From mboxrd@z Thu Jan 1 00:00:00 1970 Return-Path: Received: from ffbox0-bg.mplayerhq.hu (ffbox0-bg.ffmpeg.org [79.124.17.100]) by master.gitmailbox.com (Postfix) with ESMTP id B0FF840C5F for ; Fri, 8 Apr 2022 18:40:10 +0000 (UTC) Received: from [127.0.1.1] (localhost [127.0.0.1]) by ffbox0-bg.mplayerhq.hu (Postfix) with ESMTP id 1ADA768B221; Fri, 8 Apr 2022 21:40:07 +0300 (EEST) Received: from mail-yb1-f175.google.com (mail-yb1-f175.google.com [209.85.219.175]) by ffbox0-bg.mplayerhq.hu (Postfix) with ESMTPS id DAF6168ABA7 for ; Fri, 8 Apr 2022 21:39:59 +0300 (EEST) Received: by mail-yb1-f175.google.com with SMTP id j2so16708716ybu.0 for ; Fri, 08 Apr 2022 11:39:59 -0700 (PDT) DKIM-Signature: v=1; a=rsa-sha256; c=relaxed/relaxed; d=gmail.com; s=20210112; h=mime-version:references:in-reply-to:from:date:message-id:subject:to; bh=EbLKgV6GYwM0rNGGL/BhanXu5Fr8N8g2j7DTmO2h37g=; b=lgGLBUx530RfwAUXO/pU1mW9+HVyYRaL0Xb5ycPodn707iQRHuXA4aQGUdDpDM0yWU jI7lsrB2p+EjFVxniSivht2LZyoEVaR+Srj0HiMNIAYeQSmNAh5FC+aiY11x4EN1NtU6 hy2jZA1Iy9JC+ttQNlZBFBkWkw59tpyzomL7VKsSl5+U3p8Bfri1LSZQHJ7JbGKH6Rty GX623/3nkkB+FH79a/AtUafYRI3Aj0Mjnu9Lp1vxJqn81Y13z+UAPXSMdwrGZL1wbrOK /FgiCs6gkZ41OaNFqjv1vn8iycdhHH1An/DyyE9Hltlsr3W7Z9pWl7VO6FlVI+yUJGa2 q7JQ== X-Google-DKIM-Signature: v=1; a=rsa-sha256; c=relaxed/relaxed; d=1e100.net; s=20210112; h=x-gm-message-state:mime-version:references:in-reply-to:from:date :message-id:subject:to; bh=EbLKgV6GYwM0rNGGL/BhanXu5Fr8N8g2j7DTmO2h37g=; b=GOI3xMnk8M8ed8HwKdXsfdsY9mfN0KXZY1wEuRBltApHCQWnL3SH/qKmSLfCe+6cua /g7O0ejWgXmZ6wWKgeAZTffE0qrRdyO8waT5CVzk3xjSv/Ij6zgrzE6DkwbqqNZEEdZ/ nMrt7HovirumCo+IYycHi64BIvdhgcnjWSpKN6hUT0Pj2HCcdApGmwm0uzUIjYCQfivq HIDlFCVgq7Q87shWI5aVr6kh0wXYmmnalr7ZGbu+gzXHhtXLYKLVjrcdP/mv7ejFQ6ak Cc32PBCvZEQT2iWk5FkO89qU/jag4hBh2fYd5wcAO3ZO+byGlppY+wPmVCxrUZ47U7II aGfg== X-Gm-Message-State: AOAM531Sxviw5iIiw2YE/16ViEUTkx7LA6k1cyY9FxE17Ckj2eq0MAFO epNAMuxzP1jwOr9CSWfbAyv2Yd/4OWqhqGu6CCVxpl+9 X-Google-Smtp-Source: ABdhPJxSSdhZd6NCYGcDO9HaH6mYWILinR+ELPzqHmab4ywkPvx2kqoVk+LHh3AzZ0oDOTZSS4N+kiHZen6DvQWLDKY= X-Received: by 2002:a05:6902:50a:b0:63d:b158:a306 with SMTP id x10-20020a056902050a00b0063db158a306mr14788016ybs.571.1649443198324; Fri, 08 Apr 2022 11:39:58 -0700 (PDT) MIME-Version: 1.0 References: <20220327060800.3732289-1-wangcao@google.com> <2ec3971d-f698-1369-1ace-eac613afc980@passwd.hu> In-Reply-To: From: Paul B Mahol Date: Fri, 8 Apr 2022 20:42:10 +0200 Message-ID: To: FFmpeg development discussions and patches X-Content-Filtered-By: Mailman/MimeDel 2.1.29 Subject: Re: [FFmpeg-devel] [PATCH] avfilter/alimiter: Add "flush_buffer" option to flush the remaining valid data to the output X-BeenThere: ffmpeg-devel@ffmpeg.org X-Mailman-Version: 2.1.29 Precedence: list List-Id: FFmpeg development discussions and patches List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Reply-To: FFmpeg development discussions and patches Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: 7bit Errors-To: ffmpeg-devel-bounces@ffmpeg.org Sender: "ffmpeg-devel" Archived-At: List-Archive: List-Post: On Thu, Apr 7, 2022 at 11:56 PM Wang Cao wrote: > On Thu, Apr 7, 2022 at 12:44 AM Paul B Mahol wrote: > > > On Wed, Apr 6, 2022 at 1:49 PM Paul B Mahol wrote: > > > > > > > > > > > On Tue, Apr 5, 2022 at 8:57 PM Wang Cao < > > wangcao-at-google.com@ffmpeg.org> > > > wrote: > > > > > >> On Mon, Apr 4, 2022 at 3:28 PM Marton Balint wrote: > > >> > > >> > > > >> > > > >> > On Mon, 4 Apr 2022, Paul B Mahol wrote: > > >> > > > >> > > On Sun, Mar 27, 2022 at 11:41 PM Marton Balint > > wrote: > > >> > > > > >> > >> > > >> > >> > > >> > >> On Sat, 26 Mar 2022, Wang Cao wrote: > > >> > >> > > >> > >>> The change in the commit will add some samples to the end of the > > >> audio > > >> > >>> stream. The intention is to add a "zero_delay" option eventually > > to > > >> not > > >> > >>> have the delay in the begining the output from alimiter due to > > >> > >>> lookahead. > > >> > >> > > >> > >> I was very much suprised to see that the alimiter filter actually > > >> delays > > >> > >> the audio - as in extra samples are inserted in the beginning and > > >> some > > >> > >> samples are cut in the end. This trashes A-V sync, so it is a bug > > >> IMHO. > > >> > >> > > >> > >> So unless somebody has some valid usecase for the legacy way of > > >> > operation > > >> > >> I'd just simply change it to be "zero delay" without any > additional > > >> user > > >> > >> option, in a single patch. > > >> > >> > > >> > > > > >> > > > > >> > > This is