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gstreamer/ext/apexsink/gstapexsink.c
Tim-Philipp Müller 6eb167a176 Make build without warnings with debugging disabled
2009-06-19 16:21:28 +01:00

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/* GStreamer - AirPort Express Audio Sink -
*
* Remote Audio Access Protocol (RAOP) as used in Apple iTunes to stream music to the Airport Express (ApEx) -
* RAOP is based on the Real Time Streaming Protocol (RTSP) but with an extra challenge-response RSA based authentication step.
*
* RAW PCM input only as defined by the following GST_STATIC_PAD_TEMPLATE
*
* Copyright (C) 2008 Jérémie Bernard [GRemi] <gremimail@gmail.com>
*
* gstapexsink.c
*
* 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.
*
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include "gstapexsink.h"
GST_DEBUG_CATEGORY_STATIC (apexsink_debug);
#define GST_CAT_DEFAULT apexsink_debug
static GstStaticPadTemplate gst_apexsink_sink_factory = GST_STATIC_PAD_TEMPLATE
("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS
(GST_APEX_RAOP_INPUT_TYPE ","
"width = (int) " GST_APEX_RAOP_INPUT_WIDTH ","
"depth = (int) " GST_APEX_RAOP_INPUT_DEPTH ","
"endianness = (int) " GST_APEX_RAOP_INPUT_ENDIAN ","
"channels = (int) " GST_APEX_RAOP_INPUT_CHANNELS ","
"rate = (int) " GST_APEX_RAOP_INPUT_BIT_RATE ","
"signed = (boolean) " GST_APEX_RAOP_INPUT_SIGNED)
);
enum
{
APEX_PROP_HOST = 1,
APEX_PROP_PORT,
APEX_PROP_VOLUME,
APEX_PROP_JACK_TYPE,
APEX_PROP_JACK_STATUS,
};
#define DEFAULT_APEX_HOST ""
#define DEFAULT_APEX_PORT 5000
#define DEFAULT_APEX_VOLUME 75
#define DEFAULT_APEX_JACK_TYPE GST_APEX_JACK_TYPE_UNDEFINED
#define DEFAULT_APEX_JACK_STATUS GST_APEX_JACK_STATUS_UNDEFINED
/* genum apex jack resolution */
GType
gst_apexsink_jackstatus_get_type (void)
{
static GType jackstatus_type = 0;
static GEnumValue jackstatus[] = {
{GST_APEX_JACK_STATUS_UNDEFINED, "GST_APEX_JACK_STATUS_UNDEFINED",
"Jack status undefined"},
{GST_APEX_JACK_STATUS_DISCONNECTED, "GST_APEX_JACK_STATUS_DISCONNECTED",
"Jack disconnected"},
{GST_APEX_JACK_STATUS_CONNECTED, "GST_APEX_JACK_STATUS_CONNECTED",
"Jack connected"},
{0, NULL, NULL},
};
if (!jackstatus_type) {
jackstatus_type = g_enum_register_static ("GstApExJackStatus", jackstatus);
}
return jackstatus_type;
}
GType
gst_apexsink_jacktype_get_type (void)
{
static GType jacktype_type = 0;
static GEnumValue jacktype[] = {
{GST_APEX_JACK_TYPE_UNDEFINED, "GST_APEX_JACK_TYPE_UNDEFINED",
"Undefined jack type"},
{GST_APEX_JACK_TYPE_ANALOG, "GST_APEX_JACK_TYPE_ANALOG", "Analog jack"},
{GST_APEX_JACK_TYPE_DIGITAL, "GST_APEX_JACK_TYPE_DIGITAL", "Digital jack"},
{0, NULL, NULL},
};
if (!jacktype_type) {
jacktype_type = g_enum_register_static ("GstApExJackType", jacktype);
}
return jacktype_type;
}
static void gst_apexsink_base_init (gpointer g_class);
static void gst_apexsink_class_init (GstApExSinkClass * klass);
static void gst_apexsink_init (GstApExSink * apexsink,
GstApExSinkClass * g_class);
static void gst_apexsink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_apexsink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void gst_apexsink_finalise (GObject * object);
