From 56b208761dbd1a33fefc2cbcf60de279126991ab Mon Sep 17 00:00:00 2001
From: Harald Alvestrand <hta@webrtc.org>
Date: Fri, 1 Jul 2022 07:36:59 +0000
Subject: [PATCH] [M104] Do not allow simulcast to be turned off using SDP
 munging
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This is an error that puts the PC into an inconsistent state, so
causing a crash is the right thing to do.

(cherry picked from commit 3fe8b0d9a980642ee5ebb1f9e429378b063c1f07)

Bug: chromium:1341043
Change-Id: Ie1eb89400ad87f0c83634b7073236b07e92ec7ab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267281
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Original-Commit-Position: refs/heads/main@{#37391}
No-Presubmit: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267940
Cr-Commit-Position: refs/branch-heads/5112@{#4}
Cr-Branched-From: a976a871159f85f975fbd6f170d0a8f00a4aee49-refs/heads/main@{#37168}
---
 pc/rtp_sender.cc                   |  4 ++--
 pc/rtp_sender_receiver_unittest.cc | 38 ++++++++++++++++++++++++++++++
 2 files changed, 40 insertions(+), 2 deletions(-)

diff --git a/pc/rtp_sender.cc b/pc/rtp_sender.cc
index d8d4b520eb..3ee9145a20 100644
--- a/pc/rtp_sender.cc
+++ b/pc/rtp_sender.cc
@@ -317,8 +317,8 @@ void RtpSenderBase::SetSsrc(uint32_t ssrc) {
       // we need to copy.
       RtpParameters current_parameters =
           media_channel_->GetRtpSendParameters(ssrc_);
-      RTC_DCHECK_GE(current_parameters.encodings.size(),
-                    init_parameters_.encodings.size());
+      RTC_CHECK_GE(current_parameters.encodings.size(),
+                   init_parameters_.encodings.size());
       for (size_t i = 0; i < init_parameters_.encodings.size(); ++i) {
         init_parameters_.encodings[i].ssrc =
             current_parameters.encodings[i].ssrc;
diff --git a/pc/rtp_sender_receiver_unittest.cc b/pc/rtp_sender_receiver_unittest.cc
index 654fffc60d..59d4c606b1 100644
--- a/pc/rtp_sender_receiver_unittest.cc
+++ b/pc/rtp_sender_receiver_unittest.cc
@@ -1151,6 +1151,44 @@ TEST_F(RtpSenderReceiverTest,
   DestroyVideoRtpSender();
 }
 
+#if GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+using RtpSenderReceiverDeathTest = RtpSenderReceiverTest;
+
+TEST_F(RtpSenderReceiverDeathTest,
+       VideoSenderManualRemoveSimulcastFailsDeathTest) {
+  AddVideoTrack(false);
+
+  std::unique_ptr<MockSetStreamsObserver> set_streams_observer =
+      std::make_unique<MockSetStreamsObserver>();
+  video_rtp_sender_ = VideoRtpSender::Create(worker_thread_, video_track_->id(),
+                                             set_streams_observer.get());
+  ASSERT_TRUE(video_rtp_sender_->SetTrack(video_track_.get()));
+  EXPECT_CALL(*set_streams_observer, OnSetStreams());
+  video_rtp_sender_->SetStreams({local_stream_->id()});
+
+  std::vector<RtpEncodingParameters> init_encodings(2);
+  init_encodings[0].max_bitrate_bps = 60000;
+  init_encodings[1].max_bitrate_bps = 120000;
+  video_rtp_sender_->set_init_send_encodings(init_encodings);
+
+  RtpParameters params = video_rtp_sender_->GetParameters();
+  ASSERT_EQ(2u, params.encodings.size());
+  EXPECT_EQ(params.encodings[0].max_bitrate_bps, 60000);
+
+  // Simulate the setLocalDescription call as if the user used SDP munging
+  // to disable simulcast.
+  std::vector<uint32_t> ssrcs;
+  ssrcs.reserve(2);
+  for (int i = 0; i < 2; ++i)
+    ssrcs.push_back(kVideoSsrcSimulcast + i);
+  cricket::StreamParams stream_params =
+      cricket::StreamParams::CreateLegacy(kVideoSsrc);
+  video_media_channel()->AddSendStream(stream_params);
+  video_rtp_sender_->SetMediaChannel(video_media_channel());
+  EXPECT_DEATH(video_rtp_sender_->SetSsrc(kVideoSsrcSimulcast), "");
+}
+#endif
+
 TEST_F(RtpSenderReceiverTest,
        VideoSenderMustCallGetParametersBeforeSetParametersBeforeNegotiation) {
   video_rtp_sender_ =