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Médias (1)
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Video d’abeille en portrait
14 mai 2011, par
Mis à jour : Février 2012
Langue : français
Type : Video
Autres articles (56)
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Participer à sa traduction
10 avril 2011Vous 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 (...) -
Publier sur MédiaSpip
13 juin 2013Puis-je poster des contenus à partir d’une tablette Ipad ?
Oui, si votre Médiaspip installé est à la version 0.2 ou supérieure. Contacter au besoin l’administrateur de votre MédiaSpip pour le savoir -
Supporting all media types
13 avril 2011, parUnlike most software and media-sharing platforms, MediaSPIP aims to manage as many different media types as possible. The following are just a few examples from an ever-expanding list of supported formats : images : png, gif, jpg, bmp and more audio : MP3, Ogg, Wav and more video : AVI, MP4, OGV, mpg, mov, wmv and more text, code and other data : OpenOffice, Microsoft Office (Word, PowerPoint, Excel), web (html, CSS), LaTeX, Google Earth and (...)
Sur d’autres sites (7629)
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how to convert an MPEGTS file into an FLV file by programming with libavcodec
9 mai 2014, par Hexing BI wanna convert an MPEGTS file into an FLV file with libavcodec APIs(just transformat, video/audio codecs are not changed). Following is the code I found on web. After some hack, it can generate an FLV file, which could not be played. Any clue to fix this ?
#include
#include
#include
#include
#include <libavformat></libavformat>avformat.h>
#include <libavcodec></libavcodec>avcodec.h>
#include <libavutil></libavutil>avutil.h>
#include <libavutil></libavutil>rational.h>
#include <libavdevice></libavdevice>avdevice.h>
#include <libavutil></libavutil>mathematics.h>
#include <libswscale></libswscale>swscale.h>
static AVStream* add_output_stream(AVFormatContext* output_format_context, AVStream* input_stream) {
AVCodecContext* input_codec_context = NULL;
AVCodecContext* output_codec_context = NULL;
AVStream* output_stream = NULL;
output_stream = avformat_new_stream(output_format_context, 0);
if (!output_stream) {
printf("Call av_new_stream function failed\n");
return NULL;
}
input_codec_context = input_stream->codec;
output_codec_context = output_stream->codec;
output_codec_context->codec_id = input_codec_context->codec_id;
output_codec_context->codec_type = input_codec_context->codec_type;
// output_codec_context->codec_tag = input_codec_context->codec_tag;
output_codec_context->codec_tag = av_codec_get_tag(output_format_context->oformat->codec_tag, input_codec_context->codec_id);
output_codec_context->bit_rate = input_codec_context->bit_rate;
output_codec_context->extradata = input_codec_context->extradata;
output_codec_context->extradata_size = input_codec_context->extradata_size;
if (av_q2d(input_codec_context->time_base) * input_codec_context->ticks_per_frame > av_q2d(input_stream->time_base) && av_q2d(input_stream->time_base) < 1.0 / 1000) {
output_codec_context->time_base = input_codec_context->time_base;
output_codec_context->time_base.num *= input_codec_context->ticks_per_frame;
} else {
output_codec_context->time_base = input_stream->time_base;
}
switch (input_codec_context->codec_type) {
case AVMEDIA_TYPE_AUDIO:
output_codec_context->channel_layout = input_codec_context->channel_layout;
output_codec_context->sample_rate = input_codec_context->sample_rate;
output_codec_context->channels = input_codec_context->channels;
output_codec_context->frame_size = input_codec_context->frame_size;
