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Rennes Emotion Map 2010-11
19 octobre 2011, par
Mis à jour : Juillet 2013
Langue : français
Type : Texte
Autres articles (41)
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Les autorisations surchargées par les plugins
27 avril 2010, parMediaspip core
autoriser_auteur_modifier() afin que les visiteurs soient capables de modifier leurs informations sur la page d’auteurs -
HTML5 audio and video support
13 avril 2011, parMediaSPIP uses HTML5 video and audio tags to play multimedia files, taking advantage of the latest W3C innovations supported by modern browsers.
The MediaSPIP player used has been created specifically for MediaSPIP and can be easily adapted to fit in with a specific theme.
For older browsers the Flowplayer flash fallback is used.
MediaSPIP allows for media playback on major mobile platforms with the above (...) -
Librairies et binaires spécifiques au traitement vidéo et sonore
31 janvier 2010, parLes logiciels et librairies suivantes sont utilisées par SPIPmotion d’une manière ou d’une autre.
Binaires obligatoires FFMpeg : encodeur principal, permet de transcoder presque tous les types de fichiers vidéo et sonores dans les formats lisibles sur Internet. CF ce tutoriel pour son installation ; Oggz-tools : outils d’inspection de fichiers ogg ; Mediainfo : récupération d’informations depuis la plupart des formats vidéos et sonores ;
Binaires complémentaires et facultatifs flvtool2 : (...)
Sur d’autres sites (9042)
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TCP Connection refused error when using FFMPEG for audio stream to HTTP on macOS
26 novembre 2020, par freddyI'm trying to stream my microphone input via HTTP using ffmpeg, so I can stream it in HTML. I run the following ffmpeg command :


ffmpeg -f avfoundation -i ":1" -c:a libmp3lame -f mp3 -r 30 http://localhost:809


It, however, crashes with the following error message :


ffmpeg version 4.3.1 Copyright (c) 2000-2020 the FFmpeg developers
 built with Apple clang version 12.0.0 (clang-1200.0.32.27)
 configuration: --prefix=/usr/local/Cellar/ffmpeg/4.3.1_4 --enable-shared --enable-pthreads --enable-version3 --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-ffplay --enable-gnutls --enable-gpl --enable-libaom --enable-libbluray --enable-libdav1d --enable-libmp3lame --enable-libopus --enable-librav1e --enable-librubberband --enable-libsnappy --enable-libsrt --enable-libtesseract --enable-libtheora --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --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-libsoxr --enable-videotoolbox --disable-libjack --disable-indev=jack
 libavutil 56. 51.100 / 56. 51.100
 libavcodec 58. 91.100 / 58. 91.100
 libavformat 58. 45.100 / 58. 45.100
 libavdevice 58. 10.100 / 58. 10.100
 libavfilter 7. 85.100 / 7. 85.100
 libavresample 4. 0. 0 / 4. 0. 0
 libswscale 5. 7.100 / 5. 7.100
 libswresample 3. 7.100 / 3. 7.100
 libpostproc 55. 7.100 / 55. 7.100
Input #0, avfoundation, from ':1':
 Duration: N/A, start: 3445.340045, bitrate: 22579 kb/s
 Stream #0:0: Audio: pcm_f32le, 44100 Hz, hexadecagonal, flt, 22579 kb/s
[tcp @ 0x7fa46ec96600] Connection to tcp://localhost:8090 failed: Connection refused
http://localhost:8090: Connection refused



I've had success with streaming on that port using VLC, but it for some reason won't work using ffmpeg. Any ideas on how to fix this ?


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mp3 to wav with ffmpeg reduces quality and duration
25 novembre 2020, par NeretI took this mp3 file and converted it to wav with Audition and with this ffmpeg command :


ffmpeg -i "Casey Don’t You Fret - Dan Lebowitz.mp3" -c:a pcm_f32le "Casey Don’t You Fret - Dan Lebowitz_FFMPEG.wav"



After that I checked the statistics in Audition. The wav file which was generated with Audition has exactly the same statistics as the original mp3 file.




