Mixed audio formats

Does anyone know how much more difficult it is for FCP to have different audio formats in the timeline?
I'm guessing it takes more processing to have compressed formats like .mp3 in there, than to mix, say, .aif and .wav.

I'm still a bit new to the video world. I forget that FCP just renders all that stuff.
In my more familiar audio environments, I'm used to avoiding conversion when possible.

Similar Messages

  • Stereo Mix Audio format not supported?

    Hi
    I'm pretty new at streaming, and i need help with my audio setup, I'm selecting Stereo Mix from the dropdown menu in the FMLE 3.2 but when i do it says.
    "Audio format not supported."
    "Audio format provided by audio capture device is not supported by this application."
    I don't really understand why, since my mic device is supported. Why does this occur?
    And how can I fix it?
    I have a motherboard soundcard "VIA HD".
    EDIT: I figured out why it doesn't work.
    I went to the Recording devices menu to check the options but I noticed that Stereo Mix was "Currently unavailable" I figured maybe a program was using it but I don't know how to check what programs use the device.

    I have noticed something else which is weird : the recording device is ALWAYS called "Line 1".
    I tried renaming both the VAC playback and record devices Line 1 to other names, FMLE still displays "Line 1". This makes no sense (and Audacity sure doesn't behave that way, I have checked).
    Should I need to make a complete uninstallation to wipe old FMLE settings ?

  • Premier Elements 7 - confusion over mixing video formats

    Hi
    I have been using a DV camcorder (SD quality) for some time, originally recording in 4:3 format, and more recently in 16:9 format (as we bought a wide screen telly).
    I have just treated myself to a new AVCHD Panasonic video recorder recording High Definition footage in 1080 16:9 PAL format onto an SD memory card.
    I wanted to take all this footage and put it on my computer to edit it together and start making some films to keep for the future - on DVD for now, and at some point on Blue Ray. To this end I have just purchased Adobe Premier Element 7.0, with Photoshop Elements 7.0 bundled together.
    My confusion is that it appears I can only use one format in a "project" as I need to define the parameters for the "project" in terms of SD vs HD and 4:3 vs 16:9. I was hoping to be able to store all of my footage of various quality definitions and screen ratils into a "library" (which I see is called "Organiser" in Adobe terms) on the hard disk, and then pick and choose from the "library" what to put into each "film project" as I make them, mixing formats as I wish. I realise that in this way, the output would itself switch from 3:4 to 16:9 etc on accasions as I switch scene to use a different format, and so may not appear to be the smoothest edit.... but I thought that would be my decision to do if I wanted it. Instead it appears I cannot mix a few minutes if 4:3 SD footage of my son as a baby, with more recent 16:9 1080 HD footage of him as a boy - which is a real shame as it is what I wanted to do....
    Have I misunderstood, or am I right in thinking I am limited in this way?
    Why can I mix still photos of any format with video , but not mix different formats of PAL video?
    Many thanks

    Thanks Paul
    I have been having go, as you suggest, and as you said, I cannot edit my AVCHD footage in Scene mode as it has 5.1 audio (which I cannot change on the videocam) This is very disappointing as I really wanted the simplicity of the scene editing rather than timeline.... This is compounded by the fact that timeline editing seems even more complicated than normal as each new AVCHD clip I add is placed onto a different video track (not all on track 4 as you thought)- so it is real mess to work out on my screen.... Is this a bug, the way PRE treats 5.1 tracks - or is it just oversight and poor design? Do you think it might be addressed by Adobe?
    Anyhow, thanks for explaining what is going on, as without your warning, I would have assumed that the PRE 7 sceneline feature was broken - you would have thought there would be some warning / documentation from Adobe about this, as effectively, a big feature is now unavailable to me as an AVCHD user - and the product claims to be compatible with AVCHD!
    Would you be able to advise on a couple more of my "newbie" file format questions?
    How do the settings work between setting the initial PROJECT format, and then choosing the various OUTPUT (sharing) settings. For example, if I want to edit a project using AVCHD clips only, and output it to DVD quality to watch on a PC or burn to DVD disk, do I get the same thing if I set the project setting to PAL AVCHD High Def 1080 5.1 and then the share setting to DVD quality output, as I would if I set the project settings to DV 48hz and the share setting to DVD quality output?
    My ideal would be to retain the High Definition of the footage through the editing process and then be able to output to DVD quality now, but to be able to come back to the finished edited project in the future and choose to output it to Blueray when I have upgraded my TV and bought a blueray player. i.e. I am trying to make my filming and edited projects a bit futureproof....
    Finally, I am just not up enough on all the different format lingos (Mpeg1, MPeg2, H.116 etc) to know what I should be choosing from the list for:
    1) outputting edited footage to burn onto a DVD to be used on a DVD player
    2) outputting edited footage to save onto a PC to view in the best possible quality on a PC
    Do you know?
    Many thanks

