[Ardour-Users] "Extract LTC from audio and align video" menu option missing from "Transcode/Import Video file" popup
robertlazarski at gmail.com
Wed Apr 4 15:00:56 PDT 2018
Hi Chris, thanks for the help! See my answers inline.
On Wed, Apr 4, 2018 at 1:50 PM, Chris Caudle <chris at chriscaudle.org> wrote:
> On Tue, April 3, 2018 9:07 pm, robertlazarski . wrote:
> > The default of auto mute is off on Free Run and RTC mode and I didn't
> > change it.
> Then assuming the behavior works like it seems to be documented (which if
> I understand correctly is that the time code is continuously output as
> long as the Zoom F8 is powered on) it seems that the video recorder should
> not have incorrect time code (i.e. not matching the audio).
> > The main issue might be some missing data in the first part of the LTC on
> > CH1 of the video.
> Missing code could be because the phone has trouble adjusting to the time
> code audio. As far as I know most phones would have some type of
> automatic gain control since you don't have microphone gain knobs on the
> phone, so perhaps it takes the phone AGC a few seconds to determine an
> appropriate gain setting for LTC. The sound of LTC is definitely not like
> typical audio signals that the AGC behavior would be optimized to handle.
Its a nasty signal. It might be severally clipping but I don't have meters
on it. I have it going thru a Irig DUO which I use as sort of a preamp to
usb adapter, that accepts 1/4 line level jacks. I'll try turning the volume
down on the Irig, maybe the OpenCamera app I am using too.
I have a KT 1176 clone (among several other compressors), should I pass the
signal thru it at 20:1 limit mode? I read somewhere you shouldn't compress
> I am looking at getting a hardware slate like a Denecke TS-3 - to keep my
> > sanity I only use software in post :-) .
> You could get a hardware slate, but seems expensive. If you were
> responding to my suggestion for a clap board, I meant old school,
> literally knock two sticks of wood together. You should be able to step
> frame by frame in the video to see the frame where the pieces hit
> together, and in the audio editor it should be pretty obvious where the
> impact occurs as long as the room is otherwise relatively quiet. It lets
> you line up by hand in the editor if you want, but also gives you a
> reference point to check the timecode in that video frame, and check the
> time code where Ardour lined up that point in the audio track.
I'm doing music performance videos with a lot of piano right now, so I have
a pretty clear "first strike" moment. Clapperboards, even old school, is
something I am trying to start using in my setup.
I'm fascinated by hardware slates - I prefer hardware in general as once in
a lifetime purchases - but I can't wrap my head around why they are so
expensive. Some of these hardware slates seem over a decade old. All they
seem to do is accept a timecode signal into a 1/4 jack once a day to sync,
and display some numbers. Sure they keep it accurate but my Zoom F8 does a
lot more and costs half the price.
> On Tue, April 3, 2018 9:43 pm, robertlazarski . wrote:
> >> That sounds like both recorders are free-running.
> > Might be, I tried using time of day timestamps and it didn't help.
> No, by free running we were referring to the case where for example the
> camera generates its own time code track, and the Zoom generates its own
> time code track, and even if they start out together they could drift
> apart over time due to inaccuracies in the individual oscillators used to
> track the time. You have previously clarified that the camera does not
> generate any time code, you just cable the time code output of the zoom
> into one channel of the camera audio track.
Still trying to get the terminology right, its important.
I didn't think this issue could cause a really large delay like over 10
seconds. I was expecting it to be a possible issue as I got everything
lined up to under 1 second.
> >> Note that both audio and video may import at some late position in the
> >> timeline. e.g. at 14:00:00:00 and may not be visible with default
> >> zoom-level.
> > I tried zooming in and out and didn't see an obvious misalignment.
> It wouldn't be a misalignment per se, just that timecode is defined as
> starting at one second after midnight, and covering a full 24 hours until
> 23:59:59 (hours:min:sec). I don't remember if Ardour has a way to specify
> what hour and minute your project timeline starts, but by default it will
> just assume 00:00:00 (or maybe 00:00:01, I forget), so if you have your
> timecode set to follow wall clock time, and you record at say early
> evening at 19:22:13, Ardour will happily import your audio into a project
> starting at 00:00:00 and place the audio at 19:22:13 on the timeline so
> that you end up with a project that will play back silence for over 19
> hours and 22 minutes if you place the playhead at the project beginning
> and hit play.
> Just a detail to be aware of.
Seems like all the examples of real world uses of timecode I found via
google, do not use wall clock time but rather start at 00:00.01. That's the
default of the F8. I am going to switch to that 00:00:01 format again at
some point soon, but debugging seems easier in a wall clock format. As you
mention, there is a lot of side issues to think about.
> > On my latest files I am attempting to use "time of day" timestamps in the
> > timecode, using the F8 "Int RTC" mode.
> > The sync error pattern I am noticing, is the first entries of LTC in the
> > MP4 file seem to be missing or corrupted. The POS from ltcdump never
> > starts at zero. When I open the raw MP4 I see no such obvious problem.
