Commit 0a39f494 authored by Wim Taymans's avatar Wim Taymans
Browse files

Add RTCP docs.

Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_get_adapter):
Add RTCP docs.
Fix some more docs.
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/rtp/gstrtcpbuffer.c:
(gst_rtcp_buffer_new_take_data), (gst_rtcp_buffer_new_copy_data),
(gst_rtcp_buffer_validate_data), (gst_rtcp_buffer_validate),
(gst_rtcp_buffer_get_packet_count), (read_packet_header),
(gst_rtcp_buffer_get_first_packet), (gst_rtcp_packet_move_to_next),
(gst_rtcp_buffer_add_packet), (gst_rtcp_packet_remove),
(gst_rtcp_packet_get_padding), (gst_rtcp_packet_get_type),
(gst_rtcp_packet_get_count), (gst_rtcp_packet_get_length),
(gst_rtcp_packet_sr_get_sender_info),
(gst_rtcp_packet_sr_set_sender_info),
(gst_rtcp_packet_rr_get_ssrc), (gst_rtcp_packet_rr_set_ssrc),
(gst_rtcp_packet_get_rb_count), (gst_rtcp_packet_get_rb),
(gst_rtcp_packet_add_rb), (gst_rtcp_packet_set_rb),
(gst_rtcp_packet_sdes_get_chunk_count),
(gst_rtcp_packet_sdes_first_chunk),
(gst_rtcp_packet_sdes_next_chunk), (gst_rtcp_packet_sdes_get_ssrc),
(gst_rtcp_packet_sdes_first_item),
(gst_rtcp_packet_sdes_next_item), (gst_rtcp_packet_sdes_get_item),
(gst_rtcp_packet_bye_get_ssrc_count),
(gst_rtcp_packet_bye_get_nth_ssrc), (gst_rtcp_packet_bye_add_ssrc),
(gst_rtcp_packet_bye_add_ssrcs), (get_reason_offset),
(gst_rtcp_packet_bye_get_reason_len),
(gst_rtcp_packet_bye_get_reason), (gst_rtcp_packet_bye_set_reason):
* gst-libs/gst/rtp/gstrtcpbuffer.h:
Add new helper object for parsing and creating RTCP messages.
parent dfdd873f
2007-03-29 Wim Taymans <wim@fluendo.com>
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_get_adapter):
Add RTCP docs.
Fix some more docs.
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/rtp/gstrtcpbuffer.c:
(gst_rtcp_buffer_new_take_data), (gst_rtcp_buffer_new_copy_data),
(gst_rtcp_buffer_validate_data), (gst_rtcp_buffer_validate),
(gst_rtcp_buffer_get_packet_count), (read_packet_header),
(gst_rtcp_buffer_get_first_packet), (gst_rtcp_packet_move_to_next),
(gst_rtcp_buffer_add_packet), (gst_rtcp_packet_remove),
(gst_rtcp_packet_get_padding), (gst_rtcp_packet_get_type),
(gst_rtcp_packet_get_count), (gst_rtcp_packet_get_length),
(gst_rtcp_packet_sr_get_sender_info),
(gst_rtcp_packet_sr_set_sender_info),
(gst_rtcp_packet_rr_get_ssrc), (gst_rtcp_packet_rr_set_ssrc),
(gst_rtcp_packet_get_rb_count), (gst_rtcp_packet_get_rb),
(gst_rtcp_packet_add_rb), (gst_rtcp_packet_set_rb),
(gst_rtcp_packet_sdes_get_chunk_count),
(gst_rtcp_packet_sdes_first_chunk),
(gst_rtcp_packet_sdes_next_chunk), (gst_rtcp_packet_sdes_get_ssrc),
(gst_rtcp_packet_sdes_first_item),
(gst_rtcp_packet_sdes_next_item), (gst_rtcp_packet_sdes_get_item),
(gst_rtcp_packet_bye_get_ssrc_count),
(gst_rtcp_packet_bye_get_nth_ssrc), (gst_rtcp_packet_bye_add_ssrc),
(gst_rtcp_packet_bye_add_ssrcs), (get_reason_offset),
(gst_rtcp_packet_bye_get_reason_len),
(gst_rtcp_packet_bye_get_reason), (gst_rtcp_packet_bye_set_reason):
* gst-libs/gst/rtp/gstrtcpbuffer.h:
Add new helper object for parsing and creating RTCP messages.
