1 files changed, 47 insertions, 0 deletions
diff --git a/audio/stream.c b/audio/stream.c
new file mode 100644
index 0000000..7bab674
--- /dev/null
+++ b/audio/stream.c
@@ -0,0 +1,47 @@
+#include "stream.h"
+
+int stream_init(struct Stream *stream, AVFormatContext *s, int codec_id, int sample_rate, int format, const AVChannelLayout *layout,  AVDictionary **opts)
+{
+	int err;
+	AVStream *av_stream;
+
+	if ((av_stream = avformat_new_stream(s, NULL)) == NULL)
+		return AVERROR(ENOMEM);
+	av_stream->codecpar->codec_type  = AVMEDIA_TYPE_AUDIO;
+	av_stream->codecpar->codec_id    = codec_id;
+	av_stream->codecpar->sample_rate = sample_rate;
+	av_stream->codecpar->format      = format;
+	av_stream->codecpar->ch_layout   = *layout;
+
+	if ((err = avformat_init_output(s, opts)) < 0)
+		return err;
+
+	stream->av_stream = av_stream;
+	stream->s = s;
+	return 0;
+}
+
+int stream_send (struct Stream *stream, AVPacket *pkt)
+{
+	int err;
+	int64_t duration;
+
+	// NOTE:
+	// Before encoding;
+	// - frame_out->pts DOES matter
+	//
+	// After encoding;
+	// - pkt->pts is set to something
+	// - pkt->timebase is unset, but would be '1 / output_sample_rate'
+	// - pkt->duration is set to the amount of sample in the packet
+	// - pkt->dts seems to be pkt->pts
+	pkt->stream_index = stream->av_stream->index;
+	//av_packet_rescale_ts(pkt_out, encoder->time_base, stream->time_base);
+	//print_pts(pkt_out, &stream->time_base);
+
+	duration = pkt->duration;
+	if ((err = av_interleaved_write_frame(stream->s, pkt)) < 0)
+		return err;
+
+	return duration;
+}