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  • Real time livestreaming - RPI FFmpeg and H5 Player

    29 avril 2022, par Victor

    I work at a telehealth company and we are using connected medical devices in order to provide the doctor with real time information from these equipements, the equipements are used by a trained health Professional.

    


    Those devices work with video and audio. Right now, we are using them with peerjs (so peer to peer connection) but we are trying to move away from that and have a RPI with his only job to stream data (so streaming audio and video).

    


    Because the equipements are supposed to be used with instructions from a doctor we need the doctor to receive the data in real time.

    


    But we also need the trained health professional to see what he is doing (so we need a local feed from the equipement)

    


    How do we capture audio and video

    


    We are using ffmpeg with a go client that is in charge of managing the ffmpeg clients and stream them to a SRS server.
This works but we are having a 2-3 sec delay when streaming the data. (rtmp from ffmpeg and flv on the front end)

    


    ffmpeg settings :

    


    ("ffmpeg", "-f", "v4l2", `-i`, "*/video0", "-f", "flv", "-vcodec", "libx264", "-x264opts", "keyint=15", "-preset", "ultrafast", "-tune", "zerolatency", "-fflags", "nobuffer", "-b:a", "160k", "-threads", "0", "-g", "0", "rtmp://srs-url")


    


    My questions

    


      

    • Is there a way for this set up to achieve low latency (<1 sec) (for the nurse and for the doctor) ?
    • &#xA;

    • Is the way I want to achieve this good ? Is there a batter way ?
    • &#xA;

    &#xA;

    Flow schema

    &#xA;

    Data exchange and use case flow :

    &#xA;

    Data exchange and use case flow

    &#xA;

    &#xA;

    Note : The nurse and doctor use HTTP-FLV to play the live stream, for low latency.

    &#xA;

    &#xA;

  • Should I use aevalsrc or anullsrc ? I got flooded with "cur_dts is invalid st:0" or "Delay between the first packet", encoding stalls at frame 127

    28 avril 2022, par jokoon

    Context : this command is generated with a python script, it aims at concatenating different small videos of different format, with a 0.5s pause, considering some video don't have an audio track.

    &#xA;

    ffmpeg.exe&#xA;-report&#xA;-i&#xA;input0.mp4&#xA;-i&#xA;input1.mp4&#xA;-i&#xA;input2.mp4&#xA;-i&#xA;input3.mp4&#xA;-i&#xA;input4.mp4&#xA;-i&#xA;input5.mp4&#xA;-i&#xA;input6.mp4&#xA;-i&#xA;input7.mp4&#xA;-i&#xA;input8.mp4&#xA;-f&#xA;lavfi&#xA;-i&#xA;color=c=black:s=480x270:d=0.5:r=24,format=yuv420p&#xA;-f&#xA;lavfi&#xA;-i&#xA;anullsrc&#xA;-filter_complex_script&#xA;filter_complex_script.txt&#xA;-map&#xA;[final_video]&#xA;-map&#xA;[final_audio]&#xA;-c:a&#xA;aac&#xA;concatenated-2022-04-27--23-53.mp4&#xA;

    &#xA;

    &#xA;

