Alsa-devel February 2022

alsa-devel@alsa-project.org

121 participants
272 discussions

Re: RFC: usb: gadget: u_audio: Notifying gadget that host started playback/capture?
by Pavel Hofman 22 Sep '23

22 Sep '23

Hi, Dne 08. 09. 21 v 10:21 Pavel Hofman napsal(a): > Hi, > > The current audio gadget has no way to inform the gadget side that the > host side has started playback/capture and that gadget-side alsa > processes should be started. > > Playback/capture processes on the host side do not get stuck without the > gadget side consuming/producing data (OUT requests are ignored in > u_audio_iso_complete, IN ones send initial zeros in their req->buf). > > However, playback/capture processes on the gadget side get stuck without > the host side sending playback OUT packets or capture IN requests and > time out with error. If there was a way to inform the gadget side that > playback/capture has started on the host side, the gadget clients could > react accordingly. > I drafted a simple patch for u_audio.c which defines read-only boolean ctl elems "Capture Requested" and "Playback Requested". Their values are set/reset in methods u_audio_start_capture/playback and u_audio_stop_capture/playback, i.e. at changes of respective altsettings from 0 to 1 and back. Every ctl elem value change sends notification via snd_ctl_notify. The principle works OK for capture/playback start/stop on the host, as monitored by alsactl: pi@raspberrypi:~ $ alsactl monitor hw:UAC2Gadget node hw:UAC2Gadget, #4 (3,0,0,Capture Requested,0) VALUE node hw:UAC2Gadget, #4 (3,0,0,Capture Requested,0) VALUE node hw:UAC2Gadget, #3 (3,0,0,Playback Requested,0) VALUE node hw:UAC2Gadget, #3 (3,0,0,Playback Requested,0) VALUE However at enumeration the USB host switches both playback and capture altsettings repeatedly, generating "fake" events from the gadget side POW. The host even sends regular-sized EP-OUT packets filled with zeros during enumeration (tested on linux only for now). Please is there any way to "detect" the enumeration stage to mask out the "fake" playback/capture start/stop events? The attached patch does not apply cleanly to mainline u_audio.c because it's rebased on other patches not submitted yet but it's only a discussion inducer for now. Thanks a lot, Pavel.

2 2

[alsa-devel] [PATCH 0/6] ASoC: mediatek: mt8183: support DP audio
by Tzung-Bi Shih 24 Nov '22

24 Nov '22

(This series depends on https://lore.kernel.org/patchwork/patch/1126819/ which has not applied to the maintainer's tree yet.) This series makes mt8183-mt6358-ts3a227-max98357 support DP (display port) audio output. ASoC: hdmi-codec: add PCM trigger operator - add a PCM trigger callback if codec driver is interested in PCM events. drm: bridge: it6505: bridge to hdmi-codec - add audio feature support of it6505. - bridge to hdmi-codec so that ASoC can use the audio feature. - defer the "enabling" after receiving PCM START or RESUME event. ASoC: dt-bindings: mt8183: add a property "mediatek,hdmi-codec" ASoC: mediatek: mt8183: use hdmi-codec - bind to hdmi-codec in mt8183-mt6358-ts3a227-max98357. drm: bridge: it6505: report connector status ASoC: mediatek: mt8183: support HDMI jack reporting - use the mechanism from the following patch https://mailman.alsa-project.org/pipermail/alsa-devel/2019-September/155302… to report jack status (from DRM to ASoC). Allen Chen (1): drm: bridge: it6505: bridge to hdmi-codec Tzung-Bi Shih (5): ASoC: hdmi-codec: add PCM trigger operator ASoC: dt-bindings: mt8183: add a property "mediatek,hdmi-codec" ASoC: mediatek: mt8183: use hdmi-codec drm: bridge: it6505: report connector status ASoC: mediatek: mt8183: support HDMI jack reporting .../sound/mt8183-mt6358-ts3a227-max98357.txt | 2 + drivers/gpu/drm/bridge/ite-it6505.c | 191 +++++++++++++++++- include/sound/hdmi-codec.h | 13 ++ sound/soc/codecs/hdmi-codec.c | 31 +++ sound/soc/mediatek/Kconfig | 3 +- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 36 +++- 6 files changed, 266 insertions(+), 10 deletions(-) -- 2.23.0.866.gb869b98d4c-goog

