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Fixing some issues with the direction attribute of m-lines in offers.

Browse Source 
By default, we'll now offer to receive if already receiving
(meaning that the last remote description contained a track).

Also, m-lines that are neither receiving nor sending are now correctly
marked "inactive".

Also moved some logic relating to default tracks out of webrtcsdp.cc,
such that now the direction seen by upper layers will always be
consistent with the consumed/produced SDP.

BUG=528089

Review URL: https://codereview.webrtc.org/1406803004

Cr-Commit-Position: refs/heads/master@{#10376}
This commit is contained in:
deadbeef 2015-10-22 13:14:45 -07:00 committed by Commit bot
parent 56149e57bb
commit c80741f895
8 changed files with 216 additions and 112 deletions
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57
talk/app/webrtc/peerconnection.cc
View File

@ -463,27 +463,16 @@ bool ConvertRtcOptionsForOffer(
return false;
}
// According to the spec, offer to receive audio/video if the constraint is
// not set and there are send streams.
if (rtc_options.offer_to_receive_audio == RTCOfferAnswerOptions::kUndefined) {
session_options->recv_audio =
session_options->HasSendMediaStream(cricket::MEDIA_TYPE_AUDIO);
} else {
if (rtc_options.offer_to_receive_audio != RTCOfferAnswerOptions::kUndefined) {
session_options->recv_audio = (rtc_options.offer_to_receive_audio > 0);
}
if (rtc_options.offer_to_receive_video == RTCOfferAnswerOptions::kUndefined) {
session_options->recv_video =
session_options->HasSendMediaStream(cricket::MEDIA_TYPE_VIDEO);
} else {
if (rtc_options.offer_to_receive_video != RTCOfferAnswerOptions::kUndefined) {
session_options->recv_video = (rtc_options.offer_to_receive_video > 0);
}
session_options->vad_enabled = rtc_options.voice_activity_detection;
session_options->transport_options.ice_restart = rtc_options.ice_restart;
session_options->bundle_enabled =
rtc_options.use_rtp_mux &&
(session_options->has_audio() || session_options->has_video() ||
session_options->has_data());
session_options->bundle_enabled = rtc_options.use_rtp_mux;
return true;
}
@ -525,10 +514,6 @@ bool ParseConstraintsForAnswer(const MediaConstraintsInterface* constraints,
// kUseRtpMux defaults to true according to spec.
session_options->bundle_enabled = true;
}
session_options->bundle_enabled =
session_options->bundle_enabled &&
(session_options->has_audio() || session_options->has_video() ||
session_options->has_data());
if (FindConstraint(constraints, MediaConstraintsInterface::kIceRestart,
&value, &mandatory_constraints_satisfied)) {
@ -1043,8 +1028,8 @@ void PeerConnection::SetRemoteDescription(
audio_content->description);
UpdateRemoteStreamsList(GetActiveStreams(desc), desc->type(), new_streams);
remote_info_.default_audio_track_needed =
MediaContentDirectionHasSend(desc->direction()) &&
desc->streams().empty();
!remote_desc->msid_supported() && desc->streams().empty() &&
MediaContentDirectionHasSend(desc->direction());
}
// Find all video rtp streams and create corresponding remote VideoTracks
@ -1056,8 +1041,8 @@ void PeerConnection::SetRemoteDescription(
video_content->description);
UpdateRemoteStreamsList(GetActiveStreams(desc), desc->type(), new_streams);
remote_info_.default_video_track_needed =
MediaContentDirectionHasSend(desc->direction()) &&
desc->streams().empty();
!remote_desc->msid_supported() && desc->streams().empty() &&
MediaContentDirectionHasSend(desc->direction());
}
// Update the DataChannels with the information from the remote peer.
@ -1396,12 +1381,28 @@ void PeerConnection::PostCreateSessionDescriptionFailure(
bool PeerConnection::GetOptionsForOffer(
const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
cricket::MediaSessionOptions* session_options) {
SetStreams(session_options, local_streams_, rtp_data_channels_);
if (!ConvertRtcOptionsForOffer(rtc_options, session_options)) {
return false;
}
SetStreams(session_options, local_streams_, rtp_data_channels_);
// Offer to receive audio/video if the constraint is not set and there are
// send streams, or we're currently receiving.
if (rtc_options.offer_to_receive_audio == RTCOfferAnswerOptions::kUndefined) {
session_options->recv_audio =
session_options->HasSendMediaStream(cricket::MEDIA_TYPE_AUDIO) ||
!remote_audio_tracks_.empty();
}
if (rtc_options.offer_to_receive_video == RTCOfferAnswerOptions::kUndefined) {
session_options->recv_video =
session_options->HasSendMediaStream(cricket::MEDIA_TYPE_VIDEO) ||
!remote_video_tracks_.empty();
}
session_options->bundle_enabled =
session_options->bundle_enabled &&
(session_options->has_audio() || session_options->has_video() ||
session_options->has_data());
if (session_->data_channel_type() == cricket::DCT_SCTP && HasDataChannels()) {
session_options->data_channel_type = cricket::DCT_SCTP;
}
@ -1411,14 +1412,18 @@ bool PeerConnection::GetOptionsForOffer(
bool PeerConnection::GetOptionsForAnswer(
const MediaConstraintsInterface* constraints,
cricket::MediaSessionOptions* session_options) {
SetStreams(session_options, local_streams_, rtp_data_channels_);
session_options->recv_audio = false;
session_options->recv_video = false;
if (!ParseConstraintsForAnswer(constraints, session_options)) {
return false;
}
SetStreams(session_options, local_streams_, rtp_data_channels_);
session_options->bundle_enabled =
session_options->bundle_enabled &&
(session_options->has_audio() || session_options->has_video() ||
session_options->has_data());
// RTP data channel is handled in MediaSessionOptions::AddStream. SCTP streams
// are not signaled in the SDP so does not go through that path and must be
// handled here.

