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Somos millones 1
21 juillet 2014, par
Mis à jour : Juin 2015
Langue : français
Type : Video
Autres articles (101)
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Amélioration de la version de base
13 septembre 2013Jolie sélection multiple
Le plugin Chosen permet d’améliorer l’ergonomie des champs de sélection multiple. Voir les deux images suivantes pour comparer.
Il suffit pour cela d’activer le plugin Chosen (Configuration générale du site > Gestion des plugins), puis de configurer le plugin (Les squelettes > Chosen) en activant l’utilisation de Chosen dans le site public et en spécifiant les éléments de formulaires à améliorer, par exemple select[multiple] pour les listes à sélection multiple (...) -
MediaSPIP v0.2
21 juin 2013, parMediaSPIP 0.2 est la première version de MediaSPIP stable.
Sa date de sortie officielle est le 21 juin 2013 et est annoncée ici.
Le fichier zip ici présent contient uniquement les sources de MediaSPIP en version standalone.
Comme pour la version précédente, il est nécessaire d’installer manuellement l’ensemble des dépendances logicielles sur le serveur.
Si vous souhaitez utiliser cette archive pour une installation en mode ferme, il vous faudra également procéder à d’autres modifications (...) -
MediaSPIP version 0.1 Beta
16 avril 2011, parMediaSPIP 0.1 beta est la première version de MediaSPIP décrétée comme "utilisable".
Le fichier zip ici présent contient uniquement les sources de MediaSPIP en version standalone.
Pour avoir une installation fonctionnelle, il est nécessaire d’installer manuellement l’ensemble des dépendances logicielles sur le serveur.
Si vous souhaitez utiliser cette archive pour une installation en mode ferme, il vous faudra également procéder à d’autres modifications (...)
Sur d’autres sites (11251)
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Revert "avformat/utils : Close codec context since it is allocated by avformat_new_str...
21 juillet 2013, par Michael NiedermayerRevert "avformat/utils : Close codec context since it is allocated by avformat_new_stream in refrence to ticket 2716"
This causes a race condition with VLC. Its plausible that other
applications also would have races with it and its just fixing a memleak when
the user application forgets to free the codec. It causes more
problems than it solves in its current form, thus the revert.
Better solutions are welcomeThis reverts commit 0f229f9b91fe3c4ebd97e88f004d638819b76add.
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Muxing with libav
14 février 2014, par LordDoskiasI have a program which is supposed to demux input mpeg-ts, transcode the mpeg2 into h264 and then mux the audio alongside the transcoded video. When I open the resulting muxed file with VLC I neither get audio nor video. Here is the relevant code.
My main worker loop is as follows :
void
*writer_thread(void *thread_ctx) {
struct transcoder_ctx_t *ctx = (struct transcoder_ctx_t *) thread_ctx;
AVStream *video_stream = NULL, *audio_stream = NULL;
AVFormatContext *output_context = init_output_context(ctx, &video_stream, &audio_stream);
struct mux_state_t mux_state = {0};
//from omxtx
mux_state.pts_offset = av_rescale_q(ctx->input_context->start_time, AV_TIME_BASE_Q, output_context->streams[ctx->video_stream_index]->time_base);
//write stream header if any
avformat_write_header(output_context, NULL);
//do not start doing anything until we get an encoded packet
pthread_mutex_lock(&ctx->pipeline.video_encode.is_running_mutex);
while (!ctx->pipeline.video_encode.is_running) {
pthread_cond_wait(&ctx->pipeline.video_encode.is_running_cv, &ctx->pipeline.video_encode.is_running_mutex);
}
while (!ctx->pipeline.video_encode.eos || !ctx->processed_audio_queue->queue_finished) {
//FIXME a memory barrier is required here so that we don't race
//on above variables
//fill a buffer with video data
OERR(OMX_FillThisBuffer(ctx->pipeline.video_encode.h, omx_get_next_output_buffer(&ctx->pipeline.video_encode)));
write_audio_frame(output_context, audio_stream, ctx); //write full audio frame
//FIXME no guarantee that we have a full frame per packet?
