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examples/voip/main.cpp

This is a simple VoIP application that takes audio from a microphone and video from the video test source, encodes them with speex and h264 respectively and sends them to the other side using RTP.

The same application is both a client and a server. To make a call between them, you need to configure the destination ports of the one side to be the source ports of the other and vice versa.

/*
    Copyright (C) 2011 Collabora Ltd.
      @author George Kiagiadakis <george.kiagiadakis@collabora.co.uk>

    This library is free software; you can redistribute it and/or modify
    it under the terms of the GNU Lesser General Public License as published
    by the Free Software Foundation; either version 2.1 of the License, or
    (at your option) any later version.

    This program is distributed in the hope that it will be useful,
    but WITHOUT ANY WARRANTY; without even the implied warranty of
    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
    GNU Lesser General Public License for more details.

    You should have received a copy of the GNU Lesser General Public License
    along with this program.  If not, see <http://www.gnu.org/licenses/>.
*/

/*
    In this example, the pipeline consists of two parts, one for the audio session
    and one for the video session. Each session part is identical to the other in
    structure, the only difference being the encoder/decoder elements and the
    sources/sinks of the audio/video content.

    Here is a figure that shows the video session part of the pipeline.

.------------.   .-------.    .-------.      .----------.     .-------.
|videotestsrc|   |h264enc|    |h264pay|      |          |     |udpsink|      RTP SEND
|          src->sink    src->sink   src->send_rtp send_rtp->sink      |
'------------'   '-------'    '-------'      |          |     '-------'
                                             | rtpbin   |
                              .-------.      |          |     .---------.   .-------.   .-------------.
                RTP RECV      |udpsrc |      |          |     |h264depay|   |h264dec|   |autovideosink|
                              |      src->recv_rtp recv_rtp->sink     src->sink   src->sink           |
                              '-------'      |          |     '---------'   '-------'   '-------------'
                                             |          |
                                             |          |     .-------.
                                             |          |     |udpsink|  RTCP
                                             |    send_rtcp->sink     | sync=false
                              .-------.      |          |     '-------' async=false
                    RTCP      |udpsrc |      |          |
                              |     src->recv_rtcp      |
                              '-------'      |          |
                                             '----------'

    In the two sessions, the gstrtpbin element is common and the sessions are
    distinguished from the number at the end of rtpbin's pad names.
*/

#include "ui_voip.h"

#include <QtGui/QApplication>
#include <QtGui/QDialog>

#include <QGlib/Connect>
#include <QGlib/Error>
#include <QGst/Init>
#include <QGst/Pipeline>
#include <QGst/ElementFactory>
#include <QGst/Pad>
#include <QGst/Structure>
#include <QGst/Bus>
#include <QGst/Message>

class VoipExample : public QDialog
{
    Q_OBJECT
public:
    explicit VoipExample(QWidget *parent = 0);
    virtual ~VoipExample();

private Q_SLOTS:
    void on_startCallButton_clicked();
    void on_endCallButton_clicked();

    //keep the rtcp ports in sync with the rtp ports on the UI
    void on_audioRtpSrcSpinBox_valueChanged(int value) { m_ui.audioRtcpSrcSpinBox->setValue(value + 1); }
    void on_audioRtpDestSpinBox_valueChanged(int value) { m_ui.audioRtcpDestSpinBox->setValue(value + 1); }
    void on_videoRtpSrcSpinBox_valueChanged(int value) { m_ui.videoRtcpSrcSpinBox->setValue(value + 1); }
    void on_videoRtpDestSpinBox_valueChanged(int value) { m_ui.videoRtcpDestSpinBox->setValue(value + 1); }

private:
    void onRtpBinPadAdded(const QGst::PadPtr & pad);
    void onBusErrorMessage(const QGst::MessagePtr & msg);

private:
    Ui::VoipExample m_ui;
    QGst::PipelinePtr m_pipeline;
};

VoipExample::VoipExample(QWidget *parent)
    : QDialog(parent)
{
    m_ui.setupUi(this);
}

