Const RedundantAudioEncoderWorkerCode
RedundantAudioEncoderWorkerCode: "class RedundantAudioEncoder {\n constructor() {\n // Each payload must be less than 1024 bytes to fit the 10 bit block length\n this.maxRedPacketSizeBytes = 1 << 10;\n // Limit payload to 1000 bytes to handle small MTU. 1000 is chosen because in Chromium-based browsers, writing audio\n // payloads larger than 1000 bytes using the WebRTC Insertable Streams API (which is used to enable dynamic audio\n // redundancy) will cause an error to be thrown and cause audio flow to permanently stop. See\n // https://crbug.com/1248479.\n this.maxAudioPayloadSizeBytes = 1000;\n // Each payload can encode a timestamp delta of 14 bits\n this.maxRedTimestampOffset = 1 << 14;\n // 4 byte RED header\n this.redHeaderSizeBytes = 4;\n // reduced size for last RED header\n this.redLastHeaderSizeBytes = 1;\n // P-Time for Opus 20 msec packets\n // We do not support other p-times or clock rates\n this.redPacketizationTime = 960;\n // distance between redundant payloads, Opus FEC handles a distance of 1\n // TODO(https://issues.amazon.com/issues/ChimeSDKAudio-55):\n // Consider making this dynamic\n this.redPacketDistance = 2;\n // maximum number of redundant payloads per RTP packet\n this.maxRedEncodings = 2;\n // Maximum number of encodings that can be recovered with a single RED packet, assuming the primary and redundant\n // payloads have FEC.\n this.redMaxRecoveryDistance = this.redPacketDistance * this.maxRedEncodings + 1;\n // maximum history of prior payloads to keep\n // generally we will expire old entries based on timestamp\n // this limit is in place just to make sure the history does not\n // grow too large in the case of erroneous timestamp inputs\n this.maxEncodingHistorySize = 10;\n // Current number of encodings we want to send\n // to the remote end. This will be dynamically\n // updated through the setNumEncodingsFromPacketloss API\n this.numRedundantEncodings = 0;\n // Used to enable or disable redundancy\n // in response to very high packet loss events\n this.redundancyEnabled = true;\n // Loss stats are reported to the main thread every 5 seconds.\n // Since timestamp differences between 2 consecutive packets\n // give us the number of samples in each channel, 1 second\n // is equivalent to 48000 samples:\n // P-time * (1000ms/1s)\n // = (960 samples/20ms) * (1000ms/1s)\n // = 48000 samples/s\n this.lossReportInterval = 48000 * 5;\n // Maximum distance of a packet from the most recent packet timestamp\n // that we will consider for recovery.\n this.maxOutOfOrderPacketDistance = 16;\n /**\n * Below are Opus helper methods and constants.\n */\n this.OPUS_BAD_ARG = -1;\n this.OPUS_INVALID_PACKET = -4;\n // Max number of Opus frames in an Opus packet is 48 (https://www.rfc-editor.org/rfc/rfc6716#section-3.2.5).\n this.OPUS_MAX_OPUS_FRAMES = 48;\n // Max number of bytes that any individual Opus frame can have.\n this.OPUS_MAX_FRAME_SIZE_BYTES = 1275;\n this.encodingHistory = new Array();\n this.opusPayloadType = 0;\n this.redPayloadType = 0;\n this.initializePacketLogs();\n }\n /**\n * Creates an instance of RedundantAudioEncoder and sets up callbacks.\n */\n static initializeWorker() {\n RedundantAudioEncoder.log('Initializing RedundantAudioEncoder');\n const encoder = new RedundantAudioEncoder();\n // RED encoding is done using WebRTC Encoded Transform\n // https://github.com/w3c/webrtc-encoded-transform/blob/main/explainer.md\n // Check the DedicatedWorkerGlobalScope for existence of\n // RTCRtpScriptTransformer interface. If exists, then\n // RTCRtpScriptTransform is supported by this browser.\n // @ts-ignore\n if (self.RTCRtpScriptTransformer) {\n // @ts-ignore\n self.onrtctransform = (event) => {\n if (event.transformer.options.type === 'SenderTransform') {\n encoder.setupSenderTransform(event.transformer.readable, event.transformer.writable);\n }\n else if (event.transformer.options.type === 'ReceiverTransform') {\n encoder.setupReceiverTransform(event.transformer.readable, event.transformer.writable);\n }\n else if (event.transformer.options.type === 'PassthroughTransform') {\n encoder.setupPassthroughTransform(event.transformer.readable, event.transformer.writable);\n }\n };\n }\n self.onmessage = (event) => {\n if (event.data.msgType === 'StartRedWorker') {\n encoder.setupSenderTransform(event.data.send.readable, event.data.send.writable);\n encoder.setupReceiverTransform(event.data.receive.readable, event.data.receive.writable);\n }\n else if (event.data.msgType === 'PassthroughTransform') {\n encoder.setupPassthroughTransform(event.data.send.readable, event.data.send.writable);\n encoder.setupPassthroughTransform(event.data.receive.readable, event.data.receive.writable);\n }\n else if (event.data.msgType === 'RedPayloadType') {\n encoder.setRedPayloadType(event.data.payloadType);\n }\n else if (event.data.msgType === 'OpusPayloadType') {\n encoder.setOpusPayloadType(event.data.payloadType);\n }\n else if (event.data.msgType === 'UpdateNumRedundantEncodings') {\n encoder.setNumRedundantEncodings(event.data.numRedundantEncodings);\n }\n else if (event.data.msgType === 'Enable') {\n encoder.setRedundancyEnabled(true);\n }\n else if (event.data.msgType === 'Disable') {\n encoder.setRedundancyEnabled(false);\n }\n };\n }\n /**\n * Post logs to the main thread\n */\n static log(msg) {\n if (RedundantAudioEncoder.shouldLog) {\n // @ts-ignore\n self.postMessage({\n type: 'REDWorkerLog',\n log: `[AudioRed] ${msg}`,\n });\n }\n }\n /**\n * Returns the number of encodings based on packetLoss value. This is used by `DefaultTransceiverController` to\n * determine when to alert the encoder to update the number of encodings. It also determines if we need to\n * turn off red in cases of very high packet loss to avoid congestion collapse.\n */\n static getNumRedundantEncodingsForPacketLoss(packetLoss) {\n let recommendedRedundantEncodings = 0;\n let shouldTurnOffRed = false;\n if (packetLoss <= 8) {\n recommendedRedundantEncodings = 0;\n }\n else if (packetLoss <= 18) {\n recommendedRedundantEncodings = 1;\n }\n else if (packetLoss <= 75) {\n recommendedRedundantEncodings = 2;\n }\n else {\n recommendedRedundantEncodings = 0;\n shouldTurnOffRed = true;\n }\n return [recommendedRedundantEncodings, shouldTurnOffRed];\n }\n /**\n * Sets up a passthrough (no-op) transform for the given streams.\n */\n setupPassthroughTransform(readable, writable) {\n RedundantAudioEncoder.log('Setting up passthrough transform');\n readable.pipeTo(writable);\n }\n /**\n * Sets up the transform stream and pipes the outgoing encoded audio frames through the transform function.\n */\n setupSenderTransform(readable, writable) {\n RedundantAudioEncoder.log('Setting up sender RED transform');\n const transformStream = new TransformStream({\n transform: this.senderTransform.bind(this),\n });\n readable.pipeThrough(transformStream).pipeTo(writable);\n return;\n }\n /**\n * Sets up the transform stream and pipes the received encoded audio frames through the transform function.\n */\n setupReceiverTransform(readable, writable) {\n RedundantAudioEncoder.log('Setting up receiver RED transform');\n const transformStream = new TransformStream({\n transform: this.receivePacketLogTransform.bind(this),\n });\n readable.pipeThrough(transformStream).pipeTo(writable);\n return;\n }\n /**\n * Set the RED payload type ideally obtained from local offer.\n */\n setRedPayloadType(payloadType) {\n this.redPayloadType = payloadType;\n RedundantAudioEncoder.log(`red payload type set to ${this.redPayloadType}`);\n }\n /**\n * Set the opus payload type ideally obtained from local offer.\n */\n setOpusPayloadType(payloadType) {\n this.opusPayloadType = payloadType;\n RedundantAudioEncoder.log(`opus payload type set to ${this.opusPayloadType}`);\n }\n /**\n * Set the number of redundant encodings\n */\n setNumRedundantEncodings(numRedundantEncodings) {\n this.numRedundantEncodings = numRedundantEncodings;\n if (this.numRedundantEncodings > this.maxRedEncodings) {\n this.numRedundantEncodings = this.maxRedEncodings;\n }\n RedundantAudioEncoder.log(`Updated numRedundantEncodings to ${this.numRedundantEncodings}`);\n }\n /**\n * Enable or disable redundancy in response to\n * high packet loss event.\n */\n setRedundancyEnabled(enabled) {\n this.redundancyEnabled = enabled;\n RedundantAudioEncoder.log(`redundancy ${this.redundancyEnabled ? 'enabled' : 'disabled'}`);\n }\n /**\n * Helper function to only enqueue audio frames if they do not exceed the audio payload byte limit imposed by\n * Chromium-based browsers. Chromium will throw an error (https://crbug.com/1248479) if an audio payload larger than\n * 1000 bytes is enqueued. Any controller that attempts to enqueue an audio payload larger than 1000 bytes will\n * encounter this error and will permanently stop sending or receiving audio.\n */\n enqueueAudioFrameIfPayloadSizeIsValid(\n // @ts-ignore\n frame, controller) {\n if (frame.data.byteLength > this.maxAudioPayloadSizeBytes)\n return;\n controller.enqueue(frame);\n }\n /**\n * Receives encoded frames and modifies as needed before sending to transport.\n */\n senderTransform(\n // @ts-ignore\n frame, controller) {\n const frameMetadata = frame.getMetadata();\n // @ts-ignore\n if (frameMetadata.payloadType !== this.redPayloadType) {\n this.enqueueAudioFrameIfPayloadSizeIsValid(frame, controller);\n return;\n }\n const primaryPayloadBuffer = this.getPrimaryPayload(frame.timestamp, frame.data);\n if (!primaryPayloadBuffer) {\n this.enqueueAudioFrameIfPayloadSizeIsValid(frame, controller);\n return;\n }\n const encodedBuffer = this.encode(frame.timestamp, primaryPayloadBuffer);\n /* istanbul ignore next */\n if (!encodedBuffer) {\n this.enqueueAudioFrameIfPayloadSizeIsValid(frame, controller);\n return;\n }\n frame.data = encodedBuffer;\n this.enqueueAudioFrameIfPayloadSizeIsValid(frame, controller);\n return;\n }\n /**\n * Get the primary payload from encoding\n */\n getPrimaryPayload(primaryTimestamp, frame) {\n const encodings = this.splitEncodings(primaryTimestamp, frame);\n if (!encodings || encodings.length < 1)\n return null;\n return encodings[encodings.length - 1].payload;\n }\n /**\n * Split up the encoding received into primary and redundant encodings\n * These will be ordered oldest to newest which is the same ordering\n * in the RTP red payload.\n */\n splitEncodings(primaryTimestamp, frame, getFecInfo = false, primarySequenceNumber = undefined) {\n // process RED headers (according to RFC 2198)\n // 0 1 2 3\n // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1\n // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+\n // |F| block PT | timestamp offset | block length |\n // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+\n //\n // last header\n // 0 1 2 3 4 5 6 7\n // +-+-+-+-+-+-+-+-+\n // |0| Block PT |\n // +-+-+-+-+-+-+-+-+\n const payload = new DataView(frame);\n let payloadSizeBytes = payload.byteLength;\n let totalPayloadSizeBytes = 0;\n let totalHeaderSizeBytes = 0;\n let primaryPayloadSizeBytes = 0;\n let payloadOffset = 0;\n let gotLastBlock = false;\n const encodings = new Array();\n const redundantEncodingBlockLengths = new Array();\n const redundantEncodingTimestamps = new Array();\n while (payloadSizeBytes > 0) {\n gotLastBlock = (payload.getUint8(payloadOffset) & 0x80) === 0;\n if (gotLastBlock) {\n // Bits 1 through 7 are payload type\n const payloadType = payload.getUint8(payloadOffset) & 0x7f;\n // Unexpected payload type. This is a bad packet.\n if (payloadType !== this.opusPayloadType) {\n return null;\n }\n totalPayloadSizeBytes += this.redLastHeaderSizeBytes;\n totalHeaderSizeBytes += this.redLastHeaderSizeBytes;\n // Accumulated block lengths are equal to or larger than the buffer, which means there is no primary block. This\n // is a bad packet.\n if (totalPayloadSizeBytes >= payload.byteLength) {\n return null;\n }\n primaryPayloadSizeBytes = payload.byteLength - totalPayloadSizeBytes;\n break;\n }\n else {\n if (payloadSizeBytes < this.redHeaderSizeBytes) {\n return null;\n }\n // Bits 22 through 31 are payload length\n const blockLength = ((payload.getUint8(payloadOffset + 2) & 0x03) << 8) + payload.getUint8(payloadOffset + 3);\n redundantEncodingBlockLengths.push(blockLength);\n const timestampOffset = payload.getUint16(payloadOffset + 1) >> 2;\n const timestamp = primaryTimestamp - timestampOffset;\n redundantEncodingTimestamps.push(timestamp);\n totalPayloadSizeBytes += blockLength + this.redHeaderSizeBytes;\n totalHeaderSizeBytes += this.redHeaderSizeBytes;\n payloadOffset += this.redHeaderSizeBytes;\n payloadSizeBytes -= this.redHeaderSizeBytes;\n }\n }\n // The last block was never found. The packet we received\n // does not have a good RED payload.