Bug 1766646 - Vendor libwebrtc from 2497a27b22

Upstream commit: https://webrtc.googlesource.com/src/+/2497a27b22a3f09c701642230f83f67a4c00df13
    Store RtpPacketReceived::arrival_time as Timestamp.

    Previously this value was rounded up to a millisecond value.
    This change is complementary to another change:
    https://webrtc-review.googlesource.com/c/src/+/216398

    Bug: webrtc:12722
    Change-Id: I0fd2baceb4608132615fb6ad241ec863e343edb1
    Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217521
    Commit-Queue: Tommi <tommi@webrtc.org>
    Reviewed-by: Johannes Kron <kron@webrtc.org>
    Cr-Commit-Position: refs/heads/master@{#33928}

third_party/libwebrtc/README.moz-ff-commit

@@ -5322,3 +5322,6 @@ bb7ee95c71
 # MOZ_LIBWEBRTC_SRC=/home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src MOZ_LIBWEBRTC_COMMIT=mjfdev bash dom/media/webrtc/third_party_build/fast-forward-libwebrtc.sh
 # base of lastest vendoring
 3f418ccf99
+# MOZ_LIBWEBRTC_SRC=/home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src MOZ_LIBWEBRTC_COMMIT=mjfdev bash dom/media/webrtc/third_party_build/fast-forward-libwebrtc.sh
+# base of lastest vendoring
+2497a27b22

third_party/libwebrtc/README.mozilla

@@ -3550,3 +3550,5 @@ libwebrtc updated from /home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwe
 libwebrtc updated from /home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src commit mjfdev on 2022-05-19T20:44:40.329132.
 # python3 vendor-libwebrtc.py --from-local /home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src --commit mjfdev libwebrtc
 libwebrtc updated from /home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src commit mjfdev on 2022-05-19T20:45:29.899165.
+# python3 vendor-libwebrtc.py --from-local /home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src --commit mjfdev libwebrtc
+libwebrtc updated from /home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src commit mjfdev on 2022-05-19T20:47:15.140603.

third_party/libwebrtc/call/call.cc

@@ -1426,9 +1426,9 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
       packet_time_us = receive_time_calculator_->ReconcileReceiveTimes(
           packet_time_us, rtc::TimeUTCMicros(), clock_->TimeInMicroseconds());
     }
-    parsed_packet.set_arrival_time_ms((packet_time_us + 500) / 1000);
+    parsed_packet.set_arrival_time(Timestamp::Micros(packet_time_us));
   } else {
-    parsed_packet.set_arrival_time_ms(clock_->TimeInMilliseconds());
+    parsed_packet.set_arrival_time(clock_->CurrentTime());
   }
 
   // We might get RTP keep-alive packets in accordance with RFC6263 section 4.6.
@@ -1464,7 +1464,7 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
       received_audio_bytes_per_second_counter_.Add(length);
       event_log_->Log(
           std::make_unique<RtcEventRtpPacketIncoming>(parsed_packet));
-      const int64_t arrival_time_ms = parsed_packet.arrival_time_ms();
+      const int64_t arrival_time_ms = parsed_packet.arrival_time().ms();
       if (!first_received_rtp_audio_ms_) {
         first_received_rtp_audio_ms_.emplace(arrival_time_ms);
       }
@@ -1478,7 +1478,7 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
       received_video_bytes_per_second_counter_.Add(length);
       event_log_->Log(
           std::make_unique<RtcEventRtpPacketIncoming>(parsed_packet));
-      const int64_t arrival_time_ms = parsed_packet.arrival_time_ms();
+      const int64_t arrival_time_ms = parsed_packet.arrival_time().ms();
       if (!first_received_rtp_video_ms_) {
         first_received_rtp_video_ms_.emplace(arrival_time_ms);
       }
@@ -1566,7 +1566,7 @@ void Call::NotifyBweOfReceivedPacket(const RtpPacketReceived& packet,
 
   ReceivedPacket packet_msg;
   packet_msg.size = DataSize::Bytes(packet.payload_size());
-  packet_msg.receive_time = Timestamp::Millis(packet.arrival_time_ms());
+  packet_msg.receive_time = packet.arrival_time();
   if (header.extension.hasAbsoluteSendTime) {
     packet_msg.send_time = header.extension.GetAbsoluteSendTimestamp();
   }
@@ -1586,8 +1586,8 @@ void Call::NotifyBweOfReceivedPacket(const RtpPacketReceived& packet,
   if (media_type == MediaType::VIDEO ||
       (use_send_side_bwe && header.extension.hasTransportSequenceNumber)) {
     receive_side_cc_.OnReceivedPacket(
-        packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(),
-        header);
+        packet.arrival_time().ms(),
+        packet.payload_size() + packet.padding_size(), header);
   }
 }
 

third_party/libwebrtc/call/rtx_receive_stream.cc

@@ -64,7 +64,7 @@ void RtxReceiveStream::OnRtpPacket(const RtpPacketReceived& rtx_packet) {
   media_packet.SetSequenceNumber((payload[0] << 8) + payload[1]);
   media_packet.SetPayloadType(it->second);
   media_packet.set_recovered(true);
-  media_packet.set_arrival_time_ms(rtx_packet.arrival_time_ms());
+  media_packet.set_arrival_time(rtx_packet.arrival_time());
 
