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[gst-cvs] wtay gst-plugins-base: gst-plugins-base/ gst-plugins-base/docs/plugins/ gst-plugins-base/gst/audioconvert/ gst-plugins-base/gst/audioresample/
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From: <wt...@ke...> - 2006-03-02 18:24:08
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CVS Root: /cvs/gstreamer
Module: gst-plugins-base
Changes by: wtay
Date: Thu Mar 02 2006 18:24:07 UTC
Log message:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
Add audioresample to docs.
* gst/audioconvert/gstaudioconvert.c:
Add revision date.
* gst/audioresample/gstaudioresample.c:
(gst_audioresample_base_init), (gst_audioresample_class_init),
(gst_audioresample_init), (gst_audioresample_dispose),
(audioresample_get_unit_size), (audioresample_transform_caps),
(resample_set_state_from_caps), (audioresample_transform_size),
(audioresample_set_caps), (audioresample_event),
(audioresample_do_output), (audioresample_transform),
(audioresample_pushthrough), (gst_audioresample_set_property),
(gst_audioresample_get_property), (plugin_init):
* gst/audioresample/gstaudioresample.h:
Added docs.
Small code cleanups.
Modified files:
. : ChangeLog
docs/plugins : Makefile.am gst-plugins-base-plugins-docs.sgml
gst-plugins-base-plugins-sections.txt
gst/audioconvert: gstaudioconvert.c
gst/audioresample: gstaudioresample.c gstaudioresample.h
Links:
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-base/ChangeLog.diff?r1=1.2440&r2=1.2441
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-base/docs/plugins/Makefile.am.diff?r1=1.23&r2=1.24
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-base/docs/plugins/gst-plugins-base-plugins-docs.sgml.diff?r1=1.21&r2=1.22
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-base/docs/plugins/gst-plugins-base-plugins-sections.txt.diff?r1=1.18&r2=1.19
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-base/gst/audioconvert/gstaudioconvert.c.diff?r1=1.97&r2=1.98
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-base/gst/audioresample/gstaudioresample.c.diff?r1=1.14&r2=1.15
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-base/gst/audioresample/gstaudioresample.h.diff?r1=1.4&r2=1.5
====Begin Diffs====
Index: ChangeLog
===================================================================
RCS file: /cvs/gstreamer/gst-plugins-base/ChangeLog,v
retrieving revision 1.2440
retrieving revision 1.2441
diff -u -d -r1.2440 -r1.2441
--- ChangeLog 2 Mar 2006 17:46:35 -0000 1.2440
+++ ChangeLog 2 Mar 2006 18:23:54 -0000 1.2441
@@ -3,6 +3,29 @@
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
+ Add audioresample to docs.
+
+ * gst/audioconvert/gstaudioconvert.c:
+ Add revision date.
+ * gst/audioresample/gstaudioresample.c:
+ (gst_audioresample_base_init), (gst_audioresample_class_init),
+ (gst_audioresample_init), (gst_audioresample_dispose),
+ (audioresample_get_unit_size), (audioresample_transform_caps),
+ (resample_set_state_from_caps), (audioresample_transform_size),
+ (audioresample_set_caps), (audioresample_event),
+ (audioresample_do_output), (audioresample_transform),
+ (audioresample_pushthrough), (gst_audioresample_set_property),
+ (gst_audioresample_get_property), (plugin_init):
+ * gst/audioresample/gstaudioresample.h:
+ Added docs.
+ Small code cleanups.
+2006-03-02 Wim Taymans <wi...@fl...>
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-base-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
Added videoscale to docs.
