Implement AudioReceiveStream::GetStats().

webrtc/audio_receive_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_RECEIVE_STREAM_H_
 #define WEBRTC_AUDIO_RECEIVE_STREAM_H_
 
 #include <map>
 #include <string>
 #include <vector>
 
 #include "webrtc/config.h"
 #include "webrtc/stream.h"
 #include "webrtc/transport.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 class AudioDecoder;
 
 class AudioReceiveStream : public ReceiveStream {
  public:
-  struct Stats {};
+  struct Stats {
+    uint32_t remote_ssrc = 0;
+    int64_t bytes_rcvd = 0;
+    uint32_t packets_rcvd = 0;
+    uint32_t packets_lost = 0;
+    float fraction_lost = 0.0f;
+    std::string codec_name;
+    uint32_t ext_seqnum = 0;
+    uint32_t jitter_ms = 0;
+    uint32_t jitter_buffer_ms = 0;
+    uint32_t jitter_buffer_preferred_ms = 0;
+    uint32_t delay_estimate_ms = 0;
+    int32_t audio_level = -1;
+    float expand_rate = 0.0f;
+    float speech_expand_rate = 0.0f;
+    float secondary_decoded_rate = 0.0f;
+    float accelerate_rate = 0.0f;
+    float preemptive_expand_rate = 0.0f;
+    int32_t decoding_calls_to_silence_generator = 0;
+    int32_t decoding_calls_to_neteq = 0;
+    int32_t decoding_normal = 0;
+    int32_t decoding_plc = 0;
+    int32_t decoding_cng = 0;
+    int32_t decoding_plc_cng = 0;
+    int64_t capture_start_ntp_time_ms = 0;
+  };
 
   struct Config {
     std::string ToString() const;
 
     // Receive-stream specific RTP settings.
     struct Rtp {
       std::string ToString() const;
 
       // Synchronization source (stream identifier) to be received.
       uint32_t remote_ssrc = 0;
@@ skipping 27 matching lines @@
 
     // TODO(pbos): Remove config option once combined A/V BWE is always on.
     bool combined_audio_video_bwe = false;
   };
 
   virtual Stats GetStats() const = 0;
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_RECEIVE_STREAM_H_