[FFmpeg-devel] [PATCH] libvorbis encoder: fix non monotone timestamp problem
Nicolas George
nicolas.george
Mon Oct 13 19:34:41 CEST 2008
Hi.
Sometimes, the libvorbis encoder dies with a "non monotone timestamps"
problem. For example, with SVN revision 15614 configured with
"--enable-libvorbis --enable-debug", the following command line:
sox -t raw -s2 -c 2 -r 44100 /dev/zero -t wav - trim 0 10 synth 440 |
./ffmpeg_g-orig -f wav -i - -acodec libvorbis -y /tmp/out.ogg
(sox produces a 10-seconds WAV file with a 440 Hz sine)
ffmpeg dies with:
[libvorbis @ 0xdd5d80]error, non monotone timestamps 440960 >= 440960
av_interleaved_write_frame(): Error while opening file
(the rest of the output is just regular stuff, version and streams
identification; I can copy-paste it here, but there is really nothing
useful)
Furthermore, if the output format requires processing at the end, it is not
done; AVI index, for example, is not built.
The problem is that at the end of the stream libavcodec calls
vorbis_analysis_wrote repeatedly with a 0 size. The libvorbisenc
documentation is rather terse about it, but it seems that this should be
done only once: otherwise, this causes libvorbisenc to output the last
packet a second time, with the same granulepos, causing the timestamp
problem.
The fix seems obvious: add a flag to avoid calling vorbis_analysis_wrote
twice with an empty buffer.
The attached patch does that. The second one fixes the indentation. They do
not break the regression tests.
Regards,
--
Nicolas George
-------------- next part --------------
diff --git a/libavcodec/libvorbis.c b/libavcodec/libvorbis.c
index ce796a0..6963e70 100644
--- a/libavcodec/libvorbis.c
+++ b/libavcodec/libvorbis.c
@@ -42,6 +42,7 @@ typedef struct OggVorbisContext {
vorbis_block vb ;
uint8_t buffer[BUFFER_SIZE];
int buffer_index;
+ int eof;
/* decoder */
vorbis_comment vc ;
@@ -136,10 +137,13 @@ static int oggvorbis_encode_frame(AVCodecContext *avccontext,
int buf_size, void *data)
{
OggVorbisContext *context = avccontext->priv_data ;
- float **buffer ;
ogg_packet op ;
signed short *audio = data ;
- int l, samples = data ? OGGVORBIS_FRAME_SIZE : 0;
+ int l;
+
+ if(data) {
+ int samples = OGGVORBIS_FRAME_SIZE;
+ float **buffer ;
buffer = vorbis_analysis_buffer(&context->vd, samples) ;
@@ -154,6 +158,11 @@ static int oggvorbis_encode_frame(AVCodecContext *avccontext,
}
vorbis_analysis_wrote(&context->vd, samples) ;
+ } else {
+ if(!context->eof)
+ vorbis_analysis_wrote(&context->vd, 0) ;
+ context->eof = 1;
+ }
while(vorbis_analysis_blockout(&context->vd, &context->vb) == 1) {
vorbis_analysis(&context->vb, NULL);
-------------- next part --------------
diff --git a/libavcodec/libvorbis.c b/libavcodec/libvorbis.c
index 6963e70..03c9f88 100644
--- a/libavcodec/libvorbis.c
+++ b/libavcodec/libvorbis.c
@@ -145,19 +145,17 @@ static int oggvorbis_encode_frame(AVCodecContext *avccontext,
int samples = OGGVORBIS_FRAME_SIZE;
float **buffer ;
- buffer = vorbis_analysis_buffer(&context->vd, samples) ;
-
- if(context->vi.channels == 1) {
- for(l = 0 ; l < samples ; l++)
- buffer[0][l]=audio[l]/32768.f;
- } else {
- for(l = 0 ; l < samples ; l++){
- buffer[0][l]=audio[l*2]/32768.f;
- buffer[1][l]=audio[l*2+1]/32768.f;
+ buffer = vorbis_analysis_buffer(&context->vd, samples) ;
+ if(context->vi.channels == 1) {
+ for(l = 0 ; l < samples ; l++)
+ buffer[0][l]=audio[l]/32768.f;
+ } else {
+ for(l = 0 ; l < samples ; l++){
+ buffer[0][l]=audio[l*2]/32768.f;
+ buffer[1][l]=audio[l*2+1]/32768.f;
+ }
}
- }
-
- vorbis_analysis_wrote(&context->vd, samples) ;
+ vorbis_analysis_wrote(&context->vd, samples) ;
} else {
if(!context->eof)
vorbis_analysis_wrote(&context->vd, 0) ;
-------------- next part --------------
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