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Makes possible making video call without audio, and sharing the desktop from machine without audio device.

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Updates zrtp to use the first available stream as master, if more than one stream use the audio one.
cusax-fix
Damian Minkov 15 years ago
parent 9f8008254f
commit 4e39c83ba9
9 changed files with 264 additions and 42 deletions
Show Stats Download Patch File Download Diff File

35
src/net/java/sip/communicator/impl/neomedia/ZrtpControlImpl.java
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@ -51,6 +51,11 @@ public static enum ZRTPCustomInfoCodes
*/
private AbstractRTPConnector zrtpConnector = null;
/**
* Whether current is master session.
*/
private boolean masterSession = false;
/**
* Creates the control.
*/
@ -139,14 +144,25 @@ public ZRTPTransformEngine getTransformEngine()
}
/**
* Starts and enables zrtp in the stream holding this control.
* @param masterSession whether this stream is master for the current
* media session.
* When in multistream mode, enables the master session.
* @param masterSession whether current control, controls the master session.
*/
public void start(boolean masterSession)
public void setMasterSession(boolean masterSession)
{
// by default its not master, change only if set to be master
// sometimes (jingle) streams are re-initing and
// we must reuse old value (true) event that false is submitted
if(masterSession)
this.masterSession = masterSession;
}
boolean zrtpAutoStart = false;
/**
* Starts and enables zrtp in the stream holding this control.
* @param mediaType the media type of the stream this control controls.
*/
public void start(MediaType mediaType)
{
boolean zrtpAutoStart;
// ZRTP engine initialization
ZRTPTransformEngine engine = getTransformEngine();
@ -170,7 +186,10 @@ public void start(boolean masterSession)
// we know that audio is considered as master for zrtp
securityEventManager.setSessionType(
SecurityEventManager.AUDIO_SESSION);
mediaType.equals(MediaType.AUDIO) ?
SecurityEventManager.AUDIO_SESSION
: SecurityEventManager.VIDEO_SESSION
);
}
else
{
@ -180,7 +199,9 @@ public void start(boolean masterSession)
// initially engine has value enableZrtp = false
zrtpAutoStart = zrtpEngine.isEnableZrtp();
securityEventManager.setSessionType(
SecurityEventManager.VIDEO_SESSION);
mediaType.equals(MediaType.AUDIO) ?
SecurityEventManager.AUDIO_SESSION
: SecurityEventManager.VIDEO_SESSION);
}
// tells the engine whether to autostart(enable)

8
src/net/java/sip/communicator/impl/neomedia/notify/PortAudioClipImpl.java
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@ -193,12 +193,16 @@ private boolean runOnceInPlayThread(
* If the user has configured PortAudio to use no notification device,
* don't try to play this clip.
*/
MediaLocator rendererLocator
CaptureDeviceInfo audioNotifyDeviceInfo
= audioNotifier
.getDeviceConfiguration().getAudioNotifyDevice().getLocator();
.getDeviceConfiguration().getAudioNotifyDevice();
if(audioNotifyDeviceInfo == null)
return false;
MediaLocator rendererLocator = audioNotifyDeviceInfo.getLocator();
if (rendererLocator == null)
return false;
renderer.setLocator(rendererLocator);
AudioInputStream audioStream = null;

17
src/net/java/sip/communicator/impl/neomedia/transform/sdes/SDesControlImpl.java
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@ -14,9 +14,9 @@
import net.java.sip.communicator.impl.neomedia.*;
import net.java.sip.communicator.impl.neomedia.transform.*;
import net.java.sip.communicator.impl.neomedia.transform.zrtp.*;
import net.java.sip.communicator.service.neomedia.*;
import net.java.sip.communicator.service.neomedia.event.*;
import net.java.sip.communicator.service.protocol.event.*;
/**
* Default implementation of {@link SDesControl} that supports the crypto suites
@ -93,12 +93,19 @@ public boolean getSecureCommunicationStatus()
return engine != null;
}
public void start(boolean masterSession)
/**
* Not used.
* @param masterSession not used.
*/
public void setMasterSession(boolean masterSession)
{}
public void start(MediaType type)
{
srtpListener.securityTurnedOn(
masterSession ?
CallPeerSecurityStatusEvent.AUDIO_SESSION :
CallPeerSecurityStatusEvent.VIDEO_SESSION,
type.equals(MediaType.AUDIO) ?
SecurityEventManager.AUDIO_SESSION
: SecurityEventManager.VIDEO_SESSION,
selectedInAttribute.getCryptoSuite().encode(), this);
}

