commit 7031becbfee6bcdaa5c7a096ebb40eda0709237a
parent 8a71d04e1df9168bb16db6ca4be04394478370d3
Author: Michael Froman <mfroman@mozilla.com>
Date: Wed, 8 Oct 2025 22:00:07 -0500
Bug 1993083 - Vendor libwebrtc from 50aa2d768e
Upstream commit: https://webrtc.googlesource.com/src/+/50aa2d768ecfb5d91e3676a09d9bdb9ef72bf317
Wrap jsep_session_description_unittest in webrtc namespace
and remove a lot of webrtc:: prefixes. Follow-up from
https://webrtc-review.googlesource.com/c/src/+/398220/
Bug: None
Change-Id: I670b74828ae860558007ae3c745703c628562ea4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/398690
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#45091}
Diffstat:
2 files changed, 91 insertions(+), 96 deletions(-)
diff --git a/third_party/libwebrtc/README.mozilla.last-vendor b/third_party/libwebrtc/README.mozilla.last-vendor
@@ -1,4 +1,4 @@
# ./mach python dom/media/webrtc/third_party_build/vendor-libwebrtc.py --from-local /home/mfroman/mozilla/elm/.moz-fast-forward/moz-libwebrtc --commit mozpatches libwebrtc
-libwebrtc updated from /home/mfroman/mozilla/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2025-10-09T02:58:53.442732+00:00.
+libwebrtc updated from /home/mfroman/mozilla/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2025-10-09T02:59:57.874297+00:00.
# base of lastest vendoring
-5063e78710
+50aa2d768e
diff --git a/third_party/libwebrtc/pc/jsep_session_description_unittest.cc b/third_party/libwebrtc/pc/jsep_session_description_unittest.cc
@@ -33,15 +33,10 @@
#include "test/gmock.h"
#include "test/gtest.h"
+namespace webrtc {
+
using ::testing::NotNull;
using ::testing::Values;
-using webrtc::IceCandidate;
-using webrtc::IceCandidateCollection;
-using webrtc::IceCandidateType;
-using webrtc::JsepSessionDescription;
-using ::webrtc::MediaProtocolType;
-using webrtc::SdpType;
-using webrtc::SessionDescriptionInterface;
namespace {
const char kAudioMid[] = "audio";
@@ -58,30 +53,30 @@ const uint32_t kCandidateGeneration = 2;
// This creates a session description with both audio and video media contents.
// In SDP this is described by two m lines, one audio and one video.
-std::unique_ptr<webrtc::SessionDescription> CreateCricketSessionDescription() {
- auto desc = std::make_unique<webrtc::SessionDescription>();
+std::unique_ptr<SessionDescription> CreateCricketSessionDescription() {
+ auto desc = std::make_unique<SessionDescription>();
// AudioContentDescription
- auto audio = std::make_unique<webrtc::AudioContentDescription>();
+ auto audio = std::make_unique<AudioContentDescription>();
// VideoContentDescription
- auto video = std::make_unique<webrtc::VideoContentDescription>();
+ auto video = std::make_unique<VideoContentDescription>();
- audio->AddCodec(webrtc::CreateAudioCodec(103, "ISAC", 16000, 0));
+ audio->AddCodec(CreateAudioCodec(103, "ISAC", 16000, 0));
desc->AddContent(kAudioMid, MediaProtocolType::kRtp, std::move(audio));
- video->AddCodec(webrtc::CreateVideoCodec(120, "VP8"));
+ video->AddCodec(CreateVideoCodec(120, "VP8"));
desc->AddContent(kVideoMid, MediaProtocolType::kRtp, std::move(video));
- desc->AddTransportInfo(webrtc::TransportInfo(
+ desc->AddTransportInfo(TransportInfo(
kAudioMid,
- webrtc::TransportDescription(
- std::vector<std::string>(), kCandidateUfragVoice, kCandidatePwdVoice,
