Update libjingle to 61514460

TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5545 4adac7df-926f-26a2-2b94-8c16560cd09d

talk/app/webrtc/mediastreamhandler.cc

@@ -106,6 +106,8 @@ void LocalAudioTrackHandler::Stop() {
 void LocalAudioTrackHandler::OnEnabledChanged() {
   cricket::AudioOptions options;
   if (audio_track_->enabled() && audio_track_->GetSource()) {
+    // TODO(xians): Remove this static_cast since we should be able to connect
+    // a remote audio track to peer connection.
     options = static_cast<LocalAudioSource*>(
         audio_track_->GetSource())->options();
   }
@@ -125,10 +127,12 @@ RemoteAudioTrackHandler::RemoteAudioTrackHandler(
     : TrackHandler(track, ssrc),
       audio_track_(track),
       provider_(provider) {
+  track->GetSource()->RegisterAudioObserver(this);
   OnEnabledChanged();
 }
 
 RemoteAudioTrackHandler::~RemoteAudioTrackHandler() {
+  audio_track_->GetSource()->UnregisterAudioObserver(this);
 }
 
 void RemoteAudioTrackHandler::Stop() {
@@ -143,6 +147,14 @@ void RemoteAudioTrackHandler::OnEnabledChanged() {
                              audio_track_->GetRenderer());
 }
 
+void RemoteAudioTrackHandler::OnSetVolume(double volume) {
+  // When the track is disabled, the volume of the source, which is the
+  // corresponding WebRtc Voice Engine channel will be 0. So we do not allow
+  // setting the volume to the source when the track is disabled.
+  if (audio_track_->enabled())
+    provider_->SetAudioPlayoutVolume(ssrc(), volume);
+}
+
 LocalVideoTrackHandler::LocalVideoTrackHandler(
     VideoTrackInterface* track,
     uint32 ssrc,

talk/app/webrtc/mediastreamhandler.h

@@ -118,7 +118,8 @@ class LocalAudioTrackHandler : public TrackHandler {
 // RemoteAudioTrackHandler listen to events on a remote AudioTrack instance
 // connected to a PeerConnection and orders the |provider| to executes the
 // requested change.
-class RemoteAudioTrackHandler : public TrackHandler {
+class RemoteAudioTrackHandler : public AudioSourceInterface::AudioObserver,
+                                public TrackHandler {
  public:
   RemoteAudioTrackHandler(AudioTrackInterface* track,
                           uint32 ssrc,
@@ -131,6 +132,9 @@ class RemoteAudioTrackHandler : public TrackHandler {
   virtual void OnEnabledChanged() OVERRIDE;
 
  private:
+  // AudioSourceInterface::AudioObserver implementation.
+  virtual void OnSetVolume(double volume) OVERRIDE;
+
   AudioTrackInterface* audio_track_;
   AudioProviderInterface* provider_;
 };

talk/app/webrtc/mediastreamhandler_unittest.cc

@@ -31,6 +31,7 @@
 
 #include "talk/app/webrtc/audiotrack.h"
 #include "talk/app/webrtc/mediastream.h"
+#include "talk/app/webrtc/remoteaudiosource.h"
 #include "talk/app/webrtc/streamcollection.h"
 #include "talk/app/webrtc/videosource.h"
 #include "talk/app/webrtc/videotrack.h"
@@ -59,6 +60,7 @@ class MockAudioProvider : public AudioProviderInterface {
   MOCK_METHOD4(SetAudioSend, void(uint32 ssrc, bool enable,
                                   const cricket::AudioOptions& options,
                                   cricket::AudioRenderer* renderer));
+  MOCK_METHOD2(SetAudioPlayoutVolume, void(uint32 ssrc, double volume));
 };
 
 // Helper class to test MediaStreamHandler.
@@ -110,12 +112,11 @@ class MediaStreamHandlerTest : public testing::Test {
         FakeVideoSource::Create());
     video_track_ = VideoTrack::Create(kVideoTrackId, source);
     EXPECT_TRUE(stream_->AddTrack(video_track_));
-    audio_track_ = AudioTrack::Create(kAudioTrackId,
-                                           NULL);
-    EXPECT_TRUE(stream_->AddTrack(audio_track_));
   }
 
   void AddLocalAudioTrack() {
+    audio_track_ = AudioTrack::Create(kAudioTrackId, NULL);
+    EXPECT_TRUE(stream_->AddTrack(audio_track_));
     EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, true, _, _));
     handlers_.AddLocalAudioTrack(stream_, stream_->GetAudioTracks()[0],
                                  kAudioSsrc);
@@ -144,6 +145,9 @@ class MediaStreamHandlerTest : public testing::Test {
   }
 
   void AddRemoteAudioTrack() {
+    audio_track_ = AudioTrack::Create(kAudioTrackId,
+                                      RemoteAudioSource::Create().get());
+    EXPECT_TRUE(stream_->AddTrack(audio_track_));
     EXPECT_CALL(audio_provider_, SetAudioPlayout(kAudioSsrc, true, _));
     handlers_.AddRemoteAudioTrack(stream_, stream_->GetAudioTracks()[0],
                                   kAudioSsrc);
@@ -292,4 +296,27 @@ TEST_F(MediaStreamHandlerTest, RemoteVideoTrackDisable) {
   handlers_.TearDown();
 }
 
+TEST_F(MediaStreamHandlerTest, RemoteAudioTrackSetVolume) {
+  AddRemoteAudioTrack();
+
+  double volume = 0.5;
+  EXPECT_CALL(audio_provider_, SetAudioPlayoutVolume(kAudioSsrc, volume));
+  audio_track_->GetSource()->SetVolume(volume);
+
+  // Disable the audio track, this should prevent setting the volume.
+  EXPECT_CALL(audio_provider_, SetAudioPlayout(kAudioSsrc, false, _));
+  audio_track_->set_enabled(false);
+  audio_track_->GetSource()->SetVolume(1.0);
+
+  EXPECT_CALL(audio_provider_, SetAudioPlayout(kAudioSsrc, true, _));
+  audio_track_->set_enabled(true);
+
+  double new_volume = 0.8;
+  EXPECT_CALL(audio_provider_, SetAudioPlayoutVolume(kAudioSsrc, new_volume));
+  audio_track_->GetSource()->SetVolume(new_volume);
+
+  RemoveRemoteAudioTrack();
+  handlers_.TearDown();
+}
+
 }  // namespace webrtc

talk/app/webrtc/mediastreaminterface.h

@@ -142,9 +142,24 @@ class VideoTrackInterface : public MediaStreamTrackInterface {
 
 // AudioSourceInterface is a reference counted source used for AudioTracks.
 // The same source can be used in multiple AudioTracks.
-// TODO(perkj): Extend this class with necessary methods to allow separate
-// sources for each audio track.
 class AudioSourceInterface : public MediaSourceInterface {
+ public:
+  class AudioObserver {
+   public:
+    virtual void OnSetVolume(double volume) = 0;
+
+   protected:
+    virtual ~AudioObserver() {}
+  };
+
+  // TODO(xians): Makes all the interface pure virtual after Chrome has their
+  // implementations.
+  // Sets the volume to the source. |volume| is in  the range of [0, 10].
+  virtual void SetVolume(double volume) {}
+
+  // Registers/unregisters observer to the audio source.
+  virtual void RegisterAudioObserver(AudioObserver* observer) {}
+  virtual void UnregisterAudioObserver(AudioObserver* observer) {}
 };
 
 // Interface for receiving audio data from a AudioTrack.

talk/app/webrtc/mediastreamprovider.h

@@ -53,6 +53,10 @@ class AudioProviderInterface {
                             const cricket::AudioOptions& options,
                             cricket::AudioRenderer* renderer) = 0;
 
