212 lines
8.6 KiB
C
212 lines
8.6 KiB
C
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/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_CODING_NETEQ_NACK_TRACKER_H_
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#define MODULES_AUDIO_CODING_NETEQ_NACK_TRACKER_H_
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#include <stddef.h>
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#include <stdint.h>
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#include <map>
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#include <vector>
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#include "modules/include/module_common_types_public.h"
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#include "rtc_base/gtest_prod_util.h"
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//
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// The NackTracker class keeps track of the lost packets, an estimate of
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// time-to-play for each packet is also given.
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//
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// Every time a packet is pushed into NetEq, LastReceivedPacket() has to be
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// called to update the NACK list.
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//
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// Every time 10ms audio is pulled from NetEq LastDecodedPacket() should be
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// called, and time-to-play is updated at that moment.
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//
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// If packet N is received, any packet prior to |N - NackThreshold| which is not
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// arrived is considered lost, and should be labeled as "missing" (the size of
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// the list might be limited and older packet eliminated from the list). Packets
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// |N - NackThreshold|, |N - NackThreshold + 1|, ..., |N - 1| are considered
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// "late." A "late" packet with sequence number K is changed to "missing" any
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// time a packet with sequence number newer than |K + NackList| is arrived.
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//
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// The NackTracker class has to know about the sample rate of the packets to
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// compute time-to-play. So sample rate should be set as soon as the first
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// packet is received. If there is a change in the receive codec (sender changes
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// codec) then NackTracker should be reset. This is because NetEQ would flush
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// its buffer and re-transmission is meaning less for old packet. Therefore, in
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// that case, after reset the sampling rate has to be updated.
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//
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// Thread Safety
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// =============
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// Please note that this class in not thread safe. The class must be protected
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// if different APIs are called from different threads.
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//
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namespace webrtc {
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class NackTracker {
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public:
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// A limit for the size of the NACK list.
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static const size_t kNackListSizeLimit = 500; // 10 seconds for 20 ms frame
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// packets.
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// Factory method.
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static NackTracker* Create(int nack_threshold_packets);
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~NackTracker();
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// Set a maximum for the size of the NACK list. If the last received packet
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// has sequence number of N, then NACK list will not contain any element
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// with sequence number earlier than N - |max_nack_list_size|.
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//
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// The largest maximum size is defined by |kNackListSizeLimit|
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void SetMaxNackListSize(size_t max_nack_list_size);
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// Set the sampling rate.
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//
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// If associated sampling rate of the received packets is changed, call this
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// function to update sampling rate. Note that if there is any change in
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// received codec then NetEq will flush its buffer and NACK has to be reset.
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// After Reset() is called sampling rate has to be set.
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void UpdateSampleRate(int sample_rate_hz);
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// Update the sequence number and the timestamp of the last decoded RTP. This
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// API should be called every time 10 ms audio is pulled from NetEq.
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void UpdateLastDecodedPacket(uint16_t sequence_number, uint32_t timestamp);
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// Update the sequence number and the timestamp of the last received RTP. This
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// API should be called every time a packet pushed into ACM.
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void UpdateLastReceivedPacket(uint16_t sequence_number, uint32_t timestamp);
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// Get a list of "missing" packets which have expected time-to-play larger
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// than the given round-trip-time (in milliseconds).
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// Note: Late packets are not included.
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std::vector<uint16_t> GetNackList(int64_t round_trip_time_ms) const;
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// Reset to default values. The NACK list is cleared.
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// |nack_threshold_packets_| & |max_nack_list_size_| preserve their values.
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void Reset();
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private:
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// This test need to access the private method GetNackList().
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FRIEND_TEST_ALL_PREFIXES(NackTrackerTest, EstimateTimestampAndTimeToPlay);
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struct NackElement {
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NackElement(int64_t initial_time_to_play_ms,
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uint32_t initial_timestamp,
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bool missing)
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: time_to_play_ms(initial_time_to_play_ms),
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estimated_timestamp(initial_timestamp),
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is_missing(missing) {}
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// Estimated time (ms) left for this packet to be decoded. This estimate is
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// updated every time jitter buffer decodes a packet.
