/* GStreamer * Copyright (C) <2007> Wim Taymans * * 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 "gstrtpac3depay.h" GST_DEBUG_CATEGORY_STATIC (rtpac3depay_debug); #define GST_CAT_DEFAULT (rtpac3depay_debug) /* elementfactory information */ static const GstElementDetails gst_rtp_ac3depay_details = GST_ELEMENT_DETAILS ("RTP AC3 depayloader", "Codec/Depayloader/Network", "Extracts AC3 audio from RTP packets (RFC 4184)", "Wim Taymans "); static GstStaticPadTemplate gst_rtp_ac3_depay_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/ac3") ); static GstStaticPadTemplate gst_rtp_ac3_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 ", " "clock-rate = (int) { 32000, 44100, 48000 }, " "encoding-name = (string) \"AC3\"") ); GST_BOILERPLATE (GstRtpAC3Depay, gst_rtp_ac3_depay, GstBaseRTPDepayload, GST_TYPE_BASE_RTP_DEPAYLOAD); static gboolean gst_rtp_ac3_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps); static GstBuffer *gst_rtp_ac3_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf); static void gst_rtp_ac3_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_ac3_depay_src_template)); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_ac3_depay_sink_template)); gst_element_class_set_details (element_class, &gst_rtp_ac3depay_details); } static void gst_rtp_ac3_depay_class_init (GstRtpAC3DepayClass * klass) { GstBaseRTPDepayloadClass *gstbasertpdepayload_class; gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass; parent_class = g_type_class_peek_parent (klass); gstbasertpdepayload_class->set_caps = gst_rtp_ac3_depay_setcaps; gstbasertpdepayload_class->process = gst_rtp_ac3_depay_process; GST_DEBUG_CATEGORY_INIT (rtpac3depay_debug, "rtpac3depay", 0, "MPEG Audio RTP Depayloader"); } static void gst_rtp_ac3_depay_init (GstRtpAC3Depay * rtpac3depay, GstRtpAC3DepayClass * klass) { /* needed because of GST_BOILERPLATE */ } static gboolean gst_rtp_ac3_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps) { GstStructure *structure; gint clock_rate; GstCaps *srccaps; gboolean res; structure = gst_caps_get_structure (caps, 0); if (!gst_structure_get_int (structure, "clock-rate", &clock_rate)) clock_rate = 90000; /* default */ depayload->clock_rate = clock_rate; srccaps = gst_caps_new_simple ("audio/ac3", NULL); res = gst_pad_set_caps (depayload->srcpad, srccaps); gst_caps_unref (srccaps); return res; } struct frmsize_s { guint16 bit_rate; guint16 frm_size[3]; }; static const struct frmsize_s frmsizecod_tbl[] = { {32, {64, 69, 96}}, {32, {64, 70, 96}}, {40, {80, 87, 120}}, {40, {80, 88, 120}}, {48, {96, 104, 144}}, {48, {96, 105, 144}}, {56, {112, 121, 168}}, {56, {112, 122, 168}}, {64, {128, 139, 192}}, {64, {128, 140, 192}}, {80, {160, 174, 240}}, {80, {160, 175, 240}}, {96, {192, 208, 288}}, {96, {192, 209, 288}}, {112, {224, 243, 336}}, {112, {224, 244, 336}}, {128, {256, 278, 384}}, {128, {256, 279, 384}}, {160, {320, 348, 480}}, {160, {320, 349, 480}}, {192, {384, 417, 576}}, {192, {384, 418, 576}}, {224, {448, 487, 672}}, {224, {448, 488, 672}}, {256, {512, 557, 768}}, {256, {512, 558, 768}}, {320, {640, 696, 960}}, {320, {640, 697, 960}}, {384, {768, 835, 1152}}, {384, {768, 836, 1152}}, {448, {896, 975, 1344}}, {448, {896, 976, 1344}}, {512, {1024, 1114, 1536}}, {512, {1024, 1115, 1536}}, {576, {1152, 1253, 1728}}, {576, {1152, 1254, 1728}}, {640, {1280, 1393, 1920}}, {640, {1280, 1394, 1920}} }; static GstBuffer * gst_rtp_ac3_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf) { GstRtpAC3Depay *rtpac3depay; GstBuffer *outbuf; rtpac3depay = GST_RTP_AC3_DEPAY (depayload); { guint8 *payload; guint16 FT, NF; if (gst_rtp_buffer_get_payload_len (buf) < 2) goto empty_packet; payload = gst_rtp_buffer_get_payload (buf); /* strip off header * * 0 1 * 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ * | MBZ | FT| NF | * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ */ FT = payload[0] & 0x3; NF = payload[1]; GST_DEBUG_OBJECT (rtpac3depay, "FT: %d, NF: %d", FT, NF); /* We don't bother with fragmented packets yet */ outbuf = gst_rtp_buffer_get_payload_subbuffer (buf, 2, -1); GST_DEBUG_OBJECT (rtpac3depay, "pushing buffer of size %d", GST_BUFFER_SIZE (outbuf)); return outbuf; } return NULL; /* ERRORS */ empty_packet: { GST_ELEMENT_WARNING (rtpac3depay, STREAM, DECODE, ("Empty Payload."), (NULL)); return NULL; } } gboolean gst_rtp_ac3_depay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpac3depay", GST_RANK_MARGINAL, GST_TYPE_RTP_AC3_DEPAY); }