Implement AudioSendStream::GetStats().

talk/media/webrtc/fakewebrtccall.cc

Index: talk/media/webrtc/fakewebrtccall.cc
diff --git a/talk/media/webrtc/fakewebrtccall.cc b/talk/media/webrtc/fakewebrtccall.cc
index 04deeb4c4344d1e4f8b47414ffb96af5462ad8cf..d86bfb553c5c4bb323dcb1195a351655c2401426 100644
--- a/talk/media/webrtc/fakewebrtccall.cc
+++ b/talk/media/webrtc/fakewebrtccall.cc
@@ -39,8 +39,9 @@ FakeAudioSendStream::FakeAudioSendStream(
   RTC_DCHECK(config.voe_channel_id != -1);
 }
 
-webrtc::AudioSendStream::Stats FakeAudioSendStream::GetStats() const {
-  return webrtc::AudioSendStream::Stats();
+void FakeAudioSendStream::SetStats(
+    const webrtc::AudioSendStream::Stats& stats) {
+  stats_ = stats;

[tommi, 2015/10/23 12:50:37]

if you can have a thread checker here I think that would be a good thing even
though this is just a test class.

[the sun, 2015/10/23 15:14:16]

Hmm. Hmm. Hmm. No.

For that to have any good effects the threading assumptions must be kept in
tight sync with the real AudioSendStream. Such implicit assumptions are likely
to fall short.

This class is intended for injection in unit tests for
WebRtcVoiceEngine/WebRtcVoiceMediaChannel and should be as simple as ever
possible, else Gödel comes chasing us. Threading assumptions of the webrtc::Call
classes should be tested separately, and integration tests should verify that
the classes make the same assumptions.

 }
 
 const webrtc::AudioSendStream::Config&
@@ -48,6 +49,10 @@ const webrtc::AudioSendStream::Config&
   return config_;
 }
 
+webrtc::AudioSendStream::Stats FakeAudioSendStream::GetStats() const {
+  return stats_;

[tommi, 2015/10/23 12:50:37]

i.e. same thread as this runs on and any other call that touches stats_

[the sun, 2015/10/23 15:14:16]

Know what you're saying, but see above comment.

+}
+
 FakeAudioReceiveStream::FakeAudioReceiveStream(
     const webrtc::AudioReceiveStream::Config& config)
     : config_(config), received_packets_(0) {
@@ -68,6 +73,10 @@ void FakeAudioReceiveStream::IncrementReceivedPackets() {
   received_packets_++;
 }
 
+webrtc::AudioReceiveStream::Stats FakeAudioReceiveStream::GetStats() const {
+  return stats_;
+}
+
 FakeVideoSendStream::FakeVideoSendStream(
     const webrtc::VideoSendStream::Config& config,
     const webrtc::VideoEncoderConfig& encoder_config)