+2005-11-17 Edgard Lima <edgard.lima@indt.org.br>
+
+ * gst/rtp/Makefile.am
+ * gst/rtp/gstrtp.c
+ * gst/rtp/gstrtpg711enc.c: (gst_rtpg711enc_src_template),
+ (gst_rtpg711enc_class_init), (gst_rtpg711enc_init),
+ (gst_rtpg711enc_finalize), (gst_rtpg711enc_setcaps),
+ (gst_rtpg711enc_flush), (gst_rtpg711enc_handle_buffer):
+ * gst/rtp/gstrtpg711enc.h:
+ * gst/rtp/gstrtpg711dec.c: (gst_rtpg711dec_sink_template):
+ * gst/rtp/gstrtpspeexenc.c:
+ * gst/rtp/gstrtpspeexenc.h:
+ * gst/rtp/gstrtpspeexdec.c:
+ * gst/rtp/gstrtpspeexdec.h:
+ Created Speex payloader and depayloader; Optimize G711 payloader to
+ use adapter and send packets until MTU size.
+
2005-11-16 Wim Taymans <wim@fluendo.com>
* check/elements/matroskamux.c: (setup_src_pad), (setup_sink_pad):
gstrtph263enc.c \
gstasteriskh263.c \
gstrtpmp4venc.c \
- gstrtpmp4vdec.c
-
+ gstrtpmp4vdec.c \
+ gstrtpspeexenc.c \
+ gstrtpspeexdec.c
+
#gstrtpL16enc.c gstrtpL16parse.c gstrtpgsmenc.c gstrtpgsmparse.c
if HAVE_WINSOCK2_H
gstrtpmp4vdec.h \
gstrtpdec.h \
gstrtph263enc.h \
- gstasteriskh263.h
+ gstasteriskh263.h \
+ gstrtpspeexenc.h \
+ gstrtpspeexdec.h
#include "gstasteriskh263.h"
#include "gstrtpmp4venc.h"
#include "gstrtpmp4vdec.h"
+#include "gstrtpspeexenc.h"
+#include "gstrtpspeexdec.h"
static gboolean
plugin_init (GstPlugin * plugin)
if (!gst_rtpmp4vdec_plugin_init (plugin))
return FALSE;
+ if (!gst_rtpspeexenc_plugin_init (plugin))
+ return FALSE;
+
+ if (!gst_rtpspeexdec_plugin_init (plugin))
+ return FALSE;
+
return TRUE;
}
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\", "
- "payload = (int) 0, "
+ "payload = (int) [ 0, 255 ], "
"clock-rate = (int) 8000, "
"encoding-name = (string) \"PCMU\"; "
"application/x-rtp, "
"media = (string) \"audio\", "
- "payload = (int) 8, "
+ "payload = (int) [ 0, 255 ], "
"clock-rate = (int) 8000, " "encoding-name = (string) \"PCMA\"")
);
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\", "
- "payload = (int) 0, "
+ "payload = (int) [ 0, 255 ], "
"clock-rate = (int) 8000, "
"encoding-name = (string) \"PCMU\"; "
"application/x-rtp, "
"media = (string) \"audio\", "
- "payload = (int) 8, "
+ "payload = (int) [ 0, 255 ], "
"clock-rate = (int) 8000, " "encoding-name = (string) \"PCMA\"")
);
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\", "
- "payload = (int) 0, "
+ "payload = (int) " GST_RTP_PAYLOAD_PCMU_STRING ", "
"clock-rate = (int) 8000, "
"encoding-name = (string) \"PCMU\"; "
"application/x-rtp, "
"media = (string) \"audio\", "
- "payload = (int) 8, "
+ "payload = (int) " GST_RTP_PAYLOAD_PCMA_STRING ", "
"clock-rate = (int) 8000, " "encoding-name = (string) \"PCMA\"")
);
GstCaps * caps);
static GstFlowReturn gst_rtpg711enc_handle_buffer (GstBaseRTPPayload * payload,
GstBuffer * buffer);
+static void gst_rtpg711enc_finalize (GObject * object);
GST_BOILERPLATE (GstRtpG711Enc, gst_rtpg711enc, GstBaseRTPPayload,
GST_TYPE_BASE_RTP_PAYLOAD);
gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
parent_class = g_type_class_ref (GST_TYPE_BASE_RTP_PAYLOAD);
+ gobject_class->finalize = gst_rtpg711enc_finalize;
gstbasertppayload_class->set_caps = gst_rtpg711enc_setcaps;
gstbasertppayload_class->handle_buffer = gst_rtpg711enc_handle_buffer;
static void
gst_rtpg711enc_init (GstRtpG711Enc * rtpg711enc, GstRtpG711EncClass * klass)
{
+ rtpg711enc->adapter = gst_adapter_new ();
GST_BASE_RTP_PAYLOAD (rtpg711enc)->clock_rate = 8000;
}
+static void
+gst_rtpg711enc_finalize (GObject * object)
+{
+ GstRtpG711Enc *rtpg711enc;
+
+ rtpg711enc = GST_RTP_G711_ENC (object);
+
+ g_object_unref (rtpg711enc->adapter);
+ rtpg711enc->adapter = NULL;
+
+ G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
static gboolean
gst_rtpg711enc_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
{
stname = gst_structure_get_name (structure);
if (0 == strcmp ("audio/x-mulaw", stname)) {
- gst_basertppayload_set_options (payload, "audio", TRUE, "PCMU", 8000);
+ payload->pt = GST_RTP_PAYLOAD_PCMU;
+ gst_basertppayload_set_options (payload, "audio", FALSE, "PCMU", 8000);
} else if (0 == strcmp ("audio/x-alaw", stname)) {
- gst_basertppayload_set_options (payload, "audio", TRUE, "PCMA", 8000);
