ChanServ changed the topic of #ffmpeg to: Welcome to the FFmpeg USER support channel | Development channel: #ffmpeg-devel | Bug reports: https://ffmpeg.org/bugreports.html | Wiki: https://trac.ffmpeg.org/ | This channel is publically logged | FFmpeg 7.1.1 is released
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<Guest10> Hello, I am trying to encode 3rd Order Ambisonic Wav file to an IAMF file using FFmpeg. It works fine using Opus codec, but when using PCM, I can’t decode the created .iamf file. Any ideas why this is the case?
<Guest10> Using a webdemo decoder introduced here.
<Guest10> The encode command is as below. I changed libopus to pcm_s16le when trying.
<Guest10> ffmpeg -i foo.wav \
<Guest10>     -c:v copy \
<Guest10>     -filter_complex "[0:a]channelmap=0:mono[A0];[0:a]channelmap=1:mono[A1];[0:a]channelmap=2:mono[A2];[0:a]channelmap=3:mono[A3];[0:a]channelmap=4:mono[A4];[0:a]channelmap=5:mono[A5];[0:a]channelmap=6:mono[A6];[0:a]channelmap=7:mono[A7];[0:a]channelmap=8:mono[A8];[0:a]channelmap=9:mono[A9];[0:a]channelmap=10:mono[A10];[0:a]channelmap=11:mono[A11]
<Guest10> ;[0:a]channelmap=12:mono[A12];[0:a]channelmap=13:mono[A13];[0:a]channelmap=14:mono[A14];[0:a]channelmap=15:mono[A15]" \
<Guest10>     -map "[A0]" -map "[A1]" -map "[A2]" -map "[A3]" -map "[A4]" -map "[A5]" -map "[A6]" -map "[A7]" -map "[A8]" -map "[A9]" -map "[A10]" -map "[A11]" -map "[A12]" -map "[A13]" -map "[A14]" -map "[A15]" \
<Guest10>     -stream_group "type=iamf_audio_element:id=1:st=0:st=1:st=2:st=3:st=4:st=5:st=6:st=7:st=8:st=9:st=10:st=11:st=12:st=13:st=14:st=15:audio_element_type=scene,layer=ch_layout=ambisonic\ 3:ambisonics_mode=mono," \
<Guest10>     -stream_group
<Guest10> "type=iamf_mix_presentation:id=3:stg=0:annotations=en-us=default_mix_presentation,submix=parameter_id=100:parameter_rate=48000:default_mix_gain=0.0|element=stg=0:headphones_rendering_mode=binaural:annotations=en-us=3OA:parameter_id=101:parameter_rate=48000:default_mix_gain=0.0|layout=sound_system=stereo:integrated_loudness=0.0:digital_peak=0.0" \
<Guest10>     -streamid 0:0 -streamid 1:1 -streamid 2:2 -streamid 3:3 -streamid 4:4 -streamid 5:5 -streamid 6:6 -streamid 7:7 -streamid 8:8 -streamid 9:9 -streamid 10:10 -streamid 11:11 -streamid 12:12 -streamid 13:13 -streamid 14:14 -streamid 15:15 -streamid 16:16 \
<Guest10>     -c:a libopus bar.iamf
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<Guest52> I'm currently working on a real-time streaming project where I combine up to 16 individual RTSP camera streams into a single tiled video stream using FFmpeg, and stream this combined video to a web-based UI through the MediaSoup media server (WebRTC).
<Guest52>  Current Scenario:
<Guest52> To test the workflow, I initially set up 3 RTSP camera streams, and I'm using the following FFmpeg command to tile the video:
<Guest52>  -hide_banner -loglevel debug -report -hwaccel cuda -hwaccel_device 0 -fflags +nobuffer -flags low_delay -strict experimental -fflags +genpts -f mjpeg -thread_queue_size 1024 -i http://xx.xx.xx.xx/zm/cgi-bin/nph-zms?mode=jpeg&scale=100&maxfps=50&monitor=1&user=admin&pass=admin -f mjpeg -thread_queue_size 1024 -i
<Guest52> [0:v]scale=320:240,fps=30,format=yuv420p[v0];[1:v]scale=320:240,fps=30,format=yuv420p[v1];[2:v]scale=320:240,fps=30,format=yuv420p[v2];[v0][v1][v2]xstack=inputs=3:layout=0_0|320_0|640_0[outv] -map [outv] -an -vsync drop -err_detect ignore_err -c:v libx264 -preset veryfast -tune zerolatency -g 30 -keyint_min 30 -sc_threshold 0 -profile:v baseline
<Guest52> -x264-params keyint=30:min-keyint=30:scenecut=0:repeat-headers=1 -b:v 1500k -maxrate 1500k -bufsize 1500k -f rtp -payload_type 102 -ssrc 577362603 rtp://127.0.0.1:46144
<Guest52> The Issue:
<Guest52> The combined video stream is successfully tiled and streamed without significant overall latency.
<Guest52> However, among the tiled streams, one stream remains live and current without delay, but the other two streams have approximately a 2-second delay.
<Guest52> Individually checking each camera stream separately (via VLC), I confirmed they all have minimal latency (~200ms). The delay is only observed when combining them through FFmpeg.
<Guest52> Steps Already Taken:
<Guest52> Tried setting -fflags +nobuffer -flags low_delay on input and output.
<Guest52> Used -preset veryfast -tune zerolatency.
<Guest52> Adjusted various values for -thread_queue_size .
<Guest52> Despite these attempts, the delay remains.
<Guest52> My goal:
<Guest52> I need to have all tiled streams consistently live without the noticeable delay, especially as I scale up to 16 simultaneous camera streams.
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<futurion> Guest52 I don't know how to solve your latency issue, but I'm wondering what happens if one stream dies/drops or goes offline for a while.... does the tiled mosaic tehn still work with only two streams and when the third comes back online it just appears there as nothing happened, or do you have to restart the whole ffmpeg cmd to reinitialize all
<futurion> streams?
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<futurion> I'm asking this because I had issues with ffmpeg's UDP mpegts streams tiled into mosaic and I had to restart the whole command occasionally, otherwise some streams wouldn't come back online
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<citizendot> how to build ffmpeg statically?
<citizendot> i tried with --enable-static flag, but i keep getting dynamically linked binary
<futurion> us media-autobuild-suite... there you have an option to select full-plus (shared binaries) or just full or mimic-zeranoe where these shared binaries aren't used
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<furq> citizendot: --extra-ldflags="-static"
<furq> --enable-static just links the ffmpeg libs statically, not any external libs
<furq> or you can use BtbN's build scripts or similar
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<Guest52> futurion Currently, I am using the ZoneMinder stream API to input video streams. When one of the cameras gets disconnected, the stream on the UI side freezes. However, once the camera is turned back on, the streaming automatically resumes.
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<vlt> Hello. How can I find the video codec that will cause the lowest energy consumption when played back on a specific phone model? According to https://www.devicespecifications.com/en/model/d8903566 the phone’s CPU is a "Qualcomm Snapdragon 410 (MSM8916)". Any idea?
<JEEB> I would expect the ASIC setup is common between all of them if you mean hardware decoding
<furq> it will almost certainly be h264 on any phone
<furq> i don't know if benchmarks exist for hardware decoding power consumption but h264 is the simplest codec that is more or less universally hardware decoded on phones
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<JEEB> the ASIC hardware block should kinda be a singular one, so at that point it only depends on the complexity of the bit stream (bit rate, frame rate etc) whether it can keep up or not. but I would be surprised if the power consumption changed
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<furq> that page doesn't show support for any other codec anyway
<furq> the other codecs shown are probably decoded in software
<furq> actually i guess it does say webm but it doesn't say which codecs
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<vlt> Ok, thank you.
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<Guest4> hi
<Guest4> i love ffmpeg
<Guest4> bye
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<johnjaye> ah it seems ffmpeg doesn't enable ssml for the flite filter
<johnjaye> so there's no way to add breaks or control the tts output
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<GuestGuest> It would be much appreciated if some expert here could help me with this, I've sitting trying things but don't seem to be any closer
<johnjaye> it might help to say you're having problem with the mpegts muxer and the error is related to audio frame size.
<johnjaye> then someone might be more likely to read and answer
<GuestGuest> I seem to have a problem with mpegts muxer and the error appears to be related to audio frame size
<zeromind> usually one'd post the full command and output with the error; you mention "av1 + aac codecs", but the error is for mp3, so something's not right
<zeromind> mp3 with 0 channels even
<GuestGuest> If I mux with NUT it tells me it's Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 97 kb/s (default)
<johnjaye> right buut the paste said mp3
<johnjaye> hence the confusion
<GuestGuest> I think the error appears after there has been a problem with input udp stream, then it messes up something with timestamps, but the interesting thing is that NUT seems to be able to handle it and playback correctly, while mpegts gives that error and no sound but still plays back the video
<zeromind> did you ever post full command and out? or what streams the m3u8 actually mentions?
<zeromind> *output
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<GuestGuest> full command: ffmpeg -loglevel verbose -i https://url/stream.m3u8 -c:v libx264 -tune film -c:a copy -f mpegts pipe:1 | ffplay -
<GuestGuest> That one doesn't work, and just replace mpegts mux with nut then it works
<zeromind> okay, not getting anywhere, got other things to do, good luck
<GuestGuest> Well, thanks for trying...
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<Guest41> I'm trying to compile ffmpeg with the instructions found here: https://developer.nvidia.com/blog/nvidia-ffmpeg-transcoding-guide/ but it looks like it can't see the headers for ffnvcodec  I don't know how to fix this command to do that "./configure --enable-cuda --enable-cuvid --enable-nvdec --enable-nvenc --enable-nonfree --enable-libnpp
<Guest41> --extra-cflags=-I/usr/local/cuda/include  --extra-ldflags=-L/usr/local/cuda/lib64"
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<Guest41> The exact error I get every time is "ERROR: cuda requested, but not all dependencies are satisfied: ffnvcodec
<Guest41> "
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<BtbN> That guide is outdated and wrong in many places, just ignore it really
<Guest41> so then how do I compile with nvidia driver support?
<BtbN> read --help
<BtbN> Just --enable-ffnvcodec is usually enough
<johnjaye> if i wanted to concat 100 audio or video only files would i still use the text file method?
<johnjaye> or can i just say concat=n=100 and that method also wold join them?
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<BtbN> generate a concat file with a script, and use that
<BtbN> if it's mpegts, you can also just literally concat the files
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<johnjaye> it's parts of an mp4
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<BtbN> parts of an mp4?
<Guest41> I'm still getting "ERROR: ffnvcodec requested but not found"
<BtbN> mp4 can't usually be split
<johnjaye> i tried reencoding them all with same fps, res, sar etc but that still didn't work with a few files
<BtbN> Just don't re-encode them at all, and concat with the concat filter
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<BtbN> Much saner
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<BtbN> Guest41: you need to install ffnvcodec then
<johnjaye> no separate ones. but in my use case yes i am splitting an mp4 into segments and rejoining rn
<BtbN> most distros have it packaged nowadays
<johnjaye> > Note: Filters are incompatible with stream copying; you can't use -c copy with this method.
<Guest41> not debian
<johnjaye> is the guide i use
<BtbN> Well, you are already re-encoding though
<johnjaye> the filter failed so i tried to make them all as similar as possible but couldn't get it to work
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<BtbN> the concat demuxer is incredibly finnicky
<BtbN> it needs _perfectly_ matching content, ideally with consecutive timestamps and everything
<johnjaye> i see.
<BtbN> it's great for stitching back together stuff like segments of an hls playlist or something
<BtbN> but for combining completely unrelated stuff, it's a pain
<BtbN> it's possible, but painful, and you often end up with glitches on the seams
<johnjaye> the problem with the filter is i need a string like [0:v:0][0:a:0][1:v:0][1:a:0][2:v:0][2:a:0]
<johnjaye> which is unwieldy for 10+ files
<BtbN> yep, with 100 videos, the filter invocation will be a humongous beast
<Guest41> Still says "ERROR: ffnvcodec requested but not found"
<BtbN> Guest41: you did something wrong then. Read the config.log and see what it's missing.
<Guest41> that file is not helpful.  it is way too big and obtuse
<BtbN> Can always just manually install the latest headers. It's just a few headers and can't break anything to manually make install
<johnjaye> which avenue should i pursue then? the demuxer,protocol, or filter?
<BtbN> The protocol is literally just cat 1.mp4 2.mp4 3.mp3 .... | ffmpeg -i - ...
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<BtbN> that obviously does not work
<johnjaye> i see
<BtbN> the filter is the only realistic option to concat unrelated files
<BtbN> The humongous chain of inputs and [0:v:0][0:a:0][1:v:0][1:a:0][2:v:0][2:a:0]... can be quite easily shell-scripted
<johnjaye> can i try the original idea an rencode them to be similar enough?
<johnjaye> i couldn't get it to work
<BtbN> You can, but it's an absolute pain
<BtbN> The files need to match perfectly in every codec related parameter
<BtbN> and even then, audio frames will likely not align, so you'll end up with hitches on the seams
<johnjaye> hmm. what if i convert them all to wav 44k 1ch or something
<johnjaye> i.e. some super simple format
<Guest41> "apt-get install libffmpeg-nvenc" did not help
<BtbN> that's not the package I linked
<BtbN> for i in $(seq 100); do printf '[%d:v:0][%d:a:0]' $i $i; done
<BtbN> printf '-i %s ' *.mp4
<BtbN> should make it easy to generate a concat invocation even for rather ridiculous amounts of files
<Guest41> "apt-get install libffmpeg-nvenc-dev" I already tried.  did not help
<BtbN> the seq invocation is missing an initial 0, it starts late
<johnjaye> thanks i'll try that. i was worried about limits on command line length
<BtbN> Isn't the limit several megabytes these days?
<BtbN> Except for on Windows
<BtbN> Guest41: no idea then, install them manually, ask your distro on how to install it. If you're sure they're correctly installed, read the config.log to find out what's wrong.
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<johnjaye> google tells me 100kb to 2MB possibly
<Guest41> config.log is not telling me anything helpful.  it's way too big and I have no idea what to search for
<BtbN> the error will be the last thing it did before aborting
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<Guest41> "ar rcD /tmp/ffconf.wFSqbKDQ/test.a @/dev/null" is the second to last line
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