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author | Mark Brown <broonie@kernel.org> | 2021-08-06 01:46:24 +0100 |
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committer | Mark Brown <broonie@kernel.org> | 2021-08-06 01:46:24 +0100 |
commit | ddaa1ed52c5da64fe9adf1d5ea6202cda3a53eea (patch) | |
tree | 119c5b9dcf67841af4a92f3d3a9b5e4efcc4557b | |
parent | 585fb31c2c3aad38a5add009171d7e3277b76d45 (diff) | |
parent | e2f6867299ac85ce227eee18be11ce2c4a568447 (diff) | |
download | linux-stable-ddaa1ed52c5da64fe9adf1d5ea6202cda3a53eea.tar.gz linux-stable-ddaa1ed52c5da64fe9adf1d5ea6202cda3a53eea.tar.bz2 linux-stable-ddaa1ed52c5da64fe9adf1d5ea6202cda3a53eea.zip |
Merge some cs42l42 patches into asoc-5.15
33 files changed, 318 insertions, 189 deletions
diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml index ee936d1aa724..c2930d65728e 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml @@ -114,7 +114,7 @@ properties: ports: $ref: /schemas/graph.yaml#/properties/ports - properties: + patternProperties: port(@[0-9a-f]+)?: $ref: audio-graph-port.yaml# unevaluatedProperties: false diff --git a/MAINTAINERS b/MAINTAINERS index a61f4f3b78a9..3167fd99fc02 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -19992,7 +19992,8 @@ F: Documentation/devicetree/bindings/extcon/wlf,arizona.yaml F: Documentation/devicetree/bindings/mfd/wlf,arizona.yaml F: Documentation/devicetree/bindings/mfd/wm831x.txt F: Documentation/devicetree/bindings/regulator/wlf,arizona.yaml -F: Documentation/devicetree/bindings/sound/wlf,arizona.yaml +F: Documentation/devicetree/bindings/sound/wlf,*.yaml +F: Documentation/devicetree/bindings/sound/wm* F: Documentation/hwmon/wm83??.rst F: arch/arm/mach-s3c/mach-crag6410* F: drivers/clk/clk-wm83*.c diff --git a/include/sound/soc.h b/include/sound/soc.h index 675849d07284..8e6dd8a257c5 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -712,6 +712,12 @@ struct snd_soc_dai_link { /* Do not create a PCM for this DAI link (Backend link) */ unsigned int ignore:1; + /* This flag will reorder stop sequence. By enabling this flag + * DMA controller stop sequence will be invoked first followed by + * CPU DAI driver stop sequence + */ + unsigned int stop_dma_first:1; + #ifdef CONFIG_SND_SOC_TOPOLOGY struct snd_soc_dobj dobj; /* For topology */ #endif diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 8a13462e1a63..5dcf77af07af 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -36,6 +36,7 @@ config SND_SOC_COMPRESS config SND_SOC_TOPOLOGY bool + select SND_DYNAMIC_MINORS config SND_SOC_TOPOLOGY_KUNIT_TEST tristate "KUnit tests for SoC topology" diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index c130eeb07cdf..b3df98a9f9f3 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -525,6 +525,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { | SND_SOC_DAIFMT_CBM_CFM, .init = cz_da7219_init, .dpcm_playback = 1, + .stop_dma_first = 1, .ops = &cz_da7219_play_ops, SND_SOC_DAILINK_REG(designware1, dlgs, platform), }, @@ -534,6 +535,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, .dpcm_capture = 1, + .stop_dma_first = 1, .ops = &cz_da7219_cap_ops, SND_SOC_DAILINK_REG(designware2, dlgs, platform), }, @@ -543,6 +545,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, .dpcm_playback = 1, + .stop_dma_first = 1, .ops = &cz_max_play_ops, SND_SOC_DAILINK_REG(designware3, mx, platform), }, @@ -553,6 +556,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, .dpcm_capture = 1, + .stop_dma_first = 1, .ops = &cz_dmic0_cap_ops, SND_SOC_DAILINK_REG(designware3, adau, platform), }, @@ -563,6 +567,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, .dpcm_capture = 1, + .stop_dma_first = 1, .ops = &cz_dmic1_cap_ops, SND_SOC_DAILINK_REG(designware2, adau, platform), }, @@ -576,6 +581,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = { | SND_SOC_DAIFMT_CBM_CFM, .init = cz_rt5682_init, .dpcm_playback = 1, + .stop_dma_first = 1, .ops = &cz_rt5682_play_ops, SND_SOC_DAILINK_REG(designware1, rt5682, platform), }, @@ -585,6 +591,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = { .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, .dpcm_capture = 1, + .stop_dma_first = 1, .ops = &cz_rt5682_cap_ops, SND_SOC_DAILINK_REG(designware2, rt5682, platform), }, @@ -594,6 +601,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = { .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, .dpcm_playback = 1, + .stop_dma_first = 1, .ops = &cz_rt5682_max_play_ops, SND_SOC_DAILINK_REG(designware3, mx, platform), }, @@ -604,6 +612,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = { .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, .dpcm_capture = 1, + .stop_dma_first = 1, .ops = &cz_rt5682_dmic0_cap_ops, SND_SOC_DAILINK_REG(designware3, adau, platform), }, @@ -614,6 +623,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = { .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, .dpcm_capture = 1, + .stop_dma_first = 1, .ops = &cz_rt5682_dmic1_cap_ops, SND_SOC_DAILINK_REG(designware2, adau, platform), }, diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index ee8e9a3bcadf..11b3c4f39eba 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -969,7 +969,7 @@ static int acp_dma_hw_params(struct snd_soc_component *component, acp_set_sram_bank_state(rtd->acp_mmio, 0, true); /* Save for runtime private data */ - rtd->dma_addr = substream->dma_buffer.addr; + rtd->dma_addr = runtime->dma_addr; rtd->order = get_order(size); /* Fill the page table entries in ACP SRAM */ diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 4522d7ec22e7..75c06697fa09 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -286,7 +286,7 @@ static int acp3x_dma_hw_params(struct snd_soc_component *component, pr_err("pinfo failed\n"); } size = params_buffer_bytes(params); - rtd->dma_addr = substream->dma_buffer.addr; + rtd->dma_addr = substream->runtime->dma_addr; rtd->num_pages = (PAGE_ALIGN(size) >> PAGE_SHIFT); config_acp3x_dma(rtd, substream->stream); return 0; diff --git a/sound/soc/amd/renoir/acp3x-pdm-dma.c b/sound/soc/amd/renoir/acp3x-pdm-dma.c index 9988a50a81b0..9dd22a2fa2e5 100644 --- a/sound/soc/amd/renoir/acp3x-pdm-dma.c +++ b/sound/soc/amd/renoir/acp3x-pdm-dma.c @@ -242,7 +242,7 @@ static int acp_pdm_dma_hw_params(struct snd_soc_component *component, return -EINVAL; size = params_buffer_bytes(params); period_bytes = params_period_bytes(params); - rtd->dma_addr = substream->dma_buffer.addr; + rtd->dma_addr = substream->runtime->dma_addr; rtd->num_pages = (PAGE_ALIGN(size) >> PAGE_SHIFT); config_acp_dma(rtd, substream->stream); init_pdm_ring_buffer(MEM_WINDOW_START, size, period_bytes, diff --git a/sound/soc/amd/renoir/rn-pci-acp3x.c b/sound/soc/amd/renoir/rn-pci-acp3x.c index 19438da5dfa5..7b8040e812a1 100644 --- a/sound/soc/amd/renoir/rn-pci-acp3x.c +++ b/sound/soc/amd/renoir/rn-pci-acp3x.c @@ -382,6 +382,8 @@ static const struct dev_pm_ops rn_acp_pm = { .runtime_resume = snd_rn_acp_resume, .suspend = snd_rn_acp_suspend, .resume = snd_rn_acp_resume, + .restore = snd_rn_acp_resume, + .poweroff = snd_rn_acp_suspend, }; static void snd_rn_acp_remove(struct pci_dev *pci) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index b92b9ebad622..fe5e558635ad 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1575,7 +1575,9 @@ config SND_SOC_WCD934X Qualcomm SoCs like SDM845. config SND_SOC_WCD938X + depends on SND_SOC_WCD938X_SDW tristate + depends on SOUNDWIRE || !SOUNDWIRE config SND_SOC_WCD938X_SDW tristate "WCD9380/WCD9385 Codec - SDW" diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index d656b1405473..8dcea2c4604a 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -585,7 +585,10 @@ obj-$(CONFIG_SND_SOC_WCD_MBHC) += snd-soc-wcd-mbhc.o obj-$(CONFIG_SND_SOC_WCD9335) += snd-soc-wcd9335.o obj-$(CONFIG_SND_SOC_WCD934X) += snd-soc-wcd934x.o obj-$(CONFIG_SND_SOC_WCD938X) += snd-soc-wcd938x.o -obj-$(CONFIG_SND_SOC_WCD938X_SDW) += snd-soc-wcd938x-sdw.o +ifdef CONFIG_SND_SOC_WCD938X_SDW +# avoid link failure by forcing sdw code built-in when needed +obj-$(CONFIG_SND_SOC_WCD938X) += snd-soc-wcd938x-sdw.o +endif obj-$(CONFIG_SND_SOC_WL1273) += snd-soc-wl1273.o obj-$(CONFIG_SND_SOC_WM0010) += snd-soc-wm0010.o obj-$(CONFIG_SND_SOC_WM1250_EV1) += snd-soc-wm1250-ev1.o diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index eff013f295be..fb1e4c33e27d 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -405,7 +405,7 @@ static const struct regmap_config cs42l42_regmap = { .use_single_write = true, }; -static DECLARE_TLV_DB_SCALE(adc_tlv, -9600, 100, false); +static DECLARE_TLV_DB_SCALE(adc_tlv, -9700, 100, true); static DECLARE_TLV_DB_SCALE(mixer_tlv, -6300, 100, true); static const char * const cs42l42_hpf_freq_text[] = { @@ -425,34 +425,23 @@ static SOC_ENUM_SINGLE_DECL(cs42l42_wnf3_freq_enum, CS42L42_ADC_WNF_HPF_CTL, CS42L42_ADC_WNF_CF_SHIFT, cs42l42_wnf3_freq_text); -static const char * const cs42l42_wnf05_freq_text[] = { - "280Hz", "315Hz", "350Hz", "385Hz", - "420Hz", "455Hz", "490Hz", "525Hz" -}; - -static SOC_ENUM_SINGLE_DECL(cs42l42_wnf05_freq_enum, CS42L42_ADC_WNF_HPF_CTL, - CS42L42_ADC_WNF_CF_SHIFT, - cs42l42_wnf05_freq_text); - static const struct snd_kcontrol_new cs42l42_snd_controls[] = { /* ADC Volume and Filter Controls */ SOC_SINGLE("ADC Notch Switch", CS42L42_ADC_CTL, - CS42L42_ADC_NOTCH_DIS_SHIFT, true, false), + CS42L42_ADC_NOTCH_DIS_SHIFT, true, true), SOC_SINGLE("ADC Weak Force Switch", CS42L42_ADC_CTL, CS42L42_ADC_FORCE_WEAK_VCM_SHIFT, true, false), SOC_SINGLE("ADC Invert Switch", CS42L42_ADC_CTL, CS42L42_ADC_INV_SHIFT, true, false), SOC_SINGLE("ADC Boost Switch", CS42L42_ADC_CTL, CS42L42_ADC_DIG_BOOST_SHIFT, true, false), - SOC_SINGLE_SX_TLV("ADC Volume", CS42L42_ADC_VOLUME, - CS42L42_ADC_VOL_SHIFT, 0xA0, 0x6C, adc_tlv), + SOC_SINGLE_S8_TLV("ADC Volume", CS42L42_ADC_VOLUME, -97, 12, adc_tlv), SOC_SINGLE("ADC WNF Switch", CS42L42_ADC_WNF_HPF_CTL, CS42L42_ADC_WNF_EN_SHIFT, true, false), SOC_SINGLE("ADC HPF Switch", CS42L42_ADC_WNF_HPF_CTL, CS42L42_ADC_HPF_EN_SHIFT, true, false), SOC_ENUM("HPF Corner Freq", cs42l42_hpf_freq_enum), SOC_ENUM("WNF 3dB Freq", cs42l42_wnf3_freq_enum), - SOC_ENUM("WNF 05dB Freq", cs42l42_wnf05_freq_enum), /* DAC Volume and Filter Controls */ SOC_SINGLE("DACA Invert Switch", CS42L42_DAC_CTL1, @@ -471,8 +460,8 @@ static const struct snd_soc_dapm_widget cs42l42_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("HP"), SND_SOC_DAPM_DAC("DAC", NULL, CS42L42_PWR_CTL1, CS42L42_HP_PDN_SHIFT, 1), SND_SOC_DAPM_MIXER("MIXER", CS42L42_PWR_CTL1, CS42L42_MIXER_PDN_SHIFT, 1, NULL, 0), - SND_SOC_DAPM_AIF_IN("SDIN1", NULL, 0, CS42L42_ASP_RX_DAI0_EN, CS42L42_ASP_RX0_CH1_SHIFT, 0), - SND_SOC_DAPM_AIF_IN("SDIN2", NULL, 1, CS42L42_ASP_RX_DAI0_EN, CS42L42_ASP_RX0_CH2_SHIFT, 0), + SND_SOC_DAPM_AIF_IN("SDIN1", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("SDIN2", NULL, 1, SND_SOC_NOPM, 0, 0), /* Playback Requirements */ SND_SOC_DAPM_SUPPLY("ASP DAI0", CS42L42_PWR_CTL1, CS42L42_ASP_DAI_PDN_SHIFT, 1, NULL, 0), @@ -597,6 +586,7 @@ struct cs42l42_pll_params { * Table 4-5 from the Datasheet */ static const struct cs42l42_pll_params pll_ratio_table[] = { + { 1411200, 0, 1, 0x00, 0x80, 0x000000, 0x03, 0x10, 11289600, 128, 2}, { 1536000, 0, 1, 0x00, 0x7D, 0x000000, 0x03, 0x10, 12000000, 125, 2}, { 2304000, 0, 1, 0x00, 0x55, 0xC00000, 0x02, 0x10, 12288000, 85, 2}, { 2400000, 0, 1, 0x00, 0x50, 0x000000, 0x03, 0x10, 12000000, 80, 2}, @@ -630,6 +620,8 @@ static int cs42l42_pll_config(struct snd_soc_component *component) for (i = 0; i < ARRAY_SIZE(pll_ratio_table); i++) { if (pll_ratio_table[i].sclk == clk) { + cs42l42->pll_config = i; + /* Configure the internal sample rate */ snd_soc_component_update_bits(component, CS42L42_MCLK_CTL, CS42L42_INTERNAL_FS_MASK, @@ -638,14 +630,9 @@ static int cs42l42_pll_config(struct snd_soc_component *component) (pll_ratio_table[i].mclk_int != 24000000)) << CS42L42_INTERNAL_FS_SHIFT); - /* Set the MCLK src (PLL or SCLK) and the divide - * ratio - */ + snd_soc_component_update_bits(component, CS42L42_MCLK_SRC_SEL, - CS42L42_MCLK_SRC_SEL_MASK | CS42L42_MCLKDIV_MASK, - (pll_ratio_table[i].mclk_src_sel - << CS42L42_MCLK_SRC_SEL_SHIFT) | (pll_ratio_table[i].mclk_div << CS42L42_MCLKDIV_SHIFT)); /* Set up the LRCLK */ @@ -681,15 +668,6 @@ static int cs42l42_pll_config(struct snd_soc_component *component) CS42L42_FSYNC_PULSE_WIDTH_MASK, CS42L42_FRAC1_VAL(fsync - 1) << CS42L42_FSYNC_PULSE_WIDTH_SHIFT); - snd_soc_component_update_bits(component, - CS42L42_ASP_FRM_CFG, - CS42L42_ASP_5050_MASK, - CS42L42_ASP_5050_MASK); - /* Set the frame delay to 1.0 SCLK clocks */ - snd_soc_component_update_bits(component, CS42L42_ASP_FRM_CFG, - CS42L42_ASP_FSD_MASK, - CS42L42_ASP_FSD_1_0 << - CS42L42_ASP_FSD_SHIFT); /* Set the sample rates (96k or lower) */ snd_soc_component_update_bits(component, CS42L42_FS_RATE_EN, CS42L42_FS_EN_MASK, @@ -789,7 +767,18 @@ static int cs42l42_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) /* interface format */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - case SND_SOC_DAIFMT_LEFT_J: + /* + * 5050 mode, frame starts on falling edge of LRCLK, + * frame delayed by 1.0 SCLKs + */ + snd_soc_component_update_bits(component, + CS42L42_ASP_FRM_CFG, + CS42L42_ASP_STP_MASK | + CS42L42_ASP_5050_MASK | + CS42L42_ASP_FSD_MASK, + CS42L42_ASP_5050_MASK | + (CS42L42_ASP_FSD_1_0 << + CS42L42_ASP_FSD_SHIFT)); break; default: return -EINVAL; @@ -819,6 +808,25 @@ static int cs42l42_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) return 0; } +static int cs42l42_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component); + + /* + * Sample rates < 44.1 kHz would produce an out-of-range SCLK with + * a standard I2S frame. If the machine driver sets SCLK it must be + * legal. + */ + if (cs42l42->sclk) + return 0; + + /* Machine driver has not set a SCLK, limit bottom end to 44.1 kHz */ + return snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + 44100, 192000); +} + static int cs42l42_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -832,6 +840,17 @@ static int cs42l42_pcm_hw_params(struct snd_pcm_substream *substream, cs42l42->srate = params_rate(params); cs42l42->bclk = snd_soc_params_to_bclk(params); + /* I2S frame always has 2 channels even for mono audio */ + if (channels == 1) + cs42l42->bclk *= 2; + + /* + * Assume 24-bit samples are in 32-bit slots, to prevent SCLK being + * more than assumed (which would result in overclocking). + */ + if (params_width(params) == 24) + cs42l42->bclk = (cs42l42->bclk / 3) * 4; + switch(substream->stream) { case SNDRV_PCM_STREAM_CAPTURE: if (channels == 2) { @@ -855,6 +874,17 @@ static int cs42l42_pcm_hw_params(struct snd_pcm_substream *substream, snd_soc_component_update_bits(component, CS42L42_ASP_RX_DAI0_CH2_AP_RES, CS42L42_ASP_RX_CH_AP_MASK | CS42L42_ASP_RX_CH_RES_MASK, val); + + /* Channel B comes from the last active channel */ + snd_soc_component_update_bits(component, CS42L42_SP_RX_CH_SEL, + CS42L42_SP_RX_CHB_SEL_MASK, + (channels - 1) << CS42L42_SP_RX_CHB_SEL_SHIFT); + + /* Both LRCLK slots must be enabled */ + snd_soc_component_update_bits(component, CS42L42_ASP_RX_DAI0_EN, + CS42L42_ASP_RX0_CH_EN_MASK, + BIT(CS42L42_ASP_RX0_CH1_SHIFT) | + BIT(CS42L42_ASP_RX0_CH2_SHIFT)); break; default: break; @@ -868,10 +898,23 @@ static int cs42l42_set_sysclk(struct snd_soc_dai *dai, { struct snd_soc_component *component = dai->component; struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component); + int i; - cs42l42->sclk = freq; + if (freq == 0) { + cs42l42->sclk = 0; + return 0; + } - return 0; + for (i = 0; i < ARRAY_SIZE(pll_ratio_table); i++) { + if (pll_ratio_table[i].sclk == freq) { + cs42l42->sclk = freq; + return 0; + } + } + + dev_err(component->dev, "SCLK %u not supported\n", freq); + + return -EINVAL; } static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream) @@ -900,13 +943,21 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream) */ regmap_multi_reg_write(cs42l42->regmap, cs42l42_to_osc_seq, ARRAY_SIZE(cs42l42_to_osc_seq)); + + /* Must disconnect PLL before stopping it */ + snd_soc_component_update_bits(component, + CS42L42_MCLK_SRC_SEL, + CS42L42_MCLK_SRC_SEL_MASK, + 0); + usleep_range(100, 200); + snd_soc_component_update_bits(component, CS42L42_PLL_CTL1, CS42L42_PLL_START_MASK, 0); } } else { if (!cs42l42->stream_use) { /* SCLK must be running before codec unmute */ - if ((cs42l42->bclk < 11289600) && (cs42l42->sclk < 11289600)) { + if (pll_ratio_table[cs42l42->pll_config].mclk_src_sel) { snd_soc_component_update_bits(component, CS42L42_PLL_CTL1, CS42L42_PLL_START_MASK, 1); @@ -927,6 +978,12 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream) CS42L42_PLL_LOCK_TIMEOUT_US); if (ret < 0) dev_warn(component->dev, "PLL failed to lock: %d\n", ret); + + /* PLL must be running to drive glitchless switch logic */ + snd_soc_component_update_bits(component, + CS42L42_MCLK_SRC_SEL, + CS42L42_MCLK_SRC_SEL_MASK, + CS42L42_MCLK_SRC_SEL_MASK); } /* Mark SCLK as present, turn off internal oscillator */ @@ -960,8 +1017,8 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream) SNDRV_PCM_FMTBIT_S24_LE |\ SNDRV_PCM_FMTBIT_S32_LE ) - static const struct snd_soc_dai_ops cs42l42_ops = { + .startup = cs42l42_dai_startup, .hw_params = cs42l42_pcm_hw_params, .set_fmt = cs42l42_set_dai_fmt, .set_sysclk = cs42l42_set_sysclk, @@ -2070,4 +2127,7 @@ MODULE_DESCRIPTION("ASoC CS42L42 driver"); MODULE_AUTHOR("James Schulman, Cirrus Logic Inc, <james.schulman@cirrus.com>"); MODULE_AUTHOR("Brian Austin, Cirrus Logic Inc, <brian.austin@cirrus.com>"); MODULE_AUTHOR("Michael White, Cirrus Logic Inc, <michael.white@cirrus.com>"); +MODULE_AUTHOR("Lucas Tanure <tanureal@opensource.cirrus.com>"); +MODULE_AUTHOR("Richard Fitzgerald <rf@opensource.cirrus.com>"); +MODULE_AUTHOR("Vitaly Rodionov <vitalyr@opensource.cirrus.com>"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index 206b3c81d3e0..8734f6828f3e 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -653,6 +653,8 @@ /* Page 0x25 Audio Port Registers */ #define CS42L42_SP_RX_CH_SEL (CS42L42_PAGE_25 + 0x01) +#define CS42L42_SP_RX_CHB_SEL_SHIFT 2 +#define CS42L42_SP_RX_CHB_SEL_MASK (3 << CS42L42_SP_RX_CHB_SEL_SHIFT) #define CS42L42_SP_RX_ISOC_CTL (CS42L42_PAGE_25 + 0x02) #define CS42L42_SP_RX_RSYNC_SHIFT 6 @@ -775,6 +777,7 @@ struct cs42l42_private { struct gpio_desc *reset_gpio; struct completion pdn_done; struct snd_soc_jack *jack; + int pll_config; int bclk; u32 sclk; u32 srate; diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 15bd8335f667..db88be48c998 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -828,36 +828,6 @@ static void nau8824_int_status_clear_all(struct regmap *regmap) } } -static void nau8824_dapm_disable_pin(struct nau8824 *nau8824, const char *pin) -{ - struct snd_soc_dapm_context *dapm = nau8824->dapm; - const char *prefix = dapm->component->name_prefix; - char prefixed_pin[80]; - - if (prefix) { - snprintf(prefixed_pin, sizeof(prefixed_pin), "%s %s", - prefix, pin); - snd_soc_dapm_disable_pin(dapm, prefixed_pin); - } else { - snd_soc_dapm_disable_pin(dapm, pin); - } -} - -static void nau8824_dapm_enable_pin(struct nau8824 *nau8824, const char *pin) -{ - struct snd_soc_dapm_context *dapm = nau8824->dapm; - const char *prefix = dapm->component->name_prefix; - char prefixed_pin[80]; - - if (prefix) { - snprintf(prefixed_pin, sizeof(prefixed_pin), "%s %s", - prefix, pin); - snd_soc_dapm_force_enable_pin(dapm, prefixed_pin); - } else { - snd_soc_dapm_force_enable_pin(dapm, pin); - } -} - static void nau8824_eject_jack(struct nau8824 *nau8824) { struct snd_soc_dapm_context *dapm = nau8824->dapm; @@ -866,8 +836,8 @@ static void nau8824_eject_jack(struct nau8824 *nau8824) /* Clear all interruption status */ nau8824_int_status_clear_all(regmap); - nau8824_dapm_disable_pin(nau8824, "SAR"); - nau8824_dapm_disable_pin(nau8824, "MICBIAS"); + snd_soc_dapm_disable_pin(dapm, "SAR"); + snd_soc_dapm_disable_pin(dapm, "MICBIAS"); snd_soc_dapm_sync(dapm); /* Enable the insertion interruption, disable the ejection @@ -897,8 +867,8 @@ static void nau8824_jdet_work(struct work_struct *work) struct regmap *regmap = nau8824->regmap; int adc_value, event = 0, event_mask = 0; - nau8824_dapm_enable_pin(nau8824, "MICBIAS"); - nau8824_dapm_enable_pin(nau8824, "SAR"); + snd_soc_dapm_enable_pin(dapm, "MICBIAS"); + snd_soc_dapm_enable_pin(dapm, "SAR"); snd_soc_dapm_sync(dapm); msleep(100); @@ -909,8 +879,8 @@ static void nau8824_jdet_work(struct work_struct *work) if (adc_value < HEADSET_SARADC_THD) { event |= SND_JACK_HEADPHONE; - nau8824_dapm_disable_pin(nau8824, "SAR"); - nau8824_dapm_disable_pin(nau8824, "MICBIAS"); + snd_soc_dapm_disable_pin(dapm, "SAR"); + snd_soc_dapm_disable_pin(dapm, "MICBIAS"); snd_soc_dapm_sync(dapm); } else { event |= SND_JACK_HEADSET; diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index 3000bc128b5b..38356ea2bd6e 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -1695,6 +1695,8 @@ static const struct regmap_config rt5631_regmap_config = { .reg_defaults = rt5631_reg, .num_reg_defaults = ARRAY_SIZE(rt5631_reg), .cache_type = REGCACHE_RBTREE, + .use_single_read = true, + .use_single_write = true, }; static int rt5631_i2c_probe(struct i2c_client *i2c, diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index f50c0c8133d4..7dc01ae6bb66 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -44,6 +44,7 @@ static const struct reg_sequence patch_list[] = { {RT5682_I2C_CTRL, 0x000f}, {RT5682_PLL2_INTERNAL, 0x8266}, {RT5682_SAR_IL_CMD_3, 0x8365}, + {RT5682_SAR_IL_CMD_6, 0x0180}, }; void rt5682_apply_patch_list(struct rt5682_priv *rt5682, struct device *dev) @@ -973,10 +974,14 @@ int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert) rt5682_enable_push_button_irq(component, false); snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_LOW); - if (!snd_soc_dapm_get_pin_status(dapm, "MICBIAS")) + if (!snd_soc_dapm_get_pin_status(dapm, "MICBIAS") && + !snd_soc_dapm_get_pin_status(dapm, "PLL1") && + !snd_soc_dapm_get_pin_status(dapm, "PLL2B")) snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, RT5682_PWR_MB, 0); - if (!snd_soc_dapm_get_pin_status(dapm, "Vref2")) + if (!snd_soc_dapm_get_pin_status(dapm, "Vref2") && + !snd_soc_dapm_get_pin_status(dapm, "PLL1") && + !snd_soc_dapm_get_pin_status(dapm, "PLL2B")) snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, RT5682_PWR_VREF2, 0); snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3, diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 51870d50f419..52d2c968b5c0 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -35,6 +35,9 @@ #include "tlv320aic31xx.h" +static int aic31xx_set_jack(struct snd_soc_component *component, + struct snd_soc_jack *jack, void *data); + static const struct reg_default aic31xx_reg_defaults[] = { { AIC31XX_CLKMUX, 0x00 }, { AIC31XX_PLLPR, 0x11 }, @@ -1256,6 +1259,13 @@ static int aic31xx_power_on(struct snd_soc_component *component) return ret; } + /* + * The jack detection configuration is in the same register + * that is used to report jack detect status so is volatile + * and not covered by the cache sync, restore it