diff options
Diffstat (limited to 'sound')
54 files changed, 509 insertions, 293 deletions
diff --git a/sound/aoa/codecs/onyx.c b/sound/aoa/codecs/onyx.c index 15500b9d2da0..84bb07d39a7f 100644 --- a/sound/aoa/codecs/onyx.c +++ b/sound/aoa/codecs/onyx.c @@ -47,7 +47,7 @@ MODULE_DESCRIPTION("pcm3052 (onyx) codec driver for snd-aoa"); struct onyx { /* cache registers 65 to 80, they are write-only! */ u8 cache[16]; - struct i2c_client i2c; + struct i2c_client *i2c; struct aoa_codec codec; u32 initialised:1, spdif_locked:1, @@ -72,7 +72,7 @@ static int onyx_read_register(struct onyx *onyx, u8 reg, u8 *value) *value = onyx->cache[reg-FIRSTREGISTER]; return 0; } - v = i2c_smbus_read_byte_data(&onyx->i2c, reg); + v = i2c_smbus_read_byte_data(onyx->i2c, reg); if (v < 0) return -1; *value = (u8)v; @@ -84,7 +84,7 @@ static int onyx_write_register(struct onyx *onyx, u8 reg, u8 value) { int result; - result = i2c_smbus_write_byte_data(&onyx->i2c, reg, value); + result = i2c_smbus_write_byte_data(onyx->i2c, reg, value); if (!result) onyx->cache[reg-FIRSTREGISTER] = value; return result; @@ -996,12 +996,45 @@ static void onyx_exit_codec(struct aoa_codec *codec) onyx->codec.soundbus_dev->detach_codec(onyx->codec.soundbus_dev, onyx); } -static struct i2c_driver onyx_driver; - static int onyx_create(struct i2c_adapter *adapter, struct device_node *node, int addr) { + struct i2c_board_info info; + struct i2c_client *client; + + memset(&info, 0, sizeof(struct i2c_board_info)); + strlcpy(info.type, "aoa_codec_onyx", I2C_NAME_SIZE); + info.addr = addr; + info.platform_data = node; + client = i2c_new_device(adapter, &info); + if (!client) + return -ENODEV; + + /* + * We know the driver is already loaded, so the device should be + * already bound. If not it means binding failed, which suggests + * the device doesn't really exist and should be deleted. + * Ideally this would be replaced by better checks _before_ + * instantiating the device. + */ + if (!client->driver) { + i2c_unregister_device(client); + return -ENODEV; + } + + /* + * Let i2c-core delete that device on driver removal. + * This is safe because i2c-core holds the core_lock mutex for us. + */ + list_add_tail(&client->detected, &client->driver->clients); + return 0; +} + +static int onyx_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct device_node *node = client->dev.platform_data; struct onyx *onyx; u8 dummy; @@ -1011,20 +1044,12 @@ static int onyx_create(struct i2c_adapter *adapter, return -ENOMEM; mutex_init(&onyx->mutex); - onyx->i2c.driver = &onyx_driver; - onyx->i2c.adapter = adapter; - onyx->i2c.addr = addr & 0x7f; - strlcpy(onyx->i2c.name, "onyx audio codec", I2C_NAME_SIZE); - - if (i2c_attach_client(&onyx->i2c)) { - printk(KERN_ERR PFX "failed to attach to i2c\n"); - goto fail; - } + onyx->i2c = client; + i2c_set_clientdata(client, onyx); /* we try to read from register ONYX_REG_CONTROL * to check if the codec is present */ if (onyx_read_register(onyx, ONYX_REG_CONTROL, &dummy) != 0) { - i2c_detach_client(&onyx->i2c); printk(KERN_ERR PFX "failed to read control register\n"); goto fail; } @@ -1036,14 +1061,14 @@ static int onyx_create(struct i2c_adapter *adapter, onyx->codec.node = of_node_get(node); if (aoa_codec_register(&onyx->codec)) { - i2c_detach_client(&onyx->i2c); goto fail; } printk(KERN_DEBUG PFX "created and attached onyx instance\n"); return 0; fail: + i2c_set_clientdata(client, NULL); kfree(onyx); - return -EINVAL; + return -ENODEV; } static int onyx_i2c_attach(struct i2c_adapter *adapter) @@ -1080,28 +1105,33 @@ static int onyx_i2c_attach(struct i2c_adapter *adapter) return onyx_create(adapter, NULL, 0x47); } -static int onyx_i2c_detach(struct i2c_client *client) +static int onyx_i2c_remove(struct i2c_client *client) { - struct onyx *onyx = container_of(client, struct onyx, i2c); - int err; + struct onyx *onyx = i2c_get_clientdata(client); - if ((err = i2c_detach_client(client))) - return err; aoa_codec_unregister(&onyx->codec); of_node_put(onyx->codec.node); if (onyx->codec_info) kfree(onyx->codec_info); + i2c_set_clientdata(client, onyx); kfree(onyx); return 0; } +static const struct i2c_device_id onyx_i2c_id[] = { + { "aoa_codec_onyx", 0 }, + { } +}; + static struct i2c_driver onyx_driver = { .driver = { .name = "aoa_codec_onyx", .owner = THIS_MODULE, }, .attach_adapter = onyx_i2c_attach, - .detach_client = onyx_i2c_detach, + .probe = onyx_i2c_probe, + .remove = onyx_i2c_remove, + .id_table = onyx_i2c_id, }; static int __init onyx_init(void) diff --git a/sound/aoa/codecs/tas.c b/sound/aoa/codecs/tas.c index 008e0f85097d..f0ebc971c686 100644 --- a/sound/aoa/codecs/tas.c +++ b/sound/aoa/codecs/tas.c @@ -82,7 +82,7 @@ MODULE_DESCRIPTION("tas codec driver for snd-aoa"); struct tas { struct aoa_codec codec; - struct i2c_client i2c; + struct i2c_client *i2c; u32 mute_l:1, mute_r:1 , controls_created:1 , drc_enabled:1, @@ -108,9 +108,9 @@ static struct tas *codec_to_tas(struct aoa_codec *codec) static inline int tas_write_reg(struct tas *tas, u8 reg, u8 len, u8 *data) { if (len == 1) - return i2c_smbus_write_byte_data(&tas->i2c, reg, *data); + return i2c_smbus_write_byte_data(tas->i2c, reg, *data); else - return i2c_smbus_write_i2c_block_data(&tas->i2c, reg, len, data); + return i2c_smbus_write_i2c_block_data(tas->i2c, reg, len, data); } static void tas3004_set_drc(struct tas *tas) @@ -882,12 +882,34 @@ static void tas_exit_codec(struct aoa_codec *codec) } -static struct i2c_driver tas_driver; - static int tas_create(struct i2c_adapter *adapter, struct device_node *node, int addr) { + struct i2c_board_info info; + struct i2c_client *client; + + memset(&info, 0, sizeof(struct i2c_board_info)); + strlcpy(info.type, "aoa_codec_tas", I2C_NAME_SIZE); + info.addr = addr; + info.platform_data = node; + + client = i2c_new_device(adapter, &info); + if (!client) + return -ENODEV; + + /* + * Let i2c-core delete that device on driver removal. + * This is safe because i2c-core holds the core_lock mutex for us. + */ + list_add_tail(&client->detected, &client->driver->clients); + return 0; +} + +static int tas_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct device_node *node = client->dev.platform_data; struct tas *tas; tas = kzalloc(sizeof(struct tas), GFP_KERNEL); @@ -896,17 +918,11 @@ static int tas_create(struct i2c_adapter *adapter, return -ENOMEM; mutex_init(&tas->mtx); - tas->i2c.driver = &tas_driver; - tas->i2c.adapter = adapter; - tas->i2c.addr = addr; + tas->i2c = client; + i2c_set_clientdata(client, tas); + /* seems that half is a saner default */ tas->drc_range = TAS3004_DRC_MAX / 2; - strlcpy(tas->i2c.name, "tas audio codec", I2C_NAME_SIZE); - - if (i2c_attach_client(&tas->i2c)) { - printk(KERN_ERR PFX "failed to attach to i2c\n"); - goto fail; - } strlcpy(tas->codec.name, "tas", MAX_CODEC_NAME_LEN); tas->codec.owner = THIS_MODULE; @@ -915,14 +931,12 @@ static int tas_create(struct i2c_adapter *adapter, tas->codec.node = of_node_get(node); if (aoa_codec_register(&tas->codec)) { - goto detach; + goto fail; } printk(KERN_DEBUG "snd-aoa-codec-tas: tas found, addr 0x%02x on %s\n", - addr, node->full_name); + (unsigned int)client->addr, node->full_name); return 0; - detach: - i2c_detach_client(&tas->i2c); fail: mutex_destroy(&tas->mtx); kfree(tas); @@ -970,14 +984,11 @@ static int tas_i2c_attach(struct i2c_adapter *adapter) return -ENODEV; } -static int tas_i2c_detach(struct i2c_client *client) +static int tas_i2c_remove(struct i2c_client *client) { - struct tas *tas = container_of(client, struct tas, i2c); - int err; + struct tas *tas = i2c_get_clientdata(client); u8 tmp = TAS_ACR_ANALOG_PDOWN; - if ((err = i2c_detach_client(client))) - return err; aoa_codec_unregister(&tas->codec); of_node_put(tas->codec.node); @@ -989,13 +1000,20 @@ static int tas_i2c_detach(struct i2c_client *client) return 0; } +static const struct i2c_device_id