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

 /*
  *  Copyright (c) 2012 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.
  */

 // This is the implementation of the PacketBuffer class. It is mostly based on
 // an STL list. The list is kept sorted at all times so that the next packet to
 // decode is at the beginning of the list.

 #include "modules/audio_coding/neteq/packet_buffer.h"

 #include <algorithm>
 #include <list>
 #include <memory>
 #include <type_traits>
 #include <utility>

 #include "api/audio_codecs/audio_decoder.h"
 #include "api/neteq/tick_timer.h"
 #include "modules/audio_coding/neteq/decoder_database.h"
 #include "modules/audio_coding/neteq/statistics_calculator.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/experiments/struct_parameters_parser.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/numerics/safe_conversions.h"
 #include "system_wrappers/include/field_trial.h"

 namespace webrtc {
 namespace {
 // Predicate used when inserting packets in the buffer list.
 // Operator() returns true when `packet` goes before `new_packet`.
 class NewTimestampIsLarger {
  public:
   explicit NewTimestampIsLarger(const Packet& new_packet)
       : new_packet_(new_packet) {}
   bool operator()(const Packet& packet) { return (new_packet_ >= packet); }

  private:
   const Packet& new_packet_;
 };

 // Returns true if both payload types are known to the decoder database, and
 // have the same sample rate.
 bool EqualSampleRates(uint8_t pt1,
                       uint8_t pt2,
                       const DecoderDatabase& decoder_database) {
   auto* di1 = decoder_database.GetDecoderInfo(pt1);
   auto* di2 = decoder_database.GetDecoderInfo(pt2);
   return di1 && di2 && di1->SampleRateHz() == di2->SampleRateHz();
 }

 void LogPacketDiscarded(int codec_level, StatisticsCalculator* stats) {
   RTC_CHECK(stats);
   if (codec_level > 0) {
     stats->SecondaryPacketsDiscarded(1);
   } else {
     stats->PacketsDiscarded(1);
   }
 }

 absl::optional<SmartFlushingConfig> GetSmartflushingConfig() {
   absl::optional<SmartFlushingConfig> result;
   std::string field_trial_string =
       field_trial::FindFullName("WebRTC-Audio-NetEqSmartFlushing");
   result = SmartFlushingConfig();
   bool enabled = false;
   auto parser = StructParametersParser::Create(
       "enabled", &enabled, "target_level_threshold_ms",
       &result->target_level_threshold_ms, "target_level_multiplier",
       &result->target_level_multiplier);
   parser->Parse(field_trial_string);
   if (!enabled) {
     return absl::nullopt;
   }
   RTC_LOG(LS_INFO) << "Using smart flushing, target_level_threshold_ms: "
                    << result->target_level_threshold_ms
                    << ", target_level_multiplier: "
                    << result->target_level_multiplier;
   return result;
 }

 }  // namespace

 PacketBuffer::PacketBuffer(size_t max_number_of_packets,
                            const TickTimer* tick_timer)
     : smart_flushing_config_(GetSmartflushingConfig()),
       max_number_of_packets_(max_number_of_packets),
       tick_timer_(tick_timer) {}

 // Destructor. All packets in the buffer will be destroyed.
 PacketBuffer::~PacketBuffer() {
   buffer_.clear();
 }

 // Flush the buffer. All packets in the buffer will be destroyed.
 void PacketBuffer::Flush(StatisticsCalculator* stats) {
   for (auto& p : buffer_) {
     LogPacketDiscarded(p.priority.codec_level, stats);
   }
   buffer_.clear();
   stats->FlushedPacketBuffer();
 }

 void PacketBuffer::PartialFlush(int target_level_ms,
                                 size_t sample_rate,
                                 size_t last_decoded_length,
                                 StatisticsCalculator* stats) {
   // Make sure that at least half the packet buffer capacity will be available
   // after the flush. This is done to avoid getting stuck if the target level is
   // very high.
   int target_level_samples =
       std::min(target_level_ms * sample_rate / 1000,
                max_number_of_packets_ * last_decoded_length / 2);
   // We should avoid flushing to very low levels.
   target_level_samples = std::max(
       target_level_samples, smart_flushing_config_->target_level_threshold_ms);
   while (GetSpanSamples(last_decoded_length, sample_rate, true) >
              static_cast<size_t>(target_level_samples) ||
          buffer_.size() > max_number_of_packets_ / 2) {
     LogPacketDiscarded(PeekNextPacket()->priority.codec_level, stats);
     buffer_.pop_front();
   }
 }

