commit e1301a8b3a536935c90d1bad7cd5bd60a70babec
author Henrik Grunell <henrikg@webrtc.org> Thu Dec 13 12:13:22 2018 +0000
committer Commit Bot <commit-bot@chromium.org> Thu Dec 13 12:13:30 2018 +0000
tree 957b5d89d06f71605c6543d043b19fc672a85b7e
parent 94c0f2645e2ae6221d8cefe20a3d048387e27000

Revert "Implement read-only codecPayloadType in RtpParameters"

This reverts commit 806e06d1366b58878ced05cdd8d1d56394982fe6.

Reason for revert: Breaks WebRTC roll to Chromium. https://chromium-review.googlesource.com/c/chromium/src/+/1375538

02:52:35.346 7748   [6936:11248:1213/025234.206:ERROR:mediaengine.cc(80)] Attempted to set RtpParameters with modified codecPayloadType (INVALID_MODIFICATION)

Original change's description:
> Implement read-only codecPayloadType in RtpParameters
> 
> Bug: webrtc:7580
> Change-Id: I6d901afa97262b6c6d9fe6c7366df465ec77bfb3
> Reviewed-on: https://webrtc-review.googlesource.com/c/113944
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Seth Hampson <shampson@webrtc.org>
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25993}

TBR=steveanton@webrtc.org,sakal@webrtc.org,andersc@webrtc.org,shampson@webrtc.org,orphis@webrtc.org

Change-Id: I157f9a79ae7133395431891e15e2c053559d359b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7580
Reviewed-on: https://webrtc-review.googlesource.com/c/114300
Reviewed-by: Henrik Grunell <henrikg@webrtc.org>
Commit-Queue: Henrik Grunell <henrikg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26000}

pc/peerconnection_rtp_unittest.cc

diff --git a/pc/peerconnection_rtp_unittest.cc b/pc/peerconnection_rtp_unittest.cc
index b39b0c2..c1f7656 100644
--- a/pc/peerconnection_rtp_unittest.cc
+++ b/pc/peerconnection_rtp_unittest.cc
@@ -1424,7 +1424,7 @@
 
   auto default_send_encodings = init.send_encodings;
 
-  // Unimplemented RtpParameters: ssrc, fec, rtx, dtx,
+  // Unimplemented RtpParameters: ssrc, codec_payload_type, fec, rtx, dtx,
   // ptime, scale_resolution_down_by, scale_framerate_down_by, rid,
   // dependency_rids.
   init.send_encodings[0].ssrc = 1;
@@ -1435,6 +1435,14 @@
                 .type());
   init.send_encodings = default_send_encodings;
 
+  init.send_encodings[0].codec_payload_type = 1;
+  EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
+            caller->pc()
+                ->AddTransceiver(cricket::MEDIA_TYPE_AUDIO, init)
+                .error()
+                .type());
+  init.send_encodings = default_send_encodings;
+
   init.send_encodings[0].fec = RtpFecParameters();
   EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
             caller->pc()