 +- sound/soc/atmel/atmel-pcm-dma.c | 4 +- sound/soc/atmel/atmel-pcm-pdc.c | 2 +- sound/soc/atmel/atmel_wm8904.c | 2 +- sound/soc/atmel/mchp-i2s-mcc.c | 8 +- sound/soc/atmel/mikroe-proto.c | 2 +- sound/soc/atmel/sam9g20_wm8731.c | 2 +- sound/soc/atmel/sam9x5_wm8731.c | 2 +- sound/soc/au1x/db1200.c | 2 +- sound/soc/au1x/dbdma2.c | 2 +- sound/soc/au1x/dma.c | 2 +- sound/soc/au1x/psc-ac97.c | 2 +- sound/soc/bcm/Kconfig | 9 + sound/soc/bcm/Makefile | 4 + sound/soc/bcm/bcm63xx-i2s-whistler.c | 317 ++++ sound/soc/bcm/bcm63xx-i2s.h | 90 ++ sound/soc/bcm/bcm63xx-pcm-whistler.c | 485 ++++++ sound/soc/bcm/cygnus-pcm.c | 22 +- sound/soc/cirrus/edb93xx.c | 4 +- sound/soc/cirrus/snappercl15.c | 4 +- sound/soc/codecs/Kconfig | 627 ++++---- sound/soc/codecs/Makefile | 4 + sound/soc/codecs/cros_ec_codec.c | 25 +- sound/soc/codecs/cs4271.c | 4 +- sound/soc/codecs/cs47l15.c | 4 +- sound/soc/codecs/cs47l24.c | 6 +- sound/soc/codecs/cs47l35.c | 6 +- sound/soc/codecs/cs47l85.c | 6 +- sound/soc/codecs/cs47l90.c | 6 +- sound/soc/codecs/cs47l92.c | 4 +- sound/soc/codecs/hdac_hdmi.c | 6 +- sound/soc/codecs/max98357a.c | 37 +- sound/soc/codecs/mt6660.c | 81 +- sound/soc/codecs/rk3328_codec.c | 31 +- sound/soc/codecs/rl6231.c | 1 + sound/soc/codecs/rl6231.h | 2 +- sound/soc/codecs/rt1015.c | 10 +- sound/soc/codecs/rt1308-sdw.c | 38 +- sound/soc/codecs/rt1308-sdw.h | 2 + sound/soc/codecs/rt5659.c | 2 +- sound/soc/codecs/rt5682-sdw.c | 333 ++++ sound/soc/codecs/rt5682-sdw.h | 20 + sound/soc/codecs/rt5682.c | 1298 ++++++++++++++-- sound/soc/codecs/rt5682.h | 100 +- sound/soc/codecs/tas2562.c | 121 +- sound/soc/codecs/tas2562.h | 12 +- sound/soc/codecs/tlv320adcx140.c | 920 +++++++++++ sound/soc/codecs/tlv320adcx140.h | 131 ++ sound/soc/codecs/wcd9335.c | 18 +- sound/soc/codecs/wcd9335.h | 7 +- sound/soc/codecs/wcd934x.c | 37 +- sound/soc/codecs/wm0010.c | 2 +- sound/soc/codecs/wm5110.c | 6 +- sound/soc/codecs/wm8974.c | 8 - sound/soc/codecs/wm_adsp.c | 14 +- sound/soc/codecs/wsa881x.c | 46 +- sound/soc/dwc/dwc-i2s.c | 8 +- sound/soc/dwc/dwc-pcm.c | 2 +- sound/soc/fsl/eukrea-tlv320.c | 4 +- sound/soc/fsl/fsl-asoc-card.c | 10 +- sound/soc/fsl/fsl_asrc_dma.c | 10 +- sound/soc/fsl/fsl_spdif.c | 10 +- sound/soc/fsl/fsl_ssi.c | 8 +- sound/soc/fsl/imx-audmix.c | 8 +- sound/soc/fsl/imx-mc13783.c | 4 +- sound/soc/fsl/imx-sgtl5000.c | 2 +- sound/soc/fsl/mpc5200_dma.c | 10 +- sound/soc/fsl/mpc5200_psc_i2s.c | 2 +- sound/soc/fsl/mpc8610_hpcd.c | 4 +- sound/soc/fsl/mx27vis-aic32x4.c | 4 +- sound/soc/fsl/p1022_ds.c | 4 +- sound/soc/fsl/p1022_rdk.c | 4 +- sound/soc/fsl/wm1133-ev1.c | 6 +- sound/soc/generic/simple-card-utils.c | 60 +- sound/soc/img/img-i2s-in.c | 2 +- sound/soc/img/img-i2s-out.c | 2 +- sound/soc/intel/atom/sst-atom-controls.c | 2 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 6 +- sound/soc/intel/atom/sst/sst_pci.c | 2 +- sound/soc/intel/boards/Kconfig | 57 +- sound/soc/intel/boards/Makefile | 12 +- sound/soc/intel/boards/bdw-rt5650.c | 15 +- sound/soc/intel/boards/bdw-rt5677.c | 15 +- sound/soc/intel/boards/broadwell.c | 13 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 10 +- sound/soc/intel/boards/bxt_rt298.c | 10 +- sound/soc/intel/boards/byt-max98090.c | 2 +- sound/soc/intel/boards/byt-rt5640.c | 4 +- sound/soc/intel/boards/bytcht_cx2072x.c | 10 +- sound/soc/intel/boards/bytcht_da7213.c | 10 +- sound/soc/intel/boards/bytcht_es8316.c | 8 +- sound/soc/intel/boards/bytcht_nocodec.c | 4 +- sound/soc/intel/boards/bytcr_rt5640.c | 8 +- sound/soc/intel/boards/bytcr_rt5651.c | 8 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 8 +- sound/soc/intel/boards/cht_bsw_nau8824.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 16 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 8 +- sound/soc/intel/boards/cml_rt1011_rt5682.c | 13 +- sound/soc/intel/boards/glk_rt5682_max98357a.c | 13 +- sound/soc/intel/boards/haswell.c | 4 +- sound/soc/intel/boards/kbl_da7219_max98357a.c | 8 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 14 +- sound/soc/intel/boards/kbl_rt5660.c | 6 +- sound/soc/intel/boards/kbl_rt5663_max98927.c | 12 +- .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 12 +- sound/soc/intel/boards/skl_hda_dsp_common.h | 4 + sound/soc/intel/boards/skl_hda_dsp_generic.c | 27 +- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 14 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 19 +- sound/soc/intel/boards/skl_rt286.c | 8 +- sound/soc/intel/boards/sof_da7219_max98373.c | 83 +- sound/soc/intel/boards/sof_maxim_common.c | 80 + sound/soc/intel/boards/sof_maxim_common.h | 24 + sound/soc/intel/boards/sof_pcm512x.c | 448 ++++++ sound/soc/intel/boards/sof_rt5682.c | 137 +- sound/soc/intel/boards/sof_sdw.c | 962 ++++++++++++ sound/soc/intel/boards/sof_sdw_common.h | 114 ++ sound/soc/intel/boards/sof_sdw_dmic.c | 42 + sound/soc/intel/boards/sof_sdw_hdmi.c | 97 ++ sound/soc/intel/boards/sof_sdw_rt1308.c | 151 ++ sound/soc/intel/boards/sof_sdw_rt5682.c | 126 ++ sound/soc/intel/boards/sof_sdw_rt700.c | 125 ++ sound/soc/intel/boards/sof_sdw_rt711.c | 156 ++ sound/soc/intel/boards/sof_sdw_rt715.c | 42 + sound/soc/intel/common/soc-acpi-intel-bxt-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-cht-match.c | 7 + sound/soc/intel/common/soc-acpi-intel-cml-match.c | 111 +- sound/soc/intel/common/soc-acpi-intel-icl-match.c | 103 +- sound/soc/intel/common/soc-acpi-intel-jsl-match.c | 34 +- sound/soc/intel/common/soc-acpi-intel-tgl-match.c | 92 +- sound/soc/intel/haswell/sst-haswell-pcm.c | 26 +- sound/soc/intel/skylake/bxt-sst.c | 3 - sound/soc/intel/skylake/cnl-sst.c | 35 +- sound/soc/intel/skylake/skl-nhlt.c | 3 +- sound/soc/intel/skylake/skl-pcm.c | 20 +- sound/soc/intel/skylake/skl-sst-dsp.h | 2 + sound/soc/intel/skylake/skl.c | 33 +- sound/soc/jz4740/jz4740-i2s.c | 78 +- sound/soc/kirkwood/armada-370-db.c | 2 +- sound/soc/kirkwood/kirkwood-dma.c | 2 +- sound/soc/mediatek/common/mtk-afe-fe-dai.c | 10 +- .../soc/mediatek/common/mtk-afe-platform-driver.c | 2 +- sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 2 +- sound/soc/mediatek/mt2701/mt2701-cs42448.c | 4 +- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 4 +- sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 2 +- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 2 +- sound/soc/mediatek/mt8173/mt8173-max98090.c | 4 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c | 4 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 6 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 23 +- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 2 +- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 117 +- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 2 +- sound/soc/meson/Kconfig | 41 + sound/soc/meson/Makefile | 19 + sound/soc/meson/aiu-acodec-ctrl.c | 203 +++ sound/soc/meson/aiu-codec-ctrl.c | 151 ++ sound/soc/meson/aiu-encoder-i2s.c | 365 +++++ sound/soc/meson/aiu-encoder-spdif.c | 209 +++ sound/soc/meson/aiu-fifo-i2s.c | 153 ++ sound/soc/meson/aiu-fifo-spdif.c | 186 +++ sound/soc/meson/aiu-fifo.c | 223 +++ sound/soc/meson/aiu-fifo.h | 50 + sound/soc/meson/aiu.c | 388 +++++ sound/soc/meson/aiu.h | 89 ++ sound/soc/meson/axg-card.c | 414 +---- sound/soc/meson/axg-fifo.c | 2 +- sound/soc/meson/g12a-toacodec.c | 252 +++ sound/soc/meson/g12a-tohdmitx.c | 219 +-- sound/soc/meson/gx-card.c | 141 ++ sound/soc/meson/meson-card-utils.c | 385 +++++ sound/soc/meson/meson-card.h | 55 + sound/soc/meson/meson-codec-glue.c | 149 ++ sound/soc/meson/meson-codec-glue.h | 32 + sound/soc/meson/t9015.c | 333 ++++ sound/soc/mxs/mxs-sgtl5000.c | 4 +- sound/soc/pxa/Kconfig | 22 + sound/soc/pxa/brownstone.c | 4 +- sound/soc/pxa/corgi.c | 4 +- sound/soc/pxa/hx4700.c | 4 +- sound/soc/pxa/imote2.c | 4 +- sound/soc/pxa/magician.c | 14 +- sound/soc/pxa/mioa701_wm9713.c | 4 +- sound/soc/pxa/mmp-pcm.c | 2 +- sound/soc/pxa/mmp-sspa.c | 2 +- sound/soc/pxa/poodle.c | 4 +- sound/soc/pxa/pxa2xx-i2s.c | 2 +- sound/soc/pxa/spitz.c | 4 +- sound/soc/pxa/ttc-dkb.c | 2 +- sound/soc/pxa/z2.c | 4 +- sound/soc/pxa/zylonite.c | 6 +- sound/soc/qcom/Kconfig | 2 +- sound/soc/qcom/apq8016_sbc.c | 9 +- sound/soc/qcom/apq8096.c | 6 +- sound/soc/qcom/lpass-platform.c | 4 +- sound/soc/qcom/qdsp6/q6asm-dai.c | 173 ++- sound/soc/qcom/qdsp6/q6asm.c | 243 ++- sound/soc/qcom/qdsp6/q6asm.h | 51 +- sound/soc/qcom/qdsp6/q6routing.c | 21 +- sound/soc/qcom/sdm845.c | 105 +- sound/soc/qcom/storm.c | 2 +- sound/soc/rockchip/rk3288_hdmi_analog.c | 4 +- sound/soc/rockchip/rk3399_gru_sound.c | 16 +- sound/soc/rockchip/rockchip_max98090.c | 6 +- sound/soc/rockchip/rockchip_rt5645.c | 6 +- sound/soc/samsung/Kconfig | 4 +- sound/soc/samsung/arndale.c | 10 +- sound/soc/samsung/bells.c | 16 +- sound/soc/samsung/h1940_uda1380.c | 2 +- sound/soc/samsung/i2s.c | 2 +- sound/soc/samsung/jive_wm8750.c | 4 +- sound/soc/samsung/littlemill.c | 16 +- sound/soc/samsung/lowland.c | 6 +- sound/soc/samsung/neo1973_wm8753.c | 10 +- sound/soc/samsung/odroid.c | 6 +- sound/soc/samsung/pcm.c | 4 +- sound/soc/samsung/rx1950_uda1380.c | 2 +- sound/soc/samsung/s3c-i2s-v2.c | 2 +- sound/soc/samsung/s3c24xx_simtec.c | 4 +- sound/soc/samsung/s3c24xx_uda134x.c | 6 +- sound/soc/samsung/smartq_wm8987.c | 4 +- sound/soc/samsung/smdk_spdif.c | 2 +- sound/soc/samsung/smdk_wm8580.c | 2 +- sound/soc/samsung/smdk_wm8994.c | 4 +- sound/soc/samsung/smdk_wm8994pcm.c | 6 +- sound/soc/samsung/snow.c | 8 +- sound/soc/samsung/spdif.c | 8 +- sound/soc/samsung/speyside.c | 10 +- sound/soc/samsung/tm2_wm5110.c | 19 +- sound/soc/samsung/tobermory.c | 10 +- sound/soc/sh/dma-sh7760.c | 16 +- sound/soc/sh/fsi.c | 5 +- sound/soc/sh/migor.c | 6 +- sound/soc/sh/rcar/core.c | 2 +- sound/soc/soc-compress.c | 5 +- sound/soc/soc-core.c | 290 ++-- sound/soc/soc-dai.c | 18 +- sound/soc/soc-dapm.c | 220 +-- sound/soc/soc-generic-dmaengine-pcm.c | 26 +- sound/soc/soc-pcm.c | 1627 +++++++++----------- sound/soc/soc-topology.c | 24 +- sound/soc/sof/Kconfig | 9 + sound/soc/sof/Makefile | 1 + sound/soc/sof/compress.c | 146 ++ sound/soc/sof/compress.h | 31 + sound/soc/sof/core.c | 10 +- sound/soc/sof/debug.c | 226 +++ sound/soc/sof/imx/imx8.c | 57 +- sound/soc/sof/intel/Kconfig | 20 +- sound/soc/sof/intel/Makefile | 1 + sound/soc/sof/intel/apl.c | 9 + sound/soc/sof/intel/cnl.c | 51 +- sound/soc/sof/intel/hda-codec.c | 11 +- sound/soc/sof/intel/hda-compress.c | 114 ++ sound/soc/sof/intel/hda-ctrl.c | 40 +- sound/soc/sof/intel/hda-dai.c | 130 +- sound/soc/sof/intel/hda-dsp.c | 331 +++- sound/soc/sof/intel/hda-ipc.c | 24 +- sound/soc/sof/intel/hda-loader.c | 40 +- sound/soc/sof/intel/hda-pcm.c | 8 +- sound/soc/sof/intel/hda-stream.c | 27 +- sound/soc/sof/intel/hda.c | 433 +++++- sound/soc/sof/intel/hda.h | 120 +- sound/soc/sof/ipc.c | 41 +- sound/soc/sof/loader.c | 6 - sound/soc/sof/ops.h | 59 +- sound/soc/sof/pcm.c | 19 +- sound/soc/sof/pm.c | 176 +-- sound/soc/sof/probe.c | 290 ++++ sound/soc/sof/probe.h | 85 + sound/soc/sof/sof-audio.c | 59 +- sound/soc/sof/sof-audio.h | 6 +- sound/soc/sof/sof-of-dev.c | 10 + sound/soc/sof/sof-priv.h | 71 +- sound/soc/sof/topology.c | 25 +- sound/soc/sprd/Kconfig | 2 +- sound/soc/sprd/sprd-mcdt.h | 2 +- sound/soc/sprd/sprd-pcm-compress.c | 4 +- sound/soc/sprd/sprd-pcm-dma.c | 2 +- sound/soc/stm/stm32_adfsdm.c | 12 +- sound/soc/stm/stm32_i2s.c | 75 +- sound/soc/stm/stm32_sai.c | 26 +- sound/soc/stm/stm32_sai_sub.c | 13 +- sound/soc/stm/stm32_spdifrx.c | 89 +- sound/soc/sunxi/sun4i-spdif.c | 2 +- sound/soc/sunxi/sun8i-codec.c | 3 - sound/soc/tegra/tegra_alc5632.c | 2 +- sound/soc/tegra/tegra_max98090.c | 2 +- sound/soc/tegra/tegra_rt5640.c | 2 +- sound/soc/tegra/tegra_rt5677.c | 2 +- sound/soc/tegra/tegra_sgtl5000.c | 2 +- sound/soc/tegra/tegra_wm8753.c | 2 +- sound/soc/tegra/tegra_wm8903.c | 24 +- sound/soc/tegra/trimslice.c | 2 +- sound/soc/ti/Kconfig | 8 +- sound/soc/ti/Makefile | 2 + sound/soc/ti/ams-delta.c | 4 +- sound/soc/ti/davinci-evm.c | 4 +- sound/soc/ti/davinci-mcasp.c | 13 +- sound/soc/ti/davinci-vcif.c | 4 +- sound/soc/ti/n810.c | 2 +- sound/soc/ti/omap-abe-twl6040.c | 6 +- sound/soc/ti/omap-mcbsp-st.c | 2 +- sound/soc/ti/omap-mcbsp.c | 4 +- sound/soc/ti/omap-mcpdm.c | 2 +- sound/soc/ti/omap3pandora.c | 4 +- sound/soc/ti/osk5912.c | 2 +- sound/soc/ti/rx51.c | 2 +- sound/soc/ti/udma-pcm.c | 43 + sound/soc/ti/udma-pcm.h | 18 + sound/soc/txx9/txx9aclc.c | 2 +- sound/soc/uniphier/aio-compress.c | 22 +- sound/soc/uniphier/aio-dma.c | 6 +- sound/soc/ux500/mop500_ab8500.c | 6 +- sound/soc/ux500/ux500_pcm.c | 8 +- sound/soc/xtensa/xtfpga-i2s.c | 2 +- sound/soc/zte/zx-spdif.c | 1 - sound/soc/zte/zx-tdm.c | 3 +- sound/usb/usx2y/usbusx2yaudio.c | 9 +- 379 files changed, 17570 insertions(+), 3797 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/amlogic,aiu.yaml create mode 100644 Documentation/devicetree/bindings/sound/amlogic,g12a-toacodec.yaml create mode 100644 Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml create mode 100644 Documentation/devicetree/bindings/sound/amlogic,t9015.yaml create mode 100644 Documentation/devicetree/bindings/sound/brcm,bcm63xx-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/cirrus,cs42l51.yaml delete mode 100644 Documentation/devicetree/bindings/sound/cs42l51.txt delete mode 100644 Documentation/devicetree/bindings/sound/google,cros-ec-codec.txt create mode 100644 Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml create mode 100644 Documentation/devicetree/bindings/sound/ingenic,aic.yaml delete mode 100644 Documentation/devicetree/bindings/sound/ingenic,jz4740-i2s.txt delete mode 100644 Documentation/devicetree/bindings/sound/rockchip-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip-i2s.yaml delete mode 100644 Documentation/devicetree/bindings/sound/st,stm32-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml delete mode 100644 Documentation/devicetree/bindings/sound/st,stm32-spdifrx.txt create mode 100644 Documentation/devicetree/bindings/sound/st,stm32-spdifrx.yaml create mode 100644 Documentation/devicetree/bindings/sound/tlv320adcx140.yaml create mode 100644 include/dt-bindings/sound/meson-aiu.h create mode 100644 include/dt-bindings/sound/meson-g12a-toacodec.h create mode 100644 sound/soc/amd/acp3x-rt5682-max9836.c create mode 100644 sound/soc/bcm/bcm63xx-i2s-whistler.c create mode 100644 sound/soc/bcm/bcm63xx-i2s.h create mode 100644 sound/soc/bcm/bcm63xx-pcm-whistler.c create mode 100644 sound/soc/codecs/rt5682-sdw.c create mode 100644 sound/soc/codecs/rt5682-sdw.h create mode 100644 sound/soc/codecs/tlv320adcx140.c create mode 100644 sound/soc/codecs/tlv320adcx140.h create mode 100644 sound/soc/intel/boards/sof_maxim_common.c create mode 100644 sound/soc/intel/boards/sof_maxim_common.h create mode 100644 sound/soc/intel/boards/sof_pcm512x.c create mode 100644 sound/soc/intel/boards/sof_sdw.c create mode 100644 sound/soc/intel/boards/sof_sdw_common.h create mode 100644 sound/soc/intel/boards/sof_sdw_dmic.c create mode 100644 sound/soc/intel/boards/sof_sdw_hdmi.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt1308.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt5682.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt700.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt711.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt715.c create mode 100644 sound/soc/meson/aiu-acodec-ctrl.c create mode 100644 sound/soc/meson/aiu-codec-ctrl.c create mode 100644 sound/soc/meson/aiu-encoder-i2s.c create mode 100644 sound/soc/meson/aiu-encoder-spdif.c create mode 100644 sound/soc/meson/aiu-fifo-i2s.c create mode 100644 sound/soc/meson/aiu-fifo-spdif.c create mode 100644 sound/soc/meson/aiu-fifo.c create mode 100644 sound/soc/meson/aiu-fifo.h create mode 100644 sound/soc/meson/aiu.c create mode 100644 sound/soc/meson/aiu.h create mode 100644 sound/soc/meson/g12a-toacodec.c create mode 100644 sound/soc/meson/gx-card.c create mode 100644 sound/soc/meson/meson-card-utils.c create mode 100644 sound/soc/meson/meson-card.h create mode 100644 sound/soc/meson/meson-codec-glue.c create mode 100644 sound/soc/meson/meson-codec-glue.h create mode 100644 sound/soc/meson/t9015.c create mode 100644 sound/soc/sof/compress.c create mode 100644 sound/soc/sof/compress.h create mode 100644 sound/soc/sof/intel/hda-compress.c create mode 100644 sound/soc/sof/probe.c create mode 100644 sound/soc/sof/probe.h create mode 100644 sound/soc/ti/udma-pcm.c create mode 100644 sound/soc/ti/udma-pcm.h

