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1 /* | |
2 * Copyright (c) 2013 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 "webrtc/modules/audio_coding/neteq/nack.h" | |
12 | |
13 #include <assert.h> // For assert. | |
14 | |
15 #include <algorithm> // For std::max. | |
16 | |
17 #include "webrtc/base/checks.h" | |
18 #include "webrtc/modules/include/module_common_types.h" | |
19 #include "webrtc/system_wrappers/include/logging.h" | |
20 | |
21 namespace webrtc { | |
22 namespace { | |
23 | |
24 const int kDefaultSampleRateKhz = 48; | |
25 const int kDefaultPacketSizeMs = 20; | |
26 | |
27 } // namespace | |
28 | |
29 Nack::Nack(int nack_threshold_packets) | |
30 : nack_threshold_packets_(nack_threshold_packets), | |
31 sequence_num_last_received_rtp_(0), | |
32 timestamp_last_received_rtp_(0), | |
33 any_rtp_received_(false), | |
34 sequence_num_last_decoded_rtp_(0), | |
35 timestamp_last_decoded_rtp_(0), | |
36 any_rtp_decoded_(false), | |
37 sample_rate_khz_(kDefaultSampleRateKhz), | |
38 samples_per_packet_(sample_rate_khz_ * kDefaultPacketSizeMs), | |
39 max_nack_list_size_(kNackListSizeLimit) {} | |
40 | |
41 Nack::~Nack() = default; | |
42 | |
43 Nack* Nack::Create(int nack_threshold_packets) { | |
44 return new Nack(nack_threshold_packets); | |
45 } | |
46 | |
47 void Nack::UpdateSampleRate(int sample_rate_hz) { | |
48 assert(sample_rate_hz > 0); | |
49 sample_rate_khz_ = sample_rate_hz / 1000; | |
50 } | |
51 | |
52 void Nack::UpdateLastReceivedPacket(uint16_t sequence_number, | |
53 uint32_t timestamp) { | |
54 // Just record the value of sequence number and timestamp if this is the | |
55 // first packet. | |
56 if (!any_rtp_received_) { | |
57 sequence_num_last_received_rtp_ = sequence_number; | |
58 timestamp_last_received_rtp_ = timestamp; | |
59 any_rtp_received_ = true; | |
60 // If no packet is decoded, to have a reasonable estimate of time-to-play | |
61 // use the given values. | |
62 if (!any_rtp_decoded_) { | |
63 sequence_num_last_decoded_rtp_ = sequence_number; | |
64 timestamp_last_decoded_rtp_ = timestamp; | |
65 } | |
66 return; | |
67 } | |
68 | |
69 if (sequence_number == sequence_num_last_received_rtp_) | |
70 return; | |
71 | |
72 // Received RTP should not be in the list. | |
73 nack_list_.erase(sequence_number); | |
74 | |
75 // If this is an old sequence number, no more action is required, return. | |
76 if (IsNewerSequenceNumber(sequence_num_last_received_rtp_, sequence_number)) | |
77 return; | |
78 | |
79 UpdateSamplesPerPacket(sequence_number, timestamp); | |
80 | |
81 UpdateList(sequence_number); | |
82 | |
83 sequence_num_last_received_rtp_ = sequence_number; | |
84 timestamp_last_received_rtp_ = timestamp; | |
85 LimitNackListSize(); | |
86 } | |
87 | |
88 void Nack::UpdateSamplesPerPacket(uint16_t sequence_number_current_received_rtp, | |
89 uint32_t timestamp_current_received_rtp) { | |
90 uint32_t timestamp_increase = | |
91 timestamp_current_received_rtp - timestamp_last_received_rtp_; | |
92 uint16_t sequence_num_increase = | |
93 sequence_number_current_received_rtp - sequence_num_last_received_rtp_; | |
94 | |
95 samples_per_packet_ = timestamp_increase / sequence_num_increase; | |
96 } | |
97 | |
98 void Nack::UpdateList(uint16_t sequence_number_current_received_rtp) { | |
99 // Some of the packets which were considered late, now are considered missing. | |
100 ChangeFromLateToMissing(sequence_number_current_received_rtp); | |
101 | |
102 if (IsNewerSequenceNumber(sequence_number_current_received_rtp, | |
103 sequence_num_last_received_rtp_ + 1)) | |
104 AddToList(sequence_number_current_received_rtp); | |
105 } | |
106 | |
107 void Nack::ChangeFromLateToMissing( | |
108 uint16_t sequence_number_current_received_rtp) { | |
109 NackList::const_iterator lower_bound = | |
110 nack_list_.lower_bound(static_cast<uint16_t>( | |
111 sequence_number_current_received_rtp - nack_threshold_packets_)); | |
112 | |
113 for (NackList::iterator it = nack_list_.begin(); it != lower_bound; ++it) | |
114 it->second.is_missing = true; | |
115 } | |
116 | |
117 uint32_t Nack::EstimateTimestamp(uint16_t sequence_num) { | |
118 uint16_t sequence_num_diff = sequence_num - sequence_num_last_received_rtp_; | |
119 return sequence_num_diff * samples_per_packet_ + timestamp_last_received_rtp_; | |
120 } | |
121 | |
122 void Nack::AddToList(uint16_t sequence_number_current_received_rtp) { | |
123 assert(!any_rtp_decoded_ || | |
124 IsNewerSequenceNumber(sequence_number_current_received_rtp, | |
125 sequence_num_last_decoded_rtp_)); | |
