# All files / src/remux mp4-remuxer.js

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    12x

     /**
     * fMP4 remuxer
    */

    import AAC from './aac-helper';
    import MP4 from './mp4-generator';

    import Event from '../events';
    import { ErrorTypes, ErrorDetails } from '../errors';

    import { logger } from '../utils/logger';

    // 10 seconds
    const MAX_SILENT_FRAME_DURATION = 10 * 1000;

    class MP4Remuxer {
      constructor (observer, config, typeSupported, vendor) {
        this.observer = observer;
        this.config = config;
        this.typeSupported = typeSupported;
        const userAgent = navigator.userAgent;
        this.isSafari = vendor && vendor.indexOf('Apple') > -1 && userAgent && !userAgent.match('CriOS');
        this.ISGenerated = false;
      }

      destroy () {
      }

      resetTimeStamp (defaultTimeStamp) {
        this._initPTS = this._initDTS = defaultTimeStamp;
      }

      resetInitSegment () {
        this.ISGenerated = false;
      }

      remux (audioTrack, videoTrack, id3Track, textTrack, timeOffset, contiguous, accurateTimeOffset) {
        // generate Init Segment if needed
        if (!this.ISGenerated) {
          this.generateIS(audioTrack, videoTrack, timeOffset);
        }

        if (this.ISGenerated) {
          const nbAudioSamples = audioTrack.samples.length;
          const nbVideoSamples = videoTrack.samples.length;
          let audioTimeOffset = timeOffset;
          let videoTimeOffset = timeOffset;
          if (nbAudioSamples && nbVideoSamples) {
            // timeOffset is expected to be the offset of the first timestamp of this fragment (first DTS)
            // if first audio DTS is not aligned with first video DTS then we need to take that into account
            // when providing timeOffset to remuxAudio / remuxVideo. if we don't do that, there might be a permanent / small
            // drift between audio and video streams
            let audiovideoDeltaDts = (audioTrack.samples[0].dts - videoTrack.samples[0].dts) / videoTrack.inputTimeScale;
            audioTimeOffset += Math.max(0, audiovideoDeltaDts);
            videoTimeOffset += Math.max(0, -audiovideoDeltaDts);
          }
          // Purposefully remuxing audio before video, so that remuxVideo can use nextAudioPts, which is
          // calculated in remuxAudio.
          // logger.log('nb AAC samples:' + audioTrack.samples.length);
          if (nbAudioSamples) {
            // if initSegment was generated without video samples, regenerate it again
            if (!audioTrack.timescale) {
              logger.warn('regenerate InitSegment as audio detected');
              this.generateIS(audioTrack, videoTrack, timeOffset);
            }
            let audioData = this.remuxAudio(audioTrack, audioTimeOffset, contiguous, accurateTimeOffset);
            // logger.log('nb AVC samples:' + videoTrack.samples.length);
            if (nbVideoSamples) {
              let audioTrackLength;
              if (audioData) {
                audioTrackLength = audioData.endPTS - audioData.startPTS;
              }

              // if initSegment was generated without video samples, regenerate it again
              if (!videoTrack.timescale) {
                logger.warn('regenerate InitSegment as video detected');
                this.generateIS(audioTrack, videoTrack, timeOffset);
              }
              this.remuxVideo(videoTrack, videoTimeOffset, contiguous, audioTrackLength, accurateTimeOffset);
            }
          } else {
            // logger.log('nb AVC samples:' + videoTrack.samples.length);
            if (nbVideoSamples) {
              let videoData = this.remuxVideo(videoTrack, videoTimeOffset, contiguous, 0, accurateTimeOffset);
              if (videoData && audioTrack.codec) {
                this.remuxEmptyAudio(audioTrack, audioTimeOffset, contiguous, videoData);
              }
            }
          }
        }
        // logger.log('nb ID3 samples:' + audioTrack.samples.length);
        if (id3Track.samples.length) {
          this.remuxID3(id3Track, timeOffset);
        }

        // logger.log('nb ID3 samples:' + audioTrack.samples.length);
        if (textTrack.samples.length) {
          this.remuxText(textTrack, timeOffset);
        }

