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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.
37 lines
1.0 KiB
C++
37 lines
1.0 KiB
C++
/*
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* Copyright (c) 2016 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_RENDER_QUEUE_ITEM_VERIFIER_H_
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#define MODULES_AUDIO_PROCESSING_RENDER_QUEUE_ITEM_VERIFIER_H_
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#include <vector>
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namespace webrtc {
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// Functor to use when supplying a verifier function for the queue item
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// verifcation.
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template <typename T>
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class RenderQueueItemVerifier {
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public:
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explicit RenderQueueItemVerifier(size_t minimum_capacity)
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: minimum_capacity_(minimum_capacity) {}
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bool operator()(const std::vector<T>& v) const {
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return v.capacity() >= minimum_capacity_;
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}
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private:
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size_t minimum_capacity_;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_RENDER_QUEUE_ITEM_VERIFIER_H__
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