| /* | 
 |  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. | 
 |  * | 
 |  *  Use of this source code is governed by a BSD-style license | 
 |  *  that can be found in the LICENSE file in the root of the source | 
 |  *  tree. An additional intellectual property rights grant can be found | 
 |  *  in the file PATENTS.  All contributing project authors may | 
 |  *  be found in the AUTHORS file in the root of the source tree. | 
 |  */ | 
 |  | 
 | #include "modules/audio_coding/neteq/nack_tracker.h" | 
 |  | 
 | #include <assert.h>  // For assert. | 
 |  | 
 | #include <algorithm>  // For std::max. | 
 |  | 
 | #include "rtc_base/checks.h" | 
 |  | 
 | namespace webrtc { | 
 | namespace { | 
 |  | 
 | const int kDefaultSampleRateKhz = 48; | 
 | const int kDefaultPacketSizeMs = 20; | 
 |  | 
 | }  // namespace | 
 |  | 
 | NackTracker::NackTracker(int nack_threshold_packets) | 
 |     : nack_threshold_packets_(nack_threshold_packets), | 
 |       sequence_num_last_received_rtp_(0), | 
 |       timestamp_last_received_rtp_(0), | 
 |       any_rtp_received_(false), | 
 |       sequence_num_last_decoded_rtp_(0), | 
 |       timestamp_last_decoded_rtp_(0), | 
 |       any_rtp_decoded_(false), | 
 |       sample_rate_khz_(kDefaultSampleRateKhz), | 
 |       samples_per_packet_(sample_rate_khz_ * kDefaultPacketSizeMs), | 
 |       max_nack_list_size_(kNackListSizeLimit) {} | 
 |  | 
 | NackTracker::~NackTracker() = default; | 
 |  | 
 | NackTracker* NackTracker::Create(int nack_threshold_packets) { | 
 |   return new NackTracker(nack_threshold_packets); | 
 | } | 
 |  | 
 | void NackTracker::UpdateSampleRate(int sample_rate_hz) { | 
 |   assert(sample_rate_hz > 0); | 
 |   sample_rate_khz_ = sample_rate_hz / 1000; | 
 | } | 
 |  | 
 | void NackTracker::UpdateLastReceivedPacket(uint16_t sequence_number, | 
 |                                            uint32_t timestamp) { | 
 |   // Just record the value of sequence number and timestamp if this is the | 
 |   // first packet. | 
 |   if (!any_rtp_received_) { | 
 |     sequence_num_last_received_rtp_ = sequence_number; | 
 |     timestamp_last_received_rtp_ = timestamp; | 
 |     any_rtp_received_ = true; | 
 |     // If no packet is decoded, to have a reasonable estimate of time-to-play | 
 |     // use the given values. | 
 |     if (!any_rtp_decoded_) { | 
 |       sequence_num_last_decoded_rtp_ = sequence_number; | 
 |       timestamp_last_decoded_rtp_ = timestamp; | 
 |     } | 
 |     return; | 
 |   } | 
 |  | 
 |   if (sequence_number == sequence_num_last_received_rtp_) | 
 |     return; | 
 |  | 
 |   // Received RTP should not be in the list. | 
 |   nack_list_.erase(sequence_number); | 
 |  | 
 |   // If this is an old sequence number, no more action is required, return. | 
 |   if (IsNewerSequenceNumber(sequence_num_last_received_rtp_, sequence_number)) | 
 |     return; | 
 |  | 
 |   UpdateSamplesPerPacket(sequence_number, timestamp); | 
 |  | 
 |   UpdateList(sequence_number); | 
 |  | 
 |   sequence_num_last_received_rtp_ = sequence_number; | 
 |   timestamp_last_received_rtp_ = timestamp; | 
 |   LimitNackListSize(); | 
 | } | 
 |  | 
 | void NackTracker::UpdateSamplesPerPacket( | 
 |     uint16_t sequence_number_current_received_rtp, | 
