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Moving MAS plugin back to gst-sandbox module.

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Original commit message from CVS:
Moving MAS plugin back to gst-sandbox module.
This commit is contained in:
Zeeshan Ali 2004-09-10 00:41:12 +00:00
parent 514350e827
commit 978f5dbaee
5 changed files with 12 additions and 833 deletions
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14
configure.ac
View file

@ -1209,12 +1209,12 @@ AC_SUBST(LAME_CFLAGS)
AC_SUBST(LAME_LIBS)
dnl *** MAS (TODO) ***
translit(dnm, m, l) AM_CONDITIONAL(USE_MAS, true)
GST_CHECK_FEATURE(MAS, [mas], mas, [
GST_CHECK_CONFIGPROG(MAS, mas-config)
AC_SUBST(MAS_CFLAGS)
AC_SUBST(MAS_LIBS)
])
dnl translit(dnm, m, l) AM_CONDITIONAL(USE_MAS, true)
dnl GST_CHECK_FEATURE(MAS, [mas], mas, [
dnl GST_CHECK_CONFIGPROG(MAS, mas-config)
dnl AC_SUBST(MAS_CFLAGS)
dnl AC_SUBST(MAS_LIBS)
dnl ])
dnl *** libcaca ***
translit(dnm, m, l) AM_CONDITIONAL(USE_LIBCACA, true)
@ -1962,7 +1962,7 @@ ext/libpng/Makefile
ext/librfb/Makefile
ext/libvisual/Makefile
ext/mad/Makefile
ext/mas/Makefile
dnl ext/mas/Makefile
ext/mikmod/Makefile
ext/mpeg2dec/Makefile
ext/mpeg2enc/Makefile

12
ext/Makefile.am
View file

@ -226,11 +226,11 @@ else
MPLEX_DIR=
endif
if USE_MAS
MAS_DIR=mas
else
MAS_DIR=
endif
#if USE_MAS
#MAS_DIR=mas
#else
#MAS_DIR=
#endif
if USE_MUSICBRAINZ
MUSICBRAINZ_DIR=musicbrainz
@ -387,7 +387,6 @@ SUBDIRS=\
$(LIBRFB_DIR) \
$(LIBVISUAL_DIR) \
$(MAD_DIR) \
$(MAS_DIR) \
$(MIKMOD_DIR) \
$(MPEG2DEC_DIR) \
$(MPEG2ENC_DIR) \
@ -451,7 +450,6 @@ DIST_SUBDIRS=\
mpeg2dec \
mpeg2enc \
mplex \
mas \
musicbrainz \
nas \
ogg \

8
ext/mas/Makefile.am
View file

@ -1,8 +0,0 @@
plugin_LTLIBRARIES = libgstmassink.la
libgstmassink_la_SOURCES = gstmassink.c
libgstmassink_la_CFLAGS = $(GST_CFLAGS) $(MAS_CFLAGS)
libgstmassink_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) $(MAS_LIBS)
noinst_HEADERS = gstmassink.h

710
ext/mas/gstmassink.c
View file

@ -1,710 +0,0 @@
/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
*
* 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.
*/
/*
* Portions derived from maswavplay.c (distributed under the X11
* license):
*
* Copyright (c) 2001-2003 Shiman Associates Inc. All Rights Reserved.