separately. + */ + aic31xx_set_jack(component, aic31xx->jack, NULL); + return 0; } @@ -1604,6 +1614,8 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, ret); return ret; } + regcache_cache_only(aic31xx->regmap, true); + aic31xx->dev = &i2c->dev; aic31xx->irq = i2c->irq; diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 8c6a287927e2..d39c7d52ecfd 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -250,8 +250,8 @@ static DECLARE_TLV_DB_SCALE(tlv_pcm, -6350, 50, 0); static DECLARE_TLV_DB_SCALE(tlv_driver_gain, -600, 100, 0); /* -12dB min, 0.5dB steps */ static DECLARE_TLV_DB_SCALE(tlv_adc_vol, -1200, 50, 0); - -static DECLARE_TLV_DB_LINEAR(tlv_spk_vol, TLV_DB_GAIN_MUTE, 0); +/* -6dB min, 1dB steps */ +static DECLARE_TLV_DB_SCALE(tlv_tas_driver_gain, -5850, 50, 0); static DECLARE_TLV_DB_SCALE(tlv_amp_vol, 0, 600, 1); static const char * const lo_cm_text[] = { @@ -682,11 +682,20 @@ static int aic32x4_set_dosr(struct snd_soc_component *component, u16 dosr) static int aic32x4_set_processing_blocks(struct snd_soc_component *component, u8 r_block, u8 p_block) { - if (r_block > 18 || p_block > 25) - return -EINVAL; + struct aic32x4_priv *aic32x4 = snd_soc_component_get_drvdata(component); + + if (aic32x4->type == AIC32X4_TYPE_TAS2505) { + if (r_block || p_block > 3) + return -EINVAL; - snd_soc_component_write(component, AIC32X4_ADCSPB, r_block); - snd_soc_component_write(component, AIC32X4_DACSPB, p_block); + snd_soc_component_write(component, AIC32X4_DACSPB, p_block); + } else { /* AIC32x4 */ + if (r_block > 18 || p_block > 25) + return -EINVAL; + + snd_soc_component_write(component, AIC32X4_ADCSPB, r_block); + snd_soc_component_write(component, AIC32X4_DACSPB, p_block); + } return 0; } @@ -695,6 +704,7 @@ static int aic32x4_setup_clocks(struct snd_soc_component *component, unsigned int sample_rate, unsigned int channels, unsigned int bit_depth) { + struct aic32x4_priv *aic32x4 = snd_soc_component_get_drvdata(component); u8 aosr; u16 dosr; u8 adc_resource_class, dac_resource_class; @@ -721,19 +731,28 @@ static int aic32x4_setup_clocks(struct snd_soc_component *component, adc_resource_class = 6; dac_resource_class = 8; dosr_increment = 8; - aic32x4_set_processing_blocks(component, 1, 1); + if (aic32x4->type == AIC32X4_TYPE_TAS2505) + aic32x4_set_processing_blocks(component, 0, 1); + else + aic32x4_set_processing_blocks(component, 1, 1); } else if (sample_rate <= 96000) { aosr = 64; adc_resource_class = 6; dac_resource_class = 8; dosr_increment = 4; - aic32x4_set_processing_blocks(component, 1, 9); + if (aic32x4->type == AIC32X4_TYPE_TAS2505) + aic32x4_set_processing_blocks(component, 0, 1); + else + aic32x4_set_processing_blocks(component, 1, 9); } else if (sample_rate == 192000) { aosr = 32; adc_resource_class = 3; dac_resource_class = 4; dosr_increment = 2; - aic32x4_set_processing_blocks(component, 13, 19); + if (aic32x4->type == AIC32X4_TYPE_TAS2505) + aic32x4_set_processing_blocks(component, 0, 1); + else + aic32x4_set_processing_blocks(component, 13, 19); } else { dev_err(component->dev, "Sampling rate not supported\n"); return -EINVAL; @@ -1063,21 +1082,20 @@ static const struct snd_soc_component_driver soc_component_dev_aic32x4 = { }; static const struct snd_kcontrol_new aic32x4_tas2505_snd_controls[] = { - SOC_DOUBLE_R_S_TLV("PCM Playback Volume", AIC32X4_LDACVOL, - AIC32X4_LDACVOL, 0, -0x7f, 0x30, 7, 0, tlv_pcm), + SOC_SINGLE_S8_TLV("PCM Playback Volume", + AIC32X4_LDACVOL, -0x7f, 0x30, tlv_pcm), SOC_ENUM("DAC Playback PowerTune Switch", l_ptm_enum), - SOC_DOUBLE_R_S_TLV("HP Driver Playback Volume", AIC32X4_HPLGAIN, - AIC32X4_HPLGAIN, 0, -0x6, 0x1d, 5, 0, - tlv_driver_gain), - SOC_DOUBLE_R("HP DAC Playback Switch", AIC32X4_HPLGAIN, - AIC32X4_HPLGAIN, 6, 0x01, 1), - SOC_SINGLE("Auto-mute Switch", AIC32X4_DACMUTE, 4, 7, 0), + SOC_SINGLE_TLV("HP Driver Gain Volume", + AIC32X4_HPLGAIN, 0, 0x74, 1, tlv_tas_driver_gain), + SOC_SINGLE("HP DAC Playback Switch", AIC32X4_HPLGAIN, 6, 1, 1), - SOC_SINGLE_RANGE_TLV("Speaker Driver Playback Volume", TAS2505_SPKVOL1, - 0, 0, 117, 1, tlv_spk_vol), - SOC_SINGLE_TLV("Speaker Amplifier Playback Volume", TAS2505_SPKVOL2, - 4, 5, 0, tlv_amp_vol), + SOC_SINGLE_TLV("Speaker Driver Playback Volume", + TAS2505_SPKVOL1, 0, 0x74, 1, tlv_tas_driver_gain), + SOC_SINGLE_TLV("Speaker Amplifier Playback Volume", + TAS2505_SPKVOL2, 4, 5, 0, tlv_amp_vol), + + SOC_SINGLE("Auto-mute Switch", AIC32X4_DACMUTE, 4, 7, 0), }; static const struct snd_kcontrol_new hp_output_mixer_controls[] = { diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index 44c4bde84a67..f0daf8defcf1 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -4076,13 +4076,6 @@ static int wcd938x_soc_codec_probe(struct snd_soc_component *component) (WCD938X_DIGITAL_INTR_LEVEL_0 + i), 0); } - ret = wcd938x_irq_init(wcd938x, component->dev); - if (ret) { - dev_err(component->dev, "%s: IRQ init failed: %d\n", - __func__, ret); - return ret; - } - wcd938x->hphr_pdm_wd_int = regmap_irq_get_virq(wcd938x->irq_chip, WCD938X_IRQ_HPHR_PDM_WD_INT); wcd938x->hphl_pdm_wd_int = regmap_irq_get_virq(wcd938x->irq_chip, @@ -4342,7 +4335,6 @@ static int wcd938x_bind(struct device *dev) } wcd938x->sdw_priv[AIF1_PB] = dev_get_drvdata(wcd938x->rxdev); wcd938x->sdw_priv[AIF1_PB]->wcd938x = wcd938x; - wcd938x->sdw_priv[AIF1_PB]->slave_irq = wcd938x->virq; wcd938x->txdev = wcd938x_sdw_device_get(wcd938x->txnode); if (!wcd938x->txdev) { @@ -4351,7 +4343,6 @@ static int wcd938x_bind(struct device *dev) } wcd938x->sdw_priv[AIF1_CAP] = dev_get_drvdata(wcd938x->txdev); wcd938x->sdw_priv[AIF1_CAP]->wcd938x = wcd938x; - wcd938x->sdw_priv[AIF1_CAP]->slave_irq = wcd938x->virq; wcd938x->tx_sdw_dev = dev_to_sdw_dev(wcd938x->txdev); if (!wcd938x->tx_sdw_dev) { dev_err(dev, "could not get txslave with