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Adds the support of RTCOfferOptions for PeerConnectionInterface::CreateOffer.

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Constraints are still supported for CreateOffer, but converted to RTCOfferOptions internally.

BUG=3282
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6822 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
jiayl@webrtc.org 2014-08-04 18:34:16 +00:00
parent b01ce14b13
commit b18bf5e47d
11 changed files with 350 additions and 173 deletions
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34
talk/app/webrtc/mediastreamsignaling.cc
View File

@ -127,6 +127,12 @@ static bool MediaContentDirectionHasSend(cricket::MediaContentDirection dir) {
return dir == cricket::MD_SENDONLY || dir == cricket::MD_SENDRECV;
}
static bool IsValidOfferToReceiveMedia(int value) {
typedef PeerConnectionInterface::RTCOfferAnswerOptions Options;
return (value >= Options::kUndefined) &&
(value <= Options::kMaxOfferToReceiveMedia);
}
// Factory class for creating remote MediaStreams and MediaStreamTracks.
class RemoteMediaStreamFactory {
public:
@ -363,14 +369,32 @@ void MediaStreamSignaling::RemoveLocalStream(
}
bool MediaStreamSignaling::GetOptionsForOffer(
const MediaConstraintsInterface* constraints,
cricket::MediaSessionOptions* options) {
UpdateSessionOptions();
if (!ParseConstraints(constraints, &options_, false)) {
const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
cricket::MediaSessionOptions* session_options) {
typedef PeerConnectionInterface::RTCOfferAnswerOptions RTCOfferAnswerOptions;
if (!IsValidOfferToReceiveMedia(rtc_options.offer_to_receive_audio) ||
!IsValidOfferToReceiveMedia(rtc_options.offer_to_receive_video)) {
return false;
}
UpdateSessionOptions();
// |options.has_audio| and |options.has_video| can only change from false to
// true, but never change from true to false. This is to make sure
// CreateOffer / CreateAnswer doesn't remove a media content
// description that has been created.
if (rtc_options.offer_to_receive_audio > 0) {
options_.has_audio = true;
}
if (rtc_options.offer_to_receive_video > 0) {
options_.has_video = true;
}
options_.vad_enabled = rtc_options.voice_activity_detection;
options_.transport_options.ice_restart = rtc_options.ice_restart;
options_.bundle_enabled = rtc_options.use_rtp_mux;
options_.bundle_enabled = EvaluateNeedForBundle(options_);
*options = options_;
*session_options = options_;
return true;
}

6
talk/app/webrtc/mediastreamsignaling.h
View File

@ -200,11 +200,11 @@ class MediaStreamSignaling : public sigslot::has_slots<> {
const rtc::Buffer& payload);
void RemoveSctpDataChannel(int sid);
// Returns a MediaSessionOptions struct with options decided by |constraints|,
// Returns a MediaSessionOptions struct with options decided by |options|,
// the local MediaStreams and DataChannels.
virtual bool GetOptionsForOffer(
const MediaConstraintsInterface* constraints,
cricket::MediaSessionOptions* options);
const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
cricket::MediaSessionOptions* session_options);
// Returns a MediaSessionOptions struct with options decided by
// |constraints|, the local MediaStreams and DataChannels.

