/* GStreamer * Copyright (C) 1999 Erik Walthinsen * Copyright (C) 2005 Edgard Lima * Copyright (C) 2005 Zeeshan Ali * Copyright (C) 2008 Axis Communications * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. */ #ifdef HAVE_CONFIG_H # include "config.h" #endif #include #include #include #include #include "gstrtpg726depay.h" #define DEFAULT_BIT_RATE 32000 #define SAMPLE_RATE 8000 #define LAYOUT_G726 "g726" /* elementfactory information */ static const GstElementDetails gst_rtp_g726depay_details = GST_ELEMENT_DETAILS ("RTP G.726 depayloader", "Codec/Depayloader/Network", "Extracts G.726 audio from RTP packets", "Axis Communications "); /* RtpG726Depay signals and args */ enum { /* FILL ME */ LAST_SIGNAL }; enum { ARG_0 }; static GstStaticPadTemplate gst_rtp_g726_depay_sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("application/x-rtp, " "media = (string) \"audio\", " "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", " "encoding-name = (string) { \"G726\", \"G726-16\", \"G726-24\", \"G726-32\", \"G726-40\"}, " "clock-rate = (int) 8000;") ); static GstStaticPadTemplate gst_rtp_g726_depay_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-adpcm, " "channels = (int) 1, " "rate = (int) 8000, " "bitrate = (int) { 16000, 24000, 32000, 40000 }, " "layout = (string) \"g726\"") ); static GstBuffer *gst_rtp_g726_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf); static gboolean gst_rtp_g726_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps); GST_BOILERPLATE (GstRtpG726Depay, gst_rtp_g726_depay, GstBaseRTPDepayload, GST_TYPE_BASE_RTP_DEPAYLOAD); static void gst_rtp_g726_depay_base_init (gpointer klass) { GstElementClass *element_class = GST_ELEMENT_CLASS (klass); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_g726_depay_src_template)); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_g726_depay_sink_template)); gst_element_class_set_details (element_class, &gst_rtp_g726depay_details); } static void gst_rtp_g726_depay_class_init (GstRtpG726DepayClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; GstBaseRTPDepayloadClass *gstbasertpdepayload_class; gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass; parent_class = g_type_class_peek_parent (klass); gstbasertpdepayload_class->process = gst_rtp_g726_depay_process; gstbasertpdepayload_class->set_caps = gst_rtp_g726_depay_setcaps; } static void gst_rtp_g726_depay_init (GstRtpG726Depay * rtpG726depay, GstRtpG726DepayClass * klass) { GstBaseRTPDepayload *depayload; depayload = GST_BASE_RTP_DEPAYLOAD (rtpG726depay); gst_pad_use_fixed_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload)); } static gboolean gst_rtp_g726_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps) { GstCaps *srccaps; GstStructure *structure; gboolean ret; gint clock_rate; const gchar *encoding_name; gint bitrate; structure = gst_caps_get_structure (caps, 0); if (!gst_structure_get_int (structure, "clock-rate", &clock_rate)) clock_rate = 8000; /* default */ depayload->clock_rate = clock_rate; encoding_name = gst_structure_get_string (structure, "encoding-name"); if (encoding_name == NULL || g_ascii_strcasecmp (encoding_name, "G726") == 0) { bitrate = DEFAULT_BIT_RATE; } else if (g_ascii_strcasecmp (encoding_name, "G726-16") == 0) { bitrate = 16000; } else if (g_ascii_strcasecmp (encoding_name, "G726-24") == 0) { bitrate = 24000; } else if (g_ascii_strcasecmp (encoding_name, "G726-32") == 0) { bitrate = 32000; } else if (g_ascii_strcasecmp (encoding_name, "G726-40") == 0) { bitrate = 40000; } else { GST_WARNING ("Could not determine bitrate from encoding-name (%s)", encoding_name); ret = FALSE; goto done; } GST_DEBUG ("RTP G.726 depayloader, bitrate set to %d\n", bitrate); srccaps = gst_caps_new_simple ("audio/x-adpcm", "channels", G_TYPE_INT, 1, "rate", G_TYPE_INT, clock_rate, "bitrate", G_TYPE_INT, bitrate, "layout", G_TYPE_STRING, LAYOUT_G726, NULL); ret = gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps); gst_caps_unref (srccaps); done: return ret; } static GstBuffer * gst_rtp_g726_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf) { GstBuffer *outbuf = NULL; gboolean marker; marker = gst_rtp_buffer_get_marker (buf); GST_DEBUG ("process : got %d bytes, mark %d ts %u seqn %d", GST_BUFFER_SIZE (buf), marker, gst_rtp_buffer_get_timestamp (buf), gst_rtp_buffer_get_seq (buf)); outbuf = gst_rtp_buffer_get_payload_buffer (buf); if (marker) { /* mark start of talkspurt with discont */ GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT); } return outbuf; } gboolean gst_rtp_g726_depay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpg726depay", GST_RANK_MARGINAL, GST_TYPE_RTP_G726_DEPAY); }