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

@ -51,8 +51,6 @@ typedef std::vector<PortAllocatorFactoryInterface::TurnConfiguration>
// Populates |session_options| from |rtc_options|, and returns true if options
// are valid.
// Send streams should already be added to |session_options| before this method
// is called, as this affects the values of recv_audio and recv_video.
bool ConvertRtcOptionsForOffer(
const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
cricket::MediaSessionOptions* session_options);

166
talk/app/webrtc/peerconnectioninterface_unittest.cc
View File

@ -260,6 +260,7 @@ using webrtc::DataChannelInterface;
using webrtc::FakeConstraints;
using webrtc::FakePortAllocatorFactory;
using webrtc::IceCandidateInterface;
using webrtc::MediaConstraintsInterface;
using webrtc::MediaStream;
using webrtc::MediaStreamInterface;
using webrtc::MediaStreamTrackInterface;
@ -658,26 +659,30 @@ class PeerConnectionInterfaceTest : public testing::Test {
observer_.renegotiation_needed_ = false;
}
bool DoCreateOfferAnswer(SessionDescriptionInterface** desc, bool offer) {
bool DoCreateOfferAnswer(SessionDescriptionInterface** desc,
bool offer,
MediaConstraintsInterface* constraints) {
rtc::scoped_refptr<MockCreateSessionDescriptionObserver>
observer(new rtc::RefCountedObject<
MockCreateSessionDescriptionObserver>());
if (offer) {
pc_->CreateOffer(observer, NULL);
pc_->CreateOffer(observer, constraints);
} else {
pc_->CreateAnswer(observer, NULL);
pc_->CreateAnswer(observer, constraints);
}
EXPECT_EQ_WAIT(true, observer->called(), kTimeout);
*desc = observer->release_desc();
return observer->result();
}
bool DoCreateOffer(SessionDescriptionInterface** desc) {
return DoCreateOfferAnswer(desc, true);
bool DoCreateOffer(SessionDescriptionInterface** desc,
MediaConstraintsInterface* constraints) {
return DoCreateOfferAnswer(desc, true, constraints);
}
bool DoCreateAnswer(SessionDescriptionInterface** desc) {
return DoCreateOfferAnswer(desc, false);
bool DoCreateAnswer(SessionDescriptionInterface** desc,
MediaConstraintsInterface* constraints) {
return DoCreateOfferAnswer(desc, false, constraints);
}
bool DoSetSessionDescription(SessionDescriptionInterface* desc, bool local) {
@ -737,7 +742,7 @@ class PeerConnectionInterfaceTest : public testing::Test {
void CreateOfferAsRemoteDescription() {
rtc::scoped_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(offer.use()));
ASSERT_TRUE(DoCreateOffer(offer.use(), nullptr));
std::string sdp;
EXPECT_TRUE(offer->ToString(&sdp));
SessionDescriptionInterface* remote_offer =
@ -757,7 +762,7 @@ class PeerConnectionInterfaceTest : public testing::Test {
void CreateAnswerAsLocalDescription() {
scoped_ptr<SessionDescriptionInterface> answer;
ASSERT_TRUE(DoCreateAnswer(answer.use()));
ASSERT_TRUE(DoCreateAnswer(answer.use(), nullptr));
// TODO(perkj): Currently SetLocalDescription fails if any parameters in an
// audio codec change, even if the parameter has nothing to do with
@ -777,7 +782,7 @@ class PeerConnectionInterfaceTest : public testing::Test {
void CreatePrAnswerAsLocalDescription() {
scoped_ptr<SessionDescriptionInterface> answer;
ASSERT_TRUE(DoCreateAnswer(answer.use()));
ASSERT_TRUE(DoCreateAnswer(answer.use(), nullptr));
std::string sdp;
EXPECT_TRUE(answer->ToString(&sdp));
@ -797,7 +802,7 @@ class PeerConnectionInterfaceTest : public testing::Test {
void CreateOfferAsLocalDescription() {
