audioaggregator: Split base class from audiomixer

Also:
-  Don't modify size on early buffer
   The size is the size of the buffer, not of remaining part.
- Use the input caps when manipulating the input buffer
   Also store in in the sink pad
- Reply to the position query in bytes too
- Put GAP flag on output if all inputs are GAP data
- Only try to clip buffer if the incoming segment is in time or samples
- Use incoming segment with incoming timestamp
   Handle non-time segments and NONE timestamps
- Don't reset the position when pushing out new caps
- Make a number of member variables private
- Correctly handle case where no pad has a buffer
  If none of the pads have buffers that can be handled, don't claim to be EOS.
- Ensure proper locking
- Only support time segments

https://bugzilla.gnome.org/show_bug.cgi?id=740236

gst/audiomixer/Makefile.am

@@ -4,7 +4,7 @@ ORC_SOURCE=gstaudiomixerorc
 include $(top_srcdir)/common/orc.mak
 
 
-libgstaudiomixer_la_SOURCES = gstaudiomixer.c
+libgstaudiomixer_la_SOURCES = gstaudiomixer.c gstaudioaggregator.c
 nodist_libgstaudiomixer_la_SOURCES = $(ORC_NODIST_SOURCES)
 libgstaudiomixer_la_CFLAGS = \
 	-I$(top_srcdir)/gst-libs \
@@ -18,5 +18,5 @@ libgstaudiomixer_la_LIBADD =  \
 		$(GST_BASE_LIBS) $(GST_LIBS) $(ORC_LIBS)
 libgstaudiomixer_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
 
-noinst_HEADERS = gstaudiomixer.h
+noinst_HEADERS = gstaudiomixer.h gstaudioaggregator.h
 

gst/audiomixer/gstaudioaggregator.h

@@ -0,0 +1,171 @@
+/* GStreamer
+ * Copyright (C) 2014 Collabora
+ *   Author: Olivier Crete <olivier.crete@collabora.com>
+ *
+ * gstaudioaggregator.h:
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifndef __GST_AUDIO_AGGREGATOR_H__
+#define __GST_AUDIO_AGGREGATOR_H__
+
+#ifndef GST_USE_UNSTABLE_API
+#warning "The Base library from gst-plugins-bad is unstable API and may change in future."
+#warning "You can define GST_USE_UNSTABLE_API to avoid this warning."
+#endif
+
+#include <gst/gst.h>
+#include <gst/base/gstaggregator.h>
+#include <gst/audio/audio.h>
+
+G_BEGIN_DECLS
+
+/*******************************
+ * GstAudioAggregator Structs  *
+ *******************************/
+
+typedef struct _GstAudioAggregator GstAudioAggregator;
+typedef struct _GstAudioAggregatorPrivate GstAudioAggregatorPrivate;
+typedef struct _GstAudioAggregatorClass GstAudioAggregatorClass;
+
+
+/************************
+ * GstAudioAggregatorPad API *
+ ***********************/
+
+#define GST_TYPE_AUDIO_AGGREGATOR_PAD            (gst_audio_aggregator_pad_get_type())
+#define GST_AUDIO_AGGREGATOR_PAD(obj)            (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_AGGREGATOR_PAD, GstAudioAggregatorPad))
+#define GST_AUDIO_AGGREGATOR_PAD_CLASS(klass)    (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_AGGREGATOR_PAD, GstAudioAggregatorPadClass))
+#define GST_AUDIO_AGGREGATOR_PAD_GET_CLASS(obj)  (G_TYPE_INSTANCE_GET_CLASS ((obj),GST_TYPE_AUDIO_AGGREGATOR_PAD, GstAudioAggregatorPadClass))
+#define GST_IS_AUDIO_AGGREGATOR_PAD(obj)         (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_AGGREGATOR_PAD))
+#define GST_IS_AUDIO_AGGREGATOR_PAD_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_AGGREGATOR_PAD))
+
+/****************************
+ * GstAudioAggregatorPad Structs *
+ ***************************/
+
+typedef struct _GstAudioAggregatorPad GstAudioAggregatorPad;
+typedef struct _GstAudioAggregatorPadClass GstAudioAggregatorPadClass;
+typedef struct _GstAudioAggregatorPadPrivate GstAudioAggregatorPadPrivate;
+
+/**
+ * GstAudioAggregatorPad:
+ * @parent: The parent #GstAggregatorPad
+ * @info: The audio info for this pad set from the incoming caps
+ *
+ * The implementation the GstPad to use with #GstAudioAggregator
+ */
+struct _GstAudioAggregatorPad
+{
+  GstAggregatorPad                  parent;
+
+  GstAudioInfo                      info;
+
+  /*< private >*/
+  GstAudioAggregatorPadPrivate   *  priv;
+
+  gpointer _gst_reserved[GST_PADDING];
+};
+
+/**
+ * GstAudioAggregatorPadClass:
+ *
+ */
+struct _GstAudioAggregatorPadClass
+{
+  GstAggregatorPadClass   parent_class;
+
+  /*< private >*/
+  gpointer      _gst_reserved[GST_PADDING];
+};
+
+GType gst_audio_aggregator_pad_get_type           (void);
+
+/**************************
+ * GstAudioAggregator API *
+ **************************/
+
+#define GST_TYPE_AUDIO_AGGREGATOR            (gst_audio_aggregator_get_type())
+#define GST_AUDIO_AGGREGATOR(obj)            (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_AGGREGATOR,GstAudioAggregator))
+#define GST_AUDIO_AGGREGATOR_CLASS(klass)    (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_AGGREGATOR,GstAudioAggregatorClass))
+#define GST_AUDIO_AGGREGATOR_GET_CLASS(obj)  (G_TYPE_INSTANCE_GET_CLASS ((obj),GST_TYPE_AUDIO_AGGREGATOR,GstAudioAggregatorClass))
+#define GST_IS_AUDIO_AGGREGATOR(obj)         (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_AGGREGATOR))
+#define GST_IS_AUDIO_AGGREGATOR_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_AGGREGATOR))
+
+#define GST_FLOW_CUSTOM_SUCCESS        GST_FLOW_NOT_HANDLED
+
+/**
+ * GstAudioAggregator:
+ * @parent: The parent #GstAggregator
+ * @info: The information parsed from the current caps
+ * @current_caps: The caps set by the subclass
+ *
+ * GstAudioAggregator object
+ */
+struct _GstAudioAggregator
+{
+  GstAggregator            parent;
+
+  /* All member are read only for subclasses, must hold OBJECT lock  */
+  GstAudioInfo    info;
+
+  GstCaps *current_caps;
+
+  /*< private >*/
+  GstAudioAggregatorPrivate *priv;
+
+  gpointer                 _gst_reserved[GST_PADDING];
+};
+
+/**
+ * GstAudioAggregatorClass:
+ * @create_output_buffer: Create a new output buffer contains num_frames frames.
+ * @aggregate_one_buffer: Aggregates one input buffer to the output
+ *  buffer.  The in_offset and out_offset are in "frames", which is
+ *  the size of a sample times the number of channels. Returns TRUE if
+ *  any non-silence was added to the buffer
+ */
+struct _GstAudioAggregatorClass {
+  GstAggregatorClass   parent_class;
+
+  GstBuffer * (* create_output_buffer) (GstAudioAggregator * aagg,
+      guint num_frames);
+  gboolean (* aggregate_one_buffer) (GstAudioAggregator * aagg,
+      GstAudioAggregatorPad * pad, GstBuffer * inbuf, guint in_offset,
+      GstBuffer * outbuf, guint out_offset, guint num_frames);
+
+  /*< private >*/
+  gpointer          _gst_reserved[GST_PADDING];
+};
+
+/*************************
+ * GstAggregator methods *
+ ************************/
+
+GType gst_audio_aggregator_get_type(void);
+
+void
+gst_audio_aggregator_set_sink_caps (GstAudioAggregator * aagg,
+    GstAudioAggregatorPad * pad, GstCaps * caps);
+
+gboolean
+gst_audio_aggregator_set_src_caps (GstAudioAggregator * aagg, GstCaps * caps);
+
+
+G_END_DECLS
+
+#endif /* __GST_AUDIO_AGGREGATOR_H__ */

