/**
* {@inheritDoc}
*
* @param format unused, since this implementation records multiple streams using potentially
* different formats.
* @param dirname the path to the directory into which this <tt>Recorder</tt> will store the
* recorded media files.
*/
@Override
public void start(String format, String dirname) throws IOException, MediaException {
if (logger.isInfoEnabled()) logger.info("Starting, format=" + format + " " + hashCode());
path = dirname;
MediaService mediaService = LibJitsi.getMediaService();
/*
* Note that we use only one RTPConnector for both the RTPTranslator
* and the RTPManager instances. The this.translator will write to its
* output streams, and this.rtpManager will read from its input streams.
*/
rtpConnector = new RTPConnectorImpl(redPayloadType, ulpfecPayloadType);
rtpManager = RTPManager.newInstance();
/*
* Add the formats that we know about.
*/
rtpManager.addFormat(vp8RtpFormat, vp8PayloadType);
rtpManager.addFormat(opusFormat, opusPayloadType);
rtpManager.addReceiveStreamListener(this);
/*
* Note: When this.rtpManager sends RTCP sender/receiver reports, they
* will end up being written to its own input stream. This is not
* expected to cause problems, but might be something to keep an eye on.
*/
rtpManager.initialize(rtpConnector);
/*
* Register a fake call participant.
* TODO: can we use a more generic MediaStream here?
*/
streamRTPManager =
new StreamRTPManager(
mediaService.createMediaStream(
new MediaDeviceImpl(new CaptureDeviceInfo(), MediaType.VIDEO)),
translator);
streamRTPManager.initialize(rtpConnector);
rtcpFeedbackSender = translator.getRtcpFeedbackMessageSender();
translator.addFormat(streamRTPManager, opusFormat, opusPayloadType);
// ((RTPTranslatorImpl)videoRTPTranslator).addFormat(streamRTPManager, redFormat,
// redPayloadType);
// ((RTPTranslatorImpl)videoRTPTranslator).addFormat(streamRTPManager, ulpfecFormat,
// ulpfecPayloadType);
// ((RTPTranslatorImpl)videoRTPTranslator).addFormat(streamRTPManager,
// mediaFormatImpl.getFormat(), vp8PayloadType);
started = true;
}
/**
* Makes an <tt>RTCPREMBPacket</tt> that provides receiver feedback to the endpoint from which we
* receive.
*
* @return an <tt>RTCPREMBPacket</tt> that provides receiver feedback to the endpoint from which
* we receive.
*/
private RTCPREMBPacket makeRTCPREMBPacket() {
// TODO we should only make REMBs if REMB support has been advertised.
// Destination
RemoteBitrateEstimator remoteBitrateEstimator =
((VideoMediaStream) getStream()).getRemoteBitrateEstimator();
Collection<Integer> ssrcs = remoteBitrateEstimator.getSsrcs();
// TODO(gp) intersect with SSRCs from signaled simulcast layers
// NOTE(gp) The Google Congestion Control algorithm (sender side)
// doesn't seem to care about the SSRCs in the dest field.
long[] dest = new long[ssrcs.size()];
int i = 0;
for (Integer ssrc : ssrcs) dest[i++] = ssrc & 0xFFFFFFFFL;
// Exp & mantissa
long bitrate = remoteBitrateEstimator.getLatestEstimate();
if (bitrate == -1) return null;
if (logger.isDebugEnabled()) logger.debug("Estimated bitrate: " + bitrate);
// Create and return the packet.
// We use the stream's local source ID (SSRC) as the SSRC of packet
// sender.
long streamSSRC = getLocalSSRC();
return new RTCPREMBPacket(streamSSRC, /* mediaSSRC */ 0L, bitrate, dest);
}
private void removeReceiveStream(ReceiveStreamDesc receiveStream, boolean emptyJB) {
if (receiveStream.format instanceof VideoFormat) {
rtpConnector.packetBuffer.disable(receiveStream.ssrc);
emptyPacketBuffer(receiveStream.ssrc);
}
if (receiveStream.dataSink != null) {
try {
receiveStream.dataSink.stop();
} catch (IOException e) {
logger.error("Failed to stop DataSink " + e);
}
receiveStream.dataSink.close();
}
if (receiveStream.processor != null) {
receiveStream.processor.stop();
receiveStream.processor.close();
}
DataSource dataSource = receiveStream.receiveStream.getDataSource();
if (dataSource != null) {
try {
dataSource.stop();
} catch (IOException ioe) {
logger.warn("Failed to stop DataSource");
}
dataSource.disconnect();
}
synchronized (receiveStreams) {
receiveStreams.remove(receiveStream);
}
}
/**
* Retransmits a packet to {@link #channel}. If the destination supports the RTX format, the
* packet will be encapsulated in RTX, otherwise, the packet will be retransmitted as-is.
*
* @param pkt the packet to retransmit.
* @param after the {@code TransformEngine} in the chain of {@code TransformEngine}s of the
* associated {@code MediaStream} after which the injection of {@code pkt} is to begin
* @return {@code true} if the packet was successfully retransmitted, {@code false} otherwise.
*/
public boolean retransmit(RawPacket pkt, TransformEngine after) {
boolean destinationSupportsRtx = channel.getRtxPayloadType() != -1;
boolean retransmitPlain;
if (destinationSupportsRtx) {
long rtxSsrc = getPairedSsrc(pkt.getSSRC());
if (rtxSsrc == -1) {
logger.warn("Cannot find SSRC for RTX, retransmitting plain.");
retransmitPlain = true;
} else {
retransmitPlain = !encapsulateInRtxAndTransmit(pkt, rtxSsrc);
}
} else {
retransmitPlain = true;
}
if (retransmitPlain) {
MediaStream mediaStream = channel.getStream();
if (mediaStream != null) {
try {
mediaStream.injectPacket(pkt, /* data */ true, after);
} catch (TransmissionFailedException tfe) {
logger.warn("Failed to retransmit a packet.");
return false;
}
}
}
return true;
}
/**
* Handles a specific <tt>IOException</tt> which was thrown during the execution of {@link
* #runInConnectThread(DTLSProtocol, TlsPeer, DatagramTransport)} while trying to establish a DTLS
* connection
*
* @param ioe the <tt>IOException</tt> to handle
* @param msg the human-readable message to log about the specified <tt>ioe</tt>
* @param i the number of tries remaining after the current one
* @return <tt>true</tt> if the specified <tt>ioe</tt> was successfully handled; <tt>false</tt>,
* otherwise
*/
private boolean handleRunInConnectThreadException(IOException ioe, String msg, int i) {
// SrtpControl.start(MediaType) starts its associated TransformEngine.
// We will use that mediaType to signal the normal stop then as well
// i.e. we will ignore exception after the procedure to stop this
// PacketTransformer has begun.
if (mediaType == null) return false;
if (ioe instanceof TlsFatalAlert) {
TlsFatalAlert tfa = (TlsFatalAlert) ioe;
short alertDescription = tfa.getAlertDescription();
if (alertDescription == AlertDescription.unexpected_message) {
msg += " Received fatal unexpected message.";
if (i == 0
|| !Thread.currentThread().equals(connectThread)
|| connector == null
|| mediaType == null) {
msg += " Giving up after " + (CONNECT_TRIES - i) + " retries.";
} else {
msg += " Will retry.";
logger.error(msg, ioe);
return true;
}
} else {
msg += " Received fatal alert " + alertDescription + ".";
}
}
logger.error(msg, ioe);
return false;
}
/**
* Restarts the recording for a specific SSRC.
*
* @param ssrc the SSRC for which to restart recording. RTP packet of the new recording).
*/
private void resetRecording(long ssrc, long timestamp) {
ReceiveStreamDesc receiveStream = findReceiveStream(ssrc);
// we only restart audio recordings
if (receiveStream != null && receiveStream.format instanceof AudioFormat) {
String newFilename = getNextFilename(path + "/" + ssrc, AUDIO_FILENAME_SUFFIX);
// flush the buffer contained in the MP3 encoder
String s = "trying to flush ssrc=" + ssrc;
Processor p = receiveStream.processor;
if (p != null) {
s += " p!=null";
for (TrackControl tc : p.getTrackControls()) {
Object o = tc.getControl(FlushableControl.class.getName());
if (o != null) ((FlushableControl) o).flush();
}
}
if (logger.isInfoEnabled()) {
logger.info("Restarting recording for SSRC=" + ssrc + ". New filename: " + newFilename);
}
receiveStream.dataSink.close();
receiveStream.dataSink = null;
// flush the FMJ jitter buffer
// DataSource ds = receiveStream.receiveStream.getDataSource();
// if (ds instanceof net.sf.fmj.media.protocol.rtp.DataSource)
// ((net.sf.fmj.media.protocol.rtp.DataSource)ds).flush();
receiveStream.filename = newFilename;
try {
receiveStream.dataSink =
Manager.createDataSink(
receiveStream.dataSource, new MediaLocator("file:" + newFilename));
} catch (NoDataSinkException ndse) {
logger.warn("Could not reset recording for SSRC=" + ssrc + ": " + ndse);
removeReceiveStream(receiveStream, false);
}
try {
receiveStream.dataSink.open();
receiveStream.dataSink.start();
} catch (IOException ioe) {
logger.warn("Could not reset recording for SSRC=" + ssrc + ": " + ioe);
removeReceiveStream(receiveStream, false);
}
audioRecordingStarted(ssrc, timestamp);
}
}
/**
* Tries to find an SSRC paired with {@code ssrc} in an FID group in one of the channels from
* {@link #channel}'s {@code Content}. Returns -1 on failure.
*
* @param pkt the {@code RawPacket} that holds the RTP packet for which to find a paired SSRC.
* @return An SSRC paired with {@code ssrc} in an FID group, or -1.
*/
private long getRtxSsrc(RawPacket pkt) {
StreamRTPManager receiveRTPManager =
channel.getStream().getRTPTranslator().findStreamRTPManagerByReceiveSSRC(pkt.getSSRC());
MediaStreamTrackReceiver receiver = null;
if (receiveRTPManager != null) {
MediaStream receiveStream = receiveRTPManager.getMediaStream();
if (receiveStream != null) {
receiver = receiveStream.getMediaStreamTrackReceiver();
}
}
if (receiver == null) {
return -1;
}
RTPEncoding encoding = receiver.resolveRTPEncoding(pkt);
if (encoding == null) {
logger.warn(
"encoding_not_found"
+ ",stream_hash="
+ channel.getStream().hashCode()
+ " ssrc="
+ pkt.getSSRCAsLong());
return -1;
}
return encoding.getRTXSSRC();
}
/** Stops this <tt>PacketTransformer</tt>. */
private synchronized void stop() {
if (connectThread != null) connectThread = null;
try {
// The dtlsTransport and _srtpTransformer SHOULD be closed, of
// course. The datagramTransport MUST be closed.
if (dtlsTransport != null) {
try {
dtlsTransport.close();
} catch (IOException ioe) {
logger.error("Failed to (properly) close " + dtlsTransport.getClass(), ioe);
}
dtlsTransport = null;
}
if (_srtpTransformer != null) {
_srtpTransformer.close();
_srtpTransformer = null;
}
} finally {
try {
closeDatagramTransport();
} finally {
notifyAll();
}
}
}
/** {@inheritDoc} */
@Override
public RawPacket transform(RawPacket pkt) {
if (pkt == null) {
return pkt;
}
RTCPCompoundPacket inPacket;
try {
inPacket =
(RTCPCompoundPacket)
parser.parse(pkt.getBuffer(), pkt.getOffset(), pkt.getLength());
} catch (BadFormatException e) {
logger.warn("Failed to terminate an RTCP packet. " + "Dropping packet.");
return null;
}
// Update our RTCP stats map (timestamps). This operation is
// read-only.
remoteClockEstimator.apply(inPacket);
cnameRegistry.update(inPacket);
// Remove SRs and RRs from the RTCP packet.
pkt = feedbackGateway.gateway(inPacket);
return pkt;
}
/**
* Iterate through all the <tt>ReceiveStream</tt>s that this <tt>MediaStream</tt> has and make
* <tt>RTCPReportBlock</tt>s for all of them.
*
* @param time
* @return
*/
private RTCPReportBlock[] makeRTCPReportBlocks(long time) {
MediaStream stream = getStream();
// State validation.
if (stream == null) {
logger.warn("stream is null.");
return MIN_RTCP_REPORTS_BLOCKS_ARRAY;
}
StreamRTPManager streamRTPManager = stream.getStreamRTPManager();
if (streamRTPManager == null) {
logger.warn("streamRTPManager is null.");
return MIN_RTCP_REPORTS_BLOCKS_ARRAY;
}
Collection<ReceiveStream> receiveStreams = streamRTPManager.getReceiveStreams();
if (receiveStreams == null || receiveStreams.size() == 0) {
logger.info("There are no receive streams to build report " + "blocks for.");
return MIN_RTCP_REPORTS_BLOCKS_ARRAY;
}
SSRCCache cache = streamRTPManager.getSSRCCache();
if (cache == null) {
logger.info("cache is null.");
return MIN_RTCP_REPORTS_BLOCKS_ARRAY;
}
// Create the return object.
Collection<RTCPReportBlock> rtcpReportBlocks = new ArrayList<RTCPReportBlock>();
// Populate the return object.
for (ReceiveStream receiveStream : receiveStreams) {
// Dig into the guts of FMJ and get the stats for the current
// receiveStream.
SSRCInfo info = cache.cache.get((int) receiveStream.getSSRC());
if (!info.ours && info.sender) {
RTCPReportBlock rtcpReportBlock = info.makeReceiverReport(time);
rtcpReportBlocks.add(rtcpReportBlock);
}
}
return rtcpReportBlocks.toArray(new RTCPReportBlock[rtcpReportBlocks.size()]);
}
/**
* Removes the RTX encapsulation from a packet.
*
* @param pkt the packet to remove the RTX encapsulation from.
* @return the original media packet represented by {@code pkt}, or null if we couldn't
* reconstruct the original packet.
*/
private RawPacket deRtx(RawPacket pkt) {
boolean success = false;
if (pkt.getPayloadLength() - pkt.getPaddingSize() < 2) {
// We need at least 2 bytes to read the OSN field.
if (logger.isDebugEnabled()) {
logger.debug("Dropping an incoming RTX packet with padding only: " + pkt);
}
return null;
}
long mediaSsrc = getPrimarySsrc(pkt);
if (mediaSsrc != -1) {
if (rtxAssociatedPayloadType != -1) {
int osn = pkt.getOriginalSequenceNumber();
// Remove the RTX header by moving the RTP header two bytes
// right.
byte[] buf = pkt.getBuffer();
int off = pkt.getOffset();
System.arraycopy(buf, off, buf, off + 2, pkt.getHeaderLength());
pkt.setOffset(off + 2);
pkt.setLength(pkt.getLength() - 2);
pkt.setSSRC((int) mediaSsrc);
pkt.setSequenceNumber(osn);
pkt.setPayloadType(rtxAssociatedPayloadType);
success = true;
} else {
logger.warn(
"RTX packet received, but no APT is defined. Packet "
+ "SSRC "
+ pkt.getSSRCAsLong()
+ ", associated media"
+ " SSRC "
+ mediaSsrc);
}
}
// If we failed to handle the RTX packet, drop it.
return success ? pkt : null;
}
/**
* Returns the SSRC paired with <tt>ssrc</tt> in an FID source-group, if any. If none is found,
* returns -1.
*
* @return the SSRC paired with <tt>ssrc</tt> in an FID source-group, if any. If none is found,
* returns -1.
*/
private long getPrimarySsrc(RawPacket pkt) {
MediaStreamTrackReceiver receiver = channel.getStream().getMediaStreamTrackReceiver();
if (receiver == null) {
if (logger.isDebugEnabled()) {
logger.debug("Dropping an incoming RTX packet from an unknown source.");
}
return -1;
}
RTPEncoding encoding = receiver.resolveRTPEncoding(pkt);
if (encoding == null) {
if (logger.isDebugEnabled()) {
logger.debug("Dropping an incoming RTX packet from an unknown source.");
}
return -1;
}
return encoding.getPrimarySSRC();
}
/**
* Closes {@link #datagramTransport} if it is non-<tt>null</tt> and logs and swallows any
* <tt>IOException</tt>.
*/
private void closeDatagramTransport() {
if (datagramTransport != null) {
try {
datagramTransport.close();
} catch (IOException ioe) {
// DatagramTransportImpl has no reason to fail because it is
// merely an adapter of #connector and this PacketTransformer to
// the terms of the Bouncy Castle Crypto API.
logger.error("Failed to (properly) close " + datagramTransport.getClass(), ioe);
}
datagramTransport = null;
}
}
@Override
public void stop() {
if (started) {
if (logger.isInfoEnabled()) logger.info("Stopping " + hashCode());
// remove the recorder from the translator (e.g. stop new packets from
// being written to rtpConnector
if (streamRTPManager != null) streamRTPManager.dispose();
HashSet<ReceiveStreamDesc> streamsToRemove = new HashSet<ReceiveStreamDesc>();
synchronized (receiveStreams) {
streamsToRemove.addAll(receiveStreams);
}
for (ReceiveStreamDesc r : streamsToRemove) removeReceiveStream(r, false);
rtpConnector.rtcpPacketTransformer.close();
rtpConnector.rtpPacketTransformer.close();
rtpManager.dispose();
started = false;
}
}
private void emptyPacketBuffer(long ssrc) {
RawPacket[] pkts = rtpConnector.packetBuffer.emptyBuffer(ssrc);
RTPConnectorImpl.OutputDataStreamImpl dataStream;
try {
dataStream = rtpConnector.getDataOutputStream();
} catch (IOException ioe) {
logger.error("Failed to empty packet buffer for SSRC=" + ssrc + ": " + ioe);
return;
}
for (RawPacket pkt : pkts)
dataStream.write(
pkt.getBuffer(), pkt.getOffset(), pkt.getLength(), false /* already transformed */);
}
/**
* Encapsulates {@code pkt} in the RTX format, using {@code rtxSsrc} as its SSRC, and transmits it
* to {@link #channel} by injecting it in the {@code MediaStream}.
