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Repo info
    Ramon Caldeira
    Alex Hutnik
    Any ideas on reading a h264 mp4 file while it's being written? i'm recording from a RTSP stream to a file and want to read it "up to the most recent write" so to speak. Preferably through httpd which doesn't work and I suspect I have to process the packets through pyav to apply some metadata. Hence why I'm here :)
    Ramon Caldeira
    @alexhutnik you can read the RTSP from pyav and from there write into the file and process the packets
    I'm trying to generate my own PyAV wheel but it looks like it's not including FFMPEG... I'm trying with $PYAV_PYTHON setup.py bdist_wheel...
    Could anybody help me fix it?
    Victor Sklyar

    @alexhutnik you can read the RTSP from pyav and from there write into the file and process the packets

    seems i have the same trouble, I need support https protocol but have to rebuild ffmpeg

    Victor Sklyar
    Now I trying build 8.0.1 such as located in pip repository, but cannot put ffmpeg .so to whl.
    Can someone point me to sample or something to help solve this trouble?
    Alex Hutnik
    @ramoncaldeira I've tried doing that:
    if __name__ == "__main__":
        fh = open('video_bytes.bin', 'rb')
        codec = av.CodecContext.create('h264', 'r')
        while True:
            chunk = fh.read(1 << 16)
            packets = codec.parse(str(chunk))
            print("Parsed {} packets from {} bytes:".format(len(packets), len(chunk)))
            for packet in packets:
                print('   ', packet)
                frames = codec.decode(packet)
                for frame in frames:
                    print('       ', frame)
    Can't parse any packets
    tried to just parse the chunk directly, but parse complained that chunk was not a str
    i tried writing the packets both as raw bytes (using packet.to_bytes()) and to a stream container
    Also note that I am parsing a closed file. So this isn't even being attempted on a file that is being written to
    Alex Hutnik
    maybe there is an issue with parse
    I am trying to apply FFmpeg filters to a .wav tempfile. I am getting the error: av.error.ValueError: [Errno 22] Invalid argument on the line for p in out_stream.encode(ofr): of the offending code.
    The offending code:
            in_cont = av.open(curr_tempfile.name, mode='r')
            out_cont = av.open(output_tempfile.name, mode='w')
            in_stream = in_cont.streams.audio[0]
            out_stream = out_cont.add_stream(
            graph = Graph()
            fchain = [graph.add_abuffer(sample_rate=in_stream.rate, 
                format=in_stream.format.name, layout=in_stream.layout.name, 
            for _filter in filters:
            for ifr in in_cont.decode(in_stream):
                ofr = graph.pull()
                ofr.pts = None
                for p in out_stream.encode(ofr):
    The list filterscontains lambda functions of the format: lambda graph: graph.add("name_of_filter", "named_filter_arguments")
    For example a _filter in filters would be lambda graph: graph.add("tremolo", "f=15:d=.5")
    Hieu Trinh

    I got a VideoFrame in yuvj420p and I want to convert it to OpenCV Mat which is bgr24.

    <av.VideoFrame #3, pts=1920 yuvj420p 1920x1080 at 0x7f16f0aaf7c8>

    Currently when I run frame.to_ndarray(format='bgr24') I get deprecated pixel format used, make sure you did set range correctly. Is there a proper way to convert to prevent this warning ?
    I tried :

    arr = frame.to_ndarray()   
    mat= cv2.cvtColor(arr, cv2.COLOR_YUV2BGR_I420)

    But got ValueError: Conversion to numpy array with format yuvj420p is not yet supported.
    I know that I can use av.logging.Capture() to catch the warning, but just wondering if there is a proper way to do the conversion ?

    Ramon Caldeira
    @mhtrinhLIC try updating PyAV to a version >= 8.0.0. support for yuvj420p was recently added
    Joanna Bitton

    I'm trying to write a video containing audio using lossless codecs. I'm using "libx264rgb" for my video codec which works perfectly fine and as expected. However, I've been having trouble finding a compatible lossless audio codec.

    I tried using "flac", but I got an error saying that that is an experimental feature:

    > Traceback (most recent call last):
    >   File "/data/users/jbitton/fbsource/fbcode/buck-out/dev/gen/aml/ai_red_team/augmentations/tests/video_tests/pytorch_test#binary,link-tree/aml/ai_red_team/augmentations/tests/video_tests/pytorch_test.py", line 92, in test_compose_without_tensor
    >     audio_codec="flac",
    >   File "/data/users/jbitton/fbsource/fbcode/buck-out/dev/gen/aml/ai_red_team/augmentations/tests/video_tests/pytorch_test#binary,link-tree/torchvision/io/video.py", line 129, in write_video
    >     container.mux(packet)
    >   File "av/container/output.pyx", line 198, in av.container.output.OutputContainer.mux
    >   File "av/container/output.pyx", line 204, in av.container.output.OutputContainer.mux_one
    >   File "av/container/output.pyx", line 174, in av.container.output.OutputContainer.start_encoding
    >   File "av/container/core.pyx", line 180, in av.container.core.Container.err_check
    >   File "av/utils.pyx", line 107, in av.utils.err_check
    > av.AVError: [Errno 733130664] Experimental feature: '/tmp/tmpbwvyrh3j/aug_pt_input_1.mp4' (16: mp4)

    Additionally, I tried using "mp4als" which is a valid audio codec in ffmpeg, however PyAV raises an error saying it is an unknown codec.

