Audio drift on Constant framerate

I am experiencing an audio drift problem on a specific set of videos when using Premiere Pro CC (2014) on Windows 7. The audio starts out well, but eventually drifts way out of sync. This already happens in the Source Monitor. I read that this can be due to the use of Variable framerate video. However, the video I have is Constant framerate. At first, I thought it might have been Premiere itself or my PC, but when I dropped a Blu-ray rip into Premiere, it worked perfectly. It turns out this happen consistently with all videos from the source website (Giant Bomb).
It seems to me Premiere misinterprets the footage. When I play the video in MPC-HC, the duration reads 2:51:10 but when I drop it into Premiere, it reports the duration as 2:51:00:03. I checked the Sequence settings, but they all seem to correspond with the source video. Even when I set it manually, it doesn't help.
Screenshot:
And here's the MediaInfo output:
General
Complete name : D:\Exchange\source\problem-file.mp4
Format : MPEG-4
Format profile : Base Media
Codec ID : isom
File size : 5.12 GiB
Duration : 2h 51mn
Overall bit rate mode : Variable
Overall bit rate : 4 281 Kbps
Writing application : Lavf55.19.104
Video
ID : 1
Format : AVC
Format/Info : Advanced Video Codec
Format profile : [email protected]
Format settings, CABAC : Yes
Format settings, ReFrames : 3 frames
Format settings, GOP : M=1, N=120
Codec ID : avc1
Codec ID/Info : Advanced Video Coding
Duration : 2h 51mn
Bit rate : 4 000 Kbps
Width : 1 280 pixels
Height : 720 pixels
Display aspect ratio : 16:9
Frame rate mode : Constant
Frame rate : 59.940 fps
Color space : YUV
Chroma subsampling : 4:2:0
Bit depth : 8 bits
Scan type : Progressive
Bits/(Pixel*Frame) : 0.072
Stream size : 4.78 GiB (93%)
Writing library : x264 core 140
Encoding settings : cabac=1 / ref=3 / deblock=1:0:0 / analyse=0x1:0x111 / me=hex / subme=7 / psy=1 / psy_rd=1.00:0.00 / mixed_ref=1 / me_range=16 / chroma_me=1 / trellis=1 / 8x8dct=0 / cqm=0 / deadzone=21,11 / fast_pskip=1 / chroma_qp_offset=-2 / threads=24 / lookahead_threads=4 / sliced_threads=0 / nr=0 / decimate=1 / interlaced=0 / bluray_compat=0 / constrained_intra=0 / bframes=0 / weightp=2 / keyint=120 / keyint_min=12 / scenecut=0 / intra_refresh=0 / rc_lookahead=40 / rc=abr / mbtree=1 / bitrate=4000 / ratetol=1.0 / qcomp=0.60 / qpmin=0 / qpmax=69 / qpstep=4 / vbv_maxrate=8000 / vbv_bufsize=8000 / nal_hrd=none / filler=0 / ip_ratio=1.40 / aq=1:1.00
Language : English
Tagged date : UTC 2014-07-30 04:09:12
Audio
ID : 2
Format : AAC
Format/Info : Advanced Audio Codec
Format profile : LC
Codec ID : 40
Duration : 2h 51mn
Bit rate mode : Variable
Bit rate : 128 Kbps
Maximum bit rate : 8 000 Kbps
Channel(s) : 2 channels
Channel positions : Front: L R
Sampling rate : 48.0 KHz
Compression mode : Lossy
Stream size : 157 MiB (3%)
Language : English
Tagged date : UTC 2014-07-30 04:09:12
Other #1
ID : 65536
Type : Hint
Format : RTP
Codec ID : rtp
Duration : 2h 51mn
Encoded date : UTC 2014-07-30 03:58:42
Tagged date : UTC 2014-07-30 04:09:12
Other #2
ID : 65537
Type : Hint
Format : RTP
Codec ID : rtp
Duration : 2h 51mn
Encoded date : UTC 2014-07-30 04:09:00
Tagged date : UTC 2014-07-30 04:09:12
Bit rate mode : VBR
Final note: Kdenlive under Linux seems to have no problem handling the video correctly.
Any ideas as to why this is happening?

