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/*
* 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_PUSH_SINC_RESAMPLER_H_
#define COMMON_AUDIO_RESAMPLER_PUSH_SINC_RESAMPLER_H_
#include <stddef.h>
#include <stdint.h>
#include <memory>
#include "common_audio/resampler/sinc_resampler.h"
namespace webrtc {
// A thin wrapper over SincResampler to provide a push-based interface as
// required by WebRTC. SincResampler uses a pull-based interface, and will
// use SincResamplerCallback::Run() to request data upon a call to Resample().
// These Run() calls will happen on the same thread Resample() is called on.
class PushSincResampler : public SincResamplerCallback {
public:
// Provide the size of the source and destination blocks in samples. These
// must correspond to the same time duration (typically 10 ms) as the sample
// ratio is inferred from them.
PushSincResampler(size_t source_frames, size_t destination_frames);
~PushSincResampler() override;
PushSincResampler(const PushSincResampler&) = delete;
PushSincResampler& operator=(const PushSincResampler&) = delete;
// Perform the resampling. `source_frames` must always equal the
// `source_frames` provided at construction. `destination_capacity` must be
// at least as large as `destination_frames`. Returns the number of samples
// provided in destination (for convenience, since this will always be equal
// to `destination_frames`).
size_t Resample(const int16_t* source,
size_t source_frames,
int16_t* destination,
size_t destination_capacity);
size_t Resample(const float* source,
size_t source_frames,
float* destination,
size_t destination_capacity);
// Delay due to the filter kernel. Essentially, the time after which an input
// sample will appear in the resampled output.
static float AlgorithmicDelaySeconds(int source_rate_hz) {
return 1.f / source_rate_hz * SincResampler::kKernelSize / 2;
}
protected:
// Implements SincResamplerCallback.
void Run(size_t frames, float* destination) override;
private:
friend class PushSincResamplerTest;
SincResampler* get_resampler_for_testing() { return resampler_.get(); }
std::unique_ptr<SincResampler> resampler_;
std::unique_ptr<float[]> float_buffer_;
const float* source_ptr_;
const int16_t* source_ptr_int_;
const size_t destination_frames_;
// True on the first call to Resample(), to prime the SincResampler buffer.
bool first_pass_;
// Used to assert we are only requested for as much data as is available.
size_t source_available_;
};
} // namespace webrtc
#endif // COMMON_AUDIO_RESAMPLER_PUSH_SINC_RESAMPLER_H_
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