/**
* Sets the time in milliseconds of the last activity related to this <tt>Conference</tt> to the
* current system time.
*/
public void touch() {
long now = System.currentTimeMillis();
synchronized (this) {
if (getLastActivityTime() < now) lastActivityTime = now;
}
}
/**
* Updates rate statistics for the encodings of the tracks that this receiver is managing. Detects
* simulcast stream suspension/resuming.
*
* @param pkt the {@link RawPacket} that is causing the update of this instance.
* @param encoding the {@link RTPEncodingImpl} of the {@link RawPacket} that is passed as an
* argument.
*/
void update(RawPacket pkt, RTPEncodingImpl encoding) {
long nowMs = System.currentTimeMillis();
// Update the encoding.
FrameDesc frameDesc = encoding.update(pkt, nowMs);
if (frameDesc == null /* no frame was changed */
|| !frameDesc.isIndependent() /* frame is dependent */
// The webrtc engine is sending keyframes from high to low and less
// often than 300 millis. The first keyframe that we observe after
// we've waited for that long determines the streams that are
// streaming (not suspended).
|| nowMs - lastKeyframeMs < MIN_KEY_FRAME_WAIT_MS) {
return;
}
// media engines may decide to suspend a stream for congestion control.
// This is the case with the webrtc.org simulcast implementation. This
// behavior induces a streaming dependency between the encodings of a
// given track. The following piece of code assumes that the subjective
// quality array is ordered in a way to represent the streaming
// dependencies.
lastKeyframeMs = nowMs;
boolean isActive = false;
for (int i = rtpEncodings.length - 1; i >= 0; i--) {
if (!isActive && rtpEncodings[i].requires(encoding.getIndex())) {
isActive = true;
}
rtpEncodings[i].setActive(isActive);
}
}
/**
* Removes receiver and sender feedback from RTCP packets.
*
* @param inPacket the <tt>RTCPCompoundPacket</tt> to filter.
* @return the filtered <tt>RawPacket</tt>.
*/
public RawPacket gateway(RTCPCompoundPacket inPacket) {
if (inPacket == null || inPacket.packets == null || inPacket.packets.length == 0) {
logger.info("Ignoring empty RTCP packet.");
return null;
}
ArrayList<RTCPPacket> outPackets = new ArrayList<RTCPPacket>(inPacket.packets.length);
for (RTCPPacket p : inPacket.packets) {
switch (p.type) {
case RTCPPacket.RR:
case RTCPPacket.SR:
case RTCPPacket.SDES:
// We generate our own RR/SR/SDES packets. We only want
// to forward NACKs/PLIs/etc.
break;
case RTCPFBPacket.PSFB:
RTCPFBPacket psfb = (RTCPFBPacket) p;
switch (psfb.fmt) {
case RTCPREMBPacket.FMT:
// We generate its own REMB packets.
break;
default:
// We let through everything else, like NACK
// packets.
outPackets.add(psfb);
break;
}
break;
default:
// We let through everything else, like BYE and APP
// packets.
outPackets.add(p);
break;
}
}
if (outPackets.size() == 0) {
return null;
}
// We have feedback messages to send. Pack them in a compound
// RR and send them. TODO Use RFC5506 Reduced-Size RTCP, if the
// receiver supports it.
Collection<RTCPRRPacket> rrPackets = makeRTCPRRPackets(System.currentTimeMillis());
if (rrPackets != null && rrPackets.size() != 0) {
outPackets.addAll(0, rrPackets);
} else {
logger.warn("We might be sending invalid RTCPs.");
}
RTCPPacket[] pkts = outPackets.toArray(new RTCPPacket[outPackets.size()]);
RTCPCompoundPacket outPacket = new RTCPCompoundPacket(pkts);
return generator.apply(outPacket);
}
/**
* Implements {@link ActiveSpeakerChangedListener#activeSpeakerChanged(long)}. Notifies this
* <tt>RecorderRtpImpl</tt> that the audio <tt>ReceiveStream</tt> considered active has changed,
* and that the new active stream has SSRC <tt>ssrc</tt>.
*
* @param ssrc the SSRC of the new active stream.
