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rtp.js
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// Generated by CoffeeScript 2.5.1
(function() {
// RTP spec:
// RFC 3550 http://tools.ietf.org/html/rfc3550
// RTP payload format for H.264 video:
// RFC 6184 http://tools.ietf.org/html/rfc6184
// RTP payload format for AAC audio:
// RFC 3640 http://tools.ietf.org/html/rfc3640
// RFC 5691 http://tools.ietf.org/html/rfc5691
// TODO: Use DON (decoding order number) to carry B-frames.
// DON is to RTP what DTS is to MPEG-TS.
var Bits,
EPOCH,
MAX_PAYLOAD_SIZE,
MAX_SEQUENCE_NUMBER,
NTP_SCALE_FRAC,
RTPParser,
RTP_HEADER_LEN,
aac,
api,
logger;
Bits = require("./bits");
aac = require("./aac");
logger = require("./logger");
// Number of seconds from 1900-01-01 to 1970-01-01
EPOCH = 2208988800;
// Constant for calculating NTP fractional second
NTP_SCALE_FRAC = 4294.967295;
// Minimum length of an RTP header
RTP_HEADER_LEN = 12;
MAX_PAYLOAD_SIZE = 1360;
MAX_SEQUENCE_NUMBER = 65535;
RTPParser = class RTPParser {
constructor() {
this.eventListeners = {};
this.packetBuffers = {};
this.fragmentedH264PacketBuffer = {};
this.h264NALUnitBuffer = {};
this.aacAccessUnitBuffer = {};
// config
this.unorderedPacketBufferSize = 10;
}
emit(name, ...data) {
var j, len, listener, ref;
if (this.eventListeners[name] != null) {
ref = this.eventListeners[name];
for (j = 0, len = ref.length; j < len; j++) {
listener = ref[j];
listener(...data);
}
}
}
on(name, listener) {
if (this.eventListeners[name] != null) {
return this.eventListeners[name].push(listener);
} else {
return (this.eventListeners[name] = [listener]);
}
}
feedUnorderedAACBuffer(buf, clientId, params) {
var packet;
packet = api.parseAACPacket(buf, params);
return this.feedUnorderedPacket(`aac:${clientId}`, packet);
}
feedUnorderedH264Buffer(buf, clientId) {
var packet;
packet = api.parseH264Packet(buf);
return this.feedUnorderedPacket(`h264:${clientId}`, packet);
}
clearAllUnorderedPacketBuffers() {
return (this.packetBuffers = {});
}
clearUnorderedPacketBuffer(tag) {
delete this.packetBuffers[`h264:${tag}`];
return delete this.packetBuffers[`aac:${tag}`];
}
feedUnorderedPacket(tag, packet) {
var buffers,
diff,
discardedSequenceNumber,
firstPacket,
latestSequenceNumber,
packetBuffer,
results;
if (this.packetBuffers[tag] == null) {
this.packetBuffers[tag] = {
nextSequenceNumber: packet.rtpHeader.sequenceNumber,
minSequenceNumberInBuffer: null,
buffer: []
};
}
packetBuffer = this.packetBuffers[tag];
if (packetBuffer.nextSequenceNumber === packet.rtpHeader.sequenceNumber) {
this.onOrderedPacket(tag, packet);
packetBuffer.nextSequenceNumber++;
if (packetBuffer.nextSequenceNumber > MAX_SEQUENCE_NUMBER) {
return (packetBuffer.nextSequenceNumber = 0);
}
} else {
// stash packet in buffer
buffers = packetBuffer.buffer;
buffers.push(packet);
if (buffers.length >= 2) {
buffers.sort(function(a, b) {
var numberA, numberB;
numberA = a.rtpHeader.sequenceNumber;
numberB = b.rtpHeader.sequenceNumber;
if (numberA - numberB >= 60000) {
// large enough gap
return -1; // a comes first
} else if (numberB - numberA >= 60000) {
// large enough gap
return 1; // b comes first
} else {
return numberA - numberB;
}
});
while (
buffers.length > 0 &&
buffers[0].rtpHeader.sequenceNumber ===
packetBuffer.nextSequenceNumber
) {
this.onOrderedPacket(tag, buffers.shift());
packetBuffer.nextSequenceNumber++;
