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-rw-r--r--sound/core/pcm_native.c12
-rw-r--r--sound/core/seq/seq_timer.c8
-rw-r--r--sound/oss/sequencer.c6
-rw-r--r--sound/pci/asihpi/asihpi.c3
-rw-r--r--sound/pci/hda/hda_codec.c28
-rw-r--r--sound/pci/hda/hda_eld.c2
-rw-r--r--sound/pci/hda/hda_generic.c48
-rw-r--r--sound/pci/hda/hda_intel.c138
-rw-r--r--sound/pci/hda/patch_ca0132.c28
-rw-r--r--sound/pci/hda/patch_cirrus.c8
-rw-r--r--sound/pci/hda/patch_conexant.c16
-rw-r--r--sound/pci/hda/patch_hdmi.c2
-rw-r--r--sound/pci/hda/patch_realtek.c4
-rw-r--r--sound/pci/hda/patch_sigmatel.c29
-rw-r--r--[-rwxr-xr-x]sound/soc/codecs/max98090.c0
-rw-r--r--[-rwxr-xr-x]sound/soc/codecs/max98090.h0
-rw-r--r--sound/soc/codecs/si476x.c1
-rw-r--r--sound/soc/codecs/wm5102.c2
-rw-r--r--sound/soc/codecs/wm8903.c2
-rw-r--r--sound/soc/codecs/wm_adsp.c5
-rw-r--r--sound/soc/fsl/imx-ssi.c5
-rw-r--r--sound/soc/fsl/pcm030-audio-fabric.c2
-rw-r--r--sound/soc/samsung/i2s.c17
-rw-r--r--sound/soc/sh/dma-sh7760.c4
-rw-r--r--sound/soc/soc-compress.c14
-rw-r--r--sound/soc/soc-core.c10
-rw-r--r--sound/soc/soc-dapm.c14
-rw-r--r--sound/soc/spear/spear_pcm.c12
-rw-r--r--sound/soc/tegra/tegra_pcm.c24
-rw-r--r--sound/usb/card.c15
-rw-r--r--sound/usb/clock.c45
-rw-r--r--sound/usb/mixer.c21
-rw-r--r--sound/usb/mixer_quirks.c4
-rw-r--r--sound/usb/quirks.c2
34 files changed, 384 insertions, 147 deletions
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 71ae86ca64ac..eb560fa32321 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -3222,18 +3222,10 @@ EXPORT_SYMBOL_GPL(snd_pcm_lib_default_mmap);
int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream,
struct vm_area_struct *area)
{
- long size;
- unsigned long offset;
+ struct snd_pcm_runtime *runtime = substream->runtime;;
area->vm_page_prot = pgprot_noncached(area->vm_page_prot);
- area->vm_flags |= VM_IO;
- size = area->vm_end - area->vm_start;
- offset = area->vm_pgoff << PAGE_SHIFT;
- if (io_remap_pfn_range(area, area->vm_start,
- (substream->runtime->dma_addr + offset) >> PAGE_SHIFT,
- size, area->vm_page_prot))
- return -EAGAIN;
- return 0;
+ return vm_iomap_memory(area, runtime->dma_addr, runtime->dma_bytes);
}
EXPORT_SYMBOL(snd_pcm_lib_mmap_iomem);
diff --git a/sound/core/seq/seq_timer.c b/sound/core/seq/seq_timer.c
index 160b1bd0cd62..24d44b2f61ac 100644
--- a/sound/core/seq/seq_timer.c
+++ b/sound/core/seq/seq_timer.c
@@ -290,10 +290,10 @@ int snd_seq_timer_open(struct snd_seq_queue *q)
tid.device = SNDRV_TIMER_GLOBAL_SYSTEM;
err = snd_timer_open(&t, str, &tid, q->queue);
}
- if (err < 0) {
- snd_printk(KERN_ERR "seq fatal error: cannot create timer (%i)\n", err);
- return err;
- }
+ }
+ if (err < 0) {
+ snd_printk(KERN_ERR "seq fatal error: cannot create timer (%i)\n", err);
+ return err;
}
t->callback = snd_seq_timer_interrupt;
t->callback_data = q;
diff --git a/sound/oss/sequencer.c b/sound/oss/sequencer.c
index 30bcfe470f83..4ff60a6427d9 100644
--- a/sound/oss/sequencer.c
+++ b/sound/oss/sequencer.c
@@ -545,6 +545,9 @@ static void seq_chn_common_event(unsigned char *event_rec)
case MIDI_PGM_CHANGE:
if (seq_mode == SEQ_2)
{
+ if (chn > 15)
+ break;
+
synth_devs[dev]->chn_info[chn].pgm_num = p1;
if ((int) dev >= num_synths)
synth_devs[dev]->set_instr(dev, chn, p1);
@@ -596,6 +599,9 @@ static void seq_chn_common_event(unsigned char *event_rec)
case MIDI_PITCH_BEND:
if (seq_mode == SEQ_2)
{
+ if (chn > 15)
+ break;
+
synth_devs[dev]->chn_info[chn].bender_value = w14;
if ((int) dev < num_synths)
diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c
index 3536b076b529..0aabfedeecba 100644
--- a/sound/pci/asihpi/asihpi.c
+++ b/sound/pci/asihpi/asihpi.c
@@ -2549,7 +2549,7 @@ static int snd_asihpi_sampleclock_add(struct snd_card_asihpi *asihpi,
static int snd_card_asihpi_mixer_new(struct snd_card_asihpi *asihpi)
{
- struct snd_card *card = asihpi->card;
+ struct snd_card *card;
unsigned int idx = 0;
unsigned int subindex = 0;
int err;
@@ -2557,6 +2557,7 @@ static int snd_card_asihpi_mixer_new(struct snd_card_asihpi *asihpi)
if (snd_BUG_ON(!asihpi))
return -EINVAL;
+ card = asihpi->card;
strcpy(card->mixername, "Asihpi Mixer");
err =
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 97c68dd24ef5..4aba7646dd9c 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -173,7 +173,7 @@ const char *snd_hda_get_jack_type(u32 cfg)
"Line Out", "Speaker", "HP Out", "CD",
"SPDIF Out", "Digital Out", "Modem Line", "Modem Hand",
"Line In", "Aux", "Mic", "Telephony",
- "SPDIF In", "Digitial In", "Reserved", "Other"
+ "SPDIF In", "Digital In", "Reserved", "Other"
};
return jack_types[(cfg & AC_DEFCFG_DEVICE)
@@ -494,7 +494,7 @@ static unsigned int get_num_conns(struct hda_codec *codec, hda_nid_t nid)
