mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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665 lines
22 KiB
C
665 lines
22 KiB
C
/* GStreamer - AirPort Express Audio Sink -
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*
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* Remote Audio Access Protocol (RAOP) as used in Apple iTunes to stream music to the Airport Express (ApEx) -
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* RAOP is based on the Real Time Streaming Protocol (RTSP) but with an extra challenge-response RSA based authentication step.
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*
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* RAW PCM input only as defined by the following GST_STATIC_PAD_TEMPLATE
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*
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* Copyright (C) 2008 Jérémie Bernard [GRemi] <gremimail@gmail.com>
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*
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* gstapexsink.c
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <string.h>
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#include "gstapexsink.h"
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GST_DEBUG_CATEGORY_STATIC (apexsink_debug);
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#define GST_CAT_DEFAULT apexsink_debug
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static GstStaticPadTemplate gst_apexsink_sink_factory = GST_STATIC_PAD_TEMPLATE
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("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS
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(GST_APEX_RAOP_INPUT_TYPE ","
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"width = (int) " GST_APEX_RAOP_INPUT_WIDTH ","
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"depth = (int) " GST_APEX_RAOP_INPUT_DEPTH ","
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"endianness = (int) " GST_APEX_RAOP_INPUT_ENDIAN ","
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"channels = (int) " GST_APEX_RAOP_INPUT_CHANNELS ","
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"rate = (int) " GST_APEX_RAOP_INPUT_BIT_RATE ","
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"signed = (boolean) " GST_APEX_RAOP_INPUT_SIGNED)
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);
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enum
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{
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APEX_PROP_HOST = 1,
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APEX_PROP_PORT,
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APEX_PROP_VOLUME,
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APEX_PROP_JACK_TYPE,
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APEX_PROP_JACK_STATUS,
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APEX_PROP_GENERATION,
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APEX_PROP_TRANSPORT_PROTOCOL,
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};
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#define DEFAULT_APEX_HOST ""
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#define DEFAULT_APEX_PORT 5000
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#define DEFAULT_APEX_VOLUME 1.0
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#define DEFAULT_APEX_JACK_TYPE GST_APEX_JACK_TYPE_UNDEFINED
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#define DEFAULT_APEX_JACK_STATUS GST_APEX_JACK_STATUS_UNDEFINED
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#define DEFAULT_APEX_GENERATION GST_APEX_GENERATION_ONE
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#define DEFAULT_APEX_TRANSPORT_PROTOCOL GST_APEX_TCP
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/* genum apex jack resolution */
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GType
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gst_apexsink_jackstatus_get_type (void)
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{
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static GType jackstatus_type = 0;
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static GEnumValue jackstatus[] = {
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{GST_APEX_JACK_STATUS_UNDEFINED, "GST_APEX_JACK_STATUS_UNDEFINED",
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"Jack status undefined"},
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{GST_APEX_JACK_STATUS_DISCONNECTED, "GST_APEX_JACK_STATUS_DISCONNECTED",
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"Jack disconnected"},
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{GST_APEX_JACK_STATUS_CONNECTED, "GST_APEX_JACK_STATUS_CONNECTED",
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"Jack connected"},
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{0, NULL, NULL},
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};
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if (!jackstatus_type) {
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jackstatus_type = g_enum_register_static ("GstApExJackStatus", jackstatus);
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}
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return jackstatus_type;
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}
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GType
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gst_apexsink_jacktype_get_type (void)
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{
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static GType jacktype_type = 0;
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static GEnumValue jacktype[] = {
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{GST_APEX_JACK_TYPE_UNDEFINED, "GST_APEX_JACK_TYPE_UNDEFINED",
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"Undefined jack type"},
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{GST_APEX_JACK_TYPE_ANALOG, "GST_APEX_JACK_TYPE_ANALOG", "Analog jack"},
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{GST_APEX_JACK_TYPE_DIGITAL, "GST_APEX_JACK_TYPE_DIGITAL", "Digital jack"},
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{0, NULL, NULL},
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};
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if (!jacktype_type) {
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jacktype_type = g_enum_register_static ("GstApExJackType", jacktype);
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}
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return jacktype_type;
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}
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GType
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gst_apexsink_generation_get_type (void)
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{
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static GType generation_type = 0;