The duration of ffmpeg file has changed. Audio statistics became worse.
Why is this happening ? Can I fix it ?


I used
ffmpeg version 2020-11-22-git-0066bf4d1a-full_build-www.gyan.dev
on Windows.

UPDATE 1 :
I cut a few seconds of mp3 at the beginning and at the end :




FFMPEG added silence at the beginning and increased duration.


UPDATE 2 :
Look how ffmpeg changed the waveform of this
2.mp3
file in the middle :

ffmpeg -y -i 2.mp3 -c:a pcm_f32le 2_FFMPEG.wav





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FFMPEG Streaming to twitch low bitrate
17 juillet 2020, par El_PresidenteI have a python script that will produce frames for a video stream. To stream it to twitch I decided to use ffmpeg (as it is the only option I found). However, the bitrate of my stream is very low (70 KB), although in ffmpeg options it's set to 3000K.


# This script copies the video frame by frame
import cv2
import subprocess as sp
twitch_stream_key = 'MY_TWITCH_STREAM_KEY'
input_file = 'video.mp4'

cap = cv2.VideoCapture(input_file)
ret, frame = cap.read()
height, width, ch = frame.shape

ffmpeg = 'FFMPEG'
dimension = '{}x{}'.format(width, height)

fps = cap.get(cv2.CAP_PROP_FPS)
command = []
command.extend([
 'FFMPEG',
 '-loglevel', 'verbose',
 '-y', # overwrite previous file/stream
 '-analyzeduration', '1',
 '-f', 'rawvideo',
 '-r', '%d' % fps, # set a fixed frame rate
 '-vcodec', 'rawvideo',
 # size of one frame
 '-s', '%dx%d' % (width, height),
 '-pix_fmt', 'rgb24', # The input are raw bytes
 '-thread_queue_size', '1024',
 '-i', '-', # The input comes from a pipe
]) 
command.extend([
 '-ar', '8000',
 '-ac', '1',
 '-f', 's16le',
 '-i', 'work.mp3',
])
command.extend([
 # VIDEO CODEC PARAMETERS
 '-vcodec', 'libx264',
 '-r', '%d' % fps,
 '-b:v', '3000k',
 '-s', '%dx%d' % (width, height),
 '-preset', 'faster', '-tune', 'zerolatency',
 '-crf', '23',
 '-pix_fmt', 'yuv420p',

 '-minrate', '3000k', '-maxrate', '3000k',
 '-bufsize', '12000k',
 '-g', '60', # key frame distance
 '-keyint_min', '1',

 # AUDIO CODEC PARAMETERS
 '-acodec', 'libmp3lame', '-ar', '44100', '-b:a', '160k',
 # '-bufsize', '8192k',
 '-ac', '1',
 '-map', '0:v', '-map', '1:a',

 '-threads', '2',
 # STREAM TO TWITCH
 '-f', 'flv', 'rtmp://live-hel.twitch.tv/app/%s' %
 twitch_stream_key
])
proc = sp.Popen(command, stdin=sp.PIPE, stderr=sp.PIPE)

while True:
 ret, frame = cap.read() 
 if not ret: 
 break 
 proc.stdin.write(frame.tostring())

cap.release()
proc.stdin.close()
proc.stderr.close()
proc.wait()



How can I increase the bitrate ? Maybe you can point me towards some different solution on how I can stream python frames to twitch or any other rtmp server.


Here is the complete log, the audio is also broken, it's just noise :


