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romain8991

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  1. @andrewilley Right, I see. Then, I hope my word will spread and more users will ask for it. Thanks anyway.
  2. @andrewilley Since it's feasible and, in my honest opinion, an interesting feature to implement, could you please add it to the vote section or give it consideration in another form? Thanks again!
  3. @maxmp I see. But, you said it yourself, it allows to avoid adding any noise or intermediate clipping, even on slow devices, and I would add, by a principle of precaution and since 64 bit fixed point internal sample and DSP is the very best thing you can get quality wise, that it would be interesting. If we wan't to make Poweramp a more Hi-Fi audio player, I really think it would make sense though.
  4. @MotleyG Alright, but most Android devices currently available can handle 64 bit internal sample audio format and DSP with no issue (I can handle it with no lag whatsoever, using JRiver Media Center, even on my slow portable Acer computer, using 4 gigs of DDR3 ram and a simple 4 core low end CPU). The gain would be real, but the capacity of hearing it would depend on the audio device used (with bad quality speakers, earphones or headphones, the difference would be harder to tell, but the same goes for bypassing sample rate conversion, since SWR resample is pretty neutral and precise). Regarding development, I don't think it would be too hard to implement with libswresample (and I would dare to say that I think it would be easier than "no resampling" option), but I leave this part on the developer's hand.
  5. @Fitzian I already implied that I'm not looking for more headroom or loudness but a bigger calculation precision. Besides, the internal sample format is not what goes directly into the audio output and dithered before arriving there (the output sample format exist for this purpose, and the real available headroom after that is dependent, like you said, of the capacities of the DAC). Even though the final headroom is way bellow the mathematical values of 64 bit, the calculations can still be made in this format during sound processing/playing, while keeping alot of their precision, thanks to dithering. And that, in my sense, is why it has purpose and gives better quality results. So, then again, why not, instead of Float32, use 64 bit for internal sample format, and not just DSP, fixed or, at least, floating point?
  6. @Fitzian I didn't say otherwise and my point about the mathematical precision of 64 bit fixed point still stands. 🤔 If you use Float64 for DSP, why not at least use it for internal sample format as well or, if you go further the logic, 64 bit fixed point for both (DSP and internal sample format)?
  7. @Fitzian I disagree with you. 64 bit internal sample is indeed used in many DAW stations for DSP and the purpis to use it is to have a much more precise DSP and output of the signal, even after dithering.
  8. @andrewilley I know that, I'm not seeking for a ridiculously high dB output. I'm actually looking for a more precise sample format conversion, with more transparent dithering, DSP with greater precision and a much better overall sound quality (more audio details and lifelike sound). It's mathematical, a conversion from a much higher internal format and its usage in DSP gives greater precision compared to floating point, especially if you compare 64 bit fixed point to Float32.
  9. @andrewilley They still don't, I was referring to the computer version.
  10. @Fitzian Yes, JRiver Media Center for instance. Still, I think it would be nice to at least have Float64 for internal sample, just like for DSP effects.
  11. @andrewilley Alright, I see, thanks for the clarification. But could you please add support for 64 bit fixed point for internal sample and DSP, for better sound quality and precision?
  12. @maxmp Thanks for the response. So, is the internal format Float64 with lossless formats and most formats?
  13. Through FFmpeg libswresample commands, I think it's feasible: - https://ffmpeg.org/ffmpeg-resampler.html - https://ffmpeg.org/doxygen/trunk/group__lswr.html - https://ffmpeg.xabe.net/docs/Xabe.FFmpeg.AudioCodec.html
  14. @andrewilley Thanks for the details about DSP format. What about the internal sample format, since some websites (like Slant) suggest that it uses 32 bit floating point? And could you still add 64 bit fixed point internal sample and DSP?
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