| /* GStreamer |
| * 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_ENCODER_H__ |
| #define __GST_AUDIO_ENCODER_H__ |
| |
| #include <gst/gst.h> |
| |
| G_BEGIN_DECLS |
| |
| #define GST_TYPE_AUDIO_ENCODER (gst_audio_encoder_get_type()) |
| #define GST_AUDIO_ENCODER(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_ENCODER,GstAudioEncoder)) |
| #define GST_AUDIO_ENCODER_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_ENCODER,GstAudioEncoderClass)) |
| #define GST_AUDIO_ENCODER_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj),GST_TYPE_AUDIO_ENCODER,GstAudioEncoderClass)) |
| #define GST_IS_AUDIO_ENCODER(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_ENCODER)) |
| #define GST_IS_AUDIO_ENCODER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_ENCODER)) |
| #define GST_AUDIO_ENCODER_CAST(obj) ((GstAudioEncoder *)(obj)) |
| |
| /** |
| * GST_AUDIO_ENCODER_SINK_NAME: |
| * |
| * the name of the templates for the sink pad |
| */ |
| #define GST_AUDIO_ENCODER_SINK_NAME "sink" |
| /** |
| * GST_AUDIO_ENCODER_SRC_NAME: |
| * |
| * the name of the templates for the source pad |
| */ |
| #define GST_AUDIO_ENCODER_SRC_NAME "src" |
| |
| /** |
| * GST_AUDIO_ENCODER_SRC_PAD: |
| * @obj: audio encoder instance |
| * |
| * Gives the pointer to the source #GstPad object of the element. |
| */ |
| #define GST_AUDIO_ENCODER_SRC_PAD(obj) (GST_AUDIO_ENCODER_CAST (obj)->srcpad) |
| |
| /** |
| * GST_AUDIO_ENCODER_SINK_PAD: |
| * @obj: audio encoder instance |
| * |
| * Gives the pointer to the sink #GstPad object of the element. |
| */ |
| #define GST_AUDIO_ENCODER_SINK_PAD(obj) (GST_AUDIO_ENCODER_CAST (obj)->sinkpad) |
| |
| /** |
| * GST_AUDIO_ENCODER_INPUT_SEGMENT: |
| * @obj: base parse instance |
| * |
| * Gives the input segment of the element. |
| */ |
| #define GST_AUDIO_ENCODER_INPUT_SEGMENT(obj) (GST_AUDIO_ENCODER_CAST (obj)->input_segment) |
| |
| /** |
| * GST_AUDIO_ENCODER_OUTPUT_SEGMENT: |
| * @obj: base parse instance |
| * |
| * Gives the output segment of the element. |
| */ |
| #define GST_AUDIO_ENCODER_OUTPUT_SEGMENT(obj) (GST_AUDIO_ENCODER_CAST (obj)->output_segment) |
| |
| #define GST_AUDIO_ENCODER_STREAM_LOCK(enc) g_rec_mutex_lock (&GST_AUDIO_ENCODER (enc)->stream_lock) |
| #define GST_AUDIO_ENCODER_STREAM_UNLOCK(enc) g_rec_mutex_unlock (&GST_AUDIO_ENCODER (enc)->stream_lock) |
| |
| typedef struct _GstAudioEncoder GstAudioEncoder; |
| typedef struct _GstAudioEncoderClass GstAudioEncoderClass; |
| |
| typedef struct _GstAudioEncoderPrivate GstAudioEncoderPrivate; |
| |
| /** |
| * GstAudioEncoder: |
| * |
| * The opaque #GstAudioEncoder data structure. |
| */ |
| struct _GstAudioEncoder { |
| 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 >*/ |
| GstAudioEncoderPrivate *priv; |
| |
| gpointer _gst_reserved[GST_PADDING_LARGE]; |
| }; |
| |
| /** |
| * GstAudioEncoderClass: |
| * @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. |
| * GstAudioInfo contains the format according to provided caps. |
| * @handle_frame: Provides input samples (or NULL to clear any remaining data) |
| * according to directions as configured by the subclass |
