September 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH v8 00/22] ASoC: qcom: Add AudioReach support
by Srinivas Kandagatla 28 Sep '21

28 Sep '21

Many thanks for reviewing v7 This version addresses all the comments raised as part of v8 review. This patchset adds ASoC driver support to configure signal processing framework ("AudioReach") which is integral part of Qualcomm next generation audio SDK and will be deployed on upcoming Qualcomm chipsets. It makes use of ASoC Topology to load graphs on to the DSP which is then managed by APM (Audio Processing Manager) service to prepare/start/stop. Here is simplified high-level block diagram of AudioReach: ___________________________________________________________ | CPU (Application Processor) | | +---------+ +---------+ +----------+ | | | q6apm | | q6apm | | q6apm | | | | dais | <------> | | <-----> |lpass-dais| | | +---------+ +---------+ +----------+ | | ^ ^ | | | | +---------+ | | +---------+ v +---------->|topology | | | | q6prm | +---------+ | | | | | |<-------->| GPR | +---------+ | | +---------+ +---------+ | | ^ ^ | | | | | | +----------+ | | | | q6prm | | | | |lpass-clks| | | | +----------+ | | |____________________________|______________________________| | | RPMSG (IPC over GLINK) ____________________________|______________________________ | | | | +-----------------------+ | | | | | | v v q6 (Audio DSP) | |+-----+ +----------------------------------+ | || PRM | | APM (Audio Processing Manager) | | |+-----+ | . Graph Management | | | | . Command Handing | | | | . Event Management | | | | ... | | | +----------------------------------+ | | ^ | |____________________________|______________________________| | | LPASS AIF ____________________________|______________________________ | | Audio I/O | | v | | +--------------------------------------------------+ | | | Audio devices | | | | CODEC | HDMI-TX | PCM | SLIMBUS | I2S |MI2S |...| | | | | | | +--------------------------------------------------+ | |___________________________________________________________| AudioReach has constructs of sub-graph, container and modules. Each sub-graph can have N containers and each Container can have N Modules and connections between them can be linear or non-linear. An audio function can be realized with one or many connected sub-graphs. There are also control/event paths between modules that can be wired up while building graph to achieve various control mechanism between modules. These concepts of Sub-Graph, Containers and Modules are represented in ASoC topology. Here is simple I2S graph with a Write Shared Memory and a Volume control module within a single Subgraph (1) with one Container (1) and 5 modules. ____________________________________________________________ | Sub-Graph [1] | | _______________________________________________________ | | | Container [1] | | | | [WR_SH] -> [PCM DEC] -> [PCM CONV] -> [VOL]-> [I2S-EP]| | | |_______________________________________________________| | |____________________________________________________________| For now this graph is split into two subgraphs to achieve dpcm like below: ________________________________________________ _________________ | Sub-Graph [1] | | Sub-Graph [2] | | ____________________________________________ | | _____________ | | | Container [1] | | | |Container [2]| | | | [WR_SH] -> [PCM DEC] -> [PCM CONV] -> [VOL]| | | | [I2S-EP] | | | |____________________________________________| | | |_____________| | |________________________________________________| |_________________| _________________ | Sub-Graph [3] | | _____________ | | |Container [3]| | | | [DMA-EP] | | | |_____________| | |_________________| This patchset adds very minimal support for AudioReach which includes supporting sub-graphs containing CODEC DMA ports and simple PCM Decoder/Encoder and Logger Modules. Additional capabilities will be built over time to expose features offered by AudioReach. This patchset is Tested on SM8250 SoC based Qualcomm Robotics Platform RB5 and SM9250 MTP with WSA881X Smart Speaker Amplifiers, DMICs connected via VA Macro and WCD938x Codec connected via TX and RX Macro and HDMI audio via I2S. First 10 Patches are mostly reorganization existing Old QDSP Audio Framework code and bindings so that we could reuse them on AudioReach. ASoC topology graphs for DragonBoard RB5 and SM8250 MTP are available at https://git.linaro.org/people/srinivas.kandagatla/audioreach-topology.git/ and Qualcomm AudioReach DSP