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  • Use, discuss, criticize

    13 avril 2011, par

    Talk to people directly involved in MediaSPIP’s development, or to people around you who could use MediaSPIP to share, enhance or develop their creative projects.
    The bigger the community, the more MediaSPIP’s potential will be explored and the faster the software will evolve.
    A discussion list is available for all exchanges between users.

  • Other interesting software

    13 avril 2011, par

    We don’t claim to be the only ones doing what we do ... and especially not to assert claims to be the best either ... What we do, we just try to do it well and getting better ...
    The following list represents softwares that tend to be more or less as MediaSPIP or that MediaSPIP tries more or less to do the same, whatever ...
    We don’t know them, we didn’t try them, but you can take a peek.
    Videopress
    Website : http://videopress.com/
    License : GNU/GPL v2
    Source code : (...)

  • Menus personnalisés

    14 novembre 2010, par

    MediaSPIP utilise le plugin Menus pour gérer plusieurs menus configurables pour la navigation.
    Cela permet de laisser aux administrateurs de canaux la possibilité de configurer finement ces menus.
    Menus créés à l’initialisation du site
    Par défaut trois menus sont créés automatiquement à l’initialisation du site : Le menu principal ; Identifiant : barrenav ; Ce menu s’insère en général en haut de la page après le bloc d’entête, son identifiant le rend compatible avec les squelettes basés sur Zpip ; (...)

Sur d’autres sites (2461)

  • ffmpeg stream input sdp shows warning keyframe missing

    27 mars 2021, par Doua Beri

    I'm using ffmpeg 4.3.2.
I'm trying to forward a stream to a rtmp server having sdp file as input

    


    Opening the sdp file with VLC everything is working great. The same thing happens when I use ffplay

    


    ffplay -i rtp-forwarder.sdp -protocol_whitelist file,udp,rtp


    


    The problem starts when I start streaming to a rtmp server. The audio is good but the video is just a black screen. I event tried to stream to youtube rtmp server but it didn't work.

    


    I'm new to ffmpeg. Let me know if I'm missing something.

    


    I'm using this command

    


    ffmpeg -protocol_whitelist file,crypto,udp,rtp -re -i rtp-forwarder.sdp -c:v libx264 -b:v 3000k -maxrate 3000k -bufsize 6000k -pix_fmt yuv420p -g 50 -c:a aac -b:a 160k -ac 2 -ar 44100 -f flv rtmp://localhost/live/test


    


    The sdp file content is like this

    


    v=0
o=- 0 0 IN IP4 192.168.1.49
s=Pion WebRTC
c=IN IP4 192.168.1.49
t=0 0
m=audio 4000 RTP/AVP 111
a=rtpmap:111 OPUS/48000/2
m=video 4002 RTP/AVP 96
a=rtpmap:96 VP8/90000


    


    Here is a full log

    


