Correcting the amount of padding when send side bwe includes RTP overhead.

webrtc/modules/rtp_rtcp/source/rtp_sender.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_sender.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index 6fdac6f63fb08507ec68781c4bff1e59b6ca693e..60f8cd540dbd81ff4cc48a34b452679b26b9102e 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -442,33 +442,46 @@ size_t RTPSender::TrySendRedundantPayloads(size_t bytes_to_send,
   int bytes_left = static_cast<int>(bytes_to_send);
   while (bytes_left > 0) {
     std::unique_ptr<RtpPacketToSend> packet =
-        packet_history_.GetBestFittingPacket(bytes_left);
+        packet_history_.GetBestFittingPacket(bytes_left,
+                                             send_side_bwe_with_overhead_);
     if (!packet)
       break;
-    size_t payload_size = packet->payload_size();
+    // When WebRTC-SendSideBwe-WithOverhead is enabled, the padding budget
+    // includes overhead.
+    const size_t used_bytes =
+        send_side_bwe_with_overhead_
+            ? packet->payload_size() + packet->headers_size()
+            : packet->payload_size();
     if (!PrepareAndSendPacket(std::move(packet), true, false, pacing_info))
       break;
-    bytes_left -= payload_size;
+    bytes_left -= used_bytes;
   }
   return bytes_to_send - bytes_left;
 }
 
 size_t RTPSender::SendPadData(size_t bytes,
                               const PacedPacketInfo& pacing_info) {
-  size_t padding_bytes_in_packet;
+  // Always send full padding packets. This is accounted for by the
+  // RtpPacketSender, which will make sure we don't send too much padding even
+  // if a single packet is larger than requested.
+  // We do this to avoid frequently sending small packets on higher bitrates.
+  size_t padding_bytes_in_packet =
+      std::min(MaxPayloadSize(), kMaxPaddingLength);
+
   if (audio_configured_) {
     // Allow smaller padding packets for audio.
+
+    // When WebRTC-SendSideBwe-WithOverhead is enabled, the padding budget
+    // includes overhead.
+    const size_t padding_bytes_if_one_packet =
+        send_side_bwe_with_overhead_
+            ? std::max<int>(0, bytes - RtpHeaderLength())

[minyue-webrtc, 2017/04/11 09:35:14]

This was wrong, since bytes - RtpHeaderLength() can be negative

+            : bytes;
     padding_bytes_in_packet =
-        std::min(std::max(bytes, kMinAudioPaddingLength), MaxPayloadSize());
-    if (padding_bytes_in_packet > kMaxPaddingLength)
-      padding_bytes_in_packet = kMaxPaddingLength;
-  } else {
-    // Always send full padding packets. This is accounted for by the
-    // RtpPacketSender, which will make sure we don't send too much padding even
-    // if a single packet is larger than requested.
-    // We do this to avoid frequently sending small packets on higher bitrates.
-    padding_bytes_in_packet = std::min(MaxPayloadSize(), kMaxPaddingLength);
+        std::min(padding_bytes_in_packet,
+                 std::max(padding_bytes_if_one_packet, kMinAudioPaddingLength));
   }
+
   size_t bytes_sent = 0;
   while (bytes_sent < bytes) {
     int64_t now_ms = clock_->TimeInMilliseconds();
@@ -553,6 +566,7 @@ size_t RTPSender::SendPadData(size_t bytes,
     PacketOptions options;
     bool has_transport_seq_num =
         UpdateTransportSequenceNumber(&padding_packet, &options.packet_id);
+
     padding_packet.SetPadding(padding_bytes_in_packet, &random_);
 
     if (has_transport_seq_num) {
@@ -563,7 +577,12 @@ size_t RTPSender::SendPadData(size_t bytes,
     if (!SendPacketToNetwork(padding_packet, options, pacing_info))
       break;
 
-    bytes_sent += padding_bytes_in_packet;
+    // When WebRTC-SendSideBwe-WithOverhead is enabled, the padding budget
+    // includes overhead.
+    bytes_sent +=
+        send_side_bwe_with_overhead_
+            ? padding_packet.padding_size() + padding_packet.headers_size()
+            : padding_packet.padding_size();
     UpdateRtpStats(padding_packet, over_rtx, false);
   }
 
@@ -597,10 +616,13 @@ int32_t RTPSender::ReSendPacket(uint16_t packet_id, int64_t min_resend_time) {
     // TickTime.
     int64_t corrected_capture_tims_ms =
         packet->capture_time_ms() + clock_delta_ms_;
+    const size_t packet_size =
+        send_side_bwe_with_overhead_
+            ? packet->payload_size() + packet->headers_size()
+            : packet->payload_size();
     paced_sender_->InsertPacket(RtpPacketSender::kNormalPriority,
                                 packet->Ssrc(), packet->SequenceNumber(),
-                                corrected_capture_tims_ms,
-                                packet->payload_size(), true);
+                                corrected_capture_tims_ms, packet_size, true);
 
     return packet->size();
   }
@@ -842,7 +864,10 @@ bool RTPSender::SendToNetwork(std::unique_ptr<RtpPacketToSend> packet,
     // Correct offset between implementations of millisecond time stamps in
     // TickTime and Clock.
     int64_t corrected_time_ms = packet->capture_time_ms() + clock_delta_ms_;
-    size_t payload_length = packet->payload_size();
+    const size_t packet_size =
+        send_side_bwe_with_overhead_
+            ? packet->payload_size() + packet->headers_size()
+            : packet->payload_size();
     if (ssrc == flexfec_ssrc) {
       // Store FlexFEC packets in the history here, so they can be found
       // when the pacer calls TimeToSendPacket.
@@ -852,7 +877,8 @@ bool RTPSender::SendToNetwork(std::unique_ptr<RtpPacketToSend> packet,
     }
 
     paced_sender_->InsertPacket(priority, ssrc, seq_no, corrected_time_ms,
-                                payload_length, false);
+                                packet_size, false);
+
     if (last_capture_time_ms_sent_ == 0 ||
         corrected_time_ms > last_capture_time_ms_sent_) {
       last_capture_time_ms_sent_ = corrected_time_ms;
@@ -1237,11 +1263,10 @@ void RTPSender::AddPacketToTransportFeedback(
     uint16_t packet_id,
     const RtpPacketToSend& packet,
     const PacedPacketInfo& pacing_info) {
-  size_t packet_size = packet.payload_size() + packet.padding_size();
-  if (send_side_bwe_with_overhead_) {
-    packet_size = packet.size();
-  }
-
+  const size_t packet_size =

[minyue-webrtc, 2017/04/11 09:35:14]

this is a simple refactoring

+      send_side_bwe_with_overhead_
+          ? packet.size()
+          : packet.payload_size() + packet.padding_size();
   if (transport_feedback_observer_) {
     transport_feedback_observer_->AddPacket(SSRC(), packet_id, packet_size,
                                             pacing_info);