Jabber: initial version of Jingle support

1 files changed, 378 insertions, 0 deletions

diff --git a/include/gst/audio/gstaudioencoder.h b/include/gst/audio/gstaudioencoder.h
new file mode 100644
index 0000000000..e95938b5d5
--- /dev/null
+++ b/include/gst/audio/gstaudioencoder.h
@@ -0,0 +1,378 @@
+/* GStreamer
+ * Copyright (C) 2011 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>.
+ * Copyright (C) 2011 Nokia Corporation. All rights reserved.
+ *   Contact: Stefan Kost <stefan.kost@nokia.com>
+ *
+ * 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_AUDIO_H__
+#include <gst/audio/audio.h>
+#endif
+
+#ifndef __GST_AUDIO_ENCODER_H__
+#define __GST_AUDIO_ENCODER_H__
+
+#include <gst/gst.h>
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_AUDIO_ENCODER		   (gst_audio_encoder_get_type())
+#define GST_AUDIO_ENCODER(obj)		   (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_ENCODER,GstAudioEncoder))
+#define GST_AUDIO_ENCODER_CLASS(klass)    (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_ENCODER,GstAudioEncoderClass))
+#define GST_AUDIO_ENCODER_GET_CLASS(obj)  (G_TYPE_INSTANCE_GET_CLASS((obj),GST_TYPE_AUDIO_ENCODER,GstAudioEncoderClass))
+#define GST_IS_AUDIO_ENCODER(obj)	   (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_ENCODER))
+#define GST_IS_AUDIO_ENCODER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_ENCODER))
+#define GST_AUDIO_ENCODER_CAST(obj)	((GstAudioEncoder *)(obj))
+
+/**
+ * GST_AUDIO_ENCODER_SINK_NAME:
+ *
+ * the name of the templates for the sink pad
+ */
+#define GST_AUDIO_ENCODER_SINK_NAME	"sink"
+/**
+ * GST_AUDIO_ENCODER_SRC_NAME:
+ *
+ * the name of the templates for the source pad
+ */
+#define GST_AUDIO_ENCODER_SRC_NAME	        "src"
+
+/**
+ * GST_AUDIO_ENCODER_SRC_PAD:
+ * @obj: audio encoder instance
+ *
+ * Gives the pointer to the source #GstPad object of the element.
+ */
+#define GST_AUDIO_ENCODER_SRC_PAD(obj)	(GST_AUDIO_ENCODER_CAST (obj)->srcpad)
+
+/**
+ * GST_AUDIO_ENCODER_SINK_PAD:
+ * @obj: audio encoder instance
+ *
+ * Gives the pointer to the sink #GstPad object of the element.
+ */
+#define GST_AUDIO_ENCODER_SINK_PAD(obj)	(GST_AUDIO_ENCODER_CAST (obj)->sinkpad)
+
+/**
+ * GST_AUDIO_ENCODER_INPUT_SEGMENT:
+ * @obj: base parse instance
+ *
+ * Gives the input segment of the element.
+ */
+#define GST_AUDIO_ENCODER_INPUT_SEGMENT(obj)     (GST_AUDIO_ENCODER_CAST (obj)->input_segment)
+
+/**
+ * GST_AUDIO_ENCODER_OUTPUT_SEGMENT:
+ * @obj: base parse instance
+ *
+ * Gives the output segment of the element.
+ */
+#define GST_AUDIO_ENCODER_OUTPUT_SEGMENT(obj)     (GST_AUDIO_ENCODER_CAST (obj)->output_segment)
+
+#define GST_AUDIO_ENCODER_STREAM_LOCK(enc)   g_rec_mutex_lock (&GST_AUDIO_ENCODER (enc)->stream_lock)
+#define GST_AUDIO_ENCODER_STREAM_UNLOCK(enc) g_rec_mutex_unlock (&GST_AUDIO_ENCODER (enc)->stream_lock)
+
+typedef struct _GstAudioEncoder GstAudioEncoder;
+typedef struct _GstAudioEncoderClass GstAudioEncoderClass;
+
+typedef struct _GstAudioEncoderPrivate GstAudioEncoderPrivate;
+
+/**
+ * GstAudioEncoder:
+ *
+ * The opaque #GstAudioEncoder data structure.