int snd_hda_get_num_raw_conns(struct hda_codec *codec, hda_nid_t nid)
{
- return get_num_conns(codec, nid) & AC_CLIST_LENGTH;
+ return snd_hda_get_raw_connections(codec, nid, NULL, 0);
}
/**
@@ -517,9 +517,6 @@ int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid,
hda_nid_t prev_nid;
int null_count = 0;
- if (snd_BUG_ON(!conn_list || max_conns <= 0))
- return -EINVAL;
-
parm = get_num_conns(codec, nid);
if (!parm)
return 0;
@@ -545,7 +542,8 @@ int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid,
AC_VERB_GET_CONNECT_LIST, 0);
if (parm == -1 && codec->bus->rirb_error)
return -EIO;
- conn_list[0] = parm & mask;
+ if (conn_list)
+ conn_list[0] = parm & mask;
return 1;
}
@@ -580,14 +578,20 @@ int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid,
continue;
}
for (n = prev_nid + 1; n <= val; n++) {
+ if (conn_list) {
+ if (conns >= max_conns)
+ return -ENOSPC;
+ conn_list[conns] = n;
+ }
+ conns++;
+ }
+ } else {
+ if (conn_list) {
if (conns >= max_conns)
return -ENOSPC;
- conn_list[conns++] = n;
+ conn_list[conns] = val;
}
- } else {
- if (conns >= max_conns)
- return -ENOSPC;
- conn_list[conns++] = val;
+ conns++;
}
prev_nid = val;
}
@@ -3140,7 +3144,7 @@ static unsigned int convert_to_spdif_status(unsigned short val)
if (val & AC_DIG1_PROFESSIONAL)
sbits |= IEC958_AES0_PROFESSIONAL;
if (sbits & IEC958_AES0_PROFESSIONAL) {
- if (sbits & AC_DIG1_EMPHASIS)
+ if (val & AC_DIG1_EMPHASIS)
sbits |= IEC958_AES0_PRO_EMPHASIS_5015;
} else {
if (val & AC_DIG1_EMPHASIS)
diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index 7dd846380a50..d0d7ac1e99d2 100644
--- a/sound/pci/hda/hda_eld.c
+++ b/sound/pci/hda/hda_eld.c
@@ -320,7 +320,7 @@ int snd_hdmi_get_eld(struct hda_codec *codec, hda_nid_t nid,
unsigned char *buf, int *eld_size)
{
int i;
- int ret;
+ int ret = 0;
int size;
/*
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 78897d05d80f..2dbe767be16b 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -740,7 +740,7 @@ EXPORT_SYMBOL_HDA(snd_hda_activate_path);
static void path_power_down_sync(struct hda_codec *codec, struct nid_path *path)
{
struct hda_gen_spec *spec = codec->spec;
- bool changed;
+ bool changed = false;
int i;
if (!spec->power_down_unused || path->active)
@@ -995,6 +995,8 @@ enum {
BAD_NO_EXTRA_SURR_DAC = 0x101,
/* Primary DAC shared with main surrounds */
BAD_SHARED_SURROUND = 0x100,
+ /* No independent HP possible */
+ BAD_NO_INDEP_HP = 0x40,
/* Primary DAC shared with main CLFE */
BAD_SHARED_CLFE = 0x10,
/* Primary DAC shared with extra surrounds */
@@ -1392,6 +1394,43 @@ static int check_aamix_out_path(struct hda_codec *codec, int path_idx)
return snd_hda_get_path_idx(codec, path);
}
+/* check whether the independent HP is available with the current config */
+static bool indep_hp_possible(struct hda_codec *codec)
+{
+ struct hda_gen_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ struct nid_path *path;
+ int i, idx;
+
+ if (cfg->line_out_type == AUTO_PIN_HP_OUT)
+ idx = spec->out_paths[0];
+ else
+ idx = spec->hp_paths[0];
+ path = snd_hda_get_path_from_idx(codec, idx);
+ if (!path)
+ return false;
+
+ /* assume no path conflicts unless aamix is involved */
+ if (!spec->mixer_nid || !is_nid_contained(path, spec->mixer_nid))
+ return true;
+
+ /* check whether output paths contain aamix */
+ for (i = 0; i < cfg->line_outs; i++) {
+ if (spec->out_paths[i] == idx)
+ break;
+ path = snd_hda_get_path_from_idx(codec, spec->out_paths[i]);
+ if (path && is_nid_contained(path, spec->mixer_nid))
+ return false;
+ }
+ for (i = 0; i < cfg->speaker_outs; i++) {
+ path = snd_hda_get_path_from_idx(codec, spec->speaker_paths[i]);
+ if (path && is_nid_contained(path, spec->mixer_nid))
+ return false;
+ }
+
+ return true;
+}
+
/* fill the empty entries in the dac array for speaker/hp with the
* shared dac pointed by the paths
*/
@@ -1545,6 +1584,9 @@ static int fill_and_eval_dacs(struct hda_codec *codec,
badness += BAD_MULTI_IO;
}
+ if (spec->indep_hp && !indep_hp_possible(codec))
+ badness += BAD_NO_INDEP_HP;
+
/* re-fill the shared DAC for speaker / headphone */
if (cfg->line_out_type != AUTO_PIN_HP_OUT)
refill_shared_dacs(codec, cfg->hp_outs,
@@ -1758,6 +1800,10 @@ static int parse_output_paths(struct hda_codec *codec)
cfg->speaker_pins, val);
}
+ /* clear indep_hp flag if not available */
+ if (spec->indep_hp && !indep_hp_possible(codec))
+ spec->indep_hp = 0;
+
kfree(best_cfg);
return 0;
}
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 4cea6bb6fade..bcd40ee488e3 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -134,8 +134,8 @@ MODULE_PARM_DESC(power_save, "Automatic power-saving timeout "
* this may give more power-saving, but will take longer time to
* wake up.
