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Added docs for the audio libs.
Original commit message from CVS: * docs/libs/gst-plugins-base-libs-docs.sgml: * docs/libs/gst-plugins-base-libs-sections.txt: * gst-libs/gst/audio/gstaudioclock.c: * gst-libs/gst/audio/gstaudioclock.h: * gst-libs/gst/audio/gstaudiosink.c: * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstaudiosrc.c: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_render): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init): * gst-libs/gst/audio/gstbaseaudiosrc.h: * gst-libs/gst/audio/gstringbuffer.h: Added docs for the audio libs.
This commit is contained in:
Wim Taymans
parent
bb54e8cb1f
commit
7367722509
13 changed files with 303 additions and 26 deletions
17
ChangeLog
17
ChangeLog
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@ -1,3 +1,20 @@
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2006-09-27 Wim Taymans <wim@fluendo.com>
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* docs/libs/gst-plugins-base-libs-docs.sgml:
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* docs/libs/gst-plugins-base-libs-sections.txt:
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* gst-libs/gst/audio/gstaudioclock.c:
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* gst-libs/gst/audio/gstaudioclock.h:
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* gst-libs/gst/audio/gstaudiosink.c:
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* gst-libs/gst/audio/gstaudiosink.h:
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* gst-libs/gst/audio/gstaudiosrc.c:
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* gst-libs/gst/audio/gstbaseaudiosink.c:
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(gst_base_audio_sink_render):
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* gst-libs/gst/audio/gstbaseaudiosink.h:
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* gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init):
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* gst-libs/gst/audio/gstbaseaudiosrc.h:
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* gst-libs/gst/audio/gstringbuffer.h:
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Added docs for the audio libs.
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2006-09-27 Tim-Philipp Müller <tim at centricular dot net>
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* tests/check/Makefile.am:
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@ -124,7 +124,7 @@
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&GstTuner;
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&GstTunerChannel;
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&GstTunerNorm;
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&GstVideoorientation;
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&GstVideoOrientation;
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&GstXOverlay;
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</chapter>
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@ -23,17 +23,16 @@ gst_audio_structure_set_int
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<FILE>gstaudioclock</FILE>
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<INCLUDE>gst/audio/gstaudioclock.h</INCLUDE>
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GstAudioClock
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GstAudioClockClass
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GstAudioClockGetTimeFunc
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gst_audio_clock_new
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<SUBSECTION Standard>
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GstAudioClockClass
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GST_AUDIO_CLOCK
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GST_IS_AUDIO_CLOCK
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GST_TYPE_AUDIO_CLOCK
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gst_audio_clock_get_type
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GST_AUDIO_CLOCK_CLASS
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GST_IS_AUDIO_CLOCK_CLASS
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GST_AUDIO_CLOCK_GET_CLASS
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</SECTION>
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<SECTION>
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@ -458,6 +457,20 @@ GST_IS_TUNER_NORM_CLASS
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gst_tuner_norm_get_type
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</SECTION>
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<SECTION>
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<FILE>gstvideoorientation</FILE>
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<INCLUDE>gst/interfaces/videoorientation.h</INCLUDE>
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GstVideoOrientation
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GstVideoOrientationInterface
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<SUBSECTION Standard>
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GST_TYPE_VIDEO_ORIENTATION
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GST_VIDEO_ORIENTATION
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GST_VIDEO_ORIENTATION_CLASS
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GST_IS_VIDEO_ORIENTATION
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GST_IS_VIDEO_ORIENTATION_CLASS
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gst_video_orientation_get_type
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</SECTION>
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<SECTION>
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<FILE>gstxoverlay</FILE>
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<INCLUDE>gst/interfaces/xoverlay.h</INCLUDE>
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@ -800,7 +813,6 @@ gst_rtp_buffer_get_seq
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gst_rtp_buffer_get_ssrc
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gst_rtp_buffer_get_timestamp
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gst_rtp_buffer_get_version
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gst_rtp_buffer_new
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gst_rtp_buffer_new_allocate
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gst_rtp_buffer_new_allocate_len
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gst_rtp_buffer_new_copy_data
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@ -20,6 +20,19 @@
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* Boston, MA 02111-1307, USA.
