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d42390efd9
This extends the special case of a fixed number of samples per frame that was supported before already.
234 lines
8.5 KiB
C
234 lines
8.5 KiB
C
/* GStreamer
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* Copyright (C) 2011 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>.
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* Copyright (C) 2011 Nokia Corporation. All rights reserved.
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* Contact: Stefan Kost <stefan.kost@nokia.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#ifndef __GST_BASE_AUDIO_ENCODER_H__
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#define __GST_BASE_AUDIO_ENCODER_H__
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#ifndef GST_USE_UNSTABLE_API
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#warning "GstBaseAudioEncoder is unstable API and may change in future."
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#warning "You can define GST_USE_UNSTABLE_API to avoid this warning."
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#endif
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#include <gst/gst.h>
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#include "gstbaseaudioutils.h"
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G_BEGIN_DECLS
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#define GST_TYPE_BASE_AUDIO_ENCODER (gst_base_audio_encoder_get_type())
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#define GST_BASE_AUDIO_ENCODER(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_BASE_AUDIO_ENCODER,GstBaseAudioEncoder))
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#define GST_BASE_AUDIO_ENCODER_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_BASE_AUDIO_ENCODER,GstBaseAudioEncoderClass))
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#define GST_BASE_AUDIO_ENCODER_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj),GST_TYPE_BASE_AUDIO_ENCODER,GstBaseAudioEncoderClass))
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#define GST_IS_BASE_AUDIO_ENCODER(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_BASE_AUDIO_ENCODER))
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#define GST_IS_BASE_AUDIO_ENCODER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_BASE_AUDIO_ENCODER))
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#define GST_BASE_AUDIO_ENCODER_CAST(obj) ((GstBaseAudioEncoder *)(obj))
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/**
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* GST_BASE_AUDIO_ENCODER_SINK_NAME:
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*
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* the name of the templates for the sink pad
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*/
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#define GST_BASE_AUDIO_ENCODER_SINK_NAME "sink"
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/**
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* GST_BASE_AUDIO_ENCODER_SRC_NAME:
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*
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* the name of the templates for the source pad
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*/
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#define GST_BASE_AUDIO_ENCODER_SRC_NAME "src"
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/**
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* GST_BASE_AUDIO_ENCODER_SRC_PAD:
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* @obj: base parse instance
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*
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* Gives the pointer to the source #GstPad object of the element.
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*
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* Since: 0.10.x
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*/
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#define GST_BASE_AUDIO_ENCODER_SRC_PAD(obj) (GST_BASE_AUDIO_ENCODER_CAST (obj)->srcpad)
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/**
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* GST_BASE_AUDIO_ENCODER_SINK_PAD:
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* @obj: base parse instance
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*
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* Gives the pointer to the sink #GstPad object of the element.
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*
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* Since: 0.10.x
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*/
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#define GST_BASE_AUDIO_ENCODER_SINK_PAD(obj) (GST_BASE_AUDIO_ENCODER_CAST (obj)->sinkpad)
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/**
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* GST_BASE_AUDIO_ENCODER_SEGMENT:
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* @obj: base parse instance
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*
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* Gives the segment of the element.
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*
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* Since: 0.10.x
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*/
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#define GST_BASE_AUDIO_ENCODER_SEGMENT(obj) (GST_BASE_AUDIO_ENCODER_CAST (obj)->segment)
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#define GST_BASE_AUDIO_ENCODER_STREAM_LOCK(enc) g_static_rec_mutex_lock (&GST_BASE_AUDIO_ENCODER (enc)->stream_lock)
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#define GST_BASE_AUDIO_ENCODER_STREAM_UNLOCK(enc) g_static_rec_mutex_unlock (&GST_BASE_AUDIO_ENCODER (enc)->stream_lock)
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typedef struct _GstBaseAudioEncoder GstBaseAudioEncoder;
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typedef struct _GstBaseAudioEncoderClass GstBaseAudioEncoderClass;
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typedef struct _GstBaseAudioEncoderPrivate GstBaseAudioEncoderPrivate;
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typedef struct _GstBaseAudioEncoderContext GstBaseAudioEncoderContext;
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/**
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* GstBaseAudioEncoderContext:
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* @state: a #GstAudioState describing input audio format
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* @frame_samples_min: number of samples (per channel) subclass needs to be handed
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* at least, or will be handed all available if 0.
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* @frame_samples_max: number of samples (per channel) subclass needs to be handed
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* at most, or will be handed all available if 0.