- webrtc::ICEMODE_FULL, webrtc::CONNECTIONROLE_NONE, nullptr)));
- desc->AddTransportInfo(webrtc::TransportInfo(
+ TransportDescription(std::vector<std::string>(), kCandidateUfragVoice,
+ kCandidatePwdVoice, ICEMODE_FULL,
+ CONNECTIONROLE_NONE, nullptr)));
+ desc->AddTransportInfo(TransportInfo(
kVideoMid,
- webrtc::TransportDescription(
- std::vector<std::string>(), kCandidateUfragVideo, kCandidatePwdVideo,
- webrtc::ICEMODE_FULL, webrtc::CONNECTIONROLE_NONE, nullptr)));
+ TransportDescription(std::vector<std::string>(), kCandidateUfragVideo,
+ kCandidatePwdVideo, ICEMODE_FULL,
+ CONNECTIONROLE_NONE, nullptr)));
return desc;
}
@@ -91,13 +86,12 @@ class JsepSessionDescriptionTest : public ::testing::Test {
protected:
void SetUp() override {
int port = 1234;
- webrtc::SocketAddress address("127.0.0.1", port++);
- webrtc::Candidate candidate(webrtc::ICE_CANDIDATE_COMPONENT_RTP, "udp",
- address, 1, "", "", IceCandidateType::kHost, 0,
- "1");
+ SocketAddress address("127.0.0.1", port++);
+ Candidate candidate(ICE_CANDIDATE_COMPONENT_RTP, "udp", address, 1, "", "",
+ IceCandidateType::kHost, 0, "1");
candidate_ = candidate;
- const std::string session_id = absl::StrCat(webrtc::CreateRandomId64());
- const std::string session_version = absl::StrCat(webrtc::CreateRandomId());
+ const std::string session_id = absl::StrCat(CreateRandomId64());
+ const std::string session_version = absl::StrCat(CreateRandomId());
jsep_desc_ = std::make_unique<JsepSessionDescription>(SdpType::kOffer);
ASSERT_TRUE(jsep_desc_->Initialize(CreateCricketSessionDescription(),
session_id, session_version));
@@ -113,11 +107,11 @@ class JsepSessionDescriptionTest : public ::testing::Test {
std::unique_ptr<SessionDescriptionInterface> DeSerialize(
const std::string& sdp) {
auto jsep_desc = std::make_unique<JsepSessionDescription>(SdpType::kOffer);
- EXPECT_TRUE(webrtc::SdpDeserialize(sdp, jsep_desc.get(), nullptr));
+ EXPECT_TRUE(SdpDeserialize(sdp, jsep_desc.get(), nullptr));
return std::move(jsep_desc);
}
- webrtc::Candidate candidate_;
+ Candidate candidate_;
std::unique_ptr<JsepSessionDescription> jsep_desc_;
};
@@ -140,14 +134,14 @@ TEST_F(JsepSessionDescriptionTest, CloneRollback) {
}
TEST_F(JsepSessionDescriptionTest, CloneWithCandidates) {
- webrtc::Candidate candidate_v4(
- webrtc::ICE_CANDIDATE_COMPONENT_RTP, "udp",
- webrtc::SocketAddress("192.168.1.5", 1234), kCandidatePriority, "", "",
- IceCandidateType::kSrflx, kCandidateGeneration, kCandidateFoundation);
- webrtc::Candidate candidate_v6(
- webrtc::ICE_CANDIDATE_COMPONENT_RTP, "udp",
- webrtc::SocketAddress("::1", 1234), kCandidatePriority, "", "",
- IceCandidateType::kHost, kCandidateGeneration, kCandidateFoundation);
+ Candidate candidate_v4(ICE_CANDIDATE_COMPONENT_RTP, "udp",
+ SocketAddress("192.168.1.5", 1234), kCandidatePriority,
+ "", "", IceCandidateType::kSrflx, kCandidateGeneration,
+ kCandidateFoundation);
+ Candidate candidate_v6(ICE_CANDIDATE_COMPONENT_RTP, "udp",
+ SocketAddress("::1", 1234), kCandidatePriority, "", "",
+ IceCandidateType::kHost, kCandidateGeneration,
+ kCandidateFoundation);
IceCandidate jice_v4("audio", 0, candidate_v4);
IceCandidate jice_v6("audio", 0, candidate_v6);
@@ -232,7 +226,7 @@ TEST_F(JsepSessionDescriptionTest, AddAndRemoveCandidatesWithMid) {
// The mline index should have been updated according to mid.
