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Autres articles (100)
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List of compatible distributions
26 avril 2011, parThe table below is the list of Linux distributions compatible with the automated installation script of MediaSPIP. Distribution nameVersion nameVersion number Debian Squeeze 6.x.x Debian Weezy 7.x.x Debian Jessie 8.x.x Ubuntu The Precise Pangolin 12.04 LTS Ubuntu The Trusty Tahr 14.04
If you want to help us improve this list, you can provide us access to a machine whose distribution is not mentioned above or send the necessary fixes to add (...) -
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 -
Activation de l’inscription des visiteurs
12 avril 2011, parIl est également possible d’activer l’inscription des visiteurs ce qui permettra à tout un chacun d’ouvrir soit même un compte sur le canal en question dans le cadre de projets ouverts par exemple.
Pour ce faire, il suffit d’aller dans l’espace de configuration du site en choisissant le sous menus "Gestion des utilisateurs". Le premier formulaire visible correspond à cette fonctionnalité.
Par défaut, MediaSPIP a créé lors de son initialisation un élément de menu dans le menu du haut de la page menant (...)
Sur d’autres sites (9054)
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Create an audio file and fill it with the RTP package payload using ffmpeg
24 août 2020, par WithoutExperienceMy task is to extract the payload of an
RTPpacket and save it as an audio file. In my case, I get the payload type - 8 (PCMA). I processRTPpackets and store all the payload in a separate buffer. After that, I have to process this buffer and create an audio file. And that's my problem :(

First, the specification says that in payload 8 (PCMA), audio samples are U8(bitRate / clockRate = 64000 / 8000) in size. But I can't even find a decoder for this sample size. What might be the problem ?


Second, if I create synthetic sound, I can't record it in mka audio format.


As a cheat sheet, I use this link.


Based on this example, I wrote something similar :



class AudioGenerater
{
public:

 enum AudioFormatType
 {
 AudioFormat_WAV,
 AudioFormat_MP3
 };

public:

 AudioGenerater();
 ~AudioGenerater() = default;

 void generateAudioFileWithOptions(
 QString fileName,
 QByteArray pcmData,
 AVCodecID codecID,
 int channel,
 int bitRate,
 int sampleRate,
 AVSampleFormat format);

private:

 // init Format
 bool initFormat(QString audioFileName);

private:

 AVCodec *m_AudioCodec = nullptr;
 AVCodecContext *m_AudioCodecContext = nullptr;
 AVFormatContext *m_FormatContext = nullptr;
 AVOutputFormat *m_OutputFormat = nullptr;
};



Cpp file :


AudioGenerater::AudioGenerater()
{
 av_register_all();
 avcodec_register_all();
}

bool AudioGenerater::initFormat(QString audioFileName)
{
 // Create an output Format context
 int result = avformat_alloc_output_context2(&m_FormatContext, nullptr, nullptr, audioFileName.toLocal8Bit().data());
 if (result < 0) {
 return false;
 }

 m_OutputFormat = m_FormatContext->oformat;

 // Create an audio stream
 AVStream *audioStream = avformat_new_stream(m_FormatContext, m_AudioCodec);
 if (audioStream == nullptr) {
 avformat_free_context(m_FormatContext);
 return false;
 }

 // Set the parameters in the stream
 audioStream->id = m_FormatContext->nb_streams - 1;
 audioStream->time_base = { 1, 64000 };
 result = avcodec_parameters_from_context(audioStream->codecpar, m_AudioCodecContext);
 if (result < 0) {
 avformat_free_context(m_FormatContext);
 return false;
 }

 // Print FormatContext information
 av_dump_format(m_FormatContext, 0, audioFileName.toLocal8Bit().data(), 1);

 // Open file IO
 if (!(m_OutputFormat->flags & AVFMT_NOFILE)) {
 result = avio_open(&m_FormatContext->pb, audioFileName.toLocal8Bit().data(), AVIO_FLAG_WRITE);
 if (result < 0) {
 avformat_free_context(m_FormatContext);
 return false;
 }
 }

 return true;
}

void AudioGenerater::generateAudioFileWithOptions(
 QString _fileName,
 QByteArray _pcmData,
 AVCodecID _codecID,
 int _channel,
 int _bitRate,
 int _sampleRate,
 AVSampleFormat _format)
{
 AVCodecID codecID = _codecID;
 // Find Codec
 m_AudioCodec = avcodec_find_encoder(codecID);
 if (m_AudioCodec == nullptr) {
 return;
 }

 // Create an encoder context
 m_AudioCodecContext = avcodec_alloc_context3(m_AudioCodec);
 if (m_AudioCodecContext == nullptr){
 return;
 }

 // Setting parameters
 m_AudioCodecContext->bit_rate = _bitRate;
 m_AudioCodecContext->sample_rate = _sampleRate;
 m_AudioCodecContext->sample_fmt = _format;
 m_AudioCodecContext->channels = _channel;


 m_AudioCodecContext->channel_layout = av_get_default_channel_layout(_channel);
 m_AudioCodecContext->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;

 // Turn on the encoder
 int result = avcodec_open2(m_AudioCodecContext, m_AudioCodec, nullptr);
 if (result < 0) {
 avcodec_free_context(&m_AudioCodecContext);
 if (m_FormatContext != nullptr)
 avformat_free_context(m_FormatContext);
 return;
 }

 // Create a package
 if (!initFormat(_fileName)) {
 avcodec_free_context(&m_AudioCodecContext);
 if (m_FormatContext != nullptr)
 avformat_free_context(m_FormatContext);
 return;
 }

 // write to the file header
 result = avformat_write_header(m_FormatContext, nullptr);
 if (result < 0) {
 avcodec_free_context(&m_AudioCodecContext);
 if (m_FormatContext != nullptr)
 avformat_free_context(m_FormatContext);
 return;
 }

 // Create Frame
 AVFrame *frame = av_frame_alloc();
 if (frame == nullptr) {
 avcodec_free_context(&m_AudioCodecContext);
 if (m_FormatContext != nullptr)
 avformat_free_context(m_FormatContext);
 return;
 }

 int nb_samples = 0;
 if (m_AudioCodecContext->codec->capabilities & AV_CODEC_CAP_VARIABLE_FRAME_SIZE) {
 nb_samples = 10000;
 }
 else {
 nb_samples = m_AudioCodecContext->frame_size;
 }

 // Set the parameters of the Frame
 frame->nb_samples = nb_samples;
 frame->format = m_AudioCodecContext->sample_fmt;
 frame->channel_layout = m_AudioCodecContext->channel_layout;

 // Apply for data memory
 result = av_frame_get_buffer(frame, 0);
 if (result < 0)
 {
 av_frame_free(&frame); {
 avcodec_free_context(&m_AudioCodecContext);
 if (m_FormatContext != nullptr)
 avformat_free_context(m_FormatContext);
 return;
 }
 }

 // Set the Frame to be writable
 result = av_frame_make_writable(frame);
 if (result < 0)
 {
 av_frame_free(&frame); {
 avcodec_free_context(&m_AudioCodecContext);
 if (m_FormatContext != nullptr)
 avformat_free_context(m_FormatContext);
 return;
 }
 }

 int perFrameDataSize = frame->linesize[0];
 int count = _pcmData.size() / perFrameDataSize;
 bool needAddOne = false;
 if (_pcmData.size() % perFrameDataSize != 0)
 {
 count++;
 needAddOne = true;
 }

 int frameCount = 0;
 for (int i = 0; i < count; ++i)
 {
 // Create a Packet
 AVPacket *pkt = av_packet_alloc();
 if (pkt == nullptr) {
 avcodec_free_context(&m_AudioCodecContext);
 if (m_FormatContext != nullptr)
 avformat_free_context(m_FormatContext);
 return;
 }
 av_init_packet(pkt);

 if (i == count - 1)
 perFrameDataSize = _pcmData.size() % perFrameDataSize;

 // Synthesize WAV files
 memset(frame->data[0], 0, perFrameDataSize);
 memcpy(frame->data[0], &(_pcmData.data()[perFrameDataSize * i]), perFrameDataSize);


 frame->pts = frameCount++;
 // send Frame
 result = avcodec_send_frame(m_AudioCodecContext, frame);
 if (result < 0)
 continue;

 // Receive the encoded Packet
 result = avcodec_receive_packet(m_AudioCodecContext, pkt);
 if (result < 0)
 {
 av_packet_free(&pkt);
 continue;
 }

 // write to file
 av_packet_rescale_ts(pkt, m_AudioCodecContext->time_base, m_FormatContext->streams[0]->time_base);
 pkt->stream_index = 0;
 result = av_interleaved_write_frame(m_FormatContext, pkt);
 if (result < 0)
 continue;

 av_packet_free(&pkt);
 }

 // write to the end of the file
 av_write_trailer(m_FormatContext);
 // Close file IO
 avio_closep(&m_FormatContext->pb);
 // Release Frame memory
 av_frame_free(&frame);

 avcodec_free_context(&m_AudioCodecContext);
 if (m_FormatContext != nullptr)
 avformat_free_context(m_FormatContext);
}


Main :


int main(int argc, char *argv[])
{
 QCoreApplication a(argc, argv);
 AudioGenerater generator;
 
 av_log_set_level(AV_LOG_DEBUG);
 av_log_set_flags(av_log_get_flags());
 
 QFile file("pcma.raw");
 if (!file.open(QIODevice::ReadOnly)) {
 return EXIT_FAILURE;
 }
 QByteArray pcmaData(file.readAll());
 
 generator.generateAudioFileWithOptions(
 "test.mka",
 pcmaData,
 AV_CODEC_ID_PCM_ALAW,
 1, 64000, 8000,
 AV_SAMPLE_FMT_U8);

 return a.exec();
}



After I run the program, here are the errors that come out :


[pcm_alaw @ 0x19229a0] Specified sample format u8 is invalid or not supported


But if I set s16p instead of U8 format :


generator.generateAudioFileWithOptions(
 "test.mka",
 pcmaData,
 AV_CODEC_ID_PCM_ALAW,
 1, 64000, 8000,
 AV_SAMPLE_FMT_S16P);


I get twice as fast audio track, with a lot of noise. I need a sequential instruction encoding raw audio data into an audio file. Unfortunately, I don't understand what my error is.


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Cannot concatenate videos ffmpeg [on hold]
1er mai 2014, par Paul PrescodI have a bitmap that I would like to concatenate to the front of many videos as a sort of title screen or disclaimer screen.
I try to turn it into a video with the same attributes as the rest of the video. So first I introspect the video :
ffmpeg version 2.2.1 Copyright (c) 2000-2014 the FFmpeg developers
built on Apr 11 2014 22:50:38 with Apple LLVM version 5.1 (clang-503.0.40) (based on LLVM 3.4svn)
configuration: --prefix=/usr/local/Cellar/ffmpeg/2.2.1 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-nonfree --enable-hardcoded-tables --enable-avresample --enable-vda --cc=clang --host-cflags= --host-ldflags= --enable-libx264 --enable-libfaac --enable-libmp3lame --enable-libxvid
libavutil 52. 66.100 / 52. 66.100
libavcodec 55. 52.102 / 55. 52.102
libavformat 55. 33.100 / 55. 33.100
libavdevice 55. 10.100 / 55. 10.100
libavfilter 4. 2.100 / 4. 2.100
libavresample 1. 2. 0 / 1. 2. 0
libswscale 2. 5.102 / 2. 5.102
libswresample 0. 18.100 / 0. 18.100
libpostproc 52. 3.100 / 52. 3.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'EO1.mp4':
Metadata:
major_brand : isom
minor_version : 512
compatible_brands: isomiso2avc1mp41
creation_time : 1970-01-01 00:00:00
encoder : Lavf52.78.3
Duration: 00:00:17.77, start: 0.000000, bitrate: 582 kb/s
Stream #0:0(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 584x328 [SAR 1:1 DAR 73:41], 512 kb/s, 23.98 fps, 23.98 tbr, 1199 tbn, 47.96 tbc (default)
Metadata:
creation_time : 1970-01-01 00:00:00
handler_name : VideoHandler
Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 64 kb/s (default)
Metadata:
creation_time : 1970-01-01 00:00:00
handler_name : SoundHandlerThen I try and create a similar file :
/usr/local/Cellar/ffmpeg/2.2.1/bin/ffmpeg -y -loop 1 -i Disclaimer.png -c:v libx264 -r 23.98 -t 5 -pix_fmt yuv420p -profile:v main disclaimer.mp4
It seems to work okay. The video plays I would expect it to. The attributes turn out very similar. Here is a diff :
< Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'disclaimer.mp4':
---
> Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'EO1.mp4':
18,20c18,21
< encoder : Lavf55.33.100
< Duration: 00:00:05.01, start: 0.000000, bitrate: 21 kb/s
< Stream #0:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 584x328 [SAR 1:1 DAR 73:41], 17 kb/s, 23.98 fps, 23.98 tbr, 19184 tbn, 47.96 tbc (default)
---
> creation_time : 1970-01-01 00:00:00
> encoder : Lavf52.78.3
> Duration: 00:00:17.77, start: 0.000000, bitrate: 582 kb/s
> Stream #0:0(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 584x328 [SAR 1:1 DAR 73:41], 512 kb/s, 23.98 fps, 23.98 tbr, 1199 tbn, 47.96 tbc (default)
21a23
22a25,28
> Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 64 kb/s (default)
> Metadata:
> handler_name : SoundHandlerBut when I try to concatenate, I get errors.
> $ cat temporary.txt
file disclaimer.mp4
file EO1.mp4/usr/local/Cellar/ffmpeg/2.2.1/bin/ffmpeg -y -f concat -i temporary.txt -c copy output.mp4
[concat @ 0x7fd880806600] Estimating duration from bitrate, this may be inaccurate
Input #0, concat, from 'temporary.txt':
Duration: 00:00:00.02, start: 0.000000, bitrate: 17 kb/s
Stream #0:0: Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 584x328 [SAR 1:1 DAR 73:41], 17 kb/s, 23.98 fps, 23.98 tbr, 19184 tbn, 47.96 tbc
Output #0, mp4, to 'output.mp4':
Metadata:
encoder : Lavf55.33.100
Stream #0:0: Video: h264 ([33][0][0][0] / 0x0021), yuv420p, 584x328 [SAR 1:1 DAR 73:41], q=2-31, 17 kb/s, 23.98 fps, 19184 tbn, 19184 tbc
Stream mapping:
Stream #0:0 -> #0:0 (copy)
Press [q] to stop, [?] for help
[mp4 @ 0x7fd880829a00] Non-monotonous DTS in output stream 0:0; previous: 93600, current: 5951; changing to 93601. This may result in incorrect timestamps in the output file.
[concat @ 0x7fd880806600] Invalid stream index 1
[mp4 @ 0x7fd880829a00] Non-monotonous DTS in output stream 0:0; previous: 93601, current: 6001; changing to 93602. This may result in incorrect timestamps in the output file.
[concat @ 0x7fd880806600] Invalid stream index 1
[mp4 @ 0x7fd880829a00] Non-monotonous DTS in output stream 0:0; previous: 93602, current: 6051; changing to 93603. This may result in incorrect timestamps in the output file....
frame= 546 fps=0.0 q=-1.0 Lsize= 1127kB time=00:00:04.90 bitrate=1882.9kbits/s
video:1123kB audio:0kB subtitle:0 data:0 global headers:0kB muxing overhead 0.349865%The output looks like it only has my disclaimer file in it, not the rest of the video.
I’m also confused why it feels like it needs to "estimate" anything. It knows the input FPS and input durations. I’m not sure if this is the problem or not. Maybe its just a bug.
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Where can discord bots find the ffmpeg buildpack heroku
11 août 2020, par IsiahI'm making a bot which outputs a mp3 file to the voice chat with discord.py, It works localy by using :


vc.play(discord.FFmpegPCMAudio(executable="ffmpeg/bin/ffmpeg.exe", source=noise.mp3))

However I'm now hosting it on Heroku, i have the buildpack installed but how can my code access it in replacement of the code above


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