2 1

[PATCH 0/1] ASoC: soc-compress: lock pcm_mutex to resolve lockdep error
by 이경택 30 Mar '20

30 Mar '20

Hi, Mark I resend the patch and add recipient Vinod as your comment. Thank you. I'd like to send a patch to fix a problem using soc-compress with DPCM. soc_compr_open_fe() and soc_compr_free_fe() call snd_soc_runtime_activate() and snd_soc_runtime_deactivate() but don't lock card->pcm_mutex. It can cause lockdep error, because snd_soc_runtime_activate/deactivate() checks whether pcm_mutex is held using lockdep_assert_held(). I'd like to send a patch which adds mutex_lock/unlock() before and after of the snd_soc_runtime_activate/deactivate() call. If there is anything should be changed in my patch or email, please let me know it. I'll really appreciate it. Lee Gyeongtaek Lee (1): ASoC: soc-compress: lock pcm_mutex to resolve lockdep error sound/soc/soc-compress.c | 4 ++++ 1 file changed, 4 insertions(+) base-commit: 76ccd234269bd05debdbc12c96eafe62dd9a6180 -- 2.21.0

1 0

[PATCH 1/1] ASoC: soc-compress: lock pcm_mutex to resolve lockdep error
by 이경택 30 Mar '20

30 Mar '20

snd_soc_runtime_activate() and snd_soc_runtime_deactivate() require pcm_mutex lock. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> --- sound/soc/soc-compress.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 392a1c5b15d3..42d416ac7e9b 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -207,7 +207,9 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) fe->dpcm[stream].state = SND_SOC_DPCM_STATE_OPEN; fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + mutex_lock_nested(&fe->pcm_mutex, fe->pcm_subclass); snd_soc_runtime_activate(fe, stream); + mutex_unlock(&fe->pcm_mutex); mutex_unlock(&fe->card->mutex); @@ -285,7 +287,9 @@ static int soc_compr_free_fe(struct snd_compr_stream *cstream) else stream = SNDRV_PCM_STREAM_CAPTURE; + mutex_lock_nested(&fe->pcm_mutex, fe->pcm_subclass); snd_soc_runtime_deactivate(fe, stream); + mutex_unlock(&fe->pcm_mutex); fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; -- 2.21.0

2 2

[RFC PATCH v1 13/50] Avoid some useless msecs/jiffies conversions
by George Spelvin 30 Mar '20

