[gstreamermm] BaseAudio[Sink|Src], RingBuffer: Rename as was done in the C API.

From: José Alburquerque <jaalburqu src gnome org>
To: commits-list gnome org
Cc:
Subject: [gstreamermm] BaseAudio[Sink|Src], RingBuffer: Rename as was done in the C API.
Date: Fri, 19 Jul 2013 18:33:26 +0000 (UTC)
commit 40c445aabf9957a7a075a617ed23ee2397a7089d
Author: José Alburquerque <jaalburquerque gmail com>
Date:   Wed Jun 19 15:31:37 2013 -0400

    BaseAudio[Sink|Src],RingBuffer: Rename as was done in the C API.
    
        * gstreamer/src/baseaudiosink.{ccg,hg}: Rename these files/classes to:
        * gstreamer/src/audiobasesink.{ccg,hg}:
        * gstreamer/src/baseaudiosrc.{ccg,hg}: Rename these files/classes to:
        * gstreamer/src/audiobasesrc.{ccg,hg}:
        * gstreamer/src/ringbuffer.{ccg,hg}: Rename these files/classes to:
        * gstreamer/src/audioringbuffer.{ccg,hg}:
        * gstreamer/src/filelist.am: Rename the files here also.
        * gstreamer/src/gst_vfuncs.defs: Rename the types here also.
        * tools/m4/convert_gst.m4: Rename the types here also.

 ChangeLog                                          |   14 ++
 .../src/{baseaudiosrc.ccg => audiobasesink.ccg}    |    5 +-
 .../src/{baseaudiosink.hg => audiobasesink.hg}     |   33 +++--
 .../src/{baseaudiosink.ccg => audiobasesrc.ccg}    |    5 +-
 gstreamer/src/{baseaudiosrc.hg => audiobasesrc.hg} |   27 ++--
 .../src/{ringbuffer.ccg => audioringbuffer.ccg}    |   69 +++++-----
 .../src/{ringbuffer.hg => audioringbuffer.hg}      |  147 ++++++++++----------
 gstreamer/src/filelist.am                          |    6 +-
 gstreamer/src/gst_vfuncs.defs                      |   48 +++---
 tools/m4/convert_gst.m4                            |   14 +-
 10 files changed, 192 insertions(+), 176 deletions(-)
---
diff --git a/ChangeLog b/ChangeLog
index 3cc01ab..206a45c 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,3 +1,17 @@
+2013-06-19  José Alburquerque  <jaalburquerque gmail com>
+
+       BaseAudio[Sink|Src],RingBuffer: Rename as was done in the C API.
+
+       * gstreamer/src/baseaudiosink.{ccg,hg}: Rename these files/classes to:
+       * gstreamer/src/audiobasesink.{ccg,hg}:
+       * gstreamer/src/baseaudiosrc.{ccg,hg}: Rename these files/classes to:
+       * gstreamer/src/audiobasesrc.{ccg,hg}:
+       * gstreamer/src/ringbuffer.{ccg,hg}: Rename these files/classes to:
+       * gstreamer/src/audioringbuffer.{ccg,hg}:
+       * gstreamer/src/filelist.am: Rename the files here also.
+       * gstreamer/src/gst_vfuncs.defs: Rename the types here also.
+       * tools/m4/convert_gst.m4: Rename the types here also.
+
 2013-06-18  José Alburquerque  <jaalburquerque gmail com>
 
        Correct some general method parameter changes in existing source files.
diff --git a/gstreamer/src/baseaudiosrc.ccg b/gstreamer/src/audiobasesink.ccg
similarity index 87%
rename from gstreamer/src/baseaudiosrc.ccg
rename to gstreamer/src/audiobasesink.ccg
index c79c025..c18ad71 100644
--- a/gstreamer/src/baseaudiosrc.ccg
+++ b/gstreamer/src/audiobasesink.ccg
@@ -17,6 +17,5 @@
  * Software Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
  */
 
-#include <gst/audio/gstbaseaudiosrc.h>
-#include <gstreamermm/ringbuffer.h>
-_PINCLUDE(gstreamermm/private/pushsrc_p.h)
+#include <gst/audio/audio.h>
+#include <gstreamermm/audioringbuffer.h>
diff --git a/gstreamer/src/baseaudiosink.hg b/gstreamer/src/audiobasesink.hg
similarity index 60%
rename from gstreamer/src/baseaudiosink.hg
rename to gstreamer/src/audiobasesink.hg
index 6eface9..d2b2895 100644
--- a/gstreamer/src/baseaudiosink.hg
+++ b/gstreamer/src/audiobasesink.hg
@@ -17,17 +17,18 @@
  * Software Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
  */
 
-#include <gst/audio/gstringbuffer.h>
+#include <gst/audio/audio.h>
 #include <gstreamermm/basesink.h>
+_PINCLUDE(gstreamermm/private/basesink_p.h)
 
 _DEFS(gstreamermm,gst)
 
 namespace Gst
 {
 
-class RingBuffer;
+class AudioRingBuffer;
 
-_WRAP_ENUM(BaseAudioSinkSlaveMethod, GstBaseAudioSinkSlaveMethod)
+_WRAP_ENUM(AudioBaseSinkSlaveMethod, GstAudioBaseSinkSlaveMethod)
 
 /** The base class for audio sinks.
  * This is the base class for audio sinks. Subclasses need to implement the
@@ -38,24 +39,24 @@ _WRAP_ENUM(BaseAudioSinkSlaveMethod, GstBaseAudioSinkSlaveMethod)
  *
  * @ingroup GstBaseClasses
  */
-class BaseAudioSink : public Gst::BaseSink
+class AudioBaseSink : public Gst::BaseSink
 {
-  _CLASS_GOBJECT(BaseAudioSink, GstBaseAudioSink, GST_BASE_AUDIO_SINK, Gst::BaseSink, GstBaseSink)
+  _CLASS_GOBJECT(AudioBaseSink, GstAudioBaseSink, GST_AUDIO_BASE_SINK, Gst::BaseSink, GstBaseSink)
 
 public:
-  /** Get the Gst::Clock of the Gst::BaseAudioSink.
+  /** Get the Gst::Clock of the Gst::AudioBaseSink.
    */
   _MEMBER_GET_GOBJECT(provided_clock, provided_clock, Gst::Clock, GstClock*)
 