3 9

[PATCH v6 00/10] Add support for CS35L41 in HDA systems
by Lucas Tanure 31 Oct '22

31 Oct '22

Add support for laptops that have CS35L41 connected to an HDA codec by I2S and direct I2C connection to the CPU. Laptops that use CS35L41 and are SPI will be added in the future, after the support for it is resolved at i2c-multi-instantiate driver. i2c-multi-instantiate thread: https://lkml.org/lkml/2021/12/10/557 Hardware: - Some laptops have two CS35L41 amplifiers, connected to Realtek ALC287 by an I2S bus and by and direct I2C to the CPU. - The ALC287 codec is connected to the CPU by an HDA bus. - The CS35L41 has a DSP which will require firmware to be loaded. Architecture: - Creation of a library of shared functions for CS35L41 ASoC and HDA - HDA codec driver (RealTek) and CS35L41 HDA driver are combined by using component binding, where it uses device names to find the components and bind to the master - The HDA CS35L41 driver applies pre-defined registers sequences for each action in playback for HDA Sound card Changes from V5: - Fix build issues reported by Intel Test Bot - Check devm_kasprintf return - Add ACPI dependency to HDA drivers V5: https://lkml.org/lkml/2021/12/16/430 Changes from V4: - Save index received from ACPI for latter use in DSP Support - Move CS35L41_DSP1_CCM_CORE_CTRL to regmap_write so doesn't affect DSP hibernation feature V4: https://lkml.org/lkml/2021/12/14/487 Changes from V3: - SPI bus driver starter added - Use separate modules approach instead of liking library - Add CSC3551 ACPI _HID for more I2C laptops - Removed regulators from HDA driver - Add note about Non-conforming _HID V3: https://lkml.org/lkml/2021/11/23/723 Changes from V2: - Not an RFC - Create a new HDA driver for CS35L41 instead of using the ASoC one - Use component binding and device names to find the CS35L41 driver - Create a shared library for ASoC and HDA CS35L41 drivers v2: https://lkml.org/lkml/2021/10/8/344 Lucas Tanure (9): ASoC: cs35l41: Convert tables to shared source code ASoC: cs35l41: Move cs35l41_otp_unpack to shared code ASoC: cs35l41: Move power initializations to reg_sequence ASoC: cs35l41: Create shared function for errata patches ASoC: cs35l41: Create shared function for setting channels ASoC: cs35l41: Create shared function for boost configuration hda: cs35l41: Add support for CS35L41 in HDA systems ACPI / scan: Create platform device for CLSA0100 and CSC3551 ACPI nodes ALSA: hda/realtek: Add support for Legion 7 16ACHg6 laptop Stefan Binding (1): ALSA: hda/realtek: Add CS35L41 support for Thinkpad laptops MAINTAINERS | 2 + drivers/acpi/scan.c | 3 + drivers/platform/x86/i2c-multi-instantiate.c | 11 + include/sound/cs35l41.h | 739 ++++++++++++++++++ sound/pci/hda/Kconfig | 29 + sound/pci/hda/Makefile | 10 + sound/pci/hda/cs35l41_hda.c | 527 +++++++++++++ sound/pci/hda/cs35l41_hda.h | 69 ++ sound/pci/hda/cs35l41_hda_i2c.c | 66 ++ sound/pci/hda/cs35l41_hda_spi.c | 63 ++ sound/pci/hda/hda_component.h | 20 + sound/pci/hda/patch_realtek.c | 146 ++++ sound/soc/codecs/Kconfig | 11 +- sound/soc/codecs/Makefile | 4 +- sound/soc/codecs/cs35l41-i2c.c | 1 - .../{cs35l41-tables.c => cs35l41-lib.c} | 355 ++++++++- sound/soc/codecs/cs35l41-spi.c | 1 - sound/soc/codecs/cs35l41.c | 360 +-------- sound/soc/codecs/cs35l41.h | 734 ----------------- 19 files changed, 2055 insertions(+), 1096 deletions(-) create mode 100644 sound/pci/hda/cs35l41_hda.c create mode 100644 sound/pci/hda/cs35l41_hda.h create mode 100644 sound/pci/hda/cs35l41_hda_i2c.c create mode 100644 sound/pci/hda/cs35l41_hda_spi.c create mode 100644 sound/pci/hda/hda_component.h rename sound/soc/codecs/{cs35l41-tables.c => cs35l41-lib.c} (71%) -- 2.34.1

11 28

[PATCH 1/2] ALSA: hda: Add Intel DG2 PCI ID and HDMI codec vid
by Kai Vehmanen 14 Oct '22

14 Oct '22

Add HD Audio PCI ID and HDMI codec vendor ID for Intel DG2. Reviewed-by: Uma Shankar <uma.shankar(a)intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/pci/hda/hda_intel.c | 12 +++++++++++- sound/pci/hda/patch_hdmi.c | 1 + 2 files changed, 12 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 221afacbc7fd..b98d7975adea 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -335,7 +335,10 @@ enum { ((pci)->device == 0x0c0c) || \ ((pci)->device == 0x0d0c) || \ ((pci)->device == 0x160c) || \ - ((pci)->device == 0x490d)) + ((pci)->device == 0x490d) || \ + ((pci)->device == 0x4f90) || \ + ((pci)->device == 0x4f91) || \ + ((pci)->device == 0x4f92)) #define IS_BXT(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x5a98) @@ -2477,6 +2480,13 @@ static const struct pci_device_id azx_ids[] = { /* DG1 */ { PCI_DEVICE(0x8086, 0x490d), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* DG2 */ + { PCI_DEVICE(0x8086, 0x4f90), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + { PCI_DEVICE(0x8086, 0x4f91), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + { PCI_DEVICE(0x8086, 0x4f92), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Alderlake-S */ { PCI_DEVICE(0x8086, 0x7ad0), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 65d2c5539919..98633d2684de 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -4382,6 +4382,7 @@ HDA_CODEC_ENTRY(0x80862814, "DG1 HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862815, "Alderlake HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x8086281c, "Alderlake-P HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862816, "Rocketlake HDMI", patch_i915_tgl_hdmi), +HDA_CODEC_ENTRY(0x80862819, "DG2 HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x8086281a, "Jasperlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x8086281b, "Elkhartlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi), base-commit: d7aca8d8f548570f650d9b1dfedf6902f56f2bce -- 2.34.1

3 3

[PATCH][next] ASoC: q6asm: use kcalloc() instead of kzalloc()
by Gustavo A. R. Silva 28 Jul '22

28 Jul '22

Use 2-factor multiplication argument form kcalloc() instead of kzalloc(). Link: https://github.com/KSPP/linux/issues/162 Signed-off-by: Gustavo A. R. Silva <gustavoars(a)kernel.org> --- sound/soc/qcom/qdsp6/q6asm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c index 9251d8548965..195780f75d05 100644 --- a/sound/soc/qcom/qdsp6/q6asm.c +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -513,7 +513,7 @@ int q6asm_map_memory_regions(unsigned int dir, struct audio_client *ac, return 0; } - buf = kzalloc(((sizeof(struct audio_buffer)) * periods), GFP_ATOMIC); + buf = kcalloc(periods, sizeof(*buf), GFP_ATOMIC); if (!buf) { spin_unlock_irqrestore(&ac->lock, flags); return -ENOMEM; -- 2.27.0

3 2

[PATCH] ASoC: ops: Shift tested values in snd_soc_put_volsw() by +min
by Marek Vasut 19 May '22

