commit 70bb326fa423c3daad4916eab81cc6dea9dfb12a
author Niels Möller <nisse@webrtc.org> Wed May 23 14:15:13 2018
committer Commit Bot <commit-bot@chromium.org> Wed May 23 16:05:54 2018
tree 081d14e23ab36a9024bd2b08adad53f13773bd9e
parent 5d67f82360660b4bb5c36f719055242dbaf1945a

Delete unused argument first_payload_byte.

This was left-over after cl
https://webrtc-review.googlesource.com/c/src/+/61500.

Bug: webrtc:8995
Change-Id: Ib5ad853d67d6fc8caf72cc6d76c67b2958e4ff63
Reviewed-on: https://webrtc-review.googlesource.com/78520
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23371}

modules/rtp_rtcp/source/rtp_receiver_impl.cc

diff --git a/modules/rtp_rtcp/source/rtp_receiver_impl.cc b/modules/rtp_rtcp/source/rtp_receiver_impl.cc
index 60056c8..28d12d1 100644
--- a/modules/rtp_rtcp/source/rtp_receiver_impl.cc
+++ b/modules/rtp_rtcp/source/rtp_receiver_impl.cc
@@ -166,9 +166,7 @@
   // Trigger our callbacks.
   CheckSSRCChanged(rtp_header);
 
-  int8_t first_payload_byte = payload_length > 0 ? payload[0] : 0;
-
-  if (CheckPayloadChanged(rtp_header, first_payload_byte,
+  if (CheckPayloadChanged(rtp_header,
                           &payload_specific) == -1) {
     if (payload_length == 0) {
       // OK, keep-alive packet.
@@ -301,7 +299,6 @@
 // media_specific interface (such as CheckPayloadChange, possibly get/set
 // last known payload).
 int32_t RtpReceiverImpl::CheckPayloadChanged(const RTPHeader& rtp_header,
-                                             const int8_t first_payload_byte,
                                              PayloadUnion* specific_payload) {
   int8_t payload_type = rtp_header.payloadType;
 

modules/rtp_rtcp/source/rtp_receiver_impl.h

diff --git a/modules/rtp_rtcp/source/rtp_receiver_impl.h b/modules/rtp_rtcp/source/rtp_receiver_impl.h
index 4032318..c577d0a 100644
--- a/modules/rtp_rtcp/source/rtp_receiver_impl.h
+++ b/modules/rtp_rtcp/source/rtp_receiver_impl.h
@@ -73,7 +73,6 @@
   void CheckSSRCChanged(const RTPHeader& rtp_header);
   void CheckCSRC(const WebRtcRTPHeader& rtp_header);
   int32_t CheckPayloadChanged(const RTPHeader& rtp_header,
-                              const int8_t first_payload_byte,
                               PayloadUnion* payload);
 
   void UpdateSources(const rtc::Optional<uint8_t>& ssrc_audio_level);