+ payload->pt = GST_RTP_PAYLOAD_PCMA;
+ gst_basertppayload_set_options (payload, "audio", FALSE, "PCMA", 8000);
} else {
return FALSE;
}
}
static GstFlowReturn
+gst_rtpg711enc_flush (GstRtpG711Enc * rtpg711enc)
+{
+ guint avail;
+ GstBuffer *outbuf;
+ GstFlowReturn ret;
+
+ /* the data available in the adapter is either smaller
+ * than the MTU or bigger. In the case it is smaller, the complete
+ * adapter contents can be put in one packet. */
+ avail = gst_adapter_available (rtpg711enc->adapter);
+
+ ret = GST_FLOW_OK;
+
+ while (avail > 0) {
+ guint towrite;
+ guint8 *payload;
+ guint8 *data;
+ guint payload_len;
+ guint packet_len;
+
+ /* this will be the total lenght of the packet */
+ packet_len = gst_rtpbuffer_calc_packet_len (avail, 0, 0);
+ /* fill one MTU or all available bytes */
+ towrite = MIN (packet_len, GST_BASE_RTP_PAYLOAD_MTU (rtpg711enc));
+ /* this is the payload length */
+ payload_len = gst_rtpbuffer_calc_payload_len (towrite, 0, 0);
+ /* create buffer to hold the payload */
+ outbuf = gst_rtpbuffer_new_allocate (payload_len, 0, 0);
+
+ /* copy payload */
+ gst_rtpbuffer_set_payload_type (outbuf,
+ GST_BASE_RTP_PAYLOAD_PT (rtpg711enc));
+ payload = gst_rtpbuffer_get_payload (outbuf);
+ data = (guint8 *) gst_adapter_peek (rtpg711enc->adapter, payload_len);
+ memcpy (payload, data, payload_len);
+ gst_adapter_flush (rtpg711enc->adapter, payload_len);
+
+ avail -= payload_len;
+
+ GST_BUFFER_TIMESTAMP (outbuf) = rtpg711enc->first_ts;
+ ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpg711enc), outbuf);
+ }
+
+ return ret;
+}
+
+static GstFlowReturn
gst_rtpg711enc_handle_buffer (GstBaseRTPPayload * basepayload,
GstBuffer * buffer)
{
GstRtpG711Enc *rtpg711enc;
- guint size, payload_len;
- GstBuffer *outbuf;
- guint8 *payload, *data;
- GstClockTime timestamp;
+ guint size, packet_len, avail;
GstFlowReturn ret;
+ GstClockTime duration;
rtpg711enc = GST_RTP_G711_ENC (basepayload);
size = GST_BUFFER_SIZE (buffer);
- timestamp = GST_BUFFER_TIMESTAMP (buffer);
+ duration = GST_BUFFER_TIMESTAMP (buffer);
- /* FIXME, only one G711 frame per RTP packet for now */
- payload_len = size;
-
- outbuf = gst_rtpbuffer_new_allocate (payload_len, 0, 0);
- /* FIXME, assert for now */
- g_assert (payload_len <= GST_BASE_RTP_PAYLOAD_MTU (rtpg711enc));
-
- /* copy timestamp */
- GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
- /* get payload */
- payload = gst_rtpbuffer_get_payload (outbuf);
-
- data = GST_BUFFER_DATA (buffer);
+ avail = gst_adapter_available (rtpg711enc->adapter);
+ if (avail == 0) {
+ rtpg711enc->first_ts = GST_BUFFER_TIMESTAMP (buffer);
+ rtpg711enc->duration = 0;
+ }
- /* copy data in payload */
- memcpy (&payload[0], data, size);
+ /* get packet length of data and see if we exceeded MTU. */
+ packet_len = gst_rtpbuffer_calc_packet_len (avail + size, 0, 0);
- gst_buffer_unref (buffer);
+ /* if this buffer is going to overflow the packet, flush what we
+ * have. */
+ if (gst_basertppayload_is_filled (basepayload,
+ packet_len, rtpg711enc->duration + duration)) {
+ ret = gst_rtpg711enc_flush (rtpg711enc);
+ rtpg711enc->first_ts = GST_BUFFER_TIMESTAMP (buffer);
+ rtpg711enc->duration = 0;
+ } else {
+ ret = GST_FLOW_OK;
+ }
- ret = gst_basertppayload_push (basepayload, outbuf);
+ gst_adapter_push (rtpg711enc->adapter, buffer);
+ rtpg711enc->duration += duration;
return ret;
}
#include <gst/gst.h>
#include <gst/rtp/gstbasertppayload.h>
+#include <gst/base/gstadapter.h>
G_BEGIN_DECLS
struct _GstRtpG711Enc
{
GstBaseRTPPayload payload;
-
- gint frequency;
+ GstAdapter *adapter;
+
+ GstClockTime first_ts;
+ GstClockTime duration;
};
struct _GstRtpG711EncClass
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\", "
- "payload = (int) 0, "
+ "payload = (int) " GST_RTP_PAYLOAD_PCMU_STRING ", "
"clock-rate = (int) 8000, "
"encoding-name = (string) \"PCMU\"; "
"application/x-rtp, "
"media = (string) \"audio\", "
- "payload = (int) 8, "
+ "payload = (int) " GST_RTP_PAYLOAD_PCMA_STRING ", "
"clock-rate = (int) 8000, " "encoding-name = (string) \"PCMA\"")
);
GstCaps * caps);
static GstFlowReturn gst_rtpg711enc_handle_buffer (GstBaseRTPPayload * payload,
