I have hundreds of audio tracks and
I want to add Silence before all tracks.
I know it can be done through Chain, but I do not know how exactly it can be done.
PS: The following I have tried:
File> Edit Chains then File> Apply Chain> Apply to Files> selected required files, the output files are all silenced. I do not know coding, this is the first time I am using Audacity.
I am attaching the Edit Chains window.
Actually, Audacity provides more easy way to add silence for each of your files.
First import all the tracks and select all. Then follow Transport > Cursor to > Section Start
This path will put your cursor at the beginning (0th mili-second)
Then Generate > Silence path will ask for the duration. (My favorite is hh:mm:ss + miliseconds)
Sample screen for entering duration
After confirmation all of your sounds will be prepared like here.
Audacity's chain logic doesn't allow to set duration for silence command, it takes the length of longest track and produce all-silence sounds. This task will be more easy rather than applying chain.
Related
Im trying to make a game on Scratch that will use a feature to generate a special code, and when that code is input into a certain area it will load the stats that were there when the code was generated. I've run into a problem however, I don't know how to make it and I couldn't find a clear cut answer for how to make it.
I would prefer that the solution be:
Able to save information for as long as needed (from 1 second to however long until it's input again.)
Doesn't take too many blocks to make, so that the project won't take forever to load it.
Of course i'm willing to take any solution in order to get my game up and running, those are just preferences.
You can put all of the programs in a custom block with "Run without screen refresh" on so that the program runs instantly.
If you save the stats using variables, you could combine those variable values into one string divided by /s. i.e. join([highscore]) (join("/") (join([kills]) (/))
NOTE: Don't add any "/" in your stats, you can probably guess why.
Now "bear" (pun) with me, this is going to take a while to read
Then you need the variables:
[read] for reading the inputted code
[input] for storing the numbers
Then you could make another function that reads the code like so: letter ([read]) of (code) and stores that information to the [input] variable like this: set [input] to (letter ([read]) of (code)). Then change [read] by (1) so the function can read the next character of the code. Once it letter ([read]) of (code) equals "/", this tells the program to set [*stat variable*] to (input) (in our example, this would be [highscore] since it was the first variable we saved) and set [input] to (0), and repeat again until all of the stats variables are filled (In this case, it repeats 2 times because we saved two variables: [highscore] and [kills]).
This is the least amount of code that it takes. Jumbling it up takes more code. I will later edit this answer with a screenshot showcasing whatever I just said before, hopefully clearing up the mess of words above.
The technique you mentioned is used in many scratch games but there is two option for you when making the save/load system. You can either do it the simpler way which makes the code SUPER long(not joking). The other way is most scratchers use, encoding the data into a string as short as possible so it's easy to transfer.
If you want to do the second way, you can have a look at griffpatch's video on the mario platformer remake where he used a encode system to save levels.https://www.youtube.com/watch?v=IRtlrBnX-dY The tips is to encode your data (maybe score/items name/progress) into numbers and letters for example converting repeated letters to a shorter string which the game can still decode and read without errors
If you are worried it took too long to load, I am pretty sure it won't be a problem unless you really save a big load of data. The common compress method used by everyone works pretty well. If you want more data stored you may have to think of some other method. There is not an actual way to do that as different data have different unique methods for things working the best. Good luck.
I'd like create a class/struct/other that contains each measure of a song, complete with independent tempo and beat count, then play the entire song back (with potential updates from user input). I'm only aware of how to change those variables on an AKSequencer track as a whole; is there a way to store that data independently and then have it play back as one? And keep coherence between the measures so as not to "jump" between them? Thanks!
AKSequencer is not good at setting loop length on the fly, but it is totally fine for adding to or re-writing the contents of a track while the sequencer is running. This includes tempo events.
Why don't you set the length to something arbitrarily long, and string together your MIDI events measure after measure without ever looping? Keep track of how many beats have been written so far, and just keep adding after that point. Doing this while the sequencer is running should be no problem. You could even automate writing the next bar by triggering a callback function near the end of each measure getting it to write the next segment (which could be selected or 'cued up' at run-time). You can schedule the tempo events with addTempoEventAt(), with the starting point for the next segment.
When your user stops the sequence, clear the track(s), reset the tempo, rewind the sequence and start over.
I Have a movie sample with audio transcription (For Blind People- There is a narrator explaining what is going on in the movie). I want to extract that.
What i so far tried was:
1- I have the sample without the transcription as well so i just imported both samples in Audacity. Inverted one and mixed. But it simply doesnt work ( Normalization is also applied)
2- I tooke the sample with audio description. splitted to mono, Took one channel and inverted. and mixed again. Now i have the movie without audio transcript. My intuition was that if i invert this result file again and mix it with the Actual one the other sounds should cross out and i would have the Narrator sound. But it did not happen! what shall i do now ? any suggestions ?
I have checked the following links so far :
http://www.howtogeek.com/61250/how-to-isolate-and-save-vocals-from-music-tracks-using-audacity/
http://www.labnol.org/software/tutorials/remove-vocals-song-mp3-music-instruments/1301/
I'm trying to develop an iPhone app that will use the camera to record only the last few minutes/seconds.
For example, you record some movie for 5 minutes click "save", and only the last 30s will be saved. I don't want to actually record five minutes and then chop last 30s (this wont work for me). This idea is called "Loop recording".