*
* @param pkt the packet to transmit.
* @param rtxSsrc the SSRC for the RTX stream.
* @param after the {@code TransformEngine} in the chain of {@code TransformEngine}s of the
* associated {@code MediaStream} after which the injection of {@code pkt} is to begin
* @return {@code true} if the packet was successfully retransmitted, {@code false} otherwise.
*/
private boolean encapsulateInRtxAndTransmit(RawPacket pkt, long rtxSsrc, TransformEngine after) {
byte[] buf = pkt.getBuffer();
int len = pkt.getLength();
int off = pkt.getOffset();
byte[] newBuf = new byte[len + 2];
RawPacket rtxPkt = new RawPacket(newBuf, 0, len + 2);
int osn = pkt.getSequenceNumber();
int headerLength = pkt.getHeaderLength();
int payloadLength = pkt.getPayloadLength();
// Copy the header.
System.arraycopy(buf, off, newBuf, 0, headerLength);
// Set the OSN field.
newBuf[headerLength] = (byte) ((osn >> 8) & 0xff);
newBuf[headerLength + 1] = (byte) (osn & 0xff);
// Copy the payload.
System.arraycopy(buf, off + headerLength, newBuf, headerLength + 2, payloadLength);
MediaStream mediaStream = channel.getStream();
if (mediaStream != null) {
rtxPkt.setSSRC((int) rtxSsrc);
rtxPkt.setPayloadType(rtxPayloadType);
// Only call getNextRtxSequenceNumber() when we're sure we're going
// to transmit a packet, because it consumes a sequence number.
rtxPkt.setSequenceNumber(getNextRtxSequenceNumber(rtxSsrc));
try {
mediaStream.injectPacket(rtxPkt, /* data */ true, after);
} catch (TransmissionFailedException tfe) {
logger.warn("Failed to transmit an RTX packet.");
return false;
}
}
return true;
}
/**
* Encapsulates {@code pkt} in the RTX format, using {@code rtxSsrc} as its SSRC, and transmits it
* to {@link #channel} by injecting it in the {@code MediaStream}.
*
* @param pkt the packet to transmit.
* @param rtxSsrc the SSRC for the RTX stream.
* @return {@code true} if the packet was successfully retransmitted, {@code false} otherwise.
*/
private boolean encapsulateInRtxAndTransmit(RawPacket pkt, long rtxSsrc) {
byte[] buf = pkt.getBuffer();
int len = pkt.getLength();
int off = pkt.getOffset();
byte[] newBuf = buf;
if (buf.length < len + 2) {
// FIXME The byte array newly allocated and assigned to newBuf must
// be made known to pkt eventually.
newBuf = new byte[len + 2];
}
int osn = pkt.getSequenceNumber();
int headerLength = pkt.getHeaderLength();
int payloadLength = len - headerLength;
System.arraycopy(buf, off, newBuf, 0, headerLength);
// FIXME If newBuf is actually buf, then we will override the first two
// bytes of the payload bellow.
newBuf[headerLength] = (byte) ((osn >> 8) & 0xff);
newBuf[headerLength + 1] = (byte) (osn & 0xff);
System.arraycopy(buf, off + headerLength, newBuf, headerLength + 2, payloadLength);
// FIXME We tried to extend the payload of pkt by two bytes above but
// we never told pkt that its length has increased by these two bytes.
MediaStream mediaStream = channel.getStream();
if (mediaStream != null) {
pkt.setSSRC((int) rtxSsrc);
// Only call getNextRtxSequenceNumber() when we're sure we're going
// to transmit a packet, because it consumes a sequence number.
pkt.setSequenceNumber(getNextRtxSequenceNumber(rtxSsrc));
try {
mediaStream.injectPacket(pkt, /* data */ true, /* after */ null);
} catch (TransmissionFailedException tfe) {
logger.warn("Failed to transmit an RTX packet.");
return false;
}
}
return true;
}
/**
* Sends the data contained in a specific byte array as application data through the DTLS
* connection of this <tt>DtlsPacketTransformer</tt>.
*
* @param buf the byte array containing data to send.
* @param off the offset in <tt>buf</tt> where the data begins.
* @param len the length of data to send.
*/
public void sendApplicationData(byte[] buf, int off, int len) {
DTLSTransport dtlsTransport = this.dtlsTransport;
Throwable throwable = null;
if (dtlsTransport != null) {
try {
dtlsTransport.send(buf, off, len);
} catch (IOException ioe) {
throwable = ioe;
}
} else {
throwable = new NullPointerException("dtlsTransport");
}
if (throwable != null) {
// SrtpControl.start(MediaType) starts its associated
// TransformEngine. We will use that mediaType to signal the normal
// stop then as well i.e. we will ignore exception after the
// procedure to stop this PacketTransformer has begun.
if (mediaType != null && !tlsPeerHasRaisedCloseNotifyWarning) {
logger.error("Failed to send application data over DTLS transport: ", throwable);
}
}
}
/**
* 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);
}
}
/** Starts this <tt>PacketTransformer</tt>. */
private synchronized void start() {
if (this.datagramTransport != null) {
if (this.connectThread == null && dtlsTransport == null) {
logger.warn(
getClass().getName()
+ " has been started but has failed to establish"
+ " the DTLS connection!");
}
return;
}
if (rtcpmux && Component.RTCP == componentID) {
// In the case of rtcp-mux, the RTCP transformer does not create
// a DTLS session. The SRTP context (_srtpTransformer) will be
// initialized on demand using initializeSRTCPTransformerFromRtp().
return;
}
AbstractRTPConnector connector = this.connector;
if (connector == null) throw new NullPointerException("connector");
DtlsControl.Setup setup = this.setup;
SecureRandom secureRandom = DtlsControlImpl.createSecureRandom();
final DTLSProtocol dtlsProtocolObj;
final TlsPeer tlsPeer;
if (DtlsControl.Setup.ACTIVE.equals(setup)) {
dtlsProtocolObj = new DTLSClientProtocol(secureRandom);
tlsPeer = new TlsClientImpl(this);
} else {
dtlsProtocolObj = new DTLSServerProtocol(secureRandom);
tlsPeer = new TlsServerImpl(this);
}
tlsPeerHasRaisedCloseNotifyWarning = false;
final DatagramTransportImpl datagramTransport = new DatagramTransportImpl(componentID);
datagramTransport.setConnector(connector);
Thread connectThread =
new Thread() {
@Override
public void run() {
try {
runInConnectThread(dtlsProtocolObj, tlsPeer, datagramTransport);
} finally {
if (Thread.currentThread().equals(DtlsPacketTransformer.this.connectThread)) {
DtlsPacketTransformer.this.connectThread = null;
}
}
}
};
connectThread.setDaemon(true);
connectThread.setName(DtlsPacketTransformer.class.getName() + ".connectThread");
this.connectThread = connectThread;
this.datagramTransport = datagramTransport;
boolean started = false;
try {
connectThread.start();
started = true;
} finally {
if (!started) {
if (connectThread.equals(this.connectThread)) this.connectThread = null;
if (datagramTransport.equals(this.datagramTransport)) this.datagramTransport = null;
}
}
notifyAll();
}
/** {@inheritDoc} */
@Override
public RawPacket reverseTransform(RawPacket pkt) {
byte[] buf = pkt.getBuffer();
int off = pkt.getOffset();
int len = pkt.getLength();
if (isDtlsRecord(buf, off, len)) {
if (rtcpmux && Component.RTCP == componentID) {
// This should never happen.
logger.warn(
"Dropping a DTLS record, because it was received on the"
+ " RTCP channel while rtcpmux is in use.");
return null;
}
boolean receive;
synchronized (this) {
if (datagramTransport == null) {
receive = false;
} else {
datagramTransport.queueReceive(buf, off, len);
receive = true;
}
}
if (receive) {
DTLSTransport dtlsTransport = this.dtlsTransport;
if (dtlsTransport == null) {
// The specified pkt looks like a DTLS record and it has
// been consumed for the purposes of the secure channel
// represented by this PacketTransformer.
pkt = null;
} else {
try {
int receiveLimit = dtlsTransport.getReceiveLimit();
int delta = receiveLimit - len;
if (delta > 0) {
pkt.grow(delta);
buf = pkt.getBuffer();
off = pkt.getOffset();
len = pkt.getLength();
} else if (delta < 0) {
pkt.shrink(-delta);
buf = pkt.getBuffer();
off = pkt.getOffset();
len = pkt.getLength();
}
int received = dtlsTransport.receive(buf, off, len, DTLS_TRANSPORT_RECEIVE_WAITMILLIS);
if (received <= 0) {
// No application data was decoded.
pkt = null;
} else {
delta = len - received;
if (delta > 0) pkt.shrink(delta);
}
} catch (IOException ioe) {
pkt = null;
// SrtpControl.start(MediaType) starts its associated
// TransformEngine. We will use that mediaType to signal
// the normal stop then as well i.e. we will ignore
// exception after the procedure to stop this
// PacketTransformer has begun.
if (mediaType != null && !tlsPeerHasRaisedCloseNotifyWarning) {
logger.error("Failed to decode a DTLS record!", ioe);
}
}
}
} else {
// The specified pkt looks like a DTLS record but it is
// unexpected in the current state of the secure channel
// represented by this PacketTransformer. This PacketTransformer
// has not been started (successfully) or has been closed.
pkt = null;
}
} else if (transformEngine.isSrtpDisabled()) {
// In pure DTLS mode only DTLS records pass through.
pkt = null;
} else {
// DTLS-SRTP has not been initialized yet or has failed to
// initialize.
SinglePacketTransformer srtpTransformer = waitInitializeAndGetSRTPTransformer();
if (srtpTransformer != null) pkt = srtpTransformer.reverseTransform(pkt);
else if (DROP_UNENCRYPTED_PKTS) pkt = null;
// XXX Else, it is our explicit policy to let the received packet
// pass through and rely on the SrtpListener to notify the user that
// the session is not secured.
}
return pkt;
}
/**
* Implements {@link PacketTransformer} for DTLS-SRTP. It's capable of working in pure DTLS mode if
* appropriate flag was set in <tt>DtlsControlImpl</tt>.
*
* @author Lyubomir Marinov
*/
public class DtlsPacketTransformer extends SinglePacketTransformer {
private static final long CONNECT_RETRY_INTERVAL = 500;
/**
* The maximum number of times that {@link #runInConnectThread(DTLSProtocol, TlsPeer,
* DatagramTransport)} is to retry the invocations of {@link DTLSClientProtocol#connect(TlsClient,
* DatagramTransport)} and {@link DTLSServerProtocol#accept(TlsServer, DatagramTransport)} in
* anticipation of a successful connection.
*/
private static final int CONNECT_TRIES = 3;
/**
* The indicator which determines whether unencrypted packets sent or received through
* <tt>DtlsPacketTransformer</tt> are to be dropped. The default value is <tt>false</tt>.
*
* @see #DROP_UNENCRYPTED_PKTS_PNAME
*/
private static final boolean DROP_UNENCRYPTED_PKTS;
/**
* The name of the <tt>ConfigurationService</tt> and/or <tt>System</tt> property which indicates
* whether unencrypted packets sent or received through <tt>DtlsPacketTransformer</tt> are to be
* dropped. The default value is <tt>false</tt>.
*/
private static final String DROP_UNENCRYPTED_PKTS_PNAME =
DtlsPacketTransformer.class.getName() + ".dropUnencryptedPkts";
/** The length of the header of a DTLS record. */
static final int DTLS_RECORD_HEADER_LENGTH = 13;
/**
* The number of milliseconds a <tt>DtlsPacketTransform</tt> is to wait on its {@link
* #dtlsTransport} in order to receive a packet.
*/
private static final int DTLS_TRANSPORT_RECEIVE_WAITMILLIS = -1;
/**
* The <tt>Logger</tt> used by the <tt>DtlsPacketTransformer</tt> class and its instances to print
* debug information.
*/
private static final Logger logger = Logger.getLogger(DtlsPacketTransformer.class);
static {
ConfigurationService cfg = LibJitsi.getConfigurationService();
boolean dropUnencryptedPkts = false;
if (cfg == null) {
String s = System.getProperty(DROP_UNENCRYPTED_PKTS_PNAME);
if (s != null) dropUnencryptedPkts = Boolean.parseBoolean(s);
} else {
dropUnencryptedPkts = cfg.getBoolean(DROP_UNENCRYPTED_PKTS_PNAME, dropUnencryptedPkts);
}
DROP_UNENCRYPTED_PKTS = dropUnencryptedPkts;
}
/**
* Determines whether a specific array of <tt>byte</tt>s appears to contain a DTLS record.
*
* @param buf the array of <tt>byte</tt>s to be analyzed
* @param off the offset within <tt>buf</tt> at which the analysis is to start
* @param len the number of bytes within <tt>buf</tt> starting at <tt>off</tt> to be analyzed
* @return <tt>true</tt> if the specified <tt>buf</tt> appears to contain a DTLS record
*/
public static boolean isDtlsRecord(byte[] buf, int off, int len) {
boolean b = false;
if (len >= DTLS_RECORD_HEADER_LENGTH) {
short type = TlsUtils.readUint8(buf, off);
switch (type) {
case ContentType.alert:
case ContentType.application_data:
case ContentType.change_cipher_spec:
case ContentType.handshake:
int major = buf[off + 1] & 0xff;
int minor = buf[off + 2] & 0xff;
ProtocolVersion version = null;
if ((major == ProtocolVersion.DTLSv10.getMajorVersion())
&& (minor == ProtocolVersion.DTLSv10.getMinorVersion())) {
version = ProtocolVersion.DTLSv10;
}
if ((version == null)
&& (major == ProtocolVersion.DTLSv12.getMajorVersion())
&& (minor == ProtocolVersion.DTLSv12.getMinorVersion())) {
version = ProtocolVersion.DTLSv12;
}
if (version != null) {
int length = TlsUtils.readUint16(buf, off + 11);
if (DTLS_RECORD_HEADER_LENGTH + length <= len) b = true;
}
break;
default:
// Unless a new ContentType has been defined by the Bouncy
// Castle Crypto APIs, the specified buf does not represent a
// DTLS record.
break;
}
}
return b;
}
/** The ID of the component which this instance works for/is associated with. */
private final int componentID;
/** The <tt>RTPConnector</tt> which uses this <tt>PacketTransformer</tt>. */
private AbstractRTPConnector connector;
/** The background <tt>Thread</tt> which initializes {@link #dtlsTransport}. */
private Thread connectThread;
/**
* The <tt>DatagramTransport</tt> implementation which adapts {@link #connector} and this
* <tt>PacketTransformer</tt> to the terms of the Bouncy Castle Crypto APIs.
*/
private DatagramTransportImpl datagramTransport;
/**
* The <tt>DTLSTransport</tt> through which the actual packet transformations are being performed
* by this instance.
*/
private DTLSTransport dtlsTransport;
/** The <tt>MediaType</tt> of the stream which this instance works for/is associated with. */
private MediaType mediaType;
/**
* Whether rtcp-mux is in use.
*
* <p>If enabled, and this is the transformer for RTCP, it will not establish a DTLS session on
* its own, but rather wait for the RTP transformer to do so, and reuse it to initialize the SRTP
* transformer.
*/
private boolean rtcpmux = false;
/**
* The value of the <tt>setup</tt> SDP attribute defined by RFC 4145 "TCP-Based Media
* Transport in the Session Description Protocol (SDP)" which determines whether this
* instance acts as a DTLS client or a DTLS server.
*/
private DtlsControl.Setup setup;
/** The {@code SRTPTransformer} (to be) used by this instance. */
private SinglePacketTransformer _srtpTransformer;
/**
* The indicator which determines whether the <tt>TlsPeer</tt> employed by this
* <tt>PacketTransformer</tt> has raised an <tt>AlertDescription.close_notify</tt>
* <tt>AlertLevel.warning</tt> i.e. the remote DTLS peer has closed the write side of the
* connection.
*/
private boolean tlsPeerHasRaisedCloseNotifyWarning;
/** The <tt>TransformEngine</tt> which has initialized this instance. */
private final DtlsTransformEngine transformEngine;
/**
* Initializes a new <tt>DtlsPacketTransformer</tt> instance.
*
* @param transformEngine the <tt>TransformEngine</tt> which is initializing the new instance
* @param componentID the ID of the component for which the new instance is to work
*/
public DtlsPacketTransformer(DtlsTransformEngine transformEngine, int componentID) {
this.transformEngine = transformEngine;
this.componentID = componentID;
}
/** {@inheritDoc} */
@Override
public synchronized void close() {
// SrtpControl.start(MediaType) starts its associated TransformEngine.
// We will use that mediaType to signal the normal stop then as well
// i.e. we will call setMediaType(null) first.
setMediaType(null);
setConnector(null);
}
/**
* Closes {@link #datagramTransport} if it is non-<tt>null</tt> and logs and swallows any
* <tt>IOException</tt>.
*/
private void closeDatagramTransport() {
if (datagramTransport != null) {
try {
datagramTransport.close();
} catch (IOException ioe) {
// DatagramTransportImpl has no reason to fail because it is
// merely an adapter of #connector and this PacketTransformer to
// the terms of the Bouncy Castle Crypto API.
logger.error("Failed to (properly) close " + datagramTransport.getClass(), ioe);
}
datagramTransport = null;
}
}
/**
* Determines whether {@link #runInConnectThread(DTLSProtocol, TlsPeer, DatagramTransport)} is to
* try to establish a DTLS connection.