*/
-static int power_save_controller = -1;
-module_param(power_save_controller, bint, 0644);
+static bool power_save_controller = 1;
+module_param(power_save_controller, bool, 0644);
MODULE_PARM_DESC(power_save_controller, "Reset controller in power save mode.");
#endif /* CONFIG_PM */
@@ -415,6 +415,8 @@ struct azx_dev {
unsigned int opened :1;
unsigned int running :1;
unsigned int irq_pending :1;
+ unsigned int prepared:1;
+ unsigned int locked:1;
/*
* For VIA:
* A flag to ensure DMA position is 0
@@ -426,8 +428,25 @@ struct azx_dev {
struct timecounter azx_tc;
struct cyclecounter azx_cc;
+
+#ifdef CONFIG_SND_HDA_DSP_LOADER
+ struct mutex dsp_mutex;
+#endif
};
+/* DSP lock helpers */
+#ifdef CONFIG_SND_HDA_DSP_LOADER
+#define dsp_lock_init(dev) mutex_init(&(dev)->dsp_mutex)
+#define dsp_lock(dev) mutex_lock(&(dev)->dsp_mutex)
+#define dsp_unlock(dev) mutex_unlock(&(dev)->dsp_mutex)
+#define dsp_is_locked(dev) ((dev)->locked)
+#else
+#define dsp_lock_init(dev) do {} while (0)
+#define dsp_lock(dev) do {} while (0)
+#define dsp_unlock(dev) do {} while (0)
+#define dsp_is_locked(dev) 0
+#endif
+
/* CORB/RIRB */
struct azx_rb {
u32 *buf; /* CORB/RIRB buffer
@@ -527,6 +546,10 @@ struct azx {
/* card list (for power_save trigger) */
struct list_head list;
+
+#ifdef CONFIG_SND_HDA_DSP_LOADER
+ struct azx_dev saved_azx_dev;
+#endif
};
#define CREATE_TRACE_POINTS
@@ -1793,15 +1816,25 @@ azx_assign_device(struct azx *chip, struct snd_pcm_substream *substream)
dev = chip->capture_index_offset;
nums = chip->capture_streams;
}
- for (i = 0; i < nums; i++, dev++)
- if (!chip->azx_dev[dev].opened) {
- res = &chip->azx_dev[dev];
- if (res->assigned_key == key)
- break;
+ for (i = 0; i < nums; i++, dev++) {
+ struct azx_dev *azx_dev = &chip->azx_dev[dev];
+ dsp_lock(azx_dev);
+ if (!azx_dev->opened && !dsp_is_locked(azx_dev)) {
+ res = azx_dev;
+ if (res->assigned_key == key) {
+ res->opened = 1;
+ res->assigned_key = key;
+ dsp_unlock(azx_dev);
+ return azx_dev;
+ }
}
+ dsp_unlock(azx_dev);
+ }
if (res) {
+ dsp_lock(res);
res->opened = 1;
res->assigned_key = key;
+ dsp_unlock(res);
}
return res;
}
@@ -2009,6 +2042,12 @@ static int azx_pcm_hw_params(struct snd_pcm_substream *substream,
struct azx_dev *azx_dev = get_azx_dev(substream);
int ret;
+ dsp_lock(azx_dev);
+ if (dsp_is_locked(azx_dev)) {
+ ret = -EBUSY;
+ goto unlock;
+ }
+
mark_runtime_wc(chip, azx_dev, substream, false);
azx_dev->bufsize = 0;
azx_dev->period_bytes = 0;
@@ -2016,8 +2055,10 @@ static int azx_pcm_hw_params(struct snd_pcm_substream *substream,
ret = snd_pcm_lib_malloc_pages(substream,
params_buffer_bytes(hw_params));
if (ret < 0)
- return ret;
+ goto unlock;
mark_runtime_wc(chip, azx_dev, substream, true);
+ unlock:
+ dsp_unlock(azx_dev);
return ret;
}
@@ -2029,16 +2070,21 @@ static int azx_pcm_hw_free(struct snd_pcm_substream *substream)
struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream];
/* reset BDL address */
- azx_sd_writel(azx_dev, SD_BDLPL, 0);
- azx_sd_writel(azx_dev, SD_BDLPU, 0);
- azx_sd_writel(azx_dev, SD_CTL, 0);
- azx_dev->bufsize = 0;
- azx_dev->period_bytes = 0;
- azx_dev->format_val = 0;
+ dsp_lock(azx_dev);
+ if (!dsp_is_locked(azx_dev)) {
+ azx_sd_writel(azx_dev, SD_BDLPL, 0);
+ azx_sd_writel(azx_dev, SD_BDLPU, 0);
+ azx_sd_writel(azx_dev, SD_CTL, 0);
+ azx_dev->bufsize = 0;
+ azx_dev->period_bytes = 0;
+ azx_dev->format_val = 0;
+ }
snd_hda_codec_cleanup(apcm->codec, hinfo, substream);
mark_runtime_wc(chip, azx_dev, substream, false);
+ azx_dev->prepared = 0;
+ dsp_unlock(azx_dev);
return snd_pcm_lib_free_pages(substream);
}
@@ -2055,6 +2101,12 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream)
snd_hda_spdif_out_of_nid(apcm->codec, hinfo->nid);
unsigned short ctls = spdif ? spdif->ctls : 0;
+ dsp_lock(azx_dev);
+ if (dsp_is_locked(azx_dev)) {
+ err = -EBUSY;
+ goto unlock;
+ }
+
azx_stream_reset(chip, azx_dev);
format_val = snd_hda_calc_stream_format(runtime->rate,
runtime->channels,
@@ -2065,7 +2117,8 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream)
snd_printk(KERN_ERR SFX
"%s: invalid format_val, rate=%d, ch=%d, format=%d\n",
pci_name(chip->pci), runtime->rate, runtime->channels, runtime->format);
- return -EINVAL;
+ err = -EINVAL;
+ goto unlock;
}
bufsize = snd_pcm_lib_buffer_bytes(substream);
@@ -2084,7 +2137,7 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream)
azx_dev->no_period_wakeup = runtime->no_period_wakeup;
err = azx_setup_periods(chip, substream, azx_dev);
if (err < 0)
- return err;
+ goto unlock;
}
/* wallclk has 24Mhz clock source */
@@ -2101,8 +2154,14 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream)
if ((chip->driver_caps & AZX_DCAPS_CTX_WORKAROUND) &&
stream_tag > chip->capture_streams)
stream_tag -= chip->capture_streams;
- return snd_hda_codec_prepare(apcm->codec, hinfo, stream_tag,
+ err = snd_hda_codec_prepare(apcm->codec, hinfo, stream_tag,
azx_dev->format_val, substream);
+
+ unlock:
+ if (!err)
+ azx_dev->prepared = 1;
+ dsp_unlock(azx_dev);
+ return err;
}
static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
@@ -2117,6 +2176,9 @@ static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
