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3d7db263b902f319b5b0c828ccff59cd8664b937
platform-external-webrtc/webrtc/call/call_perf_tests.cc
mflodman 3d7db263b9 Switch voice transport to use Call and Stream instead of VoENetwork. 
VoENetwork is kept for now, but is not really used anylonger.

webrtcvoiceengine is changed to have the same behavior for unsignaled
ssrc as video has, which is reflected by disabling one test case and
this will be discussed and followed up.

BUG=webrtc:5079

TBR=tommi

Review-Url: https://codereview.webrtc.org/1909333002
Cr-Commit-Position: refs/heads/master@{#12555}
2016-04-29 07:57:21 +00:00

703 lines
26 KiB
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/*
* Copyright (c) 2013 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 <algorithm>
#include <limits>
#include <memory>
#include <sstream>
#include <string>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/call.h"
#include "webrtc/call/transport_adapter.h"
#include "webrtc/config.h"
#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/rtp_to_ntp.h"
#include "webrtc/test/call_test.h"
#include "webrtc/test/direct_transport.h"
#include "webrtc/test/drifting_clock.h"
#include "webrtc/test/encoder_settings.h"
#include "webrtc/test/fake_audio_device.h"
#include "webrtc/test/fake_decoder.h"
#include "webrtc/test/fake_encoder.h"
#include "webrtc/test/frame_generator.h"
#include "webrtc/test/frame_generator_capturer.h"
#include "webrtc/test/histogram.h"
#include "webrtc/test/rtp_rtcp_observer.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/testsupport/perf_test.h"
#include "webrtc/voice_engine/include/voe_base.h"
#include "webrtc/voice_engine/include/voe_codec.h"
#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
#include "webrtc/voice_engine/include/voe_video_sync.h"
using webrtc::test::DriftingClock;
using webrtc::test::FakeAudioDevice;
namespace webrtc {
class CallPerfTest : public test::CallTest {
protected:
enum class FecMode {
kOn, kOff
};
enum class CreateOrder {
kAudioFirst, kVideoFirst
};
void TestAudioVideoSync(FecMode fec,
CreateOrder create_first,
float video_ntp_speed,
float video_rtp_speed,
float audio_rtp_speed);
void TestCpuOveruse(LoadObserver::Load tested_load, int encode_delay_ms);
void TestMinTransmitBitrate(bool pad_to_min_bitrate);
void TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config,
int threshold_ms,
int start_time_ms,
int run_time_ms);
};
class VideoRtcpAndSyncObserver : public test::RtpRtcpObserver,
public rtc::VideoSinkInterface<VideoFrame> {
static const int kInSyncThresholdMs = 50;
static const int kStartupTimeMs = 2000;
static const int kMinRunTimeMs = 30000;
public:
explicit VideoRtcpAndSyncObserver(Clock* clock)
: test::RtpRtcpObserver(CallPerfTest::kLongTimeoutMs),
clock_(clock),
creation_time_ms_(clock_->TimeInMilliseconds()),
first_time_in_sync_(-1),
receive_stream_(nullptr) {}
void OnFrame(const VideoFrame& video_frame) override {
VideoReceiveStream::Stats stats;
{
rtc::CritScope lock(&crit_);
if (receive_stream_)
stats = receive_stream_->GetStats();
}
if (stats.sync_offset_ms == std::numeric_limits<int>::max())
return;
int64_t now_ms = clock_->TimeInMilliseconds();
std::stringstream ss;
ss << stats.sync_offset_ms;
webrtc::test::PrintResult("stream_offset",
"",
"synchronization",
ss.str(),
"ms",
false);
int64_t time_since_creation = now_ms - creation_time_ms_;
// During the first couple of seconds audio and video can falsely be
// estimated as being synchronized. We don't want to trigger on those.
