diff options
Diffstat (limited to 'sound/soc')
66 files changed, 882 insertions, 323 deletions
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 79adf360897e..0cd4f91de5db 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1,4 +1,4 @@ -# Helper to resolve issues with configs that have SPI enabled but I2C + # modular, meaning we can't build the codec driver in with I2C support. # We use an ordered list of conditional defaults to pick the appropriate # setting - SPI can't be modular so that case doesn't need to be covered. @@ -615,7 +615,7 @@ config SND_SOC_HDAC_HDMI select HDMI config SND_SOC_ICS43432 - tristate + tristate "InvenSense ICS43432 I2S microphone codec" config SND_SOC_INNO_RK3036 tristate "Inno codec driver for RK3036 SoC" @@ -1090,7 +1090,8 @@ config SND_SOC_WM8903 depends on I2C config SND_SOC_WM8904 - tristate + tristate "Wolfson Microelectronics WM8904 CODEC" + depends on I2C config SND_SOC_WM8940 tristate diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index 6e8eb1f5a041..bed64723e5d9 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -60,7 +60,7 @@ static const struct reg_default cs4265_reg_defaults[] = { static bool cs4265_readable_register(struct device *dev, unsigned int reg) { switch (reg) { - case CS4265_CHIP_ID ... CS4265_SPDIF_CTL2: + case CS4265_CHIP_ID ... CS4265_MAX_REGISTER: return true; default: return false; diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 84f86745c30e..828bc615a190 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -643,6 +643,7 @@ static const struct regmap_config cs4270_regmap = { .reg_defaults = cs4270_reg_defaults, .num_reg_defaults = ARRAY_SIZE(cs4270_reg_defaults), .cache_type = REGCACHE_RBTREE, + .write_flag_mask = CS4270_I2C_INCR, .readable_reg = cs4270_reg_is_readable, .volatile_reg = cs4270_reg_is_volatile, diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c index 2e772427b48a..cedddee67199 100644 --- a/sound/soc/codecs/cs42xx8.c +++ b/sound/soc/codecs/cs42xx8.c @@ -668,6 +668,7 @@ static int cs42xx8_runtime_resume(struct device *dev) CS42XX8_PWRCTL_PDN_MASK, 0); regcache_cache_only(cs42xx8->regmap, false); + regcache_mark_dirty(cs42xx8->regmap); ret = regcache_sync(cs42xx8->regmap); if (ret) { diff --git a/sound/soc/codecs/cs4349.c b/sound/soc/codecs/cs4349.c index 0a749c79ef57..1d38e53dc95c 100644 --- a/sound/soc/codecs/cs4349.c +++ b/sound/soc/codecs/cs4349.c @@ -380,6 +380,7 @@ static struct i2c_driver cs4349_i2c_driver = { .driver = { .name = "cs4349", .of_match_table = cs4349_of_match, + .pm = &cs4349_runtime_pm, }, .id_table = cs4349_i2c_id, .probe = cs4349_i2c_probe, diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index da2d353af5ba..949dbdc0445e 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -46,7 +46,10 @@ static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_vol_tlv, -9600, 50, 1); static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_max_gain_tlv, -650, 150, 0); static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_min_gain_tlv, -1200, 150, 0); static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_target_tlv, -1650, 150, 0); -static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(hpmixer_gain_tlv, -1200, 150, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(hpmixer_gain_tlv, + 0, 4, TLV_DB_SCALE_ITEM(-1200, 150, 0), + 8, 11, TLV_DB_SCALE_ITEM(-450, 150, 0), +); static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(adc_pga_gain_tlv, 0, 0, TLV_DB_SCALE_ITEM(-350, 0, 0), @@ -84,7 +87,7 @@ static const struct snd_kcontrol_new es8316_snd_controls[] = { SOC_DOUBLE_TLV("Headphone Playback Volume", ES8316_CPHP_ICAL_VOL, 4, 0, 3, 1, hpout_vol_tlv), SOC_DOUBLE_TLV("Headphone Mixer Volume", ES8316_HPMIX_VOL, - 0, 4, 7, 0, hpmixer_gain_tlv), + 0, 4, 11, 0, hpmixer_gain_tlv), SOC_ENUM("Playback Polarity", dacpol), SOC_DOUBLE_R_TLV("DAC Playback Volume", ES8316_DAC_VOLL, diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index bcdb8914ec16..e2f44fa46262 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -231,7 +231,7 @@ static const struct soc_enum es8328_rline_enum = ARRAY_SIZE(es8328_line_texts), es8328_line_texts); static const struct snd_kcontrol_new es8328_right_line_controls = - SOC_DAPM_ENUM("Route", es8328_lline_enum); + SOC_DAPM_ENUM("Route", es8328_rline_enum); /* Left Mixer */ static const struct snd_kcontrol_new es8328_left_mixer_controls[] = { diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index e824d47cc22b..1c3626347e12 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1408,6 +1408,12 @@ static int hdac_hdmi_create_dais(struct hdac_device *hdac, if (ret) return ret; + /* Filter out 44.1, 88.2 and 176.4Khz */ + rates &= ~(SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_176400); + if (!rates) + return -EINVAL; + sprintf(dai_name, "intel-hdmi-hifi%d", i+1); hdmi_dais[i].name = devm_kstrdup(&hdac->dev, dai_name, GFP_KERNEL); diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 5866f7332786..fc1b67f47f20 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -444,8 +444,12 @@ static int hdmi_codec_startup(struct snd_pcm_substream *substream, if (!ret) { ret = snd_pcm_hw_constraint_eld(substream->runtime, hcp->eld); - if (ret) + if (ret) { + mutex_lock(&hcp->current_stream_lock); + hcp->current_stream = NULL; + mutex_unlock(&hcp->current_stream_lock); return ret; + } } /* Select chmap supported */ hdmi_codec_eld_chmap(hcp); @@ -532,73 +536,71 @@ static int hdmi_codec_set_fmt(struct snd_soc_dai *dai, { struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai); struct hdmi_codec_daifmt cf = { 0 }; - int ret = 0; dev_dbg(dai->dev, "%s()\n", __func__); - if (dai->id == DAI_ID_SPDIF) { - cf.fmt = HDMI_SPDIF; - } else { - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - cf.bit_clk_master = 1; - cf.frame_clk_master = 1; - break; - case SND_SOC_DAIFMT_CBS_CFM: - cf.frame_clk_master = 1; - break; - case SND_SOC_DAIFMT_CBM_CFS: - cf.bit_clk_master = 1; - break; - case SND_SOC_DAIFMT_CBS_CFS: - break; - default: - return -EINVAL; - } + if (dai->id == DAI_ID_SPDIF) + return 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + cf.bit_clk_master = 1; + cf.frame_clk_master = 1; + break; + case SND_SOC_DAIFMT_CBS_CFM: + cf.frame_clk_master = 1; + break; + case SND_SOC_DAIFMT_CBM_CFS: + cf.bit_clk_master = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_NB_NF: - break; - case SND_SOC_DAIFMT_NB_IF: - cf.frame_clk_inv = 1; - break; - case SND_SOC_DAIFMT_IB_NF: - cf.bit_clk_inv = 1; - break; - case SND_SOC_DAIFMT_IB_IF: - cf.frame_clk_inv = 1; - cf.bit_clk_inv = 1; - break; - } + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_NB_IF: + cf.frame_clk_inv = 1; + break; + case SND_SOC_DAIFMT_IB_NF: + cf.bit_clk_inv = 1; + break; + case SND_SOC_DAIFMT_IB_IF: + cf.frame_clk_inv = 1; + cf.bit_clk_inv = 1; + break; + } - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_I2S: - cf.fmt = HDMI_I2S; - break; - case SND_SOC_DAIFMT_DSP_A: - cf.fmt = HDMI_DSP_A; - break; - case SND_SOC_DAIFMT_DSP_B: - cf.fmt = HDMI_DSP_B; - break; - case SND_SOC_DAIFMT_RIGHT_J: - cf.fmt = HDMI_RIGHT_J; - break; - case SND_SOC_DAIFMT_LEFT_J: - cf.fmt = HDMI_LEFT_J; - break; - case SND_SOC_DAIFMT_AC97: - cf.fmt = HDMI_AC97; - break; - default: - dev_err(dai->dev, "Invalid DAI interface format\n"); - return -EINVAL; - } + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + cf.fmt = HDMI_I2S; + break; + case SND_SOC_DAIFMT_DSP_A: + cf.fmt = HDMI_DSP_A; + break; + case SND_SOC_DAIFMT_DSP_B: + cf.fmt = HDMI_DSP_B; + break; + case SND_SOC_DAIFMT_RIGHT_J: + cf.fmt = HDMI_RIGHT_J; + break; + case SND_SOC_DAIFMT_LEFT_J: + cf.fmt = HDMI_LEFT_J; + break; + case SND_SOC_DAIFMT_AC97: + cf.fmt = HDMI_AC97; + break; + default: + dev_err(dai->dev, "Invalid DAI interface format\n"); + return -EINVAL; } hcp->daifmt[dai->id] = cf; - return ret; + return 0; } static int hdmi_codec_digital_mute(struct snd_soc_dai *dai, int mute) @@ -814,8 +816,10 @@ static int hdmi_codec_probe(struct platform_device *pdev) i++; } - if (hcd->spdif) + if (hcd->spdif) { hcp->daidrv[i] = hdmi_spdif_dai; + hcp->daifmt[DAI_ID_SPDIF].fmt = HDMI_SPDIF; + } ret = snd_soc_register_codec(dev, &hdmi_codec, hcp->daidrv, dai_count); diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 13bcfb1ef9b4..3fe09828745a 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -1209,14 +1209,14 @@ static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = { &max98090_right_rcv_mixer_controls[0], ARRAY_SIZE(max98090_right_rcv_mixer_controls)), - SND_SOC_DAPM_MUX("LINMOD Mux", M98090_REG_LOUTR_MIXER, - M98090_LINMOD_SHIFT, 0, &max98090_linmod_mux), + SND_SOC_DAPM_MUX("LINMOD Mux", SND_SOC_NOPM, 0, 0, + &max98090_linmod_mux), - SND_SOC_DAPM_MUX("MIXHPLSEL Mux", M98090_REG_HP_CONTROL, - M98090_MIXHPLSEL_SHIFT, 0, &max98090_mixhplsel_mux), + SND_SOC_DAPM_MUX("MIXHPLSEL Mux", SND_SOC_NOPM, 0, 0, + &max98090_mixhplsel_mux), - SND_SOC_DAPM_MUX("MIXHPRSEL Mux", M98090_REG_HP_CONTROL, - M98090_MIXHPRSEL_SHIFT, 0, &max98090_mixhprsel_mux), + SND_SOC_DAPM_MUX("MIXHPRSEL Mux", SND_SOC_NOPM, 0, 0, + &max98090_mixhprsel_mux), SND_SOC_DAPM_PGA("HP Left Out", M98090_REG_OUTPUT_ENABLE, M98090_HPLEN_SHIFT, 0, NULL, 0), @@ -1924,6 +1924,21 @@ static int max98090_configure_dmic(struct max98090_priv *max98090, return 0; } +static int max98090_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct max98090_priv *max98090 = snd_soc_component_get_drvdata(component); + unsigned int fmt = max98090->dai_fmt; + + /* Remove 24-bit format support if it is not in right justified mode. */ + if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_RIGHT_J) { + substream->runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; + snd_pcm_hw_constraint_msbits(substream->runtime, 0, 16, 16); + } + return 0; +} + static int max98090_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -2331,6 +2346,7 @@ EXPORT_SYMBOL_GPL(max98090_mic_detect); #define MAX98090_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) static const struct snd_soc_dai_ops max98090_dai_ops = { + .startup = max98090_dai_startup, .set_sysclk = max98090_dai_set_sysclk, .set_fmt = max98090_dai_set_fmt, .set_tdm_slot = max98090_set_tdm_slot, diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index 0b9b014b4bb6..3633eb30dd13 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -303,7 +303,7 @@ struct pm8916_wcd_analog_priv { }; static const char *const adc2_mux_text[] = { "ZERO", "INP2", "INP3" }; -static const char *const rdac2_mux_text[] = { "ZERO", "RX2", "RX1" }; +static const char *const rdac2_mux_text[] = { "RX1", "RX2" }; static const char *const hph_text[] = { "ZERO", "Switch", }; static const struct soc_enum hph_enum = SOC_ENUM_SINGLE_VIRT( @@ -318,7 +318,7 @@ static const struct soc_enum adc2_enum = SOC_ENUM_SINGLE_VIRT( /* RDAC2 MUX */ static const struct soc_enum rdac2_mux_enum = SOC_ENUM_SINGLE( - CDC_D_CDC_CONN_HPHR_DAC_CTL, 0, 3, rdac2_mux_text); + CDC_D_CDC_CONN_HPHR_DAC_CTL, 0, 2, rdac2_mux_text); static const struct snd_kcontrol_new spkr_switch[] = { SOC_DAPM_SINGLE("Switch", CDC_A_SPKR_DAC_CTL, 7, 1, 0) @@ -876,10 +876,10 @@ static const struct snd_soc_dapm_widget pm8916_wcd_analog_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("MIC BIAS External1", CDC_A_MICB_1_EN, 7, 0, pm8916_wcd_analog_enable_micbias_ext1, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("MIC BIAS External2", CDC_A_MICB_2_EN, 7, 0, pm8916_wcd_analog_enable_micbias_ext2, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_ADC_E("ADC1", NULL, CDC_A_TX_1_EN, 7, 0, pm8916_wcd_analog_enable_adc, diff --git a/sound/soc/codecs/nau8540.c b/sound/soc/codecs/nau8540.c index f9c9933acffb..c0c64f90a61b 100644 --- a/sound/soc/codecs/nau8540.c +++ b/sound/soc/codecs/nau8540.c @@ -548,7 +548,7 @@ static int nau8540_calc_fll_param(unsigned int fll_in, fvco_max = 0; fvco_sel = ARRAY_SIZE(mclk_src_scaling); for (i = 0; i < ARRAY_SIZE(mclk_src_scaling); i++) { - fvco = 256 * fs * 2 * mclk_src_scaling[i].param; + fvco = 256ULL * fs * 2 * mclk_src_scaling[i].param; if (fvco > NAU_FVCO_MIN && fvco < NAU_FVCO_MAX && fvco_max < fvco) { fvco_max = fvco; diff --git a/sound/soc/codecs/nau8810.c b/sound/soc/codecs/nau8810.c index c8e2451ae0a3..193588eb9835 100644 --- a/sound/soc/codecs/nau8810.c +++ b/sound/soc/codecs/nau8810.c @@ -414,9 +414,9 @@ static const struct snd_soc_dapm_widget nau8810_dapm_widgets[] = { SND_SOC_DAPM_MIXER("Mono Mixer", NAU8810_REG_POWER3, NAU8810_MOUTMX_EN_SFT, 0, &nau8810_mono_mixer_controls[0], ARRAY_SIZE(nau8810_mono_mixer_controls)), - SND_SOC_DAPM_DAC("DAC", "HiFi Playback", NAU8810_REG_POWER3, + SND_SOC_DAPM_DAC("DAC", "Playback", NAU8810_REG_POWER3, NAU8810_DAC_EN_SFT, 0), - SND_SOC_DAPM_ADC("ADC", "HiFi Capture", NAU8810_REG_POWER2, + SND_SOC_DAPM_ADC("ADC", "Capture", NAU8810_REG_POWER2, NAU8810_ADC_EN_SFT, 0), SND_SOC_DAPM_PGA("SpkN Out", NAU8810_REG_POWER3, NAU8810_NSPK_EN_SFT, 0, NULL, 0), diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 0240759f951c..e8ea51247b17 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -634,8 +634,8 @@ static const struct snd_soc_dapm_widget nau8824_dapm_widgets[] = { SND_SOC_DAPM_ADC("ADCR", NULL, NAU8824_REG_ANALOG_ADC_2, NAU8824_ADCR_EN_SFT, 0), - SND_SOC_DAPM_AIF_OUT("AIFTX", "HiFi Capture", 0, SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_AIF_IN("AIFRX", "HiFi Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIFTX", "Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIFRX", "Playback", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC("DACL", NULL, NAU8824_REG_RDAC, NAU8824_DACL_EN_SFT, 0), @@ -784,6 +784,36 @@ 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; @@ -792,8 +822,8 @@ static void nau8824_eject_jack(struct nau8824 *nau8824) /* Clear all interruption status */ nau8824_int_status_clear_all(regmap); - snd_soc_dapm_disable_pin(dapm, "SAR"); - snd_soc_dapm_disable_pin(dapm, "MICBIAS"); + nau8824_dapm_disable_pin(nau8824, "SAR"); + nau8824_dapm_disable_pin(nau8824, "MICBIAS"); snd_soc_dapm_sync(dapm); /* Enable the insertion interruption, disable the ejection @@ -822,8 +852,8 @@ static void nau8824_jdet_work(struct work_struct *work) struct regmap *regmap = nau8824->regmap; int adc_value, event = 0, event_mask = 0; - snd_soc_dapm_force_enable_pin(dapm, "MICBIAS"); - snd_soc_dapm_force_enable_pin(dapm, "SAR"); + nau8824_dapm_enable_pin(nau8824, "MICBIAS"); + nau8824_dapm_enable_pin(nau8824, "SAR"); snd_soc_dapm_sync(dapm); msleep(100); @@ -834,8 +864,8 @@ static void nau8824_jdet_work(struct work_struct *work) if (adc_value < HEADSET_SARADC_THD) { event |= SND_JACK_HEADPHONE; - snd_soc_dapm_disable_pin(dapm, "SAR"); - snd_soc_dapm_disable_pin(dapm, "MICBIAS"); + nau8824_dapm_disable_pin(nau8824, "SAR"); + nau8824_dapm_disable_pin(nau8824, "MICBIAS"); snd_soc_dapm_sync(dapm); } else { event |= SND_JACK_HEADSET; diff --git a/sound/soc/codecs/rt274.c b/sound/soc/codecs/rt274.c index 8f92e5c4dd9d..43086ac9ffec 100644 --- a/sound/soc/codecs/rt274.c +++ b/sound/soc/codecs/rt274.c @@ -398,6 +398,8 @@ static int rt274_mic_detect(struct snd_soc_codec *codec, { struct rt274_priv *rt274 = snd_soc_codec_get_drvdata(codec); + rt274->jack = jack; + if (jack == NULL) { /* Disable jack detection */ regmap_update_bits(rt274->regmap, RT274_EAPD_GPIO_IRQ_CTRL, @@ -405,7 +407,6 @@ static int rt274_mic_detect(struct snd_soc_codec *codec, return 0; } - rt274->jack = jack; regmap_update_bits(rt274->regmap, RT274_EAPD_GPIO_IRQ_CTRL, RT274_IRQ_EN, RT274_IRQ_EN); @@ -1128,8 +1129,11 @@ static int rt274_i2c_probe(struct i2c_client *i2c, return ret; } - regmap_read(rt274->regmap, + ret = regmap_read(rt274->regmap, RT274_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &val); + if (ret) + return ret; + if (val != RT274_VENDOR_ID) { dev_err(&i2c->dev, "Device with ID register %#x is not rt274\n", val); diff --git a/sound/soc/codecs/rt5677-spi.c b/sound/soc/codecs/rt5677-spi.c index bd51f3655ee3..06abcd017650 100644 --- a/sound/soc/codecs/rt5677-spi.c +++ b/sound/soc/codecs/rt5677-spi.c @@ -58,13 +58,15 @@ static DEFINE_MUTEX(spi_mutex); * RT5677_SPI_READ/WRITE_32: Transfer 4 bytes * RT5677_SPI_READ/WRITE_BURST: Transfer any multiples of 8 bytes * - * For example, reading 260 bytes at 0x60030002 uses the following commands: - * 0x60030002 RT5677_SPI_READ_16 2 bytes + * Note: + * 16 Bit writes and reads are restricted to the address range + * 0x18020000 ~ 0x18021000 + * + * For example, reading 256 bytes at 0x60030004 uses the following commands: * 0x60030004 RT5677_SPI_READ_32 4 bytes * 0x60030008 RT5677_SPI_READ_BURST 240 bytes * 0x600300F8 RT5677_SPI_READ_BURST 8 bytes * 0x60030100 RT5677_SPI_READ_32 4 bytes - * 0x60030104 RT5677_SPI_READ_16 2 bytes * * Input: * @read: true for read commands; false for write commands @@ -79,15 +81,13 @@ static u8 rt5677_spi_select_cmd(bool read, u32 align, u32 remain, u32 *len) { u8 cmd; - if (align == 2 || align == 6 || remain == 2) { - cmd = RT5677_SPI_READ_16; - *len = 2; - } else if (align == 4 || remain <= 6) { + if (align == 4 || remain <= 4) { cmd = RT5677_SPI_READ_32; *len = 4; } else { cmd = RT5677_SPI_READ_BURST; - *len = min_t(u32, remain & ~7, RT5677_SPI_BURST_LEN); + *len = (((remain - 1) >> 3) + 1) << 3; + *len = min_t(u32, *len, RT5677_SPI_BURST_LEN); } return read ? cmd : cmd + 1; } @@ -108,7 +108,7 @@ static void rt5677_spi_reverse(u8 *dst, u32 dstlen, const u8 *src, u32 srclen) } } -/* Read DSP address space using SPI. addr and len have to be 2-byte aligned. */ +/* Read DSP address space using SPI. addr and len have to be 4-byte aligned. */ int rt5677_spi_read(u32 addr, void *rxbuf, size_t len) { u32 offset; @@ -124,7 +124,7 @@ int rt5677_spi_read(u32 addr, void *rxbuf, size_t len) if (!g_spi) return -ENODEV; - if ((addr & 1) || (len & 1)) { + if ((addr & 3) || (len & 3)) { dev_err(&g_spi->dev, "Bad read align 0x%x(%zu)\n", addr, len); return -EACCES; } @@ -159,13 +159,13 @@ int rt5677_spi_read(u32 addr, void *rxbuf, size_t len) } EXPORT_SYMBOL_GPL(rt5677_spi_read); -/* Write DSP address space using SPI. addr has to be 2-byte aligned. - * If len is not 2-byte aligned, an extra byte of zero is written at the end +/* Write DSP address space using SPI. addr has to be 4-byte aligned. + * If len is not 4-byte aligned, then extra zeros are written at the end * as padding. */ int rt5677_spi_write(u32 addr, const void *txbuf, size_t len) { - u32 offset, len_with_pad = len; + u32 offset; int status = 0; struct spi_transfer t; struct spi_message m; @@ -178,22 +178,19 @@ int rt5677_spi_write(u32 addr, const void *txbuf, size_t len) if (!g_spi) return -ENODEV; - if (addr & 1) { + if (addr & 3) { dev_err(&g_spi->dev, "Bad write align 0x%x(%zu)\n", addr, len); return -EACCES; } - if (len & 1) - len_with_pad = len + 1; - memset(&t, 0, sizeof(t)); t.tx_buf = buf; t.speed_hz = RT5677_SPI_FREQ; spi_message_init_with_transfers(&m, &t, 1); - for (offset = 0; offset < len_with_pad;) { + for (offset = 0; offset < len;) { spi_cmd = rt5677_spi_select_cmd(false, (addr + offset) & 7, - len_with_pad - offset, &t.len); + len - offset, &t.len); /* Construct SPI message header */ buf[0] = spi_cmd; diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 1cd20b88a3a9..82ee8f4b965b 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -297,6 +297,7 @@ static bool rt5677_volatile_register(struct device *dev, unsigned int reg) case RT5677_I2C_MASTER_CTRL7: case RT5677_I2C_MASTER_CTRL8: case RT5677_HAP_GENE_CTRL2: + case RT5677_PWR_ANLG2: /* Modified by DSP firmware */ case RT5677_PWR_DSP_ST: case RT5677_PRIV_DATA: case RT5677_ASRC_22: diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index f3ffa31b5bca..5d54a4828b42 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -35,6 +35,13 @@ #define SGTL5000_DAP_REG_OFFSET 0x0100 #define SGTL5000_MAX_REG_OFFSET 0x013A +/* Delay for the VAG ramp up */ +#define SGTL5000_VAG_POWERUP_DELAY 500 /* ms */ +/* Delay for the VAG ramp down */ +#define SGTL5000_VAG_POWERDOWN_DELAY 500 /* ms */ + +#define SGTL5000_OUTPUTS_MUTE (SGTL5000_HP_MUTE | SGTL5000_LINE_OUT_MUTE) + /* default value of sgtl5000 registers */ static const struct reg_default sgtl5000_reg_defaults[] = { { SGTL5000_CHIP_DIG_POWER, 0x0000 }, @@ -120,6 +127,13 @@ enum { I2S_LRCLK_STRENGTH_HIGH, }; +enum { + HP_POWER_EVENT, + DAC_POWER_EVENT, + ADC_POWER_EVENT, + LAST_POWER_EVENT = ADC_POWER_EVENT +}; + /* sgtl5000 private structure in codec */ struct sgtl5000_priv { int sysclk; /* sysclk rate */ @@ -133,8 +147,117 @@ struct sgtl5000_priv { u8 micbias_resistor; u8 micbias_voltage; u8 lrclk_strength; + u16 mute_state[LAST_POWER_EVENT + 1]; }; +static inline int hp_sel_input(struct snd_soc_component *component) +{ + unsigned int ana_reg = 0; + + snd_soc_component_read(component, SGTL5000_CHIP_ANA_CTRL, &ana_reg); + + return (ana_reg & SGTL5000_HP_SEL_MASK) >> SGTL5000_HP_SEL_SHIFT; +} + +static inline u16 mute_output(struct snd_soc_component *component, + u16 mute_mask) +{ + unsigned int mute_reg = 0; + + snd_soc_component_read(component, SGTL5000_CHIP_ANA_CTRL, &mute_reg); + + snd_soc_component_update_bits(component, SGTL5000_CHIP_ANA_CTRL, + mute_mask, mute_mask); + return mute_reg; +} + +static inline void restore_output(struct snd_soc_component *component, + u16 mute_mask, u16 mute_reg) +{ + snd_soc_component_update_bits(component, SGTL5000_CHIP_ANA_CTRL, + mute_mask, mute_reg); +} + +static void vag_power_on(struct snd_soc_component *component, u32 source) +{ + unsigned int ana_reg = 0; + + snd_soc_component_read(component, SGTL5000_CHIP_ANA_POWER, &ana_reg); + + if (ana_reg & SGTL5000_VAG_POWERUP) + return; + + snd_soc_component_update_bits(component, SGTL5000_CHIP_ANA_POWER, + SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP); + + /* When VAG powering on to get local loop from Line-In, the sleep + * is required to avoid loud pop. + */ + if (hp_sel_input(component) == SGTL5000_HP_SEL_LINE_IN && + source == HP_POWER_EVENT) + msleep(SGTL5000_VAG_POWERUP_DELAY); +} + +static int vag_power_consumers(struct snd_soc_component *component, + u16 ana_pwr_reg, u32 source) +{ + int consumers = 0; + + /* count dac/adc consumers unconditional */ + if (ana_pwr_reg & SGTL5000_DAC_POWERUP) + consumers++; + if (ana_pwr_reg & SGTL5000_ADC_POWERUP) + consumers++; + + /* + * If the event comes from HP and Line-In is selected, + * current action is 'DAC to be powered down'. + * As HP_POWERUP is not set when HP muxed to line-in, + * we need to keep VAG power ON. + */ + if (source == HP_POWER_EVENT) { + if (hp_sel_input(component) == SGTL5000_HP_SEL_LINE_IN) + consumers++; + } else { + if (ana_pwr_reg & SGTL5000_HP_POWERUP) + consumers++; + } + + return consumers; +} + +static void vag_power_off(struct snd_soc_component *component, u32 source) +{ + unsigned int ana_pwr = SGTL5000_VAG_POWERUP; + + snd_soc_component_read(component, SGTL5000_CHIP_ANA_POWER, &ana_pwr); + + if (!