137
talk/app/webrtc/mediastreamsignaling_unittest.cc
View File

@ -62,11 +62,14 @@ using webrtc::IceCandidateInterface;
using webrtc::MediaConstraintsInterface;
using webrtc::MediaStreamInterface;
using webrtc::MediaStreamTrackInterface;
using webrtc::PeerConnectionInterface;
using webrtc::SdpParseError;
using webrtc::SessionDescriptionInterface;
using webrtc::StreamCollection;
using webrtc::StreamCollectionInterface;
typedef PeerConnectionInterface::RTCOfferAnswerOptions RTCOfferAnswerOptions;
// Reference SDP with a MediaStream with label "stream1" and audio track with
// id "audio_1" and a video track with id "video_1;
static const char kSdpStringWithStream1[] =
@ -588,115 +591,137 @@ class MediaStreamSignalingTest: public testing::Test {
rtc::scoped_ptr<FakeDataChannelProvider> data_channel_provider_;
};
TEST_F(MediaStreamSignalingTest, GetOptionsForOfferWithInvalidAudioOption) {
RTCOfferAnswerOptions rtc_options;
rtc_options.offer_to_receive_audio = RTCOfferAnswerOptions::kUndefined - 1;
cricket::MediaSessionOptions options;
EXPECT_FALSE(signaling_->GetOptionsForOffer(rtc_options, &options));
rtc_options.offer_to_receive_audio =
RTCOfferAnswerOptions::kMaxOfferToReceiveMedia + 1;
EXPECT_FALSE(signaling_->GetOptionsForOffer(rtc_options, &options));
}
TEST_F(MediaStreamSignalingTest, GetOptionsForOfferWithInvalidVideoOption) {
RTCOfferAnswerOptions rtc_options;
rtc_options.offer_to_receive_video =
RTCOfferAnswerOptions::kUndefined - 1;
cricket::MediaSessionOptions options;
EXPECT_FALSE(signaling_->GetOptionsForOffer(rtc_options, &options));
rtc_options.offer_to_receive_video =
RTCOfferAnswerOptions::kMaxOfferToReceiveMedia + 1;
EXPECT_FALSE(signaling_->GetOptionsForOffer(rtc_options, &options));
}
// Test that a MediaSessionOptions is created for an offer if
// kOfferToReceiveAudio and kOfferToReceiveVideo constraints are set but no
// OfferToReceiveAudio and OfferToReceiveVideo options are set but no
// MediaStreams are sent.
TEST_F(MediaStreamSignalingTest, GetMediaSessionOptionsForOfferWithAudioVideo) {
FakeConstraints constraints;
constraints.SetMandatoryReceiveAudio(true);
constraints.SetMandatoryReceiveVideo(true);
RTCOfferAnswerOptions rtc_options;
rtc_options.offer_to_receive_audio = 1;
rtc_options.offer_to_receive_video = 1;
cricket::MediaSessionOptions options;
EXPECT_TRUE(signaling_->GetOptionsForOffer(&constraints, &options));
EXPECT_TRUE(signaling_->GetOptionsForOffer(rtc_options, &options));
EXPECT_TRUE(options.has_audio);
EXPECT_TRUE(options.has_video);
EXPECT_TRUE(options.bundle_enabled);
}
// Test that a correct MediaSessionOptions is created for an offer if
// kOfferToReceiveAudio constraints is set but no MediaStreams are sent.
// OfferToReceiveAudio is set but no MediaStreams are sent.
TEST_F(MediaStreamSignalingTest, GetMediaSessionOptionsForOfferWithAudio) {
FakeConstraints constraints;
constraints.SetMandatoryReceiveAudio(true);
RTCOfferAnswerOptions rtc_options;
rtc_options.offer_to_receive_audio = 1;
cricket::MediaSessionOptions options;
EXPECT_TRUE(signaling_->GetOptionsForOffer(&constraints, &options));
EXPECT_TRUE(signaling_->GetOptionsForOffer(rtc_options, &options));
EXPECT_TRUE(options.has_audio);
EXPECT_FALSE(options.has_video);
EXPECT_TRUE(options.bundle_enabled);
}
// Test that a correct MediaSessionOptions is created for an offer if
// no constraints or MediaStreams are sent.
// the default OfferOptons is used or MediaStreams are sent.
TEST_F(MediaStreamSignalingTest, GetDefaultMediaSessionOptionsForOffer) {
RTCOfferAnswerOptions rtc_options;
cricket::MediaSessionOptions options;
EXPECT_TRUE(signaling_->GetOptionsForOffer(NULL, &options));
EXPECT_TRUE(options.has_audio);
EXPECT_TRUE(signaling_->GetOptionsForOffer(rtc_options, &options));
EXPECT_FALSE(options.has_audio);
EXPECT_FALSE(options.has_video);
EXPECT_TRUE(options.bundle_enabled);
EXPECT_FALSE(options.bundle_enabled);
EXPECT_TRUE(options.vad_enabled);
EXPECT_FALSE(options.transport_options.ice_restart);
}
// Test that a correct MediaSessionOptions is created for an offer if
// kOfferToReceiveVideo constraints is set but no MediaStreams are sent.
// OfferToReceiveVideo is set but no MediaStreams are sent.
TEST_F(MediaStreamSignalingTest, GetMediaSessionOptionsForOfferWithVideo) {
FakeConstraints constraints;
constraints.SetMandatoryReceiveAudio(false);
constraints.SetMandatoryReceiveVideo(true);
RTCOfferAnswerOptions rtc_options;
rtc_options.offer_to_receive_audio = 0;
rtc_options.offer_to_receive_video = 1;
cricket::MediaSessionOptions options;
EXPECT_TRUE(signaling_->GetOptionsForOffer(&constraints, &options));
EXPECT_TRUE(signaling_->GetOptionsForOffer(rtc_options, &options));
EXPECT_FALSE(options.has_audio);
EXPECT_TRUE(options.has_video);
EXPECT_TRUE(options.bundle_enabled);
}
// Test that a correct MediaSessionOptions is created for an offer if
// kUseRtpMux constraints is set to false.
// UseRtpMux is set to false.
TEST_F(MediaStreamSignalingTest,
GetMediaSessionOptionsForOfferWithBundleDisabled) {
FakeConstraints constraints;
constraints.SetMandatoryReceiveAudio(true);
constraints.SetMandatoryReceiveVideo(true);
constraints.SetMandatoryUseRtpMux(false);
RTCOfferAnswerOptions rtc_options;
rtc_options.offer_to_receive_audio = 1;
rtc_options.offer_to_receive_video = 1;
rtc_options.use_rtp_mux = false;
cricket::MediaSessionOptions options;
EXPECT_TRUE(signaling_->GetOptionsForOffer(&constraints, &options));
EXPECT_TRUE(signaling_->GetOptionsForOffer(rtc_options, &options));
EXPECT_TRUE(options.has_audio);
EXPECT_TRUE(options.has_video);
EXPECT_FALSE(options.bundle_enabled);
}
// Test that a correct MediaSessionOptions is created to restart ice if
// kIceRestart constraints is set. It also tests that subsequent
// MediaSessionOptions don't have |transport_options.ice_restart| set.
// IceRestart is set. It also tests that subsequent MediaSessionOptions don't
// have |transport_options.ice_restart| set.
TEST_F(MediaStreamSignalingTest,
GetMediaSessionOptionsForOfferWithIceRestart) {
FakeConstraints constraints;
constraints.SetMandatoryIceRestart(true);
RTCOfferAnswerOptions rtc_options;
rtc_options.ice_restart = true;
cricket::MediaSessionOptions options;
EXPECT_TRUE(signaling_->GetOptionsForOffer(&constraints, &options));
EXPECT_TRUE(signaling_->GetOptionsForOffer(rtc_options, &options));
EXPECT_TRUE(options.transport_options.ice_restart);
EXPECT_TRUE(signaling_->GetOptionsForOffer(NULL, &options));
rtc_options = RTCOfferAnswerOptions();
EXPECT_TRUE(signaling_->GetOptionsForOffer(rtc_options, &options));
EXPECT_FALSE(options.transport_options.ice_restart);
}
// Test that GetMediaSessionOptionsForOffer and GetOptionsForAnswer work as
// expected if unknown constraints are used.
TEST_F(MediaStreamSignalingTest, GetMediaSessionOptionsWithBadConstraints) {
FakeConstraints mandatory;
mandatory.AddMandatory("bad_key", "bad_value");
cricket::MediaSessionOptions options;
EXPECT_FALSE(signaling_->GetOptionsForOffer(&mandatory, &options));
EXPECT_FALSE(signaling_->GetOptionsForAnswer(&mandatory, &options));
FakeConstraints optional;
optional.AddOptional("bad_key", "bad_value");
EXPECT_TRUE(signaling_->GetOptionsForOffer(&optional, &options));
EXPECT_TRUE(signaling_->GetOptionsForAnswer(&optional, &options));
}
// Test that a correct MediaSessionOptions are created for an offer if
// a MediaStream is sent and later updated with a new track.
// MediaConstraints are not used.
TEST_F(MediaStreamSignalingTest, AddTrackToLocalMediaStream) {
RTCOfferAnswerOptions rtc_options;
rtc::scoped_refptr<StreamCollection> local_streams(
CreateStreamCollection(1));
MediaStreamInterface* local_stream = local_streams->at(0);
EXPECT_TRUE(signaling_->AddLocalStream(local_stream));
cricket::MediaSessionOptions options;
EXPECT_TRUE(signaling_->GetOptionsForOffer(NULL, &options));
EXPECT_TRUE(signaling_->GetOptionsForOffer(rtc_options, &options));
VerifyMediaOptions(local_streams, options);
cricket::MediaSessionOptions updated_options;
local_stream->AddTrack(AudioTrack::Create(kAudioTracks[1], NULL));
EXPECT_TRUE(signaling_->GetOptionsForOffer(NULL, &options));
EXPECT_TRUE(signaling_->GetOptionsForOffer(rtc_options, &options));
VerifyMediaOptions(local_streams, options);
}
@ -714,21 +739,20 @@ TEST_F(MediaStreamSignalingTest, MediaConstraintsInAnswer) {
EXPECT_TRUE(answer_options.has_audio);
EXPECT_TRUE(answer_options.has_video);
FakeConstraints offer_c;
offer_c.SetMandatoryReceiveAudio(false);
offer_c.SetMandatoryReceiveVideo(false);
RTCOfferAnswerOptions rtc_offer_optoins;
cricket::MediaSessionOptions offer_options;
EXPECT_TRUE(signaling_->GetOptionsForOffer(&offer_c, &offer_options));
EXPECT_TRUE(
signaling_->GetOptionsForOffer(rtc_offer_optoins, &offer_options));
EXPECT_FALSE(offer_options.has_audio);
EXPECT_FALSE(offer_options.has_video);
FakeConstraints updated_offer_c;
updated_offer_c.SetMandatoryReceiveAudio(true);
updated_offer_c.SetMandatoryReceiveVideo(true);
RTCOfferAnswerOptions updated_rtc_offer_optoins;
updated_rtc_offer_optoins.offer_to_receive_audio = 1;
updated_rtc_offer_optoins.offer_to_receive_video = 1;
cricket::MediaSessionOptions updated_offer_options;
EXPECT_TRUE(signaling_->GetOptionsForOffer(&updated_offer_c,
EXPECT_TRUE(signaling_->GetOptionsForOffer(updated_rtc_offer_optoins,
&updated_offer_options));
EXPECT_TRUE(updated_offer_options.has_audio);
EXPECT_TRUE(updated_offer_options.has_video);
@ -748,7 +772,8 @@ TEST_F(MediaStreamSignalingTest, MediaConstraintsInAnswer) {
EXPECT_TRUE(updated_answer_options.has_audio);
EXPECT_TRUE(updated_answer_options.has_video);
EXPECT_TRUE(signaling_->GetOptionsForOffer(NULL,
RTCOfferAnswerOptions default_rtc_options;
EXPECT_TRUE(signaling_->GetOptionsForOffer(default_rtc_options,
&updated_offer_options));
EXPECT_TRUE(updated_offer_options.has_audio);
EXPECT_TRUE(updated_offer_options.has_video);

