summaryrefslogtreecommitdiffstats
path: root/audio/out/ao_coreaudio.c
diff options
context:
space:
mode:
Diffstat (limited to 'audio/out/ao_coreaudio.c')
-rw-r--r--audio/out/ao_coreaudio.c65
1 files changed, 35 insertions, 30 deletions
diff --git a/audio/out/ao_coreaudio.c b/audio/out/ao_coreaudio.c
index 718c71187d..59e1f82f36 100644
--- a/audio/out/ao_coreaudio.c
+++ b/audio/out/ao_coreaudio.c
@@ -46,10 +46,10 @@ static void audio_pause(struct ao *ao);
static void audio_resume(struct ao *ao);
static void reset(struct ao *ao);
-static void print_buffer(struct mp_ring *buffer)
+static void print_buffer(struct ao *ao, struct mp_ring *buffer)
{
void *tctx = talloc_new(NULL);
- ca_msg(MSGL_V, "%s\n", mp_ring_repr(buffer, tctx));
+ MP_VERBOSE(ao, "%s\n", mp_ring_repr(buffer, tctx));
talloc_free(tctx);
}
@@ -109,7 +109,12 @@ static OSStatus render_cb_lpcm(void *ctx, AudioUnitRenderActionFlags *aflags,
AudioBuffer buf = buffer_list->mBuffers[0];
int requested = buf.mDataByteSize;
- buf.mDataByteSize = mp_ring_read(p->buffer, buf.mData, requested);
+ if (mp_ring_buffered(p->buffer) < requested) {
+ MP_VERBOSE(ao, "buffer underrun\n");
+ audio_pause(ao);
+ } else {
+ mp_ring_read(p->buffer, buf.mData, requested);
+ }
return noErr;
}
@@ -192,7 +197,7 @@ coreaudio_error:
return CONTROL_ERROR;
}
-static void print_list(void)
+static void print_list(struct ao *ao)
{
char *help = talloc_strdup(NULL, "Available output devices:\n");
@@ -221,7 +226,7 @@ static void print_list(void)
talloc_free(devs);
coreaudio_error:
- ca_msg(MSGL_INFO, "%s", help);
+ MP_INFO(ao, "%s", help);
talloc_free(help);
}
@@ -233,7 +238,7 @@ static int init(struct ao *ao)
OSStatus err;
struct priv *p = ao->priv;
- if (p->opt_list) print_list();
+ if (p->opt_list) print_list(ao);
struct priv_d *d = talloc_zero(p, struct priv_d);
@@ -266,9 +271,8 @@ static int init(struct ao *ao)
err = CA_GET_STR(selected_device, kAudioObjectPropertyName, &device_name);
CHECK_CA_ERROR("could not get selected audio device name");
- ca_msg(MSGL_V,
- "selected audio output device: %s (%" PRIu32 ")\n",
- device_name, selected_device);
+ MP_VERBOSE(ao, "selected audio output device: %s (%" PRIu32 ")\n",
+ device_name, selected_device);
talloc_free(device_name);
@@ -278,7 +282,7 @@ static int init(struct ao *ao)
bool supports_digital = false;
/* Probe whether device support S/PDIF stream output if input is AC3 stream. */
if (AF_FORMAT_IS_AC3(ao->format)) {
- if (ca_device_supports_digital(selected_device))
+ if (ca_device_supports_digital(ao, selected_device))
supports_digital = true;
}
@@ -293,7 +297,7 @@ static int init(struct ao *ao)
uint32_t *bitmaps;
size_t n_bitmaps;
- ca_bitmaps_from_layouts(layouts, n_layouts, &bitmaps, &n_bitmaps);
+ ca_bitmaps_from_layouts(ao, layouts, n_layouts, &bitmaps, &n_bitmaps);
talloc_free(layouts);
struct mp_chmap_sel chmap_sel = {0};
@@ -339,7 +343,7 @@ static int init(struct ao *ao)
asbd.mFramesPerPacket * asbd.mChannelsPerFrame *
(asbd.mBitsPerChannel / 8);
- ca_print_asbd("source format:", &asbd);
+ ca_print_asbd(ao, "source format:", &asbd);
if (supports_digital)
return init_digital(ao, asbd);
@@ -358,7 +362,9 @@ static int init_lpcm(struct ao *ao, AudioStreamBasicDescription asbd)
AudioComponentDescription desc = (AudioComponentDescription) {
.componentType = kAudioUnitType_Output,
- .componentSubType = kAudioUnitSubType_HALOutput,
+ .componentSubType = (p->opt_device_id < 0) ?
+ kAudioUnitSubType_DefaultOutput :
+ kAudioUnitSubType_HALOutput,
.componentManufacturer = kAudioUnitManufacturer_Apple,
.componentFlags = 0,
.componentFlagsMask = 0,
@@ -366,7 +372,7 @@ static int init_lpcm(struct ao *ao, AudioStreamBasicDescription asbd)
AudioComponent comp = AudioComponentFindNext(NULL, &desc);
if (comp == NULL) {
- ca_msg(MSGL_ERR, "unable to find audio component\n");
+ MP_ERR(ao, "unable to find audio component\n");
goto coreaudio_error;
}
@@ -410,7 +416,7 @@ static int init_lpcm(struct ao *ao, AudioStreamBasicDescription asbd)
}
p->buffer = mp_ring_new(p, get_ring_size(ao));
- print_buffer(p->buffer);
+ print_buffer(ao, p->buffer);
AURenderCallbackStruct render_cb = (AURenderCallbackStruct) {
.inputProc = render_cb_lpcm,
@@ -447,14 +453,14 @@ static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd)
CHECK_CA_WARN("could not check whether device is alive");
if (!is_alive)
- ca_msg(MSGL_WARN, "device is not alive\n");
+ MP_WARN(ao , "device is not alive\n");
p->is_digital = 1;
err = ca_lock_device(p->device, &d->hog_pid);
CHECK_CA_WARN("failed to set hogmode");
- err = ca_disable_mixing(p->device, &d->changed_mixing);
+ err = ca_disable_mixing(ao, p->device, &d->changed_mixing);
CHECK_CA_WARN("failed to disable mixing");
AudioStreamID *streams;
@@ -467,7 +473,7 @@ static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd)
CHECK_CA_ERROR("could not get number of streams");
for (int i = 0; i < n_streams && d->stream_idx < 0; i++) {
- bool digital = ca_stream_supports_digital(streams[i]);
+ bool digital = ca_stream_supports_digital(ao, streams[i]);
if (digital) {
err = CA_GET(streams[i], kAudioStreamPropertyPhysicalFormat,
@@ -518,11 +524,11 @@ static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd)
talloc_free(streams);
if (d->stream_idx < 0) {
- ca_msg(MSGL_WARN, "can't find any digital output stream format\n");
+ MP_WARN(ao , "can't find any digital output stream format\n");
goto coreaudio_error;
}
- if (!ca_change_format(d->stream, d->stream_asbd))
+ if (!ca_change_format(ao, d->stream, d->stream_asbd))
goto coreaudio_error;
void *changed = (void *) &(d->stream_asbd_changed);
@@ -537,8 +543,7 @@ static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd)
ao->format = AF_FORMAT_AC3_LE;
else if (d->stream_asbd.mFormatFlags & kAudioFormatFlagIsBigEndian)
#endif
- ca_msg(MSGL_WARN,
- "stream has non-native byte order, digital output may fail\n");
+ MP_WARN(ao, "stream has non-native byte order, output may fail\n");
ao->samplerate = d->stream_asbd.mSampleRate;
ao->bps = ao->samplerate *
@@ -546,7 +551,7 @@ static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd)
d->stream_asbd.mFramesPerPacket);
p->buffer = mp_ring_new(p, get_ring_size(ao));
- print_buffer(p->buffer);
+ print_buffer(ao, p->buffer);
err = AudioDeviceCreateIOProcID(p->device,
(AudioDeviceIOProc)render_cb_digital,
@@ -573,11 +578,11 @@ static int play(struct ao *ao, void *output_samples, int num_bytes, int flags)
// Check whether we need to reset the digital output stream.
if (p->is_digital && d->stream_asbd_changed) {
d->stream_asbd_changed = 0;
- if (ca_stream_supports_digital(d->stream)) {
- if (!ca_change_format(d->stream, d->stream_asbd)) {
- ca_msg(MSGL_WARN, "can't restore digital output\n");
+ if (ca_stream_supports_digital(ao, d->stream)) {
+ if (!ca_change_format(ao, d->stream, d->stream_asbd)) {
+ MP_WARN(ao , "can't restore digital output\n");
} else {
- ca_msg(MSGL_WARN, "restoring digital output succeeded.\n");
+ MP_WARN(ao, "restoring digital output succeeded.\n");
reset(ao);
}
}
@@ -635,10 +640,10 @@ static void uninit(struct ao *ao, bool immed)
err = AudioDeviceDestroyIOProcID(p->device, d->render_cb);
CHECK_CA_WARN("failed to remove device render callback");
- if (!ca_change_format(d->stream, d->original_asbd))
- ca_msg(MSGL_WARN, "can't revert to original device format");
+ if (!ca_change_format(ao, d->stream, d->original_asbd))
+ MP_WARN(ao, "can't revert to original device format");
- err = ca_enable_mixing(p->device, d->changed_mixing);
+ err = ca_enable_mixing(ao, p->device, d->changed_mixing);
CHECK_CA_WARN("can't re-enable mixing");
err = ca_unlock_device(p->device, &d->hog_pid);
generated by cgit v1.2.3 (git 2.25.1) at 2024-05-02 09:28:06 +0000