gstreamer/ext/mas/massink.c

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/* 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 "massink.h"
#define BUFFER_SIZE 640
/* Signals and args */
enum {
/* FILL ME */
LAST_SIGNAL
};
enum {
ARG_0,
ARG_MUTE,
ARG_DEPTH,
ARG_CHANNELS,
ARG_RATE,
ARG_HOST,
};
static GstStaticPadTemplate sink_factory =
GST_STATIC_PAD_TEMPLATE (
"sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int")
);
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);
#define GST_TYPE_MASSINK_DEPTHS (gst_massink_depths_get_type())
static GType
gst_massink_depths_get_type (void)
{
static GType massink_depths_type = 0;
static GEnumValue massink_depths[] = {
{8, "8", "8 Bits"},
{16, "16", "16 Bits"},
{0, NULL, NULL},
};
if (!massink_depths_type) {
massink_depths_type = g_enum_register_static("GstMassinkDepths", massink_depths);
}
return massink_depths_type;
}
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 (
"Esound audio sink",
"Sink/Audio",
"Plays audio to a MAS server",
"Zeeshan Ali <zak147@yahoo.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",
TRUE,G_PARAM_READWRITE)); /* CHECKME */
g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_DEPTH,
g_param_spec_enum("depth","depth","depth",
GST_TYPE_MASSINK_DEPTHS,16,G_PARAM_READWRITE)); /* CHECKME! */
g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_RATE,
g_param_spec_int("frequency","frequency","frequency",
G_MININT,G_MAXINT,0,G_PARAM_READWRITE)); /* CHECKME */
g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_HOST,
g_param_spec_string("host","host","host",
NULL, 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)
{
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->format = 16;
massink->depth = 16;
massink->channels = 2;
massink->frequency = 44100;
massink->host = NULL;
}
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);
//gst_massink_close_audio (massink);
//return gst_massink_open_audio (massink);
return 1;
}
static GstPadLinkReturn
gst_massink_sinkconnect (GstPad *pad, const GstCaps *caps)
{
GstMassink *massink;
massink = GST_MASSINK (gst_pad_get_parent (pad));
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) {
GstClockID id = gst_clock_new_single_shot_id (massink->clock, GST_BUFFER_TIMESTAMP (buf));
GST_DEBUG ("massink: clock wait: %llu\n", GST_BUFFER_TIMESTAMP (buf));
gst_element_clock_wait (GST_ELEMENT (massink), id, NULL);
gst_clock_id_free (id);
}
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) > 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) {
g_print ("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;
case ARG_DEPTH:
massink->depth = g_value_get_enum (value);
gst_massink_sync_parms (massink);
break;
case ARG_CHANNELS:
massink->channels = g_value_get_enum (value);
gst_massink_sync_parms (massink);
break;
case ARG_RATE:
massink->frequency = g_value_get_int (value);
gst_massink_sync_parms (massink);
break;
case ARG_HOST:
if (massink->host != NULL) g_free(massink->host);
if (g_value_get_string (value) == NULL)
massink->host = NULL;
else
massink->host = g_strdup (g_value_get_string (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;
case ARG_DEPTH:
g_value_set_enum (value, massink->depth);
break;
case ARG_CHANNELS:
g_value_set_enum (value, massink->channels);
break;
case ARG_RATE:
g_value_set_int (value, massink->frequency);
break;
case ARG_HOST:
g_value_set_string (value, massink->host);
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;
char *ratestring = g_malloc (16);
char *bps = g_malloc (8);
struct mas_data_characteristic* dc;
g_print ("Connecting to MAS.\n");
masc_log_verbosity (MAS_VERBLVL_DEBUG);
err = mas_init();
if (err < 0) {
GST_DEBUG ("connection with local MAS server failed.");
exit (1);
}
GST_DEBUG ("Establishing audio output channel.");
mas_make_data_channel ("Gstreamer", &massink->audio_channel, &massink->audio_source, &massink->audio_sink);
mas_asm_get_port_by_name (0, "default_mix_sink", &massink->mix_sink);
GST_DEBUG ("Instantiating endian device.");
err = mas_asm_instantiate_device ("endian", 0, 0, &massink->endian);
if (err < 0) {
GST_DEBUG ("Failed to instantiate endian converter device");
exit(1);
}
mas_asm_get_port_by_name (massink->endian, "endian_sink", &massink->endian_sink);
