Saving As Oggs Truncates Audio?

I'm having a problem I can't understand. I edit a short audio file (wav), and save it as .ogg. It's fine as long as it is still open in AA. but when I close it and open it again, the end of the file gets cut off (like the last .25 seconds). Does anyone have any idea why this could be happening? It's really frustrating. Thanks.
Ken

Durin,
A bit late getting back to this thread aren't I? Well, I don't remember what I did back then, but this problem has reared its ugly head again.
I tried switching the ogg option to "fixed bitrate (128)," but it still cuts off lid the last several milliseconds of my file. Is there, perhaps, something else I'm doing wrong?
Thanks.

Similar Messages

  • Re-saving QT files defaults audio slider in any browser to lowest setting

    Hey All,
    I create h.264 video AAC audio .mov files in QTPRO 7.2 that play audio normally in IE and Firefox but when I need to re-save a file (to add metadata information) it will not play back with any audio. You have to click on the volume icon in the browser player to boost up the audio. Normally this is already defaulted to the highest setting.
    I have noticed that any file re-saved has the same affect when played back in a browser but I have not seen any answered posts regarding this problem. Does anyone know if this has ever been addressed?

    Hey All,
    Thanks for looking into this. The good news is I have answered my own question. The bad news is the answer is "HAD TO WAIT FOR 7.3!!!!!"
    Yes QuickTime has solved this bug with the newest version. Also if anyone has created QuickTime files in h.264 (using QuickTime 7.2) and needed to re-save for any reason, you can now re-save these files again in 7.3 and the browser player's audio level will default to the highest setting as it should.
    Again thank you to everyone who spent anytime searching an answer to this bug.

  • Crashes when saving after I add audio to slides

    Hi everyone,
    I recently started using Captivate and like a lot of the
    functionality...autorecord, etc. I am having a problem though. I
    have a slide presentation that is about 11 slides long. I used auto
    record to capture the slides and then went in and cleaned up the
    slides. Everything worked great and I was able to save with no
    problems. However, last night I recorded short audio bits for 9 of
    the 11 slides and when I tried to save my computer hung-up and
    required a hard-reboot. I then tried to save after each audio bit I
    recorded, but I didn't get far the app crashed as soon as I tried
    to save after the first audio recording. I am using this locally on
    my PC/Windows XP box. I have a Pentium 4-3.4 GHz processor and 1
    gig of ram. Anyone have any ideas about why this might be happening
    or how to get this working?
    Any feedback would be appreciated.
    -Litha

    quote:
    Originally posted by:
    draggirls
    Also, I was reading somewhere that some sound cards are
    incompatible with Captivate so I need to check on that as well.
    I just thought, another question I should have asked, is,
    have you tried saving different formats MP3/Wav etc to see if that
    changes anything?
    Also, are you recording/saving the audio using Captivate? if
    so, have you tried using your sound software (Soundforge) to record
    and save, then import into Captivate?
    quote:
    Computer comes with job...I let you know if/when I
    leave...lol
    let us know how you get on
    Cheers
    Rossco

  • Saving movie and score- audio problem

    hi
    I am saving movie and score, but after I have finished doing this and then watch back the saved movie in Quicktime, the audio is sounding very muted, the bass it too heavy and the balance of the tracks is not how they play back or sound when i listen to them in Logic.
    Would anyone know why this might be and how to solve it?
    thanks

    Hey Hangtime, give you permission to kick me in the head! I feel stupid now knowing that it is a silly user error. Here I was trying to figure out if its in the 'Details' frame.
    In otherwords, 'Detail's is set to '>' and not pointing downwards.
    Funny thing is, you would think there would be a scroll bar to move up and down to reveal the Movie Preview frame if the 'Details' pane is expanded in the Track Info window.
    In any case... Thanks HT!

  • Logic Pro 9- I saved a song- the audio files did not bounce and are nowhere to be found.  The song is still there, but the audio files are missing.  Any ideas?

    I have searched the computer by song name in the audio files.
    I have opened up media files and looked everywhere.
    I tried remapping where they would be saved and hoping they were updated.
    Any help finding where they are located would be great.  The song still loads when I open it, I just can't find the audio files!

