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• ffmpeg caps audio bitrate to 128kb/s [closed]

  1er mai 2013, par Andrew Burns

  I want to upconvert a file to 256kb/s. (I understand I am losing quality and all
  that, I know that I really want to upconvert a 64kb/s=>256kb/s)

  The command I am using

  ffmpeg -i "Same Love.m4a" -acodec libfaac "Same Love.m4a" -b:a 256kb \
  
    -loglevel debug

  I have used every combination that I can think of and ffmpeg will only upconvert
  to 128kbs !

  Here is a log of my output

  $ ffmpeg -i "Same Love.m4a" -acodec libfaac "ready_to_import_to_itunes/Same Love.m4a" -b:a 256kb -loglevel debug
  ffmpeg version 1.0 Copyright (c) 2000-2012 the FFmpeg developers
    built on Oct 12 2012 12:31:30 with Apple clang version 4.1 (tags/Apple/clang-421.11.66) (based on LLVM 3.1svn)
    configuration : —prefix=/usr/local/Cellar/ffmpeg/1.0 —enable-shared —enable-gpl —enable-version3 —enable-nonfree —enable-hardcoded-tables —cc=cc —host-cflags= —host-ldflags= —enable-libx264 —enable-libfaac —enable-libmp3lame —enable-libxvid
    libavutil      51. 73.101 / 51. 73.101
    libavcodec     54. 59.100 / 54. 59.100
    libavformat    54. 29.104 / 54. 29.104
    libavdevice    54.  2.101 / 54.  2.101
    libavfilter     3. 17.100 /  3. 17.100
    libswscale      2.  1.101 /  2.  1.101
    libswresample   0. 15.100 /  0. 15.100
    libpostproc    52.  0.100 / 52.  0.100
  [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7fc6ab006600] Format mov,mp4,m4a,3gp,3g2,mj2 probed with size=2048 and score=100
  [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7fc6ab006600] ISO : File Type Major Brand : mp42
  [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7fc6ab006600] File position before avformat_find_stream_info() is 30844
  [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7fc6ab006600] All info found
  [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7fc6ab006600] File position after avformat_find_stream_info() is 31216
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'Same Love.m4a' :
    Metadata :
      major_brand : mp42
      minor_version : 0
      compatible_brands : mp42isom3gp63g2a3gp4
      creation_time : 2012-08-11 09:56:26
    Duration : 00:05:18.71, start : 0.000000, bitrate : 64 kb/s
      Stream #0:0(und), 1, 1/44100 : Audio : aac (mp4a / 0x6134706D), 44100 Hz, stereo, s16, 64 kb/s
      Metadata :
        creation_time : 2012-08-11 09:56:26
        handler_name : soun
  [abuffer @ 0x7fc6aac0e560] Setting entry with key 'time_base' to value '1/44100'
  [abuffer @ 0x7fc6aac0e560] Setting entry with key 'sample_rate' to value '44100'
  [abuffer @ 0x7fc6aac0e560] Setting entry with key 'sample_fmt' to value 's16'
  [abuffer @ 0x7fc6aac0e560] Setting entry with key 'channel_layout' to value '0x3'
  [graph 0 input from stream 0:0 @ 0x7fc6aac0e660] tb:1/44100 samplefmt:s16 samplerate:44100 chlayout:0x3
  [aformat @ 0x7fc6aac0eba0] Setting entry with key 'sample_fmts' to value 's16'
  [aformat @ 0x7fc6aac0eba0] Setting entry with key 'channel_layouts' to value '0x4,0x3,0x7,0x107,0x37,0x3f'
  Output #0, ipod, to 'ready_to_import_to_itunes/Same Love.m4a' :
    Metadata :
      major_brand : mp42
      minor_version : 0
      compatible_brands : mp42isom3gp63g2a3gp4
      encoder : Lavf54.29.104
      Stream #0:0(und), 0, 1/44100 : Audio : aac (mp4a / 0x6134706D), 44100 Hz, stereo, s16, 128 kb/s
      Metadata :
        creation_time : 2012-08-11 09:56:26
        handler_name : soun
  Stream mapping :
    Stream #0:0 -> #0:0 (aac -> libfaac)
  Press [q] to stop, [?] for help
  [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7fc6ab006600] demuxer injecting skip 1024
  [aac @ 0x7fc6ab02f600] skip 2048 samples due to side data
  [aac @ 0x7fc6ab02f600] skip whole frame, skip left : 0
  EOF on sink link output stream 0:0:default.28.0kbits/s
  No more output streams to write to, finishing.
  size=    5028kB time=00:05:18.71 bitrate= 129.2kbits/s
  video:0kB audio:4974kB subtitle:0 global headers:0kB muxing overhead 1.093817%
  [AVIOContext @ 0x7fc6aac0ca20] Statistics : 2580581 bytes read, 0 seeks
  

  What I think is happening is that ffmpeg determines that there is really no
  benifit from going to 256k so caps it at 128.

  How do I force it to output the file at the bitrate specified ? (I have tried
  with mp3 too and the samething happens)

• How to write a Live555 FramedSource to allow me to stream H.264 live

  22 juillet 2015, par Garviel

  I’ve been trying to write a class that derives from FramedSource in Live555 that will allow me to stream live data from my D3D9 application to an MP4 or similar.

  What I do each frame is grab the backbuffer into system memory as a texture, then convert it from RGB -> YUV420P, then encode it using x264, then ideally pass the NAL packets on to Live555. I made a class called H264FramedSource that derived from FramedSource basically by copying the DeviceSource file. Instead of the input being an input file, I’ve made it a NAL packet which I update each frame.

  I’m quite new to codecs and streaming, so I could be doing everything completely wrong. In each doGetNextFrame() should I be grabbing the NAL packet and doing something like

  memcpy(fTo, nal->p_payload, nal->i_payload)

  I assume that the payload is my frame data in bytes ? If anybody has an example of a class they derived from FramedSource that might at least be close to what I’m trying to do I would love to see it, this is all new to me and a little tricky to figure out what’s happening. Live555’s documentation is pretty much the code itself which doesn’t exactly make it easy for me to figure out.

• ffmpeg How to stream to my own server ?

  10 février 2018, par Ricky

  I have an app that can take in an hls source and output to an rtmp endpoint but what if I want to stream to my own server ? How would I set up my own rtmp end point. I would love to be able take a stream in and stream that stream to my own app so the user can see the stream. for example :

  if I stream to my own server like this

  ffmpeg -re -i "https://mnmedias.api.telequebec.tv/m3u8/29880.m3u8" -f flv "160.222.22.22"

  I have been googling all around and haven’t found any tutorials or guides in doing so. Any pointers would greatly be appreciated.