I am using this PyLivestream library to stream files to youtube. the problem is that once it finishes each video the scren goes down for a second until the next video starts. because it's simply just creating ffmpeg command and running then directly in a subprocess for each media file.
Is it possible to configure an instance of ffmpeg that will always be streaming to the destination. It could just be a blank screen or an image. And it also has an input, so I can point PyLivestream to the repeater.
This way the repeater will create one long un-interupted stream experience, but I can still use PyLivestream to stream the individual files.
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We are developing a stop motion app for kids and schools.
So what we have is:
A sequence of images and audio files (no overlapping audio, in v1. But there can be gaps between them)
What we need to do:
Combine the images to a video with a frame rate between 1-12 fps
Add multiple audio files at a given start times
encode with H265 to mp4 format
I would really like to avoid maintaining a VM or Azure batch jobs running ffmpeg jobs if possible.
Is there any good frameworks or third party APIs?
I have only found transloadit as the closes match but they don't have the option to add multiple audio files.
Any suggestions or experience in this area is very appreciated.
You've mentionned FFmpeg in your tag and it is a tool that checks all the boxes.
For the first part of your project (making a video from images) you should check this link. To sum up, you'll use this kind of command:
ffmpeg -r 12 -f image2 -i PATH_TO_FOLDER/frame%d.png PATH_TO_FOLDER/output.mp4
-r 12 being your framerate, here 12. You control the output format with the file extension. To control the video codec check out this link, you'll need to use the option -c:v libx265before the filename of your output.
With FFmpeg you add audio as you add video, with -i option followed by your filename. If you want to cut audio you should seek in your audio with -ss -t two options good for that. If you want and audio to start at a certain point, check out -itoffset, you can find a lot of examples.
I just want some confirmation, because I have the sneaking suspicion that I wont be able to do what I want to do, given that I already ran into some errors about ffmpeg not being able to overwrite the input file. I still have some hope that what I want to do is some kind of exception, but I doubt it.
I already used ffmpeg to extract a specific frame into its own image file, I've set the thumbnail of a video with an existing image file, but I can't seem to figure out how to set a specific frame from the video as the thumbnail. I want to do this without having to extract the frame into a separate file and I don't want to create an output file, I want to edit the video directly and change the thumbnail using a frame from the video itself. Is that possible?
You're probably better off asking it in IRC zeronode #ffmpeg-devel.
I'd look at "-ss 33.5" or a more precise filter "-vf 'select=gte(n,1000)'" both will give same or very similar result at 30 fps video.
You can pipe the image out to your own process via pipe if you like of course without saving it : "ffmpeg ... -f jpeg -|..."
I am looking for a way to play/stream to browser tag a list of mp4 files (same size, bitrate, etc) without hickups in between the files. I am hoping the following approach would work:
* convert mp4 files to m4s/m4v files
* generate MPEG-Dash MPD file (xml)
* stream MPD to dash player in browser
Is this in any way possible? I am aware the m4s/m4v files need special headers and an entry file must be made somehow, and there you have my roadblock.
Bottom-line is I want to avoid to concatenate the separate videos into one big video file and avoid the hick-ups you see when sequencing via a straightforward 'ended-event' way in JS.
Any suggestion much appreciated!
If you want a basic client side solution you can use two separate players or video tags in your web page, showing one and hiding the other.
The one that is visible plays the current video.
The other player loads starts and immediately pauses the next video.
When the first video ends, you hide that player and make the other one visible, un-pausing the video at the same time.
You then preload the next video into the original player and continue.
This technique is used successfully in some sites where ad breaks are mixed with the main video, as an example.
We have a DirectShow application where we capture video input from USB, multiplex with audio from a WAV file (backing music), overlay audio and video effects, compress and write to an MP4 file.
Originally we were using an audio input source (microphone) and mixing our backing music and sound effects over the top but the decision was made to not capture live audio, and so I thought it would make more sense to use the backing music WAV file itself as the audio source.
Here is the filter graph we have:
backing.wav is a simple WAV file (stored locally), and was added to the graph using IFilterGraph::AddSourceFilter.
The problem is that when the graph is run, no audio samples are delivered from the WAV file. The video part of the graph runs as normal, but it's as if the audio part of the graph simply isn't running.
If I stop the graph in GraphEdit, add the Default DirectSound Device audio renderer and hook that up in place of the AAC Encoder filter and then run the graph again, the audio plays as you would expect.
Additionally, if backing.wav is replaced with an audio capture source like a microphone, audio data flows through as normal.
Does anyone have any ideas why the above graph, using a WAV file as the audio source, would fail to produce any audio samples?
I suppose the title is incorrectly identifying/summarizing the problem. There is nothing to tell for the fact that audio is not produced. It is likely that it is produced equally well with DirectSound Renderer and with AAC Encoder, specifically the data is reaching output pin of Mixing Transform Filter (is this your filter? You should be able to trace its flow and see media samples passing though).
With the information given, I would say it's likely that custom AAC encoder somehow does not like the feed and somehow either drops data or switches to erroneous state. You should be able to debug this further by inserting a Sample Grabber (or alike) filter¹ before the AAC encoder and tracing the media samples. Also comparing them to data from another source. The encoder might be sensitive to small details like media sample duration or discontinuity flag on the first sample streamed.
¹ With GraphStudioNext (GraphEdit makes no longer sense compared to) you can use internal Analyzer Filter and review media sample flow interactively using filter property page.
I am rendering a video file from input pictures that come from a 3D engine at runtime (I don't pass an actual picture file, just RGB memory).
This works perfectly when outputting MP4 using CODEC_ID_H264 as video codec.
But when I want to create an MPG file using CODEC_ID_MPEG2VIDEO, the resulting file is simply broken. No player can play the video correctly and when I then want to concatenate that MPG with another MPG file, and transform the result MP4 in another step, the resulting .mp4 file has both videos, but many frames from the original MPG video (and only video! Sound works fine) are simply skipped.
At first I thought the MPG -> MP4 conversion was the problem, but then I noticed that the initial MPG, which comes from the video render engine, is already broken, which would speak for broken headers. Not sure if it is the system or sequence headers that are broken, though.
Or if it could be something totally different.
If you want to have a look, here is the file:
http://www.file-upload.net/download-7093306/broken.mpg.html
Again, the exact same muxing code works perfectly fine when directly creating an MP4 from the video render engine, so I'm pretty sure the input data, swscale(), etc. is correct. The only difference is that CODEC_ID_H264 is used and some additional variables (like qmin, qmax, etc.) are set, which are all specific to H264 so should not have an impact.
Also, neither avformat_write_header nor av_write_trailer report an error.
As an additional info, when viewing the codec data of the MPG in VLC player, it is not able to show the FPS, resolution and format (should show 640x360, 30 fps and 4:2:0 YUV).
I am using a rather new (2-3 months old, maybe) FFmpeg version, which I compiled from sources with MinGW.
Any ideas on how to resolve this would be welcome. Currently, I am out of those :)
Alright, the problem was not the avformat_write_header, but that I did not set the PTS value of each written video packet to AV_NOPTS_VALUE.
Once I do set it for each video packet, everything works fine.
I assumed that AV_NOPTS_VALUE was the default, as I never needed to set any special PTS value.