modules/audio_coding/neteq/nack_tracker.cc - src/ - Git at Google

 /*
  *  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 <cstdint>
 #include <utility>

 #include "rtc_base/checks.h"
 #include "rtc_base/experiments/struct_parameters_parser.h"
 #include "rtc_base/logging.h"
 #include "system_wrappers/include/field_trial.h"

 namespace webrtc {
 namespace {

 const int kDefaultSampleRateKhz = 48;
 const int kMaxPacketSizeMs = 120;
 constexpr char kNackTrackerConfigFieldTrial[] =
     "WebRTC-Audio-NetEqNackTrackerConfig";

 }  // namespace

 NackTracker::Config::Config() {
   auto parser = StructParametersParser::Create(
       "packet_loss_forget_factor", &packet_loss_forget_factor,
       "ms_per_loss_percent", &ms_per_loss_percent, "never_nack_multiple_times",
       &never_nack_multiple_times, "require_valid_rtt", &require_valid_rtt,
       "max_loss_rate", &max_loss_rate);
   parser->Parse(
       webrtc::field_trial::FindFullName(kNackTrackerConfigFieldTrial));
   RTC_LOG(LS_INFO) << "Nack tracker config:"
                       " packet_loss_forget_factor="
                    << packet_loss_forget_factor
                    << " ms_per_loss_percent=" << ms_per_loss_percent
                    << " never_nack_multiple_times=" << never_nack_multiple_times
                    << " require_valid_rtt=" << require_valid_rtt
                    << " max_loss_rate=" << max_loss_rate;
 }

 NackTracker::NackTracker()
     : 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),
       max_nack_list_size_(kNackListSizeLimit) {}

 NackTracker::~NackTracker() = default;

 void NackTracker::UpdateSampleRate(int sample_rate_hz) {
   RTC_DCHECK_GT(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;

   UpdatePacketLossRate(sequence_number - sequence_num_last_received_rtp_ - 1);

   UpdateList(sequence_number, timestamp);

   sequence_num_last_received_rtp_ = sequence_number;
   timestamp_last_received_rtp_ = timestamp;
   LimitNackListSize();
 }

 absl::optional<int> NackTracker::GetSamplesPerPacket(
     uint16_t sequence_number_current_received_rtp,
     uint32_t timestamp_current_received_rtp) const {
   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_;

   int samples_per_packet = timestamp_increase / sequence_num_increase;
   if (samples_per_packet == 0 ||
       samples_per_packet > kMaxPacketSizeMs * sample_rate_khz_) {
     // Not a valid samples per packet.
     return absl::nullopt;
   }
   return samples_per_packet;
 }

 void NackTracker::UpdateList(uint16_t sequence_number_current_received_rtp,
                              uint32_t timestamp_current_received_rtp) {
   if (!IsNewerSequenceNumber(sequence_number_current_received_rtp,
                              sequence_num_last_received_rtp_ + 1)) {
     return;
   }
   RTC_DCHECK(!any_rtp_decoded_ ||
              IsNewerSequenceNumber(sequence_number_current_received_rtp,
                                    sequence_num_last_decoded_rtp_));

   absl::optional<int> samples_per_packet = GetSamplesPerPacket(
       sequence_number_current_received_rtp, timestamp_current_received_rtp);
   if (!samples_per_packet) {
     return;
   }

   for (uint16_t n = sequence_num_last_received_rtp_ + 1;
        IsNewerSequenceNumber(sequence_number_current_received_rtp, n); ++n) {
     uint32_t timestamp = EstimateTimestamp(n, *samples_per_packet);
     NackElement nack_element(TimeToPlay(timestamp), timestamp);
     nack_list_.insert(nack_list_.end(), std::make_pair(n, nack_element));
   }
 }

 uint32_t NackTracker::EstimateTimestamp(uint16_t sequence_num,
                                         int samples_per_packet) {
   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::UpdateLastDecodedPacket(uint16_t sequence_number,
                                           uint32_t timestamp) {
   any_rtp_decoded_ = true;
   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);
   }
 }

