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; |