57
talk/app/webrtc/peerconnection.cc
View File

@ -508,7 +508,62 @@ void PeerConnection::CreateOffer(CreateSessionDescriptionObserver* observer,
LOG(LS_ERROR) << "CreateOffer - observer is NULL.";
return;
}
session_->CreateOffer(observer, constraints);
RTCOfferAnswerOptions options;
// Defaults to receiving audio and not receiving video.
options.offer_to_receive_audio =
RTCOfferAnswerOptions::kOfferToReceiveMediaTrue;
options.offer_to_receive_video = 0;
bool value;
size_t mandatory_constraints = 0;
if (FindConstraint(constraints,
MediaConstraintsInterface::kOfferToReceiveAudio,
&value,
&mandatory_constraints)) {
options.offer_to_receive_audio =
value ? RTCOfferAnswerOptions::kOfferToReceiveMediaTrue : 0;
}
if (FindConstraint(constraints,
MediaConstraintsInterface::kOfferToReceiveVideo,
&value,
&mandatory_constraints)) {
options.offer_to_receive_video =
value ? RTCOfferAnswerOptions::kOfferToReceiveMediaTrue : 0;
}
if (FindConstraint(constraints,
MediaConstraintsInterface::kVoiceActivityDetection,
&value,
&mandatory_constraints)) {
options.voice_activity_detection = value;
}
if (FindConstraint(constraints,
MediaConstraintsInterface::kIceRestart,
&value,
&mandatory_constraints)) {
options.ice_restart = value;
}
if (FindConstraint(constraints,
MediaConstraintsInterface::kUseRtpMux,
&value,
&mandatory_constraints)) {
options.use_rtp_mux = value;
}
CreateOffer(observer, options);
}
void PeerConnection::CreateOffer(CreateSessionDescriptionObserver* observer,
const RTCOfferAnswerOptions& options) {
if (!VERIFY(observer != NULL)) {
LOG(LS_ERROR) << "CreateOffer - observer is NULL.";
return;
}
session_->CreateOffer(observer, options);
}
void PeerConnection::CreateAnswer(

2
talk/app/webrtc/peerconnection.h
View File

@ -92,6 +92,8 @@ class PeerConnection : public PeerConnectionInterface,
// JSEP01
virtual void CreateOffer(CreateSessionDescriptionObserver* observer,
const MediaConstraintsInterface* constraints);
virtual void CreateOffer(CreateSessionDescriptionObserver* observer,
const RTCOfferAnswerOptions& options);
virtual void CreateAnswer(CreateSessionDescriptionObserver* observer,
const MediaConstraintsInterface* constraints);
virtual void SetLocalDescription(SetSessionDescriptionObserver* observer,

40
talk/app/webrtc/peerconnectioninterface.h
View File

@ -192,6 +192,38 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
explicit RTCConfiguration(IceTransportsType type) : type(type) {}
};
struct RTCOfferAnswerOptions {
static const int kUndefined = -1;
static const int kMaxOfferToReceiveMedia = 1;
// The default value for constraint offerToReceiveX:true.
static const int kOfferToReceiveMediaTrue = 1;
int offer_to_receive_video;
int offer_to_receive_audio;
bool voice_activity_detection;
bool ice_restart;
bool use_rtp_mux;
RTCOfferAnswerOptions()
: offer_to_receive_video(kUndefined),
offer_to_receive_audio(kUndefined),
voice_activity_detection(true),
ice_restart(false),
use_rtp_mux(true) {}
RTCOfferAnswerOptions(int offer_to_receive_video,
int offer_to_receive_audio,
bool voice_activity_detection,
bool ice_restart,
bool use_rtp_mux)
: offer_to_receive_video(offer_to_receive_video),
offer_to_receive_audio(offer_to_receive_audio),
voice_activity_detection(voice_activity_detection),
ice_restart(ice_restart),
use_rtp_mux(use_rtp_mux) {}
};
// Used by GetStats to decide which stats to include in the stats reports.
// |kStatsOutputLevelStandard| includes the standard stats for Javascript API;
// |kStatsOutputLevelDebug| includes both the standard stats and additional
@ -239,7 +271,13 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
// Create a new offer.
// The CreateSessionDescriptionObserver callback will be called when done.
virtual void CreateOffer(CreateSessionDescriptionObserver* observer,
const MediaConstraintsInterface* constraints) = 0;
const MediaConstraintsInterface* constraints) {}
// TODO(jiayl): remove the default impl and the old interface when chromium
// code is updated.
virtual void CreateOffer(CreateSessionDescriptionObserver* observer,
const RTCOfferAnswerOptions& options) {}
// Create an answer to an offer.
// The CreateSessionDescriptionObserver callback will be called when done.
virtual void CreateAnswer(CreateSessionDescriptionObserver* observer,

7
talk/app/webrtc/webrtcsession.cc
View File

@ -693,9 +693,10 @@ bool WebRtcSession::GetSslRole(rtc::SSLRole* role) {
return false;
}
void WebRtcSession::CreateOffer(CreateSessionDescriptionObserver* observer,
const MediaConstraintsInterface* constraints) {
webrtc_session_desc_factory_->CreateOffer(observer, constraints);
void WebRtcSession::CreateOffer(
CreateSessionDescriptionObserver* observer,
const PeerConnectionInterface::RTCOfferAnswerOptions& options) {
webrtc_session_desc_factory_->CreateOffer(observer, options);
}
void WebRtcSession::CreateAnswer(CreateSessionDescriptionObserver* observer,

5
talk/app/webrtc/webrtcsession.h
View File

@ -144,8 +144,9 @@ class WebRtcSession : public cricket::BaseSession,
// TODO - It may be necessary to supply error code as well.
sigslot::signal0<> SignalError;
void CreateOffer(CreateSessionDescriptionObserver* observer,
const MediaConstraintsInterface* constraints);
void CreateOffer(
CreateSessionDescriptionObserver* observer,
const PeerConnectionInterface::RTCOfferAnswerOptions& options);
void CreateAnswer(CreateSessionDescriptionObserver* observer,
const MediaConstraintsInterface* constraints);
// The ownership of |desc| will be transferred after this call.