if (packetBuffer.nextSequenceNumber > MAX_SEQUENCE_NUMBER) {
packetBuffer.nextSequenceNumber = 0;
}
}
results = [];
while (buffers.length >= 2) {
latestSequenceNumber =
buffers[buffers.length - 1].rtpHeader.sequenceNumber;
diff = latestSequenceNumber - packetBuffer.nextSequenceNumber;
if (diff < 0) {
diff += MAX_SEQUENCE_NUMBER + 1;
}
if (diff < this.unorderedPacketBufferSize) {
break;
}
firstPacket = buffers.shift();
if (
packetBuffer.nextSequenceNumber !==
firstPacket.rtpHeader.sequenceNumber
) {
discardedSequenceNumber =
firstPacket.rtpHeader.sequenceNumber - 1;
if (discardedSequenceNumber < 0) {
discardedSequenceNumber += MAX_SEQUENCE_NUMBER;
}
if (packetBuffer.nextSequenceNumber !== discardedSequenceNumber) {
logger.warn(
`rtp: ${tag}: incoming packet loss: sequence number ${packetBuffer.nextSequenceNumber}-${discardedSequenceNumber}`
);
} else {
logger.warn(
`rtp: ${tag}: incoming packet loss: sequence number ${discardedSequenceNumber}`
);
}
}
// consume the first packet
this.onOrderedPacket(tag, firstPacket);
packetBuffer.nextSequenceNumber =
firstPacket.rtpHeader.sequenceNumber + 1;
if (packetBuffer.nextSequenceNumber > MAX_SEQUENCE_NUMBER) {
results.push((packetBuffer.nextSequenceNumber = 0));
} else {
results.push(void 0);
}
}
return results;
}
}
}
onH264NALUnit(clientId, nalUnit, packet, timestamp) {
if (this.h264NALUnitBuffer[clientId] == null) {
this.h264NALUnitBuffer[clientId] = [];
}
this.h264NALUnitBuffer[clientId].push(nalUnit);
if (packet.rtpHeader.marker) {
this.emit(
"h264_nal_units",
clientId,
this.h264NALUnitBuffer[clientId],
timestamp
);
return (this.h264NALUnitBuffer[clientId] = []);
}
}
onAACAccessUnits(clientId, accessUnits, packet, timestamp) {
if (this.aacAccessUnitBuffer[clientId] == null) {
this.aacAccessUnitBuffer[clientId] = [];
}
this.aacAccessUnitBuffer[clientId] = this.aacAccessUnitBuffer[
clientId
].concat(accessUnits);
if (packet.rtpHeader.marker) {
this.emit(
"aac_access_units",
clientId,
this.aacAccessUnitBuffer[clientId],
timestamp
);
return (this.aacAccessUnitBuffer[clientId] = []);
}
}
onOrderedPacket(tag, packet) {
var clientId, j, len, match, nalUnit, ref, results;
if ((match = /^h264:(.*)$/.exec(tag)) != null) {
clientId = match[1];
if (packet.h264.fu_a != null) {
// startBit and endBit won't both be set to 1 in the same FU header
if (packet.h264.fu_a.fuHeader.startBit) {
this.fragmentedH264PacketBuffer[tag] = [
Buffer.from([
(packet.h264.nal_ref_idc << 5) |
packet.h264.fu_a.fuHeader.nal_unit_payload_type
]),
packet.h264.fu_a.nal_unit_fragment
];
} else if (this.fragmentedH264PacketBuffer[tag] != null) {
this.fragmentedH264PacketBuffer[tag].push(
packet.h264.fu_a.nal_unit_fragment
);
} else {
logger.warn(
`rtp: ${tag}: discarded fragmented incoming packet: ${packet.rtpHeader.sequenceNumber}`
);
return;
}
if (packet.h264.fu_a.fuHeader.endBit) {
this.onH264NALUnit(
clientId,
Buffer.concat(this.fragmentedH264PacketBuffer[tag]),
packet,
packet.rtpHeader.timestamp
);
return (this.fragmentedH264PacketBuffer[tag] = null);
}
} else if (packet.h264.stap_a != null) {
ref = packet.h264.stap_a.nalUnits;
results = [];
for (j = 0, len = ref.length; j < len; j++) {
nalUnit = ref[j];
results.push(
this.onH264NALUnit(
clientId,
nalUnit,
packet,
packet.rtpHeader.timestamp
)
); // single NAL unit
}
return results;
} else {
return this.onH264NALUnit(
clientId,
packet.h264.nal_unit,
packet,
packet.rtpHeader.timestamp
); // STAP-A
// FU-A
}
