pbos@webrtc.org | f94ccd4 | 2013-12-13 12:48:05 +0000 | [diff] [blame] | 1 | /* |
| 2 | * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. |
| 3 | * |
| 4 | * Use of this source code is governed by a BSD-style license |
| 5 | * that can be found in the LICENSE file in the root of the source |
| 6 | * tree. An additional intellectual property rights grant can be found |
| 7 | * in the file PATENTS. All contributing project authors may |
| 8 | * be found in the AUTHORS file in the root of the source tree. |
| 9 | */ |
| 10 | #include <assert.h> |
| 11 | |
| 12 | #include <algorithm> |
| 13 | #include <sstream> |
| 14 | #include <string> |
| 15 | |
| 16 | #include "testing/gtest/include/gtest/gtest.h" |
| 17 | |
| 18 | #include "webrtc/call.h" |
henrik.lundin@webrtc.org | 70e2ce9 | 2014-03-06 10:28:07 +0000 | [diff] [blame] | 19 | #include "webrtc/common.h" |
| 20 | #include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h" |
pbos@webrtc.org | f94ccd4 | 2013-12-13 12:48:05 +0000 | [diff] [blame] | 21 | #include "webrtc/modules/remote_bitrate_estimator/include/rtp_to_ntp.h" |
| 22 | #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" |
| 23 | #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" |
| 24 | #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" |
| 25 | #include "webrtc/system_wrappers/interface/scoped_ptr.h" |
| 26 | #include "webrtc/test/direct_transport.h" |
pbos@webrtc.org | e2a7a77 | 2014-03-19 08:43:57 +0000 | [diff] [blame] | 27 | #include "webrtc/test/encoder_settings.h" |
pbos@webrtc.org | f94ccd4 | 2013-12-13 12:48:05 +0000 | [diff] [blame] | 28 | #include "webrtc/test/fake_audio_device.h" |
| 29 | #include "webrtc/test/fake_decoder.h" |
| 30 | #include "webrtc/test/fake_encoder.h" |
| 31 | #include "webrtc/test/frame_generator.h" |
| 32 | #include "webrtc/test/frame_generator_capturer.h" |
| 33 | #include "webrtc/test/rtp_rtcp_observer.h" |
| 34 | #include "webrtc/test/testsupport/fileutils.h" |
| 35 | #include "webrtc/test/testsupport/perf_test.h" |
| 36 | #include "webrtc/video/transport_adapter.h" |
| 37 | #include "webrtc/voice_engine/include/voe_base.h" |
| 38 | #include "webrtc/voice_engine/include/voe_codec.h" |
| 39 | #include "webrtc/voice_engine/include/voe_network.h" |
| 40 | #include "webrtc/voice_engine/include/voe_rtp_rtcp.h" |
| 41 | #include "webrtc/voice_engine/include/voe_video_sync.h" |
| 42 | |
| 43 | namespace webrtc { |
| 44 | |
| 45 | static unsigned int kLongTimeoutMs = 120 * 1000; |
| 46 | static const uint32_t kSendSsrc = 0x654321; |
| 47 | static const uint32_t kReceiverLocalSsrc = 0x123456; |
| 48 | static const uint8_t kSendPayloadType = 125; |
| 49 | |
| 50 | class CallPerfTest : public ::testing::Test { |
asapersson@webrtc.org | 8ef6548 | 2014-01-31 10:05:07 +0000 | [diff] [blame] | 51 | public: |
| 52 | CallPerfTest() |
| 53 | : send_stream_(NULL), fake_encoder_(Clock::GetRealTimeClock()) {} |
pbos@webrtc.org | 3f83f9c | 2014-03-13 12:52:27 +0000 | [diff] [blame] | 54 | |
asapersson@webrtc.org | 8ef6548 | 2014-01-31 10:05:07 +0000 | [diff] [blame] | 55 | protected: |
| 56 | VideoSendStream::Config GetSendTestConfig(Call* call) { |
| 57 | VideoSendStream::Config config = call->GetDefaultSendConfig(); |
asapersson@webrtc.org | 8ef6548 | 2014-01-31 10:05:07 +0000 | [diff] [blame] | 58 | config.rtp.ssrcs.push_back(kSendSsrc); |
pbos@webrtc.org | e2a7a77 | 2014-03-19 08:43:57 +0000 | [diff] [blame] | 59 | config.encoder_settings = test::CreateEncoderSettings( |
