I have a txt file which contains several videos:
file '01.mp4' # No audio, only video
file '02.mp4' # Video with audio
...
All my videos are using same video codecs:
Stream #0:0[0x1](und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p(tv, bt709, progressive), 720x1280, 3167 kb/s, 30 fps, 30 tbr, 15360 tbn (default)
And all my video that contains audio are these codecs:
Stream #0:1[0x2](und): Audio: aac (HE-AAC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 43 kb/s (default)
I tried to add silent audio using:
ffmpeg -f lavfi -i anullsrc=channel_layout=stereo:sample_rate=44100 -i 01.mp4 -c:v copy -c:a aac -shortest 01-silentaudio.mp4
But when concatening all my videos, it results in no audio for all videos.
What should I do?
Maybe I can reencode all my video to a specific codec, including my video without sound, in order to have the same codecs for all? What do you think?
Thank you
Related
I'm converting some old videos to play on my Roku via a dlna server. I'm trying to understand the MP4 container better to optimize conversions. I have an ogm video:
Duration: 01:00:38.22, start: 0.000000, bitrate: 1056 kb/s
Stream #0:0: Video: mpeg4 (XVID / 0x44495658), yuv420p, 576x324 [SAR 1:1 DAR 16:9], 23.98 fps, 23.98 tbr, 23.98 tbn, 23.98 tbc
Stream #0:1(English): Audio: aac, 48000 Hz, stereo, fltp, 74 kb/s
Stream #0:2(Japanese): Audio: aac, 48000 Hz, stereo, fltp, 73 kb/s
Stream #0:3(English): Subtitle: text
From what I understand, MP4 container can contain MP4 video and aac audio. I used
-c:a copy -c:v copy
And it worked, but the video won't play. Obviously something's wrong. What I don't understand is why, the video looks like it's MP4 and the audio is aac. My guess is it's the Xvid but why?
Thank you.
Todd
MP4 is only the container but your right MP4 usually contain AAC audio and MPEG4 video. Your input file seems to be mpeg4 Xvid which is a mpeg4 derived codec but might not be supported by Roku.
Try change -c:v copy to -c:v h264 to use a more common MPEG4 based video codec.
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 have a ts file, which is created by encoding raw yuv data using x264enc and muxing it with mpegtsmux.
And this file contains more than one stream. For example it may contain two video streams, or it may contain two video and one text stream like that.
It is of 25 fps, and 720X576 resolution. I would like to change it's fps to say 10, and all the three streams must present in the new ts file also.
Is there any pipeline for this in gstreamer? or any command in ffmpeg? How can i do this?
I don't think that this is possible in GStreamer. At least is not straightforward. However, it should be possible with ffmpeg as long as all the sequences in the file can be discovered correctly.
From the wiki of ffmpeg, map option (https://trac.ffmpeg.org/wiki/Map):
The "map" command basically means "include this stream into the
immediately following OUTPUT file."
Using map option it is possible to re-encode the video stream that you want and keep untouched the others.
Using input.ts file as an example with 2 different video streams and some audio streams, the way to follow should be the next:
ffmpeg -i input.ts -map 0:0 -map 0:2 -map 0:3 -map 0:4 -map 0:23 -map 0:10 -map 0:14 -c:v:0 mpeg2video -r:v:0 10 -c:v:1 copy -c:a copy outputmini.ts
Note that the video streams are 0:0 and 0:23. The first video stream (mpeg2) is re-encoded to 10 fps whereas the 0:23 video stream (x264) is just copied to the output. Remember that if there is some unknown stream in the .ts file you may need to add -ignore_unknown flag.
As specified in the link from ffmpeg, the output stream order is specified in the command execution in the map option placement order. With -c:v:0 mpeg2video option, we select the first output video stream and re-encoding codec, while the -r:v:0 10 option puts the first video stream frame rate to 10 fps. The other options from the command line copies the input stream to output with the same parameters.
