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

[PATCH AUTOSEL 5.14 012/252] ASoC: ti: davinci-mcasp: Fix DIT mode support
by Sasha Levin 09 Sep '21

09 Sep '21

From: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> [ Upstream commit bbdd3f4dbe81e19b9123bc54e23ed54517615524 ] The DIT mode support has not been tested due to lack of platform where it can be tested. To be able to use the McASP on OMAP4/5 (only supporting DIT mode) we need to have DIT mode working in the McASP driver on a know platform. After hacking around (on BBW, mcasp1.axr1 can be routed out for this) it appeared that DIT mode is broken. This patch fixes it up and 16/24 bit audio works along with passthrough, but I have only tested with DTS example and test files. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> Link: https://lore.kernel.org/r/20210705194249.2385-2-peter.ujfalusi@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/ti/davinci-mcasp.c | 150 ++++++++++++++++++++++++++++++----- 1 file changed, 129 insertions(+), 21 deletions(-) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index 017a5a5e56cd..64ec6d485834 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -83,6 +83,8 @@ struct davinci_mcasp { struct snd_pcm_substream *substreams[2]; unsigned int dai_fmt; + u32 iec958_status; + /* Audio can not be enabled due to missing parameter(s) */ bool missing_audio_param; @@ -757,6 +759,9 @@ static int davinci_mcasp_set_tdm_slot(struct snd_soc_dai *dai, { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); + if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) + return 0; + dev_dbg(mcasp->dev, "%s() tx_mask 0x%08x rx_mask 0x%08x slots %d width %d\n", __func__, tx_mask, rx_mask, slots, slot_width); @@ -827,6 +832,20 @@ static int davinci_config_channel_size(struct davinci_mcasp *mcasp, mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, RXROT(rx_rotate), RXROT(7)); mcasp_set_reg(mcasp, DAVINCI_MCASP_RXMASK_REG, mask); + } else { + /* + * according to the TRM it should be TXROT=0, this one works: + * 16 bit to 23-8 (TXROT=6, rotate 24 bits) + * 24 bit to 23-0 (TXROT=0, rotate 0 bits) + * + * TXROT = 0 only works with 24bit samples + */ + tx_rotate = (sample_width / 4 + 2) & 0x7; + + mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, TXROT(tx_rotate), + TXROT(7)); + mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, TXSSZ(15), + TXSSZ(0x0F)); } mcasp_set_reg(mcasp, DAVINCI_MCASP_TXMASK_REG, mask); @@ -842,10 +861,16 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream, u8 tx_ser = 0; u8 rx_ser = 0; u8 slots = mcasp->tdm_slots; - u8 max_active_serializers = (channels + slots - 1) / slots; - u8 max_rx_serializers, max_tx_serializers; + u8 max_active_serializers, max_rx_serializers, max_tx_serializers; int active_serializers, numevt; u32 reg; + + /* In DIT mode we only allow maximum of one serializers for now */ + if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) + max_active_serializers = 1; + else + max_active_serializers = (channels + slots - 1) / slots; + /* Default configuration */ if (mcasp->version < MCASP_VERSION_3) mcasp_set_bits(mcasp, DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT); @@ -1031,16 +1056,18 @@ static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream, static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp, unsigned int rate) { - u32 cs_value = 0; - u8 *cs_bytes = (u8*) &cs_value; + u8 *cs_bytes = (u8 *)&mcasp->iec958_status; - /* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0 - and LSB first */ - mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, TXROT(6) | TXSSZ(15)); + if (!mcasp->dat_port) + mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, TXSEL); + else + mcasp_clr_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, TXSEL); /* Set TX frame synch : DIT Mode, 1 bit width, internal, rising edge */ mcasp_set_reg(mcasp, DAVINCI_MCASP_TXFMCTL_REG, AFSXE | FSXMOD(0x180)); + mcasp_set_reg(mcasp, DAVINCI_MCASP_TXMASK_REG, 0xFFFF); + /* Set the TX tdm : for all the slots */ mcasp_set_reg(mcasp, DAVINCI_MCASP_TXTDM_REG, 0xFFFFFFFF); @@ -1049,16 +1076,8 @@ static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp, mcasp_clr_bits(mcasp, DAVINCI_MCASP_XEVTCTL_REG, TXDATADMADIS); - /* Only 44100 and 48000 are valid, both have the same setting */ - mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXDIV(3)); - - /* Enable the DIT */ - mcasp_set_bits(mcasp, DAVINCI_MCASP_TXDITCTL_REG, DITEN); - /* Set