| /external/webrtc/webrtc/common_audio/ |
| 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,
|
| 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,
|
| 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,
|
| blocker.h | 29 size_t num_input_channels,
68 size_t num_input_channels,
76 size_t num_input_channels,
|
| lapped_transform.h | 98 size_t num_input_channels,
|
| /external/webrtc/webrtc/modules/audio_processing/test/ |
| unittest.proto | 7 optional int32 num_input_channels = 2;
|
| 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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| audio_file_processor.cc | 117 msg.num_input_channels()));
124 input_config_ = StreamConfig(msg.sample_rate(), msg.num_input_channels());
|
| 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());
1941 const size_t num_input_channels = local
2427 num_input_channels); local
[all...] |
| 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();
[all...] |
| debug_dump_test.cc | 306 input_config_ = StreamConfig(msg.sample_rate(), msg.num_input_channels());
|
| /external/webrtc/webrtc/modules/audio_processing/beamformer/ |
| covariance_matrix_generator.h | 21 // |num_input_channels| x |num_input_channels|.
|
| nonlinear_beamformer.h | 70 size_t num_input_channels,
|
| nonlinear_beamformer.cc | 411 size_t num_input_channels,
416 RTC_CHECK_EQ(num_input_channels_, num_input_channels);
|
| /external/webrtc/webrtc/modules/audio_processing/ |
| debug.proto | 10 optional int32 num_input_channels = 3;
|
| audio_buffer.cc | 47 size_t num_input_channels,
52 num_input_channels_(num_input_channels),
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| audio_buffer.h | 37 size_t num_input_channels,
|
| audio_processing_impl.h | 104 size_t num_input_channels() const override;
|
| audio_processing_impl.cc | 532 size_t AudioProcessingImpl::num_input_channels() const { function in class:webrtc::AudioProcessingImpl
[all...] |
| /external/webrtc/webrtc/modules/audio_processing/include/ |
| mock_audio_processing.h | 203 MOCK_CONST_METHOD0(num_input_channels,
|
| audio_processing.h | 296 virtual size_t num_input_channels() const = 0;
[all...] |
| /external/webrtc/talk/media/webrtc/ |
| fakewebrtcvoiceengine.h | 80 size_t num_input_channels() const override { return 0; }
|