56
src/net/java/sip/communicator/impl/protocol/jabber/CallPeerMediaHandlerGTalkImpl.java
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@ -230,7 +230,8 @@ public RtpDescriptionPacketExtension generateSessionAccept(
List<PayloadTypePacketExtension> lst = localContentMap.get("audio");
description.setNamespace(SessionIQProvider.GTALK_AUDIO_NAMESPACE);
boolean masterStreamSet = false;
for(MediaType mediaType : MediaType.values())
{
MediaFormat format = null;
@ -291,8 +292,27 @@ public RtpDescriptionPacketExtension generateSessionAccept(
new ArrayList<RTPExtension>();
MediaDirection direction = MediaDirection.SENDRECV;
boolean masterStream = false;
// if we have more than one stream, lets the audio be the master
if(!masterStreamSet)
{
if(MediaType.values().length > 1)
{
if(mediaType.equals(MediaType.AUDIO))
{
masterStream = true;
masterStreamSet = true;
}
}
else
{
masterStream = true;
masterStreamSet = true;
}
}
initStream(mediaName, connector, dev, format, target,
direction, rtpExtensions);
direction, rtpExtensions, masterStream);
}
return description;
@ -319,6 +339,7 @@ public void processAnswer(RtpDescriptionPacketExtension answer)
{
List<PayloadTypePacketExtension> lst = answer.getPayloadTypes();
boolean masterStreamSet = true;
for(MediaType mediaType : MediaType.values())
{
String ns = getNamespaceForMediaType(mediaType);
@ -360,8 +381,27 @@ public void processAnswer(RtpDescriptionPacketExtension answer)
List<RTPExtension> rtpExtensions = new ArrayList<RTPExtension>();
MediaDirection direction = MediaDirection.SENDRECV;
boolean masterStream = false;
// if we have more than one stream, lets the audio be the master
if(!masterStreamSet)
{
if(MediaType.values().length > 1)
{
if(mediaType.equals(MediaType.AUDIO))
{
masterStream = true;
masterStreamSet = true;
}
}
else
{
masterStream = true;
masterStreamSet = true;
}
}
initStream(mediaName, connector, dev, format, target,
direction, rtpExtensions);
direction, rtpExtensions, masterStream);
}
}
@ -565,8 +605,7 @@ else if(mediaType == MediaType.VIDEO)
* Create list of payload types for device.
*
* @param supportedFormats supported formats of a device
* @param direction direction
* @param supportedExtensions supported RTP extensions
* @param name name of payload type
* @return list of payload types for this device
*/
private List<PayloadTypePacketExtension> createPayloadTypesForOffer(
@ -663,6 +702,7 @@ private void wrapupConnectivityEstablishment()
* stream to use (i.e. sendonly, sendrecv, recvonly, or inactive).
* @param rtpExtensions the list of <tt>RTPExtension</tt>s that should be
* enabled for this stream.
* @param masterStream whether the stream to be used as master if secured
*
* @return the newly created <tt>MediaStream</tt>.
*
@ -675,7 +715,8 @@ protected MediaStream initStream(String streamName,
MediaFormat format,
MediaStreamTarget target,
MediaDirection direction,
List<RTPExtension> rtpExtensions)
List<RTPExtension> rtpExtensions,
boolean masterStream)
throws OperationFailedException
{
if(format instanceof VideoMediaFormat)
@ -696,7 +737,8 @@ protected MediaStream initStream(String streamName,
format,
target,
direction,
rtpExtensions);
rtpExtensions,
masterStream);
if(stream != null)
stream.setName(streamName);

