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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.
180 lines
5.3 KiB
C
180 lines
5.3 KiB
C
/* GStreamer
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* Copyright (C) <2007> 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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/**
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* SECTION:element-rtpac3depay
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* @see_also: rtpac3pay
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*
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* Extract AC3 audio from RTP packets according to RFC 4184.
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* For detailed information see: http://www.rfc-editor.org/rfc/rfc4184.txt
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*
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* <refsect2>
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* <title>Example pipeline</title>
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* |[
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* gst-launch-1.0 udpsrc caps='application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)AC3, payload=(int)96' ! rtpac3depay ! a52dec ! pulsesink
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* ]| This example pipeline will depayload and decode an RTP AC3 stream. Refer to
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* the rtpac3pay example to create the RTP stream.
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* </refsect2>
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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 "gstrtpac3depay.h"
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#include "gstrtputils.h"
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GST_DEBUG_CATEGORY_STATIC (rtpac3depay_debug);
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#define GST_CAT_DEFAULT (rtpac3depay_debug)
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static GstStaticPadTemplate gst_rtp_ac3_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/ac3")
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);
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static GstStaticPadTemplate gst_rtp_ac3_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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"clock-rate = (int) { 32000, 44100, 48000 }, "
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"encoding-name = (string) \"AC3\"")
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);
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G_DEFINE_TYPE (GstRtpAC3Depay, gst_rtp_ac3_depay, GST_TYPE_RTP_BASE_DEPAYLOAD);
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static gboolean gst_rtp_ac3_depay_setcaps (GstRTPBaseDepayload * depayload,
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GstCaps * caps);
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static GstBuffer *gst_rtp_ac3_depay_process (GstRTPBaseDepayload * depayload,
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GstRTPBuffer * rtp);
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static void
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gst_rtp_ac3_depay_class_init (GstRtpAC3DepayClass * klass)
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{
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GstElementClass *gstelement_class;
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GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
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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_ac3_depay_src_template);
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_rtp_ac3_depay_sink_template);
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gst_element_class_set_static_metadata (gstelement_class,
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"RTP AC3 depayloader", "Codec/Depayloader/Network/RTP",
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"Extracts AC3 audio from RTP packets (RFC 4184)",
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"Wim Taymans <wim.taymans@gmail.com>");
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gstrtpbasedepayload_class->set_caps = gst_rtp_ac3_depay_setcaps;
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gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_ac3_depay_process;
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GST_DEBUG_CATEGORY_INIT (rtpac3depay_debug, "rtpac3depay", 0,
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"AC3 Audio RTP Depayloader");
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}
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static void
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gst_rtp_ac3_depay_init (GstRtpAC3Depay * rtpac3depay)
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{
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/* needed because of G_DEFINE_TYPE */
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}
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static gboolean
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gst_rtp_ac3_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
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{
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GstStructure *structure;
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gint clock_rate;
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GstCaps *srccaps;
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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; /* default */
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depayload->clock_rate = clock_rate;
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srccaps = gst_caps_new_empty_simple ("audio/ac3");
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res = gst_pad_set_caps (depayload->srcpad, srccaps);
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gst_caps_unref (srccaps);
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return res;
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}
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static GstBuffer *
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gst_rtp_ac3_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
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{
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GstRtpAC3Depay *rtpac3depay;
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GstBuffer *outbuf;
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guint8 *payload;
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guint16 FT, NF;
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rtpac3depay = GST_RTP_AC3_DEPAY (depayload);
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if (gst_rtp_buffer_get_payload_len (rtp) < 2)
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goto empty_packet;
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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
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* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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* | MBZ | FT| NF |
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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*/
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FT = payload[0] & 0x3;
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NF = payload[1];
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GST_DEBUG_OBJECT (rtpac3depay, "FT: %d, NF: %d", FT, NF);
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/* We don't bother with fragmented packets yet */
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outbuf = gst_rtp_buffer_get_payload_subbuffer (rtp, 2, -1);
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if (outbuf) {
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gst_rtp_drop_non_audio_meta (rtpac3depay, outbuf);
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GST_DEBUG_OBJECT (rtpac3depay, "pushing buffer of size %" G_GSIZE_FORMAT,
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gst_buffer_get_size (outbuf));
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}
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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 (rtpac3depay, 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_ac3_depay_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtpac3depay",
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GST_RANK_SECONDARY, GST_TYPE_RTP_AC3_DEPAY);
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}
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