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*/
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/**
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* SECTION:gstaudioclock
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* @short_description: Helper object for implementing audio clocks
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* @see_also: #GstBaseAudioSink, #GstSystemClock
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*
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* #GstAudioClock makes it easy for elements to implement a #GstClock, they
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* simply need to provide a function that returns the current clock time.
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*
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* This object is internally used to implement the clock in #GstBaseAudioSink.
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*
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* Last reviewed on 2006-09-27 (0.10.12)
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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@ -86,6 +99,18 @@ gst_audio_clock_init (GstAudioClock * clock)
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GST_OBJECT_FLAG_SET (clock, GST_CLOCK_FLAG_CAN_SET_MASTER);
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}
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/**
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* gst_audio_clock_new:
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* @name: the name of the clock
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* @func: a function
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* @user_data: user data
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*
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* Create a new #GstAudioClock instance. Whenever the clock time should be
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* calculated it will call @func with @user_data. When @func returns
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* #GST_CLOCK_TIME_NONE, the clock will return the last reported time.
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*
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* Returns: a new #GstAudioClock casted to a #GstClock.
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*/
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GstClock *
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gst_audio_clock_new (gchar * name, GstAudioClockGetTimeFunc func,
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gpointer user_data)
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@ -20,7 +20,6 @@
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* Boston, MA 02111-1307, USA.
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*/
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#ifndef __GST_AUDIO_CLOCK_H__
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#define __GST_AUDIO_CLOCK_H__
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@ -42,8 +41,26 @@ G_BEGIN_DECLS
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typedef struct _GstAudioClock GstAudioClock;
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typedef struct _GstAudioClockClass GstAudioClockClass;
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/**
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* GstAudioClockGetTimeFunc:
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* @clock: the #GstAudioClock
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* @user_data: user data
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*
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* This function will be called whenever the current clock time needs to be
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* calculated. If this function returns #GST_CLOCK_TIME_NONE, the last reported
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* time will be returned by the clock.
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*
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* Returns: the current time or #GST_CLOCK_TIME_NONE if the previous time should
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* be used.
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*/
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typedef GstClockTime (*GstAudioClockGetTimeFunc) (GstClock *clock, gpointer user_data);
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/**
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* GstAudioClock:
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* @clock: parent #GstSystemClock
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*
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* Opaque #GstAudioClock.
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*/
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struct _GstAudioClock {
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GstSystemClock clock;
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@ -20,6 +20,53 @@
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* Boston, MA 02111-1307, USA.
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*/
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/**
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* SECTION:gstaudiosink
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* @short_description: Simple base class for audio sinks
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* @see_also: #GstBaseAudioSink, #GstRingBuffer, #GstAudioSink.
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*
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* This is the most simple base class for audio sinks that only requires
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* subclasses to implement a set of simple functions:
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*
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* <variablelist>
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* <varlistentry>
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* <term>open()</term>
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* <listitem><para>Open the device.</para></listitem>
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* </varlistentry>
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* <varlistentry>
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* <term>prepare()</term>
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* <listitem><para>Configure the device with the specified format.</para></listitem>
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* </varlistentry>
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* <varlistentry>
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* <term>write()</term>
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* <listitem><para>Write samples to the device.</para></listitem>
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* </varlistentry>
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* <varlistentry>
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* <term>reset()</term>
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* <listitem><para>Unblock writes and flush the device.</para></listitem>
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* </varlistentry>
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* <varlistentry>
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* <term>delay()</term>
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* <listitem><para>Get the number of samples written but not yet played
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* by the device.</para></listitem>
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* </varlistentry>
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* <varlistentry>
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* <term>unprepare()</term>
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* <listitem><para>Undo operations done by prepare.</para></listitem>
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* </varlistentry>
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* <varlistentry>
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* <term>close()</term>
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* <listitem><para>Close the device.</para></listitem>
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* </varlistentry>
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* </variablelist>
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*
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* All scheduling of samples and timestamps is done in this base class
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* together with #GstBaseAudioSink using a default implementation of a
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* #GstRingBuffer that uses threads.
