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This is required to add RTP header extensions from the meta automatically. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/674>
181 lines
5.4 KiB
C
181 lines
5.4 KiB
C
/* GStreamer
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* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
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* Copyright (C) <2005> Zeeshan Ali <zeenix@gmail.com>
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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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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#include <stdlib.h>
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#include <string.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include <gst/audio/audio.h>
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#include "gstrtpgsmpay.h"
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#include "gstrtputils.h"
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GST_DEBUG_CATEGORY_STATIC (rtpgsmpay_debug);
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#define GST_CAT_DEFAULT (rtpgsmpay_debug)
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static GstStaticPadTemplate gst_rtp_gsm_pay_sink_template =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-gsm, " "rate = (int) 8000, " "channels = (int) 1")
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);
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static GstStaticPadTemplate gst_rtp_gsm_pay_src_template =
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GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("application/x-rtp, "
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"media = (string) \"audio\", "
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"payload = (int) " GST_RTP_PAYLOAD_GSM_STRING ", "
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"clock-rate = (int) 8000, " "encoding-name = (string) \"GSM\"; "
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"application/x-rtp, "
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"media = (string) \"audio\", "
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"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
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"clock-rate = (int) 8000, " "encoding-name = (string) \"GSM\"")
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);
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static gboolean gst_rtp_gsm_pay_setcaps (GstRTPBasePayload * payload,
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GstCaps * caps);
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static GstFlowReturn gst_rtp_gsm_pay_handle_buffer (GstRTPBasePayload * payload,
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GstBuffer * buffer);
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#define gst_rtp_gsm_pay_parent_class parent_class
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G_DEFINE_TYPE (GstRTPGSMPay, gst_rtp_gsm_pay, GST_TYPE_RTP_BASE_PAYLOAD);
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static void
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gst_rtp_gsm_pay_class_init (GstRTPGSMPayClass * klass)
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{
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GstElementClass *gstelement_class;
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GstRTPBasePayloadClass *gstrtpbasepayload_class;
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GST_DEBUG_CATEGORY_INIT (rtpgsmpay_debug, "rtpgsmpay", 0,
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"GSM Audio RTP Payloader");
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gstelement_class = (GstElementClass *) klass;
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gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_rtp_gsm_pay_sink_template);
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_rtp_gsm_pay_src_template);
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gst_element_class_set_static_metadata (gstelement_class, "RTP GSM payloader",
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"Codec/Payloader/Network/RTP",
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"Payload-encodes GSM audio into a RTP packet",
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"Zeeshan Ali <zeenix@gmail.com>");
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gstrtpbasepayload_class->set_caps = gst_rtp_gsm_pay_setcaps;
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gstrtpbasepayload_class->handle_buffer = gst_rtp_gsm_pay_handle_buffer;
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}
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static void
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gst_rtp_gsm_pay_init (GstRTPGSMPay * rtpgsmpay)
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{
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GST_RTP_BASE_PAYLOAD (rtpgsmpay)->clock_rate = 8000;
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GST_RTP_BASE_PAYLOAD_PT (rtpgsmpay) = GST_RTP_PAYLOAD_GSM;
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}
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static gboolean
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gst_rtp_gsm_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
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{
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const char *stname;
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GstStructure *structure;
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gboolean res;
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structure = gst_caps_get_structure (caps, 0);
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stname = gst_structure_get_name (structure);
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if (strcmp ("audio/x-gsm", stname))
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goto invalid_type;
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gst_rtp_base_payload_set_options (payload, "audio",
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payload->pt != GST_RTP_PAYLOAD_GSM, "GSM", 8000);
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res = gst_rtp_base_payload_set_outcaps (payload, NULL);
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return res;
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/* ERRORS */
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invalid_type:
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{
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GST_WARNING_OBJECT (payload, "invalid media type received");
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return FALSE;
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}
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}
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static GstFlowReturn
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gst_rtp_gsm_pay_handle_buffer (GstRTPBasePayload * basepayload,
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GstBuffer * buffer)
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{
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GstRTPGSMPay *rtpgsmpay;
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guint payload_len;
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GstBuffer *outbuf;
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GstClockTime timestamp, duration;
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GstFlowReturn ret;
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rtpgsmpay = GST_RTP_GSM_PAY (basepayload);
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timestamp = GST_BUFFER_PTS (buffer);
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duration = GST_BUFFER_DURATION (buffer);
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/* FIXME, only one GSM frame per RTP packet for now */
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payload_len = gst_buffer_get_size (buffer);
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/* FIXME, just error out for now */
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if (payload_len > GST_RTP_BASE_PAYLOAD_MTU (rtpgsmpay))
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goto too_big;
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outbuf = gst_rtp_base_payload_allocate_output_buffer (basepayload, 0, 0, 0);
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/* copy timestamp and duration */
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GST_BUFFER_PTS (outbuf) = timestamp;
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GST_BUFFER_DURATION (outbuf) = duration;
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gst_rtp_copy_audio_meta (rtpgsmpay, outbuf, buffer);
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/* append payload */
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outbuf = gst_buffer_append (outbuf, buffer);
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GST_DEBUG ("gst_rtp_gsm_pay_chain: pushing buffer of size %" G_GSIZE_FORMAT,
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gst_buffer_get_size (outbuf));
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ret = gst_rtp_base_payload_push (basepayload, outbuf);
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return ret;
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/* ERRORS */
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too_big:
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{
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GST_ELEMENT_ERROR (rtpgsmpay, STREAM, ENCODE, (NULL),
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("payload_len %u > mtu %u", payload_len,
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GST_RTP_BASE_PAYLOAD_MTU (rtpgsmpay)));
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return GST_FLOW_ERROR;
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}
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}
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gboolean
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gst_rtp_gsm_pay_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtpgsmpay",
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GST_RANK_SECONDARY, GST_TYPE_RTP_GSM_PAY);
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}
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