Converting audio samples to EXS24 instrument

I have an audio track that contains a sequence of samples from a live instrument. The samples run from C0 to C7 in increasing order.
What's the easiest way to turn this into an EXS24 instrument?
Thanks
Eric

First slice the track using the Strip Silence function (in L9 you could also use Flex mode to splice the track), then convert all the resulting regions into new Audio files using the copy/convert function. Then drag all those files into the EXS (hit the edit button first) using the contiguous mapping feature, set the lowest note to C0 or wherever you want the mapping to start.

Similar Messages

Converting audio samples using libav*

Hello guys,
I've been looking for a solution for several days by now and have not gotten any smarter, I hope that someone of you is able to help me
What I'm currently working on is an application able to generate audio fingerprints (specifically acoustID fingerprints). To do so, I started with the sample
code that demonstrates the basic fingerprinting capability. You can find it here.
It is part of a library called chromaprint.
As you will see, a lot of this code uses deprecated functions. As a newbie to sound conversion, the sense of a lot of the calculations in the file is oblivious to me
Still, I want to renew the code to use functions that are up-to-date. Ideally I'd understand the code more thorough then, or so I thought.
I started modifiying the decode_audio_file function from fpcalc.c and got until here, where I am now stuck:
int decode_audio_file(ChromaprintContext *chromaprint_ctx, int16_t *buffer1, int16_t *buffer2, const char *file_name, int max_length, int *duration)
AVFormatContext *format_ctx = NULL;
AVCodecContext *codec_ctx = NULL;
AVAudioConvert *convert_ctx = NULL;
AVStream *stream = NULL;
AVCodec *codec = NULL;
AVPacket avpacket;
AVFrame *decoded_frame = NULL;
FILE *f;
int frameFinished = 0;
int stream_id, ok = 0;
int buffersize = AVCODEC_MAX_AUDIO_FRAME_SIZE + FF_INPUT_BUFFER_PADDING_SIZE;
uint8_t inbuf[buffersize];
/* initialize data packet that is read from the stream */
av_init_packet(&avpacket);
avpacket.data = inbuf;
avpacket.size = buffersize;
/* make space for frame that contains decoded data */
decoded_frame = avcodec_alloc_frame();
/* tell format_ctx about the input */
if (avformat_open_input(&format_ctx, file_name, NULL, NULL) < 0) {
fprintf(stderr,"ERROR: couldn't open the file");
goto done;
if (avformat_find_stream_info(format_ctx, 0) < 0) {
fprintf(stderr,"ERROR: couldn't find stream information in the file");
goto done;
for (int i = 0; i < format_ctx->nb_streams; ++i) {
codec_ctx = format_ctx->streams[i]->codec;
if (codec_ctx && codec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
stream = format_ctx->streams[i];
break;
if (!stream) {
fprintf(stderr,"ERROR: couldn't find any audio stream in the file\n");
goto done;
codec = avcodec_find_decoder(codec_ctx->codec_id);
/* chromaprint expects signed 16 bit samples */
codec_ctx->request_sample_fmt = AV_SAMPLE_FMT_S16;
if (codec_ctx->sample_fmt != AV_SAMPLE_FMT_S16) {
convert_ctx = av_audio_convert_alloc(AV_SAMPLE_FMT_S16, codec_ctx->channels,
codec_ctx->sample_fmt, codec_ctx->channels, NULL, 0);
if (!convert_ctx) {
fprintf(stderr, "ERROR: couldn't create sample format converter");
goto done;
if (!codec) {
fprintf(stderr,"ERROR: unknown codec");
goto done;
if (avcodec_open2(codec_ctx, codec, NULL) < 0) {
fprintf(stderr,"Could not open codec\n");
goto done;
chromaprint_start(chromaprint_ctx, codec_ctx->sample_rate, codec_ctx->channels);
*duration = stream->time_base.num * stream->duration / stream->time_base.den;
int len;
while (av_read_frame(format_ctx, &avpacket)>=0) {
if (avpacket.stream_index == stream->id) {
len = avcodec_decode_audio4(codec_ctx, decoded_frame, &frameFinished, &avpacket);
if(frameFinished) {
int data_size = av_samples_get_buffer_size(NULL, codec_ctx->channels,
decoded_frame->nb_samples,
codec_ctx->sample_fmt, 1);
if (convert_ctx) {
const void *ibuf[6] = { decoded_frame->data };
void *obuf[6] = { buffer2 };
int istride[6] = { av_get_bytes_per_sample(codec_ctx->sample_fmt) };
int ostride[6] = { 2 };
len = data_size / istride[0];
if (av_audio_convert(convert_ctx, obuf, ostride, ibuf, istride, 4) < 0) {
fprintf(stderr,"WARNING: unable to convert %d samples\n", len);
break;
if (!chromaprint_feed(chromaprint_ctx, buffer2, decoded_frame->nb_samples/2 )) {
fprintf(stderr,"ERROR: fingerprint calculation failed\n");
goto done;
} else if (!chromaprint_feed(chromaprint_ctx, decoded_frame->extended_data, decoded_frame->nb_samples)) {
fprintf(stderr,"ERROR: fingerprint calculation failed\n");
goto done;
ok = 1;
done:
avformat_close_input(&format_ctx);
avcodec_free_frame(&decoded_frame);
return ok;
You can find the documentation for the used chromaprint functions here.
As I said, I am not really sure whether I have understood all of what is going on in the unmodified version of the function, so please bear with me
Any answers or suggestions for reading material will be gladly appreciated!
PS: goto marks will be removed once the code works
Last edited by n0stradamus (2013-03-29 23:10:36)