 bool PacketBuffer::Empty() const {
   return buffer_.empty();
 }

 int PacketBuffer::InsertPacket(Packet&& packet,
                                StatisticsCalculator* stats,
                                size_t last_decoded_length,
                                size_t sample_rate,
                                int target_level_ms,
                                const DecoderDatabase& decoder_database) {
   if (packet.empty()) {
     RTC_LOG(LS_WARNING) << "InsertPacket invalid packet";
     return kInvalidPacket;
   }

   RTC_DCHECK_GE(packet.priority.codec_level, 0);
   RTC_DCHECK_GE(packet.priority.red_level, 0);

   int return_val = kOK;

   packet.waiting_time = tick_timer_->GetNewStopwatch();

   // Perform a smart flush if the buffer size exceeds a multiple of the target
   // level.
   const size_t span_threshold =
       smart_flushing_config_
           ? smart_flushing_config_->target_level_multiplier *
                 std::max(smart_flushing_config_->target_level_threshold_ms,
                          target_level_ms) *
                 sample_rate / 1000
           : 0;
   const bool smart_flush =
       smart_flushing_config_.has_value() &&
       GetSpanSamples(last_decoded_length, sample_rate, true) >= span_threshold;
   if (buffer_.size() >= max_number_of_packets_ || smart_flush) {
     size_t buffer_size_before_flush = buffer_.size();
     if (smart_flushing_config_.has_value()) {
       // Flush down to the target level.
       PartialFlush(target_level_ms, sample_rate, last_decoded_length, stats);
       return_val = kPartialFlush;
     } else {
       // Buffer is full.
       Flush(stats);
       return_val = kFlushed;
     }
     RTC_LOG(LS_WARNING) << "Packet buffer flushed, "
                         << (buffer_size_before_flush - buffer_.size())
                         << " packets discarded.";
   }

   // Get an iterator pointing to the place in the buffer where the new packet
   // should be inserted. The list is searched from the back, since the most
   // likely case is that the new packet should be near the end of the list.
   PacketList::reverse_iterator rit = std::find_if(
       buffer_.rbegin(), buffer_.rend(), NewTimestampIsLarger(packet));

   // The new packet is to be inserted to the right of `rit`. If it has the same
   // timestamp as `rit`, which has a higher priority, do not insert the new
   // packet to list.
   if (rit != buffer_.rend() && packet.timestamp == rit->timestamp) {
     LogPacketDiscarded(packet.priority.codec_level, stats);
     return return_val;
   }

   // The new packet is to be inserted to the left of `it`. If it has the same
   // timestamp as `it`, which has a lower priority, replace `it` with the new
   // packet.
   PacketList::iterator it = rit.base();
   if (it != buffer_.end() && packet.timestamp == it->timestamp) {
     LogPacketDiscarded(it->priority.codec_level, stats);
     it = buffer_.erase(it);
   }
   buffer_.insert(it, std::move(packet));  // Insert the packet at that position.

