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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" | |
19 #include "webrtc/base/scoped_ptr.h" | 18 #include "webrtc/base/scoped_ptr.h" |
20 #include "webrtc/call.h" | 19 #include "webrtc/call.h" |
21 #include "webrtc/frame_callback.h" | 20 #include "webrtc/frame_callback.h" |
22 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" | 21 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" |
23 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h" | 22 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h" |
24 #include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h" | 23 #include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h" |
25 #include "webrtc/modules/video_coding/main/interface/video_coding_defines.h" | 24 #include "webrtc/modules/video_coding/main/interface/video_coding_defines.h" |
26 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" | 25 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" |
27 #include "webrtc/system_wrappers/interface/event_wrapper.h" | 26 #include "webrtc/system_wrappers/interface/event_wrapper.h" |
28 #include "webrtc/system_wrappers/interface/metrics.h" | 27 #include "webrtc/system_wrappers/interface/metrics.h" |
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1439 return observer_; | 1438 return observer_; |
1440 } | 1439 } |
1441 | 1440 |
1442 private: | 1441 private: |
1443 RtpExtensionHeaderObserver* observer_; | 1442 RtpExtensionHeaderObserver* observer_; |
1444 } tester; | 1443 } tester; |
1445 | 1444 |
1446 tester.RunTest(); | 1445 tester.RunTest(); |
1447 } | 1446 } |
1448 | 1447 |
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 } | |
1511 TEST_F(EndToEndTest, ObserversEncodedFrames) { | 1448 TEST_F(EndToEndTest, ObserversEncodedFrames) { |
1512 class EncodedFrameTestObserver : public EncodedFrameObserver { | 1449 class EncodedFrameTestObserver : public EncodedFrameObserver { |
1513 public: | 1450 public: |
1514 EncodedFrameTestObserver() | 1451 EncodedFrameTestObserver() |
1515 : length_(0), | 1452 : length_(0), |
1516 frame_type_(kFrameEmpty), | 1453 frame_type_(kFrameEmpty), |
1517 called_(EventWrapper::Create()) {} | 1454 called_(EventWrapper::Create()) {} |
1518 virtual ~EncodedFrameTestObserver() {} | 1455 virtual ~EncodedFrameTestObserver() {} |
1519 | 1456 |
1520 virtual void EncodedFrameCallback(const EncodedFrame& encoded_frame) { | 1457 virtual void EncodedFrameCallback(const EncodedFrame& encoded_frame) { |
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3122 EXPECT_TRUE(default_receive_config.rtp.rtx.empty()) | 3059 EXPECT_TRUE(default_receive_config.rtp.rtx.empty()) |
3123 << "Enabling RTX requires rtpmap: rtx negotiation."; | 3060 << "Enabling RTX requires rtpmap: rtx negotiation."; |
3124 EXPECT_TRUE(default_receive_config.rtp.extensions.empty()) | 3061 EXPECT_TRUE(default_receive_config.rtp.extensions.empty()) |
3125 << "Enabling RTP extensions require negotiation."; | 3062 << "Enabling RTP extensions require negotiation."; |
3126 | 3063 |
3127 VerifyEmptyNackConfig(default_receive_config.rtp.nack); | 3064 VerifyEmptyNackConfig(default_receive_config.rtp.nack); |
3128 VerifyEmptyFecConfig(default_receive_config.rtp.fec); | 3065 VerifyEmptyFecConfig(default_receive_config.rtp.fec); |
3129 } | 3066 } |
3130 | 3067 |
3131 } // namespace webrtc | 3068 } // namespace webrtc |
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