[Libav-user] av_write_frame fails when encoding a larger audio file to .mpg

Jan drabner drabner at zoobe.com
Fri Nov 23 16:35:51 CET 2012


I am encoding live rendered video data and an existing .wav file into an

To do that I first write all audio frames, and then the video frames as
they come in from the render engine. For smaller .wav files (< 25 seconds),
everything works perfectly fine. But as soon as I use a longer .wav file,
av_write_frame (when writing the audio frame) just returns -1 after having
written some 100 frames. It is never the same frame at which it fails, also
it is never the last frame. All test .wav files can be played perfectly
with any player I tested.

I am following the muxing
example<http://ffmpeg.org/doxygen/trunk/muxing_8c-source.html>(more or

Here is my function that writes an audio frame:

void write_audio_frame( Cffmpeg_dll * ptr, AVFormatContext *oc,
AVStream *st, int16_t sample_val )
AVCodecContext *c;
AVPacket pkt = { 0 }; // data and size must be 0;
AVFrame *frame = avcodec_alloc_frame();
int got_packet;

c = st->codec;

get_audio_frame(ptr, ptr->samples, ptr->audio_input_frame_size, c->channels);
frame->nb_samples =  ptr->audio_input_frame_size;
int result = avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
                        (uint8_t *) ptr->samples,
                         ptr->audio_input_frame_size *
                        av_get_bytes_per_sample(c->sample_fmt) *
                        c->channels, 0);
if (result != 0)
    av_log(c, AV_LOG_ERROR, "Error filling audio frame. Code: %i\n", result);

result = avcodec_encode_audio2(c, &pkt, frame, &got_packet);
if (result != 0)
    av_log(c, AV_LOG_ERROR, "Error encoding audio. Code: %i\n", result);

if (c->coded_frame && c->coded_frame->pts != AV_NOPTS_VALUE)
    pkt.pts= av_rescale_q(c->coded_frame->pts, c->time_base, st->time_base);
pkt.flags |= AV_PKT_FLAG_KEY;

pkt.stream_index= st->index;
av_log(c, AV_LOG_ERROR, "Got? %i Pts: %i Dts: %i Flags: %i Side Elems:
%i Size: %i\n",
        got_packet, pkt.pts, pkt.dts, pkt.flags, pkt.side_data_elems, pkt.size);

/* write the compressed frame in the media file */
result = av_write_frame(oc, &pkt);
if (result != 0)
    av_log(c, AV_LOG_ERROR, "Error while writing audio frame. Result:
%i\n", result);


And here is my get_audio_frame function:

void get_audio_frame( Cffmpeg_dll* ptr, int16_t* samples, int
frame_size, int nb_channels, int16_t sample_val )
    fread( samples, sizeof( int16_t ), frame_size * nb_channels,
ptr->fp_sound_input );

And finally, this is the loop in which I write all audio frames (don't
worry about the .wav header, I skipped it before that loop):

while (!feof(ptr->fp_sound_input))
        write_audio_frame( ptr, ptr->oc, ptr->audio_st, -1 );

As you can see, I'm outputting almost everything in the packet and check
for any possible error. Other than av_write_frame failing after some time
when I am encoding a longer audio file, everything seems perfectly fine.
All the packet values I am tracking are 100% the same for all frames
(except the data pointer, obviously). Also, as stated, the same procedure
works flawlessly for shorter fp_sound_input files. avcodec_encode_audio2()
and avcodec_fill_audio_frame() also never fail.

The codecs I use for encoding are CODEC_ID_MPEG2VIDEO (video) and
CODEC_ID_MP2 (audio). The .wav files are saved in PCM 16 LE (all use the
exact same encoding).

What could be wrong here?

Jan Drabner ׀ TD Programming Engine & Animation

 zoobe message entertainment gmbh

kurfürstendamm 226 l 10719 berlin

  email: drabner at zoobe.com <nachname at zoobe.com> l  mob: 0172 7017640

  geschäftsführer: lenard f. krawinkel

 tel: +49 30. 288 838 88 l site: *www.zoobe.com* <http://www.zoobe.com/> ׀
email: *info at zoobe.com* <info at zoobe.com>

  amtsgericht charlottenburg, berlin ׀ hrb-nr. 11 42 79 b
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://ffmpeg.org/pipermail/libav-user/attachments/20121123/540d9083/attachment.html>

More information about the Libav-user mailing list