-  _WRAP_METHOD(Glib::RefPtr<Gst::RingBuffer> create_ring_buffer(), gst_base_audio_sink_create_ringbuffer)
-  _WRAP_METHOD(void set_provide_clock(bool provide), gst_base_audio_sink_set_provide_clock)
-  _WRAP_METHOD(bool get_provide_clock() const, gst_base_audio_sink_get_provide_clock)
+  _WRAP_METHOD(Glib::RefPtr<Gst::AudioRingBuffer> create_ring_buffer(), 
gst_audio_base_sink_create_ringbuffer)
+  _WRAP_METHOD(void set_provide_clock(bool provide), gst_audio_base_sink_set_provide_clock)
+  _WRAP_METHOD(bool get_provide_clock() const, gst_audio_base_sink_get_provide_clock)
 
-  _WRAP_METHOD(void set_slave_method(Gst::BaseAudioSinkSlaveMethod method), 
gst_base_audio_sink_set_slave_method)
-  _WRAP_METHOD(Gst::BaseAudioSinkSlaveMethod get_slave_method() const, gst_base_audio_sink_get_slave_method)
+  _WRAP_METHOD(void set_slave_method(Gst::AudioBaseSinkSlaveMethod method), 
gst_audio_base_sink_set_slave_method)
+  _WRAP_METHOD(Gst::AudioBaseSinkSlaveMethod get_slave_method() const, gst_audio_base_sink_get_slave_method)
 
-  _WRAP_METHOD(gint64 get_drift_tolerance() const, gst_base_audio_sink_get_drift_tolerance)
-  _WRAP_METHOD(void set_drift_tolerance(gint64 drift_tolerance), gst_base_audio_sink_set_drift_tolerance)
+  _WRAP_METHOD(gint64 get_drift_tolerance() const, gst_audio_base_sink_get_drift_tolerance)
+  _WRAP_METHOD(void set_drift_tolerance(gint64 drift_tolerance), gst_audio_base_sink_set_drift_tolerance)
 
   _WRAP_PROPERTY("alignment-threshold", guint64)
   _WRAP_PROPERTY("buffer-time", gint64)
@@ -64,11 +65,11 @@ public:
   _WRAP_PROPERTY("drift-tolerance", gint64)
   _WRAP_PROPERTY("latency-time", gint64)
   _WRAP_PROPERTY("provide-clock", bool)
-  _WRAP_PROPERTY("slave-method", Gst::BaseAudioSinkSlaveMethod)
+  _WRAP_PROPERTY("slave-method", Gst::AudioBaseSinkSlaveMethod)
 
-  /** vfunc to create and return a Gst::RingBuffer to write to.
+  /** vfunc to create and return a Gst::AudioRingBuffer to write to.
    */
-  _WRAP_VFUNC(Glib::RefPtr<Gst::RingBuffer> create_ring_buffer(), "create_ringbuffer")
+  _WRAP_VFUNC(Glib::RefPtr<Gst::AudioRingBuffer> create_ring_buffer(), "create_ringbuffer")
 };
 
 } // namespace Gst
diff --git a/gstreamer/src/baseaudiosink.ccg b/gstreamer/src/audiobasesrc.ccg
similarity index 87%
rename from gstreamer/src/baseaudiosink.ccg
rename to gstreamer/src/audiobasesrc.ccg
index f1ef70f..c18ad71 100644
--- a/gstreamer/src/baseaudiosink.ccg
+++ b/gstreamer/src/audiobasesrc.ccg
@@ -17,6 +17,5 @@
  * Software Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
  */
 
-#include <gst/audio/gstbaseaudiosink.h>
-#include <gstreamermm/ringbuffer.h>
-_PINCLUDE(gstreamermm/private/basesink_p.h)
+#include <gst/audio/audio.h>
+#include <gstreamermm/audioringbuffer.h>
diff --git a/gstreamer/src/baseaudiosrc.hg b/gstreamer/src/audiobasesrc.hg
similarity index 63%
rename from gstreamer/src/baseaudiosrc.hg
rename to gstreamer/src/audiobasesrc.hg
index beddade..8a70d3f 100644
--- a/gstreamer/src/baseaudiosrc.hg
+++ b/gstreamer/src/audiobasesrc.hg
@@ -17,17 +17,18 @@
  * Software Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
  */
 
-#include <gst/audio/gstringbuffer.h>
+#include <gst/audio/audio.h>
 #include <gstreamermm/pushsrc.h>
+_PINCLUDE(gstreamermm/private/pushsrc_p.h)
 
 _DEFS(gstreamermm,gst)
 
 namespace Gst
 {
 
-class RingBuffer;
+class AudioRingBuffer;
 
-_WRAP_ENUM(BaseAudioSrcSlaveMethod, GstBaseAudioSrcSlaveMethod)
+_WRAP_ENUM(AudioBaseSrcSlaveMethod, GstAudioBaseSrcSlaveMethod)
 
 /** The base class for audio sources.
  * This is the base class for audio sources. Subclasses need to implement the
@@ -38,21 +39,21 @@ _WRAP_ENUM(BaseAudioSrcSlaveMethod, GstBaseAudioSrcSlaveMethod)
  *
  * @ingroup GstBaseClasses
  */
-class BaseAudioSrc : public Gst::PushSrc
+class AudioBaseSrc : public Gst::PushSrc
 {
-  _CLASS_GOBJECT(BaseAudioSrc, GstBaseAudioSrc, GST_BASE_AUDIO_SRC, Gst::PushSrc, GstPushSrc)
+  _CLASS_GOBJECT(AudioBaseSrc, GstAudioBaseSrc, GST_AUDIO_BASE_SRC, Gst::PushSrc, GstPushSrc)
 
 public:
   /** Get the Gst::Clock of the Gst::AudioSrc.
    */
   _MEMBER_GET_GOBJECT(clock, clock, Gst::Clock, GstClock*)
 
-  _WRAP_METHOD(Glib::RefPtr<Gst::RingBuffer> create_ring_buffer(), gst_base_audio_src_create_ringbuffer)
+  _WRAP_METHOD(Glib::RefPtr<Gst::AudioRingBuffer> create_ring_buffer(), gst_audio_base_src_create_ringbuffer)
 
