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(int src_channels,
30                                                 int src_frames,
31                                                 int dst_channels,
32                                                 int dst_frames);
~AudioConverter()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 
src_channels()42   int src_channels() const { return src_channels_; }
src_frames()43   int src_frames() const { return src_frames_; }
dst_channels()44   int dst_channels() const { return dst_channels_; }
dst_frames()45   int dst_frames() const { return dst_frames_; }
46 
47  protected:
48   AudioConverter();
49   AudioConverter(int src_channels, int src_frames, int dst_channels,
50                  int dst_frames);
51 
52   // Helper to CHECK that inputs are correctly sized.
53   void CheckSizes(size_t src_size, size_t dst_capacity) const;
54 
55  private:
56   const int src_channels_;
57   const int src_frames_;
58   const int dst_channels_;
59   const int dst_frames_;
60 
61   DISALLOW_COPY_AND_ASSIGN(AudioConverter);
62 };
63 
64 }  // namespace webrtc
65 
66 #endif  // WEBRTC_COMMON_AUDIO_AUDIO_CONVERTER_H_
67