gstrtpjpegpay.c \
gstrtpL16depay.c \
gstrtpL16pay.c \
+ gstrtpL24depay.c \
+ gstrtpL24pay.c \
gstasteriskh263.c \
gstrtpmp1sdepay.c \
gstrtpmp2tdepay.c \
#include "gstrtpjpegpay.h"
#include "gstrtpL16depay.h"
#include "gstrtpL16pay.h"
+#include "gstrtpL24depay.h"
+#include "gstrtpL24pay.h"
#include "gstasteriskh263.h"
#include "gstrtpmp1sdepay.h"
#include "gstrtpmp2tdepay.h"
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;
--- /dev/null
+/* 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);
+}
--- /dev/null
+/* 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__ */
--- /dev/null
+/* 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);
+}
--- /dev/null
+/* 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__ */
}
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,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
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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