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• Pipe OpenCV and PyAudio to ffmpeg streaming youtube rtmp from python

  22 août 2021, par a0910115172

  How can I pipe openCV and PyAudio to ffmpeg streaming youtube rtmp from python.
The error message shows as following :
No such filter : 'pipe:1'
pipe:1 : Invalid argument

  


  Here is my code :

  


  Import module

  


  import cv2
import subprocess
import pyaudio


  


  Audio

  


  p = pyaudio.PyAudio()
info = p.get_host_api_info_by_index(0)
numdevices = info.get('deviceCount')
for i in range(0, numdevices):
        if (p.get_device_info_by_host_api_device_index(0, i).get('maxInputChannels')) > 0:
            print("Input Device id ", i, " - ", p.get_device_info_by_host_api_device_index(0, i).get('name'))

CHUNK = 1024
FORMAT = pyaudio.paInt16
CHANNELS = 2
RATE = 44100

stream = p.open(format=FORMAT,
                channels=CHANNELS,
                rate=RATE,
                input=True,
                frames_per_buffer=CHUNK)

# Stream Audio data here
# data = stream.read(CHUNK)


  


  Video

  


  rtmp = r'rtmp://a.rtmp.youtube.com/live2/key'

cap = cv2.VideoCapture(0)
width = int(cap.get(cv2.CAP_PROP_FRAME_WIDTH))
height = int(cap.get(cv2.CAP_PROP_FRAME_HEIGHT))
fps = 30



  


  command param here

  


  command = ['ffmpeg',
           '-y',
           '-f', 'rawvideo',
           '-pixel_format', 'bgr24',
           '-video_size', "{}x{}".format(width, height),
           '-framerate', str(fps),
           '-i', 'pipe:0',
           '-re',
           '-f', 'lavfi',
           '-i', 'pipe:1',
           '-c:v', 'libx264',
           '-c:a', 'aac',
           '-vf', 'format=yuv420p',
           '-f', 'flv',
           rtmp]


  


  Create subprocess to ffmpeg command

  


  pipe = subprocess.Popen(command, shell=False, stdin=subprocess.PIPE
)
while cap.isOpened():
    success, frame = cap.read()
    if success:
        if cv2.waitKey(1) & 0xFF == ord('q'):
            break
        pipe.stdin.write(frame.tostring())
        pipe.stdin.write(stream.read(CHUNK))


  


  Audio Stop

  


  stream.stop_stream()
stream.close()
p.terminate()


  


  Video Stop

  


  cap.release()
pipe.terminate()


  


  Thanks

  


• How to prevent screen tearing when updating stdin input to ffmpeg for youtube livestream

  8 septembre 2021, par Cameron Sima

  I am working on a livestream application that allows multiple web clients to stream video through webrtc to a 'controller' client which will add their audio track, then be able to switch between these video feeds and output the raw stream data to a server running ffmpeg, which will then send the feed to youtube live.

  


  The 'controller' client simply sets one peer video source as the 'active' stream on button press and all others as 'inactive'.

  


  The data is then sent to a node.js server like so :

  


        this.mediaSources.forEach((source, _) => {
    source.recorder.ondataavailable = (e: BlobEvent) => {
        if (source.active) {
          this.socket.emit('binarystream', e.data)
        }
      }


  


  The node server starts an ffmepg process like so :

  


  export class FfmpegService {
ffmpeg: ChildProcess
youtubeUrl = 'rtmp://a.rtmp.youtube.com/live2'

start(rtmpDestination: string) {
    this.ffmpeg = spawn('ffmpeg', this.getFfmpegOptions(rtmpDestination), 
        { shell: process.env.NODE_ENV !== 'production' }
    )
}

stop() {
    if (this.ffmpeg) {
        this.ffmpeg.stdin.end()
        this.ffmpeg.kill('SIGINT')
        this.ffmpeg = null
    }
}

feedStream(data: any) {
    this.ffmpeg.stdin.write(data)
}

private getFfmpegOptions(streamKey: string): string[] {
    const rtmpDestination = this.youtubeUrl + '/' + streamKey
    return [
        '-i','-',
        '-c:v', 'libx264', 
        '-preset', 'fast', '-tune', 'zerolatency',  // video codec config: low latency, adaptive bitrate
        '-c:a', 'aac', '-ar', '44100', '-b:a', '64k', // audio codec config: sampling frequency (11025, 22050, 44100), bitrate 64 kbits
        '-y', //force to overwrite
        '-use_wallclock_as_timestamps', '1', // used for audio sync
        '-async', '1', // used for audio sync
        //'-filter_complex', 'aresample=44100', // resample audio to 44100Hz, needed if input is not 44100
        //'-strict', 'experimental', 
        '-bufsize', '300k',
        '-pix_fmt', 'yuv420p',
        '-f', 'flv', rtmpDestination
    ];
}


  


  Everything works as expected in a youtube live session. The only problem is when switching between input streams, there is about 5 seconds of screen tearing before settling down. From all outward appearances, the switch happens immediately and seamlessly. I feel this can be attenuated/solved by tweaking the ffmpeg options but I'm pretty new to ffmpeg. I have tried increasing/decreasing -bufsize and -preset cli options, but nothing has worked so far.

  


• Live stream prerecorded video to YouTube using FFMPEG with 4500 kbps bitrate

  9 avril 2023, par Rsan

  I tried various ways to make a live streaming script with a bit rate according to YouTube's recommendation of 4500 Kbps bit rate.
The code :

  


  ffmpeg -re -stream_loop -1 -i live1.mp4 -c copy -preset veryfast -b:v 7000k -maxrate 3000k -bufsize 6000k -pix_fmt yuv420p -g 50 -c:a aac -b:a 160k -ac 2 -ar 44100 -f flv -flvflags no_duration_filesize rtmp://a.rtmp.youtube.com/live2/(streamkey)

  


  and in my current code, there is an error when live : Use a keyframe frequency of four seconds or less. Currently, keyframes are sent infrequently which can cause buffering. The current keyframe frequency is 5.0 seconds. Please note that errors in the transfer process can cause the size of the GOP (group of images) to be incorrect.

  


  How to fix my code ?

  


  I've tried several ways, but the bit rate is still high, and the error is in YouTube Studio