| 60 | &fake_encoder_, "FAKE", kSendPayloadType, 1); |
asapersson@webrtc.org | 8ef6548 | 2014-01-31 10:05:07 +0000 | [diff] [blame] | 61 | return config; |
| 62 | } |
pbos@webrtc.org | 3f83f9c | 2014-03-13 12:52:27 +0000 | [diff] [blame] | 63 | |
asapersson@webrtc.org | 8ef6548 | 2014-01-31 10:05:07 +0000 | [diff] [blame] | 64 | void RunVideoSendTest(Call* call, |
| 65 | const VideoSendStream::Config& config, |
| 66 | test::RtpRtcpObserver* observer) { |
| 67 | send_stream_ = call->CreateVideoSendStream(config); |
| 68 | scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer( |
| 69 | test::FrameGeneratorCapturer::Create( |
| 70 | send_stream_->Input(), 320, 240, 30, Clock::GetRealTimeClock())); |
| 71 | send_stream_->StartSending(); |
| 72 | frame_generator_capturer->Start(); |
| 73 | |
| 74 | EXPECT_EQ(kEventSignaled, observer->Wait()); |
| 75 | |
| 76 | observer->StopSending(); |
| 77 | frame_generator_capturer->Stop(); |
| 78 | send_stream_->StopSending(); |
| 79 | call->DestroyVideoSendStream(send_stream_); |
| 80 | } |
| 81 | |
pbos@webrtc.org | 3f83f9c | 2014-03-13 12:52:27 +0000 | [diff] [blame] | 82 | void TestMinTransmitBitrate(bool pad_to_min_bitrate); |
| 83 | |
asapersson@webrtc.org | 8ef6548 | 2014-01-31 10:05:07 +0000 | [diff] [blame] | 84 | VideoSendStream* send_stream_; |
| 85 | test::FakeEncoder fake_encoder_; |
pbos@webrtc.org | f94ccd4 | 2013-12-13 12:48:05 +0000 | [diff] [blame] | 86 | }; |
| 87 | |
| 88 | class SyncRtcpObserver : public test::RtpRtcpObserver { |
| 89 | public: |
stefan@webrtc.org | aacdb9f | 2013-12-18 20:28:25 +0000 | [diff] [blame] | 90 | explicit SyncRtcpObserver(const FakeNetworkPipe::Config& config) |
| 91 | : test::RtpRtcpObserver(kLongTimeoutMs, config), |
pbos@webrtc.org | f94ccd4 | 2013-12-13 12:48:05 +0000 | [diff] [blame] | 92 | critical_section_(CriticalSectionWrapper::CreateCriticalSection()) {} |
| 93 | |
| 94 | virtual Action OnSendRtcp(const uint8_t* packet, size_t length) OVERRIDE { |
| 95 | RTCPUtility::RTCPParserV2 parser(packet, length, true); |
| 96 | EXPECT_TRUE(parser.IsValid()); |
| 97 | |
| 98 | for (RTCPUtility::RTCPPacketTypes packet_type = parser.Begin(); |
| 99 | packet_type != RTCPUtility::kRtcpNotValidCode; |
| 100 | packet_type = parser.Iterate()) { |
| 101 | if (packet_type == RTCPUtility::kRtcpSrCode) { |
| 102 | const RTCPUtility::RTCPPacket& packet = parser.Packet(); |
| 103 | synchronization::RtcpMeasurement ntp_rtp_pair( |
| 104 | packet.SR.NTPMostSignificant, |
| 105 | packet.SR.NTPLeastSignificant, |
| 106 | packet.SR.RTPTimestamp); |
| 107 | StoreNtpRtpPair(ntp_rtp_pair); |
| 108 | } |
| 109 | } |
| 110 | return SEND_PACKET; |
| 111 | } |
| 112 | |
| 113 | int64_t RtpTimestampToNtp(uint32_t timestamp) const { |
| 114 | CriticalSectionScoped cs(critical_section_.get()); |
| 115 | int64_t timestamp_in_ms = -1; |
| 116 | if (ntp_rtp_pairs_.size() == 2) { |
| 117 | // TODO(stefan): We can't EXPECT_TRUE on this call due to a bug in the |
| 118 | // RTCP sender where it sends RTCP SR before any RTP packets, which leads |
| 119 | // to a bogus NTP/RTP mapping. |
| 120 | synchronization::RtpToNtpMs(timestamp, ntp_rtp_pairs_, ×tamp_in_ms); |
| 121 | return timestamp_in_ms; |
| 122 | } |
| 123 | return -1; |
| 124 | } |
| 125 | |
| 126 | private: |
| 127 | void StoreNtpRtpPair(synchronization::RtcpMeasurement ntp_rtp_pair) { |