* Copyright (c) 2000, 2001 by Shiman Associates Inc. and Sun
* Microsystems, Inc. All Rights Reserved.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstmassink.h"
/* Signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
ARG_0,
ARG_MUTE,
};
static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, " "endianness = (int) " G_STRINGIFY (G_BYTE_ORDER) ", " "signed = (boolean) TRUE, " /* We dont deal with unsigned creatures */
"width = (int) 16, "
"depth = (int) 16, "
"rate = [ 8000, 96000 ], "
"channels = [ 1, 2 ]; "
"audio/x-raw-int, "
"signed = (boolean) TRUE, "
"width = (int) 8, "
"depth = (int) 8, " "rate = [ 8000, 96000 ], " "channels = [ 1, 2 ]"));
static void gst_massink_base_init (gpointer g_class);
static void gst_massink_class_init (GstMassinkClass * klass);
static void gst_massink_init (GstMassink * massink);
static void gst_massink_set_clock (GstElement * element, GstClock * clock);
static gboolean gst_massink_open_audio (GstMassink * sink);
static void gst_massink_close_audio (GstMassink * sink);
static GstElementStateReturn gst_massink_change_state (GstElement * element);
static gboolean gst_massink_sync_parms (GstMassink * massink);
static GstPadLinkReturn gst_massink_sinkconnect (GstPad * pad,
const GstCaps * caps);
static void gst_massink_chain (GstPad * pad, GstData * _data);
static void gst_massink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_massink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static GstElementClass *parent_class = NULL;
/*static guint gst_massink_signals[LAST_SIGNAL] = { 0 }; */
GType
gst_massink_get_type (void)
{
static GType massink_type = 0;
if (!massink_type) {
static const GTypeInfo massink_info = {
sizeof (GstMassinkClass),
gst_massink_base_init,
NULL,
(GClassInitFunc) gst_massink_class_init,
NULL,
NULL,
sizeof (GstMassink),
0,
(GInstanceInitFunc) gst_massink_init,
};
massink_type =
g_type_register_static (GST_TYPE_ELEMENT, "GstMassink", &massink_info,
0);
}
return massink_type;
}
static void
gst_massink_base_init (gpointer g_class)
{
static GstElementDetails massink_details =
GST_ELEMENT_DETAILS ("MAS audio sink",
"Sink/Audio",
"Plays audio to a MAS server",
"Zeeshan Ali <zeenix@gmail.com>");
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sink_factory));
gst_element_class_set_details (element_class, &massink_details);
}
static void
gst_massink_class_init (GstMassinkClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_MUTE, g_param_spec_boolean ("mute", "mute", "mute", FALSE, G_PARAM_READWRITE)); /* CHECKME */
gobject_class->set_property = gst_massink_set_property;
gobject_class->get_property = gst_massink_get_property;
gstelement_class->change_state = gst_massink_change_state;
gstelement_class->set_clock = gst_massink_set_clock;
}
static void
gst_massink_set_clock (GstElement * element, GstClock * clock)
{
GstMassink *massink;
massink = GST_MASSINK (element);
massink->clock = clock;
}
static void
gst_massink_init (GstMassink * massink)
{
gint32 err;
guint32 from, to;
massink->sinkpad =
gst_pad_new_from_template (gst_static_pad_template_get (&sink_factory),
"sink");
gst_element_add_pad (GST_ELEMENT (massink), massink->sinkpad);
gst_pad_set_chain_function (massink->sinkpad,
GST_DEBUG_FUNCPTR (gst_massink_chain));
gst_pad_set_link_function (massink->sinkpad, gst_massink_sinkconnect);
massink->mute = FALSE;
massink->depth = MASSINK_DEFAULT_DEPTH;
massink->channels = MASSINK_DEFAULT_CHANNELS;
massink->frequency = MASSINK_DEFAULT_FREQUENCY;
/* MAS API connects to the MAS server on library initialization and
* there is no way of connecting/disconnecting with the server
* after the library initialization. So there is no point
* in having this code in gst_massink_open_audio
*/
GST_DEBUG ("Connecting to MAS server..\n");
masc_log_verbosity (MAS_VERBLVL_DEBUG);
err = mas_init ();
if (err < 0) {
GST_DEBUG ("Connection with MAS server failed.");
/* What else should/can I do to signal an error from an instance init
* function?
