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• Cannot open RTMP stream with FFMPEG Java Cannot assign requested address

  12 novembre 2023, par lastpeony4

  I am trying to open an RTMP stream with avformat_open_input and for some reason i keep getting Cannot assign requested address error.
  
My RTMP URL is correct because i can watch the stream using VLC Player and ffmpeg command line :
  
ffmpeg -i rtmp://1.1.1.1/app/teststream -vcodec copy -acodec copy output.mp4

  


  int timeout = 2500;
setConnectionTimeout(timeout);

AVDictionary optionsDictionary = new AVDictionary();

String timeoutStr = String.valueOf(this.timeoutMicroSeconds);
av_dict_set(optionsDictionary, "timeout", timeoutStr, 0);

int analyzeDurationUs = 1500 * 1000;
String analyzeDuration = String.valueOf(analyzeDurationUs);
av_dict_set(optionsDictionary, "analyzeduration", analyzeDuration, 0);

int ret;

if ((ret = avformat_open_input(inputFormatContext, streamUrl, null, optionsDictionary)) < 0) {
  // ERROR Cannot assign requested address
}


  


  I am using org.bytedeco:ffmpeg:5.1.2-1.5.8 (ffmpeg-5.1.2-1.5.8.jar)

  


  Why i can't open it ?

  


• Aspect ratio problems at transcoding with ffmpeg [closed]

  19 novembre 2023, par udippel

  I have a huge collection of videos from the last 20+ years, videos in all sorts of formats. I use gerbera as open source UPnP-AV media server. Our TV handles only very limited of these formats. Therefore I use the transcoding feature of gerbera (I don't want to convert the 2000+ files ; thereby avoiding loss of multiple audio tracks, (multiple) subtitles, and so forth).

  


  This is my current unified argument line for ffmpeg :
-c:v mpeg2video -maxrate 20000k -vf setdar=16/9 -r 24000/1001 -qscale:v 4 -top 1 -c:a mp2 -f mpeg -y
It works pretty okay, except of the aspect ratios. Well, I don't understand this fully, because ffprobe for File A states :
Stream #0:0: Video: mpeg4 (Simple Profile) (XVID / 0x44495658), yuv420p, 624x464 [SAR 1:1 DAR 39:29], 1500 kb/s, 25 fps, 25 tbr, 25 tbn, 25 tbc
This file displays very well.
File B comes as :
Stream #0:0(eng): Video: h264 (High), yuv420p(tv, bt709, progressive), 960x720, SAR 1:1 DAR 4:3, 23.98 fps, 23.98 tbr, 1k tbn, 180k tbc (default)
This file displays horribly squeezed vertically and doesn't fill the screen left and right neither, with the same settings of the TV. Also, playing this file (and others, naturally) the TV doesn't offer the 14:9 display option, which is available e.g. for the file further up.

  


  Both have same SAR, DAR, almost identical H:V ratios (1.345, 1.333) ; and almost identical DAR.

  


  My questions :

  


  

  1. Why, despite of almost identical pixel ratios, DAR and SAR are these files handled so differently in one and the same session on the same TV (SONY), please ?

  


  2. With which method could I instruct ffmpeg to display the second file properly, too, please ?
(I have already tried 'scale' ; but to no avail. Which could have been foreseen, since the ratios are already very close.)
My guess is, that the (tv, bt709, progressive) mess things up.
(I have already tried to add the yuv420p in the argument line, also to no avail.)

  


  


  Appreciate any help,

  


  Uwe

  


  I have already tried to add a 'scale' option ; but to no avail. Which could have been foreseen, since the ratios are already very close.
I have already tried to add the yuv420p in the argument line, also to no avail.
I have already tried force_original_aspect_ratio, but also here, nothing improving.
Also, I played with -aspect, but the aspects are okay, and would need individual corrections, which I can't and don't to do for 2000+ files. A simple 16:9 doesn't cut it.

  


• ffmpeg mp3 chunk to wav chunk adds gap in the start of the audio

  13 décembre 2023, par 1Mayur

  I have an mp3 streaming from a URL, I save the chunks in 1024 byes buffer size.
After I get all the chunks, I'm using ffmpeg to convert the incoming mp3 chunk (22050 mono) to a wav chunk.

  


  When I open/play the wav chunk I see that there is an empty gap at the start of every chunk.

  


  here is the code I'm running in Python subprocess in a loop for all the saved chunks

  


  subprocess.run(["ffmpeg", "-i",
    f"{Path.cwd()}/input/{path}",
    f"{Path.cwd()}/temp_output/{path.replace('.mp3', '')}.wav"
])


  


  here is the output in the terminal

  


  processing: test-016.mp3
ffmpeg version 6.0 Copyright (c) 2000-2023 the FFmpeg developers
  built with Apple clang version 15.0.0 (clang-1500.0.40.1)
  configuration: --prefix=/usr/local/Cellar/ffmpeg/6.0_1 --enable-shared --enable-pthreads --enable-version3 --cc=clang --host-cflags= --host-ldflags='-Wl,-ld_classic' --enable-ffplay --enable-gnutls --enable-gpl --enable-libaom --enable-libaribb24 --enable-libbluray --enable-libdav1d --enable-libjxl --enable-libmp3lame --enable-libopus --enable-librav1e --enable-librist --enable-librubberband --enable-libsnappy --enable-libsrt --enable-libsvtav1 --enable-libtesseract --enable-libtheora --enable-libvidstab --enable-libvmaf --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libxvid --enable-lzma --enable-libfontconfig --enable-libfreetype --enable-frei0r --enable-libass --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libspeex --enable-libsoxr --enable-libzmq --enable-libzimg --disable-libjack --disable-indev=jack --enable-videotoolbox --enable-audiotoolbox
  libavutil      58.  2.100 / 58.  2.100
  libavcodec     60.  3.100 / 60.  3.100
  libavformat    60.  3.100 / 60.  3.100
  libavdevice    60.  1.100 / 60.  1.100
  libavfilter     9.  3.100 /  9.  3.100
  libswscale      7.  1.100 /  7.  1.100
  libswresample   4. 10.100 /  4. 10.100
  libpostproc    57.  1.100 / 57.  1.100
[mp3 @ 0x7fd48e104480] Format mp3 detected only with low score of 25, misdetection possible!
[mp3 @ 0x7fd48e104480] Skipping 463 bytes of junk at 0.
[mp3 @ 0x7fd48e104480] Estimating duration from bitrate, this may be inaccurate
Input #0, mp3, from '/Users/mayur/Projects/input/test-016.mp3':
  Duration: 00:00:00.39, start: 0.000000, bitrate: 169 kb/s
  Stream #0:0: Audio: mp3, 22050 Hz, mono, fltp, 160 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (mp3 (mp3float) -> pcm_s16le (native))
Press [q] to stop, [?] for help
Output #0, wav, to '/Users/mayur/Projects/temp_output/test-016.wav':
  Metadata:
    ISFT            : Lavf60.3.100
  Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 22050 Hz, mono, s16, 352 kb/s
    Metadata:
      encoder         : Lavc60.3.100 pcm_s16le
size=      17kB time=00:00:00.36 bitrate= 379.7kbits/s speed= 253x    
video:0kB audio:17kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.451389%


  


  I tried the pydub as well and faced similar issue.