Commit a956a6ce authored by David Holroyd's avatar David Holroyd Committed by Wim Taymans

rtp: add L24 pay and depayloader

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=707734
parent 478f0dcf
......@@ -50,6 +50,8 @@ libgstrtp_la_SOURCES = \
gstrtpjpegpay.c \
gstrtpL16depay.c \
gstrtpL16pay.c \
gstrtpL24depay.c \
gstrtpL24pay.c \
gstasteriskh263.c \
gstrtpmp1sdepay.c \
gstrtpmp2tdepay.c \
......
......@@ -68,6 +68,8 @@
#include "gstrtpjpegpay.h"
#include "gstrtpL16depay.h"
#include "gstrtpL16pay.h"
#include "gstrtpL24depay.h"
#include "gstrtpL24pay.h"
#include "gstasteriskh263.h"
#include "gstrtpmp1sdepay.h"
#include "gstrtpmp2tdepay.h"
......@@ -236,6 +238,12 @@ plugin_init (GstPlugin * plugin)
if (!gst_rtp_L16_depay_plugin_init (plugin))
return FALSE;
if (!gst_rtp_L24_pay_plugin_init (plugin))
return FALSE;
if (!gst_rtp_L24_depay_plugin_init (plugin))
return FALSE;
if (!gst_asteriskh263_plugin_init (plugin))
return FALSE;
......
/* GStreamer
* Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
*
* 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-rtpL24depay
* @see_also: rtpL24pay
*
* Extract raw audio from RTP packets according to RFC 3190, section 4.
* For detailed information see: http://www.rfc-editor.org/rfc/rfc3190.txt
*
* <refsect2>
* <title>Example pipeline</title>
* |[
* gst-launch udpsrc caps='application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)L24, encoding-params=(string)1, channels=(int)1, payload=(int)96' ! rtpL24depay ! pulsesink
* ]| This example pipeline will depayload an RTP raw audio stream. Refer to
* the rtpL24pay example to create the RTP stream.
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include <stdlib.h>
#include <gst/audio/audio.h>
#include "gstrtpL24depay.h"
#include "gstrtpchannels.h"
GST_DEBUG_CATEGORY_STATIC (rtpL24depay_debug);
#define GST_CAT_DEFAULT (rtpL24depay_debug)
static GstStaticPadTemplate gst_rtp_L24_depay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) S24BE, "
"layout = (string) interleaved, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
);
static GstStaticPadTemplate gst_rtp_L24_depay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\", " "clock-rate = (int) [ 1, MAX ], "
/* "channels = (int) [1, MAX]" */
/* "emphasis = (string) ANY" */
/* "channel-order = (string) ANY" */
"encoding-name = (string) \"L24\";"
"application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) { " GST_RTP_PAYLOAD_DYNAMIC_STRING " },"
"clock-rate = (int) [ 1, MAX ]"
/* "channels = (int) [1, MAX]" */
/* "emphasis = (string) ANY" */
/* "channel-order = (string) ANY" */
)
);
#define gst_rtp_L24_depay_parent_class parent_class
G_DEFINE_TYPE (GstRtpL24Depay, gst_rtp_L24_depay, GST_TYPE_RTP_BASE_DEPAYLOAD);
static gboolean gst_rtp_L24_depay_setcaps (GstRTPBaseDepayload * depayload,
GstCaps * caps);
static GstBuffer *gst_rtp_L24_depay_process (GstRTPBaseDepayload * depayload,
GstBuffer * buf);
static void
gst_rtp_L24_depay_class_init (GstRtpL24DepayClass * klass)
{
GstElementClass *gstelement_class;
GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
gstelement_class = (GstElementClass *) klass;
gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;
