Implement AudioSendStream::GetStats().

webrtc/audio_send_stream.h

Index: webrtc/audio_send_stream.h
diff --git a/webrtc/audio_send_stream.h b/webrtc/audio_send_stream.h
index b96a8ef988d27762fee545d2042e15bd8731c8d3..89b73e6e3ed2a95f9ef79b6f31f0bd3e8b725b0e 100644
--- a/webrtc/audio_send_stream.h
+++ b/webrtc/audio_send_stream.h
@@ -25,7 +25,25 @@ 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;