   return return_val;
 }

 int PacketBuffer::InsertPacketList(
     PacketList* packet_list,
     const DecoderDatabase& decoder_database,
     absl::optional<uint8_t>* current_rtp_payload_type,
     absl::optional<uint8_t>* current_cng_rtp_payload_type,
     StatisticsCalculator* stats,
     size_t last_decoded_length,
     size_t sample_rate,
     int target_level_ms) {
   RTC_DCHECK(stats);
   bool flushed = false;
   for (auto& packet : *packet_list) {
     if (decoder_database.IsComfortNoise(packet.payload_type)) {
       if (*current_cng_rtp_payload_type &&
           **current_cng_rtp_payload_type != packet.payload_type) {
         // New CNG payload type implies new codec type.
         *current_rtp_payload_type = absl::nullopt;
         Flush(stats);
         flushed = true;
       }
       *current_cng_rtp_payload_type = packet.payload_type;
     } else if (!decoder_database.IsDtmf(packet.payload_type)) {
       // This must be speech.
       if ((*current_rtp_payload_type &&
            **current_rtp_payload_type != packet.payload_type) ||
           (*current_cng_rtp_payload_type &&
            !EqualSampleRates(packet.payload_type,
                              **current_cng_rtp_payload_type,
                              decoder_database))) {
         *current_cng_rtp_payload_type = absl::nullopt;
         Flush(stats);
         flushed = true;
       }
       *current_rtp_payload_type = packet.payload_type;
     }
     int return_val =
         InsertPacket(std::move(packet), stats, last_decoded_length, sample_rate,
                      target_level_ms, decoder_database);
     if (return_val == kFlushed) {
       // The buffer flushed, but this is not an error. We can still continue.
       flushed = true;
     } else if (return_val != kOK) {
       // An error occurred. Delete remaining packets in list and return.
       packet_list->clear();
       return return_val;
     }
   }
   packet_list->clear();
   return flushed ? kFlushed : kOK;
 }

 int PacketBuffer::NextTimestamp(uint32_t* next_timestamp) const {
   if (Empty()) {
     return kBufferEmpty;
   }
   if (!next_timestamp) {
     return kInvalidPointer;
   }
   *next_timestamp = buffer_.front().timestamp;
   return kOK;
 }

 int PacketBuffer::NextHigherTimestamp(uint32_t timestamp,
                                       uint32_t* next_timestamp) const {
   if (Empty()) {
     return kBufferEmpty;
   }
   if (!next_timestamp) {
     return kInvalidPointer;
   }
   PacketList::const_iterator it;
   for (it = buffer_.begin(); it != buffer_.end(); ++it) {
     if (it->timestamp >= timestamp) {
       // Found a packet matching the search.
       *next_timestamp = it->timestamp;
       return kOK;
     }
   }
   return kNotFound;
 }

 const Packet* PacketBuffer::PeekNextPacket() const {
   return buffer_.empty() ? nullptr : &buffer_.front();
 }

 absl::optional<Packet> PacketBuffer::GetNextPacket() {
   if (Empty()) {
     // Buffer is empty.
     return absl::nullopt;
   }

   absl::optional<Packet> packet(std::move(buffer_.front()));
   // Assert that the packet sanity checks in InsertPacket method works.
   RTC_DCHECK(!packet->empty());
   buffer_.pop_front();

   return packet;
 }

 int PacketBuffer::DiscardNextPacket(StatisticsCalculator* stats) {
   if (Empty()) {
     return kBufferEmpty;
   }
   // Assert that the packet sanity checks in InsertPacket method works.
   const Packet& packet = buffer_.front();
   RTC_DCHECK(!packet.empty());
   LogPacketDiscarded(packet.priority.codec_level, stats);
   buffer_.pop_front();
   return kOK;
 }

 void PacketBuffer::DiscardOldPackets(uint32_t timestamp_limit,
                                      uint32_t horizon_samples,
                                      StatisticsCalculator* stats) {
   buffer_.remove_if([timestamp_limit, horizon_samples, stats](const Packet& p) {
     if (timestamp_limit == p.timestamp ||
         !IsObsoleteTimestamp(p.timestamp, timestamp_limit, horizon_samples)) {
       return false;
     }
     LogPacketDiscarded(p.priority.codec_level, stats);
     return true;
   });
 }

 void PacketBuffer::DiscardAllOldPackets(uint32_t timestamp_limit,
                                         StatisticsCalculator* stats) {
   DiscardOldPackets(timestamp_limit, 0, stats);
 }

 void PacketBuffer::DiscardPacketsWithPayloadType(uint8_t payload_type,
                                                  StatisticsCalculator* stats) {
   buffer_.remove_if([payload_type, stats](const Packet& p) {
     if (p.payload_type != payload_type) {
       return false;
     }
     LogPacketDiscarded(p.priority.codec_level, stats);
     return true;
   });
 }