matching of dev\n"); @@ -4384,6 +4375,15 @@ static int wcd938x_bind(struct device *dev) return PTR_ERR(wcd938x->regmap); } + ret = wcd938x_irq_init(wcd938x, dev); + if (ret) { + dev_err(dev, "%s: IRQ init failed: %d\n", __func__, ret); + return ret; + } + + wcd938x->sdw_priv[AIF1_PB]->slave_irq = wcd938x->virq; + wcd938x->sdw_priv[AIF1_CAP]->slave_irq = wcd938x->virq; + ret = wcd938x_set_micbias_data(wcd938x); if (ret < 0) { dev_err(dev, "%s: bad micbias pdata\n", __func__); diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index b395df1eb72d..bbe27ab3b1fc 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -747,7 +747,6 @@ static void wm_adsp2_init_debugfs(struct wm_adsp *dsp, static void wm_adsp2_cleanup_debugfs(struct wm_adsp *dsp) { wm_adsp_debugfs_clear(dsp); - debugfs_remove_recursive(dsp->debugfs_root); } #else static inline void wm_adsp2_init_debugfs(struct wm_adsp *dsp, diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 4124aa2fc247..5db2f4865bbb 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -127,7 +127,7 @@ static void sst_fill_alloc_params(struct snd_pcm_substream *substream, snd_pcm_uframes_t period_size; ssize_t periodbytes; ssize_t buffer_bytes = snd_pcm_lib_buffer_bytes(substream); - u32 buffer_addr = virt_to_phys(substream->dma_buffer.area); + u32 buffer_addr = substream->runtime->dma_addr; channels = substream->runtime->channels; period_size = substream->runtime->period_size; @@ -233,7 +233,6 @@ static int sst_platform_alloc_stream(struct snd_pcm_substream *substream, /* set codec params and inform SST driver the same */ sst_fill_pcm_params(substream, ¶m); sst_fill_alloc_params(substream, &alloc_params); - substream->runtime->dma_area = substream->dma_buffer.area; str_params.sparams = param; str_params.aparams = alloc_params; str_params.codec = SST_CODEC_TYPE_PCM; diff --git a/sound/soc/intel/boards/sof_da7219_max98373.c b/sound/soc/intel/boards/sof_da7219_max98373.c index 896251d742fe..b7b3b0bf994a 100644 --- a/sound/soc/intel/boards/sof_da7219_max98373.c +++ b/sound/soc/intel/boards/sof_da7219_max98373.c @@ -404,7 +404,7 @@ static int audio_probe(struct platform_device *pdev) return -ENOMEM; /* By default dais[0] is configured for max98373 */ - if (!strcmp(pdev->name, "sof_da7219_max98360a")) { + if (!strcmp(pdev->name, "sof_da7219_mx98360a")) { dais[0] = (struct snd_soc_dai_link) { .name = "SSP1-Codec", .id = 0, diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index c2a5933bfcfc..700a18561a94 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -104,8 +104,6 @@ static int kirkwood_dma_open(struct snd_soc_component *component, int err; struct snd_pcm_runtime *runtime = substream->runtime; struct kirkwood_dma_data *priv = kirkwood_priv(substream); - const struct mbus_dram_target_info *dram; - unsigned long addr; snd_soc_set_runtime_hwparams(substream, &kirkwood_dma_snd_hw); @@ -142,20 +140,14 @@ static int kirkwood_dma_open(struct snd_soc_component *component, writel((unsigned int)-1, priv->io + KIRKWOOD_ERR_MASK); } - dram = mv_mbus_dram_info(); - addr = substream->dma_buffer.addr; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { if (priv->substream_play) return -EBUSY; priv->substream_play = substream; - kirkwood_dma_conf_mbus_windows(priv->io, - KIRKWOOD_PLAYBACK_WIN, addr, dram); } else { if (priv->substream_rec) return -EBUSY; priv->substream_rec = substream; - kirkwood_dma_conf_mbus_windows(priv->io, - KIRKWOOD_RECORD_WIN, addr, dram); } return 0; @@ -182,6 +174,23 @@ static int kirkwood_dma_close(struct snd_soc_component *component, return 0; } +static int kirkwood_dma_hw_params(struct snd_soc_component *component, + struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct kirkwood_dma_data *priv = kirkwood_priv(substream); + const struct mbus_dram_target_info *dram = mv_mbus_dram_info(); + unsigned long addr = substream->runtime->dma_addr; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + kirkwood_dma_conf_mbus_windows(priv->io, + KIRKWOOD_PLAYBACK_WIN, addr, dram); + else + kirkwood_dma_conf_mbus_windows(priv->io, + KIRKWOOD_RECORD_WIN, addr, dram); + return 0; +} + static int kirkwood_dma_prepare(struct snd_soc_component *component, struct snd_pcm_substream *substream) { @@ -246,6 +255,7 @@ const struct snd_soc_component_driver kirkwood_soc_component = { .name = DRV_NAME, .open = kirkwood_dma_open, .close = kirkwood_dma_close, + .hw_params = kirkwood_dma_hw_params, .prepare = kirkwood_dma_prepare, .pointer = kirkwood_dma_pointer, .pcm_construct = kirkwood_dma_new, diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c index 3a5e84e16a87..c8dfd0de30e4 100644 --- a/sound/soc/soc-component.c +++ b/sound/soc/soc-component.c @@ -148,86 +148,75 @@ int snd_soc_component_set_bias_level(struct snd_soc_component *component, return soc_component_ret(component, ret); } -static int soc_component_pin(struct snd_soc_component *component, - const char *pin, - int (*pin_func)(struct snd_soc_dapm_context *dapm, - const char *pin)) -{ - struct snd_soc_dapm_context *dapm = - snd_soc_component_get_dapm(component); - char *full_name; - int ret; - - if (!component->name_prefix) { - ret = pin_func(dapm, pin); - goto end; - } - - full_name = kasprintf(GFP_KERNEL, "%s %s", component->name_prefix, pin); - if (!full_name) { - ret = -ENOMEM; - goto end; - } - - ret = pin_func(dapm, full_name); - kfree(full_name); -end: - return soc_component_ret(component, ret); -} - int snd_soc_component_enable_pin(struct snd_soc_component *component, const char *pin) { - return soc_component_pin(component, pin, snd_soc_dapm_enable_pin); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + return snd_soc_dapm_enable_pin(dapm, pin); } EXPORT_SYMBOL_GPL(snd_soc_component_enable_pin); int snd_soc_component_enable_pin_unlocked(struct snd_soc_component *component, const char *pin) { - return soc_component_pin(component, pin, snd_soc_dapm_enable_pin_unlocked); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + return snd_soc_dapm_enable_pin_unlocked(dapm, pin); } EXPORT_SYMBOL_GPL(snd_soc_component_enable_pin_unlocked); int snd_soc_component_disable_pin(struct snd_soc_component *component, const char *pin) { - return soc_component_pin(component, pin, snd_soc_dapm_disable_pin); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + return snd_soc_dapm_disable_pin(dapm, pin); } EXPORT_SYMBOL_GPL(snd_soc_component_disable_pin); int snd_soc_component_disable_pin_unlocked(struct snd_soc_component *component, const char *pin) { - return soc_component_pin(component, pin, snd_soc_dapm_disable_pin_unlocked); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + return snd_soc_dapm_disable_pin_unlocked(dapm, pin); } EXPORT_SYMBOL_GPL(snd_soc_component_disable_pin_unlocked); int snd_soc_component_nc_pin(struct snd_soc_component *component, const char *pin) { - return soc_component_pin(component, pin, snd_soc_dapm_nc_pin); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + return snd_soc_dapm_nc_pin(dapm, pin); } EXPORT_SYMBOL_GPL(snd_soc_component_nc_pin); int snd_soc_component_nc_pin_unlocked(struct snd_soc_component *component, const char *pin) { - return soc_component_pin(component, pin, snd_soc_dapm_nc_pin_unlocked); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + return snd_soc_dapm_nc_pin_unlocked(dapm, pin); } EXPORT_SYMBOL_GPL(snd_soc_component_nc_pin_unlocked); int snd_soc_component_get_pin_status(struct snd_soc_component *component, const char *pin) { - return soc_component_pin(component, pin, snd_soc_dapm_get_pin_status); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + return snd_soc_dapm_get_pin_status(dapm, pin); } EXPORT_SYMBOL_GPL(snd_soc_component_get_pin_status); int snd_soc_component_force_enable_pin(struct snd_soc_component *component, const char *pin) { - return soc_component_pin(component, pin, snd_soc_dapm_force_enable_pin); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + return snd_soc_dapm_force_enable_pin(dapm, pin); } EXPORT_SYMBOL_GPL(snd_soc_component_force_enable_pin); @@ -235,7 +224,9 @@ int snd_soc_component_force_enable_pin_unlocked( struct snd_soc_component *component, const char *pin) { - return soc_component_pin(component, pin, snd_soc_dapm_force_enable_pin_unlocked); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + return snd_soc_dapm_force_enable_pin_unlocked(dapm, pin); } EXPORT_SYMBOL_GPL(snd_soc_component_force_enable_pin_unlocked); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 69893bd5be60..48f71bb81a2f 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1014,6 +1014,7 @@ out: static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); int ret = -EINVAL, _ret = 0; int rollback = 0; @@ -1054,14 +1055,23 @@ start_err: case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - ret = snd_soc_pcm_dai_trigger(substream, cmd, rollback); - if (ret < 0) - break; + if (rtd->dai_link->stop_dma_first) { + ret = snd_soc_pcm_component_trigger(substream, cmd, rollback); + if (ret < 0) + break; - ret = snd_soc_pcm_component_trigger(substream, cmd, rollback); - if (ret < 0) - break; + ret = snd_soc_pcm_dai_trigger(substream, cmd, rollback); + if (ret < 0) + break; + } else { + ret = snd_soc_pcm_dai_trigger(substream, cmd, rollback); + if (ret < 0) + break; + ret = snd_soc_pcm_component_trigger(substream, cmd, rollback); + if (ret < 0) + break; + } ret = snd_soc_link_trigger(substream, cmd, rollback); break; } diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index 4bce89b5ea40..4447f515e8b1 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -278,6 +278,8 @@ config SND_SOC_SOF_HDA config SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE tristate + select SOUNDWIRE_INTEL if SND_SOC_SOF_INTEL_SOUNDWIRE + select SND_INTEL_SOUNDWIRE_ACPI if SND_SOC_SOF_INTEL_SOUNDWIRE config SND_SOC_SOF_INTEL_SOUNDWIRE tristate "SOF support for SoundWire" @@ -285,8 +287,6 @@ config SND_SOC_SOF_INTEL_SOUNDWIRE depends on SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE depends on ACPI && SOUNDWIRE depends on !(SOUNDWIRE=m && SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE=y) - select SOUNDWIRE_INTEL - select SND_INTEL_SOUNDWIRE_ACPI help This adds support for SoundWire with Sound Open Firmware for Intel(R) platforms. diff --git a/sound/soc/sof/intel/hda-ipc.c b/sound/soc/sof/intel/hda-ipc.c index c91aa951df22..acfeca42604c 100644 --- a/sound/soc/sof/intel/hda-ipc.c +++ b/sound/soc/sof/intel/hda-ipc.c @@ -107,8 +107,8 @@ void hda_dsp_ipc_get_reply(struct snd_sof_dev *sdev) } else { /* reply correct size ? */ if (reply.hdr.size != msg->reply_size && - /* getter payload is never known upfront */ - !(reply.hdr.cmd & SOF_IPC_GLB_PROBE)) { + /* getter payload is never known upfront */ + ((reply.hdr.cmd & SOF_GLB_TYPE_MASK) != SOF_IPC_GLB_PROBE)) { dev_err(sdev->dev, "error: reply expected %zu got %u bytes\n", msg->reply_size, reply.hdr.size); ret = -EINVAL; diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index b4e35fbbe693..f60e2c57d3d0 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -200,12 +200,16 @@ static int hda_sdw_probe(struct snd_sof_dev *sdev) int hda_sdw_startup(struct snd_sof_dev *sdev) { struct sof_intel_hda_dev *hdev; + struct snd_sof_pdata *pdata = sdev->pdata; hdev = sdev->pdata->hw_pdata; if (!hdev->sdw) return 0; + if (pdata->machine && !pdata->machine->mach_params.link_mask) + return 0; + return sdw_intel_startup(hdev->sdw); } @@ -1015,6 +1019,14 @@ static int hda_generic_machine_select(struct snd_sof_dev *sdev) hda_mach->mach_params.dmic_num = dmic_num; pdata->machine = hda_mach; pdata->tplg_filename = tplg_filename; + + if (codec_num == 2) { + /* + * Prevent SoundWire links from starting when an external + * HDaudio codec is used + */ + hda_mach->mach_params.link_mask = 0; + } } } diff --git a/sound/soc/sof/intel/pci-tgl.c b/sound/soc/sof/intel/pci-tgl.c index a00262184efa..d04ce84fe7cc 100644 --- a/sound/soc/sof/intel/pci-tgl.c +++ b/sound/soc/sof/intel/pci-tgl.c @@ -89,6 +89,7 @@ static const struct sof_dev_desc adls_desc = { static const struct sof_dev_desc adl_desc = { .machines = snd_soc_acpi_intel_adl_machines, .alt_machines = snd_soc_acpi_intel_adl_sdw_machines, + .use_acpi_target_states = true, .resindex_lpe_base = 0, .resindex_pcicfg_base = -1, .resindex_imr_base = -1, diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c index 573374b89b10..d3276b4595af 100644 --- a/sound/soc/tegra/tegra_pcm.c +++ b/sound/soc/tegra/tegra_pcm.c @@ -213,19 +213,19 @@ snd_pcm_uframes_t tegra_pcm_pointer(struct snd_soc_component *component, } EXPORT_SYMBOL_GPL(tegra_pcm_pointer); -static int tegra_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream, +static int tegra_pcm_preallocate_dma_buffer(struct device *dev, struct snd_pcm *pcm, int stream, size_t size) { struct snd_pcm_substream *substream = pcm->streams[stream].substream; struct snd_dma_buffer *buf = &substream->dma_buffer; - buf->area = dma_alloc_wc(pcm->card->dev, size, &buf->addr, GFP_KERNEL); + buf->area = dma_alloc_wc(dev, size, &buf->addr, GFP_KERNEL); if (!buf->area) return -ENOMEM; buf->private_data = NULL; buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; + buf->dev.dev = dev; buf->bytes = size; return 0; @@ -244,31 +244,28 @@ static void tegra_pcm_deallocate_dma_buffer(struct snd_pcm *pcm, int stream) if (!buf->area) return; - dma_free_wc(pcm->card->dev, buf->bytes, buf->area, buf->addr); + dma_free_wc(buf->dev.dev, buf->bytes, buf->area, buf->addr); buf->area = NULL; } -static int tegra_pcm_dma_allocate(struct snd_soc_pcm_runtime *rtd, +static int tegra_pcm_dma_allocate(struct device *dev, struct snd_soc_pcm_runtime *rtd, size_t size) { - struct snd_card *card = rtd->card->snd_card; struct snd_pcm *pcm = rtd->pcm; int ret; - ret = dma_set_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); + ret = dma_set_mask_and_coherent(dev, DMA_BIT_MASK(32)); if (ret < 0) return ret; if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { - ret = tegra_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK, size); + ret = tegra_pcm_preallocate_dma_buffer(dev, pcm, SNDRV_PCM_STREAM_PLAYBACK, size); if (ret) goto err; } if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { - ret = tegra_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE, size); + ret = tegra_pcm_preallocate_dma_buffer(dev, pcm, SNDRV_PCM_STREAM_CAPTURE, size); if (ret) goto err_free_play; } @@ -284,7 +281,16 @@ err: int tegra_pcm_construct(struct snd_soc_component *component, struct snd_soc_pcm_runtime *rtd) { - return tegra_pcm_dma_allocate(rtd, tegra_pcm_hardware.buffer_bytes_max); + struct device *dev = component->dev; + + /* + * Fallback for backwards-compatibility with older device trees that + * have the iommus property in the virtual, top-level "sound" node. + */ + if (!of_get_property(dev->of_node, "iommus", NULL)) + dev = rtd->card->snd_card->dev; + + return tegra_pcm_dma_allocate(dev, rtd, tegra_pcm_hardware.buffer_bytes_max); } EXPORT_SYMBOL_GPL(tegra_pcm_construct); diff --git a/sound/soc/ti/j721e-evm.c b/sound/soc/ti/j721e-evm.c index 7d4e2e241f6a..9347f982c3e1 100644 --- a/sound/soc/ti/j721e-evm.c +++ b/sound/soc/ti/j721e-evm.c @@ -200,7 +200,7 @@ static int j721e_configure_refclk(struct j721e_priv *priv, return ret; } - if (priv->hsdiv_rates[domain->parent_clk_id] != scki) { + if (domain->parent_clk_id == -1 || priv->hsdiv_rates[domain->parent_clk_id] != scki) { dev_dbg(priv->dev, "domain%u configuration for %u Hz: %s, %dxFS (SCKI: %u Hz)\n", audio_domain, rate, @@ -281,23 +281,29 @@ static int j721e_audio_startup(struct snd_pcm_substream *substream) j721e_rule_rate, &priv->rate_range, SNDRV_PCM_HW_PARAM_RATE, -1); - mutex_unlock(&priv->mutex); if (ret) - return ret; + goto out; /* Reset TDM slots to 32 */ ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 32); if (ret && ret != -ENOTSUPP) - return ret; + goto out; for_each_rtd_codec_dais(rtd, i, codec_dai) { ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x3, 0x3, 2, 32); if (ret && ret != -ENOTSUPP) - return ret; + goto out; } - return 0; + if (ret == -ENOTSUPP) + ret = 0; +out: + if (ret) + domain->active--; + mutex_unlock(&priv->mutex); + + return ret; } static int j721e_audio_hw_params(struct snd_pcm_substream *substream, diff --git a/sound/soc/uniphier/aio-dma.c b/sound/soc/uniphier/aio-dma.c index 3c1628a3a1ac..3d9736e7381f 100644 --- a/sound/soc/uniphier/aio-dma.c +++ b/sound/soc/uniphier/aio-dma.c @@ -198,7 +198,7 @@ static int uniphier_aiodma_mmap(struct snd_soc_component *component, vma->vm_page_prot = pgprot_writecombine(vma->vm_page_prot); return remap_pfn_range(vma, vma->vm_start, - substream->dma_buffer.addr >> PAGE_SHIFT, + substream->runtime->dma_addr >> PAGE_SHIFT, vma->vm_end - vma->vm_start, vma->vm_page_prot); } diff --git a/sound/soc/xilinx/xlnx_formatter_pcm.c b/sound/soc/xilinx/xlnx_formatter_pcm.c index 1d59fb668c77..91afea9d5de6 100644 --- a/sound/soc/xilinx/xlnx_formatter_pcm.c +++ b/sound/soc/xilinx/xlnx_formatter_pcm.c @@ -452,8 +452,8 @@ static int xlnx_formatter_pcm_hw_params(struct snd_soc_component *component, stream_data->buffer_size = size; - low = lower_32_bits(substream->dma_buffer.addr); - high = upper_32_bits(substream->dma_buffer.addr); + low = lower_32_bits(runtime->dma_addr); + high = upper_32_bits(runtime->dma_addr); writel(low, stream_data->mmio + XLNX_AUD_BUFF_ADDR_LSB); writel(high, stream_data->mmio + XLNX_AUD_BUFF_ADDR_MSB); |