+  // Sets the audio playout volume of a remote audio track with |ssrc|.
+  // |volume| is in the range of [0, 10].
+  virtual void SetAudioPlayoutVolume(uint32 ssrc, double volume) = 0;
+
  protected:
   virtual ~AudioProviderInterface() {}
 };

talk/app/webrtc/mediastreamsignaling.cc

@@ -33,6 +33,7 @@
 #include "talk/app/webrtc/mediastreamproxy.h"
 #include "talk/app/webrtc/mediaconstraintsinterface.h"
 #include "talk/app/webrtc/mediastreamtrackproxy.h"
+#include "talk/app/webrtc/remoteaudiosource.h"
 #include "talk/app/webrtc/remotevideocapturer.h"
 #include "talk/app/webrtc/sctputils.h"
 #include "talk/app/webrtc/videosource.h"
@@ -140,7 +141,7 @@ class RemoteMediaStreamFactory {
   AudioTrackInterface* AddAudioTrack(webrtc::MediaStreamInterface* stream,
                                      const std::string& track_id) {
     return AddTrack<AudioTrackInterface, AudioTrack, AudioTrackProxy>(
-        stream, track_id, static_cast<AudioSourceInterface*>(NULL));
+        stream, track_id, RemoteAudioSource::Create().get());
   }
 
   VideoTrackInterface* AddVideoTrack(webrtc::MediaStreamInterface* stream,

talk/app/webrtc/remoteaudiosource.cc

@@ -0,0 +1,72 @@
+/*
+ * libjingle
+ * Copyright 2014, Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ *  1. Redistributions of source code must retain the above copyright notice,
+ *     this list of conditions and the following disclaimer.
+ *  2. Redistributions in binary form must reproduce the above copyright notice,
+ *     this list of conditions and the following disclaimer in the documentation
+ *     and/or other materials provided with the distribution.
+ *  3. The name of the author may not be used to endorse or promote products
+ *     derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "talk/app/webrtc/remoteaudiosource.h"
+
+#include <algorithm>
+#include <functional>
+
+#include "talk/base/logging.h"
+
+namespace webrtc {
+
+talk_base::scoped_refptr<RemoteAudioSource> RemoteAudioSource::Create() {
+  return new talk_base::RefCountedObject<RemoteAudioSource>();
+}
+
+RemoteAudioSource::RemoteAudioSource() {
+}
+
+RemoteAudioSource::~RemoteAudioSource() {
+  ASSERT(audio_observers_.empty());
+}
+
+MediaSourceInterface::SourceState RemoteAudioSource::state() const {
+  return MediaSourceInterface::kLive;
+}
+
+void RemoteAudioSource::SetVolume(double volume) {
+  ASSERT(volume >= 0 && volume <= 10);
+  for (AudioObserverList::iterator it = audio_observers_.begin();
+       it != audio_observers_.end(); ++it) {
+    (*it)->OnSetVolume(volume);
+  }
+}
+
+void RemoteAudioSource::RegisterAudioObserver(AudioObserver* observer) {
+  ASSERT(observer != NULL);
+  ASSERT(std::find(audio_observers_.begin(), audio_observers_.end(),
+                   observer) == audio_observers_.end());
+  audio_observers_.push_back(observer);
+}
+
+void RemoteAudioSource::UnregisterAudioObserver(AudioObserver* observer) {
+  ASSERT(observer != NULL);
+  audio_observers_.remove(observer);
+}
+
+}  // namespace webrtc

talk/app/webrtc/remoteaudiosource.h

@@ -0,0 +1,66 @@
+/*
+ * libjingle
+ * Copyright 2014, Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ *  1. Redistributions of source code must retain the above copyright notice,
+ *     this list of conditions and the following disclaimer.
+ *  2. Redistributions in binary form must reproduce the above copyright notice,
+ *     this list of conditions and the following disclaimer in the documentation
+ *     and/or other materials provided with the distribution.
+ *  3. The name of the author may not be used to endorse or promote products
+ *     derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef TALK_APP_WEBRTC_REMOTEAUDIOSOURCE_H_
+#define TALK_APP_WEBRTC_REMOTEAUDIOSOURCE_H_
+
+#include <list>
+
+#include "talk/app/webrtc/mediastreaminterface.h"
+#include "talk/app/webrtc/notifier.h"
+
+namespace webrtc {
+
+using webrtc::AudioSourceInterface;
+
+// This class implements the audio source used by the remote audio track.
+class RemoteAudioSource : public Notifier<AudioSourceInterface> {
+ public:
+  // Creates an instance of RemoteAudioSource.
+  static talk_base::scoped_refptr<RemoteAudioSource> Create();
+
+ protected:
+  RemoteAudioSource();
+  virtual ~RemoteAudioSource();
+
+ private:
+  typedef std::list<AudioObserver*> AudioObserverList;
+
+  // MediaSourceInterface implementation.
+  virtual MediaSourceInterface::SourceState state() const OVERRIDE;
+
+  // AudioSourceInterface implementation.
+  virtual void SetVolume(double volume) OVERRIDE;
+  virtual void RegisterAudioObserver(AudioObserver* observer) OVERRIDE;
+  virtual void UnregisterAudioObserver(AudioObserver* observer) OVERRIDE;
+
+  AudioObserverList audio_observers_;
+};
+
+}  // namespace webrtc
+
+#endif  // TALK_APP_WEBRTC_REMOTEAUDIOSOURCE_H_

talk/app/webrtc/statscollector.cc

@@ -78,6 +78,7 @@ const char StatsReport::kStatsValueNameEchoReturnLossEnhancement[] =
 
 const char StatsReport::kStatsValueNameEncodeUsagePercent[] =
     "googEncodeUsagePercent";
+const char StatsReport::kStatsValueNameExpandRate[] = "googExpandRate";
 const char StatsReport::kStatsValueNameFingerprint[] = "googFingerprint";
 const char StatsReport::kStatsValueNameFingerprintAlgorithm[] =
     "googFingerprintAlgorithm";
@@ -221,7 +222,7 @@ void ExtractStats(const cricket::VoiceReceiverInfo& info, StatsReport* report) {
                    info.bytes_rcvd);
   report->AddValue(StatsReport::kStatsValueNameJitterReceived,
                    info.jitter_ms);
-  report->AddValue(StatsReport::kStatsValueNameNetEqExpandRate,
+  report->AddValue(StatsReport::kStatsValueNameExpandRate,
                    talk_base::ToString<float>(info.expand_rate));
   report->AddValue(StatsReport::kStatsValueNamePacketsReceived,
                    info.packets_rcvd);
@@ -569,6 +570,14 @@ std::string StatsCollector::AddOneCertificateReport(
 
   talk_base::scoped_ptr<talk_base::SSLFingerprint> ssl_fingerprint(
       talk_base::SSLFingerprint::Create(digest_algorithm, cert));
+
+  // SSLFingerprint::Create can fail if the algorithm returned by
+  // SSLCertificate::GetSignatureDigestAlgorithm is not supported by the
+  // implementation of SSLCertificate::ComputeDigest.  This currently happens
+  // with MD5- and SHA-224-signed certificates when linked to libNSS.
+  if (!ssl_fingerprint)
+    return std::string();
+
   std::string fingerprint = ssl_fingerprint->GetRfc4572Fingerprint();
 
   talk_base::Buffer der_buffer;

talk/app/webrtc/statscollector_unittest.cc

@@ -302,16 +302,24 @@ class StatsCollectorTest : public testing::Test {
         webrtc::StatsReport::kStatsReportTypeComponent,
         reports,
         webrtc::StatsReport::kStatsValueNameLocalCertificateId);
-    EXPECT_NE(kNotFound, local_certificate_id);
+    if (local_ders.size() > 0) {
-    CheckCertChainReports(reports, local_ders, local_certificate_id);
+      EXPECT_NE(kNotFound, local_certificate_id);
+      CheckCertChainReports(reports, local_ders, local_certificate_id);
+    } else {
+      EXPECT_EQ(kNotFound, local_certificate_id);
+    }
 
     // Check remote certificate chain.
     std::string remote_certificate_id = ExtractStatsValue(
         webrtc::StatsReport::kStatsReportTypeComponent,
         reports,
         webrtc::StatsReport::kStatsValueNameRemoteCertificateId);
-    EXPECT_NE(kNotFound, remote_certificate_id);
+    if (remote_ders.size() > 0) {
-    CheckCertChainReports(reports, remote_ders, remote_certificate_id);
+      EXPECT_NE(kNotFound, remote_certificate_id);
+      CheckCertChainReports(reports, remote_ders, remote_certificate_id);
+    } else {
+      EXPECT_EQ(kNotFound, remote_certificate_id);
+    }
   }
 
   cricket::FakeMediaEngine* media_engine_;
@@ -774,4 +782,21 @@ TEST_F(StatsCollectorTest, NoCertificates) {
   ASSERT_EQ(kNotFound, remote_certificate_id);
 }
 
