Audio Synchronizing

I've converted quite a few VHS tapes I own to DVDs without problems, but now when I send the VHS from my VCR through my analog/digital converter to iMovie they play well. Then try to make a DVD (through IMovie) using iDVD it seems to work fine. It even plays well in the iDVD screen. But when I burn the disc and try to play it, the audio drifts and the speaker sometimes is behind the words they are saying and sometimes they are ahead of what they are saying. This, of course is very distracting. Never had this before, but on my latest project I just can't seem to get the video and audio to synchronize on the burned disc. It is synchronized on the iMovie version and the iDVD version.
Any ideas??
iMac G5 flat panel Mac OS X (10.4.8)

Hi Ron:
You may need to extract all of the audio in iMovie prior to exporting to iDVD.
Look here:
iMovie: Improving audio and video synchronization
http://docs.info.apple.com/article.html?artnum=42974
(Talks about iMovie 3, but holds true for imovie 5 as well)
:)Sue

Similar Messages

2 Cameras 1 Audio Synchronized Clips lost after timeline review

Hello. I have embarked on a film project and I am trying to sync and log the footage effectively so that a remote editor can begin work and we can utilize the same titled footage. So that he and I can work effectively from the same material, I am syncing all footage and titling it by scene and take. So far I have synced two complete scenes. Occassionally I would open a synced file in the timeline just to review and make sure everything was in order. Then I would highlight the material in the timeline, delete, it, and move on to syncing the next 3 clips. This is why I did not notice that although the title of the synced clip remained in my list of " - Synchronized Clip(s)" the clip that I had opened and then cleared from the timeline was actually blacked out completely. It was unplayable. It was as though once you added a synced clip to the timeline, you had to use it or it was gone. But as I said, the title of the clip remained so I didn't notice that some were gone.
This has proven very frustrating to me. I have tried to synchronize clips and then turn them into compound clips, but that does not seem to allow me to "break apart" the synced clips for future cutting.
My goal is to completely "sync" the two cams and 1 audio track for editorial review, and then export the entire Library to a hard drive for a two editor work flow.
What the heck am I doing wrong? Or, am I doing anything right? Thanks so much for your help.

You mean once you've deleted it? If this was the synced clip, you just have to resync it.
Once you've opened it in the timeline you don't have to do anything. Just open sonething else into the timeline or open a project.
Sync clips aren't in the Finder. They're simply two separate clips combined in the sync clip container.

Getting the audio synchronized

Hi,
If i make a recording of a flash game (works fine!) then i
encounter a problem when playing back in a browser: the audio is
totally not synchronized with the video. How come? Can i do
something in the flash video settings to get this better?
I believe i have similair problems when importing a flv file,
that the sound is not lipsynch.
Thanks for your help,
Wouter
My elearning company

Yikes, that sounds like it could be a lot of things?
You can do a "Search" on this site to see if the topic has
related posts.
Can you please try the following:
On the slide settings, turn off the fade-in and fade-out
settings so that they read as NO FADE.
If the sound is "background sound," you might want to add
this as the sound of a partiicular slide so that you can control
the timing of it.
If the sound is embedded into the slide, then you should be
able to move the sound start and stop in the timeline section at
the top of the slide editing window. In general, ledt click and
drag the entire thing or left click and drag the left end. You can
also double click on the sound object and edit from there.
Not sure this will help you, but I have seen lots of problems
with slide fade-in's and some fade-out's. It appears to me that
sound and fade-in are not always compatible.
Best of luck to you on your project. Please let this board
know if any of the above solves your problems so that others can
benefit.
Joe C.

How to keep audio synchronized with video when exporting?

I have a few .mp4 source files that I'd like to combine into a single file. I've done the editing and exported. When played, the audio is misaligned to the video. As a test, I've tried exporting a single clip rather than joining a few and the same thing happens. I end up with audio that is misaligned to the video after export even though they are perfectly synched in the source video. As another test I've tried exporting a single clip to .mpeg rather than .mp4 and the audio stays perfectly synched with the video. It seems the misalignment occurs whenever I export to .mp4. I'm obviously missing a fundemental concept here. Can anyone shed some light on it for me? Thanks in advance.
Brandon