} else if ((match = /^aac:(.*)$/.exec(tag)) != null) {
clientId = match[1];
return this.onAACAccessUnits(
clientId,
packet.aac.accessUnits,
packet,
packet.rtpHeader.timestamp
);
} else {
throw new Error(`Unknown tag: ${tag}`);
}
}
};
api = {
RTPParser: RTPParser,
// Number of bytes in RTP header
RTP_HEADER_LEN: RTP_HEADER_LEN,
RTCP_PACKET_TYPE_SENDER_REPORT: 200, // SR
RTCP_PACKET_TYPE_RECEIVER_REPORT: 201, // RR
RTCP_PACKET_TYPE_SOURCE_DESCRIPTION: 202, // SDES
RTCP_PACKET_TYPE_GOODBYE: 203, // BYE
RTCP_PACKET_TYPE_APPLICATION_DEFINED: 204, // APP
H264_NAL_UNIT_TYPE_STAP_A: 24,
H264_NAL_UNIT_TYPE_STAP_B: 25,
H264_NAL_UNIT_TYPE_MTAP16: 26,
H264_NAL_UNIT_TYPE_MTAP24: 27,
H264_NAL_UNIT_TYPE_FU_A: 28,
H264_NAL_UNIT_TYPE_FU_B: 29,
// Remove padding from the end of the buffer
removeTrailingPadding: function(bits) {
var paddingLength;
paddingLength = bits.last_get_byte_at(0);
return bits.remove_trailing_bytes(paddingLength);
},
readRTCPSenderReport: function(bits) {
var i, info, j, readBytes, ref, reportBlock, startBytePos;
// RFC 3550 - 6.4.1 SR: Sender Report RTCP Packet
startBytePos = bits.current_position().byte;
info = {};
info.version = bits.read_bits(2);
info.padding = bits.read_bit();
if (info.padding === 1) {
api.removeTrailingPadding(bits);
}
info.reportCount = bits.read_bits(5);
info.payloadType = bits.read_byte(); // == 200
if (info.payloadType !== api.RTCP_PACKET_TYPE_SENDER_REPORT) {
throw new Error(
`payload type must be ${api.RTCP_PACKET_TYPE_SENDER_REPORT}`
);
}
info.wordsMinusOne = bits.read_bits(16);
info.totalBytes = (info.wordsMinusOne + 1) * 4;
info.ssrc = bits.read_bits(32);
info.ntpTimestamp = [bits.read_bits(32), bits.read_bits(32)];
info.ntpTimestampInMs = api.ntpTimestampToTime(info.ntpTimestamp);
info.rtpTimestamp = bits.read_bits(32);
info.senderPacketCount = bits.read_bits(32);
info.senderOctetCount = bits.read_bits(32);
info.reportBlocks = [];
for (
i = j = 0, ref = info.reportCount;
0 <= ref ? j < ref : j > ref;
i = 0 <= ref ? ++j : --j
) {
reportBlock = {};
reportBlock.ssrc = bits.read_bits(32);
reportBlock.fractionLost = bits.read_byte();
reportBlock.packetsLost = bits.read_int(24);
reportBlock.highestSequenceNumber = bits.read_bits(32);
reportBlock.jitter = bits.read_bits(32);
reportBlock.lastSR = bits.read_bits(32);
reportBlock.delaySinceLastSR = bits.read_bits(32);
info.reportBlocks.push(reportBlock);
}
// skip padding bytes
readBytes = bits.current_position().byte - startBytePos;
if (readBytes < info.totalBytes) {
bits.skip_bytes(info.totalBytes - readBytes);
}
return info;
},
readRTCPReceiverReport: function(bits) {
var i, info, j, readBytes, ref, reportBlock, startBytePos;
// RFC 3550 - 6.4.2 RR: Receiver Report RTCP Packet
startBytePos = bits.current_position().byte;
info = {};
info.version = bits.read_bits(2);
info.padding = bits.read_bit();
if (info.padding === 1) {
api.removeTrailingPadding(bits);
}
info.reportCount = bits.read_bits(5);
info.payloadType = bits.read_byte(); // == 201
if (info.payloadType !== api.RTCP_PACKET_TYPE_RECEIVER_REPORT) {
throw new Error(
`payload type must be ${api.RTCP_PACKET_TYPE_RECEIVER_REPORT}`
);
}
info.wordsMinusOne = bits.read_bits(16);
info.totalBytes = (info.wordsMinusOne + 1) * 4;
info.ssrc = bits.read_bits(32);
info.reportBlocks = [];
for (
i = j = 0, ref = info.reportCount;
0 <= ref ? j < ref : j > ref;
i = 0 <= ref ? ++j : --j
) {
reportBlock = {};
