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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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5a17572119
Initialize the PT to the default value of the codec and check if it is still the default before declaring the pt to be dynamic or not when setting the caps. Also use the PT constants from the rtp lib when possible https://bugzilla.gnome.org/show_bug.cgi?id=747965
116 lines
3.8 KiB
C
116 lines
3.8 KiB
C
/* GStreamer
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* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
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* Copyright (C) <2005> Edgard Lima <edgard.lima@indt.org.br>
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* Copyright (C) <2005> Nokia Corporation <kai.vehmanen@nokia.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 "gstrtppcmapay.h"
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static GstStaticPadTemplate gst_rtp_pcma_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-alaw, channels=(int)1, rate=(int)8000")
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);
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static GstStaticPadTemplate gst_rtp_pcma_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_PCMA_STRING ", "
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"clock-rate = (int) 8000, " "encoding-name = (string) \"PCMA\"; "
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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) [1, MAX ], " "encoding-name = (string) \"PCMA\"")
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);
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static gboolean gst_rtp_pcma_pay_setcaps (GstRTPBasePayload * payload,
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GstCaps * caps);
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#define gst_rtp_pcma_pay_parent_class parent_class
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G_DEFINE_TYPE (GstRtpPcmaPay, gst_rtp_pcma_pay,
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GST_TYPE_RTP_BASE_AUDIO_PAYLOAD);
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static void
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gst_rtp_pcma_pay_class_init (GstRtpPcmaPayClass * klass)
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{
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GstElementClass *gstelement_class;
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GstRTPBasePayloadClass *gstrtpbasepayload_class;
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gstelement_class = (GstElementClass *) klass;
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gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&gst_rtp_pcma_pay_sink_template));
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&gst_rtp_pcma_pay_src_template));
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gst_element_class_set_static_metadata (gstelement_class, "RTP PCMA payloader",
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"Codec/Payloader/Network/RTP",
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"Payload-encodes PCMA audio into a RTP packet",
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"Edgard Lima <edgard.lima@indt.org.br>");
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gstrtpbasepayload_class->set_caps = gst_rtp_pcma_pay_setcaps;
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}
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static void
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gst_rtp_pcma_pay_init (GstRtpPcmaPay * rtppcmapay)
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{
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GstRTPBaseAudioPayload *rtpbaseaudiopayload;
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rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtppcmapay);
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GST_RTP_BASE_PAYLOAD (rtppcmapay)->pt = GST_RTP_PAYLOAD_PCMA;
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GST_RTP_BASE_PAYLOAD (rtppcmapay)->clock_rate = 8000;
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/* tell rtpbaseaudiopayload that this is a sample based codec */
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gst_rtp_base_audio_payload_set_sample_based (rtpbaseaudiopayload);
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/* octet-per-sample is 1 for PCM */
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gst_rtp_base_audio_payload_set_sample_options (rtpbaseaudiopayload, 1);
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}
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static gboolean
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gst_rtp_pcma_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
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{
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gboolean res;
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gst_rtp_base_payload_set_options (payload, "audio",
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payload->pt != GST_RTP_PAYLOAD_PCMA, "PCMA", 8000);
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res = gst_rtp_base_payload_set_outcaps (payload, NULL);
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return res;
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}
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gboolean
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gst_rtp_pcma_pay_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtppcmapay",
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GST_RANK_SECONDARY, GST_TYPE_RTP_PCMA_PAY);
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}
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