Commit af09257f authored by Wim Taymans's avatar Wim Taymans
Browse files

docs/plugins/: Add audioresample to docs.

Original commit message from CVS:
* 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.
parent c619495c
2006-03-02 Wim Taymans <wim@fluendo.com>
* 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 <wim@fluendo.com>
 
* docs/plugins/Makefile.am:
......
......@@ -89,6 +89,7 @@ EXTRA_HFILES = \
$(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 \
......
......@@ -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" />
......
......@@ -57,6 +57,20 @@ GST_TYPE_AUDIO_CONVERT
GstAudioConvertClass
</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>
......
......@@ -44,6 +44,8 @@
* generated by audiotestsrc.
* </para>
* </refsect2>
*
* Last reviewed on 2006-03-02 (0.10.4)
*/
/*
......
......@@ -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>
*
* Last reviewed on 2006-03-02 (0.10.4)
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
......@@ -52,8 +71,8 @@ enum
enum
{
ARG_0,
ARG_FILTERLEN
PROP_0,
PROP_FILTERLEN
};
#define SUPPORTED_CAPS \
......@@ -72,39 +91,38 @@ GST_STATIC_CAPS ( \
"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"
#endif
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_STATIC_CAPS ( \
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_element_class_add_pad_template (gstelement_class,
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_element_class_add_pad_template (gstelement_class,
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)
static void
gst_audioresample_class_init (GstAudioresampleClass * klass)
{
GObjectClass *gobject_class;
......@@ -137,7 +156,7 @@ static void gst_audioresample_class_init (GstAudioresampleClass * klass)
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_class_init (GstAudioresampleClass * klass)
}
static void
gst_audioresample_init (GstAudioresample * audioresample,
gst_audioresample_init (GstAudioresample * audioresample,
GstAudioresampleClass * klass)
{
ResampleState *r;
......@@ -181,7 +200,8 @@ static void
resample_set_format (r, RESAMPLE_FORMAT_S16);
}
static void gst_audioresample_dispose (GObject * object)
static void
gst_audioresample_dispose (GObject * object)
{
GstAudioresample *audioresample = GST_AUDIORESAMPLE (object);
......@@ -195,8 +215,9 @@ static void gst_audioresample_dispose (GObject * object)
/* vmethods */
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 @@ gboolean
return TRUE;
}
GstCaps *audioresample_transform_caps (GstBaseTransform * base,
GstCaps *
audioresample_transform_caps (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps)
{
GstCaps *res;
......@@ -230,7 +252,7 @@ GstCaps *audioresample_transform_caps (GstBaseTransform * base,
}
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)
{
GstStructure *structure;
......@@ -275,9 +297,10 @@ static gboolean
}
gboolean
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 @@ gboolean
}
gboolean
audioresample_set_caps (GstBaseTransform * base, GstCaps * incaps,
GstCaps * outcaps) {
audioresample_set_caps (GstBaseTransform * base, GstCaps * incaps,
GstCaps * outcaps)
{
gboolean ret;
gint inrate, outrate;
int channels;
......@@ -362,7 +386,8 @@ gboolean
return TRUE;
}
static gboolean audioresample_event (GstBaseTransform * base, GstEvent * event)
static gboolean
audioresample_event (GstBaseTransform * base, GstEvent * event)
{
GstAudioresample *audioresample;
......@@ -397,8 +422,7 @@ static gboolean audioresample_event (GstBaseTransform * base, GstEvent * event)
}
static GstFlowReturn
audioresample_do_output (GstAudioresample * audioresample,
GstBuffer * outbuf)
audioresample_do_output (GstAudioresample * audioresample, GstBuffer * outbuf)
{
int outsize;
int outsamples;
......@@ -475,7 +499,7 @@ static GstFlowReturn
}
static GstFlowReturn
audioresample_transform (GstBaseTransform * base, GstBuffer * inbuf,
audioresample_transform (GstBaseTransform * base, GstBuffer * inbuf,
GstBuffer * outbuf)
{
GstAudioresample *audioresample;
......@@ -522,7 +546,7 @@ static GstFlowReturn
/* push remaining data in the buffers out */
static GstFlowReturn
audioresample_pushthrough (GstAudioresample * audioresample)
audioresample_pushthrough (GstAudioresample * audioresample)
{
int outsize;
ResampleState *r;
......@@ -552,29 +576,30 @@ done:
static void
gst_audioresample_set_property (GObject * object, guint prop_id,
gst_audioresample_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAudioresample *audioresample;
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,
audioresample->filter_length);
break;
default:G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_audioresample_get_property (GObject * object, guint prop_id,
gst_audioresample_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstAudioresample *audioresample;
......@@ -583,7 +608,7 @@ static void
audioresample = GST_AUDIORESAMPLE (object);
switch (prop_id) {
case ARG_FILTERLEN:
case PROP_FILTERLEN:
g_value_set_int (value, audioresample->filter_length);
break;
default:
......@@ -593,7 +618,8 @@ static void
}
static gboolean plugin_init (GstPlugin * plugin)
static gboolean
plugin_init (GstPlugin * plugin)
{
resample_init ();
......
......@@ -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 @@ G_BEGIN_DECLS
typedef struct _GstAudioresample GstAudioresample;
typedef struct _GstAudioresampleClass GstAudioresampleClass;
/**
* GstAudioresample:
*
* Opaque data structure.
*/
struct _GstAudioresample {
GstBaseTransform element;
......@@ -70,8 +72,6 @@ struct _GstAudioresampleClass {
GType gst_audioresample_get_type(void);
G_END_DECLS
#endif /* __AUDIORESAMPLE_H__ */
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