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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 |
| 11 #include <memory> | 11 #include <memory> |
| 12 #include <vector> | 12 #include <vector> |
| 13 | 13 |
| 14 #include "webrtc/base/bind.h" | 14 #include "webrtc/base/bind.h" |
| 15 #include "webrtc/base/checks.h" | 15 #include "webrtc/base/checks.h" |
| 16 #include "webrtc/base/criticalsection.h" | 16 #include "webrtc/base/criticalsection.h" |
| 17 #include "webrtc/base/event.h" | 17 #include "webrtc/base/event.h" |
| 18 #include "webrtc/base/logging.h" | 18 #include "webrtc/base/logging.h" |
| 19 #include "webrtc/base/platform_thread.h" | 19 #include "webrtc/base/platform_thread.h" |
| 20 #include "webrtc/base/rate_limiter.h" | 20 #include "webrtc/base/rate_limiter.h" |
| 21 #include "webrtc/call.h" | 21 #include "webrtc/call.h" |
| 22 #include "webrtc/common_video/include/frame_callback.h" | 22 #include "webrtc/common_video/include/frame_callback.h" |
| 23 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" | 23 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" |
| 24 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" | 24 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" |
| 25 #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h" | 25 #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h" |
| 26 #include "webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h" | 26 #include "webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h" |
| 27 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h" | |
| 28 #include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h" | 27 #include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h" |
| 29 #include "webrtc/system_wrappers/include/sleep.h" | 28 #include "webrtc/system_wrappers/include/sleep.h" |
| 30 #include "webrtc/test/call_test.h" | 29 #include "webrtc/test/call_test.h" |
| 31 #include "webrtc/test/configurable_frame_size_encoder.h" | 30 #include "webrtc/test/configurable_frame_size_encoder.h" |
| 32 #include "webrtc/test/fake_texture_frame.h" | 31 #include "webrtc/test/fake_texture_frame.h" |
| 33 #include "webrtc/test/frame_generator.h" | 32 #include "webrtc/test/frame_generator.h" |
| 34 #include "webrtc/test/frame_utils.h" | 33 #include "webrtc/test/frame_utils.h" |
| 35 #include "webrtc/test/gtest.h" | 34 #include "webrtc/test/gtest.h" |
| 36 #include "webrtc/test/null_transport.h" | 35 #include "webrtc/test/null_transport.h" |
| 37 #include "webrtc/test/rtcp_packet_parser.h" | 36 #include "webrtc/test/rtcp_packet_parser.h" |
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| 361 use_nack_(use_nack), | 360 use_nack_(use_nack), |
| 362 expect_red_(expect_red), | 361 expect_red_(expect_red), |
| 363 expect_ulpfec_(expect_ulpfec), | 362 expect_ulpfec_(expect_ulpfec), |
| 364 send_count_(0), | 363 send_count_(0), |
| 365 received_media_(false), | 364 received_media_(false), |
| 366 received_fec_(false), | 365 received_fec_(false), |
| 367 header_extensions_enabled_(header_extensions_enabled) { | 366 header_extensions_enabled_(header_extensions_enabled) { |
| 368 if (codec == "H264") { | 367 if (codec == "H264") { |
| 369 encoder_.reset(new test::FakeH264Encoder(Clock::GetRealTimeClock())); | 368 encoder_.reset(new test::FakeH264Encoder(Clock::GetRealTimeClock())); |
| 370 } else if (codec == "VP8") { | 369 } else if (codec == "VP8") { |
| 371 encoder_.reset(VP8Encoder::Create()); | 370 encoder_.reset(VideoEncoder::Create(VideoEncoder::EncoderType::kVp8)); |
| 372 } else if (codec == "VP9") { | 371 } else if (codec == "VP9") { |
| 373 encoder_.reset(VP9Encoder::Create()); | 372 encoder_.reset(VideoEncoder::Create(VideoEncoder::EncoderType::kVp9)); |
| 374 } else { | 373 } else { |
| 375 RTC_NOTREACHED(); | 374 RTC_NOTREACHED(); |
| 376 } | 375 } |
| 377 } | 376 } |
| 378 | 377 |
| 379 private: | 378 private: |
| 380 Action OnSendRtp(const uint8_t* packet, size_t length) override { | 379 Action OnSendRtp(const uint8_t* packet, size_t length) override { |
| 381 RTPHeader header; | 380 RTPHeader header; |
| 382 EXPECT_TRUE(parser_->Parse(packet, length, &header)); | 381 EXPECT_TRUE(parser_->Parse(packet, length, &header)); |
| 383 | 382 |
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| 546 : EndToEndTest(VideoSendStreamTest::kDefaultTimeoutMs), | 545 : EndToEndTest(VideoSendStreamTest::kDefaultTimeoutMs), |
| 547 payload_name_(codec), | 546 payload_name_(codec), |
| 548 use_nack_(use_nack), | 547 use_nack_(use_nack), |
| 549 send_count_(0), | 548 send_count_(0), |
| 550 sent_media_(false), | 549 sent_media_(false), |
| 551 sent_flexfec_(false), | 550 sent_flexfec_(false), |
| 552 header_extensions_enabled_(header_extensions_enabled) { | 551 header_extensions_enabled_(header_extensions_enabled) { |
| 553 if (codec == "H264") { | 552 if (codec == "H264") { |
| 554 encoder_.reset(new test::FakeH264Encoder(Clock::GetRealTimeClock())); | 553 encoder_.reset(new test::FakeH264Encoder(Clock::GetRealTimeClock())); |
| 555 } else if (codec == "VP8") { | 554 } else if (codec == "VP8") { |
| 556 encoder_.reset(VP8Encoder::Create()); | 555 encoder_.reset(VideoEncoder::Create(VideoEncoder::EncoderType::kVp8)); |
| 557 } else if (codec == "VP9") { | 556 } else if (codec == "VP9") { |
| 558 encoder_.reset(VP9Encoder::Create()); | 557 encoder_.reset(VideoEncoder::Create(VideoEncoder::EncoderType::kVp9)); |
| 559 } else { | 558 } else { |
| 560 RTC_NOTREACHED(); | 559 RTC_NOTREACHED(); |
| 561 } | 560 } |
| 562 } | 561 } |
| 563 | 562 |
| 564 size_t GetNumFlexfecStreams() const override { return 1; } | 563 size_t GetNumFlexfecStreams() const override { return 1; } |
| 565 | 564 |
| 566 private: | 565 private: |
| 567 Action OnSendRtp(const uint8_t* packet, size_t length) override { | 566 Action OnSendRtp(const uint8_t* packet, size_t length) override { |
| 568 RTPHeader header; | 567 RTPHeader header; |
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| 3168 RequestSourceRotateIfVideoOrientationExtensionNotSupported) { | 3167 RequestSourceRotateIfVideoOrientationExtensionNotSupported) { |
| 3169 TestRequestSourceRotateVideo(false); | 3168 TestRequestSourceRotateVideo(false); |
| 3170 } | 3169 } |
| 3171 | 3170 |
| 3172 TEST_F(VideoSendStreamTest, | 3171 TEST_F(VideoSendStreamTest, |
| 3173 DoNotRequestsRotationIfVideoOrientationExtensionSupported) { | 3172 DoNotRequestsRotationIfVideoOrientationExtensionSupported) { |
| 3174 TestRequestSourceRotateVideo(true); | 3173 TestRequestSourceRotateVideo(true); |
| 3175 } | 3174 } |
| 3176 | 3175 |
| 3177 } // namespace webrtc | 3176 } // namespace webrtc |
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