    Would appreciate any advice, thanks in advance!

    Does anyone know how to tell a stream that there has been a frame drop? basically I'm receiving an audio stream through the network and re-encoding it. The problem is that as soon as there is a frame drop the encoder fails on (pts X >= expect pts Y). To fix it I set the pts to None before calling encode with the frame but then it means the played frame is considered the next which makes the video out of sync audio/video wise. Here's a code sample of what I'm doing:
    import av
    container = av.open(rtsp_url)
    video_stream = container.streams.video[0]
    audio_stream = container.streams.audio[0]
    out_container = av.open(some_file, mode="w", format="mpegts")
    video_stream_out = out_container.add_stream(template=video_stream)
    audio_stream_out = out_container.add_stream(codec_name="aac", rate=44100)
    while True:
        packet = next(container.demux(video_stream, audio_stream))
        if packet.stream.type == 'video':
            packet.stream = video_stream_out
        elif packet.stream.type == 'audio':
            audio_frames = packet.decode()
            for a_frame in audio_frames:
                a_frame.pts = None # If I don't set this to None if there is a packet drop the encode function will fail...
                packets = audio_stream_out.encode(a_frame) # When I do set it to None, audio will play ridiculously fast
                for a_packet in packets:
                    a_packet.stream = audio_stream_out
    1 reply
    Razvan Grigore
    hey, I'm trying to build https://github.com/jocover/jetson-ffmpeg codec into av, should pip3 install av --no-binary av work with the custom compiled ffmpeg? I still get this error: pkg-config returned flags we don't understand: -pthread -pthread any hints?
    Razvan Grigore
    -- worked from deb-src modified build
    Luke Wong
    hi, i want to know :how to pyav read rtmp stream nobuffer ,like ffmpeg command line : ffplay -fflag nobufer rtmp://xxxxxxx
    Brian Sherson
    Isn’t len(frame.planes) supposed to be equal to len(frame.layout.channels)?
    >>> frame = next(d) ; frame
    <av.AudioFrame 0, pts=0, 512 samples at 48000Hz, 7.1, s32p at 0x7fd146428b38>
    >>> len(frame.planes)
    >>> len(frame.layout.channels)
    I suspect this has something to do with why frame.to_ndarray() throws a segfault on 7.1 layout audio.
    Akshay Bhuradia

    I want to convert m3u8 to mp4 and i would like to do conversion as well as download on same time. for download i have used http protocol of ffmpeg.

    I am running this command

    ffmpeg -i ultra.m3u8 -c copy -listen 1 -seekable 1 -f mp4

    when i trigger this url(""), then file start's download, but i am not able to play video.

    and i get error when all chunks are read:

    [hls @ 0x55da053b4100] Opening 'ultra177.ts' for reading
    [tcp @ 0x55da0540f940] Connection to tcp:// failed: Connection refused
    [tcp @ 0x55da05520480] Connection to tcp:// failed: Connection refused
    [tcp @ 0x55da053ca780] Connection to tcp:// failed: Connection refused
    [tcp @ 0x55da05485f80] Connection to tcp:// failed: Connection refused
    [tcp @ 0x55da053ced40] Connection to tcp:// failed: Connection refused
    [tcp @ 0x55da054255c0] Connection to tcp:// failed: Connection refused
    [tcp @ 0x55da0540f940] Connection to tcp:// failed: Connection refused
    [tcp @ 0x55da05435380] Connection to tcp:// failed: Connection refused

    frame=53236 fps=7939 q=-1.0 Lsize= 476447kB time=00:29:36.30 bitrate=2197.3kbits/s speed= 265x
    video:446847kB audio:28278kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.278083%


    Hello, I'm new to PyAV. I've been trying to record my screen using 'x11grab' and then broadcast it to an RTMP stream, but I've been having no luck. Here's my code:

    import av
    if __name__ == '__main__':
        x11grab = av.open(':0.0', format='x11grab', options={'s':'1280x1024', 'framerate':'60'})
        output = av.open('rtmp://localhost', format='flv')
        ostream = output.add_stream('libx264', framerate=60, video_size='1280x1024')
        ostream.options = {}
        ostream.pix_fmt = 'yuv420p'
        istream = x11grab.streams.video[0]
        gen = x11grab.demux(istream)
        for _ in range(1000):
            packet = next(gen)
            packet.stream = ostream
            if packet.dts is None:

    When I run this, it just hangs, no output whatsoever. Could somebody please point me in the right direction on how I could go about making this work? Thank you!