2007-03-29 Sebastian Dröge <slomo@circular-chaos.org>
 
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
......@@ -41,6 +41,7 @@
<!ENTITY GstBaseRtpDepayload SYSTEM "xml/gstbasertpdepayload.xml">
<!ENTITY GstBaseRtpPayload SYSTEM "xml/gstbasertppayload.xml">
<!ENTITY GstRtpBuffer SYSTEM "xml/gstrtpbuffer.xml">
<!ENTITY GstRtcpBuffer SYSTEM "xml/gstrtcpbuffer.xml">
<!-- tag -->
<!ENTITY GstTag SYSTEM "xml/gsttag.xml">
<!ENTITY GstTagVorbis SYSTEM "xml/gsttagvorbis.xml">
......@@ -164,6 +165,7 @@
&GstBaseRtpDepayload;
&GstBaseRtpPayload;
&GstRtpBuffer;
&GstRtcpBuffer;
</chapter>
<chapter id="gstreamer-tag">
......
......@@ -93,6 +93,7 @@ GST_AUDIO_SRC_GET_CLASS
<INCLUDE>gst/audio/gstbaseaudiosink.h</INCLUDE>
GstBaseAudioSink
GstBaseAudioSinkClass
GstBaseAudioSinkSlaveMethod
GST_BASE_AUDIO_SINK_CLOCK
GST_BASE_AUDIO_SINK_PAD
......@@ -105,6 +106,7 @@ gst_base_audio_sink_get_type
GST_BASE_AUDIO_SINK_CLASS
GST_IS_BASE_AUDIO_SINK_CLASS
GST_BASE_AUDIO_SINK_GET_CLASS
GstBaseAudioSinkPrivate
</SECTION>
<SECTION>
......@@ -745,6 +747,8 @@ gst_base_rtp_audio_payload_set_frame_based
gst_base_rtp_audio_payload_set_frame_options
gst_base_rtp_audio_payload_set_sample_based
gst_base_rtp_audio_payload_set_sample_options
gst_base_rtp_audio_payload_get_adapter
gst_base_rtp_audio_payload_push
<SUBSECTION Standard>
GST_TYPE_BASE_RTP_AUDIO_PAYLOAD
GST_BASE_RTP_AUDIO_PAYLOAD
......@@ -752,6 +756,7 @@ GST_BASE_RTP_AUDIO_PAYLOAD_CLASS
GST_IS_BASE_RTP_AUDIO_PAYLOAD
GST_IS_BASE_RTP_AUDIO_PAYLOAD_CLASS
gst_base_rtp_audio_payload_get_type
GstBaseRTPAudioPayloadPrivate
</SECTION>
<SECTION>
......@@ -763,7 +768,11 @@ GstBaseRTPDepayloadClass
GST_BASE_RTP_DEPAYLOAD_SINKPAD
GST_BASE_RTP_DEPAYLOAD_SRCPAD
gst_base_rtp_depayload_push
gst_base_rtp_depayload_push_ts
<SUBSECTION Standard>
GstBaseRTPDepayloadPrivate
GST_TYPE_BASE_RTP_DEPAYLOAD
GST_BASE_RTP_DEPAYLOAD
GST_BASE_RTP_DEPAYLOAD_CLASS
......@@ -798,6 +807,58 @@ GST_IS_BASE_RTP_PAYLOAD_CLASS
gst_basertppayload_get_type
</SECTION>
<SECTION>
<FILE>gstrtcpbuffer</FILE>
<INCLUDE>gst/rtp/gstrtcpbuffer.h</INCLUDE>
GST_RTCP_VERSION
GST_RTCP_MAX_SDES
GST_RTCP_VALID_MASK
GST_RTCP_VALID_VALUE
GstRTCPType
GstRTCPPacket
GstRTCPSDESType
gst_rtcp_buffer_add_packet
gst_rtcp_buffer_get_first_packet
gst_rtcp_buffer_get_packet_count
gst_rtcp_buffer_new_copy_data
gst_rtcp_buffer_new_take_data
gst_rtcp_buffer_validate
gst_rtcp_buffer_validate_data
gst_rtcp_packet_add_rb
gst_rtcp_packet_bye_add_ssrc
gst_rtcp_packet_bye_add_ssrcs
gst_rtcp_packet_bye_get_nth_ssrc
gst_rtcp_packet_bye_get_reason
gst_rtcp_packet_bye_get_reason_len
gst_rtcp_packet_bye_get_ssrc_count
gst_rtcp_packet_bye_set_reason
gst_rtcp_packet_get_count
gst_rtcp_packet_get_length
gst_rtcp_packet_get_padding
gst_rtcp_packet_get_rb
gst_rtcp_packet_get_rb_count
gst_rtcp_packet_get_type
gst_rtcp_packet_move_to_next
gst_rtcp_packet_remove
