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27th September 2010, 00:36 | #12501 | Link | |
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27th September 2010, 02:11 | #12502 | Link |
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Don't most soundcards handle only 16 bit? I can definitely handle decoding to 32 bit fp but I was under the impression that windows or the soundcard automatically discards all that extra information or "downsizes" to 16 bit upon outputting to the speakers?
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27th September 2010, 02:15 | #12503 | Link |
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No, I don't think so. Commercial Blu-ray players will do that if they don't detect PAP hardware that they support, but, Windows won't downsample by itself.
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27th September 2010, 03:45 | #12504 | Link | |
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any kind of audio post-processing requires 32/64fp to avoid rounding errors(that's how accurate VST plugins are). I personally do a lot of post-processing on AC3/DTS in ffdshow in order to get a binaural stereo downmix. |
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27th September 2010, 04:03 | #12505 | Link |
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That's a BIG "perhaps". OK, I myself do not have reliable statistics on that, but it's a fact that even an outdated frAudigy-2 can accept a 24-bit+96kHz input and transmit this directly to its Digital2Analog Converter.
Last edited by Midzuki; 27th September 2010 at 04:07. Reason: clarification |
27th September 2010, 07:00 | #12506 | Link |
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Yeah, quite alot of sound cards actually do support 24-bit these days.
And even if they don't, decoding to 32fp and then dithering down to 16bit is far superior to just decoding in 16bit directly. Not everyone may notice the difference, but there certainly are people that do. And especially when you do alot of post-processing, a 16bit signal can be degraded quite heavily, and you might even hear it yourself.
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27th September 2010, 07:26 | #12507 | Link | |
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If someone wanted to do this, they'd have to add a flag indicating the number of bits the MP3 encoder used and then the decoder/ditherer would behave based on that - like this you'd always win, otherwise the loss from 16 -> 15 bit seems more critical to me than the potential e.g. 16 -> 18 bit win. |
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27th September 2010, 07:41 | #12508 | Link |
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You seem to miss the fact that libav decodes internally to floating point and then ROUNDS the decoded data down to int16. This is a very clear violation of digital processing laws and results in relatively high quantization errors. If libav would dither the floating point decoding results down to int16, you could argue about audiophiles bla bla. But what libav currently does is a middle sized catastrophe for audio quality.
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27th September 2010, 08:37 | #12511 | Link | |
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You clearly do not understand how decoding lossy audio works (nor the linked page, which clearly specifys how it works). The decoding process should produce as many bits of data as it can, and that does not depend on the bitdepth of the original source!
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LAV Filters - open source ffmpeg based media splitter and decoders Last edited by nevcairiel; 27th September 2010 at 08:40. |
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27th September 2010, 12:15 | #12512 | Link | |
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There's no point in decoding lossy audio in anything else than 32fp. You're making a truckload of rounding errors in 16int...There's a good reason if all the audio software always post-process in 32/64fp: avoiding useless rounding errors. Even media players such as foobar decode mp3 in 32fp, and so do liba52 and libdts...why not libavcodec? I've been whining about it since forever, but apparently the libavcodec ppl don't care...maybe STaRGaZeR could hook us up w/ 32fp DTS, I don't care much for AC3 as liba52 sounds great(in 32fp at that) but I really don't like libdts...it sounds metallic, I very much prefer libavcodec. Last edited by leeperry; 27th September 2010 at 12:26. |
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27th September 2010, 13:04 | #12513 | Link | |
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27th September 2010, 13:13 | #12514 | Link | |
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Don't be so sure that 32fp doesn't many an audible diff over 16int...especially if you're using DirectSound or Reclock resampling. Anyway, STaRGaZeR didn't confirm whether he could fix DTS in libavcodec...libmad already does 32fp MP3 and liba52 32fp AC3. I also wish the winamp2 DSP plugins in ffdshow were not converted to 16int both ways(I use a VST plugins wrapper), as 24int is entirely possible :/ Last edited by leeperry; 27th September 2010 at 13:16. |
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27th September 2010, 18:20 | #12517 | Link | ||
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Reimar, since you're a ffmpeg developer, could you consider adding an option to bypass the float to int16 conversion you have at the end of your decoders? Easy, cheap and everybody is happy. |
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27th September 2010, 18:28 | #12518 | Link |
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Custom code only sucks when you use subversion trying to manage it .. what you do .. :P
Don't blame code for your version control shortcomings, imho Anyhow, the patch to enable 32fp decoding of DTS is pretty short, but its more then a one line change to disable the downconversion. Maybe one of the ffmpeg guys could shed some light on the reasons why its downconverting everything to int16 anyway
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27th September 2010, 18:37 | #12519 | Link |
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An eternity ago I had asked both Benjamin Larsson (DTS) and Justin Ruggles (AC3 + E-AC3) about that and IIRC both said that native bitdepth output was planned for a future libav version. But that was ages ago. I had sent my eac3to patches to both of them, but neither was willing to apply them. So I keep on patching libav, year after year...
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