write_video_frame(output_context, video_stream, ctx, &mux_state); //write full video frame
//encoded_video_queue is being filled by the previous command
}
av_write_trailer(output_context);
//free all the resources
avcodec_close(video_stream->codec);
avcodec_close(audio_stream->codec);
/* Free the streams. */
for (int i = 0; i < output_context->nb_streams; i++) {
av_freep(&output_context->streams[i]->codec);
av_freep(&output_context->streams[i]);
}
if (!(output_context->oformat->flags & AVFMT_NOFILE)) {
/* Close the output file. */
avio_close(output_context->pb);
}
/* free the stream */
av_free(output_context);
free(mux_state.pps);
free(mux_state.sps);
}The code for initialising libav output context is this :
static
AVFormatContext *
init_output_context(const struct transcoder_ctx_t *ctx, AVStream **video_stream, AVStream **audio_stream) {
AVFormatContext *oc;
AVOutputFormat *fmt;
AVStream *input_stream, *output_stream;
AVCodec *c;
AVCodecContext *cc;
int audio_copied = 0; //copy just 1 stream
fmt = av_guess_format("mpegts", NULL, NULL);
if (!fmt) {
fprintf(stderr, "[DEBUG] Error guessing format, dying\n");
exit(199);
}
oc = avformat_alloc_context();
if (!oc) {
fprintf(stderr, "[DEBUG] Error allocating context, dying\n");
exit(200);
}
oc->oformat = fmt;
snprintf(oc->filename, sizeof(oc->filename), "%s", ctx->output_filename);
oc->debug = 1;
oc->start_time_realtime = ctx->input_context->start_time;
oc->start_time = ctx->input_context->start_time;
oc->duration = 0;
oc->bit_rate = 0;
for (int i = 0; i < ctx->input_context->nb_streams; i++) {
input_stream = ctx->input_context->streams[i];
output_stream = NULL;
if (input_stream->index == ctx->video_stream_index) {
//copy stuff from input video index
c = avcodec_find_encoder(CODEC_ID_H264);
output_stream = avformat_new_stream(oc, c);
*video_stream = output_stream;
cc = output_stream->codec;
cc->width = input_stream->codec->width;
cc->height = input_stream->codec->height;
cc->codec_id = CODEC_ID_H264;
cc->codec_type = AVMEDIA_TYPE_VIDEO;
cc->bit_rate = ENCODED_BITRATE;
cc->time_base = input_stream->codec->time_base;
output_stream->avg_frame_rate = input_stream->avg_frame_rate;
output_stream->r_frame_rate = input_stream->r_frame_rate;
output_stream->start_time = AV_NOPTS_VALUE;
} else if ((input_stream->codec->codec_type == AVMEDIA_TYPE_AUDIO) && !audio_copied) {
/* i care only about audio */
c = avcodec_find_encoder(input_stream->codec->codec_id);
output_stream = avformat_new_stream(oc, c);
*audio_stream = output_stream;
avcodec_copy_context(output_stream->codec, input_stream->codec);
/* Apparently fixes a crash on .mkvs with attachments: */
av_dict_copy(&output_stream->metadata, input_stream->metadata, 0);
/* Reset the codec tag so as not to cause problems with output format */
output_stream->codec->codec_tag = 0;
audio_copied = 1;
}
}
for (int i = 0; i < oc->nb_streams; i++) {
if (oc->oformat->flags & AVFMT_GLOBALHEADER)
oc->streams[i]->codec->flags |= CODEC_FLAG_GLOBAL_HEADER;
if (oc->streams[i]->codec->sample_rate == 0)
oc->streams[i]->codec->sample_rate = 48000; /* ish */
}
if (!(fmt->flags & AVFMT_NOFILE)) {
fprintf(stderr, "[DEBUG] AVFMT_NOFILE set, allocating output container\n");
if (avio_open(&oc->pb, ctx->output_filename, AVIO_FLAG_WRITE) < 0) {
fprintf(stderr, "[DEBUG] error creating the output context\n");
exit(1);
}
}
return oc;
}Finally this is the code for writing audio :
static
void
write_audio_frame(AVFormatContext *oc, AVStream *st, struct transcoder_ctx_t *ctx) {
AVPacket pkt = {0}; // data and size must be 0;
struct packet_t *source_audio;
av_init_packet(&pkt);
if (!(source_audio = packet_queue_get_next_item_asynch(ctx->processed_audio_queue))) {
return;
}
pkt.stream_index = st->index;
pkt.size = source_audio->data_length;
pkt.data = source_audio->data;
pkt.pts = source_audio->PTS;
pkt.dts = source_audio->DTS;
pkt.duration = source_audio->duration;
pkt.destruct = avpacket_destruct;
/* Write the compressed frame to the media file. */
if (av_interleaved_write_frame(oc, &pkt) != 0) {
fprintf(stderr, "[DEBUG] Error while writing audio frame\n");
}
packet_queue_free_packet(source_audio, 0);
}A resulting mpeg4 file can be obtained from here :
http://87.120.131.41/dl/mpeg4.h264
I have ommited the write_video_frame code since it is a lot more complicated and I might be making something wrong there as I'm doing timebase conversation etc. For audio however I'm doing 1:1 copy. Each packet_t packet contains data from av_read_frame from the input mpegts container. In the worst case I'd expect that my audio is working and not my video. However I cannot get either of those to work. Seems the documentation is rather vague on making things like that - I've tried both libav and ffmpeg irc channels to no avail. Any information regarding how I can debug the issue will be greatly appreciated.
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avcodec/tdsc : use ff_codec_open2_recursive()
13 mars 2015, par Michael Niedermayer