(Auto)update libjingle 72847605-> 72850595
git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6855 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/app/webrtc/mediaconstraintsinterface.cc b/app/webrtc/mediaconstraintsinterface.cc
index 874ce64..74f6542 100644
--- a/app/webrtc/mediaconstraintsinterface.cc
+++ b/app/webrtc/mediaconstraintsinterface.cc
@@ -95,6 +95,8 @@
"googSuspendBelowMinBitrate";
const char MediaConstraintsInterface::kImprovedWifiBwe[] =
"googImprovedWifiBwe";
+const char MediaConstraintsInterface::kNumUnsignalledRecvStreams[] =
+ "googNumUnsignalledRecvStreams";
const char MediaConstraintsInterface::kScreencastMinBitrate[] =
"googScreencastMinBitrate";
// TODO(ronghuawu): Remove once cpu overuse detection is stable.
diff --git a/app/webrtc/mediaconstraintsinterface.h b/app/webrtc/mediaconstraintsinterface.h
index 1304d88..b3efbb3 100644
--- a/app/webrtc/mediaconstraintsinterface.h
+++ b/app/webrtc/mediaconstraintsinterface.h
@@ -116,6 +116,8 @@
static const char kEnableVideoSuspendBelowMinBitrate[];
// googSuspendBelowMinBitrate
static const char kImprovedWifiBwe[]; // googImprovedWifiBwe
+ static const char kNumUnsignalledRecvStreams[];
+ // googNumUnsignalledRecvStreams
static const char kScreencastMinBitrate[]; // googScreencastMinBitrate
static const char kCpuOveruseDetection[]; // googCpuOveruseDetection
static const char kCpuUnderuseThreshold[]; // googCpuUnderuseThreshold
diff --git a/app/webrtc/webrtcsession.cc b/app/webrtc/webrtcsession.cc
index 3add7e1..7357a61 100644
--- a/app/webrtc/webrtcsession.cc
+++ b/app/webrtc/webrtcsession.cc
@@ -38,6 +38,7 @@
#include "talk/app/webrtc/mediastreamsignaling.h"
#include "talk/app/webrtc/peerconnectioninterface.h"
#include "talk/app/webrtc/webrtcsessiondescriptionfactory.h"
+#include "webrtc/base/basictypes.h"
#include "webrtc/base/helpers.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/stringencode.h"
@@ -74,6 +75,7 @@
"Called with SDP without ice-ufrag and ice-pwd.";
const char kSessionError[] = "Session error code: ";
const char kSessionErrorDesc[] = "Session error description: ";
+const int kMaxUnsignalledRecvStreams = 20;
// Compares |answer| against |offer|. Comparision is done
// for number of m-lines in answer against offer. If matches true will be
@@ -590,6 +592,17 @@
}
SetOptionFromOptionalConstraint(constraints,
+ MediaConstraintsInterface::kNumUnsignalledRecvStreams,
+ &video_options_.unsignalled_recv_stream_limit);
+ if (video_options_.unsignalled_recv_stream_limit.IsSet()) {
+ int stream_limit;
+ video_options_.unsignalled_recv_stream_limit.Get(&stream_limit);
+ stream_limit = rtc::_min(kMaxUnsignalledRecvStreams, stream_limit);
+ stream_limit = rtc::_max(0, stream_limit);
+ video_options_.unsignalled_recv_stream_limit.Set(stream_limit);
+ }
+
+ SetOptionFromOptionalConstraint(constraints,
MediaConstraintsInterface::kHighStartBitrate,
&video_options_.video_start_bitrate);
diff --git a/app/webrtc/webrtcsession.h b/app/webrtc/webrtcsession.h
index f5de979..564676d 100644
--- a/app/webrtc/webrtcsession.h
+++ b/app/webrtc/webrtcsession.h
@@ -73,6 +73,9 @@
extern const char kSdpWithoutSdesAndDtlsDisabled[];
extern const char kSessionError[];
extern const char kSessionErrorDesc[];
+// Maximum number of received video streams that will be processed by webrtc
+// even if they are not signalled beforehand.
+extern const int kMaxUnsignalledRecvStreams;
// ICE state callback interface.
class IceObserver {
diff --git a/app/webrtc/webrtcsession_unittest.cc b/app/webrtc/webrtcsession_unittest.cc
index b5e1dda..8bc8e8d 100644
--- a/app/webrtc/webrtcsession_unittest.cc
+++ b/app/webrtc/webrtcsession_unittest.cc
@@ -98,6 +98,7 @@
using webrtc::kSdpWithoutSdesCrypto;
using webrtc::kSessionError;
using webrtc::kSessionErrorDesc;
+using webrtc::kMaxUnsignalledRecvStreams;
typedef PeerConnectionInterface::RTCOfferAnswerOptions RTCOfferAnswerOptions;
@@ -537,6 +538,28 @@
VerifyCryptoParams(answer->description());
}
+ void SetAndVerifyNumUnsignalledRecvStreams(
+ int value_set, int value_expected) {
+ constraints_.reset(new FakeConstraints());
+ constraints_->AddOptional(
+ webrtc::MediaConstraintsInterface::kNumUnsignalledRecvStreams,
+ value_set);
+ session_.reset();
+ Init(NULL);
+ mediastream_signaling_.SendAudioVideoStream1();
+ SessionDescriptionInterface* offer = CreateOffer();
+
+ SetLocalDescriptionWithoutError(offer);
+
+ video_channel_ = media_engine_->GetVideoChannel(0);
+
+ ASSERT_TRUE(video_channel_ != NULL);
+ cricket::VideoOptions video_options;
+ EXPECT_TRUE(video_channel_->GetOptions(&video_options));
+ EXPECT_EQ(value_expected,
+ video_options.unsignalled_recv_stream_limit.GetWithDefaultIfUnset(-1));
+ }
+
void CompareIceUfragAndPassword(const cricket::SessionDescription* desc1,
const cricket::SessionDescription* desc2,
bool expect_equal) {
@@ -3279,6 +3302,15 @@
video_options.suspend_below_min_bitrate.GetWithDefaultIfUnset(false));
}
+TEST_F(WebRtcSessionTest, TestNumUnsignalledRecvStreamsConstraint) {
+ // Number of unsignalled receiving streams should be between 0 and
+ // kMaxUnsignalledRecvStreams.
+ SetAndVerifyNumUnsignalledRecvStreams(10, 10);
+ SetAndVerifyNumUnsignalledRecvStreams(kMaxUnsignalledRecvStreams + 1,
+ kMaxUnsignalledRecvStreams);
+ SetAndVerifyNumUnsignalledRecvStreams(-1, 0);
+}
+
// Tests that we can renegotiate new media content with ICE candidates in the
// new remote SDP.
TEST_F(WebRtcSessionTest, TestRenegotiateNewMediaWithCandidatesInSdp) {
