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Autres articles (108)
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MediaSPIP 0.1 Beta version
25 avril 2011, parMediaSPIP 0.1 beta is the first version of MediaSPIP proclaimed as "usable".
The zip file provided here only contains the sources of MediaSPIP in its standalone version.
To get a working installation, you must manually install all-software dependencies on the server.
If you want to use this archive for an installation in "farm mode", you will also need to proceed to other manual (...) -
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 (...) -
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 (...)
Sur d’autres sites (15965)
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How to save synthesized audio during streaming distribution
7 avril 2022, par its-ogawaI would like to stream a composite of microphone audio and digital sound sources by ffmpeg and save the delivery to an m3u8 file.


Below are the commands I have actually tried.


ffmpeg -rtbufsize 100M -f dshow -i video=<my webcam="webcam">:audio=<my microphone="microphone"> -re -stream_loop -1 -i <my sound="sound" source="source"> -filter_complex "[0]aformat=sample_fmts=fltp:sample_rates=44100:channel_layouts=stereo,adelay=2100|2100,volume@voice=volume=10dB[voice],[1]aformat=sample_fmts=fltp:sample_rates=44100:channel_layouts=stereo,azmq,volume@bgm=volume=0.2[bgm],[voice][bgm]amerge=inputs=2[out]" -map 0:v -map [out]:a -f mpegts -flush_packets 0 udp://XXX.XXX.XXX.XXX:XXX?pkt_size=1316 -f hls -hls_time 5 hls.m3u8
</my></my></my>


I have played an m3u8 created this way, but I cannot hear the digital sound source that I am supposed to have synthesized.
Even though the synthesized audio was audible when streamed.


Perhaps you need to set up something like
-map 0:v -map [out]:a
when saving to the m3u8 file, in which case theOutput with label 'out' does not exist in any defined filter graph, or was already used elsewhere.
message appears and does not work.

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avformat/jvdec : Make sizeof(JVFrame) smaller to save memory
23 septembre 2021, par Andreas Rheinhardt -
convert pcm stream data to encoded aac data
24 décembre 2019, par KumarI tried to convert pulse-audio pcm stream data to aac encoded data using ffmpeg.
But after encoding I get noise-full data, not the correct one. Here I post my code, anyone help me with some ideas.Initial configuration :
av_register_all();
int error;
if ((error = avio_open(&output_io_context,"out.aac",AVIO_FLAG_WRITE))<0) {
printf("could not open output file\n");
}
if (!(output_format_context = avformat_alloc_context())) {
printf("output_format_context error\n");
}
output_format_context->pb = output_io_context;
if(!(output_format_context->oformat = av_guess_format(NULL, "out.aac", NULL))) {
printf("guess format error\n");
}
codec = avcodec_find_encoder(AV_CODEC_ID_AAC);
if (codec == NULL) {
printf("avcodec_find_encoder: ERROR\n");
}
if (!(stream = avformat_new_stream(output_format_context, NULL))) {
printf("stream create error\n");
}
output_codec_context = avcodec_alloc_context3(codec);
if(!output_codec_context) {
printf("output_codec_context is null\n");
}
output_codec_context->channels = CHANNELS;
output_codec_context->channel_layout = av_get_default_channel_layout(CHANNELS);
output_codec_context->sample_rate = SAMPLE_RATE; //input_codec_context->sample_rate;
output_codec_context->sample_fmt = codec->sample_fmts[0];
output_codec_context->bit_rate = 48000; //OUTPUT_BIT_RATE;
stream->time_base.den = SAMPLE_RATE;//input_codec_context->sample_rate;
stream->time_base.num = 1;
if(output_format_context->oformat->flags & AVFMT_GLOBALHEADER)
output_codec_context->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
if ((error = avcodec_open2(output_codec_context, codec, NULL)) < 0) {
printf("error");
}
error = avcodec_parameters_from_context(stream->codecpar, output_codec_context);
if (write_output_file_header(output_format_context)) {
printf("write header failure...\n");
}Data encoding :
AVFrame *output_frame;
int frame_pos = 0, ctx_frame_size = output_codec_context->frame_size;
int size = av_samples_get_buffer_size(NULL, CHANNELS,
output_codec_context->frame_size,output_codec_context->sample_fmt, 1);
if((x = avcodec_fill_audio_frame(output_frame, CHANNELS,
output_codec_context->sample_fmt, data, length, 1)) < 0) {
printf("avcodec_fill_audio_frame error : %s\n", av_err2str(x));
}
int data_written;
if (encode_audio_frame(output_frame, output_format_context,
output_codec_context, &data_written)) {
printf("encode_audio_frame error\n");
}
av_frame_free(&output_frame);helper_function :
int encode_audio_frame(AVFrame *frame,AVFormatContext *output_format_context,
AVCodecContext *output_codec_context, int *data_present)
{
AVPacket output_packet;
int error;
init_packet(&output_packet);
if (frame) {
frame->pts = pts;
pts += frame->nb_samples;
}
error = avcodec_send_frame(output_codec_context, frame);
if (error == AVERROR_EOF) {
error = 0;
goto cleanup;
} else if (error < 0) {
fprintf(stderr, "Could not send packet for encoding (error '%s')\n",
av_err2str(error));
return error;
}
error = avcodec_receive_packet(output_codec_context, &output_packet);
if (error == AVERROR(EAGAIN)) {
error = 0;
goto cleanup;
} else if (error == AVERROR_EOF) {
error = 0;
goto cleanup;
} else if (error < 0) {
fprintf(stderr, "Could not encode frame (error '%s')\n",
av_err2str(error));
goto cleanup;
} else {
*data_present = 1;
}
if (*data_present &&
(error = av_write_frame(output_format_context, &output_packet)) < 0) {
fprintf(stderr, "Could not write frame (error '%s')\n",
av_err2str(error));
goto cleanup;
}
cleanup:
av_packet_unref(&output_packet);
return error;
}- Do we need to fill AVFrame with sizeof(av_samples_get_buffer_size) or context->frame_size ?
TYIA :) !!
- Do we need to fill AVFrame with sizeof(av_samples_get_buffer_size) or context->frame_size ?