+// This test verifies that a remote certificate with an unsupported digest
+// algorithm is correctly ignored.
+TEST_F(StatsCollectorTest, UnsupportedDigestIgnored) {
+  // Build a local certificate.
+  std::string local_der = "This is the local der.";
+  talk_base::FakeSSLCertificate local_cert(DerToPem(local_der));
+
+  // Build a remote certificate with an unsupported digest algorithm.
+  std::string remote_der = "This is somebody else's der.";
+  talk_base::FakeSSLCertificate remote_cert(DerToPem(remote_der));
+  remote_cert.set_digest_algorithm("foobar");
+
+  TestCertificateReports(local_cert, std::vector<std::string>(1, local_der),
+                         remote_cert, std::vector<std::string>());
+}
+
+
 }  // namespace

talk/app/webrtc/statstypes.h

@@ -141,6 +141,7 @@ class StatsReport {
   static const char kStatsValueNameEchoDelayStdDev[];
   static const char kStatsValueNameEchoReturnLoss[];
   static const char kStatsValueNameEchoReturnLossEnhancement[];
+  static const char kStatsValueNameExpandRate[];
   static const char kStatsValueNameFirsReceived[];
   static const char kStatsValueNameFirsSent[];
   static const char kStatsValueNameFrameHeightInput[];

talk/app/webrtc/webrtc.scons

@@ -31,6 +31,7 @@ if env.Bit('have_webrtc_voice') and env.Bit('have_webrtc_video'):
       'peerconnectionfactory.cc',
       'peerconnection.cc',
       'portallocatorfactory.cc',
+      'remoteaudiosource.cc',
       'roapmessages.cc',
       'roapsession.cc',
       'roapsignaling.cc',

talk/app/webrtc/webrtcsession.cc

@@ -866,6 +866,18 @@ void WebRtcSession::SetAudioSend(uint32 ssrc, bool enable,
     voice_channel_->SetChannelOptions(options);
 }
 
+void WebRtcSession::SetAudioPlayoutVolume(uint32 ssrc, double volume) {
+  ASSERT(signaling_thread()->IsCurrent());
+  ASSERT(volume >= 0 && volume <= 10);
+  if (!voice_channel_) {
+    LOG(LS_ERROR) << "SetAudioPlayoutVolume: No audio channel exists.";
+    return;
+  }
+
+  if (!voice_channel_->SetOutputScaling(ssrc, volume, volume))
+    ASSERT(false);
+}
+
 bool WebRtcSession::SetCaptureDevice(uint32 ssrc,
                                      cricket::VideoCapturer* camera) {
   ASSERT(signaling_thread()->IsCurrent());

talk/app/webrtc/webrtcsession.h

@@ -165,6 +165,7 @@ class WebRtcSession : public cricket::BaseSession,
   virtual void SetAudioSend(uint32 ssrc, bool enable,
                             const cricket::AudioOptions& options,
                             cricket::AudioRenderer* renderer) OVERRIDE;
+  virtual void SetAudioPlayoutVolume(uint32 ssrc, double volume) OVERRIDE;
 
   // Implements VideoMediaProviderInterface.
   virtual bool SetCaptureDevice(uint32 ssrc,

talk/base/asyncpacketsocket.h

@@ -28,6 +28,7 @@
 #ifndef TALK_BASE_ASYNCPACKETSOCKET_H_
 #define TALK_BASE_ASYNCPACKETSOCKET_H_
 
+#include "talk/base/buffer.h"
 #include "talk/base/dscp.h"
 #include "talk/base/sigslot.h"
 #include "talk/base/socket.h"
@@ -35,6 +36,29 @@
 
 namespace talk_base {
 
+// This structure holds the info needed to update the packet send time header
+// extension, including the information needed to update the authentication tag
+// after changing the value.
+struct PacketTimeUpdateParams {
+  PacketTimeUpdateParams()
+      : rtp_sendtime_extension_id(-1), srtp_auth_tag_len(-1),
+        srtp_packet_index(-1) {
+  }
+
+  int rtp_sendtime_extension_id;  // extension header id present in packet.
+  Buffer srtp_auth_key;           // Authentication key.
+  int srtp_auth_tag_len;          // Authentication tag length.
+  int64 srtp_packet_index;        // Required for Rtp Packet authentication.
+};
+
+// This structure holds meta information for the packet which is about to send
+// over network.
+struct PacketOptions {
+  PacketOptions() : dscp(DSCP_NO_CHANGE) {}
+  DiffServCodePoint dscp;
+  PacketTimeUpdateParams packet_time_params;
+};
+
 // This structure will have the information about when packet is actually
 // received by socket.
 struct PacketTime {

talk/base/fakenetwork.h

@@ -109,10 +109,12 @@ class FakeNetworkManager : public NetworkManagerBase,
         prefix_length = kFakeIPv6NetworkPrefixLength;
       }
       IPAddress prefix = TruncateIP(it->ipaddr(), prefix_length);
+      std::string key = MakeNetworkKey(it->hostname(), prefix, prefix_length);
       scoped_ptr<Network> net(new Network(it->hostname(),
                                           it->hostname(),
                                           prefix,
-                                          prefix_length));
+                                          prefix_length,
+                                          key));
       net->AddIP(it->ipaddr());
       networks.push_back(net.release());
     }

talk/base/fakesslidentity.h

@@ -38,9 +38,12 @@ namespace talk_base {
 
 class FakeSSLCertificate : public talk_base::SSLCertificate {
  public:
-  explicit FakeSSLCertificate(const std::string& data) : data_(data) {}
+  // SHA-1 is the default digest algorithm because it is available in all build
+  // configurations used for unit testing.
+  explicit FakeSSLCertificate(const std::string& data)
+      : data_(data), digest_algorithm_(DIGEST_SHA_1) {}
   explicit FakeSSLCertificate(const std::vector<std::string>& certs)
-      : data_(certs.front()) {
+      : data_(certs.front()), digest_algorithm_(DIGEST_SHA_1) {
     std::vector<std::string>::const_iterator it;
     // Skip certs[0].
     for (it = certs.begin() + 1; it != certs.end(); ++it) {
@@ -58,10 +61,11 @@ class FakeSSLCertificate : public talk_base::SSLCertificate {
     VERIFY(SSLIdentity::PemToDer(kPemTypeCertificate, data_, &der_string));
     der_buffer->SetData(der_string.c_str(), der_string.size());
   }
+  void set_digest_algorithm(const std::string& algorithm) {
+    digest_algorithm_ = algorithm;
+  }
   virtual bool GetSignatureDigestAlgorithm(std::string* algorithm) const {
-    // SHA-1 is chosen because it is available in all build configurations
+    *algorithm = digest_algorithm_;
-    // used for unit testing.
-    *algorithm = DIGEST_SHA_1;
     return true;
   }
   virtual bool ComputeDigest(const std::string &algorithm,
@@ -86,6 +90,7 @@ class FakeSSLCertificate : public talk_base::SSLCertificate {
   }
   std::string data_;
   std::vector<FakeSSLCertificate> certs_;
+  std::string digest_algorithm_;
 };
 
 class FakeSSLIdentity : public talk_base::SSLIdentity {

talk/base/network.cc

@@ -79,16 +79,7 @@ const uint32 kSignalNetworksMessage = 2;
 // Fetch list of networks every two seconds.
 const int kNetworksUpdateIntervalMs = 2000;
 
