Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(590)

Issues Search
    My Issues | Starred     Open | Closed | All

Unified Diff: webrtc/api/peerconnectioninterface_unittest.cc

Issue 1966333002: Refactoring some tests in peerconnectioninterface_unittest.cc. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 7 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
« no previous file with comments | « no previous file | webrtc/pc/mediasession.h » ('j') | webrtc/pc/mediasession.cc » ('J')
Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
Index: webrtc/api/peerconnectioninterface_unittest.cc
diff --git a/webrtc/api/peerconnectioninterface_unittest.cc b/webrtc/api/peerconnectioninterface_unittest.cc
index 2594b6c10637996c3a554757a188da633d5b12cb..4be690e93523e2cce1d0e76dee6bbae62c76e167 100644
--- a/webrtc/api/peerconnectioninterface_unittest.cc
+++ b/webrtc/api/peerconnectioninterface_unittest.cc
@@ -239,8 +239,9 @@ static const char kSdpStringMs1Video1[] =
return; \
}
-using rtc::scoped_refptr;
using ::testing::Exactly;
+using cricket::StreamParams;
+using rtc::scoped_refptr;
using webrtc::AudioSourceInterface;
using webrtc::AudioTrack;
using webrtc::AudioTrackInterface;
@@ -248,6 +249,7 @@ using webrtc::DataBuffer;
using webrtc::DataChannelInterface;
using webrtc::FakeConstraints;
using webrtc::IceCandidateInterface;
+using webrtc::JsepSessionDescription;
using webrtc::MediaConstraintsInterface;
using webrtc::MediaStream;
using webrtc::MediaStreamInterface;
@@ -325,12 +327,26 @@ bool ContainsSender(
return false;
}
+// Check if |senders| contains the specified sender, by id and stream id.
+bool ContainsSender(
+ const std::vector<rtc::scoped_refptr<RtpSenderInterface>>& senders,
+ const std::string& id,
+ const std::string& stream_id) {
+ for (const auto& sender : senders) {
+ if (sender->id() == id && sender->stream_id() == stream_id) {
+ return true;
+ }
+ }
+ return false;
+}
+
// Create a collection of streams.
// CreateStreamCollection(1) creates a collection that
// correspond to kSdpStringWithStream1.
// CreateStreamCollection(2) correspond to kSdpStringWithStream1And2.
rtc::scoped_refptr<StreamCollection> CreateStreamCollection(
- int number_of_streams) {
+ int number_of_streams,
+ int tracks_per_stream) {
rtc::scoped_refptr<StreamCollection> local_collection(
StreamCollection::Create());
@@ -338,16 +354,19 @@ rtc::scoped_refptr<StreamCollection> CreateStreamCollection(
rtc::scoped_refptr<webrtc::MediaStreamInterface> stream(
webrtc::MediaStream::Create(kStreams[i]));
- // Add a local audio track.
- rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
- webrtc::AudioTrack::Create(kAudioTracks[i], nullptr));
- stream->AddTrack(audio_track);
-
- // Add a local video track.
- rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track(
- webrtc::VideoTrack::Create(kVideoTracks[i],
- webrtc::FakeVideoTrackSource::Create()));
- stream->AddTrack(video_track);
+ for (int j = 0; j < tracks_per_stream; ++j) {
+ // Add a local audio track.
+ rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
+ webrtc::AudioTrack::Create(kAudioTracks[i * tracks_per_stream + j],
+ nullptr));
+ stream->AddTrack(audio_track);
+
+ // Add a local video track.
+ rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track(
+ webrtc::VideoTrack::Create(kVideoTracks[i * tracks_per_stream + j],
+ webrtc::FakeVideoTrackSource::Create()));
+ stream->AddTrack(video_track);
+ }
local_collection->AddStream(stream);
}
@@ -1984,7 +2003,7 @@ TEST_F(PeerConnectionInterfaceTest, UpdateRemoteStreams) {
CreatePeerConnection(&constraints);
CreateAndSetRemoteOffer(kSdpStringWithStream1);
- rtc::scoped_refptr<StreamCollection> reference(CreateStreamCollection(1));
+ rtc::scoped_refptr<StreamCollection> reference(CreateStreamCollection(1, 1));
EXPECT_TRUE(
CompareStreamCollections(observer_.remote_streams(), reference.get()));
MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
@@ -1994,7 +2013,7 @@ TEST_F(PeerConnectionInterfaceTest, UpdateRemoteStreams) {
// MediaStream.
