800e121dca
This CL makes it possible to change transport routes while running a scenario based test. To make this possible in a consistent manner, the scenario test framework is modified to only allow shared transport for all streams between two CallClients. This is what typically is done in practice and it is quite complex to even reason about the implications of using mixed transports for a single call. Bug: webrtc:9718 Change-Id: Ib836928feed98aa2bbbe0295e158157a6518348b Reviewed-on: https://webrtc-review.googlesource.com/c/107200 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25287}
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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