-
+const int kHighestNetworkPreference = 127;
-// Makes a string key for this network. Used in the network manager's maps.
-// Network objects are keyed on interface name, network prefix and the
-// length of that prefix.
-std::string MakeNetworkKey(const std::string& name, const IPAddress& prefix,
-                           int prefix_length) {
-  std::ostringstream ost;
-  ost << name << "%" << prefix.ToString() << "/" << prefix_length;
-  return ost.str();
-}
 
 bool CompareNetworks(const Network* a, const Network* b) {
   if (a->prefix_length() == b->prefix_length()) {
@@ -99,9 +90,36 @@ bool CompareNetworks(const Network* a, const Network* b) {
   return a->name() < b->name();
 }
 
+bool SortNetworks(const Network* a, const Network* b) {
+  // Network types will be preferred above everything else while sorting
+  // Networks.
+
+  // Networks are sorted first by type.
+  if (a->type() != b->type()) {
+    return a->type() < b->type();
+  }
+
+  // After type, networks are sorted by IP address precedence values
+  // from RFC 3484-bis
+  if (IPAddressPrecedence(a->ip()) != IPAddressPrecedence(b->ip())) {
+    return IPAddressPrecedence(a->ip()) > IPAddressPrecedence(b->ip());
+  }
+
+  // TODO(mallinath) - Add VPN and Link speed conditions while sorting.
+
+  // Networks are sorted last by key.
+  return a->key() > b->key();
+}
 
 }  // namespace
 
+std::string MakeNetworkKey(const std::string& name, const IPAddress& prefix,
+                           int prefix_length) {
+  std::ostringstream ost;
+  ost << name << "%" << prefix.ToString() << "/" << prefix_length;
+  return ost.str();
+}
+
 NetworkManager::NetworkManager() {
 }
 
@@ -180,6 +198,29 @@ void NetworkManagerBase::MergeNetworkList(const NetworkList& new_networks,
     }
   }
   networks_ = merged_list;
+
+  // If the network lists changes, we resort it.
+  if (changed) {
+    std::sort(networks_.begin(), networks_.end(), SortNetworks);
+    // Now network interfaces are sorted, we should set the preference value
+    // for each of the interfaces we are planning to use.
+    // Preference order of network interfaces might have changed from previous
+    // sorting due to addition of higher preference network interface.
+    // Since we have already sorted the network interfaces based on our
+    // requirements, we will just assign a preference value starting with 127,
+    // in decreasing order.
+    int pref = kHighestNetworkPreference;
+    for (NetworkList::const_iterator iter = networks_.begin();
+         iter != networks_.end(); ++iter) {
+      (*iter)->set_preference(pref);
+      if (pref > 0) {
+        --pref;
+      } else {
+        LOG(LS_ERROR) << "Too many network interfaces to handle!";
+        break;
+      }
+    }
+  }
 }
 
 BasicNetworkManager::BasicNetworkManager()
@@ -240,6 +281,7 @@ void BasicNetworkManager::ConvertIfAddrs(struct ifaddrs* interfaces,
         continue;
       }
     }
+
     int prefix_length = CountIPMaskBits(mask);
     prefix = TruncateIP(ip, prefix_length);
     std::string key = MakeNetworkKey(std::string(cursor->ifa_name),
@@ -249,7 +291,8 @@ void BasicNetworkManager::ConvertIfAddrs(struct ifaddrs* interfaces,
       scoped_ptr<Network> network(new Network(cursor->ifa_name,
                                               cursor->ifa_name,
                                               prefix,
-                                              prefix_length));
+                                              prefix_length,
+                                              key));
       network->set_scope_id(scope_id);
       network->AddIP(ip);
       bool ignored = ((cursor->ifa_flags & IFF_LOOPBACK) ||
@@ -386,6 +429,7 @@ bool BasicNetworkManager::CreateNetworks(bool include_ignored,
             continue;
           }
         }
+
         IPAddress prefix;
         int prefix_length = GetPrefix(prefixlist, ip, &prefix);
         std::string key = MakeNetworkKey(name, prefix, prefix_length);
@@ -394,7 +438,8 @@ bool BasicNetworkManager::CreateNetworks(bool include_ignored,
           scoped_ptr<Network> network(new Network(name,
                                                   description,
                                                   prefix,
-                                                  prefix_length));
+                                                  prefix_length,
+                                                  key));
           network->set_scope_id(scope_id);
           network->AddIP(ip);
           bool ignore = ((adapter_addrs->IfType == IF_TYPE_SOFTWARE_LOOPBACK) ||
@@ -561,12 +606,21 @@ void BasicNetworkManager::DumpNetworks(bool include_ignored) {
   }
 }
 
+Network::Network(const std::string& name, const std::string& desc,
+                 const IPAddress& prefix, int prefix_length,
+                 const std::string& key)
+    : name_(name), description_(desc), prefix_(prefix),
+      prefix_length_(prefix_length), key_(key), scope_id_(0), ignored_(false),
+      uniform_numerator_(0), uniform_denominator_(0), exponential_numerator_(0),
+      exponential_denominator_(0), type_(ADAPTER_TYPE_UNKNOWN), preference_(0) {
+}
+
 Network::Network(const std::string& name, const std::string& desc,
                  const IPAddress& prefix, int prefix_length)
     : name_(name), description_(desc), prefix_(prefix),
       prefix_length_(prefix_length), scope_id_(0), ignored_(false),
       uniform_numerator_(0), uniform_denominator_(0), exponential_numerator_(0),
-      exponential_denominator_(0) {
+      exponential_denominator_(0), type_(ADAPTER_TYPE_UNKNOWN), preference_(0) {
 }
 
 std::string Network::ToString() const {
@@ -600,4 +654,5 @@ bool Network::SetIPs(const std::vector<IPAddress>& ips, bool changed) {
   ips_ = ips;
   return changed;
 }
+
 }  // namespace talk_base

talk/base/network.h

@@ -45,9 +45,23 @@ struct ifaddrs;
 namespace talk_base {
 
 class Network;
-class NetworkSession;
 class Thread;
 
+enum AdapterType {
+  // This enum resembles the one in Chromium net::ConnectionType.
+  ADAPTER_TYPE_UNKNOWN = 0,
+  ADAPTER_TYPE_ETHERNET = 1,
+  ADAPTER_TYPE_WIFI = 2,
+  ADAPTER_TYPE_CELLULAR = 3,
+  ADAPTER_TYPE_VPN = 4
+};
+
+// Makes a string key for this network. Used in the network manager's maps.
+// Network objects are keyed on interface name, network prefix and the
+// length of that prefix.
+std::string MakeNetworkKey(const std::string& name, const IPAddress& prefix,
+                           int prefix_length);
+
 // Generic network manager interface. It provides list of local
 // networks.
 class NetworkManager {
@@ -168,7 +182,12 @@ class BasicNetworkManager : public NetworkManagerBase,
 // Represents a Unix-type network interface, with a name and single address.
 class Network {
  public:
-  Network() : prefix_(INADDR_ANY), scope_id_(0) {}
+  Network() : prefix_(INADDR_ANY), scope_id_(0),
+              type_(ADAPTER_TYPE_UNKNOWN) {}
+  Network(const std::string& name, const std::string& description,
+          const IPAddress& prefix, int prefix_length,
+          const std::string& key);
+
   Network(const std::string& name, const std::string& description,
           const IPAddress& prefix, int prefix_length);
 
@@ -184,6 +203,10 @@ class Network {
   // Returns the length, in bits, of this network's prefix.
   int prefix_length() const { return prefix_length_; }
 
+  // |key_| has unique value per network interface. Used in sorting network
+  // interfaces. Key is derived from interface name and it's prefix.
+  std::string key() const { return key_; }
+
   // Returns the Network's current idea of the 'best' IP it has.
   // 'Best' currently means the first one added.
   // TODO: We should be preferring temporary addresses.
@@ -215,27 +238,32 @@ class Network {
   bool ignored() const { return ignored_; }
   void set_ignored(bool ignored) { ignored_ = ignored; }
 