30 Mar '20

Rather than generating a random number of milliseconds in a constant range and then converting to jiffies, convert the range to jiffies (evaluated at compile time) and choose a random number of jiffies in that range. Likewise, "msecs_to_jiffies(seconds * 1000)" is more conveniently written "seconds * HZ". Signed-off-by: George Spelvin <lkml(a)sdf.org> Cc: Hannes Reinecke <hare(a)suse.de> Cc: linux-scsi(a)vger.kernel.org Cc: Marek Lindner <mareklindner(a)neomailbox.ch> Cc: Simon Wunderlich <sw(a)simonwunderlich.de> Cc: Antonio Quartulli <a(a)unstable.cc> Cc: Sven Eckelmann <sven(a)narfation.org> Cc: b.a.t.m.a.n(a)lists.open-mesh.org Cc: Johannes Berg <johannes(a)sipsolutions.net> Cc: linux-wireless(a)vger.kernel.org Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org --- drivers/scsi/fcoe/fcoe_ctlr.c | 10 +++++----- net/batman-adv/bat_iv_ogm.c | 2 +- net/batman-adv/bat_v_ogm.c | 8 ++++---- net/bluetooth/hci_request.c | 2 +- net/wireless/scan.c | 2 +- sound/core/pcm_lib.c | 2 +- sound/core/pcm_native.c | 2 +- 7 files changed, 14 insertions(+), 14 deletions(-) diff --git a/drivers/scsi/fcoe/fcoe_ctlr.c b/drivers/scsi/fcoe/fcoe_ctlr.c index 1791a393795da..9c530f8827518 100644 --- a/drivers/scsi/fcoe/fcoe_ctlr.c +++ b/drivers/scsi/fcoe/fcoe_ctlr.c @@ -2238,10 +2238,10 @@ static void fcoe_ctlr_vn_restart(struct fcoe_ctlr *fip) if (fip->probe_tries < FIP_VN_RLIM_COUNT) { fip->probe_tries++; - wait = prandom_u32() % FIP_VN_PROBE_WAIT; + wait = prandom_u32_max(msecs_to_jiffies(FIP_VN_PROBE_WAIT)); } else - wait = FIP_VN_RLIM_INT; - mod_timer(&fip->timer, jiffies + msecs_to_jiffies(wait)); + wait = msecs_to_jiffies(FIP_VN_RLIM_INT); + mod_timer(&fip->timer, jiffies + wait); fcoe_ctlr_set_state(fip, FIP_ST_VNMP_START); } @@ -3132,8 +3132,8 @@ static void fcoe_ctlr_vn_timeout(struct fcoe_ctlr *fip) fcoe_ctlr_vn_send(fip, FIP_SC_VN_BEACON, fcoe_all_vn2vn, 0); fip->port_ka_time = jiffies + - msecs_to_jiffies(FIP_VN_BEACON_INT + - (prandom_u32() % FIP_VN_BEACON_FUZZ)); + msecs_to_jiffies(FIP_VN_BEACON_INT) + + prandom_u32_max(msecs_to_jiffies(FIP_VN_BEACON_FUZZ)); } if (time_before(fip->port_ka_time, next_time)) next_time = fip->port_ka_time; diff --git a/net/batman-adv/bat_iv_ogm.c b/net/batman-adv/bat_iv_ogm.c index 46c5cd9f8019e..9efd96e91d53e 100644 --- a/net/batman-adv/bat_iv_ogm.c +++ b/net/batman-adv/bat_iv_ogm.c @@ -288,7 +288,7 @@ batadv_iv_ogm_emit_send_time(const struct batadv_priv *bat_priv) /* when do we schedule a ogm packet to be sent */ static unsigned long batadv_iv_ogm_fwd_send_time(void) { - return jiffies + msecs_to_jiffies(prandom_u32() % (BATADV_JITTER / 2)); + return jiffies + prandom_u32_max(msecs_to_jiffies(BATADV_JITTER / 2)); } /* apply hop penalty for a normal link */ diff --git a/net/batman-adv/bat_v_ogm.c b/net/batman-adv/bat_v_ogm.c index 411ce5fc6492f..61fa742ff5501 100644 --- a/net/batman-adv/bat_v_ogm.c +++ b/net/batman-adv/bat_v_ogm.c @@ -85,12 +85,12 @@ struct batadv_orig_node *batadv_v_ogm_orig_get(struct batadv_priv *bat_priv, */ static void batadv_v_ogm_start_queue_timer(struct batadv_hard_iface *hard_iface) { - unsigned int msecs = BATADV_MAX_AGGREGATION_MS * 1000; + unsigned int delay = msecs_to_jiffies(BATADV_MAX_AGGREGATION_MS); - /* msecs * [0.9, 1.1] */ - msecs += prandom_u32() % (msecs / 5) - (msecs / 10); + /* delay * [0.9, 1.1] */ + delay += prandom_u32_max(delay / 5) - (delay / 10); queue_delayed_work(batadv_event_workqueue, &hard_iface->bat_v.aggr_wq, - msecs_to_jiffies(msecs / 1000)); + delay); } /** diff --git a/net/bluetooth/hci_request.c b/net/bluetooth/hci_request.c index 2a1b64dbf76e4..8b46e23b4abe7 100644 --- a/net/bluetooth/hci_request.c +++ b/net/bluetooth/hci_request.c @@ -1505,7 +1505,7 @@ int hci_get_random_address(struct hci_dev *hdev, bool require_privacy, bacpy(rand_addr, &hdev->rpa); - to = msecs_to_jiffies(hdev->rpa_timeout * 1000); + to = hdev->rpa_timeout * HZ; if (adv_instance) queue_delayed_work(hdev->workqueue, &adv_instance->rpa_expired_cb, to); diff --git a/net/wireless/scan.c b/net/wireless/scan.c index aef240fdf8df6..b6856cbb68d3b 100644 --- a/net/wireless/scan.c +++ b/net/wireless/scan.c @@ -700,7 +700,7 @@ void cfg80211_bss_age(struct cfg80211_registered_device *rdev, unsigned long age_secs) { struct cfg80211_internal_bss *bss; - unsigned long age_jiffies = msecs_to_jiffies(age_secs * MSEC_PER_SEC); + unsigned long age_jiffies = age_secs * HZ; spin_lock_bh(&rdev->bss_lock); list_for_each_entry(bss, &rdev->bss_list, list) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 2236b5e0c1f25..8a2bf333200c1 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1839,7 +1839,7 @@ static int wait_for_avail(struct snd_pcm_substream *substream, runtime->rate; wait_time = max(t, wait_time); } - wait_time = msecs_to_jiffies(wait_time * 1000); + wait_time *= HZ; } } diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index df40d38f6e290..1ea763f9f956d 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1937,7 +1937,7 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, long t = runtime->period_size * 2 / runtime->rate; tout = max(t, tout); } - tout = msecs_to_jiffies(tout * 1000); + tout *= HZ; } tout = schedule_timeout(tout); -- 2.26.0

3 9

[PATCH 1/1] ASoC: dpcm: allow start or stop during pause for backend
by 이경택 30 Mar '20

30 Mar '20

soc_compr_trigger_fe() allows start or stop after pause_push. In dpcm_be_dai_trigger(), however, only pause_release is allowed command after pause_push. To fix this problem, SND_SOC_DPCM_STATE_PAUSED should be allowed also for start or stop command. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> --- sound/soc/soc-pcm.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 2c59b3688ca0..8f6f0ad50288 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2236,7 +2236,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, switch (cmd) { case SNDRV_PCM_TRIGGER_START: if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && - (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); @@ -2266,7 +2267,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; case SNDRV_PCM_TRIGGER_STOP: - if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) -- 2.21.0

1 0

[PATCH 0/1] ASoC: dpcm: allow started or stop during pause for backend
by 이경택 30 Mar '20

30 Mar '20

Hi, When compress offload is used with DPCM, error can be occurred on start or stop after pause. soc_compr_trigger_fe() allows start or stop after pause_push. In dpcm_be_dai_trigger(), however, only pause_release is allowed command after pause_push. Any feedback will be greatly appreciated. Gyeongtaek Lee (1): ASoC: dpcm: allow start or stop during pause for backend sound/soc/soc-pcm.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) base-commit: 76ccd234269bd05debdbc12c96eafe62dd9a6180 -- 2.21.0

1 0

[PATCH 0/4] ASoC: remove rtd->cpu/codec_dai{s} v2
by Kuninori Morimoto 30 Mar '20

30 Mar '20

Hi Mark These are rebased "remove rtd->cpu/codec_dai" patches. Kuninori Morimoto (4): ASoC: soc: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: soc-core: set rtd->num_cpu/codec at soc_new_pcm_runtime() ASoC: soc-core: tidyup soc_new_pcm_runtime() rtd setups ASoC: soc-core: remove cpu_dai/codec_dai/cpu_dais/codec_dais include/sound/soc.h | 26 +++++++++-------- sound/soc/soc-compress.c | 36 +++++++++++------------ sound/soc/soc-core.c | 42 ++++++++++----------------- sound/soc/soc-dapm.c | 4 +-- sound/soc/soc-generic-dmaengine-pcm.c | 6 ++-- sound/soc/soc-pcm.c | 30 +++++++++---------- 6 files changed, 68 insertions(+), 76 deletions(-) -- 2.17.1

1 4

[PATCH v2] fix some output pcms being namehint-ed as input pcms
by sylvain.bertrand＠gmail.com 29 Mar '20

29 Mar '20

This is additional cleanup following: commit d434638a683c01bd6decf8f41863044055237a50 namehint: remember the direction from the upper levels signed-off-by: Sylvain BERTRAND <sylvain.bertrand(a)legeek.net> --- based on arecord -L output --- a/share/alsa/pcm/front.conf +++ b/share/alsa/pcm/front.conf @@ -53,6 +53,7 @@ name defaults.namehint.basic } description "Front speakers" - device $DEV + device_output $DEV + omit_noargs true } } --- a/share/alsa/pcm/iec958.conf +++ b/share/alsa/pcm/iec958.conf @@ -78,6 +78,6 @@ name defaults.namehint.basic } description "IEC958 (S/PDIF) Digital Audio Output" - device $DEV + device_output $DEV } }

1 0

[PATCH] ASoC: bcm2835-i2s: substream alignment now independent in hwparams
by Matt Flax 29 Mar '20

29 Mar '20

Substream sample alignment was being set in hwparams for both substreams at the same time. This became a problem when the Audio Injector isolated sound card needed to offset sample alignment for high sample rates. The latency difference between playback and capture occurs because of the digital isolation chip propagation time, particularly when the codec is master and the DAC return is twice delayed. This patch sets sample alignment registers based on the substream direction in hwparams. This gives the machine driver more control over sample alignment in the bcm2835 i2s driver. Signed-off-by: Matt Flax <flatmax(a)flatmax.org> --- sound/soc/bcm/bcm2835-i2s.c | 36 +++++++++++++++++++----------------- 1 file changed, 19 insertions(+), 17 deletions(-) diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c index e6a12e271b07..9db542699a13 100644 --- a/sound/soc/bcm/bcm2835-i2s.c +++ b/sound/soc/bcm/bcm2835-i2s.c @@ -493,11 +493,6 @@ static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - bcm2835_i2s_calc_channel_pos(&rx_ch1_pos, &rx_ch2_pos, - rx_mask, slot_width, data_delay, odd_slot_offset); - bcm2835_i2s_calc_channel_pos(&tx_ch1_pos, &tx_ch2_pos, - tx_mask, slot_width, data_delay, odd_slot_offset); - /* * Transmitting data immediately after frame start, eg * in left-justified or DSP mode A, only works stable @@ -508,19 +503,26 @@ static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream, "Unstable slave config detected, L/R may be swapped"); /* - * Set format for both streams. - * We cannot set another frame length - * (and therefore word length) anyway, - * so the format will be the same. + * Set format on a per stream basis. + * The alignment format can be different depending on direction. */ - regmap_write(dev->i2s_regmap, BCM2835_I2S_RXC_A_REG, - format - | BCM2835_I2S_CH1_POS(rx_ch1_pos) - | BCM2835_I2S_CH2_POS(rx_ch2_pos)); - regmap_write(dev->i2s_regmap, BCM2835_I2S_TXC_A_REG, - format - | BCM2835_I2S_CH1_POS(tx_ch1_pos) - | BCM2835_I2S_CH2_POS(tx_ch2_pos)); + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + bcm2835_i2s_calc_channel_pos(&rx_ch1_pos, &rx_ch2_pos, + rx_mask, slot_width, data_delay, odd_slot_offset); + regmap_write(dev->i2s_regmap, BCM2835_I2S_RXC_A_REG, + format + | BCM2835_I2S_CH1_POS(rx_ch1_pos) + | BCM2835_I2S_CH2_POS(rx_ch2_pos)); + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + bcm2835_i2s_calc_channel_pos(&tx_ch1_pos, &tx_ch2_pos, + tx_mask, slot_width, data_delay, odd_slot_offset); + regmap_write(dev->i2s_regmap, BCM2835_I2S_TXC_A_REG, + format + | BCM2835_I2S_CH1_POS(tx_ch1_pos) + | BCM2835_I2S_CH2_POS(tx_ch2_pos)); + } /* Setup the I2S mode */ -- 2.20.1