    Have you tried using the built in Browser in LP?
    http://help.apple.com/logicpro/mac/10/#lgcpc788cc57
    Note: It works the same way in LP9 as in LPX

  • Cutting and saving individual video and audio files from a clip

    I have to cut 201 individual video and audio files for separate scenes from an 11-minute animated episode and save them as separate MP4 video and MP3 audio files. What is the most efficient way to do this?

    Without knowing more about your project, it's hard to even guess at the most efficient workflow.
    Like: Do these clips have audio or are you adding it? What format are these clips? What is the intended delivery for the 11 minute movie?
    As a general matter, one creates an event, imports clips to it. Then organizes the clips according to the needs of job, using keyword collections and ratings. Then creates a project and starts a rough cut by marking ranges in the event browser and editing to the time line. Some people start with the music, making that the primary storyline mark where they want the edit to be and cut to that.
    Upon finishing, one can choose one or more of the "destination" presets. It's possible to export Audio only or Video only or both. MP3 is one of the audio choices. Some of the presets are MP4. If you want possibly better quality and more control, export the files as master files and encode in Compressor.
    Good luck.
    Russ

  • Saving a file with audio

    Hi. I'm unable to save a Keynote slideshow that I made with an iTunes song associated with it. I see the "Copy Audio & Movies into Document" in the Save As box, and it's selected but greyed out.
    I need to be able to play this show, with audio, on my notebook.
    Thanks in advance for any help!
    Jim

    purchased songs are only a problem if you give the file to someone else. Your bigger problem is you only get the option to save the audio in the file when you FIRST save it. You could try doing a save as, but if that doesn't work, you might have to start a new file and drag all your slides over to it, and then save it and check the boxes.

  • Export Region As Audio File results in truncated audio file

    When I use the Export -> Region As Audio File option, the resulting file does not sound identical to the region. Specifically, the first 100 milliseconds or so are not in the resulting audio file.
    Does anyone have the same problem? Does anyone have a workaround?
    I'm using this feature to automate the mastering of hundreds of short regions for a CD-ROM title. So while theoretically I could add a little silence to all regions, this is not really practical, unfortunately.
    Powerbook G4 Mac OS X (10.4.5)

    Yes, that was exactly it, the inserted plug-in was to blame. I was using Logic's Adaptive Limiter. Now I'm using the Limiter, and the problem no longer occurs. I'm still somewhat curious if there is a simple workaround (I tried a region delay, but it has no effect on the Export Region feature, apparently), but at least now I know what's going on.
    Thanks for the responses!

  • Saving setting in the Audio MIDI utility

    Wondering if anyone knows how to save or stop the reset of the Audio MIDI utility?
    I have to look after 17 Mac audio studios and it is very anoying to have to reset it every time the mac restarts.
    Please it would seem to be a simple thing but I can nae find a skerrick of info on this.

    It's not impossible that the following script could do what you are asking for.
    *tell application "Audio MIDI Setup" to activate*
    *tell application "System Events" to tell process "Audio MIDI Setup"*
    *   if not (window "Audio Devices" exists) then*
    *      click menu item 1 of menu 1 of menu bar item 5 of menu bar 1* -- Show Audio Window
    *   end if*
    *   click button "Configure Speakers…" of tab group 1 of group 1 of window 1*
    *   tell sheet 1 of window 1*
    *      click (radio button 1 of radio group 1 whose value is 0)*
    *      if title of result is "Multichannel" then*
    *         set sampleRate to "192000.0 Hz"*
    *         set bitDepth to "8ch-24bit"*
          else
    *         set sampleRate to "48000.0 Hz"*
    *         set bitDepth to "2ch-16bit"*
    *      end if*
    *      click button "Done"*
    *   end tell*
    *   tell group 1 of tab group 1 of group 1 of window 1*
    *      click button 1 of combo box 1*
    *      delay 1*
    *      select (text field 1 of list 1 of scroll area 1 of combo box 1 whose value is sampleRate)*
    *      keystroke return*
    *      delay 1*
    *      click pop up button 2*
    *      click menu item bitDepth of menu 1 of pop up button 2*
    *   end tell*
    *   delay 1*
    *   keystroke "w" using command down*
    *end tell*

  • Why do I still get giant gray 'X's instead of an audio player, even with the correct MIME type for my Ogg Vorbis file, whether coded in the audio tag or as a source?