| 128 | CriticalSectionScoped cs(critical_section_.get()); |
| 129 | for (synchronization::RtcpList::iterator it = ntp_rtp_pairs_.begin(); |
| 130 | it != ntp_rtp_pairs_.end(); |
| 131 | ++it) { |
| 132 | if (ntp_rtp_pair.ntp_secs == it->ntp_secs && |
| 133 | ntp_rtp_pair.ntp_frac == it->ntp_frac) { |
| 134 | // This RTCP has already been added to the list. |
| 135 | return; |
| 136 | } |
| 137 | } |
| 138 | // We need two RTCP SR reports to map between RTP and NTP. More than two |
| 139 | // will not improve the mapping. |
| 140 | if (ntp_rtp_pairs_.size() == 2) { |
| 141 | ntp_rtp_pairs_.pop_back(); |
| 142 | } |
| 143 | ntp_rtp_pairs_.push_front(ntp_rtp_pair); |
| 144 | } |
| 145 | |
| 146 | scoped_ptr<CriticalSectionWrapper> critical_section_; |
| 147 | synchronization::RtcpList ntp_rtp_pairs_; |
| 148 | }; |
| 149 | |
| 150 | class VideoRtcpAndSyncObserver : public SyncRtcpObserver, public VideoRenderer { |
| 151 | static const int kInSyncThresholdMs = 50; |
| 152 | static const int kStartupTimeMs = 2000; |
| 153 | static const int kMinRunTimeMs = 30000; |
| 154 | |
| 155 | public: |
| 156 | VideoRtcpAndSyncObserver(Clock* clock, |
| 157 | int voe_channel, |
| 158 | VoEVideoSync* voe_sync, |
henrik.lundin@webrtc.org | 70e2ce9 | 2014-03-06 10:28:07 +0000 | [diff] [blame] | 159 | SyncRtcpObserver* audio_observer, |
| 160 | bool using_new_acm) |
stefan@webrtc.org | aacdb9f | 2013-12-18 20:28:25 +0000 | [diff] [blame] | 161 | : SyncRtcpObserver(FakeNetworkPipe::Config()), |
pbos@webrtc.org | f94ccd4 | 2013-12-13 12:48:05 +0000 | [diff] [blame] | 162 | clock_(clock), |
| 163 | voe_channel_(voe_channel), |
| 164 | voe_sync_(voe_sync), |
| 165 | audio_observer_(audio_observer), |
| 166 | creation_time_ms_(clock_->TimeInMilliseconds()), |
henrik.lundin@webrtc.org | 70e2ce9 | 2014-03-06 10:28:07 +0000 | [diff] [blame] | 167 | first_time_in_sync_(-1), |
| 168 | using_new_acm_(using_new_acm) {} |
pbos@webrtc.org | f94ccd4 | 2013-12-13 12:48:05 +0000 | [diff] [blame] | 169 | |
| 170 | virtual void RenderFrame(const I420VideoFrame& video_frame, |
| 171 | int time_to_render_ms) OVERRIDE { |
| 172 | int64_t now_ms = clock_->TimeInMilliseconds(); |
| 173 | uint32_t playout_timestamp = 0; |
| 174 | if (voe_sync_->GetPlayoutTimestamp(voe_channel_, playout_timestamp) != 0) |
| 175 | return; |
| 176 | int64_t latest_audio_ntp = |
| 177 | audio_observer_->RtpTimestampToNtp(playout_timestamp); |
| 178 | int64_t latest_video_ntp = RtpTimestampToNtp(video_frame.timestamp()); |
| 179 | if (latest_audio_ntp < 0 || latest_video_ntp < 0) |
| 180 | return; |
| 181 | int time_until_render_ms = |
| 182 | std::max(0, static_cast<int>(video_frame.render_time_ms() - now_ms)); |
| 183 | latest_video_ntp += time_until_render_ms; |
| 184 | int64_t stream_offset = latest_audio_ntp - latest_video_ntp; |
| 185 | std::stringstream ss; |
| 186 | ss << stream_offset; |
henrik.lundin@webrtc.org | 70e2ce9 | 2014-03-06 10:28:07 +0000 | [diff] [blame] | 187 | std::stringstream acm_type; |
| 188 | if (using_new_acm_) { |
| 189 | acm_type << "_acm2"; |
| 190 | } |
| 191 | webrtc::test::PrintResult("stream_offset", |
| 192 | acm_type.str(), |
| 193 | "synchronization", |
| 194 | ss.str(), |
| 195 | "ms", |
| 196 | false); |
pbos@webrtc.org | f94ccd4 | 2013-12-13 12:48:05 +0000 | [diff] [blame] | 197 | int64_t time_since_creation = now_ms - creation_time_ms_; |
| 198 | // During the first couple of seconds audio and video can falsely be |
