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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> |
11 #include <map> | 11 #include <map> |
12 #include <sstream> | 12 #include <sstream> |
13 #include <string> | 13 #include <string> |
14 | 14 |
15 #include "testing/gtest/include/gtest/gtest.h" | 15 #include "testing/gtest/include/gtest/gtest.h" |
16 | 16 |
17 #include "webrtc/base/checks.h" | 17 #include "webrtc/base/checks.h" |
| 18 #include "webrtc/base/event.h" |
18 #include "webrtc/base/scoped_ptr.h" | 19 #include "webrtc/base/scoped_ptr.h" |
19 #include "webrtc/call.h" | 20 #include "webrtc/call.h" |
20 #include "webrtc/frame_callback.h" | 21 #include "webrtc/frame_callback.h" |
21 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" | 22 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" |
22 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h" | 23 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h" |
23 #include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h" | 24 #include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h" |
24 #include "webrtc/modules/video_coding/main/interface/video_coding_defines.h" | 25 #include "webrtc/modules/video_coding/main/interface/video_coding_defines.h" |
25 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" | 26 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" |
26 #include "webrtc/system_wrappers/interface/event_wrapper.h" | 27 #include "webrtc/system_wrappers/interface/event_wrapper.h" |
27 #include "webrtc/system_wrappers/interface/metrics.h" | 28 #include "webrtc/system_wrappers/interface/metrics.h" |
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1438 return observer_; | 1439 return observer_; |
1439 } | 1440 } |
1440 | 1441 |
1441 private: | 1442 private: |
1442 RtpExtensionHeaderObserver* observer_; | 1443 RtpExtensionHeaderObserver* observer_; |
1443 } tester; | 1444 } tester; |
1444 | 1445 |
1445 tester.RunTest(); | 1446 tester.RunTest(); |
1446 } | 1447 } |
1447 | 1448 |
| 1449 TEST_F(EndToEndTest, ReceivesTransportFeedback) { |
| 1450 static const int kExtensionId = 5; |
| 1451 |
| 1452 class TransportFeedbackObserver : public test::DirectTransport { |
| 1453 public: |
| 1454 TransportFeedbackObserver(rtc::Event* done_event) : done_(done_event) {} |
| 1455 virtual ~TransportFeedbackObserver() {} |
| 1456 |
| 1457 bool SendRtcp(const uint8_t* data, size_t length) override { |
| 1458 RTCPUtility::RTCPParserV2 parser(data, length, true); |
| 1459 EXPECT_TRUE(parser.IsValid()); |
| 1460 |
| 1461 RTCPUtility::RTCPPacketTypes packet_type = parser.Begin(); |
| 1462 while (packet_type != RTCPUtility::RTCPPacketTypes::kInvalid) { |
| 1463 if (packet_type == RTCPUtility::RTCPPacketTypes::kTransportFeedback) { |
| 1464 done_->Set(); |
| 1465 break; |
| 1466 } |
| 1467 packet_type = parser.Iterate(); |
| 1468 } |
| 1469 |
| 1470 return test::DirectTransport::SendRtcp(data, length); |
| 1471 } |
| 1472 |
| 1473 rtc::Event* done_; |
| 1474 }; |
| 1475 |
| 1476 class TransportFeedbackTester : public MultiStreamTest { |
| 1477 public: |
| 1478 TransportFeedbackTester() : done_(false, false) {} |
| 1479 virtual ~TransportFeedbackTester() {} |
| 1480 |
| 1481 protected: |
| 1482 void Wait() override { |
| 1483 EXPECT_TRUE(done_.Wait(CallTest::kDefaultTimeoutMs)); |
| 1484 } |
| 1485 |
| 1486 void UpdateSendConfig( |
| 1487 size_t stream_index, |
| 1488 VideoSendStream::Config* send_config, |
| 1489 VideoEncoderConfig* encoder_config, |
| 1490 test::FrameGeneratorCapturer** frame_generator) override { |
| 1491 send_config->rtp.extensions.push_back( |
| 1492 RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId)); |
| 1493 } |
| 1494 |
| 1495 void UpdateReceiveConfig( |
| 1496 size_t stream_index, |
| 1497 VideoReceiveStream::Config* receive_config) override { |
| 1498 receive_config->rtp.extensions.push_back( |
| 1499 RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId)); |
| 1500 } |
| 1501 |
| 1502 virtual test::DirectTransport* CreateReceiveTransport() { |
| 1503 return new TransportFeedbackObserver(&done_); |
| 1504 } |
| 1505 |
| 1506 private: |
| 1507 rtc::Event done_; |
| 1508 } tester; |
| 1509 tester.RunTest(); |
| 1510 } |
1448 TEST_F(EndToEndTest, ObserversEncodedFrames) { | 1511 TEST_F(EndToEndTest, ObserversEncodedFrames) { |
1449 class EncodedFrameTestObserver : public EncodedFrameObserver { | 1512 class EncodedFrameTestObserver : public EncodedFrameObserver { |
1450 public: | 1513 public: |
1451 EncodedFrameTestObserver() | 1514 EncodedFrameTestObserver() |
1452 : length_(0), | 1515 : length_(0), |
1453 frame_type_(kFrameEmpty), | 1516 frame_type_(kFrameEmpty), |
1454 called_(EventWrapper::Create()) {} | 1517 called_(EventWrapper::Create()) {} |
1455 virtual ~EncodedFrameTestObserver() {} | 1518 virtual ~EncodedFrameTestObserver() {} |
1456 | 1519 |
1457 virtual void EncodedFrameCallback(const EncodedFrame& encoded_frame) { | 1520 virtual void EncodedFrameCallback(const EncodedFrame& encoded_frame) { |
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3059 EXPECT_TRUE(default_receive_config.rtp.rtx.empty()) | 3122 EXPECT_TRUE(default_receive_config.rtp.rtx.empty()) |
3060 << "Enabling RTX requires rtpmap: rtx negotiation."; | 3123 << "Enabling RTX requires rtpmap: rtx negotiation."; |
3061 EXPECT_TRUE(default_receive_config.rtp.extensions.empty()) | 3124 EXPECT_TRUE(default_receive_config.rtp.extensions.empty()) |
3062 << "Enabling RTP extensions require negotiation."; | 3125 << "Enabling RTP extensions require negotiation."; |
3063 | 3126 |
3064 VerifyEmptyNackConfig(default_receive_config.rtp.nack); | 3127 VerifyEmptyNackConfig(default_receive_config.rtp.nack); |
3065 VerifyEmptyFecConfig(default_receive_config.rtp.fec); | 3128 VerifyEmptyFecConfig(default_receive_config.rtp.fec); |
3066 } | 3129 } |
3067 | 3130 |
3068 } // namespace webrtc | 3131 } // namespace webrtc |
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