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Side by Side Diff: webrtc/modules/audio_device/android/audio_manager.cc

Issue 1316523002: Convert channel counts to size_t. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@master
Patch Set: Fix compile Created 4 years, 11 months ago
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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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