*/
exit (1);
}
/* Inititialize everything to 0 */
from = G_STRUCT_OFFSET (GstMassink, audio_channel);
to = G_STRUCT_OFFSET (GstMassink, data) + sizeof (massink->data);
memset ((gchar *) massink + from, 0, to - from);
}
static gboolean
gst_massink_sync_parms (GstMassink * massink)
{
g_return_val_if_fail (massink != NULL, FALSE);
g_return_val_if_fail (GST_IS_MASSINK (massink), FALSE);
if (GST_FLAG_IS_SET (GST_ELEMENT (massink), GST_MASSINK_OPEN)) {
gst_massink_close_audio (massink);
return gst_massink_open_audio (massink);
}
else {
return TRUE;
}
}
static GstPadLinkReturn
gst_massink_sinkconnect (GstPad * pad, const GstCaps * caps)
{
GstMassink *massink;
GstStructure *structure;
massink = GST_MASSINK (gst_pad_get_parent (pad));
structure = gst_caps_get_structure (caps, 0);
gst_structure_get_int (structure, "depth", &massink->depth);
gst_structure_get_int (structure, "endianness", &massink->endianness);
//gst_structure_get_boolean (structure, "signed", &massink->sign);
gst_structure_get_int (structure, "channels", &massink->channels);
gst_structure_get_int (structure, "rate", &massink->frequency);
if (gst_massink_sync_parms (massink)) {
return GST_PAD_LINK_OK;
}
return GST_PAD_LINK_REFUSED;
}
static void
gst_massink_chain (GstPad * pad, GstData * _data)
{
GstBuffer *buf = GST_BUFFER (_data);
gint32 err;
g_return_if_fail (pad != NULL);
g_return_if_fail (GST_IS_PAD (pad));
g_return_if_fail (buf != NULL);
GstMassink *massink = GST_MASSINK (gst_pad_get_parent (pad));
if (massink->clock && GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
GST_DEBUG ("massink: clock wait: %llu\n", GST_BUFFER_TIMESTAMP (buf));
gst_element_wait (GST_ELEMENT (massink), GST_BUFFER_TIMESTAMP (buf));
}
if (GST_BUFFER_DATA (buf) != NULL) {
if (!massink->mute) {
GST_DEBUG ("massink: data=%p size=%d", GST_BUFFER_DATA (buf),
GST_BUFFER_SIZE (buf));
if (GST_BUFFER_SIZE (buf) > MASSINK_BUFFER_SIZE) {
gst_buffer_unref (buf);
return;
}
massink->data.length = GST_BUFFER_SIZE (buf);
memcpy (massink->data.segment, GST_BUFFER_DATA (buf),
GST_BUFFER_SIZE (buf));
err = mas_send (massink->audio_channel, &massink->data);
if (err < 0) {
GST_DEBUG ("Error sending data to MAS server\n");
gst_buffer_unref (buf);
return;
}
/* FIXME: Please correct the Timestamping if its wrong */
massink->data.header.media_timestamp += massink->data.length / 4;
massink->data.header.sequence++;
}
}
gst_buffer_unref (buf);
}
static void
gst_massink_set_property (GObject * object, guint prop_id, const GValue * value,
GParamSpec * pspec)
{
GstMassink *massink;
/* it's not null if we got it, but it might not be ours */
g_return_if_fail (GST_IS_MASSINK (object));
massink = GST_MASSINK (object);
switch (prop_id) {
case ARG_MUTE:
massink->mute = g_value_get_boolean (value);
break;
default:
break;
}
}
static void
gst_massink_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstMassink *massink;
/* it's not null if we got it, but it might not be ours */
g_return_if_fail (GST_IS_MASSINK (object));
massink = GST_MASSINK (object);
switch (prop_id) {
case ARG_MUTE:
g_value_set_boolean (value, massink->mute);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static gboolean
plugin_init (GstPlugin * plugin)
{
if (!gst_element_register (plugin, "massink", GST_RANK_NONE,
GST_TYPE_MASSINK)) {
return FALSE;
}
return TRUE;
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"massink",
"uses MAS for audio output",
plugin_init, VERSION, "LGPL", GST_PACKAGE, GST_ORIGIN);
static gboolean
gst_massink_open_audio (GstMassink * massink)
{
gint32 err;