\n if (!gotLastBlock) {\n // Note that sequence numbers only exist for\n // incoming audio frames.\n if (primarySequenceNumber !== undefined) {\n // This could be a possible padding packet used\n // for BWE with a good sequence number.\n // Create a dummy encoding to make sure loss values\n // are calculated correctly by consuming sequence number.\n // Note that for the receive side, we process packets only\n // for loss/recovery calculations and forward the original\n // packet without changing it even in the error case.\n encodings.push({\n payload: frame,\n isRedundant: false,\n seq: primarySequenceNumber,\n });\n return encodings;\n }\n // This is a bad packet.\n return null;\n }\n let redundantPayloadOffset = totalHeaderSizeBytes;\n for (let i = 0; i < redundantEncodingTimestamps.length; i++) {\n const redundantPayloadBuffer = new ArrayBuffer(redundantEncodingBlockLengths[i]);\n const redundantPayloadArray = new Uint8Array(redundantPayloadBuffer);\n redundantPayloadArray.set(new Uint8Array(payload.buffer, redundantPayloadOffset, redundantEncodingBlockLengths[i]), 0);\n const encoding = {\n timestamp: redundantEncodingTimestamps[i],\n payload: redundantPayloadBuffer,\n isRedundant: true,\n };\n if (getFecInfo) {\n encoding.hasFec = this.opusPacketHasFec(new DataView(redundantPayloadBuffer), redundantPayloadBuffer.byteLength);\n }\n encodings.push(encoding);\n redundantPayloadOffset += redundantEncodingBlockLengths[i];\n }\n const primaryPayloadOffset = payload.byteLength - primaryPayloadSizeBytes;\n const primaryPayloadBuffer = new ArrayBuffer(primaryPayloadSizeBytes);\n const primaryArray = new Uint8Array(primaryPayloadBuffer);\n primaryArray.set(new Uint8Array(payload.buffer, primaryPayloadOffset, primaryPayloadSizeBytes), 0);\n const encoding = {\n timestamp: primaryTimestamp,\n payload: primaryPayloadBuffer,\n isRedundant: false,\n seq: primarySequenceNumber,\n };\n if (getFecInfo) {\n encoding.hasFec = this.opusPacketHasFec(new DataView(primaryPayloadBuffer), primaryPayloadBuffer.byteLength);\n }\n encodings.push(encoding);\n return encodings;\n }\n /**\n * Create a new encoding with current primary payload and the older payloads of choice.\n */\n encode(primaryTimestamp, primaryPayload) {\n const primaryPayloadSize = primaryPayload.byteLength;\n // Payload size needs to be valid.\n if (primaryPayloadSize === 0 ||\n primaryPayloadSize >= this.maxRedPacketSizeBytes ||\n primaryPayloadSize >= this.maxAudioPayloadSizeBytes) {\n return null;\n }\n const numRedundantEncodings = this.numRedundantEncodings;\n let headerSizeBytes = this.redLastHeaderSizeBytes;\n let payloadSizeBytes = primaryPayloadSize;\n let bytesAvailable = this.maxAudioPayloadSizeBytes - primaryPayloadSize - headerSizeBytes;\n const redundantEncodingTimestamps = new Array();\n const redundantEncodingPayloads = new Array();\n // If redundancy is disabled then only send the primary payload\n if (this.redundancyEnabled) {\n // Determine how much redundancy we can fit into our packet\n let redundantTimestamp = this.uint32WrapAround(primaryTimestamp - this.redPacketizationTime * this.redPacketDistance);\n for (let i = 0; i < numRedundantEncodings; i++) {\n // Do not add redundant encodings that are beyond the maximum timestamp offset.\n if (this.uint32WrapAround(primaryTimestamp - redundantTimestamp) >= this.maxRedTimestampOffset) {\n break;\n }\n let findTimestamp = redundantTimestamp;\n let encoding = this.encodingHistory.find(e => e.timestamp === findTimestamp);\n if (!encoding) {\n // If not found or not important then look for the previous packet.\n // The current packet may have included FEC for the previous, so just\n // use the previous packet instead provided that it has voice activity.\n findTimestamp = this.uint32WrapAround(redundantTimestamp - this.redPacketizationTime);\n encoding = this.encodingHistory.find(e => e.timestamp === findTimestamp);\n }\n if (encoding) {\n const redundantEncodingSizeBytes = encoding.payload.byteLength;\n // Only add redundancy if there are enough bytes available.\n if (bytesAvailable < this.redHeaderSizeBytes + redundantEncodingSizeBytes)\n break;\n bytesAvailable -= this.redHeaderSizeBytes + redundantEncodingSizeBytes;\n headerSizeBytes += this.redHeaderSizeBytes;\n payloadSizeBytes += redundantEncodingSizeBytes;\n redundantEncodingTimestamps.unshift(encoding.timestamp);\n redundantEncodingPayloads.unshift(encoding.payload);\n }\n redundantTimestamp -= this.redPacketizationTime * this.redPacketDistance;\n redundantTimestamp = this.uint32WrapAround(redundantTimestamp);\n }\n }\n const redPayloadBuffer = new ArrayBuffer(headerSizeBytes + payloadSizeBytes);\n const redPayloadView = new DataView(redPayloadBuffer);\n // Add redundant encoding header(s) to new buffer\n let redPayloadOffset = 0;\n for (let i = 0; i < redundantEncodingTimestamps.length; i++) {\n const timestampDelta = primaryTimestamp - redundantEncodingTimestamps[i];\n redPayloadView.setUint8(redPayloadOffset, this.opusPayloadType | 0x80);\n redPayloadView.setUint16(redPayloadOffset + 1, (timestampDelta << 2) | (redundantEncodingPayloads[i].byteLength >> 8));\n redPayloadView.setUint8(redPayloadOffset + 3, redundantEncodingPayloads[i].byteLength & 0xff);\n redPayloadOffset += this.redHeaderSizeBytes;\n }\n // Add primary encoding header to new buffer\n redPayloadView.setUint8(redPayloadOffset, this.opusPayloadType);\n redPayloadOffset += this.redLastHeaderSizeBytes;\n // Add redundant payload(s) to new buffer\n const redPayloadArray = new Uint8Array(redPayloadBuffer);\n for (let i = 0; i < redundantEncodingPayloads.length; i++) {\n redPayloadArray.set(new Uint8Array(redundantEncodingPayloads[i]), redPayloadOffset);\n redPayloadOffset += redundantEncodingPayloads[i].byteLength;\n }\n // Add primary payload to new buffer\n redPayloadArray.set(new Uint8Array(primaryPayload), redPayloadOffset);\n redPayloadOffset += primaryPayload.byteLength;\n /* istanbul ignore next */\n // Sanity check that we got the expected total payload size.\n if (redPayloadOffset !== headerSizeBytes + payloadSizeBytes)\n return null;\n this.updateEncodingHistory(primaryTimestamp, primaryPayload);\n return redPayloadBuffer;\n }\n /**\n * Update the encoding history with the latest primary encoding\n */\n updateEncodingHistory(primaryTimestamp, primaryPayload) {\n // Remove encodings from the history if they are too old.\n for (const encoding of this.encodingHistory) {\n const maxTimestampDelta = this.redPacketizationTime * this.redMaxRecoveryDistance;\n if (primaryTimestamp - encoding.timestamp >= maxTimestampDelta) {\n this.encodingHistory.shift();\n }\n else {\n break;\n }\n }\n // Only add an encoding to the history if the encoding is deemed to be important. An encoding is important if it is\n // a CELT-only packet or contains voice activity.\n const packet = new DataView(primaryPayload);\n if (this.opusPacketIsCeltOnly(packet) ||\n this.opusPacketHasVoiceActivity(packet, packet.byteLength) > 0) {\n // Check if adding an encoding will cause the length of the encoding history to exceed the maximum history size.\n // This is not expected to happen but could occur if we get incorrect timestamps. We want to make sure our memory\n // usage is bounded. In this case, just clear the history and start over from empty.\n if (this.encodingHistory.length + 1 > this.maxEncodingHistorySize)\n this.encodingHistory.length = 0;\n this.encodingHistory.push({ timestamp: primaryTimestamp, payload: primaryPayload });\n }\n }\n /**\n * Initialize packet logs and metric values.\n */\n initializePacketLogs() {\n // The extra space from the max RED recovery distance is to ensure that we do not incorrectly count recovery for\n // packets that have already been received but are outside of the max out-of-order distance.\n const packetLogSize = this.maxOutOfOrderPacketDistance + this.redMaxRecoveryDistance;\n this.primaryPacketLog = {\n window: new Array(packetLogSize),\n index: 0,\n windowSize: packetLogSize,\n };\n this.redRecoveryLog = {\n window: new Array(packetLogSize),\n index: 0,\n windowSize: packetLogSize,\n };\n this.fecRecoveryLog = {\n window: new Array(packetLogSize),\n index: 0,\n windowSize: packetLogSize,\n };\n this.totalAudioPacketsExpected = 0;\n this.totalAudioPacketsLost = 0;\n this.totalAudioPacketsRecoveredRed = 0;\n this.totalAudioPacketsRecoveredFec = 0;\n }\n /**\n * Receives encoded frames from the server\n * and adds the timestamps to a packet log\n * to calculate an approximate recovery metric.\n */\n receivePacketLogTransform(\n // @ts-ignore\n frame, controller) {\n const frameMetadata = frame.getMetadata();\n // @ts-ignore\n if (frameMetadata.payloadType !== this.redPayloadType) {\n this.enqueueAudioFrameIfPayloadSizeIsValid(frame, controller);\n return;\n }\n // @ts-ignore\n const encodings = this.splitEncodings(frame.timestamp, frame.data, \n /*getFecInfo*/ true, frameMetadata.sequenceNumber);\n if (!encodings) {\n this.enqueueAudioFrameIfPayloadSizeIsValid(frame, controller);\n return;\n }\n for (let i = encodings.length - 1; i >= 0; i--) {\n if (this.updateLossStats(encodings[i])) {\n this.updateRedStats(encodings[i]);\n this.updateFecStats(encodings[i]);\n }\n }\n this.maybeReportLossStats(frameMetadata.synchronizationSource, encodings[encodings.length - 1].timestamp);\n this.enqueueAudioFrameIfPayloadSizeIsValid(frame, controller);\n }\n /**\n * Adds a timestamp to the primary packet log.\n * This also updates totalAudioPacketsLost and totalAudioPacketsExpected by looking\n * at the difference between timestamps.\n *\n * @param encoding : The encoding to be analyzed\n * @returns false if sequence number was greater than max out of order distance\n * true otherwise\n */\n updateLossStats(encoding) {\n if (encoding.isRedundant)\n return true;\n const timestamp = encoding.timestamp;\n const seq = encoding.seq;\n if (this.totalAudioPacketsExpected === 0) {\n this.totalAudioPacketsExpected = 1;\n this.newestSequenceNumber = seq;\n this.addTimestamp(this.primaryPacketLog, timestamp);\n return true;\n }\n const diff = this.int16(seq - this.newestSequenceNumber);\n if (diff < -this.maxOutOfOrderPacketDistance)\n return false;\n if (diff < 0) {\n if (!this.hasTimestamp(this.primaryPacketLog, timestamp)) {\n if (this.totalAudioPacketsLost > 0)\n this.totalAudioPacketsLost--;\n this.addTimestamp(this.primaryPacketLog, timestamp);\n this.removeFromRecoveryWindows(timestamp);\n }\n }\n else if (diff > 1) {\n this.totalAudioPacketsLost += diff - 1;\n }\n if (diff > 0) {\n this.totalAudioPacketsExpected += diff;\n this.newestSequenceNumber = encoding.seq;\n this.addTimestamp(this.primaryPacketLog, timestamp);\n }\n return true;\n }\n /**\n * Adds a timestamp to the red recovery log if it is not present in\n * the primary packet log and if it's not too old.\n *\n * @param encoding : The encoding to be analyzed\n */\n updateRedStats(encoding) {\n if (!encoding.isRedundant || this.totalAudioPacketsLost === 0)\n return;\n const timestamp = encoding.timestamp;\n if (!this.hasTimestamp(this.primaryPacketLog, timestamp)) {\n if (!this.hasTimestamp(this.redRecoveryLog, timestamp)) {\n this.totalAudioPacketsRecoveredRed++;\n this.addTimestamp(this.redRecoveryLog, timestamp);\n }\n if (this.removeTimestamp(this.fecRecoveryLog, timestamp)) {\n /* istanbul ignore else */\n if (this.totalAudioPacketsRecoveredFec > 0)\n this.totalAudioPacketsRecoveredFec--;\n }\n }\n }\n /**\n * Adds a timestamp to the fec recovery log if it is not present in\n * the primary packet log and red recovery log and if it is not too old.