Fix playback problem with RTSP streaming for QMediaPlayer(gstreamer)
Change-Id: Ie920cbb5a377e810aee3e106bb50deb46365ce3b Reviewed-by:Michael Goddard (cherry picked from commit 05841ae6a9e0ffac623f9b00565bf33a52a22ecd) Reviewed-on: http://codereview.qt.nokia.com/983 Reviewed-by: Qt Sanity Bot <qt_sanity_bot@ovi.com>
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@@ -551,7 +551,7 @@ void QGstreamerPlayerControl::setBufferProgress(int progress)
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m_session->state() != QMediaPlayer::PlayingState)
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m_session->play();
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if (m_bufferProgress < 100 &&
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if (!m_session->isLiveSource() && m_bufferProgress < 100 &&
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(m_session->state() == QMediaPlayer::PlayingState ||
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m_session->pendingState() == QMediaPlayer::PlayingState))
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m_session->pause();
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@@ -47,6 +47,7 @@
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#include "qgstutils.h"
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#include <gst/gstvalue.h>
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#include <gst/base/gstbasesrc.h>
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#include <QtCore/qdatetime.h>
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#include <QtCore/qdebug.h>
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@@ -103,7 +104,8 @@ QGstreamerPlayerSession::QGstreamerPlayerSession(QObject *parent)
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m_duration(-1),
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m_durationQueries(0),
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m_everPlayed(false) ,
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m_sourceType(UnknownSrc)
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m_sourceType(UnknownSrc),
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m_isLiveSource(false)
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{
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#ifdef USE_PLAYBIN2
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m_playbin = gst_element_factory_make("playbin2", NULL);
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@@ -1421,22 +1423,47 @@ void QGstreamerPlayerSession::playbinNotifySource(GObject *o, GParamSpec *p, gpo
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const int timeout = 30;
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if (qstrcmp(G_OBJECT_CLASS_NAME(G_OBJECT_GET_CLASS(source)), "GstUDPSrc") == 0) {
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//udpsrc timeout unit = microsecond
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//The udpsrc is always a live source.
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g_object_set(G_OBJECT(source), "timeout", G_GUINT64_CONSTANT(timeout*1000000), NULL);
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self->m_sourceType = UDPSrc;
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self->m_isLiveSource = true;
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} else if (qstrcmp(G_OBJECT_CLASS_NAME(G_OBJECT_GET_CLASS(source)), "GstSoupHTTPSrc") == 0) {
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//souphttpsrc timeout unit = second
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g_object_set(G_OBJECT(source), "timeout", guint(timeout), NULL);
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self->m_sourceType = SoupHTTPSrc;
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//since gst_base_src_is_live is not reliable, so we check the source property directly
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gboolean isLive = false;
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g_object_get(G_OBJECT(source), "is-live", &isLive, NULL);
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self->m_isLiveSource = isLive;
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} else if (qstrcmp(G_OBJECT_CLASS_NAME(G_OBJECT_GET_CLASS(source)), "GstMMSSrc") == 0) {
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self->m_sourceType = MMSSrc;
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self->m_isLiveSource = gst_base_src_is_live(GST_BASE_SRC(source));
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g_object_set(G_OBJECT(source), "tcp-timeout", G_GUINT64_CONSTANT(timeout*1000000), NULL);
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} else if (qstrcmp(G_OBJECT_CLASS_NAME(G_OBJECT_GET_CLASS(source)), "GstRTSPSrc") == 0) {
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//rtspsrc acts like a live source and will therefore only generate data in the PLAYING state.
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self->m_sourceType = RTSPSrc;
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self->m_isLiveSource = true;
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} else {
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self->m_sourceType = UnknownSrc;
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self->m_isLiveSource = gst_base_src_is_live(GST_BASE_SRC(source));
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}
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#ifdef DEBUG_PLAYBIN
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if (self->m_isLiveSource)
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qDebug() << "Current source is a live source";
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else
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qDebug() << "Current source is a non-live source";
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#endif
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gst_object_unref(source);
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}
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bool QGstreamerPlayerSession::isLiveSource() const
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{
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return m_isLiveSource;
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}
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void QGstreamerPlayerSession::handleVolumeChange(GObject *o, GParamSpec *p, gpointer d)
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{
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Q_UNUSED(o);
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@@ -112,6 +112,8 @@ public:
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static void configureAppSrcElement(GObject*, GObject*, GParamSpec*,QGstreamerPlayerSession* _this);
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#endif
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bool isLiveSource() const;
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public slots:
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void loadFromUri(const QNetworkRequest &url);
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void loadFromStream(const QNetworkRequest &url, QIODevice *stream);
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@@ -208,10 +210,12 @@ private:
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UnknownSrc,
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SoupHTTPSrc,
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UDPSrc,
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MMSSrc
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MMSSrc,
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RTSPSrc,
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};
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SourceType m_sourceType;
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bool m_everPlayed;
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bool m_isLiveSource;
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};
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#endif // QGSTREAMERPLAYERSESSION_H
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