reportBlock.ssrc = bits.read_bits(32);
reportBlock.fractionLost = bits.read_byte();
reportBlock.packetsLost = bits.read_int(24);
reportBlock.highestSequenceNumber = bits.read_bits(32);
reportBlock.jitter = bits.read_bits(32);
reportBlock.lastSR = bits.read_bits(32);
reportBlock.delaySinceLastSR = bits.read_bits(32);
info.reportBlocks.push(reportBlock);
}
// skip padding bytes
readBytes = bits.current_position().byte - startBytePos;
if (readBytes < info.totalBytes) {
bits.skip_bytes(info.totalBytes - readBytes);
}
return info;
},
readRTCPSourceDescription: function(bits) {
var bytesPastBoundary,
chunk,
i,
info,
j,
nullOctet,
readBytes,
ref,
sdesItem,
startBytePos;
// RFC 3550 - 6.5 SDES: Source Description RTCP Packet
startBytePos = bits.current_position().byte;
info = {};
info.version = bits.read_bits(2);
info.padding = bits.read_bit();
if (info.padding === 1) {
api.removeTrailingPadding(bits);
}
info.sourceCount = bits.read_bits(5);
info.payloadType = bits.read_byte(); // == 202
if (info.payloadType !== api.RTCP_PACKET_TYPE_SOURCE_DESCRIPTION) {
throw new Error(
`payload type must be ${api.RTCP_PACKET_TYPE_SOURCE_DESCRIPTION}`
);
}
info.wordsMinusOne = bits.read_bits(16);
info.totalBytes = (info.wordsMinusOne + 1) * 4;
info.chunks = [];
for (
i = j = 0, ref = info.sourceCount;
0 <= ref ? j < ref : j > ref;
i = 0 <= ref ? ++j : --j
) {
chunk = {};
chunk.ssrc_csrc = bits.read_bits(32);
chunk.sdesItems = [];
chunk.sdes = {};
while (true) {
sdesItem = {};
sdesItem.type = bits.read_byte();
if (sdesItem.type === 0) {
// terminate the list
// skip until the next 32-bit boundary
bytesPastBoundary =
(bits.current_position().byte - startBytePos) % 4;
if (bytesPastBoundary > 0) {
while (bytesPastBoundary < 4) {
nullOctet = bits.read_byte();
if (nullOctet !== 0x00) {
throw new Error(`padding octet must be 0x00: ${nullOctet}`);
}
bytesPastBoundary++;
}
}
break;
}
sdesItem.octetCount = bits.read_byte();
if (sdesItem.octetCount > 255) {
throw new Error(
`octet count too large: ${sdesItem.octetCount} <= 255`
);
}
sdesItem.text = bits.read_bytes(sdesItem.octetCount).toString("utf8");
switch (sdesItem.type) {
case 1: // Canonical End-Point Identifier
chunk.sdes.cname = sdesItem.text;
break;
case 2: // User Name
chunk.sdes.name = sdesItem.text;
break;
case 3: // Electronic Mail Address
chunk.sdes.email = sdesItem.text;
break;
case 4: // Phone Number
chunk.sdes.phone = sdesItem.text;
break;
case 5: // Geographic User Location
chunk.sdes.loc = sdesItem.text;
break;
case 6: // Application or Tool Name
chunk.sdes.tool = sdesItem.text;
break;
case 7: // Notice/Status
chunk.sdes.note = sdesItem.text;
break;
case 8: // Private Extensions
chunk.sdes.priv = sdesItem.text;
break;
default:
throw new Error(
"unknown SDES item type in source description " +
`RTCP packet: ${chunk.type} (maybe not implemented yet)`
);
}
chunk.sdesItems.push(sdesItem);
}
info.chunks.push(chunk);
}
// skip padding bytes
readBytes = bits.current_position().byte - startBytePos;
if (readBytes < info.totalBytes) {
bits.skip_bytes(info.totalBytes - readBytes);
}
return info;
},
readRTCPGoodbye: function(bits) {
var info, readBytes, reason, startBytePos;
// RFC 3550 - 6.6 BYE: Goodbye RTCP Packet
startBytePos = bits.current_position().byte;
info = {};
info.version = bits.read_bits(2);
info.padding = bits.read_bit();
if (info.padding === 1) {
api.removeTrailingPadding(bits);
}
info.sourceCount = bits.read_bits(5);
