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segafilm: use the sample rate as the time base for audio streams

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This commit is contained in:
Justin Ruggles 2012-01-11 11:07:40 -05:00
parent ea289186f0
commit 4da374f8a9
1 changed files with 7 additions and 4 deletions
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11
libavformat/segafilm.c
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@ -196,8 +196,13 @@ static int film_read_header(AVFormatContext *s)
if (!film->sample_table)
return AVERROR(ENOMEM);
for(i=0; i<s->nb_streams; i++)
avpriv_set_pts_info(s->streams[i], 33, 1, film->base_clock);
for (i = 0; i < s->nb_streams; i++) {
st = s->streams[i];
if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO)
avpriv_set_pts_info(st, 33, 1, film->base_clock);
else
avpriv_set_pts_info(st, 64, 1, film->audio_samplerate);
}
audio_frame_counter = 0;
for (i = 0; i < film->sample_count; i++) {
@ -212,8 +217,6 @@ static int film_read_header(AVFormatContext *s)
if (AV_RB32(&scratch[8]) == 0xFFFFFFFF) {
film->sample_table[i].stream = film->audio_stream_index;
film->sample_table[i].pts = audio_frame_counter;
film->sample_table[i].pts *= film->base_clock;
film->sample_table[i].pts /= film->audio_samplerate;
if (film->audio_type == CODEC_ID_ADPCM_ADX)
audio_frame_counter += (film->sample_table[i].sample_size * 32 /