diff options
34 files changed, 957 insertions, 322 deletions
diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8962.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8962.yaml index 0e6249d7c133..5e172e9462b9 100644 --- a/Documentation/devicetree/bindings/sound/wlf,wm8962.yaml +++ b/Documentation/devicetree/bindings/sound/wlf,wm8962.yaml @@ -19,6 +19,9 @@ properties: clocks: maxItems: 1 + interrupts: + maxItems: 1 + "#sound-dai-cells": const: 0 diff --git a/include/sound/soc-acpi.h b/include/sound/soc-acpi.h index 31f4c4f9aeea..ac0893df9c76 100644 --- a/include/sound/soc-acpi.h +++ b/include/sound/soc-acpi.h @@ -147,7 +147,7 @@ struct snd_soc_acpi_link_adr { */ /* Descriptor for SST ASoC machine driver */ struct snd_soc_acpi_mach { - const u8 id[ACPI_ID_LEN]; + u8 id[ACPI_ID_LEN]; const struct snd_soc_acpi_codecs *comp_ids; const u32 link_mask; const struct snd_soc_acpi_link_adr *links; diff --git a/sound/soc/amd/yc/pci-acp6x.c b/sound/soc/amd/yc/pci-acp6x.c index 957eeb6fb8e3..7e9a9a9d8ddd 100644 --- a/sound/soc/amd/yc/pci-acp6x.c +++ b/sound/soc/amd/yc/pci-acp6x.c @@ -146,10 +146,11 @@ static int snd_acp6x_probe(struct pci_dev *pci, { struct acp6x_dev_data *adata; struct platform_device_info pdevinfo[ACP6x_DEVS]; - int ret, index; + int index = 0; int val = 0x00; u32 addr; unsigned int irqflags; + int ret; irqflags = IRQF_SHARED; /* Yellow Carp device check */ diff --git a/sound/soc/codecs/cs35l41-spi.c b/sound/soc/codecs/cs35l41-spi.c index 5d6cf39abec4..c202d9df70ee 100644 --- a/sound/soc/codecs/cs35l41-spi.c +++ b/sound/soc/codecs/cs35l41-spi.c @@ -26,34 +26,6 @@ static const struct spi_device_id cs35l41_id_spi[] = { MODULE_DEVICE_TABLE(spi, cs35l41_id_spi); -static void cs35l41_spi_otp_setup(struct cs35l41_private *cs35l41, - bool is_pre_setup, unsigned int *freq) -{ - struct spi_device *spi; - u32 orig_spi_freq; - - spi = to_spi_device(cs35l41->dev); - - if (!spi) { - dev_err(cs35l41->dev, "%s: No SPI device\n", __func__); - return; - } - - if (is_pre_setup) { - orig_spi_freq = spi->max_speed_hz; - if (orig_spi_freq > CS35L41_SPI_MAX_FREQ_OTP) { - spi->max_speed_hz = CS35L41_SPI_MAX_FREQ_OTP; - spi_setup(spi); - } - *freq = orig_spi_freq; - } else { - if (spi->max_speed_hz != *freq) { - spi->max_speed_hz = *freq; - spi_setup(spi); - } - } -} - static int cs35l41_spi_probe(struct spi_device *spi) { const struct regmap_config *regmap_config = &cs35l41_regmap_spi; @@ -65,6 +37,9 @@ static int cs35l41_spi_probe(struct spi_device *spi) if (!cs35l41) return -ENOMEM; + spi->max_speed_hz = CS35L41_SPI_MAX_FREQ; + spi_setup(spi); + spi_set_drvdata(spi, cs35l41); cs35l41->regmap = devm_regmap_init_spi(spi, regmap_config); if (IS_ERR(cs35l41->regmap)) { @@ -75,7 +50,6 @@ static int cs35l41_spi_probe(struct spi_device *spi) cs35l41->dev = &spi->dev; cs35l41->irq = spi->irq; - cs35l41->otp_setup = cs35l41_spi_otp_setup; return cs35l41_probe(cs35l41, pdata); } diff --git a/sound/soc/codecs/cs35l41.c