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Old 24th April 2024, 18:28   #2841  |  Link
FranceBB
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I'm back with some examples.
After the last discussion about SetMemoryMax() I did some more tests and this time I used a real script from a real use case scenario (although I trimmed the first 1000 frames only as I didn't want to let it filter 40 minutes worth of documentary for nothing).


Source: DNX HQX UHD 4:2:2 12bit 728 Mbit/s BT709 SDR

Video
ID : 2
Format : VC-3
Commercial name : DNxHR HQX
Format version : Version 3
Format profile : RI@HQX
Format settings, wrapping mode : Frame
Codec ID : 0D01030102110100-0401020271250000
Duration : 40 min 53 s
Bit rate mode : Constant
Bit rate : 728 Mb/s
Width : 3 840 pixels
Height : 2 160 pixels
Display aspect ratio : 16:9
Active Format Description : Full frame 16:9 image
Frame rate : 25.000 FPS
Color space : YUV
Chroma subsampling : 4:2:2
Bit depth : 12 bits
Scan type : Progressive
Bits/(Pixel*Frame) : 3.512
Stream size : 208 GiB (100%)
Color range : Limited
Color primaries : BT.709
Transfer characteristics : BT.709
Matrix coefficients : BT.709



AVS Script:


Code:
video=LWLibavVideoSource("PARTE 1.mxf")
FL=WAVSource("Parte 1 Mix OnAir.L.wav")
FR=WAVSource("Parte 1 Mix OnAir.R.wav")
CC=WAVSource("Parte 1 Mix OnAir.C.wav")
LFE=WAVSource("Parte 1 Mix OnAir.LFE.wav")
LS=WAVSource("Parte 1 Mix OnAir.Ls.wav")
RS=WAVSource("Parte 1 Mix OnAir.Rs.wav")

Dolby=MergeChannels(FL, FR, CC, LFE, LS, RS)

Stereo=WAVSource("Parte 1 MIX Lt - Rt.wav")

Mute=BlankClip(length=61335, fps=25, audio_rate=48000, channels=2)

audio=MergeChannels(Stereo, Mute, Dolby, Mute, Mute, Mute)

AudioDub(video, audio)


ConvertBits(16)

z_ConvertFormat(pixel_type="RGBP16", colorspace_op="709:709:709:limited=>rgb:709:709:full", resample_filter_uv="spline64", dither_type="error_diffusion", use_props=0)

Cube("C:\Program Files (x86)\AviSynth+\LUTs\5a_BT709_HLG_UPCONVERT_DISPLAY_mode.cube", fullrange=1, interp=1)

z_ConvertFormat(pixel_type="YUV422P10", colorspace_op="rgb:std-b67:2020:full=>2020:std-b67:2020:limited", resample_filter_uv="spline64", dither_type="error_diffusion", use_props=0)

Limiter(min_luma=64, max_luma=940, min_chroma=64, max_chroma=960)

trim(0, 1000)


AVS Meter Test:


Code:
AvsMeter64.exe "Test.avs"

pause


Hardware:

CPU: Intel Xeon Gold 6238R 2.20GHz x2 (56c/112th)
RAM: 32 x 4 DDR4 1460 MHz (128 GB)
Motherboard: HPE ProLiant XL420 Gen10
GPU: HP Matrox Matrox G200eh3 32MB
Storage 1: 447GB Hitachi
Storage2: 4471GB Hitachi
NIC: HP 562SFP+ 10 Gigabit




Test 1 - no SetMemoryMax() specified (i.e default 4GB)

Code:
AVSMeter 3.0.9.0 (x64), (c) Groucho2004, 2012-2021
AviSynth+ 3.7.3 (r4003, 3.7, x86_64) (3.7.3.0)

Number of frames:                     1001
Length (hh:mm:ss.ms):         00:00:40.040
Frame width:                          3840
Frame height:                         2160
Framerate:                          25.000 (25/1)
Colorspace:                      YUV422P10
Audio channels:                         16
Audio bits/sample:                      24
Audio sample rate:                   48000
Audio samples:                     1921920

