/* GStreamer * Copyright (C) <2005> Edgard Lima * * 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 "gstrtpspeexdepay.h" /* RtpSPEEXDepay signals and args */ enum { /* FILL ME */ LAST_SIGNAL }; enum { ARG_0 }; static GstStaticPadTemplate gst_rtp_speex_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) [6000, 48000], " "encoding-name = (string) \"SPEEX\", " "encoding-params = (string) \"1\"") ); static GstStaticPadTemplate gst_rtp_speex_depay_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-speex") ); static GstBuffer *gst_rtp_speex_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf); static gboolean gst_rtp_speex_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps); G_DEFINE_TYPE (GstRtpSPEEXDepay, gst_rtp_speex_depay, GST_TYPE_BASE_RTP_DEPAYLOAD); static void gst_rtp_speex_depay_class_init (GstRtpSPEEXDepayClass * klass) { GstElementClass *gstelement_class; GstBaseRTPDepayloadClass *gstbasertpdepayload_class; gstelement_class = (GstElementClass *) klass; gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass; gstbasertpdepayload_class->process = gst_rtp_speex_depay_process; gstbasertpdepayload_class->set_caps = gst_rtp_speex_depay_setcaps; gst_element_class_add_pad_template (gstelement_class, gst_static_pad_template_get (&gst_rtp_speex_depay_src_template)); gst_element_class_add_pad_template (gstelement_class, gst_static_pad_template_get (&gst_rtp_speex_depay_sink_template)); gst_element_class_set_details_simple (gstelement_class, "RTP Speex depayloader", "Codec/Depayloader/Network/RTP", "Extracts Speex audio from RTP packets", "Edgard Lima "); } static void gst_rtp_speex_depay_init (GstRtpSPEEXDepay * rtpspeexdepay) { } static gint gst_rtp_speex_depay_get_mode (gint rate) { if (rate > 25000) return 2; else if (rate > 12500) return 1; else return 0; } /* len 4 bytes LE, * vendor string (len bytes), * user_len 4 (0) bytes LE */ static const gchar gst_rtp_speex_comment[] = "\045\0\0\0Depayloaded with GStreamer speexdepay\0\0\0\0"; static gboolean gst_rtp_speex_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps) { GstStructure *structure; GstRtpSPEEXDepay *rtpspeexdepay; gint clock_rate, nb_channels; GstBuffer *buf; guint8 *data, *bdata; const gchar *params; GstCaps *srccaps; gboolean res; rtpspeexdepay = GST_RTP_SPEEX_DEPAY (depayload); structure = gst_caps_get_structure (caps, 0); if (!gst_structure_get_int (structure, "clock-rate", &clock_rate)) goto no_clockrate; depayload->clock_rate = clock_rate; if (!(params = gst_structure_get_string (structure, "encoding-params"))) nb_channels = 1; else { nb_channels = atoi (params); } /* construct minimal header and comment packet for the decoder */ buf = gst_buffer_new_and_alloc (80); data = bdata = gst_buffer_map (buf, NULL, NULL, GST_MAP_WRITE); memcpy (data, "Speex ", 8); data += 8; memcpy (data, "1.1.12", 7); data += 20; GST_WRITE_UINT32_LE (data, 1); /* version */ data += 4; GST_WRITE_UINT32_LE (data, 80); /* header_size */ data += 4; GST_WRITE_UINT32_LE (data, clock_rate); /* rate */ data += 4; GST_WRITE_UINT32_LE (data, gst_rtp_speex_depay_get_mode (clock_rate)); /* mode */ data += 4; GST_WRITE_UINT32_LE (data, 4); /* mode_bitstream_version */ data += 4; GST_WRITE_UINT32_LE (data, nb_channels); /* nb_channels */ data += 4; GST_WRITE_UINT32_LE (data, -1); /* bitrate */ data += 4; GST_WRITE_UINT32_LE (data, 0xa0); /* frame_size */ data += 4; GST_WRITE_UINT32_LE (data, 0); /* VBR */ data += 4; GST_WRITE_UINT32_LE (data, 1); /* frames_per_packet */ data += 4; GST_WRITE_UINT32_LE (data, 0); /* extra_headers */ data += 4; GST_WRITE_UINT32_LE (data, 0); /* reserved1 */ data += 4; GST_WRITE_UINT32_LE (data, 0); /* reserved2 */ gst_buffer_unmap (buf, bdata, -1); srccaps = gst_caps_new_empty_simple ("audio/x-speex"); res = gst_pad_set_caps (depayload->srcpad, srccaps); gst_caps_unref (srccaps); gst_base_rtp_depayload_push (GST_BASE_RTP_DEPAYLOAD (rtpspeexdepay), buf); buf = gst_buffer_new_and_alloc (sizeof (gst_rtp_speex_comment)); bdata = gst_buffer_map (buf, NULL, NULL, GST_MAP_WRITE); memcpy (bdata, gst_rtp_speex_comment, sizeof (gst_rtp_speex_comment)); gst_buffer_unmap (buf, bdata, -1); gst_base_rtp_depayload_push (GST_BASE_RTP_DEPAYLOAD (rtpspeexdepay), buf); return res; /* ERRORS */ no_clockrate: { GST_DEBUG_OBJECT (depayload, "no clock-rate specified"); return FALSE; } } static GstBuffer * gst_rtp_speex_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf) { GstBuffer *outbuf = NULL; GstRTPBuffer rtp; gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp); GST_DEBUG ("process : got %d bytes, mark %d ts %u seqn %d", gst_buffer_get_size (buf), gst_rtp_buffer_get_marker (&rtp), gst_rtp_buffer_get_timestamp (&rtp), gst_rtp_buffer_get_seq (&rtp)); /* nothing special to be done */ outbuf = gst_rtp_buffer_get_payload_buffer (&rtp); gst_rtp_buffer_unmap (&rtp); if (outbuf) GST_BUFFER_DURATION (outbuf) = 20 * GST_MSECOND; return outbuf; } gboolean gst_rtp_speex_depay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpspeexdepay", GST_RANK_SECONDARY, GST_TYPE_RTP_SPEEX_DEPAY); }