Recherche avancée

Sur d’autres sites (5881)

• Generating 64kbps audio-only mpegts for HTTP Live segmenter to meet 64kbps audio only requirement

  14 juin 2013, par Pobre

  I am trying to convert our mp4 files into mpeg-ts and segment it into .ts files for my iphone app to play. I am using Carson McDonalds's HTTP-Live-Video-Stream-Segmenter-and-Distributor to do that.

  I got his stuff complied and working correctly. I am currently trying to meet Apple's requirement where I need to provide a baseline 64 kbps audio only stream to my m3u8 playlist.
  Carson doesn't seem to have a profile for that.

  I need to be able to generate 64kbps audio-only stream from mp4, and turn that into mpeg-ts for the segmenter into ts. I am trying to find the right ffmpeg command that will validate without problem using Apple's mediastreamvalidator.

  So far I modified an existing encoding profile to try to achieve 64kbps total :

  ffmpeg -er 4 -i %s -f mpegts -acodec libmp3lame -ar 22050 -ab 32k -s 240x180 -vcodec libx264 -b 16k -flags +loop+mv4 -cmp 256 -partitions +parti4x4+partp8x8+partb8x8 -subq 7 -trellis 1 -refs 5 -coder 0 -me_range 16 -keyint_min 25 -sc_threshold 40 -i_qfactor 0.71 -bt 64k -maxrate 16k -bufsize 16k -rc_eq 'blurCplx^(1-qComp)' -qcomp 0.6 -qmin 10 -qmax 51 -qdiff 4 -level 30 -aspect 4:3 -r 10 -g 30 -async 2 - | %s %s %s %s %s

  but then when I try to validate it using mediastreamvalidator, it gives error after few ts :

  Playlist Validation : OK

  Segments :

  sample_cell_4x3_64k-00001.ts :

  WARNING : Media segment exceeds target duration of 10.00 seconds by 1.30 seconds (segment duration is 11.30 seconds)

  sample_cell_4x3_64k-00002.ts :

  WARNING : Media segment exceeds target duration of 10.00 seconds by 1.40 seconds (segment duration is 11.40 seconds)

  ....
  ....

  sample_cell_4x3_64k-00006.ts :

  ERROR : (-1) Unknown video codec : 1836069494 (program 0, track 0)
  ERROR : (-1) failed to parse segment as either an MPEG-2 TS or an ES

  sample_cell_4x3_64k-00007.ts :

  ERROR : (-1) Unknown video codec : 1836069494 (program 0, track 0)
  ERROR : (-1) failed to parse segment as either an MPEG-2 TS or an ES

  ....
  ....
  Average segment duration : 10.26 seconds
  Average segment bitrate : 376797.92 bps
  Average segment structural overhead : 349242.17 bps (92.69 %)

  Is there someway I can generate this correctly with just audio which totals 64kbps and turn it into mpeg-ts ready to be segmented and validated correctly ?

  Am I approaching the problem right ?

• Decoding an MKA audio file into raw data (MKA Audio to raw audio data)

  9 octobre 2020, par bbdd

  My task is to open an existing audio file with the mka extension (Matroska container) and extract the raw audio data. This example shows only an example of extracting raw data from an mp2 file. I do not know how to do this with the mka container. I would like to have something like this :

  


  UPD

  


  I found an option to save audio data in the format in which it was recorded in the audio file. An example is shown below.

  


  PS. This is only a test version and most likely there are memory leaks and other problems.