S/PDIF channel status bits */ - cs_bytes[0] = IEC958_AES0_CON_NOT_COPYRIGHT; - cs_bytes[1] = IEC958_AES1_CON_PCM_CODER; - + cs_bytes[3] &= ~IEC958_AES3_CON_FS; switch (rate) { case 22050: cs_bytes[3] |= IEC958_AES3_CON_FS_22050; @@ -1088,12 +1107,15 @@ static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp, cs_bytes[3] |= IEC958_AES3_CON_FS_192000; break; default: - printk(KERN_WARNING "unsupported sampling rate: %d\n", rate); + dev_err(mcasp->dev, "unsupported sampling rate: %d\n", rate); return -EINVAL; } - mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRA_REG, cs_value); - mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRB_REG, cs_value); + mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRA_REG, mcasp->iec958_status); + mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRB_REG, mcasp->iec958_status); + + /* Enable the DIT */ + mcasp_set_bits(mcasp, DAVINCI_MCASP_TXDITCTL_REG, DITEN); return 0; } @@ -1237,12 +1259,18 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, int slots = mcasp->tdm_slots; int rate = params_rate(params); int sbits = params_width(params); + unsigned int bclk_target; if (mcasp->slot_width) sbits = mcasp->slot_width; + if (mcasp->op_mode == DAVINCI_MCASP_IIS_MODE) + bclk_target = rate * sbits * slots; + else + bclk_target = rate * 128; + davinci_mcasp_calc_clk_div(mcasp, mcasp->sysclk_freq, - rate * sbits * slots, true); + bclk_target, true); } ret = mcasp_common_hw_param(mcasp, substream->stream, @@ -1598,6 +1626,77 @@ static const struct snd_soc_dai_ops davinci_mcasp_dai_ops = { .set_tdm_slot = davinci_mcasp_set_tdm_slot, }; +static int davinci_mcasp_iec958_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958; + uinfo->count = 1; + + return 0; +} + +static int davinci_mcasp_iec958_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *uctl) +{ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai); + + memcpy(uctl->value.iec958.status, &mcasp->iec958_status, + sizeof(mcasp->iec958_status)); + + return 0; +} + +static int davinci_mcasp_iec958_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *uctl) +{ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai); + + memcpy(&mcasp->iec958_status, uctl->value.iec958.status, + sizeof(mcasp->iec958_status)); + + return 0; +} + +static int davinci_mcasp_iec958_con_mask_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai); + + memset(ucontrol->value.iec958.status, 0xff, sizeof(mcasp->iec958_status)); + return 0; +} + +static const struct snd_kcontrol_new davinci_mcasp_iec958_ctls[] = { + { + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_VOLATILE), + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, DEFAULT), + .info = davinci_mcasp_iec958_info, + .get = davinci_mcasp_iec958_get, + .put = davinci_mcasp_iec958_put, + }, { + .access = SNDRV_CTL_ELEM_ACCESS_READ, + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, CON_MASK), + .info = davinci_mcasp_iec958_info, + .get = davinci_mcasp_iec958_con_mask_get, + }, +}; + +static void davinci_mcasp_init_iec958_status(struct davinci_mcasp *mcasp) +{ + unsigned char *cs = (u8 *)&mcasp->iec958_status; + + cs[0] = IEC958_AES0_CON_NOT_COPYRIGHT | IEC958_AES0_CON_EMPHASIS_NONE; + cs[1] = IEC958_AES1_CON_PCM_CODER; + cs[2] = IEC958_AES2_CON_SOURCE_UNSPEC | IEC958_AES2_CON_CHANNEL_UNSPEC; + cs[3] = IEC958_AES3_CON_CLOCK_1000PPM; +} + static int davinci_mcasp_dai_probe(struct snd_soc_dai *dai) { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); @@ -1605,6 +1704,12 @@ static int davinci_mcasp_dai_probe(struct snd_soc_dai *dai) dai->playback_dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK]; dai->capture_dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE]; + if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) { + davinci_mcasp_init_iec958_status(mcasp); + snd_soc_add_dai_controls(dai, davinci_mcasp_iec958_ctls, + ARRAY_SIZE(davinci_mcasp_iec958_ctls)); + } + return 0; } @@ -1651,7 +1756,8 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = { .channels_min = 1, .channels_max = 384, .rates = DAVINCI_MCASP_RATES, - .formats = DAVINCI_MCASP_PCM_FMTS, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, }, .ops = &davinci_mcasp_dai_ops, }, @@ -1871,6 +1977,8 @@ static int davinci_mcasp_get_config(struct davinci_mcasp *mcasp, } else { mcasp->tdm_slots = pdata->tdm_slots; } + } else { + mcasp->tdm_slots = 32; } mcasp->num_serializer = pdata->num_serializer; -- 2.30.2

