Hearing - add reverb tail?

What is the process of adding a silent area/part/section to (usually the end of) a sound file, which can then accept a reverb tail.

In soundforge use > insert silence > then treat with plug-in

Pro Tools via Audio Suite, it automatically adds the tail

In the edit view, generate/Silence and then enter the number of seconds that you'll need.

Bob

Tags: Audition

Similar Questions

  • How to add a tail to the end of a text in photoshop?

    How to add a tail to the end of a text in photoshop?

    I tested the police (from dafont.com) in the real Photoshop on a Windows 7/32 bit computer.

    It is all ok.

    Fenja

  • suggestions of reverb appropriate for an orchestral concert

    Hello good people of this forum,.

    I need some experienced (been there, done that) advice on how I did to the best approach to adding reverberation to some recordings of the classical Symphony. I came to be in this situation because I did some work video for one of the bands in the South Florida, recorded in some of the large halls from here.

    This led me to do some audio only for them to deliver as gifts from donors freebie

    Problem is I am anything but a sound engineer and have no experience, do something serious in post beyond lo - cut filter as well as the minimal standards etc.

    Conductor said that she would like to hear some reverb, so I have to add the parameter "concert hall.

    Someone here can give me some real advice on presets, settings, etc.? I use CS6.

    Thank you!

    wsmith

    How did record the concerts initially.   Usually, when you save an orchestra in a large concert hall, it is quite natural reverb that you don't need to add an extra into the mix (unless you close MICS all sections of the Orchestra).

    However, if the room you were in was particularly dead or something, I probably would work with "Reverb" (Reverb Reverb/effects/Full) in hearing.   It is an effect very requested resource so you may have to work in waveform mode and edits destructive then use "save under" after each try to keep your original.

    Anyway, there are 3 presets 'Medium Concert Hall' under the reverberation.  I would have a look for everyone to see who are closer to the desired effect - then you can play with the adjustments in the reverb (things like "Room size" 'Decay time' and some of the "first reflections" adjustments are useful).

    If any of the Concert Hall presets give you the kind of thing you want, you can try 'Church' or 'Great room' presets and, again, have a game.

  • Need a note of a song to support / extension / reverb out for the transition of the music

    Need a note of a song to hit and reverb / support / melted, so I can pass in a different song.

    I can add reverb, but how to extend the specific note of the song if possible so that it sounds and fades gradually?

    It's a question that is not the last note of the song already?

    You need the FX version (not the VSTi) of this (the one on the left):

    TimeFreezer - Audio and Music Instruments

  • None of the keyframes on Reverb

    Hello

    I imported a clip with two separate mono tracks, it's a show and I want to add reverb to the vocal numbers, but the reverb effect does not appear to be key images, y at - it another way to turn the reverb on and off at different times?

    To activate the effect and the wide use of derivation which you can keyframe.

  • What is the purpose of the dithering? Do I need?

    I understand that the dithering covers apparent audible differences when converting down types of samples, such as PCM 24 bit to 16-bit PCM. However, I edit audio using the following setings:

    Format: Wave PCM (*.wav, *.bwf)

    Type of sample: 192000 Hz, 32-bit

    Bitrate: 24576 Kbps

    Format settings: wave non-compressed, Point 64-bit floating (IEEE)

    Markers and other metadata included

    Big estimates, such as 6 min = 1 GB file sizes

    I then open it in iTunes and convert m4a, ALAC in about 1846 Kbps, 192 kHz, 16-bit files.

    Whenever I have listen subsampled audio files I've been around, I never noticed an addition of background noise, nor any difference at all, a fortiori all I didn't like.

    Should I add dither to my WAV files before exporting their?

    Apple iTunes automatically adds the dithering when converting to the ALAC, m4a? If this isn't the case, does?

    And if I were to export WAV files as files of low-bit of hearing CS6, I should add dither?

    I tend to music remaster to eliminate useless noise which has no acoustic effect. I'd be unsatisified if I added the dithering?

    Are there specifications of dithering I can add which is not audible, but serve an important purpose?

    Any thoughts are welcome! Thank you!