217
talk/app/webrtc/webrtcsession_unittest.cc
View File

@ -99,6 +99,8 @@ using webrtc::kSdpWithoutSdesCrypto;
using webrtc::kSessionError;
using webrtc::kSessionErrorDesc;
typedef PeerConnectionInterface::RTCOfferAnswerOptions RTCOfferAnswerOptions;
static const int kClientAddrPort = 0;
static const char kClientAddrHost1[] = "11.11.11.11";
static const char kClientAddrHost2[] = "22.22.22.22";
@ -378,18 +380,26 @@ class WebRtcSessionTest : public testing::Test {
// Call mediastream_signaling_.UseOptionsWithStreamX() before this function
// to decide which streams to create.
void InitiateCall() {
SessionDescriptionInterface* offer = CreateOffer(NULL);
SessionDescriptionInterface* offer = CreateOffer();
SetLocalDescriptionWithoutError(offer);
EXPECT_TRUE_WAIT(PeerConnectionInterface::kIceGatheringNew !=
observer_.ice_gathering_state_,
kIceCandidatesTimeout);
}
SessionDescriptionInterface* CreateOffer() {
PeerConnectionInterface::RTCOfferAnswerOptions options;
options.offer_to_receive_audio =
RTCOfferAnswerOptions::kOfferToReceiveMediaTrue;
return CreateOffer(options);
}
SessionDescriptionInterface* CreateOffer(
const webrtc::MediaConstraintsInterface* constraints) {
const PeerConnectionInterface::RTCOfferAnswerOptions& options) {
rtc::scoped_refptr<WebRtcSessionCreateSDPObserverForTest>
observer = new WebRtcSessionCreateSDPObserverForTest();
session_->CreateOffer(observer, constraints);
session_->CreateOffer(observer, options);
EXPECT_TRUE_WAIT(
observer->state() != WebRtcSessionCreateSDPObserverForTest::kInit,
2000);
@ -833,9 +843,11 @@ class WebRtcSessionTest : public testing::Test {
AddInterface(rtc::SocketAddress(kClientAddrHost1, kClientAddrPort));
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
FakeConstraints constraints;
constraints.SetMandatoryUseRtpMux(bundle);
SessionDescriptionInterface* offer = CreateOffer(&constraints);
PeerConnectionInterface::RTCOfferAnswerOptions options;
options.use_rtp_mux = bundle;
SessionDescriptionInterface* offer = CreateOffer(options);
// SetLocalDescription and SetRemoteDescriptions takes ownership of offer
// and answer.
SetLocalDescriptionWithoutError(offer);
@ -905,7 +917,7 @@ class WebRtcSessionTest : public testing::Test {
AddInterface(rtc::SocketAddress(kClientAddrHost1, kClientAddrPort));
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
SessionDescriptionInterface* offer = CreateOffer(NULL);
SessionDescriptionInterface* offer = CreateOffer();
EXPECT_EQ(PeerConnectionInterface::kIceGatheringNew,
observer_.ice_gathering_state_);
@ -1005,7 +1017,7 @@ class WebRtcSessionTest : public testing::Test {
webrtc::InternalDataChannelInit dci;
dci.reliable = false;
session_->CreateDataChannel("datachannel", &dci);
SessionDescriptionInterface* offer = CreateOffer(NULL);
SessionDescriptionInterface* offer = CreateOffer();
SetLocalDescriptionWithoutError(offer);
}
@ -1021,13 +1033,14 @@ class WebRtcSessionTest : public testing::Test {
SetRemoteDescriptionWithoutError(offer.release());
}
PeerConnectionInterface::RTCOfferAnswerOptions options;
const int kNumber = 3;
rtc::scoped_refptr<WebRtcSessionCreateSDPObserverForTest>
observers[kNumber];
for (int i = 0; i < kNumber; ++i) {
observers[i] = new WebRtcSessionCreateSDPObserverForTest();
if (type == CreateSessionDescriptionRequest::kOffer) {
session_->CreateOffer(observers[i], NULL);
session_->CreateOffer(observers[i], options);
} else {
session_->CreateAnswer(observers[i], NULL);
}
@ -1140,7 +1153,7 @@ TEST_F(WebRtcSessionTest, SetSdpFailedOnInvalidSdp) {
TEST_F(WebRtcSessionTest, TestCreateSdesOfferReceiveSdesAnswer) {
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
SessionDescriptionInterface* offer = CreateOffer(NULL);
SessionDescriptionInterface* offer = CreateOffer();
const std::string session_id_orig = offer->session_id();
const std::string session_version_orig = offer->session_version();
SetLocalDescriptionWithoutError(offer);
@ -1166,7 +1179,7 @@ TEST_F(WebRtcSessionTest, TestCreateSdesOfferReceiveSdesAnswer) {
// Create new offer without send streams.
mediastream_signaling_.SendNothing();
offer = CreateOffer(NULL);
offer = CreateOffer();
// Verify the session id is the same and the session version is
// increased.
@ -1195,7 +1208,7 @@ TEST_F(WebRtcSessionTest, TestCreateSdesOfferReceiveSdesAnswer) {
TEST_F(WebRtcSessionTest, TestReceiveSdesOfferCreateSdesAnswer) {
Init(NULL);
mediastream_signaling_.SendAudioVideoStream2();
SessionDescriptionInterface* offer = CreateOffer(NULL);
SessionDescriptionInterface* offer = CreateOffer();
VerifyCryptoParams(offer->description());
SetRemoteDescriptionWithoutError(offer);
@ -1222,7 +1235,7 @@ TEST_F(WebRtcSessionTest, TestReceiveSdesOfferCreateSdesAnswer) {
EXPECT_TRUE(kAudioTrack1 == voice_channel_->send_streams()[0].id);
mediastream_signaling_.SendAudioVideoStream1And2();
offer = CreateOffer(NULL);
offer = CreateOffer();
SetRemoteDescriptionWithoutError(offer);
// Answer by turning off all send streams.
@ -1252,12 +1265,12 @@ TEST_F(WebRtcSessionTest, SetLocalSdpFailedOnCreateChannel) {
Init(NULL);
media_engine_->set_fail_create_channel(true);
SessionDescriptionInterface* offer = CreateOffer(NULL);
SessionDescriptionInterface* offer = CreateOffer();
ASSERT_TRUE(offer != NULL);
// SetRemoteDescription and SetLocalDescription will take the ownership of
// the offer.
SetRemoteDescriptionOfferExpectError(kCreateChannelFailed, offer);
offer = CreateOffer(NULL);
offer = CreateOffer();
ASSERT_TRUE(offer != NULL);
SetLocalDescriptionOfferExpectError(kCreateChannelFailed, offer);
}
@ -1374,7 +1387,7 @@ TEST_F(WebRtcSessionTest, TestCreateDtlsOfferReceiveDtlsAnswer) {
SetFactoryDtlsSrtp();
// Verify that we get a crypto fingerprint in the answer.
SessionDescriptionInterface* offer = CreateOffer(NULL);
SessionDescriptionInterface* offer = CreateOffer();
ASSERT_TRUE(offer != NULL);
VerifyFingerprintStatus(offer->description(), true);
// Check that we don't have an a=crypto line in the offer.
@ -1440,7 +1453,7 @@ TEST_F(WebRtcSessionTest, TestSetLocalNonDtlsAnswerWhenDtlsOn) {
TEST_F(WebRtcSessionTest, TestSetRemoteNonDtlsAnswerWhenDtlsOn) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
InitWithDtls();
SessionDescriptionInterface* offer = CreateOffer(NULL);
SessionDescriptionInterface* offer = CreateOffer();
cricket::MediaSessionOptions options;
options.has_video = true;
JsepSessionDescription* answer =
@ -1461,7 +1474,7 @@ TEST_F(WebRtcSessionTest, TestCreateOfferReceiveAnswerWithoutEncryption) {
InitWithDtls();
// Verify that we get a crypto fingerprint in the answer.
SessionDescriptionInterface* offer = CreateOffer(NULL);
SessionDescriptionInterface* offer = CreateOffer();
ASSERT_TRUE(offer != NULL);
VerifyFingerprintStatus(offer->description(), false);
// Check that we don't have an a=crypto line in the offer.
@ -1514,11 +1527,11 @@ TEST_F(WebRtcSessionTest, TestSetLocalOfferTwice) {
Init(NULL);
