March 2022 - Alsa-devel - mailman.alsa-project.org

[PATCH v2 00/14] ASoC: Intel: avs: Topology and path management
by Cezary Rojewski 10 Jun '22

10 Jun '22

A continuation of avs-driver initial series [1]. This chapter covers path management and topology parsing part which was ealier path of the main series. The two patches that represented these two subjects in the initial series, have been split into many to allow for easier review and discussion. AVS topology is split into two major parts: dictionaries - found within ASoC topology manifest - and path templates. Dictionaries job is to reduce the total amount of memory occupied by topology elements. Rather than having every pipeline and module carry its own information, each refers to specific entry in specific dictionary by provided (from topology file) indexes. In consequence, most struct avs_tplg_xxx are made out of pointers. Path templates are similar to path descriptions found in skylake-driver and they describe how given path shall look like in runtime - number of modules and pipelines that shape it and how they are laid out. A single path template is tied either to FE or BE and thus at most to a single, user-visible endpoint when speaking of FE. Path is a software representation of its ADSP firmware equivalent. It's a logical container for pipelines which are themselves containers - this time for modules i.e. processing units. Depending on the number of audio formats supported, each path template may carry one or more descriptions of given path. During runtime, when audio format is known, description matching said format is selected and used when instantiating path on ADSP firmware side through IPCs. Changes v1 -> v2: - fixed 'set but not used' warning in patch 10/14 as mentioned by ikp builder Changes RFC -> v1: - Plenty of rewordings and spelling fixes as requested by Pierre - any sysfs-functionality-related left in RFC dropped, will be part of separate subject in the future - separated parsing of formatted strings into: "Support link_mask formatted string" patch, increasing series size to 14 patches [1]: https://lore.kernel.org/alsa-devel/20220207122108.3780926-1-cezary.rojewski… Cezary Rojewski (14): ASoC: Intel: avs: Declare vendor tokens ASoC: Intel: avs: Add topology parsing infrastructure ASoC: Intel: avs: Parse module-extension tuples ASoC: Intel: avs: Parse pplcfg and binding tuples ASoC: Intel: avs: Parse pipeline and module tuples ASoC: Intel: avs: Parse path and path templates tuples ASoC: Intel: avs: Add topology loading operations ASoC: Intel: avs: Support link_mask formatted string ASoC: Intel: avs: Declare path and its components ASoC: Intel: avs: Path creation and freeing ASoC: Intel: avs: Path state management ASoC: Intel: avs: Arm paths after creating them ASoC: Intel: avs: Prepare modules before bindings them ASoC: Intel: avs: Configure modules according to their type include/uapi/sound/intel/avs/tokens.h | 126 ++ sound/soc/intel/Kconfig | 2 + sound/soc/intel/avs/Makefile | 3 +- sound/soc/intel/avs/avs.h | 23 + sound/soc/intel/avs/path.c | 1005 ++++++++++++++++ sound/soc/intel/avs/path.h | 72 ++ sound/soc/intel/avs/topology.c | 1600 +++++++++++++++++++++++++ sound/soc/intel/avs/topology.h | 194 +++ 8 files changed, 3024 insertions(+), 1 deletion(-) create mode 100644 include/uapi/sound/intel/avs/tokens.h create mode 100644 sound/soc/intel/avs/path.c create mode 100644 sound/soc/intel/avs/path.h create mode 100644 sound/soc/intel/avs/topology.c create mode 100644 sound/soc/intel/avs/topology.h base-commit: bc55cfd5718c7c23e5524582e9fa70b4d10f2433 -- 2.25.1

3 19

[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 v3 0/6] Add driver for SAMA7G5's PDMC
by Codrin Ciubotariu 05 May '22

