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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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149 AudioFrame tempAudioFrame; | 149 AudioFrame tempAudioFrame; |
150 tempAudioFrame.samples_per_channel_ = 0; | 150 tempAudioFrame.samples_per_channel_ = 0; |
151 if( incomingAudioFrame.num_channels_ == 2 && | 151 if( incomingAudioFrame.num_channels_ == 2 && |
152 !_moduleFile->IsStereo()) | 152 !_moduleFile->IsStereo()) |
153 { | 153 { |
154 // Recording mono but incoming audio is (interleaved) stereo. | 154 // Recording mono but incoming audio is (interleaved) stereo. |
155 tempAudioFrame.num_channels_ = 1; | 155 tempAudioFrame.num_channels_ = 1; |
156 tempAudioFrame.sample_rate_hz_ = incomingAudioFrame.sample_rate_hz_; | 156 tempAudioFrame.sample_rate_hz_ = incomingAudioFrame.sample_rate_hz_; |
157 tempAudioFrame.samples_per_channel_ = | 157 tempAudioFrame.samples_per_channel_ = |
158 incomingAudioFrame.samples_per_channel_; | 158 incomingAudioFrame.samples_per_channel_; |
159 for (uint16_t i = 0; | 159 for (size_t i = 0; |
160 i < (incomingAudioFrame.samples_per_channel_); i++) | 160 i < (incomingAudioFrame.samples_per_channel_); i++) |
161 { | 161 { |
162 // Sample value is the average of left and right buffer rounded to | 162 // Sample value is the average of left and right buffer rounded to |
163 // closest integer value. Note samples can be either 1 or 2 byte. | 163 // closest integer value. Note samples can be either 1 or 2 byte. |
164 tempAudioFrame.data_[i] = | 164 tempAudioFrame.data_[i] = |
165 ((incomingAudioFrame.data_[2 * i] + | 165 ((incomingAudioFrame.data_[2 * i] + |
166 incomingAudioFrame.data_[(2 * i) + 1] + 1) >> 1); | 166 incomingAudioFrame.data_[(2 * i) + 1] + 1) >> 1); |
167 } | 167 } |
168 } | 168 } |
169 else if( incomingAudioFrame.num_channels_ == 1 && | 169 else if( incomingAudioFrame.num_channels_ == 1 && |
170 _moduleFile->IsStereo()) | 170 _moduleFile->IsStereo()) |
171 { | 171 { |
172 // Recording stereo but incoming audio is mono. | 172 // Recording stereo but incoming audio is mono. |
173 tempAudioFrame.num_channels_ = 2; | 173 tempAudioFrame.num_channels_ = 2; |
174 tempAudioFrame.sample_rate_hz_ = incomingAudioFrame.sample_rate_hz_; | 174 tempAudioFrame.sample_rate_hz_ = incomingAudioFrame.sample_rate_hz_; |
175 tempAudioFrame.samples_per_channel_ = | 175 tempAudioFrame.samples_per_channel_ = |
176 incomingAudioFrame.samples_per_channel_; | 176 incomingAudioFrame.samples_per_channel_; |
177 for (uint16_t i = 0; | 177 for (size_t i = 0; |
178 i < (incomingAudioFrame.samples_per_channel_); i++) | 178 i < (incomingAudioFrame.samples_per_channel_); i++) |
179 { | 179 { |
180 // Duplicate sample to both channels | 180 // Duplicate sample to both channels |
181 tempAudioFrame.data_[2*i] = | 181 tempAudioFrame.data_[2*i] = |
182 incomingAudioFrame.data_[i]; | 182 incomingAudioFrame.data_[i]; |
183 tempAudioFrame.data_[2*i+1] = | 183 tempAudioFrame.data_[2*i+1] = |
184 incomingAudioFrame.data_[i]; | 184 incomingAudioFrame.data_[i]; |
185 } | 185 } |
186 } | 186 } |
187 | 187 |
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203 { | 203 { |
204 if (_audioEncoder.Encode(*ptrAudioFrame, _audioBuffer, | 204 if (_audioEncoder.Encode(*ptrAudioFrame, _audioBuffer, |
205 encodedLenInBytes) == -1) | 205 encodedLenInBytes) == -1) |
206 { | 206 { |
207 LOG(LS_WARNING) << "RecordAudioToFile() codec " | 207 LOG(LS_WARNING) << "RecordAudioToFile() codec " |
208 << codec_info_.plname | 208 << codec_info_.plname |
209 << " not supported or failed to encode stream."; | 209 << " not supported or failed to encode stream."; |
210 return -1; | 210 return -1; |
211 } | 211 } |
212 } else { | 212 } else { |
213 int outLen = 0; | 213 size_t outLen = 0; |
214 _audioResampler.ResetIfNeeded(ptrAudioFrame->sample_rate_hz_, | 214 _audioResampler.ResetIfNeeded(ptrAudioFrame->sample_rate_hz_, |
215 codec_info_.plfreq, | 215 codec_info_.plfreq, |
216 ptrAudioFrame->num_channels_); | 216 ptrAudioFrame->num_channels_); |
217 _audioResampler.Push(ptrAudioFrame->data_, | 217 _audioResampler.Push(ptrAudioFrame->data_, |
218 ptrAudioFrame->samples_per_channel_ * | 218 ptrAudioFrame->samples_per_channel_ * |
219 ptrAudioFrame->num_channels_, | 219 ptrAudioFrame->num_channels_, |
220 (int16_t*)_audioBuffer, | 220 (int16_t*)_audioBuffer, |
221 MAX_AUDIO_BUFFER_IN_BYTES, outLen); | 221 MAX_AUDIO_BUFFER_IN_BYTES, outLen); |
222 encodedLenInBytes = outLen * sizeof(int16_t); | 222 encodedLenInBytes = outLen * sizeof(int16_t); |
223 } | 223 } |
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259 codecInst = codec_info_; | 259 codecInst = codec_info_; |
260 return 0; | 260 return 0; |
261 } | 261 } |
262 | 262 |
263 int32_t FileRecorderImpl::WriteEncodedAudioData(const int8_t* audioBuffer, | 263 int32_t FileRecorderImpl::WriteEncodedAudioData(const int8_t* audioBuffer, |
264 size_t bufferLength) | 264 size_t bufferLength) |
265 { | 265 { |
266 return _moduleFile->IncomingAudioData(audioBuffer, bufferLength); | 266 return _moduleFile->IncomingAudioData(audioBuffer, bufferLength); |
267 } | 267 } |
268 } // namespace webrtc | 268 } // namespace webrtc |
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