I converted some .mp4 files to .webm with Miro. They play in Chrome and Opera but they do not play in Firefox 16. Firefox has supported .webm since FF 4. I tested video.canPlayType() and it returns "probably". I know that the MIME types are correct in .htaccess. I downloaded the a test .webm file from html5rocks and uploaded it to the same server and it played fine in FF, which makes me think it must be the encoding. I tried converting the videos multiple times and still no dice. Can anyone think of what would cause the them to not work in FF? Is there a more reliable way to convert videos to .webm?
Chrome_ImF.webm <== works in FF
amber-miro.webm <== does not work in FF
costa-miro.webm <== does not work in FF
There seems to be no problem with the encoding, however there are some errors with the file format on the webm files you created with Miro. I suggest using ffmpeg.
I've tested amber.webm which you provided as a sample. I used ffmpeg to re-format it into webm, without encoding, using the commandline:
ffmpeg -i amber.webm -acodec copy -vcodec copy amber1.webm
And the resulting file seems to play well in Firefox.
To directly convert .mov to .webm via ffmpeg you can use:
ffmpeg -i amber.mov -qscale 0 amber.webm
-qscale 0 is meant to retain the highest quality. See the ffmpeg docs.
The issue is in the version of ffmpeg included with MVC 3.0 on windows. It appears that it's skipping frames at the beginning of the webm conversion.
To workaround this issue for now, you can download an updated version of ffmpeg for windows from here: http://ffmpeg.zeranoe.com/builds/.
Extract the files and copy the updated ffmpeg.exe file here:
C://Program Files/Participatory Culture Foundation/Miro Video Converter/ffmpeg/ replacing the outdated ffmpeg.exe file.
After that your webm conversions with Miro Video Converter should work fine on Firefox.
Related
I wanted to edit my .m4a voice recording from Samsung Voice Recorder using ffmpeg 2.2.2, however, I got the error Invalid data found when processing input. I tried to open it through Audacity, but it returned an error claiming that the ffmpeg library is missing, which is definitely not the case. Eventually I tried to use online .m4a to .mp3 converters, but they all returned error, so I assume there may be an issue with the encoding of the original file and ffmpeg should be configured accordingly. What settings shall I use? (The original file can be played on the phone without any problem.)
ffmpeg -ss 00:00:19 -i "C:\Your\Folder\original.m4a" edited.m4a
I'm new here.
I have a set of TIF frames that equal 1 minute and 25 seconds of a video.
I'm attempting to copy the frames without re-encoding using the "-c:v copy" function to avoid visible quality loss for a process I'm doing on my side. The command is as follows:
ffmpeg -r 23.977 -i %06d.tif -c:v copy out.mkv
However for some reason, the timing does not seem to be accurate and the video is slightly desynced from the original, ending at 1 minute and 22 seconds instead.
When I use the following command:
ffmpeg -r 23.977 -i %06d.tif out.mkv
It comes out with the proper timing at 1 minute and 25 seconds, however, I did not appreciate the quality loss that came with it.
Is there a workaround to this or is there something I'm missing?
I used both Command Line and Windows Terminal.
In general, it would make sense to transcode when you go from tiff to video format. (I'm surprised it actually works.) You can set encoding quality to your own liking. See [this FFmpeg Wiki article[(https://trac.ffmpeg.org/wiki/Encode/H.264).
I am attempting to create a online radio application using FFMPEG - an audio only DASH stream.
I have a directory of mp3 clips (all of the same bitrate and sample size) which I am encoding to the AAC format and outputting to a mpd.
This is the current command I am working with to stream a single mp3 file:
ffmpeg -re -i <input>.mp3 -c:a aac -use_timeline 1 -use_template 1 -window_size 5 -f dash <out>.mpd
(Input and output paths have been substituted for < input >.mp3 and < output >.mpd in this snippet)
I am running a web server and have made the mpd accessible on it. I am testing the stream using VLC player at the moment.
The problem:
Well, the command works, but it will only work for one clip at a time. Once the next command is run immediately proceeding the completion of the first, VLC player will halt and I need to refresh the player to continue.
I'm aiming for an uninterrupted stream wherein the clips play in sequence.
I imagine the problem is that a new mpd is being created with no reference to the previous one, and what I ought to be doing is appending segments to the existing mpd - but I don't know how to do that using FFMPEG.
The question: Is there such a command to append segments to a previously existing mpd file in FFMPEG? or am I coming at this problem all wrong? Perhaps I should be using FFMPEG to format the clips into these segments, but then adjusting the mpd file manually.
Any help or suggestions would be very much appreciated!
I cannot visualize a H264/MP4 stream generated by ffmpeg in Chrome, IE, Edge. It works only in Firefox.
My testing environment is Windows 10, all updates done, all browsers up to date.
I have a source MJPEG stream, which I need to transcode to H264/MP4 and show it in browser in a HTML5 element.
In order to provide a working example, I use here this MJPEG stream: http://200.36.58.250/mjpg/video.mjpg?resolution=320x240. In my real case I have MJPEG input from different sources like IP cameras.
I use the following command line:
ffmpeg.exe -use_wallclock_as_timestamps 1 -f mjpeg -i "http://200.36.58.250/mjpg/video.mjpg?resolution=320x240" -f mp4 -c:v libx264 -an -preset ultrafast -tune zerolatency -movflags frag_keyframe+empty_moov+faststart -reset_timestamps 1 -vsync 1 -flags global_header -r 15 "tcp://127.0.0.1:5000?listen"
If I try to visualize the output in VLC, I use this link: tcp://127.0.0.1:5000 and it works.
Then I try to visualize the stream in browser, so I put this into a html document:
<video autoplay controls>
<source src="http://127.0.0.1:5000" type="video/mp4">
</video>
If I open the document in Firefox it works just fine.
But it does not work when trying to open in Chrome, IE or Edge. It seems that the browser tries to connect to the TCP server exposed by ffmpeg, but something happens because ffmpeg exits after few seconds.
In ffmpeg console I can see this:
av_interleaved_write_frame(): Unknown error
Error writing trailer of tcp://127.0.0.1:5000?listen: Error number -10053 occurred
If I inspect the video element in Chrome is can see this error:
Failed to load resource: net::ERR_INVALID_HTTP_RESPONSE
As far as I know all these browsers should support H264 encoded streams transported in MP4 containers. If in the element I replace the link http://127.0.0.1:5000 with a local link to a mp4/H264 encoded file, it is played just fine in each browser. The problem seems to be related to live streaming.
Does anyone know why this happens and how it can be solved?
Thank you!
You're just outputting to a TCP socket. That's not HTTP. Browsers speak HTTP... you need to use an HTTP server in this case.
How to transfer metadata using FFMPEG or other tools with CMD ?
I'm trying to encode video/audio and since they already have metadata inside obviously i want to preserve them into my new file
btw since i'm using mediamonkey as main player, there's also some Custom metadata. this is the one who wont transfer
for Video output file using mp4/mkv (using x264)
for Audio output file using m4a (using neroAac)
Thank You!
ps. which container is best for neroAac and x264? since i can't seem to edit mkv metadata (when i remove from mediamonkey playlist, they're all gone), mp4 is fine though and i can't seem to play AAC, although it's fine when muxed into video
Copy all custom and global metadata tag information using the following command:
ffmpeg <inputfile> -movflags use_metadata_tags -c copy <outputfile>