| /external/webrtc/webrtc/modules/audio_device/include/ |
| audio_device_defines.h | 152 AudioParameters(int sample_rate, size_t channels, size_t frames_per_buffer)
155 frames_per_buffer_(frames_per_buffer),
157 void reset(int sample_rate, size_t channels, size_t frames_per_buffer) {
160 frames_per_buffer_ = frames_per_buffer;
173 size_t frames_per_buffer() const { return frames_per_buffer_; } function in class:webrtc::AudioParameters
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| /external/webrtc/webrtc/modules/audio_device/android/ |
| audio_manager_unittest.cc | 87 playout_parameters_.frames_per_buffer(),
94 record_parameters_.frames_per_buffer(),
123 EXPECT_EQ(0U, params.frames_per_buffer());
143 EXPECT_EQ(kFramesPerBuffer, params.frames_per_buffer());
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| audio_record_jni.cc | 133 int frames_per_buffer = j_audio_record_->InitRecording( local
135 if (frames_per_buffer < 0) {
139 frames_per_buffer_ = static_cast<size_t>(frames_per_buffer);
140 ALOGD("frames_per_buffer: %" PRIuS, frames_per_buffer_);
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| audio_device_unittest.cc | 161 explicit FifoAudioStream(size_t frames_per_buffer)
162 : frames_per_buffer_(frames_per_buffer),
254 explicit LatencyMeasuringAudioStream(size_t frames_per_buffer)
256 frames_per_buffer_(frames_per_buffer),
[all...] |
| /external/webrtc/webrtc/modules/audio_device/ios/ |
| audio_device_unittest_ios.cc | 162 explicit FifoAudioStream(size_t frames_per_buffer)
163 : frames_per_buffer_(frames_per_buffer),
248 explicit LatencyMeasuringAudioStream(size_t frames_per_buffer)
250 frames_per_buffer_(frames_per_buffer),
[all...] |
| audio_device_ios.mm | 703 << playout_parameters_.frames_per_buffer();
[all...] |