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static GEnumValue generation[] = {
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{GST_APEX_GENERATION_ONE, "generation-one",
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"First generation (e.g., original AirPort Express)"},
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{GST_APEX_GENERATION_TWO, "generation-two",
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"Second generation (e.g., Apple TV v2)"},
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{0, NULL, NULL},
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};
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if (!generation_type) {
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generation_type = g_enum_register_static ("GstApExGeneration", generation);
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}
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return generation_type;
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}
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GType
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gst_apexsink_transport_protocol_get_type (void)
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{
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static GType transport_protocol_type = 0;
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static GEnumValue transport_protocol[] = {
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{GST_APEX_TCP, "tcp", "TCP"},
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{GST_APEX_UDP, "udp", "UDP"},
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{0, NULL, NULL},
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};
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if (!transport_protocol_type) {
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transport_protocol_type =
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g_enum_register_static ("GstApExTransportProtocol", transport_protocol);
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}
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return transport_protocol_type;
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}
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static void gst_apexsink_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_apexsink_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static void gst_apexsink_finalise (GObject * object);
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static gboolean gst_apexsink_open (GstAudioSink * asink);
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static gboolean gst_apexsink_prepare (GstAudioSink * asink,
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GstRingBufferSpec * spec);
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static guint gst_apexsink_write (GstAudioSink * asink, gpointer data,
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guint length);
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static gboolean gst_apexsink_unprepare (GstAudioSink * asink);
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static guint gst_apexsink_delay (GstAudioSink * asink);
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static void gst_apexsink_reset (GstAudioSink * asink);
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static gboolean gst_apexsink_close (GstAudioSink * asink);
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static GstStateChangeReturn gst_apexsink_change_state (GstElement * element,
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GstStateChange transition);
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/* mixer interface standard api */
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static void gst_apexsink_interfaces_init (GType type);
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static void gst_apexsink_implements_interface_init (GstImplementsInterfaceClass
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* iface);
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static void gst_apexsink_mixer_interface_init (GstMixerInterface * iface);
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static gboolean gst_apexsink_interface_supported (GstImplementsInterface *
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iface, GType iface_type);
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static const GList *gst_apexsink_mixer_list_tracks (GstMixer * mixer);
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static void gst_apexsink_mixer_set_volume (GstMixer * mixer,
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GstMixerTrack * track, gint * volumes);
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static void gst_apexsink_mixer_get_volume (GstMixer * mixer,
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GstMixerTrack * track, gint * volumes);
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GST_BOILERPLATE_FULL (GstApExSink, gst_apexsink, GstAudioSink,
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GST_TYPE_AUDIO_SINK, gst_apexsink_interfaces_init);
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/* apex sink interface(s) stuff */
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static void
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gst_apexsink_interfaces_init (GType type)
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{
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static const GInterfaceInfo implements_interface_info =
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{ (GInterfaceInitFunc) gst_apexsink_implements_interface_init, NULL,
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NULL
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};
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static const GInterfaceInfo mixer_interface_info =
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{ (GInterfaceInitFunc) gst_apexsink_mixer_interface_init, NULL, NULL };
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g_type_add_interface_static (type, GST_TYPE_IMPLEMENTS_INTERFACE,
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&implements_interface_info);
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g_type_add_interface_static (type, GST_TYPE_MIXER, &mixer_interface_info);
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}
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static void
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gst_apexsink_implements_interface_init (GstImplementsInterfaceClass * iface)
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{
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iface->supported = gst_apexsink_interface_supported;
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}
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static void
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gst_apexsink_mixer_interface_init (GstMixerInterface * iface)
