Prevent updating state in the delay manager if the packet was reordered.
Currently, if the last packet was reordered (e.g. due to retransmission) then the next packet's inter-arrival time will be estimated incorrectly due to the jump in sequence numbers. This change prevents that by not resetting the stopwatch on reordered packets.
This will also better estimate inter-arrival times when we have multiple reordered packets in a burst. Currently we would only measure the iat of the first reordered packet correctly and not the ones coming after it.
There is a slight risk introducing this: If we would receive an out of order packet far into the future (in sequence numbers) and then continue getting packets in the normal order, then we would not update the current sequence number for these and incorrectly estimate their inter-arrival times since they would all be considered reordered.
Change-Id: Ic938a37cbddf1cb9c30b610218f56794568d3d01
Bug: webrtc:10178
Reviewed-on: https://webrtc-review.googlesource.com/c/119949
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26572}diff --git a/modules/audio_coding/neteq/delay_manager.cc b/modules/audio_coding/neteq/delay_manager.cc
index 4ca545d1..c97d535 100644
--- a/modules/audio_coding/neteq/delay_manager.cc
+++ b/modules/audio_coding/neteq/delay_manager.cc
@@ -34,6 +34,8 @@
// Steady-state forgetting factor for |iat_vector_|, 0.9993 in Q15.
constexpr int kIatFactor_ = 32745;
constexpr int kMaxIat = 64; // Max inter-arrival time to register.
+constexpr int kMaxReorderedPackets =
+ 10; // Max number of consecutive reordered packets.
absl::optional<int> GetForcedLimitProbability() {
constexpr char kForceTargetDelayPercentileFieldTrial[] =
@@ -63,6 +65,7 @@
DelayManager::DelayManager(size_t max_packets_in_buffer,
int base_min_target_delay_ms,
+ bool enable_rtx_handling,
DelayPeakDetector* peak_detector,
const TickTimer* tick_timer)
: first_packet_received_(false),
@@ -85,7 +88,8 @@
last_pack_cng_or_dtmf_(1),
frame_length_change_experiment_(
field_trial::IsEnabled("WebRTC-Audio-NetEqFramelengthExperiment")),
- forced_limit_probability_(GetForcedLimitProbability()) {
+ forced_limit_probability_(GetForcedLimitProbability()),
+ enable_rtx_handling_(enable_rtx_handling) {
assert(peak_detector); // Should never be NULL.
RTC_DCHECK_GE(base_min_target_delay_ms_, 0);
RTC_DCHECK_LE(minimum_delay_ms_, maximum_delay_ms_);
@@ -146,6 +150,7 @@
rtc::saturated_cast<int>(1000 * packet_len_samp / sample_rate_hz);
}
+ bool reordered = false;
if (packet_len_ms > 0) {
// Cannot update statistics unless |packet_len_ms| is valid.
// Calculate inter-arrival time (IAT) in integer "packet times"
@@ -158,7 +163,6 @@
}
// Check for discontinuous packet sequence and re-ordering.
- bool reordered = false;
if (IsNewerSequenceNumber(sequence_number, last_seq_no_ + 1)) {
// Compensate for gap in the sequence numbers. Reduce IAT with the
// expected extra time due to lost packets, but ensure that the IAT is
@@ -183,6 +187,12 @@
LimitTargetLevel();
} // End if (packet_len_ms > 0).
+ if (enable_rtx_handling_ && reordered &&
+ num_reordered_packets_ < kMaxReorderedPackets) {
+ ++num_reordered_packets_;
+ return 0;
+ }
+ num_reordered_packets_ = 0;
// Prepare for next packet arrival.
packet_iat_stopwatch_ = tick_timer_->GetNewStopwatch();
last_seq_no_ = sequence_number;