headers are available at: https://source.codeaurora.org/quic/la/platform/vendor/opensource/arspf-head… Note: There is one false positive warning in this patchset: audioreach.c:80:45: warning: array of flexible structures Thanks, srini Changes since v7: - removed returning errors in dsp command callbacks as it has no effect - removed uncessary err prints - removed default topology file as its impossible to have one for all boards. - removed references of Elite in commit log - return correct error codes returned from functions. Srinivas Kandagatla (22): soc: dt-bindings: qcom: apr: convert to yaml soc: dt-bindings: qcom: apr: deprecate qcom,apr-domain property soc: qcom: apr: make code more reuseable soc: dt-bindings: qcom: add gpr bindings soc: qcom: apr: Add GPR support ASoC: dt-bindings: move LPASS dai related bindings out of q6afe ASoC: dt-bindings: move LPASS clocks related bindings out of q6afe ASoC: dt-bindings: rename q6afe.h to q6dsp-lpass-ports.h ASoC: qdsp6: q6afe-dai: move lpass audio ports to common file ASoC: qdsp6: q6afe-clocks: move audio-clocks to common file ASoC: dt-bindings: q6dsp: add q6apm-lpass-dai compatible ASoC: dt-bindings: lpass-clocks: add q6prm clocks compatible ASoC: dt-bindings: add q6apm digital audio stream bindings ASoC: qdsp6: audioreach: add basic pkt alloc support ASoC: qdsp6: audioreach: add q6apm support ASoC: qdsp6: audioreach: add module configuration command helpers ASoC: qdsp6: audioreach: add Kconfig and Makefile ASoC: qdsp6: audioreach: add topology support ASoC: qdsp6: audioreach: add q6apm-dai support ASoC: qdsp6: audioreach: add q6apm lpass dai support ASoC: qdsp6: audioreach: add q6prm support ASoC: qdsp6: audioreach: add support for q6prm-clocks .../devicetree/bindings/soc/qcom/qcom,apr.txt | 134 -- .../bindings/soc/qcom/qcom,apr.yaml | 177 +++ .../devicetree/bindings/sound/qcom,q6afe.txt | 181 --- .../bindings/sound/qcom,q6apm-dai.yaml | 53 + .../sound/qcom,q6dsp-lpass-clocks.yaml | 77 ++ .../sound/qcom,q6dsp-lpass-ports.yaml | 205 +++ drivers/soc/qcom/Kconfig | 2 +- drivers/soc/qcom/apr.c | 287 ++++- include/dt-bindings/soc/qcom,gpr.h | 19 + include/dt-bindings/sound/qcom,q6afe.h | 203 +-- .../sound/qcom,q6dsp-lpass-ports.h | 208 +++ include/linux/soc/qcom/apr.h | 70 +- include/uapi/sound/snd_ar_tokens.h | 208 +++ sound/soc/qcom/Kconfig | 22 + sound/soc/qcom/qdsp6/Makefile | 11 +- sound/soc/qcom/qdsp6/audioreach.c | 1135 +++++++++++++++++ sound/soc/qcom/qdsp6/audioreach.h | 726 +++++++++++ sound/soc/qcom/qdsp6/q6afe-clocks.c | 187 +-- sound/soc/qcom/qdsp6/q6afe-dai.c | 687 +--------- sound/soc/qcom/qdsp6/q6apm-dai.c | 415 ++++++ sound/soc/qcom/qdsp6/q6apm-lpass-dais.c | 260 ++++ sound/soc/qcom/qdsp6/q6apm.c | 823 ++++++++++++ sound/soc/qcom/qdsp6/q6apm.h | 152 +++ sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.c | 186 +++ sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.h | 30 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 627 +++++++++ sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 22 + sound/soc/qcom/qdsp6/q6prm-clocks.c | 85 ++ sound/soc/qcom/qdsp6/q6prm.c | 202 +++ sound/soc/qcom/qdsp6/q6prm.h | 78 ++ sound/soc/qcom/qdsp6/topology.c | 1113 ++++++++++++++++ 31 files changed, 7170 insertions(+), 1415 deletions(-) delete mode 100644 Documentation/devicetree/bindings/soc/qcom/qcom,apr.txt create mode 100644 Documentation/devicetree/bindings/soc/qcom/qcom,apr.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6apm-dai.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-clocks.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-ports.yaml create mode 100644 include/dt-bindings/soc/qcom,gpr.h create mode 100644 include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h create mode 100644 include/uapi/sound/snd_ar_tokens.h create mode 100644 sound/soc/qcom/qdsp6/audioreach.c create mode 100644 sound/soc/qcom/qdsp6/audioreach.h create mode 100644 sound/soc/qcom/qdsp6/q6apm-dai.c create mode 100644 sound/soc/qcom/qdsp6/q6apm-lpass-dais.c create mode 100644 sound/soc/qcom/qdsp6/q6apm.c create mode 100644 sound/soc/qcom/qdsp6/q6apm.h create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.c create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.h create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h create mode 100644 sound/soc/qcom/qdsp6/q6prm-clocks.c create mode 100644 sound/soc/qcom/qdsp6/q6prm.c create mode 100644 sound/soc/qcom/qdsp6/q6prm.h create mode 100644 sound/soc/qcom/qdsp6/topology.c -- 2.21.0