GstBuffer * buffer);
+static void gst_rtpg711enc_finalize (GObject * object);
GST_BOILERPLATE (GstRtpG711Enc, gst_rtpg711enc, GstBaseRTPPayload,
GST_TYPE_BASE_RTP_PAYLOAD);
gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
parent_class = g_type_class_ref (GST_TYPE_BASE_RTP_PAYLOAD);
+ gobject_class->finalize = gst_rtpg711enc_finalize;
gstbasertppayload_class->set_caps = gst_rtpg711enc_setcaps;
gstbasertppayload_class->handle_buffer = gst_rtpg711enc_handle_buffer;
static void
gst_rtpg711enc_init (GstRtpG711Enc * rtpg711enc, GstRtpG711EncClass * klass)
{
+ rtpg711enc->adapter = gst_adapter_new ();
GST_BASE_RTP_PAYLOAD (rtpg711enc)->clock_rate = 8000;
}
+static void
+gst_rtpg711enc_finalize (GObject * object)
+{
+ GstRtpG711Enc *rtpg711enc;
+
+ rtpg711enc = GST_RTP_G711_ENC (object);
+
+ g_object_unref (rtpg711enc->adapter);
+ rtpg711enc->adapter = NULL;
+
+ G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
static gboolean
gst_rtpg711enc_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
{
stname = gst_structure_get_name (structure);
if (0 == strcmp ("audio/x-mulaw", stname)) {
- gst_basertppayload_set_options (payload, "audio", TRUE, "PCMU", 8000);
+ payload->pt = GST_RTP_PAYLOAD_PCMU;
+ gst_basertppayload_set_options (payload, "audio", FALSE, "PCMU", 8000);
} else if (0 == strcmp ("audio/x-alaw", stname)) {
- gst_basertppayload_set_options (payload, "audio", TRUE, "PCMA", 8000);
+ payload->pt = GST_RTP_PAYLOAD_PCMA;
+ gst_basertppayload_set_options (payload, "audio", FALSE, "PCMA", 8000);
} else {
return FALSE;
}
}
static GstFlowReturn
+gst_rtpg711enc_flush (GstRtpG711Enc * rtpg711enc)
+{
+ guint avail;
+ GstBuffer *outbuf;
+ GstFlowReturn ret;
+
+ /* the data available in the adapter is either smaller
+ * than the MTU or bigger. In the case it is smaller, the complete
+ * adapter contents can be put in one packet. */
+ avail = gst_adapter_available (rtpg711enc->adapter);
+
+ ret = GST_FLOW_OK;
+
+ while (avail > 0) {
+ guint towrite;
+ guint8 *payload;
+ guint8 *data;
+ guint payload_len;
+ guint packet_len;
+
+ /* this will be the total lenght of the packet */
+ packet_len = gst_rtpbuffer_calc_packet_len (avail, 0, 0);
+ /* fill one MTU or all available bytes */
+ towrite = MIN (packet_len, GST_BASE_RTP_PAYLOAD_MTU (rtpg711enc));
+ /* this is the payload length */
+ payload_len = gst_rtpbuffer_calc_payload_len (towrite, 0, 0);
+ /* create buffer to hold the payload */
+ outbuf = gst_rtpbuffer_new_allocate (payload_len, 0, 0);
+
+ /* copy payload */
+ gst_rtpbuffer_set_payload_type (outbuf,
+ GST_BASE_RTP_PAYLOAD_PT (rtpg711enc));
+ payload = gst_rtpbuffer_get_payload (outbuf);
+ data = (guint8 *) gst_adapter_peek (rtpg711enc->adapter, payload_len);
+ memcpy (payload, data, payload_len);
+ gst_adapter_flush (rtpg711enc->adapter, payload_len);
+
+ avail -= payload_len;
+
+ GST_BUFFER_TIMESTAMP (outbuf) = rtpg711enc->first_ts;
+ ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpg711enc), outbuf);
+ }
+
+ return ret;
+}
+
+static GstFlowReturn
gst_rtpg711enc_handle_buffer (GstBaseRTPPayload * basepayload,
GstBuffer * buffer)
{
GstRtpG711Enc *rtpg711enc;
- guint size, payload_len;
- GstBuffer *outbuf;
- guint8 *payload, *data;
- GstClockTime timestamp;
+ guint size, packet_len, avail;
GstFlowReturn ret;
+ GstClockTime duration;
rtpg711enc = GST_RTP_G711_ENC (basepayload);
size = GST_BUFFER_SIZE (buffer);
- timestamp = GST_BUFFER_TIMESTAMP (buffer);
+ duration = GST_BUFFER_TIMESTAMP (buffer);
- /* FIXME, only one G711 frame per RTP packet for now */
- payload_len = size;
-
- outbuf = gst_rtpbuffer_new_allocate (payload_len, 0, 0);
- /* FIXME, assert for now */
- g_assert (payload_len <= GST_BASE_RTP_PAYLOAD_MTU (rtpg711enc));
-
- /* copy timestamp */
- GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
- /* get payload */
- payload = gst_rtpbuffer_get_payload (outbuf);
-
- data = GST_BUFFER_DATA (buffer);
+ avail = gst_adapter_available (rtpg711enc->adapter);
+ if (avail == 0) {
+ rtpg711enc->first_ts = GST_BUFFER_TIMESTAMP (buffer);
+ rtpg711enc->duration = 0;
+ }
- /* copy data in payload */
- memcpy (&payload[0], data, size);