*
* @param i the number of tries remaining after the current one
* @param datagramTransport
* @return <tt>true</tt> to try to establish a DTLS connection; otherwise, <tt>false</tt>
*/
private boolean enterRunInConnectThreadLoop(int i, DatagramTransport datagramTransport) {
if (i < 0 || i > CONNECT_TRIES) {
return false;
} else {
Thread currentThread = Thread.currentThread();
synchronized (this) {
if (i > 0 && i < CONNECT_TRIES - 1) {
boolean interrupted = false;
try {
wait(CONNECT_RETRY_INTERVAL);
} catch (InterruptedException ie) {
interrupted = true;
}
if (interrupted) currentThread.interrupt();
}
return currentThread.equals(this.connectThread)
&& datagramTransport.equals(this.datagramTransport);
}
}
}
/**
* Gets the <tt>DtlsControl</tt> implementation associated with this instance.
*
* @return the <tt>DtlsControl</tt> implementation associated with this instance
*/
DtlsControlImpl getDtlsControl() {
return getTransformEngine().getDtlsControl();
}
/**
* Gets the <tt>TransformEngine</tt> which has initialized this instance.
*
* @return the <tt>TransformEngine</tt> which has initialized this instance
*/
DtlsTransformEngine getTransformEngine() {
return transformEngine;
}
/**
* Handles a specific <tt>IOException</tt> which was thrown during the execution of {@link
* #runInConnectThread(DTLSProtocol, TlsPeer, DatagramTransport)} while trying to establish a DTLS
* connection
*
* @param ioe the <tt>IOException</tt> to handle
* @param msg the human-readable message to log about the specified <tt>ioe</tt>
* @param i the number of tries remaining after the current one
* @return <tt>true</tt> if the specified <tt>ioe</tt> was successfully handled; <tt>false</tt>,
* otherwise
*/
private boolean handleRunInConnectThreadException(IOException ioe, String msg, int i) {
// SrtpControl.start(MediaType) starts its associated TransformEngine.
// We will use that mediaType to signal the normal stop then as well
// i.e. we will ignore exception after the procedure to stop this
// PacketTransformer has begun.
if (mediaType == null) return false;
if (ioe instanceof TlsFatalAlert) {
TlsFatalAlert tfa = (TlsFatalAlert) ioe;
short alertDescription = tfa.getAlertDescription();
if (alertDescription == AlertDescription.unexpected_message) {
msg += " Received fatal unexpected message.";
if (i == 0
|| !Thread.currentThread().equals(connectThread)
|| connector == null
|| mediaType == null) {
msg += " Giving up after " + (CONNECT_TRIES - i) + " retries.";
} else {
msg += " Will retry.";
logger.error(msg, ioe);
return true;
}
} else {
msg += " Received fatal alert " + alertDescription + ".";
}
}
logger.error(msg, ioe);
return false;
}
/**
* Tries to initialize {@link #_srtpTransformer} by using the <tt>DtlsPacketTransformer</tt> for
* RTP.
*
* @return the (possibly updated) value of {@link #_srtpTransformer}.
*/
private SinglePacketTransformer initializeSRTCPTransformerFromRtp() {
DtlsPacketTransformer rtpTransformer =
(DtlsPacketTransformer) getTransformEngine().getRTPTransformer();
// Prevent recursion (that is pretty much impossible to ever happen).
if (rtpTransformer != this) {
PacketTransformer srtpTransformer = rtpTransformer.waitInitializeAndGetSRTPTransformer();
if (srtpTransformer != null && srtpTransformer instanceof SRTPTransformer) {
synchronized (this) {
if (_srtpTransformer == null) {
_srtpTransformer = new SRTCPTransformer((SRTPTransformer) srtpTransformer);
// For the sake of completeness, we notify whenever we
// assign to _srtpTransformer.
notifyAll();
}
}
}
}
return _srtpTransformer;
}
/**
* Initializes a new <tt>SRTPTransformer</tt> instance with a specific (negotiated)
* <tt>SRTPProtectionProfile</tt> and the keying material specified by a specific
* <tt>TlsContext</tt>.
*
* @param srtpProtectionProfile the (negotiated) <tt>SRTPProtectionProfile</tt> to initialize the
* new instance with
* @param tlsContext the <tt>TlsContext</tt> which represents the keying material
* @return a new <tt>SRTPTransformer</tt> instance initialized with <tt>srtpProtectionProfile</tt>
* and <tt>tlsContext</tt>
*/
private SinglePacketTransformer initializeSRTPTransformer(
int srtpProtectionProfile, TlsContext tlsContext) {
boolean rtcp;
switch (componentID) {
case Component.RTCP:
rtcp = true;
break;
case Component.RTP:
rtcp = false;
break;
default:
throw new IllegalStateException("componentID");
}
int cipher_key_length;
int cipher_salt_length;
int cipher;
int auth_function;
int auth_key_length;
int RTCP_auth_tag_length, RTP_auth_tag_length;
switch (srtpProtectionProfile) {
case SRTPProtectionProfile.SRTP_AES128_CM_HMAC_SHA1_32:
cipher_key_length = 128 / 8;
cipher_salt_length = 112 / 8;
cipher = SRTPPolicy.AESCM_ENCRYPTION;
auth_function = SRTPPolicy.HMACSHA1_AUTHENTICATION;
auth_key_length = 160 / 8;
RTCP_auth_tag_length = 80 / 8;
RTP_auth_tag_length = 32 / 8;
break;
case SRTPProtectionProfile.SRTP_AES128_CM_HMAC_SHA1_80:
cipher_key_length = 128 / 8;
cipher_salt_length = 112 / 8;
cipher = SRTPPolicy.AESCM_ENCRYPTION;
auth_function = SRTPPolicy.HMACSHA1_AUTHENTICATION;
auth_key_length = 160 / 8;
RTCP_auth_tag_length = RTP_auth_tag_length = 80 / 8;
break;
case SRTPProtectionProfile.SRTP_NULL_HMAC_SHA1_32:
cipher_key_length = 0;
cipher_salt_length = 0;
cipher = SRTPPolicy.NULL_ENCRYPTION;
auth_function = SRTPPolicy.HMACSHA1_AUTHENTICATION;
auth_key_length = 160 / 8;
RTCP_auth_tag_length = 80 / 8;
RTP_auth_tag_length = 32 / 8;
break;
case SRTPProtectionProfile.SRTP_NULL_HMAC_SHA1_80:
cipher_key_length = 0;
cipher_salt_length = 0;
cipher = SRTPPolicy.NULL_ENCRYPTION;
auth_function = SRTPPolicy.HMACSHA1_AUTHENTICATION;
auth_key_length = 160 / 8;
RTCP_auth_tag_length = RTP_auth_tag_length = 80 / 8;
break;
default:
throw new IllegalArgumentException("srtpProtectionProfile");
}
byte[] keyingMaterial =
tlsContext.exportKeyingMaterial(
ExporterLabel.dtls_srtp, null, 2 * (cipher_key_length + cipher_salt_length));
byte[] client_write_SRTP_master_key = new byte[cipher_key_length];
byte[] server_write_SRTP_master_key = new byte[cipher_key_length];
byte[] client_write_SRTP_master_salt = new byte[cipher_salt_length];
byte[] server_write_SRTP_master_salt = new byte[cipher_salt_length];
byte[][] keyingMaterialValues = {
client_write_SRTP_master_key,
server_write_SRTP_master_key,
client_write_SRTP_master_salt,
server_write_SRTP_master_salt
};
for (int i = 0, keyingMaterialOffset = 0; i < keyingMaterialValues.length; i++) {
byte[] keyingMaterialValue = keyingMaterialValues[i];
System.arraycopy(
keyingMaterial, keyingMaterialOffset, keyingMaterialValue, 0, keyingMaterialValue.length);
keyingMaterialOffset += keyingMaterialValue.length;
}
SRTPPolicy srtcpPolicy =
new SRTPPolicy(
cipher,
cipher_key_length,
auth_function,
auth_key_length,
RTCP_auth_tag_length,
cipher_salt_length);
SRTPPolicy srtpPolicy =
new SRTPPolicy(
cipher,
cipher_key_length,
auth_function,
auth_key_length,
RTP_auth_tag_length,
cipher_salt_length);
SRTPContextFactory clientSRTPContextFactory =
new SRTPContextFactory(
/* sender */ tlsContext instanceof TlsClientContext,
client_write_SRTP_master_key,
client_write_SRTP_master_salt,
srtpPolicy,
srtcpPolicy);
SRTPContextFactory serverSRTPContextFactory =
new SRTPContextFactory(
/* sender */ tlsContext instanceof TlsServerContext,
server_write_SRTP_master_key,
server_write_SRTP_master_salt,
srtpPolicy,
srtcpPolicy);
SRTPContextFactory forwardSRTPContextFactory;
SRTPContextFactory reverseSRTPContextFactory;
if (tlsContext instanceof TlsClientContext) {
forwardSRTPContextFactory = clientSRTPContextFactory;
reverseSRTPContextFactory = serverSRTPContextFactory;
} else if (tlsContext instanceof TlsServerContext) {
forwardSRTPContextFactory = serverSRTPContextFactory;
reverseSRTPContextFactory = clientSRTPContextFactory;
} else {
throw new IllegalArgumentException("tlsContext");
}
SinglePacketTransformer srtpTransformer;
if (rtcp) {
srtpTransformer = new SRTCPTransformer(forwardSRTPContextFactory, reverseSRTPContextFactory);
} else {
srtpTransformer = new SRTPTransformer(forwardSRTPContextFactory, reverseSRTPContextFactory);
}
return srtpTransformer;
}
/**
* Notifies this instance that the DTLS record layer associated with a specific <tt>TlsPeer</tt>
* has raised an alert.
*
* @param tlsPeer the <tt>TlsPeer</tt> whose associated DTLS record layer has raised an alert
* @param alertLevel {@link AlertLevel}
* @param alertDescription {@link AlertDescription}
* @param message a human-readable message explaining what caused the alert. May be <tt>null</tt>.
* @param cause the exception that caused the alert to be raised. May be <tt>null</tt>.
*/
void notifyAlertRaised(
TlsPeer tlsPeer, short alertLevel, short alertDescription, String message, Exception cause) {
if (AlertLevel.warning == alertLevel && AlertDescription.close_notify == alertDescription) {
tlsPeerHasRaisedCloseNotifyWarning = true;
}
}
/** {@inheritDoc} */
@Override
public RawPacket reverseTransform(RawPacket pkt) {
byte[] buf = pkt.getBuffer();
int off = pkt.getOffset();
int len = pkt.getLength();
if (isDtlsRecord(buf, off, len)) {
if (rtcpmux && Component.RTCP == componentID) {
// This should never happen.
logger.warn(
"Dropping a DTLS record, because it was received on the"
+ " RTCP channel while rtcpmux is in use.");
return null;
}
boolean receive;
synchronized (this) {
if (datagramTransport == null) {
receive = false;
} else {
datagramTransport.queueReceive(buf, off, len);
receive = true;
}
}
if (receive) {
DTLSTransport dtlsTransport = this.dtlsTransport;
if (dtlsTransport == null) {
// The specified pkt looks like a DTLS record and it has
// been consumed for the purposes of the secure channel
// represented by this PacketTransformer.
pkt = null;
} else {
try {
int receiveLimit = dtlsTransport.getReceiveLimit();
int delta = receiveLimit - len;
if (delta > 0) {
pkt.grow(delta);
buf = pkt.getBuffer();
off = pkt.getOffset();
len = pkt.getLength();
} else if (delta < 0) {
pkt.shrink(-delta);
buf = pkt.getBuffer();
off = pkt.getOffset();
len = pkt.getLength();
}
int received = dtlsTransport.receive(buf, off, len, DTLS_TRANSPORT_RECEIVE_WAITMILLIS);
if (received <= 0) {
// No application data was decoded.
pkt = null;
} else {
delta = len - received;
if (delta > 0) pkt.shrink(delta);
}
} catch (IOException ioe) {
pkt = null;
// SrtpControl.start(MediaType) starts its associated
// TransformEngine. We will use that mediaType to signal
// the normal stop then as well i.e. we will ignore
// exception after the procedure to stop this
// PacketTransformer has begun.
if (mediaType != null && !tlsPeerHasRaisedCloseNotifyWarning) {
logger.error("Failed to decode a DTLS record!", ioe);
}
}
}
} else {
// The specified pkt looks like a DTLS record but it is
// unexpected in the current state of the secure channel
// represented by this PacketTransformer. This PacketTransformer
// has not been started (successfully) or has been closed.
pkt = null;
}
} else if (transformEngine.isSrtpDisabled()) {
// In pure DTLS mode only DTLS records pass through.
pkt = null;
} else {
// DTLS-SRTP has not been initialized yet or has failed to
// initialize.
SinglePacketTransformer srtpTransformer = waitInitializeAndGetSRTPTransformer();
if (srtpTransformer != null) pkt = srtpTransformer.reverseTransform(pkt);
else if (DROP_UNENCRYPTED_PKTS) pkt = null;
// XXX Else, it is our explicit policy to let the received packet
// pass through and rely on the SrtpListener to notify the user that
// the session is not secured.
}
return pkt;
}
/**
* Runs in {@link #connectThread} to initialize {@link #dtlsTransport}.
*
* @param dtlsProtocol
* @param tlsPeer
* @param datagramTransport
*/
private void runInConnectThread(
DTLSProtocol dtlsProtocol, TlsPeer tlsPeer, DatagramTransport datagramTransport) {
DTLSTransport dtlsTransport = null;
final boolean srtp = !transformEngine.isSrtpDisabled();
int srtpProtectionProfile = 0;
TlsContext tlsContext = null;
// DTLS client
if (dtlsProtocol instanceof DTLSClientProtocol) {
DTLSClientProtocol dtlsClientProtocol = (DTLSClientProtocol) dtlsProtocol;
TlsClientImpl tlsClient = (TlsClientImpl) tlsPeer;
for (int i = CONNECT_TRIES - 1; i >= 0; i--) {
if (!enterRunInConnectThreadLoop(i, datagramTransport)) break;
try {
dtlsTransport = dtlsClientProtocol.connect(tlsClient, datagramTransport);
break;
} catch (IOException ioe) {
if (!handleRunInConnectThreadException(
ioe, "Failed to connect this DTLS client to a DTLS" + " server!", i)) {
break;
}
}
}
if (dtlsTransport != null && srtp) {
srtpProtectionProfile = tlsClient.getChosenProtectionProfile();
tlsContext = tlsClient.getContext();
}
}
// DTLS server
else if (dtlsProtocol instanceof DTLSServerProtocol) {
DTLSServerProtocol dtlsServerProtocol = (DTLSServerProtocol) dtlsProtocol;
TlsServerImpl tlsServer = (TlsServerImpl) tlsPeer;
for (int i = CONNECT_TRIES - 1; i >= 0; i--) {
if (!enterRunInConnectThreadLoop(i, datagramTransport)) break;
try {
dtlsTransport = dtlsServerProtocol.accept(tlsServer, datagramTransport);
break;
} catch (IOException ioe) {
if (!handleRunInConnectThreadException(
ioe, "Failed to accept a connection from a DTLS client!", i)) {
break;
}
}
}
if (dtlsTransport != null && srtp) {
srtpProtectionProfile = tlsServer.getChosenProtectionProfile();
tlsContext = tlsServer.getContext();
}
} else {
// It MUST be either a DTLS client or a DTLS server.
throw new IllegalStateException("dtlsProtocol");
}
SinglePacketTransformer srtpTransformer =
(dtlsTransport == null || !srtp)
? null
: initializeSRTPTransformer(srtpProtectionProfile, tlsContext);
boolean closeSRTPTransformer;
synchronized (this) {
if (Thread.currentThread().equals(this.connectThread)
&& datagramTransport.equals(this.datagramTransport)) {
this.dtlsTransport = dtlsTransport;
_srtpTransformer = srtpTransformer;
notifyAll();
}
closeSRTPTransformer = (_srtpTransformer != srtpTransformer);
}
if (closeSRTPTransformer && srtpTransformer != null) srtpTransformer.close();
}
/**
* Sends the data contained in a specific byte array as application data through the DTLS
* connection of this <tt>DtlsPacketTransformer</tt>.
*
* @param buf the byte array containing data to send.
* @param off the offset in <tt>buf</tt> where the data begins.
* @param len the length of data to send.
*/
public void sendApplicationData(byte[] buf, int off, int len) {
DTLSTransport dtlsTransport = this.dtlsTransport;
Throwable throwable = null;
if (dtlsTransport != null) {
try {
dtlsTransport.send(buf, off, len);
} catch (IOException ioe) {
throwable = ioe;
}
} else {
throwable = new NullPointerException("dtlsTransport");
}
if (throwable != null) {
// SrtpControl.start(MediaType) starts its associated
// TransformEngine. We will use that mediaType to signal the normal
// stop then as well i.e. we will ignore exception after the
// procedure to stop this PacketTransformer has begun.
if (mediaType != null && !tlsPeerHasRaisedCloseNotifyWarning) {
logger.error("Failed to send application data over DTLS transport: ", throwable);
}
}
}
/**
* Sets the <tt>RTPConnector</tt> which is to use or uses this <tt>PacketTransformer</tt>.
*
* @param connector the <tt>RTPConnector</tt> which is to use or uses this
* <tt>PacketTransformer</tt>
*/
void setConnector(AbstractRTPConnector connector) {
if (this.connector != connector) {
this.connector = connector;
DatagramTransportImpl datagramTransport = this.datagramTransport;
if (datagramTransport != null) datagramTransport.setConnector(connector);
}
}
/**
* Sets the <tt>MediaType</tt> of the stream which this instance is to work for/be associated
* with.
*
* @param mediaType the <tt>MediaType</tt> of the stream which this instance is to work for/be
* associated with
*/
synchronized void setMediaType(MediaType mediaType) {
if (this.mediaType != mediaType) {
MediaType oldValue = this.mediaType;
this.mediaType = mediaType;
if (oldValue != null) stop();
if (this.mediaType != null) start();
}
}
/**
* Enables/disables rtcp-mux.
*
* @param rtcpmux whether to enable or disable.
*/
void setRtcpmux(boolean rtcpmux) {
this.rtcpmux = rtcpmux;
}
/**
* Sets the DTLS protocol according to which this <tt>DtlsPacketTransformer</tt> is to act either
* as a DTLS server or a DTLS client.
