I need to access the H264 bitstream without reading many gigabytes from the file at once.

Would theAVPacket returned from av_read_frame() in a loop provide me the h264 NAL units?

I don't know how it worked but I had a mp4 file from a HLS stream that was just a blob - no index, no ATOM. It was unplayable and did not show ANY evidence of containing any video. Even if I wanted to use untrunc to build an ATOM index, this was the only file from the source and I had no other reference of a good mp4 file from the same encoding device (i.e. Camera or streaming platform). So I just tried a vertical video I had of the walk around of my car after an accident recorded on my iPhone 15 Pro (iOS 26) to be my good mp4.

The bad file was 5 hrs long at 10 GB and the my vertical video was 9MB at 22 seconds. Yeah, no way this should have worked... and it didn't, I thought at first. untrunc spat out the first 12 seconds of the bad file as a new mp4 rotated 90 degrees. But the recovered file was actually playable! Since the first 12 seconds contained the identical frames as the bad file, I rotated the video -90 degrees and used this as the good file and fed it back into untrunc.

It worked!! All 5 hours of the original video was regenerated. It isn't perfect though. The video is jumpy in QuickTime, so I need to play it on VLC to ignore the encoder lag, and the audio does get misaligned if I jog the progress bar. But at least, the file can be played back.

I have no idea how the first untrunc even worked. But don't give up hope everyone.

For those using FFmpeg in production, how much of your encoding is still software-based vs hardware-accelerated? We’re NETINT, and every year we run the State of Video Encoding survey. We’re collecting responses for the 2027 edition now, and one of the things we’re looking at is how teams are actually using CPUs, GPUs, VPUs, and other hardware acceleration in their encoding workflows. The survey also covers codecs, encoding ladders, AI in the workflow, and cost per channel. If you’re curious, the 2026 results are public here, no form required:
netint.com/2026-state-of-video-encoding-early-access

A couple of things that stood out last year: H.264 was still in production at 84% of teams, while AV1 was at 17%. We also found that budget and team capacity were reported much more often as barriers than technical issues.

The 2027 survey takes about 10 minutes, and everyone who completes it gets the full report before it’s publicly released. Would be interesting to hear what people here are using too, especially where hardware encoding has or hasn’t made sense in your FFmpeg setup.
netint.biz/encoding-survey

Automated ffmpeg concat scripts fall apart fast when the main character morphs between consecutive five-seconds clips.I am wiring up a local assembly pipeline for a faceless storytelling channel and testing seedance 2.5 through dreamina for the character shots .

The $0.29 first month offer caught my attention, but the bigger issue for this workflow is how many rerolls I would need across twenty scenes before the clips are consistent enough to assemble automatically. To keep those rerolls down, is it better to feed a clean three- angle character sheet into the reference slots or pass the cropped final PNG frame from the previous clip ?

When I submit my film to a distributor, the system gives an error that I am missing one black frame at the beginning and at the end of the film. I have tried to tpad and other options, which did not work but I thought for sure the concat would work

First I create a solid 3840x2160 black colored png file that had the same 4K UHD HDR format as my mp4 file and saved it as black-frame.png

Followed by using the following to convert it to match the video:
ffmpeg -framerate 23.976 -i black-frame.png -c:v libx265 -x265-params "crf=0:vbv-maxrate=50000:vbv-bufsize=250000:colorprim=bt2020:transfer=smpte2084:colormatrix=bt2020nc:master-display=G(13250,34500)B(7500,3000)R(34000,16000)WP(15635,16450)L(10000000,1):max-cll=475,385:ref=2" -pix_fmt yuv420p10le -vsync 1 -b:v 59.18M black-frame.mp4

In a text file called inputs.txt:
file 'black-frame.mp4'

file 'Not-if-ac3.mp4'

file 'black-frame.mp4'

Then I used the concat command to bring them together:
ffmpeg -f concat -safe 0 -i inputs.txt -c copy Not-if-aac-w-black-frames.mp4

But this reduces the video from an overall bit rate of 59.5 MB/s to 29.5 MB/s and the framerate changed as well from 23.976 to 11.976 - I thought the -c copy was supposed to not change anything in the video quality

I have also tried:
ffmpeg -i Not-if-aac.mp4 -filter_complex "[0:v]tpad=start=1:stop=1:color=black[v]" -map "[v]" -map 0:a? -c:v libx265 -crf 0 -c:a copy Not-if-aac-w-black-frames.mp4

But then it loses it's HDR format and it changes the audio from AC3 to AAC - see second screenshot

Why isn't the "copy" working?

