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...

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 ...

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...

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 ...

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. ...

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...

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...

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...

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...

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 ...

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...

hi, i'm trying to match reference audio signals (ranging between 1 sec and 10 seconds) in source audio signal from an RTP stream. i've used...

hi, i'm trying to match reference audio signals (ranging between 1 sec and 10 seconds) in source audio signal from an RTP stream. i've used cross-correlation, which works extremely well (spikes nicely) when there is not much time-altering of the reference audio as it is passed over the RTP stream (i.e., not much packet loss or packet re-ordering, and the local packet-loss-concealment / j...

Hi, sorry to post 2 questions in the same day but Ive been searching all day and got nowhere. Im trying to compute the log energy of an...

Hi, sorry to post 2 questions in the same day but Ive been searching all day and got nowhere. Im trying to compute the log energy of an audio frame. I am aware that the result of my calculation should be a sinle value for the whole audio frame, so it is not the log of the spectrum. I know that to find the energy of an frame you get the frame from the audio stream with a window functio...

Hi everyone, I want to embed ID numbers in audio so that I can recover them on a separate device. Some of my constraints: -The audio will be...

Hi everyone, I want to embed ID numbers in audio so that I can recover them on a separate device. Some of my constraints: -The audio will be transmitted from a standard TV/DVD set-up (so DVD encoding, TV speakers). -The audio will be received and processed on a cell phone (so cell phone microphone, processing capabilities (but can be a high-end cell phone)) -There will be significant back...

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...

Hi, This is a noob question - I'm doing some research in audio & voice, and I need some advice. What are the most popular platforms to...

Hi, This is a noob question - I'm doing some research in audio & voice, and I need some advice. What are the most popular platforms to implement high end audio & voice applications on? To be more specific, I'm looking at audio/voice applications that make heavy use of (large tap) digital filters, like vocal harmonizers, digital mixing consoles, and acoustic echo cancellers. What platfor...

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

Hello, I am writing an application in which I need to be able to move trough an audio file based on where the user places the cursor, and if...

Hello, I am writing an application in which I need to be able to move trough an audio file based on where the user places the cursor, and if the user holds the cursor on a given position, I need to "freeze" the sound. In order to achieve this, I am looping 50ms of the audio from the position of the cursor. However this generates some noise in the audio, because the beginning and the end of...

Could Mathlab make sense using it as a evaluation platform for audio DSP? This means: Streaming high quality audio using a firewire audio...

Could Mathlab make sense using it as a evaluation platform for audio DSP? This means: Streaming high quality audio using a firewire audio interface for example through a experimental adaptive filter coded in mathlab. And doing measurements and visualizing results in vaious ways (and fast). And doing simple GUI's for controlling that conveniently.

In a live audio streaming system, audio pops are caused by network packet delay. How to smooth the sharp decay, and what stuff should be filled...

In a live audio streaming system, audio pops are caused by network packet delay. How to smooth the sharp decay, and what stuff should be filled in the empty buffer?