Commit 908a9ee6 authored by Tiago Katcipis's avatar Tiago Katcipis Committed by Wim Taymans

rtp: add G723 payloader

Fixes #597823
parent cc277b4a
......@@ -22,6 +22,7 @@ libgstrtp_la_SOURCES = \
gstrtppcmudepay.c \
gstrtppcmupay.c \
gstrtppcmapay.c \
gstrtpg723pay.c \
gstrtpg726pay.c \
gstrtpg726depay.c \
gstrtpg729pay.c \
......@@ -101,6 +102,7 @@ noinst_HEADERS = \
gstrtppcmudepay.h \
gstrtppcmupay.h \
gstrtppcmapay.h \
gstrtpg723pay.h \
gstrtpg726depay.h \
gstrtpg726pay.h \
gstrtpg729depay.h \
......
......@@ -35,6 +35,7 @@
#include "gstrtppcmapay.h"
#include "gstrtppcmadepay.h"
#include "gstrtppcmudepay.h"
#include "gstrtpg723pay.h"
#include "gstrtpg726depay.h"
#include "gstrtpg726pay.h"
#include "gstrtpg729depay.h"
......@@ -115,6 +116,9 @@ plugin_init (GstPlugin * plugin)
if (!gst_rtp_ilbc_depay_plugin_init (plugin))
return FALSE;
if (!gst_rtp_g723_pay_plugin_init (plugin))
return FALSE;
if (!gst_rtp_g726_depay_plugin_init (plugin))
return FALSE;
......
/* 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.
*/
/*
* This payloader assumes that the data will ALWAYS come as zero or more
* 10 bytes frame of audio followed by 0 or 1 2 byte frame of silence.
* Any other buffer format won't work
*/
#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"
/* According to RFC 3551, works only with G723 encoded with 6.3 kb/s high-rate */
#define G723_FRAME_SIZE 24
#define G723B_SID_FRAME_SIZE 4
#define G723_FRAME_DURATION (30 * GST_MSECOND)
#define G723_FRAME_DURATION_MS (30)
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 const GstElementDetails gst_rtp_g723_pay_details =
GST_ELEMENT_DETAILS ("RTP G.723 payloader",
"Codec/Payloader/Network",
"Packetize 6.3kb/s G.723 audio into RTP packets",
"Tiago Katcipis <tiago.katcipis@digitro.com.br>");
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_init (GstRTPG723Pay * pay, GstRTPG723PayClass * klass);
GST_BOILERPLATE (GstRTPG723Pay, gst_rtp_g723_pay, GstBaseRTPAudioPayload,
GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
static void
gst_rtp_g723_pay_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_rtp_g723_pay_sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_g723_pay_src_template));
gst_element_class_set_details (element_class, &gst_rtp_g723_pay_details);
}
static void
gst_rtp_g723_pay_class_init (GstRTPG723PayClass * klass)
{
GstBaseRTPPayloadClass *payload_class = GST_BASE_RTP_PAYLOAD_CLASS (klass);
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);
GstBaseRTPAudioPayload *audiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (pay);
payload->pt = GST_RTP_PAYLOAD_G723;
gst_basertppayload_set_options (payload, "audio", FALSE, "G723", 8000);
gst_base_rtp_audio_payload_set_frame_based (audiopayload);
gst_base_rtp_audio_payload_set_frame_options (audiopayload,
G723_FRAME_DURATION_MS, G723_FRAME_SIZE);
}
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_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buf)
{
GstFlowReturn ret = GST_FLOW_OK;
GstBaseRTPAudioPayload *basertpaudiopayload =
GST_BASE_RTP_AUDIO_PAYLOAD (payload);
GstAdapter *adapter = NULL;
guint payload_len;
const guint8 *data = NULL;
guint available;
guint maxptime_octets = G_MAXUINT;
guint minptime_octets = 0;
guint min_payload_len;
guint max_payload_len;
gboolean use_adapter = FALSE;
available = GST_BUFFER_SIZE (buf);
if (available % G723_FRAME_SIZE != 0 &&
available % G723_FRAME_SIZE != G723B_SID_FRAME_SIZE)
goto invalid_size;
/* max number of bytes based on given ptime, has to be multiple of
* frame_duration */
if (payload->max_ptime != -1) {
guint ptime_ms = payload->max_ptime / 1000000;
maxptime_octets = G723_FRAME_SIZE *
(int) (ptime_ms / G723_FRAME_DURATION_MS);
if (maxptime_octets < G723_FRAME_SIZE) {
GST_WARNING_OBJECT (basertpaudiopayload, "Given ptime %" G_GINT64_FORMAT
" is smaller than minimum %d ns, overwriting to minimum",
payload->max_ptime, G723_FRAME_DURATION_MS);
maxptime_octets = G723_FRAME_SIZE;
}
}
max_payload_len = MIN (
/* MTU max */
(int) (gst_rtp_buffer_calc_payload_len (GST_BASE_RTP_PAYLOAD_MTU
(basertpaudiopayload), 0, 0) / G723_FRAME_SIZE) * G723_FRAME_SIZE,
/* ptime max */
maxptime_octets);
