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  • Keeping control of your media in your hands

    13 avril 2011, par

    The vocabulary used on this site and around MediaSPIP in general, aims to avoid reference to Web 2.0 and the companies that profit from media-sharing.
    While using MediaSPIP, you are invited to avoid using words like "Brand", "Cloud" and "Market".
    MediaSPIP is designed to facilitate the sharing of creative media online, while allowing authors to retain complete control of their work.
    MediaSPIP aims to be accessible to as many people as possible and development is based on expanding the (...)

  • Participer à sa traduction

    10 avril 2011

    Vous pouvez nous aider à améliorer les locutions utilisées dans le logiciel ou à traduire celui-ci dans n’importe qu’elle nouvelle langue permettant sa diffusion à de nouvelles communautés linguistiques.
    Pour ce faire, on utilise l’interface de traduction de SPIP où l’ensemble des modules de langue de MediaSPIP sont à disposition. ll vous suffit de vous inscrire sur la liste de discussion des traducteurs pour demander plus d’informations.
    Actuellement MediaSPIP n’est disponible qu’en français et (...)

  • Creating farms of unique websites

    13 avril 2011, par

    MediaSPIP platforms can be installed as a farm, with a single "core" hosted on a dedicated server and used by multiple websites.
    This allows (among other things) : implementation costs to be shared between several different projects / individuals rapid deployment of multiple unique sites creation of groups of like-minded sites, making it possible to browse media in a more controlled and selective environment than the major "open" (...)

Sur d’autres sites (20945)

  • Taking care of silent priming frames converting mp4 to ts ?

    13 février 2019, par keepitterron
    videoOutputSettings = [
    
  AVVideoCodecKey: AVVideoCodecH264,
    
  AVVideoWidthKey: width,
    
  AVVideoHeightKey: height,
    
  AVVideoCompressionPropertiesKey: [
    
    AVVideoAverageBitRateKey: avgBitRate,
    
    AVVideoExpectedSourceFrameRateKey: fps,
    
    AVVideoProfileLevelKey: AVVideoProfileLevelH264BaselineAutoLevel
    
  ]
    
]
    
audioOutputSettings = [
    
  AVFormatIDKey: kAudioFormatMPEG4AAC,
    
  AVSampleRateKey: 44100,
    
  AVNumberOfChannelsKey: 2
    
]

    I record a video + audio with AVCaptureSession (screen, camera and mic in OSX) and i encode it with AVAssetWriter (AVFileType.mp4), swapping the writers every 6 seconds.
    Every time a new part is written, a nodeJS app will be notified and using ffmpeg will convert it to ts with

    ffmpeg -y -i generatedFile.mp4 -c copy -copyts -muxdelay 0 -muxpreload 0 outputFile.ts

    I’ll manually write a m3u8 looking like :

    #EXTM3U
    
#EXT-X-VERSION:3
    
#EXT-X-TARGETDURATION:7
    
#EXT-X-MEDIA-SEQUENCE:0
    
#EXT-X-PLAYLIST-TYPE:VOD
    
#EXTINF:6.140227,
    
11214532343c4cae953d45e94a1660ea-0.ts
    
#EXTINF:6.284218,
    
11214532343c4cae953d45e94a1660ea-1.ts
    
#EXTINF:6.099999999999998,
    
11214532343c4cae953d45e94a1660ea-2.ts
    
#EXTINF:6.133333,
    
11214532343c4cae953d45e94a1660ea-3.ts
    
#EXTINF:6.133333,
    
11214532343c4cae953d45e94a1660ea-4.ts
    
#EXTINF:6.100000000000001,
    
11214532343c4cae953d45e94a1660ea-5.ts
    
#EXTINF:6.133333,
    
11214532343c4cae953d45e94a1660ea-6.ts
    
#EXTINF:6.199999999999996,
    
11214532343c4cae953d45e94a1660ea-7.ts
    
#EXTINF:6.0666670000000025,
    
11214532343c4cae953d45e94a1660ea-8.ts
    
#EXT-X-ENDLIST

    The issue is : I have audio gaps between parts and I believe is AAC’s priming frames not being handled correctly.
    Am I correct in assuming so ?

    example file

    ❯ afinfo 11.mp4
    
File:           11.mp4
    
File type ID:   mp4f
    
Num Tracks:     1
    
----
    
Data format:     2 ch,  44100 Hz, 'aac ' (0x00000000) 0 bits/channel, 0 bytes/packet, 1024 frames/packet, 0 bytes/frame
    
                no channel layout.
    
