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henrik.lundin@webrtc.org9a400812013-01-29 12:09:21 +00001/*
2 * Copyright (c) 2012 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_MODULES_AUDIO_CODING_NETEQ4_SYNC_BUFFER_H_
12#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_SYNC_BUFFER_H_
13
14#include "webrtc/modules/audio_coding/neteq4/audio_multi_vector.h"
15#include "webrtc/system_wrappers/interface/constructor_magic.h"
16#include "webrtc/typedefs.h"
17
18namespace webrtc {
19
henrik.lundin@webrtc.org0e9c3992013-09-30 20:38:44 +000020class SyncBuffer : public AudioMultiVector {
henrik.lundin@webrtc.org9a400812013-01-29 12:09:21 +000021 public:
22 SyncBuffer(size_t channels, size_t length)
henrik.lundin@webrtc.org0e9c3992013-09-30 20:38:44 +000023 : AudioMultiVector(channels, length),
henrik.lundin@webrtc.org9a400812013-01-29 12:09:21 +000024 next_index_(length),
25 end_timestamp_(0),
26 dtmf_index_(0) {}
27
28 virtual ~SyncBuffer() {}
29
30 // Returns the number of samples yet to play out form the buffer.
31 size_t FutureLength() const;
32
33 // Adds the contents of |append_this| to the back of the SyncBuffer. Removes
34 // the same number of samples from the beginning of the SyncBuffer, to
35 // maintain a constant buffer size. The |next_index_| is updated to reflect
36 // the move of the beginning of "future" data.
henrik.lundin@webrtc.org0e9c3992013-09-30 20:38:44 +000037 void PushBack(const AudioMultiVector& append_this);
henrik.lundin@webrtc.org9a400812013-01-29 12:09:21 +000038
39 // Adds |length| zeros to the beginning of each channel. Removes
40 // the same number of samples from the end of the SyncBuffer, to
41 // maintain a constant buffer size. The |next_index_| is updated to reflect
42 // the move of the beginning of "future" data.
43 // Note that this operation may delete future samples that are waiting to
44 // be played.
45 void PushFrontZeros(size_t length);
46
47 // Inserts |length| zeros into each channel at index |position|. The size of
48 // the SyncBuffer is kept constant, which means that the last |length|
49 // elements in each channel will be purged.
50 virtual void InsertZerosAtIndex(size_t length, size_t position);
51
52 // Overwrites each channel in this SyncBuffer with values taken from
53 // |insert_this|. The values are taken from the beginning of |insert_this| and
54 // are inserted starting at |position|. |length| values are written into each
55 // channel. The size of the SyncBuffer is kept constant. That is, if |length|
56 // and |position| are selected such that the new data would extend beyond the
57 // end of the current SyncBuffer, the buffer is not extended.
58 // The |next_index_| is not updated.
henrik.lundin@webrtc.org0e9c3992013-09-30 20:38:44 +000059 virtual void ReplaceAtIndex(const AudioMultiVector& insert_this,
henrik.lundin@webrtc.org9a400812013-01-29 12:09:21 +000060 size_t length,
61 size_t position);
62
63 // Same as the above method, but where all of |insert_this| is written (with
64 // the same constraints as above, that the SyncBuffer is not extended).
henrik.lundin@webrtc.org0e9c3992013-09-30 20:38:44 +000065 virtual void ReplaceAtIndex(const AudioMultiVector& insert_this,
henrik.lundin@webrtc.org9a400812013-01-29 12:09:21 +000066 size_t position);
67
68 // Reads |requested_len| samples from each channel and writes them interleaved
69 // into |output|. The |next_index_| is updated to point to the sample to read
70 // next time.
71 size_t GetNextAudioInterleaved(size_t requested_len, int16_t* output);
72
73 // Adds |increment| to |end_timestamp_|.
74 void IncreaseEndTimestamp(uint32_t increment);
75
76 // Flushes the buffer. The buffer will contain only zeros after the flush, and
77 // |next_index_| will point to the end, like when the buffer was first
78 // created.
79 void Flush();
80
81 const AudioVector<int16_t>& Channel(size_t n) { return *channels_[n]; }
82
83 // Accessors and mutators.
84 size_t next_index() const { return next_index_; }
85 void set_next_index(size_t value);
86 uint32_t end_timestamp() const { return end_timestamp_; }
87 void set_end_timestamp(uint32_t value) { end_timestamp_ = value; }
88 size_t dtmf_index() const { return dtmf_index_; }
89 void set_dtmf_index(size_t value);
90
91 private:
92 size_t next_index_;
93 uint32_t end_timestamp_; // The timestamp of the last sample in the buffer.
94 size_t dtmf_index_; // Index to the first non-DTMF sample in the buffer.
95
96 DISALLOW_COPY_AND_ASSIGN(SyncBuffer);
97};
98
99} // namespace webrtc
100#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_SYNC_BUFFER_H_