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La conservation du net art au musée. Les stratégies à l’œuvre
26 mai 2011
Mis à jour : Juillet 2013
Langue : français
Type : Texte
Autres articles (34)
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Support audio et vidéo HTML5
10 avril 2011MediaSPIP utilise les balises HTML5 video et audio pour la lecture de documents multimedia en profitant des dernières innovations du W3C supportées par les navigateurs modernes.
Pour les navigateurs plus anciens, le lecteur flash Flowplayer est utilisé.
Le lecteur HTML5 utilisé a été spécifiquement créé pour MediaSPIP : il est complètement modifiable graphiquement pour correspondre à un thème choisi.
Ces technologies permettent de distribuer vidéo et son à la fois sur des ordinateurs conventionnels (...) -
Librairies et binaires spécifiques au traitement vidéo et sonore
31 janvier 2010, parLes logiciels et librairies suivantes sont utilisées par SPIPmotion d’une manière ou d’une autre.
Binaires obligatoires FFMpeg : encodeur principal, permet de transcoder presque tous les types de fichiers vidéo et sonores dans les formats lisibles sur Internet. CF ce tutoriel pour son installation ; Oggz-tools : outils d’inspection de fichiers ogg ; Mediainfo : récupération d’informations depuis la plupart des formats vidéos et sonores ;
Binaires complémentaires et facultatifs flvtool2 : (...) -
De l’upload à la vidéo finale [version standalone]
31 janvier 2010, parLe chemin d’un document audio ou vidéo dans SPIPMotion est divisé en trois étapes distinctes.
Upload et récupération d’informations de la vidéo source
Dans un premier temps, il est nécessaire de créer un article SPIP et de lui joindre le document vidéo "source".
Au moment où ce document est joint à l’article, deux actions supplémentaires au comportement normal sont exécutées : La récupération des informations techniques des flux audio et video du fichier ; La génération d’une vignette : extraction d’une (...)
Sur d’autres sites (5216)
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Cannot run ffmpeg in subproces.call
27 juin 2012, par Richard KnopSo, I have a simple class where I am trying to save a string response from a terminal ffmpeg command into an object property :
import os
import subprocess
class Movie(object):
absolute_path = None
movie_info = None
def __init__(self, path):
self.absolute_path = "%s/%s" % (os.getcwd(), path)
if(os.path.exists(self.absolute_path) is False):
raise IOError("File does not exist")
def get_movie_info(self):
ffmpeg_command = "ffmpeg -i %s" % self.absolute_path
self.movie_info = subprocess.call(ffmpeg_command)
print self.movie_infoWhen I then run this command in cmd :
import os
import sys
sys.path.append(os.getcwd())
from Encode.Movie import Movie
try:
movie = Movie("tests/test_1.mpg")
movie.get_movie_info()
except IOError as e:
print eI get this exception :
richard@richard-desktop:~/projects/hello-python$ python main.py
Traceback (most recent call last):
File "main.py", line 9, in <module>
movie.get_movie_info()
File "/home/richard/projects/hello-python/Encode/Movie.py", line 16, in get_movie_info
self.movie_info = subprocess.call(ffmpeg_command)
File "/usr/lib/python2.7/subprocess.py", line 493, in call
return Popen(*popenargs, **kwargs).wait()
File "/usr/lib/python2.7/subprocess.py", line 679, in __init__
errread, errwrite)
File "/usr/lib/python2.7/subprocess.py", line 1249, in _execute_child
raise child_exception
OSError: [Errno 2] No such file or directory
</module>The path is correct because when I do print self.absolute_path before subprocess.call(), I get :
/home/richard/projects/hello-python/tests/test_1.mpgAnd this file exists.
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FFMPEG audio transcoding using libav* libraries
10 février 2014, par vinvinodI am writing an audio transcoding application using ffmpeg libraries.
