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-rw-r--r--audio/filter/af.c34
-rw-r--r--audio/filter/af.h3
-rw-r--r--audio/filter/af_scaletempo.c5
-rw-r--r--audio/mixer.c4
-rw-r--r--audio/out/ao_alsa.c27
-rw-r--r--audio/out/ao_coreaudio_utils.c12
6 files changed, 52 insertions, 33 deletions
diff --git a/audio/filter/af.c b/audio/filter/af.c
index 2889e87bb7..8d29e332a8 100644
--- a/audio/filter/af.c
+++ b/audio/filter/af.c
@@ -561,6 +561,23 @@ static int af_reinit(struct af_stream *s)
retry++;
continue;
}
+ // If the format conversion is (probably) caused by spdif, then
+ // (as a feature) drop the filter, instead of failing hard.
+ int fmt_in1 = af->prev->data->format;
+ int fmt_in2 = in.format;
+ if (af_fmt_is_valid(fmt_in1) && af_fmt_is_valid(fmt_in2)) {
+ bool spd1 = AF_FORMAT_IS_IEC61937(fmt_in1);
+ bool spd2 = AF_FORMAT_IS_IEC61937(fmt_in2);
+ if (spd1 != spd2) {
+ MP_WARN(af, "Filter %s apparently cannot be used due to "
+ "spdif passthrough - removing it.\n",
+ af->info->name);
+ struct af_instance *aft = af->prev;
+ af_remove(s, af);
+ af = aft->next;
+ break;
+ }
+ }
goto negotiate_error;
}
case AF_DETACH: { // Filter is redundant and wants to be unloaded
@@ -689,8 +706,14 @@ int af_init(struct af_stream *s)
to the stream s. The filter will be inserted somewhere nice in the
list of filters. The return value is a pointer to the new filter,
If the filter couldn't be added the return value is NULL. */
-struct af_instance *af_add(struct af_stream *s, char *name, char **args)
+struct af_instance *af_add(struct af_stream *s, char *name, char *label,
+ char **args)
{
+ assert(label);
+
+ if (af_find_by_label(s, label))
+ return NULL;
+
struct af_instance *new;
// Insert the filter somewhere nice
if (af_is_conversion_filter(s->first->next))
@@ -699,17 +722,14 @@ struct af_instance *af_add(struct af_stream *s, char *name, char **args)
new = af_prepend(s, s->first->next, name, args);
if (!new)
return NULL;
+ new->label = talloc_strdup(new, label);
// Reinitalize the filter list
if (af_reinit(s) != AF_OK) {
- af_remove(s, new);
- if (af_reinit(s) != AF_OK) {
- af_uninit(s);
- af_init(s);
- }
+ af_remove_by_label(s, label);
return NULL;
}
- return new;
+ return af_find_by_label(s, label);
}
struct af_instance *af_find_by_label(struct af_stream *s, char *label)
diff --git a/audio/filter/af.h b/audio/filter/af.h
index 4c67208123..65a30f7dd1 100644
--- a/audio/filter/af.h
+++ b/audio/filter/af.h
@@ -141,7 +141,8 @@ struct af_stream *af_new(struct mpv_global *global);
void af_destroy(struct af_stream *s);
int af_init(struct af_stream *s);
void af_uninit(struct af_stream *s);
-struct af_instance *af_add(struct af_stream *s, char *name, char **args);
+struct af_instance *af_add(struct af_stream *s, char *name, char *label,
+ char **args);
int af_remove_by_label(struct af_stream *s, char *label);
struct af_instance *af_find_by_label(struct af_stream *s, char *label);
struct af_instance *af_control_any_rev(struct af_stream *s, int cmd, void *arg);
diff --git a/audio/filter/af_scaletempo.c b/audio/filter/af_scaletempo.c
index 1f187a98bc..ad8e520601 100644
--- a/audio/filter/af_scaletempo.c
+++ b/audio/filter/af_scaletempo.c
@@ -303,11 +303,6 @@ static int control(struct af_instance *af, int cmd, void *arg)
int nch = data->nch;
int use_int = 0;
- if (AF_FORMAT_IS_SPECIAL(data->format)) {
- MP_ERR(af, "Changing speed is not supported with spdif formats.\n");
- return AF_ERROR;
- }
-
mp_audio_force_interleaved_format(data);
mp_audio_copy_config(af->data, data);
diff --git a/audio/mixer.c b/audio/mixer.c
index 0eb9c86453..7ae7ed458d 100644
--- a/audio/mixer.c
+++ b/audio/mixer.c
@@ -135,7 +135,7 @@ static void setvolume_internal(struct mixer *mixer, float l, float r)
if (gain == 1.0)
return;
MP_VERBOSE(mixer, "Inserting volume filter.\n");
- if (!(af_add(mixer->af, "volume", NULL)
+ if (!(af_add(mixer->af, "volume", "softvol", NULL)
