Commit 1df2e623 authored by Wim Taymans's avatar Wim Taymans

docs: add some pay/depayloaders

See https://bugzilla.gnome.org/show_bug.cgi?id=551631
parent fb0384fa
......@@ -128,6 +128,14 @@
<xi:include href="xml/element-rgvolume.xml" />
<xi:include href="xml/element-rippletv.xml" />
<xi:include href="xml/element-rtpdec.xml" />
<xi:include href="xml/element-rtpac3depay.xml" />
<xi:include href="xml/element-rtpac3pay.xml" />
<xi:include href="xml/element-rtpamrdepay.xml" />
<xi:include href="xml/element-rtpamrpay.xml" />
<xi:include href="xml/element-rtpbvdepay.xml" />
<xi:include href="xml/element-rtpbvpay.xml" />
<xi:include href="xml/element-rtpL16depay.xml" />
<xi:include href="xml/element-rtpL16pay.xml" />
<xi:include href="xml/element-rtpj2kpay.xml" />
<xi:include href="xml/element-rtpjpegpay.xml" />
<xi:include href="xml/element-rtpsbcpay.xml" />
......
......@@ -1624,6 +1624,126 @@ GstRTPDTMFSrcEvent
GstRTPDTMFEventType
</SECTION>
<SECTION>
<FILE>element-rtpac3depay</FILE>
<TITLE>rtpac3depay</TITLE>
GstRtpAC3Depay
<SUBSECTION Standard>
GstRtpAC3DepayClass
GST_RTP_AC3_DEPAY
GST_IS_RTP_AC3_DEPAY
GST_TYPE_RTP_AC3_DEPAY
GST_RTP_AC3_DEPAY_CLASS
GST_IS_RTP_AC3_DEPAY_CLASS
gst_rtp_ac3_depay_plugin_init
gst_rtp_ac3_depay_get_type
</SECTION>
<SECTION>
<FILE>element-rtpac3pay</FILE>
<TITLE>rtpac3pay</TITLE>
GstRtpAC3Pay
<SUBSECTION Standard>
GstRtpAC3PayClass
GST_RTP_AC3_PAY
GST_IS_RTP_AC3_PAY
GST_TYPE_RTP_AC3_PAY
GST_RTP_AC3_PAY_CLASS
GST_IS_RTP_AC3_PAY_CLASS
gst_rtp_ac3_pay_plugin_init
gst_rtp_ac3_pay_get_type
</SECTION>
<SECTION>
<FILE>element-rtpamrdepay</FILE>
<TITLE>rtpamrdepay</TITLE>
GstRtpAMRDepay
<SUBSECTION Standard>
GstRtpAMRDepayClass
GST_RTP_AMR_DEPAY
GST_IS_RTP_AMR_DEPAY
GST_TYPE_RTP_AMR_DEPAY
GST_RTP_AMR_DEPAY_CLASS
GST_IS_RTP_AMR_DEPAY_CLASS
gst_rtp_amr_depay_plugin_init
gst_rtp_amr_depay_get_type
</SECTION>
<SECTION>
<FILE>element-rtpamrpay</FILE>
<TITLE>rtpamrpay</TITLE>
GstRtpAMRPay
<SUBSECTION Standard>
GstRtpAMRPayClass
GST_RTP_AMR_PAY
GST_IS_RTP_AMR_PAY
GST_TYPE_RTP_AMR_PAY
GST_RTP_AMR_PAY_CLASS
GST_IS_RTP_AMR_PAY_CLASS
gst_rtp_amr_pay_plugin_init
gst_rtp_amr_pay_get_type
</SECTION>
<SECTION>
<FILE>element-rtpbvdepay</FILE>
<TITLE>rtpbvdepay</TITLE>
GstRtpBVDepay
<SUBSECTION Standard>
GstRtpBVDepayClass
GST_RTP_BV_DEPAY
GST_IS_RTP_BV_DEPAY
GST_TYPE_RTP_BV_DEPAY
GST_RTP_BV_DEPAY_CLASS
GST_IS_RTP_BV_DEPAY_CLASS
gst_rtp_bv_depay_plugin_init
gst_rtp_bv_depay_get_type
</SECTION>
<SECTION>
<FILE>element-rtpbvpay</FILE>
<TITLE>rtpbvpay</TITLE>
GstRtpBVPay
<SUBSECTION Standard>
GstRtpBVPayClass
GST_RTP_BV_PAY
GST_IS_RTP_BV_PAY
GST_TYPE_RTP_BV_PAY
GST_RTP_BV_PAY_CLASS
GST_IS_RTP_BV_PAY_CLASS
gst_rtp_bv_pay_plugin_init
gst_rtp_bv_pay_get_type
</SECTION>
<SECTION>
<FILE>element-rtpL16depay</FILE>
<TITLE>rtpL16depay</TITLE>
GstRtpL16Depay