*/
@Override
public void activeSpeakerChanged(long ssrc) {
if (eventHandler != null) {
RecorderEvent e = new RecorderEvent();
e.setAudioSsrc(ssrc);
// TODO: how do we time this?
e.setInstant(System.currentTimeMillis());
e.setType(RecorderEvent.Type.SPEAKER_CHANGED);
e.setMediaType(MediaType.VIDEO);
eventHandler.handleEvent(e);
}
}
/**
* Implements {@link ControllerListener#controllerUpdate(ControllerEvent)}. Handles events from
* the <tt>Processor</tt>s that this instance uses to transcode media.
*
* @param ev the event to handle.
*/
public void controllerUpdate(ControllerEvent ev) {
if (ev == null || ev.getSourceController() == null) {
return;
}
Processor processor = (Processor) ev.getSourceController();
ReceiveStreamDesc desc = findReceiveStream(processor);
if (desc == null) {
logger.warn("Event from an orphaned processor, ignoring: " + ev);
return;
}
if (ev instanceof ConfigureCompleteEvent) {
if (logger.isInfoEnabled()) {
logger.info(
"Configured processor for ReceiveStream ssrc="
+ desc.ssrc
+ " ("
+ desc.format
+ ")"
+ " "
+ System.currentTimeMillis());
}
boolean audio = desc.format instanceof AudioFormat;
if (audio) {
ContentDescriptor cd = processor.setContentDescriptor(AUDIO_CONTENT_DESCRIPTOR);
if (!AUDIO_CONTENT_DESCRIPTOR.equals(cd)) {
logger.error(
"Failed to set the Processor content "
+ "descriptor to "
+ AUDIO_CONTENT_DESCRIPTOR
+ ". Actual result: "
+ cd);
removeReceiveStream(desc, false);
return;
}
}
for (TrackControl track : processor.getTrackControls()) {
Format trackFormat = track.getFormat();
if (audio) {
final long ssrc = desc.ssrc;
SilenceEffect silenceEffect;
if (Constants.OPUS_RTP.equals(desc.format.getEncoding())) {
silenceEffect = new SilenceEffect(48000);
} else {
// We haven't tested that the RTP timestamps survive
// the journey through the chain when codecs other than
// opus are in use, so for the moment we rely on FMJ's
// timestamps for non-opus formats.
silenceEffect = new SilenceEffect();
}
silenceEffect.setListener(
new SilenceEffect.Listener() {
boolean first = true;
@Override
public void onSilenceNotInserted(long timestamp) {
if (first) {
first = false;
// send event only
audioRecordingStarted(ssrc, timestamp);
} else {
// change file and send event
resetRecording(ssrc, timestamp);
}
}
});
desc.silenceEffect = silenceEffect;
AudioLevelEffect audioLevelEffect = new AudioLevelEffect();
audioLevelEffect.setAudioLevelListener(
new SimpleAudioLevelListener() {
@Override
public void audioLevelChanged(int level) {
activeSpeakerDetector.levelChanged(ssrc, level);
}
});
try {
// We add an effect, which will insert "silence" in
// place of lost packets.
track.setCodecChain(new Codec[] {silenceEffect, audioLevelEffect});
} catch (UnsupportedPlugInException upie) {
logger.warn("Failed to insert silence effect: " + upie);
// But do go on, a recording without extra silence is
// better than nothing ;)
}
} else {
// transcode vp8/rtp to vp8 (i.e. depacketize vp8)
if (trackFormat.matches(vp8RtpFormat)) track.setFormat(vp8Format);
else {
logger.error("Unsupported track format: " + trackFormat + " for ssrc=" + desc.ssrc);
// we currently only support vp8
removeReceiveStream(desc, false);
return;
}
}
}
processor.realize();
} else if (ev instanceof RealizeCompleteEvent) {
desc.dataSource = processor.getDataOutput();
long ssrc = desc.ssrc;
boolean audio = desc.format instanceof AudioFormat;
String suffix = audio ? AUDIO_FILENAME_SUFFIX : VIDEO_FILENAME_SUFFIX;
// XXX '\' on windows?