tas_i2c_id[] = { + { "aoa_codec_tas", 0 }, + { } +}; + static struct i2c_driver tas_driver = { .driver = { .name = "aoa_codec_tas", .owner = THIS_MODULE, }, .attach_adapter = tas_i2c_attach, - .detach_client = tas_i2c_detach, + .probe = tas_i2c_probe, + .remove = tas_i2c_remove, + .id_table = tas_i2c_id, }; static int __init tas_init(void) diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index 7fbd68fab944..5c48e36038f2 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -1074,7 +1074,7 @@ static unsigned int __devinit aaci_size_fifo(struct aaci *aaci) return i; } -static int __devinit aaci_probe(struct amba_device *dev, void *id) +static int __devinit aaci_probe(struct amba_device *dev, struct amba_id *id) { struct aaci *aaci; int ret, i; diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index 0afd1a8226fb..6fdca97186e7 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -65,7 +65,7 @@ static void set_resetgpio_mode(int resetgpio_action) switch (resetgpio_action) { case RESETGPIO_NORMAL_ALTFUNC: if (reset_gpio == 113) - mode = 113 | GPIO_OUT | GPIO_DFLT_LOW; + mode = 113 | GPIO_ALT_FN_2_OUT; if (reset_gpio == 95) mode = 95 | GPIO_ALT_FN_1_OUT; break; @@ -364,7 +364,7 @@ EXPORT_SYMBOL_GPL(pxa2xx_ac97_hw_resume); int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev) { int ret; - struct pxa2xx_ac97_platform_data *pdata = dev->dev.platform_data; + pxa2xx_audio_ops_t *pdata = dev->dev.platform_data; if (pdata) { switch (pdata->reset_gpio) { diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 63d088f2265f..a2a792c18c40 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -249,6 +249,12 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) new_hw_ptr = hw_base + pos; } } + /* Skip the jiffies check for hardwares with BATCH flag. + * Such hardware usually just increases the position at each IRQ, + * thus it can't give any strange position. + */ + if (runtime->hw.info & SNDRV_PCM_INFO_BATCH) + goto no_jiffies_check; hdelta = new_hw_ptr - old_hw_ptr; jdelta = jiffies - runtime->hw_ptr_jiffies; if (((hdelta * HZ) / runtime->rate) > jdelta + HZ/100) { @@ -272,6 +278,7 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) hw_base -= hw_base % runtime->buffer_size; delta = 0; } + no_jiffies_check: if (delta > runtime->period_size + runtime->period_size / 2) { hw_ptr_error(substream, "Lost interrupts? " diff --git a/sound/drivers/pcsp/pcsp_mixer.c b/sound/drivers/pcsp/pcsp_mixer.c index caeb0f57fcca..199b03377142 100644 --- a/sound/drivers/pcsp/pcsp_mixer.c +++ b/sound/drivers/pcsp/pcsp_mixer.c @@ -50,8 +50,8 @@ static int pcsp_treble_info(struct snd_kcontrol *kcontrol, uinfo->value.enumerated.items = chip->max_treble + 1; if (uinfo->value.enumerated.item > chip->max_treble) uinfo->value.enumerated.item = chip->max_treble; - sprintf(uinfo->value.enumerated.name, "%d", - PCSP_CALC_RATE(uinfo->value.enumerated.item)); + sprintf(uinfo->value.enumerated.name, "%lu", + (unsigned long)PCSP_CALC_RATE(uinfo->value.enumerated.item)); return 0; } diff --git a/sound/drivers/serial-u16550.c b/sound/drivers/serial-u16550.c index b2b6d50c9425..a25fb7b1f441 100644 --- a/sound/drivers/serial-u16550.c +++ b/sound/drivers/serial-u16550.c @@ -963,16 +963,11 @@ static int __devinit snd_serial_probe(struct platform_device *devptr) if (err < 0) goto _err; - sprintf(card->longname, "%s at 0x%lx, irq %d speed %d div %d outs %d ins %d adaptor %s droponfull %d", + sprintf(card->longname, "%s [%s] at %#lx, irq %d", card->shortname, - uart->base, - uart->irq, - uart->speed, - (int)uart->divisor, - outs[dev], - ins[dev], adaptor_names[uart->adaptor], - uart->drop_on_full); + uart->base, + uart->irq); snd_card_set_dev(card, &devptr->dev); diff --git a/sound/isa/msnd/msnd.c b/sound/isa/msnd/msnd.c index 906454413ed2..3a1526ae1729 100644 --- a/sound/isa/msnd/msnd.c +++ b/sound/isa/msnd/msnd.c @@ -438,7 +438,8 @@ static void snd_msnd_capture_reset_queue(struct snd_msnd *chip, static struct snd_pcm_hardware snd_msnd_playback = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_MMAP_VALID, + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BATCH, .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE, .rates = SNDRV_PCM_RATE_8000_48000, .rate_min = 8000, @@ -456,7 +457,8 @@ static struct snd_pcm_hardware snd_msnd_playback = { static struct snd_pcm_hardware snd_msnd_capture = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_MMAP_VALID, + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BATCH, .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE, .rates = SNDRV_PCM_RATE_8000_48000, .rate_min = 8000, diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 97ee127ac33d..78288dbfc17a 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -2122,7 +2122,7 @@ int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template, } /* nothing should be in powerdown mode */ snd_ac97_write_cache(ac97, AC97_GENERAL_PURPOSE, 0); - end_time = jiffies + msecs_to_jiffies(100); + end_time = jiffies + msecs_to_jiffies(120); do { if ((snd_ac97_read(ac97, AC97_POWERDOWN) & 0x0f) == 0x0f) goto __ready_ok; diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 81bc93e5f1e3..7337abdbe4e3 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -958,10 +958,13 @@ static int patch_sigmatel_stac9708_3d(struct snd_ac97 * ac97) } static const struct snd_kcontrol_new snd_ac97_sigmatel_4speaker = -AC97_SINGLE("Sigmatel 4-Speaker Stereo Playback Switch", AC97_SIGMATEL_DAC2INVERT, 2, 1, 0); +AC97_SINGLE("Sigmatel 4-Speaker Stereo Playback Switch", + AC97_SIGMATEL_DAC2INVERT, 2, 1, 0); +/* "Sigmatel " removed due to excessive name length: */ static const struct snd_kcontrol_new snd_ac97_sigmatel_phaseinvert = -AC97_SINGLE("Sigmatel Surround Phase Inversion Playback Switch", AC97_SIGMATEL_DAC2INVERT, 3, 1, 0); +AC97_SINGLE("Surround Phase Inversion Playback Switch", + AC97_SIGMATEL_DAC2INVERT, 3, 1, 0); static const struct snd_kcontrol_new snd_ac97_sigmatel_controls[] = { AC97_SINGLE("Sigmatel DAC 6dB Attenuate", AC97_SIGMATEL_ANALOG, 1, 1, 0), diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index 9ce8548c03e4..71515ddb4593 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -1393,6 +1393,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = { .name = "HP nx6125", .type = AC97_TUNE_MUTE_LED }, + { + .subvendor = 0x103c, + .subdevice = 0x3091, + .name = "unknown HP", + .type = AC97_TUNE_MUTE_LED + }, { } /* terminator */ }; diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c index a299340519df..ce3f2e90f4d7 100644 --- a/sound/pci/bt87x.c +++ b/sound/pci/bt87x.c @@ -349,7 +349,8 @@ static struct snd_pcm_hardware snd_bt87x_digital_hw = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP_VALID, + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BATCH, .formats = SNDRV_PCM_FMTBIT_S16_LE, .rates = 0, /* set at runtime */ .channels_min = 2, @@ -365,7 +366,8 @@ static struct snd_pcm_hardware snd_bt87x_analog_hw = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP_VALID, + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BATCH, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8, .rates = SNDRV_PCM_RATE_KNOT, .rate_min = ANALOG_CLOCK / CLOCK_DIV_MAX, diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c index ad2888705d2a..c111efe61c3c 100644 --- a/sound/pci/ca0106/ca0106_mixer.c +++ b/sound/pci/ca0106/ca0106_mixer.c @@ -800,7 +800,7 @@ int __devinit snd_ca0106_mixer(struct snd_ca0106 *emu) "Capture Volume", "External Amplifier", "Sigmatel 4-Speaker Stereo Playback Switch", - "Sigmatel Surround Phase Inversion Playback ", + "Surround Phase Inversion Playback Switch", NULL }; static char *ca0106_rename_ctls[] = { diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index c7899c32aba1..449fe02f666e 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -3014,7 +3014,7 @@ static int __devinit snd_cmipci_create(struct snd_card *card, struct pci_dev *pc .dev_free = snd_cmipci_dev_free, }; unsigned int val; - long iomidi; + long iomidi = 0; int integrated_midi = 0; char modelstr[16]; int pcm_index, pcm_spdif_index; diff --git a/sound/pci/echoaudio/indigodjx.c b/sound/pci/echoaudio/indigodjx.c index 3482ef69f491..2e44316530a2 100644 --- a/sound/pci/echoaudio/indigodjx.c +++ b/sound/pci/echoaudio/indigodjx.c @@ -88,6 +88,7 @@ static struct snd_pcm_hardware pcm_hardware_skel = { .