gstrtpbasedepayload_class->set_caps = gst_rtp_L24_depay_setcaps;
gstrtpbasedepayload_class->process = gst_rtp_L24_depay_process;
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_rtp_L24_depay_src_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_rtp_L24_depay_sink_template));
gst_element_class_set_static_metadata (gstelement_class,
"RTP audio depayloader", "Codec/Depayloader/Network/RTP",
"Extracts raw 24-bit audio from RTP packets",
"Zeeshan Ali <zak147@yahoo.com>," "Wim Taymans <wim.taymans@gmail.com>,"
"David Holroyd <dave@badgers-in-foil.co.uk>");
GST_DEBUG_CATEGORY_INIT (rtpL24depay_debug, "rtpL24depay", 0,
"Raw Audio RTP Depayloader");
}
static void
gst_rtp_L24_depay_init (GstRtpL24Depay * rtpL24depay)
{
}
static gint
gst_rtp_L24_depay_parse_int (GstStructure * structure, const gchar * field,
gint def)
{
const gchar *str;
gint res;
if ((str = gst_structure_get_string (structure, field)))
return atoi (str);
if (gst_structure_get_int (structure, field, &res))
return res;
return def;
}
static gboolean
gst_rtp_L24_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
{
GstStructure *structure;
GstRtpL24Depay *rtpL24depay;
gint clock_rate, payload;
gint channels;
GstCaps *srccaps;
gboolean res;
const gchar *channel_order;
const GstRTPChannelOrder *order;
GstAudioInfo *info;
rtpL24depay = GST_RTP_L24_DEPAY (depayload);
structure = gst_caps_get_structure (caps, 0);
payload = 96;
gst_structure_get_int (structure, "payload", &payload);
/* no fixed mapping, we need clock-rate */
channels = 0;
clock_rate = 0;
/* caps can overwrite defaults */
clock_rate =
gst_rtp_L24_depay_parse_int (structure, "clock-rate", clock_rate);
if (clock_rate == 0)
goto no_clockrate;
channels =
gst_rtp_L24_depay_parse_int (structure, "encoding-params", channels);
if (channels == 0) {
channels = gst_rtp_L24_depay_parse_int (structure, "channels", channels);
if (channels == 0) {
/* channels defaults to 1 otherwise */
channels = 1;
}
}
depayload->clock_rate = clock_rate;
info = &rtpL24depay->info;
gst_audio_info_init (info);
info->finfo = gst_audio_format_get_info (GST_AUDIO_FORMAT_S24BE);
info->rate = clock_rate;
info->channels = channels;
info->bpf = (info->finfo->width / 8) * channels;
/* add channel positions */
channel_order = gst_structure_get_string (structure, "channel-order");
order = gst_rtp_channels_get_by_order (channels, channel_order);
rtpL24depay->order = order;
if (order) {
memcpy (info->position, order->pos,
sizeof (GstAudioChannelPosition) * channels);
gst_audio_channel_positions_to_valid_order (info->position, info->channels);
} else {
GST_ELEMENT_WARNING (rtpL24depay, STREAM, DECODE,
(NULL), ("Unknown channel order '%s' for %d channels",
GST_STR_NULL (channel_order), channels));
/* create default NONE layout */
gst_rtp_channels_create_default (channels, info->position);
}
srccaps = gst_audio_info_to_caps (info);
res = gst_pad_set_caps (depayload->srcpad, srccaps);
gst_caps_unref (srccaps);
return res;
/* ERRORS */
no_clockrate:
{
GST_ERROR_OBJECT (depayload, "no clock-rate specified");
return FALSE;
}
}
static GstBuffer *
gst_rtp_L24_depay_process (GstRTPBaseDepayload * depayload, GstBuffer * buf)
{
GstRtpL24Depay *rtpL24depay;
GstBuffer *outbuf;
gint payload_len;
gboolean marker;