EXPECT_EQ(1, ice_candidate->sdp_mline_index());
- std::vector<webrtc::Candidate> candidates(1, candidate_);
+ std::vector<Candidate> candidates(1, candidate_);
candidates[0].set_transport_name(mid);
EXPECT_EQ(1u, jsep_desc_->RemoveCandidates(candidates));
EXPECT_EQ(0u, jsep_desc_->candidates(0)->count());
@@ -291,10 +285,10 @@ TEST_F(JsepSessionDescriptionTest, AddCandidateDuplicates) {
// Test that the connection address is set to a hostname address after adding a
// hostname candidate.
TEST_F(JsepSessionDescriptionTest, AddHostnameCandidate) {
- webrtc::Candidate c;
- c.set_component(webrtc::ICE_CANDIDATE_COMPONENT_RTP);
- c.set_protocol(webrtc::UDP_PROTOCOL_NAME);
- c.set_address(webrtc::SocketAddress("example.local", 1234));
+ Candidate c;
+ c.set_component(ICE_CANDIDATE_COMPONENT_RTP);
+ c.set_protocol(UDP_PROTOCOL_NAME);
+ c.set_address(SocketAddress("example.local", 1234));
c.set_type(IceCandidateType::kHost);
const size_t audio_index = 0;
IceCandidate hostname_candidate("audio", audio_index, c);
@@ -322,10 +316,10 @@ TEST_F(JsepSessionDescriptionTest, SerializeDeserialize) {
// is the default destination and deserialize it again. The connection address
// in the deserialized description should be the dummy address 0.0.0.0:9.
TEST_F(JsepSessionDescriptionTest, SerializeDeserializeWithHostnameCandidate) {
- webrtc::Candidate c;
- c.set_component(webrtc::ICE_CANDIDATE_COMPONENT_RTP);
- c.set_protocol(webrtc::UDP_PROTOCOL_NAME);
- c.set_address(webrtc::SocketAddress("example.local", 1234));
+ Candidate c;
+ c.set_component(ICE_CANDIDATE_COMPONENT_RTP);
+ c.set_protocol(UDP_PROTOCOL_NAME);
+ c.set_address(SocketAddress("example.local", 1234));
c.set_type(IceCandidateType::kHost);
const size_t audio_index = 0;
const size_t video_index = 1;
@@ -375,14 +369,14 @@ TEST_F(JsepSessionDescriptionTest, SerializeDeserializeWithCandidates) {
// is used as default address in c line according to preference.
TEST_F(JsepSessionDescriptionTest, SerializeSessionDescriptionWithIPv6Only) {
// Stun has a high preference than local host.
- webrtc::Candidate candidate1(
- webrtc::ICE_CANDIDATE_COMPONENT_RTP, "udp",
- webrtc::SocketAddress("::1", 1234), kCandidatePriority, "", "",
- IceCandidateType::kSrflx, kCandidateGeneration, kCandidateFoundation);
- webrtc::Candidate candidate2(
- webrtc::ICE_CANDIDATE_COMPONENT_RTP, "udp",
- webrtc::SocketAddress("::2", 1235), kCandidatePriority, "", "",
- IceCandidateType::kHost, kCandidateGeneration, kCandidateFoundation);
+ Candidate candidate1(ICE_CANDIDATE_COMPONENT_RTP, "udp",
+ SocketAddress("::1", 1234), kCandidatePriority, "", "",
+ IceCandidateType::kSrflx, kCandidateGeneration,
+ kCandidateFoundation);
+ Candidate candidate2(ICE_CANDIDATE_COMPONENT_RTP, "udp",
+ SocketAddress("::2", 1235), kCandidatePriority, "", "",
+ IceCandidateType::kHost, kCandidateGeneration,
+ kCandidateFoundation);
IceCandidate jice1("audio", 0, candidate1);
IceCandidate jice2("audio", 0, candidate2);
@@ -405,14 +399,14 @@ TEST_F(JsepSessionDescriptionTest, SerializeSessionDescriptionWithIPv6Only) {
// preference of IPv4 is lower.