19 May '22

While the $val/$val2 values passed in from userspace are always >= 0 integers, the limits of the control can be signed integers and the $min can be non-zero and less than zero. To correctly validate $val/$val2 against platform_max, add the $min offset to val first. Fixes: 817f7c9335ec0 ("ASoC: ops: Reject out of bounds values in snd_soc_put_volsw()") Signed-off-by: Marek Vasut <marex(a)denx.de> Cc: Mark Brown <broonie(a)kernel.org> Cc: stable(a)vger.kernel.org --- 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 f24f7354f46fe..6389a512c4dc6 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -317,7 +317,7 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, mask = BIT(sign_bit + 1) - 1; val = ucontrol->value.integer.value[0]; - if (mc->platform_max && val > mc->platform_max) + if (mc->platform_max && ((int)val + min) > mc->platform_max) return -EINVAL; if (val > max - min) return -EINVAL; @@ -330,7 +330,7 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, val = val << shift; if (snd_soc_volsw_is_stereo(mc)) { val2 = ucontrol->value.integer.value[1]; - if (mc->platform_max && val2 > mc->platform_max) + if (mc->platform_max && ((int)val2 + min) > mc->platform_max) return -EINVAL; if (val2 > max - min) return -EINVAL; -- 2.34.1

6 14

[PATCH v4] dt-bindings: dsp: mediatek: add mt8195 dsp document
by YC Hung 29 Apr '22

29 Apr '22

From: "YC Hung" <yc.hung(a)mediatek.com> This patch adds mt8195 dsp document. The dsp is used for Sound Open Firmware driver node. It includes registers, clocks, memory regions, and mailbox for dsp. Signed-off-by: yc.hung <yc.hung(a)mediatek.com> --- Changes since v3: Fix patch v3 error : v3 only provide difference between v3 and v2. Changes since v2: Remove useless watchdog interrupt. Add commit message more detail description. Changes since v1: Rename yaml file name as mediatek,mt8195-dsp.yaml Refine descriptions for mailbox, memory-region and drop unused labels in examples. --- .../bindings/dsp/mediatek,mt8195-dsp.yaml | 105 ++++++++++++++++++ 1 file changed, 105 insertions(+) create mode 100644 Documentation/devicetree/bindings/dsp/mediatek,mt8195-dsp.yaml diff --git a/Documentation/devicetree/bindings/dsp/mediatek,mt8195-dsp.yaml b/Documentation/devicetree/bindings/dsp/mediatek,mt8195-dsp.yaml new file mode 100644 index 000000000000..b7e68b0dfa13 --- /dev/null +++ b/Documentation/devicetree/bindings/dsp/mediatek,mt8195-dsp.yaml @@ -0,0 +1,105 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/dsp/mediatek,mt8195-dsp.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Mediatek mt8195 DSP core + +maintainers: + - YC Hung <yc.hung(a)mediatek.com> + +description: | + Some boards from mt8195 contain a DSP core used for + advanced pre- and post- audio processing. + +properties: + compatible: + const: mediatek,mt8195-dsp + + reg: + items: + - description: Address and size of the DSP Cfg registers + - description: Address and size of the DSP SRAM + + reg-names: + items: + - const: cfg + - const: sram + + clocks: + items: + - description: mux for audio dsp clock + - description: 26M clock + - description: mux for audio dsp local bus + - description: default audio dsp local bus clock source + - description: clock gate for audio dsp clock + - description: mux for audio dsp access external bus + + clock-names: + items: + - const: adsp_sel + - const: clk26m_ck + - const: audio_local_bus + - const: mainpll_d7_d2 + - const: scp_adsp_audiodsp + - const: audio_h + + power-domains: + maxItems: 1 + + mboxes: + items: + - description: ipc reply between host and audio DSP. + - description: ipc request between host and audio DSP. + + mbox-names: + items: + - const: mbox0 + - const: mbox1 + + memory-region: + items: + - description: dma buffer between host and DSP. + - description: DSP system memory. + +required: + - compatible + - reg + - reg-names + - clocks + - clock-names + - memory-region + - power-domains + - mbox-names + - mboxes + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + dsp@10803000 { + compatible = "mediatek,mt8195-dsp"; + reg = <0x10803000 0x1000>, + <0x10840000 0x40000>; + reg-names = "cfg", "sram"; + clocks = <&topckgen 10>, //CLK_TOP_ADSP + <&clk26m>, + <&topckgen 107>, //CLK_TOP_AUDIO_LOCAL_BUS + <&topckgen 136>, //CLK_TOP_MAINPLL_D7_D2 + <&scp_adsp 0>, //CLK_SCP_ADSP_AUDIODSP + <&topckgen 34>; //CLK_TOP_AUDIO_H + clock-names = "adsp_sel", + "clk26m_ck", + "audio_local_bus", + "mainpll_d7_d2", + "scp_adsp_audiodsp", + "audio_h"; + memory-region = <&adsp_dma_mem_reserved>, + <&adsp_mem_reserved>; + power-domains = <&spm 6>; //MT8195_POWER_DOMAIN_ADSP + mbox-names = "mbox0", "mbox1"; + mboxes = <&adsp_mailbox0>, <&adsp_mailbox1>; + }; -- 2.18.0

