| /external/webrtc/modules/audio_device/include/ |
| D | audio_device_defines.h | 86 AudioParameters(int sample_rate, size_t channels, size_t frames_per_buffer) in AudioParameters() argument
89 frames_per_buffer_(frames_per_buffer), in AudioParameters()
91 void reset(int sample_rate, size_t channels, size_t frames_per_buffer) { in reset() argument
94 frames_per_buffer_ = frames_per_buffer; in reset()
107 size_t frames_per_buffer() const { return frames_per_buffer_; } in frames_per_buffer() function
139 ss << ", frames_per_buffer=" << frames_per_buffer(); in ToString()
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| /external/webrtc/modules/audio_device/android/ |
| D | audio_manager_unittest.cc | 158 playout_parameters_.frames_per_buffer(), in TEST_F()
166 record_parameters_.frames_per_buffer(), in TEST_F()
180 EXPECT_EQ(playout_parameters_.frames_per_buffer(), in TEST_F()
181 record_parameters_.frames_per_buffer()); in TEST_F()
210 EXPECT_EQ(0U, params.frames_per_buffer()); in TEST_F()
230 EXPECT_EQ(kFramesPerBuffer, params.frames_per_buffer()); in TEST_F()
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| D | audio_record_jni.cc | 142 int frames_per_buffer = j_audio_record_->InitRecording( in InitRecording() local
144 if (frames_per_buffer < 0) { in InitRecording()
149 frames_per_buffer_ = static_cast<size_t>(frames_per_buffer); in InitRecording()
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| D | opensles_recorder.cc | 338 audio_parameters_.frames_per_buffer()); in AllocateDataBuffers()
348 audio_parameters_.frames_per_buffer() * audio_parameters_.channels(); in AllocateDataBuffers()
379 audio_parameters_.frames_per_buffer() * audio_parameters_.channels()), in ReadBufferQueue()
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| D | opensles_player.cc | 216 audio_parameters_.frames_per_buffer() * audio_parameters_.channels(); in AllocateDataBuffers()
411 audio_parameters_.frames_per_buffer() * in EnqueuePlayoutData()
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| D | audio_device_unittest.cc | 162 explicit FifoAudioStream(size_t frames_per_buffer) in FifoAudioStream() argument
163 : frames_per_buffer_(frames_per_buffer), in FifoAudioStream()
247 explicit LatencyMeasuringAudioStream(size_t frames_per_buffer) in LatencyMeasuringAudioStream() argument
248 : frames_per_buffer_(frames_per_buffer), in LatencyMeasuringAudioStream()
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| /external/webrtc/sdk/android/src/jni/audio_device/ |
| D | audio_record_jni.cc | 112 int frames_per_buffer = Java_WebRtcAudioRecord_initRecording( in InitRecording() local
115 if (frames_per_buffer < 0) { in InitRecording()
120 frames_per_buffer_ = static_cast<size_t>(frames_per_buffer); in InitRecording()
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| D | opensles_recorder.cc | 349 audio_parameters_.frames_per_buffer()); in AllocateDataBuffers()
359 audio_parameters_.frames_per_buffer() * audio_parameters_.channels(); in AllocateDataBuffers()
390 audio_parameters_.frames_per_buffer() * audio_parameters_.channels()), in ReadBufferQueue()
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| D | opensles_player.cc | 225 audio_parameters_.frames_per_buffer() * audio_parameters_.channels(); in AllocateDataBuffers()
420 audio_parameters_.frames_per_buffer() * in EnqueuePlayoutData()
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| /external/webrtc/sdk/android/native_unittests/audio_device/ |
| D | audio_device_unittest.cc | 159 explicit FifoAudioStream(size_t frames_per_buffer) in FifoAudioStream() argument
160 : frames_per_buffer_(frames_per_buffer), in FifoAudioStream()
244 explicit LatencyMeasuringAudioStream(size_t frames_per_buffer) in LatencyMeasuringAudioStream() argument
245 : frames_per_buffer_(frames_per_buffer), in LatencyMeasuringAudioStream()
964 EXPECT_EQ(output_parameters_.frames_per_buffer(), in TEST_F()
965 input_parameters_.frames_per_buffer()); in TEST_F()
977 output_parameters_.frames_per_buffer(), in TEST_F()
985 input_parameters_.frames_per_buffer(), in TEST_F()
1025 EXPECT_EQ(0U, params.frames_per_buffer()); in TEST_F()
1045 EXPECT_EQ(kFramesPerBuffer, params.frames_per_buffer()); in TEST_F()
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| /external/webrtc/modules/audio_device/win/ |
| D | core_audio_base_win.cc | 510 if (preferred_frames_per_buffer % params.frames_per_buffer()) { in Init()
511 RTC_LOG(WARNING) << "Buffer size of " << params.frames_per_buffer() in Init()
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| D | core_audio_utility_win.cc | 632 const size_t frames_per_buffer = in GetPreferredAudioParametersInternal() local
635 AudioParameters audio_params(sample_rate, channels, frames_per_buffer); in GetPreferredAudioParametersInternal()
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| /external/webrtc/sdk/objc/native/src/audio/ |
| D | audio_device_ios.mm | 574 const size_t current_frames_per_buffer = playout_parameters_.frames_per_buffer();
576 " Session sample rate: %f frames_per_buffer: %lu\n"
577 " ADM sample rate: %f frames_per_buffer: %lu",
716 RTC_LOG(LS_INFO) << " frames per I/O buffer: " << playout_parameters_.frames_per_buffer();
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| /external/webrtc/modules/audio_device/ |
| D | audio_device_unittest.cc | 385 EXPECT_EQ(samples_per_channel, record_parameters_.frames_per_buffer()); in RealRecordedDataIsAvailable()
425 EXPECT_EQ(samples_per_channel, playout_parameters_.frames_per_buffer()); in RealNeedMorePlayData()
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