2 * Copyright (C) <2005> Edgard Lima <edgard.lima@indt.org.br>
4 * This library is free software; you can redistribute it and/or
5 * modify it under the terms of the GNU Library General Public
6 * License as published by the Free Software Foundation; either
7 * version 2 of the License, or (at your option) any later version.
9 * This library is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12 * Library General Public License for more details.
14 * You should have received a copy of the GNU Library General Public
15 * License along with this library; if not, write to the
16 * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
17 * Boston, MA 02111-1307, USA.
26 #include <gst/rtp/gstrtpbuffer.h>
28 #include "gstrtpspeexpay.h"
30 GST_DEBUG_CATEGORY_STATIC (rtpspeexpay_debug);
31 #define GST_CAT_DEFAULT (rtpspeexpay_debug)
33 static GstStaticPadTemplate gst_rtp_speex_pay_sink_template =
34 GST_STATIC_PAD_TEMPLATE ("sink",
37 GST_STATIC_CAPS ("audio/x-speex, "
38 "rate = (int) [ 6000, 48000 ], " "channels = (int) 1")
41 static GstStaticPadTemplate gst_rtp_speex_pay_src_template =
42 GST_STATIC_PAD_TEMPLATE ("src",
45 GST_STATIC_CAPS ("application/x-rtp, "
46 "media = (string) \"audio\", "
47 "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
48 "clock-rate = (int) [ 6000, 48000 ], "
49 "encoding-name = (string) \"SPEEX\", "
50 "encoding-params = (string) \"1\"")
53 static GstStateChangeReturn gst_rtp_speex_pay_change_state (GstElement *
54 element, GstStateChange transition);
56 static gboolean gst_rtp_speex_pay_setcaps (GstBaseRTPPayload * payload,
58 static GstCaps *gst_rtp_speex_pay_getcaps (GstBaseRTPPayload * payload,
60 static GstFlowReturn gst_rtp_speex_pay_handle_buffer (GstBaseRTPPayload *
61 payload, GstBuffer * buffer);
63 GST_BOILERPLATE (GstRtpSPEEXPay, gst_rtp_speex_pay, GstBaseRTPPayload,
64 GST_TYPE_BASE_RTP_PAYLOAD);
67 gst_rtp_speex_pay_base_init (gpointer klass)
69 GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
71 gst_element_class_add_static_pad_template (element_class,
72 &gst_rtp_speex_pay_sink_template);
73 gst_element_class_add_static_pad_template (element_class,
74 &gst_rtp_speex_pay_src_template);
75 gst_element_class_set_details_simple (element_class, "RTP Speex payloader",
76 "Codec/Payloader/Network/RTP",
77 "Payload-encodes Speex audio into a RTP packet",
78 "Edgard Lima <edgard.lima@indt.org.br>");
80 GST_DEBUG_CATEGORY_INIT (rtpspeexpay_debug, "rtpspeexpay", 0,
81 "Speex RTP Payloader");
85 gst_rtp_speex_pay_class_init (GstRtpSPEEXPayClass * klass)
87 GstElementClass *gstelement_class;
88 GstBaseRTPPayloadClass *gstbasertppayload_class;
90 gstelement_class = (GstElementClass *) klass;
91 gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
93 gstelement_class->change_state = gst_rtp_speex_pay_change_state;
95 gstbasertppayload_class->set_caps = gst_rtp_speex_pay_setcaps;
96 gstbasertppayload_class->get_caps = gst_rtp_speex_pay_getcaps;
97 gstbasertppayload_class->handle_buffer = gst_rtp_speex_pay_handle_buffer;
101 gst_rtp_speex_pay_init (GstRtpSPEEXPay * rtpspeexpay,
102 GstRtpSPEEXPayClass * klass)
104 GST_BASE_RTP_PAYLOAD (rtpspeexpay)->clock_rate = 8000;
105 GST_BASE_RTP_PAYLOAD_PT (rtpspeexpay) = 110; /* Create String */
109 gst_rtp_speex_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
111 /* don't configure yet, we wait for the ident packet */
117 gst_rtp_speex_pay_getcaps (GstBaseRTPPayload * payload, GstPad * pad)
119 GstCaps *otherpadcaps;
122 otherpadcaps = gst_pad_get_allowed_caps (payload->srcpad);
123 caps = gst_caps_copy (gst_pad_get_pad_template_caps (pad));
126 if (!gst_caps_is_empty (otherpadcaps)) {
127 GstStructure *ps = gst_caps_get_structure (otherpadcaps, 0);
128 GstStructure *s = gst_caps_get_structure (caps, 0);
131 if (gst_structure_get_int (ps, "clock-rate", &clock_rate)) {
