Hi, I am in a process to improve the speech quality out of our current speech codec. Basically, our speech codec is an ADPCM codec with...

Hi, I am in a process to improve the speech quality out of our current speech codec. Basically, our speech codec is an ADPCM codec with bunch of interpolation and decimation filters and Sigma-Delta ADDA, and we have already adjusted all the filters so now it works just fine. However, we are still not 100% satisfied with the audio quality, we want to make the sound more clear so we are thin...

I have been working through the paper entitled "Audio Filter Banks" by J.O. Smith ( http://dafx09.como.polimi.it/proceedings/papers/paper_92.pdf...

I have been working through the paper entitled "Audio Filter Banks" by J.O. Smith ( http://dafx09.como.polimi.it/proceedings/papers/paper_92.pdf ) attempting to replicate the results presented in the paper. I am ultimately wanting to re-create the real signal non-uniform filter bank illustrated in Figure 11. My question has two main parts. The description for creating the complex non-uniform...

hello all, I am designing an application where I have an MP3 (music) decoder along with a speech encoder (not music quality). The encoder...

hello all, I am designing an application where I have an MP3 (music) decoder along with a speech encoder (not music quality). The encoder should preferably be able to store a file in audio format on a SD-card. I am looking to find a single chip / chipset with minimal number of components for my design. I did find some encoder/decoder chips from a few manufacturers (Micronas MAS3587F, VL...

Hi Guys, Need some explanation here, D = Depth et= envelope et = 1 + D sin (2*pi*fm*t); y(n) = et * sin (2*pi*fc*t) =( 1 +...

Hi Guys, Need some explanation here, D = Depth et= envelope et = 1 + D sin (2*pi*fm*t); y(n) = et * sin (2*pi*fc*t) =( 1 + D sin (2*pi*fm*t)) * sin (2*pi*fc*t) So, to get the tremolo effect, which of the following is the right one 1.Tremolo = ( 1 + D sin (2*pi*fm*t)) * audio signal or 2. Tremolo = ( 1 + D sin (2*pi*fm*t)) * sin (2*pi*fc*t) * audio signa...

Hi, We're implementing a VoIP-device for a university project. To handle the sound part of the device we're using a Blackfin 532 from Analog...

Hi, We're implementing a VoIP-device for a university project. To handle the sound part of the device we're using a Blackfin 532 from Analog Devices As a start we want to provide G.711a- and ulaw for the user and later if time permits a lower bitrate codec such as g.726. Our problem is that we're stuck in selecting an external audio codec. What we would like to see in an external au...

I'm referring to the Ultra Low Power DSP's customized for hearing aids and headsets, they are very small, very fast processors and use...

I'm referring to the Ultra Low Power DSP's customized for hearing aids and headsets, they are very small, very fast processors and use very little power (0.05mW/MIPS) and run on 1.2 volts, quite impressive here is one http://www.amis.com/products/dsp/belasigna_200.html CoolFlux in another one, but they only license the cores www.coolfluxdsp.com I'm just becoming familar with these t...

I have a question about PLESIOCHROUNOUS re-sampling of audio. It was mentioned in another thread that simply deleting or repeating a sample...

I have a question about PLESIOCHROUNOUS re-sampling of audio. It was mentioned in another thread that simply deleting or repeating a sample now and then is a poor solution because it adds clicks. It was mentioned that linear interpolation is not ideal. Why? Because it adds noise? or distortion? I presume the "ideal" digital solution is to use a higher order interpolation algo...

Hi everyone, For an audio project I'm starting from the setup of block based talkthru. Is there anyone who could give me more information how...

Hi everyone, For an audio project I'm starting from the setup of block based talkthru. Is there anyone who could give me more information how this program works? I'm connecting an audio input through the sharc and get the same output on the jack. So the program is working/running perfect. Another problem is where can I see my input/output variables? Can you display them in graph or window out...

Hi! I've been working on a program to take, say, a drum sound, then find it in another file (say, a song with that drum sound in it) by...

Hi! I've been working on a program to take, say, a drum sound, then find it in another file (say, a song with that drum sound in it) by comparing the frequency spectrum of the drum to various chunks of the song, so that it finds a close match of the spectrum charts. It compares everything in the frequency spectrum by using the Fast Fourier Transform. Once it knows where there is a du...

Hello, I recently got an ADUC831 microconverter board, and am interested in experimenting with DSP applied to an audio input and/or output....

Hello, I recently got an ADUC831 microconverter board, and am interested in experimenting with DSP applied to an audio input and/or output. A friend who is knowledgeable about these boards recommended "passing audio input through an op-amp to protect the board." His meaning appears to be that I should connect my 1/8 in. jack to an op-amp, and the op-amp to the ADC input, in such a way th...

Hi folks, On a TI c6713 DSP I have the QDMA sending a few Mbytes per second (a few kbytes per transfer) of data from SDRAM to a hard...

