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• FFMPEG .oma to .mp3 "Unsupported codec 5 !" with a big file

  27 mars 2017, par Ventura

  I’m trying to convert a .OMA file to .MP3 but no success with a specific file.

  If I try :

  ffmpeg -i audio1.oma -f mp3 output.mp3

  The file is converted successfully. The file audio1.oma is a 3 MB file.

  Full output :

  ffmpeg version 3.2.4 Copyright (c) 2000-2017 the FFmpeg developers
  
    built with Apple LLVM version 8.0.0 (clang-800.0.42.1)
  
    configuration: --prefix=/usr/local/Cellar/ffmpeg/3.2.4 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-libmp3lame --enable-libx264 --enable-libxvid --enable-opencl --disable-lzma --enable-vda
  
    libavutil      55. 34.101 / 55. 34.101
  
    libavcodec     57. 64.101 / 57. 64.101
  
    libavformat    57. 56.101 / 57. 56.101
  
    libavdevice    57.  1.100 / 57.  1.100
  
    libavfilter     6. 65.100 /  6. 65.100
  
    libavresample   3.  1.  0 /  3.  1.  0
  
    libswscale      4.  2.100 /  4.  2.100
  
    libswresample   2.  3.100 /  2.  3.100
  
    libpostproc    54.  1.100 / 54.  1.100
  
  [oma @ 0x7f8fc4000000] Estimating duration from bitrate, this may be inaccurate
  
  Input #0, oma, from 'audio1.oma':
  
    Metadata:
  
      title           : Is This It
  
      artist          : The Strokes
  
      album           : Is This It
  
      genre           : Rock
  
      OMG_TRLDA       : 2001/01/01 00:00:00
  
      TLEN            : 153000
  
    Duration: 00:02:33.36, start: 0.000000, bitrate: 128 kb/s
  
      Stream #0:0: Audio: mp3 ([3][0][0][0] / 0x0003), 44100 Hz, stereo, s16p, 128 kb/s
  
  Output #0, mp3, to 'output.mp3':
  
    Metadata:
  
      TIT2            : Is This It
  
      TPE1            : The Strokes
  
      TALB            : Is This It
  
      TCON            : Rock
  
      OMG_TRLDA       : 2001/01/01 00:00:00
  
      TLEN            : 153000
  
      TSSE            : Lavf57.56.101
  
      Stream #0:0: Audio: mp3 (libmp3lame), 44100 Hz, stereo, s16p
  
      Metadata:
  
        encoder         : Lavc57.64.101 libmp3lame
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (mp3 (native) -> mp3 (libmp3lame))
  
  Press [q] to stop, [?] for help
  
  size=    2397kB time=00:02:33.35 bitrate= 128.0kbits/s speed=37.2x    
  
  video:0kB audio:2397kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.016094%

  If I try the same with another .oma (53 MB) I’m getting the error :

  Unsupported codec 5 ! audio2.oma : Function not implemented

  Full output :

  ffmpeg version 3.2.4 Copyright (c) 2000-2017 the FFmpeg developers
  
    built with Apple LLVM version 8.0.0 (clang-800.0.42.1)
  
    configuration: --prefix=/usr/local/Cellar/ffmpeg/3.2.4 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-libmp3lame --enable-libx264 --enable-libxvid --enable-opencl --disable-lzma --enable-vda
  
    libavutil      55. 34.101 / 55. 34.101
  
    libavcodec     57. 64.101 / 57. 64.101
  
    libavformat    57. 56.101 / 57. 56.101
  
    libavdevice    57.  1.100 / 57.  1.100
  
    libavfilter     6. 65.100 /  6. 65.100
  
    libavresample   3.  1.  0 /  3.  1.  0
  
    libswscale      4.  2.100 /  4.  2.100
  
    libswresample   2.  3.100 /  2.  3.100
  
    libpostproc    54.  1.100 / 54.  1.100
  
  [oma @ 0x7f8792000000] Unsupported codec 5!
  
  audio2.OMA: Function not implemented

  Both audios works fine when using the MP3 Player.
  The first audio which works is just a random song from my MP3 player to test.
  The second file was recorded in a music studio playing live with multiple channels.

  Anything I’m missing here ?

• tcp : set socket buffer sizes before listen/connect/accept

  9 janvier 2017, par Joel Cunningham

  tcp : set socket buffer sizes before listen/connect/accept
  From e24d95c0e06a878d401ee34fd6742fcaddeeb95f Mon Sep 17 00:00:00 2001
  
  From : Joel Cunningham <joel.cunningham@me.com>
  
  Date : Mon, 9 Jan 2017 13:37:51 -0600
  
  Subject : [PATCH] tcp : set socket buffer sizes before listen/connect/accept
  Attempting to set SO_RCVBUF and SO_SNDBUF on TCP sockets after connection
  
  establishment is incorrect and some stacks ignore the set call on the socket at
  
  this point.  This has been observed on MacOS/iOS.  Windows 7 has some peculiar
  
  behavior where setting SO_RCVBUF after applies only if the buffer is increasing
  
  from the default while decreases are ignored.  This is possibly how the incorrect
  
  usage has gone unnoticed
  Unix Network Programming Vol. 1 : The Sockets Networking API (3rd edition, seciton 7.5) :
  "When setting the size of the TCP socket receive buffer, the ordering of the
  
  function calls is important.  This is because of TCP’s window scale option,
  
  which is exchanged with the peer on SYN segments when the connection is
  
  established. For a client, this means the SO_RCVBUF socket option must be
  
  set before calling connect.  For a server, this means the socket option must
  
  be set for the listening socket before calling listen.  Setting this option
  
  for the connected socket will have no effect whatsoever on the possible window
  
  scale option because accept does not return with the connected socket until
  
  TCP’s three-way handshake is complete.  This is why the option must be set on
  
  the listening socket. (The sizes of the socket buffers are always inherited from
  
  the listening socket by the newly created connected socket)"
  Signed-off-by : Joel Cunningham <joel.cunningham@me.com>
  