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{
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GST_MIXER_TYPE (iface) = GST_MIXER_SOFTWARE;
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iface->list_tracks = gst_apexsink_mixer_list_tracks;
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iface->set_volume = gst_apexsink_mixer_set_volume;
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iface->get_volume = gst_apexsink_mixer_get_volume;
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}
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static gboolean
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gst_apexsink_interface_supported (GstImplementsInterface * iface,
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GType iface_type)
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{
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g_return_val_if_fail (iface_type == GST_TYPE_MIXER, FALSE);
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return TRUE;
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}
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static const GList *
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gst_apexsink_mixer_list_tracks (GstMixer * mixer)
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{
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GstApExSink *apexsink = GST_APEX_SINK (mixer);
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return apexsink->tracks;
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}
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static void
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gst_apexsink_mixer_set_volume (GstMixer * mixer, GstMixerTrack * track,
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gint * volumes)
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{
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GstApExSink *apexsink = GST_APEX_SINK (mixer);
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apexsink->volume = volumes[0];
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if (apexsink->gst_apexraop != NULL)
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gst_apexraop_set_volume (apexsink->gst_apexraop, apexsink->volume);
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}
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static void
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gst_apexsink_mixer_get_volume (GstMixer * mixer, GstMixerTrack * track,
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gint * volumes)
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{
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GstApExSink *apexsink = GST_APEX_SINK (mixer);
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volumes[0] = apexsink->volume;
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}
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/* sink base init */
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static void
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gst_apexsink_base_init (gpointer g_class)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
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gst_element_class_set_static_metadata (element_class,
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"Apple AirPort Express Audio Sink", "Sink/Audio/Wireless",
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"Output stream to an AirPort Express",
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"Jérémie Bernard [GRemi] <gremimail@gmail.com>");
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_apexsink_sink_factory));
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}
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/* sink class init */
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static void
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gst_apexsink_class_init (GstApExSinkClass * klass)
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{
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GST_DEBUG_CATEGORY_INIT (apexsink_debug, GST_APEX_SINK_NAME, 0,
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"AirPort Express sink");
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parent_class = g_type_class_peek_parent (klass);
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((GObjectClass *) klass)->get_property =
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GST_DEBUG_FUNCPTR (gst_apexsink_get_property);
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((GObjectClass *) klass)->set_property =
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GST_DEBUG_FUNCPTR (gst_apexsink_set_property);
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((GObjectClass *) klass)->finalize =
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GST_DEBUG_FUNCPTR (gst_apexsink_finalise);
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((GstAudioSinkClass *) klass)->open = GST_DEBUG_FUNCPTR (gst_apexsink_open);
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((GstAudioSinkClass *) klass)->prepare =
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GST_DEBUG_FUNCPTR (gst_apexsink_prepare);
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((GstAudioSinkClass *) klass)->write = GST_DEBUG_FUNCPTR (gst_apexsink_write);
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((GstAudioSinkClass *) klass)->unprepare =
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GST_DEBUG_FUNCPTR (gst_apexsink_unprepare);
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((GstAudioSinkClass *) klass)->delay = GST_DEBUG_FUNCPTR (gst_apexsink_delay);
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((GstAudioSinkClass *) klass)->reset = GST_DEBUG_FUNCPTR (gst_apexsink_reset);
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((GstAudioSinkClass *) klass)->close = GST_DEBUG_FUNCPTR (gst_apexsink_close);
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((GstElementClass *) klass)->change_state =
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GST_DEBUG_FUNCPTR (gst_apexsink_change_state);
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g_object_class_install_property ((GObjectClass *) klass, APEX_PROP_HOST,
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g_param_spec_string ("host", "Host", "AirPort Express target host",
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DEFAULT_APEX_HOST, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property ((GObjectClass *) klass, APEX_PROP_PORT,
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g_param_spec_uint ("port", "Port", "AirPort Express target port", 0,
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32000, DEFAULT_APEX_PORT,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/* we need to expose the volume as a double for playbin. Internally we keep
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* it as an int between 0 and 100, where 75 corresponds to 1.0.
