I try to encode video using hevc codec
./ffmpeg -i 1234.mp4 -vcodec hevc_videotoolbox -vb 1000k -acodec aac -ab 192k -sn 2.mp4
error:
[hevc_videotoolbox # 0x7fc681813a00] Error: cannot create compression session: -12908
[hevc_videotoolbox # 0x7fc681813a00] Try -allow_sw 1. The hardware encoder may be busy, or not supported.
Error initializing output stream 0:0 -- Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height
i try change bitrate, width, height, error still exist
is it possible encode video on my macbook air 2015?
is videotoolbox can't use my old GPU and its possible on newest macbooks?
I have the same question recently, I found this when I was looking for answers:
Screenshot from this pdf: enter link description here
It's possible that old mac doesn't support HEVC hardware acceleration natively, I don't have a new mac to test if ffmpeg has anything related to it, maybe someone whose mac has 6th Gen CPU can help you address the problem.
Edit:
I tested following code on the latest 2018 mbp and it worked.
ffmpeg -i VIDEO_PATH -vcodec hevc_videotoolbox -tag:v hcv1 OUT_PATH
The size of the hevc_videotoolbox-encoded video#1 is smaller than the original test file(h264) but larger than libx265-encoded video#2 (using default parameters).
Unexpectedly, the quality of video#1 is much worse than that of the original, whereas video#2 seems untouched. Besides, hevc_videotoolbox doesn't support -crf option, so I'm still stick to libx265, even though it is really slow.
Related
I have some videos at 1080p 60fps.
These videos are with level=50 and my TV plays only videos up to level=41, so I want to convert my videos using ffmpeg and hardware acceleration.
I have a Windows 10 machine with ffmpeg and Geforce 2060, so I try run below command:
ffmpeg -i video.mp4 -vcodec h264_nvenc -preset slow -level 4.1 output.mp4
but I get this error:
[h264_nvenc # 000001dd43cd07c0] InitializeEncoder failed: invalid param (8): Invalid Level.
Error initializing output stream 0:0 -- Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height
When I try to run the same command above using "-level 4.2" or above, the encode process works! But it is useless for me, because I really need "-level 4.1".
I noticed that if use libx264 instead of h264_nvenc, the encode process accepts "-level 4.1", but it take too long to complete (I want to enjoy hardware acceleration instead of CPU).
How can I force h264_nvenc to accept "level=4.1" ?
Description
I pushed a USB camera stream by ffmpeg to a RTMP stream server which is called SRS.
The SRS had saved a MP4 file for me. The frame rate is not a common value in VLC - it's 14.939948. I've checked it out - It seems to be the 'ntsc' format.
Meanwhile, I had received the stream by OpenCV and saved it as another MP4 file.They're not synchronized.
I have tried to convert the frame rate by ffmpeg but was still not synchronized. The only way to make it is to put it in Adobe Premiere and modify the frame rate. Here is the ffmpeg commands I executed:
ffmpeg -i 1639444871684_copy.mp4 -filter:v fps=15 out.mp4
Aside from the stream server, how can I convert the frame rate to normal and keep synchronized at the same time?
Note: For live streaming, you should never depends on the FPS, because RTMP/FLV always use fixed TBN 1k, so there is always introduce some deviation, when publish stream as RTMP or record to other format like TS/MP4.
Note: For WebRTC, the fps is variant, please read Would WebRTC use a constant frame rate to capture video frame or about the Variable Frame Rate (VFR)
It's not a problem of SRS or FPS, you can also replay it by FFmpeg.
Use FFmpeg to transcode doc/source.flv from 25fps to 15fps, then publish to SRS by RTMP(15fps).
Use FFmpeg to record the RTMP(15fps) as output.mp4(15fps).
Use VLC to play the output.mp4(15fps), it show the fps IS NOT 15fps.
First, please start SRS by bellow config, note that DVR disabled:
# ./objs/srs -c test.conf
listen 1935;
daemon off;
srs_log_tank console;
vhost __defaultVhost__ {
}
Run FFmpeg to transcode and publish to SRS, change the fps to 15:
cd srs/trunk
ffmpeg -re -i doc/source.flv -c:v libx264 -r 15 -c:a copy \
-f flv rtmp://localhost/live/livestream
Record the RTMP stream(in 15fps) to output.mp4, note tat the fps is, in FFmpeg logs, it's 15fps:
ffmpeg -f flv -i rtmp://localhost/live/livestream -c copy -y output.mp4
Use VLC to play the output.mp4 which is 15fps, open the Window -> Media Information, you will find out that the fps is changing around 14.8fps, not 15fps!
It's because the TBN of RTMP/FLV, is fixed 1000(1k tbn, each frame is about 66.66666666666667ms), so the deviation is introduced when publish MP4 to RTMP stream. It's not caused by DVR, it's caused by RTMP/FLV TBN.
Note: However, for SRS, using fixed TBN 1k is not a good choice, because it's not friendly for MP4 duration, I reopen the issue srs#2790.
Ultimately, the framerate/fps is not a fixed stuff, it's just a number that give some tips about the stream. Instead, the player always use the DTS/PTS to decide when and how to render the picture.
Answer myself. Here is my method: Read by OpenCV and write frames to a new file at 15FPS. They're going to be synchronized.
with -r
ffmpeg -i 1639444871684_copy.mp4 -r 15 out.mp4
I've been testing different parameters to capture my desktop video and audio (desktop audio, not mic) and I find that no matter what settings I have, the resulting webm file's framerate is around 5fps and is horribly inconsistent. It starts at around 20fps and slowly drops over time until about 4-5fps. I'm not really sure what I'm doing wrong, but here is the basic command I'm using:
ffmpeg -y -video_size 1920x1080 -f gdigrab -framerate 60 -i desktop -c:v libvpx-vp9 -acodec libvorbis -c:a libopus -b:v 2M -threads 4 output.webm
I've tried anywhere between 30-60 fps and tested different bitrates but nothing seems to affect the output framerate.