Hi Rafeek,
One solution is to set the column width after manually or programmatically refresh the PivotTable, for example:
Private Sub Worksheet_PivotTableUpdate(ByVal Target As PivotTable)
Dim ws As Worksheet
Set ws = Application.ActiveWorkbook.ActiveSheet
ws.Columns("A").ColumnWidth = 10
End Sub
Another option is to set the column width, then protect the columns from been updated by the user, before refreshing the PivotTable, unprotect the worksheet. For example:
Public Sub LockColumnA()
Dim ws As Worksheet
Set ws = Application.ActiveWorkbook.ActiveSheet
ws.Columns("A").ColumnWidth = 10
ws.Columns("A").Locked = True
ws.Protect "123"
End Sub
Public Sub UnprotectWorksheet()
Dim ws As Worksheet
Set ws = Application.ActiveWorkbook.ActiveSheet
ws.Unprotect ("123")
End Sub
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A/V stream out of sync with when playing audio on SourceDataLine

Hi,
I'm receiving a video and an audio stream via RTP and try to play them synchronized (audio synchron with lip movement).
I'm using a DataSourceReader in order to read from a datasource and play on a JavaSound SourceDataLine. This works pretty good but unfortunately the audio is played with a delay of approx 1 second.
else if (evt instanceof NewReceiveStreamEvent) {
     try {
          stream = ((NewReceiveStreamEvent)evt).getReceiveStream();
          DataSource ds = stream.getDataSource();
          Format format = null;
          // Find out the formats.
          RTPControl ctl = (RTPControl)ds.getControl("javax.media.rtp.RTPControl");
          if (ctl != null) {
               format = ctl.getFormat();
               log.info("Received new RTP stream: " + format);
          else
               log.info("Received new RTP stream");
          if (format != null && format instanceof VideoFormat) {
               Player video = javax.media.Manager.createRealizedPlayer(ds);
               video.start();
               this.video_player = video;
          else if (format != null && format instanceof AudioFormat) {
               JavaSoundDataSourceReader dsr = new JavaSoundDataSourceReader();
               Player audio = dsr.open(ds);
               audio.start();
               this.audio_player = audio;
          }          I already tried to synchronize them with
videoPlayer.addController(audio_player);
audio_player.start();but this is throwing an Exception:
javax.media.IncompatibleTimeBaseException
     at com.sun.media.multiplexer.RawBufferMux.setTimeBase(RawBufferMux.java:377)
     at com.sun.media.BasicSinkModule.setTimeBase(BasicSinkModule.java:52)
     at com.sun.media.PlaybackEngine.setTimeBase(PlaybackEngine.java:1672)
     at com.sun.media.BasicPlayer.setTimeBase(BasicPlayer.java:272)
     at com.sun.media.BasicPlayer.addController(BasicPlayer.java:1043)
     at VideoPlayback.update(VideoPlayback.java:281)Line 281 of VidePlayback.java is the "videoPlayer.addController(audio_player);" line.
My guess is that it should rather use RawSyncBufferMux instead of RawBufferMux but I don't know how to change this. Maybe directly as a processor option?
Please help as I've no clue how to improve this.

chummer wrote:
it should also be synchronized with the video :-)The problem with getting it synchronized is that Javasound & your sound card have internal buffering, so there is some delay between writing to them and hearing the sound come out...
Normally, JMF handles this internally somehow by aligning things to a timebase, which can be syncronized, and allowing Processors to fill up their internal buffers ahead of time during the prefetching phase so that everything that needs to play together can have an instant start time. As it goes along, presumably adjustments are made as things get out of sync.
My best advise to you would be to implement your Javasound stuff inside of a Renderer class, instead of a DataSink class. DataSinks are designed to take in data and write it out as fast as they can... whereas Renderers are given data at "real-time" intervals and are designed to be played.
When implementing your Renderer class, JMF should handle the syncronization for you. IE, it'll give you data as it should be played by calling the Process function, so you'll just need to write the data to your TDL as it comes in. I believe that "Process" may be called before "Start" to allow you to prebuffer some information, but you'll need to test that as I've never actually tried it.

Experiencing *two* separate audio issues

The first problem deals with the audio amplifier powering off, and then powering back on with a beep, every time my Mac makes a sound. Since I use sticky keys, each time I hit a modifier key it beeps and then makes the normal modifier key alert sound. I suppose I could solve this problem by turning off the modifier key alert, but I find that feature helpful. Not to mention the beep still sounds before any IM event sound or mail alert chime.
Apparently I'm not the only one with this issue, judging from numerous threads like this:
http://discussions.apple.com/thread.jspa?threadID=1009334&tstart=0
The second problem involves constant crackling in my left external speaker whenever I turn on Airport. I didn't notice this at first because I usually keep Airport off, but after reading some more forum discussions, I tested it. I'm glad I do not need to rely on Airport to get on the internet because this audio noise is even more annoying!
I only noticed these problems after the latest system update, so I'm guessing they are somehow related to it.
Mac Mini Mac OS X (10.4.10) 1.83 GHz Intel Core Duo; 2 GB 667 MHz DDR2 SDRAM

