Expose RtpCodecParameters to VoiceMediaInfo stats.
Payload type -> RtpCodecParameters maps added for sender and receiver.
This is a follow-up to https://codereview.webrtc.org/2484193002/ which
did the same thing for VideoMediaInfo. This information will be used to
produce RTCCodecStats[1].
Voice[Sender/Receiver]Info is updated with current codec payload type
for every stream which can be used to look up the codec in
VoiceMediaInfo.
[1] https://w3c.github.io/webrtc-stats/#codec-dict*
BUG=chromium:659117
Review-Url: https://codereview.webrtc.org/2503383002
Cr-Commit-Position: refs/heads/master@{#15144}
diff --git a/webrtc/api/call/audio_send_stream.cc b/webrtc/api/call/audio_send_stream.cc
index 3ce35e7..b619007 100644
--- a/webrtc/api/call/audio_send_stream.cc
+++ b/webrtc/api/call/audio_send_stream.cc
@@ -30,6 +30,7 @@
namespace webrtc {
AudioSendStream::Stats::Stats() = default;
+AudioSendStream::Stats::~Stats() = default;
AudioSendStream::Config::Config(Transport* send_transport)
: send_transport(send_transport) {}