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

Issue 2682073003: Revert Make the new jitter buffer the default jitter buffer. (Closed)
Patch Set: Created 3 years, 10 months ago
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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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97 97
98 class RtpStreamReceiverTest : public testing::Test { 98 class RtpStreamReceiverTest : public testing::Test {
99 public: 99 public:
100 RtpStreamReceiverTest() 100 RtpStreamReceiverTest()
101 : config_(CreateConfig()), 101 : config_(CreateConfig()),
102 timing_(Clock::GetRealTimeClock()), 102 timing_(Clock::GetRealTimeClock()),
103 process_thread_(ProcessThread::Create("TestThread")) {} 103 process_thread_(ProcessThread::Create("TestThread")) {}
104 104
105 void SetUp() { 105 void SetUp() {
106 rtp_stream_receiver_.reset(new RtpStreamReceiver( 106 rtp_stream_receiver_.reset(new RtpStreamReceiver(
107 &mock_transport_, nullptr, &packet_router_, nullptr, &config_, 107 nullptr, &mock_transport_, nullptr, &packet_router_,
108 nullptr, process_thread_.get(), &mock_nack_sender_, 108 nullptr, &config_, nullptr, process_thread_.get(),
109 &mock_key_frame_request_sender_, &mock_on_complete_frame_callback_, 109 &mock_nack_sender_, &mock_key_frame_request_sender_,
110 &timing_)); 110 &mock_on_complete_frame_callback_, &timing_));
111 } 111 }
112 112
113 WebRtcRTPHeader GetDefaultPacket() { 113 WebRtcRTPHeader GetDefaultPacket() {
114 WebRtcRTPHeader packet; 114 WebRtcRTPHeader packet;
115 memset(&packet, 0, sizeof(packet)); 115 memset(&packet, 0, sizeof(packet));
116 packet.type.Video.codec = kRtpVideoH264; 116 packet.type.Video.codec = kRtpVideoH264;
117 return packet; 117 return packet;
118 } 118 }
119 119
120 // TODO(Johan): refactor h264_sps_pps_tracker_unittests.cc to avoid duplicate 120 // TODO(Johan): refactor h264_sps_pps_tracker_unittests.cc to avoid duplicate
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290 mock_on_complete_frame_callback_.AppendExpectedBitstream( 290 mock_on_complete_frame_callback_.AppendExpectedBitstream(
291 kH264StartCode, sizeof(kH264StartCode)); 291 kH264StartCode, sizeof(kH264StartCode));
292 mock_on_complete_frame_callback_.AppendExpectedBitstream(data.data(), 292 mock_on_complete_frame_callback_.AppendExpectedBitstream(data.data(),
293 data.size()); 293 data.size());
294 EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame(_)); 294 EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame(_));
295 rtp_stream_receiver_->OnReceivedPayloadData(data.data(), data.size(), 295 rtp_stream_receiver_->OnReceivedPayloadData(data.data(), data.size(),
296 &idr_packet); 296 &idr_packet);
297 } 297 }
298 298
299 } // namespace webrtc 299 } // namespace webrtc
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