From: Wim Taymans Date: Thu, 25 Apr 2013 11:19:35 +0000 (+0200) Subject: docs: add some pay/depayloaders X-Git-Tag: 1.19.3~509^2~5894 X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=1df2e623b530d553a5cd39f81d1462995fec2bd6;p=platform%2Fupstream%2Fgstreamer.git docs: add some pay/depayloaders See https://bugzilla.gnome.org/show_bug.cgi?id=551631 --- diff --git a/docs/plugins/gst-plugins-good-plugins-docs.sgml b/docs/plugins/gst-plugins-good-plugins-docs.sgml index 315d1b7..148c639 100644 --- a/docs/plugins/gst-plugins-good-plugins-docs.sgml +++ b/docs/plugins/gst-plugins-good-plugins-docs.sgml @@ -128,6 +128,14 @@ + + + + + + + + diff --git a/docs/plugins/gst-plugins-good-plugins-sections.txt b/docs/plugins/gst-plugins-good-plugins-sections.txt index d7a7fe7..86ecacb 100644 --- a/docs/plugins/gst-plugins-good-plugins-sections.txt +++ b/docs/plugins/gst-plugins-good-plugins-sections.txt @@ -1625,6 +1625,126 @@ GstRTPDTMFEventType
+element-rtpac3depay +rtpac3depay +GstRtpAC3Depay + +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 +
+ +
+element-rtpac3pay +rtpac3pay +GstRtpAC3Pay + +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 +
+ +
+element-rtpamrdepay +rtpamrdepay +GstRtpAMRDepay + +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 +
+ +
+element-rtpamrpay +rtpamrpay +GstRtpAMRPay + +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 +
+ +
+element-rtpbvdepay +rtpbvdepay +GstRtpBVDepay + +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 +
+ +
+element-rtpbvpay +rtpbvpay +GstRtpBVPay + +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 +
+ +
+element-rtpL16depay +rtpL16depay +GstRtpL16Depay + +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 +
+ +
+element-rtpL16pay +rtpL16pay +GstRtpL16Pay + +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 +
+ +
element-rtpj2kpay rtpj2kpay GstRtpJ2KPay diff --git a/gst/rtp/gstrtpL16depay.c b/gst/rtp/gstrtpL16depay.c index 880451f..7e96d9d 100644 --- a/gst/rtp/gstrtpL16depay.c +++ b/gst/rtp/gstrtpL16depay.c @@ -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 + * + * + * Example pipeline + * |[ + * 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. + * + * + * Last reviewed on 2013-04-25 (1.1.0) + */ + #ifdef HAVE_CONFIG_H #include "config.h" #endif diff --git a/gst/rtp/gstrtpL16pay.c b/gst/rtp/gstrtpL16pay.c index 16abf32..4a101ee 100644 --- a/gst/rtp/gstrtpL16pay.c +++ b/gst/rtp/gstrtpL16pay.c @@ -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 + * + * + * Example pipeline + * |[ + * 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. + * + * + * Last reviewed on 2013-04-25 (1.1.0) + */ + #ifdef HAVE_CONFIG_H # include "config.h" #endif diff --git a/gst/rtp/gstrtpac3depay.c b/gst/rtp/gstrtpac3depay.c index 0a2a70c..fc79b5d 100644 --- a/gst/rtp/gstrtpac3depay.c +++ b/gst/rtp/gstrtpac3depay.c @@ -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 + * + * + * Example pipeline + * |[ + * 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. + * + * + * Last reviewed on 2013-04-25 (1.1.0) + */ + #ifdef HAVE_CONFIG_H # include "config.h" #endif diff --git a/gst/rtp/gstrtpac3pay.c b/gst/rtp/gstrtpac3pay.c index 0af1ea6..175d627 100644 --- a/gst/rtp/gstrtpac3pay.c +++ b/gst/rtp/gstrtpac3pay.c @@ -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 + * + * + * Example pipeline + * |[ + * 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. + * + * + * Last reviewed on 2013-04-25 (1.1.0) + */ + #ifdef HAVE_CONFIG_H # include "config.h" #endif diff --git a/gst/rtp/gstrtpamrdepay.c b/gst/rtp/gstrtpamrdepay.c index 25a4508..e1208bf 100644 --- a/gst/rtp/gstrtpamrdepay.c +++ b/gst/rtp/gstrtpamrdepay.c @@ -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 + * + * + * Example pipeline + * |[ + * 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. + * + * + * 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 { diff --git a/gst/rtp/gstrtpamrpay.c b/gst/rtp/gstrtpamrpay.c index 29c26de..defc7f4 100644 --- a/gst/rtp/gstrtpamrpay.c +++ b/gst/rtp/gstrtpamrpay.c @@ -17,18 +17,23 @@ * Boston, MA 02110-1301, USA. */ -#ifdef HAVE_CONFIG_H -# include "config.h" -#endif - -#include - -#include - -#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 + * + * + * Example pipeline + * |[ + * 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. + * + * + * 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 + +#include + +#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, diff --git a/gst/rtp/gstrtpbvdepay.c b/gst/rtp/gstrtpbvdepay.c index 67fbb0c..7b85558 100644 --- a/gst/rtp/gstrtpbvdepay.c +++ b/gst/rtp/gstrtpbvdepay.c @@ -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 diff --git a/gst/rtp/gstrtpbvpay.c b/gst/rtp/gstrtpbvpay.c index 4a0c263..be12b38 100644 --- a/gst/rtp/gstrtpbvpay.c +++ b/gst/rtp/gstrtpbvpay.c @@ -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