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7
Lossless / Other Codecs / Re: Tested: Lossless decoding speed, multithreaded - and fast verification
Last post by Porcus -
To the extent TTA is interesting at all, the lack of flexibility suggests that differences can be put down to code quality and not to "prioritized this type of encoding strategy" and the like. And:
ffmpeg -threads 1 DEcodes much faster than the reference, which takes 42 percent more time. Tested the half-hour long file, mean of repeated runs:

533x realtime: DEcoding by ffmpeg -threads 1
375x realtime: DEcoding by tta.exe

ffmpeg also ENcodes faster, but there the difference is small. 451x realtime vs 423x realtime.
Everything to NUL.

8
Opus / Re: Opus decoding complexity
Last post by 2012 -
Quote from: Klymins on
@saratoga Looks like you're wrong. I did the decoding speed test myself and MP3 was way faster than Opus.

Different devices (and software environments) have different capabilities and properties. The same applies to decoders.

On the sort of device where this would become a concern, a fixed-point Opus decoder is used (possibly with extra optimizations applied).
On a PC, a floating-point Opus decoder (slower, but more accurate) is often used.

libopus offers both variants. A native Opus decoder exists in FFmpeg too which is maybe floating-point only (not sure).

There are many MP3 decoders out there.

Not sure which decoders would have been used in your test.

Quote
Edit: Both are 16kbps.

Opus is basically two codecs in one, SILK and CELT. One of the two, or a hybrid, is maybe used internally. At 16kbps, SILK (optimized for low-bitrate voice content) was probably used. Hybrid is a possibility too depending on input parameters and content.  In any case, that test is not sufficient.

------

The way you're approaching this almost give the impression that you're trying to solve a puzzle or answer a pop quiz question. Maybe defining a practical use-case, testing for it, then asking questions if any arise, would be a better and more fruitful approach!
9
MP3 - General / Re: Resurrecting/Preserving the Helix MP3 encoder
Last post by Kraeved -
Quote from: Case on
If there is a help text about Ctrl+C for interrupting then the program will also need a signal handler for the interruption. Otherwise Ctrl+C will just terminate the process hard leaving the output file in invalid state with incorrect headers.

I compared the behavior of different encoders when pressing Ctrl-C combo.

Monkey Audio, WavPack and NeroAAC report it was pressed:
a) Ctrl+C: MAC has been interrupted !!!
b) creating in.wv,   7% done...^C (that ^C is something, but not user-friendly)
c) Interrupted at 27 seconds... (27 seconds of what? Not the encoding time, but the source time)

WavPack does not leave an incomplete file.

Lossy encoders.

Code: [Select]
$ exhale 9 in.wav in.m4a
exhale - ecodis extended high-efficiency and low-complexity encoder
version 1.2.1 (x64, built on Dec 20 2023) - written by C.R.Helmrich
Encoding 44-kHz 2-channel 16-bit WAVE to low-complexity xHE-AAC at 192 kbit/s
Progress: --

$ faac in.wav
Freeware Advanced Audio Coder FAAC 1.30
Initial quantization quality: 100
Average bitrate: 64 kbps/channel
Bandwidth: 19293 Hz
PNS level: 4
Object type: Low Complexity(MPEG-2) + IS + PNS
Container format: Transport Stream (ADTS)
Encoding in.wav to in.aac
   frame          | bitrate | elapsed/estim | play/CPU | ETA
 2150/24275 (  8%)|  133.6  |    0.7/8.3    |   68.11x | 7.5

$ fdkaac -m5 in.wav
[3%] 00:14.838/09:23.627 (21x), ETA 00:26.297
654336/24855936 samples processed in 00:00.711

$ fhgaacenc in.wav
Progress:   7%

$ lame in.wav
LAME 3.101 (beta 3, Dec 16 2023) 64bits
CPU features: SSE (ASM used), SSE2 (ASM used)
Using polyphase lowpass filter, transition band: 16538 Hz - 17071 Hz
Encoding in.wav to in.mp3
Encoding as 44.1 kHz j-stereo MPEG-1 Layer III (11x) 128 kbps qval=3
    Frame          |  CPU time/estim | REAL time/estim | play/CPU |    ETA
   400/21578  ( 2%)|    0:00/    0:24|    0:01/    0:53|   22.864x|    0:52
--09:13------------------------------------------------------------------------
   kbps        LR    MS  %     long  %
  128.0       64.8  35.2       100.0

