mirror of
https://github.com/jellyfin/jellyfin-web
synced 2025-03-30 19:56:21 +00:00
update components
This commit is contained in:
Luke Pulverenti
parent
c6b706f48a
commit
5de2e610ce
35 changed files with 1420 additions and 1130 deletions
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@ -1,9 +1,9 @@
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/**
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* AAC demuxer
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*/
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import ADTS from './adts';
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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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@ -32,7 +32,10 @@ import {ErrorTypes, ErrorDetails} from '../errors';
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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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var track = this._aacTrack,
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id3 = new ID3(data),
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pts = 90*id3.timeStamp,
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config, adtsFrameSize, adtsStartOffset, adtsHeaderLen, stamp, nbSamples, len, 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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@ -41,7 +44,7 @@ import {ErrorTypes, ErrorDetails} from '../errors';
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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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config = ADTS.getAudioConfig(this.observer,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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@ -60,7 +63,7 @@ import {ErrorTypes, ErrorDetails} from '../errors';
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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 = Math.round(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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@ -82,124 +85,6 @@ import {ErrorTypes, ErrorDetails} from '../errors';
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this.remuxer.remux(this._aacTrack,{samples : []}, {samples : [ { pts: pts, dts : pts, unit : id3.payload} ]}, 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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132
dashboard-ui/bower_components/hls.js/src/demux/adts.js
vendored
Normal file
132
dashboard-ui/bower_components/hls.js/src/demux/adts.js
vendored
Normal file
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@ -0,0 +1,132 @@
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/**
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* ADTS parser helper
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*/
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import {logger} from '../utils/logger';
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import {ErrorTypes, ErrorDetails} from '../errors';
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class ADTS {
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static getAudioConfig(observer, 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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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 HE-AACv2) OR (manifest codec not specified AND frequency less than 24kHz)
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if ((audioCodec && ((audioCodec.indexOf('mp4a.40.29') !== -1) ||
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(audioCodec.indexOf('mp4a.40.5') !== -1))) ||
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(!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) OR (manifest codec not specified and mono audio)
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// Chrome fails to play back with AAC LC mono when initialized with HE-AAC. This is not a problem with stereo.
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if (audioCodec && audioCodec.indexOf('mp4a.40.2') !== -1 && (adtsSampleingIndex >= 6 || adtsChanelConfig === 1) ||
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(!audioCodec && 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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}
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export default ADTS;
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@ -179,7 +179,12 @@ class ExpGolomb {
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if (profileIdc === 100 ||
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profileIdc === 110 ||
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profileIdc === 122 ||
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profileIdc === 144) {
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profileIdc === 244 ||
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profileIdc === 44 ||
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profileIdc === 83 ||
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profileIdc === 86 ||
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profileIdc === 118 ||
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profileIdc === 128) {
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var chromaFormatIdc = this.readUEG();
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if (chromaFormatIdc === 3) {
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this.skipBits(1); // separate_colour_plane_flag
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@ -9,6 +9,7 @@
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* upon discontinuity or level switch detection, it will also notifies the remuxer so that it can reset its state.
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*/
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import ADTS from './adts';
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import Event from '../events';
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import ExpGolomb from './exp-golomb';
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// import Hex from '../utils/hex';
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@ -21,7 +22,6 @@
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this.observer = observer;
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this.remuxerClass = remuxerClass;
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this.lastCC = 0;
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this.PES_TIMESCALE = 90000;
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this.remuxer = new this.remuxerClass(observer);
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}
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@ -423,16 +423,19 @@
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// If NAL units are not starting right at the beginning of the PES packet, push preceding data into previous NAL unit.