Hi folks, On a TI c6713 DSP I have the QDMA sending a few Mbytes per second (a few kbytes per transfer) of data from SDRAM to a hard drive periodically. Each QDMA data transfer lasts longer than a few audio processing interrupts. The audio processing interrupt is a great time for this transfer to happen since SDRAM is hardly accessed and therefore the QDMA transfer happens almost for fr...

Hello, if I make a sample rate conversion using multi-stages (e.g. 48 kHz downto to 44.1 kHz in 3 stages with the ratios 3/6, 7/4 and 7/5), do...

Hello, if I make a sample rate conversion using multi-stages (e.g. 48 kHz downto to 44.1 kHz in 3 stages with the ratios 3/6, 7/4 and 7/5), do I need different filter coefficients for the 3 filters or can I use always the same filter, when the audio data frequence range itself is limited to something like 22 kHz??? Thanks for your help! damc

Hi. I want to make a program which let's the user to select a certain word (or sound) from an audio recording and search it for other occurence...

Hi. I want to make a program which let's the user to select a certain word (or sound) from an audio recording and search it for other occurence of it. I am not interested to make the program to recognize that word, but only to find other sounds(words) from the recording which resembles the selected portion of the sound. Please tell me how could I do this. Thank you very much. Paul

Hi all, I'm looking for WMA audio streams with differrents combination of bitrate/sampling rate and I canno find them. I need these...

Hi all, I'm looking for WMA audio streams with differrents combination of bitrate/sampling rate and I canno find them. I need these combinations because I want to check a particular player supports the L3 Profile. Also I would like to understand which are the supported streams (in term of bitrate/sampling rate) supported by WMA Std. Can anybody help ? Thanks in advance, Alberto

Hi! I wonder how to interpret the data after applying a Short Time FFT (fftw3 for real data) on audio samples (sinus tone 1400Hz). My goal...

Hi! I wonder how to interpret the data after applying a Short Time FFT (fftw3 for real data) on audio samples (sinus tone 1400Hz). My goal is to visualize the Energy in the frequency bands. First, I do a 512 point FFT on the whole audio file (frame shift without overlap) and the highest value after computing [1] re*re + imag * imag is about 9997 in a fft bin. Second, I do a 1...

Dear all, As I know, the Hilbert-Huang transform (HHT) (http://techtransfer.gsfc.nasa.gov/HHT/) is a powerfull tool declared by NASA. Compared...

Dear all, As I know, the Hilbert-Huang transform (HHT) (http://techtransfer.gsfc.nasa.gov/HHT/) is a powerfull tool declared by NASA. Compared to Fourier transform (FFT), HHT is capable of analyzing time-varying processes, nonlinear & nonstationary signals that are in a real world. It seems to be interesting if we can use it in audio signal processing like identification, noise cancellation......

Hi, I need help to understand a case of EDMA lockup because of memset() of huge memory size. I have a audio playback task which does a...

Hi, I need help to understand a case of EDMA lockup because of memset() of huge memory size. I have a audio playback task which does a SIO_issue() and SIO_reclaim() on the audio buffers. This task is running with the highest priority. The EDMA dispatcher keeps calling the ISR once the EDMA has completed the transfer. Hence SIO_reclaim is successful. Now this works fine. But if a low prio...

Hi all, I'm strugling making the right choice for our new board. I Need something like 2 GFLOPS in order to process audio. Audio is sampled...

Hi all, I'm strugling making the right choice for our new board. I Need something like 2 GFLOPS in order to process audio. Audio is sampled with 44Ks/s externally and internally at 10xFS. I have been aiming at the ADSP-21161 or ADSP-21261 as a target. I know there are 2 ways of processing; Parallel and serial. I have no clue how to create a parallel system ( bus arbiter, software ). Doe...

Hi community, I'm developing audio algorithm for speech enhancements. Unfortunately the BF533 EZ-Lite is not supporting this sampling rate. So...

Hi community, I'm developing audio algorithm for speech enhancements. Unfortunately the BF533 EZ-Lite is not supporting this sampling rate. So Analog told me to use the "Audio Extender BFAUD-EZEXT". Actually there is only a BF537 driver for this extender, but this is not usable for the BF533. Has somebody a driver to setup a AD1938 codec on the extender board? Cheers Michael

Does anyone have experience or a viewpoint on these training materials? "Apple Pro Training Series: Logic Pro 7 and Logic Express 7 Veteran...

Does anyone have experience or a viewpoint on these training materials? "Apple Pro Training Series: Logic Pro 7 and Logic Express 7 Veteran audio producer Martin Sitter uses step-by-step, project-based instruction and straightforward, jargon-free prose to guide you through the countless creative options Logic affords for audio production." It's here and cost $45-ish: http://tinyurl...