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• FFmpeg to connect to RTSP in Python, but sometimes the video stream freezes, and FFmpeg throws an error

  23 décembre 2024, par Jin Hc
  

  • Here is part of my code. I use the ffmpeg-python library in Python to receive the RTSP stream and perform subsequent processing.

  


  


      def __init__(self, parent: QWidget = None):
        super(captureVideo_Thread, self).__init__()
        self.ffmpeg_args = {
            "rtsp_transport": "tcp",
            "fflags": "+genpts+discardcorrupt+autobsf+fastseek",
            "flags": "low_delay",
            "err_detect": "ignore_err",
            "reorder_queue_size": 10000,
            "vsync": 2,
            "probesize": 10 * 1000000,
            "analyzeduration": 10 * 1000000,
            "threads": "auto",
            "correct_ts_overflow": "1",
        }

        self.parent: Video_Widgets = parent
        self.running_FLAG = True
        self.mutex = QMutex()
        self.process = None

    def run(self) -> None:
        try:
            probe = func_timeout(2, ffmpeg.probe, (self.parent.video_source,))
        except FunctionTimedOut:
            self.captureVideoEnd_signal.emit()
            return
        except ffmpeg.Error as e:
            self.captureVideoEnd_signal.emit()
            return
        cap_info = next(x for x in probe["streams"] if x["codec_type"] == "video")
        video_width = cap_info["width"]
        video_height = cap_info["height"]
        config.video_width = video_width
        config.video_height = video_height
        up, down = str(cap_info["r_frame_rate"]).split("/")
        config.video_fps = round(eval(up) / eval(down), 2)
        self.process = (
            ffmpeg.input(self.parent.video_source, **self.ffmpeg_args)
            .output(
                "pipe:",
                format="rawvideo",
                pix_fmt="rgb24",
            )
            .overwrite_output()
            .run_async(pipe_stdout=True)
        )

        while True:
            if not config.is_openVideo:
                config.is_openVideo = True
            with config.locked(self.mutex):
                if not self.running_FLAG:
                    break

            in_bytes = self.process.stdout.read(video_width * video_height * 3)
            expected_size = video_width * video_height * 3
            if not in_bytes or len(in_bytes) != expected_size:
                continue

            in_frame = np.frombuffer(in_bytes, np.uint8).reshape([video_height, video_width, 3])


  


  

  • There are two phenomena that occur, and these do not happen at fixed times. The first phenomenon is that FFmpeg throws the following error, but the video continues playing.

  


  


  # client
[vist#0:0/h264 @ 000002419fad8e40] [dec:h264 @ 00000241a01046c0] corrupt decoded frame
frame=87565 fps= 50 q=-0.0 size=336245760KiB time=00:29:11.20 bitrate=1572927.9kbits/s speed=   1x  
[h264 @ 00000241a0104a00] Invalid level prefix
[h264 @ 00000241a0104a00] error while decoding MB 25 63
[vist#0:0/h264 @ 000002419fad8e40] [dec:h264 @ 00000241a01046c0] corrupt decoded frame
frame=87593 fps= 50 q=-0.0 size=336353280KiB time=00:29:11.75 bitrate=1572938.5kbits/s speed=   1x  
[h264 @ 00000241a0104a00] Invalid level prefix
[h264 @ 00000241a0104a00] error while decoding MB 24 63
[vist#0:0/h264 @ 000002419fad8e40] [dec:h264 @ 00000241a01046c0] corrupt decoded frame


  


  # server
No errors.


  


  

  • The second phenomenon is that the receiver on my side will freeze completely, and the speed displayed by FFmpeg will gradually decrease (the trigger time for this behavior is random, sometimes after 5 minutes, sometimes after 20 minutes). Additionally, the sender will display a large number of errors.