if (time_since_creation < kStartupTimeMs)
return;
if (std::abs(stats.sync_offset_ms) < kInSyncThresholdMs) {
if (first_time_in_sync_ == -1) {
first_time_in_sync_ = now_ms;
webrtc::test::PrintResult("sync_convergence_time",
"",
"synchronization",
time_since_creation,
"ms",
false);
}
if (time_since_creation > kMinRunTimeMs)
observation_complete_.Set();
}
}
void set_receive_stream(VideoReceiveStream* receive_stream) {
rtc::CritScope lock(&crit_);
receive_stream_ = receive_stream;
}
private:
Clock* const clock_;
const int64_t creation_time_ms_;
int64_t first_time_in_sync_;
rtc::CriticalSection crit_;
VideoReceiveStream* receive_stream_ GUARDED_BY(crit_);
};
void CallPerfTest::TestAudioVideoSync(FecMode fec,
CreateOrder create_first,
float video_ntp_speed,
float video_rtp_speed,
float audio_rtp_speed) {
const char* kSyncGroup = "av_sync";
const uint32_t kAudioSendSsrc = 1234;
const uint32_t kAudioRecvSsrc = 5678;
test::ClearHistograms();
VoiceEngine* voice_engine = VoiceEngine::Create();
VoEBase* voe_base = VoEBase::GetInterface(voice_engine);
VoECodec* voe_codec = VoECodec::GetInterface(voice_engine);
const std::string audio_filename =
test::ResourcePath("voice_engine/audio_long16", "pcm");
ASSERT_STRNE("", audio_filename.c_str());
FakeAudioDevice fake_audio_device(Clock::GetRealTimeClock(), audio_filename,
audio_rtp_speed);
EXPECT_EQ(0, voe_base->Init(&fake_audio_device, nullptr));
Config voe_config;
voe_config.Set<VoicePacing>(new VoicePacing(true));
int send_channel_id = voe_base->CreateChannel(voe_config);
int recv_channel_id = voe_base->CreateChannel();
AudioState::Config send_audio_state_config;
send_audio_state_config.voice_engine = voice_engine;
Call::Config sender_config;
sender_config.audio_state = AudioState::Create(send_audio_state_config);
Call::Config receiver_config;
receiver_config.audio_state = sender_config.audio_state;
CreateCalls(sender_config, receiver_config);
VideoRtcpAndSyncObserver observer(Clock::GetRealTimeClock());
// Helper class to ensure we deliver correct media_type to the receiving call.
class MediaTypePacketReceiver : public PacketReceiver {
public:
MediaTypePacketReceiver(PacketReceiver* packet_receiver,
MediaType media_type)
: packet_receiver_(packet_receiver), media_type_(media_type) {}
DeliveryStatus DeliverPacket(MediaType media_type,
const uint8_t* packet,
size_t length,
const PacketTime& packet_time) override {
return packet_receiver_->DeliverPacket(media_type_, packet, length,
packet_time);
}
private:
PacketReceiver* packet_receiver_;
const MediaType media_type_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(MediaTypePacketReceiver);
};
FakeNetworkPipe::Config audio_net_config;
audio_net_config.queue_delay_ms = 500;
audio_net_config.loss_percent = 5;
test::PacketTransport audio_send_transport(sender_call_.get(), &observer,
test::PacketTransport::kSender,
audio_net_config);
MediaTypePacketReceiver audio_receiver(receiver_call_->Receiver(),
MediaType::AUDIO);
audio_send_transport.SetReceiver(&audio_receiver);
test::PacketTransport video_send_transport(sender_call_.get(), &observer,
test::PacketTransport::kSender,
FakeNetworkPipe::Config());
MediaTypePacketReceiver video_receiver(receiver_call_->Receiver(),
MediaType::VIDEO);
video_send_transport.SetReceiver(&video_receiver);
test::PacketTransport receive_transport(
receiver_call_.get(), &observer, test::PacketTransport::kReceiver,
FakeNetworkPipe::Config());
receive_transport.SetReceiver(sender_call_->Receiver());
test::FakeDecoder fake_decoder;
CreateSendConfig(1, 0, &video_send_transport);
CreateMatchingReceiveConfigs(&receive_transport);
AudioSendStream::Config audio_send_config(&audio_send_transport);
audio_send_config.voe_channel_id = send_channel_id;
audio_send_config.rtp.ssrc = kAudioSendSsrc;
AudioSendStream* audio_send_stream =
sender_call_->CreateAudioSendStream(audio_send_config);
CodecInst isac = {103, "ISAC", 16000, 480, 1, 32000};
EXPECT_EQ(0, voe_codec->SetSendCodec(send_channel_id, isac));
video_send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
if (fec == FecMode::kOn) {
video_send_config_.rtp.fec.red_payload_type = kRedPayloadType;