| 199 | // estimated as being synchronized. We don't want to trigger on those. |
| 200 | if (time_since_creation < kStartupTimeMs) |
| 201 | return; |
pbos@webrtc.org | 2a25b6c | 2014-03-13 08:53:39 +0000 | [diff] [blame] | 202 | if (std::abs(latest_audio_ntp - latest_video_ntp) < kInSyncThresholdMs) { |
pbos@webrtc.org | f94ccd4 | 2013-12-13 12:48:05 +0000 | [diff] [blame] | 203 | if (first_time_in_sync_ == -1) { |
| 204 | first_time_in_sync_ = now_ms; |
| 205 | webrtc::test::PrintResult("sync_convergence_time", |
henrik.lundin@webrtc.org | 70e2ce9 | 2014-03-06 10:28:07 +0000 | [diff] [blame] | 206 | acm_type.str(), |
pbos@webrtc.org | f94ccd4 | 2013-12-13 12:48:05 +0000 | [diff] [blame] | 207 | "synchronization", |
| 208 | time_since_creation, |
| 209 | "ms", |
| 210 | false); |
| 211 | } |
| 212 | if (time_since_creation > kMinRunTimeMs) |
| 213 | observation_complete_->Set(); |
| 214 | } |
| 215 | } |
| 216 | |
| 217 | private: |
| 218 | Clock* clock_; |
| 219 | int voe_channel_; |
| 220 | VoEVideoSync* voe_sync_; |
| 221 | SyncRtcpObserver* audio_observer_; |
| 222 | int64_t creation_time_ms_; |
| 223 | int64_t first_time_in_sync_; |
henrik.lundin@webrtc.org | 70e2ce9 | 2014-03-06 10:28:07 +0000 | [diff] [blame] | 224 | bool using_new_acm_; |
pbos@webrtc.org | f94ccd4 | 2013-12-13 12:48:05 +0000 | [diff] [blame] | 225 | }; |
| 226 | |
henrik.lundin@webrtc.org | 70e2ce9 | 2014-03-06 10:28:07 +0000 | [diff] [blame] | 227 | class ParamCallPerfTest : public CallPerfTest, |
| 228 | public ::testing::WithParamInterface<bool> { |
| 229 | public: |
| 230 | ParamCallPerfTest() : CallPerfTest(), use_new_acm_(GetParam()) {} |
| 231 | |
| 232 | protected: |
| 233 | bool use_new_acm_; |
| 234 | }; |
| 235 | |
| 236 | TEST_P(ParamCallPerfTest, PlaysOutAudioAndVideoInSync) { |
pbos@webrtc.org | f94ccd4 | 2013-12-13 12:48:05 +0000 | [diff] [blame] | 237 | VoiceEngine* voice_engine = VoiceEngine::Create(); |
| 238 | VoEBase* voe_base = VoEBase::GetInterface(voice_engine); |
| 239 | VoECodec* voe_codec = VoECodec::GetInterface(voice_engine); |
| 240 | VoENetwork* voe_network = VoENetwork::GetInterface(voice_engine); |
| 241 | VoEVideoSync* voe_sync = VoEVideoSync::GetInterface(voice_engine); |
| 242 | const std::string audio_filename = |
| 243 | test::ResourcePath("voice_engine/audio_long16", "pcm"); |
| 244 | ASSERT_STRNE("", audio_filename.c_str()); |
| 245 | test::FakeAudioDevice fake_audio_device(Clock::GetRealTimeClock(), |
| 246 | audio_filename); |
| 247 | EXPECT_EQ(0, voe_base->Init(&fake_audio_device, NULL)); |
henrik.lundin@webrtc.org | 70e2ce9 | 2014-03-06 10:28:07 +0000 | [diff] [blame] | 248 | Config config; |
| 249 | if (use_new_acm_) { |
| 250 | config.Set<webrtc::AudioCodingModuleFactory>( |
| 251 | new webrtc::NewAudioCodingModuleFactory()); |
| 252 | } else { |
| 253 | config.Set<webrtc::AudioCodingModuleFactory>( |
| 254 | new webrtc::AudioCodingModuleFactory()); |
| 255 | } |
| 256 | int channel = voe_base->CreateChannel(config); |
pbos@webrtc.org | f94ccd4 | 2013-12-13 12:48:05 +0000 | [diff] [blame] | 257 | |
stefan@webrtc.org | aacdb9f | 2013-12-18 20:28:25 +0000 | [diff] [blame] | 258 | FakeNetworkPipe::Config net_config; |
| 259 | net_config.queue_delay_ms = 500; |
| 260 | SyncRtcpObserver audio_observer(net_config); |
henrik.lundin@webrtc.org | 70e2ce9 | 2014-03-06 10:28:07 +0000 | [diff] [blame] | 261 | VideoRtcpAndSyncObserver observer(Clock::GetRealTimeClock(), |
