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 <gst/audio/audio.h>
29 #include "gstrtpsbcdepay.h"
30 #include "gstrtputils.h"
32 GST_DEBUG_CATEGORY_STATIC (rtpsbcdepay_debug);
33 #define GST_CAT_DEFAULT (rtpsbcdepay_debug)
35 static GstStaticPadTemplate gst_rtp_sbc_depay_src_template =
36 GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS,
37 GST_STATIC_CAPS ("audio/x-sbc, "
38 "rate = (int) { 16000, 32000, 44100, 48000 }, "
39 "channels = (int) [ 1, 2 ], "
40 "mode = (string) { mono, dual, stereo, joint }, "
41 "blocks = (int) { 4, 8, 12, 16 }, "
42 "subbands = (int) { 4, 8 }, "
43 "allocation-method = (string) { snr, loudness }, "
44 "bitpool = (int) [ 2, 64 ]")
47 static GstStaticPadTemplate gst_rtp_sbc_depay_sink_template =
48 GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
49 GST_STATIC_CAPS ("application/x-rtp, "
50 "media = (string) audio,"
51 "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
52 "clock-rate = (int) { 16000, 32000, 44100, 48000 },"
53 "encoding-name = (string) SBC")
56 #define gst_rtp_sbc_depay_parent_class parent_class
57 G_DEFINE_TYPE (GstRtpSbcDepay, gst_rtp_sbc_depay, GST_TYPE_RTP_BASE_DEPAYLOAD);
59 static void gst_rtp_sbc_depay_finalize (GObject * object);
61 static gboolean gst_rtp_sbc_depay_setcaps (GstRTPBaseDepayload * base,
63 static GstBuffer *gst_rtp_sbc_depay_process (GstRTPBaseDepayload * base,
67 gst_rtp_sbc_depay_class_init (GstRtpSbcDepayClass * klass)
69 GstRTPBaseDepayloadClass *gstbasertpdepayload_class =
70 GST_RTP_BASE_DEPAYLOAD_CLASS (klass);
71 GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
72 GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
74 gobject_class->finalize = gst_rtp_sbc_depay_finalize;
76 gstbasertpdepayload_class->set_caps = gst_rtp_sbc_depay_setcaps;
77 gstbasertpdepayload_class->process_rtp_packet = gst_rtp_sbc_depay_process;
79 gst_element_class_add_static_pad_template (element_class,
80 &gst_rtp_sbc_depay_src_template);
81 gst_element_class_add_static_pad_template (element_class,
82 &gst_rtp_sbc_depay_sink_template);
84 GST_DEBUG_CATEGORY_INIT (rtpsbcdepay_debug, "rtpsbcdepay", 0,
85 "SBC Audio RTP Depayloader");
87 gst_element_class_set_static_metadata (element_class,
88 "RTP SBC audio depayloader",
89 "Codec/Depayloader/Network/RTP",
90 "Extracts SBC audio from RTP packets",
91 "Arun Raghavan <arun.raghavan@collabora.co.uk>");
95 gst_rtp_sbc_depay_init (GstRtpSbcDepay * rtpsbcdepay)
97 rtpsbcdepay->adapter = gst_adapter_new ();
101 gst_rtp_sbc_depay_finalize (GObject * object)
103 GstRtpSbcDepay *depay = GST_RTP_SBC_DEPAY (object);
105 gst_object_unref (depay->adapter);
107 G_OBJECT_CLASS (parent_class)->finalize (object);
110 /* FIXME: This duplicates similar functionality rtpsbcpay, but there isn't a
111 * simple way to consolidate the two. This is best done by moving the function
112 * to the codec-utils library in gst-plugins-base when these elements move to
115 gst_rtp_sbc_depay_get_params (GstRtpSbcDepay * depay, const guint8 * data,
116 gint size, int *framelen, int *samples)
118 int blocks, channel_mode, channels, subbands, bitpool;
122 /* Not enough data for the header */
127 if (data[0] != 0x9c) {
128 GST_WARNING_OBJECT (depay, "Bad packet: couldn't find syncword");
132 blocks = (data[1] >> 4) & 0x3;
133 blocks = (blocks + 1) * 4;
