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Diffstat (limited to 'src/plugins/gstreamer/audiodecoder/qgstreameraudiodecodersession.cpp')
-rw-r--r--src/plugins/gstreamer/audiodecoder/qgstreameraudiodecodersession.cpp36
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) {