blob: 394e96b133b2a687a466602784ff28277442b01a [file] [log] [blame]
/*
* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_
#define COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_
#include <memory>
#include <vector>
#include "api/audio/audio_view.h"
namespace webrtc {
class PushSincResampler;
// Wraps PushSincResampler to provide stereo support.
// Note: This implementation assumes 10ms buffer sizes throughout.
template <typename T>
class PushResampler final {
public:
PushResampler();
PushResampler(size_t src_samples_per_channel,
size_t dst_samples_per_channel,
size_t num_channels);
~PushResampler();
// Returns the total number of samples provided in destination (e.g. 32 kHz,
// 2 channel audio gives 640 samples).
int Resample(InterleavedView<const T> src, InterleavedView<T> dst);
// For when a deinterleaved/mono channel already exists and we can skip the
// deinterleaved operation.
int Resample(MonoView<const T> src, MonoView<T> dst);
private:
// Ensures that source and destination buffers for deinterleaving are
// correctly configured prior to resampling that requires deinterleaving.
void EnsureInitialized(size_t src_samples_per_channel,
size_t dst_samples_per_channel,
size_t num_channels);
// Buffers used for when a deinterleaving step is necessary.
std::unique_ptr<T[]> source_;
std::unique_ptr<T[]> destination_;
DeinterleavedView<T> source_view_;
DeinterleavedView<T> destination_view_;
std::vector<std::unique_ptr<PushSincResampler>> resamplers_;
};
} // namespace webrtc
#endif // COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_