diff options
Diffstat (limited to 'sound')
| -rw-r--r-- | sound/soc/codecs/tlv320aic3x.c | 25 | ||||
| -rw-r--r-- | sound/soc/codecs/tlv320dac33.c | 223 | ||||
| -rw-r--r-- | sound/soc/codecs/tpa6130a2.c | 99 | ||||
| -rw-r--r-- | sound/soc/codecs/twl4030.c | 86 | ||||
| -rw-r--r-- | sound/soc/omap/Kconfig | 10 | ||||
| -rw-r--r-- | sound/soc/omap/Makefile | 2 | ||||
| -rw-r--r-- | sound/soc/omap/omap3pandora.c | 2 | ||||
| -rw-r--r-- | sound/soc/omap/rx51.c | 294 | ||||
| -rw-r--r-- | sound/soc/omap/zoom2.c | 3 |
9 files changed, 601 insertions, 143 deletions
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 584bc1e67f76..d57372be7a96 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -38,6 +38,7 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> +#include <linux/gpio.h> #include <linux/regulator/consumer.h> #include <linux/platform_device.h> #include <sound/core.h> @@ -47,6 +48,7 @@ #include <sound/soc-dapm.h> #include <sound/initval.h> #include <sound/tlv.h> +#include <sound/tlv320aic3x.h> #include "tlv320aic3x.h" @@ -64,6 +66,7 @@ struct aic3x_priv { struct regulator_bulk_data supplies[AIC3X_NUM_SUPPLIES]; unsigned int sysclk; int master; + int gpio_reset; }; /* @@ -1278,6 +1281,10 @@ static int aic3x_unregister(struct aic3x_priv *aic3x) snd_soc_unregister_dai(&aic3x_dai); snd_soc_unregister_codec(&aic3x->codec); + if (aic3x->gpio_reset >= 0) { + gpio_set_value(aic3x->gpio_reset, 0); + gpio_free(aic3x->gpio_reset); + } regulator_bulk_disable(ARRAY_SIZE(aic3x->supplies), aic3x->supplies); regulator_bulk_free(ARRAY_SIZE(aic3x->supplies), aic3x->supplies); @@ -1302,6 +1309,7 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, { struct snd_soc_codec *codec; struct aic3x_priv *aic3x; + struct aic3x_pdata *pdata = i2c->dev.platform_data; int ret, i; aic3x = kzalloc(sizeof(struct aic3x_priv), GFP_KERNEL); @@ -1318,6 +1326,15 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, aic3x); + aic3x->gpio_reset = -1; + if (pdata && pdata->gpio_reset >= 0) { + ret = gpio_request(pdata->gpio_reset, "tlv320aic3x reset"); + if (ret != 0) + goto err_gpio; + aic3x->gpio_reset = pdata->gpio_reset; + gpio_direction_output(aic3x->gpio_reset, 0); + } + for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) aic3x->supplies[i].supply = aic3x_supply_names[i]; @@ -1335,11 +1352,19 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, goto err_enable; } + if (aic3x->gpio_reset >= 0) { + udelay(1); + gpio_set_value(aic3x->gpio_reset, 1); + } + return aic3x_register(codec); err_enable: regulator_bulk_free(ARRAY_SIZE(aic3x->supplies), aic3x->supplies); err_get: + if (aic3x->gpio_reset >= 0) + gpio_free(aic3x->gpio_reset); +err_gpio: kfree(aic3x); return ret; } diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 3eddaec728c1..ad5e2636c944 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -61,6 +61,8 @@ #define US_TO_SAMPLES(rate, us) \ (rate / (1000000 / us)) +static void dac33_calculate_times(struct snd_pcm_substream *substream); +static int dac33_prepare_chip(struct snd_pcm_substream *substream); static struct snd_soc_codec *tlv320dac33_codec; @@ -91,6 +93,7 @@ struct tlv320dac33_priv { struct work_struct work; struct snd_soc_codec codec; struct regulator_bulk_data supplies[DAC33_NUM_SUPPLIES]; + struct snd_pcm_substream *substream; int power_gpio; int chip_power; int irq; @@ -284,45 +287,47 @@ static int dac33_write16(struct snd_soc_codec *codec, unsigned int reg, return ret; } -static void dac33_restore_regs(struct snd_soc_codec *codec) +static void dac33_init_chip(struct snd_soc_codec *codec) { struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); - u8 *cache = codec->reg_cache; - u8 data[2]; - int i, ret; - if (!dac33->chip_power) + if (unlikely(!dac33->chip_power)) return; - for (i = DAC33_PWR_CTRL; i <= DAC33_INTP_CTRL_B; i++) { - data[0] = i; - data[1] = cache[i]; - /* Skip the read only registers */ - if ((i >= DAC33_INT_OSC_STATUS && - i <= DAC33_INT_OSC_FREQ_RAT_READ_B) || - (i >= DAC33_FIFO_WPTR_MSB && i <= DAC33_FIFO_IRQ_FLAG) || - i == DAC33_DAC_STATUS_FLAGS || - i == DAC33_SRC_EST_REF_CLK_RATIO_A || - i == DAC33_SRC_EST_REF_CLK_RATIO_B) - continue; - ret = codec->hw_write(codec->control_data, data, 2); - if (ret != 2) - dev_err(codec->dev, "Write failed (%d)\n", ret); - } - for (i = DAC33_LDAC_PWR_CTRL; i <= DAC33_LINEL_TO_LLO_VOL; i++) { - data[0] = i; - data[1] = cache[i]; - ret = codec->hw_write(codec->control_data, data, 2); - if (ret != 2) - dev_err(codec->dev, "Write failed (%d)\n", ret); - } - for (i = DAC33_LINER_TO_RLO_VOL; i <= DAC33_OSC_TRIM; i++) { - data[0] = i; - data[1] = cache[i]; - ret = codec->hw_write(codec->control_data, data, 2); - if (ret != 2) - dev_err(codec->dev, "Write failed (%d)\n", ret); - } + /* 44-46: DAC Control Registers */ + /* A : DAC sample rate Fsref/1.5 */ + dac33_write(codec, DAC33_DAC_CTRL_A, DAC33_DACRATE(0)); + /* B : DAC src=normal, not muted */ + dac33_write(codec, DAC33_DAC_CTRL_B, DAC33_DACSRCR_RIGHT | + DAC33_DACSRCL_LEFT); + /* C : (defaults) */ + dac33_write(codec, DAC33_DAC_CTRL_C, 0x00); + + /* 73 : volume soft stepping control, + clock source = internal osc (?) */ + dac33_write(codec, DAC33_ANA_VOL_SOFT_STEP_CTRL, DAC33_VOLCLKEN); + + dac33_write(codec, DAC33_PWR_CTRL, DAC33_PDNALLB); + + /* Restore only selected registers (gains mostly) */ + dac33_write(codec, DAC33_LDAC_DIG_VOL_CTRL, + dac33_read_reg_cache(codec, DAC33_LDAC_DIG_VOL_CTRL)); + dac33_write(codec, DAC33_RDAC_DIG_VOL_CTRL, + dac33_read_reg_cache(codec, DAC33_RDAC_DIG_VOL_CTRL)); + + dac33_write(codec, DAC33_LINEL_TO_LLO_VOL, + dac33_read_reg_cache(codec, DAC33_LINEL_TO_LLO_VOL)); + dac33_write(codec, DAC33_LINER_TO_RLO_VOL, + dac33_read_reg_cache(codec, DAC33_LINER_TO_RLO_VOL)); +} + +static inline void dac33_read_id(struct snd_soc_codec *codec) +{ + u8 reg; + + dac33_read(codec, DAC33_DEVICE_ID_MSB, ®); + dac33_read(codec, DAC33_DEVICE_ID_LSB, ®); + dac33_read(codec, DAC33_DEVICE_REV_ID, ®); } static inline void dac33_soft_power(struct snd_soc_codec *codec, int power) @@ -341,9 +346,17 @@ static inline void dac33_soft_power(struct snd_soc_codec *codec, int power) static int dac33_hard_power(struct snd_soc_codec *codec, int power) { struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); - int ret; + int ret = 0; mutex_lock(&dac33->mutex); + + /* Safety check */ + if (unlikely(power == dac33->chip_power)) { + dev_warn(codec->dev, "Trying to set the same power state: %s\n", + power ? "ON" : "OFF"); + goto exit; + } + if (power) { ret = regulator_bulk_enable(ARRAY_SIZE(dac33->supplies), dac33->supplies); @@ -357,11 +370,6 @@ static int dac33_hard_power(struct snd_soc_codec *codec, int power) gpio_set_value(dac33->power_gpio, 1); dac33->chip_power = 1; - - /* Restore registers */ - dac33_restore_regs(codec); - - dac33_soft_power(codec, 1); } else { dac33_soft_power(codec, 0); if (dac33->power_gpio >= 0) @@ -383,6 +391,22 @@ exit: return ret; } +static int playback_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(w->codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (likely(dac33->substream)) { + dac33_calculate_times(dac33->substream); + dac33_prepare_chip(dac33->substream); + } + break; + } + return 0; +} + static int dac33_get_nsample(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -512,6 +536,8 @@ static const struct snd_soc_dapm_widget dac33_dapm_widgets[] = { DAC33_OUT_AMP_PWR_CTRL, 6, 3, 3, 0), SND_SOC_DAPM_REG(snd_soc_dapm_mixer, "Output Right Amp Power", DAC33_OUT_AMP_PWR_CTRL, 4, 3, 3, 0), + + SND_SOC_DAPM_PRE("Prepare Playback", playback_event), }; static const struct snd_soc_dapm_route audio_map[] = { @@ -554,18 +580,18 @@ static int dac33_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: if (codec->bias_level == SND_SOC_BIAS_OFF) { + /* Coming from OFF, switch on the codec */ ret = dac33_hard_power(codec, 1); if (ret != 0) return ret; - } - dac33_soft_power(codec, 0); + dac33_init_chip(codec); + } break; case SND_SOC_BIAS_OFF: ret = dac33_hard_power(codec, 0); if (ret != 0) return ret; - break; } codec->bias_level = level; @@ -708,6 +734,31 @@ static void dac33_oscwait(struct snd_soc_codec *codec) "internal