First slice the track using the Strip Silence function (in L9 you could also use Flex mode to splice the track), then convert all the resulting regions into new Audio files using the copy/convert function. Then drag all those files into the EXS (hit the edit button first) using the contiguous mapping feature, set the lowest note to C0 or wherever you want the mapping to start.

Converting Analog Sampler to Virtual Instrument

Hello there,
How do I go about sampling presets from an analog instrument, such as a Esoniq EPS 16, into Logic Express so that I can play the presets from, say, the EXSP24? If I do so, will if have the same awesome analog quality as the original synth/sampler? I could really use the help, as I'd like to be able to quantize my performances with some old-school sounding synths.
Thanks!
Sullie Pepperstein.
Powermac G5 Mac OS X (10.4.5)

I did a little research and found that sampling,
aside from third-party plugins, is a Logic Pro
feature only.
You don't need Pro in order to use Translator. It will create all the instruments that your EXSP will read.
You can find the CDs on eBay, if you're lucky. Or search the web.
It's really worth it; some sounds are useless, but drumkits for example are FANTASTIC, I've used them a lot in my productions, and many other producers have sampled that loops from vinyl! They've also ended in sample cds!
EPS16 and ASR10 sounds have a special flavor, because they mostly come from 12bit samples (Mirage). Buy all the CD collections that you will find (the Ensoniq ones at least), then fire up Translator, you'll be amazed.
my 2 eurocents

Importing audio - sample rate/bit depth

Hi forum,
I am working on project at 44.1K, 24 bit. Audio elements are being sent to me to be added. Some have come in incorrectly; at 48K, 16 bit. I can convert easily but didn't think I needed to.
I thought by selecting the "Convert Audio Sample Rate When Importing" option when creating the project that would all be worked out.
That is what I've done - and the file seems to be the correct pitch -- yet shows up in the audio window with it's original specs (48/16). Also ... will logic keep it at 16 bit and play all other files at 24 bit?
I want to be sure about this ... something seems fishy.
Cheers
Dee, Ottawa

