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1 /* | 1 /* |
2 * Copyright 2017 The WebRTC Project Authors. All rights reserved. | 2 * Copyright 2017 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 | 10 |
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119 } // namespace | 119 } // namespace |
120 | 120 |
121 class RtpVideoStreamReceiverTest : public testing::Test { | 121 class RtpVideoStreamReceiverTest : public testing::Test { |
122 public: | 122 public: |
123 RtpVideoStreamReceiverTest() | 123 RtpVideoStreamReceiverTest() |
124 : config_(CreateConfig()), | 124 : config_(CreateConfig()), |
125 timing_(Clock::GetRealTimeClock()), | 125 timing_(Clock::GetRealTimeClock()), |
126 process_thread_(ProcessThread::Create("TestThread")) {} | 126 process_thread_(ProcessThread::Create("TestThread")) {} |
127 | 127 |
128 void SetUp() { | 128 void SetUp() { |
129 rtp_receive_statistics_ = | 129 rtp_video_stream_receiver_.reset(new RtpVideoStreamReceiver( |
130 rtc::WrapUnique(ReceiveStatistics::Create(Clock::GetRealTimeClock())); | |
131 rtp_video_stream_receiver_ = rtc::MakeUnique<RtpVideoStreamReceiver>( | |
132 &mock_transport_, nullptr, &packet_router_, &config_, | 130 &mock_transport_, nullptr, &packet_router_, &config_, |
133 rtp_receive_statistics_.get(), nullptr, process_thread_.get(), | 131 nullptr, process_thread_.get(), &mock_nack_sender_, |
134 &mock_nack_sender_, | |
135 &mock_key_frame_request_sender_, &mock_on_complete_frame_callback_, | 132 &mock_key_frame_request_sender_, &mock_on_complete_frame_callback_, |
136 &timing_); | 133 &timing_)); |
137 } | 134 } |
138 | 135 |
139 WebRtcRTPHeader GetDefaultPacket() { | 136 WebRtcRTPHeader GetDefaultPacket() { |
140 WebRtcRTPHeader packet; | 137 WebRtcRTPHeader packet; |
141 memset(&packet, 0, sizeof(packet)); | 138 memset(&packet, 0, sizeof(packet)); |
142 packet.type.Video.codec = kRtpVideoH264; | 139 packet.type.Video.codec = kRtpVideoH264; |
143 return packet; | 140 return packet; |
144 } | 141 } |
145 | 142 |
146 // TODO(Johan): refactor h264_sps_pps_tracker_unittests.cc to avoid duplicate | 143 // TODO(Johan): refactor h264_sps_pps_tracker_unittests.cc to avoid duplicate |
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192 webrtc::test::ScopedFieldTrials override_field_trials_{ | 189 webrtc::test::ScopedFieldTrials override_field_trials_{ |
193 kNewJitterBufferFieldTrialEnabled}; | 190 kNewJitterBufferFieldTrialEnabled}; |
194 VideoReceiveStream::Config config_; | 191 VideoReceiveStream::Config config_; |
195 MockNackSender mock_nack_sender_; | 192 MockNackSender mock_nack_sender_; |
196 MockKeyFrameRequestSender mock_key_frame_request_sender_; | 193 MockKeyFrameRequestSender mock_key_frame_request_sender_; |
197 MockTransport mock_transport_; | 194 MockTransport mock_transport_; |
198 MockOnCompleteFrameCallback mock_on_complete_frame_callback_; | 195 MockOnCompleteFrameCallback mock_on_complete_frame_callback_; |
199 PacketRouter packet_router_; | 196 PacketRouter packet_router_; |
200 VCMTiming timing_; | 197 VCMTiming timing_; |
201 std::unique_ptr<ProcessThread> process_thread_; | 198 std::unique_ptr<ProcessThread> process_thread_; |
202 std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_; | |
203 std::unique_ptr<RtpVideoStreamReceiver> rtp_video_stream_receiver_; | 199 std::unique_ptr<RtpVideoStreamReceiver> rtp_video_stream_receiver_; |
204 }; | 200 }; |
205 | 201 |
206 TEST_F(RtpVideoStreamReceiverTest, GenericKeyFrame) { | 202 TEST_F(RtpVideoStreamReceiverTest, GenericKeyFrame) { |
207 WebRtcRTPHeader rtp_header; | 203 WebRtcRTPHeader rtp_header; |
208 const std::vector<uint8_t> data({1, 2, 3, 4}); | 204 const std::vector<uint8_t> data({1, 2, 3, 4}); |
209 memset(&rtp_header, 0, sizeof(rtp_header)); | 205 memset(&rtp_header, 0, sizeof(rtp_header)); |
210 rtp_header.header.sequenceNumber = 1; | 206 rtp_header.header.sequenceNumber = 1; |
211 rtp_header.header.markerBit = 1; | 207 rtp_header.header.markerBit = 1; |
212 rtp_header.type.Video.is_first_packet_in_frame = true; | 208 rtp_header.type.Video.is_first_packet_in_frame = true; |
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456 rtp_video_stream_receiver_->AddSecondarySink(&secondary_sink); | 452 rtp_video_stream_receiver_->AddSecondarySink(&secondary_sink); |
457 EXPECT_DEATH(rtp_video_stream_receiver_->AddSecondarySink(&secondary_sink), | 453 EXPECT_DEATH(rtp_video_stream_receiver_->AddSecondarySink(&secondary_sink), |
458 ""); | 454 ""); |
459 | 455 |
460 // Test tear-down. | 456 // Test tear-down. |
461 rtp_video_stream_receiver_->RemoveSecondarySink(&secondary_sink); | 457 rtp_video_stream_receiver_->RemoveSecondarySink(&secondary_sink); |
462 } | 458 } |
463 #endif | 459 #endif |
464 | 460 |
465 } // namespace webrtc | 461 } // namespace webrtc |
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