  • CS5 Mixed video formats in same sequence

    New user here, coming from 10 years of Pinnacle Studio.
    I have at least 3 different video formats to edit in a single sequence/timeline (iPhone, 720 from a Sony point and shoot, and 1080i from a Canon video camera).  Depending on which format I use for the sequence, when I drag the videos to the main Video1 track, the audio will go to whatever track is appropriate for the mono or stereo.  The iPhone and Sony are both mono and will be on one track and the Canon is stereo so its audio jumps to a stereo track.
    So my question is, am I to make all my audio adjustments on both mono and stereo audio tracks depending on where I am making the volume changes on the timeline.  In Pinnacle, the video and associated audio was always on the same track pair.  I was just wondering about workflow and the best way to make the audio adjustments.
    My project typically is a year's worth of video footage from the 3 sources, making a Year In Review.  There will be soundtrack music from MP3's playing most of the time, with occasional original audio from the video being raised in volume for a few seconds, then fading back down to hear the soundtrack again.  Keeps the audience more engaged.
    One more question, what format should my sequence match, the 1080 stereo Canon, 720 mono Sony, or mono iPhone.  The end result will be for viewing on widescreen big screen DVD or Blu-ray.  I would like to make the mono on both channels, and upscale to 1080.  Not sure the best way to do this.
    Any guidance will be greatly appreciated.
    Thank you.

    Thanks everyone.  I will continue practicing and learning.
    Another question.  After editing, I will be authoring with Encore, which I have used for many years now.  How should I structure the audio tracks in Premiere if I want the mixed audio track (master track) to be the main audio track in Encore (on the DVD) and the original video's audio to be the alternate audio track in Encore (on the DVD).  While playing back the DVD, the default audio track will be the mixed audio.  If the viewer wants to hear the original audio only, then they just hit the audio button on the remote to switch tracks.
    Funny how this requirement came about.  For years I always just had the mixed audio.  But while watching the videos, my kids started telling me they wanted to hear the original audio.  Knowing the video would be too boring to adults without a soundtrack, I came up with the idea for multiple audio tracks.  That requirement eliminated the consumer lever authoring tools, which is what pushed me to Encore back in the version 1.0 days.  Since then I have been using Studio to edit, and Encore to author.  Now I am trying to dump Studio and use the Premiere/Encore pair.
    The way I did this in Pinnacle Studio was to output the video and mixed audio as an AVI or MPEG2 file, and import this as the timeline in Encore.  Then from Pinnacle Studio, output the original audio only as a .wav file, and import this as a 2nd audio track in Encore.  Worked perfectly...as long as Pinnacle kept the audio in sync, which at times was difficult, but with the newer releases of Studio, it worked much better.
    Thanks for all the guidance.

  • Help with Audio Formats please

    Hello everybody,
    I have a problem with the Java AudioFormat class, can you help me please?
    The thing is Im working on a voice chat using the following audio format settings:
    AudioFormat(AudioFormat.Encoding.PCM_SIGNED, sampleRate, sampleSize, 1, 1, frameRate, false);
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    audioReadBytes = 1024;
    sampleRate = 8000.0F;
    frameRate = 8000.0F;
    sampleSize = 8;
    My problem is that these settings can cause incrementing voice delay and sometimes even block the application. It works fine on a LAN but on the internet it never works.
    I wanted to ask you if there�s a better audioformat setting that can decrease the size and quality of the sound, so it works on low bandwidths.
    I�m new to audio formats so I don�t understand them much and when I tried to change the settings by decreasing the sample rate I didn�t get what I expected.
    Please help me...
    Thanks a lot,
    Dan.

    Based on the concept of Unity in a gain chain, if you set the levels properly at the output of the analog mixer, then they should be fine coming into the software after the A/D conversion. As long as you don't clip the digital input, you can adjust levels internally on an audio timeline, so it is always better to err on the side of too little rather than too much.
    Digital maximum is -20 dB which is equivalent to 0 dB VU in the analog world. This provides sufficient headroom to avoid clipping in most circumstance.
    Try a few different levels and see what works best for your needs.
    HTH,

  • Core Audio Format -- Is anybody using it?