Hello everyone,

I am a new user of Shutter Encoder and I run into a problem when converting multiple files. I have a list of over 1k files I want to convert and leave the proces alone ( understandibly) but if one file in the list doesn't convert the whole process stops. This is annoying because I was hoping to leave the PC alone and come back to it a couple of days later and have the process done.

What am I doing wrong?

I posted this to the VLC subreddit, but it was suggested this might be a better place.

I have a video where I needed to fix the audio (the recording has no low end at all). I saved a good audio preset in the equalizer. Can someone please help me with a command to convert/save that video with the applied audio EQ?

Hey everyone, I wanted to share a major open-source development milestone that handles a complex video pipeline challenge: achieving proper Dolby Vision Profile 7 FEL (Full Enhancement Layer) reconstruction on Linux/x86 hardware without proprietary Dolby blobs or SDK constraints.

As many of you know, playing back Profile 7 FEL on PC clients usually means the pipeline strips the enhancement layer, falls back to basic HDR10, or relies on heavy software tone-mapping. A developer named CroqueMr (u/Croque_Mr) has broken through this barrier by integrating native Dolby Vision support directly into an x86 LibreELEC 13 / Kodi 22 RC1 distribution.

The Open-Source Engine & Software Patches

The project achieves this by introducing an open-source Dolby Vision processing engine called CB1 0.3, which handles the reconstruction and transport packing to output a native TV-Led RGB8 tunnel. To make this work natively in the Kodi VideoPlayer/DRM output path without external players, the developer successfully cleaned up and implemented specific patches across the underlying media stack:

• FFmpeg (3 Patches): Handles metadata preservation and AV1 container recognition.
• libplacebo (2 Patches): Focuses on OpenGL subgroup discovery and guarded affine dispatch.
• mpv/Linux display drivers (10 Patches each): Covers frame-associated metadata, display restoration, DRM/KMS validation, and HDMI signaling.

Brute-Forcing "Bit-Perfect" Fidelity with x86 Headroom

Because modern PC hardware has significantly more processing and memory bandwidth headroom than standard ARM-based streaming boxes, the build features a "Dolby Vision Master Reference" mode. It processes playback in full precision with high-end filtering to achieve a bit-perfect Dolby Vision picture, preserving original pixels and metadata as accurately as possible.

Impressive Efficiency on Budget Hardware (Intel N100)

The performance optimization across the video pipeline for low-power architectures is incredibly impressive:

• Intel N100 Mini PCs: The dev confirmed that a cheap Intel N100 box handles a 4K Profile 7 FEL remux perfectly with zero dropped frames.
• Performance Headroom: In bench testing, the N100 successfully processed a heavy FEL remux at 40fps, showing plenty of performance headroom over standard 24fps film files, and can push Dolby Vision up to 60fps using native TV-led tunneling.
• Current Hardware Support: Out-of-the-box support for Intel iGPUs (8th gen and newer, N-series, Core Ultra) and AMD iGPUs (Ryzen 2000G up to AI 300). Dedicated AMD Radeon and Nvidia GeForce GPUs are supported but currently experimental.

Project links

• Main development thread:  https://forum.libreelec.tv/thread/30362-dolby-vision-w-fel-3d-for-pc-amd-intel-nvidia/
• GitHub (Source Code & Releases): https://github.com/CroqueMr/libreelec-x86-DV/
• Original Reddit thread: https://www.reddit.com/r/libreeLEC/comments/1wq20eu/wip_native_dolby_vision_support_on_x86_intel_cpu/

Full credit to u/Croque_Mr for the incredible development work, and to the community members helping test and refine it. If you want to check out the technical code behind the engine or look through the compatible hardware lists, everything is well-documented in the links above!

