Line data Source code
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 <memory>
15 :
16 : #include "webrtc/base/constructormagic.h"
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 0 : virtual ~AudioConverter() {};
35 :
36 : // Convert |src|, containing |src_size| samples, to |dst|, having a sample
37 : // capacity of |dst_capacity|. Both point to a series of buffers containing
38 : // the samples for each channel. The sizes must correspond to the format
39 : // passed to Create().
40 : virtual void Convert(const float* const* src, size_t src_size,
41 : float* const* dst, size_t dst_capacity) = 0;
42 :
43 0 : size_t src_channels() const { return src_channels_; }
44 0 : size_t src_frames() const { return src_frames_; }
45 0 : size_t dst_channels() const { return dst_channels_; }
46 0 : size_t dst_frames() const { return dst_frames_; }
47 :
48 : protected:
49 : AudioConverter();
50 : AudioConverter(size_t src_channels, size_t src_frames, size_t dst_channels,
51 : size_t dst_frames);
52 :
53 : // Helper to RTC_CHECK that inputs are correctly sized.
54 : void CheckSizes(size_t src_size, size_t dst_capacity) const;
55 :
56 : private:
57 : const size_t src_channels_;
58 : const size_t src_frames_;
59 : const size_t dst_channels_;
60 : const size_t dst_frames_;
61 :
62 : RTC_DISALLOW_COPY_AND_ASSIGN(AudioConverter);
63 : };
64 :
65 : } // namespace webrtc
66 :
67 : #endif // WEBRTC_COMMON_AUDIO_AUDIO_CONVERTER_H_
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