I'm trying to record audio from one of my audio interfaces by using ffmpeg.
I want to do this in order to send this audio afterward via websocket with NodeJS.
I'm able to record the audio with ffmpeg and save it to an audio file, but I do not want to save it, I just want the audio stream in order to execute the ffmpeg command from NodeJS and use this audio stream with the websocket.
This is my current ffmpeg command:
ffmpeg -f alsa -i hw:0,0 -af "pan=2c|c0=c0" output.wav
Is there any way of using it without the output file?
Related
I'm working on a project that requires a live audio to be transcribed in real-time. I tried the AWS Transcribe with WebSockets using their starter code available on GitHub.
Currently, for testing I have an audio file from a YouTube which I'm streaming to an icecast2 server hosted on a Cloud VM.
The ffmpeg command for streaming to the icecast2 server is
ffmpeg -re -i yt.wav -ar 44100 -ac 1 -c:a libvorbis -aq 5 -content_type 'audio/ogg' -vn -f ogg icecast://source:hackme#serverIP:8000/mystream.ogg
I've modified the code from GitHub such that instead of reading audio data from a microphone it reads the audio from icecast2 server. The problem with this is all it sometimes doesn't return a transcript at all or returns the wrong transcript.
I'd really appreciate if anyone could help
I would like to play a separate video stream and audio stream simultaneously from the Linux command line, using e.g. cvlc or mpv.
More specifically, I would like to play a youtube video in high quality format, using youtube-dl along with a player.
More details:
I am using this command to playback a youtube video on my pc:
youtube-dl -i <youtube.com/url> -o - | mpv -
Lets say I have following formats for a youtube video available:
249 webm audio only tiny 62k , opus # 50k (48000Hz), 14.14MiB
251 webm audio only tiny 158k , opus #160k (48000Hz), 35.68MiB
303 webm 1920x1080 1080p60 4429k , vp9, 60fps, video only, 536.78MiB
299 mp4 1920x1080 1080p60 6901k , avc1.64002a, 60fps, video only, 884.09MiB
22 mp4 1280x720 720p 1339k , avc1.64001F, 30fps, mp4a.40.2#192k (44100Hz) (best)
youtube-dl would automatically choose the last entry of this list, as it is a format that includes video and audio in one file.
Is there a way I can play the formats 303 and 251 on my pc?
If I would like to download them I would use:
youtube-dl -i <youtube.com/url> -f 303+bestaudio
What youtube-dl does in this case is to download the video and the audio file seperately and merges them into one file using ffmpeg.
But I can't figure if there is a possibility to playback both streams without first downloading them into a file.
Alright I think I figured a solution.
The command I use is as follows:
ffmpeg -loglevel quiet -i $(youtube-dl -g youtube.com/url -f 303) -i $(youtube-dl -g youtube.com/url -f bestaudio) -f matroska -c copy - | mpv -
The youtube-dl -g option would just return the url to the video or audio stream.
In this case it will pass the urls to ffmpeg which is doing the merging process.
-f matroska tells ffmpeg to use the mkv container format
-c copy says that no re-encoding should be done
edit:
For some reason, on my systemm the input is broken after ffmpeg exits. For now I resolve this by typing reset, until I find a better solution to this issue.
I am trying to write a code to download YouTube videos using Pytube on Python 3.6. But for most videos progressive download(Audio and Video in same file) format is available only upto 360p. So I want to download audio and video files separately and combine it. I am able to to download the audio and video files. How can I combine the two file together?
Basically I don't find any method to marge Audio and Video in Pytube but you can use ffmpeg for muxing.
First of all you have to install ffmpeg
ffmpeg installation guide for Windows
for Ubuntu just sudo apt install ffmpeg
Add a dependency ffmpeg-python a python wrapper of ffmpeg
pip install ffmpeg-python
Now we are ready to go with this code snippet
import ffmpeg
video_stream = ffmpeg.input('Of Monsters and Men - Wild Roses.mp4')
audio_stream = ffmpeg.input('Of Monsters and Men - Wild Roses_audio.mp4')
ffmpeg.output(audio_stream, video_stream, 'out.mp4').run()
for more, ffmpeg-python API References
If you keep getting a video without audio, that's because of the adaptive streaming from pytube. A work-around is to download both video and audio... then merge them with ffpmeg.
