Create AudioSendStreams in WVoE.

talk/media/webrtc/fakewebrtccall.h

 /*
  * libjingle
  * Copyright 2015 Google Inc.
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  * modification, are permitted provided that the following conditions are met:
  *
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  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
  *     this list of conditions and the following disclaimer in the documentation
  *     and/or other materials provided with the distribution.
  *  3. The name of the author may not be used to endorse or promote products
  *     derived from this software without specific prior written permission.
  *
  * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
  * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
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+// This file contains fake implementations, for use in unit tests, of the
+// following classes:
+//
+//   webrtc::Call
+//   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_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 {
+ public:
+  explicit FakeAudioSendStream(
+      const webrtc::AudioSendStream::Config& config);
+
+  // webrtc::AudioSendStream implementation.
+  webrtc::AudioSendStream::Stats GetStats() const override;
+
+  const webrtc::AudioSendStream::Config& GetConfig() const;
+
+ 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::Config config_;
+};
+
 class FakeAudioReceiveStream : public webrtc::AudioReceiveStream {
  public:
   explicit FakeAudioReceiveStream(
       const webrtc::AudioReceiveStream::Config& config);
 
   // webrtc::AudioReceiveStream implementation.
   webrtc::AudioReceiveStream::Stats GetStats() const override;
 
   const webrtc::AudioReceiveStream::Config& GetConfig() const;
 
@@ skipping 104 matching lines @@
 
 class FakeCall : 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);
   const std::vector<FakeAudioReceiveStream*>& GetAudioReceiveStreams();
   const FakeAudioReceiveStream* GetAudioReceiveStream(uint32_t ssrc);
 
   rtc::SentPacket last_sent_packet() const { return last_sent_packet_; }
   webrtc::NetworkState GetNetworkState() const;
   int GetNumCreatedSendStreams() const;
   int GetNumCreatedReceiveStreams() const;
   void SetStats(const webrtc::Call::Stats& stats);
 
  private:
@@ skipping 27 matching lines @@
   void SetBitrateConfig(
       const webrtc::Call::Config::BitrateConfig& bitrate_config) override;
   void SignalNetworkState(webrtc::NetworkState state) override;
   void OnSentPacket(const rtc::SentPacket& sent_packet) override;
 
   webrtc::Call::Config config_;
   webrtc::NetworkState network_state_;
   rtc::SentPacket last_sent_packet_;
   webrtc::Call::Stats stats_;
   std::vector<FakeVideoSendStream*> video_send_streams_;
+  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_