rtc::scoped_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(offer.use()));
ASSERT_TRUE(DoCreateOffer(offer.use(), nullptr));
// TODO(perkj): Currently SetLocalDescription fails if any parameters in an
// audio codec change, even if the parameter has nothing to do with
// receiving. Not all parameters are serialized to SDP.
@ -967,7 +972,7 @@ TEST_F(PeerConnectionInterfaceTest, AddedStreamsPresentInOffer) {
CreatePeerConnection();
AddAudioVideoStream(kStreamLabel1, "audio_track", "video_track");
scoped_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(offer.accept()));
ASSERT_TRUE(DoCreateOffer(offer.accept(), nullptr));
const cricket::ContentInfo* audio_content =
cricket::GetFirstAudioContent(offer->description());
@ -988,7 +993,7 @@ TEST_F(PeerConnectionInterfaceTest, AddedStreamsPresentInOffer) {
// Add another stream and ensure the offer includes both the old and new
// streams.
AddAudioVideoStream(kStreamLabel2, "audio_track2", "video_track2");
ASSERT_TRUE(DoCreateOffer(offer.accept()));
ASSERT_TRUE(DoCreateOffer(offer.accept(), nullptr));
audio_content = cricket::GetFirstAudioContent(offer->description());
audio_desc = static_cast<const cricket::AudioContentDescription*>(
@ -1084,12 +1089,12 @@ TEST_F(PeerConnectionInterfaceTest, IceCandidates) {
// SetRemoteDescription takes ownership of offer.
SessionDescriptionInterface* offer = NULL;
AddVideoStream(kStreamLabel1);
EXPECT_TRUE(DoCreateOffer(&offer));
EXPECT_TRUE(DoCreateOffer(&offer, nullptr));
EXPECT_TRUE(DoSetRemoteDescription(offer));
// SetLocalDescription takes ownership of answer.
SessionDescriptionInterface* answer = NULL;
EXPECT_TRUE(DoCreateAnswer(&answer));
EXPECT_TRUE(DoCreateAnswer(&answer, nullptr));
EXPECT_TRUE(DoSetLocalDescription(answer));
EXPECT_TRUE_WAIT(observer_.last_candidate_.get() != NULL, kTimeout);
@ -1104,7 +1109,7 @@ TEST_F(PeerConnectionInterfaceTest, CreateOfferAnswerWithInvalidStream) {
CreatePeerConnection();
// Create a regular offer for the CreateAnswer test later.
SessionDescriptionInterface* offer = NULL;
EXPECT_TRUE(DoCreateOffer(&offer));
EXPECT_TRUE(DoCreateOffer(&offer, nullptr));
EXPECT_TRUE(offer != NULL);
delete offer;
offer = NULL;
@ -1113,11 +1118,11 @@ TEST_F(PeerConnectionInterfaceTest, CreateOfferAnswerWithInvalidStream) {
AddAudioVideoStream(kStreamLabel1, "track_label", "track_label");
// Test CreateOffer
EXPECT_FALSE(DoCreateOffer(&offer));
EXPECT_FALSE(DoCreateOffer(&offer, nullptr));
// Test CreateAnswer
SessionDescriptionInterface* answer = NULL;
EXPECT_FALSE(DoCreateAnswer(&answer));
EXPECT_FALSE(DoCreateAnswer(&answer, nullptr));
}
// Test that we will get different SSRCs for each tracks in the offer and answer
@ -1129,7 +1134,7 @@ TEST_F(PeerConnectionInterfaceTest, SsrcInOfferAnswer) {
// Test CreateOffer
scoped_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(offer.use()));
ASSERT_TRUE(DoCreateOffer(offer.use(), nullptr));
int audio_ssrc = 0;
int video_ssrc = 0;
EXPECT_TRUE(GetFirstSsrc(GetFirstAudioContent(offer->description()),
@ -1141,7 +1146,7 @@ TEST_F(PeerConnectionInterfaceTest, SsrcInOfferAnswer) {
// Test CreateAnswer
EXPECT_TRUE(DoSetRemoteDescription(offer.release()));
scoped_ptr<SessionDescriptionInterface> answer;
ASSERT_TRUE(DoCreateAnswer(answer.use()));
ASSERT_TRUE(DoCreateAnswer(answer.use(), nullptr));
audio_ssrc = 0;
video_ssrc = 0;
EXPECT_TRUE(GetFirstSsrc(GetFirstAudioContent(answer->description()),
