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
Related
I have an input MPEG TS file 'unit_test.ts'. This file has following content (shown by ffprobe):
Input #0, mpegts, from 'unit_test.ts':
Duration: 00:00:57.23, start: 73674.049844, bitrate: 2401 kb/s
Program 1
Metadata:
service_name : Service01
service_provider: FFmpeg
Stream #0:0[0x31]: Video: h264 (Main) ([27][0][0][0] / 0x001B), yuv420p(progressive), 852x480 [SAR 640:639 DAR 16:9], Closed Captions, 59.94 fps, 59.94 tbr, 90k tbn, 119.88 tbc
Stream #0:1[0x34](eng): Audio: ac3 ([129][0][0][0] / 0x0081), 48000 Hz, 5.1(side), fltp, 448 kb/s
Stream #0:2[0x35](spa): Audio: ac3 ([129][0][0][0] / 0x0081), 48000 Hz, stereo, fltp, 192 kb/s
I want to convert it into another MPEG TS file. Requirement is that the Video stream of the input should be directly copied to the output whereas ALL the audio streams should be transcoded "aac" format.
I tried this command:
ffmpeg -i unit_test.ts -map 0 -c copy -c:a aac maud_test.ts
It converted it into 'maud_test.ts' with following contents (shown by ffprobe)
Input #0, mpegts, from 'maud_test.ts':
Duration: 00:00:57.25, start: 1.400000, bitrate: 2211 kb/s
Program 1
Metadata:
service_name : Service01
service_provider: FFmpeg
Stream #0:0[0x100]: Video: h264 (Main) ([27][0][0][0] / 0x001B), yuv420p(progressive), 852x480 [SAR 640:639 DAR 16:9], Closed Captions, 59.94 fps, 59.94 tbr, 90k tbn, 119.88 tbc
Stream #0:1[0x101](eng): Audio: aac (LC) ([15][0][0][0] / 0x000F), 48000 Hz, 6 channels, fltp, 391 kb/s
Stream #0:2[0x102](spa): Audio: aac (LC) ([15][0][0][0] / 0x000F), 48000 Hz, stereo, fltp, 133 kb/s
So it appeared as if the command worked....However when I play the maud_test.ts file in vlc player I can see both audio streams listed in the menu; but Stream 1 (eng) remains silent............whereas Stream 2 (spa) has proper audio. (Original TS file has both audio streams properly audible)
I have tried this with different input files and have seen that same problem occurs in each case.
What that I am doing is not right?
How should I get this done? (I can write explicit stream by stream map and channel arguments to get that done; however I want the command line to be generic, in that the input file could be having any configuration with one Video and several Audios with different formats. The configuration will not be known beforehand.)
I want to overlay a video with the showcqt effect on the right corner, I know that I have to use a filter graph but don't know how, the documentation is large but not very accessible for me. ffmpeg outputs this:
Input #0, matroska,webm, from 'cover.webm':
Metadata:
ENCODER : Lavf58.20.100
Duration: 00:03:14.58, start: -0.007000, bitrate: 206 kb/s
Stream #0:0(eng): Video: vp9 (Profile 0), yuv420p(tv, bt709), 1280x720, SAR 1:1 DAR 16:9, 24 fps, 24 tbr, 1k tbn, 1k tbc (default)
Metadata:
DURATION : 00:03:14.541000000
Stream #0:1(eng): Audio: opus, 48000 Hz, stereo, fltp (default)
Metadata:
DURATION : 00:03:14.581000000
You can use the showcqt and overlay filters:
ffmpeg -i input.webm -filter_complex "showcqt=s=320x180[cqt];[0][cqt]overlay=main_w-overlay_w:main_h-overlay_h" -c:a copy output.webm
The audio is stream copied (-c:a copy) in this example to avoid re-encoding. Remove -c:a copy if you want it to automatically re-encode to an appropriate audio format for whatever output container you choose.
While using ffmpeg to concatenate similar m4a files:
ffmpeg -f concat -safe 0 -i <(for f in ./*.m4a; do echo "file '$PWD/$f'"; done) -c copy output.m4a
ffmpeg reports an error:
[ipod # 0x7f8db8014a00] Could not find tag for codec mjpeg in stream #0, codec not currently supported in container
Could not write header for output file #0 (incorrect codec parameters ?): Invalid argument
The files used are: chapter1.m4a, chapter2.m4a. Their ffprobe have no differences other than the duration. Possible related output is:
Duration: 00:13:16.72, start: 0.000000, bitrate: 48 kb/s
Stream #0:0(eng): Audio: aac (HE-AAC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 46 kb/s (default)
Metadata:
handler_name : SoundHandler
Stream #0:1: Video: mjpeg, yuvj420p(pc, bt470bg/unknown/unknown), 640x360 [SAR 100:100 DAR 16:9], 90k tbr, 90k tbn, 90k tbc
I just found out the error was due to the Stream #0, which is the cover art, and covers the actual audio track.
After removing the cover artworks in all files, I was able to concatenate them. And the speed is quite fast : speed=1.92e+03x.