134 channel_mode = (data[1] >> 2) & 0x3;
135 channels = channel_mode ? 2 : 1;
136 subbands = (data[1] & 0x1);
137 subbands = (subbands + 1) * 4;
140 length = 4 + ((4 * subbands * channels) / 8);
142 if (channel_mode == 0 || channel_mode == 1) {
143 /* Mono || Dual channel */
144 length += ((blocks * channels * bitpool)
145 + 4 /* round up */ ) / 8;
147 /* Stereo || Joint stereo */
148 gboolean joint = (channel_mode == 3);
150 length += ((joint * subbands) + (blocks * bitpool)
151 + 4 /* round up */ ) / 8;
155 *samples = blocks * subbands;
161 gst_rtp_sbc_depay_setcaps (GstRTPBaseDepayload * base, GstCaps * caps)
163 GstRtpSbcDepay *depay = GST_RTP_SBC_DEPAY (base);
164 GstStructure *structure;
165 GstCaps *outcaps, *oldcaps;
167 structure = gst_caps_get_structure (caps, 0);
169 if (!gst_structure_get_int (structure, "clock-rate", &depay->rate))
172 outcaps = gst_caps_new_simple ("audio/x-sbc", "rate", G_TYPE_INT,
175 gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (base), outcaps);
177 oldcaps = gst_pad_get_current_caps (GST_RTP_BASE_DEPAYLOAD_SINKPAD (base));
178 if (oldcaps && !gst_caps_can_intersect (oldcaps, caps)) {
179 /* Caps have changed, flush old data */
180 gst_adapter_clear (depay->adapter);
183 gst_caps_unref (outcaps);
188 GST_WARNING_OBJECT (depay, "Can't support the caps we got: %"
189 GST_PTR_FORMAT, caps);
194 gst_rtp_sbc_depay_process (GstRTPBaseDepayload * base, GstRTPBuffer * rtp)
196 GstRtpSbcDepay *depay = GST_RTP_SBC_DEPAY (base);
197 GstBuffer *data = NULL;
199 gboolean fragment, start, last;
204 GST_LOG_OBJECT (depay, "Got %" G_GSIZE_FORMAT " bytes",
205 gst_buffer_get_size (rtp->buffer));
207 if (gst_rtp_buffer_get_marker (rtp)) {
208 /* Marker isn't supposed to be set */
209 GST_WARNING_OBJECT (depay, "Marker bit was set");
213 payload = gst_rtp_buffer_get_payload (rtp);
214 payload_len = gst_rtp_buffer_get_payload_len (rtp);
216 fragment = payload[0] & 0x80;
217 start = payload[0] & 0x40;
218 last = payload[0] & 0x20;
219 nframes = payload[0] & 0x0f;
224 data = gst_rtp_buffer_get_payload_subbuffer (rtp, 1, -1);
227 /* Got a packet with a fragment */
228 GST_LOG_OBJECT (depay, "Got fragment");
230 if (start && gst_adapter_available (depay->adapter)) {
231 GST_WARNING_OBJECT (depay, "Missing last fragment");
232 gst_adapter_clear (depay->adapter);
234 } else if (!start && !gst_adapter_available (depay->adapter)) {
235 GST_WARNING_OBJECT (depay, "Missing start fragment");
236 gst_buffer_unref (data);
241 gst_adapter_push (depay->adapter, data);
244 data = gst_adapter_take_buffer (depay->adapter,
245 gst_adapter_available (depay->adapter));
246 gst_rtp_drop_meta (GST_ELEMENT_CAST (depay), data,
247 g_quark_from_static_string (GST_META_TAG_AUDIO_STR));
253 gint framelen, samples;
255 GST_LOG_OBJECT (depay, "Got %d frames", nframes);
257 if (gst_rtp_sbc_depay_get_params (depay, payload,
258 payload_len, &framelen, &samples) < 0) {
259 gst_adapter_clear (depay->adapter);
263 GST_LOG_OBJECT (depay, "Got payload of %d", payload_len);
265 if (nframes * framelen > (gint) payload_len) {
266 GST_WARNING_OBJECT (depay, "Short packet");
268 } else if (nframes * framelen < (gint) payload_len) {
269 GST_WARNING_OBJECT (depay, "Junk at end of packet");
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);