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gstreamer/subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h

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/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2005 Wim Taymans <wim@fluendo.com>
*
* gstaudioringbuffer.h:
*
* 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_RING_BUFFER_H__
#define __GST_AUDIO_RING_BUFFER_H__
G_BEGIN_DECLS
#define GST_TYPE_AUDIO_RING_BUFFER (gst_audio_ring_buffer_get_type())
#define GST_AUDIO_RING_BUFFER(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_RING_BUFFER,GstAudioRingBuffer))
#define GST_AUDIO_RING_BUFFER_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_RING_BUFFER,GstAudioRingBufferClass))
#define GST_AUDIO_RING_BUFFER_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_AUDIO_RING_BUFFER, GstAudioRingBufferClass))
#define GST_AUDIO_RING_BUFFER_CAST(obj) ((GstAudioRingBuffer *)obj)
#define GST_IS_AUDIO_RING_BUFFER(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_RING_BUFFER))
#define GST_IS_AUDIO_RING_BUFFER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_RING_BUFFER))
typedef struct _GstAudioRingBuffer GstAudioRingBuffer;
typedef struct _GstAudioRingBufferClass GstAudioRingBufferClass;
typedef struct _GstAudioRingBufferSpec GstAudioRingBufferSpec;
/**
* GstAudioRingBufferCallback:
* @rbuf: a #GstAudioRingBuffer
* @data: (array length=len): target to fill
* @len: amount to fill
* @user_data: user data
*
* This function is set with gst_audio_ring_buffer_set_callback() and is
* called to fill the memory at @data with @len bytes of samples.
*/
typedef void (*GstAudioRingBufferCallback) (GstAudioRingBuffer *rbuf, guint8* data, guint len, gpointer user_data);
/**
* GstAudioRingBufferState:
* @GST_AUDIO_RING_BUFFER_STATE_STOPPED: The ringbuffer is stopped
* @GST_AUDIO_RING_BUFFER_STATE_PAUSED: The ringbuffer is paused
* @GST_AUDIO_RING_BUFFER_STATE_STARTED: The ringbuffer is started
* @GST_AUDIO_RING_BUFFER_STATE_ERROR: The ringbuffer has encountered an
* error after it has been started, e.g. because the device was
* disconnected (Since: 1.2)
*
* The state of the ringbuffer.
*/
typedef enum {
GST_AUDIO_RING_BUFFER_STATE_STOPPED,
GST_AUDIO_RING_BUFFER_STATE_PAUSED,
GST_AUDIO_RING_BUFFER_STATE_STARTED,
GST_AUDIO_RING_BUFFER_STATE_ERROR
} GstAudioRingBufferState;
/**
* GstAudioRingBufferFormatType:
* @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW: samples in linear or float
* @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MU_LAW: samples in mulaw
* @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_A_LAW: samples in alaw
* @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IMA_ADPCM: samples in ima adpcm
* @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG: samples in mpeg audio (but not AAC) format
* @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_GSM: samples in gsm format
* @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IEC958: samples in IEC958 frames (e.g. AC3)
* @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3: samples in AC3 format
* @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_EAC3: samples in EAC3 format
* @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DTS: samples in DTS format
* @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC: samples in MPEG-2 AAC ADTS format
* @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC: samples in MPEG-4 AAC ADTS format
* @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC_RAW: samples in MPEG-2 AAC raw format (Since: 1.12)
* @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC_RAW: samples in MPEG-4 AAC raw format (Since: 1.12)
* @GST_AUDIO_RING_BUFFER_FORMAT_TYPE_FLAC: samples in FLAC format (Since: 1.12)
*
* The format of the samples in the ringbuffer.
*/
typedef enum
{
GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW,
GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MU_LAW,
GST_AUDIO_RING_BUFFER_FORMAT_TYPE_A_LAW,
GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IMA_ADPCM,
GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG,
GST_AUDIO_RING_BUFFER_FORMAT_TYPE_GSM,
GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IEC958,
GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3,
GST_AUDIO_RING_BUFFER_FORMAT_TYPE_EAC3,
GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DTS,
GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC,
GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC,
GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC_RAW,
GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC_RAW,
GST_AUDIO_RING_BUFFER_FORMAT_TYPE_FLAC
} GstAudioRingBufferFormatType;
/**
* GstAudioRingBufferSpec:
* @caps: The caps that generated the Spec.
* @type: the sample type
* @info: the #GstAudioInfo
* @latency_time: the latency in microseconds
* @buffer_time: the total buffer size in microseconds
* @segsize: the size of one segment in bytes
* @segtotal: the total number of segments
* @seglatency: number of segments queued in the lower level device,
* defaults to segtotal
*
* The structure containing the format specification of the ringbuffer.
