| /external/webrtc/webrtc/modules/audio_device/android/ |
| opensles_player.cc | 45 bytes_per_buffer_(0),
246 bytes_per_buffer_ = audio_parameters_.GetBytesPerFrame() *
248 RTC_DCHECK_GE(bytes_per_buffer_, audio_parameters_.GetBytesPerBuffer());
249 ALOGD("native buffer size: %" PRIuS, bytes_per_buffer_);
254 bytes_per_buffer_,
450 ->Enqueue(simple_buffer_queue_, audio_ptr, bytes_per_buffer_);
|
| opensles_player.h | 148 size_t bytes_per_buffer_; member in class:webrtc::OpenSLESPlayer
|
| audio_device_unittest.cc | 163 bytes_per_buffer_(frames_per_buffer_ * sizeof(int16_t)),
187 bytes_per_buffer_); variable
206 memset(destination, 0, bytes_per_buffer_);
212 bytes_per_buffer_);
241 const size_t bytes_per_buffer_; member in class:webrtc::FifoAudioStream
257 bytes_per_buffer_(frames_per_buffer_ * sizeof(int16_t)),
270 memset(destination, 0, bytes_per_buffer_);
361 const size_t bytes_per_buffer_; member in class:webrtc::LatencyMeasuringAudioStream
[all...] |
| /external/webrtc/webrtc/modules/audio_device/ios/ |
| audio_device_unittest_ios.cc | 164 bytes_per_buffer_(frames_per_buffer_ * sizeof(int16_t)),
184 memcpy(static_cast<int16_t*>(&memory[0]), source, bytes_per_buffer_); variable
203 memset(destination, 0, bytes_per_buffer_);
207 memcpy(destination, static_cast<int16_t*>(&memory[0]), bytes_per_buffer_);
235 const size_t bytes_per_buffer_; member in class:webrtc::FifoAudioStream
251 bytes_per_buffer_(frames_per_buffer_ * sizeof(int16_t)),
263 memset(destination, 0, bytes_per_buffer_);
352 const size_t bytes_per_buffer_; member in class:webrtc::LatencyMeasuringAudioStream
[all...] |