*
* @param setup the value of the <tt>setup</tt> SDP attribute to set on this instance in order to
* determine whether this instance is to act as a DTLS client or a DTLS server
*/
void setSetup(DtlsControl.Setup setup) {
if (this.setup != setup) this.setup = setup;
}
/** Starts this <tt>PacketTransformer</tt>. */
private synchronized void start() {
if (this.datagramTransport != null) {
if (this.connectThread == null && dtlsTransport == null) {
logger.warn(
getClass().getName()
+ " has been started but has failed to establish"
+ " the DTLS connection!");
}
return;
}
if (rtcpmux && Component.RTCP == componentID) {
// In the case of rtcp-mux, the RTCP transformer does not create
// a DTLS session. The SRTP context (_srtpTransformer) will be
// initialized on demand using initializeSRTCPTransformerFromRtp().
return;
}
AbstractRTPConnector connector = this.connector;
if (connector == null) throw new NullPointerException("connector");
DtlsControl.Setup setup = this.setup;
SecureRandom secureRandom = DtlsControlImpl.createSecureRandom();
final DTLSProtocol dtlsProtocolObj;
final TlsPeer tlsPeer;
if (DtlsControl.Setup.ACTIVE.equals(setup)) {
dtlsProtocolObj = new DTLSClientProtocol(secureRandom);
tlsPeer = new TlsClientImpl(this);
} else {
dtlsProtocolObj = new DTLSServerProtocol(secureRandom);
tlsPeer = new TlsServerImpl(this);
}
tlsPeerHasRaisedCloseNotifyWarning = false;
final DatagramTransportImpl datagramTransport = new DatagramTransportImpl(componentID);
datagramTransport.setConnector(connector);
Thread connectThread =
new Thread() {
@Override
public void run() {
try {
runInConnectThread(dtlsProtocolObj, tlsPeer, datagramTransport);
} finally {
if (Thread.currentThread().equals(DtlsPacketTransformer.this.connectThread)) {
DtlsPacketTransformer.this.connectThread = null;
}
}
}
};
connectThread.setDaemon(true);
connectThread.setName(DtlsPacketTransformer.class.getName() + ".connectThread");
this.connectThread = connectThread;
this.datagramTransport = datagramTransport;
boolean started = false;
try {
connectThread.start();
started = true;
} finally {
if (!started) {
if (connectThread.equals(this.connectThread)) this.connectThread = null;
if (datagramTransport.equals(this.datagramTransport)) this.datagramTransport = null;
}
}
notifyAll();
}
/** Stops this <tt>PacketTransformer</tt>. */
private synchronized void stop() {
if (connectThread != null) connectThread = null;
try {
// The dtlsTransport and _srtpTransformer SHOULD be closed, of
// course. The datagramTransport MUST be closed.
if (dtlsTransport != null) {
try {
dtlsTransport.close();
} catch (IOException ioe) {
logger.error("Failed to (properly) close " + dtlsTransport.getClass(), ioe);
}
dtlsTransport = null;
}
if (_srtpTransformer != null) {
_srtpTransformer.close();
_srtpTransformer = null;
}
} finally {
try {
closeDatagramTransport();
} finally {
notifyAll();
}
}
}
/** {@inheritDoc} */
@Override
public RawPacket transform(RawPacket pkt) {
byte[] buf = pkt.getBuffer();
int off = pkt.getOffset();
int len = pkt.getLength();
// If the specified pkt represents a DTLS record, then it should pass
// through this PacketTransformer (e.g. it has been sent through
// DatagramTransportImpl).
if (isDtlsRecord(buf, off, len)) return pkt;
// SRTP
if (!transformEngine.isSrtpDisabled()) {
// DTLS-SRTP has not been initialized yet or has failed to
// initialize.
SinglePacketTransformer srtpTransformer = waitInitializeAndGetSRTPTransformer();
if (srtpTransformer != null) pkt = srtpTransformer.transform(pkt);
else if (DROP_UNENCRYPTED_PKTS) pkt = null;
// XXX Else, it is our explicit policy to let the received packet
// pass through and rely on the SrtpListener to notify the user that
// the session is not secured.
}
// Pure/non-SRTP DTLS
else {
// The specified pkt will pass through this PacketTransformer only
// if it gets transformed into a DTLS record.
pkt = null;
sendApplicationData(buf, off, len);
}
return pkt;
}
/**
* Gets the {@code SRTPTransformer} used by this instance. If {@link #_srtpTransformer} does not
* exist (yet) and the state of this instance indicates that its initialization is in progess,
* then blocks until {@code _srtpTransformer} is initialized and returns it.
*
* @return the {@code SRTPTransformer} used by this instance
*/
private SinglePacketTransformer waitInitializeAndGetSRTPTransformer() {
SinglePacketTransformer srtpTransformer = _srtpTransformer;
if (srtpTransformer != null) return srtpTransformer;
if (rtcpmux && Component.RTCP == componentID) return initializeSRTCPTransformerFromRtp();
// XXX It is our explicit policy to rely on the SrtpListener to notify
// the user that the session is not secure. Unfortunately, (1) the
// SrtpListener is not supported by this DTLS SrtpControl implementation
// and (2) encrypted packets may arrive soon enough to be let through
// while _srtpTransformer is still initializing. Consequently, we will
// block and wait for _srtpTransformer to initialize.
boolean interrupted = false;
try {
synchronized (this) {
do {
srtpTransformer = _srtpTransformer;
if (srtpTransformer != null) break; // _srtpTransformer is initialized
if (connectThread == null) {
// Though _srtpTransformer is NOT initialized, there is
// no point in waiting because there is no one to
// initialize it.
break;
}
try {
// It does not really matter (enough) how much we wait
// here because we wait in a loop.
long timeout = CONNECT_TRIES * CONNECT_RETRY_INTERVAL;
wait(timeout);
} catch (InterruptedException ie) {
interrupted = true;
}
} while (true);
}
} finally {
if (interrupted) Thread.currentThread().interrupt();
}
return srtpTransformer;
}
}
/**
* Intercepts RTX (RFC-4588) packets coming from an {@link RtpChannel}, and removes their RTX
* encapsulation. Allows packets to be retransmitted to a channel (using the RTX format if the
* destination supports it).
*
* @author Boris Grozev
* @author George Politis
*/
public class RtxTransformer extends SinglePacketTransformerAdapter implements TransformEngine {
/**
* The {@link Logger} used by the {@link RtxTransformer} class to print debug information. Note
* that {@link Conference} instances should use {@link #logger} instead.
*/
private static final Logger classLogger = Logger.getLogger(RtxTransformer.class);
/** The <tt>RtpChannel</tt> for the transformer. */
private RtpChannel channel;
/** Maps an RTX SSRC to the last RTP sequence number sent with that SSRC. */
private final Map<Long, Integer> rtxSequenceNumbers = new HashMap<>();
/** The {@link Logger} to be used by this instance to print debug information. */
private final Logger logger;
/**
* The payload type number configured for RTX (RFC-4588), or -1 if none is configured (the other
* end does not support rtx).
*/
private byte rtxPayloadType = -1;
/** The "associated payload type" number for RTX. */
private byte rtxAssociatedPayloadType = -1;
/**
* Initializes a new <tt>RtxTransformer</tt> with a specific <tt>RtpChannel</tt>.
*
* @param channel the <tt>RtpChannel</tt> for the transformer.
*/
RtxTransformer(RtpChannel channel) {
super(RTPPacketPredicate.INSTANCE);
this.channel = channel;
this.logger = Logger.getLogger(classLogger, channel.getContent().getConference().getLogger());
}
/** Implements {@link PacketTransformer#transform(RawPacket[])}. {@inheritDoc} */
@Override
public RawPacket reverseTransform(RawPacket pkt) {
if (isRtx(pkt)) {
pkt = deRtx(pkt);
}
return pkt;
}
/**
* Determines whether {@code pkt} is an RTX packet.
*
* @param pkt the packet to check.
* @return {@code true} iff {@code pkt} is an RTX packet.
*/
private boolean isRtx(RawPacket pkt) {
byte rtxPt = rtxPayloadType;
return rtxPt != -1 && rtxPt == pkt.getPayloadType();
}
/**
* Removes the RTX encapsulation from a packet.
*
* @param pkt the packet to remove the RTX encapsulation from.
* @return the original media packet represented by {@code pkt}, or null if we couldn't
* reconstruct the original packet.
*/
private RawPacket deRtx(RawPacket pkt) {
boolean success = false;
if (pkt.getPayloadLength() - pkt.getPaddingSize() < 2) {
// We need at least 2 bytes to read the OSN field.
if (logger.isDebugEnabled()) {
logger.debug("Dropping an incoming RTX packet with padding only: " + pkt);
}
return null;
}
long mediaSsrc = getPrimarySsrc(pkt);
if (mediaSsrc != -1) {
if (rtxAssociatedPayloadType != -1) {
int osn = pkt.getOriginalSequenceNumber();
// Remove the RTX header by moving the RTP header two bytes
// right.
byte[] buf = pkt.getBuffer();
int off = pkt.getOffset();
System.arraycopy(buf, off, buf, off + 2, pkt.getHeaderLength());
pkt.setOffset(off + 2);
pkt.setLength(pkt.getLength() - 2);
pkt.setSSRC((int) mediaSsrc);
pkt.setSequenceNumber(osn);
pkt.setPayloadType(rtxAssociatedPayloadType);
success = true;
} else {
logger.warn(
"RTX packet received, but no APT is defined. Packet "
+ "SSRC "
+ pkt.getSSRCAsLong()
+ ", associated media"
+ " SSRC "
+ mediaSsrc);
}
}
// If we failed to handle the RTX packet, drop it.
return success ? pkt : null;
}
/** Implements {@link TransformEngine#getRTPTransformer()}. */
@Override
public PacketTransformer getRTPTransformer() {
return this;
}
/** Implements {@link TransformEngine#getRTCPTransformer()}. */
@Override
public PacketTransformer getRTCPTransformer() {
return null;
}
/**
* Returns the sequence number to use for a specific RTX packet, which is based on the packet's
* original sequence number.
*
* <p>Because we terminate the RTX format, and with simulcast we might translate RTX packets from
* multiple SSRCs into the same SSRC, we keep count of the RTX packets (and their sequence
* numbers) which we sent for each SSRC.
*
* @param ssrc the SSRC of the RTX stream for the packet.
* @return the sequence number which should be used for the next RTX packet sent using SSRC
* <tt>ssrc</tt>.
*/
private int getNextRtxSequenceNumber(long ssrc) {
Integer seq;
synchronized (rtxSequenceNumbers) {
seq = rtxSequenceNumbers.get(ssrc);
if (seq == null) seq = new Random().nextInt(0xffff);
else seq++;
rtxSequenceNumbers.put(ssrc, seq);
}
return seq;
}
/**
* Tries to find an SSRC paired with {@code ssrc} in an FID group in one of the channels from
* {@link #channel}'s {@code Content}. Returns -1 on failure.
*
* @param pkt the {@code RawPacket} that holds the RTP packet for which to find a paired SSRC.
* @return An SSRC paired with {@code ssrc} in an FID group, or -1.
*/
private long getRtxSsrc(RawPacket pkt) {
StreamRTPManager receiveRTPManager =
channel.getStream().getRTPTranslator().findStreamRTPManagerByReceiveSSRC(pkt.getSSRC());
MediaStreamTrackReceiver receiver = null;
if (receiveRTPManager != null) {
MediaStream receiveStream = receiveRTPManager.getMediaStream();
if (receiveStream != null) {
receiver = receiveStream.getMediaStreamTrackReceiver();
}
}
if (receiver == null) {
return -1;
}
RTPEncoding encoding = receiver.resolveRTPEncoding(pkt);
if (encoding == null) {
logger.warn(
"encoding_not_found"
+ ",stream_hash="
+ channel.getStream().hashCode()
+ " ssrc="
+ pkt.getSSRCAsLong());
return -1;
}
return encoding.getRTXSSRC();
}
/**
* Retransmits a packet to {@link #channel}. If the destination supports the RTX format, the
* packet will be encapsulated in RTX, otherwise, the packet will be retransmitted as-is.
*
* @param pkt the packet to retransmit.
* @param after the {@code TransformEngine} in the chain of {@code TransformEngine}s of the
* associated {@code MediaStream} after which the injection of {@code pkt} is to begin
* @return {@code true} if the packet was successfully retransmitted, {@code false} otherwise.
*/
public boolean retransmit(RawPacket pkt, TransformEngine after) {
boolean destinationSupportsRtx = rtxPayloadType != -1;
boolean retransmitPlain;
if (destinationSupportsRtx) {
long rtxSsrc = getRtxSsrc(pkt);
if (rtxSsrc == -1) {
logger.warn(
"Cannot find SSRC for RTX, retransmitting plain. " + "SSRC=" + pkt.getSSRCAsLong());
retransmitPlain = true;
} else {
retransmitPlain = !encapsulateInRtxAndTransmit(pkt, rtxSsrc, after);
}
} else {
retransmitPlain = true;
}
if (retransmitPlain) {
MediaStream mediaStream = channel.getStream();
if (mediaStream != null) {
try {
mediaStream.injectPacket(pkt, /* data */ true, after);
} catch (TransmissionFailedException tfe) {
logger.warn("Failed to retransmit a packet.");
return false;
}
}
}
return true;
}
/**
* Notifies this instance that the dynamic payload types of the associated {@link MediaStream}
* have changed.
*/
public void onDynamicPayloadTypesChanged() {
rtxPayloadType = -1;
rtxAssociatedPayloadType = -1;
MediaStream mediaStream = channel.getStream();
Map<Byte, MediaFormat> mediaFormatMap = mediaStream.getDynamicRTPPayloadTypes();
Iterator<Map.Entry<Byte, MediaFormat>> it = mediaFormatMap.entrySet().iterator();
while (it.hasNext() && rtxPayloadType == -1) {
Map.Entry<Byte, MediaFormat> entry = it.next();
MediaFormat format = entry.getValue();
if (!Constants.RTX.equalsIgnoreCase(format.getEncoding())) {
continue;
}
// XXX(gp) we freak out if multiple codecs with RTX support are
// present.
rtxPayloadType = entry.getKey();
rtxAssociatedPayloadType = Byte.parseByte(format.getFormatParameters().get("apt"));
}
}
/**
* Encapsulates {@code pkt} in the RTX format, using {@code rtxSsrc} as its SSRC, and transmits it
* to {@link #channel} by injecting it in the {@code MediaStream}.
*
* @param pkt the packet to transmit.
* @param rtxSsrc the SSRC for the RTX stream.
* @param after the {@code TransformEngine} in the chain of {@code TransformEngine}s of the
* associated {@code MediaStream} after which the injection of {@code pkt} is to begin
* @return {@code true} if the packet was successfully retransmitted, {@code false} otherwise.
*/
private boolean encapsulateInRtxAndTransmit(RawPacket pkt, long rtxSsrc, TransformEngine after) {
byte[] buf = pkt.getBuffer();
int len = pkt.getLength();
int off = pkt.getOffset();
byte[] newBuf = new byte[len + 2];
RawPacket rtxPkt = new RawPacket(newBuf, 0, len + 2);
int osn = pkt.getSequenceNumber();
int headerLength = pkt.getHeaderLength();
int payloadLength = pkt.getPayloadLength();
// Copy the header.
System.arraycopy(buf, off, newBuf, 0, headerLength);
// Set the OSN field.
newBuf[headerLength] = (byte) ((osn >> 8) & 0xff);
newBuf[headerLength + 1] = (byte) (osn & 0xff);
// Copy the payload.
System.arraycopy(buf, off + headerLength, newBuf, headerLength + 2, payloadLength);
MediaStream mediaStream = channel.getStream();
if (mediaStream != null) {
rtxPkt.setSSRC((int) rtxSsrc);
rtxPkt.setPayloadType(rtxPayloadType);
// Only call getNextRtxSequenceNumber() when we're sure we're going
// to transmit a packet, because it consumes a sequence number.
rtxPkt.setSequenceNumber(getNextRtxSequenceNumber(rtxSsrc));
try {
mediaStream.injectPacket(rtxPkt, /* data */ true, after);
} catch (TransmissionFailedException tfe) {
logger.warn("Failed to transmit an RTX packet.");
return false;
}
}
return true;
}
/**
* Returns the SSRC paired with <tt>ssrc</tt> in an FID source-group, if any. If none is found,
* returns -1.
*
* @return the SSRC paired with <tt>ssrc</tt> in an FID source-group, if any. If none is found,
* returns -1.
*/
private long getPrimarySsrc(RawPacket pkt) {
MediaStreamTrackReceiver receiver = channel.getStream().getMediaStreamTrackReceiver();
if (receiver == null) {
if (logger.isDebugEnabled()) {
logger.debug("Dropping an incoming RTX packet from an unknown source.");
}
return -1;
}
RTPEncoding encoding = receiver.resolveRTPEncoding(pkt);
if (encoding == null) {
if (logger.isDebugEnabled()) {
logger.debug("Dropping an incoming RTX packet from an unknown source.");
}
return -1;
}
return encoding.getPrimarySSRC();
}
}
/**
* Initializes a new <tt>RtxTransformer</tt> with a specific <tt>RtpChannel</tt>.
*
* @param channel the <tt>RtpChannel</tt> for the transformer.
*/
RtxTransformer(RtpChannel channel) {
super(RTPPacketPredicate.INSTANCE);
this.channel = channel;
this.logger = Logger.getLogger(classLogger, channel.getContent().getConference().getLogger());
}
/**
* The <tt>BasicRTCPTerminationStrategy</tt> "gateways" PLIs, FIRs, NACKs, etc, in the sense that it
* replaces the packet sender information in the PLIs, FIRs, NACKs, etc and it generates its own
* SRs/RRs/REMBs based on information that it collects and from information found in FMJ.
