I am recording AVI files with Camtasia. For some reason the video stream length is 2,3-5 seconds less than the audio stream.
When I convert the video with ffmpeg from AVI to MP4 it cuts the audio to the video length.
Would duplicating the last frame until the end of the audio be a solution? If yes how can this be done using ffmpeg?
The important thing is to convert the AVI to MP4 using ffmpeg and keep the audio stream of the video complete.
Thank you.
Edit 1: This issue is automatically solved by ffmpeg 2.x somehow but ffmpeg 4.x will cut audio. With the same settings the old version converts correctly.
Edit 2: tpad helped. Thank you very much #kesh. I used
-filter_complex 'tpad=stop=NUMBER_OF_FRAMES:stop_mode=clone'
I tried to get the duration using ffprobe and multiplied the number of seconds with number of frames per second but it was not enough. For each video I had to increase that number with 100,150 frames.
The issue is I cannot detect the exact number of frames to tell tpad. I also tried
-filter_complex 'tpad=stop=-1:stop_mode=clone'
but it freezez while processing.
Is there any other option?
In our application, we are processing audio files using ffmpeg. Specifically, we use the NodeJS library fluent-ffmpeg, (npm link).
Our audio files are generated from various text to speech providers. We recently noticed that when we converted audio using ssml to add pauses to the generated audio, the duration on the file is no longer correct. Upon further investigation, we noticed that the standard audios were also incorrect, just more accurate overall due to the more consistent data. When we put a pause at the beginning of the audio, the estimate was the worst, overshooting it by a very large margin (e.g., a 25s audio clip would read as 3 minutes long, but skip to the end when playing past the 25s mark.
I did some searching and research into the structure of MP3 files, and to me it seems like the issue is because the duration gets estimated by various audio players. Windows media player is an example, but Firefox's web player seems to also do this. I tried changing the ffmpeg command from using .audioQuality(0), which sets ffmpeg to use VBR, to .audioBitrate(320), which tells ffmpeg to use a constant bitrate.
For reference, the we are using libmp3lame, and the full command that gets run is the following, for the VBR and CBR cases respectively:
For VBR (broken durations): ffmpeg -i <URL> -acodec libmp3lame -aq 0 -f mp3 pipe:1
For CBR (correct duration): ffmpeg -i <URL> -acodec libmp3lame -b:a 320k -f mp3 pipe:1
Note: we then pipe the output to the requesting client application after sending the appropriate file headers, hence the pipe:1 output. The input is a cloud storage url where the source file is located
This fixes our problem of having a correct duration, and it makes sense to me why this would fix it if the problem was because the duration is being estimated by some of these players / audio consumers. But, this came at the cost that the file size was significantly larger, which also makes sense to me. While testing we found that compared to the same file in WAV, the VBR mp3 was about 10% of the WAV file size, while the CBR mp3 was still 50% of the WAV file size. This practically defeats the purpose of supporting the mp3 format for our use-case, which is a smaller but slightly lossy alternative to the large WAV file.
While researching, I found that there can be ID3 tags in a chunk at the beginning of the mp3 file, specifying information for the consumer of the audio to know the duration before potentially having processed the whole file. But, I also found that there doesn't seem to be a standard, at least for duration. More things like song title, album, artist, etc.
My question is, is there a way to get the proper duration onto an mp3 file, preferably via some ffmpeg mechanism, while still using VBR? Thanks!
FFmpeg does write a Xing header by default with duration info. However, that value is only known after the entire stream data has been received, so ffmpeg has to seek to the head to write it. Since you're piping the output, that can't be done.
Write the file locally or to some seekable destination, and then upload.
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.
I'd like to sync an audio and video file together so that they end at the same time. Is there an easy way to do this with ffmpeg?
The only ideas I have are 1) finding the lengths of the two (which is annoying) and delaying the shorter one by the difference between the two lengths, and 2) reversing the two, syncing them, then reversing again. There's got to be a better way...
After finding the lengths of the two and calculating the required delay, you can run
ffmpeg -i video -i audio -af "adelay=6500|6500" -c:v copy output
Here, both the channels of a stereo audio stream are delayed by 6500 milliseconds i.e. 6.5 seconds. The video stream is copied over without recompression.