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...

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, 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 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: 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'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...

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...

Hi everyone, for audio and Signal Processing related fields, which organisations do you regard as relevant? Which of them you recommend /...

Hi everyone, for audio and Signal Processing related fields, which organisations do you regard as relevant? Which of them you recommend / think it is worth to be a member? I think about IEEE Signal Processing Society, Audio Engineering Society, Acoustical Society of America, but what else...? Thanks in advance, Regards stereo

PC Audio Systems Software Architect - Location: Austin, TX (F/T) Contact: Dee Dee Dial, Pedley-Richard & Assoc., Inc. Email: ...

PC Audio Systems Software Architect - Location: Austin, TX (F/T) Contact: Dee Dee Dial, Pedley-Richard & Assoc., Inc. Email: dddial@pedley-richard.com ; Phone: 512/418-3260 Company Profile: Global leader of advanced semiconductor solutions to lead-edge communications companies that drive innovation and convergence in voice, data, and wireless networks. Responsibilities The Audio ...

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?

Hello, I'm a computer engineering student and I'm new to the world of DSPs. I've decided for a major project to design a digital audio...

Hello, I'm a computer engineering student and I'm new to the world of DSPs. I've decided for a major project to design a digital audio effects proccesor, so that I can plug my guitar into it and have it add some effect in real-time and play it out. I'm having some trouble understanding how and where to do the actual processing of the audio signal. So far I've just been doing lots of reading, an...

I'm trying to plot the amplitude envelope of an audio signal, with perfect accuracy. (the algorithm for perfect accuracy would be to take...

I'm trying to plot the amplitude envelope of an audio signal, with perfect accuracy. (the algorithm for perfect accuracy would be to take the absolute value of the audio signal, then plot a point at the location of each local maxima, then connect adjacent points with a line) In Reaktor I've tried to use the envelope follower macro, but even after tweaking the settings it doesn't seem to plot th...

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!

Hi: Decreasing the pitch of the audio in an audio linear-PCM [wave] file means the sample-rate of the wave file can be decreased without...

Hi: Decreasing the pitch of the audio in an audio linear-PCM [wave] file means the sample-rate of the wave file can be decreased without causing aliasing. If the bit-resolution and # of channels [1 in mono, 2 in stereo] of the file are kept constant, then decreasing the sample-rate will decrease the file size. Adobe Audition allows the alteration of pitch without changing speed. ht...

Hi: Clipping in an audio signal results when an audio device receives a signal that is too loud. The audio signal distorts into...

Hi: Clipping in an audio signal results when an audio device receives a signal that is too loud. The audio signal distorts into square-waves because the "tops" of the signal are flattened. The device cannot handle power levels over a certain level. When this level is exceeded, clipping occurs. Clipping is usually harsher in digital devices than in analog devices. Analog clipping tends to b...

Hi, We are going to develop Audio application using TMS320C6416. For input to AIC23 Codec, we use microphone. But AIC23 Codec is unable to detect...

Hi, We are going to develop Audio application using TMS320C6416. For input to AIC23 Codec, we use microphone. But AIC23 Codec is unable to detect the signal coming from MIC. To give microphone input, I had done all the required setting i.e INSEL bit of Analog audio path control register of AIC23 Codec is set to 1. Is any other setting is required ? Please suggest. Regards Vood00

I have a WAV file which is a complex audio signal generated in PC music software. I would like to play it back at an increased speed of 100...

I have a WAV file which is a complex audio signal generated in PC music software. I would like to play it back at an increased speed of 100 times the original. For example, if the dominant frequency was 20KHz, the upper limit of audio, at playback it would be 2MHz. That is as high as I need to go. Obviously, this cannot be done, or output, from the PC alone. I assume some type of program...

Hi All, We are working to develop an Algorithm for Enhancing the Quality of compressed audio. We are thinking of having High...

Hi All, We are working to develop an Algorithm for Enhancing the Quality of compressed audio. We are thinking of having High Frequncy Reconstrucion. We are looking for the communities suggestion or experience on "High Frequency Reconstrucion of Compressed Audio". Will it Enhance the quality is the question? We also request the community to suggest some other means of Enhan...

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...

Hello, I just did a simple audio experiment and it failed... if you could comment on it and/or suggest some changes to make it work: i have...

Hello, I just did a simple audio experiment and it failed... if you could comment on it and/or suggest some changes to make it work: i have a walkman which is in very good condition. the output sound quality of hte walkman is excellent. i connected the output of the walkman to the computer mic socket using a simple audio sterio cord and recorded the sound using sound forge in a .wav file. N...