Recherche avancée

Médias (16)

Mot : - Tags -/mp3

Autres articles (68)

  • Emballe médias : à quoi cela sert ?

    4 février 2011, par

    Ce plugin vise à gérer des sites de mise en ligne de documents de tous types.
    Il crée des "médias", à savoir : un "média" est un article au sens SPIP créé automatiquement lors du téléversement d’un document qu’il soit audio, vidéo, image ou textuel ; un seul document ne peut être lié à un article dit "média" ;

  • Le plugin : Gestion de la mutualisation

    2 mars 2010, par

    Le plugin de Gestion de mutualisation permet de gérer les différents canaux de mediaspip depuis un site maître. Il a pour but de fournir une solution pure SPIP afin de remplacer cette ancienne solution.
    Installation basique
    On installe les fichiers de SPIP sur le serveur.
    On ajoute ensuite le plugin "mutualisation" à la racine du site comme décrit ici.
    On customise le fichier mes_options.php central comme on le souhaite. Voilà pour l’exemple celui de la plateforme mediaspip.net :
    < ?php (...)

  • Gestion de la ferme

    2 mars 2010, par

    La ferme est gérée dans son ensemble par des "super admins".
    Certains réglages peuvent être fais afin de réguler les besoins des différents canaux.
    Dans un premier temps il utilise le plugin "Gestion de mutualisation"

Sur d’autres sites (8945)

  • ppc : Clarify and extend the cpuid check

    10 mai 2015, par Luca Barbato
    ppc : Clarify and extend the cpuid check
    

    Add POWER entries.

    • [DBH] libavutil/ppc/cpu.c
  • gstreamer android error : engine.GetInterface(IODeviceCapabilities) failed(0x0000000c)

    19 juin 2015, par omerjerk

    I’m trying to stream mpegts file on an Android device. This is the file.

    I’ve setup everything as done in the tutorial 5 of gstreamer android.
    But I’m getting the following error -

    06-18 23:56:49.023  27051-27132/kuchbhilabs.chestream D/GStreamer+tutorial-4﹕ 0:00:03.720735780 0xac093980 src/main/jni/main.c:104:set_ui_message Setting message to: Buffering 66%
    06-18 23:56:49.023  27051-27132/kuchbhilabs.chestream D/GStreamerPlayer﹕ Buffering 66%
    06-18 23:56:49.551  27051-27350/kuchbhilabs.chestream W/GStreamer+mpegtsbase﹕ 0:00:04.248340259 0xaec6d6f0 mpegtsbase.c:1379:query_upstream_latency:<tsdemux0> Failed to query upstream latency
    06-18 23:56:49.555  27051-27350/kuchbhilabs.chestream W/GStreamer+mpegtsbase﹕ 0:00:04.252435571 0xaec6d6f0 mpegtsbase.c:531:mpegts_base_program_add_stream Stream already present !
    06-18 23:56:49.762  27051-27351/kuchbhilabs.chestream W/libOpenSLES﹕ Leaving Object::GetInterface (SL_RESULT_FEATURE_UNSUPPORTED)
    06-18 23:56:49.762  27051-27351/kuchbhilabs.chestream E/GStreamer+opensles_sink﹕ 0:00:04.459898644 0xaec6e400 openslessink.c:152:_opensles_query_capabilities:&lt;gstopenslessink@0xac3d8408&gt; engine.GetInterface(IODeviceCapabilities) failed(0x0000000c)
    06-18 23:56:49.763  27051-27351/kuchbhilabs.chestream W/libOpenSLES﹕ Leaving OutputMix::GetDestinationOutputDeviceIDs (SL_RESULT_BUFFER_INSUFFICIENT)
    06-18 23:56:49.810  27051-27351/kuchbhilabs.chestream E/GStreamer+ffmpeg﹕ 0:00:04.507782029 0xaec6e400 :0: channel element 0.0 is not allocated
    06-18 23:56:49.810  27051-27351/kuchbhilabs.chestream W/GStreamer+ffmpeg﹕ 0:00:04.507919477 0xaec6e400 gstffmpegdec.c:2231:gst_ffmpegdec_audio_frame: error: Decoding of AAC stream by FFMPEG failed.
    06-18 23:56:49.811  27051-27351/kuchbhilabs.chestream W/GStreamer+ffmpeg﹕ 0:00:04.508238122 0xaec6e400 gstffmpegdec.c:2315:gst_ffmpegdec_frame: ffdec_aac: decoding error (len: -1, have_data: 0)
    06-18 23:56:49.811  27051-27132/kuchbhilabs.chestream D/GStreamer+tutorial-4﹕ 0:00:04.508278070 0xac093980 src/main/jni/main.c:104:set_ui_message Setting message to: Error received from element ffdec_aac0: Could not decode stream.
    06-18 23:56:49.811  27051-27132/kuchbhilabs.chestream D/GStreamerPlayer﹕ Error received from element ffdec_aac0: Could not decode stream.
    06-18 23:56:49.878  27051-27352/kuchbhilabs.chestream W/GStreamer+ffmpeg﹕ 0:00:04.575771665 0xaec6e460 gstffmpegdec.c:2315:gst_ffmpegdec_frame: ffdec_h264: decoding error (len: -1, have_data: 0)
    </tsdemux0>

