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 WEBRTC_COMMON_AUDIO_AUDIO_CONVERTER_H_ 12 #define WEBRTC_COMMON_AUDIO_AUDIO_CONVERTER_H_ 13 14 #include "webrtc/base/constructormagic.h" 15 #include "webrtc/base/scoped_ptr.h" 16 17 namespace webrtc { 18 19 // Format conversion (remixing and resampling) for audio. Only simple remixing 20 // conversions are supported: downmix to mono (i.e. |dst_channels| == 1) or 21 // upmix from mono (i.e. |src_channels == 1|). 22 // 23 // The source and destination chunks have the same duration in time; specifying 24 // the number of frames is equivalent to specifying the sample rates. 25 class AudioConverter { 26 public: 27 // Returns a new AudioConverter, which will use the supplied format for its 28 // lifetime. Caller is responsible for the memory. 29 static rtc::scoped_ptr<AudioConverter> Create(size_t src_channels, 30 size_t src_frames, 31 size_t dst_channels, 32 size_t dst_frames); 33 virtual ~AudioConverter() {}; 34 35 // Convert |src|, containing |src_size| samples, to |dst|, having a sample 36 // capacity of |dst_capacity|. Both point to a series of buffers containing 37 // the samples for each channel. The sizes must correspond to the format 38 // passed to Create(). 39 virtual void Convert(const float* const* src, size_t src_size, 40 float* const* dst, size_t dst_capacity) = 0; 41 42 size_t src_channels() const { return src_channels_; } 43 size_t src_frames() const { return src_frames_; } 44 size_t dst_channels() const { return dst_channels_; } 45 size_t dst_frames() const { return dst_frames_; } 46 47 protected: 48 AudioConverter(); 49 AudioConverter(size_t src_channels, size_t src_frames, size_t dst_channels, 50 size_t dst_frames); 51 52 // Helper to RTC_CHECK that inputs are correctly sized. 53 void CheckSizes(size_t src_size, size_t dst_capacity) const; 54 55 private: 56 const size_t src_channels_; 57 const size_t src_frames_; 58 const size_t dst_channels_; 59 const size_t dst_frames_; 60 61 RTC_DISALLOW_COPY_AND_ASSIGN(AudioConverter); 62 }; 63 64 } // namespace webrtc 65 66 #endif // WEBRTC_COMMON_AUDIO_AUDIO_CONVERTER_H_ 67