ffmpeg outputs next:
Output #0, mpegts, to 'outputmini.ts':
Metadata:
encoder : Lavf57.41.100
Stream #0:0: Video: mpeg2video (Main), yuv420p, 640x576 [SAR 6:5 DAR 4:3], q=2-31, 200 kb/s, 10 fps, 90k tbn, 10 tbc
Metadata:
encoder : Lavc57.50.100 mpeg2video
Side data:
cpb: bitrate max/min/avg: 0/0/200000 buffer size: 0 vbv_delay: -1
Stream #0:1(spa): Audio: mp2 ([3][0][0][0] / 0x0003), 48000 Hz, stereo, 192 kb/s
Stream #0:2(qaa): Audio: mp2 ([3][0][0][0] / 0x0003), 48000 Hz, stereo, 192 kb/s
Stream #0:3(spa): Audio: mp2 ([3][0][0][0] / 0x0003), 48000 Hz, mono, 96 kb/s (visual impaired)
Stream #0:4: Video: h264 ([27][0][0][0] / 0x001B), yuv420p(tv, bt709), 1280x720 [SAR 1:1 DAR 16:9], q=2-31, 50 fps, 50 tbr, 90k tbn, 90k tbc
Stream #0:5(spa): Audio: eac3 ([6][0][0][0] / 0x0006), 48000 Hz, stereo, 256 kb/s
Stream #0:6(spa): Audio: eac3 ([6][0][0][0] / 0x0006), 48000 Hz, mono, 64 kb/s (visual impaired)
Stream mapping:
Stream #0:0 -> #0:0 (mpeg2video (native) -> mpeg2video (native))
Stream #0:2 -> #0:1 (copy)
Stream #0:3 -> #0:2 (copy)
Stream #0:4 -> #0:3 (copy)
Stream #0:23 -> #0:4 (copy)
Stream #0:10 -> #0:5 (copy)
Stream #0:14 -> #0:6 (copy)
The desired output is achieved.
However, even though I'm not sure this could be done, I would get a glimpse to tsMuxeR appllication (http://www.videohelp.com/software/tsMuxeR). It is a very powerful tool to this kind of tasks.
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 have some videos either in mp4 or webm format, and I'd like to use ffmpeg to add 4 seconds to the start of each video to display some text in the center with no sound.
Some other requirements:
try to avoid re-encoding the video
need to maintain the quality (resolution, bitrate, etc)
(optional) to make the text fade in/out
I am new to ffmpeg and any help will be appreciated.
thanks in advance
Example ffprobe information for mp4 below:
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'input.mp4':
Metadata:
major_brand : isom
minor_version : 512
compatible_brands: isomiso2avc1mp41
encoder : Lavf55.33.100
Duration: 00:00:03.84, start: 0.042667, bitrate: 1117 kb/s
Stream #0:0(eng): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1280x720, 1021 kb/s, 30 fps, 30 tbr, 15360 tbn, 60 tbc (default)
Metadata:
handler_name : VideoHandler
Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 140 kb/s (default)
Metadata:
handler_name : SoundHandler
Example webm
Input #0, matroska,webm, from 'input.webm':
Metadata:
encoder : Lavf55.33.100
Duration: 00:00:03.80, start: 0.000000, bitrate: 1060 kb/s
Stream #0:0(eng): Video: vp8, yuv420p, 1280x720, SAR 1:1 DAR 16:9, 30 fps, 30 tbr, 1k tbn, 1k tbc (default)
Stream #0:1(eng): Audio: vorbis, 48000 Hz, stereo, fltp (default)
Screenshot from joined.mp4
Screenshot for step 3 console
You'll have to generate a 4 second video with dummy audio matching the parameters of the existing video, including timebase, and then use the concat demuxer with streamcopy.
For the sample files shown in Q:
Step 1 Generate text video
ffmpeg -f lavfi -r 30 -i color=black:1280x720 -f lavfi -i anullsrc -vf "drawtext=fontfile='/path/to/font.ttf':fontcolor=FFFFFF:fontsize=50:text='Your text':x='(main_w-text_w)/2':y='(main_h-text_h)/2',fade=t=in:st=0:d=1,fade=t=out:st=3:d=1" -c:v libx264 -b:v 1000k -pix_fmt yuv420p -video_track_timescale 15360 -c:a aac -ar 48000 -ac 2 -sample_fmt fltp -t 4 intro.mp4
For WebM, replace -c:v libx264 with -c:v libvpx, -c:a aac with -c:a libvorbis and intro.mp4 with intro.webm. You may remove the -video_track_timescale 15360 since WebMs tend to use a single timescale, that I've seen.
Step 2 Prepare concat file, say, list.txt
file 'intro.mp4'
file 'input.mp4'
Step 3 Concat
ffmpeg -f concat -i list.txt -c copy -fflags +genpts joined.mp4
The variables important here are video size 1280x720, frame rate -r 30, -pix_fmt yuv420p, sample rate -ar 48000, format -sample_fmt fltp, channel layout -ac 2 and of course, codecs.
Short answer is that you cannot encode new data as mp4 or webm and insert it at the front of the video stream. Those formats simply do not work like that. Both of these encoding formats are lossy, so if you decode and encode them again then additional information will be lost/changed by the second encoding. You could do something else, but what you are trying to do will not work.