video_send_config_.rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
video_receive_configs_[0].rtp.fec.red_payload_type = kRedPayloadType;
video_receive_configs_[0].rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
}
video_receive_configs_[0].rtp.nack.rtp_history_ms = 1000;
video_receive_configs_[0].renderer = &observer;
video_receive_configs_[0].sync_group = kSyncGroup;
AudioReceiveStream::Config audio_recv_config;
audio_recv_config.rtp.remote_ssrc = kAudioSendSsrc;
audio_recv_config.rtp.local_ssrc = kAudioRecvSsrc;
audio_recv_config.voe_channel_id = recv_channel_id;
audio_recv_config.sync_group = kSyncGroup;
AudioReceiveStream* audio_receive_stream;
if (create_first == CreateOrder::kAudioFirst) {
audio_receive_stream =
receiver_call_->CreateAudioReceiveStream(audio_recv_config);
CreateVideoStreams();
} else {
CreateVideoStreams();
audio_receive_stream =
receiver_call_->CreateAudioReceiveStream(audio_recv_config);
}
EXPECT_EQ(1u, video_receive_streams_.size());
observer.set_receive_stream(video_receive_streams_[0]);
DriftingClock drifting_clock(clock_, video_ntp_speed);
CreateFrameGeneratorCapturerWithDrift(&drifting_clock, video_rtp_speed);
Start();
fake_audio_device.Start();
EXPECT_EQ(0, voe_base->StartPlayout(recv_channel_id));
EXPECT_EQ(0, voe_base->StartReceive(recv_channel_id));
EXPECT_EQ(0, voe_base->StartSend(send_channel_id));
EXPECT_TRUE(observer.Wait())
<< "Timed out while waiting for audio and video to be synchronized.";
EXPECT_EQ(0, voe_base->StopSend(send_channel_id));
EXPECT_EQ(0, voe_base->StopReceive(recv_channel_id));
EXPECT_EQ(0, voe_base->StopPlayout(recv_channel_id));
fake_audio_device.Stop();
Stop();
video_send_transport.StopSending();
audio_send_transport.StopSending();
receive_transport.StopSending();
DestroyStreams();
sender_call_->DestroyAudioSendStream(audio_send_stream);
receiver_call_->DestroyAudioReceiveStream(audio_receive_stream);
voe_base->DeleteChannel(send_channel_id);
voe_base->DeleteChannel(recv_channel_id);
voe_base->Release();
voe_codec->Release();
DestroyCalls();
VoiceEngine::Delete(voice_engine);
EXPECT_EQ(1, test::NumHistogramSamples("WebRTC.Video.AVSyncOffsetInMs"));
}
TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithVideoNtpDrift) {
TestAudioVideoSync(FecMode::kOff, CreateOrder::kAudioFirst,
DriftingClock::PercentsFaster(10.0f),
DriftingClock::kNoDrift, DriftingClock::kNoDrift);
}
TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithAudioFasterThanVideoDrift) {
TestAudioVideoSync(FecMode::kOff, CreateOrder::kAudioFirst,
DriftingClock::kNoDrift,
DriftingClock::PercentsSlower(30.0f),
DriftingClock::PercentsFaster(30.0f));
}
TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithVideoFasterThanAudioDrift) {
TestAudioVideoSync(FecMode::kOn, CreateOrder::kVideoFirst,
DriftingClock::kNoDrift,
DriftingClock::PercentsFaster(30.0f),
DriftingClock::PercentsSlower(30.0f));
}
void CallPerfTest::TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config,
int threshold_ms,
int start_time_ms,
int run_time_ms) {
class CaptureNtpTimeObserver : public test::EndToEndTest,
public rtc::VideoSinkInterface<VideoFrame> {
public:
CaptureNtpTimeObserver(const FakeNetworkPipe::Config& net_config,
int threshold_ms,
int start_time_ms,
int run_time_ms)
: EndToEndTest(kLongTimeoutMs),
net_config_(net_config),
clock_(Clock::GetRealTimeClock()),
threshold_ms_(threshold_ms),
start_time_ms_(start_time_ms),
run_time_ms_(run_time_ms),
creation_time_ms_(clock_->TimeInMilliseconds()),
capturer_(nullptr),
rtp_start_timestamp_set_(false),
rtp_start_timestamp_(0) {}
private:
test::PacketTransport* CreateSendTransport(Call* sender_call) override {
return new test::PacketTransport(
sender_call, this, test::PacketTransport::kSender, net_config_);
}
test::PacketTransport* CreateReceiveTransport() override {
return new test::PacketTransport(
nullptr, this, test::PacketTransport::kReceiver, net_config_);
}
void OnFrame(const VideoFrame& video_frame) override {
rtc::CritScope lock(&crit_);
if (video_frame.ntp_time_ms() <= 0) {
// Haven't got enough RTCP SR in order to calculate the capture ntp
// time.
