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: 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...

hi all, I'm new to speech/audio processing. I would like to know if there is any free C++ library (that can work well with MS VC++ .net 2005)...

hi all, I'm new to speech/audio processing. I would like to know if there is any free C++ library (that can work well with MS VC++ .net 2005) for speech/audio processing? With this library, we can read/write .wav file, do some basic thing like filter, .... on the signal. Thks a lot!

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'm a computer engineering student and I'm new to the world of DSPs. I've decided for a major project to design a digital audio...

Hello, I'm a computer engineering student and I'm new to the world of DSPs. I've decided for a major project to design a digital audio effects proccesor, so that I can plug my guitar into it and have it add some effect in real-time and play it out. I'm having some trouble understanding how and where to do the actual processing of the audio signal. So far I've just been doing lots of reading, an...

I am trying to decode audio streams sent from an arm board, unfortunely the clock on the arm board can vary from time to time, as a result the...

I am trying to decode audio streams sent from an arm board, unfortunely the clock on the arm board can vary from time to time, as a result the wave sample rate varies. To keep high audio quality, the delay must be kept constant. Can anyone give me some advice? Do I have to design a resample algorithm whose conversion rate can be changed from time to time?

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 ...

How do you slow down a digital audio recording and keep the same pitch? For example if the audio sample was recorded at 44.1k you could, by...

How do you slow down a digital audio recording and keep the same pitch? For example if the audio sample was recorded at 44.1k you could, by S/W intervention, tell it to play at 22.05k. That will slow it down by half speed. However, it will also lower the pitch by half also. The approach I would take to slow down by 2 is: 1) Find a way to double the frequency of the audio sample. I would try...

Dear audio/DSP experts: Please go to my *Google Drive* and click once on the *AT&T* folder. http://goo.gl/Jm4bZ7 Watch the TV...

Dear audio/DSP experts: Please go to my *Google Drive* and click once on the *AT&T* folder. http://goo.gl/Jm4bZ7 Watch the TV commercial (MP4) and listen to the background music. Since my objective is to apply some *Voice Identification* process to the audio, as a preliminary step, I would like to remove the music. I have already identified the song as "You're My Best Frie...

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...

Has anyone measured the audio performance of the Freescale Soundbite audio board using simple pass through code? I am interested primarily in...

Has anyone measured the audio performance of the Freescale Soundbite audio board using simple pass through code? I am interested primarily in THD, freq response, etc....at 16/24 bits, 44.1/etc fs. Thank you , TomC

Hello everyone, it's been a while ago since someone asked here for an DSP eval board (for audio processing) that's got a SAR analog-digital...

Hello everyone, it's been a while ago since someone asked here for an DSP eval board (for audio processing) that's got a SAR analog-digital converter. Theoretical it should be possible to reach very low delay between an audio input and the output after processing by a DSP. Does anyone know a device with a fast SAR ADC (that could be used for audio processing)? A board with suiting anti-aliasi...

Hi admin, The code below to process the audio signal on red5.Audio signal on red5 is the type of data bytes.I want to change the data type to a...

Hi admin, The code below to process the audio signal on red5.Audio signal on red5 is the type of data bytes.I want to change the data type to a sample (float). public void start(IBroadcastStream broadcastStream, IScope scope) throws StreamException { if (log.isDebugEnabled()) log.debug("startTranscodingStream({},{})", broadcastStream.getPublishedName(), scope.getName()); recordStream(broa...

I'm working on a platform where the audio driver exhibits quite irregular callback patterns. My expectation was that if I configure the audio...

I'm working on a platform where the audio driver exhibits quite irregular callback patterns. My expectation was that if I configure the audio driver to give me callbacks every 10ms, I would get them approximately every 10ms (give or take some milliseconds). Ideally, the callback pattern would look something like this: t = 0ms : speaker callback t = 1ms : mic callback t =3

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 ...

I want to pursue a PhD in Electrical/Computer/Audio engineering and would like a university that has active research for 3D audio (HRTFs)....

I want to pursue a PhD in Electrical/Computer/Audio engineering and would like a university that has active research for 3D audio (HRTFs). However, I don't want the typical suggestions that I receive like UC Davis, U Maryland College Park and MIT. I am aware of these programs. I am hoping that maybe there are some less know, but good programs, that I can try. The reason is I want to apply to as ma...

I am looking for an IP camera that: 1. will allow me to programmatically connect to it (over UDP or TCP) 2. will send me the camera's...

I am looking for an IP camera that: 1. will allow me to programmatically connect to it (over UDP or TCP) 2. will send me the camera's microphone audio as RTP over UDP 3. supports the following audio sample rates: 8000Hz and any sample rate greater than or equal to 44.1kHz Any recommendations?

I have to do cross correlation of two audio file to prove they are similar. I have done FFT of two audio files and have the power spectrum value...

I have to do cross correlation of two audio file to prove they are similar. I have done FFT of two audio files and have the power spectrum value of both the audio file in a separate array.how should i proceed further to do cross correlation to prove they are similar.Any basic ideas will be helpful for me to learn and apply it.Thanks.

Is there a way of connecting a Windows PC to a satellite phone so that the following can be done from software running on the PC: 1. Remotely...

Is there a way of connecting a Windows PC to a satellite phone so that the following can be done from software running on the PC: 1. Remotely control the satellite phone from software on the PC (i.e. answer call, hang up call, adjust speaker volume, adjust mic sensitivity etc) 2. Receive speaker audio stream as a digital 16-bit signed PCM mono audio stream. 3. Send 16-bit signed PCM mono audi...

This may ultimately be a dumb question, but I need to ask it. I'm looking at using a stereo audio ADC as an I/Q baseband sampler in a low-rate...

This may ultimately be a dumb question, but I need to ask it. I'm looking at using a stereo audio ADC as an I/Q baseband sampler in a low-rate communication system. One like this: http://www.ti.com/product/tlv320adc3001 I seem to recall some horror stories of using audio ADCs for general use, but I can't really seem to see any reason to not go this way. Am I missing something? Th...