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Side by Side Diff: webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.cc

Issue 1373903003: Unify newapi::RtcpMode and RTCPMethod. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: enum class Created 5 years, 2 months ago
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1 /* 1 /*
2 * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2011 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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120 throw "_rtp->RegisterIncomingRTPCallback"; 120 throw "_rtp->RegisterIncomingRTPCallback";
121 exit(1); 121 exit(1);
122 } 122 }
123 123
124 if (_rtp->RegisterIncomingDataCallback(this) != 0) 124 if (_rtp->RegisterIncomingDataCallback(this) != 0)
125 { 125 {
126 throw "_rtp->RegisterIncomingRTPCallback"; 126 throw "_rtp->RegisterIncomingRTPCallback";
127 exit(1); 127 exit(1);
128 } 128 }
129 129
130 if (_rtp->SetRTCPStatus(kRtcpNonCompound) != 0) 130 if (_rtp->SetRTCPStatus(RtcpMode::REDUCED_SIZE) != 0)
131 { 131 {
132 throw "_rtp->SetRTCPStatus"; 132 throw "_rtp->SetRTCPStatus";
133 exit(1); 133 exit(1);
134 } 134 }
135 135
136 if (_rtp->SetTMMBRStatus(true) != 0) 136 if (_rtp->SetTMMBRStatus(true) != 0)
137 { 137 {
138 throw "_rtp->SetTMMBRStatus"; 138 throw "_rtp->SetTMMBRStatus";
139 exit(1); 139 exit(1);
140 } 140 }
(...skipping 195 matching lines...) Expand 10 before | Expand all | Expand 10 after
336 throw "_rtp->RegisterSendPayload"; 336 throw "_rtp->RegisterSendPayload";
337 exit(1); 337 exit(1);
338 } 338 }
339 339
340 if (_rtp->RegisterIncomingVideoCallback(this) != 0) 340 if (_rtp->RegisterIncomingVideoCallback(this) != 0)
341 { 341 {
342 throw "_rtp->RegisterIncomingVideoCallback"; 342 throw "_rtp->RegisterIncomingVideoCallback";
343 exit(1); 343 exit(1);
344 } 344 }
345 345
346 if (_rtp->SetRTCPStatus(kRtcpNonCompound) != 0) 346 if (_rtp->SetRTCPStatus(RtcpMode::REDUCED_SIZE) != 0)
347 { 347 {
348 throw "_rtp->SetRTCPStatus"; 348 throw "_rtp->SetRTCPStatus";
349 exit(1); 349 exit(1);
350 } 350 }
351 351
352 if (_rtp->SetSendBitrate(startBitrateKbps*1000, 0, MAX_BITRATE_KBPS) != 0) 352 if (_rtp->SetSendBitrate(startBitrateKbps*1000, 0, MAX_BITRATE_KBPS) != 0)
353 { 353 {
354 throw "_rtp->SetSendBitrate"; 354 throw "_rtp->SetSendBitrate";
355 exit(1); 355 exit(1);
356 } 356 }
(...skipping 47 matching lines...) Expand 10 before | Expand all | Expand 10 after
404 404
405 if (_sendRecCB) 405 if (_sendRecCB)
406 { 406 {
407 _sendRecCB->OnOnNetworkChanged(maxBitrateBps, 407 _sendRecCB->OnOnNetworkChanged(maxBitrateBps,
408 fractionLost, 408 fractionLost,
409 roundTripTimeMs, 409 roundTripTimeMs,
410 bwEstimateKbitMin, 410 bwEstimateKbitMin,
411 bwEstimateKbitMax); 411 bwEstimateKbitMax);
412 } 412 }
413 } 413 }
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