Fix T38768: New "audio" button in 2.70 release does not 'mixdown' audio

Issue was caused by the way how audio output works from audaspace.
Now made it much closer to what's happening in ffmpeg.c and writeffmpeg.c.

Also fixed issues with incompatible combinations of codecs and formats
in mixdown settings.

2 files changed, 38 insertions, 4 deletions

diff --git a/intern/audaspace/ffmpeg/AUD_FFMPEGWriter.cpp b/intern/audaspace/ffmpeg/AUD_FFMPEGWriter.cpp
index d8f0d837fec..859227a5006 100644
--- a/intern/audaspace/ffmpeg/AUD_FFMPEGWriter.cpp
+++ b/intern/audaspace/ffmpeg/AUD_FFMPEGWriter.cpp
@@ -187,14 +187,18 @@ AUD_FFMPEGWriter::AUD_FFMPEGWriter(std::string filename, AUD_DeviceSpecs specs,
 			m_frame = av_frame_alloc();
 			if (!m_frame)
 				AUD_THROW(AUD_ERROR_FFMPEG, codec_error);
+			avcodec_get_frame_defaults(m_frame);
 			m_frame->linesize[0]    = m_input_size * samplesize;
 			m_frame->format         = m_codecCtx->sample_fmt;
+			m_frame->nb_samples     = m_codecCtx->frame_size;
 #  ifdef FFMPEG_HAVE_AVFRAME_SAMPLE_RATE
 			m_frame->sample_rate    = m_codecCtx->sample_rate;
 #  endif
 #  ifdef FFMPEG_HAVE_FRAME_CHANNEL_LAYOUT
 			m_frame->channel_layout = m_codecCtx->channel_layout;
 #  endif
+			m_audio_sample_size = av_get_bytes_per_sample(m_codecCtx->sample_fmt);
+			m_frame_pts = 0;
 #endif
 
 			try
@@ -272,13 +276,20 @@ void AUD_FFMPEGWriter::encode(sample_t* data)
 
 #ifdef FFMPEG_HAVE_ENCODE_AUDIO2
 	int got_output, ret;
+	m_frame->pts = m_frame_pts / av_q2d(m_codecCtx->time_base);
+	m_frame_pts++;
+#ifdef FFMPEG_HAVE_FRAME_CHANNEL_LAYOUT
+	m_frame->channel_layout = m_codecCtx->channel_layout;
+#endif
+
+	avcodec_fill_audio_frame(m_frame, m_codecCtx->channels, m_codecCtx->sample_fmt, reinterpret_cast<uint8_t*>(data),
+	                         m_frame->nb_samples * av_get_bytes_per_sample(m_codecCtx->sample_fmt) * m_codecCtx->channels, 1);
 
-	m_frame->data[0] = reinterpret_cast<uint8_t*>(data);
 	ret = avcodec_encode_audio2(m_codecCtx, &packet, m_frame, &got_output);
-	if (ret < 0)
+	if(ret < 0)
 		AUD_THROW(AUD_ERROR_FFMPEG, codec_error);
 
-	if (!got_output)
+	if(!got_output)
 		return;
 #else
 	sample_t* outbuf = m_output_buffer.getBuffer();
@@ -290,10 +301,23 @@ void AUD_FFMPEGWriter::encode(sample_t* data)
 	packet.data = reinterpret_cast<uint8_t*>(outbuf);
 #endif
 
+	if(packet.pts != AV_NOPTS_VALUE)
+		packet.pts = av_rescale_q(packet.pts, m_codecCtx->time_base, m_stream->time_base);
+	if(packet.dts != AV_NOPTS_VALUE)
+		packet.dts = av_rescale_q(packet.dts, m_codecCtx->time_base, m_stream->time_base);
+	if(packet.duration > 0)
+		packet.duration = av_rescale_q(packet.duration, m_codecCtx->time_base, m_stream->time_base);
+
 	packet.stream_index = m_stream->index;
 
-	if(av_interleaved_write_frame(m_formatCtx, &packet))
+	packet.flags |= AV_PKT_FLAG_KEY;
+
+	if(av_interleaved_write_frame(m_formatCtx, &packet)) {
+		av_free_packet(&packet);
 		AUD_THROW(AUD_ERROR_FFMPEG, write_error);
+	}
+
+	av_free_packet(&packet);
 }
 
 void AUD_FFMPEGWriter::write(unsigned int length, sample_t* buffer)
diff --git a/intern/audaspace/ffmpeg/AUD_FFMPEGWriter.h b/intern/audaspace/ffmpeg/AUD_FFMPEGWriter.h
index 310f69258ea..743d885fca8 100644
--- a/intern/audaspace/ffmpeg/AUD_FFMPEGWriter.h
+++ b/intern/audaspace/ffmpeg/AUD_FFMPEGWriter.h
@@ -83,6 +83,16 @@ private:
 	AVFrame *m_frame;
 
 	/**
+	 * PTS of next frame to write.
+	 */
+	int m_frame_pts;
+
+	/**
+	 * Number of bytes per sample.
+	 */
+	int m_audio_sample_size;
+
+	/**
 	 * The input buffer for the format converted data before encoding.
 	 */
 	AUD_Buffer m_input_buffer;