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Side by Side Diff: webrtc/video/call_perf_tests.cc

Issue 1181653002: Base A/V synchronization on sync_labels. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 5 years, 5 months ago
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1 /* 1 /*
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 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 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 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 #include <algorithm> 10 #include <algorithm>
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37 #include "webrtc/voice_engine/include/voe_base.h" 37 #include "webrtc/voice_engine/include/voe_base.h"
38 #include "webrtc/voice_engine/include/voe_codec.h" 38 #include "webrtc/voice_engine/include/voe_codec.h"
39 #include "webrtc/voice_engine/include/voe_network.h" 39 #include "webrtc/voice_engine/include/voe_network.h"
40 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h" 40 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
41 #include "webrtc/voice_engine/include/voe_video_sync.h" 41 #include "webrtc/voice_engine/include/voe_video_sync.h"
42 42
43 namespace webrtc { 43 namespace webrtc {
44 44
45 class CallPerfTest : public test::CallTest { 45 class CallPerfTest : public test::CallTest {
46 protected: 46 protected:
47 void TestAudioVideoSync(bool fec); 47 void TestAudioVideoSync(bool fec, bool create_audio_first);
48 48
49 void TestCpuOveruse(LoadObserver::Load tested_load, int encode_delay_ms); 49 void TestCpuOveruse(LoadObserver::Load tested_load, int encode_delay_ms);
50 50
51 void TestMinTransmitBitrate(bool pad_to_min_bitrate); 51 void TestMinTransmitBitrate(bool pad_to_min_bitrate);
52 52
53 void TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config, 53 void TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config,
54 int threshold_ms, 54 int threshold_ms,
55 int start_time_ms, 55 int start_time_ms,
56 int run_time_ms); 56 int run_time_ms);
57 }; 57 };
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182 182
183 private: 183 private:
184 Clock* const clock_; 184 Clock* const clock_;
185 int voe_channel_; 185 int voe_channel_;
186 VoEVideoSync* voe_sync_; 186 VoEVideoSync* voe_sync_;
187 SyncRtcpObserver* audio_observer_; 187 SyncRtcpObserver* audio_observer_;
188 int64_t creation_time_ms_; 188 int64_t creation_time_ms_;
189 int64_t first_time_in_sync_; 189 int64_t first_time_in_sync_;
190 }; 190 };
191 191
192 void CallPerfTest::TestAudioVideoSync(bool fec) { 192 void CallPerfTest::TestAudioVideoSync(bool fec, bool create_audio_first) {
193 const char* kSyncGroup = "av_sync";
193 class AudioPacketReceiver : public PacketReceiver { 194 class AudioPacketReceiver : public PacketReceiver {
194 public: 195 public:
195 AudioPacketReceiver(int channel, VoENetwork* voe_network) 196 AudioPacketReceiver(int channel, VoENetwork* voe_network)
196 : channel_(channel), 197 : channel_(channel),
197 voe_network_(voe_network), 198 voe_network_(voe_network),
198 parser_(RtpHeaderParser::Create()) {} 199 parser_(RtpHeaderParser::Create()) {}
199 DeliveryStatus DeliverPacket(MediaType media_type, const uint8_t* packet, 200 DeliveryStatus DeliverPacket(MediaType media_type, const uint8_t* packet,
200 size_t length) override { 201 size_t length) override {
201 EXPECT_TRUE(media_type == MediaType::ANY || 202 EXPECT_TRUE(media_type == MediaType::ANY ||
202 media_type == MediaType::AUDIO); 203 media_type == MediaType::AUDIO);
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262 263
263 send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs; 264 send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
264 if (fec) { 265 if (fec) {
265 send_config_.rtp.fec.red_payload_type = kRedPayloadType; 266 send_config_.rtp.fec.red_payload_type = kRedPayloadType;
266 send_config_.rtp.fec.ulpfec_payload_type = kUlpfecPayloadType; 267 send_config_.rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