azx_dev = get_azx_dev(substream);
trace_azx_pcm_trigger(chip, azx_dev, cmd);
+ if (dsp_is_locked(azx_dev) || !azx_dev->prepared)
+ return -EPIPE;
+
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
rstart = 1;
@@ -2621,17 +2683,27 @@ static int azx_load_dsp_prepare(struct hda_bus *bus, unsigned int format,
struct azx_dev *azx_dev;
int err;
- if (snd_hda_lock_devices(bus))
- return -EBUSY;
+ azx_dev = azx_get_dsp_loader_dev(chip);
+
+ dsp_lock(azx_dev);
+ spin_lock_irq(&chip->reg_lock);
+ if (azx_dev->running || azx_dev->locked) {
+ spin_unlock_irq(&chip->reg_lock);
+ err = -EBUSY;
+ goto unlock;
+ }
+ azx_dev->prepared = 0;
+ chip->saved_azx_dev = *azx_dev;
+ azx_dev->locked = 1;
+ spin_unlock_irq(&chip->reg_lock);
err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV_SG,
snd_dma_pci_data(chip->pci),
byte_size, bufp);
if (err < 0)
- goto unlock;
+ goto err_alloc;
mark_pages_wc(chip, bufp, true);
- azx_dev = azx_get_dsp_loader_dev(chip);
azx_dev->bufsize = byte_size;
azx_dev->period_bytes = byte_size;
azx_dev->format_val = format;
@@ -2649,13 +2721,20 @@ static int azx_load_dsp_prepare(struct hda_bus *bus, unsigned int format,
goto error;
azx_setup_controller(chip, azx_dev);
+ dsp_unlock(azx_dev);
return azx_dev->stream_tag;
error:
mark_pages_wc(chip, bufp, false);
snd_dma_free_pages(bufp);
-unlock:
- snd_hda_unlock_devices(bus);
+ err_alloc:
+ spin_lock_irq(&chip->reg_lock);
+ if (azx_dev->opened)
+ *azx_dev = chip->saved_azx_dev;
+ azx_dev->locked = 0;
+ spin_unlock_irq(&chip->reg_lock);
+ unlock:
+ dsp_unlock(azx_dev);
return err;
}
@@ -2677,9 +2756,10 @@ static void azx_load_dsp_cleanup(struct hda_bus *bus,
struct azx *chip = bus->private_data;
struct azx_dev *azx_dev = azx_get_dsp_loader_dev(chip);
- if (!dmab->area)
+ if (!dmab->area || !azx_dev->locked)
return;
+ dsp_lock(azx_dev);
/* reset BDL address */
azx_sd_writel(azx_dev, SD_BDLPL, 0);
azx_sd_writel(azx_dev, SD_BDLPU, 0);
@@ -2692,7 +2772,12 @@ static void azx_load_dsp_cleanup(struct hda_bus *bus,
snd_dma_free_pages(dmab);
dmab->area = NULL;
- snd_hda_unlock_devices(bus);
+ spin_lock_irq(&chip->reg_lock);
+ if (azx_dev->opened)
+ *azx_dev = chip->saved_azx_dev;
+ azx_dev->locked = 0;
+ spin_unlock_irq(&chip->reg_lock);
+ dsp_unlock(azx_dev);
}
#endif /* CONFIG_SND_HDA_DSP_LOADER */
@@ -2846,8 +2931,6 @@ static int azx_runtime_idle(struct device *dev)
struct snd_card *card = dev_get_drvdata(dev);
struct azx *chip = card->private_data;
- if (power_save_controller > 0)
- return 0;
if (!power_save_controller ||
!(chip->driver_caps & AZX_DCAPS_PM_RUNTIME))
return -EBUSY;
@@ -3481,6 +3564,7 @@ static int azx_first_init(struct azx *chip)
}
for (i = 0; i < chip->num_streams; i++) {
+ dsp_lock_init(&chip->azx_dev[i]);
/* allocate memory for the BDL for each stream */
err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV,
snd_dma_pci_data(chip->pci),
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index eefc4563b2f9..0792b5725f9c 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -3239,7 +3239,7 @@ static int ca0132_set_vipsource(struct hda_codec *codec, int val)
struct ca0132_spec *spec = codec->spec;
unsigned int tmp;
- if (!dspload_is_loaded(codec))
+ if (spec->dsp_state != DSP_DOWNLOADED)
return 0;
/* if CrystalVoice if off, vipsource should be 0 */
@@ -4267,11 +4267,12 @@ static void ca0132_refresh_widget_caps(struct hda_codec *codec)
*/
static void ca0132_setup_defaults(struct hda_codec *codec)
{
+ struct ca0132_spec *spec = codec->spec;
unsigned int tmp;
int num_fx;
int idx, i;
- if (!dspload_is_loaded(codec))
+ if (spec->dsp_state != DSP_DOWNLOADED)
return;
/* out, in effects + voicefx */
@@ -4351,12 +4352,16 @@ static bool ca0132_download_dsp_images(struct hda_codec *codec)
return false;
dsp_os_image = (struct dsp_image_seg *)(fw_entry->data);
- dspload_image(codec, dsp_os_image, 0, 0, true, 0);
+ if (dspload_image(codec, dsp_os_image, 0, 0, true, 0)) {
+ pr_err("ca0132 dspload_image failed.\n");
+ goto exit_download;
+ }
+
dsp_loaded = dspload_wait_loaded(codec);
+exit_download:
release_firmware(fw_entry);
-
return dsp_loaded;
}
@@ -4367,16 +4372,13 @@ static void ca0132_download_dsp(struct hda_codec *codec)
#ifndef CONFIG_SND_HDA_CODEC_CA0132_DSP
return; /* NOP */
#endif
- spec->dsp_state = DSP_DOWNLOAD_INIT;
- if (spec->dsp_state == DSP_DOWNLOAD_INIT) {
- chipio_enable_clocks(codec);
- spec->dsp_state = DSP_DOWNLOADING;
- if (!ca0132_download_dsp_images(codec))
- spec->dsp_state = DSP_DOWNLOAD_FAILED;
- else
- spec->dsp_state = DSP_DOWNLOADED;
- }
+ chipio_enable_clocks(codec);
+ spec->dsp_state = DSP_DOWNLOADING;
+ if (!ca0132_download_dsp_images(codec))
+ spec->dsp_state = DSP_DOWNLOAD_FAILED;
+ else
+ spec->dsp_state = DSP_DOWNLOADED;
if (spec->dsp_state == DSP_DOWNLOADED)
ca0132_set_dsp_msr(codec, true);
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 72ebb8a36b13..0d9c58f13560 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -168,10 +168,10 @@ static void cs_automute(struct hda_codec *codec)
snd_hda_gen_update_outputs(codec);
if (spec->gpio_eapd_hp) {
- unsigned int gpio = spec->gen.hp_jack_present ?