Here is my code/*
* File: main.cpp
* Author: vinod
* Compile with "g++ -std=c++11 -o audiotranscode main.cpp -lavformat -lavcodec -lavutil -lavfilter"
*
*/
#if !defined PRId64 || PRI_MACROS_BROKEN
#undef PRId64
#define PRId64 "lld"
#endif
#define __STDC_FORMAT_MACROS
#ifdef __cplusplus
extern "C" {
#endif
#include
#include
#include <sys></sys>types.h>
#include
#include <libavutil></libavutil>imgutils.h>
#include <libavutil></libavutil>samplefmt.h>
#include <libavutil></libavutil>frame.h>
#include <libavutil></libavutil>timestamp.h>
#include <libavformat></libavformat>avformat.h>
#include <libavfilter></libavfilter>avfilter.h>
#include <libavfilter></libavfilter>buffersrc.h>
#include <libavfilter></libavfilter>buffersink.h>
#include <libswscale></libswscale>swscale.h>
#include <libavutil></libavutil>opt.h>
#ifdef __cplusplus
}
#endif
#include <iostream>
using namespace std;
int select_stream, got_frame, got_packet;
AVFormatContext *in_fmt_ctx = NULL, *out_fmt_ctx = NULL;
AVCodec *dec_codec = NULL, * enc_codec = NULL;
AVStream *audio_st = NULL;
AVCodecContext *enc_ctx = NULL, *dec_ctx = NULL;
AVFrame *pFrame = NULL, * pFrameFiltered = NULL;
AVFilterGraph *filter_graph = NULL;
AVFilterContext *buffersrc_ctx = NULL;
AVFilterContext *buffersink_ctx = NULL;
AVPacket packet;
string inFileName = "/home/vinod/vinod/Media/univac.webm";
string outFileName = "audio_extracted.m4a";
int target_bit_rate = 128000,
sample_rate = 22050,
channels = 1;
AVSampleFormat sample_fmt = AV_SAMPLE_FMT_S16;
string filter_description = "aresample=22050,aformat=sample_fmts=s16:channel_layouts=mono";
int log_averror(int errcode)
{
char *errbuf = (char *) calloc(AV_ERROR_MAX_STRING_SIZE, sizeof(char));
av_strerror(errcode, errbuf, AV_ERROR_MAX_STRING_SIZE);
std::cout << "Error - " << errbuf << std::endl;
delete [] errbuf;
return -1;
}
/**
* Initialize conversion filter */
int initialize_audio_filter()
{
char args[512];
int ret;
AVFilter *buffersrc = avfilter_get_by_name("abuffer");
AVFilter *buffersink = avfilter_get_by_name("abuffersink");
AVFilterInOut *outputs = avfilter_inout_alloc();
AVFilterInOut *inputs = avfilter_inout_alloc();
filter_graph = avfilter_graph_alloc();
const enum AVSampleFormat out_sample_fmts[] = {sample_fmt, AV_SAMPLE_FMT_NONE};
const int64_t out_channel_layouts[] = {av_get_default_channel_layout(out_fmt_ctx -> streams[0] -> codec -> channels), -1};
const int out_sample_rates[] = {out_fmt_ctx -> streams[0] -> codec -> sample_rate, -1};
if (!dec_ctx->channel_layout)
dec_ctx->channel_layout = av_get_default_channel_layout(dec_ctx->channels);
snprintf(args, sizeof(args), "time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%" PRIx64,
in_fmt_ctx -> streams[select_stream] -> time_base.num, in_fmt_ctx -> streams[select_stream] -> time_base.den,
dec_ctx->sample_rate,
av_get_sample_fmt_name(dec_ctx->sample_fmt),
dec_ctx->channel_layout);
ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in", args, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create buffer source\n");
return -1;
}
ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out", NULL, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create buffer sink\n");
return ret;
}
ret = av_opt_set_int_list(buffersink_ctx, "sample_fmts", out_sample_fmts, -1,
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n");
return ret;
}
