/* * Opus Payloader Gst Element * * @author: Danilo Cesar Lemes de Paula * * 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 #include #include "gstrtpopuspay.h" GST_DEBUG_CATEGORY_STATIC (rtpopuspay_debug); #define GST_CAT_DEFAULT (rtpopuspay_debug) static GstStaticPadTemplate gst_rtp_opus_pay_sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-opus, multistream = (boolean) FALSE") ); static GstStaticPadTemplate gst_rtp_opus_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) 48000, " "encoding-name = (string) \"X-GST-OPUS-DRAFT-SPITTKA-00\"") ); static gboolean gst_rtp_opus_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps); static GstFlowReturn gst_rtp_opus_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buffer); GST_BOILERPLATE (GstRtpOPUSPay, gst_rtp_opus_pay, GstBaseRTPPayload, GST_TYPE_BASE_RTP_PAYLOAD); static void gst_rtp_opus_pay_base_init (gpointer klass) { GstElementClass *element_class = GST_ELEMENT_CLASS (klass); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_opus_pay_src_template)); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_opus_pay_sink_template)); gst_element_class_set_details_simple (element_class, "RTP Opus payloader", "Codec/Payloader/Network/RTP", "Puts Opus audio in RTP packets", "Danilo Cesar Lemes de Paula "); } static void gst_rtp_opus_pay_class_init (GstRtpOPUSPayClass * klass) { GstBaseRTPPayloadClass *gstbasertppayload_class; gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass; gstbasertppayload_class->set_caps = gst_rtp_opus_pay_setcaps; gstbasertppayload_class->handle_buffer = gst_rtp_opus_pay_handle_buffer; GST_DEBUG_CATEGORY_INIT (rtpopuspay_debug, "rtpopuspay", 0, "Opus RTP Payloader"); } static void gst_rtp_opus_pay_init (GstRtpOPUSPay * rtpopuspay, GstRtpOPUSPayClass * klass) { } static gboolean gst_rtp_opus_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps) { gboolean res; gchar *capsstr; capsstr = gst_caps_to_string (caps); gst_basertppayload_set_options (payload, "audio", FALSE, "X-GST-OPUS-DRAFT-SPITTKA-00", 48000); res = gst_basertppayload_set_outcaps (payload, "caps", G_TYPE_STRING, capsstr, NULL); g_free (capsstr); return res; } static GstFlowReturn gst_rtp_opus_pay_handle_buffer (GstBaseRTPPayload * basepayload, GstBuffer * buffer) { GstBuffer *outbuf; GstClockTime timestamp; guint size; guint8 *data; guint8 *payload; size = GST_BUFFER_SIZE (buffer); data = GST_BUFFER_DATA (buffer); timestamp = GST_BUFFER_TIMESTAMP (buffer); outbuf = gst_rtp_buffer_new_allocate (size, 0, 0); payload = gst_rtp_buffer_get_payload (outbuf); memcpy (payload, data, size); gst_rtp_buffer_set_marker (outbuf, FALSE); GST_BUFFER_TIMESTAMP (outbuf) = timestamp; return gst_basertppayload_push (basepayload, outbuf); }