(ana_pwr & SGTL5000_VAG_POWERUP)) + return; + + /* + * This function calls when any of VAG power consumers is disappearing. + * Thus, if there is more than one consumer at the moment, as minimum + * one consumer will definitely stay after the end of the current + * event. + * Don't clear VAG_POWERUP if 2 or more consumers of VAG present: + * - LINE_IN (for HP events) / HP (for DAC/ADC events) + * - DAC + * - ADC + * (the current consumer is disappearing right now) + */ + if (vag_power_consumers(component, ana_pwr, source) >= 2) + return; + + snd_soc_component_update_bits(component, SGTL5000_CHIP_ANA_POWER, + SGTL5000_VAG_POWERUP, 0); + /* In power down case, we need wait 400-1000 ms + * when VAG fully ramped down. + * As longer we wait, as smaller pop we've got. + */ + msleep(SGTL5000_VAG_POWERDOWN_DELAY); +} + /* * mic_bias power on/off share the same register bits with * output impedance of mic bias, when power on mic bias, we @@ -166,36 +289,46 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w, return 0; } -/* - * As manual described, ADC/DAC only works when VAG powerup, - * So enabled VAG before ADC/DAC up. - * In power down case, we need wait 400ms when vag fully ramped down. - */ -static int power_vag_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) +static int vag_and_mute_control(struct snd_soc_component *component, + int event, int event_source) { - struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); - const u32 mask = SGTL5000_DAC_POWERUP | SGTL5000_ADC_POWERUP; + static const u16 mute_mask[] = { + /* + * Mask for HP_POWER_EVENT. + * Muxing Headphones have to be wrapped with mute/unmute + * headphones only. + */ + SGTL5000_HP_MUTE, + /* + * Masks for DAC_POWER_EVENT/ADC_POWER_EVENT. + * Muxing DAC or ADC block have to be wrapped with mute/unmute + * both headphones and line-out. + */ + SGTL5000_OUTPUTS_MUTE, + SGTL5000_OUTPUTS_MUTE + }; + + struct sgtl5000_priv *sgtl5000 = + snd_soc_component_get_drvdata(component); switch (event) { + case SND_SOC_DAPM_PRE_PMU: + sgtl5000->mute_state[event_source] = + mute_output(component, mute_mask[event_source]); + break; case SND_SOC_DAPM_POST_PMU: - snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, - SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP); - msleep(400); + vag_power_on(component, event_source); + restore_output(component, mute_mask[event_source], + sgtl5000->mute_state[event_source]); break; - case SND_SOC_DAPM_PRE_PMD: - /* - * Don't clear VAG_POWERUP, when both DAC and ADC are - * operational to prevent inadvertently starving the - * other one of them. - */ - if ((snd_soc_read(codec, SGTL5000_CHIP_ANA_POWER) & - mask) != mask) { - snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, - SGTL5000_VAG_POWERUP, 0); - msleep(400); - } + sgtl5000->mute_state[event_source] = + mute_output(component, mute_mask[event_source]); + vag_power_off(component, event_source); + break; + case SND_SOC_DAPM_POST_PMD: + restore_output(component, mute_mask[event_source], + sgtl5000->mute_state[event_source]); break; default: break; @@ -204,6 +337,41 @@ static int power_vag_event(struct snd_soc_dapm_widget *w, return 0; } +/* + * Mute Headphone when power it up/down. + * Control VAG power on HP power path. + */ +static int headphone_pga_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + + return vag_and_mute_control(component, event, HP_POWER_EVENT); +} + +/* As manual describes, ADC/DAC powering up/down requires + * to mute outputs to avoid pops. + * Control VAG power on ADC/DAC power path. + */ +static int adc_updown_depop(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + + return vag_and_mute_control(component, event, ADC_POWER_EVENT); +} + +static int dac_updown_depop(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + + return vag_and_mute_control(component, event, DAC_POWER_EVENT); +} + /* input sources for ADC */ static const char *adc_mux_text[] = { "MIC_IN", "LINE_IN" @@ -239,7 +407,10 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = { mic_bias_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), - SND_SOC_DAPM_PGA("HP", SGTL5000_CHIP_ANA_POWER, 4, 0, NULL, 0), + SND_SOC_DAPM_PGA_E("HP", SGTL5000_CHIP_ANA_POWER, 4, 0, NULL, 0, + headphone_pga_event, + SND_SOC_DAPM_PRE_POST_PMU | + SND_SOC_DAPM_PRE_POST_PMD), SND_SOC_DAPM_PGA("LO", SGTL5000_CHIP_ANA_POWER, 0, 0, NULL, 0), SND_SOC_DAPM_MUX("Capture Mux", SND_SOC_NOPM, 0, 0, &adc_mux), @@ -255,11 +426,12 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = { 0, SGTL5000_CHIP_DIG_POWER, 1, 0), - SND_SOC_DAPM_ADC("ADC", "Capture", SGTL5000_CHIP_ANA_POWER, 1, 0), - SND_SOC_DAPM_DAC("DAC", "Playback", SGTL5000_CHIP_ANA_POWER, 3, 0), - - SND_SOC_DAPM_PRE("VAG_POWER_PRE", power_vag_event), - SND_SOC_DAPM_POST("VAG_POWER_POST", power_vag_event), + SND_SOC_DAPM_ADC_E("ADC", "Capture", SGTL5000_CHIP_ANA_POWER, 1, 0, + adc_updown_depop, SND_SOC_DAPM_PRE_POST_PMU | + SND_SOC_DAPM_PRE_POST_PMD), + SND_SOC_DAPM_DAC_E("DAC", "Playback", SGTL5000_CHIP_ANA_POWER, 3, 0, + dac_updown_depop, SND_SOC_DAPM_PRE_POST_PMU | + SND_SOC_DAPM_PRE_POST_PMD), }; /* routes for sgtl5000 */ @@ -492,6 +664,7 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { SGTL5000_CHIP_ANA_ADC_CTRL, 8, 1, 0, capture_6db_attenuate), SOC_SINGLE("Capture ZC Switch", SGTL5000_CHIP_ANA_CTRL, 1, 1, 0), + SOC_SINGLE("Capture Switch", SGTL5000_CHIP_ANA_CTRL, 0, 1, 1), SOC_DOUBLE_TLV("Headphone Playback Volume", SGTL5000_CHIP_ANA_HP_CTRL, @@ -1084,12 +1257,17 @@ static int sgtl5000_set_power_regs(struct snd_soc_codec *codec) SGTL5000_INT_OSC_EN); /* Enable VDDC charge pump */ ana_pwr |= SGTL5000_VDDC_CHRGPMP_POWERUP; - } else if (vddio >= 3100 && vdda >= 3100) { + } else { ana_pwr &= ~SGTL5000_VDDC_CHRGPMP_POWERUP; - /* VDDC use VDDIO rail */ - lreg_ctrl |= SGTL5000_VDDC_ASSN_OVRD; - lreg_ctrl |= SGTL5000_VDDC_MAN_ASSN_VDDIO << - SGTL5000_VDDC_MAN_ASSN_SHIFT; + /* + * if vddio == vdda the source of charge pump should be + * assigned manually to VDDIO + */ + if (vddio == vdda) { + lreg_ctrl |= SGTL5000_VDDC_ASSN_OVRD; + lreg_ctrl |= SGTL5000_VDDC_MAN_ASSN_VDDIO << + SGTL5000_VDDC_MAN_ASSN_SHIFT; + } } snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL, lreg_ctrl); @@ -1199,6 +1377,7 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) int ret; u16 reg; struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + unsigned int zcd_mask = SGTL5000_HP_ZCD_EN | SGTL5000_ADC_ZCD_EN; /* power up sgtl5000 */ ret = sgtl5000_set_power_regs(codec); @@ -1207,7 +1386,7 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) /* enable small pop, introduce 400ms delay in turning off */ snd_soc_update_bits(codec, SGTL5000_CHIP_REF_CTRL, - SGTL5000_SMALL_POP, 1); + SGTL5000_SMALL_POP, SGTL5000_SMALL_POP); /* disable short cut detector */ snd_soc_write(codec, SGTL5000_CHIP_SHORT_CTRL, 0); @@ -1230,9 +1409,8 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) reg = ((sgtl5000->lrclk_strength) << SGTL5000_PAD_I2S_LRCLK_SHIFT | 0x5f); snd_soc_write(codec, SGTL5000_CHIP_PAD_STRENGTH, reg); - snd_soc_write(codec, SGTL5000_CHIP_ANA_CTRL, - SGTL5000_HP_ZCD_EN | - SGTL5000_ADC_ZCD_EN); + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_CTRL, + zcd_mask, zcd_mask); snd_soc_update_bits(codec, SGTL5000_CHIP_MIC_CTRL, SGTL5000_BIAS_R_MASK, @@ -1259,9 +1437,35 @@ static int sgtl5000_remove(struct snd_soc_codec *codec) return 0; } +static int sgtl5000_suspend(struct snd_soc_codec *codec) +{ + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + + clk_disable_unprepare(sgtl5000->mclk); + + return 0; +} + +static int sgtl5000_resume(struct snd_soc_codec *codec) +{ + int ret; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + + ret = clk_prepare_enable(sgtl5000->mclk); + if (ret) + dev_err(codec->dev, "Error enabling clock %d\n", ret); + + /* Need 8 clocks before I2C accesses */ + udelay(1); + + return ret; +} + static const struct snd_soc_codec_driver sgtl5000_driver = { .probe = sgtl5000_probe, .remove = sgtl5000_remove, + .suspend = sgtl5000_suspend, + .resume = sgtl5000_resume, .set_bias_level = sgtl5000_set_bias_level, .suspend_bias_off = true, .component_driver = { diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h index 22f3442af982..1c62073000de 100644 --- a/sound/soc/codecs/sgtl5000.h +++ b/sound/soc/codecs/sgtl5000.h @@ -276,7 +276,7 @@ #define SGTL5000_BIAS_CTRL_MASK 0x000e #define SGTL5000_BIAS_CTRL_SHIFT 1 #define SGTL5000_BIAS_CTRL_WIDTH 3 -#define SGTL5000_SMALL_POP 1 +#define SGTL5000_SMALL_POP 0x0001 /* * SGTL5000_CHIP_MIC_CTRL diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 54a87a905eb6..cc95c15ceceb 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -924,23 +924,31 @@ static int aic31xx_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } + /* signal polarity */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + iface_reg2 |= AIC31XX_BCLKINV_MASK; + break; + default: + dev_err(codec->dev, "Invalid DAI clock signal polarity\n"); + return -EINVAL; + } + /* interface format */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: break; case SND_SOC_DAIFMT_DSP_A: - dsp_a_val = 0x1; + dsp_a_val = 0x1; /* fall through */ case SND_SOC_DAIFMT_DSP_B: - /* NOTE: BCLKINV bit value 1 equas NB and 0 equals IB */ - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_NB_NF: - iface_reg2 |= AIC31XX_BCLKINV_MASK; - break; - case SND_SOC_DAIFMT_IB_NF: - break; - default: - return -EINVAL; - } + /* + * NOTE: This CODEC samples on the falling edge of BCLK in + * DSP mode, this is inverted compared to what most DAIs + * expect, so we invert for this mode + */ + iface_reg2 ^= AIC31XX_BCLKINV_MASK; iface_reg1 |= (AIC31XX_DSP_MODE << AIC31XX_IFACE1_DATATYPE_SHIFT); break; diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index e694f5f04eb9..628621fc3386 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -462,6 +462,8 @@ static const struct snd_soc_dapm_widget aic32x4_dapm_widgets[] = { SND_SOC_DAPM_INPUT("IN2_R"), SND_SOC_DAPM_INPUT("IN3_L"), SND_SOC_DAPM_INPUT("IN3_R"), + SND_SOC_DAPM_INPUT("CM_L"), + SND_SOC_DAPM_INPUT("CM_R"), }; static const struct snd_soc_dapm_route aic32x4_dapm_routes[] = { diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c index f0cb1c4afe3c..c5a8d758f58b 100644 --- a/sound/soc/codecs/wm8737.c +++ b/sound/soc/codecs/wm8737.c @@ -170,7 +170,7 @@ SOC_DOUBLE("Polarity Invert Switch", WM8737_ADC_CONTROL, 5, 6, 1, 0), SOC_SINGLE("3D