mediastream_signaling_.SendNothing();
// SetLocalDescription take ownership of offer.
SessionDescriptionInterface* offer = CreateOffer(NULL);
SessionDescriptionInterface* offer = CreateOffer();
SetLocalDescriptionWithoutError(offer);
// SetLocalDescription take ownership of offer.
SessionDescriptionInterface* offer2 = CreateOffer(NULL);
SessionDescriptionInterface* offer2 = CreateOffer();
SetLocalDescriptionWithoutError(offer2);
}
@ -1526,19 +1539,19 @@ TEST_F(WebRtcSessionTest, TestSetRemoteOfferTwice) {
Init(NULL);
mediastream_signaling_.SendNothing();
// SetLocalDescription take ownership of offer.
SessionDescriptionInterface* offer = CreateOffer(NULL);
SessionDescriptionInterface* offer = CreateOffer();
SetRemoteDescriptionWithoutError(offer);
SessionDescriptionInterface* offer2 = CreateOffer(NULL);
SessionDescriptionInterface* offer2 = CreateOffer();
SetRemoteDescriptionWithoutError(offer2);
}
TEST_F(WebRtcSessionTest, TestSetLocalAndRemoteOffer) {
Init(NULL);
mediastream_signaling_.SendNothing();
SessionDescriptionInterface* offer = CreateOffer(NULL);
SessionDescriptionInterface* offer = CreateOffer();
SetLocalDescriptionWithoutError(offer);
offer = CreateOffer(NULL);
offer = CreateOffer();
SetRemoteDescriptionOfferExpectError(
"Called in wrong state: STATE_SENTINITIATE", offer);
}
@ -1546,9 +1559,9 @@ TEST_F(WebRtcSessionTest, TestSetLocalAndRemoteOffer) {
TEST_F(WebRtcSessionTest, TestSetRemoteAndLocalOffer) {
Init(NULL);
mediastream_signaling_.SendNothing();
SessionDescriptionInterface* offer = CreateOffer(NULL);
SessionDescriptionInterface* offer = CreateOffer();
SetRemoteDescriptionWithoutError(offer);
offer = CreateOffer(NULL);
offer = CreateOffer();
SetLocalDescriptionOfferExpectError(
"Called in wrong state: STATE_RECEIVEDINITIATE", offer);
}
@ -1579,7 +1592,7 @@ TEST_F(WebRtcSessionTest, TestSetLocalPrAnswer) {
TEST_F(WebRtcSessionTest, TestSetRemotePrAnswer) {
Init(NULL);
mediastream_signaling_.SendNothing();
SessionDescriptionInterface* offer = CreateOffer(NULL);
SessionDescriptionInterface* offer = CreateOffer();
SetLocalDescriptionExpectState(offer, BaseSession::STATE_SENTINITIATE);
JsepSessionDescription* pranswer =
@ -1606,8 +1619,8 @@ TEST_F(WebRtcSessionTest, TestSetRemotePrAnswer) {
TEST_F(WebRtcSessionTest, TestSetLocalAnswerWithoutOffer) {
Init(NULL);
mediastream_signaling_.SendNothing();
rtc::scoped_ptr<SessionDescriptionInterface> offer(
CreateOffer(NULL));
rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
SessionDescriptionInterface* answer =
CreateRemoteAnswer(offer.get());
SetLocalDescriptionAnswerExpectError("Called in wrong state: STATE_INIT",
@ -1617,8 +1630,8 @@ TEST_F(WebRtcSessionTest, TestSetLocalAnswerWithoutOffer) {
TEST_F(WebRtcSessionTest, TestSetRemoteAnswerWithoutOffer) {
Init(NULL);
mediastream_signaling_.SendNothing();
rtc::scoped_ptr<SessionDescriptionInterface> offer(
CreateOffer(NULL));
rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
SessionDescriptionInterface* answer =
CreateRemoteAnswer(offer.get());
SetRemoteDescriptionAnswerExpectError(
@ -1636,7 +1649,7 @@ TEST_F(WebRtcSessionTest, TestAddRemoteCandidate) {
// Fail since we have not set a offer description.
EXPECT_FALSE(session_->ProcessIceMessage(&ice_candidate1));
SessionDescriptionInterface* offer = CreateOffer(NULL);
SessionDescriptionInterface* offer = CreateOffer();
SetLocalDescriptionWithoutError(offer);
// Candidate should be allowed to add before remote description.
EXPECT_TRUE(session_->ProcessIceMessage(&ice_candidate1));
@ -1770,7 +1783,7 @@ TEST_F(WebRtcSessionTest, TestSetRemoteSessionDescriptionWithCandidates) {
JsepIceCandidate ice_candidate(kMediaContentName0, kMediaContentIndex0,
candidate1);
mediastream_signaling_.SendAudioVideoStream1();
SessionDescriptionInterface* offer = CreateOffer(NULL);
SessionDescriptionInterface* offer = CreateOffer();
EXPECT_TRUE(offer->AddCandidate(&ice_candidate));
SetRemoteDescriptionWithoutError(offer);
@ -1805,8 +1818,8 @@ TEST_F(WebRtcSessionTest, TestSetLocalAndRemoteDescriptionWithCandidates) {
EXPECT_TRUE_WAIT(0u < observer_.mline_1_candidates_.size(),
kIceCandidatesTimeout);
rtc::scoped_ptr<SessionDescriptionInterface> local_offer(
CreateOffer(NULL));
rtc::scoped_ptr<SessionDescriptionInterface> local_offer(CreateOffer());
ASSERT_TRUE(local_offer->candidates(kMediaContentIndex0) != NULL);
EXPECT_LT(0u, local_offer->candidates(kMediaContentIndex0)->count());
ASSERT_TRUE(local_offer->candidates(kMediaContentIndex1) != NULL);
@ -1827,8 +1840,7 @@ TEST_F(WebRtcSessionTest, TestSetLocalAndRemoteDescriptionWithCandidates) {
TEST_F(WebRtcSessionTest, TestChannelCreationsWithContentNames) {
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
rtc::scoped_ptr<SessionDescriptionInterface> offer(
CreateOffer(NULL));
rtc::scoped_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 channel
@ -1871,8 +1883,8 @@ TEST_F(WebRtcSessionTest, TestChannelCreationsWithContentNames) {
// the send streams when no constraints have been set.
TEST_F(WebRtcSessionTest, CreateOfferWithoutConstraintsOrStreams) {
Init(NULL);
rtc::scoped_ptr<SessionDescriptionInterface> offer(
CreateOffer(NULL));
rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
ASSERT_TRUE(offer != NULL);
const cricket::ContentInfo* content =
cricket::GetFirstAudioContent(offer->description());
@ -1887,8 +1899,8 @@ TEST_F(WebRtcSessionTest, CreateOfferWithoutConstraints) {
Init(NULL);
// Test Audio only offer.
mediastream_signaling_.UseOptionsAudioOnly();
rtc::scoped_ptr<SessionDescriptionInterface> offer(
CreateOffer(NULL));
rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
const cricket::ContentInfo* content =
cricket::GetFirstAudioContent(offer->description());
EXPECT_TRUE(content != NULL);
@ -1897,7 +1909,7 @@ TEST_F(WebRtcSessionTest, CreateOfferWithoutConstraints) {
// Test Audio / Video offer.
mediastream_signaling_.SendAudioVideoStream1();
offer.reset(CreateOffer(NULL));
offer.reset(CreateOffer());
content = cricket::GetFirstAudioContent(offer->description());
EXPECT_TRUE(content != NULL);
content = cricket::GetFirstVideoContent(offer->description());
@ -1908,12 +1920,13 @@ TEST_F(WebRtcSessionTest, CreateOfferWithoutConstraints) {
// kOfferToReceiveVideo and kOfferToReceiveAudio constraints are set to false.
TEST_F(WebRtcSessionTest, CreateOfferWithConstraintsWithoutStreams) {
Init(NULL);
webrtc::FakeConstraints constraints_no_receive;
constraints_no_receive.SetMandatoryReceiveAudio(false);
constraints_no_receive.SetMandatoryReceiveVideo(false);
PeerConnectionInterface::RTCOfferAnswerOptions options;
options.offer_to_receive_audio = 0;
options.offer_to_receive_video = 0;
rtc::scoped_ptr<SessionDescriptionInterface> offer(
CreateOffer(&constraints_no_receive));
CreateOffer(options));
ASSERT_TRUE(offer != NULL);
const cricket::ContentInfo* content =
cricket::GetFirstAudioContent(offer->description());
