Ability to Trim and Add specific number of samples in dBpoweramp

[Lynx_TWO]

Some VST plugins tend to delay the input and output resulting in the effect that the original sample length is the same, but the processed waveform has been 'moved' by the same number of samples as the latency setting on the plugin. StereoTool is a perfect example of this. It's designed to be used real-time so when using it as an offline VST plugin, you get a file output that has the same number of samples, but the waveform starts 4096 samples later (or 128, 2048, all depends on the latency setting). Really annoying if the end of the track was supposed to seamlessly go to the next track since it also cuts off 4096 samples off audio at the end.

A workaround would be if dBpoweramp had an option to add a specific number of samples at the end of each file, and then the ability to trim a specific number of samples at the end. This would compensate manually for some VST Plugin latency due to the FFT size. In my particular case it would be 4096 samples, but generally you are going to be looking at powers of 2, so

64 samples
128 samples
256 samples
512 samples
1024 samples
2048 samples
4096 samples
8192 samples
16384 samples
32768 samples
65556 samples
etc.

[schmidj]

Look at the DSP Codec "Add Silence". Looks to do what you want if set up the way you need.

[Lynx_TWO]

Look at the DSP Codec "Add Silence". Looks to do what you want if set up the way you need.

Yes, unfortunately it's not granular enough. For example, here's a quick chart of samples converted to ms:

Samples	44100 Hz	48000 Hz	88200 Hz	96000 Hz	176400 Hz	192000 Hz
64	1.451247166 ms	1.333333333 ms	0.725623583 ms	0.666666667 ms	0.362811791 ms	0.333333333 ms
128	2.902494331 ms	2.666666667 ms	1.451247166 ms	1.333333333 ms	0.725623583 ms	0.666666667 ms
256	5.804988662 ms	5.333333333 ms	2.902494331 ms	2.666666667 ms	1.451247166 ms	1.333333333 ms
512	11.60997732 ms	10.66666667 ms	5.804988662 ms	5.333333333 ms	2.902494331 ms	2.666666667 ms
1024	23.21995465 ms	21.33333333 ms	11.60997732 ms	10.66666667 ms	5.804988662 ms	5.333333333 ms
2048	46.4399093 ms	42.66666667 ms	23.21995465 ms	21.33333333 ms	11.60997732 ms	10.66666667 ms
4096	92.87981859 ms	85.33333333 ms	46.4399093 ms	42.66666667 ms	23.21995465 ms	21.33333333 ms
8192	185.7596372 ms	170.6666667 ms	92.87981859 ms	85.33333333 ms	46.4399093 ms	42.66666667 ms
16384	371.5192744 ms	341.3333333 ms	185.7596372 ms	170.6666667 ms	92.87981859 ms	85.33333333 ms
32768	743.0385488 ms	682.6666667 ms	371.5192744 ms	341.3333333 ms	185.7596372 ms	170.6666667 ms
65536	1486.077098 ms	1365.333333 ms	743.0385488 ms	682.6666667 ms	371.5192744 ms	341.3333333 ms
131072	2972.154195 ms	2730.666667 ms	1486.077098 ms	1365.333333 ms	743.0385488 ms	682.6666667 ms
262144	5944.30839 ms	5461.333333 ms	2972.154195 ms	2730.666667 ms	1486.077098 ms	1365.333333 ms
524288	11888.61678 ms	10922.66667 ms	5944.30839 ms	5461.333333 ms	2972.154195 ms	2730.666667 ms
1048576	23777.23356 ms	21845.33333 ms	11888.61678 ms	10922.66667 ms	5944.30839 ms	5461.333333 ms
2097152	47554.46712 ms	43690.66667 ms	23777.23356 ms	21845.33333 ms	11888.61678 ms	10922.66667 ms
4194304	95108.93424 ms	87381.33333 ms	47554.46712 ms	43690.66667 ms	23777.23356 ms	21845.33333 ms
8388608	190217.8685 ms	174762.6667 ms	95108.93424 ms	87381.33333 ms	47554.46712 ms	43690.66667 ms

[Lynx_TWO]

Now, I did figure out I can easily add the number of samples I need to the end of a WAVE file using the 'Insert Audio'. I have created a WAVE file at 32-bit, 88.2Khz, 4096 samples Attached.

Now, I can do the exact same thing by calculating the number of samples I need to make up for the the trim at the beginning, so, for 88,200 Hz, 4096 samples = 46.439909297 ms.

1 ms - 0.439909297 ms = 0.560090703 ms

0.560090703 ms = 49.4 samples at 88,200 Hz, so I could theoretically create a wave file with 49 samples, add that to the beginning of the WAV file using the 'Insert Audio' DSP, process everything, then use Trim at 47ms.

Now, while that gets me very damned close, it's not sample-accurate which could end up causing pops in the audio when played back.

32-bit, 88.2Khz, 4096 samples.zip

[Maxpower]

Just wanted to acknowledge the discipline and depth of investigation by @Lynx_TWO.

It also highlights the advantages of sticking with 48kHz, 96kHz and 192kHz wherever possible.

[Lynx_TWO]

Just wanted to acknowledge the discipline and depth of investigation by @Lynx_TWO.

It also highlights the advantages of sticking with 48kHz, 96kHz and 192kHz wherever possible.

Haha Thanks, man! I do love getting into the maths sometimes :P