Switch", WM8737_3D_ENHANCE, 0, 1, 0), SOC_SINGLE("3D Depth", WM8737_3D_ENHANCE, 1, 15, 0), SOC_ENUM("3D Low Cut-off", low_3d), -SOC_ENUM("3D High Cut-off", low_3d), +SOC_ENUM("3D High Cut-off", high_3d), SOC_SINGLE_TLV("3D ADC Volume", WM8737_3D_ENHANCE, 7, 1, 1, adc_tlv), SOC_SINGLE("Noise Gate Switch", WM8737_NOISE_GATE, 0, 1, 0), diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 4fd350e8420d..2782b8064542 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1408,6 +1408,13 @@ static int wm8904_set_sysclk(struct snd_soc_dai *dai, int clk_id, struct snd_soc_codec *codec = dai->codec; struct wm8904_priv *priv = snd_soc_codec_get_drvdata(codec); + /* + * If using sound-simple-card this is called with clk_id fixed to 0. + * Assume we want WM8904_CLK_MCLK for now in that case. + */ + if (clk_id == 0) + clk_id = WM8904_CLK_MCLK; + switch (clk_id) { case WM8904_CLK_MCLK: priv->sysclk_src = clk_id; diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 67330b6ab204..158ce68bc9bf 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1169,8 +1169,7 @@ static unsigned int wmfw_convert_flags(unsigned int in, unsigned int len) } if (in) { - if (in & WMFW_CTL_FLAG_READABLE) - out |= rd; + out |= rd; if (in & WMFW_CTL_FLAG_WRITEABLE) out |= wr; if (in & WMFW_CTL_FLAG_VOLATILE) @@ -3711,11 +3710,13 @@ irqreturn_t wm_adsp2_bus_error(struct wm_adsp *dsp) struct regmap *regmap = dsp->regmap; int ret = 0; + mutex_lock(&dsp->pwr_lock); + ret = regmap_read(regmap, dsp->base + ADSP2_LOCK_REGION_CTRL, &val); if (ret) { adsp_err(dsp, "Failed to read Region Lock Ctrl register: %d\n", ret); - return IRQ_HANDLED; + goto error; } if (val & ADSP2_WDT_TIMEOUT_STS_MASK) { @@ -3734,7 +3735,7 @@ irqreturn_t wm_adsp2_bus_error(struct wm_adsp *dsp) adsp_err(dsp, "Failed to read Bus Err Addr register: %d\n", ret); - return IRQ_HANDLED; + goto error; } adsp_err(dsp, "bus error address = 0x%x\n", @@ -3747,7 +3748,7 @@ irqreturn_t wm_adsp2_bus_error(struct wm_adsp *dsp) adsp_err(dsp, "Failed to read Pmem Xmem Err Addr register: %d\n", ret); - return IRQ_HANDLED; + goto error; } adsp_err(dsp, "xmem error address = 0x%x\n", @@ -3760,6 +3761,9 @@ irqreturn_t wm_adsp2_bus_error(struct wm_adsp *dsp) regmap_update_bits(regmap, dsp->base + ADSP2_LOCK_REGION_CTRL, ADSP2_CTRL_ERR_EINT, ADSP2_CTRL_ERR_EINT); +error: + mutex_unlock(&dsp->pwr_lock); + return IRQ_HANDLED; } EXPORT_SYMBOL_GPL(wm_adsp2_bus_error); diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index f395bbc7c354..e10e03800cce 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -43,6 +43,7 @@ #define MCASP_MAX_AFIFO_DEPTH 64 +#ifdef CONFIG_PM static u32 context_regs[] = { DAVINCI_MCASP_TXFMCTL_REG, DAVINCI_MCASP_RXFMCTL_REG, @@ -65,6 +66,7 @@ struct davinci_mcasp_context { u32 *xrsr_regs; /* for serializer configuration */ bool pm_state; }; +#endif struct davinci_mcasp_ruledata { struct davinci_mcasp *mcasp; @@ -880,14 +882,13 @@ static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream, active_slots = hweight32(mcasp->tdm_mask[stream]); active_serializers = (channels + active_slots - 1) / active_slots; - if (active_serializers == 1) { + if (active_serializers == 1) active_slots = channels; - for (i = 0; i < total_slots; i++) { - if ((1 << i) & mcasp->tdm_mask[stream]) { - mask |= (1 << i); - if (--active_slots <= 0) - break; - } + for (i = 0; i < total_slots; i++) { + if ((1 << i) & mcasp->tdm_mask[stream]) { + mask |= (1 << i); + if (--active_slots <= 0) + break; } } } else { @@ -1156,6 +1157,28 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream, return ret; } +static int davinci_mcasp_hw_rule_slot_width(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct davinci_mcasp_ruledata *rd = rule->private; + struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + struct snd_mask nfmt; + int i, slot_width; + + snd_mask_none(&nfmt); + slot_width = rd->mcasp->slot_width; + + for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) { + if (snd_mask_test(fmt, i)) { + if (snd_pcm_format_width(i) <= slot_width) { + snd_mask_set(&nfmt, i); + } + } + } + + return snd_mask_refine(fmt, &nfmt); +} + static const unsigned int davinci_mcasp_dai_rates[] = { 8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000, 88200, 96000, 176400, 192000, @@ -1249,7 +1272,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, struct davinci_mcasp_ruledata *ruledata = &mcasp->ruledata[substream->stream]; u32 max_channels = 0; - int i, dir; + int i, dir, ret; int tdm_slots = mcasp->tdm_slots; /* Do not allow more then one stream per direction */ @@ -1278,6 +1301,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, max_channels++; } ruledata->serializers = max_channels; + ruledata->mcasp = mcasp; max_channels *= tdm_slots; /* * If the already active stream has less channels than the calculated @@ -1303,20 +1327,22 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, &mcasp->chconstr[substream->stream]); - if (mcasp->slot_width) - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_SAMPLE_BITS, - 8, mcasp->slot_width); + if (mcasp->slot_width) { + /* Only allow formats require <= slot_width bits on the bus */ + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_FORMAT, + davinci_mcasp_hw_rule_slot_width, + ruledata, + SNDRV_PCM_HW_PARAM_FORMAT, -1); + if (ret) + return ret; + } /* * If we rely on implicit BCLK divider setting we should * set constraints based on what we can provide. */ if (mcasp->bclk_master && mcasp->bclk_div == 0 && mcasp->sysclk_freq) { - int ret; - - ruledata->mcasp = mcasp; - ret = snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, davinci_mcasp_hw_rule_rate, @@ -1721,7 +1747,8 @@ static int davinci_mcasp_get_dma_type(struct davinci_mcasp *mcasp) PTR_ERR(chan)); return PTR_ERR(chan); } - BUG_ON(!chan->device || !chan->device->dev); + if (WARN_ON(!chan->device || !chan->device->dev)) + return -EINVAL; if (chan->device->dev->of_node) ret = of_property_read_string(chan->device->dev->of_node, @@ -1867,6 +1894,10 @@ static int davinci_mcasp_probe(struct platform_device *pdev) if (irq >= 0) { irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_common", dev_name(&pdev->dev)); + if (!irq_name) { + ret = -ENOMEM; + goto err; + } ret = devm_request_threaded_irq(&pdev->dev, irq, NULL, davinci_mcasp_common_irq_handler, IRQF_ONESHOT | IRQF_SHARED, @@ -1884,6 +1915,10 @@ static int davinci_mcasp_probe(struct platform_device *pdev) if (irq >= 0) { irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_rx", dev_name(&pdev->dev)); + if (!irq_name) { + ret = -ENOMEM; + goto err; + } ret = devm_request_threaded_irq(&pdev->dev, irq, NULL, davinci_mcasp_rx_irq_handler, IRQF_ONESHOT, irq_name, mcasp); @@ -1899,6 +1934,10 @@ static int davinci_mcasp_probe(struct platform_device *pdev) if (irq >= 0) { irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_tx", dev_name(&pdev->dev)); + if (!irq_name) { + ret = -ENOMEM; + goto err; + } ret = devm_request_threaded_irq(&pdev->dev, irq, NULL, davinci_mcasp_tx_irq_handler, IRQF_ONESHOT, irq_name, mcasp); @@ -1982,8 +2021,10 @@ static int davinci_mcasp_probe(struct platform_device *pdev) GFP_KERNEL); if (!mcasp->chconstr[SNDRV_PCM_STREAM_PLAYBACK].list || - !mcasp->chconstr[SNDRV_PCM_STREAM_CAPTURE].list) - return -ENOMEM; + !mcasp->chconstr[SNDRV_PCM_STREAM_CAPTURE].list) { + ret = -ENOMEM; + goto err; + } ret = davinci_mcasp_set_ch_constraints(mcasp); if (ret) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 62e89752b212..91ab4c121357 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -55,6 +55,7 @@ config SND_SOC_FSL_SSI config SND_SOC_FSL_SPDIF tristate "Sony/Philips Digital Interface (S/PDIF) module support" select REGMAP_MMIO + select SND_SOC_FSL_DMA_WORKAROUND select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && (MXC_TZIC || MXC_AVIC) select BITREVERSE @@ -67,6 +68,7 @@ config SND_SOC_FSL_SPDIF config SND_SOC_FSL_ESAI tristate "Enhanced Serial Audio Interface (ESAI) module support" select REGMAP_MMIO + select SND_SOC_FSL_DMA_WORKAROUND select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n help Say Y if you want to add Enhanced Synchronous Audio Interface @@ -111,6 +113,9 @@ config SND_SOC_FSL_DSP This option is only useful for out-of-tree drivers since in-tree drivers select it automatically. +config SND_SOC_FSL_DMA_WORKAROUND + tristate + config SND_SOC_FSL_UTILS tristate @@ -235,16 +240,17 @@ config SND_MPC52xx_SOC_EFIKA endif # SND_POWERPC_SOC +config SND_SOC_IMX_PCM_FIQ + tristate + default y if SND_SOC_IMX_SSI=y && (SND_SOC_FSL_SSI=m || SND_SOC_FSL_SPDIF=m) && (MXC_TZIC || MXC_AVIC) + select FIQ + if SND_IMX_SOC config SND_SOC_IMX_SSI tristate select SND_SOC_FSL_UTILS -config SND_SOC_IMX_PCM_FIQ - tristate - select FIQ - config SND_SOC_IMX_HDMI_DMA bool select SND_SOC_GENERIC_DMAENGINE_PCM @@ -288,7 +294,7 @@ config SND_SOC_PHYCORE_AC97 config SND_SOC_EUKREA_TLV320 tristate "Eukrea TLV320" - depends on ARCH_MXC && I2C + depends on ARCH_MXC && !ARM64 && I2C select SND_SOC_TLV320AIC23_I2C select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index c36369e9545b..b88175388741 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -23,7 +23,7 @@ snd-soc-fsl-sai-objs := fsl_sai.o snd-soc-fsl-ssi-y := fsl_ssi.o snd-soc-fsl-ssi-$(CONFIG_DEBUG_FS) += fsl_ssi_dbg.o snd-soc-fsl-spdif-objs := fsl_spdif.o -snd-soc-fsl-esai-objs := fsl_esai.o fsl_dma_workaround.o +snd-soc-fsl-esai-objs := fsl_esai.o snd-soc-fsl-utils-objs := fsl_utils.o snd-soc-fsl-dma-objs := fsl_dma.o snd-soc-fsl-rpmsg-i2s-objs := fsl_rpmsg_i2s.o @@ -35,6 +35,7 @@ snd-soc-fsl-easrc-objs := fsl_easrc.o fsl_easrc_dma.o obj-$(CONFIG_SND_SOC_FSL_ACM) += snd-soc-fsl-acm.o obj-$(CONFIG_SND_SOC_FSL_AMIX) += snd-soc-fsl-amix.o obj-$(CONFIG_SND_SOC_FSL_ASRC) += snd-soc-fsl-asrc.o +obj-$(CONFIG_SND_SOC_FSL_DMA_WORKAROUND) += snd-soc-fsl-dma-workaround.o obj-$(CONFIG_SND_SOC_FSL_DSP) += snd-soc-fsl-dsp.o obj-$(CONFIG_SND_SOC_FSL_SAI) += snd-soc-fsl-sai.o obj-$(CONFIG_SND_SOC_FSL_SSI) += snd-soc-fsl-ssi.o diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index 84ef6385736c..4c6f19ef98b2 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -119,13 +119,13 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) if (ret) { dev_err(&pdev->dev, "fsl,mux-int-port node missing or invalid.\n"); - return ret; + goto err; } ret = of_property_read_u32(np, "fsl,mux-ext-port", &ext_port); if (ret) { dev_err(&pdev->dev, "fsl,mux-ext-port node missing or invalid.