*
* @author George Politis
*/
public class BasicRTCPTerminationStrategy extends MediaStreamRTCPTerminationStrategy {
/**
* The <tt>Logger</tt> used by the <tt>BasicRTCPTerminationStrategy</tt> class and its instances
* to print debug information.
*/
private static final Logger logger = Logger.getLogger(BasicRTCPTerminationStrategy.class);
/** The maximum number of RTCP report blocks that an RR or an SR can contain. */
private static final int MAX_RTCP_REPORT_BLOCKS = 31;
/** The minimum number of RTCP report blocks that an RR or an SR can contain. */
private static final int MIN_RTCP_REPORT_BLOCKS = 0;
/**
* A reusable array that can be used to hold up to <tt>MAX_RTCP_REPORT_BLOCKS</tt>
* <tt>RTCPReportBlock</tt>s. It is assumed that a single thread is accessing this field at a
* given time.
*/
private final RTCPReportBlock[] MAX_RTCP_REPORT_BLOCKS_ARRAY =
new RTCPReportBlock[MAX_RTCP_REPORT_BLOCKS];
/** A reusable array that holds 0 <tt>RTCPReportBlock</tt>s. */
private static final RTCPReportBlock[] MIN_RTCP_REPORTS_BLOCKS_ARRAY =
new RTCPReportBlock[MIN_RTCP_REPORT_BLOCKS];
/**
* The RTP stats map that holds RTP statistics about all the streams that this
* <tt>BasicRTCPTerminationStrategy</tt> (as a <tt>TransformEngine</tt>) has observed.
*/
private final RTPStatsMap rtpStatsMap = new RTPStatsMap();
/**
* The RTCP stats map that holds RTCP statistics about all the streams that this
* <tt>BasicRTCPTerminationStrategy</tt> (as a <tt>TransformEngine</tt>) has observed.
*/
private final RemoteClockEstimator remoteClockEstimator = new RemoteClockEstimator();
/**
* The <tt>CNameRegistry</tt> holds the CNAMEs that this RTCP termination, seen as a
* TransformEngine, has seen.
*/
private final CNAMERegistry cnameRegistry = new CNAMERegistry();
/** The parser that parses <tt>RawPacket</tt>s to <tt>RTCPCompoundPacket</tt>s. */
private final RTCPPacketParserEx parser = new RTCPPacketParserEx();
/** The generator that generates <tt>RawPacket</tt>s from <tt>RTCPCompoundPacket</tt>s. */
private final RTCPGenerator generator = new RTCPGenerator();
/**
* The RTCP feedback gateway responsible for dropping all the stuff that we support in this RTCP
* termination strategy.
*/
private final FeedbackGateway feedbackGateway = new FeedbackGateway();
/** The garbage collector that cleans-up the state of this RTCP termination strategy. */
private final GarbageCollector garbageCollector = new GarbageCollector();
/** The RTP <tt>PacketTransformer</tt> of this <tt>BasicRTCPTerminationStrategy</tt>. */
private final PacketTransformer rtpTransformer =
new SinglePacketTransformer() {
/** {@inheritDoc} */
@Override
public RawPacket transform(RawPacket pkt) {
// Update our RTP stats map (packets/octet sent).
rtpStatsMap.apply(pkt);
return pkt;
}
/** {@inheritDoc} */
@Override
public RawPacket reverseTransform(RawPacket pkt) {
// Let everything pass through.
return pkt;
}
};
/** The RTCP <tt>PacketTransformer</tt> of this <tt>BasicRTCPTerminationStrategy</tt>. */
private final PacketTransformer rtcpTransformer =
new SinglePacketTransformer() {
/** {@inheritDoc} */
@Override
public RawPacket transform(RawPacket pkt) {
if (pkt == null) {
return pkt;
}
RTCPCompoundPacket inPacket;
try {
inPacket =
(RTCPCompoundPacket)
parser.parse(pkt.getBuffer(), pkt.getOffset(), pkt.getLength());
} catch (BadFormatException e) {
logger.warn("Failed to terminate an RTCP packet. " + "Dropping packet.");
return null;
}
// Update our RTCP stats map (timestamps). This operation is
// read-only.
remoteClockEstimator.apply(inPacket);
cnameRegistry.update(inPacket);
// Remove SRs and RRs from the RTCP packet.
pkt = feedbackGateway.gateway(inPacket);
return pkt;
}
/** {@inheritDoc} */
@Override
public RawPacket reverseTransform(RawPacket pkt) {
// Let everything pass through.
return pkt;
}
};
/** A counter that counts the number of times we've sent "full-blown" SDES. */
private int sdesCounter = 0;
/** {@inheritDoc} */
@Override
public PacketTransformer getRTPTransformer() {
return rtpTransformer;
}
/** {@inheritDoc} */
@Override
public PacketTransformer getRTCPTransformer() {
return rtcpTransformer;
}
/** {@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);
}
/**
* (attempts) to get the local SSRC that will be used in the media sender SSRC field of the RTCP
* reports. TAG(cat4-local-ssrc-hurricane)
*
* @return
*/
private long getLocalSSRC() {
return getStream().getStreamRTPManager().getLocalSSRC();
}
/**
* Makes <tt>RTCPRRPacket</tt>s using information in FMJ.
*
* @param time
* @return A <tt>Collection</tt> of <tt>RTCPRRPacket</tt>s to inject to the <tt>MediaStream</tt>.
*/
private Collection<RTCPRRPacket> makeRTCPRRPackets(long time) {
RTCPReportBlock[] reportBlocks = makeRTCPReportBlocks(time);
if (reportBlocks == null || reportBlocks.length == 0) {
return null;
}
Collection<RTCPRRPacket> rrPackets = new ArrayList<RTCPRRPacket>();
// We use the stream's local source ID (SSRC) as the SSRC of packet
// sender.
long streamSSRC = getLocalSSRC();
// Since a maximum of 31 reception report blocks will fit in an SR
// or RR packet, additional RR packets SHOULD be stacked after the
// initial SR or RR packet as needed to contain the reception
// reports for all sources heard during the interval since the last
// report.
if (reportBlocks.length > MAX_RTCP_REPORT_BLOCKS) {
for (int offset = 0; offset < reportBlocks.length; offset += MAX_RTCP_REPORT_BLOCKS) {
RTCPReportBlock[] blocks =
(reportBlocks.length - offset < MAX_RTCP_REPORT_BLOCKS)
? new RTCPReportBlock[reportBlocks.length - offset]
: MAX_RTCP_REPORT_BLOCKS_ARRAY;
System.arraycopy(reportBlocks, offset, blocks, 0, blocks.length);
RTCPRRPacket rr = new RTCPRRPacket((int) streamSSRC, blocks);
rrPackets.add(rr);
}
} else {
RTCPRRPacket rr = new RTCPRRPacket((int) streamSSRC, reportBlocks);
rrPackets.add(rr);
}
return rrPackets;
}
/**
* Iterate through all the <tt>ReceiveStream</tt>s that this <tt>MediaStream</tt> has and make
* <tt>RTCPReportBlock</tt>s for all of them.
*
* @param time
* @return
*/
private RTCPReportBlock[] makeRTCPReportBlocks(long time) {
MediaStream stream = getStream();
// State validation.
if (stream == null) {
logger.warn("stream is null.");
return MIN_RTCP_REPORTS_BLOCKS_ARRAY;
}
StreamRTPManager streamRTPManager = stream.getStreamRTPManager();
if (streamRTPManager == null) {
logger.warn("streamRTPManager is null.");
return MIN_RTCP_REPORTS_BLOCKS_ARRAY;
}
Collection<ReceiveStream> receiveStreams = streamRTPManager.getReceiveStreams();
if (receiveStreams == null || receiveStreams.size() == 0) {
logger.info("There are no receive streams to build report " + "blocks for.");
return MIN_RTCP_REPORTS_BLOCKS_ARRAY;
}
SSRCCache cache = streamRTPManager.getSSRCCache();
if (cache == null) {
logger.info("cache is null.");
return MIN_RTCP_REPORTS_BLOCKS_ARRAY;
}
// Create the return object.
Collection<RTCPReportBlock> rtcpReportBlocks = new ArrayList<RTCPReportBlock>();
// Populate the return object.
for (ReceiveStream receiveStream : receiveStreams) {
// Dig into the guts of FMJ and get the stats for the current
// receiveStream.
SSRCInfo info = cache.cache.get((int) receiveStream.getSSRC());
if (!info.ours && info.sender) {
RTCPReportBlock rtcpReportBlock = info.makeReceiverReport(time);
rtcpReportBlocks.add(rtcpReportBlock);
}
}
return rtcpReportBlocks.toArray(new RTCPReportBlock[rtcpReportBlocks.size()]);
}
/**
* Makes an <tt>RTCPREMBPacket</tt> that provides receiver feedback to the endpoint from which we
* receive.
*
* @return an <tt>RTCPREMBPacket</tt> that provides receiver feedback to the endpoint from which
* we receive.
*/
private RTCPREMBPacket makeRTCPREMBPacket() {
// TODO we should only make REMBs if REMB support has been advertised.
// Destination
RemoteBitrateEstimator remoteBitrateEstimator =
((VideoMediaStream) getStream()).getRemoteBitrateEstimator();
Collection<Integer> ssrcs = remoteBitrateEstimator.getSsrcs();
// TODO(gp) intersect with SSRCs from signaled simulcast layers
// NOTE(gp) The Google Congestion Control algorithm (sender side)
// doesn't seem to care about the SSRCs in the dest field.
long[] dest = new long[ssrcs.size()];
int i = 0;
for (Integer ssrc : ssrcs) dest[i++] = ssrc & 0xFFFFFFFFL;
// Exp & mantissa
long bitrate = remoteBitrateEstimator.getLatestEstimate();
if (bitrate == -1) return null;
if (logger.isDebugEnabled()) logger.debug("Estimated bitrate: " + bitrate);
// Create and return the packet.
// We use the stream's local source ID (SSRC) as the SSRC of packet
// sender.
long streamSSRC = getLocalSSRC();
return new RTCPREMBPacket(streamSSRC, /* mediaSSRC */ 0L, bitrate, dest);
}
/**
* Makes <tt>RTCPSRPacket</tt>s for all the RTP streams that we're sending.
*
* @return a <tt>List</tt> of <tt>RTCPSRPacket</tt> for all the RTP streams that we're sending.
*/
private Collection<RTCPSRPacket> makeRTCPSRPackets(long time) {
Collection<RTCPSRPacket> srPackets = new ArrayList<RTCPSRPacket>();
for (RTPStatsEntry rtpStatsEntry : rtpStatsMap.values()) {
int ssrc = rtpStatsEntry.getSsrc();
RemoteClock estimate = remoteClockEstimator.estimate(ssrc, time);
if (estimate == null) {
// We're not going to go far without an estimate..
continue;
}
RTCPSRPacket srPacket = new RTCPSRPacket(ssrc, MIN_RTCP_REPORTS_BLOCKS_ARRAY);
// Set the NTP timestamp for this SR.
long estimatedRemoteTime = estimate.getRemoteTime();
long secs = estimatedRemoteTime / 1000L;
double fraction = (estimatedRemoteTime - secs * 1000L) / 1000D;
srPacket.ntptimestamplsw = (int) (fraction * 4294967296D);
srPacket.ntptimestampmsw = secs;
// Set the RTP timestamp.
srPacket.rtptimestamp = estimate.getRtpTimestamp();
// Fill-in packet and octet send count.
srPacket.packetcount = rtpStatsEntry.getPacketsSent();
srPacket.octetcount = rtpStatsEntry.getBytesSent();
srPackets.add(srPacket);
}
return srPackets;
}
/**
* Makes <tt>RTCPSDES</tt> packets for all the RTP streams that we're sending.
*
* @return a <tt>List</tt> of <tt>RTCPSDES</tt> packets for all the RTP streams that we're
* sending.
*/
private RTCPSDESPacket makeSDESPacket() {
Collection<RTCPSDES> sdesChunks = new ArrayList<RTCPSDES>();
// Create an SDES for our own SSRC.
RTCPSDES ownSDES = new RTCPSDES();
SSRCInfo ourinfo = getStream().getStreamRTPManager().getSSRCCache().ourssrc;
ownSDES.ssrc = (int) getLocalSSRC();
Collection<RTCPSDESItem> ownItems = new ArrayList<RTCPSDESItem>();
ownItems.add(new RTCPSDESItem(RTCPSDESItem.CNAME, ourinfo.sourceInfo.getCNAME()));
// Throttle the source description bandwidth. See RFC3550#6.3.9
// Allocation of Source Description Bandwidth.
if (sdesCounter % 3 == 0) {
if (ourinfo.name != null && ourinfo.name.getDescription() != null)
ownItems.add(new RTCPSDESItem(RTCPSDESItem.NAME, ourinfo.name.getDescription()));
if (ourinfo.email != null && ourinfo.email.getDescription() != null)
ownItems.add(new RTCPSDESItem(RTCPSDESItem.EMAIL, ourinfo.email.getDescription()));
if (ourinfo.phone != null && ourinfo.phone.getDescription() != null)
ownItems.add(new RTCPSDESItem(RTCPSDESItem.PHONE, ourinfo.phone.getDescription()));
if (ourinfo.loc != null && ourinfo.loc.getDescription() != null)
ownItems.add(new RTCPSDESItem(RTCPSDESItem.LOC, ourinfo.loc.getDescription()));
if (ourinfo.tool != null && ourinfo.tool.getDescription() != null)
ownItems.add(new RTCPSDESItem(RTCPSDESItem.TOOL, ourinfo.tool.getDescription()));
if (ourinfo.note != null && ourinfo.note.getDescription() != null)
ownItems.add(new RTCPSDESItem(RTCPSDESItem.NOTE, ourinfo.note.getDescription()));
}
sdesCounter++;
ownSDES.items = ownItems.toArray(new RTCPSDESItem[ownItems.size()]);
sdesChunks.add(ownSDES);
for (Map.Entry<Integer, byte[]> entry : cnameRegistry.entrySet()) {
RTCPSDES sdes = new RTCPSDES();
sdes.ssrc = entry.getKey();
sdes.items = new RTCPSDESItem[] {new RTCPSDESItem(RTCPSDESItem.CNAME, entry.getValue())};
}
RTCPSDES[] sps = sdesChunks.toArray(new RTCPSDES[sdesChunks.size()]);
RTCPSDESPacket sp = new RTCPSDESPacket(sps);
return sp;
}
/**
* The garbage collector runs at each reporting interval and cleans up the data structures of this
* RTCP termination strategy based on the SSRCs that the owner <tt>MediaStream</tt> is still
* sending.
*/
class GarbageCollector {
public void cleanup() {
// TODO We need to fix TAG(cat4-local-ssrc-hurricane) and
// TAG(cat4-remote-ssrc-hurricane) first. The idea is to remove
// from our data structures everything that is not listed in as
// a remote SSRC.
}
}
/**
* Removes receiver and sender feedback from RTCP packets. Typically this means dropping SRs, RR
* report blocks and REMBs. It needs to pass through PLIs, FIRs, NACKs, etc.
*/
class FeedbackGateway {
/**
* 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);
}
}
/** Holds the NTP timestamp and the associated RTP timestamp for a given RTP stream. */
class RemoteClock {
/**
* Ctor.
*
* @param remoteTime
* @param rtpTimestamp
*/
public RemoteClock(long remoteTime, int rtpTimestamp) {
this.remoteTime = remoteTime;
this.rtpTimestamp = rtpTimestamp;
}
/**
* The last NTP timestamp that we received for {@link this.ssrc} expressed in millis. Should be
* treated a signed long.
*/
private final long remoteTime;
/**
* The RTP timestamp associated to {@link this.ntpTimestamp}. The RTP timestamp is an unsigned
* int.
*/
private final int rtpTimestamp;
/** @return */
public int getRtpTimestamp() {
return rtpTimestamp;
}
/** @return */
public long getRemoteTime() {
return remoteTime;
}
}
/** */
class ReceivedRemoteClock {
/** The SSRC. */
private final int ssrc;
/**
* The <tt>RemoteClock</tt> which was received at {@link this.receivedTime} for this RTP stream.
*/
private final RemoteClock remoteClock;
/**
* The local time in millis when we received the RTCP report with the RTP/NTP timestamps. It's a
* signed long.
*/
private final long receivedTime;
/**
* The clock rate for {@link.ssrc}. We need to have received at least two SRs in order to be
* able to calculate this. Unsigned short.
*/
private final int frequencyHz;
/**
* 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();
}
/** @return */
public RemoteClock getRemoteClock() {
return remoteClock;
}
/** @return */
public long getReceivedTime() {
return receivedTime;
}
/** @return */
public int getSsrc() {
return ssrc;
}
/** @return */
public int getFrequencyHz() {
return frequencyHz;
}
}
/** The <tt>RTPStatsEntry</tt> class contains information about an outgoing SSRC. */
class RTPStatsEntry {
/** The SSRC of the stream that this instance tracks. */
private final int ssrc;
/**
* The total number of _payload_ octets (i.e., not including header or padding) transmitted in
* RTP data packets by the sender since starting transmission up until the time this SR packet
* was generated. This should be treated as an unsigned int.
*/
private final int bytesSent;
/**
* The total number of RTP data packets transmitted by the sender (including re-transmissions)
* since starting transmission up until the time this SR packet was generated. Re-transmissions
* using an RTX stream are tracked in the RTX SSRC. This should be treated as an unsigned int.
*/
private final int packetsSent;
/** @return */
public int getSsrc() {
return ssrc;
}
/** @return */
public int getBytesSent() {
return bytesSent;
}
/** @return */
public int getPacketsSent() {
return packetsSent;
}
/**
* Ctor.
*
* @param ssrc
* @param bytesSent
*/
RTPStatsEntry(int ssrc, int bytesSent, int packetsSent) {
this.ssrc = ssrc;
this.bytesSent = bytesSent;
this.packetsSent = packetsSent;
}
}
/**
* The <tt>RtpStatsMap</tt> gathers stats from RTP packets that the <tt>RTCPReportBuilder</tt>
* uses to build its reports.
