update components
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35f69d09ea
commit
51fcbbf744
15 changed files with 114 additions and 83 deletions
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@ -673,7 +673,7 @@ class StreamController extends EventHandler {
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media.addEventListener('seeked', this.onvseeked);
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media.addEventListener('ended', this.onvended);
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if(this.levels && this.config.autoStartLoad) {
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this.startLoad();
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this.hls.startLoad();
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}
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}
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@ -782,7 +782,7 @@ class StreamController extends EventHandler {
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this.startLevelLoaded = false;
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this.startFragRequested = false;
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if (this.config.autoStartLoad) {
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this.startLoad();
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this.hls.startLoad();
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}
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}
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@ -49,8 +49,7 @@ import ID3 from '../demux/id3';
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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 = config.samplerate;
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track.duration = config.samplerate * duration;
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track.duration = duration;
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logger.log(`parsed codec:${track.codec},rate:${config.samplerate},nb channel:${config.channelCount}`);
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}
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frameIndex = 0;
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@ -427,8 +427,7 @@
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track.width = config.width;
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track.height = config.height;
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track.sps = [unit.data];
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track.timescale = this.remuxer.timescale;
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track.duration = this.remuxer.timescale * this._duration;
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track.duration = this._duration;
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var codecarray = unit.data.subarray(1, 4);
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var codecstring = 'avc1.';
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for (i = 0; i < 3; i++) {
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@ -598,8 +597,7 @@
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track.audiosamplerate = config.samplerate;
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track.channelCount = config.channelCount;
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track.codec = config.codec;
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track.timescale = config.samplerate;
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track.duration = config.samplerate * duration;
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track.duration = duration;
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logger.log(`parsed codec:${track.codec},rate:${config.samplerate},nb channel:${config.channelCount}`);
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}
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frameIndex = 0;
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@ -12,10 +12,6 @@ class DummyRemuxer {
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return false;
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}
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get timescale() {
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return this.PES_TIMESCALE;
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}
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destroy() {
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}
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@ -172,6 +172,7 @@ class MP4 {
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}
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static mdhd(timescale, duration) {
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duration *= timescale;
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return MP4.box(MP4.types.mdhd, new Uint8Array([
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0x00, // version 0
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0x00, 0x00, 0x00, // flags
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@ -243,6 +244,7 @@ class MP4 {
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}
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static mvhd(timescale,duration) {
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duration*=timescale;
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var
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bytes = new Uint8Array([
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0x00, // version 0
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@ -424,7 +426,7 @@ class MP4 {
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static tkhd(track) {
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var id = track.id,
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duration = track.duration,
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duration = track.duration*track.timescale,
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width = track.width,
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height = track.height;
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return MP4.box(MP4.types.tkhd, new Uint8Array([
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@ -21,10 +21,6 @@ class MP4Remuxer {
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return false;
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}
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get timescale() {
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return this.MP4_TIMESCALE;
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}
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destroy() {
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}
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@ -74,8 +70,22 @@ class MP4Remuxer {
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if (computePTSDTS) {
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initPTS = initDTS = Infinity;
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}
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if (audioTrack.config && audioSamples.length) {
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audioTrack.timescale = audioTrack.audiosamplerate;
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// MP4 duration (track duration in seconds multiplied by timescale) is coded on 32 bits
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// we know that each AAC sample contains 1024 frames....
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// in order to avoid overflowing the 32 bit counter for large duration, we use smaller timescale (timescale/gcd)
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// we just need to ensure that AAC sample duration will still be an integer (will be 1024/gcd)
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if (audioTrack.timescale * audioTrack.duration > Math.pow(2, 32)) {
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let greatestCommonDivisor = function(a, b) {
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if ( ! b) {
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return a;
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}
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return greatestCommonDivisor(b, a % b);
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};
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audioTrack.timescale = audioTrack.audiosamplerate / greatestCommonDivisor(audioTrack.audiosamplerate,1024);
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}
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logger.log ('audio mp4 timescale :'+ audioTrack.timescale);
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tracks.audio = {
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container : 'audio/mp4',
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codec : audioTrack.codec,
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@ -91,6 +101,7 @@ class MP4Remuxer {
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}
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if (videoTrack.sps && videoTrack.pps && videoSamples.length) {
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videoTrack.timescale = this.MP4_TIMESCALE;
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tracks.video = {
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container : 'video/mp4',
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codec : videoTrack.codec,
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@ -167,7 +178,12 @@ class MP4Remuxer {
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}
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mp4Sample.duration = sampleDuration;
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} else {
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var nextAvcDts = this.nextAvcDts,delta;
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let nextAvcDts, delta;
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if (contiguous) {
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nextAvcDts = this.nextAvcDts;
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} else {
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nextAvcDts = timeOffset*pesTimeScale;
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}
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// first AVC sample of video track, normalize PTS/DTS
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ptsnorm = this._PTSNormalize(pts, nextAvcDts);
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dtsnorm = this._PTSNormalize(dts, nextAvcDts);
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@ -252,6 +268,7 @@ class MP4Remuxer {
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pesTimeScale = this.PES_TIMESCALE,
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mp4timeScale = track.timescale,
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pes2mp4ScaleFactor = pesTimeScale/mp4timeScale,
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expectedSampleDuration = track.timescale * 1024 / track.audiosamplerate,
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aacSample, mp4Sample,
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unit,
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mdat, moof,
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@ -276,16 +293,22 @@ class MP4Remuxer {
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ptsnorm = this._PTSNormalize(pts, lastDTS);
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dtsnorm = this._PTSNormalize(dts, lastDTS);
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// let's compute sample duration.
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// there should be 1024 audio samples in one AAC frame
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// sample Duration should be close to expectedSampleDuration
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mp4Sample.duration = (dtsnorm - lastDTS) / pes2mp4ScaleFactor;
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if(Math.abs(mp4Sample.duration - 1024) > 10) {
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// not expected to happen ...
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logger.log(`invalid AAC sample duration at PTS ${Math.round(pts/90)},should be 1024,found :${Math.round(mp4Sample.duration)}`);
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if(Math.abs(mp4Sample.duration - expectedSampleDuration) > expectedSampleDuration/10) {
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// more than 10% diff between sample duration and expectedSampleDuration .... lets log that
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logger.log(`invalid AAC sample duration at PTS ${Math.round(pts/90)},should be 1024,found :${Math.round(mp4Sample.duration*track.audiosamplerate/track.timescale)}`);
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}
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mp4Sample.duration = 1024;
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dtsnorm = 1024 * pes2mp4ScaleFactor + lastDTS;
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// always adjust sample duration to avoid av sync issue
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mp4Sample.duration = expectedSampleDuration;
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dtsnorm = expectedSampleDuration * pes2mp4ScaleFactor + lastDTS;
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} else {
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var nextAacPts = this.nextAacPts,delta;
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let nextAacPts, delta;
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if (contiguous) {
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nextAacPts = this.nextAacPts;
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} else {
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nextAacPts = timeOffset*pesTimeScale;
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}
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ptsnorm = this._PTSNormalize(pts, nextAacPts);
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dtsnorm = this._PTSNormalize(dts, nextAacPts);
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delta = Math.round(1000 * (ptsnorm - nextAacPts) / pesTimeScale);
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@ -9,15 +9,10 @@ class PassThroughRemuxer {
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this.ISGenerated = false;
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}
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get passthrough() {
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return true;
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}
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get timescale() {
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return 0;
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}
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destroy() {
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}
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