diff options
Diffstat (limited to 'sound/soc')
30 files changed, 108 insertions, 68 deletions
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 5d230cee3fa7..7fbfa051f6e1 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -672,7 +672,7 @@ static int atmel_ssc_resume(struct snd_soc_dai *cpu_dai) /* re-enable interrupts */ ssc_writel(ssc_p->ssc->regs, IER, ssc_p->ssc_state.ssc_imr); - /* Re-enable recieve and transmit as appropriate */ + /* Re-enable receive and transmit as appropriate */ cr = 0; cr |= (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_RXEN)) ? SSC_BIT(CR_RXEN) : 0; diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index 4f377c9e868d..eecffb548947 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -481,7 +481,7 @@ struct _pll_div { }; /* Note : pll code from original alc5623 driver. Not sure of how good it is */ -/* usefull only for master mode */ +/* useful only for master mode */ static const struct _pll_div codec_master_pll_div[] = { { 2048000, 8192000, 0x0ea0}, diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c index f7cd346fd727..f5ccdbf7ebc6 100644 --- a/sound/soc/codecs/jz4740.c +++ b/sound/soc/codecs/jz4740.c @@ -308,8 +308,6 @@ static int jz4740_codec_dev_probe(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(dapm, jz4740_codec_dapm_routes, ARRAY_SIZE(jz4740_codec_dapm_routes)); - snd_soc_dapm_new_widgets(codec); - jz4740_codec_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c index 72de47e5d040..2c2a681da0d7 100644 --- a/sound/soc/codecs/lm4857.c +++ b/sound/soc/codecs/lm4857.c @@ -161,7 +161,7 @@ static const struct snd_kcontrol_new lm4857_controls[] = { lm4857_get_mode, lm4857_set_mode), }; -/* There is a demux inbetween the the input signal and the output signals. +/* There is a demux between the input signal and the output signals. * Currently there is no easy way to model it in ASoC and since it does not make * much of a difference in practice simply connect the input direclty to the * outputs. */ diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 2a30eae1881c..4d9fb279e146 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -26,7 +26,9 @@ #define pr_fmt(fmt) KBUILD_MODNAME ": " fmt #include <linux/platform_device.h> +#include <linux/delay.h> #include <linux/slab.h> + #include <asm/intel_scu_ipc.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -925,7 +927,7 @@ static struct platform_driver sn95031_codec_driver = { .owner = THIS_MODULE, }, .probe = sn95031_device_probe, - .remove = sn95031_device_remove, + .remove = __devexit_p(sn95031_device_remove), }; static int __init sn95031_init(void) diff --git a/sound/soc/codecs/tlv320aic26.h b/sound/soc/codecs/tlv320aic26.h index 62b1f2261429..67f19c3bebe6 100644 --- a/sound/soc/codecs/tlv320aic26.h +++ b/sound/soc/codecs/tlv320aic26.h @@ -14,14 +14,14 @@ #define AIC26_PAGE_ADDR(page, offset) ((page << 6) | offset) #define AIC26_NUM_REGS AIC26_PAGE_ADDR(3, 0) -/* Page 0: Auxillary data registers */ +/* Page 0: Auxiliary data registers */ #define AIC26_REG_BAT1 AIC26_PAGE_ADDR(0, 0x05) #define AIC26_REG_BAT2 AIC26_PAGE_ADDR(0, 0x06) #define AIC26_REG_AUX AIC26_PAGE_ADDR(0, 0x07) #define AIC26_REG_TEMP1 AIC26_PAGE_ADDR(0, 0x09) #define AIC26_REG_TEMP2 AIC26_PAGE_ADDR(0, 0x0A) -/* Page 1: Auxillary control registers */ +/* Page 1: Auxiliary control registers */ #define AIC26_REG_AUX_ADC AIC26_PAGE_ADDR(1, 0x00) #define AIC26_REG_STATUS AIC26_PAGE_ADDR(1, 0x01) #define AIC26_REG_REFERENCE AIC26_PAGE_ADDR(1, 0x03) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 3bedab26892f..6c43c13f0430 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -884,7 +884,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, if (bypass_pll) return 0; - /* Use PLL, compute apropriate setup for j, d, r and p, the closest + /* Use PLL, compute appropriate setup for j, d, r and p, the closest * one wins the game. Try with d==0 first, next with d!