1 0

[PATCH] ASoC: SOF: fix a typo in put operations for kcontrol
by Pierre-Louis Bossart 08 Sep '21

08 Sep '21

From: Rander Wang <rander.wang(a)intel.com> SOF_CTRL_TYPE_VALUE_CHAN_SET should be used for put operations for consistency. The current use of _GET is obviously incorrect but _GET and _SET result in the same action so there is no functional change introduced by this patch. Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Rander Wang <rander.wang(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/control.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/control.c b/sound/soc/sof/control.c index a5dd728c580a..504500dd4d43 100644 --- a/sound/soc/sof/control.c +++ b/sound/soc/sof/control.c @@ -108,7 +108,7 @@ int snd_sof_volume_put(struct snd_kcontrol *kcontrol, if (pm_runtime_active(scomp->dev)) snd_sof_ipc_set_get_comp_data(scontrol, SOF_IPC_COMP_SET_VALUE, - SOF_CTRL_TYPE_VALUE_CHAN_GET, + SOF_CTRL_TYPE_VALUE_CHAN_SET, SOF_CTRL_CMD_VOLUME, true); return change; @@ -179,7 +179,7 @@ int snd_sof_switch_put(struct snd_kcontrol *kcontrol, if (pm_runtime_active(scomp->dev)) snd_sof_ipc_set_get_comp_data(scontrol, SOF_IPC_COMP_SET_VALUE, - SOF_CTRL_TYPE_VALUE_CHAN_GET, + SOF_CTRL_TYPE_VALUE_CHAN_SET, SOF_CTRL_CMD_SWITCH, true); @@ -226,7 +226,7 @@ int snd_sof_enum_put(struct snd_kcontrol *kcontrol, if (pm_runtime_active(scomp->dev)) snd_sof_ipc_set_get_comp_data(scontrol, SOF_IPC_COMP_SET_VALUE, - SOF_CTRL_TYPE_VALUE_CHAN_GET, + SOF_CTRL_TYPE_VALUE_CHAN_SET, SOF_CTRL_CMD_ENUM, true); -- 2.25.1

1 1

[GIT PULL] ASoC fixes for v5.15-rc1
by Mark Brown 08 Sep '21

08 Sep '21

The following changes since commit 38b7673000949ca784fcb8a9feb70d2a802befa6: Merge remote-tracking branch 'asoc/for-5.15' into asoc-linus (2021-08-30 12:30:33 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.15-rc1 for you to fetch changes up to fcb958ee8e832e9cdf43408535207e15f14af755: ASoC: rockchip: i2s: Fix concurrency between tx/rx (2021-09-03 16:19:01 +0100) ---------------------------------------------------------------- ASoC: Fixes for v5.15 A collection of fixes that came in during the merge window, nothing too remarkable but a reasonably large number of fixes. ---------------------------------------------------------------- Colin Ian King (1): ASoC: samsung: s3c24xx_simtec: fix spelling mistake "devicec" -> "device" Geert Uytterhoeven (1): ASoC: mediatek: SND_SOC_MT8195 should depend on ARCH_MEDIATEK Kuninori Morimoto (1): ASoC: audio-graph: respawn Platform Support Mark Brown (1): Merge tag 'asoc-v5.15' into asoc-5.15 Nathan Chancellor (1): ASoC: Intel: boards: Fix CONFIG_SND_SOC_SDW_MOCKUP select Shengjiu Wang (1): ASoC: dt-bindings: fsl_rpmsg: Add compatible string for i.MX8ULP Shuming Fan (1): ASoC: rt5682: fix headset background noise when S3 state Sugar Zhang (1): ASoC: rockchip: i2s: Fix concurrency between tx/rx Trevor Wu (3): ASoC: mediatek: mt8195: add MTK_PMIC_WRAP dependency ASoC: dt-bindings: mt8195: remove dependent headers in the example ASoC: mt8195: correct the dts parsing logic about DPTX and HDMITX .../devicetree/bindings/sound/fsl,rpmsg.yaml | 1 + .../devicetree/bindings/sound/mt8195-afe-pcm.yaml | 40 ++++++++++------------ sound/soc/codecs/rt5682.c | 3 -- sound/soc/generic/audio-graph-card.c | 6 ++++ sound/soc/intel/boards/Kconfig | 2 +- sound/soc/mediatek/Kconfig | 3 +- .../mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c | 22 ++++++------ sound/soc/rockchip/rockchip_i2s.c | 7 ++++ sound/soc/samsung/s3c24xx_simtec.c | 2 +- 9 files changed, 47 insertions(+), 39 deletions(-)