@ -1588,6 +1593,73 @@ TEST_F(PeerConnectionInterfaceTest, ReceiveUpdatedAudioOfferWithBadCodecs) {
CreateAnswerAsLocalDescription();
}
// Test that if we're receiving (but not sending) a track, subsequent offers
// will have m-lines with a=recvonly.
TEST_F(PeerConnectionInterfaceTest, CreateSubsequentRecvOnlyOffer) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
CreatePeerConnection(&constraints);
CreateAndSetRemoteOffer(kSdpStringWithStream1);
CreateAnswerAsLocalDescription();
// At this point we should be receiving stream 1, but not sending anything.
// A new offer should be recvonly.
SessionDescriptionInterface* offer;
DoCreateOffer(&offer, nullptr);
const cricket::ContentInfo* video_content =
cricket::GetFirstVideoContent(offer->description());
const cricket::VideoContentDescription* video_desc =
static_cast<const cricket::VideoContentDescription*>(
video_content->description);
ASSERT_EQ(cricket::MD_RECVONLY, video_desc->direction());
const cricket::ContentInfo* audio_content =
cricket::GetFirstAudioContent(offer->description());
const cricket::AudioContentDescription* audio_desc =
static_cast<const cricket::AudioContentDescription*>(
audio_content->description);
ASSERT_EQ(cricket::MD_RECVONLY, audio_desc->direction());
}
// Test that if we're receiving (but not sending) a track, and the
// offerToReceiveVideo/offerToReceiveAudio constraints are explicitly set to
// false, the generated m-lines will be a=inactive.
TEST_F(PeerConnectionInterfaceTest, CreateSubsequentInactiveOffer) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
CreatePeerConnection(&constraints);
CreateAndSetRemoteOffer(kSdpStringWithStream1);
CreateAnswerAsLocalDescription();
// At this point we should be receiving stream 1, but not sending anything.
// A new offer would be recvonly, but we'll set the "no receive" constraints
// to make it inactive.
SessionDescriptionInterface* offer;
FakeConstraints offer_constraints;
offer_constraints.AddMandatory(
webrtc::MediaConstraintsInterface::kOfferToReceiveVideo, false);
offer_constraints.AddMandatory(
webrtc::MediaConstraintsInterface::kOfferToReceiveAudio, false);
DoCreateOffer(&offer, &offer_constraints);
const cricket::ContentInfo* video_content =
cricket::GetFirstVideoContent(offer->description());
const cricket::VideoContentDescription* video_desc =
static_cast<const cricket::VideoContentDescription*>(
video_content->description);
ASSERT_EQ(cricket::MD_INACTIVE, video_desc->direction());
const cricket::ContentInfo* audio_content =
cricket::GetFirstAudioContent(offer->description());
const cricket::AudioContentDescription* audio_desc =
static_cast<const cricket::AudioContentDescription*>(
audio_content->description);
ASSERT_EQ(cricket::MD_INACTIVE, audio_desc->direction());
}
// Test that PeerConnection::Close changes the states to closed and all remote
// tracks change state to ended.
TEST_F(PeerConnectionInterfaceTest, CloseAndTestStreamsAndStates) {
@ -1644,9 +1716,9 @@ TEST_F(PeerConnectionInterfaceTest, CloseAndTestMethods) {
EXPECT_TRUE(pc_->remote_description() != NULL);
rtc::scoped_ptr<SessionDescriptionInterface> offer;
EXPECT_TRUE(DoCreateOffer(offer.use()));
EXPECT_TRUE(DoCreateOffer(offer.use(), nullptr));
rtc::scoped_ptr<SessionDescriptionInterface> answer;
EXPECT_TRUE(DoCreateAnswer(answer.use()));
EXPECT_TRUE(DoCreateAnswer(answer.use(), nullptr));
std::string sdp;