 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;
 }

 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) {
   RTC_DCHECK_GE(round_trip_time_ms, 0);
   std::vector<uint16_t> sequence_numbers;
   if (round_trip_time_ms == 0) {
     if (config_.require_valid_rtt) {
       return sequence_numbers;
     } else {
       round_trip_time_ms = config_.default_rtt_ms;
     }
   }
   if (packet_loss_rate_ >
       static_cast<uint32_t>(config_.max_loss_rate * (1 << 30))) {
     return sequence_numbers;
   }
   // The estimated packet loss is between 0 and 1, so we need to multiply by 100
   // here.
   int max_wait_ms =
       100.0 * config_.ms_per_loss_percent * packet_loss_rate_ / (1 << 30);
   for (NackList::const_iterator it = nack_list_.begin(); it != nack_list_.end();
        ++it) {
     int64_t time_since_packet_ms =
         (timestamp_last_received_rtp_ - it->second.estimated_timestamp) /
         sample_rate_khz_;
     if (it->second.time_to_play_ms > round_trip_time_ms ||
         time_since_packet_ms + round_trip_time_ms < max_wait_ms)
       sequence_numbers.push_back(it->first);
   }
   if (config_.never_nack_multiple_times) {
     nack_list_.clear();
   }
   return sequence_numbers;
 }

 void NackTracker::UpdatePacketLossRate(int packets_lost) {
   const uint64_t alpha_q30 = (1 << 30) * config_.packet_loss_forget_factor;
   // Exponential filter.
   packet_loss_rate_ = (alpha_q30 * packet_loss_rate_) >> 30;
   for (int i = 0; i < packets_lost; ++i) {
     packet_loss_rate_ =
         ((alpha_q30 * packet_loss_rate_) >> 30) + ((1 << 30) - alpha_q30);
   }
 }
 }  // namespace webrtc
	/*
	* 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 <cstdint>
	#include <utility>

	#include "rtc_base/checks.h"
	#include "rtc_base/experiments/struct_parameters_parser.h"
	#include "rtc_base/logging.h"
	#include "system_wrappers/include/field_trial.h"

	namespace webrtc {
	namespace {

	const int kDefaultSampleRateKhz = 48;
	const int kMaxPacketSizeMs = 120;
	constexpr char kNackTrackerConfigFieldTrial[] =
	"WebRTC-Audio-NetEqNackTrackerConfig";

	} // namespace

	NackTracker::Config::Config() {
	auto parser = StructParametersParser::Create(
	"packet_loss_forget_factor", &packet_loss_forget_factor,
	"ms_per_loss_percent", &ms_per_loss_percent, "never_nack_multiple_times",
	&never_nack_multiple_times, "require_valid_rtt", &require_valid_rtt,
	"max_loss_rate", &max_loss_rate);
	parser->Parse(
	webrtc::field_trial::FindFullName(kNackTrackerConfigFieldTrial));
	RTC_LOG(LS_INFO) << "Nack tracker config:"
	" packet_loss_forget_factor="
	<< packet_loss_forget_factor
	<< " ms_per_loss_percent=" << ms_per_loss_percent
	<< " never_nack_multiple_times=" << never_nack_multiple_times
	<< " require_valid_rtt=" << require_valid_rtt
	<< " max_loss_rate=" << max_loss_rate;
	}

	NackTracker::NackTracker()
	: 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),
	max_nack_list_size_(kNackListSizeLimit) {}

	NackTracker::~NackTracker() = default;

	void NackTracker::UpdateSampleRate(int sample_rate_hz) {
	RTC_DCHECK_GT(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;

	UpdatePacketLossRate(sequence_number - sequence_num_last_received_rtp_ - 1);

