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SPIP - plugins - embed code - Exemple
2 septembre 2013, par
Mis à jour : Septembre 2013
Langue : français
Type : Image
Autres articles (68)
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Les autorisations surchargées par les plugins
27 avril 2010, parMediaspip core
autoriser_auteur_modifier() afin que les visiteurs soient capables de modifier leurs informations sur la page d’auteurs -
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 -
Organiser par catégorie
17 mai 2013, parDans MédiaSPIP, une rubrique a 2 noms : catégorie et rubrique.
Les différents documents stockés dans MédiaSPIP peuvent être rangés dans différentes catégories. On peut créer une catégorie en cliquant sur "publier une catégorie" dans le menu publier en haut à droite ( après authentification ). Une catégorie peut être rangée dans une autre catégorie aussi ce qui fait qu’on peut construire une arborescence de catégories.
Lors de la publication prochaine d’un document, la nouvelle catégorie créée sera proposée (...)
Sur d’autres sites (10965)
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Google Lato Font Display discrepancies between platforms/software [closed]
5 juin 2024, par CRAIGUsing Google's Lato font with 400 weight, is giving me some inconsistent results between Photoshop/Web on MacOS and Linux Imagemagick & Linux ffmpeg.


I have printed out some text on my linux server using the font using both imagemagick (v6.9.10-68) and ffmpeg's (v6.0) drawtext feature and in both cases, I get some extra space between the T and the e. (Image below)


At first I was only testing ffmpeg and thought it was related to drawtext, but now that imagemagick is doing the same, I am thinking it may be somehow related to how the server is managing the font ?


I did download the exact same font to test on my Mac and installed it, but I still get the original photoshop results I was getting before. So I don't think it is inherently the font unless I missed a step. I did completely uninstall the font and then install the downloaded one to be sure.


However, the same word, font, font-size combo used in Photoshop and on the web does not produce the same space between the T and the e.


Here is an image I put together showing the results :




For reference, here is the imagemagick command :


convert -size 900x900 xc:black -font /fonts/Lato:400.ttf -pointsize 261.6 -fill white -gravity center -annotate +0+0 'Text' testoutput.png

And the ffmpeg command :


/usr/local/bin/ffmpeg -f lavfi -i color=c=black@0.0:s=953x953:r=30:d=5,format=rgba -filter_complex "drawtext=expansion=none:enable='between(t,0,5)':fontfile='/fonts/Lato\:400.ttf':fontcolor=0xffffffff:fontsize=294:text='Text':x=(w-text_w)/2+0:y=476.5-(max_glyph_a)+(294/2.8)" -copyts -threads 1 -c:v png -pix_fmt rgba -movflags faststart -r 30 -t 8 -y testoutput.mov

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Finding File Matches & Variable Assignment using a .BAT Script
6 octobre 2019, par A Person — batch-file, ffmpegI am trying to assign a file to a variable in a batch-file and then also assign another 2 files into anoter variable.
However, I am having an issue.
From research, I found how I can do the assigning but does anyone know how I can do the below.
From a folder or text file, (either is fine), find the
.m2vvideo file and assign that toVar1then find matching audio in.wavand put that inVar2and the third is also an audio.wavwith mathcing name and assign that toVar3.The problem I am having is trying to find the matching
2audio files to the video.The video file is named as :
PAV_PRG_13683Highc450277201906251802090353.m2vAudio 1 is :
PAV_PRG_13683High01c450211201906251802090376.wavAudio 2 is :
PAV_PRG_13683High00c450211201906251802090368.wavThe file name matches until it sees the word
High. Everything afterHighis not needed, (it is just a random string), so trying to match is an issue.Is there a way to find the match by comparing everything before
High.Also as I will be using the variable and putting them through ffmpeg to merge, is there way to do it so that when the
ffmpegcommand has completed, it moves to the next matching files and assigns them to the variable.The files are store in
2folders, one folder has all the video files*.m2vand another folder has all the*.wavaudio files in pairs of2. Each video has exactly2audios, (left right).is there any help on this subject, I have already come up empty in my research and have been checking for this over the last week spent almost 30 hours.
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ffmpeg error "Could not allocate picture : Invalid argument Found Video Stream Found Audio Stream"
26 octobre 2020, par DinkanI am trying to write a C program to stream AV by copying both AV codecs with
rtp_mpegtsusing RTP over network

ffmpeg -re -i Sample_AV_15min.ts -acodec copy -vcodec copy -f rtp_mpegts rtp://192.168.1.1:5004


using muxing.c as example which used ffmpeg libraries.
ffmpeg application works fine.


