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ask for help on Hi Fidelity Audio Compression with ADPCM

Hi all, Currently, I am studying on how to get Hi Fidelity effects with ADPCM related compression methods. But it seems that the traditional IMA ADPCM and MS ADPCM will not get the satisfied results. Any one can give me some suggestions? 16 bit PCM to 5 bit PCM will be accepted by m...

ask for help on Hi Fidelity Audio Compression with ADPCM

Hi all, Currently, I am studying on how to get Hi Fidelity effects with ADPCM related compression methods. But it seems that the traditional IMA ADPCM and MS ADPCM will not get the satisfied results. Any one can give me some suggestions? 16 bit PCM to 5 bit PCM will be accepted by me. ...

adpcm code

Hi There, I have been searching for the Matlab code for 40, 32, 24, 16 KBits/s ADPCM code (G.726) for the last one week but could not find a good working/ compliant code. Can anyone tell me where to get free version of the ADPCM code in Matlab Thanks, jagadeesh ...

ADPCM Matlab Code

Hi There, I have been searching for the Matlab code for 40, 32, 24, 16 KBits/s ADPCM code (G.726) for the last one week but could not find a good working/ compliant code. Can anyone tell me where to get free version of the ADPCM code in Matlab compliant to the G.726 standard? Thanks, Farha...

switched ADPCM

Hi All, I am preparing to develop an real time audio compression codec for high quality audio signals. Due to the heavy calculation burden, I will not adopt MP3 as my compression algorithm. I get from google search engine that there is an algorithm called switched ADPCM that can code high q...

number of ADPCM channels in E1 ???

Hi fellows im newcomer . could anyone reply to my Question please .... How many ADPCM channels are transmitted in E1 (2mb) stream if the bitrate is 40kbps , 32kbps , 24kbps, 16kbps Is their any standard frame structure for it as their is for PCM . Kindly help me out With regards Sahar ...

Interpolated ADPCM

Hi all, I am developping an real time audio compression codec system.In my system I adopted 16-bit 48kHz to 5 bit 48kHz IMA-ADPCM algorithm,but could not get the satisfied results,audio result was not good. I know from someone,there is an tech call "interpolated" will do the job. I would li...

Re: ADPCM Wav File Specification

dbell wrote: > What I am looking for is the algorithm description including data > packing format for > > WAVE_FORMAT_MEDIASPACE_ADPCM (0x0012) All I can offer you is good luck as that is one of the many ADPCM algorithms I have never seen documentation or code for. However, if you ...

ADPCM g.722 old ADI Code

There is some g.722 ADPCM code in the Digital Signal Processing Applications Using the ADSP-2100 Family book. Does anyone know if it actually works? We are translating the code into a newer DSP and we think we have found some mistakes. Can anyone point to some working sample code, examples,...

G.722 ADPCM Help

We have an application that uses a sub band ADPCM (G.722) coder and decoder. Our goal is data reduction with good quality, minimal computation overhead and extended speech bandwidth. We chose G.722 because we didn't want to reinvent the wheel and it appeared to be a good candidate. We have ...

Re: WAV to PCM conversion

Michel Rouzic wrote: > > When you're sticking to the PCM WAVE, it's quite simple, isn't it? No. > I mean you have a fixed 44 bytes of tags, right? No. I ac supply you with a vaild file where this is not the case. > and you only have 8-bit > unsigned, 16-bit signed and 32-bit IEE...

.wav-> G.723

Hello, I am a beginner and trying to understand the working of G.723 ADPCM. I have been able to get all the C files from ITU for the G.723 ADPCM. I have a basic question. According to the documentation and block diagram the first block converts the input signal from linear or A-law or U-l...

Re: console based gsm encoder

rg wrote: > > Hi all, > > Can anyone tell me whether there is a CLI / Console / Command Prompt based > gsm encoder (pcm wav to gsm) for the windows platform. Currently, I am using > a program called AVS AudioConverter, but it has a graphical interface and I > need to automate a proce...

distinguishing file formats in libsndfile

Hi, I was using libsndfile to read files encoded in g.726(is it alright to do so?) and write them to an oki adpcm at 6k. the problem is that libsndfile only accepts a combo of au and g.721 and my encoded file are headerless. any ideas about how to work around this problem? Thanx in advance...