 |     uint32_t timestamp_current_received_rtp) { | 
 |   uint32_t timestamp_increase = | 
 |       timestamp_current_received_rtp - timestamp_last_received_rtp_; | 
 |   uint16_t sequence_num_increase = | 
 |       sequence_number_current_received_rtp - sequence_num_last_received_rtp_; | 
 |  | 
 |   samples_per_packet_ = timestamp_increase / sequence_num_increase; | 
 | } | 
 |  | 
 | void NackTracker::UpdateList(uint16_t sequence_number_current_received_rtp) { | 
 |   // Some of the packets which were considered late, now are considered missing. | 
 |   ChangeFromLateToMissing(sequence_number_current_received_rtp); | 
 |  | 
 |   if (IsNewerSequenceNumber(sequence_number_current_received_rtp, | 
 |                             sequence_num_last_received_rtp_ + 1)) | 
 |     AddToList(sequence_number_current_received_rtp); | 
 | } | 
 |  | 
 | void NackTracker::ChangeFromLateToMissing( | 
 |     uint16_t sequence_number_current_received_rtp) { | 
 |   NackList::const_iterator lower_bound = | 
 |       nack_list_.lower_bound(static_cast<uint16_t>( | 
 |           sequence_number_current_received_rtp - nack_threshold_packets_)); | 
 |  | 
 |   for (NackList::iterator it = nack_list_.begin(); it != lower_bound; ++it) | 
 |     it->second.is_missing = true; | 
 | } | 
 |  | 
 | uint32_t NackTracker::EstimateTimestamp(uint16_t sequence_num) { | 
 |   uint16_t sequence_num_diff = sequence_num - sequence_num_last_received_rtp_; | 
 |   return sequence_num_diff * samples_per_packet_ + timestamp_last_received_rtp_; | 
 | } | 
 |  | 
 | void NackTracker::AddToList(uint16_t sequence_number_current_received_rtp) { | 
 |   assert(!any_rtp_decoded_ || | 
 |          IsNewerSequenceNumber(sequence_number_current_received_rtp, | 
 |                                sequence_num_last_decoded_rtp_)); | 
 |  | 
 |   // Packets with sequence numbers older than |upper_bound_missing| are | 
 |   // considered missing, and the rest are considered late. | 
 |   uint16_t upper_bound_missing = | 
 |       sequence_number_current_received_rtp - nack_threshold_packets_; | 
 |  | 
 |   for (uint16_t n = sequence_num_last_received_rtp_ + 1; | 
 |        IsNewerSequenceNumber(sequence_number_current_received_rtp, n); ++n) { | 
 |     bool is_missing = IsNewerSequenceNumber(upper_bound_missing, n); | 
 |     uint32_t timestamp = EstimateTimestamp(n); | 
 |     NackElement nack_element(TimeToPlay(timestamp), timestamp, is_missing); | 
 |     nack_list_.insert(nack_list_.end(), std::make_pair(n, nack_element)); | 
 |   } | 
 | } | 
 |  | 
 | void NackTracker::UpdateEstimatedPlayoutTimeBy10ms() { | 
 |   while (!nack_list_.empty() && | 
 |          nack_list_.begin()->second.time_to_play_ms <= 10) | 
 |     nack_list_.erase(nack_list_.begin()); | 
 |  | 
 |   for (NackList::iterator it = nack_list_.begin(); it != nack_list_.end(); ++it) | 
 |     it->second.time_to_play_ms -= 10; | 
 | } | 
 |  | 
 | void NackTracker::UpdateLastDecodedPacket(uint16_t sequence_number, | 
 |                                           uint32_t timestamp) { | 
 |   if (IsNewerSequenceNumber(sequence_number, sequence_num_last_decoded_rtp_) || | 
 |       !any_rtp_decoded_) { | 
 |     sequence_num_last_decoded_rtp_ = sequence_number; | 
