update components

This commit is contained in:
Luke Pulverenti 2016-02-03 18:00:01 -05:00
parent 59ea1c2f7d
commit cf577ba8eb
1136 changed files with 59263 additions and 576 deletions

View file

@ -54,13 +54,14 @@ class MSEMediaController extends EventHandler {
}
startLoad() {
if (this.levels && this.media) {
if (this.levels) {
this.startInternal();
if (this.lastCurrentTime) {
logger.log(`seeking @ ${this.lastCurrentTime}`);
var media = this.media, lastCurrentTime = this.lastCurrentTime;
if (media && lastCurrentTime) {
logger.log(`seeking @ ${lastCurrentTime}`);
if (!this.lastPaused) {
logger.log('resuming video');
this.media.play();
media.play();
}
this.state = State.IDLE;
} else {
@ -70,7 +71,7 @@ class MSEMediaController extends EventHandler {
this.nextLoadPosition = this.startPosition = this.lastCurrentTime;
this.tick();
} else {
logger.warn('cannot start loading as either manifest not parsed or video not attached');
logger.warn('cannot start loading as manifest not parsed yet');
}
}
@ -666,7 +667,7 @@ class MSEMediaController extends EventHandler {
to avoid rounding issues/infinite loop,
only flush buffer range of length greater than 500ms.
*/
if (flushEnd - flushStart > 0.5) {
if (Math.min(flushEnd,bufEnd) - flushStart > 0.5) {
logger.log(`flush ${type} [${flushStart},${flushEnd}], of [${bufStart},${bufEnd}], pos:${this.media.currentTime}`);
sb.remove(flushStart, flushEnd);
return false;
@ -679,6 +680,8 @@ class MSEMediaController extends EventHandler {
return false;
}
}
} else {
logger.warn('abort flushing too many retries');
}
/* after successful buffer flushing, rebuild buffer Range array
@ -1221,7 +1224,7 @@ _checkBuffer() {
jumpThreshold = 0;
} else {
// playhead not moving AND media playing
logger.log('playback seems stuck');
logger.log(`playback seems stuck @${currentTime}`);
if(!this.stalled) {
this.hls.trigger(Event.ERROR, {type: ErrorTypes.MEDIA_ERROR, details: ErrorDetails.BUFFER_STALLED_ERROR, fatal: false});
this.stalled = true;

View file

@ -16,11 +16,11 @@ import ID3 from '../demux/id3';
static probe(data) {
// check if data contains ID3 timestamp and ADTS sync worc
var id3 = new ID3(data), adtsStartOffset,len;
var id3 = new ID3(data), offset,len;
if(id3.hasTimeStamp) {
// look for ADTS header (0xFFFx)
for (adtsStartOffset = id3.length, len = data.length; adtsStartOffset < len - 1; adtsStartOffset++) {
if ((data[adtsStartOffset] === 0xff) && (data[adtsStartOffset+1] & 0xf0) === 0xf0) {
for (offset = id3.length, len = data.length; offset < len - 1; offset++) {
if ((data[offset] === 0xff) && (data[offset+1] & 0xf0) === 0xf0) {
//logger.log('ADTS sync word found !');
return true;
}
@ -35,46 +35,47 @@ import ID3 from '../demux/id3';
var track = this._aacTrack,
id3 = new ID3(data),
pts = 90*id3.timeStamp,
config, adtsFrameSize, adtsStartOffset, adtsHeaderLen, stamp, nbSamples, len, aacSample;
config, frameLength, frameDuration, frameIndex, offset, headerLength, stamp, len, aacSample;
// look for ADTS header (0xFFFx)
for (adtsStartOffset = id3.length, len = data.length; adtsStartOffset < len - 1; adtsStartOffset++) {
if ((data[adtsStartOffset] === 0xff) && (data[adtsStartOffset+1] & 0xf0) === 0xf0) {
for (offset = id3.length, len = data.length; offset < len - 1; offset++) {
if ((data[offset] === 0xff) && (data[offset+1] & 0xf0) === 0xf0) {
break;
}
}
if (!track.audiosamplerate) {
config = ADTS.getAudioConfig(this.observer,data, adtsStartOffset, 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 * duration;
track.timescale = config.samplerate;
track.duration = config.samplerate * duration;
logger.log(`parsed codec:${track.codec},rate:${config.samplerate},nb channel:${config.channelCount}`);
}
nbSamples = 0;
while ((adtsStartOffset + 5) < len) {
frameIndex = 0;
frameDuration = 1024 * 90000 / track.audiosamplerate;
while ((offset + 5) < len) {
// The protection skip bit tells us if we have 2 bytes of CRC data at the end of the ADTS header
headerLength = (!!(data[offset + 1] & 0x01) ? 7 : 9);
// retrieve frame size
adtsFrameSize = ((data[adtsStartOffset + 3] & 0x03) << 11);
// byte 4
adtsFrameSize |= (data[adtsStartOffset + 4] << 3);
// byte 5
adtsFrameSize |= ((data[adtsStartOffset + 5] & 0xE0) >>> 5);
adtsHeaderLen = (!!(data[adtsStartOffset + 1] & 0x01) ? 7 : 9);
adtsFrameSize -= adtsHeaderLen;
stamp = Math.round(pts + nbSamples * 1024 * 90000 / track.audiosamplerate);
frameLength = ((data[offset + 3] & 0x03) << 11) |
(data[offset + 4] << 3) |
((data[offset + 5] & 0xE0) >>> 5);
frameLength -= headerLength;
//stamp = pes.pts;
//console.log('AAC frame, offset/length/pts:' + (adtsStartOffset+7) + '/' + adtsFrameSize + '/' + stamp.toFixed(0));
if ((adtsFrameSize > 0) && ((adtsStartOffset + adtsHeaderLen + adtsFrameSize) <= len)) {
aacSample = {unit: data.subarray(adtsStartOffset + adtsHeaderLen, adtsStartOffset + adtsHeaderLen + adtsFrameSize), pts: stamp, dts: stamp};
if ((frameLength > 0) && ((offset + headerLength + frameLength) <= len)) {
stamp = pts + frameIndex * frameDuration;
//logger.log(`AAC frame, offset/length/total/pts:${offset+headerLength}/${frameLength}/${data.byteLength}/${(stamp/90).toFixed(0)}`);
aacSample = {unit: data.subarray(offset + headerLength, offset + headerLength + frameLength), pts: stamp, dts: stamp};
track.samples.push(aacSample);
track.len += adtsFrameSize;
adtsStartOffset += adtsFrameSize + adtsHeaderLen;
nbSamples++;
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;
}
}