=0. * Constraints for j are according to the datasheet. * The sysclk is divided by 1000 to prevent integer overflows. diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index eb1a0b4e09b6..082e9d51963f 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -1027,7 +1027,7 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream) /* * For FIFO bypass mode: * Enable the FIFO bypass (Disable the FIFO use) - * Set the BCLK as continous + * Set the BCLK as continuous */ fifoctrl_a |= DAC33_FBYPAS; aictrl_b |= DAC33_BCLKON; diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 8512800f6326..575238d68e5e 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -281,7 +281,7 @@ static inline void twl4030_check_defaults(struct snd_soc_codec *codec) i, val, twl4030_reg[i]); } } - dev_dbg(codec->dev, "Found %d non maching registers. %s\n", + dev_dbg(codec->dev, "Found %d non-matching registers. %s\n", difference, difference ? "Not OK" : "OK"); } @@ -2018,7 +2018,7 @@ static int twl4030_voice_startup(struct snd_pcm_substream *substream, u8 mode; /* If the system master clock is not 26MHz, the voice PCM interface is - * not avilable. + * not available. */ if (twl4030->sysclk != 26000) { dev_err(codec->dev, "The board is configured for %u Hz, while" @@ -2028,7 +2028,7 @@ static int twl4030_voice_startup(struct snd_pcm_substream *substream, } /* If the codec mode is not option2, the voice PCM interface is not - * avilable. + * available. */ mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE) & TWL4030_OPT_MODE; diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 8f6b5ee6645b..4bbc0a79f01e 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -772,7 +772,7 @@ static int wm8580_set_bias_level(struct snd_soc_codec *codec, reg &= ~(WM8580_PWRDN1_PWDN | WM8580_PWRDN1_ALLDACPD); snd_soc_write(codec, WM8580_PWRDN1, reg); - /* Make VMID high impedence */ + /* Make VMID high impedance */ reg = snd_soc_read(codec, WM8580_ADC_CONTROL1); reg &= ~0x100; snd_soc_write(codec, WM8580_ADC_CONTROL1, reg); diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 3f09deea8d9d..ffa2ffe5ec11 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1312,7 +1312,7 @@ static int wm8753_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_FMTBIT_S24_LE) /* - * The WM8753 supports upto 4 different and mutually exclusive DAI + * The WM8753 supports up to 4 different and mutually exclusive DAI * configurations. This gives 2 PCM's available for use, hifi and voice. * NOTE: The Voice PCM cannot play or capture audio to the CPU as it's DAI * is connected between the wm8753 and a BT codec or GSM modem. diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index ae1cadfae84c..f52b623bb692 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -247,8 +247,6 @@ static int wm8903_volatile_register(struct snd_soc_codec *codec, unsigned int re case WM8903_REVISION_NUMBER: case WM8903_INTERRUPT_STATUS_1: case WM8903_WRITE_SEQUENCER_4: - case WM8903_POWER_MANAGEMENT_3: - case WM8903_POWER_MANAGEMENT_2: case WM8903_DC_SERVO_READBACK_1: case WM8903_DC_SERVO_READBACK_2: case WM8903_DC_SERVO_READBACK_3: @@ -875,34 +873,40 @@ SND_SOC_DAPM_MIXER("Left Speaker Mixer", WM8903_POWER_MANAGEMENT_4, 1, 0, SND_SOC_DAPM_MIXER("Right Speaker Mixer", WM8903_POWER_MANAGEMENT_4, 0, 0, right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer)), -SND_SOC_DAPM_PGA_S("Left Headphone Output PGA", 0, WM8903_ANALOGUE_HP_0, - 4, 0, NULL, 0), -SND_SOC_DAPM_PGA_S("Right Headphone Output PGA", 0, WM8903_ANALOGUE_HP_0, +SND_SOC_DAPM_PGA_S("Left Headphone Output PGA", 0, WM8903_POWER_MANAGEMENT_2, + 1, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("Right Headphone Output PGA", 0, WM8903_POWER_MANAGEMENT_2, 0, 0, NULL, 0), -SND_SOC_DAPM_PGA_S("Left