*/
class RTPStatsMap extends ConcurrentHashMap<Integer, RTPStatsEntry> {
/**
* Updates this <tt>RTPStatsMap</tt> with information it gets from the <tt>RawPacket</tt>.
*
* @param pkt the <tt>RawPacket</tt> that is being transmitted.
*/
public void apply(RawPacket pkt) {
int ssrc = pkt.getSSRC();
if (this.containsKey(ssrc)) {
RTPStatsEntry oldRtpStatsEntry = this.get(ssrc);
// Replace whatever was in there before. A feature of the two's
// complement encoding (which is used by Java integers) is that
// the bitwise results for add, subtract, and multiply are the
// same if both inputs are interpreted as signed values or both
// inputs are interpreted as unsigned values. (Other encodings
// like one's complement and signed magnitude don't have this
// properly.)
this.put(
ssrc,
new RTPStatsEntry(
ssrc,
oldRtpStatsEntry.getBytesSent()
+ pkt.getLength()
- pkt.getHeaderLength()
- pkt.getPaddingSize(),
oldRtpStatsEntry.getPacketsSent() + 1));
} else {
// Add a new <tt>RTPStatsEntry</tt> in this map.
this.put(
ssrc,
new RTPStatsEntry(
ssrc, pkt.getLength() - pkt.getHeaderLength() - pkt.getPaddingSize(), 1));
}
}
}
/** A class that can be used to estimate the remote time at a given local time. */
class RemoteClockEstimator {
/** base: 7-Feb-2036 @ 06:28:16 UTC */
private static final long msb0baseTime = 2085978496000L;
/** base: 1-Jan-1900 @ 01:00:00 UTC */
private static final long msb1baseTime = -2208988800000L;
/** A map holding the received remote clocks. */
private Map<Integer, ReceivedRemoteClock> receivedClocks =
new ConcurrentHashMap<Integer, ReceivedRemoteClock>();
/**
* Inspect an <tt>RTCPCompoundPacket</tt> and build-up the state for future estimations.
*
* @param pkt
*/
public void apply(RTCPCompoundPacket pkt) {
if (pkt == null || pkt.packets == null || pkt.packets.length == 0) {
return;
}
for (RTCPPacket rtcpPacket : pkt.packets) {
switch (rtcpPacket.type) {
case RTCPPacket.SR:
RTCPSRPacket srPacket = (RTCPSRPacket) rtcpPacket;
// The media sender SSRC.
int ssrc = srPacket.ssrc;
// Convert 64-bit NTP timestamp to Java standard time.
// Note that java time (milliseconds) by definition has
// less precision then NTP time (picoseconds) so
// converting NTP timestamp to java time and back to NTP
// timestamp loses precision. For example, Tue, Dec 17
// 2002 09:07:24.810 EST is represented by a single
// Java-based time value of f22cd1fc8a, but its NTP
// equivalent are all values ranging from
// c1a9ae1c.cf5c28f5 to c1a9ae1c.cf9db22c.
// Use round-off on fractional part to preserve going to
// lower precision
long fraction = Math.round(1000D * srPacket.ntptimestamplsw / 0x100000000L);
/*
* If the most significant bit (MSB) on the seconds
* field is set we use a different time base. The
* following text is a quote from RFC-2030 (SNTP v4):
*
* If bit 0 is set, the UTC time is in the range
* 1968-2036 and UTC time is reckoned from 0h 0m 0s UTC
* on 1 January 1900. If bit 0 is not set, the time is
* in the range 2036-2104 and UTC time is reckoned from
* 6h 28m 16s UTC on 7 February 2036.
*/
long msb = srPacket.ntptimestampmsw & 0x80000000L;
long remoteTime =
(msb == 0)
// use base: 7-Feb-2036 @ 06:28:16 UTC
? msb0baseTime + (srPacket.ntptimestampmsw * 1000) + fraction
// use base: 1-Jan-1900 @ 01:00:00 UTC
: msb1baseTime + (srPacket.ntptimestampmsw * 1000) + fraction;
// Estimate the clock rate of the sender.
int frequencyHz = -1;
if (receivedClocks.containsKey(ssrc)) {
// Calculate the clock rate.
ReceivedRemoteClock oldStats = receivedClocks.get(ssrc);
RemoteClock oldRemoteClock = oldStats.getRemoteClock();
frequencyHz =
Math.round(
(float)
(((int) srPacket.rtptimestamp - oldRemoteClock.getRtpTimestamp())
& 0xffffffffl)
/ (remoteTime - oldRemoteClock.getRemoteTime()));
}
// Replace whatever was in there before.
receivedClocks.put(
ssrc,
new ReceivedRemoteClock(
ssrc, remoteTime, (int) srPacket.rtptimestamp, frequencyHz));
break;
case RTCPPacket.SDES:
break;
}
}
}
/**
* Estimate the <tt>RemoteClock</tt> of a given RTP stream (identified by its SSRC) at a given
* time.
*
* @param ssrc the SSRC of the RTP stream whose <tt>RemoteClock</tt> we want to estimate.
* @param time the local time that will be mapped to a remote time.
* @return An estimation of the <tt>RemoteClock</tt> at time "time".
*/
public RemoteClock estimate(int ssrc, long time) {
ReceivedRemoteClock receivedRemoteClock = receivedClocks.get(ssrc);
if (receivedRemoteClock == null || receivedRemoteClock.getFrequencyHz() == -1) {
// We can't continue if we don't have NTP and RTP timestamps
// and/or the original sender frequency, so move to the next
// one.
return null;
}
long delayMillis = time - receivedRemoteClock.getReceivedTime();
// Estimate the remote wall clock.
long remoteTime = receivedRemoteClock.getRemoteClock().getRemoteTime();
long estimatedRemoteTime = remoteTime + delayMillis;
// Drift the RTP timestamp.
int rtpTimestamp =
receivedRemoteClock.getRemoteClock().getRtpTimestamp()
+ ((int) delayMillis) * (receivedRemoteClock.getFrequencyHz() / 1000);
return new RemoteClock(estimatedRemoteTime, rtpTimestamp);
}
}
/** Keeps track of the CNAMEs of the RTP streams that we've seen. */
class CNAMERegistry extends ConcurrentHashMap<Integer, byte[]> {
/** @param inPacket */
public void update(RTCPCompoundPacket inPacket) {
// Update CNAMEs.
if (inPacket == null || inPacket.packets == null || inPacket.packets.length == 0) {
return;
}
for (RTCPPacket p : inPacket.packets) {
switch (p.type) {
case RTCPPacket.SDES:
RTCPSDESPacket sdesPacket = (RTCPSDESPacket) p;
if (sdesPacket.sdes == null || sdesPacket.sdes.length == 0) {
continue;
}
for (RTCPSDES chunk : sdesPacket.sdes) {
if (chunk.items == null || chunk.items.length == 0) {
continue;
}
for (RTCPSDESItem sdesItm : chunk.items) {
if (sdesItm.type != RTCPSDESItem.CNAME) {
continue;
}
this.put(chunk.ssrc, sdesItm.data);
}
}
break;
}
}
}
}
}
/**
* Implements {@link ReceiveStreamListener#update(ReceiveStreamEvent)}.
*
* <p>{@link #rtpManager} will use this to notify us of <tt>ReceiveStreamEvent</tt>s.
*/
@Override
public void update(ReceiveStreamEvent event) {
if (event == null) return;
ReceiveStream receiveStream = event.getReceiveStream();
if (event instanceof NewReceiveStreamEvent) {
if (receiveStream == null) {
logger.warn("NewReceiveStreamEvent: null");
return;
}
final long ssrc = getReceiveStreamSSRC(receiveStream);
ReceiveStreamDesc receiveStreamDesc = findReceiveStream(ssrc);
if (receiveStreamDesc != null) {
String s = "NewReceiveStreamEvent for an existing SSRC. ";
if (receiveStream != receiveStreamDesc.receiveStream)
s += "(but different ReceiveStream object)";
logger.warn(s);
return;
} else receiveStreamDesc = new ReceiveStreamDesc(receiveStream);
if (logger.isInfoEnabled()) logger.info("New ReceiveStream, ssrc=" + ssrc);
// Find the format of the ReceiveStream
DataSource dataSource = receiveStream.getDataSource();
if (dataSource instanceof PushBufferDataSource) {
Format format = null;
PushBufferDataSource pbds = (PushBufferDataSource) dataSource;
for (PushBufferStream pbs : pbds.getStreams()) {
if ((format = pbs.getFormat()) != null) break;
}
if (format == null) {
logger.error("Failed to handle new ReceiveStream: " + "Failed to determine format");
return;
}
receiveStreamDesc.format = format;
} else {
logger.error("Failed to handle new ReceiveStream: " + "Unsupported DataSource");
return;
}
int rtpClockRate = -1;
if (receiveStreamDesc.format instanceof AudioFormat)
rtpClockRate = (int) ((AudioFormat) receiveStreamDesc.format).getSampleRate();
else if (receiveStreamDesc.format instanceof VideoFormat) rtpClockRate = 90000;
getSynchronizer().setRtpClockRate(ssrc, rtpClockRate);
// create a Processor and configure it
Processor processor = null;
try {
processor = Manager.createProcessor(receiveStream.getDataSource());
} catch (NoProcessorException npe) {
logger.error("Failed to create Processor: ", npe);
return;
} catch (IOException ioe) {
logger.error("Failed to create Processor: ", ioe);
return;
}
if (logger.isInfoEnabled()) logger.info("Created processor for SSRC=" + ssrc);
processor.addControllerListener(this);
receiveStreamDesc.processor = processor;
final int streamCount;
synchronized (receiveStreams) {
receiveStreams.add(receiveStreamDesc);
streamCount = receiveStreams.size();
}
/*
* XXX TODO IRBABOON
* This is a terrible hack which works around a failure to realize()
* some of the Processor-s for audio streams, when multiple streams
* start nearly simultaneously. The cause of the problem is currently
* unknown (and synchronizing all FMJ calls in RecorderRtpImpl
* does not help).
* XXX TODO NOOBABRI
*/
if (receiveStreamDesc.format instanceof AudioFormat) {
final Processor p = processor;
new Thread() {
@Override
public void run() {
// delay configuring the processors for the different
// audio streams to decrease the probability that they
// run together.
try {
int ms = 450 * (streamCount - 1);
logger.warn(
"Sleeping for "
+ ms
+ "ms before"
+ " configuring processor for SSRC="
+ ssrc
+ " "
+ System.currentTimeMillis());
Thread.sleep(ms);
} catch (Exception e) {
}
p.configure();
}
}.run();
} else {
processor.configure();
}
} else if (event instanceof TimeoutEvent) {
if (receiveStream == null) {
// TODO: we might want to get the list of ReceiveStream-s from
// rtpManager and compare it to our list, to see if we should
// remove a stream.
logger.warn("TimeoutEvent: null.");
return;
}
// FMJ silently creates new ReceiveStream instances, so we have to
// recognize them by the SSRC.
ReceiveStreamDesc receiveStreamDesc = findReceiveStream(getReceiveStreamSSRC(receiveStream));
if (receiveStreamDesc != null) {
if (logger.isInfoEnabled()) {
logger.info("ReceiveStream timeout, ssrc=" + receiveStreamDesc.ssrc);
}
removeReceiveStream(receiveStreamDesc, true);
}
} else if (event != null && logger.isInfoEnabled()) {
logger.info("Unhandled ReceiveStreamEvent (" + event.getClass().getName() + "): " + event);
}
}
/**
* Intercepts and handles outgoing RTX (RFC-4588) packets for an <tt>RtpChannel</tt>. Depending on
* whether the destination supports the RTX format (RFC-4588) either removes the RTX encapsulation
* (thus effectively retransmitting packets bit-by-bit) or updates the sequence number and SSRC
* fields taking into account the data sent to the particular <tt>RtpChannel</tt>.
*
* @author Boris Grozev
*/
public class RtxTransformer extends SinglePacketTransformerAdapter implements TransformEngine {
/**
* The <tt>Logger</tt> used by the <tt>RtxTransformer</tt> class and its instances to print debug
* information.
*/
private static final Logger logger = Logger.getLogger(RtxTransformer.class);
/** The <tt>RtpChannel</tt> for the transformer. */
private RtpChannel channel;
/** Maps an RTX SSRC to the last RTP sequence number sent with that SSRC. */
private final Map<Long, Integer> rtxSequenceNumbers = new HashMap<>();
/**
* Initializes a new <tt>RtxTransformer</tt> with a specific <tt>RtpChannel</tt>.
*
* @param channel the <tt>RtpChannel</tt> for the transformer.
*/
RtxTransformer(RtpChannel channel) {
this.channel = channel;
}
/** Implements {@link PacketTransformer#transform(RawPacket[])}. {@inheritDoc} */
@Override
public RawPacket transform(RawPacket pkt) {
byte rtxPt;
if (pkt != null
&& (rtxPt = channel.getRtxPayloadType()) != -1
&& pkt.getPayloadType() == rtxPt) {
pkt = handleRtxPacket(pkt);
}
return pkt;
}
/**
* Handles an RTX packet and returns it.
*
* @param pkt the packet to handle.
* @return the packet
*/
private RawPacket handleRtxPacket(RawPacket pkt) {
boolean destinationSupportsRtx = channel.getRtxPayloadType() != -1;
RawPacket mediaPacket = createMediaPacket(pkt);
if (mediaPacket != null) {
RawPacketCache cache = channel.getStream().getPacketCache();
if (cache != null) {
cache.cachePacket(mediaPacket);
}
}
if (destinationSupportsRtx) {
pkt.setSequenceNumber(
getNextRtxSequenceNumber(pkt.getSSRC() & 0xffffffffL, pkt.getSequenceNumber()));
} else {
// If the media packet was not reconstructed, drop the RTX packet
// (by returning null).
return mediaPacket;
}
return pkt;
}
/**
* Creates a {@code RawPacket} which represents the original packet encapsulated in {@code pkt}
* using the RTX format.
*
* @param pkt the packet from which to extract a media packet.
* @return the extracted media packet.
*/
private RawPacket createMediaPacket(RawPacket pkt) {
RawPacket mediaPacket = null;
long rtxSsrc = pkt.getSSRC() & 0xffffffffL;
// We need to know the SSRC paired with rtxSsrc *as seen by the
// receiver (i.e. this.channel)*. However, we only store SSRCs
// that endpoints *send* with.
// We therefore assume that SSRC re-writing has not introduced any
// new SSRCs and therefor the FID mappings known to the senders
// also apply to receivers.
RtpChannel sourceChannel = channel.getContent().findChannelByFidSsrc(rtxSsrc);
if (sourceChannel != null) {
long mediaSsrc = sourceChannel.getFidPairedSsrc(rtxSsrc);
if (mediaSsrc != -1) {
byte apt = sourceChannel.getRtxAssociatedPayloadType();
if (apt != -1) {
mediaPacket = new RawPacket(pkt.getBuffer().clone(), pkt.getOffset(), pkt.getLength());
// Remove the RTX header by moving the RTP header two bytes
// right.
byte[] buf = mediaPacket.getBuffer();
int off = mediaPacket.getOffset();
System.arraycopy(buf, off, buf, off + 2, mediaPacket.getHeaderLength());
mediaPacket.setOffset(off + 2);
mediaPacket.setLength(pkt.getLength() - 2);
mediaPacket.setSSRC((int) mediaSsrc);
mediaPacket.setSequenceNumber(pkt.getOriginalSequenceNumber());
mediaPacket.setPayloadType(apt);
}
}
}
return mediaPacket;
}
/** Implements {@link TransformEngine#getRTPTransformer()}. */
@Override
public PacketTransformer getRTPTransformer() {
return this;
}
/** Implements {@link TransformEngine#getRTCPTransformer()}. */
@Override
public PacketTransformer getRTCPTransformer() {
return null;
}
/**
* Returns the sequence number to use for a specific RTX packet, which is based on the packet's
* original sequence number.
*
* <p>Because we terminate the RTX format, and with simulcast we might translate RTX packets from
* multiple SSRCs into the same SSRC, we keep count of the RTX packets (and their sequence
* numbers) which we sent for each SSRC.
*
* @param ssrc the SSRC of the RTX stream for the packet.
* @param defaultSeq the default sequence number to use in case we don't (yet) have any
* information about <tt>ssrc</tt>.
* @return the sequence number which should be used for the next RTX packet sent using SSRC
* <tt>ssrc</tt>.
*/
private int getNextRtxSequenceNumber(long ssrc, int defaultSeq) {
Integer seq;
synchronized (rtxSequenceNumbers) {
seq = rtxSequenceNumbers.get(ssrc);
if (seq == null) seq = defaultSeq;
else seq++;
rtxSequenceNumbers.put(ssrc, seq);
}
return seq;
}
/**
* Returns the next RTP sequence number to use for the RTX stream for a particular SSRC.
*
* @param ssrc the SSRC.
* @return the next sequence number to use for SSRC <tt>ssrc</tt>.
*/
private int getNextRtxSequenceNumber(long ssrc) {
return getNextRtxSequenceNumber(ssrc, new Random().nextInt(1 << 16));
}
/**
* Tries to find an SSRC paired with {@code ssrc} in an FID group in one of the channels from
* {@link #channel}'s {@code Content}. Returns -1 on failure.
*
* @param ssrc the SSRC for which to find a paired SSRC.
* @return An SSRC paired with {@code ssrc} in an FID group, or -1.
*/
private long getPairedSsrc(long ssrc) {
RtpChannel sourceChannel = channel.getContent().findChannelByFidSsrc(ssrc);
if (sourceChannel != null) {
return sourceChannel.getFidPairedSsrc(ssrc);
}
return -1;
}
/**
* Retransmits a packet to {@link #channel}. If the destination supports the RTX format, the
* packet will be encapsulated in RTX, otherwise, the packet will be retransmitted as-is.
*
* @param pkt the packet to retransmit.