Frames processed:                   1001 (0 - 1000)
FPS (min | max | average):          3.484 | 4.231 | 4.084
Process memory usage (max):         808 MiB
Thread count:                       73
CPU usage (average):                2.3%

Time (elapsed):                     00:04:05.089


Test 2 - SetMemoryMax(25000) specified (i.e 25GB)

Code:
AVSMeter 3.0.9.0 (x64), (c) Groucho2004, 2012-2021
AviSynth+ 3.7.3 (r4003, 3.7, x86_64) (3.7.3.0)

Number of frames:                     1001
Length (hh:mm:ss.ms):         00:00:40.040
Frame width:                          3840
Frame height:                         2160
Framerate:                          25.000 (25/1)
Colorspace:                      YUV422P10
Audio channels:                         16
Audio bits/sample:                      24
Audio sample rate:                   48000
Audio samples:                     1921920

Frames processed:                   1001 (0 - 1000)
FPS (min | max | average):          3.574 | 4.406 | 4.201
Process memory usage (max):         841 MiB
Thread count:                       73
CPU usage (average):                2.3%

Time (elapsed):                     00:03:58.282


Test 3 - SetMemoryMax(90000) specified (i.e 90GB)

Code:
AVSMeter 3.0.9.0 (x64), (c) Groucho2004, 2012-2021
AviSynth+ 3.7.3 (r4003, 3.7, x86_64) (3.7.3.0)

Number of frames:                     1001
Length (hh:mm:ss.ms):         00:00:40.040
Frame width:                          3840
Frame height:                         2160
Framerate:                          25.000 (25/1)
Colorspace:                      YUV422P10
Audio channels:                         16
Audio bits/sample:                      24
Audio sample rate:                   48000
Audio samples:                     1921920

Frames processed:                   1001 (0 - 1000)
FPS (min | max | average):          3.580 | 4.405 | 4.209
Process memory usage (max):         841 MiB
Thread count:                       73
CPU usage (average):                2.3%

Time (elapsed):                     00:03:57.812

As we can see from the tests above, even when working on UHD 12bit contents with 16bit precision and doing very heavyweight operations like converting to RGB full range, applying a LUT with tetrahedral interpolation, going back to YUV limited tv range and dithering down to 10bit, the RAM usage never exceeds 4GB, in fact it doesn't matter how much RAM I tell Avisynth to allocate with SetMemoryMax(), it never really uses it anyway and the performances stay the same. This is - of course - without using Prefetch() and I never use it in any of my encodes anyway, so I guess I'm safe to say that I don't really need to specify SetMemoryMax() and I'm pretty much fine with the default value.


Here's another test with a different kind of conversion, this time instead of going from BT709 SDR to BT2020 HLG with highlights expansions to 420 nits using the BBC LUT, we're using HDR Tools by Jean Philippe Scotto di Rinaldi to go to BT2020 SDR 100 nits:

AVS Script:

Code:
video=LWLibavVideoSource("PARTE 1.mxf")
FL=WAVSource("Parte 1 Mix OnAir.L.wav")
FR=WAVSource("Parte 1 Mix OnAir.R.wav")
CC=WAVSource("Parte 1 Mix OnAir.C.wav")
LFE=WAVSource("Parte 1 Mix OnAir.LFE.wav")
LS=WAVSource("Parte 1 Mix OnAir.Ls.wav")
RS=WAVSource("Parte 1 Mix OnAir.Rs.wav")

Dolby=MergeChannels(FL, FR, CC, LFE, LS, RS)

Stereo=WAVSource("Parte 1 MIX Lt - Rt.wav")

Mute=BlankClip(length=61335, fps=25, audio_rate=48000, channels=2)

audio=MergeChannels(Stereo, Mute, Dolby, Mute, Mute, Mute)

AudioDub(video, audio)


ConvertBits(16)

#BT709 SDR to BT2020 SDR

ConvertYUVtoXYZ()
ConvertXYZtoYUV(Color=1, pColor=2)

ConverttoYUV422(matrix="Rec.2020", interlaced=false)

ConvertBits(bits=10, dither=1)

