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")
59 PROP_IGNORE_TIMESTAMPS,
63 #define DEFAULT_IGNORE_TIMESTAMPS FALSE
65 #define gst_rtp_sbc_depay_parent_class parent_class
66 G_DEFINE_TYPE (GstRtpSbcDepay, gst_rtp_sbc_depay, GST_TYPE_RTP_BASE_DEPAYLOAD);
68 static void gst_rtp_sbc_depay_set_property (GObject * object,
69 guint prop_id, const GValue * value, GParamSpec * pspec);
70 static void gst_rtp_sbc_depay_get_property (GObject * object,
71 guint prop_id, GValue * value, GParamSpec * pspec);
72 static void gst_rtp_sbc_depay_finalize (GObject * object);
74 static gboolean gst_rtp_sbc_depay_setcaps (GstRTPBaseDepayload * base,
76 static GstBuffer *gst_rtp_sbc_depay_process (GstRTPBaseDepayload * base,
80 gst_rtp_sbc_depay_class_init (GstRtpSbcDepayClass * klass)
82 GstRTPBaseDepayloadClass *gstbasertpdepayload_class =
83 GST_RTP_BASE_DEPAYLOAD_CLASS (klass);
84 GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
85 GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
87 gobject_class->finalize = gst_rtp_sbc_depay_finalize;
88 gobject_class->set_property = gst_rtp_sbc_depay_set_property;
89 gobject_class->get_property = gst_rtp_sbc_depay_get_property;
91 g_object_class_install_property (gobject_class, PROP_IGNORE_TIMESTAMPS,
92 g_param_spec_boolean ("ignore-timestamps", "Ignore Timestamps",
93 "Various statistics", DEFAULT_IGNORE_TIMESTAMPS,
94 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
96 gstbasertpdepayload_class->set_caps = gst_rtp_sbc_depay_setcaps;
97 gstbasertpdepayload_class->process_rtp_packet = gst_rtp_sbc_depay_process;
99 gst_element_class_add_static_pad_template (element_class,
100 &gst_rtp_sbc_depay_src_template);
101 gst_element_class_add_static_pad_template (element_class,
102 &gst_rtp_sbc_depay_sink_template);
104 GST_DEBUG_CATEGORY_INIT (rtpsbcdepay_debug, "rtpsbcdepay", 0,
105 "SBC Audio RTP Depayloader");
107 gst_element_class_set_static_metadata (element_class,
108 "RTP SBC audio depayloader",
109 "Codec/Depayloader/Network/RTP",
110 "Extracts SBC audio from RTP packets",
111 "Arun Raghavan <arun.raghavan@collabora.co.uk>");
115 gst_rtp_sbc_depay_init (GstRtpSbcDepay * rtpsbcdepay)
117 rtpsbcdepay->adapter = gst_adapter_new ();
118 rtpsbcdepay->stream_align =
119 gst_audio_stream_align_new (48000, 40 * GST_MSECOND, 1 * GST_SECOND);
120 rtpsbcdepay->ignore_timestamps = DEFAULT_IGNORE_TIMESTAMPS;
124 gst_rtp_sbc_depay_finalize (GObject * object)
126 GstRtpSbcDepay *depay = GST_RTP_SBC_DEPAY (object);
128 gst_audio_stream_align_free (depay->stream_align);
129 gst_object_unref (depay->adapter);
131 G_OBJECT_CLASS (parent_class)->finalize (object);
135 gst_rtp_sbc_depay_set_property (GObject * object,
136 guint prop_id, const GValue * value, GParamSpec * pspec)
138 GstRtpSbcDepay *depay = GST_RTP_SBC_DEPAY (object);
141 case PROP_IGNORE_TIMESTAMPS:
142 depay->ignore_timestamps = g_value_get_boolean (value);
145 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
151 gst_rtp_sbc_depay_get_property (GObject * object,
152 guint prop_id, GValue * value, GParamSpec * pspec)
154 GstRtpSbcDepay *depay = GST_RTP_SBC_DEPAY (object);
157 case PROP_IGNORE_TIMESTAMPS:
158 g_value_set_boolean (value, depay->ignore_timestamps);
161 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
166 /* FIXME: This duplicates similar functionality rtpsbcpay, but there isn't a
167 * simple way to consolidate the two. This is best done by moving the function
168 * to the codec-utils library in gst-plugins-base when these elements move to
171 gst_rtp_sbc_depay_get_params (GstRtpSbcDepay * depay, const guint8 * data,
172 gint size, int *framelen, int *samples)
174 int blocks, channel_mode, channels, subbands, bitpool;
178 /* Not enough data for the header */
183 if (data[0] != 0x9c) {
184 GST_WARNING_OBJECT (depay, "Bad packet: couldn't find syncword");
188 blocks = (data[1] >> 4) & 0x3;
189 blocks = (blocks + 1) * 4;
190 channel_mode = (data[1] >> 2) & 0x3;
191 channels = channel_mode ? 2 : 1;
192 subbands = (data[1] & 0x1);
193 subbands = (subbands + 1) * 4;
196 length = 4 + ((4 * subbands * channels) / 8);
198 if (channel_mode == 0 || channel_mode == 1) {
199 /* Mono || Dual channel */
200 length += ((blocks * channels * bitpool)
201 + 4 /* round up */ ) / 8;
203 /* Stereo || Joint stereo */
