GStreamer: port to 1.0.

0.10 is still used by default.
To enable GStreamer 1.0, pass GST_VERSION=1.0 to qmake
for qtmultimedia.pro.

Contributions from:
Andrew den Exter <andrew.den.exter@qinetic.com.au>
Ilya Smelykh <ilya@videoexpertsgroup.com>
Jim Hodapp <jim.hodapp@canonical.com>
Sergio Schvezov <sergio.schvezov@canonical.com>

Change-Id: I72a46d1170a8794a149bdb5e20767afcc5b7587c
Reviewed-by: Andrew den Exter <andrew.den.exter@qinetic.com.au>

1 files changed, 29 insertions, 7 deletions

diff --git a/src/plugins/gstreamer/audiodecoder/qgstreameraudiodecodersession.cpp b/src/plugins/gstreamer/audiodecoder/qgstreameraudiodecodersession.cpp
index f944a60ac..69876b963 100644
--- a/src/plugins/gstreamer/audiodecoder/qgstreameraudiodecodersession.cpp
+++ b/src/plugins/gstreamer/audiodecoder/qgstreameraudiodecodersession.cpp
@@ -85,7 +85,7 @@ QGstreamerAudioDecoderSession::QGstreamerAudioDecoderSession(QObject *parent)
      m_durationQueries(0)
 {
     // Create pipeline here
-    m_playbin = gst_element_factory_make("playbin2", NULL);
+    m_playbin = gst_element_factory_make(QT_GSTREAMER_PLAYBIN_ELEMENT_NAME, NULL);
 
     if (m_playbin != 0) {
         // Sort out messages
@@ -446,21 +446,40 @@ QAudioBuffer QGstreamerAudioDecoderSession::read()
         if (buffersAvailable == 1)
             emit bufferAvailableChanged(false);
 
+        const char* bufferData = 0;
+        int bufferSize = 0;
+
+#if GST_CHECK_VERSION(1,0,0)
+        GstSample *sample = gst_app_sink_pull_sample(m_appSink);
+        GstBuffer *buffer = gst_sample_get_buffer(sample);
+        GstMapInfo mapInfo;
+        gst_buffer_map(buffer, &mapInfo, GST_MAP_READ);
+        bufferData = (const char*)mapInfo.data;
+        bufferSize = mapInfo.size;
+        QAudioFormat format = QGstUtils::audioFormatForSample(sample);
+#else
         GstBuffer *buffer = gst_app_sink_pull_buffer(m_appSink);
-
+        bufferData = (const char*)buffer->data;
+        bufferSize = buffer->size;
         QAudioFormat format = QGstUtils::audioFormatForBuffer(buffer);
+#endif
+
         if (format.isValid()) {
             // XXX At the moment we have to copy data from GstBuffer into QAudioBuffer.
             // We could improve performance by implementing QAbstractAudioBuffer for GstBuffer.
             qint64 position = getPositionFromBuffer(buffer);
-            audioBuffer = QAudioBuffer(QByteArray((const char*)buffer->data, buffer->size), format, position);
+            audioBuffer = QAudioBuffer(QByteArray((const char*)bufferData, bufferSize), format, position);
             position /= 1000; // convert to milliseconds
             if (position != m_position) {
                 m_position = position;
                 emit positionChanged(m_position);
             }
         }
+#if GST_CHECK_VERSION(1,0,0)
+        gst_sample_unref(sample);
+#else
         gst_buffer_unref(buffer);
+#endif
     }
 
     return audioBuffer;
@@ -488,7 +507,7 @@ void QGstreamerAudioDecoderSession::processInvalidMedia(QAudioDecoder::Error err
     emit error(int(errorCode), errorString);
 }
 
-GstFlowReturn QGstreamerAudioDecoderSession::new_buffer(GstAppSink *, gpointer user_data)
+GstFlowReturn QGstreamerAudioDecoderSession::new_sample(GstAppSink *, gpointer user_data)
 {
     // "Note that the preroll buffer will also be returned as the first buffer when calling gst_app_sink_pull_buffer()."
     QGstreamerAudioDecoderSession *session = reinterpret_cast<QGstreamerAudioDecoderSession*>(user_data);
@@ -531,7 +550,11 @@ void QGstreamerAudioDecoderSession::addAppSink()
 
     GstAppSinkCallbacks callbacks;
     memset(&callbacks, 0, sizeof(callbacks));
-    callbacks.new_buffer = &new_buffer;
+#if GST_CHECK_VERSION(1,0,0)
+    callbacks.new_sample = &new_sample;
+#else
+    callbacks.new_buffer = &new_sample;
+#endif
     gst_app_sink_set_callbacks(m_appSink, &callbacks, this, NULL);
     gst_app_sink_set_max_buffers(m_appSink, MAX_BUFFERS_IN_QUEUE);
     gst_base_sink_set_sync(GST_BASE_SINK(m_appSink), FALSE);
@@ -553,11 +576,10 @@ void QGstreamerAudioDecoderSession::removeAppSink()
 
 void QGstreamerAudioDecoderSession::updateDuration()
 {
-    GstFormat format = GST_FORMAT_TIME;
     gint64 gstDuration = 0;
     int duration = -1;
 
-    if (m_playbin && gst_element_query_duration(m_playbin, &format, &gstDuration))
+    if (m_playbin && qt_gst_element_query_duration(m_playbin, GST_FORMAT_TIME, &gstDuration))
         duration = gstDuration / 1000000;
 
     if (m_duration != duration) {