2 * Opus Payloader Gst Element
4 * @author: Danilo Cesar Lemes de Paula <danilo.cesar@collabora.co.uk>
6 * This library is free software; you can redistribute it and/or
7 * modify it under the terms of the GNU Library General Public
8 * License as published by the Free Software Foundation; either
9 * version 2 of the License, or (at your option) any later version.
11 * This library is distributed in the hope that it will be useful,
12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14 * Library General Public License for more details.
16 * You should have received a copy of the GNU Library General Public
17 * License along with this library; if not, write to the
18 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
19 * Boston, MA 02110-1301, USA.
28 #include <gst/rtp/gstrtpbuffer.h>
30 #include "gstrtpopuspay.h"
32 GST_DEBUG_CATEGORY_STATIC (rtpopuspay_debug);
33 #define GST_CAT_DEFAULT (rtpopuspay_debug)
36 static GstStaticPadTemplate gst_rtp_opus_pay_sink_template =
37 GST_STATIC_PAD_TEMPLATE ("sink",
40 GST_STATIC_CAPS ("audio/x-opus, multistream = (boolean) FALSE")
43 static GstStaticPadTemplate gst_rtp_opus_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) 48000, "
51 "encoding-params = (string) \"2\", "
52 "encoding-name = (string) { \"OPUS\", \"X-GST-OPUS-DRAFT-SPITTKA-00\" }")
55 static gboolean gst_rtp_opus_pay_setcaps (GstRTPBasePayload * payload,
57 static GstCaps *gst_rtp_opus_pay_getcaps (GstRTPBasePayload * payload,
58 GstPad * pad, GstCaps * filter);
59 static GstFlowReturn gst_rtp_opus_pay_handle_buffer (GstRTPBasePayload *
60 payload, GstBuffer * buffer);
62 G_DEFINE_TYPE (GstRtpOPUSPay, gst_rtp_opus_pay, GST_TYPE_RTP_BASE_PAYLOAD);
65 gst_rtp_opus_pay_class_init (GstRtpOPUSPayClass * klass)
67 GstRTPBasePayloadClass *gstbasertppayload_class;
68 GstElementClass *element_class;
70 gstbasertppayload_class = (GstRTPBasePayloadClass *) klass;
71 element_class = GST_ELEMENT_CLASS (klass);
73 gstbasertppayload_class->set_caps = gst_rtp_opus_pay_setcaps;
74 gstbasertppayload_class->get_caps = gst_rtp_opus_pay_getcaps;
75 gstbasertppayload_class->handle_buffer = gst_rtp_opus_pay_handle_buffer;
77 gst_element_class_add_pad_template (element_class,
78 gst_static_pad_template_get (&gst_rtp_opus_pay_src_template));
79 gst_element_class_add_pad_template (element_class,
80 gst_static_pad_template_get (&gst_rtp_opus_pay_sink_template));
82 gst_element_class_set_static_metadata (element_class,
84 "Codec/Payloader/Network/RTP",
85 "Puts Opus audio in RTP packets",
86 "Danilo Cesar Lemes de Paula <danilo.cesar@collabora.co.uk>");
88 GST_DEBUG_CATEGORY_INIT (rtpopuspay_debug, "rtpopuspay", 0,
89 "Opus RTP Payloader");
93 gst_rtp_opus_pay_init (GstRtpOPUSPay * rtpopuspay)
98 gst_rtp_opus_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
105 const char *sprop_stereo = NULL;
106 char *sprop_maxcapturerate = NULL;
108 src_caps = gst_pad_get_allowed_caps (GST_RTP_BASE_PAYLOAD_SRCPAD (payload));
110 src_caps = gst_caps_make_writable (src_caps);
111 src_caps = gst_caps_truncate (src_caps);
112 s = gst_caps_get_structure (src_caps, 0);
113 gst_structure_fixate_field_string (s, "encoding-name", "OPUS");
114 encoding_name = g_strdup (gst_structure_get_string (s, "encoding-name"));
