59844ce57e
This reverts commit be142178aaf6ab4089b4d81c88c3d59c12cca567. Reason for revert: breaking internal projects Original change's description: > Use the factory instead of using the builtin code path in `VideoCodecInitializer`. > > Bug: webrtc:9513 > Change-Id: Ia299ae1044a3ff4c91e208200938cba540bdcea6 > Reviewed-on: https://webrtc-review.googlesource.com/c/94782 > Commit-Queue: Jiawei Ou <ouj@fb.com> > Reviewed-by: Kári Helgason <kthelgason@webrtc.org> > Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > Reviewed-by: Anders Carlsson <andersc@webrtc.org> > Reviewed-by: Seth Hampson <shampson@webrtc.org> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Reviewed-by: Niels Moller <nisse@webrtc.org> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#25456} TBR=brandtr@webrtc.org,magjed@webrtc.org,sakal@webrtc.org,nisse@webrtc.org,andersc@webrtc.org,tommi@webrtc.org,kthelgason@webrtc.org,sprang@webrtc.org,srte@webrtc.org,perkj@webrtc.org,tkchin@webrtc.org,shampson@webrtc.org,glaznev@webrtc.org,ouj@fb.com,qingsi@webrtc.org Change-Id: I8040ccabe3ae6464d72c7696adb663c1dd275b63 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:9513 Reviewed-on: https://webrtc-review.googlesource.com/c/108980 Reviewed-by: Qingsi Wang <qingsi@webrtc.org> Commit-Queue: Qingsi Wang <qingsi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25459}
107 lines
3.4 KiB
C++
107 lines
3.4 KiB
C++
/*
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* Copyright 2018 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 TEST_SCENARIO_VIDEO_STREAM_H_
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#define TEST_SCENARIO_VIDEO_STREAM_H_
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#include <memory>
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#include <string>
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#include <vector>
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#include "rtc_base/constructormagic.h"
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#include "test/frame_generator_capturer.h"
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#include "test/scenario/call_client.h"
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#include "test/scenario/column_printer.h"
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#include "test/scenario/network_node.h"
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#include "test/scenario/scenario_config.h"
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#include "test/test_video_capturer.h"
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namespace webrtc {
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namespace test {
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// SendVideoStream provides an interface for changing parameters and retrieving
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// states at run time.
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class SendVideoStream {
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public:
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RTC_DISALLOW_COPY_AND_ASSIGN(SendVideoStream);
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~SendVideoStream();
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void SetCaptureFramerate(int framerate);
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void SetMaxFramerate(absl::optional<int> max_framerate);
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VideoSendStream::Stats GetStats() const;
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ColumnPrinter StatsPrinter();
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void Start();
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private:
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friend class Scenario;
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friend class VideoStreamPair;
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friend class ReceiveVideoStream;
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// Handles RTCP feedback for this stream.
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SendVideoStream(CallClient* sender,
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VideoStreamConfig config,
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Transport* send_transport);
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std::vector<uint32_t> ssrcs_;
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std::vector<uint32_t> rtx_ssrcs_;
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VideoSendStream* send_stream_ = nullptr;
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CallClient* const sender_;
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const VideoStreamConfig config_;
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std::unique_ptr<VideoEncoderFactory> encoder_factory_;
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std::unique_ptr<TestVideoCapturer> video_capturer_;
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FrameGeneratorCapturer* frame_generator_ = nullptr;
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int next_local_network_id_ = 0;
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int next_remote_network_id_ = 0;
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};
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// ReceiveVideoStream represents a video receiver. It can't be used directly.
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class ReceiveVideoStream {
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public:
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RTC_DISALLOW_COPY_AND_ASSIGN(ReceiveVideoStream);
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~ReceiveVideoStream();
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private:
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friend class Scenario;
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friend class VideoStreamPair;
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ReceiveVideoStream(CallClient* receiver,
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VideoStreamConfig config,
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SendVideoStream* send_stream,
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size_t chosen_stream,
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Transport* feedback_transport);
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VideoReceiveStream* receive_stream_ = nullptr;
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FlexfecReceiveStream* flecfec_stream_ = nullptr;
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std::unique_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> renderer_;
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CallClient* const receiver_;
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const VideoStreamConfig config_;
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std::unique_ptr<VideoDecoderFactory> decoder_factory_;
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};
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// VideoStreamPair represents a video streaming session. It can be used to
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// access underlying send and receive classes. It can also be used in calls to
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// the Scenario class.
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class VideoStreamPair {
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public:
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RTC_DISALLOW_COPY_AND_ASSIGN(VideoStreamPair);
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~VideoStreamPair();
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SendVideoStream* send() { return &send_stream_; }
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ReceiveVideoStream* receive() { return &receive_stream_; }
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private:
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friend class Scenario;
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VideoStreamPair(CallClient* sender,
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CallClient* receiver,
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VideoStreamConfig config);
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const VideoStreamConfig config_;
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SendVideoStream send_stream_;
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ReceiveVideoStream receive_stream_;
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};
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} // namespace test
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} // namespace webrtc
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#endif // TEST_SCENARIO_VIDEO_STREAM_H_
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