| /external/webrtc/modules/audio_processing/aec3/ |
| D | echo_remover.cc | 126 std::vector<std::vector<std::vector<float>>>* linear_output,
239 std::vector<std::vector<std::vector<float>>>* linear_output, in ProcessCapture() argument
373 if (linear_output) { in ProcessCapture()
374 RTC_DCHECK_GE(1, linear_output->size()); in ProcessCapture()
375 RTC_DCHECK_EQ(num_capture_channels_, linear_output[0].size()); in ProcessCapture()
377 RTC_DCHECK_EQ(kBlockSize, (*linear_output)[0][ch].size()); in ProcessCapture()
378 std::copy(e[ch].begin(), e[ch].end(), (*linear_output)[0][ch].begin()); in ProcessCapture()
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| D | echo_canceller3.cc | 107 AudioBuffer* linear_output, in ProcessCaptureFrameContent() argument
123 if (linear_output) { in ProcessCaptureFrameContent()
127 FillSubFrameView(linear_output, sub_frame_index, in ProcessCaptureFrameContent()
138 if (linear_output) { in ProcessCaptureFrameContent()
740 AudioBuffer* linear_output, in ProcessCapture() argument
750 if (linear_output && !linear_output_framer_) { in ProcessCapture()
773 ProcessCaptureFrameContent(linear_output, capture, level_change, in ProcessCapture()
780 ProcessCaptureFrameContent(linear_output, capture, level_change, in ProcessCapture()
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| D | block_processor.cc | 55 std::vector<std::vector<std::vector<float>>>* linear_output,
109 std::vector<std::vector<std::vector<float>>>* linear_output, in ProcessCapture() argument
196 render_buffer_->GetRenderBuffer(), linear_output, capture_block); in ProcessCapture()
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| D | echo_remover.h | 45 std::vector<std::vector<std::vector<float>>>* linear_output,
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| D | block_processor.h | 62 std::vector<std::vector<std::vector<float>>>* linear_output,
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| D | echo_canceller3.h | 114 AudioBuffer* linear_output,
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| D | echo_canceller3_unittest.cc | 116 std::vector<std::vector<std::vector<float>>>* linear_output, in ProcessCapture() argument
142 std::vector<std::vector<std::vector<float>>>* linear_output, in ProcessCapture() argument
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| /external/webrtc/modules/audio_processing/include/ |
| D | mock_audio_processing.h | 62 AudioBuffer* linear_output,
137 ((rtc::ArrayView<std::array<float, 160>> linear_output)),
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| D | audio_processing.h | 576 rtc::ArrayView<std::array<float, 160>> linear_output) const = 0;
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| /external/libgsm/src/ |
| D | toast_lin.c | 21 int linear_output P1((buf), gsm_signal * buf)
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| D | toast.c | 89 linear_output
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| /external/webrtc/modules/audio_processing/test/ |
| D | echo_control_mock.h | 32 AudioBuffer* linear_output,
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| /external/webrtc/modules/audio_processing/aec3/mock/ |
| D | mock_echo_remover.h | 36 std::vector<std::vector<std::vector<float>>>* linear_output,
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| D | mock_block_processor.h | 31 std::vector<std::vector<std::vector<float>>>* linear_output,
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| /external/webrtc/api/audio/ |
| D | echo_control.h | 36 AudioBuffer* linear_output,
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| /external/libgsm/inc/ |
| D | toast.h | 111 extern int linear_input P((gsm_signal*)), linear_output P((gsm_signal *));
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| /external/tensorflow/tensorflow/python/keras/premade/ |
| D | wide_deep.py | 98 linear_output = self.linear_model(linear_inputs)
106 output = nest.map_structure(lambda x, y: (x + y), linear_output, dnn_output)
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| /external/webrtc/modules/audio_processing/ |
| D | audio_processing_impl.cc | 1480 rtc::ArrayView<std::array<float, 160>> linear_output) const { in GetLinearAecOutput()
1487 RTC_DCHECK_EQ(linear_output.size(), linear_aec_buffer->num_channels()); in GetLinearAecOutput()
1490 RTC_DCHECK_EQ(linear_output[ch].size(), linear_aec_buffer->num_frames()); in GetLinearAecOutput()
1495 linear_output[ch].begin()); in GetLinearAecOutput()
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| D | audio_processing_impl.h | 98 rtc::ArrayView<std::array<float, 160>> linear_output) const override;
|