Use the segment start time when calculating the timestamp of data buffers in the scope. Fixes a memory leak and makes the scope work again when playing ogg streams. Fixes issue #495.

src/engines/gstengine.cpp

@@ -308,7 +308,8 @@ void GstEngine::UpdateScope() {
   typedef int16_t sampletype;
 
   // prune the scope and get the current pos of the audio device
-  quint64 pos = PruneScope();
+  const quint64 pos = PruneScope();
+  const quint64 segment_start = current_pipeline_->segment_start();
 
   // head of the delay queue is the most delayed, so we work with that one
   GstBuffer *buf = reinterpret_cast<GstBuffer *>( g_queue_peek_head(delayq_) );
@@ -316,7 +317,7 @@ void GstEngine::UpdateScope() {
     return;
 
   // start time for this buffer
-  quint64 stime = GST_BUFFER_TIMESTAMP(buf);
+  quint64 stime = GST_BUFFER_TIMESTAMP(buf) - segment_start;
   // duration of the buffer...
   quint64 dur = GST_BUFFER_DURATION(buf);
   // therefore we can calculate the end time for the buffer
@@ -716,7 +717,8 @@ qint64 GstEngine::PruneScope() {
     return 0;
 
   // get the position playing in the audio device
-  qint64 pos = current_pipeline_->position();
+  const qint64 pos = current_pipeline_->position();
+  const qint64 segment_start = current_pipeline_->segment_start();
 
   GstBuffer *buf = 0;
   quint64 etime = 0;
@@ -727,7 +729,7 @@ qint64 GstEngine::PruneScope() {
     buf = reinterpret_cast<GstBuffer *>( g_queue_peek_head(delayq_) );
     if (buf) {
       // the start time of the buffer
-      quint64 stime = GST_BUFFER_TIMESTAMP(buf);
+      quint64 stime = GST_BUFFER_TIMESTAMP(buf) - segment_start;
       // the duration of the buffer
       quint64 dur = GST_BUFFER_DURATION(buf);
       // therefore we can calculate the end time of the buffer

src/engines/gstenginepipeline.cpp

@@ -29,6 +29,7 @@ GstEnginePipeline::GstEnginePipeline(GstEngine* engine)
     engine_(engine),
     valid_(false),
     sink_(GstEngine::kAutoSink),
+    segment_start_(0),
     rg_enabled_(false),
     rg_mode_(0),
     rg_preamp_(0.0),
@@ -145,6 +146,7 @@ bool GstEnginePipeline::Init(const QUrl &url) {
   // so that our visualization are not affected by them
   pad = gst_element_get_pad(scope_element, "src");
   gst_pad_add_buffer_probe(pad, G_CALLBACK(HandoffCallback), this);
+  gst_pad_add_event_probe(pad, G_CALLBACK(EventHandoffCallback), this);
   gst_object_unref (pad);
 
   // Ensure we get the right type out of audioconvert for our scope
@@ -321,6 +323,20 @@ bool GstEnginePipeline::HandoffCallback(GstPad*, GstBuffer* buf, gpointer self)
   return true;
 }
 
+bool GstEnginePipeline::EventHandoffCallback(GstPad*, GstEvent* e, gpointer self) {
+  GstEnginePipeline* instance = reinterpret_cast<GstEnginePipeline*>(self);
+
+  if (GST_EVENT_TYPE(e) == GST_EVENT_NEWSEGMENT) {
+    // The segment start time is used to calculate the proper offset of data
+    // buffers from the start of the stream
+    gint64 start = 0;
+    gst_event_parse_new_segment(e, NULL, NULL, NULL, &start, NULL, NULL);
+    instance->segment_start_ = start;
+  }
+
+  return true;
+}
+
 void GstEnginePipeline::SourceDrainedCallback(GstURIDecodeBin* bin, gpointer self) {
   GstEnginePipeline* instance = reinterpret_cast<GstEnginePipeline*>(self);
 

src/engines/gstenginepipeline.h

@@ -72,6 +72,7 @@ class GstEnginePipeline : public QObject {
   qint64 position() const;
   qint64 length() const;
   GstState state() const;
+  qint64 segment_start() const { return segment_start_; }
 
   QUrl redirect_url() const { return redirect_url_; }
 
@@ -94,6 +95,7 @@ class GstEnginePipeline : public QObject {
   static gboolean BusCallback(GstBus*, GstMessage*, gpointer);
   static void NewPadCallback(GstElement*, GstPad*, gpointer);
   static bool HandoffCallback(GstPad*, GstBuffer*, gpointer);
+  static bool EventHandoffCallback(GstPad*, GstEvent*, gpointer);
   static void SourceDrainedCallback(GstURIDecodeBin*, gpointer);
   static bool StopUriDecodeBin(gpointer bin);
   void TagMessageReceived(GstMessage*);
@@ -121,6 +123,7 @@ class GstEnginePipeline : public QObject {
   // These get called when there is a new audio buffer available
   QList<BufferConsumer*> buffer_consumers_;
   QMutex buffer_consumers_mutex_;
+  qint64 segment_start_;
 
   // ReplayGain
   bool rg_enabled_;