/external/webrtc/webrtc/common_audio/ |
H A D | blocker.h | 29 size_t num_input_channels, 68 size_t num_input_channels, 76 size_t num_input_channels,
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H A D | blocker.cc | 103 size_t num_input_channels, 110 num_input_channels_(num_input_channels), 169 size_t num_input_channels, 173 RTC_CHECK_EQ(num_input_channels, num_input_channels_); 176 input_buffer_.Write(input, num_input_channels, chunk_size_); 181 input_buffer_.Read(input_block_.channels(), num_input_channels, 101 Blocker(size_t chunk_size, size_t block_size, size_t num_input_channels, size_t num_output_channels, const float* window, size_t shift_amount, BlockerCallback* callback) argument 167 ProcessChunk(const float* const* input, size_t chunk_size, size_t num_input_channels, size_t num_output_channels, float* const* output) argument
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H A D | lapped_transform.cc | 24 size_t num_input_channels, 27 RTC_CHECK_EQ(num_input_channels, parent_->num_in_channels_); 31 for (size_t i = 0; i < num_input_channels; ++i) { 42 num_input_channels, 22 ProcessBlock(const float* const* input, size_t num_frames, size_t num_input_channels, size_t num_output_channels, float* const* output) argument
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H A D | lapped_transform.h | 98 size_t num_input_channels,
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H A D | blocker_unittest.cc | 23 size_t num_input_channels, 39 size_t num_input_channels, 66 size_t num_input_channels, 71 CopyTo(input_chunk, 0, start, num_input_channels, chunk_size, input); 74 num_input_channels, 59 RunTest(Blocker* blocker, size_t chunk_size, size_t num_frames, const float* const* input, float* const* input_chunk, float* const* output, float* const* output_chunk, size_t num_input_channels, size_t num_output_channels) argument
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/external/webrtc/webrtc/modules/audio_processing/test/ |
H A D | unpack.cc | 84 size_t num_input_channels = 0; local 143 num_input_channels * input_samples_per_channel, 151 new const float* [num_input_channels]); 152 for (size_t i = 0; i < num_input_channels; ++i) { 157 num_input_channels, 271 num_input_channels = msg.num_input_channels(); 273 num_input_channels); 305 num_input_channels));
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H A D | audio_file_processor.cc | 117 msg.num_input_channels())); 124 input_config_ = StreamConfig(msg.sample_rate(), msg.num_input_channels());
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H A D | audio_processing_unittest.cc | 262 size_t num_input_channels, 268 ss << name << "_i" << num_input_channels << "_" << input_rate / 1000 << "_ir" 360 size_t num_input_channels, 485 size_t num_input_channels, 489 SetContainerFormat(sample_rate_hz, num_input_channels, frame_, &float_cb_); 519 reverse_sample_rate_hz, num_input_channels, num_output_channels, 842 {{frame_->sample_rate_hz_, apm_->num_input_channels()}, 862 EXPECT_EQ(i, apm_->num_input_channels()); 882 EXPECT_EQ(i, apm_->num_input_channels()); 1734 msg.num_input_channels(), 257 OutputFilePath(std::string name, int input_rate, int output_rate, int reverse_input_rate, int reverse_output_rate, size_t num_input_channels, size_t num_output_channels, size_t num_reverse_input_channels, size_t num_reverse_output_channels, StreamDirection file_direction) argument 482 Init(int sample_rate_hz, int output_sample_rate_hz, int reverse_sample_rate_hz, size_t num_input_channels, size_t num_output_channels, size_t num_reverse_channels, bool open_output_file) argument 1941 const size_t num_input_channels = local 2383 ProcessFormat(int input_rate, int output_rate, int reverse_input_rate, int reverse_output_rate, size_t num_input_channels, size_t num_output_channels, size_t num_reverse_input_channels, size_t num_reverse_output_channels, std::string output_file_prefix) argument 2427 num_input_channels); local [all...] |
H A D | process_test.cc | 613 static_cast<size_t>(msg.num_input_channels())), 624 near_frame.num_channels_ = msg.num_input_channels(); 629 msg.num_input_channels())); 637 msg.num_input_channels(), 704 near_frame.num_channels_ = apm->num_input_channels();
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H A D | debug_dump_test.cc | 306 input_config_ = StreamConfig(msg.sample_rate(), msg.num_input_channels());
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/external/webrtc/webrtc/modules/audio_processing/ |
H A D | audio_buffer.h | 37 size_t num_input_channels,
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H A D | audio_buffer.cc | 47 size_t num_input_channels, 52 num_input_channels_(num_input_channels), 46 AudioBuffer(size_t input_num_frames, size_t num_input_channels, size_t process_num_frames, size_t num_process_channels, size_t output_num_frames) argument
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H A D | audio_processing_impl.h | 104 size_t num_input_channels() const override;
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H A D | audio_processing_impl.cc | 532 size_t AudioProcessingImpl::num_input_channels() const { function in class:webrtc::AudioProcessingImpl
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/external/webrtc/webrtc/modules/audio_processing/beamformer/ |
H A D | nonlinear_beamformer.h | 70 size_t num_input_channels,
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H A D | nonlinear_beamformer.cc | 411 size_t num_input_channels, 416 RTC_CHECK_EQ(num_input_channels_, num_input_channels); 410 ProcessAudioBlock(const complex_f* const* input, size_t num_input_channels, size_t num_freq_bins, size_t num_output_channels, complex_f* const* output) argument
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/external/webrtc/webrtc/modules/audio_processing/include/ |
H A D | mock_audio_processing.h | 203 MOCK_CONST_METHOD0(num_input_channels,
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H A D | audio_processing.h | 296 virtual size_t num_input_channels() const = 0;
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/external/webrtc/talk/media/webrtc/ |
H A D | fakewebrtcvoiceengine.h | 80 size_t num_input_channels() const override { return 0; }
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