audio_converter.h (2643B)
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 namespace webrtc { 19 20 // Format conversion (remixing and resampling) for audio. Only simple remixing 21 // conversions are supported: downmix to mono (i.e. `dst_channels` == 1) or 22 // upmix from mono (i.e. |src_channels == 1|). 23 // 24 // The source and destination chunks have the same duration in time; specifying 25 // the number of frames is equivalent to specifying the sample rates. 26 class AudioConverter { 27 public: 28 // Returns a new AudioConverter, which will use the supplied format for its 29 // lifetime. Caller is responsible for the memory. 30 static std::unique_ptr<AudioConverter> Create(size_t src_channels, 31 size_t src_frames, 32 size_t dst_channels, 33 size_t dst_frames); 34 virtual ~AudioConverter() {} 35 36 AudioConverter(const AudioConverter&) = delete; 37 AudioConverter& operator=(const AudioConverter&) = delete; 38 39 // Convert `src`, containing `src_size` samples, to `dst`, having a sample 40 // capacity of `dst_capacity`. Both point to a series of buffers containing 41 // the samples for each channel. The sizes must correspond to the format 42 // passed to Create(). 43 virtual void Convert(const float* const* src, 44 size_t src_size, 45 float* const* dst, 46 size_t dst_capacity) = 0; 47 48 size_t src_channels() const { return src_channels_; } 49 size_t src_frames() const { return src_frames_; } 50 size_t dst_channels() const { return dst_channels_; } 51 size_t dst_frames() const { return dst_frames_; } 52 53 protected: 54 AudioConverter(); 55 AudioConverter(size_t src_channels, 56 size_t src_frames, 57 size_t dst_channels, 58 size_t dst_frames); 59 60 // Helper to RTC_CHECK that inputs are correctly sized. 61 void CheckSizes(size_t src_size, size_t dst_capacity) const; 62 63 private: 64 const size_t src_channels_; 65 const size_t src_frames_; 66 const size_t dst_channels_; 67 const size_t dst_frames_; 68 }; 69 70 } // namespace webrtc 71 72 #endif // COMMON_AUDIO_AUDIO_CONVERTER_H_