Director of DSP Audio Software Engineering (PC Systems-Driver level) Location: Austin, Tx Contact: Dee Dee Dial Pedley-Richard & Assoc.,...

Director of DSP Audio Software Engineering (PC Systems-Driver level) Location: Austin, Tx Contact: Dee Dee Dial Pedley-Richard & Assoc., Inc. email: dddial@pedley-richard.com Our client is a global leader of advanced semiconductor solutions to lead-edge communications companies that drive innovation and convergence in voice, data, and wireless networks. Responsibilities Director...

Hi all, I am currently on a design of an equalizer for Medium/Short wave AM receiver. Problem statement: Medium /Short waver AM receiver...

Hi all, I am currently on a design of an equalizer for Medium/Short wave AM receiver. Problem statement: Medium /Short waver AM receiver suffers from sudden volume reduction irrespective of the Automatic gain control in receiver. How does this can be addressed. I am trying to control this audio reduction by an equalizer. But how does the actual reduction of audio in the radio program ...

Hi, I'm using a C program to output a set of audio samples to my sound card. The samples belong to a marimba I sampled and I have each...

Hi, I'm using a C program to output a set of audio samples to my sound card. The samples belong to a marimba I sampled and I have each note from 220 to 440Hz sampled. At the moment I have a buffer, which I can update in real time, which contains what note to play and at what time to play them. I can update this buffer in realtime. I have each of my samples stored in a seperate buffer, and ...

Hi, I'm having a bit of trouble modifying one of the TalkThru projects that comes with SHARC 21469 EZKit from AD. I am trying to implement...

Hi, I'm having a bit of trouble modifying one of the TalkThru projects that comes with SHARC 21469 EZKit from AD. I am trying to implement this signal chain: ADC (48kHz) -> SRC (48kHz -> 16kHz) -> Audio Processing -> SRC (16kHz - > 48kHz) -> DAC (48kHz) where audio is sampled at 48kHz. The signal is routed through the SRC and resampled to 16kHz. The resampled signal is then processe

I'm writing a guitar tuner application for mobile. Captured audio samples are passed to fft routine. HSP is done for frequency spectrum. HSP step...

I'm writing a guitar tuner application for mobile. Captured audio samples are passed to fft routine. HSP is done for frequency spectrum. HSP step by step: -audio samples are passed to fft -frequency spectrum magnitudes are computed -downsample 2x, 3x , 4x power spectrum - multiply downsampled spectra It recognizes frequencies between 440+ down to 150. Below 150hz frequencies are recognized ...

Can anyone recommend a good starting point for creating code that does Sound Identification/Matching? I was going to start creating a library...

Can anyone recommend a good starting point for creating code that does Sound Identification/Matching? I was going to start creating a library of FFT snapshots consisting of varying time window lengths for the sounds I want to identify in audio streams. Then I was going to start matching up sounds that way by comparing snapshots taken from incoming audio seeing how close they are to the libr...

Hi all, Where can I find audio samples (*.wav or *.mp3) of low and high frequencies? I am looking for particular frequencies like 2-4 kHz...

Hi all, Where can I find audio samples (*.wav or *.mp3) of low and high frequencies? I am looking for particular frequencies like 2-4 kHz and 18-20 kHz. Thanks, Abhishek

hi all, i am searching for the ISO test samples which are used for the conformance tests of AAC codecs. we have the complete set of ISO samples...

hi all, i am searching for the ISO test samples which are used for the conformance tests of AAC codecs. we have the complete set of ISO samples which ISO provides but the sine sweep signal is compressed as a mpg file. we need an AAC file with an ADTS header. if someone has found or used these AAC ADTS sine sweep audio samples please let us know where we can find them. we could obviously...

Hi, I am trying to process audio came from microphone using FFT. But audio driver gives only 512 bytes in a buffer. Sampling rate...

Hi, I am trying to process audio came from microphone using FFT. But audio driver gives only 512 bytes in a buffer. Sampling rate is 44Khz. Whether to do FFT complete sample for 1 sec is required ? Or I can process these 512 Bytes buffer one by one ? If yes any examples on net ? I already have a Function which takes buffer and sampling rate and does FFT. (i.e from Numerical Recipies usin...

Im trying to implement a biquad cascade, but I keep having problems and I think it has to do with the fixed point implementation of...