struct mas_data_characteristic *dc;
GST_DEBUG ("Establishing audio output channel.");
err =
mas_make_data_channel ("Gstreamer", &massink->audio_channel,
&massink->audio_source, &massink->audio_sink);
if (err < 0) {
massink->audio_channel = 0;
massink->audio_source = massink->audio_sink = 0;
GST_DEBUG ("Failed to create data channel");
return FALSE;
}
err = mas_asm_get_port_by_name (0, "default_mix_sink", &massink->mix_sink);
if (err < 0) {
GST_DEBUG ("Failed to get default_mix_sink");
massink->mix_sink = 0;
return FALSE;
}
GST_DEBUG ("Instantiating endian device.");
err = mas_asm_instantiate_device ("endian", 0, 0, &massink->endian);
if (err < 0) {
GST_DEBUG ("Failed to instantiate endian device");
massink->endian = 0;
return FALSE;
}
err =
mas_asm_get_port_by_name (massink->endian, "sink", &massink->endian_sink);
if (err < 0) {
GST_DEBUG ("Failed to get sink port from endian device");
massink->endian_sink = 0;
return FALSE;
}
err =
mas_asm_get_port_by_name (massink->endian, "source",
&massink->endian_source);
if (err < 0) {
GST_DEBUG ("Failed to get source port from endian device");
massink->endian_source = 0;
return FALSE;
}
GST_DEBUG ("Connecting net -> endian.");
dc = masc_make_audio_basic_dc (
(massink->depth == 8) ? MAS_ULINEAR_FMT : MAS_LINEAR_FMT,
massink->frequency,
massink->depth,
massink->channels,
(massink->endianness == G_LITTLE_ENDIAN) ? MAS_LITTLE_ENDIAN_FMT :
MAS_BIG_ENDIAN_FMT);
mas_assert (dc != 0, "Memory allocation error");
err =
mas_asm_connect_source_sink (massink->audio_source, massink->endian_sink,
dc);
masc_strike_dc (dc);
masc_rtfree (dc);
if (err < 0) {
GST_DEBUG ("Failed to connect Channel Converter output to endian");
return FALSE;
}
/* The next device is 'if needed' only. After the following if()
statement, open_source will contain the current unconnected
source in the path (will be either endian_source or
squant_source in this case)
*/
massink->open_source = massink->endian_source;
if (massink->channels != 2) {
GST_DEBUG ("Instantiating Channel Converter device.");
err =
mas_asm_instantiate_device ("channelconv", 0, 0, &massink->channelconv);
if (err < 0) {
GST_DEBUG ("Failed to instantiate Channel Converter device");
massink->channelconv = 0;
return FALSE;
}
err =
mas_asm_get_port_by_name (massink->channelconv, "sink",
&massink->channelconv_sink);
if (err < 0) {
GST_DEBUG ("Failed to get sink port from Channel Converter device");
massink->channelconv_sink = 0;
return FALSE;
}
err =
mas_asm_get_port_by_name (massink->channelconv, "source",
&massink->channelconv_source);
if (err < 0) {
GST_DEBUG ("Failed to get source port from Channel Converter device");
massink->channelconv_source = 0;
return FALSE;
}
GST_DEBUG ("Connecting audio -> channelconv.");
dc = masc_make_audio_basic_dc (
(massink->depth == 8) ? MAS_ULINEAR_FMT : MAS_LINEAR_FMT,
massink->frequency,
massink->depth, massink->channels, MAS_HOST_ENDIAN_FMT);
mas_assert (dc != 0, "Memory allocation error");
err =
mas_asm_connect_source_sink (massink->open_source,
massink->channelconv_sink, dc);
masc_strike_dc (dc);
masc_rtfree (dc);
if (err < 0) {
GST_DEBUG ("Failed to connect endian device to Channel converter");
return FALSE;
}
massink->open_source = massink->channelconv_source;
}
if (massink->depth != 16) {
GST_DEBUG
("Sample resolution is not 16 bit/sample, instantiating squant device.");
err = mas_asm_instantiate_device ("squant", 0, 0, &massink->squant);
if (err < 0) {
massink->squant = 0;
GST_DEBUG ("Failed to instantiate squant device");
return FALSE;
}
err =
mas_asm_get_port_by_name (massink->squant, "sink",
&massink->squant_sink);
if (err < 0) {
massink->squant_sink = 0;
GST_DEBUG ("Failed to get sink port from squant device");
return FALSE;
}
err =
mas_asm_get_port_by_name (massink->squant, "source",
&massink->squant_source);
if (err < 0) {
massink->squant_source = 0;
GST_DEBUG ("Failed to get source port from squant device");
return FALSE;
}
GST_DEBUG ("Connecting endian -> squant.");
dc = masc_make_audio_basic_dc (
(massink->depth == 8) ? MAS_ULINEAR_FMT : MAS_LINEAR_FMT,
massink->frequency, massink->depth, 2, MAS_HOST_ENDIAN_FMT);
mas_assert (dc != 0, "Memory allocation error");
err =
mas_asm_connect_source_sink (massink->open_source, massink->squant_sink,
dc);
masc_strike_dc (dc);
masc_rtfree (dc);
if (err < 0) {
GST_DEBUG ("Failed to connect endian output to squant");
return FALSE;
}
/* sneaky: the squant device is optional -> pretend it isn't there */
massink->open_source = massink->squant_source;
}
/* Another 'if necessary' device, as above */
if (massink->frequency != 44100) {
GST_DEBUG ("Sample rate is not 44100, instantiating srate device.");
err = mas_asm_instantiate_device ("srate", 0, 0, &massink->srate);
if (err < 0) {
massink->srate = 0;
GST_DEBUG ("Failed to instantiate srate device");
return FALSE;
}
mas_asm_get_port_by_name (massink->srate, "sink", &massink->srate_sink);
if (err < 0) {
massink->srate_sink = 0;
GST_DEBUG ("Failed to get sink port from srate device");
return FALSE;
}
mas_asm_get_port_by_name (massink->srate, "source", &massink->srate_source);
if (err < 0) {
massink->srate_source = 0;
GST_DEBUG ("Failed to get source port from srate device");
return FALSE;
}
GST_DEBUG ("Connecting to srate.");
dc = masc_make_audio_basic_dc (MAS_LINEAR_FMT,
massink->frequency, 16, 2, MAS_HOST_ENDIAN_FMT);
mas_assert (dc != 0, "Memory allocation error");
err =
mas_asm_connect_source_sink (massink->open_source, massink->srate_sink,
dc);
masc_strike_dc (dc);
masc_rtfree (dc);
if (err < 0) {
GST_DEBUG ("Failed to connect to srate");
return FALSE;
}
massink->open_source = massink->srate_source;
}
GST_DEBUG ("Connecting to mix.");
dc = masc_make_audio_basic_dc (MAS_LINEAR_FMT,
44100, 16, 2, MAS_HOST_ENDIAN_FMT);
mas_assert (dc != 0, "Memory allocation error");
err =
mas_asm_connect_source_sink (massink->open_source, massink->mix_sink, dc);
masc_strike_dc (dc);
masc_rtfree (dc);
if (err < 0) {
GST_DEBUG ("Failed to connect to mixer");
return FALSE;
}
massink->data.segment = masc_rtalloc (MASSINK_BUFFER_SIZE);
massink->data.length = MASSINK_BUFFER_SIZE;
massink->data.allocated_length = MASSINK_BUFFER_SIZE;
massink->data.header.type = 10;
massink->data.header.media_timestamp = 0;
massink->data.header.sequence = 0;
GST_FLAG_SET (massink, GST_MASSINK_OPEN);
return TRUE;
}
static void
gst_massink_close_audio (GstMassink * massink)
{
if (massink->mix_sink) {
mas_free_port (massink->mix_sink);
massink->mix_sink = 0;
}
if (massink->channelconv_source) {
mas_free_port (massink->channelconv_source);
massink->channelconv_source = 0;
}
if (massink->channelconv_sink) {
mas_free_port (massink->channelconv_sink);
massink->channelconv_sink = 0;
}
if (massink->srate_source) {
mas_free_port (massink->srate_source);
massink->srate_source = 0;
}
if (massink->srate_sink) {
mas_free_port (massink->srate_sink);
massink->srate_sink = 0;
}
if (massink->audio_source) {
mas_free_port (massink->audio_source);
massink->audio_source = 0;
}
if (massink->audio_sink) {
mas_free_port (massink->audio_sink);
massink->audio_sink = 0;
}
if (massink->endian_source) {