Stream details


Input #0, mpegts, from 'Weather_Nation_10min.ts':
 Duration: 00:10:00.38, start: 41313.400811, bitrate: 2840 kb/s
 Program 1
 Stream #0:0[0x11]: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p, 1440x1080 [SAR 4:3 DAR 16:9], 29.97 fps, 59.94 tbr, 90k tbn, 59.94 tbc
 Stream #0:1[0x14]: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, stereo, fltp, 448 kb/s
Output #0, rtp_mpegts, to 'rtp://192.168.1.1:5004':
 Metadata:
 encoder : Lavf54.63.104
 Stream #0:0: Video: h264 ([27][0][0][0] / 0x001B), yuv420p, 1440x1080 [SAR 4:3 DAR 16:9], q=2-31, 29.97 fps, 90k tbn, 29.97 tbc
 Stream #0:1: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, stereo, 448 kb/s
Stream mapping:
 Stream #0:0 -> #0:0 (copy)
 Stream #0:1 -> #0:1 (copy)


However, my application fails with


./my_test_app Sample_AV_15min.ts rtp://192.168.1.1:5004 
[h264 @ 0x800b30] non-existing PPS referenced 
[h264 @ 0x800b30] non-existing PPS 0 referenced 
[h264 @ 0x800b30] decode_slice_header error 
[h264 @ 0x800b30] no frame! 

[....snipped...]
[h264 @ 0x800b30] non-existing PPS 0 referenced 
[h264 @ 0x800b30] non-existing PPS referenced 
[h264 @ 0x800b30] non-existing PPS 0 referenced 
[h264 @ 0x800b30] decode_slice_header error 
[h264 @ 0x800b30] no frame! 
[h264 @ 0x800b30] mmco: unref short failure 
[h264 @ 0x800b30] mmco: unref short failure

[mpegts @ 0x800020] max_analyze_duration 5000000 reached at 5024000 microseconds 
[mpegts @ 0x800020] PES packet size mismatch could not find codec tag for codec id 
17075200, default to 0. could not find codec tag for codec id 86019, default to 0. 
Could not allocate picture: Invalid argument 
Found Video Stream Found Audio Stream


How do I fix this ? My complete source code based on muxing.c


/**
 * @file
 * libavformat API example.
 *
 * Output a media file in any supported libavformat format.
 * The default codecs are used.
 * @example doc/examples/muxing.c
 */

#include 
#include 
#include 
#include 

#include <libavutil></libavutil>mathematics.h>
#include <libavformat></libavformat>avformat.h>
#include <libswscale></libswscale>swscale.h>

/* 5 seconds stream duration */
#define STREAM_DURATION 200.0
#define STREAM_FRAME_RATE 25 /* 25 images/s */
#define STREAM_NB_FRAMES ((int)(STREAM_DURATION * STREAM_FRAME_RATE))
#define STREAM_PIX_FMT AV_PIX_FMT_YUV420P /* default pix_fmt */

static int sws_flags = SWS_BICUBIC;

/**************************************************************/
/* audio output */

static float t, tincr, tincr2;
static int16_t *samples;
static int audio_input_frame_size;
#if 0
/* Add an output stream. */
static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec,
 enum AVCodecID codec_id)
{
 AVCodecContext *c;
 AVStream *st;

