Audio Connection for BF 537 EZkit lite board

Started by isnithin in comp.dsp11 years ago

Hello Everyone, I am developing an Audio application project on ADSP Bf537 EZ-LITE board. I have developed compression and...

Hello Everyone, I am developing an Audio application project on ADSP Bf537 EZ-LITE board. I have developed compression and decompression algorithms for the Audio samples. The algorithm is working fine and I have tested it. Now I want to achieve live recording and output the recorded samples at the speaker/headphone connected at "LINE_OUT" port of the board. I have a Mike f...


Wireless Home Video/Audio Component Interfaces

Started by Randy Yates in comp.dsp13 years ago

Folks, This is a survey question. What devices/technology is currently available to a designer for implementing wireless interfaces?...

Folks, This is a survey question. What devices/technology is currently available to a designer for implementing wireless interfaces? For example, if I wanted to implement a wireless DVD player that transmits the decoded video and audio streams to a TV and/or audio receiver system, what sort of technology is available to me today? (Assuming the TV also had such an interface...) For examp...


Importance of phase in Audio

Started by Anonymous in comp.dsp13 years ago 12 replies

Hello all, i am designing a low pass filter and high pass filter to pass my audio samples. in matlab i checked out both butterworth...

Hello all, i am designing a low pass filter and high pass filter to pass my audio samples. in matlab i checked out both butterworth and chebyshev(cheby2) 3rd order filter. I have observed there is a lot of differance in the phase plot of both the filters. Now my query went to what actuall effect of phase in audio? please anybody help me in solving this. Looking for the repl...


Converting voltages to PCM audio (programming related)

Started by Anonymous in comp.dsp13 years ago 8 replies

Hello everyone, I'm recording audio with a data acquisition card from the speaker cables. When recording, I get an array of unsigned short...

Hello everyone, I'm recording audio with a data acquisition card from the speaker cables. When recording, I get an array of unsigned short integer values. These I need to convert to PCM audio. Everything works quite nicely, except the fact that a lot of noise occurs in the PCM file. I'm using Measurement Computings PCI-DAS6036 -card, and I have tried this also with a simulated National Ins...


FFT of audio signal resolution

Started by vivu91 in comp.dsp8 years ago 13 replies

Hi, I have a doubt which might be very basic. I am currently working on a project. In that i input a test audio signal of 1.5 seconds in which...

Hi, I have a doubt which might be very basic. I am currently working on a project. In that i input a test audio signal of 1.5 seconds in which the actual audio is of only 0.5 seconds or so. The sampling frequency is 44100. I am currently using a hanning window and the pwelch command in matlab to find the frequency spectrum. However I am using a hanning window of size 128 and that gives me a resol...


Multi-Sharc architecture

Started by Jerome in comp.dsp14 years ago 4 replies

Hi All We are ready to start a new audio design featuring 4 DSPs 21262 from ADI. I was wondering if anyone has experience in multiple...

Hi All We are ready to start a new audio design featuring 4 DSPs 21262 from ADI. I was wondering if anyone has experience in multiple dsps architectures, more precisely in inter-communication between the dsps. Our architecture is currently as follow : - 1 dsp to interface inputs / outputs (audio codecs, spdif, usb audio ..) - 3 dsp to generate sounds based on our algorithms. the dsp 1...


headless audio file - identify and repair?

Started by prefetch in comp.dsp14 years ago 2 replies

hi all, i've got a ~10MB file which i think is a realmedia audio file (because the filename ended with .rm and i found it with other realmedia...

hi all, i've got a ~10MB file which i think is a realmedia audio file (because the filename ended with .rm and i found it with other realmedia audio files) but the index header is missing. it all seems to be just "data". could anyone give me a clue as to how to: 1-identify what format this raw data is in? 2-plop a header onto it so i can use it? any experts out there?


Audio Compression

Started by Udhay in comp.dsp12 years ago 5 replies

Sir, I am udhay. I am a student and i am working on Audio compression for my exam. I want to know how does the compression take place in...

Sir, I am udhay. I am a student and i am working on Audio compression for my exam. I want to know how does the compression take place in audio?? I have doubt in human earing capacity.The maximun hearing capacity of a human is 20-20khz. if a take a sample wave file,its coming around 40khz. what it mean? can anyone help me to clear this doubt.. udhay


Off-Topic : Looking for IP camera which can send audio via RTP to remote client

Started by Mauritz Jameson in comp.dsp2 years ago 6 replies

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?


