Implement AudioReceiveStream::GetStats().

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.
  */
 #include "webrtc/test/call_test.h"
@@ skipping 71 matching lines @@
 }
 
 void CallTest::CreateSenderCall(const Call::Config& config) {
   sender_call_.reset(Call::Create(config));
 }
 
 void CallTest::CreateReceiverCall(const Call::Config& config) {
   receiver_call_.reset(Call::Create(config));
 }
 
+void CallTest::DestroyCalls() {
+  sender_call_.reset(nullptr);
+  receiver_call_.reset(nullptr);
+}
+
 void CallTest::CreateSendConfig(size_t num_streams,
                                 Transport* send_transport) {
   assert(num_streams <= kNumSsrcs);
   send_config_ = VideoSendStream::Config(send_transport);
   send_config_.encoder_settings.encoder = &fake_encoder_;
   send_config_.encoder_settings.payload_name = "FAKE";
   send_config_.encoder_settings.payload_type = kFakeSendPayloadType;
   send_config_.rtp.extensions.push_back(
       RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId));
   encoder_config_.streams = test::CreateVideoStreams(num_streams);
@@ skipping 132 matching lines @@
                            const FakeNetworkPipe::Config& config)
     : BaseTest(timeout_ms, config) {
 }
 
 bool EndToEndTest::ShouldCreateReceivers() const {
   return true;
 }
 
 }  // namespace test
 }  // namespace webrtc