  


  &#xA;#include <qfile>&#xA;#include <qdebug>&#xA;#include "audiodecoder.h"&#xA;&#xA;int main(int argc, char* argv[])&#xA;{&#xA;    AudioDecoder decoder("/home/test/test.mka");&#xA;    bool started = decoder.start();&#xA;    if (!started) {&#xA;        return EXIT_FAILURE;&#xA;    }&#xA;&#xA;    QFile file("/home/test/rawData.bin");&#xA;    file.open(QIODevice::WriteOnly);&#xA;&#xA;    while (true) {&#xA;        auto data = decoder.getData(255);&#xA;        if (data.isEmpty()) {&#xA;            break;&#xA;        }&#xA;        file.write(data.data(), data.size());&#xA;    }&#xA;    file.close();&#xA;    return EXIT_SUCCESS;&#xA;}&#xA;&#xA;</qdebug></qfile>

  


  audiodecoder.h

  


  class AudioDecoder {
public:
    AudioDecoder(const QString& fileName);
    AudioDecoder& operator=(const AudioDecoder& rhs) = delete;
    AudioDecoder& operator=(AudioDecoder&& rhs) = delete;
    AudioDecoder(const AudioDecoder& rhs) = delete;
    AudioDecoder(AudioDecoder&& rhs) = delete;
    virtual ~AudioDecoder(void);

    virtual bool start(void) noexcept;
    virtual QByteArray getData(const quint16& size) noexcept;
    virtual bool stop(void) noexcept;

protected:
    bool m_initialized;
    QString m_fileName;

    AVFrame* p_frame = nullptr;
    AVPacket* p_packet = nullptr;
    AVCodecContext* p_cdcCtx = nullptr;
    AVFormatContext* p_frmCtx = nullptr;
};


  


  audiodecoder.cpp

  


  
static void logging(const char* message)
{
    qDebug() << message;
}

AudioDecoder::AudioDecoder(const QString& fileName)
    : m_initialized(false)
    , m_fileName(fileName)
    , p_cdcCtx(nullptr)
    , p_frmCtx(nullptr)
{
    av_register_all();
}

QByteArray AudioDecoder::getData(const quint16& dataSize) noexcept
{
    QByteArray data;
    qint32 response = 0;
    if (av_read_frame(p_frmCtx, p_packet) >= 0) {
        //logging(QString("AVPacket->pts %1").arg(p_packet->pts).toStdString().c_str());
        //response = decode_packet(p_packet, p_cdcCtx, p_frame);
        response = avcodec_send_packet(p_cdcCtx, p_packet);
        if (response < 0) {
            logging("Error while sending a packet to the decoder");
            return {};
        }
        while (response >= 0) {
            response = avcodec_receive_frame(p_cdcCtx, p_frame);
            if (response == AVERROR(EAGAIN) || response == AVERROR_EOF) {
                break;
            }
            else if (response < 0) {
                logging("Error while receiving a frame from the decoder");
                return {};
            }
            if (response >= 0) {
                logging(QString("Frame %1 (type=%2, size=%3 bytes) pts %4 key_frame %5 [DTS %6], duration[%7]")
                            .arg(p_cdcCtx->frame_number)
                            .arg(av_get_picture_type_char(p_frame->pict_type))
                            .arg(p_frame->pkt_size)
                            .arg(p_frame->pts)
                            .arg(p_frame->key_frame)
                            .arg(p_frame->coded_picture_number)
                            .arg(p_frame->pkt_duration)
                            .toStdString()
                            .c_str());

                for (int i = 0; i < p_frame->linesize[0]; ++i) {
                    data.push_back(p_frame->data[0][i]);
                }
            }
        }
        av_packet_unref(p_packet);
        return data;
    }
    return {};
}

bool AudioDecoder::start(void) noexcept
{
    if (m_initialized) {
        return true;
    }

    int error;
    // Open the input file to read from it.
    if ((error = avformat_open_input(&p_frmCtx,
             m_fileName.toStdString().c_str(), nullptr, nullptr))
        < 0) {
        qDebug() << "Could not open input file: " << m_fileName;
        p_frmCtx = nullptr;
        return false;
    }
    // Get information on the input file (number of streams etc.).
    if ((error = avformat_find_stream_info(p_frmCtx, nullptr)) < 0) {
        avformat_close_input(&p_frmCtx);
        qDebug() << __LINE__;
        return false;
    }
    // Make sure that there is only one stream in the input file.
    if ((p_frmCtx)->nb_streams != 1) {
        avformat_close_input(&p_frmCtx);
        qDebug() << __LINE__;
        return false;
    }

    if (p_frmCtx->streams[0]->codecpar->codec_type != AVMEDIA_TYPE_AUDIO) {
        avformat_close_input(&p_frmCtx);
        qDebug() << __LINE__;
        return false;
    }