&& af_control_any_rev(mixer->af, AF_CONTROL_SET_VOLUME, &gain)))
MP_ERR(mixer, "No volume control available.\n");
}
@@ -222,7 +222,7 @@ void mixer_setbalance(struct mixer *mixer, float val)
if (val == 0)
return;
- if (!(af_pan_balance = af_add(mixer->af, "pan", NULL))) {
+ if (!(af_pan_balance = af_add(mixer->af, "pan", "autopan", NULL))) {
MP_ERR(mixer, "No balance control available.\n");
return;
}
diff --git a/audio/out/ao_alsa.c b/audio/out/ao_alsa.c
index f84a781e8a..b60c8363f1 100644
--- a/audio/out/ao_alsa.c
+++ b/audio/out/ao_alsa.c
@@ -376,6 +376,7 @@ static int try_open_device(struct ao *ao, const char *device)
IEC958_AES1_CON_ORIGINAL | IEC958_AES1_CON_PCM_CODER,
map_iec958_srate(ao->samplerate));
const char *ac3_device = append_params(tmp, device, params);
+ MP_VERBOSE(ao, "opening device '%s' => '%s'\n", device, ac3_device);
int err = snd_pcm_open
(&p->alsa, ac3_device, SND_PCM_STREAM_PLAYBACK, 0);
talloc_free(tmp);
@@ -383,6 +384,7 @@ static int try_open_device(struct ao *ao, const char *device)
return 0;
}
+ MP_VERBOSE(ao, "opening device '%s'\n", device);
return snd_pcm_open(&p->alsa, device, SND_PCM_STREAM_PLAYBACK, 0);
}
@@ -408,23 +410,14 @@ static int init_device(struct ao *ao)
struct priv *p = ao->priv;
int err;
- if (!p->cfg_ni)
- ao->format = af_fmt_from_planar(ao->format);
-
const char *device = "default";
- if (AF_FORMAT_IS_IEC61937(ao->format)) {
+ if (AF_FORMAT_IS_IEC61937(ao->format))
device = "iec958";
- MP_VERBOSE(ao, "playing AC3/iec61937/iec958, %i channels\n",
- ao->channels.num);
- }
if (ao->device)
device = ao->device;
if (p->cfg_device && p->cfg_device[0])
device = p->cfg_device;
- MP_VERBOSE(ao, "using device: %s\n", device);
- MP_VERBOSE(ao, "using ALSA version: %s\n", snd_asoundlib_version());
-
err = try_open_device(ao, device);
CHECK_ALSA_ERROR("Playback open error");
@@ -551,8 +544,11 @@ static int init_device(struct ao *ao)
err = snd_pcm_set_chmap(p->alsa, alsa_chmap);
if (err == -ENXIO) {
- MP_WARN(ao, "Device does not support requested channel map (%s)\n",
- mp_chmap_to_str(&dev_chmap));
+ // I consider this an ALSA bug: the channel map was reported as
+ // supported, but we still can't set it. It happens virtually
+ // always with dmix, though.
+ MP_VERBOSE(ao, "Device does not support requested channel map (%s)\n",
+ mp_chmap_to_str(&dev_chmap));
} else {
CHECK_ALSA_WARN("Channel map setup failed");
}
@@ -607,6 +603,7 @@ static int init_device(struct ao *ao)
// the number of channels to 2 either, because the hw params
// are already set! So just fuck it and reopen the device with
// the chmap "cleaned out" of NA entries.
+ MP_VERBOSE(ao, "Working around braindead ALSA behavior.\n");
err = snd_pcm_close(p->alsa);
p->alsa = NULL;
CHECK_ALSA_ERROR("pcm close error");
@@ -678,6 +675,12 @@ alsa_error:
static int init(struct ao *ao)
{
+ struct priv *p = ao->priv;
+ if (!p->cfg_ni)
+ ao->format = af_fmt_from_planar(ao->format);
+
+ MP_VERBOSE(ao, "using ALSA version: %s\n", snd_asoundlib_version());
+
int r = init_device(ao);
if (r == INIT_BRAINDEATH)
r = init_device(ao); // retry with normalized channel layout
diff --git a/audio/out/ao_coreaudio_utils.c b/audio/out/ao_coreaudio_utils.c
index 64d98e5ece..8485011722 100644
--- a/audio/out/ao_coreaudio_utils.c
+++ b/audio/out/ao_coreaudio_utils.c
@@ -117,12 +117,12 @@ OSStatus ca_select_device(struct ao *ao, char* name, AudioDeviceID *device)
if (mp_msg_test(ao->log, MSGL_V)) {
char *desc;
err = CA_GET_STR(*device, kAudioObjectPropertyName, &desc);
- CHECK_CA_ERROR("could not get selected audio device name");
-
- MP_VERBOSE(ao, "selected audio output device: %s (%" PRIu32 ")\n",
- desc, *device);
-
- talloc_free(desc);
+ CHECK_CA_WARN("could not get selected audio device name");
+ if (err == noErr) {
+ MP_VERBOSE(ao, "selected audio output device: %s (%" PRIu32 ")\n",
+ desc, *device);
+ talloc_free(desc);
+ }
}
coreaudio_error:
generated by cgit v1.2.3 (git 2.25.1) at 2024-06-06 14:22:09 +0000