Forum Replies Created

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  • Michael Gissing

    January 19, 2009 at 5:08 am in reply to: FCP & Audio Normalization

    “Rich’s experience, and the audio post he works with, is clearly different from yours. It’s also more appropriate to the original poster.

    I think it should be made clear that I am not saying the original poster shouldn’t use the Levelator. It may be perfect for his application. I didn’t respond to any of the arguments except factually incorrect statements and to remind all FCP users of the need to consider sound post before considering any audio processing of the raw file data.

    For many, audio is misunderstood and I will always challenge errors or omissions that further obfuscate.

  • Michael Gissing

    January 19, 2009 at 4:57 am in reply to: FCP & Audio Normalization

    Oh and thanks for the appropriate user pic…

  • Michael Gissing

    January 19, 2009 at 4:42 am in reply to: FCP & Audio Normalization

    “And, now we all know that the next time we’re considering heading off to Tasmania for a final mix with Michael, that he might blow a gasket if we premix our V.O.

    Here he is going after the last offender…”

    Frankly David, I have no interest in getting personal with you and as a Cow host, I expect you to play the ball and not the man. Your statement – “In the digital domain there are no ill effects from adjusting amplitude or pre-mixing,” remains factually incorrect.

    If you wish to stand by it, then fine. I suspect you have confused amplitude with gain adjustment. If you adjust amplitude, you change the relationship between the peaks and troughs of a waveform. If you change that relationship, you are either compressing or expanding the sound. Compression, if done incorrectly will raise the noise floor. That’s why good compressors have a threshold setting. Even so, applying compression will change the noise floor relative to the maximum peak.

    If you meant gain adjustment, unfortunately you are again incorrect. Gain changes automatically raises or lowers the noise floor. Relative to the sound, the noise floor remains constant. Because digital signal fall into their own ugly noise floor, it is important than they be gained by a much higher bit rate than their native rate. Best solution is to use floating point processing so they you don’t introduce too many artefacts from that digital noise floor when you gain.

    Lots of simple normalising programs don’t do floating point, which is why I constantly warn people against changing the raw media as digitised.

  • Michael Gissing

    January 19, 2009 at 1:18 am in reply to: FCP & Audio Normalization

    Just so that I wasn’t speaking without knowledge, I downloaded the Levelator and dropped a wav file of a short film final mix. Levels do change so broadcast spec is no longer correct.

    Secondly the compression/limiting is what I would classify as soft character. For me this is not good as it flattens a good dynamic. A great limiter caps peak levels but doesn’t soften or flatten the sound. I won’t get to technical on compressor limiters characteristics, but suffice to say this product will help smooth some vocals, particularly narration, but it is not ideal for sharp transient effects. It might help some music, but most music is pre processed anyway.

    I also think it would not be ideal for final mixes that are complex and have strong dynamic sound effects, but Ok for music & dialog simple tracks.

  • Michael Gissing

    January 19, 2009 at 12:57 am in reply to: FCP & Audio Normalization

    Firstly, the desire to get audio that has not been pre processed is not old school or billable hours based. As an audio post pro who also runs FCP for grading & online, I can assure you that material that has been normalised or compressed in any way is not what I want. The reason is simple. My gaining, EQing and compressing tools are vastly superior. For example, on a Fairlight all EQ & dynamics are done at over 70 bit floating point resolution. 32 bit floating point resolution is the minimum for mix processing.

    Most disturbing are ill informed ideas. David Roth Weiss’s quote -“In the digital domain there are no ill effects from adjusting amplitude or pre-mixing” is so wrong and the sort of miss education that makes my job harder. Please don’t ever think you are helping a sound mixer by premixing. Think how hard it is to color correct a composite layered shot with titles. You can’t unpick layers or avoid affecting text. I am sure you will all understand that this is a mistake and an audio premix is no different.

    By using FCPs filters to change gain or EQ, this leaves the base sound unaffected and so when it exports via OMF, us sound pros have the raw file, not a normalised or pre compressed signal. Most normalising doesn’t use sophisticated floating point processing. If you send files out to something like the Levelator, you are firstly baking in compression and also losing timecode reference on the files. It also means you have to export wav or aif files, process them and then reimport. I presume this means manually resyncing unless, horror of horrors, you are making long aifs of whole tracks which means handles are gone when it finally gets to sound post. Either way it sounds like a lot of manual work to round trip, just to use an automated leveler.

    If you need to use this tool to gain match your final mixes prior to podcasting then it says to me that your original sound post was poor. Don’t use it on broadcast finalised mixes as it will likely make your mix illegal and it should be totally unnecessary anyway.

    Finally I can see why these guys developed the product and it makes sense in their application, but please don’t use it if you are going to sound post from FCP. I am not bagging the Levelator. It has applications for creating a standard loudness on finished material for the web that is not already broadcast standard.

  • Michael Gissing

    January 19, 2009 at 12:08 am in reply to: Audiofilter “Away sound”

    By far the best way to do this is with a convolution reverb using a sample that is recorded at a distance and through a wall. FCP doesn’t have the ability to use this type of software but it can be run stand alone on a VST software host. SIR2 is a brilliant and cheap convolution reverb and there are heaps of sampled environments to use freely available on the web.

    The main thing that is attenuated by distance and walls is high frequency so rolling off low frequencies is optional. Reverb should be set for a short decay but an initial delay of at least 40 msecs.

    If all of this sounds like double dutch, then you begin to appreciate the value of going to sound post pros who not only understand but have these tools. FCP is not an audio editor or mixer. For audio it is only an offline tool compared with dedicated audio systems like Fairlight etc.

  • Michael Gissing

    January 18, 2009 at 11:59 pm in reply to: sharper footage

    Whilst I have the Nattress filters, I also recommend Lyric’s Masked Sharpen filter.

    https://www.lyric.com/fcp-plugins/lyric-effects/masked-effects/index.html

    It works much better than the FCP sharpen filter and can help soft shots without going too edgy.

  • Michael Gissing

    January 14, 2009 at 10:56 pm in reply to: Final Cut Pro Export to HDCAM

    Too many conflicting issues here. To go to HDCam tape, you should export a quicktime. I would recommend that you talk to a dub house and they will advise what codec is best for them to take your final file and record to HDCam via a card (Kona/ Decklink etc)as you can’t do this yourself without an SDI card and hiring in a complicated machine.

    As for making a DVD, you need an mpeg2 file which is standard def. If you use the Compressor pre-sets then you are doing fine.

    (For Shane)- Strictly speaking HDCam is a codec, designed to record compressed data to tape. Neither FCP or AVID can create or read files in this codec however, but I believe Vegas can.

  • Michael Gissing

    January 14, 2009 at 10:46 pm in reply to: DigiBeta Output has Audio Distortion

    Firstly make sure the deck is set to reference to input signal and not external reference. It may be a sync issue but I suspect the problem is tape. When you get a red channel condition it means the machine is madly trying to read and correct corrupt data off tape.

    Try a different tape and if the problem is still happening, then it may be a sync issue. It has been known that bad tape batches can happen.

  • Is your sequence set to 4 channel? That’s the first thing to check and then you also have to assign your channel outputs.

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