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docs: add some pay/depayloaders
See https://bugzilla.gnome.org/show_bug.cgi?id=551631
This commit is contained in:
parent
fb0384fa0d
commit
1df2e623b5
10 changed files with 274 additions and 19 deletions
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@ -128,6 +128,14 @@
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<xi:include href="xml/element-rgvolume.xml" />
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<xi:include href="xml/element-rippletv.xml" />
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<xi:include href="xml/element-rtpdec.xml" />
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<xi:include href="xml/element-rtpac3depay.xml" />
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<xi:include href="xml/element-rtpac3pay.xml" />
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<xi:include href="xml/element-rtpamrdepay.xml" />
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<xi:include href="xml/element-rtpamrpay.xml" />
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<xi:include href="xml/element-rtpbvdepay.xml" />
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<xi:include href="xml/element-rtpbvpay.xml" />
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<xi:include href="xml/element-rtpL16depay.xml" />
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<xi:include href="xml/element-rtpL16pay.xml" />
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<xi:include href="xml/element-rtpj2kpay.xml" />
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<xi:include href="xml/element-rtpjpegpay.xml" />
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<xi:include href="xml/element-rtpsbcpay.xml" />
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@ -1624,6 +1624,126 @@ GstRTPDTMFSrcEvent
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GstRTPDTMFEventType
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</SECTION>
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<SECTION>
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<FILE>element-rtpac3depay</FILE>
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<TITLE>rtpac3depay</TITLE>
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GstRtpAC3Depay
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<SUBSECTION Standard>
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GstRtpAC3DepayClass
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GST_RTP_AC3_DEPAY
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GST_IS_RTP_AC3_DEPAY
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GST_TYPE_RTP_AC3_DEPAY
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GST_RTP_AC3_DEPAY_CLASS
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GST_IS_RTP_AC3_DEPAY_CLASS
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gst_rtp_ac3_depay_plugin_init
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gst_rtp_ac3_depay_get_type
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</SECTION>
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<SECTION>
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<FILE>element-rtpac3pay</FILE>
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<TITLE>rtpac3pay</TITLE>
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GstRtpAC3Pay
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<SUBSECTION Standard>
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GstRtpAC3PayClass
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GST_RTP_AC3_PAY
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GST_IS_RTP_AC3_PAY
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GST_TYPE_RTP_AC3_PAY
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GST_RTP_AC3_PAY_CLASS
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GST_IS_RTP_AC3_PAY_CLASS
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gst_rtp_ac3_pay_plugin_init
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gst_rtp_ac3_pay_get_type
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</SECTION>
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<SECTION>
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<FILE>element-rtpamrdepay</FILE>
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<TITLE>rtpamrdepay</TITLE>
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GstRtpAMRDepay
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<SUBSECTION Standard>
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GstRtpAMRDepayClass
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GST_RTP_AMR_DEPAY
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GST_IS_RTP_AMR_DEPAY
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GST_TYPE_RTP_AMR_DEPAY
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GST_RTP_AMR_DEPAY_CLASS
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GST_IS_RTP_AMR_DEPAY_CLASS
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gst_rtp_amr_depay_plugin_init
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gst_rtp_amr_depay_get_type
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</SECTION>
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<SECTION>
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<FILE>element-rtpamrpay</FILE>
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<TITLE>rtpamrpay</TITLE>
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GstRtpAMRPay
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<SUBSECTION Standard>
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GstRtpAMRPayClass
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GST_RTP_AMR_PAY
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GST_IS_RTP_AMR_PAY
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GST_TYPE_RTP_AMR_PAY
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GST_RTP_AMR_PAY_CLASS
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GST_IS_RTP_AMR_PAY_CLASS
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gst_rtp_amr_pay_plugin_init
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gst_rtp_amr_pay_get_type
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</SECTION>
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<SECTION>
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<FILE>element-rtpbvdepay</FILE>
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<TITLE>rtpbvdepay</TITLE>
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GstRtpBVDepay
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<SUBSECTION Standard>
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GstRtpBVDepayClass
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GST_RTP_BV_DEPAY
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GST_IS_RTP_BV_DEPAY
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GST_TYPE_RTP_BV_DEPAY
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GST_RTP_BV_DEPAY_CLASS
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GST_IS_RTP_BV_DEPAY_CLASS
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gst_rtp_bv_depay_plugin_init
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gst_rtp_bv_depay_get_type
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</SECTION>
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<SECTION>