\n"); - return ret; + goto err; } /* diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 2db4d0c80d33..393100edd5fd 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -689,6 +689,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) asrc_fail: of_node_put(asrc_np); of_node_put(codec_np); + put_device(&cpu_pdev->dev); fail: of_node_put(cpu_np); diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index 771467dfc503..c65bffe99cb9 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -378,8 +378,8 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair, bool p2p_in, bool p2 return -EINVAL; } - if ((outrate > 8000 && outrate < 30000) && - (outrate/inrate > 24 || inrate/outrate > 8)) { + if ((outrate >= 8000 && outrate <= 30000) && + (outrate > 24 * inrate || inrate > 8 * outrate)) { pair_err("exceed supported ratio range [1/24, 8] for \ inrate/outrate: %d/%d\n", inrate, outrate); return -EINVAL; diff --git a/sound/soc/fsl/fsl_dsp.c b/sound/soc/fsl/fsl_dsp.c index 35733841046c..39a5b13c6cb6 100644 --- a/sound/soc/fsl/fsl_dsp.c +++ b/sound/soc/fsl/fsl_dsp.c @@ -71,6 +71,45 @@ #include "fsl_dsp_pool.h" #include "fsl_dsp_xaf_api.h" +#define DSP_DISABLE_FUSE 0x8 +#define DSP_DISABLE_MASK 0x1 + +static int check_dsp_is_available(void) +{ + sc_ipc_t mu_ipc; + sc_ipc_id_t mu_id; + uint32_t fuse = 0xffff; + int ret; + + ret = sc_ipc_getMuID(&mu_id); + if (ret) { + /* We're not running on a iMX8 or iMX8X, so there is no DSP */ + return -EINVAL; + } + + ret = sc_ipc_open(&mu_ipc, mu_id); + if (ret) { + pr_err("sc_ipc_getMuID() can't open MU channel to SCU! %d\n", + ret); + return -EINVAL; + } + + ret = sc_misc_otp_fuse_read(mu_ipc, DSP_DISABLE_FUSE, &fuse); + sc_ipc_close(mu_ipc); + if (ret) { + pr_err("sc_misc_otp_fuse_read fail! %d\n", ret); + return -EINVAL; + } + + pr_debug("mu_id = %d, fuse[%i] = 0x%x\n", mu_id, DSP_DISABLE_FUSE, fuse); + if (fuse & DSP_DISABLE_MASK) { + pr_info("%s: HiFi4 DSP not available on this silicon\n", __func__); + return -EINVAL; + } + + return 0; +} + /* ...allocate new client */ struct xf_client *xf_client_alloc(struct fsl_dsp *dsp_priv) { @@ -1159,7 +1198,22 @@ static struct platform_driver fsl_dsp_driver = { .pm = &fsl_dsp_pm, }, }; -module_platform_driver(fsl_dsp_driver); + +static int __init fsl_dsp_driver_init(void) +{ + /* do not install the driver if no DSP is found */ + if (check_dsp_is_available()) + return -EINVAL; + + return platform_driver_register(&fsl_dsp_driver); +} +module_init(fsl_dsp_driver_init); + +static void __exit fsl_dsp_driver_exit(void) +{ + platform_driver_unregister(&fsl_dsp_driver); +} +module_exit(fsl_dsp_driver_exit); MODULE_DESCRIPTION("Freescale DSP driver"); MODULE_ALIAS("platform:fsl-dsp"); diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 8bcd3f918a5a..44e1af2e38ef 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -438,8 +438,8 @@ static int fsl_esai_set_dai_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, esai_priv->slot_width = slot_width; esai_priv->slots = slots; - esai_priv->tx_mask = tx_mask; - esai_priv->rx_mask = rx_mask; + esai_priv->tx_mask = tx_mask; + esai_priv->rx_mask = rx_mask; return 0; } @@ -556,6 +556,11 @@ static int fsl_esai_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, mask, xcr); regmap_update_bits(esai_priv->regmap, REG_ESAI_RCR, mask, xcr); + mask = ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP | ESAI_xCCR_xFSP | + ESAI_xCCR_xFSD | ESAI_xCCR_xCKD; + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, mask, xccr); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, mask, xccr); + return 0; } @@ -699,6 +704,18 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, for (i = 0; tx && i < channels; i++) regmap_write(esai_priv->regmap, REG_ESAI_ETDR, 0x0); + /* + * When set the TE/RE in the end of enablement flow, there + * will be channel swap issue for multi data line case. + * In order to workaround this issue, we switch the bit + * enablement sequence to below sequence + * 1) clear the xSMB & xSMA: which is done in probe and + * stop state. + * 2) set TE/RE + * 3) set xSMB + * 4) set xSMA: xSMA is the last one in this flow, which + * will trigger esai to start. + */ regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), tx ? ESAI_xCR_TE_MASK : ESAI_xCR_RE_MASK, tx ? ESAI_xCR_TE(pins) : ESAI_xCR_RE(pins)); @@ -1143,6 +1160,9 @@ static int fsl_esai_probe(struct platform_device *pdev) return ret; } + esai_priv->tx_mask = 0xFFFFFFFF; + esai_priv->rx_mask = 0xFFFFFFFF; + /* Clear the TSMA, TSMB, RSMA, RSMB */ regmap_write(esai_priv->regmap, REG_ESAI_TSMA, 0); regmap_write(esai_priv->regmap, REG_ESAI_TSMB, 0); diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 00537c96bbd1..a13d5fbc285d 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -423,12 +423,14 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, case SND_SOC_DAIFMT_CBS_CFS: val_cr2 |= FSL_SAI_CR2_BCD_MSTR; val_cr4 |= FSL_SAI_CR4_FSD_MSTR; + sai->slave_mode[tx] = false; break; case SND_SOC_DAIFMT_CBM_CFM: sai->slave_mode[tx] = true; break; case SND_SOC_DAIFMT_CBS_CFM: val_cr2 |= FSL_SAI_CR2_BCD_MSTR; + sai->slave_mode[tx] = false; break; case SND_SOC_DAIFMT_CBM_CFS: val_cr4 |= FSL_SAI_CR4_FSD_MSTR; diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 3ab5c7fc286c..b93da5a1822c 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -548,10 +548,10 @@ static int fsl_spdif_startup(struct snd_pcm_substream *substream, if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { scr = SCR_TXFIFO_AUTOSYNC | SCR_TXFIFO_CTRL_NORMAL | SCR_TXSEL_NORMAL | SCR_USRC_SEL_CHIP | - SCR_TXFIFO_FSEL_IF8; + SCR_TXFIFO_FSEL_IF8 | SCR_VAL_CLEAR; mask = SCR_TXFIFO_AUTOSYNC_MASK | SCR_TXFIFO_CTRL_MASK | SCR_TXSEL_MASK | SCR_USRC_SEL_MASK | - SCR_TXFIFO_FSEL_MASK; + SCR_TXFIFO_FSEL_MASK | SCR_VAL_MASK; } else { scr = SCR_RXFIFO_FSEL_IF8 | SCR_RXFIFO_AUTOSYNC; mask = SCR_RXFIFO_FSEL_MASK | SCR_RXFIFO_AUTOSYNC_MASK| diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 45e9de81cea9..1245db8451a1 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1459,6 +1459,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) struct fsl_ssi_private *ssi_private; int ret = 0; struct device_node *np = pdev->dev.of_node; + struct device_node *root; const struct of_device_id *of_id; const char *p, *sprop; const uint32_t *iprop; @@ -1648,7 +1649,9 @@ static int fsl_ssi_probe(struct platform_device *pdev) * device tree. We also pass the address of the CPU DAI driver * structure. */ - sprop = of_get_property(of_find_node_by_path("/"), "compatible", NULL); + root = of_find_node_by_path("/"); + sprop = of_get_property(root, "compatible", NULL); + of_node_put(root); /* Sometimes the compatible name has a "fsl," prefix, so we strip it. */ p = strrchr(sprop, ','); if (p) diff --git a/sound/soc/fsl/fsl_utils.c b/sound/soc/fsl/fsl_utils.c index b9e42b503a37..4f8bdb7650e8 100644 --- a/sound/soc/fsl/fsl_utils.c +++ b/sound/soc/fsl/fsl_utils.c @@ -75,6 +75,7 @@ int fsl_asoc_get_dma_channel(struct device_node *ssi_np, iprop = of_get_property(dma_np, "cell-index", NULL); if (!iprop) { of_node_put(dma_np); + of_node_put(dma_channel_np); return -EINVAL; } *dma_id = be32_to_cpup(iprop); diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index 016a863204f2..f01a13ee02aa 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -88,49 +88,49 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf, if (!buf) return -ENOMEM; - ret = snprintf(buf, PAGE_SIZE, "PDCR: %08x\nPTCR: %08x\n", + ret = scnprintf(buf, PAGE_SIZE, "PDCR: %08x\nPTCR: %08x\n", pdcr, ptcr); if (ptcr & IMX_AUDMUX_V2_PTCR_TFSDIR) - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "TxFS output from %s, ", audmux_port_string((ptcr >> 27) & 0x7)); else - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "TxFS input, "); if (ptcr & IMX_AUDMUX_V2_PTCR_TCLKDIR) - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "TxClk output from %s", audmux_port_string((ptcr >> 22) & 0x7)); else - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "TxClk input"); - ret += snprintf(buf + ret, PAGE_SIZE - ret, "\n"); + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "\n"); if (ptcr & IMX_AUDMUX_V2_PTCR_SYN) { - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "Port is symmetric"); } else { if (ptcr & IMX_AUDMUX_V2_PTCR_RFSDIR) - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "RxFS output from %s, ", audmux_port_string((ptcr >> 17) & 0x7)); else - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "RxFS input, "); if (ptcr & IMX_AUDMUX_V2_PTCR_RCLKDIR) - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "RxClk output from %s", audmux_port_string((ptcr >> 12) & 0x7)); else - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "RxClk input"); } - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "\nData received from %s\n", audmux_port_string((pdcr >> 13) & 0x7)); diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index b99e0b5e00e9..ab5c62f2d240 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -1,5 +1,5 @@ /* - * Copyright 2012 Freescale Semiconductor, Inc. + * Copyright 2012, 2014 Freescale Semiconductor, Inc. * Copyright 2012 Linaro Ltd. * * The code contained herein is licensed under the GNU General Public @@ -55,13 +55,9 @@ static const struct snd_soc_dapm_widget imx_sgtl5000_dapm_widgets[] = { SND_SOC_DAPM_SPK("Ext Spk", NULL), }; -static int imx_sgtl5000_probe(struct platform_device *pdev) +static int imx_sgtl5000_audmux_config(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; - struct device_node *ssi_np, *codec_np; - struct platform_device *ssi_pdev; - struct i2c_client *codec_dev; - struct imx_sgtl5000_data *data = NULL; int int_port, ext_port; int ret; @@ -101,24 +97,43 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) return ret; } - ssi_np = of_parse_phandle(pdev->dev.of_node, "ssi-controller", 0); + return 0; +} + +static int imx_sgtl5000_probe(struct platform_device *pdev) +{ + struct device_node *cpu_np, *codec_np; + struct platform_device *cpu_pdev; + struct i2c_client *codec_dev; + struct imx_sgtl5000_data *data = NULL; + int ret; + + cpu_np = of_parse_phandle(pdev->dev.of_node, "cpu-dai", 0); codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0); - if (!ssi_np || !codec_np) { + if (!cpu_np || !codec_np) { dev_err(&pdev->dev, "phandle missing or invalid\n"); ret = -EINVAL; goto fail; } - ssi_pdev = of_find_device_by_node(ssi_np); - if (!ssi_pdev) { + if (strstr(cpu_np->name, "ssi")) { + ret = imx_sgtl5000_audmux_config(pdev); + if (ret) + goto fail; + } + + cpu_pdev = of_find_device_by_node(cpu_np); + if (!cpu_pdev) { dev_err(&pdev->dev, "failed to find SSI platform device\n"); ret = -EPROBE_DEFER; goto fail; } + put_device(&cpu_pdev->dev); codec_dev = of_find_i2c_device_by_node(codec_np); if (!codec_dev) { dev_err(&pdev->dev, "failed to find codec platform device\n"); - return -EPROBE_DEFER; + ret = -EPROBE_DEFER; + goto fail; } data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); @@ -139,8 +154,8 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) data->dai.stream_name = "HiFi"; data->dai.codec_dai_name = "sgtl5000"; data->dai.codec_of_node = codec_np; - data->dai.cpu_of_node = ssi_np; - data->dai.platform_of_node = ssi_np; + data->dai.cpu_of_node = cpu_np; + data->dai.platform_of_node = cpu_np; data->dai.init = &imx_sgtl5000_dai_init; data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM; @@ -167,7 +182,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) goto fail; } - of_node_put(ssi_np); + of_node_put(cpu_np); of_node_put(codec_np); return 0; @@ -175,8 +190,10 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) fail: if (data && !IS_ERR(data->codec_clk)) clk_put(data->codec_clk); - of_node_put(ssi_np); - of_node_put(codec_np); + if (cpu_np) + of_node_put(cpu_np); + if (codec_np) + of_node_put(codec_np); return ret; } diff --git a/sound/soc/intel/atom/sst/sst_loader.c b/sound/soc/intel/atom/sst/sst_loader.c index 33917146d9c4..054b1d514e8a 100644 --- a/sound/soc/intel/atom/sst/sst_loader.c +++ b/sound/soc/intel/atom/sst/sst_loader.c @@ -354,14 +354,14 @@ static int sst_request_fw(struct intel_sst_drv *sst) const struct firmware *fw; retval = request_firmware(&fw, sst->firmware_name, sst->dev); - if (fw == NULL) { - dev_err(sst->dev, "fw is returning as null\n"); - return -EINVAL; - } if (retval) { dev_err(sst->dev, "request fw failed %d\n", retval); return retval; } + if (fw == NULL) { + dev_err(sst->dev, "fw is returning as null\n"); + return -EINVAL; + } mutex_lock(&sst->sst_lock); retval = sst_cache_and_parse_fw(sst, fw); mutex_unlock(&sst->sst_lock); diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c index 6dcbbcefc25b..88c26ab7b027 100644 --- a/sound/soc/intel/boards/broadwell.c +++ b/sound/soc/intel/boards/broadwell.c @@ -191,7 +191,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { .stream_name = "Loopback", .cpu_dai_name = "Loopback Pin", .platform_name = "haswell-pcm-audio", - .dynamic = 0, + .dynamic = 1, .