Line Output PGA", 0, WM8903_ANALOGUE_LINEOUT_0, 4, 0, +SND_SOC_DAPM_PGA_S("Left Line Output PGA", 0, WM8903_POWER_MANAGEMENT_3, 1, 0, NULL, 0), -SND_SOC_DAPM_PGA_S("Right Line Output PGA", 0, WM8903_ANALOGUE_LINEOUT_0, 0, 0, +SND_SOC_DAPM_PGA_S("Right Line Output PGA", 0, WM8903_POWER_MANAGEMENT_3, 0, 0, NULL, 0), SND_SOC_DAPM_PGA_S("HPL_RMV_SHORT", 4, WM8903_ANALOGUE_HP_0, 7, 0, NULL, 0), SND_SOC_DAPM_PGA_S("HPL_ENA_OUTP", 3, WM8903_ANALOGUE_HP_0, 6, 0, NULL, 0), -SND_SOC_DAPM_PGA_S("HPL_ENA_DLY", 1, WM8903_ANALOGUE_HP_0, 5, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPL_ENA_DLY", 2, WM8903_ANALOGUE_HP_0, 5, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPL_ENA", 1, WM8903_ANALOGUE_HP_0, 4, 0, NULL, 0), SND_SOC_DAPM_PGA_S("HPR_RMV_SHORT", 4, WM8903_ANALOGUE_HP_0, 3, 0, NULL, 0), SND_SOC_DAPM_PGA_S("HPR_ENA_OUTP", 3, WM8903_ANALOGUE_HP_0, 2, 0, NULL, 0), -SND_SOC_DAPM_PGA_S("HPR_ENA_DLY", 1, WM8903_ANALOGUE_HP_0, 1, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPR_ENA_DLY", 2, WM8903_ANALOGUE_HP_0, 1, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPR_ENA", 1, WM8903_ANALOGUE_HP_0, 0, 0, NULL, 0), SND_SOC_DAPM_PGA_S("LINEOUTL_RMV_SHORT", 4, WM8903_ANALOGUE_LINEOUT_0, 7, 0, NULL, 0), SND_SOC_DAPM_PGA_S("LINEOUTL_ENA_OUTP", 3, WM8903_ANALOGUE_LINEOUT_0, 6, 0, NULL, 0), -SND_SOC_DAPM_PGA_S("LINEOUTL_ENA_DLY", 1, WM8903_ANALOGUE_LINEOUT_0, 5, 0, +SND_SOC_DAPM_PGA_S("LINEOUTL_ENA_DLY", 2, WM8903_ANALOGUE_LINEOUT_0, 5, 0, + NULL, 0), +SND_SOC_DAPM_PGA_S("LINEOUTL_ENA", 1, WM8903_ANALOGUE_LINEOUT_0, 4, 0, NULL, 0), SND_SOC_DAPM_PGA_S("LINEOUTR_RMV_SHORT", 4, WM8903_ANALOGUE_LINEOUT_0, 3, 0, NULL, 0), SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_OUTP", 3, WM8903_ANALOGUE_LINEOUT_0, 2, 0, NULL, 0), -SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_DLY", 1, WM8903_ANALOGUE_LINEOUT_0, 1, 0, +SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_DLY", 2, WM8903_ANALOGUE_LINEOUT_0, 1, 0, + NULL, 0), +SND_SOC_DAPM_PGA_S("LINEOUTR_ENA", 1, WM8903_ANALOGUE_LINEOUT_0, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("DCS Master", WM8903_DC_SERVO_0, 4, 0, NULL, 0), @@ -1037,10 +1041,14 @@ static const struct snd_soc_dapm_route intercon[] = { { "Left Speaker PGA", NULL, "Left Speaker Mixer" }, { "Right Speaker PGA", NULL, "Right Speaker Mixer" }, - { "HPL_ENA_DLY", NULL, "Left Headphone Output PGA" }, - { "HPR_ENA_DLY", NULL, "Right Headphone Output PGA" }, - { "LINEOUTL_ENA_DLY", NULL, "Left Line Output PGA" }, - { "LINEOUTR_ENA_DLY", NULL, "Right Line Output PGA" }, + { "HPL_ENA", NULL, "Left Headphone Output PGA" }, + { "HPR_ENA", NULL, "Right Headphone Output PGA" }, + { "HPL_ENA_DLY", NULL, "HPL_ENA" }, + { "HPR_ENA_DLY", NULL, "HPR_ENA" }, + { "LINEOUTL_ENA", NULL, "Left Line Output PGA" }, + { "LINEOUTR_ENA", NULL, "Right Line Output PGA" }, + { "LINEOUTL_ENA_DLY", NULL, "LINEOUTL_ENA" }, + { "LINEOUTR_ENA_DLY", NULL, "LINEOUTR_ENA" }, { "HPL_DCS", NULL, "DCS Master" }, { "HPR_DCS", NULL, "DCS Master" }, diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 443ae580445c..9b3bba4df5b3 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1895,7 +1895,7 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref, pr_debug("Fvco=%dHz\n", target); - /* Find an appropraite FLL_FRATIO and factor it out of the target */ + /* Find an appropriate FLL_FRATIO and factor it out of the target */ for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) { if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) { fll_div->fll_fratio = fll_fratios[i].fll_fratio; diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index 5e0214d6293e..3c7198779c31 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -176,7 +176,7 @@ static int wm8995_pll_factors(struct device *dev, return 0; } -/* Lookup table specifiying SRATE (table 25 in