\n *\n * @param encoding : The encoding to be analyzed\n */\n updateFecStats(encoding) {\n if (!encoding.hasFec || this.totalAudioPacketsLost === 0)\n return;\n const fecTimestamp = encoding.timestamp - this.redPacketizationTime;\n if (this.hasTimestamp(this.primaryPacketLog, fecTimestamp) ||\n this.hasTimestamp(this.redRecoveryLog, fecTimestamp) ||\n this.hasTimestamp(this.fecRecoveryLog, fecTimestamp)) {\n return;\n }\n this.totalAudioPacketsRecoveredFec++;\n this.addTimestamp(this.fecRecoveryLog, fecTimestamp);\n }\n /**\n * Reports loss metrics to DefaultTransceiverController\n *\n * @param timestamp : Timestamp of most recent primary packet\n */\n maybeReportLossStats(ssrc, timestamp) {\n if (timestamp === undefined ||\n timestamp - this.lastLossReportTimestamp < this.lossReportInterval)\n return;\n /* istanbul ignore next */\n if (RedundantAudioEncoder.shouldReportStats) {\n // @ts-ignore\n self.postMessage({\n type: 'RedundantAudioEncoderStats',\n ssrc,\n totalAudioPacketsLost: this.totalAudioPacketsLost,\n totalAudioPacketsExpected: this.totalAudioPacketsExpected,\n totalAudioPacketsRecoveredRed: this.totalAudioPacketsRecoveredRed,\n totalAudioPacketsRecoveredFec: this.totalAudioPacketsRecoveredFec,\n });\n }\n this.lastLossReportTimestamp = timestamp;\n }\n /**\n * Adds a timestamp to a packet log\n *\n * @param packetLog : The packetlog to add the timestamp to\n * @param timestamp : The timestamp that should be added\n */\n addTimestamp(packetLog, timestamp) {\n if (timestamp === undefined) {\n return;\n }\n packetLog.window[packetLog.index] = timestamp;\n packetLog.index = (packetLog.index + 1) % packetLog.windowSize;\n }\n /**\n * Checks if a timestamp is in a packetlog\n *\n * @param packetLog : The packetlog to search\n * @param timestamp : The timestamp to search for\n * @returns true if timestamp is present, false otherwise\n */\n hasTimestamp(packetLog, timestamp) {\n const element = packetLog.window.find(t => t === timestamp);\n return !!element;\n }\n /**\n * Removes a timestamp from a packet log\n *\n * @param packetLog : The packetlog from which the timestamp should be removed\n * @param timestamp : The timestamp to be removed\n * @returns true if timestamp was present in the log and removed, false otherwise\n */\n removeTimestamp(packetLog, timestamp) {\n const index = packetLog.window.indexOf(timestamp);\n if (index >= 0) {\n packetLog.window[index] = undefined;\n return true;\n }\n return false;\n }\n /**\n * Removes a timestamp from red and fec recovery windows.\n *\n * @param timestamp : The timestamp to be removed\n */\n removeFromRecoveryWindows(timestamp) {\n let removed = this.removeTimestamp(this.redRecoveryLog, timestamp);\n if (removed) {\n if (this.totalAudioPacketsRecoveredRed > 0)\n this.totalAudioPacketsRecoveredRed--;\n }\n removed = this.removeTimestamp(this.fecRecoveryLog, timestamp);\n if (removed) {\n if (this.totalAudioPacketsRecoveredFec > 0)\n this.totalAudioPacketsRecoveredFec--;\n }\n }\n /**\n * Converts the supplied argument to 32-bit unsigned integer\n */\n uint32WrapAround(num) {\n const mod = 4294967296; // 2^32\n let res = num;\n if (num >= mod) {\n res = num - mod;\n }\n else if (num < 0) {\n res = mod + num;\n }\n return res;\n }\n /**\n * Converts the supplied argument to 16-bit signed integer\n */\n int16(num) {\n return (num << 16) >> 16;\n }\n /**\n * Determines if an Opus packet is in CELT-only mode.\n *\n * @param packet Opus packet.\n * @returns `true` if the packet is in CELT-only mode.\n */\n opusPacketIsCeltOnly(packet) {\n // TOC byte format (https://www.rfc-editor.org/rfc/rfc6716#section-3.1):\n // 0\n // 0 1 2 3 4 5 6 7\n // +-+-+-+-+-+-+-+-+\n // | config |s| c |\n // +-+-+-+-+-+-+-+-+\n // Since CELT-only packets are represented using configurations 16 to 31, the highest 'config' bit will always be 1\n // for CELT-only packets.\n return !!(packet.getUint8(0) & 0x80);\n }\n /**\n * Gets the number of samples per frame from an Opus packet.\n *\n * @param packet Opus packet. This must contain at least one byte of data.\n * @param sampleRateHz 32-bit integer sampling rate in Hz. This must be a multiple of 400 or inaccurate results will\n * be returned.\n * @returns Number of samples per frame.\n */\n opusPacketGetSamplesPerFrame(packet, sampleRateHz) {\n // Sample rate must be a 32-bit integer.\n sampleRateHz = Math.round(sampleRateHz);\n sampleRateHz = Math.min(Math.max(sampleRateHz, -(Math.pow(2, 32))), Math.pow(2, 32) - 1);\n // TOC byte format (https://www.rfc-editor.org/rfc/rfc6716#section-3.1):\n // 0\n // 0 1 2 3 4 5 6 7\n // +-+-+-+-+-+-+-+-+\n // | config |s| c |\n // +-+-+-+-+-+-+-+-+\n let numSamples;\n let frameSizeOption;\n // Case for CELT-only packet.\n if (this.opusPacketIsCeltOnly(packet)) {\n // The lower 3 'config' bits indicate the frame size option.\n frameSizeOption = (packet.getUint8(0) >> 3) & 0x3;\n // The frame size options 0, 1, 2, 3 correspond to frame sizes of 2.5, 5, 10, 20 ms. Notice that the frame sizes\n // can be represented as (2.5 * 2^0), (2.5 * 2^1), (2.5 * 2^2), (2.5 * 2^3) ms. So, the number of samples can be\n // calculated as follows:\n // (sample/s) * (1s/1000ms) * (2.5ms) * 2^(frameSizeOption)\n // = (sample/s) * (1s/400) * 2^(frameSizeOption)\n // = (sample/s) * 2^(frameSizeOption) * (1s/400)\n numSamples = (sampleRateHz << frameSizeOption) / 400;\n }\n // Case for Hybrid packet. Since Hybrid packets are represented using configurations 12 to 15, bits 1 and 2 in the\n // above TOC byte diagram will both be 1.\n else if ((packet.getUint8(0) & 0x60) === 0x60) {\n // In the case of configuration 13 or 15, bit 4 in the above TOC byte diagram will be 1. Configurations 13 and 15\n // correspond to a 20ms frame size, so the number of samples is calculated as follows:\n // (sample/s) * (1s/1000ms) * (20ms)\n // = (sample/s) * (1s/50)\n //\n // In the case of configuration 12 or 14, bit 4 in the above TOC byte diagram will be 0. Configurations 12 and 14\n // correspond to a 10ms frame size, so the number of samples is calculated as follows:\n // (sample/s) * (1s/1000ms) * (10ms)\n // = (sample/s) * (1s/100)\n numSamples = packet.getUint8(0) & 0x08 ? sampleRateHz / 50 : sampleRateHz / 100;\n }\n // Case for SILK-only packet.\n else {\n // The lower 3 'config' bits indicate the frame size option for SILK-only packets.\n frameSizeOption = (packet.getUint8(0) >> 3) & 0x3;\n if (frameSizeOption === 3) {\n // Frame size option 3 corresponds to a frame size of 60ms, so the number of samples is calculated as follows:\n // (sample/s) * (1s/1000ms) * (60ms)\n // = (sample/s) * (60ms) * (1s/1000ms)\n numSamples = (sampleRateHz * 60) / 1000;\n }\n else {\n // The frame size options 0, 1, 2 correspond to frame sizes of 10, 20, 40 ms. Notice that the frame sizes can be\n // represented as (10 * 2^0), (10 * 2^1), (10 * 2^2) ms. So, the number of samples can be calculated as follows:\n // (sample/s) * (1s/1000ms) * (10ms) * 2^(frameSizeOption)\n // = (sample/s) * (1s/100) * 2^(frameSizeOption)\n // = (sample/s) * 2^(frameSizeOption) * (1s/100)\n numSamples = (sampleRateHz << frameSizeOption) / 100;\n }\n }\n return numSamples;\n }\n /**\n * Gets the number of SILK frames per Opus frame.\n *\n * @param packet Opus packet.\n * @returns Number of SILK frames per Opus frame.\n */\n opusNumSilkFrames(packet) {\n // For computing the frame length in ms, the sample rate is not important since it cancels out. We use 48 kHz, but\n // any valid sample rate would work.\n //\n // To calculate the length of a frame (with a 48kHz sample rate) in ms:\n // (samples/frame) * (1s/48000 samples) * (1000ms/s)\n // = (samples/frame) * (1000ms/48000 samples)\n // = (samples/frame) * (1ms/48 samples)\n let frameLengthMs = this.opusPacketGetSamplesPerFrame(packet, 48000) / 48;\n if (frameLengthMs < 10)\n frameLengthMs = 10;\n // The number of SILK frames per Opus frame is described in https://www.rfc-editor.org/rfc/rfc6716#section-4.2.2.\n switch (frameLengthMs) {\n case 10:\n case 20:\n return 1;\n case 40:\n return 2;\n case 60:\n return 3;\n // It is not possible to reach the default case since an Opus packet can only encode sizes of 2.5, 5, 10, 20, 40,\n // or 60 ms, so we ignore the default case for test coverage.\n /* istanbul ignore next */\n default:\n return 0;\n }\n }\n /**\n * Gets the number of channels from an Opus packet.\n *\n * @param packet Opus packet.\n * @returns Number of channels.\n */\n opusPacketGetNumChannels(packet) {\n // TOC byte format (https://www.rfc-editor.org/rfc/rfc6716#section-3.1):\n // 0\n // 0 1 2 3 4 5 6 7\n // +-+-+-+-+-+-+-+-+\n // | config |s| c |\n // +-+-+-+-+-+-+-+-+\n // The 's' bit indicates mono or stereo audio, with 0 indicating mono and 1 indicating stereo.\n return packet.getUint8(0) & 0x4 ? 2 : 1;\n }\n /**\n * Determine the size (in bytes) of an Opus frame.\n *\n * @param packet Opus packet.\n * @param byteOffset Offset (from the start of the packet) to the byte containing the size information.\n * @param remainingBytes Remaining number of bytes to parse from the Opus packet.\n * @param sizeBytes Variable to store the parsed frame size (in bytes).\n * @returns Number of bytes that were parsed to determine the frame size.\n */\n opusParseSize(packet, byteOffset, remainingBytes, sizeBytes) {\n // See https://www.rfc-editor.org/rfc/rfc6716#section-3.2.1 for an explanation of how frame size is represented.\n // If there are no remaining bytes to parse the size from, then the size cannot be determined.\n if (remainingBytes < 1) {\n sizeBytes[0] = -1;\n return -1;\n }\n // If the first byte is in the range 0...251, then this value is the size of the frame.\n else if (packet.getUint8(byteOffset) < 252) {\n sizeBytes[0] = packet.getUint8(byteOffset);\n return 1;\n }\n // If the first byte is in the range 252...255, a second byte is needed. If there is no second byte, then the size\n // cannot be determined.\n else if (remainingBytes < 2) {\n sizeBytes[0] = -1;\n return -1;\n }\n // The total size of the frame given two size bytes is:\n // (4 * secondSizeByte) + firstSizeByte\n else {\n sizeBytes[0] = 4 * packet.getUint8(byteOffset + 1) + packet.getUint8(byteOffset);\n return 2;\n }\n }\n /**\n * Parse binary data containing an Opus packet into one or more Opus frames.\n *\n * @param data Binary data containing an Opus packet to be parsed. The data should begin with the first byte (i.e the\n * TOC byte) of an Opus packet. Note that the size of the data does not have to equal the size of the\n * contained Opus packet.\n * @param lenBytes Size of the data (in bytes).\n * @param selfDelimited Indicates if the Opus packet is self-delimiting\n * (https://www.rfc-editor.org/rfc/rfc6716#appendix-B).\n * @param tocByte Optional variable to store the TOC (table of contents) byte.\n * @param frameOffsets Optional variable to store the offsets (from the start of the data) to the first bytes of each\n * Opus frame.\n * @param frameSizes Required variable to store the sizes (in bytes) of each Opus frame.\n * @param payloadOffset Optional variable to store the offset (from the start of the data) to the first byte of the\n * payload.\n * @param packetLenBytes Optional variable to store the length of the Opus packet (in bytes).\n * @returns Number of Opus frames.\n */\n opusPacketParseImpl(data, lenBytes, selfDelimited, tocByte, frameOffsets, frameSizes, payloadOffset, packetLenBytes) {\n if (!frameSizes || lenBytes < 0)\n return this.OPUS_BAD_ARG;\n if (lenBytes === 0)\n return this.OPUS_INVALID_PACKET;\n // The number of Opus frames in the packet.\n let numFrames;\n // Intermediate storage for the number of bytes parsed to determine the size of a frame.\n let numBytesParsed;\n // The number of the padding bytes (excluding the padding count bytes) in the packet.\n let paddingBytes = 0;\n // Indicates whether CBR (constant bitrate) framing is used.\n let cbr = false;\n // The TOC (table of contents) byte (https://www.rfc-editor.org/rfc/rfc6716#section-3.1).\n const toc = data.getUint8(0);\n // Store the TOC byte.\n if (tocByte)\n tocByte[0] = toc;\n // The remaining number of bytes to parse from the packet. Note that the TOC byte has already been parsed, hence the\n // minus 1.\n let remainingBytes = lenBytes - 1;\n // This keeps track of where we are in the packet. This starts at 1 since the TOC byte has already been read.\n let byteOffset = 1;\n // The size of the last Opus frame in bytes.\n let lastSizeBytes = remainingBytes;\n // Read the `c` bits (i.e. code bits) from the TOC byte.\n switch (toc & 0x3) {\n // A code 0 packet (https://www.rfc-editor.org/rfc/rfc6716#section-3.2.2) has one frame.\n case 0:\n numFrames = 1;\n break;\n // A code 1 packet (https://www.rfc-editor.org/rfc/rfc6716#section-3.2.3) has two CBR (constant bitrate) frames.\n case 1:\n numFrames = 2;\n cbr = true;\n if (!selfDelimited) {\n // Undelimited code 1 packets must be an even number of data bytes, otherwise the packet is invalid.\n if (remainingBytes & 0x1)\n return this.OPUS_INVALID_PACKET;\n // The sizes of both frames are equal (i.e. half of the number of data bytes).\n lastSizeBytes = remainingBytes / 2;\n // If `lastSizeBytes` is too large, we will catch it later.\n frameSizes[0][0] = lastSizeBytes;\n }\n break;\n // A code 2 packet (https://www.rfc-editor.org/rfc/rfc6716#section-3.2.4) has two VBR (variable bitrate) frames.\n case 2:\n numFrames = 2;\n numBytesParsed = this.opusParseSize(data, byteOffset, remainingBytes, frameSizes[0]);\n remainingBytes -= numBytesParsed;\n // The parsed size of the first frame cannot be larger than the number of remaining bytes in the packet.\n if (frameSizes[0][0] < 0 || frameSizes[0][0] > remainingBytes) {\n return this.OPUS_INVALID_PACKET;\n }\n byteOffset += numBytesParsed;\n // The size of the second frame is the remaining number of bytes after the first frame.\n lastSizeBytes = remainingBytes - frameSizes[0][0];\n break;\n // A code 3 packet (https://www.rfc-editor.org/rfc/rfc6716#section-3.2.5) has multiple CBR/VBR frames (from 0 to\n // 120 ms).\n default:\n // Code 3 packets must have at least 2 bytes (i.e. at least 1 byte after the TOC byte).\n if (remainingBytes < 1)\n return this.OPUS_INVALID_PACKET;\n // Frame count byte format:\n // 0\n // 0 1 2 3 4 5 6 7\n // +-+-+-+-+-+-+-+-+\n // |v|p| M |\n // +-+-+-+-+-+-+-+-+\n //\n // Read the frame count byte, which immediately follows the TOC byte.\n const frameCountByte = data.getUint8(byteOffset++);\n --remainingBytes;\n // Read the 'M' bits of the frame count byte, which encode the number of frames.\n numFrames = frameCountByte & 0x3f;\n // The number of frames in a code 3 packet must not be 0.\n if (numFrames <= 0)\n return this.OPUS_INVALID_PACKET;\n const samplesPerFrame = this.opusPacketGetSamplesPerFrame(data, 48000);\n // A single frame can have at most 2880 samples, which happens in the case where 60ms of 48kHz audio is encoded\n // per frame. A code 3 packet cannot contain more than 120ms of audio, so the total number of samples cannot\n // exceed 2880 * 2 = 5760.\n if (samplesPerFrame * numFrames > 5760)\n return this.OPUS_INVALID_PACKET;\n // Parse padding bytes if the 'p' bit is 1.\n if (frameCountByte & 0x40) {\n let paddingCountByte;\n let numPaddingBytes;\n // Remove padding bytes (including padding count bytes) from the remaining byte count.\n do {\n // Sanity check that there are enough bytes to parse and remove the padding.\n if (remainingBytes <= 0)\n return this.OPUS_INVALID_PACKET;\n // Get the next padding count byte.\n paddingCountByte = data.getUint8(byteOffset++);\n --remainingBytes;\n // If the padding count byte has a value in the range 0...254, then the total size of the padding is the\n // value in the padding count byte.\n //\n // If the padding count byte has value 255, then the total size of the padding is 254 plus the value in the\n // next padding count byte. Therefore, keep reading padding count bytes while the value is 255.\n numPaddingBytes = paddingCountByte === 255 ? 254 : paddingCountByte;\n remainingBytes -= numPaddingBytes;\n paddingBytes += numPaddingBytes;\n } while (paddingCountByte === 255);\n }\n // Sanity check that the remaining number of bytes is not negative after removing the padding.\n if (remainingBytes < 0)\n return this.OPUS_INVALID_PACKET;\n // Read the 'v' bit (i.e. VBR bit).\n cbr = !(frameCountByte & 0x80);\n // VBR case\n if (!cbr) {\n lastSizeBytes = remainingBytes;\n // Let M be the number of frames. There will be M - 1 frame length indicators (which can be 1 or 2 bytes)\n // corresponding to the lengths of frames 0 to M - 2. The size of the last frame (i.e. frame M - 1) is the\n // number of data bytes after the end of frame M - 2 and before the start of the padding bytes.\n for (let i = 0; i < numFrames - 1; ++i) {\n numBytesParsed = this.opusParseSize(data, byteOffset, remainingBytes, frameSizes[i]);\n remainingBytes -= numBytesParsed;\n // The remaining number of data bytes must be enough to contain each frame.\n if (frameSizes[i][0] < 0 || frameSizes[i][0] > remainingBytes) {\n return this.OPUS_INVALID_PACKET;\n }\n byteOffset += numBytesParsed;\n lastSizeBytes -= numBytesParsed + frameSizes[i][0];\n }\n // Sanity check that the size of the last frame is not negative.\n if (lastSizeBytes < 0)\n return this.OPUS_INVALID_PACKET;\n }\n // CBR case\n else if (!selfDelimited) {\n // The size of each frame is the number of data bytes divided by the number of frames.\n lastSizeBytes = Math.trunc(remainingBytes / numFrames);\n // The number of data bytes must be a non-negative integer multiple of the number of frames.\n if (lastSizeBytes * numFrames !== remainingBytes)\n return this.OPUS_INVALID_PACKET;\n // All frames have equal size in the undelimited CBR case.\n for (let i = 0; i < numFrames - 1; ++i) {\n frameSizes[i][0] = lastSizeBytes;\n }\n }\n }\n // Self-delimited framing uses an extra 1 or 2 bytes, immediately preceding the data bytes, to indicate either the\n // size of the last frame (for code 0, code 2, and VBR code 3 packets) or the size of all the frames (for code 1 and\n // CBR code 3 packets). See https://www.rfc-editor.org/rfc/rfc6716#appendix-B.\n if (selfDelimited) {\n // The extra frame size byte(s) will always indicate the size of the last frame.\n numBytesParsed = this.opusParseSize(data, byteOffset, remainingBytes, frameSizes[numFrames - 1]);\n remainingBytes -= numBytesParsed;\n // There must be enough data bytes for the last frame.\n if (frameSizes[numFrames - 1][0] < 0 || frameSizes[numFrames - 1][0] > remainingBytes) {\n return this.OPUS_INVALID_PACKET;\n }\n byteOffset += numBytesParsed;\n // For CBR packets, the sizes of all the frames are equal.\n if (cbr) {\n // There must be enough data bytes for all the frames.\n if (frameSizes[numFrames - 1][0] * numFrames > remainingBytes) {\n return this.OPUS_INVALID_PACKET;\n }\n for (let i = 0; i < numFrames - 1; ++i) {\n frameSizes[i][0] = frameSizes[numFrames - 1][0];\n }\n }\n // At this point, `lastSizeBytes` contains the size of the last frame plus the size of the extra frame size\n // byte(s), so sanity check that `lastSizeBytes` is the upper bound for the size of the last frame.\n else if (!(numBytesParsed + frameSizes[numFrames - 1][0] <= lastSizeBytes)) {\n return this.OPUS_INVALID_PACKET;\n }\n }\n // Undelimited case\n else {\n // Because the size of the last packet is not encoded explicitly, it is possible that the size of the last packet\n // (or of all the packets, for the CBR case) is larger than maximum frame size.\n if (lastSizeBytes > this.OPUS_MAX_FRAME_SIZE_BYTES)\n return this.OPUS_INVALID_PACKET;\n frameSizes[numFrames - 1][0] = lastSizeBytes;\n }\n // Store the offset to the start of the payload.\n if (payloadOffset)\n payloadOffset[0] = byteOffset;\n // Store the offsets to the start of each frame.\n for (let i = 0; i < numFrames; ++i) {\n if (frameOffsets)\n frameOffsets[i][0] = byteOffset;\n byteOffset += frameSizes[i][0];\n }\n // Store the length of the Opus packet.\n if (packetLenBytes)\n packetLenBytes[0] = byteOffset + paddingBytes;\n return numFrames;\n }\n /**\n * Parse a single undelimited Opus packet into one or more Opus frames.\n *\n * @param packet Opus packet to be parsed.\n * @param lenBytes Size of the packet (in bytes).\n * @param tocByte Optional variable to store the TOC (table of contents) byte.\n * @param frameOffsets Optional variable to store the offsets (from the start of the packet) to the first bytes of\n * each Opus frame.\n * @param frameSizes Required variable to store the sizes (in bytes) of each Opus frame.\n * @param payloadOffset Optional variable to store the offset (from the start of the packet) to the first byte of the\n * payload.\n * @returns Number of Opus frames.\n */\n opusPacketParse(packet, lenBytes, tocByte, frameOffsets, frameSizes, payloadOffset) {\n return this.opusPacketParseImpl(packet, lenBytes, \n /* selfDelimited */ false, tocByte, frameOffsets, frameSizes, payloadOffset, null);\n }\n /**\n * This function returns the SILK VAD (voice activity detection) information encoded in the Opus packet. For CELT-only\n * packets that do not have VAD information, it returns -1.\n *\n * @param packet Opus packet.\n * @param lenBytes Size of the packet (in bytes).\n * @returns 0: no frame had the VAD flag set.\n * 1: at least one frame had the VAD flag set.\n * -1: VAD status could not be determined.\n */\n opusPacketHasVoiceActivity(packet, lenBytes) {\n if (!packet || lenBytes <= 0)\n return 0;\n // In CELT-only mode, we can not determine whether there is VAD.\n if (this.opusPacketIsCeltOnly(packet))\n return -1;\n const numSilkFrames = this.opusNumSilkFrames(packet);\n // It is not possible for `opusNumSilkFrames()` to return 0, so we ignore the next sanity check for test coverage.\n /* istanbul ignore next */\n if (numSilkFrames === 0)\n return -1;\n const opusFrameOffsets = new Array(this.OPUS_MAX_OPUS_FRAMES);\n const opusFrameSizes = new Array(this.OPUS_MAX_OPUS_FRAMES);\n for (let i = 0; i < this.OPUS_MAX_OPUS_FRAMES; ++i) {\n opusFrameOffsets[i] = [undefined];\n opusFrameSizes[i] = [undefined];\n }\n // Parse packet to get the Opus frames.\n const numOpusFrames = this.opusPacketParse(packet, lenBytes, null, opusFrameOffsets, opusFrameSizes, null);\n // VAD status cannot be determined for invalid packets.\n if (numOpusFrames < 0)\n return -1;\n // Iterate over all Opus frames, which may contain multiple SILK frames, to determine the VAD status.\n for (let i = 0; i < numOpusFrames; ++i) {\n if (opusFrameSizes[i][0] < 1)\n continue;\n // LP layer header bits format (https://www.rfc-editor.org/rfc/rfc6716#section-4.2.3):\n //\n // Mono case:\n // +-----------------+----------+\n // | 1 to 3 VAD bits | LBRR bit |\n // +-----------------+----------+\n //\n // Stereo case:\n // +---------------------+--------------+----------------------+---------------+\n // | 1 to 3 mid VAD bits | mid LBRR bit | 1 to 3 side VAD bits | side LBRR bit |\n // +---------------------+--------------+----------------------+---------------+\n // The upper 1 to 3 bits (dependent on the number of SILK frames) of the LP layer contain VAD bits. If any of\n // these VAD bits are 1, then voice activity is present.\n if (packet.getUint8(opusFrameOffsets[i][0]) >> (8 - numSilkFrames))\n return 1;\n // In the stereo case, there is a second set of 1 to 3 VAD bits, so also check these VAD bits.\n const channels = this.opusPacketGetNumChannels(packet);\n if (channels === 2 &&\n (packet.getUint8(opusFrameOffsets[i][0]) << (numSilkFrames + 1)) >> (8 - numSilkFrames)) {\n return 1;\n }\n }\n // No voice activity was detected.\n return 0;\n }\n /**\n * This method is based on Definition of the Opus Audio Codec\n * (https://tools.ietf.org/html/rfc6716). Basically, this method is based on\n * parsing the LP layer of an Opus packet, particularly the LBRR flag.\n *\n * @param packet Opus packet.\n * @param lenBytes Size of the packet (in bytes).\n * @returns true: packet has fec encoding about previous packet.\n * false: no fec encoding present.\n */\n opusPacketHasFec(packet, lenBytes) {\n if (!packet || lenBytes <= 0)\n return false;\n // In CELT-only mode, packets should not have FEC.\n if (this.opusPacketIsCeltOnly(packet))\n return false;\n const opusFrameOffsets = new Array(this.OPUS_MAX_OPUS_FRAMES);\n const opusFrameSizes = new Array(this.OPUS_MAX_OPUS_FRAMES);\n for (let i = 0; i < this.OPUS_MAX_OPUS_FRAMES; ++i) {\n opusFrameOffsets[i] = [undefined];\n opusFrameSizes[i] = [undefined];\n }\n // Parse packet to get the Opus frames.\n const numOpusFrames = this.opusPacketParse(packet, lenBytes, null, opusFrameOffsets, opusFrameSizes, null);\n if (numOpusFrames < 0)\n return false;\n /* istanbul ignore next */\n if (opusFrameSizes[0][0] <= 1)\n return false;\n const numSilkFrames = this.opusNumSilkFrames(packet);\n /* istanbul ignore next */\n if (numSilkFrames === 0)\n return false;\n const channels = this.opusPacketGetNumChannels(packet);\n /* istanbul ignore next */\n if (channels !== 1 && channels !== 2)\n return false;\n // A frame starts with the LP layer. The LP layer begins with two to eight\n // header bits.These consist of one VAD bit per SILK frame (up to 3),\n // followed by a single flag indicating the presence of LBRR frames.\n // For a stereo packet, these first flags correspond to the mid channel, and\n // a second set of flags is included for the side channel. Because these are\n // the first symbols decoded by the range coder and because they are coded\n // as binary values with uniform probability, they can be extracted directly\n // from the most significant bits of the first byte of compressed data.