VoipExample::~VoipExample()
{
    //cleanup if the pipeline is still active
    if (m_pipeline) {
        on_endCallButton_clicked();
    }
}

void VoipExample::on_startCallButton_clicked()
{
    m_pipeline = QGst::Pipeline::create();

    QGst::ElementPtr rtpbin = QGst::ElementFactory::make("gstrtpbin");
    if (!rtpbin) {
        qFatal("Failed to create gstrtpbin");
    }
    m_pipeline->add(rtpbin);

    //audio content
    if (m_ui.audioGroupBox->isChecked()) {
        //sending side
        QGst::ElementPtr audiosrc;
        try {
            audiosrc = QGst::Bin::fromDescription(
                "autoaudiosrc ! queue ! audioconvert ! audiorate ! audio/x-raw-int,rate=8000 "
                "! speexenc ! rtpspeexpay"
            );
        } catch (const QGlib::Error & error) {
            qCritical() << error;
            qFatal("One ore more required elements are missing. Aborting...");
        }
        m_pipeline->add(audiosrc);
        audiosrc->link(rtpbin, "send_rtp_sink_1");

        QGst::ElementPtr rtpudpsink = QGst::ElementFactory::make("udpsink");
        if (!rtpudpsink) {
            qFatal("Failed to create udpsink. Aborting...");
        }
        rtpudpsink->setProperty("host", m_ui.remoteHostLineEdit->text());
        rtpudpsink->setProperty("port", m_ui.audioRtpDestSpinBox->value());
        m_pipeline->add(rtpudpsink);
        rtpbin->link("send_rtp_src_1", rtpudpsink);

        //receiving side
        QGst::ElementPtr rtpudpsrc = QGst::ElementFactory::make("udpsrc");
        if (!rtpudpsrc) {
            qFatal("Failed to create udpsrc. Aborting...");
        }
        rtpudpsrc->setProperty("port", m_ui.audioRtpSrcSpinBox->value());
        rtpudpsrc->setProperty("caps", QGst::Caps::fromString("application/x-rtp,"
                                                              "media=(string)audio,"
                                                              "clock-rate=(int)8000,"
                                                              "encoding-name=(string)SPEEX"));
        m_pipeline->add(rtpudpsrc);
        rtpudpsrc->link(rtpbin, "recv_rtp_sink_1");

        //recv sink will be added when the pad becomes available

        //rtcp connection
        QGst::ElementPtr rtcpudpsrc = QGst::ElementFactory::make("udpsrc");
        rtcpudpsrc->setProperty("port", m_ui.audioRtcpSrcSpinBox->value());
        m_pipeline->add(rtcpudpsrc);
        rtcpudpsrc->link(rtpbin, "recv_rtcp_sink_1");

        QGst::ElementPtr rtcpudpsink = QGst::ElementFactory::make("udpsink");
        rtcpudpsink->setProperty("host", m_ui.remoteHostLineEdit->text());
        rtcpudpsink->setProperty("port", m_ui.audioRtcpDestSpinBox->value());
        rtcpudpsink->setProperty("sync", false);
        rtcpudpsink->setProperty("async", false);
        m_pipeline->add(rtcpudpsink);
        rtpbin->link("send_rtcp_src_1", rtcpudpsink);
    }

    //video content
    if (m_ui.videoGroupBox->isChecked()) {
        //sending side
        QGst::ElementPtr videosrc;
        try {
            videosrc = QGst::Bin::fromDescription(
                "videotestsrc is-live=true ! video/x-raw-yuv,width=320,height=240,framerate=15/1 "
                "! x264enc tune=zerolatency byte-stream=true bitrate=300 ! rtph264pay"
            );
        } catch (const QGlib::Error & error) {
            qCritical() << error;
            qFatal("One ore more required elements are missing. Aborting...");
        }
        m_pipeline->add(videosrc);
        videosrc->link(rtpbin, "send_rtp_sink_2");

