mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-19 14:56:36 +00:00
220 lines
8.2 KiB
C
220 lines
8.2 KiB
C
/* GStreamer
|
|
* Copyright (C) 2009 Igalia S.L.
|
|
* Author: Iago Toral Quiroga <itoral@igalia.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., 59 Temple Place - Suite 330,
|
|
* Boston, MA 02111-1307, USA.
|
|
*/
|
|
|
|
#ifndef _GST_BASE_AUDIO_DECODER_H_
|
|
#define _GST_BASE_AUDIO_DECODER_H_
|
|
|
|
#ifndef GST_USE_UNSTABLE_API
|
|
#warning "GstBaseAudioDecoder 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/gstadapter.h>
|
|
|
|
G_BEGIN_DECLS
|
|
|
|
#define GST_TYPE_BASE_AUDIO_DECODER \
|
|
(gst_base_audio_decoder_get_type())
|
|
#define GST_BASE_AUDIO_DECODER(obj) \
|
|
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_BASE_AUDIO_DECODER,GstBaseAudioDecoder))
|
|
#define GST_BASE_AUDIO_DECODER_CLASS(klass) \
|
|
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_BASE_AUDIO_DECODER,GstBaseAudioDecoderClass))
|
|
#define GST_BASE_AUDIO_DECODER_GET_CLASS(obj) \
|
|
(G_TYPE_INSTANCE_GET_CLASS((obj),GST_TYPE_BASE_AUDIO_DECODER,GstBaseAudioDecoderClass))
|
|
#define GST_IS_BASE_AUDIO_DECODER(obj) \
|
|
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_BASE_AUDIO_DECODER))
|
|
#define GST_IS_BASE_AUDIO_DECODER_CLASS(obj) \
|
|
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_BASE_AUDIO_DECODER))
|
|
|
|
/**
|
|
* GST_BASE_AUDIO_DECODER_SINK_NAME:
|
|
*
|
|
* The name of the templates for the sink pad.
|
|
*/
|
|
#define GST_BASE_AUDIO_DECODER_SINK_NAME "sink"
|
|
/**
|
|
* GST_BASE_AUDIO_DECODER_SRC_NAME:
|
|
*
|
|
* The name of the templates for the source pad.
|
|
*/
|
|
#define GST_BASE_AUDIO_DECODER_SRC_NAME "src"
|
|
|
|
/**
|
|
* GST_BASE_AUDIO_DECODER_SRC_PAD:
|
|
* @obj: base audio codec instance
|
|
*
|
|
* Gives the pointer to the source #GstPad object of the element.
|
|
*/
|
|
#define GST_BASE_AUDIO_DECODER_SRC_PAD(obj) (((GstBaseAudioDecoder *) (obj))->srcpad)
|
|
|
|
/**
|
|
* GST_BASE_AUDIO_DECODER_SINK_PAD:
|
|
* @obj: base audio codec instance
|
|
*
|
|
* Gives the pointer to the sink #GstPad object of the element.
|
|
*/
|
|
#define GST_BASE_AUDIO_DECODER_SINK_PAD(obj) (((GstBaseAudioDecoder *) (obj))->sinkpad)
|
|
|
|
/**
|
|
* GST_BASE_AUDIO_DECODER_INPUT_ADAPTER:
|
|
* @obj: base audio codec instance
|
|
*
|
|
* Gives the pointer to the input #GstAdapter object of the element.
|
|
*/
|
|
#define GST_BASE_AUDIO_DECODER_INPUT_ADAPTER(obj) (((GstBaseAudioDecoder *) (obj))->input_adapter)
|
|
|
|
/**
|
|
* GST_BASE_AUDIO_DECODER_OUTPUT_ADAPTER:
|
|
* @obj: base audio codec instance
|
|
*
|
|
* Gives the pointer to the output #GstAdapter object of the element.
|
|
*/
|
|
#define GST_BASE_AUDIO_DECODER_OUTPUT_ADAPTER(obj) (((GstBaseAudioDecoder *) (obj))->output_adapter)
|
|
|
|
typedef struct _GstBaseAudioDecoder GstBaseAudioDecoder;
|
|
typedef struct _GstBaseAudioDecoderClass GstBaseAudioDecoderClass;
|
|
typedef struct _GstAudioState GstAudioState;
|
|
|
|
struct _GstAudioState
|
|
{
|
|
gint channels;
|
|
gint rate;
|
|
gint bytes_per_sample;
|
|
gint sample_depth;
|
|
gint frame_size;
|
|
GstSegment segment;
|
|
};
|
|
|
|
/**
|
|
* GstBaseAudioDecoder:
|
|
* @element: the parent element.
|
|
* @caps_set: whether caps have been set on the codec's source pad.
|
|
* @sinkpad: the sink pad.
|
|
* @srcpad: the source pad.
|
|
* @input_adapter: the input adapter that will be filled with the input buffers.
|
|
* @output_adapter: the output adapter. Subclasses will read from the input
|
|
* adapter, process the data and fill the output adapter with the result.
|
|
* @input_buffer_size: The minimum amount of data that should be present on the
|
|
* input adapter for the codec to process it.
|
|
* @output_buffer_size: The minimum amount of data that should be present on the
|
|
* output adapter for the codec to push buffers out.