@ -1926,10 +1939,12 @@ TEST_F(WebRtcSessionTest, CreateOfferWithConstraintsWithoutStreams) {
// kOfferToReceiveAudio constraints are set to true.
TEST_F(WebRtcSessionTest, CreateAudioOnlyOfferWithConstraints) {
Init(NULL);
webrtc::FakeConstraints constraints_audio_only;
constraints_audio_only.SetMandatoryReceiveAudio(true);
PeerConnectionInterface::RTCOfferAnswerOptions options;
options.offer_to_receive_audio =
RTCOfferAnswerOptions::kOfferToReceiveMediaTrue;
rtc::scoped_ptr<SessionDescriptionInterface> offer(
CreateOffer(&constraints_audio_only));
CreateOffer(options));
const cricket::ContentInfo* content =
cricket::GetFirstAudioContent(offer->description());
@ -1943,11 +1958,15 @@ TEST_F(WebRtcSessionTest, CreateAudioOnlyOfferWithConstraints) {
TEST_F(WebRtcSessionTest, CreateOfferWithConstraints) {
Init(NULL);
// Test Audio / Video offer.
webrtc::FakeConstraints constraints_audio_video;
constraints_audio_video.SetMandatoryReceiveAudio(true);
constraints_audio_video.SetMandatoryReceiveVideo(true);
PeerConnectionInterface::RTCOfferAnswerOptions options;
options.offer_to_receive_audio =
RTCOfferAnswerOptions::kOfferToReceiveMediaTrue;
options.offer_to_receive_video =
RTCOfferAnswerOptions::kOfferToReceiveMediaTrue;
rtc::scoped_ptr<SessionDescriptionInterface> offer(
CreateOffer(&constraints_audio_video));
CreateOffer(options));
const cricket::ContentInfo* content =
cricket::GetFirstAudioContent(offer->description());
@ -1963,7 +1982,7 @@ TEST_F(WebRtcSessionTest, CreateOfferWithConstraints) {
// an offer.
TEST_F(WebRtcSessionTest, CreateAnswerWithoutAnOffer) {
Init(NULL);
SessionDescriptionInterface* offer = CreateOffer(NULL);
SessionDescriptionInterface* offer = CreateOffer();
SetLocalDescriptionWithoutError(offer);
SessionDescriptionInterface* answer = CreateRemoteAnswer(offer);
SetRemoteDescriptionWithoutError(answer);
@ -2090,10 +2109,14 @@ TEST_F(WebRtcSessionTest, CreateAnswerWithConstraints) {
TEST_F(WebRtcSessionTest, CreateOfferWithoutCNCodecs) {
AddCNCodecs();
Init(NULL);
webrtc::FakeConstraints constraints;
constraints.SetOptionalVAD(false);
PeerConnectionInterface::RTCOfferAnswerOptions options;
options.offer_to_receive_audio =
RTCOfferAnswerOptions::kOfferToReceiveMediaTrue;
options.voice_activity_detection = false;
rtc::scoped_ptr<SessionDescriptionInterface> offer(
CreateOffer(&constraints));
CreateOffer(options));
const cricket::ContentInfo* content =
cricket::GetFirstAudioContent(offer->description());
EXPECT_TRUE(content != NULL);
@ -2125,7 +2148,7 @@ TEST_F(WebRtcSessionTest, TestAVOfferWithAudioOnlyAnswer) {
EXPECT_TRUE(media_engine_->GetVoiceChannel(0) == NULL);
mediastream_signaling_.SendAudioVideoStream1();
SessionDescriptionInterface* offer = CreateOffer(NULL);
SessionDescriptionInterface* offer = CreateOffer();
cricket::MediaSessionOptions options;
options.has_video = false;
@ -2182,7 +2205,7 @@ TEST_F(WebRtcSessionTest, TestAVOfferWithVideoOnlyAnswer) {
EXPECT_TRUE(media_engine_->GetVideoChannel(0) == NULL);
EXPECT_TRUE(media_engine_->GetVoiceChannel(0) == NULL);
mediastream_signaling_.SendAudioVideoStream1();
SessionDescriptionInterface* offer = CreateOffer(NULL);
SessionDescriptionInterface* offer = CreateOffer();
cricket::MediaSessionOptions options;
options.has_audio = false;
@ -2233,8 +2256,7 @@ TEST_F(WebRtcSessionTest, TestAVOfferWithVideoOnlyAnswer) {
TEST_F(WebRtcSessionTest, VerifyCryptoParamsInSDP) {
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
scoped_ptr<SessionDescriptionInterface> offer(
CreateOffer(NULL));
scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
VerifyCryptoParams(offer->description());
SetRemoteDescriptionWithoutError(offer.release());
scoped_ptr<SessionDescriptionInterface> answer(CreateAnswer(NULL));
@ -2245,8 +2267,7 @@ TEST_F(WebRtcSessionTest, VerifyNoCryptoParamsInSDP) {
options_.disable_encryption = true;
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
scoped_ptr<SessionDescriptionInterface> offer(
CreateOffer(NULL));
scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
VerifyNoCryptoParams(offer->description(), false);
}
@ -2265,7 +2286,8 @@ TEST_F(WebRtcSessionTest, VerifyAnswerFromCryptoOffer) {
TEST_F(WebRtcSessionTest, TestSetLocalDescriptionWithoutIce) {
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer(NULL));
rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
std::string sdp;
RemoveIceUfragPwdLines(offer.get(), &sdp);
SessionDescriptionInterface* modified_offer =
@ -2291,7 +2313,8 @@ TEST_F(WebRtcSessionTest, TestSetLocalDescriptionInvalidIceCredentials) {
Init(NULL);
tdesc_factory_->set_protocol(cricket::ICEPROTO_RFC5245);
mediastream_signaling_.SendAudioVideoStream1();
rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer(NULL));
rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
std::string sdp;
// Modifying ice ufrag and pwd in local offer with strings smaller than the
// recommended values of 4 and 22 bytes respectively.
@ -2340,8 +2363,8 @@ TEST_F(WebRtcSessionTest, VerifyBundleFlagInPA) {
Init(NULL);
EXPECT_TRUE((cricket::PORTALLOCATOR_ENABLE_BUNDLE &
allocator_->flags()) == cricket::PORTALLOCATOR_ENABLE_BUNDLE);
rtc::scoped_ptr<SessionDescriptionInterface> offer(
CreateOffer(NULL));
rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
cricket::SessionDescription* offer_copy =
offer->description()->Copy();
offer_copy->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
@ -2358,9 +2381,12 @@ TEST_F(WebRtcSessionTest, TestDisabledBundleInAnswer) {
mediastream_signaling_.SendAudioVideoStream1();
EXPECT_TRUE((cricket::PORTALLOCATOR_ENABLE_BUNDLE &
allocator_->flags()) == cricket::PORTALLOCATOR_ENABLE_BUNDLE);
FakeConstraints constraints;
constraints.SetMandatoryUseRtpMux(true);
SessionDescriptionInterface* offer = CreateOffer(&constraints);
PeerConnectionInterface::RTCOfferAnswerOptions options;
options.use_rtp_mux = true;
SessionDescriptionInterface* offer = CreateOffer(options);
SetLocalDescriptionWithoutError(offer);
mediastream_signaling_.SendAudioVideoStream2();
rtc::scoped_ptr<SessionDescriptionInterface> answer(
@ -2396,9 +2422,11 @@ TEST_F(WebRtcSessionTest, TestDisabledRtcpMuxWithBundleEnabled) {
mediastream_signaling_.SendAudioVideoStream1();
EXPECT_TRUE((cricket::PORTALLOCATOR_ENABLE_BUNDLE &
allocator_->flags()) == cricket::PORTALLOCATOR_ENABLE_BUNDLE);
FakeConstraints constraints;
constraints.SetMandatoryUseRtpMux(true);
SessionDescriptionInterface* offer = CreateOffer(&constraints);
PeerConnectionInterface::RTCOfferAnswerOptions options;
options.use_rtp_mux = true;
SessionDescriptionInterface* offer = CreateOffer(options);
std::string offer_str;
offer->ToString(&offer_str);
// Disable rtcp-mux
@ -2568,7 +2596,7 @@ TEST_F(WebRtcSessionTest, InsertDtmf) {
TEST_F(WebRtcSessionTest, TestInitiatorFlagAsOriginator) {
Init(NULL);
EXPECT_FALSE(session_->initiator());
SessionDescriptionInterface* offer = CreateOffer(NULL);
SessionDescriptionInterface* offer = CreateOffer();