ret = av_opt_set_int_list(buffersink_ctx, "channel_layouts", out_channel_layouts, -1,
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
return ret;
}
ret = av_opt_set_int_list(buffersink_ctx, "sample_rates", out_sample_rates, -1,
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n");
return ret;
}
/* Endpoints for the filter graph. */
outputs -> name = av_strdup("in");
outputs -> filter_ctx = buffersrc_ctx;
outputs -> pad_idx = 0;
outputs -> next = NULL;
/* Endpoints for the filter graph. */
inputs -> name = av_strdup("out");
inputs -> filter_ctx = buffersink_ctx;
inputs -> pad_idx = 0;
inputs -> next = NULL;
string filter_desc = filter_description;
if ((ret = avfilter_graph_parse_ptr(filter_graph, filter_desc.c_str(), &inputs, &outputs, NULL)) < 0) {
log_averror(ret);
exit(1);
}
if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0) {
log_averror(ret);
exit(1);
}
/* Print summary of the sink buffer
* Note: args buffer is reused to store channel layout string */
AVFilterLink *outlink = buffersink_ctx->inputs[0];
av_get_channel_layout_string(args, sizeof(args), -1, outlink->channel_layout);
av_log(NULL, AV_LOG_INFO, "Output: srate:%dHz fmt:%s chlayout:%s\n",
(int) outlink->sample_rate,
(char *) av_x_if_null(av_get_sample_fmt_name((AVSampleFormat) outlink->format), "?"),
args);
return 0;
}
/*
*
*/
int main(int argc, char **argv)
{
int ret;
cout << "Hello World" << endl;
printf("abcd");
avcodec_register_all();
av_register_all();
avfilter_register_all();
/* open input file, and allocate format context */
if (avformat_open_input(&in_fmt_ctx, inFileName.c_str(), NULL, NULL) < 0) {
std::cout << "error opening input file - " << inFileName << std::endl;
return -1;
}
/* retrieve stream information */
if (avformat_find_stream_info(in_fmt_ctx, NULL) < 0) {
std::cerr << "Could not find stream information in the input file " << inFileName << std::endl;
}
/* Dump format details */
printf("\n ---------------------------------------------------------------------- \n");
av_dump_format(in_fmt_ctx, 0, inFileName.c_str(), 0);
printf("\n ---------------------------------------------------------------------- \n");
/* Choose a audio stream */
select_stream = av_find_best_stream(in_fmt_ctx, AVMEDIA_TYPE_AUDIO, -1, -1, &dec_codec, 0);
if (select_stream == AVERROR_STREAM_NOT_FOUND) {
std::cerr << "No audio stream found" << std::endl;
return -1;
}
if (select_stream == AVERROR_DECODER_NOT_FOUND) {
std::cerr << "No suitable decoder found" << std::endl;
return -1;
}
dec_ctx = in_fmt_ctx -> streams[ select_stream] -> codec;
av_opt_set_int(dec_ctx, "refcounted_frames", 1, 0);
/* init the audio decoder */
if ((ret = avcodec_open2(dec_ctx, dec_codec, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open audio decoder\n");
return ret;
}
/* allocate output context */
ret = avformat_alloc_output_context2(&out_fmt_ctx, NULL, NULL,
outFileName.c_str());
if (ret < 0) {
std::cerr << "Could not create output context for the file " << outFileName << std::endl;
return -1;
}
/* find the encoder */
enum AVCodecID codec_id = out_fmt_ctx -> oformat -> audio_codec;
enc_codec = avcodec_find_encoder(codec_id);
if (!(enc_codec)) {
std::cerr << "Could not find encoder for - " << avcodec_get_name(codec_id) << std::endl;
return -1;
}
/* add a new stream */
audio_st = avformat_new_stream(out_fmt_ctx, enc_codec);
if (!audio_st) {
