| /* |
| * Opus Payloader Gst Element |
| * |
| * @author: Danilo Cesar Lemes de Paula <danilo.cesar@collabora.co.uk> |
| * |
| * 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 <gst/rtp/gstrtpbuffer.h> |
| |
| #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) { \"OPUS\", \"X-GST-OPUS-DRAFT-SPITTKA-00\" }") |
| ); |
| |
| static gboolean gst_rtp_opus_pay_setcaps (GstRTPBasePayload * payload, |
| GstCaps * caps); |
| static GstFlowReturn gst_rtp_opus_pay_handle_buffer (GstRTPBasePayload * |
| payload, GstBuffer * buffer); |
| |
| G_DEFINE_TYPE (GstRtpOPUSPay, gst_rtp_opus_pay, GST_TYPE_RTP_BASE_PAYLOAD); |
| |
| static void |
| gst_rtp_opus_pay_class_init (GstRtpOPUSPayClass * klass) |
| { |
| GstRTPBasePayloadClass *gstbasertppayload_class; |
| GstElementClass *element_class; |
| |
| gstbasertppayload_class = (GstRTPBasePayloadClass *) klass; |
| element_class = GST_ELEMENT_CLASS (klass); |
| |
| gstbasertppayload_class->set_caps = gst_rtp_opus_pay_setcaps; |
| gstbasertppayload_class->handle_buffer = gst_rtp_opus_pay_handle_buffer; |
| |
| 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_static_metadata (element_class, |
| "RTP Opus payloader", |
| "Codec/Payloader/Network/RTP", |
| "Puts Opus audio in RTP packets", |
| "Danilo Cesar Lemes de Paula <danilo.cesar@collabora.co.uk>"); |
| |
| GST_DEBUG_CATEGORY_INIT (rtpopuspay_debug, "rtpopuspay", 0, |
| "Opus RTP Payloader"); |
| } |
| |
| static void |
| gst_rtp_opus_pay_init (GstRtpOPUSPay * rtpopuspay) |
| { |
| } |
| |
| static gboolean |
| gst_rtp_opus_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps) |
| { |
| gboolean res; |
| GstCaps *src_caps; |
| GstStructure *s; |
| char *encoding_name; |
| |
| src_caps = gst_pad_get_allowed_caps (GST_RTP_BASE_PAYLOAD_SRCPAD (payload)); |
| if (src_caps) { |
| src_caps = gst_caps_make_writable (src_caps); |
| src_caps = gst_caps_truncate (src_caps); |
| s = gst_caps_get_structure (src_caps, 0); |
| gst_structure_fixate_field_string (s, "encoding-name", "OPUS"); |
| encoding_name = g_strdup (gst_structure_get_string (s, "encoding-name")); |
| gst_caps_unref (src_caps); |
| } else { |
| encoding_name = g_strdup ("X-GST-OPUS-DRAFT-SPITTKA-00"); |
| } |
| |
| gst_rtp_base_payload_set_options (payload, "audio", FALSE, |
| encoding_name, 48000); |
| g_free (encoding_name); |
| res = gst_rtp_base_payload_set_outcaps (payload, NULL); |
| |
| return res; |
| } |
| |
| static GstFlowReturn |
| gst_rtp_opus_pay_handle_buffer (GstRTPBasePayload * basepayload, |
| GstBuffer * buffer) |
| { |
| GstBuffer *outbuf; |
| GstClockTime pts, dts, duration; |
| |
| pts = GST_BUFFER_PTS (buffer); |
| dts = GST_BUFFER_DTS (buffer); |
| duration = GST_BUFFER_DURATION (buffer); |
| |
| outbuf = gst_rtp_buffer_new_allocate (0, 0, 0); |
| outbuf = gst_buffer_append (outbuf, buffer); |
| |
| GST_BUFFER_PTS (outbuf) = pts; |
| GST_BUFFER_DTS (outbuf) = dts; |
| GST_BUFFER_DURATION (outbuf) = duration; |
| |
| /* Push out */ |
| return gst_rtp_base_payload_push (basepayload, outbuf); |
| } |