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_64000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, .rate_min = 32000, diff --git a/sound/pci/echoaudio/indigoiox.c b/sound/pci/echoaudio/indigoiox.c index aebee27a40ff..eb3819f9654a 100644 --- a/sound/pci/echoaudio/indigoiox.c +++ b/sound/pci/echoaudio/indigoiox.c @@ -89,6 +89,7 @@ static struct snd_pcm_hardware pcm_hardware_skel = { .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_64000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, .rate_min = 32000, diff --git a/sound/pci/emu10k1/io.c b/sound/pci/emu10k1/io.c index 4bfc31d1b281..c1a5aa15af8f 100644 --- a/sound/pci/emu10k1/io.c +++ b/sound/pci/emu10k1/io.c @@ -490,7 +490,7 @@ void snd_emu10k1_wait(struct snd_emu10k1 *emu, unsigned int wait) if (newtime != curtime) break; } - if (count >= 16384) + if (count > 16384) break; curtime = newtime; } diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 5dced5b79387..173bebf9f51d 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -1854,6 +1854,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = { }, { .subvendor = 0x1028, + .subdevice = 0x016a, + .name = "Dell Inspiron 8600", /* STAC9750/51 */ + .type = AC97_TUNE_HP_ONLY + }, + { + .subvendor = 0x1028, .subdevice = 0x0186, .name = "Dell Latitude D810", /* cf. Malone #41015 */ .type = AC97_TUNE_HP_MUTE_LED @@ -1896,12 +1902,6 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = { }, { .subvendor = 0x103c, - .subdevice = 0x0934, - .name = "HP nx8220", - .type = AC97_TUNE_MUTE_LED - }, - { - .subvendor = 0x103c, .subdevice = 0x129d, .name = "HP xw8000", .type = AC97_TUNE_HP_ONLY @@ -2751,11 +2751,12 @@ static void __devinit intel8x0_measure_ac97_clock(struct intel8x0 *chip) if (pos == 0) { snd_printk(KERN_ERR "intel8x0: measure - unreliable DMA position..\n"); __retry: - if (attempt < 2) { + if (attempt < 3) { + msleep(300); attempt++; goto __again; } - return; + goto __end; } pos /= 4; @@ -2782,6 +2783,7 @@ static void __devinit intel8x0_measure_ac97_clock(struct intel8x0 *chip) else if (pos < 47500 || pos > 48500) /* not 48000Hz, tuning the clock.. */ chip->ac97_bus->clock = (chip->ac97_bus->clock * 48000) / pos; + __end: printk(KERN_INFO "intel8x0: clocking to %d\n", chip->ac97_bus->clock); snd_ac97_update_power(chip->ac97[0], AC97_PCM_FRONT_DAC_RATE, 0); } diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index 8b79969034be..7cc38a11e997 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -1238,7 +1238,8 @@ static struct snd_pcm_hardware snd_korg1212_playback_info = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_INTERLEAVED), + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BATCH), .formats = SNDRV_PCM_FMTBIT_S16_LE, .rates = (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000), @@ -1258,7 +1259,8 @@ static struct snd_pcm_hardware snd_korg1212_capture_info = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_INTERLEAVED), + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BATCH), .formats = SNDRV_PCM_FMTBIT_S16_LE, .rates = (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000), diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index 6f1034417a02..e51a5ef1954d 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -889,7 +889,7 @@ static int sendcmd(struct cmdif *cif, u32 flags, u32 cmd, u32 parm, spin_lock_irqsave(&cif->lock, irqflags); while (i++ < CMDIF_TIMEOUT && !IS_READY(cif->hwport)) udelay(10); - if (i >= CMDIF_TIMEOUT) { + if (i > CMDIF_TIMEOUT) { err = -EBUSY; goto errout; } @@ -907,8 +907,10 @@ static int sendcmd(struct cmdif *cif, u32 flags, u32 cmd, u32 parm, WRITE_PORT_ULONG(cmdport->data1, cmd); /* write cmd */ if ((flags & RESP) && ret) { while (!IS_DATF(cmdport) && - time++ < CMDIF_TIMEOUT) + time < CMDIF_TIMEOUT) { udelay(10); + time++; + } if (time < CMDIF_TIMEOUT) { /* read response */ ret->retlongs[0] = READ_PORT_ULONG(cmdport->data1); @@ -1454,7 +1456,7 @@ static int snd_riptide_trigger(struct snd_pcm_substream *substream, int cmd) SEND_GPOS(cif, 0, data->id, &rptr); udelay(1); } while (i != rptr.retlongs[1] && j++ < MAX_WRITE_RETRY); - if (j >= MAX_WRITE_RETRY) + if (j > MAX_WRITE_RETRY) snd_printk(KERN_ERR "Riptide: Could not stop stream!"); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: @@ -1783,7 +1785,7 @@ snd_riptide_codec_write(struct snd_ac97 *ac97, unsigned short reg, SEND_SACR(cif, val, reg); SEND_RACR(cif, reg, &rptr); } while (rptr.retwords[1] != val && i++ < MAX_WRITE_RETRY); - if (i == MAX_WRITE_RETRY) + if (i > MAX_WRITE_RETRY) snd_printdd("Write AC97 reg failed\n"); } diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 809b233dd4a3..1ef58c51c213 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -1687,7 +1687,7 @@ static int snd_via8233_pcmdxs_volume_put(struct snd_kcontrol *kcontrol, return change; } -static const DECLARE_TLV_DB_SCALE(db_scale_dxs, -9450, 150, 1); +static const DECLARE_TLV_DB_SCALE(db_scale_dxs, -4650, 150, 1); static struct snd_kcontrol_new snd_via8233_pcmdxs_volume_control __devinitdata = { .name = "PCM Playback Volume", diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c index 01066c95580e..d057e6489643 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c @@ -240,7 +240,8 @@ static int pdacf_pcm_prepare(struct snd_pcm_substream *subs) static struct snd_pcm_hardware pdacf_pcm_capture_hw = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME | - SNDRV_PCM_INFO_MMAP_VALID), + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BATCH), .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE, diff --git a/sound/ppc/keywest.c b/sound/ppc/keywest.c index 6ff99ed77516..a5afb2682e7f 100644 --- a/sound/ppc/keywest.c +++ b/sound/ppc/keywest.c @@ -33,26 +33,25 @@ static struct pmac_keywest *keywest_ctx; -static int keywest_attach_adapter(struct i2c_adapter *adapter); -static int keywest_detach_client(struct i2c_client *client); - -struct i2c_driver keywest_driver = { - .driver = { - .name = "PMac Keywest Audio", - }, - .attach_adapter = &keywest_attach_adapter, - .detach_client = &keywest_detach_client, -}; - - #ifndef i2c_device_name #define i2c_device_name(x) ((x)->name) #endif +static int keywest_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + i2c_set_clientdata(client, keywest_ctx); + return 0; +} + +/* + * This is kind of a hack, best would be to turn powermac to fixed i2c + * bus numbers and declare the sound device as part of platform + * initialization + */ static int keywest_attach_adapter(struct i2c_adapter *adapter) { - int err; - struct i2c_client *new_client; + struct i2c_board_info info; if (! keywest_ctx) return -EINVAL; @@ -60,46 +59,47 @@ static int keywest_attach_adapter(struct i2c_adapter *adapter) if (strncmp(i2c_device_name(adapter), "mac-io", 6)) return 0; /* ignored */ - new_client = kzalloc(sizeof(struct i2c_client), GFP_KERNEL); - if (! new_client) - return -ENOMEM; - - new_client->addr = keywest_ctx->addr; - i2c_set_clientdata(new_client, keywest_ctx); - new_client->adapter = adapter; - new_client->driver = &keywest_driver; - new_client->flags = 0; - - strcpy(i2c_device_name(new_client), keywest_ctx->name); - keywest_ctx->client = new_client; + memset(&info, 0, sizeof(struct i2c_board_info)); + strlcpy(info.type, "keywest", I2C_NAME_SIZE); + info.addr = keywest_ctx->addr; + keywest_ctx->client = i2c_new_device(adapter, &info); - /* Tell the i2c layer a new client has arrived */ - if (i2c_attach_client(new_client)) { - snd_printk(KERN_ERR "tumbler: cannot attach i2c client\n"); - err = -ENODEV; - goto __err; - } - + /* + * Let i2c-core delete that device on driver removal. + * This is safe because i2c-core holds the core_lock mutex for us. + */ + list_add_tail(&keywest_ctx->client->detected, + &keywest_ctx->client->driver->clients); return 0; - - __err: - kfree(new_client); - keywest_ctx->client = NULL; - return err; } -static int keywest_detach_client(struct i2c_client *client) +static int keywest_remove(struct i2c_client *client) { + i2c_set_clientdata(client, NULL); if (! keywest_ctx) return 0; if (client == keywest_ctx->client) keywest_ctx->client = NULL; - i2c_detach_client(client); - kfree(client); return 0; } + +static const struct i2c_device_id keywest_i2c_id[] = { + { "keywest", 0 }, + { } +}; + +struct i2c_driver keywest_driver = { + .driver = { + .name = "PMac Keywest Audio", + }, + .attach_adapter = keywest_attach_adapter, + .probe = keywest_probe, + .remove = keywest_remove, + .id_table = keywest_i2c_id, +}; + /* exported */ void snd_pmac_keywest_cleanup(struct pmac_keywest *i2c) { diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index 30490a259148..594c6c5b7838 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -82,7 +82,7 @@ static struct au1xpsc_audio_dmadata *au1xpsc_audio_pcmdma[2]; /* PCM hardware DMA capabilities - platform specific */ static const struct snd_pcm_hardware au1xpsc_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_INTERLEAVED, + SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BATCH, .formats = AU1XPSC_PCM_FMTS, .period_bytes_min = AU1XPSC_PERIOD_MIN_BYTES, .period_bytes_max = 4096 * 1024 - 1, diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 030d2454725f..f2653803ede8 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -56,7 +56,6 @@ obj-$(CONFIG_SND_SOC_WM8900) += snd-soc-wm8900.o obj-$(CONFIG_SND_SOC_WM8903) += snd-soc-wm8903.o obj-$(CONFIG_SND_SOC_WM8971) += snd-soc-wm8971.o obj-$(CONFIG_SND_SOC_WM8990) += snd-soc-wm8990.o -obj-$(CONFIG_SND_SOC_WM8991) += snd-soc-wm8991.o obj-$(CONFIG_SND_SOC_WM9705) += snd-soc-wm9705.o obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 921b205de28a..df7c8c281d2f 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -836,6 +836,12 @@ static DECLARE_TLV_DB_SCALE(analog_tlv, -2400, 200, 0); static DECLARE_TLV_DB_SCALE(output_tvl, -1200, 600, 1); /* + * Gain control for earpiece amplifier + * 0 dB to 12 dB in 6 dB steps (mute instead of -6) + */ +static DECLARE_TLV_DB_SCALE(output_ear_tvl, -600, 600, 1); + +/* * Capture gain after the ADCs * from 0 dB to 31 dB in 1 dB steps */ @@ -900,7 +906,7 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = { 4, 3, 0, output_tvl), SOC_SINGLE_TLV_TWL4030("Earpiece Playback Volume", - TWL4030_REG_EAR_CTL, 4, 3, 0, output_tvl), + TWL4030_REG_EAR_CTL, 4, 3, 0, output_ear_tvl), /* Common capture gain controls */ SOC_DOUBLE_R_TLV("TX1 Digital Capture Volume", diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 3b1d0993bed9..0275321ff8ab 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -968,7 +968,7 @@ static int wm8350_pcm_trigger(struct snd_pcm_substream *substream, * required for LRC in master mode. The DACs or ADCs need a * valid audio path i.e. pin -> ADC or DAC -> pin before * the LRC will be enabled in master mode. */ - if (!master && cmd != SNDRV_PCM_TRIGGER_START) + if (!master || cmd != SNDRV_PCM_TRIGGER_START) return 0; if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 442ea6f160fc..9f6be3d31ac0 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -268,9 +268,11 @@ static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1); static int wm8580_out_vu(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - int reg = kcontrol->private_value & 0xff; - int reg2 = (kcontrol->private_value >> 24) & 0xff; + unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; int ret; u16 val; @@ -292,15 +294,17 @@ static int wm8580_out_vu(struct snd_kcontrol *kcontrol, return 0; } -#define SOC_WM8580_OUT_DOUBLE_R_TLV(xname, reg_left, reg_right, shift, max, invert, tlv_array) \ +#define SOC_WM8580_OUT_DOUBLE_R_TLV(xname, reg_left, reg_right, 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 = snd_soc_info_volsw_2r, \ .get = snd_soc_get_volsw_2r, .put = wm8580_out_vu, \ - .private_value = (reg_left) | ((shift) << 8) | \ - ((max) << 12) | ((invert) << 20) | ((reg_right) << 24) } + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = reg_left, .rreg = reg_right, .shift = xshift, \ + .max = xmax, .invert = xinvert} } static const struct snd_kcontrol_new wm8580_snd_controls[] = { SOC_WM8580_OUT_DOUBLE_R_TLV("DAC1 Playback Volume", @@ -522,7 +526,7 @@ static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai, reg = wm8580_read(codec, WM8580_PLLA4 + offset); reg &= ~0x3f; reg |= pll_div.prescale | pll_div.postscale << 1 | - pll_div.freqmode << 4; + pll_div.freqmode << 3; wm8580_write(codec, WM8580_PLLA4 + offset, reg); diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index c518c3e5aa3f..40cd274eb1ef 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -729,7 +729,7 @@ SND_SOC_DAPM_MIXER_E("INMIXL", WM8990_INTDRIVBITS, WM8990_INMIXL_PWR_BIT, 0, inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), /* AINLMUX */ -SND_SOC_DAPM_MUX_E("AILNMUX", WM8990_INTDRIVBITS, WM8990_AINLMUX_PWR_BIT, 0, +SND_SOC_DAPM_MUX_E("AINLMUX", WM8990_INTDRIVBITS, WM8990_AINLMUX_PWR_BIT, 0, &wm8990_dapm_ainlmux_controls, inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), @@ -740,7 +740,7 @@ SND_SOC_DAPM_MIXER_E("INMIXR", WM8990_INTDRIVBITS, WM8990_INMIXR_PWR_BIT, 0, inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), /* AINRMUX */ -SND_SOC_DAPM_MUX_E("AIRNMUX", WM8990_INTDRIVBITS, WM8990_AINRMUX_PWR_BIT, 0, +SND_SOC_DAPM_MUX_E("AINRMUX", WM8990_INTDRIVBITS, WM8990_AINRMUX_PWR_BIT, 0, &wm8990_dapm_ainrmux_controls, inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), @@ -848,40 +848,40 @@ static const struct snd_soc_dapm_route audio_map[] = { {"LIN12 PGA", "LIN2 Switch", "LIN2"}, /* LIN34 PGA */ {"LIN34 PGA", "LIN3 Switch", "LIN3"}, - {"LIN34 PGA", "LIN4 Switch", "LIN4"}, + {"LIN34 PGA", "LIN4 Switch", "LIN4/RXN"}, /* INMIXL */ {"INMIXL", "Record Left Volume", "LOMIX"}, {"INMIXL", "LIN2 Volume", "LIN2"}, {"INMIXL", "LINPGA12 Switch", "LIN12 PGA"}, {"INMIXL", "LINPGA34 Switch", "LIN34 PGA"}, - /* AILNMUX */ - {"AILNMUX", "INMIXL Mix", "INMIXL"}, - {"AILNMUX", "DIFFINL Mix", "LIN12PGA"}, - {"AILNMUX", "DIFFINL Mix", "LIN34PGA"}, - {"AILNMUX", "RXVOICE Mix", "LIN4/RXN"}, - {"AILNMUX", "RXVOICE Mix", "RIN4/RXP"}, + /* AINLMUX */ + {"AINLMUX", "INMIXL Mix", "INMIXL"}, + {"AINLMUX", "DIFFINL Mix", "LIN12 PGA"}, + {"AINLMUX", "DIFFINL Mix", "LIN34 PGA"}, + {"AINLMUX", "RXVOICE Mix", "LIN4/RXN"}, + {"AINLMUX", "RXVOICE Mix", "RIN4/RXP"}, /* ADC */ - {"Left ADC", NULL, "AILNMUX"}, + {"Left ADC", NULL, "AINLMUX"}, /* RIN12 PGA */ {"RIN12 PGA", "RIN1 Switch", "RIN1"}, {"RIN12 PGA", "RIN2 Switch", "RIN2"}, /* RIN34 PGA */ {"RIN34 PGA", "RIN3 Switch", "RIN3"}, - {"RIN34 PGA", "RIN4 Switch", "RIN4"}, + {"RIN34 PGA", "RIN4 Switch", "RIN4/RXP"}, /* INMIXL */ {"INMIXR", "Record Right Volume", "ROMIX"}, {"INMIXR", "RIN2 Volume", "RIN2"}, {"INMIXR", "RINPGA12 Switch", "RIN12 PGA"}, {"INMIXR", "RINPGA34 Switch", "RIN34 PGA"}, - /* AIRNMUX */ - {"AIRNMUX", "INMIXR Mix", "INMIXR"}, - {"AIRNMUX", "DIFFINR Mix", "RIN12PGA"}, - {"AIRNMUX", "DIFFINR Mix", "RIN34PGA"}, - {"AIRNMUX", "RXVOICE Mix", "RIN4/RXN"}, - {"AIRNMUX", "RXVOICE Mix", "RIN4/RXP"}, + /* AINRMUX */ + {"AINRMUX", "INMIXR Mix", "INMIXR"}, + {"AINRMUX", "DIFFINR Mix", "RIN12 PGA"}, + {"AINRMUX", "DIFFINR Mix", "RIN34 PGA"}, + {"AINRMUX", "RXVOICE Mix", "LIN4/RXN"}, + {"AINRMUX", "RXVOICE Mix", "RIN4/RXP"}, /* ADC */ - {"Right ADC", NULL, "AIRNMUX"}, + {"Right ADC", NULL, "AINRMUX"}, /* LOMIX */ {"LOMIX", "LOMIX RIN3 Bypass