   // Skip the RTX header.
   rtc::ArrayView<const uint8_t> rtx_payload = payload.subview(kRtxHeaderSize);

third_party/libwebrtc/call/rtx_receive_stream_unittest.cc

@@ -194,9 +194,9 @@ TEST(RtxReceiveStreamTest, PropagatesArrivalTime) {
   RtxReceiveStream rtx_sink(&media_sink, PayloadTypeMapping(), kMediaSSRC);
   RtpPacketReceived rtx_packet(nullptr);
   EXPECT_TRUE(rtx_packet.Parse(rtc::ArrayView<const uint8_t>(kRtxPacket)));
-  rtx_packet.set_arrival_time_ms(123);
-  EXPECT_CALL(media_sink,
-              OnRtpPacket(Property(&RtpPacketReceived::arrival_time_ms, 123)));
+  rtx_packet.set_arrival_time(Timestamp::Millis(123));
+  EXPECT_CALL(media_sink, OnRtpPacket(Property(&RtpPacketReceived::arrival_time,
+                                               Timestamp::Millis(123))));
   rtx_sink.OnRtpPacket(rtx_packet);
 }
 

third_party/libwebrtc/modules/rtp_rtcp/BUILD.gn

@@ -111,6 +111,7 @@ rtc_library("rtp_rtcp_format") {
     "../../api/transport:network_control",
     "../../api/transport/rtp:dependency_descriptor",
     "../../api/units:time_delta",
+    "../../api/units:timestamp",
     "../../api/video:video_frame",
     "../../api/video:video_layers_allocation",
     "../../api/video:video_rtp_headers",

third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_received.cc

@@ -21,8 +21,10 @@
 namespace webrtc {
 
 RtpPacketReceived::RtpPacketReceived() = default;
-RtpPacketReceived::RtpPacketReceived(const ExtensionManager* extensions)
-    : RtpPacket(extensions) {}
+RtpPacketReceived::RtpPacketReceived(
+    const ExtensionManager* extensions,
+    webrtc::Timestamp arrival_time /*= webrtc::Timestamp::MinusInfinity()*/)
+    : RtpPacket(extensions), arrival_time_(arrival_time) {}
 RtpPacketReceived::RtpPacketReceived(const RtpPacketReceived& packet) = default;
 RtpPacketReceived::RtpPacketReceived(RtpPacketReceived&& packet) = default;
 

third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_received.h

@@ -14,10 +14,12 @@
 
 #include <utility>
 
+#include "absl/base/attributes.h"
 #include "api/array_view.h"
 #include "api/ref_counted_base.h"
 #include "api/rtp_headers.h"
 #include "api/scoped_refptr.h"
+#include "api/units/timestamp.h"
 #include "modules/rtp_rtcp/source/rtp_packet.h"
 
 namespace webrtc {
@@ -27,7 +29,9 @@ namespace webrtc {
 class RtpPacketReceived : public RtpPacket {
  public:
   RtpPacketReceived();
-  explicit RtpPacketReceived(const ExtensionManager* extensions);
+  explicit RtpPacketReceived(
+      const ExtensionManager* extensions,
+      webrtc::Timestamp arrival_time = webrtc::Timestamp::MinusInfinity());
   RtpPacketReceived(const RtpPacketReceived& packet);
   RtpPacketReceived(RtpPacketReceived&& packet);
 
@@ -42,8 +46,17 @@ class RtpPacketReceived : public RtpPacket {
 
   // Time in local time base as close as it can to packet arrived on the
   // network.
-  int64_t arrival_time_ms() const { return arrival_time_ms_; }
-  void set_arrival_time_ms(int64_t time) { arrival_time_ms_ = time; }
+  webrtc::Timestamp arrival_time() const { return arrival_time_; }
+  void set_arrival_time(webrtc::Timestamp time) { arrival_time_ = time; }
+
+  ABSL_DEPRECATED("Use arrival_time() instead")
+  int64_t arrival_time_ms() const {
+    return arrival_time_.IsMinusInfinity() ? -1 : arrival_time_.ms();
+  }
+  ABSL_DEPRECATED("Use set_arrival_time() instead")
+  void set_arrival_time_ms(int64_t time) {
+    arrival_time_ = webrtc::Timestamp::Millis(time);
+  }
 