3 8

[PATCH v2] ALSA: hda/realtek - a fake key event is triggered by running shutup
by Hui Wang 29 Mar '20

29 Mar '20

On the Lenovo X1C7 machines, after we plug the headset, the rt_resume() and rt_suspend() of the codec driver will be called periodically, the driver can't stay in the rt_suspend state even users doen't use the sound card. Through debugging, I found when running rt_suspend(), it will call alc225_shutup(), in this function, it will change 3k pull down control by alc_update_coef_idx(codec, 0x4a, 0, 3 << 10), this will trigger a fake key event and that event will resume the codec, when codec suspend agin, it will trigger the fake key event one more time, this process will repeat. If disable the key event before changing the pull down control, it will not trigger fake key event. It also needs to restore the pull down control and re-enable the key event, otherwise the system can't get key event when codec is in rt_suspend state. Also move some functions ahead of alc225_shutup(), this can save the function declaration. Fixes: 76f7dec08fd6 (ALSA: hda/realtek - Add Headset Button supported for ThinkPad X1) Cc: Kailang Yang <kailang(a)realtek.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 170 +++++++++++++++++++++------------- 1 file changed, 107 insertions(+), 63 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1ad8c2e2d1af..0af33f00617a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -107,6 +107,7 @@ struct alc_spec { unsigned int done_hp_init:1; unsigned int no_shutup_pins:1; unsigned int ultra_low_power:1; + unsigned int has_hs_key:1; /* for PLL fix */ hda_nid_t pll_nid; @@ -2982,6 +2983,107 @@ static int alc269_parse_auto_config(struct hda_codec *codec) return alc_parse_auto_config(codec, alc269_ignore, ssids); } +static const struct hda_jack_keymap alc_headset_btn_keymap[] = { + { SND_JACK_BTN_0, KEY_PLAYPAUSE }, + { SND_JACK_BTN_1, KEY_VOICECOMMAND }, + { SND_JACK_BTN_2, KEY_VOLUMEUP }, + { SND_JACK_BTN_3, KEY_VOLUMEDOWN }, + {} +}; + +static void alc_headset_btn_callback(struct hda_codec *codec, + struct hda_jack_callback *jack) +{ + int report = 0; + + if (jack->unsol_res & (7 << 13)) + report |= SND_JACK_BTN_0; + + if (jack->unsol_res & (1 << 16 | 3 << 8)) + report |= SND_JACK_BTN_1; + + /* Volume up key */ + if (jack->unsol_res & (7 << 23)) + report |= SND_JACK_BTN_2; + + /* Volume down key */ + if (jack->unsol_res & (7 << 10)) + report |= SND_JACK_BTN_3; + + jack->jack->button_state = report; +} + +static void alc_disable_headset_jack_key(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + if (!spec->has_hs_key) + return; + + switch (codec->core.vendor_id) { + case 0x10ec0215: + case 0x10ec0225: + case 0x10ec0285: + case 0x10ec0295: + case 0x10ec0289: + case 0x10ec0299: + alc_write_coef_idx(codec, 0x48, 0x0); + alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); + alc_update_coef_idx(codec, 0x44, 0x0045 << 8, 0x0); + break; + case 0x10ec0236: + case 0x10ec0256: + alc_write_coef_idx(codec, 0x48, 0x0); + alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); + break; + } +} + +static void alc_enable_headset_jack_key(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + if (!spec->has_hs_key) + return; + + switch (codec->core.vendor_id) { + case 0x10ec0215: + case 0x10ec0225: + case 0x10ec0285: + case 0x10ec0295: + case 0x10ec0289: + case 0x10ec0299: + alc_write_coef_idx(codec, 0x48, 0xd011); + alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); + alc_update_coef_idx(codec, 0x44, 0x007f << 8, 0x0045 << 8); + break; + case 0x10ec0236: + case 0x10ec0256: + alc_write_coef_idx(codec, 0x48, 0xd011); + alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); + break; + } +} + +static void alc_fixup_headset_jack(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + spec->has_hs_key = 1; + snd_hda_jack_detect_enable_callback(codec, 0x55, + alc_headset_btn_callback); + snd_hda_jack_add_kctl(codec, 0x55, "Headset Jack", false, + SND_JACK_HEADSET, alc_headset_btn_keymap); + break; + case HDA_FIXUP_ACT_INIT: + alc_enable_headset_jack_key(codec); + break; + } +} + static void alc269vb_toggle_power_output(struct hda_codec *codec, int power_up) { alc_update_coef_idx(codec, 0x04, 1 << 11, power_up ? (1 << 11) : 0); @@ -3372,6 +3474,8 @@ static void alc225_shutup(struct hda_codec *codec) if (!hp_pin) hp_pin = 0x21; + + alc_disable_headset_jack_key(codec); /* 3k pull low control for Headset jack. */ alc_update_coef_idx(codec, 0x4a, 0, 3 << 10); @@ -3411,6 +3515,9 @@ static void alc225_shutup(struct hda_codec *codec) alc_update_coef_idx(codec, 0x4a, 3<<4, 2<<4); msleep(30); } + + alc_update_coef_idx(codec, 0x4a, 3 << 10, 0); + alc_enable_headset_jack_key(codec); } static void alc_default_init(struct hda_codec *codec) @@ -5668,69 +5775,6 @@ static void alc285_fixup_invalidate_dacs(struct hda_codec *codec, snd_hda_override_wcaps(codec, 0x03, 0); } -static const struct hda_jack_keymap alc_headset_btn_keymap[] = { - { SND_JACK_BTN_0, KEY_PLAYPAUSE }, - { SND_JACK_BTN_1, KEY_VOICECOMMAND }, - { SND_JACK_BTN_2, KEY_VOLUMEUP }, - { SND_JACK_BTN_3, KEY_VOLUMEDOWN }, - {} -}; - -static void alc_headset_btn_callback(struct hda_codec *codec, - struct hda_jack_callback *jack) -{ - int report = 0; - - if (jack->unsol_res & (7 << 13)) - report |= SND_JACK_BTN_0; - - if (jack->unsol_res & (1 << 16 | 3 << 8)) - report |= SND_JACK_BTN_1; - - /* Volume up key */ - if (jack->unsol_res & (7 << 23)) - report |= SND_JACK_BTN_2; - - /* Volume down key */ - if (jack->unsol_res & (7 << 10)) - report |= SND_JACK_BTN_3; - - jack->jack->button_state = report; -} - -static void alc_fixup_headset_jack(struct hda_codec *codec, - const struct hda_fixup *fix, int action) -{ - - switch (action) { - case HDA_FIXUP_ACT_PRE_PROBE: - snd_hda_jack_detect_enable_callback(codec, 0x55, - alc_headset_btn_callback); - snd_hda_jack_add_kctl(codec, 0x55, "Headset Jack", false, - SND_JACK_HEADSET, alc_headset_btn_keymap); - break; - case HDA_FIXUP_ACT_INIT: - switch (codec->core.vendor_id) { - case 0x10ec0215: - case 0x10ec0225: - case 0x10ec0285: - case 0x10ec0295: - case 0x10ec0289: - case 0x10ec0299: - alc_write_coef_idx(codec, 0x48, 0xd011); - alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); - alc_update_coef_idx(codec, 0x44, 0x007f << 8, 0x0045 << 8); - break; - case 0x10ec0236: - case 0x10ec0256: - alc_write_coef_idx(codec, 0x48, 0xd011); - alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); - break; - } - break; - } -} - static void alc295_fixup_chromebook(struct hda_codec *codec, const struct hda_fixup *fix, int action) { -- 2.17.1

2 1

about alc880 no sound and eapd coef
by 尤晓杰 29 Mar '20

29 Mar '20

dear friend, I am using ecs g420 laptop, use options snd-hda-intel model=generic the speaker and mic correctly, 2 pop sounds,when the kernel load(left channel) and the x11 load(right channel), but with standard way only one pop sound when kernel load and silent always. the mic is ok. I have got datasheet of alc880 and see the eapd function, but in your code I see this need coef parameter. so what should I write using verb to open eapd? the attachment is the schematic of sound part.in summary: the 0x14 hp out, 0x18 mic in,0x1c cd in, 0x1e spif out, no jack sense. eapd shared with spdif in. all are correct with default of datasheet's describe. I use verb to dectt each pin widget's capability , take 0x14 as exaple, no eapd capability.To set 0x14's eapd with hda-verb /dev/snd/hwC0D0 0x70c 0x2, failed to read it again alwas 0. and bios also reported 0x16, I think it is bios's bug. I what to use "Early Patching" method to deal with it without changing source code, is it feasible? could you give me some idea? thanks!!!! dmesg: [ 8.009944] snd_hda_codec_realtek hdaudioC0D0: autoconfig for ALC880: line_outs=2 (0x14/0x16/0x0/0x0/0x0) type:hp [ 8.014023] snd_hda_codec_realtek hdaudioC0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 8.017999] snd_hda_codec_realtek hdaudioC0D0: hp_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 8.021625] snd_hda_codec_realtek hdaudioC0D0: mono: mono_out=0x0 [ 8.024255] snd_hda_codec_realtek hdaudioC0D0: dig-out=0x1e/0x0 [ 8.026741] snd_hda_codec_realtek hdaudioC0D0: inputs: [ 8.038468] mc: Linux media interface: v0.10 [ 8.067515] snd_hda_codec_realtek hdaudioC0D0: Mic=0x18 [ 8.078984] snd_hda_codec_realtek hdaudioC0D0: CD=0x1c [ 8.290306] input: HDA Digital PCBeep as /devices/pci0000:00/0000:00:1b.0/sound/card0/input13

2 1

Port HDA Analyzer to python 3
by andythe_great 29 Mar '20

29 Mar '20

Hello, Is it possible to port HDA Analyzer to python 3? https://www.alsa-project.org/main/index.php/HDA_Analyzer Because openSUSE is retiring python 2. I tried using python-modernize https://python-modernize.readthedocs.io/en/latest/ but many could not get it to run. Thanks. Andy

2 1

[PATCH] ALSA: hda: default enable CA0132 DSP support
by Rouven Czerwinski 29 Mar '20

29 Mar '20

From: Rouven Czerwinski <rouven(a)czerwinskis.de> If SND_HDA_CODEC_CA0132 is enabled, the DSP support should be enabled as well. Disabled DSP support leads to a hanging alsa system and no sound output on the card otherwise. Tested on: 06:00.0 Audio device: Creative Labs Sound Core3D [Sound Blaster Recon3D / Z-Series] (rev 01) Signed-off-by: Rouven Czerwinski <rouven(a)czerwinskis.de> --- sound/pci/hda/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index bd48335d09d7..e1d3082a4fe9 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -184,6 +184,7 @@ comment "Set to Y if you want auto-loading the codec driver" config SND_HDA_CODEC_CA0132_DSP bool "Support new DSP code for CA0132 codec" depends on SND_HDA_CODEC_CA0132 + default y select SND_HDA_DSP_LOADER select FW_LOADER help base-commit: e595dd94515ed6bc5ba38fce0f9598db8c0ee9a9 -- 2.25.1

2 1

[alsa-devel] [PATCH 0/3] ALSA: bebob: misc improvements for starting/stopping packet streaming
by Takashi Sakamoto 27 Mar '20

27 Mar '20

Hi, This patchset improves model-specific issues in a point of protocol version supported by the model. The changes are relevant to packet streaming, thus it can brings some regressions. For testers, I prepare for a remote branch to backport the latest codes to Linux v4.17 or later. I'm happy to receive any report. https://github.com/takaswie/snd-firewire-improve/tree/topic/bebob-connectio… Takashi Sakamoto (3): ALSA: bebob: expand sleep just after breaking connections for protocol version 1 ALSA: bebob: expand delay of start for IR context just for version 3 firmware ALSA: bebob: link the order of establishing connections and Syt-match clock mode sound/firewire/bebob/bebob_stream.c | 76 ++++++++++++++--------------- 1 file changed, 38 insertions(+), 38 deletions(-) -- 2.20.1

3 7

[PATCH] ASoC: topology: Add missing memory checks
by Amadeusz Sławiński 27 Mar '20

