/* GStreamer * Copyright (C) <2009> Wim Taymans * * 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. */ /** * SECTION:element-rtpbvpay * @see_also: rtpbvdepay * * Payload BroadcomVoice audio into RTP packets according to RFC 4298. * For detailed information see: http://www.rfc-editor.org/rfc/rfc4298.txt */ #ifdef HAVE_CONFIG_H # include "config.h" #endif #include #include #include #include "gstrtpbvpay.h" GST_DEBUG_CATEGORY_STATIC (rtpbvpay_debug); #define GST_CAT_DEFAULT (rtpbvpay_debug) static GstStaticPadTemplate gst_rtp_bv_pay_sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-bv, " "mode = (int) {16, 32}") ); static GstStaticPadTemplate gst_rtp_bv_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) \"BV16\";" "application/x-rtp, " "media = (string) \"audio\", " "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", " "clock-rate = (int) 16000, " "encoding-name = (string) \"BV32\"") ); static GstCaps *gst_rtp_bv_pay_sink_getcaps (GstRTPBasePayload * payload, GstPad * pad, GstCaps * filter); static gboolean gst_rtp_bv_pay_sink_setcaps (GstRTPBasePayload * payload, GstCaps * caps); #define gst_rtp_bv_pay_parent_class parent_class G_DEFINE_TYPE (GstRTPBVPay, gst_rtp_bv_pay, GST_TYPE_RTP_BASE_AUDIO_PAYLOAD); static void gst_rtp_bv_pay_class_init (GstRTPBVPayClass * klass) { GstElementClass *gstelement_class; GstRTPBasePayloadClass *gstrtpbasepayload_class; GST_DEBUG_CATEGORY_INIT (rtpbvpay_debug, "rtpbvpay", 0, "BroadcomVoice audio RTP payloader"); gstelement_class = (GstElementClass *) klass; gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass; gst_element_class_add_static_pad_template (gstelement_class, &gst_rtp_bv_pay_sink_template); gst_element_class_add_static_pad_template (gstelement_class, &gst_rtp_bv_pay_src_template); gst_element_class_set_static_metadata (gstelement_class, "RTP BV Payloader", "Codec/Payloader/Network/RTP", "Packetize BroadcomVoice audio streams into RTP packets (RFC 4298)", "Wim Taymans "); gstrtpbasepayload_class->set_caps = gst_rtp_bv_pay_sink_setcaps; gstrtpbasepayload_class->get_caps = gst_rtp_bv_pay_sink_getcaps; } static void gst_rtp_bv_pay_init (GstRTPBVPay * rtpbvpay) { GstRTPBaseAudioPayload *rtpbaseaudiopayload; rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpbvpay); rtpbvpay->mode = -1; /* tell rtpbaseaudiopayload that this is a frame based codec */ gst_rtp_base_audio_payload_set_frame_based (rtpbaseaudiopayload); } static gboolean gst_rtp_bv_pay_sink_setcaps (GstRTPBasePayload * rtpbasepayload, GstCaps * caps) { GstRTPBVPay *rtpbvpay; GstRTPBaseAudioPayload *rtpbaseaudiopayload; gint mode; GstStructure *structure; const char *payload_name; rtpbvpay = GST_RTP_BV_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-bv", payload_name)) goto wrong_caps; if (!gst_structure_get_int (structure, "mode", &mode)) goto no_mode; if (mode != 16 && mode != 32) goto wrong_mode; if (mode == 16) { gst_rtp_base_payload_set_options (rtpbasepayload, "audio", TRUE, "BV16", 8000); rtpbasepayload->clock_rate = 8000; } else { gst_rtp_base_payload_set_options (rtpbasepayload, "audio", TRUE, "BV32", 16000); rtpbasepayload->clock_rate = 16000; } /* set options for this frame based audio codec */ gst_rtp_base_audio_payload_set_frame_options (rtpbaseaudiopayload, mode, mode == 16 ? 10 : 20); if (mode != rtpbvpay->mode && rtpbvpay->mode != -1) goto mode_changed; rtpbvpay->mode = mode; return TRUE; /* ERRORS */ wrong_caps: { GST_ERROR_OBJECT (rtpbvpay, "expected audio/x-bv, received %s", payload_name); return FALSE; } no_mode: { GST_ERROR_OBJECT (rtpbvpay, "did not receive a mode"); return FALSE; } wrong_mode: { GST_ERROR_OBJECT (rtpbvpay, "mode must be 16 or 32, received %d", mode); return FALSE; } mode_changed: { GST_ERROR_OBJECT (rtpbvpay, "Mode has changed from %d to %d! " "Mode cannot change while streaming", rtpbvpay->mode, mode); return FALSE; } } /* we return the padtemplate caps with the mode field fixated to a value if we * can */ static GstCaps * gst_rtp_bv_pay_sink_getcaps (GstRTPBasePayload * rtppayload, GstPad * pad, GstCaps * filter) { GstCaps *otherpadcaps; GstCaps *caps; caps = gst_pad_get_pad_template_caps (pad); otherpadcaps = gst_pad_get_allowed_caps (rtppayload->srcpad); if (otherpadcaps) { if (!gst_caps_is_empty (otherpadcaps)) { GstStructure *structure; const gchar *mode_str; gint mode; structure = gst_caps_get_structure (otherpadcaps, 0); /* construct mode, if we can */ mode_str = gst_structure_get_string (structure, "encoding-name"); if (mode_str) { if (!strcmp (mode_str, "BV16")) mode = 16; else if (!strcmp (mode_str, "BV32")) mode = 32; else mode = -1; if (mode == 16 || mode == 32) { 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_bv_pay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpbvpay", GST_RANK_SECONDARY, GST_TYPE_RTP_BV_PAY); }