Limiter(min_luma=64, max_luma=940, min_chroma=64, max_chroma=960)

trim(0, 1000)
Test 1 - no SetMemoryMax() specified (i.e default 4GB)

Code:
AVSMeter 3.0.9.0 (x64), (c) Groucho2004, 2012-2021
AviSynth+ 3.7.3 (r4003, 3.7, x86_64) (3.7.3.0)

Number of frames:                     1001
Length (hh:mm:ss.ms):         00:00:40.040
Frame width:                          3840
Frame height:                         2160
Framerate:                          25.000 (25/1)
Colorspace:                      YUV422P10
Audio channels:                         16
Audio bits/sample:                      24
Audio sample rate:                   48000
Audio samples:                     1921920

Frames processed:                   1001 (0 - 1000)
FPS (min | max | average):          4.594 | 8.925 | 5.896
Process memory usage (max):         1377 MiB
Thread count:                       129
CPU usage (average):                16.5%

Time (elapsed):                     00:02:49.782

Test 2 - SetMemoryMax(25000) specified (i.e 25GB)

Code:
AVSMeter 3.0.9.0 (x64), (c) Groucho2004, 2012-2021
AviSynth+ 3.7.3 (r4003, 3.7, x86_64) (3.7.3.0)

Number of frames:                     1001
Length (hh:mm:ss.ms):         00:00:40.040
Frame width:                          3840
Frame height:                         2160
Framerate:                          25.000 (25/1)
Colorspace:                      YUV422P10
Audio channels:                         16
Audio bits/sample:                      24
Audio sample rate:                   48000
Audio samples:                     1921920

Frames processed:                   1001 (0 - 1000)
FPS (min | max | average):          4.550 | 8.933 | 5.884
Process memory usage (max):         1377 MiB
Thread count:                       129
CPU usage (average):                16.3%

Time (elapsed):                     00:02:50.134

Test 3 - SetMemoryMax(90000) specified (i.e 90GB)

Code:
AVSMeter 3.0.9.0 (x64), (c) Groucho2004, 2012-2021
AviSynth+ 3.7.3 (r4003, 3.7, x86_64) (3.7.3.0)

Number of frames:                     1001
Length (hh:mm:ss.ms):         00:00:40.040
Frame width:                          3840
Frame height:                         2160
Framerate:                          25.000 (25/1)
Colorspace:                      YUV422P10
Audio channels:                         16
Audio bits/sample:                      24
Audio sample rate:                   48000
Audio samples:                     1921920

Frames processed:                   1001 (0 - 1000)
FPS (min | max | average):          4.655 | 8.845 | 5.877
Process memory usage (max):         1377 MiB
Thread count:                       129
CPU usage (average):                16.0%

Time (elapsed):                     00:02:50.337
For those wondering why I'm "happy" with this, please consider that this server is part of a wider farm and would typically run several encodes at the same time, so the CPU is gonna be pegged at 100% anyway. Besides, those tests were made only considering Avisynth, but when x264 is encoding the files in XAVC Intra Class 300 then the usage consumption is of course much higher.



For reference, today the farm encoded 284 files:

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Old 25th April 2024, 01:49   #2842  |  Link
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Quote:
Originally Posted by FranceBB View Post
This is - of course - without using Prefetch() and I never use it in any of my encodes anyway, so I guess I'm safe to say that I don't really need to specify SetMemoryMax() and I'm pretty much fine with the default value.
If you don't add Prefetch, Avisynth+ runs in single threaded mode. Adding Prefetch can make a huge difference to encoding speed, depending on the filtering in the script.

There's a file here that can be saved as an avsi script and auto-loaded by Avisynth+. It tells Avisynth+ the type of multi-threading to use for most of the common plugins. Many plugins register their multi-threading mode with Avisynth+ automatically these days, but they still require you to add Prefetch to the script to activate multi-threading.

https://publishwith.me/ep/pad/view/r...cdWn4k9/latest

More info
http://avisynth.nl/index.php/AviSynt...lling_MT_modes

I assume for plugins with "internal" multi-threading it's a different thing to enabling Avisynth's own multi-threading. Ideally for some plugins, such as the ones in the JPSDR plugins pack, you should use the plugin arguments to specify how many threads Avisynth+ is using when you've added Prefetch to the script. And sometimes it's better to disable a plugin's multi-threading. Apparently it's not a good idea to use avstp.dll for multi-threading when Avisynth's multi-threading is enabled. The script I linked to above also tells avstp.dll to run in single threaded mode, assuming it's loaded.