Audio sampling in C6713 SDK

Started by haiyan in comp.dsp12 years ago

Dear all, I'm sampling the audio signal through mic jack on C6713 board. The code is based on the example project "dsk_app". The audio can...

Dear all, I'm sampling the audio signal through mic jack on C6713 board. The code is based on the example project "dsk_app". The audio can be played back from the speaker jack. However, when I tranform the signal in the buffer to double precision values, it seemed that (later) half of the data in the buffer has been truncked (all zeored). Is there any one can give me some advice to solve this...


Live audio resampling problem!

Started by in comp.dsp11 years ago 3 replies

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?


counterphase detection in stereo audio

Started by kork in comp.dsp8 years ago 20 replies

Hi folks, I'm going to develop a quality control application that inspects recently imported audio files for a number of checks. One of them is...

Hi folks, I'm going to develop a quality control application that inspects recently imported audio files for a number of checks. One of them is the detection of counterphase fragments in the file. With counterphase I mean a 180 degrees (or pi rad, if you prefer) phase shift between the two audio channels in the (stereo) file. In a radio broadcast of the file this is killing when it is listene...


Cutting and Resampling

Started by kingdavid3 in comp.dsp12 years ago 2 replies

I am reading a full length (duration of the speech) audio wave file sampled at 44.1KHz, 16 bit stereo. I want to cut a 15 second segment from...

I am reading a full length (duration of the speech) audio wave file sampled at 44.1KHz, 16 bit stereo. I want to cut a 15 second segment from the audio and resample it to 16000 Hz. So far I managed to do the resampling. Please help on how to cut 15 second length portion of the audio. Thank you [s, fs]=wavread('test.wav); %%cut to 15 seconds s=resample(s, 16000, fs); %downsam...


16 bit stereo Audio to 8 Bit Mono

Started by kingdavid3 in comp.dsp12 years ago 4 replies

How do I convert a 16 BIT, stereo wave audio.... into an 8 BIT mono audio in MATLAB. Please help [s, fs]=wavread('test.wav'); snew=(:,1);...

How do I convert a 16 BIT, stereo wave audio.... into an 8 BIT mono audio in MATLAB. Please help [s, fs]=wavread('test.wav'); snew=(:,1); %This converts it to mono


Project in Digital Audio Effects

Started by Balaji in comp.dsp14 years ago 6 replies

Hi .. I have to do my project in Audio(Digital Audio Effects) I am confusing from where I have to start? Can any body guide how to follow in a...

Hi .. I have to do my project in Audio(Digital Audio Effects) I am confusing from where I have to start? Can any body guide how to follow in a simple manner..and in order. I want to know from where I have start and how to follow.. what can I do in this feild(ie some project titles) waiting for reply Thanks.......... bye


free C++ library for speech/audio processing

Started by ldkha1979 in comp.dsp11 years ago

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!


Any PCM audio amplitude manipulation libraries available?

Started by Verified by Kerberos in comp.dsp16 years ago 4 replies

Hi, I'm working on an embedded ARM board running Linux 2.4.18. The processor is a Hynix 7201 based on ARM720T core, running at 60MHz...

Hi, I'm working on an embedded ARM board running Linux 2.4.18. The processor is a Hynix 7201 based on ARM720T core, running at 60MHz (25 BogoMIPS), with a Piccolo DSP. I want to change audio volume by software, however the board doesnt have any hardware mixer for audio. So changing the volume via OSS etc. is not possible. Since I require only to playback WAV PCM files, i figure that i...


Slow Motion Recording

Started by Stacy in comp.dsp11 years ago 10 replies

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...


Standard audio compression (LA-2A, not MP3) algorithm

Started by Pat Farrell in comp.dsp13 years ago 1 reply

This may be a FAQ, but I don't see one, and you guys seem pretty smart. Is there a standard algorithm for doing audio compression in a DSP...

This may be a FAQ, but I don't see one, and you guys seem pretty smart. Is there a standard algorithm for doing audio compression in a DSP context? I mean the type of compression that recording studios and radio stations used to use a Universal LA2A or Universal Audio 1176LN. While these are typically thought of as compression (making the loud ampliture software) they usually work by making...


Audio equalizer!

Started by tasi in comp.dsp15 years ago 2 replies

Hi, Greeting, I am interesting with a fully digital audio equalizer. For I am a fresh in this field, I don't how to start. Does somebody can...

Hi, Greeting, I am interesting with a fully digital audio equalizer. For I am a fresh in this field, I don't how to start. Does somebody can tell me how to get/find a useful information? Thx