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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
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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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