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c index 5e1ea0371c90..8158409921e0 100644 --- a/sound/soc/intel/boards/haswell.c +++ b/sound/soc/intel/boards/haswell.c @@ -145,7 +145,7 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = { .stream_name = "Loopback", .cpu_dai_name = "Loopback Pin", .platform_name = "haswell-pcm-audio", - .dynamic = 0, + .dynamic = 1, .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 69ab55956492..405196283688 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -405,7 +405,7 @@ static const struct snd_pcm_hw_constraint_list constraints_dmic_channels = { }; static const unsigned int dmic_2ch[] = { - 4, + 2, }; static const struct snd_pcm_hw_constraint_list constraints_dmic_2ch = { @@ -422,6 +422,9 @@ static int kabylake_dmic_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, dmic_constraints); + runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; + snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16); + return snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); } diff --git a/sound/soc/intel/common/sst-firmware.c b/sound/soc/intel/common/sst-firmware.c index 79a9fdf94d38..582b30a5118d 100644 --- a/sound/soc/intel/common/sst-firmware.c +++ b/sound/soc/intel/common/sst-firmware.c @@ -1252,11 +1252,15 @@ struct sst_dsp *sst_dsp_new(struct device *dev, goto irq_err; err = sst_dma_new(sst); - if (err) - dev_warn(dev, "sst_dma_new failed %d\n", err); + if (err) { + dev_err(dev, "sst_dma_new failed %d\n", err); + goto dma_err; + } return sst; +dma_err: + free_irq(sst->irq, sst); irq_err: if (sst->ops->free) sst->ops->free(sst); diff --git a/sound/soc/intel/common/sst-ipc.c b/sound/soc/intel/common/sst-ipc.c index 62f3a8e0ec87..fedce78675e8 100644 --- a/sound/soc/intel/common/sst-ipc.c +++ b/sound/soc/intel/common/sst-ipc.c @@ -231,6 +231,8 @@ struct ipc_message *sst_ipc_reply_find_msg(struct sst_generic_ipc *ipc, if (ipc->ops.reply_msg_match != NULL) header = ipc->ops.reply_msg_match(header, &mask); + else + mask = (u64)-1; if (list_empty(&ipc->rx_list)) { dev_err(ipc->dev, "error: rx list empty but received 0x%llx\n", diff --git a/sound/soc/intel/skylake/skl-debug.c b/sound/soc/intel/skylake/skl-debug.c index dc20d91f62e6..1987f78ea91e 100644 --- a/sound/soc/intel/skylake/skl-debug.c +++ b/sound/soc/intel/skylake/skl-debug.c @@ -196,7 +196,7 @@ static ssize_t fw_softreg_read(struct file *file, char __user *user_buf, memset(d->fw_read_buff, 0, FW_REG_BUF); if (w0_stat_sz > 0) - __iowrite32_copy(d->fw_read_buff, fw_reg_addr, w0_stat_sz >> 2); + __ioread32_copy(d->fw_read_buff, fw_reg_addr, w0_stat_sz >> 2); for (offset = 0; offset < FW_REG_SIZE; offset += 16) { ret += snprintf(tmp + ret, FW_REG_BUF - ret, "%#.4x: ", offset); diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c index 55859c5b456f..1b0129478a7f 100644 --- a/sound/soc/intel/skylake/skl-nhlt.c +++ b/sound/soc/intel/skylake/skl-nhlt.c @@ -215,7 +215,7 @@ int skl_nhlt_update_topology_bin(struct skl *skl) struct hdac_bus *bus = ebus_to_hbus(&skl->ebus); struct device *dev = bus->dev; - dev_dbg(dev, "oem_id %.6s, oem_table_id %8s oem_revision %d\n", + dev_dbg(dev, "oem_id %.6s, oem_table_id %.8s oem_revision %d\n", nhlt->header.oem_id, nhlt->header.oem_table_id, nhlt->header.oem_revision); diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 105a73cc5158..149b7cba10fb 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -569,10 +569,6 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) return PTR_ERR(priv->clk); } - err = clk_prepare_enable(priv->clk); - if (err < 0) - return err; - priv->extclk = devm_clk_get(&pdev->dev, "extclk"); if (IS_ERR(priv->extclk)) { if (PTR_ERR(priv->extclk) == -EPROBE_DEFER) @@ -588,6 +584,10 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) } } + err = clk_prepare_enable(priv->clk); + if (err < 0) + return err; + /* Some sensible defaults - this reflects the powerup values */ priv->ctl_play = KIRKWOOD_PLAYCTL_SIZE_24; priv->ctl_rec = KIRKWOOD_RECCTL_SIZE_24; diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c index d49adc822a11..8e6b88d68ca6 100644 --- a/sound/soc/qcom/apq8016_sbc.c +++ b/sound/soc/qcom/apq8016_sbc.c @@ -163,41 +163,52 @@ static struct apq8016_sbc_data *apq8016_sbc_parse_of(struct snd_soc_card *card) if (!cpu || !codec) { dev_err(dev, "Can't find cpu/codec DT node\n"); - return ERR_PTR(-EINVAL); + ret = -EINVAL; + goto error; } link->cpu_of_node = of_parse_phandle(cpu, "sound-dai", 0); if (!link->cpu_of_node) { dev_err(card->dev, "error getting cpu phandle\n"); - return ERR_PTR(-EINVAL); + ret = -EINVAL; + goto error; } ret = snd_soc_of_get_dai_name(cpu, &link->cpu_dai_name); if (ret) { dev_err(card->dev, "error getting cpu dai name\n"); - return ERR_PTR(ret); + goto error; } ret = snd_soc_of_get_dai_link_codecs(dev, codec, link); if (ret < 0) { dev_err(card->dev, "error getting codec dai name\n"); - return ERR_PTR(ret); + goto error; } link->platform_of_node = link->cpu_of_node; ret = of_property_read_string(np, "link-name", &link->name); if (ret) { dev_err(card->dev, "error getting codec dai_link name\n"); - return ERR_PTR(ret); + goto error; } link->stream_name = link->name; link->init = apq8016_sbc_dai_init; link++; + + of_node_put(cpu); + of_node_put(codec); } return data; + + error: + of_node_put(np); + of_node_put(cpu); + of_node_put(codec); + return ERR_PTR(ret); } static const struct snd_soc_dapm_widget apq8016_sbc_dapm_widgets[] = { diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 66fc13a2396a..0e07e3dea7de 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -676,7 +676,7 @@ static int rockchip_i2s_probe(struct platform_device *pdev) ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); if (ret) { dev_err(&pdev->dev, "Could not register PCM\n"); - return ret; + goto err_suspend; } return 0; diff --git a/sound/soc/rockchip/rockchip_pdm.c b/sound/soc/rockchip/rockchip_pdm.c index 400e29edb1c9..8a2e3bbce3a1 100644 --- a/sound/soc/rockchip/rockchip_pdm.c +++ b/sound/soc/rockchip/rockchip_pdm.c @@ -208,7 +208,9 @@ static int rockchip_pdm_set_fmt(struct snd_soc_dai *cpu_dai, return -EINVAL; } + pm_runtime_get_sync(cpu_dai->dev); regmap_update_bits(pdm->regmap, PDM_CLK_CTRL, mask, val); + pm_runtime_put(cpu_dai->dev); return 0; } diff --git a/sound/soc/samsung/odroid.c b/sound/soc/samsung/odroid.c index 06a31a9585a0..32c9e197ca95 100644 --- a/sound/soc/samsung/odroid.c +++ b/sound/soc/samsung/odroid.c @@ -66,11 +66,11 @@ static int odroid_card_hw_params(struct snd_pcm_substream *substream, return ret; /* - * We add 1 to the rclk_freq value in order to avoid too low clock + * We add 2 to the rclk_freq value in order to avoid too low clock * frequency values due to the EPLL output frequency not being exact * multiple of the audio sampling rate. */ - rclk_freq = params_rate(params) * rfs + 1; + rclk_freq = params_rate(params) * rfs + 2; ret = clk_set_rate(priv->sclk_i2s, rclk_freq); if (ret < 0) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index eb7879bcc6a7..686401bcd1f5 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -33,6 +33,7 @@ struct rsnd_adg { struct clk *clkout[CLKOUTMAX]; struct clk_onecell_data onecell; struct rsnd_mod mod; + int clk_rate[CLKMAX]; u32 flags; u32 ckr; u32 rbga; @@ -110,9 +111,9 @@ static void __rsnd_adg_get_timesel_ratio(struct rsnd_priv *priv, unsigned int val, en; unsigned int min, diff; unsigned int sel_rate[] = { - clk_get_rate(adg->clk[CLKA]), /* 0000: CLKA */ - clk_get_rate(adg->clk[CLKB]), /* 0001: CLKB */ - clk_get_rate(adg->clk[CLKC]), /* 0010: CLKC */ + adg->clk_rate[CLKA], /* 0000: CLKA */ + adg->clk_rate[CLKB], /* 0001: CLKB */ + adg->clk_rate[CLKC], /* 0010: CLKC */ adg->rbga_rate_for_441khz, /* 0011: RBGA */ adg->rbgb_rate_for_48khz, /* 0100: RBGB */ }; @@ -328,7 +329,7 @@ int rsnd_adg_clk_query(struct rsnd_priv *priv, unsigned int rate) * AUDIO_CLKA/AUDIO_CLKB/AUDIO_CLKC/AUDIO_CLKI. */ for_each_rsnd_clk(clk, adg, i) { - if (rate == clk_get_rate(clk)) + if (rate == adg->clk_rate[i]) return sel_table[i]; } @@ -394,10 +395,18 @@ void rsnd_adg_clk_control(struct rsnd_priv *priv, int enable) for_each_rsnd_clk(clk, adg, i) { ret = 0; - if (enable) + if (enable) { ret = clk_prepare_enable(clk); - else + + /* + * We shouldn't use clk_get_rate() under + * atomic context. Let's keep it when + * rsnd_adg_clk_enable() was called + */ + adg->clk_rate[i] = clk_get_rate(adg->clk[i]); + } else { clk_disable_unprepare(clk); + } if (ret < 0) dev_warn(dev, "can't use clk %d\n", i); diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 710c01cd2ad2..f203c0878e69 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -676,6 +676,7 @@ static int rsnd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) } /* set format */ + rdai->bit_clk_inv = 0; switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: rdai->sys_delay = 0; @@ -1277,6 +1278,18 @@ int rsnd_kctrl_new(struct rsnd_mod *mod, }; int ret; + /* + * 1) Avoid duplicate register for DVC with MIX case + * 2) Allow duplicate register for MIX + * 3) re-register if card was rebinded + */ + list_for_each_entry(kctrl, &card->controls, list) { + struct rsnd_kctrl_cfg *c = kctrl->private_data; + + if (c == cfg) + return 0; + } + if (size > RSND_MAX_CHANNELS) return -EINVAL; diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 1768a0ae469d..c68b31483c7b 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -432,6 +432,7 @@ struct rsnd_dai_stream { char name[RSND_DAI_NAME_SIZE]; struct snd_pcm_substream *substream; struct rsnd_mod *mod[RSND_MOD_MAX]; + struct rsnd_mod *dma; struct rsnd_dai *rdai; u32 parent_ssi_status; }; diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 0db2791f7035..cae9ed6a0cdb 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -66,7 +66,6 @@ struct rsnd_ssi { struct rsnd_mod mod; - struct rsnd_mod *dma; u32 flags; u32 cr_own; @@ -280,7 +279,7 