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Clean up 32 byte mid support from the demuxer

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the SDP parser is restricting this to 16 bytes already

BUG=webrtc:42222663

Change-Id: I26f8f15a3f15cb5a5c8eab64f9dc3e26c0ea1bbb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359780
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42809}
This commit is contained in:
Philipp Hancke 2024-08-16 16:55:12 -07:00 committed by WebRTC LUCI CQ
parent b1ffa9bd4e
commit 37c406a1fd
2 changed files with 7 additions and 30 deletions
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17
call/rtp_demuxer.cc
View File

@ -40,25 +40,14 @@ size_t RemoveFromMapByValue(Map* map, const Value& value) {
return EraseIf(*map, [&](const auto& elem) { return elem.second == value; });
}
// Temp fix: MID in SDP is allowed to be slightly longer than what's allowed
// in the RTP demuxer. Truncate if needed; this won't match, but it only
// makes sense in places that wouldn't use this for matching anyway.
// TODO(bugs.webrtc.org/12517): remove when length 16 is policed by parser.
std::string CheckMidLength(absl::string_view mid) {
std::string new_mid(mid);
if (new_mid.length() > BaseRtpStringExtension::kMaxValueSizeBytes) {
RTC_LOG(LS_WARNING) << "`mid` attribute too long. Truncating.";
new_mid.resize(BaseRtpStringExtension::kMaxValueSizeBytes);
}
return new_mid;
}
} // namespace
RtpDemuxerCriteria::RtpDemuxerCriteria(
absl::string_view mid,
absl::string_view rsid /*= absl::string_view()*/)
: mid_(CheckMidLength(mid)), rsid_(rsid) {}
: mid_(mid), rsid_(rsid) {
RTC_DCHECK(mid.length() <= BaseRtpStringExtension::kMaxValueSizeBytes);
}
RtpDemuxerCriteria::RtpDemuxerCriteria() = default;
RtpDemuxerCriteria::~RtpDemuxerCriteria() = default;

20
call/rtp_demuxer_unittest.cc
View File

@ -1238,26 +1238,14 @@ TEST_F(RtpDemuxerTest, PacketWithMidAndUnknownRsidIsNotRoutedByPayloadType) {
EXPECT_FALSE(demuxer_.OnRtpPacket(*packet));
}
TEST_F(RtpDemuxerTest, MidMustNotExceedMaximumLength) {
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
TEST_F(RtpDemuxerDeathTest, MidMustNotExceedMaximumLength) {
MockRtpPacketSink sink1;
std::string mid1(BaseRtpStringExtension::kMaxValueSizeBytes + 1, 'a');
// Adding the sink should pass even though the supplied mid is too long.
// The mid will be truncated though.
EXPECT_TRUE(AddSinkOnlyMid(mid1, &sink1));
// Adding a second sink with a mid that matches the truncated mid that was
// just added, should fail.
MockRtpPacketSink sink2;
std::string mid2(mid1.substr(0, BaseRtpStringExtension::kMaxValueSizeBytes));
EXPECT_FALSE(AddSinkOnlyMid(mid2, &sink2));
EXPECT_FALSE(RemoveSink(&sink2));
// Remove the original sink.
EXPECT_TRUE(RemoveSink(&sink1));
EXPECT_DEATH(AddSinkOnlyMid(mid1, &sink1), "");
}
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
TEST_F(RtpDemuxerDeathTest, CriteriaMustBeNonEmpty) {
MockRtpPacketSink sink;
RtpDemuxerCriteria criteria;