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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> // max | 10 #include <algorithm> // max |
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| 330 void PerformTest() override { | 330 void PerformTest() override { |
| 331 EXPECT_TRUE(Wait()) << "Timed out while waiting for single RTP packet."; | 331 EXPECT_TRUE(Wait()) << "Timed out while waiting for single RTP packet."; |
| 332 } | 332 } |
| 333 } test; | 333 } test; |
| 334 | 334 |
| 335 test::ScopedFieldTrials override_field_trials( | 335 test::ScopedFieldTrials override_field_trials( |
| 336 "WebRTC-VideoContentTypeExtension/Enabled/"); | 336 "WebRTC-VideoContentTypeExtension/Enabled/"); |
| 337 RunBaseTest(&test); | 337 RunBaseTest(&test); |
| 338 } | 338 } |
| 339 | 339 |
| 340 TEST_F(VideoSendStreamTest, SupportsVideoTimingFrames) { | |
| 341 class VideoRotationObserver : public test::SendTest { | |
| 342 public: | |
| 343 VideoRotationObserver() | |
| 344 : SendTest(kDefaultTimeoutMs), timing_frame_observed_(false) { | |
| 345 EXPECT_TRUE(parser_->RegisterRtpHeaderExtension( | |
| 346 kRtpExtensionVideoTiming, test::kVideoTimingExtensionId)); | |
| 347 } | |
| 348 | |
| 349 Action OnSendRtp(const uint8_t* packet, size_t length) override { | |
| 350 RTPHeader header; | |
| 351 EXPECT_TRUE(parser_->Parse(packet, length, &header)); | |
| 352 if (header.extension.hasVideoTiming) { | |
| 353 observation_complete_.Set(); | |
| 354 timing_frame_observed_ = true; | |
|
sprang_webrtc
2017/06/13 14:14:14
Do you need this variable?
ilnik
2017/06/13 14:55:44
You are right, it's not needed, because if observa
| |
| 355 } | |
| 356 return SEND_PACKET; | |
| 357 } | |
| 358 | |
| 359 void ModifyVideoConfigs( | |
| 360 VideoSendStream::Config* send_config, | |
| 361 std::vector<VideoReceiveStream::Config>* receive_configs, | |
| 362 VideoEncoderConfig* encoder_config) override { | |
| 363 send_config->rtp.extensions.clear(); | |
| 364 send_config->rtp.extensions.push_back(RtpExtension( | |
| 365 RtpExtension::kVideoTimingUri, test::kVideoTimingExtensionId)); | |
| 366 } | |
| 367 | |
| 368 void PerformTest() override { | |
| 369 EXPECT_TRUE(Wait()) << "Timed out while waiting for timing frames."; | |
| 370 } | |
| 371 | |
| 372 bool timing_frame_observed_; | |
| 373 } test; | |
| 374 | |
| 375 RunBaseTest(&test); | |
| 376 EXPECT_TRUE(test.timing_frame_observed_); | |
| 377 } | |
| 378 | |
| 340 class FakeReceiveStatistics : public NullReceiveStatistics { | 379 class FakeReceiveStatistics : public NullReceiveStatistics { |
| 341 public: | 380 public: |
| 342 FakeReceiveStatistics(uint32_t send_ssrc, | 381 FakeReceiveStatistics(uint32_t send_ssrc, |
| 343 uint32_t last_sequence_number, | 382 uint32_t last_sequence_number, |
| 344 uint32_t cumulative_lost, | 383 uint32_t cumulative_lost, |
| 345 uint8_t fraction_lost) | 384 uint8_t fraction_lost) |
| 346 : lossy_stats_(new LossyStatistician(last_sequence_number, | 385 : lossy_stats_(new LossyStatistician(last_sequence_number, |
| 347 cumulative_lost, | 386 cumulative_lost, |
| 348 fraction_lost)) { | 387 fraction_lost)) { |
| 349 stats_map_[send_ssrc] = lossy_stats_.get(); | 388 stats_map_[send_ssrc] = lossy_stats_.get(); |
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| 3334 rtc::CriticalSection crit_; | 3373 rtc::CriticalSection crit_; |
| 3335 uint32_t max_bitrate_bps_ GUARDED_BY(&crit_); | 3374 uint32_t max_bitrate_bps_ GUARDED_BY(&crit_); |
| 3336 bool first_packet_sent_ GUARDED_BY(&crit_); | 3375 bool first_packet_sent_ GUARDED_BY(&crit_); |
| 3337 rtc::Event bitrate_changed_event_; | 3376 rtc::Event bitrate_changed_event_; |
| 3338 } test; | 3377 } test; |
| 3339 | 3378 |
| 3340 RunBaseTest(&test); | 3379 RunBaseTest(&test); |
| 3341 } | 3380 } |
| 3342 | 3381 |
| 3343 } // namespace webrtc | 3382 } // namespace webrtc |
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