114
src/net/java/sip/communicator/impl/protocol/jabber/CallPeerMediaHandlerJabberImpl.java
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@ -193,6 +193,7 @@ public ContentPacketExtension getLocalContent(String contentType)
* stream to use (i.e. sendonly, sendrecv, recvonly, or inactive).
* @param rtpExtensions the list of <tt>RTPExtension</tt>s that should be
* enabled for this stream.
* @param masterStream whether the stream to be used as master if secured
*
* @return the newly created <tt>MediaStream</tt>.
*
@ -205,7 +206,8 @@ protected MediaStream initStream(String streamName,
MediaFormat format,
MediaStreamTarget target,
MediaDirection direction,
List<RTPExtension> rtpExtensions)
List<RTPExtension> rtpExtensions,
boolean masterStream)
throws OperationFailedException
{
MediaStream stream
@ -215,7 +217,8 @@ protected MediaStream initStream(String streamName,
format,
target,
direction,
rtpExtensions);
rtpExtensions,
masterStream);
if(stream != null)
stream.setName(streamName);
@ -457,10 +460,27 @@ public Iterable<ContentPacketExtension> generateSessionAccept()
//user answered an incoming call so we go through whatever content
//entries we are initializing and init their corresponding streams
// First parse content so we know how may streams,
// and what type of content we have
Map<ContentPacketExtension,
RtpDescriptionPacketExtension> contents
= new HashMap<ContentPacketExtension,
RtpDescriptionPacketExtension>();
for(ContentPacketExtension ourContent : sessAccept)
{
RtpDescriptionPacketExtension description
= JingleUtils.getRtpDescription(ourContent);
= JingleUtils.getRtpDescription(ourContent);
contents.put(ourContent, description);
}
boolean masterStreamSet = false;
for(Map.Entry<ContentPacketExtension, RtpDescriptionPacketExtension> en
: contents.entrySet())
{
ContentPacketExtension ourContent = en.getKey();
RtpDescriptionPacketExtension description = en.getValue();
MediaType type = MediaType.parseString(description.getMedia());
// stream connector
@ -532,9 +552,28 @@ public Iterable<ContentPacketExtension> generateSessionAccept()
}
}
boolean masterStream = false;
// if we have more than one stream, lets the audio be the master
if(!masterStreamSet)
{
if(contents.size() > 1)
{
if(type.equals(MediaType.AUDIO))
{
masterStream = true;
masterStreamSet = true;
}
}
else
{
masterStream = true;
masterStreamSet = true;
}
}
// create the corresponding stream...
initStream(ourContent.getName(), connector, dev, format, target,
direction, rtpExtensions);
direction, rtpExtensions, masterStream);
// if remote peer requires inputevt, notify UI to capture mouse
// and keyboard events
@ -887,12 +926,37 @@ public void reinitAllContents()
throws OperationFailedException,
IllegalArgumentException
{
boolean masterStreamSet = false;
for(String key : remoteContentMap.keySet())
{
ContentPacketExtension ext = remoteContentMap.get(key);
boolean masterStream = false;
// if we have more than one stream, lets the audio be the master
if(!masterStreamSet)
{
RtpDescriptionPacketExtension description
= JingleUtils.getRtpDescription(ext);
MediaType mediaType
= MediaType.parseString( description.getMedia() );
if(remoteContentMap.size() > 1)
{
if(mediaType.equals(MediaType.AUDIO))
{
masterStream = true;
masterStreamSet = true;
}
}
else
{
masterStream = true;
masterStreamSet = true;
}
}
if(ext != null)
processContent(ext, false);
processContent(ext, false, masterStream);
}
}
@ -924,13 +988,13 @@ public void reinitContent(
{
if(modify)
{
processContent(content, modify);
processContent(content, modify, false);
remoteContentMap.put(name, content);
}
else
{
ext.setSenders(content.getSenders());
processContent(ext, modify);
processContent(ext, modify, false);
remoteContentMap.put(name, ext);
}
}
@ -982,6 +1046,7 @@ private void removeContent(
*
* @param content a <tt>ContentPacketExtension</tt>
* @param modify if it correspond to a content-modify for resolution change
* @param masterStream whether the stream to be used as master
* @throws OperationFailedException if we fail to handle <tt>content</tt>
* for reasons like failing to initialize media devices or streams.
* @throws IllegalArgumentException if there's a problem with the syntax or
@ -991,7 +1056,8 @@ private void removeContent(
* in this operation can synchronize to the mediaHandler instance to wait
* processing to stop (method setState in CallPeer).
*/
private void processContent(ContentPacketExtension content, boolean modify)
private void processContent(ContentPacketExtension content, boolean modify,
boolean masterStream)
throws OperationFailedException,
IllegalArgumentException
{
@ -1114,7 +1180,8 @@ private void processContent(ContentPacketExtension content, boolean modify)
// create the corresponding stream...
initStream(content.getName(), connector, dev,
supportedFormats.get(0), target, direction, rtpExtensions);
supportedFormats.get(0), target, direction, rtpExtensions,
masterStream);
}
/**
@ -1141,12 +1208,37 @@ public void processAnswer(List<ContentPacketExtension> answer)
* information compatible with that carried in transport-info.
*/
processTransportInfo(answer);
boolean masterStreamSet = false;
for (ContentPacketExtension content : answer)
{
remoteContentMap.put(content.getName(), content);
processContent(content, false);
boolean masterStream = false;
// if we have more than one stream, lets the audio be the master
if(!masterStreamSet)
{
RtpDescriptionPacketExtension description
= JingleUtils.getRtpDescription(content);
MediaType mediaType
= MediaType.parseString( description.getMedia() );
if(answer.size() > 1)
{
if(mediaType.equals(MediaType.AUDIO))
{
masterStream = true;
masterStreamSet = true;
}
}
else
{
masterStream = true;
masterStreamSet = true;
}
}
processContent(content, false, masterStream);
}
}