4 6

[PATCH] ALSA: usb-audio: Always apply the hw constraints for implicit fb sync
by Takashi Iwai 22 Apr '22

22 Apr '22

Since the commit 5a6c3e11c9c9 ("ALSA: usb-audio: Add hw constraint for implicit fb sync"), we apply the hw constraints for the implicit feedback sync to make the secondary open aligned with the already opened stream setup. This change assumed that the secondary open is performed after the first stream has been already set up, and adds the hw constraints to sync with the first stream's parameters only when the EP setup for the first stream was confirmed at the open time. However, most of applications handling the full-duplex operations do open both playback and capture streams at first, then set up both streams. This results in skipping the additional hw constraints since the counter-part stream hasn't been set up yet at the open of the second stream, and it eventually leads to "incompatible EP" error in the end. This patch corrects the behavior by always applying the hw constraints for the implicit fb sync. The hw constraint rules are defined so that they check the sync EP dynamically at each invocation, instead. This covers the concurrent stream setups better and lets the hw refine calls resolving to the right configuration. Also this patch corrects a minor error that has existed in the debug print that isn't built as default. Fixes: 5a6c3e11c9c9 ("ALSA: usb-audio: Add hw constraint for implicit fb sync") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- This patched was developed primarily as an attemp for the regression of Pioneer devices. Although Pioneer devices have still some issues after this patch, this one itself should be good for all devices in general. Keith, Dylan, could you verify that this doesn't break things for your devices? sound/usb/pcm.c | 171 ++++++++++++++++++++++++++++++------------------ 1 file changed, 108 insertions(+), 63 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 56079901769f..f71965bf815f 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -663,7 +663,7 @@ static int hw_check_valid_format(struct snd_usb_substream *subs, check_fmts.bits[1] = (u32)(fp->formats >> 32); snd_mask_intersect(&check_fmts, fmts); if (snd_mask_empty(&check_fmts)) { - hwc_debug(" > check: no supported format %d\n", fp->format); + hwc_debug(" > check: no supported format 0x%llx\n", fp->formats); return 0; } /* check the channels */ @@ -775,24 +775,11 @@ static int hw_rule_channels(struct snd_pcm_hw_params *params, return apply_hw_params_minmax(it, rmin, rmax); } -static int hw_rule_format(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) +static int apply_hw_params_format_bits(struct snd_mask *fmt, u64 fbits) { - struct snd_usb_substream *subs = rule->private; - const struct audioformat *fp; - struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - u64 fbits; u32 oldbits[2]; int changed; - hwc_debug("hw_rule_format: %x:%x\n", fmt->bits[0], fmt->bits[1]); - fbits = 0; - list_for_each_entry(fp, &subs->fmt_list, list) { - if (!hw_check_valid_format(subs, params, fp)) - continue; - fbits |= fp->formats; - } - oldbits[0] = fmt->bits[0]; oldbits[1] = fmt->bits[1]; fmt->bits[0] &= (u32)fbits; @@ -806,6 +793,24 @@ static int hw_rule_format(struct snd_pcm_hw_params *params, return changed; } +static int hw_rule_format(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_usb_substream *subs = rule->private; + const struct audioformat *fp; + struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + u64 fbits; + + hwc_debug("hw_rule_format: %x:%x\n", fmt->bits[0], fmt->bits[1]); + fbits = 0; + list_for_each_entry(fp, &subs->fmt_list, list) { + if (!hw_check_valid_format(subs, params, fp)) + continue; + fbits |= fp->formats; + } + return apply_hw_params_format_bits(fmt, fbits); +} + static int hw_rule_period_time(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { @@ -833,64 +838,92 @@ static int hw_rule_period_time(struct snd_pcm_hw_params *params, return apply_hw_params_minmax(it, pmin, UINT_MAX); } -/* apply PCM hw constraints from the concurrent sync EP */ -static int apply_hw_constraint_from_sync(struct snd_pcm_runtime *runtime, - struct snd_usb_substream *subs) +/* get the EP or the sync EP for implicit fb when it's already set up */ +static const struct snd_usb_endpoint * +get_sync_ep_from_substream(struct snd_usb_substream *subs) { struct snd_usb_audio *chip = subs->stream->chip; - struct snd_usb_endpoint *ep; const struct audioformat *fp; - int err; + const struct snd_usb_endpoint *ep; list_for_each_entry(fp, &subs->fmt_list, list) { ep = snd_usb_get_endpoint(chip, fp->endpoint); if (ep && ep->cur_rate) - goto found; + return ep; if (!fp->implicit_fb) continue; /* for the implicit fb, check the sync ep as well */ ep = snd_usb_get_endpoint(chip, fp->sync_ep); if (ep && ep->cur_rate) - goto found; + return ep; } - return 0; + return NULL; +} + +/* additional hw constraints for implicit feedback mode */ +static int hw_rule_format_implicit_fb(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_usb_substream *subs = rule->private; + const struct snd_usb_endpoint *ep; + struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - found: - if (!find_format(&subs->fmt_list, ep->cur_format, ep->cur_rate, - ep->cur_channels, false, NULL)) { - usb_audio_dbg(chip, "EP 0x%x being used, but not applicable\n", - ep->ep_num); + ep = get_sync_ep_from_substream(subs); + if (!ep) return 0; - } - usb_audio_dbg(chip, "EP 0x%x being used, using fixed params:\n", - ep->ep_num); - usb_audio_dbg(chip, "rate=%d, period_size=%d, periods=%d\n", - ep->cur_rate, ep->cur_period_frames, - ep->cur_buffer_periods); + hwc_debug("applying %s\n", __func__); + return apply_hw_params_format_bits(fmt, pcm_format_to_bits(ep->cur_format)); +} - runtime->hw.formats = subs->formats; - runtime->hw.rate_min = runtime->hw.rate_max = ep->cur_rate; - runtime->hw.rates = SNDRV_PCM_RATE_KNOT; - runtime->hw.periods_min = runtime->hw.periods_max = - ep->cur_buffer_periods; +static int hw_rule_rate_implicit_fb(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_usb_substream *subs = rule->private; + const struct snd_usb_endpoint *ep; + struct snd_interval *it; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - hw_rule_channels, subs, - SNDRV_PCM_HW_PARAM_FORMAT, - SNDRV_PCM_HW_PARAM_RATE, - -1); - if (err < 0) - return err; + ep = get_sync_ep_from_substream(subs); + if (!ep) + return 0; - err = snd_pcm_hw_constraint_minmax(runtime, - SNDRV_PCM_HW_PARAM_PERIOD_SIZE, - ep->cur_period_frames, - ep->cur_period_frames); - if (err < 0) - return err; + hwc_debug("applying %s\n", __func__); + it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + return apply_hw_params_minmax(it, ep->cur_rate, ep->cur_rate); +} + +static int hw_rule_period_size_implicit_fb(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_usb_substream *subs = rule->private; + const struct snd_usb_endpoint *ep; + struct snd_interval *it; - return 1; /* notify the finding */ + ep = get_sync_ep_from_substream(subs); + if (!ep) + return 0; + + hwc_debug("applying %s\n", __func__); + it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE); + return apply_hw_params_minmax(it, ep->cur_period_frames, + ep->cur_period_frames); +} + +static int hw_rule_periods_implicit_fb(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_usb_substream *subs = rule->private; + const struct snd_usb_endpoint *ep; + struct snd_interval *it; + + ep = get_sync_ep_from_substream(subs); + if (!ep) + return 0; + + hwc_debug("applying %s\n", __func__); + it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_PERIODS); + return apply_hw_params_minmax(it, ep->cur_buffer_periods, + ep->cur_buffer_periods); } /* @@ -899,20 +932,11 @@ static int apply_hw_constraint_from_sync(struct snd_pcm_runtime *runtime, static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substream *subs) { - struct snd_usb_audio *chip = subs->stream->chip; const struct audioformat *fp; unsigned int pt, ptmin; int param_period_time_if_needed = -1; int err; - mutex_lock(&chip->mutex); - err = apply_hw_constraint_from_sync(runtime, subs); - mutex_unlock(&chip->mutex); - if (err < 0) - return err; - if (err > 0) /* found the matching? */ - goto add_extra_rules; - runtime->hw.formats = subs->formats; runtime->hw.rate_min = 0x7fffffff; @@ -964,7 +988,6 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre if (err < 0) return err; -add_extra_rules: err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, hw_rule_channels, subs, SNDRV_PCM_HW_PARAM_FORMAT, @@ -993,6 +1016,28 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre return err; } + /* additional hw constraints for implicit fb */ + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, + hw_rule_format_implicit_fb, subs, + SNDRV_PCM_HW_PARAM_FORMAT, -1); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + hw_rule_rate_implicit_fb, subs, + SNDRV_PCM_HW_PARAM_RATE, -1); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + hw_rule_period_size_implicit_fb, subs, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIODS, + hw_rule_periods_implicit_fb, subs, + SNDRV_PCM_HW_PARAM_PERIODS, -1); + if (err < 0) + return err; + return 0; } -- 2.26.2

