diff options
Diffstat (limited to 'sound/soc')
45 files changed, 525 insertions, 175 deletions
diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index 22aec9a1e9a4..838d03a138ca 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -25,6 +25,8 @@ config SND_ATMEL_SOC_DMA config SND_ATMEL_SOC_SSC_DMA tristate + select SND_ATMEL_SOC_DMA + select SND_ATMEL_SOC_PDC config SND_ATMEL_SOC_SSC 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 43cdd06309d7..7cbbe64e3509 100644 --- a/sound/soc/codecs/cs42xx8.c +++ b/sound/soc/codecs/cs42xx8.c @@ -654,6 +654,7 @@ 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 231ca935cdf3..c232c42ccead 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/es8328.c b/sound/soc/codecs/es8328.c index 37722194b107..6b22700842e2 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -234,7 +234,7 @@ 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 c602c4960924..88355d1719a3 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1267,6 +1267,12 @@ 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 511d09cb2181..ca86eefdb9eb 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -140,8 +140,12 @@ 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; + } } } return 0; diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 584aab83e478..e7aef841f87d 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/nau8810.c b/sound/soc/codecs/nau8810.c index e45518629968..2234d0c04165 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/pcm512x.c b/sound/soc/codecs/pcm512x.c index 72b19e62f626..c0807b82399a 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -1441,13 +1441,15 @@ int pcm512x_probe(struct device *dev, struct regmap *regmap) } pcm512x->sclk = devm_clk_get(dev, NULL); - if (PTR_ERR(pcm512x->sclk) == -EPROBE_DEFER) - return -EPROBE_DEFER; + if (PTR_ERR(pcm512x->sclk) == -EPROBE_DEFER) { + ret = -EPROBE_DEFER; + goto err; + } if (!IS_ERR(pcm512x->sclk)) { ret = clk_prepare_enable(pcm512x->sclk); if (ret != 0) { dev_err(dev, "Failed to enable SCLK: %d\n", ret); - return ret; + goto err; } } diff --git a/sound/soc/codecs/rt5677-spi.c b/sound/soc/codecs/rt5677-spi.c index 91879ea95415..01aa75cde571 100644 --- a/sound/soc/codecs/rt5677-spi.c +++ b/sound/soc/codecs/rt5677-spi.c @@ -60,13 +60,15 @@ 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 @@ -81,15 +83,13 @@ 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; } @@ -110,7 +110,7 @@ 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; @@ -126,7 +126,7 @@ 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; } @@ -161,13 +161,13 @@ 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; @@ -180,22 +180,19 @@ 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 65ac4518ad06..49ab26e69f2f 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -305,6 +305,7 @@ 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 3dba5550a665..39810b713d5f 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 }, @@ -99,6 +106,13 @@ enum sgtl5000_micbias_resistor { SGTL5000_MICBIAS_8K = 8, }; +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 */ @@ -111,8 +125,117 @@ struct sgtl5000_priv { int revision; u8 micbias_resistor; u8 micbias_voltage; + 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 @@ -144,36 +267,46 @@ 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; @@ -182,6 +315,41 @@ 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" @@ -217,7 +385,10 @@ 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), @@ -233,11 +404,12 @@ 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 */ @@ -987,12 +1159,17 @@ 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); @@ -1040,7 +1217,7 @@ static int sgtl5000_set_power_regs(struct snd_soc_codec *codec) * Searching for a suitable index solving this formula: * idx = 40 * log10(vag_val / lo_cagcntrl) + 15 */ - vol_quot = (vag * 100) / lo_vag; + vol_quot = lo_vag ? (vag * 100) / lo_vag : 0; lo_vol = 0; for (i = 0; i < ARRAY_SIZE(vol_quot_table); i++) { if (vol_quot >= vol_quot_table[i]) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 28fdfc5ec544..c27e3476848a 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -316,6 +316,8 @@ 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/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index c03c9da076c2..28eb55bc4663 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -948,8 +948,7 @@ 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) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 3c5a9804d3f5..d1935c5c3602 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, diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 