L5730
06-14-2017, 09:50 AM
Hi all,
Firstly, I apologise if this is in the wrong section of the forum. I couldn't find one specific to the command line coreconverter.exe.
OK, what I am doing is piping from ffmpeg to coreconverter to make use of ffmpeg's audio filters and coreconverter's VST DSP support.
I have found that my converted files have some issue with regards to their position and are also cut a little short.
Here is the command line:
ffmpeg.exe > input file > decode 32bit fp and apply gain var (dB) > pipe 32bit fp lpcm audio data out to coreconverter.exe > accept input pipe attributes from pre-defined vars > set output file location > apply VST limiter > dither down to 16bit > convert to output file.
FFMPEG.exe -hide_banner -loglevel panic -i %1 -af volume=volume=%GAIN%dB:precision=float -acodec pcm_f32le -f f32le - | "c:\program files (x86)\illustrate\dbpoweramp\coreconverter.exe" -infile="-" -sourcefreq="!SAMPLERATE!" -sourcechannels="!CHANNELS!" -sourcebits="32" -sourceisfloat -silent -priority="high" -outfile="%OUT_FOLDER%\%~n1.wav" -dspeffect1="VST PlugIn=George Yohng's W1 Limiter=X3A000000433A5C50726F6772616D2046696C65732 028783836295C56535433325C47656F72676520596F686E672 773205731204C696D697465722E646C6C4C0000000C0000817 4687265736800004A196A3F0C0000816365696C696E6700963 2713F0C00008172656C65617365002EF5253F1400008161646 1707452656C65617365000000000000000000000080" -dspeffect2="Bit Depth=-depth={qt}16{qt} -dither={qt}tpdf{qt}"
Everything works, I get output files exactly how I expect them, but with one problem - the sample lengths aren't the same as their input file.
The Problem:
*1: 64 samples of silence are added to the front of the audio data. This is no major concern, I can live with it just fine.
*2: 64 samples are truncated or 'chopped off' of the end of the audio data. This doesn't seem right to me, and I would like my samples back.
I've tested ffmpeg.exe independently and it outputs exactly the same length as went in.
When I tested coreconverter.exe independently I found the following:
*a: 64 samples of silence were added at the front of the audio data
*b: 448 samples of silence were added at the end of the audio data
*c: The audio data was not truncated.
Is there something I am missing, or some kind of trick to preventing truncation?
I'd rather not specify the time in ms of the pipe, because that's a potential source of error when reading lossy formats, and would require a slower scan to determine the value more accurately (ffprobbe.exe).
I am not averse to just bolting on another 64 samples of silence to the end of the audio from ffmpeg.exe, it doesn't seem hard to do and fairly swift.
Thanks for any help.
dBPowerAmp ref: 15.1
dsp release: 11
Firstly, I apologise if this is in the wrong section of the forum. I couldn't find one specific to the command line coreconverter.exe.
OK, what I am doing is piping from ffmpeg to coreconverter to make use of ffmpeg's audio filters and coreconverter's VST DSP support.
I have found that my converted files have some issue with regards to their position and are also cut a little short.
Here is the command line:
ffmpeg.exe > input file > decode 32bit fp and apply gain var (dB) > pipe 32bit fp lpcm audio data out to coreconverter.exe > accept input pipe attributes from pre-defined vars > set output file location > apply VST limiter > dither down to 16bit > convert to output file.
FFMPEG.exe -hide_banner -loglevel panic -i %1 -af volume=volume=%GAIN%dB:precision=float -acodec pcm_f32le -f f32le - | "c:\program files (x86)\illustrate\dbpoweramp\coreconverter.exe" -infile="-" -sourcefreq="!SAMPLERATE!" -sourcechannels="!CHANNELS!" -sourcebits="32" -sourceisfloat -silent -priority="high" -outfile="%OUT_FOLDER%\%~n1.wav" -dspeffect1="VST PlugIn=George Yohng's W1 Limiter=X3A000000433A5C50726F6772616D2046696C65732 028783836295C56535433325C47656F72676520596F686E672 773205731204C696D697465722E646C6C4C0000000C0000817 4687265736800004A196A3F0C0000816365696C696E6700963 2713F0C00008172656C65617365002EF5253F1400008161646 1707452656C65617365000000000000000000000080" -dspeffect2="Bit Depth=-depth={qt}16{qt} -dither={qt}tpdf{qt}"
Everything works, I get output files exactly how I expect them, but with one problem - the sample lengths aren't the same as their input file.
The Problem:
*1: 64 samples of silence are added to the front of the audio data. This is no major concern, I can live with it just fine.
*2: 64 samples are truncated or 'chopped off' of the end of the audio data. This doesn't seem right to me, and I would like my samples back.
I've tested ffmpeg.exe independently and it outputs exactly the same length as went in.
When I tested coreconverter.exe independently I found the following:
*a: 64 samples of silence were added at the front of the audio data
*b: 448 samples of silence were added at the end of the audio data
*c: The audio data was not truncated.
Is there something I am missing, or some kind of trick to preventing truncation?
I'd rather not specify the time in ms of the pipe, because that's a potential source of error when reading lossy formats, and would require a slower scan to determine the value more accurately (ffprobbe.exe).
I am not averse to just bolting on another 64 samples of silence to the end of the audio from ffmpeg.exe, it doesn't seem hard to do and fairly swift.
Thanks for any help.
dBPowerAmp ref: 15.1
dsp release: 11