summaryrefslogtreecommitdiff
path: root/gst/rtp/gstrtpg723pay.c
diff options
context:
space:
mode:
Diffstat (limited to 'gst/rtp/gstrtpg723pay.c')
-rw-r--r--gst/rtp/gstrtpg723pay.c316
1 files changed, 316 insertions, 0 deletions
diff --git a/gst/rtp/gstrtpg723pay.c b/gst/rtp/gstrtpg723pay.c
new file mode 100644
index 0000000..3fe906f
--- /dev/null
+++ b/gst/rtp/gstrtpg723pay.c
@@ -0,0 +1,316 @@
+/* GStreamer
+ * Copyright (C) <2007> Nokia Corporation
+ * Copyright (C) <2007> Collabora Ltd
+ * @author: Olivier Crete <olivier.crete@collabora.co.uk>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <string.h>
+#include <gst/rtp/gstrtpbuffer.h>
+#include <gst/base/gstadapter.h>
+
+#include "gstrtpg723pay.h"
+
+#define GST_RTP_PAYLOAD_G723 4
+#define GST_RTP_PAYLOAD_G723_STRING "4"
+
+#define G723_FRAME_DURATION (30 * GST_MSECOND)
+
+static gboolean gst_rtp_g723_pay_set_caps (GstBaseRTPPayload * payload,
+ GstCaps * caps);
+static GstFlowReturn gst_rtp_g723_pay_handle_buffer (GstBaseRTPPayload *
+ payload, GstBuffer * buf);
+
+static GstStaticPadTemplate gst_rtp_g723_pay_sink_template =
+GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/G723, " /* according to RFC 3551 */
+ "channels = (int) 1, " "rate = (int) 8000")
+ );
+
+static GstStaticPadTemplate gst_rtp_g723_pay_src_template =
+ GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("application/x-rtp, "
+ "media = (string) \"audio\", "
+ "payload = (int) " GST_RTP_PAYLOAD_G723_STRING ", "
+ "clock-rate = (int) 8000, "
+ "encoding-name = (string) \"G723\"; "
+ "application/x-rtp, "
+ "media = (string) \"audio\", "
+ "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
+ "clock-rate = (int) 8000, " "encoding-name = (string) \"G723\"")
+ );
+
+static void gst_rtp_g723_pay_finalize (GObject * object);
+
+static GstStateChangeReturn gst_rtp_g723_pay_change_state (GstElement * element,
+ GstStateChange transition);
+
+GST_BOILERPLATE (GstRTPG723Pay, gst_rtp_g723_pay, GstBaseRTPPayload,
+ GST_TYPE_BASE_RTP_PAYLOAD);
+
+static void
+gst_rtp_g723_pay_base_init (gpointer klass)
+{
+ GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+
+ gst_element_class_add_static_pad_template (element_class,
+ &gst_rtp_g723_pay_sink_template);
+ gst_element_class_add_static_pad_template (element_class,
+ &gst_rtp_g723_pay_src_template);
+ gst_element_class_set_details_simple (element_class, "RTP G.723 payloader",
+ "Codec/Payloader/Network/RTP",
+ "Packetize G.723 audio into RTP packets",
+ "Wim Taymans <wim.taymans@gmail.com>");
+}
+
+static void
+gst_rtp_g723_pay_class_init (GstRTPG723PayClass * klass)
+{
+ GObjectClass *gobject_class;
+ GstElementClass *gstelement_class;
+ GstBaseRTPPayloadClass *payload_class;
+
+ gobject_class = (GObjectClass *) klass;
+ gstelement_class = (GstElementClass *) klass;
+ payload_class = (GstBaseRTPPayloadClass *) klass;
+
+ gobject_class->finalize = gst_rtp_g723_pay_finalize;
+
+ gstelement_class->change_state = gst_rtp_g723_pay_change_state;
+
+ payload_class->set_caps = gst_rtp_g723_pay_set_caps;
+ payload_class->handle_buffer = gst_rtp_g723_pay_handle_buffer;
+}
+
+static void
+gst_rtp_g723_pay_init (GstRTPG723Pay * pay, GstRTPG723PayClass * klass)
+{
+ GstBaseRTPPayload *payload = GST_BASE_RTP_PAYLOAD (pay);
+
+ pay->adapter = gst_adapter_new ();
+
+ payload->pt = GST_RTP_PAYLOAD_G723;
+ gst_basertppayload_set_options (payload, "audio", FALSE, "G723", 8000);
+}
+
+static void
+gst_rtp_g723_pay_finalize (GObject * object)
+{
+ GstRTPG723Pay *pay;
+
+ pay = GST_RTP_G723_PAY (object);
+
+ g_object_unref (pay->adapter);
+ pay->adapter = NULL;
+
+ G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+
+static gboolean
+gst_rtp_g723_pay_set_caps (GstBaseRTPPayload * payload, GstCaps * caps)
+{
+ gboolean res;
+ GstStructure *structure;
+ gint pt;
+
+ structure = gst_caps_get_structure (caps, 0);
+ if (!gst_structure_get_int (structure, "payload", &pt))
+ pt = GST_RTP_PAYLOAD_G723;
+
+ payload->pt = pt;
+ payload->dynamic = pt != GST_RTP_PAYLOAD_G723;
+
+ res = gst_basertppayload_set_outcaps (payload, NULL);
+
+ return res;
+}
+
+static GstFlowReturn
+gst_rtp_g723_pay_flush (GstRTPG723Pay * pay)
+{
+ GstBuffer *outbuf;