    Daniel Takacs
    I want to generate a video with changing resolution (e.g.: first frame should be 640x480 but the next is 320x240 or 480x640). When I naively started changing the stream.width and stream.height I got Input picture width (xxx) is greater than stride (yyy) error during the next stream.encode() so I suppose I need to do something more. Any idea what is missing? Is it possible to do with pyav or I need to go down to the C api?
    Jimmy Berry
    Is it possible (and if so how) to access frame metadata like lavfi.astats.Overall.RMS_level when using a av.filter.Graph()configured with astats=metadata=1? frame.side_data is seemingly always empty after push() frame into graph and pull() back out.
    Paul Wieland
    Hello - I'm using OSX Catalina and having a problem installing av.
    Any insight?

    Hi all. I am recording a RTP live stream into disk. The program is working correctly but the start time of the video files does not start in 0. I've checked the input container and effectively the start_time attribute is not set to 0. I've tried to force the start_time of the input container to 0 but it is protected.

    On the other hand, I've found that if those segments with the start_time not set to 0 are muxed using ffmpeg cli with the flag '-c copy' the offset is correctly removed. So, I've been playing with the audio and video packet pts and dts to remove the initial offset but after some segments the program crashes because problems in the decoding process...

    How do you use to restart or set the start_time to 0?

    @PaulWieland You're using Python 2 there; you need Python 3 instead. If you have it installed, it should be set up so you can run 'python3' instead of 'python' and 'pip3' instead of 'pip'. If not, get it from https://www.python.org/downloads/
    Also, have a look at how to create a venv: https://docs.python.org/3/tutorial/venv.html
    Once you have a virtual environment (that has been made with the correct Python installation) active, the plain 'python' and 'pip' commands will refer to the correct programs within that terminal session (and not to the deprecated legacy version that comes with macOS for backwards compatibility).
    Peter Hausamann
    Hi, I have a camera that produces frames in the Bayer RGGB format which I would like to encode with PyAV. With plain ffmpeg this works fine with the -pix_fmt bayer_rggb8 flag, however it seems like this is not supported by PyAV:
    av.VideoFrame.from_ndarray(img, format="bayer_rggb8")
    ValueError: Conversion from numpy array with format `bayer_rggb8` is not yet supported
    Samuel Smith

    So I'm recording an online radio stream and dumping it to a file. So the file is raw AAC audio data. From there, I put the raw data into a container with:

    in_container = av.open('raw_stream.aac', mode='r')
    in_stream = in_container.streams.audio[0]
     out_container = av.open('stream.m4a', mode='w')
      out_stream = out_container.add_stream(template=in_stream)
       for packet in in_container.demux(in_stream):
            # skip flush packet
            if packet.dts is None:
            # We need to assign the packet to the new stream.
            packet.stream = out_stream

    But my question is, could I perhaps bypass having to write the file and instead directly insert the raw audio data into the output container incrementally?

    Samuel Smith
    Well, I guess I could just pass the http stream straight into av.open.. but what fun is that?
    Hi all. Im using PyAV to re-mux on air masters from a client (HBO) - Im re-muxing ProRes 422HQ, and masters with up to 16 discrete, mono 48Khz LPCM tracks with no viable audio layout / channel info (dont ask, its just how it is) - Now I am able to parse, re-mux and do that just fine, however audio channels after the 1st appear to be disabled, and it appears to be a default behavior in FFMPEG (see https://trac.ffmpeg.org/ticket/2626 & https://trac.ffmpeg.org/ticket/3622 ) - now usually doing AV dev in AVFoundation / CoreMedia / CoreVideo CoreAudio on macOS - and enabling tracks is trivial as is making track group associations). My Question: is there a way in PyAV to change this behavior so discrete audio tracks are enabled by default? Is there a way to associate audio tracks to belong to an audio group so they belong to the same language?
    Thank you

    You can see the tracks here https://gist.github.com/vade/139c60b7ba57485a2c0c93f17095bc40

    Note that audio track 1 - default is YES, on all others its NO. This was produced by PyAV with FFMPEG

    ffmpeg version 4.3-2~18.04.york0 Copyright (c) 2000-2020 the FFmpeg developers

    Ah - I think FFMPEG calls this 'disposition' : https://ffmpeg.org/ffmpeg-all.html#Main-options
    Evan Clark
    Does anyone have srt working with pyav, get a protocol not supported with the latest pyav?
    Ramzan Bekbulatov
    Hello! How can I split audio to chunks of fixed length using PyAV?
    Anyone has ffmpeg animation code example of char by char animation?
    Thank you in advance for help...