-  _WRAP_METHOD(void set_provide_clock(bool provide), gst_base_audio_src_set_provide_clock)
-  _WRAP_METHOD(bool get_provide_clock() const, gst_base_audio_src_get_provide_clock)
-  _WRAP_METHOD(void set_slave_method(Gst::BaseAudioSrcSlaveMethod method), 
gst_base_audio_src_set_slave_method)
-  _WRAP_METHOD(Gst::BaseAudioSrcSlaveMethod get_slave_method() const, gst_base_audio_src_get_slave_method)
+  _WRAP_METHOD(void set_provide_clock(bool provide), gst_audio_base_src_set_provide_clock)
+  _WRAP_METHOD(bool get_provide_clock() const, gst_audio_base_src_get_provide_clock)
+  _WRAP_METHOD(void set_slave_method(Gst::AudioBaseSrcSlaveMethod method), 
gst_audio_base_src_set_slave_method)
+  _WRAP_METHOD(Gst::AudioBaseSrcSlaveMethod get_slave_method() const, gst_audio_base_src_get_slave_method)
 
   _WRAP_PROPERTY("actual-buffer-time", gint64)
   _WRAP_PROPERTY("actual-latency-time", gint64)
@@ -60,11 +61,11 @@ public:
   _WRAP_PROPERTY("latency-time", gint64)
   _WRAP_PROPERTY("provide-clock", bool)
 
-  _WRAP_PROPERTY("slave-method", BaseAudioSrcSlaveMethod)
+  _WRAP_PROPERTY("slave-method", AudioBaseSrcSlaveMethod)
 
-  /** vfunc to create and return a Gst::RingBuffer to read from.
+  /** vfunc to create and return a Gst::AudioRingBuffer to read from.
    */
-  _WRAP_VFUNC(Glib::RefPtr<Gst::RingBuffer> create_ring_buffer(), "create_ringbuffer")
+  _WRAP_VFUNC(Glib::RefPtr<Gst::AudioRingBuffer> create_ring_buffer(), "create_ringbuffer")
 };
 
 } // namespace Gst
diff --git a/gstreamer/src/ringbuffer.ccg b/gstreamer/src/audioringbuffer.ccg
similarity index 75%
rename from gstreamer/src/ringbuffer.ccg
rename to gstreamer/src/audioringbuffer.ccg
index 91a740b..2868f5e 100644
--- a/gstreamer/src/ringbuffer.ccg
+++ b/gstreamer/src/audioringbuffer.ccg
@@ -17,7 +17,6 @@
  * Software Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
  */
 
-#include <gst/audio/audio-enumtypes.h>
 #include <gstreamermm/handle_error.h>
 _PINCLUDE(gstreamermm/private/object_p.h)
 
@@ -27,10 +26,11 @@ namespace
 extern "C"
 {
 
-static void RingBuffer_Fill_gstreamermm_callback(GstRingBuffer*, guint8* data, guint len,
-                                                 gpointer user_data)
+static void AudioRingBuffer_Fill_gstreamermm_callback(GstAudioRingBuffer*,
+  guint8* data, guint len, gpointer user_data)
 {
-  Gst::RingBuffer::SlotFill& slot_fill = *static_cast<Gst::RingBuffer::SlotFill*>(user_data);
+  Gst::AudioRingBuffer::SlotFill& slot_fill =
+    *static_cast<Gst::AudioRingBuffer::SlotFill*>(user_data);
 
   try
   {
@@ -50,35 +50,36 @@ static void RingBuffer_Fill_gstreamermm_callback(GstRingBuffer*, guint8* data, g
 namespace Gst
 {
 
-RingBufferSpec::RingBufferSpec()
-: m_spec(g_try_new(GstRingBufferSpec, 1)),
+AudioRingBufferSpec::AudioRingBufferSpec()
+: m_spec(g_try_new(GstAudioRingBufferSpec, 1)),
   take_ownership(true)
 {
   // Handle possible memory allocation failure.
   if(!m_spec)
   {
-    gstreamermm_handle_error("Failed to allocate a new Gst::RingBufferSpec.");
+    gstreamermm_handle_error(
+      "Failed to allocate a new Gst::AudioRingBufferSpec.");
     return;
   }
 }
 
-RingBufferSpec::RingBufferSpec(const Glib::RefPtr<Gst::Caps>& caps,
+AudioRingBufferSpec::AudioRingBufferSpec(const Glib::RefPtr<Gst::Caps>& caps,
   Gst::BufferFormatType type, Gst::BufferFormat format, bool sign,
   bool bigend, int width, int depth, int rate, int channels,
   guint64 latency_time, guint64 buffer_time, int segsize, int segtotal,
   int seglatency)
-: m_spec(g_try_new(GstRingBufferSpec, 1)),
+: m_spec(g_try_new(GstAudioRingBufferSpec, 1)),
   take_ownership(true)
   
 {
   // Handle possible memory allocation failure.
   if(!m_spec)
   {
-    gstreamermm_handle_error("Failed to allocate a new Gst::RingBufferSpec.");
+    gstreamermm_handle_error("Failed to allocate a new Gst::AudioRingBufferSpec.");
     return;
   }
 
-  GstRingBufferSpec* spec = gobj();
+  GstAudioRingBufferSpec* spec = gobj();
 
   spec->caps = Glib::unwrap(caps);
 
@@ -102,7 +103,7 @@ RingBufferSpec::RingBufferSpec(const Glib::RefPtr<Gst::Caps>& caps,
 }
 
 
-RingBufferSpec::RingBufferSpec(GstRingBufferSpec& castitem, bool take_ownership)
+AudioRingBufferSpec::AudioRingBufferSpec(GstAudioRingBufferSpec& castitem, bool take_ownership)
 : m_spec(&castitem),
   take_ownership(take_ownership)
 {
@@ -111,14 +112,14 @@ RingBufferSpec::RingBufferSpec(GstRingBufferSpec& castitem, bool take_ownership)
       gst_caps_ref(castitem.caps);
 }
 
-RingBufferSpec::RingBufferSpec(const RingBufferSpec& other)
-: m_spec(g_try_new(GstRingBufferSpec, 1)),
+AudioRingBufferSpec::AudioRingBufferSpec(const AudioRingBufferSpec& other)
+: m_spec(g_try_new(GstAudioRingBufferSpec, 1)),
   take_ownership(true)
 {
   // Handle possible memory allocation failure.
   if(!m_spec)
   {
-    gstreamermm_handle_error("Failed to allocate a new Gst::RingBufferSpec.");
+    gstreamermm_handle_error("Failed to allocate a new Gst::AudioRingBufferSpec.");
     return;
   }
 
@@ -146,14 +147,14 @@ RingBufferSpec::RingBufferSpec(const RingBufferSpec& other)
   std::copy(other.m_spec->silence_sample, other.m_spec->silence_sample + 
GSTREAMERMM_RING_BUFFER_SPEC_SILENCE_SAMPLE, m_spec->silence_sample);
 }
 
