mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-07-10 08:35:58 +00:00
Code Issues 1 Projects Releases Wiki Activity

audioencoders: baseaudioencoder and ported encoders

Browse source
This commit is contained in:
Mark Nauwelaerts 2011-01-27 16:52:50 +01:00 committed by Tim-Philipp Müller
parent 2b495769dc
commit 80241fde8d
2 changed files with 1844 additions and 0 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

1593
gst-libs/gst/audio/gstbaseaudioencoder.c Normal file
View file

File diff suppressed because it is too large Load diff

251
gst-libs/gst/audio/gstbaseaudioencoder.h Normal file
View file

@ -0,0 +1,251 @@
/* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifndef __GST_BASE_AUDIO_ENCODER_H__
#define __GST_BASE_AUDIO_ENCODER_H__
#include <gst/gst.h>
#include <gst/base/gstadapter.h>
#include <gst/audio/multichannel.h>
G_BEGIN_DECLS
#define GST_TYPE_BASE_AUDIO_ENCODER (gst_base_audio_encoder_get_type())
#define GST_BASE_AUDIO_ENCODER(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_BASE_AUDIO_ENCODER,GstBaseAudioEncoder))
#define GST_BASE_AUDIO_ENCODER_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_BASE_AUDIO_ENCODER,GstBaseAudioEncoderClass))
#define GST_BASE_AUDIO_ENCODER_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj),GST_TYPE_BASE_AUDIO_ENCODER,GstBaseAudioEncoderClass))
#define GST_IS_BASE_AUDIO_ENCODER(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_BASE_AUDIO_ENCODER))
#define GST_IS_BASE_AUDIO_ENCODER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_BASE_AUDIO_ENCODER))
#define GST_BASE_AUDIO_ENCODER_CAST(obj) ((GstBaseAudioEncoder *)(obj))
/**
* GST_BASE_AUDIO_ENCODER_SINK_NAME:
*
* the name of the templates for the sink pad
*/
#define GST_BASE_AUDIO_ENCODER_SINK_NAME "sink"
/**
* GST_BASE_AUDIO_ENCODER_SRC_NAME:
*
* the name of the templates for the source pad
*/
#define GST_BASE_AUDIO_ENCODER_SRC_NAME "src"
/**
* GST_BASE_AUDIO_ENCODER_SRC_PAD:
* @obj: base parse instance
*
* Gives the pointer to the source #GstPad object of the element.
*
* Since: 0.10.x
*/
#define GST_BASE_AUDIO_ENCODER_SRC_PAD(obj) (GST_BASE_AUDIO_ENCODER_CAST (obj)->srcpad)
/**
* GST_BASE_AUDIO_ENCODER_SINK_PAD:
* @obj: base parse instance
*
* Gives the pointer to the sink #GstPad object of the element.
*
* Since: 0.10.x
*/
#define GST_BASE_AUDIO_ENCODER_SINK_PAD(obj) (GST_BASE_AUDIO_ENCODER_CAST (obj)->sinkpad)
/**
* GST_BASE_AUDIO_ENCODER_SEGMENT:
* @obj: base parse instance
*
* Gives the segment of the element.
*
* Since: 0.10.x
*/
#define GST_BASE_AUDIO_ENCODER_SEGMENT(obj) (GST_BASE_AUDIO_ENCODER_CAST (obj)->segment)
/**
* GST_BASE_AUDIO_ENCODER_FLOW_DROPPED:
*
* A #GstFlowReturn that can be returned from parse_frame to
* indicate that no output buffer was generated, or from pre_push_buffer to
* to forego pushing buffer.
*
* Since: 0.10.x
*/
#define GST_BASE_AUDIO_ENCODER_FLOW_DROPPED GST_FLOW_CUSTOM_SUCCESS
typedef struct _GstBaseAudioEncoder GstBaseAudioEncoder;
typedef struct _GstBaseAudioEncoderClass GstBaseAudioEncoderClass;
typedef struct _GstBaseAudioEncoderPrivate GstBaseAudioEncoderPrivate;
typedef struct _GstBaseAudioEncoderContext GstBaseAudioEncoderContext;
/**
* GstAudioState:
* @xint: whether sample data is int or float
* @rate: rate of sample data
* @channels: number of channels in sample data