    Steve.KR wrote:

    I understand that the dithering covers apparent audible differences when converting down types of samples, such as PCM 24 bit to 16-bit PCM. However, I edit audio using the following setings:

    This isn't quite like that. It is all based on statistics, and to do with what happens when you hit the lower part of the dynamic range of a file. This only ever happen with the 16-bit files or less, because they are the ones when it could potentially occur within the range of human hearing. That dither is to obscure the point at which a signal goes from a determined point (about - 96dB) to the digital 'silence' at least the infinite dB. This, in terms of 16-bit is the next step down, and it is quite a leap. By dithering the signal (basically adding noise to it in a very controlled manner) you can not only hide it, but to increase the bit depth apparent through the noise. The full explanation for this is quite complex, and you can do all kinds of things with it - mainly involving the formatting of the noise profile.

    This makes a difference? Mainly of reverb tails, it must be said. Do you need it for a 24 - bit signal? No you haven't - it can solve up to lower way to the human ear can. Do you need it when you compress files using an encoder? You should not - the encoder will take care of it, and if it isn't, you probably shouldn't use it!

  • Make custom arrowhead. Cannot use spacing customized to an anchored object that is "path"?

    Hello world

    I build a model to use for the preparation of the documents for the brand of the company. I'm trying to find an easy way to make custom arrows that we will use for the legends (see below).

    Screen Shot 2015-05-26 at 10.09.54 AM.png

    What I want to do is to find a quick way to create this arrow to use for captions in my document. The main elements are the tip of the arrow 2-tone and the circle for a tail. I know that Indesign has a tail of the arrow circle but the problem with its use, it is that the color should be the same as the race and also the size of the circle is determined by the size of the race. As far as I know, Indesign does allow you to change those.

    I tried to use this type of path:

    http://InDesignSecrets.com/making-custom-arrawheads.php

    It works very well, but you can still see the path because the path is flush with the edge of the anchored object and the end of the path is visible behind the anchored objects. I think to be able to define these objects anchored with custom positioning would solve this problem. The problem is that I don't think that Indesign lets you define objects anchored to the positioning personalized text on a path, because this option is grayed out in the anchored object options.

    I was wondering if there is another way to create an offset for anchored objects or if there is a better way to make these custom arrow heads.

    Could there be a way to use objects from a library to create custom arrowheads? Also, is there a way to do this while a style object so that arrows can to shoot quickly and then simply apply the style of the object to get the correct formatting?

    I know that this is not possible and may have to be done manually, but I'll try to find a quick and easy way to make these arrows so that other people who have to use this model won't screw up their.

    Any help/ideas would be appreciated! Thank you all!

    Jacob F.

    Well, if anyone is interested, I sort of solved the problem.

    First, it was decided that the rounded tail of the arrow and the line should use the same darker green color. I still had solved the problem of adjusting the size of the circle but this trick of Eugene Tyson that:

    change the size of the arrowhead?

    Once I did, I made a style of object out of it, so that I can quickly draw lines, and at least add the tail quickly.

    As far as the head is concerned, I still have not found an automated way to do, but I have an easier way than what makes from scratch. I was directed in the right direction with the help of type on a path, but I was able to get the arrow flush with the end of the path and not past it. I saw a tutorial on Lynda.com, that shows you how make arrowhead custom and found a missing piece of the puzzle. It turns out that InDesign uses the same grips that you get when you add the type on a path in Illustrator. The trick to get the anchored object overlap the end of the path is to adjust the object anchored using these handles... After that, the axis of the anchored object y can be adjusted manually with the selection tool. Note that only the y axis (which is actually your baseline) can be modified because InDesign does not allow embedded objects that are typed to a path using a custom positioning. At this time, I don't bother centering the text on the path because the y-axis for my form must be adjusted in any case.

    So that's the solution I came up with that. If someone wants to share a solution better or more automated, I would really like to hear it.

    Thank you!

  • Can I mix up to 32 bits at a higher than 44.1 kHz sampling frequency?