CreateAndSetRemoteOffer(kSdpStringWithStream1And2);
- rtc::scoped_refptr<StreamCollection> reference2(CreateStreamCollection(2));
+ rtc::scoped_refptr<StreamCollection> reference2(CreateStreamCollection(2, 1));
EXPECT_TRUE(
CompareStreamCollections(observer_.remote_streams(), reference2.get()));
}
@@ -2260,7 +2279,7 @@ TEST_F(PeerConnectionInterfaceTest, VerifyDefaultStreamIsNotCreated) {
true);
CreatePeerConnection(&constraints);
CreateAndSetRemoteOffer(kSdpStringWithStream1);
- rtc::scoped_refptr<StreamCollection> reference(CreateStreamCollection(1));
+ rtc::scoped_refptr<StreamCollection> reference(CreateStreamCollection(1, 1));
EXPECT_TRUE(
CompareStreamCollections(observer_.remote_streams(), reference.get()));
@@ -2277,16 +2296,15 @@ TEST_F(PeerConnectionInterfaceTest, LocalDescriptionChanged) {
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
CreatePeerConnection(&constraints);
- // Create an offer just to ensure we have an identity before we manually
- // call SetLocalDescription.
- std::unique_ptr<SessionDescriptionInterface> throwaway;
- ASSERT_TRUE(DoCreateOffer(&throwaway, nullptr));
- std::unique_ptr<SessionDescriptionInterface> desc_1 =
- CreateSessionDescriptionAndReference(2, 2);
+ // Create an offer with 1 stream with 2 tracks of each type.
+ rtc::scoped_refptr<StreamCollection> stream_collection =
+ CreateStreamCollection(1, 2);
+ pc_->AddStream(stream_collection->at(0));
+ std::unique_ptr<SessionDescriptionInterface> offer;
+ ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
+ EXPECT_TRUE(DoSetLocalDescription(offer.release()));
- pc_->AddStream(reference_collection_->at(0));
- EXPECT_TRUE(DoSetLocalDescription(desc_1.release()));
auto senders = pc_->GetSenders();
EXPECT_EQ(4u, senders.size());
EXPECT_TRUE(ContainsSender(senders, kAudioTracks[0]));
@@ -2295,11 +2313,12 @@ TEST_F(PeerConnectionInterfaceTest, LocalDescriptionChanged) {
EXPECT_TRUE(ContainsSender(senders, kVideoTracks[1]));
// Remove an audio and video track.
- pc_->RemoveStream(reference_collection_->at(0));
- std::unique_ptr<SessionDescriptionInterface> desc_2 =
- CreateSessionDescriptionAndReference(1, 1);
- pc_->AddStream(reference_collection_->at(0));
- EXPECT_TRUE(DoSetLocalDescription(desc_2.release()));
+ pc_->RemoveStream(stream_collection->at(0));
+ stream_collection = CreateStreamCollection(1, 1);
+ pc_->AddStream(stream_collection->at(0));
+ ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
+ EXPECT_TRUE(DoSetLocalDescription(offer.release()));
+
senders = pc_->GetSenders();
EXPECT_EQ(2u, senders.size());
EXPECT_TRUE(ContainsSender(senders, kAudioTracks[0]));
@@ -2316,19 +2335,20 @@ TEST_F(PeerConnectionInterfaceTest,
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
CreatePeerConnection(&constraints);
- // Create an offer just to ensure we have an identity before we manually
- // call SetLocalDescription.