I not sure if this is OK to post, but I think this program I have been working on, mostly for myself that was really vibe coded with AI, so took many hours to fixing bugs and adding changes to things to follow the way the work flow was being done files I had on my server and also any new video files.

It uses handbrakecli and ffprobe to process a queue of files. You give it a default directory, drag files, or browse to any directory local or on a NAS and it will scan it and let you sort them all by size, it fills in media info per each file and flags files it thinks looks like they could save space by being re-encoded. it then goes through each file and samples them 3 time at different locations in that file and if it determines it wont gain more than a 10% (you can change that in settings) of saving from the re-encode it skips it. You can tell it to force files to be encoded, or have them reanalyzed more times. After each file is encoded, it created a backup, you can restore if needed, it can also auto clean the files (You have to have MKVToolNix installed on system) or you can pick what files to be cleaned, this just removes all tags that are not needed from the files, gets rid of ugly tags. There is a also a rename tab you can send the files to and it will help you rename selected files or all files in a same directory using quick ways that are helpful to common video files.

The program basically streamlines the process I usually did manually. I find that I have really used it a lot. I made it because I had a need. Not sure if something like existed already, but I needed this and worked to make it do what I needed.

Feel free to check it out if you think it would help, I would love to hear if anyone else used it or found it helpful.

https://github.com/jsdeprey/MKVEncoderQ

I feel like I kind of already know the answer to this question, but I just wanted a sanity check.

I have been encoding my lossless audio to AC-3 5.1. @640kbps for both compatibility and space saving purposes. If a AC-3 core/lossy track already exists, I just use that. I know it is not as good as passthrough but for my setup and to my ears, unless I am doing an AxB comparison, it’s good enough.

Historically, what I’ve done is decode the track to PCM, split the channels and run them through DME to make an “official” AC-3 track. I am happy with the results, but I’d be lying if I said the process wasn’t tedious, especially when doing multiple encodes at a time!

I remember reading a LONG time ago that “libavcodec” was mediocre at encoding AC-3, which is what lead me to use DME. But, I was wondering if that is still true today or if the community considers it “good enough”.

It’s not a make or break, but I would love to be able to eliminate one extra step in my work flow and encode the track in program as opposed to having to take the extra steps to do so in another one, you know?

[ Removed by Reddit on account of violating the content policy. ]

Never used ffmpeg through the command line before, but I have a niche case where I need to expand 2.0 stereo into 7.1 audio but with all the extra channels muted for compatibility reasons (I need to merge the log lady intros with the episodes in Twin Peaks s1&2). I know this is possible because Clever FFMPEG can do it, but it's a slow process there. I'm hoping to speed it up. Idk what I'm doing. I tried to make a .bat but that just immediately opens and closes. Instead I made a bat with just "cmd" that opens the command line and I enter this:

ffmpeg -i input.mkv -c:v copy -filter_complex "pan=7.1|FL=FL|FR=FR|FC=0|LFE=0|BL=0|BR=0|SL=0|SR=0" -c:a wav output.mkv

Which I stole from someone else online and modified but then it spits out this error after parsing. I'm such a noob at this I don't even know how to get it to read my mkv file so I just rename it to "input.mkv" to get ffmpeg to take it.

