Implement AudioSendStream::GetStats().

webrtc/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_AUDIO_SEND_STREAM_H_
 #define WEBRTC_AUDIO_SEND_STREAM_H_
 
 #include <string>
 #include <vector>
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/config.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
 #include "webrtc/stream.h"
 #include "webrtc/transport.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 class AudioSendStream : public SendStream {
  public:
-  struct Stats {};
+  struct 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;
+    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;
+    bool typing_noise_detected = false;
+  };
 
   struct Config {
     Config() = delete;
     explicit Config(Transport* send_transport)
         : send_transport(send_transport) {}
 
     std::string ToString() const;
 
     // Receive-stream specific RTP settings.
     struct Rtp {
@@ skipping 22 matching lines @@
     // rtc::scoped_ptr<AudioEncoder> encoder;
     int cng_payload_type = -1;  // pt, or -1 to disable Comfort Noise Generator.
     int red_payload_type = -1;  // pt, or -1 to disable REDundant coding.
   };
 
   virtual Stats GetStats() const = 0;
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_SEND_STREAM_H_