oscillator calibration failed\n"); } +static int dac33_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); + + /* Stream started, save the substream pointer */ + dac33->substream = substream; + + return 0; +} + +static void dac33_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); + + dac33->substream = NULL; +} + static int dac33_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -791,6 +842,16 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream) } mutex_lock(&dac33->mutex); + + if (!dac33->chip_power) { + /* + * Chip is not powered yet. + * Do the init in the dac33_set_bias_level later. + */ + mutex_unlock(&dac33->mutex); + return 0; + } + dac33_soft_power(codec, 0); dac33_soft_power(codec, 1); @@ -997,15 +1058,6 @@ static void dac33_calculate_times(struct snd_pcm_substream *substream) } -static int dac33_pcm_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - dac33_calculate_times(substream); - dac33_prepare_chip(substream); - - return 0; -} - static int dac33_pcm_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -1269,35 +1321,6 @@ static int dac33_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } -static void dac33_init_chip(struct snd_soc_codec *codec) -{ - /* 44-46: DAC Control Registers */ - /* A : DAC sample rate Fsref/1.5 */ - dac33_write(codec, DAC33_DAC_CTRL_A, DAC33_DACRATE(0)); - /* B : DAC src=normal, not muted */ - dac33_write(codec, DAC33_DAC_CTRL_B, DAC33_DACSRCR_RIGHT | - DAC33_DACSRCL_LEFT); - /* C : (defaults) */ - dac33_write(codec, DAC33_DAC_CTRL_C, 0x00); - - /* 64-65 : L&R DAC power control - Line In -> OUT 1V/V Gain, DAC -> OUT 4V/V Gain*/ - dac33_write(codec, DAC33_LDAC_PWR_CTRL, DAC33_LROUT_GAIN(2)); - dac33_write(codec, DAC33_RDAC_PWR_CTRL, DAC33_LROUT_GAIN(2)); - - /* 73 : volume soft stepping control, - clock source = internal osc (?) */ - dac33_write(codec, DAC33_ANA_VOL_SOFT_STEP_CTRL, DAC33_VOLCLKEN); - - /* 66 : LOP/LOM Modes */ - dac33_write(codec, DAC33_OUT_AMP_CM_CTRL, 0xff); - - /* 68 : LOM inverted from LOP */ - dac33_write(codec, DAC33_OUT_AMP_CTRL, (3<<2)); - - dac33_write(codec, DAC33_PWR_CTRL, DAC33_PDNALLB); -} - static int dac33_soc_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); @@ -1311,11 +1334,6 @@ static int dac33_soc_probe(struct platform_device *pdev) socdev->card->codec = codec; dac33 = snd_soc_codec_get_drvdata(codec); - /* Power up the codec */ - dac33_hard_power(codec, 1); - /* Set default configuration */ - dac33_init_chip(codec); - /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) { @@ -1332,12 +1350,6 @@ static int dac33_soc_probe(struct platform_device *pdev) dac33_add_widgets(codec); - /* power on device */ - dac33_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - /* Bias level configuration has enabled regulator an extra time */ - regulator_bulk_disable(ARRAY_SIZE(dac33->supplies), dac33->supplies); - return 0; pcm_err: @@ -1374,6 +1386,8 @@ static int dac33_soc_resume(struct platform_device *pdev) struct snd_soc_codec *codec = socdev->card->codec; dac33_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + if (codec->suspend_bias_level == SND_SOC_BIAS_ON) + dac33_set_bias_level(codec, SND_SOC_BIAS_PREPARE); dac33_set_bias_level(codec, codec->suspend_bias_level); return 0; @@ -1392,8 +1406,9 @@ EXPORT_SYMBOL_GPL(soc_codec_dev_tlv320dac33); #define DAC33_FORMATS SNDRV_PCM_FMTBIT_S16_LE static struct snd_soc_dai_ops dac33_dai_ops = { + .startup = dac33_startup, + .shutdown = dac33_shutdown, .hw_params = dac33_hw_params, - .prepare = dac33_pcm_prepare, .trigger = dac33_pcm_trigger, .delay = dac33_dai_delay, .set_sysclk = dac33_set_dai_sysclk, @@ -1447,6 +1462,7 @@ static int __devinit dac33_i2c_probe(struct i2c_client *client, codec->hw_write = (hw_write_t) i2c_master_send; codec->bias_level = SND_SOC_BIAS_OFF; codec->set_bias_level = dac33_set_bias_level; + codec->idle_bias_off = 1; codec->dai = &dac33_dai; codec->num_dai = 1; codec->reg_cache_size = ARRAY_SIZE(dac33_reg); @@ -1487,8 +1503,6 @@ static int __devinit dac33_i2c_probe(struct i2c_client *client, goto