137 lines
4.1 KiB
C++
137 lines
4.1 KiB
C++
/*
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* Copyright (c) 2014 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_CALL_RAMPUP_TESTS_H_
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#define WEBRTC_CALL_RAMPUP_TESTS_H_
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#include <map>
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#include <string>
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#include <vector>
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#include "webrtc/base/event.h"
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#include "webrtc/call.h"
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#include "webrtc/test/call_test.h"
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namespace webrtc {
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static const int kTransmissionTimeOffsetExtensionId = 6;
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static const int kAbsSendTimeExtensionId = 7;
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static const int kTransportSequenceNumberExtensionId = 8;
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static const unsigned int kSingleStreamTargetBps = 1000000;
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class Clock;
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class RampUpTester : public test::EndToEndTest {
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public:
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RampUpTester(size_t num_video_streams,
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size_t num_audio_streams,
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unsigned int start_bitrate_bps,
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const std::string& extension_type,
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bool rtx,
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bool red);
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~RampUpTester() override;
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size_t GetNumVideoStreams() const override;
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size_t GetNumAudioStreams() const override;
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void PerformTest() override;
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protected:
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virtual bool PollStats();
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void AccumulateStats(const VideoSendStream::StreamStats& stream,
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size_t* total_packets_sent,
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size_t* total_sent,
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size_t* padding_sent,
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size_t* media_sent) const;
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void ReportResult(const std::string& measurement,
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size_t value,
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const std::string& units) const;
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void TriggerTestDone();
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rtc::Event event_;
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Clock* const clock_;
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FakeNetworkPipe::Config forward_transport_config_;
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const size_t num_video_streams_;
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const size_t num_audio_streams_;
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const bool rtx_;
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const bool red_;
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VideoSendStream* send_stream_;
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test::PacketTransport* send_transport_;
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private:
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typedef std::map<uint32_t, uint32_t> SsrcMap;
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Call::Config GetSenderCallConfig() override;
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void OnVideoStreamsCreated(
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VideoSendStream* send_stream,
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const std::vector<VideoReceiveStream*>& receive_streams) override;
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test::PacketTransport* CreateSendTransport(Call* sender_call) override;
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void ModifyVideoConfigs(
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VideoSendStream::Config* send_config,
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std::vector<VideoReceiveStream::Config>* receive_configs,
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VideoEncoderConfig* encoder_config) override;
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void ModifyAudioConfigs(
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AudioSendStream::Config* send_config,
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std::vector<AudioReceiveStream::Config>* receive_configs) override;
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void OnCallsCreated(Call* sender_call, Call* receiver_call) override;
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static bool BitrateStatsPollingThread(void* obj);
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const int start_bitrate_bps_;
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bool start_bitrate_verified_;
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int expected_bitrate_bps_;
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int64_t test_start_ms_;
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int64_t ramp_up_finished_ms_;
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const std::string extension_type_;
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std::vector<uint32_t> video_ssrcs_;
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std::vector<uint32_t> video_rtx_ssrcs_;
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std::vector<uint32_t> audio_ssrcs_;
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SsrcMap rtx_ssrc_map_;
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rtc::PlatformThread poller_thread_;
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Call* sender_call_;
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};
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class RampUpDownUpTester : public RampUpTester {
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public:
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RampUpDownUpTester(size_t num_video_streams,
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size_t num_audio_streams,
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unsigned int start_bitrate_bps,
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const std::string& extension_type,
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bool rtx,
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bool red);
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~RampUpDownUpTester() override;
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protected:
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bool PollStats() override;
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private:
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static const int kHighBandwidthLimitBps = 80000;
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static const int kExpectedHighBitrateBps = 60000;
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static const int kLowBandwidthLimitBps = 20000;
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static const int kExpectedLowBitrateBps = 20000;
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enum TestStates { kFirstRampup, kLowRate, kSecondRampup };
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Call::Config GetReceiverCallConfig() override;
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std::string GetModifierString() const;
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void EvolveTestState(int bitrate_bps, bool suspended);
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TestStates test_state_;
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int64_t state_start_ms_;
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int64_t interval_start_ms_;
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int sent_bytes_;
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};
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} // namespace webrtc
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#endif // WEBRTC_CALL_RAMPUP_TESTS_H_
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