Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly i…

webrtc/call/bitrate_estimator_tests.cc

 /*
  *  Copyright (c) 2013 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 <functional>
 #include <list>
 #include <string>
 
 #include "testing/gtest/include/gtest/gtest.h"
 
+#include "webrtc/audio_state.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/call.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/call_test.h"
 #include "webrtc/test/direct_transport.h"
 #include "webrtc/test/encoder_settings.h"
@@ skipping 90 matching lines @@
 
 class BitrateEstimatorTest : public test::CallTest {
  public:
   BitrateEstimatorTest() : receive_config_(nullptr) {}
 
   virtual ~BitrateEstimatorTest() {
     EXPECT_TRUE(streams_.empty());
   }
 
   virtual void SetUp() {
+    AudioState::Config audio_state_config;
+    audio_state_config.voice_engine = &fake_voice_engine_;
     Call::Config config;
-    config.voice_engine = &fake_voice_engine_;
+    config.audio_state = AudioState::Create(audio_state_config);
     receiver_call_.reset(Call::Create(config));
     sender_call_.reset(Call::Create(config));
 
     send_transport_.reset(new test::DirectTransport(sender_call_.get()));
     send_transport_->SetReceiver(receiver_call_->Receiver());
     receive_transport_.reset(new test::DirectTransport(receiver_call_.get()));
     receive_transport_->SetReceiver(sender_call_->Receiver());
 
     send_config_ = VideoSendStream::Config(send_transport_.get());
     send_config_.rtp.ssrcs.push_back(kSendSsrcs[0]);
@@ skipping 20 matching lines @@
 
     send_transport_->StopSending();
     receive_transport_->StopSending();
 
     while (!streams_.empty()) {
       delete streams_.back();
       streams_.pop_back();
     }
 
     receiver_call_.reset();
+    sender_call_.reset();
   }
 
  protected:
   friend class Stream;
 
   class Stream {
    public:
     Stream(BitrateEstimatorTest* test, bool receive_audio)
         : test_(test),
           is_sending_receiving_(false),
@@ skipping 179 matching lines @@
       RtpExtension(RtpExtension::kTOffset, kTOFExtensionId);
   receiver_trace_.PushExpectedLogLine(
       "WrappingBitrateEstimator: Switching to transmission time offset RBE.");
   receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
   streams_.push_back(new Stream(this, false));
   streams_[0]->StopSending();
   streams_[1]->StopSending();
   EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
 }
 }  // namespace webrtc