gst_rtcp_packet_rr_get_ssrc
gst_rtcp_packet_rr_set_ssrc
gst_rtcp_packet_sdes_first_chunk
gst_rtcp_packet_sdes_first_item
gst_rtcp_packet_sdes_get_chunk_count
gst_rtcp_packet_sdes_get_item
gst_rtcp_packet_sdes_get_ssrc
gst_rtcp_packet_sdes_next_chunk
gst_rtcp_packet_sdes_next_item
gst_rtcp_packet_set_rb
gst_rtcp_packet_sr_get_sender_info
gst_rtcp_packet_sr_set_sender_info
<SUBSECTION Standard>
</SECTION>
<SECTION>
<FILE>gstrtpbuffer</FILE>
<INCLUDE>gst/rtp/gstrtpbuffer.h</INCLUDE>
......@@ -809,6 +870,7 @@ gst_rtp_buffer_allocate_data
gst_rtp_buffer_calc_header_len
gst_rtp_buffer_calc_packet_len
gst_rtp_buffer_calc_payload_len
gst_rtp_buffer_default_clock_rate
gst_rtp_buffer_get_csrc
gst_rtp_buffer_get_csrc_count
gst_rtp_buffer_get_extension
......@@ -857,6 +919,26 @@ GST_RTP_PAYLOAD_PCMA_STRING
GST_RTP_PAYLOAD_PCMU_STRING
GST_RTP_PAYLOAD_TS41_STRING
GST_RTP_PAYLOAD_TS48_STRING
GST_RTP_PAYLOAD_1016_STRING
GST_RTP_PAYLOAD_CELLB_STRING
GST_RTP_PAYLOAD_CN_STRING
GST_RTP_PAYLOAD_DVI4_11025_STRING
GST_RTP_PAYLOAD_DVI4_16000_STRING
GST_RTP_PAYLOAD_DVI4_22050_STRING
GST_RTP_PAYLOAD_DVI4_8000_STRING
GST_RTP_PAYLOAD_G721_STRING
GST_RTP_PAYLOAD_G722_STRING
GST_RTP_PAYLOAD_G723_53
GST_RTP_PAYLOAD_G723_63
GST_RTP_PAYLOAD_G723_STRING
GST_RTP_PAYLOAD_H261_STRING
GST_RTP_PAYLOAD_JPEG_STRING
GST_RTP_PAYLOAD_LPC_STRING
GST_RTP_PAYLOAD_MP2T_STRING
GST_RTP_PAYLOAD_NV_STRING
GST_RTP_PAYLOAD_QCELP_STRING
GST_RTP_PAYLOAD_TS41
GST_RTP_PAYLOAD_TS48
</SECTION>
# tag
......
libgstrtpincludedir = $(includedir)/gstreamer-@GST_MAJORMINOR@/gst/rtp
libgstrtpinclude_HEADERS = gstrtpbuffer.h \
gstrtcpbuffer.h \
gstbasertpaudiopayload.h \
gstbasertppayload.h \
gstbasertpdepayload.h
......@@ -8,6 +9,7 @@ libgstrtpinclude_HEADERS = gstrtpbuffer.h \
lib_LTLIBRARIES = libgstrtp-@GST_MAJORMINOR@.la
libgstrtp_@GST_MAJORMINOR@_la_SOURCES = gstrtpbuffer.c \
gstrtcpbuffer.c \
gstbasertpaudiopayload.c \
gstbasertppayload.c \
gstbasertpdepayload.c
......
......@@ -598,6 +598,19 @@ gst_base_rtp_audio_payload_handle_sample_based_buffer (GstBaseRTPPayload *
return ret;
}
/**
* gst_base_rtp_audio_payload_push:
* @basepayload: a #GstBaseRTPPayload
* @data: data to set as payload
* @payload_len: length of payload
* @timestamp: a #GstClockTime
*
* Create an RTP buffer and store @payload_len bytes of @data as the
* payload. Set the timestamp on the new buffer to @timestamp before pushing
* the buffer downstream.
*
* Returns: a #GstFlowReturn
*/
GstFlowReturn
gst_base_rtp_audio_payload_push (GstBaseRTPPayload * basepayload,
const guint8 * data, guint payload_len, GstClockTime timestamp)
......@@ -712,14 +725,22 @@ gst_base_rtp_payload_audio_handle_event (GstPad * pad, GstEvent * event,
return res;
}
/**
* gst_base_rtp_audio_payload_get_adapter:
* @basertpaudiopayload: a #GstBaseRTPAudioPayload
*
* Gets the internal adapter used by the depayloader.
