Adaptive Filter | ADPCM | ADSP | ADSP-2181 | Aliasing | AMR | Anti-Aliasing | ARMA | Autocorrelation | AutoCovariance | Beamforming | Bessel | Blackfin | Butterworth | C6713 | CCS | Chebyshev | CIC Filter | Circular Convolution | Code Composer Studio | Comb Filter | Compression | Convolution | Cross Correlation | DCT | Decimation | Deconvolution | Demodulation | DM642 | DSP Boards | DSP/BIOS | DTMF | Echo Cancellation | Equalization | Equalizer | ETSI | EZLITE (Ez-kit Lite) | FFT | FFTW | FIR Filter | Fixed Point | FSK | G.711 | G.723 | G.729 | Gaussian Noise | Goertzel | GPIO | Hilbert Transform | IFFT | IIR Filter | Interpolation | Invariance | JTAG | Kalman | Laplace Transform | Levinson | LPC | McBSP | MIPS | Modulation | MPEG | Multirate | Notch Filter | Nyquist | OFDM | Oversampling | Pink Noise | Pitch | PLL | Polyphase | QAM | QDMA | Quantization | Quantizer | Radar | Random Noise | Reed Solomon | Remez | Resampling | RTDX | Sampling | Sharc | TI C6711 | Undersampling | Viterbi | Wavelets | White Noise | Wiener Filter | Windowing | XDS510PP | Z Transform

Re: Mu-Law Algorithm

On Sun, 05 Feb 2006 09:00:37 -0600, "vikaspatil123" wrote: > Hi, > > I am doing project named Implementation of Mu-law Speech companding on > TMS320vc5510 using DSK5510 board. > > I need help regarding algorithm for mu-law My data is 16-bit samples that > i am getting from reading .W...

Merging G.711 files

I have two G.711 files/streams and I want to merge them. They represent each side of a telephone conversation. I know I can create a 'stereo' .wav file but I would rather create a single channel (mono) file. Can I just add together each contemporaneous data point? If not, how should I do this....

Do I really need a DSP? If yes, how to choose one?

Hi everybody, For our next project, we need to develop & engineer a VOIP gateway which must support 8 radio channels. The codec is G.711 . The requirement for the latency of voice on the gateway is 1/4 seconds. My gut feeling says that we can accomplish this with by using a 8 channel professional...

G.711 implementation

Hello. I have read the G.711 spec from ITU and I have seen some implementations written in C. There is something I don't understand because I cannot understand it from the spec and the different implementations I have seen produce different results! So, my question is in the bit ordering an...

Coding distortion for a G.711 codec.

I have implemented a G.711 encoder and decoder over the StarCore 1400. I need to determine the distortion produced by the conversion of linear 16 bit samples to 8 bit non linear PCM (encoding) and back to 16 bit linear samples (decoding). I was looking for a standard look-up table or something...

Re: Echo Cancellation

> I had a discussion with a collague about echo cancellation. > > He claims that an echo canceller during double-talk is able to > completely remove the far-end speech component in the microphone > signal even if the far-end speech component occupies the same > frequency as the near-end speech...

vocoder transcoding question

hello, i am endeavoring to transcode G.729 r8 to G.711 as my first serious DSP project for a production environment. I am seeking any relevant resources including how-tos, reference docs, tutorials, working code, ..annnything I am fairly new to DSP and don't really have much of an idea of ho...

Re: ADPCM g.722 old ADI Code

The book "C Algorithms for Real-time DSP" by Paul Embree includes code for g.711 and g.722. I think it also comes with code for the ADSP-21020/21062. I can't find my copy right now so I'm not 100% sure, but if you still need it email me privately (smb [AT] dspdimension [DOT] com) and I'll search...

audio transcoding

Hi all, I was wondering if anyone can give me some information (links, forums, possible books) on audio transcoding topic. I've been assigned this task and I need a starting point. I'm generally interested in audio transcoding but more specific in G.729.A or G.729.AB to G.723.1A transcoding....

