2 * GStreamer RTP SBC depayloader
4 * Copyright (C) 2012 Collabora Ltd.
5 * @author: Arun Raghavan <arun.raghavan@collabora.co.uk>
7 * This library is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Library General Public
9 * License as published by the Free Software Foundation; either
10 * version 2 of the License, or (at your option) any later version.
12 * This library is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Library General Public License for more details.
17 * You should have received a copy of the GNU Library General Public
18 * License along with this library; if not, write to the
19 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
20 * Boston, MA 02110-1301, USA.
27 #include <gst/rtp/gstrtpbuffer.h>
28 #include "gstrtpsbcdepay.h"
30 GST_DEBUG_CATEGORY_STATIC (rtpsbcdepay_debug);
31 #define GST_CAT_DEFAULT (rtpsbcdepay_debug)
33 static GstStaticPadTemplate gst_rtp_sbc_depay_src_template =
34 GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS,
35 GST_STATIC_CAPS ("audio/x-sbc, "
36 "rate = (int) { 16000, 32000, 44100, 48000 }, "
37 "channels = (int) [ 1, 2 ], "
38 "mode = (string) { mono, dual, stereo, joint }, "
39 "blocks = (int) { 4, 8, 12, 16 }, "
40 "subbands = (int) { 4, 8 }, "
41 "allocation-method = (string) { snr, loudness }, "
42 "bitpool = (int) [ 2, 64 ]")
45 static GstStaticPadTemplate gst_rtp_sbc_depay_sink_template =
46 GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
47 GST_STATIC_CAPS ("application/x-rtp, "
48 "media = (string) audio,"
49 "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
50 "clock-rate = (int) { 16000, 32000, 44100, 48000 },"
51 "encoding-name = (string) SBC")
54 #define gst_rtp_sbc_depay_parent_class parent_class
55 G_DEFINE_TYPE (GstRtpSbcDepay, gst_rtp_sbc_depay, GST_TYPE_RTP_BASE_DEPAYLOAD);
57 static void gst_rtp_sbc_depay_finalize (GObject * object);
59 static gboolean gst_rtp_sbc_depay_setcaps (GstRTPBaseDepayload * base,
61 static GstBuffer *gst_rtp_sbc_depay_process (GstRTPBaseDepayload * base,
65 gst_rtp_sbc_depay_class_init (GstRtpSbcDepayClass * klass)
67 GstRTPBaseDepayloadClass *gstbasertpdepayload_class =
68 GST_RTP_BASE_DEPAYLOAD_CLASS (klass);
69 GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
70 GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
72 gobject_class->finalize = gst_rtp_sbc_depay_finalize;
74 gstbasertpdepayload_class->set_caps = gst_rtp_sbc_depay_setcaps;
75 gstbasertpdepayload_class->process = gst_rtp_sbc_depay_process;
77 gst_element_class_add_pad_template (element_class,
78 gst_static_pad_template_get (&gst_rtp_sbc_depay_src_template));
79 gst_element_class_add_pad_template (element_class,
80 gst_static_pad_template_get (&gst_rtp_sbc_depay_sink_template));
82 GST_DEBUG_CATEGORY_INIT (rtpsbcdepay_debug, "rtpsbcdepay", 0,
83 "SBC Audio RTP Depayloader");
85 gst_element_class_set_static_metadata (element_class,
86 "RTP SBC audio depayloader",
87 "Codec/Depayloader/Network/RTP",
88 "Extracts SBC audio from RTP packets",
89 "Arun Raghavan <arun.raghavan@collabora.co.uk>");
93 gst_rtp_sbc_depay_init (GstRtpSbcDepay * rtpsbcdepay)
95 rtpsbcdepay->adapter = gst_adapter_new ();
99 gst_rtp_sbc_depay_finalize (GObject * object)
101 GstRtpSbcDepay *depay = GST_RTP_SBC_DEPAY (object);
103 gst_object_unref (depay->adapter);
105 G_OBJECT_CLASS (parent_class)->finalize (object);
108 /* FIXME: This duplicates similar functionality rtpsbcpay, but there isn't a
109 * simple way to consolidate the two. This is best done by moving the function
110 * to the codec-utils library in gst-plugins-base when these elements move to
113 gst_rtp_sbc_depay_get_params (GstRtpSbcDepay * depay, const guint8 * data,
114 gint size, int *framelen, int *samples)
