I have a .wtv file, recorded from Windows Media Center, that I'd like to extract the subtitles from. The video is encrypted, but the subtitles are not (something I've been able to verify by using CCExtractor with it). FFMpeg lists the video as such:
Duration: 00:07:01.72, start: 2.214551, bitrate: 9154 kb/s
Stream #0:0[0xcc](eng): Audio: ac3, 48000 Hz, 5.1(side), fltp, 384 kb/s
Stream #0:1[0xcd]: Video: mpeg2video (Main), yuv420p(tv), 1920x1080 [SAR 1:1 DAR 16:9], Closed Captions, max. 25000 kb/s, 29.97 fps, 29.97 tbr, 10000k tbn, 59.94 tbc
Stream #0:2[0xce]: Subtitle: eia_608
When I try to run
ffmpeg -i myvideofile.wtv -an -vn -map 0:2 -c:s:0 srt test.srt
I see a lot of the following errors:
[wtv # 0x7fef79806e00] encrypted stream detected (st:1), decoding will likely fail
Last message repeated 8 times
[Closed caption Decoder # 0x7fef7982cc00] Data Ignored since exceeding screen width
and eventually:
Output file is empty, nothing was encoded (check -ss / -t / -frames parameters if used)
I don't mind it being able to decode the video stream, but is that causing the closed caption error? If it can't decode the video it doesn't know the screen width, perhaps? Is it possible to set the closed caption decoder to ignore such errors and output anyway (it's just in text format after all)?
Related
I'm ripping video from a bunch of ancient MiniDV tapes using, after much trial and error, some almost as ancient Mac hardware and iMovie HD 6.0.5. This is working well except that it will only create a contiguous video clip of about 12.6 GB in size. If the total video is larger than that, it creates a second clip that is usually about 500 MB.
I want to join these two clips in the "best" way possible - meaning with ffmpeg throwing as few errors as possible, and the audio / video staying in sync.
I'm currently using the following command line in a bash shell:
for f in *.dv ; do echo file '$f' >> list.txt; done && ffmpeg -f concat -safe 0 -i list.txt -c copy stitched-video.dv && rm list.txt
This seems to be working well, and using the 'eyeball' check, sync seems to be preserved.
However, I do get the following error message when ffmpeg starts in on the second file:
Non-monotonous DTS in output stream 0:1; previous: 107844491, current: 107843736; changing to 107844492. This may result in incorrect timestamps in the output file.
Since I know just enough about ffmpeg to be dangerous, I don't understand the significance of this message.
Can anyone suggest changes to my ffmpeg command that will fix whatever ffmpeg is telling me is going wrong?
I'm going to be working on HD MiniDV tapes next, and, because they suffer from numerous dropouts, my task is going to become more complex, so I'd like to nail this one.
Thanks!
as suggested below ffprobe for the two files
Input #0, dv, from 'file1.dv': Metadata: timecode : 00:00:00;22 Duration: 00:59:54.79, start: 0.000000, bitrate: 28771 kb/s Stream #0:0: Video: dvvideo, yuv411p, 720x480 [SAR 8:9 DAR 4:3], 25000 kb/s, 29.97 fps, 29.97 tbr, 29.97 tbn Stream #0:1: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
Input #0, dv, from 'file2.dv': Metadata: timecode : 00:15:06;19 Duration: 00:02:04.09, start: 0.000000, bitrate: 28771 kb/s Stream #0:0: Video: dvvideo, yuv411p, 720x480 [SAR 8:9 DAR 4:3], 25000 kb/s, 29.97 fps, 29.97 tbr, 29.97 tbn Stream #0:1: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
I currently use windows free software called: AnyVideoConverter to convert my iPhone huge MOV files to MP4s that can be played on other devices via my plex server.
I want to automate that process so it runs in the background on one of my linux machines.
However I am struggling to get it working. Here is what I have so far.
