
R=mflodman@webrtc.org, stefan@webrtc.org BUG=1667,1788 Review URL: https://webrtc-codereview.appspot.com/37489004 Cr-Commit-Position: refs/heads/master@{#8429} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8429 4adac7df-926f-26a2-2b94-8c16560cd09d
159 lines
5.6 KiB
C++
159 lines
5.6 KiB
C++
/*
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* libjingle
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* Copyright 2014 Google Inc.
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions are met:
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*
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* 1. Redistributions of source code must retain the above copyright notice,
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* this list of conditions and the following disclaimer.
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* 2. Redistributions in binary form must reproduce the above copyright notice,
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* this list of conditions and the following disclaimer in the documentation
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* and/or other materials provided with the distribution.
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* 3. The name of the author may not be used to endorse or promote products
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* derived from this software without specific prior written permission.
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*
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* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
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* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
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* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
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* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
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* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
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* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
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* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
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* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
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* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
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* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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*/
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#ifndef TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_UNITTEST_H_
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#define TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_UNITTEST_H_
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#include <map>
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#include <vector>
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#include "webrtc/call.h"
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#include "webrtc/video_receive_stream.h"
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#include "webrtc/video_send_stream.h"
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namespace cricket {
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class FakeVideoSendStream : public webrtc::VideoSendStream,
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public webrtc::VideoSendStreamInput {
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public:
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FakeVideoSendStream(const webrtc::VideoSendStream::Config& config,
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const webrtc::VideoEncoderConfig& encoder_config);
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webrtc::VideoSendStream::Config GetConfig() const;
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webrtc::VideoEncoderConfig GetEncoderConfig() const;
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std::vector<webrtc::VideoStream> GetVideoStreams();
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bool IsSending() const;
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bool GetVp8Settings(webrtc::VideoCodecVP8* settings) const;
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int GetNumberOfSwappedFrames() const;
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int GetLastWidth() const;
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int GetLastHeight() const;
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void SetStats(const webrtc::VideoSendStream::Stats& stats);
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private:
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virtual void SwapFrame(webrtc::I420VideoFrame* frame) OVERRIDE;
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virtual webrtc::VideoSendStream::Stats GetStats() OVERRIDE;
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virtual bool ReconfigureVideoEncoder(
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const webrtc::VideoEncoderConfig& config) OVERRIDE;
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virtual webrtc::VideoSendStreamInput* Input() OVERRIDE;
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virtual void Start() OVERRIDE;
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virtual void Stop() OVERRIDE;
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bool sending_;
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webrtc::VideoSendStream::Config config_;
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webrtc::VideoEncoderConfig encoder_config_;
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bool codec_settings_set_;
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webrtc::VideoCodecVP8 vp8_settings_;
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int num_swapped_frames_;
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webrtc::I420VideoFrame last_frame_;
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webrtc::VideoSendStream::Stats stats_;
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};
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class FakeVideoReceiveStream : public webrtc::VideoReceiveStream {
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public:
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explicit FakeVideoReceiveStream(
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const webrtc::VideoReceiveStream::Config& config);
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webrtc::VideoReceiveStream::Config GetConfig();
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bool IsReceiving() const;
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void InjectFrame(const webrtc::I420VideoFrame& frame, int time_to_render_ms);
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void SetStats(const webrtc::VideoReceiveStream::Stats& stats);
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private:
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virtual webrtc::VideoReceiveStream::Stats GetStats() const OVERRIDE;
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virtual void Start() OVERRIDE;
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virtual void Stop() OVERRIDE;
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webrtc::VideoReceiveStream::Config config_;
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bool receiving_;
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webrtc::VideoReceiveStream::Stats stats_;
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};
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class FakeCall : public webrtc::Call {
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public:
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FakeCall(const webrtc::Call::Config& config);
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~FakeCall();
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void SetVideoCodecs(const std::vector<webrtc::VideoCodec> codecs);
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webrtc::Call::Config GetConfig() const;
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std::vector<FakeVideoSendStream*> GetVideoSendStreams();
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std::vector<FakeVideoReceiveStream*> GetVideoReceiveStreams();
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webrtc::VideoCodec GetEmptyVideoCodec();
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webrtc::VideoCodec GetVideoCodecVp8();
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webrtc::VideoCodec GetVideoCodecVp9();
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std::vector<webrtc::VideoCodec> GetDefaultVideoCodecs();
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webrtc::Call::NetworkState GetNetworkState() const;
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int GetNumCreatedSendStreams() const;
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int GetNumCreatedReceiveStreams() const;
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void SetStats(const webrtc::Call::Stats& stats);
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private:
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virtual webrtc::VideoSendStream* CreateVideoSendStream(
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const webrtc::VideoSendStream::Config& config,
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const webrtc::VideoEncoderConfig& encoder_config) OVERRIDE;
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virtual void DestroyVideoSendStream(
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webrtc::VideoSendStream* send_stream) OVERRIDE;
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virtual webrtc::VideoReceiveStream* CreateVideoReceiveStream(
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const webrtc::VideoReceiveStream::Config& config) OVERRIDE;
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virtual void DestroyVideoReceiveStream(
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webrtc::VideoReceiveStream* receive_stream) OVERRIDE;
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virtual webrtc::PacketReceiver* Receiver() OVERRIDE;
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virtual webrtc::Call::Stats GetStats() const OVERRIDE;
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virtual void SetBitrateConfig(
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const webrtc::Call::Config::BitrateConfig& bitrate_config) OVERRIDE;
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virtual void SignalNetworkState(webrtc::Call::NetworkState state) OVERRIDE;
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webrtc::Call::Config config_;
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webrtc::Call::NetworkState network_state_;
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webrtc::Call::Stats stats_;
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std::vector<webrtc::VideoCodec> codecs_;
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std::vector<FakeVideoSendStream*> video_send_streams_;
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std::vector<FakeVideoReceiveStream*> video_receive_streams_;
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int num_created_send_streams_;
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int num_created_receive_streams_;
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};
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} // namespace cricket
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#endif // TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_UNITTEST_H_
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