Hi Zach,
Soundbooth was not really designed as a full-featured multitrack recorder, and the recording model does not currently support playback of other audio synchronized with new recordings. You would probably be interested in a more suitable application for this workflow. If you are a Windows user, Adobe Audition is a full multitrack recording and mastering studio. You may also want to try Audacity (http://audacity.sourceforge.net/) which is a free, open-source recording tool which works on Windows and Mac.
Good luck!
Durin

Recording two separate audio question

I want to be able to record a sound, and then record another sound while being able to hear the first sound to match it up but I have no idea where or how to do that. Any help would be appreciative.

Hi Zach,
Soundbooth was not really designed as a full-featured multitrack recorder, and the recording model does not currently support playback of other audio synchronized with new recordings. You would probably be interested in a more suitable application for this workflow. If you are a Windows user, Adobe Audition is a full multitrack recording and mastering studio. You may also want to try Audacity (http://audacity.sourceforge.net/) which is a free, open-source recording tool which works on Windows and Mac.
Good luck!
Durin

Audio Late Binding Sync

Hi,
I want to use audio late binding for a live video stream along with different live audio streams.
I succeeded to set up the manifest as described here: http://www.adobe.com/devnet/adobe-media-server/articles/late-binding-audio-ams.html
Everything works fine, but one thing I don't understand is: Upon which information is the audio synchronized with the video at the receiver player?
Are there some timestamps included in metadata to do this or how is this done?
I ask, because my task is to manually adjust the delay of different live incoming audio streams to match the video.
Thanks,
ueler13

Hi Scott,
in simple terms, switching rules are using the f4m declared bitrate to compare to the actual bandwidth. Declaring the same bitrate for all MBR assets will prevent them from working correctly - correct me if I misunderstood what you wrote.
Switching or changing the alternate audio track will be affected by the buffer size(s). You could expect a maximum lag equivalent, in the worst case, with the sum of all buffers.
Regarding the seek and change audio bug, I think it was fixed. http://bugs.adobe.com/jira/browse/FM-1326 is a similar case. You might want to resync with the trunk to get the latest bugfixes, since the sprint 5 was relatively in the middle of the Alternate audio feature development.
S.

Synchronize on video stream with mplayer

Hello,
I recently bought an EasyCap DC60+ USB adapter. It is an RCA -> USB adapter which is recognized as an USB webcam by the kernel.
I installed the easycapd60 module and it works fine for video with no sound, but I can't get audio working correctly. It looks like my audio stream is 1 second late on the video stream (I get the sound one second after the image).
The problem is that mplayer is synchronizing the playback on the audio stream (I think). That means that my video stream is delayed in order for it to be in sync with the audio stream. As I want to use this DC60+ adapter to play video games on my computer screen, I can't tolerate a 1 second delay.
A solution to this problem would be to ask mplayer to synchronize on the video stream instead of the audio stream, then play with the A/V sync parameters to get the audio synchronized. But I can't find the parameter to do this in the mplayer manpage.
So my question is the following : does this parameter exist ? And is there another solution I didn't think of which could solve my problem ?
Last edited by delroth (2010-06-06 00:26:14)

Did you have to do anything special to get the DC60+ to work? I've got one installed using the em28xx driver and it eventually makes my USB mouse stop responding, and sometimes the whole system crashes after that.

How should i develop my p2p streaming system

i have tried some way as following to develop in half year. But it still not works as i wish.
h1. 1.
h3. main idea:
Use jmf RTP api to link each peers.
h3. implement:
RTPManager( transmitter) ---> RTPManager (receiver )
receiver create a merged CloneableDataSource (2 tracks) to RTPManager for tramsmit on receiver.
And create SendStream to next peer on each RTPManager for tramsmit .
h3. result:
The first receiver have good quality. But the next peer's result is bad and lag. The third peer is worse that second peer.
There is video/audio synchronous problem sometimes. Maybe fail recovery can't be perfect done due to new stream realize time.
h1. 2.
h3. main idea:
Use custom DataSource to buffer a few seconds.
h3. implement:
RTPManager/custom rtp sender( transmitter) ---> DataCatcher ---(sychronized queue)--> PullBufferDataSource ----> Player (create by Manager)
h3. result:
Data reached PullBufferDataSource and buffer for 5 seconds(timestamp subtraction). if player starts, all buffer in PullBufferDataSource will be eated by the player. Screen displayed by the player is quickly go through. Does player do not support auto time synchronization?
h1. 3.
h3. main idea:
Direct transmit data in disk to peer. Use SeekableStream to play.
h3. implement:
Custom data chunk sender( transmitter) ---> catcher(write to data) /////--> ByteDataSource(seekable read from not complete data)
h3. result:
it's good for playing from start to end. But if data chunk sender doesn't sends from first position, player remains sound and video stoped. if sended data play in KMplayer, it works.
Sorry for poor English, and any suggestion?
Thank you in anvanced..