Im trying to implement a biquad cascade, but I keep having problems and I think it has to do with the fixed point implementation of filter coefficients. I was getting filter coefficients from the rbj audio cookbook for simple high shelf, low shelf, etc. The following code is how I normalize them and convert to fixed point. I'm processing pcm audio data and I tried to convert the filter coeffici...

Hi, We welcome audio/speech/voice quality professionals to evaluate our tool that performs automated voice quality testing providing quantative...

Hi, We welcome audio/speech/voice quality professionals to evaluate our tool that performs automated voice quality testing providing quantative reasons for voice quality loss. Here one can find command line version of the tool: http://www.sevana.fi/speech/VQcmd.zip and here one can find more information about the program and technology: http://www.sevana.fi/voice_quality_testing_measur...

I'm creating an FFT-based real-time analyzer for audio (i.e. live sound,) and have pretty much everything figured out except for this one thing:...

I'm creating an FFT-based real-time analyzer for audio (i.e. live sound,) and have pretty much everything figured out except for this one thing: should I plot the power spectrum (log of the _square_ of the magnitude of FFT outputs,) or the amplitude spectrum (log of the "plain old" magnitude of the FFT outputs)? (I think I'm using these terms correctly; correct me if I'm wrong.) Every desc...

I have a DAC with 12 bit resolution that I want to use to playback an audio clip with 16 bits/sample. Obviously, I'm going to loose...

I have a DAC with 12 bit resolution that I want to use to playback an audio clip with 16 bits/sample. Obviously, I'm going to loose some quality, but I was wondering if it's possible to improve the quality a little by increasing the playback rate. For instance, my audio clip has 44,100 samples pers second, 16 bits per sample: If I played back with 12 bits per sample, but, say, 88,200 samples ...

hi, I'm trying to interface audio codec and SDcard with DSP[DM642]... SDcard is connected with MCBSP port and audio codec is connected...

hi, I'm trying to interface audio codec and SDcard with DSP[DM642]... SDcard is connected with MCBSP port and audio codec is connected with MCASP0.... I'm also using Video port 1 and video port 2.. In the datasheet, it is given that MCASP0's control is in videoport 0 and data is in videoport 1... My doubt is i'm using videoport 1 for video in interfacing... If i connect the MCASP...

Hi: What are the specs of uncompressed linear PCM video? The specs of uncompressed linear PCM audio are bit-rate, sample-rate,...

Hi: What are the specs of uncompressed linear PCM video? The specs of uncompressed linear PCM audio are bit-rate, sample-rate, bit-resolution, and number of channels [1 if mono, 2 if stereo] In LPCM audio, Bit-rate = sample-rate X bit-resolution X # of channels In CD-audio, sample-rate is 44.1 kHz, bit-resolution is 16-bit, and with 2 channels [stereo] For CD, Bit-rate = 4...

Hi: Why does DVD-Audio use 192 kHz sample rate? What's the advantage over 44.1 kHz? Humans can't hear the full range of a 192 kHz sample...

Hi: Why does DVD-Audio use 192 kHz sample rate? What's the advantage over 44.1 kHz? Humans can't hear the full range of a 192 kHz sample rate? On average, what is the minimum sample rate for a guy in his early to mid 20s who likes treble? I agree there are a small percentage of humans who can hear above 20 kHz. However, DVD-audio uses a sample-rate of 192 kHz which allows a maximu...

A post-doctoral research position in the field of speech and audio coding is open at Orange Labs in Lannion, France. About Orange Labs: The...

A post-doctoral research position in the field of speech and audio coding is open at Orange Labs in Lannion, France. About Orange Labs: The network of Orange Labs represents the France Telecom-Orange Group's global innovation network. It is made up of 5,000 employees on four continents. R&D is the principal source of innovation for the Group, consisting of 3,600 researchers in 17 laboratories...

I have the following query in relation to Cross-correlation and general Audio DSP signalA is 512 samples @ 44100 samples/sec. length 0.0116...

I have the following query in relation to Cross-correlation and general Audio DSP signalA is 512 samples @ 44100 samples/sec. length 0.0116 seconds I take a recording and play signalA at 0.5 seconds. The recording is 1 second long or 44100 samples @ 44100 samples/sec. When i do a Cross-correlation of signalA and the recording. i plot the result in MatLab and i get the result i was expe...

Hi: Why is it that uncompressed digital video can have so many format [such and RGB as well as linear PCM] but uncompressed digital audio can...

Hi: Why is it that uncompressed digital video can have so many format [such and RGB as well as linear PCM] but uncompressed digital audio can only have one format -- PCM?? Thanks, Radium

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