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301 lines
12 KiB
C
301 lines
12 KiB
C
/* GStreamer
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* Copyright (C) 2011 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>.
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* Copyright (C) 2011 Nokia Corporation. All rights reserved.
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* Contact: Stefan Kost <stefan.kost@nokia.com>
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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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#ifndef __GST_AUDIO_ENCODER_H__
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#define __GST_AUDIO_ENCODER_H__
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#include <gst/gst.h>
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#include <gst/audio/audio.h>
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G_BEGIN_DECLS
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#define GST_TYPE_AUDIO_ENCODER (gst_audio_encoder_get_type())
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#define GST_AUDIO_ENCODER(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_ENCODER,GstAudioEncoder))
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#define GST_AUDIO_ENCODER_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_ENCODER,GstAudioEncoderClass))
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#define GST_AUDIO_ENCODER_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj),GST_TYPE_AUDIO_ENCODER,GstAudioEncoderClass))
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#define GST_IS_AUDIO_ENCODER(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_ENCODER))
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#define GST_IS_AUDIO_ENCODER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_ENCODER))
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#define GST_AUDIO_ENCODER_CAST(obj) ((GstAudioEncoder *)(obj))
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/**
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* GST_AUDIO_ENCODER_SINK_NAME:
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*
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* the name of the templates for the sink pad
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*/
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#define GST_AUDIO_ENCODER_SINK_NAME "sink"
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/**
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* GST_AUDIO_ENCODER_SRC_NAME:
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*
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* the name of the templates for the source pad
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*/
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#define GST_AUDIO_ENCODER_SRC_NAME "src"
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/**
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* GST_AUDIO_ENCODER_SRC_PAD:
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* @obj: audio encoder instance
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*
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* Gives the pointer to the source #GstPad object of the element.
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*/
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#define GST_AUDIO_ENCODER_SRC_PAD(obj) (GST_AUDIO_ENCODER_CAST (obj)->srcpad)
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/**
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* GST_AUDIO_ENCODER_SINK_PAD:
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* @obj: audio encoder instance
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*
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* Gives the pointer to the sink #GstPad object of the element.
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*/
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#define GST_AUDIO_ENCODER_SINK_PAD(obj) (GST_AUDIO_ENCODER_CAST (obj)->sinkpad)
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/**
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* GST_AUDIO_ENCODER_INPUT_SEGMENT:
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* @obj: base parse instance
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*
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* Gives the input segment of the element.
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*/
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#define GST_AUDIO_ENCODER_INPUT_SEGMENT(obj) (GST_AUDIO_ENCODER_CAST (obj)->input_segment)
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/**
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* GST_AUDIO_ENCODER_OUTPUT_SEGMENT:
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* @obj: base parse instance
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*
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* Gives the output segment of the element.
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*/
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#define GST_AUDIO_ENCODER_OUTPUT_SEGMENT(obj) (GST_AUDIO_ENCODER_CAST (obj)->output_segment)
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#define GST_AUDIO_ENCODER_STREAM_LOCK(enc) g_rec_mutex_lock (&GST_AUDIO_ENCODER (enc)->stream_lock)
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#define GST_AUDIO_ENCODER_STREAM_UNLOCK(enc) g_rec_mutex_unlock (&GST_AUDIO_ENCODER (enc)->stream_lock)
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typedef struct _GstAudioEncoder GstAudioEncoder;
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typedef struct _GstAudioEncoderClass GstAudioEncoderClass;
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typedef struct _GstAudioEncoderPrivate GstAudioEncoderPrivate;
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/**
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* GstAudioEncoder:
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*
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* The opaque #GstAudioEncoder data structure.
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*/
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struct _GstAudioEncoder {
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GstElement element;
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/*< protected >*/
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/* source and sink pads */
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GstPad *sinkpad;
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GstPad *srcpad;
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/* protects all data processing, i.e. is locked
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* in the chain function, finish_frame and when
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* processing serialized events */
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GRecMutex stream_lock;
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/* MT-protected (with STREAM_LOCK) */
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GstSegment input_segment;
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GstSegment output_segment;
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/*< private >*/
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GstAudioEncoderPrivate *priv;
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gpointer _gst_reserved[GST_PADDING_LARGE];
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};
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/**
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* GstAudioEncoderClass:
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* @element_class: The parent class structure
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* @start: Optional.
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* Called when the element starts processing.
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* Allows opening external resources.
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* @stop: Optional.
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* Called when the element stops processing.
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* Allows closing external resources.
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* @set_format: Notifies subclass of incoming data format.
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* GstAudioInfo contains the format according to provided caps.
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* @handle_frame: Provides input samples (or NULL to clear any remaining data)
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* according to directions as configured by the subclass
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* using the API. Input data ref management is performed
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* by base class, subclass should not care or intervene,
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* and input data is only valid until next call to base class,
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* most notably a call to gst_audio_encoder_finish_frame().
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* @flush: Optional.
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* Instructs subclass to clear any codec caches and discard
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* any pending samples and not yet returned encoded data.
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* @sink_event: Optional.
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* Event handler on the sink pad. Subclasses should chain up to
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* the parent implementation to invoke the default handler.
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* @src_event: Optional.
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* Event handler on the src pad. Subclasses should chain up to
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* the parent implementation to invoke the default handler.
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* @pre_push: Optional.
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* Called just prior to pushing (encoded data) buffer downstream.
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* Subclass has full discretionary access to buffer,
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* and a not OK flow return will abort downstream pushing.
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* @getcaps: Optional.
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* Allows for a custom sink getcaps implementation (e.g.
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* for multichannel input specification). If not implemented,
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* default returns gst_audio_encoder_proxy_getcaps
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* applied to sink template caps.
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* @open: Optional.
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* Called when the element changes to GST_STATE_READY.
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* Allows opening external resources.
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* @close: Optional.
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* Called when the element changes to GST_STATE_NULL.
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* Allows closing external resources.
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* @negotiate: Optional.
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* Negotiate with downstream and configure buffer pools, etc.
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* @decide_allocation: Optional.
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* Setup the allocation parameters for allocating output
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* buffers. The passed in query contains the result of the
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* downstream allocation query.
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* @propose_allocation: Optional.
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* Propose buffer allocation parameters for upstream elements.
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*
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* Subclasses can override any of the available virtual methods or not, as
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* needed. At minimum @set_format and @handle_frame needs to be overridden.
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*/
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struct _GstAudioEncoderClass {
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GstElementClass element_class;
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/*< public >*/
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/* virtual methods for subclasses */
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gboolean (*start) (GstAudioEncoder *enc);
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gboolean (*stop) (GstAudioEncoder *enc);
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gboolean (*set_format) (GstAudioEncoder *enc,
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GstAudioInfo *info);
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GstFlowReturn (*handle_frame) (GstAudioEncoder *enc,
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GstBuffer *buffer);
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void (*flush) (GstAudioEncoder *enc);
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GstFlowReturn (*pre_push) (GstAudioEncoder *enc,
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GstBuffer **buffer);
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gboolean (*sink_event) (GstAudioEncoder *enc,
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GstEvent *event);
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gboolean (*src_event) (GstAudioEncoder *enc,
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GstEvent *event);
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GstCaps * (*getcaps) (GstAudioEncoder *enc, GstCaps *filter);
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gboolean (*open) (GstAudioEncoder *enc);
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gboolean (*close) (GstAudioEncoder *enc);
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gboolean (*negotiate) (GstAudioEncoder *enc);
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gboolean (*decide_allocation) (GstAudioEncoder *enc, GstQuery *query);
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gboolean (*propose_allocation) (GstAudioEncoder * enc,
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GstQuery * query);
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/*< private >*/
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gpointer _gst_reserved[GST_PADDING_LARGE-2];
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};
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GType gst_audio_encoder_get_type (void);
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GstFlowReturn gst_audio_encoder_finish_frame (GstAudioEncoder * enc,
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GstBuffer * buffer,
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gint samples);
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GstCaps * gst_audio_encoder_proxy_getcaps (GstAudioEncoder * enc,
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GstCaps * caps,
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GstCaps * filter);
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gboolean gst_audio_encoder_set_output_format (GstAudioEncoder * enc,
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GstCaps * caps);
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gboolean gst_audio_encoder_negotiate (GstAudioEncoder * enc);
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GstBuffer * gst_audio_encoder_allocate_output_buffer (GstAudioEncoder * encoder,
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gsize size);
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/* context parameters */
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GstAudioInfo * gst_audio_encoder_get_audio_info (GstAudioEncoder * enc);
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gint gst_audio_encoder_get_frame_samples_min (GstAudioEncoder * enc);
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void gst_audio_encoder_set_frame_samples_min (GstAudioEncoder * enc, gint num);
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gint gst_audio_encoder_get_frame_samples_max (GstAudioEncoder * enc);
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void gst_audio_encoder_set_frame_samples_max (GstAudioEncoder * enc, gint num);
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gint gst_audio_encoder_get_frame_max (GstAudioEncoder * enc);
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void gst_audio_encoder_set_frame_max (GstAudioEncoder * enc, gint num);
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gint gst_audio_encoder_get_lookahead (GstAudioEncoder * enc);
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void gst_audio_encoder_set_lookahead (GstAudioEncoder * enc, gint num);
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void gst_audio_encoder_get_latency (GstAudioEncoder * enc,
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GstClockTime * min,
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GstClockTime * max);
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void gst_audio_encoder_set_latency (GstAudioEncoder * enc,
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GstClockTime min,
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GstClockTime max);
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void gst_audio_encoder_set_headers (GstAudioEncoder * enc,
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GList * headers);
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/* object properties */
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void gst_audio_encoder_set_mark_granule (GstAudioEncoder * enc,
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gboolean enabled);
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gboolean gst_audio_encoder_get_mark_granule (GstAudioEncoder * enc);
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void gst_audio_encoder_set_perfect_timestamp (GstAudioEncoder * enc,
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gboolean enabled);
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gboolean gst_audio_encoder_get_perfect_timestamp (GstAudioEncoder * enc);
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void gst_audio_encoder_set_hard_resync (GstAudioEncoder * enc,
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gboolean enabled);
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gboolean gst_audio_encoder_get_hard_resync (GstAudioEncoder * enc);
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void gst_audio_encoder_set_tolerance (GstAudioEncoder * enc,
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gint64 tolerance);
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gint64 gst_audio_encoder_get_tolerance (GstAudioEncoder * enc);
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void gst_audio_encoder_set_hard_min (GstAudioEncoder * enc,
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gboolean enabled);
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gboolean gst_audio_encoder_get_hard_min (GstAudioEncoder * enc);
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void gst_audio_encoder_set_drainable (GstAudioEncoder * enc,
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gboolean enabled);
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gboolean gst_audio_encoder_get_drainable (GstAudioEncoder * enc);
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void gst_audio_encoder_merge_tags (GstAudioEncoder * enc,
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const GstTagList * tags, GstTagMergeMode mode);
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G_END_DECLS
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#endif /* __GST_AUDIO_ENCODER_H__ */
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