Original file details:
Duration: 00:00:13.53
Original Video: 40MB
Video details: Stream #0:0(und): Video: hevc (Main) (hvc1 / 0x31637668), yuv420p(tv, bt709), 3840x2160, 23453 kb/s, 30 fps, 30 tbr, 600 tbn, 600 tbc (default)
AnyVideoConverted file:
File Size: 5MB
Video Details: Stream #0:0(und): Video: h264 (Constrained Baseline) (avc1 / 0x31637661), yuv420p, 3840x2160, 2870 kb/s, 24 fps, 24 tbr, 12288 tbn, 48 tbc (default)
This format works great on all my devices
So far the closes I managed to get to that is this ffmpeg command:
ffmpeg -i original.mov -c:v libx264 -profile:v baseline -maxrate 3M -bufsize 3M -c:a aac -b:a 128k x264.mp4
File size:5MB
Video details: Stream #0:0(und): Video: h264 (Constrained Baseline) (avc1 / 0x31637661), yuv420p, 2160x3840, 3063 kb/s, 30 fps, 30 tbr, 15360 tbn, 60 tbc (default)
problem is that when I try to play it via plex I get:
Playback Error: This server is not powerful enough to convert video
What am I doing wrong here?
My knowledge of ffmpeg or video encoding/transcoding is zero. Can someone advise me how to get my mov files converted to mp4 so they can be played via plex without additional transcoding by plex and without reducing the resolution size of the converted video.
Any pointers ?
Thanks
Transcoding depends on the type of Plex client that is requesting the movie. You have two options here:
Set video streaming quality to maximum in all of your Plex clients. This way a transcoding is not necessary in most of the cases.
Disable video transcoding in your Plex server (there is an option for that in your server settings). If your server is not capable of transcoding a normal video at all (like you described above), then it makes no sense to leave it enabled.
I am beginning to be more serious about video. I am processing my videos with ffmpeg in a fully updated Linux into mp4 to use it in HTML5 directly.
Now, I have old AVI videos that I want to convert to mp4 with ffmpeg for use with HTML5. In particular, I have this one:
http://luis.impa.br/photo/1101_aves_ce/caneleiro-de-chapeu-preto_femea_Quixada-CE-110126-E_05662+7a.avi
(I know, terrible quality... sorry). According to ffprobe:
Duration: 00:01:35.30, start: 0.000000, bitrate: 1284 kb/s
Stream #0:0: Video: mpeg4 (Simple Profile) (DX50 / 0x30355844), yuv420p, 640x480 [SAR 1:1 DAR 4:3], 1144 kb/s, 30 fps, 30 tbr, 30 tbn, 30 tbc
Stream #0:1: Audio: mp3 (U[0][0][0] / 0x0055), 44100 Hz, stereo, s16p, 128 kb/s
That seems perfect: mpeg4 video and mp3 audio. So I tried:
ffmpeg -i input.avi -acodec copy -vcodec copy output.mp4
It generates a file that plays nicely in mplayer, but not in firefox getting an error:
Video format or MIME type not supported.
Chrome plays the audio, but no video is shown... Now, if I do:
ffmpeg -i input.avi output.mp4
firefox works, but the video is reencoded in another one with half the size (half the bitrate). This is what ffprobe says about the reencoded video:
major_brand : isom
minor_version : 512
compatible_brands: isomiso2avc1mp41
encoder : Lavf57.71.100
Duration: 00:01:35.30, start: 0.000000, bitrate: 685 kb/s
Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 640x480 [SAR 1:1 DAR 4:3], 548 kb/s, 30 fps, 30 tbr, 15360 tbn, 60 tbc (default)
Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 128 kb/s (default)
I suppose that I am loosing lots of quality (and time processing the video). So, my questions:
Why are browsers not playing my video with the copy codecs ?
Can I work with ffmpeg in this particular file without reencoding? If yes, how?
If I need to reencode, which are "reasonable" parameters to keep close to the original quality? Would something like
ffmpeg -i input.avi -b:v 1024k -bufsize 1024k output.mp4
suffice for this video? This generates a new video with size closer to the original one.
Thanks!
According to ffprobe and if I see it correctly, you have a DivX (5) video file. Do not use it for web!! ;)
mpeg4 (Simple Profile) (DX50 / 0x30355844)
So I don't see any chance to use this video without reencoding. Not if you wish to support firefox.
Use WebM or h264: https://developer.mozilla.org/en-US/docs/Web/HTML/Supported_media_formats
UPDATE
Good settings for reencode depends on your input (bitrate, resolution, fps, kind of material ...), so there is no standard answer.
But you have to specify a codec or ffmpeg choose one depending on your output file extension (so it can be the wrong one).
You can try this:
ffmpeg -i input.avi -c:v libx264 -preset slow -crf 22 -c:a copy output.mkv
Presets and tunes can help to find the best choice: https://trac.ffmpeg.org/wiki/Encode/H.264
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