/external/webrtc/webrtc/modules/audio_processing/test/ |
D | unpack.cc | 84 size_t num_input_channels = 0; in do_main() local 143 num_input_channels * input_samples_per_channel, in do_main() 151 new const float* [num_input_channels]); in do_main() 152 for (size_t i = 0; i < num_input_channels; ++i) { in do_main() 157 num_input_channels, in do_main() 271 num_input_channels = msg.num_input_channels(); in do_main() 273 num_input_channels); in do_main() 305 num_input_channels)); in do_main()
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D | audio_processing_unittest.cc | 262 size_t num_input_channels, in OutputFilePath() argument 268 ss << name << "_i" << num_input_channels << "_" << input_rate / 1000 << "_ir" in OutputFilePath() 360 size_t num_input_channels, 485 size_t num_input_channels, in Init() argument 489 SetContainerFormat(sample_rate_hz, num_input_channels, frame_, &float_cb_); in Init() 519 reverse_sample_rate_hz, num_input_channels, num_output_channels, in Init() 842 {{frame_->sample_rate_hz_, apm_->num_input_channels()}, in TestChangingReverseChannels() 862 EXPECT_EQ(i, apm_->num_input_channels()); in TEST_F() 882 EXPECT_EQ(i, apm_->num_input_channels()); in TEST_F() 1734 msg.num_input_channels(), in ProcessDebugDump() [all …]
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D | audio_file_processor.cc | 117 msg.num_input_channels())); in HandleMessage() 124 input_config_ = StreamConfig(msg.sample_rate(), msg.num_input_channels()); in HandleMessage()
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D | unittest.proto | 7 optional int32 num_input_channels = 2; field
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D | process_test.cc | 613 static_cast<size_t>(msg.num_input_channels())), in void_main() 624 near_frame.num_channels_ = msg.num_input_channels(); in void_main() 629 msg.num_input_channels())); in void_main() 637 msg.num_input_channels(), in void_main() 704 near_frame.num_channels_ = apm->num_input_channels(); in void_main()
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D | debug_dump_test.cc | 306 input_config_ = StreamConfig(msg.sample_rate(), msg.num_input_channels()); in OnInitEvent()
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/external/webrtc/webrtc/common_audio/ |
D | blocker.cc | 103 size_t num_input_channels, in Blocker() argument 110 num_input_channels_(num_input_channels), in Blocker() 169 size_t num_input_channels, in ProcessChunk() argument 173 RTC_CHECK_EQ(num_input_channels, num_input_channels_); in ProcessChunk() 176 input_buffer_.Write(input, num_input_channels, chunk_size_); in ProcessChunk() 181 input_buffer_.Read(input_block_.channels(), num_input_channels, in ProcessChunk()
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D | lapped_transform.cc | 24 size_t num_input_channels, in ProcessBlock() argument 27 RTC_CHECK_EQ(num_input_channels, parent_->num_in_channels_); in ProcessBlock() 31 for (size_t i = 0; i < num_input_channels; ++i) { in ProcessBlock() 42 num_input_channels, in ProcessBlock()
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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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D | blocker_unittest.cc | 23 size_t num_input_channels, in ProcessBlock() argument 39 size_t num_input_channels, in ProcessBlock() argument 66 size_t num_input_channels, in RunTest() argument 71 CopyTo(input_chunk, 0, start, num_input_channels, chunk_size, input); in RunTest() 74 num_input_channels, in RunTest()
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D | lapped_transform.h | 98 size_t num_input_channels,
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/external/webrtc/webrtc/modules/audio_processing/beamformer/ |
D | nonlinear_beamformer.h | 70 size_t num_input_channels,
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D | nonlinear_beamformer.cc | 411 size_t num_input_channels, in ProcessAudioBlock() argument 416 RTC_CHECK_EQ(num_input_channels_, num_input_channels); in ProcessAudioBlock()
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/external/webrtc/webrtc/modules/audio_processing/ |
D | audio_buffer.h | 37 size_t num_input_channels,
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D | audio_buffer.cc | 47 size_t num_input_channels, in AudioBuffer() argument 52 num_input_channels_(num_input_channels), in AudioBuffer()
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D | debug.proto | 10 optional int32 num_input_channels = 3; field
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D | audio_processing_impl.h | 104 size_t num_input_channels() const override;
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D | audio_processing_impl.cc | 532 size_t AudioProcessingImpl::num_input_channels() const { in num_input_channels() function in webrtc::AudioProcessingImpl
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/external/webrtc/webrtc/modules/audio_processing/include/ |
D | mock_audio_processing.h | 203 MOCK_CONST_METHOD0(num_input_channels,
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D | audio_processing.h | 296 virtual size_t num_input_channels() const = 0;
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/external/webrtc/talk/media/webrtc/ |
D | fakewebrtcvoiceengine.h | 80 size_t num_input_channels() const override { return 0; } in num_input_channels() function
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