diff options
Diffstat (limited to 'sound/soc')
44 files changed, 185 insertions, 108 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/Kconfig b/sound/soc/codecs/Kconfig index d63c1754e05f..6943e24a74a1 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -51,7 +51,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TWL6040 if TWL4030_CORE select SND_SOC_UDA134X select SND_SOC_UDA1380 if I2C - select SND_SOC_WL1273 if RADIO_WL1273 + select SND_SOC_WL1273 if MFD_WL1273_CORE select SND_SOC_WM2000 if I2C select SND_SOC_WM8350 if MFD_WM8350 select SND_SOC_WM8400 if MFD_WM8400 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/cq93vc.c b/sound/soc/codecs/cq93vc.c index 347a567b01e1..b8066ef10bb0 100644 --- a/sound/soc/codecs/cq93vc.c +++ b/sound/soc/codecs/cq93vc.c @@ -153,7 +153,8 @@ static int cq93vc_resume(struct snd_soc_codec *codec) static int cq93vc_probe(struct snd_soc_codec *codec) { - struct davinci_vc *davinci_vc = snd_soc_codec_get_drvdata(codec); + struct davinci_vc *davinci_vc = + mfd_get_data(to_platform_device(codec->dev)); davinci_vc->cq93vc.codec = codec; codec->control_data = davinci_vc; 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/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 1f7217f703ee..ff29380c9ed3 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -772,6 +772,7 @@ static int sgtl5000_pcm_hw_params(struct snd_pcm_substream *substream, return 0; } +#ifdef CONFIG_REGULATOR static int ldo_regulator_is_enabled(struct regulator_dev *dev) { struct ldo_regulator *ldo = rdev_get_drvdata(dev); @@ -901,6 +902,19 @@ static int ldo_regulator_remove(struct snd_soc_codec *codec) return 0; } +#else +static int ldo_regulator_register(struct snd_soc_codec *codec, + struct regulator_init_data *init_data, + int voltage) +{ + return -EINVAL; +} + +static int ldo_regulator_remove(struct snd_soc_codec *codec) +{ + return 0; +} +#endif /* * set dac bias 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 00b6d87e7bdb..082e9d51963f 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -324,6 +324,10 @@ static void dac33_init_chip(struct snd_soc_codec *codec) dac33_write(codec, DAC33_OUT_AMP_CTRL, dac33_read_reg_cache(codec, DAC33_OUT_AMP_CTRL)); + dac33_write(codec, DAC33_LDAC_PWR_CTRL, + dac33_read_reg_cache(codec, DAC33_LDAC_PWR_CTRL)); + dac33_write(codec, DAC33_RDAC_PWR_CTRL, + dac33_read_reg_cache(codec, DAC33_RDAC_PWR_CTRL)); } static inline int dac33_read_id(struct snd_soc_codec *codec) @@ -670,6 +674,7 @@ static inline void dac33_prefill_handler(struct tlv320dac33_priv *dac33) { struct snd_soc_codec *codec = dac33->codec; unsigned int delay; + unsigned long flags; switch (dac33->fifo_mode) { case DAC33_FIFO_MODE1: @@ -677,10 +682,10 @@ static inline void dac33_prefill_handler(struct tlv320dac33_priv *dac33) DAC33_THRREG(dac33->nsample)); /* Take the timestamps */ - spin_lock_irq(&dac33->lock); + spin_lock_irqsave(&dac33->lock, flags); dac33->t_stamp2 = ktime_to_us(ktime_get()); dac33->t_stamp1 = dac33->t_stamp2; - spin_unlock_irq(&dac33->lock); + spin_unlock_irqrestore(&dac33->lock, flags); dac33_write16(codec, DAC33_PREFILL_MSB, DAC33_THRREG(dac33->alarm_threshold)); @@ -692,11 +697,11 @@ static inline void dac33_prefill_handler(struct tlv320dac33_priv *dac33) break; case DAC33_FIFO_MODE7: /* Take the timestamp */ - spin_lock_irq(&dac33->lock); + spin_lock_irqsave(&dac33->lock, flags); dac33->t_stamp1 = ktime_to_us(ktime_get()); /* Move back the timestamp with drain time */ dac33->t_stamp1 -= dac33->mode7_us_to_lthr; - spin_unlock_irq(&dac33->lock); + spin_unlock_irqrestore(&dac33->lock, flags); dac33_write16(codec, DAC33_PREFILL_MSB, DAC33_THRREG(DAC33_MODE7_MARGIN)); @@ -714,13 +719,14 @@ static inline void dac33_prefill_handler(struct tlv320dac33_priv *dac33) static inline void dac33_playback_handler(struct tlv320dac33_priv *dac33) { struct snd_soc_codec *codec = dac33->codec; + unsigned long flags; switch (dac33->fifo_mode) { case DAC33_FIFO_MODE1: /* Take the timestamp */ - spin_lock_irq(&dac33->lock); + spin_lock_irqsave(&dac33->lock, flags); dac33->t_stamp2 = ktime_to_us(ktime_get()); - spin_unlock_irq(&dac33->lock); + spin_unlock_irqrestore(&dac33->lock, flags); dac33_write16(codec, DAC33_NSAMPLE_MSB, DAC33_THRREG(dac33->nsample)); @@ -773,10 +779,11 @@ static irqreturn_t dac33_interrupt_handler(int irq, void *dev) { struct snd_soc_codec *codec = dev; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); + unsigned long flags; - spin_lock(&dac33->lock); + spin_lock_irqsave(&dac33->lock, flags); dac33->t_stamp1 = ktime_to_us(ktime_get()); - spin_unlock(&dac33->lock); + spin_unlock_irqrestore(&dac33->lock, flags); /* Do not schedule the workqueue in Mode7 */ if (dac33->fifo_mode != DAC33_FIFO_MODE7) @@ -1020,7 +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; @@ -1173,15 +1180,16 @@ static snd_pcm_sframes_t dac33_dai_delay( unsigned int time_delta, uthr; int samples_out, samples_in, samples; snd_pcm_sframes_t delay = 0; + unsigned long flags; switch (dac33->fifo_mode) { case DAC33_FIFO_BYPASS: break; case DAC33_FIFO_MODE1: - spin_lock(&dac33->lock); + spin_lock_irqsave(&dac33->lock, flags); t0 = dac33->t_stamp1; t1 = dac33->t_stamp2; - spin_unlock(&dac33->lock); + spin_unlock_irqrestore(&dac33->lock, flags); t_now = ktime_to_us(ktime_get()); /* We have not started to fill the FIFO yet, delay is 0 */ @@ -1246,10 +1254,10 @@ static snd_pcm_sframes_t dac33_dai_delay( } break; case DAC33_FIFO_MODE7: - spin_lock(&dac33->lock); + spin_lock_irqsave(&dac33->lock, flags); t0 = dac33->t_stamp1; uthr = dac33->uthr; - spin_unlock(&dac33->lock); + spin_unlock_irqrestore(&dac33->lock, flags); t_now = ktime_to_us(ktime_get()); /* We have not started to fill the FIFO yet, delay is 0 */ diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index e4d464b937d6..575238d68e5e 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -26,6 +26,7 @@ #include <linux/pm.h> #include <linux/i2c.h> #include <linux/platform_device.h> +#include <linux/mfd/core.h> #include <linux/i2c/twl.h> #include <linux/slab.h> #include <sound/core.h> @@ -280,7 +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"); } @@ -732,7 +733,8 @@ static int aif_event(struct snd_soc_dapm_widget *w, static void headset_ramp(struct snd_soc_codec *codec, int ramp) { - struct twl4030_codec_audio_data *pdata = codec->dev->platform_data; + struct twl4030_codec_audio_data *pdata = + mfd_get_data(to_platform_device(codec->dev)); unsigned char hs_gain, hs_pop; struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); /* Base values for ramp delay calculation: 2^19 - 2^26 */ @@ -2016,7 +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" @@ -2026,7 +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; @@ -2297,7 +2299,7 @@ static struct snd_soc_codec_driver soc_codec_dev_twl4030 = { static int __devinit twl4030_codec_probe(struct platform_device *pdev) { - struct twl4030_codec_audio_data *pdata = pdev->dev.platform_data; + struct twl4030_codec_audio_data *pdata = mfd_get_data(pdev); if (!pdata) { dev_err(&pdev->dev, "platform_data is missing\n"); diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 482fcdb59bfa..255901c4460d 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1629,8 +1629,10 @@ static int twl6040_probe(struct snd_soc_codec *codec) priv->naudint = naudint; priv->workqueue = create_singlethread_workqueue("twl6040-codec"); - if (!priv->workqueue) + if (!priv->workqueue) { + ret = -ENOMEM; goto work_err; + } INIT_DELAYED_WORK(&priv->delayed_work, twl6040_accessory_work); diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index e76847a9438b..48ffd406a71d 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -486,7 +486,8 @@ static struct snd_soc_dai_driver uda134x_dai = { static int uda134x_soc_probe(struct snd_soc_codec *codec) { struct uda134x_priv *uda134x; - struct uda134x_platform_data *pd = dev_get_drvdata(codec->card->dev); + struct uda134x_platform_data *pd = codec->card->dev->platform_data; + int ret; printk(KERN_INFO "UDA134X SoC Audio Codec\n"); diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c index 861b28f543d2..c8a874d0d4ca 100644 --- a/sound/soc/codecs/wl1273.c +++ b/sound/soc/codecs/wl1273.c @@ -3,7 +3,7 @@ * * Author: Matti