/* 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., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, USA. */ #ifndef __GST_AUDIO_AUDIO_H__ #include #endif #ifndef _GST_AUDIO_DECODER_H_ #define _GST_AUDIO_DECODER_H_ #include #include G_BEGIN_DECLS #define GST_TYPE_AUDIO_DECODER \ (gst_audio_decoder_get_type()) #define GST_AUDIO_DECODER(obj) \ (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_DECODER,GstAudioDecoder)) #define GST_AUDIO_DECODER_CLASS(klass) \ (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_DECODER,GstAudioDecoderClass)) #define GST_AUDIO_DECODER_GET_CLASS(obj) \ (G_TYPE_INSTANCE_GET_CLASS((obj),GST_TYPE_AUDIO_DECODER,GstAudioDecoderClass)) #define GST_IS_AUDIO_DECODER(obj) \ (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_DECODER)) #define GST_IS_AUDIO_DECODER_CLASS(obj) \ (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_DECODER)) #define GST_AUDIO_DECODER_CAST(obj) \ ((GstAudioDecoder *)(obj)) /** * GST_AUDIO_DECODER_SINK_NAME: * * The name of the templates for the sink pad. */ #define GST_AUDIO_DECODER_SINK_NAME "sink" /** * GST_AUDIO_DECODER_SRC_NAME: * * The name of the templates for the source pad. */ #define GST_AUDIO_DECODER_SRC_NAME "src" /** * GST_AUDIO_DECODER_SRC_PAD: * @obj: base audio codec instance * * Gives the pointer to the source #GstPad object of the element. */ #define GST_AUDIO_DECODER_SRC_PAD(obj) (((GstAudioDecoder *) (obj))->srcpad) /** * GST_AUDIO_DECODER_SINK_PAD: * @obj: base audio codec instance * * Gives the pointer to the sink #GstPad object of the element. */ #define GST_AUDIO_DECODER_SINK_PAD(obj) (((GstAudioDecoder *) (obj))->sinkpad) #define GST_AUDIO_DECODER_STREAM_LOCK(dec) g_rec_mutex_lock (&GST_AUDIO_DECODER (dec)->stream_lock) #define GST_AUDIO_DECODER_STREAM_UNLOCK(dec) g_rec_mutex_unlock (&GST_AUDIO_DECODER (dec)->stream_lock) /** * GST_AUDIO_DECODER_INPUT_SEGMENT: * @obj: audio decoder instance * * Gives the input segment of the element. */ #define GST_AUDIO_DECODER_INPUT_SEGMENT(obj) (GST_AUDIO_DECODER_CAST (obj)->input_segment) /** * GST_AUDIO_DECODER_OUTPUT_SEGMENT: * @obj: audio decoder instance * * Gives the output segment of the element. */ #define GST_AUDIO_DECODER_OUTPUT_SEGMENT(obj) (GST_AUDIO_DECODER_CAST (obj)->output_segment) typedef struct _GstAudioDecoder GstAudioDecoder; typedef struct _GstAudioDecoderClass GstAudioDecoderClass; typedef struct _GstAudioDecoderPrivate GstAudioDecoderPrivate; /* do not use this one, use macro below */ GST_AUDIO_API GstFlowReturn _gst_audio_decoder_error (GstAudioDecoder *dec, gint weight, GQuark domain, gint code, gchar *txt, gchar *debug, const gchar *file, const gchar *function, gint line); /** * GST_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_AUDIO_DECODER_ERROR(el, weight, domain, code, text, debug, ret) \ G_STMT_START { \ gchar *__txt = _gst_element_error_printf text; \ gchar *__dbg = _gst_element_error_printf debug; \ GstAudioDecoder *__dec = GST_AUDIO_DECODER (el); \ ret = _gst_audio_decoder_error (__dec, weight, GST_ ## domain ## _ERROR, \ GST_ ## domain ## _ERROR_ ## code, __txt, __dbg, __FILE__, \ GST_FUNCTION, __LINE__); \ } G_STMT_END /** * GST_AUDIO_DECODER_MAX_ERRORS: * * Default maximum number of errors tolerated before signaling error. */ #define GST_AUDIO_DECODER_MAX_ERRORS 10 /** * GstAudioDecoder: * * The opaque #GstAudioDecoder data structure. */ struct _GstAudioDecoder { GstElement element; /*< protected >*/ /* source and sink pads */ GstPad *sinkpad; GstPad *srcpad; /* protects all data processing, i.e. is locked * in the chain function, finish_frame and when * processing serialized events */ GRecMutex stream_lock; /* MT-protected (with STREAM_LOCK) */ GstSegment input_segment; GstSegment