Forums - AudioLab

Allavaudioplayer buffer clear

Wed, 26 Dec 2018 12:25:59 -0500

I have a program which will play Mp3s,wma, and wavs. You give it the filename of the first track and it plays fine. When you click the Next Track button it plays about 3 notes of the previous track before playing the new track correctly. I have tried flushing the AlAudioout several times but it makes no difference. Is there a buffer attached to the Allavaudioplayer and if so how do I clear it?

How do you move a track bar while playing audio

Thu, 08 Dec 2016 18:18:49 -0500

When using Delphi7 and Audiolabs ALDSAudioplayer I could find the total number of samples in a file and using Currentsample (which then gave the number of the sample between Startsample to Endsample) I could calculate where the trackbar should be moved to in the Onprogress procedure. Using Berlin,and ALLAVAudioPlayer, the value of Currentsample soon goes above Endsample so I am not sure what it is measuring - certainly not the count of samples. Samplecount gives the same value as Endsample (why?) I can find no way of finding the position reached in the file so I cannot move my trackbar. Any help would be appreciated

SLResizeDispatcher and IPPPTypes.IppiRect error

Wed, 21 Sep 2016 18:36:41 -0400

Hi Dave,

I just installed V7.8.2.0 audio (and openwire) VCL packages for use into Delphi 10.1 Berlin Update 1. I Use your packages since several years and actually try to update an old program where Audiolab components gave full satisfaction.

With new version, I get an error message that say SLResizeDispatcher unit has been compiled with a different version of IPPPTypes.IppiRect.

Message (in French) :
[dcc32 Erreur fatale] SLBasicDispatcher.pas(1593): F2051 L'unité SLResizeDispatcher a été compilée avec une version différente de IPPPTypes.IppiRect

Does this problem can be due to not correctly removed old package ?

Edit : for information, problem also occur when trying compile an new project that contain only one Audiolab components (and without writing a single line of code).

Thanks in advance for your answer,
kindest regards,
Remy

ALAudioOut.Device.DeviceName problem

Tue, 20 Sep 2016 19:23:59 -0400

I am having problems routing the audio to a different sound device.

Here is the code line I am using:

SCInterface_Form.ALAudioOut.Device.DeviceName := 'HP w2408-8 (NVIDIA High Definition Audio)';

Using the debugger, the next line shows the devicename property in ALAudioOut has indeed been changed. But the audio is still routed to the default audio device.

If I set the w2408 as the default device then sound does go out that device, so the device is okay, seems ALAudioOut does not make the change.

Using Windows 10, Delphi 10 Seattle.

Craig

MIDI Output to VST Host PlugIn

Sun, 24 Jan 2016 22:31:20 -0500

I am coming from "somewhat successful use" of the ASIO VST 3.x project on SourceForge and would like to use AudioLab for my project but am having one heck of a time getting the PlugInDispatch to recognize and/or process MIDI provided data.

I can send data out to midi devices... I do have simple things like the signal generator and VST host hooked up to an AudioLab mixer followed by the ASIO Output. The signal generators produce output so I am certain the hardware communication layer is fine.

Over in the ASIO VST solution you push the VST Events via the ASIO HostBufferSwitch... basically calling ProcessEvents which iterates the events array and outputs the sound. You then detect if the MIDI thru is enabled and pass along the data to the MIDI device down the chain.

In Mitov I cannot seem to figure out the equivalent approach. I have my MIDI player pushing notes to both MIDI output and filling my events buffer, but when I forward the array to PlugInDispatch nothing all all occurs. No errors, no output, nothing.

I'm passing 25 (ProcessEvents) to the call along with my event array.. I'm just stumped.

iRes := ALVSTHost1.PlugInDispatch(25,0,0,@fMyEvents,0.0);

Has anyone done this and can you provide a dumbed-down example of just sending a note-on then note-off through a VST instrument so I can stop driving myself nuts?