h1. 1.
h3. main idea:
Use jmf RTP api to link each peers.
h3. implement:
RTPManager( transmitter) ---> RTPManager (receiver )
receiver create a merged CloneableDataSource (2 tracks) to RTPManager for tramsmit on receiver.
And create SendStream to next peer on each RTPManager for tramsmit .
h3. result:
The first receiver have good quality. But the next peer's result is bad and lag. The third peer is worse that second peer.
There is video/audio synchronous problem sometimes. Maybe fail recovery can't be perfect done due to new stream realize time.There is a lot of processing involved in doing it this way, most of it unnecessary. That'll add delay, which will result in dropped packets.
A better design would be to implement the RTP forwarding at the network level.
Suggestion: [http://java.sun.com/javase/technologies/desktop/media/jmf/2.1.1/solutions/RTPConnector.html]
Write a custom RTP connector (sender and receiver) such that the receiver will not only receive the RTP packets, but also forward them to whoever it forwards them to. This will minimum the processing on each peer, which will reduce the delay and thus minimize packet loss.
h1. 2.
h3. main idea:
Use custom DataSource to buffer a few seconds.
h3. implement:
RTPManager/custom rtp sender( transmitter) ---> DataCatcher ---(sychronized queue)--> PullBufferDataSource ----> Player (create by Manager)
h3. result:
Data reached PullBufferDataSource and buffer for 5 seconds(timestamp subtraction). if player starts, all buffer in PullBufferDataSource will be eated by the player. Screen displayed by the player is quickly go through. Does player do not support auto time synchronization?RTP systems have a built-in mechanism to prevent the player from going too fast, which is simply, the packets come in real-time, so the player's job is to play them as fast as it can. If you're dropping 30% of your packets, and buffering the packets, you'd expect the player to play the data 30% faster than it should...because those missing packets aren't replaced with blank packets or duplicate packets or anything, they're just simply not in the stream...
Suggestion: Use a PullBufferDataSource that will manage how fast data is given to the player based on the timestamps of the available frames.
For instance, if your player is playing packets at 30 FPS, but you're only getting every other frame (50% frame loss), then you'd want to deliver data one frame at a time with a delay equal to one frame... so read, sleep, read, sleep...etc...
With a PushBufferDataSource, the player will not read data until told to do so with a call to transferData...
h1. 3.
h3. main idea:
Direct transmit data in disk to peer. Use SeekableStream to play.
h3. implement:
Custom data chunk sender( transmitter) ---> catcher(write to data) /////--> ByteDataSource(seekable read from not complete data)
h3. result:
it's good for playing from start to end. But if data chunk sender doesn't sends from first position, player remains sound and video stoped. if sended data play in KMplayer, it works.If you're doing it directly from disk, then you're not including the file header that JMF needs to figure out how to play the file.
Suggestion: If you want to send from some point other than the middle, just grab the file header and append it to the beginning of the stream...

CC and FLV files

Hi -
Help please - I have tried a bunch pf workarounds and nothing
seems to work.....I am doing a soft skill course in captiavte. I am
importing flv files that contain avatars, video and audio. Well,
since the audio is embedded in the flv file - I am obviously having
a problem with adding cc. Any workarounds?

quote:
I am obviously having a problem with adding cc. Any
workarounds?
Nope, not really. All you can do is approximate it,
and it may be difficult to get timing right (or not).
The first issue you will face is that Captivate "grays out"
the CC button if there is no audio on the slide. The workaround for
that is to create a tiny audio file that contains an inaudible
waveform. You won't be able to hear it, but it will fool Captivate
into accepting your closed captioning.
As to the timing issues with having your closed captioning
and the FLV audio synchronized, that is going to have to be a trial
and error process. Best of luck to you!! Larry

