Make Qt Player codec independent

Question

I develop Qt application which can play more then one video file using bellow code.

QMediaPlayer *player;
QString fileName = "C:/username/test.h264";
player->setmedia(QUrl::fromLocalFile(fileName));

In starting I cannot play all types of video file, so I install codec on my system, now when my player start codec decoder start, and my CPU usage reach at high.(Show the bellow Image)

You can see in above image right side bottom corner LAW(Red label) which saw external decoder started.

Now, I want to make my Qt Player codec independent, means I know my player have to play only .h264 file, so I will use only h264 decoder and no need of audio so I will not use audio decoder.

As per my knowledge, QMediaPlayer start decoder when it come in picture, correct me if i am wrong. So What can I do to stop external decoder and decode frame internally and play successfully?

EDIT : code for audio decode using FFmpeg

FFmpegAudio.pro

TARGET = fooAudioFFMPEG
QT       += core gui qml quick widgets
TEMPLATE = app
SOURCES += main.cpp \
    mainwindow.cpp
HEADERS += mainwindow.h \
    wrapper.h
FORMS += mainwindow.ui
QMAKE_CXXFLAGS += -D__STDC_CONSTANT_MACROS

LIBS += -pthread
LIBS += -L/usr/local/lib
LIBS += -lavdevice
LIBS += -lavfilter
LIBS += -lpostproc
LIBS += -lavformat
LIBS += -lavcodec
LIBS += -ldl
LIBS += -lXfixes
LIBS += -lXext
LIBS += -lX11
LIBS += -lasound
LIBS += -lSDL
LIBS += -lx264
LIBS += -lvpx
LIBS += -lvorbisenc
LIBS += -lvorbis
LIBS += -logg
LIBS += -lopencore-amrwb
LIBS += -lopencore-amrnb
LIBS += -lmp3lame
LIBS += -lfaac
LIBS += -lz
LIBS += -lrt
LIBS += -lswscale
LIBS += -lavutil
LIBS += -lm

mainwindow.h

#ifndef MAINWINDOW_H
#define MAINWINDOW_H

#include <QMainWindow>

namespace Ui {
    class MainWindow;
}

class MainWindow : public QMainWindow {
    Q_OBJECT
public:
    MainWindow(QWidget *parent = 0);
    ~MainWindow();

protected:
    void changeEvent(QEvent *e);

private:
    Ui::MainWindow *ui;

private slots:
    void on_pushButton_clicked();
};

#endif // MAINWINDOW_H

wrapper.h

#ifndef WRAPPER_H_
#define WRAPPER_H_

#include <math.h>

#include <libavutil/opt.h>
#include <libavcodec/avcodec.h>
#include <libavutil/channel_layout.h>
#include <libavutil/common.h>
#include <libavutil/imgutils.h>
#include <libavutil/mathematics.h>
#include <libavutil/samplefmt.h>

#define INBUF_SIZE 4096
#define AUDIO_INBUF_SIZE 20480
#define AUDIO_REFILL_THRESH 4096



/* check that a given sample format is supported by the encoder */
static int check_sample_fmt(AVCodec *codec, enum AVSampleFormat sample_fmt)
{
    const enum AVSampleFormat *p = codec->sample_fmts;

    while (*p != AV_SAMPLE_FMT_NONE) {
        if (*p == sample_fmt)
            return 1;
        p++;
    }
    return 0;
}

/* just pick the highest supported samplerate */
static int select_sample_rate(AVCodec *codec)
{
    const int *p;
    int best_samplerate = 0;

    if (!codec->supported_samplerates)
        return 44100;

    p = codec->supported_samplerates;
    while (*p) {
        best_samplerate = FFMAX(*p, best_samplerate);
        p++;
    }
    return best_samplerate;
}

/* select layout with the highest channel count */
static int select_channel_layout(AVCodec *codec)
{
    const uint64_t *p;
    uint64_t best_ch_layout = 0;
    int best_nb_channells   = 0;

    if (!codec->channel_layouts)
        return AV_CH_LAYOUT_STEREO;

    p = codec->channel_layouts;
    while (*p) {
        int nb_channels = av_get_channel_layout_nb_channels(*p);

        if (nb_channels > best_nb_channells) {
            best_ch_layout    = *p;
            best_nb_channells = nb_channels;
        }
        p++;
    }
    return best_ch_layout;
}

