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Audio Equalizer

Started by akin...@hotmail.com in Audio Signal Processing16 years ago 2 replies

Hi I want build a multiband ( 10 band) 2-channel audio equalizer with FIR or IIR filters. Can you suggest any sample project or ready library...

Hi I want build a multiband ( 10 band) 2-channel audio equalizer with FIR or IIR filters. Can you suggest any sample project or ready library for that purpose ? Regards, Akin Ocal


Noise Cancellation using Auto and Cross-Correlation

Started by dwch...@gmail.com in Audio Signal Processing16 years ago

Hello, I'm trying to create an FIR filter to cancel some noise in an audio sample by looking back at the previous samples (some m samples back)...

Hello, I'm trying to create an FIR filter to cancel some noise in an audio sample by looking back at the previous samples (some m samples back) to create the filter weight for the current sample and then incrementing (similar to a queue where a new sample enters). I am using the auto and cross-correlation between the noise and the noisy signal to find the filter coefficients and not having ...


FIR & IIR : design,filtering

Started by akin...@hotmail.com in Audio Signal Processing16 years ago

Hello I am using some FIR and IIR filters. I use coefficients from Dsptutor's java applets. First of all , have you ever experienced Dpstutor's...

Hello I am using some FIR and IIR filters. I use coefficients from Dsptutor's java applets. First of all , have you ever experienced Dpstutor's designs on projects ? I am using fir and iir filtering functions from Paul Embree's book. I am building a PC\Windows application using Directshow and Winapi. In my application , I use these filtering functions with coefficients I already got , bu...


Data types in Audio Signal Processing

Started by akin...@hotmail.com in Audio Signal Processing16 years ago 1 reply

Hello I want to implement my own audio equalizer. First of all I create FIR& IIR bandpass filters but they process"doubl" or "float"...

Hello I want to implement my own audio equalizer. First of all I create FIR& IIR bandpass filters but they process"doubl" or "float" samples. But I get sound samples as shorts ( 16bit) from both Directshow and Waveout API. How can I handle data type conversion between floaats and shorts ? They say that i must divide shorts by 32768 to get floats. But how should I put my processed flo...


About Implementing A Sound Equalizer

Started by akin...@hotmail.com in Audio Signal Processing16 years ago 1 reply

Hello , I want to build a multiband equalizer. I know that I must create bandpass filters for that aim. So should they be FIR or IIR...

Hello , I want to build a multiband equalizer. I know that I must create bandpass filters for that aim. So should they be FIR or IIR filters and why ? Is there any sample code in C\C++ for building an equalizer ? Best Regards, Ak?n Ă–cal


Question on Sampling of signals

Started by Pradeep Hegde in Audio Signal Processing16 years ago 1 reply

hello all, I am answering a questionnaire and came across this question "When is the effective bandwidth of the signal Fs instead of...

hello all, I am answering a questionnaire and came across this question "When is the effective bandwidth of the signal Fs instead of Fs/2?" can someone please help me with that? Pradeep


Store audio data from DSK6711

Started by naam...@hotmail.com in Audio Signal Processing16 years ago 2 replies

Hi guys, Im implementing noise cancelling in DSK6711 using PCM3003. Could anyone tell me if there is any way i can store or save my filtered...

Hi guys, Im implementing noise cancelling in DSK6711 using PCM3003. Could anyone tell me if there is any way i can store or save my filtered audio speech directly from CCS and not from the PCM3003 output. Thanks


Equalizers

Started by sham...@gmail.com in Audio Signal Processing16 years ago 7 replies

Hi Group, Please tell can we connect filters in equalizers in parallel instead of connecting them in cascade. Because we can achieve same effect...

Hi Group, Please tell can we connect filters in equalizers in parallel instead of connecting them in cascade. Because we can achieve same effect in both the case(|| or cascade) with adjusting gains of individual filters. Different gains for parallel and cascade.


speech enhancement with the 6711 DSK and the PCM3033

Started by naam...@hotmail.com in Audio Signal Processing16 years ago 4 replies

Hi guyz This is my first DSP project and its a undergrad thesis project, so forgive my stupid question. Im implementing an adaptive LMS...

Hi guyz This is my first DSP project and its a undergrad thesis project, so forgive my stupid question. Im implementing an adaptive LMS algorithm to cancel background noise from a speech. my primary mic picks up a noisy speech and the secondary mic will pick up the reference noise. in my PCM 3003 theres only one input port. how can i connect two mics into the PCM. since this is real time so...


Need Long Term Assistance for Pro P.A. Line Array Crossover Calculations

Started by oldmics in Audio Signal Processing16 years ago

Hello New to the group. Did not even know this group of geeks exisited. Heres my story. Pro P.A. supplier here.Big regional rig line...

Hello New to the group. Did not even know this group of geeks exisited. Heres my story. Pro P.A. supplier here.Big regional rig line array system. No Harry hi fi stuff here! Want to explore some different filter uses/crossover points,etc. Have XTA/DP 448 processing capable of different above functions. http://www.xta.uk.com/dp428/Bevel%20Frameset.htm I have tweeked the sys...


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