Re: Convert 4 bit to 8 bit audio

Jerry Avins wrote: > clayton.aa@gmail.com wrote: > > > How can I converto a 4 bit/sample raw audio file to an 8 bit/sample > > file? > > Extend each nibble to a byte. Do you need to be told how? Jerry, your answer may just be that simple, but I have never seen a 4 bit per sample f...

Re: Audio Artifacts: WAV vs. WMA

> So it is possible to have a lower bit-rate [20,000 bits per second] > with a higher sample rate [44.1 KHz]. Then wouldn't it be possible to > have WMA with a bit-rate of 1 bit per second with a sample-rate of 1 > GHz? No, there are limits to how much compression you can get. Audio comp...

WAV file header corrupted - help needed

Hi, I'm in a bit of trouble with a WAV-file. I spoke at a symposium earlier this year and recorded my talk with a MP3 player. However, to my devastation I later found out that the resulting WAV file is corrupted and will not play, and somebody thought that it is because the header has a fault. No...

G.729 with different sampling rates?

I'm trying to understand G.729. The only compression algorithm I've coded before is ADPCM, and it wasn't keyed to a certain sampling rate - that is, it would be just as happy with 8100 samples per second or 7900 samples per second as it would have been with 8000, just the quality would be slight...

Re: Linear PCM audio: 44.1 KHz, monaural, 1-bit-per-second

On Sun, 22 Jul 2007 22:00:48 +0200, andre wrote: > Radium wrote: > > what would such audio sound like? Bad-quality? > > tic tac tic tac ..... Or some variation on that. Doesn't have to be a boring as a square wave. Could be encoding centre frequency of a two-tone oscilator (ambulance...

[Q]Audio processing technique to increase speech quality?

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 s...

[Q]Audio processing technique to increase speech quality?

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 s...

Re: Audio Mixing using DSP

On Jul 7, 8:35=A0am, "jadhav_rahul" wrote: > Hello everyone, > =A0 =A0 =A0 =A0 =A0 =A0 =A0 i want to implement audio mixing for call con= ference with > PCM encoded voice channels, > I want to know how audio mixing is done by DSP ? =A0and any suitable > algorithm or document for it. >...

VDVI and DVI4

Hi all, RFC3551 defines the way in which the output from a number of codecs should be packed into RTP packets. Amongst these is the DVI ADPCM codec, commonly found on PCs. They refer to this as DVI4. Also, there is VDVI, which is a kind of run length encoded variable bit rate version of DVI...

Re: ITUT G729 preprocessing question

steveu,thank you. 1. Removing DC is reasonable,for it is not a part of speech at all.But what is the reason to remove bass? How and why it can optimise the performance ? We should be to noted that G729 is not a waveform coder,not a frequency domain waveform coder also, it is a paramete...

Re: Why can Matlab wavread() not read AAC .wav reference files?

Ah, that's where your understanding is incomplete. PCM is only one of many forms in which audio data may be stored in such a file. Other forms are A-law and mu-law 8-bit logarithmic compression, several ADPCM algorithms, A-law and mu-law 8-bit logarithmic compression, and others. I don't have...

Re: Mu-Law Algorithm

> Robert, there is no point to argue with the offended individual. It only > makes difficult to the other folks to distinguish you and him. Vladimir, but it's easy to distinguish you from a speech coding expert... :) > Precisely. The differentiation of the signal to compensate for the >...

Re: Channel Coding: approximation instead of correction

dudelmann wrote: > Hi all > > I have a question regarding > channel coding for the wireless channel. > > All the codes I know have error detection and (some) error correction > capabilities. So far so good. But for audio transmission I don't really > need to get the exact trans...

Re: How to account for filter delay

On Jul 3, 4:44=A0pm, Piergiorgio Sartor wrote: > Jerry Avins wrote: > > Do your samples have a variable number of bits? How does that work? > > Huffman coding, maybe? > > bye, > > -- > > piergiorgio More like this. Say you have 16 bit samples at 8Ksps for input. You u...

