Hi
I checked doc/examples/muxingc.c file where an example generating MPEG AV file
is explained. Then I try to change AV format to "webm" (video is encoded with
VP8 and audio with vorbis encoder). I have problems with setting the right
parameters for vorbis encoder. When I debugging through the code I figured out
that vorbis only accepts (AV_SAMPLE_FMT_FLTP) sample format type. Example was
done with (AV_SAMPLE_FMT_S16). I looked into source code to figure out that
when I call
avcodec_fill_audio_frame(AVFrame, channels, sample_format, buffer, buf_size,
align),
in case of AV_SAMPLE_FMT_FLTP the buffer should be type float* and audio
samples should have values between -1.0 to 1.0. For more than one channel
values are not interleaved (FLTP - float plain) but they are followed by each
other (array: all values for channel0, all values for channel 1, ...).
When I accept those changes in my code, unfortunately I still don't get correct
result. If I use mono (1 channel only) then the flag (got_packet) returned from
function avcodec_encode_audio2 is set only once (after around 5 consecutive
calls), with AVPacket->pts timestamp set to some huge values. Because of that
only video is encoded.
When I set stereo mode I get error from function av_interleaved_write_frame
(-12).
I tested the same code and setting AV format to "asf", where audio is encoded
with WMA2 encoder and also accepts AV_SAMPLE_FMT_FLTP sample format type. I got
correct AV file which can be played with VLC player or Windows media player.
I think I still need to set some flags for vorbis encoder, but I can't figure
out. I would appreciate any suggestions.
Best regards
For audio encoder I set those parameters:
c->sample_fmt = AV_SAMPLE_FMT_FLTP;
c->sample_rate = 44100;
c->channels = 1;
My code to prepare samples in AV_SAMPLE_FMT_FLTP sample format:
void TT_Transcode::get_audio_frame_fltp(float *fsamples, int frame_size, int
nb_channels)
{
int j, i;
float v;
float *q;
for (j = 0; j < frame_size; j++) {
v = (sin(t) * 10000.0) / 32767.0f; //values between -1.0 and +1.0
fsamples[ j ] = v;
for (i = 1; i < nb_channels; i++)
{
fsamples[i * in_linesize + j] = v;
}
t += tincr;
tincr += tincr2;
}
}_______________________________________________
Libav-user mailing list
[email protected]
http://ffmpeg.org/mailman/listinfo/libav-user