Last edited by hello_hello; 25th April 2024 at 02:25.
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Old 25th June 2024, 10:13   #2843  |  Link
cretindesalpes
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Quote:
Originally Posted by FranceBB View Post
Yep, Ferenc fixed it in avstp.
I mean, of course he did, he's amazing.
Sorry I’m a bit late to the party. I read the Ferenc debugging report. Very interesting, nice catch! I saw that an assert occured in:
Code:
		++ loop_cnt;
		if (loop_cnt >= max_loop)
		{
			// This could indicate that the queue is:
			// - corrupted
			// - or in heavy contention
			assert (false);
			return nullptr;
		}
Obviously returning 0 is not expected at all by the calling code and makes everything fail. But have you tried to increase max_loop? The value in the code is totally arbitrary, I haven’t done any statistics on the maximum loop_cnt I could encounter in a heavy use, but maybe we are sometimes close to the limit (contention case)? (edit: the GRAME paper doesn’t even have this check in the presented algorithm, I probably added it during development).

Anyway, I love the irony of the fix: enclosing a “lock-free” code in a std::lock_guard
BTW this could be achieved more simply using a trivial std::list or std::deque for storage with guarded access instead of the complicated lock-free procedure. But I’m glad you fixed it!
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Last edited by cretindesalpes; 25th June 2024 at 10:30.
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Old 4th July 2024, 13:31   #2844  |  Link
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Currently, when a Convertto function is called, it uses the matrix from the frame properties.
This is ok, however it only really works when the file is right and the indexer populates the frame properties correctly, which is not always the case.
Here's a sample file: https://we.tl/t-aJB9L84pmw (link valid for 7 days)
(keep in mind that the file ain't important, it's just an example to make a point).

When it's being indexed by FFVideoSource() it shows the following frame properties:



Clearly Matrix(3) doesn't mean anything, but you can bet everything you want that ConverttoRGB24() is gonna try to use that and fail, in fact it fails with "Unknown matrix".

Of course one could always nuke frame properties with propClearAll() before using ConverttoRGB24() and call it a day, however that's not exactly ideal.

What I'm proposing here is to still allow the Convertto functions to use the matrix populated by the frame properties as it is today, but, if it's populated automatically and not passed by the user explicitly, to fallback to a safe default instead of throwing an error if what is passed through is garbage.
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Old 8th July 2024, 04:01   #2845  |  Link
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Quote:
Originally Posted by tebasuna51 View Post
I can't reproduce the problem with last test 14 (r4066), what is your installed version?
If it is the same version please upload the .avs used and the source (or a sample)
Sorry for the months late reply.

I just tried it again, and I'm still having the issue. It's not happening with every script, but some.

Source video

Source audio

Script:
Code:
v = lwlibavvideoSource("f:\virtualdub\tape transfers\opening to swing time 1996 vhs - edit.mkv", fpsnum=30000,fpsden=1001)
a = lwlibavaudiosource("d:\recordings\opening to swing time 1996 vhs - edit.flac")
audiodub(v,a)
#delayaudio(-.150)
assumetff().converttoyuv422(matrix="rec601", interlaced=true).convertbits(10)
#Crop(8, 4, -24, -6_
Levels(50, 1,920, 0, 1020, coring=false,dither=true).tweak(bright=0, cont=1.00, hue=-0, sat=1.00, coring=false, dither=true).convertbits(8)
#turnRight().Histogram().TurnLeft()
normalize(0.8912)
Trim(415, 5286)
prefetch(8)
Example WARNING: LOUD!!!
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Old 8th July 2024, 11:00   #2846  |  Link
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Yes, there are a problem and I can't understand where.

