Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(58)

Issues Search
    My Issues | Starred     Open | Closed | All

Unified Diff: webrtc/video/call_perf_tests.cc

Issue 1227923005: Split webrtc/video into webrtc/{audio,call,video}. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 5 years, 3 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
« no previous file with comments | « webrtc/video/call.cc ('k') | webrtc/video/end_to_end_tests.cc » ('j') | no next file with comments »
Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
Index: webrtc/video/call_perf_tests.cc
diff --git a/webrtc/video/call_perf_tests.cc b/webrtc/video/call_perf_tests.cc
deleted file mode 100644
index bbf4caaebd7ff8736955ff29368f9256bdfc1734..0000000000000000000000000000000000000000
--- a/webrtc/video/call_perf_tests.cc
+++ /dev/null
@@ -1,712 +0,0 @@
-/*
- * 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 <sstream>
-#include <string>
-
-#include "testing/gtest/include/gtest/gtest.h"
-
-#include "webrtc/base/checks.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/base/thread_annotations.h"
-#include "webrtc/call.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/rtp_to_ntp.h"
-#include "webrtc/test/call_test.h"
-#include "webrtc/test/direct_transport.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/rtp_rtcp_observer.h"
-#include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/perf_test.h"
-#include "webrtc/video/transport_adapter.h"
-#include "webrtc/voice_engine/include/voe_base.h"
-#include "webrtc/voice_engine/include/voe_codec.h"
-#include "webrtc/voice_engine/include/voe_network.h"
-#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
-#include "webrtc/voice_engine/include/voe_video_sync.h"
-
-namespace webrtc {
-
-class CallPerfTest : public test::CallTest {
- protected:
- void TestAudioVideoSync(bool fec, bool create_audio_first);
-
- 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 SyncRtcpObserver : public test::RtpRtcpObserver {
- public:
- explicit SyncRtcpObserver(const FakeNetworkPipe::Config& config)
- : test::RtpRtcpObserver(CallPerfTest::kLongTimeoutMs, config) {}
-
- Action OnSendRtcp(const uint8_t* packet, size_t length) override {
- RTCPUtility::RTCPParserV2 parser(packet, length, true);
- EXPECT_TRUE(parser.IsValid());
-
- for (RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
- packet_type != RTCPUtility::RTCPPacketTypes::kInvalid;
- packet_type = parser.Iterate()) {
- if (packet_type == RTCPUtility::RTCPPacketTypes::kSr) {
- const RTCPUtility::RTCPPacket& packet = parser.Packet();
- RtcpMeasurement ntp_rtp_pair(
- packet.SR.NTPMostSignificant,
- packet.SR.NTPLeastSignificant,
- packet.SR.RTPTimestamp);
- StoreNtpRtpPair(ntp_rtp_pair);
- }
- }
- return SEND_PACKET;
- }
-
- int64_t RtpTimestampToNtp(uint32_t timestamp) const {
- rtc::CritScope lock(&crit_);
- int64_t timestamp_in_ms = -1;
- if (ntp_rtp_pairs_.size() == 2) {
- // TODO(stefan): We can't EXPECT_TRUE on this call due to a bug in the
- // RTCP sender where it sends RTCP SR before any RTP packets, which leads
- // to a bogus NTP/RTP mapping.
- RtpToNtpMs(timestamp, ntp_rtp_pairs_, &timestamp_in_ms);
- return timestamp_in_ms;
- }
- return -1;
- }
-
- private:
- void StoreNtpRtpPair(RtcpMeasurement ntp_rtp_pair) {
- rtc::CritScope lock(&crit_);
- for (RtcpList::iterator it = ntp_rtp_pairs_.begin();
- it != ntp_rtp_pairs_.end();
- ++it) {
- if (ntp_rtp_pair.ntp_secs == it->ntp_secs &&
- ntp_rtp_pair.ntp_frac == it->ntp_frac) {
- // This RTCP has already been added to the list.
- return;
- }
- }
- // We need two RTCP SR reports to map between RTP and NTP. More than two
- // will not improve the mapping.
