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ffmpegenc: Fix timestamp resyncing

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Properly convert bytes into time using sample size, sample rate
and channels number, instead of sample rate only.

This can cause huge timestamp discontinuities (even though the
durations remain correct) and might cause problems to muxers.

Fixes #623388
This commit is contained in:
Thiago Santos 2010-07-02 18:38:06 -03:00
parent 2f6086e495
commit b8f556a566
1 changed files with 13 additions and 5 deletions
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18
ext/ffmpeg/gstffmpegenc.c
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@ -930,15 +930,18 @@ gst_ffmpegenc_chain_audio (GstPad * pad, GstBuffer * inbuf)
ffmpegenc->adapter_consumed = 0;
} else {
GstClockTime upstream_time;
GstClockTime consumed_time;
guint64 bytes;
/* use timestamp at head of the adapter */
GST_LOG_OBJECT (ffmpegenc, "taking adapter timestamp %" GST_TIME_FORMAT,
GST_TIME_ARGS (ffmpegenc->adapter_ts));
timestamp = ffmpegenc->adapter_ts;
timestamp +=
consumed_time =
gst_util_uint64_scale (ffmpegenc->adapter_consumed, GST_SECOND,
ctx->sample_rate);
timestamp = ffmpegenc->adapter_ts + consumed_time;
GST_LOG_OBJECT (ffmpegenc, "taking adapter timestamp %" GST_TIME_FORMAT
" and adding consumed time %" GST_TIME_FORMAT,
GST_TIME_ARGS (ffmpegenc->adapter_ts), GST_TIME_ARGS (consumed_time));
/* check with upstream timestamps, if too much deviation,
* forego some timestamp perfection in favour of upstream syncing
* (particularly in case these do not happen to come in multiple
@ -948,7 +951,8 @@ gst_ffmpegenc_chain_audio (GstPad * pad, GstBuffer * inbuf)
GstClockTimeDiff diff;
upstream_time +=
gst_util_uint64_scale (bytes, GST_SECOND, ctx->sample_rate);
gst_util_uint64_scale (bytes, GST_SECOND,
ctx->sample_rate * osize * ctx->channels);
diff = upstream_time - timestamp;
/* relaxed difference, rather than half a sample or so ... */
if (diff > GST_SECOND / 10 || diff < -GST_SECOND / 10) {
@ -978,6 +982,10 @@ gst_ffmpegenc_chain_audio (GstPad * pad, GstBuffer * inbuf)
GST_LOG_OBJECT (ffmpegenc, "taking %u bytes from the adapter",
frame_bytes);
/* Note that we take frame_bytes and add frame_size.
* Makes sense when resyncing because you don't have to count channels
* or samplesize to divide by the samplerate */
/* take an audio buffer out of the adapter */
in_data = (guint8 *) gst_adapter_peek (ffmpegenc->adapter, frame_bytes);
ffmpegenc->adapter_consumed += frame_size;