48f795dfc3ed..0fcf4065d0e4 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -223,16 +223,17 @@ 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 diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index 883087f2b092..38132143b7d5 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 dffd549a0e2a..705d2524ec31 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 969c5c20605a..bc3fe4a3fc84 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -350,8 +350,8 @@ 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_esai.c b/sound/soc/fsl/fsl_esai.c index 0742aff1ab31..efe4904bb6f2 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -439,8 +439,8 @@ 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; } @@ -673,6 +673,18 @@ 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)); @@ -1106,6 +1118,9 @@ 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 d6c211921cee..8f45305a1e9a 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_ssi.c b/sound/soc/fsl/fsl_ssi.c index c184590d5d8e..342c360f2a65 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1447,6 +1447,7 @@ 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; @@ -1638,7 +1639,9 @@ 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-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index b99e0b5e00e9..3d99a8579c99 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -115,10 +115,12 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) ret = -EPROBE_DEFER; goto fail; } + put_device(&ssi_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); diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c index 0838478c4c3f..b3464ac99b99 100644 --- a/sound/soc/intel/atom/sst-atom-controls.c +++ b/sound/soc/intel/atom/sst-atom-controls.c @@ -1343,7 +1343,7 @@ int sst_send_pipe_gains(struct snd_soc_dai *dai, int stream, int mute) dai->capture_widget->name); w = dai->capture_widget; snd_soc_dapm_widget_for_each_source_path(w, p) { - if (p->connected && !p->connected(w, p->sink)) + if (p->connected && !p->connected(w, p->source)) continue; if (p->connect && p->source->power && diff --git a/sound/soc/intel/atom/sst/sst_pci.c b/sound/soc/intel/atom/sst/sst_pci.c index 3a0b3bf0af97..e9c6894cc27f 100644 --- a/sound/soc/intel/atom/sst/sst_pci.c +++ b/sound/soc/intel/atom/sst/sst_pci.c @@ -107,7 +107,7 @@ static int sst_platform_get_resources(struct intel_sst_drv *ctx) dev_dbg(ctx->dev, "DRAM Ptr %p\n", ctx->dram); do_release_regions: pci_release_regions(pci); - return 0; + return ret; } /* 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 6c672ac79cce..9d24cb719f69 100644 --- a/sound/soc/intel/common/sst-ipc.c +++ b/sound/soc/intel/common/sst-ipc.c @@ -211,6 +211,8 @@ 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-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c index dcf03691ebc8..d8cf37a0f696 100644 --- a/sound/soc/intel/skylake/skl-nhlt.c +++ b/sound/soc/intel/skylake/skl-nhlt.c @@ -211,7 +211,7 @@ 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/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index 794a3499e567..0dc1ab48fceb 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -92,7 +92,7 @@ #define JZ_AIC_I2S_STATUS_BUSY BIT(2) #define JZ_AIC_CLK_DIV_MASK 0xf -#define I2SDIV_DV_SHIFT 8 +#define I2SDIV_DV_SHIFT 0 #define I2SDIV_DV_MASK (0xf << I2SDIV_DV_SHIFT) #define I2SDIV_IDV_SHIFT 8 #define I2SDIV_IDV_MASK (0xf << I2SDIV_IDV_SHIFT) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 3a36d60e1785..0a5d9fb6fc84 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -570,10 +570,6 @@ 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) @@ -589,6 +585,10 @@ 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 07f91e918b23..3d91cef3704a 100644 --- a/sound/soc/qcom/apq8016_sbc.c +++ b/sound/soc/qcom/apq8016_sbc.c @@ -114,46 +114,58 @@ 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; } link->codec_of_node = of_parse_phandle(codec, "sound-dai", 0); if (!link->codec_of_node) { dev_err(card->dev, "error getting codec 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_name(codec, &link->codec_dai_name); if (ret) { 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 08bfee447a36..94b6f9c7dd6b 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -649,7 +649,7 @@ 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/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 91b444db575e..5346b3cafc67 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -629,6 +629,7 @@ 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; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index