GstRTPBuffer rtp = { NULL };
rtpL24depay = GST_RTP_L24_DEPAY (depayload);
gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp);
payload_len = gst_rtp_buffer_get_payload_len (&rtp);
if (payload_len <= 0)
goto empty_packet;
GST_DEBUG_OBJECT (rtpL24depay, "got payload of %d bytes", payload_len);
outbuf = gst_rtp_buffer_get_payload_buffer (&rtp);
marker = gst_rtp_buffer_get_marker (&rtp);
if (marker) {
/* mark talk spurt with RESYNC */
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
}
outbuf = gst_buffer_make_writable (outbuf);
if (rtpL24depay->order &&
!gst_audio_buffer_reorder_channels (outbuf,
rtpL24depay->info.finfo->format, rtpL24depay->info.channels,
rtpL24depay->info.position, rtpL24depay->order->pos)) {
goto reorder_failed;
}
gst_rtp_buffer_unmap (&rtp);
return outbuf;
/* ERRORS */
empty_packet:
{
GST_ELEMENT_WARNING (rtpL24depay, STREAM, DECODE,
("Empty Payload."), (NULL));
gst_rtp_buffer_unmap (&rtp);
return NULL;
}
reorder_failed:
{
GST_ELEMENT_ERROR (rtpL24depay, STREAM, DECODE,
("Channel reordering failed."), (NULL));
gst_rtp_buffer_unmap (&rtp);
return NULL;
}
}
gboolean
gst_rtp_L24_depay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpL24depay",
GST_RANK_SECONDARY, GST_TYPE_RTP_L24_DEPAY);
}
/* GStreamer
* Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
*
* 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifndef __GST_RTP_L24_DEPAY_H__
#define __GST_RTP_L24_DEPAY_H__
#include <gst/gst.h>
#include <gst/rtp/gstrtpbasedepayload.h>
#include <gst/audio/audio.h>
#include "gstrtpchannels.h"
G_BEGIN_DECLS
/* Standard macros for defining types for this element. */
#define GST_TYPE_RTP_L24_DEPAY \
(gst_rtp_L24_depay_get_type())
#define GST_RTP_L24_DEPAY(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_L24_DEPAY,GstRtpL24Depay))
#define GST_RTP_L24_DEPAY_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_L24_DEPAY,GstRtpL24DepayClass))
#define GST_IS_RTP_L24_DEPAY(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_L24_DEPAY))
#define GST_IS_RTP_L24_DEPAY_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_L24_DEPAY))
typedef struct _GstRtpL24Depay GstRtpL24Depay;
typedef struct _GstRtpL24DepayClass GstRtpL24DepayClass;
/* Definition of structure storing data for this element. */
struct _GstRtpL24Depay
{
GstRTPBaseDepayload depayload;
GstAudioInfo info;
const GstRTPChannelOrder *order;
};
/* Standard definition defining a class for this element. */
struct _GstRtpL24DepayClass
{
GstRTPBaseDepayloadClass parent_class;
};
GType gst_rtp_L24_depay_get_type (void);
gboolean gst_rtp_L24_depay_plugin_init (GstPlugin * plugin);
G_END_DECLS
#endif /* __GST_RTP_L24_DEPAY_H__ */
/* GStreamer
* Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
*
* 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-rtpL24pay
* @see_also: rtpL24depay
*
* Payload raw 24-bit audio into RTP packets according to RFC 3190, section 4.
* For detailed information see: http://www.rfc-editor.org/rfc/rfc3190.txt
*
* <refsect2>
* <title>Example pipeline</title>
* |[
* gst-launch -v audiotestsrc ! audioconvert ! rtpL24pay ! udpsink
* ]| This example pipeline will payload raw audio. Refer to
* the rtpL24depay example to depayload and play the RTP stream.