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int64_t time_to_play_ms;
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// A guess about the timestamp of the missing packet, it is used for
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// estimation of |time_to_play_ms|. The estimate might be slightly wrong if
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// there has been frame-size change since the last received packet and the
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// missing packet. However, the risk of this is low, and in case of such
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// errors, there will be a minor misestimation in time-to-play of missing
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// packets. This will have a very minor effect on NACK performance.
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uint32_t estimated_timestamp;
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// True if the packet is considered missing. Otherwise indicates packet is
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// late.
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bool is_missing;
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};
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class NackListCompare {
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public:
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bool operator()(uint16_t sequence_number_old,
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uint16_t sequence_number_new) const {
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return IsNewerSequenceNumber(sequence_number_new, sequence_number_old);
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}
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};
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typedef std::map<uint16_t, NackElement, NackListCompare> NackList;
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// Constructor.
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explicit NackTracker(int nack_threshold_packets);
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// This API is used only for testing to assess whether time-to-play is
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// computed correctly.
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NackList GetNackList() const;
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// Given the |sequence_number_current_received_rtp| of currently received RTP,
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// recognize packets which are not arrive and add to the list.
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void AddToList(uint16_t sequence_number_current_received_rtp);
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// This function subtracts 10 ms of time-to-play for all packets in NACK list.
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// This is called when 10 ms elapsed with no new RTP packet decoded.
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void UpdateEstimatedPlayoutTimeBy10ms();
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// Given the |sequence_number_current_received_rtp| and
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// |timestamp_current_received_rtp| of currently received RTP update number
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// of samples per packet.
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void UpdateSamplesPerPacket(uint16_t sequence_number_current_received_rtp,
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uint32_t timestamp_current_received_rtp);
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// Given the |sequence_number_current_received_rtp| of currently received RTP
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// update the list. That is; some packets will change from late to missing,
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// some packets are inserted as missing and some inserted as late.
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void UpdateList(uint16_t sequence_number_current_received_rtp);
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// Packets which are considered late for too long (according to
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// |nack_threshold_packets_|) are flagged as missing.
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void ChangeFromLateToMissing(uint16_t sequence_number_current_received_rtp);
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// Packets which have sequence number older that
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// |sequence_num_last_received_rtp_| - |max_nack_list_size_| are removed
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// from the NACK list.
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void LimitNackListSize();
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// Estimate timestamp of a missing packet given its sequence number.
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uint32_t EstimateTimestamp(uint16_t sequence_number);
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// Compute time-to-play given a timestamp.
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int64_t TimeToPlay(uint32_t timestamp) const;
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// If packet N is arrived, any packet prior to N - |nack_threshold_packets_|
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// which is not arrived is considered missing, and should be in NACK list.
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// Also any packet in the range of N-1 and N - |nack_threshold_packets_|,
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// exclusive, which is not arrived is considered late, and should should be
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// in the list of late packets.
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const int nack_threshold_packets_;
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// Valid if a packet is received.
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uint16_t sequence_num_last_received_rtp_;
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uint32_t timestamp_last_received_rtp_;
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bool any_rtp_received_; // If any packet received.
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// Valid if a packet is decoded.
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uint16_t sequence_num_last_decoded_rtp_;
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uint32_t timestamp_last_decoded_rtp_;
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bool any_rtp_decoded_; // If any packet decoded.
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int sample_rate_khz_; // Sample rate in kHz.
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// Number of samples per packet. We update this every time we receive a
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// packet, not only for consecutive packets.
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int samples_per_packet_;
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// A list of missing packets to be retransmitted. Components of the list
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// contain the sequence number of missing packets and the estimated time that
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// each pack is going to be played out.
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NackList nack_list_;
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// NACK list will not keep track of missing packets prior to
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// |sequence_num_last_received_rtp_| - |max_nack_list_size_|.
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size_t max_nack_list_size_;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_NETEQ_NACK_TRACKER_H_
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