Using Ntp times for the absolute send time.

Bug: webrtc:15930
Change-Id: Ie460ac6e3561efafeb11bf36735cb6f33bdfd8a3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349162
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Lionel Koenig Gélas <lionelk@google.com>
Cr-Commit-Position: refs/heads/main@{#42183}

audio/channel_send.cc

@@ -430,6 +430,7 @@ ChannelSend::ChannelSend(
   configuration.rtcp_packet_type_counter_observer = this;
 
   configuration.local_media_ssrc = ssrc;
+  configuration.field_trials = &field_trials;
 
   rtp_rtcp_ = ModuleRtpRtcpImpl2::Create(configuration);
   rtp_rtcp_->SetSendingMediaStatus(false);

experiments/field_trials.py

@@ -123,6 +123,9 @@ ACTIVE_FIELD_TRIALS: FrozenSet[FieldTrial] = frozenset([
     FieldTrial('WebRTC-TaskQueue-ReplaceLibeventWithStdlib',
                'webrtc:14389',
                date(2024, 4, 1)),
+    FieldTrial('WebRTC-UseNtpTimeAbsoluteSendTime',
+               'webrtc:15930',
+               date(2024, 9, 1)),
     FieldTrial('WebRTC-VP8-MaxFrameInterval',
                'webrtc:15530',
                date(2024, 4, 1)),

modules/rtp_rtcp/source/rtp_header_extensions.h

@@ -27,6 +27,7 @@
 #include "api/video/video_rotation.h"
 #include "api/video/video_timing.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "system_wrappers/include/ntp_time.h"
 
 // This file contains class definitions for reading/writing each RTP header
 // extension. Each class must be defined such that it is compatible with being
@@ -58,6 +59,11 @@ class AbsoluteSendTime {
     return static_cast<uint32_t>(time6x18);
   }
 
+  static uint32_t To24Bits(NtpTime ntp_time) {
+    uint64_t ntp_time32x32 = static_cast<uint64_t>(ntp_time);
+    return (ntp_time32x32 >> 14) & 0x00FF'FFFF;
+  }
+
   static constexpr Timestamp ToTimestamp(uint32_t time_24bits) {
     RTC_DCHECK_LT(time_24bits, (1 << 24));
     return Timestamp::Micros((time_24bits* int64_t{1'000'000}) >> 18);

modules/rtp_rtcp/source/rtp_sender_egress.cc

@@ -26,6 +26,15 @@ constexpr uint32_t kTimestampTicksPerMs = 90;
 constexpr TimeDelta kBitrateStatisticsWindow = TimeDelta::Seconds(1);
 constexpr size_t kRtpSequenceNumberMapMaxEntries = 1 << 13;
 constexpr TimeDelta kUpdateInterval = kBitrateStatisticsWindow;
+
+bool GetUseNtpTimeForAbsoluteSendTime(const FieldTrialsView* field_trials) {
+  if (field_trials != nullptr &&
+      field_trials->IsDisabled("WebRTC-UseNtpTimeAbsoluteSendTime")) {
+    return false;
+  }
+  return true;
+}
+
 }  // namespace
 
 RtpSenderEgress::NonPacedPacketSender::NonPacedPacketSender(
@@ -104,7 +113,9 @@ RtpSenderEgress::RtpSenderEgress(const RtpRtcpInterface::Configuration& config,
       rtp_sequence_number_map_(need_rtp_packet_infos_
                                    ? std::make_unique<RtpSequenceNumberMap>(
                                          kRtpSequenceNumberMapMaxEntries)
-                                   : nullptr) {
+                                   : nullptr),
+      use_ntp_time_for_absolute_send_time_(
+          GetUseNtpTimeForAbsoluteSendTime(config.field_trials)) {
   RTC_DCHECK(worker_queue_);
   RTC_DCHECK(config.transport_feedback_callback == nullptr)
       << "transport_feedback_callback is no longer used and will soon be "
@@ -209,7 +220,12 @@ void RtpSenderEgress::SendPacket(std::unique_ptr<RtpPacketToSend> packet,
     packet->SetExtension<TransmissionOffset>(kTimestampTicksPerMs * diff.ms());
   }
   if (packet->HasExtension<AbsoluteSendTime>()) {
-    packet->SetExtension<AbsoluteSendTime>(AbsoluteSendTime::To24Bits(now));
+    if (use_ntp_time_for_absolute_send_time_) {
+      packet->SetExtension<AbsoluteSendTime>(
+          AbsoluteSendTime::To24Bits(clock_->ConvertTimestampToNtpTime(now)));
+    } else {
+      packet->SetExtension<AbsoluteSendTime>(AbsoluteSendTime::To24Bits(now));
+    }
   }
   if (packet->HasExtension<TransportSequenceNumber>() &&
       packet->transport_sequence_number()) {
@@ -457,16 +473,16 @@ void RtpSenderEgress::UpdateRtpStats(Timestamp now,
   } else if (packet_type == RtpPacketMediaType::kRetransmission) {
     counters->retransmitted.Add(counter);
   }
-    counters->transmitted.Add(counter);
+  counters->transmitted.Add(counter);
 
-    send_rates_[static_cast<size_t>(packet_type)].Update(packet_size, now);
-    if (bitrate_callback_) {
+  send_rates_[static_cast<size_t>(packet_type)].Update(packet_size, now);
+  if (bitrate_callback_) {
     send_rates = GetSendRates(now);
-    }
+  }
 
-    if (rtp_stats_callback_) {
-      rtp_stats_callback_->DataCountersUpdated(*counters, packet_ssrc);
-    }
+  if (rtp_stats_callback_) {
+    rtp_stats_callback_->DataCountersUpdated(*counters, packet_ssrc);
+  }
 
   // The bitrate_callback_ and rtp_stats_callback_ pointers in practice point
   // to the same object, so these callbacks could be consolidated into one.

modules/rtp_rtcp/source/rtp_sender_egress.h

@@ -167,6 +167,7 @@ class RtpSenderEgress {
   RepeatingTaskHandle update_task_ RTC_GUARDED_BY(worker_queue_);
   std::vector<Packet> packets_to_send_ RTC_GUARDED_BY(worker_queue_);
   ScopedTaskSafety task_safety_;
+  const bool use_ntp_time_for_absolute_send_time_;
 };
 
 }  // namespace webrtc