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <string.h>
#include <gst/audio/audio.h>
#include <gst/rtp/gstrtpbuffer.h>
#include "gstrtpL24pay.h"
#include "gstrtpchannels.h"
GST_DEBUG_CATEGORY_STATIC (rtpL24pay_debug);
#define GST_CAT_DEFAULT (rtpL24pay_debug)
static GstStaticPadTemplate gst_rtp_L24_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) S24BE, "
"layout = (string) interleaved, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
);
static GstStaticPadTemplate gst_rtp_L24_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) [ 96, 127 ], "
"clock-rate = (int) [ 1, MAX ], "
"encoding-name = (string) \"L24\", " "channels = (int) [ 1, MAX ];")
);
static gboolean gst_rtp_L24_pay_setcaps (GstRTPBasePayload * basepayload,
GstCaps * caps);
static GstCaps *gst_rtp_L24_pay_getcaps (GstRTPBasePayload * rtppayload,
GstPad * pad, GstCaps * filter);
static GstFlowReturn
gst_rtp_L24_pay_handle_buffer (GstRTPBasePayload * basepayload,
GstBuffer * buffer);
#define gst_rtp_L24_pay_parent_class parent_class
G_DEFINE_TYPE (GstRtpL24Pay, gst_rtp_L24_pay, GST_TYPE_RTP_BASE_AUDIO_PAYLOAD);
static void
gst_rtp_L24_pay_class_init (GstRtpL24PayClass * klass)
{
GstElementClass *gstelement_class;
GstRTPBasePayloadClass *gstrtpbasepayload_class;
gstelement_class = (GstElementClass *) klass;
gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
gstrtpbasepayload_class->set_caps = gst_rtp_L24_pay_setcaps;
gstrtpbasepayload_class->get_caps = gst_rtp_L24_pay_getcaps;
gstrtpbasepayload_class->handle_buffer = gst_rtp_L24_pay_handle_buffer;
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_rtp_L24_pay_src_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_rtp_L24_pay_sink_template));
gst_element_class_set_static_metadata (gstelement_class,
"RTP audio payloader", "Codec/Payloader/Network/RTP",
"Payload-encode Raw 24-bit audio into RTP packets (RFC 3190)",
"Wim Taymans <wim.taymans@gmail.com>,"
"David Holroyd <dave@badgers-in-foil.co.uk>");
GST_DEBUG_CATEGORY_INIT (rtpL24pay_debug, "rtpL24pay", 0,
"L24 RTP Payloader");
}
static void
gst_rtp_L24_pay_init (GstRtpL24Pay * rtpL24pay)
{
GstRTPBaseAudioPayload *rtpbaseaudiopayload;
rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpL24pay);
/* tell rtpbaseaudiopayload that this is a sample based codec */
gst_rtp_base_audio_payload_set_sample_based (rtpbaseaudiopayload);
}
static gboolean
gst_rtp_L24_pay_setcaps (GstRTPBasePayload * basepayload, GstCaps * caps)
{
GstRtpL24Pay *rtpL24pay;
gboolean res;
gchar *params;
GstAudioInfo *info;
const GstRTPChannelOrder *order;
GstRTPBaseAudioPayload *rtpbaseaudiopayload;
rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (basepayload);
rtpL24pay = GST_RTP_L24_PAY (basepayload);
info = &rtpL24pay->info;
gst_audio_info_init (info);
if (!gst_audio_info_from_caps (info, caps))
goto invalid_caps;
order = gst_rtp_channels_get_by_pos (info->channels, info->position);
rtpL24pay->order = order;
gst_rtp_base_payload_set_options (basepayload, "audio", TRUE, "L24",
info->rate);
params = g_strdup_printf ("%d", info->channels);
if (!order && info->channels > 2) {
GST_ELEMENT_WARNING (rtpL24pay, STREAM, DECODE,
(NULL), ("Unknown channel order for %d channels", info->channels));
}
if (order && order->name) {
res = gst_rtp_base_payload_set_outcaps (basepayload,
"encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
info->channels, "channel-order", G_TYPE_STRING, order->name, NULL);
} else {
res = gst_rtp_base_payload_set_outcaps (basepayload,
"encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
info->channels, NULL);
}
g_free (params);
/* octet-per-sample is 3 * channels for L24 */
gst_rtp_base_audio_payload_set_sample_options (rtpbaseaudiopayload,
3 * info->channels);
return res;
/* ERRORS */
invalid_caps:
{
GST_DEBUG_OBJECT (rtpL24pay, "invalid caps");
return FALSE;
}
}
static GstCaps *
gst_rtp_L24_pay_getcaps (GstRTPBasePayload * rtppayload, GstPad * pad,
GstCaps * filter)
{
GstCaps *otherpadcaps;
GstCaps *caps;
caps = gst_pad_get_pad_template_caps (pad);
otherpadcaps = gst_pad_get_allowed_caps (rtppayload->srcpad);
if (otherpadcaps) {
if (!gst_caps_is_empty (otherpadcaps)) {
GstStructure *structure;
gint channels;
gint rate;
structure = gst_caps_get_structure (otherpadcaps, 0);
caps = gst_caps_make_writable (caps);
if (gst_structure_get_int (structure, "channels", &channels)) {
gst_caps_set_simple (caps, "channels", G_TYPE_INT, channels, NULL);
}
if (gst_structure_get_int (structure, "clock-rate", &rate)) {
gst_caps_set_simple (caps, "rate", G_TYPE_INT, rate, NULL);
}
}
gst_caps_unref (otherpadcaps);
}
if (filter) {
GstCaps *tcaps = caps;
caps = gst_caps_intersect_full (filter, tcaps, GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (tcaps);
}
return caps;
}
static GstFlowReturn
gst_rtp_L24_pay_handle_buffer (GstRTPBasePayload * basepayload,
GstBuffer * buffer)
{
GstRtpL24Pay *rtpL24pay;
rtpL24pay = GST_RTP_L24_PAY (basepayload);
buffer = gst_buffer_make_writable (buffer);
if (rtpL24pay->order &&
!gst_audio_buffer_reorder_channels (buffer, rtpL24pay->info.finfo->format,
rtpL24pay->info.channels, rtpL24pay->info.position,
rtpL24pay->order->pos)) {
return GST_FLOW_ERROR;
}
return GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->handle_buffer (basepayload,
buffer);
}
gboolean
gst_rtp_L24_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpL24pay",
GST_RANK_SECONDARY, GST_TYPE_RTP_L24_PAY);
}
/* GStreamer
* Copyright (C) <2005> Wim Taymans <wim.taymans@gmail.com>
*
* 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifndef __GST_RTP_L24_PAY_H__
#define __GST_RTP_L24_PAY_H__
#include <gst/gst.h>
#include <gst/rtp/gstrtpbaseaudiopayload.h>
#include "gstrtpchannels.h"
G_BEGIN_DECLS
#define GST_TYPE_RTP_L24_PAY \
(gst_rtp_L24_pay_get_type())
#define GST_RTP_L24_PAY(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_L24_PAY,GstRtpL24Pay))
#define GST_RTP_L24_PAY_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_L24_PAY,GstRtpL24PayClass))
#define GST_IS_RTP_L24_PAY(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_L24_PAY))
#define GST_IS_RTP_L24_PAY_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_L24_PAY))
typedef struct _GstRtpL24Pay GstRtpL24Pay;
typedef struct _GstRtpL24PayClass GstRtpL24PayClass;
struct _GstRtpL24Pay
{
GstRTPBaseAudioPayload payload;
GstAudioInfo info;
const GstRTPChannelOrder *order;
};
struct _GstRtpL24PayClass
{
GstRTPBaseAudioPayloadClass parent_class;
};
GType gst_rtp_L24_pay_get_type (void);
gboolean gst_rtp_L24_pay_plugin_init (GstPlugin * plugin);
G_END_DECLS
#endif /* __GST_RTP_L24_PAY_H__ */
......@@ -650,6 +650,25 @@ GST_START_TEST (rtp_L16)
"rtpL16pay", "rtpL16depay", 0, 0, FALSE);
}
GST_END_TEST;
static const guint8 rtp_L24_frame_data[] =
{ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
};
static int rtp_L24_frame_data_size = 24;
static int rtp_L24_frame_count = 1;
GST_START_TEST (rtp_L24)
{
rtp_pipeline_test (rtp_L24_frame_data, rtp_L24_frame_data_size,
rtp_L24_frame_count,
"audio/x-raw,format=S24BE,rate=1,channels=1,layout=(string)interleaved",
"rtpL24pay", "rtpL24depay", 0, 0, FALSE);
}
GST_END_TEST;
static const guint8 rtp_mp2t_frame_data[] =
{ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
......@@ -917,6 +936,7 @@ rtp_payloading_suite (void)
tcase_add_test (tc_chain, rtp_h264_list_gt_mtu);
tcase_add_test (tc_chain, rtp_h264_list_gt_mtu_avc);
tcase_add_test (tc_chain, rtp_L16);
tcase_add_test (tc_chain, rtp_L24);
tcase_add_test (tc_chain, rtp_mp2t);
tcase_add_test (tc_chain, rtp_mp4v);
tcase_add_test (tc_chain, rtp_mp4v_list);
......
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