|
|
* @bytes_in: total bytes that have been received.
|
|
* @bytes_out: total bytes that have been pushed out.
|
|
* @discont: whether the next buffer to push represents a discontinuity in the
|
|
* stream.
|
|
* @state: Audio stream information. See #GstAudioState for details.
|
|
* @codec_data: The codec data.
|
|
* @started: Whether the codec has been started and is ready to process data
|
|
* or not.
|
|
* @first_ts: timestamp of the first buffer in the input adapter.
|
|
* @last_ts: timestamp of the last buffer in the input adapter.
|
|
*
|
|
* The opaque #GstBaseAudioDecoder data structure.
|
|
*/
|
|
struct _GstBaseAudioDecoder
|
|
{
|
|
GstElement element;
|
|
|
|
/*< private >*/
|
|
gboolean caps_set;
|
|
|
|
/*< protected >*/
|
|
GstPad *sinkpad;
|
|
GstPad *srcpad;
|
|
GstAdapter *input_adapter;
|
|
GstAdapter *output_adapter;
|
|
guint input_buffer_size;
|
|
guint output_buffer_size;
|
|
guint64 bytes_in;
|
|
guint64 bytes_out;
|
|
gboolean discont;
|
|
GstAudioState state;
|
|
GstBuffer *codec_data;
|
|
gboolean started;
|
|
|
|
guint64 first_ts;
|
|
guint64 last_ts;
|
|
};
|
|
|
|
/**
|
|
* GstBaseAudioDecoderClass:
|
|
* @parent_class: Element parent class
|
|
* @start: Start processing. Ideal for opening resources in the subclass
|
|
* @stop: Stop processing. Subclasses should use this to close resources.
|
|
* @reset: Resets the codec. Called on discontinuities, etc.
|
|
* @event: Override this to handle events arriving on the sink pad.
|
|
* @handle_discont: Override to be notified on discontinuities.
|
|
* @flush_input: Subclasses may implement this to flush the input adapter,
|
|
* processing any data present in it and filling the output adapter with the
|
|
* result. This could be necessary if it is possible for the codec to
|
|
* receive an end-of-stream event before all the data in the input
|
|
* adapter has been processed.
|
|
* @flush_output: Subclasses may implement this to flush the output adapter,
|
|
* pushing buffers out through the codec's source pad when the end-of-stream
|
|
* event is received and there is data waiting to be processed in the
|
|
* adapters.
|
|
* @process_data: Subclasses must implement this. They should read from the
|
|
* input adapter, encode/decode the data present in it and fill the
|
|
* output adapter with the result.
|
|
* @push_data: Normally, #GstBaseAudioDecoder will handle pushing buffers out.
|
|
* However, it is possible for developers to take control of when and how
|
|
* buffers are pushed out by overriding this method. If subclasses provide
|
|
* an implementation, #GstBaseAudioDecoder will not push any buffers,
|
|
* instead, whenever there is data on the output adapter, it will call this
|
|
* method on the subclass, which would be the sole responsible for
|
|
* pushing the buffers out when appropriate.
|
|
* @negotiate_src_caps: Subclasses can implement this method to provide
|
|
* appropriate caps to be set on the codec's source pad. If they don't
|
|
* provide this, they will be responsible for calling
|
|
* gst_base_audio_decoder_set_src_caps when appropriate.
|
|
*/
|
|
struct _GstBaseAudioDecoderClass
|
|
{
|
|
GstElementClass parent_class;
|
|
|
|
gboolean (*start) (GstBaseAudioDecoder *codec);
|
|
gboolean (*stop) (GstBaseAudioDecoder *codec);
|
|
gboolean (*reset) (GstBaseAudioDecoder *codec);
|
|
|
|
GstFlowReturn (*event) (GstBaseAudioDecoder *codec, GstEvent *event);
|
|
void (*handle_discont) (GstBaseAudioDecoder *codec, GstBuffer *buffer);
|
|
gboolean (*flush_input) (GstBaseAudioDecoder *codec);
|
|
gboolean (*flush_output) (GstBaseAudioDecoder *codec);
|
|
GstFlowReturn (*process_data) (GstBaseAudioDecoder *codec);
|
|
GstFlowReturn (*push_data) (GstBaseAudioDecoder *codec);
|
|
GstCaps * (*negotiate_src_caps) (GstBaseAudioDecoder *codec,
|
|
GstCaps *sink_caps);
|
|
};
|
|
|
|
GType gst_base_audio_decoder_get_type (void);
|
|
gboolean gst_base_audio_decoder_reset (GstBaseAudioDecoder *codec);
|
|
gboolean gst_base_audio_decoder_stop (GstBaseAudioDecoder *codec);
|
|
gboolean gst_base_audio_decoder_start (GstBaseAudioDecoder *codec);
|
|
gboolean gst_base_audio_decoder_flush (GstBaseAudioDecoder *codec);
|
|
gboolean gst_base_audio_decoder_set_src_caps (GstBaseAudioDecoder *codec,
|
|
GstCaps *caps);
|
|
GstFlowReturn gst_base_audio_decoder_push_buffer (GstBaseAudioDecoder *codec,
|
|
GstBuffer *buffer);
|
|
|
|
G_END_DECLS
|
|
|
|
#endif
|
|
|