05 May '22

This patch series adds support for Pulse Density Microphone Controller (PDMC), present on Microchip's SAMA7G5. The PDMC interfaces up to 4 digital microphones having Pulse Density Modulated (PDM) outputs. It generates a single clock line and samples 1 or 2 data lines. The signal path includes an audio grade programmable decimation filter and outputs 24-bit audio words. The source of each channel can be independently defined as PDMC_DS0 or PDMC_DS1, sampled at the rising or falling edge of PDMC_CLK. The patch series starts with a fix on the ASoC DMA engine support. Then continues with the bindings and the driver of PDMC. It is followed by the DT nodes for SAMA7G5 and SAMA7G5-EK. In the end, the drivers for PDMC and PDM microphones are enabled in sama7_defconfig. Changes in v3: - addressed new DT bindings comments from Krzysztof Kozlowski Changes in v2: - addressed DT bindings comments from Krzysztof Kozlowski Codrin Ciubotariu (6): ASoC: dmaengine: do not use a NULL prepare_slave_config() callback ASoC: dt-bindings: Document Microchip's PDMC ASoC: atmel: mchp-pdmc: add PDMC driver ARM: dts: at91: sama7g5: add nodes for PDMC ARM: dts: at91: sama7g5ek: add node for PDMC0 ARM: configs: at91: sama7_defconfig: add MCHP PDMC and DMIC drivers .../bindings/sound/microchip,pdmc.yaml | 100 ++ arch/arm/boot/dts/at91-sama7g5ek.dts | 21 +- arch/arm/boot/dts/sama7g5.dtsi | 24 + arch/arm/configs/sama7_defconfig | 2 + include/dt-bindings/sound/microchip,pdmc.h | 13 + sound/soc/atmel/Kconfig | 16 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/mchp-pdmc.c | 1084 +++++++++++++++++ sound/soc/soc-generic-dmaengine-pcm.c | 6 +- 9 files changed, 1264 insertions(+), 4 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/microchip,pdmc.yaml create mode 100644 include/dt-bindings/sound/microchip,pdmc.h create mode 100644 sound/soc/atmel/mchp-pdmc.c -- 2.32.0

6 16

[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

[RFC PATCH 0/5] Apple Macs machine-level ASoC driver
by Martin Povišer 25 Apr '22

25 Apr '22

Hi, I put together a machine-level ASoC driver for recent Apple Macs (the ones with ARM64 SoCs) and want to gauge opinions. Commit 1 is the binding. It is some subset of simple-audio-card with the extra distinction of allowing multiple CPU/CODEC DAIs per a DAI link. I want to draw special attention to the issue of describing speaker topologies. The way it now works is that the driver expects the speakers to be declared in a fixed order in the sound-dai= list. This populates a topology the driver expects on a particular machine model. Mark (in CC) has made the suggestion of keeping the topology descriptions with the codec nodes themselves in some generic manner, akin to how sound-name-prefix= already helps identify codecs to the user. Commit 2 adds a new ASoC card method (filter_controls) to let the card prevent some codec kcontrols from being visible to userspace. For example the TAS2770 speaker amp driver would be happy to expose TDM slot selection and ISENSE/VSENSE enables which is ridiculous. I am all ears on how to make the patch acceptable to upstream. Commit 3 makes ASoC tolerate N-to-M DAI links, not sure what the right (simple) approach should be there. Commit 4 adds some utility function and commit 5 is the driver itself. Let me know what you think. Martin Martin Povišer (5): dt-bindings: sound: Add Apple Macs sound system HACK: ASoC: Add card->filter_controls hook HACK: ASoC: Tolerate N-cpus-to-M-codecs links ASoC: Introduce snd_soc_of_get_dai_link_cpus ASoC: Add macaudio machine driver .../bindings/sound/apple,macaudio.yaml | 103 +++ include/sound/soc.h | 7 + sound/soc/apple/Kconfig | 10 + sound/soc/apple/Makefile | 3 + sound/soc/apple/macaudio.c | 597 ++++++++++++++++++ sound/soc/soc-core.c | 125 +++- sound/soc/soc-dapm.c | 34 +- sound/soc/soc-pcm.c | 3 + 8 files changed, 860 insertions(+), 22 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/apple,macaudio.yaml create mode 100644 sound/soc/apple/Kconfig create mode 100644 sound/soc/apple/Makefile create mode 100644 sound/soc/apple/macaudio.c -- 2.33.0

