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

talk/media/webrtc/fakewebrtccall.h

 /*
  * libjingle
  * Copyright 2015 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 11 matching lines @@
  * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
  * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
  * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
  * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
 // This file contains fake implementations, for use in unit tests, of the
 // following classes:
 //
 //   webrtc::Call
+//   webrtc::AudioState
 //   webrtc::AudioSendStream
 //   webrtc::AudioReceiveStream
 //   webrtc::VideoSendStream
 //   webrtc::VideoReceiveStream
 
 #ifndef TALK_MEDIA_WEBRTC_FAKEWEBRTCCALL_H_
 #define TALK_MEDIA_WEBRTC_FAKEWEBRTCCALL_H_
 
 #include <vector>
 
 #include "webrtc/call.h"
+#include "webrtc/audio_state.h"
 #include "webrtc/audio_receive_stream.h"
 #include "webrtc/audio_send_stream.h"
 #include "webrtc/video_frame.h"
 #include "webrtc/video_receive_stream.h"
 #include "webrtc/video_send_stream.h"
 
 namespace cricket {
 
-class FakeAudioSendStream : public webrtc::AudioSendStream {
+class FakeAudioState final : public webrtc::AudioState {
+ public:
+  FakeAudioState() {}
+  ~FakeAudioState() override {}
+};
+
+class FakeAudioSendStream final : public webrtc::AudioSendStream {
  public:
   explicit FakeAudioSendStream(
       const webrtc::AudioSendStream::Config& config);
 
   const webrtc::AudioSendStream::Config& GetConfig() const;
   void SetStats(const webrtc::AudioSendStream::Stats& stats);
 
  private:
   // webrtc::SendStream implementation.
   void Start() override {}
   void Stop() override {}
   void SignalNetworkState(webrtc::NetworkState state) override {}
   bool DeliverRtcp(const uint8_t* packet, size_t length) override {
     return true;
   }
 
   // webrtc::AudioSendStream implementation.
   webrtc::AudioSendStream::Stats GetStats() const override;
 
   webrtc::AudioSendStream::Config config_;
   webrtc::AudioSendStream::Stats stats_;
 };
 
-class FakeAudioReceiveStream : public webrtc::AudioReceiveStream {
+class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream {
  public:
   explicit FakeAudioReceiveStream(
       const webrtc::AudioReceiveStream::Config& config);
 
   const webrtc::AudioReceiveStream::Config& GetConfig() const;
   void SetStats(const webrtc::AudioReceiveStream::Stats& stats);
   int received_packets() const { return received_packets_; }
   void IncrementReceivedPackets();
 
  private:
@@ skipping 11 matching lines @@
   }
 
   // webrtc::AudioReceiveStream implementation.
   webrtc::AudioReceiveStream::Stats GetStats() const override;
 
   webrtc::AudioReceiveStream::Config config_;
   webrtc::AudioReceiveStream::Stats stats_;
   int received_packets_;
 };
 
-class FakeVideoSendStream : public webrtc::VideoSendStream,
-                            public webrtc::VideoCaptureInput {
+class FakeVideoSendStream final : public webrtc::VideoSendStream,
+                                  public webrtc::VideoCaptureInput {
  public:
   FakeVideoSendStream(const webrtc::VideoSendStream::Config& config,
                       const webrtc::VideoEncoderConfig& encoder_config);
   webrtc::VideoSendStream::Config GetConfig() const;
   webrtc::VideoEncoderConfig GetEncoderConfig() const;
   std::vector<webrtc::VideoStream> GetVideoStreams();
 
   bool IsSending() const;
   bool GetVp8Settings(webrtc::VideoCodecVP8* settings) const;
   bool GetVp9Settings(webrtc::VideoCodecVP9* settings) const;
@@ skipping 27 matching lines @@
   bool codec_settings_set_;
   union VpxSettings {
     webrtc::VideoCodecVP8 vp8;
     webrtc::VideoCodecVP9 vp9;
   } vpx_settings_;
   int num_swapped_frames_;
   webrtc::VideoFrame last_frame_;
   webrtc::VideoSendStream::Stats stats_;
 };
 
-class FakeVideoReceiveStream : public webrtc::VideoReceiveStream {
+class FakeVideoReceiveStream final : public webrtc::VideoReceiveStream {
  public:
   explicit FakeVideoReceiveStream(
       const webrtc::VideoReceiveStream::Config& config);
 
   webrtc::VideoReceiveStream::Config GetConfig();
 
   bool IsReceiving() const;
 
   void InjectFrame(const webrtc::VideoFrame& frame, int time_to_render_ms);
 
@@ skipping 14 matching lines @@
   }
 
   // webrtc::VideoReceiveStream implementation.
   webrtc::VideoReceiveStream::Stats GetStats() const override;
 
   webrtc::VideoReceiveStream::Config config_;
   bool receiving_;
   webrtc::VideoReceiveStream::Stats stats_;
 };
 
-class FakeCall : public webrtc::Call, public webrtc::PacketReceiver {
+class FakeCall final : public webrtc::Call, public webrtc::PacketReceiver {
  public:
   explicit FakeCall(const webrtc::Call::Config& config);
   ~FakeCall() override;
 
   webrtc::Call::Config GetConfig() const;
   const std::vector<FakeVideoSendStream*>& GetVideoSendStreams();
   const std::vector<FakeVideoReceiveStream*>& GetVideoReceiveStreams();
 
   const std::vector<FakeAudioSendStream*>& GetAudioSendStreams();
   const FakeAudioSendStream* GetAudioSendStream(uint32_t ssrc);
@@ skipping 47 matching lines @@
   std::vector<FakeAudioSendStream*> audio_send_streams_;
   std::vector<FakeVideoReceiveStream*> video_receive_streams_;
   std::vector<FakeAudioReceiveStream*> audio_receive_streams_;
 
   int num_created_send_streams_;
   int num_created_receive_streams_;
 };
 
 }  // namespace cricket
 #endif  // TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_UNITTEST_H_