Aaltonen, <matti.j.aaltonen@nokia.com> * - * Copyright: (C) 2010 Nokia Corporation + * Copyright: (C) 2010, 2011 Nokia Corporation * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License @@ -179,7 +179,12 @@ static int snd_wl1273_get_audio_route(struct snd_kcontrol *kcontrol, return 0; } -static const char *wl1273_audio_route[] = { "Bt", "FmRx", "FmTx" }; +/* + * TODO: Implement the audio routing in the driver. Now this control + * only indicates the setting that has been done elsewhere (in the user + * space). + */ +static const char * const wl1273_audio_route[] = { "Bt", "FmRx", "FmTx" }; static int snd_wl1273_set_audio_route(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -239,7 +244,7 @@ static int snd_wl1273_fm_audio_put(struct snd_kcontrol *kcontrol, return 1; } -static const char *wl1273_audio_strings[] = { "Digital", "Analog" }; +static const char * const wl1273_audio_strings[] = { "Digital", "Analog" }; static const struct soc_enum wl1273_audio_enum = SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(wl1273_audio_strings), @@ -436,7 +441,8 @@ EXPORT_SYMBOL_GPL(wl1273_get_format); static int wl1273_probe(struct snd_soc_codec *codec) { - struct wl1273_core **core = codec->dev->platform_data; + struct wl1273_core **core = + mfd_get_data(to_platform_device(codec->dev)); struct wl1273_priv *wl1273; int r; diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 3c3bc079167e..736b785e3756 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -22,6 +22,7 @@ #include <linux/regulator/consumer.h> #include <linux/mfd/wm8400-audio.h> #include <linux/mfd/wm8400-private.h> +#include <linux/mfd/core.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -1377,7 +1378,7 @@ static void wm8400_probe_deferred(struct work_struct *work) static int wm8400_codec_probe(struct snd_soc_codec *codec) { - struct wm8400 *wm8400 = dev_get_platdata(codec->dev); + struct wm8400 *wm8400 = mfd_get_data(to_platform_device(codec->dev)); struct wm8400_priv *priv; int ret; u16 reg; 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/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c index 9d2afccc3a2d..13e05a302a92 100644 --- a/sound/soc/davinci/davinci-vcif.c +++ b/sound/soc/davinci/davinci-vcif.c @@ -205,7 +205,7 @@ static struct snd_soc_dai_driver davinci_vcif_dai = { static int davinci_vcif_probe(struct platform_device *pdev) { - struct davinci_vc *davinci_vc = platform_get_drvdata(pdev); + struct davinci_vc *davinci_vc = mfd_get_data(pdev); struct davinci_vcif_dev *davinci_vcif_dev; int ret; diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c index 671ef8dd524c..aab7765f401a 100644 --- a/sound/soc/imx/imx-pcm-dma-mx2.c +++ b/sound/soc/imx/imx-pcm-dma-mx2.c @@ -110,12 +110,12 @@ static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream, slave_config.direction = DMA_TO_DEVICE; slave_config.dst_addr = dma_params->dma_addr; slave_config.dst_addr_width = buswidth; - slave_config.dst_maxburst = dma_params->burstsize; + slave_config.dst_maxburst = dma_params->burstsize * buswidth; } else { slave_config.direction = DMA_FROM_DEVICE; slave_config.src_addr = dma_params->dma_addr; slave_config.src_addr_width = buswidth; - slave_config.src_maxburst = dma_params->burstsize; + slave_config.src_maxburst = dma_params->burstsize * buswidth; } ret = dmaengine_slave_config(iprtd->dma_chan, &slave_config); @@ -303,6 +303,11 @@ static struct snd_soc_platform_driver imx_soc_platform_mx2 = { static int __devinit imx_soc_platform_probe(struct platform_device *pdev) { + struct imx_ssi *ssi = platform_get_drvdata(pdev); + + ssi->dma_params_tx.burstsize = 6; + ssi->dma_params_rx.burstsize = 4; + return snd_soc_register_platform(&pdev->dev, &imx_soc_platform_mx2); } 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/imx/imx-ssi.h b/sound/soc/imx/imx-ssi.h index a4406a134892..dc8a87530e3e 100644 --- a/sound/soc/imx/imx-ssi.h +++ b/sound/soc/imx/imx-ssi.h @@ -234,7 +234,4 @@ void imx_pcm_free(struct snd_pcm *pcm); */ #define IMX_SSI_DMABUF_SIZE (64 * 1024) -#define DMA_RXFIFO_BURST 0x4 -#define DMA_TXFIFO_BURST 0x6 - #endif /* _IMX_SSI_H */ 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/pxa/corgi.c b/sound/soc/pxa/corgi.c index 784cff5f67e8..9027da466cae 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -310,7 +310,7 @@ static struct snd_soc_dai_link corgi_dai = { .