Hello everybody! Here is a kind of silly question, you know. I write a radio station program that reacts on some sound events...

Hello everybody! Here is a kind of silly question, you know. I write a radio station program that reacts on some sound events detected within a continious audio stream received through a sound card input channel or whatever. A program runs and at the specified moment starts to listen to the input audio stream and tries to detect a sound pattern that it's been trained before (a PCM file or so...

I'm trying to plot the amplitude envelope of an audio signal, with perfect accuracy. (the algorithm for perfect accuracy would be to take...

I'm trying to plot the amplitude envelope of an audio signal, with perfect accuracy. (the algorithm for perfect accuracy would be to take the absolute value of the audio signal, then plot a point at the location of each local maxima, then connect adjacent points with a line) In Reaktor I've tried to use the envelope follower macro, but even after tweaking the settings it doesn't seem to plot th...

Hello, I want to change the sample rate of an audio signal. Where can I find some information about it? Some mathematical model and some...

Hello, I want to change the sample rate of an audio signal. Where can I find some information about it? Some mathematical model and some source code would be perfect. What I want to do is: I have a device that can play audio signal with the sampling rate of 8KSPS. If I want to play a signal that is samples at say 44KSPS I need to change the sampling rate. How can I d...

Hello, i'm new to this group I'm trying to demix two audio signal. My audio signals are y1 and y2 (Y=[y1;y2]) The impulse resonses: H11, H12,...

Hello, i'm new to this group I'm trying to demix two audio signal. My audio signals are y1 and y2 (Y=[y1;y2]) The impulse resonses: H11, H12, H21, H22. The matrix form of the system is: y=H*x I'm searching for x: x=(H^-1)*y The problem is: det(H)=0, also invertible! i'll be grateful for any idea.

Hi, I am going to do a research project which needs to cancel feedback data signal of an amplifier. The data signal frequency is 10MHz which...

Hi, I am going to do a research project which needs to cancel feedback data signal of an amplifier. The data signal frequency is 10MHz which is not audio type. I have searched the internet that most of the feedback algorithm are for audio signal, which allows a little distortion in the output signal. However, as the input data type is not audio, the output data must not be distorted in any...

Hi: Clipping in an audio signal results when an audio device receives a signal that is too loud. The audio signal distorts into...

Hi: Clipping in an audio signal results when an audio device receives a signal that is too loud. The audio signal distorts into square-waves because the "tops" of the signal are flattened. The device cannot handle power levels over a certain level. When this level is exceeded, clipping occurs. Clipping is usually harsher in digital devices than in analog devices. Analog clipping tends to b...

Hi guys, When I graph individual audio samples what's the best way to zoom out ? I can think of several methods but I'm not sure which will...

Hi guys, When I graph individual audio samples what's the best way to zoom out ? I can think of several methods but I'm not sure which will work best. Say I am zoomed out to level 8, meaning for every eight samples I plot one of them, I can simply ignore seven of them, I can average them, I can use the maximum value, and so on. What do you think ? Thanks, Robert A.

Hello I am a biologist and i am trying to eliminate certain sounds from a stereo recording eg the sound of cars from the recording of a...

Hello I am a biologist and i am trying to eliminate certain sounds from a stereo recording eg the sound of cars from the recording of a park, without loosing a lot in audio quality. I have some experience in audio and computers though i am not familiar with programming I need suggestions on some programms i can use for this purpose Thanks Dimitris

In my experience 1st and 2nd order all-pass filters are usually specified by their frequency, i.e. the frequency at which the phase shift is half...

In my experience 1st and 2nd order all-pass filters are usually specified by their frequency, i.e. the frequency at which the phase shift is half of the maximum. For example, see r b-j's audio cookbook (http://www.harmony-central.com/Computer/Programming/Audio-EQ-Cookbook.txt) for a 2nd order digital implementation. However, often times in audio, I see all-pass filters specified by the amo...

I have a program that has 64 bit audio of type double(range:-1.0 to 1.0) held in memory. I want to write it to disk as 16 bit audio. Do I fill...

I have a program that has 64 bit audio of type double(range:-1.0 to 1.0) held in memory. I want to write it to disk as 16 bit audio. Do I fill the range of the type short and convert my data to integers with a range of -32768 to 32767? Or, do I try to preserve symmetry and convert to a range of -32767 to 32767? Thanks.