b/sound/soc/codecs/cs35l41.c index e04924526883..60332eae1162 100644 --- a/sound/soc/codecs/cs35l41.c +++ b/sound/soc/codecs/cs35l41.c @@ -438,7 +438,6 @@ static int cs35l41_otp_unpack(void *data) const struct cs35l41_otp_packed_element_t *otp_map; struct cs35l41_private *cs35l41 = data; int bit_offset, word_offset, ret, i; - unsigned int orig_spi_freq; unsigned int bit_sum = 8; u32 otp_val, otp_id_reg; u32 *otp_mem; @@ -462,9 +461,6 @@ static int cs35l41_otp_unpack(void *data) goto err_otp_unpack; } - if (cs35l41->otp_setup) - cs35l41->otp_setup(cs35l41, true, &orig_spi_freq); - ret = regmap_bulk_read(cs35l41->regmap, CS35L41_OTP_MEM0, otp_mem, CS35L41_OTP_SIZE_WORDS); if (ret < 0) { @@ -472,9 +468,6 @@ static int cs35l41_otp_unpack(void *data) goto err_otp_unpack; } - if (cs35l41->otp_setup) - cs35l41->otp_setup(cs35l41, false, &orig_spi_freq); - otp_map = otp_map_match->map; bit_offset = otp_map_match->bit_offset; diff --git a/sound/soc/codecs/cs35l41.h b/sound/soc/codecs/cs35l41.h index f82075ea855f..c7c45f19754b 100644 --- a/sound/soc/codecs/cs35l41.h +++ b/sound/soc/codecs/cs35l41.h @@ -728,7 +728,7 @@ #define CS35L41_FS2_WINDOW_MASK 0x00FFF800 #define CS35L41_FS2_WINDOW_SHIFT 12 -#define CS35L41_SPI_MAX_FREQ_OTP 4000000 +#define CS35L41_SPI_MAX_FREQ 4000000 #define CS35L41_RX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) #define CS35L41_TX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) @@ -781,8 +781,6 @@ struct cs35l41_private { int irq; /* GPIO for /RST */ struct gpio_desc *reset_gpio; - void (*otp_setup)(struct cs35l41_private *cs35l41, bool is_pre_setup, - unsigned int *freq); }; int cs35l41_probe(struct cs35l41_private *cs35l41, diff --git a/sound/soc/codecs/lpass-rx-macro.c b/sound/soc/codecs/lpass-rx-macro.c index 2bed5cf229be..aec5127260fd 100644 --- a/sound/soc/codecs/lpass-rx-macro.c +++ b/sound/soc/codecs/lpass-rx-macro.c @@ -2188,7 +2188,7 @@ static int rx_macro_config_classh(struct snd_soc_component *component, snd_soc_component_update_bits(component, CDC_RX_CLSH_DECAY_CTRL, CDC_RX_CLSH_DECAY_RATE_MASK, 0x0); - snd_soc_component_update_bits(component, + snd_soc_component_write_field(component, CDC_RX_RX1_RX_PATH_CFG0, CDC_RX_RXn_CLSH_EN_MASK, 0x1); break; diff --git a/sound/soc/codecs/rk817_codec.c b/sound/soc/codecs/rk817_codec.c index 943d7d933e81..03f24edfe4f6 100644 --- a/sound/soc/codecs/rk817_codec.c +++ b/sound/soc/codecs/rk817_codec.c @@ -539,3 +539,4 @@ module_platform_driver(rk817_codec_driver); MODULE_DESCRIPTION("ASoC RK817 codec driver"); MODULE_AUTHOR("binyuan <kevan.lan@rock-chips.com>"); MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:rk817-codec"); diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 04cb747c2b12..5224123d0d3b 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -2858,6 +2858,8 @@ int rt5682_register_dai_clks(struct rt5682_priv *rt5682) for (i = 0; i < RT5682_DAI_NUM_CLKS; ++i) { struct clk_init_data init = { }; + struct