error_gpio; } gpio_direction_output(dac33->power_gpio, 0); - } else { - dac33->chip_power = 1; } /* Check if the IRQ number is valid and request it */ @@ -1526,12 +1540,14 @@ static int __devinit dac33_i2c_probe(struct i2c_client *client, goto err_get; } - ret = regulator_bulk_enable(ARRAY_SIZE(dac33->supplies), - dac33->supplies); + /* Read the tlv320dac33 ID registers */ + ret = dac33_hard_power(codec, 1); if (ret != 0) { - dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); - goto err_enable; + dev_err(codec->dev, "Failed to power up codec: %d\n", ret); + goto error_codec; } + dac33_read_id(codec); + dac33_hard_power(codec, 0); ret = snd_soc_register_codec(codec); if (ret != 0) { @@ -1546,14 +1562,9 @@ static int __devinit dac33_i2c_probe(struct i2c_client *client, goto error_codec; } - /* Shut down the codec for now */ - dac33_hard_power(codec, 0); - return ret; error_codec: - regulator_bulk_disable(ARRAY_SIZE(dac33->supplies), dac33->supplies); -err_enable: regulator_bulk_free(ARRAY_SIZE(dac33->supplies), dac33->supplies); err_get: if (dac33->irq >= 0) { @@ -1577,7 +1588,9 @@ static int __devexit dac33_i2c_remove(struct i2c_client *client) struct tlv320dac33_priv *dac33; dac33 = i2c_get_clientdata(client); - dac33_hard_power(&dac33->codec, 0); + + if (unlikely(dac33->chip_power)) + dac33_hard_power(&dac33->codec, 0); if (dac33->power_gpio >= 0) gpio_free(dac33->power_gpio); diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 958d49c969ac..31f67b527ca1 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -46,6 +46,9 @@ static const char *tpa6140a2_supply_names[TPA6130A2_NUM_SUPPLIES] = { "AVdd", }; +#define TPA6130A2_GAIN_MAX 0x3f +#define TPA6140A2_GAIN_MAX 0x1f + /* This struct is used to save the context */ struct tpa6130a2_data { struct mutex mutex; @@ -53,6 +56,8 @@ struct tpa6130a2_data { struct regulator_bulk_data supplies[TPA6130A2_NUM_SUPPLIES]; int power_gpio; unsigned char power_state; + enum tpa_model id; + int gain_limit; }; static int tpa6130a2_i2c_read(int reg) @@ -175,6 +180,40 @@ exit: return ret; } +static int tpa6130a2_info_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct tpa6130a2_data *data; + + BUG_ON(tpa6130a2_client == NULL); + data = i2c_get_clientdata(tpa6130a2_client); + + mutex_lock(&data->mutex); + switch (mc->reg) { + case TPA6130A2_REG_VOL_MUTE: + if (data->gain_limit != mc->max) + mc->max = data->gain_limit; + break; + default: + dev_err(&tpa6130a2_client->dev, + "Invalid register: 0x02%x\n", mc->reg); + goto out; + } + if (unlikely(mc->max == 1)) + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + else + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = mc->max; +out: + mutex_unlock(&data->mutex); + return 0; +} + static int tpa6130a2_get_reg(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -238,6 +277,15 @@ static int tpa6130a2_set_reg(struct snd_kcontrol *kcontrol, return 1; } +#define SOC_SINGLE_EXT_TLV_TPA(xname, xreg, xshift, xmax, xinvert, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ + SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .info = tpa6130a2_info_volsw, \ + .get = tpa6130a2_get_reg, .put = tpa6130a2_set_reg, \ + .private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) } + /* * TPA6130 volume. From -59.5 to 4 dB with increasing step size when going * down in gain. @@ -257,10 +305,22 @@ static const unsigned int tpa6130_tlv[] = { }; static const struct snd_kcontrol_new tpa6130a2_controls[] = { - SOC_SINGLE_EXT_TLV("TPA6130A2 Headphone Playback Volume", - TPA6130A2_REG_VOL_MUTE, 0, 0x3f, 0, - tpa6130a2_get_reg, tpa6130a2_set_reg, - tpa6130_tlv), + SOC_SINGLE_EXT_TLV_TPA("TPA6130A2 Headphone Playback Volume", + TPA6130A2_REG_VOL_MUTE, 0, TPA6130A2_GAIN_MAX, 0, + tpa6130_tlv), +}; + +static const unsigned int tpa6140_tlv[] = { + TLV_DB_RANGE_HEAD(3), + 0, 8, TLV_DB_SCALE_ITEM(-5900, 400, 0), + 9, 16, TLV_DB_SCALE_ITEM(-2500, 200, 0), + 17, 31, TLV_DB_SCALE_ITEM(-1000, 100, 0), +}; + +static const struct snd_kcontrol_new tpa6140a2_controls[] = { + SOC_SINGLE_EXT_TLV_TPA("TPA6140A2 Headphone Playback