Hi,
I'm having some issues with AAC decoding with recent version of ffmpeg:
I'm using this code to decode AAC Frames (from a stream).
I had some troubles with the decoding - everything runs fine but the
after a few seconds playing
the decoded buffer (m_avAACFrame->data) only contains zeros.
int gotframe=0;
if(!m_avAACFrame)
m_avAACFrame=avcodec_alloc_frame();
else
avcodec_get_frame_defaults(m_avAACFrame);
int buff=avcodec_decode_audio4(pAACCodecCtx, m_avAACFrame,
&gotframe, &avAACpkt);
if (gotframe==0 || buff<0)
{
*outputsize=0;
return;
}
*outputsize=av_samples_get_buffer_size(NULL, 2,
m_avAACFrame->nb_samples, pAACCodecCtx->sample_fmt, 1);
memcpy(outbuffer,*(byte**)&m_avAACFrame->data,*outputsize);
So i switched to the latest version of ffmpeg (Zeranoe FFmpeg for
Windows) and now i have the problem that the output sample format is
always AV_SAMPLE_FMT_FLT instead of AV_SAMPLE_FMT_S16 ?
pAACCodecCtx->bit_rate = 64000;
pAACCodecCtx->sample_rate = 44100;
pAACCodecCtx->channels = 2;
pAACCodecCtx->channel_layout = 3;
pAACCodecCtx->sample_fmt = AV_SAMPLE_FMT_S16;
//pAACCodecCtx->bit_rate = 64000;
//pAACCodecCtx->sample_rate = fmtp[11];
//pAACCodecCtx->channels = fmtp[7];
//c->sample_fmt = AV_SAMPLE_FMT_DBL;
//c->profile = FF_PROFILE_AAC_SSR;
/* open it */
if (avcodec_open2(pAACCodecCtx, AACCodec,NULL) < 0)
{
return false;
}
Is AAC not properly supported on ffmpeg ?
Thanks for your help
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