| 262 | channel, |
| 263 | voe_sync, |
| 264 | &audio_observer, |
| 265 | use_new_acm_); |
pbos@webrtc.org | f94ccd4 | 2013-12-13 12:48:05 +0000 | [diff] [blame] | 266 | |
| 267 | Call::Config receiver_config(observer.ReceiveTransport()); |
| 268 | receiver_config.voice_engine = voice_engine; |
| 269 | scoped_ptr<Call> sender_call( |
| 270 | Call::Create(Call::Config(observer.SendTransport()))); |
| 271 | scoped_ptr<Call> receiver_call(Call::Create(receiver_config)); |
| 272 | CodecInst isac = {103, "ISAC", 16000, 480, 1, 32000}; |
| 273 | EXPECT_EQ(0, voe_codec->SetSendCodec(channel, isac)); |
| 274 | |
| 275 | class VoicePacketReceiver : public PacketReceiver { |
| 276 | public: |
| 277 | VoicePacketReceiver(int channel, VoENetwork* voe_network) |
| 278 | : channel_(channel), |
| 279 | voe_network_(voe_network), |
| 280 | parser_(RtpHeaderParser::Create()) {} |
| 281 | virtual bool DeliverPacket(const uint8_t* packet, size_t length) { |
| 282 | int ret; |
| 283 | if (parser_->IsRtcp(packet, static_cast<int>(length))) { |
| 284 | ret = voe_network_->ReceivedRTCPPacket( |
| 285 | channel_, packet, static_cast<unsigned int>(length)); |
| 286 | } else { |
| 287 | ret = voe_network_->ReceivedRTPPacket( |
solenberg@webrtc.org | fec6b6e | 2014-03-24 10:38:25 +0000 | [diff] [blame] | 288 | channel_, packet, static_cast<unsigned int>(length), PacketTime()); |
pbos@webrtc.org | f94ccd4 | 2013-12-13 12:48:05 +0000 | [diff] [blame] | 289 | } |
| 290 | return ret == 0; |
| 291 | } |
| 292 | |
| 293 | private: |
| 294 | int channel_; |
| 295 | VoENetwork* voe_network_; |
| 296 | scoped_ptr<RtpHeaderParser> parser_; |
| 297 | } voe_packet_receiver(channel, voe_network); |
| 298 | |
| 299 | audio_observer.SetReceivers(&voe_packet_receiver, &voe_packet_receiver); |
| 300 | |
| 301 | internal::TransportAdapter transport_adapter(audio_observer.SendTransport()); |
sprang@webrtc.org | 0a29815 | 2014-01-27 13:03:02 +0000 | [diff] [blame] | 302 | transport_adapter.Enable(); |
pbos@webrtc.org | f94ccd4 | 2013-12-13 12:48:05 +0000 | [diff] [blame] | 303 | EXPECT_EQ(0, |
| 304 | voe_network->RegisterExternalTransport(channel, transport_adapter)); |
| 305 | |
| 306 | observer.SetReceivers(receiver_call->Receiver(), sender_call->Receiver()); |
| 307 | |
pbos@webrtc.org | f94ccd4 | 2013-12-13 12:48:05 +0000 | [diff] [blame] | 308 | test::FakeDecoder fake_decoder; |
| 309 | |
asapersson@webrtc.org | 8ef6548 | 2014-01-31 10:05:07 +0000 | [diff] [blame] | 310 | VideoSendStream::Config send_config = GetSendTestConfig(sender_call.get()); |
pbos@webrtc.org | f94ccd4 | 2013-12-13 12:48:05 +0000 | [diff] [blame] | 311 | |
| 312 | VideoReceiveStream::Config receive_config = |
| 313 | receiver_call->GetDefaultReceiveConfig(); |
pbos@webrtc.org | e2a7a77 | 2014-03-19 08:43:57 +0000 | [diff] [blame] | 314 | assert(receive_config.codecs.empty()); |
| 315 | VideoCodec codec = |
| 316 | test::CreateDecoderVideoCodec(send_config.encoder_settings); |
| 317 | receive_config.codecs.push_back(codec); |
| 318 | assert(receive_config.external_decoders.empty()); |
pbos@webrtc.org | f94ccd4 | 2013-12-13 12:48:05 +0000 | [diff] [blame] | 319 | ExternalVideoDecoder decoder; |
| 320 | decoder.decoder = &fake_decoder; |
pbos@webrtc.org | e2a7a77 | 2014-03-19 08:43:57 +0000 | [diff] [blame] | 321 | decoder.payload_type = send_config.encoder_settings.payload_type; |
