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• ts video via UDP from ffmpeg to gstreamer [closed]

  23 janvier 2024, par aron.h

  Hardware : Jetson AGX ORIN
Software : Jetpack 5.0.2

  


  I have been attempting to send a video file locally via UDP using ffmpeg :

  


  ffmpeg -stream_loop -1 -re -i test.ts -map 0 -c copy -preset ultrafast -f mpegts "udp://127.0.0.1:5000"

  


  And receiving the same stream via UDP using gstreamer :

  


  gst-launch-1.0 udpsrc port=5000 ! application/x-rtp, media=video, clock-rate=90000, encoding-name=H264 ! rtph264depay ! decodebin ! videoconvert ! aasink

  


  But I get an error on the receiving gstreamer end :

  


  /GstPipeline:pipeline0/GstRtpH264Depay:rtph264depay0: Could not decode stream.
Additional debug info:
gstrtpbasedepayload.c(505): gst_rtp_base_depayload_handle_buffer (): 
/GstPipeline:pipeline0/GstRtpH264Depay:rtph264depay0:
Received invalid RTP payload, dropping
ERROR: from element /GstPipeline:pipeline0/GstRtpH264Depay:rtph264depay0:
The stream is in the wrong format.
Additional debug info:
gstrtph264depay.c(1298): gst_rtp_h264_depay_process ():
/GstPipeline:pipeline0/GstRtpH264Depay:rtph264depay0:
NAL unit type 27 not supported yet


  


  More detailed information on the video file :

  


  Original ID: 1002
Codec: H264 - MPEG-4 AVC (part 10) (h264)
Type: Video
Video resolution: 1920x1080
Buffer dimensions: 1920x1088
Frame rate: 30
Decoded format: 
Orientation: Top left
Chroma location: left


  


  When I listen with the command gst-launch-1.0 -v udpsrc port=5000 ! fakesink dump=1, it is quite apparent that the packets from FFMPEG are being received.
I am not sure why gstreamer's rtph264depay says the stream is in the wrong format.

  


  Would I have to check some details on the FFMPEG side ?
This is what information FFMPEG shows by default while running.

  


  Input #0, mpegts, from 'test.ts':
  Duration: 00:00:57.36, start: 20902.827056, bitrate: 2504 kb/s
  Program 1
    Stream #0:0[0x3ea]: Video: h264 (Main) ([27][0][0][0] / 0x001B), yuv420p(progressive), 1920x1080 [SAR 1:1 DAR 16:9], 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc
Output #0, mpegts, to 'udp://127.0.0.1:5000':
  Metadata:
    encoder         : Lavf58.29.100
    Stream #0:0: Video: h264 (Main) ([27][0][0][0] / 0x001B), yuv420p(progressive), 1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 29.97 fps, 29.97 tbr, 90k tbn, 90k tbc
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
Press [q] to stop, [?] for help
frame=  611 fps= 30 q=-1.0 Lsize=    5847kB time=00:00:20.62 bitrate=2323.0kbits/s speed=   1x
video:5350kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 9.301388%


  


  Any advice would be appreciated.

  


• FFmpeg - ducking music for voiceover (sidechain)

  7 décembre 2023, par kaushal

  I am trying to duck the background music from a voiceover input file using FFmpeg (I tried doing this using pydub with no success).

  


  The below command is predominantly working, except that it is truncating my background music file (main.mp3, over 3 min long) to the length of the voiceover file (taekwondo.mp3, about a min long). Any idea or suggestion on what can I change in below command so that the output file is full 3 min long, with background music ducked when the voiceover is playing. Once the voiceover completes, background music continues to play as normal.

  


  In below complex filter, this is what I'm doing :
adelay filter : to start voiceover playing after 5 seconds into background music
volume filter : Since my background music is too loud and voiceover is too low, I'm increasing voiceover volume by 1.5 times, and lowering background music volume to 0.7 times
sidechaincompress filter : Then I'm applying sidechain filter to duck background music whenever voiceover is playing, with usual threshold, ratio & release filters

  


  I've tried both amix & amerge, but results in same behaviour.

  


  ffmpeg -i main.mp3 -i taekwondo.mp3 -filter_complex "[1:a]adelay=5000|5000,volume=1.5[a];[0:a]volume=0.7[b];[a]asplit=2[sc][mix];[b][sc]sidechaincompress=threshold=0.05:ratio=20:level_sc=1:release=500:attack=1[compr];[compr][mix]amix" output.mp3


  


• ffmpeg avformat_open_input() function causes memory leak when receiving live stream

  12 septembre 2023, par george_d

  I have live streams (can be UDP or HLS, video codec is H264), from which I grab frames for further processing.

  


  For this purpose, I use ffmpeg + nvjpeg + cuda libraries.

  


  However I noticed memory leak - memory usage periodically (every 10-20 seconds) is increased by 100-400 KB, the amount and period may vary.

  


  After disabling pieces of code one by one, I realized that it is avformat_open_input() which causes memory leak.

  


  No matter which buffer settings (https://ffmpeg.org/ffmpeg-protocols.html#udp) I choose for UDP, the leak still persists. Same goes for HLS streams.

  


  I tried to find anything related to this problem, but all the sources I found claimed that this problem took place in the past and has been fixed.

  


  Is there some mysterious setting I am missing, so that memory could be freed properly ?

  


  Or is this memory supposed to be freed when processing frames (i.e. using av_read_frame() and av_packet_unref(), etc) ?

  


  Minimal example of code to reproduce the problem :

  


  avformat_example.cpp

  


  #include &#xA;extern "C" {&#xA;    #include <libavformat></libavformat>avformat.h>&#xA;    #include <libavcodec></libavcodec>avcodec.h>&#xA;}&#xA;&#xA;int main(int argc, char *argv[]){&#xA;    if (argc &lt; 2) {&#xA;      return 1;&#xA;    }&#xA;&#xA;    char* inputSource = argv[1];&#xA;    AVFormatContext *ctx = NULL;&#xA;&#xA;    if (avformat_open_input(&amp;ctx, inputSource, NULL, NULL) != 0) {&#xA;        av_log(NULL,&#xA;               AV_LOG_ERROR,&#xA;               "Cannot open &#x27;%s&#x27;",&#xA;               inputSource);&#xA;        return 1;&#xA;    }&#xA;&#xA;    /*&#xA;    This loop is placed here to demonstrate&#xA;    avformat_open_input() causing leak.&#xA;    Actually, instead of noop loop there is logic of getting and processing frames,&#xA;    but it doesn&#x27;t matter now.&#xA;    As loop goes on, the amount of leaked memory increases.&#xA;    */&#xA;    while(true) {&#xA;      sleep(1);&#xA;    }&#xA;&#xA;    return 0;&#xA;}&#xA;

  


  Compile with :

  


  g++ avformat_example.cpp -lavcodec -lavutil -lavformat -I/usr/include/ffmpeg-cuda -o avformat_open_input_example


  


  Run :

  


  ./avformat_open_input_example "udp://127.0.0.1:5000?reuse=1&pkt_size=1316&buffer_size=1310720&fifo_size=40000"


  


  Version of ffmpeg underlying libraries :

  


  libavutil      58.  7.100 / 58.  7.100
libavcodec     60. 11.100 / 60. 11.100
libavformat    60.  5.100 / 60.  5.100
libavdevice    60.  2.100 / 60.  2.100
libavfilter     9.  8.100 /  9.  8.100
libswscale      7.  2.100 /  7.  2.100
libswresample   4. 11.100 /  4. 11.100