  


  


  # client
frame=21131 fps= 48 q=-0.0 size=81139200KiB time=00:07:10.97 bitrate=1542290.7kbits/s speed=0.984x


  


  # server
[WARN  rtsp_lib/rtsp_demo.c:1966:rtsp_sever_tx_video] client 192.168.0.102 will lost video packet
[WARN  rtsp_lib/rtsp_demo.c:1966:rtsp_sever_tx_video] client 192.168.0.102 will lost video packet
[WARN  rtsp_lib/rtsp_demo.c:1966:rtsp_sever_tx_video] client 192.168.0.102 will lost video packet
[WARN  rtsp_lib/rtsp_demo.c:1966:rtsp_sever_tx_video] client 192.168.0.102 will lost video packet
[WARN  rtsp_lib/rtsp_demo.c:1966:rtsp_sever_tx_video] client 192.168.0.102 will lost video packet
[DEBUG rtsp_lib/rtsp_msg.c:865:rtsp_msg_parse_from_array]
RTSP/1.0 501 Not Implemented


  


  


  • I am sure the issue is not with the stream source, because there are no problems when using VLC or other software.

    


  


  • I think the issue might be with my FFmpeg settings.???

    


  


  • I found that the average bitrate of VLC is 19000kb/s, but the FFmpeg in my code shows 1500000kb/s. Why is this happening ? Is there something wrong with my settings ?

    


  


  


• Stream specifier ':a' in filtergraph description [0:v][0:a][1:v][1:a]concat=a=1:n=2:v=1[s0][s1] matches no streams

  6 septembre 2022, par Mash

  I am using ffmpeg-python and I am trying to make a code that gets all the clips from a folder, converts all of them to the same format then merge all of them with their audio to one final video file.

  


  Here is how the python script looks like

  


  import os
import ffmpeg

dir = "./clips/"

# Reshape video
for f in os.listdir(dir):
  (
    ffmpeg
    .input(os.path.join(dir, f))
    .filter("scale", size="hd1080", force_original_aspect_ratio="increase")
    .filter('fps', fps=60, round='up')
    .output(str(os.path.join(dir, f))[:-4] + "_reshaped.mp4")
    .run()
  )

# Select reshaped video
files = []

for f in os.listdir(dir):
  if "_reshaped.mp4" in str(os.path.join(dir, f)):
    files.append(ffmpeg.input(str(os.path.join(dir, f))))

# eparate video and audio, then flat the array
video_and_audios_files = [item for sublist in map(lambda f: [f.video, f.audio], files) for item in sublist]

# concat all
joined = (
    ffmpeg
    .concat(*video_and_audios_files, v=1, a=1)
    .node
)

# merge video and audio
(
    ffmpeg
    .output(joined[0], joined[1], "./out.mp4")
    .run()
)



  


  Now here is my output, this is the output of the part of the code that merges every clip into one (aka the merge video and audio section of the code)

  


