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MediaSPIP Simple : futur thème graphique par défaut ?
26 septembre 2013, par
Mis à jour : Octobre 2013
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Type : Video
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Les autorisations surchargées par les plugins
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autoriser_auteur_modifier() afin que les visiteurs soient capables de modifier leurs informations sur la page d’auteurs -
Personnaliser les catégories
21 juin 2013, parFormulaire de création d’une catégorie
Pour ceux qui connaissent bien SPIP, une catégorie peut être assimilée à une rubrique.
Dans le cas d’un document de type catégorie, les champs proposés par défaut sont : Texte
On peut modifier ce formulaire dans la partie :
Administration > Configuration des masques de formulaire.
Dans le cas d’un document de type média, les champs non affichés par défaut sont : Descriptif rapide
Par ailleurs, c’est dans cette partie configuration qu’on peut indiquer le (...) -
Des sites réalisés avec MediaSPIP
2 mai 2011, parCette page présente quelques-uns des sites fonctionnant sous MediaSPIP.
Vous pouvez bien entendu ajouter le votre grâce au formulaire en bas de page.
Sur d’autres sites (11409)
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using ffmpeg libraray to write to a mp4, ffprobe shows there are 100 frames and 100 packets, but av_interleaved_write_frame only called 50 times
2 mai 2023, par ollydbg23here is my code to generate a mp4 file by using ffmpeg and opencv library. The opencv library is only try to generate 100 images(frames), and ffmpeg library is to compress the images to a mp4 files.


Here is the working code :


#include <iostream>
#include <vector>
#include <cstring>
#include <fstream>
#include <sstream>
#include <stdexcept>
#include <opencv2></opencv2>opencv.hpp>
extern "C" {
#include <libavutil></libavutil>imgutils.h>
#include <libavcodec></libavcodec>avcodec.h>
#include <libavformat></libavformat>avformat.h>
#include <libavutil></libavutil>opt.h>
}

#include<cstdlib> // to generate time stamps

using namespace std;
using namespace cv;

int main()
{
 // Set up input frames as BGR byte arrays
 vector<mat> frames;

 int width = 640;
 int height = 480;
 int num_frames = 100;
 Scalar black(0, 0, 0);
 Scalar white(255, 255, 255);
 int font = FONT_HERSHEY_SIMPLEX;
 double font_scale = 1.0;
 int thickness = 2;

 for (int i = 0; i < num_frames; i++) {
 Mat frame = Mat::zeros(height, width, CV_8UC3);
 putText(frame, std::to_string(i), Point(width / 2 - 50, height / 2), font, font_scale, white, thickness);
 frames.push_back(frame);
 }

 // generate a serial of time stamps which is used to set the PTS value
 // suppose they are in ms unit, the time interval is between 30ms to 59ms
 vector<int> timestamps;

 for (int i = 0; i < num_frames; i++) {
 int timestamp;
 if (i == 0)
 timestamp = 0;
 else
 {
 int random = 30 + (rand() % 30);
 timestamp = timestamps[i-0] + random;
 }

 timestamps.push_back(timestamp);
 }

 // Populate frames with BGR byte arrays

 // Initialize FFmpeg
 //av_register_all();

 // Set up output file
 AVFormatContext* outFormatCtx = nullptr;
 //AVCodec* outCodec = nullptr;
 AVCodecContext* outCodecCtx = nullptr;
 //AVStream* outStream = nullptr;
 //AVPacket outPacket;

 const char* outFile = "output.mp4";
 int outWidth = frames[0].cols;
 int outHeight = frames[0].rows;
 int fps = 25;

 // Open the output file context
 avformat_alloc_output_context2(&outFormatCtx, nullptr, nullptr, outFile);
 if (!outFormatCtx) {
 cerr << "Error: Could not allocate output format context" << endl;
 return -1;
 }

 // Open the output file
 if (avio_open(&outFormatCtx->pb, outFile, AVIO_FLAG_WRITE) < 0) {
 cerr << "Error opening output file" << std::endl;
 return -1;
 }