Volume", + TPA6130A2_REG_VOL_MUTE, 1, TPA6140A2_GAIN_MAX, 0, + tpa6140_tlv), }; /* @@ -368,13 +428,22 @@ static const struct snd_soc_dapm_route audio_map[] = { int tpa6130a2_add_controls(struct snd_soc_codec *codec) { + struct tpa6130a2_data *data; + + BUG_ON(tpa6130a2_client == NULL); + data = i2c_get_clientdata(tpa6130a2_client); + snd_soc_dapm_new_controls(codec, tpa6130a2_dapm_widgets, ARRAY_SIZE(tpa6130a2_dapm_widgets)); snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - return snd_soc_add_controls(codec, tpa6130a2_controls, - ARRAY_SIZE(tpa6130a2_controls)); + if (data->id == TPA6140A2) + return snd_soc_add_controls(codec, tpa6140a2_controls, + ARRAY_SIZE(tpa6140a2_controls)); + else + return snd_soc_add_controls(codec, tpa6130a2_controls, + ARRAY_SIZE(tpa6130a2_controls)); } EXPORT_SYMBOL_GPL(tpa6130a2_add_controls); @@ -407,6 +476,7 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client, pdata = client->dev.platform_data; data->power_gpio = pdata->power_gpio; + data->id = pdata->id; mutex_init(&data->mutex); @@ -425,20 +495,35 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client, gpio_direction_output(data->power_gpio, 0); } - switch (pdata->id) { + switch (data->id) { case TPA6130A2: for (i = 0; i < ARRAY_SIZE(data->supplies); i++) data->supplies[i].supply = tpa6130a2_supply_names[i]; + if (pdata->limit_gain > 0 && + pdata->limit_gain < TPA6130A2_GAIN_MAX) + data->gain_limit = pdata->limit_gain; + else + data->gain_limit = TPA6130A2_GAIN_MAX; break; case TPA6140A2: for (i = 0; i < ARRAY_SIZE(data->supplies); i++) data->supplies[i].supply = tpa6140a2_supply_names[i];; + if (pdata->limit_gain > 0 && + pdata->limit_gain < TPA6140A2_GAIN_MAX) + data->gain_limit = pdata->limit_gain; + else + data->gain_limit = TPA6140A2_GAIN_MAX; break; default: dev_warn(dev, "Unknown TPA model (%d). Assuming 6130A2\n", pdata->id); for (i = 0; i < ARRAY_SIZE(data->supplies); i++) data->supplies[i].supply = tpa6130a2_supply_names[i]; + if (pdata->limit_gain > 0 && + pdata->limit_gain < TPA6130A2_GAIN_MAX) + data->gain_limit = pdata->limit_gain; + else + data->gain_limit = TPA6130A2_GAIN_MAX; } ret = regulator_bulk_get(dev, ARRAY_SIZE(data->supplies), diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 2e025a3a2618..b717a03dfacf 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -123,6 +123,8 @@ struct twl4030_priv { struct snd_soc_codec codec; unsigned int codec_powered; + + /* reference counts of AIF/APLL users */ unsigned int apll_enabled; struct snd_pcm_substream *master_substream; @@ -259,22 +261,22 @@ static void twl4030_init_chip(struct snd_soc_codec *codec) static void twl4030_apll_enable(struct snd_soc_codec *codec, int enable) { struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); - int status; - - if (enable == twl4030->apll_enabled) - return; + int status = -1; - if (enable) - /* Enable PLL */ - status = twl4030_codec_enable_resource(TWL4030_CODEC_RES_APLL); - else - /* Disable PLL */ - status = twl4030_codec_disable_resource(TWL4030_CODEC_RES_APLL); + if (enable) { + twl4030->apll_enabled++; + if (twl4030->apll_enabled == 1) + status = twl4030_codec_enable_resource( + TWL4030_CODEC_RES_APLL); + } else { + twl4030->apll_enabled--; + if (!twl4030->apll_enabled) + status = twl4030_codec_disable_resource( + TWL4030_CODEC_RES_APLL); + } if (status >= 0) twl4030_write_reg_cache(codec, TWL4030_REG_APLL_CTL, status); - - twl4030->apll_enabled = enable; } static void twl4030_power_up(struct snd_soc_codec *codec) @@ -672,6 +674,31 @@ static int apll_event(struct snd_soc_dapm_widget *w, return 0; } +static int aif_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + u8 audio_if; + + audio_if = twl4030_read_reg_cache(w->codec, TWL4030_REG_AUDIO_IF); + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + /* Enable AIF */ + /* enable the PLL before we use it to clock the DAI */ + twl4030_apll_enable(w->codec, 1); + + twl4030_write(w->codec, TWL4030_REG_AUDIO_IF, + audio_if | TWL4030_AIF_EN); + break; + case SND_SOC_DAPM_POST_PMD: + /* disable the DAI before we stop