static gboolean gst_apexsink_open (GstAudioSink * asink);
static gboolean gst_apexsink_prepare (GstAudioSink * asink,
GstRingBufferSpec * spec);
static guint gst_apexsink_write (GstAudioSink * asink, gpointer data,
guint length);
static gboolean gst_apexsink_unprepare (GstAudioSink * asink);
static guint gst_apexsink_delay (GstAudioSink * asink);
static void gst_apexsink_reset (GstAudioSink * asink);
static gboolean gst_apexsink_close (GstAudioSink * asink);
/* mixer interface standard api */
static void gst_apexsink_interfaces_init (GType type);
static void gst_apexsink_implements_interface_init (GstImplementsInterfaceClass
* iface);
static void gst_apexsink_mixer_interface_init (GstMixerClass * iface);
static gboolean gst_apexsink_interface_supported (GstImplementsInterface *
iface, GType iface_type);
static const GList *gst_apexsink_mixer_list_tracks (GstMixer * mixer);
static void gst_apexsink_mixer_set_volume (GstMixer * mixer,
GstMixerTrack * track, gint * volumes);
static void gst_apexsink_mixer_get_volume (GstMixer * mixer,
GstMixerTrack * track, gint * volumes);
GST_BOILERPLATE_FULL (GstApExSink, gst_apexsink, GstAudioSink,
GST_TYPE_AUDIO_SINK, gst_apexsink_interfaces_init);
/* apex sink interface(s) stuff */
static void
gst_apexsink_interfaces_init (GType type)
{
static const GInterfaceInfo implements_interface_info =
{ (GInterfaceInitFunc) gst_apexsink_implements_interface_init, NULL,
NULL
};
static const GInterfaceInfo mixer_interface_info =
{ (GInterfaceInitFunc) gst_apexsink_mixer_interface_init, NULL, NULL };
g_type_add_interface_static (type, GST_TYPE_IMPLEMENTS_INTERFACE,
&implements_interface_info);
g_type_add_interface_static (type, GST_TYPE_MIXER, &mixer_interface_info);
}
static void
gst_apexsink_implements_interface_init (GstImplementsInterfaceClass * iface)
{
iface->supported = gst_apexsink_interface_supported;
}
static void
gst_apexsink_mixer_interface_init (GstMixerClass * iface)
{
GST_MIXER_TYPE (iface) = GST_MIXER_SOFTWARE;
iface->list_tracks = gst_apexsink_mixer_list_tracks;
iface->set_volume = gst_apexsink_mixer_set_volume;
iface->get_volume = gst_apexsink_mixer_get_volume;
}
static gboolean
gst_apexsink_interface_supported (GstImplementsInterface * iface,
GType iface_type)
{
g_return_val_if_fail (iface_type == GST_TYPE_MIXER, FALSE);
return TRUE;
}
static const GList *
gst_apexsink_mixer_list_tracks (GstMixer * mixer)
{
GstApExSink *apexsink = GST_APEX_SINK (mixer);
return apexsink->tracks;
}
static void
gst_apexsink_mixer_set_volume (GstMixer * mixer, GstMixerTrack * track,
gint * volumes)
{
GstApExSink *apexsink = GST_APEX_SINK (mixer);
apexsink->volume = volumes[0];
if (apexsink->gst_apexraop != NULL)
gst_apexraop_set_volume (apexsink->gst_apexraop, apexsink->volume);
}
static void
gst_apexsink_mixer_get_volume (GstMixer * mixer, GstMixerTrack * track,
gint * volumes)
{
GstApExSink *apexsink = GST_APEX_SINK (mixer);
volumes[0] = apexsink->volume;
}
/* sink base init */
static void
gst_apexsink_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_set_details_simple (element_class,