-RingBufferSpec& RingBufferSpec::operator=(const RingBufferSpec& other)
+AudioRingBufferSpec& AudioRingBufferSpec::operator=(const AudioRingBufferSpec& other)
 {
-  RingBufferSpec temp(other);
+  AudioRingBufferSpec temp(other);
   swap(temp);
   return *this;
 }
 
-RingBufferSpec::~RingBufferSpec()
+AudioRingBufferSpec::~AudioRingBufferSpec()
 {
   // Unref existing reference to caps.
   if(m_spec->caps)
@@ -164,9 +165,9 @@ RingBufferSpec::~RingBufferSpec()
     g_free(m_spec);
 }
 
-void RingBufferSpec::swap(RingBufferSpec& other)
+void AudioRingBufferSpec::swap(AudioRingBufferSpec& other)
 {
-  GstRingBufferSpec* spec_temp = m_spec;
+  GstAudioRingBufferSpec* spec_temp = m_spec;
   m_spec = other.m_spec;
   other.m_spec = spec_temp;
 
@@ -175,31 +176,32 @@ void RingBufferSpec::swap(RingBufferSpec& other)
   other.take_ownership = take_temp;
 }
 
-void RingBufferSpec::set_silence_sample(const Glib::ArrayHandle<guint8>& silence_sample)
+void AudioRingBufferSpec::set_silence_sample(const Glib::ArrayHandle<guint8>& silence_sample)
 {
   std::copy(silence_sample.data(),
     silence_sample.data() + GSTREAMERMM_RING_BUFFER_SPEC_SILENCE_SAMPLE,
     m_spec->silence_sample);
 }
 
-void RingBuffer::set_fill_slot(const SlotFill& slot)
+void AudioRingBuffer::set_fill_slot(const SlotFill& slot)
 {
   // The slot is dynamically allocated because according to the
-  // gst_ring_buffer_set_callback() docs, the slot may be called several times.
+  // gst_audio_ring_buffer_set_callback() docs, the slot may be called several
+  // times.
   // A copy is hence created to avoid losing the slot.  The copy is kept in an
   // auto_ptr<> so that is released upon the ring buffer's destruction.
   m_slot.reset(new SlotFill(slot));
 
-  gst_ring_buffer_set_callback(gobj(), &RingBuffer_Fill_gstreamermm_callback,
+  gst_audio_ring_buffer_set_callback(gobj(), &AudioRingBuffer_Fill_gstreamermm_callback,
     m_slot.get());
 }
 
-bool RingBuffer::prepare_read(int& segment, std::vector<guint8>& readptr,
+bool AudioRingBuffer::prepare_read(int& segment, std::vector<guint8>& readptr,
   int& len)
 {
   guint8* c_readptr = 0;
 
-  const bool result = static_cast<bool>(gst_ring_buffer_prepare_read(gobj(), &segment, &c_readptr, &len));
+  const bool result = static_cast<bool>(gst_audio_ring_buffer_prepare_read(gobj(), &segment, &c_readptr, 
&len));
   readptr.assign(c_readptr, c_readptr + len);
 
   if (c_readptr)
@@ -208,7 +210,7 @@ bool RingBuffer::prepare_read(int& segment, std::vector<guint8>& readptr,
   return result;
 }
 
-gboolean RingBuffer_Class::acquire_vfunc_callback(GstRingBuffer* self, GstRingBufferSpec* spec)
+gboolean AudioRingBuffer_Class::acquire_vfunc_callback(GstAudioRingBuffer* self, GstAudioRingBufferSpec* 
spec)
 {
   Glib::ObjectBase *const obj_base = static_cast<Glib::ObjectBase*>(
       Glib::ObjectBase::_get_current_wrapper((GObject*)self));
@@ -225,7 +227,7 @@ gboolean RingBuffer_Class::acquire_vfunc_callback(GstRingBuffer* self, GstRingBu
     {
       try // Trap C++ exceptions which would normally be lost because this is a C callback.
       {
-        Gst::RingBufferSpec cpp_spec(*spec);
+        Gst::AudioRingBufferSpec cpp_spec(*spec);
 
         // Call the virtual member method, which derived classes might override.
         return static_cast<int>(obj->acquire_vfunc(cpp_spec));
@@ -250,7 +252,7 @@ gboolean RingBuffer_Class::acquire_vfunc_callback(GstRingBuffer* self, GstRingBu
   typedef gboolean RType;
   return RType();
 }
-bool Gst::RingBuffer::acquire_vfunc(Gst::RingBufferSpec& spec) 
+bool Gst::AudioRingBuffer::acquire_vfunc(Gst::AudioRingBufferSpec& spec) 
 {
   BaseClassType *const base = static_cast<BaseClassType*>(
       g_type_class_peek_parent(G_OBJECT_GET_CLASS(gobject_)) // Get the parent class of the object class 
(The original underlying C class).
@@ -262,7 +264,7 @@ bool Gst::RingBuffer::acquire_vfunc(Gst::RingBufferSpec& spec)
   typedef bool RType;
   return RType();
 }
-guint RingBuffer_Class::commit_vfunc_callback(GstRingBuffer* self, guint64* sample, guchar* data, gint 
in_samples, gint out_samples, gint* accum)
+guint AudioRingBuffer_Class::commit_vfunc_callback(GstAudioRingBuffer* self, guint64* sample, guchar* data, 
gint in_samples, gint out_samples, gint* accum)
 {
   Glib::ObjectBase *const obj_base = static_cast<Glib::ObjectBase*>(
       Glib::ObjectBase::_get_current_wrapper((GObject*)self));
@@ -287,8 +289,7 @@ guint RingBuffer_Class::commit_vfunc_callback(GstRingBuffer* self, guint64* samp
         // Declare the ArrayHandle for the C++ vfunc.
         Glib::ArrayHandle<guchar> cpp_data(data, data_length, Glib::OWNERSHIP_NONE);
         // Call the virtual member method, which derived classes might override.
-        return obj->commit_vfunc(*(sample), cpp_data, in_samples, out_samples,
-          *(accum));
+        return obj->commit_vfunc(*(sample), Glib::ArrayHandler<guint8>::array_to_vector(data, data_length, 
Glib::OWNERSHIP_NONE), in_samples, out_samples, *(accum));
       }
       catch(...)
       {
@@ -309,14 +310,14 @@ guint RingBuffer_Class::commit_vfunc_callback(GstRingBuffer* self, guint64* samp
   typedef guint RType;
   return RType();
 }
-guint Gst::RingBuffer::commit_vfunc(guint64& sample, const Glib::ArrayHandle<guchar>& data, int in_samples, 
int out_samples, int& accum) 
+guint Gst::AudioRingBuffer::commit_vfunc(guint64& sample, const std::vector<guint8>& data, int in_samples, 
int out_samples, int& accum) 
 {
   BaseClassType *const base = static_cast<BaseClassType*>(
       g_type_class_peek_parent(G_OBJECT_GET_CLASS(gobject_)) // Get the parent class of the object class 
(The original underlying C class).
   );
 
