Merge tag 'sound-3.9' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Mostly HD-audio and USB-audio regression fixes:
- Oops fix at unloading of snd-hda-codec-conexant module
- A few trivial regression fixes for Cirrus and Conexant HD-audio
codecs
- Relax the USB-audio descriptor parse errors as non-fatal
- Fix locking of HD-audio CA0132 DSP loader
- Fix the generic HD-audio parser for VIA codecs"
* tag 'sound-3.9' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda - Fix DAC assignment for independent HP
ALSA: hda - Fix abuse of snd_hda_lock_devices() for DSP loader
ALSA: hda - Fix typo in checking IEC958 emphasis bit
ALSA: snd-usb: mixer: ignore -EINVAL in snd_usb_mixer_controls()
ALSA: snd-usb: mixer: propagate errors up the call chain
ALSA: usb: Parse UAC2 extension unit like for UAC1
ALSA: hda - Fix yet missing GPIO/EAPD setup in cirrus driver
ALSA: hda/cirrus - Fix the digital beep registration
ALSA: hda - Fix missing beep detach in patch_conexant.c
ALSA: documentation: Fix typo in Documentation/sound
diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index ce6581c..4499bd9 100644
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -912,7 +912,7 @@
models depending on the codec chip. The list of available models
is found in HD-Audio-Models.txt
- The model name "genric" is treated as a special case. When this
+ The model name "generic" is treated as a special case. When this
model is given, the driver uses the generic codec parser without
"codec-patch". It's sometimes good for testing and debugging.
diff --git a/Documentation/sound/alsa/seq_oss.html b/Documentation/sound/alsa/seq_oss.html
index d9776cf..9663b45 100644
--- a/Documentation/sound/alsa/seq_oss.html
+++ b/Documentation/sound/alsa/seq_oss.html
@@ -285,7 +285,7 @@
<H4>
7.2.4 Close Callback</H4>
The <TT>close</TT> callback is called when this device is closed by the
-applicaion. If any private data was allocated in open callback, it must
+application. If any private data was allocated in open callback, it must
be released in the close callback. The deletion of ALSA port should be
done here, too. This callback must not be NULL.
<H4>
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index a9ebcf9..ecdf30e 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -3144,7 +3144,7 @@
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_generic.c b/sound/pci/hda/hda_generic.c
index 78897d0..43c2ea5 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -995,6 +995,8 @@
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 @@
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 @@
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 @@
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 4cea6bb6..418bfc0 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -415,6 +415,8 @@
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 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 @@
/* 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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 */
@@ -3481,6 +3566,7 @@
}
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_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 60d08f6..0d9c58f 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -168,10 +168,10 @@
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);
}
}
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 941bf6c..2a89d1ee 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -1142,7 +1142,7 @@
}
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 @@
}
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 @@
}
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 @@
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 @@
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/usb/mixer.c b/sound/usb/mixer.c
index 638e7f73..ca4739c 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -715,8 +715,9 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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;
}
}