4 19

Re: [RFC PATCH 0/5] Apple Macs machine-level ASoC driver
by Mark Brown 22 Apr '22

22 Apr '22

On Thu, Mar 31, 2022 at 05:04:32PM +0200, Martin Povišer wrote: > > On 31. 3. 2022, at 16:18, Mark Brown <broonie(a)kernel.org> wrote: > > Yes, having two devices driving the bus at the same time wouldn't be > > great. How is the TDM slot selection for the signals done in the > > hardware, I'm not seeing anything immediately obvious in the driver? > > I'd have thought that things would be implemented such that you could > > implement speaker protection on all speakers simultaneously but perhaps > > not. > I don’t know. I would have to go study the details of this. Should I see > if I can find a combination of ‘ASI1 Sel’ ‘VSENSE’ ‘ISENSE’ settings > that would lead to driver conflict on one of the models, or is there > a chance we could hide those controls just on the basis of ‘it doesn’t > do anything usable and is possibly dangerous’? If ISENSE and VSENSE output are controlled by the same mux as routing then we should lock one of the controls out for at least stereo devices (it might be a good idea to check if the output is actually high Z when ISENSE and VSENSE are off rather than just driving zeros, if not it definitely has to be the routing control). My instinct is that it's better to preserve the ability to implement speaker protection in future since that is something that'd be broadly useful, especially if someone comes up with a generic speaker protection implementation in which case there should be an awful lot of systems out there which could benefit. > >> That’s the reasoning anyway. To reiterate, seems to me the controls > >> are useless/confusing at best and dangerous at worst. > > I'm just not seeing an issue for the slot selection. > Yeah, agreed there’s no (damage) issue as we should to proper volume > caps anyway. Though see above about how ISENSE/VSENSE output slot is controlled I guess :/

2 11

[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

[V2 1/2] ASoC: max98390: Add reset gpio control
by Steve Lee 21 Apr '22

21 Apr '22

Add reset gpio control to support RESET PIN connected to gpio. Signed-off-by: Steve Lee <steve.lee.analog(a)gmail.com> --- sound/soc/codecs/max98390.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c index b392567c2b3e..5c6162701667 100644 --- a/sound/soc/codecs/max98390.c +++ b/sound/soc/codecs/max98390.c @@ -1022,6 +1022,7 @@ static int max98390_i2c_probe(struct i2c_client *i2c, struct max98390_priv *max98390 = NULL; struct i2c_adapter *adapter = to_i2c_adapter(i2c->dev.parent); + struct gpio_desc *reset_gpio; ret = i2c_check_functionality(adapter, I2C_FUNC_SMBUS_BYTE @@ -1073,6 +1074,17 @@ static int max98390_i2c_probe(struct i2c_client *i2c, return ret; } + reset_gpio = devm_gpiod_get_optional(&i2c->dev, + "maxim,reset-gpios", GPIOD_OUT_LOW); + + /* Power on device */ + if (reset_gpio) { + usleep_range(1000, 2000); + /* bring out of reset */ + gpiod_set_value_cansleep(reset_gpio, 1); + usleep_range(1000, 2000); + } + /* Check Revision ID */ ret = regmap_read(max98390->regmap, MAX98390_R24FF_REV_ID, &reg); -- 2.17.1

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[PATCH v12 0/7] Add pin control support for lpass sc7280
by Srinivasa Rao Mandadapu 21 Apr '22