ffmpeg version 5.1-essentials_build-www.gyan.dev Copyright (c) 2000-2022 the FFmpeg developers built with gcc 12.1.0 (Rev2, Built by MSYS2 project) configuration: --enable-gpl --enable-version3 --enable-static --disable-w32threads --disable-autodetect --enable-fontconfig --enable-iconv --enable-gnutls --enable-libxml2 --enable-gmp --enable-lzma --enable-zlib --enable-libsrt --enable-libssh --enable-libzmq --enable-avisynth --enable-sdl2 --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxvid --enable-libaom --enable-libopenjpeg --enable-libvpx --enable-libass --enable-libfreetype --enable-libfribidi --enable-libvidstab --enable-libvmaf --enable-libzimg --enable-amf --enable-cuda-llvm --enable-cuvid --enable-ffnvcodec --enable-nvdec --enable-nvenc --enable-d3d11va --enable-dxva2 --enable-libmfx --enable-libgme --enable-libopenmpt --enable-libopencore-amrwb --enable-libmp3lame --enable-libtheora --enable-libvo-amrwbenc --enable-libgsm --enable-libopencore-amrnb --enable-libopus --enable-libspeex --enable-libvorbis --enable-librubberband libavutil 57. 28.100 / 57. 28.100 libavcodec 59. 37.100 / 59. 37.100 libavformat 59. 27.100 / 59. 27.100 libavdevice 59. 7.100 / 59. 7.100 libavfilter 8. 44.100 / 8. 44.100 libswscale 6. 7.100 / 6. 7.100 libswresample 4. 7.100 / 4. 7.100 libpostproc 56. 6.100 / 56. 6.100 Input #0, matroska,webm, from 'input.mkv': Metadata: title : Twin Peaks: The Entire Mystery - Disc 2 encoder : libmakemkv v2.0.0 (1.3.10/1.5.2) win(x64-release) creation_time : 2026-10-22T03:14:37.000000Z Duration: 00:00:44.88, start: 0.000000, bitrate: 11866 kb/s Stream #0:0(eng): Video: h264 (High), yuv420p(progressive), 1920x1080 [SAR 1:1 DAR 16:9], 23.98 fps, 23.98 tbr, 1k tbn Metadata: BPS-eng : 11633678 DURATION-eng : 00:00:44.878166666 NUMBER_OF_FRAMES-eng: 1076 NUMBER_OF_BYTES-eng: 65262026 SOURCE_ID-eng : 001011 _STATISTICS_WRITING_APP-eng: MakeMKV v2.0.0 win(x64-release) _STATISTICS_WRITING_DATE_UTC-eng: 2026-10-22 03:14:37 _STATISTICS_TAGS-eng: BPS DURATION NUMBER_OF_FRAMES NUMBER_OF_BYTES SOURCE_ID Stream #0:1(eng): Audio: ac3, 48000 Hz, stereo, fltp, 192 kb/s (default) Metadata: title : Stereo BPS-eng : 192000 DURATION-eng : 00:00:44.928000000 NUMBER_OF_FRAMES-eng: 1404 NUMBER_OF_BYTES-eng: 1078272 SOURCE_ID-eng : 001100 _STATISTICS_WRITING_APP-eng: MakeMKV v2.0.0 win(x64-release) _STATISTICS_WRITING_DATE_UTC-eng: 2026-10-22 03:14:37 _STATISTICS_TAGS-eng: BPS DURATION NUMBER_OF_FRAMES NUMBER_OF_BYTES SOURCE_ID Stream #0:2(eng): Subtitle: hdmv_pgs_subtitle, 1920x1080 Metadata: BPS-eng : 44867 DURATION-eng : 00:00:37.057854166 NUMBER_OF_FRAMES-eng: 20 NUMBER_OF_BYTES-eng: 207831 SOURCE_ID-eng : 001200 _STATISTICS_WRITING_APP-eng: MakeMKV v2.0.0 win(x64-release) _STATISTICS_WRITING_DATE_UTC-eng: 2026-10-22 03:14:37 _STATISTICS_TAGS-eng: BPS DURATION NUMBER_OF_FRAMES NUMBER_OF_BYTES SOURCE_ID [Parsed_pan_0 @ 000002b4d002d700] Expected in channel name, got "" [AVFilterGraph @ 000002b4cf571200] Error initializing filter 'pan' with args '7.1|FL=FL|FR=FR|FC=0|LFE=0|BL=0|BR=0|SL=0|SR=0' Error initializing complex filters. Invalid argument

Hi! F3D maintainer here!

