Replace voe_conference_test.

webrtc/test/call_test.cc

 /*
  *  Copyright (c) 2014 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.
  */
 
@@ skipping 135 matching lines @@
       test->ModifyVideoCaptureStartResolution(&width, &height, &frame_rate);
       CreateFrameGeneratorCapturer(frame_rate, width, height);
       test->OnFrameGeneratorCapturerCreated(frame_generator_capturer_.get());
     }
 
     Start();
   });
 
   test->PerformTest();
 
-  task_queue_.SendTask([this]() {
+  task_queue_.SendTask([this, test]() {
     Stop();
+    test->OnStreamsStopped();
     DestroyStreams();
     send_transport_.reset();
     receive_transport_.reset();
     DestroyCalls();
     if (num_audio_streams_ > 0)
       DestroyVoiceEngines();
   });
-
-  test->OnTestFinished();
 }
 
 void CallTest::CreateCalls(const Call::Config& sender_config,
                            const Call::Config& receiver_config) {
   CreateSenderCall(sender_config);
   CreateReceiverCall(receiver_config);
 }
 
 void CallTest::CreateSenderCall(const Call::Config& config) {
   sender_call_transport_controller_ = new RtpTransportControllerSend(
@@ skipping 39 matching lines @@
     video_send_config_.rtp.extensions.push_back(RtpExtension(
         RtpExtension::kVideoRotationUri, kVideoRotationRtpExtensionId));
   }
 
   if (num_audio_streams > 0) {
     audio_send_config_ = AudioSendStream::Config(send_transport);
     audio_send_config_.voe_channel_id = voe_send_.channel_id;
     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_;
   }
 
   // TODO(brandtr): Update this when we support multistream protection.
   if (num_flexfec_streams > 0) {
     video_send_config_.rtp.flexfec.payload_type = kFlexfecPayloadType;
     video_send_config_.rtp.flexfec.ssrc = kFlexfecSendSsrc;
     video_send_config_.rtp.flexfec.protected_media_ssrcs = {kVideoSendSsrcs[0]};
   }
 }
@@ skipping 346 matching lines @@
 void BaseTest::ModifyFlexfecConfigs(
     std::vector<FlexfecReceiveStream::Config>* receive_configs) {}
 
 void BaseTest::OnFlexfecStreamsCreated(
     const std::vector<FlexfecReceiveStream*>& receive_streams) {}
 
 void BaseTest::OnFrameGeneratorCapturerCreated(
     FrameGeneratorCapturer* frame_generator_capturer) {
 }
 
-void BaseTest::OnTestFinished() {
+void BaseTest::OnStreamsStopped() {
 }
 
 SendTest::SendTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
 }
 
 bool SendTest::ShouldCreateReceivers() const {
   return false;
 }
 
 EndToEndTest::EndToEndTest() {}
 
 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
 }
 
 bool EndToEndTest::ShouldCreateReceivers() const {
   return true;
 }
 
 }  // namespace test
 }  // namespace webrtc