\n for (let i = 0; i < channels; i++) {\n if (packet.getUint8(opusFrameOffsets[0][0]) & (0x80 >> ((i + 1) * (numSilkFrames + 1) - 1)))\n return true;\n }\n return false;\n }\n}\nRedundantAudioEncoder.shouldLog = true;\nRedundantAudioEncoder.shouldReportStats = true;\nRedundantAudioEncoder.initializeWorker();\n" = "class RedundantAudioEncoder {\n constructor() {\n // Each payload must be less than 1024 bytes to fit the 10 bit block length\n this.maxRedPacketSizeBytes = 1 << 10;\n // Limit payload to 1000 bytes to handle small MTU. 1000 is chosen because in Chromium-based browsers, writing audio\n // payloads larger than 1000 bytes using the WebRTC Insertable Streams API (which is used to enable dynamic audio\n // redundancy) will cause an error to be thrown and cause audio flow to permanently stop. See\n // https://crbug.com/1248479.\n this.maxAudioPayloadSizeBytes = 1000;\n // Each payload can encode a timestamp delta of 14 bits\n this.maxRedTimestampOffset = 1 << 14;\n // 4 byte RED header\n this.redHeaderSizeBytes = 4;\n // reduced size for last RED header\n this.redLastHeaderSizeBytes = 1;\n // P-Time for Opus 20 msec packets\n // We do not support other p-times or clock rates\n this.redPacketizationTime = 960;\n // distance between redundant payloads, Opus FEC handles a distance of 1\n // TODO(https://issues.amazon.com/issues/ChimeSDKAudio-55):\n // Consider making this dynamic\n this.redPacketDistance = 2;\n // maximum number of redundant payloads per RTP packet\n this.maxRedEncodings = 2;\n // Maximum number of encodings that can be recovered with a single RED packet, assuming the primary and redundant\n // payloads have FEC.\n this.redMaxRecoveryDistance = this.redPacketDistance * this.maxRedEncodings + 1;\n // maximum history of prior payloads to keep\n // generally we will expire old entries based on timestamp\n // this limit is in place just to make sure the history does not\n // grow too large in the case of erroneous timestamp inputs\n this.maxEncodingHistorySize = 10;\n // Current number of encodings we want to send\n // to the remote end. This will be dynamically\n // updated through the setNumEncodingsFromPacketloss API\n this.numRedundantEncodings = 0;\n // Used to enable or disable redundancy\n // in response to very high packet loss events\n this.redundancyEnabled = true;\n // Loss stats are reported to the main thread every 5 seconds.\n // Since timestamp differences between 2 consecutive packets\n // give us the number of samples in each channel, 1 second\n // is equivalent to 48000 samples:\n // P-time * (1000ms/1s)\n // = (960 samples/20ms) * (1000ms/1s)\n // = 48000 samples/s\n this.lossReportInterval = 48000 * 5;\n // Maximum distance of a packet from the most recent packet timestamp\n // that we will consider for recovery.\n this.maxOutOfOrderPacketDistance = 16;\n /**\n * Below are Opus helper methods and constants.\n */\n this.OPUS_BAD_ARG = -1;\n this.OPUS_INVALID_PACKET = -4;\n // Max number of Opus frames in an Opus packet is 48 (https://www.rfc-editor.org/rfc/rfc6716#section-3.2.5).\n this.OPUS_MAX_OPUS_FRAMES = 48;\n // Max number of bytes that any individual Opus frame can have.\n this.OPUS_MAX_FRAME_SIZE_BYTES = 1275;\n this.encodingHistory = new Array();\n this.opusPayloadType = 0;\n this.redPayloadType = 0;\n this.initializePacketLogs();\n }\n /**\n * Creates an instance of RedundantAudioEncoder and sets up callbacks.\n */\n static initializeWorker() {\n RedundantAudioEncoder.log('Initializing RedundantAudioEncoder');\n const encoder = new RedundantAudioEncoder();\n // RED encoding is done using WebRTC Encoded Transform\n // https://github.com/w3c/webrtc-encoded-transform/blob/main/explainer.md\n // Check the DedicatedWorkerGlobalScope for existence of\n // RTCRtpScriptTransformer interface. If exists, then\n // RTCRtpScriptTransform is supported by this browser.\n // @ts-ignore\n if (self.RTCRtpScriptTransformer) {\n // @ts-ignore\n self.onrtctransform = (event) => {\n if (event.transformer.options.type === 'SenderTransform') {\n encoder.setupSenderTransform(event.transformer.readable, event.transformer.writable);\n }\n else if (event.transformer.options.type === 'ReceiverTransform') {\n encoder.setupReceiverTransform(event.transformer.readable, event.transformer.writable);\n }\n else if (event.transformer.options.type === 'PassthroughTransform') {\n encoder.setupPassthroughTransform(event.transformer.readable, event.transformer.writable);\n }\n };\n }\n self.onmessage = (event) => {\n if (event.data.msgType === 'StartRedWorker') {\n encoder.setupSenderTransform(event.data.send.readable, event.data.send.writable);\n encoder.setupReceiverTransform(event.data.receive.readable, event.data.receive.writable);\n }\n else if (event.data.msgType === 'PassthroughTransform') {\n encoder.setupPassthroughTransform(event.data.send.readable, event.data.send.writable);\n encoder.setupPassthroughTransform(event.data.receive.readable, event.data.receive.writable);\n }\n else if (event.data.msgType === 'RedPayloadType') {\n encoder.setRedPayloadType(event.data.payloadType);\n }\n else if (event.data.msgType === 'OpusPayloadType') {\n encoder.setOpusPayloadType(event.data.payloadType);\n }\n else if (event.data.msgType === 'UpdateNumRedundantEncodings') {\n encoder.setNumRedundantEncodings(event.data.numRedundantEncodings);\n }\n else if (event.data.msgType === 'Enable') {\n encoder.setRedundancyEnabled(true);\n }\n else if (event.data.msgType === 'Disable') {\n encoder.setRedundancyEnabled(false);\n }\n };\n }\n /**\n * Post logs to the main thread\n */\n static log(msg) {\n if (RedundantAudioEncoder.shouldLog) {\n // @ts-ignore\n self.postMessage({\n type: 'REDWorkerLog',\n log: `[AudioRed] ${msg}`,\n });\n }\n }\n /**\n * Returns the number of encodings based on packetLoss value. This is used by `DefaultTransceiverController` to\n * determine when to alert the encoder to update the number of encodings. It also determines if we need to\n * turn off red in cases of very high packet loss to avoid congestion collapse.\n */\n static getNumRedundantEncodingsForPacketLoss(packetLoss) {\n let recommendedRedundantEncodings = 0;\n let shouldTurnOffRed = false;\n if (packetLoss <= 8) {\n recommendedRedundantEncodings = 0;\n }\n else if (packetLoss <= 18) {\n recommendedRedundantEncodings = 1;\n }\n else if (packetLoss <= 75) {\n recommendedRedundantEncodings = 2;\n }\n else {\n recommendedRedundantEncodings = 0;\n shouldTurnOffRed = true;\n }\n return [recommendedRedundantEncodings, shouldTurnOffRed];\n }\n /**\n * Sets up a passthrough (no-op) transform for the given streams.\n */\n setupPassthroughTransform(readable, writable) {\n RedundantAudioEncoder.log('Setting up passthrough transform');\n readable.pipeTo(writable);\n }\n /**\n * Sets up the transform stream and pipes the outgoing encoded audio frames through the transform function.\n */\n setupSenderTransform(readable, writable) {\n RedundantAudioEncoder.log('Setting up sender RED transform');\n const transformStream = new TransformStream({\n transform: this.senderTransform.bind(this),\n });\n readable.pipeThrough(transformStream).pipeTo(writable);\n return;\n }\n /**\n * Sets up the transform stream and pipes the received encoded audio frames through the transform function.\n */\n setupReceiverTransform(readable, writable) {\n RedundantAudioEncoder.log('Setting up receiver RED transform');\n const transformStream = new TransformStream({\n transform: this.receivePacketLogTransform.bind(this),\n });\n readable.pipeThrough(transformStream).pipeTo(writable);\n return;\n }\n /**\n * Set the RED payload type ideally obtained from local offer.\n */\n setRedPayloadType(payloadType) {\n this.redPayloadType = payloadType;\n RedundantAudioEncoder.log(`red payload type set to ${this.redPayloadType}`);\n }\n /**\n * Set the opus payload type ideally obtained from local offer.\n */\n setOpusPayloadType(payloadType) {\n this.opusPayloadType = payloadType;\n RedundantAudioEncoder.log(`opus payload type set to ${this.opusPayloadType}`);\n }\n /**\n * Set the number of redundant encodings\n */\n setNumRedundantEncodings(numRedundantEncodings) {\n this.numRedundantEncodings = numRedundantEncodings;\n if (this.numRedundantEncodings > this.maxRedEncodings) {\n this.numRedundantEncodings = this.maxRedEncodings;\n }\n RedundantAudioEncoder.log(`Updated numRedundantEncodings to ${this.numRedundantEncodings}`);\n }\n /**\n * Enable or disable redundancy in response to\n * high packet loss event.\n */\n setRedundancyEnabled(enabled) {\n this.redundancyEnabled = enabled;\n RedundantAudioEncoder.log(`redundancy ${this.redundancyEnabled ? 'enabled' : 'disabled'}`);\n }\n /**\n * Helper function to only enqueue audio frames if they do not exceed the audio payload byte limit imposed by\n * Chromium-based browsers. Chromium will throw an error (https://crbug.com/1248479) if an audio payload larger than\n * 1000 bytes is enqueued. Any controller that attempts to enqueue an audio payload larger than 1000 bytes will\n * encounter this error and will permanently stop sending or receiving audio.\n */\n enqueueAudioFrameIfPayloadSizeIsValid(\n // @ts-ignore\n frame, controller) {\n if (frame.data.byteLength > this.maxAudioPayloadSizeBytes)\n return;\n controller.enqueue(frame);\n }\n /**\n * Receives encoded frames and modifies as needed before sending to transport.\n */\n senderTransform(\n // @ts-ignore\n frame, controller) {\n const frameMetadata = frame.getMetadata();\n // @ts-ignore\n if (frameMetadata.payloadType !== this.redPayloadType) {\n this.enqueueAudioFrameIfPayloadSizeIsValid(frame, controller);\n return;\n }\n const primaryPayloadBuffer = this.getPrimaryPayload(frame.timestamp, frame.data);\n if (!primaryPayloadBuffer) {\n this.enqueueAudioFrameIfPayloadSizeIsValid(frame, controller);\n return;\n }\n const encodedBuffer = this.encode(frame.timestamp, primaryPayloadBuffer);\n /* istanbul ignore next */\n if (!encodedBuffer) {\n this.enqueueAudioFrameIfPayloadSizeIsValid(frame, controller);\n return;\n }\n frame.data = encodedBuffer;\n this.enqueueAudioFrameIfPayloadSizeIsValid(frame, controller);\n return;\n }\n /**\n * Get the primary payload from encoding\n */\n getPrimaryPayload(primaryTimestamp, frame) {\n const encodings = this.splitEncodings(primaryTimestamp, frame);\n if (!encodings || encodings.length < 1)\n return null;\n return encodings[encodings.length - 1].payload;\n }\n /**\n * Split up the encoding received into primary and redundant encodings\n * These will be ordered oldest to newest which is the same ordering\n * in the RTP red payload.\n */\n splitEncodings(primaryTimestamp, frame, getFecInfo = false, primarySequenceNumber = undefined) {\n // process RED headers (according to RFC 2198)\n // 0 1 2 3\n // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1\n // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+\n // |F| block PT | timestamp offset | block length |\n // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+\n //\n // last header\n // 0 1 2 3 4 5 6 7\n // +-+-+-+-+-+-+-+-+\n // |0| Block PT |\n // +-+-+-+-+-+-+-+-+\n const payload = new DataView(frame);\n let payloadSizeBytes = payload.byteLength;\n let totalPayloadSizeBytes = 0;\n let totalHeaderSizeBytes = 0;\n let primaryPayloadSizeBytes = 0;\n let payloadOffset = 0;\n let gotLastBlock = false;\n const encodings = new Array();\n const redundantEncodingBlockLengths = new Array();\n const redundantEncodingTimestamps = new Array();\n while (payloadSizeBytes > 0) {\n gotLastBlock = (payload.getUint8(payloadOffset) & 0x80) === 0;\n if (gotLastBlock) {\n // Bits 1 through 7 are payload type\n const payloadType = payload.getUint8(payloadOffset) & 0x7f;\n // Unexpected payload type. This is a bad packet.\n if (payloadType !== this.opusPayloadType) {\n return null;\n }\n totalPayloadSizeBytes += this.redLastHeaderSizeBytes;\n totalHeaderSizeBytes += this.redLastHeaderSizeBytes;\n // Accumulated block lengths are equal to or larger than the buffer, which means there is no primary block. This\n // is a bad packet.\n if (totalPayloadSizeBytes >= payload.byteLength) {\n return null;\n }\n primaryPayloadSizeBytes = payload.byteLength - totalPayloadSizeBytes;\n break;\n }\n else {\n if (payloadSizeBytes < this.redHeaderSizeBytes) {\n return null;\n }\n // Bits 22 through 31 are payload length\n const blockLength = ((payload.getUint8(payloadOffset + 2) & 0x03) << 8) + payload.getUint8(payloadOffset + 3);\n redundantEncodingBlockLengths.push(blockLength);\n const timestampOffset = payload.getUint16(payloadOffset + 1) >> 2;\n const timestamp = primaryTimestamp - timestampOffset;\n redundantEncodingTimestamps.push(timestamp);\n totalPayloadSizeBytes += blockLength + this.redHeaderSizeBytes;\n totalHeaderSizeBytes += this.redHeaderSizeBytes;\n payloadOffset += this.redHeaderSizeBytes;\n payloadSizeBytes -= this.redHeaderSizeBytes;\n }\n }\n // The last block was never found. The packet we received\n // does not have a good RED payload.