Hi guys..where can i get the sourcecodes(matlab code) for different audio encoders??where can i find a good tutorial on present audio coding...

Hi guys..where can i get the sourcecodes(matlab code) for different audio encoders??where can i find a good tutorial on present audio coding progress in matlab????

How did the interviews for the audio engineering position go - did you find your dream job / candidate?

How did the interviews for the audio engineering position go - did you find your dream job / candidate?

Hi, I am working on a project where the output is a 24 bit audio signal. I am using an FPGA chip to generate this audio signal. The input to...

Hi, I am working on a project where the output is a 24 bit audio signal. I am using an FPGA chip to generate this audio signal. The input to the FPGA is a frequency value ( anywhere between 400hz to 10,000hz). Inside the FPGA, the frequency is converted to its amplitude by giving it to a sine look up table. The output 'A'is then manipulated in such a way( dividing it by 2) that the ampl...

I just watched a video on Microsoft's Channel 9 site about the new audio processing code that they are putting in Windows Vista (the...

I just watched a video on Microsoft's Channel 9 site about the new audio processing code that they are putting in Windows Vista (the next version): http://channel9.msdn.com/Showpost.aspx?postid=145665 They are changing from an "int16 pipeline" to a "float32 pipeline". They assert that using floating point for audio processing yields far better fidelity than int16, and also point out that...

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

Hello, I wonder if somebody would be able to help me with the manufacturer name for some easy to use audio DSP design software. The software...

Hello, I wonder if somebody would be able to help me with the manufacturer name for some easy to use audio DSP design software. The software in question was a windows based drag-and-drop design suite for some line of audio DSP cores, so you could basically drag a 'Equaliser' block onto the screen and set it's properties in a handy little window then join it to a 'Delay' block, for example, ...

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...

I'm working on a platform where the audio driver exhibits quite irregular callback patterns. My expectation was that if I configure the audio...

I'm working on a platform where the audio driver exhibits quite irregular callback patterns. My expectation was that if I configure the audio driver to give me callbacks every 10ms, I would get them approximately every 10ms (give or take some milliseconds). Ideally, the callback pattern would look something like this: t = 0ms : speaker callback t = 1ms : mic callback t =3

Hi: Is the following possible?: A linear uncompressed PCM audio file whose sample-rate is 44.1 KHz, monaural, and with a bit-rate of...

Hi: Is the following possible?: A linear uncompressed PCM audio file whose sample-rate is 44.1 KHz, monaural, and with a bit-rate of 1-bit-per-second. The 1-bit-per-second is because the bit-resolution [normally 16-bit in CD audio] is only 1-bit-per-44,100 samples in this hypothetical case. Since the sample-rate is 44.1 KHz, and there is only 1 channel [due to the monaural audio], the...

I have to do cross correlation of two audio file to prove they are similar. I have done FFT of two audio files and have the power spectrum value...

I have to do cross correlation of two audio file to prove they are similar. I have done FFT of two audio files and have the power spectrum value of both the audio file in a separate array.how should i proceed further to do cross correlation to prove they are similar.Any basic ideas will be helpful for me to learn and apply it.Thanks.

A new site on DSP audio that you might be interested in. http://aspress.co.uk/ds

A new site on DSP audio that you might be interested in. http://aspress.co.uk/ds

My goal is to implement a template structure on the DSK 6713. This template is to contain a structure for audio input/output, so that...

My goal is to implement a template structure on the DSK 6713. This template is to contain a structure for audio input/output, so that an algorithm can easily be fit into that template structure (for example a filter or a controller). To implement this a tutorial might be helpful, where step by step is explained how to send audio data to the line out/head phone jag and on the other way how to read...

Hi, below is a sample of the filter to process de-emphasis on audio signals on a receiver card with a ADSP2101. When i run the code without...

Hi, below is a sample of the filter to process de-emphasis on audio signals on a receiver card with a ADSP2101. When i run the code without this filter the linearity of the audio signals is good down to 80dB below full scale which is around -60dBm. So -10dBm input equals -10dBm output for good linearity and so on. What i notice is that when i add this filter to compensate from the pre-emphas...

France Telecom R&D is looking for a Ph.D. student in the field of speech and audio compression. This 3-year PhD studentship will be funded...

France Telecom R&D is looking for a Ph.D. student in the field of speech and audio compression. This 3-year PhD studentship will be funded by France Telecom. The subject of this thesis consists in exploring new techniques for audio coding allowing quality improvements, especially at low bit rates for both speech and music signals. Based on an analysis of existing codecs, and in strong relations...

Dear All !! **************************************************** Any shed of the Kowledge on this will help my me out ...

Dear All !! **************************************************** Any shed of the Kowledge on this will help my me out ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ I am working on the module in which i have to mix the two (audio/speech) files Its look simple to add the each samples of the two diffrent audio file and then write into the Mixed file. But here c...

http://www.celemony.com/en/start I have heard that this tool can split an orchestral audio into its component instruments. I was unaware that...

http://www.celemony.com/en/start I have heard that this tool can split an orchestral audio into its component instruments. I was unaware that we had the technology as yet but I think the trick is in how it is recorded with two mics near the front and two at the back or something like that.

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...

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 quality audio signals. I would like to hear from some one that what switch ADPCM is or get some u...

Hello everyone, it's been a while ago since someone asked here for an DSP eval board (for audio processing) that's got a SAR analog-digital...

Hello everyone, it's been a while ago since someone asked here for an DSP eval board (for audio processing) that's got a SAR analog-digital converter. Theoretical it should be possible to reach very low delay between an audio input and the output after processing by a DSP. Does anyone know a device with a fast SAR ADC (that could be used for audio processing)? A board with suiting anti-aliasi...

http://createdigitalmusic.com/2009/10/the-speaking-piano-and-transforming-audio-to-midi/ -- Randy Yates Digital Signal...

http://createdigitalmusic.com/2009/10/the-speaking-piano-and-transforming-audio-to-midi/ -- Randy Yates Digital Signal Labs http://www.digitalsignallabs.com

Hello everyone. I have to make a plugin that detects and analyzes the noise in it, in order to classify the audio tracks depending on...

Hello everyone. I have to make a plugin that detects and analyzes the noise in it, in order to classify the audio tracks depending on quality. What is the most efficient way to do this? Any suggestions? Thank you. --------------------------------------- Posted through http://www.DSPRelated.com