ASSERT_TRUE(pc_->remote_description()->ToString(&sdp));
@ -1750,7 +1822,7 @@ TEST_F(PeerConnectionInterfaceTest, RejectMediaContent) {
EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, remote_audio->state());
rtc::scoped_ptr<SessionDescriptionInterface> local_answer;
EXPECT_TRUE(DoCreateAnswer(local_answer.accept()));
EXPECT_TRUE(DoCreateAnswer(local_answer.accept(), nullptr));
cricket::ContentInfo* video_info =
local_answer->description()->GetContentByName("video");
video_info->rejected = true;
@ -1760,7 +1832,7 @@ TEST_F(PeerConnectionInterfaceTest, RejectMediaContent) {
// Now create an offer where we reject both video and audio.
rtc::scoped_ptr<SessionDescriptionInterface> local_offer;
EXPECT_TRUE(DoCreateOffer(local_offer.accept()));
EXPECT_TRUE(DoCreateOffer(local_offer.accept(), nullptr));
video_info = local_offer->description()->GetContentByName("video");
ASSERT_TRUE(video_info != nullptr);
video_info->rejected = true;
@ -1933,7 +2005,7 @@ TEST_F(PeerConnectionInterfaceTest, LocalDescriptionChanged) {
// Create an offer just to ensure we have an identity before we manually
// call SetLocalDescription.
rtc::scoped_ptr<SessionDescriptionInterface> throwaway;
ASSERT_TRUE(DoCreateOffer(throwaway.accept()));
ASSERT_TRUE(DoCreateOffer(throwaway.accept(), nullptr));
rtc::scoped_ptr<SessionDescriptionInterface> desc_1;
CreateSessionDescriptionAndReference(2, 2, desc_1.accept());
@ -1970,7 +2042,7 @@ TEST_F(PeerConnectionInterfaceTest,
// Create an offer just to ensure we have an identity before we manually
// call SetLocalDescription.
rtc::scoped_ptr<SessionDescriptionInterface> throwaway;
ASSERT_TRUE(DoCreateOffer(throwaway.accept()));
ASSERT_TRUE(DoCreateOffer(throwaway.accept(), nullptr));
rtc::scoped_ptr<SessionDescriptionInterface> desc_1;
CreateSessionDescriptionAndReference(2, 2, desc_1.accept());
@ -1999,7 +2071,7 @@ TEST_F(PeerConnectionInterfaceTest,
// Create an offer just to ensure we have an identity before we manually
// call SetLocalDescription.
rtc::scoped_ptr<SessionDescriptionInterface> throwaway;
ASSERT_TRUE(DoCreateOffer(throwaway.accept()));
ASSERT_TRUE(DoCreateOffer(throwaway.accept(), nullptr));
rtc::scoped_ptr<SessionDescriptionInterface> desc;
CreateSessionDescriptionAndReference(1, 1, desc.accept());
@ -2045,7 +2117,7 @@ TEST_F(PeerConnectionInterfaceTest, SignalSameTracksInSeparateMediaStream) {
// Create an offer just to ensure we have an identity before we manually
// call SetLocalDescription.
rtc::scoped_ptr<SessionDescriptionInterface> throwaway;
ASSERT_TRUE(DoCreateOffer(throwaway.accept()));
ASSERT_TRUE(DoCreateOffer(throwaway.accept(), nullptr));
rtc::scoped_ptr<SessionDescriptionInterface> desc;
CreateSessionDescriptionAndReference(1, 1, desc.accept());
@ -2082,6 +2154,9 @@ TEST_F(PeerConnectionInterfaceTest, SignalSameTracksInSeparateMediaStream) {
}
// The following tests verify that session options are created correctly.
// TODO(deadbeef): Convert these tests to be more end-to-end. Instead of
// "verify options are converted correctly", should be "pass options into
// CreateOffer and verify the correct offer is produced."
TEST(CreateSessionOptionsTest, GetOptionsForOfferWithInvalidAudioOption) {
RTCOfferAnswerOptions rtc_options;
@ -2108,8 +2183,7 @@ TEST(CreateSessionOptionsTest, GetOptionsForOfferWithInvalidVideoOption) {
}
// Test that a MediaSessionOptions is created for an offer if
