/* GStreamer * Copyright (C) 2009 Igalia S.L. * Author: Iago Toral Quiroga * Copyright (C) 2011 Mark Nauwelaerts . * Copyright (C) 2011 Nokia Corporation. All rights reserved. * Contact: Stefan Kost * * 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 #include #include 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) typedef struct _GstBaseAudioDecoder GstBaseAudioDecoder; typedef struct _GstBaseAudioDecoderClass GstBaseAudioDecoderClass; typedef struct _GstBaseAudioDecoderPrivate GstBaseAudioDecoderPrivate; typedef struct _GstBaseAudioDecoderContext GstBaseAudioDecoderContext; /* do not use this one, use macro below */ GstFlowReturn _gst_base_audio_decoder_error (GstBaseAudioDecoder *dec, gint weight, GQuark domain, gint code, gchar *txt, gchar *debug, const gchar *file, const gchar *function, gint line); /** * GST_BASE_AUDIO_DECODER_ERROR: * @el: the base audio decoder element that generates the error * @weight: element defined weight of the error, added to error count * @domain: like CORE, LIBRARY, RESOURCE or STREAM (see #gstreamer-GstGError) * @code: error code defined for that domain (see #gstreamer-GstGError) * @text: the message to display (format string and args enclosed in * parentheses) * @debug: debugging information for the message (format string and args * enclosed in parentheses) * @ret: variable to receive return value * * Utility function that audio decoder elements can use in case they encountered * a data processing error that may be fatal for the current "data unit" but * need not prevent subsequent decoding. Such errors are counted and if there * are too many, as configured in the context's max_errors, the pipeline will * post an error message and the application will be requested to stop further * media processing. Otherwise, it is considered a "glitch" and only a warning * is logged. In either case, @ret is set to the proper value to * return to upstream/caller (indicating either GST_FLOW_ERROR or GST_FLOW_OK). */ #define GST_BASE_AUDIO_DECODER_ERROR(el, w, domain, code, text, debug, ret) \ G_STMT_START { \ gchar *__txt = _gst_element_error_printf text; \ gchar *__dbg = _gst_element_error_printf debug; \ GstBaseAudioDecoder *dec = GST_BASE_AUDIO_DECODER (el); \ ret = _gst_base_audio_decoder_error (dec, w, GST_ ## domain ## _ERROR, \ GST_ ## domain ## _ERROR_ ## code, __txt, __dbg, __FILE__, \ GST_FUNCTION, __LINE__); \ } G_STMT_END /** * GstBaseAudioDecoderContext: * @state: a #GstAudioState describing input audio format * @eos: no (immediate) subsequent data in stream * @sync: stream parsing in sync * @delay: number of frames pending decoding (typically at least 1 for current) * @do_plc: whether subclass is prepared to handle (packet) loss concealment * @min_latency: min latency of element * @max_latency: max latency of element * @lookahead: decoder lookahead (in units of input rate samples) * * Transparent #GstBaseAudioEncoderContext data structure. */ struct _GstBaseAudioDecoderContext { /* input */ /* (output) audio format */ GstAudioState state; /* parsing state */ gboolean eos; gboolean sync; /* misc */ gint delay; /* output */ gboolean do_plc; gboolean do_byte_time; gint max_errors; /* MT-protected (with LOCK) */ GstClockTime min_latency; GstClockTime max_latency; }; /** * GstBaseAudioDecoder: * * The opaque #GstBaseAudioDecoder data structure. */ struct _GstBaseAudioDecoder { GstElement element; /*< protected >*/ /* source and sink pads */ GstPad *sinkpad; GstPad *srcpad; /* MT-protected (with STREAM_LOCK) */ GstSegment segment; GstBaseAudioDecoderContext *ctx; /* properties */ GstClockTime latency; GstClockTime tolerance; gboolean plc; /*< private >*/ GstBaseAudioDecoderPrivate *priv; gpointer _gst_reserved[GST_PADDING_LARGE]; }; /** * GstBaseAudioDecoderClass: * @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 (caps). * @parse: Optional. * Allows chopping incoming data into manageable units (frames) * for subsequent decoding. This division is at subclass * discretion and may or may not correspond to 1 (or more) * frames as defined by audio format. * @handle_frame: Provides input data (or NULL to clear any remaining data) * to subclass. Input data ref management is performed by * base class, subclass should not care or intervene. * @flush: Optional. * Instructs subclass to clear any codec caches and discard * any pending samples and not yet returned encoded data. * @hard indicates whether a FLUSH is being processed, * or otherwise a DISCONT (or conceptually similar). * @event: Optional. * Event handler on the sink pad. This function should return * TRUE if the event was handled and should be discarded * (i.e. not unref'ed). * @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. * * Subclasses can override any of the available virtual methods or not, as * needed. At minimum @handle_frame (and likely @set_format) needs to be * overridden. */ struct _GstBaseAudioDecoderClass { GstElementClass parent_class; /*< public >*/ /* virtual methods for subclasses */ gboolean (*start) (GstBaseAudioDecoder *dec); gboolean (*stop) (GstBaseAudioDecoder *dec); gboolean (*set_format) (GstBaseAudioDecoder *dec, GstCaps *caps); GstFlowReturn (*parse) (GstBaseAudioDecoder *dec, GstAdapter *adapter, gint *offset, gint *length); GstFlowReturn (*handle_frame) (GstBaseAudioDecoder *dec, GstBuffer *buffer); void (*flush) (GstBaseAudioDecoder *dec, gboolean hard); GstFlowReturn (*pre_push) (GstBaseAudioDecoder *dec, GstBuffer **buffer); gboolean (*event) (GstBaseAudioDecoder *dec, GstEvent *event); /*< private >*/ gpointer _gst_reserved[GST_PADDING_LARGE]; }; GstFlowReturn gst_base_audio_decoder_finish_frame (GstBaseAudioDecoder * dec, GstBuffer * buf, gint frames); GType gst_base_audio_decoder_get_type (void); G_END_DECLS #endif