I have a problem using ffmpeg:
when i trying to record video+audio from my webcam in result i got only video stream, wthout audio at all.
I have tried different codecs and nothing..
Maybe, someone can give me advice?
ffmpeg -f dshow -i video="Logitech HD Webcam C270" -r 25 -s 800x600 -acodec libmp3lame -vcodec mpeg4 -b 3000k -f avi D:\1.avi
Btw: virtualdub grabs both well.
Thanks.
Assuming that you have installed the driver and codecs, use something like:
ffmpeg -f dshow -i video="Logitech HD Webcam C270" [path]out.mp4
A short explanation is given in capture a webcam input. For using DirectShow you have this examples.
Related
I am trying to use ffmpeg to combine 1 audio file (ADPCM) and 1 video file (h264) into single mp4. Video by file conversion works fine but ffmpeg chokes on guessing audio input. I can't figure out how to tell ffmpeg which params to use to decode raw audio file.
Currently I first run sox to convert raw audio to wav:
sox -t ima -r 8000 audio.raw audio.wav
... then feed audio.wav from sox as ffmpeg input
ffmpeg -i video.raw -i audio.wav movie.mp4
I am trying to avoid sox step and use audio.raw in ffmpeg.
Thank you
Since you have headless audio, you should tell ffmpeg about the sample format and (optionally) sample rate, audio channels, e.g.:
ffmpeg -i video.raw -f s16le -ar 22050 -ac 1 -i audio.raw movie.mp4
To check supported PCM formats you may use this command:
ffmpeg -formats 2>&1 | grep -i pcm
I am using ffmpeg to extract the audio from a video. Below code downlaods the audio from a video file. I'm not sure how efficient this program is but I do know that it downloaods it in 48KHZ.
How do I use this program to extract audio from a video in 8Khz because the file is getting too big.
ffmpeg -i video_link -vn output.wav
Use -ar option to change frequency rate
ffmpeg -i video_link -vn -ar 8000 output.wav
If you want to try different formats of audio check the available formats in ffmpeg using ffmpeg -formats and available codecs using ffmpeg -codecs
Here's an example to extract to mp3 file
ffmpeg -i video_link -vn -ar 8000 -f mp3 output.mp3
Edit: as #llogan pointed out, -f option is not needed, ffmpeg automatically mux mp3 file.
ffmpeg -i video_link -vn -ar 8000 output.mp3
I'm trying to calculate the audio + visual difference between a harshly compressed video file and one that hasn't been.
I'm using pipes because ultimately I wish this to take src from a camera stream.
I've managed to get the video results that I'm looking for, but I'm struggling with the audio.
I've added a line to invert the phase of the compressed audio, so that when they add up in the blend they should almost cancel each other out, but that doesn't happen.
ffmpeg -i input.avi -f avi -c:v libxvid -qscale:v 30 -c:a wmav1 - | \
ffmpeg -i - -f avi -af "aeval='-val(0)':c=same" - | \
ffmpeg -i input.avi -i - -filter_complex "blend=all_mode=difference" -c:v libx264 -crf 18 -f avi - | \
ffplay -
I can still hear all the audio, when what I should be hearing are solely compression artifacts. thx
To preface, I'm not sure your method would identify audio compression 'artifacts'
Your command doesn't perform any audio comparison, it only inverts a single channel. Also, the audio and video are compressed twice and the codecs the last ffmpeg command receives are the default AVI codecs of mpeg4 and mp3.
Use
ffmpeg -i input.avi -f matroska -c:v libxvid -qscale:v 30 -c:a wmav1 - |\
ffmpeg -i input.avi -i - -filter_complex "[0][1]blend=all_mode=difference;[1]aselect=gt(n\,0),asetpts=PTS-STARTPTS[1a];[0][1a]amerge,aeval=val(0)-val(1):c=mono" -c:v rawvideo -c:a pcm_s16le -f matroska - |\
ffplay -
I assume your audio is mono. If your audio has N channels, your aeval will need N expressions where the Mth expression is val(M-1)-val(N+M-1)
I also trim out the first encoded audio frame in order to mitigate encoder delay that Paul mentioned, and it seems to work here.
There might be some delay introduced with encoded audio samples. Also your command is incorrect.
I have a encoded Audio File(.aac file). I want to stream this file over RTP using FFMPEG without any transcoding. I am using following command :
ffmpeg -i input_file.aac -re -vn -acodec copy -strict experimental -f rtp rtp://225.0.0.1:1234
But above command gives below error:
AAC with no global headers is currently not supported
Can anyone point out any corrections ?
Thanks for the help.
The warning means it wants global headers to be set on your audio stream, like this:
ffmpeg -re -i input_file.aac -c:a copy -flags:a +global_headers -f rtp rtp://225.0.0.1:1234
I have a webcam and a separate mic. I want to record what is happening.
It almost works, however the audio seems to play quickly and parts missing while playing over the video.
This is the command I am currently using to get it partially working
ffmpeg -thread_queue_size 1024 -f alsa -ac 1 -i plughw:1,0 -f video4linux2 -thread_queue_size 1024 -re -s 1280x720 -i /dev/video0 -r 25 -f avi -q:a 2 -acodec libmp3lame -ab 96k out.mp4
I have tried other arguments, but unsure if it has to do with the formats I am using or incorrect parameter settings.
Also, the next part would be how to stream it. Everytime I try going through rtp it complains about multiple streams. I tried doing html as well, but didn't like the format. html html://localhost:50000/live_feed or rts rts://localhost:5000
edit:
I am running this on a rpi 3.