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

Issue 2500743002: Revert of H.264 packetization mode 0 (try 2) (Closed)
Patch Set: Created 4 years, 1 month 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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123 void VerifyHistogramStats(bool use_rtx, bool use_red, bool screenshare); 123 void VerifyHistogramStats(bool use_rtx, bool use_red, bool screenshare);
124 void VerifyNewVideoSendStreamsRespectNetworkState( 124 void VerifyNewVideoSendStreamsRespectNetworkState(
125 MediaType network_to_bring_down, 125 MediaType network_to_bring_down,
126 VideoEncoder* encoder, 126 VideoEncoder* encoder,
127 Transport* transport); 127 Transport* transport);
128 void VerifyNewVideoReceiveStreamsRespectNetworkState( 128 void VerifyNewVideoReceiveStreamsRespectNetworkState(
129 MediaType network_to_bring_down, 129 MediaType network_to_bring_down,
130 Transport* transport); 130 Transport* transport);
131 }; 131 };
132 132
133 void SetPacketizationMode(H264PacketizationMode mode, VideoEncoder* encoder) {
134 VideoCodec codec_settings;
135 codec_settings.codecType = kVideoCodecH264;
136 codec_settings.H264()->packetization_mode = mode;
137 // TODO(hta): Determine appropriate value for max packet size.
138 static const int kMaxPacketSize = 1024;
139 encoder->InitEncode(&codec_settings, 0, kMaxPacketSize);
140 }
141
142 TEST_F(EndToEndTest, ReceiverCanBeStartedTwice) { 133 TEST_F(EndToEndTest, ReceiverCanBeStartedTwice) {
143 CreateCalls(Call::Config(&event_log_), Call::Config(&event_log_)); 134 CreateCalls(Call::Config(&event_log_), Call::Config(&event_log_));
144 135
145 test::NullTransport transport; 136 test::NullTransport transport;
146 CreateSendConfig(1, 0, &transport); 137 CreateSendConfig(1, 0, &transport);
147 CreateMatchingReceiveConfigs(&transport); 138 CreateMatchingReceiveConfigs(&transport);
148 139
149 CreateVideoStreams(); 140 CreateVideoStreams();
150 141
151 video_receive_streams_[0]->Start(); 142 video_receive_streams_[0]->Start();
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377 } 368 }
378 369
379 TEST_F(EndToEndTest, SendsAndReceivesVP9VideoRotation90) { 370 TEST_F(EndToEndTest, SendsAndReceivesVP9VideoRotation90) {
380 CodecObserver test(5, kVideoRotation_90, "VP9", 371 CodecObserver test(5, kVideoRotation_90, "VP9",
381 VideoEncoder::Create(VideoEncoder::kVp9), 372 VideoEncoder::Create(VideoEncoder::kVp9),
382 VP9Decoder::Create()); 373 VP9Decoder::Create());
383 RunBaseTest(&test); 374 RunBaseTest(&test);
384 } 375 }
385 #endif // !defined(RTC_DISABLE_VP9) 376 #endif // !defined(RTC_DISABLE_VP9)
386 377
387 #if defined(WEBRTC_USE_H264) 378 #if defined(WEBRTC_END_TO_END_H264_TESTS)
388 379
389 TEST_F(EndToEndTest, SendsAndReceivesH264) { 380 TEST_F(EndToEndTest, SendsAndReceivesH264) {
390 CodecObserver test(500, kVideoRotation_0, "H264", 381 CodecObserver test(500, kVideoRotation_0, "H264",
391 VideoEncoder::Create(VideoEncoder::kH264), 382 VideoEncoder::Create(VideoEncoder::kH264),
392 H264Decoder::Create()); 383 H264Decoder::Create());
393 RunBaseTest(&test); 384 RunBaseTest(&test);
394 } 385 }
395 386
396 TEST_F(EndToEndTest, SendsAndReceivesH264VideoRotation90) { 387 TEST_F(EndToEndTest, SendsAndReceivesH264VideoRotation90) {
397 CodecObserver test(5, kVideoRotation_90, "H264", 388 CodecObserver test(5, kVideoRotation_90, "H264",
398 VideoEncoder::Create(VideoEncoder::kH264), 389 VideoEncoder::Create(VideoEncoder::kH264),
399 H264Decoder::Create()); 390 H264Decoder::Create());
400 RunBaseTest(&test); 391 RunBaseTest(&test);
401 } 392 }
402 393
403 TEST_F(EndToEndTest, SendsAndReceivesH264PacketizationMode0) { 394 #endif // defined(WEBRTC_END_TO_END_H264_TESTS)
404 VideoEncoder* encoder = VideoEncoder::Create(VideoEncoder::kH264);
405 SetPacketizationMode(kH264PacketizationMode0, encoder);
406 // The CodecObserver takes ownership of the encoder.
407 CodecObserver test(500, kVideoRotation_0, "H264", encoder,
408 H264Decoder::Create());
409 RunBaseTest(&test);
410 }
411
412 TEST_F(EndToEndTest, SendsAndReceivesH264PacketizationMode1) {
413 VideoEncoder* encoder = VideoEncoder::Create(VideoEncoder::kH264);
414 SetPacketizationMode(kH264PacketizationMode1, encoder);
415 // The CodecObserver takes ownership of the encoder.
416 CodecObserver test(500, kVideoRotation_0, "H264", encoder,
417 H264Decoder::Create());
418 RunBaseTest(&test);
419 }
420
421 #endif // defined(WEBRTC_USE_H264)
422 395
423 TEST_F(EndToEndTest, ReceiverUsesLocalSsrc) { 396 TEST_F(EndToEndTest, ReceiverUsesLocalSsrc) {
424 class SyncRtcpObserver : public test::EndToEndTest { 397 class SyncRtcpObserver : public test::EndToEndTest {
425 public: 398 public:
426 SyncRtcpObserver() : EndToEndTest(kDefaultTimeoutMs) {} 399 SyncRtcpObserver() : EndToEndTest(kDefaultTimeoutMs) {}
427 400
428 Action OnReceiveRtcp(const uint8_t* packet, size_t length) override { 401 Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
429 test::RtcpPacketParser parser; 402 test::RtcpPacketParser parser;
430 EXPECT_TRUE(parser.Parse(packet, length)); 403 EXPECT_TRUE(parser.Parse(packet, length));
431 EXPECT_EQ(kReceiverLocalVideoSsrc, parser.sender_ssrc()); 404 EXPECT_EQ(kReceiverLocalVideoSsrc, parser.sender_ssrc());
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3936 std::unique_ptr<VideoEncoder> encoder_; 3909 std::unique_ptr<VideoEncoder> encoder_;
3937 std::unique_ptr<VideoDecoder> decoder_; 3910 std::unique_ptr<VideoDecoder> decoder_;
3938 rtc::CriticalSection crit_; 3911 rtc::CriticalSection crit_;
3939 int recorded_frames_ GUARDED_BY(crit_); 3912 int recorded_frames_ GUARDED_BY(crit_);
3940 } test(this); 3913 } test(this);
3941 3914
3942 RunBaseTest(&test); 3915 RunBaseTest(&test);
3943 } 3916 }
3944 3917
3945 } // namespace webrtc 3918 } // namespace webrtc
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