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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.
This commit is contained in:
Wim Taymans
parent
dfdd873f6a
commit
0a39f494b5
7 changed files with 1536 additions and 5 deletions
36
ChangeLog
36
ChangeLog
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@ -1,3 +1,39 @@
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2007-03-29 Wim Taymans <wim@fluendo.com>
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* docs/libs/gst-plugins-base-libs-docs.sgml:
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* docs/libs/gst-plugins-base-libs-sections.txt:
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* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
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(gst_base_rtp_audio_payload_get_adapter):
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Add RTCP docs.
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Fix some more docs.
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* gst-libs/gst/rtp/Makefile.am:
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* gst-libs/gst/rtp/gstrtcpbuffer.c:
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(gst_rtcp_buffer_new_take_data), (gst_rtcp_buffer_new_copy_data),
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(gst_rtcp_buffer_validate_data), (gst_rtcp_buffer_validate),
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(gst_rtcp_buffer_get_packet_count), (read_packet_header),
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(gst_rtcp_buffer_get_first_packet), (gst_rtcp_packet_move_to_next),
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(gst_rtcp_buffer_add_packet), (gst_rtcp_packet_remove),
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(gst_rtcp_packet_get_padding), (gst_rtcp_packet_get_type),
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(gst_rtcp_packet_get_count), (gst_rtcp_packet_get_length),
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(gst_rtcp_packet_sr_get_sender_info),
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(gst_rtcp_packet_sr_set_sender_info),
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(gst_rtcp_packet_rr_get_ssrc), (gst_rtcp_packet_rr_set_ssrc),
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(gst_rtcp_packet_get_rb_count), (gst_rtcp_packet_get_rb),
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(gst_rtcp_packet_add_rb), (gst_rtcp_packet_set_rb),
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(gst_rtcp_packet_sdes_get_chunk_count),
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(gst_rtcp_packet_sdes_first_chunk),
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(gst_rtcp_packet_sdes_next_chunk), (gst_rtcp_packet_sdes_get_ssrc),
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(gst_rtcp_packet_sdes_first_item),
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(gst_rtcp_packet_sdes_next_item), (gst_rtcp_packet_sdes_get_item),
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(gst_rtcp_packet_bye_get_ssrc_count),
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(gst_rtcp_packet_bye_get_nth_ssrc), (gst_rtcp_packet_bye_add_ssrc),
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(gst_rtcp_packet_bye_add_ssrcs), (get_reason_offset),
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(gst_rtcp_packet_bye_get_reason_len),
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(gst_rtcp_packet_bye_get_reason), (gst_rtcp_packet_bye_set_reason):
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* gst-libs/gst/rtp/gstrtcpbuffer.h:
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Add new helper object for parsing and creating RTCP messages.
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2007-03-29 Sebastian Dröge <slomo@circular-chaos.org>
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2007-03-29 Sebastian Dröge <slomo@circular-chaos.org>
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* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
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* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
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@ -41,6 +41,7 @@
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<!ENTITY GstBaseRtpDepayload SYSTEM "xml/gstbasertpdepayload.xml">
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<!ENTITY GstBaseRtpDepayload SYSTEM "xml/gstbasertpdepayload.xml">
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<!ENTITY GstBaseRtpPayload SYSTEM "xml/gstbasertppayload.xml">
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<!ENTITY GstBaseRtpPayload SYSTEM "xml/gstbasertppayload.xml">
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<!ENTITY GstRtpBuffer SYSTEM "xml/gstrtpbuffer.xml">
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<!ENTITY GstRtpBuffer SYSTEM "xml/gstrtpbuffer.xml">
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<!ENTITY GstRtcpBuffer SYSTEM "xml/gstrtcpbuffer.xml">
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<!-- tag -->
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<!-- tag -->
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<!ENTITY GstTag SYSTEM "xml/gsttag.xml">
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<!ENTITY GstTag SYSTEM "xml/gsttag.xml">
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<!ENTITY GstTagVorbis SYSTEM "xml/gsttagvorbis.xml">
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<!ENTITY GstTagVorbis SYSTEM "xml/gsttagvorbis.xml">
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@ -164,6 +165,7 @@
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&GstBaseRtpDepayload;
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&GstBaseRtpDepayload;
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&GstBaseRtpPayload;
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&GstBaseRtpPayload;
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&GstRtpBuffer;
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&GstRtpBuffer;
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&GstRtcpBuffer;
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</chapter>
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</chapter>
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<chapter id="gstreamer-tag">
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<chapter id="gstreamer-tag">
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@ -93,6 +93,7 @@ GST_AUDIO_SRC_GET_CLASS
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<INCLUDE>gst/audio/gstbaseaudiosink.h</INCLUDE>
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<INCLUDE>gst/audio/gstbaseaudiosink.h</INCLUDE>
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GstBaseAudioSink
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GstBaseAudioSink
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GstBaseAudioSinkClass
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GstBaseAudioSinkClass
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GstBaseAudioSinkSlaveMethod
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GST_BASE_AUDIO_SINK_CLOCK
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GST_BASE_AUDIO_SINK_CLOCK
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GST_BASE_AUDIO_SINK_PAD
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GST_BASE_AUDIO_SINK_PAD
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@ -105,6 +106,7 @@ gst_base_audio_sink_get_type
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GST_BASE_AUDIO_SINK_CLASS
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GST_BASE_AUDIO_SINK_CLASS
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GST_IS_BASE_AUDIO_SINK_CLASS
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GST_IS_BASE_AUDIO_SINK_CLASS
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GST_BASE_AUDIO_SINK_GET_CLASS
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GST_BASE_AUDIO_SINK_GET_CLASS
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GstBaseAudioSinkPrivate
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</SECTION>
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</SECTION>
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<SECTION>
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<SECTION>
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@ -745,6 +747,8 @@ gst_base_rtp_audio_payload_set_frame_based
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gst_base_rtp_audio_payload_set_frame_options
