Fix ADTS canParse (last sample), typos, and logging

Fixes PIPELINE_ERROR_DECODE errors in certain streams with acc file segments

src/demux/adts.ts

@@ -29,7 +29,7 @@ export function getAudioConfig(
   audioCodec: string
 ): AudioConfig | void {
   let adtsObjectType: number;
-  let adtsExtensionSampleingIndex: number;
+  let adtsExtensionSamplingIndex: number;
   let adtsChanelConfig: number;
   let config: number[];
   const userAgent = navigator.userAgent.toLowerCase();
@@ -51,41 +51,41 @@ export function getAudioConfig(
   ];
   // byte 2
   adtsObjectType = ((data[offset + 2] & 0xc0) >>> 6) + 1;
-  const adtsSampleingIndex = (data[offset + 2] & 0x3c) >>> 2;
-  if (adtsSampleingIndex > adtsSampleingRates.length - 1) {
+  const adtsSamplingIndex = (data[offset + 2] & 0x3c) >>> 2;
+  if (adtsSamplingIndex > adtsSampleingRates.length - 1) {
     observer.trigger(Events.ERROR, {
       type: ErrorTypes.MEDIA_ERROR,
       details: ErrorDetails.FRAG_PARSING_ERROR,
       fatal: true,
-      reason: `invalid ADTS sampling index:${adtsSampleingIndex}`,
+      reason: `invalid ADTS sampling index:${adtsSamplingIndex}`,
     });
     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:${adtsChanelConfig}`
+    `manifest codec:${audioCodec}, ADTS type:${adtsObjectType}, samplingIndex:${adtsSamplingIndex}`
   );
   // firefox: freq less than 24kHz = AAC SBR (HE-AAC)
   if (/firefox/i.test(userAgent)) {
-    if (adtsSampleingIndex >= 6) {
+    if (adtsSamplingIndex >= 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;
+      adtsExtensionSamplingIndex = adtsSamplingIndex - 3;
     } else {
       adtsObjectType = 2;
       config = new Array(2);
-      adtsExtensionSampleingIndex = adtsSampleingIndex;
+      adtsExtensionSamplingIndex = adtsSamplingIndex;
     }
     // Android : always use AAC
   } else if (userAgent.indexOf('android') !== -1) {
     adtsObjectType = 2;
     config = new Array(2);
-    adtsExtensionSampleingIndex = adtsSampleingIndex;
+    adtsExtensionSamplingIndex = adtsSamplingIndex;
   } 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 ...)
@@ -97,26 +97,26 @@ export function getAudioConfig(
       (audioCodec &&
         (audioCodec.indexOf('mp4a.40.29') !== -1 ||
           audioCodec.indexOf('mp4a.40.5') !== -1)) ||
-      (!audioCodec && adtsSampleingIndex >= 6)
+      (!audioCodec && adtsSamplingIndex >= 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;
+      adtsExtensionSamplingIndex = adtsSamplingIndex - 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) ||
+          ((adtsSamplingIndex >= 6 && adtsChanelConfig === 1) ||
             /vivaldi/i.test(userAgent))) ||
         (!audioCodec && adtsChanelConfig === 1)
       ) {
         adtsObjectType = 2;
         config = new Array(2);
       }
-      adtsExtensionSampleingIndex = adtsSampleingIndex;
+      adtsExtensionSamplingIndex = adtsSamplingIndex;
     }
   }
   /* refer to http://wiki.multimedia.cx/index.php?title=MPEG-4_Audio#Audio_Specific_Config
@@ -155,22 +155,22 @@ export function getAudioConfig(
   // 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;
+  config[0] |= (adtsSamplingIndex & 0x0e) >> 1;
+  config[1] |= (adtsSamplingIndex & 0x01) << 7;
   // channelConfiguration
   config[1] |= adtsChanelConfig << 3;
   if (adtsObjectType === 5) {
     // adtsExtensionSampleingIndex
-    config[1] |= (adtsExtensionSampleingIndex & 0x0e) >> 1;
-    config[2] = (adtsExtensionSampleingIndex & 0x01) << 7;
+    config[1] |= (adtsExtensionSamplingIndex & 0x0e) >> 1;
+    config[2] = (adtsExtensionSamplingIndex & 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,
-    samplerate: adtsSampleingRates[adtsSampleingIndex],
+    samplerate: adtsSampleingRates[adtsSamplingIndex],
     channelCount: adtsChanelConfig,
     codec: 'mp4a.40.' + adtsObjectType,
     manifestCodec,
@@ -208,7 +208,7 @@ export function canParse(data: Uint8Array, offset: number): boolean {
   return (
     canGetFrameLength(data, offset) &&
     isHeaderPattern(data, offset) &&
-    getFullFrameLength(data, offset) < data.length - offset
+    getFullFrameLength(data, offset) <= data.length - offset
   );
 }
 
@@ -251,7 +251,7 @@ export function initTrackConfig(
     track.codec = config.codec;
     track.manifestCodec = config.manifestCodec;
     logger.log(
-      `parsed codec:${track.codec},rate:${config.samplerate},nb channel:${config.channelCount}`
+      `parsed codec:${track.codec}, rate:${config.samplerate}, channels:${config.channelCount}`
     );
   }
 }

src/demux/base-audio-demuxer.ts

@@ -127,12 +127,13 @@ class BaseAudioDemuxer implements Demuxer {
 
   flush(timeOffset: number): DemuxerResult {
     // Parse cache in case of remaining frames.
-    if (this.cachedData) {
-      this.demux(this.cachedData, 0);
+    const cachedData = this.cachedData;
+    if (cachedData) {
+      this.cachedData = null;
+      this.demux(cachedData, 0);
     }
 
     this.frameIndex = 0;
-    this.cachedData = null;
 
     return {
       audioTrack: this._audioTrack,

src/demux/mpegaudio.ts

@@ -227,7 +227,7 @@ export function isHeader(data: Uint8Array, offset: number): boolean {
 export function canParse(data: Uint8Array, offset: number): boolean {
   const headerSize = 4;
 
-  return isHeaderPattern(data, offset) && data.length - offset >= headerSize;
+  return isHeaderPattern(data, offset) && headerSize <= data.length - offset;
 }
 
 export function probe(data: Uint8Array, offset: number): boolean {

src/remux/mp4-remuxer.ts

@@ -281,9 +281,6 @@ export default class MP4Remuxer implements Remuxer {
       // using audio sampling rate here helps having an integer MP4 frame duration
       // this avoids potential rounding issue and AV sync issue
       audioTrack.timescale = audioTrack.samplerate;
-      logger.log(
-        `[mp4-remuxer]: audio sampling rate : ${audioTrack.samplerate}`
-      );
       if (!audioTrack.isAAC) {
         if (typeSupported.mpeg) {
           // Chrome and Safari