 |     timestamp_last_decoded_rtp_ = timestamp; | 
 |     // Packets in the list with sequence numbers less than the | 
 |     // sequence number of the decoded RTP should be removed from the lists. | 
 |     // They will be discarded by the jitter buffer if they arrive. | 
 |     nack_list_.erase(nack_list_.begin(), | 
 |                      nack_list_.upper_bound(sequence_num_last_decoded_rtp_)); | 
 |  | 
 |     // Update estimated time-to-play. | 
 |     for (NackList::iterator it = nack_list_.begin(); it != nack_list_.end(); | 
 |          ++it) | 
 |       it->second.time_to_play_ms = TimeToPlay(it->second.estimated_timestamp); | 
 |   } else { | 
 |     assert(sequence_number == sequence_num_last_decoded_rtp_); | 
 |  | 
 |     // Same sequence number as before. 10 ms is elapsed, update estimations for | 
 |     // time-to-play. | 
 |     UpdateEstimatedPlayoutTimeBy10ms(); | 
 |  | 
 |     // Update timestamp for better estimate of time-to-play, for packets which | 
 |     // are added to NACK list later on. | 
 |     timestamp_last_decoded_rtp_ += sample_rate_khz_ * 10; | 
 |   } | 
 |   any_rtp_decoded_ = true; | 
 | } | 
 |  | 
 | NackTracker::NackList NackTracker::GetNackList() const { | 
 |   return nack_list_; | 
 | } | 
 |  | 
 | void NackTracker::Reset() { | 
 |   nack_list_.clear(); | 
 |  | 
 |   sequence_num_last_received_rtp_ = 0; | 
 |   timestamp_last_received_rtp_ = 0; | 
 |   any_rtp_received_ = false; | 
 |   sequence_num_last_decoded_rtp_ = 0; | 
 |   timestamp_last_decoded_rtp_ = 0; | 
 |   any_rtp_decoded_ = false; | 
 |   sample_rate_khz_ = kDefaultSampleRateKhz; | 
 |   samples_per_packet_ = sample_rate_khz_ * kDefaultPacketSizeMs; | 
 | } | 
 |  | 
 | void NackTracker::SetMaxNackListSize(size_t max_nack_list_size) { | 
 |   RTC_CHECK_GT(max_nack_list_size, 0); | 
 |   // Ugly hack to get around the problem of passing static consts by reference. | 
 |   const size_t kNackListSizeLimitLocal = NackTracker::kNackListSizeLimit; | 
 |   RTC_CHECK_LE(max_nack_list_size, kNackListSizeLimitLocal); | 
 |  | 
 |   max_nack_list_size_ = max_nack_list_size; | 
 |   LimitNackListSize(); | 
 | } | 
 |  | 
 | void NackTracker::LimitNackListSize() { | 
 |   uint16_t limit = sequence_num_last_received_rtp_ - | 
 |                    static_cast<uint16_t>(max_nack_list_size_) - 1; | 
 |   nack_list_.erase(nack_list_.begin(), nack_list_.upper_bound(limit)); | 
 | } | 
 |  | 
 | int64_t NackTracker::TimeToPlay(uint32_t timestamp) const { | 
 |   uint32_t timestamp_increase = timestamp - timestamp_last_decoded_rtp_; | 
 |   return timestamp_increase / sample_rate_khz_; | 
 | } | 
 |  | 
 | // We don't erase elements with time-to-play shorter than round-trip-time. | 
 | std::vector<uint16_t> NackTracker::GetNackList( | 
 |     int64_t round_trip_time_ms) const { | 
 |   RTC_DCHECK_GE(round_trip_time_ms, 0); | 
 |   std::vector<uint16_t> sequence_numbers; | 
 |   for (NackList::const_iterator it = nack_list_.begin(); it != nack_list_.end(); | 
 |        ++it) { | 
 |     if (it->second.is_missing && | 
 |         it->second.time_to_play_ms > round_trip_time_ms) | 
 |       sequence_numbers.push_back(it->first); | 
 |   } | 
 |   return sequence_numbers; | 
 | } | 
 |  | 
 | }  // namespace webrtc |