	UpdateList(sequence_number, timestamp);

	sequence_num_last_received_rtp_ = sequence_number;
	timestamp_last_received_rtp_ = timestamp;
	LimitNackListSize();
	}

	absl::optional<int> NackTracker::GetSamplesPerPacket(
	uint16_t sequence_number_current_received_rtp,
	uint32_t timestamp_current_received_rtp) const {
	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_;

	int samples_per_packet = timestamp_increase / sequence_num_increase;
	if (samples_per_packet == 0 \|\|
	samples_per_packet > kMaxPacketSizeMs * sample_rate_khz_) {
	// Not a valid samples per packet.
	return absl::nullopt;
	}
	return samples_per_packet;
	}

	void NackTracker::UpdateList(uint16_t sequence_number_current_received_rtp,
	uint32_t timestamp_current_received_rtp) {
	if (!IsNewerSequenceNumber(sequence_number_current_received_rtp,
	sequence_num_last_received_rtp_ + 1)) {
	return;
	}
	RTC_DCHECK(!any_rtp_decoded_ \|\|
	IsNewerSequenceNumber(sequence_number_current_received_rtp,
	sequence_num_last_decoded_rtp_));

	absl::optional<int> samples_per_packet = GetSamplesPerPacket(
	sequence_number_current_received_rtp, timestamp_current_received_rtp);
	if (!samples_per_packet) {
	return;
	}

	for (uint16_t n = sequence_num_last_received_rtp_ + 1;
	IsNewerSequenceNumber(sequence_number_current_received_rtp, n); ++n) {
	uint32_t timestamp = EstimateTimestamp(n, *samples_per_packet);
	NackElement nack_element(TimeToPlay(timestamp), timestamp);
	nack_list_.insert(nack_list_.end(), std::make_pair(n, nack_element));
	}
	}

	uint32_t NackTracker::EstimateTimestamp(uint16_t sequence_num,
	int samples_per_packet) {
	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::UpdateLastDecodedPacket(uint16_t sequence_number,
	uint32_t timestamp) {
	any_rtp_decoded_ = true;
	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);
	}
	}

	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;
	}

	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) {
	RTC_DCHECK_GE(round_trip_time_ms, 0);
	std::vector<uint16_t> sequence_numbers;
	if (round_trip_time_ms == 0) {
	if (config_.require_valid_rtt) {
	return sequence_numbers;
	} else {
	round_trip_time_ms = config_.default_rtt_ms;
	}
	}
	if (packet_loss_rate_ >
	static_cast<uint32_t>(config_.max_loss_rate * (1 << 30))) {
	return sequence_numbers;
	}
	// The estimated packet loss is between 0 and 1, so we need to multiply by 100
	// here.
	int max_wait_ms =
	100.0 * config_.ms_per_loss_percent * packet_loss_rate_ / (1 << 30);
	for (NackList::const_iterator it = nack_list_.begin(); it != nack_list_.end();
	++it) {
	int64_t time_since_packet_ms =
	(timestamp_last_received_rtp_ - it->second.estimated_timestamp) /
	sample_rate_khz_;
	if (it->second.time_to_play_ms > round_trip_time_ms \|\|
	time_since_packet_ms + round_trip_time_ms < max_wait_ms)
	sequence_numbers.push_back(it->first);
	}
	if (config_.never_nack_multiple_times) {
	nack_list_.clear();
	}
	return sequence_numbers;
	}

	void NackTracker::UpdatePacketLossRate(int packets_lost) {
	const uint64_t alpha_q30 = (1 << 30) * config_.packet_loss_forget_factor;
	// Exponential filter.
	packet_loss_rate_ = (alpha_q30 * packet_loss_rate_) >> 30;
	for (int i = 0; i < packets_lost; ++i) {
	packet_loss_rate_ =
	((alpha_q30 * packet_loss_rate_) >> 30) + ((1 << 30) - alpha_q30);
	}
	}
	} // namespace webrtc