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| author | Takashi Iwai <tiwai@suse.de> | 2011-03-28 13:03:58 +0200 |
|---|---|---|
| committer | Takashi Iwai <tiwai@suse.de> | 2011-03-28 13:03:58 +0200 |
| commit | cdccfc8dc0bf62a1da327324a8d639139acc9279 (patch) | |
| tree | dca7934b27d510c9c006558979ebc48e07a531cf /sound | |
| parent | b21a8ee67013372f439fbd1591e91d09de49bb9c (diff) | |
| parent | c6b358748e19ce7e230b0926ac42696bc485a562 (diff) | |
| download | kernel_samsung_smdk4412-cdccfc8dc0bf62a1da327324a8d639139acc9279.tar.gz kernel_samsung_smdk4412-cdccfc8dc0bf62a1da327324a8d639139acc9279.tar.bz2 kernel_samsung_smdk4412-cdccfc8dc0bf62a1da327324a8d639139acc9279.zip | |
Merge branch 'fix/misc' into topic/misc
Diffstat (limited to 'sound')
38 files changed, 539 insertions, 485 deletions
diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index 7c1fc64cb53..d0cead38d5f 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -210,6 +210,7 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask) if (mask & ISR_RXINTR) { struct aaci_runtime *aacirun = &aaci->capture; + bool period_elapsed = false; void *ptr; if (!aacirun->substream || !aacirun->start) { @@ -222,15 +223,12 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask) ptr = aacirun->ptr; do { - unsigned int len = aacirun->fifosz; + unsigned int len = aacirun->fifo_bytes; u32 val; if (aacirun->bytes <= 0) { aacirun->bytes += aacirun->period; - aacirun->ptr = ptr; - spin_unlock(&aacirun->lock); - snd_pcm_period_elapsed(aacirun->substream); - spin_lock(&aacirun->lock); + period_elapsed = true; } if (!(aacirun->cr & CR_EN)) break; @@ -260,6 +258,9 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask) aacirun->ptr = ptr; spin_unlock(&aacirun->lock); + + if (period_elapsed) + snd_pcm_period_elapsed(aacirun->substream); } if (mask & ISR_URINTR) { @@ -269,6 +270,7 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask) if (mask & ISR_TXINTR) { struct aaci_runtime *aacirun = &aaci->playback; + bool period_elapsed = false; void *ptr; if (!aacirun->substream || !aacirun->start) { @@ -281,15 +283,12 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask) ptr = aacirun->ptr; do { - unsigned int len = aacirun->fifosz; + unsigned int len = aacirun->fifo_bytes; u32 val; if (aacirun->bytes <= 0) { aacirun->bytes += aacirun->period; - aacirun->ptr = ptr; - spin_unlock(&aacirun->lock); - snd_pcm_period_elapsed(aacirun->substream); - spin_lock(&aacirun->lock); + period_elapsed = true; } if (!(aacirun->cr & CR_EN)) break; @@ -319,6 +318,9 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask) aacirun->ptr = ptr; spin_unlock(&aacirun->lock); + + if (period_elapsed) + snd_pcm_period_elapsed(aacirun->substream); } } @@ -361,7 +363,7 @@ static struct snd_pcm_hardware aaci_hw_info = { /* rates are setup from the AC'97 codec */ .channels_min = 2, - .channels_max = 6, + .channels_max = 2, .buffer_bytes_max = 64 * 1024, .period_bytes_min = 256, .period_bytes_max = PAGE_SIZE, @@ -369,12 +371,46 @@ static struct snd_pcm_hardware aaci_hw_info = { .periods_max = PAGE_SIZE / 16, }; -static int __aaci_pcm_open(struct aaci *aaci, - struct snd_pcm_substream *substream, - struct aaci_runtime *aacirun) +/* + * We can support two and four channel audio. Unfortunately + * six channel audio requires a non-standard channel ordering: + * 2 -> FL(3), FR(4) + * 4 -> FL(3), FR(4), SL(7), SR(8) + * 6 -> FL(3), FR(4), SL(7), SR(8), C(6), LFE(9) (required) + * FL(3), FR(4), C(6), SL(7), SR(8), LFE(9) (actual) + * This requires an ALSA configuration file to correct. + */ +static int aaci_rule_channels(struct snd_pcm_hw_params *p, + struct snd_pcm_hw_rule *rule) +{ + static unsigned int channel_list[] = { 2, 4, 6 }; + struct aaci *aaci = rule->private; + unsigned int mask = 1 << 0, slots; + + /* pcms[0] is the our 5.1 PCM instance. */ + slots = aaci->ac97_bus->pcms[0].r[0].slots; + if (slots & (1 << AC97_SLOT_PCM_SLEFT)) { + mask |= 1 << 1; + if (slots & (1 << AC97_SLOT_LFE)) + mask |= 1 << 2; + } + + return snd_interval_list(hw_param_interval(p, rule->var), + ARRAY_SIZE(channel_list), channel_list, mask); +} + +static int aaci_pcm_open(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; - int ret; + struct aaci *aaci = substream->private_data; + struct aaci_runtime *aacirun; + int ret = 0; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + aacirun = &aaci->playback; + } else { + aacirun = &aaci->capture; + } aacirun->substream = substream; runtime->private_data = aacirun; @@ -382,27 +418,37 @@ static int __aaci_pcm_open(struct aaci *aaci, runtime->hw.rates = aacirun->pcm->rates; snd_pcm_limit_hw_rates(runtime); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && - aacirun->pcm->r[1].slots) - snd_ac97_pcm_double_rate_rules(runtime); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + runtime->hw.channels_max = 6; + + /* Add rule describing channel dependency. */ + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + aaci_rule_channels, aaci, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (ret) + return ret; + + if (aacirun->pcm->r[1].slots) + snd_ac97_pcm_double_rate_rules(runtime); + } /* - * FIXME: ALSA specifies fifo_size in bytes. If we're in normal - * mode, each 32-bit word contains one sample. If we're in - * compact mode, each 32-bit word contains two samples, effectively - * halving the FIFO size. However, we don't know for sure which - * we'll be using at this point. We set this to the lower limit. + * ALSA wants the byte-size of the FIFOs. As we only support + * 16-bit samples, this is twice the FIFO depth irrespective + * of whether it's in compact mode or not. */ - runtime->hw.fifo_size = aaci->fifosize * 2; - - ret = request_irq(aaci->dev->irq[0], aaci_irq, IRQF_SHARED|IRQF_DISABLED, - DRIVER_NAME, aaci); - if (ret) - goto out; - - return 0; + runtime->hw.fifo_size = aaci->fifo_depth * 2; + + mutex_lock(&aaci->irq_lock); + if (!aaci->users++) { + ret = request_irq(aaci->dev->irq[0], aaci_irq, + IRQF_SHARED | IRQF_DISABLED, DRIVER_NAME, aaci); + if (ret != 0) + aaci->users--; + } + mutex_unlock(&aaci->irq_lock); - out: return ret; } @@ -418,7 +464,11 @@ static int aaci_pcm_close(struct snd_pcm_substream *substream) WARN_ON(aacirun->cr & CR_EN); aacirun->substream = NULL; - free_irq(aaci->dev->irq[0], aaci); + + mutex_lock(&aaci->irq_lock); + if (!--aaci->users) + free_irq(aaci->dev->irq[0], aaci); + mutex_unlock(&aaci->irq_lock); return 0; } @@ -444,12 +494,21 @@ static int aaci_pcm_hw_free(struct snd_pcm_substream *substream) return 0; } +/* Channel to slot mask */ +static const u32 channels_to_slotmask[] = { + [2] = CR_SL3 | CR_SL4, + [4] = CR_SL3 | CR_SL4 | CR_SL7 | CR_SL8, + [6] = CR_SL3 | CR_SL4 | CR_SL7 | CR_SL8 | CR_SL6 | CR_SL9, +}; + static int aaci_pcm_hw_params(struct snd_pcm_substream *substream, - struct aaci_runtime *aacirun, struct snd_pcm_hw_params *params) { + struct aaci_runtime *aacirun = substream->runtime->private_data; + unsigned int channels = params_channels(params); + unsigned int rate = params_rate(params); + int dbl = rate > 48000; int err; - struct aaci *aaci = substream->private_data; aaci_pcm_hw_free(substream); if (aacirun->pcm_open) { @@ -457,22 +516,28 @@ static int aaci_pcm_hw_params(struct snd_pcm_substream *substream, aacirun->pcm_open = 0; } + /* channels is already limited to 2, 4, or 6 by aaci_rule_channels */ + if (dbl && channels != 2) + return -EINVAL; + err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); if (err >= 0) { - unsigned int rate = params_rate(params); - int dbl = rate > 48000; + struct aaci *aaci = substream->private_data; - err = snd_ac97_pcm_open(aacirun->pcm, rate, - params_channels(params), + err = snd_ac97_pcm_open(aacirun->pcm, rate, channels, aacirun->pcm->r[dbl].slots); aacirun->pcm_open = err == 0; aacirun->cr = CR_FEN | CR_COMPACT | CR_SZ16; - aacirun->fifosz = aaci->fifosize * 4; - - if (aacirun->cr & CR_COMPACT) - aacirun->fifosz >>= 1; + aacirun->cr |= channels_to_slotmask[channels + dbl * 2]; + + /* + * fifo_bytes is the number of bytes we transfer to/from + * the FIFO, including padding. So that's x4. As we're + * in compact mode, the FIFO is half the size. + */ + aacirun->fifo_bytes = aaci->fifo_depth * 4 / 2; } return err; @@ -483,11 +548,11 @@ static int aaci_pcm_prepare(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct aaci_runtime *aacirun = runtime->private_data; + aacirun->period = snd_pcm_lib_period_bytes(substream); aacirun->start = runtime->dma_area; aacirun->end = aacirun->start + snd_pcm_lib_buffer_bytes(substream); aacirun->ptr = aacirun->start; - aacirun->period = - aacirun->bytes = frames_to_bytes(runtime, runtime->period_size); + aacirun->bytes = aacirun->period; return 0; } @@ -505,89 +570,6 @@ static snd_pcm_uframes_t aaci_pcm_pointer(struct snd_pcm_substream *substream) /* * Playback specific ALSA stuff */ -static const u32 channels_to_txmask[] = { - [2] = CR_SL3 | CR_SL4, - [4] = CR_SL3 | CR_SL4 | CR_SL7 | CR_SL8, - [6] = CR_SL3 | CR_SL4 | CR_SL7 | CR_SL8 | CR_SL6 | CR_SL9, -}; - -/* - * We can support two and four channel audio. Unfortunately - * six channel audio requires a non-standard channel ordering: - * 2 -> FL(3), FR(4) - * 4 -> FL(3), FR(4), SL(7), SR(8) - * 6 -> FL(3), FR(4), SL(7), SR(8), C(6), LFE(9) (required) - * FL(3), FR(4), C(6), SL(7), SR(8), LFE(9) (actual) - * This requires an ALSA configuration file to correct. - */ -static unsigned int channel_list[] = { 2, 4, 6 }; - -static int -aaci_rule_channels(struct snd_pcm_hw_params *p, struct snd_pcm_hw_rule *rule) -{ - struct aaci *aaci = rule->private; - unsigned int chan_mask = 1 << 0, slots; - - /* - * pcms[0] is the our 5.1 PCM instance. - */ - slots = aaci->ac97_bus->pcms[0].r[0].slots; - if (slots & (1 << AC97_SLOT_PCM_SLEFT)) { - chan_mask |= 1 << 1; - if (slots & (1 << AC97_SLOT_LFE)) - chan_mask |= 1 << 2; - } - - return snd_interval_list(hw_param_interval(p, rule->var), - ARRAY_SIZE(channel_list), channel_list, - chan_mask); -} - -static int aaci_pcm_open(struct snd_pcm_substream *substream) -{ - struct aaci *aaci = substream->private_data; - int ret; - - /* - * Add rule describing channel dependency. - */ - ret = snd_pcm_hw_rule_add(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_CHANNELS, - aaci_rule_channels, aaci, - SNDRV_PCM_HW_PARAM_CHANNELS, -1); - if (ret) - return ret; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - ret = __aaci_pcm_open(aaci, substream, &aaci->playback); - } else { - ret = __aaci_pcm_open(aaci, substream, &aaci->capture); - } - return ret; -} - -static int aaci_pcm_playback_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct aaci_runtime *aacirun = substream->runtime->private_data; - unsigned int channels = params_channels(params); - int ret; - - WARN_ON(channels >= ARRAY_SIZE(channels_to_txmask) || - !channels_to_txmask[channels]); - - ret = aaci_pcm_hw_params(substream, aacirun, params); - - /* - * Enable FIFO, compact mode, 16 bits per sample. - * FIXME: double rate slots? - */ - if (ret >= 0) - aacirun->cr |= channels_to_txmask[channels]; - - return ret; -} - static void aaci_pcm_playback_stop(struct aaci_runtime *aacirun) { u32 ie; @@ -657,27 +639,13 @@ static struct snd_pcm_ops aaci_playback_ops = { .open = aaci_pcm_open, .close = aaci_pcm_close, .ioctl = snd_pcm_lib_ioctl, - .hw_params = aaci_pcm_playback_hw_params, + .hw_params = aaci_pcm_hw_params, .hw_free = aaci_pcm_hw_free, .prepare = aaci_pcm_prepare, .trigger = aaci_pcm_playback_trigger, .pointer = aaci_pcm_pointer, }; -static int aaci_pcm_capture_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct aaci_runtime *aacirun = substream->runtime->private_data; - int ret; - - ret = aaci_pcm_hw_params(substream, aacirun, params); - if (ret >= 0) - /* Line in record: slot 3 and 4 */ - aacirun->cr |= CR_SL3 | CR_SL4; - - return ret; -} - static void aaci_pcm_capture_stop(struct aaci_runtime *aacirun) { u32 ie; @@ -774,7 +742,7 @@ static struct snd_pcm_ops aaci_capture_ops = { .open = aaci_pcm_open, .close = aaci_pcm_close, .ioctl = snd_pcm_lib_ioctl, - .hw_params = aaci_pcm_capture_hw_params, + .hw_params = aaci_pcm_hw_params, .hw_free = aaci_pcm_hw_free, .prepare = aaci_pcm_capture_prepare, .trigger = aaci_pcm_capture_trigger, @@ -941,12 +909,13 @@ static struct aaci * __devinit aaci_init_card(struct amba_device *dev) strlcpy(card->driver, DRIVER_NAME, sizeof(card->driver)); strlcpy(card->shortname, "ARM AC'97 Interface", sizeof(card->shortname)); snprintf(card->longname, sizeof(card->longname), - "%s at 0x%016llx, irq %d", - card->shortname, (unsigned long long)dev->res.start, - dev->irq[0]); + "%s PL%03x rev%u at 0x%08llx, irq %d", + card->shortname, amba_part(dev), amba_rev(dev), + (unsigned long long)dev->res.start, dev->irq[0]); aaci = card->private_data; mutex_init(&aaci->ac97_sem); + mutex_init(&aaci->irq_lock); aaci->card = card; aaci->dev = dev; @@ -984,6 +953,10 @@ static unsigned int __devinit aaci_size_fifo(struct aaci *aaci) struct aaci_runtime *aacirun = &aaci->playback; int i; + /* + * Enable the channel, but don't assign it to any slots, so + * it won't empty onto the AC'97 link. + */ writel(CR_FEN | CR_SZ16 | CR_EN, aacirun->base + AACI_TXCR); for (i = 0; !(readl(aacirun->base + AACI_SR) & SR_TXFF) && i < 4096; i++) @@ -1002,7 +975,7 @@ static unsigned int __devinit aaci_size_fifo(struct aaci *aaci) writel(aaci->maincr, aaci->base + AACI_MAINCR); /* - * If we hit 4096, we failed. Go back to the specified + * If we hit 4096 entries, we failed. Go back to the specified * fifo depth. */ if (i == 4096) @@ -1011,7 +984,8 @@ static unsigned int __devinit aaci_size_fifo(struct aaci *aaci) return i; } -static int __devinit aaci_probe(struct amba_device *dev, struct amba_id *id) +static int __devinit aaci_probe(struct amba_device *dev, + const struct amba_id *id) { struct aaci *aaci; int ret, i; @@ -1067,11 +1041,12 @@ static int __devinit aaci_probe(struct amba_device *dev, struct amba_id *id) /* * Size the FIFOs (must be multiple of 16). + * This is the number of entries in the FIFO. */ - aaci->fifosize = aaci_size_fifo(aaci); - if (aaci->fifosize & 15) { - printk(KERN_WARNING "AACI: fifosize = %d not supported\n", - aaci->fifosize); + aaci->fifo_depth = aaci_size_fifo(aaci); + if (aaci->fifo_depth & 15) { + printk(KERN_WARNING "AACI: FIFO depth %d not supported\n", + aaci->fifo_depth); ret = -ENODEV; goto out; } @@ -1084,8 +1059,8 @@ static int __devinit aaci_probe(struct amba_device *dev, struct amba_id *id) ret = snd_card_register(aaci->card); if (ret == 0) { - dev_info(&dev->dev, "%s, fifo %d\n", aaci->card->longname, - aaci->fifosize); + dev_info(&dev->dev, "%s\n", aaci->card->longname); + dev_info(&dev->dev, "FIFO %u entries\n", aaci->fifo_depth); amba_set_drvdata(dev, aaci->card); return ret; } diff --git a/sound/arm/aaci.h b/sound/arm/aaci.h index 6a4a2eebdda..5791bd5bd2a 100644 --- a/sound/arm/aaci.h +++ b/sound/arm/aaci.h @@ -210,6 +210,8 @@ struct aaci_runtime { u32 cr; struct snd_pcm_substream *substream; + unsigned int period; /* byte size of a "period" */ + /* * PIO support */ @@ -217,15 +219,16 @@ struct aaci_runtime { void *end; void *ptr; int bytes; - unsigned int period; - unsigned int fifosz; + unsigned int fifo_bytes; }; struct aaci { struct amba_device *dev; struct snd_card *card; void __iomem *base; - unsigned int fifosize; + unsigned int fifo_depth; + unsigned int users; + struct mutex irq_lock; /* AC'97 */ struct mutex ac97_sem; diff --git a/sound/core/init.c b/sound/core/init.c index 3e65da21a08..a0080aa45ae 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -848,6 +848,7 @@ int snd_card_file_add(struct snd_card *card, struct file *file) return -ENOMEM; mfile->file = file; mfile->disconnected_f_op = NULL; + INIT_LIST_HEAD(&mfile->shutdown_list); spin_lock(&card->files_lock); if (card->shutdown) { spin_unlock(&card->files_lock); @@ -883,6 +884,9 @@ int snd_card_file_remove(struct snd_card *card, struct file *file) list_for_each_entry(mfile, &card->files_list, list) { if (mfile->file == file) { list_del(&mfile->list); + spin_lock(&shutdown_lock); + list_del(&mfile->shutdown_list); + spin_unlock(&shutdown_lock); if (mfile->disconnected_f_op) fops_put(mfile->disconnected_f_op); found = mfile; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index ae42b6509ce..fe5c8036beb 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3201,15 +3201,6 @@ int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, EXPORT_SYMBOL(snd_pcm_lib_mmap_iomem); #endif /* SNDRV_PCM_INFO_MMAP */ -/* mmap callback with pgprot_noncached */ -int snd_pcm_lib_mmap_noncached(struct snd_pcm_substream *substream, - struct vm_area_struct *area) -{ - area->vm_page_prot = pgprot_noncached(area->vm_page_prot); - return snd_pcm_default_mmap(substream, area); -} -EXPORT_SYMBOL(snd_pcm_lib_mmap_noncached); - /* * mmap DMA buffer */ diff --git a/sound/oss/dev_table.h b/sound/oss/dev_table.h index b7617bee638..0199a317c5a 100644 --- a/sound/oss/dev_table.h +++ b/sound/oss/dev_table.h @@ -271,7 +271,7 @@ struct synth_operations void (*reset) (int dev); void (*hw_control) (int dev, unsigned char *event); int (*load_patch) (int dev, int format, const char __user *addr, - int offs, int count, int pmgr_flag); + int count, int pmgr_flag); void (*aftertouch) (int dev, int voice, int pressure); void (*controller) (int dev, int voice, int ctrl_num, int value); void (*panning) (int dev, int voice, int value); diff --git a/sound/oss/midi_synth.c b/sound/oss/midi_synth.c index 3c09374ea5b..2292c230d7e 100644 --- a/sound/oss/midi_synth.c +++ b/sound/oss/midi_synth.c @@ -476,7 +476,7 @@ EXPORT_SYMBOL(midi_synth_hw_control); int midi_synth_load_patch(int dev, int format, const char __user *addr, - int offs, int count, int pmgr_flag) + int count, int pmgr_flag) { int orig_dev = synth_devs[dev]->midi_dev; @@ -491,33 +491,29 @@ midi_synth_load_patch(int dev, int format, const char __user *addr, if (!prefix_cmd(orig_dev, 0xf0)) return 0; + /* Invalid patch format */ if (format != SYSEX_PATCH) - { -/* printk("MIDI Error: Invalid patch format (key) 0x%x\n", format);*/ return -EINVAL; - } + + /* Patch header too short */ if (count < hdr_size) - { -/* printk("MIDI Error: Patch header too short\n");*/ return -EINVAL; - } + count -= hdr_size; /* - * Copy the header from user space but ignore the first bytes which have - * been transferred already. + * Copy the header from user space */ - if(copy_from_user(&((char *) &sysex)[offs], &(addr)[offs], hdr_size - offs)) + if (copy_from_user(&sysex, addr, hdr_size)) return -EFAULT; - - if (count < sysex.len) - { -/* printk(KERN_WARNING "MIDI Warning: Sysex record too short (%d<%d)\n", count, (int) sysex.len);*/ + + /* Sysex record too short */ + if ((unsigned)count < (unsigned)sysex.len) sysex.len = count; - } - left = sysex.len; - src_offs = 0; + + left = sysex.len; + src_offs = 0; for (i = 0; i < left && !signal_pending(current); i++) { diff --git a/sound/oss/midi_synth.h b/sound/oss/midi_synth.h index 6bc9d00bc77..b64ddd6c4ab 100644 --- a/sound/oss/midi_synth.h +++ b/sound/oss/midi_synth.h @@ -8,7 +8,7 @@ int midi_synth_open (int dev, int mode); void midi_synth_close (int dev); void midi_synth_hw_control (int dev, unsigned char *event); int midi_synth_load_patch (int dev, int format, const char __user * addr, - int offs, int count, int pmgr_flag); + int count, int pmgr_flag); void midi_synth_panning (int dev, int channel, int pressure); void midi_synth_aftertouch (int dev, int channel, int pressure); void midi_synth_controller (int dev, int channel, int ctrl_num, int value); diff --git a/sound/oss/opl3.c b/sound/oss/opl3.c index 938c48c4358..407cd677950 100644 --- a/sound/oss/opl3.c +++ b/sound/oss/opl3.c @@ -820,7 +820,7 @@ static void opl3_hw_control(int dev, unsigned char *event) } static int opl3_load_patch(int dev, int format, const char __user *addr, - int offs, int count, int pmgr_flag) + int count, int pmgr_flag) { struct sbi_instrument ins; @@ -830,11 +830,7 @@ static int opl3_load_patch(int dev, int format, const char __user *addr, return -EINVAL; } - /* - * What the fuck is going on here? We leave junk in the beginning - * of ins and then check the field pretty close to that beginning? - */ - if(copy_from_user(&((char *) &ins)[offs], addr + offs, sizeof(ins) - offs)) + if (copy_from_user(&ins, addr, sizeof(ins))) return -EFAULT; if (ins.channel < 0 || ins.channel >= SBFM_MAXINSTR) @@ -849,6 +845,10 @@ static int opl3_load_patch(int dev, int format, const char __user *addr, static void opl3_panning(int dev, int voice, int value) { + + if (voice < 0 || voice >= devc->nr_voice) + return; + devc->voc[voice].panning = value; } @@ -1066,8 +1066,15 @@ static int opl3_alloc_voice(int dev, int chn, int note, struct voice_alloc_info static void opl3_setup_voice(int dev, int voice, int chn) { - struct channel_info *info = - &synth_devs[dev]->chn_info[chn]; + struct channel_info *info; + + if (voice < 0 || voice >= devc->nr_voice) + return; + + if (chn < 0 || chn > 15) + return; + + info = &synth_devs[dev]->chn_info[chn]; opl3_set_instr(dev, voice, info->pgm_num); diff --git a/sound/oss/sequencer.c b/sound/oss/sequencer.c index 5ea1098ac42..30bcfe470f8 100644 --- a/sound/oss/sequencer.c +++ b/sound/oss/sequencer.c @@ -241,7 +241,7 @@ int sequencer_write(int dev, struct file *file, const char __user *buf, int coun return -ENXIO; fmt = (*(short *) &event_rec[0]) & 0xffff; - err = synth_devs[dev]->load_patch(dev, fmt, buf, p + 4, c, 0); + err = synth_devs[dev]->load_patch(dev, fmt, buf + p, c, 0); if (err < 0) return err; diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 0ac1f98d91a..f53a31e939c 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -22,21 +22,6 @@ * for any purpose including commercial applications. */ -/* >0: print Hw params, timer vars. >1: print stream write/copy sizes */ -#define REALLY_VERBOSE_LOGGING 0 - -#if REALLY_VERBOSE_LOGGING -#define VPRINTK1 snd_printd -#else -#define VPRINTK1(...) -#endif - -#if REALLY_VERBOSE_LOGGING > 1 -#define VPRINTK2 snd_printd -#else -#define VPRINTK2(...) -#endif - #include "hpi_internal.h" #include "hpimsginit.h" #include "hpioctl.h" @@ -57,11 +42,25 @@ #include <sound/tlv.h> #include <sound/hwdep.h> - MODULE_LICENSE("GPL"); MODULE_AUTHOR("AudioScience inc. <support@audioscience.com>"); MODULE_DESCRIPTION("AudioScience ALSA ASI5000 ASI6000 ASI87xx ASI89xx"); +#if defined CONFIG_SND_DEBUG_VERBOSE +/** + * snd_printddd - very verbose debug printk + * @format: format string + * + * Works like snd_printk() for debugging purposes. + * Ignored when CONFIG_SND_DEBUG_VERBOSE is not set. + * Must set snd module debug parameter to 3 to enable at runtime. + */ +#define snd_printddd(format, args...) \ + __snd_printk(3, __FILE__, __LINE__, format, ##args) +#else +#define snd_printddd(format, args...) do { } while (0) +#endif + static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; @@ -289,7 +288,6 @@ static u16 handle_error(u16 err, int line, char *filename) #define hpi_handle_error(x) handle_error(x, __LINE__, __FILE__) /***************************** GENERAL PCM ****************/ -#if REALLY_VERBOSE_LOGGING static void print_hwparams(struct snd_pcm_hw_params *p) { snd_printd("HWPARAMS \n"); @@ -304,9 +302,6 @@ static void print_hwparams(struct snd_pcm_hw_params *p) snd_printd("periods %d \n", params_periods(p)); snd_printd("buffer_size %d \n", params_buffer_size(p)); } -#else -#define print_hwparams(x) -#endif static snd_pcm_format_t hpi_to_alsa_formats[] = { -1, /* INVALID */ @@ -381,13 +376,13 @@ static void snd_card_asihpi_pcm_samplerates(struct snd_card_asihpi *asihpi, "No local sampleclock, err %d\n", err); } - for (idx = 0; idx < 100; idx++) { - if (hpi_sample_clock_query_local_rate( - h_control, idx, &sample_rate)) { - if (!idx) - snd_printk(KERN_ERR - "Local rate query failed\n"); - + for (idx = -1; idx < 100; idx++) { + if (idx == -1) { + if (hpi_sample_clock_get_sample_rate(h_control, + &sample_rate)) + continue; + } else if (hpi_sample_clock_query_local_rate(h_control, + idx, &sample_rate)) { break; } @@ -440,8 +435,6 @@ static void snd_card_asihpi_pcm_samplerates(struct snd_card_asihpi *asihpi, } } - /* printk(KERN_INFO "Supported rates %X %d %d\n", - rates, rate_min, rate_max); */ pcmhw->rates = rates; pcmhw->rate_min = rate_min; pcmhw->rate_max = rate_max; @@ -466,7 +459,7 @@ static int snd_card_asihpi_pcm_hw_params(struct snd_pcm_substream *substream, if (err) return err; - VPRINTK1(KERN_INFO "format %d, %d chans, %d_hz\n", + snd_printdd("format %d, %d chans, %d_hz\n", format, params_channels(params), params_rate(params)); @@ -489,13 +482,12 @@ static int snd_card_asihpi_pcm_hw_params(struct snd_pcm_substream *substream, err = hpi_stream_host_buffer_attach(dpcm->h_stream, params_buffer_bytes(params), runtime->dma_addr); if (err == 0) { - VPRINTK1(KERN_INFO + snd_printdd( "stream_host_buffer_attach succeeded %u %lu\n", params_buffer_bytes(params), (unsigned long)runtime->dma_addr); } else { - snd_printd(KERN_INFO - "stream_host_buffer_attach error %d\n", + snd_printd("stream_host_buffer_attach error %d\n", err); return -ENOMEM; } @@ -504,7 +496,7 @@ static int snd_card_asihpi_pcm_hw_params(struct snd_pcm_substream *substream, &dpcm->hpi_buffer_attached, NULL, NULL, NULL); - VPRINTK1(KERN_INFO "stream_host_buffer_attach status 0x%x\n", + snd_printdd("stream_host_buffer_attach status 0x%x\n", dpcm->hpi_buffer_attached); } bytes_per_sec = params_rate(params) * params_channels(params); @@ -517,7 +509,7 @@ static int snd_card_asihpi_pcm_hw_params(struct snd_pcm_substream *substream, dpcm->bytes_per_sec = bytes_per_sec; dpcm->buffer_bytes = params_buffer_bytes(params); dpcm->period_bytes = params_period_bytes(params); - VPRINTK1(KERN_INFO "buffer_bytes=%d, period_bytes=%d, bps=%d\n", + snd_printdd("buffer_bytes=%d, period_bytes=%d, bps=%d\n", dpcm->buffer_bytes, dpcm->period_bytes, bytes_per_sec); return 0; @@ -573,7 +565,7 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream, struct snd_pcm_substream *s; u16 e; - VPRINTK1(KERN_INFO "%c%d trigger\n", + snd_printdd("%c%d trigger\n", SCHR(substream->stream), substream->number); switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -597,7 +589,7 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream, * data?? */ unsigned int preload = ds->period_bytes * 1; - VPRINTK2(KERN_INFO "%d preload x%x\n", s->number, preload); + snd_printddd("%d preload x%x\n", s->number, preload); hpi_handle_error(hpi_outstream_write_buf( ds->h_stream, &runtime->dma_area[0], @@ -607,7 +599,7 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream, } if (card->support_grouping) { - VPRINTK1(KERN_INFO "\t%c%d group\n", + snd_printdd("\t%c%d group\n", SCHR(s->stream), s->number); e = hpi_stream_group_add( @@ -622,7 +614,7 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream, } else break; } - VPRINTK1(KERN_INFO "start\n"); + snd_printdd("start\n"); /* start the master stream */ snd_card_asihpi_pcm_timer_start(substream); if ((substream->stream == SNDRV_PCM_STREAM_CAPTURE) || @@ -644,14 +636,14 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream, s->runtime->status->state = SNDRV_PCM_STATE_SETUP; if (card->support_grouping) { - VPRINTK1(KERN_INFO "\t%c%d group\n", + snd_printdd("\t%c%d group\n", SCHR(s->stream), s->number); snd_pcm_trigger_done(s, substream); } else break; } - VPRINTK1(KERN_INFO "stop\n"); + snd_printdd("stop\n"); /* _prepare and _hwparams reset the stream */ hpi_handle_error(hpi_stream_stop(dpcm->h_stream)); @@ -664,12 +656,12 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream, break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - VPRINTK1(KERN_INFO "pause release\n"); + snd_printdd("pause release\n"); hpi_handle_error(hpi_stream_start(dpcm->h_stream)); snd_card_asihpi_pcm_timer_start(substream); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - VPRINTK1(KERN_INFO "pause\n"); + snd_printdd("pause\n"); snd_card_asihpi_pcm_timer_stop(substream); hpi_handle_error(hpi_stream_stop(dpcm->h_stream)); break; @@ -741,7 +733,7 @@ static void snd_card_asihpi_timer_function(unsigned long data) u16 state; u32 buffer_size, bytes_avail, samples_played, on_card_bytes; - VPRINTK1(KERN_INFO "%c%d snd_card_asihpi_timer_function\n", + snd_printdd("%c%d snd_card_asihpi_timer_function\n", SCHR(substream->stream), substream->number); /* find minimum newdata and buffer pos in group */ @@ -770,10 +762,10 @@ static void snd_card_asihpi_timer_function(unsigned long data) if ((bytes_avail == 0) && (on_card_bytes < ds->pcm_buf_host_rw_ofs)) { hpi_handle_error(hpi_stream_start(ds->h_stream)); - VPRINTK1(KERN_INFO "P%d start\n", s->number); + snd_printdd("P%d