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get api libs from bower
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Luke Pulverenti
parent
def418714f
commit
f36e664503
97 changed files with 16860 additions and 197 deletions
208
dashboard-ui/bower_components/hls.js/src/demux/aacdemuxer.js
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208
dashboard-ui/bower_components/hls.js/src/demux/aacdemuxer.js
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/**
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* AAC demuxer
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*/
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import {logger} from '../utils/logger';
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import ID3 from '../demux/id3';
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import {ErrorTypes, ErrorDetails} from '../errors';
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class AACDemuxer {
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constructor(observer,remuxerClass) {
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this.observer = observer;
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this.remuxerClass = remuxerClass;
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this.remuxer = new this.remuxerClass(observer);
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this._aacTrack = {type: 'audio', id :-1, sequenceNumber: 0, samples : [], len : 0};
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}
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static probe(data) {
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// check if data contains ID3 timestamp and ADTS sync worc
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var id3 = new ID3(data), adtsStartOffset,len;
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if(id3.hasTimeStamp) {
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// look for ADTS header (0xFFFx)
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for (adtsStartOffset = id3.length, len = data.length; adtsStartOffset < len - 1; adtsStartOffset++) {
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if ((data[adtsStartOffset] === 0xff) && (data[adtsStartOffset+1] & 0xf0) === 0xf0) {
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//logger.log('ADTS sync word found !');
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return true;
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}
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}
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}
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return false;
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}
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// feed incoming data to the front of the parsing pipeline
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push(data, audioCodec, videoCodec, timeOffset, cc, level, sn, duration) {
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var id3 = new ID3(data), adtsStartOffset,len, track = this._aacTrack, pts = id3.timeStamp, config, nbSamples,adtsFrameSize,adtsHeaderLen,stamp,aacSample;
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// look for ADTS header (0xFFFx)
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for (adtsStartOffset = id3.length, len = data.length; adtsStartOffset < len - 1; adtsStartOffset++) {
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if ((data[adtsStartOffset] === 0xff) && (data[adtsStartOffset+1] & 0xf0) === 0xf0) {
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break;
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}
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}
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if (!track.audiosamplerate) {
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config = this._ADTStoAudioConfig(data, adtsStartOffset, audioCodec);
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track.config = config.config;
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track.audiosamplerate = config.samplerate;
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track.channelCount = config.channelCount;
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track.codec = config.codec;
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track.timescale = this.remuxer.timescale;
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track.duration = this.remuxer.timescale * duration;
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logger.log(`parsed codec:${track.codec},rate:${config.samplerate},nb channel:${config.channelCount}`);
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}
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nbSamples = 0;
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while ((adtsStartOffset + 5) < len) {
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// retrieve frame size
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adtsFrameSize = ((data[adtsStartOffset + 3] & 0x03) << 11);
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// byte 4
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adtsFrameSize |= (data[adtsStartOffset + 4] << 3);
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// byte 5
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adtsFrameSize |= ((data[adtsStartOffset + 5] & 0xE0) >>> 5);
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adtsHeaderLen = (!!(data[adtsStartOffset + 1] & 0x01) ? 7 : 9);
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adtsFrameSize -= adtsHeaderLen;
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stamp = Math.round(90*pts + nbSamples * 1024 * 90000 / track.audiosamplerate);
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//stamp = pes.pts;
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//console.log('AAC frame, offset/length/pts:' + (adtsStartOffset+7) + '/' + adtsFrameSize + '/' + stamp.toFixed(0));
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if ((adtsFrameSize > 0) && ((adtsStartOffset + adtsHeaderLen + adtsFrameSize) <= len)) {
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aacSample = {unit: data.subarray(adtsStartOffset + adtsHeaderLen, adtsStartOffset + adtsHeaderLen + adtsFrameSize), pts: stamp, dts: stamp};
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track.samples.push(aacSample);
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track.len += adtsFrameSize;
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adtsStartOffset += adtsFrameSize + adtsHeaderLen;
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nbSamples++;
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// look for ADTS header (0xFFFx)
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for ( ; adtsStartOffset < (len - 1); adtsStartOffset++) {
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if ((data[adtsStartOffset] === 0xff) && ((data[adtsStartOffset + 1] & 0xf0) === 0xf0)) {
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break;
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}
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}
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} else {
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break;
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}
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}
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this.remuxer.remux(this._aacTrack,{samples : []}, {samples : []}, timeOffset);
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}
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_ADTStoAudioConfig(data, offset, audioCodec) {
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var adtsObjectType, // :int
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adtsSampleingIndex, // :int
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adtsExtensionSampleingIndex, // :int
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adtsChanelConfig, // :int
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config,
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userAgent = navigator.userAgent.toLowerCase(),
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adtsSampleingRates = [
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96000, 88200,
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64000, 48000,
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44100, 32000,
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24000, 22050,
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16000, 12000,
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11025, 8000,
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7350];
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// byte 2
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adtsObjectType = ((data[offset + 2] & 0xC0) >>> 6) + 1;
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adtsSampleingIndex = ((data[offset + 2] & 0x3C) >>> 2);
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if(adtsSampleingIndex > adtsSampleingRates.length-1) {
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this.observer.trigger(Event.ERROR, {type: ErrorTypes.MEDIA_ERROR, details: ErrorDetails.FRAG_PARSING_ERROR, fatal: true, reason: `invalid ADTS sampling index:${adtsSampleingIndex}`});