*/
struct _GstAudioRingBufferSpec
{
/*< public >*/
/* in */
GstCaps *caps; /* the caps of the buffer */
/* in/out */
GstAudioRingBufferFormatType type;
GstAudioInfo info;
guint64 latency_time; /* the required/actual latency time, this is the
* actual the size of one segment and the
* minimum possible latency we can achieve. */
guint64 buffer_time; /* the required/actual time of the buffer, this is
* the total size of the buffer and maximum
* latency we can compensate for. */
gint segsize; /* size of one buffer segment in bytes, this value
* should be chosen to match latency_time as
* well as possible. */
gint segtotal; /* total number of segments, this value is the
* number of segments of @segsize and should be
* chosen so that it matches buffer_time as
* close as possible. */
/* ABI added 0.10.20 */
gint seglatency; /* number of segments queued in the lower
* level device, defaults to segtotal. */
/*< private >*/
gpointer _gst_reserved[GST_PADDING];
};
#define GST_AUDIO_RING_BUFFER_GET_COND(buf) (&(((GstAudioRingBuffer *)buf)->cond))
#define GST_AUDIO_RING_BUFFER_WAIT(buf) (g_cond_wait (GST_AUDIO_RING_BUFFER_GET_COND (buf), GST_OBJECT_GET_LOCK (buf)))
#define GST_AUDIO_RING_BUFFER_SIGNAL(buf) (g_cond_signal (GST_AUDIO_RING_BUFFER_GET_COND (buf)))
#define GST_AUDIO_RING_BUFFER_BROADCAST(buf)(g_cond_broadcast (GST_AUDIO_RING_BUFFER_GET_COND (buf)))
/**
* GstAudioRingBuffer:
* @cond: used to signal start/stop/pause/resume actions
* @open: boolean indicating that the ringbuffer is open
* @acquired: boolean indicating that the ringbuffer is acquired
* @memory: data in the ringbuffer
* @size: size of data in the ringbuffer
* @spec: format and layout of the ringbuffer data
* @samples_per_seg: number of samples in one segment
* @empty_seg: pointer to memory holding one segment of silence samples
* @state: state of the buffer
* @segdone: readpointer in the ringbuffer
* @segbase: segment corresponding to segment 0 (unused)
* @waiting: is a reader or writer waiting for a free segment
*
* The ringbuffer base class structure.
*/
struct _GstAudioRingBuffer {
GstObject object;
/*< public >*/ /* with LOCK */
GCond cond;
gboolean open;
gboolean acquired;
guint8 *memory;
gsize size;
/*< private >*/
GstClockTime *timestamps;
/*< public >*/ /* with LOCK */
GstAudioRingBufferSpec spec;
gint samples_per_seg;
guint8 *empty_seg;
/*< public >*/ /* ATOMIC */
gint state;
gint segdone;
gint segbase;
gint waiting;
/*< private >*/
GstAudioRingBufferCallback callback;
gpointer cb_data;
gboolean need_reorder;
/* gst[channel_reorder_map[i]] = device[i] */
gint channel_reorder_map[64];
gboolean flushing;
/* ATOMIC */
gint may_start;
gboolean active;
GDestroyNotify cb_data_notify;
/*< private >*/
gpointer _gst_reserved[GST_PADDING - 1];
};
/**
* GstAudioRingBufferClass:
* @parent_class: parent class
* @open_device: open the device, don't set any params or allocate anything
* @acquire: allocate the resources for the ringbuffer using the given spec
* @release: free resources of the ringbuffer
* @close_device: close the device
* @start: start processing of samples
* @pause: pause processing of samples
* @resume: resume processing of samples after pause
* @stop: stop processing of samples
* @delay: get number of frames queued in device
* @activate: activate the thread that starts pulling and monitoring the
* consumed segments in the device.
* @commit: write samples into the ringbuffer
* @clear_all: Optional.
* Clear the entire ringbuffer.
* Subclasses should chain up to the parent implementation to
* invoke the default handler.
*
* The vmethods that subclasses can override to implement the ringbuffer.
