2 * Copyright (C) <2009> Wim Taymans <wim.taymans@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 "gstrtpelements.h"
30 #include "gstrtpceltdepay.h"
31 #include "gstrtputils.h"
33 GST_DEBUG_CATEGORY_STATIC (rtpceltdepay_debug);
34 #define GST_CAT_DEFAULT (rtpceltdepay_debug)
36 /* RtpCELTDepay signals and args */
38 #define DEFAULT_FRAMESIZE 480
39 #define DEFAULT_CHANNELS 1
40 #define DEFAULT_CLOCKRATE 32000
53 static GstStaticPadTemplate gst_rtp_celt_depay_sink_template =
54 GST_STATIC_PAD_TEMPLATE ("sink",
57 GST_STATIC_CAPS ("application/x-rtp, "
58 "media = (string) \"audio\", "
59 "clock-rate = (int) [32000, 48000], "
60 "encoding-name = (string) \"CELT\"")
63 static GstStaticPadTemplate gst_rtp_celt_depay_src_template =
64 GST_STATIC_PAD_TEMPLATE ("src",
67 GST_STATIC_CAPS ("audio/x-celt")
70 static GstBuffer *gst_rtp_celt_depay_process (GstRTPBaseDepayload * depayload,
72 static gboolean gst_rtp_celt_depay_setcaps (GstRTPBaseDepayload * depayload,
75 #define gst_rtp_celt_depay_parent_class parent_class
76 G_DEFINE_TYPE (GstRtpCELTDepay, gst_rtp_celt_depay,
77 GST_TYPE_RTP_BASE_DEPAYLOAD);
78 GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpceltdepay, "rtpceltdepay",
79 GST_RANK_SECONDARY, GST_TYPE_RTP_CELT_DEPAY, rtp_element_init (plugin));
81 gst_rtp_celt_depay_class_init (GstRtpCELTDepayClass * klass)
83 GstElementClass *gstelement_class;
84 GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
86 GST_DEBUG_CATEGORY_INIT (rtpceltdepay_debug, "rtpceltdepay", 0,
87 "CELT RTP Depayloader");
89 gstelement_class = (GstElementClass *) klass;
90 gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;
92 gst_element_class_add_static_pad_template (gstelement_class,
93 &gst_rtp_celt_depay_src_template);
94 gst_element_class_add_static_pad_template (gstelement_class,
95 &gst_rtp_celt_depay_sink_template);
97 gst_element_class_set_static_metadata (gstelement_class,
98 "RTP CELT depayloader", "Codec/Depayloader/Network/RTP",
99 "Extracts CELT audio from RTP packets",
100 "Wim Taymans <wim.taymans@gmail.com>");
102 gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_celt_depay_process;
103 gstrtpbasedepayload_class->set_caps = gst_rtp_celt_depay_setcaps;
107 gst_rtp_celt_depay_init (GstRtpCELTDepay * rtpceltdepay)
112 * vendor string (len bytes),
113 * user_len 4 (0) bytes LE
115 static const gchar gst_rtp_celt_comment[] =
116 "\045\0\0\0Depayloaded with GStreamer celtdepay\0\0\0\0";
119 gst_rtp_celt_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
121 GstStructure *structure;
122 GstRtpCELTDepay *rtpceltdepay;
123 gint clock_rate, nb_channels = 0, frame_size = 0;
131 rtpceltdepay = GST_RTP_CELT_DEPAY (depayload);
133 structure = gst_caps_get_structure (caps, 0);
135 if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
137 depayload->clock_rate = clock_rate;
139 if ((params = gst_structure_get_string (structure, "encoding-params")))
140 nb_channels = atoi (params);
142 nb_channels = DEFAULT_CHANNELS;
144 if ((params = gst_structure_get_string (structure, "frame-size")))
145 frame_size = atoi (params);
147 frame_size = DEFAULT_FRAMESIZE;
148 rtpceltdepay->frame_size = frame_size;
150 GST_DEBUG_OBJECT (depayload, "clock-rate=%d channels=%d frame-size=%d",
