mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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177 lines
5.2 KiB
C
177 lines
5.2 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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, 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 "gstrtpgsmpay.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 (GstBaseRTPPayload * payload,
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GstCaps * caps);
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static GstFlowReturn gst_rtp_gsm_pay_handle_buffer (GstBaseRTPPayload * payload,
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GstBuffer * buffer);
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GST_BOILERPLATE (GstRTPGSMPay, gst_rtp_gsm_pay, GstBaseRTPPayload,
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GST_TYPE_BASE_RTP_PAYLOAD);
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static void
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gst_rtp_gsm_pay_base_init (gpointer klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_rtp_gsm_pay_sink_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_rtp_gsm_pay_src_template));
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gst_element_class_set_details_simple (element_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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}
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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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GstBaseRTPPayloadClass *gstbasertppayload_class;
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gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
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gstbasertppayload_class->set_caps = gst_rtp_gsm_pay_setcaps;
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gstbasertppayload_class->handle_buffer = gst_rtp_gsm_pay_handle_buffer;
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GST_DEBUG_CATEGORY_INIT (rtpgsmpay_debug, "rtpgsmpay", 0,
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"GSM Audio RTP Payloader");
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}
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static void
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gst_rtp_gsm_pay_init (GstRTPGSMPay * rtpgsmpay, GstRTPGSMPayClass * klass)
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{
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GST_BASE_RTP_PAYLOAD (rtpgsmpay)->clock_rate = 8000;
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GST_BASE_RTP_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 (GstBaseRTPPayload * 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_basertppayload_set_options (payload, "audio", FALSE, "GSM", 8000);
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res = gst_basertppayload_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 (GstBaseRTPPayload * basepayload,
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GstBuffer * buffer)
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{
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GstRTPGSMPay *rtpgsmpay;
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guint size, payload_len;
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GstBuffer *outbuf;
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guint8 *payload, *data;
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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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size = GST_BUFFER_SIZE (buffer);
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timestamp = GST_BUFFER_TIMESTAMP (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 = size;
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outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
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/* FIXME, assert for now */
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g_assert (payload_len <= GST_BASE_RTP_PAYLOAD_MTU (rtpgsmpay));
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/* copy timestamp and duration */
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GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
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GST_BUFFER_DURATION (outbuf) = duration;
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/* get payload */
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payload = gst_rtp_buffer_get_payload (outbuf);
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data = GST_BUFFER_DATA (buffer);
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/* copy data in payload */
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memcpy (&payload[0], data, size);
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gst_buffer_unref (buffer);
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GST_DEBUG ("gst_rtp_gsm_pay_chain: pushing buffer of size %d",
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GST_BUFFER_SIZE (outbuf));
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ret = gst_basertppayload_push (basepayload, outbuf);
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return ret;
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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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