// OfferToReceiveAudio and OfferToReceiveVideo options are set but no
// MediaStreams are sent.
// OfferToReceiveAudio and OfferToReceiveVideo options are set.
TEST(CreateSessionOptionsTest, GetMediaSessionOptionsForOfferWithAudioVideo) {
RTCOfferAnswerOptions rtc_options;
rtc_options.offer_to_receive_audio = 1;
@ -2123,7 +2197,7 @@ TEST(CreateSessionOptionsTest, GetMediaSessionOptionsForOfferWithAudioVideo) {
}
// Test that a correct MediaSessionOptions is created for an offer if
// OfferToReceiveAudio is set but no MediaStreams are sent.
// OfferToReceiveAudio is set.
TEST(CreateSessionOptionsTest, GetMediaSessionOptionsForOfferWithAudio) {
RTCOfferAnswerOptions rtc_options;
rtc_options.offer_to_receive_audio = 1;
@ -2136,21 +2210,21 @@ TEST(CreateSessionOptionsTest, GetMediaSessionOptionsForOfferWithAudio) {
}
// Test that a correct MediaSessionOptions is created for an offer if
// the default OfferOptons is used or MediaStreams are sent.
// the default OfferOptions are used.
TEST(CreateSessionOptionsTest, GetDefaultMediaSessionOptionsForOffer) {
RTCOfferAnswerOptions rtc_options;
cricket::MediaSessionOptions options;
EXPECT_TRUE(ConvertRtcOptionsForOffer(rtc_options, &options));
EXPECT_FALSE(options.has_audio());
EXPECT_TRUE(options.has_audio());
EXPECT_FALSE(options.has_video());
EXPECT_FALSE(options.bundle_enabled);
EXPECT_TRUE(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
// OfferToReceiveVideo is set but no MediaStreams are sent.
// OfferToReceiveVideo is set.
TEST(CreateSessionOptionsTest, GetMediaSessionOptionsForOfferWithVideo) {
RTCOfferAnswerOptions rtc_options;
rtc_options.offer_to_receive_audio = 0;
@ -2209,19 +2283,19 @@ TEST(CreateSessionOptionsTest, MediaConstraintsInAnswer) {
EXPECT_TRUE(answer_options.has_audio());
EXPECT_TRUE(answer_options.has_video());
RTCOfferAnswerOptions rtc_offer_optoins;
RTCOfferAnswerOptions rtc_offer_options;
cricket::MediaSessionOptions offer_options;
EXPECT_TRUE(ConvertRtcOptionsForOffer(rtc_offer_optoins, &offer_options));
EXPECT_FALSE(offer_options.has_audio());
EXPECT_TRUE(ConvertRtcOptionsForOffer(rtc_offer_options, &offer_options));
EXPECT_TRUE(offer_options.has_audio());
EXPECT_FALSE(offer_options.has_video());
RTCOfferAnswerOptions updated_rtc_offer_optoins;
updated_rtc_offer_optoins.offer_to_receive_audio = 1;
updated_rtc_offer_optoins.offer_to_receive_video = 1;
RTCOfferAnswerOptions updated_rtc_offer_options;
updated_rtc_offer_options.offer_to_receive_audio = 1;
updated_rtc_offer_options.offer_to_receive_video = 1;
cricket::MediaSessionOptions updated_offer_options;
EXPECT_TRUE(ConvertRtcOptionsForOffer(updated_rtc_offer_optoins,
EXPECT_TRUE(ConvertRtcOptionsForOffer(updated_rtc_offer_options,
&updated_offer_options));
EXPECT_TRUE(updated_offer_options.has_audio());
EXPECT_TRUE(updated_offer_options.has_video());
@ -2240,12 +2314,4 @@ TEST(CreateSessionOptionsTest, MediaConstraintsInAnswer) {
ParseConstraintsForAnswer(&updated_answer_c, &updated_answer_options));
EXPECT_TRUE(updated_answer_options.has_audio());
EXPECT_TRUE(updated_answer_options.has_video());
RTCOfferAnswerOptions default_rtc_options;
EXPECT_TRUE(
ConvertRtcOptionsForOffer(default_rtc_options, &updated_offer_options));
// By default, |has_audio| or |has_video| are false if there is no media
// track.
EXPECT_FALSE(updated_offer_options.has_audio());
EXPECT_FALSE(updated_offer_options.has_video());
}

