Revert "Rewrite WebRtcSession media tests as PeerConnection tests"
This reverts commit 3df5dcac9b339ba4d3f4969602f094c2c8035b51.
Reason for revert: suspected of breaking chromium.webrtc.fyi:
WebRtcBrowserTest.NegotiateUnsupportedVideoCodec
WebRtcBrowserTest.NegotiateNonCryptoCall
android https://uberchromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Android%20Tests%20%28dbg%29%20%28L%20Nexus5%29/builds/25506
linux https://uberchromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Linux%20Tester/builds/38809
mac
https://uberchromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Mac%20Tester/builds/44120
windows
https://uberchromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Win10%20Tester/builds/9236
Original change's description:
> Rewrite WebRtcSession media tests as PeerConnection tests
>
> Bug: webrtc:8222
> Change-Id: I782a3227e30de70eb8f6c26a48723cb3510a84ad
> Reviewed-on: https://webrtc-review.googlesource.com/6640
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20364}
TBR=steveanton@webrtc.org,deadbeef@webrtc.org
Change-Id: Iaacc950d050ba2835d262908658dc045f234ef5b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8222
Reviewed-on: https://webrtc-review.googlesource.com/14160
Commit-Queue: Olga Sharonova <olka@webrtc.org>
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20375}diff --git a/media/base/fakemediaengine.h b/media/base/fakemediaengine.h
index 7b09dd4..29a129f 100644
--- a/media/base/fakemediaengine.h
+++ b/media/base/fakemediaengine.h
@@ -488,11 +488,12 @@
if (it != local_sinks_.end()) {
RTC_CHECK(it->second->source() == source);
} else {
- local_sinks_.insert(std::make_pair(
- ssrc, rtc::MakeUnique<VoiceChannelAudioSink>(source)));
+ local_sinks_.insert(
+ std::make_pair(ssrc, new VoiceChannelAudioSink(source)));
}
} else {
if (it != local_sinks_.end()) {
+ delete it->second;
local_sinks_.erase(it);
}
}
@@ -505,7 +506,7 @@
std::map<uint32_t, double> output_scalings_;
std::vector<DtmfInfo> dtmf_info_queue_;
AudioOptions options_;
- std::map<uint32_t, std::unique_ptr<VoiceChannelAudioSink>> local_sinks_;
+ std::map<uint32_t, VoiceChannelAudioSink*> local_sinks_;
std::unique_ptr<webrtc::AudioSinkInterface> sink_;
int max_bps_;
};
diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index 02a8e9a..c77a638 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -394,9 +394,7 @@
"peerconnection_crypto_unittest.cc",
"peerconnection_ice_unittest.cc",
"peerconnection_integrationtest.cc",
- "peerconnection_media_unittest.cc",
"peerconnection_rtp_unittest.cc",
- "peerconnection_signaling_unittest.cc",
"peerconnectionendtoend_unittest.cc",
"peerconnectionfactory_unittest.cc",
"peerconnectioninterface_unittest.cc",
@@ -465,9 +463,7 @@
"../api/audio_codecs:builtin_audio_encoder_factory",
"../api/audio_codecs/L16:audio_decoder_L16",
"../api/audio_codecs/L16:audio_encoder_L16",
- "../call:call_interfaces",
"../logging:rtc_event_log_api",
- "../logging:rtc_event_log_impl",
"../media:rtc_audio_video",
"../media:rtc_data", # TODO(phoglund): AFAIK only used for one sctp constant.
"../media:rtc_media_base",
diff --git a/pc/peerconnection.cc b/pc/peerconnection.cc
index 7cad0e1..9eb12d6 100644
--- a/pc/peerconnection.cc
+++ b/pc/peerconnection.cc
@@ -828,7 +828,10 @@
void PeerConnection::CreateOffer(CreateSessionDescriptionObserver* observer,
const MediaConstraintsInterface* constraints) {
TRACE_EVENT0("webrtc", "PeerConnection::CreateOffer");
-
+ if (!observer) {
+ LOG(LS_ERROR) << "CreateOffer - observer is NULL.";
+ return;
+ }
PeerConnectionInterface::RTCOfferAnswerOptions offer_answer_options;
// Always create an offer even if |ConvertConstraintsToOfferAnswerOptions|
// returns false for now. Because |ConvertConstraintsToOfferAnswerOptions|
@@ -845,19 +848,11 @@
void PeerConnection::CreateOffer(CreateSessionDescriptionObserver* observer,
const RTCOfferAnswerOptions& options) {
TRACE_EVENT0("webrtc", "PeerConnection::CreateOffer");
-
if (!observer) {
LOG(LS_ERROR) << "CreateOffer - observer is NULL.";
return;
}
- if (IsClosed()) {
- std::string error = "CreateOffer called when PeerConnection is closed.";
- LOG(LS_ERROR) << error;
- PostCreateSessionDescriptionFailure(observer, error);
- return;
- }
-
if (!ValidateOfferAnswerOptions(options)) {
std::string error = "CreateOffer called with invalid options.";
LOG(LS_ERROR) << error;
@@ -874,12 +869,20 @@
CreateSessionDescriptionObserver* observer,
const MediaConstraintsInterface* constraints) {
TRACE_EVENT0("webrtc", "PeerConnection::CreateAnswer");
-
if (!observer) {
LOG(LS_ERROR) << "CreateAnswer - observer is NULL.";
return;
}
+ if (!session_->remote_description() ||
+ session_->remote_description()->type() !=
+ SessionDescriptionInterface::kOffer) {
+ std::string error = "CreateAnswer called without remote offer.";
+ LOG(LS_ERROR) << error;
+ PostCreateSessionDescriptionFailure(observer, error);
+ return;
+ }
+
PeerConnectionInterface::RTCOfferAnswerOptions offer_answer_options;
if (!ConvertConstraintsToOfferAnswerOptions(constraints,
&offer_answer_options)) {
@@ -889,7 +892,9 @@
return;
}
- CreateAnswer(observer, offer_answer_options);
+ cricket::MediaSessionOptions session_options;
+ GetOptionsForAnswer(offer_answer_options, &session_options);
+ session_->CreateAnswer(observer, session_options);
}
void PeerConnection::CreateAnswer(CreateSessionDescriptionObserver* observer,
@@ -900,22 +905,6 @@
return;
}
- if (IsClosed()) {
- std::string error = "CreateAnswer called when PeerConnection is closed.";
- LOG(LS_ERROR) << error;
- PostCreateSessionDescriptionFailure(observer, error);
- return;
- }
-
- if (!session_->remote_description() ||
- session_->remote_description()->type() !=
- SessionDescriptionInterface::kOffer) {
- std::string error = "CreateAnswer called without remote offer.";
- LOG(LS_ERROR) << error;
- PostCreateSessionDescriptionFailure(observer, error);
- return;
- }
-
cricket::MediaSessionOptions session_options;
GetOptionsForAnswer(options, &session_options);
@@ -926,6 +915,9 @@
SetSessionDescriptionObserver* observer,
SessionDescriptionInterface* desc) {
TRACE_EVENT0("webrtc", "PeerConnection::SetLocalDescription");
+ if (IsClosed()) {
+ return;
+ }
if (!observer) {
LOG(LS_ERROR) << "SetLocalDescription - observer is NULL.";
return;
@@ -934,23 +926,11 @@
PostSetSessionDescriptionFailure(observer, "SessionDescription is NULL.");
return;
}
-
- // Takes the ownership of |desc| regardless of the result.
- std::unique_ptr<SessionDescriptionInterface> desc_temp(desc);
-
- if (IsClosed()) {
- std::string error = "Failed to set local " + desc->type() +
- " SDP: Called in wrong state: STATE_CLOSED";
- LOG(LS_ERROR) << error;
- PostSetSessionDescriptionFailure(observer, error);
- return;
- }
-
// Update stats here so that we have the most recent stats for tracks and
// streams that might be removed by updating the session description.
stats_->UpdateStats(kStatsOutputLevelStandard);
std::string error;
- if (!session_->SetLocalDescription(std::move(desc_temp), &error)) {
+ if (!session_->SetLocalDescription(desc, &error)) {
PostSetSessionDescriptionFailure(observer, error);
return;
}
@@ -1031,6 +1011,9 @@
SetSessionDescriptionObserver* observer,
SessionDescriptionInterface* desc) {
TRACE_EVENT0("webrtc", "PeerConnection::SetRemoteDescription");
+ if (IsClosed()) {
+ return;
+ }
if (!observer) {
LOG(LS_ERROR) << "SetRemoteDescription - observer is NULL.";
return;
@@ -1039,23 +1022,11 @@
PostSetSessionDescriptionFailure(observer, "SessionDescription is NULL.");
return;
}
-
- // Takes the ownership of |desc| regardless of the result.
- std::unique_ptr<SessionDescriptionInterface> desc_temp(desc);
-
- if (IsClosed()) {
- std::string error = "Failed to set remote " + desc->type() +
- " SDP: Called in wrong state: STATE_CLOSED";
- LOG(LS_ERROR) << error;
- PostSetSessionDescriptionFailure(observer, error);
- return;
- }
-
// Update stats here so that we have the most recent stats for tracks and
// streams that might be removed by updating the session description.
stats_->UpdateStats(kStatsOutputLevelStandard);
std::string error;
- if (!session_->SetRemoteDescription(std::move(desc_temp), &error)) {
+ if (!session_->SetRemoteDescription(desc, &error)) {
PostSetSessionDescriptionFailure(observer, error);
return;
}
@@ -1090,15 +1061,6 @@
// since only at that point will new streams have all their tracks.
rtc::scoped_refptr<StreamCollection> new_streams(StreamCollection::Create());
- // TODO(steveanton): When removing RTP senders/receivers in response to a
- // rejected media section, there is some cleanup logic that expects the voice/
- // video channel to still be set. But in this method the voice/video channel
- // would have been destroyed by WebRtcSession's SetRemoteDescription method
- // above, so the cleanup that relies on them fails to run. This is hard to fix
- // with WebRtcSession and PeerConnection separated, but once the classes are
- // merged it will be easy to call RemoveTracks right before destroying the
- // voice/video channels.
-
// Find all audio rtp streams and create corresponding remote AudioTracks
// and MediaStreams.
if (audio_content) {
diff --git a/pc/peerconnection_crypto_unittest.cc b/pc/peerconnection_crypto_unittest.cc
index 68eec08..081e11a 100644
--- a/pc/peerconnection_crypto_unittest.cc
+++ b/pc/peerconnection_crypto_unittest.cc
@@ -75,8 +75,7 @@
if (!wrapper) {
return nullptr;
}
- wrapper->AddAudioTrack("a");
- wrapper->AddVideoTrack("v");
+ wrapper->AddAudioVideoStream("s", "a", "v");
return wrapper;
}
diff --git a/pc/peerconnection_ice_unittest.cc b/pc/peerconnection_ice_unittest.cc
index 3ab9acb..0880018 100644
--- a/pc/peerconnection_ice_unittest.cc
+++ b/pc/peerconnection_ice_unittest.cc
@@ -120,8 +120,7 @@
if (!wrapper) {
return nullptr;
}
- wrapper->AddAudioTrack("a");
- wrapper->AddVideoTrack("v");
+ wrapper->AddAudioVideoStream("s", "a", "v");
return wrapper;
}
diff --git a/pc/peerconnection_media_unittest.cc b/pc/peerconnection_media_unittest.cc
deleted file mode 100644
index f106bbe..0000000
--- a/pc/peerconnection_media_unittest.cc
+++ /dev/null
@@ -1,889 +0,0 @@
-/*
- * Copyright 2017 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// This file contains tests that check the interaction between the
-// PeerConnection and the underlying media engine, as well as tests that check
-// the media-related aspects of SDP.
-
-#include <tuple>
-
-#include "call/callfactoryinterface.h"
-#include "logging/rtc_event_log/rtc_event_log_factory.h"
-#include "media/base/fakemediaengine.h"
-#include "p2p/base/fakeportallocator.h"
-#include "pc/mediasession.h"
-#include "pc/peerconnectionwrapper.h"
-#include "pc/sdputils.h"
-#ifdef WEBRTC_ANDROID
-#include "pc/test/androidtestinitializer.h"
-#endif
-#include "pc/test/fakertccertificategenerator.h"
-#include "rtc_base/gunit.h"
-#include "rtc_base/ptr_util.h"
-#include "rtc_base/virtualsocketserver.h"
-#include "test/gmock.h"
-
-namespace webrtc {
-
-using cricket::FakeMediaEngine;
-using RTCConfiguration = PeerConnectionInterface::RTCConfiguration;
-using RTCOfferAnswerOptions = PeerConnectionInterface::RTCOfferAnswerOptions;
-using ::testing::Bool;
-using ::testing::Combine;
-using ::testing::Values;
-using ::testing::ElementsAre;
-
-class PeerConnectionWrapperForMediaTest : public PeerConnectionWrapper {
- public:
- using PeerConnectionWrapper::PeerConnectionWrapper;
-
- FakeMediaEngine* media_engine() { return media_engine_; }
- void set_media_engine(FakeMediaEngine* media_engine) {
- media_engine_ = media_engine;
- }
-
- private:
- FakeMediaEngine* media_engine_;
-};
-
-class PeerConnectionMediaTest : public ::testing::Test {
- protected:
- typedef std::unique_ptr<PeerConnectionWrapperForMediaTest> WrapperPtr;
-
- PeerConnectionMediaTest()
- : vss_(new rtc::VirtualSocketServer()), main_(vss_.get()) {
-#ifdef WEBRTC_ANDROID
- InitializeAndroidObjects();
-#endif
- }
-
- WrapperPtr CreatePeerConnection() {
- return CreatePeerConnection(RTCConfiguration());
- }
-
- WrapperPtr CreatePeerConnection(const RTCConfiguration& config) {
- auto media_engine = rtc::MakeUnique<FakeMediaEngine>();
- auto* media_engine_ptr = media_engine.get();
- auto pc_factory = CreateModularPeerConnectionFactory(
- rtc::Thread::Current(), rtc::Thread::Current(), rtc::Thread::Current(),
- std::move(media_engine), CreateCallFactory(),
- CreateRtcEventLogFactory());
-
- auto fake_port_allocator = rtc::MakeUnique<cricket::FakePortAllocator>(
- rtc::Thread::Current(), nullptr);
- auto observer = rtc::MakeUnique<MockPeerConnectionObserver>();
- auto pc = pc_factory->CreatePeerConnection(
- config, std::move(fake_port_allocator), nullptr, observer.get());
- if (!pc) {
- return nullptr;
- }
-
- auto wrapper = rtc::MakeUnique<PeerConnectionWrapperForMediaTest>(
- pc_factory, pc, std::move(observer));
- wrapper->set_media_engine(media_engine_ptr);
- return wrapper;
- }
-
- // Accepts the same arguments as CreatePeerConnection and adds default audio
- // and video tracks.
- template <typename... Args>
- WrapperPtr CreatePeerConnectionWithAudioVideo(Args&&... args) {
- auto wrapper = CreatePeerConnection(std::forward<Args>(args)...);
- if (!wrapper) {
- return nullptr;
- }
- wrapper->AddAudioTrack("a");
- wrapper->AddVideoTrack("v");
- return wrapper;
- }
-
- const cricket::MediaContentDescription* GetMediaContent(
- const SessionDescriptionInterface* sdesc,
- const std::string& mid) {
- const auto* content_desc =
- sdesc->description()->GetContentDescriptionByName(mid);
- return static_cast<const cricket::MediaContentDescription*>(content_desc);
- }
-
- cricket::MediaContentDirection GetMediaContentDirection(
- const SessionDescriptionInterface* sdesc,
- const std::string& mid) {
- auto* media_content = GetMediaContent(sdesc, mid);
- RTC_DCHECK(media_content);
- return media_content->direction();
- }
-
- std::unique_ptr<rtc::VirtualSocketServer> vss_;
- rtc::AutoSocketServerThread main_;
-};
-
-TEST_F(PeerConnectionMediaTest,
- FailToSetRemoteDescriptionIfCreateMediaChannelFails) {
- auto caller = CreatePeerConnectionWithAudioVideo();
- auto callee = CreatePeerConnectionWithAudioVideo();
- callee->media_engine()->set_fail_create_channel(true);
-
- std::string error;
- ASSERT_FALSE(callee->SetRemoteDescription(caller->CreateOffer(), &error));
- EXPECT_EQ("Failed to set remote offer SDP: Failed to create channels.",
- error);
-}
-
-TEST_F(PeerConnectionMediaTest,
- FailToSetLocalDescriptionIfCreateMediaChannelFails) {
- auto caller = CreatePeerConnectionWithAudioVideo();
- caller->media_engine()->set_fail_create_channel(true);
-
- std::string error;
- ASSERT_FALSE(caller->SetLocalDescription(caller->CreateOffer(), &error));
- EXPECT_EQ("Failed to set local offer SDP: Failed to create channels.", error);
-}
-
-std::vector<std::string> GetIds(
- const std::vector<cricket::StreamParams>& streams) {
- std::vector<std::string> ids;
- for (const auto& stream : streams) {
- ids.push_back(stream.id);
- }
- return ids;
-}
-
-// Test that exchanging an offer and answer with each side having an audio and
-// video stream creates the appropriate send/recv streams in the underlying
-// media engine on both sides.
-TEST_F(PeerConnectionMediaTest, AudioVideoOfferAnswerCreateSendRecvStreams) {
- const std::string kCallerAudioId = "caller_a";
- const std::string kCallerVideoId = "caller_v";
- const std::string kCalleeAudioId = "callee_a";
- const std::string kCalleeVideoId = "callee_v";
-
- auto caller = CreatePeerConnection();
- caller->AddAudioTrack(kCallerAudioId);
- caller->AddVideoTrack(kCallerVideoId);
-
- auto callee = CreatePeerConnection();
- callee->AddAudioTrack(kCalleeAudioId);
- callee->AddVideoTrack(kCalleeVideoId);
-
- ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
- ASSERT_TRUE(
- caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
-
- auto* caller_voice = caller->media_engine()->GetVoiceChannel(0);
- EXPECT_THAT(GetIds(caller_voice->recv_streams()),
- ElementsAre(kCalleeAudioId));
- EXPECT_THAT(GetIds(caller_voice->send_streams()),
- ElementsAre(kCallerAudioId));
-
- auto* caller_video = caller->media_engine()->GetVideoChannel(0);
- EXPECT_THAT(GetIds(caller_video->recv_streams()),
- ElementsAre(kCalleeVideoId));
- EXPECT_THAT(GetIds(caller_video->send_streams()),
- ElementsAre(kCallerVideoId));
-
- auto* callee_voice = callee->media_engine()->GetVoiceChannel(0);
- EXPECT_THAT(GetIds(callee_voice->recv_streams()),
- ElementsAre(kCallerAudioId));
- EXPECT_THAT(GetIds(callee_voice->send_streams()),
- ElementsAre(kCalleeAudioId));
-
- auto* callee_video = callee->media_engine()->GetVideoChannel(0);
- EXPECT_THAT(GetIds(callee_video->recv_streams()),
- ElementsAre(kCallerVideoId));
- EXPECT_THAT(GetIds(callee_video->send_streams()),
- ElementsAre(kCalleeVideoId));
-}
-
-// Test that removing streams from a subsequent offer causes the receive streams
-// on the callee to be removed.