start\n", s->number); } } else if (state == HPI_STATE_DRAINED) { - VPRINTK1(KERN_WARNING "P%d drained\n", + snd_printd(KERN_WARNING "P%d drained\n", s->number); /*snd_pcm_stop(s, SNDRV_PCM_STATE_XRUN); continue; */ @@ -794,13 +786,13 @@ static void snd_card_asihpi_timer_function(unsigned long data) newdata); } - VPRINTK1(KERN_INFO "PB timer hw_ptr x%04lX, appl_ptr x%04lX\n", + snd_printdd("hw_ptr x%04lX, appl_ptr x%04lX\n", (unsigned long)frames_to_bytes(runtime, runtime->status->hw_ptr), (unsigned long)frames_to_bytes(runtime, runtime->control->appl_ptr)); - VPRINTK1(KERN_INFO "%d %c%d S=%d, rw=%04X, dma=x%04X, left=x%04X," + snd_printdd("%d %c%d S=%d, rw=%04X, dma=x%04X, left=x%04X," " aux=x%04X space=x%04X\n", loops, SCHR(s->stream), s->number, state, ds->pcm_buf_host_rw_ofs, pcm_buf_dma_ofs, (int)bytes_avail, @@ -822,7 +814,7 @@ static void snd_card_asihpi_timer_function(unsigned long data) next_jiffies = max(next_jiffies, 1U); dpcm->timer.expires = jiffies + next_jiffies; - VPRINTK1(KERN_INFO "jif %d buf pos x%04X newdata x%04X xfer x%04X\n", + snd_printdd("jif %d buf pos x%04X newdata x%04X xfer x%04X\n", next_jiffies, pcm_buf_dma_ofs, newdata, xfercount); snd_pcm_group_for_each_entry(s, substream) { @@ -837,7 +829,7 @@ static void snd_card_asihpi_timer_function(unsigned long data) if (xfercount && (on_card_bytes <= ds->period_bytes)) { if (card->support_mmap) { if (s->stream == SNDRV_PCM_STREAM_PLAYBACK) { - VPRINTK2(KERN_INFO "P%d write x%04x\n", + snd_printddd("P%d write x%04x\n", s->number, ds->period_bytes); hpi_handle_error( @@ -848,7 +840,7 @@ static void snd_card_asihpi_timer_function(unsigned long data) xfercount, &ds->format)); } else { - VPRINTK2(KERN_INFO "C%d read x%04x\n", + snd_printddd("C%d read x%04x\n", s->number, xfercount); hpi_handle_error( @@ -871,7 +863,7 @@ static void snd_card_asihpi_timer_function(unsigned long data) static int snd_card_asihpi_playback_ioctl(struct snd_pcm_substream *substream, unsigned int cmd, void *arg) { - /* snd_printd(KERN_INFO "Playback ioctl %d\n", cmd); */ + snd_printdd(KERN_INFO "Playback ioctl %d\n", cmd); return snd_pcm_lib_ioctl(substream, cmd, arg); } @@ -881,7 +873,7 @@ static int snd_card_asihpi_playback_prepare(struct snd_pcm_substream * struct snd_pcm_runtime *runtime = substream->runtime; struct snd_card_asihpi_pcm *dpcm = runtime->private_data; - VPRINTK1(KERN_INFO "playback prepare %d\n", substream->number); + snd_printdd("playback prepare %d\n", substream->number); hpi_handle_error(hpi_outstream_reset(dpcm->h_stream)); dpcm->pcm_buf_host_rw_ofs = 0; @@ -898,7 +890,7 @@ snd_card_asihpi_playback_pointer(struct snd_pcm_substream *substream) snd_pcm_uframes_t ptr; ptr = bytes_to_frames(runtime, dpcm->pcm_buf_dma_ofs % dpcm->buffer_bytes); - /* VPRINTK2(KERN_INFO "playback_pointer=x%04lx\n", (unsigned long)ptr); */ + snd_printddd("playback_pointer=x%04lx\n", (unsigned long)ptr); return ptr; } @@ -1014,12 +1006,13 @@ static int snd_card_asihpi_playback_open(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, card->update_interval_frames); + snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, card->update_interval_frames * 2, UINT_MAX); snd_pcm_set_sync(substream); - VPRINTK1(KERN_INFO "playback open\n"); + snd_printdd("playback open\n"); return 0; } @@ -1030,7 +1023,7 @@ static int snd_card_asihpi_playback_close(struct snd_pcm_substream *substream) struct snd_card_asihpi_pcm *dpcm = runtime->private_data; hpi_handle_error(hpi_outstream_close(dpcm->h_stream)); - VPRINTK1(KERN_INFO "playback close\n"); + snd_printdd("playback close\n"); return 0; } @@ -1050,13 +1043,13 @@ static int snd_card_asihpi_playback_copy(struct snd_pcm_substream *substream, if (copy_from_user(runtime->dma_area, src, len)) return -EFAULT; - VPRINTK2(KERN_DEBUG "playback copy%d %u bytes\n", + snd_printddd("playback copy%d %u bytes\n", substream->number, len); hpi_handle_error(hpi_outstream_write_buf(dpcm->h_stream, runtime->dma_area, len, &dpcm->format)); - dpcm->pcm_buf_host_rw_ofs = dpcm->pcm_buf_host_rw_ofs + len; + dpcm->pcm_buf_host_rw_ofs += len; return 0; } @@ -1066,16 +1059,11 @@ static int snd_card_asihpi_playback_silence(struct snd_pcm_substream * snd_pcm_uframes_t pos, snd_pcm_uframes_t count) { - unsigned int len; - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_card_asihpi_pcm *dpcm = runtime->private_data; - - len = frames_to_bytes(runtime, count); - VPRINTK1(KERN_INFO "playback silence %u bytes\n", len); - - memset(runtime->dma_area, 0, len); - hpi_handle_error(hpi_outstream_write_buf(dpcm->h_stream, - runtime->dma_area, len, &dpcm->format)); + /* Usually writes silence to DMA buffer, which should be overwritten + by real audio later. Our fifos cannot be overwritten, and are not + free-running DMAs. Silence is output on fifo underflow. + This callback is still required to allow the copy callback to be used. + */ return 0; } @@ -1110,7 +1098,7 @@ snd_card_asihpi_capture_pointer(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct snd_card_asihpi_pcm *dpcm = runtime->private_data; - VPRINTK2(KERN_INFO "capture pointer %d=%d\n", + snd_printddd("capture pointer %d=%d\n", substream->number, dpcm->pcm_buf_dma_ofs); /* NOTE Unlike playback can't use actual samples_played for the capture position, because those samples aren't yet in @@ -1135,7 +1123,7 @@ static int snd_card_asihpi_capture_prepare(struct snd_pcm_substream *substream) dpcm->pcm_buf_dma_ofs = 0; dpcm->pcm_buf_elapsed_dma_ofs = 0; - VPRINTK1("Capture Prepare %d\n", substream->number); + snd_printdd("Capture Prepare %d\n", substream->number); return 0; } @@ -1198,7 +1186,7 @@ static int snd_card_asihpi_capture_open(struct snd_pcm_substream *substream) if (dpcm == NULL) return -ENOMEM; - VPRINTK1("hpi_instream_open adapter %d stream %d\n", + snd_printdd("capture open adapter %d stream %d\n", card->adapter_index, substream->number); err = hpi_handle_error( @@ -1268,7 +1256,7 @@ static int snd_card_asihpi_capture_copy(struct snd_pcm_substream *substream, len = frames_to_bytes(runtime, count); - VPRINTK2(KERN_INFO "capture copy%d %d bytes\n", substream->number, len); + snd_printddd("capture copy%d %d bytes\n", substream->number, len); hpi_handle_error(hpi_instream_read_buf(dpcm->h_stream, runtime->dma_area, len)); @@ -2887,6 +2875,9 @@ static int __devinit snd_asihpi_probe(struct pci_dev *pci_dev, if (err) asihpi->update_interval_frames = 512; + if (!asihpi->support_mmap) + asihpi->update_interval_frames *= 2; + hpi_handle_error(hpi_instream_open(asihpi->adapter_index, 0, &h_stream)); @@ -2909,7 +2900,6 @@ static int __devinit snd_asihpi_probe(struct pci_dev *pci_dev, asihpi->support_mrx ); - err = snd_card_asihpi_pcm_new(asihpi, 0, pcm_substreams); if (err < 0) { snd_printk(KERN_ERR "pcm_new failed\n"); @@ -2944,6 +2934,7 @@ static int __devinit snd_asihpi_probe(struct pci_dev *pci_dev, sprintf(card->longname, "%s %i", card->shortname, asihpi->adapter_index); err = snd_card_register(card); + if (!err) { hpi_card->snd_card_asihpi = card; dev++; diff --git a/sound/pci/au88x0/au88x0.h b/sound/pci/au88x0/au88x0.h index cf46bba563c..ecb8f4daf40 100644 --- a/sound/pci/au88x0/au88x0.h +++ b/sound/pci/au88x0/au88x0.h @@ -211,7 +211,7 @@ static void vortex_adbdma_startfifo(vortex_t * vortex, int adbdma); //static void vortex_adbdma_stopfifo(vortex_t *vortex, int adbdma); static void vortex_adbdma_pausefifo(vortex_t * vortex, int adbdma); static void vortex_adbdma_resumefifo(vortex_t * vortex, int adbdma); -static int inline vortex_adbdma_getlinearpos(vortex_t * vortex, int adbdma); +static inline int vortex_adbdma_getlinearpos(vortex_t * vortex, int adbdma); static void vortex_adbdma_resetup(vortex_t *vortex, int adbdma); #ifndef CHIP_AU8810 @@ -219,7 +219,7 @@ static void vortex_wtdma_startfifo(vortex_t * vortex, int wtdma); static void vortex_wtdma_stopfifo(vortex_t * vortex, int wtdma); static void vortex_wtdma_pausefifo(vortex_t * vortex, int wtdma); static void vortex_wtdma_resumefifo(vortex_t * vortex, int wtdma); -static int inline vortex_wtdma_getlinearpos(vortex_t * vortex, int wtdma); +static inline int vortex_wtdma_getlinearpos(vortex_t * vortex, int wtdma); #endif /* global stuff. */ diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c index 16c0bdfbb16..489150380ea 100644 --- a/sound/pci/au88x0/au88x0_core.c +++ b/sound/pci/au88x0/au88x0_core.c @@ -1249,7 +1249,7 @@ static void vortex_adbdma_resetup(vortex_t *vortex, int adbdma) { } } -static int inline vortex_adbdma_getlinearpos(vortex_t * vortex, int adbdma) +static inline int vortex_adbdma_getlinearpos(vortex_t * vortex, int adbdma) { stream_t *dma = &vortex->dma_adb[adbdma]; int temp, page, delta; @@ -1506,7 +1506,7 @@ static int vortex_wtdma_getcursubuffer(vortex_t * vortex, int wtdma) POS_SHIFT) & POS_MASK); } #endif -static int inline vortex_wtdma_getlinearpos(vortex_t * vortex, int wtdma) +static inline int vortex_wtdma_getlinearpos(vortex_t * vortex, int wtdma) { stream_t *dma = &vortex->dma_wt[wtdma]; int temp; diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 734c6ee55d8..2942d2a9ea1 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -4256,6 +4256,84 @@ static int ad1984a_thinkpad_init(struct hda_codec *codec) } /* + * Precision R5500 + * 0x12 - HP/line-out + * 0x13 - speaker (mono) + * 0x15 - mic-in + */ + +static struct hda_verb ad1984a_precision_verbs[] = { + /* Unmute main output path */ + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */ + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE + 0x1f}, /* 0dB */ + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5) + 0x17}, /* 0dB */ + /* Analog mixer; mute as default */ + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + /* Select mic as input */ + {0x0c, AC_VERB_SET_CONNECT_SEL, 0x1}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE + 0x27}, /* 0dB */ + /* Configure as mic */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */ + /* HP unmute */ + {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* turn on EAPD */ + {0x13, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, + /* unsolicited event for pin-sense */ + {0x12, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT}, + { } /* end */ +}; + +static struct snd_kcontrol_new ad1984a_precision_mixers[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT), + HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x15, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Front Playback Switch", 0x12, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x13, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), + { } /* end */ +}; + + +/* mute internal speaker if HP is plugged */ +static void ad1984a_precision_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_jack_detect(codec, 0x12); + snd_hda_codec_amp_stereo(codec, 0x13, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); +} + + +/* unsolicited event for HP jack sensing */ +static void ad1984a_precision_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if ((res >> 26) != AD1884A_HP_EVENT) + return; + ad1984a_precision_automute(codec); +} + +/* initialize jack-sensing, too */ +static int ad1984a_precision_init(struct hda_codec *codec) +{ + ad198x_init(codec); + ad1984a_precision_automute(codec); + return 0; +} + + +/* * HP Touchsmart * port-A (0x11) - front hp-out * port-B (0x14) - unused @@ -4384,6 +4462,7 @@ enum { AD1884A_MOBILE, AD1884A_THINKPAD, AD1984A_TOUCHSMART, + AD1984A_PRECISION, AD1884A_MODELS }; @@ -4393,9 +4472,11 @@ static const char * const ad1884a_models[AD1884A_MODELS] = { [AD1884A_MOBILE] = "mobile", [AD1884A_THINKPAD] = "thinkpad", [AD1984A_TOUCHSMART] = "touchsmart", + [AD1984A_PRECISION] = "precision", }; static struct snd_pci_quirk ad1884a_cfg_tbl[] = { + SND_PCI_QUIRK(0x1028, 0x04ac, "Precision R5500", AD1984A_PRECISION), SND_PCI_QUIRK(0x103c, 0x3030, "HP", AD1884A_MOBILE), SND_PCI_QUIRK(0x103c, 0x3037, "HP 2230s", AD1884A_LAPTOP), SND_PCI_QUIRK(0x103c, 0x3056, "HP", AD1884A_MOBILE), @@ -4489,6 +4570,14 @@ static int patch_ad1884a(struct hda_codec *codec) codec->patch_ops.unsol_event = ad1984a_thinkpad_unsol_event; codec->patch_ops.init = ad1984a_thinkpad_init; break; + case AD1984A_PRECISION: + spec->mixers[0] = ad1984a_precision_mixers; + spec->init_verbs[spec->num_init_verbs++] = + ad1984a_precision_verbs; + spec->multiout.dig_out_nid = 0; + codec->patch_ops.unsol_event = ad1984a_precision_unsol_event; + codec->patch_ops.init = ad1984a_precision_init; + break; case AD1984A_TOUCHSMART: spec->mixers[0] = ad1984a_touchsmart_mixers; spec->init_verbs[0] = ad1984a_touchsmart_verbs; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 28f95d14ba6..12c6f4508c5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1290,7 +1290,7 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type) case 0x10ec0883: case 0x10ec0885: case 0x10ec0887: - case 0x10ec0889: + /*case 0x10ec0889:*/ /* this causes an SPDIF problem */ alc889_coef_init(codec); break; case 0x10ec0888: @@ -9863,7 +9863,6 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC883_LAPTOP_EAPD), SND_PCI_QUIRK(0x10f1, 0x2350, "TYAN-S2350", ALC888_6ST_DELL), SND_PCI_QUIRK(0x108e, 0x534d, NULL, ALC883_3ST_6ch), - SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte P35 DS3R", ALC882_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x0349, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x1462, 0x040d, "MSI", ALC883_TARGA_2ch_DIG), @@ -10700,6 +10699,7 @@ enum { PINFIX_LENOVO_Y530, PINFIX_PB_M5210, PINFIX_ACER_ASPIRE_7736, + PINFIX_GIGABYTE_880GM, }; static const struct alc_fixup alc882_fixups[] = { @@ -10731,6 +10731,13 @@ static const struct alc_fixup alc882_fixups[] = { .type = ALC_FIXUP_SKU, .v.sku = ALC_FIXUP_SKU_IGNORE, }, + [PINFIX_GIGABYTE_880GM] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x14, 0x1114410 }, /* set as speaker */ + { } + } + }, }; static struct snd_pci_quirk alc882_fixup_tbl[] = { @@ -10738,6 +10745,7 @@ static struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Y530", PINFIX_LENOVO_Y530), SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", PINFIX_ABIT_AW9D_MAX), SND_PCI_QUIRK(0x1025, 0x0296, "Acer Aspire 7736z", PINFIX_ACER_ASPIRE_7736), + SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte", PINFIX_GIGABYTE_880GM), {} }; @@ -16008,9 +16016,12 @@ static int alc861_auto_create_multi_out_ctls(struct hda_codec *codec, return err; } else { const char *name = pfx; - if (!name) + int index = i; + if (!name) { name = chname[i]; - err = __alc861_create_out_sw(codec, name, nid, i, 3); + index = 0; + } + err = __alc861_create_out_sw(codec, name, nid, index, 3); if (err < 0) return err; } @@ -17161,16 +17172,19 @@ static int alc861vd_auto_create_multi_out_ctls(struct alc_spec *spec, return err; } else { const char *name = pfx; - if (!name) + int index = i; + if (!name) { name = chname[i]; + index = 0; + } err = __add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, - name, i, + name, index, HDA_COMPOSE_AMP_VAL(nid_v, 3, 0, HDA_OUTPUT)); if (err < 0) return err; err = __add_pb_sw_ctrl(spec, ALC_CTL_BIND_MUTE, - name, i, + name, index, HDA_COMPOSE_AMP_VAL(nid_s, 3, 2, HDA_INPUT)); if (err < 0) @@ -18768,8 +18782,6 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = { ALC662_3ST_6ch_DIG), SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB20x", ALC662_AUTO), SND_PCI_QUIRK(0x144d, 0xca00, "Samsung NC10", ALC272_SAMSUNG_NC10), - SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L", - ALC662_3ST_6ch_DIG), SND_PCI_QUIRK(0x152d, 0x2304, "Quanta WH1", ALC663_ASUS_H13), SND_PCI_QUIRK(0x1565, 0x820f, "Biostar TA780G M2+", ALC662_3ST_6ch_DIG), SND_PCI_QUIRK(0x1631, 0xc10c, "PB RS65", ALC663_ASUS_M51VA), @@ -19219,12 +19231,15 @@ static int alc662_auto_create_multi_out_ctls(struct hda_codec *codec, return err; } else { const char *name = pfx; - if (!name) + int index = i; + if (!name) { name = chname[i]; - err = __alc662_add_vol_ctl(spec, name, nid, i, 3); + index = 0; + } + err = __alc662_add_vol_ctl(spec, name, nid, index, 3); if (err < 0) return err; - err = __alc662_add_sw_ctl(spec, name, mix, i, 3); + err = __alc662_add_sw_ctl(spec, name, mix, index, 3); if (err < 0) return err; } @@ -19440,6 +19455,7 @@ enum { ALC662_FIXUP_IDEAPAD, ALC272_FIXUP_MARIO, ALC662_FIXUP_CZC_P10T, + ALC662_FIXUP_GIGABYTE, }; static const struct alc_fixup alc662_fixups[] = { @@ -19468,12 +19484,20 @@ static const struct alc_fixup alc662_fixups[] = { {} } }, + [ALC662_FIXUP_GIGABYTE] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x14, 0x1114410 }, /* set as speaker */ + { } + } + }, }; static struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0308, "Acer Aspire 8942G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD), + SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte", ALC662_FIXUP_GIGABYTE), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Ideapad Y550", ALC662_FIXUP_IDEAPAD), SND_PCI_QUIRK(0x1b35, 0x2206, "CZC P10T", ALC662_FIXUP_CZC_P10T), diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 63b0054200a..1371b57c11e 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -159,6 +159,7 @@ struct via_spec { #endif }; +static enum VIA_HDA_CODEC get_codec_type(struct hda_codec *codec); static struct via_spec * via_new_spec(struct hda_codec *codec) { struct via_spec *spec; @@ -169,6 +170,10 @@ static struct via_spec * via_new_spec(struct hda_codec *codec) codec->spec = spec; spec->codec = codec; + spec->codec_type = get_codec_type(codec); + /* VT1708BCE & VT1708S are almost same */ + if (spec->codec_type == VT1708BCE) + spec->codec_type = VT1708S; return spec; } @@ -1101,6 +1106,7 @@ static int via_mux_enum_put(struct snd_kcontrol *kcontrol, struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct via_spec *spec = codec->spec; unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + int ret; if (!spec->mux_nids[adc_idx]) return -EINVAL; @@ -1109,12 +1115,14 @@ static int via_mux_enum_put(struct snd_kcontrol *kcontrol, AC_VERB_GET_POWER_STATE, 0x00) != AC_PWRST_D0) snd_hda_codec_write(codec, spec->mux_nids[adc_idx], 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - /* update jack power state */ - set_jack_power_state(codec); - return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, + ret = snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, spec->mux_nids[adc_idx], &spec->cur_mux[adc_idx]); + /* update jack power state */ + set_jack_power_state(codec); + + return ret; } static int via_independent_hp_info(struct snd_kcontrol *kcontrol, @@ -1188,8 +1196,16 @@ static int via_independent_hp_put(struct snd_kcontrol *kcontrol, /* Get Independent Mode index of headphone pin widget */ spec->hp_independent_mode = spec->hp_independent_mode_index == pinsel ? 1 : 0; - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, pinsel); + if (spec->codec_type == VT1718S) + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_CONNECT_SEL, pinsel ? 