/*
 * Audio encoding example
 */
static void audio_encode_example(const char *filename)
{
    AVCodec *codec;
    AVCodecContext *c= NULL;
    AVFrame *frame;
    AVPacket pkt;
    int i, j, k, ret, got_output;
    int buffer_size;
    FILE *f;
    uint16_t *samples;
    float t, tincr;

    printf("Encode audio file %s\n", filename);

    /* find the MP2 encoder */
    codec = avcodec_find_encoder(AV_CODEC_ID_MP2);
    if (!codec) {
        fprintf(stderr, "Codec not found\n");
        exit(1);
    }

    c = avcodec_alloc_context3(codec);
    if (!c) {
        fprintf(stderr, "Could not allocate audio codec context\n");
        exit(1);
    }

    /* put sample parameters */
    c->bit_rate = 64000;

    /* check that the encoder supports s16 pcm input */
    c->sample_fmt = AV_SAMPLE_FMT_S16;
    if (!check_sample_fmt(codec, c->sample_fmt)) {
        fprintf(stderr, "Encoder does not support sample format %s",
                av_get_sample_fmt_name(c->sample_fmt));
        exit(1);
    }

    /* select other audio parameters supported by the encoder */
    c->sample_rate    = select_sample_rate(codec);
    c->channel_layout = select_channel_layout(codec);
    c->channels       = av_get_channel_layout_nb_channels(c->channel_layout);

    /* open it */
    if (avcodec_open2(c, codec, NULL) < 0) {
        fprintf(stderr, "Could not open codec\n");
        exit(1);
    }

    f = fopen(filename, "wb");
    if (!f) {
        fprintf(stderr, "Could not open %s\n", filename);
        exit(1);
    }

    /* frame containing input raw audio */
    frame = avcodec_alloc_frame();
    if (!frame) {
        fprintf(stderr, "Could not allocate audio frame\n");
        exit(1);
    }

    frame->nb_samples     = c->frame_size;
    frame->format         = c->sample_fmt;
    frame->channel_layout = c->channel_layout;

    /* the codec gives us the frame size, in samples,
     * we calculate the size of the samples buffer in bytes */
    buffer_size = av_samples_get_buffer_size(NULL, c->channels, c->frame_size,
                                             c->sample_fmt, 0);
    samples = (uint16_t *)av_malloc(buffer_size);
    if (!samples) {
        fprintf(stderr, "Could not allocate %d bytes for samples buffer\n",
                buffer_size);
        exit(1);
    }
    /* setup the data pointers in the AVFrame */
    ret = avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
                                   (const uint8_t*)samples, buffer_size, 0);
    if (ret < 0) {
        fprintf(stderr, "Could not setup audio frame\n");
        exit(1);
    }

    /* encode a single tone sound */
    t = 0;
    tincr = 2 * M_PI * 440.0 / c->sample_rate;
    for(i=0;i<200;i++) {
        av_init_packet(&pkt);
        pkt.data = NULL; // packet data will be allocated by the encoder
        pkt.size = 0;

        for (j = 0; j < c->frame_size; j++) {
            samples[2*j] = (int)(sin(t) * 10000);

            for (k = 1; k < c->channels; k++)
                samples[2*j + k] = samples[2*j];
            t += tincr;
        }
        /* encode the samples */
        ret = avcodec_encode_audio2(c, &pkt, frame, &got_output);
        if (ret < 0) {
            fprintf(stderr, "Error encoding audio frame\n");
            exit(1);
        }
        if (got_output) {
            fwrite(pkt.data, 1, pkt.size, f);
            av_free_packet(&pkt);
        }
    }

    /* get the delayed frames */
    for (got_output = 1; got_output; i++) {
        ret = avcodec_encode_audio2(c, &pkt, NULL, &got_output);
        if (ret < 0) {
            fprintf(stderr, "Error encoding frame\n");
            exit(1);
        }

        if (got_output) {
            fwrite(pkt.data, 1, pkt.size, f);
            av_free_packet(&pkt);
        }
    }
    fclose(f);

    av_freep(&samples);
    avcodec_free_frame(&frame);
    avcodec_close(c);
    av_free(c);
}

/*
 * Audio decoding.
 */
static void audio_decode_example(const char *outfilename, const char *filename)
{
    AVCodec *codec;
    AVCodecContext *c= NULL;
    int len;
    FILE *f, *outfile;
    uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
    AVPacket avpkt;
    AVFrame *decoded_frame = NULL;

    av_init_packet(&avpkt);

    printf("Decode audio file %s to %s\n", filename, outfilename);