Can anyone point me to a general solution for measuring the zero crossing rate of an audio signal if I have the individual...

Can anyone point me to a general solution for measuring the zero crossing rate of an audio signal if I have the individual samples?? Thanks Carmen Branje

I have two 16-bit wav files and am using Microsoft DirectSound to play and mix the files. I'm able to "mix" the two audio files by...

I have two 16-bit wav files and am using Microsoft DirectSound to play and mix the files. I'm able to "mix" the two audio files by simply adding the wav data values together. I have a problem however, when attempting to "fade" from one audio file to the other. Algorithm is listed below: Get two bytes of WAV data from the "right file" Put second byte into uint (32 bit) and bit-shift 8 b...

Audio Compression Optimization: MP3, AAC, OGG This paper will give you understanding on how one can achieve better compression ratio by bit...

Audio Compression Optimization: MP3, AAC, OGG This paper will give you understanding on how one can achieve better compression ratio by bit rate optimization. The key point is that our approach describes a fully automated manner of choosing the bit rate that will preserve the audio quality you define. Read this paper through and find out how to save on size when encoding your podcasts, sav...

Hello, I am working on an Open Source HW/SW project, DMFX-1, Audio Digital Multi Effects, originally conceived as a guitar pedal that will...

Hello, I am working on an Open Source HW/SW project, DMFX-1, Audio Digital Multi Effects, originally conceived as a guitar pedal that will include multitude of effects: Distorsion, Overdrive, Fuzz, Compressor, Equalizer, Phaser, Chorus, Delay, Reverberation, Flanger, Tremolo, Pitch Shift, Wah-wah ... HW is based on two TI TMS320C5535 DSP, one is used for digital audio and the other one as user ...

Vladimir Vassilevsky wrote: > Yes, of course. Pretty much any good algorithm of the frequency > estimation can be used. It will make the...

Vladimir Vassilevsky wrote: > Yes, of course. Pretty much any good algorithm of the frequency > estimation can be used. It will make the FM demodulator which performs > better then the PLL. Also, the statistics of the audio signal can be > exploited... However the radical solution will be encoding the audio > into MP3 and transmitting it digitally. No analog solu

Can anyone point me in the direction of some good technical literature on approaches one can take in developing digital audio mixing software? ...

Can anyone point me in the direction of some good technical literature on approaches one can take in developing digital audio mixing software? In particular, I'd like to know the most common and/or effective methods of avoiding clipping after addition. Thanks very much!

Hi: Decreasing the pitch of the audio in an audio linear-PCM [wave] file means the sample-rate of the wave file can be decreased without...

Hi: Decreasing the pitch of the audio in an audio linear-PCM [wave] file means the sample-rate of the wave file can be decreased without causing aliasing. If the bit-resolution and # of channels [1 in mono, 2 in stereo] of the file are kept constant, then decreasing the sample-rate will decrease the file size. Adobe Audition allows the alteration of pitch without changing speed. ht...

PC Audio Systems Software Architect - Location: Austin, TX (F/T) Contact: Dee Dee Dial, Pedley-Richard & Assoc., Inc. Email: ...

PC Audio Systems Software Architect - Location: Austin, TX (F/T) Contact: Dee Dee Dial, Pedley-Richard & Assoc., Inc. Email: dddial@pedley-richard.com ; Phone: 512/418-3260 Company Profile: Global leader of advanced semiconductor solutions to lead-edge communications companies that drive innovation and convergence in voice, data, and wireless networks. Responsibilities The Audio ...

Hello everybody, I have Denon AVR-2307CI (Audio/Video reciever) with me. I am feeding graphics card's output to the HDMI input for Denon...

Hello everybody, I have Denon AVR-2307CI (Audio/Video reciever) with me. I am feeding graphics card's output to the HDMI input for Denon reciever. Now the problem is the sampling frequency of the signal is stuck to 48KHz on Denon screen. I tried changing the sampling frequency of signal using sound card's control panel (Sound cards tried: X-Mystique and M-Audio) but it didnt help . Can any...

Hi, I am trying to implement an IVR functionality as a part of Windows application. I don't want to use TAPI. So, I am looking for a source...

Hi, I am trying to implement an IVR functionality as a part of Windows application. I don't want to use TAPI. So, I am looking for a source code detecting DTMF signals within a PCM audio stream. Can anybody help me on this? Any references? Thanks in advance, Mark