    When I'm using Ableton Live, it allows me to choose 16, 24 or 32-bit, and then I can choose a up to 192000 sample rate.  Is this possible in the hearing?  I've been through all the preferences and all tabs and I can't find this option.  Everything I find is a convert, or the option adjust.  But this isn't what I want.  I want to mix down this way.

    The closest thing is when I go to "Export Audio Mix Down", I found, I can select 32-bit.  Then, there is a box for the sampling rate, with all the different values.  But it doesn't allow me to change of 44100.

    ?

    sleepwalk1000 wrote:

    Thanks for the responses guys.  Hmm, well the reason why I ask this question is because I am preparing a cd to be sent to a mastering studio.  The engineer told me to mix down to 24 bits instead of 16 bit.  I asked him why, because everything happens to 16 bit cd anyway.  He said even if my session has 16 bit files, if I have them running through the beaches with effects bus vst as Altiverb (which I do), it will improve the sound effects in mixdown, if treated according to a higher bitrate.  Would poll better mix so he could work in the mastering session.  I assumed that this meant that a higher sampling frequency would also be beneficial.  Maybe not?  I use a lot of high-quality vst effects, so I want to make sure I get the best possible results.  I mean if you have this option and space disk hard isn't a problem, why wouldn't you use it?

    It is interesting that the export Audio Mix Down allows me to mix a session from 16-bit to 32-bit files, but if doesn't let me change the sampling frequency.  Maybe I'm not understanding these terms exactly.  I always thought that, for two, more the number, the higher sound quality.

    SteveG - tell you it would be a waste of more than 32 bit 44.1 kHz mixing?  What about k of 48 or 96 k?  There is not a noticeable difference?

    The sample rate only affects the highest frequency that can be solved - it has no impact on the quality of all, once the rate is high enough to fix all that humans can hear. You determine the highest frequency that can be resolved in any case of the sample given by half to get the "Nyquist" frequency - so 44.1 k the highest frequency that can be fully resolved is 22.05 kHz. The human ear extends up to 20 kHz, but it's only in young children - by the time you reach your teenage years he began to fall, especially if you listen to a lot of loud music... then 44.1 k is already greater than any human condition to the response at high frequency. This overall been proved? You bet it has!

    To discover the truth about the hype about the sound quality in general and more bits/sample rate are concerned, you should first read this thread of AudioMasters - and the links it contains. It's of high quality academic research, and all attempts to discredit have been completely ransacked.

    Your mastering engineer is correct up to a point, but not really for the right reasons - mixtures of 32 bits are usually more specific, simply because the sums do more accurately and scale of signal is significantly improved. You must keep in mind that the only place where you will hear no appreciable difference with one more high bit depth would be in really quiet parts of reverb tails. The best way to do a mix in Audition is to do all this in 32-bit (which is floating point version of 24-bit anyway) and if he really wants a 24 bit file, you can convert the 32-bit mix after that you have to create a copy of 24-bit integer. As much as the 16-bit CD is concerned, if you (or engineer) procrastinate it properly when you do the master of the 16 bits of the final mix of 32-bit CD, then the effective resolution is higher than 16 - bit anyway - there is also a thread of AudioMasters explaining all this too (it's complicated).

    What all the above implies, is that if you keep your mix 32 bit 44.1 k files as they are, if you remaster at a higher sample rate, all you have to do is up-conversion files in hearing. You don't win a single thing in doing that, but then again, you won't lose anything either - and person don't will be able to make a difference! This is not a case of "why don't use you it?" - this is really a case of ' why would you? "

    Either way, it should be noted that due to some misinformation presented by people who should really have known better long, understanding of most of the people of sampling is completely and totally false. Once more, a search around AudioMasters will give you a better understanding of the present. If I get a chance later, I'll look on all appropriate threads.

  • Tecra S1 - can't get sound to work after a new installation

    Hello

    I have a Tecra S1 and the reformat. Did not have the original driver CD.

    All right, the drivers installed from the Web from Toshiba website. SoundMAX driver for sound. But I don't know that I hear some reverb when the music plays. VERY annoying.
    I think that a sound environment "Preset" must be turned on, but I've looked EVERYWHERE and can't find how to enter or adjust any environment sound presets.

    HELP PLEASE!

