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audio seems correct, but need more elegant way, and still bad interleaving
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@ -237,6 +237,11 @@ void CVideo::nextVideoFrame()
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void CVideo::nextAudioFrame(short* pData)
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{
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if (m_Recording && m_HasAudio)
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{
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m_aseq += 1;
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for(int i=0; i < 1024; i++)
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m_aBuffer[i+(m_aseq%2)*1024] = pData[i];
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if(m_aseq % 2)
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{
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m_ProcessingAudioFrame = true;
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m_AudioStream.frame->pts = m_AudioStream.enc->frame_number;
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@ -262,14 +267,14 @@ void CVideo::nextAudioFrame(short* pData)
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// );
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// dbg_msg("video_recorder", "dst_nb_samples: %d", dst_nb_samples);
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fwrite(pData, sizeof(short), 1024, m_dbgfile);
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//fwrite(pData, sizeof(short), 1024, m_dbgfile);
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av_samples_fill_arrays(
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(uint8_t**)m_AudioStream.tmp_frame->data,
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0, // pointer to linesize (int*)
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(const uint8_t*)pData,
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(const uint8_t*)m_aBuffer,
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2, // channels
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m_AudioStream.tmp_frame->nb_samples,
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m_AudioStream.tmp_frame->nb_samples * 2,
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AV_SAMPLE_FMT_S16,
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0 // align
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);
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@ -283,9 +288,9 @@ void CVideo::nextAudioFrame(short* pData)
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ret = swr_convert(
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m_AudioStream.swr_ctx,
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m_AudioStream.frame->data,
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m_AudioStream.frame->nb_samples,
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m_AudioStream.frame->nb_samples * 2,
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(const uint8_t **)m_AudioStream.tmp_frame->data,
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m_AudioStream.tmp_frame->nb_samples
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m_AudioStream.tmp_frame->nb_samples * 2
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);
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if (ret < 0)
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@ -305,6 +310,7 @@ void CVideo::nextAudioFrame(short* pData)
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m_ProcessingAudioFrame = false;
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}
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}
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}
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void CVideo::fill_audio_frame()
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{
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@ -510,6 +516,8 @@ void CVideo::open_audio()
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dbg_msg("video_recorder", "Failed to initialize the resampling context");
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exit(1);
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}
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m_aseq = 0;
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}
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@ -94,6 +94,8 @@ private:
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int m_Height;
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char m_Name[256];
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FILE *m_dbgfile;
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int m_aseq;
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short m_aBuffer[2048];
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int m_FPS;
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