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1 /* | 1 /* |
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2015 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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184 ALOGD("hardware_aec: %d", hardware_aec); | 184 ALOGD("hardware_aec: %d", hardware_aec); |
185 ALOGD("low_latency_output: %d", low_latency_output); | 185 ALOGD("low_latency_output: %d", low_latency_output); |
186 ALOGD("sample_rate: %d", sample_rate); | 186 ALOGD("sample_rate: %d", sample_rate); |
187 ALOGD("channels: %d", channels); | 187 ALOGD("channels: %d", channels); |
188 ALOGD("output_buffer_size: %d", output_buffer_size); | 188 ALOGD("output_buffer_size: %d", output_buffer_size); |
189 ALOGD("input_buffer_size: %d", input_buffer_size); | 189 ALOGD("input_buffer_size: %d", input_buffer_size); |
190 DCHECK(thread_checker_.CalledOnValidThread()); | 190 DCHECK(thread_checker_.CalledOnValidThread()); |
191 hardware_aec_ = hardware_aec; | 191 hardware_aec_ = hardware_aec; |
192 low_latency_playout_ = low_latency_output; | 192 low_latency_playout_ = low_latency_output; |
193 // TODO(henrika): add support for stereo output. | 193 // TODO(henrika): add support for stereo output. |
194 playout_parameters_.reset(sample_rate, channels, output_buffer_size); | 194 playout_parameters_.reset(sample_rate, channels, |
195 record_parameters_.reset(sample_rate, channels, input_buffer_size); | 195 static_cast<size_t>(output_buffer_size)); |
| 196 record_parameters_.reset(sample_rate, channels, |
| 197 static_cast<size_t>(input_buffer_size)); |
196 } | 198 } |
197 | 199 |
198 const AudioParameters& AudioManager::GetPlayoutAudioParameters() { | 200 const AudioParameters& AudioManager::GetPlayoutAudioParameters() { |
199 CHECK(playout_parameters_.is_valid()); | 201 CHECK(playout_parameters_.is_valid()); |
200 DCHECK(thread_checker_.CalledOnValidThread()); | 202 DCHECK(thread_checker_.CalledOnValidThread()); |
201 return playout_parameters_; | 203 return playout_parameters_; |
202 } | 204 } |
203 | 205 |
204 const AudioParameters& AudioManager::GetRecordAudioParameters() { | 206 const AudioParameters& AudioManager::GetRecordAudioParameters() { |
205 CHECK(record_parameters_.is_valid()); | 207 CHECK(record_parameters_.is_valid()); |
206 DCHECK(thread_checker_.CalledOnValidThread()); | 208 DCHECK(thread_checker_.CalledOnValidThread()); |
207 return record_parameters_; | 209 return record_parameters_; |
208 } | 210 } |
209 | 211 |
210 } // namespace webrtc | 212 } // namespace webrtc |
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