$ lossywav in.wav
lossyWAV 1.4.2, Copyright (C) 2007-2016 Nick Currie. Copyleft.
Filename  : in.wav
Settings  : --quality standard
File Info : 44.10kHz; 2 channel; 16 bit; 09:23.62, 94.81MiB
Progress  :  2.11%; 1.2178 bits; 15.74x; 00:00.76/00:35.81

$ mpcenc in.wav
MPC Encoder 1.30.0 --stable-- (C) 1999-2009 Buschmann/Klemm/Piecha/MDT
 encoding file 'in.wav'
       to file 'in.mpc'
 SV 8, Profile 'Standard'
    %|avg.bitrate| speed|play time (proc/tot)| CPU time (proc/tot)| ETA
  7.4  164.1 kbps 26.94x     0:41.5    9:23.6     0:01.5    0:20.9     0:19.3

$ neroaacenc -if in.wav -of in.m4a
Nero AAC Encoder
Package build date: Feb 18 2010
Package version:    1.5.4.0
Interrupted at 27 seconds...

$ oggenc in.wav
Opening with wav module: WAV file reader
Encoding "in.wav" to
         "in.ogg"
at quality 3,00
        [  2,0%] [ 0m48s remaining] \

$ opusenc in.wav in.opus
Encoding using libopus 1.5.2-2-gdfd4175 (audio)
-----------------------------------------------------
   Input: WAV, 44.1 kHz, 2 channels, stereo
  Output: Opus, 2 channels (2 coupled), stereo
          20ms packets, 96 kbit/s VBR
 Preskip: 312
[|] 10% 00:01:00.12   20x realtime, 95.94 kbit/s

$ qaac in.wav
qaac 2.82, CoreAudioToolbox 7.10.9.0
in.m4a
AAC-LC Encoder, TVBR q91, Quality 96
[2.9%] 0:16.370/9:23.626 (21.0x), ETA 0:26.075
721920/24855936 samples processed in 0:00.780
Overall bitrate: 201.654kbps
Optimizing...done

$ qoa in.wav in.qoa
in.wav: channels: 2, samplerate: 44100 hz, samples per channel: 24855936, duration: 563 sec

Lossless encoders.

Code: [Select]
$ flac -7 in.wav
flac git-d2b24410 20240309
in.wav: 9% complete, ratio=0,512

$ mac in.wav auto -c2000
--- Monkey's Audio Console Front End (v 10.61) (c) Matthew T. Ashland ---
Compressing (normal)...
Progress: 8.3% (10.7 seconds remaining, 1.0 seconds total)
Ctrl+C: MAC has been interrupted !!!

$ takc -e in.wav
in.wav                              ...

$ tta -e in.wav in.tta
TTA1 lossless audio encoder/decoder, version 2.3
Encoding: "in.wav" to "in.tta"
Progress: 20%

$ wavpack -x3m --threads in.wav
WAVPACK  Hybrid Lossless Audio Compressor  Win64 Version 5.7.0
creating in.wv,   7% done...^C
10
Audio Hardware / Re: audio device on Beelink u59 / AZW U59 (U3E1)
Last post by stanley.tweedle -
It's so unnerving I always forget. The audio device actually works "fine" when I boot into Linux. There's still issues. I hesitate to ever buy a product from that manufacurer again. I hate to say that because I'm just not the type of person to go around trying to dis' on someone. I'm from way back when you could fit 6 hard drives in the case! the old IDE kind. but stuff worked back then somehow.  my fingers can barely get the screws out to swap the SSD on occasion. haha. We made some pretty bad-ash recordings with this Event Electronics GINA though. I think it was 16bit 44.1k?

From my professional experience, albeit twenty years ago by now, you're always going to get a better sound from outboard gear, even if the processor isn't being taxed because it's simply what it's made for. quality outboard gear (e.g. ad/da conversion processes) makes for night and day of course! Cables etc. When Windows tries to use it, it sounds like there's broken wires on the INSide! Obviously, it comes down to not being able to afford the proper gear at this time. So, I'm writing as much to hear myself say it.

I would say Linux powers it in fairly high fidelity. I personally never liked the way any audio sounds on Linux. Never studied it.

Thank you, cordially!