/* min number of bytes based on a given ptime, has to be a multiple
of frame duration */
{
guint64 min_ptime;
g_object_get (G_OBJECT (payload), "min-ptime", &min_ptime, NULL);
min_ptime = min_ptime / 1000000;
minptime_octets = G723_FRAME_SIZE *
(int) (min_ptime / G723_FRAME_DURATION_MS);
}
min_payload_len = MAX (minptime_octets, G723_FRAME_SIZE);
if (min_payload_len > max_payload_len) {
min_payload_len = max_payload_len;
}
GST_DEBUG_OBJECT (basertpaudiopayload,
"Calculated min_payload_len %u and max_payload_len %u",
min_payload_len, max_payload_len);
adapter = gst_base_rtp_audio_payload_get_adapter (basertpaudiopayload);
if (adapter && gst_adapter_available (adapter)) {
/* If there is always data in the adapter, we have to use it */
gst_adapter_push (adapter, buf);
available = gst_adapter_available (adapter);
use_adapter = TRUE;
} else {
/* let's set the base timestamp */
basertpaudiopayload->base_ts = GST_BUFFER_TIMESTAMP (buf);
/* If buffer fits on an RTP packet, let's just push it through */
/* this will check against max_ptime and max_mtu */
if (GST_BUFFER_SIZE (buf) >= min_payload_len &&
GST_BUFFER_SIZE (buf) <= max_payload_len) {
ret = gst_base_rtp_audio_payload_push (basertpaudiopayload,
GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf),
GST_BUFFER_TIMESTAMP (buf));
gst_buffer_unref (buf);
return ret;
}
available = GST_BUFFER_SIZE (buf);
data = (guint8 *) GST_BUFFER_DATA (buf);
}
/* as long as we have full frames */
/* this loop will push all available buffers till the last frame */
while (available >= min_payload_len ||
available % G723_FRAME_SIZE == G723B_SID_FRAME_SIZE) {
guint num;
/* We send as much as we can */
if (available <= max_payload_len) {
payload_len = available;
} else {
payload_len = MIN (max_payload_len,
(available / G723_FRAME_SIZE) * G723_FRAME_SIZE);
}
if (use_adapter) {
data = gst_adapter_peek (adapter, payload_len);
}
ret = gst_base_rtp_audio_payload_push (basertpaudiopayload, data,
payload_len, basertpaudiopayload->base_ts);
num = payload_len / G723_FRAME_SIZE;
basertpaudiopayload->base_ts += G723_FRAME_DURATION * num;
if (use_adapter) {
gst_adapter_flush (adapter, payload_len);
available = gst_adapter_available (adapter);
} else {
available -= payload_len;
data += payload_len;
}
}
if (!use_adapter) {
if (available != 0 && adapter) {
GstBuffer *buf2;
buf2 = gst_buffer_create_sub (buf,
GST_BUFFER_SIZE (buf) - available, available);
gst_adapter_push (adapter, buf2);
} else {
gst_buffer_unref (buf);
}
}
if (adapter) {
g_object_unref (adapter);
}
return ret;
/* ERRORS */
invalid_size:
{
GST_ELEMENT_ERROR (payload, STREAM, WRONG_TYPE,
("Invalid input buffer size"),
("Invalid buffer size, should be a multiple of"
" G723_FRAME_SIZE(24) with an optional G723B_SID_FRAME_SIZE(4)"
" added to it, but it is %u", available));
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
}
/*Plugin init functions*/
gboolean
gst_rtp_g723_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpg723pay", GST_RANK_NONE,
gst_rtp_g723_pay_get_type ());
}
/* GStreamer
* Copyright (C) <2007> Nokia Corporation
* Copyright (C) <2007> Collabora Ltd
*
* 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.
*/
#ifndef __GST_RTP_G723_PAY_H__
#define __GST_RTP_G723_PAY_H__
#include <gst/gst.h>
#include <gst/rtp/gstbasertpaudiopayload.h>
G_BEGIN_DECLS
#define GST_TYPE_RTP_G723_PAY \
(gst_rtp_g723_pay_get_type())
#define GST_RTP_G723_PAY(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_G723_PAY,GstRTPG723Pay))
#define GST_RTP_G723_PAY_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_G723_PAY,GstRTPG723PayClass))
#define GST_IS_RTP_G723_PAY(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_G723_PAY))
#define GST_IS_RTP_G723_PAY_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_G723_PAY))
typedef struct _GstRTPG723Pay GstRTPG723Pay;
typedef struct _GstRTPG723PayClass GstRTPG723PayClass;
struct _GstRTPG723Pay
{
GstBaseRTPAudioPayload audiopayload;
};
struct _GstRTPG723PayClass
{
GstBaseRTPAudioPayloadClass parent_class;
};
gboolean gst_rtp_g723_pay_plugin_init (GstPlugin * plugin);
GType gst_rtp_g723_pay_get_type (void);
G_END_DECLS
#endif /* __GST_RTP_G723_PAY_H__ */
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