2 1

[PATCH v5 00/21] ASoC: qcom: Add AudioReach support
by Srinivas Kandagatla 08 Sep '21

08 Sep '21

Many thanks for reviewing v4. This version addresses all the comments raised as part of v4 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/ Thanks, srini Changes since v4: only apr dt-binding changes. - deprecated qcom,apr-domain dt-bindings to make it more reusable - added more protection-domain list that have been missed during yaml conversion. - remove "qcom,apr" compatible that was added by mistake during yaml conversion. Srinivas Kandagatla (21): 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 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 | 169 +++ .../devicetree/bindings/sound/qcom,q6afe.txt | 181 --- .../bindings/sound/qcom,q6apm-dai.yaml | 50 + .../sound/qcom,q6dsp-lpass-clocks.yaml | 74 ++ .../sound/qcom,q6dsp-lpass-ports.yaml | 202 +++ drivers/soc/qcom/Kconfig | 8 + drivers/soc/qcom/apr.c | 289 +++- include/dt-bindings/soc/qcom,gpr.h | 18 + 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 | 203 +++ sound/soc/qcom/Kconfig | 22 + sound/soc/qcom/qdsp6/Makefile | 11 +- sound/soc/qcom/qdsp6/audioreach.c | 1170 +++++++++++++++++ sound/soc/qcom/qdsp6/audioreach.h | 692 ++++++++++ sound/soc/qcom/qdsp6/q6afe-clocks.c | 187 +-- sound/soc/qcom/qdsp6/q6afe-dai.c | 687 +--------- sound/soc/qcom/qdsp6/q6apm-dai.c | 504 +++++++ sound/soc/qcom/qdsp6/q6apm-lpass-dais.c | 266 ++++ sound/soc/qcom/qdsp6/q6apm.c | 916 +++++++++++++ sound/soc/qcom/qdsp6/q6apm.h | 154 +++ 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 | 256 ++++ sound/soc/qcom/qdsp6/q6prm.h | 78 ++ sound/soc/qcom/qdsp6/topology.c | 1118 ++++++++++++++++ 31 files changed, 7406 insertions(+), 1414 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

5 53

[PATCH] ASoC: qcom: lpass-platform: Reset irq clear reg post handling interrupts
by Srinivasa Rao Mandadapu 07 Sep '21

07 Sep '21

Update interrupt clear register with reset value after addressing all interrupts. This is to fix playback or capture hanging issue in simultaneous playback and capture usecase. Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- sound/soc/qcom/lpass-platform.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index f9df76d37858..1a0a4b0b1a03 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -749,6 +749,12 @@ static irqreturn_t lpass_platform_lpaif_irq(int irq, void *data) } } + rv = regmap_write(drvdata->lpaif_map, LPAIF_IRQCLEAR_REG(v, LPAIF_IRQ_PORT_HOST), 0x0); + if (rv) { + pr_err("error writing to irqstat reg: %d\n", rv); + return IRQ_NONE; + } + return IRQ_HANDLED; } -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

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[PATCH v6 0/2] Cirrus Logic CS35L41 Amplifier
by David Rhodes 07 Sep '21

07 Sep '21

ASoC driver and devicetree documentation for a new Cirrus Logic amplifier CS35L41 v6 changes: Fix header comment style Remove some unsafe mixer controls Remove some fields in regmap readable_reg Remove unneccesssary regmap setup in component_probe Remove extra whitespace Reorder in probe: handle_pdata before regulator_enable Set initial analog gain to 0dB David Rhodes (2): ASoC: cs35l41: CS35L41 Boosted Smart Amplifier ASoC: cs35l41: Add bindings for CS35L41 .../devicetree/bindings/sound/cs35l41.yaml | 162 ++ include/sound/cs35l41.h | 35 + sound/soc/codecs/Kconfig | 12 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/cs35l41-i2c.c | 114 ++ sound/soc/codecs/cs35l41-spi.c | 143 ++ sound/soc/codecs/cs35l41-tables.c | 597 +++++++ sound/soc/codecs/cs35l41.c | 1561 +++++++++++++++++ sound/soc/codecs/cs35l41.h | 775 ++++++++ 9 files changed, 3403 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs35l41.yaml create mode 100644 include/sound/cs35l41.h create mode 100644 sound/soc/codecs/cs35l41-i2c.c create mode 100644 sound/soc/codecs/cs35l41-spi.c create mode 100644 sound/soc/codecs/cs35l41-tables.c create mode 100644 sound/soc/codecs/cs35l41.c create mode 100644 sound/soc/codecs/cs35l41.h -- 2.25.1