TEST_F(JsepSessionDescriptionTest,
SerializeSessionDescriptionWithBothIPFamilies) {
- webrtc::Candidate candidate_v4(
- webrtc::ICE_CANDIDATE_COMPONENT_RTP, "udp",
- webrtc::SocketAddress("192.168.1.5", 1234), kCandidatePriority, "", "",
- IceCandidateType::kSrflx, kCandidateGeneration, kCandidateFoundation);
- webrtc::Candidate candidate_v6(
- webrtc::ICE_CANDIDATE_COMPONENT_RTP, "udp",
- webrtc::SocketAddress("::1", 1234), kCandidatePriority, "", "",
- IceCandidateType::kHost, kCandidateGeneration, kCandidateFoundation);
+ Candidate candidate_v4(ICE_CANDIDATE_COMPONENT_RTP, "udp",
+ SocketAddress("192.168.1.5", 1234), kCandidatePriority,
+ "", "", IceCandidateType::kSrflx, kCandidateGeneration,
+ kCandidateFoundation);
+ Candidate candidate_v6(ICE_CANDIDATE_COMPONENT_RTP, "udp",
+ SocketAddress("::1", 1234), kCandidatePriority, "", "",
+ IceCandidateType::kHost, kCandidateGeneration,
+ kCandidateFoundation);
IceCandidate jice_v4("audio", 0, candidate_v4);
IceCandidate jice_v6("audio", 0, candidate_v6);
@@ -436,15 +430,14 @@ TEST_F(JsepSessionDescriptionTest,
TEST_F(JsepSessionDescriptionTest,
SerializeSessionDescriptionWithBothProtocols) {
// Stun has a high preference than local host.
- webrtc::Candidate candidate1(
- webrtc::ICE_CANDIDATE_COMPONENT_RTP, "tcp",
- webrtc::SocketAddress("::1", 1234), kCandidatePriority, "", "",
- IceCandidateType::kSrflx, kCandidateGeneration, kCandidateFoundation);
- webrtc::Candidate candidate2(
- webrtc::ICE_CANDIDATE_COMPONENT_RTP, "udp",
- webrtc::SocketAddress("fe80::1234:5678:abcd:ef12", 1235),
- kCandidatePriority, "", "", IceCandidateType::kHost, kCandidateGeneration,
- kCandidateFoundation);
+ Candidate candidate1(ICE_CANDIDATE_COMPONENT_RTP, "tcp",
+ SocketAddress("::1", 1234), kCandidatePriority, "", "",
+ IceCandidateType::kSrflx, kCandidateGeneration,
+ kCandidateFoundation);
+ Candidate candidate2(ICE_CANDIDATE_COMPONENT_RTP, "udp",
+ SocketAddress("fe80::1234:5678:abcd:ef12", 1235),
+ kCandidatePriority, "", "", IceCandidateType::kHost,
+ kCandidateGeneration, kCandidateFoundation);
IceCandidate jice1("audio", 0, candidate1);
IceCandidate jice2("audio", 0, candidate2);
@@ -467,14 +460,14 @@ TEST_F(JsepSessionDescriptionTest,
// null IPv4 is used as default address in c line.
TEST_F(JsepSessionDescriptionTest, SerializeSessionDescriptionWithTCPOnly) {
// Stun has a high preference than local host.