        // notify end of parsing
        this.observer.trigger(Event.FRAG_PARSED);
      }

      generateIS (audioTrack, videoTrack, timeOffset) {
        let observer = this.observer,
          audioSamples = audioTrack.samples,
          videoSamples = videoTrack.samples,
          typeSupported = this.typeSupported,
          container = 'audio/mp4',
          tracks = {},
          data = { tracks: tracks },
          computePTSDTS = (this._initPTS === undefined),
          initPTS, initDTS;

        if (computePTSDTS) {
          initPTS = initDTS = Infinity;
        }

        if (audioTrack.config && audioSamples.length) {
          // let's use audio sampling rate as MP4 time scale.
          // rationale is that there is a integer nb of audio frames per audio sample (1024 for AAC)
          // 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(`audio sampling rate : ${audioTrack.samplerate}`);
          if (!audioTrack.isAAC) {
            if (typeSupported.mpeg) { // Chrome and Safari
              container = 'audio/mpeg';
              audioTrack.codec = '';
            } else if (typeSupported.mp3) { // Firefox
              audioTrack.codec = 'mp3';
            }
          }
          tracks.audio = {
            container: container,
            codec: audioTrack.codec,
            initSegment: !audioTrack.isAAC && typeSupported.mpeg ? new Uint8Array() : MP4.initSegment([audioTrack]),
            metadata: {
              channelCount: audioTrack.channelCount
            }
          };
          if (computePTSDTS) {
            // remember first PTS of this demuxing context. for audio, PTS = DTS
            initPTS = initDTS = audioSamples[0].pts - audioTrack.inputTimeScale * timeOffset;
          }
        }

        if (videoTrack.sps && videoTrack.pps && videoSamples.length) {
          // let's use input time scale as MP4 video timescale
          // we use input time scale straight away to avoid rounding issues on frame duration / cts computation
          const inputTimeScale = videoTrack.inputTimeScale;
          videoTrack.timescale = inputTimeScale;
          tracks.video = {
            container: 'video/mp4',
            codec: videoTrack.codec,
            initSegment: MP4.initSegment([videoTrack]),
            metadata: {
              width: videoTrack.width,
              height: videoTrack.height
            }
          };
          if (computePTSDTS) {
            initPTS = Math.min(initPTS, videoSamples[0].pts - inputTimeScale * timeOffset);
            initDTS = Math.min(initDTS, videoSamples[0].dts - inputTimeScale * timeOffset);
            this.observer.trigger(Event.INIT_PTS_FOUND, { initPTS: initPTS });
          }
        }

        if (Object.keys(tracks).length) {
          observer.trigger(Event.FRAG_PARSING_INIT_SEGMENT, data);
          this.ISGenerated = true;
          if (computePTSDTS) {
            this._initPTS = initPTS;
            this._initDTS = initDTS;
          }
        } else {
          observer.trigger(Event.ERROR, { type: ErrorTypes.MEDIA_ERROR, details: ErrorDetails.FRAG_PARSING_ERROR, fatal: false, reason: 'no audio/video samples found' });
        }
      }

      remuxVideo (track, timeOffset, contiguous, audioTrackLength, accurateTimeOffset) {
        let offset = 8,
          timeScale = track.timescale,
          mp4SampleDuration,
          mdat, moof,
          firstPTS, firstDTS,
          nextDTS,
          lastPTS, lastDTS,
          inputSamples = track.samples,
          outputSamples = [],
          nbSamples = inputSamples.length,
          ptsNormalize = this._PTSNormalize,
          initDTS = this._initDTS;

        // for (let i = 0; i < track.samples.length; i++) {
        //   let avcSample = track.samples[i];
        //   let units = avcSample.units;
        //   let unitsString = '';
        //   for (let j = 0; j < units.length ; j++) {
        //     unitsString += units[j].type + ',';
        //     if (units[j].data.length < 500) {
        //       unitsString += Hex.hexDump(units[j].data);
        //     }
        //   }
        //   logger.log(avcSample.pts + '/' + avcSample.dts + ',' + unitsString + avcSample.units.length);
        // }