"Apple AirPort Express Audio Sink", "Sink/Audio/Wireless",
"Output stream to an AirPort Express",
"Jérémie Bernard [GRemi] <gremimail@gmail.com>");
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_apexsink_sink_factory));
}
/* sink class init */
static void
gst_apexsink_class_init (GstApExSinkClass * klass)
{
GST_DEBUG_CATEGORY_INIT (apexsink_debug, GST_APEX_SINK_NAME, 0,
"AirPort Express sink");
parent_class = g_type_class_peek_parent (klass);
((GObjectClass *) klass)->get_property =
GST_DEBUG_FUNCPTR (gst_apexsink_get_property);
((GObjectClass *) klass)->set_property =
GST_DEBUG_FUNCPTR (gst_apexsink_set_property);
((GObjectClass *) klass)->finalize =
GST_DEBUG_FUNCPTR (gst_apexsink_finalise);
((GstAudioSinkClass *) klass)->open = GST_DEBUG_FUNCPTR (gst_apexsink_open);
((GstAudioSinkClass *) klass)->prepare =
GST_DEBUG_FUNCPTR (gst_apexsink_prepare);
((GstAudioSinkClass *) klass)->write = GST_DEBUG_FUNCPTR (gst_apexsink_write);
((GstAudioSinkClass *) klass)->unprepare =
GST_DEBUG_FUNCPTR (gst_apexsink_unprepare);
((GstAudioSinkClass *) klass)->delay = GST_DEBUG_FUNCPTR (gst_apexsink_delay);
((GstAudioSinkClass *) klass)->reset = GST_DEBUG_FUNCPTR (gst_apexsink_reset);
((GstAudioSinkClass *) klass)->close = GST_DEBUG_FUNCPTR (gst_apexsink_close);
g_object_class_install_property ((GObjectClass *) klass, APEX_PROP_HOST,
g_param_spec_string ("host", "Host", "AirPort Express target host",
DEFAULT_APEX_HOST, G_PARAM_READWRITE));
g_object_class_install_property ((GObjectClass *) klass, APEX_PROP_PORT,
g_param_spec_uint ("port", "Port", "AirPort Express target port", 0,
32000, DEFAULT_APEX_PORT, G_PARAM_READWRITE));
g_object_class_install_property ((GObjectClass *) klass, APEX_PROP_VOLUME,
g_param_spec_uint ("volume", "Volume", "AirPort Express target volume", 0,
100, DEFAULT_APEX_VOLUME, G_PARAM_READWRITE));
g_object_class_install_property ((GObjectClass *) klass, APEX_PROP_JACK_TYPE,
g_param_spec_enum ("jack_type", "Jack Type",
"AirPort Express connected jack type", GST_APEX_SINK_JACKTYPE_TYPE,
DEFAULT_APEX_JACK_TYPE, G_PARAM_READABLE));
g_object_class_install_property ((GObjectClass *) klass,
APEX_PROP_JACK_STATUS, g_param_spec_enum ("jack_status", "Jack Status",
"AirPort Express jack connection status",
GST_APEX_SINK_JACKSTATUS_TYPE, DEFAULT_APEX_JACK_STATUS,
G_PARAM_READABLE));
}
/* sink plugin instance init */
static void
gst_apexsink_init (GstApExSink * apexsink, GstApExSinkClass * g_class)
{
GstMixerTrack *track = NULL;
track = g_object_new (GST_TYPE_MIXER_TRACK, NULL);
track->label = g_strdup ("Airport Express");
track->num_channels = GST_APEX_RAOP_CHANNELS;
track->min_volume = 0;
track->max_volume = 100;
track->flags = GST_MIXER_TRACK_OUTPUT;
apexsink->host = g_strdup (DEFAULT_APEX_HOST);
apexsink->port = DEFAULT_APEX_PORT;
apexsink->volume = DEFAULT_APEX_VOLUME;
apexsink->gst_apexraop = NULL;
apexsink->tracks = g_list_append (apexsink->tracks, track);
GST_INFO_OBJECT (apexsink,
"ApEx sink default initialization, target=\"%s\", port=\"%d\", volume=\"%d%%\"",
apexsink->host, apexsink->port, apexsink->volume);
