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Hi All,<br>
<br>
Here is my code for encoding Audio. Can anyone help and point me in
the direction I need to fix this issue?<br>
<br>
   AVFrame* frame;<br>
   AVPacket pkt = { 0 };<br>
   AVCodecContext *c = m_audio_st.st->codec;<br>
   int got_packet;<br>
   int dst_nb_samples;<br>
   int bytesReady = audioBuffer.size();<br>
   QByteArray samples;<br>
   int sampleSize;<br>
<br>
   if(m_errorCode != NoError){<br>
       return;<br>
   }<br>
<br>
   frame = m_audio_st.tmp_frame;<br>
   if(m_audio_st.swr_ctx == NULL || c == NULL<br>
      || frame == NULL){<br>
       return;<br>
   }<br>
<br>
   sampleSize = av_samples_get_buffer_size(NULL, c->channels,<br>
                                           c->frame_size,<br>
                                           c->sample_fmt, 0);<br>
<br>
   do{<br>
       samples.clear();<br>
<br>
       if(m_audioBuffer.size() + bytesReady == m_sampleSize){<br>
           samples.append(m_audioBuffer + audioBuffer);<br>
           m_audioBuffer.clear();<br>
       }<br>
       else if(m_audioBuffer.size() + bytesReady >
m_sampleSize){<br>
           m_audioBuffer.append(audioBuffer);<br>
           samples.append(m_audioBuffer.left(m_sampleSize));<br>
           m_audioBuffer.remove(0, m_sampleSize);<br>
       }<br>
       else if(m_audioBuffer.size() + bytesReady <
m_sampleSize){<br>
           m_audioBuffer.append(audioBuffer);<br>
           return;<br>
       }<br>
<br>
       bytesReady = 0;<br>
       audioBuffer.clear();<br>
       int ret;<br>
       if((ret = avcodec_fill_audio_frame(frame, c->channels,<br>
                                c->sample_fmt,<br>
                                (uint8_t*) samples.data(),<br>
                                sampleSize, 0)) < 0){<br>
           m_errorCode = AudioEncoderError;<br>
           Q_EMIT(m_errorCode);<br>
           continue;<br>
       }<br>
       frame->pts = m_audio_st.next_pts;<br>
       m_audio_st.next_pts += frame->nb_samples;<br>
       dst_nb_samples =
av_rescale_rnd(swr_get_delay(m_audio_st.swr_ctx, c->sample_rate)
+ frame->nb_samples,<br>
                                       c->sample_rate,
c->sample_rate, AV_ROUND_UP);<br>
       av_assert0(dst_nb_samples == frame->nb_samples);<br>
<br>
       ret = av_frame_make_writable(m_audio_st.frame);<br>
       if(ret < 0){<br>
           m_errorCode = AudioEncoderError;<br>
           Q_EMIT(m_errorCode);<br>
           continue;<br>
       }<br>
<br>
       ret = swr_convert(m_audio_st.swr_ctx,<br>
                         m_audio_st.frame->data, dst_nb_samples,<br>
                         (const uint8_t **)frame->data,
frame->nb_samples);<br>
       if(ret < 0){<br>
           m_errorCode = AudioEncoderError;<br>
           Q_EMIT(m_errorCode);<br>
           return;<br>
       }<br>
<br>
<br>
       frame = m_audio_st.frame;<br>
       frame->pts = av_rescale_q(m_audio_st.samples_count, {1,
c->sample_rate}, c->time_base);<br>
<br>
       m_audio_st.samples_count += dst_nb_samples;<br>
       av_init_packet(&pkt);<br>
       ret = avcodec_encode_audio2(c, &pkt, frame,
&got_packet);<br>
       if(ret < 0){<br>
           m_errorCode = AudioEncoderError;<br>
           Q_EMIT(m_errorCode);<br>
           continue;<br>
       }<br>
<br>
       if(got_packet){<br>
           ret = write_frame(m_oc, &c->time_base,
m_audio_st.st, &pkt);<br>
           if(ret < 0){<br>
               m_errorCode = AudioEncoderError;<br>
               Q_EMIT(m_errorCode);<br>
               continue;<br>
           }<br>
       }<br>
   }while(m_audioBuffer.size() > m_sampleSize);<br>
<br>
<div class="moz-cite-prefix">
<meta http-equiv="content-type" content="text/html; charset=utf-8">
<title></title>
Sincerely,<span style="font-size:12.0pt;color:#0070C0"><br>
Kevin J. Brooks<o:p></o:p></span><span
style="font-size:12.0pt;color:#0070C0"></span><br>
<br>
On 9/22/2015 8:34 AM, Kevin J. Brooks wrote:<br>
</div>
<blockquote cite="mid:560158FE.8090502@r2c-ss.com" type="cite">
<meta http-equiv="content-type" content="text/html; charset=utf-8">
Hi All,<br>
<br>
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.Â
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.<br>
<br>
Can anyone give me and idea where to start to fix this problem?<br>
<div class="moz-signature">-- <br>
<meta http-equiv="content-type" content="text/html;
charset=utf-8">
<title></title>
Sincerely,<span style="font-size:12.0pt;color:#0070C0"><br>
Kevin J. Brooks<o:p></o:p></span><span
style="font-size:12.0pt;color:#0070C0"><br>
Senior Software Engineer<o:p></o:p></span><span
style="font-size:14.0pt;color:black;mso-themecolor: text1"><br>
</span><br>
</div>
<br>
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<br>
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