return;
}
int64_t now_ms = clock_->TimeInMilliseconds();
int64_t time_since_creation = now_ms - creation_time_ms_;
if (time_since_creation < start_time_ms_) {
// Wait for |start_time_ms_| before start measuring.
return;
}
if (time_since_creation > run_time_ms_) {
observation_complete_.Set();
}
FrameCaptureTimeList::iterator iter =
capture_time_list_.find(video_frame.timestamp());
EXPECT_TRUE(iter != capture_time_list_.end());
// The real capture time has been wrapped to uint32_t before converted
// to rtp timestamp in the sender side. So here we convert the estimated
// capture time to a uint32_t 90k timestamp also for comparing.
uint32_t estimated_capture_timestamp =
90 * static_cast<uint32_t>(video_frame.ntp_time_ms());
uint32_t real_capture_timestamp = iter->second;
int time_offset_ms = real_capture_timestamp - estimated_capture_timestamp;
time_offset_ms = time_offset_ms / 90;
std::stringstream ss;
ss << time_offset_ms;
webrtc::test::PrintResult(
"capture_ntp_time", "", "real - estimated", ss.str(), "ms", true);
EXPECT_TRUE(std::abs(time_offset_ms) < threshold_ms_);
}
virtual Action OnSendRtp(const uint8_t* packet, size_t length) {
rtc::CritScope lock(&crit_);
RTPHeader header;
EXPECT_TRUE(parser_->Parse(packet, length, &header));
if (!rtp_start_timestamp_set_) {
// Calculate the rtp timestamp offset in order to calculate the real
// capture time.
uint32_t first_capture_timestamp =
90 * static_cast<uint32_t>(capturer_->first_frame_capture_time());
rtp_start_timestamp_ = header.timestamp - first_capture_timestamp;
rtp_start_timestamp_set_ = true;
}
uint32_t capture_timestamp = header.timestamp - rtp_start_timestamp_;
capture_time_list_.insert(
capture_time_list_.end(),
std::make_pair(header.timestamp, capture_timestamp));
return SEND_PACKET;
}
void OnFrameGeneratorCapturerCreated(
test::FrameGeneratorCapturer* frame_generator_capturer) override {
capturer_ = frame_generator_capturer;
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
(*receive_configs)[0].renderer = this;
// Enable the receiver side rtt calculation.
(*receive_configs)[0].rtp.rtcp_xr.receiver_reference_time_report = true;
}
void PerformTest() override {
EXPECT_TRUE(Wait()) << "Timed out while waiting for "
"estimated capture NTP time to be "
"within bounds.";
}
rtc::CriticalSection crit_;
const FakeNetworkPipe::Config net_config_;
Clock* const clock_;
int threshold_ms_;
int start_time_ms_;
int run_time_ms_;
int64_t creation_time_ms_;
test::FrameGeneratorCapturer* capturer_;
bool rtp_start_timestamp_set_;
uint32_t rtp_start_timestamp_;
typedef std::map<uint32_t, uint32_t> FrameCaptureTimeList;
FrameCaptureTimeList capture_time_list_ GUARDED_BY(&crit_);
} test(net_config, threshold_ms, start_time_ms, run_time_ms);
RunBaseTest(&test);
}
TEST_F(CallPerfTest, CaptureNtpTimeWithNetworkDelay) {
FakeNetworkPipe::Config net_config;
net_config.queue_delay_ms = 100;
// TODO(wu): lower the threshold as the calculation/estimatation becomes more
// accurate.