+ */
+struct _GstAudioEncoder {
+  GstElement     element;
+
+  /*< protected >*/
+  /* source and sink pads */
+  GstPad         *sinkpad;
+  GstPad         *srcpad;
+
+  /* protects all data processing, i.e. is locked
+   * in the chain function, finish_frame and when
+   * processing serialized events */
+  GRecMutex       stream_lock;
+
+  /* MT-protected (with STREAM_LOCK) */
+  GstSegment      input_segment;
+  GstSegment      output_segment;
+
+  /*< private >*/
+  GstAudioEncoderPrivate *priv;
+
+  gpointer       _gst_reserved[GST_PADDING_LARGE];
+};
+
+/**
+ * GstAudioEncoderClass:
+ * @element_class:  The parent class structure
+ * @start:          Optional.
+ *                  Called when the element starts processing.
+ *                  Allows opening external resources.
+ * @stop:           Optional.
+ *                  Called when the element stops processing.
+ *                  Allows closing external resources.
+ * @set_format:     Notifies subclass of incoming data format.
+ *                  GstAudioInfo contains the format according to provided caps.
+ * @handle_frame:   Provides input samples (or NULL to clear any remaining data)
+ *                  according to directions as configured by the subclass
+ *                  using the API.  Input data ref management is performed
+ *                  by base class, subclass should not care or intervene,
+ *                  and input data is only valid until next call to base class,
+ *                  most notably a call to gst_audio_encoder_finish_frame().
+ * @flush:          Optional.
+ *                  Instructs subclass to clear any codec caches and discard
+ *                  any pending samples and not yet returned encoded data.
+ * @sink_event:     Optional.
+ *                  Event handler on the sink pad. Subclasses should chain up to
+ *                  the parent implementation to invoke the default handler.
+ * @src_event:      Optional.
+ *                  Event handler on the src pad. Subclasses should chain up to
+ *                  the parent implementation to invoke the default handler.
+ * @pre_push:       Optional.
+ *                  Called just prior to pushing (encoded data) buffer downstream.
+ *                  Subclass has full discretionary access to buffer,
+ *                  and a not OK flow return will abort downstream pushing.
+ * @getcaps:        Optional.
+ *                  Allows for a custom sink getcaps implementation (e.g.
+ *                  for multichannel input specification).  If not implemented,
+ *                  default returns gst_audio_encoder_proxy_getcaps
+ *                  applied to sink template caps.
+ * @open:           Optional.
+ *                  Called when the element changes to GST_STATE_READY.
+ *                  Allows opening external resources.
+ * @close:          Optional.
+ *                  Called when the element changes to GST_STATE_NULL.
+ *                  Allows closing external resources.
+ * @negotiate:      Optional.
+ *                  Negotiate with downstream and configure buffer pools, etc.
+ *                  Subclasses should chain up to the parent implementation to
+ *                  invoke the default handler.
+ * @decide_allocation: Optional.
+ *                     Setup the allocation parameters for allocating output
+ *                     buffers. The passed in query contains the result of the
+ *                     downstream allocation query.
+ *                     Subclasses should chain up to the parent implementation to
+ *                     invoke the default handler.
+ * @propose_allocation: Optional.
+ *                      Propose buffer allocation parameters for upstream elements.
+ *                      Subclasses should chain up to the parent implementation to
+ *                      invoke the default handler.
+ * @transform_meta: Optional. Transform the metadata on the input buffer to the
+ *                  output buffer. By default this method copies all meta without
+ *                  tags and meta with only the "audio" tag. subclasses can
+ *                  implement this method and return %TRUE if the metadata is to be
+ *                  copied. Since: 1.6
+ * @sink_query:     Optional.