+ spec->gpio_data = spec->gen.hp_jack_present ?
spec->gpio_eapd_hp : spec->gpio_eapd_speaker;
snd_hda_codec_write(codec, 0x01, 0,
- AC_VERB_SET_GPIO_DATA, gpio);
+ AC_VERB_SET_GPIO_DATA, spec->gpio_data);
}
}
@@ -506,6 +506,8 @@ static int patch_cs420x(struct hda_codec *codec)
if (!spec)
return -ENOMEM;
+ spec->gen.automute_hook = cs_automute;
+
snd_hda_pick_fixup(codec, cs420x_models, cs420x_fixup_tbl,
cs420x_fixups);
snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
@@ -893,6 +895,8 @@ static int patch_cs4210(struct hda_codec *codec)
if (!spec)
return -ENOMEM;
+ spec->gen.automute_hook = cs_automute;
+
snd_hda_pick_fixup(codec, cs421x_models, cs421x_fixup_tbl,
cs421x_fixups);
snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 941bf6c766ec..2a89d1eefeb6 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -1142,7 +1142,7 @@ static int patch_cxt5045(struct hda_codec *codec)
}
if (spec->beep_amp)
- snd_hda_attach_beep_device(codec, spec->beep_amp);
+ snd_hda_attach_beep_device(codec, get_amp_nid_(spec->beep_amp));
return 0;
}
@@ -1921,7 +1921,7 @@ static int patch_cxt5051(struct hda_codec *codec)
}
if (spec->beep_amp)
- snd_hda_attach_beep_device(codec, spec->beep_amp);
+ snd_hda_attach_beep_device(codec, get_amp_nid_(spec->beep_amp));
return 0;
}
@@ -3099,7 +3099,7 @@ static int patch_cxt5066(struct hda_codec *codec)
}
if (spec->beep_amp)
- snd_hda_attach_beep_device(codec, spec->beep_amp);
+ snd_hda_attach_beep_device(codec, get_amp_nid_(spec->beep_amp));
return 0;
}
@@ -3191,11 +3191,17 @@ static int cx_auto_build_controls(struct hda_codec *codec)
return 0;
}
+static void cx_auto_free(struct hda_codec *codec)
+{
+ snd_hda_detach_beep_device(codec);
+ snd_hda_gen_free(codec);
+}
+
static const struct hda_codec_ops cx_auto_patch_ops = {
.build_controls = cx_auto_build_controls,
.build_pcms = snd_hda_gen_build_pcms,
.init = snd_hda_gen_init,
- .free = snd_hda_gen_free,
+ .free = cx_auto_free,
.unsol_event = snd_hda_jack_unsol_event,
#ifdef CONFIG_PM
.check_power_status = snd_hda_gen_check_power_status,
@@ -3391,7 +3397,7 @@ static int patch_conexant_auto(struct hda_codec *codec)
codec->patch_ops = cx_auto_patch_ops;
if (spec->beep_amp)
- snd_hda_attach_beep_device(codec, spec->beep_amp);
+ snd_hda_attach_beep_device(codec, get_amp_nid_(spec->beep_amp));
/* Some laptops with Conexant chips show stalls in S3 resume,
* which falls into the single-cmd mode.
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 78e1827d0a95..de8ac5c07fd0 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1196,7 +1196,7 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll)
_snd_printd(SND_PR_VERBOSE,
"HDMI status: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n",
- codec->addr, pin_nid, eld->monitor_present, eld->eld_valid);
+ codec->addr, pin_nid, pin_eld->monitor_present, eld->eld_valid);
if (eld->eld_valid) {
if (snd_hdmi_get_eld(codec, pin_nid, eld->eld_buffer,
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 563c24df4d6f..f15c36bde540 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -3440,7 +3440,8 @@ static int alc662_parse_auto_config(struct hda_codec *codec)
const hda_nid_t *ssids;
if (codec->vendor_id == 0x10ec0272 || codec->vendor_id == 0x10ec0663 ||
- codec->vendor_id == 0x10ec0665 || codec->vendor_id == 0x10ec0670)
+ codec->vendor_id == 0x10ec0665 || codec->vendor_id == 0x10ec0670 ||
+ codec->vendor_id == 0x10ec0671)
ssids = alc663_ssids;
else
ssids = alc662_ssids;
@@ -3894,6 +3895,7 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = {
{ .id = 0x10ec0665, .name = "ALC665", .patch = patch_alc662 },
{ .id = 0x10ec0668, .name = "ALC668", .patch = patch_alc662 },
{ .id = 0x10ec0670, .name = "ALC670", .patch = patch_alc662 },
+ { .id = 0x10ec0671, .name = "ALC671", .patch = patch_alc662 },
{ .id = 0x10ec0680, .name = "ALC680", .patch = patch_alc680 },
{ .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 },
{ .id = 0x10ec0882, .name = "ALC882", .patch = patch_alc882 },
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 83d5335ac348..dafe04ae8c72 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -815,6 +815,29 @@ static int find_mute_led_cfg(struct hda_codec *codec, int default_polarity)
return 0;
}
+/* check whether a built-in speaker is included in parsed pins */
+static bool has_builtin_speaker(struct hda_codec *codec)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ hda_nid_t *nid_pin;
+ int nids, i;
+
+ if (spec->gen.autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT) {
+ nid_pin = spec->gen.autocfg.line_out_pins;
+ nids = spec->gen.autocfg.line_outs;
+ } else {
+ nid_pin = spec->gen.autocfg.speaker_pins;
+ nids = spec->gen.autocfg.speaker_outs;
+ }
+
+ for (i = 0; i < nids; i++) {
+ unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid_pin[i]);
+ if (snd_hda_get_input_pin_attr(def_conf) == INPUT_PIN_ATTR_INT)
+ return true;
+ }
+ return false;
+}
+
/*
* PC beep controls
*/
@@ -3890,6 +3913,12 @@ static int patch_stac92hd73xx(struct hda_codec *codec)
return err;
}
+ /* Don't GPIO-mute speakers if there are no internal speakers, because
+ * the GPIO might be necessary for Headphone
+ */
+ if (spec->eapd_switch && !has_builtin_speaker(codec))
+ spec->eapd_switch = 0;
+
codec->proc_widget_hook = stac92hd7x_proc_hook;
snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE);
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index fc176044994d..fc176044994d 100755..100644
--- a/sound/soc/codecs/max98090.c
+++ b/sound/soc/codecs/max98090.c
diff --git a/sound/soc/codecs/max98090.h b/sound/soc/codecs/max98090.h
index 7e103f249053..7e103f249053 100755..100644
--- a/sound/soc/codecs/max98090.h
+++ b/sound/soc/codecs/max98090.h
diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c
index f2d61a187830..566ea3256e2d 100644
--- a/sound/soc/codecs/si476x.c
+++ b/sound/soc/codecs/si476x.c