        // if parsed fragment is contiguous with last one, let's use last DTS value as reference
        let nextAvcDts = this.nextAvcDts;

        const isSafari = this.isSafari;

        if (nbSamples === 0) {
          return;
        }

        // Safari does not like overlapping DTS on consecutive fragments. let's use nextAvcDts to overcome this if fragments are consecutive
        if (isSafari) {
          // also consider consecutive fragments as being contiguous (even if a level switch occurs),
          // for sake of clarity:
          // consecutive fragments are frags with
          //  - less than 100ms gaps between new time offset (if accurate) and next expected PTS OR
          //  - less than 200 ms PTS gaps (timeScale/5)
          contiguous |= (inputSamples.length && nextAvcDts &&
                         ((accurateTimeOffset && Math.abs(timeOffset - nextAvcDts / timeScale) < 0.1) ||
                          Math.abs((inputSamples[0].pts - nextAvcDts - initDTS)) < timeScale / 5)
          );
        }

        if (!contiguous) {
          // if not contiguous, let's use target timeOffset
          nextAvcDts = timeOffset * timeScale;
        }

        // PTS is coded on 33bits, and can loop from -2^32 to 2^32
        // ptsNormalize will make PTS/DTS value monotonic, we use last known DTS value as reference value
        inputSamples.forEach(function (sample) {
          sample.pts = ptsNormalize(sample.pts - initDTS, nextAvcDts);
          sample.dts = ptsNormalize(sample.dts - initDTS, nextAvcDts);
        });

        // sort video samples by DTS then PTS then demux id order
        inputSamples.sort(function (a, b) {
          const deltadts = a.dts - b.dts;
          const deltapts = a.pts - b.pts;
          return deltadts || (deltapts || (a.id - b.id));
        });

        // handle broken streams with PTS < DTS, tolerance up 200ms (18000 in 90kHz timescale)
        let PTSDTSshift = inputSamples.reduce((prev, curr) => Math.max(Math.min(prev, curr.pts - curr.dts), -18000), 0);
        if (PTSDTSshift < 0) {
          logger.warn(`PTS < DTS detected in video samples, shifting DTS by ${Math.round(PTSDTSshift / 90)} ms to overcome this issue`);
          for (let i = 0; i < inputSamples.length; i++) {
            inputSamples[i].dts += PTSDTSshift;
          }
        }

        // compute first DTS and last DTS, normalize them against reference value
        let sample = inputSamples[0];
        firstDTS = Math.max(sample.dts, 0);
        firstPTS = Math.max(sample.pts, 0);

        // check timestamp continuity accross consecutive fragments (this is to remove inter-fragment gap/hole)
        let delta = Math.round((firstDTS - nextAvcDts) / 90);
        // if fragment are contiguous, detect hole/overlapping between fragments
        if (contiguous) {
          if (delta) {
            if (delta > 1) {
              logger.log(`AVC:${delta} ms hole between fragments detected,filling it`);
            } else if (delta < -1) {
              logger.log(`AVC:${(-delta)} ms overlapping between fragments detected`);
            }

            // remove hole/gap : set DTS to next expected DTS
            firstDTS = nextAvcDts;
            inputSamples[0].dts = firstDTS;
            // offset PTS as well, ensure that PTS is smaller or equal than new DTS
            firstPTS = Math.max(firstPTS - delta, nextAvcDts);
            inputSamples[0].pts = firstPTS;
            logger.log(`Video/PTS/DTS adjusted: ${Math.round(firstPTS / 90)}/${Math.round(firstDTS / 90)},delta:${delta} ms`);
          }
        }
        nextDTS = firstDTS;

        // compute lastPTS/lastDTS
        sample = inputSamples[inputSamples.length - 1];
        lastDTS = Math.max(sample.dts, 0);
        lastPTS = Math.max(sample.pts, 0, lastDTS);

        // on Safari let's signal the same sample duration for all samples
        // sample duration (as expected by trun MP4 boxes), should be the delta between sample DTS
        // set this constant duration as being the avg delta between consecutive DTS.
        if (isSafari) {
          mp4SampleDuration = Math.round((lastDTS - firstDTS) / (inputSamples.length - 1));
        }

        let nbNalu = 0, naluLen = 0;
        for (let i = 0; i < nbSamples; i++) {
          // compute total/avc sample length and nb of NAL units
          let sample = inputSamples[i], units = sample.units, nbUnits = units.length, sampleLen = 0;
          for (let j = 0; j < nbUnits; j++) {
            sampleLen += units[j].data.length;
          }

          naluLen += sampleLen;
          nbNalu += nbUnits;
          sample.length = sampleLen;

