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Side by Side Diff: webrtc/modules/audio_coding/test/EncodeDecodeTest.cc

Issue 2750783004: Add mute state field to AudioFrame. (Closed)
Patch Set: Update new usages of AudioFrame::data_ Created 3 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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216 ADD_FAILURE(); 216 ADD_FAILURE();
217 return false; 217 return false;
218 } 218 }
219 EXPECT_EQ(0, ok); 219 EXPECT_EQ(0, ok);
220 if (ok < 0){ 220 if (ok < 0){
221 return false; 221 return false;
222 } 222 }
223 if (_playoutLengthSmpls == 0) { 223 if (_playoutLengthSmpls == 0) {
224 return false; 224 return false;
225 } 225 }
226 _pcmFile.Write10MsData(audioFrame.data_, 226 _pcmFile.Write10MsData(audioFrame.data(),
227 audioFrame.samples_per_channel_ * audioFrame.num_channels_); 227 audioFrame.samples_per_channel_ * audioFrame.num_channels_);
228 return true; 228 return true;
229 } 229 }
230 230
231 void Receiver::Run() { 231 void Receiver::Run() {
232 uint8_t counter500Ms = 50; 232 uint8_t counter500Ms = 50;
233 uint32_t clock = 0; 233 uint32_t clock = 0;
234 234
235 while (counter500Ms > 0) { 235 while (counter500Ms > 0) {
236 if (clock == 0 || clock >= _nextTime) { 236 if (clock == 0 || clock >= _nextTime) {
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352 if (acm->SendCodec()) { 352 if (acm->SendCodec()) {
353 _sender.Run(); 353 _sender.Run();
354 } 354 }
355 _sender.Teardown(); 355 _sender.Teardown();
356 rtpFile.Close(); 356 rtpFile.Close();
357 357
358 return fileName; 358 return fileName;
359 } 359 }
360 360
361 } // namespace webrtc 361 } // namespace webrtc
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