-TEST_F(PeerConnectionMediaTest, EmptyRemoteOfferRemovesRecvStreams) {
- auto caller = CreatePeerConnection();
- auto caller_audio_track = caller->AddAudioTrack("a");
- auto caller_video_track = caller->AddVideoTrack("v");
- auto callee = CreatePeerConnectionWithAudioVideo();
-
- ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
- ASSERT_TRUE(
- caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
-
- // Remove both tracks from caller.
- caller->pc()->RemoveTrack(caller_audio_track);
- caller->pc()->RemoveTrack(caller_video_track);
-
- ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
- ASSERT_TRUE(
- caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
-
- auto callee_voice = callee->media_engine()->GetVoiceChannel(0);
- EXPECT_EQ(1u, callee_voice->send_streams().size());
- EXPECT_EQ(0u, callee_voice->recv_streams().size());
-
- auto callee_video = callee->media_engine()->GetVideoChannel(0);
- EXPECT_EQ(1u, callee_video->send_streams().size());
- EXPECT_EQ(0u, callee_video->recv_streams().size());
-}
-
-// Test that removing streams from a subsequent answer causes the send streams
-// on the callee to be removed when applied locally.
-TEST_F(PeerConnectionMediaTest, EmptyLocalAnswerRemovesSendStreams) {
- auto caller = CreatePeerConnectionWithAudioVideo();
- auto callee = CreatePeerConnection();
- auto callee_audio_track = callee->AddAudioTrack("a");
- auto callee_video_track = callee->AddVideoTrack("v");
-
- ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
- ASSERT_TRUE(
- caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
-
- // Remove both tracks from callee.
- callee->pc()->RemoveTrack(callee_audio_track);
- callee->pc()->RemoveTrack(callee_video_track);
-
- ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
- ASSERT_TRUE(
- caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
-
- auto callee_voice = callee->media_engine()->GetVoiceChannel(0);
- EXPECT_EQ(0u, callee_voice->send_streams().size());
- EXPECT_EQ(1u, callee_voice->recv_streams().size());
-
- auto callee_video = callee->media_engine()->GetVideoChannel(0);
- EXPECT_EQ(0u, callee_video->send_streams().size());
- EXPECT_EQ(1u, callee_video->recv_streams().size());
-}
-
-// Test that a new stream in a subsequent offer causes a new receive stream to
-// be created on the callee.
-TEST_F(PeerConnectionMediaTest, NewStreamInRemoteOfferAddsRecvStreams) {
- auto caller = CreatePeerConnectionWithAudioVideo();
- auto callee = CreatePeerConnection();
-
- ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
- ASSERT_TRUE(
- caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
-
- // Add second set of tracks to the caller.
- caller->AddAudioTrack("a2");
- caller->AddVideoTrack("v2");
-
- ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
- ASSERT_TRUE(
- caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
-
- auto callee_voice = callee->media_engine()->GetVoiceChannel(0);
- EXPECT_EQ(2u, callee_voice->recv_streams().size());
- auto callee_video = callee->media_engine()->GetVideoChannel(0);
- EXPECT_EQ(2u, callee_video->recv_streams().size());
-}
-
-// Test that a new stream in a subsequent answer causes a new send stream to be
-// created on the callee when added locally.
-TEST_F(PeerConnectionMediaTest, NewStreamInLocalAnswerAddsSendStreams) {
- auto caller = CreatePeerConnection();
- auto callee = CreatePeerConnectionWithAudioVideo();
-
- RTCOfferAnswerOptions options;
- options.offer_to_receive_audio =
- RTCOfferAnswerOptions::kOfferToReceiveMediaTrue;
- options.offer_to_receive_video =
- RTCOfferAnswerOptions::kOfferToReceiveMediaTrue;
-
- ASSERT_TRUE(
- callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(options)));
- ASSERT_TRUE(
- caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
-
- // Add second set of tracks to the callee.
- callee->AddAudioTrack("a2");
- callee->AddVideoTrack("v2");
-
- ASSERT_TRUE(
- callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(options)));
- ASSERT_TRUE(
- caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
-
- auto callee_voice = callee->media_engine()->GetVoiceChannel(0);
- EXPECT_EQ(2u, callee_voice->send_streams().size());
- auto callee_video = callee->media_engine()->GetVideoChannel(0);
- EXPECT_EQ(2u, callee_video->send_streams().size());
-}
-
-// A PeerConnection with no local streams and no explicit answer constraints
-// should not reject any offered media sections.
-TEST_F(PeerConnectionMediaTest,
- CreateAnswerWithNoStreamsAndDefaultOptionsDoesNotReject) {
- auto caller = CreatePeerConnectionWithAudioVideo();
- auto callee = CreatePeerConnection();
- ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
- auto answer = callee->CreateAnswer();
-
- const auto* audio_content =
- cricket::GetFirstAudioContent(answer->description());
- ASSERT_TRUE(audio_content);
- EXPECT_FALSE(audio_content->rejected);
-
- const auto* video_content =
- cricket::GetFirstVideoContent(answer->description());
- ASSERT_TRUE(video_content);
- EXPECT_FALSE(video_content->rejected);
-}
-
-class PeerConnectionMediaOfferDirectionTest
- : public PeerConnectionMediaTest,
- public ::testing::WithParamInterface<
- std::tuple<bool, int, cricket::MediaContentDirection>> {
- protected:
- PeerConnectionMediaOfferDirectionTest() {
- send_media_ = std::get<0>(GetParam());
- offer_to_receive_ = std::get<1>(GetParam());
- expected_direction_ = std::get<2>(GetParam());
- }
-
- bool send_media_;
- int offer_to_receive_;
- cricket::MediaContentDirection expected_direction_;
-};
-
-// Tests that the correct direction is set on the media description according
-// to the presence of a local media track and the offer_to_receive setting.
-TEST_P(PeerConnectionMediaOfferDirectionTest, VerifyDirection) {
- auto caller = CreatePeerConnection();
- if (send_media_) {
- caller->AddAudioTrack("a");
- }
-
- RTCOfferAnswerOptions options;
- options.offer_to_receive_audio = offer_to_receive_;
- auto offer = caller->CreateOffer(options);
-
- auto* media_content = GetMediaContent(offer.get(), cricket::CN_AUDIO);
- if (expected_direction_ == cricket::MD_INACTIVE) {
- EXPECT_FALSE(media_content);
- } else {
- EXPECT_EQ(expected_direction_, media_content->direction());
- }
-}
-
-// Note that in these tests, MD_INACTIVE indicates that no media section is
-// included in the offer, not that the media direction is inactive.
-INSTANTIATE_TEST_CASE_P(PeerConnectionMediaTest,
- PeerConnectionMediaOfferDirectionTest,
- Values(std::make_tuple(false, -1, cricket::MD_INACTIVE),
- std::make_tuple(false, 0, cricket::MD_INACTIVE),
- std::make_tuple(false, 1, cricket::MD_RECVONLY),
- std::make_tuple(true, -1, cricket::MD_SENDRECV),
- std::make_tuple(true, 0, cricket::MD_SENDONLY),
- std::make_tuple(true, 1, cricket::MD_SENDRECV)));
-
-class PeerConnectionMediaAnswerDirectionTest
- : public PeerConnectionMediaTest,
- public ::testing::WithParamInterface<
- std::tuple<cricket::MediaContentDirection, bool, int>> {
- protected:
- PeerConnectionMediaAnswerDirectionTest() {
- offer_direction_ = std::get<0>(GetParam());
- send_media_ = std::get<1>(GetParam());
- offer_to_receive_ = std::get<2>(GetParam());
- }
-
- cricket::MediaContentDirection offer_direction_;
- bool send_media_;
- int offer_to_receive_;
-};
-
-// Tests that the direction in an answer is correct according to direction sent
-// in the offer, the presence of a local media track on the receive side and the
-// offer_to_receive setting.
-TEST_P(PeerConnectionMediaAnswerDirectionTest, VerifyDirection) {
- auto caller = CreatePeerConnection();
- caller->AddAudioTrack("a");
-
- // Create the offer with an audio section and set its direction.
- auto offer = caller->CreateOffer();
- cricket::GetFirstAudioContentDescription(offer->description())
- ->set_direction(offer_direction_);
-
- auto callee = CreatePeerConnection();
- if (send_media_) {
- callee->AddAudioTrack("a");
- }
- ASSERT_TRUE(callee->SetRemoteDescription(std::move(offer)));
-
- // Create the answer according to the test parameters.
- RTCOfferAnswerOptions options;
- options.offer_to_receive_audio = offer_to_receive_;
- auto answer = callee->CreateAnswer(options);
-
- // The expected direction in the answer is the intersection of each side's
- // capability to send/recv media.
- // For the offerer, the direction is given in the offer (offer_direction_).
- // For the answerer, the direction has two components:
- // 1. Send if the answerer has a local track to send.
- // 2. Receive if the answerer has explicitly set the offer_to_receive to 1 or
- // if it has been left as default.
- auto offer_direction =
- cricket::RtpTransceiverDirection::FromMediaContentDirection(
- offer_direction_);
-
- // The negotiated components determine the direction set in the answer.
- bool negotiate_send = (send_media_ && offer_direction.recv);
- bool negotiate_recv = ((offer_to_receive_ != 0) && offer_direction.send);
-
- auto expected_direction =
- cricket::RtpTransceiverDirection(negotiate_send, negotiate_recv)
- .ToMediaContentDirection();
- EXPECT_EQ(expected_direction,
- GetMediaContentDirection(answer.get(), cricket::CN_AUDIO));
-}
-
-// Tests that the media section is rejected if and only if the callee has no
-// local media track and has set offer_to_receive to 0, no matter which
-// direction the caller indicated in the offer.
-TEST_P(PeerConnectionMediaAnswerDirectionTest, VerifyRejected) {
- auto caller = CreatePeerConnection();
- caller->AddAudioTrack("a");
-
- // Create the offer with an audio section and set its direction.
- auto offer = caller->CreateOffer();
- cricket::GetFirstAudioContentDescription(offer->description())
- ->set_direction(offer_direction_);
-
- auto callee = CreatePeerConnection();
- if (send_media_) {
- callee->AddAudioTrack("a");
- }
- ASSERT_TRUE(callee->SetRemoteDescription(std::move(offer)));
-
- // Create the answer according to the test parameters.
- RTCOfferAnswerOptions options;
- options.offer_to_receive_audio = offer_to_receive_;
- auto answer = callee->CreateAnswer(options);
-
- // The media section is rejected if and only if offer_to_receive is explicitly
- // set to 0 and there is no media to send.
- auto* audio_content = cricket::GetFirstAudioContent(answer->description());
- ASSERT_TRUE(audio_content);
- EXPECT_EQ((offer_to_receive_ == 0 && !send_media_), audio_content->rejected);
-}
-
-INSTANTIATE_TEST_CASE_P(PeerConnectionMediaTest,
- PeerConnectionMediaAnswerDirectionTest,
- Combine(Values(cricket::MD_INACTIVE,
- cricket::MD_SENDONLY,
- cricket::MD_RECVONLY,
- cricket::MD_SENDRECV),
- Bool(),
- Values(-1, 0, 1)));
-
-TEST_F(PeerConnectionMediaTest, OfferHasDifferentDirectionForAudioVideo) {
- auto caller = CreatePeerConnection();
- caller->AddVideoTrack("v");
-
- RTCOfferAnswerOptions options;
- options.offer_to_receive_audio = 1;
- options.offer_to_receive_video = 0;
- auto offer = caller->CreateOffer(options);
-
- EXPECT_EQ(cricket::MD_RECVONLY,
- GetMediaContentDirection(offer.get(), cricket::CN_AUDIO));
- EXPECT_EQ(cricket::MD_SENDONLY,
- GetMediaContentDirection(offer.get(), cricket::CN_VIDEO));
-}
-
-TEST_F(PeerConnectionMediaTest, AnswerHasDifferentDirectionsForAudioVideo) {
- auto caller = CreatePeerConnectionWithAudioVideo();
- auto callee = CreatePeerConnection();
- callee->AddVideoTrack("v");
-
- ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
-
- RTCOfferAnswerOptions options;
- options.offer_to_receive_audio = 1;
- options.offer_to_receive_video = 0;
- auto answer = callee->CreateAnswer(options);
-
- EXPECT_EQ(cricket::MD_RECVONLY,
- GetMediaContentDirection(answer.get(), cricket::CN_AUDIO));
- EXPECT_EQ(cricket::MD_SENDONLY,
- GetMediaContentDirection(answer.get(), cricket::CN_VIDEO));
-}
-
-void AddComfortNoiseCodecsToSend(cricket::FakeMediaEngine* media_engine) {
- const cricket::AudioCodec kComfortNoiseCodec8k(102, "CN", 8000, 0, 1);
- const cricket::AudioCodec kComfortNoiseCodec16k(103, "CN", 16000, 0, 1);
-
- auto codecs = media_engine->audio_send_codecs();
- codecs.push_back(kComfortNoiseCodec8k);
- codecs.push_back(kComfortNoiseCodec16k);
- media_engine->SetAudioCodecs(codecs);
-}
-
-bool HasAnyComfortNoiseCodecs(const cricket::SessionDescription* desc) {
- const auto* audio_desc = cricket::GetFirstAudioContentDescription(desc);
- for (const auto& codec : audio_desc->codecs()) {
- if (codec.name == "CN") {
- return true;
- }
- }
- return false;
-}
-
-TEST_F(PeerConnectionMediaTest,
- CreateOfferWithNoVoiceActivityDetectionIncludesNoComfortNoiseCodecs) {
- auto caller = CreatePeerConnectionWithAudioVideo();
- AddComfortNoiseCodecsToSend(caller->media_engine());
-
- RTCOfferAnswerOptions options;
- options.voice_activity_detection = false;
- auto offer = caller->CreateOffer(options);
-
- EXPECT_FALSE(HasAnyComfortNoiseCodecs(offer->description()));
-}
-
-TEST_F(PeerConnectionMediaTest,
- CreateAnswerWithNoVoiceActivityDetectionIncludesNoComfortNoiseCodecs) {
- auto caller = CreatePeerConnectionWithAudioVideo();
- AddComfortNoiseCodecsToSend(caller->media_engine());
- auto callee = CreatePeerConnectionWithAudioVideo();
- AddComfortNoiseCodecsToSend(callee->media_engine());
-
- ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
-
- RTCOfferAnswerOptions options;
- options.voice_activity_detection = false;
- auto answer = callee->CreateAnswer(options);
-
- EXPECT_FALSE(HasAnyComfortNoiseCodecs(answer->description()));
-}
-
-// The following test group verifies that we reject answers with invalid media
-// sections as per RFC 3264.
-
-class PeerConnectionMediaInvalidMediaTest
- : public PeerConnectionMediaTest,
- public ::testing::WithParamInterface<
- std::tuple<std::string,
- std::function<void(cricket::SessionDescription*)>,
- std::string>> {
- protected:
- PeerConnectionMediaInvalidMediaTest() {
- mutator_ = std::get<1>(GetParam());
- expected_error_ = std::get<2>(GetParam());
- }
-
- std::function<void(cricket::SessionDescription*)> mutator_;
- std::string expected_error_;
-};
-
-TEST_P(PeerConnectionMediaInvalidMediaTest, FailToSetRemoteAnswer) {
- auto caller = CreatePeerConnectionWithAudioVideo();
- auto callee = CreatePeerConnectionWithAudioVideo();
-
- ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
-
- auto answer = callee->CreateAnswer();
- mutator_(answer->description());
-
- std::string error;
- ASSERT_FALSE(caller->SetRemoteDescription(std::move(answer), &error));
- EXPECT_EQ("Failed to set remote answer SDP: " + expected_error_, error);
-}
-
-TEST_P(PeerConnectionMediaInvalidMediaTest, FailToSetLocalAnswer) {
- auto caller = CreatePeerConnectionWithAudioVideo();
- auto callee = CreatePeerConnectionWithAudioVideo();
-
- ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
-
- auto answer = callee->CreateAnswer();
- mutator_(answer->description());
-
- std::string error;
- ASSERT_FALSE(callee->SetLocalDescription(std::move(answer), &error));
- EXPECT_EQ("Failed to set local answer SDP: " + expected_error_, error);
-}
-
-void RemoveVideoContent(cricket::SessionDescription* desc) {
- auto content_name = cricket::GetFirstVideoContent(desc)->name;
- desc->RemoveContentByName(content_name);
- desc->RemoveTransportInfoByName(content_name);
-}
-
-void RenameVideoContent(cricket::SessionDescription* desc) {
- auto* video_content = cricket::GetFirstVideoContent(desc);
- auto* transport_info = desc->GetTransportInfoByName(video_content->name);
- video_content->name = "video_renamed";
- transport_info->content_name = video_content->name;
-}
-
-void ReverseMediaContent(cricket::SessionDescription* desc) {
- std::reverse(desc->contents().begin(), desc->contents().end());
- std::reverse(desc->transport_infos().begin(), desc->transport_infos().end());
-}
-
-void ChangeMediaTypeAudioToVideo(cricket::SessionDescription* desc) {
- desc->RemoveContentByName(cricket::CN_AUDIO);
- auto* video_content = desc->GetContentByName(cricket::CN_VIDEO);
- desc->AddContent(cricket::CN_AUDIO, cricket::NS_JINGLE_RTP,
- video_content->description->Copy());
-}
-
-constexpr char kMLinesOutOfOrder[] =
- "The order of m-lines in answer doesn't match order in offer. Rejecting "
- "answer.";
-
-INSTANTIATE_TEST_CASE_P(
- PeerConnectionMediaTest,
- PeerConnectionMediaInvalidMediaTest,
- Values(
- std::make_tuple("remove video", RemoveVideoContent, kMLinesOutOfOrder),
- std::make_tuple("rename video", RenameVideoContent, kMLinesOutOfOrder),
- std::make_tuple("reverse media sections",
- ReverseMediaContent,
- kMLinesOutOfOrder),
- std::make_tuple("change audio type to video type",
- ChangeMediaTypeAudioToVideo,
- kMLinesOutOfOrder)));
-
-// Test that the correct media engine send/recv streams are created when doing
-// a series of offer/answers where audio/video are both sent, then audio is
-// rejected, then both audio/video sent again.
-TEST_F(PeerConnectionMediaTest, TestAVOfferWithAudioOnlyAnswer) {
- RTCOfferAnswerOptions options_reject_video;
- options_reject_video.offer_to_receive_audio =
- RTCOfferAnswerOptions::kOfferToReceiveMediaTrue;
- options_reject_video.offer_to_receive_video = 0;
-
- auto caller = CreatePeerConnection();
- caller->AddAudioTrack("a");
- caller->AddVideoTrack("v");
- auto callee = CreatePeerConnection();
-
- // Caller initially offers to send/recv audio and video.
- ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
- // Callee accepts the audio as recv only but rejects the video.
- ASSERT_TRUE(caller->SetRemoteDescription(
- callee->CreateAnswerAndSetAsLocal(options_reject_video)));
-
- auto caller_voice = caller->media_engine()->GetVoiceChannel(0);
- ASSERT_TRUE(caller_voice);
- EXPECT_EQ(0u, caller_voice->recv_streams().size());
- EXPECT_EQ(1u, caller_voice->send_streams().size());
- auto caller_video = caller->media_engine()->GetVideoChannel(0);
- EXPECT_FALSE(caller_video);
-
- // Callee adds its own audio/video stream and offers to receive audio/video
- // too.