  ffmpeg version 5.1.1-essentials_build-www.gyan.dev Copyright (c) 2000-2022 the FFmpeg developers&#xA;  built with gcc 12.1.0 (Rev2, Built by MSYS2 project)&#xA;  configuration: --enable-gpl --enable-version3 --enable-static --disable-w32threads --disable-autodetect --enable-fontconfig --enable-iconv --enable-gnutls --enable-libxml2 --enable-gmp --enable-lzma --enable-zlib --enable-libsrt --enable-libssh --enable-libzmq --enable-avisynth --enable-sdl2 --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxvid --enable-libaom --enable-libopenjpeg --enable-libvpx --enable-libass --enable-libfreetype --enable-libfribidi --enable-libvidstab --enable-libvmaf --enable-libzimg --enable-amf &#xA;--enable-cuda-llvm --enable-cuvid --enable-ffnvcodec --enable-nvdec --enable-nvenc --enable-d3d11va --enable-dxva2 --enable-libmfx --enable-libgme --enable-libopenmpt --enable-libopencore-amrwb --enable-libmp3lame --enable-libtheora --enable-libvo-amrwbenc --enable-libgsm --enable-libopencore-amrnb --enable-libopus --enable-libspeex --enable-libvorbis --enable-librubberband&#xA;  libavutil      57. 28.100 / 57. 28.100&#xA;  libavcodec     59. 37.100 / 59. 37.100&#xA;  libavformat    59. 27.100 / 59. 27.100&#xA;  libavdevice    59.  7.100 / 59.  7.100&#xA;  libavfilter     8. 44.100 /  8. 44.100&#xA;  libswscale      6.  7.100 /  6.  7.100&#xA;  libswresample   4.  7.100 /  4.  7.100&#xA;  libpostproc    56.  6.100 / 56.  6.100&#xA;Input #0, mov,mp4,m4a,3gp,3g2,mj2, from &#x27;./clips/clip1_reshaped.mp4&#x27;:&#xA;  Metadata:&#xA;    major_brand     : isom&#xA;    minor_version   : 512&#xA;    compatible_brands: isomiso2avc1mp41&#xA;    encoder         : Lavf59.27.100&#xA;  Duration: 00:00:26.02, start: 0.000000, bitrate: 7432 kb/s&#xA;  Stream #0:0[0x1](und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, bt709, progressive), 1920x1080 [SAR 1:1 DAR 16:9], 7425 kb/s, 60 fps, 60 tbr, 15360 tbn (default)      &#xA;    Metadata:&#xA;      handler_name    : VideoHandler&#xA;      vendor_id       : [0][0][0][0]&#xA;      encoder         : Lavc59.37.100 libx264&#xA;Input #1, mov,mp4,m4a,3gp,3g2,mj2, from &#x27;./clips/clip2_reshaped.mp4&#x27;:&#xA;  Metadata:&#xA;    major_brand     : isom&#xA;    minor_version   : 512&#xA;    compatible_brands: isomiso2avc1mp41&#xA;    encoder         : Lavf59.27.100&#xA;  Duration: 00:00:05.00, start: 0.000000, bitrate: 11445 kb/s&#xA;  Stream #1:0[0x1](und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, unknown/bt709/unknown, progressive), 1920x1080, 11438 kb/s, 60 fps, 60 tbr, 15360 tbn (default)        &#xA;    Metadata:&#xA;      handler_name    : VideoHandler&#xA;      vendor_id       : [0][0][0][0]&#xA;      encoder         : Lavc59.37.100 libx264&#xA;Stream specifier &#x27;:a&#x27; in filtergraph description [0:v][0:a][1:v][1:a]concat=a=1:n=2:v=1[s0][s1] matches no streams.&#xA;Traceback (most recent call last):&#xA;  File "c:\Users\USERNAME\Desktop\auto_poster\concat_vids.py", line 38, in <module>&#xA;    .run()&#xA;  File "C:\Users\USERNAME\AppData\Local\Programs\Python\Python310\lib\site-packages\ffmpeg\_run.py", line 325, in run&#xA;    raise Error(&#x27;ffmpeg&#x27;, out, err)&#xA;ffmpeg._run.Error: ffmpeg error (see stderr output for detail)&#xA;</module>

  


  Can I get some help please ? I am on Windows 10

  


• ffmpeg - Build HLS stream with all tracks of the input source

  25 juillet 2021, par Kaydee Dunlop

  I would like to create HLS streams for videos. I'm working since more than 2 days now to find a solution that might work but no luck till now to get all tracks of the input inside my HLS output.

  


  What I want in the end is a small script that can merge all tracks form a mp4 imput together and create a proper master.m3u8 playlist referencing all other m3u8 playlists of my m4s, webvtt and audio snippets. Currently I don't have adaptive streaming implemented. To save bandwidth I only want to have one video stream output at a max resolution of 1080p @ 3 Mbit/s.

  


  This is my current state of work, please feel free to to use it :

  


  #!/usr/bin/env bash
LC_NUMERIC="en_US.UTF-8"
START_TIME=$SECONDS
set -e

echo "-----START GENERATING HLS STREAM-----"
# Usage create-vod-hls.sh SOURCE_FILE [OUTPUT_NAME]
[[ ! "${1}" ]] && echo "Usage: create-vod-hls.sh SOURCE_FILE [OUTPUT_NAME]" && exit 1

# comment/add lines here to control which renditions will be created
renditions=(
# resolution  bitrate  audio-rate
  "1920x1080  3000k    128k"
)

segment_target_duration=20      # try to create a new segment every 10 seconds
max_bitrate_ratio=1.07          # maximum accepted bitrate fluctuations
rate_monitor_buffer_ratio=1.5   # maximum buffer size between bitrate conformance checks

#########################################################################

source="${1}"
target="${2}"
if [[ ! "${target}" ]]; then
  target="${source##*/}" # leave only last component of path
  target="${target%.*}"  # strip extension
fi
mkdir -p ${target}

