Support planar sample formats in audio mixdown

Now libav-10 should work for output of ac3 container.

2 files changed, 36 insertions, 2 deletions

diff --git a/intern/audaspace/ffmpeg/AUD_FFMPEGWriter.cpp b/intern/audaspace/ffmpeg/AUD_FFMPEGWriter.cpp
index dcacf71a2d3..d30835da4e5 100644
--- a/intern/audaspace/ffmpeg/AUD_FFMPEGWriter.cpp
+++ b/intern/audaspace/ffmpeg/AUD_FFMPEGWriter.cpp
@@ -169,6 +169,19 @@ AUD_FFMPEGWriter::AUD_FFMPEGWriter(std::string filename, AUD_DeviceSpecs specs,
 			if(!codec)
 				AUD_THROW(AUD_ERROR_FFMPEG, codec_error);
 
+			if(codec->sample_fmts) {
+				// Check if the prefered sample format for this codec is supported.
+				const enum AVSampleFormat *p = codec->sample_fmts;
+				for(; *p != -1; p++) {
+					if(*p == m_stream->codec->sample_fmt)
+						break;
+				}
+				if(*p == -1) {
+					// Sample format incompatible with codec. Defaulting to a format known to work.
+					m_stream->codec->sample_fmt = codec->sample_fmts[0];
+				}
+			}
+
 			if(avcodec_open2(m_codecCtx, codec, NULL))
 				AUD_THROW(AUD_ERROR_FFMPEG, codec_error);
 
@@ -199,8 +212,11 @@ AUD_FFMPEGWriter::AUD_FFMPEGWriter(std::string filename, AUD_DeviceSpecs specs,
 #  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_sample_size = av_get_bytes_per_sample(m_codecCtx->sample_fmt);
 			m_frame_pts = 0;
+			m_deinterleave = av_sample_fmt_is_planar(m_codecCtx->sample_fmt);
+			if(m_deinterleave)
+				m_deinterleave_buffer.resize(m_input_size * m_codecCtx->channels * m_sample_size);
 #endif
 
 			try
@@ -284,6 +300,17 @@ void AUD_FFMPEGWriter::encode(sample_t* data)
 	m_frame->channel_layout = m_codecCtx->channel_layout;
 #endif
 
+	if(m_deinterleave) {
+		for(int channel = 0; channel < m_codecCtx->channels; channel++) {
+			for(int i = 0; i < m_frame->nb_samples; i++) {
+				memcpy(reinterpret_cast<uint8_t*>(m_deinterleave_buffer.getBuffer()) + (i + channel * m_frame->nb_samples) * m_sample_size,
+					   reinterpret_cast<uint8_t*>(data) + (m_codecCtx->channels * i + channel) * m_sample_size, m_sample_size);
+			}
+		}
+
+		data = m_deinterleave_buffer.getBuffer();
+	}
+
 	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);
 
diff --git a/intern/audaspace/ffmpeg/AUD_FFMPEGWriter.h b/intern/audaspace/ffmpeg/AUD_FFMPEGWriter.h
index 743d885fca8..492aa35ff12 100644
--- a/intern/audaspace/ffmpeg/AUD_FFMPEGWriter.h
+++ b/intern/audaspace/ffmpeg/AUD_FFMPEGWriter.h
@@ -90,7 +90,14 @@ private:
 	/**
 	 * Number of bytes per sample.
 	 */
-	int m_audio_sample_size;
+	int m_sample_size;
+
+	/**
+	 * Need to de-interleave audio for planar sample formats.
+	 */
+	bool m_deinterleave;
+
+	AUD_Buffer m_deinterleave_buffer;
 
 	/**
 	 * The input buffer for the format converted data before encoding.