| /* GStreamer |
| * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu> |
| * 2005 Wim Taymans <wim@fluendo.com> |
| * |
| * gstbaseaudiosrc.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., 59 Temple Place - Suite 330, |
| * Boston, MA 02111-1307, USA. |
| */ |
| |
| /* a base class for audio sources. |
| */ |
| |
| #ifndef __GST_BASE_AUDIO_SRC_H__ |
| #define __GST_BASE_AUDIO_SRC_H__ |
| |
| #include <gst/gst.h> |
| #include <gst/base/gstpushsrc.h> |
| #include "gstringbuffer.h" |
| #include "gstaudioclock.h" |
| |
| G_BEGIN_DECLS |
| |
| #define GST_TYPE_BASE_AUDIO_SRC (gst_base_audio_src_get_type()) |
| #define GST_BASE_AUDIO_SRC(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_BASE_AUDIO_SRC,GstBaseAudioSrc)) |
| #define GST_BASE_AUDIO_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_BASE_AUDIO_SRC,GstBaseAudioSrcClass)) |
| #define GST_BASE_AUDIO_SRC_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_BASE_AUDIO_SRC, GstBaseAudioSrcClass)) |
| #define GST_IS_BASE_AUDIO_SRC(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_BASE_AUDIO_SRC)) |
| #define GST_IS_BASE_AUDIO_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_BASE_AUDIO_SRC)) |
| |
| /** |
| * GST_BASE_AUDIO_SRC_CLOCK: |
| * @obj: a #GstBaseAudioSrc |
| * |
| * Get the #GstClock of @obj. |
| */ |
| #define GST_BASE_AUDIO_SRC_CLOCK(obj) (GST_BASE_AUDIO_SRC (obj)->clock) |
| /** |
| * GST_BASE_AUDIO_SRC_PAD: |
| * @obj: a #GstBaseAudioSrc |
| * |
| * Get the source #GstPad of @obj. |
| */ |
| #define GST_BASE_AUDIO_SRC_PAD(obj) (GST_BASE_SRC (obj)->srcpad) |
| |
| typedef struct _GstBaseAudioSrc GstBaseAudioSrc; |
| typedef struct _GstBaseAudioSrcClass GstBaseAudioSrcClass; |
| typedef struct _GstBaseAudioSrcPrivate GstBaseAudioSrcPrivate; |
| |
| /** |
| * GstBaseAudioSrcSlaveMethod: |
| * @GST_BASE_AUDIO_SRC_SLAVE_RESAMPLE: Resample to match the master clock. |
| * @GST_BASE_AUDIO_SRC_SLAVE_RETIMESTAMP: Retimestamp output buffers with master |
| * clock time. |
| * @GST_BASE_AUDIO_SRC_SLAVE_SKEW: Adjust capture pointer when master clock |
| * drifts too much. |
| * @GST_BASE_AUDIO_SRC_SLAVE_NONE: No adjustment is done. |
| * |
| * Different possible clock slaving algorithms when the internal audio clock was |
| * not selected as the pipeline clock. |
| */ |
| typedef enum |
| { |
| GST_BASE_AUDIO_SRC_SLAVE_RESAMPLE, |
| GST_BASE_AUDIO_SRC_SLAVE_RETIMESTAMP, |
| GST_BASE_AUDIO_SRC_SLAVE_SKEW, |
| GST_BASE_AUDIO_SRC_SLAVE_NONE |
| } GstBaseAudioSrcSlaveMethod; |
| |
| #define GST_TYPE_BASE_AUDIO_SRC_SLAVE_METHOD (gst_base_audio_src_slave_method_get_type ()) |
| |
| /** |
| * GstBaseAudioSrc: |
| * |
| * Opaque #GstBaseAudioSrc. |
| */ |
| struct _GstBaseAudioSrc { |
| GstPushSrc element; |
| |
| /*< protected >*/ /* with LOCK */ |
| /* our ringbuffer */ |
| GstRingBuffer *ringbuffer; |
| |
| /* required buffer and latency */ |
| GstClockTime buffer_time; |
| GstClockTime latency_time; |
| |
| /* the next sample to write */ |
| guint64 next_sample; |
| |
| /* clock */ |
| GstClock *clock; |
| |
| /*< private >*/ |
| GstBaseAudioSrcPrivate *priv; |
| |
| gpointer _gst_reserved[GST_PADDING - 1]; |
| }; |
| |
| /** |
| * GstBaseAudioSrcClass: |
| * @parent_class: the parent class. |
| * @create_ringbuffer: create and return a #GstRingBuffer to read from. |
| * |
| * #GstBaseAudioSrc class. Override the vmethod to implement |
| * functionality. |
| */ |
| struct _GstBaseAudioSrcClass { |
| GstPushSrcClass parent_class; |
| |
| /* subclass ringbuffer allocation */ |
| GstRingBuffer* (*create_ringbuffer) (GstBaseAudioSrc *src); |
| |
| /*< private >*/ |
| gpointer _gst_reserved[GST_PADDING]; |
| }; |
| |
| GType gst_base_audio_src_get_type(void); |
| GType gst_base_audio_src_slave_method_get_type (void); |
| |
| GstRingBuffer *gst_base_audio_src_create_ringbuffer (GstBaseAudioSrc *src); |
| |
| void gst_base_audio_src_set_provide_clock (GstBaseAudioSrc *src, gboolean provide); |
| gboolean gst_base_audio_src_get_provide_clock (GstBaseAudioSrc *src); |
| |
| void gst_base_audio_src_set_slave_method (GstBaseAudioSrc *src, |
| GstBaseAudioSrcSlaveMethod method); |
| GstBaseAudioSrcSlaveMethod |
| gst_base_audio_src_get_slave_method (GstBaseAudioSrc *src); |
| |
| |
| G_END_DECLS |
| |
| #endif /* __GST_BASE_AUDIO_SRC_H__ */ |