126 | |
127 // Packets with sequence numbers older than |upper_bound_missing| are | |
128 // considered missing, and the rest are considered late. | |
129 uint16_t upper_bound_missing = | |
130 sequence_number_current_received_rtp - nack_threshold_packets_; | |
131 | |
132 for (uint16_t n = sequence_num_last_received_rtp_ + 1; | |
133 IsNewerSequenceNumber(sequence_number_current_received_rtp, n); ++n) { | |
134 bool is_missing = IsNewerSequenceNumber(upper_bound_missing, n); | |
135 uint32_t timestamp = EstimateTimestamp(n); | |
136 NackElement nack_element(TimeToPlay(timestamp), timestamp, is_missing); | |
137 nack_list_.insert(nack_list_.end(), std::make_pair(n, nack_element)); | |
138 } | |
139 } | |
140 | |
141 void Nack::UpdateEstimatedPlayoutTimeBy10ms() { | |
142 while (!nack_list_.empty() && | |
143 nack_list_.begin()->second.time_to_play_ms <= 10) | |
144 nack_list_.erase(nack_list_.begin()); | |
145 | |
146 for (NackList::iterator it = nack_list_.begin(); it != nack_list_.end(); ++it) | |
147 it->second.time_to_play_ms -= 10; | |
148 } | |
149 | |
150 void Nack::UpdateLastDecodedPacket(uint16_t sequence_number, | |
151 uint32_t timestamp) { | |
152 if (IsNewerSequenceNumber(sequence_number, sequence_num_last_decoded_rtp_) || | |
153 !any_rtp_decoded_) { | |
154 sequence_num_last_decoded_rtp_ = sequence_number; | |
155 timestamp_last_decoded_rtp_ = timestamp; | |
156 // Packets in the list with sequence numbers less than the | |
157 // sequence number of the decoded RTP should be removed from the lists. | |
158 // They will be discarded by the jitter buffer if they arrive. | |
159 nack_list_.erase(nack_list_.begin(), | |
160 nack_list_.upper_bound(sequence_num_last_decoded_rtp_)); | |
161 | |
162 // Update estimated time-to-play. | |
163 for (NackList::iterator it = nack_list_.begin(); it != nack_list_.end(); | |
164 ++it) | |
165 it->second.time_to_play_ms = TimeToPlay(it->second.estimated_timestamp); | |
166 } else { | |
167 assert(sequence_number == sequence_num_last_decoded_rtp_); | |
168 | |
169 // Same sequence number as before. 10 ms is elapsed, update estimations for | |
170 // time-to-play. | |
171 UpdateEstimatedPlayoutTimeBy10ms(); | |
172 | |
173 // Update timestamp for better estimate of time-to-play, for packets which | |
174 // are added to NACK list later on. | |
175 timestamp_last_decoded_rtp_ += sample_rate_khz_ * 10; | |
176 } | |
177 any_rtp_decoded_ = true; | |
178 } | |
179 | |
180 Nack::NackList Nack::GetNackList() const { | |
181 return nack_list_; | |
182 } | |
183 | |
184 void Nack::Reset() { | |
185 nack_list_.clear(); | |
186 | |
187 sequence_num_last_received_rtp_ = 0; | |
188 timestamp_last_received_rtp_ = 0; | |
189 any_rtp_received_ = false; | |
190 sequence_num_last_decoded_rtp_ = 0; | |
191 timestamp_last_decoded_rtp_ = 0; | |
192 any_rtp_decoded_ = false; | |
193 sample_rate_khz_ = kDefaultSampleRateKhz; | |
194 samples_per_packet_ = sample_rate_khz_ * kDefaultPacketSizeMs; | |
195 } | |
196 | |
197 void Nack::SetMaxNackListSize(size_t max_nack_list_size) { | |
198 RTC_CHECK_GT(max_nack_list_size, 0u); | |
199 // Ugly hack to get around the problem of passing static consts by reference. | |
200 const size_t kNackListSizeLimitLocal = Nack::kNackListSizeLimit; | |
201 RTC_CHECK_LE(max_nack_list_size, kNackListSizeLimitLocal); | |
202 | |
203 max_nack_list_size_ = max_nack_list_size; | |
204 LimitNackListSize(); | |
205 } | |
206 | |
207 void Nack::LimitNackListSize() { | |
208 uint16_t limit = sequence_num_last_received_rtp_ - | |
209 static_cast<uint16_t>(max_nack_list_size_) - 1; | |
210 nack_list_.erase(nack_list_.begin(), nack_list_.upper_bound(limit)); | |
211 } | |
212 | |
213 int64_t Nack::TimeToPlay(uint32_t timestamp) const { | |
214 uint32_t timestamp_increase = timestamp - timestamp_last_decoded_rtp_; | |
215 return timestamp_increase / sample_rate_khz_; | |
216 } | |
217 | |
218 // We don't erase elements with time-to-play shorter than round-trip-time. | |
219 std::vector<uint16_t> Nack::GetNackList(int64_t round_trip_time_ms) const { | |
220 RTC_DCHECK_GE(round_trip_time_ms, 0); | |
221 std::vector<uint16_t> sequence_numbers; | |
222 for (NackList::const_iterator it = nack_list_.begin(); it != nack_list_.end(); | |
223 ++it) { | |
224 if (it->second.is_missing && | |
225 it->second.time_to_play_ms > round_trip_time_ms) | |
226 sequence_numbers.push_back(it->first); | |
227 } | |
228 return sequence_numbers; | |
229 } | |
230 | |
231 } // namespace webrtc | |
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