I'm trying to generate h.264 video from raw 2-byte gray video (14-bit range encoded in 16-bit values). I can do something like:
ffmpeg -f rawvideo -pix_fmt gray16le -s:v 1280x720 -r 60 -i input.raw -c:v libx264 output.mp4
And I get video but it's pretty dark, not sure if it's clipping, doing a linear remap, or storing the 16-bit data and VLC is doing the remap. ffprobe is reporting Video: h264 (High 4:4:4 Predictive) (avc1 / 0x31637661), yuvj444p(pc), 1280x720, 108 kb/s, 60 fps, 60 tbr, 15360 tbn, 120 tbc
I was figuring I'd use the tonemap filter to make a better mapping. I added a filter before the output file with -vf.
tonemap=hable errors Impossible to convert between the formats supported by the filter 'graph 0 input from stream 0:0' and the filter 'auto_scaler_0'
zscale=transfer=linear,tonemap=hable errors Impossible to convert between the formats supported by the filter 'Parsed_tonemap_1' and the filter 'auto_scaler_1'
zscale=transfer=linear,tonemap=hable,zscale=transfer=bt709,format=yuvj444p errors code 3074: no path between colorspaces
I'm not sure where to proceed from here...
Related
We have a .ts input file that contains (among other streams) a video stream and MISB 0604-compliant KLV data stream. The output of ffprobe for these stream are:
Stream #0:0[0x111]: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p(progressive), 1280x720 [SAR 1:1 DAR 16:9], 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc
...
Stream #0:2[0x1001]: Data: klv (KLVA / 0x41564C4B)
We are hoping to extract every Nth frame of the video as a .tiff. We also need to associate each of these frames with a corresponding KLV packet from the MISB 0604-compliant data stream.
The following command that select filters and adjusts the original FPS by the corresponding ratio does result in the expected frames being saved out as TIFF (in this case the original video has 1187 frames, and I expect to get 12 frames from the select filter).
ffmpeg -y -i 2205Z.ts -map 0:0 -vf "select='not(mod(n,100))'",fps=30000/1001/100 -compression_algo raw -pix_fmt rgb24 %05d.tif
However I can't seem to get any filters working on the data stream. For example using filter:d does not throw an error, but also doesn't seem to actually filter. My question is whether ffmpeg can be used to save out a "downsampled" data stream corresponding to the downsampling operations performed on the video stream above?
Using a recent git master build, run
ffmpeg -i 2205Z.ts -map 0:2 -bsf "noise=drop=mod(n\,100)" -f segment -segment_format data -segment_time 0.01 %d.bin
The noise bsf uses an expression to drop, not select i.e. select=EXPR === drop=not(EXPR).
Add -loglevel verbose to see details about which packets are kept, in a format like this:
[noise # 000001cd418a68c0] Stream #2 packet 1099 pts 3420417 - amount 0 drop 1
[noise # 000001cd418a68c0] Stream #2 packet 1100 pts 3423419 - amount 0 drop 0
[noise # 000001cd418a68c0] Stream #2 packet 1101 pts 3426423 - amount 0 drop 1
I'm trying to convert HEVC videos to VP9 so they can be played in a web browser while keeping file size roughly the same.
I am struggling to create a video with similar quality/file size.
Here's the stream info for one of the HEVC videos the video is is 22:49 and 168.7mb:
Stream #0:0(und): Video: hevc (Main) (hev1 / 0x31766568), yuv420p(tv, bt709), 1920x1080, 900 kb/s, 23.98 fps, 23.98 tbr, 90k tbn, 23.98 tbc (default)
The bitrate is 900K so I had thought that the following command would use the same bitrate and give a roughly similar image quality:
ffmpeg -hwaccel vaapi -hwaccel_device /dev/dri/renderD128 -vaapi_device /dev/dri/renderD128 -i "$infile" -vf 'format=nv12,hwupload' -c:v vp9_vaapi -b:v 900K -bf 2 -bsf:v vp9_raw_reorder,vp9_superframe -c:a libvorbis "$outfile"
Using this, the quality is noticeably much, much worse and busy scenes look incredibly blocky though the file size is roughly equivalent to the HEVC source.
If I omit the bitrate and let VP9 work it out
ffmpeg -hwaccel vaapi -hwaccel_device /dev/dri/renderD128 -vaapi_device /dev/dri/renderD128 -i "$infile" -vf 'format=nv12,hwupload' -c:v vp9_vaapi -b:v 0 -bf 2 -bsf:v vp9_raw_reorder,vp9_superframe -c:a libvorbis "$outfile"
The quality is visually indistinguishable but the file size of the VP9 converted video reaches 401mb, up from 168mb of the HEVC file and during encoding the bitrate is over 3m for most of the video.
