Use sample builder data for webm audio

Previously it was using the raw rtp payload

Author: tarrencev

README.md

@@ -63,6 +63,7 @@ Check out the **[contributing wiki](https://github.com/pion/webrtc/wiki/Contribu
 * [EricSong](https://github.com/xsephiroth) - *Implement GstV4l2Alsa example*
 * [Tristan Matthews](https://github.com/tmatth)
 * [Alexey Kravtsov](https://github.com/alexey-kravtsov) - *GStreamer encoder tune*
+* [Tarrence van As](https://github.com/tarrencev) - *Webm saver fix*
 
 ### License
 MIT License - see [LICENSE](LICENSE) for full text

save-to-webm/main.go

@@ -68,7 +68,7 @@ func (s *webmSaver) PushOpus(rtpPacket *rtp.Packet) {
 		if s.audioWriter != nil {
 			s.audioTimestamp += sample.Samples
 			t := s.audioTimestamp / 48
-			if _, err := s.audioWriter.Write(true, int64(t), rtpPacket.Payload); err != nil {
+			if _, err := s.audioWriter.Write(true, int64(t), sample.Data); err != nil {
 				panic(err)
 			}
 		}