\n if (!gotLastBlock) {\n // Note that sequence numbers only exist for\n // incoming audio frames.\n if (primarySequenceNumber !== undefined) {\n // This could be a possible padding packet used\n // for BWE with a good sequence number.\n // Create a dummy encoding to make sure loss values\n // are calculated correctly by consuming sequence number.\n // Note that for the receive side, we process packets only\n // for loss/recovery calculations and forward the original\n // packet without changing it even in the error case.\n encodings.push({\n payload: frame,\n isRedundant: false,\n seq: primarySequenceNumber,\n });\n return encodings;\n }\n // This is a bad packet.\n return null;\n }\n let redundantPayloadOffset = totalHeaderSizeBytes;\n for (let i = 0; i < redundantEncodingTimestamps.length; i++) {\n const redundantPayloadBuffer = new ArrayBuffer(redundantEncodingBlockLengths[i]);\n const redundantPayloadArray = new Uint8Array(redundantPayloadBuffer);\n redundantPayloadArray.set(new Uint8Array(payload.buffer, redundantPayloadOffset, redundantEncodingBlockLengths[i]), 0);\n const encoding = {\n timestamp: redundantEncodingTimestamps[i],\n payload: redundantPayloadBuffer,\n isRedundant: true,\n };\n if (getFecInfo) {\n encoding.hasFec = this.opusPacketHasFec(new DataView(redundantPayloadBuffer), redundantPayloadBuffer.byteLength);\n }\n encodings.push(encoding);\n redundantPayloadOffset += redundantEncodingBlockLengths[i];\n }\n const primaryPayloadOffset = payload.byteLength - primaryPayloadSizeBytes;\n const primaryPayloadBuffer = new ArrayBuffer(primaryPayloadSizeBytes);\n const primaryArray = new Uint8Array(primaryPayloadBuffer);\n primaryArray.set(new Uint8Array(payload.buffer, primaryPayloadOffset, primaryPayloadSizeBytes), 0);\n const encoding = {\n timestamp: primaryTimestamp,\n payload: primaryPayloadBuffer,\n isRedundant: false,\n seq: primarySequenceNumber,\n };\n if (getFecInfo) {\n encoding.hasFec = this.opusPacketHasFec(new DataView(primaryPayloadBuffer), primaryPayloadBuffer.byteLength);\n }\n encodings.push(encoding);\n return encodings;\n }\n /**\n * Create a new encoding with current primary payload and the older payloads of choice.\n */\n encode(primaryTimestamp, primaryPayload) {\n const primaryPayloadSize = primaryPayload.byteLength;\n // Payload size needs to be valid.\n if (primaryPayloadSize === 0 ||\n primaryPayloadSize >= this.maxRedPacketSizeBytes ||\n primaryPayloadSize >= this.maxAudioPayloadSizeBytes) {\n return null;\n }\n const numRedundantEncodings = this.numRedundantEncodings;\n let headerSizeBytes = this.redLastHeaderSizeBytes;\n let payloadSizeBytes = primaryPayloadSize;\n let bytesAvailable = this.maxAudioPayloadSizeBytes - primaryPayloadSize - headerSizeBytes;\n const redundantEncodingTimestamps = new Array();\n const redundantEncodingPayloads = new Array();\n // If redundancy is disabled then only send the primary payload\n if (this.redundancyEnabled) {\n // Determine how much redundancy we can fit into our packet\n let redundantTimestamp = this.uint32WrapAround(primaryTimestamp - this.redPacketizationTime * this.redPacketDistance);\n for (let i = 0; i < numRedundantEncodings; i++) {\n // Do not add redundant encodings that are beyond the maximum timestamp offset.\n if (this.uint32WrapAround(primaryTimestamp - redundantTimestamp) >= this.maxRedTimestampOffset) {\n break;\n }\n let findTimestamp = redundantTimestamp;\n let encoding = this.encodingHistory.find(e => e.timestamp === findTimestamp);\n if (!encoding) {\n // If not found or not important then look for the previous packet.\n // The current packet may have included FEC for the previous, so just\n // use the previous packet instead provided that it has voice activity.\n findTimestamp = this.uint32WrapAround(redundantTimestamp - this.redPacketizationTime);\n encoding = this.encodingHistory.find(e => e.timestamp === findTimestamp);\n }\n if (encoding) {\n const redundantEncodingSizeBytes = encoding.payload.byteLength;\n // Only add redundancy if there are enough bytes available.\n if (bytesAvailable < this.redHeaderSizeBytes + redundantEncodingSizeBytes)\n break;\n bytesAvailable -= this.redHeaderSizeBytes + redundantEncodingSizeBytes;\n headerSizeBytes += this.redHeaderSizeBytes;\n payloadSizeBytes += redundantEncodingSizeBytes;\n redundantEncodingTimestamps.unshift(encoding.timestamp);\n redundantEncodingPayloads.unshift(encoding.payload);\n }\n redundantTimestamp -= this.redPacketizationTime * this.redPacketDistance;\n redundantTimestamp = this.uint32WrapAround(redundantTimestamp);\n }\n }\n const redPayloadBuffer = new ArrayBuffer(headerSizeBytes + payloadSizeBytes);\n const redPayloadView = new DataView(redPayloadBuffer);\n // Add redundant encoding header(s) to new buffer\n let redPayloadOffset = 0;\n for (let i = 0; i < redundantEncodingTimestamps.length; i++) {\n const timestampDelta = primaryTimestamp - redundantEncodingTimestamps[i];\n redPayloadView.setUint8(redPayloadOffset, this.opusPayloadType | 0x80);\n redPayloadView.setUint16(redPayloadOffset + 1, (timestampDelta << 2) | (redundantEncodingPayloads[i].byteLength >> 8));\n redPayloadView.setUint8(redPayloadOffset + 3, redundantEncodingPayloads[i].byteLength & 0xff);\n redPayloadOffset += this.redHeaderSizeBytes;\n }\n // Add primary encoding header to new buffer\n redPayloadView.setUint8(redPayloadOffset, this.opusPayloadType);\n redPayloadOffset += this.redLastHeaderSizeBytes;\n // Add redundant payload(s) to new buffer\n const redPayloadArray = new Uint8Array(redPayloadBuffer);\n for (let i = 0; i < redundantEncodingPayloads.length; i++) {\n redPayloadArray.set(new Uint8Array(redundantEncodingPayloads[i]), redPayloadOffset);\n redPayloadOffset += redundantEncodingPayloads[i].byteLength;\n }\n // Add primary payload to new buffer\n redPayloadArray.set(new Uint8Array(primaryPayload), redPayloadOffset);\n redPayloadOffset += primaryPayload.byteLength;\n /* istanbul ignore next */\n // Sanity check that we got the expected total payload size.\n if (redPayloadOffset !== headerSizeBytes + payloadSizeBytes)\n return null;\n this.updateEncodingHistory(primaryTimestamp, primaryPayload);\n return redPayloadBuffer;\n }\n /**\n * Update the encoding history with the latest primary encoding\n */\n updateEncodingHistory(primaryTimestamp, primaryPayload) {\n // Remove encodings from the history if they are too old.\n for (const encoding of this.encodingHistory) {\n const maxTimestampDelta = this.redPacketizationTime * this.redMaxRecoveryDistance;\n if (primaryTimestamp - encoding.timestamp >= maxTimestampDelta) {\n this.encodingHistory.shift();\n }\n else {\n break;\n }\n }\n // Only add an encoding to the history if the encoding is deemed to be important. An encoding is important if it is\n // a CELT-only packet or contains voice activity.\n const packet = new DataView(primaryPayload);\n if (this.opusPacketIsCeltOnly(packet) ||\n this.opusPacketHasVoiceActivity(packet, packet.byteLength) > 0) {\n // Check if adding an encoding will cause the length of the encoding history to exceed the maximum history size.\n // This is not expected to happen but could occur if we get incorrect timestamps. We want to make sure our memory\n // usage is bounded. In this case, just clear the history and start over from empty.\n if (this.encodingHistory.length + 1 > this.maxEncodingHistorySize)\n this.encodingHistory.length = 0;\n this.encodingHistory.push({ timestamp: primaryTimestamp, payload: primaryPayload });\n }\n }\n /**\n * Initialize packet logs and metric values.\n */\n initializePacketLogs() {\n // The extra space from the max RED recovery distance is to ensure that we do not incorrectly count recovery for\n // packets that have already been received but are outside of the max out-of-order distance.\n const packetLogSize = this.maxOutOfOrderPacketDistance + this.redMaxRecoveryDistance;\n this.primaryPacketLog = {\n window: new Array(packetLogSize),\n index: 0,\n windowSize: packetLogSize,\n };\n this.redRecoveryLog = {\n window: new Array(packetLogSize),\n index: 0,\n windowSize: packetLogSize,\n };\n this.fecRecoveryLog = {\n window: new Array(packetLogSize),\n index: 0,\n windowSize: packetLogSize,\n };\n this.totalAudioPacketsExpected = 0;\n this.totalAudioPacketsLost = 0;\n this.totalAudioPacketsRecoveredRed = 0;\n this.totalAudioPacketsRecoveredFec = 0;\n }\n /**\n * Receives encoded frames from the server\n * and adds the timestamps to a packet log\n * to calculate an approximate recovery metric.\n */\n receivePacketLogTransform(\n // @ts-ignore\n frame, controller) {\n const frameMetadata = frame.getMetadata();\n // @ts-ignore\n if (frameMetadata.payloadType !== this.redPayloadType) {\n this.enqueueAudioFrameIfPayloadSizeIsValid(frame, controller);\n return;\n }\n // @ts-ignore\n const encodings = this.splitEncodings(frame.timestamp, frame.data, \n /*getFecInfo*/ true, frameMetadata.sequenceNumber);\n if (!encodings) {\n this.enqueueAudioFrameIfPayloadSizeIsValid(frame, controller);\n return;\n }\n for (let i = encodings.length - 1; i >= 0; i--) {\n if (this.updateLossStats(encodings[i])) {\n this.updateRedStats(encodings[i]);\n this.updateFecStats(encodings[i]);\n }\n }\n this.maybeReportLossStats(frameMetadata.synchronizationSource, encodings[encodings.length - 1].timestamp);\n this.enqueueAudioFrameIfPayloadSizeIsValid(frame, controller);\n }\n /**\n * Adds a timestamp to the primary packet log.\n * This also updates totalAudioPacketsLost and totalAudioPacketsExpected by looking\n * at the difference between timestamps.\n *\n * @param encoding : The encoding to be analyzed\n * @returns false if sequence number was greater than max out of order distance\n * true otherwise\n */\n updateLossStats(encoding) {\n if (encoding.isRedundant)\n return true;\n const timestamp = encoding.timestamp;\n const seq = encoding.seq;\n if (this.totalAudioPacketsExpected === 0) {\n this.totalAudioPacketsExpected = 1;\n this.newestSequenceNumber = seq;\n this.addTimestamp(this.primaryPacketLog, timestamp);\n return true;\n }\n const diff = this.int16(seq - this.newestSequenceNumber);\n if (diff < -this.maxOutOfOrderPacketDistance)\n return false;\n if (diff < 0) {\n if (!this.hasTimestamp(this.primaryPacketLog, timestamp)) {\n if (this.totalAudioPacketsLost > 0)\n this.totalAudioPacketsLost--;\n this.addTimestamp(this.primaryPacketLog, timestamp);\n this.removeFromRecoveryWindows(timestamp);\n }\n }\n else if (diff > 1) {\n this.totalAudioPacketsLost += diff - 1;\n }\n if (diff > 0) {\n this.totalAudioPacketsExpected += diff;\n this.newestSequenceNumber = encoding.seq;\n this.addTimestamp(this.primaryPacketLog, timestamp);\n }\n return true;\n }\n /**\n * Adds a timestamp to the red recovery log if it is not present in\n * the primary packet log and if it's not too old.\n *\n * @param encoding : The encoding to be analyzed\n */\n updateRedStats(encoding) {\n if (!encoding.isRedundant || this.totalAudioPacketsLost === 0)\n return;\n const timestamp = encoding.timestamp;\n if (!this.hasTimestamp(this.primaryPacketLog, timestamp)) {\n if (!this.hasTimestamp(this.redRecoveryLog, timestamp)) {\n this.totalAudioPacketsRecoveredRed++;\n this.addTimestamp(this.redRecoveryLog, timestamp);\n }\n if (this.removeTimestamp(this.fecRecoveryLog, timestamp)) {\n /* istanbul ignore else */\n if (this.totalAudioPacketsRecoveredFec > 0)\n this.totalAudioPacketsRecoveredFec--;\n }\n }\n }\n /**\n * Adds a timestamp to the fec recovery log if it is not present in\n * the primary packet log and red recovery log and if it is not too old.