Hi,
I am re posting.
Regarding the same project: must everything in the arrange window of a project be the same sample rate (and bit depth). My understanding is that there is real- time conversion during playback. That all file types supported by logic - and all virtual instrument samples are converted in real time to conform to the selected bit depth and sample rate of project.
I ask only as I recieved reference sound files to temporarily place in a mix to see how mix will sit when going to Post. Two audio files are almost a semi tone higher than they are supposed to be (which is odd - so I am pretty sure it was just quickly sung in the wrong key at there end). And one file which was supposed to be timed out is not lining up.
I can work around this on this project. And I can simply convert in the sample editor and re import to compare.
But again ... I just want to check my understanding for future reference. The manual indicates differing rates etc should not be a problem. (ie: That Logic allows one to have differing rates and bit depths. Inversely, the M Sitter video implies just the opposite.
I just want to be sure of this for future reference. Any adive?
Thanks in advance.
Cheers
Dee

EXS24 instrument created from audio region- How to move files?

In Logic Pro X. I have created an EXS24 instrument from an audio region with 3 well-separated sounds. The instrument in EXS24 editor shows a zone with 3 files. Each file is actually the same physical file, but, when examining the three separate "files" (what should I call them??), a different section is highlighted for each tone. I can set a note for each of these "files". This instrument plays perfectly.
So now I have 2 files in my project folder - a .exs one for the instrument, and ONE SINGLE .wav file somehow representing three different samples. I want to move them to ~Library/Application Support/Logic/folderName, where folderName is , say, myInstruments (for .exs) and mySamples (for .wav). When I do this, and erase the files in the project folder, I can load the instrument and the sample file into EXS24, but the sample file is recognized as one single tone - that is, in the EXS24 editor, there is only one file rather than 3, and that file has all three audio sounds.
How do I move these files correctly? Is there some way to get them created originally in the correct place?

Hi keystrike,
This article will give you information about how to move files in Logic -
Logic Pro X: Move, copy, and convert audio files
Thanks for using Apple Support Communities.
Best,
Brett L

Logic Pro X - Is there a way to convert a sample kick from audio to midi ?

Is there a way to convert a sample that you drag into logic pro for example im using a sample kick for a dance track, and want to use it as MIDI because it a **** of a lot easier to edit than a audio file, is this possible?

Not exactly.
This site explains MIDI in a straightforward and eay way:
http://morganstudios.com/Tutorials/MIDI.htm
But for a "solution" of sorts, you can use the EXS24 to sample the kick.
Video of how to add the EXS24 on a track in your project (Logic X):
http://www.youtube.com/watch?v=3vbjlZZs_24
How to add your own sample to the EXS24 (Don't worry that it's Logic 9):
http://www.youtube.com/watch?feature=player_detailpage&v=tB4_2IYBess#t=53
Hopefully this helps you out!

How do you save sample regions in the EXS24 instrument editor?

Hi There, I don't if my memory has failed me or something was changed in Logic pro 9 but I can't get a EXS24 instrument to save the sample region sections I set?? I open EXS24 click on edit, click on load zone, load a sample, click on edit in sample editor, set the region and then the new region doesn't set inside the EXS24 sample editor window? Tried saving the instrument, but it doesn't up date the selection? It just keeps the entire waveform as the sample? What am I doing wrong here? In the past this was really simple, you just moved the region on the sample area you want and the sample begin and end setting automatically update inside the EXS24 instrument editor window and then you just saved the instrument and each piece of the audio file as and how you set it was then on your keyboard ready to be used. Can anyone help me out here?
Thanks

Got it to work, seems I had to resave the audio file again... weird glitch...

Multi-sampled ProSessions Premium Instruments for EXS24

Hey,
did any of you are using Multi-Sampled which directly Load from EXS24.
what i know is some of the sampling must/neccesary Load from their own sampler...
i wish to get something more easier stuff which can directly load from EXS24 sampler since we r using Logic Pro.
Below is the sound that i found online,is it suit my needs?please offer me your suggestion and idea...i might need your helps buddy!!
http://www.m-audio.com/index.php?do=products.list&ID=Prosessions+Premium
thanks ya!