- callee->AddAudioTrack("a");
- auto callee_video_track = callee->AddVideoTrack("v");
- ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
- ASSERT_TRUE(
- caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
-
- auto callee_voice = callee->media_engine()->GetVoiceChannel(0);
- ASSERT_TRUE(callee_voice);
- EXPECT_EQ(1u, callee_voice->recv_streams().size());
- EXPECT_EQ(1u, callee_voice->send_streams().size());
- auto callee_video = callee->media_engine()->GetVideoChannel(0);
- ASSERT_TRUE(callee_video);
- EXPECT_EQ(1u, callee_video->recv_streams().size());
- EXPECT_EQ(1u, callee_video->send_streams().size());
-
- // Callee removes video but keeps audio and rejects the video once again.
- callee->pc()->RemoveTrack(callee_video_track);
- ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
- ASSERT_TRUE(
- callee->SetLocalDescription(callee->CreateAnswer(options_reject_video)));
-
- callee_voice = callee->media_engine()->GetVoiceChannel(0);
- ASSERT_TRUE(callee_voice);
- EXPECT_EQ(1u, callee_voice->recv_streams().size());
- EXPECT_EQ(1u, callee_voice->send_streams().size());
- callee_video = callee->media_engine()->GetVideoChannel(0);
- EXPECT_FALSE(callee_video);
-}
-
-// Test that the correct media engine send/recv streams are created when doing
-// a series of offer/answers where audio/video are both sent, then video is
-// rejected, then both audio/video sent again.
-TEST_F(PeerConnectionMediaTest, TestAVOfferWithVideoOnlyAnswer) {
- // Disable the bundling here. If the media is bundled on audio
- // transport, then we can't reject the audio because switching the bundled
- // transport is not currently supported.
- // (https://bugs.chromium.org/p/webrtc/issues/detail?id=6704)
- RTCOfferAnswerOptions options_no_bundle;
- options_no_bundle.use_rtp_mux = false;
- RTCOfferAnswerOptions options_reject_audio = options_no_bundle;
- options_reject_audio.offer_to_receive_audio = 0;
- options_reject_audio.offer_to_receive_video =
- RTCOfferAnswerOptions::kMaxOfferToReceiveMedia;
-
- auto caller = CreatePeerConnection();
- caller->AddAudioTrack("a");
- caller->AddVideoTrack("v");
- auto callee = CreatePeerConnection();
-
- // Caller initially offers to send/recv audio and video.
- ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
- // Callee accepts the video as recv only but rejects the audio.
- ASSERT_TRUE(caller->SetRemoteDescription(
- callee->CreateAnswerAndSetAsLocal(options_reject_audio)));
-
- auto caller_voice = caller->media_engine()->GetVoiceChannel(0);
- EXPECT_FALSE(caller_voice);
- auto caller_video = caller->media_engine()->GetVideoChannel(0);
- ASSERT_TRUE(caller_video);
- EXPECT_EQ(0u, caller_video->recv_streams().size());
- EXPECT_EQ(1u, caller_video->send_streams().size());
-
- // Callee adds its own audio/video stream and offers to receive audio/video
- // too.
- auto callee_audio_track = callee->AddAudioTrack("a");
- callee->AddVideoTrack("v");
- ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
- ASSERT_TRUE(caller->SetRemoteDescription(
- callee->CreateAnswerAndSetAsLocal(options_no_bundle)));
-
- auto callee_voice = callee->media_engine()->GetVoiceChannel(0);
- ASSERT_TRUE(callee_voice);
- EXPECT_EQ(1u, callee_voice->recv_streams().size());
- EXPECT_EQ(1u, callee_voice->send_streams().size());
- auto callee_video = callee->media_engine()->GetVideoChannel(0);
- ASSERT_TRUE(callee_video);
- EXPECT_EQ(1u, callee_video->recv_streams().size());
- EXPECT_EQ(1u, callee_video->send_streams().size());
-
- // Callee removes audio but keeps video and rejects the audio once again.
- callee->pc()->RemoveTrack(callee_audio_track);
- ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
- ASSERT_TRUE(
- callee->SetLocalDescription(callee->CreateAnswer(options_reject_audio)));
-
- callee_voice = callee->media_engine()->GetVoiceChannel(0);
- EXPECT_FALSE(callee_voice);
- callee_video = callee->media_engine()->GetVideoChannel(0);
- ASSERT_TRUE(callee_video);
- EXPECT_EQ(1u, callee_video->recv_streams().size());
- EXPECT_EQ(1u, callee_video->send_streams().size());
-}
-
-// Tests that if the underlying video encoder fails to be initialized (signaled
-// by failing to set send codecs), the PeerConnection signals the error to the
-// client.
-TEST_F(PeerConnectionMediaTest, MediaEngineErrorPropagatedToClients) {
- auto caller = CreatePeerConnectionWithAudioVideo();
- auto callee = CreatePeerConnectionWithAudioVideo();
-
- ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
-
- auto video_channel = caller->media_engine()->GetVideoChannel(0);
- video_channel->set_fail_set_send_codecs(true);
-
- std::string error;
- ASSERT_FALSE(caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal(),
- &error));
- EXPECT_EQ(
- "Failed to set remote answer SDP: Session error code: ERROR_CONTENT. "
- "Session error description: Failed to set remote video description send "
- "parameters..",
- error);
-}
-
-// Tests that if the underlying video encoder fails once then subsequent
-// attempts at setting the local/remote description will also fail, even if
-// SetSendCodecs no longer fails.
-TEST_F(PeerConnectionMediaTest,
- FailToApplyDescriptionIfVideoEncoderHasEverFailed) {
- auto caller = CreatePeerConnectionWithAudioVideo();
- auto callee = CreatePeerConnectionWithAudioVideo();
-
- ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
-
- auto video_channel = caller->media_engine()->GetVideoChannel(0);
- video_channel->set_fail_set_send_codecs(true);
-
- EXPECT_FALSE(
- caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
-
- video_channel->set_fail_set_send_codecs(false);
-
- EXPECT_FALSE(caller->SetRemoteDescription(callee->CreateAnswer()));
- EXPECT_FALSE(caller->SetLocalDescription(caller->CreateOffer()));
-}
-
-void RenameContent(cricket::SessionDescription* desc,
- const std::string& old_name,
- const std::string& new_name) {
- auto* content = desc->GetContentByName(old_name);
- RTC_DCHECK(content);
- content->name = new_name;
- auto* transport = desc->GetTransportInfoByName(old_name);
- RTC_DCHECK(transport);
- transport->content_name = new_name;
-}
-
-// Tests that an answer responds with the same MIDs as the offer.
-TEST_F(PeerConnectionMediaTest, AnswerHasSameMidsAsOffer) {
- const std::string kAudioMid = "not default1";
- const std::string kVideoMid = "not default2";
-
- auto caller = CreatePeerConnectionWithAudioVideo();
- auto callee = CreatePeerConnectionWithAudioVideo();
-
- auto offer = caller->CreateOffer();
- RenameContent(offer->description(), cricket::CN_AUDIO, kAudioMid);
- RenameContent(offer->description(), cricket::CN_VIDEO, kVideoMid);
- ASSERT_TRUE(callee->SetRemoteDescription(std::move(offer)));
-
- auto answer = callee->CreateAnswer();
- EXPECT_EQ(kAudioMid,
- cricket::GetFirstAudioContent(answer->description())->name);
- EXPECT_EQ(kVideoMid,
- cricket::GetFirstVideoContent(answer->description())->name);
-}
-
-// Test that if the callee creates a re-offer, the MIDs are the same as the
-// original offer.
-TEST_F(PeerConnectionMediaTest, ReOfferHasSameMidsAsFirstOffer) {
- const std::string kAudioMid = "not default1";
- const std::string kVideoMid = "not default2";
-
- auto caller = CreatePeerConnectionWithAudioVideo();
- auto callee = CreatePeerConnectionWithAudioVideo();
-
- auto offer = caller->CreateOffer();
- RenameContent(offer->description(), cricket::CN_AUDIO, kAudioMid);
- RenameContent(offer->description(), cricket::CN_VIDEO, kVideoMid);
- ASSERT_TRUE(callee->SetRemoteDescription(std::move(offer)));
- ASSERT_TRUE(callee->SetLocalDescription(callee->CreateAnswer()));
-
- auto reoffer = callee->CreateOffer();
- EXPECT_EQ(kAudioMid,
- cricket::GetFirstAudioContent(reoffer->description())->name);
- EXPECT_EQ(kVideoMid,
- cricket::GetFirstVideoContent(reoffer->description())->name);
-}
-
-TEST_F(PeerConnectionMediaTest,
- CombinedAudioVideoBweConfigPropagatedToMediaEngine) {
- RTCConfiguration config;
- config.combined_audio_video_bwe.emplace(true);
- auto caller = CreatePeerConnectionWithAudioVideo(config);
-
- ASSERT_TRUE(caller->SetLocalDescription(caller->CreateOffer()));
-
- auto caller_voice = caller->media_engine()->GetVoiceChannel(0);
- ASSERT_TRUE(caller_voice);
- const cricket::AudioOptions& audio_options = caller_voice->options();
- EXPECT_EQ(config.combined_audio_video_bwe,
- audio_options.combined_audio_video_bwe);
-}
-
-} // namespace webrtc
diff --git a/pc/peerconnection_signaling_unittest.cc b/pc/peerconnection_signaling_unittest.cc
deleted file mode 100644
index caaac4c..0000000
--- a/pc/peerconnection_signaling_unittest.cc
+++ /dev/null
@@ -1,501 +0,0 @@
-/*
- * Copyright 2017 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// This file contains tests that check the PeerConnection's signaling state
-// machine, as well as tests that check basic, media-agnostic aspects of SDP.
-
-#include <tuple>
-
-#include "api/audio_codecs/builtin_audio_decoder_factory.h"
-#include "api/audio_codecs/builtin_audio_encoder_factory.h"
-#include "api/peerconnectionproxy.h"
-#include "pc/peerconnection.h"
-#include "pc/peerconnectionwrapper.h"
-#include "pc/sdputils.h"
-#ifdef WEBRTC_ANDROID
-#include "pc/test/androidtestinitializer.h"
-#endif
-#include "pc/test/fakeaudiocapturemodule.h"
-#include "pc/test/fakertccertificategenerator.h"
-#include "rtc_base/gunit.h"
-#include "rtc_base/ptr_util.h"
-#include "rtc_base/stringutils.h"
-#include "rtc_base/virtualsocketserver.h"
-#include "test/gmock.h"
-
-namespace webrtc {
-
-using SignalingState = PeerConnectionInterface::SignalingState;
-using RTCConfiguration = PeerConnectionInterface::RTCConfiguration;
-using RTCOfferAnswerOptions = PeerConnectionInterface::RTCOfferAnswerOptions;
-using ::testing::Bool;
-using ::testing::Combine;
-using ::testing::Values;
-
-class PeerConnectionWrapperForSignalingTest : public PeerConnectionWrapper {
- public:
- using PeerConnectionWrapper::PeerConnectionWrapper;
-
- bool initial_offerer() {
- return GetInternalPeerConnection()->initial_offerer();
- }
-
- PeerConnection* GetInternalPeerConnection() {
- auto* pci = reinterpret_cast<
- PeerConnectionProxyWithInternal<PeerConnectionInterface>*>(pc());
- return reinterpret_cast<PeerConnection*>(pci->internal());
- }
-};
-
-class PeerConnectionSignalingTest : public ::testing::Test {
- protected:
- typedef std::unique_ptr<PeerConnectionWrapperForSignalingTest> WrapperPtr;
-
- PeerConnectionSignalingTest()
- : vss_(new rtc::VirtualSocketServer()), main_(vss_.get()) {
-#ifdef WEBRTC_ANDROID
- InitializeAndroidObjects();
-#endif
- pc_factory_ = CreatePeerConnectionFactory(
- rtc::Thread::Current(), rtc::Thread::Current(), rtc::Thread::Current(),
- FakeAudioCaptureModule::Create(), CreateBuiltinAudioEncoderFactory(),
- CreateBuiltinAudioDecoderFactory(), nullptr, nullptr);
- }
-
- WrapperPtr CreatePeerConnection() {
- return CreatePeerConnection(RTCConfiguration());
- }
-
- WrapperPtr CreatePeerConnection(const RTCConfiguration& config) {
- auto observer = rtc::MakeUnique<MockPeerConnectionObserver>();
- auto pc = pc_factory_->CreatePeerConnection(config, nullptr, nullptr,
- observer.get());
- if (!pc) {
- return nullptr;
- }
-
- return rtc::MakeUnique<PeerConnectionWrapperForSignalingTest>(
- pc_factory_, pc, std::move(observer));
- }
-
- // Accepts the same arguments as CreatePeerConnection and adds default audio
- // and video tracks.
- template <typename... Args>
- WrapperPtr CreatePeerConnectionWithAudioVideo(Args&&... args) {
- auto wrapper = CreatePeerConnection(std::forward<Args>(args)...);
- if (!wrapper) {
- return nullptr;
- }
- wrapper->AddAudioTrack("a");
- wrapper->AddVideoTrack("v");
- return wrapper;
- }
-
- std::unique_ptr<rtc::VirtualSocketServer> vss_;
- rtc::AutoSocketServerThread main_;
- rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory_;
-};
-
-TEST_F(PeerConnectionSignalingTest, SetLocalOfferTwiceWorks) {
- auto caller = CreatePeerConnection();
-
- EXPECT_TRUE(caller->SetLocalDescription(caller->CreateOffer()));
- EXPECT_TRUE(caller->SetLocalDescription(caller->CreateOffer()));
-}
-
-TEST_F(PeerConnectionSignalingTest, SetRemoteOfferTwiceWorks) {
- auto caller = CreatePeerConnection();
- auto callee = CreatePeerConnection();
-
- EXPECT_TRUE(callee->SetRemoteDescription(caller->CreateOffer()));
- EXPECT_TRUE(callee->SetRemoteDescription(caller->CreateOffer()));
-}
-
-TEST_F(PeerConnectionSignalingTest, FailToSetNullLocalDescription) {
- auto caller = CreatePeerConnection();
- std::string error;
- ASSERT_FALSE(caller->SetLocalDescription(nullptr, &error));
- EXPECT_EQ("SessionDescription is NULL.", error);
-}
-
-TEST_F(PeerConnectionSignalingTest, FailToSetNullRemoteDescription) {
- auto caller = CreatePeerConnection();
- std::string error;
- ASSERT_FALSE(caller->SetRemoteDescription(nullptr, &error));
- EXPECT_EQ("SessionDescription is NULL.", error);
-}
-
-// The following parameterized test verifies that calls to various signaling
-// methods on PeerConnection will succeed/fail depending on what is the
-// PeerConnection's signaling state. Note that the test tries many different
-// forms of SignalingState::kClosed by arriving at a valid state then calling
-// |Close()|. This is intended to catch cases where the PeerConnection signaling
-// method ignores the closed flag but may work/not work because of the single
-// state the PeerConnection was created in before it was closed.
-
-class PeerConnectionSignalingStateTest
- : public PeerConnectionSignalingTest,
- public ::testing::WithParamInterface<std::tuple<SignalingState, bool>> {
- protected:
- RTCConfiguration GetConfig() {
- RTCConfiguration config;
- config.certificates.push_back(
- FakeRTCCertificateGenerator::GenerateCertificate());
- return config;
- }
-
- WrapperPtr CreatePeerConnectionInState(SignalingState state) {
- return CreatePeerConnectionInState(std::make_tuple(state, false));
- }
-
- WrapperPtr CreatePeerConnectionInState(
- std::tuple<SignalingState, bool> state_tuple) {
- SignalingState state = std::get<0>(state_tuple);
- bool closed = std::get<1>(state_tuple);
-
- auto wrapper = CreatePeerConnectionWithAudioVideo(GetConfig());
- switch (state) {
- case SignalingState::kStable: {
- break;
- }
- case SignalingState::kHaveLocalOffer: {
- wrapper->SetLocalDescription(wrapper->CreateOffer());
- break;
- }
- case SignalingState::kHaveLocalPrAnswer: {
- auto caller = CreatePeerConnectionWithAudioVideo(GetConfig());
- wrapper->SetRemoteDescription(caller->CreateOffer());
- auto answer = wrapper->CreateAnswer();
- wrapper->SetLocalDescription(CloneSessionDescriptionAsType(
- answer.get(), SessionDescriptionInterface::kPrAnswer));
- break;
- }
- case SignalingState::kHaveRemoteOffer: {
- auto caller = CreatePeerConnectionWithAudioVideo(GetConfig());
- wrapper->SetRemoteDescription(caller->CreateOffer());
- break;
- }
- case SignalingState::kHaveRemotePrAnswer: {
- auto callee = CreatePeerConnectionWithAudioVideo(GetConfig());
- callee->SetRemoteDescription(wrapper->CreateOfferAndSetAsLocal());
- auto answer = callee->CreateAnswer();
- wrapper->SetRemoteDescription(CloneSessionDescriptionAsType(
- answer.get(), SessionDescriptionInterface::kPrAnswer));
- break;
- }
- case SignalingState::kClosed: {
- RTC_NOTREACHED() << "Set the second member of the tuple to true to "
- "achieve a closed state from an existing, valid "
- "state.";
- }
- }
-
- RTC_DCHECK_EQ(state, wrapper->pc()->signaling_state());
-
- if (closed) {
- wrapper->pc()->Close();
- RTC_DCHECK_EQ(SignalingState::kClosed, wrapper->signaling_state());
- }
-
- return wrapper;
- }
-};
-
-::testing::AssertionResult AssertStartsWith(const char* str_expr,
- const char* prefix_expr,
- const std::string& str,
- const std::string& prefix) {
- if (rtc::starts_with(str.c_str(), prefix.c_str())) {
- return ::testing::AssertionSuccess();
- } else {
- return ::testing::AssertionFailure()
- << str_expr << "\nwhich is\n\"" << str << "\"\ndoes not start with\n"
- << prefix_expr << "\nwhich is\n\"" << prefix << "\"";
- }
-}
-
-TEST_P(PeerConnectionSignalingStateTest, CreateOffer) {
- auto wrapper = CreatePeerConnectionInState(GetParam());
- if (wrapper->signaling_state() != SignalingState::kClosed) {
- EXPECT_TRUE(wrapper->CreateOffer());
- } else {
- std::string error;
- ASSERT_FALSE(wrapper->CreateOffer(RTCOfferAnswerOptions(), &error));
- EXPECT_PRED_FORMAT2(AssertStartsWith, error,
- "CreateOffer called when PeerConnection is closed.");
- }
-}
-
-TEST_P(PeerConnectionSignalingStateTest, CreateAnswer) {
- auto wrapper = CreatePeerConnectionInState(GetParam());
- if (wrapper->signaling_state() == SignalingState::kHaveLocalPrAnswer ||
- wrapper->signaling_state() == SignalingState::kHaveRemoteOffer) {
- EXPECT_TRUE(wrapper->CreateAnswer());
- } else {
- std::string error;
- ASSERT_FALSE(wrapper->CreateAnswer(RTCOfferAnswerOptions(), &error));
- if (wrapper->signaling_state() == SignalingState::kClosed) {
- EXPECT_PRED_FORMAT2(AssertStartsWith, error,
- "CreateAnswer called when PeerConnection is closed.");
- } else {
- EXPECT_PRED_FORMAT2(AssertStartsWith, error,
- "CreateAnswer called without remote offer.");
- }
- }
-}
-
-TEST_P(PeerConnectionSignalingStateTest, SetLocalOffer) {
- auto wrapper = CreatePeerConnectionInState(GetParam());
- if (wrapper->signaling_state() == SignalingState::kStable ||
- wrapper->signaling_state() == SignalingState::kHaveLocalOffer) {
- // Need to call CreateOffer on the PeerConnection under test, otherwise when
- // setting the local offer it will want to verify the DTLS fingerprint
- // against the locally generated certificate, but without a call to
- // CreateOffer the certificate will never be generated.