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod, if (rsnd_ssi_is_multi_slave(mod, io)) return 0; - if (ssi->usrcnt > 1) { + if (ssi->usrcnt > 0) { if (ssi->rate != rate) { dev_err(dev, "SSI parent/child should use same rate\n"); return -EINVAL; @@ -868,7 +867,6 @@ static int rsnd_ssi_dma_probe(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv) { - struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); int ret; /* @@ -883,7 +881,7 @@ static int rsnd_ssi_dma_probe(struct rsnd_mod *mod, return ret; /* SSI probe might be called many times in MUX multi path */ - ret = rsnd_dma_attach(io, mod, &ssi->dma); + ret = rsnd_dma_attach(io, mod, &io->dma); return ret; } diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 53c9d7525639..104d5f487c7d 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -75,12 +75,16 @@ static int dapm_up_seq[] = { [snd_soc_dapm_clock_supply] = 1, [snd_soc_dapm_supply] = 2, [snd_soc_dapm_micbias] = 3, + [snd_soc_dapm_vmid] = 3, [snd_soc_dapm_dai_link] = 2, [snd_soc_dapm_dai_in] = 4, [snd_soc_dapm_dai_out] = 4, [snd_soc_dapm_aif_in] = 4, [snd_soc_dapm_aif_out] = 4, [snd_soc_dapm_mic] = 5, + [snd_soc_dapm_siggen] = 5, + [snd_soc_dapm_input] = 5, + [snd_soc_dapm_output] = 5, [snd_soc_dapm_mux] = 6, [snd_soc_dapm_demux] = 6, [snd_soc_dapm_dac] = 7, @@ -88,11 +92,19 @@ static int dapm_up_seq[] = { [snd_soc_dapm_mixer] = 8, [snd_soc_dapm_mixer_named_ctl] = 8, [snd_soc_dapm_pga] = 9, + [snd_soc_dapm_buffer] = 9, + [snd_soc_dapm_scheduler] = 9, + [snd_soc_dapm_effect] = 9, + [snd_soc_dapm_src] = 9, + [snd_soc_dapm_asrc] = 9, + [snd_soc_dapm_encoder] = 9, + [snd_soc_dapm_decoder] = 9, [snd_soc_dapm_adc] = 10, [snd_soc_dapm_out_drv] = 11, [snd_soc_dapm_hp] = 11, [snd_soc_dapm_spk] = 11, [snd_soc_dapm_line] = 11, + [snd_soc_dapm_sink] = 11, [snd_soc_dapm_kcontrol] = 12, [snd_soc_dapm_post] = 13, }; @@ -105,13 +117,25 @@ static int dapm_down_seq[] = { [snd_soc_dapm_spk] = 3, [snd_soc_dapm_line] = 3, [snd_soc_dapm_out_drv] = 3, + [snd_soc_dapm_sink] = 3, [snd_soc_dapm_pga] = 4, + [snd_soc_dapm_buffer] = 4, + [snd_soc_dapm_scheduler] = 4, + [snd_soc_dapm_effect] = 4, + [snd_soc_dapm_src] = 4, + [snd_soc_dapm_asrc] = 4, + [snd_soc_dapm_encoder] = 4, + [snd_soc_dapm_decoder] = 4, [snd_soc_dapm_switch] = 5, [snd_soc_dapm_mixer_named_ctl] = 5, [snd_soc_dapm_mixer] = 5, [snd_soc_dapm_dac] = 6, [snd_soc_dapm_mic] = 7, + [snd_soc_dapm_siggen] = 7, + [snd_soc_dapm_input] = 7, + [snd_soc_dapm_output] = 7, [snd_soc_dapm_micbias] = 8, + [snd_soc_dapm_vmid] = 8, [snd_soc_dapm_mux] = 9, [snd_soc_dapm_demux] = 9, [snd_soc_dapm_aif_in] = 10, @@ -1128,8 +1152,8 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, list_add_tail(&widget->work_list, list); if (custom_stop_condition && custom_stop_condition(widget, dir)) { - widget->endpoints[dir] = 1; - return widget->endpoints[dir]; + list = NULL; + custom_stop_condition = NULL; } if ((widget->is_ep & SND_SOC_DAPM_DIR_TO_EP(dir)) && widget->connected) { @@ -1166,8 +1190,8 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, * * Optionally, can be supplied with a function acting as a stopping condition. * This function takes the dapm widget currently being examined and the walk - * direction as an arguments, it should return true if the walk should be - * stopped and false otherwise. + * direction as an arguments, it should return true if widgets from that point + * in the graph onwards should not be added to the widget list. */ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, struct list_head *list, @@ -2009,19 +2033,19 @@ static ssize_t dapm_widget_power_read_file(struct file *file, out = is_connected_output_ep(w, NULL, NULL); } - ret = snprintf(buf, PAGE_SIZE, "%s: %s%s in %d out %d", + ret = scnprintf(buf, PAGE_SIZE, "%s: %s%s in %d out %d", w->name, w->power ? "On" : "Off", w->force ? " (forced)" : "", in, out); if (w->reg >= 0) - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, " - R%d(0x%x) mask 0x%x", w->reg, w->reg, w->mask << w->shift); - ret += snprintf(buf + ret, PAGE_SIZE - ret, "\n"); + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "\n"); if (w->sname) - ret += snprintf(buf + ret, PAGE_SIZE - ret, " stream %s %s\n", + ret += scnprintf(buf + ret, PAGE_SIZE - ret, " stream %s %s\n", w->sname, w->active ? "active" : "inactive"); @@ -2034,7 +2058,7 @@ static ssize_t dapm_widget_power_read_file(struct file *file, if (!p->connect) continue; - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, " %s \"%s\" \"%s\"\n", (rdir == SND_SOC_DAPM_DIR_IN) ? "in" : "out", p->name ? p->name : "static", @@ -2096,23 +2120,25 @@ void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm, { struct dentry *d; - if (!parent) + if (!parent || IS_ERR(parent)) return; dapm->debugfs_dapm = debugfs_create_dir("dapm", parent); - if (!dapm->debugfs_dapm) { + if (IS_ERR(dapm->debugfs_dapm)) { dev_warn(dapm->dev, - "ASoC: Failed to create DAPM debugfs directory\n"); + "ASoC: Failed to create DAPM debugfs directory %ld\n", + PTR_ERR(dapm->debugfs_dapm)); return; } d = debugfs_create_file("bias_level", 0444, dapm->debugfs_dapm, dapm, &dapm_bias_fops); - if (!d) + if (IS_ERR(d)) dev_warn(dapm->dev, - "ASoC: Failed to create bias level debugfs file\n"); + "ASoC: Failed to create bias level debugfs file: %ld\n", + PTR_ERR(d)); } static void dapm_debugfs_add_widget(struct snd_soc_dapm_widget *w) @@ -2126,10 +2152,10 @@ static void dapm_debugfs_add_widget(struct snd_soc_dapm_widget *w) d = debugfs_create_file(w->name, 0444, dapm->debugfs_dapm, w, &dapm_widget_power_fops); - if (!d) + if (IS_ERR(d)) dev_warn(w->dapm->dev, - "ASoC: Failed to create %s debugfs file\n", - w->name); + "ASoC: Failed to create %s debugfs file: %ld\n", + w->name, PTR_ERR(d)); } static void dapm_debugfs_cleanup(struct snd_soc_dapm_context *dapm) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index d53786498b61..052778c6afad 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -311,6 +311,12 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) if (!dmaengine_pcm_can_report_residue(dev, pcm->chan[i])) pcm->flags |= SND_DMAENGINE_PCM_FLAG_NO_RESIDUE; + + if (rtd->pcm->streams[i].pcm->name[0] == '\0') { + strncpy(rtd->pcm->streams[i].pcm->name, + rtd->pcm->streams[i].pcm->id, + sizeof(rtd->pcm->streams[i].pcm->name)); + } } return 0; diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 99902ae1a2d9..b04ecc633da3 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -127,10 +127,9 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) unsigned int sync = 0; int enable; - trace_snd_soc_jack_report(jack, mask, status); - if (!jack) return; + trace_snd_soc_jack_report(jack, mask, status); dapm = &jack->card->dapm; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 7d021de2cd1b..a422c4ad9616 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -48,8 +48,8 @@ static bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream) else codec_stream = &dai->driver->capture; - /* If the codec specifies any rate at all, it supports the stream. */ - return codec_stream->rates; + /* If the codec specifies any channels at all, it supports the stream */ + return codec_stream->channels_min; } /** @@ -894,10 +894,13 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, codec_params = *params; /* fixup params based on TDM slot masks */ - if (codec_dai->tx_mask) + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && + codec_dai->tx_mask) soc_pcm_codec_params_fixup(&codec_params, codec_dai->tx_mask); - if (codec_dai->rx_mask) + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE && + codec_dai->rx_mask) soc_pcm_codec_params_fixup(&codec_params, codec_dai->rx_mask); @@ -1575,7 +1578,7 @@ static void dpcm_init_runtime_hw(struct snd_pcm_runtime *runtime, u64 formats) { runtime->hw.rate_min = stream->rate_min; - runtime->hw.rate_max = stream->rate_max; + runtime->hw.rate_max = min_not_zero(stream->rate_max, UINT_MAX); runtime->hw.channels_min = stream->channels_min; runtime->hw.channels_max = stream->channels_max; if (runtime->hw.formats) @@ -2273,7 +2276,8 @@ int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream) if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) && - (be->dpcm[stream].state != SND_SOC_DPCM_STATE_SUSPEND)) + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_SUSPEND) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; dev_dbg(be->dev, "ASoC: prepare BE %s\n", diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index c1619860a5de..72301bcad3bd 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1921,6 +1921,7 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg, int count = hdr->count; int i; bool abi_match; + int ret; if (tplg->pass != SOC_TPLG_PASS_PCM_DAI) return 0; @@ -1957,7 +1958,12 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg, } /* create the FE DAIs and DAI links */ - soc_tplg_pcm_create(tplg, _pcm); + ret = soc_tplg_pcm_create(tplg, _pcm); + if (ret < 0) { + if (!abi_match) + kfree(_pcm); + return ret; + } /* offset by version-specific struct size and * real priv data size @@ -2513,6 +2519,7 @@ int snd_soc_tplg_component_load(struct snd_soc_component *comp, struct snd_soc_tplg_ops *ops, const struct firmware *fw, u32 id) { struct soc_tplg tplg; + int ret; /* setup parsing context */ memset(&tplg, 0, sizeof(tplg)); @@ -2526,7 +2533,12 @@ int snd_soc_tplg_component_load(struct snd_soc_component *comp, tplg.bytes_ext_ops = ops->bytes_ext_ops; tplg.bytes_ext_ops_count = ops->bytes_ext_ops_count; - return soc_tplg_load(&tplg); + ret = soc_tplg_load(&tplg); + /* free the created components if fail to load topology */ + if (ret) + snd_soc_tplg_component_remove(comp, SND_SOC_TPLG_INDEX_ALL); + + return ret; } EXPORT_SYMBOL_GPL(snd_soc_tplg_component_load); diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c index d8b6936e544e..908f13623f8c 100644 --- a/sound/soc/sti/uniperif_player.c +++ b/sound/soc/sti/uniperif_player.c @@ -226,7 +226,6 @@ static void uni_player_set_channel_status(struct uniperif *player, * sampling frequency. If no sample rate is already specified, then * set one. */ - mutex_lock(&player->ctrl_lock); if (runtime) { switch (runtime->rate) { case 22050: @@ -303,7 +302,6 @@ static void uni_player_set_channel_status(struct uniperif *player, player->stream_settings.iec958.status[3 + (n * 4)] << 24; SET_UNIPERIF_CHANNEL_STA_REGN(player, n, status); } - mutex_unlock(&player->ctrl_lock); /* Update the channel status */ if (player->ver < SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0) @@ -365,8 +363,10 @@ static int uni_player_prepare_iec958(struct uniperif *player, SET_UNIPERIF_CTRL_ZERO_STUFF_HW(player); + mutex_lock(&player->ctrl_lock); /* Update the channel status */ uni_player_set_channel_status(player, runtime); + mutex_unlock(&player->ctrl_lock); /* Clear the user validity user bits */ SET_UNIPERIF_USER_VALIDITY_VALIDITY_LR(player, 0); @@ -598,7 +598,6 @@ static int uni_player_ctl_iec958_put(struct snd_kcontrol *kcontrol, iec958->status[1] = ucontrol->value.iec958.status[1]; iec958->status[2] = ucontrol->value.iec958.status[2]; iec958->status[3] = ucontrol->value.iec958.status[3]; - mutex_unlock(&player->ctrl_lock); spin_lock_irqsave(&player->irq_lock, flags); if (player->substream && player->substream->runtime) @@ -608,6 +607,8 @@ static int uni_player_ctl_iec958_put(struct snd_kcontrol *kcontrol, uni_player_set_channel_status(player, NULL); spin_unlock_irqrestore(&player->irq_lock, flags); + mutex_unlock(&player->ctrl_lock); + return 0; } diff --git a/sound/soc/stm/stm32_i2s.c b/sound/soc/stm/stm32_i2s.c index 6d0bf78d114d..aa2b1196171a 100644 --- a/sound/soc/stm/stm32_i2s.c +++ b/sound/soc/stm/stm32_i2s.c @@ -246,8 +246,8 @@ static irqreturn_t stm32_i2s_isr(int irq, void *devid) return IRQ_NONE; } - regmap_update_bits(i2s->regmap, STM32_I2S_IFCR_REG, - I2S_IFCR_MASK, flags); + regmap_write_bits(i2s->regmap, STM32_I2S_IFCR_REG, + I2S_IFCR_MASK, flags); if (flags & I2S_SR_OVR) { dev_dbg(&pdev->dev, "Overrun\n"); @@ -276,7 +276,6 @@ static bool stm32_i2s_readable_reg(struct device *dev, unsigned int reg) case STM32_I2S_CFG2_REG: case STM32_I2S_IER_REG: case STM32_I2S_SR_REG: - case STM32_I2S_IFCR_REG: case STM32_I2S_TXDR_REG: case STM32_I2S_RXDR_REG: case STM32_I2S_CGFR_REG: @@ -488,7 +487,7 @@ static int stm32_i2s_configure(struct snd_soc_dai *cpu_dai, { struct stm32_i2s_data *i2s = snd_soc_dai_get_drvdata(cpu_dai); int format = params_width(params); - u32 cfgr, cfgr_mask, cfg1, cfg1_mask; + u32 cfgr, cfgr_mask, cfg1; unsigned int fthlv; int ret; @@ -501,7 +500,7 @@ static int stm32_i2s_configure(struct snd_soc_dai *cpu_dai, switch (format) { case 16: cfgr = I2S_CGFR_DATLEN_SET(I2S_I2SMOD_DATLEN_16); - cfgr_mask = I2S_CGFR_DATLEN_MASK; + cfgr_mask = I2S_CGFR_DATLEN_MASK | I2S_CGFR_CHLEN; break; case 32: cfgr = I2S_CGFR_DATLEN_SET(I2S_I2SMOD_DATLEN_32) | @@ -529,15 +528,11 @@ static int stm32_i2s_configure(struct snd_soc_dai *cpu_dai, if (ret < 0) return ret; - cfg1 = I2S_CFG1_RXDMAEN | I2S_CFG1_TXDMAEN; - cfg1_mask = cfg1; - fthlv = STM32_I2S_FIFO_SIZE * I2S_FIFO_TH_ONE_QUARTER / 4; - cfg1 |= I2S_CFG1_FTHVL_SET(fthlv - 1); - cfg1_mask |= I2S_CFG1_FTHVL_MASK; + cfg1 = I2S_CFG1_FTHVL_SET(fthlv - 1); return regmap_update_bits(i2s->regmap, STM32_I2S_CFG1_REG, - cfg1_mask, cfg1); + I2S_CFG1_FTHVL_MASK, cfg1); } static int stm32_i2s_startup(struct snd_pcm_substream *substream, @@ -551,8 +546,8 @@ static int stm32_i2s_startup(struct snd_pcm_substream *substream, i2s->refcount++; spin_unlock(&i2s->lock_fd); - return regmap_update_bits(i2s->regmap, STM32_I2S_IFCR_REG, - I2S_IFCR_MASK, I2S_IFCR_MASK); + return regmap_write_bits(i2s->regmap, STM32_I2S_IFCR_REG, + I2S_IFCR_MASK, I2S_IFCR_MASK); } static int stm32_i2s_hw_params(struct snd_pcm_substream *substream, @@ -589,6 +584,10 @@ static int stm32_i2s_trigger(struct snd_pcm_substream *substream, int cmd, /* Enable i2s */ dev_dbg(cpu_dai->dev, "start I2S\n"); + cfg1_mask = I2S_CFG1_RXDMAEN | I2S_CFG1_TXDMAEN; + regmap_update_bits(i2s->regmap, STM32_I2S_CFG1_REG, + cfg1_mask, cfg1_mask); + ret = regmap_update_bits(i2s->regmap, STM32_I2S_CR1_REG, I2S_CR1_SPE, I2S_CR1_SPE); if (ret < 0) { @@ -603,8 +602,8 @@ static int stm32_i2s_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } - regmap_update_bits(i2s->regmap, STM32_I2S_IFCR_REG, - I2S_IFCR_MASK, I2S_IFCR_MASK); + regmap_write_bits(i2s->regmap, STM32_I2S_IFCR_REG, + I2S_IFCR_MASK, I2S_IFCR_MASK); if (playback_flg) { ier = I2S_IER_UDRIE; diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 90d439613899..48b4286100d4 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -873,7 +873,6 @@ static int stm32_sai_sub_dais_init(struct platform_device *pdev, if (!sai->cpu_dai_drv) return -ENOMEM; - sai->cpu_dai_drv->name = dev_name(&pdev->dev); if (STM_SAI_IS_PLAYBACK(sai)) { memcpy(sai->cpu_dai_drv, &stm32_sai_playback_dai, sizeof(stm32_sai_playback_dai)); @@ -883,6 +882,7 @@ static int stm32_sai_sub_dais_init(struct platform_device *pdev, sizeof(stm32_sai_capture_dai)); sai->cpu_dai_drv->capture.stream_name = sai->cpu_dai_drv->name; } + sai->cpu_dai_drv->name = dev_name(&pdev->dev); return 0; } diff --git a/sound/soc/stm/stm32_spdifrx.c b/sound/soc/stm/stm32_spdifrx.c index 84cc5678beba..7bc57651e186 100644 --- a/sound/soc/stm/stm32_spdifrx.c +++ b/sound/soc/stm/stm32_spdifrx.c @@ -213,6 +213,7 @@ * @slave_config: dma slave channel runtime config pointer * @phys_addr: SPDIFRX registers physical base address * @lock: synchronization enabling lock + * @irq_lock: prevent race condition with IRQ on stream state * @cs: channel status buffer * @ub: user data buffer * @irq: SPDIFRX interrupt line @@ -233,6 +234,7 @@ struct stm32_spdifrx_data { struct dma_slave_config slave_config; dma_addr_t phys_addr; spinlock_t lock; /* Sync enabling lock */ + spinlock_t irq_lock; /* Prevent race condition on stream state */ unsigned char cs[SPDIFRX_CS_BYTES_NB]; unsigned char ub[SPDIFRX_UB_BYTES_NB]; int irq; @@ -313,6 +315,7 @@ static void stm32_spdifrx_dma_ctrl_stop(struct stm32_spdifrx_data *spdifrx) static int stm32_spdifrx_start_sync(struct stm32_spdifrx_data *spdifrx) { int cr, cr_mask, imr, ret; + unsigned long flags; /* Enable IRQs */ imr = SPDIFRX_IMR_IFEIE | SPDIFRX_IMR_SYNCDIE | SPDIFRX_IMR_PERRIE; @@ -320,7 +323,7 @@ static int stm32_spdifrx_start_sync(struct stm32_spdifrx_data *spdifrx) if (ret) return ret; - spin_lock(&spdifrx->lock); + spin_lock_irqsave(&spdifrx->lock, flags); spdifrx->refcount++; @@ -353,7 +356,7 @@ static int stm32_spdifrx_start_sync(struct stm32_spdifrx_data *spdifrx) "Failed to start synchronization\n"); } - spin_unlock(&spdifrx->lock); + spin_unlock_irqrestore(&spdifrx->lock, flags); return ret; } @@ -361,11 +364,12 @@ static int stm32_spdifrx_start_sync(struct stm32_spdifrx_data *spdifrx) static void stm32_spdifrx_stop(struct stm32_spdifrx_data *spdifrx) { int cr, cr_mask, reg; + unsigned long flags; - spin_lock(&spdifrx->lock); + spin_lock_irqsave(&spdifrx->lock, flags); if (--spdifrx->refcount) { - spin_unlock(&spdifrx->lock); + spin_unlock_irqrestore(&spdifrx->lock, flags); return; } @@ -384,7 +388,7 @@ static void stm32_spdifrx_stop(struct stm32_spdifrx_data *spdifrx) regmap_read(spdifrx->regmap, STM32_SPDIFRX_DR, ®); regmap_read(spdifrx->regmap, STM32_SPDIFRX_CSR, ®); - spin_unlock(&spdifrx->lock); + spin_unlock_irqrestore(&spdifrx->lock, flags); } static int stm32_spdifrx_dma_ctrl_register(struct device *dev, @@ -644,7 +648,6 @@ static const struct regmap_config stm32_h7_spdifrx_regmap_conf = { static irqreturn_t stm32_spdifrx_isr(int irq, void *devid) { struct stm32_spdifrx_data *spdifrx = (struct stm32_spdifrx_data *)devid; - struct snd_pcm_substream *substream = spdifrx->substream; struct platform_device *pdev = spdifrx->pdev; unsigned int cr, mask, sr, imr; unsigned int flags; @@ -712,14 +715,19 @@ static irqreturn_t stm32_spdifrx_isr(int irq, void *devid) regmap_update_bits(spdifrx->regmap, STM32_SPDIFRX_CR, SPDIFRX_CR_SPDIFEN_MASK, cr); - if (substream) - snd_pcm_stop(substream, SNDRV_PCM_STATE_DISCONNECTED); + spin_lock(&spdifrx->irq_lock); + if (spdifrx->substream) + snd_pcm_stop(spdifrx->substream, + SNDRV_PCM_STATE_DISCONNECTED); + spin_unlock(&spdifrx->irq_lock); return IRQ_HANDLED; } - if (err_xrun && substream) - snd_pcm_stop_xrun(substream); + spin_lock(&spdifrx->irq_lock); + if (err_xrun && spdifrx->substream) + snd_pcm_stop_xrun(spdifrx->substream); + spin_unlock(&spdifrx->irq_lock); return IRQ_HANDLED; } @@ -728,9 +736,12 @@ static int stm32_spdifrx_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct stm32_spdifrx_data *spdifrx = snd_soc_dai_get_drvdata(cpu_dai); + unsigned long flags; int ret; + spin_lock_irqsave(&spdifrx->irq_lock, flags); spdifrx->substream = substream; + spin_unlock_irqrestore(&spdifrx->irq_lock, flags); ret = clk_prepare_enable(spdifrx->kclk); if (ret) @@ -802,8 +813,12 @@ static void stm32_spdifrx_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct stm32_spdifrx_data *spdifrx = snd_soc_dai_get_drvdata(cpu_dai); + unsigned long flags; + spin_lock_irqsave(&spdifrx->irq_lock, flags); spdifrx->substream = NULL; + spin_unlock_irqrestore(&spdifrx->irq_lock, flags); + clk_disable_unprepare(spdifrx->kclk); } @@ -908,6 +923,7 @@ static int stm32_spdifrx_probe(struct platform_device *pdev) spdifrx->pdev = pdev; init_completion(&spdifrx->cs_completion); spin_lock_init(&spdifrx->lock); + spin_lock_init(&spdifrx->irq_lock); platform_set_drvdata(pdev, spdifrx); diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index b4af5ce78ecb..d2802fd8c1dd 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -80,8 +80,8 @@ #define SUN4I_I2S_CLK_DIV_MCLK_MASK GENMASK(3, 0) #define SUN4I_I2S_CLK_DIV_MCLK(mclk) ((mclk) << 0) -#define SUN4I_I2S_RX_CNT_REG 0x28 -#define SUN4I_I2S_TX_CNT_REG 0x2c +#define SUN4I_I2S_TX_CNT_REG 0x28 +#define SUN4I_I2S_RX_CNT_REG 0x2c #define SUN4I_I2S_TX_CHAN_SEL_REG 0x30 #define SUN4I_I2S_CHAN_SEL(num_chan) (((num_chan) - 1) << 0) @@ -110,7 +110,7 @@ #define SUN8I_I2S_TX_CHAN_MAP_REG 0x44 #define SUN8I_I2S_TX_CHAN_SEL_REG 0x34 -#define SUN8I_I2S_TX_CHAN_OFFSET_MASK GENMASK(13, 11) +#define SUN8I_I2S_TX_CHAN_OFFSET_MASK GENMASK(13, 12) #define SUN8I_I2S_TX_CHAN_OFFSET(offset) (offset << 12) #define SUN8I_I2S_TX_CHAN_EN_MASK GENMASK(11, 4) #define SUN8I_I2S_TX_CHAN_EN(num_chan) (((1 << num_chan) - 1) << 4) @@ -442,6 +442,10 @@ static int sun4i_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) regmap_update_bits(i2s->regmap, SUN8I_I2S_TX_CHAN_SEL_REG, SUN8I_I2S_TX_CHAN_OFFSET_MASK, SUN8I_I2S_TX_CHAN_OFFSET(offset)); + + regmap_update_bits(i2s->regmap, SUN8I_I2S_RX_CHAN_SEL_REG, + SUN8I_I2S_TX_CHAN_OFFSET_MASK, + SUN8I_I2S_TX_CHAN_OFFSET(offset)); } regmap_field_write(i2s->field_fmt_mode, val); diff --git a/sound/soc/tegra/tegra_sgtl5000.c b/sound/soc/tegra/tegra_sgtl5000.c index 45a4aa9d2a47..901457da25ec 100644 --- a/sound/soc/tegra/tegra_sgtl5000.c +++ b/sound/soc/tegra/tegra_sgtl5000.c @@ -149,14 +149,14 @@ static int tegra_sgtl5000_driver_probe(struct platform_device *pdev) dev_err(&pdev->dev, "Property 'nvidia,i2s-controller' missing/invalid\n"); ret = -EINVAL; - goto err; + goto err_put_codec_of_node; } tegra_sgtl5000_dai.platform_of_node = tegra_sgtl5000_dai.cpu_of_node; ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev); if (ret) - goto err; + goto err_put_cpu_of_node; ret = snd_soc_register_card(card); if (ret) { @@ -169,6 +169,13 @@ static int tegra_sgtl5000_driver_probe(struct platform_device *pdev) err_fini_utils: tegra_asoc_utils_fini(&machine->util_data); +err_put_cpu_of_node: + of_node_put(tegra_sgtl5000_dai.cpu_of_node); + tegra_sgtl5000_dai.cpu_of_node = NULL; + tegra_sgtl5000_dai.platform_of_node = NULL; +err_put_codec_of_node: + of_node_put(tegra_sgtl5000_dai.codec_of_node); + tegra_sgtl5000_dai.codec_of_node = NULL; err: return ret; } @@ -183,6 +190,12 @@ static int tegra_sgtl5000_driver_remove(struct platform_device *pdev) tegra_asoc_utils_fini(&machine->util_data); + of_node_put(tegra_sgtl5000_dai.cpu_of_node); + tegra_sgtl5000_dai.cpu_of_node = NULL; + tegra_sgtl5000_dai.platform_of_node = NULL; + of_node_put(tegra_sgtl5000_dai.codec_of_node); + tegra_sgtl5000_dai.codec_of_node = NULL; + return ret; } |