pbos@webrtc.org | f94ccd4 | 2013-12-13 12:48:05 +0000 | [diff] [blame] | 322 | receive_config.external_decoders.push_back(decoder); |
| 323 | receive_config.rtp.remote_ssrc = send_config.rtp.ssrcs[0]; |
| 324 | receive_config.rtp.local_ssrc = kReceiverLocalSsrc; |
| 325 | receive_config.renderer = &observer; |
| 326 | receive_config.audio_channel_id = channel; |
| 327 | |
| 328 | VideoSendStream* send_stream = |
| 329 | sender_call->CreateVideoSendStream(send_config); |
| 330 | VideoReceiveStream* receive_stream = |
| 331 | receiver_call->CreateVideoReceiveStream(receive_config); |
| 332 | scoped_ptr<test::FrameGeneratorCapturer> capturer( |
pbos@webrtc.org | e2a7a77 | 2014-03-19 08:43:57 +0000 | [diff] [blame] | 333 | test::FrameGeneratorCapturer::Create( |
| 334 | send_stream->Input(), |
| 335 | send_config.encoder_settings.streams[0].width, |
| 336 | send_config.encoder_settings.streams[0].height, |
| 337 | 30, |
| 338 | Clock::GetRealTimeClock())); |
pbos@webrtc.org | f94ccd4 | 2013-12-13 12:48:05 +0000 | [diff] [blame] | 339 | receive_stream->StartReceiving(); |
| 340 | send_stream->StartSending(); |
| 341 | capturer->Start(); |
| 342 | |
| 343 | fake_audio_device.Start(); |
| 344 | EXPECT_EQ(0, voe_base->StartPlayout(channel)); |
| 345 | EXPECT_EQ(0, voe_base->StartReceive(channel)); |
| 346 | EXPECT_EQ(0, voe_base->StartSend(channel)); |
| 347 | |
| 348 | EXPECT_EQ(kEventSignaled, observer.Wait()) |
| 349 | << "Timed out while waiting for audio and video to be synchronized."; |
| 350 | |
| 351 | EXPECT_EQ(0, voe_base->StopSend(channel)); |
| 352 | EXPECT_EQ(0, voe_base->StopReceive(channel)); |
| 353 | EXPECT_EQ(0, voe_base->StopPlayout(channel)); |
| 354 | fake_audio_device.Stop(); |
| 355 | |
| 356 | capturer->Stop(); |
| 357 | send_stream->StopSending(); |
| 358 | receive_stream->StopReceiving(); |
| 359 | observer.StopSending(); |
| 360 | audio_observer.StopSending(); |
| 361 | |
| 362 | voe_base->DeleteChannel(channel); |
| 363 | voe_base->Release(); |
| 364 | voe_codec->Release(); |
| 365 | voe_network->Release(); |
| 366 | voe_sync->Release(); |
| 367 | sender_call->DestroyVideoSendStream(send_stream); |
| 368 | receiver_call->DestroyVideoReceiveStream(receive_stream); |
| 369 | VoiceEngine::Delete(voice_engine); |
| 370 | } |
asapersson@webrtc.org | 8ef6548 | 2014-01-31 10:05:07 +0000 | [diff] [blame] | 371 | |
henrik.lundin@webrtc.org | 70e2ce9 | 2014-03-06 10:28:07 +0000 | [diff] [blame] | 372 | // Test with both ACM1 and ACM2. |
| 373 | INSTANTIATE_TEST_CASE_P(SwitchAcm, ParamCallPerfTest, ::testing::Bool()); |
| 374 | |
asapersson@webrtc.org | 8ef6548 | 2014-01-31 10:05:07 +0000 | [diff] [blame] | 375 | TEST_F(CallPerfTest, RegisterCpuOveruseObserver) { |
| 376 | // Verifies that either a normal or overuse callback is triggered. |
| 377 | class OveruseCallbackObserver : public test::RtpRtcpObserver, |
| 378 | public webrtc::OveruseCallback { |
| 379 | public: |
| 380 | OveruseCallbackObserver() : RtpRtcpObserver(kLongTimeoutMs) {} |
| 381 | |
| 382 | virtual void OnOveruse() OVERRIDE { |
| 383 | observation_complete_->Set(); |
| 384 | } |
| 385 | virtual void OnNormalUse() OVERRIDE { |
| 386 | observation_complete_->Set(); |
| 387 | } |
| 388 | }; |
| 389 | |
| 390 | OveruseCallbackObserver observer; |
| 391 | Call::Config call_config(observer.SendTransport()); |
| 392 | call_config.overuse_callback = &observer; |
| 393 | scoped_ptr<Call> call(Call::Create(call_config)); |
| 394 | |