<SUBSECTION Standard>
GstRtpL16DepayClass
GST_RTP_L16_DEPAY
GST_IS_RTP_L16_DEPAY
GST_TYPE_RTP_L16_DEPAY
GST_RTP_L16_DEPAY_CLASS
GST_IS_RTP_L16_DEPAY_CLASS
gst_rtp_L16_depay_plugin_init
gst_rtp_L16_depay_get_type
</SECTION>
<SECTION>
<FILE>element-rtpL16pay</FILE>
<TITLE>rtpL16pay</TITLE>
GstRtpL16Pay
<SUBSECTION Standard>
GstRtpL16PayClass
GST_RTP_L16_PAY
GST_IS_RTP_L16_PAY
GST_TYPE_RTP_L16_PAY
GST_RTP_L16_PAY_CLASS
GST_IS_RTP_L16_PAY_CLASS
gst_rtp_L16_pay_plugin_init
gst_rtp_L16_pay_get_type
</SECTION>
<SECTION>
<FILE>element-rtpj2kpay</FILE>
<TITLE>rtpj2kpay</TITLE>
......
......@@ -17,6 +17,24 @@
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-rtpL16depay
* @see_also: rtpL16pay
*
* Extract raw audio from RTP packets according to RFC 3551.
* For detailed information see: http://www.rfc-editor.org/rfc/rfc3551.txt
*
* <refsect2>
* <title>Example pipeline</title>
* |[
* gst-launch udpsrc caps='application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)L16, encoding-params=(string)1, channels=(int)1, payload=(int)96' ! rtpL16depay ! pulsesink
* ]| This example pipeline will depayload an RTP raw audio stream. Refer to
* the rtpL16pay example to create the RTP stream.
* </refsect2>
*
* Last reviewed on 2013-04-25 (1.1.0)
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
......
......@@ -17,6 +17,24 @@
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-rtpL16pay
* @see_also: rtpL16depay
*
* Payload raw audio into RTP packets according to RFC 3551.
* For detailed information see: http://www.rfc-editor.org/rfc/rfc3551.txt
*
* <refsect2>
* <title>Example pipeline</title>
* |[
* gst-launch -v audiotestsrc ! audioconvert ! rtpL16pay ! udpsink
* ]| This example pipeline will payload raw audio. Refer to
* the rtpL16depay example to depayload and play the RTP stream.
* </refsect2>
*
* Last reviewed on 2013-04-25 (1.1.0)
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
......
......@@ -17,6 +17,24 @@
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-rtpac3depay
* @see_also: rtpac3pay
*
* Extract AC3 audio from RTP packets according to RFC 4184.
* For detailed information see: http://www.rfc-editor.org/rfc/rfc4184.txt
*
* <refsect2>
* <title>Example pipeline</title>
* |[
* gst-launch-1.0 udpsrc caps='application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)AC3, payload=(int)96' ! rtpac3depay ! a52dec ! pulsesink
* ]| This example pipeline will depayload and decode an RTP AC3 stream. Refer to
* the rtpac3pay example to create the RTP stream.
* </refsect2>
*
* Last reviewed on 2013-04-25 (1.1.0)
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
......
......@@ -17,6 +17,24 @@
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-rtpac3pay
* @see_also: rtpac3depay
*
* Payload AC3 audio into RTP packets according to RFC 4184.