ffmpeg version git-2020-06-01-dd76226 Copyright (c) 2000-2020 the FFmpeg developers
 built with gcc 9.3.1 (GCC) 20200523
 configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libsrt --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvmaf --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --disable-w32threads --enable-libmfx --enable-ffnvcodec --enable-cuda-llvm --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt --enable-amf
 libavutil 56. 49.100 / 56. 49.100
 libavcodec 58. 90.100 / 58. 90.100
 libavformat 58. 44.100 / 58. 44.100
 libavdevice 58. 9.103 / 58. 9.103
 libavfilter 7. 84.100 / 7. 84.100
 libswscale 5. 6.101 / 5. 6.101
 libswresample 3. 6.100 / 3. 6.100
 libpostproc 55. 6.100 / 55. 6.100
Input #0, rawvideo, from 'pipe:':
 Duration: N/A, start: 0.000000, bitrate: 1443225 kb/s
 Stream #0:0: Video: rawvideo, 1 reference frame (RGB[24] / 0x18424752), rgb24, 1920x1080, 1443225 kb/s, 29 tbr, 29 tbn, 29 tbc
[s16le @ 0000026d64eb5340] Estimating duration from bitrate, this may be inaccurate
Guessed Channel Layout for Input Stream #1.0 : mono
Input #1, s16le, from 'work.mp3':
 Metadata:
 encoded_by : iTunes v7.0
 Duration: 00:09:36.13, bitrate: 128 kb/s
 Stream #1:0: Audio: pcm_s16le, 8000 Hz, mono, s16, 128 kb/s
[tcp @ 0000026d64ee34c0] Starting connection attempt to 99.181.64.78 port 1935
[tcp @ 0000026d64ee34c0] Successfully connected to 99.181.64.78 port 1935
Stream mapping:
 Stream #0:0 -> #0:0 (rawvideo (native) -> h264 (libx264))
 Stream #1:0 -> #0:1 (pcm_s16le (native) -> mp3 (libmp3lame))
[graph 0 input from stream 0:0 @ 0000026d64f47c00] w:1920 h:1080 pixfmt:rgb24 tb:1/29 fr:29/1 sar:0/1
[scaler_out_0_0 @ 0000026d64f4c780] w:1920 h:1080 flags:'bicubic' interl:0
[scaler_out_0_0 @ 0000026d64f4c780] w:1920 h:1080 fmt:rgb24 sar:0/1 -> w:1920 h:1080 fmt:yuv420p sar:0/1 flags:0x4
[libx264 @ 0000026d64edf840] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX FMA3 BMI2 AVX2
[libx264 @ 0000026d64edf840] profile High, level 4.0, 4:2:0, 8-bit
[libx264 @ 0000026d64edf840] 264 - core 160 - H.264/MPEG-4 AVC codec - Copyleft 2003-2020 - http://www.videolan.org/x264.html - options: cabac=1 ref=2 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=4 psy=1 psy_rd=1.00:0.00 mixed_ref=0 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=0 threads=2 lookahead_threads=2 sliced_threads=1 slices=2 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=0 weightp=1 keyint=60 keyint_min=1 scenecut=40 intra_refresh=0 rc_lookahead=0 rc=crf mbtree=0 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 vbv_maxrate=3000 vbv_bufsize=12000 crf_max=0.0 nal_hrd=none filler=0 ip_ratio=1.40 aq=1:1.00
[graph_1_in_1_0 @ 0000026d651319c0] tb:1/8000 samplefmt:s16 samplerate:8000 chlayout:0x4
[format_out_0_1 @ 0000026d65132d80] auto-inserting filter 'auto_resampler_0' between the filter 'Parsed_anull_0' and the filter 'format_out_0_1'
[auto_resampler_0 @ 0000026d651331c0] ch:1 chl:mono fmt:s16 r:8000Hz -> ch:1 chl:mono fmt:s16p r:44100Hz
Output #0, flv, to 'rtmp://live-hel.twitch.tv/app/live_*************':
 Metadata:
 encoder : Lavf58.44.100
 Stream #0:0: Video: h264 (libx264), 1 reference frame ([7][0][0][0] / 0x0007), yuv420p(progressive), 1920x1080, q=-1--1, 3000 kb/s, 29 fps, 1k tbn, 29 tbc
 Metadata:
 encoder : Lavc58.90.100 libx264
 Side data:
 cpb: bitrate max/min/avg: 3000000/0/3000000 buffer size: 12000000 vbv_delay: N/A
 Stream #0:1: Audio: mp3 (libmp3lame) ([2][0][0][0] / 0x0002), 44100 Hz, mono, s16p, delay 1105, 160 kb/s
 Metadata:
 encoder : Lavc58.90.100 libmp3lame