Need info on speech CODECS

Hi, I work for a Multimedia startup company. we are luking for speech codecs implemented on ARM9 platform for our end system.Can any one give some info abt good companies who does Porting of speech codecs. we need ITU G.723.1 or 3gpp AMR-NB and G.726, G.711. Also cud any tell me how much ...

Wimax Baseband Algorithms

Hi, How complicated are Wimax baseband algorithms as compared to voice algorithms (eg G.711). I am new to wimax and need to have an estimate of the time required to code and optimize wimax baseband algorithms. Walter ...

Re: A-law, u-law equations

On Wed, 09 Nov 2011 00:21:12 -0800, Daku wrote: > Could some DSP guru please direst me to a good online reference fro > A-law, mu-law -- especially the equations for companding, de-companding. > My old DSP text does not have it, and those on Wikipedia do not look > authentic. > Any help ...

Re: System Identification

On Wed, 09 Jul 2008 19:49:09 -0500, Vladimir Vassilevsky wrote: > The FDLS worked wonderfully. > Pretty accurate results from the first try. By the way, any FDLS solution that has both poles and frequency response measurement errors will be biased. You might try the Total Least Squares...

Comfort Noise Generator

Dear Experts Well I've read a few previous posts on this aspect. The documents or references I used were: RTP Payload for Comfort Noise RFC3389. Appendix II of G.711 Annex B of G.729 Well my query is regarding implementation of a codec independent CNG: 1) Which excitation is best for ...

Re: Combining G711 streams

Adrian Hey wrote in message news: ... > Erik de Castro Lopo wrote: > > > kevinmr wrote: > > > > > > Hi, I have two streams of G711 coming into my process. I have a need > > > to combine the streams and write the result to a file. Is this > > > possible? Any ideas would be ap...

Re: Definition of dBOV

Ben Bradley wrote in message news: ... > Back to quoting Steve Underwood: > > > That is defined as 8 specific samples > > which will produce a 1kHz sine wave that is considered 0dBO. The clip > > point turns about to be about +3.14dBO for a sine wave. I would think > > That...

Re: LINEAR PCM>> u/A law standard I/O files.

jadhav_rahul schrieb: > Hello all, > I have a code of linear pcm to u/A law conversion and viceversa > to simulate on CCS, and i want to test it with standard input sequence and > verify the output with expected result value. where can i get the standard > test input and output ta...

Re: Image and Video References

Randy Yates wrote: > Hi Folks, > > I'm looking for authoritative and in-depth on-line or hardcopy > references to the following topics: > > 1. The YUV color scheme and the reason for the choices in > its various bit-depths (i.e., 6-2-2, etc.), how it maps > to RGB, and any ot...

Re: Nearest IIR System to FIR?

Andor wrote: > > Hahahahahaha ... if you only knew ... ;-) > > I'm all ears! I'm really an artificial intelligence program running on an IBM PC XT. > > If you have ideas about how to reformulate using the infinity-norm, I > > encourage you to pursue them! > > I have. Just rec...

Re: Acoustic echo cancellation

> On Sep 4, 8:32=A0am, Vladimir Vassilevsky wrote: > > createdon2003 wrote: > > > may be it will be more clear by a diagram. > > > > > --far_end-------------------|ENC|-------|DEC|--------// > > > =A0 =A0| =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 = > =A0 =A0 =A0 =...

Re: ITUT G729 preprocessing question

> > > HyeeWang wrote: > > > About ITUT G729, I have two questions. > > > > 1. In the processing section of G729 spec, it says to apply a high > > pass filter with a cut-off frequency of 140 Hz. As we know, the > > range of pitch is between 55 and 400 Hz. Then, how to search the p...

Re: Modem question

> On Wed, 8 Sep 2010 00:33:45 +0000 (UTC), glen herrmannsfeldt > wrote: > > > Eric Jacobsen wrote: > > > On Tue, 07 Sep 2010 19:10:08 -0400, Randy Yates > > > > > or FDD within the capabilities of the wire. > > (snip) > > > > > > Couldn't you use cancellation techniques? > > (s...