Switch", "RIN3"}, @@ -922,7 +922,7 @@ static const struct snd_soc_dapm_route audio_map[] = { {"LOPMIX", "LOPMIX Left Mixer PGA Switch", "LOPGA"}, /* OUT3MIX */ - {"OUT3MIX", "OUT3MIX LIN4/RXP Bypass Switch", "LIN4/RXP"}, + {"OUT3MIX", "OUT3MIX LIN4/RXP Bypass Switch", "LIN4/RXN"}, {"OUT3MIX", "OUT3MIX Left Out PGA Switch", "LOPGA"}, /* OUT4MIX */ @@ -949,7 +949,7 @@ static const struct snd_soc_dapm_route audio_map[] = { /* Output Pins */ {"LON", NULL, "LONMIX"}, {"LOP", NULL, "LOPMIX"}, - {"OUT", NULL, "OUT3MIX"}, + {"OUT3", NULL, "OUT3MIX"}, {"LOUT", NULL, "LOUT PGA"}, {"SPKN", NULL, "SPKMIX"}, {"ROUT", NULL, "ROUT PGA"}, diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index 6e23a81dba78..c2d1a7a18fa3 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -318,7 +318,7 @@ static int wm9705_reset(struct snd_soc_codec *codec) } #ifdef CONFIG_PM -static int wm9705_soc_suspend(struct platform_device *pdev) +static int wm9705_soc_suspend(struct platform_device *pdev, pm_message_t msg) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index bd7392c9657e..411a710be660 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -10,13 +10,14 @@ config SND_DAVINCI_SOC_I2S tristate config SND_DAVINCI_SOC_EVM - tristate "SoC Audio support for DaVinci EVM" - depends on SND_DAVINCI_SOC && MACH_DAVINCI_EVM + tristate "SoC Audio support for DaVinci DM6446 or DM355 EVM" + depends on SND_DAVINCI_SOC + depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM select SND_DAVINCI_SOC_I2S select SND_SOC_TLV320AIC3X help Say Y if you want to add support for SoC audio on TI - DaVinci EVM platform. + DaVinci DM6446 or DM355 EVM platforms. config SND_DAVINCI_SOC_SFFSDR tristate "SoC Audio support for SFFSDR" diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 9b90b347007c..58fd1cbedd88 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -20,7 +20,11 @@ #include <sound/soc-dapm.h> #include <asm/dma.h> -#include <mach/hardware.h> +#include <asm/mach-types.h> + +#include <mach/asp.h> +#include <mach/edma.h> +#include <mach/mux.h> #include "../codecs/tlv320aic3x.h" #include "davinci-pcm.h" @@ -150,7 +154,7 @@ static struct snd_soc_card snd_soc_card_evm = { /* evm audio private data */ static struct aic3x_setup_data evm_aic3x_setup = { - .i2c_bus = 0, + .i2c_bus = 1, .i2c_address = 0x1b, }; @@ -161,36 +165,73 @@ static struct snd_soc_device evm_snd_devdata = { .codec_data = &evm_aic3x_setup, }; +/* DM6446 EVM uses ASP0; line-out is a pair of RCA jacks */ static struct resource evm_snd_resources[] = { { - .start = DAVINCI_MCBSP_BASE, - .end = DAVINCI_MCBSP_BASE + SZ_8K - 1, + .start = DAVINCI_ASP0_BASE, + .end = DAVINCI_ASP0_BASE + SZ_8K - 1, .flags = IORESOURCE_MEM, }, }; static struct evm_snd_platform_data evm_snd_data = { - .tx_dma_ch = DM644X_DMACH_MCBSP_TX, - .rx_dma_ch = DM644X_DMACH_MCBSP_RX, + .tx_dma_ch = DAVINCI_DMA_ASP0_TX, + .rx_dma_ch = DAVINCI_DMA_ASP0_RX, +}; + +/* DM335 EVM uses ASP1; line-out is a stereo mini-jack */ +static struct resource dm335evm_snd_resources[] = { + { + .start = DAVINCI_ASP1_BASE, + .end = DAVINCI_ASP1_BASE + SZ_8K - 1, + .flags = IORESOURCE_MEM, + }, +}; + +static struct evm_snd_platform_data dm335evm_snd_data = { + .tx_dma_ch = DAVINCI_DMA_ASP1_TX, + .rx_dma_ch = DAVINCI_DMA_ASP1_RX, }; static struct platform_device *evm_snd_device; static int __init evm_init(void) { + struct resource *resources; + unsigned num_resources; + struct evm_snd_platform_data *data; + int index; int ret; - evm_snd_device = platform_device_alloc("soc-audio", 0); + if (machine_is_davinci_evm()) { + davinci_cfg_reg(DM644X_MCBSP); + + resources = evm_snd_resources; + num_resources = ARRAY_SIZE(evm_snd_resources); + data = &evm_snd_data; + index = 0; + } else if (machine_is_davinci_dm355_evm()) { + /* we don't use ASP1 IRQs, or we'd need to mux them ... */ + davinci_cfg_reg(DM355_EVT8_ASP1_TX); + davinci_cfg_reg(DM355_EVT9_ASP1_RX); + + resources = dm335evm_snd_resources; + num_resources = ARRAY_SIZE(dm335evm_snd_resources); + data = &dm335evm_snd_data; + index = 1; + } else + return -EINVAL; + + evm_snd_device = platform_device_alloc("soc-audio", index); if (!evm_snd_device) return -ENOMEM; platform_set_drvdata(evm_snd_device, &evm_snd_devdata); evm_snd_devdata.dev = &evm_snd_device->dev; - platform_device_add_data(evm_snd_device, &evm_snd_data, - sizeof(evm_snd_data)); + platform_device_add_data(evm_snd_device, data, sizeof(*data)); - ret = platform_device_add_resources(evm_snd_device, evm_snd_resources, - ARRAY_SIZE(evm_snd_resources)); + ret = platform_device_add_resources(evm_snd_device, resources, + num_resources); if (ret) { platform_device_put(evm_snd_device); return ret; diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index ffdb9439d3d8..b1ea52fc83c7 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -24,6 +24,26 @@ #include "davinci-pcm.h" + +/* + * NOTE: terminology here is confusing. + * + * - This driver supports the "Audio Serial Port" (ASP), + * found on dm6446, dm355, and other DaVinci chips. + * + * - But it labels it a "Multi-channel Buffered Serial Port" + * (McBSP) as on older chips like the dm642 ... which was + * backward-compatible, possibly explaining that confusion. + * + * - OMAP chips have a controller called McBSP, which is + * incompatible with the DaVinci flavor of McBSP. + * + * - Newer DaVinci chips have a controller called McASP, + * incompatible with ASP and with either McBSP. + * + * In short: this uses ASP to implement I2S, not McBSP. + * And it won't be the only DaVinci implemention of I2S. + */ #define DAVINCI_MCBSP_DRR_REG 0x00 #define DAVINCI_MCBSP_DXR_REG 0x04 #define DAVINCI_MCBSP_SPCR_REG 0x08 @@ -421,7 +441,7 @@ static int davinci_i2s_probe(struct platform_device *pdev, { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_card *card = socdev->card; - struct snd_soc_dai *cpu_dai = card->dai_link[pdev->id].cpu_dai; + struct snd_soc_dai *cpu_dai = card->dai_link->cpu_dai; struct davinci_mcbsp_dev *dev; struct resource *mem, *ioarea; struct evm_snd_platform_data *pdata; @@ -448,7 +468,7 @@ static int davinci_i2s_probe(struct platform_device *pdev, cpu_dai->private_data = dev; - dev->clk = clk_get(&pdev->dev, "McBSPCLK"); + dev->clk = clk_get(&pdev->dev, NULL); if (IS_ERR(dev->clk)) { ret = -ENODEV; goto err_free_mem; @@ -483,7 +503,7 @@ static void davinci_i2s_remove(struct platform_device *pdev, { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_card *card = socdev->card; - struct snd_soc_dai *cpu_dai = card->dai_link[pdev->id].cpu_dai; + struct snd_soc_dai *cpu_dai = card->dai_link->cpu_dai; struct davinci_mcbsp_dev *dev = cpu_dai->private_data; struct resource *mem; diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 7af3b5b3a53d..a05996588489 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -22,6 +22,7 @@ #include <sound/soc.h> #include <asm/dma.h> +#include <mach/edma.h> #include "davinci-pcm.h" @@ -51,7 +52,7 @@ struct davinci_runtime_data { spinlock_t lock; int period; /* current DMA period */ int master_lch; /* Master DMA channel */ - int slave_lch; /* Slave DMA channel */ + int slave_lch; /* linked parameter RAM reload slot */ struct davinci_pcm_dma_params *params; /* DMA params */ }; @@ -90,18 +91,18 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) dst_bidx = data_type; } - davinci_set_dma_src_params(lch, src, INCR, W8BIT); - davinci_set_dma_dest_params(lch, dst, INCR, W8BIT); - davinci_set_dma_src_index(lch, src_bidx, 0); - davinci_set_dma_dest_index(lch, dst_bidx, 0); - davinci_set_dma_transfer_params(lch, data_type, count, 1, 0, ASYNC); + edma_set_src(lch, src, INCR, W8BIT); + edma_set_dest(lch, dst, INCR, W8BIT); + edma_set_src_index(lch, src_bidx, 0); + edma_set_dest_index(lch, dst_bidx, 0); + edma_set_transfer_params(lch, data_type, count, 1, 0, ASYNC); prtd->period++; if (unlikely(prtd->period >= runtime->periods)) prtd->period = 