| * using the API. 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_encoder_finish_frame(). |
| * @flush: Optional. |
| * Instructs subclass to clear any codec caches and discard |
| * any pending samples and not yet returned encoded data. |
| * @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. |
| * @getcaps: Optional. |
| * Allows for a custom sink getcaps implementation (e.g. |
| * for multichannel input specification). If not implemented, |
| * default returns gst_audio_encoder_proxy_getcaps |
| * applied to sink template caps. |
| * @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. |
| * @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 |
| * @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 |
| * |
| * Subclasses can override any of the available virtual methods or not, as |
| * needed. At minimum @set_format and @handle_frame needs to be overridden. |
| */ |
| struct _GstAudioEncoderClass { |
| GstElementClass element_class; |
| |
| /*< public >*/ |
| /* virtual methods for subclasses */ |
| |
| gboolean (*start) (GstAudioEncoder *enc); |
| |
| gboolean (*stop) (GstAudioEncoder *enc); |
| |
| gboolean (*set_format) (GstAudioEncoder *enc, |
| GstAudioInfo *info); |
| |
| GstFlowReturn (*handle_frame) (GstAudioEncoder *enc, |
| GstBuffer *buffer); |
| |
| void (*flush) (GstAudioEncoder *enc); |
| |
| GstFlowReturn (*pre_push) (GstAudioEncoder *enc, |
| GstBuffer **buffer); |
| |
| gboolean (*sink_event) (GstAudioEncoder *enc, |
| GstEvent *event); |
| |
| gboolean (*src_event) (GstAudioEncoder *enc, |
| GstEvent *event); |
| |
| GstCaps * (*getcaps) (GstAudioEncoder *enc, GstCaps *filter); |
| |
| gboolean (*open) (GstAudioEncoder *enc); |
| |
| gboolean (*close) (GstAudioEncoder *enc); |
| |
| gboolean (*negotiate) (GstAudioEncoder *enc); |
| |
| gboolean (*decide_allocation) (GstAudioEncoder *enc, GstQuery *query); |
| |
| gboolean (*propose_allocation) (GstAudioEncoder * enc, |
| GstQuery * query); |
| |
| gboolean (*transform_meta) (GstAudioEncoder *enc, GstBuffer *outbuf, |
| GstMeta *meta, GstBuffer *inbuf); |
| |
| gboolean (*sink_query) (GstAudioEncoder *encoder, |
| GstQuery *query); |
| |
| gboolean (*src_query) (GstAudioEncoder *encoder, |
| GstQuery *query); |
| |
| |
| /*< private >*/ |
| gpointer _gst_reserved[GST_PADDING_LARGE-3]; |
| }; |
| |
| GST_AUDIO_API |
| GType gst_audio_encoder_get_type (void); |
| |
| GST_AUDIO_API |
| GstFlowReturn gst_audio_encoder_finish_frame (GstAudioEncoder * enc, |
| GstBuffer * buffer, |
| gint samples); |
| |
| GST_AUDIO_API |
| GstCaps * gst_audio_encoder_proxy_getcaps (GstAudioEncoder * enc, |
| GstCaps * caps, |
| GstCaps * filter); |
| |
| GST_AUDIO_API |
| gboolean gst_audio_encoder_set_output_format (GstAudioEncoder * enc, |
| GstCaps * caps); |
| |
| GST_AUDIO_API |
| gboolean gst_audio_encoder_negotiate (GstAudioEncoder * enc); |
| |
| GST_AUDIO_API |
| GstBuffer * gst_audio_encoder_allocate_output_buffer (GstAudioEncoder * enc, |
| gsize size); |
| |
| /* context parameters */ |
| |
| GST_AUDIO_API |