*/
struct _GstAudioRingBufferClass {
GstObjectClass parent_class;
/*< public >*/
gboolean (*open_device) (GstAudioRingBuffer *buf);
gboolean (*acquire) (GstAudioRingBuffer *buf, GstAudioRingBufferSpec *spec);
gboolean (*release) (GstAudioRingBuffer *buf);
gboolean (*close_device) (GstAudioRingBuffer *buf);
gboolean (*start) (GstAudioRingBuffer *buf);
gboolean (*pause) (GstAudioRingBuffer *buf);
gboolean (*resume) (GstAudioRingBuffer *buf);
gboolean (*stop) (GstAudioRingBuffer *buf);
guint (*delay) (GstAudioRingBuffer *buf);
/* ABI added */
gboolean (*activate) (GstAudioRingBuffer *buf, gboolean active);
guint (*commit) (GstAudioRingBuffer * buf, guint64 *sample,
guint8 * data, gint in_samples,
gint out_samples, gint * accum);
void (*clear_all) (GstAudioRingBuffer * buf);
/*< private >*/
gpointer _gst_reserved[GST_PADDING];
};
GST_AUDIO_API
GType gst_audio_ring_buffer_get_type(void);
/* callback stuff */
GST_AUDIO_API
void gst_audio_ring_buffer_set_callback (GstAudioRingBuffer *buf,
GstAudioRingBufferCallback cb,
gpointer user_data);
GST_AUDIO_API
void gst_audio_ring_buffer_set_callback_full (GstAudioRingBuffer *buf,
GstAudioRingBufferCallback cb,
gpointer user_data,
GDestroyNotify notify);
GST_AUDIO_API
gboolean gst_audio_ring_buffer_parse_caps (GstAudioRingBufferSpec *spec, GstCaps *caps);
GST_AUDIO_API
void gst_audio_ring_buffer_debug_spec_caps (GstAudioRingBufferSpec *spec);
GST_AUDIO_API
void gst_audio_ring_buffer_debug_spec_buff (GstAudioRingBufferSpec *spec);
GST_AUDIO_API
gboolean gst_audio_ring_buffer_convert (GstAudioRingBuffer * buf, GstFormat src_fmt,
gint64 src_val, GstFormat dest_fmt,
gint64 * dest_val);
/* device state */
GST_AUDIO_API
gboolean gst_audio_ring_buffer_open_device (GstAudioRingBuffer *buf);
GST_AUDIO_API
gboolean gst_audio_ring_buffer_close_device (GstAudioRingBuffer *buf);
GST_AUDIO_API
gboolean gst_audio_ring_buffer_device_is_open (GstAudioRingBuffer *buf);
/* allocate resources */
GST_AUDIO_API
gboolean gst_audio_ring_buffer_acquire (GstAudioRingBuffer *buf, GstAudioRingBufferSpec *spec);
GST_AUDIO_API
gboolean gst_audio_ring_buffer_release (GstAudioRingBuffer *buf);
GST_AUDIO_API
gboolean gst_audio_ring_buffer_is_acquired (GstAudioRingBuffer *buf);
/* set the device channel positions */
GST_AUDIO_API
void gst_audio_ring_buffer_set_channel_positions (GstAudioRingBuffer *buf, const GstAudioChannelPosition *position);
/* activating */
GST_AUDIO_API
gboolean gst_audio_ring_buffer_activate (GstAudioRingBuffer *buf, gboolean active);
GST_AUDIO_API
gboolean gst_audio_ring_buffer_is_active (GstAudioRingBuffer *buf);
/* flushing */
GST_AUDIO_API
void gst_audio_ring_buffer_set_flushing (GstAudioRingBuffer *buf, gboolean flushing);
GST_AUDIO_API
gboolean gst_audio_ring_buffer_is_flushing (GstAudioRingBuffer *buf);
/* playback/pause */
GST_AUDIO_API
gboolean gst_audio_ring_buffer_start (GstAudioRingBuffer *buf);
GST_AUDIO_API
gboolean gst_audio_ring_buffer_pause (GstAudioRingBuffer *buf);
GST_AUDIO_API
gboolean gst_audio_ring_buffer_stop (GstAudioRingBuffer *buf);
/* get status */
GST_AUDIO_API
guint gst_audio_ring_buffer_delay (GstAudioRingBuffer *buf);
GST_AUDIO_API
guint64 gst_audio_ring_buffer_samples_done (GstAudioRingBuffer *buf);
GST_AUDIO_API
void gst_audio_ring_buffer_set_sample (GstAudioRingBuffer *buf, guint64 sample);
/* clear all segments */
GST_AUDIO_API
void gst_audio_ring_buffer_clear_all (GstAudioRingBuffer *buf);
/* commit samples */
GST_AUDIO_API
guint gst_audio_ring_buffer_commit (GstAudioRingBuffer * buf, guint64 *sample,
guint8 * data, gint in_samples,
gint out_samples, gint * accum);
/* read samples */
GST_AUDIO_API
guint gst_audio_ring_buffer_read (GstAudioRingBuffer *buf, guint64 sample,
guint8 *data, guint len, GstClockTime *timestamp);
/* Set timestamp on buffer */
GST_AUDIO_API
void gst_audio_ring_buffer_set_timestamp (GstAudioRingBuffer * buf, gint readseg, GstClockTime
timestamp);
/* mostly protected */
/* not yet implemented
gboolean gst_audio_ring_buffer_prepare_write (GstAudioRingBuffer *buf, gint *segment, guint8 **writeptr, gint *len);
*/
GST_AUDIO_API
gboolean gst_audio_ring_buffer_prepare_read (GstAudioRingBuffer *buf, gint *segment,
guint8 **readptr, gint *len);
GST_AUDIO_API
void gst_audio_ring_buffer_clear (GstAudioRingBuffer *buf, gint segment);
GST_AUDIO_API
void gst_audio_ring_buffer_advance (GstAudioRingBuffer *buf, guint advance);
GST_AUDIO_API
void gst_audio_ring_buffer_may_start (GstAudioRingBuffer *buf, gboolean allowed);
G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstAudioRingBuffer, gst_object_unref)
G_END_DECLS
#endif /* __GST_AUDIO_RING_BUFFER_H__ */
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