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Add setting for jitter buffer max packets

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Co-authored-by: Jim Gustafson <jim@signal.org>
This commit is contained in:
Rashad Sookram 2023-06-08 10:11:53 -04:00 committed by GitHub
parent 79676c3da1
commit db3d421794
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GPG key ID: 4AEE18F83AFDEB23
4 changed files with 12 additions and 9 deletions
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16
modules/audio_coding/neteq/packet_buffer.cc
View file

@ -102,8 +102,8 @@ void PacketBuffer::Flush(StatisticsCalculator* stats) {
// RingRTC change to log more information around audio jitter buffer flushes
auto prev_recv_ts = Timestamp::Micros(0);
auto num_out_of_order = 0;
auto num_gaps_below_15ms = 0;
auto num_gaps_above_40ms = 0;
auto num_gaps_below_40ms = 0;
auto num_gaps_above_90ms = 0;
for (auto& p : buffer_) {
LogPacketDiscarded(p.priority.codec_level, stats);
@ -112,10 +112,10 @@ void PacketBuffer::Flush(StatisticsCalculator* stats) {
if (gap_us < 0) {
num_out_of_order++;
} else if (gap_us < 15000) {
num_gaps_below_15ms++;
} else if (gap_us > 40000) {
num_gaps_above_40ms++;
} else if (gap_us < 40000) {
num_gaps_below_40ms++;
} else if (gap_us > 90000) {
num_gaps_above_90ms++;
}
}
prev_recv_ts = p.packet_info.receive_time();
@ -131,8 +131,8 @@ void PacketBuffer::Flush(StatisticsCalculator* stats) {
<< ", ms_since_first_insert=" << first.waiting_time->ElapsedMs()
<< ", ms_since_last_insert=" << last.waiting_time->ElapsedMs()
<< ", num_out_of_order=" << num_out_of_order
<< ", num_gaps_below_15ms=" << num_gaps_below_15ms
<< ", num_gaps_above_40ms=" << num_gaps_above_40ms;
<< ", num_gaps_below_40ms=" << num_gaps_below_40ms
<< ", num_gaps_above_90ms=" << num_gaps_above_90ms;
}
buffer_.clear();
stats->FlushedPacketBuffer();

2
modules/audio_processing/agc2/input_volume_stats_reporter.cc
View file

@ -95,7 +95,7 @@ InputVolumeStatsReporter::InputVolumeStatsReporter(InputVolumeType type)
.update_average = CreateAverageHistogram(type, "UpdateAverage")}),
cannot_log_stats_(!histograms_.AllPointersSet()) {
if (cannot_log_stats_) {
RTC_LOG(LS_WARNING) << "Will not log any `" << MetricNamePrefix(type)
RTC_LOG(LS_INFO) << "Will not log any `" << MetricNamePrefix(type)
<< "*` histogram stats.";
}
}

1
ringrtc/rffi/api/peer_connection_factory.h
View file

@ -92,6 +92,7 @@ RUSTEXPORT webrtc::PeerConnectionInterface* Rust_createPeerConnection(
webrtc::PeerConnectionFactoryOwner* factory_owner_borrowed_rc,
webrtc::rffi::PeerConnectionObserverRffi* observer_borrowed,
RffiPeerConnectionKind kind,
int audio_jitter_buffer_max_packets,
RffiIceServer ice_server,
webrtc::AudioTrackInterface* outgoing_audio_track_borrowed_rc,
webrtc::VideoTrackInterface* outgoing_video_track_borrowed_rc);

2
ringrtc/rffi/src/peer_connection_factory.cc
View file

@ -297,6 +297,7 @@ RUSTEXPORT PeerConnectionInterface* Rust_createPeerConnection(
PeerConnectionFactoryOwner* factory_owner_borrowed_rc,
PeerConnectionObserverRffi* observer_borrowed,
RffiPeerConnectionKind kind,
int audio_jitter_buffer_max_packets,
RffiIceServer ice_server,
webrtc::AudioTrackInterface* outgoing_audio_track_borrowed_rc,
webrtc::VideoTrackInterface* outgoing_video_track_borrowed_rc) {
@ -311,6 +312,7 @@ RUSTEXPORT PeerConnectionInterface* Rust_createPeerConnection(
} else if (kind == RffiPeerConnectionKind::kGroupCall) {
config.tcp_candidate_policy = PeerConnectionInterface::kTcpCandidatePolicyEnabled;
}
config.audio_jitter_buffer_max_packets = audio_jitter_buffer_max_packets;
config.sdp_semantics = SdpSemantics::kPlanB_DEPRECATED;
if (ice_server.urls_size > 0) {
webrtc::PeerConnectionInterface::IceServer rtc_ice_server;