if ((input_codec_context->block_align == 1 && input_codec_context->codec_id == CODEC_ID_MP3) || input_codec_context->codec_id == CODEC_ID_AC3) {
output_codec_context->block_align = 0;
} else {
output_codec_context->block_align = input_codec_context->block_align;
}
break;
case AVMEDIA_TYPE_VIDEO:
output_codec_context->pix_fmt = input_codec_context->pix_fmt;
output_codec_context->width = input_codec_context->width;
output_codec_context->height = input_codec_context->height;
output_codec_context->has_b_frames = input_codec_context->has_b_frames;
if (output_format_context->oformat->flags & AVFMT_GLOBALHEADER) {
output_codec_context->flags |= CODEC_FLAG_GLOBAL_HEADER;
}
// output_codec_context->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
break;
default:
break;
}
return output_stream;
}
int main(int argc, char* argv[]) {
const char* input = argv[1];
const char* output_prefix = NULL;
char* segment_duration_check = 0;
const char* index = NULL;
char* tmp_index = NULL;
const char* http_prefix = NULL;
long max_tsfiles = 0;
double prev_segment_time = 0;
double segment_duration = 0;
AVInputFormat* ifmt = NULL;
AVOutputFormat* ofmt = NULL;
AVFormatContext* ic = NULL;
AVFormatContext* oc = NULL;
AVStream* video_st = NULL;
AVStream* audio_st = NULL;
AVCodec* codec = NULL;
AVDictionary* pAVDictionary = NULL;
av_register_all();
av_log_set_level(AV_LOG_DEBUG);
char szError[256] = {0};
int nRet = avformat_open_input(&ic, input, ifmt, &pAVDictionary);
if (nRet != 0) {
av_strerror(nRet, szError, 256);
printf(szError);
printf("\n");
printf("Call avformat_open_input function failed!\n");
return 0;
}
if (avformat_find_stream_info(ic, NULL) < 0) {
printf("Call av_find_stream_info function failed!\n");
return 0;
}
ofmt = av_guess_format(NULL, argv[2], NULL);
if (!ofmt) {
printf("Call av_guess_format function failed!\n");
return 0;
}
oc = avformat_alloc_context();
if (!oc) {
printf("Call av_guess_format function failed!\n");
return 0;
}
oc->oformat = ofmt;
int video_index = -1, audio_index = -1;
unsigned int i;
for (i = 0; i < ic->nb_streams && (video_index < 0 || audio_index < 0); i++) {
switch (ic->streams[i]->codec->codec_type) {
case AVMEDIA_TYPE_VIDEO:
video_index = i;
ic->streams[i]->discard = AVDISCARD_NONE;
video_st = add_output_stream(oc, ic->streams[i]);
if (video_st->codec->codec_id == CODEC_ID_H264) {
video_st->codec->opaque = av_bitstream_filter_init("h264_mp4toannexb");
}
break;
case AVMEDIA_TYPE_AUDIO:
audio_index = i;
ic->streams[i]->discard = AVDISCARD_NONE;
audio_st = add_output_stream(oc, ic->streams[i]);
if (audio_st->codec->codec_id == CODEC_ID_AAC && !audio_st->codec->extradata_size) {
audio_st->codec->opaque = av_bitstream_filter_init("aac_adtstoasc");
}
break;
default:
ic->streams[i]->discard = AVDISCARD_ALL;
break;
}
}
codec = avcodec_find_decoder(video_st->codec->codec_id);
if (codec == NULL) {
printf("Call avcodec_find_decoder function failed!\n");
return 0;
}
if (avcodec_open2(video_st->codec, codec, NULL) < 0) {
printf("Call avcodec_open function failed !\n");
return 0;
}
if (avio_open(&oc->pb, argv[2], AVIO_FLAG_WRITE) < 0) {
return 0;
}
if (avformat_write_header(oc, &pAVDictionary)) {
printf("Call avformat_write_header function failed.\n");
return 0;
}
int decode_done = 0;
do {
AVPacket packet;
decode_done = av_read_frame(ic, &packet);
if (decode_done < 0) {
break;
}
if (packet.stream_index == video_index && (packet.flags & AV_PKT_FLAG_KEY) && video_st->codec->opaque != NULL) {
AVPacket pkt = packet;
int a = av_bitstream_filter_filter(video_st->codec->opaque, video_st->codec, NULL, &pkt.data, &pkt.size,
packet.data, packet.size, packet.flags & AV_PKT_FLAG_KEY);
if (a == 0) {
memmove(packet.data, pkt.data, pkt.size);
packet.size = pkt.size;
} else if (a > 0) {
