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  • avatar
    44 sounds
    99 posts

    yes, i know that the noise floor will be present from the preamp no matter what gain setting i use. the problem there is also noise created in the ACD. This is why if you record silence with no preamp running to your adc you can still get noise if you crank the signal up enough. So when I normalize a low level signal i am not only turning up the noise from my preamp but I am also turn up the noise from the ADC!!!!!! Double whammy when you are recording at low volume levels. This is why you don't do it. Use the dynamic range that 24 bits gives you.

    NEVER NEVER NEVER use a compressor in your signal chain when tracking. That's audio basics 101!!!!!!! Of course clipping your peaks isn't good either but I would rather take a second run at something to get a better signal than to record at low levels or use a compressor during tracking.

  • avatar
    0 sounds
    49 posts

    It doesn't work like that because you've already normalised it.

    You're boosting the gain on ur pre-amp which is also raising the noise level. Where do you think the noise is coming from?

    Consider this.. You get ur pre-amp and you record a singer. You think to yourself 'I must make this as loud as possible because I want to make use of my 24bits' So you turn up the gain.. find yourself clipping at the peaks... so you add a compressor so you can record to your DAW at a higher volume - permanently altering the vocal whilst ur at it.

    What you've just done is boosted the noise floor... by raising the gain on ur pre-amp... you've then compressed the signal.. making the difference the loud bits and quiets bits smaller.. which has infact raised the noise floor again.

    Just forget about the compressor, reduce the volume.. record in to the DAW. apply a gate removing the noise and don't worry that ur signal is representing 20bits because you now thats plently of dynamic range and you fine well know chances are its going to be mixed down to 16bit anyway.

  • avatar
    0 sounds
    5 posts

    hello! you can download some stuff... here it is cool
    http://download.yousendit.com/D55C01610DE716B3
    goodluck to ya

  • avatar
    3097 sounds
    480 posts

    Yes, it sounds much like one. You can compare with this other sample in which Acrocephalus arundinaceus dominates:
    http://freesound.iua.upf.edu/samplesViewSingle.php?id=18970

    D

  • avatar
    121 sounds
    1552 posts

    you seem to have succesfully uploaded some files:
    http://freesound.iua.upf.edu/packsViewSingle.php?id=2661

    are you still experiencing this problem?

    try using directories/filenames without strange characters, don't use spaces either...

    "rev,s metal guitar loops" is a bad idea for a directory name wink

    try
    directory: loops
    files:
    loop1.mp3
    loop2.mp3
    loop3.mp3
    loop4.mp3
    loop5.mp3
    etc...

    - bram

    PS: please do not write in ALL CAPS. it looks like you are shouting.

  • avatar
    72 sounds
    1 post

    I FOLLOWD ALL THE DIRECTIONS AND FILEZILLA REPORTED ALL FILES UPLOADED BUT THERE NOT ON FREESOUND WERE ARE THEY AT HELP!!!!!!

  • avatar
    44 sounds
    99 posts

    mike-t3
    You've not answered my questions.

    Go on try it. Question your technique. Why do you do it?

    Glad to answer your questions:

    I push the levels hot in order to keep the noise floor low. It is just good practice especially if you plan on normalizing all of your recordings and mixing them to a soundtrack, musical composition, video game, etc...... Let's say you normalize a file that you had peak values at -12db. This raises the noise floor 12db for this particular sound. Now you add 40 or so tracks to a multi-track session, which you have used the same 12 db normalize. Even at 24 bit there will be audible noise present in your mix.

    When you are recording at 24 bit and you have your levels peaking at -12db you are really only useing 22 bits of the signal. If you are going to record in 24 bit why not use ALL 24 bits!!!!!!

  • avatar
    44 sounds
    99 posts

    Sorry, I chose 0 to -3. sad I didn't know which one to pick, because I submit to both options, but I could only choose one. Right now I am working as a Sound Designer for a video game development studio.

  • avatar
    0 sounds
    2 posts

    hallo,
    I am new to the freesoundproject and I am very curious about the work you are doing here. I very much like the idea of showing the waveform (or a piece of it) of every sound. I am currently working on a project where I am trying to find visual representations for voice sounds (not a music visualizer). I am interested whether it is possible to "read" a waveform (or other kinds of sound visualisations) in order to get information about the content or structure of human conversations (e.g. number of speakers, overlappings, emotional state..). If anyone knows if thats possible please tell me.
    Another question: the waveforms you are showing to each sound file have different colors, why is that? they range from dark blue to yellow. Maybe it is obvious, but I am totally new to this field, thanks for any answer, Juli

  • avatar
    859 sounds
    73 posts

    Any ideas, what bird sings here?

    ERH suggested a Reed warbler and he might be right! :lol:

    http://freesound.iua.upf.edu/samplesViewSingle.php?id=53697

    Inchadney

  • avatar
    26 sounds
    6 posts

    keep at it, you'll do good in few years

  • avatar
    128 sounds
    198 posts

    Hear hear. From your description I hope you chose "..based on source and circumstance...", but I also know you might have been answering on the basis of what is desireable for the overall project. I admit a poll cannot allow all options at once.