Played your script in VirtualDub2 sound noise and distorted audio.

Without the normalize() play fine, but also without the normalize play noise in mpc_hc.

Tested also the FFAudioSource and BSAudioSource decoders, and only the audio without video, normalize, etc., always noise and distort.

Maybe something related with AviSynth flac decoders?
Decoded with ffmpeg or flac seems fine.
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Old 8th July 2024, 13:14   #2847  |  Link
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There is FLAC with 24 bit resolution. That might be unusual for an Avisynth audio decoder and sound wrong when it does not report the correct sample layout.

Can you check technical details of your audio source, e.g. with MediaInfo or FLAC tools?
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Old 8th July 2024, 21:23   #2848  |  Link
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Quote:
Originally Posted by tebasuna51 View Post
Yes, there are a problem and I can't understand where.

Played your script in VirtualDub2 sound noise and distorted audio.

Without the normalize() play fine, but also without the normalize play noise in mpc_hc.

Tested also the FFAudioSource and BSAudioSource decoders, and only the audio without video, normalize, etc., always noise and distort.

Maybe something related with AviSynth flac decoders?
Decoded with ffmpeg or flac seems fine.
I had the same issue with an ac3 file, but as I said before, it seems to happen on some files, but not all.

Quote:
Originally Posted by LigH View Post
There is FLAC with 24 bit resolution. That might be unusual for an Avisynth audio decoder and sound wrong when it does not report the correct sample layout.

Can you check technical details of your audio source, e.g. with MediaInfo or FLAC tools?
Code:
General
Complete name                            : D:\recordings\opening to swing time 1996 vhs - edit.flac
Format                                   : FLAC
Format/Info                              : Free Lossless Audio Codec
File size                                : 15.9 MiB
Duration                                 : 3 min 0 s
Overall bit rate mode                    : Variable
Overall bit rate                         : 740 kb/s

Audio
Format                                   : FLAC
Format/Info                              : Free Lossless Audio Codec
Duration                                 : 3 min 0 s
Bit rate mode                            : Variable
Bit rate                                 : 740 kb/s
Channel(s)                               : 2 channels
Channel layout                           : L R
Sampling rate                            : 48.0 kHz
Bit depth                                : 16 bits
Compression mode                         : Lossless
Stream size                              : 15.9 MiB (100%)
Writing library                          : libFLAC 1.3.2 (2017-01-01)
MD5 of the unencoded content             : 3C5F271BC902D1430FDC4F3D062F4A21
This was a FLAC created by Adobe Audition. I usually do some dehumming and denoising. I imported the raw audio, also in FLAC (but this time converted with ffmpeg), and the same issue happens. I did try converting it to PCM with ffmpeg as well, and the issue persists. This doesn't happen with r4003.

Last edited by ENunn; 8th July 2024 at 22:22.
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Old 8th July 2024, 22:43   #2849  |  Link
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I can't reproduce it here. Samples as posted, script as posted. LSMASHSource built on May 1, freshly built AviSynth+ r4076 plugnew, Ubuntu 24.04. There's no noise or distortion.
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Old 8th July 2024, 23:46   #2850  |  Link
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Just to be safe, I updated LSMASHWorks, and the issue persists.
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Old 9th July 2024, 01:27   #2851  |  Link
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What happens if you use SetMaxCPU("None")?

Is this with 32-bit or 64-bit AviSynth+?
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Old 9th July 2024, 03:37   #2852  |  Link
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Quote:
Originally Posted by qyot27 View Post
What happens if you use SetMaxCPU("None")?
Still loud unfortunately.

Quote:
Originally Posted by qyot27 View Post
Is this with 32-bit or 64-bit AviSynth+?
64-bit.
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Old 9th July 2024, 07:48   #2853  |  Link
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Confirmed on AviSynth r4066.
LWLibavAudioSource, normalize(0.8912) on 64 bit Avisynth 4066 -> too much gain, negative values only, positive sample values clipped at 0
BestAudioSource: the same -> too much gain, negative values only, positive sample values clipped at 0
FFAudioSource: the same -> too much gain, negative values only, positive sample values clipped at 0
normalize(0.001) the same -> too much gain, negative values only, positive sample values clipped at 0
normalize(0.001, show=true) tells amplify DB: -58.9998 while still clipping heavily.

