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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
67 lines
2.5 KiB
C++
67 lines
2.5 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 COMMON_AUDIO_AUDIO_CONVERTER_H_
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#define COMMON_AUDIO_AUDIO_CONVERTER_H_
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#include <memory>
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#include "rtc_base/constructormagic.h"
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namespace webrtc {
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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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// Returns a new AudioConverter, which will use the supplied format for its
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// lifetime. Caller is responsible for the memory.
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static std::unique_ptr<AudioConverter> Create(size_t src_channels,
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size_t src_frames,
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size_t dst_channels,
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size_t dst_frames);
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virtual ~AudioConverter() {}
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// Convert |src|, containing |src_size| samples, to |dst|, having a sample
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// capacity of |dst_capacity|. Both point to a series of buffers containing
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// the samples for each channel. The sizes must correspond to the format
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// passed to Create().
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virtual void Convert(const float* const* src, size_t src_size,
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float* const* dst, size_t dst_capacity) = 0;
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size_t src_channels() const { return src_channels_; }
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size_t src_frames() const { return src_frames_; }
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size_t dst_channels() const { return dst_channels_; }
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size_t dst_frames() const { return dst_frames_; }
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protected:
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AudioConverter();
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AudioConverter(size_t src_channels, size_t src_frames, size_t dst_channels,
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size_t dst_frames);
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// Helper to RTC_CHECK that inputs are correctly sized.
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void CheckSizes(size_t src_size, size_t dst_capacity) const;
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private:
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const size_t src_channels_;
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const size_t src_frames_;
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const size_t dst_channels_;
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const size_t dst_frames_;
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RTC_DISALLOW_COPY_AND_ASSIGN(AudioConverter);
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};
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} // namespace webrtc
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#endif // COMMON_AUDIO_AUDIO_CONVERTER_H_
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