0; } -static void davinci_pcm_dma_irq(int lch, u16 ch_status, void *data) +static void davinci_pcm_dma_irq(unsigned lch, u16 ch_status, void *data) { struct snd_pcm_substream *substream = data; struct davinci_runtime_data *prtd = substream->runtime->private_data; @@ -125,7 +126,7 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) struct davinci_runtime_data *prtd = substream->runtime->private_data; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct davinci_pcm_dma_params *dma_data = rtd->dai->cpu_dai->dma_data; - int tcc = TCC_ANY; + struct edmacc_param p_ram; int ret; if (!dma_data) @@ -134,22 +135,34 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) prtd->params = dma_data; /* Request master DMA channel */ - ret = davinci_request_dma(prtd->params->channel, prtd->params->name, + ret = edma_alloc_channel(prtd->params->channel, davinci_pcm_dma_irq, substream, - &prtd->master_lch, &tcc, EVENTQ_0); - if (ret) + EVENTQ_0); + if (ret < 0) return ret; + prtd->master_lch = ret; - /* Request slave DMA channel */ - ret = davinci_request_dma(PARAM_ANY, "Link", - NULL, NULL, &prtd->slave_lch, &tcc, EVENTQ_0); - if (ret) { - davinci_free_dma(prtd->master_lch); + /* Request parameter RAM reload slot */ + ret = edma_alloc_slot(EDMA_SLOT_ANY); + if (ret < 0) { + edma_free_channel(prtd->master_lch); return ret; } - - /* Link slave DMA channel in loopback */ - davinci_dma_link_lch(prtd->slave_lch, prtd->slave_lch); + prtd->slave_lch = ret; + + /* Issue transfer completion IRQ when the channel completes a + * transfer, then always reload from the same slot (by a kind + * of loopback link). The completion IRQ handler will update + * the reload slot with a new buffer. + * + * REVISIT save p_ram here after setting up everything except + * the buffer and its length (ccnt) ... use it as a template + * so davinci_pcm_enqueue_dma() takes less time in IRQ. + */ + edma_read_slot(prtd->slave_lch, &p_ram); + p_ram.opt |= TCINTEN | EDMA_TCC(prtd->master_lch); + p_ram.link_bcntrld = prtd->slave_lch << 5; + edma_write_slot(prtd->slave_lch, &p_ram); return 0; } @@ -165,12 +178,12 @@ static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - davinci_start_dma(prtd->master_lch); + edma_start(prtd->master_lch); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - davinci_stop_dma(prtd->master_lch); + edma_stop(prtd->master_lch); break; default: ret = -EINVAL; @@ -185,14 +198,14 @@ static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd) static int davinci_pcm_prepare(struct snd_pcm_substream *substream) { struct davinci_runtime_data *prtd = substream->runtime->private_data; - struct paramentry_descriptor temp; + struct edmacc_param temp; prtd->period = 0; davinci_pcm_enqueue_dma(substream); - /* Get slave channel dma params for master channel startup */ - davinci_get_dma_params(prtd->slave_lch, &temp); - davinci_set_dma_params(prtd->master_lch, &temp); + /* Copy self-linked parameter RAM entry into master channel */ + edma_read_slot(prtd->slave_lch, &temp); + edma_write_slot(prtd->master_lch, &temp); return 0; } @@ -208,7 +221,7 @@ davinci_pcm_pointer(struct snd_pcm_substream *substream) spin_lock(&prtd->lock); - davinci_dma_getposition(prtd->master_lch, &src, &dst); + edma_get_position(prtd->master_lch, &src, &dst); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) count = src - runtime->dma_addr; else @@ -253,10 +266,10 @@ static int davinci_pcm_close(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct davinci_runtime_data *prtd = runtime->private_data; - davinci_dma_unlink_lch(prtd->slave_lch, prtd->slave_lch); + edma_unlink(prtd->slave_lch); - davinci_free_dma(prtd->slave_lch); - davinci_free_dma(prtd->master_lch); + edma_free_slot(prtd->slave_lch); + edma_free_channel(prtd->master_lch); kfree(prtd); diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c index 3aa729df27b5..1111c710118a 100644 --- a/sound/soc/fsl/mpc5200_psc_i2s.c +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -504,7 +504,8 @@ static struct snd_soc_dai psc_i2s_dai_template = { static const struct snd_pcm_hardware psc_i2s_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER, + SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_BATCH, .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE, .rate_min = 8000, diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index a6d1178ce128..91ef17992de5 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -3,7 +3,7 @@ * * Copyright (C) 2008 Nokia Corporation * - * Contact: Jarkko Nikula <jarkko.nikula@nokia.com> + * Contact: 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 @@ -417,6 +417,6 @@ static void __exit n810_soc_exit(void) module_init(n810_soc_init); module_exit(n810_soc_exit); -MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>"); +MODULE_AUTHOR("Jarkko Nikula <jhnikula@gmail.com>"); MODULE_DESCRIPTION("ALSA SoC Nokia N810"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 9c09b94f0cf8..912614283848 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -3,7 +3,8 @@ * * Copyright (C) 2008 Nokia Corporation * - * Contact: Jarkko Nikula <jarkko.nikula@nokia.com> + * Contact: Jarkko Nikula <jhnikula@gmail.com> + * Peter Ujfalusi <peter.ujfalusi@nokia.com> * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License @@ -283,7 +284,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, break; case SND_SOC_DAIFMT_DSP_B: regs->srgr2 |= FPER(wlen * channels - 1); - regs->srgr1 |= FWID(wlen * channels - 2); + regs->srgr1 |= FWID(0); break; } @@ -302,6 +303,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, { struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; + unsigned int temp_fmt = fmt; if (mcbsp_data->configured) return 0; @@ -328,6 +330,8 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, /* 0-bit data delay */ regs->rcr2 |= RDATDLY(0); regs->xcr2 |= XDATDLY(0); + /* Invert FS polarity configuration */ + temp_fmt ^= SND_SOC_DAIFMT_NB_IF; break; default: /* Unsupported data format */ @@ -351,7 +355,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, } /* Set bit clock (CLKX/CLKR) and FS polarities */ - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + switch (temp_fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: /* * Normal BCLK + FS. @@ -529,6 +533,6 @@ static void __exit snd_omap_mcbsp_exit(void) } module_exit(snd_omap_mcbsp_exit); -MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>"); +MODULE_AUTHOR("Jarkko Nikula <jhnikula@gmail.com>"); MODULE_DESCRIPTION("OMAP I2S SoC Interface"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h index df7ad13ba73d..c8147aace813 100644 --- a/sound/soc/omap/omap-mcbsp.h +++ b/sound/soc/omap/omap-mcbsp.h @@ -3,7 +3,8 @@ * * Copyright (C) 2008 Nokia Corporation * - * Contact: Jarkko Nikula <jarkko.nikula@nokia.com> + * Contact: Jarkko Nikula <jhnikula@gmail.com> + * Peter Ujfalusi <peter.ujfalusi@nokia.com> * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 1bdbb0427183..07cf7f46b584 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -3,7 +3,8 @@ * * Copyright (C) 2008 Nokia Corporation * - * Contact: Jarkko Nikula <jarkko.nikula@nokia.com> + * Contact: Jarkko Nikula <jhnikula@gmail.com> + * Peter Ujfalusi <peter.ujfalusi@nokia.com> * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License @@ -367,6 +368,6 @@ static void __exit omap_soc_platform_exit(void) } module_exit(omap_soc_platform_exit); -MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>"); +MODULE_AUTHOR("Jarkko Nikula <jhnikula@gmail.com>"); MODULE_DESCRIPTION("OMAP PCM DMA module"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap-pcm.h b/sound/soc/omap/omap-pcm.h index e4369bdfd77d..8d9d26916b05 100644 --- a/sound/soc/omap/omap-pcm.h +++ b/sound/soc/omap/omap-pcm.h @@ -3,7 +3,8 @@ * * Copyright (C) 2008 Nokia Corporation * - * Contact: Jarkko Nikula <jarkko.nikula@nokia.com> + * Contact: Jarkko Nikula <jhnikula@gmail.com> + * Peter Ujfalusi <peter.ujfalusi@nokia.com> * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c index a952a4eb3361..a4e149b7f0eb 100644 --- a/sound/soc/omap/osk5912.c +++ b/sound/soc/omap/osk5912.c @@ -62,7 +62,7 @@ static int osk_hw_params(struct snd_pcm_substream *substream, /* Set codec DAI configuration */ err = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B | - SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (err < 0) { printk(KERN_ERR "can't set codec DAI configuration\n"); @@ -72,7 +72,7 @@ static int osk_hw_params(struct snd_pcm_substream *substream, /* Set cpu DAI configuration */ err = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_B | - SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (err < 0) { printk(KERN_ERR "can't set cpu DAI configuration\n"); diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c index 48a73f64500b..44fcc4e01e08 100644 --- a/sound/soc/pxa/palm27x.c +++ b/sound/soc/pxa/palm27x.c @@ -200,7 +200,7 @@ static struct snd_soc_device palm27x_snd_devdata = { static struct platform_device *palm27x_snd_device; -static int __init palm27x_asoc_init(void) +static int palm27x_asoc_probe(struct platform_device *pdev) { int ret; @@ -208,6 +208,10 @@ static int __init palm27x_asoc_init(void) machine_is_palmld())) return -ENODEV; + if (pdev->dev.platform_data) + palm27x_ep_gpio = ((struct palm27x_asoc_info *) + (pdev->dev.platform_data))->jack_gpio; + ret = gpio_request(palm27x_ep_gpio, "Headphone Jack"); if (ret) return ret; @@ -245,16 +249,31 @@ err_alloc: return ret; } -static void __exit palm27x_asoc_exit(void) +static int __devexit palm27x_asoc_remove(struct platform_device *pdev) { free_irq(gpio_to_irq(palm27x_ep_gpio), NULL); gpio_free(palm27x_ep_gpio); platform_device_unregister(palm27x_snd_device); + return 0; } -void __init palm27x_asoc_set_pdata(struct palm27x_asoc_info *data) +static struct platform_driver palm27x_wm9712_driver = { + .probe = palm27x_asoc_probe, + .remove = __devexit_p(palm27x_asoc_remove), + .driver = { + .name = "palm27x-asoc", + .owner = THIS_MODULE, + }, +}; + +static int __init palm27x_asoc_init(void) +{ + return platform_driver_register(&palm27x_wm9712_driver); +} + +static void __exit palm27x_asoc_exit(void) { - palm27x_ep_gpio = data->jack_gpio; + platform_driver_unregister(&palm27x_wm9712_driver); } module_init(palm27x_asoc_init); diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 308a657928d2..286be31545df 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -280,12 +280,33 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai) * ssp_set_clkdiv - set SSP clock divider * @div: serial clock rate divider */ -static void ssp_set_scr(struct ssp_dev *dev, u32 div) +static void ssp_set_scr(struct ssp_device *ssp, u32 div) { - struct ssp_device *ssp = dev->ssp; - u32 sscr0 = ssp_read_reg(dev->ssp, SSCR0) & ~SSCR0_SCR; + u32 sscr0 = ssp_read_reg(ssp, SSCR0); + + if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) { + sscr0 &= ~0x0000ff00; + sscr0 |= ((div - 2)/2) << 8; /* 2..512 */ + } else { + sscr0 &= ~0x000fff00; + sscr0 |= (div - 1) << 8; /* 1..4096 */ + } + ssp_write_reg(ssp, SSCR0, sscr0); +} + +/** + * ssp_get_clkdiv - get SSP clock divider + */ +static u32 ssp_get_scr(struct ssp_device *ssp) +{ + u32 sscr0 = ssp_read_reg(ssp, SSCR0); + u32 div; - ssp_write_reg(ssp, SSCR0, (sscr0 | SSCR0_SerClkDiv(div))); + if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) + div = ((sscr0 >> 8) & 0xff) * 2 + 2; + else + div = ((sscr0 >> 8) & 0xfff) + 1; + return div; } /* @@ -326,7 +347,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, break; case PXA_SSP_CLK_AUDIO: priv->sysclk = 0; - ssp_set_scr(&priv->dev, 1); + ssp_set_scr(ssp, 1); sscr0 |= SSCR0_ACS; break; default: @@ -387,7 +408,7 @@ static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, ssp_write_reg(ssp, SSACD, val); break; case PXA_SSP_DIV_SCR: - ssp_set_scr(&priv->dev, div); + ssp_set_scr(ssp, div); break; default: return -ENODEV; @@ -674,8 +695,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, case SND_SOC_DAIFMT_I2S: sspsp = ssp_read_reg(ssp, SSPSP); - if (((sscr0 & SSCR0_SCR) == SSCR0_SerClkDiv(4)) && - (width == 16)) { + if ((ssp_get_scr(ssp) == 4) && (width == 16)) { /* This is a special case where the bitclk is 64fs * and we're not dealing with 2*32 bits of audio * samples. @@ -806,6 +826,7 @@ static int pxa_ssp_probe(struct platform_device *pdev, goto err_priv; } + priv->dai_fmt = (unsigned int) -1; dai->private_data = priv; return 0; diff --git a/sound/soc/s3c24xx/jive_wm8750.c b/sound/soc/s3c24xx/jive_wm8750.c index 32063790d95b..93e6c87b7399 100644 --- a/sound/soc/s3c24xx/jive_wm8750.c +++ b/sound/soc/s3c24xx/jive_wm8750.c @@ -69,8 +69,8 @@ static int jive_hw_params(struct snd_pcm_substream *substream, break; } - s3c_i2sv2_calc_rate(&div, NULL, params_rate(params), - s3c2412_get_iisclk()); + s3c_i2sv2_iis_calc_rate(&div, NULL, params_rate(params), + s3c2412_get_iisclk()); /* set codec DAI configuration */ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | @@ -145,8 +145,9 @@ static struct snd_soc_dai_link jive_dai = { }; /* jive audio machine driver */ -static struct snd_soc_machine snd_soc_machine_jive = { +static struct snd_soc_card snd_soc_machine_jive = { .name = "Jive", + .platform = &s3c24xx_soc_platform, .dai_link = &jive_dai, .num_links = 1, }; @@ -157,9 +158,8 @@ static struct wm8750_setup_data jive_wm8750_setup = { /* jive audio subsystem */ static struct snd_soc_device jive_snd_devdata = { - .machine = &snd_soc_machine_jive, - .platform = &s3c24xx_soc_platform, - .codec_dev = &soc_codec_dev_wm8750_spi, + .card = &snd_soc_machine_jive, + .codec_dev = &soc_codec_dev_wm8750, .codec_data = &jive_wm8750_setup, }; diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index 295a4c910262..ab680aac3fcb 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -473,9 +473,9 @@ static int s3c2412_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai, /* default table of all avaialable root fs divisors */ static unsigned int iis_fs_tab[] = { 256, 512, 384, 768 }; -int s3c2412_iis_calc_rate(struct s3c_i2sv2_rate_calc *info, - unsigned int *fstab, - unsigned int rate, struct clk *clk) +int s3c_i2sv2_iis_calc_rate(struct s3c_i2sv2_rate_calc *info, + unsigned int *fstab, + unsigned int rate, struct clk *clk) { unsigned long clkrate = clk_get_rate(clk); unsigned int div; @@ -531,7 +531,7 @@ int s3c2412_iis_calc_rate(struct s3c_i2sv2_rate_calc *info, return 0; } -EXPORT_SYMBOL_GPL(s3c2412_iis_calc_rate); +EXPORT_SYMBOL_GPL(s3c_i2sv2_iis_calc_rate); int s3c_i2sv2_probe(struct platform_device *pdev, struct snd_soc_dai *dai, @@ -624,15 +624,18 @@ static int s3c2412_i2s_resume(struct snd_soc_dai *dai) int s3c_i2sv2_register_dai(struct snd_soc_dai *dai) { - dai->ops.trigger = s3c2412_i2s_trigger; - dai->ops.hw_params = s3c2412_i2s_hw_params; - dai->ops.set_fmt = s3c2412_i2s_set_fmt; - dai->ops.set_clkdiv = s3c2412_i2s_set_clkdiv; + struct snd_soc_dai_ops *ops = dai->ops; + + ops->trigger = s3c2412_i2s_trigger; + ops->hw_params = s3c2412_i2s_hw_params; + ops->set_fmt = s3c2412_i2s_set_fmt; + ops->set_clkdiv = s3c2412_i2s_set_clkdiv; dai->suspend = s3c2412_i2s_suspend; dai->resume = s3c2412_i2s_resume; return snd_soc_register_dai(dai); } - EXPORT_SYMBOL_GPL(s3c_i2sv2_register_dai); + +MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c index 1ca3cdaa8213..b7e0b3f0bfc8 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -33,8 +33,8 @@ #include <plat/regs-s3c2412-iis.h> -#include <plat/regs-gpio.h> #include <plat/audio.h> +#include <mach/regs-gpio.h> #include <mach/dma.h> #include "s3c24xx-pcm.h" diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c index 0dad3a0bb920..baddb1242c71 100644 --- a/sound/soc/sh/dma-sh7760.c +++ b/sound/soc/sh/dma-sh7760.c @@ -103,7 +103,8 @@ static struct snd_pcm_hardware camelot_pcm_hardware = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP_VALID), + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BATCH), .formats = DMABRG_FMTS, .rates = DMABRG_RATES, .rate_min = 8000, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 99712f652d0d..1cd149b9ce69 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -954,6 +954,9 @@ static int soc_remove(struct platform_device *pdev) struct snd_soc_platform *platform = card->platform; struct snd_soc_codec_device *codec_dev = socdev->codec_dev; + if (!card->instantiated) + return 0; + run_delayed_work(&card->delayed_work); if (platform->remove) diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index af95ff1e126c..1d2e51b3f918 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -1975,7 +1975,8 @@ static struct snd_pcm_hardware snd_dbri_pcm_hw = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP_VALID, + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BATCH, .formats = SNDRV_PCM_FMTBIT_MU_LAW | SNDRV_PCM_FMTBIT_A_LAW | SNDRV_PCM_FMTBIT_U8 | diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index 3f45c0fe61ab..b13ce767ac72 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -195,11 +195,14 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) debug("%s(%p)\n", __func__, substream); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + dev->period_out_count[index] = BYTES_PER_SAMPLE + 1; dev->audio_out_buf_pos[index] = BYTES_PER_SAMPLE + 1; - else + } else { + dev->period_in_count[index] = BYTES_PER_SAMPLE; dev->audio_in_buf_pos[index] = BYTES_PER_SAMPLE; - + } + if (dev->streaming) return 0; @@ -300,8 +303,7 @@ static void check_for_elapsed_periods(struct snd_usb_caiaqdev *dev, if (!sub) continue; - pb = frames_to_bytes(sub->runtime, - sub->runtime->period_size); + pb = snd_pcm_lib_period_bytes(sub); cnt = (sub->stream == SNDRV_PCM_STREAM_PLAYBACK) ? &dev->period_out_count[stream] : &dev->period_in_count[stream]; diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index 6d517705da0e..515de1cd2a3e 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -35,7 +35,7 @@ #include "input.h" MODULE_AUTHOR("Daniel Mack <daniel@caiaq.de>"); -MODULE_DESCRIPTION("caiaq USB audio, version 1.3.13"); +MODULE_DESCRIPTION("caiaq USB audio, version 1.3.14"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, RigKontrol3}," diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c index 012ff1f6f8af..a5aae9d67f31 100644 --- a/sound/usb/usx2y/us122l.c +++ b/sound/usb/usx2y/us122l.c @@ -474,6 +474,14 @@ static bool us122l_create_card(struct snd_card *card) return true; } +static void snd_us122l_free(struct snd_card *card) +{ + struct us122l *us122l = US122L(card); + int index = us122l->chip.index; + if (index >= 0 && index < SNDRV_CARDS) + snd_us122l_card_used[index] = 0; +} + static int usx2y_create_card(struct usb_device *device, struct snd_card **cardp) { int dev; @@ -490,7 +498,7 @@ static int usx2y_create_card(struct usb_device *device, struct snd_card **cardp) if (err < 0) return err; snd_us122l_card_used[US122L(card)->chip.index = dev] = 1; - + card->private_free = snd_us122l_free; US122L(card)->chip.dev = device; US122L(card)->chip.card = card; mutex_init(&US122L(card)->mutex); @@ -584,7 +592,7 @@ static void snd_us122l_disconnect(struct usb_interface *intf) } usb_put_intf(intf); - usb_put_dev(US122L(card)->chip.dev); + usb_put_dev(us122l->chip.dev); while (atomic_read(&us122l->mmap_count)) msleep(500); diff --git a/sound/usb/usx2y/usb_stream.c b/sound/usb/usx2y/usb_stream.c index 24393dafcb6e..12ae0340adc0 100644 --- a/sound/usb/usx2y/usb_stream.c +++ b/sound/usb/usx2y/usb_stream.c @@ -33,32 +33,26 @@ static unsigned usb_stream_next_packet_size(struct usb_stream_kernel *sk) static void playback_prep_freqn(struct usb_stream_kernel *sk, struct urb *urb) { struct usb_stream *s = sk->s; - unsigned l = 0; - int pack; - - urb->iso_frame_desc[0].offset = 0; - urb->iso_frame_desc[0].length = usb_stream_next_packet_size(sk); - sk->out_phase = sk->out_phase_peeked; - urb->transfer_buffer_length = urb->iso_frame_desc[0].length; - - for (pack = 1; pack < sk->n_o_ps; pack++) { - l = usb_stream_next_packet_size(sk); - if (s->idle_outsize + urb->transfer_buffer_length + l > - s->period_size) + int pack, lb = 0; + + for (pack = 0; pack < sk->n_o_ps; pack++) { + int l = usb_stream_next_packet_size(sk); + if (s->idle_outsize + lb + l > s->period_size) goto check; sk->out_phase = sk->out_phase_peeked; - urb->iso_frame_desc[pack].offset = urb->transfer_buffer_length; + urb->iso_frame_desc[pack].offset = lb; urb->iso_frame_desc[pack].length = l; - urb->transfer_buffer_length += l; + lb += l; } - snd_printdd(KERN_DEBUG "%i\n", urb->transfer_buffer_length); + snd_printdd(KERN_DEBUG "%i\n", lb); check: urb->number_of_packets = pack; - s->idle_outsize += urb->transfer_buffer_length - s->period_size; + urb->transfer_buffer_length = lb; + s->idle_outsize += lb - s->period_size; snd_printdd(KERN_DEBUG "idle=%i ul=%i ps=%i\n", s->idle_outsize, - urb->transfer_buffer_length, s->period_size); + lb, s->period_size); } static void init_pipe_urbs(struct usb_stream_kernel *sk, unsigned use_packsize, @@ -282,21 +276,20 @@ static int usb_stream_prepare_playback(struct usb_stream_kernel *sk, struct usb_stream *s = sk->s; struct urb *io; struct usb_iso_packet_descriptor *id, *od; - int p, l = 0; + int p = 0, lb = 0, l = 0; io = sk->idle_outurb; od = io->iso_frame_desc; - io->transfer_buffer_length = 0; - for (p = 0; s->sync_packet < 0; ++p, ++s->sync_packet) { + for (; s->sync_packet < 0; ++p, ++s->sync_packet) { struct urb *ii = sk->completed_inurb; id = ii->iso_frame_desc + ii->number_of_packets + s->sync_packet; l = id->actual_length; od[p].length = l; - od[p].offset = io->transfer_buffer_length; - io->transfer_buffer_length += l; + od[p].offset = lb; + lb += l; } for (; @@ -304,38 +297,38 @@ static int usb_stream_prepare_playback(struct usb_stream_kernel *sk, ++p, ++s->sync_packet) { l = inurb->iso_frame_desc[s->sync_packet].actual_length; - if (s->idle_outsize + io->transfer_buffer_length + l > - s->period_size) + if (s->idle_outsize + lb + l > s->period_size) goto check_ok; od[p].length = l; - od[p].offset = io->transfer_buffer_length; - io->transfer_buffer_length += l; + od[p].offset = lb; + lb += l; } check_ok: s->sync_packet -= inurb->number_of_packets; - if (s->sync_packet < -2 || s->sync_packet > 0) { + if (unlikely(s->sync_packet < -2 || s->sync_packet > 0)) { snd_printk(KERN_WARNING "invalid sync_packet = %i;" " p=%i nop=%i %i %x %x %x > %x\n", s->sync_packet, p, inurb->number_of_packets, - s->idle_outsize + io->transfer_buffer_length + l, - s->idle_outsize, io->transfer_buffer_length, l, + s->idle_outsize + lb + l, + s->idle_outsize, lb, l, s->period_size); return -1; } - if (io->transfer_buffer_length % s->cfg.frame_size) { + if (unlikely(lb % s->cfg.frame_size)) { snd_printk(KERN_WARNING"invalid outsize = %i\n", - io->transfer_buffer_length); + lb); return -1; } - s->idle_outsize += io->transfer_buffer_length - s->period_size; + s->idle_outsize += lb - s->period_size; io->number_of_packets = p; - if (s->idle_outsize > 0) { - snd_printk(KERN_WARNING "idle=%i\n", s->idle_outsize); - return -1; - } - return 0; + io->transfer_buffer_length = lb; + if (s->idle_outsize <= 0) + return 0; + + snd_printk(KERN_WARNING "idle=%i\n", s->idle_outsize); + return -1; } static void prepare_inurb(int number_of_packets, struct urb *iu) diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index 9a608fa85155..dd1ab6177840 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -870,7 +870,8 @@ static struct snd_pcm_hardware snd_usX2Y_2c = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP_VALID), + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BATCH), .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_3LE, .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, .rate_min = 44100, |