| /* GStreamer |
| * Copyright (C) 2009 Igalia S.L. |
| * Author: Iago Toral Quiroga <itoral@igalia.com> |
| * Copyright (C) 2011 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>. |
| * Copyright (C) 2011 Nokia Corporation. All rights reserved. |
| * Contact: Stefan Kost <stefan.kost@nokia.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., 51 Franklin St, Fifth Floor, |
| * Boston, MA 02110-1301, USA. |
| */ |
| |
| #ifndef __GST_AUDIO_AUDIO_H__ |
| #include <gst/audio/audio.h> |
| #endif |
| |
| #ifndef _GST_AUDIO_DECODER_H_ |
| #define _GST_AUDIO_DECODER_H_ |
| |
| #include <gst/gst.h> |
| #include <gst/base/gstadapter.h> |
| |
| 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); |
| |
| 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 |
| 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_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); |
| |
| #ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC |
| G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstAudioDecoder, gst_object_unref) |
| #endif |
| |
| G_END_DECLS |
| |
| #endif /* _GST_AUDIO_DECODER_H_ */ |
