5804936052
And use a std::min. Post-commit fixes after: https://review.webrtc.org/30779004/ TBR=kwiberg Review URL: https://webrtc-codereview.appspot.com/25059004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7600 4adac7df-926f-26a2-2b94-8c16560cd09d
56 lines
1.8 KiB
C++
56 lines
1.8 KiB
C++
/*
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* Copyright (c) 2014 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_AUDIO_CONVERTER_H_
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#define WEBRTC_COMMON_AUDIO_AUDIO_CONVERTER_H_
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// TODO(ajm): Move channel buffer to common_audio.
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#include "webrtc/base/constructormagic.h"
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#include "webrtc/modules/audio_processing/common.h"
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#include "webrtc/system_wrappers/interface/scoped_ptr.h"
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#include "webrtc/system_wrappers/interface/scoped_vector.h"
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namespace webrtc {
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class PushSincResampler;
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// Format conversion (remixing and resampling) for audio. Only simple remixing
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// conversions are supported: downmix to mono (i.e. |dst_channels| == 1) or
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// upmix from mono (i.e. |src_channels == 1|).
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//
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// The source and destination chunks have the same duration in time; specifying
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// the number of frames is equivalent to specifying the sample rates.
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class AudioConverter {
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public:
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AudioConverter(int src_channels, int src_frames,
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int dst_channels, int dst_frames);
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void Convert(const float* const* src,
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int src_channels,
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int src_frames,
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int dst_channels,
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int dst_frames,
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float* const* dest);
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private:
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const int src_channels_;
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const int src_frames_;
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const int dst_channels_;
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const int dst_frames_;
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scoped_ptr<ChannelBuffer<float>> downmix_buffer_;
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ScopedVector<PushSincResampler> resamplers_;
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DISALLOW_COPY_AND_ASSIGN(AudioConverter);
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};
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} // namespace webrtc
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#endif // WEBRTC_COMMON_AUDIO_AUDIO_CONVERTER_H_
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