Reland "Batch assign RTP seq# for all packets of a frame."

This is a reland of 5cc99570620890edc3989b2cae1d1ee0669a021c

Original change's description:
> Batch assign RTP seq# for all packets of a frame.
>
> This avoids a potential race where other call sites could assign
> sequence numbers while the video frame is mid packetization - resulting
> in a non-contiguous video sequence.
>
> Avoiding the tight lock-unlock within the loop also couldn't hurt from
> a performance standpoint.
>
> Bug: webrtc:12448
> Change-Id: I6cc31c7743d2ca75caeaeffb98651a480dbe08e2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207867
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33291}

Bug: webrtc:12448
Change-Id: I7c5a5e00a5e08330ff24b58af9f090c327eeeaa2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208221
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33296}

modules/rtp_rtcp/source/rtp_sender.cc

@@ -47,6 +47,9 @@ constexpr uint32_t kTimestampTicksPerMs = 90;
 // Min size needed to get payload padding from packet history.
 constexpr int kMinPayloadPaddingBytes = 50;
 
+// RED header if first byte of payload.
+constexpr size_t kRedForFecHeaderLength = 1;
+
 template <typename Extension>
 constexpr RtpExtensionSize CreateExtensionSize() {
   return {Extension::kId, Extension::kValueSizeBytes};
@@ -611,16 +614,40 @@ bool RTPSender::AssignSequenceNumber(RtpPacketToSend* packet) {
   RTC_DCHECK(packet->Ssrc() == ssrc_);
   packet->SetSequenceNumber(sequence_number_++);
 
+  UpdateLastPacketState(*packet);
+  return true;
+}
+
+bool RTPSender::AssignSequenceNumbersAndStoreLastPacketState(
+    rtc::ArrayView<std::unique_ptr<RtpPacketToSend>> packets) {
+  RTC_DCHECK(!packets.empty());
+  MutexLock lock(&send_mutex_);
+  if (!sending_media_)
+    return false;
+  for (auto& packet : packets) {
+    RTC_DCHECK_EQ(packet->Ssrc(), ssrc_);
+    packet->SetSequenceNumber(sequence_number_++);
+  }
+  UpdateLastPacketState(**packets.rbegin());
+  return true;
+}
+
+void RTPSender::UpdateLastPacketState(const RtpPacketToSend& packet) {
   // Remember marker bit to determine if padding can be inserted with
   // sequence number following |packet|.
-  last_packet_marker_bit_ = packet->Marker();
-  // Remember payload type to use in the padding packet if rtx is disabled.
-  last_payload_type_ = packet->PayloadType();
+  last_packet_marker_bit_ = packet.Marker();
+  // Remember media payload type to use in the padding packet if rtx is
+  // disabled.
+  if (packet.is_red()) {
+    RTC_DCHECK_GE(packet.payload_size(), kRedForFecHeaderLength);
+    last_payload_type_ = packet.PayloadBuffer()[0];
+  } else {
+    last_payload_type_ = packet.PayloadType();
+  }
   // Save timestamps to generate timestamp field and extensions for the padding.
-  last_rtp_timestamp_ = packet->Timestamp();
+  last_rtp_timestamp_ = packet.Timestamp();
   last_timestamp_time_ms_ = clock_->TimeInMilliseconds();
-  capture_time_ms_ = packet->capture_time_ms();
-  return true;
+  capture_time_ms_ = packet.capture_time_ms();
 }
 
 void RTPSender::SetSendingMediaStatus(bool enabled) {

modules/rtp_rtcp/source/rtp_sender.h

@@ -137,6 +137,11 @@ class RTPSender {
   // Return false if sending was turned off.
   bool AssignSequenceNumber(RtpPacketToSend* packet)
       RTC_LOCKS_EXCLUDED(send_mutex_);
+  // Same as AssignSequenceNumber(), but applies sequence numbers atomically to
+  // a batch of packets.
+  bool AssignSequenceNumbersAndStoreLastPacketState(
+      rtc::ArrayView<std::unique_ptr<RtpPacketToSend>> packets)
+      RTC_LOCKS_EXCLUDED(send_mutex_);
   // Maximum header overhead per fec/padding packet.
   size_t FecOrPaddingPacketMaxRtpHeaderLength() const
       RTC_LOCKS_EXCLUDED(send_mutex_);
@@ -179,6 +184,9 @@ class RTPSender {
 
   void UpdateHeaderSizes() RTC_EXCLUSIVE_LOCKS_REQUIRED(send_mutex_);
 
+  void UpdateLastPacketState(const RtpPacketToSend& packet)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(send_mutex_);
+
   Clock* const clock_;
   Random random_ RTC_GUARDED_BY(send_mutex_);
 

modules/rtp_rtcp/source/rtp_sender_video.cc

@@ -648,8 +648,6 @@ bool RTPSenderVideo::SendVideo(
     if (!packetizer->NextPacket(packet.get()))
       return false;
     RTC_DCHECK_LE(packet->payload_size(), expected_payload_capacity);
-    if (!rtp_sender_->AssignSequenceNumber(packet.get()))
-      return false;
 
     packet->set_allow_retransmission(allow_retransmission);
     packet->set_is_key_frame(video_header.frame_type ==
@@ -670,7 +668,7 @@ bool RTPSenderVideo::SendVideo(
       red_packet->SetPayloadType(*red_payload_type_);
       red_packet->set_is_red(true);
 
-      // Send |red_packet| instead of |packet| for allocated sequence number.
+      // Append |red_packet| instead of |packet| to output.
       red_packet->set_packet_type(RtpPacketMediaType::kVideo);
       red_packet->set_allow_retransmission(packet->allow_retransmission());
       rtp_packets.emplace_back(std::move(red_packet));
@@ -691,6 +689,11 @@ bool RTPSenderVideo::SendVideo(
     }
   }
 
+  if (!rtp_sender_->AssignSequenceNumbersAndStoreLastPacketState(rtp_packets)) {
+    // Media not being sent.
+    return false;
+  }
+
   LogAndSendToNetwork(std::move(rtp_packets), payload.size());
 
   // Update details about the last sent frame.