46
src/net/java/sip/communicator/impl/protocol/sip/CallPeerMediaHandlerSipImpl.java
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@ -415,6 +415,7 @@ private Vector<MediaDescription> createMediaDescriptionsForAnswer(
.getAccountPropertyInt(ProtocolProviderFactory.SAVP_OPTION,
ProtocolProviderFactory.SAVP_OFF);
boolean masterStreamSet = false;
List<MediaType> seenMediaTypes = new ArrayList<MediaType>();
for (MediaDescription mediaDescription : remoteDescriptions)
{
@ -590,7 +591,28 @@ private Vector<MediaDescription> createMediaDescriptionsForAnswer(
// create the corresponding stream...
MediaFormat fmt = findMediaFormat(remoteFormats,
mutuallySupportedFormats.get(0));
initStream(connector, dev, fmt, target, direction, rtpExtensions);
boolean masterStream = false;
// if we have more than one stream, lets the audio be the master
if(!masterStreamSet)
{
if(remoteDescriptions.size() > 1)
{
if(mediaType.equals(MediaType.AUDIO))
{
masterStream = true;
masterStreamSet = true;
}
}
else
{
masterStream = true;
masterStreamSet = true;
}
}
initStream(connector, dev, fmt, target, direction, rtpExtensions,
masterStream);
// create the answer description
answerDescriptions.add(md);
@ -847,6 +869,7 @@ private synchronized void processAnswer(SessionDescription answer)
this.setCallInfoURL(SdpUtils.getCallInfoURL(answer));
boolean masterStreamSet = false;
List<MediaType> seenMediaTypes = new ArrayList<MediaType>();
for (MediaDescription mediaDescription : remoteDescriptions)
{
@ -992,9 +1015,28 @@ private synchronized void processAnswer(SessionDescription answer)
}
}
boolean masterStream = false;
// if we have more than one stream, lets the audio be the master
if(!masterStreamSet)
{
if(remoteDescriptions.size() > 1)
{
if(mediaType.equals(MediaType.AUDIO))
{
masterStream = true;
masterStreamSet = true;
}
}
else
{
masterStream = true;
masterStreamSet = true;
}
}
// create the corresponding stream...
initStream(connector, dev, supportedFormats.get(0), target,
direction, rtpExtensions);
direction, rtpExtensions, masterStream);
}
}

