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Provide a default RTT for audio NACK.

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This is used until the first RTT measurement becomes available.
100ms is a reasonable default and used in other places.

Bug: webrtc:10178
Change-Id: I14f530504a4866fbe75f025dfe184fd6e296b75e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256861
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36384}
This commit is contained in:
Jakob Ivarsson
2022-03-30 16:26:19 +02:00
committed by WebRTC LUCI CQ
parent b2be4eff92
commit 4d5eb6507a
2 changed files with 8 additions and 2 deletions
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6
modules/audio_coding/neteq/nack_tracker.cc
View File

@ -225,8 +225,12 @@ int64_t NackTracker::TimeToPlay(uint32_t timestamp) const {
std::vector<uint16_t> NackTracker::GetNackList(int64_t round_trip_time_ms) { std::vector<uint16_t> NackTracker::GetNackList(int64_t round_trip_time_ms) {
RTC_DCHECK_GE(round_trip_time_ms, 0); RTC_DCHECK_GE(round_trip_time_ms, 0);
std::vector<uint16_t> sequence_numbers; std::vector<uint16_t> sequence_numbers;
if (config_.require_valid_rtt && round_trip_time_ms == 0) { if (round_trip_time_ms == 0) {
if (config_.require_valid_rtt) {
return sequence_numbers; return sequence_numbers;
} else {
round_trip_time_ms = config_.default_rtt_ms;
}
} }
if (packet_loss_rate_ > if (packet_loss_rate_ >
static_cast<uint32_t>(config_.max_loss_rate * (1 << 30))) { static_cast<uint32_t>(config_.max_loss_rate * (1 << 30))) {

2
modules/audio_coding/neteq/nack_tracker.h
View File

@ -111,6 +111,8 @@ class NackTracker {
bool never_nack_multiple_times = false; bool never_nack_multiple_times = false;
// Only nack if the RTT is valid. // Only nack if the RTT is valid.
bool require_valid_rtt = false; bool require_valid_rtt = false;
// Default RTT to use unless `require_valid_rtt` is set.
int default_rtt_ms = 100;
// Do not nack if the loss rate is above this value. // Do not nack if the loss rate is above this value.
double max_loss_rate = 1.0; double max_loss_rate = 1.0;
}; };