| 395 | VideoSendStream::Config send_config = GetSendTestConfig(call.get()); |
| 396 | RunVideoSendTest(call.get(), send_config, &observer); |
| 397 | } |
pbos@webrtc.org | 3f83f9c | 2014-03-13 12:52:27 +0000 | [diff] [blame] | 398 | |
| 399 | void CallPerfTest::TestMinTransmitBitrate(bool pad_to_min_bitrate) { |
| 400 | static const int kMaxEncodeBitrateKbps = 30; |
pbos@webrtc.org | 1d61e3a | 2014-03-19 10:59:52 +0000 | [diff] [blame] | 401 | static const int kMinTransmitBitrateBps = 150000; |
pbos@webrtc.org | 3f83f9c | 2014-03-13 12:52:27 +0000 | [diff] [blame] | 402 | static const int kMinAcceptableTransmitBitrate = 130; |
| 403 | static const int kMaxAcceptableTransmitBitrate = 170; |
| 404 | static const int kNumBitrateObservationsInRange = 100; |
| 405 | class BitrateObserver : public test::RtpRtcpObserver, public PacketReceiver { |
| 406 | public: |
| 407 | explicit BitrateObserver(bool using_min_transmit_bitrate) |
| 408 | : test::RtpRtcpObserver(kLongTimeoutMs), |
| 409 | send_stream_(NULL), |
| 410 | send_transport_receiver_(NULL), |
| 411 | using_min_transmit_bitrate_(using_min_transmit_bitrate), |
| 412 | num_bitrate_observations_in_range_(0) {} |
| 413 | |
| 414 | virtual void SetReceivers(PacketReceiver* send_transport_receiver, |
| 415 | PacketReceiver* receive_transport_receiver) |
| 416 | OVERRIDE { |
| 417 | send_transport_receiver_ = send_transport_receiver; |
| 418 | test::RtpRtcpObserver::SetReceivers(this, receive_transport_receiver); |
| 419 | } |
| 420 | |
| 421 | void SetSendStream(VideoSendStream* send_stream) { |
| 422 | send_stream_ = send_stream; |
| 423 | } |
| 424 | |
| 425 | private: |
| 426 | virtual bool DeliverPacket(const uint8_t* packet, size_t length) OVERRIDE { |
| 427 | VideoSendStream::Stats stats = send_stream_->GetStats(); |
| 428 | if (stats.substreams.size() > 0) { |
| 429 | assert(stats.substreams.size() == 1); |
| 430 | int bitrate_kbps = stats.substreams.begin()->second.bitrate_bps / 1000; |
| 431 | if (bitrate_kbps > 0) { |
| 432 | test::PrintResult( |
| 433 | "bitrate_stats_", |
| 434 | (using_min_transmit_bitrate_ ? "min_transmit_bitrate" |
| 435 | : "without_min_transmit_bitrate"), |
| 436 | "bitrate_kbps", |
| 437 | static_cast<size_t>(bitrate_kbps), |
| 438 | "kbps", |
| 439 | false); |
| 440 | if (using_min_transmit_bitrate_) { |
| 441 | if (bitrate_kbps > kMinAcceptableTransmitBitrate && |
| 442 | bitrate_kbps < kMaxAcceptableTransmitBitrate) { |
| 443 | ++num_bitrate_observations_in_range_; |
| 444 | } |
| 445 | } else { |
| 446 | // Expect bitrate stats to roughly match the max encode bitrate. |
| 447 | if (bitrate_kbps > kMaxEncodeBitrateKbps - 5 && |
| 448 | bitrate_kbps < kMaxEncodeBitrateKbps + 5) { |
| 449 | ++num_bitrate_observations_in_range_; |
| 450 | } |
| 451 | } |
| 452 | if (num_bitrate_observations_in_range_ == |
| 453 | kNumBitrateObservationsInRange) |
| 454 | observation_complete_->Set(); |
| 455 | } |
| 456 | } |
| 457 | return send_transport_receiver_->DeliverPacket(packet, length); |
| 458 | } |
| 459 | |
| 460 | VideoSendStream* send_stream_; |
| 461 | PacketReceiver* send_transport_receiver_; |
| 462 | const bool using_min_transmit_bitrate_; |
| 463 | int num_bitrate_observations_in_range_; |
| 464 | } observer(pad_to_min_bitrate); |
| 465 | |
| 466 | scoped_ptr<Call> sender_call( |
| 467 | Call::Create(Call::Config(observer.SendTransport()))); |
| 468 | scoped_ptr<Call> receiver_call( |