Play music from Audio Buffer

Sat, 26 Sep 2015 18:27:52 -0400

Hello
How can I play music direct from buffer

DataBuffer := TALAudioBuffer.CreateSize( 44100, 16, 2, 1024000 );
DataBuffer.LoadFromFile('d:\jjj');
ALGenericFilter1.SendData( DataBuffer );
ALGenericFilter1.SendStartCommand(16,2,44100);

ALGenericFilter1 output pin connect alds audio out input pin

This is not work. why?

PLEASE help.

audio buffer can load just genericfilter or direct_?

Low latency audio capture and output

Wed, 11 Mar 2015 16:47:34 -0400

I need to capture microphone audio, mix a reverb signal, plus the audio from a playing video, and output the mixed signal in real time for a karaoke application.

I've done this fine in the past using DirectShow on WinXP, but Windows 7 and later no longer has the ability to feed the live mic input to the output via hardware.

Any capturing I do with DirectShow or WaveIn has an unacceptable delay, no matter how low I set the buffers. There is also no way to control the render buffers, as far as I can find, and I think this is where most of the delay comes from.

So it looks like I'll need to use Kernel streaming or ASIO. I have no idea how, or even if I can extract the audio from a playing video file to mix with the captured signal.

Is this something AudioLab could be used for? If so, is there any documentation that can help me get started?

Voice Changer with audio lab

Wed, 17 Dec 2014 02:41:43 -0500

Hi,

I'm evaluating Mitov SDK for a future proyect and I need to add voice changer capability to my app.

Something like speach be heard distorcioned..I need to hide the identity of the speaker by adding some voice distortion to the source.

It is possible, any hints?

Thanks in advance,

Omar Zelaya

Hi,

Actually looking for something like lowing the frequency of the input signal.

Thanks in advance,

Omar Zelaya

ALWaveLogger Stream

Sun, 02 Nov 2014 12:46:34 -0500

Hello how it's possible to save wav into a Stream ( Memory ) instead of saving to a File ? or even the ALRawLogger ?

i searched the forum but i really didn't find any suggestion .

yours,Randy

Audio Buffer . Edit and Cut wave

Fri, 29 Aug 2014 07:50:24 -0400

Hello. Pls. Help

How to load wav file in buffer and cut ranger x pos to y pos .

How to play buffer range.

How to plot this buffer in scope.

connecting a TALSpeexDecompressor output to a TALAudioMixer input at runtime

Wed, 20 Aug 2014 18:49:22 -0400

What do I have to do to connect a TALSpeexDecompressor output pin to a TALAudioMixer input pin in code?

I can successfully connect a TALWavePlayer output pin to a TALAudioMixer input pin in code.

playerWave.OutputPin.Connect(mixer.InputPins[0]);

Connecting a TALAudioMixer output pin to a TALAudioOut input pin works also.

mixer.OutputPin.Connect(audioOut.InputPin);

But I am trying the whole day now to connect a TALSpeexDecompressor output pin to a TALAudioMixer input pin in code. But there is no signal. Please note: There is a channel and an input in the mixer available for connecting. Connecting the same two components visually at design time works fine, so the audio format settings should be fine.

any chance for answers?

Mon, 04 Aug 2014 14:38:59 -0400

Hi,

does it make any sense to ask my question about AudioLab?
I found many questions but no answers in the last months, so I guess this forum (and perhaps the product) is dead more or less.

Before I post a longish text here, I would like to know if it will be to no avail, so I could save my efforts.

Setting compression (codec) bitrate

Tue, 11 Jun 2013 09:35:56 -0400

Hi there,

I'm trying to build a simple audio capture program in Delphi with a compressor ( in this case QDesign MPEG) to capture directly to mpeg2-audio, but I can't find how to set the codec's bitrate. I want it to be set to 48K, 256 kbps. Now it only captures in 256 kbps if I set it in another audio program.

I've connected: ALDSAudioin1 > ALDSAudioLogger1

In ALDSAudiologger1 I've selected the codec as compressor and capturing works fine. But the params options says [No Config]. The codec has no dialog.