On the page that you point to, it says:
Each volume includes EXS24, HALion, Kontakt, MachFive
and Reason's NN-XT formats.
And that's what you shold be looking for. You are
right that some have their own interfaces but there
may sometimes be advantages in using those.
Hi Pete,
Thanks for the reply.
according to your opinion above,that mean the ProSessions Premium Instruments by M-Audio (for example Drum Kit) is posible to Load by EXS24 and apply to my programming with Velocity on each Shot,am i right??
2) is that mean,every sampling which do mention support EXS24 or Multi format (included EXS24) or multi-sampled with EXS24 are able to Load by EXS24 without using 3rd party sampler (unless we need the facility of the 3rd party sampler),am i right?
Thanks

Q: Convert audio bit & sample rate during import

I have a project that's in 24bit/48KHz, and want to import an audio file (mp3) that's in 16bit/44.1KHz.
When I import the mp3 file into the project either through drag & drop or the Add Audio File dialog box, it doesn't convert the file into the project's native bit & sample rate. Consequently, when I play it back in the Arrange window it plays in chipmunk voice.
I thought Logic used to automatically convert audio files into native resolution. How do I do this in Logic 8?
Thanks,
Steven

Thanks for the response. My project currently does have those options checked, and it's still not converting.
To double check, the mp3 file does get copied to the project's Audio Files directory. It also creates a file with the same name and the file extension .ovw - but doesn't seem to be converted.
Any ideas? Thanks again -

EXS24 instrument or samples of telephone ringtones?

Anyone have an EXS24 instrument or samples of telephone ringtones?
Possibly created by combining the actual multi-tones used in each telephone key?
Thanks.

try ES1's factory preset>Effects>Crazy toy phone if it helps

Question for EXS24 instrument editor experts!

Can anybody help with this?
here's what I'm trying to do:
I'm trying to combine a solo violin instrument and a solo cello instrument into one EXS24 instrument that has the violin in the higher octaves and the cello in the lower octaves, not overlapping (to use with my EWI).
These are VSL (Vienna Symphonic) instruments, a different sample every half step, 3 velocity layers (different group for each velocity layer). So each instrument has around 150 audio samples.
Here's what i tried so far:
I opened the violin instrument, I added a new group and named it "solo cello pp". I drag dropped all the cello pp samples into the new group. I chose "auto map" in the dialog box.
OK, when I look in the new group it got all the zone key notes right (zone key note matches the audio sample name)
But... every zone has a key range of "Lo C-2, hi G8" so when I play the patch I'm hearing all the samples playing at once (because every zone has the full key range)
OK ... I can just go type in the correct key range for each zone (like for the A#3 sample: "Lo A#3, hi A#3" etc.) but with 150 samples per inst., that's typing in a field 300 times! Kind of insane! and I plan to build several instruments like this (trombone/trumpet, bassoon/oboe etc.)
Is there a smarter way to do this?
I suppose I could probably just keep the violin & cello instruments separate, constrict their ranges so they don't overlap, have them on separate tracks, record enable both tracks, and play it that way. But it would just be more idiot proof to have it all in one EXS24 instrument.
Any help would be appreciated.
Tripp

EXS is not the easiest to work with.........
but , do this:
bring two EXS's up on two different channels.
load the vlns in one, the cello in the other.
open the cello edit pane. switch to "group" and select all groups.
open the edit menu and click on "copy".
next, open the vln editor, then click on edit and select "paste".
this should copy the cello in the vln pane.
if they overlap, you'll have to move them an octave and probably limit the range.
save as....... a new patch.
DR9.

Some EXS24 instruments loads slow

Hi,
Why is it that some of my EXS24 instruments load very slow. Ironicly the worst ones is some which contains only a few very short samples (drums). What can I do to speed up the process?
When I load a song containing the instruments there's no problem. It's when I open the instruments in a new song.