+ GstFlowReturn ret;
+ guint8 *payload;
+ guint avail;
+
+ avail = gst_adapter_available (pay->adapter);
+
+ outbuf = gst_rtp_buffer_new_allocate (avail, 0, 0);
+ payload = gst_rtp_buffer_get_payload (outbuf);
+
+ GST_BUFFER_TIMESTAMP (outbuf) = pay->timestamp;
+ GST_BUFFER_DURATION (outbuf) = pay->duration;
+
+ /* copy G723 data as payload */
+ gst_adapter_copy (pay->adapter, payload, 0, avail);
+
+ /* flush bytes from adapter */
+ gst_adapter_flush (pay->adapter, avail);
+ pay->timestamp = GST_CLOCK_TIME_NONE;
+ pay->duration = 0;
+
+ /* set discont and marker */
+ if (pay->discont) {
+ GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
+ gst_rtp_buffer_set_marker (outbuf, TRUE);
+ pay->discont = FALSE;
+ }
+
+ ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (pay), outbuf);
+
+ return ret;
+}
+
+/* 00 high-rate speech (6.3 kb/s) 24
+ * 01 low-rate speech (5.3 kb/s) 20
+ * 10 SID frame 4
+ * 11 reserved 0 */
+static const guint size_tab[4] = {
+ 24, 20, 4, 0
+};
+
+static GstFlowReturn
+gst_rtp_g723_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buf)
+{
+ GstFlowReturn ret = GST_FLOW_OK;
+ guint8 *data;
+ guint size;
+ guint8 HDR;
+ GstRTPG723Pay *pay;
+ GstClockTime packet_dur, timestamp;
+ guint payload_len, packet_len;
+
+ pay = GST_RTP_G723_PAY (payload);
+
+ size = GST_BUFFER_SIZE (buf);
+ data = GST_BUFFER_DATA (buf);
+ timestamp = GST_BUFFER_TIMESTAMP (buf);
+
+ if (GST_BUFFER_IS_DISCONT (buf)) {
+ /* flush everything on discont */
+ gst_adapter_clear (pay->adapter);
+ pay->timestamp = GST_CLOCK_TIME_NONE;
+ pay->duration = 0;
+ pay->discont = TRUE;
+ }
+
+ /* should be one of these sizes */
+ if (size != 4 && size != 20 && size != 24)
+ goto invalid_size;
+
+ /* check size by looking at the header bits */
+ HDR = data[0] & 0x3;
+ if (size_tab[HDR] != size)
+ goto wrong_size;
+
+ /* calculate packet size and duration */
+ payload_len = gst_adapter_available (pay->adapter) + size;
+ packet_dur = pay->duration + G723_FRAME_DURATION;
+ packet_len = gst_rtp_buffer_calc_packet_len (payload_len, 0, 0);
+
+ if (gst_basertppayload_is_filled (payload, packet_len, packet_dur)) {
+ /* size or duration would overflow the packet, flush the queued data */
+ ret = gst_rtp_g723_pay_flush (pay);
+ }
+
+ /* update timestamp, we keep the timestamp for the first packet in the adapter
+ * but are able to calculate it from next packets. */
+ if (timestamp != GST_CLOCK_TIME_NONE && pay->timestamp == GST_CLOCK_TIME_NONE) {
+ if (timestamp > pay->duration)
+ pay->timestamp = timestamp - pay->duration;
+ else
+ pay->timestamp = 0;
+ }
+
+ /* add packet to the queue */
+ gst_adapter_push (pay->adapter, buf);
+ pay->duration = packet_dur;
+
+ /* check if we can flush now */
+ if (pay->duration >= payload->min_ptime) {
+ ret = gst_rtp_g723_pay_flush (pay);
+ }
+
+ return ret;
+
+ /* WARNINGS */
+invalid_size:
+ {
+ GST_ELEMENT_WARNING (pay, STREAM, WRONG_TYPE,
+ ("Invalid input buffer size"),
+ ("Input size should be 4, 20 or 24, got %u", size));
+ gst_buffer_unref (buf);
+ return GST_FLOW_OK;
+ }
+wrong_size:
+ {
+ GST_ELEMENT_WARNING (pay, STREAM, WRONG_TYPE,
+ ("Wrong input buffer size"),
+ ("Expected input buffer size %u but got %u", size_tab[HDR], size));
+ gst_buffer_unref (buf);
+ return GST_FLOW_OK;
+ }
+}
+
+static GstStateChangeReturn
+gst_rtp_g723_pay_change_state (GstElement * element, GstStateChange transition)
+{
+ GstStateChangeReturn ret;
+ GstRTPG723Pay *pay;
+
+ pay = GST_RTP_G723_PAY (element);
+
+ switch (transition) {
+ case GST_STATE_CHANGE_READY_TO_PAUSED:
+ gst_adapter_clear (pay->adapter);
+ pay->timestamp = GST_CLOCK_TIME_NONE;
+ pay->duration = 0;
+ pay->discont = TRUE;
+ break;
+ default:
+ break;
+ }
+
+ ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+
+ switch (transition) {
+ case GST_STATE_CHANGE_PAUSED_TO_READY:
+ gst_adapter_clear (pay->adapter);
+ break;
+ default:
+ break;
+ }
+
+ return ret;
+}
+
+/*Plugin init functions*/
+gboolean
+gst_rtp_g723_pay_plugin_init (GstPlugin * plugin)
+{
+ return gst_element_register (plugin, "rtpg723pay", GST_RANK_SECONDARY,
+ gst_rtp_g723_pay_get_type ());
+}
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-24 02:19:09 +0000