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overflow = i - state - 1;
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if (overflow) {
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var track = this._avcTrack,
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samples = track.samples;
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//logger.log('first NALU found with overflow:' + overflow);
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if (this._avcTrack.samples.length) {
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var lastavcSample = this._avcTrack.samples[this._avcTrack.samples.length - 1];
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var lastUnit = lastavcSample.units.units[lastavcSample.units.units.length - 1];
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var tmp = new Uint8Array(lastUnit.data.byteLength + overflow);
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if (samples.length) {
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var lastavcSample = samples[samples.length - 1],
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lastUnits = lastavcSample.units.units,
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lastUnit = lastUnits[lastUnits.length - 1],
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tmp = new Uint8Array(lastUnit.data.byteLength + overflow);
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tmp.set(lastUnit.data, 0);
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tmp.set(array.subarray(0, overflow), lastUnit.data.byteLength);
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lastUnit.data = tmp;
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lastavcSample.units.length += overflow;
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this._avcTrack.len += overflow;
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track.len += overflow;
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}
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}
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}
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@ -460,7 +463,13 @@
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}
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_parseAACPES(pes) {
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var track = this._aacTrack, aacSample, data = pes.data, config, adtsFrameSize, adtsStartOffset, adtsHeaderLen, stamp, nbSamples, len;
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var track = this._aacTrack,
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data = pes.data,
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pts = pes.pts,
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startOffset = 0,
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duration = this._duration,
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audioCodec = this.audioCodec,
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config, frameLength, frameDuration, frameIndex, offset, headerLength, stamp, len, aacSample;
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if (this.aacOverFlow) {
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var tmp = new Uint8Array(this.aacOverFlow.byteLength + data.byteLength);
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tmp.set(this.aacOverFlow, 0);
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@ -468,16 +477,16 @@
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data = tmp;
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}
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// look for ADTS header (0xFFFx)
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for (adtsStartOffset = 0, len = data.length; adtsStartOffset < len - 1; adtsStartOffset++) {
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if ((data[adtsStartOffset] === 0xff) && (data[adtsStartOffset+1] & 0xf0) === 0xf0) {
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for (offset = startOffset, len = data.length; offset < len - 1; offset++) {
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if ((data[offset] === 0xff) && (data[offset+1] & 0xf0) === 0xf0) {
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break;
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}
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}
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// if ADTS header does not start straight from the beginning of the PES payload, raise an error
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if (adtsStartOffset) {
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if (offset) {
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var reason, fatal;
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if (adtsStartOffset < len - 1) {
|
||||
reason = `AAC PES did not start with ADTS header,offset:${adtsStartOffset}`;
|
||||
if (offset < len - 1) {
|
||||
reason = `AAC PES did not start with ADTS header,offset:${offset}`;
|
||||
fatal = false;
|
||||
} else {
|
||||
reason = 'no ADTS header found in AAC PES';
|
||||
|
|
@ -489,37 +498,38 @@
|
|||
}
|
||||
}
|
||||
if (!track.audiosamplerate) {
|
||||
config = this._ADTStoAudioConfig(data, adtsStartOffset, this.audioCodec);
|
||||
config = ADTS.getAudioConfig(this.observer,data, offset, audioCodec);
|
||||
track.config = config.config;