+  AdapterType type() const { return type_; }
+  int preference() const { return preference_; }
+  void set_preference(int preference) { preference_ = preference; }
+
   // Debugging description of this network
   std::string ToString() const;
 
  private:
-  typedef std::vector<NetworkSession*> SessionList;
-
   std::string name_;
   std::string description_;
   IPAddress prefix_;
   int prefix_length_;
+  std::string key_;
   std::vector<IPAddress> ips_;
   int scope_id_;
   bool ignored_;
-  SessionList sessions_;
   double uniform_numerator_;
   double uniform_denominator_;
   double exponential_numerator_;
   double exponential_denominator_;
+  AdapterType type_;
+  int preference_;
 
   friend class NetworkManager;
 };
+
 }  // namespace talk_base
 
 #endif  // TALK_BASE_NETWORK_H_

talk/base/network_unittest.cc

@@ -527,6 +527,33 @@ TEST_F(NetworkTest, TestIPv6Toggle) {
   }
 }
 
+TEST_F(NetworkTest, TestNetworkListSorting) {
+  BasicNetworkManager manager;
+  Network ipv4_network1("test_eth0", "Test Network Adapter 1",
+                        IPAddress(0x12345600U), 24);
+  ipv4_network1.AddIP(IPAddress(0x12345600U));
+
+  IPAddress ip;
+  IPAddress prefix;
+  EXPECT_TRUE(IPFromString("2400:4030:1:2c00:be30:abcd:efab:cdef", &ip));
+  prefix = TruncateIP(ip, 64);
+  Network ipv6_eth1_publicnetwork1_ip1("test_eth1", "Test NetworkAdapter 2",
+                                       prefix, 64);
+  ipv6_eth1_publicnetwork1_ip1.AddIP(ip);
+
+  NetworkManager::NetworkList list;
+  list.push_back(new Network(ipv4_network1));
+  list.push_back(new Network(ipv6_eth1_publicnetwork1_ip1));
+  Network* net1 = list[0];
+  Network* net2 = list[1];
+
+  bool changed = false;
+  MergeNetworkList(manager, list, &changed);
+  ASSERT_TRUE(changed);
+  // After sorting IPv6 network should be higher order than IPv4 networks.
+  EXPECT_TRUE(net1->preference() < net2->preference());
+}
+
 #if defined(POSIX)
 // Verify that we correctly handle interfaces with no address.
 TEST_F(NetworkTest, TestConvertIfAddrsNoAddress) {

talk/base/openssl.h

@@ -0,0 +1,37 @@
+/*
+ * libjingle
+ * Copyright 2013, Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ *  1. Redistributions of source code must retain the above copyright notice,
+ *     this list of conditions and the following disclaimer.
+ *  2. Redistributions in binary form must reproduce the above copyright notice,
+ *     this list of conditions and the following disclaimer in the documentation
+ *     and/or other materials provided with the distribution.
+ *  3. The name of the author may not be used to endorse or promote products
+ *     derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef TALK_BASE_OPENSSL_H_
+#define TALK_BASE_OPENSSL_H_
+
+#include <openssl/ssl.h>
+
+#if (OPENSSL_VERSION_NUMBER < 0x10001000L)
+#error OpenSSL is older than 1.0.1, which is the minimum supported version.
+#endif
+
+#endif  // TALK_BASE_OPENSSL_H_

talk/base/openssladapter.cc

@@ -41,7 +41,6 @@
 #include <openssl/err.h>
 #include <openssl/opensslv.h>
 #include <openssl/rand.h>
-#include <openssl/ssl.h>
 #include <openssl/x509v3.h>
 
 #if HAVE_CONFIG_H
@@ -50,6 +49,7 @@
 
 #include "talk/base/common.h"
 #include "talk/base/logging.h"
+#include "talk/base/openssl.h"
 #include "talk/base/sslroots.h"
 #include "talk/base/stringutils.h"
 
@@ -688,11 +688,7 @@ bool OpenSSLAdapter::VerifyServerName(SSL* ssl, const char* host,
     int extension_nid = OBJ_obj2nid(X509_EXTENSION_get_object(extension));
 
     if (extension_nid == NID_subject_alt_name) {
-#if OPENSSL_VERSION_NUMBER >= 0x10000000L
       const X509V3_EXT_METHOD* meth = X509V3_EXT_get(extension);
-#else
-      X509V3_EXT_METHOD* meth = X509V3_EXT_get(extension);
-#endif
       if (!meth)
         break;
 
@@ -703,12 +699,8 @@ bool OpenSSLAdapter::VerifyServerName(SSL* ssl, const char* host,
       // See http://readlist.com/lists/openssl.org/openssl-users/0/4761.html.
       unsigned char* ext_value_data = extension->value->data;
 
-#if OPENSSL_VERSION_NUMBER >= 0x0090800fL
       const unsigned char **ext_value_data_ptr =
           (const_cast<const unsigned char **>(&ext_value_data));
-#else
-      unsigned char **ext_value_data_ptr = &ext_value_data;
-#endif
 
       if (meth->it) {
         ext_str = ASN1_item_d2i(NULL, ext_value_data_ptr,

talk/base/openssldigest.cc

@@ -30,6 +30,7 @@
 #include "talk/base/openssldigest.h"
 
 #include "talk/base/common.h"
+#include "talk/base/openssl.h"
 
 namespace talk_base {
 
@@ -78,7 +79,6 @@ bool OpenSSLDigest::GetDigestEVP(const std::string& algorithm,
     md = EVP_md5();
   } else if (algorithm == DIGEST_SHA_1) {
     md = EVP_sha1();
-#if OPENSSL_VERSION_NUMBER >= 0x00908000L
   } else if (algorithm == DIGEST_SHA_224) {
     md = EVP_sha224();
   } else if (algorithm == DIGEST_SHA_256) {
@@ -87,7 +87,6 @@ bool OpenSSLDigest::GetDigestEVP(const std::string& algorithm,
     md = EVP_sha384();
   } else if (algorithm == DIGEST_SHA_512) {
     md = EVP_sha512();
-#endif
   } else {
     return false;
   }
@@ -108,7 +107,6 @@ bool OpenSSLDigest::GetDigestName(const EVP_MD* md,
     *algorithm = DIGEST_MD5;
   } else if (md_type == NID_sha1) {
     *algorithm = DIGEST_SHA_1;
-#if OPENSSL_VERSION_NUMBER >= 0x00908000L
   } else if (md_type == NID_sha224) {
     *algorithm = DIGEST_SHA_224;
   } else if (md_type == NID_sha256) {
@@ -117,7 +115,6 @@ bool OpenSSLDigest::GetDigestName(const EVP_MD* md,
     *algorithm = DIGEST_SHA_384;
   } else if (md_type == NID_sha512) {
     *algorithm = DIGEST_SHA_512;
-#endif
   } else {
     algorithm->clear();
     return false;

talk/base/opensslidentity.cc

@@ -32,7 +32,6 @@
 // Must be included first before openssl headers.
 #include "talk/base/win32.h"  // NOLINT
 
-#include <openssl/ssl.h>
 #include <openssl/bio.h>
 #include <openssl/err.h>
 #include <openssl/pem.h>
@@ -43,6 +42,7 @@
 #include "talk/base/checks.h"
 #include "talk/base/helpers.h"
 #include "talk/base/logging.h"
+#include "talk/base/openssl.h"
 #include "talk/base/openssldigest.h"
 
 namespace talk_base {
@@ -66,15 +66,6 @@ static const int CERTIFICATE_WINDOW = -60*60*24;
 static EVP_PKEY* MakeKey() {
   LOG(LS_INFO) << "Making key pair";
   EVP_PKEY* pkey = EVP_PKEY_new();
-#if OPENSSL_VERSION_NUMBER < 0x00908000l
-  // Only RSA_generate_key is available. Use that.
-  RSA* rsa = RSA_generate_key(KEY_LENGTH, 0x10001, NULL, NULL);
-  if (!EVP_PKEY_assign_RSA(pkey, rsa)) {
-    EVP_PKEY_free(pkey);
-    RSA_free(rsa);
-    return NULL;
-  }
-#else
   // RSA_generate_key is deprecated. Use _ex version.
   BIGNUM* exponent = BN_new();
   RSA* rsa = RSA_new();
@@ -89,7 +80,6 @@ static EVP_PKEY* MakeKey() {
   }
   // ownership of rsa struct was assigned, don't free it.
   BN_free(exponent);
-#endif
   LOG(LS_INFO) << "Returning key pair";
   return pkey;
 }

talk/base/opensslstreamadapter.cc

@@ -37,7 +37,6 @@
 #include <openssl/crypto.h>
 #include <openssl/err.h>
 #include <openssl/rand.h>
-#include <openssl/ssl.h>
 #include <openssl/x509v3.h>
 