27 Mar '20

kstrdup is an allocation function and it can fail, so its return value should be checked and handled appropriately. In order to check all cases, we need to modify set_stream_info to return a value, so check that everything went correctly when doing kstrdup(). Later add proper checks and error handlers. Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> --- sound/soc/soc-topology.c | 65 +++++++++++++++++++++++++++++++--------- 1 file changed, 51 insertions(+), 14 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 575da6aba807..0bec3ff782c1 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1766,10 +1766,13 @@ static int soc_tplg_dapm_complete(struct soc_tplg *tplg) return 0; } -static void set_stream_info(struct snd_soc_pcm_stream *stream, +static int set_stream_info(struct snd_soc_pcm_stream *stream, struct snd_soc_tplg_stream_caps *caps) { stream->stream_name = kstrdup(caps->name, GFP_KERNEL); + if (!stream->stream_name) + return -ENOMEM; + stream->channels_min = le32_to_cpu(caps->channels_min); stream->channels_max = le32_to_cpu(caps->channels_max); stream->rates = le32_to_cpu(caps->rates); @@ -1777,6 +1780,8 @@ static void set_stream_info(struct snd_soc_pcm_stream *stream, stream->rate_max = le32_to_cpu(caps->rate_max); stream->formats = le64_to_cpu(caps->formats); stream->sig_bits = le32_to_cpu(caps->sig_bits); + + return 0; } static void set_dai_flags(struct snd_soc_dai_driver *dai_drv, @@ -1812,20 +1817,29 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg, if (dai_drv == NULL) return -ENOMEM; - if (strlen(pcm->dai_name)) + if (strlen(pcm->dai_name)) { dai_drv->name = kstrdup(pcm->dai_name, GFP_KERNEL); + if (!dai_drv->name) { + ret = -ENOMEM; + goto err; + } + } dai_drv->id = le32_to_cpu(pcm->dai_id); if (pcm->playback) { stream = &dai_drv->playback; caps = &pcm->caps[SND_SOC_TPLG_STREAM_PLAYBACK]; - set_stream_info(stream, caps); + ret = set_stream_info(stream, caps); + if (ret < 0) + goto err; } if (pcm->capture) { stream = &dai_drv->capture; caps = &pcm->caps[SND_SOC_TPLG_STREAM_CAPTURE]; - set_stream_info(stream, caps); + ret = set_stream_info(stream, caps); + if (ret < 0) + goto err; } if (pcm->compress) @@ -1835,11 +1849,7 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg, ret = soc_tplg_dai_load(tplg, dai_drv, pcm, NULL); if (ret < 0) { dev_err(tplg->comp->dev, "ASoC: DAI loading failed\n"); - kfree(dai_drv->playback.stream_name); - kfree(dai_drv->capture.stream_name); - kfree(dai_drv->name); - kfree(dai_drv); - return ret; + goto err; } dai_drv->dobj.index = tplg->index; @@ -1857,9 +1867,17 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg, if (ret != 0) { dev_err(dai->dev, "Failed to create DAI widgets %d\n", ret); snd_soc_unregister_dai(dai); - return ret; + goto err; } + return 0; + +err: + kfree(dai_drv->playback.stream_name); + kfree(dai_drv->capture.stream_name); + kfree(dai_drv->name); + kfree(dai_drv); + return ret; } @@ -1916,11 +1934,20 @@ static int soc_tplg_fe_link_create(struct soc_tplg *tplg, if (strlen(pcm->pcm_name)) { link->name = kstrdup(pcm->pcm_name, GFP_KERNEL); link->stream_name = kstrdup(pcm->pcm_name, GFP_KERNEL); + if (!link->name || !link->stream_name) { + ret = -ENOMEM; + goto err; + } } link->id = le32_to_cpu(pcm->pcm_id); - if (strlen(pcm->dai_name)) + if (strlen(pcm->dai_name)) { link->cpus->dai_name = kstrdup(pcm->dai_name, GFP_KERNEL); + if (!link->cpus->dai_name) { + ret = -ENOMEM; + goto err; + } + } link->codecs->name = "snd-soc-dummy"; link->codecs->dai_name = "snd-soc-dummy-dai"; @@ -2436,13 +2463,17 @@ static int soc_tplg_dai_config(struct soc_tplg *tplg, if (d->playback) { stream = &dai_drv->playback; caps = &d->caps[SND_SOC_TPLG_STREAM_PLAYBACK]; - set_stream_info(stream, caps); + ret = set_stream_info(stream, caps); + if (ret < 0) + goto err; } if (d->capture) { stream = &dai_drv->capture; caps = &d->caps[SND_SOC_TPLG_STREAM_CAPTURE]; - set_stream_info(stream, caps); + ret = set_stream_info(stream, caps); + if (ret < 0) + goto err; } if (d->flag_mask) @@ -2454,10 +2485,16 @@ static int soc_tplg_dai_config(struct soc_tplg *tplg, ret = soc_tplg_dai_load(tplg, dai_drv, NULL, dai); if (ret < 0) { dev_err(tplg->comp->dev, "ASoC: DAI loading failed\n"); - return ret; + goto err; } return 0; + +err: + kfree(dai_drv->playback.stream_name); + kfree(dai_drv->capture.stream_name); + + return ret; } /* load physical DAI elements */ -- 2.17.1

2 2

[PATCH 0/4] ASoC: Intel: add SoundWire machine driver
by Pierre-Louis Bossart 27 Mar '20

27 Mar '20

To handle multiple hardware combinations, this patchset suggests a single machine driver which will create and initialize dailinks dynamically. This allows us to support new configurations easily, as shown with the TigerLake rt5682 example. Each configuration updates the card component string, and UCM can test for the presence of components to configure them as needed. Since we use a single the machine driver name, all previous ACPI tables need to be updated. That should have no impact since the machine drivers listed at the time were not upstreamed and are no longer maintained. Naveen Manohar (2): ASoC: Intel: common: add match table for TGL RT5682 SoundWire driver ASoC: Intel: sof_sdw: Add Volteer support with RT5682 SNDW helper function Pierre-Louis Bossart (1): ASoC: Intel: boards: add sof_sdw machine driver Rander Wang (1): ASoC: Intel: soc-acpi: update topology and driver name for SoundWire platforms sound/soc/intel/boards/Kconfig | 24 + sound/soc/intel/boards/Makefile | 8 +- sound/soc/intel/boards/sof_sdw.c | 962 ++++++++++++++++++ sound/soc/intel/boards/sof_sdw_common.h | 114 +++ sound/soc/intel/boards/sof_sdw_dmic.c | 42 + sound/soc/intel/boards/sof_sdw_hdmi.c | 97 ++ sound/soc/intel/boards/sof_sdw_rt1308.c | 151 +++ sound/soc/intel/boards/sof_sdw_rt5682.c | 126 +++ sound/soc/intel/boards/sof_sdw_rt700.c | 125 +++ sound/soc/intel/boards/sof_sdw_rt711.c | 156 +++ sound/soc/intel/boards/sof_sdw_rt715.c | 42 + .../intel/common/soc-acpi-intel-cml-match.c | 24 +- .../intel/common/soc-acpi-intel-icl-match.c | 6 +- .../intel/common/soc-acpi-intel-tgl-match.c | 30 +- 14 files changed, 1896 insertions(+), 11 deletions(-) create mode 100644 sound/soc/intel/boards/sof_sdw.c create mode 100644 sound/soc/intel/boards/sof_sdw_common.h create mode 100644 sound/soc/intel/boards/sof_sdw_dmic.c create mode 100644 sound/soc/intel/boards/sof_sdw_hdmi.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt1308.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt5682.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt700.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt711.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt715.c -- 2.20.1

2 11

[PATCH] dt-bindings: sound: tlv320adcx140: Remove undocumented property
by Dan Murphy 27 Mar '20

27 Mar '20

Remove undocumented and unneeded ti,use-internal-reg from the example as it was an artifact from initial development. The code does not query for this property and as the document indicates if areg-supply is undefined then the internal regulator is used. Fixes: 302c0b7490cd ("dt-bindings: sound: Add TLV320ADCx140 dt bindings") CC: Rob Herring <robh(a)kernel.org> Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- Documentation/devicetree/bindings/sound/tlv320adcx140.yaml | 1 - 1 file changed, 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml index 1433ff62b14f..ab2268c0ee67 100644 --- a/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml +++ b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml @@ -76,7 +76,6 @@ examples: codec: codec@4c { compatible = "ti,tlv320adc5140"; reg = <0x4c>; - ti,use-internal-areg; ti,mic-bias-source = <6>; reset-gpios = <&gpio0 14 GPIO_ACTIVE_HIGH>; }; -- 2.25.1

2 1

[PATCH] ASoC: amd: acp3x-pcm-dma: clean up two indentation issues
by Colin King 27 Mar '20

27 Mar '20

From: Colin Ian King <colin.king(a)canonical.com> There are a couple of statements that are not indented correctly, add in the missing tab and break the lines to address a checkpatch warning. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index d62c0d90c41e..e362f0bc9e46 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -458,7 +458,8 @@ static int acp3x_resume(struct device *dev) reg_val = mmACP_I2STDM_ITER; frmt_val = mmACP_I2STDM_TXFRMT; } - rv_writel((rtd->xfer_resolution << 3), rtd->acp3x_base + reg_val); + rv_writel((rtd->xfer_resolution << 3), + rtd->acp3x_base + reg_val); } if (adata->capture_stream && adata->capture_stream->runtime) { struct i2s_stream_instance *rtd = @@ -474,7 +475,8 @@ static int acp3x_resume(struct device *dev) reg_val = mmACP_I2STDM_IRER; frmt_val = mmACP_I2STDM_RXFRMT; } - rv_writel((rtd->xfer_resolution << 3), rtd->acp3x_base + reg_val); + rv_writel((rtd->xfer_resolution << 3), + rtd->acp3x_base + reg_val); } if (adata->tdm_mode == TDM_ENABLE) { rv_writel(adata->tdm_fmt, adata->acp3x_base + frmt_val); -- 2.25.1

2 1

[PATCH] additional cleanup for "arecord -L show entries which are not capture pcms"
by sylvain.bertrand＠gmail.com 27 Mar '20

27 Mar '20

I did post to github, my bad. On Fri, Mar 27, 2020 at 04:01:57AM -0700, Jaroslav Kysela wrote: > Closed #39 via d434638a683c01bd6decf8f41863044055237a50. Tested with the mic from my usb webcam, and it actually needs a little bit more tidying. -- Sylvain --- a/share/alsa/pcm/front.conf +++ b/share/alsa/pcm/front.conf @@ -53,6 +53,7 @@ name defaults.namehint.basic } description "Front speakers" - device $DEV + device_output $DEV + omit_noargs true } } --- a/share/alsa/pcm/iec958.conf +++ b/share/alsa/pcm/iec958.conf @@ -78,6 +78,6 @@ name defaults.namehint.basic } description "IEC958 (S/PDIF) Digital Audio Output" - device $DEV + device_output $DEV } }

1 0

route and chmap
by Jonas Holmberg 27 Mar '20

27 Mar '20

Hi, I use the route plugin to copy mono to both FL and FR (front left and front right stereo) of the slave during playback. When get_chmap() is called in the route plugin it will call get_chmap() of the slave, which returns 2 channels with FL and FR positions, and then route itself returns 1 channel with FL position. But the application that called snd_pcm_get_chmap() on the playback device expected to get position MONO instead of FL since the number of channels is 1. Is the application wrong in not accepting the chmap or should the route plugin have changed position from FL to MONO? Is it possible to add something to the config (asound.conf) to make FL become MONO in the reply of get_chmap() or solve the problem in any other way? BR /Jonas