    CAF files are a new(ish) Audio File Format/Wrapper type. Apple made it.
    Qs
    Is anybody using it with Logic? -- They do import
    How? Why? -- They have several advantages but are relatively new, and it is not acknowlodged in Logic's documentation AFAIK.
    Soooo...... There are other apps that support this Format, many including Logic will just plop them in as AIFF, for example. The format actually supports many audio file formats.
    So within a CAF there can be audio of several types, sample rates, resolutions... They even support multichannel, markers, regions, music chunks.
    In practice this could be a very useful format. Logic doesn't support all of it's features/formats but could be extended through QT or could hold limitations, probably why it is not listed as a supported format.
    So is anybody taking advantage of this stuff, or hitting walls, having problems using these with Logic Pro?
    Just Curious.
    J

    I have. I mentioned it imports -- prolly just certain cases where the underlying format is Logic approved. Haven't really worked with it much yet though with Logic as it was never cited that it was supported AFAIR. So I am guessing that it's underlying format needs to be a Logic approved one, but it imports 16, 24 and rejects 8 and 32. It also creates independent waveform overview files. I compared a 16 CAF to 16 AIFF, there were 3-6 lines which are different. Of course the header was in that list every time.
    I was asking if there was anybody else using it so.... shrugs. It imports but I don't know of any advantage to use it, at the moment.
    Maybe this will be the de facto 32/64 mixed file format when we get a new arrange window with sample resolution : ^)
    Cheers, J
    Gr8-1. ja, right -- is that a fat joke?

  • How To Get Supported Audio Formats?

    If I have a particular Mixer how do I get the supported Audio Formats
    for every line?
    I know DataLine.Info has a getFormats() method.
    How do I get a DataLine.Info from a Line?
    I know how to get the Source and Target Lines and they subinterface
    DataLine, I just dont know how to get the DataLine.Info from DataLine.
    Thanks a lot!

    Mr Evil, thanks a lot! That was very helpful.
    Your code is going to help me quite a bit when I get to 24 bit audio.
    I love that JSR page. I did miss that relevant FAQ question though.
    Instead of checking for instanceOf i was casting within a try catch.
    It should work now.
    Is there anyone alive in this forum?
    I'm guessing that sound is not a popular application for Java. That was funny. I feel your pain.
    I cant wait to get out of the nuts and bolts of java sound so i can
    move on to the actual DSP fun : )

  • HT3775 This file contains a track in the Dolby AC3 Audio (code "8192") format. You may need to install a QuickTime component for this audio format in order to hear the soundtrack of this file.

    This message appeared and does this just mean I need to upgrade Quick time player? This file contains a track in the Dolby AC3 Audio (code "8192") format. You may need to install a QuickTime™ component for this audio format in order to hear the soundtrack of this file.

    Luckily, there are DOZENS of video/media players and extensions avaliable for the Mac. Many of them are FREE, some of the better FREE ones are...
    PERIAN (quicktime extension)
    http://perian.org/
    QUICKTIME 7.6.6
    http://support.apple.com/kb/DL923
    VLC media player
    http://www.videolan.org/vlc/
    NICEPLAYER
    https://code.google.com/p/niceplayer/
    And if none of those work, there is a possibility that those video files may be corrupted.

  • Finder media encoder, how can I add audio formats support?

    I've noticed the feature where if you right click on a media file, you get an option in the popup menu to encode the file to some Apple-friendly format, I suppose.
    I am trying to find a way to enhance this program's support for audio formats. I have tons of music in FLAC and a little bit in other formats like OGG and -gosh- wma, and I'd rather convert them with a native, Apple-written app than with anything else.
    Any clues as to how we can enhance this Encoder feature's support for audio formats?

    Many video and audio files can be added to keynote slides using the Insert Media browser - the sounds and videos need to be in a format recognized by iPad and need to be placed in the browser via iPhoto then synched to the iPad using iTunes. Sound files can be created using Garageband or Quicktime and appear as movies in the browser so it is best to place them in a separate album with descriptive names.
    If you would like Apple to add additional features, give them somme feedback:
    http://www.apple.com/feedback/

  • Macbook Pro's Mini Toslink Output LPCM 2ch Audio format?

    Hi, I have a Sony Amplifier that accept optical input, but it only accept "LPCM 2ch audio format", so does Macbook Pro output this format?
    http://www.sony-europe.com/support/emanual/HAP-S1/HG/EN/contents/TP0000221964.ht ml
    Thanks!!