@@ -159,6 +159,7 @@ static int si476x_codec_hw_params(struct snd_pcm_substream *substream,
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S8:
width = SI476X_PCM_FORMAT_S8;
+ break;
case SNDRV_PCM_FORMAT_S16_LE:
width = SI476X_PCM_FORMAT_S16_LE;
break;
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index b82bbf584146..34d0201d6a78 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -584,7 +584,7 @@ static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_codec *codec = w->codec;
- struct arizona *arizona = dev_get_drvdata(codec->dev);
+ struct arizona *arizona = dev_get_drvdata(codec->dev->parent);
struct regmap *regmap = codec->control_data;
const struct reg_default *patch = NULL;
int i, patch_size;
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index 134e41c870b9..f8a31ad0b203 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -1083,6 +1083,8 @@ static const struct snd_soc_dapm_route wm8903_intercon[] = {
{ "ROP", NULL, "Right Speaker PGA" },
{ "RON", NULL, "Right Speaker PGA" },
+ { "Charge Pump", NULL, "CLK_DSP" },
+
{ "Left Headphone Output PGA", NULL, "Charge Pump" },
{ "Right Headphone Output PGA", NULL, "Charge Pump" },
{ "Left Line Output PGA", NULL, "Charge Pump" },
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index f3f7e75f8628..9af1bddc4c62 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -828,7 +828,8 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp)
&buf_list);
if (!buf) {
adsp_err(dsp, "Out of memory\n");
- return -ENOMEM;
+ ret = -ENOMEM;
+ goto out_fw;
}
adsp_dbg(dsp, "%s.%d: Writing %d bytes at %x\n",
@@ -865,7 +866,7 @@ out_fw:
wm_adsp_buf_free(&buf_list);
out:
kfree(file);
- return 0;
+ return ret;
}
int wm_adsp1_init(struct wm_adsp *adsp)
diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c
index 55464a5b0706..810c7eeb7b03 100644
--- a/sound/soc/fsl/imx-ssi.c
+++ b/sound/soc/fsl/imx-ssi.c
@@ -496,6 +496,8 @@ static void imx_ssi_ac97_reset(struct snd_ac97 *ac97)
if (imx_ssi->ac97_reset)
imx_ssi->ac97_reset(ac97);
+ /* First read sometimes fails, do a dummy read */
+ imx_ssi_ac97_read(ac97, 0);
}
static void imx_ssi_ac97_warm_reset(struct snd_ac97 *ac97)
@@ -504,6 +506,9 @@ static void imx_ssi_ac97_warm_reset(struct snd_ac97 *ac97)
if (imx_ssi->ac97_warm_reset)
imx_ssi->ac97_warm_reset(ac97);
+
+ /* First read sometimes fails, do a dummy read */
+ imx_ssi_ac97_read(ac97, 0);
}
struct snd_ac97_bus_ops soc_ac97_ops = {
diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c
index 8e52c1485df3..eb4373840bb6 100644
--- a/sound/soc/fsl/pcm030-audio-fabric.c
+++ b/sound/soc/fsl/pcm030-audio-fabric.c
@@ -51,7 +51,7 @@ static struct snd_soc_card pcm030_card = {
.num_links = ARRAY_SIZE(pcm030_fabric_dai),
};
-static int __init pcm030_fabric_probe(struct platform_device *op)
+static int pcm030_fabric_probe(struct platform_device *op)
{
struct device_node *np = op->dev.of_node;
struct device_node *platform_np;
diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c
index d7231e336a7c..6bbeb0bf1a73 100644
--- a/sound/soc/samsung/i2s.c
+++ b/sound/soc/samsung/i2s.c
@@ -972,6 +972,7 @@ static const struct snd_soc_dai_ops samsung_i2s_dai_ops = {
static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec)
{
struct i2s_dai *i2s;
+ int ret;
i2s = devm_kzalloc(&pdev->dev, sizeof(struct i2s_dai), GFP_KERNEL);
if (i2s == NULL)
@@ -996,15 +997,17 @@ static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec)
i2s->i2s_dai_drv.capture.channels_max = 2;
i2s->i2s_dai_drv.capture.rates = SAMSUNG_I2S_RATES;
i2s->i2s_dai_drv.capture.formats = SAMSUNG_I2S_FMTS;
+ dev_set_drvdata(&i2s->pdev->dev, i2s);
} else { /* Create a new platform_device for Secondary */
- i2s->pdev = platform_device_register_resndata(NULL,
- "samsung-i2s-sec", -1, NULL, 0, NULL, 0);
+ i2s->pdev = platform_device_alloc("samsung-i2s-sec", -1);
if (IS_ERR(i2s->pdev))
return NULL;
- }
- /* Pre-assign snd_soc_dai_set_drvdata */
- dev_set_drvdata(&i2s->pdev->dev, i2s);
+ platform_set_drvdata(i2s->pdev, i2s);
+ ret = platform_device_add(i2s->pdev);
+ if (ret < 0)
+ return NULL;
+ }
return i2s;
}
@@ -1107,6 +1110,10 @@ static int samsung_i2s_probe(struct platform_device *pdev)
if (samsung_dai_type == TYPE_SEC) {
sec_dai = dev_get_drvdata(&pdev->dev);
+ if (!sec_dai) {
+ dev_err(&pdev->dev, "Unable to get drvdata\n");
+ return -EFAULT;
+ }
snd_soc_register_dai(&sec_dai->pdev->dev,
&sec_dai->i2s_dai_drv);
asoc_dma_platform_register(&pdev->dev);
diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c
index 19eff8fc4fdd..1a8b03e4b41b 100644
--- a/sound/soc/sh/dma-sh7760.c
+++ b/sound/soc/sh/dma-sh7760.c
@@ -342,8 +342,8 @@ static int camelot_pcm_new(struct snd_soc_pcm_runtime *rtd)
return 0;
}
-static struct snd_soc_platform sh7760_soc_platform = {
- .pcm_ops = &camelot_pcm_ops,
+static struct snd_soc_platform_driver sh7760_soc_platform = {
+ .ops = &camelot_pcm_ops,
.pcm_new = camelot_pcm_new,
.pcm_free = camelot_pcm_free,
};
diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
index b5b3db71e253..ed0bfb0ddb96 100644
--- a/sound/soc/soc-compress.c
+++ b/sound/soc/soc-compress.c
@@ -211,19 +211,27 @@ static int soc_compr_set_params(struct snd_compr_stream *cstream,
if (platform->driver->compr_ops && platform->driver->compr_ops->set_params) {
ret = platform->driver->compr_ops->set_params(cstream, params);
if (ret < 0)
- goto out;
+ goto err;
}
if (rtd->dai_link->compr_ops && rtd->dai_link->compr_ops->set_params) {
ret = rtd->dai_link->compr_ops->set_params(cstream);
if (ret < 0)
- goto out;
+ goto err;
}
snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK,
SND_SOC_DAPM_STREAM_START);
-out:
+ /* cancel any delayed stream shutdown that is pending */
+ rtd->pop_wait = 0;
+ mutex_unlock(&rtd->pcm_mutex);
+
+ cancel_delayed_work_sync(&rtd->delayed_work);
+
+ return ret;
+
+err:
mutex_unlock(&rtd->pcm_mutex);
return ret;
}
