/* GStreamer * Copyright (C) <2005> Edgard Lima <edgard.lima@indt.org.br> * * 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., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, USA. */ #ifdef HAVE_CONFIG_H # include "config.h" #endif #include <stdlib.h> #include <string.h> #include <gst/rtp/gstrtpbuffer.h> #include "gstrtpspeexpay.h" GST_DEBUG_CATEGORY_STATIC (rtpspeexpay_debug); #define GST_CAT_DEFAULT (rtpspeexpay_debug) static GstStaticPadTemplate gst_rtp_speex_pay_sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-speex, " "rate = (int) [ 6000, 48000 ], " "channels = (int) 1") ); static GstStaticPadTemplate gst_rtp_speex_pay_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, 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 GstStateChangeReturn gst_rtp_speex_pay_change_state (GstElement * element, GstStateChange transition); static gboolean gst_rtp_speex_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps); static GstCaps *gst_rtp_speex_pay_getcaps (GstRTPBasePayload * payload, GstPad * pad, GstCaps * filter); static GstFlowReturn gst_rtp_speex_pay_handle_buffer (GstRTPBasePayload * payload, GstBuffer * buffer); #define gst_rtp_speex_pay_parent_class parent_class G_DEFINE_TYPE (GstRtpSPEEXPay, gst_rtp_speex_pay, GST_TYPE_RTP_BASE_PAYLOAD); static void gst_rtp_speex_pay_class_init (GstRtpSPEEXPayClass * klass) { GstElementClass *gstelement_class; GstRTPBasePayloadClass *gstrtpbasepayload_class; gstelement_class = (GstElementClass *) klass; gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass; gstelement_class->change_state = gst_rtp_speex_pay_change_state; gstrtpbasepayload_class->set_caps = gst_rtp_speex_pay_setcaps; gstrtpbasepayload_class->get_caps = gst_rtp_speex_pay_getcaps; gstrtpbasepayload_class->handle_buffer = gst_rtp_speex_pay_handle_buffer; gst_element_class_add_pad_template (gstelement_class, gst_static_pad_template_get (&gst_rtp_speex_pay_sink_template)); gst_element_class_add_pad_template (gstelement_class, gst_static_pad_template_get (&gst_rtp_speex_pay_src_template)); gst_element_class_set_static_metadata (gstelement_class, "RTP Speex payloader", "Codec/Payloader/Network/RTP", "Payload-encodes Speex audio into a RTP packet", "Edgard Lima <edgard.lima@indt.org.br>"); GST_DEBUG_CATEGORY_INIT (rtpspeexpay_debug, "rtpspeexpay", 0, "Speex RTP Payloader"); } static void gst_rtp_speex_pay_init (GstRtpSPEEXPay * rtpspeexpay) { GST_RTP_BASE_PAYLOAD (rtpspeexpay)->clock_rate = 8000; GST_RTP_BASE_PAYLOAD_PT (rtpspeexpay) = 110; /* Create String */ } static gboolean gst_rtp_speex_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps) { /* don't configure yet, we wait for the ident packet */ return TRUE; } static GstCaps * gst_rtp_speex_pay_getcaps (GstRTPBasePayload * payload, GstPad * pad, GstCaps * filter) { GstCaps *otherpadcaps; GstCaps *caps; otherpadcaps = gst_pad_get_allowed_caps (payload->srcpad); caps = gst_pad_get_pad_template_caps (pad); if (otherpadcaps) { if (!gst_caps_is_empty (otherpadcaps)) { GstStructure *ps; GstStructure *s; gint clock_rate; ps = gst_caps_get_structure (otherpadcaps, 0); caps = gst_caps_make_writable (caps); s = gst_caps_get_structure (caps, 0); if (gst_structure_get_int (ps, "clock-rate", &clock_rate)) { gst_structure_fixate_field_nearest_int (s, "rate", clock_rate); } } gst_caps_unref (otherpadcaps); } if (filter) { GstCaps *tcaps = caps; caps = gst_caps_intersect_full (filter, tcaps, GST_CAPS_INTERSECT_FIRST); gst_caps_unref (tcaps); } return caps; } static gboolean gst_rtp_speex_pay_parse_ident (GstRtpSPEEXPay * rtpspeexpay, const guint8 * data, guint size) { guint32 version, header_size, rate, mode, nb_channels; GstRTPBasePayload *payload; gchar *cstr; gboolean res; /* we need the header string (8), the version string (20), the version * and the header length. */ if (size < 36) goto too_small; if (!g_str_has_prefix ((const gchar *) data, "Speex ")) goto wrong_header; /* skip header and version string */ data += 28; version = GST_READ_UINT32_LE (data); if (version != 1) goto wrong_version; data += 4; /* ensure sizes */ header_size = GST_READ_UINT32_LE (data); if (header_size < 80) goto header_too_small; if (size < header_size) goto payload_too_small; data += 4; rate = GST_READ_UINT32_LE (data); data += 4; mode = GST_READ_UINT32_LE (data); data += 8; nb_channels = GST_READ_UINT32_LE (data); GST_DEBUG_OBJECT (rtpspeexpay, "rate %d, mode %d, nb_channels %d", rate, mode, nb_channels); payload = GST_RTP_BASE_PAYLOAD (rtpspeexpay); gst_rtp_base_payload_set_options (payload, "audio", FALSE, "SPEEX", rate); cstr = g_strdup_printf ("%d", nb_channels); res = gst_rtp_base_payload_set_outcaps (payload, "encoding-params", G_TYPE_STRING, cstr, NULL); g_free (cstr); return res; /* ERRORS */ too_small: { GST_DEBUG_OBJECT (rtpspeexpay, "ident packet too small, need at least 32 bytes"); return FALSE; } wrong_header: { GST_DEBUG_OBJECT (rtpspeexpay, "ident packet does not start with \"Speex \""); return FALSE; } wrong_version: { GST_DEBUG_OBJECT (rtpspeexpay, "can only handle version 1, have version %d", version); return FALSE; } header_too_small: { GST_DEBUG_OBJECT (rtpspeexpay, "header size too small, need at least 80 bytes, " "got only %d", header_size); return FALSE; } payload_too_small: { GST_DEBUG_OBJECT (rtpspeexpay, "payload too small, need at least %d bytes, got only %d", header_size, size); return FALSE; } } static GstFlowReturn gst_rtp_speex_pay_handle_buffer (GstRTPBasePayload * basepayload, GstBuffer * buffer) { GstRtpSPEEXPay *rtpspeexpay; guint payload_len; GstMapInfo map; GstBuffer *outbuf; guint8 *payload; GstClockTime timestamp, duration; GstFlowReturn ret; GstRTPBuffer rtp = { NULL }; rtpspeexpay = GST_RTP_SPEEX_PAY (basepayload); gst_buffer_map (buffer, &map, GST_MAP_READ); switch (rtpspeexpay->packet) { case 0: /* ident packet. We need to parse the headers to construct the RTP * properties. */ if (!gst_rtp_speex_pay_parse_ident (rtpspeexpay, map.data, map.size)) goto parse_error; ret = GST_FLOW_OK; goto done; case 1: /* comment packet, we ignore it */ ret = GST_FLOW_OK; goto done; default: /* other packets go in the payload */ break; } if (GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_GAP)) { ret = GST_FLOW_OK; goto done; } timestamp = GST_BUFFER_TIMESTAMP (buffer); duration = GST_BUFFER_DURATION (buffer); /* FIXME, only one SPEEX frame per RTP packet for now */ payload_len = map.size; outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0); /* FIXME, assert for now */ g_assert (payload_len <= GST_RTP_BASE_PAYLOAD_MTU (rtpspeexpay)); /* copy timestamp and duration */ GST_BUFFER_TIMESTAMP (outbuf) = timestamp; GST_BUFFER_DURATION (outbuf) = duration; gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp); /* get payload */ payload = gst_rtp_buffer_get_payload (&rtp); /* copy data in payload */ memcpy (&payload[0], map.data, map.size); gst_rtp_buffer_unmap (&rtp); ret = gst_rtp_base_payload_push (basepayload, outbuf); done: gst_buffer_unmap (buffer, &map); gst_buffer_unref (buffer); rtpspeexpay->packet++; return ret; /* ERRORS */ parse_error: { GST_ELEMENT_ERROR (rtpspeexpay, STREAM, DECODE, (NULL), ("Error parsing first identification packet.")); gst_buffer_unmap (buffer, &map); gst_buffer_unref (buffer); return GST_FLOW_ERROR; } } static GstStateChangeReturn gst_rtp_speex_pay_change_state (GstElement * element, GstStateChange transition) { GstRtpSPEEXPay *rtpspeexpay; GstStateChangeReturn ret; rtpspeexpay = GST_RTP_SPEEX_PAY (element); switch (transition) { case GST_STATE_CHANGE_NULL_TO_READY: break; case GST_STATE_CHANGE_READY_TO_PAUSED: rtpspeexpay->packet = 0; break; default: break; } ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); switch (transition) { case GST_STATE_CHANGE_READY_TO_NULL: break; default: break; } return ret; } gboolean gst_rtp_speex_pay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpspeexpay", GST_RANK_SECONDARY, GST_TYPE_RTP_SPEEX_PAY); }