clk_parent_data parent_data; + const struct clk_hw *parent; dai_clk_hw = &rt5682->dai_clks_hw[i]; @@ -2865,17 +2867,17 @@ int rt5682_register_dai_clks(struct rt5682_priv *rt5682) case RT5682_DAI_WCLK_IDX: /* Make MCLK the parent of WCLK */ if (rt5682->mclk) { - init.parent_data = &(struct clk_parent_data){ + parent_data = (struct clk_parent_data){ .fw_name = "mclk", }; + init.parent_data = &parent_data; init.num_parents = 1; } break; case RT5682_DAI_BCLK_IDX: /* Make WCLK the parent of BCLK */ - init.parent_hws = &(const struct clk_hw *){ - &rt5682->dai_clks_hw[RT5682_DAI_WCLK_IDX] - }; + parent = &rt5682->dai_clks_hw[RT5682_DAI_WCLK_IDX]; + init.parent_hws = &parent; init.num_parents = 1; break; default: diff --git a/sound/soc/codecs/rt5682s.c b/sound/soc/codecs/rt5682s.c index 470957fcad6b..d49a4f68566d 100644 --- a/sound/soc/codecs/rt5682s.c +++ b/sound/soc/codecs/rt5682s.c @@ -2693,6 +2693,8 @@ static int rt5682s_register_dai_clks(struct snd_soc_component *component) for (i = 0; i < RT5682S_DAI_NUM_CLKS; ++i) { struct clk_init_data init = { }; + struct clk_parent_data parent_data; + const struct clk_hw *parent; dai_clk_hw = &rt5682s->dai_clks_hw[i]; @@ -2700,17 +2702,17 @@ static int rt5682s_register_dai_clks(struct snd_soc_component *component) case RT5682S_DAI_WCLK_IDX: /* Make MCLK the parent of WCLK */ if (rt5682s->mclk) { - init.parent_data = &(struct clk_parent_data){ + parent_data = (struct clk_parent_data){ .fw_name = "mclk", }; + init.parent_data = &parent_data; init.num_parents = 1; } break; case RT5682S_DAI_BCLK_IDX: /* Make WCLK the parent of BCLK */ - init.parent_hws = &(const struct clk_hw *){ - &rt5682s->dai_clks_hw[RT5682S_DAI_WCLK_IDX] - }; + parent = &rt5682s->dai_clks_hw[RT5682S_DAI_WCLK_IDX]; + init.parent_hws = &parent; init.num_parents = 1; break; default: diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c index c496b359f2f4..4f568abd59e2 100644 --- a/sound/soc/codecs/wcd934x.c +++ b/sound/soc/codecs/wcd934x.c @@ -1896,9 +1896,8 @@ static int wcd934x_hw_params(struct snd_pcm_substream *substream, } wcd->dai[dai->id].sconfig.rate = params_rate(params); - wcd934x_slim_set_hw_params(wcd, &wcd->dai[dai->id], substream->stream); - return 0; + return wcd934x_slim_set_hw_params(wcd, &wcd->dai[dai->id], substream->stream); } static int wcd934x_hw_free(struct snd_pcm_substream *substream, diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index 52de7d14b139..67151c7770c6 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -1174,6 +1174,9 @@ static bool wcd938x_readonly_register(struct device *dev, unsigned int reg) case WCD938X_DIGITAL_INTR_STATUS_0: case WCD938X_DIGITAL_INTR_STATUS_1: case WCD938X_DIGITAL_INTR_STATUS_2: + case WCD938X_DIGITAL_INTR_CLEAR_0: + case WCD938X_DIGITAL_INTR_CLEAR_1: + case WCD938X_DIGITAL_INTR_CLEAR_2: case WCD938X_DIGITAL_SWR_HM_TEST_0: case WCD938X_DIGITAL_SWR_HM_TEST_1: case WCD938X_DIGITAL_EFUSE_T_DATA_0: diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index