* gst/videorate/gstvideorate.c: (gst_video_rate_flush_prev),
Index: Makefile.am
RCS file: /cvs/gstreamer/gst-plugins-base/docs/plugins/Makefile.am,v
retrieving revision 1.23
retrieving revision 1.24
diff -u -d -r1.23 -r1.24
--- Makefile.am 2 Mar 2006 17:46:36 -0000 1.23
+++ Makefile.am 2 Mar 2006 18:23:55 -0000 1.24
@@ -89,6 +89,7 @@
$(top_srcdir)/ext/vorbis/vorbisenc.h \
$(top_srcdir)/ext/vorbis/vorbisparse.h \
$(top_srcdir)/gst/audioconvert/gstaudioconvert.h \
+ $(top_srcdir)/gst/audioresample/gstaudioresample.h \
$(top_srcdir)/gst/audiotestsrc/gstaudiotestsrc.h \
$(top_srcdir)/gst/ffmpegcolorspace/gstffmpegcolorspace.h \
$(top_srcdir)/gst/tcp/gstmultifdsink.h \
Index: gst-plugins-base-plugins-docs.sgml
RCS file: /cvs/gstreamer/gst-plugins-base/docs/plugins/gst-plugins-base-plugins-docs.sgml,v
retrieving revision 1.21
retrieving revision 1.22
diff -u -d -r1.21 -r1.22
--- gst-plugins-base-plugins-docs.sgml 2 Mar 2006 17:46:36 -0000 1.21
+++ gst-plugins-base-plugins-docs.sgml 2 Mar 2006 18:23:55 -0000 1.22
@@ -16,6 +16,7 @@
<xi:include href="xml/element-alsasink.xml" />
<xi:include href="xml/element-alsasrc.xml" />
<xi:include href="xml/element-audioconvert.xml" />
+ <xi:include href="xml/element-audioresample.xml" />
<xi:include href="xml/element-audiotestsrc.xml" />
<xi:include href="xml/element-clockoverlay.xml" />
<xi:include href="xml/element-ffmpegcolorspace.xml" />
Index: gst-plugins-base-plugins-sections.txt
RCS file: /cvs/gstreamer/gst-plugins-base/docs/plugins/gst-plugins-base-plugins-sections.txt,v
retrieving revision 1.18
retrieving revision 1.19
diff -u -d -r1.18 -r1.19
--- gst-plugins-base-plugins-sections.txt 2 Mar 2006 17:46:36 -0000 1.18
+++ gst-plugins-base-plugins-sections.txt 2 Mar 2006 18:23:55 -0000 1.19
@@ -58,6 +58,20 @@
</SECTION>
<SECTION>
+<FILE>element-audioresample</FILE>
+<TITLE>audioresample</TITLE>
+GstAudioresample
+<SUBSECTION Standard>
+GST_AUDIORESAMPLE
+GST_IS_AUDIORESAMPLE
+GST_TYPE_AUDIORESAMPLE
+gst_audioresample_get_type
+GST_AUDIORESAMPLE_CLASS
+GST_IS_AUDIORESAMPLE_CLASS
+GstAudioresampleClass
+</SECTION>
+<SECTION>
<FILE>element-audiotestsrc</FILE>
<TITLE>audiotestsrc</TITLE>
GstAudioTestSrc
Index: gstaudioconvert.c
RCS file: /cvs/gstreamer/gst-plugins-base/gst/audioconvert/gstaudioconvert.c,v
retrieving revision 1.97
retrieving revision 1.98
diff -u -d -r1.97 -r1.98
--- gstaudioconvert.c 6 Dec 2005 19:41:56 -0000 1.97
+++ gstaudioconvert.c 2 Mar 2006 18:23:55 -0000 1.98
@@ -44,6 +44,8 @@
* generated by audiotestsrc.
* </para>
* </refsect2>
+ *
+ * Last reviewed on 2006-03-02 (0.10.4)
*/
/*
Index: gstaudioresample.c
RCS file: /cvs/gstreamer/gst-plugins-base/gst/audioresample/gstaudioresample.c,v
retrieving revision 1.14
retrieving revision 1.15
diff -u -d -r1.14 -r1.15
--- gstaudioresample.c 15 Dec 2005 10:30:14 -0000 1.14
+++ gstaudioresample.c 2 Mar 2006 18:23:55 -0000 1.15
@@ -19,6 +19,25 @@
/* Element-Checklist-Version: 5 */
+/**
+ * SECTION:element-audioresample
+ * <refsect2>
+ * Audioresample resamples raw audio buffers to different sample rates using
+ * a configurable windowing function to enhance quality.