|
||||
track.audiosamplerate = config.samplerate;
|
||||
track.channelCount = config.channelCount;
|
||||
track.codec = config.codec;
|
||||
track.timescale = this.remuxer.timescale;
|
||||
track.duration = this.remuxer.timescale * this._duration;
|
||||
track.duration = track.timescale * duration;
|
||||
logger.log(`parsed codec:${track.codec},rate:${config.samplerate},nb channel:${config.channelCount}`);
|
||||
}
|
||||
nbSamples = 0;
|
||||
while ((adtsStartOffset + 5) < len) {
|
||||
frameIndex = 0;
|
||||
frameDuration = 1024 * 90000 / track.audiosamplerate;
|
||||
while ((offset + 5) < len) {
|
||||
// The protection skip bit tells us if we have 2 bytes of CRC data at the end of the ADTS header
|
||||
headerLength = (!!(data[offset + 1] & 0x01) ? 7 : 9);
|
||||
// retrieve frame size
|
||||
adtsFrameSize = ((data[adtsStartOffset + 3] & 0x03) << 11);
|
||||
// byte 4
|
||||
adtsFrameSize |= (data[adtsStartOffset + 4] << 3);
|
||||
// byte 5
|
||||
adtsFrameSize |= ((data[adtsStartOffset + 5] & 0xE0) >>> 5);
|
||||
adtsHeaderLen = (!!(data[adtsStartOffset + 1] & 0x01) ? 7 : 9);
|
||||
adtsFrameSize -= adtsHeaderLen;
|
||||
stamp = Math.round(pes.pts + nbSamples * 1024 * this.PES_TIMESCALE / track.audiosamplerate);
|
||||
frameLength = ((data[offset + 3] & 0x03) << 11) |
|
||||
(data[offset + 4] << 3) |
|
||||
((data[offset + 5] & 0xE0) >>> 5);
|
||||
frameLength -= headerLength;
|
||||
stamp = Math.round(pts + frameIndex * frameDuration);
|
||||
//stamp = pes.pts;
|
||||
//console.log('AAC frame, offset/length/pts:' + (adtsStartOffset+7) + '/' + adtsFrameSize + '/' + stamp.toFixed(0));
|
||||
if ((adtsFrameSize > 0) && ((adtsStartOffset + adtsHeaderLen + adtsFrameSize) <= len)) {
|
||||
aacSample = {unit: data.subarray(adtsStartOffset + adtsHeaderLen, adtsStartOffset + adtsHeaderLen + adtsFrameSize), pts: stamp, dts: stamp};
|
||||
this._aacTrack.samples.push(aacSample);
|
||||
this._aacTrack.len += adtsFrameSize;
|
||||
adtsStartOffset += adtsFrameSize + adtsHeaderLen;
|
||||
nbSamples++;
|
||||
|
||||
//console.log('AAC frame, offset/length/pts:' + (offset+headerLength) + '/' + frameLength + '/' + stamp.toFixed(0));
|
||||
if ((frameLength > 0) && ((offset + headerLength + frameLength) <= len)) {
|
||||
aacSample = {unit: data.subarray(offset + headerLength, offset + headerLength + frameLength), pts: stamp, dts: stamp};
|
||||
track.samples.push(aacSample);
|
||||
track.len += frameLength;
|
||||
offset += frameLength + headerLength;
|
||||
frameIndex++;
|
||||
// look for ADTS header (0xFFFx)
|
||||
for ( ; adtsStartOffset < (len - 1); adtsStartOffset++) {
|
||||
if ((data[adtsStartOffset] === 0xff) && ((data[adtsStartOffset + 1] & 0xf0) === 0xf0)) {
|
||||
for ( ; offset < (len - 1); offset++) {
|
||||
if ((data[offset] === 0xff) && ((data[offset + 1] & 0xf0) === 0xf0)) {
|
||||
break;
|
||||
}
|
||||
}
|
||||
|
|
@ -527,135 +537,13 @@
|
|||
break;
|
||||
}
|
||||
}
|
||||
if (adtsStartOffset < len) {
|
||||
this.aacOverFlow = data.subarray(adtsStartOffset, len);
|
||||
if (offset < len) {
|
||||
this.aacOverFlow = data.subarray(offset, len);
|
||||
} else {
|
||||
this.aacOverFlow = null;
|
||||
}
|
||||
}
|
||||
|
||||
_ADTStoAudioConfig(data, offset, audioCodec) {
|
||||
var adtsObjectType, // :int
|
||||
adtsSampleingIndex, // :int
|
||||
adtsExtensionSampleingIndex, // :int
|
||||
adtsChanelConfig, // :int
|
||||
config,
|
||||
userAgent = navigator.userAgent.toLowerCase(),
|
||||
adtsSampleingRates = [
|
||||
96000, 88200,
|
||||
64000, 48000,
|
||||
44100, 32000,
|
||||
24000, 22050,
|
||||
16000, 12000,
|
||||
11025, 8000,
|
||||
7350];
|
||||
// byte 2
|
||||
adtsObjectType = ((data[offset + 2] & 0xC0) >>> 6) + 1;
|
||||
adtsSampleingIndex = ((data[offset + 2] & 0x3C) >>> 2);
|
||||
if(adtsSampleingIndex > adtsSampleingRates.length-1) {
|
||||
this.observer.trigger(Event.ERROR, {type: ErrorTypes.MEDIA_ERROR, details: ErrorDetails.FRAG_PARSING_ERROR, fatal: true, reason: `invalid ADTS sampling index:${adtsSampleingIndex}`});
|
||||
return;
|
||||
}
|
||||
adtsChanelConfig = ((data[offset + 2] & 0x01) << 2);
|
||||
// byte 3
|
||||
adtsChanelConfig |= ((data[offset + 3] & 0xC0) >>> 6);
|
||||
logger.log(`manifest codec:${audioCodec},ADTS data:type:${adtsObjectType},sampleingIndex:${adtsSampleingIndex}[${adtsSampleingRates[adtsSampleingIndex]}Hz],channelConfig:${adtsChanelConfig}`);
|
||||
// firefox: freq less than 24kHz = AAC SBR (HE-AAC)
|
||||
if (userAgent.indexOf('firefox') !== -1) {
|
||||
if (adtsSampleingIndex >= 6) {
|
||||
adtsObjectType = 5;
|
||||
config = new Array(4);
|
||||
// HE-AAC uses SBR (Spectral Band Replication) , high frequencies are constructed from low frequencies
|
||||
// there is a factor 2 between frame sample rate and output sample rate
|
||||
// multiply frequency by 2 (see table below, equivalent to substract 3)
|
||||
adtsExtensionSampleingIndex = adtsSampleingIndex - 3;
|
||||
} else {
|
||||
adtsObjectType = 2;
|
||||
config = new Array(2);
|
||||
adtsExtensionSampleingIndex = adtsSampleingIndex;
|
||||
}
|
||||
// Android : always use AAC
|
||||
} else if (userAgent.indexOf('android') !== -1) {
|
||||
adtsObjectType = 2;
|
||||
config = new Array(2);
|
||||
adtsExtensionSampleingIndex = adtsSampleingIndex;
|
||||
} else {
|
||||
/* for other browsers (chrome ...)
