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Sur d’autres sites (8444)
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How to convert rtmp hevc video stream to srt av1 endpoint with ffmpeg ?
20 juin 2024, par Lulíki want use ffmpeg to listen rtmp stream and send to srt endpoint.


Flow : smartphone (camera) -> laptop (ffmpeg script) -> desktop (obs studio)


ffmpeg script show warning message and in obs stuido i can see any video only audio.


Thank you in advance.


Console output while running script (error in the end is bcs i stoped sending data from phone) :


ffmpeg version git-2024-06-20-8d6014d Copyright (c) 2000-2024 the FFmpeg developers
 built with gcc 12 (Debian 12.2.0-14)
 configuration: --enable-libsvtav1 --enable-libsrt
 libavutil 59. 24.100 / 59. 24.100
 libavcodec 61. 8.100 / 61. 8.100
 libavformat 61. 3.104 / 61. 3.104
 libavdevice 61. 2.100 / 61. 2.100
 libavfilter 10. 2.102 / 10. 2.102
 libswscale 8. 2.100 / 8. 2.100
 libswresample 5. 2.100 / 5. 2.100
Input #0, flv, from 'rtmp://192.168.0.194/s/streamKey':
 Duration: 00:00:00.00, start: 0.000000, bitrate: N/A
 Stream #0:0: Video: hevc (Main), yuv420p(tv, smpte170m/bt470bg/smpte170m), 1080x1920, 10240 kb/s, 30 fps, 120 tbr, 1k tbn
 Stream #0:1: Audio: aac (LC), 44100 Hz, stereo, fltp, 131 kb/s
Stream mapping:
 Stream #0:0 -> #0:0 (hevc (native) -> av1 (libsvtav1))
 Stream #0:1 -> #0:1 (aac (native) -> mp2 (native))
Press [q] to stop, [?] for help
Svt[info]: -------------------------------------------
Svt[info]: SVT [version]: SVT-AV1 Encoder Lib 595a874
Svt[info]: SVT [build] : GCC 12.2.0 64 bit
Svt[info]: LIB Build date: Jun 20 2024 14:25:08
Svt[info]: -------------------------------------------
Svt[info]: Number of logical cores available: 12
Svt[info]: Number of PPCS 76
Svt[info]: [asm level on system : up to avx2]
Svt[info]: [asm level selected : up to avx2]
Svt[info]: -------------------------------------------
Svt[info]: SVT [config]: main profile tier (auto) level (auto)
Svt[info]: SVT [config]: width / height / fps numerator / fps denominator : 1080 / 1920 / 120 / 1
Svt[info]: SVT [config]: bit-depth / color format : 8 / YUV420
Svt[info]: SVT [config]: preset / tune / pred struct : 10 / PSNR / random access
Svt[info]: SVT [config]: gop size / mini-gop size / key-frame type : 641 / 16 / key frame
Svt[info]: SVT [config]: BRC mode / rate factor : CRF / 35 
Svt[info]: SVT [config]: AQ mode / variance boost : 2 / 0
Svt[info]: -------------------------------------------
Svt[warn]: Failed to set thread priority: Invalid argument
Output #0, mpegts, to 'srt://192.168.0.167:9998?mode=caller':
 Metadata:
 encoder : Lavf61.3.104
 Stream #0:0: Video: av1, yuv420p(tv, smpte170m/bt470bg/smpte170m, progressive), 1080x1920, q=2-31, 120 fps, 90k tbn
 Metadata:
 encoder : Lavc61.8.100 libsvtav1
 Stream #0:1: Audio: mp2, 44100 Hz, stereo, s16, 384 kb/s
 Metadata:
 encoder : Lavc61.8.100 mp2
[mpegts @ 0x55ec921d9540] Stream 0, codec av1, is muxed as a private data stream and may not be recognized upon reading.
[in#0/flv @ 0x55ec9219cc40] Error during demuxing: Input/output error1990.7kbits/s speed=0.967x 
[out#0/mpegts @ 0x55ec922247c0] video:4431KiB audio:1138KiB subtitle:0KiB other streams:0KiB global headers:0KiB muxing overhead: 6.374870%
frame= 723 fps= 31 q=35.0 Lsize= 5923KiB time=00:00:24.12 bitrate=2011.3kbits/s speed=1.04x



I send video stream from mobile app over rtmp encoded with hevc to my laptop where running script
ffmpeg -f flv -listen 1 -i rtmp://192.168.0.194/s/streamKey -c:v libsvtav1 -f mpegts srt://192.168.0.167:9998?mode=caller
. On the desktop i have obs with media source inputsrt://192.168.0.167:9998?mode=listener
.