info.payloadType = bits.read_byte(); // == 203
if (info.payloadType !== api.RTCP_PACKET_TYPE_GOODBYE) {
throw new Error(`payload type must be ${api.RTCP_PACKET_TYPE_GOODBYE}`);
}
info.wordsMinusOne = bits.read_bits(16);
info.totalBytes = (info.wordsMinusOne + 1) * 4;
info.ssrc = bits.read_bits(32);
if (bits.has_more_data()) {
info.reasonOctetCount = bits.read_byte();
reason = bits.read_bytes(info.reasonOctetCount);
}
// skip padding bytes
readBytes = bits.current_position().byte - startBytePos;
if (readBytes < info.totalBytes) {
bits.skip_bytes(info.totalBytes - readBytes);
}
return info;
},
readRTCPApplicationDefined: function(bits) {
var info, readBytes, startBytePos;
// RFC 3550 - 6.7 APP: Application-Defined RTCP Packet
startBytePos = bits.current_position().byte;
info = {};
info.version = bits.read_bits(2);
info.padding = bits.read_bit();
if (info.padding === 1) {
api.removeTrailingPadding(bits);
}
info.subtype = bits.read_bits(5);
info.payloadType = bits.read_byte(); // == 204
if (info.payloadType !== api.RTCP_PACKET_TYPE_APPLICATION_DEFINED) {
throw new Error(
`payload type must be ${api.RTCP_PACKET_TYPE_APPLICATION_DEFINED}`
);
}
info.wordsMinusOne = bits.read_bits(16);
info.totalBytes = (info.wordsMinusOne + 1) * 4;
info.ssrc_csrc = bits.read_bits(32);
info.name = bits.read_bytes(4).toString("ascii");
// read the application-dependent data (remaining bytes)
readBytes = bits.current_position().byte - startBytePos;
if (readBytes < info.totalBytes) {
info.applicationData = bits.read_bytes(info.totalBytes - readBytes);
} else {
info.applicationData = null;
}
return info;
},
readRTPFixedHeader: function(bits) {
var i, info, j, ref;
// RFC 3550 - 5.1 RTP Fixed Header Fields
info = {};
info.version = bits.read_bits(2);
info.padding = bits.read_bit();
if (info.padding === 1) {
api.removeTrailingPadding(bits);
}
info.extension = bits.read_bit();
info.csrcCount = bits.read_bits(4);
info.marker = bits.read_bit();
info.payloadType = bits.read_bits(7);
info.sequenceNumber = bits.read_bits(16);
info.timestamp = bits.read_bits(32);
info.ssrc = bits.read_bits(32);
info.csrc = [];
for (
i = j = 0, ref = info.csrcCount;
0 <= ref ? j < ref : j > ref;
i = 0 <= ref ? ++j : --j
) {
info.csrc.push(bits.read_bits(32));
}
return info;
},
parseAACPacket: function(buf, params) {
var bits, packet;
bits = new Bits(buf);
packet = {};
packet.rtpHeader = api.readRTPFixedHeader(bits);
packet.aac = api.readAACPayload(bits, params);
return packet;
},
parseH264Packet: function(buf) {
var bits, packet;
bits = new Bits(buf);
packet = {};
packet.rtpHeader = api.readRTPFixedHeader(bits);
packet.h264 = api.readH264Payload(bits);
return packet;
},
readH264Payload: function(bits) {
var info, ref, ref1;
info = {};
info.forbidden_zero_bit = bits.read_bit(); // 1 indicates error
if (info.forbidden_zero_bit !== 0) {
throw new Error(
`forbidden_zero_bit must be 0 (got ${info.forbidden_zero_bit})`
);
}
info.nal_ref_idc = bits.read_bits(2); // == 00: not important, > 00: important
info.nal_unit_type = bits.read_bits(5);
if (1 <= (ref = info.nal_unit_type) && ref <= 23) {
// Single NAL unit packet
bits.push_back_byte();
info.nal_unit = bits.remaining_buffer();
} else if (24 <= (ref1 = info.nal_unit_type) && ref1 <= 29) {
switch (info.nal_unit_type) {
case api.H264_NAL_UNIT_TYPE_STAP_A: // STAP-A (24)
info.stap_a = api.readH264STAP_A(bits);
break;
case api.H264_NAL_UNIT_TYPE_FU_A: // FU-A (28)