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[PATCH] ASoC: wm8960: Fix clock configuration on slave mode
by Shengjiu Wang 07 Sep '21

07 Sep '21

There is a noise issue for 8kHz sample rate on slave mode. Compared with master mode, the difference is the DACDIV setting, after correcting the DACDIV, the noise is gone. There is no noise issue for 48kHz sample rate, because the default value of DACDIV is correct for 48kHz. So wm8960_configure_clocking() should be functional for ADC and DAC function even if it is slave mode. In order to be compatible for old use case, just add condition for checking that sysclk is zero with slave mode. Fixes: 0e50b51aa22f ("ASoC: wm8960: Let wm8960 driver configure its bit clock and frame clock") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/codecs/wm8960.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 9e621a254392..9c6af76a60fd 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -742,7 +742,7 @@ static int wm8960_configure_clocking(struct snd_soc_component *component) int i, j, k; int ret; - if (!(iface1 & (1<<6))) { + if (!(iface1 & (1 << 6)) && !wm8960->sysclk) { dev_dbg(component->dev, "Codec is slave mode, no need to configure clock\n"); return 0; -- 2.17.1

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[PATCH 1/2] ALSA: gus: Fix repeated probes of snd_gus_create()
by Takashi Iwai 07 Sep '21

07 Sep '21

GUS card object may be repeatedly probed for the legacy ISA devices, and the behavior doesn't fit with the devres resource management. Revert partially back to the classical way for the snd_gus_card object, so that the repeated calls of snd_gus_create() are allowed. Fixes: 5b88da3c800f ("ALSA: gus: Allocate resources with device-managed APIs") Reported-by: kernel test robot <oliver.sang(a)intel.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/isa/gus/gus_main.c | 44 ++++++++++++++++++++++++++++------------ 1 file changed, 31 insertions(+), 13 deletions(-) diff --git a/sound/isa/gus/gus_main.c b/sound/isa/gus/gus_main.c index ae1e2542ee4a..3b46490271fe 100644 --- a/sound/isa/gus/gus_main.c +++ b/sound/isa/gus/gus_main.c @@ -87,10 +87,24 @@ static void snd_gus_init_control(struct snd_gus_card *gus) static int snd_gus_free(struct snd_gus_card *gus) { - if (gus->gf1.res_port2) { - snd_gf1_stop(gus); - snd_gus_init_dma_irq(gus, 0); + if (gus->gf1.res_port2 == NULL) + goto __hw_end; + snd_gf1_stop(gus); + snd_gus_init_dma_irq(gus, 0); + __hw_end: + release_and_free_resource(gus->gf1.res_port1); + release_and_free_resource(gus->gf1.res_port2); + if (gus->gf1.irq >= 0) + free_irq(gus->gf1.irq, (void *) gus); + if (gus->gf1.dma1 >= 0) { + disable_dma(gus->gf1.dma1); + free_dma(gus->gf1.dma1); } + if (!gus->equal_dma && gus->gf1.dma2 >= 0) { + disable_dma(gus->gf1.dma2); + free_dma(gus->gf1.dma2); + } + kfree(gus); return 0; } @@ -116,7 +130,7 @@ int snd_gus_create(struct snd_card *card, }; *rgus = NULL; - gus = devm_kzalloc(card->dev, sizeof(*gus), GFP_KERNEL); + gus = kzalloc(sizeof(*gus), GFP_KERNEL); if (gus == NULL) return -ENOMEM; spin_lock_init(&gus->reg_lock); @@ -142,33 +156,35 @@ int snd_gus_create(struct snd_card *card, gus->gf1.reg_timerctrl = GUSP(gus, TIMERCNTRL); gus->gf1.reg_timerdata = GUSP(gus, TIMERDATA); /* allocate resources */ - gus->gf1.res_port1 = devm_request_region(card->dev, port, 16, - "GUS GF1 (Adlib/SB)"); + gus->gf1.res_port1 = request_region(port, 16, "GUS GF1 (Adlib/SB)"); if (!gus->gf1.res_port1) { snd_printk(KERN_ERR "gus: can't grab SB port 0x%lx\n", port); + snd_gus_free(gus); return -EBUSY; } - gus->gf1.res_port2 = devm_request_region(card->dev, port + 0x100, 12, - "GUS GF1 (Synth)"); + gus->gf1.res_port2 = request_region(port + 0x100, 12, "GUS GF1 (Synth)"); if (!gus->gf1.res_port2) { snd_printk(KERN_ERR "gus: can't grab synth port 0x%lx\n", port + 0x100); + snd_gus_free(gus); return -EBUSY; } - if (irq >= 0 && devm_request_irq(card->dev, irq, snd_gus_interrupt, 0, - "GUS GF1", (void *) gus)) { + if (irq >= 0 && request_irq(irq, snd_gus_interrupt, 0, "GUS GF1", (void *) gus)) { snd_printk(KERN_ERR "gus: can't grab irq %d\n", irq); + snd_gus_free(gus); return -EBUSY; } gus->gf1.irq = irq; card->sync_irq = irq; - if (snd_devm_request_dma(card->dev, dma1, "GUS - 1")) { + if (request_dma(dma1, "GUS - 1")) { snd_printk(KERN_ERR "gus: can't grab DMA1 %d\n", dma1); + snd_gus_free(gus); return -EBUSY; } gus->gf1.dma1 = dma1; if (dma2 >= 0 && dma1 != dma2) { - if (snd_devm_request_dma(card->dev, dma2, "GUS - 2")) { + if (request_dma(dma2, "GUS - 2")) { snd_printk(KERN_ERR "gus: can't grab DMA2 %d\n", dma2); + snd_gus_free(gus); return -EBUSY; } gus->gf1.dma2 = dma2; @@ -193,8 +209,10 @@ int snd_gus_create(struct snd_card *card, gus->gf1.volume_ramp = 25; gus->gf1.smooth_pan = 1; err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, gus, &ops); - if (err < 0) + if (err < 0) { + snd_gus_free(gus); return err; + } *rgus = gus; return 0; } -- 2.26.2