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[patch 00/11] hrtimers: Cleanup hrtimer_forward() [ab]use
by Thomas Gleixner 28 Sep '21

28 Sep '21

A recent syzbot report unearthed abuse of hrtimer_forward() which can cause runaway timers hogging the CPU in timer expiry context by rearming the timer in the past over and over. This happens when the caller uses timer->expiry for the 'now' argument of hrtimer_forward(). That works as long as the timer expiry is on time, but can cause a long period of rearm/fire loops which hog the CPU. Expiring late can have various causes, but obviously virtualization is prone to that due to VCPU scheduling. The correct usage of hrtimer_forward() is to hand the current time to the 'now' argument which ensures that the next event on the periodic time line is past now. This is what hrtimer_forward_now() provides. The following series addresses this: 1) Add a debug mechanism to the hrtimer expiry loop 2) Convert all hrtimer_forward() usage outside of kernel/time/ to use hrtimer_forward_now(). 3) Confine hrtimer_forward() to kernel/time/ core code. The mac80211_hwsim patch has already been picked up by the wireless maintainer and all other patches which affect usage outside the core code can be picked up by the relevant subsystems. If a maintainer wants me to pick a particular patch up, please let me know. The last patch which confines hrtimer_forward() will be postponed until all other patches have been merged into Linus tree. The series is also available from git: git://git.kernel.org/pub/scm/linux/kernel/git/tglx/devel.git hrtimer Thanks, tglx --- drivers/gpu/drm/i915/i915_pmu.c | 2 - drivers/net/wireless/mac80211_hwsim.c | 4 +- drivers/net/wwan/iosm/iosm_ipc_imem.c | 4 +- drivers/power/reset/ltc2952-poweroff.c | 4 -- include/linux/hrtimer.h | 26 ----------------- include/linux/posix-timers.h | 3 ++ kernel/signal.c | 14 +-------- kernel/time/hrtimer.c | 48 ++++++++++++++++++++++++++++++++- kernel/time/itimer.c | 13 ++++++++ kernel/time/posix-timers.c | 42 +++++++++++----------------- kernel/time/tick-internal.h | 1 net/can/bcm.c | 2 - sound/drivers/pcsp/pcsp_lib.c | 2 - 13 files changed, 92 insertions(+), 73 deletions(-)