 size_t PacketBuffer::NumPacketsInBuffer() const {
   return buffer_.size();
 }

 size_t PacketBuffer::NumSamplesInBuffer(size_t last_decoded_length) const {
   size_t num_samples = 0;
   size_t last_duration = last_decoded_length;
   for (const Packet& packet : buffer_) {
     if (packet.frame) {
       // TODO(hlundin): Verify that it's fine to count all packets and remove
       // this check.
       if (packet.priority != Packet::Priority(0, 0)) {
         continue;
       }
       size_t duration = packet.frame->Duration();
       if (duration > 0) {
         last_duration = duration;  // Save the most up-to-date (valid) duration.
       }
     }
     num_samples += last_duration;
   }
   return num_samples;
 }

 size_t PacketBuffer::GetSpanSamples(size_t last_decoded_length,
                                     size_t sample_rate,
                                     bool count_dtx_waiting_time) const {
   if (buffer_.size() == 0) {
     return 0;
   }

   size_t span = buffer_.back().timestamp - buffer_.front().timestamp;
   if (buffer_.back().frame && buffer_.back().frame->Duration() > 0) {
     size_t duration = buffer_.back().frame->Duration();
     if (count_dtx_waiting_time && buffer_.back().frame->IsDtxPacket()) {
       size_t waiting_time_samples = rtc::dchecked_cast<size_t>(
           buffer_.back().waiting_time->ElapsedMs() * (sample_rate / 1000));
       duration = std::max(duration, waiting_time_samples);
     }
     span += duration;
   } else {
     span += last_decoded_length;
   }
   return span;
 }

 bool PacketBuffer::ContainsDtxOrCngPacket(
     const DecoderDatabase* decoder_database) const {
   RTC_DCHECK(decoder_database);
   for (const Packet& packet : buffer_) {
     if ((packet.frame && packet.frame->IsDtxPacket()) ||
         decoder_database->IsComfortNoise(packet.payload_type)) {
       return true;
     }
   }
   return false;
 }

 }  // namespace webrtc
	/*
	* Copyright (c) 2012 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.
	*/

	// This is the implementation of the PacketBuffer class. It is mostly based on
	// an STL list. The list is kept sorted at all times so that the next packet to
	// decode is at the beginning of the list.

	#include "modules/audio_coding/neteq/packet_buffer.h"

	#include <algorithm>
	#include <list>
	#include <memory>
	#include <type_traits>
	#include <utility>

	#include "api/audio_codecs/audio_decoder.h"
	#include "api/neteq/tick_timer.h"
	#include "modules/audio_coding/neteq/decoder_database.h"
	#include "modules/audio_coding/neteq/statistics_calculator.h"
	#include "rtc_base/checks.h"
	#include "rtc_base/experiments/struct_parameters_parser.h"
	#include "rtc_base/logging.h"
	#include "rtc_base/numerics/safe_conversions.h"
	#include "system_wrappers/include/field_trial.h"

	namespace webrtc {
	namespace {
	// Predicate used when inserting packets in the buffer list.
	// Operator() returns true when `packet` goes before `new_packet`.
	class NewTimestampIsLarger {
	public:
	explicit NewTimestampIsLarger(const Packet& new_packet)
	: new_packet_(new_packet) {}
	bool operator()(const Packet& packet) { return (new_packet_ >= packet); }

	private:
	const Packet& new_packet_;
	};

	// Returns true if both payload types are known to the decoder database, and
	// have the same sample rate.
	bool EqualSampleRates(uint8_t pt1,
	uint8_t pt2,
	const DecoderDatabase& decoder_database) {
	auto* di1 = decoder_database.GetDecoderInfo(pt1);
	auto* di2 = decoder_database.GetDecoderInfo(pt2);
	return di1 && di2 && di1->SampleRateHz() == di2->SampleRateHz();
	}

	void LogPacketDiscarded(int codec_level, StatisticsCalculator* stats) {
	RTC_CHECK(stats);
	if (codec_level > 0) {
	stats->SecondaryPacketsDiscarded(1);
	} else {
	stats->PacketsDiscarded(1);
	}
	}