 /* find the encoder */
 *codec = avcodec_find_encoder(codec_id);
 if (!(*codec)) {
 fprintf(stderr, "Could not find encoder for '%s'\n",
 avcodec_get_name(codec_id));
 exit(1);
 }

 st = avformat_new_stream(oc, *codec);
 if (!st) {
 fprintf(stderr, "Could not allocate stream\n");
 exit(1);
 }
 st->id = oc->nb_streams-1;
 c = st->codec;

 switch ((*codec)->type) {
 case AVMEDIA_TYPE_AUDIO:
 st->id = 1;
 c->sample_fmt = AV_SAMPLE_FMT_S16;
 c->bit_rate = 64000;
 c->sample_rate = 44100;
 c->channels = 2;
 break;

 case AVMEDIA_TYPE_VIDEO:
 c->codec_id = codec_id;

 c->bit_rate = 400000;
 /* Resolution must be a multiple of two. */
 c->width = 352;
 c->height = 288;
 /* timebase: This is the fundamental unit of time (in seconds) in terms
 * of which frame timestamps are represented. For fixed-fps content,
 * timebase should be 1/framerate and timestamp increments should be
 * identical to 1. */
 c->time_base.den = STREAM_FRAME_RATE;
 c->time_base.num = 1;
 c->gop_size = 12; /* emit one intra frame every twelve frames at most */
 c->pix_fmt = STREAM_PIX_FMT;
 if (c->codec_id == AV_CODEC_ID_MPEG2VIDEO) {
 /* just for testing, we also add B frames */
 c->max_b_frames = 2;
 }
 if (c->codec_id == AV_CODEC_ID_MPEG1VIDEO) {
 /* Needed to avoid using macroblocks in which some coeffs overflow.
 * This does not happen with normal video, it just happens here as
 * the motion of the chroma plane does not match the luma plane. */
 c->mb_decision = 2;
 }
 break;

 default:
 break;
 }

 /* Some formats want stream headers to be separate. */
 if (oc->oformat->flags & AVFMT_GLOBALHEADER)
 c->flags |= CODEC_FLAG_GLOBAL_HEADER;

 return st;
}
#endif 
/**************************************************************/
/* audio output */

static float t, tincr, tincr2;
static int16_t *samples;
static int audio_input_frame_size;

static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st)
{
 AVCodecContext *c;
 int ret;

 c = st->codec;

 /* open it */
 ret = avcodec_open2(c, codec, NULL);
 if (ret < 0) {
 fprintf(stderr, "Could not open audio codec: %s\n", av_err2str(ret));
 exit(1);
 }

 /* init signal generator */
 t = 0;
 tincr = 2 * M_PI * 110.0 / c->sample_rate;
 /* increment frequency by 110 Hz per second */
 tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;

 if (c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)
 audio_input_frame_size = 10000;
 else
 audio_input_frame_size = c->frame_size;
 samples = av_malloc(audio_input_frame_size *
 av_get_bytes_per_sample(c->sample_fmt) *
 c->channels);
 if (!samples) {
 fprintf(stderr, "Could not allocate audio samples buffer\n");
 exit(1);
 }
}

/* Prepare a 16 bit dummy audio frame of 'frame_size' samples and
 * 'nb_channels' channels. */
static void get_audio_frame(int16_t *samples, int frame_size, int nb_channels)
{
 int j, i, v;
 int16_t *q;

 q = samples;
 for (j = 0; j < frame_size; j++) {
 v = (int)(sin(t) * 10000);
 for (i = 0; i < nb_channels; i++)
 *q++ = v;
 t += tincr;
 tincr += tincr2;
 }
}

static void write_audio_frame(AVFormatContext *oc, AVStream *st)
{
 AVCodecContext *c;
 AVPacket pkt = { 0 }; // data and size must be 0;
 AVFrame *frame = avcodec_alloc_frame();
 int got_packet, ret;

 av_init_packet(&pkt);
 c = st->codec;

 get_audio_frame(samples, audio_input_frame_size, c->channels);
 frame->nb_samples = audio_input_frame_size;
 avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
 (uint8_t *)samples,
 audio_input_frame_size *
 av_get_bytes_per_sample(c->sample_fmt) *
 c->channels, 1);

 ret = avcodec_encode_audio2(c, &pkt, frame, &got_packet);
 if (ret < 0) {
 fprintf(stderr, "Error encoding audio frame: %s\n", av_err2str(ret));
 exit(1);
 }

 if (!got_packet)
 return;

 pkt.stream_index = st->index;

