/* GStreamer * Copyright (C) <2006> Philippe Khalaf <burger@speedy.org> * * 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 <stdlib.h> #include <gst/rtp/gstrtpbuffer.h> #include "gstrtpilbcpay.h" GST_DEBUG_CATEGORY_STATIC (rtpilbcpay_debug); #define GST_CAT_DEFAULT (rtpilbcpay_debug) static GstStaticPadTemplate gst_rtp_ilbc_pay_sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-iLBC, " "mode = (int) {20, 30}") ); static GstStaticPadTemplate gst_rtp_ilbc_pay_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("application/x-rtp, " "media = (string) \"audio\", " "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", " "clock-rate = (int) 8000, " "encoding-name = (string) \"ILBC\", " "mode = (string) { \"20\", \"30\" }") ); static GstCaps *gst_rtp_ilbc_pay_sink_getcaps (GstRTPBasePayload * payload, GstPad * pad, GstCaps * filter); static gboolean gst_rtp_ilbc_pay_sink_setcaps (GstRTPBasePayload * payload, GstCaps * caps); #define gst_rtp_ilbc_pay_parent_class parent_class G_DEFINE_TYPE (GstRTPILBCPay, gst_rtp_ilbc_pay, GST_TYPE_RTP_BASE_AUDIO_PAYLOAD); static void gst_rtp_ilbc_pay_class_init (GstRTPILBCPayClass * klass) { GstElementClass *gstelement_class; GstRTPBasePayloadClass *gstrtpbasepayload_class; GST_DEBUG_CATEGORY_INIT (rtpilbcpay_debug, "rtpilbcpay", 0, "iLBC audio RTP payloader"); gstelement_class = (GstElementClass *) klass; gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass; gst_element_class_add_pad_template (gstelement_class, gst_static_pad_template_get (&gst_rtp_ilbc_pay_sink_template)); gst_element_class_add_pad_template (gstelement_class, gst_static_pad_template_get (&gst_rtp_ilbc_pay_src_template)); gst_element_class_set_static_metadata (gstelement_class, "RTP iLBC Payloader", "Codec/Payloader/Network/RTP", "Packetize iLBC audio streams into RTP packets", "Philippe Kalaf <philippe.kalaf@collabora.co.uk>"); gstrtpbasepayload_class->set_caps = gst_rtp_ilbc_pay_sink_setcaps; gstrtpbasepayload_class->get_caps = gst_rtp_ilbc_pay_sink_getcaps; } static void gst_rtp_ilbc_pay_init (GstRTPILBCPay * rtpilbcpay) { GstRTPBasePayload *rtpbasepayload; GstRTPBaseAudioPayload *rtpbaseaudiopayload; rtpbasepayload = GST_RTP_BASE_PAYLOAD (rtpilbcpay); rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpilbcpay); /* we don't set the payload type, it should be set by the application using * the pt property or the default 96 will be used */ rtpbasepayload->clock_rate = 8000; rtpilbcpay->mode = -1; /* tell rtpbaseaudiopayload that this is a frame based codec */ gst_rtp_base_audio_payload_set_frame_based (rtpbaseaudiopayload); } static gboolean gst_rtp_ilbc_pay_sink_setcaps (GstRTPBasePayload * rtpbasepayload, GstCaps * caps) { GstRTPILBCPay *rtpilbcpay; GstRTPBaseAudioPayload *rtpbaseaudiopayload; gboolean ret; gint mode; gchar *mode_str; GstStructure *structure; const char *payload_name; rtpilbcpay = GST_RTP_ILBC_PAY (rtpbasepayload); rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpbasepayload); structure = gst_caps_get_structure (caps, 0); payload_name = gst_structure_get_name (structure); if (g_ascii_strcasecmp ("audio/x-iLBC", payload_name)) goto wrong_caps; if (!gst_structure_get_int (structure, "mode", &mode)) goto no_mode; if (mode != 20 && mode != 30) goto wrong_mode; gst_rtp_base_payload_set_options (rtpbasepayload, "audio", TRUE, "ILBC", 8000); /* set options for this frame based audio codec */ gst_rtp_base_audio_payload_set_frame_options (rtpbaseaudiopayload, mode, mode == 30 ? 50 : 38); mode_str = g_strdup_printf ("%d", mode); ret = gst_rtp_base_payload_set_outcaps (rtpbasepayload, "mode", G_TYPE_STRING, mode_str, NULL); g_free (mode_str); if (mode != rtpilbcpay->mode && rtpilbcpay->mode != -1) goto mode_changed; rtpilbcpay->mode = mode; return ret; /* ERRORS */ wrong_caps: { GST_ERROR_OBJECT (rtpilbcpay, "expected audio/x-iLBC, received %s", payload_name); return FALSE; } no_mode: { GST_ERROR_OBJECT (rtpilbcpay, "did not receive a mode"); return FALSE; } wrong_mode: { GST_ERROR_OBJECT (rtpilbcpay, "mode must be 20 or 30, received %d", mode); return FALSE; } mode_changed: { GST_ERROR_OBJECT (rtpilbcpay, "Mode has changed from %d to %d! " "Mode cannot change while streaming", rtpilbcpay->mode, mode); return FALSE; } } /* we return the padtemplate caps with the mode field fixated to a value if we * can */ static GstCaps * gst_rtp_ilbc_pay_sink_getcaps (GstRTPBasePayload * rtppayload, GstPad * pad, GstCaps * filter) { GstCaps *otherpadcaps; GstCaps *caps; otherpadcaps = gst_pad_get_allowed_caps (rtppayload->srcpad); caps = gst_pad_get_pad_template_caps (pad); if (otherpadcaps) { if (!gst_caps_is_empty (otherpadcaps)) { GstStructure *structure; const gchar *mode_str; gint mode; structure = gst_caps_get_structure (otherpadcaps, 0); /* parse mode, if we can */ mode_str = gst_structure_get_string (structure, "mode"); if (mode_str) { mode = strtol (mode_str, NULL, 10); if (mode == 20 || mode == 30) { caps = gst_caps_make_writable (caps); structure = gst_caps_get_structure (caps, 0); gst_structure_set (structure, "mode", G_TYPE_INT, mode, NULL); } } } gst_caps_unref (otherpadcaps); } return caps; } gboolean gst_rtp_ilbc_pay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpilbcpay", GST_RANK_SECONDARY, GST_TYPE_RTP_ILBC_PAY); }