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Fix codec collision on reoffer after munged codec on offer.

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Bug: chromium:395077842
Change-Id: I7665e593fa0f6883150363cb75103facd62f4fea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/377141
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43889}
This commit is contained in:
Harald Alvestrand 2025-02-13 13:46:31 +00:00 committed by WebRTC LUCI CQ
parent a3b2dd4adb
commit b9ddaa154b
3 changed files with 67 additions and 25 deletions
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14
pc/codec_vendor.cc
View File

@ -454,6 +454,14 @@ webrtc::RTCErrorOr<std::vector<Codec>> CodecVendor::GetNegotiatedCodecsForOffer(
}
}
}
// Note what PTs are already in use.
UsedPayloadTypes
used_pltypes; // Used to avoid pt collisions in filtered_codecs
for (auto& codec : filtered_codecs) {
// Note: This may change PTs. Doing so woud indicate an error, but
// UsedPayloadTypes doesn't offer a means to make the distinction.
used_pltypes.FindAndSetIdUsed(&codec);
}
// Add other supported codecs.
for (const Codec& codec : supported_codecs) {
std::optional<Codec> found_codec =
@ -461,12 +469,12 @@ webrtc::RTCErrorOr<std::vector<Codec>> CodecVendor::GetNegotiatedCodecsForOffer(
if (found_codec &&
!FindMatchingCodec(supported_codecs, filtered_codecs, codec)) {
// Use the `found_codec` from `codecs` because it has the
// correctly mapped payload type.
// This is only done for video since we do not yet have rtx for audio.
// correctly mapped payload type (most of the time).
if (media_description_options.type == MEDIA_TYPE_VIDEO &&
found_codec->GetResiliencyType() == Codec::ResiliencyType::kRtx) {
// For RTX we might need to adjust the apt parameter if we got a
// remote offer without RTX for a codec for which we support RTX.
// This is only done for video since we do not yet have rtx for audio.
auto referenced_codec =
GetAssociatedCodecForRtx(supported_codecs, codec);
RTC_DCHECK(referenced_codec);
@ -479,6 +487,8 @@ webrtc::RTCErrorOr<std::vector<Codec>> CodecVendor::GetNegotiatedCodecsForOffer(
changed_referenced_codec->id);
}
}
// Quick fix for b/395077842: Remap the codec if it collides.
used_pltypes.FindAndSetIdUsed(&(*found_codec));
filtered_codecs.push_back(*found_codec);
}
}

31
pc/peer_connection_integrationtest.cc
View File

@ -4731,6 +4731,37 @@ TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
EXPECT_LT(metrics::MinSample("WebRTC.Call.AbsCapture.Offset"), 2);
}
// Test that when SDP is munged to use a PT for a different codec,
// the old codec is added to a subsequent offer with a different PT
// Regression test for https://issues.chromium.org/395077824
TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
MungeOfferCodecAndReOfferWorks) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddVideoTrack();
auto munger = [](std::unique_ptr<SessionDescriptionInterface>& sdp) {
auto video = GetFirstVideoContentDescription(sdp->description());
auto codecs = video->codecs();
for (auto&& codec : codecs) {
if (codec.name == "VP9") {
RTC_LOG(LS_ERROR) << "Remapping VP9 codec " << codec << " to AV1";
codec.name = "AV1";
}
}
video->set_codecs(codecs);
};
caller()->SetGeneratedSdpMunger(munger);
caller()->CreateAndSetAndSignalOffer();
ASSERT_THAT(
WaitUntil([&] { return SignalingStateStable(); }, ::testing::IsTrue()),
IsRtcOk());
caller()->SetGeneratedSdpMunger(nullptr);
auto offer = caller()->CreateOfferAndWait();
ASSERT_NE(nullptr, offer);
// The offer should be acceptable.
EXPECT_TRUE(caller()->SetLocalDescriptionAndSendSdpMessage(std::move(offer)));
}
} // namespace
} // namespace webrtc

47
pc/test/integration_test_helpers.h
View File

@ -746,6 +746,30 @@ class PeerConnectionIntegrationWrapper : public PeerConnectionObserver,
});
}
// Setting the local description and sending the SDP message over the fake
// signaling channel are combined into the same method because the SDP
// message needs to be sent as soon as SetLocalDescription finishes, without
// waiting for the observer to be called. This ensures that ICE candidates
// don't outrace the description.
bool SetLocalDescriptionAndSendSdpMessage(
std::unique_ptr<SessionDescriptionInterface> desc) {
auto observer = rtc::make_ref_counted<MockSetSessionDescriptionObserver>();
RTC_LOG(LS_INFO) << debug_name_ << ": SetLocalDescriptionAndSendSdpMessage";
SdpType type = desc->GetType();
std::string sdp;
EXPECT_TRUE(desc->ToString(&sdp));
RTC_LOG(LS_INFO) << debug_name_ << ": local SDP contents=\n" << sdp;
pc()->SetLocalDescription(observer.get(), desc.release());
RemoveUnusedVideoRenderers();
// As mentioned above, we need to send the message immediately after
// SetLocalDescription.
SendSdpMessage(type, sdp);
EXPECT_THAT(
WaitUntil([&] { return observer->called(); }, ::testing::IsTrue()),
IsRtcOk());
return true;
}
private:
// Constructor used by friend class PeerConnectionIntegrationBaseTest.
explicit PeerConnectionIntegrationWrapper(const std::string& debug_name)
@ -868,29 +892,6 @@ class PeerConnectionIntegrationWrapper : public PeerConnectionObserver,
return description;
}
// Setting the local description and sending the SDP message over the fake
// signaling channel are combined into the same method because the SDP
// message needs to be sent as soon as SetLocalDescription finishes, without
// waiting for the observer to be called. This ensures that ICE candidates
// don't outrace the description.
bool SetLocalDescriptionAndSendSdpMessage(
std::unique_ptr<SessionDescriptionInterface> desc) {
auto observer = rtc::make_ref_counted<MockSetSessionDescriptionObserver>();
RTC_LOG(LS_INFO) << debug_name_ << ": SetLocalDescriptionAndSendSdpMessage";
SdpType type = desc->GetType();
std::string sdp;
EXPECT_TRUE(desc->ToString(&sdp));
RTC_LOG(LS_INFO) << debug_name_ << ": local SDP contents=\n" << sdp;
pc()->SetLocalDescription(observer.get(), desc.release());
RemoveUnusedVideoRenderers();
// As mentioned above, we need to send the message immediately after
// SetLocalDescription.
SendSdpMessage(type, sdp);
EXPECT_THAT(
WaitUntil([&] { return observer->called(); }, ::testing::IsTrue()),
IsRtcOk());
return true;
}
// This is a work around to remove unused fake_video_renderers from
// transceivers that have either stopped or are no longer receiving.