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Add a wrapper around PushSincResampler and the old Resampler.

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The old resampler is used whenever it supports the requested rates. Otherwise
the sinc resampler is enabled.

Integrated with output_mixer in order to test the change through
output_mixer_unittest. The sinc resampler will not yet be used, since we don't
feed VoE with any rates that trigger it.

BUG=webrtc:1395
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1355004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3915 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
andrew@webrtc.org
2013-04-29 17:27:29 +00:00
parent 5b7120c81b
commit 50b2efef6e
12 changed files with 513 additions and 77 deletions
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61
webrtc/common_audio/resampler/include/push_resampler.h Normal file
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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 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_