132 gst_structure_fixate_field_nearest_int (s, "rate", clock_rate);
135 gst_caps_unref (otherpadcaps);
142 gst_rtp_speex_pay_parse_ident (GstRtpSPEEXPay * rtpspeexpay,
143 const guint8 * data, guint size)
145 guint32 version, header_size, rate, mode, nb_channels;
146 GstBaseRTPPayload *payload;
150 /* we need the header string (8), the version string (20), the version
151 * and the header length. */
155 if (!g_str_has_prefix ((const gchar *) data, "Speex "))
158 /* skip header and version string */
161 version = GST_READ_UINT32_LE (data);
167 header_size = GST_READ_UINT32_LE (data);
168 if (header_size < 80)
169 goto header_too_small;
171 if (size < header_size)
172 goto payload_too_small;
175 rate = GST_READ_UINT32_LE (data);
177 mode = GST_READ_UINT32_LE (data);
179 nb_channels = GST_READ_UINT32_LE (data);
181 GST_DEBUG_OBJECT (rtpspeexpay, "rate %d, mode %d, nb_channels %d",
182 rate, mode, nb_channels);
184 payload = GST_BASE_RTP_PAYLOAD (rtpspeexpay);
186 gst_basertppayload_set_options (payload, "audio", FALSE, "SPEEX", rate);
187 cstr = g_strdup_printf ("%d", nb_channels);
188 res = gst_basertppayload_set_outcaps (payload, "encoding-params",
189 G_TYPE_STRING, cstr, NULL);
197 GST_DEBUG_OBJECT (rtpspeexpay,
198 "ident packet too small, need at least 32 bytes");
203 GST_DEBUG_OBJECT (rtpspeexpay,
204 "ident packet does not start with \"Speex \"");
209 GST_DEBUG_OBJECT (rtpspeexpay, "can only handle version 1, have version %d",
215 GST_DEBUG_OBJECT (rtpspeexpay,
216 "header size too small, need at least 80 bytes, " "got only %d",
222 GST_DEBUG_OBJECT (rtpspeexpay,
223 "payload too small, need at least %d bytes, got only %d", header_size,
230 gst_rtp_speex_pay_handle_buffer (GstBaseRTPPayload * basepayload,
233 GstRtpSPEEXPay *rtpspeexpay;
234 guint size, payload_len;
236 guint8 *payload, *data;
237 GstClockTime timestamp, duration;
240 rtpspeexpay = GST_RTP_SPEEX_PAY (basepayload);
242 size = GST_BUFFER_SIZE (buffer);
243 data = GST_BUFFER_DATA (buffer);
245 switch (rtpspeexpay->packet) {
247 /* ident packet. We need to parse the headers to construct the RTP
249 if (!gst_rtp_speex_pay_parse_ident (rtpspeexpay, data, size))
255 /* comment packet, we ignore it */
259 /* other packets go in the payload */
263 if (GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_GAP)) {
268 timestamp = GST_BUFFER_TIMESTAMP (buffer);
269 duration = GST_BUFFER_DURATION (buffer);
271 /* FIXME, only one SPEEX frame per RTP packet for now */
274 outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
275 /* FIXME, assert for now */
276 g_assert (payload_len <= GST_BASE_RTP_PAYLOAD_MTU (rtpspeexpay));
278 /* copy timestamp and duration */
279 GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
280 GST_BUFFER_DURATION (outbuf) = duration;
283 payload = gst_rtp_buffer_get_payload (outbuf);
285 /* copy data in payload */
286 memcpy (&payload[0], data, size);
288 ret = gst_basertppayload_push (basepayload, outbuf);
291 gst_buffer_unref (buffer);
293 rtpspeexpay->packet++;
300 GST_ELEMENT_ERROR (rtpspeexpay, STREAM, DECODE, (NULL),
301 ("Error parsing first identification packet."));
302 gst_buffer_unref (buffer);
303 return GST_FLOW_ERROR;
307 static GstStateChangeReturn
308 gst_rtp_speex_pay_change_state (GstElement * element, GstStateChange transition)
310 GstRtpSPEEXPay *rtpspeexpay;
311 GstStateChangeReturn ret;
313 rtpspeexpay = GST_RTP_SPEEX_PAY (element);
315 switch (transition) {
316 case GST_STATE_CHANGE_NULL_TO_READY:
318 case GST_STATE_CHANGE_READY_TO_PAUSED:
319 rtpspeexpay->packet = 0;
325 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
327 switch (transition) {
328 case GST_STATE_CHANGE_READY_TO_NULL:
337 gst_rtp_speex_pay_plugin_init (GstPlugin * plugin)
339 return gst_element_register (plugin, "rtpspeexpay",
340 GST_RANK_SECONDARY, GST_TYPE_RTP_SPEEX_PAY);