f084b093cff6..c3112bf23866 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -602,8 +602,9 @@ static int wm_adsp_control_add(struct cs_dsp_coeff_ctl *cs_ctl) switch (cs_dsp->fw_ver) { case 0: case 1: - snprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "%s %s %x", - cs_dsp->name, region_name, cs_ctl->alg_region.alg); + ret = scnprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + "%s %s %x", cs_dsp->name, region_name, + cs_ctl->alg_region.alg); break; case 2: ret = scnprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, diff --git a/sound/soc/intel/common/soc-acpi-intel-cml-match.c b/sound/soc/intel/common/soc-acpi-intel-cml-match.c index b4eb0c97edf1..4eebc79d4b48 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cml-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cml-match.c @@ -81,6 +81,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cml_machines[] = { .sof_fw_filename = "sof-cml.ri", .sof_tplg_filename = "sof-cml-da7219-max98390.tplg", }, + { + .id = "ESSX8336", + .drv_name = "sof-essx8336", + .sof_fw_filename = "sof-cml.ri", + .sof_tplg_filename = "sof-cml-es8336.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_cml_machines); diff --git a/sound/soc/qcom/qdsp6/audioreach.h b/sound/soc/qcom/qdsp6/audioreach.h index 4f693a2660b5..3ee8bfcd0121 100644 --- a/sound/soc/qcom/qdsp6/audioreach.h +++ b/sound/soc/qcom/qdsp6/audioreach.h @@ -550,6 +550,10 @@ struct audio_hw_clk_cfg { uint32_t clock_root; } __packed; +struct audio_hw_clk_rel_cfg { + uint32_t clock_id; +} __packed; + #define PARAM_ID_HW_EP_POWER_MODE_CFG 0x8001176 #define AR_HW_EP_POWER_MODE_0 0 /* default */ #define AR_HW_EP_POWER_MODE_1 1 /* XO Shutdown allowed */ diff --git a/sound/soc/qcom/qdsp6/q6adm.c b/sound/soc/qcom/qdsp6/q6adm.c index 3d831b635524..72c5719f1d25 100644 --- a/sound/soc/qcom/qdsp6/q6adm.c +++ b/sound/soc/qcom/qdsp6/q6adm.c @@ -390,7 +390,7 @@ struct q6copp *q6adm_open(struct device *dev, int port_id, int path, int rate, int ret = 0; if (port_id < 0) { - dev_err(dev, "Invalid port_id 0x%x\n", port_id); + dev_err(dev, "Invalid port_id %d\n", port_id); return ERR_PTR(-EINVAL); } @@ -508,7 +508,7 @@ int q6adm_matrix_map(struct device *dev, int path, int port_idx = payload_map.port_id[i]; if (port_idx < 0) { - dev_err(dev, "Invalid port_id 0x%x\n", + dev_err(dev, "Invalid port_id %d\n", payload_map.port_id[i]); kfree(pkt); return -EINVAL; diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index 46f365528d50..b74b67720ef4 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -269,9 +269,7 @@ static int q6asm_dai_prepare(struct snd_soc_component *component, if (ret < 0) { dev_err(dev, "%s: q6asm_open_write failed\n", __func__); - q6asm_audio_client_free(prtd->audio_client); - prtd->audio_client = NULL; - return -ENOMEM; + goto open_err; } prtd->session_id = q6asm_get_session_id(prtd->audio_client); @@ -279,7 +277,7 @@ static int q6asm_dai_prepare(struct