2 * Copyright (C) <2005> Edgard Lima <edgard.lima@gmail.com>
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., 51 Franklin St, Fifth Floor,
17 * Boston, MA 02110-1301, USA.
26 #include <gst/rtp/gstrtpbuffer.h>
27 #include <gst/audio/audio.h>
29 #include "gstrtpspeexpay.h"
30 #include "gstrtputils.h"
32 GST_DEBUG_CATEGORY_STATIC (rtpspeexpay_debug);
33 #define GST_CAT_DEFAULT (rtpspeexpay_debug)
35 static GstStaticPadTemplate gst_rtp_speex_pay_sink_template =
36 GST_STATIC_PAD_TEMPLATE ("sink",
39 GST_STATIC_CAPS ("audio/x-speex, "
40 "rate = (int) [ 6000, 48000 ], " "channels = (int) 1")
43 static GstStaticPadTemplate gst_rtp_speex_pay_src_template =
44 GST_STATIC_PAD_TEMPLATE ("src",
47 GST_STATIC_CAPS ("application/x-rtp, "
48 "media = (string) \"audio\", "
49 "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
50 "clock-rate = (int) [ 6000, 48000 ], "
51 "encoding-name = (string) \"SPEEX\", "
52 "encoding-params = (string) \"1\"")
55 static GstStateChangeReturn gst_rtp_speex_pay_change_state (GstElement *
56 element, GstStateChange transition);
58 static gboolean gst_rtp_speex_pay_setcaps (GstRTPBasePayload * payload,
60 static GstCaps *gst_rtp_speex_pay_getcaps (GstRTPBasePayload * payload,
61 GstPad * pad, GstCaps * filter);
62 static GstFlowReturn gst_rtp_speex_pay_handle_buffer (GstRTPBasePayload *
63 payload, GstBuffer * buffer);
65 #define gst_rtp_speex_pay_parent_class parent_class
66 G_DEFINE_TYPE (GstRtpSPEEXPay, gst_rtp_speex_pay, GST_TYPE_RTP_BASE_PAYLOAD);
69 gst_rtp_speex_pay_class_init (GstRtpSPEEXPayClass * klass)
71 GstElementClass *gstelement_class;
72 GstRTPBasePayloadClass *gstrtpbasepayload_class;
74 gstelement_class = (GstElementClass *) klass;
75 gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
77 gstelement_class->change_state = gst_rtp_speex_pay_change_state;
79 gstrtpbasepayload_class->set_caps = gst_rtp_speex_pay_setcaps;
80 gstrtpbasepayload_class->get_caps = gst_rtp_speex_pay_getcaps;
81 gstrtpbasepayload_class->handle_buffer = gst_rtp_speex_pay_handle_buffer;
83 gst_element_class_add_static_pad_template (gstelement_class,
84 &gst_rtp_speex_pay_sink_template);
85 gst_element_class_add_static_pad_template (gstelement_class,
86 &gst_rtp_speex_pay_src_template);
87 gst_element_class_set_static_metadata (gstelement_class,
88 "RTP Speex payloader", "Codec/Payloader/Network/RTP",
89 "Payload-encodes Speex audio into a RTP packet",
90 "Edgard Lima <edgard.lima@gmail.com>");
92 GST_DEBUG_CATEGORY_INIT (rtpspeexpay_debug, "rtpspeexpay", 0,
93 "Speex RTP Payloader");
97 gst_rtp_speex_pay_init (GstRtpSPEEXPay * rtpspeexpay)
99 GST_RTP_BASE_PAYLOAD (rtpspeexpay)->clock_rate = 8000;
100 GST_RTP_BASE_PAYLOAD_PT (rtpspeexpay) = 110; /* Create String */
104 gst_rtp_speex_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
106 /* don't configure yet, we wait for the ident packet */
112 gst_rtp_speex_pay_getcaps (GstRTPBasePayload * payload, GstPad * pad,
115 GstCaps *otherpadcaps;
118 otherpadcaps = gst_pad_get_allowed_caps (payload->srcpad);
119 caps = gst_pad_get_pad_template_caps (pad);
122 if (!gst_caps_is_empty (otherpadcaps)) {
127 ps = gst_caps_get_structure (otherpadcaps, 0);
128 caps = gst_caps_make_writable (caps);
129 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);
139 GstCaps *tcaps = caps;
141 caps = gst_caps_intersect_full (filter, tcaps, GST_CAPS_INTERSECT_FIRST);
142 gst_caps_unref (tcaps);
149 gst_rtp_speex_pay_parse_ident (GstRtpSPEEXPay * rtpspeexpay,
150 const guint8 * data, guint size)
152 guint32 version, header_size, rate, mode, nb_channels;
153 GstRTPBasePayload *payload;
