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• Concat two mp4 files with MP4Box error : Bad Parameter [on hold]

  6 février 2014, par user3279646

  I'm trying to concat two mp4 files. But I get an error : Bad Parameter

  Can someone help me with this problem ?

  I'm using de debian packages. On a previously version it worked fine. I moved from a old debian version to the new one and I also had to change the ffmpeg to avconv which I use to create these mp4 files. I also had to change the audio codec to '-acodec aac -strict experimental'

  Kind Regards,
  Marcel

      MP4Box -force-cat -cat conv1-1391608417314.mp4 -add conv1-1391608417345.mp4 -new output.mp4
  
      IsoMedia import - track ID 1 - Video (size 384 x 288)
  
      Error importing conv1-1391608417345.mp4: Bad Parameter

  ---

      MP4Box -info conv1-1391608417314.mp4
  
      Movie Info 
  
      Timescale 1000 - Duration 00:00:23.985
  
      Fragmented File no - 2 track(s)
  
      File Brand isom - version 512
  
      Created: UNKNOWN DATE
  
      File has root IOD (9 bytes)
  
      Scene PL 0xff - Graphics PL 0xff - OD PL 0xff
  
      Visual PL: Not part of MPEG-4 Visual profiles (0xfe)
  
      Audio PL: Not part of MPEG-4 audio profiles (0xfe)
  
      No streams included in root OD
  
  
  
      iTunes Info:
  
      Encoder Software: Lavf53.21.1
  
  
  
      Track # 1 Info - TrackID 1 - TimeScale 1000 - Duration 00:00:23.985
  
      Media Info: Language "Undetermined" - Type "vide:avc1" - 23988 samples
  
      Visual Track layout: x=0 y=0 width=384 height=288
  
      MPEG-4 Config: Visual Stream - ObjectTypeIndication 0x21
  
      AVC/H264 Video - Visual Size 384 x 288
  
      AVC Info: 1 SPS - 1 PPS - Profile High @ Level 4.2
  
      NAL Unit length bits: 32
  
      Self-synchronized
  
  
  
      Track # 2 Info - TrackID 2 - TimeScale 44100 - Duration 00:00:23.962
  
      Media Info: Language "Undetermined" - Type "soun:mp4a" - 1032 samples
  
      MPEG-4 Config: Audio Stream - ObjectTypeIndication 0x40
  
      MPEG-4 Audio MPEG-4 Audio AAC LC - 2 Channel(s) - SampleRate 44100
  
      Synchronized on stream 1
  
      Alternate Group ID 1

  ---

      MP4Box -version
  
      MP4Box - GPAC version 0.5.0-rev
  
      GPAC Copyright: (c) Jean Le Feuvre 2000-2005
  
      (c) ENST 2005-200X
  
      GPAC Configuration: --build=x86_64-linux-gnu --prefix=/usr --includedir=${prefix}/include --mandir=${prefix}/share/man --infodir=${prefix}/share/info --sysconfdir=/etc --localstatedir=/var --libdir=${prefix}/lib/x86_64-linux-gnu --libexecdir=${prefix}/lib/x86_64-linux-gnu --disable-maintainer-mode --disable-dependency-tracking --prefix=/usr --mandir=${prefix}/share/man --libdir=lib/x86_64-linux-gnu --extra-cflags='-Wall -fPIC -DPIC -I/usr/include/mozjs -DXP_UNIX' --enable-joystick --enable-debug --disable-ssl
  
      Features: GPAC_HAS_JPEG GPAC_HAS_PNG

• Raspberry Pi : how to configure audio sampling rate in ALSA, ffmpeg, mp3 encoding pipeline

  30 janvier 2014, par user3252022

  I have a Pi with the camera module and a USB sound card. The intent is to use this combination to generate the MPEG2-TS segments for HTTP Live Streaming (Apple's HLS.)

  raspivid -ih -e -w 1280 -h 720 -fps 25 -vf -hf -t 0 -b 1800000 -o - | \
  
  ffmpeg -y \
  
    -i - \
  
    -ac 1 \
  
    -f alsa -ar 22050 -i hw:0,0 \
  
    -c:v copy \
  
    -c:a libshine -strict -2 \
  
    -map 0:0 -map 1:0 \
  
    -f ssegment \
  
    -segment_list_entry_prefix live/ \
  
    -segment_time 10 \
  
    -segment_format mpegts \
  
    -segment_list "$base/live.m3u8" \
  
    -segment_list_size 20 \
  
    -segment_list_flags +live \
  
    -segment_list_type m3u8 \
  
    -segment_wrap 200 \
  
    live/%08d.ts"

  The above basically works. The heavy lifting of H.264 video encoding is done in hardware. However, MP3 encoding in software brings the ARM core to its knees. The option :

  -f alsa -ar 22050 -i hw:0,0

  is intended to lighten the CPU loading by reducing the ALSA audio sampling rate - from the 48kHz default down to 22.05kHz. The sampling rate, however, won't budge to anything below 44.1kHz.

  Any idea what I'm doing wrong ? Why can't I tell ALSA to sample at 22.05kHz (or 8kHz for that matter ?)

  Here's an excerpt output from running the encoding pipeline :

  ffmpeg version N-60125-g4b8c599 Copyright (c) 2000-2014 the FFmpeg developers
  
    built on Jan 29 2014 10:16:02 with gcc 4.6 (Debian 4.6.3-14+rpi1)
  
    configuration: --enable-gpl --enable-nonfree --enable-libshine
  
    ...   
  
  Input #0, h264, from 'pipe:':
  
    Duration: N/A, bitrate: N/A
  
      Stream #0:0: Video: h264 (High), yuv420p, 1280x720, 25 fps, 25 tbr, 1200k tbn, 50 tbc
  
  Guessed Channel Layout for  Input Stream #1.0 : mono
  
  Input #1, alsa, from 'hw:0,0':
  
    Duration: N/A, start: 1391066215.835499, bitrate: 705 kb/s
  
      Stream #1:0: Audio: pcm_s16le, 44100 Hz, mono, s16, 705 kb/s
  
      ...

• linphone on windows building error

  21 novembre 2013, par chhameed

  i have downloaded the linphone for Android from Github.
  i have also followed all the instructions in readme file. the problem is that when i run ndk-build command it produce below error .

  Android NDK: ERROR:jni/..//submodules/externals/build/ffmpeg/Android.mk:avutil-linphone: LOCAL_SRC_FILES points to a missing file
  
  Android NDK: Check that jni/..//submodules/externals/build/ffmpeg/arm/libavutil/libavutil-linphone-arm.so exists  or that its path is correct
  
  /cygdrive/c/development/android-ndk-r9b/build/core/prebuilt-library.mk:45: *** Android NDK: Aborting    .  Stop.

  what this error mean ? & how can i resolve this? i already spent many hours to searchout this error but no success...
  my OS is window 7 & i used cygwin for commands.

  i have found same question on SO
  Android NDK error when using FFmpeg in Android ?

  it says Basically problem was with ffmpeg lib.I was not able to compile ffmpeg lib on Windows 7. I switched to Ubuntu & tried AppUnite & after spending so many hours it was working.

  is it not possible to build linphone on windows ???