const int kThresholdMs = 100;
const int kStartTimeMs = 10000;
const int kRunTimeMs = 20000;
TestCaptureNtpTime(net_config, kThresholdMs, kStartTimeMs, kRunTimeMs);
}
TEST_F(CallPerfTest, CaptureNtpTimeWithNetworkJitter) {
FakeNetworkPipe::Config net_config;
net_config.queue_delay_ms = 100;
net_config.delay_standard_deviation_ms = 10;
// TODO(wu): lower the threshold as the calculation/estimatation becomes more
// accurate.
const int kThresholdMs = 100;
const int kStartTimeMs = 10000;
const int kRunTimeMs = 20000;
TestCaptureNtpTime(net_config, kThresholdMs, kStartTimeMs, kRunTimeMs);
}
void CallPerfTest::TestCpuOveruse(LoadObserver::Load tested_load,
int encode_delay_ms) {
class LoadObserver : public test::SendTest, public webrtc::LoadObserver {
public:
LoadObserver(LoadObserver::Load tested_load, int encode_delay_ms)
: SendTest(kLongTimeoutMs),
tested_load_(tested_load),
encoder_(Clock::GetRealTimeClock(), encode_delay_ms) {}
void OnLoadUpdate(Load load) override {
if (load == tested_load_)
observation_complete_.Set();
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
send_config->overuse_callback = this;
send_config->encoder_settings.encoder = &encoder_;
}
void PerformTest() override {
EXPECT_TRUE(Wait()) << "Timed out before receiving an overuse callback.";
}
LoadObserver::Load tested_load_;
test::DelayedEncoder encoder_;
} test(tested_load, encode_delay_ms);
RunBaseTest(&test);
}
TEST_F(CallPerfTest, ReceivesCpuUnderuse) {
const int kEncodeDelayMs = 2;
TestCpuOveruse(LoadObserver::kUnderuse, kEncodeDelayMs);
}
TEST_F(CallPerfTest, ReceivesCpuOveruse) {
const int kEncodeDelayMs = 35;
TestCpuOveruse(LoadObserver::kOveruse, kEncodeDelayMs);
}
void CallPerfTest::TestMinTransmitBitrate(bool pad_to_min_bitrate) {
static const int kMaxEncodeBitrateKbps = 30;
static const int kMinTransmitBitrateBps = 150000;
static const int kMinAcceptableTransmitBitrate = 130;
static const int kMaxAcceptableTransmitBitrate = 170;
static const int kNumBitrateObservationsInRange = 100;
static const int kAcceptableBitrateErrorMargin = 15; // +- 7
class BitrateObserver : public test::EndToEndTest {
public:
explicit BitrateObserver(bool using_min_transmit_bitrate)
: EndToEndTest(kLongTimeoutMs),
send_stream_(nullptr),
pad_to_min_bitrate_(using_min_transmit_bitrate),
num_bitrate_observations_in_range_(0) {}
private:
// TODO(holmer): Run this with a timer instead of once per packet.
Action OnSendRtp(const uint8_t* packet, size_t length) override {
VideoSendStream::Stats stats = send_stream_->GetStats();
if (stats.substreams.size() > 0) {
RTC_DCHECK_EQ(1u, stats.substreams.size());
int bitrate_kbps =
stats.substreams.begin()->second.total_bitrate_bps / 1000;
if (bitrate_kbps > 0) {
test::PrintResult(
"bitrate_stats_",
(pad_to_min_bitrate_ ? "min_transmit_bitrate"
: "without_min_transmit_bitrate"),
"bitrate_kbps",
static_cast<size_t>(bitrate_kbps),
"kbps",
false);
if (pad_to_min_bitrate_) {
if (bitrate_kbps > kMinAcceptableTransmitBitrate &&
bitrate_kbps < kMaxAcceptableTransmitBitrate) {
++num_bitrate_observations_in_range_;
}
} else {
// Expect bitrate stats to roughly match the max encode bitrate.