+ *                  Query handler on the sink pad. This function should
+ *                  return TRUE if the query could be performed. Subclasses
+ *                  should chain up to the parent implementation to invoke the
+ *                  default handler. Since: 1.6
+ * @src_query:      Optional.
+ *                  Query handler on the source pad. This function should
+ *                  return TRUE if the query could be performed. Subclasses
+ *                  should chain up to the parent implementation to invoke the
+ *                  default handler. Since: 1.6
+ *
+ * Subclasses can override any of the available virtual methods or not, as
+ * needed. At minimum @set_format and @handle_frame needs to be overridden.
+ */
+struct _GstAudioEncoderClass {
+  GstElementClass element_class;
+
+  /*< public >*/
+  /* virtual methods for subclasses */
+
+  gboolean      (*start)              (GstAudioEncoder *enc);
+
+  gboolean      (*stop)               (GstAudioEncoder *enc);
+
+  gboolean      (*set_format)         (GstAudioEncoder *enc,
+                                       GstAudioInfo        *info);
+
+  GstFlowReturn (*handle_frame)       (GstAudioEncoder *enc,
+                                       GstBuffer *buffer);
+
+  void          (*flush)              (GstAudioEncoder *enc);
+
+  GstFlowReturn (*pre_push)           (GstAudioEncoder *enc,
+                                       GstBuffer **buffer);
+
+  gboolean      (*sink_event)         (GstAudioEncoder *enc,
+                                       GstEvent *event);
+
+  gboolean      (*src_event)          (GstAudioEncoder *enc,
+                                       GstEvent *event);
+
+  GstCaps *     (*getcaps)            (GstAudioEncoder *enc, GstCaps *filter);
+
+  gboolean      (*open)               (GstAudioEncoder *enc);
+
+  gboolean      (*close)              (GstAudioEncoder *enc);
+
+  gboolean      (*negotiate)          (GstAudioEncoder *enc);
+
+  gboolean      (*decide_allocation)  (GstAudioEncoder *enc, GstQuery *query);
+
+  gboolean      (*propose_allocation) (GstAudioEncoder * enc,
+                                       GstQuery * query);
+
+  gboolean      (*transform_meta)     (GstAudioEncoder *enc, GstBuffer *outbuf,
+                                       GstMeta *meta, GstBuffer *inbuf);
+
+  gboolean      (*sink_query)         (GstAudioEncoder *encoder,
+				       GstQuery *query);
+
+  gboolean      (*src_query)          (GstAudioEncoder *encoder,
+				       GstQuery *query);
+
+
+  /*< private >*/
+  gpointer       _gst_reserved[GST_PADDING_LARGE-3];
+};
+
+GST_AUDIO_API
+GType           gst_audio_encoder_get_type         (void);
+
+GST_AUDIO_API
+GstFlowReturn   gst_audio_encoder_finish_frame (GstAudioEncoder * enc,
+                                                GstBuffer       * buffer,
+                                                gint              samples);
+
+GST_AUDIO_API
+GstCaps *       gst_audio_encoder_proxy_getcaps (GstAudioEncoder * enc,
+                                                 GstCaps         * caps,
+                                                 GstCaps         * filter);
+
+GST_AUDIO_API
+gboolean        gst_audio_encoder_set_output_format  (GstAudioEncoder    * enc,
+                                                      GstCaps            * caps);
+
+GST_AUDIO_API
+gboolean        gst_audio_encoder_negotiate          (GstAudioEncoder * enc);
+
+GST_AUDIO_API
+GstBuffer *     gst_audio_encoder_allocate_output_buffer (GstAudioEncoder * enc,
+                                                          gsize             size);
+
+/* context parameters */
+
+GST_AUDIO_API
+GstAudioInfo  * gst_audio_encoder_get_audio_info (GstAudioEncoder * enc);
+
+GST_AUDIO_API
