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La file d’attente de SPIPmotion
28 novembre 2010, parUne file d’attente stockée dans la base de donnée
Lors de son installation, SPIPmotion crée une nouvelle table dans la base de donnée intitulée spip_spipmotion_attentes.
Cette nouvelle table est constituée des champs suivants : id_spipmotion_attente, l’identifiant numérique unique de la tâche à traiter ; id_document, l’identifiant numérique du document original à encoder ; id_objet l’identifiant unique de l’objet auquel le document encodé devra être attaché automatiquement ; objet, le type d’objet auquel (...) -
Les vidéos
21 avril 2011, parComme les documents de type "audio", Mediaspip affiche dans la mesure du possible les vidéos grâce à la balise html5 .
Un des inconvénients de cette balise est qu’elle n’est pas reconnue correctement par certains navigateurs (Internet Explorer pour ne pas le nommer) et que chaque navigateur ne gère en natif que certains formats de vidéos.
Son avantage principal quant à lui est de bénéficier de la prise en charge native de vidéos dans les navigateur et donc de se passer de l’utilisation de Flash et (...) -
Websites made with MediaSPIP
2 mai 2011, parThis page lists some websites based on MediaSPIP.
Sur d’autres sites (6831)
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Bat file to create a list of files and there paths in the same directory as the bat file
4 janvier 2020, par punktilendI want to create a bat file that creates a list of files and there paths (MyList.txt). Saves that file to the same directory the bat file is being run in.
This is what I am trying to accomplish with the MyList.txt. I would like to have the files in order as well.
file 'E:\Adobe\Projects\Work\191215 December\1\CONTENTS\CLIPS001\AA3023\AA302301.MXF'
file 'E:\Adobe\Projects\Work\191215 December\1\CONTENTS\CLIPS001\AA3023\AA302302.MXF'
file 'E:\Adobe\Projects\Work\191215 December\1\CONTENTS\CLIPS001\AA3023\AA302303.MXF'
file 'E:\Adobe\Projects\Work\191215 December\1\CONTENTS\CLIPS001\AA3023\AA302304.MXF'After this, I would like FFMPEG to concat those files
ffmpeg -f concat -safe 0 -i MyList.txt -c copy -map 0 Output.mxf
and output that to the same directory as well, but I think it will do that currently.This is what I have come up with so far :
@echo off
ECHO Creating MXF Concat List
dir /a /-p /o:gen > "%~dp1MXF_list.txt"
ffmpeg -f concat -safe 0 -i MXF_list.txt -c copy -map 0 Output.mxf -
flutter :can i make the input file & the output file the same in ffmpeg command ?
17 juillet 2022, par Khaled MortajaI am using this command for applying a crop operation on a video stored on my internal storage $x is the location of the video , I do not want a new output file to be created, i just want the cropped video to overwrite the input video, so any changes on the video will be applied to the original video location.
i used this command but it did not work


FFmpegKit.execute("-i $x -ss 00:00:01 -t 00:00:10 -c copy $x")



any suggestions...
thanks in advance


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I want to take any Audio from a file and encode it as PCM_ALAW. My Example is a .m4a file to .wav file
22 novembre 2023, par ClockmanI have been working on this for a while now while am generally new to ffmpeg library, I have studied it a bit. The challenge I have that at the point of witting to file I get the following exception.


"Exception thrown at 0x00007FFACA8305B3 (avformat-60.dll) in FfmpegPractice.exe : 0xC0000005 : Access violation writing location 0x0000000000000000.". I understand this means am writing to an uninitialized buffer am unable to discover why this is happening. The exception call stack shows the following


avformat-60.dll!avformat_write_header() C
avformat-60.dll!ff_write_chained() C
avformat-60.dll!ff_write_chained() C
avformat-60.dll!av_write_frame() C
FfmpegPractice.exe!main() Line 215 C++



Some things I have tried


This code is part of a larger project built with CMake but for some reason I could no step into ffmpeg library while debugging, So I recompiled ffmpeg ensured debugging was enabled so I could drill down to the root cause but I still could not step into the ffmpeg library.


I then created a minimal project using visual studio c++ console project and I still could not step into the code.


I have read through many ffmpeg docs and some I could find on the internet and I still could not solve it.


This is the code


#include <iostream>

extern "C" {
#include <libavcodec></libavcodec>avcodec.h>
#include <libavformat></libavformat>avformat.h>
#include <libswresample></libswresample>swresample.h>
#include <libavutil></libavutil>opt.h>
#include <libavutil></libavutil>audio_fifo.h>
}

using namespace std;

//in audio file
string filename{ "rapid_caller_test.m4a" };
AVFormatContext* pFormatCtx{};
AVCodecContext* pCodecCtx{};
AVStream* pStream{};

//out audio file
string outFilename{ "output.wav" };
AVFormatContext* pOutFormatCtx{ nullptr };
AVCodecContext* pOutCodecCtx{ nullptr };
AVIOContext* pOutIoContext{ nullptr };
const AVCodec* pOutCodec{ nullptr };
AVStream* pOutStream{ nullptr };
const int OUTPUT_CHANNELS = 1;
const int SAMPLE_RATE = 8000;
const int OUT_BIT_RATE = 64000;
uint8_t** convertedSamplesBuffer{ nullptr };
int64_t dstNmbrSamples{ 0 };
int dstLineSize{ 0 };
static int64_t pts{ 0 };

//conversion context;
SwrContext* swr{};

uint32_t i{ 0 };
int audiostream{ -1 };


void cleanUp() 
{
 avcodec_free_context(&pOutCodecCtx);;
 avio_closep(&(pOutFormatCtx)->pb);
 avformat_free_context(pOutFormatCtx);
 pOutFormatCtx = nullptr;
}

int main()
{

/*
* section to setup input file
*/
if (avformat_open_input(&pFormatCtx, filename.data(), nullptr, nullptr) != 0) {
 cout << "could not open file " << filename << endl;
 return -1;
}
if (avformat_find_stream_info(pFormatCtx, nullptr) < 0) {
 cout << "Could not retrieve stream information from file " << filename << endl;
 return -1;
}
av_dump_format(pFormatCtx, 0, filename.c_str(), 0);

for (i = 0; i < pFormatCtx->nb_streams; i++) {
 if (pFormatCtx->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
 audiostream = i;
 break;
 }
}
if (audiostream == -1) {
 cout << "did not find audio stream" << endl;
 return -1;
}

pStream = pFormatCtx->streams[audiostream];
const AVCodec* pCodec{ avcodec_find_decoder(pStream->codecpar->codec_id) };
pCodecCtx = avcodec_alloc_context3(pCodec);
avcodec_parameters_to_context(pCodecCtx, pStream->codecpar);
if (avcodec_open2(pCodecCtx, pCodec, nullptr)) {
 cout << "could not open codec" << endl;
 return -1;
}

/*
* section to set up output file which is a G711 audio
*/
if (avio_open(&pOutIoContext, outFilename.data(), AVIO_FLAG_WRITE)) {
 cout << "could not open out put file" << endl;
 return -1;
}
if (!(pOutFormatCtx = avformat_alloc_context())) {
 cout << "could not create format conext" << endl;
 cleanUp();
 return -1;
}
pOutFormatCtx->pb = pOutIoContext;
if (!(pOutFormatCtx->oformat = av_guess_format(nullptr, outFilename.data(), nullptr))) {
 cout << "could not find output file format" << endl;
 cleanUp();
 return -1;
}
if (!(pOutFormatCtx->url = av_strdup(outFilename.data()))) {
 cout << "could not allocate file name" << endl;
 cleanUp();
 return -1;
}
if (!(pOutCodec = avcodec_find_encoder(AV_CODEC_ID_PCM_ALAW))) {
 cout << "codec not found" << endl;
 cleanUp();
 return -1;
}
if (!(pOutStream = avformat_new_stream(pOutFormatCtx, nullptr))) {
 cout << "could not create new stream" << endl;
 cleanUp();
 return -1;
}
if (!(pOutCodecCtx = avcodec_alloc_context3(pOutCodec))) {
 cout << "could not allocate codec context" << endl;
 return -1;
}
av_channel_layout_default(&pOutCodecCtx->ch_layout, OUTPUT_CHANNELS);
pOutCodecCtx->sample_rate = SAMPLE_RATE;
pOutCodecCtx->sample_fmt = pOutCodec->sample_fmts[0];
pOutCodecCtx->bit_rate = OUT_BIT_RATE;

//setting sample rate for the container
pOutStream->time_base.den = SAMPLE_RATE;
pOutStream->time_base.num = 1;
if (pOutFormatCtx->oformat->flags & AVFMT_GLOBALHEADER)
 pOutCodecCtx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;

if (avcodec_open2(pOutCodecCtx, pOutCodec, nullptr)) {
 cout << "could not open output codec" << endl;
 cleanUp();
 return -1;
}
if ((avcodec_parameters_from_context(pOutStream->codecpar, pOutCodecCtx)) < 0) {
 cout << "could not initialize stream parameters" << endl;
} 

AVPacket* packet = av_packet_alloc();

swr = swr_alloc();
swr_alloc_set_opts2(&swr, &pOutCodecCtx->ch_layout, pOutCodecCtx->sample_fmt, pOutCodecCtx->sample_rate,&pCodecCtx->ch_layout, pCodecCtx->sample_fmt, pCodecCtx->sample_rate, 0, nullptr);
swr_init(swr);

int ret{};
int bSize{};
while (av_read_frame(pFormatCtx, packet) >= 0) {
 AVFrame* pFrame = av_frame_alloc();
 AVFrame* pOutFrame = av_frame_alloc();
 if (packet->stream_index == audiostream) {
 ret = avcodec_send_packet(pCodecCtx, packet);
 while (ret >= 0) {
 ret = avcodec_receive_frame(pCodecCtx, pFrame);
 if (ret == AVERROR(EAGAIN))
 continue;
 else if (ret == AVERROR_EOF)
 break;
 dstNmbrSamples = av_rescale_rnd(swr_get_delay(swr, pCodecCtx->sample_rate) + pFrame->nb_samples, pOutCodecCtx->sample_rate, pCodecCtx->sample_rate, AV_ROUND_UP);
 if ((av_samples_alloc_array_and_samples(&convertedSamplesBuffer, &dstLineSize, pOutCodecCtx->ch_layout.nb_channels,dstNmbrSamples, pOutCodecCtx->sample_fmt, 0)) < 0) {
 cout << "coult not allocate samples array and buffer" << endl;
 }
 int channel_samples_count{ 0 };
 channel_samples_count = swr_convert(swr, convertedSamplesBuffer, dstNmbrSamples, (const uint8_t**)pFrame->data, pFrame->nb_samples);
 bSize = av_samples_get_buffer_size(&dstLineSize, pOutCodecCtx->ch_layout.nb_channels, channel_samples_count, pOutCodecCtx->sample_fmt, 0);
 cout << "no of samples is " << channel_samples_count << " the buffer size " << bSize << endl;
 pOutFrame->nb_samples = channel_samples_count;
 av_channel_layout_copy(&pOutFrame->ch_layout, &pOutCodecCtx->ch_layout);
 pOutFrame->format = pOutCodecCtx->sample_fmt;
 pOutFrame->sample_rate = pOutCodecCtx->sample_rate;
 if ((av_frame_get_buffer(pOutFrame, 0)) < 0) {
 cout << "could not allocate output frame samples " << endl;
 av_frame_free(&pOutFrame);
 }
 
 //populate out frame buffer
 av_frame_make_writable(pOutFrame);
 for (int i{ 0 }; i < bSize; i++) {
 pOutFrame->data[0][i] = convertedSamplesBuffer[0][i];
 cout << pOutFrame->data[0][i];
 }
 if (pOutFrame) {
 pOutFrame->pts = pts;
 pts += pOutFrame->nb_samples;
 }
 int res = avcodec_send_frame(pOutCodecCtx, pOutFrame);
 if (res < 0) {
 cout << "error sending frame to encoder" << endl;
 cleanUp();
 return -1;
 }
 //int er = avformat_write_header(pOutFormatCtx,nullptr);
 AVPacket* pOutPacket = av_packet_alloc();
 pOutPacket->time_base.num = 1;
 pOutPacket->time_base.den = 8000;
 if (pOutPacket == nullptr) {
 cout << "unable to allocate packet" << endl;
 }
 while (res >= 0) {
 res = avcodec_receive_packet(pOutCodecCtx, pOutPacket);
 if (res == AVERROR(EAGAIN))
 continue;
 else if (ret == AVERROR_EOF)
 break;
 av_packet_rescale_ts(pOutPacket, pOutCodecCtx->time_base, pOutFormatCtx->streams[0]->time_base);
 //av_dump_format(pOutFormatCtx, 0, outFilename.c_str(), 1);
 if (av_write_frame(pOutFormatCtx, pOutPacket) < 0) {
 cout << "could not write frame" << endl;
 }
 }
 }
}
 av_frame_free(&pFrame);
 av_frame_free(&pOutFrame);
}
if (av_write_trailer(pOutFormatCtx) < 0) {
 cout << "could not write file trailer" << endl;
}
swr_free(&swr);
avcodec_free_context(&pOutCodecCtx);
av_packet_free(&packet);
}
</iostream>


Error/Exception


The exception is thrown when I call


if (av_write_frame(pOutFormatCtx, pOutPacket) < 0) { cout << "could not write frame" << endl; }

I also called this line

//int er = avformat_write_header(pOutFormatCtx,nullptr);


to see if I will get an exception but it did not throw any exception.


I have spent weeks on this issue with no success.
My goal is to take any audio from a file an be able to resample it if need be, and transcode it to PCM_ALAW.
I will appreciate any help I can get.