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henrik.lundin@webrtc.orgd94659d2013-01-29 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
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020011#ifndef MODULES_AUDIO_CODING_NETEQ_SYNC_BUFFER_H_
12#define MODULES_AUDIO_CODING_NETEQ_SYNC_BUFFER_H_
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000013
Fredrik Solenbergbbf21a32018-04-12 22:44:09 +020014#include "api/audio/audio_frame.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020015#include "modules/audio_coding/neteq/audio_multi_vector.h"
Henrik Lundin00eb12a2018-09-05 18:14:52 +020016#include "rtc_base/buffer.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020017#include "rtc_base/constructormagic.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000018
19namespace webrtc {
20
henrik.lundin@webrtc.orgfd11bbf2013-09-30 20:38:44 +000021class SyncBuffer : public AudioMultiVector {
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000022 public:
23 SyncBuffer(size_t channels, size_t length)
henrik.lundin@webrtc.orgfd11bbf2013-09-30 20:38:44 +000024 : AudioMultiVector(channels, length),
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000025 next_index_(length),
26 end_timestamp_(0),
27 dtmf_index_(0) {}
28
henrik.lundin114c1b32017-04-26 07:47:32 -070029 // Returns the number of samples yet to play out from the buffer.
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000030 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.
Karl Wiberg7f6c4d42015-04-09 15:44:22 +020036 void PushBack(const AudioMultiVector& append_this) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000037
Henrik Lundin00eb12a2018-09-05 18:14:52 +020038 // Like PushBack, but reads the samples channel-interleaved from the input.
39 void PushBackInterleaved(const rtc::BufferT<int16_t>& append_this);
40
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000041 // Adds |length| zeros to the beginning of each channel. Removes
42 // the same number of samples from the end of the SyncBuffer, to
43 // maintain a constant buffer size. The |next_index_| is updated to reflect
44 // the move of the beginning of "future" data.
45 // Note that this operation may delete future samples that are waiting to
46 // be played.
47 void PushFrontZeros(size_t length);
48
49 // Inserts |length| zeros into each channel at index |position|. The size of
50 // the SyncBuffer is kept constant, which means that the last |length|
51 // elements in each channel will be purged.
52 virtual void InsertZerosAtIndex(size_t length, size_t position);
53
54 // Overwrites each channel in this SyncBuffer with values taken from
55 // |insert_this|. The values are taken from the beginning of |insert_this| and
56 // are inserted starting at |position|. |length| values are written into each
57 // channel. The size of the SyncBuffer is kept constant. That is, if |length|
58 // and |position| are selected such that the new data would extend beyond the
59 // end of the current SyncBuffer, the buffer is not extended.
60 // The |next_index_| is not updated.
henrik.lundin@webrtc.orgfd11bbf2013-09-30 20:38:44 +000061 virtual void ReplaceAtIndex(const AudioMultiVector& insert_this,
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000062 size_t length,
63 size_t position);
64
65 // Same as the above method, but where all of |insert_this| is written (with
66 // the same constraints as above, that the SyncBuffer is not extended).
henrik.lundin@webrtc.orgfd11bbf2013-09-30 20:38:44 +000067 virtual void ReplaceAtIndex(const AudioMultiVector& insert_this,
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000068 size_t position);
69
70 // Reads |requested_len| samples from each channel and writes them interleaved
71 // into |output|. The |next_index_| is updated to point to the sample to read
henrik.lundin6d8e0112016-03-04 10:34:21 -080072 // next time. The AudioFrame |output| is first reset, and the |data_|,
henrik.lundin7dc68892016-04-06 01:03:02 -070073 // |num_channels_|, and |samples_per_channel_| fields are updated.
henrik.lundin6d8e0112016-03-04 10:34:21 -080074 void GetNextAudioInterleaved(size_t requested_len, AudioFrame* output);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000075
76 // Adds |increment| to |end_timestamp_|.
77 void IncreaseEndTimestamp(uint32_t increment);
78
79 // Flushes the buffer. The buffer will contain only zeros after the flush, and
80 // |next_index_| will point to the end, like when the buffer was first
81 // created.
82 void Flush();
83
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +000084 const AudioVector& Channel(size_t n) const { return *channels_[n]; }
85 AudioVector& Channel(size_t n) { return *channels_[n]; }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000086
87 // Accessors and mutators.
88 size_t next_index() const { return next_index_; }
89 void set_next_index(size_t value);
90 uint32_t end_timestamp() const { return end_timestamp_; }
91 void set_end_timestamp(uint32_t value) { end_timestamp_ = value; }
92 size_t dtmf_index() const { return dtmf_index_; }
93 void set_dtmf_index(size_t value);
94
95 private:
96 size_t next_index_;
97 uint32_t end_timestamp_; // The timestamp of the last sample in the buffer.
Yves Gerey665174f2018-06-19 15:03:05 +020098 size_t dtmf_index_; // Index to the first non-DTMF sample in the buffer.
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000099
henrikg3c089d72015-09-16 05:37:44 -0700100 RTC_DISALLOW_COPY_AND_ASSIGN(SyncBuffer);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000101};
102
103} // namespace webrtc
Mirko Bonadei92ea95e2017-09-15 06:47:31 +0200104#endif // MODULES_AUDIO_CODING_NETEQ_SYNC_BUFFER_H_