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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.
This commit is contained in:
wm4 2013-07-11 19:15:09 +02:00
parent 7a4f9cc4d2
commit 23e303859a
4 changed files with 9 additions and 12 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

10
audio/decode/dec_audio.c
View File

@ -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);

8
core/mplayer.c
View File

@ -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);

1
demux/stheader.h
View File

@ -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

2
stream/tv.c
View File

@ -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;