datasheet); some of the +/* Lookup table specifying SRATE (table 25 in datasheet); some of the * output frequencies have been rounded to the standard frequencies * they are intended to match where the error is slight. */ static struct { diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 3b71dd65c966..500011eb8b2b 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3137,7 +3137,7 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref, pr_debug("FLL Fvco=%dHz\n", target); - /* Find an appropraite FLL_FRATIO and factor it out of the target */ + /* Find an appropriate FLL_FRATIO and factor it out of the target */ for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) { if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) { fll_div->fll_fratio = fll_fratios[i].fll_fratio; diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index 28fdfd66661d..3c2ee1bb73cd 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -981,7 +981,7 @@ static int wm8991_set_dai_pll(struct snd_soc_dai *codec_dai, reg = snd_soc_read(codec, WM8991_CLOCKING_2); snd_soc_write(codec, WM8991_CLOCKING_2, reg | WM8991_SYSCLK_SRC); - /* set up N , fractional mode and pre-divisor if neccessary */ + /* set up N , fractional mode and pre-divisor if necessary */ snd_soc_write(codec, WM8991_PLL1, pll_div.n | WM8991_SDM | (pll_div.div2 ? WM8991_PRESCALE : 0)); snd_soc_write(codec, WM8991_PLL2, (u8)(pll_div.k>>8)); diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 379fa22c5b6c..056aef904347 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -324,7 +324,7 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref, pr_debug("Fvco=%dHz\n", target); - /* Find an appropraite FLL_FRATIO and factor it out of the target */ + /* Find an appropriate FLL_FRATIO and factor it out of the target */ for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) { if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) { fll_div->fll_fratio = fll_fratios[i].fll_fratio; diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 3dc64c8b6a5c..84e1bd1d2822 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -82,18 +82,18 @@ struct wm8994_priv { int mbc_ena[3]; - /* Platform dependant DRC configuration */ + /* Platform dependent DRC configuration */ const char **drc_texts; int drc_cfg[WM8994_NUM_DRC]; struct soc_enum drc_enum; - /* Platform dependant ReTune mobile configuration */ + /* Platform dependent ReTune mobile configuration */ int num_retune_mobile_texts; const char **retune_mobile_texts; int retune_mobile_cfg[WM8994_NUM_EQ]; struct soc_enum retune_mobile_enum; - /* Platform dependant MBC configuration */ + /* Platform dependent MBC configuration */ int mbc_cfg; const char **mbc_texts; struct soc_enum mbc_enum; @@ -3261,20 +3261,36 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Latch volume updates (right only; we always do left then right). */ + snd_soc_update_bits(codec, WM8994_AIF1_DAC1_LEFT_VOLUME, + WM8994_AIF1DAC1_VU, WM8994_AIF1DAC1_VU); snd_soc_update_bits(codec, WM8994_AIF1_DAC1_RIGHT_VOLUME, WM8994_AIF1DAC1_VU, WM8994_AIF1DAC1_VU); + snd_soc_update_bits(codec, WM8994_AIF1_DAC2_LEFT_VOLUME, + WM8994_AIF1DAC2_VU, WM8994_AIF1DAC2_VU); snd_soc_update_bits(codec, WM8994_AIF1_DAC2_RIGHT_VOLUME, WM8994_AIF1DAC2_VU, WM8994_AIF1DAC2_VU); + snd_soc_update_bits(codec, WM8994_AIF2_DAC_LEFT_VOLUME, + WM8994_AIF2DAC_VU, WM8994_AIF2DAC_VU); snd_soc_update_bits(codec, WM8994_AIF2_DAC_RIGHT_VOLUME, WM8994_AIF2DAC_VU, WM8994_AIF2DAC_VU); + snd_soc_update_bits(codec, WM8994_AIF1_ADC1_LEFT_VOLUME, + WM8994_AIF1ADC1_VU, WM8994_AIF1ADC1_VU); snd_soc_update_bits(codec, WM8994_AIF1_ADC1_RIGHT_VOLUME, WM8994_AIF1ADC1_VU, WM8994_AIF1ADC1_VU); + snd_soc_update_bits(codec, WM8994_AIF1_ADC2_LEFT_VOLUME, + WM8994_AIF1ADC2_VU, WM8994_AIF1ADC2_VU); snd_soc_update_bits(codec, WM8994_AIF1_ADC2_RIGHT_VOLUME, WM8994_AIF1ADC2_VU, WM8994_AIF1ADC2_VU); + snd_soc_update_bits(codec, WM8994_AIF2_ADC_LEFT_VOLUME, + WM8994_AIF2ADC_VU, WM8994_AIF1ADC2_VU); snd_soc_update_bits(codec, WM8994_AIF2_ADC_RIGHT_VOLUME, WM8994_AIF2ADC_VU, WM8994_AIF1ADC2_VU); + snd_soc_update_bits(codec, WM8994_DAC1_LEFT_VOLUME, + WM8994_DAC1_VU, WM8994_DAC1_VU); snd_soc_update_bits(codec, WM8994_DAC1_RIGHT_VOLUME, WM8994_DAC1_VU, WM8994_DAC1_VU); + snd_soc_update_bits(codec, WM8994_DAC2_LEFT_VOLUME, + WM8994_DAC2_VU, WM8994_DAC2_VU); snd_soc_update_bits(codec, WM8994_DAC2_RIGHT_VOLUME, WM8994_DAC2_VU, WM8994_DAC2_VU); diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 55cdf2982020..91c6b39de50c 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -305,7 +305,7 @@ static int speaker_mode_get(struct snd_kcontrol *kcontrol, /* * Stop any attempts to change speaker mode while the speaker is enabled. * - * We also have some special anti-pop controls dependant on speaker + * We also have some special anti-pop controls dependent on speaker * mode which must be changed along with the mode. */ static int speaker_mode_put(struct snd_kcontrol *kcontrol, @@ -456,7 +456,7 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref, pr_debug("Fvco=%dHz\n", target); - /* Find an appropraite FLL_FRATIO and factor it out of the target */ + /* Find an appropriate FLL_FRATIO and factor it out of the target */ for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) { if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) { fll_div->fll_fratio = fll_fratios[i].fll_fratio; diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 7b6b3c18e299..4005e9af5d61 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -740,12 +740,12 @@ static const struct snd_soc_dapm_route analogue_routes[] = { { "SPKL", "Input Switch", "MIXINL" }, { "SPKL", "IN1LP Switch", "IN1LP" }, - { "SPKL", "Output Switch", "Left Output Mixer" }, + { "SPKL", "Output Switch", "Left Output PGA" }, { "SPKL", NULL, "TOCLK" }, { "SPKR", "Input Switch", "MIXINR" }, { "SPKR", "IN1RP Switch", "IN1RP" }, - { "SPKR", "Output Switch", "Right Output Mixer" }, + { "SPKR", "Output Switch", "Right Output PGA" }, { "SPKR", NULL, "TOCLK" }, { "SPKL Boost", "Direct Voice Switch", "Direct Voice" }, @@ -767,8 +767,8 @@ static const struct snd_soc_dapm_route analogue_routes[] = { { "SPKOUTRP", NULL, "SPKR Driver" }, { "SPKOUTRN", NULL, "SPKR Driver" }, - { "Left Headphone Mux", "Mixer", "Left Output Mixer" }, - { "Right Headphone Mux", "Mixer", "Right Output Mixer" }, + { "Left Headphone Mux", "Mixer", "Left Output PGA" }, + { "Right Headphone Mux", "Mixer", "Right Output PGA" }, { "Headphone PGA", NULL, "Left Headphone Mux" }, { "Headphone PGA", NULL, "Right Headphone Mux" }, diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index bc92ec620004..ac2ded969253 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -16,7 +16,7 @@ * sane processor vendors have a FIFO per AC97 slot, the i.MX has only * one FIFO which combines all valid receive slots. We cannot even select * which slots we want to receive. The WM9712 with which this driver - * was developped with always sends GPIO status data in slot 12 which + * was developed with always sends GPIO status data in slot 12 which * we receive in our (PCM-) data stream. The only chance we have is to * manually skip this data in the FIQ handler. With sampling rates different * from 48000Hz not every frame has valid receive data, so the ratio diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index 0fd6a630db01..e13c6ce46328 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -132,7 +132,7 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream) priv = snd_soc_dai_get_dma_data(cpu_dai, substream); snd_soc_set_runtime_hwparams(substream, &kirkwood_dma_snd_hw); - /* Ensure that all constraints linked to dma burst are fullfilled */ + /* Ensure that all constraints linked to dma burst are fulfilled */ err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, priv->burst * 2, @@ -170,7 +170,7 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream) /* * Enable Error interrupts. We're only ack'ing them but - * it's usefull for diagnostics + * it's useful for diagnostics */ writel((unsigned long)-1, priv->io + KIRKWOOD_ERR_MASK); } diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c index ee2c22475a76..d567c322a2fb 100644 --- a/sound/soc/mid-x86/sst_platform.c +++ b/sound/soc/mid-x86/sst_platform.c @@ -116,18 +116,20 @@ struct snd_soc_dai_driver sst_platform_dai[] = { static inline void sst_set_stream_status(struct sst_runtime_stream *stream, int state) { - spin_lock(&stream->status_lock); + unsigned long flags; + spin_lock_irqsave(&stream->status_lock, flags); stream->stream_status = state; - spin_unlock(&stream->status_lock); + spin_unlock_irqrestore(&stream->status_lock, flags); } static inline int sst_get_stream_status(struct sst_runtime_stream *stream) { int state; + unsigned long flags; - spin_lock(&stream->status_lock); + spin_lock_irqsave(&stream->status_lock, flags); state = stream->stream_status; - spin_unlock(&stream->status_lock); + spin_unlock_irqrestore(&stream->status_lock, flags); return state; } @@ -440,7 +442,7 @@ static int sst_platform_remove(struct platform_device *pdev) snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(sst_platform_dai)); snd_soc_unregister_platform(&pdev->dev); - pr_debug("sst_platform_remove sucess\n"); + pr_debug("sst_platform_remove success\n"); return 0; } @@ -463,7 +465,7 @@ module_init(sst_soc_platform_init); static void __exit sst_soc_platform_exit(void) { platform_driver_unregister(&sst_platform_driver); - pr_debug("sst_soc_platform_exit sucess\n"); + pr_debug("sst_soc_platform_exit success\n"); } module_exit(sst_soc_platform_exit); diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 3167be689621..462cbcbea74a 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -248,7 +248,7 @@ static struct snd_soc_jack_pin ams_delta_hook_switch_pins[] = { */ /* To actually apply any modem controlled configuration changes to the codec, - * we must connect codec DAI pins to the modem for a moment. Be carefull not + * we must connect codec DAI pins to the modem for a moment. Be careful not * to interfere with our digital mute function that shares the same hardware. */ static struct timer_list cx81801_timer; static bool cx81801_cmd_pending; @@ -402,9 +402,9 @@ static struct tty_ldisc_ops cx81801_ops = { /* - * Even if not very usefull, the sound card can still work without any of the + * Even if not very useful, the sound card can still work without any of the * above functonality activated. You can still control its audio input/output - * constellation and speakerphone gain from userspace by issueing AT commands + * constellation and speakerphone gain from userspace by issuing AT commands * over the modem port. */ diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index 78bfdb3f5d7e..452230975632 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -228,7 +228,7 @@ static const struct snd_kcontrol_new neo1973_wm8753_controls[] = { SOC_DAPM_PIN_SWITCH("Handset Mic"), }; -/* GTA02 specific routes and controlls */ +/* GTA02 specific routes and controls */ #ifdef CONFIG_MACH_NEO1973_GTA02 @@ -372,7 +372,7 @@ static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd) return 0; } -/* GTA01 specific controlls */ +/* GTA01 specific controls */ #ifdef CONFIG_MACH_NEO1973_GTA01 diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index 38aac7d57a59..9c7e8b48aed6 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -350,8 +350,8 @@ static int s3c_pcm_set_fmt(struct snd_soc_dai *cpu_dai, ctl = readl(regs + S3C_PCM_CTL); switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_NB_NF: - /* Nothing to do, NB_NF by default */ + case SND_SOC_DAIFMT_IB_NF: + /* Nothing to do, IB_NF by default */ break; default: dev_err(pcm->dev, "Unsupported clock inversion!