mas_asm_get_port_by_name (massink->endian, "endian_source", &massink->endian_source);
sprintf (bps, "%u", massink->depth);
sprintf (ratestring, "%u", massink->frequency);
GST_DEBUG ("Connecting net -> endian.");
masc_make_dc (&dc, 6);
/* wav weirdness: 8 bit data is unsigned, >8 data is signed. */
masc_append_dc_key_value (dc, "format", (massink->depth==8) ? "ulinear":"linear");
masc_append_dc_key_value (dc, "resolution", bps);
masc_append_dc_key_value (dc, "sampling rate", ratestring);
masc_append_dc_key_value (dc, "channels", "2");
masc_append_dc_key_value (dc, "endian", "little");
err = mas_asm_connect_source_sink (massink->audio_source, massink->endian_sink, dc);
if ( err < 0 ) {
GST_DEBUG ("Failed to connect net audio output to endian");
return -1;
}
/* 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->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) {
GST_DEBUG ("Failed to instantiate squant device");
return -1;
}
mas_asm_get_port_by_name (massink->squant, "squant_sink", &massink->squant_sink);
mas_asm_get_port_by_name (massink->squant, "squant_source", &massink->squant_source);
GST_DEBUG ("Connecting endian -> squant.");
masc_make_dc (&dc, 6);
masc_append_dc_key_value (dc,"format",(massink->depth==8) ? "ulinear":"linear");
masc_append_dc_key_value (dc, "resolution", bps);
masc_append_dc_key_value (dc, "sampling rate", ratestring);
masc_append_dc_key_value (dc, "channels", "2");
masc_append_dc_key_value (dc, "endian", "host");
err = mas_asm_connect_source_sink (massink->endian_source, massink->squant_sink, dc);
if (err < 0) {
GST_DEBUG ("Failed to connect endian output to squant");
return -1;
}
/* 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) {
GST_DEBUG ("Failed to instantiate srate device");
return -1;
}
mas_asm_get_port_by_name (massink->srate, "sink", &massink->srate_sink);
mas_asm_get_port_by_name (massink->srate, "source", &massink->srate_source);
GST_DEBUG ("Connecting to srate.");
masc_make_dc (&dc, 6);
masc_append_dc_key_value (dc, "format", "linear");
masc_append_dc_key_value (dc, "resolution", "16");
masc_append_dc_key_value (dc, "sampling rate", ratestring);
masc_append_dc_key_value (dc, "channels", "2");
masc_append_dc_key_value (dc, "endian", "host");
err = mas_asm_connect_source_sink (massink->open_source, massink->srate_sink, dc);
if ( err < 0 ) {
GST_DEBUG ("Failed to connect to srate");
return -1;
}
massink->open_source = massink->srate_source;
}
GST_DEBUG ("Connecting to mix.");
masc_make_dc(&dc, 6);
masc_append_dc_key_value (dc, "format", "linear");
masc_append_dc_key_value (dc, "resolution", "16");
masc_append_dc_key_value (dc, "sampling rate", "44100");
masc_append_dc_key_value (dc, "channels", "2");
masc_append_dc_key_value (dc, "endian", "host");
err = mas_asm_connect_source_sink (massink->open_source, massink->mix_sink, dc);
if ( err < 0 ) {
GST_DEBUG ("Failed to connect to mixer");
return -1;
}
GST_FLAG_SET (massink, GST_MASSINK_OPEN);
masc_make_mas_data (&massink->data, BUFFER_SIZE);
massink->data->header.type = 10;
massink->data->header.media_timestamp = 0;
massink->data->header.sequence = 0;
return TRUE;
}
/*static void
gst_massink_close_audio (GstMassink *massink)
{
mas_free_device(massink->endian);
mas_free_device(massink->srate);
mas_free_device(massink->squant);
mas_free_port(massink->mix_sink);
mas_free_port(massink->srate_source);
mas_free_port(massink->srate_sink);
mas_free_port(massink->audio_source);
mas_free_port(massink->audio_sink);
mas_free_port(massink->endian_source);
mas_free_port(massink->endian_sink);
mas_free_port(massink->squant_source);
mas_free_port(massink->squant_sink);
mas_free_port(massink->open_source);
mas_free_channel (massink->audio_channel);
masc_destroy_mas_data (massink->data);
g_free (ratestring);
g_free (bps);
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 going down into NULL state, close the fd if it's open */
if (GST_STATE_PENDING (element) == GST_STATE_NULL) {
//if (GST_FLAG_IS_SET (element, GST_MASSINK_OPEN))
//gst_massink_close_audio (GST_MASSINK (element));
/* otherwise (READY or higher) we need to open the fd */
} else {
if (!GST_FLAG_IS_SET (element, GST_MASSINK_OPEN)) {
if (!gst_massink_open_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;
}