 /* Write the compressed frame to the media file. */
 ret = av_interleaved_write_frame(oc, &pkt);
 if (ret != 0) {
 fprintf(stderr, "Error while writing audio frame: %s\n",
 av_err2str(ret));
 exit(1);
 }
 avcodec_free_frame(&frame);
}

static void close_audio(AVFormatContext *oc, AVStream *st)
{
 avcodec_close(st->codec);

 av_free(samples);
}

/**************************************************************/
/* video output */

static AVFrame *frame;
static AVPicture src_picture, dst_picture;
static int frame_count;

static void open_video(AVFormatContext *oc, AVCodec *codec, AVStream *st)
{
 int ret;
 AVCodecContext *c = st->codec;

 /* open the codec */
 ret = avcodec_open2(c, codec, NULL);
 if (ret < 0) {
 fprintf(stderr, "Could not open video codec: %s\n", av_err2str(ret));
 exit(1);
 }

 /* allocate and init a re-usable frame */
 frame = avcodec_alloc_frame();
 if (!frame) {
 fprintf(stderr, "Could not allocate video frame\n");
 exit(1);
 }

 /* Allocate the encoded raw picture. */
 ret = avpicture_alloc(&dst_picture, c->pix_fmt, c->width, c->height);
 if (ret < 0) {
 fprintf(stderr, "Could not allocate picture: %s\n", av_err2str(ret));
 exit(1);
 }

 /* If the output format is not YUV420P, then a temporary YUV420P
 * picture is needed too. It is then converted to the required
 * output format. */
 if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
 ret = avpicture_alloc(&src_picture, AV_PIX_FMT_YUV420P, c->width, c->height);
 if (ret < 0) {
 fprintf(stderr, "Could not allocate temporary picture: %s\n",
 av_err2str(ret));
 exit(1);
 }
 }

 /* copy data and linesize picture pointers to frame */
 *((AVPicture *)frame) = dst_picture;
}

/* Prepare a dummy image. */
static void fill_yuv_image(AVPicture *pict, int frame_index,
 int width, int height)
{
 int x, y, i;

 i = frame_index;

 /* Y */
 for (y = 0; y < height; y++)
 for (x = 0; x < width; x++)
 pict->data[0][y * pict->linesize[0] + x] = x + y + i * 3;

 /* Cb and Cr */
 for (y = 0; y < height / 2; y++) {
 for (x = 0; x < width / 2; x++) {
 pict->data[1][y * pict->linesize[1] + x] = 128 + y + i * 2;
 pict->data[2][y * pict->linesize[2] + x] = 64 + x + i * 5;
 }
 }
}

static void write_video_frame(AVFormatContext *oc, AVStream *st)
{
 int ret;
 static struct SwsContext *sws_ctx;
 AVCodecContext *c = st->codec;

 if (frame_count >= STREAM_NB_FRAMES) {
 /* No more frames to compress. The codec has a latency of a few
 * frames if using B-frames, so we get the last frames by
 * passing the same picture again. */
 } else {
 if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
 /* as we only generate a YUV420P picture, we must convert it
 * to the codec pixel format if needed */
 if (!sws_ctx) {
 sws_ctx = sws_getContext(c->width, c->height, AV_PIX_FMT_YUV420P,
 c->width, c->height, c->pix_fmt,
 sws_flags, NULL, NULL, NULL);
 if (!sws_ctx) {
 fprintf(stderr,
 "Could not initialize the conversion context\n");
 exit(1);
 }
 }
 fill_yuv_image(&src_picture, frame_count, c->width, c->height);
 sws_scale(sws_ctx,
 (const uint8_t * const *)src_picture.data, src_picture.linesize,
 0, c->height, dst_picture.data, dst_picture.linesize);
 } else {
 fill_yuv_image(&dst_picture, frame_count, c->width, c->height);
 }
 }