if (bitrate_kbps > (kMaxEncodeBitrateKbps -
kAcceptableBitrateErrorMargin / 2) &&
bitrate_kbps < (kMaxEncodeBitrateKbps +
kAcceptableBitrateErrorMargin / 2)) {
++num_bitrate_observations_in_range_;
}
}
if (num_bitrate_observations_in_range_ ==
kNumBitrateObservationsInRange)
observation_complete_.Set();
}
}
return SEND_PACKET;
}
void OnVideoStreamsCreated(
VideoSendStream* send_stream,
const std::vector<VideoReceiveStream*>& receive_streams) override {
send_stream_ = send_stream;
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
if (pad_to_min_bitrate_) {
encoder_config->min_transmit_bitrate_bps = kMinTransmitBitrateBps;
} else {
RTC_DCHECK_EQ(0, encoder_config->min_transmit_bitrate_bps);
}
}
void PerformTest() override {
EXPECT_TRUE(Wait()) << "Timeout while waiting for send-bitrate stats.";
}
VideoSendStream* send_stream_;
const bool pad_to_min_bitrate_;
int num_bitrate_observations_in_range_;
} test(pad_to_min_bitrate);
fake_encoder_.SetMaxBitrate(kMaxEncodeBitrateKbps);
RunBaseTest(&test);
}
TEST_F(CallPerfTest, PadsToMinTransmitBitrate) { TestMinTransmitBitrate(true); }
TEST_F(CallPerfTest, NoPadWithoutMinTransmitBitrate) {
TestMinTransmitBitrate(false);
}
TEST_F(CallPerfTest, KeepsHighBitrateWhenReconfiguringSender) {
static const uint32_t kInitialBitrateKbps = 400;
static const uint32_t kReconfigureThresholdKbps = 600;
static const uint32_t kPermittedReconfiguredBitrateDiffKbps = 100;
class BitrateObserver : public test::EndToEndTest, public test::FakeEncoder {
public:
BitrateObserver()
: EndToEndTest(kDefaultTimeoutMs),
FakeEncoder(Clock::GetRealTimeClock()),
time_to_reconfigure_(false, false),
encoder_inits_(0),
last_set_bitrate_(0),
send_stream_(nullptr) {}
int32_t InitEncode(const VideoCodec* config,
int32_t number_of_cores,
size_t max_payload_size) override {
if (encoder_inits_ == 0) {
EXPECT_EQ(kInitialBitrateKbps, config->startBitrate)
<< "Encoder not initialized at expected bitrate.";
}
++encoder_inits_;
if (encoder_inits_ == 2) {
EXPECT_GE(last_set_bitrate_, kReconfigureThresholdKbps);
EXPECT_NEAR(config->startBitrate,
last_set_bitrate_,
kPermittedReconfiguredBitrateDiffKbps)
<< "Encoder reconfigured with bitrate too far away from last set.";
observation_complete_.Set();
}
return FakeEncoder::InitEncode(config, number_of_cores, max_payload_size);
}
int32_t SetRates(uint32_t new_target_bitrate_kbps,
uint32_t framerate) override {
last_set_bitrate_ = new_target_bitrate_kbps;
if (encoder_inits_ == 1 &&
new_target_bitrate_kbps > kReconfigureThresholdKbps) {
time_to_reconfigure_.Set();
}
return FakeEncoder::SetRates(new_target_bitrate_kbps, framerate);
}
Call::Config GetSenderCallConfig() override {
Call::Config config = EndToEndTest::GetSenderCallConfig();
config.bitrate_config.start_bitrate_bps = kInitialBitrateKbps * 1000;
return config;
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
send_config->encoder_settings.encoder = this;
encoder_config->streams[0].min_bitrate_bps = 50000;
encoder_config->streams[0].target_bitrate_bps =
encoder_config->streams[0].max_bitrate_bps = 2000000;
encoder_config_ = *encoder_config;
}
void OnVideoStreamsCreated(
VideoSendStream* send_stream,
const std::vector<VideoReceiveStream*>& receive_streams) override {
send_stream_ = send_stream;
}
void PerformTest() override {
ASSERT_TRUE(time_to_reconfigure_.Wait(kDefaultTimeoutMs))
<< "Timed out before receiving an initial high bitrate.";
encoder_config_.streams[0].width *= 2;
encoder_config_.streams[0].height *= 2;
send_stream_->ReconfigureVideoEncoder(encoder_config_);
EXPECT_TRUE(Wait())
<< "Timed out while waiting for a couple of high bitrate estimates "
"after reconfiguring the send stream.";
}
private:
rtc::Event time_to_reconfigure_;
int encoder_inits_;
uint32_t last_set_bitrate_;
VideoSendStream* send_stream_;
VideoEncoderConfig encoder_config_;
} test;
RunBaseTest(&test);
}
} // namespace webrtc
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