Hi All,
Here is my code for encoding Audio. Can anyone help and point me in the
direction I need to fix this issue?
AVFrame* frame;
AVPacket pkt = { 0 };
AVCodecContext *c = m_audio_st.st->codec;
int got_packet;
int dst_nb_samples;
int bytesReady = audioBuffer.size();
QByteArray samples;
int sampleSize;
if(m_errorCode != NoError){
return;
}
frame = m_audio_st.tmp_frame;
if(m_audio_st.swr_ctx == NULL || c == NULL
|| frame == NULL){
return;
}
sampleSize = av_samples_get_buffer_size(NULL, c->channels,
c->frame_size,
c->sample_fmt, 0);
do{
samples.clear();
if(m_audioBuffer.size() + bytesReady == m_sampleSize){
samples.append(m_audioBuffer + audioBuffer);
m_audioBuffer.clear();
}
else if(m_audioBuffer.size() + bytesReady > m_sampleSize){
m_audioBuffer.append(audioBuffer);
samples.append(m_audioBuffer.left(m_sampleSize));
m_audioBuffer.remove(0, m_sampleSize);
}
else if(m_audioBuffer.size() + bytesReady < m_sampleSize){
m_audioBuffer.append(audioBuffer);
return;
}
bytesReady = 0;
audioBuffer.clear();
int ret;
if((ret = avcodec_fill_audio_frame(frame, c->channels,
c->sample_fmt,
(uint8_t*) samples.data(),
sampleSize, 0)) < 0){
m_errorCode = AudioEncoderError;
Q_EMIT(m_errorCode);
continue;
}
frame->pts = m_audio_st.next_pts;
m_audio_st.next_pts += frame->nb_samples;
dst_nb_samples =
av_rescale_rnd(swr_get_delay(m_audio_st.swr_ctx, c->sample_rate) +
frame->nb_samples,
c->sample_rate, c->sample_rate,
AV_ROUND_UP);
av_assert0(dst_nb_samples == frame->nb_samples);
ret = av_frame_make_writable(m_audio_st.frame);
if(ret < 0){
m_errorCode = AudioEncoderError;
Q_EMIT(m_errorCode);
continue;
}
ret = swr_convert(m_audio_st.swr_ctx,
m_audio_st.frame->data, dst_nb_samples,
(const uint8_t **)frame->data,
frame->nb_samples);
if(ret < 0){
m_errorCode = AudioEncoderError;
Q_EMIT(m_errorCode);
return;
}
frame = m_audio_st.frame;
frame->pts = av_rescale_q(m_audio_st.samples_count, {1,
c->sample_rate}, c->time_base);
m_audio_st.samples_count += dst_nb_samples;
av_init_packet(&pkt);
ret = avcodec_encode_audio2(c, &pkt, frame, &got_packet);
if(ret < 0){
m_errorCode = AudioEncoderError;
Q_EMIT(m_errorCode);
continue;
}
if(got_packet){
ret = write_frame(m_oc, &c->time_base, m_audio_st.st, &pkt);
if(ret < 0){
m_errorCode = AudioEncoderError;
Q_EMIT(m_errorCode);
continue;
}
}
}while(m_audioBuffer.size() > m_sampleSize);
Sincerely,
Kevin J. Brooks
On 9/22/2015 8:34 AM, Kevin J. Brooks wrote:
Hi All,
Obviously I don't understand enough about encoding video with audio to
figure out how to fix this. If I take the captured sound from the
microphone and pipe it directly back to my headphones, it is crystal
clear. Well maybe a little tinny sounding, but clear. However, when I
try to encode it into the video, it is not synced up with the images
and there is a bunch of white noise. It is only after the video has
ended that you finally can barely hear my voice and whistles.
Can anyone give me and idea where to start to fix this problem?
--
Sincerely,
Kevin J. Brooks
Senior Software Engineer
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http://ffmpeg.org/mailman/listinfo/libav-user