- webrtc::Candidate candidate1(
- webrtc::ICE_CANDIDATE_COMPONENT_RTP, "tcp",
- webrtc::SocketAddress("::1", 1234), kCandidatePriority, "", "",
- IceCandidateType::kSrflx, kCandidateGeneration, kCandidateFoundation);
- webrtc::Candidate candidate2(
- webrtc::ICE_CANDIDATE_COMPONENT_RTP, "tcp",
- webrtc::SocketAddress("::2", 1235), kCandidatePriority, "", "",
- IceCandidateType::kHost, kCandidateGeneration, kCandidateFoundation);
+ Candidate candidate1(ICE_CANDIDATE_COMPONENT_RTP, "tcp",
+ SocketAddress("::1", 1234), kCandidatePriority, "", "",
+ IceCandidateType::kSrflx, kCandidateGeneration,
+ kCandidateFoundation);
+ Candidate candidate2(ICE_CANDIDATE_COMPONENT_RTP, "tcp",
+ SocketAddress("::2", 1235), kCandidatePriority, "", "",
+ IceCandidateType::kHost, kCandidateGeneration,
+ kCandidateFoundation);
IceCandidate jice1("audio", 0, candidate1);
IceCandidate jice2("audio", 0, candidate2);
@@ -494,22 +487,22 @@ TEST_F(JsepSessionDescriptionTest, SerializeSessionDescriptionWithTCPOnly) {
// Tests that the connection address will be correctly set when the Candidate is
// removed.
TEST_F(JsepSessionDescriptionTest, RemoveCandidateAndSetConnectionAddress) {
- webrtc::Candidate candidate1(
- webrtc::ICE_CANDIDATE_COMPONENT_RTP, "udp",
- webrtc::SocketAddress("::1", 1234), kCandidatePriority, "", "",
- IceCandidateType::kHost, kCandidateGeneration, kCandidateFoundation);
+ Candidate candidate1(ICE_CANDIDATE_COMPONENT_RTP, "udp",
+ SocketAddress("::1", 1234), kCandidatePriority, "", "",
+ IceCandidateType::kHost, kCandidateGeneration,
+ kCandidateFoundation);
candidate1.set_transport_name("audio");
- webrtc::Candidate candidate2(
- webrtc::ICE_CANDIDATE_COMPONENT_RTP, "tcp",
- webrtc::SocketAddress("::2", 1235), kCandidatePriority, "", "",
- IceCandidateType::kHost, kCandidateGeneration, kCandidateFoundation);
+ Candidate candidate2(ICE_CANDIDATE_COMPONENT_RTP, "tcp",
+ SocketAddress("::2", 1235), kCandidatePriority, "", "",
+ IceCandidateType::kHost, kCandidateGeneration,
+ kCandidateFoundation);
candidate2.set_transport_name("audio");
- webrtc::Candidate candidate3(
- webrtc::ICE_CANDIDATE_COMPONENT_RTP, "udp",
- webrtc::SocketAddress("192.168.1.1", 1236), kCandidatePriority, "", "",
- IceCandidateType::kHost, kCandidateGeneration, kCandidateFoundation);
+ Candidate candidate3(ICE_CANDIDATE_COMPONENT_RTP, "udp",
+ SocketAddress("192.168.1.1", 1236), kCandidatePriority,
+ "", "", IceCandidateType::kHost, kCandidateGeneration,
+ kCandidateFoundation);
candidate3.set_transport_name("audio");
IceCandidate jice1("audio", 0, candidate1);
@@ -524,7 +517,7 @@ TEST_F(JsepSessionDescriptionTest, RemoveCandidateAndSetConnectionAddress) {
ASSERT_TRUE(jsep_desc_->AddCandidate(&jice2));
ASSERT_TRUE(jsep_desc_->AddCandidate(&jice3));
- std::vector<webrtc::Candidate> candidates;
+ std::vector<Candidate> candidates;
EXPECT_EQ("192.168.1.1:1236", media_desc->connection_address().ToString());
candidates.push_back(candidate3);
@@ -549,8 +542,8 @@ class EnumerateAllSdpTypesTest : public ::testing::Test,
TEST_P(EnumerateAllSdpTypesTest, TestIdentity) {
SdpType type = GetParam();
- const char* str = webrtc::SdpTypeToString(type);
- EXPECT_EQ(type, webrtc::SdpTypeFromString(str));
+ const char* str = SdpTypeToString(type);
+ EXPECT_EQ(type, SdpTypeFromString(str));
}
INSTANTIATE_TEST_SUITE_P(JsepSessionDescriptionTest,
@@ -558,3 +551,5 @@ INSTANTIATE_TEST_SUITE_P(JsepSessionDescriptionTest,
Values(SdpType::kOffer,
SdpType::kPrAnswer,
SdpType::kAnswer));
+
+} // namespace webrtc
