I am using moviepy (Python) to read video and audio frames of a video and after making some changes I am writing them back to a videofile, say new.avi, to preserve the changes, or to avoid compression, I am using codec= 'rawvideo' in write_videofile function. But when I read the video and audio frames back, the number of video and audio frames are different than when they were when written, they are usually increased.
Can anybody tell me the reason,? is it because of the ffmpeg used or some other reason? Does it happen always or there is some problem in my machine? Thank you :-)
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I use a video player called MPV to transcode a dynamic playlist of media files.
I pipe MPV's encoded output into FFMPEG and format it for rtmp delivery.
However the playlist may contain media with misaligned audio and video, ie - the audio track may be shorter / longer than the video track.
No matter what MPV will only output what it's given. So if my media file has audio that is 1 second long and video that is 2 seconds long, it will output a media stream with exactly the same misalignment, rather than generating null audio or skipping to the next item in the playlist when it first encounters an active stream ending (eof).
For example, assuming my playlist was full of problematic media where the audio and video of each file was misaligned:
If I output this media stream to a popular streaming service's server, it could lead to stuttering and/or loss of a/v sync.
Similarly, if I output this media stream to a file and played it back in MPV or another video player, the result appears to be more like this:
I have tried to fix this in MPV in all sorts of ways, trying every relevant command line option available. I even wrote a user script that detects 'eof' audio and skips to the next item in the playlist, but it is not fast enough and still leads to small gaps of audio.
So my only hope is correcting it in ffmpeg. In the event of null audio/video, I need a fallback or a generative filter that can fill these empty gaps with silence (audio) or a colour/image (video).
I'm open to any ideas, and if my understanding in a/v encoding is a little off please educate me.
I want to add a 5.1 .flac audio track to a .ts file that already has three audio tracks. I tried with tsMuxer and ffmpeg with unsuccessful results. In tsMuxeR the .flac track is not recognized and in ffmpeg everything seems to work fine until the very last moment when I check the file and the .flac audio track is not included in the "output.ts". The .flac track is about 3GB and its lenght is around two and a half hours.
Thank you so much.
I don't think you'll find any existing software that maps FLAC into a MPEG-2 Transport Stream.
This gives you an idea what sort of issues you run into: https://xiph.org/flac/ogg_mapping.html
Let's say you came up with a reasonable way of mapping FLAC into a MPEG-2 Transport Stream - there won't be anything reading it.
Unless there is a specified way of mapping FLAC into a MPEG-2 Tranport Stream - you are on your own.
But PCM is supported in a MPEG-2 Transport Stream (for example Blu-Ray).
I'd use ffmpeg to transcode your audio from FLAC to PCM and then mux it into your transport stream.
Your audio transcode (FLAC to PCM) is lossless.
During capturing from some audio and video sources and encoding at AVI container for synchronizing audio & video I set audio as a master stream and this gave best result for synchronizing.
http://msdn.microsoft.com/en-us/library/windows/desktop/dd312034(v=vs.85).aspx
But this method gives a higher FPS value as a result. About 40 or 50 instead of 30 FPS.
If this media file just playback - all OK, but if try to recode at different software to another video format appears out of sync.
How can I programmatically set dwScale and dwRate values in the AVISTREAMHEADER structure at AVI muxing?
How can I programmatically set dwScale and dwRate values in the AVISTREAMHEADER structure at AVI muxing?
MSDN:
This method works by adjusting the dwScale and dwRate values in the AVISTREAMHEADER structure.
You requested that multiplexer manages the scale/rate values, so you cannot adjust them. You should be seeing more odd things in your file, not just higher FPS. The file itself is perhaps out of sync and as soon as you process it with other applciations that don't do playback fine tuning, you start seeing issues. You might be having video media type showing one frame rate and effectively the rate is different.
I'm working on a project that requires me to sync an audio playback(preferably an mp3 file) with my program.
My program reads a motion file from a txt file and output's it onto the serial port at a particular rate. At the same time an audio file has to be played back on the speaker. This audio file has to be in sync with the data..that is to say after say transmittin 100 bytes of data, the audio mustve played back to a predefined time.
What would be the tools used to play and control audio like this?
a tutorial would be great!
Thanks!!
In general, when working with audio, you want to synchronize other sources to audio. This is for several reasons, but most important is that audio runs on a clock running on its own hardware. You'll have to get timing information from that clock. There is a guide here written for using portaudio, but the principles apply to other situations:
http://www.portaudio.com/docs/portaudio_sync_acmc2003.pdf
Has anyone tried to change the audio frame size in WebRTC? It uses 10ms frames. I need a different size, and the code doesn't look promising...
Fundamentally there's no reason you can't use non-10ms frames, but much of the code is written with that assumption. It would indeed likely be a serious undertaking to change it.
On the device you can use other audio frame (e.g. 20，40ms). For the codec，you can use other audio frame because codec has a audio buffer. I used silk codec and ios device.