packet = pkt;
}
}
else if (packet.stream_index == audio_index && audio_st->codec->opaque != NULL) {
AVPacket pkt = packet;
int a = av_bitstream_filter_filter(audio_st->codec->opaque, audio_st->codec, NULL, &pkt.data, &pkt.size,
packet.data, packet.size, packet.flags & AV_PKT_FLAG_KEY);
if (a == 0) {
memmove(packet.data, pkt.data, pkt.size);
packet.size = pkt.size;
} else if (a > 0) {
packet = pkt;
}
}
nRet = av_interleaved_write_frame(oc, &packet);
if (nRet < 0) {
printf("Call av_interleaved_write_frame function failed\n");
} else if (nRet > 0) {
printf("End of stream requested\n");
av_free_packet(&packet);
break;
}
av_free_packet(&packet);
} while(!decode_done);
av_write_trailer(oc);
av_bitstream_filter_close(video_st->codec->opaque);
av_bitstream_filter_close(audio_st->codec->opaque);
avcodec_close(video_st->codec);
unsigned int k;
for(k = 0; k < oc->nb_streams; k++) {
av_freep(&oc->streams[k]->codec);
av_freep(&oc->streams[k]);
}
av_free(oc);
getchar();
return 0;
} -
Qt Video Recorder
11 mai 2014, par DavlogI am trying to create a video recorder with Qt. What I did so far was taking a screenshot of a rectangle on the screen and save it. At the end I use ffmpeg to get a video file out of the images.
I connected a timer’s signal
timeout()to my custom slot which takes the snapshot and saves it to my tmp folder. The timer has an intervall of 1000 / 30. That should be 30 times per second. But 1000 / 30 is a little bit more than 33 milliseconds so I cannot really get 30 fps. It’s a bit more.I recorded a youtube video with my recorder and everything was smooth but a little bit faster / slower depending on the intervall.
So my question basically is how do I get really 30 / 40 / 50 / ... fps ?
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Correct command to transmit audio to ip camera using ffmpeg ?
4 novembre 2016, par the_naiveSo I found some hints in this discussion on the correct command to transmit audio to Axis IP camera through using ffmpeg in windows, but still I have not managed to successfully transmit audio to the camera.
The command I’m using is the following :
ffmpeg -v debug -y -re -f dshow -i "audio=Microphone (2- High Definition Audio Device)" -c:a pcm_mulaw -ac 1 -ar 16000 -b:a 128k -f flv http://oper
ator:operator@10.10.210.2/axis-cgi/audio/transmit.cgi -multiple_requests 1 -reconnect_at_eof 1 -reconnect_streamed 1 -content_type "audio/basic" -reportThe ouput I get following this command is the following :
ffmpeg started on 2016-11-04 at 17:32:13
Report written to "ffmpeg-20161104-173213.log"
Command line:
ffmpeg -v debug -y -re -f dshow -i "audio=Microphone (2- High Definition Audio Device)" -c:a pcm_mulaw -ac 1 -ar 16000 -b:a 128k -f flv http://operator:operator@10.10.210.2/axis-cgi/audio/transmit.cgi -content_type audio/basic -multiple_requests 1 -reconnect 1 -reconnect_at_eof 1 -reconnect_streamed 1 -report
ffmpeg version N-82225-gb4e9252 Copyright (c) 2000-2016 the FFmpeg developers
built with gcc 5.4.0 (GCC)
configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-dxva2 --enable-libmfx --enable-nvenc --enable-avisynth --enable-bzlib --enable-libebur128 --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-decklink --enable-zlib
libavutil 55. 35.100 / 55. 35.100
libavcodec 57. 66.101 / 57. 66.101
libavformat 57. 57.100 / 57. 57.100
libavdevice 57. 2.100 / 57. 2.100
libavfilter 6. 66.100 / 6. 66.100
libswscale 4. 3.100 / 4. 3.100
libswresample 2. 4.100 / 2. 4.100
libpostproc 54. 2.100 / 54. 2.100
Splitting the commandline.
Reading option '-v' ... matched as option 'v' (set logging level) with argument 'debug'.
Reading option '-y' ... matched as option 'y' (overwrite output files) with argument '1'.