2 : 0); + else + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_CONNECT_SEL, pinsel); + if (spec->codec_type == VT1812) + snd_hda_codec_write(codec, 0x35, 0, + AC_VERB_SET_CONNECT_SEL, pinsel); if (spec->multiout.hp_nid && spec->multiout.hp_nid != spec->multiout.dac_nids[HDA_FRONT]) snd_hda_codec_setup_stream(codec, spec->multiout.hp_nid, @@ -1208,6 +1224,8 @@ static int via_independent_hp_put(struct snd_kcontrol *kcontrol, activate_ctl(codec, "Headphone Playback Switch", spec->hp_independent_mode); } + /* update jack power state */ + set_jack_power_state(codec); return 0; } @@ -1248,9 +1266,12 @@ static int via_hp_build(struct hda_codec *codec) break; } - nums = snd_hda_get_connections(codec, nid, conn, HDA_MAX_CONNECTIONS); - if (nums <= 1) - return 0; + if (spec->codec_type != VT1708) { + nums = snd_hda_get_connections(codec, nid, + conn, HDA_MAX_CONNECTIONS); + if (nums <= 1) + return 0; + } knew = via_clone_control(spec, &via_hp_mixer[0]); if (knew == NULL) @@ -1310,6 +1331,11 @@ static void mute_aa_path(struct hda_codec *codec, int mute) start_idx = 2; end_idx = 4; break; + case VT1718S: + nid_mixer = 0x21; + start_idx = 1; + end_idx = 3; + break; default: return; } @@ -2185,10 +2211,6 @@ static int via_init(struct hda_codec *codec) for (i = 0; i < spec->num_iverbs; i++) snd_hda_sequence_write(codec, spec->init_verbs[i]); - spec->codec_type = get_codec_type(codec); - if (spec->codec_type == VT1708BCE) - spec->codec_type = VT1708S; /* VT1708BCE & VT1708S are almost - same */ /* Lydia Add for EAPD enable */ if (!spec->dig_in_nid) { /* No Digital In connection */ if (spec->dig_in_pin) { @@ -2438,7 +2460,14 @@ static int vt_auto_create_analog_input_ctls(struct hda_codec *codec, else type_idx = 0; label = hda_get_autocfg_input_label(codec, cfg, i); - err = via_new_analog_input(spec, label, type_idx, idx, cap_nid); + if (spec->codec_type == VT1708S || + spec->codec_type == VT1702 || + spec->codec_type == VT1716S) + err = via_new_analog_input(spec, label, type_idx, + idx+1, cap_nid); + else + err = via_new_analog_input(spec, label, type_idx, + idx, cap_nid); if (err < 0) return err; snd_hda_add_imux_item(imux, label, idx, NULL); @@ -4147,6 +4176,11 @@ static int patch_vt1708S(struct hda_codec *codec) spec->stream_name_analog = "VT1708BCE Analog"; spec->stream_name_digital = "VT1708BCE Digital"; } + /* correct names for VT1818S */ + if (codec->vendor_id == 0x11060440) { + spec->stream_name_analog = "VT1818S Analog"; + spec->stream_name_digital = "VT1818S Digital"; + } return 0; } diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index d63c1754e05..6943e24a74a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -51,7 +51,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TWL6040 if TWL4030_CORE select SND_SOC_UDA134X select SND_SOC_UDA1380 if I2C - select SND_SOC_WL1273 if RADIO_WL1273 + select SND_SOC_WL1273 if MFD_WL1273_CORE select SND_SOC_WM2000 if I2C select SND_SOC_WM8350 if MFD_WM8350 select SND_SOC_WM8400 if MFD_WM8400 diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c index 347a567b01e..b8066ef10bb 100644 --- a/sound/soc/codecs/cq93vc.c +++ b/sound/soc/codecs/cq93vc.c @@ -153,7 +153,8 @@ static int cq93vc_resume(struct snd_soc_codec *codec) static int cq93vc_probe(struct snd_soc_codec *codec) { - struct davinci_vc *davinci_vc = snd_soc_codec_get_drvdata(codec); + struct davinci_vc *davinci_vc = + mfd_get_data(to_platform_device(codec->dev)); davinci_vc->cq93vc.codec = codec; codec->control_data = davinci_vc; diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 1f7217f703e..ff29380c9ed 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -772,6 +772,7 @@ static int sgtl5000_pcm_hw_params(struct snd_pcm_substream *substream, return 0; } +#ifdef CONFIG_REGULATOR static int ldo_regulator_is_enabled(struct regulator_dev *dev) { struct ldo_regulator *ldo = rdev_get_drvdata(dev); @@ -901,6 +902,19 @@ static int ldo_regulator_remove(struct snd_soc_codec *codec) return 0; } +#else +static int ldo_regulator_register(struct snd_soc_codec *codec, + struct regulator_init_data *init_data, + int voltage) +{ + return -EINVAL; +} + +static int ldo_regulator_remove(struct snd_soc_codec *codec) +{ + return 0; +} +#endif /* * set dac bias diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index e4d464b937d..8512800f632 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -26,6 +26,7 @@ #include <linux/pm.h> #include <linux/i2c.h> #include <linux/platform_device.h> +#include <linux/mfd/core.h> #include <linux/i2c/twl.h> #include <linux/slab.h> #include <sound/core.h> @@ -732,7 +733,8 @@ static int aif_event(struct snd_soc_dapm_widget *w, static void headset_ramp(struct snd_soc_codec *codec, int ramp) { - struct twl4030_codec_audio_data *pdata = codec->dev->platform_data; + struct twl4030_codec_audio_data *pdata = + mfd_get_data(to_platform_device(codec->dev)); unsigned char hs_gain, hs_pop; struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); /* Base values for ramp delay calculation: 2^19 - 2^26 */ @@ -2297,7 +2299,7 @@ static struct snd_soc_codec_driver soc_codec_dev_twl4030 = { static int __devinit twl4030_codec_probe(struct platform_device *pdev) { - struct twl4030_codec_audio_data *pdata = pdev->dev.platform_data; + struct twl4030_codec_audio_data *pdata = mfd_get_data(pdev); if (!pdata) { dev_err(&pdev->dev, "platform_data is missing\n"); diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index e76847a9438..48ffd406a71 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -486,7 +486,8 @@ static struct snd_soc_dai_driver uda134x_dai = { static int uda134x_soc_probe(struct snd_soc_codec *codec) { struct uda134x_priv *uda134x; - struct uda134x_platform_data *pd = dev_get_drvdata(codec->card->dev); + struct uda134x_platform_data *pd = codec->card->dev->platform_data; + int ret; printk(KERN_INFO "UDA134X SoC Audio Codec\n"); diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c index 861b28f543d..c8a874d0d4c 100644 --- a/sound/soc/codecs/wl1273.c +++ b/sound/soc/codecs/wl1273.c @@ -3,7 +3,7 @@ * * Author: Matti Aaltonen, <matti.j.aaltonen@nokia.com> * - * Copyright: (C) 2010 Nokia Corporation + * Copyright: (C) 2010, 2011 Nokia Corporation * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License @@ -179,7 +179,12 @@ static int snd_wl1273_get_audio_route(struct snd_kcontrol *kcontrol, return 0; } -static const char *wl1273_audio_route[] = { "Bt", "FmRx", "FmTx" }; +/* + * TODO: Implement the audio routing in the driver. Now this control + * only indicates the setting that has been done elsewhere (in the user + * space). + */ +static const char * const wl1273_audio_route[] = { "Bt", "FmRx", "FmTx" }; static int snd_wl1273_set_audio_route(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -239,7 +244,7 @@ static int snd_wl1273_fm_audio_put(struct snd_kcontrol *kcontrol, return 1; } -static const char *wl1273_audio_strings[] = { "Digital", "Analog" }; +static const char * const wl1273_audio_strings[] = { "Digital", "Analog" }; static const struct soc_enum wl1273_audio_enum = SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(wl1273_audio_strings), @@ -436,7 +441,8 @@ EXPORT_SYMBOL_GPL(wl1273_get_format); static int wl1273_probe(struct snd_soc_codec *codec) { - struct wl1273_core **core = codec->dev->platform_data; + struct wl1273_core **core = + mfd_get_data(to_platform_device(codec->dev)); struct wl1273_priv *wl1273; int r; diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 3c3bc079167..736b785e375 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -22,6 +22,7 @@ #include <linux/regulator/consumer.h> #include <linux/mfd/wm8400-audio.h> #include <linux/mfd/wm8400-private.h> +#include <linux/mfd/core.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -1377,7 +1378,7 @@ static void wm8400_probe_deferred(struct work_struct *work) static int wm8400_codec_probe(struct snd_soc_codec *codec) { - struct wm8400 *wm8400 = dev_get_platdata(codec->dev); + struct wm8400 *wm8400 = mfd_get_data(to_platform_device(codec->dev)); struct wm8400_priv *priv; int ret; u16 reg; diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c index 9d2afccc3a2..13e05a302a9 100644 --- a/sound/soc/davinci/davinci-vcif.c +++ b/sound/soc/davinci/davinci-vcif.c @@ -205,7 +205,7 @@ static struct snd_soc_dai_driver davinci_vcif_dai = { static int davinci_vcif_probe(struct platform_device *pdev) { - struct davinci_vc *davinci_vc = platform_get_drvdata(pdev); + struct davinci_vc *davinci_vc = mfd_get_data(pdev); struct davinci_vcif_dev *davinci_vcif_dev; int ret; diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 4cf98c03af2..15dac0f20cd 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -896,8 +896,7 @@ static struct snd_pcm_ops fsl_dma_ops = { .pointer = fsl_dma_pointer, }; -static int __devinit fsl_soc_dma_probe(struct platform_device *pdev, - const struct of_device_id *match) +static int __devinit fsl_soc_dma_probe(struct platform_device *pdev) { struct dma_object *dma; struct device_node *np = pdev->dev.of_node; @@ -979,7 +978,7 @@ static const struct of_device_id fsl_soc_dma_ids[] = { }; MODULE_DEVICE_TABLE(of, fsl_soc_dma_ids); -static struct of_platform_driver fsl_soc_dma_driver = { +static struct platform_driver fsl_soc_dma_driver = { .driver = { .name = "fsl-pcm-audio", .owner = THIS_MODULE, @@ -993,12 +992,12 @@ static int __init fsl_soc_dma_init(void) { pr_info("Freescale Elo DMA ASoC PCM Driver\n"); - return of_register_platform_driver(&fsl_soc_dma_driver); + return platform_driver_register(&fsl_soc_dma_driver); } static void __exit fsl_soc_dma_exit(void) { - of_unregister_platform_driver(&fsl_soc_dma_driver); + platform_driver_unregister(&fsl_soc_dma_driver); } module_init(fsl_soc_dma_init); diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 