* @width: width (in bits) of sample data
* @depth: used bits in sample data (if integer)
* @sign: rate of sample data (if integer)
* @endian: endianness of sample data
* @bpf: bytes per audio frame
*/
typedef struct _GstAudioState {
gboolean xint;
gint rate;
gint channels;
gint width;
gint depth;
gboolean sign;
gint endian;
GstAudioChannelPosition *channel_pos;
gint bpf;
} GstAudioState;
/**
* GstBaseAudioEncoderContext:
* @state: a #GstAudioState describing input audio format
* @frame_samples: number of samples (per channel) subclass needs to be handed,
* or will be handed all available if 0.
* @frame_max: max number of frames of size @frame_bytes accepted at once
* (assumed minimally 1)
* @latency: latency of element; should only be changed during configure
* @lookahead: encoder lookahead (in units of input rate samples)
*
* Transparent #GstBaseAudioEncoderContext data structure.
*/
struct _GstBaseAudioEncoderContext {
/* input */
GstAudioState state;
/* output */
gint frame_samples;
gint frame_max;
GstClockTime min_latency;
GstClockTime max_latency;
gint lookahead;
};
/**
* GstBaseAudioEncoder:
* @element: the parent element.
*
* The opaque #GstBaseAudioEncoder data structure.
*/
struct _GstBaseAudioEncoder {
GstElement element;
/*< protected >*/
/* source and sink pads */
GstPad *sinkpad;
GstPad *srcpad;
/* MT-protected (with STREAM_LOCK) */
GstSegment segment;
GstBaseAudioEncoderContext *ctx;
/* properties */
gint64 tolerance;
gboolean perfect_ts;
gboolean hard_resync;
gboolean granule;
/*< private >*/
GstBaseAudioEncoderPrivate *priv;
gpointer _gst_reserved[GST_PADDING_LARGE];
};
/**
* GstBaseAudioEncoderClass:
* @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.
* GstBaseAudioEncoderContext fields have already been
* set according to provided caps.
* @handle_frame: Provides input samples (or NULL to clear any remaining data)
* according to directions as provided by subclass in the
* #GstBaseAudioEncoderContext. 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.
* @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.
* @getcaps: Optional.
* Allows for a custom sink getcaps implementation (e.g.
* for multichannel input specification). If not implemented,
* default returns gst_base_audio_encoder_proxy_getcaps
* applied to sink template caps.
*
* 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 _GstBaseAudioEncoderClass {
GstElementClass parent_class;
/*< public >*/
/* virtual methods for subclasses */
gboolean (*start) (GstBaseAudioEncoder *enc);
gboolean (*stop) (GstBaseAudioEncoder *enc);
gboolean (*set_format) (GstBaseAudioEncoder *enc,
GstAudioState *state);
GstFlowReturn (*handle_frame) (GstBaseAudioEncoder *enc,
GstBuffer *buffer);
void (*flush) (GstBaseAudioEncoder *enc);
GstFlowReturn (*pre_push) (GstBaseAudioEncoder *enc,
GstBuffer *buffer);
gboolean (*event) (GstBaseAudioEncoder *enc,
GstEvent *event);
GstCaps * (*getcaps) (GstBaseAudioEncoder *enc);
/*< private >*/
gpointer _gst_reserved[GST_PADDING_LARGE];
};
GType gst_base_audio_encoder_get_type (void);
GstFlowReturn gst_base_audio_encoder_finish_frame (GstBaseAudioEncoder * enc,
GstBuffer *buffer, gint samples);
GstCaps * gst_base_audio_encoder_proxy_getcaps (GstBaseAudioEncoder * enc,
GstCaps * caps);
G_END_DECLS
#endif /* __GST_BASE_AUDIO_ENCODER_H__ */