Basic 3 min song loading takes 20 sec and 37 ms (
this is:
EXS24 = 8
TRILOGY =1
SONIK SYNTH =1
COM,EQ,DELAY,GATE,SPACE DESIGN,AD-LIMITER ect =25
Full song 7 min song loading takes 1min 3sec and 53ms
this is:
40 audio tracks
5 Stylus RMX all load with full presets
3 EXS24
2 Hypersonic each loading fully
60 COM,EQ,DELAY,GATE,SPACE DESIGN,ADLIMITER ect
now this will be different from all systems
I know for sure my G5 dual 1.8 with 1gb of rams will take longer to do this
and if add more rams or get a better processor than my G5 dual 2.5 with 3gb of rams will improve the loading time as well
but just having this setup and running the project manager have solved alot of problems when it comes to loading time
without the pro. man. it'll have taking more than 1
not to mention that this is not only EXS24, all the plug-ins are included in the song loading time
especially Hypersonic which takes like 20 sec just to load plus RMX which also takes like 10-15 sec to load as well
so how about you do a test
create 15 instrument channels with just EXS24 then load samples into them, save the project, close logic down then launch again then time it
just incase you don't know, the project manager only works for
EXS24
Movies
Audio samples
Song files
and logic's plug-ins settings
it does not include 3rd party plug-ins such as SAMPLETANK, STYLUS RMX,HYPERSONIC BFD and all them big dat. plug-ins
so try the test with just EXS24 and see
stash

Please Help. How do I assign a short audio sample to a specific note?

Hi. Happy new year to all!!
If any one can please please offer advise it would help to get my new year off to a much better start.
I am wanting to assign/map individual audio samples to individual 'note' or midi events, same kind of set up as how say any of the drum sets work in ultrabeat.
Is it something to do with ch assignment?
The result I'm looking for is if I hit C0 on my keyboard, I get one sample. If I hit G4 on my keyboard, I get a different sample and so on.
I'm working with a novation 49 sl compact.
I have searched extensively in the manuals and the discussions and suspect the 'way' I'm thinking about this might be wrong. I'm sure it must be possible.
Is it possible? How?
Any help/thoughts etc greatly appreciated in advance.

Hi Thanks for that.
Since my posting, I discovered another option, by using the Sampler exs24 and creating a new key map, this seems to be exactly what I was after. Like all things, its straight forward when you know how.
I looked into the ultra beat sampling a bit further as I understood it, before seeing your reply, to discover that the sampler window was about chopping/or linking a sample to a preexisting rhythmic sequence. This was not quite what I was after but I'm interested to look further in to your suggestion which seems a bit of a different thing again.
Where theres a will theres a way and I googled my question leading me to the macprovideo discussion group with the following link:
http://www.macprovideo.com/forum/logic/logic-pro-express&id=5356
so this was enough to send me in the right direction.
Not being inundated with answers form the forums, your responce is very appreciated.
Thankyou.

Mapping Samples on EXS24

Hi.
I've just moved from Gigastudio/cubase/pc to Logic/Mac and I'm having trouble using new ESX24 samples I'm installing onto the ESX24 sampler.
The notes are not mapped out properly across the keyboard - it only seems to use/see two of the samples within an instrument, and then maps thos across the keyboard with no changes in pitch. I'm installing them according to the instructions. Do I have a bug? Am I overlooking something? Any help much appreciated.
Cheers.

I'm using the conversion with .gig samples. But it's also doing the same thing with EXS24 samples that I install myself (the ones supplied with logic are mapped fine). I'm starting to think there's a bug in my installation somewhere.

Sending audio samples to midi controller

i have logic pro 8 and i was wondering how to send chopped up samples of an audio file to my midi controller which is an axiom 49. for example, i wanna chop up some vocals and instrument loops from a song called dazed and confused to create a hip hop song and i ve seen it done before on youtube just wondering how to do it.

Not sure what you mean by sending audio samples to your MIDI controller. Maybe you'd like to play or trigger these samples from your controller? One way is to cut them up (lots of ways to do this in Logic) and then assign the resultant bits of audio as samples in the EXS sample player. You can also assign these samples in Ultrabeat.
You'll need a basic familiarity with editing in Logic, as well as some fundamental knowledge of the EXS sampler or Ultrabeat.

Converting audio samples to EXS24 instrument

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