SessionDescriptionInterface* answer = CreateRemoteAnswer(offer);
SetLocalDescriptionWithoutError(offer);
EXPECT_TRUE(session_->initiator());
@ -2594,7 +2622,7 @@ TEST_F(WebRtcSessionTest, TestInitiatorFlagAsReceiver) {
TEST_F(WebRtcSessionTest, TestInitiatorGIceInAnswer) {
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
SessionDescriptionInterface* offer = CreateOffer(NULL);
SessionDescriptionInterface* offer = CreateOffer();
rtc::scoped_ptr<SessionDescriptionInterface> answer(
CreateRemoteAnswer(offer));
SetLocalDescriptionWithoutError(offer);
@ -2616,7 +2644,7 @@ TEST_F(WebRtcSessionTest, TestInitiatorGIceInAnswer) {
TEST_F(WebRtcSessionTest, TestInitiatorIceInAnswer) {
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
SessionDescriptionInterface* offer = CreateOffer(NULL);
SessionDescriptionInterface* offer = CreateOffer();
SessionDescriptionInterface* answer = CreateRemoteAnswer(offer);
SetLocalDescriptionWithoutError(offer);
@ -2630,7 +2658,7 @@ TEST_F(WebRtcSessionTest, TestInitiatorIceInAnswer) {
TEST_F(WebRtcSessionTest, TestReceiverGIceInOffer) {
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
SessionDescriptionInterface* offer = CreateOffer(NULL);
SessionDescriptionInterface* offer = CreateOffer();
SetRemoteDescriptionWithoutError(offer);
rtc::scoped_ptr<SessionDescriptionInterface> answer(
CreateAnswer(NULL));
@ -2652,7 +2680,7 @@ TEST_F(WebRtcSessionTest, TestReceiverGIceInOffer) {
TEST_F(WebRtcSessionTest, TestReceiverIceInOffer) {
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
SessionDescriptionInterface* offer = CreateOffer(NULL);
SessionDescriptionInterface* offer = CreateOffer();
SetRemoteDescriptionWithoutError(offer);
SessionDescriptionInterface* answer = CreateAnswer(NULL);
SetLocalDescriptionWithoutError(answer);
@ -2665,8 +2693,8 @@ TEST_F(WebRtcSessionTest, TestReceiverIceInOffer) {
TEST_F(WebRtcSessionTest, TestIceOfferGIceOnlyAnswer) {
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
rtc::scoped_ptr<SessionDescriptionInterface> offer(
CreateOffer(NULL));
rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
std::string offer_str;
offer->ToString(&offer_str);
// Disable google-ice
@ -2697,7 +2725,7 @@ TEST_F(WebRtcSessionTest, TestIceOfferGIceOnlyAnswer) {
TEST_F(WebRtcSessionTest, TestIncorrectMLinesInRemoteAnswer) {
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
SessionDescriptionInterface* offer = CreateOffer(NULL);
SessionDescriptionInterface* offer = CreateOffer();
SetLocalDescriptionWithoutError(offer);
rtc::scoped_ptr<SessionDescriptionInterface> answer(
CreateRemoteAnswer(session_->local_description()));
@ -2798,8 +2826,7 @@ TEST_F(WebRtcSessionTest, TestIceStartAfterSetLocalDescriptionOnly) {
TEST_F(WebRtcSessionTest, TestCryptoAfterSetLocalDescription) {
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
rtc::scoped_ptr<SessionDescriptionInterface> offer(
CreateOffer(NULL));
rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
// Making sure SetLocalDescription correctly sets crypto value in
// SessionDescription object after de-serialization of sdp string. The value
@ -2818,8 +2845,7 @@ TEST_F(WebRtcSessionTest, TestCryptoAfterSetLocalDescriptionWithDisabled) {
options_.disable_encryption = true;
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
rtc::scoped_ptr<SessionDescriptionInterface> offer(
CreateOffer(NULL));
rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
// Making sure SetLocalDescription correctly sets crypto value in
// SessionDescription object after de-serialization of sdp string. The value
@ -2900,7 +2926,7 @@ TEST_F(WebRtcSessionTest, TestCreateAnswerWithOldUfragAndPassword) {
TEST_F(WebRtcSessionTest, TestSessionContentError) {
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
SessionDescriptionInterface* offer = CreateOffer(NULL);
SessionDescriptionInterface* offer = CreateOffer();
const std::string session_id_orig = offer->session_id();
const std::string session_version_orig = offer->session_version();
SetLocalDescriptionWithoutError(offer);
@ -2985,7 +3011,7 @@ TEST_F(WebRtcSessionTest, TestCreateOfferWithSctpEnabledWithoutStreams) {
InitWithDtls();
rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer(NULL));
rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
EXPECT_TRUE(offer->description()->GetContentByName("data") == NULL);
EXPECT_TRUE(offer->description()->GetTransportInfoByName("data") == NULL);
}
@ -3099,7 +3125,8 @@ TEST_F(WebRtcSessionTest, TestCreateOfferBeforeIdentityRequestReturnSuccess) {
EXPECT_TRUE(session_->waiting_for_identity());
mediastream_signaling_.SendAudioVideoStream1();
rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer(NULL));
rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
EXPECT_TRUE(offer != NULL);
VerifyNoCryptoParams(offer->description(), true);
VerifyFingerprintStatus(offer->description(), true);
@ -3132,7 +3159,8 @@ TEST_F(WebRtcSessionTest, TestCreateOfferAfterIdentityRequestReturnSuccess) {
InitWithDtls();
EXPECT_TRUE_WAIT(!session_->waiting_for_identity(), 1000);
rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer(NULL));
rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
EXPECT_TRUE(offer != NULL);
}
@ -3143,7 +3171,8 @@ TEST_F(WebRtcSessionTest, TestCreateOfferAfterIdentityRequestReturnFailure) {
InitWithDtls(true);
EXPECT_TRUE_WAIT(!session_->waiting_for_identity(), 1000);
rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer(NULL));
rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
EXPECT_TRUE(offer == NULL);
}
@ -3211,7 +3240,7 @@ TEST_F(WebRtcSessionTest, TestDscpConstraint) {
webrtc::MediaConstraintsInterface::kEnableDscp, true);
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
SessionDescriptionInterface* offer = CreateOffer(NULL);
SessionDescriptionInterface* offer = CreateOffer();
SetLocalDescriptionWithoutError(offer);
@ -3237,7 +3266,7 @@ TEST_F(WebRtcSessionTest, TestSuspendBelowMinBitrateConstraint) {
true);
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
SessionDescriptionInterface* offer = CreateOffer(NULL);
SessionDescriptionInterface* offer = CreateOffer();
SetLocalDescriptionWithoutError(offer);
@ -3258,7 +3287,7 @@ TEST_F(WebRtcSessionTest, TestRenegotiateNewMediaWithCandidatesInSdp) {
SetFactoryDtlsSrtp();
mediastream_signaling_.UseOptionsAudioOnly();
SessionDescriptionInterface* offer = CreateOffer(NULL);
SessionDescriptionInterface* offer = CreateOffer();
SetLocalDescriptionWithoutError(offer);
SessionDescriptionInterface* answer = CreateRemoteAnswer(offer);
@ -3288,7 +3317,7 @@ TEST_F(WebRtcSessionTest, TestRenegotiateNewMediaWithCandidatesSeparated) {
SetFactoryDtlsSrtp();
mediastream_signaling_.UseOptionsAudioOnly();
SessionDescriptionInterface* offer = CreateOffer(NULL);
SessionDescriptionInterface* offer = CreateOffer();
SetLocalDescriptionWithoutError(offer);
SessionDescriptionInterface* answer = CreateRemoteAnswer(offer);

