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• Converting a rtsp stream to hls but got error "Non-monotonous DTS in output stream 0:0"

  8 mai 2023, par Devi

  I'm converting a rtsp live camera stream to hls using ffmpeg, it is all good the first few hours, each segment file sizes around 1.1M, which match the setting of segment_time of 5 seconds. However, when I check back a few days later, the size of each segment file became 20G, and I found that some error are logged.

  


  My current implementation is like this :

  


  ffmpeg -fflags nobuffer \
           -rtsp_transport tcp \
           -i rtsp://abc.com/live.sdp \
           -fps_mode 0 \
           -copyts \
           -c copy \
           -movflags frag_keyframe+empty_moov \
           -hls_flags delete_segments+append_list \
           -f segment \
           -segment_list_flags live \
           -segment_time 5 \
           -segment_list_size 3 \
           -segment_format mpegts \
           -segment_list /app/mount/test/live.m3u8 \
           -segment_list_type m3u8 \
           -segment_wrap 10 \
           /app/mount/test/%04d.ts \
           2>&1 


  


  Here's the logged error :

  


  [segment @ 0x7fe5123101c0] Non-monotonous DTS in output stream 0:0; previous: 14179898636, current: 11332822107; changing to 14179898637. This may result in incorrect timestamps in the output file.
[segment @ 0x7fe5123101c0] stream:0 start_pts_time:157550 pts:14179898637 pts_time:157554 dts:14179898637 dts_time:157554 -> pts:14179898637 pts_time:157554 dts:14179898637 dts_time:157554
[NULL @ 0x7fe510744880] unknown SEI type 229
[segment @ 0x7fe5123101c0] Non-monotonous DTS in output stream 0:0; previous: 14179898637, current: 11332823907; changing to 14179898638. This may result in incorrect timestamps in the output file.
[segment @ 0x7fe5123101c0] stream:0 start_pts_time:157550 pts:14179898638 pts_time:157554 dts:14179898638 dts_time:157554 -> pts:14179898638 pts_time:157554 dts:14179898638 dts_time:157554
[segment @ 0x7fe5123101c0] Non-monotonous DTS in output stream 0:0; previous: 14179898638, current: 11332825707; changing to 14179898639. This may result in incorrect timestamps in the output file.
[segment @ 0x7fe5123101c0] stream:0 start_pts_time:157550 pts:14179898639 pts_time:157554 dts:14179898639 dts_time:157554 -> pts:14179898639 pts_time:157554 dts:14179898639 dts_time:157554


  


  I've tried :

  


  

  1. Changing the option -fps_mode with passthrough, vfr, drop.

  


  2. Changing the option -c with -c copy, -c:a copy -c:v libx264.

  


  


  None of the above works for me.

  


• I am phasing a issue of " ffmpeg attribute 'error'" in my project which is a voice command assistant using GPT 3.5 Turbo and BING API [closed]

  2 mai 2023, par Rudraksh Singh Bhadauria

  I ma working on a pyhton project as want to make a program which can work as fluent as GPT and can also adapt latest news through BING API but after trying a lot I am stuck with a issue where "Error transcribing audio : module 'ffmpeg' has no attribute 'Error'" I have tried installing these ffmpeg files again and again in the way already mentioned in stack overflow but for me its not working.

  


  I have tried using :
installing ffmpeg-pyhton in my terminal and also ffmpeg module and also I have reinstalled the whole files and also have put them in my environment variables.
If anyone could help me to sort this problem out then it will be great help.

  


• "Error : more samples than frame size" while encoding audio to opus codec using FFMPEG

  28 avril 2023, par lokit khemka

  I am converting audio from codec AAC to Opus using libavcodec library of FFMPEG. The input codec details are as follows : Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, 6 channels, fltp, 391 kb/s (default)

  


  The codec options that I have used for the output encoding are as follows :

  


      int OUTPUT_CHANNELS = 2;
    int OUTPUT_BIT_RATE = 32000;
int sample_rate = 48000;
    encoder_sc->audio_avcc->channels = OUTPUT_CHANNELS;
    encoder_sc->audio_avcc->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS);
    encoder_sc->audio_avcc->sample_rate = sample_rate;
    encoder_sc->audio_avcc->sample_fmt = encoder_sc->audio_avc->sample_fmts[0];
    encoder_sc->audio_avcc->bit_rate = OUTPUT_BIT_RATE;
    encoder_sc->audio_avcc->time_base = (AVRational){1, sample_rate};


  


  I am using the code in the file as it is, with minimal changes : https://github.com/leandromoreira/ffmpeg-libav-tutorial/blob/master/3_transcoding.c for reference. Look for the function prepare_audio_encoder in the file.

  


  When the run the program, I keep getting the error : " more samples than frame size". I don't know much about Audio Processing, so I cannot debug this error. Any help is greatly appreciated.