String filename = getNextFilename(path + "/" + ssrc, suffix);
desc.filename = filename;
DataSink dataSink;
if (audio) {
try {
dataSink = Manager.createDataSink(desc.dataSource, new MediaLocator("file:" + filename));
} catch (NoDataSinkException ndse) {
logger.error("Could not create DataSink: " + ndse);
removeReceiveStream(desc, false);
return;
}
} else {
dataSink = new WebmDataSink(filename, desc.dataSource);
}
if (logger.isInfoEnabled())
logger.info(
"Created DataSink ("
+ dataSink
+ ") for SSRC="
+ ssrc
+ ". Output filename: "
+ filename);
try {
dataSink.open();
} catch (IOException e) {
logger.error("Failed to open DataSink (" + dataSink + ") for" + " SSRC=" + ssrc + ": " + e);
removeReceiveStream(desc, false);
return;
}
if (!audio) {
final WebmDataSink webmDataSink = (WebmDataSink) dataSink;
webmDataSink.setSsrc(ssrc);
webmDataSink.setEventHandler(eventHandler);
webmDataSink.setKeyFrameControl(
new KeyFrameControlAdapter() {
@Override
public boolean requestKeyFrame(boolean urgent) {
return requestFIR(webmDataSink);
}
});
}
try {
dataSink.start();
} catch (IOException e) {
logger.error(
"Failed to start DataSink (" + dataSink + ") for" + " SSRC=" + ssrc + ". " + e);
removeReceiveStream(desc, false);
return;
}
if (logger.isInfoEnabled()) logger.info("Started DataSink for SSRC=" + ssrc);
desc.dataSink = dataSink;
processor.start();
} else if (logger.isDebugEnabled()) {
logger.debug(
"Unhandled ControllerEvent from the Processor for ssrc=" + desc.ssrc + ": " + ev);
}
}
/**
* Ctor.
*
* @param ssrc
* @param remoteTime
* @param rtpTimestamp
* @param frequencyHz
*/
ReceivedRemoteClock(int ssrc, long remoteTime, int rtpTimestamp, int frequencyHz) {
this.ssrc = ssrc;
this.remoteClock = new RemoteClock(remoteTime, rtpTimestamp);
this.frequencyHz = frequencyHz;
this.receivedTime = System.currentTimeMillis();
}
/** {@inheritDoc} */
@Override
public RawPacket report() {
garbageCollector.cleanup();
// TODO Compound RTCP packets should not exceed the MTU of the network
// path.
//
// An individual RTP participant should send only one compound RTCP
// packet per report interval in order for the RTCP bandwidth per
// participant to be estimated correctly, except when the compound
// RTCP packet is split for partial encryption.
//
// If there are too many sources to fit all the necessary RR packets
// into one compound RTCP packet without exceeding the maximum
// transmission unit (MTU) of the network path, then only the subset
// that will fit into one MTU should be included in each interval. The
// subsets should be selected round-robin across multiple intervals so
// that all sources are reported.
//
// It is impossible to know in advance what the MTU of path will be.
// There are various algorithms for experimenting to find out, but many
// devices do not properly implement (or deliberately ignore) the
// necessary standards so it all comes down to trial and error. For that
// reason, we can just guess 1200 or 1500 bytes per message.
long time = System.currentTimeMillis();
Collection<RTCPPacket> packets = new ArrayList<RTCPPacket>();
// First, we build the RRs.
Collection<RTCPRRPacket> rrPackets = makeRTCPRRPackets(time);
if (rrPackets != null && rrPackets.size() != 0) {
packets.addAll(rrPackets);
}
// Next, we build the SRs.
Collection<RTCPSRPacket> srPackets = makeRTCPSRPackets(time);
if (srPackets != null && srPackets.size() != 0) {
packets.addAll(srPackets);
}
// Bail out if we have nothing to report.
if (packets.size() == 0) {
return null;
}
// Next, we build the REMB.
RTCPREMBPacket rembPacket = makeRTCPREMBPacket();
if (rembPacket != null) {
packets.add(rembPacket);
}
// Finally, we add an SDES packet.
RTCPSDESPacket sdesPacket = makeSDESPacket();
if (sdesPacket != null) {
packets.add(sdesPacket);
}
// Prepare the <tt>RTCPCompoundPacket</tt> to return.
RTCPPacket rtcpPackets[] = packets.toArray(new RTCPPacket[packets.size()]);
RTCPCompoundPacket cp = new RTCPCompoundPacket(rtcpPackets);
// Build the <tt>RTCPCompoundPacket</tt> and return the
// <tt>RawPacket</tt> to inject to the <tt>MediaStream</tt>.
return generator.apply(cp);
}