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-rw-r--r--player/audio.c12
1 files changed, 6 insertions, 6 deletions
diff --git a/player/audio.c b/player/audio.c
index 413cbf4a32..e3dbe6577c 100644
--- a/player/audio.c
+++ b/player/audio.c
@@ -192,6 +192,7 @@ void reinit_audio_chain(struct MPContext *mpctx)
mpctx->d_audio->global = mpctx->global;
mpctx->d_audio->opts = opts;
mpctx->d_audio->header = sh;
+ mpctx->d_audio->pool = mp_audio_pool_create(mpctx->d_audio);
mpctx->d_audio->afilter = af_new(mpctx->global);
mpctx->d_audio->afilter->replaygain_data = sh->audio->replaygain_data;
mpctx->ao_buffer = mp_audio_buffer_create(NULL);
@@ -207,8 +208,7 @@ void reinit_audio_chain(struct MPContext *mpctx)
}
assert(mpctx->d_audio);
- struct mp_audio in_format;
- mp_audio_buffer_get_format(mpctx->d_audio->decode_buffer, &in_format);
+ struct mp_audio in_format = mpctx->d_audio->decode_format;
if (!mp_audio_config_valid(&in_format)) {
// We don't know the audio format yet - so configure it later as we're
@@ -238,7 +238,7 @@ void reinit_audio_chain(struct MPContext *mpctx)
}
// filter input format: same as codec's output format:
- mp_audio_buffer_get_format(mpctx->d_audio->decode_buffer, &afs->input);
+ afs->input = in_format;
// Determine what the filter chain outputs. recreate_audio_filters() also
// needs this for testing whether playback speed is changed by resampling
@@ -304,8 +304,7 @@ double written_audio_pts(struct MPContext *mpctx)
if (!d_audio)
return MP_NOPTS_VALUE;
- struct mp_audio in_format;
- mp_audio_buffer_get_format(d_audio->decode_buffer, &in_format);
+ struct mp_audio in_format = d_audio->decode_format;
if (!mp_audio_config_valid(&in_format) || d_audio->afilter->initialized < 1)
return MP_NOPTS_VALUE;
@@ -324,7 +323,8 @@ double written_audio_pts(struct MPContext *mpctx)
// Subtract data in buffers between decoder and audio out.
// Decoded but not filtered
- a_pts -= mp_audio_buffer_seconds(d_audio->decode_buffer);
+ if (d_audio->waiting)
+ a_pts -= d_audio->waiting->samples / (double)in_format.rate;
// Data buffered in audio filters, measured in seconds of "missing" output
double buffered_output = af_calc_delay(d_audio->afilter);