*
* Returns: a #GstAdapter.
*/
GstAdapter *
gst_base_rtp_audio_payload_get_adapter (GstBaseRTPAudioPayload
* basertpaudiopayload)
{
if (basertpaudiopayload->priv->adapter == NULL) {
return NULL;
}
GstAdapter *adapter;
if ((adapter = basertpaudiopayload->priv->adapter))
g_object_ref (adapter);
g_object_ref (G_OBJECT (basertpaudiopayload->priv->adapter));
return basertpaudiopayload->priv->adapter;
return adapter;
}
This diff is collapsed.
/* GStreamer
* Copyright (C) <2007> Wim Taymans <wim@fluendo.com>
*
* gstrtcpbuffer.h: various helper functions to manipulate buffers
* with RTCP payload.
*
* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifndef __GST_RTCPBUFFER_H__
#define __GST_RTCPBUFFER_H__
#include <gst/gst.h>
G_BEGIN_DECLS
/**
* GST_RTCP_VERSION:
*
* The supported RTCP version 2.
*/
#define GST_RTCP_VERSION 2
/**
* GstRTCPType:
* @GST_RTCP_TYPE_INVALID: Invalid type
* @GST_RTCP_TYPE_SR: Sender report
* @GST_RTCP_TYPE_RR: Receiver report
* @GST_RTCP_TYPE_SDES: Source description
* @GST_RTCP_TYPE_BYE: Goodbye
* @GST_RTCP_TYPE_APP: Application defined
*
* Different RTCP packet types.
*/
typedef enum
{
GST_RTCP_TYPE_INVALID = 0,
GST_RTCP_TYPE_SR = 200,
GST_RTCP_TYPE_RR = 201,
GST_RTCP_TYPE_SDES = 202,
GST_RTCP_TYPE_BYE = 203,
GST_RTCP_TYPE_APP = 204
} GstRTCPType;
/**
* GstRTCPSDESType:
* @GST_RTCP_SDES_INVALID: Invalid SDES entry
* @GST_RTCP_SDES_END: End of SDES list
* @GST_RTCP_SDES_CNAME: Canonical name
* @GST_RTCP_SDES_NAME: User name
* @GST_RTCP_SDES_EMAIL: User's electronic mail address
* @GST_RTCP_SDES_PHONE: User's phone number
* @GST_RTCP_SDES_LOC: Geographic user location
* @GST_RTCP_SDES_TOOL: Name of application or tool
* @GST_RTCP_SDES_NOTE: Notice about the source
* @GST_RTCP_SDES_PRIV: Private extensions
*
* Different types of SDES content.
*/
typedef enum
{
GST_RTCP_SDES_INVALID = -1,
GST_RTCP_SDES_END = 0,
GST_RTCP_SDES_CNAME = 1,
GST_RTCP_SDES_NAME = 2,
GST_RTCP_SDES_EMAIL = 3,
GST_RTCP_SDES_PHONE = 4,
GST_RTCP_SDES_LOC = 5,
GST_RTCP_SDES_TOOL = 6,
GST_RTCP_SDES_NOTE = 7,
GST_RTCP_SDES_PRIV = 8
} GstRTCPSDESType;
/**
* GST_RTCP_MAX_SDES:
*
* The maximum text length for an SDES item.
*/
#define GST_RTCP_MAX_SDES 255
/**
* GST_RTCP_VALID_MASK:
*
* Mask for version, padding bit and packet type pair
*/
#define GST_RTCP_VALID_MASK (0xc000 | 0x2000 | 0xfe)
/**
* GST_RTCP_VALID_VALUE:
*
* Valid value for the first two bytes of an RTCP packet after applying
* #GST_RTCP_VALID_MASK to them.
*/
#define GST_RTCP_VALID_VALUE ((GST_RTCP_VERSION << 14) | GST_RTCP_TYPE_SR)
typedef struct _GstRTCPPacket GstRTCPPacket;
/**
* GstRTCPPacket:
* @buffer: pointer to RTCP buffer
* @offset: offset of packet in buffer data
*
* Data structure that points to a packet at @offset in @buffer.
* The size of the structure is made public to allow stack allocations.
