1 /* 2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 3 * 4 * Use of this source code is governed by a BSD-style license 5 * that can be found in the LICENSE file in the root of the source 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #ifndef COMMON_AUDIO_AUDIO_CONVERTER_H_ 12 #define COMMON_AUDIO_AUDIO_CONVERTER_H_ 13 14 #include <stddef.h> 15 16 #include <memory> 17 18 #include "rtc_base/constructor_magic.h" 19 20 namespace webrtc { 21 22 // Format conversion (remixing and resampling) for audio. Only simple remixing 23 // conversions are supported: downmix to mono (i.e. |dst_channels| == 1) or 24 // upmix from mono (i.e. |src_channels == 1|). 25 // 26 // The source and destination chunks have the same duration in time; specifying 27 // the number of frames is equivalent to specifying the sample rates. 28 class AudioConverter { 29 public: 30 // Returns a new AudioConverter, which will use the supplied format for its 31 // lifetime. Caller is responsible for the memory. 32 static std::unique_ptr<AudioConverter> Create(size_t src_channels, 33 size_t src_frames, 34 size_t dst_channels, 35 size_t dst_frames); ~AudioConverter()36 virtual ~AudioConverter() {} 37 38 // Convert |src|, containing |src_size| samples, to |dst|, having a sample 39 // capacity of |dst_capacity|. Both point to a series of buffers containing 40 // the samples for each channel. The sizes must correspond to the format 41 // passed to Create(). 42 virtual void Convert(const float* const* src, 43 size_t src_size, 44 float* const* dst, 45 size_t dst_capacity) = 0; 46 src_channels()47 size_t src_channels() const { return src_channels_; } src_frames()48 size_t src_frames() const { return src_frames_; } dst_channels()49 size_t dst_channels() const { return dst_channels_; } dst_frames()50 size_t dst_frames() const { return dst_frames_; } 51 52 protected: 53 AudioConverter(); 54 AudioConverter(size_t src_channels, 55 size_t src_frames, 56 size_t dst_channels, 57 size_t dst_frames); 58 59 // Helper to RTC_CHECK that inputs are correctly sized. 60 void CheckSizes(size_t src_size, size_t dst_capacity) const; 61 62 private: 63 const size_t src_channels_; 64 const size_t src_frames_; 65 const size_t dst_channels_; 66 const size_t dst_frames_; 67 68 RTC_DISALLOW_COPY_AND_ASSIGN(AudioConverter); 69 }; 70 71 } // namespace webrtc 72 73 #endif // COMMON_AUDIO_AUDIO_CONVERTER_H_ 74