Reading option '-re' ... matched as option 're' (read input at native frame rate) with argument '1'.
Reading option '-f' ... matched as option 'f' (force format) with argument 'dshow'.
Reading option '-i' ... matched as input file with argument 'audio=Microphone (2- High Definition Audio Device)'.
Reading option '-c:a' ... matched as option 'c' (codec name) with argument 'pcm_mulaw'.
Reading option '-ac' ... matched as option 'ac' (set number of audio channels) with argument '1'.
Reading option '-ar' ... matched as option 'ar' (set audio sampling rate (in Hz)) with argument '16000'.
Reading option '-b:a' ... matched as option 'b' (video bitrate (please use -b:v)) with argument '128k'.
Reading option '-f' ... matched as option 'f' (force format) with argument 'flv'.
Reading option 'http://operator:operator@10.10.210.2/axis-cgi/audio/transmit.cgi' ... matched as output file.
Reading option '-content_type' ... matched as AVOption 'content_type' with argument 'audio/basic'.
Reading option '-multiple_requests' ... matched as AVOption 'multiple_requests' with argument '1'.
Reading option '-reconnect' ... matched as AVOption 'reconnect' with argument '1'.
Reading option '-reconnect_at_eof' ... matched as AVOption 'reconnect_at_eof' with argument '1'.
Reading option '-reconnect_streamed' ... matched as AVOption 'reconnect_streamed' with argument '1'.
Reading option '-report' ... matched as option 'report' (generate a report) with argument '1'.
Trailing options were found on the commandline.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option v (set logging level) with argument debug.
Applying option y (overwrite output files) with argument 1.
Applying option report (generate a report) with argument 1.
Successfully parsed a group of options.
Parsing a group of options: input file audio=Microphone (2- High Definition Audio Device).
Applying option re (read input at native frame rate) with argument 1.
Applying option f (force format) with argument dshow.
Successfully parsed a group of options.
Opening an input file: audio=Microphone (2- High Definition Audio Device).
[dshow @ 00000000000279e0] Selecting pin Capture on audio only
dshow passing through packet of type audio size 88200 timestamp 310221040000 orig timestamp 310221040000 graph timestamp 310226130000 diff 5090000 Microphone (2- High Definition Audio Device)
[dshow @ 00000000000279e0] All info found
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, dshow, from 'audio=Microphone (2- High Definition Audio Device)':
Duration: N/A, start: 31022.104000, bitrate: 1411 kb/s
Stream #0:0, 1, 1/10000000: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s
Successfully opened the file.
Parsing a group of options: output file http://operator:operator@10.10.210.2/axis-cgi/audio/transmit.cgi.
Applying option c:a (codec name) with argument pcm_mulaw.
Applying option ac (set number of audio channels) with argument 1.
Applying option ar (set audio sampling rate (in Hz)) with argument 16000.
Applying option b:a (video bitrate (please use -b:v)) with argument 128k.
Applying option f (force format) with argument flv.
Successfully parsed a group of options.
Opening an output file: http://operator:operator@10.10.210.2/axis-cgi/audio/transmit.cgi.
[http @ 0000000001c94040] Setting default whitelist 'http,https,tls,rtp,tcp,udp,crypto,httpproxy'
[http @ 0000000001c94040] request: POST /axis-cgi/audio/transmit.cgi HTTP/1.1
Transfer-Encoding: chunked
User-Agent: Lavf/57.57.100
Accept: */*
Expect: 100-continue
Connection: close
Host: 10.10.210.2
Icy-MetaData: 1
[http @ 0000000001c94040] request: POST /axis-cgi/audio/transmit.cgi HTTP/1.1
Transfer-Encoding: chunked
User-Agent: Lavf/57.57.100
Accept: */*
Connection: close
Host: 10.10.210.2
Icy-MetaData: 1
Authorization: Digest username="operator", realm="AXIS_ACCC8E027F47", nonce="0EcsO3xABQA=ab5efc4740a6c625ecf6a6729d0d67d2b62b615a", uri="/axis-cgi/audio/transmit.cgi", response="4bd3a627b20d6bcaba9e2f595ef6cd2a", algorithm="MD5", qop="auth", cnonce="6a579dd6664b57eb", nc=00000001
Successfully opened the file.