I tried going up to 1.2M (33% higher than the source video) and VP9 still gave a very blocky looking video.
Is VP9 really that much worse than HEVC or is there an option I am missing? or is it because I am converting from HEVC?
I have about 1200 pngs that I'm converting into a movie. Some of them are missing: i.e. - _00003.png, _00005.png exist, but 1, 2, and 4 do not.
The following command works for other datasets, but not my current set of pngs:
ffmpeg -i pngs/_*.png -y -vcodec mpeg4 -pix_fmt yuv420p -r 25 -filter:v 'setpts=1.2*PTS' p3SN.mp4
I get this error:
Output #61, image2, to 'pngs/_00096.png':
Metadata:
encoder : Lavf57.83.100
Stream #61:0: Video: png, rgba, 3240x2160 [SAR 3937:3937 DAR 3:2], q=2-31, 200 kb/s, 25 fps, 25 tbn, 25 tbc
Metadata:
encoder : Lavc57.107.100 png
Output #62, image2, to 'pngs/_00097.png':
Metadata:
encoder : Lavf57.83.100
Stream #62:0: Video: png, rgba, 3240x2160 [SAR 3937:3937 DAR 3:2], q=2-31, 200 kb/s, 25 fps, 25 tbn, 25 tbc
Metadata:
encoder : Lavc57.107.100 png
[png # 0x7fae93170e00] ff_frame_thread_encoder_init failed
Error initializing output stream 63:0 -- Error while opening encoder for output stream #63:0 - maybe incorrect parameters such as bit_rate, rate, width or height
Conversion failed!
at image _00097.png. If I remove it, it just happens a little later (105).
I've checked the images by looking at their dimensions, etc. and all of them in this range look the same (I checked all those with _0009?.png).
Any idea why this is happening?
Here's the offending file (middle) and the one before/after:
Your command will overwrite all of the input files with the first input. This is an example of why to use caution when using -y which will automatically overwrite files without asking you.
You need to tell ffmpeg to use the glob pattern:
ffmpeg -y -pattern_type glob -framerate 25/1.2 -i "pngs/_*.png" -vcodec mpeg4 -pix_fmt yuv420p -r 25 p3SN.mp4
I believe the glob pattern option does not work on Windows, but if it has an equivalent to the Linux cat command you can pipe the output: cat *.png | ffmpeg -i - output.mp4
You can use -framerate and -r instead of setpts if desired.
I am blocked trying to do something, and I'm ready to make a donation if somebody can help me:
I try to concat http://s.serero.free.fr/rolex.mp4 video and http://s.serero.free.fr/video.mp4 video in one output mp4 file and I tried during a big time without results.
I want to concat http://s.serero.free.fr/rolex.mp4 + http://s.serero.free.fr/video.mp4
or http://s.serero.free.fr/video.mp4 + http://s.serero.free.fr/rolex.mp4.
I tried with ffmpeg command line software and with mp4box command line software, I think that I don't have the good method.
I tried to transform http://s.serero.free.fr/video.mp4 in the same format of http://s.serero.free.fr/rolex.mp4 (and vice versa):
I transformed http://s.serero.free.fr/rolex.mp4 with the same frame rate of http://s.serero.free.fr/video.mp4
I transformed http://s.serero.free.fr/rolex.mp4 with the same video bitrate of http://s.serero.free.fr/video.mp4
I transformed http://s.serero.free.fr/rolex.mp4 with the same video audio bitrate of http://s.serero.free.fr/video.mp4
Can somebody help me?
Explain to me what is wrong in my strategy?
Your input parameters vary, so you have to make them similar before concatenation.
rolex.mp4
Video: h264 (Main) (avc1 / 0x31637661), yuv420p(tv, bt709), 1280x720 [SAR 1:1 DAR 16:9], 835 kb/s, 25 fps, 25 tbr, 90k tbn, 50 tbc (default)
Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 125 kb/s (default)
video.mp4
Video: h264 (High) (avc1 / 0x31637661), yuvj420p(pc), 1152x720, 1749 kb/s, 30 fps, 30 tbr, 15360 tbn, 60 tbc (default)
Audio: mp3 (mp4a / 0x6134706D), 44100 Hz, stereo, s16p, 127 kb/s (default)
This example will make video.mp4 more like rolex.mp4 then concat them:
ffmpeg -i rolex.mp4 -i video.mp4 -filter_complex \
"[1:v]pad=1280:720:(ow-iw)/2:0,fps=25,format=yuv420p[v1]; \
[0:v][0:a][v1][1:a]concat=n=2:v=1:a=1[v][a]" \
-map "[v]" -map "[a]" output.mp4
You don't actually need to declare fps or format because, as the concat filter documentation states:
All corresponding streams must have the same parameters in all
segments; the filtering system will automatically select a common
pixel format for video streams, and a common sample format, sample
rate and channel layout for audio streams, but other settings, such as
resolution, must be converted explicitly by the user.
