Expose RtpCodecParameters to VoiceMediaInfo stats.

webrtc/api/call/audio_receive_stream.h

Index: webrtc/api/call/audio_receive_stream.h
diff --git a/webrtc/api/call/audio_receive_stream.h b/webrtc/api/call/audio_receive_stream.h
index ec80a9e3f0cdc9f87121799827fd8ebfeac9c0b9..2fd6760d8641bc71e47108b249705ae299ab4d33 100644
--- a/webrtc/api/call/audio_receive_stream.h
+++ b/webrtc/api/call/audio_receive_stream.h
@@ -16,6 +16,7 @@
 #include <string>
 #include <vector>
 
+#include "webrtc/base/optional.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/modules/audio_coding/codecs/audio_decoder_factory.h"
 #include "webrtc/common_types.h"
@@ -40,6 +41,7 @@ class AudioReceiveStream {
     uint32_t packets_lost = 0;
     float fraction_lost = 0.0f;
     std::string codec_name;
+    rtc::Optional<int> codec_payload_type;
     uint32_t ext_seqnum = 0;
     uint32_t jitter_ms = 0;
     uint32_t jitter_buffer_ms = 0;