}
/* apex sink set property */
static void
gst_apexsink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstApExSink *sink = GST_APEX_SINK (object);
switch (prop_id) {
case APEX_PROP_HOST:
{
if (sink->gst_apexraop == NULL) {
g_free (sink->host);
sink->host = g_value_dup_string (value);
GST_INFO_OBJECT (sink, "ApEx sink target set to \"%s\"", sink->host);
} else
G_OBJECT_WARN_INVALID_PSPEC (object, "host", prop_id, pspec);
}
break;
case APEX_PROP_PORT:
{
if (sink->gst_apexraop == NULL) {
sink->port = g_value_get_uint (value);
GST_INFO_OBJECT (sink, "ApEx port set to \"%d\"", sink->port);
} else
G_OBJECT_WARN_INVALID_PSPEC (object, "port", prop_id, pspec);
}
break;
case APEX_PROP_VOLUME:
{
sink->volume = g_value_get_uint (value);
if (sink->gst_apexraop != NULL)
gst_apexraop_set_volume (sink->gst_apexraop, sink->volume);
GST_INFO_OBJECT (sink, "ApEx volume set to \"%d%%\"", sink->volume);
}
break;
default:
{
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
}
break;
}
}
/* apex sink get property */
static void
gst_apexsink_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstApExSink *sink = GST_APEX_SINK (object);
switch (prop_id) {
case APEX_PROP_HOST:
{
g_value_set_string (value, sink->host);
}
break;
case APEX_PROP_PORT:
{
g_value_set_uint (value, sink->port);
}
break;
case APEX_PROP_VOLUME:
{
g_value_set_uint (value, sink->volume);
}
break;
case APEX_PROP_JACK_TYPE:
{
g_value_set_enum (value, gst_apexraop_get_jacktype (sink->gst_apexraop));
}
break;
case APEX_PROP_JACK_STATUS:
{
g_value_set_enum (value,
gst_apexraop_get_jackstatus (sink->gst_apexraop));
}
break;
default:
{
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
}
break;
}
}
/* apex sink finalize */
static void
gst_apexsink_finalise (GObject * object)
{
GstApExSink *sink = GST_APEX_SINK (object);
if (sink->tracks) {
g_list_foreach (sink->tracks, (GFunc) g_object_unref, NULL);
g_list_free (sink->tracks);
sink->tracks = NULL;
}
g_free (sink->host);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
/* sink open : open the device */
static gboolean
gst_apexsink_open (GstAudioSink * asink)
{
int res;
GstApExSink *apexsink = (GstApExSink *) asink;
apexsink->gst_apexraop = gst_apexraop_new (apexsink->host, apexsink->port);
if ((res = gst_apexraop_connect (apexsink->gst_apexraop)) != GST_RTSP_STS_OK) {
GST_ERROR_OBJECT (apexsink,
"%s : network or RAOP failure, connection refused or timeout, RTSP code=%d",
apexsink->host, res);
return FALSE;
}
GST_INFO_OBJECT (apexsink,
"OPEN : ApEx sink successfully connected to \"%s:%d\", ANNOUNCE, SETUP and RECORD requests performed",
apexsink->host, apexsink->port);
switch (gst_apexraop_get_jackstatus (apexsink->gst_apexraop)) {
case GST_APEX_JACK_STATUS_CONNECTED:
{
GST_INFO_OBJECT (apexsink, "OPEN : ApEx jack is connected");
}
break;
case GST_APEX_JACK_STATUS_DISCONNECTED:
{
GST_WARNING_OBJECT (apexsink, "OPEN : ApEx jack is disconnected !");
}
break;
default:
{
GST_WARNING_OBJECT (apexsink, "OPEN : ApEx jack status is undefined !");
}
break;
}
switch (gst_apexraop_get_jacktype (apexsink->gst_apexraop)) {
case GST_APEX_JACK_TYPE_ANALOG:
{
GST_INFO_OBJECT (apexsink, "OPEN : ApEx jack type is analog");
}
break;
case GST_APEX_JACK_TYPE_DIGITAL:
{
GST_INFO_OBJECT (apexsink, "OPEN : ApEx jack type is digital");
}
break;
default:
{
GST_WARNING_OBJECT (apexsink, "OPEN : ApEx jack type is undefined !");
}
break;
}
if ((res =
gst_apexraop_set_volume (apexsink->gst_apexraop,
apexsink->volume)) != GST_RTSP_STS_OK) {
GST_WARNING_OBJECT (apexsink,
"%s : could not set initial volume to \"%d%%\", RTSP code=%d",
apexsink->host, apexsink->volume, res);
} else {
GST_INFO_OBJECT (apexsink,
"OPEN : ApEx sink successfully set volume to \"%d%%\"",
apexsink->volume);
}
return TRUE;
}
/* prepare sink : configure the device with the specified format */
static gboolean
gst_apexsink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec)
{
GstApExSink *apexsink = (GstApExSink *) asink;
apexsink->latency_time = spec->latency_time;
spec->segsize =
GST_APEX_RAOP_SAMPLES_PER_FRAME * GST_APEX_RAOP_BYTES_PER_SAMPLE;
spec->segtotal = 1;
memset (spec->silence_sample, 0, sizeof (spec->silence_sample));
GST_INFO_OBJECT (apexsink,
"PREPARE : ApEx sink ready to stream at %dHz, %d bytes per sample, %d channels, %d bytes segments (%dkB/s)",
spec->rate, spec->bytes_per_sample, spec->channels, spec->segsize,
spec->rate * spec->bytes_per_sample / 1000);
return TRUE;
}
/* sink write : write samples to the device */
static guint
gst_apexsink_write (GstAudioSink * asink, gpointer data, guint length)
{
GstApExSink *apexsink = (GstApExSink *) asink;
if (gst_apexraop_write (apexsink->gst_apexraop, data, length) != length) {
GST_INFO_OBJECT (apexsink,
"WRITE : %d bytes not fully sended, skipping frame samples...", length);
} else {
GST_INFO_OBJECT (apexsink, "WRITE : %d bytes sended", length);
usleep ((gulong) ((length * 1000000.) / (GST_APEX_RAOP_BITRATE *
GST_APEX_RAOP_BYTES_PER_SAMPLE) - apexsink->latency_time));
}
return length;
}
/* unprepare sink : undo operations done by prepare */
static gboolean
gst_apexsink_unprepare (GstAudioSink * asink)
{
GST_INFO_OBJECT (asink, "UNPREPARE");
return TRUE;
}
/* delay sink : get the estimated number of samples written but not played yet by the device */
static guint
gst_apexsink_delay (GstAudioSink * asink)
{
GST_LOG_OBJECT (asink, "DELAY");
return 0;
}
/* reset sink : unblock writes and flush the device */
static void
gst_apexsink_reset (GstAudioSink * asink)
{
int res;
GstApExSink *apexsink = (GstApExSink *) asink;
GST_INFO_OBJECT (apexsink, "RESET : flushing buffer...");
if ((res = gst_apexraop_flush (apexsink->gst_apexraop)) == GST_RTSP_STS_OK) {
GST_INFO_OBJECT (apexsink, "RESET : ApEx buffer flush success");
} else {
GST_WARNING_OBJECT (apexsink,
"RESET : could not flush ApEx buffer, RTSP code=%d", res);
}
}
/* sink close : close the device */
static gboolean
gst_apexsink_close (GstAudioSink * asink)
{
GstApExSink *apexsink = (GstApExSink *) asink;
gst_apexraop_close (apexsink->gst_apexraop);
gst_apexraop_free (apexsink->gst_apexraop);
GST_INFO_OBJECT (apexsink, "CLOSE : ApEx sink closed connection");
return TRUE;
}
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