Re: GSM 6.10 Freq Other than 8000

Steve Underwood wrote: > Vladimir Vassilevsky wrote: > > > "KWhat4" wrote in message > > news:1187845207.085353.67250@e9g2000prf.googlegroups.com... > > > > > I have been playing around with the GSM 6.10 codec and noticed that > > > the source code i found does not support an...

Re: ping: Jim Thompson

On 4/25/06 9:56 AM, in article e2lkbv$nt2$1@nnews.pacific.net.hk, "Steve Underwood" wrote: > Don Bowey wrote: > > On 4/25/06 12:14 AM, in article e2ki8j$cuh$1@home.itg.ti.com, "Steve > > Underwood" wrote: > > > > > > > Joerg wrote: > > > > > > > Hello Jim, > > > > > >...

Re: QMF banks in G.722

Thanks, Steve. Please allow me to specify my simulation here. +-----+ +------------------- + +--------------------+ +----+ +-----> | H0 |----> |Decimation by 2|----> | Interpolation by 2 |----> | G0 |------+ +-...

Re: I am back

On 1 Oct 2005 19:45:56 -0700, "Radium" wrote: > dpierce@cartchunk.org wrote: > > > WAV is simply a specific container format for audio information. > > WAV files can hold linear PCM, non-linear PCM such as A-law or > > u-law, encodings such as MPEG, AC-3 and such. > > > > Thus, if ...

Re: Compress/Decompress Wave file to MP3 and vice versa

wrote in message news:1145261306.489910.204460@u72g2000cwu.googlegroups.com... > Hello. Sorry to disturb you all. I hope to get your opinion. For your > information, I have a Visual C++ 6.0 software to record and playback > human speech. All the speech is recorded with sampling rate of ...

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 =A0 =A0 =A0 =A0...

Re: Audio programming - WAV data chunk format issues

alessandro.saccoia@gmail.com wrote: > > > - read the original 44 bytes WAV header and I copy it to the output > > > file > > > > Are you sure the wave files are 16 bit PCM and the total header is 44 bytes? > > Yes I am. Here you go > > /********* WAV BigEndian File Version *****...

Re: Floating Point Data Compression?

On Fri, 07 Nov 2008 00:10:26 -0600, Tim Wescott wrote: > On Thu, 06 Nov 2008 14:19:42 -0800, DigitalSignal wrote: > > > Hi there, A quick question: Is there any way to compress the single > > point floating point data? Apparently most of the research and > > development work focuses o...

Re: New Digital Audio Standard?

Hi Everyone > It also reduces the input slew rate. This doesn't matter if the > system is perfectly linear, but do we know that it is? > > Regards, > Allan. As long as this thread is still active and talking about new audio standards, this might peak some interest. Interest...

Re: What does PCM mean to you?

On Tue, 04 Oct 2005 17:41:34 -0700, Paulo Castello da Costa wrote: > Jon Harris wrote: > > "Jerry Avins" wrote in message > > news:sMydnckE6OIQFN_eRVn-jg@rcn.net... > > > > > Matt Timmermans wrote: > > > > > > > These days, PCM is the name given to the common uncompressed > > > >...

Re: How can digital be more spectrum efficient than analog ?

On 10/27/06 8:36 PM, in article kmg5k21ljhik3sgfecg6br34u2llkko0cn@4ax.com, "Howard Eisenhauer" wrote: > On Fri, 27 Oct 2006 10:29:06 -0500, no-top-post wrote: > > > It's common knowledge that digital technology gives more telephone > > [4 Khz wide] channels than analog technology - f...

comp.dsp FAQ [1 of 4]

Archive-name: dsp-faq/part1 Last-modified: Mon May 8 2006 URL: http://www.bdti.com/faq/ FAQs (Frequently asked questions with answers) on Digital Signal Processing The world-wide web version of the comp.dsp FAQ is maintained and sponsored by Berkeley Design Technology, Inc. For inf...