| GstAudioInfo * gst_audio_encoder_get_audio_info (GstAudioEncoder * enc); |
| |
| GST_AUDIO_API |
| gint gst_audio_encoder_get_frame_samples_min (GstAudioEncoder * enc); |
| |
| GST_AUDIO_API |
| void gst_audio_encoder_set_frame_samples_min (GstAudioEncoder * enc, gint num); |
| |
| GST_AUDIO_API |
| gint gst_audio_encoder_get_frame_samples_max (GstAudioEncoder * enc); |
| |
| GST_AUDIO_API |
| void gst_audio_encoder_set_frame_samples_max (GstAudioEncoder * enc, gint num); |
| |
| GST_AUDIO_API |
| gint gst_audio_encoder_get_frame_max (GstAudioEncoder * enc); |
| |
| GST_AUDIO_API |
| void gst_audio_encoder_set_frame_max (GstAudioEncoder * enc, gint num); |
| |
| GST_AUDIO_API |
| gint gst_audio_encoder_get_lookahead (GstAudioEncoder * enc); |
| |
| GST_AUDIO_API |
| void gst_audio_encoder_set_lookahead (GstAudioEncoder * enc, gint num); |
| |
| GST_AUDIO_API |
| void gst_audio_encoder_get_latency (GstAudioEncoder * enc, |
| GstClockTime * min, |
| GstClockTime * max); |
| |
| GST_AUDIO_API |
| void gst_audio_encoder_set_latency (GstAudioEncoder * enc, |
| GstClockTime min, |
| GstClockTime max); |
| |
| GST_AUDIO_API |
| void gst_audio_encoder_set_headers (GstAudioEncoder * enc, |
| GList * headers); |
| |
| GST_AUDIO_API |
| void gst_audio_encoder_set_allocation_caps (GstAudioEncoder * enc, |
| GstCaps * allocation_caps); |
| |
| /* object properties */ |
| |
| GST_AUDIO_API |
| void gst_audio_encoder_set_mark_granule (GstAudioEncoder * enc, |
| gboolean enabled); |
| |
| GST_AUDIO_API |
| gboolean gst_audio_encoder_get_mark_granule (GstAudioEncoder * enc); |
| |
| GST_AUDIO_API |
| void gst_audio_encoder_set_perfect_timestamp (GstAudioEncoder * enc, |
| gboolean enabled); |
| |
| GST_AUDIO_API |
| gboolean gst_audio_encoder_get_perfect_timestamp (GstAudioEncoder * enc); |
| |
| GST_AUDIO_API |
| void gst_audio_encoder_set_hard_resync (GstAudioEncoder * enc, |
| gboolean enabled); |
| |
| GST_AUDIO_API |
| gboolean gst_audio_encoder_get_hard_resync (GstAudioEncoder * enc); |
| |
| GST_AUDIO_API |
| void gst_audio_encoder_set_tolerance (GstAudioEncoder * enc, |
| GstClockTime tolerance); |
| |
| GST_AUDIO_API |
| GstClockTime gst_audio_encoder_get_tolerance (GstAudioEncoder * enc); |
| |
| GST_AUDIO_API |
| void gst_audio_encoder_set_hard_min (GstAudioEncoder * enc, |
| gboolean enabled); |
| |
| GST_AUDIO_API |
| gboolean gst_audio_encoder_get_hard_min (GstAudioEncoder * enc); |
| |
| GST_AUDIO_API |
| void gst_audio_encoder_set_drainable (GstAudioEncoder * enc, |
| gboolean enabled); |
| |
| GST_AUDIO_API |
| gboolean gst_audio_encoder_get_drainable (GstAudioEncoder * enc); |
| |
| GST_AUDIO_API |
| void gst_audio_encoder_get_allocator (GstAudioEncoder * enc, |
| GstAllocator ** allocator, |
| GstAllocationParams * params); |
| |
| GST_AUDIO_API |
| void gst_audio_encoder_merge_tags (GstAudioEncoder * enc, |
| const GstTagList * tags, GstTagMergeMode mode); |
| |
| #ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC |
| G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstAudioEncoder, gst_object_unref) |
| #endif |
| |
| G_END_DECLS |
| |
| #endif /* __GST_AUDIO_ENCODER_H__ */ |