          // normalize PTS/DTS
          if (isSafari) {
            // sample DTS is computed using a constant decoding offset (mp4SampleDuration) between samples
            sample.dts = firstDTS + i * mp4SampleDuration;
          } else {
            // ensure sample monotonic DTS
            sample.dts = Math.max(sample.dts, firstDTS);
          }
          // ensure that computed value is greater or equal than sample DTS
          sample.pts = Math.max(sample.pts, sample.dts);
        }

        /* concatenate the video data and construct the mdat in place
          (need 8 more bytes to fill length and mpdat type) */
        let mdatSize = naluLen + (4 * nbNalu) + 8;
        try {
          mdat = new Uint8Array(mdatSize);
        } catch (err) {
          this.observer.trigger(Event.ERROR, { type: ErrorTypes.MUX_ERROR, details: ErrorDetails.REMUX_ALLOC_ERROR, fatal: false, bytes: mdatSize, reason: `fail allocating video mdat ${mdatSize}` });
          return;
        }
        let view = new DataView(mdat.buffer);
        view.setUint32(0, mdatSize);
        mdat.set(MP4.types.mdat, 4);

        for (let i = 0; i < nbSamples; i++) {
          let avcSample = inputSamples[i],
            avcSampleUnits = avcSample.units,
            mp4SampleLength = 0,
            compositionTimeOffset;
          // convert NALU bitstream to MP4 format (prepend NALU with size field)
          for (let j = 0, nbUnits = avcSampleUnits.length; j < nbUnits; j++) {
            let unit = avcSampleUnits[j],
              unitData = unit.data,
              unitDataLen = unit.data.byteLength;
            view.setUint32(offset, unitDataLen);
            offset += 4;
            mdat.set(unitData, offset);
            offset += unitDataLen;
            mp4SampleLength += 4 + unitDataLen;
          }

          if (!isSafari) {
            // expected sample duration is the Decoding Timestamp diff of consecutive samples
            if (i < nbSamples - 1) {
              mp4SampleDuration = inputSamples[i + 1].dts - avcSample.dts;
            } else {
              let config = this.config,
                lastFrameDuration = avcSample.dts - inputSamples[i > 0 ? i - 1 : i].dts;
              if (config.stretchShortVideoTrack) {
                // In some cases, a segment's audio track duration may exceed the video track duration.
                // Since we've already remuxed audio, and we know how long the audio track is, we look to
                // see if the delta to the next segment is longer than maxBufferHole.
                // If so, playback would potentially get stuck, so we artificially inflate
                // the duration of the last frame to minimize any potential gap between segments.
                let maxBufferHole = config.maxBufferHole,
                  gapTolerance = Math.floor(maxBufferHole * timeScale),
                  deltaToFrameEnd = (audioTrackLength ? firstPTS + audioTrackLength * timeScale : this.nextAudioPts) - avcSample.pts;
                if (deltaToFrameEnd > gapTolerance) {
                  // We subtract lastFrameDuration from deltaToFrameEnd to try to prevent any video
                  // frame overlap. maxBufferHole should be >> lastFrameDuration anyway.
                  mp4SampleDuration = deltaToFrameEnd - lastFrameDuration;
                  if (mp4SampleDuration < 0) {
                    mp4SampleDuration = lastFrameDuration;
                  }

                  logger.log(`It is approximately ${deltaToFrameEnd / 90} ms to the next segment; using duration ${mp4SampleDuration / 90} ms for the last video frame.`);
                } else {
                  mp4SampleDuration = lastFrameDuration;
                }
              } else {
                mp4SampleDuration = lastFrameDuration;
              }
            }
            compositionTimeOffset = Math.round(avcSample.pts - avcSample.dts);
          } else {
            compositionTimeOffset = Math.max(0, mp4SampleDuration * Math.round((avcSample.pts - avcSample.dts) / mp4SampleDuration));
          }