* For detailed information see: http://www.rfc-editor.org/rfc/rfc4184.txt
*
* <refsect2>
* <title>Example pipeline</title>
* |[
* gst-launch -v audiotestsrc ! avenc_ac3 ! rtpac3pay ! udpsink
* ]| This example pipeline will encode and payload AC3 stream. Refer to
* the rtpac3depay example to depayload and decode the RTP stream.
* </refsect2>
*
* Last reviewed on 2013-04-25 (1.1.0)
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
......
......@@ -17,6 +17,30 @@
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-rtpamrdepay
* @see_also: rtpamrpay
*
* Extract AMR audio from RTP packets according to RFC 3267.
* For detailed information see: http://www.rfc-editor.org/rfc/rfc3267.txt
*
* <refsect2>
* <title>Example pipeline</title>
* |[
* gst-launch-1.0 udpsrc caps='application/x-rtp, media=(string)audio, clock-rate=(int)8000, encoding-name=(string)AMR, encoding-params=(string)1, octet-align=(string)1, payload=(int)96' ! rtpamrdepay ! amrnbdec ! pulsesink
* ]| This example pipeline will depayload and decode an RTP AMR stream. Refer to
* the rtpamrpay example to create the RTP stream.
* </refsect2>
*
* Last reviewed on 2013-04-25 (1.1.0)
*/
/*
* RFC 3267 - Real-Time Transport Protocol (RTP) Payload Format and File
* Storage Format for the Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate
* Wideband (AMR-WB) Audio Codecs.
*
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
......@@ -30,13 +54,6 @@
GST_DEBUG_CATEGORY_STATIC (rtpamrdepay_debug);
#define GST_CAT_DEFAULT (rtpamrdepay_debug)
/* references:
*
* RFC 3267 - Real-Time Transport Protocol (RTP) Payload Format and File
* Storage Format for the Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate
* Wideband (AMR-WB) Audio Codecs.
*/
/* RtpAMRDepay signals and args */
enum
{
......
......@@ -17,18 +17,23 @@
* Boston, MA 02110-1301, USA.
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include "gstrtpamrpay.h"
GST_DEBUG_CATEGORY_STATIC (rtpamrpay_debug);
#define GST_CAT_DEFAULT (rtpamrpay_debug)
/**
* SECTION:element-rtpamrpay
* @see_also: rtpamrdepay
*
* Payload AMR audio into RTP packets according to RFC 3267.
* For detailed information see: http://www.rfc-editor.org/rfc/rfc3267.txt
*
* <refsect2>
* <title>Example pipeline</title>
* |[
* gst-launch -v audiotestsrc ! amrnbenc ! rtpamrpay ! udpsink
* ]| This example pipeline will encode and payload an AMR stream. Refer to
* the rtpamrdepay example to depayload and decode the RTP stream.
* </refsect2>
*
* Last reviewed on 2013-04-25 (1.1.0)
*/
/* references:
*
......@@ -43,6 +48,19 @@ GST_DEBUG_CATEGORY_STATIC (rtpamrpay_debug);
* (3GPP TS 26.201 version 6.0.0 Release 6)
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include "gstrtpamrpay.h"
GST_DEBUG_CATEGORY_STATIC (rtpamrpay_debug);
#define GST_CAT_DEFAULT (rtpamrpay_debug)
static GstStaticPadTemplate gst_rtp_amr_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
......
......@@ -17,6 +17,16 @@
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-rtpbvdepay
* @see_also: rtpbvpay
*
* Extract BroadcomVoice audio from RTP packets according to RFC 4298.
* For detailed information see: http://www.rfc-editor.org/rfc/rfc4298.txt
*
* Last reviewed on 2013-04-25 (1.1.0)
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
......
......@@ -17,6 +17,16 @@
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-rtpbvpay
* @see_also: rtpbvdepay
*
* Payload BroadcomVoice audio into RTP packets according to RFC 4298.
* For detailed information see: http://www.rfc-editor.org/rfc/rfc4298.txt
*
* Last reviewed on 2013-04-25 (1.1.0)
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
......
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