it's source PLL */ + twl4030_write(w->codec, TWL4030_REG_AUDIO_IF, + audio_if & ~TWL4030_AIF_EN); + twl4030_apll_enable(w->codec, 0); + break; + } + return 0; +} + static void headset_ramp(struct snd_soc_codec *codec, int ramp) { struct snd_soc_device *socdev = codec->socdev; @@ -1167,8 +1194,6 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_INPUT("DIGIMIC1"), /* Outputs */ - SND_SOC_DAPM_OUTPUT("OUTL"), - SND_SOC_DAPM_OUTPUT("OUTR"), SND_SOC_DAPM_OUTPUT("EARPIECE"), SND_SOC_DAPM_OUTPUT("PREDRIVEL"), SND_SOC_DAPM_OUTPUT("PREDRIVER"), @@ -1180,6 +1205,11 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("HFR"), SND_SOC_DAPM_OUTPUT("VIBRA"), + /* AIF and APLL clocks for running DAIs (including loopback) */ + SND_SOC_DAPM_OUTPUT("Virtual HiFi OUT"), + SND_SOC_DAPM_INPUT("Virtual HiFi IN"), + SND_SOC_DAPM_OUTPUT("Virtual Voice OUT"), + /* DACs */ SND_SOC_DAPM_DAC("DAC Right1", "Right Front HiFi Playback", SND_SOC_NOPM, 0, 0), @@ -1243,7 +1273,8 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("APLL Enable", SND_SOC_NOPM, 0, 0, apll_event, SND_SOC_DAPM_PRE_PMU|SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_SUPPLY("AIF Enable", TWL4030_REG_AUDIO_IF, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("AIF Enable", SND_SOC_NOPM, 0, 0, aif_event, + SND_SOC_DAPM_PRE_PMU|SND_SOC_DAPM_POST_PMD), /* Output MIXER controls */ /* Earpiece */ @@ -1373,10 +1404,6 @@ static const struct snd_soc_dapm_route intercon[] = { {"Digital Voice Playback Mixer", NULL, "DAC Voice"}, /* Supply for the digital part (APLL) */ - {"Digital R1 Playback Mixer", NULL, "APLL Enable"}, - {"Digital L1 Playback Mixer", NULL, "APLL Enable"}, - {"Digital R2 Playback Mixer", NULL, "APLL Enable"}, - {"Digital L2 Playback Mixer", NULL, "APLL Enable"}, {"Digital Voice Playback Mixer", NULL, "APLL Enable"}, {"Digital R1 Playback Mixer", NULL, "AIF Enable"}, @@ -1450,8 +1477,14 @@ static const struct snd_soc_dapm_route intercon[] = { {"Vibra Mux", "AudioR2", "DAC Right2"}, /* outputs */ - {"OUTL", NULL, "Analog L2 Playback Mixer"}, - {"OUTR", NULL, "Analog R2 Playback Mixer"}, + /* Must be always connected (for AIF and APLL) */ + {"Virtual HiFi OUT", NULL, "Digital L1 Playback Mixer"}, + {"Virtual HiFi OUT", NULL, "Digital R1 Playback Mixer"}, + {"Virtual HiFi OUT", NULL, "Digital L2 Playback Mixer"}, + {"Virtual HiFi OUT", NULL, "Digital R2 Playback Mixer"}, + /* Must be always connected (for APLL) */ + {"Virtual Voice OUT", NULL, "Digital Voice Playback Mixer"}, + /* Physical outputs */ {"EARPIECE", NULL, "Earpiece PGA"}, {"PREDRIVEL", NULL, "PredriveL PGA"}, {"PREDRIVER", NULL, "PredriveR PGA"}, @@ -1465,6 +1498,12 @@ static const struct snd_soc_dapm_route intercon[] = { {"VIBRA", NULL, "Vibra Route"}, /* Capture path */ + /* Must be always connected (for AIF and APLL) */ + {"ADC Virtual Left1", NULL, "Virtual HiFi IN"}, + {"ADC Virtual Right1", NULL, "Virtual HiFi IN"}, + {"ADC Virtual Left2", NULL, "Virtual HiFi IN"}, + {"ADC Virtual Right2", NULL, "Virtual HiFi IN"}, + /* Physical inputs */ {"Analog Left", "Main Mic Capture Switch", "MAINMIC"}, {"Analog Left", "Headset Mic Capture Switch", "HSMIC"}, {"Analog Left", "AUXL Capture Switch", "AUXL"}, @@ -1497,11 +1536,6 @@ static const struct snd_soc_dapm_route intercon[] = { {"ADC Virtual Left2", NULL, "TX2 Capture Route"}, {"ADC Virtual Right2", NULL, "TX2 Capture Route"}, - {"ADC Virtual Left1", NULL, "APLL Enable"}, - {"ADC Virtual Right1", NULL, "APLL Enable"}, - {"ADC Virtual Left2", NULL, "APLL Enable"}, - {"ADC Virtual Right2", NULL, "APLL Enable"}, - {"ADC Virtual Left1", NULL, "AIF Enable"}, {"ADC Virtual Right1", NULL, "AIF Enable"}, {"ADC Virtual Left2", NULL, "AIF Enable"}, diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index f11963c21873..83be4a76d2bb 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -18,6 +18,16 @@ config SND_OMAP_SOC_N810 help Say Y if you want to add support for SoC audio on Nokia N810. +config SND_OMAP_SOC_RX51 + tristate "SoC Audio support for Nokia RX-51" + depends on SND_OMAP_SOC && MACH_NOKIA_RX51 + select