|
||||
always force audio type to be HE-AAC SBR, as some browsers do not support audio codec switch properly (like Chrome ...)
|
||||
*/
|
||||
adtsObjectType = 5;
|
||||
config = new Array(4);
|
||||
// if (manifest codec is HE-AAC or HE-AACv2) OR (manifest codec not specified AND frequency less than 24kHz)
|
||||
if ((audioCodec && ((audioCodec.indexOf('mp4a.40.29') !== -1) ||
|
||||
(audioCodec.indexOf('mp4a.40.5') !== -1))) ||
|
||||
(!audioCodec && adtsSampleingIndex >= 6)) {
|
||||
// HE-AAC uses SBR (Spectral Band Replication) , high frequencies are constructed from low frequencies
|
||||
// there is a factor 2 between frame sample rate and output sample rate
|
||||
// multiply frequency by 2 (see table below, equivalent to substract 3)
|
||||
adtsExtensionSampleingIndex = adtsSampleingIndex - 3;
|
||||
} else {
|
||||
// if (manifest codec is AAC) AND (frequency less than 24kHz OR nb channel is 1) OR (manifest codec not specified and mono audio)
|
||||
// Chrome fails to play back with AAC LC mono when initialized with HE-AAC. This is not a problem with stereo.
|
||||
if (audioCodec && audioCodec.indexOf('mp4a.40.2') !== -1 && (adtsSampleingIndex >= 6 || adtsChanelConfig === 1) ||
|
||||
(!audioCodec && adtsChanelConfig === 1)) {
|
||||
adtsObjectType = 2;
|
||||
config = new Array(2);
|
||||
}
|
||||
adtsExtensionSampleingIndex = adtsSampleingIndex;
|
||||
}
|
||||
}
|
||||
/* refer to http://wiki.multimedia.cx/index.php?title=MPEG-4_Audio#Audio_Specific_Config
|
||||
ISO 14496-3 (AAC).pdf - Table 1.13 — Syntax of AudioSpecificConfig()
|
||||
Audio Profile / Audio Object Type
|
||||
0: Null
|
||||
1: AAC Main
|
||||
2: AAC LC (Low Complexity)
|
||||
3: AAC SSR (Scalable Sample Rate)
|
||||
4: AAC LTP (Long Term Prediction)
|
||||
5: SBR (Spectral Band Replication)
|
||||
6: AAC Scalable
|
||||
sampling freq
|
||||
0: 96000 Hz
|
||||
1: 88200 Hz
|
||||
2: 64000 Hz
|
||||
3: 48000 Hz
|
||||
4: 44100 Hz
|
||||
5: 32000 Hz
|
||||
6: 24000 Hz
|
||||
7: 22050 Hz
|
||||
8: 16000 Hz
|
||||
9: 12000 Hz
|
||||
10: 11025 Hz
|
||||
11: 8000 Hz
|
||||
12: 7350 Hz
|
||||
13: Reserved
|
||||
14: Reserved
|
||||
15: frequency is written explictly
|
||||
Channel Configurations
|
||||
These are the channel configurations:
|
||||
0: Defined in AOT Specifc Config
|
||||
1: 1 channel: front-center
|
||||
2: 2 channels: front-left, front-right
|
||||
*/
|
||||
// audioObjectType = profile => profile, the MPEG-4 Audio Object Type minus 1
|
||||
config[0] = adtsObjectType << 3;
|
||||
// samplingFrequencyIndex
|
||||
config[0] |= (adtsSampleingIndex & 0x0E) >> 1;
|
||||
config[1] |= (adtsSampleingIndex & 0x01) << 7;
|
||||
// channelConfiguration
|
||||
config[1] |= adtsChanelConfig << 3;
|
||||
if (adtsObjectType === 5) {
|
||||
// adtsExtensionSampleingIndex
|
||||
config[1] |= (adtsExtensionSampleingIndex & 0x0E) >> 1;
|
||||
config[2] = (adtsExtensionSampleingIndex & 0x01) << 7;
|
||||
// adtsObjectType (force to 2, chrome is checking that object type is less than 5 ???
|
||||
// https://chromium.googlesource.com/chromium/src.git/+/master/media/formats/mp4/aac.cc
|
||||
config[2] |= 2 << 2;
|
||||
config[3] = 0;
|
||||
}
|
||||
return {config: config, samplerate: adtsSampleingRates[adtsSampleingIndex], channelCount: adtsChanelConfig, codec: ('mp4a.40.' + adtsObjectType)};
|
||||
}
|
||||
|
||||
_parseID3PES(pes) {
|
||||
this._id3Track.samples.push(pes);
|
||||
}
|
||||
|
|
|
|||
Loading…
Add table
Add a link
Reference in a new issue