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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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207 ALOGD("sample_rate: %d", sample_rate); | 207 ALOGD("sample_rate: %d", sample_rate); |
208 ALOGD("channels: %d", channels); | 208 ALOGD("channels: %d", channels); |
209 ALOGD("output_buffer_size: %d", output_buffer_size); | 209 ALOGD("output_buffer_size: %d", output_buffer_size); |
210 ALOGD("input_buffer_size: %d", input_buffer_size); | 210 ALOGD("input_buffer_size: %d", input_buffer_size); |
211 RTC_DCHECK(thread_checker_.CalledOnValidThread()); | 211 RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
212 hardware_aec_ = hardware_aec; | 212 hardware_aec_ = hardware_aec; |
213 hardware_agc_ = hardware_agc; | 213 hardware_agc_ = hardware_agc; |
214 hardware_ns_ = hardware_ns; | 214 hardware_ns_ = hardware_ns; |
215 low_latency_playout_ = low_latency_output; | 215 low_latency_playout_ = low_latency_output; |
216 // TODO(henrika): add support for stereo output. | 216 // TODO(henrika): add support for stereo output. |
217 playout_parameters_.reset(sample_rate, channels, | 217 playout_parameters_.reset(sample_rate, static_cast<size_t>(channels), |
218 static_cast<size_t>(output_buffer_size)); | 218 static_cast<size_t>(output_buffer_size)); |
219 record_parameters_.reset(sample_rate, channels, | 219 record_parameters_.reset(sample_rate, static_cast<size_t>(channels), |
220 static_cast<size_t>(input_buffer_size)); | 220 static_cast<size_t>(input_buffer_size)); |
221 } | 221 } |
222 | 222 |
223 const AudioParameters& AudioManager::GetPlayoutAudioParameters() { | 223 const AudioParameters& AudioManager::GetPlayoutAudioParameters() { |
224 RTC_CHECK(playout_parameters_.is_valid()); | 224 RTC_CHECK(playout_parameters_.is_valid()); |
225 RTC_DCHECK(thread_checker_.CalledOnValidThread()); | 225 RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
226 return playout_parameters_; | 226 return playout_parameters_; |
227 } | 227 } |
228 | 228 |
229 const AudioParameters& AudioManager::GetRecordAudioParameters() { | 229 const AudioParameters& AudioManager::GetRecordAudioParameters() { |
230 RTC_CHECK(record_parameters_.is_valid()); | 230 RTC_CHECK(record_parameters_.is_valid()); |
231 RTC_DCHECK(thread_checker_.CalledOnValidThread()); | 231 RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
232 return record_parameters_; | 232 return record_parameters_; |
233 } | 233 } |
234 | 234 |
235 } // namespace webrtc | 235 } // namespace webrtc |
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