info.fu_a = api.readH264FragmentationUnitA(bits);
break;
default:
throw new Error(
`Not implemented: nal_unit_type=${info.nal_unit_type} (please report this bug)`
);
}
} else {
throw new Error(`Invalid nal_unit_type=${info.nal_unit_type}`);
}
return info;
},
// Read Single-Time Aggregation Packet type A (STAP-A)
readH264STAP_A: function(bits) {
var info, nalUnitSize;
info = {
nalUnits: []
};
while (bits.get_remaining_bytes() >= 2) {
nalUnitSize = bits.read_bits(16);
info.nalUnits.push(bits.read_bytes(nalUnitSize));
}
if (info.nalUnits.length < 1) {
logger.error("rtp: error: STAP-A does not contain a NAL unit");
}
return info;
},
readH264FragmentationUnitA: function(bits) {
var info;
info = {};
info.fuHeader = api.readH264FragmentationUnitHeader(bits);
info.nal_unit_fragment = bits.remaining_buffer();
return info;
},
// FU header
readH264FragmentationUnitHeader: function(bits) {
var info, reservedBit;
info = {};
info.startBit = bits.read_bit();
info.endBit = bits.read_bit();
reservedBit = bits.read_bit();
if (reservedBit !== 0) {
throw new Error(`reserved bit must be 0 (got ${reservedBit})`);
}
info.nal_unit_payload_type = bits.read_bits(5);
return info;
},
parsePacket: function(buf) {
var bits, packet, payloadValue;
bits = new Bits(buf);
packet = {};
payloadValue = bits.get_byte_at(1); // including marker bit
switch (payloadValue) {
case api.RTCP_PACKET_TYPE_SENDER_REPORT:
packet.rtcpSenderReport = api.readRTCPSenderReport(bits);
break;
case api.RTCP_PACKET_TYPE_RECEIVER_REPORT:
packet.rtcpReceiverReport = api.readRTCPReceiverReport(bits);
break;
case api.RTCP_PACKET_TYPE_SOURCE_DESCRIPTION:
packet.rtcpSourceDescription = api.readRTCPSourceDescription(bits);
break;
case api.RTCP_PACKET_TYPE_GOODBYE:
packet.rtcpGoodbye = api.readRTCPGoodbye(bits);
break;
case api.RTCP_PACKET_TYPE_APPLICATION_DEFINED:
packet.rtcpApplicationDefined = api.readRTCPApplicationDefined(bits); // RTP data transfer protocol - fixed header
break;
default:
packet.rtpHeader = api.readRTPFixedHeader(bits);
}
return packet;
},
parsePackets: function(buf) {
var bits, packet, packets, payloadValue;
bits = new Bits(buf);
packets = [];
while (bits.has_more_data()) {
packet = {};
payloadValue = bits.get_byte_at(1); // including marker bit
switch (payloadValue) {
case api.RTCP_PACKET_TYPE_SENDER_REPORT:
packet.rtcpSenderReport = api.readRTCPSenderReport(bits);
break;
case api.RTCP_PACKET_TYPE_RECEIVER_REPORT:
packet.rtcpReceiverReport = api.readRTCPReceiverReport(bits);
break;
case api.RTCP_PACKET_TYPE_SOURCE_DESCRIPTION:
packet.rtcpSourceDescription = api.readRTCPSourceDescription(bits);
break;
case api.RTCP_PACKET_TYPE_GOODBYE:
packet.rtcpGoodbye = api.readRTCPGoodbye(bits);
break;
case api.RTCP_PACKET_TYPE_APPLICATION_DEFINED:
packet.rtcpApplicationDefined = api.readRTCPApplicationDefined(
bits
); // RTP data transfer protocol - fixed header
break;
default:
packet.rtpHeader = api.readRTPFixedHeader(bits);
}
packets.push(packet);
}
return packets;
},
// Replace SSRC in-place in the given RTP header
replaceSSRCInRTP: function(buf, ssrc) {
buf[8] = (ssrc >>> 24) & 0xff;
buf[9] = (ssrc >>> 16) & 0xff;
buf[10] = (ssrc >>> 8) & 0xff;
buf[11] = ssrc & 0xff;
},
// ntpTimestamp: [ <32-bit second part>, <32-bit fractional second part> ]
ntpTimestampToTime: function(ntpTimestamp) {
var ms, sec;
sec = ntpTimestamp[0] - EPOCH;
ms = ntpTimestamp[1] / NTP_SCALE_FRAC / 1000;