Well, ladies and/or gentlemen, I am now content to call the file in my original post an anomaly. I downloaded two versions of a different video: one from Youtube and one from the site. Both worked fine and seemed perfectly synced up. Even the one from Youtube, which MediaInfo reports is Variable frame rate! That sort of blew my mind, as I expected that not to work. I then thought that the length might have something to do with it, so I downloaded another video of equal length and content (again with camera footage and gameplay footage). It was also synced. So there must just be some problem with the first file after all. I did spot some glitching in the video, so perhaps it's corrupted somewhere. Problem "solved" I suppose. Some additional notes/questions below. Thanks for your help in getting to the bottom of this mystery!
RoninEdits: Before doing all this, I tried your method. I made markers in the Source Monitor, but they didn't transfer to the timeline when I made a Sequence. I tried making those markers on the timeline itself, but then they didn't move when I used the Rate Stretch Tool. Do you have (a link to) instructions on how to properly do this? I'm a beginner in Premiere Pro and I couldn't figure it out.
J. Simon: I guess what I meant by standardized is that it fits the Base profile and 4.1 level standards of H.264. Incidentally, if I check the Sequence settings on the file that originate from the site -- the 60FPS HD files -- Premiere automatically assumes they are in fact AVC-Intra 100 720p. So maybe they did standardize these videos after all.

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    Looks like I left out an important piece of info, though...When I demux the audio stream to AC3 (its original format), and try to import it into DVD Studio Pro, I get the error message: "Variable bit rate audio assets cannot be imported." I am not sure how the client originally encoded their audio, but the fact that it stays in sync on the DVD led me to believe they didn't do anything wrong. Could this VBR be throwing the audio off during the demux? The only instances where I've noticed it's out of sync are when I export to an uncompressed format like AIFF or WAV, and can actually view it.

  • Audio drifts out of synch

    For me, this is a very inexplicable problem.
    Concert video. Video taken with 5 HD cameras, camera's audio at 48KHz. Audio was also separately recorded in a Pro Tools system, also 48 KHz.
    The video was later captured into FCP using the standard HD 1080 50i. Syncing up the videos was no big hassle. When adding the ProTools audio, it drifts out of sync.
    I found in the FCP manual about the fact that if audio and video is not synced using a master sync / time generator, the audio can drift.
    But how is this possible? I want to understand. I can understand the need for an external sync during recording, to more easily sync up the start point for each track when editing, but one second is one second long, exactly and 48000 samples is 48000 samples, wether recorded on one device or another. 48KHz is 48KHz and 1 second is one second, whatever the device.
    In my mind, the master program (in this case FCP) should read exactly 48000 / second regardless if the audio is captured on the cam or on any other 48KHz device. But it does not do this... it reads maybe 48090 - or whatnot - samples per second. The result is that the audio drifts ahead of the video. Why does it do this?
    This is not just in FCP. My friend is composing music on Logic Pro. The bas player (who lives in another city) records his bas lines on Logic Pro and sends the files (uncompressed). When my friend imports the bas recording into the master Logic project, the same thing happens... the imported file drifts.
    In my mind, a bit is a bit, a byte is a byte, a sample is a sample... the master-program should read each sample in its place, 48000 / second, in a quantified "time grid". So, in my mind, this just cant happen, but it does. Can anyone explain why, and if there is anything one can to in the program (FCP or Logic) when reading the files to ensure actual sample-time accuracy when reading imported files?

    Hi Per:
    I have been through this many times. The reason that the devices go out of sync is that their master clocks (the digital equivalent of a crystal in an analog device) are all beating to their own drummers, so to speak. The internal master clocks are all off from each other the second that they began recording. It's just how long does it take before those offsets become greater than one frame of video or film, right?
    This is why genlock and video TC sync were invented. What is more frustrating is that the devices do not all slip at the same speed either so usually, one nudge of the audio in one direction or another may line it up for a while, but later down the timeline, you may discover that the devices (camera tracks and the ProTools track) have all lost sync again. On a long show, you may have to do a lot of slicing and dicing to make everything stay in sync. This is the price you pay for not using really big, expensive, professional cameras, there has to be a trade off somewhere.
    The best way to mitigate this is to shoot with genlocked cameras with locked TC with the PT rig or a quality TC generator as the master clock. If the projects are shot with non-genlock prosumer type cameras, the next best solution is to rent a giant TC display and position just out of frame in a position that all five cameras can easily pan to, but won't pan to it accidentally as they frame their shots. The master TC display is driven by the PT system. You instruct the cameras ops to quickly photograph just a second or two of it every "whatever length you had before it slipped, probably once every five to ten minutes". In this way, when using multicame to edit, you can quickly locate the TC display frames and slip and slide the tracks accordingly.
    This can happen with any non-genlocked, non-TC reading devices. Usually I can get about 20 minutes between my cheap MD recorder and prosumer cameras before I see sync slip. Some devices have better, more steady master clocks, some have really lousy ones that slip quickly.
    You just have to bite the bullet and deal with it.
    Dan

  • Audio drift on FCP self-contained movie output

    I just output a 25 minute show as a self-contained FCP QT file, which I was was then going to throw into Compressor. The exported QT file started experiencing audio drift at about 7 minutes, and it looks like it's about a second off by the end of the show.
    In FCP, the sequence is in sync, al the way through the end. The sequence is 1920x1080 XDCAM 24p.
    I've never had this happen in a self-contained movie, and ideas what could be happening here?
    Thanks,
    Patrick
    Message was edited by: P.N. Barnes
    UPDATE: It looks like the audio may be off by the same amount starting at one point in the QT file through the rest of the show.

    Is the total duration of your sequence and your exported QuickTime movie exactly the same?
    I have seen FCP actually skipping frames on export of XDCAM HD422 sequences.
    Try importing your self-contained QuickTime movie and put it on a video layer above your original sequence. Zoom in at the end of your sequence and see if the total duration matches up. If not, you know you have issues with frames getting skipped on export to QuickTime Movie. If you look closely in the QuickTime Movie file you might even notice by looking at the timecode in QuickTime Player that certain frames are skipped. It is possible to spot these areas by looking for places with motion where you easily can see any jarring jumps caused by frame skipping.
    Here is a possible workaround. Copy the clips from your XDCAM HD sequence, paste them in a ProRes HQ sequence and export a QuickTime movie of that. It might also help just deleting all render files before exporting. Seems like FCP (or rather the XDCAM HD codec) gets confused by the render files on export when calculating the new GOP structure.

  • Audio drift in FEC (2)

    Sorry guys, more details
    Using compressor QuickTime Setting
    Apple Intermediate Codec (AIC)
    converted .AVI files to .mov
    Recorded Stereo and Mono files on Zoom + Sennheiser
    Audio at 48000 16 bit
    Using Logic Audio did Stereo mixdown to 44100 24 bit
    In Final Cut Express
    Using HDV-Apple Intermediate Codec 720p30 as Project Setting
    Video and audio files sync up fine visually in FCE but there is audio drift
    Fine at beginning but after say 3 minutes distinct drift
    I’ve tried various settings for Audio, changing 44100 to 48000. 24 bit to 43 & 16, same result
    Any ideas, please?

    Early Zooms were notorious for drift even when the audio was recorded at 48 khz/16 bit.
    There are only two ways to use 2nd system audio and be assured it will sync correctly.
    1. jam Timecode to camera and recorder (most consumer cams and recorders don't support this)
    2. loop the audio back to the camera and have it recorded with the video (take the feed off of the field mixer)
    If you do end up with out of sync material you have a couple of ways to deal with it.
    • if the error is consistent over a long take, speed up/slow down the audio by the correct percentage
    • if you have cuts in the video, adjust the audio back into sync at each cut.
    good luck.
    x

  • Audio drift in FEC

    Using compressor QuickTime Setting
    Apple Intermediate Codec (AIC)
    converted Video .AVI files to .mov
    Recorded Stereo and Mono files on Zoom + Sennheiser
    Audio at 48000 16 bit
    Using Logic Audio did Stereo mixdown to 44100 24 bit
    Video and audio files sync up fine visually in Final Cut Express but there is audio drift, fine at beginning but after say 3 minutes distinct drift
    Any ideas out there?

    I have had this exact same issue in Final Cut Pro with DV CAM footage and Zoom H4n / ProTools audio. The internal clock in the recording devices are not exactly 48000Hz (they may be 48003Hz etc) and so eventually the audio is out of sync with the image.
    I have found a solution by opening the audio file in Adobe Soundbooth and saving it as a 48KHz file. FCP then reads this perfectly and everything is in sync (this is with a 45minute duration file).
    Hope this helps!

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