- std::unique_ptr<SessionDescriptionInterface> throwaway;
- ASSERT_TRUE(DoCreateOffer(&throwaway, nullptr));
- std::unique_ptr<SessionDescriptionInterface> desc_1 =
- CreateSessionDescriptionAndReference(2, 2);
+ rtc::scoped_refptr<StreamCollection> stream_collection =
+ CreateStreamCollection(1, 2);
+ // Add a stream to create the offer, but remove it afterwards.
+ pc_->AddStream(stream_collection->at(0));
+ std::unique_ptr<SessionDescriptionInterface> offer;
+ ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
+ pc_->RemoveStream(stream_collection->at(0));
- EXPECT_TRUE(DoSetLocalDescription(desc_1.release()));
+ EXPECT_TRUE(DoSetLocalDescription(offer.release()));
auto senders = pc_->GetSenders();
EXPECT_EQ(0u, senders.size());
- pc_->AddStream(reference_collection_->at(0));
+ pc_->AddStream(stream_collection->at(0));
senders = pc_->GetSenders();
EXPECT_EQ(4u, senders.size());
EXPECT_TRUE(ContainsSender(senders, kAudioTracks[0]));
@@ -2345,37 +2365,44 @@ TEST_F(PeerConnectionInterfaceTest,
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
CreatePeerConnection(&constraints);
- // Create an offer just to ensure we have an identity before we manually
- // call SetLocalDescription.
- std::unique_ptr<SessionDescriptionInterface> throwaway;
- ASSERT_TRUE(DoCreateOffer(&throwaway, nullptr));
- std::unique_ptr<SessionDescriptionInterface> desc =
- CreateSessionDescriptionAndReference(1, 1);
- std::string sdp;
- desc->ToString(&sdp);
+ rtc::scoped_refptr<StreamCollection> stream_collection =
+ CreateStreamCollection(2, 1);
+ pc_->AddStream(stream_collection->at(0));
+ std::unique_ptr<SessionDescriptionInterface> offer;
+ ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
+ // Grab a copy of the offer before it gets passed into the PC.
+ std::unique_ptr<JsepSessionDescription> modified_offer(
+ new JsepSessionDescription(JsepSessionDescription::kOffer));
+ modified_offer->Initialize(offer->description()->Copy(), offer->session_id(),
+ offer->session_version());
+ EXPECT_TRUE(DoSetLocalDescription(offer.release()));
- pc_->AddStream(reference_collection_->at(0));
- EXPECT_TRUE(DoSetLocalDescription(desc.release()));
auto senders = pc_->GetSenders();
EXPECT_EQ(2u, senders.size());
EXPECT_TRUE(ContainsSender(senders, kAudioTracks[0]));
EXPECT_TRUE(ContainsSender(senders, kVideoTracks[0]));
// Change the ssrc of the audio and video track.
- std::string ssrc_org = "a=ssrc:1";
- std::string ssrc_to = "a=ssrc:97";
- rtc::replace_substrs(ssrc_org.c_str(), ssrc_org.length(), ssrc_to.c_str(),
- ssrc_to.length(), &sdp);
- ssrc_org = "a=ssrc:2";
- ssrc_to = "a=ssrc:98";
- rtc::replace_substrs(ssrc_org.c_str(), ssrc_org.length(), ssrc_to.c_str(),
- ssrc_to.length(), &sdp);
- std::unique_ptr<SessionDescriptionInterface> updated_desc(
- webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer, sdp,
- nullptr));
-
- EXPECT_TRUE(DoSetLocalDescription(updated_desc.release()));
+ cricket::MediaContentDescription* desc =
+ cricket::GetFirstAudioContentDescription(modified_offer->description());
+ ASSERT_TRUE(desc != NULL);
+ for (StreamParams& stream : desc->mutable_streams()) {
+ for (unsigned int& ssrc : stream.ssrcs) {
tommi 2016/05/12 11:41:44 nit: since we rarely use unsigned int, and should
+ ++ssrc;
+ }
+ }
+
+ desc =
+ cricket::GetFirstVideoContentDescription(modified_offer->description());
+ ASSERT_TRUE(desc != NULL);
+ for (StreamParams& stream : desc->mutable_streams()) {
+ for (unsigned int& ssrc : stream.ssrcs) {
+ ++ssrc;
+ }
+ }
+
+ EXPECT_TRUE(DoSetLocalDescription(modified_offer.release()));
senders = pc_->GetSenders();
EXPECT_EQ(2u, senders.size());
EXPECT_TRUE(ContainsSender(senders, kAudioTracks[0]));
@@ -2392,43 +2419,36 @@ TEST_F(PeerConnectionInterfaceTest,
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
CreatePeerConnection(&constraints);
- // Create an offer just to ensure we have an identity before we manually
- // call SetLocalDescription.