return sec * 1000 + ms;
},
// Get NTP timestamp for a time
// time is expressed the same as Date.now()
getNTPTimestamp: function(time) {
var ms, ntp_sec, ntp_usec, sec;
sec = parseInt(time / 1000);
ms = time - sec * 1000;
ntp_sec = sec + EPOCH;
ntp_usec = Math.round(ms * 1000 * NTP_SCALE_FRAC);
return [ntp_sec, ntp_usec];
},
readAACPayload: function(bits, params) {
var accessUnit, auHeader, auHeaders, i, info, j, k, len, ref;
info = {};
info.auHeadersLengthBits = bits.read_bits(16); // in bits
info.numAUHeaders = info.auHeadersLengthBits / 16;
auHeaders = [];
for (
i = j = 0, ref = info.numAUHeaders;
0 <= ref ? j < ref : j > ref;
i = 0 <= ref ? ++j : --j
) {
params.index = i;
auHeaders.push(api.readAACAUHeader(bits, params));
}
info.auHeaders = auHeaders;
info.accessUnits = [];
for (k = 0, len = auHeaders.length; k < len; k++) {
auHeader = auHeaders[k];
info.accessUnits.push(bits.read_bytes(auHeader.auSize));
accessUnit = info.accessUnits[info.accessUnits.length - 1];
}
return info;
},
readAACAUHeader: function(bits, params) {
var info;
if (params.sizelength == null) {
throw new Error("sizelength is not defined in params");
}
info = {};
// size in octets of the associated Access Unit in the
// Access Unit Data Section in the same RTP packet
info.auSize = bits.read_bits(params.sizelength);
// serial number of the associated Access Unit (fragment).
if (params.index == null) {
throw new Error("index is not defined in params");
}
if (params.index > 0) {
if (params.indexdeltalength == null) {
throw new Error("indexdeltalength is not defined in params");
}
info.auIndexDelta = bits.read_bits(params.indexdeltalength);
} else {
if (params.indexlength == null) {
throw new Error("indexlength is not defined in params");
}
info.auIndex = bits.read_bits(params.indexlength);
}
return info;
},
// Used for encapsulating AAC audio data
// opts:
// accessUnitLength (number): number of bytes in the access unit
createAudioHeader: function(opts) {
var accessUnit, header, j, len, numBits, ref;
if (opts.accessUnits.length > 4095) {
throw new Error(
`too many audio access units: ${opts.accessUnits.length} (must be <= 4095)`
);
}
numBits = opts.accessUnits.length * 16; // 2 bytes per access unit
//# payload
//# See section 3.2.1 and 3.3.6 of RFC 3640 for details
//# AU Header Section
// AU-headers-length(16) for AAC-hbr
// Number of bits in the AU-headers
header = [(numBits >> 8) & 0xff, numBits & 0xff];
ref = opts.accessUnits;
for (j = 0, len = ref.length; j < len; j++) {
accessUnit = ref[j];
header = header.concat(api.createAudioAUHeader(accessUnit.length));
}
return header;
},
groupAudioFrames: function(adtsFrames) {
var adtsFrame, currentGroup, groups, i, j, len, packetSize;
packetSize = RTP_HEADER_LEN;
groups = [];
currentGroup = [];
for (i = j = 0, len = adtsFrames.length; j < len; i = ++j) {
adtsFrame = adtsFrames[i];
packetSize += adtsFrame.length + 2; // 2 bytes for AU-Header
if (packetSize > MAX_PAYLOAD_SIZE) {
groups.push(currentGroup);
currentGroup = [];
packetSize = RTP_HEADER_LEN + adtsFrame.length + 2;
}
currentGroup.push(adtsFrame);
}
if (currentGroup.length > 0) {
groups.push(currentGroup);
}
return groups;
},
createAudioAUHeader: function(accessUnitLength) {
// AU Header
// AU-size(13) by SDP
// AU-Index(3) or AU-Index-Delta(3)
// AU-Index is used for the first access unit, and the value must be 0.
// AU-Index-Delta is used for the consecutive access units.
