blob: efc56ad2631caa1b1049ff71cfc75375e2ae43be [file] [log] [blame]
* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
#include <map>
#include <string>
#include <vector>
#include "webrtc/base/event.h"
#include "webrtc/call.h"
#include "webrtc/test/call_test.h"
namespace webrtc {
static const int kTransmissionTimeOffsetExtensionId = 6;
static const int kAbsSendTimeExtensionId = 7;
static const int kTransportSequenceNumberExtensionId = 8;
static const unsigned int kSingleStreamTargetBps = 1000000;
class Clock;
class RampUpTester : public test::EndToEndTest {
RampUpTester(size_t num_video_streams,
size_t num_audio_streams,
unsigned int start_bitrate_bps,
const std::string& extension_type,
bool rtx,
bool red);
~RampUpTester() override;
size_t GetNumVideoStreams() const override;
size_t GetNumAudioStreams() const override;
void PerformTest() override;
virtual bool PollStats();
void AccumulateStats(const VideoSendStream::StreamStats& stream,
size_t* total_packets_sent,
size_t* total_sent,
size_t* padding_sent,
size_t* media_sent) const;
void ReportResult(const std::string& measurement,
size_t value,
const std::string& units) const;
void TriggerTestDone();
rtc::Event event_;
Clock* const clock_;
FakeNetworkPipe::Config forward_transport_config_;
const size_t num_video_streams_;
const size_t num_audio_streams_;
const bool rtx_;
const bool red_;
Call* sender_call_;
VideoSendStream* send_stream_;
test::PacketTransport* send_transport_;
typedef std::map<uint32_t, uint32_t> SsrcMap;
Call::Config GetSenderCallConfig() override;
void OnVideoStreamsCreated(
VideoSendStream* send_stream,
const std::vector<VideoReceiveStream*>& receive_streams) override;
test::PacketTransport* CreateSendTransport(Call* sender_call) override;
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override;
void ModifyAudioConfigs(
AudioSendStream::Config* send_config,
std::vector<AudioReceiveStream::Config>* receive_configs) override;
void OnCallsCreated(Call* sender_call, Call* receiver_call) override;
static bool BitrateStatsPollingThread(void* obj);
const int start_bitrate_bps_;
bool start_bitrate_verified_;
int expected_bitrate_bps_;
int64_t test_start_ms_;
int64_t ramp_up_finished_ms_;
const std::string extension_type_;
std::vector<uint32_t> video_ssrcs_;
std::vector<uint32_t> video_rtx_ssrcs_;
std::vector<uint32_t> audio_ssrcs_;
SsrcMap rtx_ssrc_map_;
rtc::PlatformThread poller_thread_;
class RampUpDownUpTester : public RampUpTester {
RampUpDownUpTester(size_t num_video_streams,
size_t num_audio_streams,
unsigned int start_bitrate_bps,
const std::string& extension_type,
bool rtx,
bool red);
~RampUpDownUpTester() override;
bool PollStats() override;
static const int kHighBandwidthLimitBps = 80000;
static const int kExpectedHighBitrateBps = 60000;
static const int kLowBandwidthLimitBps = 20000;
static const int kExpectedLowBitrateBps = 20000;
enum TestStates { kFirstRampup, kLowRate, kSecondRampup };
Call::Config GetReceiverCallConfig() override;
std::string GetModifierString() const;
void EvolveTestState(int bitrate_bps, bool suspended);
TestStates test_state_;
int64_t state_start_ms_;
int64_t interval_start_ms_;
int sent_bytes_;
} // namespace webrtc