I am trying to put a simple text on the bottom of video using ffmpeg on Ubuntu 12.04 . I tried this which is suggested in several places:
ffmpeg -i input.avi -vf drawtext="fontfile=/usr/share/fonts/truetype/ttf-dejavu/DejaVuSerif.ttf:text='Text to write':fontsize=20:fontcolor=black:x=100:y=100" output.avi
But I get this error each time:
ffmpeg version 0.8.3-4:0.8.3-0ubuntu0.12.04.1, Copyright (c) 2000-2012 the Libav developers
built on Jun 12 2012 16:37:58 with gcc 4.6.3
*** THIS PROGRAM IS DEPRECATED ***
This program is only provided for compatibility and will be removed in a future release. Please use avconv instead.
Input #0, avi, from 'input.avi':
Duration: 04:09:09.66, start: 0.000000, bitrate: 480 kb/s
Stream #0.0: Video: mpeg4 (Advanced Simple Profile), yuv420p, 320x240 [PAR 1:1 DAR 4:3], 45 tbr, 45 tbn, 45 tbc
Stream #0.1: Audio: mp3, 48000 Hz, stereo, s16, 64 kb/s
[buffer # 0x860d5a0] w:320 h:240 pixfmt:yuv420p
Incompatible sample format 's16' for codec 'ac3', auto-selecting format 'flt'
[ac3 # 0x8607a00] invalid bit rate
Output #0, avi, to 'output.avi':
Stream #0.0: Video: mpeg4, yuv420p, 320x240 [PAR 1:1 DAR 4:3], q=2-31, 200 kb/s, 90k tbn, 45 tbc
Stream #0.1: Audio: ac3, 48000 Hz, stereo, flt, 200 kb/s
Stream mapping:
Stream #0.0 -> #0.0
Stream #0.1 -> #0.1
Error while opening encoder for output stream #0.1 - maybe incorrect parameters such as bit_rate, rate, width or height
Appreciate your help.
The documentation shows that you can use other parameters with x or y such as input video height and width and text width and height. To place the text on the bottom one method is y=main_h-text_h. If you want a little padding on the bottom you can use y=main_h-(text_h*2) To center it horizontally use x=(main_w/2-text_w/2).
I have created a server system based on Xuggle to encode an incoming file to H264 and segment it. However, when playing the video back in Quicktime it almost works (with a small hiccup in the audio sometimes) but when changing fro one quality stream to another the image gets lost.
So I ran the 'mediastreamvalidator'and got the following error:
ERROR: (-1) Unknown video codec: 1836069494 (program 0, track 0)
ERROR: (-1) failed to parse segment as either an MPEG-2 TS or an ES
So I used FFMPEG to get some info on the codex:
The result of my Xuggler encoding:
Input #0, mpegts, from 'segment_0.ts':
Duration: 00:00:09.40, start: 0.000000, bitrate: 3618 kb/s
Program 1
Metadata:
service_name : Service01
service_provider: FFmpeg
Stream #0.0[0x100]: Video: mpeg2video (Main), yuv420p, 960x540 [PAR 1:1 DAR 16:9], 104857 kb/s, 25 fps, 25 tbr, 90k tbn, 50 tbc
Stream #0.1[0x101]: Audio: mp2, 48000 Hz, stereo, s16, 128 kb/s
The result of a file created by Compressor:
Seems stream 0 codec frame rate differs from container frame rate: 180000.00 (180000/1) -> 25.00 (25/1)
Input #0, mpegts, from 'fileSequence1.ts':
Duration: 00:00:09.97, start: 19.984578, bitrate: 5308 kb/s
Program 1
Stream #0.0[0x101]: Video: h264 (Main), yuv420p, 960x540, 25 tbr, 90k tbn, 180k tbc
Stream #0.1[0x102]: Audio: aac, 22050 Hz, stereo, s16, 32 kb/s
The main difference seems to me that for the Xuggler encoded file it says Video: mpeg2video instead of h264. However, while encoding I did specifically set the Coder to ICodec.ID.CODEC_ID_H264.
How can I force it to use h264. The same with audio. I specified AAC and get MP2.
I subsequent used FFMPEG directly and that results in:
Input #0, mpegts, from 'encoded.ts':
Duration: 00:00:24.16, start: 1.400000, bitrate: 360 kb/s
Program 1
Metadata:
service_name : Service01
service_provider: FFmpeg
Stream #0.0[0x100]: Video: h264 (Main), yuv420p, 1920x1080 [PAR 1:1 DAR 16:9], 25 fps, 25 tbr, 90k tbn, 50 tbc
Stream #0.1[0x101](eng): Audio: aac, 48000 Hz, stereo, s16, 57 kb/s
That looks better. I could use FFMPEG directly, but by using Xuggler I can segment the file while easier keep track of progress of the process.
I moved away form Xuggle for the moment and use FFMPEG for the encoding and the segmenting and only use it to get encoding info etc.
Currently two processes needed (first encode, then segment with FFMPEG), but hopefully soon FFMPEG will allow to segment on the fly while encoding