> What do you use to open the raw MP4?
mplayer on Linux. Immediately as expected, I hear the super loud LTC on
ch1, and a scratch track with a constant drum machine on ch2. The video and
audio both seems fine, but that ch1 LTC could be clipping the ADC on the
way in. For my next step, I am going to focus on lowering the ch1 audio
Speaking of MP4, it just occurred to me that the phone is probably going
> to be encoding the audio tracks with AAC (or some other lossy codec). I
> have no idea what effect that has on LTC decoding, and I could not find
> any reference to the effect of AAC on LTC recording with a quick google
> search. Maybe Robin knows, or I can ask around to see if anyone else has
> experience with that. Maybe the AAC encoding is causing some problems for
> the LTC decoder.
> I'll see if I can figure out a way to script up some tests for that. Can
> you check to see if you phone is using AAC for sure, and what bit rate?
> Should be able to tell pretty easily with ffmpeg -i on one of the phone
> video files, that should dump the codec and bit rate detected.
Seems like its using AAC, this is from 'ffmpeg -i' , let me know if it
would be helpul to paste the entire output - its big.
creation_time : 2018-04-04T00:33:00.000000Z
handler_name : VideoHandle
Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz,
stereo, fltp, 128 kb/s (default)
I see later that the audio you extracted from the phone video is wav, was
> that recorded uncompressed native, or was the original audio compressed
> and you converted to wav when you extracted from the video file?
The command that Robin recommended I use, 'sndfile-info -b' , only works on
wav = not mp4. I'm just using it for debugging.
1) (phone video in MP4 format) ffmpeg -i VID_20180403_183123.mp4 -vn
-acodec pcm_s16le -ar 44100 -ac 2 output_audio.wav
2) sndfile-info -b output_audio.wav
> > [linux-7cab(iksrazal)]
> > /home/iksrazal> sndfile-info -b F8.wav
> > Version : libsndfile-1.0.25-exp
> > Description : SPEED=29.970D
> > TAKE=004
> > UBITS=00000000
> > SCENE=180403
> > TAPE=180403
> > CIRCLED=FALSE
> > TR1=Tr1
> > TR2=Tr2
> > TR7=Tr7
> > TR8=Tr8
> > NOTE=EVE
> > Originator : ZOOM F8
> > Origination ref :
> > Origination date : 2018-04-03
> > Origination time : 18:31:09
> > Time ref : 0x0bebdd5f1 (66669.002354 seconds)
> > BWF version : 1
> > UMID :
> > Coding history : A=PCM,F=48000,W=24,M=multi,T=F8;VERSION=1.10;1:1 0 0
> > 1 00;2:1 0 0 CNTR 00;3:0 0 0 CNTR 00;4:0 0 0 CNTR 00;5:0 0 0 CNTR
> > 00;6:0 0 0 CNTR 00;7:1 0 0 CNTR 00;8:1 0 0 CNTR 00;L:0 1 0 CNTR
> > 00;R:0 1 0 CNTR 00;
> > [linux-7cab(iksrazal)]
> > /home/iksrazal> ltcdump phone_video_with_ltc.wav | head
> > Note: This is not a mono audio file - using channel 1
> > #User bits Timecode | Pos. (samples)
> > #DISCONTINUITY
> > 00000000 18:31:21.29 | 2639 4239
> > #DISCONTINUITY
> > 00000000 18:31:22.00 | 4240 5841
> > 00000000 18:31:22.01 | 5842 7442
> > 00000000 18:31:22.02 | 7443 9044
> > 00000000 18:31:22.03 | 9045 10646
> > 00000000 18:31:22.04 | 10647 12247
> > 00000000 18:31:22.05 | 12248 13849
> So the audio file begins at 18:31:09.
> The video seems to be missing the beginning, but position 4240 audio
> samples is decoded as 18:31:22.00.
> 4240 samples is 4240/48000 = 88ms.
I am starting the F8 first, then the video ... there should only be a few
second delay but way under the 13 second gap. I am not sure if the video
should start first or the audio recording on the F8 should start first. At
the moment I have settled on the F8 starting first.
> 13 seconds should be 624 000 samples, so it does seem that the video
> starts later than the F8 audio, but the question is whether that part of
> the video still lines up with the appropriate part of the audio recording,
> or if they are offset by that many (or some other non-zero) seconds. That
> is where some common reference point, like wood blocks knocking together,
> would come in useful.
I'm using a big piano octave with an extra arm gesture at the moment, to
start things off. The music seems off by the same amount as the missing
LTC, but I lack exact numbers since I am eye balling the video / audio sync
> What is on the second channel of the phone/camera audio? You could also
> send a copy of the microphone feed with the clap board audio to F8 and
> camera second channel simultaneously, that would give you another
> reference point for comparison.
Yep, my scratch track for testing has a drum machine and a piano going to
ch2 from one of the F8 master outputs.
> Chris Caudle
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