\n"); diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 0c9997e2d8c0..23c0e83d4c19 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1200,10 +1200,11 @@ static int fsi_probe(struct platform_device *pdev) master->fsib.master = master; pm_runtime_enable(&pdev->dev); - pm_runtime_resume(&pdev->dev); dev_set_drvdata(&pdev->dev, master); + pm_runtime_get_sync(&pdev->dev); fsi_soft_all_reset(master); + pm_runtime_put_sync(&pdev->dev); ret = request_irq(irq, &fsi_interrupt, IRQF_DISABLED, id_entry->name, master); @@ -1218,8 +1219,17 @@ static int fsi_probe(struct platform_device *pdev) goto exit_free_irq; } - return snd_soc_register_dais(&pdev->dev, fsi_soc_dai, ARRAY_SIZE(fsi_soc_dai)); + ret = snd_soc_register_dais(&pdev->dev, fsi_soc_dai, + ARRAY_SIZE(fsi_soc_dai)); + if (ret < 0) { + dev_err(&pdev->dev, "cannot snd dai register\n"); + goto exit_snd_soc; + } + + return ret; +exit_snd_soc: + snd_soc_unregister_platform(&pdev->dev); exit_free_irq: free_irq(irq, master); exit_iounmap: @@ -1238,12 +1248,11 @@ static int fsi_remove(struct platform_device *pdev) master = dev_get_drvdata(&pdev->dev); - snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(fsi_soc_dai)); - snd_soc_unregister_platform(&pdev->dev); - + free_irq(master->irq, master); pm_runtime_disable(&pdev->dev); - free_irq(master->irq, master); + snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(fsi_soc_dai)); + snd_soc_unregister_platform(&pdev->dev); iounmap(master->base); kfree(master); @@ -1321,3 +1330,4 @@ module_exit(fsi_mobile_exit); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("SuperH onchip FSI audio driver"); MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>"); +MODULE_ALIAS("platform:fsi-pcm-audio"); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b76b74db0968..d8562ce4de7a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -629,6 +629,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) runtime->hw.rates |= codec_dai_drv->capture.rates; } + ret = -EINVAL; snd_pcm_limit_hw_rates(runtime); if (!runtime->hw.rates) { printk(KERN_ERR "asoc: %s <-> %s No matching rates\n", @@ -640,7 +641,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) codec_dai->name, cpu_dai->name); goto config_err; } - if (!runtime->hw.channels_min || !runtime->hw.channels_max) { + if (!runtime->hw.channels_min || !runtime->hw.channels_max || + runtime->hw.channels_min > runtime->hw.channels_max) { printk(KERN_ERR "asoc: %s <-> %s No matching channels\n", codec_dai->name, cpu_dai->name); goto config_err; @@ -2060,6 +2062,7 @@ const struct dev_pm_ops snd_soc_pm_ops = { .resume = snd_soc_resume, .poweroff = snd_soc_poweroff, }; +EXPORT_SYMBOL_GPL(snd_soc_pm_ops); /* ASoC platform driver */ static struct platform_driver soc_driver = { diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index fcab80b36a37..fc017c0a7b5d 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -331,7 +331,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, goto err; if (gpios[i].wake) { - ret = set_irq_wake(gpio_to_irq(gpios[i].gpio), 1); + ret = irq_set_irq_wake(gpio_to_irq(gpios[i].gpio), 1); if (ret != 0) printk(KERN_ERR "Failed to mark GPIO %d as wake source: %d\n", diff --git a/sound/soc/tegra/harmony.c b/sound/soc/tegra/harmony.c index 8585957477eb..556a57133925 100644 --- a/sound/soc/tegra/harmony.c +++ b/sound/soc/tegra/harmony.c @@ -370,6 +370,7 @@ static struct platform_driver tegra_snd_harmony_driver = { .driver = { .name = DRV_NAME, .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, }, .probe = tegra_snd_harmony_probe, .remove = __devexit_p(tegra_snd_harmony_remove), |