I'm trying to record from a microphone and webcam on MacOS with the following command:
ffmpeg -f avfoundation -framerate 30 -i "0:0" ~/recorded.mp4
My result has crackling in the audio.
I'm familiar with this problem when you use a DAW: you solve it by increasing the sample buffer. The idea is that audio samples coming from your interface/mic are not coming in a consistent or fast enough rate, so the missing samples being filled with zeroes causes the crackling sound. To avoid missing samples you want the recording software to wait longer for samples accumulating in a buffer before they're processed.
How can you configure such buffer for ffmpeg?
Version 4.3 seems to have this issue. Try with 4.2.
It seems to me that giving the -aq 0 audio quality parameter reduces this issue.
ffmpeg -f avfoundation -i 0:0 -acodec pcm_f32le -ar 48000 -aq 0 output.wav
Related
The situation is kind of complex. I was archiving several CCTV camera feeds (rtsp, h264, no audio) through OpenCV, which worked but the CPU utilization was too high and started to lose some frames time by time.
To reduce the CPU utilization, I started to use FFMPEG to skip the decoding and encoding processes, which worked perfectly on my home machine. However, when I connected to my university VPN and tried to deploy it on our lab server, FFmpeg couldn't read any frame, ffplay couldn't get anything either. However, OpenCV, VLC Player and IINA Player could still read and display the feed.
In Summary,
1 FFMPEG/ffplay
1.1 can only read the feed from my home network(Wi-Fi, optimum)
1.2 from other two networks, the error message says: "Could not find codec parameters for stream 0 (Video: h264, none): unspecified size
Consider increasing the value for the 'analyzeduration' (0) and 'probesize' (5000000) options"
2 IINA/VLC Player, OpenCV
These tools can get the video all the time.
I'm wondering whether it's related to some specific port access, that the ffmpeg required but the others don't. I'd appreciate it if anyone can provide any suggestions.
As references, the tested ffplay command is simple:
ffplay 'the rtsp address'
Thanks
Update
More tests have been performed.
By specifying rtsp_transport as TCP, ffplay can play the video, but FFmpeg can't access the video. (In the beginning, when both FFmpeg and ffplay worked through my home network, it was UDP)
The FFmpeg command is as follows:
ffmpeg -i rtsp://the_ip_address/axis-media/media.amp -hide_banner -c:v copy -s 1920x1080 -segment_time 00:30:00 -f segment -strftime 1 -reset_timestamps 1 -rtsp_transport tcp "%Y-%m-%d-%H-%M-%S_Test.mp4"
Please help...
Solved by forcing it to use "-rtsp_transport tcp" right before -i.
ffmpeg -rtsp_transport tcp -i rtsp://the_ip_address/axis-media/media.amp -hide_banner -c:v copy -s 1920x1080 -segment_time 00:30:00 -f segment -strftime 1 -reset_timestamps 1 "%Y-%m-%d-%H-%M-%S_Test.mp4"
i try to concat multiple videos to one video and add an background music to it.
for some reason the background music is perfectly added to the output video but the audio of each part of the output is speed up to a chipmunk version of the video itself. this results in an output video of 7 minutes with about 5 minutes of silence since everything is so fast that all the audio finishes after about 2 minutes.
my command is:
ffmpeg -safe 0 -i videolist.ffconcat -i bg_loop.mp3 -y -filter_complex "[1:0]volume=0.3[a1];[0:a][a1]amix=inputs=2" -vcodec libx264 -r 25 -filter:v scale=w=1920:h=1080 -map 0:v:0 output.mp4
i tried to remove the background music (since i wasn't able to loop it through the video i thought maybe that's the issue) and still.. all the audio of the video clips is still speed up resulting in chaotic audio at the beginning and silence at the end.
my video list looks like this:
ffconcat version 1.0
file intro.mp4
file clip-x.mp4
file clip-y.mp4
file clip-x.mp4
file clip-y.mp4
[... and so on]
i hope somebody can tell me what i'm doing wrong here (and maybe how to adjust my command to loop the background music through all the clips)
i googled a bit and found the adjustment of my command to add amix=inputs=2:duration=first but that doesn't do the trick and if i add duration=shortest or duration=longest nothing changes the output audio
The concat demuxer requires that all streams in inputs have the same properties. For audio, that includes codec, sampling rate, channel layout, sample format..
If audio of some inputs is sounding funny after concat, that usually indicates a sampling rate mismatch. Run ffprobe -show_streams -select_streams a -v 0 "input-file" on each input to check. For those which are different, you can re-encode only the audio by adding -ar X where X is the most common sampling rate found among your inputs e.g. -ar 44100. Other parameters will depend on format details. Keep video by using -c:v copy.
I generated a waveform of the raw audio with ffmpeg and it looks as perfectly normal audio on the picture. But when I imported it in Audacity, I can see and hear that the audio is clipping.