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 4d24b5ea3ba7..c70f9e072043 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -2963,7 +2963,7 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol,
val = val << shift;
ret = snd_soc_update_bits_locked(codec, reg, val_mask, val);
- if (ret != 0)
+ if (ret < 0)
return ret;
if (snd_soc_volsw_is_stereo(mc)) {
@@ -3140,7 +3140,7 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol,
if (params->mask) {
ret = regmap_read(codec->control_data, params->base, &val);
if (ret != 0)
- return ret;
+ goto out;
val &= params->mask;
@@ -3158,13 +3158,15 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol,
((u32 *)data)[0] |= cpu_to_be32(val);
break;
default:
- return -EINVAL;
+ ret = -EINVAL;
+ goto out;
}
}
ret = regmap_raw_write(codec->control_data, params->base,
data, len);
+out:
kfree(data);
return ret;
@@ -4198,7 +4200,6 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card,
dev_err(card->dev,
"ASoC: Property '%s' index %d could not be read: %d\n",
propname, 2 * i, ret);
- kfree(routes);
return -EINVAL;
}
ret = of_property_read_string_index(np, propname,
@@ -4207,7 +4208,6 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card,
dev_err(card->dev,
"ASoC: Property '%s' index %d could not be read: %d\n",
propname, (2 * i) + 1, ret);
- kfree(routes);
return -EINVAL;
}
}
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 625d4824abbb..33acd8b892dc 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -831,6 +831,9 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget,
if (path->weak)
continue;
+ if (path->walking)
+ return 1;
+
if (path->walked)
continue;
@@ -838,6 +841,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget,
if (path->sink && path->connect) {
path->walked = 1;
+ path->walking = 1;
/* do we need to add this widget to the list ? */
if (list) {
@@ -847,11 +851,14 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget,
dev_err(widget->dapm->dev,
"ASoC: could not add widget %s\n",
widget->name);
+ path->walking = 0;
return con;
}
}
con += is_connected_output_ep(path->sink, list);
+
+ path->walking = 0;
}
}
@@ -931,6 +938,9 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget,
if (path->weak)
continue;
+ if (path->walking)
+ return 1;
+
if (path->walked)
continue;
@@ -938,6 +948,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget,
if (path->source && path->connect) {
path->walked = 1;
+ path->walking = 1;
/* do we need to add this widget to the list ? */
if (list) {
@@ -947,11 +958,14 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget,
dev_err(widget->dapm->dev,
"ASoC: could not add widget %s\n",
widget->name);
+ path->walking = 0;
return con;
}
}
con += is_connected_input_ep(path->source, list);
+
+ path->walking = 0;
}
}
diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c
index 9b76cc5a1148..5e7aebe1e664 100644
--- a/sound/soc/spear/spear_pcm.c
+++ b/sound/soc/spear/spear_pcm.c
@@ -149,9 +149,9 @@ static void spear_pcm_free(struct snd_pcm *pcm)
static u64 spear_pcm_dmamask = DMA_BIT_MASK(32);
-static int spear_pcm_new(struct snd_card *card,
- struct snd_soc_dai *dai, struct snd_pcm *pcm)
+static int spear_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_card *card = rtd->card->snd_card;
int ret;
if (!card->dev->dma_mask)
@@ -159,16 +159,16 @@ static int spear_pcm_new(struct snd_card *card,
if (!card->dev->coherent_dma_mask)
card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
- if (dai->driver->playback.channels_min) {
- ret = spear_pcm_preallocate_dma_buffer(pcm,
+ if (rtd->cpu_dai->driver->playback.channels_min) {
+ ret = spear_pcm_preallocate_dma_buffer(rtd->pcm,
SNDRV_PCM_STREAM_PLAYBACK,
spear_pcm_hardware.buffer_bytes_max);
if (ret)
return ret;
}
- if (dai->driver->capture.channels_min) {
- ret = spear_pcm_preallocate_dma_buffer(pcm,
+ if (rtd->cpu_dai->driver->capture.channels_min) {
+ ret = spear_pcm_preallocate_dma_buffer(rtd->pcm,
SNDRV_PCM_STREAM_CAPTURE,
spear_pcm_hardware.buffer_bytes_max);
if (ret)
diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c
index c925ab0adeb6..5e2c55c5b255 100644
--- a/sound/soc/tegra/tegra_pcm.c
+++ b/sound/soc/tegra/tegra_pcm.c
@@ -43,8 +43,6 @@
static const struct snd_pcm_hardware tegra_pcm_hardware = {
.info = SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_MMAP_VALID |
- SNDRV_PCM_INFO_PAUSE |
- SNDRV_PCM_INFO_RESUME |
SNDRV_PCM_INFO_INTERLEAVED,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
.channels_min = 2,
@@ -127,26 +125,6 @@ static int tegra_pcm_hw_free(struct snd_pcm_substream *substream)
return 0;
}
-static int tegra_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
-{
- switch (cmd) {
- case SNDRV_PCM_TRIGGER_START:
- case SNDRV_PCM_TRIGGER_RESUME:
- case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- return snd_dmaengine_pcm_trigger(substream,
- SNDRV_PCM_TRIGGER_START);
-
- case SNDRV_PCM_TRIGGER_STOP:
- case SNDRV_PCM_TRIGGER_SUSPEND:
- case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
- return snd_dmaengine_pcm_trigger(substream,
- SNDRV_PCM_TRIGGER_STOP);
- default:
- return -EINVAL;
- }
- return 0;
-}
-
static int tegra_pcm_mmap(struct snd_pcm_substream *substream,
struct vm_area_struct *vma)
{
@@ -164,7 +142,7 @@ static struct snd_pcm_ops tegra_pcm_ops = {
.ioctl = snd_pcm_lib_ioctl,
.hw_params = tegra_pcm_hw_params,
.hw_free = tegra_pcm_hw_free,
- .trigger = tegra_pcm_trigger,
+ .trigger = snd_dmaengine_pcm_trigger,
.pointer = snd_dmaengine_pcm_pointer,
.mmap = tegra_pcm_mmap,
};
diff --git a/sound/usb/card.c b/sound/usb/card.c
index 803953a9bff3..2da8ad75fd96 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -244,6 +244,21 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif)
usb_ifnum_to_if(dev, ctrlif)->intf_assoc;
if (!assoc) {
+ /*
+ * Firmware writers cannot count to three. So to find
+ * the IAD on the NuForce UDH-100, also check the next
+ * interface.