+ * <title>Example launch line</title>
+ * <para>
+ * <programlisting>
+ * gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! audio/x-raw-int, rate=8000 ! alsasink
+ * </programlisting>
+ * Decode an Ogg/Vorbis downsample to 8Khz and play sound through alsa.
+ * To create the Ogg/Vorbis file refer to the documentation of vorbisenc.
+ * </para>
+ * </refsect2>
+ */
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
@@ -52,8 +71,8 @@
enum
{
- ARG_0,
- ARG_FILTERLEN
+ PROP_0,
+ PROP_FILTERLEN
};
#define SUPPORTED_CAPS \
@@ -72,39 +91,38 @@
"audio/x-raw-float, "
"rate = (int) [ 1, MAX ], "
"channels = (int) [ 1, MAX ], "
- "endianness = (int) BYTE_ORDER, " "width = (int) 32")
+ "endianness = (int) BYTE_ORDER, " "width = (int) 32"
- static GstStaticPadTemplate gst_audioresample_sink_template =
- GST_STATIC_PAD_TEMPLATE ("sink",
- GST_PAD_SINK, GST_PAD_ALWAYS, SUPPORTED_CAPS);
+static GstStaticPadTemplate gst_audioresample_sink_template =
+GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK, GST_PAD_ALWAYS, SUPPORTED_CAPS);
- static GstStaticPadTemplate gst_audioresample_src_template =
- GST_STATIC_PAD_TEMPLATE ("src",
- GST_PAD_SRC, GST_PAD_ALWAYS, SUPPORTED_CAPS);
+static GstStaticPadTemplate gst_audioresample_src_template =
+GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC, GST_PAD_ALWAYS, SUPPORTED_CAPS);
- static void gst_audioresample_dispose (GObject * object);
+static void gst_audioresample_dispose (GObject * object);
- static void gst_audioresample_set_property (GObject * object,
- guint prop_id, const GValue * value, GParamSpec * pspec);
- static void gst_audioresample_get_property (GObject * object,
- guint prop_id, GValue * value, GParamSpec * pspec);
+static void gst_audioresample_set_property (GObject * object,
+ guint prop_id, const GValue * value, GParamSpec * pspec);
+static void gst_audioresample_get_property (GObject * object,
+ guint prop_id, GValue * value, GParamSpec * pspec);
/* vmethods */
- gboolean audioresample_get_unit_size (GstBaseTransform * base,
- GstCaps * caps, guint * size);
- GstCaps *audioresample_transform_caps (GstBaseTransform * base,
- GstPadDirection direction, GstCaps * caps);
- gboolean audioresample_transform_size (GstBaseTransform * trans,
- GstPadDirection direction, GstCaps * incaps, guint insize,
- GstCaps * outcaps, guint * outsize);
- gboolean audioresample_set_caps (GstBaseTransform * base, GstCaps * incaps,
- GstCaps * outcaps);
- static GstFlowReturn audioresample_pushthrough (GstAudioresample *
- audioresample);
- static GstFlowReturn audioresample_transform (GstBaseTransform * base,
- GstBuffer * inbuf, GstBuffer * outbuf);
- static gboolean audioresample_event (GstBaseTransform * base,
- GstEvent * event);
+gboolean audioresample_get_unit_size (GstBaseTransform * base,
+ GstCaps * caps, guint * size);
+GstCaps *audioresample_transform_caps (GstBaseTransform * base,
+ GstPadDirection direction, GstCaps * caps);
+gboolean audioresample_transform_size (GstBaseTransform * trans,
+ GstPadDirection direction, GstCaps * incaps, guint insize,
+ GstCaps * outcaps, guint * outsize);
+gboolean audioresample_set_caps (GstBaseTransform * base, GstCaps * incaps,
+ GstCaps * outcaps);