    [0]fps=24[fps0];&#xA;[fps0]scale=480:270:force_original_aspect_ratio=decrease,pad=480:270:-1:-1,setsar=1[rescaled0];&#xA;[1]fps=24[fps1];&#xA;[fps1]scale=480:270:force_original_aspect_ratio=decrease,pad=480:270:-1:-1,setsar=1[rescaled1];&#xA;[2]fps=24[fps2];&#xA;[fps2]scale=480:270:force_original_aspect_ratio=decrease,pad=480:270:-1:-1,setsar=1[rescaled2];&#xA;[3]fps=24[fps3];&#xA;[fps3]scale=480:270:force_original_aspect_ratio=decrease,pad=480:270:-1:-1,setsar=1[rescaled3];&#xA;[4]fps=24[fps4];&#xA;[fps4]scale=480:270:force_original_aspect_ratio=decrease,pad=480:270:-1:-1,setsar=1[rescaled4];&#xA;[5]fps=24[fps5];&#xA;[fps5]scale=480:270:force_original_aspect_ratio=decrease,pad=480:270:-1:-1,setsar=1[rescaled5];&#xA;[6]fps=24[fps6];&#xA;[fps6]scale=480:270:force_original_aspect_ratio=decrease,pad=480:270:-1:-1,setsar=1[rescaled6];&#xA;[7]fps=24[fps7];&#xA;[fps7]scale=480:270:force_original_aspect_ratio=decrease,pad=480:270:-1:-1,setsar=1[rescaled7];&#xA;[8]fps=24[fps8];&#xA;[fps8]scale=480:270:force_original_aspect_ratio=decrease,pad=480:270:-1:-1,setsar=1[rescaled8];&#xA;[rescaled0][10:a][9:v][10:a][rescaled1][10:a][9:v][10:a][rescaled2][10:a][9:v][10:a][rescaled3][10:a][9:v][10:a][rescaled4][10:a][9:v][10:a][rescaled5][5:a][9:v][10:a][rescaled6][6:a][9:v][10:a][rescaled7][7:a][9:v][10:a][rescaled8][8:a]concat=n=17:v=1:a=1[final_video][final_audio]&#xA;

    &#xA;

    I either have thousands of&#xA;[mp4 @ 000002132cefe100] Delay between the first packet and last packet in the muxing queue is 10007800 > 10000000: forcing output

    &#xA;

    when I use

    &#xA;

    -f lavfi -i anullsrc&#xA;so I tried using -f lavfi -i aevalsrc=0:n=1

    &#xA;

    instead, I have have thousands of

    &#xA;

    cur_dts is invalid st:0 (0) [init:1 i_done:0 finish:0] (this is harmless if it occurs once at the start per stream)

    &#xA;

    The encoding "stalls" at frame 127, it keeps doing things, but the frame counter doesn't increment.

    &#xA;

    Here are the inputs :

    &#xA;

    I skipped the yuvj420p input as it generated warnings (looks like it's not a big problem)

    &#xA;

    enter image description here

    &#xA;

    Here is a log (I trimmed it a bit) : (I think it's a bit large for stackoverflow)

    &#xA;

    https://bpa.st/HB7A

    &#xA;

  • Real time livestreaming - RPI

    24 avril 2022, par Victor

    I work at a telehealth company and we are using connected medical devices in order to provide the doctor with real time information from these equipements, the equipements are used by a trained health Professional.

    &#xA;

    Those devices work with video and audio. Right now, we are using them with peerjs (so peer to peer connection) but we are trying to move away from that and have a RPI with his only job to stream data (so streaming audio and video).

    &#xA;

    Because the equipements are supposed to be used with instructions from a doctor we need the doctor to receive the data in real time.

    &#xA;

    But we also need the trained health professional to see what he is doing (so we need a local feed from the equipement)

    &#xA;

    How do we capture audio and video

    &#xA;

    We are using ffmpeg with a go client that is in charge of managing the ffmpeg clients and stream them to a SRS server.&#xA;This works but we are having a 2-3 sec delay when streaming the data. (rtmp from ffmpeg and flv on the front end)

    &#xA;

    ffmpeg settings :

    &#xA;

    ("ffmpeg", "-f", "v4l2", `-i`, "*/video0", "-f", "flv", "-vcodec", "libx264", "-x264opts", "keyint=15", "-preset", "ultrafast", "-tune", "zerolatency", "-fflags", "nobuffer", "-b:a", "160k", "-threads", "0", "-g", "0", "rtmp://srs-url")&#xA;

    &#xA;

    My questions

    &#xA;

      &#xA;
    • Is there a way for this set up to achieve low latency (<1 sec) (for the nurse and for the doctor) ?
    • &#xA;

    • Is the way I want to achieve this good ? Is there a batter way ?
    • &#xA;

    &#xA;

    Flow schema

    &#xA;

    Data exchange and use case flow

    &#xA;