 // Set up output codec
 const AVCodec* outCodec = avcodec_find_encoder(AV_CODEC_ID_H264);
 if (!outCodec) {
 cerr << "Error: Could not find H.264 codec" << endl;
 return -1;
 }

 outCodecCtx = avcodec_alloc_context3(outCodec);
 if (!outCodecCtx) {
 cerr << "Error: Could not allocate output codec context" << endl;
 return -1;
 }
 outCodecCtx->codec_id = AV_CODEC_ID_H264;
 outCodecCtx->codec_type = AVMEDIA_TYPE_VIDEO;
 outCodecCtx->pix_fmt = AV_PIX_FMT_YUV420P;
 outCodecCtx->width = outWidth;
 outCodecCtx->height = outHeight;
 //outCodecCtx->time_base = { 1, fps*1000 }; // 25000
 outCodecCtx->time_base = { 1, fps}; // 25000
 outCodecCtx->framerate = {fps, 1}; // 25
 outCodecCtx->bit_rate = 4000000;

 //https://github.com/leandromoreira/ffmpeg-libav-tutorial
 //We set the flag AV_CODEC_FLAG_GLOBAL_HEADER which tells the encoder that it can use the global headers.
 if (outFormatCtx->oformat->flags & AVFMT_GLOBALHEADER)
 {
 outCodecCtx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER; //
 }

 // Open output codec
 if (avcodec_open2(outCodecCtx, outCodec, nullptr) < 0) {
 cerr << "Error: Could not open output codec" << endl;
 return -1;
 }

 // Create output stream
 AVStream* outStream = avformat_new_stream(outFormatCtx, outCodec);
 if (!outStream) {
 cerr << "Error: Could not allocate output stream" << endl;
 return -1;
 }

 // Configure output stream parameters (e.g., time base, codec parameters, etc.)
 // ...

 // Connect output stream to format context
 outStream->codecpar->codec_id = outCodecCtx->codec_id;
 outStream->codecpar->codec_type = AVMEDIA_TYPE_VIDEO;
 outStream->codecpar->width = outCodecCtx->width;
 outStream->codecpar->height = outCodecCtx->height;
 outStream->codecpar->format = outCodecCtx->pix_fmt;
 outStream->time_base = outCodecCtx->time_base;

 int ret = avcodec_parameters_from_context(outStream->codecpar, outCodecCtx);
 if (ret < 0) {
 cerr << "Error: Could not copy codec parameters to output stream" << endl;
 return -1;
 }

 outStream->avg_frame_rate = outCodecCtx->framerate;
 //outStream->id = outFormatCtx->nb_streams++; <--- We shouldn't modify outStream->id

 ret = avformat_write_header(outFormatCtx, nullptr);
 if (ret < 0) {
 cerr << "Error: Could not write output header" << endl;
 return -1;
 }

 // Convert frames to YUV format and write to output file
 int frame_count = -1;
 for (const auto& frame : frames) {
 frame_count++;
 AVFrame* yuvFrame = av_frame_alloc();
 if (!yuvFrame) {
 cerr << "Error: Could not allocate YUV frame" << endl;
 return -1;
 }
 av_image_alloc(yuvFrame->data, yuvFrame->linesize, outWidth, outHeight, AV_PIX_FMT_YUV420P, 32);

 yuvFrame->width = outWidth;
 yuvFrame->height = outHeight;
 yuvFrame->format = AV_PIX_FMT_YUV420P;

 // Convert BGR frame to YUV format
 Mat yuvMat;
 cvtColor(frame, yuvMat, COLOR_BGR2YUV_I420);
 memcpy(yuvFrame->data[0], yuvMat.data, outWidth * outHeight);
 memcpy(yuvFrame->data[1], yuvMat.data + outWidth * outHeight, outWidth * outHeight / 4);
 memcpy(yuvFrame->data[2], yuvMat.data + outWidth * outHeight * 5 / 4, outWidth * outHeight / 4);

 // Set up output packet
 //av_init_packet(&outPacket); //error C4996: 'av_init_packet': was declared deprecated
 AVPacket* outPacket = av_packet_alloc();
 memset(outPacket, 0, sizeof(outPacket)); //Use memset instead of av_init_packet (probably unnecessary).
 //outPacket->data = nullptr;
 //outPacket->size = 0;

 // set the frame pts, do I have to set the package pts?