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "wm8731-hifi", .platform_name = "pxa-pcm-audio", - .codec_name = "wm8731-codec-0.001b", + .codec_name = "wm8731-codec.0-001b", .init = corgi_wm8731_init, .ops = &corgi_ops, }; diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index 02fb66416ddc..2ce0b2d891d5 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -65,6 +65,7 @@ static int pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream) if (prtd->dma_ch >= 0) { pxa_free_dma(prtd->dma_ch); prtd->dma_ch = -1; + prtd->params = NULL; } return 0; diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index ac577263b3e3..b6445757fc54 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -167,7 +167,7 @@ static struct snd_soc_dai_link zylonite_dai[] = { .codec_name = "wm9713-codec", .platform_name = "pxa-pcm-audio", .cpu_dai_name = "pxa2xx-ac97", - .codec_name = "wm9713-hifi", + .codec_dai_name = "wm9713-hifi", .init = zylonite_wm9713_init, }, { @@ -176,7 +176,7 @@ static struct snd_soc_dai_link zylonite_dai[] = { .codec_name = "wm9713-codec", .platform_name = "pxa-pcm-audio", .cpu_dai_name = "pxa2xx-ac97-aux", - .codec_name = "wm9713-aux", + .codec_dai_name = "wm9713-aux", }, { .name = "WM9713 Voice", @@ -184,7 +184,7 @@ static struct snd_soc_dai_link zylonite_dai[] = { .codec_name = "wm9713-codec", .platform_name = "pxa-pcm-audio", .cpu_dai_name = "pxa-ssp-dai.2", - .codec_name = "wm9713-voice", + .codec_dai_name = "wm9713-voice", .ops = &zylonite_voice_ops, }, }; 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/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c index 3cb700751078..dc9d551f6788 100644 --- a/sound/soc/samsung/s3c24xx_uda134x.c +++ b/sound/soc/samsung/s3c24xx_uda134x.c @@ -219,7 +219,7 @@ static struct snd_soc_ops s3c24xx_uda134x_ops = { static struct snd_soc_dai_link s3c24xx_uda134x_dai_link = { .name = "UDA134X", .stream_name = "UDA134X", - .codec_name = "uda134x-hifi", + .codec_name = "uda134x-codec", .codec_dai_name = "uda134x-hifi", .cpu_dai_name = "s3c24xx-iis", .ops = &s3c24xx_uda134x_ops, @@ -314,6 +314,7 @@ static int s3c24xx_uda134x_probe(struct platform_device *pdev) platform_set_drvdata(s3c24xx_uda134x_snd_device, &snd_soc_s3c24xx_uda134x); + platform_device_add_data(s3c24xx_uda134x_snd_device, &s3c24xx_uda134x, sizeof(s3c24xx_uda134x)); ret = platform_device_add(s3c24xx_uda134x_snd_device); if (ret) { printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: Unable to add\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 17efacdb248a..d8562ce4de7a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -92,8 +92,8 @@ static int min_bytes_needed(unsigned long val) static int format_register_str(struct snd_soc_codec *codec, unsigned int reg, char *buf, size_t len) { - int wordsize = codec->driver->reg_word_size * 2; - int regsize = min_bytes_needed(codec->driver->reg_cache_size) * 2; + int wordsize = min_bytes_needed(codec->driver->reg_cache_size) * 2; + int regsize = codec->driver->reg_word_size * 2; int ret; char tmpbuf[len + 1]; char regbuf[regsize + 1]; @@ -132,8 +132,8 @@ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf, size_t total = 0; loff_t p = 0; - wordsize = codec->driver->reg_word_size * 2; - regsize = min_bytes_needed(codec->driver->reg_cache_size) * 2; + wordsize = min_bytes_needed(codec->driver->reg_cache_size) * 2; + regsize = codec->driver->reg_word_size * 2; len = wordsize + regsize + 2 + 1; @@ -259,8 +259,6 @@ static ssize_t codec_reg_write_file(struct file *file, while (*start == ' ') start++; reg = simple_strtoul(start, &start, 16); - if ((reg >= codec->driver->reg_cache_size) || (reg % step)) - return -EINVAL; while (*start == ' ') start++; if (strict_strtoul(start, 16, &value)) @@ -631,6 +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", @@ -642,7 +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; @@ -2062,6 +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), |