For instance, something like this to get both audio and video (audio part adapted from here)
from pytube import YouTube
import os
youtube = YouTube('https://youtu.be/ksu-zTG9HHg')
video = youtube.streams.filter(res="1080p").first().download()
os.rename(video,"video_1080.mp4")
audio = youtube.streams.filter(only_audio=True)
audio[0].download()
and then the ffmpeg part (adapted from both here and here) you can set it up on Windows following this procedure and then run something like
ffmpeg -i video.mp4 -i audio.mp4 -c:v copy -c:a aac output.mp4
Merging audio and video using ffmpeg
Once you have downloaded both video and audio files (‘videoplayback.mp4’ and ‘videoplayback.m4a’ respectively), here’s how you can merge them into a single file:
In case of MP4 format (all, except 1440p 60fps & 2160p 60fps):
ffmpeg -i videoplayback.mp4 -i videoplayback.m4a -c:v copy -c:a copy output.mp4
In case of WebM format (1440p 60fps and 2160p 60fps):
ffmpeg -i videoplayback.webm -i videoplayback.m4a -c:v copy -c:a copy output.mkv
Wait until ffmpeg finishes merging audio and video into a single file named "output.mp4".
How do I convert the downloaded audio file to mp3?
you need to execute the following command in the Command Prompt window:
ffmpeg -i INPUT_FILE -ab BITRATE -vn OUTPUT_FILE
Example:
ffmpeg -i videoplayback.m4a -ab 128000 -vn music.mp3
Example:2 (without bit rate)
ffmpeg -i videoplayback.m4a -vn music.mp3
I have a dummy client that is suppose to simulate a video recorder, on this client i want to simulate a video stream; I have gotten so far that i can create a video from bitmap images that i create in code.
The dummy client is a nodejs application running on an Raspberry Pi 3 with the latest version of raspian lite.
In order to use the video I have created, I need to get ffmpeg to dump the video to pipe:1. The problem is that I need the -f rawvideo as a input parameter, else ffmpeg can't understand my video, but when i have that parameter set ffmpeg refuses to write anything to stdio
ffmpeg is running with these parameters
ffmpeg -r 15 -f rawvideo -s 3840x2160 -pixel_format rgba -i pipe:0 -r 15 -vcodec h264 pipe:1
Can anybody help with a solution to my problem?
--Edit
Maybe i sould explain a bit more.
The system i am creating is to be set up in a way, where instead of my stream server ask the video recorder for a video stream, it will be the recorder that tells the server that there is a stream.
I have have slowed my problem on my own. (-:
i now have 2 solutions.
Is to change my -f rawvideo to -f data that works for me anyways.
I can encode my bitmaps as jpeg in code and pipe my jpeg images to stdin. This also requires me to change the ffmpeg parameters to -r 4 -f mjpeg -i pipe:0 -r 4 -vcodec copy -f mjpeg pipe:1 and is by far the slowest thing i have ever done. and i can't use a 4k input
thanks #Mulvya for trying to help.
#eFox Thanks for editing my stupid spelling and grammar mistakes
I'm developing Android application, and im using ffmpeg for conversion of files.
I want my binary file to be as slim as possible since i don't have many input formats and output formats, and my operation is quite basic.And of course not to bloat the APK.
In my program ffmpeg receives a file, and copys the audio stream (-acodec copy), the audio stream will always be aac (mp4a). What i need is to save the stream to file.
My command looks like this : ffmpeg -i {Input} -vn -acodec copy output.aac.
What muxer do i need to for muxing aac to file? I have tried flv,mp3,mov but i always get
Unable to find a suitable output format for 'output.aac', so these options are wrong.
I don't need an encoder for stream copy btw.
Side note: this command work flawlessly on full installation of ffmpeg , but I don't know which muxer it uses. If there is a way to output the muxer it uses from regular ffmpeg run, it would work too.
A common file format for AAC is BMFF/MOV/MP4/M4A. If you specify the m4a file extension, FFmpeg will take care of it for you.
ffmpeg -i {input} -vn -acodec copy output.m4a
If you just want raw AAC, you can use ADTS as a lightweight container of sorts, as Mulvya suggested.
ffmpeg -i {input} -vn -acodec copy -f adts output.aac
I had to add the -f for me to work (on FFmpeg 3.22):
ffmpeg -i {input} -vn -acodec copy -f adts output.m4a
You have to add adts to --enable-muxer when configuring ffmpeg, eg. ./configure --disable-everything (...) --enable-muxer=adts. Then you will be able to save to .aac file