35
talk/app/webrtc/webrtcsdp.cc
View File

@ -268,7 +268,6 @@ static bool IsDtlsSctp(const std::string& protocol);
static bool ParseSessionDescription(const std::string& message, size_t* pos,
std::string* session_id,
std::string* session_version,
bool* supports_msid,
TransportDescription* session_td,
RtpHeaderExtensions* session_extmaps,
cricket::SessionDescription* desc,
@ -280,7 +279,6 @@ static bool ParseMediaDescription(
const std::string& message,
const TransportDescription& session_td,
const RtpHeaderExtensions& session_extmaps,
bool supports_msid,
size_t* pos, cricket::SessionDescription* desc,
std::vector<JsepIceCandidate*>* candidates,
SdpParseError* error);
@ -898,20 +896,18 @@ bool SdpDeserialize(const std::string& message,
cricket::SessionDescription* desc = new cricket::SessionDescription();
std::vector<JsepIceCandidate*> candidates;
size_t current_pos = 0;
bool supports_msid = false;
// Session Description
if (!ParseSessionDescription(message, &current_pos, &session_id,
&session_version, &supports_msid, &session_td,
&session_extmaps, desc, error)) {
&session_version, &session_td, &session_extmaps,
desc, error)) {
delete desc;
return false;
}
// Media Description
if (!ParseMediaDescription(message, session_td, session_extmaps,
supports_msid, &current_pos, desc, &candidates,
error)) {
if (!ParseMediaDescription(message, session_td, session_extmaps, &current_pos,
desc, &candidates, error)) {
delete desc;
for (std::vector<JsepIceCandidate*>::const_iterator
it = candidates.begin(); it != candidates.end(); ++it) {
@ -1379,13 +1375,7 @@ void BuildRtpContentAttributes(
// RFC 3264
// a=sendrecv || a=sendonly || a=sendrecv || a=inactive
cricket::MediaContentDirection direction = media_desc->direction();
if (media_desc->streams().empty() && direction == cricket::MD_SENDRECV) {
direction = cricket::MD_RECVONLY;
}
switch (direction) {
switch (media_desc->direction()) {
case cricket::MD_INACTIVE:
InitAttrLine(kAttributeInactive, &os);
break;
@ -1798,13 +1788,14 @@ bool IsDtlsSctp(const std::string& protocol) {
bool ParseSessionDescription(const std::string& message, size_t* pos,
std::string* session_id,
std::string* session_version,
bool* supports_msid,
TransportDescription* session_td,
RtpHeaderExtensions* session_extmaps,
cricket::SessionDescription* desc,
SdpParseError* error) {
std::string line;
desc->set_msid_supported(false);
// RFC 4566
// v= (protocol version)
if (!GetLineWithType(message, pos, &line, kLineTypeVersion)) {
@ -1936,7 +1927,8 @@ bool ParseSessionDescription(const std::string& message, size_t* pos,
if (!GetValue(line, kAttributeMsidSemantics, &semantics, error)) {
return false;
}
*supports_msid = CaseInsensitiveFind(semantics, kMediaStreamSemantic);
desc->set_msid_supported(
CaseInsensitiveFind(semantics, kMediaStreamSemantic));
} else if (HasAttribute(line, kAttributeExtmap)) {
RtpHeaderExtension extmap;
if (!ParseExtmap(line, &extmap, error)) {
@ -2146,7 +2138,6 @@ static C* ParseContentDescription(const std::string& message,
bool ParseMediaDescription(const std::string& message,
const TransportDescription& session_td,
const RtpHeaderExtensions& session_extmaps,
bool supports_msid,
size_t* pos,
cricket::SessionDescription* desc,
std::vector<JsepIceCandidate*>* candidates,
@ -2241,14 +2232,6 @@ bool ParseMediaDescription(const std::string& message,
}
if (IsRtp(protocol)) {
// Make sure to set the media direction correctly. If the direction is not
// MD_RECVONLY or Inactive and no streams are parsed,
// a default MediaStream will be created to prepare for receiving media.
if (supports_msid && content->streams().empty() &&
content->direction() == cricket::MD_SENDRECV) {
content->set_direction(cricket::MD_RECVONLY);
}
// Set the extmap.
if (!session_extmaps.empty() &&
!content->rtp_header_extensions().empty()) {

4
talk/app/webrtc/webrtcsdp_unittest.cc
View File

@ -883,6 +883,9 @@ class WebRtcSdpTest : public testing::Test {
EXPECT_TRUE(CompareCandidates(transport1.description.candidates,
transport2.description.candidates));
}
// global attributes
EXPECT_EQ(desc1.msid_supported(), desc2.msid_supported());
}
bool CompareCandidates(const Candidates& cs1, const Candidates& cs2) {
@ -1967,6 +1970,7 @@ TEST_F(WebRtcSdpTest, DeserializeSessionDescriptionWithRejectedAudioVideo) {
// Tests that we can still handle the sdp uses mslabel and label instead of
// msid for backward compatibility.
TEST_F(WebRtcSdpTest, DeserializeSessionDescriptionWithoutMsid) {
jdesc_.description()->set_msid_supported(false);
JsepSessionDescription jdesc(kDummyString);
std::string sdp_without_msid = kSdpFullString;
Replace("msid", "xmsid", &sdp_without_msid);

