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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.
225 lines
5.9 KiB
C
225 lines
5.9 KiB
C
/* GStreamer
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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 <stdlib.h>
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#include <string.h>
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#include "gstrtpg729depay.h"
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#include "gstrtputils.h"
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GST_DEBUG_CATEGORY_STATIC (rtpg729depay_debug);
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#define GST_CAT_DEFAULT (rtpg729depay_debug)
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/* references:
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*
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* RFC 3551 (4.5.6)
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*/
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enum
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{
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/* FILL ME */
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LAST_SIGNAL
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};
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enum
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{
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PROP_0
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};
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/* input is an RTP packet
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*
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*/
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static GstStaticPadTemplate gst_rtp_g729_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) 8000, "
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"encoding-name = (string) \"G729\"; "
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"application/x-rtp, "
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"media = (string) \"audio\", "
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"payload = (int) " GST_RTP_PAYLOAD_G729_STRING ", "
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"clock-rate = (int) 8000")
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);
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static GstStaticPadTemplate gst_rtp_g729_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/G729, " "channels = (int) 1," "rate = (int) 8000")
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);
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static gboolean gst_rtp_g729_depay_setcaps (GstRTPBaseDepayload * depayload,
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GstCaps * caps);
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static GstBuffer *gst_rtp_g729_depay_process (GstRTPBaseDepayload * depayload,
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GstRTPBuffer * rtp);
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#define gst_rtp_g729_depay_parent_class parent_class
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G_DEFINE_TYPE (GstRtpG729Depay, gst_rtp_g729_depay,
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GST_TYPE_RTP_BASE_DEPAYLOAD);
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static void
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gst_rtp_g729_depay_class_init (GstRtpG729DepayClass * 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 (rtpg729depay_debug, "rtpg729depay", 0,
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"G.729 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_g729_depay_src_template);
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_rtp_g729_depay_sink_template);
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gst_element_class_set_static_metadata (gstelement_class,
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"RTP G.729 depayloader", "Codec/Depayloader/Network/RTP",
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"Extracts G.729 audio from RTP packets (RFC 3551)",
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"Laurent Glayal <spglegle@yahoo.fr>");
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gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_g729_depay_process;
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gstrtpbasedepayload_class->set_caps = gst_rtp_g729_depay_setcaps;
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}
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static void
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gst_rtp_g729_depay_init (GstRtpG729Depay * rtpg729depay)
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{
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GstRTPBaseDepayload *depayload;
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depayload = GST_RTP_BASE_DEPAYLOAD (rtpg729depay);
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gst_pad_use_fixed_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload));
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}
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static gboolean
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gst_rtp_g729_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
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{
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GstStructure *structure;
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GstCaps *srccaps;
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GstRtpG729Depay *rtpg729depay;
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const gchar *params;
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gint clock_rate, channels;
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gboolean ret;
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rtpg729depay = GST_RTP_G729_DEPAY (depayload);
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structure = gst_caps_get_structure (caps, 0);
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if (!(params = gst_structure_get_string (structure, "encoding-params")))
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channels = 1;
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else {
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channels = atoi (params);
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}
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if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
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clock_rate = 8000;
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if (channels != 1)
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goto wrong_channels;
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if (clock_rate != 8000)
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goto wrong_clock_rate;
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depayload->clock_rate = clock_rate;
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srccaps = gst_caps_new_simple ("audio/G729",
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"channels", G_TYPE_INT, channels, "rate", G_TYPE_INT, clock_rate, NULL);
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ret = gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload), srccaps);
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gst_caps_unref (srccaps);
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return ret;
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/* ERRORS */
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wrong_channels:
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{
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GST_DEBUG_OBJECT (rtpg729depay, "expected 1 channel, got %d", channels);
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return FALSE;
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}
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wrong_clock_rate:
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{
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GST_DEBUG_OBJECT (rtpg729depay, "expected 8000 clock-rate, got %d",
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clock_rate);
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return FALSE;
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}
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}
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static GstBuffer *
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gst_rtp_g729_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
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{
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GstRtpG729Depay *rtpg729depay;
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GstBuffer *outbuf = NULL;
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gint payload_len;
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gboolean marker;
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rtpg729depay = GST_RTP_G729_DEPAY (depayload);
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payload_len = gst_rtp_buffer_get_payload_len (rtp);
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/* At least 2 bytes (CNG from G729 Annex B) */
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if (payload_len < 2) {
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GST_ELEMENT_WARNING (rtpg729depay, STREAM, DECODE,
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(NULL), ("G729 RTP payload too small (%d)", payload_len));
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goto bad_packet;
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}
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GST_LOG_OBJECT (rtpg729depay, "payload len %d", payload_len);
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if ((payload_len % 10) == 2) {
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GST_LOG_OBJECT (rtpg729depay, "G729 payload contains CNG frame");
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}
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outbuf = gst_rtp_buffer_get_payload_buffer (rtp);
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marker = gst_rtp_buffer_get_marker (rtp);
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if (marker) {
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/* marker bit starts talkspurt */
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GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
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}
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gst_rtp_drop_non_audio_meta (depayload, outbuf);
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GST_LOG_OBJECT (depayload, "pushing buffer of size %" G_GSIZE_FORMAT,
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gst_buffer_get_size (outbuf));
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return outbuf;
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/* ERRORS */
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bad_packet:
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{
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/* no fatal error */
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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_g729_depay_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtpg729depay",
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GST_RANK_SECONDARY, GST_TYPE_RTP_G729_DEPAY);
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}
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