11
src/net/java/sip/communicator/service/neomedia/SrtpControl.java
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@ -46,12 +46,17 @@ public interface SrtpControl
*/
public boolean getSecureCommunicationStatus();
/**
* When in multistream mode, enables the master session.
* @param masterSession whether current control, controls the master session.
*/
public void setMasterSession(boolean masterSession);
/**
* Starts and enables zrtp in the stream holding this control.
* @param masterSession whether this stream is master for the current
* media session.
* @param mediaType the media type of the stream this control controls.
*/
public void start(boolean masterSession);
public void start(MediaType mediaType);
/**
* Sets the multistream data, which means that the master stream

14
src/net/java/sip/communicator/service/protocol/media/CallPeerMediaHandler.java
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@ -1236,6 +1236,7 @@ protected Map<MediaTypeSrtpControl, SrtpControl> getSrtpControls()
* stream to use (i.e. sendonly, sendrecv, recvonly, or inactive).
* @param rtpExtensions the list of <tt>RTPExtension</tt>s that should be
* enabled for this stream.
* @param masterStream whether the stream to be used as master if secured
*
* @return the newly created <tt>MediaStream</tt>.
*
@ -1247,7 +1248,8 @@ protected MediaStream initStream(StreamConnector connector,
MediaFormat format,
MediaStreamTarget target,
MediaDirection direction,
List<RTPExtension> rtpExtensions)
List<RTPExtension> rtpExtensions,
boolean masterStream)
throws OperationFailedException
{
MediaType mediaType = device.getMediaType();
@ -1284,7 +1286,8 @@ protected MediaStream initStream(StreamConnector connector,
return
configureStream(
device, format, target, direction, rtpExtensions, stream);
device, format, target, direction, rtpExtensions, stream,
masterStream);
}
/**
@ -1303,6 +1306,7 @@ protected MediaStream initStream(StreamConnector connector,
* @param rtpExtensions the list of <tt>RTPExtension</tt>s that should be
* enabled for this stream.
* @param stream the <tt>MediaStream</tt> that we'd like to configure.
* @param masterStream whether the stream to be used as master if secured
*
* @return the <tt>MediaStream</tt> that we received as a parameter (for
* convenience reasons).
@ -1316,7 +1320,8 @@ protected MediaStream configureStream( MediaDevice device,
MediaStreamTarget target,
MediaDirection direction,
List<RTPExtension> rtpExtensions,
MediaStream stream)
MediaStream stream,
boolean masterStream)
throws OperationFailedException
{
registerDynamicPTsWithStream(stream);
@ -1355,8 +1360,9 @@ protected MediaStream configureStream( MediaDevice device,
*/
SrtpControl srtpControl = stream.getSrtpControl();
srtpControl.setMasterSession(masterStream);
srtpControl.setSrtpListener(srtpListener);
srtpControl.start(MediaType.AUDIO.equals(mediaType));
srtpControl.start(mediaType);
}
return stream;

5
src/net/java/sip/communicator/service/protocol/media/MediaAwareCall.java
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@ -495,7 +495,10 @@ public MediaDevice getDefaultDevice(MediaType mediaType)
*/
if ((conferenceAudioMixer == null)
&& (device != null)
&& (!OSUtils.IS_ANDROID || isConferenceFocus()))
&& (!OSUtils.IS_ANDROID || isConferenceFocus())
// we can use audio mixer only if we
// have capture device (device can send)
&& (device.getDirection().allowsSending()))
conferenceAudioMixer = mediaService.createMixer(device);
if (conferenceAudioMixer != null)
device = conferenceAudioMixer;

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