+ /* get packet length of data and see if we exceeded MTU. */
+ packet_len = gst_rtpbuffer_calc_packet_len (avail + size, 0, 0);
- gst_buffer_unref (buffer);
+ /* if this buffer is going to overflow the packet, flush what we
+ * have. */
+ if (gst_basertppayload_is_filled (basepayload,
+ packet_len, rtpg711enc->duration + duration)) {
+ ret = gst_rtpg711enc_flush (rtpg711enc);
+ rtpg711enc->first_ts = GST_BUFFER_TIMESTAMP (buffer);
+ rtpg711enc->duration = 0;
+ } else {
+ ret = GST_FLOW_OK;
+ }
- ret = gst_basertppayload_push (basepayload, outbuf);
+ gst_adapter_push (rtpg711enc->adapter, buffer);
+ rtpg711enc->duration += duration;
return ret;
}
#include <gst/gst.h>
#include <gst/rtp/gstbasertppayload.h>
+#include <gst/base/gstadapter.h>
G_BEGIN_DECLS
struct _GstRtpG711Enc
{
GstBaseRTPPayload payload;
-
- gint frequency;
+ GstAdapter *adapter;
+
+ GstClockTime first_ts;
+ GstClockTime duration;
};
struct _GstRtpG711EncClass
--- /dev/null
+/* GStreamer
+ * Copyright (C) <2005> Edgard Lima <edgard.lima@indt.org.br>
+ *
+ * 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
+ */
+
+#ifdef HAVE_CONFIG_H
+# include "config.h"
+#endif
+
+#include <string.h>
+#include <gst/rtp/gstrtpbuffer.h>
+#include "gstrtpspeexdec.h"
+
+/* elementfactory information */
+static GstElementDetails gst_rtp_speexdec_details = {
+ "RTP packet parser",
+ "Codec/Parser/Network",
+ "Extracts Speex audio from RTP packets",
+ "Edgard Lima <edgard.lima@indt.org.br>"
+};
+
+/* RtpSPEEXDec signals and args */
+enum
+{
+ /* FILL ME */
+ LAST_SIGNAL
+};
+
+enum
+{
+ ARG_0
+};
+
+static GstStaticPadTemplate gst_rtpspeexdec_sink_template =
+GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("application/x-rtp, "
+ "media = (string) \"audio\", "
+ "payload = (int) [ 96, 127 ], "
+ "clock-rate = (int) [6000, 48000], "
+ "encoding-name = (string) \"speex\", "
+ "encoding-params = (string) \"1\"")
+ );
+
+static GstStaticPadTemplate gst_rtpspeexdec_src_template =
+GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/x-speex")
+ );
+
+static GstBuffer *gst_rtpspeexdec_process (GstBaseRTPDepayload * depayload,
+ GstBuffer * buf);
+static gboolean gst_rtpspeexdec_setcaps (GstBaseRTPDepayload * depayload,
+ GstCaps * caps);
+
+GST_BOILERPLATE (GstRtpSPEEXDec, gst_rtpspeexdec, GstBaseRTPDepayload,
+ GST_TYPE_BASE_RTP_DEPAYLOAD);
+
+static void
+gst_rtpspeexdec_base_init (gpointer klass)
+{
+ GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&gst_rtpspeexdec_src_template));
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&gst_rtpspeexdec_sink_template));
+ gst_element_class_set_details (element_class, &gst_rtp_speexdec_details);
+}
+
+static void
+gst_rtpspeexdec_class_init (GstRtpSPEEXDecClass * klass)
+{
+ GObjectClass *gobject_class;
+ GstElementClass *gstelement_class;
+ GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
+
+ gobject_class = (GObjectClass *) klass;
+ gstelement_class = (GstElementClass *) klass;
+ gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
+
+ parent_class = g_type_class_ref (GST_TYPE_BASE_RTP_DEPAYLOAD);
+
+ gstbasertpdepayload_class->process = gst_rtpspeexdec_process;
+ gstbasertpdepayload_class->set_caps = gst_rtpspeexdec_setcaps;
+}
+
+static void
+gst_rtpspeexdec_init (GstRtpSPEEXDec * rtpspeexdec, GstRtpSPEEXDecClass * klass)
+{
+ GST_BASE_RTP_DEPAYLOAD (rtpspeexdec)->clock_rate = 8000;
+}
+
+static gboolean
+gst_rtpspeexdec_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
+{
+ GstCaps *srccaps;
+ gboolean ret;
+
+ srccaps = gst_static_pad_template_get_caps (&gst_rtpspeexdec_src_template);
+ ret = gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps);
+
+ gst_caps_unref (srccaps);
+ return ret;
+}
+
+static GstBuffer *
+gst_rtpspeexdec_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
+{
+ GstBuffer *outbuf = NULL;
+ gint payload_len;
+ guint8 *payload;
+
+ GST_DEBUG ("process : got %d bytes, mark %d ts %u seqn %d",
+ GST_BUFFER_SIZE (buf),
+ gst_rtpbuffer_get_marker (buf),
+ gst_rtpbuffer_get_timestamp (buf), gst_rtpbuffer_get_seq (buf));
+
+ payload_len = gst_rtpbuffer_get_payload_len (buf);
+ payload = gst_rtpbuffer_get_payload (buf);
