8fc05feed4
Includes a unittest to ensure we meet the same quality thresholds as SincResampler (modulo quantization error). BUG=webrtc:1395 Review URL: https://webrtc-codereview.appspot.com/1323011 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3909 4adac7df-926f-26a2-2b94-8c16560cd09d
54 lines
2.0 KiB
C++
54 lines
2.0 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_PUSH_SINC_RESAMPLER_H_
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#define WEBRTC_COMMON_AUDIO_RESAMPLER_PUSH_SINC_RESAMPLER_H_
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#include "webrtc/common_audio/resampler/sinc_resampler.h"
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#include "webrtc/system_wrappers/interface/constructor_magic.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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// A thin wrapper over SincResampler to provide a push-based interface as
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// required by WebRTC.
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class PushSincResampler : public SincResamplerCallback {
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public:
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// Provide the size of the source and destination blocks in samples. These
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// must correspond to the same time duration (typically 10 ms) as the sample
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// ratio is inferred from them.
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PushSincResampler(int src_block_size, int dst_block_size);
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virtual ~PushSincResampler();
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// Perform the resampling. |source_length| must always equal the
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// |src_block_size| provided at construction. |destination_capacity| must be
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// at least as large as |dst_block_size|. Returns the number of samples
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// provided in destination (for convenience, since this will always be equal
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// to |dst_block_size|).
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int Resample(const int16_t* source, int source_length,
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int16_t* destination, int destination_capacity);
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// Implements SincResamplerCallback.
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virtual void Run(float* destination, int frames);
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private:
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scoped_ptr<SincResampler> resampler_;
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scoped_array<float> float_buffer_;
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const int16_t* source_ptr_;
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const int dst_size_;
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DISALLOW_COPY_AND_ASSIGN(PushSincResampler);
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};
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} // namespace webrtc
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#endif // WEBRTC_COMMON_AUDIO_RESAMPLER_PUSH_SINC_RESAMPLER_H_
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