Hi

Started by ap1234_singh in Audio Signal Processing17 years ago

Hi all I am working on MP3 decoder for a 16 bit Fixed pt DSP. I am having trouble in dequantizer implementation please let me know in case...

Hi all I am working on MP3 decoder for a 16 bit Fixed pt DSP. I am having trouble in dequantizer implementation please let me know in case there is any information or implemenation available. Thanks and Regards, AP


Re: Why is this group so silent?

Started by Eliot Blennerhassett in Audio Signal Processing19 years ago 1 reply

On 5 Sep 2000, at 21:57, Krishnakumar G wrote: > Hi friends, > Can anyone tell me a way to understand the quantization block found in the >...

On 5 Sep 2000, at 21:57, Krishnakumar G wrote: > Hi friends, > Can anyone tell me a way to understand the quantization block found in the > MP3 standard in some detail.Please give me some pointers to some material you > know. Please be more specific. Do you want to know about the encoder or decoder? Do you have a copy of the standards document? Eliot ===================


Bandpass filters for color organ

Started by Ord Millar in Audio Signal Processing8 years ago 5 replies

Hello all, I am working on a project that plays MP3 or WAV files, and part of it involved having LED's that vary in intensity according to the...

Hello all, I am working on a project that plays MP3 or WAV files, and part of it involved having LED's that vary in intensity according to the audio level in 4 bands. The bands are to be adjustable by the user during operation. There is no need for great precision, but I will need to keep the bands relatively constant even if a file at a different sample rate is played. I have read ...


Synthesis Filter in libmad

Started by Karthika in Audio Signal Processing12 years ago

HI all, I am studying the libmad code for mp3 decoder and I have a doubt in the file ?synth.h?. I could not understand the ?filter? element...

HI all, I am studying the libmad code for mp3 decoder and I have a doubt in the file ?synth.h?. I could not understand the ?filter? element of the structure ?mad_synth?. It says... struct mad_synth { mad_fixed_t filter[2][2][2][16][8]; /* polyphase filterbank outputs */ /* [ch][eo][peo][s][v] */ unsigned int phase; ...


Frequency inversion

Started by raj raj in Audio Signal Processing14 years ago

Hello everybody, Iam implementing MP3 decoder. I would be very happy if any body gives answer for 1. What is frequency...

Hello everybody, Iam implementing MP3 decoder. I would be very happy if any body gives answer for 1. What is frequency inversion 2. Why frequency inversion is done before subband sythesis Thanks Regards Seenu


Minimad output

Started by Karthika in Audio Signal Processing13 years ago

HI all, I am trying to decode an mp3 bit stream using the library libmad. I am using the program "minimad.c". But, I am not knowing where to...

HI all, I am trying to decode an mp3 bit stream using the library libmad. I am using the program "minimad.c". But, I am not knowing where to look for the output data i.e the output PCM samples. I am interested in printing the PCM samples.If anyone has any idea please help me. Thanks and regards, Karthika .


Near-realtime lossy compression of many PCM streams

Started by Iain Surgey in Audio Signal Processing9 years ago

Hi, I want to build a server that can compress many different PCM streams simultaneously. This laptop can encode mp3 at 18x the speed of...

Hi, I want to build a server that can compress many different PCM streams simultaneously. This laptop can encode mp3 at 18x the speed of playback, which theoretically means it could encode up to 18 different streams in near-realtime (in reality this would probably be lower). With faster hardware this would improve, perhaps to the region of 50 streams. However I want to be able to handle ...


building LibMpeg3

Started by YOSHITA NANDA in Audio Signal Processing14 years ago

Dear All, I am doing masters project on audio classifcation system. A freely available package LibMpeg3 was downloaded by me from the web,...

Dear All, I am doing masters project on audio classifcation system. A freely available package LibMpeg3 was downloaded by me from the web, required to extract mp3 audio stream. As written in doucmentation it is required to run make in the LibMpeg3 folder. I cannot build the library as yet as its too confusing. Has anyone used the library before and can set guideline how it can be used.


Time warping of audio signals

Started by raja...@yahoo.com in Audio Signal Processing14 years ago 2 replies

Hi, I'm working on implementing Trick Modes (fast forward, rewind etc) in MP3. In this context, I need to play audio at higher speeds (2x,...

Hi, I'm working on implementing Trick Modes (fast forward, rewind etc) in MP3. In this context, I need to play audio at higher speeds (2x, 4x). Skipping audio samples distorts the sound (it sounds very squeaky). Is there any good way of doing this? I'm assuming time warping is what I need. If yes, could anyone suggest a good article on this? Thanks, M


how to drive pcm samples directly to speaker

Started by raghurh in Audio Signal Processing17 years ago 1 reply

Hi all, I have developed MP3 Audio Decoder on windows NT platform . Previously i used to put the ouptput of Decoder i.e, PCM samples into...

Hi all, I have developed MP3 Audio Decoder on windows NT platform . Previously i used to put the ouptput of Decoder i.e, PCM samples into the file and play it through the Cool Edit software. Now i am trying to play the PCM samples directly through the speaker.Eeven though i have suceeded to quite an extent, i am facing some problems which are as follows: Issues encountered when a


AC3 bitstreams

Started by Sameer Kibey in Audio Signal Processing15 years ago 1 reply

hi all can anyone tell me from where i can get either 1. AC-3 bitstreams or 2. an AC3 'encoder' or 3. mp3/wav/wma/pcm to "AC3"...

hi all can anyone tell me from where i can get either 1. AC-3 bitstreams or 2. an AC3 'encoder' or 3. mp3/wav/wma/pcm to "AC3" converter basically i am in need of an AC3 bitstream or some utility to generate AC3 bitstreams. thanks and regards, Sameer --- "To understand recursion you must first understand recursion."


filter bank : design n application

Started by Sanjay S in Audio Signal Processing14 years ago 1 reply

I'm currently doing my major project on Non-uniform filter bank theory and application. I need your help and suggestions based the design of a...

I'm currently doing my major project on Non-uniform filter bank theory and application. I need your help and suggestions based the design of a non-uniform cosine-modulated filter bank (a 27 or 32 channel filter bank)and find it's application in the existing MPEG-1, Layer-III model ~ "MP3" audio codec, thereby replacing the hybrid filter bank which is employed in the existing one.


Problem in decoding g729 audio samples

Started by krav...@gmail.com in Audio Signal Processing11 years ago 1 reply

My aim is to dump the G.729 encoded audio samples packetized in RTP payload and then convert the G.729 encoded samples to MP3 format. What i'm...

My aim is to dump the G.729 encoded audio samples packetized in RTP payload and then convert the G.729 encoded samples to MP3 format. What i'm doing: 1. I got the G.729 encoded audio samples from RTP payload and saved as g729audio.ul on my machine using my c programm. 2. I'm using SOX as my converter to convert g729audio.ul to audio.wav(wav format). 3. Then, i'm using bladeencoder to co...


unpredictability equation in psychoacoustic model 2

Started by sandeep hublikar in Audio Signal Processing17 years ago 2 replies

Hi all, Can anybody tell me the Theory and derivation behind the?? unpredictability equation in psychoacoustic model 2 used in...

Hi all, Can anybody tell me the Theory and derivation behind the?? unpredictability equation in psychoacoustic model 2 used in mp3 audio encoder ? Thanks in advance. Regards Sandeep?R Hublikar