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Side by Side Diff: webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc

Issue 2064523002: GN: Add video_engine_tests (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: audio_receive_stream_unittest compile Created 4 years, 6 months ago
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1 /* 1 /*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2012 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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62 62
63 bool LoopBackTransport::SendRtcp(const uint8_t* data, size_t len) { 63 bool LoopBackTransport::SendRtcp(const uint8_t* data, size_t len) {
64 if (rtp_rtcp_module_->IncomingRtcpPacket((const uint8_t*)data, len) < 0) { 64 if (rtp_rtcp_module_->IncomingRtcpPacket((const uint8_t*)data, len) < 0) {
65 return false; 65 return false;
66 } 66 }
67 return true; 67 return true;
68 } 68 }
69 69
70 int32_t TestRtpReceiver::OnReceivedPayloadData( 70 int32_t TestRtpReceiver::OnReceivedPayloadData(
71 const uint8_t* payload_data, 71 const uint8_t* payload_data,
72 const size_t payload_size, 72 size_t payload_size,
73 const webrtc::WebRtcRTPHeader* rtp_header) { 73 const webrtc::WebRtcRTPHeader* rtp_header) {
74 EXPECT_LE(payload_size, sizeof(payload_data_)); 74 EXPECT_LE(payload_size, sizeof(payload_data_));
75 memcpy(payload_data_, payload_data, payload_size); 75 memcpy(payload_data_, payload_data, payload_size);
76 memcpy(&rtp_header_, rtp_header, sizeof(rtp_header_)); 76 memcpy(&rtp_header_, rtp_header, sizeof(rtp_header_));
77 payload_size_ = payload_size; 77 payload_size_ = payload_size;
78 return 0; 78 return 0;
79 } 79 }
80 80
81 class RtpRtcpAPITest : public ::testing::Test { 81 class RtpRtcpAPITest : public ::testing::Test {
82 protected: 82 protected:
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178 rtx_header.payloadType = kRtxPayloadType; 178 rtx_header.payloadType = kRtxPayloadType;
179 EXPECT_TRUE(rtp_payload_registry_->IsRtx(rtx_header)); 179 EXPECT_TRUE(rtp_payload_registry_->IsRtx(rtx_header));
180 rtx_header.ssrc = 0; 180 rtx_header.ssrc = 0;
181 EXPECT_FALSE(rtp_payload_registry_->IsRtx(rtx_header)); 181 EXPECT_FALSE(rtp_payload_registry_->IsRtx(rtx_header));
182 rtx_header.ssrc = kRtxSsrc; 182 rtx_header.ssrc = kRtxSsrc;
183 rtx_header.payloadType = 0; 183 rtx_header.payloadType = 0;
184 EXPECT_TRUE(rtp_payload_registry_->IsRtx(rtx_header)); 184 EXPECT_TRUE(rtp_payload_registry_->IsRtx(rtx_header));
185 } 185 }
186 186
187 } // namespace webrtc 187 } // namespace webrtc
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