| 469 | Call::Create(Call::Config(observer.ReceiveTransport()))); |
| 470 | |
| 471 | VideoSendStream::Config send_config = GetSendTestConfig(sender_call.get()); |
| 472 | fake_encoder_.SetMaxBitrate(kMaxEncodeBitrateKbps); |
| 473 | |
| 474 | observer.SetReceivers(receiver_call->Receiver(), sender_call->Receiver()); |
| 475 | |
| 476 | send_config.pacing = true; |
| 477 | if (pad_to_min_bitrate) { |
pbos@webrtc.org | 1d61e3a | 2014-03-19 10:59:52 +0000 | [diff] [blame] | 478 | send_config.rtp.min_transmit_bitrate_bps = kMinTransmitBitrateBps; |
pbos@webrtc.org | 3f83f9c | 2014-03-13 12:52:27 +0000 | [diff] [blame] | 479 | } else { |
pbos@webrtc.org | 1d61e3a | 2014-03-19 10:59:52 +0000 | [diff] [blame] | 480 | assert(send_config.rtp.min_transmit_bitrate_bps == 0); |
pbos@webrtc.org | 3f83f9c | 2014-03-13 12:52:27 +0000 | [diff] [blame] | 481 | } |
| 482 | |
| 483 | VideoReceiveStream::Config receive_config = |
| 484 | receiver_call->GetDefaultReceiveConfig(); |
| 485 | receive_config.codecs.clear(); |
pbos@webrtc.org | e2a7a77 | 2014-03-19 08:43:57 +0000 | [diff] [blame] | 486 | VideoCodec codec = |
| 487 | test::CreateDecoderVideoCodec(send_config.encoder_settings); |
| 488 | receive_config.codecs.push_back(codec); |
pbos@webrtc.org | 3f83f9c | 2014-03-13 12:52:27 +0000 | [diff] [blame] | 489 | test::FakeDecoder fake_decoder; |
| 490 | ExternalVideoDecoder decoder; |
| 491 | decoder.decoder = &fake_decoder; |
pbos@webrtc.org | e2a7a77 | 2014-03-19 08:43:57 +0000 | [diff] [blame] | 492 | decoder.payload_type = send_config.encoder_settings.payload_type; |
pbos@webrtc.org | 3f83f9c | 2014-03-13 12:52:27 +0000 | [diff] [blame] | 493 | receive_config.external_decoders.push_back(decoder); |
| 494 | receive_config.rtp.remote_ssrc = send_config.rtp.ssrcs[0]; |
| 495 | receive_config.rtp.local_ssrc = kReceiverLocalSsrc; |
| 496 | |
| 497 | VideoSendStream* send_stream = |
| 498 | sender_call->CreateVideoSendStream(send_config); |
| 499 | VideoReceiveStream* receive_stream = |
| 500 | receiver_call->CreateVideoReceiveStream(receive_config); |
| 501 | scoped_ptr<test::FrameGeneratorCapturer> capturer( |
pbos@webrtc.org | e2a7a77 | 2014-03-19 08:43:57 +0000 | [diff] [blame] | 502 | test::FrameGeneratorCapturer::Create( |
| 503 | send_stream->Input(), |
| 504 | send_config.encoder_settings.streams[0].width, |
| 505 | send_config.encoder_settings.streams[0].height, |
| 506 | 30, |
| 507 | Clock::GetRealTimeClock())); |
pbos@webrtc.org | 3f83f9c | 2014-03-13 12:52:27 +0000 | [diff] [blame] | 508 | observer.SetSendStream(send_stream); |
| 509 | receive_stream->StartReceiving(); |
| 510 | send_stream->StartSending(); |
| 511 | capturer->Start(); |
| 512 | |
| 513 | EXPECT_EQ(kEventSignaled, observer.Wait()) |
| 514 | << "Timeout while waiting for send-bitrate stats."; |
| 515 | |
| 516 | send_stream->StopSending(); |
| 517 | receive_stream->StopReceiving(); |
| 518 | observer.StopSending(); |
| 519 | capturer->Stop(); |
| 520 | sender_call->DestroyVideoSendStream(send_stream); |
| 521 | receiver_call->DestroyVideoReceiveStream(receive_stream); |
| 522 | } |
| 523 | |
| 524 | TEST_F(CallPerfTest, PadsToMinTransmitBitrate) { TestMinTransmitBitrate(true); } |
| 525 | |
| 526 | TEST_F(CallPerfTest, NoPadWithoutMinTransmitBitrate) { |
| 527 | TestMinTransmitBitrate(false); |
| 528 | } |
| 529 | |
pbos@webrtc.org | f94ccd4 | 2013-12-13 12:48:05 +0000 | [diff] [blame] | 530 | } // namespace webrtc |