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

Issue 2523843002: Send audio and video codecs to RTPPayloadRegistry (Closed)
Patch Set: Change strcpy to strncpy Created 4 years 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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139 139
140 EXPECT_EQ(0, module1->SetSendingStatus(true)); 140 EXPECT_EQ(0, module1->SetSendingStatus(true));
141 141
142 CodecInst voice_codec; 142 CodecInst voice_codec;
143 voice_codec.pltype = 96; 143 voice_codec.pltype = 96;
144 voice_codec.plfreq = 8000; 144 voice_codec.plfreq = 8000;
145 voice_codec.rate = 64000; 145 voice_codec.rate = 64000;
146 memcpy(voice_codec.plname, "PCMU", 5); 146 memcpy(voice_codec.plname, "PCMU", 5);
147 147
148 EXPECT_EQ(0, module1->RegisterSendPayload(voice_codec)); 148 EXPECT_EQ(0, module1->RegisterSendPayload(voice_codec));
149 EXPECT_EQ(0, rtp_receiver1_->RegisterReceivePayload( 149 EXPECT_EQ(0, rtp_receiver1_->RegisterReceivePayload(voice_codec));
150 voice_codec.plname,
151 voice_codec.pltype,
152 voice_codec.plfreq,
153 voice_codec.channels,
154 (voice_codec.rate < 0) ? 0 : voice_codec.rate));
155 EXPECT_EQ(0, module2->RegisterSendPayload(voice_codec)); 150 EXPECT_EQ(0, module2->RegisterSendPayload(voice_codec));
156 EXPECT_EQ(0, rtp_receiver2_->RegisterReceivePayload( 151 EXPECT_EQ(0, rtp_receiver2_->RegisterReceivePayload(voice_codec));
157 voice_codec.plname,
158 voice_codec.pltype,
159 voice_codec.plfreq,
160 voice_codec.channels,
161 (voice_codec.rate < 0) ? 0 : voice_codec.rate));
162 152
163 // We need to send one RTP packet to get the RTCP packet to be accepted by 153 // We need to send one RTP packet to get the RTCP packet to be accepted by
164 // the receiving module. 154 // the receiving module.
165 // send RTP packet with the data "testtest" 155 // send RTP packet with the data "testtest"
166 const uint8_t test[9] = "testtest"; 156 const uint8_t test[9] = "testtest";
167 EXPECT_EQ(true, 157 EXPECT_EQ(true,
168 module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96, 0, -1, 158 module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96, 0, -1,
169 test, 8, nullptr, nullptr, nullptr)); 159 test, 8, nullptr, nullptr, nullptr));
170 } 160 }
171 161
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267 EXPECT_EQ(test_ssrc, report_blocks[0].sourceSSRC); 257 EXPECT_EQ(test_ssrc, report_blocks[0].sourceSSRC);
268 258
269 EXPECT_EQ(0u, report_blocks[0].cumulativeLost); 259 EXPECT_EQ(0u, report_blocks[0].cumulativeLost);
270 EXPECT_LT(0u, report_blocks[0].delaySinceLastSR); 260 EXPECT_LT(0u, report_blocks[0].delaySinceLastSR);
271 EXPECT_EQ(test_sequence_number, report_blocks[0].extendedHighSeqNum); 261 EXPECT_EQ(test_sequence_number, report_blocks[0].extendedHighSeqNum);
272 EXPECT_EQ(0u, report_blocks[0].fractionLost); 262 EXPECT_EQ(0u, report_blocks[0].fractionLost);
273 } 263 }
274 264
275 } // namespace 265 } // namespace
276 } // namespace webrtc 266 } // namespace webrtc
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