When i run ffmpeg script without video codec option (-c:v libsvtav1) its working fine and in obs i can see video from my phone camera. With the option i can not see video only audio.
I clearly dont understand warning message :
[mpegts @ 0x55ec921d9540] Stream 0, codec av1, is muxed as a private data stream and may not be recognized upon reading.
.
Do I need specify codec (av1) in obs media source or my ffmpeg script is wrong ?

-
How to get webam frames one by one but also compressed ?
29 mars, par VoracI need to grab frames from the webcam of a laptop, transmit them one by one and the receiving side stitch them into a video. I picked
ffmpeg-python
as wrapper of choice and the example from the docs works right away :

#!/usr/bin/env python

# In this file: reading frames one by one from the webcam.


import ffmpeg

width = 640
height = 480


reader = (
 ffmpeg
 .input('/dev/video0', s='{}x{}'.format(width, height))
 .output('pipe:', format='rawvideo', pix_fmt='yuv420p')
 .run_async(pipe_stdout=True)
)

# This is here only to test the reader.
writer = (
 ffmpeg
 .input('pipe:', format='rawvideo', pix_fmt='yuv420p', s='{}x{}'.format(width, height))
 .output('/tmp/test.mp4', format='h264', pix_fmt='yuv420p')
 .overwrite_output()
 .run_async(pipe_stdin=True)
)


while True:
 chunk = reader.stdout.read(width * height * 1.5) # yuv
 print(len(chunk))
 writer.stdin.write(chunk)



Now for the compression part.


My reading of the docs is that the input to the reader perhaps needs be
rawvideo
but nothing else does. I tried replacingrawvideo
withh264
in my code but that resulted in empty frames. I'm considering a third invocation looking like this but is that really the correct approach ?

encoder = ( 
 ffmpeg 
 .input('pipe:', format='rawvideo', pix_fmt='yuv420p', s='{}x{}'.format(width, height))
 .output('pipe:', format='h264', pix_fmt='yuv420p') 
 .run_async(pipe_stdin=True, pipe_stdout=True) 



-
ffmpeg command in my Python code is not working in Docker
12 juin 2024, par Akhil VargheseI am developing a Python (Django-REST) app for video processing. I need to extract the frames of videos and need to save a live stream(RSTP stream) to the desired directory.
I am using
ffmpeg
for saving the RTSP stream, and I have also Dockerized the app.

- 

- Live Stream save method




import subprocess
@api_view(['POST'])
def saveRtsp(request):
 logger.info("Saving RTSP URL")
 rtsp_url = 'rtsp://192.168.1.117/sample2.mkv'
 rtsp_video_path = 'rtsp.mp4'
 try:
 ffmpeg_command = [
 'ffmpeg',
 '-y',
 "-analyzeduration", "20000000",
 "-probesize", "10000000",
 "-rtsp_transport", "tcp",
 "-r", "30",
 "-i", rtsp_url,
 "-c:v", "copy",
 "-b:v", "1M",
 "-c:a", "copy",
 "-b:a", "128k",
 "-f", "mp4",
 rtsp_video_path
 ]
 
 subprocess.run(ffmpeg_command)
 return True
 except Exception as e:
 logger.error(f"Failed to save RTSP URL: {e}")
 return False



This is actually saving a video file in my directory with properties (as shown by "Get Info" in the MacOS Finder) :




As shown in the image, the video doesn't have the resolution and codec information, so it's not compatible with players.


However, If I execute the following command directly in the Docker container shell, I get a compatible video.


A valid video information :




As shown in the image, a compatible video has the Dimension, codecs, and Duration details in the more info section.


ffmpeg -analyzeduration 20000000 -probesize 10000000 -rtsp_transport tcp -r 30 -i rtsp://xx.xx.xx.xx/sample2.mkv -c:v copy -b:v 1M -c:a copy -b:a 128k -f mp4 testoutput2.mp4



My
Dockerfile
is :

FROM python:3.9-slim

# Set environment variables
ENV PYTHONUNBUFFERED 1

# Set the working directory in the container
WORKDIR /app

# Install system dependencies
RUN apt-get update && apt-get install -y \
 build-essential \
 ffmpeg \
 libsm6 \
 libxext6 \
 libx264-dev \
 libopencv-dev \
 && apt-get clean

# Copy the requirements file into the container
COPY requirements.txt /app/requirements.txt

# Install Python dependencies
RUN pip install --no-cache-dir -r /app/requirements.txt

# Copy the current directory contents into the container at /app
COPY . /app/

# Expose the port the app runs on
EXPOSE 8000

# Run the Django development server
CMD ["python", "manage.py", "runserver", "0.0.0.0:8000"]



Why does the strange behaviour happen ?
How can I fix this issue ?


I tried using different codecs and even the
ffmpeg-python
library,
but still getting the same result.