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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