ab647f1fe11b..73523cf0329b 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -751,7 +751,13 @@ static void dapm_set_mixer_path_status(struct snd_soc_dapm_path *p, int i) val = max - val; p->connect = !!val; } else { - p->connect = 0; + /* since a virtual mixer has no backing registers to + * decide which path to connect, it will try to match + * with initial state. This is to ensure + * that the default mixer choice will be + * correctly powered up during initialization. + */ + p->connect = invert; } } @@ -1104,8 +1110,8 @@ 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) { @@ -1142,8 +1148,8 @@ 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, @@ -4363,7 +4369,7 @@ static void soc_dapm_shutdown_dapm(struct snd_soc_dapm_context *dapm) continue; if (w->power) { dapm_seq_insert(w, &down_list, false); - w->power = 0; + w->new_power = 0; powerdown = 1; } } diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 6cef3977507a..67d22b4baeb0 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -312,6 +312,12 @@ 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 fbaa1bb41102..00d7902ad427 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -80,10 +80,9 @@ 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-ops.c b/sound/soc/soc-ops.c index 9fc1a7bb8b95..90acdf4d90ed 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -837,7 +837,7 @@ int snd_soc_get_xr_sx(struct snd_kcontrol *kcontrol, unsigned int regbase = mc->regbase; unsigned int regcount = mc->regcount; unsigned int regwshift = component->val_bytes * BITS_PER_BYTE; - unsigned int regwmask = (1<<regwshift)-1; + unsigned int regwmask = (1UL<<regwshift)-1; unsigned int invert = mc->invert; unsigned long mask = (1UL<<mc->nbits)-1; long min = mc->min; @@ -886,7 +886,7 @@ int snd_soc_put_xr_sx(struct snd_kcontrol *kcontrol, unsigned int regbase = mc->regbase; unsigned int regcount = mc->regcount; unsigned int regwshift = component->val_bytes * BITS_PER_BYTE; - unsigned int regwmask = (1<<regwshift)-1; + unsigned int regwmask = (1UL<<regwshift)-1; unsigned int invert = mc->invert; unsigned long mask = (1UL<<mc->nbits)-1; long max = mc->max; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index f35ea743171b..c1d41d048dba 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); @@ -1589,7 +1592,7 @@ 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) @@ -2102,7 +2105,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, switch (cmd) { case SNDRV_PCM_TRIGGER_START: if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && - (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); @@ -2132,7 +2136,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; case SNDRV_PCM_TRIGGER_STOP: - if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) @@ -2177,42 +2182,81 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, } EXPORT_SYMBOL_GPL(dpcm_be_dai_trigger); +static int dpcm_dai_trigger_fe_be(struct snd_pcm_substream *substream, + int cmd, bool fe_first) +{ + struct snd_soc_pcm_runtime *fe = substream->private_data; + int ret; + + /* call trigger on the frontend before the backend. */ + if (fe_first) { + dev_dbg(fe->dev, "ASoC: pre trigger FE %s cmd %d\n", + fe->dai_link->name, cmd); + + ret = soc_pcm_trigger(substream, cmd); + if (ret < 0) + return ret; + + ret = dpcm_be_dai_trigger(fe, substream->stream, cmd); + return ret; + } + + /* call trigger on the frontend after the backend. */ + ret = dpcm_be_dai_trigger(fe, substream->stream, cmd); + if (ret < 0) + return ret; + + dev_dbg(fe->dev, "ASoC: post trigger FE %s cmd %d\n", + fe->dai_link->name, cmd); + + ret = soc_pcm_trigger(substream, cmd); + + return ret; +} + static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *fe = substream->private_data; - int stream = substream->stream, ret; + int stream = substream->stream; + int ret = 0; enum snd_soc_dpcm_trigger trigger = fe->dai_link->trigger[stream]; fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; switch (trigger) { case SND_SOC_DPCM_TRIGGER_PRE: - /* call trigger on the frontend before the backend. */ - - dev_dbg(fe->dev, "ASoC: pre trigger FE %s cmd %d\n", - fe->dai_link->name, cmd); - - ret = soc_pcm_trigger(substream, cmd); - if (ret < 0) { - dev_err(fe->dev,"ASoC: trigger FE failed %d\n", ret); - goto out; + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ret = dpcm_dai_trigger_fe_be(substream, cmd, true); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ret = dpcm_dai_trigger_fe_be(substream, cmd, false); + break; + default: + ret = -EINVAL; + break; } - - ret = dpcm_be_dai_trigger(fe, substream->stream, cmd); break; case SND_SOC_DPCM_TRIGGER_POST: - /* call trigger on the frontend after the backend. */ - - ret = dpcm_be_dai_trigger(fe, substream->stream, cmd); - if (ret < 0) { - dev_err(fe->dev,"ASoC: trigger FE failed %d\n", ret); - goto out; + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ret = dpcm_dai_trigger_fe_be(substream, cmd, false); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ret = dpcm_dai_trigger_fe_be(substream, cmd, true); + break; + default: + ret = -EINVAL; + break; } - - dev_dbg(fe->dev, "ASoC: post trigger FE %s cmd %d\n", - fe->dai_link->name, cmd); - - ret = soc_pcm_trigger(substream, cmd); break; case SND_SOC_DPCM_TRIGGER_BESPOKE: /* bespoke trigger() - handles both FE and BEs */ @@ -2221,10 +2265,6 @@ static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd) fe->dai_link->name, cmd); ret = soc_pcm_bespoke_trigger(substream, cmd); - if (ret < 0) { - dev_err(fe->dev,"ASoC: trigger FE failed %d\n", ret); - goto out; - } break; default: dev_err(fe->dev, "ASoC: invalid trigger cmd %d for %s\n", cmd, @@ -2233,6 +2273,12 @@ static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd) goto out; } + if (ret < 0) { + dev_err(fe->dev, "ASoC: trigger FE cmd: %d failed: %d\n", + cmd, ret); + goto out; + } + switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: @@ -2287,7 +2333,8 @@ 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", @@ -2977,16 +3024,16 @@ static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe, ssize_t offset = 0; /* FE state */ - offset += snprintf(buf + offset, size - offset, + offset += scnprintf(buf + offset, size - offset, "[%s - %s]\n", fe->dai_link->name, stream ? "Capture" : "Playback"); - offset += snprintf(buf + offset, size - offset, "State: %s\n", + offset += scnprintf(buf + offset, size - offset, "State: %s\n", dpcm_state_string(fe->dpcm[stream].state)); if ((fe->dpcm[stream].state >= SND_SOC_DPCM_STATE_HW_PARAMS) && (fe->dpcm[stream].state <= SND_SOC_DPCM_STATE_STOP)) - offset += snprintf(buf + offset, size - offset, + offset += scnprintf(buf + offset, size - offset, "Hardware Params: " "Format = %s, Channels = %d, Rate = %d\n", snd_pcm_format_name(params_format(params)), @@ -2994,10 +3041,10 @@ static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe, params_rate(params)); /* BEs state */ - offset += snprintf(buf + offset, size - offset, "Backends:\n"); + offset += scnprintf(buf + offset, size - offset, "Backends:\n"); if (list_empty(&fe->dpcm[stream].be_clients)) { - offset += snprintf(buf + offset, size - offset, + offset += scnprintf(buf + offset, size - offset, " No active DSP links\n"); goto out; } @@ -3006,16 +3053,16 @@ static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be = dpcm->be; params = &dpcm->hw_params; - offset += snprintf(buf + offset, size - offset, + offset += scnprintf(buf + offset, size - offset, "- %s\n", be->dai_link->name); - offset += snprintf(buf + offset, size - offset, + offset += scnprintf(buf + offset, size - offset, " State: %s\n", dpcm_state_string(be->dpcm[stream].state)); if ((be->dpcm[stream].state >= SND_SOC_DPCM_STATE_HW_PARAMS) && (be->dpcm[stream].state <= SND_SOC_DPCM_STATE_STOP)) - offset += snprintf(buf + offset, size - offset, + offset += scnprintf(buf + offset, size - offset, " Hardware Params: " "Format = %s, Channels = %d, Rate = %d\n", snd_pcm_format_name(params_format(params)), diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 086fe4d27f60..e9c57bd3c02b 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -344,7 +344,7 @@ static int soc_tplg_add_kcontrol(struct soc_tplg *tplg, struct snd_soc_component *comp = tplg->comp; return soc_tplg_add_dcontrol(comp->card->snd_card, - comp->dev, k, NULL, comp, kcontrol); + comp->dev, k, comp->name_prefix, comp, kcontrol); } /* remove a mixer kcontrol */ diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index 15c92400cea4..02c373c65e19 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -78,8 +78,8 @@ #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_TX_CHAN_SEL(num_chan) (((num_chan) - 1) << 0) diff --git a/sound/soc/tegra/tegra_sgtl5000.c b/sound/soc/tegra/tegra_sgtl5000.c index 1e76869dd488..863e04809a6b 100644 --- a/sound/soc/tegra/tegra_sgtl5000.c +++ b/sound/soc/tegra/tegra_sgtl5000.c @@ -152,14 +152,14 @@ 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) { @@ -172,6 +172,13 @@ 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; } @@ -186,6 +193,12 @@ 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; } |