LWLibavAudioSource, normalize(0.8912) on 32 bit Avisynth 4066 -> fine
All these sourcefilters decode that .flac file as 2.0 16bit int, SoundForgePro 11.0 see it as 2.0 16bit too.

In SoundForgePro 11.0 I see a ± disbalance, not just a DC offset towards negative values.
Negative peaks coming from the narrator's mike are much stronger.
May be that is it what dips normalize64bit in the ocean ?

BTW, I had a overused Shure SM58 here which delivered the same waveform.
As a sound engineer I had to retire this one from duty.
Misadjusted coil vs gap, so a diaphragm dip gave less efficient voltage than a lift.
Nice to see on a scope or from the waveform.

AviSynth 4073 Clang is a no-starter for me. Gotta look out for AviSynth r4076...
Ah, AviSynth+ r4073 Intel LVM 64bit works ! And normalize fault is gone. Phew.
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Last edited by Emulgator; 9th July 2024 at 08:51.
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Old 9th July 2024, 10:48   #2854  |  Link
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Quote:
Originally Posted by Emulgator View Post
Ah, AviSynth+ r4073 Intel LVM 64bit works ! And normalize fault is gone. Phew.
With:
Code:
AviSynth+ 3.7.3 (r4073, 3.7, x86_64)  IntelLLVM
BestSource.dll  [n/a, 2024-04-25]
ffms2.dll  [2390.0.0.0, 2024-03-06]
LSMASHSource.dll  [1194.0.0.0, 2024-04-08]
And using (tested 3 decoders, same behavior):
Code:
#BSAudioSource("opening to swing time 1996 vhs - edit.flac")
#FFAudioSource("opening to swing time 1996 vhs - edit.flac")
LWLibavAudioSource("opening to swing time 1996 vhs - edit.flac", drc_scale=0)
AudioDubEx(BlankClip(length=Int(1000*AudioLengthF(last)/Audiorate(last)), width=720, height=720, fps=25), last)
Info()
Normalize(0.95)
Play fine in VirtualDub2 (stereo 16 bits).
Using AviSynth test 14 r4066 the output is like the attached image.

But, with r4073, still is wrong played in mpc-hc
Attached Images
 
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Last edited by tebasuna51; 9th July 2024 at 11:09. Reason: add info
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Old 9th July 2024, 11:21   #2855  |  Link
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tebasuna51, did you try to play a .avs script in MPC-HC served from r4073 ?
Or did you play a transcoded .wav as the title bar suggests, maybe from SoundOut() AviSynth r4073?
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Old 10th July 2024, 08:24   #2856  |  Link
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Quote:
Originally Posted by Emulgator View Post
tebasuna51, did you try to play a .avs script in MPC-HC served from r4073 ?
Or did you play a transcoded .wav as the title bar suggests, maybe from SoundOut() AviSynth r4073?
Of course I listen noise playing the .avs directly over mpc-hc with r4073 active, the wav is the VirtualDub2->File->Save Audio with r4066 active
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Old 10th July 2024, 10:35   #2857  |  Link
Emulgator
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So this would mean that r4073 still has the fault inherited.
AviSynth64+ r4073
Just throwing the script on MPC-HC64 2.3.0 here: plays fine.
LAVSplitter -> SaneAR Render

MPC-BEx64 1.7.2 using MPC Audio renderer: missing audio source filter, plays video only.
Code:
AVI/WAV File Source::Avisynth audio #1

Media Type 0:
--------------------------
Audio: 0xfffe 48000Hz 2.0 chn 1536 kbit/s

AM_MEDIA_TYPE: 
majortype: MEDIATYPE_Audio {73647561-0000-0010-8000-00AA00389B71}
subtype: MEDIASUBTYPE_0xfffe {0000FFFE-0000-0010-8000-00AA00389B71}
formattype: FORMAT_WaveFormatEx {05589F81-C356-11CE-BF01-00AA0055595A}
bFixedSizeSamples: 1
bTemporalCompression: 0
lSampleSize: 4
cbFormat: 40