157 /* we need the header string (8), the version string (20), the version
158 * and the header length. */
162 if (!g_str_has_prefix ((const gchar *) data, "Speex "))
165 /* skip header and version string */
168 version = GST_READ_UINT32_LE (data);
174 header_size = GST_READ_UINT32_LE (data);
175 if (header_size < 80)
176 goto header_too_small;
178 if (size < header_size)
179 goto payload_too_small;
182 rate = GST_READ_UINT32_LE (data);
184 mode = GST_READ_UINT32_LE (data);
186 nb_channels = GST_READ_UINT32_LE (data);
188 GST_DEBUG_OBJECT (rtpspeexpay, "rate %d, mode %d, nb_channels %d",
189 rate, mode, nb_channels);
191 payload = GST_RTP_BASE_PAYLOAD (rtpspeexpay);
193 gst_rtp_base_payload_set_options (payload, "audio", FALSE, "SPEEX", rate);
194 cstr = g_strdup_printf ("%d", nb_channels);
195 res = gst_rtp_base_payload_set_outcaps (payload, "encoding-params",
196 G_TYPE_STRING, cstr, NULL);
204 GST_DEBUG_OBJECT (rtpspeexpay,
205 "ident packet too small, need at least 32 bytes");
210 GST_DEBUG_OBJECT (rtpspeexpay,
211 "ident packet does not start with \"Speex \"");
216 GST_DEBUG_OBJECT (rtpspeexpay, "can only handle version 1, have version %d",
222 GST_DEBUG_OBJECT (rtpspeexpay,
223 "header size too small, need at least 80 bytes, " "got only %d",
229 GST_DEBUG_OBJECT (rtpspeexpay,
230 "payload too small, need at least %d bytes, got only %d", header_size,
237 gst_rtp_speex_pay_handle_buffer (GstRTPBasePayload * basepayload,
240 GstRtpSPEEXPay *rtpspeexpay;
243 GstClockTime timestamp, duration;
246 rtpspeexpay = GST_RTP_SPEEX_PAY (basepayload);
248 gst_buffer_map (buffer, &map, GST_MAP_READ);
250 switch (rtpspeexpay->packet) {
252 /* ident packet. We need to parse the headers to construct the RTP
254 if (!gst_rtp_speex_pay_parse_ident (rtpspeexpay, map.data, map.size)) {
255 gst_buffer_unmap (buffer, &map);
260 gst_buffer_unmap (buffer, &map);
263 /* comment packet, we ignore it */
265 gst_buffer_unmap (buffer, &map);
268 /* other packets go in the payload */
271 gst_buffer_unmap (buffer, &map);
273 if (GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_GAP)) {
278 timestamp = GST_BUFFER_PTS (buffer);
279 duration = GST_BUFFER_DURATION (buffer);
281 /* FIXME, only one SPEEX frame per RTP packet for now */
282 outbuf = gst_rtp_base_payload_allocate_output_buffer (basepayload, 0, 0, 0);
284 /* FIXME, assert for now */
285 g_assert (gst_buffer_get_size (buffer) <=
286 GST_RTP_BASE_PAYLOAD_MTU (rtpspeexpay));
288 /* copy timestamp and duration */
289 GST_BUFFER_PTS (outbuf) = timestamp;
290 GST_BUFFER_DURATION (outbuf) = duration;
292 gst_rtp_copy_audio_meta (basepayload, outbuf, buffer);
293 outbuf = gst_buffer_append (outbuf, buffer);
296 ret = gst_rtp_base_payload_push (basepayload, outbuf);
300 gst_buffer_unref (buffer);
302 rtpspeexpay->packet++;
309 GST_ELEMENT_ERROR (rtpspeexpay, STREAM, DECODE, (NULL),
310 ("Error parsing first identification packet."));
311 gst_buffer_unref (buffer);
312 return GST_FLOW_ERROR;
316 static GstStateChangeReturn
317 gst_rtp_speex_pay_change_state (GstElement * element, GstStateChange transition)
319 GstRtpSPEEXPay *rtpspeexpay;
320 GstStateChangeReturn ret;
322 rtpspeexpay = GST_RTP_SPEEX_PAY (element);
324 switch (transition) {
325 case GST_STATE_CHANGE_NULL_TO_READY:
327 case GST_STATE_CHANGE_READY_TO_PAUSED:
328 rtpspeexpay->packet = 0;
334 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
336 switch (transition) {
337 case GST_STATE_CHANGE_READY_TO_NULL:
346 gst_rtp_speex_pay_plugin_init (GstPlugin * plugin)
348 return gst_element_register (plugin, "rtpspeexpay",
349 GST_RANK_SECONDARY, GST_TYPE_RTP_SPEEX_PAY);