*/
struct _GstRTCPPacket
{
GstBuffer *buffer;
guint offset;
/*< private >*/
gboolean padding; /* padding field of current packet */
guint8 count; /* count field of current packet */
GstRTCPType type; /* type of current packet */
guint16 length; /* length of current packet in 32-bits words */
guint chunk_offset; /* current chunk offset for navigating SDES */
guint item_offset; /* current item offset for navigating SDE */
};
/* creating buffers */
GstBuffer* gst_rtcp_buffer_new_take_data (gpointer data, guint len);
GstBuffer* gst_rtcp_buffer_new_copy_data (gpointer data, guint len);
gboolean gst_rtcp_buffer_validate_data (guint8 *data, guint len);
gboolean gst_rtcp_buffer_validate (GstBuffer *buffer);
/* adding/retrieving packets */
guint gst_rtcp_buffer_get_packet_count (GstBuffer *buffer);
gboolean gst_rtcp_buffer_get_first_packet (GstBuffer *buffer, GstRTCPPacket *packet);
gboolean gst_rtcp_packet_move_to_next (GstRTCPPacket *packet);
void gst_rtcp_buffer_add_packet (GstBuffer *buffer, GstRTCPType type,
GstRTCPPacket *packet);
void gst_rtcp_packet_remove (GstRTCPPacket *packet);
/* working with packets */
gboolean gst_rtcp_packet_get_padding (GstRTCPPacket *packet);
guint8 gst_rtcp_packet_get_count (GstRTCPPacket *packet);
GstRTCPType gst_rtcp_packet_get_type (GstRTCPPacket *packet);
guint16 gst_rtcp_packet_get_length (GstRTCPPacket *packet);
/* sender reports */
void gst_rtcp_packet_sr_get_sender_info (GstRTCPPacket *packet, guint32 *ssrc,
guint64 *ntptime, guint32 *rtptime,
guint32 *packet_count, guint32 *octet_count);
void gst_rtcp_packet_sr_set_sender_info (GstRTCPPacket *packet, guint32 ssrc,
guint64 ntptime, guint32 rtptime,
guint32 packet_count, guint32 octet_count);
/* receiver reports */
guint32 gst_rtcp_packet_rr_get_ssrc (GstRTCPPacket *packet);
void gst_rtcp_packet_rr_set_ssrc (GstRTCPPacket *packet, guint32 ssrc);
/* report blocks for SR and RR */
guint gst_rtcp_packet_get_rb_count (GstRTCPPacket *packet);
void gst_rtcp_packet_get_rb (GstRTCPPacket *packet, guint nth, guint32 *ssrc,
guint8 *fractionlost, gint32 *packetslost,
guint32 *exthighestseq, guint32 *jitter,
guint32 *lsr, guint32 *dlsr);
void gst_rtcp_packet_add_rb (GstRTCPPacket *packet, guint32 ssrc,
guint8 fractionlost, gint32 packetslost,
guint32 exthighestseq, guint32 jitter,
guint32 lsr, guint32 dlsr);
void gst_rtcp_packet_set_rb (GstRTCPPacket *packet, guint nth, guint32 ssrc,
guint8 fractionlost, gint32 packetslost,
guint32 exthighestseq, guint32 jitter,
guint32 lsr, guint32 dlsr);
/* source description packet */
guint gst_rtcp_packet_sdes_get_chunk_count (GstRTCPPacket *packet);
gboolean gst_rtcp_packet_sdes_first_chunk (GstRTCPPacket *packet);
gboolean gst_rtcp_packet_sdes_next_chunk (GstRTCPPacket *packet);
guint32 gst_rtcp_packet_sdes_get_ssrc (GstRTCPPacket *packet);
gboolean gst_rtcp_packet_sdes_first_item (GstRTCPPacket *packet);
gboolean gst_rtcp_packet_sdes_next_item (GstRTCPPacket *packet);
gboolean gst_rtcp_packet_sdes_get_item (GstRTCPPacket *packet,
GstRTCPSDESType *type, guint8 *len,
gchar **data);
/* bye packet */
guint gst_rtcp_packet_bye_get_ssrc_count (GstRTCPPacket *packet);
guint32 gst_rtcp_packet_bye_get_nth_ssrc (GstRTCPPacket *packet, guint nth);
void gst_rtcp_packet_bye_add_ssrc (GstRTCPPacket *packet, guint32 ssrc);
void gst_rtcp_packet_bye_add_ssrcs (GstRTCPPacket *packet, guint32 *ssrc, guint len);
guint8 gst_rtcp_packet_bye_get_reason_len (GstRTCPPacket *packet);
gchar* gst_rtcp_packet_bye_get_reason (GstRTCPPacket *packet);
gboolean gst_rtcp_packet_bye_set_reason (GstRTCPPacket *packet, const gchar *reason);
G_END_DECLS
#endif /* __GST_RTCPBUFFER_H__ */
Markdown is supported
0% or .
You are about to add 0 people to the discussion. Proceed with caution.
Finish editing this message first!
Please register or to comment