          // console.log('PTS/DTS/initDTS/normPTS/normDTS/relative PTS : ${avcSample.pts}/${avcSample.dts}/${initDTS}/${ptsnorm}/${dtsnorm}/${(avcSample.pts/4294967296).toFixed(3)}');
          outputSamples.push({
            size: mp4SampleLength,
            // constant duration
            duration: mp4SampleDuration,
            cts: compositionTimeOffset,
            flags: {
              isLeading: 0,
              isDependedOn: 0,
              hasRedundancy: 0,
              degradPrio: 0,
              dependsOn: avcSample.key ? 2 : 1,
              isNonSync: avcSample.key ? 0 : 1
            }
          });
        }
        // next AVC sample DTS should be equal to last sample DTS + last sample duration (in PES timescale)
        this.nextAvcDts = lastDTS + mp4SampleDuration;
        let dropped = track.dropped;
        track.len = 0;
        track.nbNalu = 0;
        track.dropped = 0;
        if (outputSamples.length && navigator.userAgent.toLowerCase().indexOf('chrome') > -1) {
          let flags = outputSamples[0].flags;
          // chrome workaround, mark first sample as being a Random Access Point to avoid sourcebuffer append issue
          // https://code.google.com/p/chromium/issues/detail?id=229412
          flags.dependsOn = 2;
          flags.isNonSync = 0;
        }
        track.samples = outputSamples;
        moof = MP4.moof(track.sequenceNumber++, firstDTS, track);
        track.samples = [];

        let data = {
          data1: moof,
          data2: mdat,
          startPTS: firstPTS / timeScale,
          endPTS: (lastPTS + mp4SampleDuration) / timeScale,
          startDTS: firstDTS / timeScale,
          endDTS: this.nextAvcDts / timeScale,
          type: 'video',
          hasAudio: false,
          hasVideo: true,
          nb: outputSamples.length,
          dropped: dropped
        };
        this.observer.trigger(Event.FRAG_PARSING_DATA, data);
        return data;
      }

      remuxAudio (track, timeOffset, contiguous, accurateTimeOffset) {
        const inputTimeScale = track.inputTimeScale,
          mp4timeScale = track.timescale,
          scaleFactor = inputTimeScale / mp4timeScale,
          mp4SampleDuration = track.isAAC ? 1024 : 1152,
          inputSampleDuration = mp4SampleDuration * scaleFactor,
          ptsNormalize = this._PTSNormalize,
          initDTS = this._initDTS,
          rawMPEG = !track.isAAC && this.typeSupported.mpeg;

        let offset,
          mp4Sample,
          fillFrame,
          mdat, moof,
          firstPTS, lastPTS,
          inputSamples = track.samples,
          outputSamples = [],
          nextAudioPts = this.nextAudioPts;

        // for audio samples, also consider consecutive fragments as being contiguous (even if a level switch occurs),
        // for sake of clarity:
        // consecutive fragments are frags with
        //  - less than 100ms gaps between new time offset (if accurate) and next expected PTS OR
        //  - less than 20 audio frames distance
        // contiguous fragments are consecutive fragments from same quality level (same level, new SN = old SN + 1)
        // this helps ensuring audio continuity
        // and this also avoids audio glitches/cut when switching quality, or reporting wrong duration on first audio frame
        contiguous |= (inputSamples.length && nextAudioPts &&
                       ((accurateTimeOffset && Math.abs(timeOffset - nextAudioPts / inputTimeScale) < 0.1) ||
                        Math.abs((inputSamples[0].pts - nextAudioPts - initDTS)) < 20 * inputSampleDuration)
        );

        // compute normalized PTS
        inputSamples.forEach(function (sample) {
          sample.pts = sample.dts = ptsNormalize(sample.pts - initDTS, timeOffset * inputTimeScale);
        });

        // filter out sample with negative PTS that are not playable anyway
        // if we don't remove these negative samples, they will shift all audio samples forward.
        // leading to audio overlap between current / next fragment
        inputSamples = inputSamples.filter(function (sample) {
          return sample.pts >= 0;
        });