    I uploaded Saturnalia.ogg to my site and used the audio tag to try to display it on my test page. I'd love to ditch the ultra-slow Flash players I currently use, but I don't want to leave most of my site's users wondering what the files sound like, where they are, and why are there giant gray Xs ruining my layout. The audio files work fine in Safari and Opera, and IE uses the correct fallback text. I've tried both coding the file into the audio tag and into source tags. I've set controls="controls" and tried both type="audio/ogg" and type="audio/vorbis". I've tried with and without autoplay.
    According to Mozilla, audio tags are supported, but I'm not seeing any of that support on my site.

    Where are you seeing that?
    Earlier:
    <audio src="dsh.ogg" controls="controls" type="audio/ogg">Your browser does not support the HTML5 audio tag.</audio>
    Now:
    [<object type="application/x-java-applet" width="580" height="15px" controls="controls">
    <param name="archive" value="cortado.jar" />
    <param name="code" value="com.fluendo.player.Cortado.class" />
    <param name="url" value="http://desolosubhumus.webs.com/dsh.ogg" />
    You need to install Java to play this file.
    </object>]<br />
    <audio src="dsh.ogg" type="audio/ogg">
    You need to install Java to play this file.
    </audio>
    *p tags removed for readability
    I know the audio tag should open before the object tag opens and close after the object tag closes, and that I shouldn't need duplicate fallback text, but it wouldn't show up that way. The way I have it set up now finally works on Firefox, IE (plays, not fallback text), Opera, Safari, and Chrome. I'm still trying to tweak the code to make it more standard (audio tags in the proper places), to clean it up so it's not such a huge chunk, and to see if I can get the applet to stay visible (which may not be possible, as the only way I've made it work is as a Cortado VIDEO instead of a purely audio file), and to make it stop autoplaying instead of playing when the user chooses play. Perhaps a Kate stream for track labels and a play button image set directly behind the applet when the play button actually is.

  • Why is audio truncated when "shared"

    I have an hour of keynote presentations that have correctly working audio. (Recorded and dragged and dropped to individual slides.)
    One short presentation (2 min) works perfectly when played in Keynote, but invariably truncates when shared to either iTunes or iWeb. (please see link)
    http://www.brentcbrolin.com/BrentCBrolin/TestPodcast/TestPodcast.html
    This has a video podcast and a Keynote download link. The audio works on the latter; not on the former.
    I have tried various kinds of transition changes, but nothing helps. And, besides, the existing transitions DO WORK IN KEYNOTE playback and when KN file is shared (and downloaded).
    Any suggestions appreciated. Thanks.

    This is awkward, but it did solve the problem.
    Goal: To have the same picture (call it A) continue on screen as the show goes from slide 1 to slide 2. That is, to not have A “blink” when the slides change between 1 & 2.
    (N.B. All slides have MP3 narrations; these were recorded in audio program and dropped onto each slide.
    Problem: As long as I did not give picture A an EXIT transition, the audio on slide 1 was truncated when I “shared” the slide show to iTunes or iWeb.
    I had thought I could just have NO EXIT transition for A on slide 1, and have no entry transition for A on slide 2 (which would avoid the “blink” between 1 & 2.
    And when I previewed it in Keynote, THE AUDIO WORKED PERFECTLY, even when picture A on slide 1 had NO EXIT transition.
    However, the truncation of audio on slide 1 did occur whenever I shared the same show to iTunes or iWeb (.mov).
    I tried many variations including:
    a “stop audio” transition with a lot of extra time at end of slide 1;
    delaying the “start transition” in the “slide” window of inspector of slide 1;
    ...but all resulted in truncated audio on slide 1, as long as picture A DID NOT HAVE AN EXIT TRANSITION.
    However, if I did give picture A an EXIT transition on slide 1, the audio was not truncated (but, of course, I had a “blink” between slide 1 and 2).
    Solution:
    Hide a rectangle (or any graphic) behind picture A on slide 1.
    Give that blind rectangle an EXIT transition…
    …but don’t give picture A an exit transition.
    This fools Keynote: it segues from slide 1 to 2 without a “blink,” and the audio narration is not truncated.
    But it took a day to figure it out. (I removed the link to the examples mentioned in first post.)

  • Soundtrack Pro losing audio!