std::cerr << "Could not add audio stream - " << std::endl;
}
/* Initialise audio codec */
audio_st -> id = out_fmt_ctx -> nb_streams - 1;
enc_ctx = audio_st -> codec;
enc_ctx -> codec_id = codec_id;
enc_ctx -> codec_type = AVMEDIA_TYPE_AUDIO;
enc_ctx -> bit_rate = target_bit_rate;
enc_ctx -> sample_rate = sample_rate;
enc_ctx -> sample_fmt = sample_fmt;
enc_ctx -> channels = channels;
enc_ctx -> channel_layout = av_get_default_channel_layout(enc_ctx -> channels);
/* Some formats want stream headers to be separate. */
if (out_fmt_ctx -> oformat -> flags & AVFMT_GLOBALHEADER) {
enc_ctx -> flags |= CODEC_FLAG_GLOBAL_HEADER;
}
ret = avcodec_open2(out_fmt_ctx -> streams[0] -> codec, enc_codec, NULL);
if (ret < 0) {
std::cerr << "Could not create codec context for the file " << outFileName << std::endl;
return -1;
}
/* Initialize filter */
initialize_audio_filter();
if (!(out_fmt_ctx -> oformat -> flags & AVFMT_NOFILE)) {
int ret = avio_open(& out_fmt_ctx -> pb, outFileName.c_str(),
AVIO_FLAG_WRITE);
if (ret < 0) {
log_averror(ret);
return -1;
}
}
/* Write header */
if (avformat_write_header(out_fmt_ctx, NULL) < 0) {
if (ret < 0) {
log_averror(ret);
return -1;
}
}
/* Allocate frame */
pFrame = av_frame_alloc();
if (!pFrame) {
std::cerr << "Could not allocate frame\n";
return -1;
}
pFrameFiltered = av_frame_alloc();
if (!pFrameFiltered) {
std::cerr << "Could not allocate frame\n";
return -1;
}
av_init_packet(&packet);
packet.data = NULL;
packet.size = 0;
/* Read packet from the stream */
while (av_read_frame(in_fmt_ctx, &packet) >= 0) {
if (packet.stream_index == select_stream) {
avcodec_get_frame_defaults(pFrame);
ret = avcodec_decode_audio4(dec_ctx, pFrame, &got_frame, &packet);
if (ret < 0) {
log_averror(ret);
return ret;
}
printf("Decoded packet pts : %ld ", packet.pts);
printf("Frame Best Effor pts : %ld \n", pFrame->best_effort_timestamp);
/* Set frame pts */
pFrame -> pts = av_frame_get_best_effort_timestamp(pFrame);
if (got_frame) {
/* push the decoded frame into the filtergraph */
ret = av_buffersrc_add_frame_flags(buffersrc_ctx, pFrame, AV_BUFFERSRC_FLAG_KEEP_REF);
if (ret < 0) {
log_averror(ret);
return ret;
}
/* pull filtered frames from the filtergraph */
while (1) {
ret = av_buffersink_get_frame(buffersink_ctx, pFrameFiltered);
if ((ret == AVERROR(EAGAIN)) || (ret == AVERROR_EOF)) {
break;
}
if (ret < 0) {
printf("Error while getting filtered frames from filtergraph\n");
log_averror(ret);
return -1;
}
/* Initialize the packets */
AVPacket encodedPacket = {0};
av_init_packet(&encodedPacket);
ret = avcodec_encode_audio2(out_fmt_ctx -> streams[0] -> codec, &encodedPacket, pFrameFiltered, &got_packet);
if (!ret && got_packet && encodedPacket.size) {
/* Set correct pts and dts */
if (encodedPacket.pts != AV_NOPTS_VALUE) {
encodedPacket.pts = av_rescale_q(encodedPacket.pts, buffersink_ctx -> inputs[0] -> time_base,
out_fmt_ctx -> streams[0] -> time_base);
}
if (encodedPacket.dts != AV_NOPTS_VALUE) {
encodedPacket.dts = av_rescale_q(encodedPacket.dts, buffersink_ctx -> inputs[0] -> time_base,
out_fmt_ctx -> streams[0] -> time_base);
}
printf("Encoded packet pts %ld\n", encodedPacket.pts);
/* Write the compressed frame to the media file. */
ret = av_interleaved_write_frame(out_fmt_ctx, &encodedPacket);
if (ret < 0) {
log_averror(ret);
return -1;
}
} else if (ret < 0) {