2 2

[PATCH] snd-usb-audio: fix comment reference in __uac_clock_find_source
by Geraldo Nascimento 28 Sep '21

28 Sep '21

snd_usb_find_clock_source and snd_usb_find_clock_selector are helper macros that look at an entity id and validate that this entity id is in fact a clock source or a clock selector. The present comments inside __uac_clock_find_source give the reader the impression we're looking for an entity id. We're looking for an entity id indeed, the clock source, but since __uac_clock_find_source is recursive, we're also looking *at* the entity ids, in the search for the one clock source. Fix the comment so we don't give readers a wrong idea. Signed-off-by: Geraldo Nascimento <geraldogabriel(a)gmail.com> diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 81d5ce07d548..7dd71d342443 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -271,7 +271,7 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, return -EINVAL; } - /* first, see if the ID we're looking for is a clock source already */ + /* first, see if the ID we're looking at is a clock source already */ source = snd_usb_find_clock_source(chip, entity_id, proto); if (source) { entity_id = GET_VAL(source, proto, bClockID); @@ -297,7 +297,7 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, goto find_source; } - /* the entity ID we are looking for is a selector. + /* the entity ID we are looking at is a selector. * find out what it currently selects */ ret = uac_clock_selector_get_val(chip, clock_id); if (ret < 0) {

2 1

[RFC PATCH 0/4] ALSA: hda: potential hdac_stream locking issues?
by Pierre-Louis Bossart 28 Sep '21

28 Sep '21

While reviewing the HDAudio DMA handling, I found a number of inconsistencies in how spin_locks are used. It's not clear what the HDaudio bus->reg_lock is supposed to protect. In most cases only the writes to specific boolean status flags are protected, and there are multiple cases of taking the lock after testing a status flag. This patchset suggests a more consistent locking pattern, but it's entirely possible that the bus->reg_lock is only intented to protect register read/write access on the HDaudio bus, and not the status flags, and that this entire piece of code is completely over-engineered. On the Intel side no one knows why this spinlock was used, the reasons are lost to history. I set the 'RFC' status on purpose in the hope that Takashi might recall what this lock is supposed to protect. The diff format makes this patchset difficult to review, it's recommended to apply the patches and look at entire functions with changes to get a better idea of the suggested changes. Pierre-Louis Bossart (4): ALSA: hda: hdac_stream: fix potential locking issue in snd_hdac_stream_assign() ALSA: hda: hdac_stream: reset assigned_key in stream_release() ALSA: hda: hdac_ext_stream: fix potential locking issues ASoC: SOF: Intel: hda-dai: fix potential locking issue include/sound/hdaudio_ext.h | 2 ++ sound/hda/ext/hdac_ext_stream.c | 46 ++++++++++++++++++++------------- sound/hda/hdac_stream.c | 5 ++-- sound/soc/sof/intel/hda-dai.c | 7 ++--- 4 files changed, 37 insertions(+), 23 deletions(-) -- 2.25.1

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[RESEND PATCH v4] ASoC: dt-bindings: mediatek: mt8192: re-add audio afe document
by Tzung-Bi Shih 28 Sep '21