2 1

[PATCH v4] ASoC: codecs: wsa881x: add runtime pm support
by Srinivas Kandagatla 21 Apr '22

21 Apr '22

WSA SoundWire Controller does not support Clock stop and performs a soft reset on suspend resume path. Its recommended that WSA881x codecs connected to this are also reset using a hard reset during suspend resume. So this codec driver performs a hard reset during suspend resume cycle. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- Changes since v3: - added missing sign off. - no code changes sound/soc/codecs/wsa881x.c | 53 ++++++++++++++++++++++++++++++++++++++ 1 file changed, 53 insertions(+) diff --git a/sound/soc/codecs/wsa881x.c b/sound/soc/codecs/wsa881x.c index 0222370ff95d..616b26c70c3b 100644 --- a/sound/soc/codecs/wsa881x.c +++ b/sound/soc/codecs/wsa881x.c @@ -11,6 +11,7 @@ #include <linux/of_gpio.h> #include <linux/regmap.h> #include <linux/slab.h> +#include <linux/pm_runtime.h> #include <linux/soundwire/sdw.h> #include <linux/soundwire/sdw_registers.h> #include <linux/soundwire/sdw_type.h> @@ -198,6 +199,7 @@ #define WSA881X_OCP_CTL_TIMER_SEC 2 #define WSA881X_OCP_CTL_TEMP_CELSIUS 25 #define WSA881X_OCP_CTL_POLL_TIMER_SEC 60 +#define WSA881X_PROBE_TIMEOUT 1000 #define WSA881X_PA_GAIN_TLV(xname, reg, shift, max, invert, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ @@ -747,6 +749,12 @@ static int wsa881x_put_pa_gain(struct snd_kcontrol *kc, unsigned int mask = (1 << fls(max)) - 1; int val, ret, min_gain, max_gain; + ret = pm_runtime_get_sync(comp->dev); + if (ret < 0 && ret != -EACCES) { + pm_runtime_put_noidle(comp->dev); + return ret; + } + max_gain = (max - ucontrol->value.integer.value[0]) & mask; /* * Gain has to set incrementally in 4 steps @@ -773,6 +781,9 @@ static int wsa881x_put_pa_gain(struct snd_kcontrol *kc, usleep_range(1000, 1010); } + pm_runtime_mark_last_busy(comp->dev); + pm_runtime_put_autosuspend(comp->dev); + return 1; } @@ -1101,6 +1112,7 @@ static int wsa881x_probe(struct sdw_slave *pdev, const struct sdw_device_id *id) { struct wsa881x_priv *wsa881x; + struct device *dev = &pdev->dev; wsa881x = devm_kzalloc(&pdev->dev, sizeof(*wsa881x), GFP_KERNEL); if (!wsa881x) @@ -1132,12 +1144,52 @@ static int wsa881x_probe(struct sdw_slave *pdev, return PTR_ERR(wsa881x->regmap); } + pm_runtime_set_autosuspend_delay(dev, 3000); + pm_runtime_use_autosuspend(dev); + pm_runtime_mark_last_busy(dev); + pm_runtime_set_active(dev); + pm_runtime_enable(dev); + return