estimated duration: 6.128617 sec
    
audio bytes: 87558
    
audio packets: 266
    
bit rate: 113407 bits per second
    
packet size upper bound: 401
    
maximum packet size: 401
    
audio data file offset: 577163
    
optimized
    
audio 270272 valid frames + 2112 priming + 0 remainder = 272384
    
format list:
    
[ 0] format:      2 ch,  44100 Hz, 'aac ' (0x00000000) 0 bits/channel, 0 bytes/packet, 1024 frames/packet, 0 bytes/frame
    
Channel layout: Stereo (L R)
    
----

    example file

    ❯ ffprobe -v error -show_format -show_streams 1.mp4
    
[STREAM]
    
index=0
    
codec_name=h264
    
codec_long_name=H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10
    
profile=Baseline
    
codec_type=video
    
codec_time_base=1/60
    
codec_tag_string=avc1
    
codec_tag=0x31637661
    
width=1920
    
height=1080
    
coded_width=1920
    
coded_height=1088
    
has_b_frames=0
    
sample_aspect_ratio=1:1
    
display_aspect_ratio=16:9
    
pix_fmt=yuv420p
    
level=40
    
color_range=tv
    
color_space=bt709
    
color_transfer=bt709
    
color_primaries=bt709
    
chroma_location=bottom
    
field_order=unknown
    
timecode=N/A
    
refs=1
    
is_avc=true
    
nal_length_size=4
    
id=N/A
    
r_frame_rate=30/1
    
avg_frame_rate=30/1
    
time_base=1/600
    
start_pts=3600
    
start_time=6.000000
    
duration_ts=7260
    
duration=12.100000
    
bit_rate=4227137
    
max_bit_rate=N/A
    
bits_per_raw_sample=8
    
nb_frames=183
    
nb_read_frames=N/A
    
nb_read_packets=N/A
    
DISPOSITION:default=1
    
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:creation_time=2019-02-13T13:21:51.000000Z
    
TAG:language=und
    
TAG:handler_name=Core Media Video
    
[/STREAM]
    
[STREAM]
    
index=1
    
codec_name=aac
    
codec_long_name=AAC (Advanced Audio Coding)
    
profile=LC
    
codec_type=audio
    
codec_time_base=1/44100
    
codec_tag_string=mp4a
    
codec_tag=0x6134706d
    
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=1/44100
    
start_pts=272991
    
start_time=6.190272
    
duration_ts=545352
    
duration=12.366259
    
bit_rate=115688
    
max_bit_rate=128000
    
bits_per_raw_sample=N/A
    
nb_frames=269
    
nb_read_frames=N/A
    
nb_read_packets=N/A
    
DISPOSITION:default=1
    
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:creation_time=2019-02-13T13:21:51.000000Z
    
TAG:language=und
    
TAG:handler_name=Core Media Audio
    
[/STREAM]
    
[FORMAT]
    
filename=1.mp4
    
nb_streams=2
    
nb_programs=0
    
format_name=mov,mp4,m4a,3gp,3g2,mj2
    
format_long_name=QuickTime / MOV
    
start_time=6.000000
    
duration=12.366259
    
size=3317034
    
bit_rate=2145860
    
probe_score=100
    
TAG:major_brand=mp42
    
TAG:minor_version=1
    
TAG:compatible_brands=mp41mp42isom
    
TAG:creation_time=2019-02-13T13:21:51.000000Z
    
[/FORMAT]

    Is there a way I can generate ts files with the encoder delay being taken care of ?

  • asfdec : remove the wrong condition

    22 juillet 2015, par Alexandra Hájková
    asfdec : remove the wrong condition

    this condition breaks reading from the pipe as data_reached variable
    
have to be set to break while in the asf_read_header just after the Data
    
Object is found
    

    Signed-off-by : Anton Khirnov <anton@khirnov.net>
    • [DBH] libavformat/asfdec.c

  • asfdec : remove the wrong condition

    22 juillet 2015, par Alexandra Hájková
    asfdec : remove the wrong condition

    this condition breaks reading from the pipe as data_reached variable
    
have to be set to break while in the asf_read_header just after the Data
    
Object is found
    

    Signed-off-by : Anton Khirnov <anton@khirnov.net>
    • [DH] libavformat/asfdec.c

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