	absl::optional<SmartFlushingConfig> GetSmartflushingConfig() {
	absl::optional<SmartFlushingConfig> result;
	std::string field_trial_string =
	field_trial::FindFullName("WebRTC-Audio-NetEqSmartFlushing");
	result = SmartFlushingConfig();
	bool enabled = false;
	auto parser = StructParametersParser::Create(
	"enabled", &enabled, "target_level_threshold_ms",
	&result->target_level_threshold_ms, "target_level_multiplier",
	&result->target_level_multiplier);
	parser->Parse(field_trial_string);
	if (!enabled) {
	return absl::nullopt;
	}
	RTC_LOG(LS_INFO) << "Using smart flushing, target_level_threshold_ms: "
	<< result->target_level_threshold_ms
	<< ", target_level_multiplier: "
	<< result->target_level_multiplier;
	return result;
	}

	} // namespace

	PacketBuffer::PacketBuffer(size_t max_number_of_packets,
	const TickTimer* tick_timer)
	: smart_flushing_config_(GetSmartflushingConfig()),
	max_number_of_packets_(max_number_of_packets),
	tick_timer_(tick_timer) {}

	// Destructor. All packets in the buffer will be destroyed.
	PacketBuffer::~PacketBuffer() {
	buffer_.clear();
	}

	// Flush the buffer. All packets in the buffer will be destroyed.
	void PacketBuffer::Flush(StatisticsCalculator* stats) {
	for (auto& p : buffer_) {
	LogPacketDiscarded(p.priority.codec_level, stats);
	}
	buffer_.clear();
	stats->FlushedPacketBuffer();
	}

	void PacketBuffer::PartialFlush(int target_level_ms,
	size_t sample_rate,
	size_t last_decoded_length,
	StatisticsCalculator* stats) {
	// Make sure that at least half the packet buffer capacity will be available
	// after the flush. This is done to avoid getting stuck if the target level is
	// very high.
	int target_level_samples =
	std::min(target_level_ms * sample_rate / 1000,
	max_number_of_packets_ * last_decoded_length / 2);
	// We should avoid flushing to very low levels.
	target_level_samples = std::max(
	target_level_samples, smart_flushing_config_->target_level_threshold_ms);
	while (GetSpanSamples(last_decoded_length, sample_rate, true) >
	static_cast<size_t>(target_level_samples) \|\|
	buffer_.size() > max_number_of_packets_ / 2) {
	LogPacketDiscarded(PeekNextPacket()->priority.codec_level, stats);
	buffer_.pop_front();
	}
	}

	bool PacketBuffer::Empty() const {
	return buffer_.empty();
	}

	int PacketBuffer::InsertPacket(Packet&& packet,
	StatisticsCalculator* stats,
	size_t last_decoded_length,
	size_t sample_rate,
	int target_level_ms,
	const DecoderDatabase& decoder_database) {
	if (packet.empty()) {
	RTC_LOG(LS_WARNING) << "InsertPacket invalid packet";
	return kInvalidPacket;
	}

	RTC_DCHECK_GE(packet.priority.codec_level, 0);
	RTC_DCHECK_GE(packet.priority.red_level, 0);

	int return_val = kOK;

	packet.waiting_time = tick_timer_->GetNewStopwatch();

	// Perform a smart flush if the buffer size exceeds a multiple of the target
	// level.
	const size_t span_threshold =
	smart_flushing_config_
	? smart_flushing_config_->target_level_multiplier *
	std::max(smart_flushing_config_->target_level_threshold_ms,
	target_level_ms) *
	sample_rate / 1000
	: 0;
	const bool smart_flush =
	smart_flushing_config_.has_value() &&
	GetSpanSamples(last_decoded_length, sample_rate, true) >= span_threshold;
	if (buffer_.size() >= max_number_of_packets_ \|\| smart_flush) {
	size_t buffer_size_before_flush = buffer_.size();
	if (smart_flushing_config_.has_value()) {
	// Flush down to the target level.
	PartialFlush(target_level_ms, sample_rate, last_decoded_length, stats);
	return_val = kPartialFlush;
	} else {
	// Buffer is full.
	Flush(stats);
	return_val = kFlushed;
	}
	RTC_LOG(LS_WARNING) << "Packet buffer flushed, "
	<< (buffer_size_before_flush - buffer_.size())
	<< " packets discarded.";
	}