        // in case all samples have negative PTS, and have been filtered out, return now
        if (inputSamples.length === 0) {
          return;
        }

        if (!contiguous) {
          if (!accurateTimeOffset) {
            // if frag are mot contiguous and if we cant trust time offset, let's use first sample PTS as next audio PTS
            nextAudioPts = inputSamples[0].pts;
          } else {
            // if timeOffset is accurate, let's use it as predicted next audio PTS
            nextAudioPts = timeOffset * inputTimeScale;
          }
        }

        // If the audio track is missing samples, the frames seem to get "left-shifted" within the
        // resulting mp4 segment, causing sync issues and leaving gaps at the end of the audio segment.
        // In an effort to prevent this from happening, we inject frames here where there are gaps.
        // When possible, we inject a silent frame; when that's not possible, we duplicate the last
        // frame.

        if (track.isAAC) {
          const maxAudioFramesDrift = this.config.maxAudioFramesDrift;
          for (let i = 0, nextPts = nextAudioPts; i < inputSamples.length;) {
            // First, let's see how far off this frame is from where we expect it to be
            var sample = inputSamples[i], delta;
            let pts = sample.pts;
            delta = pts - nextPts;

            const duration = Math.abs(1000 * delta / inputTimeScale);

            // If we're overlapping by more than a duration, drop this sample
            if (delta <= -maxAudioFramesDrift * inputSampleDuration) {
              logger.warn(`Dropping 1 audio frame @ ${(nextPts / inputTimeScale).toFixed(3)}s due to ${Math.round(duration)} ms overlap.`);
              inputSamples.splice(i, 1);
              track.len -= sample.unit.length;
              // Don't touch nextPtsNorm or i
            } // eslint-disable-line brace-style

            // Insert missing frames if:
            // 1: We're more than maxAudioFramesDrift frame away
            // 2: Not more than MAX_SILENT_FRAME_DURATION away
            // 3: currentTime (aka nextPtsNorm) is not 0
            else if (delta >= maxAudioFramesDrift * inputSampleDuration && duration < MAX_SILENT_FRAME_DURATION && nextPts) {
              let missing = Math.round(delta / inputSampleDuration);
              logger.warn(`Injecting ${missing} audio frame @ ${(nextPts / inputTimeScale).toFixed(3)}s due to ${Math.round(1000 * delta / inputTimeScale)} ms gap.`);
              for (let j = 0; j < missing; j++) {
                let newStamp = Math.max(nextPts, 0);
                fillFrame = AAC.getSilentFrame(track.manifestCodec || track.codec, track.channelCount);
                if (!fillFrame) {
                  logger.log('Unable to get silent frame for given audio codec; duplicating last frame instead.');
                  fillFrame = sample.unit.subarray();
                }
                inputSamples.splice(i, 0, { unit: fillFrame, pts: newStamp, dts: newStamp });
                track.len += fillFrame.length;
                nextPts += inputSampleDuration;
                i++;
              }

              // Adjust sample to next expected pts
              sample.pts = sample.dts = nextPts;
              nextPts += inputSampleDuration;
              i++;
            } else {
            // Otherwise, just adjust pts
              if (Math.abs(delta) > (0.1 * inputSampleDuration)) {
                // logger.log(`Invalid frame delta ${Math.round(delta + inputSampleDuration)} at PTS ${Math.round(pts / 90)} (should be ${Math.round(inputSampleDuration)}).`);
              }
              sample.pts = sample.dts = nextPts;
              nextPts += inputSampleDuration;
              i++;
            }
          }
        }

        for (let j = 0, nbSamples = inputSamples.length; j < nbSamples; j++) {
          let audioSample = inputSamples[j];
          let unit = audioSample.unit;
          let pts = audioSample.pts;
          // logger.log(`Audio/PTS:${Math.round(pts/90)}`);
          // if not first sample
          if (lastPTS !== undefined) {
            mp4Sample.duration = Math.round((pts - lastPTS) / scaleFactor);
          } else {
            let delta = Math.round(1000 * (pts - nextAudioPts) / inputTimeScale),
              numMissingFrames = 0;
            // if fragment are contiguous, detect hole/overlapping between fragments
            // contiguous fragments are consecutive fragments from same quality level (same level, new SN = old SN + 1)
            if (contiguous && track.isAAC) {
              // log delta
              if (delta) {
                if (delta > 0 && delta < MAX_SILENT_FRAME_DURATION) {
                  numMissingFrames = Math.round((pts - nextAudioPts) / inputSampleDuration);
                  logger.log(`${delta} ms hole between AAC samples detected,filling it`);
                  if (numMissingFrames > 0) {
                    fillFrame = AAC.getSilentFrame(track.manifestCodec || track.codec, track.channelCount);
                    if (!fillFrame) {
                      fillFrame = unit.subarray();
                    }