28 Sep '21

The document was merged as commit 1afc60e00de3 ("dt-bindings: mediatek: mt8192: add audio afe document"). However, [1] revealed that the commit 1afc60e00de3 breaks dt_binding_check due to dt-bindings/clock/mt8192-clk.h doesn't exist. As a temporary fix, commit 7d94ca3c8acd ("ASoC: mt8192: revert add audio afe document") reverted commit 1afc60e00de3. dt-bindings/clock/mt8192-clk.h is in mainline per commit f35f1a23e0e1 ("clk: mediatek: Add dt-bindings of MT8192 clocks"). Re-adds the document back. [1]: https://mailman.alsa-project.org/pipermail/alsa-devel/2020-November/176873.… Reviewed-by: Rob Herring <robh(a)kernel.org> Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- .../bindings/sound/mt8192-afe-pcm.yaml | 100 ++++++++++++++++++ 1 file changed, 100 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml diff --git a/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml b/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml new file mode 100644 index 000000000000..5e9fe067f440 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml @@ -0,0 +1,100 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mt8192-afe-pcm.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Mediatek AFE PCM controller for mt8192 + +maintainers: + - Jiaxin Yu <jiaxin.yu(a)mediatek.com> + - Shane Chien <shane.chien(a)mediatek.com> + +properties: + compatible: + const: mediatek,mt8192-audio + + interrupts: + maxItems: 1 + + resets: + maxItems: 1 + + reset-names: + const: audiosys + + mediatek,apmixedsys: + $ref: "/schemas/types.yaml#/definitions/phandle" + description: The phandle of the mediatek apmixedsys controller + + mediatek,infracfg: + $ref: "/schemas/types.yaml#/definitions/phandle" + description: The phandle of the mediatek infracfg controller + + mediatek,topckgen: + $ref: "/schemas/types.yaml#/definitions/phandle" + description: The phandle of the mediatek topckgen controller + + power-domains: + maxItems: 1 + + clocks: + items: + - description: AFE clock + - description: ADDA DAC clock + - description: ADDA DAC pre-distortion clock + - description: audio infra sys clock + - description: audio infra 26M clock + + clock-names: + items: + - const: aud_afe_clk + - const: aud_dac_clk + - const: aud_dac_predis_clk + - const: aud_infra_clk + - const: aud_infra_26m_clk + +required: + - compatible + - interrupts + - resets + - reset-names + - mediatek,apmixedsys + - mediatek,infracfg + - mediatek,topckgen + - power-domains + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/mt8192-clk.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + #include <dt-bindings/power/mt8192-power.h> + #include <dt-bindings/reset-controller/mt8192-resets.h> + + afe: mt8192-afe-pcm { + compatible = "mediatek,mt8192-audio"; + interrupts = <GIC_SPI 202 IRQ_TYPE_LEVEL_HIGH>; + resets = <&watchdog MT8192_TOPRGU_AUDIO_SW_RST>; + reset-names = "audiosys"; + mediatek,apmixedsys = <&apmixedsys>; + mediatek,infracfg = <&infracfg>; + mediatek,topckgen = <&topckgen>; + power-domains = <&scpsys MT8192_POWER_DOMAIN_AUDIO>; + clocks = <&audsys CLK_AUD_AFE>, + <&audsys CLK_AUD_DAC>, + <&audsys CLK_AUD_DAC_PREDIS>, + <&infracfg CLK_INFRA_AUDIO>, + <&infracfg CLK_INFRA_AUDIO_26M_B>; + clock-names = "aud_afe_clk", + "aud_dac_clk", + "aud_dac_predis_clk", + "aud_infra_clk", + "aud_infra_26m_clk"; + }; + +... -- 2.33.0.685.g46640cef36-goog

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[PATCH] ASoC: zl38060: Update to modern clocking terminology
by Mark Brown 27 Sep '21

27 Sep '21

As part of moving to remove the old style defines for the bus clocks update the zl38060 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/zl38060.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/zl38060.c b/sound/soc/codecs/zl38060.c index d21a72314d37..d20ec1571010 100644 --- a/sound/soc/codecs/zl38060.c +++ b/sound/soc/codecs/zl38060.c @@ -250,8 +250,8 @@ static int zl38_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: /* always 32 bits per frame (= 16 bits/channel, 2 channels) */ err = regmap_update_bits(priv->regmap, REG_TDMA_CFG_CLK, CFG_CLK_MASTER | CFG_CLK_PCLK_MASK, -- 2.20.1

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[PATCH] ASoC: bcm: Convert to modern clocking terminology
by Mark Brown 27 Sep '21

