Recherche avancée

Médias (39)

Mot : - Tags -/audio

Autres articles (52)

  • List of compatible distributions

    26 avril 2011, par

    The table below is the list of Linux distributions compatible with the automated installation script of MediaSPIP. Distribution nameVersion nameVersion number Debian Squeeze 6.x.x Debian Weezy 7.x.x Debian Jessie 8.x.x Ubuntu The Precise Pangolin 12.04 LTS Ubuntu The Trusty Tahr 14.04
    If you want to help us improve this list, you can provide us access to a machine whose distribution is not mentioned above or send the necessary fixes to add (...)

  • Automated installation script of MediaSPIP

    25 avril 2011, par

    To overcome the difficulties mainly due to the installation of server side software dependencies, an "all-in-one" installation script written in bash was created to facilitate this step on a server with a compatible Linux distribution.
    You must have access to your server via SSH and a root account to use it, which will install the dependencies. Contact your provider if you do not have that.
    The documentation of the use of this installation script is available here.
    The code of this (...)

  • Sélection de projets utilisant MediaSPIP

    29 avril 2011, par

    Les exemples cités ci-dessous sont des éléments représentatifs d’usages spécifiques de MediaSPIP pour certains projets.
    Vous pensez avoir un site "remarquable" réalisé avec MediaSPIP ? Faites le nous savoir ici.
    Ferme MediaSPIP @ Infini
    L’Association Infini développe des activités d’accueil, de point d’accès internet, de formation, de conduite de projets innovants dans le domaine des Technologies de l’Information et de la Communication, et l’hébergement de sites. Elle joue en la matière un rôle unique (...)

Sur d’autres sites (9589)

  • Audio stuttering every couple of seconds

    27 juin 2019, par Glutch

    I’m merging a video (recorded with ffmpeg, good quality, all solid), with a musicfile.mp3. However every couple of seconds the music stutters and skips slightly. Which seems very strange since simply adding music on top of a video sounds like the engine could relax and take its time, creating no artifacts. (In comparison to recording live desktop footage). Can anyone help me sort this out ?

    System : MacOS MBP 2015, 16gb ram 2.7ghz i5

    ffmpeg -i "temp/1561246948349.mkv" -i "music/happy.mp3" -vcodec copy -filter_complex amix -map 0:v -map 0:a -map 1:a -shortest -b:a 144k "finished/2019-06-22/1561246948349/output.mkv"
    
ffmpeg version 4.1.3 Copyright (c) 2000-2019 the FFmpeg developers
    
  built with Apple LLVM version 10.0.1 (clang-1001.0.46.4)
    
  configuration: --prefix=/usr/local/Cellar/ffmpeg/4.1.3_1 --enable-shared --enable-pthreads --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags='-I/Library/Java/JavaVirtualMachines/adoptopenjdk-11.0.2.jdk/Contents/Home/include -I/Library/Java/JavaVirtualMachines/adoptopenjdk-11.0.2.jdk/Contents/Home/include/darwin' --host-ldflags= --enable-ffplay --enable-gnutls --enable-gpl --enable-libaom --enable-libbluray --enable-libmp3lame --enable-libopus --enable-librubberband --enable-libsnappy --enable-libtesseract --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-libxvid --enable-lzma --enable-libfontconfig --enable-libfreetype --enable-frei0r --enable-libass --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librtmp --enable-libspeex --enable-videotoolbox --disable-libjack --disable-indev=jack --enable-libaom --enable-libsoxr
    
  libavutil      56. 22.100 / 56. 22.100
    
  libavcodec     58. 35.100 / 58. 35.100
    
  libavformat    58. 20.100 / 58. 20.100
    
  libavdevice    58.  5.100 / 58.  5.100
    
  libavfilter     7. 40.101 /  7. 40.101
    
  libavresample   4.  0.  0 /  4.  0.  0
    
  libswscale      5.  3.100 /  5.  3.100
    
  libswresample   3.  3.100 /  3.  3.100
    
  libpostproc    55.  3.100 / 55.  3.100
    
Input #0, matroska,webm, from 'temp/1561246948349.mkv':
    
  Metadata:
    