267 receive_configs_[0].rtp.fec.red_payload_type = kRedPayloadType; 268 receive_configs_[0].rtp.fec.red_payload_type = kRedPayloadType;
268 receive_configs_[0].rtp.fec.ulpfec_payload_type = kUlpfecPayloadType; 269 receive_configs_[0].rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
269 } 270 }
270 receive_configs_[0].rtp.nack.rtp_history_ms = 1000; 271 receive_configs_[0].rtp.nack.rtp_history_ms = 1000;
271 receive_configs_[0].renderer = &observer; 272 receive_configs_[0].renderer = &observer;
272 receive_configs_[0].audio_channel_id = channel; 273 receive_configs_[0].sync_group = kSyncGroup;
273 274
274 CreateStreams(); 275 AudioReceiveStream::Config audio_config;
276 audio_config.voe_channel_id = channel;
277 audio_config.sync_group = kSyncGroup;
278
279 AudioReceiveStream* audio_receive_stream = nullptr;
280
281 if (create_audio_first) {
282 audio_receive_stream =
283 receiver_call_->CreateAudioReceiveStream(audio_config);
284 CreateStreams();
285 } else {
286 CreateStreams();
287 audio_receive_stream =
288 receiver_call_->CreateAudioReceiveStream(audio_config);
289 }
275 290
276 CreateFrameGeneratorCapturer(); 291 CreateFrameGeneratorCapturer();
277 292
278 Start(); 293 Start();
279 294
280 fake_audio_device.Start(); 295 fake_audio_device.Start();
281 EXPECT_EQ(0, voe_base->StartPlayout(channel)); 296 EXPECT_EQ(0, voe_base->StartPlayout(channel));
282 EXPECT_EQ(0, voe_base->StartReceive(channel)); 297 EXPECT_EQ(0, voe_base->StartReceive(channel));
283 EXPECT_EQ(0, voe_base->StartSend(channel)); 298 EXPECT_EQ(0, voe_base->StartSend(channel));
284 299
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295 audio_observer.StopSending(); 310 audio_observer.StopSending();
296 311
297 voe_base->DeleteChannel(channel); 312 voe_base->DeleteChannel(channel);
298 voe_base->Release(); 313 voe_base->Release();
299 voe_codec->Release(); 314 voe_codec->Release();
300 voe_network->Release(); 315 voe_network->Release();
301 voe_sync->Release(); 316 voe_sync->Release();
302 317
303 DestroyStreams(); 318 DestroyStreams();
304 319
320 receiver_call_->DestroyAudioReceiveStream(audio_receive_stream);
321
305 VoiceEngine::Delete(voice_engine); 322 VoiceEngine::Delete(voice_engine);
306 } 323 }
307 324
308 TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSync) { 325 TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithAudioCreatedFirst) {
309 TestAudioVideoSync(false); 326 TestAudioVideoSync(false, true);
327 }
328
329 TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithVideoCreatedFirst) {
330 TestAudioVideoSync(false, false);
310 } 331 }
311 332
312 TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithFec) { 333 TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithFec) {
313 TestAudioVideoSync(true); 334 TestAudioVideoSync(true, false);
314 } 335 }
315 336
316 void CallPerfTest::TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config, 337 void CallPerfTest::TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config,
317 int threshold_ms, 338 int threshold_ms,
318 int start_time_ms, 339 int start_time_ms,
319 int run_time_ms) { 340 int run_time_ms) {
320 class CaptureNtpTimeObserver : public test::EndToEndTest, 341 class CaptureNtpTimeObserver : public test::EndToEndTest,
321 public VideoRenderer { 342 public VideoRenderer {
322 public: 343 public:
323 CaptureNtpTimeObserver(const FakeNetworkPipe::Config& config, 344 CaptureNtpTimeObserver(const FakeNetworkPipe::Config& config,
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678 int encoder_inits_; 699 int encoder_inits_;
679 uint32_t last_set_bitrate_; 700 uint32_t last_set_bitrate_;
680 VideoSendStream* send_stream_; 701 VideoSendStream* send_stream_;
681 VideoEncoderConfig encoder_config_; 702 VideoEncoderConfig encoder_config_;
682 } test; 703 } test;
683 704
684 RunBaseTest(&test); 705 RunBaseTest(&test);
685 } 706 }
686 707
687 } // namespace webrtc 708 } // namespace webrtc
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