Expose RtpCodecParameters to VoiceMediaInfo stats.

webrtc/api/call/audio_send_stream.h

 /*
  *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #ifndef WEBRTC_API_CALL_AUDIO_SEND_STREAM_H_
 #define WEBRTC_API_CALL_AUDIO_SEND_STREAM_H_
 
 #include <memory>
 #include <string>
 #include <vector>
 
+#include "webrtc/base/optional.h"
 #include "webrtc/config.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
 #include "webrtc/transport.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 // WORK IN PROGRESS
 // This class is under development and is not yet intended for for use outside
 // of WebRtc/Libjingle. Please use the VoiceEngine API instead.
 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690
 
 class AudioSendStream {
  public:
   struct Stats {
     Stats();
+    ~Stats();
 
     // TODO(solenberg): Harmonize naming and defaults with receive stream stats.
     uint32_t local_ssrc = 0;
     int64_t bytes_sent = 0;
     int32_t packets_sent = 0;
     int32_t packets_lost = -1;
     float fraction_lost = -1.0f;
     std::string codec_name;
+    rtc::Optional<int> codec_payload_type;
     int32_t ext_seqnum = -1;
     int32_t jitter_ms = -1;
     int64_t rtt_ms = -1;
     int32_t audio_level = -1;
     float aec_quality_min = -1.0f;
     int32_t echo_delay_median_ms = -1;
     int32_t echo_delay_std_ms = -1;
     int32_t echo_return_loss = -100;
     int32_t echo_return_loss_enhancement = -100;
     float residual_echo_likelihood = -1.0f;
@@ skipping 81 matching lines @@
   virtual void SetMuted(bool muted) = 0;
 
   virtual Stats GetStats() const = 0;
 
  protected:
   virtual ~AudioSendStream() {}
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_API_CALL_AUDIO_SEND_STREAM_H_