   if(base && base->commit)
-    return (*base->commit)(gobj(),&sample,const_cast<guchar*>(data.data()),in_samples,out_samples,&accum);
+    return 
(*base->commit)(gobj(),&sample,Glib::ArrayHandler<guint8>::vector_to_array(data).data(),,in_samples,out_samples,&accum);
 
   typedef guint RType;
   return RType();
diff --git a/gstreamer/src/ringbuffer.hg b/gstreamer/src/audioringbuffer.hg
similarity index 67%
rename from gstreamer/src/ringbuffer.hg
rename to gstreamer/src/audioringbuffer.hg
index ebcd437..ab39807 100644
--- a/gstreamer/src/ringbuffer.hg
+++ b/gstreamer/src/audioringbuffer.hg
@@ -17,7 +17,7 @@
  * Software Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
  */
 
-#include <gst/audio/gstringbuffer.h>
+#include <gst/audio/audio.h>
 #include <gstreamermm/caps.h>
 #include <gstreamermm/object.h>
 #include <gstreamermm/format.h>
@@ -29,33 +29,33 @@ _DEFS(gstreamermm,gst)
 namespace Gst
 {
 
-_WRAP_ENUM(RingBufferSegState, GstRingBufferSegState)
-_WRAP_ENUM(RingBufferState, GstRingBufferState)
+_WRAP_ENUM(AudioRingBufferSegState, GstAudioRingBufferSegState)
+_WRAP_ENUM(AudioRingBufferState, GstAudioRingBufferState)
 _WRAP_ENUM(BufferFormat, GstBufferFormat)
 _WRAP_ENUM(BufferFormatType, GstBufferFormatType)
 
-/** A class containing the format specification of a Gst::RingBuffer.
- * Gst::RingBufferSpec contains the format specification of a Gst::RingBuffer.
- * The "in" members should be specified by the caller of the
- * Gst::RingBuffer::acquire() method while the "in/out" members may be set by
- * the caller but are also modifiable by Gst::RingBuffer::acquire().  The
- * "out" members are generated as a result of the call to
- * Gst::RingBuffer::acquire().
- * @see Gst::RingBuffer::acquire().
+/** A class containing the format specification of a Gst::AudioRingBuffer.
+ * Gst::AudioRingBufferSpec contains the format specification of a
+ * Gst::AudioRingBuffer.  The "in" members should be specified by the caller of
+ * the Gst::AudioRingBuffer::acquire() method while the "in/out" members may be
+ * set by the caller but are also modifiable by
+ * Gst::AudioRingBuffer::acquire().  The "out" members are generated as a
+ * result of the call to Gst::AudioRingBuffer::acquire().
+ * @see Gst::AudioRingBuffer::acquire().
  */
-class RingBufferSpec
+class AudioRingBufferSpec
 {
-    _CLASS_GENERIC(RingBufferSpec, GstRingBufferSpec)
+    _CLASS_GENERIC(AudioRingBufferSpec, GstAudioRingBufferSpec)
 public:
   /** Default constructor.
    * @throw std::runtime_error if memory is unavailable for the new
-   * Gst::RingBufferSpec.
+   * Gst::AudioRingBufferSpec.
    */
-  RingBufferSpec();
+  AudioRingBufferSpec();
 
-  /** Fully construct a Gst::RingBufferSpec.  Only the "(in)" parameters are
-   * required.  The "(in/out)" parameters are optional and may be modified by
-   * the call to the Gst::RingBuffer::acquire() method.
+  /** Fully construct a Gst::AudioRingBufferSpec.  Only the "(in)" parameters
+   * are required.  The "(in/out)" parameters are optional and may be modified
+   * by the call to the Gst::AudioRingBuffer::acquire() method.
    * @param caps The caps of the buffer (in).
    * @param type The sample type (in/out).
    * @param format The sample format (in/out).
@@ -73,39 +73,39 @@ public:
    * defaults to @a segtotal in the C API (in/out).
    *
    * @throw std::runtime_error if memory is unavailable for the new
-   * Gst::RingBufferSpec.
+   * Gst::AudioRingBufferSpec.
    */
-  RingBufferSpec(const Glib::RefPtr<Gst::Caps>& caps,
+  AudioRingBufferSpec(const Glib::RefPtr<Gst::Caps>& caps,
     Gst::BufferFormatType type = Gst::BUFTYPE_LINEAR,
     Gst::BufferFormat format = Gst::UNKNOWN, bool sign = false,
     bool bigend = false, int width = 0, int depth = 0, int rate = 0,
     int channels = 0, guint64 latency_time = 0, guint64 buffer_time = 0,
     int segsize = 0, int segtotal = 0, int seglatency = 0);
 
-  /// Construct a Gst::RingBufferSpec from a GstRingBufferSpec.
-  explicit RingBufferSpec(GstRingBufferSpec& castitem,
+  /// Construct a Gst::AudioRingBufferSpec from a GstAudioRingBufferSpec.
+  explicit AudioRingBufferSpec(GstAudioRingBufferSpec& castitem,
     bool take_ownership = false);
 
   /** Copy constructor.
    *
    * @throw std::runtime_error if memory is unavailable for the new
-   * Gst::RingBufferSpec.
+   * Gst::AudioRingBufferSpec.
    */
-  RingBufferSpec(const RingBufferSpec& other);
+  AudioRingBufferSpec(const AudioRingBufferSpec& other);
 
   /// Assignment operator.
-  RingBufferSpec& operator=(const RingBufferSpec& other);
+  AudioRingBufferSpec& operator=(const AudioRingBufferSpec& other);
 
   /// Destructor.
-  virtual ~RingBufferSpec();
+  virtual ~AudioRingBufferSpec();
 
-  void swap(RingBufferSpec& other);
+  void swap(AudioRingBufferSpec& other);
 
