| /* |
| * 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 "modules/include/module_common_types.h" |
| #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 |