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