    Hi, I have a Sony Amplifier that accept optical input, but it only accept "LPCM 2ch audio format", so does Macbook Pro output this format?
    http://www.sony-europe.com/support/emanual/HAP-S1/HG/EN/contents/TP0000221964.ht ml
    Thanks!!

  • What audio format encoded into iTunes HD Movies and TV shows?

    What audio format encoded into iTunes HD Movies and TV shows, AC3, 5.1 DolbyDigital? Its been a long long while since I rented an HD movie on ATV since netflix came around with streaming and BD. Now they have rentals and purchases. Anyone know if the purchases play with 5.1 or better audio?

    I think if you pull up the movie in the iTunes store it should tell you. For example, I pulled up the Star Trek movie that came out a few months ago, and it reports "Dolby Digital 5.1 surround sound". Clearly you need to make sure that your hardware supports whatever sound configuration the movie employs.

  • What's the highest quality audio format to import CD?

    I've a 40 GB iPod. the system default to AAC. But i'm planning to buy a Shure e4c ear-plug and want to make sure i've the best quality audio format recording in the ipod.
    could someone tell me what audio format yields the best audio quality to import CD? i'm only used less than 20GB of the ipod.
    so i don't mind using the best audio format which may take the most space.

    Simple answer: Apple Lossless. You will hear the song at the same audio quality level as on the CD--at about half the size.

  • Some advice on audio formats and converting tracks

    _*Some thoughts about audio formats and conversions. These are all supported by iTunes unless stated otherwise.*_
    General principles : there is no point in converting an already compressed track into a higher bit rate - it may seem as if this would give an improved quality, but once information has been lost through compression, there is no recovering it. The sound MAY be improved subjectively through the use of an appropriate EQ setting.
    This is not a definitive list : other formats, for example the one used by Real Player, and the ATRAC minidisc codec used by Sony, are not listed : many of the following can be used in iTunes, or are reasonably well-known.
    To use one of the available formats, they should be set in the Importing tag in *iTunes Preferences* (Advanced prior to iTunes 8, now in General). Once set, the format is used for importing CDs, but can also be used to convert tracks already in iTunes (using the Advanced menu : +Convert Selection to xxx+ ). It is not used when dropping files onto iTunes, or using Add To Library.
    1. _Lossless formats_
    Certain high-quality audio formats are "Lossless". This means that if the track is re-converted into the same format (after editing, for example), it will not degrade no matter how many times that is done.
    .AIFF - a format (common in Macs) representing full CD quality. Bit-rates are over 1400 kbps; +1 minute of music is approx 10MB.+
    .WAV - more or less the same as .AIFF but a Windows format.
    *Apple Lossless* - a 'once-only compression', applied to .AIFF or .WAV tracks to retain quality but reduce file sizes. A little (totally inaudible) information is lost, but also a more efficient 'codec' (compression algorithm) is involved. +1 minute = approx 5MB.+
    .FLAC - similar to Apple Lossless, but less proprietary (a more open standard). Not supported by iTunes.
    2. _'Lossy' formats_
    These formats are so-called because each time the conversion is done, the track is re-compressed, and more information is lost even if the file size remains the same. Various quality settings can be chosen, from around 320 kbps (high) down to below 128 kbps, though 128 is reckoned to be the lowest acceptable bit rate to listen to music in stereo.
    In general, the higher the input quality, the higher the output quality when 'stepping down'. Therefore an .AIFF track converted directly to 128kbps 'lossy' will sound better than a 256kbps track re-compressed to the same 128kbps.
    .MP3 - a standard audio compression format that has been around for many years. Common everywhere, and supported by virtually all contemporary music players. The codec has been improved so that modern MP3s sound markedly better than earlier versions. +1 minute @ 128 kbps = approx 1MB.+
    .AAC - a superior codec to MP3 (though the quality gap has narrowed), which has been chosen by Apple as the iTunes format of choice. Nowadays, a track at 128 kbps .AAC will sound around the same quality as the same track at 160 kbps MP3. +1 minute @ 128 kbps = approx 1MB.+
    .WMA - Windows' own proprietary lossy codec. It seems to have a bright and sparkly sound compared to MP3, but lacks a decent 'bottom end', i.e. the bass tones are somewhat lacking. Not supported by iTunes. (Can be played using VLC).
    .OGG - the open standard Ogg Vorbis format, often held to be superior to MP3 but little-used and therefore not supported in many platforms and players, including iTunes.
    When considering which format to use, the most important things to remember are
    1. how will these tracks be played back? (on a hi-fi? an iPod? expensive earphones?)
    2. the quality will depend finally on the judgement of the listener (some ears are better at telling small differences in quality than others)
    3. how much space is available to store (and expand) the music library?
    As for quality, a good way to decide is to "audition" different formats. Using one CD track known to you, try importing it at several different formats and bit rates (using iTunes Preferences). Then just play them, see which sounds best to you. In this way I came to decide for myself that AAC @ 256kbps was the best for me +(1 minute = approx 2MB).+