Also, I know that acodec and c:a are for audio but I'm not sure how to specify the audio device to use.
So my issues are horrible framerate for webm and how to include desktop audio in the recording.
You can use arecord and pipe it through stdout and ffmpeg can read it from stdin.
aplay piping to arecord using a file instead of stdin and stdout
Replacing the aplay command with your ffmpeg. Dont forget to add '-i -' in ffmpeg.
A doubt: why are you defining audio encoder two times?
It's impossible to say why the video frame rate is low from the question. It can be an issue with encoder. Or issue in reading input. Remove the video encoding option. See if the issue persists. If it's working fine, try some other encoders.
Use -c:v libx264 instead of -c:v libvpx-vp9. libvpx-vp9's realtime encoding quality is really bad, even regular libvpx (i.e. VP8) is much better. If you insist on using libvpx, use options like -deadline realtime and -cpu-used -4
Hi i have a ffmpeg hardware related problem.
Simply put: The conversion works on mac, but on some TV screens the converted h.265 video will give a few seconds of delay when it starts playing!
A response will be much appreciated!
Input:
UHD(3840x2160) Prores 422(HQ) master file
What we used to recompile to a 16bit renderer:
brew uninstall ffmpeg
brew uninstall x265
brew uninstall --force x265
brew install x265 --16-bit
brew install ffmpeg --with-fdk-aac --with-ffplay --with-freetype --with-libass --with-libquvi --with-libvorbis --with-libvpx --with-opus --with-x265
Code we used in FFMpeg:
-b:v 36000k -maxrate 38000k -c:v libx265 -pix_fmt yuv420p10le -x265-params "profile=main10:level=5.1:b=36000k" -c:a aac -strict experimental -b:a 256k
Preferred Output
3840x2160 resolution
10bit 4:2:0 colordepth
Main 10#L5.1#High format profile
MP4 container
HEVC codec
24p framerate
38Mb Variable bitrate
256Kb Variable bitrate/AAC audio
Output device is Tarakan UHD Stream Generator T7
Our problem:
The encode file that FFmpeg gives is correct, on spec level.
I would like the file to have a Main 10 High level 5.1 format profile.
Based on the specs I've put into FFmpeg, it automatically makes a file with a level 5.0 profile, as it doesn't need a higher level based on the specs.
The file plays correctly on a Mac.
The file must also play correctly on a Tarakan UHD Stream Generator(media player with multiple HDMI outputs). The file takes a while to load, chops off 5-6 seconds of the beginning of the film, then plays correctly.
I guess this might be an decoder performance issue.
First, the bitrate. 38Mbps, is very high so the decoding takes more time for the first several frames.
Second, 10 bit also increases the loading of the decoder.
You may add "-t fast-decode" to x265 for better decoding performance.
Hello i need to have two versions of the same file stored on my server, medium and HD quality, the thing is that don't really know ffmpeg that well so im just trying this is code at random, i'm using the code belo but I end up with a much larger file, however it works,it plays.
ffmpeg -i inputfile.wmv -vcodec libx264 -ar 44100 -b 200 -ab 56 -crf 22 -s 360x288 -vpre medium -f flv tmp.flv
Just need the two commands to create the 2 different files
You need to give more information about what bitrate, quality or target file size you are aiming for and the size and quality of your source material preferably including codecs used and relevant parameters.
You should read the manual or ffmpeg -h or both. There are several problems with your command line:
You are using constant rate factor, crf = 22, while still trying to limit the bitrate using -b 200.
Bitrate is specified in bits/s (unless you are using a very old ffmpeg), and 200 bps is not usable for anything, add k to get kilobits/s.
You have not specified an audio codec, but you have specified an audio bitrate, ffmpeg will try to guess the audio codec for you but I don't know what codec is the default for .flv-files.
I'm assuming that the command line you posted is supposed to be for the 'medium' quality file.
Some suggestions that you can try:
Try this first: specify audio codec, e.g. -acodec libmp3lame, or if the audio is already in a good format you can just copy it without modification using -acodec copy
Try a different rate factor, e.g. -crf 30, higher numbers mean uglier picture quality, but also smaller file size.
Try a different encoder preset, e.g. -vpre slow, in general, the slower presets enable features that require more CPU cycles when encoding but results in a better picture quality, see x264 --fullhelp or this page to see what each preset contains.
Do a 2-pass encode, link.
If you don't want to read all the documentation for ffmpeg and the codec parameters that you need I suggest you look at this cheat sheet, although the command line switches have changed over the different versions of ffmpeg so the examples might not work.
An example command line:
ffmpeg -i inputfile.wmv -vcodec libx264 -crf 25 -s 360x288 -vpre veryslow -acodec libmp3lame -ar 44100 -ab 56k -f flv tmp.flv
The parameter -s [size] is the size of the output video, in pixels, for the HD file you probably want something around 1280x720, if your material is 5:4 ratio (as 360x288 is) you'll want to try 1280x1024, 960x768 or 900x720. Don't set a size larger than the source material as that will simply upscale the video and you will (probably) end up with a larger file without any noticeable improvement in quality. The -ab parameter is the audio bitrate, you'll probably want to increase this parameter on the HD version as well.