+
+ outbuf = gst_buffer_new_and_alloc (payload_len);
+ memcpy (GST_BUFFER_DATA (outbuf), payload, payload_len);
+ return outbuf;
+}
+
+gboolean
+gst_rtpspeexdec_plugin_init (GstPlugin * plugin)
+{
+ return gst_element_register (plugin, "rtpspeexdec",
+ GST_RANK_NONE, GST_TYPE_RTP_SPEEX_DEC);
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) <2005> Edgard Lima <edgard.lima@indt.org.br>
+ *
+ * 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
+ */
+
+#ifndef __GST_RTP_SPEEX_DEC_H__
+#define __GST_RTP_SPEEX_DEC_H__
+
+#include <gst/gst.h>
+#include <gst/rtp/gstbasertpdepayload.h>
+
+G_BEGIN_DECLS
+
+typedef struct _GstRtpSPEEXDec GstRtpSPEEXDec;
+typedef struct _GstRtpSPEEXDecClass GstRtpSPEEXDecClass;
+
+#define GST_TYPE_RTP_SPEEX_DEC \
+ (gst_rtpspeexdec_get_type())
+#define GST_RTP_SPEEX_DEC(obj) \
+ (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_SPEEX_DEC,GstRtpSPEEXDec))
+#define GST_RTP_SPEEX_DEC_CLASS(klass) \
+ (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_SPEEX_DEC,GstRtpSPEEXDec))
+#define GST_IS_RTP_SPEEX_DEC(obj) \
+ (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_SPEEX_DEC))
+#define GST_IS_RTP_SPEEX_DEC_CLASS(obj) \
+ (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_SPEEX_DEC))
+
+struct _GstRtpSPEEXDec
+{
+ GstBaseRTPDepayload depayload;
+};
+
+struct _GstRtpSPEEXDecClass
+{
+ GstBaseRTPDepayloadClass parent_class;
+};
+
+gboolean gst_rtpspeexdec_plugin_init (GstPlugin * plugin);
+
+G_END_DECLS
+
+#endif /* __GST_RTP_SPEEX_DEC_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) <2005> Edgard Lima <edgard.lima@indt.org.br>
+ *
+ * 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
+ */
+
+#ifdef HAVE_CONFIG_H
+# include "config.h"
+#endif
+
+#include <string.h>
+#include <gst/rtp/gstrtpbuffer.h>
+#include "gstrtpspeexdec.h"
+
+/* elementfactory information */
+static GstElementDetails gst_rtp_speexdec_details = {
+ "RTP packet parser",
+ "Codec/Parser/Network",
+ "Extracts Speex audio from RTP packets",
+ "Edgard Lima <edgard.lima@indt.org.br>"
+};
+
+/* RtpSPEEXDec signals and args */
+enum
+{
+ /* FILL ME */
+ LAST_SIGNAL
+};
+
+enum
+{
+ ARG_0
+};
+
+static GstStaticPadTemplate gst_rtpspeexdec_sink_template =
+GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("application/x-rtp, "
+ "media = (string) \"audio\", "
+ "payload = (int) [ 96, 127 ], "
+ "clock-rate = (int) [6000, 48000], "
+ "encoding-name = (string) \"speex\", "
+ "encoding-params = (string) \"1\"")
+ );
+
+static GstStaticPadTemplate gst_rtpspeexdec_src_template =
+GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/x-speex")
+ );
+
+static GstBuffer *gst_rtpspeexdec_process (GstBaseRTPDepayload * depayload,
+ GstBuffer * buf);
+static gboolean gst_rtpspeexdec_setcaps (GstBaseRTPDepayload * depayload,
+ GstCaps * caps);
+
+GST_BOILERPLATE (GstRtpSPEEXDec, gst_rtpspeexdec, GstBaseRTPDepayload,
+ GST_TYPE_BASE_RTP_DEPAYLOAD);
+
+static void
+gst_rtpspeexdec_base_init (gpointer klass)
+{
+ GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&gst_rtpspeexdec_src_template));
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&gst_rtpspeexdec_sink_template));
+ gst_element_class_set_details (element_class, &gst_rtp_speexdec_details);
+}
+
+static void
+gst_rtpspeexdec_class_init (GstRtpSPEEXDecClass * klass)
+{
+ GObjectClass *gobject_class;
+ GstElementClass *gstelement_class;
+ GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
+
+ gobject_class = (GObjectClass *) klass;
+ gstelement_class = (GstElementClass *) klass;
+ gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
+
+ parent_class = g_type_class_ref (GST_TYPE_BASE_RTP_DEPAYLOAD);
+
+ gstbasertpdepayload_class->process = gst_rtpspeexdec_process;
+ gstbasertpdepayload_class->set_caps = gst_rtpspeexdec_setcaps;
+}
+
+static void
+gst_rtpspeexdec_init (GstRtpSPEEXDec * rtpspeexdec, GstRtpSPEEXDecClass * klass)
+{
+ GST_BASE_RTP_DEPAYLOAD (rtpspeexdec)->clock_rate = 8000;
+}
+
+static gboolean
+gst_rtpspeexdec_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
+{
+ GstCaps *srccaps;
+ gboolean ret;
+
+ srccaps = gst_static_pad_template_get_caps (&gst_rtpspeexdec_src_template);