21 Apr '22

This patch series is to split lpass variant common pin control functions and SoC specific functions and to add lpass sc7280 pincontrol support. It also Adds dt-bindings for lpass sc7280 lpass lpi pincontrol. Changes Since V11: -- Add pinctrl_generic_remove_group in lpass lpi driver. -- Make proper error handling in lpass lpi driver. Changes Since V10: -- Modify driver's custom functions with pin control framework generic functions. -- Update sm8250 and sc7280 pin control depedency list in Kconfig. -- Update commit description of few patches. Changes Since V9: -- Add pinctrl groups macro to Kconfig. Changes Since V8: -- Remove redundant headers included in v8. Changes Since V7: -- Update optional clock voting with conditional check. -- Add const to lpi_pinctrl_variant_data structure. -- Update required headers and remove redundant. -- Change EXPORT_SYMBOL to EXPORT_SYMBOL_GPL -- Fix typo errors. Changes Since V6: -- Update conditional clock voting to optional clock voting. -- Update Kconfig depends on field with select. -- Fix typo errors. Changes Since V5: -- Create new patch by updating macro name to lpi specific. -- Create new patch by updating lpi pin group structure with core group_desc structure. -- Fix typo errors. -- Sort macros in the make file and configuration file. Changes Since V4: -- Update commit message and description of the chip specific extraction patch. -- Sort macros in kconfig and makefile. -- Update optional clock voting to conditional clock voting. -- Fix typo errors. -- Move to quicinc domain email id's. Changes Since V3: -- Update separate Kconfig fields for sm8250 and sc7280. -- Update module license and description. -- Move static variables to corresponding .c files from header file. Changes Since V2: -- Add new dt-bindings for sc7280 lpi driver. -- Make clock voting change as separate patch. -- Split existing pincontrol driver and make common functions as part of separate file. -- Rename lpass pincontrol lpi dt-bindings to sm8250 specific dt-bindings Changes Since V1: -- Make lpi pinctrl variant data structure as constant -- Add appropriate commit message -- Change signedoff by sequence. Srinivasa Rao Mandadapu (7): dt-bindings: pinctrl: qcom: Update lpass lpi file name to SoC specific dt-bindings: pinctrl: qcom: Add sc7280 lpass lpi pinctrl bindings pinctrl: qcom: Update macro name to LPI specific pinctrl: qcom: Update lpi pin group custiom functions with framework generic functions pinctrl: qcom: Extract chip specific LPASS LPI code pinctrl: qcom: Add SC7280 lpass pin configuration pinctrl: qcom: Update clock voting as optional Tested this on SM8250 MTP with WSA and WCD codecs. Tested-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> .../bindings/pinctrl/qcom,lpass-lpi-pinctrl.yaml | 133 --------- .../pinctrl/qcom,sc7280-lpass-lpi-pinctrl.yaml | 115 ++++++++ .../pinctrl/qcom,sm8250-lpass-lpi-pinctrl.yaml | 133 +++++++++ drivers/pinctrl/qcom/Kconfig | 19 ++ drivers/pinctrl/qcom/Makefile | 2 + drivers/pinctrl/qcom/pinctrl-lpass-lpi.c | 309 +++------------------ drivers/pinctrl/qcom/pinctrl-lpass-lpi.h | 86 ++++++ drivers/pinctrl/qcom/pinctrl-sc7280-lpass-lpi.c | 168 +++++++++++ drivers/pinctrl/qcom/pinctrl-sm8250-lpass-lpi.c | 164 +++++++++++ 9 files changed, 732 insertions(+), 397 deletions(-) delete mode 100644 Documentation/devicetree/bindings/pinctrl/qcom,lpass-lpi-pinctrl.yaml create mode 100644 Documentation/devicetree/bindings/pinctrl/qcom,sc7280-lpass-lpi-pinctrl.yaml create mode 100644 Documentation/devicetree/bindings/pinctrl/qcom,sm8250-lpass-lpi-pinctrl.yaml create mode 100644 drivers/pinctrl/qcom/pinctrl-lpass-lpi.h create mode 100644 drivers/pinctrl/qcom/pinctrl-sc7280-lpass-lpi.c create mode 100644 drivers/pinctrl/qcom/pinctrl-sm8250-lpass-lpi.c -- 2.7.4

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[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

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