- if (ntp_rtp_pairs_.size() == 2) {
- ntp_rtp_pairs_.pop_back();
- }
- ntp_rtp_pairs_.push_front(ntp_rtp_pair);
- }
-
- mutable rtc::CriticalSection crit_;
- RtcpList ntp_rtp_pairs_ GUARDED_BY(crit_);
-};
-
-class VideoRtcpAndSyncObserver : public SyncRtcpObserver, public VideoRenderer {
- static const int kInSyncThresholdMs = 50;
- static const int kStartupTimeMs = 2000;
- static const int kMinRunTimeMs = 30000;
-
- public:
- VideoRtcpAndSyncObserver(Clock* clock,
- int voe_channel,
- VoEVideoSync* voe_sync,
- SyncRtcpObserver* audio_observer)
- : SyncRtcpObserver(FakeNetworkPipe::Config()),
- clock_(clock),
- voe_channel_(voe_channel),
- voe_sync_(voe_sync),
- audio_observer_(audio_observer),
- creation_time_ms_(clock_->TimeInMilliseconds()),
- first_time_in_sync_(-1) {}
-
- void RenderFrame(const VideoFrame& video_frame,
- int time_to_render_ms) override {
- int64_t now_ms = clock_->TimeInMilliseconds();
- uint32_t playout_timestamp = 0;
- if (voe_sync_->GetPlayoutTimestamp(voe_channel_, playout_timestamp) != 0)
- return;
- int64_t latest_audio_ntp =
- audio_observer_->RtpTimestampToNtp(playout_timestamp);
- int64_t latest_video_ntp = RtpTimestampToNtp(video_frame.timestamp());
- if (latest_audio_ntp < 0 || latest_video_ntp < 0)
- return;
- int time_until_render_ms =
- std::max(0, static_cast<int>(video_frame.render_time_ms() - now_ms));
- latest_video_ntp += time_until_render_ms;
- int64_t stream_offset = latest_audio_ntp - latest_video_ntp;
- std::stringstream ss;
- ss << stream_offset;
- 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(latest_audio_ntp - latest_video_ntp) < 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();
- }
- }
-
- bool IsTextureSupported() const override { return false; }
-
- private:
- Clock* const clock_;
- int voe_channel_;
- VoEVideoSync* voe_sync_;
- SyncRtcpObserver* audio_observer_;
- int64_t creation_time_ms_;
- int64_t first_time_in_sync_;
-};
-
-void CallPerfTest::TestAudioVideoSync(bool fec, bool create_audio_first) {
- const char* kSyncGroup = "av_sync";
- class AudioPacketReceiver : public PacketReceiver {
- public:
- AudioPacketReceiver(int channel, VoENetwork* voe_network)
- : channel_(channel),
- voe_network_(voe_network),
- parser_(RtpHeaderParser::Create()) {}
- DeliveryStatus DeliverPacket(MediaType media_type,
- const uint8_t* packet,
- size_t length,
- const PacketTime& packet_time) override {
- EXPECT_TRUE(media_type == MediaType::ANY ||
- media_type == MediaType::AUDIO);
- int ret;
- if (parser_->IsRtcp(packet, length)) {
- ret = voe_network_->ReceivedRTCPPacket(channel_, packet, length);
- } else {
- ret = voe_network_->ReceivedRTPPacket(channel_, packet, length,
- PacketTime());
- }
- return ret == 0 ? DELIVERY_OK : DELIVERY_PACKET_ERROR;
- }
-
- private:
- int channel_;
- VoENetwork* voe_network_;
- rtc::scoped_ptr<RtpHeaderParser> parser_;
- };
-
- VoiceEngine* voice_engine = VoiceEngine::Create();
- VoEBase* voe_base = VoEBase::GetInterface(voice_engine);
- VoECodec* voe_codec = VoECodec::GetInterface(voice_engine);
- VoENetwork* voe_network = VoENetwork::GetInterface(voice_engine);
- VoEVideoSync* voe_sync = VoEVideoSync::GetInterface(voice_engine);
- const std::string audio_filename =
- test::ResourcePath("voice_engine/audio_long16", "pcm");
- ASSERT_STRNE("", audio_filename.c_str());
- test::FakeAudioDevice fake_audio_device(Clock::GetRealTimeClock(),
- audio_filename);
- EXPECT_EQ(0, voe_base->Init(&fake_audio_device, nullptr));
- int channel = voe_base->CreateChannel();
-
- FakeNetworkPipe::Config net_config;
- net_config.queue_delay_ms = 500;
- net_config.loss_percent = 5;
- SyncRtcpObserver audio_observer(net_config);
- VideoRtcpAndSyncObserver observer(Clock::GetRealTimeClock(),
- channel,
- voe_sync,
- &audio_observer);
-
- Call::Config receiver_config;
- receiver_config.voice_engine = voice_engine;
- CreateCalls(Call::Config(), receiver_config);
-
- CodecInst isac = {103, "ISAC", 16000, 480, 1, 32000};
- EXPECT_EQ(0, voe_codec->SetSendCodec(channel, isac));
-
- AudioPacketReceiver voe_packet_receiver(channel, voe_network);
- audio_observer.SetReceivers(&voe_packet_receiver, &voe_packet_receiver);
-
- internal::TransportAdapter transport_adapter(audio_observer.SendTransport());
- transport_adapter.Enable();
- EXPECT_EQ(0,
- voe_network->RegisterExternalTransport(channel, transport_adapter));
-
- observer.SetReceivers(receiver_call_->Receiver(), sender_call_->Receiver());
-
- test::FakeDecoder fake_decoder;
-
- CreateSendConfig(1, observer.SendTransport());
- CreateMatchingReceiveConfigs(observer.ReceiveTransport());
-
- send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
- if (fec) {
- send_config_.rtp.fec.red_payload_type = kRedPayloadType;
- send_config_.rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
- receive_configs_[0].rtp.fec.red_payload_type = kRedPayloadType;
- receive_configs_[0].rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
- }
- receive_configs_[0].rtp.nack.rtp_history_ms = 1000;
- receive_configs_[0].renderer = &observer;
- receive_configs_[0].sync_group = kSyncGroup;
