PushResampler now uses a single buffer for the deinterleaved channel sources and another for the deinterleaved destinations. Before, there was a dedicated buffer per channel (source and dest). This reduces allocations and allows for using DeinterleavedView for both which simplifies some checks. Bug: chromium:335805780 Change-Id: I553a36164109127fa332ab17918d53832d442303 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351542 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42415}
52 lines
1.6 KiB
C++
52 lines
1.6 KiB
C++
/*
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* Copyright (c) 2013 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 COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_
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#define COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_
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#include <memory>
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#include <vector>
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#include "api/audio/audio_view.h"
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namespace webrtc {
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class PushSincResampler;
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// Wraps PushSincResampler to provide stereo support.
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// Note: This implementation assumes 10ms buffer sizes throughout.
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template <typename T>
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class PushResampler {
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public:
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PushResampler();
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virtual ~PushResampler();
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// Must be called whenever the parameters change. Free to be called at any
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// time as it is a no-op if parameters have not changed since the last call.
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int InitializeIfNeeded(int src_sample_rate_hz,
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int dst_sample_rate_hz,
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size_t num_channels);
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// Returns the total number of samples provided in destination (e.g. 32 kHz,
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// 2 channel audio gives 640 samples).
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int Resample(InterleavedView<const T> src, InterleavedView<T> dst);
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private:
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std::unique_ptr<T[]> source_;
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std::unique_ptr<T[]> destination_;
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DeinterleavedView<T> source_view_;
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DeinterleavedView<T> destination_view_;
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std::vector<std::unique_ptr<PushSincResampler>> resamplers_;
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};
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} // namespace webrtc
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#endif // COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_
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