	// Get an iterator pointing to the place in the buffer where the new packet
	// should be inserted. The list is searched from the back, since the most
	// likely case is that the new packet should be near the end of the list.
	PacketList::reverse_iterator rit = std::find_if(
	buffer_.rbegin(), buffer_.rend(), NewTimestampIsLarger(packet));

	// The new packet is to be inserted to the right of `rit`. If it has the same
	// timestamp as `rit`, which has a higher priority, do not insert the new
	// packet to list.
	if (rit != buffer_.rend() && packet.timestamp == rit->timestamp) {
	LogPacketDiscarded(packet.priority.codec_level, stats);
	return return_val;
	}

	// The new packet is to be inserted to the left of `it`. If it has the same
	// timestamp as `it`, which has a lower priority, replace `it` with the new
	// packet.
	PacketList::iterator it = rit.base();
	if (it != buffer_.end() && packet.timestamp == it->timestamp) {
	LogPacketDiscarded(it->priority.codec_level, stats);
	it = buffer_.erase(it);
	}
	buffer_.insert(it, std::move(packet)); // Insert the packet at that position.

	return return_val;
	}

	int PacketBuffer::InsertPacketList(
	PacketList* packet_list,
	const DecoderDatabase& decoder_database,
	absl::optional<uint8_t>* current_rtp_payload_type,
	absl::optional<uint8_t>* current_cng_rtp_payload_type,
	StatisticsCalculator* stats,
	size_t last_decoded_length,
	size_t sample_rate,
	int target_level_ms) {
	RTC_DCHECK(stats);
	bool flushed = false;
	for (auto& packet : *packet_list) {
	if (decoder_database.IsComfortNoise(packet.payload_type)) {
	if (*current_cng_rtp_payload_type &&
	**current_cng_rtp_payload_type != packet.payload_type) {
	// New CNG payload type implies new codec type.
	*current_rtp_payload_type = absl::nullopt;
	Flush(stats);
	flushed = true;
	}
	*current_cng_rtp_payload_type = packet.payload_type;
	} else if (!decoder_database.IsDtmf(packet.payload_type)) {
	// This must be speech.
	if ((*current_rtp_payload_type &&
	**current_rtp_payload_type != packet.payload_type) \|\|
	(*current_cng_rtp_payload_type &&
	!EqualSampleRates(packet.payload_type,
	**current_cng_rtp_payload_type,
	decoder_database))) {
	*current_cng_rtp_payload_type = absl::nullopt;
	Flush(stats);
	flushed = true;
	}
	*current_rtp_payload_type = packet.payload_type;
	}
	int return_val =
	InsertPacket(std::move(packet), stats, last_decoded_length, sample_rate,
	target_level_ms, decoder_database);
	if (return_val == kFlushed) {
	// The buffer flushed, but this is not an error. We can still continue.
	flushed = true;
	} else if (return_val != kOK) {
	// An error occurred. Delete remaining packets in list and return.
	packet_list->clear();
	return return_val;
	}
	}
	packet_list->clear();
	return flushed ? kFlushed : kOK;
	}

	int PacketBuffer::NextTimestamp(uint32_t* next_timestamp) const {
	if (Empty()) {
	return kBufferEmpty;
	}
	if (!next_timestamp) {
	return kInvalidPointer;
	}
	*next_timestamp = buffer_.front().timestamp;
	return kOK;
	}

	int PacketBuffer::NextHigherTimestamp(uint32_t timestamp,
	uint32_t* next_timestamp) const {
	if (Empty()) {
	return kBufferEmpty;
	}
	if (!next_timestamp) {
	return kInvalidPointer;
	}
	PacketList::const_iterator it;
	for (it = buffer_.begin(); it != buffer_.end(); ++it) {
	if (it->timestamp >= timestamp) {
	// Found a packet matching the search.
	*next_timestamp = it->timestamp;
	return kOK;
	}
	}
	return kNotFound;
	}