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* FIXME we should store the volume as a double. */
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g_object_class_install_property ((GObjectClass *) klass, APEX_PROP_VOLUME,
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g_param_spec_double ("volume", "Volume", "AirPort Express target volume",
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0.0, 10.0, DEFAULT_APEX_VOLUME,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property ((GObjectClass *) klass, APEX_PROP_JACK_TYPE,
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g_param_spec_enum ("jack-type", "Jack Type",
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"AirPort Express connected jack type", GST_APEX_SINK_JACKTYPE_TYPE,
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DEFAULT_APEX_JACK_TYPE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property ((GObjectClass *) klass,
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APEX_PROP_JACK_STATUS, g_param_spec_enum ("jack-status", "Jack Status",
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"AirPort Express jack connection status",
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GST_APEX_SINK_JACKSTATUS_TYPE, DEFAULT_APEX_JACK_STATUS,
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G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property ((GObjectClass *) klass,
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APEX_PROP_GENERATION, g_param_spec_enum ("generation", "Generation",
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"AirPort device generation",
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GST_APEX_SINK_GENERATION_TYPE, DEFAULT_APEX_GENERATION,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property ((GObjectClass *) klass,
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APEX_PROP_TRANSPORT_PROTOCOL, g_param_spec_enum ("transport-protocol",
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"Transport Protocol", "AirPort transport protocol",
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GST_APEX_SINK_TRANSPORT_PROTOCOL_TYPE,
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DEFAULT_APEX_TRANSPORT_PROTOCOL,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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}
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/* sink plugin instance init */
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static void
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gst_apexsink_init (GstApExSink * apexsink, GstApExSinkClass * g_class)
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{
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GstMixerTrack *track = NULL;
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track = g_object_new (GST_TYPE_MIXER_TRACK, NULL);
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track->label = g_strdup ("Airport Express");
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track->num_channels = GST_APEX_RAOP_CHANNELS;
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track->min_volume = 0;
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track->max_volume = 100;
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track->flags = GST_MIXER_TRACK_OUTPUT;
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apexsink->host = g_strdup (DEFAULT_APEX_HOST);
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apexsink->port = DEFAULT_APEX_PORT;
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apexsink->volume = CLAMP (DEFAULT_APEX_VOLUME * 75, 0, 100);
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apexsink->gst_apexraop = NULL;
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apexsink->tracks = g_list_append (apexsink->tracks, track);
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apexsink->clock = gst_system_clock_obtain ();
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apexsink->clock_id = NULL;
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GST_INFO_OBJECT (apexsink,
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"ApEx sink default initialization, target=\"%s\", port=\"%d\", volume=\"%d%%\"",
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apexsink->host, apexsink->port, apexsink->volume);
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}
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/* apex sink set property */
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static void
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gst_apexsink_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstApExSink *sink = GST_APEX_SINK (object);
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switch (prop_id) {
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case APEX_PROP_HOST:
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if (sink->gst_apexraop == NULL) {
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g_free (sink->host);
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sink->host = g_value_dup_string (value);
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GST_INFO_OBJECT (sink, "ApEx sink target set to \"%s\"", sink->host);
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} else {
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G_OBJECT_WARN_INVALID_PSPEC (object, "host", prop_id, pspec);
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}
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break;
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case APEX_PROP_PORT:
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if (sink->gst_apexraop == NULL) {
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sink->port = g_value_get_uint (value);
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GST_INFO_OBJECT (sink, "ApEx port set to \"%d\"", sink->port);
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} else {
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G_OBJECT_WARN_INVALID_PSPEC (object, "port", prop_id, pspec);
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}
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break;
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case APEX_PROP_VOLUME:
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{
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gdouble volume;
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volume = g_value_get_double (value);
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volume *= 75.0;
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sink->volume = CLAMP (volume, 0, 100);
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if (sink->gst_apexraop != NULL)
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gst_apexraop_set_volume (sink->gst_apexraop, sink->volume);
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GST_INFO_OBJECT (sink, "ApEx volume set to \"%d%%\"", sink->volume);
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break;
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}
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case APEX_PROP_GENERATION:
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if (sink->gst_apexraop == NULL) {
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sink->generation = g_value_get_enum (value);
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GST_INFO_OBJECT (sink, "ApEx generation set to \"%d\"",
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sink->generation);
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} else {
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GST_WARNING_OBJECT (sink,
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"SET-PROPERTY : generation property may not be set when apexsink opened !");
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}
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break;
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case APEX_PROP_TRANSPORT_PROTOCOL:
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if (sink->gst_apexraop == NULL) {
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sink->transport_protocol = g_value_get_enum (value);
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GST_INFO_OBJECT (sink, "ApEx transport protocol set to \"%d\"",
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sink->transport_protocol);
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} else {
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GST_WARNING_OBJECT (sink,
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"SET-PROPERTY : transport protocol property may not be set when apexsink opened !");
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}
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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/* apex sink get property */
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static void
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gst_apexsink_get_property (GObject * object, guint prop_id, GValue * value,
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GParamSpec * pspec)
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{
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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);
|
|
}
|