+gint            gst_audio_encoder_get_frame_samples_min (GstAudioEncoder * enc);
+
+GST_AUDIO_API
+void            gst_audio_encoder_set_frame_samples_min (GstAudioEncoder * enc, gint num);
+
+GST_AUDIO_API
+gint            gst_audio_encoder_get_frame_samples_max (GstAudioEncoder * enc);
+
+GST_AUDIO_API
+void            gst_audio_encoder_set_frame_samples_max (GstAudioEncoder * enc, gint num);
+
+GST_AUDIO_API
+gint            gst_audio_encoder_get_frame_max (GstAudioEncoder * enc);
+
+GST_AUDIO_API
+void            gst_audio_encoder_set_frame_max (GstAudioEncoder * enc, gint num);
+
+GST_AUDIO_API
+gint            gst_audio_encoder_get_lookahead (GstAudioEncoder * enc);
+
+GST_AUDIO_API
+void            gst_audio_encoder_set_lookahead (GstAudioEncoder * enc, gint num);
+
+GST_AUDIO_API
+void            gst_audio_encoder_get_latency (GstAudioEncoder * enc,
+                                               GstClockTime    * min,
+                                               GstClockTime    * max);
+
+GST_AUDIO_API
+void            gst_audio_encoder_set_latency (GstAudioEncoder * enc,
+                                               GstClockTime      min,
+                                               GstClockTime      max);
+
+GST_AUDIO_API
+void            gst_audio_encoder_set_headers (GstAudioEncoder * enc,
+                                               GList           * headers);
+
+GST_AUDIO_API
+void            gst_audio_encoder_set_allocation_caps (GstAudioEncoder * enc,
+                                                       GstCaps         * allocation_caps);
+
+/* object properties */
+
+GST_AUDIO_API
+void            gst_audio_encoder_set_mark_granule (GstAudioEncoder * enc,
+                                                    gboolean enabled);
+
+GST_AUDIO_API
+gboolean        gst_audio_encoder_get_mark_granule (GstAudioEncoder * enc);
+
+GST_AUDIO_API
+void            gst_audio_encoder_set_perfect_timestamp (GstAudioEncoder * enc,
+                                                         gboolean          enabled);
+
+GST_AUDIO_API
+gboolean        gst_audio_encoder_get_perfect_timestamp (GstAudioEncoder * enc);
+
+GST_AUDIO_API
+void            gst_audio_encoder_set_hard_resync (GstAudioEncoder * enc,
+                                                   gboolean          enabled);
+
+GST_AUDIO_API
+gboolean        gst_audio_encoder_get_hard_resync (GstAudioEncoder * enc);
+
+GST_AUDIO_API
+void            gst_audio_encoder_set_tolerance (GstAudioEncoder * enc,
+                                                 GstClockTime      tolerance);
+
+GST_AUDIO_API
+GstClockTime    gst_audio_encoder_get_tolerance (GstAudioEncoder * enc);
+
+GST_AUDIO_API
+void            gst_audio_encoder_set_hard_min (GstAudioEncoder * enc,
+                                                gboolean enabled);
+
+GST_AUDIO_API
+gboolean        gst_audio_encoder_get_hard_min (GstAudioEncoder * enc);
+
+GST_AUDIO_API
+void            gst_audio_encoder_set_drainable (GstAudioEncoder * enc,
+                                                 gboolean enabled);
+
+GST_AUDIO_API
+gboolean        gst_audio_encoder_get_drainable (GstAudioEncoder * enc);
+
+GST_AUDIO_API
+void            gst_audio_encoder_get_allocator (GstAudioEncoder * enc,
+                                                 GstAllocator ** allocator,
+                                                 GstAllocationParams * params);
+
+GST_AUDIO_API
+void            gst_audio_encoder_merge_tags (GstAudioEncoder * enc,
+                                              const GstTagList * tags, GstTagMergeMode mode);
+
+G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstAudioEncoder, gst_object_unref)
+
+G_END_DECLS
+
+#endif /* __GST_AUDIO_ENCODER_H__ */