Is Raspberry PI 2 B suitable for a real-time audio effects application?

Started by Mauritz Jameson in comp.dsp2 years ago 19 replies

I am about to start working on a project which involves real-time audio processing for an audio effects application. The application is supposed...

I am about to start working on a project which involves real-time audio processing for an audio effects application. The application is supposed to acquire and process audio from 8 individual audio input channels at a sample rate of 48kHz. The audio processing algorithm generates audio for 8 individual audio output channels. The audio sample format is 16-bit signed LPCM. Since it's for a re...


Converters

Started by Dan NITA in comp.dsp14 years ago 2 replies

Hello everybody, I'm wonder why the audio A to D Converters are more expensive and fewer that the audio D to A Converters. Analog Devices...

Hello everybody, I'm wonder why the audio A to D Converters are more expensive and fewer that the audio D to A Converters. Analog Devices produce only 3 audio ADCs and 13 audio DACs. Texas Instruments manufacture 7 audio ADCs and 56 audio DACs. For a low-cost and multi-channel application I found PCM1606 quite reliable with 6 DAC channels for less that 3$ price. Can anyone point me ...


instantaneous audio streaming

Started by louis in comp.dsp11 years ago

Hi there, I was wondering if anyone out there might know of any good references/books/journals etc on what is involved in audio...

Hi there, I was wondering if anyone out there might know of any good references/books/journals etc on what is involved in audio streaming.. I am programming an application where I synthesize a particular audio sound. I was able to precompute everything and then dump out the audio file. However I now woudl like to compute the audio file "on the fly" and have a streamed output. Computati...


instantaneous audio streaming

Started by louis in comp.dsp11 years ago

Hi there, I was wondering if anyone out there might know of any good references/books/journals etc on what is involved in audio...

Hi there, I was wondering if anyone out there might know of any good references/books/journals etc on what is involved in audio streaming.. I am programming an application where I synthesize a particular audio sound. I was able to precompute everything and then dump out the audio file. However I now woudl like to compute the audio file "on the fly" and have a streamed output. Computati...


what is audio shapping?

Started by Anonymous in comp.dsp13 years ago 1 reply

Hi, What is audio shapping? and why we want to shape the audio signal before sending it to speaker? what are the main components of...

Hi, What is audio shapping? and why we want to shape the audio signal before sending it to speaker? what are the main components of audio shapping? What kind of IIR filter should be used? Thanks in advance for your informations. Gold


Question about audio artifacts; WMA vs. MP3

Started by Radium in comp.dsp10 years ago 6 replies

Hi: Why is it that MP3s needs some amount of encoded audio in order to have any audio at all, while WMA can simply make its own audio?...

Hi: Why is it that MP3s needs some amount of encoded audio in order to have any audio at all, while WMA can simply make its own audio? [To better understand this question, read below] I have Adobe Audition 1.5 in which I do audio experiments. Below is my first experiment: 1. I make a silent 44.1 KHz-sample-rate, 16-bit-resolution, monaural wave file that is 4 seconds long. [To do th...


Audio sampling question

Started by Steamer in comp.dsp10 years ago 9 replies

Hi all, I need to sample a 100 Hz - 15 kHz analog FM audio (music and speech) channel for an unusual purpose - to measure the time that...

Hi all, I need to sample a 100 Hz - 15 kHz analog FM audio (music and speech) channel for an unusual purpose - to measure the time that certain audio events occur. Audio quality, resolution, distortion are only important to the extent they affect the time accuracy of event detections. I'd like to be able to detect the occurrence of specific audio events in the channel with an accu...


Audio filtering

Started by Sreeram in comp.dsp13 years ago 17 replies

Hello, I am trying to do an audio filter program. I need to filter out certain frequencies from the audio data. I am doing FFT, Bandpass &...

Hello, I am trying to do an audio filter program. I need to filter out certain frequencies from the audio data. I am doing FFT, Bandpass & IFFT. But after doing FFT, I am 0'ing out the bins for my desired frequency range. After that I am doing Inverse FFT. But the playing audio is not proper and a "tick tick" is comming in the audio output. I need to do the filtering for all PCM form...


Identify PCM samples as audio

Started by Anonymous in comp.dsp11 years ago 9 replies

Hi, I have a system that records audio from line in, and stores it as a PCM file. Audio is recorded every once in a while since it would make...

Hi, I have a system that records audio from line in, and stores it as a PCM file. Audio is recorded every once in a while since it would make no sense for me to store it continuosly. What I would need to do is to detect if there's audio coming to line in. I'll try to explain certain situations that I hope that would clarify my needs. The main point is endurance testing, meaning that I ju...


Audio frequency

Started by mahathi in comp.dsp12 years ago 6 replies

Hi, I have some questions in the basics of audio engineering as I am new to this field. It would be great if somebody could help me...

Hi, I have some questions in the basics of audio engineering as I am new to this field. It would be great if somebody could help me out. What is the format of 24 bit audio. How does it represent the frequency and amplitude information? Is there any relation between audio frequency and the number 1.3652125? How do we determine amplitude from frequency.. Thanks... Mahathi


Audio Input on DSP Kit

Started by mehboob_iiui in comp.dsp10 years ago

I have a DSP kit , that has c5416 processor, one audio in one line in one audio out one line out I want to find out, if I connect a...