* @param after the {@code TransformEngine} in the chain of {@code TransformEngine}s of the
* associated {@code MediaStream} after which the injection of {@code pkt} is to begin
* @return {@code true} if the packet was successfully retransmitted, {@code false} otherwise.
*/
public boolean retransmit(RawPacket pkt, TransformEngine after) {
boolean destinationSupportsRtx = channel.getRtxPayloadType() != -1;
boolean retransmitPlain;
if (destinationSupportsRtx) {
long rtxSsrc = getPairedSsrc(pkt.getSSRC());
if (rtxSsrc == -1) {
logger.warn("Cannot find SSRC for RTX, retransmitting plain.");
retransmitPlain = true;
} else {
retransmitPlain = !encapsulateInRtxAndTransmit(pkt, rtxSsrc);
}
} else {
retransmitPlain = true;
}
if (retransmitPlain) {
MediaStream mediaStream = channel.getStream();
if (mediaStream != null) {
try {
mediaStream.injectPacket(pkt, /* data */ true, after);
} catch (TransmissionFailedException tfe) {
logger.warn("Failed to retransmit a packet.");
return false;
}
}
}
return true;
}
/**
* Encapsulates {@code pkt} in the RTX format, using {@code rtxSsrc} as its SSRC, and transmits it
* to {@link #channel} by injecting it in the {@code MediaStream}.
*
* @param pkt the packet to transmit.
* @param rtxSsrc the SSRC for the RTX stream.
* @return {@code true} if the packet was successfully retransmitted, {@code false} otherwise.
*/
private boolean encapsulateInRtxAndTransmit(RawPacket pkt, long rtxSsrc) {
byte[] buf = pkt.getBuffer();
int len = pkt.getLength();
int off = pkt.getOffset();
byte[] newBuf = buf;
if (buf.length < len + 2) {
// FIXME The byte array newly allocated and assigned to newBuf must
// be made known to pkt eventually.
newBuf = new byte[len + 2];
}
int osn = pkt.getSequenceNumber();
int headerLength = pkt.getHeaderLength();
int payloadLength = len - headerLength;
System.arraycopy(buf, off, newBuf, 0, headerLength);
// FIXME If newBuf is actually buf, then we will override the first two
// bytes of the payload bellow.
newBuf[headerLength] = (byte) ((osn >> 8) & 0xff);
newBuf[headerLength + 1] = (byte) (osn & 0xff);
System.arraycopy(buf, off + headerLength, newBuf, headerLength + 2, payloadLength);
// FIXME We tried to extend the payload of pkt by two bytes above but
// we never told pkt that its length has increased by these two bytes.
MediaStream mediaStream = channel.getStream();
if (mediaStream != null) {
pkt.setSSRC((int) rtxSsrc);
// Only call getNextRtxSequenceNumber() when we're sure we're going
// to transmit a packet, because it consumes a sequence number.
pkt.setSequenceNumber(getNextRtxSequenceNumber(rtxSsrc));
try {
mediaStream.injectPacket(pkt, /* data */ true, /* after */ null);
} catch (TransmissionFailedException tfe) {
logger.warn("Failed to transmit an RTX packet.");
return false;
}
}
return true;
}
}
/**
* A <tt>Recorder</tt> implementation which attaches to an <tt>RTPTranslator</tt>.
*
* @author Vladimir Marinov
* @author Boris Grozev
*/
public class RecorderRtpImpl
implements Recorder, ReceiveStreamListener, ActiveSpeakerChangedListener, ControllerListener {
/**
* The <tt>Logger</tt> used by the <tt>RecorderRtpImpl</tt> class and its instances for logging
* output.
*/
private static final Logger logger = Logger.getLogger(RecorderRtpImpl.class);
// values hard-coded to match chrome
// TODO: allow to set them dynamically
private static final byte redPayloadType = 116;
private static final byte ulpfecPayloadType = 117;
private static final byte vp8PayloadType = 100;
private static final byte opusPayloadType = 111;
private static final Format redFormat = new VideoFormat(Constants.RED);
private static final Format ulpfecFormat = new VideoFormat(Constants.ULPFEC);
private static final Format vp8RtpFormat = new VideoFormat(Constants.VP8_RTP);
private static final Format vp8Format = new VideoFormat(Constants.VP8);
private static final Format opusFormat =
new AudioFormat(Constants.OPUS_RTP, 48000, Format.NOT_SPECIFIED, Format.NOT_SPECIFIED);
private static final int FMJ_VIDEO_JITTER_BUFFER_MIN_SIZE = 300;
/** The <tt>ContentDescriptor</tt> to use when saving audio. */
private static final ContentDescriptor AUDIO_CONTENT_DESCRIPTOR =
new ContentDescriptor(FileTypeDescriptor.MPEG_AUDIO);
/** The suffix for audio file names. */
private static final String AUDIO_FILENAME_SUFFIX = ".mp3";
/** The suffix for video file names. */
private static final String VIDEO_FILENAME_SUFFIX = ".webm";
static {
Registry.set("video_jitter_buffer_MIN_SIZE", FMJ_VIDEO_JITTER_BUFFER_MIN_SIZE);
}
/** The <tt>RTPTranslator</tt> that this recorder is/will be attached to. */
private RTPTranslatorImpl translator;
/**
* The custom <tt>RTPConnector</tt> that this instance uses to read from {@link #translator} and
* write to {@link #rtpManager}.
*/
private RTPConnectorImpl rtpConnector;
/** Path to the directory where the output files will be stored. */
private String path;
/** The <tt>RTCPFeedbackMessageSender</tt> that we use to send RTCP FIR messages. */
private RTCPFeedbackMessageSender rtcpFeedbackSender;
/**
* The {@link RTPManager} instance we use to handle the packets coming from
* <tt>RTPTranslator</tt>.
*/
private RTPManager rtpManager;
/**
* The instance which should be notified when events related to recordings (such as the start or
* end of a recording) occur.
*/
private RecorderEventHandlerImpl eventHandler;
/**
* Holds the <tt>ReceiveStreams</tt> added to this instance by {@link #rtpManager} and additional
* information associated with each one (e.g. the <tt>Processor</tt>, if any, used for it).
*/
private final HashSet<ReceiveStreamDesc> receiveStreams = new HashSet<ReceiveStreamDesc>();
private final Set<Long> activeVideoSsrcs = new HashSet<Long>();
/**
* The <tt>ActiveSpeakerDetector</tt> which will listen to the audio receive streams of this
* <tt>RecorderRtpImpl</tt> and notify it about changes to the active speaker via calls to {@link
* #activeSpeakerChanged(long)}
*/
private ActiveSpeakerDetector activeSpeakerDetector = null;
StreamRTPManager streamRTPManager;
private SynchronizerImpl synchronizer;
private boolean started = false;
/**
* Constructor.
*
* @param translator the <tt>RTPTranslator</tt> to which this instance will attach in order to
* record media.
*/
public RecorderRtpImpl(RTPTranslator translator) {
this.translator = (RTPTranslatorImpl) translator;
activeSpeakerDetector = new ActiveSpeakerDetectorImpl();
activeSpeakerDetector.addActiveSpeakerChangedListener(this);
}
/** Implements {@link Recorder#addListener(Recorder.Listener)}. */
@Override
public void addListener(Listener listener) {}
/** Implements {@link Recorder#removeListener(Recorder.Listener)}. */
@Override
public void removeListener(Listener listener) {}
/** Implements {@link Recorder#getSupportedFormats()}. */
@Override
public List<String> getSupportedFormats() {
return null;
}
/** Implements {@link Recorder#setMute(boolean)}. */
@Override
public void setMute(boolean mute) {}
/**
* Implements {@link Recorder#getFilename()}. Returns null, since we don't have a (single)
* associated filename.
*/
@Override
public String getFilename() {
return null;
}
/**
* Sets the instance which should be notified when events related to recordings (such as the start
* or end of a recording) occur.
*/
public void setEventHandler(RecorderEventHandler eventHandler) {
if (this.eventHandler == null
|| (this.eventHandler != eventHandler && this.eventHandler.handler != eventHandler)) {
if (this.eventHandler == null) this.eventHandler = new RecorderEventHandlerImpl(eventHandler);
else this.eventHandler.handler = eventHandler;
}
}
/**
* {@inheritDoc}
*
* @param format unused, since this implementation records multiple streams using potentially
* different formats.
* @param dirname the path to the directory into which this <tt>Recorder</tt> will store the
* recorded media files.
*/
@Override
public void start(String format, String dirname) throws IOException, MediaException {
if (logger.isInfoEnabled()) logger.info("Starting, format=" + format + " " + hashCode());
path = dirname;
MediaService mediaService = LibJitsi.getMediaService();
/*
* Note that we use only one RTPConnector for both the RTPTranslator
* and the RTPManager instances. The this.translator will write to its
* output streams, and this.rtpManager will read from its input streams.
*/
rtpConnector = new RTPConnectorImpl(redPayloadType, ulpfecPayloadType);
rtpManager = RTPManager.newInstance();
/*
* Add the formats that we know about.
*/
rtpManager.addFormat(vp8RtpFormat, vp8PayloadType);
rtpManager.addFormat(opusFormat, opusPayloadType);
rtpManager.addReceiveStreamListener(this);
/*
* Note: When this.rtpManager sends RTCP sender/receiver reports, they
* will end up being written to its own input stream. This is not
* expected to cause problems, but might be something to keep an eye on.
*/
rtpManager.initialize(rtpConnector);
/*
* Register a fake call participant.
* TODO: can we use a more generic MediaStream here?
*/
streamRTPManager =
new StreamRTPManager(
mediaService.createMediaStream(
new MediaDeviceImpl(new CaptureDeviceInfo(), MediaType.VIDEO)),
translator);
streamRTPManager.initialize(rtpConnector);
rtcpFeedbackSender = translator.getRtcpFeedbackMessageSender();
translator.addFormat(streamRTPManager, opusFormat, opusPayloadType);
// ((RTPTranslatorImpl)videoRTPTranslator).addFormat(streamRTPManager, redFormat,
// redPayloadType);
// ((RTPTranslatorImpl)videoRTPTranslator).addFormat(streamRTPManager, ulpfecFormat,
// ulpfecPayloadType);
// ((RTPTranslatorImpl)videoRTPTranslator).addFormat(streamRTPManager,
// mediaFormatImpl.getFormat(), vp8PayloadType);
started = true;
}
@Override
public void stop() {
if (started) {
if (logger.isInfoEnabled()) logger.info("Stopping " + hashCode());
// remove the recorder from the translator (e.g. stop new packets from
// being written to rtpConnector
if (streamRTPManager != null) streamRTPManager.dispose();
HashSet<ReceiveStreamDesc> streamsToRemove = new HashSet<ReceiveStreamDesc>();
synchronized (receiveStreams) {
streamsToRemove.addAll(receiveStreams);
}
for (ReceiveStreamDesc r : streamsToRemove) removeReceiveStream(r, false);
rtpConnector.rtcpPacketTransformer.close();
rtpConnector.rtpPacketTransformer.close();
rtpManager.dispose();
started = false;
}
}
/**
* Implements {@link ReceiveStreamListener#update(ReceiveStreamEvent)}.
*
* <p>{@link #rtpManager} will use this to notify us of <tt>ReceiveStreamEvent</tt>s.
*/
@Override
public void update(ReceiveStreamEvent event) {
if (event == null) return;
ReceiveStream receiveStream = event.getReceiveStream();
if (event instanceof NewReceiveStreamEvent) {
if (receiveStream == null) {
logger.warn("NewReceiveStreamEvent: null");
return;
}
final long ssrc = getReceiveStreamSSRC(receiveStream);
ReceiveStreamDesc receiveStreamDesc = findReceiveStream(ssrc);
if (receiveStreamDesc != null) {
String s = "NewReceiveStreamEvent for an existing SSRC. ";
if (receiveStream != receiveStreamDesc.receiveStream)
s += "(but different ReceiveStream object)";
logger.warn(s);
return;
} else receiveStreamDesc = new ReceiveStreamDesc(receiveStream);
if (logger.isInfoEnabled()) logger.info("New ReceiveStream, ssrc=" + ssrc);
// Find the format of the ReceiveStream
DataSource dataSource = receiveStream.getDataSource();
if (dataSource instanceof PushBufferDataSource) {
Format format = null;
PushBufferDataSource pbds = (PushBufferDataSource) dataSource;
for (PushBufferStream pbs : pbds.getStreams()) {
if ((format = pbs.getFormat()) != null) break;
}
if (format == null) {
logger.error("Failed to handle new ReceiveStream: " + "Failed to determine format");
return;
}
receiveStreamDesc.format = format;
} else {
logger.error("Failed to handle new ReceiveStream: " + "Unsupported DataSource");
return;
}
int rtpClockRate = -1;
if (receiveStreamDesc.format instanceof AudioFormat)
rtpClockRate = (int) ((AudioFormat) receiveStreamDesc.format).getSampleRate();
else if (receiveStreamDesc.format instanceof VideoFormat) rtpClockRate = 90000;
getSynchronizer().setRtpClockRate(ssrc, rtpClockRate);
// create a Processor and configure it
Processor processor = null;
try {
processor = Manager.createProcessor(receiveStream.getDataSource());
} catch (NoProcessorException npe) {
logger.error("Failed to create Processor: ", npe);
return;
} catch (IOException ioe) {
logger.error("Failed to create Processor: ", ioe);
return;
}
if (logger.isInfoEnabled()) logger.info("Created processor for SSRC=" + ssrc);
processor.addControllerListener(this);
receiveStreamDesc.processor = processor;
final int streamCount;
synchronized (receiveStreams) {
receiveStreams.add(receiveStreamDesc);
streamCount = receiveStreams.size();
}
/*
* XXX TODO IRBABOON
* This is a terrible hack which works around a failure to realize()
* some of the Processor-s for audio streams, when multiple streams
* start nearly simultaneously. The cause of the problem is currently
* unknown (and synchronizing all FMJ calls in RecorderRtpImpl
* does not help).
* XXX TODO NOOBABRI
*/
if (receiveStreamDesc.format instanceof AudioFormat) {
final Processor p = processor;
new Thread() {
@Override
public void run() {
// delay configuring the processors for the different
// audio streams to decrease the probability that they
// run together.
try {
int ms = 450 * (streamCount - 1);
logger.warn(
"Sleeping for "
+ ms
+ "ms before"
+ " configuring processor for SSRC="
+ ssrc
+ " "
+ System.currentTimeMillis());
Thread.sleep(ms);
} catch (Exception e) {
}
p.configure();
}
}.run();
} else {
processor.configure();
}
} else if (event instanceof TimeoutEvent) {
if (receiveStream == null) {
// TODO: we might want to get the list of ReceiveStream-s from
// rtpManager and compare it to our list, to see if we should
// remove a stream.
logger.warn("TimeoutEvent: null.");
return;
}
// FMJ silently creates new ReceiveStream instances, so we have to
// recognize them by the SSRC.
ReceiveStreamDesc receiveStreamDesc = findReceiveStream(getReceiveStreamSSRC(receiveStream));
if (receiveStreamDesc != null) {
if (logger.isInfoEnabled()) {
logger.info("ReceiveStream timeout, ssrc=" + receiveStreamDesc.ssrc);
}
removeReceiveStream(receiveStreamDesc, true);
}
} else if (event != null && logger.isInfoEnabled()) {
logger.info("Unhandled ReceiveStreamEvent (" + event.getClass().getName() + "): " + event);
}
}
private void removeReceiveStream(ReceiveStreamDesc receiveStream, boolean emptyJB) {
if (receiveStream.format instanceof VideoFormat) {
rtpConnector.packetBuffer.disable(receiveStream.ssrc);
emptyPacketBuffer(receiveStream.ssrc);
}
if (receiveStream.dataSink != null) {
try {
receiveStream.dataSink.stop();
} catch (IOException e) {
logger.error("Failed to stop DataSink " + e);
}
receiveStream.dataSink.close();
}
if (receiveStream.processor != null) {
receiveStream.processor.stop();
receiveStream.processor.close();
}
DataSource dataSource = receiveStream.receiveStream.getDataSource();
if (dataSource != null) {
try {
dataSource.stop();
} catch (IOException ioe) {
logger.warn("Failed to stop DataSource");
}
dataSource.disconnect();
}
synchronized (receiveStreams) {
receiveStreams.remove(receiveStream);
}
}
/**
* 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);
}
}
/**
* Restarts the recording for a specific SSRC.
*
* @param ssrc the SSRC for which to restart recording. RTP packet of the new recording).
*/
private void resetRecording(long ssrc, long timestamp) {
ReceiveStreamDesc receiveStream = findReceiveStream(ssrc);
// we only restart audio recordings
if (receiveStream != null && receiveStream.format instanceof AudioFormat) {
String newFilename = getNextFilename(path + "/" + ssrc, AUDIO_FILENAME_SUFFIX);
// flush the buffer contained in the MP3 encoder
String s = "trying to flush ssrc=" + ssrc;
Processor p = receiveStream.processor;
if (p != null) {
s += " p!=null";
for (TrackControl tc : p.getTrackControls()) {
Object o = tc.getControl(FlushableControl.class.getName());
if (o != null) ((FlushableControl) o).flush();
}
}
if (logger.isInfoEnabled()) {
logger.info("Restarting recording for SSRC=" + ssrc + ". New filename: " + newFilename);
}
receiveStream.dataSink.close();
receiveStream.dataSink = null;
// flush the FMJ jitter buffer
// DataSource ds = receiveStream.receiveStream.getDataSource();
// if (ds instanceof net.sf.fmj.media.protocol.rtp.DataSource)
// ((net.sf.fmj.media.protocol.rtp.DataSource)ds).flush();
receiveStream.filename = newFilename;
try {
receiveStream.dataSink =
Manager.createDataSink(
receiveStream.dataSource, new MediaLocator("file:" + newFilename));
} catch (NoDataSinkException ndse) {
logger.warn("Could not reset recording for SSRC=" + ssrc + ": " + ndse);
removeReceiveStream(receiveStream, false);
}
try {
receiveStream.dataSink.open();
receiveStream.dataSink.start();
} catch (IOException ioe) {
logger.warn("Could not reset recording for SSRC=" + ssrc + ": " + ioe);
removeReceiveStream(receiveStream, false);
}
audioRecordingStarted(ssrc, timestamp);
}
}
private void audioRecordingStarted(long ssrc, long timestamp) {
ReceiveStreamDesc desc = findReceiveStream(ssrc);
if (desc == null) return;
RecorderEvent event = new RecorderEvent();
event.setType(RecorderEvent.Type.RECORDING_STARTED);
event.setMediaType(MediaType.AUDIO);
event.setSsrc(ssrc);
event.setRtpTimestamp(timestamp);
event.setFilename(desc.filename);
if (eventHandler != null) eventHandler.handleEvent(event);
}
/**
* Handles a request from a specific <tt>DataSink</tt> to request a keyframe by sending an RTCP
* feedback FIR message to the media source.