   /// Gets the underlying gobject.
-  GstRingBufferSpec* gobj() { return m_spec; };
+  GstAudioRingBufferSpec* gobj() { return m_spec; };
 
   /// Gets the underlying gobject.
-  const GstRingBufferSpec* gobj() const { return m_spec; };
+  const GstAudioRingBufferSpec* gobj() const { return m_spec; };
 
   /** Get the caps of the buffer (in). */
   _MEMBER_GET_GOBJECT(caps, caps, Gst::Caps, GstCaps*)
@@ -220,33 +220,33 @@ public:
   _MEMBER_SET(seglatency, seglatency, int, gint)
 
   /** Get the number of bytes of one sample (out).  This is set by the call to
-   * Gst::RingBuffer::acquire().
+   * Gst::AudioRingBuffer::acquire().
    */
   _MEMBER_GET(bytes_per_sample, bytes_per_sample, int, gint)
 
   /** Set the number of bytes of one sample (out).  This is set by the call to
-   * Gst::RingBuffer::acquire() and should probably not be set.
+   * Gst::AudioRingBuffer::acquire() and should probably not be set.
    */
   _MEMBER_SET(bytes_per_sample, bytes_per_sample, int, gint)
 
 // This constant is obtained from the "silence_sample" member in the
-// GstRingBufferSpec structure definition.
+// GstAudioRingBufferSpec structure definition.
 #define GSTREAMERMM_RING_BUFFER_SPEC_SILENCE_SAMPLE 32
 
 #m4 _CONVERSION(`guint8*', `Glib::ArrayHandle<guint8>', `Glib::ArrayHandle<guint8>($3, 
GSTREAMERMM_RING_BUFFER_SPEC_SILENCE_SAMPLE, Glib::OWNERSHIP_NONE)')
   /** Get the bytes representing one sample of silence (out).  This is set by
-   * the call to Gst::RingBuffer::acquire().
+   * the call to Gst::AudioRingBuffer::acquire().
    */
   _MEMBER_GET(silence_sample, silence_sample, Glib::ArrayHandle<guint8>, guint8*)
 
   /** Sets the bytes representing one sample of silence (out).  This is set by
-   * the call to Gst::RingBuffer::acquire() and probably should not be set.
+   * the call to Gst::AudioRingBuffer::acquire() and probably should not be set.
    */
   void set_silence_sample(const Glib::ArrayHandle<guint8>& silence_sample);
 
 protected:
 #ifndef DOXYGEN_SHOULD_SKIP_THIS
-  GstRingBufferSpec* m_spec;
+  GstAudioRingBufferSpec* m_spec;
   // Tells whether the m_spec member should be freed upon destruction.
   bool take_ownership;
 #endif /* DOXYGEN_SHOULD_SKIP_THIS */
@@ -264,13 +264,13 @@ protected:
  * Last reviewed on 2006-02-02 (0.10.4).
  * @ingroup GstBaseClasses
  */
-class RingBuffer : public Gst::Object
+class AudioRingBuffer : public Gst::Object
 {
-  _CLASS_GOBJECT(RingBuffer, GstRingBuffer, GST_RING_BUFFER, Gst::Object, GstObject)
+  _CLASS_GOBJECT(AudioRingBuffer, GstAudioRingBuffer, GST_RING_BUFFER, Gst::Object, GstObject)
 
 public:
   /** For example,
-   * bool on_fill(const Glib::RefPtr<Gst::RingBuffer>& rbuf,
+   * bool on_fill(const Glib::RefPtr<Gst::AudioRingBuffer>& rbuf,
    * const std::vector<guint8>& data, guint len);.
    * This slot is set with set_fill_slot() and is called to fill the memory at
    * data with len bytes of samples.
@@ -287,40 +287,40 @@ public:
    * @param slot The fill slot to set.
    */
   void set_fill_slot(const SlotFill& slot);
-  _IGNORE(gst_ring_buffer_set_callback)
-
-  _WRAP_METHOD(bool acquire(Gst::RingBufferSpec& spec), gst_ring_buffer_acquire)
-  _WRAP_METHOD(bool release(), gst_ring_buffer_release)
-  _WRAP_METHOD(bool is_acquired() const, gst_ring_buffer_is_acquired)
-  _WRAP_METHOD(bool activate(bool active), gst_ring_buffer_activate)
-  _WRAP_METHOD(bool is_active() const, gst_ring_buffer_is_active)
-  _WRAP_METHOD(bool start(), gst_ring_buffer_start)
-  _WRAP_METHOD(bool pause(), gst_ring_buffer_pause)
-  _WRAP_METHOD(bool stop(), gst_ring_buffer_stop)
-  _WRAP_METHOD(guint get_delay() const, gst_ring_buffer_delay)
-  _WRAP_METHOD(guint64 get_samples_done() const, gst_ring_buffer_samples_done)
-  _WRAP_METHOD(void set_sample(guint64 sample), gst_ring_buffer_set_sample)
-
-#m4 _CONVERSION(`const Glib::ArrayHandle<guchar>&', `guchar*', `const_cast<guchar*>($3.data())')
-  _WRAP_METHOD(guint commit(guint64 sample, const Glib::ArrayHandle<guchar>& data, guint len), 
gst_ring_buffer_commit)
-  _WRAP_METHOD(guint commit(guint64& sample, const Glib::ArrayHandle<guchar>& data, int in_samples, int 
out_samples, int& accum), gst_ring_buffer_commit_full)
-
-  _WRAP_METHOD(bool convert(Gst::Format src_fmt, gint64 src_val, Gst::Format dest_fmt, gint64& dest_val) 
const, gst_ring_buffer_convert)
-
-  _WRAP_METHOD_DOCS_ONLY(gst_ring_buffer_prepare_read)
+  _IGNORE(gst_audio_ring_buffer_set_callback)
+
+  _WRAP_METHOD(bool acquire(Gst::AudioRingBufferSpec& spec), gst_audio_ring_buffer_acquire)
+  _WRAP_METHOD(bool release(), gst_audio_ring_buffer_release)
+  _WRAP_METHOD(bool is_acquired() const, gst_audio_ring_buffer_is_acquired)
+  _WRAP_METHOD(bool activate(bool active), gst_audio_ring_buffer_activate)
+  _WRAP_METHOD(bool is_active() const, gst_audio_ring_buffer_is_active)
+  _WRAP_METHOD(bool start(), gst_audio_ring_buffer_start)
+  _WRAP_METHOD(bool pause(), gst_audio_ring_buffer_pause)
+  _WRAP_METHOD(bool stop(), gst_audio_ring_buffer_stop)
+  _WRAP_METHOD(guint get_delay() const, gst_audio_ring_buffer_delay)
+  _WRAP_METHOD(guint64 get_samples_done() const, gst_audio_ring_buffer_samples_done)
+  _WRAP_METHOD(void set_sample(guint64 sample), gst_audio_ring_buffer_set_sample)
+
+#m4 _CONVERSION(`const std::vector<guint8>&', `guint8*', 
`Glib::ArrayHandler<guint8>::vector_to_array($3).data()')
+
+  _WRAP_METHOD(guint commit(guint64& sample, const std::vector<guint8>& data, int in_samples, int 
out_samples, int& accum), gst_audio_ring_buffer_commit)
+
+  _WRAP_METHOD(bool convert(Gst::Format src_fmt, gint64 src_val, Gst::Format dest_fmt, gint64& dest_val) 
const, gst_audio_ring_buffer_convert)
+
+  _WRAP_METHOD_DOCS_ONLY(gst_audio_ring_buffer_prepare_read)
   bool prepare_read(int& segment, std::vector<guint8>& readptr, int& len);
 