   // Flag if packet was recovered via RTX or FEC.
   bool recovered() const { return recovered_; }
@@ -64,7 +77,7 @@ class RtpPacketReceived : public RtpPacket {
   }
 
  private:
-  int64_t arrival_time_ms_ = 0;
+  webrtc::Timestamp arrival_time_ = Timestamp::MinusInfinity();
   int payload_type_frequency_ = 0;
   bool recovered_ = false;
   rtc::scoped_refptr<rtc::RefCountedBase> additional_data_;

third_party/libwebrtc/pc/channel.cc

@@ -471,13 +471,6 @@ bool BaseChannel::SendPacket(bool rtcp,
 void BaseChannel::OnRtpPacket(const webrtc::RtpPacketReceived& parsed_packet) {
   RTC_DCHECK_RUN_ON(network_thread());
 
-  // Take packet time from the |parsed_packet|.
-  // RtpPacketReceived.arrival_time_ms = (timestamp_us + 500) / 1000;
-  int64_t packet_time_us = -1;
-  if (parsed_packet.arrival_time_ms() > 0) {
-    packet_time_us = parsed_packet.arrival_time_ms() * 1000;
-  }
-
   if (on_first_packet_received_) {
     on_first_packet_received_();
     on_first_packet_received_ = nullptr;
@@ -501,7 +494,10 @@ void BaseChannel::OnRtpPacket(const webrtc::RtpPacketReceived& parsed_packet) {
     return;
   }
 
-  media_channel_->OnPacketReceived(parsed_packet.Buffer(), packet_time_us);
+  webrtc::Timestamp packet_time = parsed_packet.arrival_time();
+  media_channel_->OnPacketReceived(
+      parsed_packet.Buffer(),
+      packet_time.IsMinusInfinity() ? -1 : packet_time.us());
 }
 
 void BaseChannel::UpdateRtpHeaderExtensionMap(

third_party/libwebrtc/pc/rtp_transport.cc

@@ -181,16 +181,16 @@ bool RtpTransport::UnregisterRtpDemuxerSink(RtpPacketSinkInterface* sink) {
 
 void RtpTransport::DemuxPacket(rtc::CopyOnWriteBuffer packet,
                                int64_t packet_time_us) {
-  webrtc::RtpPacketReceived parsed_packet(&header_extension_map_);
+  webrtc::RtpPacketReceived parsed_packet(
+      &header_extension_map_, packet_time_us == -1
+                                  ? Timestamp::MinusInfinity()
+                                  : Timestamp::Micros(packet_time_us));
   if (!parsed_packet.Parse(std::move(packet))) {
     RTC_LOG(LS_ERROR)
         << "Failed to parse the incoming RTP packet before demuxing. Drop it.";
     return;
   }
 
-  if (packet_time_us != -1) {
-    parsed_packet.set_arrival_time_ms((packet_time_us + 500) / 1000);
-  }
   if (!rtp_demuxer_.OnRtpPacket(parsed_packet)) {
     RTC_LOG(LS_WARNING) << "Failed to demux RTP packet: "
                         << RtpDemuxer::DescribePacket(parsed_packet);

third_party/libwebrtc/video/rtp_video_stream_receiver.cc

@@ -1214,7 +1214,7 @@ void RtpVideoStreamReceiver::UpdatePacketReceiveTimestamps(
        << " with payload type: " << static_cast<int>(packet.PayloadType())
        << ", timestamp: " << packet.Timestamp()
        << ", sequence number: " << packet.SequenceNumber()
-       << ", arrival time: " << packet.arrival_time_ms();
+       << ", arrival time: " << ToString(packet.arrival_time());
     int32_t time_offset;
     if (packet.GetExtension<TransmissionOffset>(&time_offset)) {
       ss << ", toffset: " << time_offset;

third_party/libwebrtc/video/rtp_video_stream_receiver2.cc

@@ -1177,7 +1177,7 @@ void RtpVideoStreamReceiver2::UpdatePacketReceiveTimestamps(
        << " with payload type: " << static_cast<int>(packet.PayloadType())
        << ", timestamp: " << packet.Timestamp()
        << ", sequence number: " << packet.SequenceNumber()
-       << ", arrival time: " << packet.arrival_time_ms();
+       << ", arrival time: " << ToString(packet.arrival_time());
     int32_t time_offset;
     if (packet.GetExtension<TransmissionOffset>(&time_offset)) {
       ss << ", toffset: " << time_offset;