16
talk/app/webrtc/webrtcsessiondescriptionfactory.cc
View File

@ -166,8 +166,9 @@ WebRtcSessionDescriptionFactory::~WebRtcSessionDescriptionFactory() {
void WebRtcSessionDescriptionFactory::CreateOffer(
CreateSessionDescriptionObserver* observer,
const MediaConstraintsInterface* constraints) {
cricket::MediaSessionOptions options;
const PeerConnectionInterface::RTCOfferAnswerOptions& options) {
cricket::MediaSessionOptions session_options;
std::string error = "CreateOffer";
if (identity_request_state_ == IDENTITY_FAILED) {
error += kFailedDueToIdentityFailed;
@ -176,14 +177,15 @@ void WebRtcSessionDescriptionFactory::CreateOffer(
return;
}
if (!mediastream_signaling_->GetOptionsForOffer(constraints, &options)) {
error += " called with invalid constraints.";
if (!mediastream_signaling_->GetOptionsForOffer(options,
&session_options)) {
error += " called with invalid options.";
LOG(LS_ERROR) << error;
PostCreateSessionDescriptionFailed(observer, error);
return;
}
if (!ValidStreams(options.streams)) {
if (!ValidStreams(session_options.streams)) {
error += " called with invalid media streams.";
LOG(LS_ERROR) << error;
PostCreateSessionDescriptionFailed(observer, error);
@ -192,11 +194,11 @@ void WebRtcSessionDescriptionFactory::CreateOffer(
if (data_channel_type_ == cricket::DCT_SCTP &&
mediastream_signaling_->HasDataChannels()) {
options.data_channel_type = cricket::DCT_SCTP;
session_options.data_channel_type = cricket::DCT_SCTP;
}
CreateSessionDescriptionRequest request(
CreateSessionDescriptionRequest::kOffer, observer, options);
CreateSessionDescriptionRequest::kOffer, observer, session_options);
if (identity_request_state_ == IDENTITY_WAITING) {
create_session_description_requests_.push(request);
} else {

2
talk/app/webrtc/webrtcsessiondescriptionfactory.h
View File

@ -107,7 +107,7 @@ class WebRtcSessionDescriptionFactory : public rtc::MessageHandler,
void CreateOffer(
CreateSessionDescriptionObserver* observer,
const MediaConstraintsInterface* constraints);
const PeerConnectionInterface::RTCOfferAnswerOptions& options);
void CreateAnswer(
CreateSessionDescriptionObserver* observer,
const MediaConstraintsInterface* constraints);