I tried simple command:
ffmpeg -f s16le -ar 16000 -ac 1 -i audio.raw -lavfi showwavespic audio.png
With this result: waveform image 1
And also tried this solution https://stackoverflow.com/a/32276471/12253501
Looks much prettier, but still can't see that audio actually clipped: waveform image 2.
For the second one I made sure to delete "compand=gain=-6" to get the actual audio level.
And here's what I see in Audacity: screenshot from Audacity
The audio I'm importing is raw data (16-bit signed PCM, Little-endian, 16000Hz Sample Rate). I also tried converting it to WAV first, but got the same results with ffmpeg and Audacity.
I'm wondering what I'm doing wrong and how to I get to see clipping on ffmpeg waveform output?
Here's the link for the audio file: raw and wav
Your audio is definitely clipping. Try setting scale=log so you can see this better.
ffmpeg -i audio.wav -lavfi showwavespic=scale=log audio.png
Also keep in mind that some of this clipping may have occurred before this version of the file. If this is clipping on the input stage of the recorder, but the recorded output was down a few dB, then you can hear clipping but not necessarily see it in the raw levels.
I couldn't achieve the results I wanted with just the ffmpeg, and I end up using audiowaveform, which gave me result similar to Audacity. Not sure if it's RMS or Peak, but works for me. Unfortunately, audiowaveform can't work with raw audio:
ffmpeg -f s16le -ar 16000 -ac 1 -i audio.raw audio.wav
audiowaveform -i audio.wav -o audio.png -e 105
Other downside of audiowaveform is that by default it only generates waveform for the first 15 seconds. Needs to calculate the length of the audio and give it as a parameter (-e seconds).
Here's the result:
I've been testing different parameters to capture my desktop video and audio (desktop audio, not mic) and I find that no matter what settings I have, the resulting webm file's framerate is around 5fps and is horribly inconsistent. It starts at around 20fps and slowly drops over time until about 4-5fps. I'm not really sure what I'm doing wrong, but here is the basic command I'm using:
ffmpeg -y -video_size 1920x1080 -f gdigrab -framerate 60 -i desktop -c:v libvpx-vp9 -acodec libvorbis -c:a libopus -b:v 2M -threads 4 output.webm
I've tried anywhere between 30-60 fps and tested different bitrates but nothing seems to affect the output framerate.
Also, I know that acodec and c:a are for audio but I'm not sure how to specify the audio device to use.
So my issues are horrible framerate for webm and how to include desktop audio in the recording.
You can use arecord and pipe it through stdout and ffmpeg can read it from stdin.
aplay piping to arecord using a file instead of stdin and stdout
Replacing the aplay command with your ffmpeg. Dont forget to add '-i -' in ffmpeg.
A doubt: why are you defining audio encoder two times?
It's impossible to say why the video frame rate is low from the question. It can be an issue with encoder. Or issue in reading input. Remove the video encoding option. See if the issue persists. If it's working fine, try some other encoders.
Use -c:v libx264 instead of -c:v libvpx-vp9. libvpx-vp9's realtime encoding quality is really bad, even regular libvpx (i.e. VP8) is much better. If you insist on using libvpx, use options like -deadline realtime and -cpu-used -4
everyone.
I'm trying to use FFmpeg to record video and 3 audio sources and use it to generate 3 different video files - each file should contain the same video stream but the different audio stream. The problem is that I got audio sync issues. The first audio stream is synced perfectly, but the second one has 1 sec lag, and the third one has like 2 sec lag.
I've made a few tests so far and it seems that root cause of the issue is initialization time of video/audio devices. So, one device is already recording something but the second is still being opened and so on. I've tried to change input devices order and after that audio streams still have the same issue BUT if before 2nd and 3rd audio streams were some time ahead of video, after reordering they became to lag after the audio (audio for the same event appears with some delay). So this test confirms my version about device initialization times.
But the question still, why the first audio stream is synchronized properly, while other 2 are not. and also, how could I overcome this issues? Any workarounds and ideas are highly appreciated.
Here is FFmpeg command I'm using and it's output.
ffmpeg.exe -f dshow -video_size 1920x1080 -i video="Logitech HD Webcam
C615" -f dshow -i audio="Microphone (HD Webcam C615)" -f dshow -i
audio="Microphone Array (Realtek High Definition Audio)"
-filter_complex "[1:a]volume=1[a1];[2:a]volume=1[a2]" -vf scale=h=1080:force_original_aspect_ratio=decrease -vcodec libx264
-pix_fmt yuv420p -crf 23 -preset ultrafast -acodec aac -vbr 5 -threads 0 -map v:0 -map [a1] -map [a2] -f tee
"[select=\'v,a:0\']C:/Users/vshevchu/Desktop/123/111/111_jjj1.avi|
[select=\'v,a:1\']C:/Users/vshevchu/Desktop/123/111/111_jjj2.avi"
OUTPUT
PS. Actually, the issue is exactly the same when I'm not using "tee" muxer but writing all the audio streams to one container. So, "tee" isn't a suspect.