    The Windows version of iTunes supports importing of WMA files, converting them to one of the other formats. Mac iTunes offers now support of WMA files at all.
    Good point - and if I knew more about iTunes on Windows I would amend that and post a different version of the article in their forum.
    I accept the point about .WAV files being used also for compressed formats but didn't want to make it too complicated; the most common use for .WAV I believe, is full quality audio.
    On the subject of Lossless compressed audio (FLAC and AL), Wikipedia has this :
    _*Difficulties in lossless compression of audio data*_
    +It is difficult to maintain all the data in an audio stream and achieve substantial compression.+ +First, the vast majority of sound recordings are highly complex, recorded from the real world. As one of the key methods of compression is to find patterns and repetition, more chaotic data such as audio doesn't compress well. In a similar manner, photographs compress less efficiently with lossless methods than simpler computer-generated images do. But interestingly, even computer generated sounds can contain very complicated waveforms that present a challenge to many compression algorithms. This is due to the nature of audio waveforms, which are generally difficult to simplify without a (necessarily lossy) conversion to frequency information, as performed by the human ear.+
    +The second reason is that values of audio samples change very quickly, so generic data compression algorithms don't work well for audio, and strings of consecutive bytes don't generally appear very often. However, convolution with the filter [-1 1] (that is, taking the first difference) tends to slightly whiten (decorrelate, make flat) the spectrum, thereby allowing traditional lossless compression at the encoder to do its job; integration at the decoder restores the original signal.+
    +Codecs such as FLAC, Shorten and TTA use linear prediction to estimate the spectrum of the signal. At the encoder, the estimator's inverse is used to whiten the signal by removing spectral peaks while the estimator is used to reconstruct the original signal at the decoder.+
    This seems to imply that even Lossless codecs make changes, but it IS a very complex subject. If I could still edit the main article I would remove that bit about AL losing information.

  • Best audio format for recording speech?

    Hello everybody,
    I was wondering which audio format (encoding, sample rate, sample size) is considered best for recording speech in regard of memory usage and perceptible/acceptable quality loss.
    What do you suggest, what are your experiences, what reasons can you state?
    Thanks!

    I finally found something:
    [sample rate for speech|http://wiki.audacityteam.org/index.php?title=Sample_Rates#32_kHz_.2F_14.5_kHz]
    [bit depth|http://wiki.audacityteam.org/wiki/Bit_Depth#16_Bit]
    I know audacity is not an application based on the java sound api but these wiki articles provide good common information on signal processing.
    According to the article I believe 32kHz ist the optimal sample rate for recording speech. But I still don't feel satisfied by the information given about the bit depth. I think 8 bit is too low but 16 bit (CD quality) might be too sumptuous. Unfortunately the quality I can record with is limited my recording and output hardware so it might become hard for me to top that out. Maybe someone can help.
    Is it true, that the encoding can be customized with codecs by extending the audio service provider? How would something like that work?

  • What can i do to fix the following : You may need an additional audio decoder to play the soundtrack of this file. This file contains a track in the Dolby AC3 Audio (code "8192") format. You may need to install a DirectShow decoder for this audio format i

    You may need an additional audio decoder to play the soundtrack of this file.
    This file contains a track in the Dolby AC3 Audio (code "8192") format. You may need to install a DirectShow decoder for this audio format in order to hear the soundtrack of this file.

    Luckily, there are DOZENS of video/media players and extensions avaliable for the Mac. Many of them are FREE, some of the better FREE ones are...
    PERIAN (quicktime extension)
    http://perian.org/
    QUICKTIME 7.6.6
    http://support.apple.com/kb/DL923
    VLC media player
    http://www.videolan.org/vlc/
    NICEPLAYER
    https://code.google.com/p/niceplayer/
    And if none of those work, there is a possibility that those video files may be corrupted.

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