\n *\n * @param encoding : The encoding to be analyzed\n */\n updateFecStats(encoding) {\n if (!encoding.hasFec || this.totalAudioPacketsLost === 0)\n return;\n const fecTimestamp = encoding.timestamp - this.redPacketizationTime;\n if (this.hasTimestamp(this.primaryPacketLog, fecTimestamp) ||\n this.hasTimestamp(this.redRecoveryLog, fecTimestamp) ||\n this.hasTimestamp(this.fecRecoveryLog, fecTimestamp)) {\n return;\n }\n this.totalAudioPacketsRecoveredFec++;\n this.addTimestamp(this.fecRecoveryLog, fecTimestamp);\n }\n /**\n * Reports loss metrics to DefaultTransceiverController\n *\n * @param timestamp : Timestamp of most recent primary packet\n */\n maybeReportLossStats(ssrc, timestamp) {\n if (timestamp === undefined ||\n timestamp - this.lastLossReportTimestamp < this.lossReportInterval)\n return;\n /* istanbul ignore next */\n if (RedundantAudioEncoder.shouldReportStats) {\n // @ts-ignore\n self.postMessage({\n type: 'RedundantAudioEncoderStats',\n ssrc,\n totalAudioPacketsLost: this.totalAudioPacketsLost,\n totalAudioPacketsExpected: this.totalAudioPacketsExpected,\n totalAudioPacketsRecoveredRed: this.totalAudioPacketsRecoveredRed,\n totalAudioPacketsRecoveredFec: this.totalAudioPacketsRecoveredFec,\n });\n }\n this.lastLossReportTimestamp = timestamp;\n }\n /**\n * Adds a timestamp to a packet log\n *\n * @param packetLog : The packetlog to add the timestamp to\n * @param timestamp : The timestamp that should be added\n */\n addTimestamp(packetLog, timestamp) {\n if (timestamp === undefined) {\n return;\n }\n packetLog.window[packetLog.index] = timestamp;\n packetLog.index = (packetLog.index + 1) % packetLog.windowSize;\n }\n /**\n * Checks if a timestamp is in a packetlog\n *\n * @param packetLog : The packetlog to search\n * @param timestamp : The timestamp to search for\n * @returns true if timestamp is present, false otherwise\n */\n hasTimestamp(packetLog, timestamp) {\n const element = packetLog.window.find(t => t === timestamp);\n return !!element;\n }\n /**\n * Removes a timestamp from a packet log\n *\n * @param packetLog : The packetlog from which the timestamp should be removed\n * @param timestamp : The timestamp to be removed\n * @returns true if timestamp was present in the log and removed, false otherwise\n */\n removeTimestamp(packetLog, timestamp) {\n const index = packetLog.window.indexOf(timestamp);\n if (index >= 0) {\n packetLog.window[index] = undefined;\n return true;\n }\n return false;\n }\n /**\n * Removes a timestamp from red and fec recovery windows.\n *\n * @param timestamp : The timestamp to be removed\n */\n removeFromRecoveryWindows(timestamp) {\n let removed = this.removeTimestamp(this.redRecoveryLog, timestamp);\n if (removed) {\n if (this.totalAudioPacketsRecoveredRed > 0)\n this.totalAudioPacketsRecoveredRed--;\n }\n removed = this.removeTimestamp(this.fecRecoveryLog, timestamp);\n if (removed) {\n if (this.totalAudioPacketsRecoveredFec > 0)\n this.totalAudioPacketsRecoveredFec--;\n }\n }\n /**\n * Converts the supplied argument to 32-bit unsigned integer\n */\n uint32WrapAround(num) {\n const mod = 4294967296; // 2^32\n let res = num;\n if (num >= mod) {\n res = num - mod;\n }\n else if (num < 0) {\n res = mod + num;\n }\n return res;\n }\n /**\n * Converts the supplied argument to 16-bit signed integer\n */\n int16(num) {\n return (num << 16) >> 16;\n }\n /**\n * Determines if an Opus packet is in CELT-only mode.\n *\n * @param packet Opus packet.\n * @returns `true` if the packet is in CELT-only mode.\n */\n opusPacketIsCeltOnly(packet) {\n // TOC byte format (https://www.rfc-editor.org/rfc/rfc6716#section-3.1):\n // 0\n // 0 1 2 3 4 5 6 7\n // +-+-+-+-+-+-+-+-+\n // | config |s| c |\n // +-+-+-+-+-+-+-+-+\n // Since CELT-only packets are represented using configurations 16 to 31, the highest 'config' bit will always be 1\n // for CELT-only packets.\n return !!(packet.getUint8(0) & 0x80);\n }\n /**\n * Gets the number of samples per frame from an Opus packet.\n *\n * @param packet Opus packet. This must contain at least one byte of data.\n * @param sampleRateHz 32-bit integer sampling rate in Hz. This must be a multiple of 400 or inaccurate results will\n * be returned.\n * @returns Number of samples per frame.\n */\n opusPacketGetSamplesPerFrame(packet, sampleRateHz) {\n // Sample rate must be a 32-bit integer.\n sampleRateHz = Math.round(sampleRateHz);\n sampleRateHz = Math.min(Math.max(sampleRateHz, -(Math.pow(2, 32))), Math.pow(2, 32) - 1);\n // TOC byte format (https://www.rfc-editor.org/rfc/rfc6716#section-3.1):\n // 0\n // 0 1 2 3 4 5 6 7\n // +-+-+-+-+-+-+-+-+\n // | config |s| c |\n // +-+-+-+-+-+-+-+-+\n let numSamples;\n let frameSizeOption;\n // Case for CELT-only packet.\n if (this.opusPacketIsCeltOnly(packet)) {\n // The lower 3 'config' bits indicate the frame size option.\n frameSizeOption = (packet.getUint8(0) >> 3) & 0x3;\n // The frame size options 0, 1, 2, 3 correspond to frame sizes of 2.5, 5, 10, 20 ms. Notice that the frame sizes\n // can be represented as (2.5 * 2^0), (2.5 * 2^1), (2.5 * 2^2), (2.5 * 2^3) ms. So, the number of samples can be\n // calculated as follows:\n // (sample/s) * (1s/1000ms) * (2.5ms) * 2^(frameSizeOption)\n // = (sample/s) * (1s/400) * 2^(frameSizeOption)\n // = (sample/s) * 2^(frameSizeOption) * (1s/400)\n numSamples = (sampleRateHz << frameSizeOption) / 400;\n }\n // Case for Hybrid packet. Since Hybrid packets are represented using configurations 12 to 15, bits 1 and 2 in the\n // above TOC byte diagram will both be 1.\n else if ((packet.getUint8(0) & 0x60) === 0x60) {\n // In the case of configuration 13 or 15, bit 4 in the above TOC byte diagram will be 1. Configurations 13 and 15\n // correspond to a 20ms frame size, so the number of samples is calculated as follows:\n // (sample/s) * (1s/1000ms) * (20ms)\n // = (sample/s) * (1s/50)\n //\n // In the case of configuration 12 or 14, bit 4 in the above TOC byte diagram will be 0. Configurations 12 and 14\n // correspond to a 10ms frame size, so the number of samples is calculated as follows:\n // (sample/s) * (1s/1000ms) * (10ms)\n // = (sample/s) * (1s/100)\n numSamples = packet.getUint8(0) & 0x08 ? sampleRateHz / 50 : sampleRateHz / 100;\n }\n // Case for SILK-only packet.\n else {\n // The lower 3 'config' bits indicate the frame size option for SILK-only packets.\n frameSizeOption = (packet.getUint8(0) >> 3) & 0x3;\n if (frameSizeOption === 3) {\n // Frame size option 3 corresponds to a frame size of 60ms, so the number of samples is calculated as follows:\n // (sample/s) * (1s/1000ms) * (60ms)\n // = (sample/s) * (60ms) * (1s/1000ms)\n numSamples = (sampleRateHz * 60) / 1000;\n }\n else {\n // The frame size options 0, 1, 2 correspond to frame sizes of 10, 20, 40 ms. Notice that the frame sizes can be\n // represented as (10 * 2^0), (10 * 2^1), (10 * 2^2) ms. So, the number of samples can be calculated as follows:\n // (sample/s) * (1s/1000ms) * (10ms) * 2^(frameSizeOption)\n // = (sample/s) * (1s/100) * 2^(frameSizeOption)\n // = (sample/s) * 2^(frameSizeOption) * (1s/100)\n numSamples = (sampleRateHz << frameSizeOption) / 100;\n }\n }\n return numSamples;\n }\n /**\n * Gets the number of SILK frames per Opus frame.\n *\n * @param packet Opus packet.\n * @returns Number of SILK frames per Opus frame.\n */\n opusNumSilkFrames(packet) {\n // For computing the frame length in ms, the sample rate is not important since it cancels out. We use 48 kHz, but\n // any valid sample rate would work.\n //\n // To calculate the length of a frame (with a 48kHz sample rate) in ms:\n // (samples/frame) * (1s/48000 samples) * (1000ms/s)\n // = (samples/frame) * (1000ms/48000 samples)\n // = (samples/frame) * (1ms/48 samples)\n let frameLengthMs = this.opusPacketGetSamplesPerFrame(packet, 48000) / 48;\n if (frameLengthMs < 10)\n frameLengthMs = 10;\n // The number of SILK frames per Opus frame is described in https://www.rfc-editor.org/rfc/rfc6716#section-4.2.2.\n switch (frameLengthMs) {\n case 10:\n case 20:\n return 1;\n case 40:\n return 2;\n case 60:\n return 3;\n // It is not possible to reach the default case since an Opus packet can only encode sizes of 2.5, 5, 10, 20, 40,\n // or 60 ms, so we ignore the default case for test coverage.\n /* istanbul ignore next */\n default:\n return 0;\n }\n }\n /**\n * Gets the number of channels from an Opus packet.\n *\n * @param packet Opus packet.\n * @returns Number of channels.\n */\n opusPacketGetNumChannels(packet) {\n // TOC byte format (https://www.rfc-editor.org/rfc/rfc6716#section-3.1):\n // 0\n // 0 1 2 3 4 5 6 7\n // +-+-+-+-+-+-+-+-+\n // | config |s| c |\n // +-+-+-+-+-+-+-+-+\n // The 's' bit indicates mono or stereo audio, with 0 indicating mono and 1 indicating stereo.\n return packet.getUint8(0) & 0x4 ? 2 : 1;\n }\n /**\n * Determine the size (in bytes) of an Opus frame.\n *\n * @param packet Opus packet.\n * @param byteOffset Offset (from the start of the packet) to the byte containing the size information.\n * @param remainingBytes Remaining number of bytes to parse from the Opus packet.\n * @param sizeBytes Variable to store the parsed frame size (in bytes).\n * @returns Number of bytes that were parsed to determine the frame size.\n */\n opusParseSize(packet, byteOffset, remainingBytes, sizeBytes) {\n // See https://www.rfc-editor.org/rfc/rfc6716#section-3.2.1 for an explanation of how frame size is represented.\n // If there are no remaining bytes to parse the size from, then the size cannot be determined.\n if (remainingBytes < 1) {\n sizeBytes[0] = -1;\n return -1;\n }\n // If the first byte is in the range 0...251, then this value is the size of the frame.\n else if (packet.getUint8(byteOffset) < 252) {\n sizeBytes[0] = packet.getUint8(byteOffset);\n return 1;\n }\n // If the first byte is in the range 252...255, a second byte is needed. If there is no second byte, then the size\n // cannot be determined.\n else if (remainingBytes < 2) {\n sizeBytes[0] = -1;\n return -1;\n }\n // The total size of the frame given two size bytes is:\n // (4 * secondSizeByte) + firstSizeByte\n else {\n sizeBytes[0] = 4 * packet.getUint8(byteOffset + 1) + packet.getUint8(byteOffset);\n return 2;\n }\n }\n /**\n * Parse binary data containing an Opus packet into one or more Opus frames.\n *\n * @param data Binary data containing an Opus packet to be parsed. The data should begin with the first byte (i.e the\n * TOC byte) of an Opus packet. Note that the size of the data does not have to equal the size of the\n * contained Opus packet.\n * @param lenBytes Size of the data (in bytes).\n * @param selfDelimited Indicates if the Opus packet is self-delimiting\n * (https://www.rfc-editor.org/rfc/rfc6716#appendix-B).\n * @param tocByte Optional variable to store the TOC (table of contents) byte.\n * @param frameOffsets Optional variable to store the offsets (from the start of the data) to the first bytes of each\n * Opus frame.\n * @param frameSizes Required variable to store the sizes (in bytes) of each Opus frame.\n * @param payloadOffset Optional variable to store the offset (from the start of the data) to the first byte of the\n * payload.\n * @param packetLenBytes Optional variable to store the length of the Opus packet (in bytes).\n * @returns Number of Opus frames.\n */\n opusPacketParseImpl(data, lenBytes, selfDelimited, tocByte, frameOffsets, frameSizes, payloadOffset, packetLenBytes) {\n if (!frameSizes || lenBytes < 0)\n return this.OPUS_BAD_ARG;\n if (lenBytes === 0)\n return this.OPUS_INVALID_PACKET;\n // The number of Opus frames in the packet.\n let numFrames;\n // Intermediate storage for the number of bytes parsed to determine the size of a frame.\n let numBytesParsed;\n // The number of the padding bytes (excluding the padding count bytes) in the packet.\n let paddingBytes = 0;\n // Indicates whether CBR (constant bitrate) framing is used.\n let cbr = false;\n // The TOC (table of contents) byte (https://www.rfc-editor.org/rfc/rfc6716#section-3.1).\n const toc = data.getUint8(0);\n // Store the TOC byte.\n if (tocByte)\n tocByte[0] = toc;\n // The remaining number of bytes to parse from the packet. Note that the TOC byte has already been parsed, hence the\n // minus 1.\n let remainingBytes = lenBytes - 1;\n // This keeps track of where we are in the packet. This starts at 1 since the TOC byte has already been read.\n let byteOffset = 1;\n // The size of the last Opus frame in bytes.\n let lastSizeBytes = remainingBytes;\n // Read the `c` bits (i.e. code bits) from the TOC byte.\n switch (toc & 0x3) {\n // A code 0 packet (https://www.rfc-editor.org/rfc/rfc6716#section-3.2.2) has one frame.\n case 0:\n numFrames = 1;\n break;\n // A code 1 packet (https://www.rfc-editor.org/rfc/rfc6716#section-3.2.3) has two CBR (constant bitrate) frames.\n case 1:\n numFrames = 2;\n cbr = true;\n if (!selfDelimited) {\n // Undelimited code 1 packets must be an even number of data bytes, otherwise the packet is invalid.\n if (remainingBytes & 0x1)\n return this.OPUS_INVALID_PACKET;\n // The sizes of both frames are equal (i.e. half of the number of data bytes).\n lastSizeBytes = remainingBytes / 2;\n // If `lastSizeBytes` is too large, we will catch it later.\n frameSizes[0][0] = lastSizeBytes;\n }\n break;\n // A code 2 packet (https://www.rfc-editor.org/rfc/rfc6716#section-3.2.4) has two VBR (variable bitrate) frames.\n case 2:\n numFrames = 2;\n numBytesParsed = this.opusParseSize(data, byteOffset, remainingBytes, frameSizes[0]);\n remainingBytes -= numBytesParsed;\n // The parsed size of the first frame cannot be larger than the number of remaining bytes in the packet.