detected 8 logical cores
[graph 0 input from stream 0:0 @ 0000000001c9f6e0] Setting 'time_base' to value '1/44100'
[graph 0 input from stream 0:0 @ 0000000001c9f6e0] Setting 'sample_rate' to value '44100'
[graph 0 input from stream 0:0 @ 0000000001c9f6e0] Setting 'sample_fmt' to value 's16'
[graph 0 input from stream 0:0 @ 0000000001c9f6e0] Setting 'channel_layout' to value '0x3'
[graph 0 input from stream 0:0 @ 0000000001c9f6e0] tb:1/44100 samplefmt:s16 samplerate:44100 chlayout:0x3
[audio format for output stream 0:0 @ 0000000001c9fa20] Setting 'sample_fmts' to value 's16'
[audio format for output stream 0:0 @ 0000000001c9fa20] Setting 'sample_rates' to value '16000'
[audio format for output stream 0:0 @ 0000000001c9fa20] Setting 'channel_layouts' to value '0x4'
[audio format for output stream 0:0 @ 0000000001c9fa20] auto-inserting filter 'auto-inserted resampler 0' between the filter 'Parsed_anull_0' and the filter 'audio format for output stream 0:0'
[AVFilterGraph @ 000000000002ab20] query_formats: 4 queried, 6 merged, 3 already done, 0 delayed
[auto-inserted resampler 0 @ 0000000001ca4060] [SWR @ 0000000001ca4a80] Using s16p internally between filters
[auto-inserted resampler 0 @ 0000000001ca4060] [SWR @ 0000000001ca4a80] Matrix coefficients:
[auto-inserted resampler 0 @ 0000000001ca4060] [SWR @ 0000000001ca4a80] FC: FL:0.500000 FR:0.500000
[auto-inserted resampler 0 @ 0000000001ca4060] ch:2 chl:stereo fmt:s16 r:44100Hz -> ch:1 chl:mono fmt:s16 r:16000Hz
Output #0, flv, to 'http://operator:operator@10.10.210.2/axis-cgi/audio/transmit.cgi':
Metadata:
encoder : Lavf57.57.100
Stream #0:0, 0, 1/1000: Audio: pcm_mulaw ([8][0][0][0] / 0x0008), 16000 Hz, mono, s16, 128 kb/s
Metadata:
encoder : Lavc57.66.101 pcm_mulaw
Stream mapping:
Stream #0:0 -> #0:0 (pcm_s16le (native) -> pcm_mulaw (native))
Press [q] to stop, [?] for help
cur_dts is invalid (this is harmless if it occurs once at the start per stream)
av_interleaved_write_frame(): Unknown error
No more output streams to write to, finishing.
Error writing trailer of http://operator:operator@10.10.210.2/axis-cgi/audio/transmit.cgi: Error number -10053 occurredsize= 8kB time=00:00:00.49 bitrate= 131.2kbits/s speed=79.6x
video:0kB audio:8kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 2.492485%
Input file #0 (audio=Microphone (2- High Definition Audio Device)):
Input stream #0:0 (audio): 1 packets read (88200 bytes); 1 frames decoded (22050 samples);
Total: 1 packets (88200 bytes) demuxed
Output file #0 (http://operator:operator@10.10.210.2/axis-cgi/audio/transmit.cgi):
Output stream #0:0 (audio): 1 frames encoded (7984 samples); 1 packets muxed (7984 bytes);
Total: 1 packets (7984 bytes) muxed
1 frames successfully decoded, 0 decoding errors
[AVIOContext @ 0000000001c9e4c0] Statistics: 0 seeks, 2 writeouts
dshow passing through packet of type audio size 12152 timestamp 310226130000 orig timestamp 310226130000 graph timestamp 310226820000 diff 690000 Microphone (2- High Definition Audio Device)
Conversion failed!For some reason, despite setting
multiple_requests,reconnect_eof,reconnect_streamedall to true, connection becomes closed.Could you please tell me what I’m doing wrong ?
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