+ ret = gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps);
+
+ gst_caps_unref (srccaps);
+ return ret;
+}
+
+static GstBuffer *
+gst_rtpspeexdec_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
+{
+ GstBuffer *outbuf = NULL;
+ gint payload_len;
+ guint8 *payload;
+
+ GST_DEBUG ("process : got %d bytes, mark %d ts %u seqn %d",
+ GST_BUFFER_SIZE (buf),
+ gst_rtpbuffer_get_marker (buf),
+ gst_rtpbuffer_get_timestamp (buf), gst_rtpbuffer_get_seq (buf));
+
+ payload_len = gst_rtpbuffer_get_payload_len (buf);
+ payload = gst_rtpbuffer_get_payload (buf);
+
+ outbuf = gst_buffer_new_and_alloc (payload_len);
+ memcpy (GST_BUFFER_DATA (outbuf), payload, payload_len);
+ return outbuf;
+}
+
+gboolean
+gst_rtpspeexdec_plugin_init (GstPlugin * plugin)
+{
+ return gst_element_register (plugin, "rtpspeexdec",
+ GST_RANK_NONE, GST_TYPE_RTP_SPEEX_DEC);
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) <2005> Edgard Lima <edgard.lima@indt.org.br>
+ *
+ * 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
+ */
+
+#ifndef __GST_RTP_SPEEX_DEC_H__
+#define __GST_RTP_SPEEX_DEC_H__
+
+#include <gst/gst.h>
+#include <gst/rtp/gstbasertpdepayload.h>
+
+G_BEGIN_DECLS
+
+typedef struct _GstRtpSPEEXDec GstRtpSPEEXDec;
+typedef struct _GstRtpSPEEXDecClass GstRtpSPEEXDecClass;
+
+#define GST_TYPE_RTP_SPEEX_DEC \
+ (gst_rtpspeexdec_get_type())
+#define GST_RTP_SPEEX_DEC(obj) \
+ (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_SPEEX_DEC,GstRtpSPEEXDec))
+#define GST_RTP_SPEEX_DEC_CLASS(klass) \
+ (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_SPEEX_DEC,GstRtpSPEEXDec))
+#define GST_IS_RTP_SPEEX_DEC(obj) \
+ (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_SPEEX_DEC))
+#define GST_IS_RTP_SPEEX_DEC_CLASS(obj) \
+ (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_SPEEX_DEC))
+
+struct _GstRtpSPEEXDec
+{
+ GstBaseRTPDepayload depayload;
+};
+
+struct _GstRtpSPEEXDecClass
+{
+ GstBaseRTPDepayloadClass parent_class;
+};
+
+gboolean gst_rtpspeexdec_plugin_init (GstPlugin * plugin);
+
+G_END_DECLS
+
+#endif /* __GST_RTP_SPEEX_DEC_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) <2005> Edgard Lima <edgard.lima@indt.org.br>
+ *
+ * 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
+ */
+
+#ifdef HAVE_CONFIG_H
+# include "config.h"
+#endif
+
+#include <stdlib.h>
+#include <string.h>
+#include <gst/rtp/gstrtpbuffer.h>
+
+#include "gstrtpspeexenc.h"
+
+/* elementfactory information */
+static GstElementDetails gst_rtpspeexenc_details = {
+ "RTP packet parser",
+ "Codec/Encoder/Network",
+ "Encodes Speex audio into a RTP packet",
+ "Edgard Lima <edgard.lima@indt.org.br>"
+};
+
+static GstStaticPadTemplate gst_rtpspeexenc_sink_template =
+GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/x-speex")
+ );
+
+static GstStaticPadTemplate gst_rtpspeexenc_src_template =
+GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("application/x-rtp, " "media = (string) \"audio\", " "payload = (int) 110, " /* guaranties compatibility with Linphone
+ Could be [96,127] See page 34 at http://www.ietf.org/rfc/rfc3551.txt */
+ "clock-rate = (int) [6000, 48000], "
+ "encoding-name = (string) \"speex\", "
+ "encoding-params = (string) \"1\"")
+ );
+
+static gboolean gst_rtpspeexenc_setcaps (GstBaseRTPPayload * payload,
+ GstCaps * caps);
+static GstFlowReturn gst_rtpspeexenc_handle_buffer (GstBaseRTPPayload * payload,
+ GstBuffer * buffer);
+
+GST_BOILERPLATE (GstRtpSPEEXEnc, gst_rtpspeexenc, GstBaseRTPPayload,
+ GST_TYPE_BASE_RTP_PAYLOAD);
+
+static void
+gst_rtpspeexenc_base_init (gpointer klass)
+{
+ GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&gst_rtpspeexenc_sink_template));
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&gst_rtpspeexenc_src_template));
+ gst_element_class_set_details (element_class, &gst_rtpspeexenc_details);
+}
+
+static void
+gst_rtpspeexenc_class_init (GstRtpSPEEXEncClass * klass)
+{
+ GObjectClass *gobject_class;
+ GstElementClass *gstelement_class;
+ GstBaseRTPPayloadClass *gstbasertppayload_class;
+
+ gobject_class = (GObjectClass *) klass;
+ gstelement_class = (GstElementClass *) klass;
+ gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
+
+ parent_class = g_type_class_ref (GST_TYPE_BASE_RTP_PAYLOAD);
+
+ gstbasertppayload_class->set_caps = gst_rtpspeexenc_setcaps;
+ gstbasertppayload_class->handle_buffer = gst_rtpspeexenc_handle_buffer;
+}
+
+static void
+gst_rtpspeexenc_init (GstRtpSPEEXEnc * rtpspeexenc, GstRtpSPEEXEncClass * klass)
+{
+ GST_BASE_RTP_PAYLOAD (rtpspeexenc)->clock_rate = 8000;
+ GST_BASE_RTP_PAYLOAD_PT (rtpspeexenc) = 110; /* Create String */
+}
+
+static gboolean
+gst_rtpspeexenc_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
+{
+ gst_basertppayload_set_options (payload, "audio", FALSE, "speex", 8000);
+ gst_basertppayload_set_outcaps (payload, NULL);
+
+ return TRUE;
+}
+
+static GstFlowReturn
+gst_rtpspeexenc_handle_buffer (GstBaseRTPPayload * basepayload,
+ GstBuffer * buffer)
+{
+ GstRtpSPEEXEnc *rtpspeexenc;
+ guint size, payload_len;
+ GstBuffer *outbuf;
+ guint8 *payload, *data;
+ GstClockTime timestamp;
+ GstFlowReturn ret;
+
+ rtpspeexenc = GST_RTP_SPEEX_ENC (basepayload);
+
+ size = GST_BUFFER_SIZE (buffer);
+ timestamp = GST_BUFFER_TIMESTAMP (buffer);
+
+ /* FIXME, only one SPEEX frame per RTP packet for now */
+ payload_len = size;
+
+ outbuf = gst_rtpbuffer_new_allocate (payload_len, 0, 0);
+ /* FIXME, assert for now */
+ g_assert (payload_len <= GST_BASE_RTP_PAYLOAD_MTU (rtpspeexenc));
+
+ /* copy timestamp */
+ GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
+ /* get payload */
+ payload = gst_rtpbuffer_get_payload (outbuf);
+
+ data = GST_BUFFER_DATA (buffer);
+
+ /* copy data in payload */
+ memcpy (&payload[0], data, size);
+
+ gst_buffer_unref (buffer);
+
+ ret = gst_basertppayload_push (basepayload, outbuf);
+
+ return ret;
+}
+
+gboolean
+gst_rtpspeexenc_plugin_init (GstPlugin * plugin)
+{
+ return gst_element_register (plugin, "rtpspeexenc",
+ GST_RANK_NONE, GST_TYPE_RTP_SPEEX_ENC);
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) <2005> Edgard Lima <edgard.lima@indt.org.br>
+ *
+ * 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
+ */
+
+
+#ifndef __GST_RTP_SPEEX_ENC_H__
+#define __GST_RTP_SPEEX_ENC_H__
+
+#include <gst/gst.h>
+#include <gst/rtp/gstbasertppayload.h>
+
+G_BEGIN_DECLS
+
+typedef struct _GstRtpSPEEXEnc GstRtpSPEEXEnc;
+typedef struct _GstRtpSPEEXEncClass GstRtpSPEEXEncClass;
+
+#define GST_TYPE_RTP_SPEEX_ENC \
+ (gst_rtpspeexenc_get_type())
+#define GST_RTP_SPEEX_ENC(obj) \
+ (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_SPEEX_ENC,GstRtpSPEEXEnc))
+#define GST_RTP_SPEEX_ENC_CLASS(klass) \
+ (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_SPEEX_ENC,GstRtpSPEEXEnc))
+#define GST_IS_RTP_SPEEX_ENC(obj) \
+ (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_SPEEX_ENC))
+#define GST_IS_RTP_SPEEX_ENC_CLASS(obj) \
+ (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_SPEEX_ENC))
+
+struct _GstRtpSPEEXEnc
+{
+ GstBaseRTPPayload payload;
+};
+
+struct _GstRtpSPEEXEncClass
+{
+ GstBaseRTPPayloadClass parent_class;
+};
+
+gboolean gst_rtpspeexenc_plugin_init (GstPlugin * plugin);
+
+G_END_DECLS
+
+#endif /* __GST_RTP_SPEEX_ENC_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) <2005> Edgard Lima <edgard.lima@indt.org.br>
+ *
+ * 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
+ */
+
+#ifdef HAVE_CONFIG_H
+# include "config.h"
+#endif
+
+#include <stdlib.h>
+#include <string.h>
+#include <gst/rtp/gstrtpbuffer.h>
+
+#include "gstrtpspeexenc.h"
+
+/* elementfactory information */
+static GstElementDetails gst_rtpspeexenc_details = {
+ "RTP packet parser",
+ "Codec/Encoder/Network",
+ "Encodes Speex audio into a RTP packet",
+ "Edgard Lima <edgard.lima@indt.org.br>"
+};
+
+static GstStaticPadTemplate gst_rtpspeexenc_sink_template =
+GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/x-speex")
+ );
+
+static GstStaticPadTemplate gst_rtpspeexenc_src_template =
+GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("application/x-rtp, " "media = (string) \"audio\", " "payload = (int) 110, " /* guaranties compatibility with Linphone
+ Could be [96,127] See page 34 at http://www.ietf.org/rfc/rfc3551.txt */
+ "clock-rate = (int) [6000, 48000], "
