Move talk/media to webrtc/media

talk/media/webrtc/fakewebrtccall.h

Index: talk/media/webrtc/fakewebrtccall.h
diff --git a/talk/media/webrtc/fakewebrtccall.h b/talk/media/webrtc/fakewebrtccall.h
deleted file mode 100644
index ab1e3b6a5202af4265b17eef48d263c25fd7eaa4..0000000000000000000000000000000000000000
--- a/talk/media/webrtc/fakewebrtccall.h
+++ /dev/null
@@ -1,269 +0,0 @@
-/*
- * 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,
- *     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
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * 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::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 final : public webrtc::AudioSendStream {
- public:
-  struct TelephoneEvent {
-    int payload_type = -1;
-    uint8_t event_code = 0;
-    uint32_t duration_ms = 0;
-  };
-
-  explicit FakeAudioSendStream(const webrtc::AudioSendStream::Config& config);
-
-  const webrtc::AudioSendStream::Config& GetConfig() const;
-  void SetStats(const webrtc::AudioSendStream::Stats& stats);
-  TelephoneEvent GetLatestTelephoneEvent() 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 implementation.
-  bool SendTelephoneEvent(int payload_type, uint8_t event,
-                          uint32_t duration_ms) override;
-  webrtc::AudioSendStream::Stats GetStats() const override;
-
-  TelephoneEvent latest_telephone_event_;
-  webrtc::AudioSendStream::Config config_;
-  webrtc::AudioSendStream::Stats stats_;
-};
-
-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();
-  const webrtc::AudioSinkInterface* sink() const { return sink_.get(); }
-
- private:
-  // webrtc::ReceiveStream 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;
-  }
-  bool DeliverRtp(const uint8_t* packet,
-                  size_t length,
-                  const webrtc::PacketTime& packet_time) override {
-    return true;
-  }
-
-  // webrtc::AudioReceiveStream implementation.
-  webrtc::AudioReceiveStream::Stats GetStats() const override;
-  void SetSink(rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) override;
-
-  webrtc::AudioReceiveStream::Config config_;
-  webrtc::AudioReceiveStream::Stats stats_;
-  int received_packets_;
-  rtc::scoped_ptr<webrtc::AudioSinkInterface> sink_;
-};
-
-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;
-
-  int GetNumberOfSwappedFrames() const;
-  int GetLastWidth() const;
-  int GetLastHeight() const;
-  int64_t GetLastTimestamp() const;
-  void SetStats(const webrtc::VideoSendStream::Stats& stats);
-
- private:
-  void IncomingCapturedFrame(const webrtc::VideoFrame& frame) override;
-
-  // 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::VideoSendStream implementation.
-  webrtc::VideoSendStream::Stats GetStats() override;
-  bool ReconfigureVideoEncoder(
-      const webrtc::VideoEncoderConfig& config) override;
-  webrtc::VideoCaptureInput* Input() override;
-
-  bool sending_;
-  webrtc::VideoSendStream::Config config_;
-  webrtc::VideoEncoderConfig encoder_config_;
-  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 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);
-
-  void SetStats(const webrtc::VideoReceiveStream::Stats& stats);
-
- private:
-  // webrtc::ReceiveStream 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;
-  }
-  bool DeliverRtp(const uint8_t* packet,
-                  size_t length,
-                  const webrtc::PacketTime& packet_time) override {
-    return true;
-  }
-
-  // webrtc::VideoReceiveStream implementation.
-  webrtc::VideoReceiveStream::Stats GetStats() const override;
-
-  webrtc::VideoReceiveStream::Config config_;
-  bool receiving_;
-  webrtc::VideoReceiveStream::Stats stats_;
-};
-
-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);
-  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:
-  webrtc::AudioSendStream* CreateAudioSendStream(
-      const webrtc::AudioSendStream::Config& config) override;
-  void DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) override;
-
-  webrtc::AudioReceiveStream* CreateAudioReceiveStream(
-      const webrtc::AudioReceiveStream::Config& config) override;
-  void DestroyAudioReceiveStream(
-      webrtc::AudioReceiveStream* receive_stream) override;
-
-  webrtc::VideoSendStream* CreateVideoSendStream(
-      const webrtc::VideoSendStream::Config& config,
-      const webrtc::VideoEncoderConfig& encoder_config) override;
-  void DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) override;
-
-  webrtc::VideoReceiveStream* CreateVideoReceiveStream(
-      const webrtc::VideoReceiveStream::Config& config) override;
-  void DestroyVideoReceiveStream(
-      webrtc::VideoReceiveStream* receive_stream) override;
-  webrtc::PacketReceiver* Receiver() override;
-
-  DeliveryStatus DeliverPacket(webrtc::MediaType media_type,
-                               const uint8_t* packet,
-                               size_t length,
-                               const webrtc::PacketTime& packet_time) override;
-
-  webrtc::Call::Stats GetStats() const override;
-
-  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_