24
talk/app/webrtc/webrtcsession_unittest.cc
View File

@ -571,9 +571,24 @@ class WebRtcSessionTest
void GetOptionsForOffer(
const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
cricket::MediaSessionOptions* session_options) {
AddStreamsToOptions(session_options);
ASSERT_TRUE(ConvertRtcOptionsForOffer(rtc_options, session_options));
AddStreamsToOptions(session_options);
if (rtc_options.offer_to_receive_audio ==
RTCOfferAnswerOptions::kUndefined) {
session_options->recv_audio =
session_options->HasSendMediaStream(cricket::MEDIA_TYPE_AUDIO);
}
if (rtc_options.offer_to_receive_video ==
RTCOfferAnswerOptions::kUndefined) {
session_options->recv_video =
session_options->HasSendMediaStream(cricket::MEDIA_TYPE_VIDEO);
}
session_options->bundle_enabled =
session_options->bundle_enabled &&
(session_options->has_audio() || session_options->has_video() ||
session_options->has_data());
if (session_->data_channel_type() == cricket::DCT_SCTP && data_channel_) {
session_options->data_channel_type = cricket::DCT_SCTP;
}
@ -581,11 +596,16 @@ class WebRtcSessionTest
void GetOptionsForAnswer(const webrtc::MediaConstraintsInterface* constraints,
cricket::MediaSessionOptions* session_options) {
AddStreamsToOptions(session_options);
session_options->recv_audio = false;
session_options->recv_video = false;
ASSERT_TRUE(ParseConstraintsForAnswer(constraints, session_options));
AddStreamsToOptions(session_options);
session_options->bundle_enabled =
session_options->bundle_enabled &&
(session_options->has_audio() || session_options->has_video() ||
session_options->has_data());
if (session_->data_channel_type() == cricket::DCT_SCTP) {
session_options->data_channel_type = cricket::DCT_SCTP;
}

35
talk/session/media/mediasession.cc
View File

@ -1036,12 +1036,15 @@ static bool CreateMediaContentAnswer(
answer->set_direction(MD_RECVONLY);
break;
case MD_RECVONLY:
answer->set_direction(MD_SENDONLY);
answer->set_direction(answer->streams().empty() ? MD_INACTIVE
: MD_SENDONLY);
break;
case MD_SENDRECV:
answer->set_direction(MD_SENDRECV);
answer->set_direction(answer->streams().empty() ? MD_RECVONLY
: MD_SENDRECV);
break;
default:
RTC_DCHECK(false && "MediaContentDescription has unexpected direction.");
break;
}
@ -1529,8 +1532,18 @@ bool MediaSessionDescriptionFactory::AddAudioContentForOffer(
bool secure_transport = (transport_desc_factory_->secure() != SEC_DISABLED);
SetMediaProtocol(secure_transport, audio.get());
if (!options.recv_audio) {
audio->set_direction(MD_SENDONLY);
if (!audio->streams().empty()) {
if (options.recv_audio) {
audio->set_direction(MD_SENDRECV);
} else {
audio->set_direction(MD_SENDONLY);
}
} else {
if (options.recv_audio) {
audio->set_direction(MD_RECVONLY);
} else {
audio->set_direction(MD_INACTIVE);
}
}
desc->AddContent(CN_AUDIO, NS_JINGLE_RTP, audio.release());
@ -1574,8 +1587,18 @@ bool MediaSessionDescriptionFactory::AddVideoContentForOffer(
bool secure_transport = (transport_desc_factory_->secure() != SEC_DISABLED);
SetMediaProtocol(secure_transport, video.get());
if (!options.recv_video) {
video->set_direction(MD_SENDONLY);
if (!video->streams().empty()) {
if (options.recv_video) {
video->set_direction(MD_SENDRECV);
} else {
video->set_direction(MD_SENDONLY);
}
} else {
if (options.recv_video) {
video->set_direction(MD_RECVONLY);
} else {
video->set_direction(MD_INACTIVE);
}
}
desc->AddContent(CN_VIDEO, NS_JINGLE_RTP, video.release());

5
webrtc/p2p/base/sessiondescription.h
View File

@ -160,10 +160,15 @@ class SessionDescription {
// Remove the first group with the same semantics specified by |name|.
void RemoveGroupByName(const std::string& name);
// Global attributes.
void set_msid_supported(bool supported) { msid_supported_ = supported; }
bool msid_supported() const { return msid_supported_; }
private:
ContentInfos contents_;
TransportInfos transport_infos_;
ContentGroups content_groups_;
bool msid_supported_ = true;
};
// Indicates whether a ContentDescription was an offer or an answer, as