 // yuvFrame->pts = av_rescale_q(timestamps[frame_count]*25, outCodecCtx->time_base, outStream->time_base); //Set PTS timestamp
 yuvFrame->pts = av_rescale_q(frame_count*frame_count, outCodecCtx->time_base, outStream->time_base); //Set PTS timestamp

 // Encode frame and write to output file
 int ret = avcodec_send_frame(outCodecCtx, yuvFrame);
 if (ret < 0) {
 cerr << "Error: Could not send frame to output codec" << endl;
 return -1;
 }
 while (ret >= 0)
 {
 ret = avcodec_receive_packet(outCodecCtx, outPacket);

 if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
 {
 int abc;
 abc++;
 break;
 }
 else if (ret < 0)
 {
 cerr << "Error: Could not receive packet from output codec" << endl;
 return -1;
 }

 //av_packet_rescale_ts(&outPacket, outCodecCtx->time_base, outStream->time_base);

 outPacket->stream_index = outStream->index;

 outPacket->duration = av_rescale_q(1, outCodecCtx->time_base, outStream->time_base); // Set packet duration

 ret = av_interleaved_write_frame(outFormatCtx, outPacket);

 static int call_write = 0;

 call_write++;
 printf("av_interleaved_write_frame %d\n", call_write);

 av_packet_unref(outPacket);
 if (ret < 0) {
 cerr << "Error: Could not write packet to output file" << endl;
 return -1;
 }
 }

 av_frame_free(&yuvFrame);
 }

 // Flush the encoder
 ret = avcodec_send_frame(outCodecCtx, nullptr);
 if (ret < 0) {
 std::cerr << "Error flushing encoder: " << std::endl;
 return -1;
 }

 while (ret >= 0) {
 AVPacket* pkt = av_packet_alloc();
 if (!pkt) {
 std::cerr << "Error allocating packet" << std::endl;
 return -1;
 }
 ret = avcodec_receive_packet(outCodecCtx, pkt);

 // Write the packet to the output file
 if (ret == 0)
 {
 pkt->stream_index = outStream->index;
 pkt->duration = av_rescale_q(1, outCodecCtx->time_base, outStream->time_base); // <---- Set packet duration
 ret = av_interleaved_write_frame(outFormatCtx, pkt);
 av_packet_unref(pkt);
 if (ret < 0) {
 std::cerr << "Error writing packet to output file: " << std::endl;
 return -1;
 }
 }
 }


 // Write output trailer
 av_write_trailer(outFormatCtx);

 // Clean up
 avcodec_close(outCodecCtx);
 avcodec_free_context(&outCodecCtx);
 avformat_free_context(outFormatCtx);

 return 0;
}

</int></mat></cstdlib></stdexcept></sstream></fstream></cstring></vector></iostream>

Note that I have used the
ffprobetool(one of the tool from ffmpeg) to inspect the generated mp4 files.

I see that the mp4 file has 100 frames and 100 packets, but in my code, I have such lines :


static int call_write = 0;

 call_write++;
 printf("av_interleaved_write_frame %d\n", call_write);


I just see that the
av_interleaved_write_framefunction is only called 50 times, not the expected 100 times, anyone can explain it ?

Thanks.


BTW, from the ffmpeg document( see here : For video, it should typically contain one compressed frame ), I see that a
packetmainly has one videoframe, so theffprobe's result looks correct.

Here is the command I used to inspect the mp4 file :


ffprobe -show_frames output.mp4 >> frames.txt
ffprobe -show_packets output.mp4 >> packets.txt


My testing code is derived from an answer in another question here : avformat_write_header() function call crashed when I try to save several RGB data to a output.mp4 file


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How to record the bluetooth audio from my phone in my pc using ffmpeg ?
12 septembre 2023, par Juan InzunzaI have my phone connected via Bluetooth to my PC and playing audio in my phone so i can hear that audio in my pc and i was trying to record that audio in my PC using ffmpeg this is the command i got so far


ffmpeg -f pulse -ac 1 -ar 44100 -i alsa_input.usb-KTMicro_KT_USB_Audio_2021-06-25-0000-0000-0000--00.mono-fallback -f pulse -ac 2 -ar 44100 -i bluez_source.84_5F_04_75_B9_F6.a2dp_source -filter_complex amix=inputs=2 -i /dev/video5 -s 640x480 -vcodec libx264 -preset veryfast -crf 18 -acodec libmp3lame -ar 44100 -q:a 1 -pix_fmt yuv420p -aq 0 ~/Videos


"bluez_source.84_5F_04_75_B9_F6.a2dp_source" is the name of the bluetooth audio source (my phone) but the problem is, if i am not playing audio on the device, just disappears when i execute this command "pactl list sources", and appears again when i play audio, so is listed when i run "pactl list sources" and i have to keep the phone playing audio so ffmpeg recognizes that device and not fail when executed.


Finally my question is how can i keep that device in running state or suspended state without need to play audio all the time.


gandalf@Mordor:~$ pactl list sources
Source #64
 State: IDLE
 Name: alsa_output.usb-KTMicro_KT_USB_Audio_2021-06-25-0000-0000-0000--00.analog-stereo.monitor
 Description: Monitor of KT USB Audio Analog Stereo
 Driver: module-alsa-card.c
 Sample Specification: s16le 2ch 44100Hz
 Channel Map: front-left,front-right
 Owner Module: 85
 Mute: no
 Volume: front-left: 65536 / 100% / 0.00 dB, front-right: 65536 / 100% / 0.00 dB
 balance 0.00
 Base Volume: 65536 / 100% / 0.00 dB
 Monitor of Sink: alsa_output.usb-KTMicro_KT_USB_Audio_2021-06-25-0000-0000-0000--00.analog-stereo
 Latency: 0 usec, configured 2000000 usec
 Flags: DECIBEL_VOLUME LATENCY 
 Properties:
 device.description = "Monitor of KT USB Audio Analog Stereo"
 device.class = "monitor"
 alsa.card = "1"
 alsa.card_name = "KT USB Audio"
 alsa.long_card_name = "KTMicro KT USB Audio at usb-0000:03:00.3-3, full speed"
 alsa.driver_name = "snd_usb_audio"
 device.bus_path = "pci-0000:03:00.3-usb-0:3:1.0"
 sysfs.path = "/devices/pci0000:00/0000:00:08.1/0000:03:00.3/usb1/1-3/1-3:1.0/sound/card1"
 udev.id = "usb-KTMicro_KT_USB_Audio_2021-06-25-0000-0000-0000--00"
 device.bus = "usb"
 device.vendor.id = "12d1"
 device.vendor.name = "Huawei Technologies Co., Ltd."
 device.product.id = "0010"
 device.product.name = "KT USB Audio"
 device.serial = "KTMicro_KT_USB_Audio_2021-06-25-0000-0000-0000-"
 device.string = "1"
 module-udev-detect.discovered = "1"
 device.icon_name = "audio-card-usb"
 Formats:
 pcm

Source #65
 State: SUSPENDED
 Name: alsa_input.usb-KTMicro_KT_USB_Audio_2021-06-25-0000-0000-0000--00.mono-fallback
 Description: KT USB Audio Mono
 Driver: module-alsa-card.c
 Sample Specification: s16le 1ch 44100Hz
 Channel Map: mono
 Owner Module: 85
 Mute: no
 Volume: mono: 65536 / 100% / 0.00 dB
 balance 0.00
 Base Volume: 65536 / 100% / 0.00 dB
 Monitor of Sink: n/a
 Latency: 0 usec, configured 0 usec
 Flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY 
 Properties:
 alsa.resolution_bits = "16"
 device.api = "alsa"
 device.class = "sound"
 alsa.class = "generic"
 alsa.subclass = "generic-mix"
 alsa.name = "USB Audio"
 alsa.id = "USB Audio"
 alsa.subdevice = "0"
 alsa.subdevice_name = "subdevice #0"
 alsa.device = "0"
 alsa.card = "1"
 alsa.card_name = "KT USB Audio"
 alsa.long_card_name = "KTMicro KT USB Audio at usb-0000:03:00.3-3, full speed"
 alsa.driver_name = "snd_usb_audio"
 device.bus_path = "pci-0000:03:00.3-usb-0:3:1.0"
 sysfs.path = "/devices/pci0000:00/0000:00:08.1/0000:03:00.3/usb1/1-3/1-3:1.0/sound/card1"
 udev.id = "usb-KTMicro_KT_USB_Audio_2021-06-25-0000-0000-0000--00"
 device.bus = "usb"
 device.vendor.id = "12d1"
 device.vendor.name = "Huawei Technologies Co., Ltd."
 device.product.id = "0010"
 device.product.name = "KT USB Audio"
 device.serial = "KTMicro_KT_USB_Audio_2021-06-25-0000-0000-0000-"
 device.string = "hw:1"
 device.buffering.buffer_size = "176400"
 device.buffering.fragment_size = "88200"
 device.access_mode = "mmap+timer"
 device.profile.name = "mono-fallback"
 device.profile.description = "Mono"
 device.description = "KT USB Audio Mono"
 module-udev-detect.discovered = "1"
 device.icon_name = "audio-card-usb"
 Ports:
 analog-input-mic: Microphone (type: Mic, priority: 8700, availability unknown)
 Active Port: analog-input-mic
 Formats:
 pcm

Source #68
 State: RUNNING
 Name: bluez_source.84_5F_04_75_B9_F6.a2dp_source
 Description: Juan's S20 FE
 Driver: module-bluez5-device.c
 Sample Specification: s16le 2ch 44100Hz
 Channel Map: front-left,front-right
 Owner Module: 23
 Mute: no
 Volume: front-left: 53151 / 81% / -5.46 dB, front-right: 53151 / 81% / -5.46 dB
 balance 0.00
 Base Volume: 65536 / 100% / 0.00 dB
 Monitor of Sink: n/a
 Latency: 68928 usec, configured 68537 usec
 Flags: HARDWARE DECIBEL_VOLUME LATENCY 
 Properties:
 bluetooth.protocol = "a2dp_source"
 bluetooth.codec = "sbc"
 device.description = "Juan's S20 FE"
 device.string = "84:5F:04:75:B9:F6"
 device.api = "bluez"
 device.class = "sound"
 device.bus = "bluetooth"
 device.form_factor = "phone"
 bluez.path = "/org/bluez/hci0/dev_84_5F_04_75_B9_F6"
 bluez.class = "0x5a020c"
 bluez.alias = "Juan's S20 FE"
 device.icon_name = "audio-card-bluetooth"
 Ports:
 phone-input: Phone (type: Phone, priority: 0, available)
 Active Port: phone-input
 Formats:
 pcm


This is the error i am getting when there's no playing audio in my phone.


ffmpeg version 4.4.2-0ubuntu0.22.04.1 Copyright (c) 2000-2021 the FFmpeg developers
 built with gcc 11 (Ubuntu 11.2.0-19ubuntu1)
 configuration: --prefix=/usr --extra-version=0ubuntu0.22.04.1 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --arch=amd64 --enable-gpl --disable-stripping --enable-gnutls --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libcodec2 --enable-libdav1d --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libjack --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librabbitmq --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzimg --enable-libzmq --enable-libzvbi --enable-lv2 --enable-omx --enable-openal --enable-opencl --enable-opengl --enable-sdl2 --enable-pocketsphinx --enable-librsvg --enable-libmfx --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libx264 --enable-shared
 libavutil 56. 70.100 / 56. 70.100
 libavcodec 58.134.100 / 58.134.100
 libavformat 58. 76.100 / 58. 76.100
 libavdevice 58. 13.100 / 58. 13.100
 libavfilter 7.110.100 / 7.110.100
 libswscale 5. 9.100 / 5. 9.100
 libswresample 3. 9.100 / 3. 9.100
 libpostproc 55. 9.100 / 55. 9.100
Guessed Channel Layout for Input Stream #0.0 : mono
Input #0, pulse, from 'alsa_input.usb-KTMicro_KT_USB_Audio_2021-06-25-0000-0000-0000--00.mono-fallback':
 Duration: N/A, start: 1694557964.268337, bitrate: 705 kb/s
 Stream #0:0: Audio: pcm_s16le, 44100 Hz, mono, s16, 705 kb/s
bluez_source.84_5F_04_75_B9_F6.a2dp_source: Input/output error



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lavc/speedhqdec : Add AV_CODEC_CAP_SLICE_THREADS
13 mai 2024, par Tomas Härdin
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