      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
  libswscale      5.  7.100 /  5.  7.100
  libswresample   3.  7.100 /  3.  7.100
  libpostproc    55.  7.100 / 55.  7.100
[sdp @ 0000015a81ede6c0] Keyframe missing
Input #0, sdp, from 'rtp-forwarder.sdp':
  Metadata:
    title           : Pion WebRTC
  Duration: N/A, start: 0.000000, bitrate: N/A
    Stream #0:0: Audio: opus, 48000 Hz, stereo, fltp
    Stream #0:1: Video: vp8, yuv420p(tv, bt470bg/unknown/unknown), 640x480, 90k tbr, 90k tbn, 90k tbc
Stream mapping:
  Stream #0:1 -> #0:0 (vp8 (native) -> h264 (libx264))
  Stream #0:0 -> #0:1 (opus (native) -> aac (native))
Press [q] to stop, [?] for help
[libx264 @ 0000015a81f9e880] MB rate (108000000) > level limit (16711680) -0.0kbits/s speed=N/A
[libx264 @ 0000015a81f9e880] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX FMA3 BMI2 AVX2
[libx264 @ 0000015a81f9e880] profile High, level 6.2, 4:2:0, 8-bit
[libx264 @ 0000015a81f9e880] 264 - core 161 r3048 b86ae3c - H.264/MPEG-4 AVC codec - Copyleft 2003-2021 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=12 lookahead_threads=2 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=50 keyint_min=5 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=cbr mbtree=1 bitrate=3000 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 vbv_maxrate=3000 vbv_bufsize=6000 nal_hrd=none filler=0 ip_ratio=1.40 aq=1:1.00
Output #0, flv, to 'rtmp://localhost/live/test':
  Metadata:
    title           : Pion WebRTC
    encoder         : Lavf58.45.100
    Stream #0:0: Video: h264 (libx264) ([7][0][0][0] / 0x0007), yuv420p, 640x480, q=-1--1, 3000 kb/s, 90k fps, 1k tbn, 90k tbc
    Metadata:
      encoder         : Lavc58.91.100 libx264
    Side data:
      cpb: bitrate max/min/avg: 3000000/0/3000000 buffer size: 6000000 vbv_delay: N/A
    Stream #0:1: Audio: aac (LC) ([10][0][0][0] / 0x000A), 44100 Hz, stereo, fltp, 160 kb/s
    Metadata:
      encoder         : Lavc58.91.100 aac
[flv @ 0000015a820dd200] Failed to update header with correct duration..1kbits/s speed=   1x
[flv @ 0000015a820dd200] Failed to update header with correct filesize.
frame=   13 fps=0.1 q=-1.0 Lsize=    4233kB time=00:03:07.12 bitrate= 185.3kbits/s speed=0.999x
video:440kB audio:3658kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 3.281771%
[libx264 @ 0000015a81f9e880] frame I:1     Avg QP: 1.61  size: 59263
[libx264 @ 0000015a81f9e880] frame P:3     Avg QP: 9.10  size: 44500
[libx264 @ 0000015a81f9e880] frame B:9     Avg QP: 9.80  size: 28605
[libx264 @ 0000015a81f9e880] consecutive B-frames:  7.7%  0.0%  0.0% 92.3%
[libx264 @ 0000015a81f9e880] mb I  I16..4: 36.5% 19.5% 44.0%
[libx264 @ 0000015a81f9e880] mb P  I16..4: 10.7% 50.3% 12.3%  P16..4: 11.6%  9.6%  5.5%  0.0%  0.0%    skip: 0.1%
[libx264 @ 0000015a81f9e880] mb B  I16..4:  2.2% 10.4%  3.6%  B16..8: 45.7% 21.2%  6.1%  direct:10.6%  skip: 0.1%  L0:60.0% L1:25.5% BI:14.5%
[libx264 @ 0000015a81f9e880] final ratefactor: 13.03
[libx264 @ 0000015a81f9e880] 8x8 transform intra:56.7% inter:41.9%
[libx264 @ 0000015a81f9e880] coded y,uvDC,uvAC intra: 90.9% 99.6% 98.9% inter: 66.1% 99.0% 93.2%
[libx264 @ 0000015a81f9e880] i16 v,h,dc,p: 23% 22% 18% 38%
[libx264 @ 0000015a81f9e880] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 31% 26% 23%  4%  2%  3%  3%  4%  4%
[libx264 @ 0000015a81f9e880] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 36% 34% 18%  2%  2%  2%  2%  2%  2%
[libx264 @ 0000015a81f9e880] i8c dc,h,v,p: 55% 29% 11%  5%
[libx264 @ 0000015a81f9e880] Weighted P-Frames: Y:0.0% UV:0.0%
[libx264 @ 0000015a81f9e880] ref P L0: 59.2%  5.0% 13.4% 22.4%
[libx264 @ 0000015a81f9e880] ref B L0: 87.1%  7.5%  5.4%
[libx264 @ 0000015a81f9e880] ref B L1: 95.1%  4.9%
[libx264 @ 0000015a81f9e880] kb/s:4028.74
[aac @ 0000015a81f9a880] Qavg: 7535.380


    


    I get this warning [sdp @ 0000015a81ede6c0] Keyframe missing
and this one after a stop it

    


    [flv @ 0000015a820dd200] Failed to update header with correct duration..1kbits/s speed=   1x
[flv @ 0000015a820dd200] Failed to update header with correct filesize.