    Has anyone encountered (and please god, fixed) a problem involving Soundtrack Pro 2 actually truncating audio files (.aiff) so that data after a certain point in the track up and vanishes? For example, say a 10:00 audio file get some edits, is saved, then reopened, and voila—the 10:00 is now 05:00, with everything that existed after that point in the track simply gone. I've experienced this several times over the past few days, twice when saving, then reopening the file, and another time after flattening, which was really weird, because the newly flattened file suddenly consisted of only the first half of the file I had just been working on in the editing window. Great feature: Flatten the file and lose half your track. Anyone have anything on this? Thanks in advance.

    You'll probably get a better answer if you post this in the Soundtrack Pro forum instead of the Mail and Address Book one
    You can find it here:
    http://discussions.apple.com/forum.jspa?forumID=743

  • Audio Cuts Out in my clips

    I imported video clips into a project, and started putting my video together in the timeline.
    I noticed that the audio just cuts out on some of the clips...both in the timeline, and in the clip preview.
    The files are AVCHD video. I chose a project setting under the AVCHD folder that seemed close to what my clips are (as far as frame rate & frame size).
    I just updated my QuickTime player, but that did not help.
    I am using Premiere Elements 11, on a Windows 7 PC. The project and all the clips are saved in a folder on my desktop that I am working off of. I played the videos from the folder in Windows Media Player, and the audio is there.
    Please give me suggestions! Thanks!

    One of the main causes for missing, or truncated Audio is a failure to allow Conforming of the Audio to complete. This Tips & Tricks article goes into much more detail: http://forums.adobe.com/thread/726693?tstart=60
    I would close PrE, then navigate to the Media Cache (the location can be set in Edit>Preferences>Scratch Disks, but is within the Project's folder hierarchy by default), and Delete all the CFA and PEK files for that Project. Then, launch PrE, and Open that Project. The Conforming will be done again. Be very patient, and do not touch PrE, until Conforming has completed 100%.
    Does the Audio now appear?
    Good luck,
    Hunt

  • [SOLVED] How to stream audio output over wifi?

    Hello,
    I have following problem. I would like to listen to the music from my laptop on my home Onkyo AV receiver (amplifier). Both laptop and receiver are connected on the same network, laptop via wifi, receiver using LAN. Receiver is capable to play internet radio streams in various format - e.g. ogg or mp3.
    My idea is to produce ogg stream on laptop and listen to this stream through amplifier. I was searching for this solution a found something that has always some disadvantage, like MPD + MPC (can play only from database of local files, not from jamendo or other network services), DLNA can play only local files, Icecast + Ices can play from predefined playlist.
    My idea is to have something independent - something what takes audio data normally played on soundcard and feeds it to network stream, something what can work with any player - audio or video, or even with desktop notifications.
    Does anybody here know how to do it?
    Thanks!
    Last edited by KejPi (2012-03-14 20:18:28)