We just released F3D 4.0.0, fast and minimalist 3D viewer. Obviously free and open source (and not vibecoded).

One of the key new feature is ffmpeg compatibility with the possibility to output animated 3d data directly into a video file from the terminal.

f3d animated_file.glb --output-video=- --resolution=600,600 | ffmpeg -i - out.mp4

Let us know what you think!

https://github.com/f3d-app/f3d/releases/tag/v4.0.0

is it possbile .mp4 to support pcm files like .wav and .aiff using -c:a copy?

Hi, I am trying to condense 24 hours of video into (for example) 2 minutes - so a factor of 720. My source is a collection of 10 minute .mkv recordings, (made by AgentDVR from a Wyze Cam v3 via go2rtc). I have tried compressing each video using both of the methods described here
https://trac.ffmpeg.org/wiki/How%20to%20speed%20up%20/%20slow%20down%20a%20video
then concatenating them using ffmpeg -f concat -i list.txt -c copy etc.
The result is a video with jerky motion - each component video is sped up but there seems almost to be a pause between them.

Is there a way to do this and get a smooth output, or is this doomed?

The alternative is to concatenate the raw input videos into one and then compress that. That is probably a better way to do it, but I was hoping to avoid this, as the concatenated file would be at least 10GB.

As part of a GIS (Geographic Information System) workflow that I developed, I have taken GoPro 360° video footage collected with a MAX2 and exported it to MP4. I have also extracted the GPS telemetry from the footage and slimmed the number of camera locations down to a predefined distance interval. Each point has an associated time offset from the start of the MP4.

For a particular MP4, I could be extracting say 200 frames. At present I have 200 ffmpeg lines in a batch file (.bat) that I am running to extract the images. This means that ffmpeg is started, seeks through the MP4 to get to the time offset and extracts the frame, then the process ends. That gets repeated another 199 times.

The time offsets are not at regular intervals. I am presuming that there must be a more efficient way of extracting the frames by only opening the MP4 file once? Hoping someone can shed some light on this for me.

What settings would you recommend for transcoding the lossless audio track to native aac? The video stream would just be copied.

I heard that the native version of aac has been rewritten and is supposed to be much better than it was. I hope it can be transparent at a reasonable bitrate.

As I am building my digital library, I have found that, while some of my devices support THD and DTS, most, if not all, of my devices support FLAC and ALAC. That coupled with the possibility of a 10-15% size reduction in audio stream, I am wondering if there are any drawbacks to converting my lossless audio to FLAC/ALAC. Like am I losing any fancy metadata/audio processing queues that would worsen the listening experience?

I was mixing a couple of audio tracks and used apad on the shorter one so it wouldn't cut off early, together with amix=inputs=2:duration=longest. ffmpeg didn't complain at all, it just kept encoding. By the time I noticed, the WAV was several hundred GB and the disk was almost full.

The reason is that apad with no arguments pads forever. It turns that input into an endless stream, and duration=longest tells amix to keep going until the longest input ends, which is now never. Nothing errors, because from ffmpeg's point of view it's doing exactly what it was asked.

The fix is to give the padding an end. Use apad=whole_dur=90 (total length in seconds) or apad=pad_dur=2 (extra silence after the audio), or put -t 90 on the output. If you just want the mix to end with the longer track, amix=duration=first with that track as the first input works too, and then you don't need apad at all.

A related one if you script ffmpeg in zsh: writing a filter label like [$i:a] inside a loop doesn't do what it looks like, because zsh reads :a as a modifier on the variable. ffmpeg can still exit with 0 and quietly use a different stream than you meant. Writing [${i}:a] avoids it.

Hope this saves someone a full disk.

This ffmpeg issue has recently passed the 10 year open mark. The tldr is that ffmpeg doesn't know how to read the matroska way of mkv embedded webvtt subtitles.