Choosing external audio codec for VoIP-solution using a Blackfin

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 t...

DSP contractor position in Sydney, AU

Hello to everybody, I apologize for getting off topic but it looks like this is the only place you can meet an experienced DSP engineer, the jobs forums are very generic and partially hopeless. I am deeply sorry. We have a position for DSP contractor (6 months to 1 year contract) in Sydn...

Re: Newbie Q: Goertzel threshold testing, speech detection

Zack Angelo wrote: > Hi, > > I'm currently using the Goertzel algorithm to do DTMF detection. > Currently, I get the magnitude for each DTMF frequency, save the sums > of the magnitudes that correspond to each number, then I assume > whichever sum is the highest must the be the most l...

Re: VOIP Clock rate adaption

Rocky wrote: > We currently have an E1 channel bank but have found that some of the > available E1 cards for Asterisk will lock up to the extent that a > power down reset is required to get it up again. Also, the E1 cards > are relatively expensive so we are exploring the idea of using a LAN...

Multiple openings for Telecom

Dear All Many of our esteemed MNC clients are looking out for Oracle professionals with multiple positions opening. Please go through the list below and forward your updated word document CV if any of the profile matches your requirement. Please email your CVs to sandhya@amartyaplacements.com...

Multiple openings for Telecom

Dear All Many of our esteemed MNC clients are looking out for Oracle professionals with multiple positions opening. Please go through the list below and forward your updated word document CV if any of the profile matches your requirement. Please email your CVs to sandhya@amartyaplacements.com...

Re: Wavforms and principle component analysis

I know that this is supper annoying but I have no other idea of how to share some examples. Wave1 Wave2 Wave3 Wave4 Wave5 Wave6 Wave7 1259 1267 949 1324 1154 1245 1191 736 666 0 736 1429 638 281 0 69 649 76 69 62 0 767 783 36 833 467 763 558 0 0 213 0 380 3 111 425 454 510 467 0 427 154 ...

comp.dsp FAQ [2 of 4]

Archive-name: dsp-faq/part2 Last-modified: Wed Apr 11 2007 URL: http://www.bdti.com/faq/ Previous section (1) Next section (3) Q2: Algorithms and standards Q2.1: Where can I get public domain algorithms for general-purpose DSP? Updated ...

comp.dsp FAQ [2 of 4]

Archive-name: dsp-faq/part2 Last-modified: Mon May 8 2006 URL: http://www.bdti.com/faq/ Previous section (1) Next section (3) Q2: Algorithms and standards Q2.1: Where can I get public domain algorithms for general-purpose DSP? Updated ...

comp.dsp FAQ [2 of 4]

Archive-name: dsp-faq/part2 Last-modified: Wed May 3 2006 URL: http://www.bdti.com/faq/ Previous section (1) Next section (3) Q2: Algorithms and standards Q2.1: Where can I get public domain algorithms for general-purpose DSP? Updated ...

comp.dsp FAQ [2 of 4]

Archive-name: dsp-faq/part2 Last-modified: Thu May 4 2006 URL: http://www.bdti.com/faq/ Previous section (1) Next section (3) Q2: Algorithms and standards Q2.1: Where can I get public domain algorithms for general-purpose DSP? Updated ...

comp.dsp FAQ [2 of 4]

Archive-name: dsp-faq/part2 Last-modified: Tue Oct 19 2004 URL: http://www.bdti.com/faq/ Previous section (1) Next section (3) Q2: Algorithms and standards Q2.1: Where can I get public domain algorithms for general-purpose DSP? Updated 1...

Re: IIR design and stability

On Sun, 04 Jan 2004 05:40:32 +0000, Martin Eisenberg wrote: > > Thanks for reading ! I would like to design a 1st order digital > > high pass filter with a 50 us time constant. The analog transfer > > function is Ha(s)=50E-6*s/(1+50E-6*s). > > > Assuming that by "pre-warped pulsation"...