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gst/qtdemux/qtdemux.c: fix audio chunk size/timestamp calculation

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Original commit message from CVS:
2004-01-11  Benjamin Otte  <in7y118@public.uni-hamburg.de>

* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header),
(qtdemux_parse_trak):
fix audio chunk size/timestamp calculation
This commit is contained in:
Benjamin Otte 2004-01-11 17:49:56 +00:00
parent fa66fa64c5
commit 96bbccf5ef
2 changed files with 23 additions and 5 deletions
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6
ChangeLog
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@ -1,3 +1,9 @@
2004-01-11 Benjamin Otte <in7y118@public.uni-hamburg.de>
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header),
(qtdemux_parse_trak):
fix audio chunk size/timestamp calculation
2004-01-11 Benjamin Otte <in7y118@public.uni-hamburg.de>
* ext/ffmpeg/gstffmpegcodecmap.c: (gst_ffmpeg_codecid_to_caps):

22
gst/qtdemux/qtdemux.c
View file

@ -26,7 +26,7 @@
#include <string.h>
#include <zlib.h>
#define g_print(x...)
#define g_print(...)
#define QTDEMUX_GUINT32_GET(a) GUINT32_FROM_BE(*(guint32 *)(a))
#define QTDEMUX_GUINT16_GET(a) GUINT16_FROM_BE(*(guint16 *)(a))
@ -77,8 +77,11 @@ struct _QtDemuxStream {
int width;
int height;
float fps;
double rate;
int n_channels;
guint bytes_per_frame;
guint samples_per_packet;
};
enum QtDemuxState {
@ -477,6 +480,7 @@ static void gst_qtdemux_loop_header (GstElement *element)
cur_offset = gst_bytestream_tell(qtdemux->bs);
if(offset != cur_offset){
GST_DEBUG ("seeking to offset %d",offset);
g_print ("seeking to offset %d\n",offset);
ret = gst_bytestream_seek(qtdemux->bs, offset, GST_SEEK_METHOD_SET);
GST_DEBUG ("seek returned %d",ret);
return;
@ -1296,9 +1300,14 @@ static void qtdemux_parse_trak(GstQTDemux *qtdemux, GNode *trak)
if(version == 0x00010000){
g_print("samples/packet: %d\n", QTDEMUX_GUINT32_GET(stsd->data+offset + 20));
stream->samples_per_packet = QTDEMUX_GUINT32_GET(stsd->data+offset + 20);
g_print("bytes/packet: %d\n", QTDEMUX_GUINT32_GET(stsd->data+offset + 24));
g_print("bytes/frame: %d\n", QTDEMUX_GUINT32_GET(stsd->data+offset + 28));
stream->bytes_per_frame = QTDEMUX_GUINT32_GET(stsd->data+offset + 28);
g_print("bytes/sample: %d\n", QTDEMUX_GUINT32_GET(stsd->data+offset + 32));
} else {
stream->bytes_per_frame = stream->n_channels * QTDEMUX_GUINT16_GET(stsd->data+offset + 10);
stream->samples_per_packet = 1;
}
stream->caps = qtdemux_audio_caps(qtdemux,
@ -1425,14 +1434,16 @@ done:
}
samples[j].chunk = j;
samples[j].offset = chunk_offset;
samples[j].size = samples_per_chunk * stream->n_channels * sample_width;
samples[j].size = samples_per_chunk * stream->bytes_per_frame / stream->samples_per_packet;
samples[j].duration = samples_per_chunk * GST_SECOND / stream->rate;
samples[j].timestamp = j == 0 ? 0 : samples[j - 1].timestamp + samples[j - 1].duration;
samples[j].sample_index = sample_index;
sample_index += samples_per_chunk;
if(j>=n_samples)goto done2;
}
}
/*
done2:
n_sample_times = QTDEMUX_GUINT32_GET(stts->data + 12);
g_print("n_sample_times = %d\n",n_sample_times);
timestamp = 0;
@ -1449,13 +1460,14 @@ done2:
samples[index].timestamp = timestamp;
size = samples[index+1].sample_index - samples[index].sample_index;
time = (GST_SECOND * duration * samples[index].size)/stream->timescale ;
time = GST_SECOND / stream->rate; //(GST_SECOND * duration * samples[index].size)/stream->timescale ;
timestamp += time;
samples[index].duration = time;
}
}
*/
}
done2:
#if 0
for(i=0;i<n_samples;i++){
g_print("%d: %d %d %d %d %" G_GUINT64_FORMAT "\n",i,