+static GstFlowReturn audioresample_pushthrough (GstAudioresample *
+ audioresample);
+static GstFlowReturn audioresample_transform (GstBaseTransform * base,
+ GstBuffer * inbuf, GstBuffer * outbuf);
+static gboolean audioresample_event (GstBaseTransform * base, GstEvent * event);
/*static guint gst_audioresample_signals[LAST_SIGNAL] = { 0 }; */
@@ -114,20 +132,21 @@
GST_BOILERPLATE_FULL (GstAudioresample, gst_audioresample, GstBaseTransform,
GST_TYPE_BASE_TRANSFORM, DEBUG_INIT);
- static void gst_audioresample_base_init (gpointer g_class)
- {
- GstElementClass *gstelement_class = GST_ELEMENT_CLASS (g_class);
+static void
+gst_audioresample_base_init (gpointer g_class)
+{
+ GstElementClass *gstelement_class = GST_ELEMENT_CLASS (g_class);
- gst_element_class_add_pad_template (gstelement_class,
- gst_static_pad_template_get (&gst_audioresample_src_template));
- gst_static_pad_template_get (&gst_audioresample_sink_template));
+ gst_element_class_add_pad_template (gstelement_class,
+ gst_static_pad_template_get (&gst_audioresample_src_template));
+ gst_static_pad_template_get (&gst_audioresample_sink_template));
- gst_element_class_set_details (gstelement_class,
- &gst_audioresample_details);
- }
+ gst_element_class_set_details (gstelement_class, &gst_audioresample_details);
+}
-static void gst_audioresample_class_init (GstAudioresampleClass * klass)
+gst_audioresample_class_init (GstAudioresampleClass * klass)
GObjectClass *gobject_class;
@@ -137,7 +156,7 @@
gobject_class->get_property = gst_audioresample_get_property;
gobject_class->dispose = gst_audioresample_dispose;
- g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_FILTERLEN,
+ g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_FILTERLEN,
g_param_spec_int ("filter_length", "filter_length", "filter_length",
0, G_MAXINT, DEFAULT_FILTERLEN,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
@@ -159,7 +178,7 @@
}
static void
- gst_audioresample_init (GstAudioresample * audioresample,
+gst_audioresample_init (GstAudioresample * audioresample,
GstAudioresampleClass * klass)
ResampleState *r;
@@ -181,7 +200,8 @@
resample_set_format (r, RESAMPLE_FORMAT_S16);
-static void gst_audioresample_dispose (GObject * object)
+gst_audioresample_dispose (GObject * object)
GstAudioresample *audioresample = GST_AUDIORESAMPLE (object);
@@ -195,8 +215,9 @@
gboolean
- audioresample_get_unit_size (GstBaseTransform * base, GstCaps * caps,
- guint * size) {
+audioresample_get_unit_size (GstBaseTransform * base, GstCaps * caps,
+ guint * size)
gint width, channels;
GstStructure *structure;
gboolean ret;
@@ -214,7 +235,8 @@
return TRUE;
-GstCaps *audioresample_transform_caps (GstBaseTransform * base,
+GstCaps *
+audioresample_transform_caps (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps)
GstCaps *res;
@@ -230,7 +252,7 @@
static gboolean
- resample_set_state_from_caps (ResampleState * state, GstCaps * incaps,
+resample_set_state_from_caps (ResampleState * state, GstCaps * incaps,
GstCaps * outcaps, gint * channels, gint * inrate, gint * outrate)
@@ -275,9 +297,10 @@
- audioresample_transform_size (GstBaseTransform * base,
+audioresample_transform_size (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps, guint size, GstCaps * othercaps,
- guint * othersize) {