+ "encoding-name = (string) \"speex\", "
+ "encoding-params = (string) \"1\"")
+ );
+
+static gboolean gst_rtpspeexenc_setcaps (GstBaseRTPPayload * payload,
+ GstCaps * caps);
+static GstFlowReturn gst_rtpspeexenc_handle_buffer (GstBaseRTPPayload * payload,
+ GstBuffer * buffer);
+
+GST_BOILERPLATE (GstRtpSPEEXEnc, gst_rtpspeexenc, GstBaseRTPPayload,
+ GST_TYPE_BASE_RTP_PAYLOAD);
+
+static void
+gst_rtpspeexenc_base_init (gpointer klass)
+{
+ GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&gst_rtpspeexenc_sink_template));
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&gst_rtpspeexenc_src_template));
+ gst_element_class_set_details (element_class, &gst_rtpspeexenc_details);
+}
+
+static void
+gst_rtpspeexenc_class_init (GstRtpSPEEXEncClass * klass)
+{
+ GObjectClass *gobject_class;
+ GstElementClass *gstelement_class;
+ GstBaseRTPPayloadClass *gstbasertppayload_class;
+
+ gobject_class = (GObjectClass *) klass;
+ gstelement_class = (GstElementClass *) klass;
+ gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
+
+ parent_class = g_type_class_ref (GST_TYPE_BASE_RTP_PAYLOAD);
+
+ gstbasertppayload_class->set_caps = gst_rtpspeexenc_setcaps;
+ gstbasertppayload_class->handle_buffer = gst_rtpspeexenc_handle_buffer;
+}
+
+static void
+gst_rtpspeexenc_init (GstRtpSPEEXEnc * rtpspeexenc, GstRtpSPEEXEncClass * klass)
+{
+ GST_BASE_RTP_PAYLOAD (rtpspeexenc)->clock_rate = 8000;
+ GST_BASE_RTP_PAYLOAD_PT (rtpspeexenc) = 110; /* Create String */
+}
+
+static gboolean
+gst_rtpspeexenc_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
+{
+ gst_basertppayload_set_options (payload, "audio", FALSE, "speex", 8000);
+ gst_basertppayload_set_outcaps (payload, NULL);
+
+ return TRUE;
+}
+
+static GstFlowReturn
+gst_rtpspeexenc_handle_buffer (GstBaseRTPPayload * basepayload,
+ GstBuffer * buffer)
+{
+ GstRtpSPEEXEnc *rtpspeexenc;
+ guint size, payload_len;
+ GstBuffer *outbuf;
+ guint8 *payload, *data;
+ GstClockTime timestamp;
+ GstFlowReturn ret;
+
+ rtpspeexenc = GST_RTP_SPEEX_ENC (basepayload);
+
+ size = GST_BUFFER_SIZE (buffer);
+ timestamp = GST_BUFFER_TIMESTAMP (buffer);
+
+ /* FIXME, only one SPEEX frame per RTP packet for now */
+ payload_len = size;
+
+ outbuf = gst_rtpbuffer_new_allocate (payload_len, 0, 0);
+ /* FIXME, assert for now */
+ g_assert (payload_len <= GST_BASE_RTP_PAYLOAD_MTU (rtpspeexenc));
+
+ /* copy timestamp */
+ GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
+ /* get payload */
+ payload = gst_rtpbuffer_get_payload (outbuf);
+
+ data = GST_BUFFER_DATA (buffer);
+
+ /* copy data in payload */
+ memcpy (&payload[0], data, size);
+
+ gst_buffer_unref (buffer);
+
+ ret = gst_basertppayload_push (basepayload, outbuf);
+
+ return ret;
+}
+
+gboolean
+gst_rtpspeexenc_plugin_init (GstPlugin * plugin)
+{
+ return gst_element_register (plugin, "rtpspeexenc",
+ GST_RANK_NONE, GST_TYPE_RTP_SPEEX_ENC);
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) <2005> Edgard Lima <edgard.lima@indt.org.br>
+ *
+ * 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
+ */
+
+
+#ifndef __GST_RTP_SPEEX_ENC_H__
+#define __GST_RTP_SPEEX_ENC_H__
+
+#include <gst/gst.h>
+#include <gst/rtp/gstbasertppayload.h>
+
+G_BEGIN_DECLS
+
+typedef struct _GstRtpSPEEXEnc GstRtpSPEEXEnc;
+typedef struct _GstRtpSPEEXEncClass GstRtpSPEEXEncClass;
+
+#define GST_TYPE_RTP_SPEEX_ENC \
+ (gst_rtpspeexenc_get_type())
+#define GST_RTP_SPEEX_ENC(obj) \
+ (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_SPEEX_ENC,GstRtpSPEEXEnc))
+#define GST_RTP_SPEEX_ENC_CLASS(klass) \
+ (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_SPEEX_ENC,GstRtpSPEEXEnc))
+#define GST_IS_RTP_SPEEX_ENC(obj) \
+ (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_SPEEX_ENC))
+#define GST_IS_RTP_SPEEX_ENC_CLASS(obj) \
+ (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_SPEEX_ENC))
+
+struct _GstRtpSPEEXEnc
+{
+ GstBaseRTPPayload payload;
+};
+
+struct _GstRtpSPEEXEncClass
+{
+ GstBaseRTPPayloadClass parent_class;
+};
+
+gboolean gst_rtpspeexenc_plugin_init (GstPlugin * plugin);
+
+G_END_DECLS
+
+#endif /* __GST_RTP_SPEEX_ENC_H__ */
projects / platform / upstream / gst-plugins-good.git / commitdiff