I am working on a project where I need to playback some sounds depending on specific inputs.
The system contains a BRAM containing pre-recorded audio samples (pronunciation of numbers from 1 to 10). From that memory, and depending on input dip switches (4 LSB bits), the appropriate sound sample is loaded from the memory and transferred to the audio output.
My target FPGA board is the Nexys 4 DDR, the IP will be integrated with a softcore that already uses the DDR2 Memory cannot use is to store the samples, I have to instaciate memory from BRAM. The softcore already uses many resources (65% of available BRAM is already taken).
1- I would like to ask for some guidance on how to convert a typical sound (.wav or mp3 or whatever) to an appropriate format that I can store (.Coe for example)?
2- Assuming the sounds take around 1 second, how much memory will I need?
3- What parameters should I use (output frequency ....)? Any pointers on how to compute that?
I have no background on audio processing but I am open to learn, so any pointers are appreciated.
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noureddine-as
Hello,
I am working on a project where I need to playback some sounds depending on specific inputs.
The system contains a BRAM containing pre-recorded audio samples (pronunciation of numbers from 1 to 10). From that memory, and depending on input dip switches (4 LSB bits), the appropriate sound sample is loaded from the memory and transferred to the audio output.
My target FPGA board is the Nexys 4 DDR, the IP will be integrated with a softcore that already uses the DDR2 Memory cannot use is to store the samples, I have to instaciate memory from BRAM. The softcore already uses many resources (65% of available BRAM is already taken).
1- I would like to ask for some guidance on how to convert a typical sound (.wav or mp3 or whatever) to an appropriate format that I can store (.Coe for example)?
2- Assuming the sounds take around 1 second, how much memory will I need?
3- What parameters should I use (output frequency ....)? Any pointers on how to compute that?
I have no background on audio processing but I am open to learn, so any pointers are appreciated.
Best regards.
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