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* @frame_max: max number of frames of size @frame_samples accepted at once
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* (assumed minimally 1). Requires @frame_samples_min and @frame_samples_max
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* to be the equal.
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* @min_latency: min latency of element
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* @max_latency: max latency of element
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* @lookahead: encoder lookahead (in units of input rate samples)
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*
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* Transparent #GstBaseAudioEncoderContext data structure.
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*/
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struct _GstBaseAudioEncoderContext {
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/* input */
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GstAudioState state;
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/* output */
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gint frame_samples_min, frame_samples_max;
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gint frame_max;
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gint lookahead;
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/* MT-protected (with LOCK) */
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GstClockTime min_latency;
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GstClockTime max_latency;
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};
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/**
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* GstBaseAudioEncoder:
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* @element: the parent element.
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*
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* The opaque #GstBaseAudioEncoder data structure.
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*/
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struct _GstBaseAudioEncoder {
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GstElement element;
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/*< protected >*/
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/* source and sink pads */
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GstPad *sinkpad;
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GstPad *srcpad;
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/* protects all data processing, i.e. is locked
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* in the chain function, finish_frame and when
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* processing serialized events */
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GStaticRecMutex stream_lock;
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/* MT-protected (with STREAM_LOCK) */
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GstSegment segment;
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GstBaseAudioEncoderContext *ctx;
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/* properties */
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gint64 tolerance;
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gboolean perfect_ts;
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gboolean hard_resync;
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gboolean granule;
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/*< private >*/
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GstBaseAudioEncoderPrivate *priv;
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gpointer _gst_reserved[GST_PADDING_LARGE];
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};
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/**
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* GstBaseAudioEncoderClass:
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* @start: Optional.
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* Called when the element starts processing.
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* Allows opening external resources.
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* @stop: Optional.
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* Called when the element stops processing.
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* Allows closing external resources.
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* @set_format: Notifies subclass of incoming data format.
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* GstBaseAudioEncoderContext fields have already been
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* set according to provided caps.
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* @handle_frame: Provides input samples (or NULL to clear any remaining data)
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* according to directions as provided by subclass in the
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* #GstBaseAudioEncoderContext. Input data ref management
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* is performed by base class, subclass should not care or
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* intervene.
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* @flush: Optional.
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* Instructs subclass to clear any codec caches and discard
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* any pending samples and not yet returned encoded data.
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* @event: Optional.
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* Event handler on the sink pad. This function should return
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* TRUE if the event was handled and should be discarded
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* (i.e. not unref'ed).
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* @pre_push: Optional.
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* Called just prior to pushing (encoded data) buffer downstream.
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* Subclass has full discretionary access to buffer,
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* and a not OK flow return will abort downstream pushing.
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* @getcaps: Optional.
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* Allows for a custom sink getcaps implementation (e.g.
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* for multichannel input specification). If not implemented,
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* default returns gst_base_audio_encoder_proxy_getcaps
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* applied to sink template caps.
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*
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* Subclasses can override any of the available virtual methods or not, as
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* needed. At minimum @set_format and @handle_frame needs to be overridden.
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*/
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struct _GstBaseAudioEncoderClass {
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GstElementClass parent_class;
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/*< public >*/
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/* virtual methods for subclasses */
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gboolean (*start) (GstBaseAudioEncoder *enc);
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gboolean (*stop) (GstBaseAudioEncoder *enc);
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gboolean (*set_format) (GstBaseAudioEncoder *enc,
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GstAudioState *state);
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GstFlowReturn (*handle_frame) (GstBaseAudioEncoder *enc,
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GstBuffer *buffer);
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void (*flush) (GstBaseAudioEncoder *enc);
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GstFlowReturn (*pre_push) (GstBaseAudioEncoder *enc,
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GstBuffer **buffer);
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gboolean (*event) (GstBaseAudioEncoder *enc,
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GstEvent *event);
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GstCaps * (*getcaps) (GstBaseAudioEncoder *enc);
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/*< private >*/
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gpointer _gst_reserved[GST_PADDING_LARGE];
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};
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GType gst_base_audio_encoder_get_type (void);
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GstFlowReturn gst_base_audio_encoder_finish_frame (GstBaseAudioEncoder * enc,
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GstBuffer *buffer, gint samples);
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GstCaps * gst_base_audio_encoder_proxy_getcaps (GstBaseAudioEncoder * enc,
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GstCaps * caps);
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G_END_DECLS
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#endif /* __GST_BASE_AUDIO_ENCODER_H__ */
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