    Any help appreciated.

    Pete

    Good work, buddy! :)

    How did you solve this problem?

  • Inspiron 7000 point 8.1: how to mix the mic input with audio from YouTube (for DIY karaoke)

    Hello

    I want to use my Inspiron 7000 (wIn 8.1) as DIY karaoke machine.  Maybe there's karaoke apps or software, but they are usually expensive and the choice of songs is limited while on the mountain, there are clips from karaoke practically for each song and I want to mix the audio with a microphone on my laptop.  How I can do and it is possible at all?  Is there another solution?  Is it possible to add a reverb effect to the microphone entry?

    Thanks for the ideas.

    Hello. The recording of the sound Properties tab and right-click on your PC to open the context menu, and then click Properties.

    In properties, select the listen check "listen to this device". That will make your input mic through your speakers set. If you get your comments, and then click the levels tab and turn down the mic gain.

    You cannot add reverb on the side of the entrance. You can try to add it to the reading. Open the playback of the sound properties. Right click on the speaker to open the context menu and then properties. In properties, click the enhancements tab, and see what you have there. Below is a screenshot of an old computer with Realtek.

    As you can see he has an accessory called 'Environment' with fanciful names for the different quantities & types of reverb. I don't know whether or not your Realtek driver has a similar improvement. If so, I don't know if it is applied to the mic 'listen' function. If it works it will apply to audio from youtube too.

    If you can not reverberation through the configuration of the Realtek driver, then another solution would be to use a multitrack recording as the free Audacity program. You can save the audio from youtube on a track and your voice on another, and then use the effects of reverberation in the program on your voice.

    This version of Realtek in the screenshot has a function called "voice cancellation". Its purpose is to try to eliminate the voice of a record so that any record can then be a karaoke. I tried it and it's impressive. On a recording where the voice is at the Center, he gets completely eliminated. If the voices are mixed in a different way then it does not. If you want to access the feature, but your version of Realtek does not have it, download the program Audacity - it has an effect like this. You can open a record in the program, and then apply the effect of karaoke recording to eliminate the voice if possible.

  • SOUL - medium not in front of After Effects

    Hello

    I have a new iMac that has just been installed with CC software that I use, as LR; Photoshop and After Effects.

    My problem is that when I create a new After Effects project and then click Composition > Add to tail me, nothing happens outside of ME opening.

    I don't want to use the in AE render queue that I find MYSELF much more useful with the amount of work that I do.

    Can someone advise?

    Thank you!

    Edit: thought I'd try the AE render queue, just to see how fast is my new purchase and that does not work either!

    Hello

    I found the answer by chance last night. It was primer AE and I now hold down shift.

    The two versions are the most recent of the Adobe Web site from Saturday.

  • Audition Question Please

    When I import a file containing markers (using "Split at markers", recorded in twisted wave) hearing adds an additional marker for EACH file (labeled "marker 2").  A reflection as to why?

    Thank you very much!

    Stu Norfleet

    Where in the file is it puts 2 marker? It is possible that hearing is expected of the range markers and so requiring a marker beginning and end for each division.

  • "selected clip bounce."

    This new feature multitrack - Bounce the selected Clip - t - it the possibility to extend the scope of the new clip?  For example, if the clip is shorter than the entire session, the new clip to rebound is possible as long as the session?

    Question: what are some of the more obvious target uses this function?

    "Selected Clips" is one of several options "Bounce to new Track":

    Selected track: Bounce the entire contents of a piece to a single item in a new track.

    Selection time: Bounce contained all tracks in the selection of the duration of a clip in a new track.

    Selected clips in time selection: Make a selection of time, select or deselect clips and bounce back to a single item in a new track.  (This may be the option you are asking about.  The generated video will be the duration of your choice of time and generate silence or reverb tails, as appropriate.)

    Only selected Clips: Will bounce back content of the clips selected to an element in a new track.  Selected clips can come from any number of titles and positions on the timeline.

  • New on Premiere Pro CS5.5

    Hi all

    Recently moved upward to CS5.5 of PE8 and hoping for some advice before starting my main project.