snd_soc_component *component, prtd->session_id, substream->stream); if (ret) { dev_err(dev, "%s: stream reg failed ret:%d\n", __func__, ret); - return ret; + goto routing_err; } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { @@ -301,10 +299,19 @@ static int q6asm_dai_prepare(struct snd_soc_component *component, } if (ret < 0) dev_info(dev, "%s: CMD Format block failed\n", __func__); + else + prtd->state = Q6ASM_STREAM_RUNNING; - prtd->state = Q6ASM_STREAM_RUNNING; + return ret; - return 0; +routing_err: + q6asm_cmd(prtd->audio_client, prtd->stream_id, CMD_CLOSE); +open_err: + q6asm_unmap_memory_regions(substream->stream, prtd->audio_client); + q6asm_audio_client_free(prtd->audio_client); + prtd->audio_client = NULL; + + return ret; } static int q6asm_dai_trigger(struct snd_soc_component *component, diff --git a/sound/soc/qcom/qdsp6/q6prm.c b/sound/soc/qcom/qdsp6/q6prm.c index 82c40f2d4e1d..cda33ded29be 100644 --- a/sound/soc/qcom/qdsp6/q6prm.c +++ b/sound/soc/qcom/qdsp6/q6prm.c @@ -42,6 +42,12 @@ struct prm_cmd_request_rsc { struct audio_hw_clk_cfg clock_id; } __packed; +struct prm_cmd_release_rsc { + struct apm_module_param_data param_data; + uint32_t num_clk_id; + struct audio_hw_clk_rel_cfg clock_id; +} __packed; + static int q6prm_send_cmd_sync(struct q6prm *prm, struct gpr_pkt *pkt, uint32_t rsp_opcode) { return audioreach_send_cmd_sync(prm->dev, prm->gdev, &prm->result, &prm->lock, @@ -102,8 +108,8 @@ int q6prm_unvote_lpass_core_hw(struct device *dev, uint32_t hw_block_id, uint32_ } EXPORT_SYMBOL_GPL(q6prm_unvote_lpass_core_hw); -int q6prm_set_lpass_clock(struct device *dev, int clk_id, int clk_attr, int clk_root, - unsigned int freq) +static int q6prm_request_lpass_clock(struct device *dev, int clk_id, int clk_attr, int clk_root, + unsigned int freq) { struct q6prm *prm = dev_get_drvdata(dev->parent); struct apm_module_param_data *param_data; @@ -138,6 +144,49 @@ int q6prm_set_lpass_clock(struct device *dev, int clk_id, int clk_attr, int clk_ return rc; } + +static int q6prm_release_lpass_clock(struct device *dev, int clk_id, int clk_attr, int clk_root, + unsigned int freq) +{ + struct q6prm *prm = dev_get_drvdata(dev->parent); + struct apm_module_param_data *param_data; + struct prm_cmd_release_rsc *rel; + gpr_device_t *gdev = prm->gdev; + struct gpr_pkt *pkt; + int rc; + + pkt = audioreach_alloc_cmd_pkt(sizeof(*rel), PRM_CMD_RELEASE_HW_RSC, 0, gdev->svc.id, + GPR_PRM_MODULE_IID); + if (IS_ERR(pkt)) + return PTR_ERR(pkt); + + rel = (void *)pkt + GPR_HDR_SIZE + APM_CMD_HDR_SIZE; + + param_data = &rel->param_data; + + param_data->module_instance_id = GPR_PRM_MODULE_IID; + param_data->error_code = 0; + param_data->param_id = PARAM_ID_RSC_AUDIO_HW_CLK; + param_data->param_size = sizeof(*rel) - APM_MODULE_PARAM_DATA_SIZE; + + rel->num_clk_id = 1; + rel->clock_id.clock_id = clk_id; + + rc = q6prm_send_cmd_sync(prm, pkt, PRM_CMD_RSP_RELEASE_HW_RSC); + + kfree(pkt); + + return rc; +} + +int