50b2efef6e
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
62 lines
2.1 KiB
C++
62 lines
2.1 KiB
C++
/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_
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#define WEBRTC_COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_
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#include "webrtc/system_wrappers/interface/scoped_ptr.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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class Resampler;
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class PushSincResampler;
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// Wraps the old resampler and new arbitrary rate conversion resampler. The
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// old resampler will be used whenever it supports the requested rates, and
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// otherwise the sinc resampler will be enabled.
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class PushResampler {
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public:
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PushResampler();
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virtual ~PushResampler();
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// Must be called whenever the parameters change. Free to be called at any
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// time as it is a no-op if parameters have not changed since the last call.
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int InitializeIfNeeded(int src_sample_rate_hz, int dst_sample_rate_hz,
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int num_channels);
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// Returns the total number of samples provided in destination (e.g. 32 kHz,
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// 2 channel audio gives 640 samples).
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int Resample(const int16_t* src, int src_length, int16_t* dst,
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int dst_capacity);
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bool use_sinc_resampler() const { return use_sinc_resampler_; }
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private:
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int ResampleSinc(const int16_t* src, int src_length, int16_t* dst,
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int dst_capacity);
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scoped_ptr<Resampler> resampler_;
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scoped_ptr<PushSincResampler> sinc_resampler_;
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scoped_ptr<PushSincResampler> sinc_resampler_right_;
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int src_sample_rate_hz_;
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int dst_sample_rate_hz_;
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int num_channels_;
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bool use_sinc_resampler_;
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scoped_array<int16_t> src_left_;
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scoped_array<int16_t> src_right_;
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scoped_array<int16_t> dst_left_;
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scoped_array<int16_t> dst_right_;
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};
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} // namespace webrtc
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#endif // WEBRTC_COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_
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