    I too regularly do both controlled and un-controlled environments. If I had my way I would do controlled ones, but nature calls sometimes.
    (You and I should probably share some material someday soon) What kind of studio do you have going on ejfortin?

  • avatar
    44 sounds
    99 posts

    Of course it depends on the material you are recording. If it is going to be something you only get one shot at without any type of sound check you are gonna have to be on the safe side and keep the lower. I have recorded other live events where I do get a sound check, but I don't always trust the performers to be at the same level (they usually get a bit more excited during the performance) so I push the levels a bit higher but still need to keep a good bit of headroom for any sudden craziness the performer decides to throw at me.

    On a controlled recording I always push the levels as hot as I can. By controlled I mean I can take as many takes as needed, and the environment is controlled (such as in a studio). For these types of studio recordings I try to get everything between 0 and -3. I try and use every bit available, to give myself the best / cleanest signal possible.

  • avatar
    128 sounds
    198 posts

    Given the recent discussion about how much Headroom should be observed for recordings of every kind, what do you think? In general terms only, please.

  • avatar
    128 sounds
    198 posts

    You're sounds are fine. Part of the reason I apologlized for various highjackings was this thread has become something differnet than the original intention.

  • avatar
    121 sounds
    1552 posts

    Mart[1001],

    could you try another FTP application like filezilla (pc) or cyberduck (mac) and see if that helps?

    - bram

  • avatar
    0 sounds
    1 post

    Hi everyone, I want to run my own music studio and for that I need a little funding from reliable sources. I am 28 years old and I am looking for flexible options for going about this. Can you people suggest me any ideas.

    Thanks in advance!!!

  • avatar
    26 sounds
    6 posts

    http://zamp.kicks-ass.org/mus/analog_alienation.mp3
    its a preview, kinda lost ideas at what it is in that mp3. But I've come up with some new stuff to it.
    I've been working on that same song for quite a while now.. its about 9 mins long atm.. no idea where to stop :lol:

    Samples probably used, cant remember which one actually are

    By patchen (http://freesound.iua.upf.edu/usersViewSingle.php?id=6997)
    Rhino-11.wav (http://freesound.iua.upf.edu/samplesViewSingle.php?id=3385)
    By Erratic (http://freesound.iua.upf.edu/usersViewSingle.php?id=15)
    ufo_atmosphere.wav (http://freesound.iua.upf.edu/samplesViewSingle.php?id=235)
    ---------------------------------------
    May 12, 2008
    By pulse00 (http://freesound.iua.upf.edu/usersViewSingle.php?id=14604)
    p_ambient_beat1.wav (http://freesound.iua.upf.edu/samplesViewSingle.php?id=53404)
    p_glitchbeat.wav (http://freesound.iua.upf.edu/samplesViewSingle.php?id=53408)
    ---------------------------------------
    May 6, 2008
    By NoiseCollector (http://freesound.iua.upf.edu/usersViewSingle.php?id=4948)
    DootDaDoot.wav (http://freesound.iua.upf.edu/samplesViewSingle.php?id=2819)

  • avatar
    0 sounds
    1 post

    I think that the first step you should do is to gather a few friends of yours and test the beats on their ears…because they will be able to provide a judgment on your composition…and that too free of cost. The serious part about going on this would be getting to vmusicbook.com which can really help you to make it through this by giving you the necessary sources and plans to approach the management companies.

  • avatar
    20 sounds
    25 posts

    Re the hijacking: No problem -- there's much to learn in here smile
    Re bit rate/depth: I did this error myself in my post tongue

    mike-t3
    I'm saying that ur hardware such as preamps wont perform as well at higher volumes.

    I want to understand you and must therefore question this for verification: Is this the core basis for your argumentation?

    If this behaviour of the preamps is the case, then because of the implication I'd have generally a problem: Because this is in the analogue circuitry. (The digital recorder wont do anything as soon as the signal became discrete other than saving it to disk)
    And spinning this thought further, because this is common analogue circuitry, I'd also have this problem when I plug in a microphone into a mixer and preamp its peak to unity - this is always done as the first step when using a mixer.

    This goes hand in hand with the question as to why this is done: To have a unified peak level of all signals in my mixer (= unity) which is at what is labelled "0dB" and describes the 'best' operating level of the mixer according to the manufacturer. I would be very worried if the hardware does not perform as it should at what is defined as the best operating level - especially if it's a common recorder like the Fostex-FR2 which I have.
    (If it were for the preamp alone, I'd also expect it to have a controlable range that controls its behaviour linearly)

    Or, looking at it from another side: If this problem lies in the hardware before the ADC (signal flow wise), then I'd have it always, regardless of bit depth, right? Is this high-peak problem then 'lesser' than the problems of quantization errors and high noise floor in 16bit?