q6prm_set_lpass_clock(struct device *dev, int clk_id, int clk_attr, int clk_root, + unsigned int freq) +{ + if (freq) + return q6prm_request_lpass_clock(dev, clk_id, clk_attr, clk_attr, freq); + + return q6prm_release_lpass_clock(dev, clk_id, clk_attr, clk_attr, freq); +} EXPORT_SYMBOL_GPL(q6prm_set_lpass_clock); static int prm_callback(struct gpr_resp_pkt *data, void *priv, int op) diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c index 3390ebef9549..cd74681e811e 100644 --- a/sound/soc/qcom/qdsp6/q6routing.c +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -372,6 +372,12 @@ int q6routing_stream_open(int fedai_id, int perf_mode, } session = &routing_data->sessions[stream_id - 1]; + if (session->port_id < 0) { + dev_err(routing_data->dev, "Routing not setup for MultiMedia%d Session\n", + session->fedai_id); + return -EINVAL; + } + pdata = &routing_data->port_data[session->port_id]; mutex_lock(&routing_data->lock); @@ -495,7 +501,11 @@ static int msm_routing_put_audio_mixer(struct snd_kcontrol *kcontrol, session->port_id = be_id; snd_soc_dapm_mixer_update_power(dapm, kcontrol, 1, update); } else { - session->port_id = -1; + if (session->port_id == be_id) { + session->port_id = -1; + return 0; + } + snd_soc_dapm_mixer_update_power(dapm, kcontrol, 0, update); } diff --git a/sound/soc/rockchip/rockchip_i2s_tdm.c b/sound/soc/rockchip/rockchip_i2s_tdm.c index 17b9b287853a..5f9cb5c4c7f0 100644 --- a/sound/soc/rockchip/rockchip_i2s_tdm.c +++ b/sound/soc/rockchip/rockchip_i2s_tdm.c @@ -95,6 +95,7 @@ struct rk_i2s_tdm_dev { spinlock_t lock; /* xfer lock */ bool has_playback; bool has_capture; + struct snd_soc_dai_driver *dai; }; static int to_ch_num(unsigned int val) @@ -1310,19 +1311,14 @@ static const struct of_device_id rockchip_i2s_tdm_match[] = { {}, }; -static struct snd_soc_dai_driver i2s_tdm_dai = { +static const struct snd_soc_dai_driver i2s_tdm_dai = { .probe = rockchip_i2s_tdm_dai_probe, - .playback = { - .stream_name = "Playback", - }, - .capture = { - .stream_name = "Capture", - }, .ops = &rockchip_i2s_tdm_dai_ops, }; -static void rockchip_i2s_tdm_init_dai(struct rk_i2s_tdm_dev *i2s_tdm) +static int rockchip_i2s_tdm_init_dai(struct rk_i2s_tdm_dev *i2s_tdm) { + struct snd_soc_dai_driver *dai; struct property *dma_names; const char *dma_name; u64 formats = (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | @@ -1337,19 +1333,33 @@ static void rockchip_i2s_tdm_init_dai(struct rk_i2s_tdm_dev *i2s_tdm) i2s_tdm->has_capture = true; } + dai = devm_kmemdup(i2s_tdm->dev, &i2s_tdm_dai, + sizeof(*dai), GFP_KERNEL); + if (!dai) + return -ENOMEM; + if (i2s_tdm->has_playback) { - i2s_tdm_dai.playback.channels_min = 2; - i2s_tdm_dai.playback.channels_max = 8; - i2s_tdm_dai.playback.rates = SNDRV_PCM_RATE_8000_192000; - i2s_tdm_dai.playback.formats = formats; + dai->playback.stream_name = "Playback"; + dai->playback.channels_min = 2; + dai->playback.channels_max = 8; + dai->playback.rates = SNDRV_PCM_RATE_8000_192000; + dai->playback.formats = formats; } if (i2s_tdm->has_capture) { - i2s_tdm_dai.capture.channels_min = 2; - i2s_tdm_dai.capture.channels_max = 8; - i2s_tdm_dai.capture.rates = SNDRV_PCM_RATE_8000_192000; - i2s_tdm_dai.capture.formats = formats; + dai->capture.stream_name = "Capture"; + dai->capture.channels_min = 2; + dai->capture.channels_max = 8; + dai->capture.rates = SNDRV_PCM_RATE_8000_192000; + dai->capture.formats = formats; } + + if (i2s_tdm->clk_trcm != TRCM_TXRX) + dai->symmetric_rate = 1; + + i2s_tdm->dai = dai; + + return 0; } static int rockchip_i2s_tdm_path_check(struct rk_i2s_tdm_dev *i2s_tdm, @@ -1541,8 +1551,6 @@ static int rockchip_i2s_tdm_probe(struct platform_device *pdev) spin_lock_init(&i2s_tdm->lock); i2s_tdm->soc_data = (struct rk_i2s_soc_data *)of_id->data; - rockchip_i2s_tdm_init_dai(i2s_tdm); - i2s_tdm->frame_width = 64; i2s_tdm->clk_trcm = TRCM_TXRX; @@ -1555,8 +1563,10 @@ static int rockchip_i2s_tdm_probe(struct platform_device *pdev) } i2s_tdm->clk_trcm = TRCM_RX; } - if (i2s_tdm->clk_trcm != TRCM_TXRX) - i2s_tdm_dai.symmetric_rate = 1; + + ret = rockchip_i2s_tdm_init_dai(i2s_tdm); + if (ret) + return ret; i2s_tdm->grf = syscon_regmap_lookup_by_phandle(node, "rockchip,grf"); if (IS_ERR(i2s_tdm->grf)) @@ -1678,7 +1688,7 @@ static int rockchip_i2s_tdm_probe(struct platform_device *pdev) ret = devm_snd_soc_register_component(&pdev->dev, &rockchip_i2s_tdm_component, - &i2s_tdm_dai, 1); + i2s_tdm->dai, 1); if (ret) { dev_err(&pdev->dev, "Could not register DAI\n"); diff --git a/sound/soc/soc-acpi.c b/sound/soc/soc-acpi.c index 2ae99b49d3f5..cbd7ea48837b 100644 --- a/sound/soc/soc-acpi.c +++ b/sound/soc/soc-acpi.c @@ -20,8 +20,10 @@ static bool snd_soc_acpi_id_present(struct snd_soc_acpi_mach *machine) if (comp_ids) { for (i = 0; i < comp_ids->num_codecs; i++) { - if (acpi_dev_present(comp_ids->codecs[i], NULL, -1)) + if (acpi_dev_present(comp_ids->codecs[i], NULL, -1)) { + strscpy(machine->id, comp_ids->codecs[i], ACPI_ID_LEN); return true; + } } } diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 557e22c5254c..f5b9e66ac3b8 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -2700,6 +2700,7 @@ EXPORT_SYMBOL_GPL(snd_soc_tplg_component_load); /* remove dynamic controls from the component driver */ int snd_soc_tplg_component_remove(struct snd_soc_component *comp) { + struct snd_card *card = comp->card->snd_card; struct snd_soc_dobj *dobj, *next_dobj; int pass = SOC_TPLG_PASS_END; @@ -2707,6 +2708,7 @@ int snd_soc_tplg_component_remove(struct snd_soc_component *comp) while (pass >= SOC_TPLG_PASS_START) { /* remove mixer controls */ + down_write(&card->controls_rwsem); list_for_each_entry_safe(dobj, next_dobj, &comp->dobj_list, list) { @@ -2745,6 +2747,7 @@ int snd_soc_tplg_component_remove(struct snd_soc_component *comp) break; } } + up_write(&card->controls_rwsem); pass--; } diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index 6744318de612..13cd96e6724a 