This has major downstream implications, for example jellyfin cant stream these subtitles if they're embedded, because ffmpeg cant read them. Ive seen so so many tools have the default of converting webvtt subtitles into srt or even ass just because webvtt ends up not working with ffmpeg based projects when the subtitles are muxed into an mkv. I've looked at this issue so so many times each time thinking, somebody will get to it eventually. There are even implementations to add support for this out there, but for some reason none of them have been merged. I have thought about tackling the issue myself but am doubtful of my abilities of contributing quality code considering the sheer downstream implications of any changes made to ffmpeg.

Just had to vent about this somewhere after a webvtt to srt conversion lost some important formatting in a movie once again that annoyed me. I could of course do seperate subtitle files but I must say single mkv files are something I value quite a bit. If anyone got some motivation and some pull in the ffmpeg project, I'd be very grateful to see this implemented & merged. Also shoutout to all open source developers who do all this for free, I know I'm not owed anything.

I'm remuxing iOS .MOV videos to .MP4 using ffmpeg. The command I'm using, below, retains the Dolby Vision metadata (just to try to keep as much of the original info as possible). Anyone care to comment on if this seems like a reasonable "best practice" for remuxing these? Any suggestions on other communities to ask?

ffmpeg -n -i "$f" -c copy -tag:v hvc1 -strict unofficial -map_metadata 0 -movflags +use_metadata_tags "$f.mp4"

Hi

I have 2 videos of a natively english movie, one is super high quality video and audio in english. Another is a much watered down video and audio in hindi.

I want to be able to overlay the hindi audio track on the super high quality video as a separate track.

Both the videos have different frame rates and different lengths (difference is in minutes).

I have tried various tools and various permutations and combinations on MKVToolNix but it isnt giving me the result I want.

Any helpful pointers are appreciated!

Seeing a strange issue when transcoding from h.264 files to h.265 files using ffmpeg and the nvidia encoder, whilst the output looks files and plays fine, if you pause the video at any point (I'm using VLC for play back) for a few seconds and resume playback, the video freezes entirely, and even seeking to a new position does not unfreeze it (audio continues to play) - if you seek normally without having paused it then the video loads instantly. You have to stop playback of the file entirely and play it again for the video to come back. The h.264 file before transcoding does not have this issue, only the output h.265 file. The command I am using is: ffmpeg -hwaccel cuda -hwaccel_output_format cuda -i "<input>" -filter:v fps=25 -c:v hevc_nvenc -b_ref_mode each -bf 4 -b_adapt 1 -preset p7 -profile:v main -tier high -tune hq -rc:v vbr -lookahead_level:v 14 -multipass:v 2 -cq:v 36 -c:a copy "<output>", I've tried with combinations of -intra-refresh 1, -single-slice-intra-refresh 1, -no-scenecut 1, -g 120, -forced-idr 1 and -force_key_frames "250" but nothing seems to fix it.

Output from running it:

ffmpeg version 9.0.1-full_build-www.gyan.dev Copyright (c) 2000-2026 the FFmpeg developers                                built with gcc 16.1.0 (Rev2, Built by MSYS2 project)                                                                    configuration: --enable-gpl --enable-version3 --enable-static --disable-w32threads --disable-autodetect --enable-cairo --enable-fontconfig --enable-iconv --enable-gnutls --enable-lcms2 --enable-libxml2 --enable-gmp --enable-bzlib --enable-lzma --enable-libsnappy --enable-zlib --enable-librist --enable-libsrt --enable-libssh --enable-libzmq --enable-avisynth --enable-libbluray --enable-libcaca --enable-libdvdnav --enable-libdvdread --enable-sdl2 --enable-libaribb24 --enable-libaribcaption --enable-libdav1d --enable-libdavs2 --enable-libopenjpeg --enable-libquirc --enable-libuavs3d --enable-libxevd --enable-libzvbi --enable-liboapv --enable-libqrencode --enable-librav1e --enable-libsvtav1 --enable-libvvenc --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs2 --enable-libxeve --enable-libxvid --enable-libaom --enable-libjxl --enable-libsvtjpegxs --enable-libvpx --enable-mediafoundation --enable-libass --enable-frei0r --enable-libfreetype --enable-libfribidi --enable-libharfbuzz --enable-liblensfun --enable-libvidstab --enable-libvmaf --enable-libzimg --enable-amf --enable-cuda-llvm --enable-cuvid --enable-dxva2 --enable-d3d11va --enable-d3d12va --enable-ffnvcodec --enable-libvpl --enable-nvdec --enable-nvenc --enable-vaapi --enable-vulkan --enable-libplacebo --enable-opencl --enable-libcdio --enable-openal --enable-libgme --enable-libmodplug --enable-libopenmpt --enable-libopencore-amrwb --enable-libmp3lame --enable-libshine --enable-libtheora --enable-libtwolame --enable-libvo-amrwbenc --enable-libcodec2 --enable-libilbc --enable-libgsm --enable-liblc3 --enable-libopencore-amrnb --enable-libopus --enable-libspeex --enable-libvorbis --enable-ladspa --enable-libbs2b --enable-libflite --enable-libmysofa --enable-librubberband --enable-libsoxr --enable-chromaprint --enable-whisper                                                                                                   libavutil      61.  1.101 / 61.  1.101                                                                                  libavcodec     63.  1.101 / 63.  1.101                                                                                  libavformat    63.  1.101 / 63.  1.101                                                                                  libavdevice    63.  1.101 / 63.  1.101                                                                                  libavfilter    12.  1.101 / 12.  1.101                                                                                  libswscale     10.  1.101 / 10.  1.101                                                                                  libswresample   7.  1.101 /  7.  1.101                                                                                Input #0, matroska,webm, from 'D:\file.mkv':                     Metadata:                                                                                                                 encoder         : libebml v1.3.4 + libmatroska v1.4.5                                                                   creation_time   : 2017-08-05T00:18:14.000000Z                                                                         Duration: 00:28:35.96, start: 0.000000, bitrate: 3707 kb/s                                                              Stream #0:0: Video: h264 (High), yuv420p(progressive), 1280x720, SAR 1:1 DAR 16:9, 25 fps, 25 tbr, 1k tbn (default)       Metadata:                                                                                                                 BPS             : 3449048                                                                                               BPS-eng         : 3449048                                                                                               DURATION        : 00:28:35.960000000                                                                                    DURATION-eng    : 00:28:35.960000000                                                                                    NUMBER_OF_FRAMES: 42899                                                                                                 NUMBER_OF_FRAMES-eng: 42899                                                                                             NUMBER_OF_BYTES : 739803566                                                                                             NUMBER_OF_BYTES-eng: 739803566                                                                                          _STATISTICS_WRITING_APP: mkvmerge v14.0.0 ('Flow') 64bit                                                                _STATISTICS_WRITING_APP-eng: mkvmerge v14.0.0 ('Flow') 64bit                                                            _STATISTICS_WRITING_DATE_UTC: 2017-08-05 00:18:14                                                                       _STATISTICS_WRITING_DATE_UTC-eng: 2017-08-05 00:18:14                                                                   _STATISTICS_TAGS: BPS DURATION NUMBER_OF_FRAMES NUMBER_OF_BYTES                                                         _STATISTICS_TAGS-eng: BPS DURATION NUMBER_OF_FRAMES NUMBER_OF_BYTES                                                 Stream #0:1(eng): Audio: mp3 (mp3float), 48000 Hz, stereo, fltp, 128 kb/s (default)                                       Metadata:                                                                                                                 BPS             : 128000                                                                                                BPS-eng         : 128000                                                                                                DURATION        : 00:28:35.928000000                                                                                    DURATION-eng    : 00:28:35.928000000                                                                                    NUMBER_OF_FRAMES: 71497                                                                                                 NUMBER_OF_FRAMES-eng: 71497                                                                                             NUMBER_OF_BYTES : 27454848                                                                                              NUMBER_OF_BYTES-eng: 27454848                                                                                           _STATISTICS_WRITING_APP: mkvmerge v14.0.0 ('Flow') 64bit                                                                _STATISTICS_WRITING_APP-eng: mkvmerge v14.0.0 ('Flow') 64bit                                                            _STATISTICS_WRITING_DATE_UTC: 2017-08-05 00:18:14                                                                       _STATISTICS_WRITING_DATE_UTC-eng: 2017-08-05 00:18:14                                                                   _STATISTICS_TAGS: BPS DURATION NUMBER_OF_FRAMES NUMBER_OF_BYTES                                                         _STATISTICS_TAGS-eng: BPS DURATION NUMBER_OF_FRAMES NUMBER_OF_BYTES
Stream mapping:
  Stream #0:0 -> #0:0 (h264 (native) -> hevc (hevc_nvenc))
  Stream #0:1 -> #0:1 (copy)
Press [q] to stop, [?] for help
Output #0, mp4, to 'D:\output.mp4':
  Metadata:
    encoder         : Lavf63.1.101
  Stream #0:0: Video: hevc (Main) (hev1 / 0x31766568), nv12(tv, progressive), 1280x720 [SAR 1:1 DAR 16:9], q=2-31, 25 fps, 12800 tbn (default)
    Metadata:
      encoder         : Lavc63.1.101 hevc_nvenc
      BPS             : 3449048
      BPS-eng         : 3449048
      DURATION        : 00:28:35.960000000
      DURATION-eng    : 00:28:35.960000000
      NUMBER_OF_FRAMES: 42899
      NUMBER_OF_FRAMES-eng: 42899
      NUMBER_OF_BYTES : 739803566
      NUMBER_OF_BYTES-eng: 739803566
      _STATISTICS_WRITING_APP: mkvmerge v14.0.0 ('Flow') 64bit
      _STATISTICS_WRITING_APP-eng: mkvmerge v14.0.0 ('Flow') 64bit
      _STATISTICS_WRITING_DATE_UTC: 2017-08-05 00:18:14
      _STATISTICS_WRITING_DATE_UTC-eng: 2017-08-05 00:18:14
      _STATISTICS_TAGS: BPS DURATION NUMBER_OF_FRAMES NUMBER_OF_BYTES
      _STATISTICS_TAGS-eng: BPS DURATION NUMBER_OF_FRAMES NUMBER_OF_BYTES
    Side data:
      CPB properties: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: N/A
  Stream #0:1(eng): Audio: mp3 (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 128 kb/s (default)
    Metadata:
      BPS             : 128000
      BPS-eng         : 128000
      DURATION        : 00:28:35.928000000
      DURATION-eng    : 00:28:35.928000000
      NUMBER_OF_FRAMES: 71497
      NUMBER_OF_FRAMES-eng: 71497
      NUMBER_OF_BYTES : 27454848
      NUMBER_OF_BYTES-eng: 27454848
      _STATISTICS_WRITING_APP: mkvmerge v14.0.0 ('Flow') 64bit
      _STATISTICS_WRITING_APP-eng: mkvmerge v14.0.0 ('Flow') 64bit
      _STATISTICS_WRITING_DATE_UTC: 2017-08-05 00:18:14
      _STATISTICS_WRITING_DATE_UTC-eng: 2017-08-05 00:18:14
      _STATISTICS_TAGS: BPS DURATION NUMBER_OF_FRAMES NUMBER_OF_BYTES
      _STATISTICS_TAGS-eng: BPS DURATION NUMBER_OF_FRAMES NUMBER_OF_BYTES

Using ffmpeg 9.0.1 on windows (with VLC 3.0.24 on windows for playback). Any idea as to why this is happening and how to fix it?

Basically what the title says is there any video editor that I can use to switch the main audio track to the language I want from a video and then convert that video to mkv or mp4?

I tried one called avidemux but the videos glitched or lagged in some parts 😪

I always and i mean always see these types of clips online that are very clearly very low quality, but they dont really have the noisy low quality look. They instead look really bland, lacking contrast and depth. Its a specific look i adore somehow and i want to know what exactly achieves this look and if its possible to recreate in ffmpeg. If your not sure what command i should use to recreate it, then sharing info on what compression method this is would be fine too.