REMOVE_FROM_STAGE issue

I have a movie with a four frame timeline. Each frame contains the same thing: a movie clip of 10 frames length that slides an animation clip in, then when the slide is complete, tells it to play. So, in short, each clip contains another clip, which it animates before playing: finally, this clip that plays after sliding in has embedded timeline audio synchronized to an animation.
My problems began when I applied the navigation to the main clip, in order to move between the four frames. The audio would repeat: when I moved from say frame 1 to frame 2 -- EVEN THOUGH the frame 1 clip was not longer on stage -- it's timeline audio continued to play out, and at some point would repeat.There was no means of controlling it. Simply calling a command to stop the clip and stop all audio playing upon navigating would only stop the previous CLIP timeline: while it seems as though the embedded timeline SOUND itself has a separate existence/stream.
Which brings me to the REMOVE_FROM_STAGE issue. At first, placing a listener for this seemed to be the answer to my problem. Here's the code I placed on the clip that contained the timeline audio I wanted to stop:
this.addEventListener(Event.REMOVE_FROM_STAGE, stopIt);
function stopIt(e:Event) {
e.target.gotoAndStop(1);
This code worked, as long as I navigated FORWARD. As soon as I navigate BACKWARDS, the Flash Player crashes, every time. I've even done a very simple clip to test this: I place a clip with the above code in it on a timeline, then a button to navigate to an empty second frame in the main timeline. On navigating to the second frame, the sound stops, and on navigating back it plays again without problem. I then created another clip, this time with an empty FIRST frame on the timeline, and the clip with the REMOVE_FROM_STAGE listener on the second frame. I play this movie, click to go to the second frame, and the clip plays normally. I then click to navigate BACK, and the crash occurs.

Found solution here:
http://board.flashkit.com/board/showthread.php?t=671514
It's a bug with customizeable seekbar component.

Doku nachvertonen

Hallo,
Habe mit PremierePro CS5 eine Art Doku (Lehrvideo) erstellt.
Jetzt möchte ich noch teilweise etwas Senf dazugeben (wie man das so aus den Dokus der dritten Programme kennt).
Direkte Aufzeichnung in die PremierePro Audiospur geht ja wohl nicht (hab zumindest keine Möglichkeit gefunden).
Könnte jetzt mit Soundbooth mehrere einzelne "Senfdateien" aufnehmen und die dann in PP einfügen.
Macht man das wirklich so? Geht das auch irgendwie einfacher?
Vielen Dank für euere Unterstützung

Könnte jetzt mit Soundbooth mehrere einzelne "Senfdateien" aufnehmen und die dann in PP einfügen.
Yupp, so isses auch gedacht. Premiere Projekt as DynamiLink öffnen für Vorschau und dann Audio synchron dazu in SB aufzeichnen, dann Datei in Premiere importieren.
Mylenium

"pop-up" smart object?

Hello,
Can anyone tell me how I can create a pop-up object (Smart Object or otherwise).
My goal is to do the following:
1. The learner clicks a button
2. The resulting action is a pop up message which is displayed for 3 seconds and then
3. Automatically executes a line of javascript
Is this even possible?
Currently, I am doing something similar, only I am using a 'phantom' slide where the message resides. Unfortunately, this method messes up my lesson page count display by: "Page $$cpInfoCurrentSlide$$ of $$cpInfoSlideCount$$"
I'd rather do this with a pop-up, if possible?
Thank you,
Doug

Would be possible with micronavigation: put the pop up message (in a shape if you want) at the end of the slide for the wanted duration and hide it. When clicking the button, show the message, navigate to the start frame of the message and execute the JS. If the button is not clicked, the message will remain invisible. Of course I do not know what has to happen more on that slide, or if there is some audio synchronized.
Lilybiri

Let fragment's music play at start of fade

Hi guys
I'm working on a project in iMovie for which I would like to have this "script":
1. Text with name of the artist and the song. Black background, white text.
2. Fade from black background immediately to the fragment.
3. Fragment
4...
I would like the music of the fragment to start playing at the very beginning of the movie, while the images of the singer that sings the song live on stage, appear after the text and the fade.
So the question: how can I put the text and the fade ON (not before) the fragment so that the music can start at 00:00:00 and the appears after 4 or 5 seconds with the music/audio synchrone with the fragment?
Is my question clear enough?
Thanks!
Bob

iPod issues? Try these five steps first: Reset, Retry, Restart, Reinstall, Restore.
From the Troubleshooting Section Here
http://www.apple.com/support/ipodclassic/

Audio Synchronizing

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