 #include <vector>
@@ -45,6 +44,7 @@
 #include "talk/base/common.h"
 #include "talk/base/logging.h"
 #include "talk/base/stream.h"
+#include "talk/base/openssl.h"
 #include "talk/base/openssladapter.h"
 #include "talk/base/openssldigest.h"
 #include "talk/base/opensslidentity.h"
@@ -53,15 +53,6 @@
 
 namespace talk_base {
 
-#if (OPENSSL_VERSION_NUMBER >= 0x10001000L)
-#define HAVE_DTLS_SRTP
-#endif
-
-#if (OPENSSL_VERSION_NUMBER >= 0x10000000L)
-#define HAVE_DTLS
-#endif
-
-#ifdef HAVE_DTLS_SRTP
 // SRTP cipher suite table
 struct SrtpCipherMapEntry {
   const char* external_name;
@@ -74,7 +65,6 @@ static SrtpCipherMapEntry SrtpCipherMap[] = {
   {"AES_CM_128_HMAC_SHA1_32", "SRTP_AES128_CM_SHA1_32"},
   {NULL, NULL}
 };
-#endif
 
 //////////////////////////////////////////////////////////////////////
 // StreamBIO
@@ -248,7 +238,6 @@ bool OpenSSLStreamAdapter::ExportKeyingMaterial(const std::string& label,
                                                 bool use_context,
                                                 uint8* result,
                                                 size_t result_len) {
-#ifdef HAVE_DTLS_SRTP
   int i;
 
   i = SSL_export_keying_material(ssl_, result, result_len,
@@ -260,9 +249,6 @@ bool OpenSSLStreamAdapter::ExportKeyingMaterial(const std::string& label,
     return false;
 
   return true;
-#else
-  return false;
-#endif
 }
 
 bool OpenSSLStreamAdapter::SetDtlsSrtpCiphers(
@@ -272,7 +258,6 @@ bool OpenSSLStreamAdapter::SetDtlsSrtpCiphers(
   if (state_ != SSL_NONE)
     return false;
 
-#ifdef HAVE_DTLS_SRTP
   for (std::vector<std::string>::const_iterator cipher = ciphers.begin();
        cipher != ciphers.end(); ++cipher) {
     bool found = false;
@@ -298,13 +283,9 @@ bool OpenSSLStreamAdapter::SetDtlsSrtpCiphers(
 
   srtp_ciphers_ = internal_ciphers;
   return true;
-#else
-  return false;
-#endif
 }
 
 bool OpenSSLStreamAdapter::GetDtlsSrtpCipher(std::string* cipher) {
-#ifdef HAVE_DTLS_SRTP
   ASSERT(state_ == SSL_CONNECTED);
   if (state_ != SSL_CONNECTED)
     return false;
@@ -326,9 +307,6 @@ bool OpenSSLStreamAdapter::GetDtlsSrtpCipher(std::string* cipher) {
   ASSERT(false);  // This should never happen
 
   return false;
-#else
-  return false;
-#endif
 }
 
 int OpenSSLStreamAdapter::StartSSLWithServer(const char* server_name) {
@@ -665,14 +643,12 @@ int OpenSSLStreamAdapter::ContinueSSL() {
 
     case SSL_ERROR_WANT_READ: {
         LOG(LS_VERBOSE) << " -- error want read";
-#ifdef HAVE_DTLS
         struct timeval timeout;
         if (DTLSv1_get_timeout(ssl_, &timeout)) {
           int delay = timeout.tv_sec * 1000 + timeout.tv_usec/1000;
 
           Thread::Current()->PostDelayed(delay, this, MSG_TIMEOUT, 0);
         }
-#endif
       }
       break;
 
@@ -727,9 +703,7 @@ void OpenSSLStreamAdapter::OnMessage(Message* msg) {
   // Process our own messages and then pass others to the superclass
   if (MSG_TIMEOUT == msg->message_id) {
     LOG(LS_INFO) << "DTLS timeout expired";
-#ifdef HAVE_DTLS
     DTLSv1_handle_timeout(ssl_);
-#endif
     ContinueSSL();
   } else {
     StreamInterface::OnMessage(msg);
@@ -740,19 +714,11 @@ SSL_CTX* OpenSSLStreamAdapter::SetupSSLContext() {
   SSL_CTX *ctx = NULL;
 
   if (role_ == SSL_CLIENT) {
-#ifdef HAVE_DTLS
     ctx = SSL_CTX_new(ssl_mode_ == SSL_MODE_DTLS ?
         DTLSv1_client_method() : TLSv1_client_method());
-#else
-    ctx = SSL_CTX_new(TLSv1_client_method());
-#endif
   } else {
-#ifdef HAVE_DTLS
     ctx = SSL_CTX_new(ssl_mode_ == SSL_MODE_DTLS ?
         DTLSv1_server_method() : TLSv1_server_method());
-#else
-    ctx = SSL_CTX_new(TLSv1_server_method());
-#endif
   }
   if (ctx == NULL)
     return NULL;
@@ -771,14 +737,12 @@ SSL_CTX* OpenSSLStreamAdapter::SetupSSLContext() {
   SSL_CTX_set_verify_depth(ctx, 4);
   SSL_CTX_set_cipher_list(ctx, "ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH");
 
-#ifdef HAVE_DTLS_SRTP
   if (!srtp_ciphers_.empty()) {
     if (SSL_CTX_set_tlsext_use_srtp(ctx, srtp_ciphers_.c_str())) {
       SSL_CTX_free(ctx);
       return NULL;
     }
   }
-#endif
 
   return ctx;
 }
@@ -852,27 +816,15 @@ bool OpenSSLStreamAdapter::SSLPostConnectionCheck(SSL* ssl,
 }
 
 bool OpenSSLStreamAdapter::HaveDtls() {
-#ifdef HAVE_DTLS
   return true;
-#else
-  return false;
-#endif
 }
 
 bool OpenSSLStreamAdapter::HaveDtlsSrtp() {
-#ifdef HAVE_DTLS_SRTP
   return true;
-#else
-  return false;
-#endif
 }
 
 bool OpenSSLStreamAdapter::HaveExporter() {
-#ifdef HAVE_DTLS_SRTP
   return true;
-#else
-  return false;
-#endif
 }
 
 }  // namespace talk_base

talk/base/physicalsocketserver.cc

@@ -541,6 +541,8 @@ class PhysicalSocket : public AsyncSocket, public sigslot::has_slots<> {
       case OPT_DSCP:
         LOG(LS_WARNING) << "Socket::OPT_DSCP not supported.";
         return -1;
+      case OPT_RTP_SENDTIME_EXTN_ID:
+        return -1;  // No logging is necessary as this not a OS socket option.
       default:
         ASSERT(false);
         return -1;

talk/base/socket.h

@@ -185,7 +185,10 @@ class Socket {
     OPT_SNDBUF,      // send buffer size
     OPT_NODELAY,     // whether Nagle algorithm is enabled
     OPT_IPV6_V6ONLY, // Whether the socket is IPv6 only.
-    OPT_DSCP         // DSCP code
+    OPT_DSCP,        // DSCP code
+    OPT_RTP_SENDTIME_EXTN_ID,  // This is a non-traditional socket option param.
+                               // This is specific to libjingle and will be used
+                               // if SendTime option is needed at socket level.
   };
   virtual int GetOption(Option opt, int* value) = 0;
   virtual int SetOption(Option opt, int value) = 0;

talk/base/thread_unittest.cc

@@ -339,7 +339,7 @@ class AsyncInvokeTest : public testing::Test {
   Thread* expected_thread_;
 };
 
-TEST_F(AsyncInvokeTest, FireAndForget) {
+TEST_F(AsyncInvokeTest, DISABLED_FireAndForget) {
   AsyncInvoker invoker;
   // Create and start the thread.
   Thread thread;
@@ -350,7 +350,7 @@ TEST_F(AsyncInvokeTest, FireAndForget) {
   EXPECT_TRUE_WAIT(called, kWaitTimeout);
 }
 