2 2

[PATCH 00/36] ASoC: remove rtd->cpu/codec_dai{s}
by Kuninori Morimoto 27 Mar '20

27 Mar '20

Hi Mark Now, CPU/Codec DAI(s) were replaced by rtd->dais. Thus, We don't need rtd->cpu/codec_dai{s} anymore. This pathset replaces it by new macro. Kuninori Morimoto (36): ASoC: soc-core: add asoc_rtd_to_cpu/codec() macro ASoC: amd: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: atmel: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: au1x: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: bcm: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: cirrus: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: dwc: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: fsl: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: generic: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: img: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: intel: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: kirkwood: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: mediatek: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: meson: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: mxs: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: pxa: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: qcom: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: rockchip: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: samsung: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: sh: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: sof: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: sprd: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: stm: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: sunxi: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: tegra: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: ti: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: txx9: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: uniphier: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: ux500: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: xtensa: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: arm: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: codecs: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: soc: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: soc-core: set rtd->num_cpu/codec at soc_new_pcm_runtime() ASoC: soc-core: tidyup soc_new_pcm_runtime() rtd setups ASoC: soc-core: remove cpu_dai/codec_dai/cpu_dais/codec_dais include/sound/soc.h | 30 +++++++------ sound/arm/pxa2xx-pcm-lib.c | 8 ++-- sound/soc/amd/acp-da7219-max98357a.c | 2 +- sound/soc/amd/acp-rt5645.c | 4 +- sound/soc/amd/acp3x-rt5682-max9836.c | 6 +-- sound/soc/atmel/atmel-pcm-dma.c | 4 +- sound/soc/atmel/atmel-pcm-pdc.c | 2 +- sound/soc/atmel/atmel_wm8904.c | 2 +- sound/soc/atmel/mikroe-proto.c | 2 +- sound/soc/atmel/sam9g20_wm8731.c | 2 +- sound/soc/atmel/sam9x5_wm8731.c | 2 +- sound/soc/au1x/db1200.c | 2 +- sound/soc/au1x/dbdma2.c | 2 +- sound/soc/au1x/dma.c | 2 +- sound/soc/au1x/psc-ac97.c | 2 +- sound/soc/bcm/bcm63xx-pcm-whistler.c | 16 +++---- sound/soc/bcm/cygnus-pcm.c | 22 +++++----- sound/soc/cirrus/edb93xx.c | 4 +- sound/soc/cirrus/snappercl15.c | 4 +- sound/soc/codecs/cs47l15.c | 4 +- sound/soc/codecs/cs47l24.c | 6 +-- sound/soc/codecs/cs47l35.c | 6 +-- sound/soc/codecs/cs47l85.c | 6 +-- sound/soc/codecs/cs47l90.c | 6 +-- sound/soc/codecs/cs47l92.c | 4 +- sound/soc/codecs/wm5110.c | 6 +-- sound/soc/codecs/wm_adsp.c | 10 ++--- sound/soc/dwc/dwc-pcm.c | 2 +- sound/soc/fsl/eukrea-tlv320.c | 4 +- sound/soc/fsl/fsl-asoc-card.c | 10 ++--- sound/soc/fsl/fsl_asrc_dma.c | 6 +-- sound/soc/fsl/fsl_spdif.c | 10 ++--- sound/soc/fsl/fsl_ssi.c | 8 ++-- sound/soc/fsl/imx-audmix.c | 8 ++-- sound/soc/fsl/imx-mc13783.c | 4 +- sound/soc/fsl/imx-sgtl5000.c | 2 +- sound/soc/fsl/mpc5200_dma.c | 10 ++--- sound/soc/fsl/mpc5200_psc_i2s.c | 2 +- sound/soc/fsl/mpc8610_hpcd.c | 4 +- sound/soc/fsl/mx27vis-aic32x4.c | 4 +- sound/soc/fsl/p1022_ds.c | 4 +- sound/soc/fsl/p1022_rdk.c | 4 +- sound/soc/fsl/wm1133-ev1.c | 6 +-- sound/soc/generic/simple-card-utils.c | 12 +++--- sound/soc/img/img-i2s-in.c | 2 +- sound/soc/img/img-i2s-out.c | 2 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 6 +-- sound/soc/intel/boards/bdw-rt5650.c | 6 +-- sound/soc/intel/boards/bdw-rt5677.c | 6 +-- sound/soc/intel/boards/broadwell.c | 4 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 8 ++-- sound/soc/intel/boards/bxt_rt298.c | 8 ++-- sound/soc/intel/boards/byt-max98090.c | 2 +- sound/soc/intel/boards/byt-rt5640.c | 4 +- sound/soc/intel/boards/bytcht_cx2072x.c | 10 ++--- sound/soc/intel/boards/bytcht_da7213.c | 8 ++-- sound/soc/intel/boards/bytcht_es8316.c | 8 ++-- sound/soc/intel/boards/bytcht_nocodec.c | 4 +- sound/soc/intel/boards/bytcr_rt5640.c | 8 ++-- sound/soc/intel/boards/bytcr_rt5651.c | 8 ++-- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 6 +-- sound/soc/intel/boards/cht_bsw_nau8824.c | 4 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 14 +++---- sound/soc/intel/boards/cht_bsw_rt5672.c | 8 ++-- sound/soc/intel/boards/cml_rt1011_rt5682.c | 6 +-- sound/soc/intel/boards/glk_rt5682_max98357a.c | 10 ++--- sound/soc/intel/boards/haswell.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98357a.c | 8 ++-- sound/soc/intel/boards/kbl_da7219_max98927.c | 6 +-- sound/soc/intel/boards/kbl_rt5660.c | 6 +-- sound/soc/intel/boards/kbl_rt5663_max98927.c | 8 ++-- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 8 ++-- .../soc/intel/boards/skl_nau88l25_max98357a.c | 12 +++--- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 16 +++---- sound/soc/intel/boards/skl_rt286.c | 8 ++-- sound/soc/intel/boards/sof_da7219_max98373.c | 8 ++-- sound/soc/intel/boards/sof_pcm512x.c | 8 ++-- sound/soc/intel/boards/sof_rt5682.c | 6 +-- sound/soc/intel/haswell/sst-haswell-pcm.c | 26 ++++++------ sound/soc/intel/skylake/skl-pcm.c | 10 ++--- sound/soc/kirkwood/armada-370-db.c | 2 +- sound/soc/kirkwood/kirkwood-dma.c | 2 +- sound/soc/mediatek/common/mtk-afe-fe-dai.c | 10 ++--- .../mediatek/common/mtk-afe-platform-driver.c | 2 +- sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 2 +- sound/soc/mediatek/mt2701/mt2701-cs42448.c | 4 +- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 4 +- sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 2 +- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 2 +- sound/soc/mediatek/mt8173/mt8173-max98090.c | 4 +- .../mediatek/mt8173/mt8173-rt5650-rt5514.c | 2 +- .../mediatek/mt8173/mt8173-rt5650-rt5676.c | 4 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 6 +-- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 2 +- .../mediatek/mt8183/mt8183-da7219-max98357.c | 4 +- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 2 +- sound/soc/meson/aiu-fifo.c | 2 +- sound/soc/meson/axg-card.c | 8 ++-- sound/soc/meson/axg-fifo.c | 2 +- sound/soc/meson/meson-card-utils.c | 2 +- sound/soc/mxs/mxs-sgtl5000.c | 4 +- sound/soc/pxa/brownstone.c | 4 +- sound/soc/pxa/corgi.c | 4 +- sound/soc/pxa/hx4700.c | 4 +- sound/soc/pxa/imote2.c | 4 +- sound/soc/pxa/magician.c | 8 ++-- sound/soc/pxa/mioa701_wm9713.c | 4 +- sound/soc/pxa/mmp-pcm.c | 2 +- sound/soc/pxa/mmp-sspa.c | 2 +- sound/soc/pxa/poodle.c | 4 +- sound/soc/pxa/pxa2xx-i2s.c | 2 +- sound/soc/pxa/spitz.c | 4 +- sound/soc/pxa/ttc-dkb.c | 2 +- sound/soc/pxa/z2.c | 4 +- sound/soc/pxa/zylonite.c | 6 +-- sound/soc/qcom/apq8016_sbc.c | 2 +- sound/soc/qcom/apq8096.c | 6 +-- sound/soc/qcom/lpass-platform.c | 2 +- sound/soc/qcom/qdsp6/q6asm-dai.c | 4 +- sound/soc/qcom/qdsp6/q6routing.c | 2 +- sound/soc/qcom/sdm845.c | 22 +++++----- sound/soc/qcom/storm.c | 2 +- sound/soc/rockchip/rk3288_hdmi_analog.c | 4 +- sound/soc/rockchip/rk3399_gru_sound.c | 16 +++---- sound/soc/rockchip/rockchip_max98090.c | 6 +-- sound/soc/rockchip/rockchip_rt5645.c | 6 +-- sound/soc/samsung/arndale.c | 6 +-- sound/soc/samsung/bells.c | 16 +++---- sound/soc/samsung/h1940_uda1380.c | 2 +- sound/soc/samsung/i2s.c | 2 +- sound/soc/samsung/jive_wm8750.c | 4 +- sound/soc/samsung/littlemill.c | 14 +++---- sound/soc/samsung/lowland.c | 4 +- sound/soc/samsung/neo1973_wm8753.c | 10 ++--- sound/soc/samsung/odroid.c | 2 +- sound/soc/samsung/pcm.c | 4 +- sound/soc/samsung/rx1950_uda1380.c | 2 +- sound/soc/samsung/s3c-i2s-v2.c | 2 +- sound/soc/samsung/s3c24xx_simtec.c | 4 +- sound/soc/samsung/s3c24xx_uda134x.c | 6 +-- sound/soc/samsung/smartq_wm8987.c | 4 +- sound/soc/samsung/smdk_spdif.c | 2 +- sound/soc/samsung/smdk_wm8580.c | 2 +- sound/soc/samsung/smdk_wm8994.c | 2 +- sound/soc/samsung/smdk_wm8994pcm.c | 4 +- sound/soc/samsung/snow.c | 4 +- sound/soc/samsung/spdif.c | 8 ++-- sound/soc/samsung/speyside.c | 8 ++-- sound/soc/samsung/tm2_wm5110.c | 16 +++---- sound/soc/samsung/tobermory.c | 8 ++-- sound/soc/sh/dma-sh7760.c | 16 +++---- sound/soc/sh/fsi.c | 2 +- sound/soc/sh/migor.c | 6 +-- sound/soc/sh/rcar/core.c | 2 +- sound/soc/soc-compress.c | 36 ++++++++-------- sound/soc/soc-core.c | 42 +++++++------------ sound/soc/soc-dapm.c | 4 +- sound/soc/soc-generic-dmaengine-pcm.c | 6 +-- sound/soc/soc-pcm.c | 30 ++++++------- sound/soc/sof/intel/hda-dai.c | 6 +-- sound/soc/sof/intel/hda-dsp.c | 2 +- sound/soc/sprd/sprd-pcm-compress.c | 4 +- sound/soc/sprd/sprd-pcm-dma.c | 2 +- sound/soc/stm/stm32_adfsdm.c | 12 +++--- sound/soc/stm/stm32_sai_sub.c | 2 +- sound/soc/sunxi/sun4i-spdif.c | 2 +- sound/soc/tegra/tegra_alc5632.c | 2 +- sound/soc/tegra/tegra_max98090.c | 2 +- sound/soc/tegra/tegra_rt5640.c | 2 +- sound/soc/tegra/tegra_rt5677.c | 2 +- sound/soc/tegra/tegra_sgtl5000.c | 2 +- sound/soc/tegra/tegra_wm8753.c | 2 +- sound/soc/tegra/tegra_wm8903.c | 6 +-- sound/soc/tegra/trimslice.c | 2 +- sound/soc/ti/ams-delta.c | 4 +- sound/soc/ti/davinci-evm.c | 4 +- sound/soc/ti/davinci-vcif.c | 4 +- sound/soc/ti/n810.c | 2 +- sound/soc/ti/omap-abe-twl6040.c | 6 +-- sound/soc/ti/omap-mcbsp-st.c | 2 +- sound/soc/ti/omap-mcbsp.c | 4 +- sound/soc/ti/omap-mcpdm.c | 2 +- sound/soc/ti/omap3pandora.c | 4 +- sound/soc/ti/osk5912.c | 2 +- sound/soc/ti/rx51.c | 2 +- sound/soc/txx9/txx9aclc.c | 2 +- sound/soc/uniphier/aio-compress.c | 22 +++++----- sound/soc/uniphier/aio-dma.c | 6 +-- sound/soc/ux500/mop500_ab8500.c | 6 +-- sound/soc/ux500/ux500_pcm.c | 8 ++-- sound/soc/xtensa/xtfpga-i2s.c | 2 +- 191 files changed, 573 insertions(+), 577 deletions(-) -- 2.17.1

3 72

[PATCH v2 00/11] ASoC: SOF: Intel: add SoundWire support
by Pierre-Louis Bossart 27 Mar '20