devm_snd_soc_register_component(&pdev->dev, &wsa881x_component_drv, wsa881x_dais, ARRAY_SIZE(wsa881x_dais)); } +static int __maybe_unused wsa881x_runtime_suspend(struct device *dev) +{ + struct regmap *regmap = dev_get_regmap(dev, NULL); + struct wsa881x_priv *wsa881x = dev_get_drvdata(dev); + + gpiod_direction_output(wsa881x->sd_n, 0); + + regcache_cache_only(regmap, true); + regcache_mark_dirty(regmap); + + return 0; +} + +static int __maybe_unused wsa881x_runtime_resume(struct device *dev) +{ + struct sdw_slave *slave = dev_to_sdw_dev(dev); + struct regmap *regmap = dev_get_regmap(dev, NULL); + struct wsa881x_priv *wsa881x = dev_get_drvdata(dev); + + gpiod_direction_output(wsa881x->sd_n, 1); + + wait_for_completion_timeout(&slave->initialization_complete, + msecs_to_jiffies(WSA881X_PROBE_TIMEOUT)); + + regcache_cache_only(regmap, false); + regcache_sync(regmap); + + return 0; +} + +static const struct dev_pm_ops wsa881x_pm_ops = { + SET_RUNTIME_PM_OPS(wsa881x_runtime_suspend, wsa881x_runtime_resume, NULL) +}; + static const struct sdw_device_id wsa881x_slave_id[] = { SDW_SLAVE_ENTRY(0x0217, 0x2010, 0), SDW_SLAVE_ENTRY(0x0217, 0x2110, 0), @@ -1151,6 +1203,7 @@ static struct sdw_driver wsa881x_codec_driver = { .id_table = wsa881x_slave_id, .driver = { .name = "wsa881x-codec", + .pm = &wsa881x_pm_ops, } }; module_sdw_driver(wsa881x_codec_driver); -- 2.21.0

3 3

[PATCH v3 0/3] soundwire: qcom: add pm runtime support
by Srinivas Kandagatla 20 Apr '22

20 Apr '22

This patchset adds pm runtime support to Qualcomm SounWire Controller using SoundWire Clock Stop and Wake up using Headset events on supported instances and instances like WSA which do not support clock stop a soft reset of controller along with full rest of slaves is done to resume from a low power state. Tested it on SM8250 MTP and Dragon Board DB845c Changes since v2: - update log as suggested by Pierre - removing handling clk stop for cases where the controller is soft reset. - add more error checks when calling sdw_bus_prep_clk_stop - update dt-bindings with wakeup-source and interrupt-names properties. --srini Srinivas Kandagatla (3): soundwire: qcom: add runtime pm support dt-bindings: soundwire: qcom: document optional wake irq soundwire: qcom: add in-band wake up interrupt support .../bindings/soundwire/qcom,sdw.txt | 14 +- drivers/soundwire/qcom.c | 206 +++++++++++++++++- 2 files changed, 218 insertions(+), 2 deletions(-) -- 2.21.0

5 15

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Alsa-devel February 2022