62 lines
2.1 KiB
C
62 lines
2.1 KiB
C
|
/*
|
||
|
|
* 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 WEBRTC_COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_
|
||
|
|
#define WEBRTC_COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_
|
||
|
|
|
||
|
|
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
|
||
|
|
#include "webrtc/typedefs.h"
|
||
|
|
|
||
|
|
namespace webrtc {
|
||
|
|
|
||
|
|
class Resampler;
|
||
|
|
class PushSincResampler;
|
||
|
|
|
||
|
|
// Wraps the old resampler and new arbitrary rate conversion resampler. The
|
||
|
|
// old resampler will be used whenever it supports the requested rates, and
|
||
|
|
// otherwise the sinc resampler will be enabled.
|
||
|
|
class PushResampler {
|
||
|
|
public:
|
||
|
|
PushResampler();
|
||
|
|
virtual ~PushResampler();
|
||
|
|
|
||
|
|
// Must be called whenever the parameters change. Free to be called at any
|
||
|
|
// time as it is a no-op if parameters have not changed since the last call.
|
||
|
|
int InitializeIfNeeded(int src_sample_rate_hz, int dst_sample_rate_hz,
|
||
|
|
int num_channels);
|
||
|
|
|
||
|
|
// Returns the total number of samples provided in destination (e.g. 32 kHz,
|
||
|
|
// 2 channel audio gives 640 samples).
|
||
|
|
int Resample(const int16_t* src, int src_length, int16_t* dst,
|
||
|
|
int dst_capacity);
|
||
|
|
|
||
|
|
bool use_sinc_resampler() const { return use_sinc_resampler_; }
|
||
|
|
|
||
|
|
private:
|
||
|
|
int ResampleSinc(const int16_t* src, int src_length, int16_t* dst,
|
||
|
|
int dst_capacity);
|
||
|
|
|
||
|
|
scoped_ptr<Resampler> resampler_;
|
||
|
|
scoped_ptr<PushSincResampler> sinc_resampler_;
|
||
|
|
scoped_ptr<PushSincResampler> sinc_resampler_right_;
|
||
|
|
int src_sample_rate_hz_;
|
||
|
|
int dst_sample_rate_hz_;
|
||
|
|
int num_channels_;
|
||
|
|
bool use_sinc_resampler_;
|
||
|
|
scoped_array<int16_t> src_left_;
|
||
|
|
scoped_array<int16_t> src_right_;
|
||
|
|
scoped_array<int16_t> dst_left_;
|
||
|
|
scoped_array<int16_t> dst_right_;
|
||
|
|
};
|
||
|
|
|
||
|
|
} // namespace webrtc
|
||
|
|
|
||
|
|
#endif // WEBRTC_COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_
|