    Thank all of you for your hints. Finally I have decided for ALSA based solution that simply works for all applications smoothly in background and it is here when I need it - in other words, it is something that doesn't borther me when I do not need it. I use ALSA loopback with ffmpeg and ffserver for streaming in mp3 format.This is my final solution:
    You need following:
    alsa
    ffmpeg
    if you use KDE4, then you need phonon-gstreamer backend, as in my case phonon-vlc didn't work (it was heavily distorted)
    Procedure:
    Modprobe snd-aloop (2 loopback channels are enough)
    modprobe snd-aloop pcm_substreams=2
    For permanent solution follow arch wiki https://wiki.archlinux.org/index.php/Modprobe
    Create ALSA configuration file and store it either to ~/.asoundrc (just for user) or to /etc/asound.conf as system wide:
    pcm.!default {
    type asym
    playback.pcm "LoopAndReal"
    capture.pcm "hw:0,0"
    hint {
    show on
    description "Default with loopback"
    #"type plug" is mandatory to convert sample type
    pcm.LoopAndReal {
    type plug
    slave.pcm mdev
    route_policy "duplicate"
    hint {
    show on
    description "LoopAndReal"
    pcm.mdev {
    type multi
    slaves.a.pcm pcm.MixReal
    slaves.a.channels 2
    slaves.b.pcm pcm.MixLoopback
    slaves.b.channels 2
    bindings.0.slave a
    bindings.0.channel 0
    bindings.1.slave a
    bindings.1.channel 1
    bindings.2.slave b
    bindings.2.channel 0
    bindings.3.slave b
    bindings.3.channel 1
    pcm.MixReal {
    type dmix
    ipc_key 1024
    slave {
    pcm "hw:0,0"
    #rate 48000
    #rate 44100
    #periods 128
    #period_time 0
    #period_size 1024 # must be power of 2
    #buffer_size 8192
    pcm.MixLoopback {
    type dmix
    ipc_key 1025
    slave {
    pcm "hw:Loopback,0,0"
    #rate 48000
    #rate 44100
    #periods 128
    #period_time 0
    #period_size 1024 # must be power of 2
    #buffer_size 8192
    You can play with sample rates and buffer sizes you you have any problem. This configuration works on my system.
    Prepare ffserver configration and store either to default location /etc/ffserver.conf as system wide setup or anywhere to your home:
    # Port on which the server is listening. You must select a different
    # port from your standard HTTP web server if it is running on the same
    # computer.
    Port 8090
    # Address on which the server is bound. Only useful if you have
    # several network interfaces.
    BindAddress 0.0.0.0
    # Number of simultaneous HTTP connections that can be handled. It has
    # to be defined *before* the MaxClients parameter, since it defines the
    # MaxClients maximum limit.
    MaxHTTPConnections 2000
    # Number of simultaneous requests that can be handled. Since FFServer
    # is very fast, it is more likely that you will want to leave this high
    # and use MaxBandwidth, below.
    MaxClients 1000
    # This the maximum amount of kbit/sec that you are prepared to
    # consume when streaming to clients.
    MaxBandwidth 1000
    # Access log file (uses standard Apache log file format)
    # '-' is the standard output.
    CustomLog -
    # Suppress that if you want to launch ffserver as a daemon.
    NoDaemon
    # Definition of the live feeds. Each live feed contains one video
    # and/or audio sequence coming from an ffmpeg encoder or another
    # ffserver. This sequence may be encoded simultaneously with several
    # codecs at several resolutions.
    <Feed feed1.ffm>
    # You must use 'ffmpeg' to send a live feed to ffserver. In this
    # example, you can type:
    # ffmpeg http://localhost:8090/feed1.ffm
    # ffserver can also do time shifting. It means that it can stream any
    # previously recorded live stream. The request should contain:
    # "http://xxxx?date=[YYYY-MM-DDT][[HH:]MM:]SS[.m...]".You must specify
    # a path where the feed is stored on disk. You also specify the
    # maximum size of the feed, where zero means unlimited. Default:
    # File=/tmp/feed_name.ffm FileMaxSize=5M
    File /tmp/feed1.ffm
    FileMaxSize 200K
    # You could specify
    # ReadOnlyFile /saved/specialvideo.ffm
    # This marks the file as readonly and it will not be deleted or updated.
    # Specify launch in order to start ffmpeg automatically.
    # First ffmpeg must be defined with an appropriate path if needed,
    # after that options can follow, but avoid adding the http:// field
    #Launch ffmpeg
    # Only allow connections from localhost to the feed.
    #ACL allow 127.0.0.1
    </Feed>
    # Now you can define each stream which will be generated from the
    # original audio and video stream. Each format has a filename (here
    # 'test1.mpg'). FFServer will send this stream when answering a
    # request containing this filename.
    # MP3 audio
    <Stream stream.mp3>
    Feed feed1.ffm
    Format mp2
    AudioCodec libmp3lame
    AudioBitRate 320
    AudioChannels 2
    AudioSampleRate 44100
    NoVideo
    </Stream>
    # Ogg Vorbis audio
    #<Stream test.ogg>
    #Feed feed1.ffm
    #Format ogg
    #AudioCodec libvorbis
    #Title "Stream title"
    #AudioBitRate 64
    #AudioChannels 2
    #AudioSampleRate 44100
    #NoVideo
    #</Stream>
    # Special streams
    # Server status
    <Stream stat.html>
    Format status
    # Only allow local people to get the status
    ACL allow localhost
    ACL allow 192.168.1.0 192.168.1.255
    #FaviconURL http://pond1.gladstonefamily.net:8080/favicon.ico
    </Stream>
    # Redirect index.html to the appropriate site
    <Redirect index.html>
    URL http://www.ffmpeg.org/
    </Redirect>
    This sets ffserver for streaming in MP3 format, stereo, 320kbps. Unfortunately I haven't succeeded with OGG Vorbis streaming.
    Now you have all configuration you need and if you want to stream following two commands do that:
    ffserver -f ffserver.conf
    ffmpeg -f alsa -ac 2 -i hw:Loopback,1,0 http://localhost:8090/feed1.ffm
    You can test it for example by mplayer:
    mplayer http://YourLinuxBox:8090/stream.mp3
    And that's it. Sound is played by normal sound card and sent to stream simultaneously. If you do not want to listen sound from computer you can mute your soundcard. It has an advantage that one can normally listen to music on the computer with or without streaming and in both cases without any reconfiguration. To start streaming just call ffserver and ffmpeg.
    Advantages:
    + very simple solution without any special sound server
    + no special SW required (in my case I had already instaled all I need for that)
    + streaming on request by two simple commands
    + normal soundcard function
    + streaming in MP3 format that is supported by many home AV receivers
    Disadvantages
    - phonon-vlc backend not compatible (also VLC does not work)
    - OGG streaming does not work
    - some latency (~ 5 sec)
    - all sounds are sent to stream, including various desktop notifications (in KDE could be managed by phonon)