4cc167a7aeb..313e0ccedd5 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -624,8 +624,7 @@ static void make_lowercase(char *s) } } -static int __devinit fsl_ssi_probe(struct platform_device *pdev, - const struct of_device_id *match) +static int __devinit fsl_ssi_probe(struct platform_device *pdev) { struct fsl_ssi_private *ssi_private; int ret = 0; @@ -774,7 +773,7 @@ static const struct of_device_id fsl_ssi_ids[] = { }; MODULE_DEVICE_TABLE(of, fsl_ssi_ids); -static struct of_platform_driver fsl_ssi_driver = { +static struct platform_driver fsl_ssi_driver = { .driver = { .name = "fsl-ssi-dai", .owner = THIS_MODULE, @@ -788,12 +787,12 @@ static int __init fsl_ssi_init(void) { printk(KERN_INFO "Freescale Synchronous Serial Interface (SSI) ASoC Driver\n"); - return of_register_platform_driver(&fsl_ssi_driver); + return platform_driver_register(&fsl_ssi_driver); } static void __exit fsl_ssi_exit(void) { - of_unregister_platform_driver(&fsl_ssi_driver); + platform_driver_unregister(&fsl_ssi_driver); } module_init(fsl_ssi_init); diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index f92dca07cd3..fff695ccdd3 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -368,8 +368,7 @@ static struct snd_soc_platform_driver mpc5200_audio_dma_platform = { .pcm_free = &psc_dma_free, }; -static int mpc5200_hpcd_probe(struct of_device *op, - const struct of_device_id *match) +static int mpc5200_hpcd_probe(struct of_device *op) { phys_addr_t fifo; struct psc_dma *psc_dma; @@ -511,32 +510,31 @@ static int mpc5200_hpcd_remove(struct of_device *op) } static struct of_device_id mpc5200_hpcd_match[] = { - { - .compatible = "fsl,mpc5200-pcm", - }, + { .compatible = "fsl,mpc5200-pcm", }, {} }; MODULE_DEVICE_TABLE(of, mpc5200_hpcd_match); -static struct of_platform_driver mpc5200_hpcd_of_driver = { - .owner = THIS_MODULE, - .name = "mpc5200-pcm-audio", - .match_table = mpc5200_hpcd_match, +static struct platform_driver mpc5200_hpcd_of_driver = { .probe = mpc5200_hpcd_probe, .remove = mpc5200_hpcd_remove, + .dev = { + .owner = THIS_MODULE, + .name = "mpc5200-pcm-audio", + .of_match_table = mpc5200_hpcd_match, + } }; static int __init mpc5200_hpcd_init(void) { - return of_register_platform_driver(&mpc5200_hpcd_of_driver); + return platform_driver_register(&mpc5200_hpcd_of_driver); } +module_init(mpc5200_hpcd_init); static void __exit mpc5200_hpcd_exit(void) { - of_unregister_platform_driver(&mpc5200_hpcd_of_driver); + platform_driver_unregister(&mpc5200_hpcd_of_driver); } - -module_init(mpc5200_hpcd_init); module_exit(mpc5200_hpcd_exit); MODULE_AUTHOR("Grant Likely <grant.likely@secretlab.ca>"); diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c index 40acc8e2b1c..ad36b095bb7 100644 --- a/sound/soc/fsl/mpc5200_psc_ac97.c +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -272,8 +272,7 @@ static struct snd_soc_dai_driver psc_ac97_dai[] = { * - Probe/remove operations * - OF device match table */ -static int __devinit psc_ac97_of_probe(struct platform_device *op, - const struct of_device_id *match) +static int __devinit psc_ac97_of_probe(struct platform_device *op) { int rc; struct snd_ac97 ac97; @@ -316,7 +315,7 @@ static struct of_device_id psc_ac97_match[] __devinitdata = { }; MODULE_DEVICE_TABLE(of, psc_ac97_match); -static struct of_platform_driver psc_ac97_driver = { +static struct platform_driver psc_ac97_driver = { .probe = psc_ac97_of_probe, .remove = __devexit_p(psc_ac97_of_remove), .driver = { @@ -332,13 +331,13 @@ static struct of_platform_driver psc_ac97_driver = { */ static int __init psc_ac97_init(void) { - return of_register_platform_driver(&psc_ac97_driver); + return platform_driver_register(&psc_ac97_driver); } module_init(psc_ac97_init); static void __exit psc_ac97_exit(void) { - of_unregister_platform_driver(&psc_ac97_driver); + platform_driver_unregister(&psc_ac97_driver); } module_exit(psc_ac97_exit); diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c index 9018fa5bf0d..87cf2a5c2b2 100644 --- a/sound/soc/fsl/mpc5200_psc_i2s.c +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -150,8 +150,7 @@ static struct snd_soc_dai_driver psc_i2s_dai[] = {{ * - Probe/remove operations * - OF device match table */ -static int __devinit psc_i2s_of_probe(struct platform_device *op, - const struct of_device_id *match) +static int __devinit psc_i2s_of_probe(struct platform_device *op) { int rc; struct psc_dma *psc_dma; @@ -213,7 +212,7 @@ static struct of_device_id psc_i2s_match[] __devinitdata = { }; MODULE_DEVICE_TABLE(of, psc_i2s_match); -static struct of_platform_driver psc_i2s_driver = { +static struct platform_driver psc_i2s_driver = { .probe = psc_i2s_of_probe, .remove = __devexit_p(psc_i2s_of_remove), .driver = { @@ -229,13 +228,13 @@ static struct of_platform_driver psc_i2s_driver = { */ static int __init psc_i2s_init(void) { - return of_register_platform_driver(&psc_i2s_driver); + return platform_driver_register(&psc_i2s_driver); } module_init(psc_i2s_init); static void __exit psc_i2s_exit(void) { - of_unregister_platform_driver(&psc_i2s_driver); + platform_driver_unregister(&psc_i2s_driver); } module_exit(psc_i2s_exit); diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c index 161750443eb..73dde4a1adc 100644 --- a/sound/soc/omap/am3517evm.c +++ b/sound/soc/omap/am3517evm.c @@ -139,7 +139,7 @@ static struct snd_soc_dai_link am3517evm_dai = { .cpu_dai_name ="omap-mcbsp-dai.0", .codec_dai_name = "tlv320aic23-hifi", .platform_name = "omap-pcm-audio", - .codec_name = "tlv320aic23-codec", + .codec_name = "tlv320aic23-codec.2-001a", .init = am3517evm_aic23_init, .ops = &am3517evm_ops, }; diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index d203f4da18a..2175f09e57b 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -69,110 +69,6 @@ static struct omap_mcbsp_data mcbsp_data[NUM_LINKS]; */ static struct omap_pcm_dma_data omap_mcbsp_dai_dma_params[NUM_LINKS][2]; -#if defined(CONFIG_ARCH_OMAP15XX) || defined(CONFIG_ARCH_OMAP16XX) -static const int omap1_dma_reqs[][2] = { - { OMAP_DMA_MCBSP1_TX, OMAP_DMA_MCBSP1_RX }, - { OMAP_DMA_MCBSP2_TX, OMAP_DMA_MCBSP2_RX }, - { OMAP_DMA_MCBSP3_TX, OMAP_DMA_MCBSP3_RX }, -}; -static const unsigned long omap1_mcbsp_port[][2] = { - { OMAP1510_MCBSP1_BASE + OMAP_MCBSP_REG_DXR1, - OMAP1510_MCBSP1_BASE + OMAP_MCBSP_REG_DRR1 }, - { OMAP1510_MCBSP2_BASE + OMAP_MCBSP_REG_DXR1, - OMAP1510_MCBSP2_BASE + OMAP_MCBSP_REG_DRR1 }, - { OMAP1510_MCBSP3_BASE + OMAP_MCBSP_REG_DXR1, - OMAP1510_MCBSP3_BASE + OMAP_MCBSP_REG_DRR1 }, -}; -#else -static const int omap1_dma_reqs[][2] = {}; -static const unsigned long omap1_mcbsp_port[][2] = {}; -#endif - -#if defined(CONFIG_ARCH_OMAP2) || defined(CONFIG_ARCH_OMAP3) -static const int omap24xx_dma_reqs[][2] = { - { OMAP24XX_DMA_MCBSP1_TX, OMAP24XX_DMA_MCBSP1_RX }, - { OMAP24XX_DMA_MCBSP2_TX, OMAP24XX_DMA_MCBSP2_RX }, -#if defined(CONFIG_ARCH_OMAP2430) || defined(CONFIG_ARCH_OMAP3) - { OMAP24XX_DMA_MCBSP3_TX, OMAP24XX_DMA_MCBSP3_RX }, - { OMAP24XX_DMA_MCBSP4_TX, OMAP24XX_DMA_MCBSP4_RX }, - { OMAP24XX_DMA_MCBSP5_TX, OMAP24XX_DMA_MCBSP5_RX }, -#endif -}; -#else -static const int omap24xx_dma_reqs[][2] = {}; -#endif - -#if defined(CONFIG_ARCH_OMAP4) -static const int omap44xx_dma_reqs[][2] = { - { OMAP44XX_DMA_MCBSP1_TX, OMAP44XX_DMA_MCBSP1_RX }, - { OMAP44XX_DMA_MCBSP2_TX, OMAP44XX_DMA_MCBSP2_RX }, - { OMAP44XX_DMA_MCBSP3_TX, OMAP44XX_DMA_MCBSP3_RX }, - { OMAP44XX_DMA_MCBSP4_TX, OMAP44XX_DMA_MCBSP4_RX }, -}; -#else -static const int omap44xx_dma_reqs[][2] = {}; -#endif - -#if defined(CONFIG_ARCH_OMAP2420) -static const unsigned long omap2420_mcbsp_port[][2] = { - { OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR1, - OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR1 }, - { OMAP24XX_MCBSP2_BASE + OMAP_MCBSP_REG_DXR1, - OMAP24XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR1 }, -}; -#else -static const unsigned long omap2420_mcbsp_port[][2] = {}; -#endif - -#if defined(CONFIG_ARCH_OMAP2430) -static const unsigned long omap2430_mcbsp_port[][2] = { - { OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR, - OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR }, - { OMAP24XX_MCBSP2_BASE + OMAP_MCBSP_REG_DXR, - OMAP24XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR }, - { OMAP2430_MCBSP3_BASE + OMAP_MCBSP_REG_DXR, - OMAP2430_MCBSP3_BASE + OMAP_MCBSP_REG_DRR }, - { OMAP2430_MCBSP4_BASE + OMAP_MCBSP_REG_DXR, - OMAP2430_MCBSP4_BASE + OMAP_MCBSP_REG_DRR }, - { OMAP2430_MCBSP5_BASE + OMAP_MCBSP_REG_DXR, - OMAP2430_MCBSP5_BASE + OMAP_MCBSP_REG_DRR }, -}; -#else -static const unsigned long omap2430_mcbsp_port[][2] = {}; -#endif - -#if defined(CONFIG_ARCH_OMAP3) -static const unsigned long omap34xx_mcbsp_port[][2] = { - { OMAP34XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR, - OMAP34XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR }, - { OMAP34XX_MCBSP2_BASE + OMAP_MCBSP_REG_DXR, - OMAP34XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR }, - { OMAP34XX_MCBSP3_BASE + OMAP_MCBSP_REG_DXR, - OMAP34XX_MCBSP3_BASE + OMAP_MCBSP_REG_DRR }, - { OMAP34XX_MCBSP4_BASE + OMAP_MCBSP_REG_DXR, - OMAP34XX_MCBSP4_BASE + OMAP_MCBSP_REG_DRR }, - { OMAP34XX_MCBSP5_BASE + OMAP_MCBSP_REG_DXR, - OMAP34XX_MCBSP5_BASE + OMAP_MCBSP_REG_DRR }, -}; -#else -static const unsigned long omap34xx_mcbsp_port[][2] = {}; -#endif - -#if defined(CONFIG_ARCH_OMAP4) -static const unsigned long omap44xx_mcbsp_port[][2] = { - { OMAP44XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR, - OMAP44XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR }, - { OMAP44XX_MCBSP2_BASE + OMAP_MCBSP_REG_DXR, - OMAP44XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR }, - { OMAP44XX_MCBSP3_BASE + OMAP_MCBSP_REG_DXR, - OMAP44XX_MCBSP3_BASE + OMAP_MCBSP_REG_DRR }, - { OMAP44XX_MCBSP4_BASE + OMAP_MCBSP_REG_DXR, - OMAP44XX_MCBSP4_BASE + OMAP_MCBSP_REG_DRR }, -}; -#else -static const unsigned long omap44xx_mcbsp_port[][2] = {}; -#endif - static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -346,24 +242,10 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, unsigned int format, div, framesize, master; dma_data = &omap_mcbsp_dai_dma_params[cpu_dai->id][substream->stream]; - if (cpu_class_is_omap1()) { - dma = omap1_dma_reqs[bus_id][substream->stream]; - port = omap1_mcbsp_port[bus_id][substream->stream]; - } else if (cpu_is_omap2420()) { - dma = omap24xx_dma_reqs[bus_id][substream->stream]; - port = omap2420_mcbsp_port[bus_id][substream->stream]; - } else if (cpu_is_omap2430()) { - dma = omap24xx_dma_reqs[bus_id][substream->stream]; - port = omap2430_mcbsp_port[bus_id][substream->stream]; - } else if (cpu_is_omap343x()) { - dma = omap24xx_dma_reqs[bus_id][substream->stream]; - port = omap34xx_mcbsp_port[bus_id][substream->stream]; - } else if (cpu_is_omap44xx()) { - dma = omap44xx_dma_reqs[bus_id][substream->stream]; - port = omap44xx_mcbsp_port[bus_id][substream->stream]; - } else { - return -ENODEV; - } + + dma = omap_mcbsp_dma_ch_params(bus_id, substream->stream); + port = omap_mcbsp_dma_reg_params(bus_id, substream->stream); + switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: dma_data->data_type = OMAP_DMA_DATA_TYPE_S16; diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h index 110c106611d..37dc7211ed3 100644 --- a/sound/soc/omap/omap-mcbsp.h +++ b/sound/soc/omap/omap-mcbsp.h @@ -43,7 +43,7 @@ enum omap_mcbsp_div { OMAP_MCBSP_CLKGDV, /* Sample rate generator divider */ }; -#if defined(CONFIG_ARCH_OMAP2420) +#if defined(CONFIG_SOC_OMAP2420) #define NUM_LINKS 2 #endif #if defined(CONFIG_ARCH_OMAP15XX) || defined(CONFIG_ARCH_OMAP16XX) @@ -54,7 +54,7 @@ enum omap_mcbsp_div { #undef NUM_LINKS #define NUM_LINKS 4 #endif -#if defined(CONFIG_ARCH_OMAP2430) || defined(CONFIG_ARCH_OMAP3) +#if defined(CONFIG_ARCH_OMAP3) || defined(CONFIG_SOC_OMAP2430) #undef NUM_LINKS #define NUM_LINKS 5 #endif diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index a08237acc53..a3fdfb63146 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -1,6 +1,6 @@ config SND_SOC_SAMSUNG tristate "ASoC support for Samsung" - depends on ARCH_S3C2410 || ARCH_S3C64XX || ARCH_S5PC100 || ARCH_S5PV210 || ARCH_S5P64X0 || ARCH_S5P6442 || ARCH_S5PV310 + depends on ARCH_S3C2410 || ARCH_S3C64XX || ARCH_S5PC100 || ARCH_S5PV210 || ARCH_S5P64X0 || ARCH_S5P6442 || ARCH_EXYNOS4 select S3C64XX_DMA if ARCH_S3C64XX select S3C2410_DMA if ARCH_S3C2410 help diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c index 3cb70075107..dc9d551f678 100644 --- a/sound/soc/samsung/s3c24xx_uda134x.c +++ b/sound/soc/samsung/s3c24xx_uda134x.c @@ -219,7 +219,7 @@ static struct snd_soc_ops s3c24xx_uda134x_ops = { static struct snd_soc_dai_link s3c24xx_uda134x_dai_link = { .name = "UDA134X", .stream_name = "UDA134X", - .codec_name = "uda134x-hifi", + .codec_name = "uda134x-codec", .codec_dai_name = "uda134x-hifi", .cpu_dai_name = "s3c24xx-iis", .ops = &s3c24xx_uda134x_ops, @@ -314,6 +314,7 @@ static int s3c24xx_uda134x_probe(struct platform_device *pdev) platform_set_drvdata(s3c24xx_uda134x_snd_device, &snd_soc_s3c24xx_uda134x); + platform_device_add_data(s3c24xx_uda134x_snd_device, &s3c24xx_uda134x, sizeof(s3c24xx_uda134x)); ret = platform_device_add(s3c24xx_uda134x_snd_device); if (ret) { printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: Unable to add\n"); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 17efacdb248..4dda58926bc 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -259,8 +259,6 @@ static ssize_t codec_reg_write_file(struct file *file, while (*start == ' ') start++; reg = simple_strtoul(start, &start, 16); - if ((reg >= codec->driver->reg_cache_size) || (reg % step)) - return -EINVAL; while (*start == ' ') start++; if (strict_strtoul(start, 16, &value)) diff --git a/sound/sparc/amd7930.c b/sound/sparc/amd7930.c index f8bcfc30f80..ad7d4d7d923 100644 --- a/sound/sparc/amd7930.c +++ b/sound/sparc/amd7930.c @@ -1002,7 +1002,7 @@ static int __devinit snd_amd7930_create(struct snd_card *card, return 0; } -static int __devinit amd7930_sbus_probe(struct platform_device *op, const struct of_device_id *match) +static int __devinit amd7930_sbus_probe(struct platform_device *op) { struct resource *rp = &op->resource[0]; static int dev_num; @@ -1064,7 +1064,7 @@ static const struct of_device_id amd7930_match[] = { {}, }; -static struct of_platform_driver amd7930_sbus_driver = { +static struct platform_driver amd7930_sbus_driver = { .driver = { .name = "audio", .owner = THIS_MODULE, @@ -1075,7 +1075,7 @@ static struct of_platform_driver amd7930_sbus_driver = { static int __init amd7930_init(void) { - return of_register_platform_driver(&amd7930_sbus_driver); + return platform_driver_register(&amd7930_sbus_driver); } static void __exit amd7930_exit(void) @@ -1092,7 +1092,7 @@ static void __exit amd7930_exit(void) amd7930_list = NULL; - of_unregister_platform_driver(&amd7930_sbus_driver); + platform_driver_unregister(&amd7930_sbus_driver); } module_init(amd7930_init); diff --git a/sound/sparc/cs4231.c b/sound/sparc/cs4231.c index c276086c3b5..0e618f82808 100644 --- a/sound/sparc/cs4231.c +++ b/sound/sparc/cs4231.c @@ -1856,7 +1856,7 @@ static int __devinit snd_cs4231_sbus_create(struct snd_card *card, return 0; } -static int __devinit cs4231_sbus_probe(struct platform_device *op, const struct of_device_id *match) +static int __devinit cs4231_sbus_probe(struct platform_device *op) { struct resource *rp = &op->resource[0]; struct snd_card *card; @@ -2048,7 +2048,7 @@ static int __devinit snd_cs4231_ebus_create(struct snd_card *card, return 0; } -static int __devinit cs4231_ebus_probe(struct platform_device *op, const struct of_device_id *match) +static int __devinit cs4231_ebus_probe(struct platform_device *op) { struct snd_card *card; int err; @@ -2072,16 +2072,16 @@ static int __devinit cs4231_ebus_probe(struct platform_device *op, const struct } #endif -static int __devinit cs4231_probe(struct platform_device *op, const struct of_device_id *match) +static int __devinit cs4231_probe(struct platform_device *op) { #ifdef EBUS_SUPPORT if (!strcmp(op->dev.of_node->parent->name, "ebus")) - return cs4231_ebus_probe(op, match); + return cs4231_ebus_probe(op); #endif #ifdef SBUS_SUPPORT if (!strcmp(op->dev.of_node->parent->name, "sbus") || !strcmp(op->dev.of_node->parent->name, "sbi")) - return cs4231_sbus_probe(op, match); + return cs4231_sbus_probe(op); #endif return -ENODEV; } @@ -2108,7 +2108,7 @@ static const struct of_device_id cs4231_match[] = { MODULE_DEVICE_TABLE(of, cs4231_match); -static struct of_platform_driver cs4231_driver = { +static struct platform_driver cs4231_driver = { .driver = { .name = "audio", .owner = THIS_MODULE, @@ -2120,12 +2120,12 @@ static struct of_platform_driver cs4231_driver = { static int __init cs4231_init(void) { - return of_register_platform_driver(&cs4231_driver); + return platform_driver_register(&cs4231_driver); } static void __exit cs4231_exit(void) { - of_unregister_platform_driver(&cs4231_driver); + platform_driver_unregister(&cs4231_driver); } module_init(cs4231_init); diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index 39cd5d69d05..73f9cbacc07 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -2592,7 +2592,7 @@ static void snd_dbri_free(struct snd_dbri *dbri) (void *)dbri->dma, dbri->dma_dvma); } -static int __devinit dbri_probe(struct platform_device *op, const struct of_device_id *match) +static int __devinit dbri_probe(struct platform_device *op) { struct snd_dbri *dbri; struct resource *rp; @@ -2686,7 +2686,7 @@ static const struct of_device_id dbri_match[] = { MODULE_DEVICE_TABLE(of, dbri_match); -static struct of_platform_driver dbri_sbus_driver = { +static struct platform_driver dbri_sbus_driver = { .driver = { .name = "dbri", .owner = THIS_MODULE, @@ -2699,12 +2699,12 @@ static struct of_platform_driver dbri_sbus_driver = { /* Probe for the dbri chip and then attach the driver. */ static int __init dbri_init(void) { - return of_register_platform_driver(&dbri_sbus_driver); + return platform_driver_register(&dbri_sbus_driver); } static void __exit dbri_exit(void) { - of_unregister_platform_driver(&dbri_sbus_driver); + platform_driver_unregister(&dbri_sbus_driver); } module_init(dbri_init); diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index c0dcfca9b5b..c66d3f64dcf 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1568,6 +1568,46 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { + USB_DEVICE_VENDOR_SPEC(0x0582, 0x0104), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "Roland", */ + /* .product_name = "UM-1G", */ + .ifnum = 0, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0001, + .in_cables = 0x0001 + } + } +}, +{ + /* Boss JS-8 Jam Station */ + USB_DEVICE(0x0582, 0x0109), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "BOSS", */ + /* .product_name = "JS-8", */ + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_MIDI_STANDARD_INTERFACE + }, + { + .ifnum = -1 + } + } + } +}, +{ /* has ID 0x0110 when not in Advanced Driver mode */ USB_DEVICE_VENDOR_SPEC(0x0582, 0x010f), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { |