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audio/out: fix previous commit 

This didn't quite work. The main issue was that get_space tries to be
clever to reduce overall buffering, so it will cause the playloop to
decode and queue only as much audio as is needed to refill the AO in
reasonable time. Also, even if ignoring the problem, the logic of the
previous commit was slightly broken. (This required a few retries,
because I couldn't reproduce the issue on my own machine.)
This commit is contained in:
wm4 2014-05-11 20:51:49 +02:00
parent 665c8b59be
commit c78b8b2c0c
1 changed files with 11 additions and 9 deletions
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20
audio/out/push.c
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@ -49,7 +49,6 @@ struct ao_push_state {
struct mp_audio_buffer *buffer;
bool newdata;
bool terminate;
bool playing;
@ -148,10 +147,9 @@ static void drain(struct ao *ao)
}
}
static int get_space(struct ao *ao)
static int unlocked_get_space(struct ao *ao)
{
struct ao_push_state *p = ao->api_priv;
pthread_mutex_lock(&p->lock);
int space = mp_audio_buffer_get_write_available(p->buffer);
if (ao->driver->get_space) {
// The following code attempts to keep the total buffered audio to
@ -167,6 +165,14 @@ static int get_space(struct ao *ao)
space = MPMIN(space, missing);
space = MPMAX(0, space);
}
return space;
}
static int get_space(struct ao *ao)
{
struct ao_push_state *p = ao->api_priv;
pthread_mutex_lock(&p->lock);
int space = unlocked_get_space(ao);
pthread_mutex_unlock(&p->lock);
return space;
}
@ -186,8 +192,6 @@ static int play(struct ao *ao, void **data, int samples, int flags)
audio.planes[n] = data[n];
audio.samples = write_samples;
mp_audio_buffer_append(p->buffer, &audio);
if (write_samples > 0)
p->newdata = true;
p->final_chunk = !!(flags & AOPLAY_FINAL_CHUNK);
p->playing = true;
@ -219,10 +223,8 @@ static int ao_play_data(struct ao *ao)
MP_WARN(ao, "Audio device returned non-sense value.\n");
r = data.samples;
}
if (r > 0) {
if (r > 0)
mp_audio_buffer_skip(p->buffer, r);
p->newdata = false;
}
if (p->final_chunk && mp_audio_buffer_samples(p->buffer) == 0) {
p->playing = false;
p->expected_end_time = mp_time_sec() + AO_EOF_DELAY + 0.25; // + margin
@ -261,7 +263,7 @@ static void *playthread(void *arg)
}
// Half of the buffer played -> wakeup playback thread to get more.
double min_wait = ao->device_buffer / (double)ao->samplerate;
if (timeout <= min_wait / 2 + 0.001 && !p->newdata)
if (timeout <= min_wait / 2 + 0.001 && unlocked_get_space(ao) > 0)
mp_input_wakeup(ao->input_ctx);
// Avoid wasting CPU - this assumes ao_play_data() usually fills the
// audio buffer as far as possible, so even if the device buffer