WAVEFORMATEX:
wFormatTag: 0xfffe
nChannels: 2
nSamplesPerSec: 48000
nAvgBytesPerSec: 192000
nBlockAlign: 4
wBitsPerSample: 16
cbSize: 22 (extra bytes)

WAVEFORMATEXTENSIBLE:
wValidBitsPerSample: 16
dwChannelMask: 0x00000003
SubFormat: {00000001-0000-0010-8000-00AA00389B71}
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Last edited by Emulgator; 10th July 2024 at 11:03.
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Old 10th July 2024, 11:27   #2858  |  Link
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AviSynth64+ r4073 (both clang and Intel LLVM) + MPC-BE x64 1.7.2 (MPC Audio Renderer, LAV Filters 0.79.2) no issues with the following (also no issues with mpv 0.38.0 x86_64):
Code:
v = lwlibavvideoSource("opening to swing time 1996 vhs - edit.mkv", fpsnum=30000,fpsden=1001)

a = lwlibavaudiosource("opening to swing time 1996 vhs - edit.flac")
audiodub(v,a)
#delayaudio(-.150)
assumetff().converttoyuv422(matrix="rec601", interlaced=true).convertbits(10)
#Crop(8, 4, -24, -6_
Levels(50, 1,920, 0, 1020, coring=false,dither=true).tweak(bright=0, cont=1.00, hue=-0, sat=1.00, coring=false, dither=true).convertbits(8)
#turnRight().Histogram().TurnLeft()
normalize(0.8912)
Trim(415, 5286)
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Old 11th July 2024, 10:04   #2859  |  Link
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Test with some player options:
Code:
                            r4073               r4066
                      ------------------  ------------------
Player                without  Normalize  without  Normalize
--------------------- -------  ---------  -------  ---------
mpc-hc 2.3.2 DirectS    Ok       Noise     Noise     Noise
mpc-hc 2.3.2 + SaneAR   Ok        Ok        Ok       Noise
VirtualDub2 2.1.1.607   Ok        Ok        Ok       Noise
mpc-be 1.7.1 MPC-Aud    Ok        Ok        Ok        Ok
mpc-be 1.7.1 DirectS    Ok        Ok        Ok       Noise
vlc 3.0.21              Ok        Ok        Ok       Noise
@pinterf
Maybe we need a new test15 and/or a new release 3.7.4, the 3.7.3 is 1 year old.
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Last edited by tebasuna51; 11th July 2024 at 10:12.
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Old 11th July 2024, 20:02   #2860  |  Link
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Between r4066 and r4073 these were the changes:
Code:
Add cache for string heap to avoid duplicates. related to #389
pinterf committed Apr 17, 2024

Add basic ArraySort
pinterf committed Apr 17, 2024

Update documentation and changelog (break/continue)
pinterf committed Apr 10, 2024

SetLogParams defaults to follow doc: stderr and LOG_INFO (mentioned in #391)
pinterf committed Apr 9, 2024

Add "continue" for for/while loops Implement #392
pinterf committed Apr 9, 2024

Fix: "break" to return up-to-date "last" in the expression chain
pinterf committed Apr 9, 2024

fix when building with absolute paths
jopejoe1 authored and qyot27 committed Feb 3, 2024
None of these are a fix for anything audio-related. 1 is related to pkg-config, 3 are related to script syntax constructs, 1 adjusts the defaults for logging, 1 is a documentation change, and 1 relates to string handling.



What everyone is actually comparing here - and a couple of the posts acknowledge this around the periphery without directly calling it out - are different compilers. Ubuntu uses GCC 13, which is unaffected. ClangCL and Intel are reported to be [mostly] unaffected. 3.7.3 was reported to be unaffected; it was built with MSVC 2019.

It's pointing at there being a problem with MSVC 2022, at least as of the time of the last test build having been built. In the time since, maybe the issue has been resolved with an update to the compiler, or it would produce a build of r4073 that's still affected by the same problem, even when all the other compilers produce working builds.
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