204 gboolean joint = (channel_mode == 3);
206 length += ((joint * subbands) + (blocks * bitpool)
207 + 4 /* round up */ ) / 8;
211 *samples = blocks * subbands;
217 gst_rtp_sbc_depay_setcaps (GstRTPBaseDepayload * base, GstCaps * caps)
219 GstRtpSbcDepay *depay = GST_RTP_SBC_DEPAY (base);
220 GstStructure *structure;
221 GstCaps *outcaps, *oldcaps;
223 structure = gst_caps_get_structure (caps, 0);
225 if (!gst_structure_get_int (structure, "clock-rate", &depay->rate))
228 outcaps = gst_caps_new_simple ("audio/x-sbc", "rate", G_TYPE_INT,
231 gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (base), outcaps);
233 oldcaps = gst_pad_get_current_caps (GST_RTP_BASE_DEPAYLOAD_SINKPAD (base));
234 if (oldcaps && !gst_caps_can_intersect (oldcaps, caps)) {
235 /* Caps have changed, flush old data */
236 gst_adapter_clear (depay->adapter);
239 gst_caps_unref (outcaps);
241 gst_caps_unref (oldcaps);
243 /* Reset when the caps are changing */
244 gst_audio_stream_align_set_rate (depay->stream_align, depay->rate);
249 GST_WARNING_OBJECT (depay, "Can't support the caps we got: %"
250 GST_PTR_FORMAT, caps);
255 gst_rtp_sbc_depay_process (GstRTPBaseDepayload * base, GstRTPBuffer * rtp)
257 GstRtpSbcDepay *depay = GST_RTP_SBC_DEPAY (base);
258 GstBuffer *data = NULL;
260 gboolean fragment, start, last;
266 GstClockTime timestamp;
268 GST_LOG_OBJECT (depay, "Got %" G_GSIZE_FORMAT " bytes",
269 gst_buffer_get_size (rtp->buffer));
271 if (gst_rtp_buffer_get_marker (rtp)) {
272 /* Marker isn't supposed to be set */
273 GST_WARNING_OBJECT (depay, "Marker bit was set");
277 timestamp = GST_BUFFER_DTS_OR_PTS (rtp->buffer);
278 if (depay->ignore_timestamps && timestamp == GST_CLOCK_TIME_NONE) {
279 GstClockTime initial_timestamp;
283 gst_audio_stream_align_get_timestamp_at_discont (depay->stream_align);
285 gst_audio_stream_align_get_samples_since_discont (depay->stream_align);
287 if (initial_timestamp == GST_CLOCK_TIME_NONE) {
288 GST_ERROR_OBJECT (depay,
289 "Can only ignore timestamps on streams without valid initial timestamp");
294 initial_timestamp + gst_util_uint64_scale (n_samples, GST_SECOND,
298 payload = gst_rtp_buffer_get_payload (rtp);
299 payload_len = gst_rtp_buffer_get_payload_len (rtp);
301 fragment = payload[0] & 0x80;
302 start = payload[0] & 0x40;
303 last = payload[0] & 0x20;
304 nframes = payload[0] & 0x0f;
309 data = gst_rtp_buffer_get_payload_subbuffer (rtp, 1, -1);
312 /* Got a packet with a fragment */
313 GST_LOG_OBJECT (depay, "Got fragment");
315 if (start && gst_adapter_available (depay->adapter)) {
316 GST_WARNING_OBJECT (depay, "Missing last fragment");
317 gst_adapter_clear (depay->adapter);
319 } else if (!start && !gst_adapter_available (depay->adapter)) {
320 GST_WARNING_OBJECT (depay, "Missing start fragment");
321 gst_buffer_unref (data);
326 gst_adapter_push (depay->adapter, data);
329 gint framelen, samples;
332 data = gst_adapter_take_buffer (depay->adapter,
333 gst_adapter_available (depay->adapter));
334 gst_rtp_drop_non_audio_meta (depay, data);
336 if (gst_buffer_extract (data, 0, &header, 4) != 4 ||
337 gst_rtp_sbc_depay_get_params (depay, header,
338 payload_len, &framelen, &samples) < 0) {
339 gst_buffer_unref (data);
349 GST_LOG_OBJECT (depay, "Got %d frames", nframes);
351 if (gst_rtp_sbc_depay_get_params (depay, payload,
352 payload_len, &framelen, &samples) < 0) {
353 gst_adapter_clear (depay->adapter);
359 GST_LOG_OBJECT (depay, "Got payload of %d", payload_len);
361 if (nframes * framelen > (gint) payload_len) {
362 GST_WARNING_OBJECT (depay, "Short packet");
364 } else if (nframes * framelen < (gint) payload_len) {
365 GST_WARNING_OBJECT (depay, "Junk at end of packet");
369 if (depay->ignore_timestamps && data) {
370 GstClockTime duration;
372 gst_audio_stream_align_process (depay->stream_align,
373 GST_BUFFER_IS_DISCONT (rtp->buffer), timestamp, samples, ×tamp,
376 GST_BUFFER_PTS (data) = timestamp;
377 GST_BUFFER_DTS (data) = GST_CLOCK_TIME_NONE;
378 GST_BUFFER_DURATION (data) = duration;
385 GST_ELEMENT_WARNING (depay, STREAM, DECODE,
386 ("Received invalid RTP payload, dropping"), (NULL));
391 gst_rtp_sbc_depay_plugin_init (GstPlugin * plugin)
393 return gst_element_register (plugin, "rtpsbcdepay", GST_RANK_SECONDARY,
394 GST_TYPE_RTP_SBC_DEPAY);