- std::unique_ptr<SessionDescriptionInterface> throwaway;
- ASSERT_TRUE(DoCreateOffer(&throwaway, nullptr));
- std::unique_ptr<SessionDescriptionInterface> desc =
- CreateSessionDescriptionAndReference(1, 1);
- std::string sdp;
- desc->ToString(&sdp);
+ rtc::scoped_refptr<StreamCollection> stream_collection =
+ CreateStreamCollection(2, 1);
+ pc_->AddStream(stream_collection->at(0));
+ std::unique_ptr<SessionDescriptionInterface> offer;
+ ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
+ EXPECT_TRUE(DoSetLocalDescription(offer.release()));
- pc_->AddStream(reference_collection_->at(0));
- EXPECT_TRUE(DoSetLocalDescription(desc.release()));
auto senders = pc_->GetSenders();
EXPECT_EQ(2u, senders.size());
- EXPECT_TRUE(ContainsSender(senders, kAudioTracks[0]));
- EXPECT_TRUE(ContainsSender(senders, kVideoTracks[0]));
+ EXPECT_TRUE(ContainsSender(senders, kAudioTracks[0], kStreams[0]));
+ EXPECT_TRUE(ContainsSender(senders, kVideoTracks[0], kStreams[0]));
// Add a new MediaStream but with the same tracks as in the first stream.
rtc::scoped_refptr<webrtc::MediaStreamInterface> stream_1(
webrtc::MediaStream::Create(kStreams[1]));
- stream_1->AddTrack(reference_collection_->at(0)->GetVideoTracks()[0]);
- stream_1->AddTrack(reference_collection_->at(0)->GetAudioTracks()[0]);
+ stream_1->AddTrack(stream_collection->at(0)->GetVideoTracks()[0]);
+ stream_1->AddTrack(stream_collection->at(0)->GetAudioTracks()[0]);
pc_->AddStream(stream_1);
- // Replace msid in the original SDP.
- rtc::replace_substrs(kStreams[0], strlen(kStreams[0]), kStreams[1],
- strlen(kStreams[1]), &sdp);
-
- std::unique_ptr<SessionDescriptionInterface> updated_desc(
- webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer, sdp,
- nullptr));
+ ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
+ EXPECT_TRUE(DoSetLocalDescription(offer.release()));
- EXPECT_TRUE(DoSetLocalDescription(updated_desc.release()));
- senders = pc_->GetSenders();
- EXPECT_EQ(2u, senders.size());
- EXPECT_TRUE(ContainsSender(senders, kAudioTracks[0]));
- EXPECT_TRUE(ContainsSender(senders, kVideoTracks[0]));
+ auto new_senders = pc_->GetSenders();
+ // Should be the same senders as before, but with updated stream id.
+ // Note that this behavior is subject to change in the future.
+ // We may decide the PC should ignore existing tracks in AddStream.
+ EXPECT_EQ(senders, new_senders);
+ EXPECT_TRUE(ContainsSender(new_senders, kAudioTracks[0], kStreams[1]));
+ EXPECT_TRUE(ContainsSender(new_senders, kVideoTracks[0], kStreams[1]));
}
// The PeerConnectionMediaConfig tests below verify that configuration
« no previous file with comments | « no previous file | webrtc/pc/mediasession.h » ('j') | webrtc/pc/mediasession.cc » ('J')

Powered by Google App Engine
This is Rietveld 408576698