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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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