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