7623ce4aeb
Reason for revert: Bot breakage caused by TickTime::UseFakeClock has been removed. Original issue's description: > Revert of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:20001 of https://codereview.webrtc.org/1506773002/ ) > > Reason for revert: > Breaks Dr Memory Light https://build.chromium.org/p/client.webrtc/builders/Win%20DrMemory%20Light/builds/4643 and all the Android Tests bots. > > Original issue's description: > > Merge webrtc/video_engine/ into webrtc/video/ > > > > BUG=webrtc:1695 > > R=mflodman@webrtc.org > > > > Committed: https://crrev.com/03ef053202bc5d5ab43460eebf5403232f157646 > > Cr-Commit-Position: refs/heads/master@{#10926} > > TBR=mflodman@webrtc.org,pbos@webrtc.org > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:1695 > > Committed: https://crrev.com/8237abf563bf4782ee104408b53cc8e55ce44518 > Cr-Commit-Position: refs/heads/master@{#10937} BUG=webrtc:1695 TBR=mflodman@webrtc.org,kjellander@webrtc.org NOPRESUBMIT=true Review URL: https://codereview.webrtc.org/1510183002 . Cr-Commit-Position: refs/heads/master@{#10948}
96 lines
3.3 KiB
C++
96 lines
3.3 KiB
C++
/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_VIDEO_VIDEO_RECEIVE_STREAM_H_
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#define WEBRTC_VIDEO_VIDEO_RECEIVE_STREAM_H_
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#include <vector>
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#include "webrtc/base/scoped_ptr.h"
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#include "webrtc/call.h"
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#include "webrtc/call/transport_adapter.h"
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#include "webrtc/common_video/include/incoming_video_stream.h"
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#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
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#include "webrtc/modules/video_render/video_render_defines.h"
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#include "webrtc/system_wrappers/include/clock.h"
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#include "webrtc/video/encoded_frame_callback_adapter.h"
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#include "webrtc/video/receive_statistics_proxy.h"
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#include "webrtc/video/vie_channel.h"
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#include "webrtc/video/vie_encoder.h"
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#include "webrtc/video_encoder.h"
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#include "webrtc/video_receive_stream.h"
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namespace webrtc {
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class CallStats;
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class CongestionController;
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class VoiceEngine;
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namespace internal {
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class VideoReceiveStream : public webrtc::VideoReceiveStream,
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public I420FrameCallback,
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public VideoRenderCallback,
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public EncodedImageCallback {
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public:
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VideoReceiveStream(int num_cpu_cores,
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CongestionController* congestion_controller,
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const VideoReceiveStream::Config& config,
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webrtc::VoiceEngine* voice_engine,
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ProcessThread* process_thread,
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CallStats* call_stats);
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~VideoReceiveStream() override;
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// webrtc::ReceiveStream implementation.
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void Start() override;
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void Stop() override;
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void SignalNetworkState(NetworkState state) override;
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bool DeliverRtcp(const uint8_t* packet, size_t length) override;
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bool DeliverRtp(const uint8_t* packet,
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size_t length,
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const PacketTime& packet_time) override;
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// webrtc::VideoReceiveStream implementation.
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webrtc::VideoReceiveStream::Stats GetStats() const override;
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// Overrides I420FrameCallback.
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void FrameCallback(VideoFrame* video_frame) override;
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// Overrides VideoRenderCallback.
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int RenderFrame(const uint32_t /*stream_id*/,
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const VideoFrame& video_frame) override;
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// Overrides EncodedImageCallback.
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int32_t Encoded(const EncodedImage& encoded_image,
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const CodecSpecificInfo* codec_specific_info,
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const RTPFragmentationHeader* fragmentation) override;
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const Config& config() const { return config_; }
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void SetSyncChannel(VoiceEngine* voice_engine, int audio_channel_id);
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private:
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TransportAdapter transport_adapter_;
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EncodedFrameCallbackAdapter encoded_frame_proxy_;
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const VideoReceiveStream::Config config_;
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Clock* const clock_;
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CongestionController* const congestion_controller_;
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CallStats* const call_stats_;
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rtc::scoped_ptr<IncomingVideoStream> incoming_video_stream_;
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rtc::scoped_ptr<ReceiveStatisticsProxy> stats_proxy_;
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rtc::scoped_ptr<ViEChannel> vie_channel_;
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};
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} // namespace internal
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} // namespace webrtc
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#endif // WEBRTC_VIDEO_VIDEO_RECEIVE_STREAM_H_
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