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}
130 lines
4.4 KiB
C++
130 lines
4.4 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_CALL_CLIENT_H_
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#define TEST_SCENARIO_CALL_CLIENT_H_
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#include <memory>
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#include <string>
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#include <vector>
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#include "call/call.h"
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#include "logging/rtc_event_log/rtc_event_log.h"
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#include "modules/audio_device/include/test_audio_device.h"
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#include "modules/congestion_controller/test/controller_printer.h"
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#include "modules/rtp_rtcp/include/rtp_header_parser.h"
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#include "rtc_base/constructormagic.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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namespace webrtc {
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namespace test {
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class LoggingNetworkControllerFactory
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: public NetworkControllerFactoryInterface {
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public:
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LoggingNetworkControllerFactory(std::string filename,
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TransportControllerConfig config);
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RTC_DISALLOW_COPY_AND_ASSIGN(LoggingNetworkControllerFactory);
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~LoggingNetworkControllerFactory();
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std::unique_ptr<NetworkControllerInterface> Create(
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NetworkControllerConfig config) override;
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TimeDelta GetProcessInterval() const override;
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// TODO(srte): Consider using the Columnprinter interface for this.
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void LogCongestionControllerStats(Timestamp at_time);
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RtcEventLog* GetEventLog() const;
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private:
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std::unique_ptr<RtcEventLog> event_log_;
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std::unique_ptr<NetworkControllerFactoryInterface> cc_factory_;
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std::unique_ptr<ControlStatePrinter> cc_printer_;
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FILE* cc_out_ = nullptr;
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};
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struct CallClientFakeAudio {
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rtc::scoped_refptr<AudioProcessing> apm;
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rtc::scoped_refptr<TestAudioDeviceModule> fake_audio_device;
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rtc::scoped_refptr<AudioState> audio_state;
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};
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// CallClient represents a participant in a call scenario. It is created by the
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// Scenario class and is used as sender and receiver when setting up a media
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// stream session.
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class CallClient : public NetworkReceiverInterface {
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public:
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CallClient(Clock* clock, std::string log_filename, CallClientConfig config);
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RTC_DISALLOW_COPY_AND_ASSIGN(CallClient);
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~CallClient();
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ColumnPrinter StatsPrinter();
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Call::Stats GetStats();
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DataRate send_bandwidth() {
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return DataRate::bps(GetStats().send_bandwidth_bps);
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}
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bool TryDeliverPacket(rtc::CopyOnWriteBuffer packet,
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uint64_t receiver,
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Timestamp at_time) override;
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private:
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friend class Scenario;
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friend class CallClientPair;
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friend class SendVideoStream;
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friend class VideoStreamPair;
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friend class ReceiveVideoStream;
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friend class SendAudioStream;
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friend class ReceiveAudioStream;
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friend class AudioStreamPair;
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friend class NetworkNodeTransport;
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void DeliverPacket(MediaType media_type,
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rtc::CopyOnWriteBuffer packet,
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Timestamp at_time);
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uint32_t GetNextVideoSsrc();
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uint32_t GetNextAudioSsrc();
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uint32_t GetNextRtxSsrc();
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std::string GetNextPriorityId();
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Clock* clock_;
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LoggingNetworkControllerFactory network_controller_factory_;
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CallClientFakeAudio fake_audio_setup_;
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std::unique_ptr<Call> call_;
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NetworkNodeTransport transport_;
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RtpHeaderParser* const header_parser_;
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std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory_;
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// Stores the configured overhead per known incomming route. This is used to
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// subtract the overhead before processing.
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std::map<uint64_t, DataSize> route_overhead_;
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int next_video_ssrc_index_ = 0;
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int next_rtx_ssrc_index_ = 0;
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int next_audio_ssrc_index_ = 0;
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int next_priority_index_ = 0;
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std::map<uint32_t, MediaType> ssrc_media_types_;
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};
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class CallClientPair {
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public:
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RTC_DISALLOW_COPY_AND_ASSIGN(CallClientPair);
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~CallClientPair();
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CallClient* first() { return first_; }
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CallClient* second() { return second_; }
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std::pair<CallClient*, CallClient*> forward() { return {first(), second()}; }
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std::pair<CallClient*, CallClient*> reverse() { return {second(), first()}; }
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private:
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friend class Scenario;
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CallClientPair(CallClient* first, CallClient* second)
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: first_(first), second_(second) {}
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CallClient* const first_;
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CallClient* const second_;
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};
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} // namespace test
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} // namespace webrtc
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#endif // TEST_SCENARIO_CALL_CLIENT_H_
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