Fix packetization logic to leave space for extensions in the last packet

webrtc/modules/rtp_rtcp/source/rtp_format.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_format.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format.cc b/webrtc/modules/rtp_rtcp/source/rtp_format.cc
index 753fc2ec41795684f1b7d709416ed0ad1e94a931..47ecfb6177bb0f880ff12e384be451d029deb7b4 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format.cc
@@ -20,21 +20,25 @@
 namespace webrtc {
 RtpPacketizer* RtpPacketizer::Create(RtpVideoCodecTypes type,
                                      size_t max_payload_len,
+                                     size_t last_packet_reduction_len,
                                      const RTPVideoTypeHeader* rtp_type_header,
                                      FrameType frame_type) {
   switch (type) {
     case kRtpVideoH264:
       RTC_CHECK(rtp_type_header);
-      return new RtpPacketizerH264(max_payload_len,
+      return new RtpPacketizerH264(max_payload_len, last_packet_reduction_len,
                                    rtp_type_header->H264.packetization_mode);
     case kRtpVideoVp8:
       RTC_CHECK(rtp_type_header);
-      return new RtpPacketizerVp8(rtp_type_header->VP8, max_payload_len);
+      return new RtpPacketizerVp8(rtp_type_header->VP8, max_payload_len,
+                                  last_packet_reduction_len);
     case kRtpVideoVp9:
       RTC_CHECK(rtp_type_header);
-      return new RtpPacketizerVp9(rtp_type_header->VP9, max_payload_len);
+      return new RtpPacketizerVp9(rtp_type_header->VP9, max_payload_len,
+                                  last_packet_reduction_len);
     case kRtpVideoGeneric:
-      return new RtpPacketizerGeneric(frame_type, max_payload_len);
+      return new RtpPacketizerGeneric(frame_type, max_payload_len,
+                                      last_packet_reduction_len);
     case kRtpVideoNone:
       RTC_NOTREACHED();
   }