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https://github.com/danog/libtgvoip.git
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5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
68 lines
2.1 KiB
C++
68 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,
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size_t num_channels,
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size_t 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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// 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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size_t num_channels() const { return num_channels_; }
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size_t samples_per_channel() const { return channel_size_; }
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rtc::ArrayView<T> channel(size_t 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(size_t 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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size_t num_channels_;
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size_t 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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