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  /external/webrtc/webrtc/modules/audio_processing/test/
test_utils.cc 41 interleaved_.resize(buffer->size());
42 if (file_->ReadSamples(interleaved_.size(), &interleaved_[0]) !=
43 interleaved_.size()) {
47 FloatS16ToFloat(&interleaved_[0], interleaved_.size(), &interleaved_[0]);
48 Deinterleave(&interleaved_[0], buffer->num_frames(), buffer->num_channels(),
58 interleaved_.resize(buffer.size());
60 &interleaved_[0])
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test_utils.h 57 std::vector<float> interleaved_; member in class:webrtc::final
70 std::vector<float> interleaved_; member in class:webrtc::final
  /external/webrtc/webrtc/modules/include/
module_common_types.h 507 // |interleaved_| is not changed by this method.
543 bool interleaved_; member in class:webrtc::AudioFrame
567 interleaved_ = true;
610 interleaved_ = src.interleaved_;
634 assert(interleaved_ == rhs.interleaved_);
670 assert(interleaved_ == rhs.interleaved_);
711 assert(interleaved_ == rhs.interleaved_)
    [all...]

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