View file

@ -568,8 +568,8 @@
track.audiosamplerate = config.samplerate;
track.channelCount = config.channelCount;
track.codec = config.codec;
track.timescale = this.remuxer.timescale;
track.duration = track.timescale * duration;
track.timescale = config.samplerate;
track.duration = config.samplerate * duration;
logger.log(`parsed codec:${track.codec},rate:${config.samplerate},nb channel:${config.channelCount}`);
}
frameIndex = 0;
@ -596,7 +596,7 @@
//stamp = pes.pts;
if ((frameLength > 0) && ((offset + headerLength + frameLength) <= len)) {
stamp = Math.round(pts + frameIndex * frameDuration);
stamp = pts + frameIndex * frameDuration;
//logger.log(`AAC frame, offset/length/total/pts:${offset+headerLength}/${frameLength}/${data.byteLength}/${(stamp/90).toFixed(0)}`);
aacSample = {unit: data.subarray(offset + headerLength, offset + headerLength + frameLength), pts: stamp, dts: stamp};
track.samples.push(aacSample);

View file

@ -42,7 +42,7 @@ class Hls {
debug: false,
maxBufferLength: 30,
maxBufferSize: 60 * 1000 * 1000,
maxBufferHole: 0.3,
maxBufferHole: 0.5,
maxSeekHole: 2,
liveSyncDurationCount:3,
liveMaxLatencyDurationCount: Infinity,

View file

@ -253,7 +253,8 @@ class MP4Remuxer {
var view,
offset = 8,
pesTimeScale = this.PES_TIMESCALE,
pes2mp4ScaleFactor = this.PES2MP4SCALEFACTOR,
mp4timeScale = track.timescale,
pes2mp4ScaleFactor = pesTimeScale/mp4timeScale,
aacSample, mp4Sample,
unit,
mdat, moof,
@ -262,15 +263,10 @@ class MP4Remuxer {
samples = [],
samples0 = [];
track.samples.forEach(aacSample => {
if(pts === undefined || aacSample.pts > pts) {
samples0.push(aacSample);
pts = aacSample.pts;
} else {
logger.warn('dropping past audio frame');
track.len -= aacSample.unit.byteLength;
}
track.samples.sort(function(a, b) {
return (a.pts-b.pts);
});
samples0 = track.samples;
while (samples0.length) {
aacSample = samples0.shift();
@ -282,13 +278,15 @@ class MP4Remuxer {
if (lastDTS !== undefined) {
ptsnorm = this._PTSNormalize(pts, lastDTS);
dtsnorm = this._PTSNormalize(dts, lastDTS);
// let's compute sample duration
// let's compute sample duration.
// there should be 1024 audio samples in one AAC frame
mp4Sample.duration = (dtsnorm - lastDTS) / pes2mp4ScaleFactor;
if (mp4Sample.duration < 0) {
if(Math.abs(mp4Sample.duration - 1024) > 10) {
// not expected to happen ...
logger.log(`invalid AAC sample duration at PTS:${aacSample.pts}:${mp4Sample.duration}`);
mp4Sample.duration = 0;
logger.log(`invalid AAC sample duration at PTS ${Math.round(pts/90)},should be 1024,found :${Math.round(mp4Sample.duration)}`);
}
mp4Sample.duration = 1024;
dtsnorm = 1024 * pes2mp4ScaleFactor + lastDTS;
} else {
var nextAacPts = this.nextAacPts,delta;
ptsnorm = this._PTSNormalize(pts, nextAacPts);