- EXPECT_TRUE(wrapper->SetLocalDescription(wrapper->CreateOffer()));
- } else {
- auto wrapper_for_offer =
- CreatePeerConnectionInState(SignalingState::kHaveLocalOffer);
- auto offer =
- CloneSessionDescription(wrapper_for_offer->pc()->local_description());
-
- std::string error;
- ASSERT_FALSE(wrapper->SetLocalDescription(std::move(offer), &error));
- EXPECT_PRED_FORMAT2(
- AssertStartsWith, error,
- "Failed to set local offer SDP: Called in wrong state:");
- }
-}
-
-TEST_P(PeerConnectionSignalingStateTest, SetLocalPrAnswer) {
- auto wrapper_for_pranswer =
- CreatePeerConnectionInState(SignalingState::kHaveLocalPrAnswer);
- auto pranswer =
- CloneSessionDescription(wrapper_for_pranswer->pc()->local_description());
-
- auto wrapper = CreatePeerConnectionInState(GetParam());
- if (wrapper->signaling_state() == SignalingState::kHaveLocalPrAnswer ||
- wrapper->signaling_state() == SignalingState::kHaveRemoteOffer) {
- EXPECT_TRUE(wrapper->SetLocalDescription(std::move(pranswer)));
- } else {
- std::string error;
- ASSERT_FALSE(wrapper->SetLocalDescription(std::move(pranswer), &error));
- EXPECT_PRED_FORMAT2(
- AssertStartsWith, error,
- "Failed to set local pranswer SDP: Called in wrong state:");
- }
-}
-
-TEST_P(PeerConnectionSignalingStateTest, SetLocalAnswer) {
- auto wrapper_for_answer =
- CreatePeerConnectionInState(SignalingState::kHaveRemoteOffer);
- auto answer = wrapper_for_answer->CreateAnswer();
-
- auto wrapper = CreatePeerConnectionInState(GetParam());
- if (wrapper->signaling_state() == SignalingState::kHaveLocalPrAnswer ||
- wrapper->signaling_state() == SignalingState::kHaveRemoteOffer) {
- EXPECT_TRUE(wrapper->SetLocalDescription(std::move(answer)));
- } else {
- std::string error;
- ASSERT_FALSE(wrapper->SetLocalDescription(std::move(answer), &error));
- EXPECT_PRED_FORMAT2(
- AssertStartsWith, error,
- "Failed to set local answer SDP: Called in wrong state:");
- }
-}
-
-TEST_P(PeerConnectionSignalingStateTest, SetRemoteOffer) {
- auto wrapper_for_offer =
- CreatePeerConnectionInState(SignalingState::kHaveRemoteOffer);
- auto offer =
- CloneSessionDescription(wrapper_for_offer->pc()->remote_description());
-
- auto wrapper = CreatePeerConnectionInState(GetParam());
- if (wrapper->signaling_state() == SignalingState::kStable ||
- wrapper->signaling_state() == SignalingState::kHaveRemoteOffer) {
- EXPECT_TRUE(wrapper->SetRemoteDescription(std::move(offer)));
- } else {
- std::string error;
- ASSERT_FALSE(wrapper->SetRemoteDescription(std::move(offer), &error));
- EXPECT_PRED_FORMAT2(
- AssertStartsWith, error,
- "Failed to set remote offer SDP: Called in wrong state:");
- }
-}
-
-TEST_P(PeerConnectionSignalingStateTest, SetRemotePrAnswer) {
- auto wrapper_for_pranswer =
- CreatePeerConnectionInState(SignalingState::kHaveRemotePrAnswer);
- auto pranswer =
- CloneSessionDescription(wrapper_for_pranswer->pc()->remote_description());
-
- auto wrapper = CreatePeerConnectionInState(GetParam());
- if (wrapper->signaling_state() == SignalingState::kHaveLocalOffer ||
- wrapper->signaling_state() == SignalingState::kHaveRemotePrAnswer) {
- EXPECT_TRUE(wrapper->SetRemoteDescription(std::move(pranswer)));
- } else {
- std::string error;
- ASSERT_FALSE(wrapper->SetRemoteDescription(std::move(pranswer), &error));
- EXPECT_PRED_FORMAT2(
- AssertStartsWith, error,
- "Failed to set remote pranswer SDP: Called in wrong state:");
- }
-}
-
-TEST_P(PeerConnectionSignalingStateTest, SetRemoteAnswer) {
- auto wrapper_for_answer =
- CreatePeerConnectionInState(SignalingState::kHaveRemoteOffer);
- auto answer = wrapper_for_answer->CreateAnswer();
-
- auto wrapper = CreatePeerConnectionInState(GetParam());
- if (wrapper->signaling_state() == SignalingState::kHaveLocalOffer ||
- wrapper->signaling_state() == SignalingState::kHaveRemotePrAnswer) {
- EXPECT_TRUE(wrapper->SetRemoteDescription(std::move(answer)));
- } else {
- std::string error;
- ASSERT_FALSE(wrapper->SetRemoteDescription(std::move(answer), &error));
- EXPECT_PRED_FORMAT2(
- AssertStartsWith, error,
- "Failed to set remote answer SDP: Called in wrong state:");
- }
-}
-
-INSTANTIATE_TEST_CASE_P(PeerConnectionSignalingTest,
- PeerConnectionSignalingStateTest,
- Combine(Values(SignalingState::kStable,
- SignalingState::kHaveLocalOffer,
- SignalingState::kHaveLocalPrAnswer,
- SignalingState::kHaveRemoteOffer,
- SignalingState::kHaveRemotePrAnswer),
- Bool()));
-
-TEST_F(PeerConnectionSignalingTest,
- CreateAnswerSucceedsIfStableAndRemoteDescriptionIsOffer) {
- auto caller = CreatePeerConnection();
- auto callee = CreatePeerConnection();
-
- ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
- ASSERT_TRUE(
- caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
-
- ASSERT_EQ(SignalingState::kStable, callee->signaling_state());
- EXPECT_TRUE(callee->CreateAnswer());
-}
-
-TEST_F(PeerConnectionSignalingTest,
- CreateAnswerFailsIfStableButRemoteDescriptionIsAnswer) {
- auto caller = CreatePeerConnection();
- auto callee = CreatePeerConnection();
-
- ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
- ASSERT_TRUE(
- caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
-
- ASSERT_EQ(SignalingState::kStable, caller->signaling_state());
- std::string error;
- ASSERT_FALSE(caller->CreateAnswer(RTCOfferAnswerOptions(), &error));
- EXPECT_EQ("CreateAnswer called without remote offer.", error);
-}
-
-// According to https://tools.ietf.org/html/rfc3264#section-8, the session id
-// stays the same but the version must be incremented if a later, different
-// session description is generated. These two tests verify that is the case for
-// both offers and answers.
-TEST_F(PeerConnectionSignalingTest,
- SessionVersionIncrementedInSubsequentDifferentOffer) {
- auto caller = CreatePeerConnection();
- auto callee = CreatePeerConnection();
-
- auto original_offer = caller->CreateOfferAndSetAsLocal();
- const std::string original_id = original_offer->session_id();
- const std::string original_version = original_offer->session_version();
-
- ASSERT_TRUE(callee->SetRemoteDescription(std::move(original_offer)));
- ASSERT_TRUE(caller->SetRemoteDescription(callee->CreateAnswer()));
-
- // Add track to get a different offer.
- caller->AddAudioTrack("a");
-
- auto later_offer = caller->CreateOffer();
-
- EXPECT_EQ(original_id, later_offer->session_id());
- EXPECT_LT(rtc::FromString<uint64_t>(original_version),
- rtc::FromString<uint64_t>(later_offer->session_version()));
-}
-TEST_F(PeerConnectionSignalingTest,
- SessionVersionIncrementedInSubsequentDifferentAnswer) {
- auto caller = CreatePeerConnection();
- auto callee = CreatePeerConnection();
-
- ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
-
- auto original_answer = callee->CreateAnswerAndSetAsLocal();
- const std::string original_id = original_answer->session_id();
- const std::string original_version = original_answer->session_version();
-
- // Add track to get a different answer.
- callee->AddAudioTrack("a");
-
- auto later_answer = callee->CreateAnswer();
-
- EXPECT_EQ(original_id, later_answer->session_id());
- EXPECT_LT(rtc::FromString<uint64_t>(original_version),
- rtc::FromString<uint64_t>(later_answer->session_version()));
-}
-
-TEST_F(PeerConnectionSignalingTest, InitiatorFlagSetOnCallerAndNotOnCallee) {
- auto caller = CreatePeerConnectionWithAudioVideo();
- auto callee = CreatePeerConnectionWithAudioVideo();
-
- EXPECT_FALSE(caller->initial_offerer());
- EXPECT_FALSE(callee->initial_offerer());
-
- ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
-
- EXPECT_TRUE(caller->initial_offerer());
- EXPECT_FALSE(callee->initial_offerer());
-
- ASSERT_TRUE(
- caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
-
- EXPECT_TRUE(caller->initial_offerer());
- EXPECT_FALSE(callee->initial_offerer());
-}
-
-// Test creating a PeerConnection, request multiple offers, destroy the
-// PeerConnection and make sure we get success/failure callbacks for all of the
-// requests.
-// Background: crbug.com/507307
-TEST_F(PeerConnectionSignalingTest, CreateOffersAndShutdown) {
- auto caller = CreatePeerConnection();
-
- RTCOfferAnswerOptions options;
- options.offer_to_receive_audio =
- RTCOfferAnswerOptions::kOfferToReceiveMediaTrue;
-
- rtc::scoped_refptr<MockCreateSessionDescriptionObserver> observers[100];
- for (auto& observer : observers) {
- observer =
- new rtc::RefCountedObject<MockCreateSessionDescriptionObserver>();
- caller->pc()->CreateOffer(observer, options);
- }
-
- // Destroy the PeerConnection.
- caller.reset(nullptr);
-
- for (auto& observer : observers) {
- // We expect to have received a notification now even if the PeerConnection
- // was terminated. The offer creation may or may not have succeeded, but we
- // must have received a notification.
- EXPECT_TRUE(observer->called());
- }
-}
-
-} // namespace webrtc
diff --git a/pc/peerconnectioninterface_unittest.cc b/pc/peerconnectioninterface_unittest.cc
index a8b4f72..99f3301 100644
--- a/pc/peerconnectioninterface_unittest.cc
+++ b/pc/peerconnectioninterface_unittest.cc
@@ -2517,9 +2517,9 @@
EXPECT_TRUE(pc_->remote_description() != NULL);
std::unique_ptr<SessionDescriptionInterface> offer;
- EXPECT_FALSE(DoCreateOffer(&offer, nullptr));
+ EXPECT_TRUE(DoCreateOffer(&offer, nullptr));
std::unique_ptr<SessionDescriptionInterface> answer;
- EXPECT_FALSE(DoCreateAnswer(&answer, nullptr));
+ EXPECT_TRUE(DoCreateAnswer(&answer, nullptr));
std::string sdp;
ASSERT_TRUE(pc_->remote_description()->ToString(&sdp));
@@ -3558,6 +3558,32 @@
EXPECT_NE(nullptr, GetFirstVideoContent(offer->description()));
}
+// Test that if |voice_activity_detection| is false, no CN codec is added to the
+// offer.
+TEST_F(PeerConnectionInterfaceTest, CreateOfferWithVADOptions) {
+ RTCOfferAnswerOptions rtc_options;
+ rtc_options.offer_to_receive_audio = 1;
+ rtc_options.offer_to_receive_video = 0;
+
+ std::unique_ptr<SessionDescriptionInterface> offer;
+ CreatePeerConnection();
+ offer = CreateOfferWithOptions(rtc_options);
+ ASSERT_TRUE(offer);
+ const cricket::ContentInfo* audio_content =
+ offer->description()->GetContentByName(cricket::CN_AUDIO);
+ ASSERT_TRUE(audio_content);
+ // |voice_activity_detection| is true by default.
+ EXPECT_TRUE(HasCNCodecs(audio_content));
+
+ rtc_options.voice_activity_detection = false;
+ CreatePeerConnection();
+ offer = CreateOfferWithOptions(rtc_options);
+ ASSERT_TRUE(offer);
+ audio_content = offer->description()->GetContentByName(cricket::CN_AUDIO);
+ ASSERT_TRUE(audio_content);
+ EXPECT_FALSE(HasCNCodecs(audio_content));
+}
+
// Test that no media content will be added to the offer if using default
// RTCOfferAnswerOptions.
TEST_F(PeerConnectionInterfaceTest, CreateOfferWithDefaultOfferAnswerOptions) {
@@ -3638,6 +3664,42 @@
EXPECT_FALSE(offer->description()->HasGroup(cricket::GROUP_TYPE_BUNDLE));
}
+// If SetMandatoryReceiveAudio(false) and SetMandatoryReceiveVideo(false) are
+// called for the answer constraints, but an audio and a video section were
+// offered, there will still be an audio and a video section in the answer.
+TEST_F(PeerConnectionInterfaceTest,
+ RejectAudioAndVideoInAnswerWithConstraints) {
+ // Offer both audio and video.
+ RTCOfferAnswerOptions rtc_offer_options;
+ rtc_offer_options.offer_to_receive_audio = 1;
+ rtc_offer_options.offer_to_receive_video = 1;
+
+ CreatePeerConnection();
+ std::unique_ptr<SessionDescriptionInterface> offer;
+ CreateOfferWithOptionsAsRemoteDescription(&offer, rtc_offer_options);
+ EXPECT_NE(nullptr, GetFirstAudioContent(offer->description()));
+ EXPECT_NE(nullptr, GetFirstVideoContent(offer->description()));
+
+ // Since an offer has been created with both audio and video,
+ // Answers will contain the media types that exist in the offer regardless of
+ // the value of |answer_options.has_audio| and |answer_options.has_video|.
+ FakeConstraints answer_c;
+ // Reject both audio and video.
+ answer_c.SetMandatoryReceiveAudio(false);
+ answer_c.SetMandatoryReceiveVideo(false);
+
+ std::unique_ptr<SessionDescriptionInterface> answer;
+ ASSERT_TRUE(DoCreateAnswer(&answer, &answer_c));
+ const cricket::ContentInfo* audio_content =
+ GetFirstAudioContent(answer->description());
+ const cricket::ContentInfo* video_content =
+ GetFirstVideoContent(answer->description());
+ ASSERT_NE(nullptr, audio_content);
+ ASSERT_NE(nullptr, video_content);
+ EXPECT_TRUE(audio_content->rejected);
+ EXPECT_TRUE(video_content->rejected);
+}
+
// This test ensures OnRenegotiationNeeded is called when we add track with
// MediaStream -> AddTrack in the same way it is called when we add track with
// PeerConnection -> AddTrack.
@@ -3672,6 +3734,52 @@
observer_.renegotiation_needed_ = false;
}
+// Tests that creating answer would fail gracefully without being crashed if the
+// remote description is unset.
+TEST_F(PeerConnectionInterfaceTest, CreateAnswerWithoutRemoteDescription) {
+ CreatePeerConnection();
+ // Creating answer fails because the remote description is unset.
+ std::unique_ptr<SessionDescriptionInterface> answer;
+ EXPECT_FALSE(DoCreateAnswer(&answer, nullptr));
+
+ // Createing answer succeeds when the remote description is set.
+ CreateOfferAsRemoteDescription();
+ EXPECT_TRUE(DoCreateAnswer(&answer, nullptr));
+}
+
+// Test that an error is returned if a description is applied that doesn't
+// respect the order of existing media sections.
+TEST_F(PeerConnectionInterfaceTest,
+ MediaSectionOrderEnforcedForSubsequentOffers) {
+ CreatePeerConnection();
+ FakeConstraints constraints;
+ constraints.SetMandatoryReceiveAudio(true);
+ constraints.SetMandatoryReceiveVideo(true);
+ std::unique_ptr<SessionDescriptionInterface> offer;
+ ASSERT_TRUE(DoCreateOffer(&offer, &constraints));
+ EXPECT_TRUE(DoSetRemoteDescription(std::move(offer)));
+
+ std::unique_ptr<SessionDescriptionInterface> answer;
+ ASSERT_TRUE(DoCreateAnswer(&answer, nullptr));
+ EXPECT_TRUE(DoSetLocalDescription(std::move(answer)));
+
+ // A remote offer with different m=line order should be rejected.
+ ASSERT_TRUE(DoCreateOffer(&offer, &constraints));
+ std::reverse(offer->description()->contents().begin(),
+ offer->description()->contents().end());
+ std::reverse(offer->description()->transport_infos().begin(),
+ offer->description()->transport_infos().end());
+ EXPECT_FALSE(DoSetRemoteDescription(std::move(offer)));
+
+ // A subsequent local offer with different m=line order should be rejected.
