1/*
2 *  Copyright (c) 2015 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#ifndef WEBRTC_COMMON_AUDIO_AUDIO_RING_BUFFER_H_
11#define WEBRTC_COMMON_AUDIO_AUDIO_RING_BUFFER_H_
12
13#include <stddef.h>
14#include <vector>
15
16struct RingBuffer;
17
18namespace webrtc {
19
20// A ring buffer tailored for float deinterleaved audio. Any operation that
21// cannot be performed as requested will cause a crash (e.g. insufficient data
22// in the buffer to fulfill a read request.)
23class AudioRingBuffer final {
24 public:
25  // Specify the number of channels and maximum number of frames the buffer will
26  // contain.
27  AudioRingBuffer(size_t channels, size_t max_frames);
28  ~AudioRingBuffer();
29
30  // Copies |data| to the buffer and advances the write pointer. |channels| must
31  // be the same as at creation time.
32  void Write(const float* const* data, size_t channels, size_t frames);
33
34  // Copies from the buffer to |data| and advances the read pointer. |channels|
35  // must be the same as at creation time.
36  void Read(float* const* data, size_t channels, size_t frames);
37
38  size_t ReadFramesAvailable() const;
39  size_t WriteFramesAvailable() const;
40
41  // Moves the read position. The forward version advances the read pointer
42  // towards the write pointer and the backward verison withdraws the read
43  // pointer away from the write pointer (i.e. flushing and stuffing the buffer
44  // respectively.)
45  void MoveReadPositionForward(size_t frames);
46  void MoveReadPositionBackward(size_t frames);
47
48 private:
49  // We don't use a ScopedVector because it doesn't support a specialized
50  // deleter (like scoped_ptr for instance.)
51  std::vector<RingBuffer*> buffers_;
52};
53
54}  // namespace webrtc
55
56#endif  // WEBRTC_COMMON_AUDIO_AUDIO_RING_BUFFER_H_
57