// When interleaving is not applied, AU-Index-Delta is 0.
return [accessUnitLength >> 5, (accessUnitLength & 0b11111) << 3];
},
// Used for encapsulating H.264 video data
// There is no Auxiliary Section for AAC-hbr
createFragmentationUnitHeader: function(opts) {
return [
// Fragmentation Unit
// See section 5.8 of RFC 6184 for details
// FU indicator
// forbidden_zero_bit(1), nal_ref_idc(2), type(5)
// type is 28 for FU-A
opts.nal_ref_idc | 28,
// FU header
// start bit(1) == 0, end bit(1) == 1, reserved bit(1), type(5)
(opts.isStart << 7) | (opts.isEnd << 6) | opts.nal_unit_type
];
},
// Create RTP header
// opts:
// marker (boolean): true if this is the last packet of the
// access unit indicated by the RTP timestamp
// payloadType (number): payload type
// sequenceNumber (number): sequence number
// timestamp (number): timestamp in 90 kHz clock rate
// ssrc (number): SSRC (can be null)
createRTPHeader: function(opts) {
var ref, seqNum, ssrc, ts;
seqNum = opts.sequenceNumber;
ts = opts.timestamp;
ssrc = (ref = opts.ssrc) != null ? ref : 0;
return [
// version(2): 2
// padding(1): 0
// extension(1): 0
// CSRC count(4): 0
0b10000000,
// marker(1)
// payload type(7)
(opts.marker << 7) | opts.payloadType,
// sequence number(16)
seqNum >>> 8,
seqNum & 0xff,
// timestamp(32) in 90 kHz clock rate
(ts >>> 24) & 0xff,
(ts >>> 16) & 0xff,
(ts >>> 8) & 0xff,
ts & 0xff,
// SSRC(32)
(ssrc >>> 24) & 0xff,
(ssrc >>> 16) & 0xff,
(ssrc >>> 8) & 0xff,
ssrc & 0xff
];
},
// Create RTCP BYE (Goodbye) packet
createGoodbye: function(opts) {
var data, j, len, length, padLen, reason, reasonLen, ssrc, ssrcs;
if ((opts != null ? opts.ssrcs : void 0) == null) {
throw new Error("createGoodbye: ssrcs is required");
}
ssrcs = opts.ssrcs;
if (ssrcs.length > 0b11111) {
throw new Error(
`createGoodbye: too many ssrcs: ${ssrcs.length} (must be <= 31)`
);
}
// Reason for leaving
reason = [...Buffer.from("End of stream", "utf8")]; // Convert Buffer to array
reasonLen = reason.length;
// Number of bytes until the next 32-bit boundary
padLen = 4 - ((1 + reasonLen) % 4);
if (reason.length > 0xff) {
throw new Error(
`createGoodbye: reason is too long: ${reason.length} (must be <= 255)`
);
}
// Length of this RTCP packet in 32-bit words minus one
// including the header and any padding
length = (4 + ssrcs.length * 4 + 1 + reasonLen + padLen) / 4 - 1;
data = [
// See section 6.6 for details
// version(2): 2 (RTP version 2)
// padding(1): 0 (padding doesn't exist)
// source count(5): number of SSRC/CSRC identifiers
0b10000000 | ssrcs.length,
// packet type(8): 203 (RTCP BYE)
203,
// length(16)
length >> 8,
length & 0xff
];
for (j = 0, len = ssrcs.length; j < len; j++) {
ssrc = ssrcs[j];
// Append SSRC
data.push(
(ssrc >>> 24) & 0xff,
(ssrc >>> 16) & 0xff,
(ssrc >>> 8) & 0xff,
ssrc & 0xff
);
}
data.push(reason.length);
data = data.concat(reason);
while (padLen-- > 0) {
data.push(0x00);
}
return data;
},
// Create RTCP Sender Report packet
// opts:
// time: timestamp of the packet
// rtpTime: timestamp relative to the start point of media
// ssrc: SSRC
// packetCount: packet count
// octetCount: octetCount
createSenderReport: function(opts) {
var length, ntp_ts, octetCount, packetCount, rtp_ts, ssrc;
if ((opts != null ? opts.ssrc : void 0) == null) {
throw new Error("createSenderReport: ssrc is required");
}
ssrc = opts.ssrc;
if ((opts != null ? opts.packetCount : void 0) == null) {
throw new Error("createSenderReport: packetCount is required");
}
packetCount = opts.packetCount;
if ((opts != null ? opts.octetCount : void 0) == null) {
throw new Error("createSenderReport: octetCount is required");
}
octetCount = opts.octetCount;
if ((opts != null ? opts.time : void 0) == null) {
throw new Error("createSenderReport: time is required");
}
ntp_ts = api.getNTPTimestamp(opts.time);