+ ASSERT_TRUE(DoCreateOffer(&offer, &constraints));
+ std::reverse(offer->description()->contents().begin(),
+ offer->description()->contents().end());
+ std::reverse(offer->description()->transport_infos().begin(),
+ offer->description()->transport_infos().end());
+ EXPECT_FALSE(DoSetLocalDescription(std::move(offer)));
+}
+
class PeerConnectionMediaConfigTest : public testing::Test {
protected:
void SetUp() override {
diff --git a/pc/peerconnectionwrapper.cc b/pc/peerconnectionwrapper.cc
index 9be9309..dd11460 100644
--- a/pc/peerconnectionwrapper.cc
+++ b/pc/peerconnectionwrapper.cc
@@ -30,7 +30,7 @@
rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory,
rtc::scoped_refptr<PeerConnectionInterface> pc,
std::unique_ptr<MockPeerConnectionObserver> observer)
- : pc_factory_(pc_factory), observer_(std::move(observer)), pc_(pc) {
+ : pc_factory_(pc_factory), pc_(pc), observer_(std::move(observer)) {
RTC_DCHECK(pc_factory_);
RTC_DCHECK(pc_);
RTC_DCHECK(observer_);
@@ -57,25 +57,15 @@
}
std::unique_ptr<SessionDescriptionInterface> PeerConnectionWrapper::CreateOffer(
- const PeerConnectionInterface::RTCOfferAnswerOptions& options,
- std::string* error_out) {
- return CreateSdp(
- [this, options](CreateSessionDescriptionObserver* observer) {
- pc()->CreateOffer(observer, options);
- },
- error_out);
+ const PeerConnectionInterface::RTCOfferAnswerOptions& options) {
+ return CreateSdp([this, options](CreateSessionDescriptionObserver* observer) {
+ pc()->CreateOffer(observer, options);
+ });
}
std::unique_ptr<SessionDescriptionInterface>
PeerConnectionWrapper::CreateOfferAndSetAsLocal() {
- return CreateOfferAndSetAsLocal(
- PeerConnectionInterface::RTCOfferAnswerOptions());
-}
-
-std::unique_ptr<SessionDescriptionInterface>
-PeerConnectionWrapper::CreateOfferAndSetAsLocal(
- const PeerConnectionInterface::RTCOfferAnswerOptions& options) {
- auto offer = CreateOffer(options);
+ auto offer = CreateOffer();
if (!offer) {
return nullptr;
}
@@ -90,25 +80,15 @@
std::unique_ptr<SessionDescriptionInterface>
PeerConnectionWrapper::CreateAnswer(
- const PeerConnectionInterface::RTCOfferAnswerOptions& options,
- std::string* error_out) {
- return CreateSdp(
- [this, options](CreateSessionDescriptionObserver* observer) {
- pc()->CreateAnswer(observer, options);
- },
- error_out);
+ const PeerConnectionInterface::RTCOfferAnswerOptions& options) {
+ return CreateSdp([this, options](CreateSessionDescriptionObserver* observer) {
+ pc()->CreateAnswer(observer, options);
+ });
}
std::unique_ptr<SessionDescriptionInterface>
PeerConnectionWrapper::CreateAnswerAndSetAsLocal() {
- return CreateAnswerAndSetAsLocal(
- PeerConnectionInterface::RTCOfferAnswerOptions());
-}
-
-std::unique_ptr<SessionDescriptionInterface>
-PeerConnectionWrapper::CreateAnswerAndSetAsLocal(
- const PeerConnectionInterface::RTCOfferAnswerOptions& options) {
- auto answer = CreateAnswer(options);
+ auto answer = CreateAnswer();
if (!answer) {
return nullptr;
}
@@ -117,72 +97,73 @@
}
std::unique_ptr<SessionDescriptionInterface> PeerConnectionWrapper::CreateSdp(
- std::function<void(CreateSessionDescriptionObserver*)> fn,
- std::string* error_out) {
+ std::function<void(CreateSessionDescriptionObserver*)> fn) {
rtc::scoped_refptr<MockCreateSessionDescriptionObserver> observer(
new rtc::RefCountedObject<MockCreateSessionDescriptionObserver>());
fn(observer);
EXPECT_EQ_WAIT(true, observer->called(), kWaitTimeout);
- if (error_out && !observer->result()) {
- *error_out = observer->error();
- }
return observer->MoveDescription();
}
bool PeerConnectionWrapper::SetLocalDescription(
- std::unique_ptr<SessionDescriptionInterface> desc,
- std::string* error_out) {
- return SetSdp(
- [this, &desc](SetSessionDescriptionObserver* observer) {
- pc()->SetLocalDescription(observer, desc.release());
- },
- error_out);
+ std::unique_ptr<SessionDescriptionInterface> desc) {
+ return SetSdp([this, &desc](SetSessionDescriptionObserver* observer) {
+ pc()->SetLocalDescription(observer, desc.release());
+ });
}
bool PeerConnectionWrapper::SetRemoteDescription(
- std::unique_ptr<SessionDescriptionInterface> desc,
- std::string* error_out) {
- return SetSdp(
- [this, &desc](SetSessionDescriptionObserver* observer) {
- pc()->SetRemoteDescription(observer, desc.release());
- },
- error_out);
+ std::unique_ptr<SessionDescriptionInterface> desc) {
+ return SetSdp([this, &desc](SetSessionDescriptionObserver* observer) {
+ pc()->SetRemoteDescription(observer, desc.release());
+ });
}
bool PeerConnectionWrapper::SetSdp(
- std::function<void(SetSessionDescriptionObserver*)> fn,
- std::string* error_out) {
+ std::function<void(SetSessionDescriptionObserver*)> fn) {
rtc::scoped_refptr<MockSetSessionDescriptionObserver> observer(
new rtc::RefCountedObject<MockSetSessionDescriptionObserver>());
fn(observer);
- EXPECT_EQ_WAIT(true, observer->called(), kWaitTimeout);
- if (error_out && !observer->result()) {
- *error_out = observer->error();
+ if (pc()->signaling_state() != PeerConnectionInterface::kClosed) {
+ EXPECT_EQ_WAIT(true, observer->called(), kWaitTimeout);
}
return observer->result();
}
-rtc::scoped_refptr<RtpSenderInterface> PeerConnectionWrapper::AddAudioTrack(
- const std::string& track_label,
- std::vector<MediaStreamInterface*> streams) {
- auto media_stream_track =
- pc_factory()->CreateAudioTrack(track_label, nullptr);
- return pc()->AddTrack(media_stream_track, streams);
+void PeerConnectionWrapper::AddAudioStream(const std::string& stream_label,
+ const std::string& track_label) {
+ auto stream = pc_factory()->CreateLocalMediaStream(stream_label);
+ auto audio_track = pc_factory()->CreateAudioTrack(track_label, nullptr);
+ EXPECT_TRUE(pc()->AddTrack(audio_track, {stream}));
+ EXPECT_TRUE_WAIT(observer()->renegotiation_needed_, kWaitTimeout);
+ observer()->renegotiation_needed_ = false;
}
-rtc::scoped_refptr<RtpSenderInterface> PeerConnectionWrapper::AddVideoTrack(
- const std::string& track_label,
- std::vector<MediaStreamInterface*> streams) {
+void PeerConnectionWrapper::AddVideoStream(const std::string& stream_label,
+ const std::string& track_label) {
+ auto stream = pc_factory()->CreateLocalMediaStream(stream_label);
auto video_source = pc_factory()->CreateVideoSource(
rtc::MakeUnique<cricket::FakeVideoCapturer>());
- auto media_stream_track =
- pc_factory()->CreateVideoTrack(track_label, video_source);
- return pc()->AddTrack(media_stream_track, streams);
+ auto video_track = pc_factory()->CreateVideoTrack(track_label, video_source);
+ EXPECT_TRUE(pc()->AddTrack(video_track, {stream}));
+ EXPECT_TRUE_WAIT(observer()->renegotiation_needed_, kWaitTimeout);
+ observer()->renegotiation_needed_ = false;
}
-PeerConnectionInterface::SignalingState
-PeerConnectionWrapper::signaling_state() {
- return pc()->signaling_state();
+void PeerConnectionWrapper::AddAudioVideoStream(
+ const std::string& stream_label,
+ const std::string& audio_track_label,
+ const std::string& video_track_label) {
+ auto stream = pc_factory()->CreateLocalMediaStream(stream_label);
+ auto audio_track = pc_factory()->CreateAudioTrack(audio_track_label, nullptr);
+ EXPECT_TRUE(pc()->AddTrack(audio_track, {stream}));
+ auto video_source = pc_factory()->CreateVideoSource(
+ rtc::MakeUnique<cricket::FakeVideoCapturer>());
+ auto video_track =
+ pc_factory()->CreateVideoTrack(video_track_label, video_source);
+ EXPECT_TRUE(pc()->AddTrack(video_track, {stream}));
+ EXPECT_TRUE_WAIT(observer()->renegotiation_needed_, kWaitTimeout);
+ observer()->renegotiation_needed_ = false;
}
bool PeerConnectionWrapper::IsIceGatheringDone() {
diff --git a/pc/peerconnectionwrapper.h b/pc/peerconnectionwrapper.h
index f74fcdb..783ae38 100644
--- a/pc/peerconnectionwrapper.h
+++ b/pc/peerconnectionwrapper.h
@@ -54,69 +54,54 @@
// resulting SessionDescription once it is available. If the method call
// failed, null is returned.
std::unique_ptr<SessionDescriptionInterface> CreateOffer(
- const PeerConnectionInterface::RTCOfferAnswerOptions& options,
- std::string* error_out = nullptr);
+ const PeerConnectionInterface::RTCOfferAnswerOptions& options);
// Calls CreateOffer with default options.
std::unique_ptr<SessionDescriptionInterface> CreateOffer();
// Calls CreateOffer and sets a copy of the offer as the local description.
- std::unique_ptr<SessionDescriptionInterface> CreateOfferAndSetAsLocal(
- const PeerConnectionInterface::RTCOfferAnswerOptions& options);
- // Calls CreateOfferAndSetAsLocal with default options.
std::unique_ptr<SessionDescriptionInterface> CreateOfferAndSetAsLocal();
// Calls the underlying PeerConnection's CreateAnswer method and returns the
// resulting SessionDescription once it is available. If the method call
// failed, null is returned.
std::unique_ptr<SessionDescriptionInterface> CreateAnswer(
- const PeerConnectionInterface::RTCOfferAnswerOptions& options,
- std::string* error_out = nullptr);
+ const PeerConnectionInterface::RTCOfferAnswerOptions& options);
// Calls CreateAnswer with the default options.
std::unique_ptr<SessionDescriptionInterface> CreateAnswer();
// Calls CreateAnswer and sets a copy of the offer as the local description.
- std::unique_ptr<SessionDescriptionInterface> CreateAnswerAndSetAsLocal(
- const PeerConnectionInterface::RTCOfferAnswerOptions& options);
- // Calls CreateAnswerAndSetAsLocal with default options.
std::unique_ptr<SessionDescriptionInterface> CreateAnswerAndSetAsLocal();
// Calls the underlying PeerConnection's SetLocalDescription method with the
// given session description and waits for the success/failure response.
// Returns true if the description was successfully set.
- bool SetLocalDescription(std::unique_ptr<SessionDescriptionInterface> desc,
- std::string* error_out = nullptr);
+ bool SetLocalDescription(std::unique_ptr<SessionDescriptionInterface> desc);
// Calls the underlying PeerConnection's SetRemoteDescription method with the
// given session description and waits for the success/failure response.
// Returns true if the description was successfully set.
- bool SetRemoteDescription(std::unique_ptr<SessionDescriptionInterface> desc,
- std::string* error_out = nullptr);
+ bool SetRemoteDescription(std::unique_ptr<SessionDescriptionInterface> desc);
- // Calls the underlying PeerConnection's AddTrack method with an audio media
- // stream track not bound to any source.
- rtc::scoped_refptr<RtpSenderInterface> AddAudioTrack(
- const std::string& track_label,
- std::vector<MediaStreamInterface*> streams = {});
-
- // Calls the underlying PeerConnection's AddTrack method with a video media
- // stream track fed by a fake video capturer.
- rtc::scoped_refptr<RtpSenderInterface> AddVideoTrack(
- const std::string& track_label,
- std::vector<MediaStreamInterface*> streams = {});
-
- // Returns the signaling state of the underlying PeerConnection.
- PeerConnectionInterface::SignalingState signaling_state();
+ // Adds a new stream with one audio track to the underlying PeerConnection.
+ void AddAudioStream(const std::string& stream_label,
+ const std::string& track_label);
+ // Adds a new stream with one video track to the underlying PeerConnection.
+ void AddVideoStream(const std::string& stream_label,
+ const std::string& track_label);
+ // Adds a new stream with one audio and one video track to the underlying
+ // PeerConnection.
+ void AddAudioVideoStream(const std::string& stream_label,
+ const std::string& audio_track_label,
+ const std::string& video_track_label);
// Returns true if ICE has finished gathering candidates.
bool IsIceGatheringDone();
private:
std::unique_ptr<SessionDescriptionInterface> CreateSdp(
- std::function<void(CreateSessionDescriptionObserver*)> fn,
- std::string* error_out);
- bool SetSdp(std::function<void(SetSessionDescriptionObserver*)> fn,
- std::string* error_out);
+ std::function<void(CreateSessionDescriptionObserver*)> fn);
+ bool SetSdp(std::function<void(SetSessionDescriptionObserver*)> fn);
rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory_;
- std::unique_ptr<MockPeerConnectionObserver> observer_;
rtc::scoped_refptr<PeerConnectionInterface> pc_;
+ std::unique_ptr<MockPeerConnectionObserver> observer_;
};
} // namespace webrtc
diff --git a/pc/sdputils.cc b/pc/sdputils.cc
index 8932bea..9339fdb 100644
--- a/pc/sdputils.cc
+++ b/pc/sdputils.cc
@@ -20,14 +20,7 @@
std::unique_ptr<SessionDescriptionInterface> CloneSessionDescription(
const SessionDescriptionInterface* sdesc) {
RTC_DCHECK(sdesc);
- return CloneSessionDescriptionAsType(sdesc, sdesc->type());
-}
-
-std::unique_ptr<SessionDescriptionInterface> CloneSessionDescriptionAsType(
- const SessionDescriptionInterface* sdesc,
- const std::string& type) {
- RTC_DCHECK(sdesc);
- auto clone = rtc::MakeUnique<JsepSessionDescription>(type);
+ auto clone = rtc::MakeUnique<JsepSessionDescription>(sdesc->type());
clone->Initialize(sdesc->description()->Copy(), sdesc->session_id(),
sdesc->session_version());
// As of writing, our version of GCC does not allow returning a unique_ptr of
diff --git a/pc/sdputils.h b/pc/sdputils.h
index 3a53a41..7d67fd8 100644
--- a/pc/sdputils.h
+++ b/pc/sdputils.h
@@ -23,11 +23,6 @@
std::unique_ptr<SessionDescriptionInterface> CloneSessionDescription(
const SessionDescriptionInterface* sdesc);
-// Returns a copy of the given session description with the type changed.
-std::unique_ptr<SessionDescriptionInterface> CloneSessionDescriptionAsType(
- const SessionDescriptionInterface* sdesc,
- const std::string& type);
-
// Function that takes a single session description content with its
// corresponding transport and produces a boolean.
typedef std::function<bool(const cricket::ContentInfo*,
diff --git a/pc/test/mockpeerconnectionobservers.h b/pc/test/mockpeerconnectionobservers.h
index 82098ca..9b077f5 100644
--- a/pc/test/mockpeerconnectionobservers.h
+++ b/pc/test/mockpeerconnectionobservers.h
@@ -177,27 +177,26 @@
public:
MockCreateSessionDescriptionObserver()
: called_(false),
- error_("MockCreateSessionDescriptionObserver not called") {}
+ result_(false) {}
virtual ~MockCreateSessionDescriptionObserver() {}
virtual void OnSuccess(SessionDescriptionInterface* desc) {
called_ = true;
- error_ = "";
+ result_ = true;
desc_.reset(desc);
}
virtual void OnFailure(const std::string& error) {
called_ = true;
- error_ = error;
+ result_ = false;
}
bool called() const { return called_; }
- bool result() const { return error_.empty(); }
- const std::string& error() const { return error_; }
+ bool result() const { return result_; }
std::unique_ptr<SessionDescriptionInterface> MoveDescription() {
return std::move(desc_);
}
private:
bool called_;
- std::string error_;
+ bool result_;
std::unique_ptr<SessionDescriptionInterface> desc_;
};
@@ -206,23 +205,22 @@
public:
MockSetSessionDescriptionObserver()
: called_(false),
- error_("MockSetSessionDescriptionObserver not called") {}
+ result_(false) {}
virtual ~MockSetSessionDescriptionObserver() {}
virtual void OnSuccess() {
called_ = true;
- error_ = "";
+ result_ = true;
}
virtual void OnFailure(const std::string& error) {
called_ = true;
- error_ = error;
+ result_ = false;
}
bool called() const { return called_; }
- bool result() const { return error_.empty(); }
- const std::string& error() const { return error_; }
+ bool result() const { return result_; }
private:
bool called_;
- std::string error_;
+ bool result_;
};
class MockDataChannelObserver : public webrtc::DataChannelObserver {
diff --git a/pc/webrtcsession.cc b/pc/webrtcsession.cc
index 2e0ae50..f556204 100644
--- a/pc/webrtcsession.cc
+++ b/pc/webrtcsession.cc
@@ -337,7 +337,7 @@
if (!type.empty()) {
desc << " " << type;
}
- desc << " SDP: " << reason;
+ desc << " sdp: " << reason;
if (err_desc) {
*err_desc = desc.str();
@@ -707,13 +707,15 @@
webrtc_session_desc_factory_->CreateAnswer(observer, session_options);
}
-bool WebRtcSession::SetLocalDescription(
- std::unique_ptr<SessionDescriptionInterface> desc,
- std::string* err_desc) {
+bool WebRtcSession::SetLocalDescription(SessionDescriptionInterface* desc,
+ std::string* err_desc) {
RTC_DCHECK(signaling_thread()->IsCurrent());
+ // Takes the ownership of |desc| regardless of the result.
+ std::unique_ptr<SessionDescriptionInterface> desc_temp(desc);
+
// Validate SDP.
- if (!ValidateSessionDescription(desc.get(), cricket::CS_LOCAL, err_desc)) {
+ if (!ValidateSessionDescription(desc, cricket::CS_LOCAL, err_desc)) {
return false;
}
@@ -725,19 +727,18 @@
}
if (action == kAnswer) {
- current_local_description_ = std::move(desc);
- pending_local_description_ = nullptr;
- current_remote_description_ = std::move(pending_remote_description_);
+ current_local_description_.reset(desc_temp.release());
+ pending_local_description_.reset(nullptr);
+ current_remote_description_.reset(pending_remote_description_.release());
} else {
- pending_local_description_ = std::move(desc);
+ pending_local_description_.reset(desc_temp.release());
}
// Transport and Media channels will be created only when offer is set.
if (action == kOffer && !CreateChannels(local_description()->description())) {
// TODO(mallinath) - Handle CreateChannel failure, as new local description
// is applied. Restore back to old description.
- return BadLocalSdp(local_description()->type(), kCreateChannelFailed,
- err_desc);
+ return BadLocalSdp(desc->type(), kCreateChannelFailed, err_desc);
}
// Remove unused channels if MediaContentDescription is rejected.
@@ -753,54 +754,50 @@
pending_ice_restarts_.clear();
if (error() != ERROR_NONE) {
- return BadLocalSdp(local_description()->type(), GetSessionErrorMsg(),
- err_desc);
+ return BadLocalSdp(desc->type(), GetSessionErrorMsg(), err_desc);
}
return true;
}
-bool WebRtcSession::SetRemoteDescription(
- std::unique_ptr<SessionDescriptionInterface> desc,
- std::string* err_desc) {
+bool WebRtcSession::SetRemoteDescription(SessionDescriptionInterface* desc,
+ std::string* err_desc) {
RTC_DCHECK(signaling_thread()->IsCurrent());
+ // Takes the ownership of |desc| regardless of the result.
+ std::unique_ptr<SessionDescriptionInterface> desc_temp(desc);
+
// Validate SDP.
- if (!ValidateSessionDescription(desc.get(), cricket::CS_REMOTE, err_desc)) {
+ if (!ValidateSessionDescription(desc, cricket::CS_REMOTE, err_desc)) {
return false;
}
- // Hold this pointer so candidates can be copied to it later in the method.
- SessionDescriptionInterface* desc_ptr = desc.get();
-
const SessionDescriptionInterface* old_remote_description =
remote_description();
// Grab ownership of the description being replaced for the remainder of this
- // method, since it's used below as |old_remote_description|.
+ // method, since it's used below.
std::unique_ptr<SessionDescriptionInterface> replaced_remote_description;
Action action = GetAction(desc->type());
if (action == kAnswer) {
- replaced_remote_description = pending_remote_description_
- ? std::move(pending_remote_description_)
- : std::move(current_remote_description_);
- current_remote_description_ = std::move(desc);
- pending_remote_description_ = nullptr;
- current_local_description_ = std::move(pending_local_description_);
+ replaced_remote_description.reset(
+ pending_remote_description_ ? pending_remote_description_.release()
+ : current_remote_description_.release());
+ current_remote_description_.reset(desc_temp.release());
+ pending_remote_description_.reset(nullptr);
+ current_local_description_.reset(pending_local_description_.release());
} else {
- replaced_remote_description = std::move(pending_remote_description_);
- pending_remote_description_ = std::move(desc);
+ replaced_remote_description.reset(pending_remote_description_.release());
+ pending_remote_description_.reset(desc_temp.release());
}
// Transport and Media channels will be created only when offer is set.
