/**
* 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);
}
}
}
/**
* 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;
}
}
}
}
}