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<FILE>element-rtpbvpay</FILE>
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<TITLE>rtpbvpay</TITLE>
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GstRtpBVPay
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<SUBSECTION Standard>
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GstRtpBVPayClass
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GST_RTP_BV_PAY
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GST_IS_RTP_BV_PAY
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GST_TYPE_RTP_BV_PAY
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GST_RTP_BV_PAY_CLASS
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GST_IS_RTP_BV_PAY_CLASS
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gst_rtp_bv_pay_plugin_init
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gst_rtp_bv_pay_get_type
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</SECTION>
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<SECTION>
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<FILE>element-rtpL16depay</FILE>
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<TITLE>rtpL16depay</TITLE>
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GstRtpL16Depay
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<SUBSECTION Standard>
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GstRtpL16DepayClass
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GST_RTP_L16_DEPAY
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GST_IS_RTP_L16_DEPAY
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GST_TYPE_RTP_L16_DEPAY
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GST_RTP_L16_DEPAY_CLASS
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GST_IS_RTP_L16_DEPAY_CLASS
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gst_rtp_L16_depay_plugin_init
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gst_rtp_L16_depay_get_type
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</SECTION>
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<SECTION>
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<FILE>element-rtpL16pay</FILE>
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<TITLE>rtpL16pay</TITLE>
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GstRtpL16Pay
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<SUBSECTION Standard>
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GstRtpL16PayClass
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GST_RTP_L16_PAY
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GST_IS_RTP_L16_PAY
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GST_TYPE_RTP_L16_PAY
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GST_RTP_L16_PAY_CLASS
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GST_IS_RTP_L16_PAY_CLASS
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gst_rtp_L16_pay_plugin_init
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gst_rtp_L16_pay_get_type
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</SECTION>
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<SECTION>
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<FILE>element-rtpj2kpay</FILE>
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<TITLE>rtpj2kpay</TITLE>
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@ -17,6 +17,24 @@
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:element-rtpL16depay
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* @see_also: rtpL16pay
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*
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* Extract raw audio from RTP packets according to RFC 3551.
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* For detailed information see: http://www.rfc-editor.org/rfc/rfc3551.txt
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*
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* <refsect2>
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* <title>Example pipeline</title>
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* |[
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* 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
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* ]| This example pipeline will depayload an RTP raw audio stream. Refer to
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* the rtpL16pay example to create the RTP stream.
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* </refsect2>
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*
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* Last reviewed on 2013-04-25 (1.1.0)
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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@ -17,6 +17,24 @@
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:element-rtpL16pay
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* @see_also: rtpL16depay
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*
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* Payload raw audio into RTP packets according to RFC 3551.
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* For detailed information see: http://www.rfc-editor.org/rfc/rfc3551.txt
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*
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* <refsect2>
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* <title>Example pipeline</title>
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* |[
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* gst-launch -v audiotestsrc ! audioconvert ! rtpL16pay ! udpsink
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* ]| This example pipeline will payload raw audio. Refer to
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* the rtpL16depay example to depayload and play the RTP stream.
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* </refsect2>
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*
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* Last reviewed on 2013-04-25 (1.1.0)
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*/
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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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@ -17,6 +17,24 @@
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:element-rtpac3depay
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* @see_also: rtpac3pay
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*
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* Extract AC3 audio from RTP packets according to RFC 4184.
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* For detailed information see: http://www.rfc-editor.org/rfc/rfc4184.txt
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*
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* <refsect2>
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* <title>Example pipeline</title>
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* |[
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* 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
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* ]| This example pipeline will depayload and decode an RTP AC3 stream. Refer to
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* the rtpac3pay example to create the RTP stream.
