src/demux/adts.js
/**
* ADTS parser helper
* @link https://wiki.multimedia.cx/index.php?title=ADTS
*/
import { logger } from '../utils/logger';
import { ErrorTypes, ErrorDetails } from '../errors';
import Event from '../events';
import { getSelfScope } from '../utils/get-self-scope';
export function getAudioConfig (observer, data, offset, audioCodec) {
let adtsObjectType, // :int
adtsSampleingIndex, // :int
adtsExtensionSampleingIndex, // :int
adtsChanelConfig, // :int
config,
userAgent = navigator.userAgent.toLowerCase(),
manifestCodec = audioCodec,
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) {
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 (/firefox/i.test(userAgent)) {
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/Vivaldi/Opera ...)
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 AND nb channel is 1) OR (manifest codec not specified and mono audio)
// Chrome fails to play back with low frequency 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) ||
/vivaldi/i.test(userAgent)) ||
(!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), manifestCodec: manifestCodec };
}
export function isHeaderPattern (data, offset) {
return data[offset] === 0xff && (data[offset + 1] & 0xf6) === 0xf0;
}
export function getHeaderLength (data, offset) {
return (data[offset + 1] & 0x01 ? 7 : 9);
}
export function getFullFrameLength (data, offset) {
return ((data[offset + 3] & 0x03) << 11) |
(data[offset + 4] << 3) |
((data[offset + 5] & 0xE0) >>> 5);
}
export function isHeader (data, offset) {
// Look for ADTS header | 1111 1111 | 1111 X00X | where X can be either 0 or 1
// Layer bits (position 14 and 15) in header should be always 0 for ADTS
// More info https://wiki.multimedia.cx/index.php?title=ADTS
if (offset + 1 < data.length && isHeaderPattern(data, offset)) {
return true;
}
return false;
}
export function probe (data, offset) {
// same as isHeader but we also check that ADTS frame follows last ADTS frame
// or end of data is reached
if (isHeader(data, offset)) {
// ADTS header Length
let headerLength = getHeaderLength(data, offset);
if (offset + headerLength >= data.length) {
return false;
}
// ADTS frame Length
let frameLength = getFullFrameLength(data, offset);
if (frameLength <= headerLength) {
return false;
}
let newOffset = offset + frameLength;
if (newOffset === data.length || (newOffset + 1 < data.length && isHeaderPattern(data, newOffset))) {
return true;
}
}
return false;
}
export function initTrackConfig (track, observer, data, offset, audioCodec) {
if (!track.samplerate) {
let config = getAudioConfig(observer, data, offset, audioCodec);
track.config = config.config;
track.samplerate = config.samplerate;
track.channelCount = config.channelCount;
track.codec = config.codec;
track.manifestCodec = config.manifestCodec;
logger.log(`parsed codec:${track.codec},rate:${config.samplerate},nb channel:${config.channelCount}`);
}
}
export function getFrameDuration (samplerate) {
return 1024 * 90000 / samplerate;
}
export function parseFrameHeader (data, offset, pts, frameIndex, frameDuration) {
let headerLength, frameLength, stamp;
let length = data.length;
// The protection skip bit tells us if we have 2 bytes of CRC data at the end of the ADTS header
headerLength = getHeaderLength(data, offset);
// retrieve frame size
frameLength = getFullFrameLength(data, offset);
frameLength -= headerLength;
if ((frameLength > 0) && ((offset + headerLength + frameLength) <= length)) {
stamp = pts + frameIndex * frameDuration;
// logger.log(`AAC frame, offset/length/total/pts:${offset+headerLength}/${frameLength}/${data.byteLength}/${(stamp/90).toFixed(0)}`);
return { headerLength, frameLength, stamp };
}
return undefined;
}
export function appendFrame (track, data, offset, pts, frameIndex) {
let frameDuration = getFrameDuration(track.samplerate);
let header = parseFrameHeader(data, offset, pts, frameIndex, frameDuration);
if (header) {
let stamp = header.stamp;
let headerLength = header.headerLength;
let frameLength = header.frameLength;
// logger.log(`AAC frame, offset/length/total/pts:${offset+headerLength}/${frameLength}/${data.byteLength}/${(stamp/90).toFixed(0)}`);
let aacSample = {
unit: data.subarray(offset + headerLength, offset + headerLength + frameLength),
pts: stamp,
dts: stamp
};
track.samples.push(aacSample);
return { sample: aacSample, length: frameLength + headerLength };
}
return undefined;
}