	const Packet* PacketBuffer::PeekNextPacket() const {
	return buffer_.empty() ? nullptr : &buffer_.front();
	}

	absl::optional<Packet> PacketBuffer::GetNextPacket() {
	if (Empty()) {
	// Buffer is empty.
	return absl::nullopt;
	}

	absl::optional<Packet> packet(std::move(buffer_.front()));
	// Assert that the packet sanity checks in InsertPacket method works.
	RTC_DCHECK(!packet->empty());
	buffer_.pop_front();

	return packet;
	}

	int PacketBuffer::DiscardNextPacket(StatisticsCalculator* stats) {
	if (Empty()) {
	return kBufferEmpty;
	}
	// Assert that the packet sanity checks in InsertPacket method works.
	const Packet& packet = buffer_.front();
	RTC_DCHECK(!packet.empty());
	LogPacketDiscarded(packet.priority.codec_level, stats);
	buffer_.pop_front();
	return kOK;
	}

	void PacketBuffer::DiscardOldPackets(uint32_t timestamp_limit,
	uint32_t horizon_samples,
	StatisticsCalculator* stats) {
	buffer_.remove_if([timestamp_limit, horizon_samples, stats](const Packet& p) {
	if (timestamp_limit == p.timestamp \|\|
	!IsObsoleteTimestamp(p.timestamp, timestamp_limit, horizon_samples)) {
	return false;
	}
	LogPacketDiscarded(p.priority.codec_level, stats);
	return true;
	});
	}

	void PacketBuffer::DiscardAllOldPackets(uint32_t timestamp_limit,
	StatisticsCalculator* stats) {
	DiscardOldPackets(timestamp_limit, 0, stats);
	}

	void PacketBuffer::DiscardPacketsWithPayloadType(uint8_t payload_type,
	StatisticsCalculator* stats) {
	buffer_.remove_if([payload_type, stats](const Packet& p) {
	if (p.payload_type != payload_type) {
	return false;
	}
	LogPacketDiscarded(p.priority.codec_level, stats);
	return true;
	});
	}

	size_t PacketBuffer::NumPacketsInBuffer() const {
	return buffer_.size();
	}

	size_t PacketBuffer::NumSamplesInBuffer(size_t last_decoded_length) const {
	size_t num_samples = 0;
	size_t last_duration = last_decoded_length;
	for (const Packet& packet : buffer_) {
	if (packet.frame) {
	// TODO(hlundin): Verify that it's fine to count all packets and remove
	// this check.
	if (packet.priority != Packet::Priority(0, 0)) {
	continue;
	}
	size_t duration = packet.frame->Duration();
	if (duration > 0) {
	last_duration = duration; // Save the most up-to-date (valid) duration.
	}
	}
	num_samples += last_duration;
	}
	return num_samples;
	}

	size_t PacketBuffer::GetSpanSamples(size_t last_decoded_length,
	size_t sample_rate,
	bool count_dtx_waiting_time) const {
	if (buffer_.size() == 0) {
	return 0;
	}

	size_t span = buffer_.back().timestamp - buffer_.front().timestamp;
	if (buffer_.back().frame && buffer_.back().frame->Duration() > 0) {
	size_t duration = buffer_.back().frame->Duration();
	if (count_dtx_waiting_time && buffer_.back().frame->IsDtxPacket()) {
	size_t waiting_time_samples = rtc::dchecked_cast<size_t>(
	buffer_.back().waiting_time->ElapsedMs() * (sample_rate / 1000));
	duration = std::max(duration, waiting_time_samples);
	}
	span += duration;
	} else {
	span += last_decoded_length;
	}
	return span;
	}

	bool PacketBuffer::ContainsDtxOrCngPacket(
	const DecoderDatabase* decoder_database) const {
	RTC_DCHECK(decoder_database);
	for (const Packet& packet : buffer_) {
	if ((packet.frame && packet.frame->IsDtxPacket()) \|\|
	decoder_database->IsComfortNoise(packet.payload_type)) {
	return true;
	}
	}
	return false;
	}

	} // namespace webrtc