OMAP_MCBSP + select SND_OMAP_SOC_MCBSP + select SND_SOC_TLV320AIC3X + help + Say Y if you want to add support for SoC audio on Nokia RX-51 + hardware. This is also known as Nokia N900 product. + config SND_OMAP_SOC_AMS_DELTA tristate "SoC Audio support for Amstrad E3 (Delta) videophone" depends on SND_OMAP_SOC && MACH_AMS_DELTA diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 0bc00ca14b37..3a75755f25e4 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -9,6 +9,7 @@ obj-$(CONFIG_SND_OMAP_SOC_MCPDM) += snd-soc-omap-mcpdm.o # OMAP Machine Support snd-soc-n810-objs := n810.o +snd-soc-rx51-objs := rx51.o snd-soc-ams-delta-objs := ams-delta.o snd-soc-osk5912-objs := osk5912.o snd-soc-overo-objs := overo.o @@ -22,6 +23,7 @@ snd-soc-zoom2-objs := zoom2.o snd-soc-igep0020-objs := igep0020.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o +obj-$(CONFIG_SND_OMAP_SOC_RX51) += snd-soc-rx51.o obj-$(CONFIG_SND_OMAP_SOC_AMS_DELTA) += snd-soc-ams-delta.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index de10f76baded..87ce842fa2e8 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -188,8 +188,6 @@ static int omap3pandora_out_init(struct snd_soc_codec *codec) int ret; /* All TWL4030 output pins are floating */ - snd_soc_dapm_nc_pin(codec, "OUTL"); - snd_soc_dapm_nc_pin(codec, "OUTR"); snd_soc_dapm_nc_pin(codec, "EARPIECE"); snd_soc_dapm_nc_pin(codec, "PREDRIVEL"); snd_soc_dapm_nc_pin(codec, "PREDRIVER"); diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c new file mode 100644 index 000000000000..47d831ef2dbb --- /dev/null +++ b/sound/soc/omap/rx51.c @@ -0,0 +1,294 @@ +/* + * rx51.c -- SoC audio for Nokia RX-51 + * + * Copyright (C) 2008 - 2009 Nokia Corporation + * + * Contact: Peter Ujfalusi <peter.ujfalusi@nokia.com> + * Eduardo Valentin <eduardo.valentin@nokia.com> + * Jarkko Nikula <jhnikula@gmail.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/delay.h> +#include <linux/gpio.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/tlv320aic3x.h" + +/* + * REVISIT: TWL4030 GPIO base in RX-51. Now statically defined to 192. This + * gpio is reserved in arch/arm/mach-omap2/board-rx51-peripherals.c + */ +#define RX51_SPEAKER_AMP_TWL_GPIO (192 + 7) + +static int rx51_spk_func; +static int rx51_dmic_func; + +static void rx51_ext_control(struct snd_soc_codec *codec) +{ + if (rx51_spk_func) + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + else + snd_soc_dapm_disable_pin(codec, "Ext Spk"); + if (rx51_dmic_func) + snd_soc_dapm_enable_pin(codec, "DMic"); + else + snd_soc_dapm_disable_pin(codec, "DMic"); + + snd_soc_dapm_sync(codec); +} + +static int rx51_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->socdev->card->codec; + + snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2); + rx51_ext_control(codec); + + return 0; +} + +static int rx51_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int err; + + /* Set codec DAI configuration */ + err = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (err < 0) + return err; + + /* Set cpu DAI configuration */ + err = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (err < 0) + return err; + + /* Set the codec system clock for DAC and ADC */ + return snd_soc_dai_set_sysclk(codec_dai, 0, 19200000, + SND_SOC_CLOCK_IN); +} + +static struct snd_soc_ops rx51_ops = { + .startup = rx51_startup, + .hw_params = rx51_hw_params, +}; + +static int rx51_get_spk(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = rx51_spk_func; + + return 0; +} + +static int rx51_set_spk(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + if (rx51_spk_func == ucontrol->value.integer.value[0]) + return 0; + + rx51_spk_func = ucontrol->value.integer.value[0]; + rx51_ext_control(codec); + + return 1; +} + +static int rx51_spk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + if (SND_SOC_DAPM_EVENT_ON(event)) + gpio_set_value(RX51_SPEAKER_AMP_TWL_GPIO, 1); + else + gpio_set_value(RX51_SPEAKER_AMP_TWL_GPIO, 0); + + return 0; +} + +static int rx51_get_input(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = rx51_dmic_func; + + return 