| /external/webrtc/webrtc/modules/audio_device/include/ |
| audio_device_defines.h | 150 frames_per_buffer_(0),
155 frames_per_buffer_(frames_per_buffer),
160 frames_per_buffer_ = frames_per_buffer;
173 size_t frames_per_buffer() const { return frames_per_buffer_; }
177 return frames_per_buffer_ * GetBytesPerFrame();
186 bool is_complete() const { return (is_valid() && (frames_per_buffer_ > 0)); }
193 return frames_per_buffer_ / (sample_rate_ / 1000.0);
198 return static_cast<double>(frames_per_buffer_) / (sample_rate_);
204 size_t frames_per_buffer_; member in class:webrtc::AudioParameters
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| /external/webrtc/webrtc/modules/audio_device/android/ |
| audio_record_jni.cc | 83 frames_per_buffer_(0),
139 frames_per_buffer_ = static_cast<size_t>(frames_per_buffer);
140 ALOGD("frames_per_buffer: %" PRIuS, frames_per_buffer_);
142 frames_per_buffer_ * kBytesPerFrame);
143 RTC_CHECK_EQ(frames_per_buffer_, audio_parameters_.frames_per_10ms_buffer());
250 frames_per_buffer_);
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| audio_track_jni.cc | 76 frames_per_buffer_(0),
227 frames_per_buffer_ = direct_buffer_capacity_in_bytes_ / kBytesPerFrame;
228 ALOGD("frames_per_buffer: %" PRIuS, frames_per_buffer_);
242 RTC_DCHECK_EQ(frames_per_buffer_, length / kBytesPerFrame);
248 int samples = audio_device_buffer_->RequestPlayoutData(frames_per_buffer_);
253 RTC_DCHECK_EQ(static_cast<size_t>(samples), frames_per_buffer_); local
|
| audio_record_jni.h | 150 size_t frames_per_buffer_; member in class:webrtc::AudioRecordJni
|
| audio_track_jni.h | 138 size_t frames_per_buffer_; member in class:webrtc::AudioTrackJni
|
| audio_device_unittest.cc | 162 : frames_per_buffer_(frames_per_buffer),
163 bytes_per_buffer_(frames_per_buffer_ * sizeof(int16_t)),
179 ASSERT_EQ(num_frames, frames_per_buffer_);
184 int16_t* memory = new int16_t[frames_per_buffer_];
202 ASSERT_EQ(num_frames, frames_per_buffer_);
240 const size_t frames_per_buffer_; member in class:webrtc::FifoAudioStream
256 frames_per_buffer_(frames_per_buffer),
257 bytes_per_buffer_(frames_per_buffer_ * sizeof(int16_t)),
265 ASSERT_EQ(num_frames, frames_per_buffer_);
287 ASSERT_EQ(num_frames, frames_per_buffer_);
360 const size_t frames_per_buffer_; member in class:webrtc::LatencyMeasuringAudioStream
[all...] |
| /external/webrtc/webrtc/modules/audio_device/ios/ |
| audio_device_unittest_ios.cc | 163 : frames_per_buffer_(frames_per_buffer),
164 bytes_per_buffer_(frames_per_buffer_ * sizeof(int16_t)),
178 ASSERT_EQ(num_frames, frames_per_buffer_);
183 int16_t* memory = new int16_t[frames_per_buffer_];
199 ASSERT_EQ(num_frames, frames_per_buffer_);
234 const size_t frames_per_buffer_; member in class:webrtc::FifoAudioStream
250 frames_per_buffer_(frames_per_buffer),
251 bytes_per_buffer_(frames_per_buffer_ * sizeof(int16_t)),
258 ASSERT_EQ(num_frames, frames_per_buffer_);
280 ASSERT_EQ(num_frames, frames_per_buffer_);
351 const size_t frames_per_buffer_; member in class:webrtc::LatencyMeasuringAudioStream
[all...] |