\n if (frameSizes[0][0] < 0 || frameSizes[0][0] > remainingBytes) {\n return this.OPUS_INVALID_PACKET;\n }\n byteOffset += numBytesParsed;\n // The size of the second frame is the remaining number of bytes after the first frame.\n lastSizeBytes = remainingBytes - frameSizes[0][0];\n break;\n // A code 3 packet (https://www.rfc-editor.org/rfc/rfc6716#section-3.2.5) has multiple CBR/VBR frames (from 0 to\n // 120 ms).\n default:\n // Code 3 packets must have at least 2 bytes (i.e. at least 1 byte after the TOC byte).\n if (remainingBytes < 1)\n return this.OPUS_INVALID_PACKET;\n // Frame count byte format:\n // 0\n // 0 1 2 3 4 5 6 7\n // +-+-+-+-+-+-+-+-+\n // |v|p| M |\n // +-+-+-+-+-+-+-+-+\n //\n // Read the frame count byte, which immediately follows the TOC byte.\n const frameCountByte = data.getUint8(byteOffset++);\n --remainingBytes;\n // Read the 'M' bits of the frame count byte, which encode the number of frames.\n numFrames = frameCountByte & 0x3f;\n // The number of frames in a code 3 packet must not be 0.\n if (numFrames <= 0)\n return this.OPUS_INVALID_PACKET;\n const samplesPerFrame = this.opusPacketGetSamplesPerFrame(data, 48000);\n // A single frame can have at most 2880 samples, which happens in the case where 60ms of 48kHz audio is encoded\n // per frame. A code 3 packet cannot contain more than 120ms of audio, so the total number of samples cannot\n // exceed 2880 * 2 = 5760.\n if (samplesPerFrame * numFrames > 5760)\n return this.OPUS_INVALID_PACKET;\n // Parse padding bytes if the 'p' bit is 1.\n if (frameCountByte & 0x40) {\n let paddingCountByte;\n let numPaddingBytes;\n // Remove padding bytes (including padding count bytes) from the remaining byte count.\n do {\n // Sanity check that there are enough bytes to parse and remove the padding.\n if (remainingBytes <= 0)\n return this.OPUS_INVALID_PACKET;\n // Get the next padding count byte.\n paddingCountByte = data.getUint8(byteOffset++);\n --remainingBytes;\n // If the padding count byte has a value in the range 0...254, then the total size of the padding is the\n // value in the padding count byte.\n //\n // If the padding count byte has value 255, then the total size of the padding is 254 plus the value in the\n // next padding count byte. Therefore, keep reading padding count bytes while the value is 255.\n numPaddingBytes = paddingCountByte === 255 ? 254 : paddingCountByte;\n remainingBytes -= numPaddingBytes;\n paddingBytes += numPaddingBytes;\n } while (paddingCountByte === 255);\n }\n // Sanity check that the remaining number of bytes is not negative after removing the padding.\n if (remainingBytes < 0)\n return this.OPUS_INVALID_PACKET;\n // Read the 'v' bit (i.e. VBR bit).\n cbr = !(frameCountByte & 0x80);\n // VBR case\n if (!cbr) {\n lastSizeBytes = remainingBytes;\n // Let M be the number of frames. There will be M - 1 frame length indicators (which can be 1 or 2 bytes)\n // corresponding to the lengths of frames 0 to M - 2. The size of the last frame (i.e. frame M - 1) is the\n // number of data bytes after the end of frame M - 2 and before the start of the padding bytes.\n for (let i = 0; i < numFrames - 1; ++i) {\n numBytesParsed = this.opusParseSize(data, byteOffset, remainingBytes, frameSizes[i]);\n remainingBytes -= numBytesParsed;\n // The remaining number of data bytes must be enough to contain each frame.\n if (frameSizes[i][0] < 0 || frameSizes[i][0] > remainingBytes) {\n return this.OPUS_INVALID_PACKET;\n }\n byteOffset += numBytesParsed;\n lastSizeBytes -= numBytesParsed + frameSizes[i][0];\n }\n // Sanity check that the size of the last frame is not negative.\n if (lastSizeBytes < 0)\n return this.OPUS_INVALID_PACKET;\n }\n // CBR case\n else if (!selfDelimited) {\n // The size of each frame is the number of data bytes divided by the number of frames.\n lastSizeBytes = Math.trunc(remainingBytes / numFrames);\n // The number of data bytes must be a non-negative integer multiple of the number of frames.\n if (lastSizeBytes * numFrames !== remainingBytes)\n return this.OPUS_INVALID_PACKET;\n // All frames have equal size in the undelimited CBR case.\n for (let i = 0; i < numFrames - 1; ++i) {\n frameSizes[i][0] = lastSizeBytes;\n }\n }\n }\n // Self-delimited framing uses an extra 1 or 2 bytes, immediately preceding the data bytes, to indicate either the\n // size of the last frame (for code 0, code 2, and VBR code 3 packets) or the size of all the frames (for code 1 and\n // CBR code 3 packets). See https://www.rfc-editor.org/rfc/rfc6716#appendix-B.\n if (selfDelimited) {\n // The extra frame size byte(s) will always indicate the size of the last frame.\n numBytesParsed = this.opusParseSize(data, byteOffset, remainingBytes, frameSizes[numFrames - 1]);\n remainingBytes -= numBytesParsed;\n // There must be enough data bytes for the last frame.\n if (frameSizes[numFrames - 1][0] < 0 || frameSizes[numFrames - 1][0] > remainingBytes) {\n return this.OPUS_INVALID_PACKET;\n }\n byteOffset += numBytesParsed;\n // For CBR packets, the sizes of all the frames are equal.\n if (cbr) {\n // There must be enough data bytes for all the frames.\n if (frameSizes[numFrames - 1][0] * numFrames > remainingBytes) {\n return this.OPUS_INVALID_PACKET;\n }\n for (let i = 0; i < numFrames - 1; ++i) {\n frameSizes[i][0] = frameSizes[numFrames - 1][0];\n }\n }\n // At this point, `lastSizeBytes` contains the size of the last frame plus the size of the extra frame size\n // byte(s), so sanity check that `lastSizeBytes` is the upper bound for the size of the last frame.\n else if (!(numBytesParsed + frameSizes[numFrames - 1][0] <= lastSizeBytes)) {\n return this.OPUS_INVALID_PACKET;\n }\n }\n // Undelimited case\n else {\n // Because the size of the last packet is not encoded explicitly, it is possible that the size of the last packet\n // (or of all the packets, for the CBR case) is larger than maximum frame size.\n if (lastSizeBytes > this.OPUS_MAX_FRAME_SIZE_BYTES)\n return this.OPUS_INVALID_PACKET;\n frameSizes[numFrames - 1][0] = lastSizeBytes;\n }\n // Store the offset to the start of the payload.\n if (payloadOffset)\n payloadOffset[0] = byteOffset;\n // Store the offsets to the start of each frame.\n for (let i = 0; i < numFrames; ++i) {\n if (frameOffsets)\n frameOffsets[i][0] = byteOffset;\n byteOffset += frameSizes[i][0];\n }\n // Store the length of the Opus packet.\n if (packetLenBytes)\n packetLenBytes[0] = byteOffset + paddingBytes;\n return numFrames;\n }\n /**\n * Parse a single undelimited Opus packet into one or more Opus frames.\n *\n * @param packet Opus packet to be parsed.\n * @param lenBytes Size of the packet (in bytes).\n * @param tocByte Optional variable to store the TOC (table of contents) byte.\n * @param frameOffsets Optional variable to store the offsets (from the start of the packet) to the first bytes of\n * each Opus frame.\n * @param frameSizes Required variable to store the sizes (in bytes) of each Opus frame.\n * @param payloadOffset Optional variable to store the offset (from the start of the packet) to the first byte of the\n * payload.\n * @returns Number of Opus frames.\n */\n opusPacketParse(packet, lenBytes, tocByte, frameOffsets, frameSizes, payloadOffset) {\n return this.opusPacketParseImpl(packet, lenBytes, \n /* selfDelimited */ false, tocByte, frameOffsets, frameSizes, payloadOffset, null);\n }\n /**\n * This function returns the SILK VAD (voice activity detection) information encoded in the Opus packet. For CELT-only\n * packets that do not have VAD information, it returns -1.\n *\n * @param packet Opus packet.\n * @param lenBytes Size of the packet (in bytes).\n * @returns 0: no frame had the VAD flag set.\n * 1: at least one frame had the VAD flag set.\n * -1: VAD status could not be determined.\n */\n opusPacketHasVoiceActivity(packet, lenBytes) {\n if (!packet || lenBytes <= 0)\n return 0;\n // In CELT-only mode, we can not determine whether there is VAD.\n if (this.opusPacketIsCeltOnly(packet))\n return -1;\n const numSilkFrames = this.opusNumSilkFrames(packet);\n // It is not possible for `opusNumSilkFrames()` to return 0, so we ignore the next sanity check for test coverage.\n /* istanbul ignore next */\n if (numSilkFrames === 0)\n return -1;\n const opusFrameOffsets = new Array(this.OPUS_MAX_OPUS_FRAMES);\n const opusFrameSizes = new Array(this.OPUS_MAX_OPUS_FRAMES);\n for (let i = 0; i < this.OPUS_MAX_OPUS_FRAMES; ++i) {\n opusFrameOffsets[i] = [undefined];\n opusFrameSizes[i] = [undefined];\n }\n // Parse packet to get the Opus frames.\n const numOpusFrames = this.opusPacketParse(packet, lenBytes, null, opusFrameOffsets, opusFrameSizes, null);\n // VAD status cannot be determined for invalid packets.\n if (numOpusFrames < 0)\n return -1;\n // Iterate over all Opus frames, which may contain multiple SILK frames, to determine the VAD status.\n for (let i = 0; i < numOpusFrames; ++i) {\n if (opusFrameSizes[i][0] < 1)\n continue;\n // LP layer header bits format (https://www.rfc-editor.org/rfc/rfc6716#section-4.2.3):\n //\n // Mono case:\n // +-----------------+----------+\n // | 1 to 3 VAD bits | LBRR bit |\n // +-----------------+----------+\n //\n // Stereo case:\n // +---------------------+--------------+----------------------+---------------+\n // | 1 to 3 mid VAD bits | mid LBRR bit | 1 to 3 side VAD bits | side LBRR bit |\n // +---------------------+--------------+----------------------+---------------+\n // The upper 1 to 3 bits (dependent on the number of SILK frames) of the LP layer contain VAD bits. If any of\n // these VAD bits are 1, then voice activity is present.\n if (packet.getUint8(opusFrameOffsets[i][0]) >> (8 - numSilkFrames))\n return 1;\n // In the stereo case, there is a second set of 1 to 3 VAD bits, so also check these VAD bits.\n const channels = this.opusPacketGetNumChannels(packet);\n if (channels === 2 &&\n (packet.getUint8(opusFrameOffsets[i][0]) << (numSilkFrames + 1)) >> (8 - numSilkFrames)) {\n return 1;\n }\n }\n // No voice activity was detected.\n return 0;\n }\n /**\n * This method is based on Definition of the Opus Audio Codec\n * (https://tools.ietf.org/html/rfc6716). Basically, this method is based on\n * parsing the LP layer of an Opus packet, particularly the LBRR flag.\n *\n * @param packet Opus packet.\n * @param lenBytes Size of the packet (in bytes).\n * @returns true: packet has fec encoding about previous packet.\n * false: no fec encoding present.\n */\n opusPacketHasFec(packet, lenBytes) {\n if (!packet || lenBytes <= 0)\n return false;\n // In CELT-only mode, packets should not have FEC.\n if (this.opusPacketIsCeltOnly(packet))\n return false;\n const opusFrameOffsets = new Array(this.OPUS_MAX_OPUS_FRAMES);\n const opusFrameSizes = new Array(this.OPUS_MAX_OPUS_FRAMES);\n for (let i = 0; i < this.OPUS_MAX_OPUS_FRAMES; ++i) {\n opusFrameOffsets[i] = [undefined];\n opusFrameSizes[i] = [undefined];\n }\n // Parse packet to get the Opus frames.\n const numOpusFrames = this.opusPacketParse(packet, lenBytes, null, opusFrameOffsets, opusFrameSizes, null);\n if (numOpusFrames < 0)\n return false;\n /* istanbul ignore next */\n if (opusFrameSizes[0][0] <= 1)\n return false;\n const numSilkFrames = this.opusNumSilkFrames(packet);\n /* istanbul ignore next */\n if (numSilkFrames === 0)\n return false;\n const channels = this.opusPacketGetNumChannels(packet);\n /* istanbul ignore next */\n if (channels !== 1 && channels !== 2)\n return false;\n // A frame starts with the LP layer. The LP layer begins with two to eight\n // header bits.These consist of one VAD bit per SILK frame (up to 3),\n // followed by a single flag indicating the presence of LBRR frames.\n // For a stereo packet, these first flags correspond to the mid channel, and\n // a second set of flags is included for the side channel. Because these are\n // the first symbols decoded by the range coder and because they are coded\n // as binary values with uniform probability, they can be extracted directly\n // from the most significant bits of the first byte of compressed data.\n for (let i = 0; i < channels; i++) {\n if (packet.getUint8(opusFrameOffsets[0][0]) & (0x80 >> ((i + 1) * (numSilkFrames + 1) - 1)))\n return true;\n }\n return false;\n }\n}\nRedundantAudioEncoder.shouldLog = true;\nRedundantAudioEncoder.shouldReportStats = true;\nRedundantAudioEncoder.initializeWorker();\n"
A specifier for how to obtain a media stream from the browser. This can be a
MediaStreamitself, a set of constraints, a device ID.