116 int blocks, channel_mode, channels, subbands, bitpool;
120 /* Not enough data for the header */
125 if (data[0] != 0x9c) {
126 GST_WARNING_OBJECT (depay, "Bad packet: couldn't find syncword");
130 blocks = (data[1] >> 4) & 0x3;
131 blocks = (blocks + 1) * 4;
132 channel_mode = (data[1] >> 2) & 0x3;
133 channels = channel_mode ? 2 : 1;
134 subbands = (data[1] & 0x1);
135 subbands = (subbands + 1) * 4;
138 length = 4 + ((4 * subbands * channels) / 8);
140 if (channel_mode == 0 || channel_mode == 1) {
141 /* Mono || Dual channel */
142 length += ((blocks * channels * bitpool)
143 + 4 /* round up */ ) / 8;
145 /* Stereo || Joint stereo */
146 gboolean joint = (channel_mode == 3);
148 length += ((joint * subbands) + (blocks * bitpool)
149 + 4 /* round up */ ) / 8;
153 *samples = blocks * subbands;
159 gst_rtp_sbc_depay_setcaps (GstRTPBaseDepayload * base, GstCaps * caps)
161 GstRtpSbcDepay *depay = GST_RTP_SBC_DEPAY (base);
162 GstStructure *structure;
163 GstCaps *outcaps, *oldcaps;
165 structure = gst_caps_get_structure (caps, 0);
167 if (!gst_structure_get_int (structure, "clock-rate", &depay->rate))
170 outcaps = gst_caps_new_simple ("audio/x-sbc", "rate", G_TYPE_INT,
173 gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (base), outcaps);
175 oldcaps = gst_pad_get_current_caps (GST_RTP_BASE_DEPAYLOAD_SINKPAD (base));
176 if (oldcaps && !gst_caps_can_intersect (oldcaps, caps)) {
177 /* Caps have changed, flush old data */
178 gst_adapter_clear (depay->adapter);
181 gst_caps_unref (outcaps);
186 GST_WARNING_OBJECT (depay, "Can't support the caps we got: %"
187 GST_PTR_FORMAT, caps);
192 gst_rtp_sbc_depay_process (GstRTPBaseDepayload * base, GstBuffer * in)
194 GstRtpSbcDepay *depay = GST_RTP_SBC_DEPAY (base);
195 GstBuffer *data = NULL;
196 GstRTPBuffer rtp = { NULL };
198 gboolean fragment, start, last;
203 gst_rtp_buffer_map (in, GST_MAP_READ, &rtp);
205 GST_LOG_OBJECT (depay, "Got %" G_GSIZE_FORMAT " bytes",
206 gst_buffer_get_size (in));
208 if (gst_rtp_buffer_get_marker (&rtp)) {
209 /* Marker isn't supposed to be set */
210 GST_WARNING_OBJECT (depay, "Marker bit was set");
214 payload = gst_rtp_buffer_get_payload (&rtp);
215 payload_len = gst_rtp_buffer_get_payload_len (&rtp);
217 fragment = payload[0] & 0x80;
218 start = payload[0] & 0x40;
219 last = payload[0] & 0x20;
220 nframes = payload[0] & 0x0f;
225 data = gst_rtp_buffer_get_payload_subbuffer (&rtp, 1, -1);
228 /* Got a packet with a fragment */
229 GST_LOG_OBJECT (depay, "Got fragment");
231 if (start && gst_adapter_available (depay->adapter)) {
232 GST_WARNING_OBJECT (depay, "Missing last fragment");
233 gst_adapter_clear (depay->adapter);
235 } else if (!start && !gst_adapter_available (depay->adapter)) {
236 GST_WARNING_OBJECT (depay, "Missing start fragment");
237 gst_buffer_unref (data);
242 gst_adapter_push (depay->adapter, data);
245 data = gst_adapter_take_buffer (depay->adapter,
246 gst_adapter_available (depay->adapter));
252 gint framelen, samples;
254 GST_LOG_OBJECT (depay, "Got %d frames", nframes);
256 if (gst_rtp_sbc_depay_get_params (depay, payload,
257 payload_len, &framelen, &samples) < 0) {
258 gst_adapter_clear (depay->adapter);
262 GST_LOG_OBJECT (depay, "Got payload of %d", payload_len);
264 if (nframes * framelen > (gint) payload_len) {
265 GST_WARNING_OBJECT (depay, "Short packet");
267 } else if (nframes * framelen < (gint) payload_len) {
268 GST_WARNING_OBJECT (depay, "Junk at end of packet");
273 gst_rtp_buffer_unmap (&rtp);
277 GST_ELEMENT_WARNING (depay, STREAM, DECODE,
278 ("Received invalid RTP payload, dropping"), (NULL));
283 gst_rtp_sbc_depay_plugin_init (GstPlugin * plugin)
285 return gst_element_register (plugin, "rtpsbcdepay", GST_RANK_SECONDARY,
286 GST_TYPE_RTP_SBC_DEPAY);