output_segment; /*< private >*/ GstAudioDecoderPrivate *priv; gpointer _gst_reserved[GST_PADDING_LARGE]; }; /** * GstAudioDecoderClass: * @element_class: The parent class structure * @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, * and input data is only valid until next call to base class, * most notably a call to gst_audio_decoder_finish_frame(). * @flush: Optional. * Instructs subclass to clear any codec caches and discard * any pending samples and not yet returned decoded data. * @hard indicates whether a FLUSH is being processed, * or otherwise a DISCONT (or conceptually similar). * @sink_event: Optional. * Event handler on the sink pad. Subclasses should chain up to * the parent implementation to invoke the default handler. * @src_event: Optional. * Event handler on the src pad. Subclasses should chain up to * the parent implementation to invoke the default handler. * @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. * @open: Optional. * Called when the element changes to GST_STATE_READY. * Allows opening external resources. * @close: Optional. * Called when the element changes to GST_STATE_NULL. * Allows closing external resources. * @negotiate: Optional. * Negotiate with downstream and configure buffer pools, etc. * Subclasses should chain up to the parent implementation to * invoke the default handler. * @decide_allocation: Optional. * Setup the allocation parameters for allocating output * buffers. The passed in query contains the result of the * downstream allocation query. * Subclasses should chain up to the parent implementation to * invoke the default handler. * @propose_allocation: Optional. * Propose buffer allocation parameters for upstream elements. * Subclasses should chain up to the parent implementation to * invoke the default handler. * @sink_query: Optional. * Query handler on the sink pad. This function should * return TRUE if the query could be performed. Subclasses * should chain up to the parent implementation to invoke the * default handler. Since: 1.6 * @src_query: Optional. * Query handler on the source pad. This function should * return TRUE if the query could be performed. Subclasses * should chain up to the parent implementation to invoke the * default handler. Since: 1.6 * @getcaps: Optional. * Allows for a custom sink getcaps implementation. * If not implemented, * default returns gst_audio_decoder_proxy_getcaps * applied to sink template caps. * @transform_meta: Optional. Transform the metadata on the input buffer to the * output buffer. By default this method copies all meta without * tags and meta with only the "audio" tag. subclasses can * implement this method and return %TRUE if the metadata is to be * copied. Since: 1.6 * * 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 _GstAudioDecoderClass { GstElementClass element_class; /*< public >*/ /* virtual methods for subclasses */ gboolean (*start) (GstAudioDecoder *dec); gboolean (*stop) (GstAudioDecoder *dec); gboolean (*set_format) (GstAudioDecoder *dec, GstCaps *caps); /** * GstAudioDecoderClass::parse: * @offset: (out): * @length: (out): */ GstFlowReturn (*parse) (GstAudioDecoder *dec, GstAdapter *adapter, gint *offset, gint *length); GstFlowReturn (*handle_frame) (GstAudioDecoder *dec, GstBuffer *buffer); void (*flush) (GstAudioDecoder *dec, gboolean hard); GstFlowReturn (*pre_push) (GstAudioDecoder *dec, GstBuffer **buffer); gboolean (*sink_event) (GstAudioDecoder *dec, GstEvent *event); gboolean (*src_event) (GstAudioDecoder *dec, GstEvent *event); gboolean (*open) (GstAudioDecoder *dec); gboolean (*close) (GstAudioDecoder *dec); gboolean (*negotiate) (GstAudioDecoder *dec); gboolean (*decide_allocation) (GstAudioDecoder *dec, GstQuery *query); gboolean (*propose_allocation) (GstAudioDecoder *dec, GstQuery * query); gboolean (*sink_query) (GstAudioDecoder *dec, GstQuery *query); gboolean (*src_query) (GstAudioDecoder *dec, GstQuery *query); GstCaps * (*getcaps) (GstAudioDecoder * dec, GstCaps * filter); gboolean (*transform_meta) (GstAudioDecoder *enc, GstBuffer *outbuf, GstMeta *meta, GstBuffer *inbuf); /*< private >*/ gpointer _gst_reserved[GST_PADDING_LARGE - 4]; }; GST_AUDIO_API GType gst_audio_decoder_get_type (void); GST_AUDIO_API gboolean gst_audio_decoder_set_output_format (GstAudioDecoder * dec, const GstAudioInfo * info); GST_AUDIO_API gboolean gst_audio_decoder_set_output_caps (GstAudioDecoder * dec, GstCaps * caps); GST_AUDIO_API GstCaps * gst_audio_decoder_proxy_getcaps (GstAudioDecoder * decoder, GstCaps * caps, GstCaps * filter); GST_AUDIO_API gboolean gst_audio_decoder_negotiate (GstAudioDecoder * dec); GST_AUDIO_API GstFlowReturn gst_audio_decoder_finish_subframe (GstAudioDecoder * dec, GstBuffer * buf); GST_AUDIO_API GstFlowReturn gst_audio_decoder_finish_frame (GstAudioDecoder * dec, GstBuffer * buf, gint frames); GST_AUDIO_API GstBuffer * gst_audio_decoder_allocate_output_buffer (GstAudioDecoder * dec, gsize size); /* context parameters */ GST_AUDIO_API GstAudioInfo * gst_audio_decoder_get_audio_info (GstAudioDecoder * dec); GST_AUDIO_API void gst_audio_decoder_set_plc_aware (GstAudioDecoder * dec, gboolean plc); GST_AUDIO_API gint gst_audio_decoder_get_plc_aware (GstAudioDecoder * dec); GST_AUDIO_API void gst_audio_decoder_set_estimate_rate (GstAudioDecoder * dec, gboolean enabled); GST_AUDIO_API gint gst_audio_decoder_get_estimate_rate (GstAudioDecoder * dec); GST_AUDIO_API gint gst_audio_decoder_get_delay (GstAudioDecoder * dec); GST_AUDIO_API void gst_audio_decoder_set_max_errors (GstAudioDecoder * dec, gint num); GST_AUDIO_API gint gst_audio_decoder_get_max_errors (GstAudioDecoder * dec); GST_AUDIO_API void gst_audio_decoder_set_latency (GstAudioDecoder * dec, GstClockTime min, GstClockTime max); GST_AUDIO_API void gst_audio_decoder_get_latency (GstAudioDecoder * dec, GstClockTime * min, GstClockTime * max); GST_AUDIO_API void gst_audio_decoder_get_parse_state (GstAudioDecoder * dec, gboolean * sync, gboolean * eos); GST_AUDIO_API void gst_audio_decoder_set_allocation_caps (GstAudioDecoder * dec, GstCaps * allocation_caps); /* object properties */ GST_AUDIO_API void gst_audio_decoder_set_plc (GstAudioDecoder * dec, gboolean enabled); GST_AUDIO_API gboolean gst_audio_decoder_get_plc (GstAudioDecoder * dec); GST_AUDIO_API void gst_audio_decoder_set_min_latency (GstAudioDecoder * dec, GstClockTime num); GST_AUDIO_API GstClockTime gst_audio_decoder_get_min_latency (GstAudioDecoder * dec); GST_AUDIO_API void gst_audio_decoder_set_tolerance (GstAudioDecoder * dec, GstClockTime tolerance); GST_AUDIO_API GstClockTime gst_audio_decoder_get_tolerance (GstAudioDecoder * dec); GST_AUDIO_API void gst_audio_decoder_set_drainable (GstAudioDecoder * dec, gboolean enabled); GST_AUDIO_API gboolean gst_audio_decoder_get_drainable (GstAudioDecoder * dec); GST_AUDIO_API void gst_audio_decoder_set_needs_format (GstAudioDecoder * dec, gboolean enabled); GST_AUDIO_API gboolean gst_audio_decoder_get_needs_format (GstAudioDecoder * dec); GST_AUDIO_API void gst_audio_decoder_get_allocator (GstAudioDecoder * dec, GstAllocator ** allocator, GstAllocationParams * params); GST_AUDIO_API void gst_audio_decoder_merge_tags (GstAudioDecoder * dec, const GstTagList * tags, GstTagMergeMode mode); GST_AUDIO_API void gst_audio_decoder_set_use_default_pad_acceptcaps (GstAudioDecoder * decoder, gboolean use); G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstAudioDecoder, gst_object_unref) G_END_DECLS #endif /* _GST_AUDIO_DECODER_H_ */