+ */
+ struct usb_interface *iface =
+ usb_ifnum_to_if(dev, ctrlif + 1);
+ if (iface &&
+ iface->intf_assoc &&
+ iface->intf_assoc->bFunctionClass == USB_CLASS_AUDIO &&
+ iface->intf_assoc->bFunctionProtocol == UAC_VERSION_2)
+ assoc = iface->intf_assoc;
+ }
+
+ if (!assoc) {
snd_printk(KERN_ERR "Audio class v2 interfaces need an interface association\n");
return -EINVAL;
}
diff --git a/sound/usb/clock.c b/sound/usb/clock.c
index 5e634a2eb282..9e2703a25156 100644
--- a/sound/usb/clock.c
+++ b/sound/usb/clock.c
@@ -253,7 +253,7 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface,
{
struct usb_device *dev = chip->dev;
unsigned char data[4];
- int err, crate;
+ int err, cur_rate, prev_rate;
int clock = snd_usb_clock_find_source(chip, fmt->clock);
if (clock < 0)
@@ -266,6 +266,19 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface,
return -ENXIO;
}
+ err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR,
+ USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
+ UAC2_CS_CONTROL_SAM_FREQ << 8,
+ snd_usb_ctrl_intf(chip) | (clock << 8),
+ data, sizeof(data));
+ if (err < 0) {
+ snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq (v2)\n",
+ dev->devnum, iface, fmt->altsetting);
+ prev_rate = 0;
+ } else {
+ prev_rate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24);
+ }
+
data[0] = rate;
data[1] = rate >> 8;
data[2] = rate >> 16;
@@ -280,19 +293,31 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface,
return err;
}
- if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR,
- USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
- UAC2_CS_CONTROL_SAM_FREQ << 8,
- snd_usb_ctrl_intf(chip) | (clock << 8),
- data, sizeof(data))) < 0) {
+ err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR,
+ USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
+ UAC2_CS_CONTROL_SAM_FREQ << 8,
+ snd_usb_ctrl_intf(chip) | (clock << 8),
+ data, sizeof(data));
+ if (err < 0) {
snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq (v2)\n",
dev->devnum, iface, fmt->altsetting);
- return err;
+ cur_rate = 0;
+ } else {
+ cur_rate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24);
}
- crate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24);
- if (crate != rate)
- snd_printd(KERN_WARNING "current rate %d is different from the runtime rate %d\n", crate, rate);
+ if (cur_rate != rate) {
+ snd_printd(KERN_WARNING
+ "current rate %d is different from the runtime rate %d\n",
+ cur_rate, rate);
+ }
+
+ /* Some devices doesn't respond to sample rate changes while the
+ * interface is active. */
+ if (rate != prev_rate) {
+ usb_set_interface(dev, iface, 0);
+ usb_set_interface(dev, iface, fmt->altsetting);
+ }
return 0;
}
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index 638e7f738018..ca4739c3f650 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -715,8 +715,9 @@ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_
case UAC2_CLOCK_SELECTOR: {
struct uac_selector_unit_descriptor *d = p1;
/* call recursively to retrieve the channel info */
- if (check_input_term(state, d->baSourceID[0], term) < 0)
- return -ENODEV;
+ err = check_input_term(state, d->baSourceID[0], term);
+ if (err < 0)
+ return err;
term->type = d->bDescriptorSubtype << 16; /* virtual type */
term->id = id;
term->name = uac_selector_unit_iSelector(d);
@@ -725,7 +726,8 @@ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_
case UAC1_PROCESSING_UNIT:
case UAC1_EXTENSION_UNIT:
/* UAC2_PROCESSING_UNIT_V2 */
- /* UAC2_EFFECT_UNIT */ {
+ /* UAC2_EFFECT_UNIT */
+ case UAC2_EXTENSION_UNIT_V2: {
struct uac_processing_unit_descriptor *d = p1;
if (state->mixer->protocol == UAC_VERSION_2 &&
@@ -1356,8 +1358,9 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void
return err;
/* determine the input source type and name */
- if (check_input_term(state, hdr->bSourceID, &iterm) < 0)
- return -EINVAL;
+ err = check_input_term(state, hdr->bSourceID, &iterm);
+ if (err < 0)
+ return err;
master_bits = snd_usb_combine_bytes(bmaControls, csize);
/* master configuration quirks */
@@ -2052,6 +2055,8 @@ static int parse_audio_unit(struct mixer_build *state, int unitid)
return parse_audio_extension_unit(state, unitid, p1);
else /* UAC_VERSION_2 */
return parse_audio_processing_unit(state, unitid, p1);
+ case UAC2_EXTENSION_UNIT_V2:
+ return parse_audio_extension_unit(state, unitid, p1);
default:
snd_printk(KERN_ERR "usbaudio: unit %u: unexpected type 0x%02x\n", unitid, p1[2]);
return -EINVAL;
@@ -2118,7 +2123,7 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer)
state.oterm.type = le16_to_cpu(desc->wTerminalType);
state.oterm.name = desc->iTerminal;
err = parse_audio_unit(&state, desc->bSourceID);
- if (err < 0)
+ if (err < 0 && err != -EINVAL)
return err;
} else { /* UAC_VERSION_2 */
struct uac2_output_terminal_descriptor *desc = p;
@@ -2130,12 +2135,12 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer)
state.oterm.type = le16_to_cpu(desc->wTerminalType);
state.oterm.name = desc->iTerminal;
err = parse_audio_unit(&state, desc->bSourceID);
- if (err < 0)
+ if (err < 0 && err != -EINVAL)
return err;
/* for UAC2, use the same approach to also add the clock selectors */
err = parse_audio_unit(&state, desc->bCSourceID);
- if (err < 0)
+ if (err < 0 && err != -EINVAL)
return err;
}
}
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index 497d2741d119..ebe91440a068 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -509,7 +509,7 @@ static int snd_nativeinstruments_control_get(struct snd_kcontrol *kcontrol,
else
ret = usb_control_msg(dev, usb_rcvctrlpipe(dev, 0), bRequest,
USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_IN,
- 0, cpu_to_le16(wIndex),
+ 0, wIndex,
&tmp, sizeof(tmp), 1000);
up_read(&mixer->chip->shutdown_rwsem);
@@ -540,7 +540,7 @@ static int snd_nativeinstruments_control_put(struct snd_kcontrol *kcontrol,
else
ret = usb_control_msg(dev, usb_sndctrlpipe(dev, 0), bRequest,
USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_OUT,
- cpu_to_le16(wValue), cpu_to_le16(wIndex),
+ wValue, wIndex,
NULL, 0, 1000);
up_read(&mixer->chip->shutdown_rwsem);
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 5325a3869bb7..9c5ab22358b1 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -486,7 +486,7 @@ static int snd_usb_nativeinstruments_boot_quirk(struct usb_device *dev)
{
int ret = usb_control_msg(dev, usb_sndctrlpipe(dev, 0),
0xaf, USB_TYPE_VENDOR | USB_RECIP_DEVICE,
- cpu_to_le16(1), 0, NULL, 0, 1000);
+ 1, 0, NULL, 0, 1000);
if (ret < 0)
return ret;
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