    


  • pydub.exceptions.CouldntDecodeError : Couldn't find fmt header in wav data

    26 mai 2021, par Jaswanth

    I am trying to make an audiofile into chunks and converting into text but, pydub is refusing to read my wav file.
Here is the code

    


    #from speakerDiarization import main,fmtTime
from pydub import AudioSegment
import os
from speech_to_text import wav_to_text

meet_audio = 'UK.wav'
out_file = r'test.txt'
#spkrs = main(meet_audio)

spkrs = {0: [{'start': 0, 'stop': 6000}, {'start': 15000, 'stop': 15500}], 
1: [{'start': 6000, 'stop': 11000}, {'start': 15500, 'stop': 18500}, {'start':27500, 'stop': 34500}], 
2: [{'start': 11000, 'stop': 15000}, {'start': 18500, 'stop': 27500}, {'start': 34500, 'stop': 41000}]}

new_dict = {}
for spkr in spkrs:
    for i in range(len(spkrs[spkr])):
        new_dict[spkrs[spkr][i]['start']] = [spkr,i]
new_dict = sorted(new_dict)

audio = AudioSegment(meet_audio)

for i in new_dict:
    spkr,ind = new_dict[i][0],new_dict[i][1]
    start,end = spkrs[spkr][ind]['start'],spkrs[spkr][ind]['stop']
    chunk = audio[start:end]
    chunk_file = 'Chunks\chunk'+str(spkr)+str(ind)+'.wav'
    chunk.export(chunk_file,format='.wav')
    wav_to_text(chunk_file,out_file,spkr)


    


    output :

    


    (sprk-diaz) H:\Btech-Proj\Speaker_Diarization>split_audio.py&#xA;H:\Btech-Proj\Speaker_Diarization\sprk-diaz\lib\site-packages\pydub\utils.py:170: RuntimeWarning: Couldn&#x27;t find ffmpeg or avconv - defaulting to ffmpeg, but may not work&#xA;  warn("Couldn&#x27;t find ffmpeg or avconv - defaulting to ffmpeg, but may not work", RuntimeWarning)&#xA;Traceback (most recent call last):&#xA;  File "H:\Btech-Proj\Speaker_Diarization\split_audio.py", line 20, in <module>&#xA;    audio = AudioSegment(meet_audio)&#xA;  File "H:\Btech-Proj\Speaker_Diarization\sprk-diaz\lib\site-packages\pydub\audio_segment.py", line 222, in __init__&#xA;    wav_data = read_wav_audio(data)&#xA;  File "H:\Btech-Proj\Speaker_Diarization\sprk-diaz\lib\site-packages\pydub\audio_segment.py", line 114, in read_wav_audio&#xA;    raise CouldntDecodeError("Couldn&#x27;t find fmt header in wav data")&#xA;pydub.exceptions.CouldntDecodeError: Couldn&#x27;t find fmt header in wav data&#xA;</module>

    &#xA;

    I don't know what's wrong, can some solve this please.&#xA;Thank you.

    &#xA;

    My audiofile : around 40sec

    &#xA;

  • avformat/rtpenc_rfc4175 : support for interlace format

    1er janvier 2022, par Limin Wang
    avformat/rtpenc_rfc4175 : support for interlace format

    Below are steps how to test on your local host :
    
wget —no-check-certificate https://samples.ffmpeg.org/MPEG2/interlaced/burosch1.mpg
    

    1. interlace format :
    
ffmpeg  -re -i ./burosch1.mpg  -c:v bitpacked  -pix_fmt yuv422p10  -f rtp  rtp ://239.255.0.1:6000
    
copy and create sdp file test.sdp
    
ffplay -buffer_size 671088640 -protocol_whitelist "file,rtp,udp"  test.sdp
    

    2. progressive format :
    
ffmpeg  -re -i ./burosch1.mpg  -vf yadif -c:v bitpacked  -pix_fmt yuv422p10  -f rtp  rtp ://239.255.0.1:6000
    
copy and create sdp file test.sdp
    
ffplay -buffer_size 671088640 -protocol_whitelist "file,rtp,udp"  test.sdp
    

    Signed-off-by : Limin Wang <lance.lmwang@gmail.com>
    • [DH] libavformat/rtpenc.c
    • [DH] libavformat/rtpenc.h
    • [DH] libavformat/rtpenc_rfc4175.c
    • [DH] libavformat/sdp.c

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