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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 #include "modules/audio_coding/neteq/sync_buffer.h"
12 
13 #include <algorithm>  // Access to min.
14 
15 #include "rtc_base/checks.h"
16 
17 namespace webrtc {
18 
FutureLength() const19 size_t SyncBuffer::FutureLength() const {
20   return Size() - next_index_;
21 }
22 
PushBack(const AudioMultiVector & append_this)23 void SyncBuffer::PushBack(const AudioMultiVector& append_this) {
24   size_t samples_added = append_this.Size();
25   AudioMultiVector::PushBack(append_this);
26   AudioMultiVector::PopFront(samples_added);
27   if (samples_added <= next_index_) {
28     next_index_ -= samples_added;
29   } else {
30     // This means that we are pushing out future data that was never used.
31     //    assert(false);
32     // TODO(hlundin): This assert must be disabled to support 60 ms frames.
33     // This should not happen even for 60 ms frames, but it does. Investigate
34     // why.
35     next_index_ = 0;
36   }
37   dtmf_index_ -= std::min(dtmf_index_, samples_added);
38 }
39 
PushBackInterleaved(const rtc::BufferT<int16_t> & append_this)40 void SyncBuffer::PushBackInterleaved(const rtc::BufferT<int16_t>& append_this) {
41   const size_t size_before_adding = Size();
42   AudioMultiVector::PushBackInterleaved(append_this);
43   const size_t samples_added_per_channel = Size() - size_before_adding;
44   RTC_DCHECK_EQ(samples_added_per_channel * Channels(), append_this.size());
45   AudioMultiVector::PopFront(samples_added_per_channel);
46   next_index_ -= std::min(next_index_, samples_added_per_channel);
47   dtmf_index_ -= std::min(dtmf_index_, samples_added_per_channel);
48 }
49 
PushFrontZeros(size_t length)50 void SyncBuffer::PushFrontZeros(size_t length) {
51   InsertZerosAtIndex(length, 0);
52 }
53 
InsertZerosAtIndex(size_t length,size_t position)54 void SyncBuffer::InsertZerosAtIndex(size_t length, size_t position) {
55   position = std::min(position, Size());
56   length = std::min(length, Size() - position);
57   AudioMultiVector::PopBack(length);
58   for (size_t channel = 0; channel < Channels(); ++channel) {
59     channels_[channel]->InsertZerosAt(length, position);
60   }
61   if (next_index_ >= position) {
62     // We are moving the |next_index_| sample.
63     set_next_index(next_index_ + length);  // Overflow handled by subfunction.
64   }
65   if (dtmf_index_ > 0 && dtmf_index_ >= position) {
66     // We are moving the |dtmf_index_| sample.
67     set_dtmf_index(dtmf_index_ + length);  // Overflow handled by subfunction.
68   }
69 }
70 
ReplaceAtIndex(const AudioMultiVector & insert_this,size_t length,size_t position)71 void SyncBuffer::ReplaceAtIndex(const AudioMultiVector& insert_this,
72                                 size_t length,
73                                 size_t position) {
74   position = std::min(position, Size());  // Cap |position| in the valid range.
75   length = std::min(length, Size() - position);
76   AudioMultiVector::OverwriteAt(insert_this, length, position);
77 }
78 
ReplaceAtIndex(const AudioMultiVector & insert_this,size_t position)79 void SyncBuffer::ReplaceAtIndex(const AudioMultiVector& insert_this,
80                                 size_t position) {
81   ReplaceAtIndex(insert_this, insert_this.Size(), position);
82 }
83 
GetNextAudioInterleaved(size_t requested_len,AudioFrame * output)84 void SyncBuffer::GetNextAudioInterleaved(size_t requested_len,
85                                          AudioFrame* output) {
86   RTC_DCHECK(output);
87   const size_t samples_to_read = std::min(FutureLength(), requested_len);
88   output->ResetWithoutMuting();
89   const size_t tot_samples_read = ReadInterleavedFromIndex(
90       next_index_, samples_to_read, output->mutable_data());
91   const size_t samples_read_per_channel = tot_samples_read / Channels();
92   next_index_ += samples_read_per_channel;
93   output->num_channels_ = Channels();
94   output->samples_per_channel_ = samples_read_per_channel;
95 }
96 
IncreaseEndTimestamp(uint32_t increment)97 void SyncBuffer::IncreaseEndTimestamp(uint32_t increment) {
98   end_timestamp_ += increment;
99 }
100 
Flush()101 void SyncBuffer::Flush() {
102   Zeros(Size());
103   next_index_ = Size();
104   end_timestamp_ = 0;
105   dtmf_index_ = 0;
106 }
107 
set_next_index(size_t value)108 void SyncBuffer::set_next_index(size_t value) {
109   // Cannot set |next_index_| larger than the size of the buffer.
110   next_index_ = std::min(value, Size());
111 }
112 
set_dtmf_index(size_t value)113 void SyncBuffer::set_dtmf_index(size_t value) {
114   // Cannot set |dtmf_index_| larger than the size of the buffer.
115   dtmf_index_ = std::min(value, Size());
116 }
117 
118 }  // namespace webrtc
119