This results in an endless video recording, but you remember only last part.
Precorder app do what I want to do. (I want use this feature in other context)
I think this should be easily simulated with a Circular buffer.
I started a project with AVFoundation. It would be awesome if I could somehow redirect video data to a circular buffer (which I will implement). I found information only on how to write it to a file.
I know I can chop video into intervals and save them, but saving it and restarting camera to record another part will take time and it is possible to lose some important moments in the movie.
Any clues how to redirect data from camera would be appreciated.
Important! As of iOS 8 you can use VTCompressionSession and have direct access to the NAL units instead of having to dig through the container.
Well luckily you can do this and I'll tell you how, but you're going to have to get your hands dirty with either the MP4 or MOV container. A helpful resource for this (though, more MOV-specific) is Apple's Quicktime File Format Introduction manual
http://developer.apple.com/library/mac/#documentation/QuickTime/QTFF/QTFFPreface/qtffPreface.html#//apple_ref/doc/uid/TP40000939-CH202-TPXREF101
First thing's first, you're not going to be able to start your saved movie from an arbitrary point 30 seconds before the end of the recording, you'll have to use some I-Frame at approximately 30 seconds. Depending on what your Keyframe Interval is, it may be several seconds before or after that 30 second mark. You could use all I-frames and start from an arbitrary point, but then you'll probably want to re-encode the video afterward because it will be quite large.
SO knowing that, let's move on.
First step is when you set up your AVAssetWriter, you will want to set its AVAssetWriterInput's expectsMediaDataInRealTime property to YES.
In the captureOutput callback you'll be able to do an fread from the file you are writing to. The first fread will get you a little bit of MP4/MOV (whatever format you're using) header (i.e. 'ftyp' atom, 'wide' atom, and the beginning of the 'mdat' atom). You want what's inside the 'mdat' section. So the offset you'll start saving data from will be 36 or so.
Each read will get you 0 or more AVC NAL Units. You can find a listing of NAL unit types from ISO/IEC 14496-10 Table 7-1. They will be in a slightly different format than specified in Annex B, but it's fine. Additionally, there will only be IDR slices and non-IDR slices in the MP4/MOV file. IDR will be the I-Frame you're looking to hang onto.
The NAL unit format in the MP4/MOV container is as follows:
4 bytes - Size
[Size] bytes - NALU Data
data[0] & 0x1F - NALU Type
So now you have the data you're looking for. When you go to save this file, you'll have to update the MPV/MOV container with the correct length, sample count, you'll have to update the 'stsz' atom with the correct sizes for each sample and things like updating the media headers and track headers with the correct duration of the movie and so on. What I would probably recommend doing is creating a sample container on first run that you can more or less just overwrite/augment with the appropriate data for that particular movie. You'll want to do this because the encoders on the various iDevices don't all have the same settings and the 'avcC' atom contains encoder information.
You don't really need to know much about the AVC stream in this case, so you'll probably want to concentrate your experimenting around updating the container format you choose correctly. Good luck.
I need to append MIDI files: leave header (same for all files) and other meta information, just copy music/score part.
I already have MIDI files in appropriate bytearrays, as I guessed I need to use writeBytes, but unfortunately couldn't find which bytes I need to take and copy.
Something like this:
var newFileBytes:ByteArray=new ByteArray();
newFileBytes.writeBytes(firstMIDIBytes);
newFileBytes.writeBytes(secondMIDIBytes,8);
Works only partially, file is playable; first part fully and second - only some notes (then player hangs out)
To say truth byteArrays aren't my strong side, as the MIDI file structure.
Can you suggest how to solve this?
Thanks in advance.
As per my comment, you probably mean to append these files, not merge them. Assuming that to be the case, you can't simply slap the data from the second file to the end of the first. As the MIDI protocol is bandwidth-optimized, it makes a number of assumptions regarding the streaming of events. These behaviors mean that you must take special care when appending MIDI data.
MIDI files can (and usually) use running status, which means that an even may omit the status byte, in which case the event should use the status byte of the previous event. This may not be the cause of your problems, but are you absolutely sure that you are only parsing raw MIDI data, and not the file headers and such? If this were the case, all sorts of weird data would be erroneously interpreted as valid MIDI events.
Events in MIDI files use relative offsets to the previous event in the sequence. The way that this is calculated is a bit complicated, but it involves a few properties (such as tempo, number of pulses/sec, etc) which are defined in the MIDI file header. If you stripped these events, and the properties are different for the second file, then the timing of these events will be wrong.
Basically, the only safe way to append the two MIDI files is to play them through a sequencer and re-write them to a new stream. Appending the byte arrays will probably be the cause of many mysterious bugs.
The structure of a MIDI file doesn't allow you to just "append" more data to it, for the following reasons:
Each track ends with an End of Track event, rendering all notes after that event meaningless.
Each track header chunk defines the size of the data that follows. Even if you append new data, any reader will only read [size] bytes before it starts looking for a new track.
A MIDI file defines how many tracks are present in the file, so even if you appended the byte array of a single MIDI track, unless you also update the track count of the header data, any reader would simply ignore the track you added.
If you add data to a MIDI file, you need to make sure the structural integrity of the file format is maintained. Simply appending data does not accomplish this.