                    track.len += numMissingFrames * fillFrame.length;
                  }
                  // if we have frame overlap, overlapping for more than half a frame duraion
                } else if (delta < -12) {
                  // drop overlapping audio frames... browser will deal with it
                  logger.log(`drop overlapping AAC sample, expected/parsed/delta:${(nextAudioPts / inputTimeScale).toFixed(3)}s/${(pts / inputTimeScale).toFixed(3)}s/${(-delta)}ms`);
                  track.len -= unit.byteLength;
                  continue;
                }
                // set PTS/DTS to expected PTS/DTS
                pts = nextAudioPts;
              }
            }
            // remember first PTS of our audioSamples
            firstPTS = pts;
            if (track.len > 0) {
              /* concatenate the audio data and construct the mdat in place
                (need 8 more bytes to fill length and mdat type) */
              let mdatSize = rawMPEG ? track.len : track.len + 8;
              offset = rawMPEG ? 0 : 8;
              try {
                mdat = new Uint8Array(mdatSize);
              } catch (err) {
                this.observer.trigger(Event.ERROR, { type: ErrorTypes.MUX_ERROR, details: ErrorDetails.REMUX_ALLOC_ERROR, fatal: false, bytes: mdatSize, reason: `fail allocating audio mdat ${mdatSize}` });
                return;
              }
              if (!rawMPEG) {
                const view = new DataView(mdat.buffer);
                view.setUint32(0, mdatSize);
                mdat.set(MP4.types.mdat, 4);
              }
            } else {
              // no audio samples
              return;
            }
            for (let i = 0; i < numMissingFrames; i++) {
              fillFrame = AAC.getSilentFrame(track.manifestCodec || track.codec, track.channelCount);
              if (!fillFrame) {
                logger.log('Unable to get silent frame for given audio codec; duplicating this frame instead.');
                fillFrame = unit.subarray();
              }
              mdat.set(fillFrame, offset);
              offset += fillFrame.byteLength;
              mp4Sample = {
                size: fillFrame.byteLength,
                cts: 0,
                duration: 1024,
                flags: {
                  isLeading: 0,
                  isDependedOn: 0,
                  hasRedundancy: 0,
                  degradPrio: 0,
                  dependsOn: 1
                }
              };
              outputSamples.push(mp4Sample);
            }
          }
          mdat.set(unit, offset);
          let unitLen = unit.byteLength;
          offset += unitLen;
          // console.log('PTS/DTS/initDTS/normPTS/normDTS/relative PTS : ${audioSample.pts}/${audioSample.dts}/${initDTS}/${ptsnorm}/${dtsnorm}/${(audioSample.pts/4294967296).toFixed(3)}');
          mp4Sample = {
            size: unitLen,
            cts: 0,
            duration: 0,
            flags: {
              isLeading: 0,
              isDependedOn: 0,
              hasRedundancy: 0,
              degradPrio: 0,
              dependsOn: 1
            }
          };
          outputSamples.push(mp4Sample);
          lastPTS = pts;
        }
        let lastSampleDuration = 0;
        let nbSamples = outputSamples.length;
        // set last sample duration as being identical to previous sample
        if (nbSamples >= 2) {
          lastSampleDuration = outputSamples[nbSamples - 2].duration;
          mp4Sample.duration = lastSampleDuration;
        }
        if (nbSamples) {
          // next audio sample PTS should be equal to last sample PTS + duration
          this.nextAudioPts = nextAudioPts = lastPTS + scaleFactor * lastSampleDuration;
          // logger.log('Audio/PTS/PTSend:' + audioSample.pts.toFixed(0) + '/' + this.nextAacDts.toFixed(0));
          track.len = 0;
          track.samples = outputSamples;
          if (rawMPEG) {
            moof = new Uint8Array();
          } else {
            moof = MP4.moof(track.sequenceNumber++, firstPTS / scaleFactor, track);
          }