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*
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* Last reviewed on 2006-09-27 (0.10.12)
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*/
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#include <string.h>
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#include "gstaudiosink.h"
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@ -20,22 +20,6 @@
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* Boston, MA 02111-1307, USA.
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*/
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/* a base class for simple audio sinks.
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*
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* This base class only requires subclasses to implement a set
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* of simple functions.
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*
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* - open: open the device with the specified caps
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* - write: write the samples to the audio device
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* - close: close the device
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* - delay: the number of samples queued in the device
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* - reset: unblock a write to the device and reset.
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*
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* All scheduling of samples and timestamps is done in this
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* base class together with the GstBaseAudioSink using a
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* default implementation of a ringbuffer that uses threads.
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*/
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#ifndef __GST_AUDIO_SINK_H__
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#define __GST_AUDIO_SINK_H__
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@ -54,6 +38,11 @@ G_BEGIN_DECLS
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typedef struct _GstAudioSink GstAudioSink;
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typedef struct _GstAudioSinkClass GstAudioSinkClass;
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/**
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* GstAudioSink:
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*
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* Opaque #GstAudioSink.
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*/
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struct _GstAudioSink {
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GstBaseAudioSink element;
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@ -64,6 +53,22 @@ struct _GstAudioSink {
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gpointer _gst_reserved[GST_PADDING];
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};
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/**
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* GstAudioSinkClass:
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* @parent_class: the parent class structure.
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* @open: Open the device. No configuration needs to be done at this point.
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* This function is also used to check if the device is available.
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* @prepare: Prepare the device to operate with the specified parameters.
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* @unprepare: Undo operations done in prepare.
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* @close: Close the device.
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* @write: Write data to the device.
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* @delay: Return how many samples are still in the device. This is used to
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* drive the synchronisation.
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* @reset: Returns as quickly as possible from a write and flush any pending
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* samples from the device.
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*
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* #GstAudioSink class. Override the vmethods to implement functionality.
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*/
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struct _GstAudioSinkClass {
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GstBaseAudioSinkClass parent_class;
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@ -20,6 +20,53 @@
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* Boston, MA 02111-1307, USA.
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*/
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/**
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* SECTION:gstaudiosrc
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* @short_description: Simple base class for audio sources
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* @see_also: #GstBaseAudioSrc, #GstRingBuffer, #GstAudioSrc.
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*
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* This is the most simple base class for audio sources that only requires
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* subclasses to implement a set of simple functions:
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*
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* <variablelist>
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* <varlistentry>
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* <term>open()</term>
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* <listitem><para>Open the device.</para></listitem>
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* </varlistentry>
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* <varlistentry>
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* <term>prepare()</term>
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* <listitem><para>Configure the device with the specified format.</para></listitem>
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* </varlistentry>
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* <varlistentry>
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* <term>read()</term>
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* <listitem><para>Read samples from the device.</para></listitem>
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* </varlistentry>
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* <varlistentry>
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* <term>reset()</term>
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* <listitem><para>Unblock reads and flush the device.</para></listitem>
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* </varlistentry>
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* <varlistentry>
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* <term>delay()</term>
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* <listitem><para>Get the number of samples in the device but not yet read.
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* </para></listitem>
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* </varlistentry>
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* <varlistentry>
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* <term>unprepare()</term>
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* <listitem><para>Undo operations done by prepare.</para></listitem>
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* </varlistentry>
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* <varlistentry>
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* <term>close()</term>
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* <listitem><para>Close the device.</para></listitem>
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* </varlistentry>
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* </variablelist>
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*
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* All scheduling of samples and timestamps is done in this base class
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* together with #GstBaseAudioSrc using a default implementation of a
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* #GstRingBuffer that uses threads.
