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

Issue 1985743002: Propagate muted parameter to VoE::Channel (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Resurrect the PlayoutData10Ms(int, AudioFrame*) method Created 4 years, 7 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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201 _payloadSizeBytes, &_nextTime); 201 _payloadSizeBytes, &_nextTime);
202 if (_realPayloadSizeBytes == 0 && _rtpStream->EndOfFile()) { 202 if (_realPayloadSizeBytes == 0 && _rtpStream->EndOfFile()) {
203 _firstTime = true; 203 _firstTime = true;
204 } 204 }
205 } 205 }
206 return true; 206 return true;
207 } 207 }
208 208
209 bool Receiver::PlayoutData() { 209 bool Receiver::PlayoutData() {
210 AudioFrame audioFrame; 210 AudioFrame audioFrame;
211 211 bool muted;
212 int32_t ok =_acm->PlayoutData10Ms(_frequency, &audioFrame); 212 int32_t ok = _acm->PlayoutData10Ms(_frequency, &audioFrame, &muted);
213 if (muted) {
214 ADD_FAILURE();
215 return false;
216 }
213 EXPECT_EQ(0, ok); 217 EXPECT_EQ(0, ok);
214 if (ok < 0){ 218 if (ok < 0){
215 return false; 219 return false;
216 } 220 }
217 if (_playoutLengthSmpls == 0) { 221 if (_playoutLengthSmpls == 0) {
218 return false; 222 return false;
219 } 223 }
220 _pcmFile.Write10MsData(audioFrame.data_, 224 _pcmFile.Write10MsData(audioFrame.data_,
221 audioFrame.samples_per_channel_ * audioFrame.num_channels_); 225 audioFrame.samples_per_channel_ * audioFrame.num_channels_);
222 return true; 226 return true;
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346 if (acm->SendCodec()) { 350 if (acm->SendCodec()) {
347 _sender.Run(); 351 _sender.Run();
348 } 352 }
349 _sender.Teardown(); 353 _sender.Teardown();
350 rtpFile.Close(); 354 rtpFile.Close();
351 355
352 return fileName; 356 return fileName;
353 } 357 }
354 358
355 } // namespace webrtc 359 } // namespace webrtc
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