Recherche avancée

Sur d’autres sites (5324)

• How to use scale filter in doc/examples/transcoding.c

  19 janvier 2018, par siods333333

  I tried to replace the "null" filter with the "scale=320x180" filter in the doc/examples/transcoding.c file, it only resulted into the this error message :

  [libx264 @ 03303ee0] Input picture width (640) is greater than stride (256)
  
  Error occurred: Generic error in an external library

  What is wrong ? Look that init_filters happens after open_output_file, the encoder is already set before it even knows the resolution of the output.

  How to do this properly ?

  Look at this piece of code, I don’t get what it’s talking about, fiters ain’t going to magically set the correct resolution :

          /* In this example, we transcode to same properties (picture size,
  
           * sample rate etc.). These properties can be changed for output
  
           * streams easily using filters */
  
          if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO) {
  
              enc_ctx->height = dec_ctx->height;
  
              enc_ctx->width = dec_ctx->width;

  Found this in the ffmpeg.c itself :

      enc_ctx->width  = av_buffersink_get_w(ost->filter->filter);
  
      enc_ctx->height = av_buffersink_get_h(ost->filter->filter);

• FFmpeg -> JSMpeg Websocket Closes Repeatedly

  13 mars 2018, par Kyle Martin

  I’m trying to create a fairly simple streaming server/site. Here’s the current flow :

  • OBS streams to an RTMP URL
  • Nginx accepts the RTMP stream and uses exec-push to have FFmpeg pick up the stream and transcode it
  • FFmpeg transcodes the stream and outputs it to a JSMpeg application, which displays the stream on a webpage.

  When I have my exec_push statement as follows, everything seems to work perfectly, except the browser says Possible garbage data. Skipping. on every frame it receives :

  exec_push /usr/bin/ffmpeg -re -i rtmp://127.0.0.1:1935/$app/$name -f mpeg1video  http://localhost:8080/supersecret;

  This behavior is understandable, because JSMpeg must receive MPEG-TS data, not MPEG1 data. It sees the MPEG1 frames and thinks they’re garbage.

  So through some online research, I found this :

  exec_push /usr/bin/ffmpeg -re -i rtmp://127.0.0.1:1935/$app/$name -c:v copy -c:a copy -f mpegts http://localhost:8080/supersecret;

  Supposedly, this is supposed to transcode my RTMP stream into an MPEG-TS format, which should be compatible with JSMpeg.

  However, with the second version of the command, my FFmpeg -> JSMpeg stream keeps connecting and disconnecting, connecting and disconnecting, and so on. This behavior is observed in terminal :

  Stream Connected: ::1:40208
  
  close
  
  Stream Connected: ::1:40212
  
  close
  
  Stream Connected: ::1:40216
  
  close
  
  Stream Connected: ::1:40220
  
  close
  
  Stream Connected: ::1:40224
  
  close
  
  ...

  What would cause this ? I am pretty certain the issue is in my exec_push command. OBS is perfectly content, which tells me that the stream is making it to the server, and if I do a push, I can do a test push to Ustream just fine, which tells me that Nginx is at least processing the stream with some reasonable degree of success.

  ---

  Disclaimer : I have no idea what I’m talking about. Everything I know about FFmpeg and JSMpeg/Node is from snippets of code that I found online.

• convert G.723.1 to normal wav and split channels ?

  10 août 2019, par user9316498

  I have a wav file who’s info is

  ion@aurora:~/Inbound$ mediainfo 48401-3405-48403--18042018170000.wav 
  
  General
  
  Complete name                            : 48401-3405-48403--18042018170000.wav
  
  Format                                   : Wave
  
  File size                                : 327 KiB
  
  Duration                                 : 4mn 11s
  
  Overall bit rate                         : 10.7 Kbps
  
  
  
  Audio
  
  Format                                   : G.723.1
  
  Codec ID                                 : A100
  
  Duration                                 : 4mn 11s
  
  Bit rate                                 : 10.7 Kbps
  
  Channel(s)                               : 2 channels
  
  Sampling rate                            : 8 000 Hz
  
  Stream size                              : 327 KiB (100%)

  and the audacity shows like this
  

  To tackle it I tried

  ffmpeg -i 48401-3405-48403--18042018170000.wav  -f wav test.wav

  But it just appends the second channel at the back of first channel , So I get only single channel.
  and its media info is :
  ion@aurora : /Inbound$ mediainfo test.wav

  General
  
  Complete name                            : test.wav
  
  Format                                   : Wave
  
  File size                                : 7.67 MiB
  
  Duration                                 : 8mn 22s
  
  Overall bit rate mode                    : Constant
  
  Overall bit rate                         : 128 Kbps
  
  Writing application                      : Lavf56.40.101
  
  
  
  Audio
  
  Format                                   : PCM
  
  Format settings, Endianness              : Little
  
  Format settings, Sign                    : Signed
  
  Codec ID                                 : 1
  
  Duration                                 : 8mn 22s
  
  Bit rate mode                            : Constant
  
  Bit rate                                 : 128 Kbps
  
  Channel(s)                               : 1 channel
  
  Sampling rate                            : 8 000 Hz
  
  Bit depth                                : 16 bits
  
  Stream size                              : 7.67 MiB (100%)

  Any idea how to solve this ?

  PS : When I hear the original wav file, It feels like the talking is too fast.

  This is how the media info of processed file should look like.

  ion@aurora:~/Downloads/Call Recordings$ mediainfo 9417648939\ \(G\ Call\)_Done.wav 
  
  General
  
  Complete name                            : 9417648939 (G Call)_Done.wav
  
  Format                                   : Wave
  
  File size                                : 8.01 MiB
  
  Duration                                 : 8mn 44s
  
  Overall bit rate mode                    : Constant
  
  Overall bit rate                         : 128 Kbps
  
  
  
  Audio
  
  Format                                   : ADPCM
  
  Format profile                           : U-Law
  
  Codec ID                                 : 7
  
  Codec ID/Hint                            : CCITT
  
  Duration                                 : 8mn 44s
  
  Bit rate mode                            : Constant
  
  Bit rate                                 : 128 Kbps
  
  Channel(s)                               : 2 channels
  
  Sampling rate                            : 8 000 Hz
  
  Bit depth                                : 8 bits
  
  Stream size                              : 8.01 MiB (100%)