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*
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* Last reviewed on 2006-09-27 (0.10.12)
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*/
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#include <string.h>
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#include "gstaudiosrc.h"
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@ -20,6 +20,18 @@
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* Boston, MA 02111-1307, USA.
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*/
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/**
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* SECTION:gstbaseaudiosink
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* @short_description: Base class for audio sinks
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* @see_also: #GstAudioSink, #GstRingBuffer.
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*
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* This is the base class for audio sinks. Subclasses need to implement the
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* ::create_ringbuffer vmethod. This base class will then take care of
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* writing samples to the ringbuffer, synchronisation, clipping and flushing.
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*
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* Last reviewed on 2006-09-27 (0.10.12)
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*/
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#include <string.h>
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#include "gstbaseaudiosink.h"
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@ -629,8 +641,11 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
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} else {
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/* timestamps drifted apart from previous samples too much, we need to
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* resync. We log this as an element warning. */
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GST_ELEMENT_WARNING (sink, CORE, CLOCK, (NULL),
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("Unexpected discontinuity in audio timestamps of more than half a second"));
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GST_ELEMENT_WARNING (sink, CORE, CLOCK,
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("Compensating for audio synchronisation problems"),
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("Unexpected discontinuity in audio timestamps of more "
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"than half a second (%" GST_TIME_FORMAT "), resyncing",
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GST_TIME_ARGS (diff)));
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}
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no_align:
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@ -701,6 +716,16 @@ stopping:
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}
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}
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/**
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* gst_base_audio_sink_create_ringbuffer:
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* @sink: a #GstBaseAudioSink.
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*
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* Create and return the #GstRingBuffer for @sink. This function will call the
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* ::create_ringbuffer vmethod and will set @sink as the parent of the returned
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* buffer (see gst_object_set_parent()).
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*
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* Returns: The new ringbuffer of @sink.
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*/
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GstRingBuffer *
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gst_base_audio_sink_create_ringbuffer (GstBaseAudioSink * sink)
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{
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|
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@ -63,12 +63,29 @@ G_BEGIN_DECLS
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#define GST_IS_BASE_AUDIO_SINK(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_BASE_AUDIO_SINK))
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#define GST_IS_BASE_AUDIO_SINK_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_BASE_AUDIO_SINK))
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/**
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* GST_BASE_AUDIO_SINK_CLOCK:
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* @obj: a #GstBaseAudioSink
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*
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* Get the #GstClock of @obj.
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*/
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#define GST_BASE_AUDIO_SINK_CLOCK(obj) (GST_BASE_AUDIO_SINK (obj)->clock)
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/**
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* GST_BASE_AUDIO_SINK_PAD:
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* @obj: a #GstBaseAudioSink
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*
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* Get the sink #GstPad of @obj.
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*/
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#define GST_BASE_AUDIO_SINK_PAD(obj) (GST_BASE_SINK (obj)->sinkpad)
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typedef struct _GstBaseAudioSink GstBaseAudioSink;
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typedef struct _GstBaseAudioSinkClass GstBaseAudioSinkClass;
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/**
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* GstBaseAudioSink:
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*
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* Opaque #GstBaseAudioSink.
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*/
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struct _GstBaseAudioSink {
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GstBaseSink element;
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@ -77,8 +94,8 @@ struct _GstBaseAudioSink {
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GstRingBuffer *ringbuffer;
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/* required buffer and latency in microseconds */
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guint64 buffer_time;
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guint64 latency_time;
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guint64 buffer_time;
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guint64 latency_time;
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/* the next sample to write */
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guint64 next_sample;
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@ -91,6 +108,14 @@ struct _GstBaseAudioSink {
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gpointer _gst_reserved[GST_PADDING];
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};
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/**
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* GstBaseAudioSinkClass:
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* @parent_class: the parent class.
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* @create_ringbuffer: create and return a #GstRingBuffer to write to.