# ----CUSTOM----
sourceResolution="$(ffprobe -v error -select_streams v:0 -show_entries stream=width,height -of csv=s=x:p=0 ${source})"
# echo ${sourceResolution}
arrIN=(${sourceResolution//x/ })
sourceWidth="${arrIN[0]}"
sourceHeight="${arrIN[1]}"

echo ${sourceWidth}
echo ${sourceHeight}

sourceAudioBitRate="$(ffprobe -v error -select_streams a:0 -show_entries stream=bit_rate -of csv=s=x:p=0 ${source})"
sourceAudioBitRateFormatted=$((sourceAudioBitRate / 1000))
# ----END CUSTOM----

key_frames_interval="$(echo `ffprobe ${source} 2>&1 | grep -oE '[[:digit:]]+(.[[:digit:]]+)? fps' | grep -oE '[[:digit:]]+(.[[:digit:]]+)?'`*2 | bc || echo '')"
key_frames_interval=${key_frames_interval:-50}
key_frames_interval=$(echo `printf "%.1f\n" $(bc -l <<<"$key_frames_interval/10")`*10 | bc) # round
key_frames_interval=${key_frames_interval%.*} # truncate to integer

# static parameters that are similar for all renditions
static_params="-c:s webvtt -c:a aac -ar 48000 -c:v copy -sc_threshold 0"
static_params+=" -g ${key_frames_interval} -keyint_min ${key_frames_interval} -hls_time ${segment_target_duration}"
static_params+=" -hls_playlist_type vod -hls_segment_type fmp4"
static_params+=" -var_stream_map v:0,a:0,s:0"

# misc params
misc_params="-hide_banner -y"

master_playlist="#EXTM3U
#EXT-X-VERSION:3
"
cmd=""
resolutionValid=0
prevHeight=0
for rendition in "${renditions[@]}"; do
  # drop extraneous spaces
  rendition="${rendition/[[:space:]]+/ }"

  # rendition fields
  resolution="$(echo ${rendition} | cut -d ' ' -f 1)"
  bitrate="$(echo ${rendition} | cut -d ' ' -f 2)"
  audiorate="$(echo ${rendition} | cut -d ' ' -f 3)"

  audioBitRateFormatted=${audiorate%?} # remove "k" at the last index

  # take highest possible audio bit rate
  if [ $audioBitRateFormatted -gt $sourceAudioBitRateFormatted ]; then
      audiorate=${sourceAudioBitRateFormatted}k
  fi

  # calculated fields
  width="$(echo ${resolution} | grep -oE '^[[:digit:]]+')"
  height="$(echo ${resolution} | grep -oE '[[:digit:]]+$')"
  maxrate="$(echo "`echo ${bitrate} | grep -oE '[[:digit:]]+'`*${max_bitrate_ratio}" | bc)"
  bufsize="$(echo "`echo ${bitrate} | grep -oE '[[:digit:]]+'`*${rate_monitor_buffer_ratio}" | bc)"
  bandwidth="$(echo ${bitrate} | grep -oE '[[:digit:]]+')000"
  name="stream"
  
  widthParam=0
  heightParam=0

  if [ $(((width / sourceWidth) * sourceHeight)) -gt $height ]; then
    widthParam=-2
    heightParam=$height
  else
    widthParam=$width
    heightParam=-2
  fi
  
  #cmd+=" ${static_params} -vf scale=w=${widthParam}:h=${heightParam}"
  cmd+=" -b:v ${bitrate} -maxrate ${maxrate%.*}k -bufsize ${bufsize%.*}k -b:a ${audiorate}"
  cmd+=" -hls_segment_filename ${target}/${name}_%03d.m4s ${target}/${name}.m3u8"
  
  # add rendition entry in the master playlist
  master_playlist+="#EXT-X-STREAM-INF:BANDWIDTH=${bandwidth},RESOLUTION=${resolution}\n${name}.m3u8\n"

  resolutionValid=1
  prevHeight=${height}
done

if [ $resolutionValid -eq 1 ]; then
  # start conversion
  echo -e "Executing command:\nffmpeg ${misc_params} -i ${source} ${cmd}\n"
  ffmpeg ${misc_params} -i ${source} ${cmd}
  # create master playlist file
  echo -e "${master_playlist}" > ${target}/playlist.m3u8
  echo "Done - encoded HLS is at ${target}/"
else
  echo "Video source is too small"
  exit 1
fi

ELAPSED_TIME=$(($SECONDS - $START_TIME))

echo "Elapsed time: ${ELAPSED_TIME}"
echo "-----FINISH GENERATING HLS STREAM-----"