-  _WRAP_METHOD(guint read(guint64 sample, const Glib::ArrayHandle<guchar>& data, guint len), 
gst_ring_buffer_read)
+  _WRAP_METHOD(guint read(guint64 sample, const std::vector<guint8>& data, guint len, ClockTime& timestamp), 
gst_audio_ring_buffer_read)
 
-  _WRAP_METHOD(void clear(int segment), gst_ring_buffer_clear)
-  _WRAP_METHOD(void clear_all(), gst_ring_buffer_clear_all)
-  _WRAP_METHOD(void advance(guint advance), gst_ring_buffer_advance)
-  _WRAP_METHOD(bool close_device(), gst_ring_buffer_close_device)
-  _WRAP_METHOD(bool open_device(), gst_ring_buffer_open_device)
-  _WRAP_METHOD(bool device_is_open() const, gst_ring_buffer_device_is_open)
-  _WRAP_METHOD(void set_may_start(bool allowed), gst_ring_buffer_may_start)
-  _WRAP_METHOD(static bool parse_caps(Gst::RingBufferSpec&, const Glib::RefPtr<Gst::Caps>& caps), 
gst_ring_buffer_parse_caps)
-  _WRAP_METHOD(void set_flushing(bool flushing), gst_ring_buffer_set_flushing)
+  _WRAP_METHOD(void clear(int segment), gst_audio_ring_buffer_clear)
+  _WRAP_METHOD(void clear_all(), gst_audio_ring_buffer_clear_all)
+  _WRAP_METHOD(void advance(guint advance), gst_audio_ring_buffer_advance)
+  _WRAP_METHOD(bool close_device(), gst_audio_ring_buffer_close_device)
+  _WRAP_METHOD(bool open_device(), gst_audio_ring_buffer_open_device)
+  _WRAP_METHOD(bool device_is_open() const, gst_audio_ring_buffer_device_is_open)
+  _WRAP_METHOD(void set_may_start(bool allowed), gst_audio_ring_buffer_may_start)
+  _WRAP_METHOD(static bool parse_caps(Gst::AudioRingBufferSpec&, const Glib::RefPtr<Gst::Caps>& caps), 
gst_audio_ring_buffer_parse_caps)
+  _WRAP_METHOD(void set_flushing(bool flushing), gst_audio_ring_buffer_set_flushing)
 
   /** Virtual function to open the device.  Don't set any params or allocate
    * anything.
@@ -330,7 +330,7 @@ public:
   /** Virtual function to allocate the resources for the ring buffer using the
    * given spec.
    */
-  virtual bool acquire_vfunc(Gst::RingBufferSpec& spec);
+  virtual bool acquire_vfunc(Gst::AudioRingBufferSpec& spec);
 
   /** Virtual function to free resources of the ring buffer.
    */
@@ -367,7 +367,8 @@ public:
 
   /** Virtual function to write samples into the ring buffer.
    */
-  virtual guint commit_vfunc(guint64& sample, const Glib::ArrayHandle<guchar>& data, int in_samples, int 
out_samples, int& accum);
+  virtual guint commit_vfunc(guint64& sample, const std::vector<guint8>& data,
+    int in_samples, int out_samples, int& accum);
 
   /** Virtual function to clear the entire ringbuffer Since 0.10.24.
    */
@@ -379,8 +380,8 @@ protected:
   klass->acquire = &acquire_vfunc_callback;
   klass->commit = &commit_vfunc_callback;
   _SECTION(SECTION_PH_VFUNCS)
-  static gboolean acquire_vfunc_callback(GstRingBuffer* self, GstRingBufferSpec* spec);
-  static guint commit_vfunc_callback(GstRingBuffer* self, guint64* sample, guchar* data, gint in_samples, 
gint out_samples, gint* accum);
+  static gboolean acquire_vfunc_callback(GstAudioRingBuffer* self, GstAudioRingBufferSpec* spec);
+  static guint commit_vfunc_callback(GstAudioRingBuffer* self, guint64* sample, guchar* data, gint 
in_samples, gint out_samples, gint* accum);
   _POP()
 #m4end
 
diff --git a/gstreamer/src/filelist.am b/gstreamer/src/filelist.am
index 6e3b168..ff45dc9 100644
--- a/gstreamer/src/filelist.am
+++ b/gstreamer/src/filelist.am
@@ -84,12 +84,13 @@ plugins_hg =                    \
 plugins_ccg = $(plugins_hg:.hg=.ccg)
 
 files_hg  =                     \
+        audiobasesink.hg        \
+        audiobasesrc.hg         \
         audioclock.hg           \
         audiofilter.hg          \
+        audioringbuffer.hg      \
         audiosink.hg            \
         audiosrc.hg             \
-        baseaudiosink.hg        \
-        baseaudiosrc.hg         \
         basesink.hg             \
         basesrc.hg              \
         basetransform.hg        \
@@ -134,7 +135,6 @@ files_hg  =                     \
         pushsrc.hg              \
         query.hg                \
         registry.hg             \
-        ringbuffer.hg           \
         segment.hg              \
         streamvolume.hg         \
         structure.hg            \
diff --git a/gstreamer/src/gst_vfuncs.defs b/gstreamer/src/gst_vfuncs.defs
index 6b19145..ba1d9ea 100644
--- a/gstreamer/src/gst_vfuncs.defs
+++ b/gstreamer/src/gst_vfuncs.defs
@@ -9,7 +9,7 @@
   (of-object "GstAudioFilter")
   (return-type "gboolean")
   (parameters
-   '("GstRingBufferSpec*" "format")
+   '("GstAudioRingBufferSpec*" "format")
   )
 )
 