In GrapheditPlus (filtergraph editor) I can set the output format through IAMStreamConfig::SetFormat

Is there a way to set or enumerate and select these codec properties in Audioloab?
(aldsaudiologger1.Compression.Compressions.Items[0].???).

Any help, solutions or samples to capture audio with a compressor and bitrate/samplerate settings are welcome!

Best regards,
Paul

how to use EnablePin

Tue, 29 Jan 2013 08:19:40 -0500

Kind time of day!
Prompt how to use, for example,
ALAmplifier1.EnablePin

Thanks.

AudioLab component for beat detection in song

Mon, 17 Dec 2012 04:06:52 -0500

hello,
i'm a student developing project related to music
and i need to detect bpm (beats per minute) in song
Are there example of AudioLab VCL to detect bpm ?
appreciate for any advise

thank you

Asio clock drift

Sat, 24 Nov 2012 19:41:35 -0500

Hi
I'm using 4 TALWavePlayers into a TALMixer. The players are synced by external
Clock. I'm outputting through. A TALAsio component. Everything sounds ok
However I slowly lose sync and players become slow to play. Any suggestions?

Bobo

Property Charset does not exist Error

Thu, 15 Nov 2012 06:29:34 -0500

After trying to compile my project got an exception:
Exception EReadError in module ALWaterfallDemo.exe at 00052E24.
Error reading SLWaterfall1.Title.Font.Charset: Property Charset does not exist.
Can anybody help me with that?
Thank U very much in advance..

Buffer

Thu, 11 Oct 2012 15:08:52 -0400

Hi,

how i load a file.mp3 to memory buffer using audiolab and play it when i press a button?
i need load to memory buffer because i cant have delay when press play button.

i will have 5 buttons and each button plays a different file.mp3

explaining better:
when the software loads, i load the five file.mp3 to memory buffer.
when i press button, the file.mp3 plays with no delay.

thank you.

Rafael

how to make a recoding and player machine complete

Mon, 10 Sep 2012 19:08:46 -0400

good day, Mitov Software

I am trying to develop a form where
can record an audio file when you stop and record again
that store all the audio and not record a new audio?

I also record and play back the file and continue recording in the same file

how I do this?

thanks
Audio program.zip (Size: 5 KB / Downloads: 4)

2 questions: AudioIn sample rate 44100, OMP affinity

Wed, 05 Sep 2012 12:35:59 -0400

Hello!
1) Delphi XE2 + AudioLab_VCL_5_0_2 have OMP affinity error:
OMP: Warning #72: KMP_AFFINITY: affinity only supported for Intel® processors.
OMP: Warning #71: KMP_AFFINITY: affinity not supported, using "disabled".

2) AudioIn.SampleRate = 44100 - worked only in Windows 7 with HD Audio 192000Hz.

Object in form:

object ALAudioIn1: TALAudioIn
Device.AlternativeDevices = <
item
end>
Device.DeviceName = 'Default'
AudioFormat.Channels = 1
AudioFormat.SampleRate = 44100
AudioFormat.BufferSize = 16384
OutputPin.Form = Form1
OutputPin.SinkPins = (
Form1.ALGenericFilter1.InputPin
Form1.ALWaveLogger1.InputPin)
Left = 176
end
object ALGenericFilter1: TALGenericFilter
Threading.Queue.FullMode = fmDropData
InputPin.Form = Form1
InputPin.SourcePin = Form1.ALAudioIn1.OutputPin
OutputPin.Form = Form1
OutputPin.SinkPins = (
Form1.ALAudioOut1.InputPin)
SynchronizeType = stSingleBuffer
OnProcessData = ALGenericFilter1ProcessData
Left = 248
end

if Audio In device has not bigger than 48000Hz sample rate - my program is not working! Install last RealTek HD Audio driver on notebook Lenovo - failed to capture audio.