 if (oc->oformat->flags & AVFMT_RAWPICTURE) {
 /* Raw video case - directly store the picture in the packet */
 AVPacket pkt;
 av_init_packet(&pkt);

 pkt.flags |= AV_PKT_FLAG_KEY;
 pkt.stream_index = st->index;
 pkt.data = dst_picture.data[0];
 pkt.size = sizeof(AVPicture);

 ret = av_interleaved_write_frame(oc, &pkt);
 } else {
 /* encode the image */
 AVPacket pkt;
 int got_output;

 av_init_packet(&pkt);
 pkt.data = NULL; // packet data will be allocated by the encoder
 pkt.size = 0;

 ret = avcodec_encode_video2(c, &pkt, frame, &got_output);
 if (ret < 0) {
 fprintf(stderr, "Error encoding video frame: %s\n", av_err2str(ret));
 exit(1);
 }

 /* If size is zero, it means the image was buffered. */
 if (got_output) {
 if (c->coded_frame->key_frame)
 pkt.flags |= AV_PKT_FLAG_KEY;

 pkt.stream_index = st->index;

 /* Write the compressed frame to the media file. */
 ret = av_interleaved_write_frame(oc, &pkt);
 } else {
 ret = 0;
 }
 }
 if (ret != 0) {
 fprintf(stderr, "Error while writing video frame: %s\n", av_err2str(ret));
 exit(1);
 }
 frame_count++;
}

static void close_video(AVFormatContext *oc, AVStream *st)
{
 avcodec_close(st->codec);
 av_free(src_picture.data[0]);
 av_free(dst_picture.data[0]);
 av_free(frame);
}

/**************************************************************/
/* media file output */

int main(int argc, char **argv)
{
 const char *filename;
 AVOutputFormat *fmt;
 AVFormatContext *oc;
 AVStream *audio_st, *video_st;
 AVCodec *audio_codec, *video_codec;
 double audio_pts, video_pts;
 int ret;
 char errbuf[50];
 int i = 0;
 /* Initialize libavcodec, and register all codecs and formats. */
 av_register_all();

 if (argc != 3) {
 printf("usage: %s input_file out_file|stream\n"
 "API example program to output a media file with libavformat.\n"
 "This program generates a synthetic audio and video stream, encodes and\n"
 "muxes them into a file named output_file.\n"
 "The output format is automatically guessed according to the file extension.\n"
 "Raw images can also be output by using '%%d' in the filename.\n"
 "\n", argv[0]);
 return 1;
 }

 filename = argv[2];

 /* allocate the output media context */
 avformat_alloc_output_context2(&oc, NULL, "rtp_mpegts", filename);
 if (!oc) {
 printf("Could not deduce output format from file extension: using MPEG.\n");
 avformat_alloc_output_context2(&oc, NULL, "mpeg", filename);
 }
 if (!oc) {
 return 1;
 }
 fmt = oc->oformat;
 //Find input stream info.

 video_st = NULL;
 audio_st = NULL;

 avformat_open_input( &oc, argv[1], 0, 0);

 if ((ret = avformat_find_stream_info(oc, 0))< 0)
 {
 av_strerror(ret, errbuf,sizeof(errbuf));
 printf("Not Able to find stream info::%s ", errbuf);
 ret = -1;
 return ret;
 }
 for (i = 0; i < oc->nb_streams; i++)
 {
 if(oc->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO)
 {
 AVCodecContext *codec_ctx;
 unsigned int tag = 0;

 printf("Found Video Stream ");
 video_st = oc->streams[i];
 codec_ctx = video_st->codec;
 // m_num_frames = oc->streams[i]->nb_frames;
 video_codec = avcodec_find_decoder(codec_ctx->codec_id);
 ret = avcodec_open2(codec_ctx, video_codec, NULL);
 if (ret < 0) 
 {
 av_log(NULL, AV_LOG_ERROR, "Failed to open decoder for stream #%u\n", i);
 return ret;
 }
 if (av_codec_get_tag2(oc->oformat->codec_tag, video_codec->id, &tag) == 0) 
 {
 av_log(NULL, AV_LOG_ERROR, "could not find codec tag for codec id %d, default to 0.\n", audio_codec->id);
 }
 video_st->codec = avcodec_alloc_context3(video_codec);
 video_st->codec->codec_tag = tag;
 }

 if(oc->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO)
 {
 AVCodecContext *codec_ctx;
 unsigned int tag = 0;

 printf("Found Audio Stream ");
 audio_st = oc->streams[i];
 // aud_dts = audio_st->cur_dts;
 // aud_pts = audio_st->last_IP_pts; 
 codec_ctx = audio_st->codec;
 audio_codec = avcodec_find_decoder(codec_ctx->codec_id);
 ret = avcodec_open2(codec_ctx, audio_codec, NULL);
 if (ret < 0) 
 {
 av_log(NULL, AV_LOG_ERROR, "Failed to open decoder for stream #%u\n", i);
 return ret;
 }
 if (av_codec_get_tag2(oc->oformat->codec_tag, audio_codec->id, &tag) == 0) 
 {
 av_log(NULL, AV_LOG_ERROR, "could not find codec tag for codec id %d, default to 0.\n", audio_codec->id);
 }
 audio_st->codec = avcodec_alloc_context3(audio_codec);
 audio_st->codec->codec_tag = tag;
 }
 }
 /* Add the audio and video streams using the default format codecs
 * and initialize the codecs. */
 /*
 if (fmt->video_codec != AV_CODEC_ID_NONE) {
 video_st = add_stream(oc, &video_codec, fmt->video_codec);
 }
 if (fmt->audio_codec != AV_CODEC_ID_NONE) {
 audio_st = add_stream(oc, &audio_codec, fmt->audio_codec);
 }
 */

 /* Now that all the parameters are set, we can open the audio and
 * video codecs and allocate the necessary encode buffers. */
 if (video_st)
 open_video(oc, video_codec, video_st);
 if (audio_st)
 open_audio(oc, audio_codec, audio_st);

 av_dump_format(oc, 0, filename, 1);

 /* open the output file, if needed */
 if (!(fmt->flags & AVFMT_NOFILE)) {
 ret = avio_open(&oc->pb, filename, AVIO_FLAG_WRITE);
 if (ret < 0) {
 fprintf(stderr, "Could not open '%s': %s\n", filename,
 av_err2str(ret));
 return 1;
 }
 }

 /* Write the stream header, if any. */
 ret = avformat_write_header(oc, NULL);
 if (ret < 0) {
 fprintf(stderr, "Error occurred when opening output file: %s\n",
 av_err2str(ret));
 return 1;
 }

 if (frame)
 frame->pts = 0;
 for (;;) {
 /* Compute current audio and video time. */
 if (audio_st)
 audio_pts = (double)audio_st->pts.val * audio_st->time_base.num / audio_st->time_base.den;
 else
 audio_pts = 0.0;

 if (video_st)
 video_pts = (double)video_st->pts.val * video_st->time_base.num /
 video_st->time_base.den;
 else
 video_pts = 0.0;

 if ((!audio_st || audio_pts >= STREAM_DURATION) &&
 (!video_st || video_pts >= STREAM_DURATION))
 break;

 /* write interleaved audio and video frames */
 if (!video_st || (video_st && audio_st && audio_pts < video_pts)) {
 write_audio_frame(oc, audio_st);
 } else {
 write_video_frame(oc, video_st);
 frame->pts += av_rescale_q(1, video_st->codec->time_base, video_st->time_base);
 }
 }

 /* Write the trailer, if any. The trailer must be written before you
 * close the CodecContexts open when you wrote the header; otherwise
 * av_write_trailer() may try to use memory that was freed on
 * av_codec_close(). */
 av_write_trailer(oc);

 /* Close each codec. */
 if (video_st)
 close_video(oc, video_st);
 if (audio_st)
 close_audio(oc, audio_st);

 if (!(fmt->flags & AVFMT_NOFILE))
 /* Close the output file. */
 avio_close(oc->pb);

 /* free the stream */
 avformat_free_context(oc);

 return 0;
}


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