c6abf6cd3f
Some API deprecation -- ExperimentalAgc and ExperimentalNs are gone. We're continuing to carry iSAC even though it's gone upstream, but maybe we'll want to drop that soon.
69 lines
2.1 KiB
C++
69 lines
2.1 KiB
C++
/*
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* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_FRAME_VIEW_H_
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#define MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_FRAME_VIEW_H_
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#include "api/array_view.h"
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namespace webrtc {
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// Class to pass audio data in T** format, where T is a numeric type.
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template <class T>
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class AudioFrameView {
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public:
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// `num_channels` and `channel_size` describe the T**
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// `audio_samples`. `audio_samples` is assumed to point to a
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// two-dimensional |num_channels * channel_size| array of floats.
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AudioFrameView(T* const* audio_samples, int num_channels, int channel_size)
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: audio_samples_(audio_samples),
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num_channels_(num_channels),
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channel_size_(channel_size) {
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RTC_DCHECK_GE(num_channels_, 0);
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RTC_DCHECK_GE(channel_size_, 0);
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}
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// Implicit cast to allow converting Frame<float> to
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// Frame<const float>.
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template <class U>
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AudioFrameView(AudioFrameView<U> other)
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: audio_samples_(other.data()),
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num_channels_(other.num_channels()),
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channel_size_(other.samples_per_channel()) {}
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AudioFrameView() = delete;
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int num_channels() const { return num_channels_; }
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int samples_per_channel() const { return channel_size_; }
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rtc::ArrayView<T> channel(int idx) {
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RTC_DCHECK_LE(0, idx);
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RTC_DCHECK_LE(idx, num_channels_);
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return rtc::ArrayView<T>(audio_samples_[idx], channel_size_);
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}
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rtc::ArrayView<const T> channel(int idx) const {
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RTC_DCHECK_LE(0, idx);
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RTC_DCHECK_LE(idx, num_channels_);
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return rtc::ArrayView<const T>(audio_samples_[idx], channel_size_);
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}
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T* const* data() { return audio_samples_; }
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private:
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T* const* audio_samples_;
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int num_channels_;
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int channel_size_;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_FRAME_VIEW_H_
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