        QGst::ElementPtr rtpudpsink = QGst::ElementFactory::make("udpsink");
        if (!rtpudpsink) {
            qFatal("Failed to create udpsink. Aborting...");
        }
        rtpudpsink->setProperty("host", m_ui.remoteHostLineEdit->text());
        rtpudpsink->setProperty("port", m_ui.videoRtpDestSpinBox->value());
        m_pipeline->add(rtpudpsink);
        rtpbin->link("send_rtp_src_2", rtpudpsink);

        //receiving side
        QGst::ElementPtr rtpudpsrc = QGst::ElementFactory::make("udpsrc");
        if (!rtpudpsrc) {
            qFatal("Failed to create udpsrc. Aborting...");
        }
        rtpudpsrc->setProperty("port", m_ui.videoRtpSrcSpinBox->value());
        rtpudpsrc->setProperty("caps", QGst::Caps::fromString("application/x-rtp,"
                                                              "media=(string)video,"
                                                              "clock-rate=(int)90000,"
                                                              "encoding-name=(string)H264"));
        m_pipeline->add(rtpudpsrc);
        rtpudpsrc->link(rtpbin, "recv_rtp_sink_2");

        //recv sink will be added when the pad becomes available

        //rtcp connection
        QGst::ElementPtr rtcpudpsrc = QGst::ElementFactory::make("udpsrc");
        rtcpudpsrc->setProperty("port", m_ui.videoRtcpSrcSpinBox->value());
        m_pipeline->add(rtcpudpsrc);
        rtcpudpsrc->link(rtpbin, "recv_rtcp_sink_2");

        QGst::ElementPtr rtcpudpsink = QGst::ElementFactory::make("udpsink");
        rtcpudpsink->setProperty("host", m_ui.remoteHostLineEdit->text());
        rtcpudpsink->setProperty("port", m_ui.videoRtcpDestSpinBox->value());
        rtcpudpsink->setProperty("sync", false);
        rtcpudpsink->setProperty("async", false);
        m_pipeline->add(rtcpudpsink);
        rtpbin->link("send_rtcp_src_2", rtcpudpsink);
    }

    //watch for the receiving side src pads
    QGlib::connect(rtpbin, "pad-added", this, &VoipExample::onRtpBinPadAdded);

    //watch the bus
    m_pipeline->bus()->addSignalWatch();
    QGlib::connect(m_pipeline->bus(), "message::error", this, &VoipExample::onBusErrorMessage);

    //switch to the video view and connect the video widget
    m_ui.stackedWidget->setCurrentIndex(1);
    m_ui.videoWidget->watchPipeline(m_pipeline);

    //go!
    m_pipeline->setState(QGst::StatePlaying);
}

void VoipExample::onRtpBinPadAdded(const QGst::PadPtr & pad)
{
    QGst::ElementPtr bin;

    try {
        if (pad->caps()->internalStructure(0)->value("media").toString() == QLatin1String("audio")) {
            bin = QGst::Bin::fromDescription(
                "rtpspeexdepay ! speexdec ! audioconvert ! autoaudiosink"
            );
        } else {
            bin = QGst::Bin::fromDescription(
                "rtph264depay ! ffdec_h264 ! ffmpegcolorspace ! autovideosink"
            );
        }
    } catch (const QGlib::Error & error) {
        qCritical() << error;
        qFatal("One ore more required elements are missing. Aborting...");
    }

    m_pipeline->add(bin);
    bin->syncStateWithParent();
    pad->link(bin->getStaticPad("sink"));
}

void VoipExample::onBusErrorMessage(const QGst::MessagePtr & msg)
{
    qCritical() << "Error from bus:" << msg.staticCast<QGst::ErrorMessage>()->error();
    on_endCallButton_clicked(); //stop the call
}

void VoipExample::on_endCallButton_clicked()
{
    m_pipeline->setState(QGst::StateNull);
    m_ui.videoWidget->stopPipelineWatch();
    m_ui.stackedWidget->setCurrentIndex(0);
    m_pipeline.clear();
}


int main(int argc, char *argv[])
{
    QApplication app(argc, argv);
    QGst::init(&argc, &argv);

    VoipExample gui;
    gui.show();

    return app.exec();
}

#include "main.moc"