mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-11 02:46:33 +00:00
959f8e4a3e
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
171 lines
5.7 KiB
C
171 lines
5.7 KiB
C
/* 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__ */
|