27 Mar '20

This patchset provides the support for SoundWire support on Intel CometLake, IcelLake and TigerLake RVP platforms and form-factor devices to be released 'soon'. The bulk of the code is about detecting a valid SoundWire configuration from ACPI, and implementing the interfaces suggested in '[PATCH 0/8] soundwire: remove platform devices, add SOF interfaces' for interrupts, PCI wakes and clock-stop configurations. Since that SoundWire series will not be in 5.7, the build support for SOF w/ SoundWire is not provided for now, and fall-back functions will be used. This code is tested on a daily basis in the SOF tree and is not expected to change in significant ways. Changes since v2: Corrected error in ACPI table (thanks Amadeusz) Added patch 11 to add reset cycle required on some SoundWire platforms Bard Liao (1): ASoC: SOF: Intel: hda: merge IPC, stream and SoundWire interrupt handlers Pierre-Louis Bossart (8): ASoC: soc-acpi: expand description of _ADR-based devices ASoC: SOF: Intel: add SoundWire configuration interface ASoC: SOF: IPC: dai-intel: move ALH declarations in header file ASoC: SOF: Intel: hda: add SoundWire stream config/free callbacks ASoC: SOF: Intel: hda: initial SoundWire machine driver autodetect ASoC: SOF: Intel: hda: disable SoundWire interrupts on suspend ASoC: SOF: Intel: hda: add parameter to control SoundWire clock stop quirks ASoC: SOF: Intel: hda-ctrl: add reset cycle before parsing capabilities Rander Wang (2): ASoC: SOF: Intel: hda: add WAKEEN interrupt support for SoundWire Asoc: SOF: Intel: hda: check SoundWire wakeen interrupt in irq thread include/sound/soc-acpi.h | 39 +- include/sound/sof/dai-intel.h | 18 +- .../intel/common/soc-acpi-intel-cml-match.c | 87 +++- .../intel/common/soc-acpi-intel-icl-match.c | 97 ++++- .../intel/common/soc-acpi-intel-tgl-match.c | 49 ++- sound/soc/sof/intel/hda-ctrl.c | 25 +- sound/soc/sof/intel/hda-dsp.c | 2 + sound/soc/sof/intel/hda-loader.c | 31 ++ sound/soc/sof/intel/hda.c | 400 ++++++++++++++++++ sound/soc/sof/intel/hda.h | 66 +++ 10 files changed, 750 insertions(+), 64 deletions(-) -- 2.20.1

2 14

Applied "ASoC: SOF: Intel: add SoundWire configuration interface" to the asoc tree
by Mark Brown 27 Mar '20

27 Mar '20

The patch ASoC: SOF: Intel: add SoundWire configuration interface has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 51dfed1e178a38202960b98f6e29df009a06050f Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Date: Wed, 25 Mar 2020 16:50:18 -0500 Subject: [PATCH] ASoC: SOF: Intel: add SoundWire configuration interface Now that the SoundWire core supports the multi-step initialization, call the relevant APIs. The actual hardware enablement can be done in two places, ideally we'd want to startup the SoundWire IP as soon as possible (while still taking power rail dependencies into account) However when suspend/resume is implemented, the DSP device will be resumed first, and only when the DSP firmware is downloaded/booted would the SoundWire child devices be resumed, so there are only marginal benefits in starting the IP earlier for the first probe. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200325215027.28716-3-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/intel/hda-loader.c | 13 ++++ sound/soc/sof/intel/hda.c | 120 +++++++++++++++++++++++++++++++ sound/soc/sof/intel/hda.h | 44 ++++++++++++ 3 files changed, 177 insertions(+) diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index 0633b76dab49..2ae94ea53122 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -399,6 +399,19 @@ int hda_dsp_pre_fw_run(struct snd_sof_dev *sdev) /* post fw run operations */ int hda_dsp_post_fw_run(struct snd_sof_dev *sdev) { + int ret; + + if (sdev->first_boot) { + ret = hda_sdw_startup(sdev); + if (ret < 0) { + dev_err(sdev->dev, + "error: could not startup SoundWire links\n"); + return ret; + } + } + + hda_sdw_int_enable(sdev, true); + /* re-enable clock gating and power gating */ return hda_dsp_ctrl_clock_power_gating(sdev, true); } diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index ee1edb53840c..c1fe94800da1 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -18,7 +18,9 @@ #include <sound/hdaudio_ext.h> #include <sound/hda_register.h> +#include <linux/acpi.h> #include <linux/module.h> +#include <linux/soundwire/sdw_intel.h> #include <sound/intel-nhlt.h> #include <sound/sof.h> #include <sound/sof/xtensa.h> @@ -34,6 +36,98 @@ #define EXCEPT_MAX_HDR_SIZE 0x400 +#if IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE) + +void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable) +{ + sdw_intel_enable_irq(sdev->bar[HDA_DSP_BAR], enable); +} + +static int hda_sdw_acpi_scan(struct snd_sof_dev *sdev) +{ + struct sof_intel_hda_dev *hdev; + acpi_handle handle; + int ret; + + handle = ACPI_HANDLE(sdev->dev); + + /* save ACPI info for the probe step */ + hdev = sdev->pdata->hw_pdata; + + ret = sdw_intel_acpi_scan(handle, &hdev->info); + if (ret < 0) { + dev_err(sdev->dev, "%s failed\n", __func__); + return -EINVAL; + } + + return 0; +} + +static int hda_sdw_probe(struct snd_sof_dev *sdev) +{ + struct sof_intel_hda_dev *hdev; + struct sdw_intel_res res; + acpi_handle handle; + void *sdw; + + handle = ACPI_HANDLE(sdev->dev); + + hdev = sdev->pdata->hw_pdata; + + memset(&res, 0, sizeof(res)); + + res.mmio_base = sdev->bar[HDA_DSP_BAR]; + res.irq = sdev->ipc_irq; + res.handle = hdev->info.handle; + res.parent = sdev->dev; + + /* + * ops and arg fields are not populated for now, + * they will be needed when the DAI callbacks are + * provided + */ + + /* we could filter links here if needed, e.g for quirks */ + res.count = hdev->info.count; + res.link_mask = hdev->info.link_mask; + + sdw = sdw_intel_probe(&res); + if (!sdw) { + dev_err(sdev->dev, "error: SoundWire probe failed\n"); + return -EINVAL; + } + + /* save context */ + hdev->sdw = sdw; + + return 0; +} + +int hda_sdw_startup(struct snd_sof_dev *sdev) +{ + struct sof_intel_hda_dev *hdev; + + hdev = sdev->pdata->hw_pdata; + + return sdw_intel_startup(hdev->sdw); +} + +static int hda_sdw_exit(struct snd_sof_dev *sdev) +{ + struct sof_intel_hda_dev *hdev; + + hdev = sdev->pdata->hw_pdata; + + hda_sdw_int_enable(sdev, false); + + if (hdev->sdw) + sdw_intel_exit(hdev->sdw); + hdev->sdw = NULL; + + return 0; +} +#endif + /* * Debug */ @@ -346,9 +440,12 @@ static const char *fixup_tplg_name(struct snd_sof_dev *sdev, static int hda_init_caps(struct snd_sof_dev *sdev) { struct hdac_bus *bus = sof_to_bus(sdev); + struct snd_sof_pdata *pdata = sdev->pdata; #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) struct hdac_ext_link *hlink; #endif + struct sof_intel_hda_dev *hdev = pdata->hw_pdata; + u32 link_mask; int ret = 0; device_disable_async_suspend(bus->dev); @@ -365,6 +462,27 @@ static int hda_init_caps(struct snd_sof_dev *sdev) return ret; } + /* scan SoundWire capabilities exposed by DSDT */ + ret = hda_sdw_acpi_scan(sdev); + if (ret < 0) { + dev_err(sdev->dev, "error: SoundWire ACPI scan error\n"); + return ret; + } + + link_mask = hdev->info.link_mask; + if (!link_mask) { + /* + * probe/allocated SoundWire resources. + * The hardware configuration takes place in hda_sdw_startup + * after power rails are enabled. + */ + ret = hda_sdw_probe(sdev); + if (ret < 0) { + dev_err(sdev->dev, "error: SoundWire probe error\n"); + return ret; + } + } + #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) if (bus->mlcap) snd_hdac_ext_bus_get_ml_capabilities(bus); @@ -622,6 +740,8 @@ int hda_dsp_remove(struct snd_sof_dev *sdev) snd_hdac_ext_bus_device_remove(bus); #endif + hda_sdw_exit(sdev); + if (!IS_ERR_OR_NULL(hda->dmic_dev)) platform_device_unregister(hda->dmic_dev); diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 2a57fc9f3e58..928a3432e9e6 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -11,6 +11,8 @@ #ifndef __SOF_INTEL_HDA_H #define __SOF_INTEL_HDA_H +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_intel.h> #include <sound/compress_driver.h> #include <sound/hda_codec.h> #include <sound/hdaudio_ext.h> @@ -436,6 +438,12 @@ struct sof_intel_hda_dev { /* delayed work to enter D0I3 opportunistically */ struct delayed_work d0i3_work; + + /* ACPI information stored between scan and probe steps */ + struct sdw_intel_acpi_info info; + + /* sdw context allocated by SoundWire driver */ + struct sdw_intel_ctx *sdw; }; static inline struct hdac_bus *sof_to_bus(struct snd_sof_dev *s) @@ -654,6 +662,42 @@ int hda_dsp_trace_init(struct snd_sof_dev *sdev, u32 *stream_tag); int hda_dsp_trace_release(struct snd_sof_dev *sdev); int hda_dsp_trace_trigger(struct snd_sof_dev *sdev, int cmd); +/* + * SoundWire support + */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE) + +int hda_sdw_startup(struct snd_sof_dev *sdev); +void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable); + +#else + +static inline int hda_sdw_acpi_scan(struct snd_sof_dev *sdev) +{ + return 0; +} + +static inline int hda_sdw_probe(struct snd_sof_dev *sdev) +{ + return 0; +} + +static inline int hda_sdw_startup(struct snd_sof_dev *sdev) +{ + return 0; +} + +static inline int hda_sdw_exit(struct snd_sof_dev *sdev) +{ + return 0; +} + +static inline void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable) +{ +} + +#endif + /* common dai driver */ extern struct snd_soc_dai_driver skl_dai[]; -- 2.20.1

1 0

Applied "ASoC: SOF: IPC: dai-intel: move ALH declarations in header file" to the asoc tree
by Mark Brown 27 Mar '20

27 Mar '20

The patch ASoC: SOF: IPC: dai-intel: move ALH declarations in header file has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From f8e25018801548314b3d908315cbf271e8eceba8 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Date: Wed, 25 Mar 2020 16:50:19 -0500 Subject: [PATCH] ASoC: SOF: IPC: dai-intel: move ALH declarations in header file ALH was inserted in the wrong place during integration, add after DMIC to mirror the file used by SOF firmware. No functional change, just text move in the same file to better track changes, if any. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200325215027.28716-4-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/sof/dai-intel.h | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/include/sound/sof/dai-intel.h b/include/sound/sof/dai-intel.h index 5f1ef5565be6..04e48227f542 100644 --- a/include/sound/sof/dai-intel.h +++ b/include/sound/sof/dai-intel.h @@ -87,6 +87,15 @@ struct sof_ipc_dai_hda_params { uint32_t link_dma_ch; } __packed; +/* ALH Configuration Request - SOF_IPC_DAI_ALH_CONFIG */ +struct sof_ipc_dai_alh_params { + struct sof_ipc_hdr hdr; + uint32_t stream_id; + + /* reserved for future use */ + uint32_t reserved[15]; +} __packed; + /* DMIC Configuration Request - SOF_IPC_DAI_DMIC_CONFIG */ /* This struct is defined per 2ch PDM controller available in the platform. @@ -179,13 +188,4 @@ struct sof_ipc_dai_dmic_params { struct sof_ipc_dai_dmic_pdm_ctrl pdm[0]; } __packed; -/* ALH Configuration Request - SOF_IPC_DAI_ALH_CONFIG */ -struct sof_ipc_dai_alh_params { - struct sof_ipc_hdr hdr; - uint32_t stream_id; - - /* reserved for future use */ - uint32_t reserved[15]; -} __packed; - #endif -- 2.20.1

1 0

Applied "ASoC: SOF: Intel: hda: add SoundWire stream config/free callbacks" to the asoc tree
by Mark Brown 27 Mar '20