    ENCODER         : Lavf58.20.100
    
  Duration: 00:00:21.50, start: 0.000000, bitrate: 5834 kb/s
    
    Stream #0:0: Video: h264 (High 4:4:4 Predictive), yuv422p(progressive), 2880x1800, 30 fps, 30 tbr, 1k tbn, 2000k tbc (default)
    
    Metadata:
    
      ENCODER         : Lavc58.35.100 libx264
    
      DURATION        : 00:00:21.467000000
    
    Stream #0:1: Audio: vorbis, 44100 Hz, stereo, fltp (default)
    
    Metadata:
    
      ENCODER         : Lavc58.35.100 libvorbis
    
      DURATION        : 00:00:21.496000000
    
Input #1, mp3, from 'music/happy.mp3':
    
  Metadata:
    
    album           : Random
    
    genre           : Jazz & Blues
    
  Duration: 00:15:59.84, start: 0.025057, bitrate: 186 kb/s
    
    Stream #1:0: Audio: mp3, 44100 Hz, stereo, fltp, 186 kb/s
    
    Metadata:
    
      encoder         : LAME3.100
    
Stream mapping:
    
  Stream #0:1 (vorbis) -> amix:input0
    
  Stream #1:0 (mp3float) -> amix:input1
    
  amix -> Stream #0:0 (libvorbis)
    
  Stream #0:0 -> #0:1 (copy)
    
Press [q] to stop, [?] for help
    
Output #0, matroska, to 'finished/2019-06-22/1561246948349/output.mkv':
    
  Metadata:
    
    encoder         : Lavf58.20.100
    
    Stream #0:0: Audio: vorbis (libvorbis) (oV[0][0] / 0x566F), 44100 Hz, stereo, fltp, 144 kb/s (default)
    
    Metadata:
    
      encoder         : Lavc58.35.100 libvorbis
    
    Stream #0:1: Video: h264 (High 4:4:4 Predictive) (H264 / 0x34363248), yuv422p(progressive), 2880x1800, q=2-31, 30 fps, 30 tbr, 1k tbn, 1k tbc (default)
    
    Metadata:
    
      ENCODER         : Lavc58.35.100 libx264
    
      DURATION        : 00:00:21.467000000
    
frame=  640 fps=0.0 q=-1.0 Lsize=   15227kB time=00:00:21.46 bitrate=5810.3kbits/s speed=33.8x    
    
video:14888kB audio:318kB subtitle:0kB other streams:0kB global headers:4kB muxing overhead: 0.139864%

  • ffmpeg Output file #0 does not contain any stream when trying to access 1 of 2 audio streams

    7 juillet 2019, par nulltorpedo
    ffmpeg -i input.mkv  -map 0:2 -c copy -strict -2  audio.mkv

    Hi I have the above command. The output shows that there are 2 audio streams. I want to copy just the ac3 audio (actually I want to convert it but even this copy does not work). I have truncated the output print where there is metadata

    NEW updated sample with full log which results in same message

    ffmpeg -i input.mka -map 0:0 -c:a libfdk_aac   aac_out.m4a
    
ffmpeg version 2.7.1 Copyright (c) 2000-2015 the FFmpeg developers
    
  built with gcc 4.9.3 (crosstool-NG 1.20.0) 20150311 (prerelease)
    
  configuration: --prefix=/usr --incdir='${prefix}/include/ffmpeg' --arch=i686 --target-os=linux --cross-prefix=/usr/local/x86_64-pc-linux-gnu/bin/x86_64-pc-linux-gnu- --enable-cross-compile --enable-optimizations --enable-pic --enable-gpl --enable-shared --disable-static --enable-version3 --enable-nonfree --enable-libfaac --enable-encoders --enable-pthreads --disable-bzlib --disable-protocol=rtp --disable-muxer=image2 --disable-muxer=image2pipe --disable-swscale-alpha --disable-ffserver --disable-ffplay --disable-devices --disable-bzlib --disable-altivec --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libmp3lame --disable-vaapi --disable-decoder=amrnb --disable-decoder=ac3 --disable-decoder=ac3_fixed --disable-encoder=zmbv --disable-encoder=dca --disable-encoder=ac3 --disable-encoder=ac3_fixed --disable-encoder=eac3 --disable-decoder=dca --disable-decoder=eac3 --disable-decoder=truehd --cc=/usr/local/x86_64-pc-linux-gnu/bin/x86_64-pc-linux-gnu-ccache-gcc --enable-yasm --enable-libx264 --enable-encoder=libx264
    