done by this patch in very complicated way and also it > > really > > >> > > should be optional. > > >> > > > >> > But why does it make sense to keep the current (IMHO buggy) > > operational > > >> > mode which adds silence in the beginning and trims the end? I > > understand > > >> > that the original implementation worked like this, but libavfilter > has > > >> > packet timestamps and N:M filtering so there is absolutely no reason > > to > > >> > use an 1:1 implementation and live with its limitations. > > >> > > > >> Hello Paul and Marton, thank you so much for taking time to review my > > >> patch. > > >> I totally understand that my patch may seem a little bit complicated > > but I > > >> can > > >> show with a FATE test that if we set the alimiter to behave as a > > >> passthrough filter, > > >> the output frames will be the same from "framecrc" with my patch. The > > >> existing > > >> behavior will not work for all gapless audio processing. > > >> > > >> The complete patch to fix this issue is at > > >> > > >> > > > https://patchwork.ffmpeg.org/project/ffmpeg/patch/20220330210314.2055201-1-wangcao@google.com/ > > >> > > >> Regarding Paul's concern, I personally don't have any preference > whether > > >> to > > >> put > > >> the patch as an extra option or not. With respect to the > implementation, > > >> the patch > > >> is the best I can think of by preserving as much information as > possible > > >> from input > > >> frames. I also understand it may break concept that "filter_frame" > > outputs > > >> one frame > > >> at a time. For alimiter with my patch, depending on the size of the > > >> lookahead buffer, > > >> it may take a few frames before one output frame can be generated. > This > > is > > >> inevitable > > >> to compensate for the delay of the lookahead buffer. > > >> > > >> Thanks again for reviewing my patch and I'm looking forward to hearing > > >> from > > >> you :) > > >> > > > > > > Better than (because its no more 1 frame X nb_samples in, 1 frame X > > > nb_samples out) to replace .filter_frame/.request_frame with .activate > > > logic. > > > > > > And make this output delay compensation filtering optional. > > > > > > In this process make sure that output PTS frame timestamps are > unchanged > > > from input one, by keeping reference of needed frames in filter queue. > > > > > > Look how speechnorm/dynaudnorm does it. > > > > > > > > > Alternatively, use current logic in ladspa filter, (also add option with > > same name). > > > > This would need less code, and probably better approach, as there is no > > extra latency introduced. > > > > Than mapping 1:1 between same number of samples per frame is not hold any > > more, but I think that is not much important any more. > > > Thank you for replying to me with your valuable feedback! I have checked > af_ladspa > and the "request_frame" function in af_ladspa looks similar to what I'm > doing. The > difference comes from the fact that I need an internal frame buffer to keep > track of > output frames. Essentially I add a frame to the internal buffer when an > input frame > comes in. The frames in this buffer will be the future output frames. We > start writing > these output frame data buffers only when the internal lookahead buffer has > been filled. > When there is no more input frames, we flushing all the remaining data in > the internal > frame buffers and lookahead buffers. Can you advise on my approach here? > Thank you! > > I can put my current implementation of "filter_frame" and "request_frame" > into the "activate" approach as you suggested with speechnorm/dynaudnorm. > No need to wait for all buffers to fill up, only lookahead buffer. Just trim initial samples that is size of lookahead, and than start outputing samples just once you get whatever modulo of current frame number of samples. So there should not be need for extra buffers to keep audio samples. Just buffers to keep input pts and number of samples of previous input frames, like in ladspa filter. > > -- > > Wang Cao | Software Engineer | wangcao@google.com | 650-203-7807 > _______________________________________________ > ffmpeg-devel mailing list > ffmpeg-devel@ffmpeg.org > https://ffmpeg.org/mailman/listinfo/ffmpeg-devel > > To unsubscribe, visit link above, or email > ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe". > _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-devel To unsubscribe, visit link above, or email ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe".