commit 73276ad7ed1e70ab764cd02d7189ed5839fadc20
author Fredrik Solenberg <solenberg@webrtc.org> Thu Sep 14 14:46:47 2017 +0200
committer Fredrik Solenberg <solenberg@webrtc.org> Thu Sep 14 12:46:50 2017 +0000
tree 2a2e9edab7a943bea5d6189befacf1d66d8aadff
parent 7d1f493a8b042b9ccf6993bb078c83bb0d992094

- Removes voe_conference_test.
- Adds a new AudioStatsTest, with better coverage of the same features, based on call_test.
- Adds an AudioEndToEndTest utility, which AudioStatsTest and LowBandwidthAudioTest uses.

BUG=webrtc:4690
R=kwiberg@webrtc.org

Review-Url: https://codereview.webrtc.org/3008273002 .
Cr-Commit-Position: refs/heads/master@{#19833}

webrtc/test/call_test.cc

diff --git a/webrtc/test/call_test.cc b/webrtc/test/call_test.cc
index b5d7236..d4084d5 100644
--- a/webrtc/test/call_test.cc
+++ b/webrtc/test/call_test.cc
@@ -153,8 +153,9 @@
 
   test->PerformTest();
 
-  task_queue_.SendTask([this]() {
+  task_queue_.SendTask([this, test]() {
     Stop();
+    test->OnStreamsStopped();
     DestroyStreams();
     send_transport_.reset();
     receive_transport_.reset();
@@ -162,8 +163,6 @@
     if (num_audio_streams_ > 0)
       DestroyVoiceEngines();
   });
-
-  test->OnTestFinished();
 }
 
 void CallTest::CreateCalls(const Call::Config& sender_config,
@@ -223,7 +222,7 @@
     audio_send_config_.rtp.ssrc = kAudioSendSsrc;
     audio_send_config_.send_codec_spec =
         rtc::Optional<AudioSendStream::Config::SendCodecSpec>(
-            {kAudioSendPayloadType, {"OPUS", 48000, 2, {{"stereo", "1"}}}});
+            {kAudioSendPayloadType, {"opus", 48000, 2, {{"stereo", "1"}}}});
     audio_send_config_.encoder_factory = encoder_factory_;
   }
 
@@ -590,7 +589,7 @@
     FrameGeneratorCapturer* frame_generator_capturer) {
 }
 
-void BaseTest::OnTestFinished() {
+void BaseTest::OnStreamsStopped() {
 }
 
 SendTest::SendTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {