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<SomeoneElse>
Is there a documentation that explains what is expected to send to a pipe before to start sending data to the next pipe? I'm using rawvideo on the first pipe but I'm not sure what it actually expects me to send, I thought it expected whole frames as bytes but I must be wrong cause it is expecting me to send more bytes
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<BtbN>
rawvideo is literally raw data in whatever pixel format you specified, densely packed (linesize == width)
<BtbN>
after one frame is done, it'll immediately expect the next one
<BtbN>
until it gets EOF
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<SomeoneElse>
not to me tho
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<SomeoneElse>
I'm using 2 pipes, in order a rawvideo and a s16le pipes, the frame I want to send to rawvideo is a grayscale 1x1, so literally just a single pixel, yet FFmpeg doesn't start reading from the second pipe until I have sent exactly 11 bytes
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<SomeoneElse>
if I increase the video_size the expected bytes increase, 1=11, 2=17, 3=25, 4=33, 5=41, 6=49
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<SomeoneElse>
the framerate I set is 1, so it should just expect a single frame per each second, not sure if this is needed to know
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<Traneptora>
SomeoneElse: it's possible that avio is reading more than one byte at once from the buffer for i/o efficiency
<Traneptora>
and if you don't send enough it blocks on input
<Traneptora>
naturally if you send EOF that'll work fine
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<SomeoneElse>
Would EOF just prevent FFmpeg from reading from that pipe at all until the whole process is finished?
<Traneptora>
no, whatever i/o is blocking wouldn't block if it receives EOF
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<Traneptora>
what pixel format are you reading in, anyway?
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<SomeoneElse>
grayscale, I did it on purpose to test this issue
<Traneptora>
as long as you keep feeding ffmpeg data it'll keep reading it
<Traneptora>
do you need it to output these very small videos in realtime?
<Traneptora>
I don't see why this is an "issue" exactly
<SomeoneElse>
for 1x1 it stops reading after 11 bytes and start reading from the second pipe
<Traneptora>
full command?
<SomeoneElse>
yes it's in real time
<Traneptora>
currently ffmpeg.c doesn't multithread its I/O iirc
<Traneptora>
so it goes back and forth between the two inputs
<Traneptora>
you can always use the -re input flag to read at realtime rate
<SomeoneElse>
and that's ok to me
<SomeoneElse>
as long as I can understand how to send just a single frame instead of being forced to send multiple for a 1 second video of 1 fps
<Traneptora>
-re will read in realtime
<SomeoneElse>
I tried it, still waits for 11 bytes to be sent
<Traneptora>
full command?
<SomeoneElse>
I have never sent EOF, I thought it was a character. I'll send the whole command now
<Traneptora>
you also need to put -re before the audio input as well, because otherwise it won't read that one in realtime
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<SomeoneElse>
same issue
<SomeoneElse>
it doesn't start reading from the second pipe by sending just one byte
<SomeoneElse>
I'll check how to send EOF through a pipe
<Traneptora>
EOF will cause it to close the input
<Traneptora>
you don't want to do that if you have more data to send
<SomeoneElse>
yep, just read about it, I can't do it that way then
<Traneptora>
anyway, I can reproduce the issue on Linux as well
<Traneptora>
sounds like it might be a bug worth reporting
<SomeoneElse>
I could ask it on StackOverflow before to report it
<Traneptora>
?
<Traneptora>
the issue appears to be that avio is blocking on more input than it needs to output very small AVFrames
<Traneptora>
asking on stackoverflow won't change that?
<SomeoneElse>
unless somebody knows of an option to change that behaviour
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<SomeoneElse>
Just found out that adding -fflags +nofillin it multiplies the amount of required bytes by about 3.8
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<sh4>
i'm trying to convert hdmv_pgs_subtitle to dvdsub, there's a superuser thread suggesting to use output format matroska, but then one needs mkvextract to get sub/idx files. is there a way to achieve it with ffmpeg directly ?
<sh4>
something like ffmpeg -i input.mkv -map 0:s:0 -c:s dvdsub -f vobsub output, where output.sub and output.idx are created
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<shashanks>
Hello folks, I need some information on using the FFMPeg API avcodec_receive_frame(). I am doing vaapi-accelerated-decode of a file, and referring to hw_decode.c example from FFMpeg sample code. I am receiving the decoded data into a AVFrame, and then copying the data from AVFrame into a gbm-drm-buffer using av_hwframe_transfer_data.
<shashanks>
Is there any way where I can setup my gbm-buffer as backing buffer for my AVFrame, and receive the data directly on it, instead of one extra step of copy.
<shashanks>
my gbm-buffer is a AMDGPU-DRM GPU buffer, VAAPI driver is AMDGPU
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<Guest1>
seeing an error I'm not sure what to do with:
<Guest1>
cl.exe is unable to create an executable file.
<Guest1>
If cl.exe is a cross-compiler, use the --enable-cross-compile option.
