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chore: cherry-pick 763d847f1e5a from webrtc (#34882)

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* chore: cherry-pick 763d847f1e5a from webrtc

* chore: update patches

Co-authored-by: PatchUp <73610968+patchup[bot]@users.noreply.github.com>
This commit is contained in:
Pedro Pontes
2022-07-12 02:08:19 +02:00
committed by GitHub
parent b21dae6bee
commit a70431f8db
2 changed files with 94 additions and 0 deletions
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1
patches/webrtc/.patches
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@@ -1,2 +1,3 @@
add_thread_local_to_x_error_trap_cc.patch
cherry-pick-a18fddcb53e6.patch
cherry-pick-763d847f1e5a.patch

93
patches/webrtc/cherry-pick-763d847f1e5a.patch Normal file
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@@ -0,0 +1,93 @@
From 0000000000000000000000000000000000000000 Mon Sep 17 00:00:00 2001
From: Taylor Brandstetter <deadbeef@webrtc.org>
Date: Fri, 1 Jul 2022 12:20:05 -0700
Subject: Do not allow simulcast to be turned off using SDP munging
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
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)
TBR=hta@webrtc.org
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
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267427
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/branch-heads/5005@{#9}
Cr-Branched-From: 8f9b44ba38c134e1c69126196e4d9e566ca5d5e3-refs/heads/main@{#36552}
diff --git a/pc/rtp_sender.cc b/pc/rtp_sender.cc
index 110b5aae0a01a5d612eefdc3262b20e4dd5504fe..d1355e31ab6331058cd557b0f164c903c9bc8f46 100644
--- a/pc/rtp_sender.cc
+++ b/pc/rtp_sender.cc
@@ -299,8 +299,8 @@ 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 2c6bc7b71a1e5e30609f382fee0b4cc30f03512e..388edbaf73f0570d895b4855bc2207afb8eff1a5 100644
--- a/pc/rtp_sender_receiver_unittest.cc
+++ b/pc/rtp_sender_receiver_unittest.cc
@@ -1157,6 +1157,44 @@ 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_ =