mas_free_port (massink->endian_source);
massink->endian_source = 0;
}
if (massink->endian_sink) {
mas_free_port (massink->endian_sink);
massink->endian_sink = 0;
}
if (massink->squant_source) {
mas_free_port (massink->squant_source);
massink->squant_source = 0;
}
if (massink->squant_sink) {
mas_free_port (massink->squant_sink);
massink->squant_sink = 0;
}
if (massink->channelconv) {
mas_free_device (massink->channelconv);
massink->channelconv = 0;
}
if (massink->endian) {
mas_free_device (massink->endian);
massink->endian = 0;
}
if (massink->srate) {
mas_free_device (massink->srate);
massink->srate = 0;
}
if (massink->squant) {
mas_free_device (massink->squant);
massink->squant = 0;
}
if (massink->audio_channel) {
mas_free_channel (massink->audio_channel);
massink->audio_channel = 0;
}
if (massink->data.segment) {
masc_rtfree (massink->data.segment);
memset (&massink->data, 0, sizeof (massink->data));
}
GST_FLAG_UNSET (massink, GST_MASSINK_OPEN);
GST_DEBUG ("massink: closed sound channel");
}
static GstElementStateReturn
gst_massink_change_state (GstElement * element)
{
g_return_val_if_fail (GST_IS_MASSINK (element), FALSE);
if (GST_STATE_PENDING (element) == GST_STATE_NULL) {
if (GST_FLAG_IS_SET (element, GST_MASSINK_OPEN))
gst_massink_close_audio (GST_MASSINK (element));
} else {
if (!GST_FLAG_IS_SET (element, GST_MASSINK_OPEN)) {
if (!gst_massink_open_audio (GST_MASSINK (element))) {
gst_massink_close_audio (GST_MASSINK (element));
return GST_STATE_FAILURE;
}
}
}
if (GST_ELEMENT_CLASS (parent_class)->change_state)
return GST_ELEMENT_CLASS (parent_class)->change_state (element);
return GST_STATE_SUCCESS;
}

101
ext/mas/gstmassink.h
View file

@ -1,101 +0,0 @@
/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
*
* 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.
*/
#ifndef __GST_MASSINK_H__
#define __GST_MASSINK_H__
#include <gst/gst.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <mas/mas.h>
#ifdef __cplusplus
extern "C" {
#endif /* __cplusplus */
#define GST_TYPE_MASSINK \
(gst_massink_get_type())
#define GST_MASSINK(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_MASSINK,GstMassink))
#define GST_MASSINK_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_MASSINK,GstMassink))
#define GST_IS_MASSINK(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_MASSINK))
#define GST_IS_MASSINK_CLASS(obj) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_MASSINK))
typedef enum {
GST_MASSINK_OPEN = GST_ELEMENT_FLAG_LAST,
GST_MASSINK_FLAG_LAST = GST_ELEMENT_FLAG_LAST+2,
} GstMasSinkFlags;
typedef struct _GstMassink GstMassink;
typedef struct _GstMassinkClass GstMassinkClass;
struct _GstMassink {
GstElement element;
GstPad *sinkpad;
gboolean mute;
gint depth;
gint channels;
gint frequency;
gint endianness;
/* MAS Data Structures */
mas_channel_t audio_channel;
mas_port_t mix_sink;
mas_port_t channelconv_source, channelconv_sink;
mas_port_t srate_source, srate_sink;
mas_port_t audio_source, audio_sink;
mas_port_t endian_sink, endian_source;
mas_port_t squant_sink, squant_source;
mas_port_t open_source; /* (!) */
mas_device_t channelconv;
mas_device_t endian;
mas_device_t srate;
mas_device_t squant;
struct mas_data data;
GstClock *clock;
};
struct _GstMassinkClass {
GstElementClass parent_class;
};
GType gst_massink_get_type(void);
#define MASSINK_DEFAULT_DEPTH 16
#define MASSINK_DEFAULT_CHANNELS 2
#define MASSINK_DEFAULT_FREQUENCY 44100
#define MASSINK_BUFFER_SIZE 10240
#ifdef __cplusplus
}
#endif /* __cplusplus */
#endif /* __GST_MASSINK_H__ */