diff --git a/Documentation/devicetree/bindings/sound/tdm-slot.txt b/Documentation/devicetree/bindings/sound/tdm-slot.txt index 34cf70e2cbc4..4bb513ae62fc 100644 --- a/Documentation/devicetree/bindings/sound/tdm-slot.txt +++ b/Documentation/devicetree/bindings/sound/tdm-slot.txt @@ -14,8 +14,8 @@ For instance: dai-tdm-slot-tx-mask = <0 1>; dai-tdm-slot-rx-mask = <1 0>; -And for each spcified driver, there could be one .of_xlate_tdm_slot_mask() -to specify a explicit mapping of the channels and the slots. If it's absent +And for each specified driver, there could be one .of_xlate_tdm_slot_mask() +to specify an explicit mapping of the channels and the slots. If it's absent the default snd_soc_of_xlate_tdm_slot_mask() will be used to generating the tx and rx masks. diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c index b9ce44dda886..0d63ebfbff2f 100644 --- a/sound/soc/codecs/max98390.c +++ b/sound/soc/codecs/max98390.c @@ -754,6 +754,7 @@ static struct snd_soc_dai_driver max98390_dai[] = { static int max98390_dsm_init(struct snd_soc_component *component) { int ret; + int param_size, param_start_addr; char filename[128]; const char *vendor, *product; struct max98390_priv *max98390 = @@ -778,16 +779,31 @@ static int max98390_dsm_init(struct snd_soc_component *component) } dev_dbg(component->dev, - "max98390: param fw size %ld\n", + "max98390: param fw size %zd\n", fw->size); + if (fw->size < MAX98390_DSM_PARAM_MIN_SIZE) { + dev_err(component->dev, + "param fw is invalid.\n"); + goto err_alloc; + } dsm_param = (char *)fw->data; + param_start_addr = (dsm_param[0] & 0xff) | (dsm_param[1] & 0xff) << 8; + param_size = (dsm_param[2] & 0xff) | (dsm_param[3] & 0xff) << 8; + if (param_size > MAX98390_DSM_PARAM_MAX_SIZE || + param_start_addr < DSM_STBASS_HPF_B0_BYTE0 || + fw->size < param_size + MAX98390_DSM_PAYLOAD_OFFSET) { + dev_err(component->dev, + "param fw is invalid.\n"); + goto err_alloc; + } + regmap_write(max98390->regmap, MAX98390_R203A_AMP_EN, 0x80); dsm_param += MAX98390_DSM_PAYLOAD_OFFSET; - regmap_bulk_write(max98390->regmap, DSM_EQ_BQ1_B0_BYTE0, - dsm_param, - fw->size - MAX98390_DSM_PAYLOAD_OFFSET); - release_firmware(fw); + regmap_bulk_write(max98390->regmap, param_start_addr, + dsm_param, param_size); regmap_write(max98390->regmap, MAX98390_R23E1_DSP_GLOBAL_EN, 0x01); +err_alloc: + release_firmware(fw); err: return ret; } diff --git a/sound/soc/codecs/max98390.h b/sound/soc/codecs/max98390.h index f59cb114d957..5f444e7779b0 100644 --- a/sound/soc/codecs/max98390.h +++ b/sound/soc/codecs/max98390.h @@ -650,7 +650,8 @@ /* DSM register offset */ #define MAX98390_DSM_PAYLOAD_OFFSET 16 -#define MAX98390_DSM_PAYLOAD_OFFSET_2 495 +#define MAX98390_DSM_PARAM_MAX_SIZE 770 +#define MAX98390_DSM_PARAM_MIN_SIZE 670 struct max98390_priv { struct regmap *regmap; diff --git a/sound/soc/codecs/rl6231.c b/sound/soc/codecs/rl6231.c index 2586d1cafc0c..8c9daf32bab8 100644 --- a/sound/soc/codecs/rl6231.c +++ b/sound/soc/codecs/rl6231.c @@ -80,8 +80,8 @@ int rl6231_calc_dmic_clk(int rate) for (i = 0; i < ARRAY_SIZE(div); i++) { if ((div[i] % 3) == 0) continue; - /* find divider that gives DMIC frequency below 3.072MHz */ - if (3072000 * div[i] >= rate) + /* find divider that gives DMIC frequency below 1.536MHz */ + if (1536000 * div[i] >= rate) return i; } diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 6ba1849a77b0..e2e1d5b03b38 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3625,6 +3625,12 @@ static const struct rt5645_platform_data asus_t100ha_platform_data = { .inv_jd1_1 = true, }; +static const struct rt5645_platform_data asus_t101ha_platform_data = { + .dmic1_data_pin = RT5645_DMIC_DATA_IN2N, + .dmic2_data_pin = RT5645_DMIC2_DISABLE, + .jd_mode = 3, +}; + static const struct rt5645_platform_data lenovo_ideapad_miix_310_pdata = { .jd_mode = 3, .in2_diff = true, @@ -3708,6 +3714,14 @@ static const struct dmi_system_id dmi_platform_data[] = { }, .driver_data = (void *)&asus_t100ha_platform_data, }, + { + .ident = "ASUS T101HA", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), + DMI_MATCH(DMI_PRODUCT_NAME, "T101HA"), + }, + .driver_data = (void *)&asus_t101ha_platform_data, + }, { .ident = "MINIX Z83-4", .matches = { diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index cf4feb835743..00be73900888 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -581,7 +581,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) if (!fsl_asoc_card_is_ac97(priv) && !codec_dev) { dev_err(&pdev->dev, "failed to find codec device\n"); - ret = -EINVAL; + ret = -EPROBE_DEFER; goto asrc_fail; } diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 30f70bbdf89c..1fdb70b9e478 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -754,6 +754,18 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_JD_NOT_INV | BYT_RT5640_MCLK_EN), }, + { /* Toshiba Encore WT10-A */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "TOSHIBA"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "TOSHIBA WT10-A-103"), + }, + .driver_data = (void *)(BYT_RT5640_DMIC1_MAP | + BYT_RT5640_JD_SRC_JD1_IN4P | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_SSP0_AIF2 | + BYT_RT5640_MCLK_EN), + }, { /* Catch-all for generic Insyde tablets, must be last */ .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Insyde"), diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c index 48eda1a8aa6c..954ab01f695b 100644 --- a/sound/soc/intel/boards/glk_rt5682_max98357a.c +++ b/sound/soc/intel/boards/glk_rt5682_max98357a.c @@ -407,7 +407,7 @@ static struct snd_soc_dai_link geminilake_dais[] = { .name = "Glk Audio Echo Reference cap", .stream_name = "Echoreference Capture", .init = NULL, - .capture_only = 1, + .dpcm_capture = 1, .nonatomic = 1, .dynamic = 1, SND_SOC_DAILINK_REG(echoref, dummy, platform), diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index cc9b5eab8b4a..e29c31ffd241 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -692,7 +692,7 @@ static struct snd_soc_dai_link kabylake_dais[] = { .name = "Kbl Audio Echo Reference cap", .stream_name = "Echoreference Capture", .init = NULL, - .capture_only = 1, + .dpcm_capture = 1, .nonatomic = 1, SND_SOC_DAILINK_REG(echoref, dummy, platform), }, @@ -858,7 +858,7 @@ static struct snd_soc_dai_link kabylake_max98_927_373_dais[] = { .name = "Kbl Audio Echo Reference cap", .stream_name = "Echoreference Capture", .init = NULL, - .capture_only = 1, + .dpcm_capture = 1, .nonatomic = 1, SND_SOC_DAILINK_REG(echoref, dummy, platform), }, diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index 658a9da3a40f..09ba55fc36d5 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -672,7 +672,7 @@ static struct snd_soc_dai_link kabylake_dais[] = { .name = "Kbl Audio Echo Reference cap", .stream_name = "Echoreference Capture", .init = NULL, - .capture_only = 1, + .dpcm_capture = 1, .nonatomic = 1, SND_SOC_DAILINK_REG(echoref, dummy, platform), }, diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 1b1f8d7a4ea3..b34cf6cf1139 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -566,7 +566,7 @@ static struct snd_soc_dai_link kabylake_dais[] = { .name = "Kbl Audio Echo Reference cap", .stream_name = "Echoreference Capture", .init = NULL, - .capture_only = 1, + .dpcm_capture = 1, .nonatomic = 1, SND_SOC_DAILINK_REG(echoref, dummy, platform), }, diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c index 2e9b56b29d31..b2e867113226 100644 --- a/sound/soc/meson/axg-fifo.c +++ b/sound/soc/meson/axg-fifo.c @@ -249,7 +249,7 @@ int axg_fifo_pcm_open(struct snd_soc_component *component, /* Enable pclk to access registers and clock the fifo ip */ ret = clk_prepare_enable(fifo->pclk); if (ret) - return ret; + goto free_irq; /* Setup status2 so it reports the memory pointer */ regmap_update_bits(fifo->map, FIFO_CTRL1, @@ -269,8 +269,14 @@ int axg_fifo_pcm_open(struct snd_soc_component *component, /* Take memory arbitror out of reset */ ret = reset_control_deassert(fifo->arb); if (ret) - clk_disable_unprepare(fifo->pclk); + goto free_clk; + return 0; + +free_clk: + clk_disable_unprepare(fifo->pclk); +free_irq: + free_irq(fifo->irq, ss); return ret; } EXPORT_SYMBOL_GPL(axg_fifo_pcm_open); diff --git a/sound/soc/meson/meson-card-utils.c b/sound/soc/meson/meson-card-utils.c index 