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