-
- AudioReceiveStream::Config audio_config;
- audio_config.voe_channel_id = channel;
- audio_config.sync_group = kSyncGroup;
-
- AudioReceiveStream* audio_receive_stream = nullptr;
-
- if (create_audio_first) {
- audio_receive_stream =
- receiver_call_->CreateAudioReceiveStream(audio_config);
- CreateStreams();
- } else {
- CreateStreams();
- audio_receive_stream =
- receiver_call_->CreateAudioReceiveStream(audio_config);
- }
-
- CreateFrameGeneratorCapturer();
-
- Start();
-
- fake_audio_device.Start();
- EXPECT_EQ(0, voe_base->StartPlayout(channel));
- EXPECT_EQ(0, voe_base->StartReceive(channel));
- EXPECT_EQ(0, voe_base->StartSend(channel));
-
- EXPECT_EQ(kEventSignaled, observer.Wait())
- << "Timed out while waiting for audio and video to be synchronized.";
-
- EXPECT_EQ(0, voe_base->StopSend(channel));
- EXPECT_EQ(0, voe_base->StopReceive(channel));
- EXPECT_EQ(0, voe_base->StopPlayout(channel));
- fake_audio_device.Stop();
-
- Stop();
- observer.StopSending();
- audio_observer.StopSending();
-
- voe_base->DeleteChannel(channel);
- voe_base->Release();
- voe_codec->Release();
- voe_network->Release();
- voe_sync->Release();
-
- DestroyStreams();
-
- receiver_call_->DestroyAudioReceiveStream(audio_receive_stream);
-
- VoiceEngine::Delete(voice_engine);
-}
-
-TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithAudioCreatedFirst) {
- TestAudioVideoSync(false, true);
-}
-
-TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithVideoCreatedFirst) {
- TestAudioVideoSync(false, false);
-}
-
-TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithFec) {
- TestAudioVideoSync(true, false);
-}
-
-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 VideoRenderer {
- public:
- CaptureNtpTimeObserver(const FakeNetworkPipe::Config& config,
- int threshold_ms,
- int start_time_ms,
- int run_time_ms)
- : EndToEndTest(kLongTimeoutMs, 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:
- void RenderFrame(const VideoFrame& video_frame,
- int time_to_render_ms) override {
- 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_);
- }
-
- bool IsTextureSupported() const override { return false; }
-
- virtual Action OnSendRtp(const uint8_t* packet, size_t length) {
- 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 ModifyConfigs(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_EQ(kEventSignaled, Wait()) << "Timed out while waiting for "
- "estimated capture NTP time to be "
- "within bounds.";
- }
-
- Clock* 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_;
- } 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 ModifyConfigs(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_EQ(kEventSignaled, 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 PacketReceiver {
- public:
- explicit BitrateObserver(bool using_min_transmit_bitrate)
- : EndToEndTest(kLongTimeoutMs),
- send_stream_(nullptr),
- send_transport_receiver_(nullptr),
- pad_to_min_bitrate_(using_min_transmit_bitrate),
- num_bitrate_observations_in_range_(0) {}
-
- private:
- void SetReceivers(PacketReceiver* send_transport_receiver,
- PacketReceiver* receive_transport_receiver) override {
- send_transport_receiver_ = send_transport_receiver;
- test::RtpRtcpObserver::SetReceivers(this, receive_transport_receiver);
- }
-
- DeliveryStatus DeliverPacket(MediaType media_type,
- const uint8_t* packet,
- size_t length,
- const PacketTime& packet_time) 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_transport_receiver_->DeliverPacket(media_type, packet, length,
- packet_time);
- }
-
- void OnStreamsCreated(
- VideoSendStream* send_stream,
- const std::vector<VideoReceiveStream*>& receive_streams) override {
- send_stream_ = send_stream;
- }
-
- void ModifyConfigs(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_EQ(kEventSignaled, Wait())
- << "Timeout while waiting for send-bitrate stats.";
- }
-
- VideoSendStream* send_stream_;
- PacketReceiver* send_transport_receiver_;
- 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_(webrtc::EventWrapper::Create()),
- 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 ModifyConfigs(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 OnStreamsCreated(
- VideoSendStream* send_stream,
- const std::vector<VideoReceiveStream*>& receive_streams) override {
- send_stream_ = send_stream;
- }
-
- void PerformTest() override {
- ASSERT_EQ(kEventSignaled, 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;
- EXPECT_TRUE(send_stream_->ReconfigureVideoEncoder(encoder_config_));
- EXPECT_EQ(kEventSignaled, Wait())
- << "Timed out while waiting for a couple of high bitrate estimates "
- "after reconfiguring the send stream.";
- }
-
- private:
- rtc::scoped_ptr<webrtc::EventWrapper> time_to_reconfigure_;
- int encoder_inits_;
- uint32_t last_set_bitrate_;
- VideoSendStream* send_stream_;
- VideoEncoderConfig encoder_config_;
- } test;
-
- RunBaseTest(&test);
-}
-
-} // namespace webrtc
« no previous file with comments | « webrtc/video/call.cc ('k') | webrtc/video/end_to_end_tests.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698