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[PATCH v1] ALSA: vx222: fix null-ptr-deref
by Tong Zhang 07 Sep '21

07 Sep '21

a recent refactor created a null pointer vx in snd_vx222_probe(). The vx pointer should have been populated in snd_vx222_create() as suggested in earlier version, otherwise vx->core.ibl.size will throw an error. [ 1.298398] BUG: kernel NULL pointer dereference, address: 00000000000001d8 [ 1.316799] RIP: 0010:snd_vx222_probe+0x155/0x290 [snd_vx222] Fixes: 3bde3359aa16 ("ALSA: vx222: Allocate resources with device-managed APIs") Signed-off-by: Tong Zhang <ztong0001(a)gmail.com> --- sound/pci/vx222/vx222.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/vx222/vx222.c b/sound/pci/vx222/vx222.c index f48cc20b9e8a..468a6a20dc1e 100644 --- a/sound/pci/vx222/vx222.c +++ b/sound/pci/vx222/vx222.c @@ -137,6 +137,7 @@ static int snd_vx222_create(struct snd_card *card, struct pci_dev *pci, } chip->irq = pci->irq; card->sync_irq = chip->irq; + *rchip = vx; return 0; } -- 2.25.1

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ucm2: Add UCM support for rockchip_es8316 on Pinebook Pro
by GitHub pull_request - opened 07 Sep '21

07 Sep '21

alsa-project/alsa-ucm-conf pull request #112 was opened from starnight: Add UCM config files for Pinebook Pro's audio. The config files are transformed and tuned from Manjaro ARM's packages: * pinebookpro-post-install's asound.state [1] * pinebookpro-audio's audio_jack_plugged_in.sh [2] By the way, with these config files, the Speaker and Headphones will be switched according to the (un)plugged state automatically. [1] https://gitlab.manjaro.org/manjaro-arm/packages/community/pinebookpro-post-… [2] https://gitlab.manjaro.org/manjaro-arm/packages/community/pinebookpro-audio… Fixes: https://github.com/alsa-project/alsa-ucm-conf/issues/110 Signed-off-by: Jian-Hong Pan <jhp(a)endlessos.org> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/112 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/112.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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