@@ -24,7 +24,7 @@
   (of-object "GstAudioSink")
   (return-type "gboolean")
   (parameters
-   '("GstRingBufferSpec*" "spec")
+   '("GstAudioRingBufferSpec*" "spec")
   )
 )
 
@@ -68,7 +68,7 @@
   (of-object "GstAudioSrc")
   (return-type "gboolean")
   (parameters
-   '("GstRingBufferSpec*" "spec")
+   '("GstAudioRingBufferSpec*" "spec")
   )
 )
 
@@ -101,18 +101,18 @@
   (return-type "void")
 )
 
-; GstBaseAudioSink
+; GstAudioBaseSink
 
 (define-vfunc create_ringbuffer
-  (of-object "GstBaseAudioSink")
-  (return-type "GstRingBuffer*")
+  (of-object "GstAudioBaseSink")
+  (return-type "GstAudioRingBuffer*")
 )
 
-; GstBaseAudioSrc
+; GstAudioBaseSrc
 
 (define-vfunc create_ringbuffer
-  (of-object "GstBaseAudioSrc")
-  (return-type "GstRingBuffer*")
+  (of-object "GstAudioBaseSrc")
+  (return-type "GstAudioRingBuffer*")
 )
 
 ; GstBaseSink
@@ -971,58 +971,58 @@
   )
 )
 
-; GstRingBuffer
+; GstAudioRingBuffer
 
 (define-vfunc open_device
-  (of-object "GstRingBuffer")
+  (of-object "GstAudioRingBuffer")
   (return-type "gboolean")
 )
 
 (define-vfunc acquire
-  (of-object "GstRingBuffer")
+  (of-object "GstAudioRingBuffer")
   (return-type "gboolean")
   (parameters
-   '("GstRingBufferSpec*" "spec")
+   '("GstAudioRingBufferSpec*" "spec")
   )
 )
 
 (define-vfunc release
-  (of-object "GstRingBuffer")
+  (of-object "GstAudioRingBuffer")
   (return-type "gboolean")
 )
 
 (define-vfunc close_device
-  (of-object "GstRingBuffer")
+  (of-object "GstAudioRingBuffer")
   (return-type "gboolean")
 )
 
 (define-vfunc start
-  (of-object "GstRingBuffer")
+  (of-object "GstAudioRingBuffer")
   (return-type "gboolean")
 )
 
 (define-vfunc pause
-  (of-object "GstRingBuffer")
+  (of-object "GstAudioRingBuffer")
   (return-type "gboolean")
 )
 
 (define-vfunc resume
-  (of-object "GstRingBuffer")
+  (of-object "GstAudioRingBuffer")
   (return-type "gboolean")
 )
 
 (define-vfunc stop
-  (of-object "GstRingBuffer")
+  (of-object "GstAudioRingBuffer")
   (return-type "gboolean")
 )
 
 (define-vfunc delay
-  (of-object "GstRingBuffer")
+  (of-object "GstAudioRingBuffer")
   (return-type "guint")
 )
 
 (define-vfunc activate
-  (of-object "GstRingBuffer")
+  (of-object "GstAudioRingBuffer")
   (return-type "gboolean")
   (parameters
    '("gboolean" "active")
@@ -1030,11 +1030,11 @@
 )
 
 (define-vfunc commit
-  (of-object "GstRingBuffer")
+  (of-object "GstAudioRingBuffer")
   (return-type "guint")
   (parameters
    '("guint64*" "sample")
-   '("guchar*" "data")
+   '("guint8*" "data")
    '("gint" "in_samples")
    '("gint" "out_samples")
    '("gint*" "accum")
@@ -1042,7 +1042,7 @@
 )
 
 (define-vfunc clear_all
-  (of-object "GstRingBuffer")
+  (of-object "GstAudioRingBuffer")
   (return-type "void")
 )
 
diff --git a/tools/m4/convert_gst.m4 b/tools/m4/convert_gst.m4
index 1c25abb..03fc7f9 100644
--- a/tools/m4/convert_gst.m4
+++ b/tools/m4/convert_gst.m4
@@ -3,8 +3,8 @@ dnl Copyright 2008 The gstreamermm Development Team
 dnl Enums
 _CONV_ENUM(Gst,AssocFlags)
 _CONV_ENUM(Gst,AutoplugSelectResult)
-_CONV_ENUM(Gst,BaseAudioSinkSlaveMethod)
-_CONV_ENUM(Gst,BaseAudioSrcSlaveMethod)
+_CONV_ENUM(Gst,AudioBaseSinkSlaveMethod)
+_CONV_ENUM(Gst,AudioBaseSrcSlaveMethod)
 _CONV_ENUM(Gst,BufferCopyFlags)
 _CONV_ENUM(Gst,BufferFlag)
 _CONV_ENUM(Gst,BufferFormat)
@@ -176,12 +176,12 @@ _CONVERSION(`const Glib::RefPtr<Gst::Query>&',`GstQuery*', `Gst::unwrap($3)')
 dnl Registry
 _CONVERSION(`GstRegistry*',`Glib::RefPtr<Gst::Registry>', `Glib::wrap($3)')
 
-dnl RingBuffer
-_CONVERSION(`GstRingBuffer*',`Glib::RefPtr<Gst::RingBuffer>',`Glib::wrap($3)')
-_CONVERSION(`Glib::RefPtr<Gst::RingBuffer>',`GstRingBuffer*',`Glib::unwrap($3)')
+dnl AudioRingBuffer
+_CONVERSION(`GstAudioRingBuffer*',`Glib::RefPtr<Gst::AudioRingBuffer>',`Glib::wrap($3)')
+_CONVERSION(`Glib::RefPtr<Gst::AudioRingBuffer>',`GstAudioRingBuffer*',`Glib::unwrap($3)')
 
-dnl RingBufferSpec
-_CONVERSION(`Gst::RingBufferSpec&', `GstRingBufferSpec*', `$3.gobj()')
+dnl AudioRingBufferSpec
+_CONVERSION(`Gst::AudioRingBufferSpec&', `GstAudioRingBufferSpec*', `$3.gobj()')
 
 dnl Structure
 _CONVERSION(`Gst::Structure&',`GstStructure*',`$3.gobj()')
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