Line data Source code
1 : /*
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_NETEQ_SYNC_BUFFER_H_
12 : #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_SYNC_BUFFER_H_
13 :
14 : #include "webrtc/base/constructormagic.h"
15 : #include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
16 : #include "webrtc/modules/include/module_common_types.h"
17 : #include "webrtc/typedefs.h"
18 :
19 : namespace webrtc {
20 :
21 0 : class SyncBuffer : public AudioMultiVector {
22 : public:
23 0 : SyncBuffer(size_t channels, size_t length)
24 0 : : AudioMultiVector(channels, length),
25 : next_index_(length),
26 : end_timestamp_(0),
27 0 : dtmf_index_(0) {}
28 :
29 : // Returns the number of samples yet to play out form the buffer.
30 : size_t FutureLength() const;
31 :
32 : // Adds the contents of |append_this| to the back of the SyncBuffer. Removes
33 : // the same number of samples from the beginning of the SyncBuffer, to
34 : // maintain a constant buffer size. The |next_index_| is updated to reflect
35 : // the move of the beginning of "future" data.
36 : void PushBack(const AudioMultiVector& append_this) override;
37 :
38 : // Adds |length| zeros to the beginning of each channel. Removes
39 : // the same number of samples from the end of the SyncBuffer, to
40 : // maintain a constant buffer size. The |next_index_| is updated to reflect
41 : // the move of the beginning of "future" data.
42 : // Note that this operation may delete future samples that are waiting to
43 : // be played.
44 : void PushFrontZeros(size_t length);
45 :
46 : // Inserts |length| zeros into each channel at index |position|. The size of
47 : // the SyncBuffer is kept constant, which means that the last |length|
48 : // elements in each channel will be purged.
49 : virtual void InsertZerosAtIndex(size_t length, size_t position);
50 :
51 : // Overwrites each channel in this SyncBuffer with values taken from
52 : // |insert_this|. The values are taken from the beginning of |insert_this| and
53 : // are inserted starting at |position|. |length| values are written into each
54 : // channel. The size of the SyncBuffer is kept constant. That is, if |length|
55 : // and |position| are selected such that the new data would extend beyond the
56 : // end of the current SyncBuffer, the buffer is not extended.
57 : // The |next_index_| is not updated.
58 : virtual void ReplaceAtIndex(const AudioMultiVector& insert_this,
59 : size_t length,
60 : size_t position);
61 :
62 : // Same as the above method, but where all of |insert_this| is written (with
63 : // the same constraints as above, that the SyncBuffer is not extended).
64 : virtual void ReplaceAtIndex(const AudioMultiVector& insert_this,
65 : size_t position);
66 :
67 : // Reads |requested_len| samples from each channel and writes them interleaved
68 : // into |output|. The |next_index_| is updated to point to the sample to read
69 : // next time. The AudioFrame |output| is first reset, and the |data_|,
70 : // |num_channels_|, and |samples_per_channel_| fields are updated.
71 : void GetNextAudioInterleaved(size_t requested_len, AudioFrame* 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& Channel(size_t n) const { return *channels_[n]; }
82 : AudioVector& Channel(size_t n) { return *channels_[n]; }
83 :
84 : // Accessors and mutators.
85 0 : size_t next_index() const { return next_index_; }
86 : void set_next_index(size_t value);
87 0 : uint32_t end_timestamp() const { return end_timestamp_; }
88 0 : void set_end_timestamp(uint32_t value) { end_timestamp_ = value; }
89 0 : size_t dtmf_index() const { return dtmf_index_; }
90 : void set_dtmf_index(size_t value);
91 :
92 : private:
93 : size_t next_index_;
94 : uint32_t end_timestamp_; // The timestamp of the last sample in the buffer.
95 : size_t dtmf_index_; // Index to the first non-DTMF sample in the buffer.
96 :
97 : RTC_DISALLOW_COPY_AND_ASSIGN(SyncBuffer);
98 : };
99 :
100 : } // namespace webrtc
101 : #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_SYNC_BUFFER_H_
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