*
* @param dataSink the <tt>DataSink</tt> which requests that a keyframe be requested with a FIR
* message.
* @return <tt>true</tt> if a keyframe was successfully requested, <tt>false</tt> otherwise
*/
private boolean requestFIR(WebmDataSink dataSink) {
ReceiveStreamDesc desc = findReceiveStream(dataSink);
if (desc != null && rtcpFeedbackSender != null) {
return rtcpFeedbackSender.sendFIR((int) desc.ssrc);
}
return false;
}
/**
* Returns "prefix"+"suffix" if the file with this name does not exist. Otherwise, returns the
* first inexistant filename of the form "prefix-"+i+"suffix", for an integer i. i is bounded by
* 100 to prevent hanging, and on failure to find an inexistant filename the method will return
* null.
*
* @param prefix
* @param suffix
* @return
*/
private String getNextFilename(String prefix, String suffix) {
if (!new File(prefix + suffix).exists()) return prefix + suffix;
int i = 1;
String s;
do {
s = prefix + "-" + i + suffix;
if (!new File(s).exists()) return s;
i++;
} while (i < 1000); // don't hang indefinitely...
return null;
}
/**
* Finds the <tt>ReceiveStreamDesc</tt> with a particular <tt>Processor</tt>
*
* @param processor The <tt>Processor</tt> to match.
* @return the <tt>ReceiveStreamDesc</tt> with a particular <tt>Processor</tt>, or <tt>null</tt>.
*/
private ReceiveStreamDesc findReceiveStream(Processor processor) {
if (processor == null) return null;
synchronized (receiveStreams) {
for (ReceiveStreamDesc r : receiveStreams) if (processor.equals(r.processor)) return r;
}
return null;
}
/**
* Finds the <tt>ReceiveStreamDesc</tt> with a particular <tt>DataSink</tt>
*
* @param dataSink The <tt>DataSink</tt> to match.
* @return the <tt>ReceiveStreamDesc</tt> with a particular <tt>DataSink</tt>, or <tt>null</tt>.
*/
private ReceiveStreamDesc findReceiveStream(DataSink dataSink) {
if (dataSink == null) return null;
synchronized (receiveStreams) {
for (ReceiveStreamDesc r : receiveStreams) if (dataSink.equals(r.dataSink)) return r;
}
return null;
}
/**
* Finds the <tt>ReceiveStreamDesc</tt> with a particular SSRC.
*
* @param ssrc The SSRC to match.
* @return the <tt>ReceiveStreamDesc</tt> with a particular SSRC, or <tt>null</tt>.
*/
private ReceiveStreamDesc findReceiveStream(long ssrc) {
synchronized (receiveStreams) {
for (ReceiveStreamDesc r : receiveStreams) if (ssrc == r.ssrc) return r;
}
return null;
}
/**
* Gets the SSRC of a <tt>ReceiveStream</tt> as a (non-negative) <tt>long</tt>.
*
* <p>FMJ stores the 32-bit SSRC values in <tt>int</tt>s, and the <tt>ReceiveStream.getSSRC()</tt>
* implementation(s) don't take care of converting the negative <tt>int</tt> values sometimes
* resulting from reading of a 32-bit field into the correct unsigned <tt>long</tt> value. So do
* the conversion here.
*
* @param receiveStream the <tt>ReceiveStream</tt> for which to get the SSRC.
* @return the SSRC of <tt>receiveStream</tt> an a (non-negative) <tt>long</tt>.
*/
private long getReceiveStreamSSRC(ReceiveStream receiveStream) {
return 0xffffffffL & receiveStream.getSSRC();
}
/**
* 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);
}
}
private void handleRtpPacket(RawPacket pkt) {
if (pkt != null && pkt.getPayloadType() == vp8PayloadType) {
int ssrc = pkt.getSSRC();
if (!activeVideoSsrcs.contains(ssrc & 0xffffffffL)) {
synchronized (activeVideoSsrcs) {
if (!activeVideoSsrcs.contains(ssrc & 0xffffffffL)) {
activeVideoSsrcs.add(ssrc & 0xffffffffL);
rtcpFeedbackSender.sendFIR(ssrc);
}
}
}
}
}
private void handleRtcpPacket(RawPacket pkt) {
getSynchronizer().addRTCPPacket(pkt);
eventHandler.nudge();
}
public SynchronizerImpl getSynchronizer() {
if (synchronizer == null) synchronizer = new SynchronizerImpl();
return synchronizer;
}
public void setSynchronizer(Synchronizer synchronizer) {
if (synchronizer instanceof SynchronizerImpl) {
this.synchronizer = (SynchronizerImpl) synchronizer;
}
}
public void connect(Recorder recorder) {
if (!(recorder instanceof RecorderRtpImpl)) return;
((RecorderRtpImpl) recorder).setSynchronizer(getSynchronizer());
}
private void emptyPacketBuffer(long ssrc) {
RawPacket[] pkts = rtpConnector.packetBuffer.emptyBuffer(ssrc);
RTPConnectorImpl.OutputDataStreamImpl dataStream;
try {
dataStream = rtpConnector.getDataOutputStream();
} catch (IOException ioe) {
logger.error("Failed to empty packet buffer for SSRC=" + ssrc + ": " + ioe);
return;
}
for (RawPacket pkt : pkts)
dataStream.write(
pkt.getBuffer(), pkt.getOffset(), pkt.getLength(), false /* already transformed */);
}
/** The <tt>RTPConnector</tt> implementation used by this <tt>RecorderRtpImpl</tt>. */
private class RTPConnectorImpl implements RTPConnector {
private PushSourceStreamImpl controlInputStream;
private OutputDataStreamImpl controlOutputStream;
private PushSourceStreamImpl dataInputStream;
private OutputDataStreamImpl dataOutputStream;
private SourceTransferHandler dataTransferHandler;
private SourceTransferHandler controlTransferHandler;
private RawPacket pendingDataPacket = new RawPacket();
private RawPacket pendingControlPacket = new RawPacket();
private PacketTransformer rtpPacketTransformer = null;
private PacketTransformer rtcpPacketTransformer = null;
/** The PacketBuffer instance which we use as a jitter buffer. */
private PacketBuffer packetBuffer;
private RTPConnectorImpl(byte redPT, byte ulpfecPT) {
packetBuffer = new PacketBuffer();
// The chain of transformers will be applied in reverse order for
// incoming packets.
TransformEngine transformEngine =
new TransformEngineChain(
new TransformEngine[] {
packetBuffer,
new TransformEngineImpl(),
new CompoundPacketEngine(),
new FECTransformEngine(ulpfecPT, (byte) -1),
new REDTransformEngine(redPT, (byte) -1)
});
rtpPacketTransformer = transformEngine.getRTPTransformer();
rtcpPacketTransformer = transformEngine.getRTCPTransformer();
}
private RTPConnectorImpl() {}
@Override
public void close() {
try {
if (dataOutputStream != null) dataOutputStream.close();
if (controlOutputStream != null) controlOutputStream.close();
} catch (IOException ioe) {
throw new UndeclaredThrowableException(ioe);
}
}
@Override
public PushSourceStream getControlInputStream() throws IOException {
if (controlInputStream == null) {
controlInputStream = new PushSourceStreamImpl(true);
}
return controlInputStream;
}
@Override
public OutputDataStream getControlOutputStream() throws IOException {
if (controlOutputStream == null) {
controlOutputStream = new OutputDataStreamImpl(true);
}
return controlOutputStream;
}
@Override
public PushSourceStream getDataInputStream() throws IOException {
if (dataInputStream == null) {
dataInputStream = new PushSourceStreamImpl(false);
}
return dataInputStream;
}
@Override
public OutputDataStreamImpl getDataOutputStream() throws IOException {
if (dataOutputStream == null) {
dataOutputStream = new OutputDataStreamImpl(false);
}
return dataOutputStream;
}
@Override
public double getRTCPBandwidthFraction() {
return -1;
}
@Override
public double getRTCPSenderBandwidthFraction() {
return -1;
}
@Override
public int getReceiveBufferSize() {
// TODO Auto-generated method stub
return 0;
}
@Override
public int getSendBufferSize() {
// TODO Auto-generated method stub
return 0;
}
@Override
public void setReceiveBufferSize(int arg0) throws IOException {
// TODO Auto-generated method stub
}
@Override
public void setSendBufferSize(int arg0) throws IOException {
// TODO Auto-generated method stub
}
private class OutputDataStreamImpl implements OutputDataStream {
boolean isControlStream;
private RawPacket[] rawPacketArray = new RawPacket[1];
public OutputDataStreamImpl(boolean isControlStream) {
this.isControlStream = isControlStream;
}
public int write(byte[] buffer, int offset, int length) {
return write(buffer, offset, length, true);
}
public int write(byte[] buffer, int offset, int length, boolean transform) {
RawPacket pkt = rawPacketArray[0];
if (pkt == null) pkt = new RawPacket();
rawPacketArray[0] = pkt;
byte[] pktBuf = pkt.getBuffer();
if (pktBuf == null || pktBuf.length < length) {
pktBuf = new byte[length];
pkt.setBuffer(pktBuf);
}
System.arraycopy(buffer, offset, pktBuf, 0, length);
pkt.setOffset(0);
pkt.setLength(length);
if (transform) {
PacketTransformer packetTransformer =
isControlStream ? rtcpPacketTransformer : rtpPacketTransformer;
if (packetTransformer != null)
rawPacketArray = packetTransformer.reverseTransform(rawPacketArray);
}
SourceTransferHandler transferHandler;
PushSourceStream pushSourceStream;
try {
if (isControlStream) {
transferHandler = controlTransferHandler;
pushSourceStream = getControlInputStream();
} else {
transferHandler = dataTransferHandler;
pushSourceStream = getDataInputStream();
}
} catch (IOException ioe) {
throw new UndeclaredThrowableException(ioe);
}
for (int i = 0; i < rawPacketArray.length; i++) {
RawPacket packet = rawPacketArray[i];
// keep the first element for reuse
if (i != 0) rawPacketArray[i] = null;
if (packet != null) {
if (isControlStream) pendingControlPacket = packet;
else pendingDataPacket = packet;
if (transferHandler != null) {
transferHandler.transferData(pushSourceStream);
}
}
}
return length;
}
public void close() throws IOException {}
}
/**
* A dummy implementation of {@link PushSourceStream}.
*
* @author Vladimir Marinov
*/
private class PushSourceStreamImpl implements PushSourceStream {
private boolean isControlStream = false;
public PushSourceStreamImpl(boolean isControlStream) {
this.isControlStream = isControlStream;
}
/** Not implemented because there are currently no uses of the underlying functionality. */
@Override
public boolean endOfStream() {
return false;
}
/** Not implemented because there are currently no uses of the underlying functionality. */
@Override
public ContentDescriptor getContentDescriptor() {
return null;
}
/** Not implemented because there are currently no uses of the underlying functionality. */
@Override
public long getContentLength() {
return 0;
}
/** Not implemented because there are currently no uses of the underlying functionality. */
@Override
public Object getControl(String arg0) {
return null;
}
/** Not implemented because there are currently no uses of the underlying functionality. */
@Override
public Object[] getControls() {
return null;
}
/** Not implemented because there are currently no uses of the underlying functionality. */
@Override
public int getMinimumTransferSize() {
if (isControlStream) {
if (pendingControlPacket.getBuffer() != null) {
return pendingControlPacket.getLength();
}
} else {
if (pendingDataPacket.getBuffer() != null) {
return pendingDataPacket.getLength();
}
}
return 0;
}
@Override
public int read(byte[] buffer, int offset, int length) throws IOException {
RawPacket pendingPacket;
if (isControlStream) {
pendingPacket = pendingControlPacket;
} else {
pendingPacket = pendingDataPacket;
}
int bytesToRead = 0;
byte[] pendingPacketBuffer = pendingPacket.getBuffer();
if (pendingPacketBuffer != null) {
int pendingPacketLength = pendingPacket.getLength();
bytesToRead = length > pendingPacketLength ? pendingPacketLength : length;
System.arraycopy(
pendingPacketBuffer, pendingPacket.getOffset(), buffer, offset, bytesToRead);
}
return bytesToRead;
}
/**
* {@inheritDoc}
*
* <p>We keep the first non-null <tt>SourceTransferHandler</tt> that was set, because we don't
* want it to be overwritten when we initialize a second <tt>RTPManager</tt> with this
* <tt>RTPConnector</tt>.
*
* <p>See {@link RecorderRtpImpl#start(String, String)}
*/
@Override
public void setTransferHandler(SourceTransferHandler transferHandler) {
if (isControlStream) {
if (RTPConnectorImpl.this.controlTransferHandler == null) {
RTPConnectorImpl.this.controlTransferHandler = transferHandler;
}
} else {
if (RTPConnectorImpl.this.dataTransferHandler == null) {
RTPConnectorImpl.this.dataTransferHandler = transferHandler;
}
}
}
}
/**
* A transform engine implementation which allows <tt>RecorderRtpImpl</tt> to intercept RTP and
* RTCP packets in.
*/
private class TransformEngineImpl implements TransformEngine {
SinglePacketTransformer rtpTransformer =
new SinglePacketTransformer() {
@Override
public RawPacket transform(RawPacket pkt) {
return pkt;
}
@Override
public RawPacket reverseTransform(RawPacket pkt) {
RecorderRtpImpl.this.handleRtpPacket(pkt);
return pkt;
}
@Override
public void close() {}
};
SinglePacketTransformer rtcpTransformer =
new SinglePacketTransformer() {
@Override
public RawPacket transform(RawPacket pkt) {
return pkt;
}
@Override
public RawPacket reverseTransform(RawPacket pkt) {
RecorderRtpImpl.this.handleRtcpPacket(pkt);
if (pkt != null && pkt.getRTCPPayloadType() == 203) {
// An RTCP BYE packet. Remove the receive stream before
// it gets to FMJ, because we want to, for example,
// flush the packet buffer before that.
long ssrc = pkt.getRTCPSSRC() & 0xffffffffl;
if (logger.isInfoEnabled()) logger.info("RTCP BYE for SSRC=" + ssrc);
ReceiveStreamDesc receiveStream = findReceiveStream(ssrc);
if (receiveStream != null) removeReceiveStream(receiveStream, false);
}
return pkt;
}
@Override
public void close() {}
};
@Override
public PacketTransformer getRTPTransformer() {
return rtpTransformer;
}
@Override
public PacketTransformer getRTCPTransformer() {
return rtcpTransformer;
}
}
}
private class RecorderEventHandlerImpl implements RecorderEventHandler {
private RecorderEventHandler handler;
private final Set<RecorderEvent> pendingEvents = new HashSet<RecorderEvent>();
private RecorderEventHandlerImpl(RecorderEventHandler handler) {
this.handler = handler;
}
@Override
public boolean handleEvent(RecorderEvent ev) {
if (ev == null) return true;
if (RecorderEvent.Type.RECORDING_STARTED.equals(ev.getType())) {
long instant = getSynchronizer().getLocalTime(ev.getSsrc(), ev.getRtpTimestamp());
if (instant != -1) {
ev.setInstant(instant);
return handler.handleEvent(ev);
} else {
pendingEvents.add(ev);
return true;
}
}
return handler.handleEvent(ev);
}
private void nudge() {
for (Iterator<RecorderEvent> iter = pendingEvents.iterator(); iter.hasNext(); ) {
RecorderEvent ev = iter.next();
long instant = getSynchronizer().getLocalTime(ev.getSsrc(), ev.getRtpTimestamp());
if (instant != -1) {
iter.remove();
ev.setInstant(instant);
handler.handleEvent(ev);
}
}
}
@Override
public void close() {
for (RecorderEvent ev : pendingEvents) handler.handleEvent(ev);
}
}
/** Represents a <tt>ReceiveStream</tt> for the purposes of this <tt>RecorderRtpImpl</tt>. */
private class ReceiveStreamDesc {
/**
* The actual <tt>ReceiveStream</tt> which is represented by this <tt>ReceiveStreamDesc</tt>.
*/
private ReceiveStream receiveStream;
/** The SSRC of the stream. */
long ssrc;
/**
* The <tt>Processor</tt> used to transcode this receive stream into a format appropriate for
* saving to a file.
*/
private Processor processor;
/** The <tt>DataSink</tt> which saves the <tt>this.dataSource</tt> to a file. */
private DataSink dataSink;
/**
* The <tt>DataSource</tt> for this receive stream which is to be saved using a
* <tt>DataSink</tt> (i.e. the <tt>DataSource</tt> "after" all needed transcoding is done).
*/
private DataSource dataSource;
/** The name of the file into which this stream is being saved. */
private String filename;
/** The (original) format of this receive stream. */
private Format format;
/** The <tt>SilenceEffect</tt> used for this stream (for audio streams only). */
private SilenceEffect silenceEffect;
private ReceiveStreamDesc(ReceiveStream receiveStream) {
this.receiveStream = receiveStream;
this.ssrc = getReceiveStreamSSRC(receiveStream);
}
}
}