  • How can I match sound quality between video clips and audio clips?

    Hi guys,
    First of all let me say that I enjoy using the latest Media Encoder. However, as a novice, I have two related problems.
    1.I am recording video to use with PowerPoint slides in Adobe Captivate. I am encoding the video with best quality FLV. This produces very good video,
          but with a degraded audio track, even though I had filtered the audio in soundbooth. The post encoding sounds like it did before Soundbooth corrections.
    2. Some of the video recording has been set cut into audio clips to be used as background narration behind PowerPoint. The logic behind this is: that if both are
          recorded at the same time, the voice quality should match.
    This is far from the outcome. I have tried MP3 for the audio which of course produced the best sound- but is entirely unlike the rest of the video. I have also encoded the audio with Windows Waveform preset figuring that since the soundbooth edit was a .wav file, that there should be a reasonable match. Wrong again!
    So, the audio quality of all clips is poor, and the difference between the encoded video clips, and the audio encoded clips spoils the otherwise professional result in Captivate.
    Please help, all of you experts.

    Thank you.
    OS Windows 7 x 64; 3.2 MHz; 2 TB; 12 GB Ram-9GB ram allocated to Adobe.
    My process is this:
    1. The whole video and audio narration material is shot in one continuous video sesseion.
    2. The microphone is only mono, but the camera is set to stereo. Noone can work out how to change the camera to mono.
    3. In Premiere Pro 5, I select render and replace in SoundBooth.
    4. In Soundbooth, I choose Add Multichannels.
    5. In Multichannels, I delete the unwanted noisy channel, and any other unused channels.
    6. With the remaining mono channel, I then select Export and Save As option, then Stereo.
    7. Next I edit the stereo channels as needed.
    8. The imported stereo won't automatically update in Premiere Pro, so in PPR I import the new edited stereo track.
    9. I test this track in PPR, to check that it is playing R & L channels. It is. OK.
    10 On the timeline, I proceed to cut apart this single stereo wavelength .
    11.These clips are saved as Video, or Audio subclips.
    12 From the video subclips, I export to Media Encoder, and choose either FLV4 match attributes (High Q). This I modify to custom 2 pass (Custom save).
         The summary says that this is saved to MP3 Stereo.
    13. I import this into Captivate 5, but the Video outcome is Left channel only. I have tested this on Media Player/Real Player where it is also L channel only.
    14. I export my Audio subclips to Media player, and select MP3 option and this summary also reports stereo. The Audio outcome is stereo, when played in any player and in Captivate 5.
    I cannot understand how a single, continuous track edited uniformly in Soundbooth to stereo, can produce different outcomes in Media player.
    I would so appreciate a solution to this problem, because, in the Captivate 5 e-learning production, the switch between stereo audio (behind PowerPoints), and l. channel video is distracting. Naturally, the l. channel only is a thin sound. It is as if it had never been edited in SoundBooth at all.

Maybe you are looking for