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* </refsect2>
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*
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* Last reviewed on 2013-04-25 (1.1.0)
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*/
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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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@ -17,6 +17,24 @@
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:element-rtpac3pay
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* @see_also: rtpac3depay
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*
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* Payload AC3 audio into RTP packets according to RFC 4184.
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* For detailed information see: http://www.rfc-editor.org/rfc/rfc4184.txt
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*
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* <refsect2>
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* <title>Example pipeline</title>
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* |[
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* gst-launch -v audiotestsrc ! avenc_ac3 ! rtpac3pay ! udpsink
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* ]| This example pipeline will encode and payload AC3 stream. Refer to
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* the rtpac3depay example to depayload and decode the RTP stream.
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* </refsect2>
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*
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* Last reviewed on 2013-04-25 (1.1.0)
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*/
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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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@ -17,6 +17,30 @@
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:element-rtpamrdepay
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* @see_also: rtpamrpay
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*
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* Extract AMR audio from RTP packets according to RFC 3267.
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* For detailed information see: http://www.rfc-editor.org/rfc/rfc3267.txt
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*
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* <refsect2>
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* <title>Example pipeline</title>
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* |[
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* 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
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* ]| This example pipeline will depayload and decode an RTP AMR stream. Refer to
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* the rtpamrpay example to create the RTP stream.
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* </refsect2>
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*
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* Last reviewed on 2013-04-25 (1.1.0)
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*/
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/*
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* RFC 3267 - Real-Time Transport Protocol (RTP) Payload Format and File
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* Storage Format for the Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate
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* Wideband (AMR-WB) Audio Codecs.
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*
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*/
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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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GST_DEBUG_CATEGORY_STATIC (rtpamrdepay_debug);
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#define GST_CAT_DEFAULT (rtpamrdepay_debug)
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/* references:
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*
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* RFC 3267 - Real-Time Transport Protocol (RTP) Payload Format and File
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* Storage Format for the Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate
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* Wideband (AMR-WB) Audio Codecs.
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*/
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/* RtpAMRDepay signals and args */
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enum
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{
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* Boston, MA 02110-1301, USA.
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*/
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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#include <string.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include "gstrtpamrpay.h"
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GST_DEBUG_CATEGORY_STATIC (rtpamrpay_debug);
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#define GST_CAT_DEFAULT (rtpamrpay_debug)
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/**
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* SECTION:element-rtpamrpay
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* @see_also: rtpamrdepay
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*
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* Payload AMR audio into RTP packets according to RFC 3267.
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* For detailed information see: http://www.rfc-editor.org/rfc/rfc3267.txt
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*
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* <refsect2>
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* <title>Example pipeline</title>
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* |[
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* gst-launch -v audiotestsrc ! amrnbenc ! rtpamrpay ! udpsink
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* ]| This example pipeline will encode and payload an AMR stream. Refer to
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* the rtpamrdepay example to depayload and decode the RTP stream.
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* </refsect2>
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*
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* Last reviewed on 2013-04-25 (1.1.0)
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*/
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/* references:
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*
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* (3GPP TS 26.201 version 6.0.0 Release 6)
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*/
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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#include <string.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include "gstrtpamrpay.h"
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GST_DEBUG_CATEGORY_STATIC (rtpamrpay_debug);
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#define GST_CAT_DEFAULT (rtpamrpay_debug)
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static GstStaticPadTemplate gst_rtp_amr_pay_sink_template =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:element-rtpbvdepay
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* @see_also: rtpbvpay
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*
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* Extract BroadcomVoice audio from RTP packets according to RFC 4298.
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* For detailed information see: http://www.rfc-editor.org/rfc/rfc4298.txt
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*
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* Last reviewed on 2013-04-25 (1.1.0)
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*/
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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:element-rtpbvpay
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* @see_also: rtpbvdepay
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*
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* Payload BroadcomVoice audio into RTP packets according to RFC 4298.
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* For detailed information see: http://www.rfc-editor.org/rfc/rfc4298.txt
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*
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* Last reviewed on 2013-04-25 (1.1.0)
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*/
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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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