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FFmpeg: Fix encoing AAC

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This commit is contained in:
Jeffrey Pfau 2016-09-21 11:05:51 -07:00
parent 5c744a3f13
commit cf6006f9f8
2 changed files with 26 additions and 10 deletions
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1
CHANGES
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@ -5,6 +5,7 @@ Bugfixes:
- GBA Cheats: Fix holding onto pointers that may get invalidated
- Qt: Fix "close" button on Overrides view
- Qt: Fix saving overrides
- FFmpeg: Fix encoding AAC
Misc:
- All: Only update version info if needed

35
src/feature/ffmpeg/ffmpeg-encoder.c
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@ -381,6 +381,11 @@ void _ffmpegPostAudioFrame(struct mAVStream* stream, int16_t left, int16_t right
return;
}
if (encoder->absf && !left) {
// XXX: AVBSF doesn't like silence. Figure out why.
left = 1;
}
encoder->audioBuffer[encoder->currentAudioSample * 2] = left;
encoder->audioBuffer[encoder->currentAudioSample * 2 + 1] = right;
@ -415,40 +420,50 @@ void _ffmpegPostAudioFrame(struct mAVStream* stream, int16_t left, int16_t right
int gotData;
#ifdef FFMPEG_USE_PACKETS
avcodec_send_frame(encoder->audio, encoder->audioFrame);
gotData = avcodec_receive_packet(encoder->audio, &packet) == 0;
gotData = avcodec_receive_packet(encoder->audio, &packet);
gotData = (gotData == 0) && packet.size;
#else
avcodec_encode_audio2(encoder->audio, &packet, encoder->audioFrame, &gotData);
#endif
if (gotData) {
if (encoder->absf) {
AVPacket tempPacket = packet;
AVPacket tempPacket;
#ifdef FFMPEG_USE_NEW_BSF
int success = av_bsf_send_packet(encoder->absf, &packet) && av_bsf_receive_packet(encoder->absf, &packet);
int success = av_bsf_send_packet(encoder->absf, &packet);
if (success >= 0) {
success = av_bsf_receive_packet(encoder->absf, &tempPacket);
}
#else
int success = av_bitstream_filter_filter(encoder->absf, encoder->audio, 0,
&tempPacket.data, &tempPacket.size,
packet.data, packet.size, 0);
#endif
if (success > 0) {
if (success >= 0) {
#if LIBAVUTIL_VERSION_MAJOR >= 53
tempPacket.buf = av_buffer_create(tempPacket.data, tempPacket.size, av_buffer_default_free, 0, 0);
#endif
#ifdef FFMPEG_USE_PACKET_UNREF
av_packet_unref(&packet);
av_packet_move_ref(&packet, &tempPacket);
#else
av_free_packet(&packet);
packet = tempPacket;
#endif
packet.stream_index = encoder->audioStream->index;
av_interleaved_write_frame(encoder->context, &packet);
}
packet = tempPacket;
} else {
packet.stream_index = encoder->audioStream->index;
av_interleaved_write_frame(encoder->context, &packet);
}
packet.stream_index = encoder->audioStream->index;
av_interleaved_write_frame(encoder->context, &packet);
}
#ifdef FFMPEG_USE_PACKET_UNREF
av_packet_unref(&packet);
av_packet_unref(&packet);
#else
av_free_packet(&packet);
av_free_packet(&packet);
#endif
}