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authorwm4 <wm4@nowhere>2013-07-11 19:15:09 +0200
committerwm4 <wm4@nowhere>2013-07-11 19:15:09 +0200
commit23e303859aa93572f00b17e3b2bc0a552ad7c348 (patch)
tree155f4f429170b770e54bdd630886830168e220bc
parent7a4f9cc4d2509a0d9afa68ea53ed97b72525d001 (diff)
downloadmpv-23e303859aa93572f00b17e3b2bc0a552ad7c348.tar.bz2
mpv-23e303859aa93572f00b17e3b2bc0a552ad7c348.tar.xz
mplayer: fix incorrect audio sync after format changes
This is not directly related to the handling of format changes itself, but playing audio normally after the change. This was broken: the output byte rate was not recalculated, so audio-video sync was simply broken. Fix this by calculating the byte rate on the fly, instead of storing it in sh_audio. Format changes are relatively common (switches between stereo and 5.1 in TV recordings), so this fixes a somewhat critical bug.
-rw-r--r--audio/decode/dec_audio.c10
-rw-r--r--core/mplayer.c8
-rw-r--r--demux/stheader.h1
-rw-r--r--stream/tv.c2
4 files changed, 9 insertions, 12 deletions
diff --git a/audio/decode/dec_audio.c b/audio/decode/dec_audio.c
index 8c80a0b119..d6b57dd2b5 100644
--- a/audio/decode/dec_audio.c
+++ b/audio/decode/dec_audio.c
@@ -84,9 +84,6 @@ static int init_audio_codec(sh_audio_t *sh_audio, const char *decoder)
return 0;
}
- if (!sh_audio->o_bps)
- sh_audio->o_bps = sh_audio->channels.num * sh_audio->samplerate
- * sh_audio->samplesize;
return 1;
}
@@ -150,12 +147,9 @@ int init_best_audio_codec(sh_audio_t *sh_audio, char *audio_decoders)
mp_msg(MSGT_DECAUDIO, MSGL_INFO, "Selected audio codec: %s\n",
sh_audio->gsh->decoder_desc);
mp_msg(MSGT_DECAUDIO, MSGL_V,
- "AUDIO: %d Hz, %d ch, %s, %3.1f kbit/%3.2f%% (ratio: %d->%d)\n",
+ "AUDIO: %d Hz, %d ch, %s\n",
sh_audio->samplerate, sh_audio->channels.num,
- af_fmt2str_short(sh_audio->sample_format),
- sh_audio->i_bps * 8 * 0.001,
- ((float) sh_audio->i_bps / sh_audio->o_bps) * 100.0,
- sh_audio->i_bps, sh_audio->o_bps);
+ af_fmt2str_short(sh_audio->sample_format));
mp_msg(MSGT_IDENTIFY, MSGL_INFO,
"ID_AUDIO_BITRATE=%d\nID_AUDIO_RATE=%d\n" "ID_AUDIO_NCH=%d\n",
sh_audio->i_bps * 8, sh_audio->samplerate, sh_audio->channels.num);
diff --git a/core/mplayer.c b/core/mplayer.c
index 87bef0e3c5..7b3cf65026 100644
--- a/core/mplayer.c
+++ b/core/mplayer.c
@@ -1668,6 +1668,10 @@ static double written_audio_pts(struct MPContext *mpctx)
sh_audio_t *sh_audio = mpctx->sh_audio;
if (!sh_audio)
return MP_NOPTS_VALUE;
+
+ double bps = sh_audio->channels.num * sh_audio->samplerate *
+ sh_audio->samplesize;
+
// first calculate the end pts of audio that has been output by decoder
double a_pts = sh_audio->pts;
if (a_pts == MP_NOPTS_VALUE)
@@ -1676,13 +1680,13 @@ static double written_audio_pts(struct MPContext *mpctx)
// sh_audio->pts is the timestamp of the latest input packet with
// known pts that the decoder has decoded. sh_audio->pts_bytes is
// the amount of bytes the decoder has written after that timestamp.
- a_pts += sh_audio->pts_bytes / (double) sh_audio->o_bps;
+ a_pts += sh_audio->pts_bytes / bps;
// Now a_pts hopefully holds the pts for end of audio from decoder.
// Subtract data in buffers between decoder and audio out.
// Decoded but not filtered
- a_pts -= sh_audio->a_buffer_len / (double)sh_audio->o_bps;
+ a_pts -= sh_audio->a_buffer_len / bps;
// Data buffered in audio filters, measured in bytes of "missing" output
double buffered_output = af_calc_delay(sh_audio->afilter);
diff --git a/demux/stheader.h b/demux/stheader.h
index 8b584396ca..447cacfc02 100644
--- a/demux/stheader.h
+++ b/demux/stheader.h
@@ -92,7 +92,6 @@ typedef struct sh_audio {
int container_out_samplerate;
int samplesize;
struct mp_chmap channels;
- int o_bps; // == samplerate*samplesize*channels.num (uncompr. bytes/sec)
int i_bps; // == bitrate (compressed bytes/sec)
// decoder buffers:
int audio_out_minsize; // minimal output from decoder may be this much
diff --git a/stream/tv.c b/stream/tv.c
index e5768a350a..72b67b7f9e 100644
--- a/stream/tv.c
+++ b/stream/tv.c
@@ -794,7 +794,7 @@ static demuxer_t* demux_open_tv(demuxer_t *demuxer)
sh_audio->gsh->codec = "mp-pcm";
sh_audio->format = audio_format;
- sh_audio->i_bps = sh_audio->o_bps =
+ sh_audio->i_bps =
sh_audio->samplerate * sh_audio->samplesize *
sh_audio->channels.num;