mirror of
https://github.com/danog/libtgvoip.git
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5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
73 lines
2.6 KiB
C++
73 lines
2.6 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 <stddef.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,
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size_t src_size,
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float* const* dst,
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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,
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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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// 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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