-TEST_F(AsyncInvokeTest, WithCallback) {
+TEST_F(AsyncInvokeTest, DISABLED_WithCallback) {
   AsyncInvoker invoker;
   // Create and start the thread.
   Thread thread;
@@ -379,7 +379,7 @@ TEST_F(AsyncInvokeTest, DISABLED_CancelInvoker) {
   EXPECT_EQ(0, int_value_);
 }
 
-TEST_F(AsyncInvokeTest, CancelCallingThread) {
+TEST_F(AsyncInvokeTest, DISABLED_CancelCallingThread) {
   AsyncInvoker invoker;
   { // Create and start the thread.
     Thread thread;
@@ -396,7 +396,7 @@ TEST_F(AsyncInvokeTest, CancelCallingThread) {
   EXPECT_EQ(0, int_value_);
 }
 
-TEST_F(AsyncInvokeTest, KillInvokerBeforeExecute) {
+TEST_F(AsyncInvokeTest, DISABLED_KillInvokerBeforeExecute) {
   Thread thread;
   thread.Start();
   {
@@ -413,7 +413,7 @@ TEST_F(AsyncInvokeTest, KillInvokerBeforeExecute) {
   EXPECT_EQ(0, int_value_);
 }
 
-TEST_F(AsyncInvokeTest, Flush) {
+TEST_F(AsyncInvokeTest, DISABLED_Flush) {
   AsyncInvoker invoker;
   bool flag1 = false;
   bool flag2 = false;
@@ -431,7 +431,7 @@ TEST_F(AsyncInvokeTest, Flush) {
   EXPECT_TRUE(flag2);
 }
 
-TEST_F(AsyncInvokeTest, FlushWithIds) {
+TEST_F(AsyncInvokeTest, DISABLED_FlushWithIds) {
   AsyncInvoker invoker;
   bool flag1 = false;
   bool flag2 = false;

talk/libjingle.gyp

@@ -715,6 +715,7 @@
           'conditions': [
             ['OS!="ios"', {
               'sources': [
+                'base/openssl.h',
                 'base/openssladapter.cc',
                 'base/openssladapter.h',
                 'base/openssldigest.cc',
@@ -1175,6 +1176,8 @@
         'app/webrtc/portallocatorfactory.cc',
         'app/webrtc/portallocatorfactory.h',
         'app/webrtc/proxy.h',
+        'app/webrtc/remoteaudiosource.cc',
+        'app/webrtc/remoteaudiosource.h',
         'app/webrtc/remotevideocapturer.cc',
         'app/webrtc/remotevideocapturer.h',
         'app/webrtc/sctputils.cc',

talk/media/base/videoadapter.cc

@@ -183,7 +183,8 @@ void VideoAdapter::SetInputFormat(const VideoFormat& format) {
   output_format_.interval = talk_base::_max(
       output_format_.interval, input_format_.interval);
   if (old_input_interval != input_format_.interval) {
-    LOG(LS_INFO) << "VAdapt Input Interval: " << input_format_.interval;
+    LOG(LS_INFO) << "VAdapt input interval changed from "
+      << old_input_interval << " to " << input_format_.interval;
   }
 }
 
@@ -218,7 +219,8 @@ void VideoAdapter::SetOutputFormat(const VideoFormat& format) {
   output_format_.interval = talk_base::_max(
       output_format_.interval, input_format_.interval);
   if (old_output_interval != output_format_.interval) {
-    LOG(LS_INFO) << "VAdapt Output Interval: " << output_format_.interval;
+    LOG(LS_INFO) << "VAdapt output interval changed from "
+      << old_output_interval << " to " << output_format_.interval;
   }
 }
 
@@ -283,16 +285,12 @@ bool VideoAdapter::AdaptFrame(const VideoFrame* in_frame,
   }
   if (should_drop) {
     // Show VAdapt log every 90 frames dropped. (3 seconds)
-    // TODO(fbarchard): Consider GetLogSeverity() to change interval to less
+    if ((frames_in_ - frames_out_) % 90 == 0) {
-    // for LS_VERBOSE and more for LS_INFO.
-    bool show = (frames_in_ - frames_out_) % 90 == 0;
-
-    if (show) {
       // TODO(fbarchard): Reduce to LS_VERBOSE when adapter info is not needed
       // in default calls.
-      LOG(LS_INFO) << "VAdapt Drop Frame: " << frames_scaled_
+      LOG(LS_INFO) << "VAdapt Drop Frame: scaled " << frames_scaled_
-                   << " / " << frames_out_
+                   << " / out " << frames_out_
-                   << " / " << frames_in_
+                   << " / in " << frames_in_
                    << " Changes: " << adaption_changes_
                    << " Input: " << in_frame->GetWidth()
                    << "x" << in_frame->GetHeight()
@@ -344,9 +342,9 @@ bool VideoAdapter::AdaptFrame(const VideoFrame* in_frame,
   if (show) {
     // TODO(fbarchard): Reduce to LS_VERBOSE when adapter info is not needed
     // in default calls.
-    LOG(LS_INFO) << "VAdapt Frame: " << frames_scaled_
+    LOG(LS_INFO) << "VAdapt Frame: scaled " << frames_scaled_
-                 << " / " << frames_out_
+                 << " / out " << frames_out_
-                 << " / " << frames_in_
+                 << " / in " << frames_in_
                  << " Changes: " << adaption_changes_
                  << " Input: " << in_frame->GetWidth()
                  << "x" << in_frame->GetHeight()

talk/media/sctp/sctpdataengine.cc

@@ -237,8 +237,10 @@ static int OnSctpInboundPacket(struct socket* sock, union sctp_sockstore addr,
 // Set the initial value of the static SCTP Data Engines reference count.
 int SctpDataEngine::usrsctp_engines_count = 0;
 
-SctpDataEngine::SctpDataEngine() {
+void SctpDataEngine::AddRefEngine() {
+  LOG(LS_VERBOSE) << "usrsctp_engines_count:" << usrsctp_engines_count;
   if (usrsctp_engines_count == 0) {
+    LOG(LS_INFO) << "SctpDataEngine: Initializing usrsctp";
     // First argument is udp_encapsulation_port, which is not releveant for our
     // AF_CONN use of sctp.
     usrsctp_init(0, cricket::OnSctpOutboundPacket, debug_sctp_printf);
@@ -276,26 +278,25 @@ SctpDataEngine::SctpDataEngine() {
         cricket::kMaxSctpSid);
   }
   usrsctp_engines_count++;
+}
 
-  // We don't put in a codec because we don't want one offered when we
+void SctpDataEngine::ReleaseEngine() {
-  // use the hybrid data engine.
+  usrsctp_engines_count--;
-  // codecs_.push_back(cricket::DataCodec( kGoogleSctpDataCodecId,
+  if (usrsctp_engines_count == 0) {
-  // kGoogleSctpDataCodecName, 0));
+    LOG(LS_INFO) << "SctpDataEngine: Shutting down";
+    if (usrsctp_finish() != 0) {
+      LOG_ERRNO(LS_ERROR) << "SctpDataEngine: usrsctp_finish failed: ";
+    }
+  }
+  LOG(LS_VERBOSE) << "usrsctp_engines_count:" << usrsctp_engines_count;
+}
+
+SctpDataEngine::SctpDataEngine() {
+  AddRefEngine();
 }
 
 SctpDataEngine::~SctpDataEngine() {
-  // TODO(ldixon): There is currently a bug in teardown of usrsctp that blocks
+  ReleaseEngine();
-  // indefintely if a finish call made too soon after close calls. So teardown
-  // has been skipped. Once the bug is fixed, retest and enable teardown.
-  // Tracked in webrtc issue 2749.
-  //
-  // usrsctp_engines_count--;
-  // LOG(LS_VERBOSE) << "usrsctp_engines_count:" << usrsctp_engines_count;
-  // if (usrsctp_engines_count == 0) {
-  //   if (usrsctp_finish() != 0) {
-  //     LOG(LS_WARNING) << "usrsctp_finish.";
-  //   }
-  // }
 }
 