I have a DSP kit , that has c5416 processor, one audio in one line in one audio out one line out I want to find out, if I connect a microphone to the audio in, how do I get the data in my program, I am programming in C language on code composer studio 3.1


Audio Engineering Journals?

Started by davidross in comp.dsp11 years ago 2 replies

Hi folks, What are the best journals for Audio Engineering? I'm looking mainly for filter designs techniques targeted at audio. Journal of...

Hi folks, What are the best journals for Audio Engineering? I'm looking mainly for filter designs techniques targeted at audio. Journal of Audio Engineering is one, but What else is good? Thanks very much for any tips. Also, is it possible to buy a subscription to online downloadable journals where you can access a LOT of material over the net? Best Regards David


Audio sequencer development - newbie resources?

Started by Anonymous in comp.dsp9 years ago 1 reply

Hi, I'm starting up my first audio project, adding realtime audio support to a MIDI program I've developed. I'm looking for resources on...

Hi, I'm starting up my first audio project, adding realtime audio support to a MIDI program I've developed. I'm looking for resources on the principles of audio sequencer development, so I don't have to reinvent the wheel. All I have to do at first is take in realtime input and do some basic DSP on it. I can find lots of books and resources on DSP topics, but not audio-sequencer specific. ...


Audio - Increasing output sample rate increases frequency ?

Started by ABM 200 in comp.dsp2 years ago 14 replies

Suppose, i record audio at say, 10000 samples/sec for some amount of time. I store these samples in a file(WAV file). When i play back this audio,...

Suppose, i record audio at say, 10000 samples/sec for some amount of time. I store these samples in a file(WAV file). When i play back this audio, at a different rate, say 30000 samples/ sec. The audio will definitely be fast. But, theoretically more samples/ sec implies higher frequency. So, does this mean that the play back audio will be at 3 times the original (recorded) audio frequency or more...


Is 8-bit Audio Unsigned?

Started by Chris Barrett in comp.dsp10 years ago 10 replies

I've been working on this problem for some time now, and I can't seem to understand what's going on. I have data in the range of -1.0 to 1.0....

I've been working on this problem for some time now, and I can't seem to understand what's going on. I have data in the range of -1.0 to 1.0. For 8-bit audio I scale it to range of 0 to 256 and for 16-bit audio I scale it to the range of -32768 to 32767. Why can't I scale 8-bit audio to the range of -128 to 127? Why does it appear that 8-bit audio is not signed? Below is some code ...


Compare two audio signals

Started by nagarajan karunakaran in comp.dsp7 years ago 12 replies

Hi, i am working on a concept where i want to compare two audio file (content of audio file)is identical to some extent.For ex if it's...

Hi, i am working on a concept where i want to compare two audio file (content of audio file)is identical to some extent.For ex if it's two different people saying the same words .I have to say they are equal. I have done goggling but i not able to predict what is really needed for my project. I don't know ,where to start and how to start.How can i compare audio files in that manner.Please he...


Audio delay circuit

Started by George W in comp.dsp13 years ago 5 replies

I have a need to measure the relative delay between two audio signals arriving at a receiver via different paths. The audio signals are...

I have a need to measure the relative delay between two audio signals arriving at a receiver via different paths. The audio signals are standard 3 kHz audio bandwidths containing voice or music. One could arrive as much as 1 msec before or after the other. I'd like to be able to measure the difference in time of arrival between the two signals down to 1 usec if that's possible. Any sugg...


TV stereo audio decoding

Started by Anonymous in comp.dsp11 years ago 2 replies

I am trying to design a professional broadcast television product based upon a Linux server which would incorporate decoding of a digitized BTSC...

I am trying to design a professional broadcast television product based upon a Linux server which would incorporate decoding of a digitized BTSC signal (the stereo audio and second audio channel on a NTSC TV) into its component audio streams. My preferred implementation would be in a C language SDK which implements that functionality. Is anyone familiar with such an SDK or know of a consul...


Regarding comparison of audio quality

Started by vasindagi in comp.dsp9 years ago 10 replies

Hi, I m working on QMF filters wherein I split the input audio into different bands and then reconstruct them. I wanted to know how can I compare...

Hi, I m working on QMF filters wherein I split the input audio into different bands and then reconstruct them. I wanted to know how can I compare the quality of the input audio and the reconstructed audio. Right now I m using mean square error which I feel is not sensible because the more the filter length I use more is the error that is showing up. Can I use cross correlation to compare the q...


Upsample, Downsample in the DSP audio Processing

Started by Kathy_pdx in comp.dsp2 years ago 14 replies

The audio system sample rate is 192 kbits/s. The DSP audio processing involves the ASIC hardware and DSP processor. Due to the...

The audio system sample rate is 192 kbits/s. The DSP audio processing involves the ASIC hardware and DSP processor. Due to the real-time constrains, the equalizer need be implemented at 16 kbits/s by a DSP processor. The routine way is to first down sample the audio signal to 16 kbits/s, design an equalizer, then apply the equalizer to the audio signal at 16 kbits. Then the equalized audio signal ...