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24th September 2024, 23:06 | #341 | Link | |
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AVSRepoGUI // VSRepoGUI - Package Manager for AviSynth // VapourSynth VapourSynth Portable FATPACK || VapourSynth Database |
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25th September 2024, 16:14 | #343 | Link |
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But the plugin is written in cpp...
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AVSRepoGUI // VSRepoGUI - Package Manager for AviSynth // VapourSynth VapourSynth Portable FATPACK || VapourSynth Database |
27th September 2024, 12:19 | #344 | Link |
German doom9/Gleitz SuMo
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But it is for a VapourSynth plugin interface. The AviSynth plugin interface is probably different, I would assume.
I saw that the AviSynthPlus repo comes with sources of the Shibatch Sampling Rate Converter (SSRC), that might be a convenient base to derive from. But I also noticed that I won't be able to do that, with all the C++ interface syntax I am not used to, and related compiler control files... |
1st October 2024, 09:04 | #345 | Link | |
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I am annoyed by plugins in ffmpeg AviSynth/VapourSynt opened as shared AviSynth/VapourSynt DLL. In AviSynth_c you have all the functions you can open but warmigs are. Code:
avisynth_c.cpp: In function 'int avs_is_yv24(const AVS_VideoInfo*)': avisynth_c.cpp:66:63: warning: bitwise operation between different enumeration types '<unnamed enum>' and '<unnamed enum>' is deprecated [-Wdeprecated-enum-enum-conversion] 66 | return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV24 & AVS_CS_PLANAR_FILTER); | ~~~~~~~~~~~~^~~~~~~~~~~~~~~~~~~~~~ Code:
static int avisynth_create_stream_audio(AVFormatContext *s, AVStream *st) { AviSynthContext *avs = (AviSynthContext *)s->priv_data; st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO; st->codecpar->sample_rate = avs->vi->audio_samples_per_second; st->codecpar->ch_layout.nb_channels = avs->vi->nchannels; st->duration = avs->vi->num_audio_samples; avpriv_set_pts_info(st, 64, 1, avs->vi->audio_samples_per_second); if (avs_get_version(avs->clip) >= 10) av_channel_layout_from_mask(&st->codecpar->ch_layout, avs_get_channel_mask(avs->vi)); switch (avs->vi->sample_type) { case AVS_SAMPLE_INT8: st->codecpar->codec_id = AV_CODEC_ID_PCM_U8; break; case AVS_SAMPLE_INT16: st->codecpar->codec_id = PCM(S16); break; case AVS_SAMPLE_INT24: st->codecpar->codec_id = PCM(S24); break; case AVS_SAMPLE_INT32: st->codecpar->codec_id = PCM(S32); break; case AVS_SAMPLE_FLOAT: st->codecpar->codec_id = PCM(F32); break; default: av_log(s, AV_LOG_ERROR, "unknown AviSynth sample type %d\n", avs->vi->sample_type); avs->error = 1; return AVERROR_UNKNOWN; } return 0; } https://www.sendspace.com/file/znalx3 Last edited by Jamaika; 1st October 2024 at 11:00. |
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10th October 2024, 14:25 | #347 | Link |
German doom9/Gleitz SuMo
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A little teaser of what I would like to get implemented as AviSynth audio plugin; it already works as CLI application:
I wrote that in FreePascal (Lazarus). I cannot write it in C++. Last edited by LigH; 10th October 2024 at 14:28. |
14th October 2024, 06:30 | #348 | Link |
German doom9/Gleitz SuMo
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I am now halfway successful. Could build plugins in MSYS2/MinGW and GCC 14.2 (M-AB-S interactive shell).
Only flaw: Despite linker flags containing parameters to statically link some GCC libs, the resulting plugins still depend on them. CMakeCache contains Code:
CMAKE_MODULE_LINKER_FLAGS:STRING=-D_FORTIFY_SOURCE=2 -fstack-protector-strong -mtune=generic -O2 -pipe -D__USE_MINGW_ANSI_STDIO=1 -static-libgcc -static-libstdc++ CMAKE_SHARED_LINKER_FLAGS:STRING=-D_FORTIFY_SOURCE=2 -fstack-protector-strong -mtune=generic -O2 -pipe -D__USE_MINGW_ANSI_STDIO=1 -static-libgcc -static-libstdc++ _ PS: Even when adding those (and libwinpthread-1.dll) to the plugins directory, it is still not useable as AviSynth plugin. Last edited by LigH; 14th October 2024 at 07:51. |
15th October 2024, 08:44 | #350 | Link |
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Hi LigH, congrats !
I did not test it yet, so just a tip about the wording to avoid ambiguities: In audio the terms "Limiter, Compressor" are reserved for devices/algorithms which regulate gain and avoid distortion at any cost (the regulation phase excluded, here a carefully chosen lookahead/time constant combo may serve to make distortion unnoticeable) Going from the graphs you published I would guess that your algo remaps sample values, not gain, so can be seen now as a soft clipper with selectable curves which is in any case nice to have. I see arbitrary curves coming, or introduction of a sidechain input. In the end you will be on the way to LigH's Arbitrary Audio Modulator allowing stranger audio FX like ring modulation etc.
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"To bypass shortcuts and find suffering...is called QUALity" (Die toten Augen von Friedrichshain) "Data reduction ? Yep, Sir. We're that issue working on. Synce invntoin uf lingöage..." Last edited by Emulgator; 15th October 2024 at 08:53. |
15th October 2024, 10:14 | #353 | Link |
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I don't know if it works correctly but you can test it.
https://www.sendspace.com/file/1vnpbw AVISource("AudioBoost.avi") # resize the dimensions of the video frame to 320x240 LanczosResize(320, 240) ConvertAudioToFloat() AudioBoost(4.0, 0.95, 1, true) |
15th October 2024, 19:39 | #354 | Link |
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Is there any modern filter to dehalo properly?
I have tried many and some don't accept YV24 or they simply work so and so. Any help is welcome. P.S: Tried FineDehalo and DeHaloAlpha and, oh my god, parameters are such a mess for me.
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@turment on Telegram Last edited by tormento; 15th October 2024 at 21:14. |
18th October 2024, 13:47 | #359 | Link | |
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I think I also denoised chroma (for DeRainbow) so FFT3D really came in handy here. I'd share a clip, but I need to rebuild my desktop, which will take me some months. (Waiting for Black Friday or similar good deal on some parts) Last edited by takla; 18th October 2024 at 13:50. |
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