...but doing so will allow you to manually choose the "common" settings instead of relying on the filter automatically doing so and potentially selecting a setting you don't want.
Thanks for LordNeckbeard for his excellent answer, he just let a little mistake on the command, i just want to a little explanation :
If I want to concat video.mp4(1152X720) with rolex.mp4(1280X720), we must understand that "video.mp4" is the main video so the video(s) to concatene must have exactly the same frame size.
So before to do this operation you need to resize rolex.mp4 video with the same size like video.mp4 with ffmpeg :
ffmpeg -i rolex.mp4 -s 1152x720 -c:a copy newrolexsized.mp4
No video.mp4 and newrolexsized.mp4 has the same frame size, and you can use the command (spcifying pad=1152:720 => size of the main video):
ffmpeg -i video.mp4 -i newrolexsized.mp4 -filter_complex "[1:v]pad=1152:720:(ow-iw)/2:0,fps=25,format=yuv420p[v1];[0:v][0:a][v1][1:a]concat=n=2:v=1:a=1[v][a]" -map "[v]" -map "[a]" out.mp4
i am trying to convert some different video formats to flv using ffmpeg. But it seems that only some videos go through.
ffmpeg -i /var/www/tmp/91640.avi -ar 22050 -ab 32 -f flv /var/www/videos/91640.flv
here is some debug info:
Seems stream 0 codec frame rate differs from container frame rate: 23.98 (65535/2733) -> 23.98 (5000000/208541)
Input #0, avi, from '/var/www/tmp/91640.avi':
Duration: 00:01:12.82, start: 0.000000, bitrate: 5022 kb/s
Stream #0.0: Video: mpeg4, yuv420p, 1280x528 [PAR 1:1 DAR 80:33], 23.98 tbr, 23.98 tbn, 23.98 tbc
Stream #0.1: Audio: ac3, 48000 Hz, 5.1, s16, 448 kb/s
WARNING: The bitrate parameter is set too low. It takes bits/s as argument, not kbits/s
Output #0, flv, to '/var/www/videos/91640.flv':
Stream #0.0: Video: flv, yuv420p, 1280x528 [PAR 1:1 DAR 80:33], q=2-31, 200 kb/s, 90k tbn, 23.98 tbc
Stream #0.1: Audio: adpcm_swf, 22050 Hz, 5.1, s16, 0 kb/s
Stream mapping:
Stream #0.0 -> #0.0
Stream #0.1 -> #0.1
Error while opening codec for output stream #0.1 - maybe incorrect parameters such as bit_rate, rate, width or height
also, if i try to grab one frame ad convert it to jpeg i get an error as well
ffmpeg -i /var/www/tmp/91640.avi -an -ss 00:00:03 -t 00:00:01 -r 1 -y /var/www/videos/91640.jpg
debug info
...
[mpeg4 # 0x1d7d810]Invalid and inefficient vfw-avi packed B frames detected
av_interleaved_write_frame(): I/O error occurred
Usually that means that input file is truncated and/or corrupted.
im thinking that the image fails because the video conversion failed in the first place, not sure though
any ideas what goes wrong?
Bits, not kbits
From your console output:
WARNING: The bitrate parameter is set too low. It takes bits/s as argument, not kbits/s
Use 32k, not just 32.
Only stereo or mono is supported
The encoder adpcm_swf ony supports mono or stereo, so add -ac 2 as an output option. The console output would have suggested this if you were using a recent ffmpeg build.
Use -vframes 1 for single image outputs
Instead of -t 00:00:01 -r 1 use -vframes 1.
A better encoder
Instead of using the encoders flv and adpcm_swf, I recommend libx264 and libmp3lame:
ffmpeg -i input -vcodec libx264 -preset medium -crf 23 -acodec libmp3lame -ar 44100 -q:a 5 output.flv
-preset – Controls the encoding speed to compression ratio. Use the slowest preset you have patience for: ultrafast,superfast, veryfast, faster, fast, medium, slow, slower, veryslow.
-crf – Constant Rate Factor. A lower value is a higher quality. Range is 0-51 for this encoder. 0 is lossless, 18 is roughly "visually lossless", 23 is default, and 51 is worst quality. Use the highest value that still gives an acceptable quality.
-q:a – Audio quality for libmp3lame. Range is 0-9 for this encoder. A lower value is a higher quality.
Also see
FFmpeg and x264 Encoding Guide
Encoding VBR (Variable Bit Rate) mp3 audio