27 Mar '20

The patch ASoC: SOF: Intel: hda: add SoundWire stream config/free callbacks has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From d2c383aa49dce9c5b59930c999099ecc580857fe Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Date: Wed, 25 Mar 2020 16:50:20 -0500 Subject: [PATCH] ASoC: SOF: Intel: hda: add SoundWire stream config/free callbacks These callbacks are invoked when a matching hw_params/hw_free() DAI operation takes place, and will result in IPC operations with the SOF firmware. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200325215027.28716-5-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/intel/hda.c | 71 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 71 insertions(+) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index c1fe94800da1..1f93124a63cd 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -24,6 +24,7 @@ #include <sound/intel-nhlt.h> #include <sound/sof.h> #include <sound/sof/xtensa.h> +#include "../sof-audio.h" #include "../ops.h" #include "hda.h" @@ -38,6 +39,74 @@ #if IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE) +static int sdw_params_stream(struct device *dev, + struct sdw_intel_stream_params_data *params_data) +{ + struct snd_sof_dev *sdev = dev_get_drvdata(dev); + struct snd_soc_dai *d = params_data->dai; + struct sof_ipc_dai_config config; + struct sof_ipc_reply reply; + int link_id = params_data->link_id; + int alh_stream_id = params_data->alh_stream_id; + int ret; + u32 size = sizeof(config); + + memset(&config, 0, size); + config.hdr.size = size; + config.hdr.cmd = SOF_IPC_GLB_DAI_MSG | SOF_IPC_DAI_CONFIG; + config.type = SOF_DAI_INTEL_ALH; + config.dai_index = (link_id << 8) | (d->id); + config.alh.stream_id = alh_stream_id; + + /* send message to DSP */ + ret = sof_ipc_tx_message(sdev->ipc, + config.hdr.cmd, &config, size, &reply, + sizeof(reply)); + if (ret < 0) { + dev_err(sdev->dev, + "error: failed to set DAI hw_params for link %d dai->id %d ALH %d\n", + link_id, d->id, alh_stream_id); + } + + return ret; +} + +static int sdw_free_stream(struct device *dev, + struct sdw_intel_stream_free_data *free_data) +{ + struct snd_sof_dev *sdev = dev_get_drvdata(dev); + struct snd_soc_dai *d = free_data->dai; + struct sof_ipc_dai_config config; + struct sof_ipc_reply reply; + int link_id = free_data->link_id; + int ret; + u32 size = sizeof(config); + + memset(&config, 0, size); + config.hdr.size = size; + config.hdr.cmd = SOF_IPC_GLB_DAI_MSG | SOF_IPC_DAI_CONFIG; + config.type = SOF_DAI_INTEL_ALH; + config.dai_index = (link_id << 8) | d->id; + config.alh.stream_id = 0xFFFF; /* invalid value on purpose */ + + /* send message to DSP */ + ret = sof_ipc_tx_message(sdev->ipc, + config.hdr.cmd, &config, size, &reply, + sizeof(reply)); + if (ret < 0) { + dev_err(sdev->dev, + "error: failed to free stream for link %d dai->id %d\n", + link_id, d->id); + } + + return ret; +} + +static const struct sdw_intel_ops sdw_callback = { + .params_stream = sdw_params_stream, + .free_stream = sdw_free_stream, +}; + void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable) { sdw_intel_enable_irq(sdev->bar[HDA_DSP_BAR], enable); @@ -80,6 +149,8 @@ static int hda_sdw_probe(struct snd_sof_dev *sdev) res.irq = sdev->ipc_irq; res.handle = hdev->info.handle; res.parent = sdev->dev; + res.ops = &sdw_callback; + res.dev = sdev->dev; /* * ops and arg fields are not populated for now, -- 2.20.1

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Applied "ASoC: SOF: Intel: hda: disable SoundWire interrupts on suspend" to the asoc tree
by Mark Brown 27 Mar '20

27 Mar '20

The patch ASoC: SOF: Intel: hda: disable SoundWire interrupts on suspend has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 3eadff5639b01c17f5f2ffeb209d05cc19706687 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Date: Wed, 25 Mar 2020 16:50:22 -0500 Subject: [PATCH] ASoC: SOF: Intel: hda: disable SoundWire interrupts on suspend Doing this avoid conflicts and errors reported on the bus. The interrupts are only re-enabled on resume after the firmware is downloaded, so the behavior is not fully symmetric Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200325215027.28716-7-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/intel/hda-dsp.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index c396b7ef0328..1aff90042694 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -594,6 +594,8 @@ static int hda_suspend(struct snd_sof_dev *sdev, bool runtime_suspend) #endif int ret; + hda_sdw_int_enable(sdev, false); + /* disable IPC interrupts */ hda_dsp_ipc_int_disable(sdev); -- 2.20.1

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Applied "ASoC: SOF: Intel: hda: merge IPC, stream and SoundWire interrupt handlers" to the asoc tree
by Mark Brown 27 Mar '20

27 Mar '20

The patch ASoC: SOF: Intel: hda: merge IPC, stream and SoundWire interrupt handlers has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 722ba5f1f530a919d28a0dd9e8e0ec63af18270d Mon Sep 17 00:00:00 2001 From: Bard Liao <yung-chuan.liao(a)linux.intel.com> Date: Wed, 25 Mar 2020 16:50:23 -0500 Subject: [PATCH] ASoC: SOF: Intel: hda: merge IPC, stream and SoundWire interrupt handlers We have a single irq handler for SOF interrupts. We can further merge SoundWire ones to completely remove MSI interrupts handling issues leading to timeouts. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://lore.kernel.org/r/20200325215027.28716-8-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/intel/hda.c | 36 ++++++++++++++++++++++++++++++++++++ sound/soc/sof/intel/hda.h | 11 +++++++++++ 2 files changed, 47 insertions(+) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index dea3c385b664..ee4f1ceca883 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -198,6 +198,38 @@ static int hda_sdw_exit(struct snd_sof_dev *sdev) return 0; } + +static bool hda_dsp_check_sdw_irq(struct snd_sof_dev *sdev) +{ + struct sof_intel_hda_dev *hdev; + bool ret = false; + u32 irq_status; + + hdev = sdev->pdata->hw_pdata; + + if (!hdev->sdw) + return ret; + + /* store status */ + irq_status = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_ADSPIS2); + + /* invalid message ? */ + if (irq_status == 0xffffffff) + goto out; + + /* SDW message ? */ + if (irq_status & HDA_DSP_REG_ADSPIS2_SNDW) + ret = true; + +out: + return ret; +} + +static irqreturn_t hda_dsp_sdw_thread(int irq, void *context) +{ + return sdw_intel_thread(irq, context); +} + #endif /* @@ -618,6 +650,7 @@ static irqreturn_t hda_dsp_interrupt_handler(int irq, void *context) static irqreturn_t hda_dsp_interrupt_thread(int irq, void *context) { struct snd_sof_dev *sdev = context; + struct sof_intel_hda_dev *hdev = sdev->pdata->hw_pdata; /* deal with streams and controller first */ if (hda_dsp_check_stream_irq(sdev)) @@ -626,6 +659,9 @@ static irqreturn_t hda_dsp_interrupt_thread(int irq, void *context) if (hda_dsp_check_ipc_irq(sdev)) sof_ops(sdev)->irq_thread(irq, sdev); + if (hda_dsp_check_sdw_irq(sdev)) + hda_dsp_sdw_thread(irq, hdev->sdw); + /* enable GIE interrupt */ snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTCTL, diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 928a3432e9e6..fc104c5ba006 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -232,6 +232,8 @@ #define HDA_DSP_REG_ADSPIC2 (HDA_DSP_GEN_BASE + 0x10) #define HDA_DSP_REG_ADSPIS2 (HDA_DSP_GEN_BASE + 0x14) +#define HDA_DSP_REG_ADSPIS2_SNDW BIT(5) + /* Intel HD Audio Inter-Processor Communication Registers */ #define HDA_DSP_IPC_BASE 0x40 #define HDA_DSP_REG_HIPCT (HDA_DSP_IPC_BASE + 0x00) @@ -696,6 +698,15 @@ static inline void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable) { } +static inline bool hda_dsp_check_sdw_irq(struct snd_sof_dev *sdev) +{ + return false; +} + +static inline irqreturn_t hda_dsp_sdw_thread(int irq, void *context) +{ + return IRQ_HANDLED; +} #endif /* common dai driver */ -- 2.20.1

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Applied "ASoC: SOF: Intel: hda: add parameter to control SoundWire clock stop quirks" to the asoc tree
by Mark Brown 27 Mar '20

27 Mar '20

The patch ASoC: SOF: Intel: hda: add parameter to control SoundWire clock stop quirks has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 02df8f4364b070428af0e5b6c8739c884c8ad4e7 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Date: Wed, 25 Mar 2020 16:50:24 -0500 Subject: [PATCH] ASoC: SOF: Intel: hda: add parameter to control SoundWire clock stop quirks Add module parameter so that the different modes can be quickly tested. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200325215027.28716-9-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/intel/hda.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index ee4f1ceca883..1e69cfcee8e0 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -40,6 +40,16 @@ #if IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE) +/* + * The default for SoundWire clock stop quirks is to power gate the IP + * and do a Bus Reset, this will need to be modified when the DSP + * needs to remain in D0i3 so that the Master does not lose context + * and enumeration is not required on clock restart + */ +static int sdw_clock_stop_quirks = SDW_INTEL_CLK_STOP_BUS_RESET; +module_param(sdw_clock_stop_quirks, int, 0444); +MODULE_PARM_DESC(sdw_clock_stop_quirks, "SOF SoundWire clock stop quirks"); + static int sdw_params_stream(struct device *dev, struct sdw_intel_stream_params_data *params_data) { @@ -149,6 +159,7 @@ static int hda_sdw_probe(struct snd_sof_dev *sdev) res.parent = sdev->dev; res.ops = &sdw_callback; res.dev = sdev->dev; + res.clock_stop_quirks = sdw_clock_stop_quirks; /* * ops and arg fields are not populated for now, -- 2.20.1

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Applied "ASoC: SOF: Intel: hda: add WAKEEN interrupt support for SoundWire" to the asoc tree
by Mark Brown 27 Mar '20

27 Mar '20

The patch ASoC: SOF: Intel: hda: add WAKEEN interrupt support for SoundWire has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From bbd19cdca8279cf244a301c6a13ae5ec9e4ef976 Mon Sep 17 00:00:00 2001 From: Rander Wang <rander.wang(a)intel.com> Date: Wed, 25 Mar 2020 16:50:25 -0500 Subject: [PATCH] ASoC: SOF: Intel: hda: add WAKEEN interrupt support for SoundWire When a SoundWire link is in clock stop state, a Slave device may wake up the Master for some events such as jack detection. The WAKEEN interrupt will be triggered and processed by the audio pci device. If audio device is in D3, the interrupt will be routed to PME, or aggregated at cAVS level as interrupt when audio device is in D0. This patch only supports D3 case, where the audio pci device will be resumed by a PME event and the WAKEEN interrupt will be processed after audio pci device is powered up and ROM is initialized successfully. The WAKEEN handling is only enabled after the first boot due to dependencies on a shim_lock mutex being initialized. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Rander Wang <rander.wang(a)intel.com> Link: https://lore.kernel.org/r/20200325215027.28716-10-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/intel/hda-loader.c | 18 ++++++++++++++++++ sound/soc/sof/intel/hda.c | 11 +++++++++++ sound/soc/sof/intel/hda.h | 5 +++++ 3 files changed, 34 insertions(+) diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index 2ae94ea53122..e1550ccd0a49 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -346,6 +346,24 @@ int hda_dsp_cl_boot_firmware(struct snd_sof_dev *sdev) goto cleanup; } + /* + * When a SoundWire link is in clock stop state, a Slave + * device may trigger in-band wakes for events such as jack + * insertion or acoustic event detection. This event will lead + * to a WAKEEN interrupt, handled by the PCI device and routed + * to PME if the PCI device is in D3. The resume function in + * audio PCI driver will be invoked by ACPI for PME event and + * initialize the device and process WAKEEN interrupt. + * + * The WAKEEN interrupt should be processed ASAP to prevent an + * interrupt flood, otherwise other interrupts, such IPC, + * cannot work normally. The WAKEEN is handled after the ROM + * is initialized successfully, which ensures power rails are + * enabled before accessing the SoundWire SHIM registers + */ + if (!sdev->first_boot) + hda_sdw_process_wakeen(sdev); + /* * at this point DSP ROM has been initialized and * should be ready for code loading and firmware boot diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 1e69cfcee8e0..7d1aa4c7d82c 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -241,6 +241,17 @@ static irqreturn_t hda_dsp_sdw_thread(int irq, void *context) return sdw_intel_thread(irq, context); } +void hda_sdw_process_wakeen(struct snd_sof_dev *sdev) +{ + struct sof_intel_hda_dev *hdev; + + hdev = sdev->pdata->hw_pdata; + if (!hdev->sdw) + return; + + sdw_intel_process_wakeen_event(hdev->sdw); +} + #endif /* diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index fc104c5ba006..6f1765b1ed1d 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -671,6 +671,7 @@ int hda_dsp_trace_trigger(struct snd_sof_dev *sdev, int cmd); int hda_sdw_startup(struct snd_sof_dev *sdev); void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable); +void hda_sdw_process_wakeen(struct snd_sof_dev *sdev); #else @@ -707,6 +708,10 @@ static inline irqreturn_t hda_dsp_sdw_thread(int irq, void *context) { return IRQ_HANDLED; } + +static inline void hda_sdw_process_wakeen(struct snd_sof_dev *sdev) +{ +} #endif /* common dai driver */ -- 2.20.1

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