    This is my Android.mk file :

    LOCAL_PATH := $(call my-dir)

    include $(CLEAR_VARS)

    LOCAL_MODULE    := chestream
    LOCAL_SRC_FILES := main.c
    LOCAL_SHARED_LIBRARIES := gstreamer_android
    LOCAL_LDLIBS := -llog -landroid
    include $(BUILD_SHARED_LIBRARY)

    ifndef GSTREAMER_SDK_ROOT
    ifndef GSTREAMER_SDK_ROOT_ANDROID
    $(error GSTREAMER_SDK_ROOT_ANDROID is not defined!)
    endif
    GSTREAMER_SDK_ROOT        := $(GSTREAMER_SDK_ROOT_ANDROID)
    endif
    GSTREAMER_NDK_BUILD_PATH  := $(GSTREAMER_SDK_ROOT)/share/gst-android/ndk-build/
    include $(GSTREAMER_NDK_BUILD_PATH)/plugins.mk
    GSTREAMER_PLUGINS         := $(GSTREAMER_PLUGINS_CORE) $(GSTREAMER_PLUGINS_PLAYBACK) $(GSTREAMER_PLUGINS_CODECS) $(GSTREAMER_PLUGINS_NET) $(GSTREAMER_PLUGINS_SYS)\
    mpeg2dec mpegdemux2 mpegtsdemux mpegtsmux ffmpeg
    GSTREAMER_EXTRA_DEPS      := gstreamer-interfaces-0.10 gstreamer-video-0.10
    include $(GSTREAMER_NDK_BUILD_PATH)/gstreamer.mk

    The error is probably that gstreamer is not able to decode AAC file.
    Any help would be appreciated.

  • Encode live audio stream with ffmpeg

    11 novembre 2015, par Victor Canezin de Oliveira

    Someone please help me.

    I’m trying to encode a live audio stream with ffmpeg. I’m using it Chrome’s native client.

    It’s getting the audio from GetUserMedia() and sending it to pnacl. It receives the audio and send it to ffmpeg to encode. The problem is the final audio is chopped.

    I think the problem is that the codec frame_size is larger than the stream nb_samples (1152 and 480 respectively) and it’s leaving a "blank" audio data at the end of each frame (I’m no audio expert).

    I am creating the AVFrame like this :

    AVFrame *frame = av_frame_alloc();
    frame->nb_samples = nb_samples; //(I get it from the stream)
    AVSampleFormat sample_fmt = ost->st->codec->sample_fmt;
    // nb_channels, buffer and buffer_size I get from the stream

    avcodec_fill_audio_frame(frame, nb_channels, sample_fmt, buffer, buffer_size, 0);

    And encoding with avcodec_encode_audio2

    I am using a mix of nacl example and muxing.c example from ffmpeg.

    Can someone help me ?