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...

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?

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

Hi: Is the following possible?: A linear uncompressed PCM audio file whose sample-rate is 44.1 KHz, monaural, and with a bit-rate of...

Hi: Is the following possible?: A linear uncompressed PCM audio file whose sample-rate is 44.1 KHz, monaural, and with a bit-rate of 1-bit-per-second. The 1-bit-per-second is because the bit-resolution [normally 16-bit in CD audio] is only 1-bit-per-44,100 samples in this hypothetical case. Since the sample-rate is 44.1 KHz, and there is only 1 channel [due to the monaural audio], the...

I'm working on a hobby project that processes several communcations grade audio channels and I'm looking for a *cheap* way to get eight full...

I'm working on a hobby project that processes several communcations grade audio channels and I'm looking for a *cheap* way to get eight full duplex audio paths. The on-board interface today is usually full deplx stereo, so that's two. I've been looking at cheap USB audio interfaces and they all seem to have mic rather than line inputs, so a single channel input. Does anyone know of o...

Hi: I=2E Audio vs. Video Digitized (mono) audio has a single sample per each sampling interval. In the case of digital video, we could...

Hi: I=2E Audio vs. Video Digitized (mono) audio has a single sample per each sampling interval. In the case of digital video, we could treat each individual sample point location in the sampling grid (each pixel position in a frame) the same way as if it was a sample from an individual (mono) audio signal that continues on the same position in the next frame. For example, a 640=D7480...

I have big problem with my app. I'm using dm642 evm (Texas Instruments), and I need to use both video and audio input (in rf5 environment) ,...

I have big problem with my app. I'm using dm642 evm (Texas Instruments), and I need to use both video and audio input (in rf5 environment) , but I can't make them work together. I've added dio codec driver for aic23 audio codec and gio video(fvid interface) driver in dsp/bios module (i've used .cdb file from rf5 audio example and modyfied it to match my needs) but when i'm trying to run my ...

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

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...

Hi, all I believe this is a very basic and old question. I know there are a number of advantages of subband processing. Here I'd like to ask...

Hi, all I believe this is a very basic and old question. I know there are a number of advantages of subband processing. Here I'd like to ask for your opinions of why subband is so widely used in audio signal processing applications, not only in audio compression, but on many other audio related topics, e.g BSS, surround effects... Researchers also talked about octave-band filter bank...

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

dear all, I am a computer enggr. doing masters. The final Project involves lot of DSP. The project is on audio classification. The initial step...

dear all, I am a computer enggr. doing masters. The final Project involves lot of DSP. The project is on audio classification. The initial step is to obtain mp3 audio stream. Is there any freely available library to obtain the audio data as floating point values. This would help to reduce work load. I have tried using the LibMpeg3 but to i am stuck trying to convert to library. It would be kind...

Does anyone know of a way to send a high-quality (better than telephone-quality) audio stream over the Internet? I would like to stream audio...

Does anyone know of a way to send a high-quality (better than telephone-quality) audio stream over the Internet? I would like to stream audio from my work to my home. I have a broadband Internet connection at both ends, and VPN is possible as well. There seems to be plenty of products that deal with telephone-quality audio (VOIP, voice-chat program, etc.), but I haven't found anything f...

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...

Hi All, I am just about to start a reasonably simple DSP project processing audio (mainly filtering), I need 24 bit data, on board 8kw FLASH...

Hi All, I am just about to start a reasonably simple DSP project processing audio (mainly filtering), I need 24 bit data, on board 8kw FLASH and 4kw RAM, > 30MIPS and low cost development environment. I need a small parts count so I can interface an AES/EBU (S/PDIF) audio RX to a DAC -- I2S is the obvious interface. Are there any newer DSPs that support this audio interface directly as