andrew@webrtc.org f5a33f145b Resampler modifications in preparation for arbitrary audioproc rates.
- Templatize PushResampler to support int16 and float.
- Add a helper method to PushSincResampler to compute the algorithmic
delay.

This is a prerequisite of:
http://review.webrtc.org/9919004/

BUG=2894
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5943 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-19 00:32:07 +00:00

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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 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 PushSincResampler;
// Wraps PushSincResampler to provide stereo support.
// TODO(ajm): add support for an arbitrary number of channels.
template <typename T>
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 T* src, int src_length, T* dst, int dst_capacity);
private:
scoped_ptr<PushSincResampler> sinc_resampler_;
scoped_ptr<PushSincResampler> sinc_resampler_right_;
int src_sample_rate_hz_;
int dst_sample_rate_hz_;
int num_channels_;
scoped_ptr<T[]> src_left_;
scoped_ptr<T[]> src_right_;
scoped_ptr<T[]> dst_left_;
scoped_ptr<T[]> dst_right_;
};
} // namespace webrtc
#endif // WEBRTC_COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_