    /* find the mpeg audio decoder */
    codec = avcodec_find_decoder(AV_CODEC_ID_MP2);
    if (!codec) {
        fprintf(stderr, "Codec not found\n");
        exit(1);
    }

    c = avcodec_alloc_context3(codec);
    if (!c) {
        fprintf(stderr, "Could not allocate audio codec context\n");
        exit(1);
    }

    /* open it */
    if (avcodec_open2(c, codec, NULL) < 0) {
        fprintf(stderr, "Could not open codec\n");
        exit(1);
    }

    f = fopen(filename, "rb");
    if (!f) {
        fprintf(stderr, "Could not open %s\n", filename);
        exit(1);
    }
    outfile = fopen(outfilename, "wb");
    if (!outfile) {
        av_free(c);
        exit(1);
    }

    /* decode until eof */
    avpkt.data = inbuf;
    avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);

    while (avpkt.size > 0) {
        int got_frame = 0;

        if (!decoded_frame) {
            if (!(decoded_frame = avcodec_alloc_frame())) {
                fprintf(stderr, "Could not allocate audio frame\n");
                exit(1);
            }
        } else
            avcodec_get_frame_defaults(decoded_frame);

        len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt);
        if (len < 0) {
            fprintf(stderr, "Error while decoding\n");
            exit(1);
        }
        if (got_frame) {
            /* if a frame has been decoded, output it */
            int data_size = av_samples_get_buffer_size(NULL, c->channels,
                                                       decoded_frame->nb_samples,
                                                       c->sample_fmt, 1);
            fwrite(decoded_frame->data[0], 1, data_size, outfile);
        }
        avpkt.size -= len;
        avpkt.data += len;
        avpkt.dts =
        avpkt.pts = AV_NOPTS_VALUE;
        if (avpkt.size < AUDIO_REFILL_THRESH) {
            /* Refill the input buffer, to avoid trying to decode
             * incomplete frames. Instead of this, one could also use
             * a parser, or use a proper container format through
             * libavformat. */
            memmove(inbuf, avpkt.data, avpkt.size);
            avpkt.data = inbuf;
            len = fread(avpkt.data + avpkt.size, 1,
                        AUDIO_INBUF_SIZE - avpkt.size, f);
            if (len > 0)
                avpkt.size += len;
        }
    }

    fclose(outfile);
    fclose(f);

    avcodec_close(c);
    av_free(c);
    avcodec_free_frame(&decoded_frame);
}

/*
 * Main WRAPPER function
 */
void service(){


    /* register all the codecs */
    avcodec_register_all();


    audio_encode_example("test.mp2");
    audio_decode_example("test.sw", "test.mp2");

}

#endif 

main.cpp

#include <QApplication>
#include "mainwindow.h"

extern "C"{
    #include "wrapper.h"
}

int main(int argc, char *argv[])
{
    service(); //calling the function service inside the wrapper

    QApplication a(argc, argv);
    MainWindow w;
    w.show();
    return a.exec();
}

mainwindow.cpp

#include "mainwindow.h"
#include "ui_mainwindow.h"

MainWindow::MainWindow(QWidget *parent) :
    QMainWindow(parent),
    ui(new Ui::MainWindow)
{
    ui->setupUi(this);
}

MainWindow::~MainWindow()
{
    delete ui;
}

void MainWindow::changeEvent(QEvent *e)
{
    QMainWindow::changeEvent(e);
    switch (e->type()) {
    case QEvent::LanguageChange:
        ui->retranslateUi(this);
        break;
    default:
        break;
    }
}

void MainWindow::on_pushButton_clicked()
{
        this->close();
}

mainwindow.ui //Nothing important

Thanks.

Answer 1

Just for playing video files in Qt, you may also use libvlc, after the famous media player (http://www.videolan.org)

Initialization:

libvlc_instance_t *vlcinstance = libvlc_new(0, NULL);
libvlc_media_player_t *player = libvlc_media_player_new(vlcinstance);
libvlc_media_t *media = libvlc_media_new_path(vlcinstance, file_path);

Use some qt widget for displaying the video:

libvlc_media_player_set_drawable(player, some_widget->winId());

Play the video:

libvlc_media_player_set_media(player, media);
libvlc_media_player_play(player);

This example code of course omits error-checking.

Refer to the following link for a more complete example: https://wiki.videolan.org/LibVLC_SampleCode_Qt

Answer 2

Qt's built-in media player uses system codecs. You will need to use something else if you want to subvert that.

I haven't used it, but there are mentions around of QtAV, which uses FFmpeg.

The QtAV website observes: QtMultimedia's implemention is platform dependent. It uses dshow on windows, gstream on linux etc. May be not an easy work to support a new platform.