Forum Replies Created

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  • Michael Gissing

    July 24, 2007 at 11:58 pm in reply to: Audio Filter

    You need both a high pass and a parametric or notch style filter. Rooms have resonances that amplify certain frequencies. Find that frequency by first adding + gain with a notch Q around 10. (Q is the width of the notch. The lower the Q the wider the frequency band being affected.)

    Trim the frequency up and down until you hear the resonance amplify. Trim – gain to taste. There may be a secondary or harmonic resonance. If the primary resonance is around 120hz (very common frequency in large buildings)then it is likely that a harmonic resonance may occur at 240hz and possibly 480hz. Once you find the low primary resonance try a second notch at double the primary frequency. It is always easiest to tune in to resonance by + gain, then trim -.

  • Michael Gissing

    July 24, 2007 at 11:51 pm in reply to: removing word in interview technique

    Dialogue editing really needs a sub frame capable editor that can also do dissolves of any length, anywhere within the frame – in short an audio DAW. I have seen editors struggle with neat dialogue edits in FCP and AVID that would be simple in an audio editor.

    Filling with matching room tone is essential. The best room tone comes from a clean section as close to the dialogue that you are editing. It is nice when a room atmos is done after an interview, but in my experience, it rarely matches the hole you want to fill. This can be because something changes in the room, the mic is slightly differently placed or the level has been changed throughout the interview.

    If the job is going to sound post then getting an edit close is nice for the offline screenings, but let the sound editor with proper tools get the final edit right. If you have to do it within FCP then try Sound Track Pro. I never use it because I have audio DAWs here that are much better but if it lets you do sub frame accurate edits then that is an improvement.

    Also invest in a good pair of speakers or headphones. Most picture editors when they get into sound post say.. “oh I never heard that before”.. when their dialogue edits and atmos fills bump all over the place.

  • David, exceeding the bandwidth that set top players are throttled at is possible with any DVD if you don’t get the balance of audio and video data rates correct. AC3 is not the magic bullet to getting that balance right. Knowledge of data limitations and ways to get the balance right using any of the audio options is important. If a desire to correct oversimplifications is going to confuse the forum then my apologies.

    If it was about bandwidth, then no-one would use AC3 over MPEG variable bit rate audio. AC3 is fixed bit rate and uses far too much data for the ordinary sound it produces. Years ago (1997 from memory,)I did a test disk with an AC3 2.0 and an mpeg stereo track. This was played to the Australian Screen Sound Guild in a blind listening test. 100% of the audience chose the mpeg stereo over the AC3. The files sizes? AC3 file was twice the size of the mpeg.

    The reason that the Motion Pictures Expert Group original rejected Dolby AC3 should not be forgotten. There are many out there that have forced Dolby encoding on DVDs and digital broadcasts. It never should be considered anything other than optional.

    By all means share your winning formula for making bullet proof DVDs, but please don’t proclaim that Dolby is rightly the only audio format for DVDs.

  • Just make your aif file from FCP directly not using Compressor. Use the same in out markers and the file length will be the same as your video mpeg2 file. If Compressor 3 has dropped aif, then mores the pity.

    I have both 5.1.4 and 6.0.1 but haven’t used Compressor 3 yet so I was unaware that they had dropped the aif option.

  • David, what a strange comment to make. AC3 is but one of four options for audio on a DVD. Uncompressed PCM, DTS and Mpeg2 multichannel are the other three perfectly valid options to put on a DVD.

    When I put my Led Zepllin DVD on, I love to use the DTS or PCM sound tracks that they have. AC3 is a poor codec and my least favourite as an audio professional. For the record, the Compressor presets that I have used so far include aif and AC3. Not sure which one you are trying but many of mine in 5.1.4 and 6.0.1 have aif.

    If you can’t find them, make an aif mixdown in FCP and then import that into DVD SP. You can happily put an aif file in DVD SP.

  • Michael Gissing

    July 21, 2007 at 7:08 am in reply to: layback off by one frame

    I have the same issue with HDCam decks. No problem with digi beta. I trim the in frame in the edit to tape box 1 frame to fix it.

  • Michael Gissing

    July 19, 2007 at 1:28 pm in reply to: calculating space

    https://www.dvguru.com/2005/10/05/dashboard-widgets-calculating-hd-space-for-video/

    three seconds to find this in Google by pasting in DV space on hard drive.

  • Michael Gissing

    July 18, 2007 at 2:10 am in reply to: Export Questions! Please help

    Here https://www.blackmagic-design.com/support/software/
    is the Blackmagic codecs for Mac & Win XP. My understanding is that the Blackmagic 10 bit 422 SD codec is the same as the Apple 10 bit 422 SD.

    Worth a try as you already have the file rendered and an Win XP codec should be easy to download.

  • Michael Gissing

    July 18, 2007 at 2:05 am in reply to: Digibeta audio bitrate

    The 24 bit gets truncated to 20 bits unless you pre dither it. I don’t think the digi beta does dithering. Don’t fret about it.

    Some dithering and noise shaping can create problems if you don’t know what you are doing. I have not heard any problems dropping a 24 bit file and laying off to digi beta and we are a sound post with FCP facility. Similarly 16 bit files work fine as well. A typical broadcast sound track won’t have anything like the dynamic range that would need a 24 bit file anyway.

  • Michael Gissing

    July 16, 2007 at 8:43 pm in reply to: “normalizing” final audio tracks?

    If you are sending an OMFi to sound post then they won’t want you to normalise clips. When you put level adjustments and various FX like EQ in the FCP timeline then the clip remains unprocessed for sound post, just the way they like it.

    Processes like normalise render a new clip. If you do this then you might find yourself up all night replacing clips with original audio to make the OMFi for the sound post people.

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