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*
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* #GstBaseAudioSink class. Override the vmethod to implement
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* functionality.
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*/
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struct _GstBaseAudioSinkClass {
|
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GstBaseSinkClass parent_class;
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|
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|||
|
|
@ -20,6 +20,18 @@
|
|||
* Boston, MA 02111-1307, USA.
|
||||
*/
|
||||
|
||||
/**
|
||||
* SECTION:gstbaseaudiosrc
|
||||
* @short_description: Base class for audio sources
|
||||
* @see_also: #GstAudioSrc, #GstRingBuffer.
|
||||
*
|
||||
* This is the base class for audio sources. Subclasses need to implement the
|
||||
* ::create_ringbuffer vmethod. This base class will then take care of
|
||||
* reading samples from the ringbuffer, synchronisation and flushing.
|
||||
*
|
||||
* Last reviewed on 2006-09-27 (0.10.12)
|
||||
*/
|
||||
|
||||
#include <string.h>
|
||||
|
||||
#include "gstbaseaudiosrc.h"
|
||||
|
|
@ -144,6 +156,7 @@ gst_base_audio_src_init (GstBaseAudioSrc * baseaudiosrc,
|
|||
|
||||
/* we are always a live source */
|
||||
gst_base_src_set_live (GST_BASE_SRC (baseaudiosrc), TRUE);
|
||||
/* we operate in time */
|
||||
gst_base_src_set_format (GST_BASE_SRC (baseaudiosrc), GST_FORMAT_TIME);
|
||||
}
|
||||
|
||||
|
|
@ -527,6 +540,16 @@ stopped:
|
|||
}
|
||||
}
|
||||
|
||||
/**
|
||||
* gst_base_audio_src_create_ringbuffer:
|
||||
* @src: a #GstBaseAudioSrc.
|
||||
*
|
||||
* Create and return the #GstRingBuffer for @src. This function will call the
|
||||
* ::create_ringbuffer vmethod and will set @src as the parent of the returned
|
||||
* buffer (see gst_object_set_parent()).
|
||||
*
|
||||
* Returns: The new ringbuffer of @src.
|
||||
*/
|
||||
GstRingBuffer *
|
||||
gst_base_audio_src_create_ringbuffer (GstBaseAudioSrc * src)
|
||||
{
|
||||
|
|
|
|||
|
|
@ -40,12 +40,29 @@ G_BEGIN_DECLS
|
|||
#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;
|
||||
|
||||
/**
|
||||
* GstBaseAudioSrc:
|
||||
*
|
||||
* Opaque #GstBaseAudioSrc.
|
||||
*/
|
||||
struct _GstBaseAudioSrc {
|
||||
GstPushSrc element;
|
||||
|
||||
|
|
@ -67,6 +84,14 @@ struct _GstBaseAudioSrc {
|
|||
gpointer _gst_reserved[GST_PADDING];
|
||||
};
|
||||
|
||||
/**
|
||||
* 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;
|
||||
|
||||
|
|
|
|||
|
|
@ -39,7 +39,16 @@ typedef struct _GstRingBuffer GstRingBuffer;
|
|||
typedef struct _GstRingBufferClass GstRingBufferClass;
|
||||
typedef struct _GstRingBufferSpec GstRingBufferSpec;
|
||||
|
||||
/* called to fill data with len bytes of samples */
|
||||
/**
|
||||
* GstRingBufferCallback:
|
||||
* @rbuf: a #GstRingBuffer
|
||||
* @data: target to fill
|
||||
* @len: amount to fill
|
||||
* @user_data: user data
|
||||
*
|
||||
* This function is set with gst_ring_buffer_set_callback() and is
|
||||
* called to fill the memory at @data with @len bytes of samples.
|
||||
*/
|
||||
typedef void (*GstRingBufferCallback) (GstRingBuffer *rbuf, guint8* data, guint len, gpointer user_data);
|
||||
|
||||
/**
|
||||
|
|
|
|||
Loading…
Reference in a new issue