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28th September 2010, 23:38 | #12561 | Link | |
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Every option in sound processing was checked. Every option in the output section was checked. It seems you don't get that I've modified ffdshow to process winamp plugins in 24-bit (output is zeroed to 32-bit) instead of 16-bit here in my system, by changing "tons of "int16_t"/"SF_PCM16" in those sections" into something else following your own words. Too bad I couln't get 32-bit to work, everything was fine but the audio returned from the plugin was untouched. Just a FYI: Code:
int (*ModifySamples)(struct winampDSPModule *this_mod, int16_t *samples, int numsamples, int bps, int nch, int srate) Someone wanna bet on this? |
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29th September 2010, 00:04 | #12563 | Link | |
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We can output the wav files and compare them right? so whats the difference? When madshi had to prove his renderer does better chroma scaling he made a screenshot comparison, and i've seen tools that can highlight the pixels that the better scaling affects. the same should be possible for audio, no need to involve ears or speakers. just the pure wave, is it different or not, and if so where.
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29th September 2010, 00:54 | #12564 | Link |
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If you want to show better chroma upsampling then you show a picture; if you want to show better audio quality you have to do it with an audio track.
Last edited by kieranrk; 29th September 2010 at 01:01. |
29th September 2010, 02:53 | #12565 | Link |
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Some new tests with O2 and O3:
Sample 1: Blu-ray, 1080p24, ~35Mbps O2 --> 178,12fps O3 --> 182,43fps Sample 2: Blu-ray, 1080i30, ~35Mbps O2 --> 166,9fps O3 --> 169,9fps Sample 3: Big Buck Bunny, 720p24, ~2,5Mbps O2 --> 672.06fps O3 --> 677.42fps 1-2% difference. What do you think about changing it? Binaries are ~25% smaller with GCC 4.5.1 |
29th September 2010, 08:05 | #12566 | Link | |
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Next, you're saying that most non-HD formats are 16bit, which is simply not true. AC3 and DTS are non-HD formats and perfectly support 24bit encodings. There's even a DTS header field which tells you which bitdepth was fed to the encoder. And from my experience there are plenty of 24bit DTS encodings out there, and I'm not talking about DTS-HD here. For AC3 we will never know which bitdepth was fed to the encoder, but the encoder also happily encodes 24bit sources. So it's a fair guess that many studio created AC3 tracks were also created from 24bit sources. I've done tests myself with rounding vs. dithering. Ok, I've done them at a lower bitdepth than 16bit to make the difference very easily audible and the difference was big. Of course with 16bit the difference is much smaller. But still, rounding is scientifically and mathematically worse than dithering. And libav rounds down every DTS and AC3 audio track to 16bit, no questions asked. Easy, the web is full of proof that dithering is needed. E.g. this is a nice starting point: http://www.users.qwest.net/~volt42/c...rExplained.pdf |
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29th September 2010, 08:18 | #12567 | Link |
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That document clearly shows that a 24bit (lossless?) source benefits from 24bit output or dithering over truncation.
However what kierank seems to suggest is that lossless codecs seem to expect the truncation, the encoder/decoder psy optimises the stream to not become ugly like the examples in the document. For the argument to be settled, the same test as explained in that document should be repeated (by either side of the argument) but this time using a typical lossy sample.
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29th September 2010, 08:30 | #12568 | Link | |
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> If the signal has had any processing happen to it > at higher bit depths then choose to add dither This is exactly what we're talking about here: "16bit int source -> lossy encoding -> lossy decoding -> 16bit int output". This processing chain has processing steps in a higher bit depth (floating point in the case of DTS and AC3), so dithering needs to be added when converting the floating point decoding results back to int16. Furthermore, what kieranrk conveniently ignores is that AC3 and DTS tracks were often encoded directly from true 24bit studio masters. |
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29th September 2010, 08:39 | #12569 | Link | |
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By the way, what effect has O2 vs. O3 on compile/build time? |
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29th September 2010, 09:51 | #12570 | Link | |
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What I expect the developers to want you to do is to only support float output for decoders that use float internally, but also add scale/bias options so that decoders that are capable of it will apply these to the output "for free", and then use that to implement a very fast float -> 16 bit int conversion outside the codec like the one that is currently done inside it (also means you get volume control for free). As you can see this will be a bit of effort, just deciding on the precise design will take some time. |
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29th September 2010, 10:16 | #12571 | Link |
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Adding scale/bias options sounds like a useful idea if you really need to squeeze out the last bit of performance on very slow hardware. But scale/bias options would also be useful if the decoders keep on outputting int16. So I don't think that changing the output data format and adding scale/bias options are (or have to be) connected in any way.
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29th September 2010, 11:16 | #12572 | Link | |
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That is very unlikely to be accepted by FFmpeg developers. Selectable codec output format (either at run- or compiletime) might be accepted, but it is not an ideal solution (more a quick hack) and is likely to be treated as such during review. |
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29th September 2010, 12:18 | #12573 | Link | |
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And I can't really measure compile time, I'm bottlenecked by the hard drive Last edited by STaRGaZeR; 29th September 2010 at 12:20. |
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29th September 2010, 13:45 | #12574 | Link |
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ffdshow_installer.iss
Line 326 is obsolete: Code:
Name: "audio\mp3\mp3lib"; Description: "mp3lib"; Components: ffdshow; Flags: unchecked exclusive |
29th September 2010, 19:11 | #12576 | Link | |
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30th September 2010, 07:12 | #12577 | Link | |
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well, ... Code:
now, the "multi-purpose" pin of the raw video filter connects to the output pin of the audio processor! P.S.: This post is dedicated to Keiyakusha Last edited by Midzuki; 1st October 2010 at 02:25. |
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30th September 2010, 12:16 | #12579 | Link |
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Hey,
i noticed the other day that the audio delay doesn't work with bitstreaming, at least not with all bitstreaming formats. I set a delay to fix A/V sync with my TV, and it was all peachy with PCM. But when bitstreaming AC3 for example, i get no sound at all anymore. It seemed to work when streaming DTS-HD, but not with DTS.. I can run more tests on other formats if that helps.
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30th September 2010, 12:25 | #12580 | Link | |
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Doesnt know what happen lol.
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