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Every g_quark_from_static_string() is a hash table lookup serialised on the global quark lock in GLib. Let's just look up the two quarks we need once and cache them locally for future use. While we're at it, add new utility functions for the two most commonly used tags (audio + video). Make first argument a gpointer so we don't have to cast and make the code ugly. These are used for logging purposes only anyway.
181 lines
5.4 KiB
C
181 lines
5.4 KiB
C
/* GStreamer
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* Copyright (C) <2005> Wim Taymans <wim.taymans@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 <gst/rtp/gstrtpbuffer.h>
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#include <gst/audio/audio.h>
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#include <string.h>
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#include "gstrtpmpadepay.h"
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#include "gstrtputils.h"
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GST_DEBUG_CATEGORY_STATIC (rtpmpadepay_debug);
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#define GST_CAT_DEFAULT (rtpmpadepay_debug)
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static GstStaticPadTemplate gst_rtp_mpa_depay_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 ("audio/mpeg, " "mpegversion = (int) 1")
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);
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static GstStaticPadTemplate gst_rtp_mpa_depay_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 ("application/x-rtp, "
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"media = (string) \"audio\", "
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"payload = (int) " GST_RTP_PAYLOAD_MPA_STRING ", "
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"clock-rate = (int) 90000 ;"
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"application/x-rtp, "
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"media = (string) \"audio\", "
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"encoding-name = (string) \"MPA\", clock-rate = (int) [1, MAX]")
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);
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#define gst_rtp_mpa_depay_parent_class parent_class
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G_DEFINE_TYPE (GstRtpMPADepay, gst_rtp_mpa_depay, GST_TYPE_RTP_BASE_DEPAYLOAD);
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static gboolean gst_rtp_mpa_depay_setcaps (GstRTPBaseDepayload * depayload,
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GstCaps * caps);
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static GstBuffer *gst_rtp_mpa_depay_process (GstRTPBaseDepayload * depayload,
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GstRTPBuffer * rtp);
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static void
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gst_rtp_mpa_depay_class_init (GstRtpMPADepayClass * klass)
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{
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GstElementClass *gstelement_class;
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GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
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GST_DEBUG_CATEGORY_INIT (rtpmpadepay_debug, "rtpmpadepay", 0,
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"MPEG Audio RTP Depayloader");
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gstelement_class = (GstElementClass *) klass;
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gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_rtp_mpa_depay_src_template);
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_rtp_mpa_depay_sink_template);
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gst_element_class_set_static_metadata (gstelement_class,
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"RTP MPEG audio depayloader", "Codec/Depayloader/Network/RTP",
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"Extracts MPEG audio from RTP packets (RFC 2038)",
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"Wim Taymans <wim.taymans@gmail.com>");
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gstrtpbasedepayload_class->set_caps = gst_rtp_mpa_depay_setcaps;
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gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_mpa_depay_process;
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}
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static void
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gst_rtp_mpa_depay_init (GstRtpMPADepay * rtpmpadepay)
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{
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}
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static gboolean
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gst_rtp_mpa_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
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{
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GstStructure *structure;
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GstCaps *outcaps;
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gint clock_rate;
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gboolean res;
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structure = gst_caps_get_structure (caps, 0);
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if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
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clock_rate = 90000;
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depayload->clock_rate = clock_rate;
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outcaps =
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gst_caps_new_simple ("audio/mpeg", "mpegversion", G_TYPE_INT, 1, NULL);
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res = gst_pad_set_caps (depayload->srcpad, outcaps);
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gst_caps_unref (outcaps);
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return res;
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}
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static GstBuffer *
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gst_rtp_mpa_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
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{
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GstRtpMPADepay *rtpmpadepay;
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GstBuffer *outbuf;
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gint payload_len;
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#if 0
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guint8 *payload;
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guint16 frag_offset;
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#endif
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gboolean marker;
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rtpmpadepay = GST_RTP_MPA_DEPAY (depayload);
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payload_len = gst_rtp_buffer_get_payload_len (rtp);
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if (payload_len <= 4)
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goto empty_packet;
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#if 0
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payload = gst_rtp_buffer_get_payload (&rtp);
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/* strip off header
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*
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* 0 1 2 3
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* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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* | MBZ | Frag_offset |
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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*/
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frag_offset = (payload[2] << 8) | payload[3];
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#endif
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/* subbuffer skipping the 4 header bytes */
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outbuf = gst_rtp_buffer_get_payload_subbuffer (rtp, 4, -1);
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marker = gst_rtp_buffer_get_marker (rtp);
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if (marker) {
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/* mark start of talkspurt with RESYNC */
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GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
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}
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GST_DEBUG_OBJECT (rtpmpadepay,
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"gst_rtp_mpa_depay_chain: pushing buffer of size %" G_GSIZE_FORMAT "",
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gst_buffer_get_size (outbuf));
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if (outbuf) {
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gst_rtp_drop_non_audio_meta (rtpmpadepay, outbuf);
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}
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/* FIXME, we can push half mpeg frames when they are split over multiple
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* RTP packets */
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return outbuf;
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/* ERRORS */
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empty_packet:
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{
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GST_ELEMENT_WARNING (rtpmpadepay, STREAM, DECODE,
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("Empty Payload."), (NULL));
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return NULL;
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}
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}
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gboolean
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gst_rtp_mpa_depay_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtpmpadepay",
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GST_RANK_SECONDARY, GST_TYPE_RTP_MPA_DEPAY);
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}
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