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return;
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}
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adtsChanelConfig = ((data[offset + 2] & 0x01) << 2);
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// byte 3
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adtsChanelConfig |= ((data[offset + 3] & 0xC0) >>> 6);
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logger.log(`manifest codec:${audioCodec},ADTS data:type:${adtsObjectType},sampleingIndex:${adtsSampleingIndex}[${adtsSampleingRates[adtsSampleingIndex]}Hz],channelConfig:${adtsChanelConfig}`);
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// firefox: freq less than 24kHz = AAC SBR (HE-AAC)
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if (userAgent.indexOf('firefox') !== -1) {
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if (adtsSampleingIndex >= 6) {
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adtsObjectType = 5;
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config = new Array(4);
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// HE-AAC uses SBR (Spectral Band Replication) , high frequencies are constructed from low frequencies
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// there is a factor 2 between frame sample rate and output sample rate
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// multiply frequency by 2 (see table below, equivalent to substract 3)
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adtsExtensionSampleingIndex = adtsSampleingIndex - 3;
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} else {
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adtsObjectType = 2;
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config = new Array(2);
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adtsExtensionSampleingIndex = adtsSampleingIndex;
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}
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// Android : always use AAC
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} else if (userAgent.indexOf('android') !== -1) {
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adtsObjectType = 2;
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config = new Array(2);
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adtsExtensionSampleingIndex = adtsSampleingIndex;
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} else {
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/* for other browsers (chrome ...)
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always force audio type to be HE-AAC SBR, as some browsers do not support audio codec switch properly (like Chrome ...)
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*/
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adtsObjectType = 5;
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config = new Array(4);
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// if (manifest codec is HE-AAC) OR (manifest codec not specified AND frequency less than 24kHz)
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if ((audioCodec && audioCodec.indexOf('mp4a.40.5') !== -1) || (!audioCodec && adtsSampleingIndex >= 6)) {
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// HE-AAC uses SBR (Spectral Band Replication) , high frequencies are constructed from low frequencies
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// there is a factor 2 between frame sample rate and output sample rate
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// multiply frequency by 2 (see table below, equivalent to substract 3)
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adtsExtensionSampleingIndex = adtsSampleingIndex - 3;
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} else {
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// if (manifest codec is AAC) AND (frequency less than 24kHz OR nb channel is 1)
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if (audioCodec && audioCodec.indexOf('mp4a.40.2') !== -1 && (adtsSampleingIndex >= 6 || adtsChanelConfig === 1)) {
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adtsObjectType = 2;
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config = new Array(2);
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}
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adtsExtensionSampleingIndex = adtsSampleingIndex;
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}
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}
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/* refer to http://wiki.multimedia.cx/index.php?title=MPEG-4_Audio#Audio_Specific_Config
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ISO 14496-3 (AAC).pdf - Table 1.13 — Syntax of AudioSpecificConfig()
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Audio Profile / Audio Object Type
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0: Null
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1: AAC Main
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2: AAC LC (Low Complexity)
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3: AAC SSR (Scalable Sample Rate)
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4: AAC LTP (Long Term Prediction)
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5: SBR (Spectral Band Replication)
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6: AAC Scalable
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sampling freq
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0: 96000 Hz
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1: 88200 Hz
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2: 64000 Hz
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3: 48000 Hz
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4: 44100 Hz
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5: 32000 Hz
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6: 24000 Hz
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7: 22050 Hz
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8: 16000 Hz
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9: 12000 Hz
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10: 11025 Hz
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11: 8000 Hz
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12: 7350 Hz
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13: Reserved
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14: Reserved
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15: frequency is written explictly
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Channel Configurations
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These are the channel configurations:
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0: Defined in AOT Specifc Config
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1: 1 channel: front-center
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2: 2 channels: front-left, front-right
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*/
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// audioObjectType = profile => profile, the MPEG-4 Audio Object Type minus 1
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config[0] = adtsObjectType << 3;
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// samplingFrequencyIndex
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config[0] |= (adtsSampleingIndex & 0x0E) >> 1;
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config[1] |= (adtsSampleingIndex & 0x01) << 7;
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// channelConfiguration
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config[1] |= adtsChanelConfig << 3;
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if (adtsObjectType === 5) {
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// adtsExtensionSampleingIndex
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config[1] |= (adtsExtensionSampleingIndex & 0x0E) >> 1;
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config[2] = (adtsExtensionSampleingIndex & 0x01) << 7;
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// adtsObjectType (force to 2, chrome is checking that object type is less than 5 ???
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// https://chromium.googlesource.com/chromium/src.git/+/master/media/formats/mp4/aac.cc
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config[2] |= 2 << 2;
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config[3] = 0;
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}
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return {config: config, samplerate: adtsSampleingRates[adtsSampleingIndex], channelCount: adtsChanelConfig, codec: ('mp4a.40.' + adtsObjectType)};
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}
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destroy() {
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}
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}
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export default AACDemuxer;
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