- if (action == kOffer &&
- !CreateChannels(remote_description()->description())) {
+ if (action == kOffer && !CreateChannels(desc->description())) {
// TODO(mallinath) - Handle CreateChannel failure, as new local description
// is applied. Restore back to old description.
- return BadRemoteSdp(remote_description()->type(), kCreateChannelFailed,
- err_desc);
+ return BadRemoteSdp(desc->type(), kCreateChannelFailed, err_desc);
}
// Remove unused channels if MediaContentDescription is rejected.
- RemoveUnusedChannels(remote_description()->description());
+ RemoveUnusedChannels(desc->description());
// NOTE: Candidates allocation will be initiated only when SetLocalDescription
// is called.
@@ -808,10 +805,8 @@
return false;
}
- if (local_description() &&
- !UseCandidatesInSessionDescription(remote_description())) {
- return BadRemoteSdp(remote_description()->type(), kInvalidCandidates,
- err_desc);
+ if (local_description() && !UseCandidatesInSessionDescription(desc)) {
+ return BadRemoteSdp(desc->type(), kInvalidCandidates, err_desc);
}
if (old_remote_description) {
@@ -822,7 +817,7 @@
// TODO(deadbeef): When we start storing both the current and pending
// remote description, this should reset pending_ice_restarts and compare
// against the current description.
- if (CheckForRemoteIceRestart(old_remote_description, remote_description(),
+ if (CheckForRemoteIceRestart(old_remote_description, desc,
content.name)) {
if (action == kOffer) {
pending_ice_restarts_.insert(content.name);
@@ -836,14 +831,13 @@
// description plus any candidates added since then. We should remove
// this once we're sure it won't break anything.
WebRtcSessionDescriptionFactory::CopyCandidatesFromSessionDescription(
- old_remote_description, content.name, desc_ptr);
+ old_remote_description, content.name, desc);
}
}
}
if (error() != ERROR_NONE) {
- return BadRemoteSdp(remote_description()->type(), GetSessionErrorMsg(),
- err_desc);
+ return BadRemoteSdp(desc->type(), GetSessionErrorMsg(), err_desc);
}
// Set the the ICE connection state to connecting since the connection may
@@ -854,7 +848,7 @@
// transport and expose a new checking() member from transport that can be
// read to determine the current checking state. The existing SignalConnecting
// actually means "gathering candidates", so cannot be be used here.
- if (remote_description()->type() != SessionDescriptionInterface::kOffer &&
+ if (desc->type() != SessionDescriptionInterface::kOffer &&
ice_connection_state_ == PeerConnectionInterface::kIceConnectionNew) {
SetIceConnectionState(PeerConnectionInterface::kIceConnectionChecking);
}
diff --git a/pc/webrtcsession.h b/pc/webrtcsession.h
index 185fa05..16c3931 100644
--- a/pc/webrtcsession.h
+++ b/pc/webrtcsession.h
@@ -254,9 +254,11 @@
const cricket::MediaSessionOptions& session_options);
void CreateAnswer(CreateSessionDescriptionObserver* observer,
const cricket::MediaSessionOptions& session_options);
- bool SetLocalDescription(std::unique_ptr<SessionDescriptionInterface> desc,
+ // The ownership of |desc| will be transferred after this call.
+ bool SetLocalDescription(SessionDescriptionInterface* desc,
std::string* err_desc);
- bool SetRemoteDescription(std::unique_ptr<SessionDescriptionInterface> desc,
+ // The ownership of |desc| will be transferred after this call.
+ bool SetRemoteDescription(SessionDescriptionInterface* desc,
std::string* err_desc);
bool ProcessIceMessage(const IceCandidateInterface* ice_candidate);
diff --git a/pc/webrtcsession_unittest.cc b/pc/webrtcsession_unittest.cc
index 0c54abc..25cb35a 100644
--- a/pc/webrtcsession_unittest.cc
+++ b/pc/webrtcsession_unittest.cc
@@ -92,6 +92,26 @@
static const int kDefaultTimeout = 10000; // 10 seconds.
static const int kIceCandidatesTimeout = 10000;
+static const char kSdpWithRtx[] =
+ "v=0\r\n"
+ "o=- 4104004319237231850 2 IN IP4 127.0.0.1\r\n"
+ "s=-\r\n"
+ "t=0 0\r\n"
+ "a=msid-semantic: WMS stream1\r\n"
+ "m=video 9 RTP/SAVPF 0 96\r\n"
+ "c=IN IP4 0.0.0.0\r\n"
+ "a=rtcp:9 IN IP4 0.0.0.0\r\n"
+ "a=ice-ufrag:CerjGp19G7wpXwl7\r\n"
+ "a=ice-pwd:cMvOlFvQ6ochez1ZOoC2uBEC\r\n"
+ "a=mid:video\r\n"
+ "a=sendrecv\r\n"
+ "a=rtcp-mux\r\n"
+ "a=crypto:1 AES_CM_128_HMAC_SHA1_80 "
+ "inline:5/4N5CDvMiyDArHtBByUM71VIkguH17ZNoX60GrA\r\n"
+ "a=rtpmap:0 fake_video_codec/90000\r\n"
+ "a=rtpmap:96 rtx/90000\r\n"
+ "a=fmtp:96 apt=0\r\n";
+
static const char kStream1[] = "stream1";
static const char kVideoTrack1[] = "video1";
static const char kAudioTrack1[] = "audio1";
@@ -100,6 +120,7 @@
static const char kVideoTrack2[] = "video2";
static const char kAudioTrack2[] = "audio2";
+static constexpr bool kStopped = true;
static constexpr bool kActive = false;
enum RTCCertificateGenerationMethod { ALREADY_GENERATED, DTLS_IDENTITY_STORE };
@@ -444,6 +465,24 @@
remote_send_video_ = true;
}
+ void SendAudioVideoStream1And2() {
+ send_stream_1_ = true;
+ send_stream_2_ = true;
+ local_send_audio_ = true;
+ local_send_video_ = true;
+ remote_send_audio_ = true;
+ remote_send_video_ = true;
+ }
+
+ void SendNothing() {
+ send_stream_1_ = false;
+ send_stream_2_ = false;
+ local_send_audio_ = false;
+ local_send_video_ = false;
+ remote_send_audio_ = false;
+ remote_send_video_ = false;
+ }
+
void SendAudioOnlyStream2() {
send_stream_1_ = false;
send_stream_2_ = true;
@@ -462,6 +501,19 @@
remote_send_video_ = true;
}
+ // Helper function used to add a specific media section to the
+ // |session_options|.
+ void AddMediaSection(cricket::MediaType type,
+ const std::string& mid,
+ cricket::MediaContentDirection direction,
+ bool stopped,
+ cricket::MediaSessionOptions* opts) {
+ opts->media_description_options.push_back(cricket::MediaDescriptionOptions(
+ type, mid,
+ cricket::RtpTransceiverDirection::FromMediaContentDirection(direction),
+ stopped));
+ }
+
// Add the media sections to the options from |offered_media_sections_| when
// creating an answer or a new offer.
// This duplicates a lot of logic from PeerConnection but this can be fixed
@@ -636,6 +688,13 @@
session_options->crypto_options = crypto_options_;
}
+ void GetOptionsForAudioOnlyRemoteOffer(
+ cricket::MediaSessionOptions* session_options) {
+ remote_recv_audio_ = true;
+ remote_recv_video_ = false;
+ GetOptionsForRemoteOffer(session_options);
+ }
+
void GetOptionsForRemoteOffer(cricket::MediaSessionOptions* session_options) {
AddMediaSectionsAndSendersToOptions(session_options, remote_send_audio_,
remote_recv_audio_, remote_send_video_,
@@ -752,15 +811,30 @@
transport_desc->ice_pwd = pwd;
}
+ // Creates a remote offer and and applies it as a remote description,
+ // creates a local answer and applies is as a local description.
+ // Call SendAudioVideoStreamX() before this function
+ // to decide which local and remote streams to create.
+ void CreateAndSetRemoteOfferAndLocalAnswer() {
+ SessionDescriptionInterface* offer = CreateRemoteOffer();
+ SetRemoteDescriptionWithoutError(offer);
+ SessionDescriptionInterface* answer = CreateAnswer();
+ SetLocalDescriptionWithoutError(answer);
+ }
void SetLocalDescriptionWithoutError(SessionDescriptionInterface* desc) {
- ASSERT_TRUE(session_->SetLocalDescription(rtc::WrapUnique(desc), nullptr));
+ ASSERT_TRUE(session_->SetLocalDescription(desc, nullptr));
session_->MaybeStartGathering();
}
+ void SetLocalDescriptionExpectState(SessionDescriptionInterface* desc,
+ WebRtcSession::State expected_state) {
+ SetLocalDescriptionWithoutError(desc);
+ EXPECT_EQ(expected_state, session_->state());
+ }
void SetLocalDescriptionExpectError(const std::string& action,
const std::string& expected_error,
SessionDescriptionInterface* desc) {
std::string error;
- EXPECT_FALSE(session_->SetLocalDescription(rtc::WrapUnique(desc), &error));
+ EXPECT_FALSE(session_->SetLocalDescription(desc, &error));
std::string sdp_type = "local ";
sdp_type.append(action);
EXPECT_NE(std::string::npos, error.find(sdp_type));
@@ -771,14 +845,24 @@
SetLocalDescriptionExpectError(SessionDescriptionInterface::kOffer,
expected_error, desc);
}
+ void SetLocalDescriptionAnswerExpectError(const std::string& expected_error,
+ SessionDescriptionInterface* desc) {
+ SetLocalDescriptionExpectError(SessionDescriptionInterface::kAnswer,
+ expected_error, desc);
+ }
void SetRemoteDescriptionWithoutError(SessionDescriptionInterface* desc) {
- ASSERT_TRUE(session_->SetRemoteDescription(rtc::WrapUnique(desc), nullptr));
+ ASSERT_TRUE(session_->SetRemoteDescription(desc, nullptr));
+ }
+ void SetRemoteDescriptionExpectState(SessionDescriptionInterface* desc,
+ WebRtcSession::State expected_state) {
+ SetRemoteDescriptionWithoutError(desc);
+ EXPECT_EQ(expected_state, session_->state());
}
void SetRemoteDescriptionExpectError(const std::string& action,
const std::string& expected_error,
SessionDescriptionInterface* desc) {
std::string error;
- EXPECT_FALSE(session_->SetRemoteDescription(rtc::WrapUnique(desc), &error));
+ EXPECT_FALSE(session_->SetRemoteDescription(desc, &error));
std::string sdp_type = "remote ";
sdp_type.append(action);
EXPECT_NE(std::string::npos, error.find(sdp_type));
@@ -789,6 +873,11 @@
SetRemoteDescriptionExpectError(SessionDescriptionInterface::kOffer,
expected_error, desc);
}
+ void SetRemoteDescriptionAnswerExpectError(
+ const std::string& expected_error, SessionDescriptionInterface* desc) {
+ SetRemoteDescriptionExpectError(SessionDescriptionInterface::kAnswer,
+ expected_error, desc);
+ }
JsepSessionDescription* CreateRemoteOfferWithVersion(
cricket::MediaSessionOptions options,
@@ -946,6 +1035,23 @@
}
}
+ bool ContainsVideoCodecWithName(const SessionDescriptionInterface* desc,
+ const std::string& codec_name) {
+ for (const auto& content : desc->description()->contents()) {
+ if (static_cast<cricket::MediaContentDescription*>(content.description)
+ ->type() == cricket::MEDIA_TYPE_VIDEO) {
+ const auto* mdesc =
+ static_cast<cricket::VideoContentDescription*>(content.description);
+ for (const auto& codec : mdesc->codecs()) {
+ if (codec.name == codec_name) {
+ return true;
+ }
+ }
+ }
+ }
+ return false;
+ }
+
// The method sets up a call from the session to itself, in a loopback
// arrangement. It also uses a firewall rule to create a temporary
// disconnection, and then a permanent disconnection.
@@ -1009,6 +1115,33 @@
EXPECT_GT(fake_call_.last_sent_packet().send_time_ms, -1);
}
+ // Adds CN codecs to FakeMediaEngine and MediaDescriptionFactory.
+ void AddCNCodecs() {
+ const cricket::AudioCodec kCNCodec1(102, "CN", 8000, 0, 1);
+ const cricket::AudioCodec kCNCodec2(103, "CN", 16000, 0, 1);
+
+ // Add kCNCodec for dtmf test.
+ std::vector<cricket::AudioCodec> codecs =
+ media_engine_->audio_send_codecs();
+ codecs.push_back(kCNCodec1);
+ codecs.push_back(kCNCodec2);
+ media_engine_->SetAudioCodecs(codecs);
+ desc_factory_->set_audio_codecs(codecs, codecs);
+ }
+
+ bool VerifyNoCNCodecs(const cricket::ContentInfo* content) {
+ const cricket::ContentDescription* description = content->description;
+ RTC_CHECK(description != NULL);
+ const cricket::AudioContentDescription* audio_content_desc =
+ static_cast<const cricket::AudioContentDescription*>(description);
+ RTC_CHECK(audio_content_desc != NULL);
+ for (size_t i = 0; i < audio_content_desc->codecs().size(); ++i) {
+ if (audio_content_desc->codecs()[i].name == "CN")
+ return false;
+ }
+ return true;
+ }
+
void CreateDataChannel() {
webrtc::InternalDataChannelInit dci;
RTC_CHECK(session_.get());
@@ -1081,6 +1214,141 @@
TestSessionCandidatesWithBundleRtcpMux(true, true);
}
+TEST_F(WebRtcSessionTest, SetSdpFailedOnInvalidSdp) {
+ Init();
+ SessionDescriptionInterface* offer = NULL;
+ // Since |offer| is NULL, there's no way to tell if it's an offer or answer.
+ std::string unknown_action;
+ SetLocalDescriptionExpectError(unknown_action, kInvalidSdp, offer);
+ SetRemoteDescriptionExpectError(unknown_action, kInvalidSdp, offer);
+}
+
+// Test creating offers and receive answers and make sure the
+// media engine creates the expected send and receive streams.
+TEST_F(WebRtcSessionTest, TestCreateSdesOfferReceiveSdesAnswer) {
+ Init();
+ SendAudioVideoStream1();
+ SessionDescriptionInterface* offer = CreateOffer();
+ const std::string session_id_orig = offer->session_id();
+ const std::string session_version_orig = offer->session_version();
+ SetLocalDescriptionWithoutError(offer);
+
+ SendAudioVideoStream2();
+ SessionDescriptionInterface* answer =
+ CreateRemoteAnswer(session_->local_description());
+ SetRemoteDescriptionWithoutError(answer);
+
+ video_channel_ = media_engine_->GetVideoChannel(0);
+ voice_channel_ = media_engine_->GetVoiceChannel(0);
+
+ ASSERT_EQ(1u, video_channel_->recv_streams().size());
+ EXPECT_TRUE(kVideoTrack2 == video_channel_->recv_streams()[0].id);
+
+ ASSERT_EQ(1u, voice_channel_->recv_streams().size());
+ EXPECT_TRUE(kAudioTrack2 == voice_channel_->recv_streams()[0].id);
+
+ ASSERT_EQ(1u, video_channel_->send_streams().size());
+ EXPECT_TRUE(kVideoTrack1 == video_channel_->send_streams()[0].id);
+ ASSERT_EQ(1u, voice_channel_->send_streams().size());
+ EXPECT_TRUE(kAudioTrack1 == voice_channel_->send_streams()[0].id);
+
+ // Create new offer without send streams.
+ SendNothing();
+ offer = CreateOffer();
+
+ // Verify the session id is the same and the session version is
+ // increased.
+ EXPECT_EQ(session_id_orig, offer->session_id());
+ EXPECT_LT(rtc::FromString<uint64_t>(session_version_orig),
+ rtc::FromString<uint64_t>(offer->session_version()));
+
+ SetLocalDescriptionWithoutError(offer);
+ EXPECT_EQ(0u, video_channel_->send_streams().size());
+ EXPECT_EQ(0u, voice_channel_->send_streams().size());
+
+ SendAudioVideoStream2();
+ answer = CreateRemoteAnswer(session_->local_description());
+ SetRemoteDescriptionWithoutError(answer);
+
+ // Make sure the receive streams have not changed.
+ ASSERT_EQ(1u, video_channel_->recv_streams().size());
+ EXPECT_TRUE(kVideoTrack2 == video_channel_->recv_streams()[0].id);
+ ASSERT_EQ(1u, voice_channel_->recv_streams().size());
+ EXPECT_TRUE(kAudioTrack2 == voice_channel_->recv_streams()[0].id);
+}
+
+// Test receiving offers and creating answers and make sure the
+// media engine creates the expected send and receive streams.
+TEST_F(WebRtcSessionTest, TestReceiveSdesOfferCreateSdesAnswer) {
+ Init();
+ SendAudioVideoStream2();
+ SessionDescriptionInterface* offer = CreateOffer();
+ SetRemoteDescriptionWithoutError(offer);
+
+ SendAudioVideoStream1();
+ SessionDescriptionInterface* answer = CreateAnswer();
+ SetLocalDescriptionWithoutError(answer);
+
+ const std::string session_id_orig = answer->session_id();
+ const std::string session_version_orig = answer->session_version();
+
+ video_channel_ = media_engine_->GetVideoChannel(0);
+ voice_channel_ = media_engine_->GetVoiceChannel(0);
+
+ ASSERT_TRUE(video_channel_);
+ ASSERT_TRUE(voice_channel_);
+ ASSERT_EQ(1u, video_channel_->recv_streams().size());
+ EXPECT_TRUE(kVideoTrack2 == video_channel_->recv_streams()[0].id);
+
+ ASSERT_EQ(1u, voice_channel_->recv_streams().size());
+ EXPECT_TRUE(kAudioTrack2 == voice_channel_->recv_streams()[0].id);
+
+ ASSERT_EQ(1u, video_channel_->send_streams().size());
+ EXPECT_TRUE(kVideoTrack1 == video_channel_->send_streams()[0].id);
+ ASSERT_EQ(1u, voice_channel_->send_streams().size());
+ EXPECT_TRUE(kAudioTrack1 == voice_channel_->send_streams()[0].id);
+
+ SendAudioVideoStream1And2();
+ offer = CreateOffer();
+ SetRemoteDescriptionWithoutError(offer);
+
+ // Answer by turning off all send streams.
+ SendNothing();
+ answer = CreateAnswer();
+
+ // Verify the session id is the same and the session version is
+ // increased.