  libavutil      54. 27.100 / 54. 27.100
    
  libavcodec     56. 41.100 / 56. 41.100
    
  libavformat    56. 36.100 / 56. 36.100
    
  libavdevice    56.  4.100 / 56.  4.100
    
  libavfilter     5. 16.101 /  5. 16.101
    
  libswscale      3.  1.101 /  3.  1.101
    
  libswresample   1.  2.100 /  1.  2.100
    
  libpostproc    53.  3.100 / 53.  3.100
    
Input #0, matroska,webm, from '/volume1/..../input.mka':
    
  Metadata:
    
    encoder         : libebml v1.3.9 + libmatroska v1.5.2
    
    creation_time   : 2019-07-07 06:19:20
    
  Duration: 02:29:21.98, start: 0.000000, bitrate: 640 kb/s
    
    Stream #0:0(eng): Audio: ac3, 48000 Hz, 5.1(side), 640 kb/s
    
    Metadata:
    
      BPS-eng         : 640000
    
      DURATION-eng    : 02:29:21.984000480
    
      NUMBER_OF_FRAMES-eng: 280062
    
      NUMBER_OF_BYTES-eng: 716958720
    
      _STATISTICS_WRITING_APP-eng: mkvmerge v35.0.0 ('All The Love In The World') 64-bit
    
      _STATISTICS_WRITING_DATE_UTC-eng: 2019-07-07 06:19:20
    
      _STATISTICS_TAGS-eng: BPS DURATION NUMBER_OF_FRAMES NUMBER_OF_BYTES
    
Output #0, ipod, to 'aac_out.m4a':
    
  Metadata:
    
    encoder         : libebml v1.3.9 + libmatroska v1.5.2
    
Output file #0 does not contain any stream

  • How to convert aac to ogg opus keeping bit rate and sample rate unchanged

    25 juin 2019, par Doovi

    I’m trying to convert a .aac file to .opus but after inspecting with ffprobe I get different bit and sample rates.

    While input file’s audio stream bit rate is 245995, the output file’s audio stream has no bit rate specified - "format" shows bit rate of 118788.

    While input file’s audio stream sample rate is 44100, the output’s is 48000.

    ffprobe  -v error -show_format -show_streams input.aac
    
[STREAM]
    
index=0
    
codec_name=aac
    
codec_long_name=AAC (Advanced Audio Coding)
    
profile=LC
    
codec_type=audio
    
codec_time_base=1/44100
    
codec_tag_string=[0][0][0][0]
    
codec_tag=0x0000
    
sample_fmt=fltp
    
sample_rate=44100
    
channels=2
    
channel_layout=stereo
    
bits_per_sample=0
    
id=N/A
    
r_frame_rate=0/0
    
avg_frame_rate=0/0
    
time_base=1/28224000
    
start_pts=N/A
    
start_time=N/A
    
duration_ts=106533390807
    
duration=3774.567418
    
bit_rate=245995
    
max_bit_rate=N/A
    
bits_per_raw_sample=N/A
    
nb_frames=N/A
    
nb_read_frames=N/A
    
nb_read_packets=N/A
    
DISPOSITION:default=0
    
DISPOSITION:dub=0
    
DISPOSITION:original=0
    
DISPOSITION:comment=0
    
DISPOSITION:lyrics=0
    
DISPOSITION:karaoke=0
    
DISPOSITION:forced=0
    
DISPOSITION:hearing_impaired=0
    
DISPOSITION:visual_impaired=0
    
DISPOSITION:clean_effects=0
    
DISPOSITION:attached_pic=0
    
DISPOSITION:timed_thumbnails=0
    
[/STREAM]
    