  Signed-off-by : Michael Niedermayer <michael@niedermayer.cc>
  

  • [DH] libavformat/tcp.c

• Writing A Dreamcast Media Player

  6 janvier 2017, par Multimedia Mike — Sega Dreamcast

  I know I’m not the only person to have the idea to port a media player to the Sega Dreamcast video game console. But I did make significant progress on an implementation. I’m a little surprised to realize that I haven’t written anything about it on this blog yet, given my propensity for publishing my programming misadventures.

  
  
  

  This old effort had been on my mind lately due to its architectural similarities to something else I was recently brainstorming.

  Early Days
  Porting a multimedia player was one of the earliest endeavors that I embarked upon in the multimedia domain. It’s a bit fuzzy for me now, but I’m pretty sure that my first exposure to the MPlayer project in 2001 arose from looking for a multimedia player to port. I fed it through the Dreamcast development toolchain but encountered roadblocks pretty quickly. However, this got me looking at the MPlayer source code and made me wonder how I could contribute, which is how I finally broke into practical open source multimedia hacking after studying the concepts and technology for more than a year at that point.

  Eventually, I jumped over to the xine project. After hacking on that for awhile, I remembered my DC media player efforts and endeavored to compile xine to the console. The first attempt was to simply compile the codebase using the Dreamcast hobbyist community’s toolchain. This is when I came to fear the multithreaded snake pit in xine’s core. Again, my memories are hazy on the specifics, but I remember the engine having a bunch of threading hacks with comments along the lines of “this code deadlocks sometimes, so on shutdown, monitor this lock and deliberately break it if it has been more than 3 seconds”.

  Something Workable
  Eventually, I settled on a combination of FFmpeg’s libavcodec library for audio and video decoders, xine’s demuxer library, and xine’s input API, combined with my own engine code to tie it all together along with video and output drivers provided by the KallistiOS hobbyist OS for Dreamcast. Here is a simple diagram of the data movement through this player :

  
  
  

  Details and Challenges
  This is a rare occasion when I actually got to write the core of a media player engine. I made some mistakes.

  xine’s internal clock ran at 90000 Hz. At least, its internal timestamps were all in reference to a 90 kHz clock. I got this brilliant idea to trigger timer interrupts at 6000 Hz to drive the engine. Whatever the timer facilities on the Dreamcast, I found that 6 kHz was the greatest common divisor with 90 kHz. This means that if I could have found an even higher GCD frequency, I would have used that instead.

  So the idea was that, for a 30 fps video, the engine would know to render a frame on every 200th timer interrupt. I eventually realized that servicing 6000 timer interrupts every second would incur a ridiculous amount of overhead. After that, my engine’s philosophy was to set a timer to fire for the next frame while beginning to process the current frame. I.e., when rendering a frame, set a timer to call back in 1/30th of a second. That worked a lot better.

  As I was still keen on 8-bit paletted image codecs at the time (especially since they were simple and small for bootstrapping this project), I got to use output palette images directly thanks to the Dreamcast’s paletted textures. So that was exciting. The engine didn’t need to convert the paletted images to a different colorspace before rendering. However, I seem to recall that the Dreamcast’s PowerVR graphics hardware required that 8-bit textures be twiddled/swizzled. Thus, it was still required to manipulate the 8-bit image before rendering.

  I made good progress on this player concept. However, a huge blocker for me was that I didn’t know how to make a proper user interface for the media player. Obviously, programming the Dreamcast occurred at a very low level (at least with the approach I was using), so there were no UI widgets easily available.

  This was circa 2003. I assumed there must have been some embedded UI widget libraries with amenable open source licenses that I could leverage. I remember searching and checking out a library named libSTK. I think STK stood for “set-top toolkit” and was positioned specifically for doing things like media player UIs on low-spec embedded computing devices. The domain hosting the project is no longer useful but this appears to be a backup of the core code.

  It sounded promising, but the libSTK developers had a different definition of “low-spec embedded” device than I did. I seem to recall that they were targeting something along with likes of a Pentium III clocked at 800 MHz with 128 MB RAM. The Dreamcast, by contrast, has a 200 MHz SH-4 CPU and 16 MB RAM. LibSTK was also authored in C++ and leveraged the Boost library (my first exposure to that code), and this all had the effect of making binaries quite large while I was trying to keep the player in lean C.

  Regrettably, I never made any serious progress on a proper user interface. I think that’s when the player effort ran out of steam.

  The Code
  So, that’s another project that I never got around to finishing or publishing. I was able to find the source code so I decided to toss it up on github, along with 2 old architecture outlines that I was able to dig up. It looks like I was starting small, just porting over a few of the demuxers and decoders that I knew well.

  I’m wondering if it would still be as straightforward to separate out such components now, more than 13 years later ?

  The post Writing A Dreamcast Media Player first appeared on Breaking Eggs And Making Omelettes.