27 Sep '21

As part of the effort to remove our old APIs based on outdated terminology update the Broadcom drivers to use modern terminology. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/bcm/bcm2835-i2s.c | 56 ++++++++++++++++++------------------- sound/soc/bcm/cygnus-ssp.c | 6 ++-- 2 files changed, 31 insertions(+), 31 deletions(-) diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c index 3d668f449bc1..e3fc4bee8cfd 100644 --- a/sound/soc/bcm/bcm2835-i2s.c +++ b/sound/soc/bcm/bcm2835-i2s.c @@ -127,14 +127,14 @@ struct bcm2835_i2s_dev { static void bcm2835_i2s_start_clock(struct bcm2835_i2s_dev *dev) { - unsigned int master = dev->fmt & SND_SOC_DAIFMT_MASTER_MASK; + unsigned int provider = dev->fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK; if (dev->clk_prepared) return; - switch (master) { - case SND_SOC_DAIFMT_CBS_CFS: - case SND_SOC_DAIFMT_CBS_CFM: + switch (provider) { + case SND_SOC_DAIFMT_CBC_CFC: + case SND_SOC_DAIFMT_CBC_CFP: clk_prepare_enable(dev->clk); dev->clk_prepared = true; break; @@ -337,8 +337,8 @@ static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream, unsigned int rx_mask, tx_mask; unsigned int rx_ch1_pos, rx_ch2_pos, tx_ch1_pos, tx_ch2_pos; unsigned int mode, format; - bool bit_clock_master = false; - bool frame_sync_master = false; + bool bit_clock_provider = false; + bool frame_sync_provider = false; bool frame_start_falling_edge = false; uint32_t csreg; int ret = 0; @@ -383,36 +383,36 @@ static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream, if (data_length > slot_width) return -EINVAL; - /* Check if CPU is bit clock master */ - switch (dev->fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: - case SND_SOC_DAIFMT_CBS_CFM: - bit_clock_master = true; + /* Check if CPU is bit clock provider */ + switch (dev->fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBC_CFC: + case SND_SOC_DAIFMT_CBC_CFP: + bit_clock_provider = true; break; - case SND_SOC_DAIFMT_CBM_CFS: - case SND_SOC_DAIFMT_CBM_CFM: - bit_clock_master = false; + case SND_SOC_DAIFMT_CBP_CFC: + case SND_SOC_DAIFMT_CBP_CFP: + bit_clock_provider = false; break; default: return -EINVAL; } - /* Check if CPU is frame sync master */ - switch (dev->fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: - case SND_SOC_DAIFMT_CBM_CFS: - frame_sync_master = true; + /* Check if CPU is frame sync provider */ + switch (dev->fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBC_CFC: + case SND_SOC_DAIFMT_CBP_CFC: + frame_sync_provider = true; break; - case SND_SOC_DAIFMT_CBS_CFM: - case SND_SOC_DAIFMT_CBM_CFM: - frame_sync_master = false; + case SND_SOC_DAIFMT_CBC_CFP: + case SND_SOC_DAIFMT_CBP_CFP: + frame_sync_provider = false; break; default: return -EINVAL; } /* Clock should only be set up here if CPU is clock master */ - if (bit_clock_master && + if (bit_clock_provider && (!dev->clk_prepared || dev->clk_rate != bclk_rate)) { if (dev->clk_prepared) bcm2835_i2s_stop_clock(dev); @@ -501,11 +501,11 @@ static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream, /* * Transmitting data immediately after frame start, eg * in left-justified or DSP mode A, only works stable - * if bcm2835 is the frame clock master. + * if bcm2835 is the frame clock provider. */ - if ((!rx_ch1_pos || !tx_ch1_pos) && !frame_sync_master) + if ((!rx_ch1_pos || !tx_ch1_pos) && !frame_sync_provider) dev_warn(dev->dev, - "Unstable slave config detected, L/R may be swapped"); + "Unstable consumer config detected, L/R may be swapped"); /* * Set format for both streams. @@ -538,11 +538,11 @@ static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream, mode |= BCM2835_I2S_FSLEN(framesync_length); /* CLKM selects bcm2835 clock slave mode */ - if (!bit_clock_master) + if (!bit_clock_provider) mode |= BCM2835_I2S_CLKM; /* FSM selects bcm2835 frame sync slave mode */ - if (!frame_sync_master) + if (!frame_sync_provider) mode |= BCM2835_I2S_FSM; /* CLKI selects normal clocking mode, sampling on rising edge */ diff --git a/sound/soc/bcm/cygnus-ssp.c b/sound/soc/bcm/cygnus-ssp.c index fca5a3f2eec5..9698f4531c90 100644 --- a/sound/soc/bcm/cygnus-ssp.c +++ b/sound/soc/bcm/cygnus-ssp.c @@ -848,12 +848,12 @@ static int cygnus_ssp_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) ssp_newcfg = 0; - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: ssp_newcfg |= BIT(I2S_OUT_CFGX_SLAVE_MODE); aio->is_slave = 1; break; - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBC_CFC: ssp_newcfg &= ~BIT(I2S_OUT_CFGX_SLAVE_MODE); aio->is_slave = 0; break; -- 2.20.1