    // Find a decoder for the audio stream.
    AVCodec* input_codec = nullptr;
    if (!(input_codec = avcodec_find_decoder((p_frmCtx)->streams[0]->codecpar->codec_id))) {
        avformat_close_input(&p_frmCtx);
        qDebug() << __LINE__;
        return false;
    }
    // Allocate a new decoding context.
    AVCodecContext* avctx = avcodec_alloc_context3(input_codec);
    if (!avctx) {
        avformat_close_input(&p_frmCtx);
        qDebug() << __LINE__;
        return false;
    }
    // Initialize the stream parameters with demuxer information.
    error = avcodec_parameters_to_context(avctx, (p_frmCtx)->streams[0]->codecpar);
    if (error < 0) {
        avformat_close_input(&p_frmCtx);
        avcodec_free_context(&avctx);
        qDebug() << __LINE__;
        return false;
    }
    /* Open the decoder for the audio stream to use it later. */
    if ((error = avcodec_open2(avctx, input_codec, NULL)) < 0) {
        avcodec_free_context(&avctx);
        avformat_close_input(&p_frmCtx);
        qDebug() << __LINE__;
        return false;
    }
    /* Save the decoder context for easier access later. */
    p_cdcCtx = avctx;
    av_dump_format(p_frmCtx, 0, m_fileName.toStdString().c_str(), 0);

    p_frame = av_frame_alloc();
    if (!p_frame) {
        logging("failed to allocated memory for AVFrame");
        return false;
    }
    p_packet = av_packet_alloc();
    if (!p_packet) {
        logging("failed to allocated memory for AVPacket");
        return false;
    }
    return m_initialized = true;
}

bool AudioDecoder::stop(void) noexcept
{
    if (p_cdcCtx != nullptr) {
        avcodec_free_context(&p_cdcCtx);
    }
    if (p_frmCtx != nullptr) {
        avformat_close_input(&p_frmCtx);
    }
    return true;
}

AudioDecoder::~AudioDecoder(void)
{
    stop();
}


  


  But the problem in this example is that I didn't implement the ability to get exactly the requested size of audio data. In my case, it's just ignored.

  


• ffmpeg javacv audio - integrate audio into video javacv

  16 juillet 2014, par Cyril Lequeux

  I’ve looked for this issue since a while but I didn’t find out any solution...
  I want to create a video from pictures with javacv (great, it works). But now I want to put in a sync way audio into my video. I heard I had to make a byte array with the sound data and then record() according to the audiobitrate/8 = audiobyterate but i didn’t find a simple sample...I’m not the first asking this question but all the same questions are still unanswered

      IplImage Image = cvLoadImage("data/photo.jpg");
  
      FrameRecorder recorder = FFmpegFrameRecorder.createDefault("out.avi", 1920, 1080);
  
      recorder.setVideoCodec(AV_CODEC_ID_MPEG4);
  
      recorder.setFrameRate(30);
  
      recorder.setFormat("avi");
  
      recorder.start();
  
          for(int s=0; s<10; s++){
  
              for(int i=0; i<30; i++){
  
                  recorder.record(Image);
  
  //Here I want to record my music.mp3 stream
  
              }
  
          }
  
          recorder.stop();

  Please help me, I’m desesperate :p
  Ps : I’m using javacv 0.3 FFMPEG 1.06 OpenCv 2.4.3