Do you know ffmpeg audio filter settings to remove sharp signal spikes?
I already tried some ffmpeg filters (adeclip etc.) but nothing worked.
I have spikes on one of my microphone audio channels, screenshot attached..
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I have a recording as a collection of files in mpegts format, like
audio: a-1.ts, a-2.ts, a-3.ts, a-4.ts
video: v-1.ts, v-2.ts, v-3.ts
I need to make a single video clip in mp4 or mkv format.
However, there are two problems:
audio and video segments have different duration each, number of audio segments is different from number of video segments. Total duration of audio and video matches. Hence I can not concat pairwise audio video segments using mpeg and merge them afterwards, I get sync issues increasing progressively
few segments are corrupt or missing. So if I concat audio and video streams separately using ffmpeg I get streams of different lengths. When I merge these streams using ffmpeg I have correct a/v synchronization until time when first missing packet is encountered.
It's OK if video freezes for a while or there is silence for a while as long as most of the video is in sync with audio.
I've checked with tsduck and PCR seems to be present in all audio and video segments yet I could not find a way to merge streams using mpegTS PCR as sync reference. Please advise how can I achieve this.
I am using FFmpeg to record audio using pulse audio at a 48k sampling rate and 32k bitrate. Eventually, I perform encoding these recorded samples through opus which is configured on default settings such as frame_size set to 120, initial padding set to 120 and etc. I am not sure about the recording callback settings for pulse audio but I have read in opus RFC that it uses 120ms by default and can support 10, 20, 40, 60, and any other which is a multiple of 2.5.
I have a project where I have added WebRTC p2p connection support to send FFmpeg recorded audio/video on the audio/video channel of WebRTC to the next connected peer. In the case of sending plain audio as well as plain video which is recorded through FFmpeg, there is no issue with sending these over an audio/video channel (obviously) however, in the case of FFmpeg encoded data, I have to modify WebRTC audio/video encoder factories (clear to me).
I want to capture raw audio samples using FFmpeg after every 10MS callback.
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?
I'm doing voiceover and since Sony Vegas does not support sidechaining, I render voiceover into voices.wav and then use sidechain_compress filter, as per ffmpeg documentation:
ffmpeg -y -i background.m4a -i voices.wav -filter_complex \
"[1:a]asplit=2[sc][mix];\
[0:a][sc]sidechaincompress=threshold=0.015:ratio=2:level_sc=0.8:release=500:attack=1[compr];\
[compr][mix]amerge" sidechain_1.wav
voices.wav is a stereo audio file, as well as background.m4a. But here's how the result file looks like when loaded into Sony Vegas:
This shows that in channels 1/2 I get the compressed background, while in channel 3 and 4 I get two mono tracks that somehow differ (probably, that's the original voices input and somewhat altered voices input, both in mono). UPD: I don't want to further process resulting tracks in Sony Vegas, I'd prefer ffmpeg to be the last step in my production process. The screenshot above is for illustration purposes only.
Is the background gets sidechain compressed with only left or right channel of voices? If so, how to change that to make it compressed by both channels (some voices are panned into left or right, so there might be actual difference in compressed result)
What are those channels 3 and 4? Why are they mono?
How do I get single 1/2 stereo track in the output wav file instead of this weird 4 channels in 3 tracks? (I've looked at pan complex filter, but didn't figure out how to set it up in my case).
amerge adds the channels of the inputs. amix uses the channel count of the input with the most channels. So, switch to amix.
ffmpeg -y -i background.m4a -i voices.wav -filter_complex \
"[1:a]asplit=2[sc][mix];\
[0:a][sc]sidechaincompress=threshold=0.015:ratio=2:level_sc=0.8:release=500:attack=1[compr];\
[compr][mix]amix" sidechain_1.wav
For the past few days, I have been trying to get my lossless .mov video(that has an audio track) to a .webm format.
Some info on the video & audio is that the fps is 30. Also the audio track has about 3-5 seconds of silence/blank audio before you start hearing some music.
My problem is that is seems during the transcoding to webm, it strips away this blank audio because when I go to play the video, the audio starts right away.I've also notice that it jumps right away to ~4 seconds in the video. When i play it on the browser, it jumps to that moment in the timeline. If I try to scrub to the beginning, the video ends.
I've have figured somethings out.
This is just a webm problem. This does not happen with ogv or mp4
It only happens if they is blank audio in the beginning of the audio track.
I am using ffmpeg with the libvpx and libvorbis librarys and I am doing just the basic command line setup
ffmpeg -i "infile" "outfile.webm"