51b2459d37
BUG=163 R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1899004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4473 4adac7df-926f-26a2-2b94-8c16560cd09d
62 lines
2.3 KiB
C++
62 lines
2.3 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 source_frames, int destination_frames);
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virtual ~PushSincResampler();
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// Perform the resampling. |source_frames| must always equal the
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// |source_frames| provided at construction. |destination_capacity| must be
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// at least as large as |destination_frames|. 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 |destination_frames|).
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int Resample(const int16_t* source, int source_frames,
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int16_t* destination, int destination_capacity);
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// Implements SincResamplerCallback.
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virtual void Run(int frames, float* destination) OVERRIDE;
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SincResampler* get_resampler_for_testing() { return resampler_.get(); }
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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 destination_frames_;
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// True on the first call to Resample(), to prime the SincResampler buffer.
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bool first_pass_;
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// Used to assert we are only requested for as much data as is available.
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int source_available_;
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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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