0; +} + +static int rx51_set_input(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + if (rx51_dmic_func == ucontrol->value.integer.value[0]) + return 0; + + rx51_dmic_func = ucontrol->value.integer.value[0]; + rx51_ext_control(codec); + + return 1; +} + +static const struct snd_soc_dapm_widget aic34_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Ext Spk", rx51_spk_event), + SND_SOC_DAPM_MIC("DMic", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + {"Ext Spk", NULL, "HPLOUT"}, + {"Ext Spk", NULL, "HPROUT"}, + + {"DMic Rate 64", NULL, "Mic Bias 2V"}, + {"Mic Bias 2V", NULL, "DMic"}, +}; + +static const char *spk_function[] = {"Off", "On"}; +static const char *input_function[] = {"ADC", "Digital Mic"}; + +static const struct soc_enum rx51_enum[] = { + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(spk_function), spk_function), + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(input_function), input_function), +}; + +static const struct snd_kcontrol_new aic34_rx51_controls[] = { + SOC_ENUM_EXT("Speaker Function", rx51_enum[0], + rx51_get_spk, rx51_set_spk), + SOC_ENUM_EXT("Input Select", rx51_enum[1], + rx51_get_input, rx51_set_input), +}; + +static int rx51_aic34_init(struct snd_soc_codec *codec) +{ + int err; + + /* Set up NC codec pins */ + snd_soc_dapm_nc_pin(codec, "MIC3L"); + snd_soc_dapm_nc_pin(codec, "MIC3R"); + snd_soc_dapm_nc_pin(codec, "LINE1R"); + + /* Add RX-51 specific controls */ + err = snd_soc_add_controls(codec, aic34_rx51_controls, + ARRAY_SIZE(aic34_rx51_controls)); + if (err < 0) + return err; + + /* Add RX-51 specific widgets */ + snd_soc_dapm_new_controls(codec, aic34_dapm_widgets, + ARRAY_SIZE(aic34_dapm_widgets)); + + /* Set up RX-51 specific audio path audio_map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_sync(codec); + + return 0; +} + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link rx51_dai[] = { + { + .name = "TLV320AIC34", + .stream_name = "AIC34", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &aic3x_dai, + .init = rx51_aic34_init, + .ops = &rx51_ops, + }, +}; + +/* Audio private data */ +static struct aic3x_setup_data rx51_aic34_setup = { + .gpio_func[0] = AIC3X_GPIO1_FUNC_DISABLED, + .gpio_func[1] = AIC3X_GPIO2_FUNC_DIGITAL_MIC_INPUT, +}; + +/* Audio card */ +static struct snd_soc_card rx51_sound_card = { + .name = "RX-51", + .dai_link = rx51_dai, + .num_links = ARRAY_SIZE(rx51_dai), + .platform = &omap_soc_platform, +}; + +/* Audio subsystem */ +static struct snd_soc_device rx51_snd_devdata = { + .card = &rx51_sound_card, + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &rx51_aic34_setup, +}; + +static struct platform_device *rx51_snd_device; + +static int __init rx51_soc_init(void) +{ + int err; + + if (!machine_is_nokia_rx51()) + return -ENODEV; + + rx51_snd_device = platform_device_alloc("soc-audio", -1); + if (!rx51_snd_device) { + err = -ENOMEM; + goto err1; + } + + platform_set_drvdata(rx51_snd_device, &rx51_snd_devdata); + rx51_snd_devdata.dev = &rx51_snd_device->dev; + *(unsigned int *)rx51_dai[0].cpu_dai->private_data = 1; /* McBSP2 */ + + err = platform_device_add(rx51_snd_device); + if (err) + goto err2; + + return 0; +err2: + platform_device_put(rx51_snd_device); +err1: + + return err; +} + +static void __exit rx51_soc_exit(void) +{ + platform_device_unregister(rx51_snd_device); +} + +module_init(rx51_soc_init); +module_exit(rx51_soc_exit); + +MODULE_AUTHOR("Nokia Corporation"); +MODULE_DESCRIPTION("ALSA SoC Nokia RX-51"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/zoom2.c b/sound/soc/omap/zoom2.c index f90a2ac888cf..50a94ee76ecc 100644 --- a/sound/soc/omap/zoom2.c +++ b/sound/soc/omap/zoom2.c @@ -181,9 +181,6 @@ static int zoom2_twl4030_init(struct snd_soc_codec *codec) snd_soc_dapm_nc_pin(codec, "CARKITMIC"); snd_soc_dapm_nc_pin(codec, "DIGIMIC0"); snd_soc_dapm_nc_pin(codec, "DIGIMIC1"); - - snd_soc_dapm_nc_pin(codec, "OUTL"); - snd_soc_dapm_nc_pin(codec, "OUTR"); snd_soc_dapm_nc_pin(codec, "EARPIECE"); snd_soc_dapm_nc_pin(codec, "PREDRIVEL"); snd_soc_dapm_nc_pin(codec, "PREDRIVER"); |