+ EXPECT_EQ(session_id_orig, answer->session_id());
+ EXPECT_LT(rtc::FromString<uint64_t>(session_version_orig),
+ rtc::FromString<uint64_t>(answer->session_version()));
+ SetLocalDescriptionWithoutError(answer);
+
+ ASSERT_EQ(2u, video_channel_->recv_streams().size());
+ EXPECT_TRUE(kVideoTrack1 == video_channel_->recv_streams()[0].id);
+ EXPECT_TRUE(kVideoTrack2 == video_channel_->recv_streams()[1].id);
+ ASSERT_EQ(2u, voice_channel_->recv_streams().size());
+ EXPECT_TRUE(kAudioTrack1 == voice_channel_->recv_streams()[0].id);
+ EXPECT_TRUE(kAudioTrack2 == voice_channel_->recv_streams()[1].id);
+
+ // Make sure we have no send streams.
+ EXPECT_EQ(0u, video_channel_->send_streams().size());
+ EXPECT_EQ(0u, voice_channel_->send_streams().size());
+}
+
+TEST_F(WebRtcSessionTest, SetLocalSdpFailedOnCreateChannel) {
+ Init();
+ media_engine_->set_fail_create_channel(true);
+
+ SessionDescriptionInterface* offer = CreateOffer();
+ ASSERT_TRUE(offer != NULL);
+ // SetRemoteDescription and SetLocalDescription will take the ownership of
+ // the offer.
+ SetRemoteDescriptionOfferExpectError(kCreateChannelFailed, offer);
+ offer = CreateOffer();
+ ASSERT_TRUE(offer != NULL);
+ SetLocalDescriptionOfferExpectError(kCreateChannelFailed, offer);
+}
+
// Test that we can create and set an answer correctly when different
// SSL roles have been negotiated for different transports.
// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4525
@@ -1144,6 +1412,592 @@
SetLocalDescriptionWithoutError(answer);
}
+TEST_F(WebRtcSessionTest, TestSetLocalOfferTwice) {
+ Init();
+ SendNothing();
+ // SetLocalDescription take ownership of offer.
+ SessionDescriptionInterface* offer = CreateOffer();
+ SetLocalDescriptionWithoutError(offer);
+
+ // SetLocalDescription take ownership of offer.
+ SessionDescriptionInterface* offer2 = CreateOffer();
+ SetLocalDescriptionWithoutError(offer2);
+}
+
+TEST_F(WebRtcSessionTest, TestSetRemoteOfferTwice) {
+ Init();
+ SendNothing();
+ // SetLocalDescription take ownership of offer.
+ SessionDescriptionInterface* offer = CreateOffer();
+ SetRemoteDescriptionWithoutError(offer);
+
+ SessionDescriptionInterface* offer2 = CreateOffer();
+ SetRemoteDescriptionWithoutError(offer2);
+}
+
+TEST_F(WebRtcSessionTest, TestSetLocalAndRemoteOffer) {
+ Init();
+ SendNothing();
+ SessionDescriptionInterface* offer = CreateOffer();
+ SetLocalDescriptionWithoutError(offer);
+ offer = CreateOffer();
+ SetRemoteDescriptionOfferExpectError("Called in wrong state: STATE_SENTOFFER",
+ offer);
+}
+
+TEST_F(WebRtcSessionTest, TestSetRemoteAndLocalOffer) {
+ Init();
+ SendNothing();
+ SessionDescriptionInterface* offer = CreateOffer();
+ SetRemoteDescriptionWithoutError(offer);
+ offer = CreateOffer();
+ SetLocalDescriptionOfferExpectError(
+ "Called in wrong state: STATE_RECEIVEDOFFER", offer);
+}
+
+TEST_F(WebRtcSessionTest, TestSetLocalPrAnswer) {
+ Init();
+ SendNothing();
+ SessionDescriptionInterface* offer = CreateRemoteOffer();
+ SetRemoteDescriptionExpectState(offer, WebRtcSession::STATE_RECEIVEDOFFER);
+
+ JsepSessionDescription* pranswer =
+ static_cast<JsepSessionDescription*>(CreateAnswer());
+ pranswer->set_type(SessionDescriptionInterface::kPrAnswer);
+ SetLocalDescriptionExpectState(pranswer, WebRtcSession::STATE_SENTPRANSWER);
+
+ SendAudioVideoStream1();
+ JsepSessionDescription* pranswer2 =
+ static_cast<JsepSessionDescription*>(CreateAnswer());
+ pranswer2->set_type(SessionDescriptionInterface::kPrAnswer);
+
+ SetLocalDescriptionExpectState(pranswer2, WebRtcSession::STATE_SENTPRANSWER);
+
+ SendAudioVideoStream2();
+ SessionDescriptionInterface* answer = CreateAnswer();
+ SetLocalDescriptionExpectState(answer, WebRtcSession::STATE_INPROGRESS);
+}
+
+TEST_F(WebRtcSessionTest, TestSetRemotePrAnswer) {
+ Init();
+ SendNothing();
+ SessionDescriptionInterface* offer = CreateOffer();
+ SetLocalDescriptionExpectState(offer, WebRtcSession::STATE_SENTOFFER);
+
+ JsepSessionDescription* pranswer =
+ CreateRemoteAnswer(session_->local_description());
+ pranswer->set_type(SessionDescriptionInterface::kPrAnswer);
+
+ SetRemoteDescriptionExpectState(pranswer,
+ WebRtcSession::STATE_RECEIVEDPRANSWER);
+
+ SendAudioVideoStream1();
+ JsepSessionDescription* pranswer2 =
+ CreateRemoteAnswer(session_->local_description());
+ pranswer2->set_type(SessionDescriptionInterface::kPrAnswer);
+
+ SetRemoteDescriptionExpectState(pranswer2,
+ WebRtcSession::STATE_RECEIVEDPRANSWER);
+
+ SendAudioVideoStream2();
+ SessionDescriptionInterface* answer =
+ CreateRemoteAnswer(session_->local_description());
+ SetRemoteDescriptionExpectState(answer, WebRtcSession::STATE_INPROGRESS);
+}
+
+TEST_F(WebRtcSessionTest, TestSetLocalAnswerWithoutOffer) {
+ Init();
+ SendNothing();
+ std::unique_ptr<SessionDescriptionInterface> offer(CreateOffer());
+
+ SessionDescriptionInterface* answer =
+ CreateRemoteAnswer(offer.get());
+ SetLocalDescriptionAnswerExpectError("Called in wrong state: STATE_INIT",
+ answer);
+}
+
+TEST_F(WebRtcSessionTest, TestSetRemoteAnswerWithoutOffer) {
+ Init();
+ SendNothing();
+ std::unique_ptr<SessionDescriptionInterface> offer(CreateOffer());
+
+ SessionDescriptionInterface* answer =
+ CreateRemoteAnswer(offer.get());
+ SetRemoteDescriptionAnswerExpectError(
+ "Called in wrong state: STATE_INIT", answer);
+}
+
+// Verifies TransportProxy and media channels are created with content names
+// present in the SessionDescription.
+TEST_F(WebRtcSessionTest, TestChannelCreationsWithContentNames) {
+ Init();
+ SendAudioVideoStream1();
+ std::unique_ptr<SessionDescriptionInterface> offer(CreateOffer());
+
+ // CreateOffer creates session description with the content names "audio" and
+ // "video". Goal is to modify these content names and verify transport
+ // channels
+ // in the WebRtcSession, as channels are created with the content names
+ // present in SDP.
+ std::string sdp;
+ EXPECT_TRUE(offer->ToString(&sdp));
+
+ SessionDescriptionInterface* modified_offer =
+ CreateSessionDescription(JsepSessionDescription::kOffer, sdp, NULL);
+
+ SetRemoteDescriptionWithoutError(modified_offer);
+
+ cricket::MediaSessionOptions answer_options;
+ answer_options.bundle_enabled = false;
+ SessionDescriptionInterface* answer = CreateAnswer(answer_options);
+ SetLocalDescriptionWithoutError(answer);
+
+ rtc::PacketTransportInternal* voice_transport_channel =
+ session_->voice_rtp_transport_channel();
+ EXPECT_TRUE(voice_transport_channel != NULL);
+ EXPECT_EQ(voice_transport_channel->debug_name(),
+ "audio " + std::to_string(cricket::ICE_CANDIDATE_COMPONENT_RTP));
+ rtc::PacketTransportInternal* video_transport_channel =
+ session_->video_rtp_transport_channel();
+ ASSERT_TRUE(video_transport_channel != NULL);
+ EXPECT_EQ(video_transport_channel->debug_name(),
+ "video " + std::to_string(cricket::ICE_CANDIDATE_COMPONENT_RTP));
+ EXPECT_TRUE((video_channel_ = media_engine_->GetVideoChannel(0)) != NULL);
+ EXPECT_TRUE((voice_channel_ = media_engine_->GetVoiceChannel(0)) != NULL);
+}
+
+// Test that an offer contains the correct media content descriptions based on
+// the send streams when no constraints have been set.
+TEST_F(WebRtcSessionTest, CreateOfferWithoutConstraintsOrStreams) {
+ Init();
+ std::unique_ptr<SessionDescriptionInterface> offer(CreateOffer());
+
+ ASSERT_TRUE(offer != NULL);
+ const cricket::ContentInfo* content =
+ cricket::GetFirstAudioContent(offer->description());
+ ASSERT_TRUE(content != NULL);
+ EXPECT_EQ(
+ cricket::MD_RECVONLY,
+ static_cast<const cricket::AudioContentDescription*>(content->description)
+ ->direction());
+ content = cricket::GetFirstVideoContent(offer->description());
+ ASSERT_TRUE(content != NULL);
+ EXPECT_EQ(
+ cricket::MD_RECVONLY,
+ static_cast<const cricket::VideoContentDescription*>(content->description)
+ ->direction());
+}
+
+// Test that an offer contains the correct media content descriptions based on
+// the send streams when no constraints have been set.
+TEST_F(WebRtcSessionTest, CreateOfferWithoutConstraints) {
+ Init();
+ // Test Audio only offer.
+ SendAudioOnlyStream2();
+ std::unique_ptr<SessionDescriptionInterface> offer(CreateOffer());
+
+ const cricket::ContentInfo* content =
+ cricket::GetFirstAudioContent(offer->description());
+ ASSERT_TRUE(content != NULL);
+ EXPECT_EQ(
+ cricket::MD_SENDRECV,
+ static_cast<const cricket::AudioContentDescription*>(content->description)
+ ->direction());
+ content = cricket::GetFirstVideoContent(offer->description());
+ ASSERT_TRUE(content != NULL);
+ EXPECT_EQ(
+ cricket::MD_RECVONLY,
+ static_cast<const cricket::VideoContentDescription*>(content->description)
+ ->direction());
+
+ // Test Audio / Video offer.
+ SendAudioVideoStream1();
+ offer.reset(CreateOffer());
+ content = cricket::GetFirstAudioContent(offer->description());
+ ASSERT_TRUE(content != NULL);
+ EXPECT_EQ(
+ cricket::MD_SENDRECV,
+ static_cast<const cricket::AudioContentDescription*>(content->description)
+ ->direction());
+
+ content = cricket::GetFirstVideoContent(offer->description());
+ ASSERT_TRUE(content != NULL);
+ EXPECT_EQ(
+ cricket::MD_SENDRECV,
+ static_cast<const cricket::VideoContentDescription*>(content->description)
+ ->direction());
+}
+
+// Test that an offer contains no media content descriptions if
+// kOfferToReceiveVideo and kOfferToReceiveAudio constraints are set to false.
+TEST_F(WebRtcSessionTest, CreateOfferWithConstraintsWithoutStreams) {
+ Init();
+ PeerConnectionInterface::RTCOfferAnswerOptions options;
+ options.offer_to_receive_audio = 0;
+ options.offer_to_receive_video = 0;
+
+ std::unique_ptr<SessionDescriptionInterface> offer(CreateOffer(options));
+
+ ASSERT_TRUE(offer != NULL);
+ const cricket::ContentInfo* content =
+ cricket::GetFirstAudioContent(offer->description());
+ EXPECT_TRUE(content == NULL);
+ content = cricket::GetFirstVideoContent(offer->description());
+ EXPECT_TRUE(content == NULL);
+}
+
+// Test that an offer contains only audio media content descriptions if
+// kOfferToReceiveAudio constraints are set to true.
+TEST_F(WebRtcSessionTest, CreateAudioOnlyOfferWithConstraints) {
+ Init();
+ PeerConnectionInterface::RTCOfferAnswerOptions options;
+ options.offer_to_receive_audio =
+ RTCOfferAnswerOptions::kOfferToReceiveMediaTrue;
+ options.offer_to_receive_video = 0;
+
+ std::unique_ptr<SessionDescriptionInterface> offer(CreateOffer(options));
+
+ const cricket::ContentInfo* content =
+ cricket::GetFirstAudioContent(offer->description());
+ EXPECT_TRUE(content != NULL);
+ content = cricket::GetFirstVideoContent(offer->description());
+ EXPECT_TRUE(content == NULL);
+}
+
+// Test that an offer contains audio and video media content descriptions if
+// kOfferToReceiveAudio and kOfferToReceiveVideo constraints are set to true.
+TEST_F(WebRtcSessionTest, CreateOfferWithConstraints) {
+ Init();
+ // Test Audio / Video offer.
+ PeerConnectionInterface::RTCOfferAnswerOptions options;
+ options.offer_to_receive_audio =
+ RTCOfferAnswerOptions::kOfferToReceiveMediaTrue;
+ options.offer_to_receive_video =
+ RTCOfferAnswerOptions::kOfferToReceiveMediaTrue;
+
+ std::unique_ptr<SessionDescriptionInterface> offer(CreateOffer(options));
+
+ const cricket::ContentInfo* content =
+ cricket::GetFirstAudioContent(offer->description());
+ EXPECT_TRUE(content != NULL);
+
+ content = cricket::GetFirstVideoContent(offer->description());
+ EXPECT_TRUE(content != NULL);
+
+ // Sets constraints to false and verifies that audio/video contents are
+ // removed.
+ options.offer_to_receive_audio = 0;
+ options.offer_to_receive_video = 0;
+ // Remove the media sections added in previous offer.
+ offered_media_sections_.clear();
+ offer.reset(CreateOffer(options));
+
+ content = cricket::GetFirstAudioContent(offer->description());
+ EXPECT_TRUE(content == NULL);
+ content = cricket::GetFirstVideoContent(offer->description());
+ EXPECT_TRUE(content == NULL);
+}
+
+// Test that an answer can not be created if the last remote description is not
+// an offer.
+TEST_F(WebRtcSessionTest, CreateAnswerWithoutAnOffer) {
+ Init();
+ SessionDescriptionInterface* offer = CreateOffer();
+ SetLocalDescriptionWithoutError(offer);
+ SessionDescriptionInterface* answer = CreateRemoteAnswer(offer);
+ SetRemoteDescriptionWithoutError(answer);
+ EXPECT_TRUE(CreateAnswer() == NULL);
+}
+
+// Test that an answer contains the correct media content descriptions when no
+// constraints have been set.
+TEST_F(WebRtcSessionTest, CreateAnswerWithoutConstraintsOrStreams) {
+ Init();
+ // Create a remote offer with audio and video content.
+ std::unique_ptr<JsepSessionDescription> offer(CreateRemoteOffer());
+ SetRemoteDescriptionWithoutError(offer.release());
+ std::unique_ptr<SessionDescriptionInterface> answer(CreateAnswer());
+ const cricket::ContentInfo* content =
+ cricket::GetFirstAudioContent(answer->description());
+ ASSERT_TRUE(content != NULL);
+ EXPECT_FALSE(content->rejected);
+
+ content = cricket::GetFirstVideoContent(answer->description());
+ ASSERT_TRUE(content != NULL);
+ EXPECT_FALSE(content->rejected);
+}
+
+// Test that an answer contains the correct media content descriptions when no
+// constraints have been set and the offer only contain audio.
+TEST_F(WebRtcSessionTest, CreateAudioAnswerWithoutConstraintsOrStreams) {
+ Init();
+ // Create a remote offer with audio only.
+ cricket::MediaSessionOptions options;
+ GetOptionsForAudioOnlyRemoteOffer(&options);
+
+ std::unique_ptr<JsepSessionDescription> offer(CreateRemoteOffer(options));
+ ASSERT_TRUE(cricket::GetFirstVideoContent(offer->description()) == NULL);
+ ASSERT_TRUE(cricket::GetFirstAudioContent(offer->description()) != NULL);
+
+ SetRemoteDescriptionWithoutError(offer.release());
+ std::unique_ptr<SessionDescriptionInterface> answer(CreateAnswer());
+ const cricket::ContentInfo* content =
+ cricket::GetFirstAudioContent(answer->description());
+ ASSERT_TRUE(content != NULL);
+ EXPECT_FALSE(content->rejected);
+
+ EXPECT_TRUE(cricket::GetFirstVideoContent(answer->description()) == NULL);
+}
+
+// Test that an answer contains the correct media content descriptions when no
+// constraints have been set.
+TEST_F(WebRtcSessionTest, CreateAnswerWithoutConstraints) {
+ Init();
+ // Create a remote offer with audio and video content.
+ std::unique_ptr<JsepSessionDescription> offer(CreateRemoteOffer());
+ SetRemoteDescriptionWithoutError(offer.release());
+ // Test with a stream with tracks.
+ SendAudioVideoStream1();
+ std::unique_ptr<SessionDescriptionInterface> answer(CreateAnswer());
+ const cricket::ContentInfo* content =
+ cricket::GetFirstAudioContent(answer->description());
+ ASSERT_TRUE(content != NULL);
+ EXPECT_FALSE(content->rejected);
+
+ content = cricket::GetFirstVideoContent(answer->description());
+ ASSERT_TRUE(content != NULL);
+ EXPECT_FALSE(content->rejected);
+}
+
+// Test that an answer contains the correct media content descriptions when
+// constraints have been set but no stream is sent.
+TEST_F(WebRtcSessionTest, CreateAnswerWithConstraintsWithoutStreams) {
+ Init();
+ // Create a remote offer with audio and video content.
+ std::unique_ptr<JsepSessionDescription> offer(CreateRemoteOffer());
+ SetRemoteDescriptionWithoutError(offer.release());
+
+ cricket::MediaSessionOptions session_options;
+ remote_send_audio_ = false;
+ remote_send_video_ = false;
+ local_recv_audio_ = false;
+ local_recv_video_ = false;
+ std::unique_ptr<SessionDescriptionInterface> answer(
+ CreateAnswer(session_options));
+
+ const cricket::ContentInfo* content =
+ cricket::GetFirstAudioContent(answer->description());
+ ASSERT_TRUE(content != NULL);
+ EXPECT_TRUE(content->rejected);
+
+ content = cricket::GetFirstVideoContent(answer->description());
+ ASSERT_TRUE(content != NULL);
+ EXPECT_TRUE(content->rejected);
+}
+
+// Test that an answer contains the correct media content descriptions when
+// constraints have been set and streams are sent.
+TEST_F(WebRtcSessionTest, CreateAnswerWithConstraints) {
+ Init();
+ // Create a remote offer with audio and video content.
+ std::unique_ptr<JsepSessionDescription> offer(CreateRemoteOffer());
+ SetRemoteDescriptionWithoutError(offer.release());
+
+ cricket::MediaSessionOptions options;
+ // Test with a stream with tracks.
+ SendAudioVideoStream1();
+ std::unique_ptr<SessionDescriptionInterface> answer(CreateAnswer(options));
+
+ // TODO(perkj): Should the direction be set to SEND_ONLY?
+ const cricket::ContentInfo* content =
+ cricket::GetFirstAudioContent(answer->description());
+ ASSERT_TRUE(content != NULL);
+ EXPECT_FALSE(content->rejected);
+
+ // TODO(perkj): Should the direction be set to SEND_ONLY?