    Project will be approximately 1 HR long, want to be able to burn any DVD Blu Ray as well as the finished product. Most of the footage was shot so far with / Handycam Sony HDR - CX, AVCHD located on FH, (should I use FX?). Some also have film shot with a standard DV camera and I would use both in project?

    I am able to create the project with 5.1 SS and always use footage with 2 stereo channels also?

    Played around looked suspicious, I have an incorrect setting, from the tutorials a bit edition with some success, until he comes to burn on DVD, seem to say don't bother the settings of the project at the beginning?

    My main issues so far are - should I set the necessary parameters at the beginning of my project? and if so what settings should I choose for desired finished products?  Also should what settings I choose when recording on HDD to burn it to DVD and then BR discs?

    Thanks in advance for any help.

    You can do a test of Red Giant Magic Bullet instant HD up-rezzing the SD material to HD. Some users really like it for their story, but others are the result not in their standards. Only a trial will tell you if it helps with your images.

    With a master 5.1 and the Minnetonka Audio over-codes DD 5.1 SS plugin, you can work with stereo equipment. There are many ways to use it, just leave the "Audio" Stadium collapse of 5.1 to stereo or using the Panel mixer Audio to place the apparent around the Audio scene source, or still to duplicate this stereo stream and may add Reverb and perhaps delay the dupe, then adjust the locations on the stage of (fake 5.1 Audio).

    Now, you do not have a 5.1 Master and also the SurCode plugin or program/CPU very similar to the output of DD 5.1 SS, as an AC3 track.

    Good luck and being new to PrPro, don't forget to spend some time with the First FAQ section. There are a ton of good advice, with links to the full articles, the file of help and tutorials. Great and useful to read here.

    Hunt

  • Record in the .mp3 file, but the file is displayed as .wav

    Been using AA 3.0.1 for a long time, but can't explain why this is happening. I usually record a song in wave format, no problem, but then I do a "copy as save", in order to make an MP3 of the song, but now my choice to save is this:

    mp3PRO (FhG) (*.wav, *.mp3).

    First of all, I never recall the *.wav never belong to with *.mp3 file extension. Second, once I recorded the song file appears as "mysong.wav", instead of "mysong.mp3", but looking at the size of the file, all about 5 k for a 256 Kbps that runs approximately 02:30, makes it look to be recorded as MP3's... a wav file size would be around 25000 kb.

    Thus, it seems that it is saved as an .mp3 file, but to show that it has been saved as a .wav file.

    How hell did this happen, and what can I do to make sure that the editor, I send an .mp3 file, actually becomes a .mp3 file and will be able to open it?

    Another crazy thing ' don't mess with me and use until some time I have...

    Harry

    Usually because an mp3 player has accidentally recorded with .wav left at the end of the name, a fairly common occurrence. Just make an mp3 save under and actually type .mp3 at the end of the name. This should fix hearing Add .mp3 at the end of mp3s automatically registered again.

    I suspect that your saved file is actually an mp3 player so just rename it correcting the suffix.

Maybe you are looking for

  • removal of clips video iOS iMovie

    I import video into iMovie on my iPad air 2 on USB. I edit the project, and then export it back. I delete the project, to make room for the next project, but none of the clips are removed. I must therefore delete iMovie on the iPad and then back - pr

  • need to disable the download in firefox, so I can use my download manager

    I want to record the download links in my download manager, but firefox downloader is the only option I get

  • Satellite A200 - some drivers Vista not installing

    Some drivers don't install software, why? Drivers are listed on Toshiba alerts. Thak you

  • NB200 BIOS upgraded to V1.9 - good or bad?

    My new NB200 - 10G has BIOS V1.2. Site Web de Toshiba EU shows V1.9 ready for download. Anyone know the advantages or disadvantages of the installation of this upgrade? There are lists of changes to each version anywhere? The upgrade process is relia

  • Re: Equium A60-199 stops intermittently

    Hi, hope you could help me with a problem iv been with my laptop recently, it seems to go off at different stages. prechecks- 1. take battery out and running AC adapter (problem)2 changed a work known and verified adapter adapter (problem) -(la premi