<Guest1>
Only do this if you know what cross compiling means.
<Guest1>
C compiler test failed.
<Guest1>
If you think configure made a mistake, make sure you are using the latest
<Guest1>
version from Git. If the latest version fails, report the problem to the
<Guest1>
ffmpeg-user@ffmpeg.org mailing list or IRC #ffmpeg on irc.libera.chat.
<Guest1>
Include the log file "ffbuild/config.log" produced by configure as this will help
<Guest1>
solve the problem.
<Guest1>
__
<Guest1>
I just grabbed the latest version of ffmpeg from git, so I don't think that's the problem. I also had a problem with the -disable-shared flag (Unknown option "–disable-shared". See ./configure --help for available options.), but deleting that flag got me to my current issue.
<Guest1>
Any ideas for things to check or validate?
<BtbN>
Don't bother with MSVC really
<BtbN>
pointless headache
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<Guest1>
what do you mean?
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<BtbN>
Quite literally what I said
<BtbN>
it's barely working, and has a lot of issues. And has no practical advantage
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<Guest1>
is there a way to do this without MSVC? Or are you saying NVIDIA hardware acceleration isn't worth it? I need on the order of <300 ms latency generating an rtsp stream, and after much ffmpeg configuration, I couldn't bring it lower than ~800 ms average. Here are my ffmpeg arguments if that's where we want to go: -f rawvideo -pixel_format rgba
<BtbN>
<300ms latency is very tight though, you will likely want to implement the software for that yourself
<BtbN>
ffmpeg is optimized for reliability, so it likes to buffer
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<BtbN>
I don't think nvenc and x264 are all that different in terms of latency, as long as the CPU can keep up
<JEEB>
yea, x264 might even be lower with multi-slice and zerolatency etc
<JEEB>
not sure how low nvenc can go?
<JEEB>
of course if you can get a dumb in&out thing then of course nvenc will also be similar
<JEEB>
in any case, technically you can attempt to optimize for low latency via the FFmpeg APIs, but you're definitely going to hit the issue of 99% of people not attempting to do that
<JEEB>
so you're going to more or less be on your own trying to figure out where exactly in your flow you have buffering
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<JEEB>
is it your input reader? is it your decoder? is it your filtering? is it your encoder? what about muxer? or the output I/O protocol handler?
<BtbN>
<300ms might be possible, that's still somewhat realistic
<BtbN>
A lot of people want to do remote-play stuff though, which needs <10ms or something. That ain't happening
<Guest1>
gotcha. I'm pretty sure after debugging it's being bogged down by the encoding, hence trying to change the encoding piece; not sure about the muxer(?)
<JEEB>
if you were with x264 before then I'm going to blame the user
<JEEB>
becuase x264 is most definitely capable of one in-one out
<JEEB>
I'm not 100% sure if you can feed it slices separately to get sub-frame latency
<JEEB>
but one frame latency should be totally possible
<Guest1>
interesting...the frames are generated in Unity within the FixedUpdate method, and throwing timers in there, it's able to go through flushing the rendered scene to the ffmpeg process in ~2-4 ms for each frame. I think the process isn't sending updates to the rtsp server (using rtsp-simple-server) quick enough based on Wireshark logs, which made me
<Guest1>
think it was the encoder bogging it down. Do you have any recommendations on how to check latency for the muxer?
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<JEEB>
I'd probably add a log line in two parts of avformat: the API function which receives the AVPacket ("this is where avformat put it into its queue/logic") and then in the module that you're utilizing that then gets passed to AVPacket, in the location where it writes out that packet into output data. you can attempt to utilize something like AV_PKT_DATA_PRFT to attempt to pass through some timing information
<JEEB>
that then can be logged at those points
<Guest1>
I'll try it out, thanks!
<JEEB>
but you really need to attempt to figure out your latency at each step
<JEEB>
if you think your encoding latency is high, then you actually measure it
<JEEB>
you know when AVFrames are pushed to an encoder, and you know when they're pulled from it
<JEEB>
s/they're/AVPackets/
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<JEEB>
also I really hope you're not calculating latency as in the time that it takes for a client to start decoding, since those two are orthogonal to each other :P
<JEEB>
you can have it take 10 seconds for a client to get its first image, but the latency could still be under a second
<Guest1>
right, I don't care about the time it takes to get the first frame, just the latency once the stream is properly setup. I'll dig deeper into where the latencies are coming from, thanks for the help!
<BtbN>
I can also just be all on the client side
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<LimeOn>
Hello, the previously provided suggestions about music cds worked, many thanks.
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<LimeOn>
Now i have another question: how do i remove the 'sound' produced in some audio files when changing the audio position?
<LimeOn>
some low quality audio files have this 'problem', in where when you change the position of the play, it produces a small sound and then plays correctly
<LimeOn>
i do not know how it is called and either i do not know how its even called
<LimeOn>
I tried changing the audio frequency and the audio bitrate, which does not seem to alter the result
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