115 gst_caps_unref (src_caps);
117 encoding_name = g_strdup ("X-GST-OPUS-DRAFT-SPITTKA-00");
120 s = gst_caps_get_structure (caps, 0);
121 if (gst_structure_get_int (s, "channels", &channels)) {
123 GST_ERROR_OBJECT (payload,
124 "More than 2 channels with multistream=FALSE is invalid");
126 } else if (channels == 2) {
133 if (gst_structure_get_int (s, "rate", &rate)) {
134 sprop_maxcapturerate = g_strdup_printf ("%d", rate);
137 gst_rtp_base_payload_set_options (payload, "audio", FALSE,
138 encoding_name, 48000);
139 g_free (encoding_name);
141 if (sprop_maxcapturerate && sprop_stereo) {
143 gst_rtp_base_payload_set_outcaps (payload, "sprop-maxcapturerate",
144 G_TYPE_STRING, sprop_maxcapturerate, "sprop-stereo", G_TYPE_STRING,
146 } else if (sprop_maxcapturerate) {
148 gst_rtp_base_payload_set_outcaps (payload, "sprop-maxcapturerate",
149 G_TYPE_STRING, sprop_maxcapturerate, NULL);
150 } else if (sprop_stereo) {
152 gst_rtp_base_payload_set_outcaps (payload, "sprop-stereo",
153 G_TYPE_STRING, sprop_stereo, NULL);
155 res = gst_rtp_base_payload_set_outcaps (payload, NULL);
158 g_free (sprop_maxcapturerate);
164 gst_rtp_opus_pay_handle_buffer (GstRTPBasePayload * basepayload,
168 GstClockTime pts, dts, duration;
170 pts = GST_BUFFER_PTS (buffer);
171 dts = GST_BUFFER_DTS (buffer);
172 duration = GST_BUFFER_DURATION (buffer);
174 outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);
175 outbuf = gst_buffer_append (outbuf, buffer);
177 GST_BUFFER_PTS (outbuf) = pts;
178 GST_BUFFER_DTS (outbuf) = dts;
179 GST_BUFFER_DURATION (outbuf) = duration;
182 return gst_rtp_base_payload_push (basepayload, outbuf);
186 gst_rtp_opus_pay_getcaps (GstRTPBasePayload * payload,
187 GstPad * pad, GstCaps * filter)
189 GstCaps *caps, *peercaps, *tcaps;
193 if (pad == GST_RTP_BASE_PAYLOAD_SRCPAD (payload))
195 GST_RTP_BASE_PAYLOAD_CLASS (gst_rtp_opus_pay_parent_class)->get_caps
196 (payload, pad, filter);
198 tcaps = gst_pad_get_pad_template_caps (GST_RTP_BASE_PAYLOAD_SRCPAD (payload));
199 peercaps = gst_pad_peer_query_caps (GST_RTP_BASE_PAYLOAD_SRCPAD (payload),
201 gst_caps_unref (tcaps);
204 GST_RTP_BASE_PAYLOAD_CLASS (gst_rtp_opus_pay_parent_class)->get_caps
205 (payload, pad, filter);
207 if (gst_caps_is_empty (peercaps))
210 caps = gst_pad_get_pad_template_caps (GST_RTP_BASE_PAYLOAD_SINKPAD (payload));
212 s = gst_caps_get_structure (peercaps, 0);
213 stereo = gst_structure_get_string (s, "stereo");
214 if (stereo != NULL) {
215 caps = gst_caps_make_writable (caps);
217 if (!strcmp (stereo, "1")) {
218 GstCaps *caps2 = gst_caps_copy (caps);
220 gst_caps_set_simple (caps, "channels", G_TYPE_INT, 2, NULL);
221 gst_caps_set_simple (caps2, "channels", G_TYPE_INT, 1, NULL);
222 caps = gst_caps_merge (caps, caps2);
223 } else if (!strcmp (stereo, "0")) {
224 GstCaps *caps2 = gst_caps_copy (caps);
226 gst_caps_set_simple (caps, "channels", G_TYPE_INT, 1, NULL);
227 gst_caps_set_simple (caps2, "channels", G_TYPE_INT, 2, NULL);
228 caps = gst_caps_merge (caps, caps2);
231 gst_caps_unref (peercaps);
234 GstCaps *tmp = gst_caps_intersect_full (caps, filter,
235 GST_CAPS_INTERSECT_FIRST);
236 gst_caps_unref (caps);
240 GST_DEBUG_OBJECT (payload, "Returning caps: %" GST_PTR_FORMAT, caps);