2ca8c98e204f..5a4a91c88734 100644 --- a/sound/soc/meson/meson-card-utils.c +++ b/sound/soc/meson/meson-card-utils.c @@ -49,19 +49,26 @@ int meson_card_reallocate_links(struct snd_soc_card *card, links = krealloc(priv->card.dai_link, num_links * sizeof(*priv->card.dai_link), GFP_KERNEL | __GFP_ZERO); + if (!links) + goto err_links; + ldata = krealloc(priv->link_data, num_links * sizeof(*priv->link_data), GFP_KERNEL | __GFP_ZERO); - - if (!links || !ldata) { - dev_err(priv->card.dev, "failed to allocate links\n"); - return -ENOMEM; - } + if (!ldata) + goto err_ldata; priv->card.dai_link = links; priv->link_data = ldata; priv->card.num_links = num_links; return 0; + +err_ldata: + kfree(links); +err_links: + dev_err(priv->card.dev, "failed to allocate links\n"); + return -ENOMEM; + } EXPORT_SYMBOL_GPL(meson_card_reallocate_links); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b07eca2c6ccc..7b387202c5db 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1648,9 +1648,25 @@ static void soc_check_tplg_fes(struct snd_soc_card *card) dai_link->platforms->name = component->name; /* convert non BE into BE */ - dai_link->no_pcm = 1; - dai_link->dpcm_playback = 1; - dai_link->dpcm_capture = 1; + if (!dai_link->no_pcm) { + dai_link->no_pcm = 1; + + if (dai_link->dpcm_playback) + dev_warn(card->dev, + "invalid configuration, dailink %s has flags no_pcm=0 and dpcm_playback=1\n", + dai_link->name); + if (dai_link->dpcm_capture) + dev_warn(card->dev, + "invalid configuration, dailink %s has flags no_pcm=0 and dpcm_capture=1\n", + dai_link->name); + + /* convert normal link into DPCM one */ + if (!(dai_link->dpcm_playback || + dai_link->dpcm_capture)) { + dai_link->dpcm_playback = !dai_link->capture_only; + dai_link->dpcm_capture = !dai_link->playback_only; + } + } /* * override any BE fixups diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 276505fb9d50..2c114b4542ce 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2789,20 +2789,44 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) struct snd_pcm *pcm; char new_name[64]; int ret = 0, playback = 0, capture = 0; + int stream; int i; - if (rtd->dai_link->dynamic || rtd->dai_link->no_pcm) { - cpu_dai = asoc_rtd_to_cpu(rtd, 0); - if (rtd->num_cpus > 1) { - dev_err(rtd->dev, - "DPCM doesn't support Multi CPU yet\n"); - return -EINVAL; - } + if (rtd->dai_link->dynamic && rtd->num_cpus > 1) { + dev_err(rtd->dev, + "DPCM doesn't support Multi CPU for Front-Ends yet\n"); + return -EINVAL; + } - playback = rtd->dai_link->dpcm_playback && - snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_PLAYBACK); - capture = rtd->dai_link->dpcm_capture && - snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_CAPTURE); + if (rtd->dai_link->dynamic || rtd->dai_link->no_pcm) { + if (rtd->dai_link->dpcm_playback) { + stream = SNDRV_PCM_STREAM_PLAYBACK; + + for_each_rtd_cpu_dais(rtd, i, cpu_dai) + if (!snd_soc_dai_stream_valid(cpu_dai, + stream)) { + dev_err(rtd->card->dev, + "CPU DAI %s for rtd %s does not support playback\n", + cpu_dai->name, + rtd->dai_link->stream_name); + return -EINVAL; + } + playback = 1; + } + if (rtd->dai_link->dpcm_capture) { + stream = SNDRV_PCM_STREAM_CAPTURE; + + for_each_rtd_cpu_dais(rtd, i, cpu_dai) + if (!snd_soc_dai_stream_valid(cpu_dai, + stream)) { + dev_err(rtd->card->dev, + "CPU DAI %s for rtd %s does not support capture\n", + cpu_dai->name, + rtd->dai_link->stream_name); + return -EINVAL; + } + capture = 1; + } } else { /* Adapt stream for codec2codec links */ int cpu_capture = rtd->dai_link->params ? diff --git a/sound/soc/sof/nocodec.c b/sound/soc/sof/nocodec.c index ce053ba8f2e8..d03b5be31255 100644 --- a/sound/soc/sof/nocodec.c +++ b/sound/soc/sof/nocodec.c @@ -52,8 +52,10 @@ static int sof_nocodec_bes_setup(struct device *dev, links[i].platforms->name = dev_name(dev); links[i].codecs->dai_name = "snd-soc-dummy-dai"; links[i].codecs->name = "snd-soc-dummy"; - links[i].dpcm_playback = 1; - links[i].dpcm_capture = 1; + if (ops->drv[i].playback.channels_min) + links[i].dpcm_playback = 1; + if (ops->drv[i].capture.channels_min) + links[i].dpcm_capture = 1; } card->dai_link = links;