/* 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., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, 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, APEX_PROP_GENERATION, APEX_PROP_TRANSPORT_PROTOCOL, }; #define DEFAULT_APEX_HOST "" #define DEFAULT_APEX_PORT 5000 #define DEFAULT_APEX_VOLUME 1.0 #define DEFAULT_APEX_JACK_TYPE GST_APEX_JACK_TYPE_UNDEFINED #define DEFAULT_APEX_JACK_STATUS GST_APEX_JACK_STATUS_UNDEFINED #define DEFAULT_APEX_GENERATION GST_APEX_GENERATION_ONE #define DEFAULT_APEX_TRANSPORT_PROTOCOL GST_APEX_TCP /* genum apex jack resolution */ GType gst_apexsink_jackstatus_get_type (void) { static GType jackstatus_type = 0; static const 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 const 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; } GType gst_apexsink_generation_get_type (void) { static GType generation_type = 0; static const GEnumValue generation[] = { {GST_APEX_GENERATION_ONE, "generation-one", "First generation (e.g., original AirPort Express)"}, {GST_APEX_GENERATION_TWO, "generation-two", "Second generation (e.g., Apple TV v2)"}, {0, NULL, NULL}, }; if (!generation_type) { generation_type = g_enum_register_static ("GstApExGeneration", generation); } return generation_type; } GType gst_apexsink_transport_protocol_get_type (void) { static GType transport_protocol_type = 0; static const GEnumValue transport_protocol[] = { {GST_APEX_TCP, "tcp", "TCP"}, {GST_APEX_UDP, "udp", "UDP"}, {0, NULL, NULL}, }; if (!transport_protocol_type) { transport_protocol_type = g_enum_register_static ("GstApExTransportProtocol", transport_protocol); } return transport_protocol_type; } 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); static GstStateChangeReturn gst_apexsink_change_state (GstElement * element, GstStateChange transition); /* 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 (GstMixerInterface * 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 (GstMixerInterface * 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_static_metadata (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); ((GstElementClass *) klass)->change_state = GST_DEBUG_FUNCPTR (gst_apexsink_change_state); 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_PARAM_STATIC_STRINGS)); 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_PARAM_STATIC_STRINGS)); /* we need to expose the volume as a double for playbin. Internally we keep * it as an int between 0 and 100, where 75 corresponds to 1.0. * FIXME we should store the volume as a double. */ g_object_class_install_property ((GObjectClass *) klass, APEX_PROP_VOLUME, g_param_spec_double ("volume", "Volume", "AirPort Express target volume", 0.0, 10.0, DEFAULT_APEX_VOLUME, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); 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_PARAM_STATIC_STRINGS)); 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 | G_PARAM_STATIC_STRINGS)); g_object_class_install_property ((GObjectClass *) klass, APEX_PROP_GENERATION, g_param_spec_enum ("generation", "Generation", "AirPort device generation", GST_APEX_SINK_GENERATION_TYPE, DEFAULT_APEX_GENERATION, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property ((GObjectClass *) klass, APEX_PROP_TRANSPORT_PROTOCOL, g_param_spec_enum ("transport-protocol", "Transport Protocol", "AirPort transport protocol", GST_APEX_SINK_TRANSPORT_PROTOCOL_TYPE, DEFAULT_APEX_TRANSPORT_PROTOCOL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); } /* 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 = CLAMP (DEFAULT_APEX_VOLUME * 75, 0, 100); apexsink->gst_apexraop = NULL; apexsink->tracks = g_list_append (apexsink->tracks, track); apexsink->clock = gst_system_clock_obtain (); apexsink->clock_id = NULL; 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: { gdouble volume; volume = g_value_get_double (value); volume *= 75.0; sink->volume = CLAMP (volume, 0, 100); 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; } case APEX_PROP_GENERATION: if (sink->gst_apexraop == NULL) { sink->generation = g_value_get_enum (value); GST_INFO_OBJECT (sink, "ApEx generation set to \"%d\"", sink->generation); } else { GST_WARNING_OBJECT (sink, "SET-PROPERTY : generation property may not be set when apexsink opened !"); } break; case APEX_PROP_TRANSPORT_PROTOCOL: if (sink->gst_apexraop == NULL) { sink->transport_protocol = g_value_get_enum (value); GST_INFO_OBJECT (sink, "ApEx transport protocol set to \"%d\"", sink->transport_protocol); } else { GST_WARNING_OBJECT (sink, "SET-PROPERTY : transport protocol property may not be set when apexsink opened !"); } 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_double (value, ((gdouble) sink->volume) / 75.0); 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; case APEX_PROP_GENERATION: g_value_set_enum (value, gst_apexraop_get_generation (sink->gst_apexraop)); break; case APEX_PROP_TRANSPORT_PROTOCOL: g_value_set_enum (value, gst_apexraop_get_transport_protocol (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; } gst_object_unref (sink->clock); 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, apexsink->generation, apexsink->transport_protocol); 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; GstApExGeneration gen = gst_apexraop_get_generation (apexsink->gst_apexraop); apexsink->latency_time = spec->latency_time; spec->segsize = gen == GST_APEX_GENERATION_ONE ? GST_APEX_RAOP_V1_SAMPLES_PER_FRAME * GST_APEX_RAOP_BYTES_PER_SAMPLE : GST_APEX_RAOP_V2_SAMPLES_PER_FRAME * GST_APEX_RAOP_BYTES_PER_SAMPLE; spec->segtotal = 2; 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) { guint written; GstApExSink *apexsink = (GstApExSink *) asink; if ((written = gst_apexraop_write (apexsink->gst_apexraop, data, length)) != length) { GST_INFO_OBJECT (apexsink, "WRITE : %d of %d bytes sent, skipping frame samples...", written, length); } else { GST_INFO_OBJECT (apexsink, "WRITE : %d bytes sent", length); /* NOTE, previous calculation subtracted apexsink->latency_time from this; * however, the value below is less than apexsink->latency_time for generation 2. * In this case, the number went negative (actualy wrapped around into a big number). */ apexsink->clock_id = gst_clock_new_single_shot_id (apexsink->clock, (GstClockTime) (gst_clock_get_time (apexsink->clock) + ((length * 1000000000.) / (GST_APEX_RAOP_BITRATE * GST_APEX_RAOP_BYTES_PER_SAMPLE)))); gst_clock_id_wait (apexsink->clock_id, NULL); gst_clock_id_unref (apexsink->clock_id); apexsink->clock_id = NULL; } 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; } static GstStateChangeReturn gst_apexsink_change_state (GstElement * element, GstStateChange transition) { GstApExSink *apexsink = (GstApExSink *) element; if (apexsink->clock_id && transition == GST_STATE_CHANGE_PAUSED_TO_READY) { gst_clock_id_unschedule (apexsink->clock_id); gst_clock_id_unref (apexsink->clock_id); apexsink->clock_id = NULL; } return GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); }