[FORMAT]
    
filename=input.aac
    
nb_streams=1
    
nb_programs=0
    
format_name=aac
    
format_long_name=raw ADTS AAC (Advanced Audio Coding)
    
start_time=N/A
    
duration=3774.567418
    
size=116065589
    
bit_rate=245995
    
probe_score=51
    
[/FORMAT]
    ffmpeg -nostdin -i input.aac -c:a libopus output.opus
    
ffmpeg version N-93449-g013f714 Copyright (c) 2000-2019 the FFmpeg developers
    
  built with gcc 7 (Ubuntu 7.3.0-27ubuntu1~18.04)
    
  configuration: --prefix=/home/vagrant/ffmpeg_build --pkg-config-flags=--static --extra-cflags=-I/home/vagrant/ffmpeg_build/include --extra-ldflags=-L/home/vagrant/ffmpeg_build/lib --extra-libs='-lpthread -lm' --bindir=/home/vagrant/bin --enable-gpl --enable-libass --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-libopus --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-nonfree
    
  libavutil      56. 26.100 / 56. 26.100
    
  libavcodec     58. 47.105 / 58. 47.105
    
  libavformat    58. 26.101 / 58. 26.101
    
  libavdevice    58.  7.100 / 58.  7.100
    
  libavfilter     7. 48.100 /  7. 48.100
    
  libswscale      5.  4.100 /  5.  4.100
    
  libswresample   3.  4.100 /  3.  4.100
    
  libpostproc    55.  4.100 / 55.  4.100
    
[aac @ 0x55d4b7e21d80] Estimating duration from bitrate, this may be inaccurate
    
Input #0, aac, from 'input.aac':
    
  Duration: 01:02:54.57, bitrate: 245 kb/s
    
    Stream #0:0: Audio: aac (LC), 44100 Hz, stereo, fltp, 245 kb/s
    
Stream mapping:
    
  Stream #0:0 -> #0:0 (aac (native) -> opus (libopus))
    
[libopus @ 0x55d4b7e3f8c0] No bit rate set. Defaulting to 96000 bps.
    
Output #0, opus, to 'output.opus':
    
  Metadata:
    
    encoder         : Lavf58.26.101
    
    Stream #0:0: Audio: opus (libopus), 48000 Hz, stereo, flt, 96 kb/s
    
    Metadata:
    
      encoder         : Lavc58.47.105 libopus
    
size=   52103kB time=00:59:53.21 bitrate= 118.8kbits/s speed=66.2x
    
video:0kB audio:51733kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.715930%
    ffprobe -v error -show_format -show_streams output.opus
    
[STREAM]
    
index=0
    
codec_name=opus
    
codec_long_name=Opus (Opus Interactive Audio Codec)
    
profile=unknown
    
codec_type=audio
    
codec_time_base=1/48000
    
codec_tag_string=[0][0][0][0]
    
codec_tag=0x0000
    
sample_fmt=fltp
    
sample_rate=48000
    
channels=2
    
channel_layout=stereo
    
bits_per_sample=0
    
id=N/A
    
r_frame_rate=0/0
    
avg_frame_rate=0/0
    
time_base=1/48000
    
start_pts=0
    
start_time=0.000000
    
duration_ts=172473677
    
duration=3593.201604
    
bit_rate=N/A
    
max_bit_rate=N/A
    
bits_per_raw_sample=N/A
    
nb_frames=N/A
    
nb_read_frames=N/A
    
nb_read_packets=N/A
    
DISPOSITION:default=0
    
DISPOSITION:dub=0
    
DISPOSITION:original=0
    
DISPOSITION:comment=0
    
DISPOSITION:lyrics=0
    
DISPOSITION:karaoke=0
    
DISPOSITION:forced=0
    
DISPOSITION:hearing_impaired=0
    
DISPOSITION:visual_impaired=0
    
DISPOSITION:clean_effects=0
    
DISPOSITION:attached_pic=0
    
DISPOSITION:timed_thumbnails=0
    
TAG:ENCODER=Lavc58.47.105 libopus
    
[/STREAM]
    
[FORMAT]
    
filename=output.opus
    
nb_streams=1
    
nb_programs=0
    
format_name=ogg
    
format_long_name=Ogg
    
start_time=0.000000
    
duration=3593.201604
    
size=53353867
    
bit_rate=118788
    
probe_score=100
    
[/FORMAT]

    How can I preserve the quality of the input file ? Am I missing something in the ffmpeg cmd ?

Navigation