          track.samples = [];
          const start = firstPTS / inputTimeScale;
          const end = nextAudioPts / inputTimeScale;
          const audioData = {
            data1: moof,
            data2: mdat,
            startPTS: start,
            endPTS: end,
            startDTS: start,
            endDTS: end,
            type: 'audio',
            hasAudio: true,
            hasVideo: false,
            nb: nbSamples
          };
          this.observer.trigger(Event.FRAG_PARSING_DATA, audioData);
          return audioData;
        }
        return null;
      }

      remuxEmptyAudio (track, timeOffset, contiguous, videoData) {
        let inputTimeScale = track.inputTimeScale,
          mp4timeScale = track.samplerate ? track.samplerate : inputTimeScale,
          scaleFactor = inputTimeScale / mp4timeScale,
          nextAudioPts = this.nextAudioPts,

          // sync with video's timestamp
          startDTS = (nextAudioPts !== undefined ? nextAudioPts : videoData.startDTS * inputTimeScale) + this._initDTS,
          endDTS = videoData.endDTS * inputTimeScale + this._initDTS,
          // one sample's duration value
          sampleDuration = 1024,
          frameDuration = scaleFactor * sampleDuration,

          // samples count of this segment's duration
          nbSamples = Math.ceil((endDTS - startDTS) / frameDuration),

          // silent frame
          silentFrame = AAC.getSilentFrame(track.manifestCodec || track.codec, track.channelCount);

        logger.warn('remux empty Audio');
        // Can't remux if we can't generate a silent frame...
        if (!silentFrame) {
          logger.trace('Unable to remuxEmptyAudio since we were unable to get a silent frame for given audio codec!');
          return;
        }

        let samples = [];
        for (let i = 0; i < nbSamples; i++) {
          let stamp = startDTS + i * frameDuration;
          samples.push({ unit: silentFrame, pts: stamp, dts: stamp });
          track.len += silentFrame.length;
        }
        track.samples = samples;

        this.remuxAudio(track, timeOffset, contiguous);
      }

      remuxID3 (track, timeOffset) {
        let length = track.samples.length, sample;
        const inputTimeScale = track.inputTimeScale;
        const initPTS = this._initPTS;
        const initDTS = this._initDTS;
        // consume samples
        if (length) {
          for (let index = 0; index < length; index++) {
            sample = track.samples[index];
            // setting id3 pts, dts to relative time
            // using this._initPTS and this._initDTS to calculate relative time
            sample.pts = ((sample.pts - initPTS) / inputTimeScale);
            sample.dts = ((sample.dts - initDTS) / inputTimeScale);
          }
          this.observer.trigger(Event.FRAG_PARSING_METADATA, {
            samples: track.samples
          });
        }

        track.samples = [];
        timeOffset = timeOffset;
      }

      remuxText (track, timeOffset) {
        track.samples.sort(function (a, b) {
          return (a.pts - b.pts);
        });

        let length = track.samples.length, sample;
        const inputTimeScale = track.inputTimeScale;
        const initPTS = this._initPTS;
        // consume samples
        if (length) {
          for (let index = 0; index < length; index++) {
            sample = track.samples[index];
            // setting text pts, dts to relative time
            // using this._initPTS and this._initDTS to calculate relative time
            sample.pts = ((sample.pts - initPTS) / inputTimeScale);
          }
          this.observer.trigger(Event.FRAG_PARSING_USERDATA, {
            samples: track.samples
          });
        }

        track.samples = [];
        timeOffset = timeOffset;
      }

      _PTSNormalize (value, reference) {
        let offset;
        if (reference === undefined) {
          return value;
        }

        if (reference < value) {
          // - 2^33
          offset = -8589934592;
        } else {
          // + 2^33
          offset = 8589934592;
        }
        /* PTS is 33bit (from 0 to 2^33 -1)
          if diff between value and reference is bigger than half of the amplitude (2^32) then it means that
          PTS looping occured. fill the gap */
        while (Math.abs(value - reference) > 4294967296) {
          value += offset;
        }

        return value;
      }
    }

    export default MP4Remuxer;