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/external/webrtc/webrtc/modules/audio_processing/test/
Daudio_processing_unittest.cc2277 ChannelBuffer<float> out_cb(SamplesFromRate(out_rate), in TEST_F() local
2289 out_cb.channels())); in TEST_F()
2430 ChannelBuffer<float> out_cb(SamplesFromRate(output_rate), in ProcessFormat() local
2437 2 * std::max(std::max(out_cb.num_frames(), rev_out_cb.num_frames()), in ProcessFormat()
2460 out_cb.channels())); in ProcessFormat()
2463 Interleave(out_cb.channels(), out_cb.num_frames(), out_cb.num_channels(), in ProcessFormat()
2465 size_t out_length = out_cb.num_channels() * out_cb.num_frames(); in ProcessFormat()