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I have a 20 second-long WAV file (sample rate = 48kHz, bit-depth presumed to be 16-bit) of a 1 kHz sine wave recorded at -20dBFS. We wish to use it to check signal levels through our Axia broadcast console. Most of our sources are only 44.1kHz sources derived from CDs either directly or by ripping with dBpoweramp to FLAC files for on-air play.
I tried converting this file to 44.1kHz and 96kHz but I see no changes in sample period (exactly 1ms) when opening the resulting files in Adobe Audition. The output file is identical to the input file.
What is there that I don't understand about this DSP effect? If it doesn't work, why is it included in the effects plug-in for dBpoweramp Music Converter toolset?
If anyone knows of a tool to do this easily, please let me know. (I also want to be able to convert this reference-quality WAV file to FLAC for the same purposes. I don't need to change the bit-depth, although I wish the source file were 102/24 to begin with.)
d2b
I tried converting this file to 44.1kHz and 96kHz but I see no changes in sample period (exactly 1ms) when opening the resulting files in Adobe Audition. The output file is identical to the input file.
What is there that I don't understand about this DSP effect? If it doesn't work, why is it included in the effects plug-in for dBpoweramp Music Converter toolset?
If anyone knows of a tool to do this easily, please let me know. (I also want to be able to convert this reference-quality WAV file to FLAC for the same purposes. I don't need to change the bit-depth, although I wish the source file were 102/24 to begin with.)
d2b
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