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -22,6 +22,7 @@ #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC) #define IDISP_VID_INTEL 0x80860000 +#define CODEC_PROBE_RETRIES 3 /* load the legacy HDA codec driver */ static int request_codec_module(struct hda_codec *codec) @@ -121,12 +122,15 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, u32 hda_cmd = (address << 28) | (AC_NODE_ROOT << 20) | (AC_VERB_PARAMETERS << 8) | AC_PAR_VENDOR_ID; u32 resp = -1; - int ret; + int ret, retry = 0; + + do { + mutex_lock(&hbus->core.cmd_mutex); + snd_hdac_bus_send_cmd(&hbus->core, hda_cmd); + snd_hdac_bus_get_response(&hbus->core, address, &resp); + mutex_unlock(&hbus->core.cmd_mutex); + } while (resp == -1 && retry++ < CODEC_PROBE_RETRIES); - mutex_lock(&hbus->core.cmd_mutex); - snd_hdac_bus_send_cmd(&hbus->core, hda_cmd); - snd_hdac_bus_get_response(&hbus->core, address, &resp); - mutex_unlock(&hbus->core.cmd_mutex); if (resp == -1) return -EIO; dev_dbg(sdev->dev, "HDA codec #%d probed OK: response: %x\n", diff --git a/sound/soc/stm/stm32_i2s.c b/sound/soc/stm/stm32_i2s.c index 68c5de040df8..24327cabd32a 100644 --- a/sound/soc/stm/stm32_i2s.c +++ b/sound/soc/stm/stm32_i2s.c @@ -701,7 +701,7 @@ static int stm32_i2s_configure_clock(struct snd_soc_dai *cpu_dai, if (ret < 0) return ret; - nb_bits = frame_len * ((cgfr & I2S_CGFR_CHLEN) + 1); + nb_bits = frame_len * (FIELD_GET(I2S_CGFR_CHLEN, cgfr) + 1); ret = stm32_i2s_calc_clk_div(i2s, i2s_clock_rate, (nb_bits * rate)); if (ret) diff --git a/sound/soc/tegra/tegra186_dspk.c b/sound/soc/tegra/tegra186_dspk.c index 8ee9a77bd83d..a74c980ee775 100644 --- a/sound/soc/tegra/tegra186_dspk.c +++ b/sound/soc/tegra/tegra186_dspk.c @@ -26,51 +26,162 @@ static const struct reg_default tegra186_dspk_reg_defaults[] = { { TEGRA186_DSPK_CODEC_CTRL, 0x03000000 }, }; -static int tegra186_dspk_get_control(struct snd_kcontrol *kcontrol, +static int tegra186_dspk_get_fifo_th(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); struct tegra186_dspk *dspk = snd_soc_component_get_drvdata(codec); - if (strstr(kcontrol->id.name, "FIFO Threshold")) - ucontrol->value.integer.value[0] = dspk->rx_fifo_th; - else if (strstr(kcontrol->id.name, "OSR Value")) - ucontrol->value.integer.value[0] = dspk->osr_val; - else if (strstr(kcontrol->id.name, "LR Polarity Select")) - ucontrol->value.integer.value[0] = dspk->lrsel; - else if (strstr(kcontrol->id.name, "Channel Select")) - ucontrol->value.integer.value[0] = dspk->ch_sel; - else if (strstr(kcontrol->id.name, "Mono To Stereo")) - ucontrol->value.integer.value[0] = dspk->mono_to_stereo; - else if (strstr(kcontrol->id.name, "Stereo To Mono")) - ucontrol->value.integer.value[0] = dspk->stereo_to_mono; + ucontrol->value.integer.value[0] = dspk->rx_fifo_th; return 0; } -static int tegra186_dspk_put_control(struct snd_kcontrol *kcontrol, +static int tegra186_dspk_put_fifo_th(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { |