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• getting video duration with ffmpeg, php function

  17 novembre 2014, par keithp

  I have FFMpeg installed and I know it’s functional, but i’m trying to get the duration time from a flv video through PHP but when I use this code :

  function mbmGetFLVDuration($file)

  /*  
  
  * Determine video duration with ffmpeg   
  
  * ffmpeg should be installed on your server.  
  
  */  
  
  
  
  //$time = 00:00:00.000 format   
  
  $ffmpeg = "../ffmpeg/ffmpeg";
  
  
  
  $time =  exec("$ffmpeg -i $file 2>&1 | grep 'Duration' | cut -d ' ' -f 4 | sed s/,//");   
  
  
  
  $duration = explode(":",$time);   
  
  $duration_in_seconds = $duration[0]*3600 + $duration[1]*60+ round($duration[2]);   
  
  
  
  return $duration_in_seconds;   

  and :

  $duration = mbmGetFLVDuration(’http://www.videoaddsite.com/videos/intro.flv’) ;
  echo $duration ;

  I get an output of 220. THe video is 3:40. Can any help me on what i’m doing wrong, or if there’s something else I can use ?

• using ffmpeg headers with a c program

  15 avril 2014, par orpgol

  i'm trying to learn ffmpeg from drangers guide (for school), i have a mac so the first thing i did was to use Macports and get ffmpeg and sdl...

  now when i'm tried to compile drangers code, the compiler didnt recognize the headers...
  so i used -I on gnu when compiling, and also gave the "whole/path/name..." to the headers,
  but i always get an error on some header missing...

  below i will put the code with my corrections.

  i also tried including all the headers that i get an error about but there is always another header the compiler can't find

  i tried both gnu (on console) and on xcode.

  // tutorial01.c
  
  // Code based on a tutorial by Martin Bohme (boehme@inb.uni-luebeckREMOVETHIS.de)
  
  // Tested on Gentoo, CVS version 5/01/07 compiled with GCC 4.1.1
  
  
  
  // A small sample program that shows how to use libavformat and libavcodec to
  
  // read video from a file.
  
  //
  
  // Use
  
  //
  
  // gcc -o tutorial01 tutorial01.c -lavformat -lavcodec -lz
  
  //
  
  // to build (assuming libavformat and libavcodec are correctly installed
  
  // your system).
  
  //
  
  // Run using
  
  //
  
  // tutorial01 myvideofile.mpg
  
  //
  
  // to write the first five frames from "myvideofile.mpg" to disk in PPM
  
  // format.
  
  
  
  #include "/opt/local/include/libavcodec/avcodec.h"
  
  #include "/opt/local/include/libavformat/avformat.h"
  
  
  
  #include 
  
  
  
  void SaveFrame(AVFrame *pFrame, int width, int height, int iFrame) {
  
    FILE *pFile;
  
    char szFilename[32];
  
    int  y;
  
  
  
    // Open file
  
    sprintf(szFilename, "frame%d.ppm", iFrame);
  
    pFile=fopen(szFilename, "wb");
  
    if(pFile==NULL)
  
      return;
  
  
  
    // Write header
  
    fprintf(pFile, "P6\n%d %d\n255\n", width, height);
  
  
  
    // Write pixel data
  
    for(y=0; ydata[0]+y*pFrame->linesize[0], 1, width*3, pFile);
  
  
  
    // Close file
  
    fclose(pFile);
  
  }
  
  
  
  int main(int argc, char *argv[]) {
  
    AVFormatContext *pFormatCtx;
  
    int             i, videoStream;
  
    AVCodecContext  *pCodecCtx;
  
    AVCodec         *pCodec;
  
    AVFrame         *pFrame; 
  
    AVFrame         *pFrameRGB;
  
    AVPacket        packet;
  
    int             frameFinished;
  
    int             numBytes;
  
    uint8_t         *buffer;
  
  
  
    if(argc < 2) {
  
      printf("Please provide a movie file\n");
  
      return -1;
  
    }
  
    // Register all formats and codecs
  
    av_register_all();
  
  
  
    // Open video file
  
    if(av_open_input_file(&pFormatCtx, argv[1], NULL, 0, NULL)!=0)
  
      return -1; // Couldn't open file
  
  
  
    // Retrieve stream information
  
    if(av_find_stream_info(pFormatCtx)<0)
  
      return -1; // Couldn't find stream information
  
  
  
    // Dump information about file onto standard error
  
    dump_format(pFormatCtx, 0, argv[1], 0);
  
  
  
    // Find the first video stream
  
    videoStream=-1;
  
    for(i=0; inb_streams; i++)
  
      if(pFormatCtx->streams[i]->codec->codec_type==CODEC_TYPE_VIDEO) {
  
        videoStream=i;
  
        break;
  
      }
  
    if(videoStream==-1)
  
      return -1; // Didn't find a video stream
  
  
  
    // Get a pointer to the codec context for the video stream
  
    pCodecCtx=pFormatCtx->streams[videoStream]->codec;
  
  
  
    // Find the decoder for the video stream
  
    pCodec=avcodec_find_decoder(pCodecCtx->codec_id);
  
    if(pCodec==NULL) {
  
      fprintf(stderr, "Unsupported codec!\n");
  
      return -1; // Codec not found
  
    }
  
    // Open codec
  
    if(avcodec_open(pCodecCtx, pCodec)<0)
  
      return -1; // Could not open codec
  
  
  
    // Allocate video frame
  
    pFrame=avcodec_alloc_frame();
  
  
  
    // Allocate an AVFrame structure
  
    pFrameRGB=avcodec_alloc_frame();
  
    if(pFrameRGB==NULL)
  
      return -1;
  
  
  
    // Determine required buffer size and allocate buffer
  
    numBytes=avpicture_get_size(PIX_FMT_RGB24, pCodecCtx->width,
  
                    pCodecCtx->height);
  
    buffer=(uint8_t *)av_malloc(numBytes*sizeof(uint8_t));
  
  
  
    // Assign appropriate parts of buffer to image planes in pFrameRGB
  
    // Note that pFrameRGB is an AVFrame, but AVFrame is a superset
  
    // of AVPicture
  
    avpicture_fill((AVPicture *)pFrameRGB, buffer, PIX_FMT_RGB24,
  
           pCodecCtx->width, pCodecCtx->height);
  
  
  
    // Read frames and save first five frames to disk
  
    i=0;
  
    while(av_read_frame(pFormatCtx, &packet)>=0) {
  
      // Is this a packet from the video stream?
  
      if(packet.stream_index==videoStream) {
  
        // Decode video frame
  
        avcodec_decode_video(pCodecCtx, pFrame, &frameFinished, 
  
                 packet.data, packet.size);
  
  
  
        // Did we get a video frame?
  
        if(frameFinished) {
  
      // Convert the image from its native format to RGB
  
      img_convert((AVPicture *)pFrameRGB, PIX_FMT_RGB24, 
  
                      (AVPicture*)pFrame, pCodecCtx->pix_fmt, pCodecCtx->width, 
  
                      pCodecCtx->height);
  
  
  
      // Save the frame to disk
  
      if(++i<=5)
  
        SaveFrame(pFrameRGB, pCodecCtx->width, pCodecCtx->height, 
  
              i);
  
        }
  
      }
  
  
  
      // Free the packet that was allocated by av_read_frame
  
      av_free_packet(&packet);
  
    }
  
  
  
    // Free the RGB image
  
    av_free(buffer);
  
    av_free(pFrameRGB);
  
  
  
    // Free the YUV frame
  
    av_free(pFrame);
  
  
  
    // Close the codec
  
    avcodec_close(pCodecCtx);
  
  
  
    // Close the video file
  
    av_close_input_file(pFormatCtx);
  
  
  
    return 0;
  
  }

• Wrap audio data of the pcm_alaw type into an MKA audio file using the ffmpeg API

  19 septembre 2020, par bbdd

  Imagine that in my project, I receive RTP packets with the payload type-8, for later saving this load as the Nth part of the audio track. I extract this load from the RTP packet and save it to a temporary buffer :

  


  ...

while ((rtp = receiveRtpPackets()).withoutErrors()) {
   payloadData.push(rtp.getPayloadData());
}

audioGenerator.setPayloadData(payloadData);
audioGenerator.recordToFile();

...


  


  After filling a temporary buffer of a certain size with this payload, I process this buffer, namely, extract the entire payload and encode it using ffmpeg for further saving to an audio file in Matroska format. But I have a problem. Since the payload of the RTP packet is type 8, I have to save the raw audio data of the pcm_alaw format to mka audio format. But when saving raw data pcm_alaw to an audio file, I get these messages from the library :

  


  ...

[libopus @ 0x18eff60] Queue input is backward in time
[libopus @ 0x18eff60] Queue input is backward in time
[libopus @ 0x18eff60] Queue input is backward in time
[libopus @ 0x18eff60] Queue input is backward in time

...


  


  When you open an audio file in vlc, nothing is played (the audio track timestamp is missing).

  


  The task of my project is to simply take pcm_alaw data and pack it in a container, in mka format. The best way to determine the codec is to use the av_guess_codec() function, which in turn automatically selects the desired codec ID. But how do I pack the raw data into the container correctly, I do not know.

  


  It is important to note that I can get as raw data any format of this data (audio formats only) defined by the RTP packet type (All types of RTP packet payload). All I know is that in any case, I have to pack the audio data in an mka container.

  


  I also attach the code (borrowed from this resource) that I use :

  


  audiogenerater.h

  


  extern "C"
{
#include "libavformat/avformat.h"
#include "libavcodec/avcodec.h"
#include "libswresample/swresample.h"
}

class AudioGenerater
{
public:

    AudioGenerater();
   ~AudioGenerater() = default;

    void generateAudioFileWithOptions(
            QString        fileName,
            QByteArray     pcmData,
            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;
};


  


  audiogenerater.cpp

  


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

AudioGenerater::~AudioGenerater()
{
    // ... 
}

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, 8000 };
    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,
    int _channel,
    int _bitRate,
    int _sampleRate,
    AVSampleFormat _format)
{
    AVFormatContext* oc;
    if (avformat_alloc_output_context2(
            &oc, nullptr, nullptr, _fileName.toStdString().c_str())
        < 0) {
        qDebug() << "Error in line: " << __LINE__;
        return;
    }
    if (!oc) {
        printf("Could not deduce output format from file extension: using mka.\n");
        avformat_alloc_output_context2(
            &oc, nullptr, "mka", _fileName.toStdString().c_str());
    }
    if (!oc) {
        qDebug() << "Error in line: " << __LINE__;
        return;
    }
    AVOutputFormat* fmt = oc->oformat;
    if (fmt->audio_codec == AV_CODEC_ID_NONE) {
        qDebug() << "Error in line: " << __LINE__;
        return;
    }

    AVCodecID codecID = av_guess_codec(
        fmt, nullptr, _fileName.toStdString().c_str(), nullptr, AVMEDIA_TYPE_AUDIO);
    // Find Codec
    m_AudioCodec = avcodec_find_encoder(codecID);
    if (m_AudioCodec == nullptr) {
        qDebug() << "Error in line: " << __LINE__;
        return;
    }
    // Create an encoder context
    m_AudioCodecContext = avcodec_alloc_context3(m_AudioCodec);
    if (m_AudioCodecContext == nullptr) {
        qDebug() << "Error in line: " << __LINE__;
        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.cpp

  


  int main(int argc, char **argv)
{
    av_log_set_level(AV_LOG_TRACE);

    QFile file("rawDataOfPcmAlawType.bin");
    if (!file.open(QIODevice::ReadOnly)) {
        return EXIT_FAILURE;
    }
    QByteArray rawData(file.readAll());

    AudioGenerater generator;
    generator.generateAudioFileWithOptions(
               "test.mka",
               rawData,
               1, 
               64000, 
               8000,
               AV_SAMPLE_FMT_S16);

    return 0;
}


  


  It is IMPORTANT you help me find the most appropriate way to record pcm_alaw or a different data format in an MKA audio file.

  


  I ask everyone who knows anything to help (there is too little time left to implement this project)