151 clock_rate, nb_channels, frame_size);
153 /* construct minimal header and comment packet for the decoder */
154 buf = gst_buffer_new_and_alloc (60);
155 gst_buffer_map (buf, &map, GST_MAP_WRITE);
157 memcpy (ptr, "CELT ", 8);
159 memcpy (ptr, "1.1.12", 7);
161 GST_WRITE_UINT32_LE (ptr, 0x80000006); /* version */
163 GST_WRITE_UINT32_LE (ptr, 56); /* header_size */
165 GST_WRITE_UINT32_LE (ptr, clock_rate); /* rate */
167 GST_WRITE_UINT32_LE (ptr, nb_channels); /* channels */
169 GST_WRITE_UINT32_LE (ptr, frame_size); /* frame-size */
171 GST_WRITE_UINT32_LE (ptr, -1); /* overlap */
173 GST_WRITE_UINT32_LE (ptr, -1); /* bytes_per_packet */
175 GST_WRITE_UINT32_LE (ptr, 0); /* extra headers */
176 gst_buffer_unmap (buf, &map);
178 srccaps = gst_caps_new_empty_simple ("audio/x-celt");
179 res = gst_pad_set_caps (depayload->srcpad, srccaps);
180 gst_caps_unref (srccaps);
182 gst_rtp_base_depayload_push (GST_RTP_BASE_DEPAYLOAD (rtpceltdepay), buf);
184 buf = gst_buffer_new_and_alloc (sizeof (gst_rtp_celt_comment));
185 gst_buffer_fill (buf, 0, gst_rtp_celt_comment, sizeof (gst_rtp_celt_comment));
187 gst_rtp_base_depayload_push (GST_RTP_BASE_DEPAYLOAD (rtpceltdepay), buf);
194 GST_ERROR_OBJECT (depayload, "no clock-rate specified");
200 gst_rtp_celt_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
202 GstBuffer *outbuf = NULL;
204 guint offset, pos, payload_len, total_size, size;
206 gint clock_rate = 0, frame_size = 0;
207 GstClockTime framesize_ns = 0, timestamp;
209 GstRtpCELTDepay *rtpceltdepay;
211 rtpceltdepay = GST_RTP_CELT_DEPAY (depayload);
212 clock_rate = depayload->clock_rate;
213 frame_size = rtpceltdepay->frame_size;
214 framesize_ns = gst_util_uint64_scale_int (frame_size, GST_SECOND, clock_rate);
216 timestamp = GST_BUFFER_PTS (rtp->buffer);
218 GST_LOG_OBJECT (depayload,
219 "got %" G_GSIZE_FORMAT " bytes, mark %d ts %u seqn %d",
220 gst_buffer_get_size (rtp->buffer), gst_rtp_buffer_get_marker (rtp),
221 gst_rtp_buffer_get_timestamp (rtp), gst_rtp_buffer_get_seq (rtp));
223 GST_LOG_OBJECT (depayload, "got clock-rate=%d, frame_size=%d, "
224 "_ns=%" GST_TIME_FORMAT ", timestamp=%" GST_TIME_FORMAT, clock_rate,
225 frame_size, GST_TIME_ARGS (framesize_ns), GST_TIME_ARGS (timestamp));
227 payload = gst_rtp_buffer_get_payload (rtp);
228 payload_len = gst_rtp_buffer_get_payload_len (rtp);
230 /* first count how many bytes are consumed by the size headers and make offset
231 * point to the first data byte */
234 while (total_size < payload_len) {
236 s = payload[offset++];
241 /* offset is now pointing to the payload */
244 while (total_size < payload_len) {
253 outbuf = gst_rtp_buffer_get_payload_subbuffer (rtp, offset, size);
256 if (frame_size != -1 && clock_rate != -1) {
257 GST_BUFFER_PTS (outbuf) = timestamp + framesize_ns * n;
258 GST_BUFFER_DURATION (outbuf) = framesize_ns;
260 GST_LOG_OBJECT (depayload, "push timestamp=%"
261 GST_TIME_FORMAT ", duration=%" GST_TIME_FORMAT,
262 GST_TIME_ARGS (GST_BUFFER_PTS (outbuf)),
263 GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)));
265 gst_rtp_drop_non_audio_meta (depayload, outbuf);
267 gst_rtp_base_depayload_push (depayload, outbuf);