+ guint * othersize)
GstAudioresample *audioresample = GST_AUDIORESAMPLE (base);
ResampleState *state;
GstCaps *srccaps, *sinkcaps;
@@ -329,8 +352,9 @@
- audioresample_set_caps (GstBaseTransform * base, GstCaps * incaps,
- GstCaps * outcaps) {
+audioresample_set_caps (GstBaseTransform * base, GstCaps * incaps,
+ GstCaps * outcaps)
gint inrate, outrate;
int channels;
@@ -362,7 +386,8 @@
-static gboolean audioresample_event (GstBaseTransform * base, GstEvent * event)
+static gboolean
+audioresample_event (GstBaseTransform * base, GstEvent * event)
GstAudioresample *audioresample;
@@ -397,8 +422,7 @@
static GstFlowReturn
- audioresample_do_output (GstAudioresample * audioresample,
- GstBuffer * outbuf)
+audioresample_do_output (GstAudioresample * audioresample, GstBuffer * outbuf)
int outsize;
int outsamples;
@@ -475,7 +499,7 @@
- audioresample_transform (GstBaseTransform * base, GstBuffer * inbuf,
+audioresample_transform (GstBaseTransform * base, GstBuffer * inbuf,
GstBuffer * outbuf)
@@ -522,7 +546,7 @@
/* push remaining data in the buffers out */
- audioresample_pushthrough (GstAudioresample * audioresample)
+audioresample_pushthrough (GstAudioresample * audioresample)
@@ -552,29 +576,30 @@
- gst_audioresample_set_property (GObject * object, guint prop_id,
+gst_audioresample_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
- g_return_if_fail (GST_IS_AUDIORESAMPLE (object));
- audioresample = GST_AUDIORESAMPLE (object);
+ g_return_if_fail (GST_IS_AUDIORESAMPLE (object));
+ audioresample = GST_AUDIORESAMPLE (object);
switch (prop_id) {
- case ARG_FILTERLEN:
+ case PROP_FILTERLEN:
audioresample->filter_length = g_value_get_int (value);
GST_DEBUG_OBJECT (GST_ELEMENT (audioresample), "new filter length %d",
audioresample->filter_length);
resample_set_filter_length (audioresample->resample,
break;
- default:G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
}
- gst_audioresample_get_property (GObject * object, guint prop_id,
+gst_audioresample_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
@@ -583,7 +608,7 @@
audioresample = GST_AUDIORESAMPLE (object);
g_value_set_int (value, audioresample->filter_length);
default:
@@ -593,7 +618,8 @@
-static gboolean plugin_init (GstPlugin * plugin)
+plugin_init (GstPlugin * plugin)
resample_init ();
Index: gstaudioresample.h
RCS file: /cvs/gstreamer/gst-plugins-base/gst/audioresample/gstaudioresample.h,v
retrieving revision 1.4
retrieving revision 1.5
diff -u -d -r1.4 -r1.5
--- gstaudioresample.h 2 Dec 2005 11:34:49 -0000 1.4
+++ gstaudioresample.h 2 Mar 2006 18:23:55 -0000 1.5
@@ -21,16 +21,13 @@
#ifndef __AUDIORESAMPLE_H__
#define __AUDIORESAMPLE_H__
-
#include <gst/gst.h>
#include <gst/base/gstbasetransform.h>
#include "resample.h"
G_BEGIN_DECLS
#define GST_TYPE_AUDIORESAMPLE \
(gst_audioresample_get_type())
#define GST_AUDIORESAMPLE(obj) \
@@ -45,6 +42,11 @@
typedef struct _GstAudioresample GstAudioresample;
typedef struct _GstAudioresampleClass GstAudioresampleClass;
+ * GstAudioresample:
+ * Opaque data structure.
struct _GstAudioresample {
GstBaseTransform element;
@@ -70,8 +72,6 @@
GType gst_audioresample_get_type(void);
G_END_DECLS
#endif /* __AUDIORESAMPLE_H__ */
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