log_averror(ret);
return -1;
}
av_frame_unref(pFrameFiltered);
}
av_frame_unref(pFrame);
}
}
}
/* Flush delayed frames from encoder*/
got_packet=1;
while (got_packet) {
AVPacket encodedPacket = {0};
av_init_packet(&encodedPacket);
ret = avcodec_encode_audio2(out_fmt_ctx -> streams[0] -> codec, &encodedPacket, NULL, &got_packet);
if (!ret && got_packet && encodedPacket.size) {
/* Set correct pts and dts */
if (encodedPacket.pts != AV_NOPTS_VALUE) {
encodedPacket.pts = av_rescale_q(encodedPacket.pts, buffersink_ctx -> inputs[0] -> time_base,
out_fmt_ctx -> streams[0] -> time_base);
}
if (encodedPacket.dts != AV_NOPTS_VALUE) {
encodedPacket.dts = av_rescale_q(encodedPacket.dts, buffersink_ctx -> inputs[0] -> time_base,
out_fmt_ctx -> streams[0] -> time_base);
}
printf("Encoded packet pts %ld\n", encodedPacket.pts);
/* Write the compressed frame to the media file. */
ret = av_interleaved_write_frame(out_fmt_ctx, &encodedPacket);
if (ret < 0) {
log_averror(ret);
return -1;
}
} else if (ret < 0) {
log_averror(ret);
return -1;
}
}
/* Write Trailer */
av_write_trailer(out_fmt_ctx);
avfilter_graph_free(&filter_graph);
if (dec_ctx)
avcodec_close(dec_ctx);
avformat_close_input(&in_fmt_ctx);
av_frame_free(&pFrame);
av_frame_free(&pFrameFiltered);
if (!(out_fmt_ctx -> oformat -> flags & AVFMT_NOFILE))
avio_close(out_fmt_ctx -> pb);
avcodec_close(out_fmt_ctx->streams[0]->codec);
avformat_free_context(out_fmt_ctx);
return 0;
}
</iostream>The audio file after transcoding is same duration as the input. But its completely noisy. Can somebody tell me what I am doing wrong here !
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Size Discrepany in the ‘du’ Command
22 juin 2012, par Multimedia Mike — GeneralI had a problem today while using the common Unix command ’du’. As a refresher, ’du’ stands for disk usage and is a handy tool for understanding how much disk space is being occupied.
I think ’du’ is probably doing the right thing. The problem might be that I’m getting strange (read : 1/2 the expected number) when running the tool against directories on vmhgfs, the VMware filesystem.
Science Project
On an Ubuntu Linux VMware session, my home directory is on the main file system, which is ext4. The directory /mnt/hgfs is reported by ’mount’ to be of type vmhgfs and is shared with the host machine.Create a directory in the home directory and generate a 10 MiB file :
mkdir /home/melanson/dir dd if=/dev/urandom of=/home/melanson/dir/random-file bs=1048576 count=10
Create a directory on the shared drive and copy the same file :
mkdir /mnt/hgfs/vmshare/dir cp /home/melanson/dir/random-file /mnt/hgfs/vmshare/dir
Run ’du’ on each directory using the -k and -h options :
du -k /home/melanson/dir /mnt/hgfs/vmshare/dir 10244 /home/melanson/dir 5120 /mnt/hgfs/vmshare/dir
du -h /home/melanson/dir /mnt/hgfs/vmshare/dir
11M /home/melanson/directory
5.0M /mnt/hgfs/vmshare/directoryI noticed this discrepancy when I was trying to pack a set of files (akin to ’tar’-ing) living in a directory in the shared location. I was going mad trying to understand why the original directory was only 2 MB as reported by ’du’ but the final packed file was 4 MB.
To be fair, the man page for ’du’ succinctly states that the tool’s purpose is merely to "estimate file space usage".
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