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[PATCH] ASoC: cpcap: Use modern ASoC DAI format terminology
by Mark Brown 27 Sep '21

27 Sep '21

As part of moving to remove the old style defines for the bus clocks update the cpcap driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/cpcap.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cpcap.c b/sound/soc/codecs/cpcap.c index 05bbacd0d174..9a932d088fb5 100644 --- a/sound/soc/codecs/cpcap.c +++ b/sound/soc/codecs/cpcap.c @@ -1168,15 +1168,15 @@ static int cpcap_hifi_set_dai_fmt(struct snd_soc_dai *codec_dai, /* * "HiFi Playback" should always be configured as - * SND_SOC_DAIFMT_CBM_CFM - codec clk & frm master + * SND_SOC_DAIFMT_CBP_CFP - codec clk & frm provider * SND_SOC_DAIFMT_I2S - I2S mode */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: val &= ~BIT(CPCAP_BIT_SMB_ST_DAC); break; default: - dev_err(dev, "HiFi dai fmt failed: CPCAP should be master"); + dev_err(dev, "HiFi dai fmt failed: CPCAP should be provider"); return -EINVAL; } -- 2.20.1

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[PATCH] ASoC: dt-bindings: wlf,wm8978: Fix I2C address in example
by Geert Uytterhoeven 27 Sep '21

27 Sep '21

According to the WM8978 datasheet, the address of the device is fixed as 0011010 in 2-wire mode. Fixes: 5bd5699c494f42a3 ("ASoC: dt-bindings: Add WM8978 Binding") Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- Documentation/devicetree/bindings/sound/wlf,wm8978.yaml | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8978.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8978.yaml index 944e5859c962e634..0b80ac3a7106658c 100644 --- a/Documentation/devicetree/bindings/sound/wlf,wm8978.yaml +++ b/Documentation/devicetree/bindings/sound/wlf,wm8978.yaml @@ -48,10 +48,10 @@ examples: #address-cells = <1>; #size-cells = <0>; - codec@0 { + codec@1a { #sound-dai-cells = <0>; compatible = "wlf,wm8978"; - reg = <0>; + reg = <1a>; }; }; -- 2.25.1

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[PATCH] ASoC: ak4104: Update to modern clocking terminology
by Mark Brown 27 Sep '21

27 Sep '21

As part of moving to remove the old style defines for the bus clocks update the ak4104 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/ak4104.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index 979cfb165eed..dc4747c77a7a 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -81,8 +81,8 @@ static int ak4104_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - /* This device can only be slave */ - if ((format & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) + /* This device can only be consumer */ + if ((format & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) != SND_SOC_DAIFMT_CBC_CFC) return -EINVAL; ret = regmap_update_bits(ak4104->regmap, AK4104_REG_CONTROL1, -- 2.20.1

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