gst/audiomixer/gstaudiomixer.h

@@ -25,8 +25,8 @@
 #define __GST_AUDIO_MIXER_H__
 
 #include <gst/gst.h>
-#include <gst/base/gstaggregator.h>
 #include <gst/audio/audio.h>
+#include "gstaudioaggregator.h"
 
 G_BEGIN_DECLS
 
@@ -49,32 +49,14 @@ typedef struct _GstAudioMixerPadClass GstAudioMixerPadClass;
  * The audiomixer object structure.
  */
 struct _GstAudioMixer {
-  GstAggregator   aggregator;
-
-  /* the next are valid for both int and float */
-  GstAudioInfo    info;
-
-  /* counters to keep track of timestamps */
-  gint64          offset;
-  /* Buffer starting at offset containing block_size samples */
-  GstBuffer      *current_buffer;
-
-  /* current caps */
-  GstCaps *current_caps;
-  gboolean send_caps;
+  GstAudioAggregator element;
 
   /* target caps (set via property) */
   GstCaps *filter_caps;
-
-  GstClockTime alignment_threshold;
-  GstClockTime discont_wait;
-
-  /* Duration of every output buffer */
-  GstClockTime output_buffer_duration;
 };
 
 struct _GstAudioMixerClass {
-  GstAggregatorClass parent_class;
+  GstAudioAggregatorClass parent_class;
 };
 
 GType    gst_audiomixer_get_type (void);
@@ -87,36 +69,17 @@ GType    gst_audiomixer_get_type (void);
 #define GST_AUDIO_MIXER_PAD_GET_CLASS(obj)  (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_AUDIO_MIXER_PAD,GstAudioMixerPadClass))
 
 struct _GstAudioMixerPad {
-  GstAggregatorPad parent;
+  GstAudioAggregatorPad parent;
 
   gdouble volume;
   gint volume_i32;
   gint volume_i16;
   gint volume_i8;
   gboolean mute;
-
-  /* < private > */
-  GstBuffer *buffer;            /* current buffer we're mixing,
-                                   for comparison with collect.buffer
-                                   to see if we need to update our
-                                   cached values. */
-  guint position, size;
-
-  guint64 output_offset;        /* Offset in output segment that
-                                   collect.pos refers to in the
-                                   current buffer. */
-
-  guint64 next_offset;          /* Next expected offset in the input segment */
-
-  /* Last time we noticed a discont */
-  GstClockTime discont_time;
-
-  /* A new unhandled segment event has been received */
-  gboolean new_segment;
 };
 
 struct _GstAudioMixerPadClass {
-  GstAggregatorPadClass parent_class;
+  GstAudioAggregatorPadClass parent_class;
 };
 
 GType gst_audiomixer_pad_get_type (void);