 DataMediaChannel* SctpDataEngine::CreateChannel(
@@ -314,10 +315,12 @@ SctpDataMediaChannel::SctpDataMediaChannel(talk_base::Thread* thread)
       sending_(false),
       receiving_(false),
       debug_name_("SctpDataMediaChannel") {
+  SctpDataEngine::AddRefEngine();
 }
 
 SctpDataMediaChannel::~SctpDataMediaChannel() {
   CloseSctpSocket();
+  SctpDataEngine::ReleaseEngine();
 }
 
 sockaddr_conn SctpDataMediaChannel::GetSctpSockAddr(int port) {

talk/media/sctp/sctpdataengine.h

@@ -91,6 +91,12 @@ class SctpDataEngine : public DataEngineInterface {
 
   virtual const std::vector<DataCodec>& data_codecs() { return codecs_; }
 
+  // Manages the lifetime of the usrsctp library data.  Each SctpDataEngine
+  // and SctpDataMediaChannel AddRefEngine the library at construction, and
+  // ReleaseEngine at shutdown.
+  static void AddRefEngine();
+  static void ReleaseEngine();
+
  private:
   static int usrsctp_engines_count;
   std::vector<DataCodec> codecs_;

talk/media/webrtc/webrtcvideoengine.cc

@@ -2632,6 +2632,15 @@ bool WebRtcVideoMediaChannel::SetSendRtpHeaderExtensions(
       return false;
     }
   }
+
+  if (send_time_extension) {
+    // For video RTP packets, we would like to update AbsoluteSendTimeHeader
+    // Extension closer to the network, @ socket level before sending.
+    // Pushing the extension id to socket layer.
+    MediaChannel::SetOption(NetworkInterface::ST_RTP,
+                            talk_base::Socket::OPT_RTP_SENDTIME_EXTN_ID,
+                            send_time_extension->id);
+  }
   return true;
 }
 

talk/p2p/base/candidate.h

@@ -33,6 +33,7 @@
 #include <string>
 #include <sstream>
 #include <iomanip>
+
 #include "talk/base/basictypes.h"
 #include "talk/base/socketaddress.h"
 #include "talk/p2p/base/constants.h"
@@ -163,13 +164,30 @@ class Candidate {
     return ToStringInternal(true);
   }
 
-  uint32 GetPriority(uint32 type_preference) const {
+  uint32 GetPriority(uint32 type_preference,
+                     int network_adapter_preference) const {
     // RFC 5245 - 4.1.2.1.
     // priority = (2^24)*(type preference) +
     //            (2^8)*(local preference) +
     //            (2^0)*(256 - component ID)
+
+    // |local_preference| length is 2 bytes, 0-65535 inclusive.
+    // In our implemenation we will partion local_preference into
+    //              0                 1
+    //       0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
+    //      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+    //      |  NIC Pref     |    Addr Pref  |
+    //      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+    // NIC Type - Type of the network adapter e.g. 3G/Wifi/Wired.
+    // Addr Pref - Address preference value as per RFC 3484.
+    // local preference is calculated as - NIC Type << 8 | Addr_Pref.
+
     int addr_pref = IPAddressPrecedence(address_.ipaddr());
-    return (type_preference << 24) | (addr_pref << 8) | (256 - component_);
+    int local_preference = (network_adapter_preference << 8) | addr_pref;
+
+    return (type_preference << 24) |
+           (local_preference << 8) |
+           (256 - component_);
   }
 
  private:
@@ -177,9 +195,9 @@ class Candidate {
     std::ostringstream ost;
     std::string address = sensitive ? address_.ToSensitiveString() :
                                       address_.ToString();
-    ost << "Cand[" << id_ << ":" << component_ << ":"
+    ost << "Cand[" << foundation_ << ":" << component_ << ":"
-        << type_ << ":" << protocol_ << ":"
+        << protocol_ << ":" << priority_ << ":"
-        << network_name_ << ":" << address << ":"
+        << address << ":" << type_ << ":" << related_address_ << ":"
         << username_ << ":" << password_ << "]";
     return ost.str();
   }

talk/p2p/base/p2ptransportchannel.cc

@@ -493,7 +493,8 @@ void P2PTransportChannel::OnUnknownAddress(
         port->Network()->name(), 0U,
         talk_base::ToString<uint32>(talk_base::ComputeCrc32(id)));
     new_remote_candidate.set_priority(
-        new_remote_candidate.GetPriority(ICE_TYPE_PREFERENCE_SRFLX));
+        new_remote_candidate.GetPriority(ICE_TYPE_PREFERENCE_SRFLX,
+                                         port->Network()->preference()));
   }
 
   if (port->IceProtocol() == ICEPROTO_RFC5245) {

talk/p2p/base/p2ptransportchannel_unittest.cc

@@ -1559,8 +1559,11 @@ TEST_F(P2PTransportChannelMultihomedTest, DISABLED_TestBasic) {
 // Test that we can quickly switch links if an interface goes down.
 TEST_F(P2PTransportChannelMultihomedTest, TestFailover) {
   AddAddress(0, kPublicAddrs[0]);
-  AddAddress(1, kPublicAddrs[1]);
+  // Adding alternate address will make sure |kPublicAddrs| has the higher
+  // priority than others. This is due to FakeNetwork::AddInterface method.
   AddAddress(1, kAlternateAddrs[1]);
+  AddAddress(1, kPublicAddrs[1]);
+
   // Use only local ports for simplicity.
   SetAllocatorFlags(0, kOnlyLocalPorts);
   SetAllocatorFlags(1, kOnlyLocalPorts);

talk/p2p/base/port.cc

@@ -258,7 +258,7 @@ void Port::AddAddress(const talk_base::SocketAddress& address,
   c.set_type(type);
   c.set_protocol(protocol);
   c.set_address(address);
-  c.set_priority(c.GetPriority(type_preference));
+  c.set_priority(c.GetPriority(type_preference, network_->preference()));
   c.set_username(username_fragment());
   c.set_password(password_);
   c.set_network_name(network_->name());

talk/p2p/client/portallocator_unittest.cc

@@ -53,8 +53,8 @@ using talk_base::Thread;
 static const SocketAddress kClientAddr("11.11.11.11", 0);
 static const SocketAddress kClientIPv6Addr(
     "2401:fa00:4:1000:be30:5bff:fee5:c3", 0);
+static const SocketAddress kClientAddr2("22.22.22.22", 0);
 static const SocketAddress kNatAddr("77.77.77.77", talk_base::NAT_SERVER_PORT);
-static const SocketAddress kRemoteClientAddr("22.22.22.22", 0);
 static const SocketAddress kStunAddr("99.99.99.1", cricket::STUN_SERVER_PORT);
 static const SocketAddress kRelayUdpIntAddr("99.99.99.2", 5000);
 static const SocketAddress kRelayUdpExtAddr("99.99.99.3", 5001);
@@ -492,6 +492,23 @@ TEST_F(PortAllocatorTest, TestGetAllPortsNoUdpAllowed) {
   EXPECT_TRUE_WAIT(candidate_allocation_done_, 9000);
 }
 
+TEST_F(PortAllocatorTest, TestCandidatePriorityOfMultipleInterfaces) {
+  AddInterface(kClientAddr);
+  AddInterface(kClientAddr2);
+  // Allocating only host UDP ports. This is done purely for testing
+  // convenience.
+  allocator().set_flags(cricket::PORTALLOCATOR_DISABLE_TCP |
+                        cricket::PORTALLOCATOR_DISABLE_STUN |
+                        cricket::PORTALLOCATOR_DISABLE_RELAY);
+  EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
+  session_->StartGettingPorts();
+  EXPECT_TRUE_WAIT(candidate_allocation_done_, kDefaultAllocationTimeout);
+  ASSERT_EQ(2U, candidates_.size());
+  EXPECT_EQ(2U, ports_.size());
+  // Candidates priorities should be different.
+  EXPECT_NE(candidates_[0].priority(), candidates_[1].priority());
+}
+
 // Test to verify ICE restart process.
 TEST_F(PortAllocatorTest, TestGetAllPortsRestarts) {
   AddInterface(kClientAddr);