+ content = cricket::GetFirstVideoContent(answer->description());
+ ASSERT_TRUE(content != NULL);
+ EXPECT_FALSE(content->rejected);
+}
+
+TEST_F(WebRtcSessionTest, CreateOfferWithoutCNCodecs) {
+ AddCNCodecs();
+ Init();
+ PeerConnectionInterface::RTCOfferAnswerOptions options;
+ options.offer_to_receive_audio =
+ RTCOfferAnswerOptions::kOfferToReceiveMediaTrue;
+ options.voice_activity_detection = false;
+
+ std::unique_ptr<SessionDescriptionInterface> offer(CreateOffer(options));
+
+ const cricket::ContentInfo* content =
+ cricket::GetFirstAudioContent(offer->description());
+ EXPECT_TRUE(content != NULL);
+ EXPECT_TRUE(VerifyNoCNCodecs(content));
+}
+
+TEST_F(WebRtcSessionTest, CreateAnswerWithoutCNCodecs) {
+ AddCNCodecs();
+ Init();
+ // Create a remote offer with audio and video content.
+ std::unique_ptr<JsepSessionDescription> offer(CreateRemoteOffer());
+ SetRemoteDescriptionWithoutError(offer.release());
+
+ cricket::MediaSessionOptions options;
+ options.vad_enabled = false;
+ std::unique_ptr<SessionDescriptionInterface> answer(CreateAnswer(options));
+ const cricket::ContentInfo* content =
+ cricket::GetFirstAudioContent(answer->description());
+ ASSERT_TRUE(content != NULL);
+ EXPECT_TRUE(VerifyNoCNCodecs(content));
+}
+
+// This test verifies the call setup when remote answer with audio only and
+// later updates with video.
+TEST_F(WebRtcSessionTest, TestAVOfferWithAudioOnlyAnswer) {
+ Init();
+ EXPECT_TRUE(media_engine_->GetVideoChannel(0) == NULL);
+ EXPECT_TRUE(media_engine_->GetVoiceChannel(0) == NULL);
+
+ SendAudioVideoStream1();
+ SessionDescriptionInterface* offer = CreateOffer();
+
+ cricket::MediaSessionOptions options;
+ AddMediaSection(cricket::MEDIA_TYPE_AUDIO, cricket::CN_AUDIO,
+ cricket::MD_RECVONLY, kActive, &options);
+ AddMediaSection(cricket::MEDIA_TYPE_VIDEO, cricket::CN_VIDEO,
+ cricket::MD_INACTIVE, kStopped, &options);
+ local_recv_video_ = false;
+ SessionDescriptionInterface* answer = CreateRemoteAnswer(offer, options);
+
+ // SetLocalDescription and SetRemoteDescriptions takes ownership of offer
+ // and answer;
+ SetLocalDescriptionWithoutError(offer);
+ SetRemoteDescriptionWithoutError(answer);
+
+ video_channel_ = media_engine_->GetVideoChannel(0);
+ voice_channel_ = media_engine_->GetVoiceChannel(0);
+
+ ASSERT_TRUE(video_channel_ == nullptr);
+
+ ASSERT_EQ(0u, voice_channel_->recv_streams().size());
+ ASSERT_EQ(1u, voice_channel_->send_streams().size());
+ EXPECT_EQ(kAudioTrack1, voice_channel_->send_streams()[0].id);
+
+ // Let the remote end update the session descriptions, with Audio and Video.
+ SendAudioVideoStream2();
+ local_recv_video_ = true;
+ CreateAndSetRemoteOfferAndLocalAnswer();
+
+ video_channel_ = media_engine_->GetVideoChannel(0);
+ voice_channel_ = media_engine_->GetVoiceChannel(0);
+
+ ASSERT_TRUE(video_channel_ != nullptr);
+ ASSERT_TRUE(voice_channel_ != nullptr);
+
+ ASSERT_EQ(1u, video_channel_->recv_streams().size());
+ ASSERT_EQ(1u, video_channel_->send_streams().size());
+ EXPECT_EQ(kVideoTrack2, video_channel_->recv_streams()[0].id);
+ EXPECT_EQ(kVideoTrack2, video_channel_->send_streams()[0].id);
+ ASSERT_EQ(1u, voice_channel_->recv_streams().size());
+ ASSERT_EQ(1u, voice_channel_->send_streams().size());
+ EXPECT_EQ(kAudioTrack2, voice_channel_->recv_streams()[0].id);
+ EXPECT_EQ(kAudioTrack2, voice_channel_->send_streams()[0].id);
+
+ // Change session back to audio only.
+ // The remote side doesn't send and recv video.
+ SendAudioOnlyStream2();
+ remote_recv_video_ = false;
+ CreateAndSetRemoteOfferAndLocalAnswer();
+
+ video_channel_ = media_engine_->GetVideoChannel(0);
+ voice_channel_ = media_engine_->GetVoiceChannel(0);
+
+ // The audio is expected to be rejected.
+ EXPECT_TRUE(video_channel_ == nullptr);
+
+ ASSERT_EQ(1u, voice_channel_->recv_streams().size());
+ EXPECT_EQ(kAudioTrack2, voice_channel_->recv_streams()[0].id);
+ ASSERT_EQ(1u, voice_channel_->send_streams().size());
+ EXPECT_EQ(kAudioTrack2, voice_channel_->send_streams()[0].id);
+}
+
+// This test verifies the call setup when remote answer with video only and
+// later updates with audio.
+TEST_F(WebRtcSessionTest, TestAVOfferWithVideoOnlyAnswer) {
+ Init();
+ EXPECT_TRUE(media_engine_->GetVideoChannel(0) == NULL);
+ EXPECT_TRUE(media_engine_->GetVoiceChannel(0) == NULL);
+ SendAudioVideoStream1();
+ SessionDescriptionInterface* offer = CreateOffer();
+
+ cricket::MediaSessionOptions options;
+ AddMediaSection(cricket::MEDIA_TYPE_AUDIO, cricket::CN_AUDIO,
+ cricket::MD_INACTIVE, kStopped, &options);
+ AddMediaSection(cricket::MEDIA_TYPE_VIDEO, cricket::CN_VIDEO,
+ cricket::MD_RECVONLY, kActive, &options);
+ local_recv_audio_ = false;
+ SessionDescriptionInterface* answer =
+ CreateRemoteAnswer(offer, options, cricket::SEC_ENABLED);
+
+ // SetLocalDescription and SetRemoteDescriptions takes ownership of offer
+ // and answer.
+ SetLocalDescriptionWithoutError(offer);
+ SetRemoteDescriptionWithoutError(answer);
+
+ video_channel_ = media_engine_->GetVideoChannel(0);
+ voice_channel_ = media_engine_->GetVoiceChannel(0);
+
+ ASSERT_TRUE(voice_channel_ == NULL);
+ ASSERT_TRUE(video_channel_ != NULL);
+
+ EXPECT_EQ(0u, video_channel_->recv_streams().size());
+ ASSERT_EQ(1u, video_channel_->send_streams().size());
+ EXPECT_EQ(kVideoTrack1, video_channel_->send_streams()[0].id);
+
+ // Update the session descriptions, with Audio and Video.
+ SendAudioVideoStream2();
+ local_recv_audio_ = true;
+ SessionDescriptionInterface* offer2 = CreateRemoteOffer();
+ SetRemoteDescriptionWithoutError(offer2);
+ cricket::MediaSessionOptions answer_options;
+ // Disable the bundling here. If the media is bundled on audio
+ // transport, then we can't reject the audio because switching the bundled
+ // transport is not currently supported.
+ // (https://bugs.chromium.org/p/webrtc/issues/detail?id=6704)
+ answer_options.bundle_enabled = false;
+ SessionDescriptionInterface* answer2 = CreateAnswer(answer_options);
+ SetLocalDescriptionWithoutError(answer2);
+
+ voice_channel_ = media_engine_->GetVoiceChannel(0);
+
+ ASSERT_TRUE(voice_channel_ != NULL);
+ ASSERT_EQ(1u, voice_channel_->recv_streams().size());
+ ASSERT_EQ(1u, voice_channel_->send_streams().size());
+ EXPECT_EQ(kAudioTrack2, voice_channel_->recv_streams()[0].id);
+ EXPECT_EQ(kAudioTrack2, voice_channel_->send_streams()[0].id);
+
+ // Change session back to video only.
+ // The remote side doesn't send and recv audio.
+ SendVideoOnlyStream2();
+ remote_recv_audio_ = false;
+ SessionDescriptionInterface* offer3 = CreateRemoteOffer();
+ SetRemoteDescriptionWithoutError(offer3);
+ SessionDescriptionInterface* answer3 = CreateAnswer(answer_options);
+ SetLocalDescriptionWithoutError(answer3);
+
+ video_channel_ = media_engine_->GetVideoChannel(0);
+ voice_channel_ = media_engine_->GetVoiceChannel(0);
+
+ // The video is expected to be rejected.
+ EXPECT_TRUE(voice_channel_ == nullptr);
+
+ ASSERT_EQ(1u, video_channel_->recv_streams().size());
+ EXPECT_EQ(kVideoTrack2, video_channel_->recv_streams()[0].id);
+ ASSERT_EQ(1u, video_channel_->send_streams().size());
+ EXPECT_EQ(kVideoTrack2, video_channel_->send_streams()[0].id);
+}
+
// Test that candidates sent to the "video" transport do not get pushed down to
// the "audio" transport channel when bundling.
TEST_F(WebRtcSessionTest, TestIgnoreCandidatesForUnusedTransportWhenBundling) {
@@ -1604,6 +2458,127 @@
SetLocalDescriptionWithoutError(offer);
}
+// This test verifies the |initial_offerer| flag when session initiates the
+// call.
+TEST_F(WebRtcSessionTest, TestInitiatorFlagAsOriginator) {
+ Init();
+ EXPECT_FALSE(session_->initial_offerer());
+ SessionDescriptionInterface* offer = CreateOffer();
+ SessionDescriptionInterface* answer = CreateRemoteAnswer(offer);
+ SetLocalDescriptionWithoutError(offer);
+ EXPECT_TRUE(session_->initial_offerer());
+ SetRemoteDescriptionWithoutError(answer);
+ EXPECT_TRUE(session_->initial_offerer());
+}
+
+// This test verifies the |initial_offerer| flag when session receives the call.
+TEST_F(WebRtcSessionTest, TestInitiatorFlagAsReceiver) {
+ Init();
+ EXPECT_FALSE(session_->initial_offerer());
+ SessionDescriptionInterface* offer = CreateRemoteOffer();
+ SetRemoteDescriptionWithoutError(offer);
+ SessionDescriptionInterface* answer = CreateAnswer();
+
+ EXPECT_FALSE(session_->initial_offerer());
+ SetLocalDescriptionWithoutError(answer);
+ EXPECT_FALSE(session_->initial_offerer());
+}
+
+// Verifing local offer and remote answer have matching m-lines as per RFC 3264.
+TEST_F(WebRtcSessionTest, TestIncorrectMLinesInRemoteAnswer) {
+ Init();
+ SendAudioVideoStream1();
+ SessionDescriptionInterface* offer = CreateOffer();
+ SetLocalDescriptionWithoutError(offer);
+ std::unique_ptr<SessionDescriptionInterface> answer(
+ CreateRemoteAnswer(session_->local_description()));
+
+ cricket::SessionDescription* answer_copy = answer->description()->Copy();
+ answer_copy->RemoveContentByName("video");
+ JsepSessionDescription* modified_answer =
+ new JsepSessionDescription(JsepSessionDescription::kAnswer);
+
+ EXPECT_TRUE(modified_answer->Initialize(answer_copy,
+ answer->session_id(),
+ answer->session_version()));
+ SetRemoteDescriptionAnswerExpectError(kMlineMismatchInAnswer,
+ modified_answer);
+
+ // Different content names.
+ std::string sdp;
+ EXPECT_TRUE(answer->ToString(&sdp));
+ const std::string kAudioMid = "a=mid:audio";
+ const std::string kAudioMidReplaceStr = "a=mid:audio_content_name";
+ rtc::replace_substrs(kAudioMid.c_str(), kAudioMid.length(),
+ kAudioMidReplaceStr.c_str(),
+ kAudioMidReplaceStr.length(),
+ &sdp);
+ SessionDescriptionInterface* modified_answer1 =
+ CreateSessionDescription(JsepSessionDescription::kAnswer, sdp, NULL);
+ SetRemoteDescriptionAnswerExpectError(kMlineMismatchInAnswer,
+ modified_answer1);
+
+ // Different media types.
+ EXPECT_TRUE(answer->ToString(&sdp));
+ const std::string kAudioMline = "m=audio";
+ const std::string kAudioMlineReplaceStr = "m=video";
+ rtc::replace_substrs(kAudioMline.c_str(), kAudioMline.length(),
+ kAudioMlineReplaceStr.c_str(),
+ kAudioMlineReplaceStr.length(),
+ &sdp);
+ SessionDescriptionInterface* modified_answer2 =
+ CreateSessionDescription(JsepSessionDescription::kAnswer, sdp, NULL);
+ SetRemoteDescriptionAnswerExpectError(kMlineMismatchInAnswer,
+ modified_answer2);
+
+ SetRemoteDescriptionWithoutError(answer.release());
+}
+
+// Verifying remote offer and local answer have matching m-lines as per
+// RFC 3264.
+TEST_F(WebRtcSessionTest, TestIncorrectMLinesInLocalAnswer) {
+ Init();
+ SendAudioVideoStream1();
+ SessionDescriptionInterface* offer = CreateRemoteOffer();
+ SetRemoteDescriptionWithoutError(offer);
+ SessionDescriptionInterface* answer = CreateAnswer();
+
+ cricket::SessionDescription* answer_copy = answer->description()->Copy();
+ answer_copy->RemoveContentByName("video");
+ JsepSessionDescription* modified_answer =
+ new JsepSessionDescription(JsepSessionDescription::kAnswer);
+
+ EXPECT_TRUE(modified_answer->Initialize(answer_copy,
+ answer->session_id(),
+ answer->session_version()));
+ SetLocalDescriptionAnswerExpectError(kMlineMismatchInAnswer, modified_answer);
+ SetLocalDescriptionWithoutError(answer);
+}
+
+TEST_F(WebRtcSessionTest, TestSessionContentError) {
+ Init();
+ SendAudioVideoStream1();
+ SessionDescriptionInterface* offer = CreateOffer();
+ const std::string session_id_orig = offer->session_id();
+ const std::string session_version_orig = offer->session_version();
+ SetLocalDescriptionWithoutError(offer);
+
+ video_channel_ = media_engine_->GetVideoChannel(0);
+ video_channel_->set_fail_set_send_codecs(true);
+
+ SessionDescriptionInterface* answer =
+ CreateRemoteAnswer(session_->local_description());
+ SetRemoteDescriptionAnswerExpectError("ERROR_CONTENT", answer);
+
+ // Test that after a content error, setting any description will
+ // result in an error.
+ video_channel_->set_fail_set_send_codecs(false);
+ answer = CreateRemoteAnswer(session_->local_description());
+ SetRemoteDescriptionExpectError("", "ERROR_CONTENT", answer);
+ offer = CreateRemoteOffer();
+ SetLocalDescriptionExpectError("", "ERROR_CONTENT", offer);
+}
+
TEST_F(WebRtcSessionTest, TestRtpDataChannel) {
configuration_.enable_rtp_data_channel = true;
Init();
@@ -1782,6 +2757,21 @@
last_data_channel_config_.open_handshake_role);
}
+TEST_F(WebRtcSessionTest, TestCombinedAudioVideoBweConstraint) {
+ configuration_.combined_audio_video_bwe = rtc::Optional<bool>(true);
+ Init();
+ SendAudioVideoStream1();
+ SessionDescriptionInterface* offer = CreateOffer();
+
+ SetLocalDescriptionWithoutError(offer);
+
+ voice_channel_ = media_engine_->GetVoiceChannel(0);
+
+ ASSERT_TRUE(voice_channel_ != NULL);
+ const cricket::AudioOptions& audio_options = voice_channel_->options();
+ EXPECT_EQ(rtc::Optional<bool>(true), audio_options.combined_audio_video_bwe);
+}
+
#ifdef HAVE_QUIC
TEST_P(WebRtcSessionTest, TestNegotiateQuic) {
configuration_.enable_quic = true;
@@ -1801,6 +2791,33 @@
}
#endif // HAVE_QUIC
+// Tests that RTX codec is removed from the answer when it isn't supported
+// by local side.
+TEST_F(WebRtcSessionTest, TestRtxRemovedByCreateAnswer) {
+ Init();
+ // Send video only to match the |kSdpWithRtx|.
+ SendVideoOnlyStream2();
+ std::string offer_sdp(kSdpWithRtx);
+
+ SessionDescriptionInterface* offer =
+ CreateSessionDescription(JsepSessionDescription::kOffer, offer_sdp, NULL);
+ EXPECT_TRUE(offer->ToString(&offer_sdp));
+
+ // Offer SDP contains the RTX codec.
+ EXPECT_TRUE(ContainsVideoCodecWithName(offer, "rtx"));
+ SetRemoteDescriptionWithoutError(offer);
+
+ // |offered_media_sections_| is used when creating answer.
+ offered_media_sections_.push_back(cricket::MediaDescriptionOptions(
+ cricket::MEDIA_TYPE_VIDEO, cricket::CN_VIDEO,
+ cricket::RtpTransceiverDirection(true, true), false));
+ // Don't create media section for audio in the answer.
+ SessionDescriptionInterface* answer = CreateAnswer();
+ // Answer SDP does not contain the RTX codec.
+ EXPECT_FALSE(ContainsVideoCodecWithName(answer, "rtx"));
+ SetLocalDescriptionWithoutError(answer);
+}
+
// This verifies that the voice channel after bundle has both options from video
// and voice channels.
TEST_F(WebRtcSessionTest, TestSetSocketOptionBeforeBundle) {
@@ -1849,6 +2866,34 @@
EXPECT_EQ(8000, option_val);
}
+// Test creating a session, request multiple offers, destroy the session
+// and make sure we got success/failure callbacks for all of the requests.
+// Background: crbug.com/507307
+TEST_F(WebRtcSessionTest, CreateOffersAndShutdown) {
+ Init();
+
+ rtc::scoped_refptr<WebRtcSessionCreateSDPObserverForTest> observers[100];
+ PeerConnectionInterface::RTCOfferAnswerOptions options;
+ options.offer_to_receive_audio =
+ RTCOfferAnswerOptions::kOfferToReceiveMediaTrue;
+ cricket::MediaSessionOptions session_options;
+ GetOptionsForOffer(options, &session_options);
+ for (auto& o : observers) {
+ o = new WebRtcSessionCreateSDPObserverForTest();
+ session_->CreateOffer(o, options, session_options);
+ }
+
+ session_.reset();
+
+ for (auto& o : observers) {
+ // We expect to have received a notification now even if the session was
+ // terminated. The offer creation may or may not have succeeded, but we
+ // must have received a notification which, so the only invalid state
+ // is kInit.
+ EXPECT_NE(WebRtcSessionCreateSDPObserverForTest::kInit, o->state());
+ }
+}
+
TEST_F(WebRtcSessionTest, TestPacketOptionsAndOnPacketSent) {
TestPacketOptions();
}