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gst_base_rtp_audio_payload_set_frame_options
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gst_base_rtp_audio_payload_set_sample_based
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gst_base_rtp_audio_payload_set_sample_based
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gst_base_rtp_audio_payload_set_sample_options
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gst_base_rtp_audio_payload_set_sample_options
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gst_base_rtp_audio_payload_get_adapter
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gst_base_rtp_audio_payload_push
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<SUBSECTION Standard>
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<SUBSECTION Standard>
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GST_TYPE_BASE_RTP_AUDIO_PAYLOAD
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GST_TYPE_BASE_RTP_AUDIO_PAYLOAD
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GST_BASE_RTP_AUDIO_PAYLOAD
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GST_BASE_RTP_AUDIO_PAYLOAD
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@ -752,6 +756,7 @@ GST_BASE_RTP_AUDIO_PAYLOAD_CLASS
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GST_IS_BASE_RTP_AUDIO_PAYLOAD
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GST_IS_BASE_RTP_AUDIO_PAYLOAD
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GST_IS_BASE_RTP_AUDIO_PAYLOAD_CLASS
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GST_IS_BASE_RTP_AUDIO_PAYLOAD_CLASS
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gst_base_rtp_audio_payload_get_type
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gst_base_rtp_audio_payload_get_type
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GstBaseRTPAudioPayloadPrivate
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</SECTION>
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</SECTION>
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<SECTION>
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<SECTION>
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@ -763,7 +768,11 @@ GstBaseRTPDepayloadClass
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GST_BASE_RTP_DEPAYLOAD_SINKPAD
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GST_BASE_RTP_DEPAYLOAD_SINKPAD
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GST_BASE_RTP_DEPAYLOAD_SRCPAD
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GST_BASE_RTP_DEPAYLOAD_SRCPAD
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gst_base_rtp_depayload_push
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gst_base_rtp_depayload_push_ts
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<SUBSECTION Standard>
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<SUBSECTION Standard>
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GstBaseRTPDepayloadPrivate
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GST_TYPE_BASE_RTP_DEPAYLOAD
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GST_TYPE_BASE_RTP_DEPAYLOAD
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GST_BASE_RTP_DEPAYLOAD
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GST_BASE_RTP_DEPAYLOAD
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GST_BASE_RTP_DEPAYLOAD_CLASS
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GST_BASE_RTP_DEPAYLOAD_CLASS
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@ -798,6 +807,58 @@ GST_IS_BASE_RTP_PAYLOAD_CLASS
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gst_basertppayload_get_type
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gst_basertppayload_get_type
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</SECTION>
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</SECTION>
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<SECTION>
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<FILE>gstrtcpbuffer</FILE>
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<INCLUDE>gst/rtp/gstrtcpbuffer.h</INCLUDE>
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GST_RTCP_VERSION
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GST_RTCP_MAX_SDES
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GST_RTCP_VALID_MASK
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GST_RTCP_VALID_VALUE
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GstRTCPType
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GstRTCPPacket
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GstRTCPSDESType
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gst_rtcp_buffer_add_packet
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gst_rtcp_buffer_get_first_packet
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gst_rtcp_buffer_get_packet_count
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gst_rtcp_buffer_new_copy_data
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gst_rtcp_buffer_new_take_data
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gst_rtcp_buffer_validate
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gst_rtcp_buffer_validate_data
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gst_rtcp_packet_add_rb
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gst_rtcp_packet_bye_add_ssrc
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gst_rtcp_packet_bye_add_ssrcs
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gst_rtcp_packet_bye_get_nth_ssrc
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gst_rtcp_packet_bye_get_reason
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gst_rtcp_packet_bye_get_reason_len
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gst_rtcp_packet_bye_get_ssrc_count
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gst_rtcp_packet_bye_set_reason
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gst_rtcp_packet_get_count
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gst_rtcp_packet_get_length
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gst_rtcp_packet_get_padding
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gst_rtcp_packet_get_rb
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gst_rtcp_packet_get_rb_count
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gst_rtcp_packet_get_type
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gst_rtcp_packet_move_to_next
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gst_rtcp_packet_remove
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gst_rtcp_packet_rr_get_ssrc
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gst_rtcp_packet_rr_set_ssrc
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gst_rtcp_packet_sdes_first_chunk
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gst_rtcp_packet_sdes_first_item
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gst_rtcp_packet_sdes_get_chunk_count
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gst_rtcp_packet_sdes_get_item
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gst_rtcp_packet_sdes_get_ssrc
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gst_rtcp_packet_sdes_next_chunk
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gst_rtcp_packet_sdes_next_item
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gst_rtcp_packet_set_rb
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gst_rtcp_packet_sr_get_sender_info
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gst_rtcp_packet_sr_set_sender_info
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<SUBSECTION Standard>
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</SECTION>
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<SECTION>
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<SECTION>
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<FILE>gstrtpbuffer</FILE>
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<FILE>gstrtpbuffer</FILE>
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<INCLUDE>gst/rtp/gstrtpbuffer.h</INCLUDE>
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<INCLUDE>gst/rtp/gstrtpbuffer.h</INCLUDE>
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@ -809,6 +870,7 @@ gst_rtp_buffer_allocate_data
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gst_rtp_buffer_calc_header_len
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gst_rtp_buffer_calc_header_len
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gst_rtp_buffer_calc_packet_len
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gst_rtp_buffer_calc_packet_len
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gst_rtp_buffer_calc_payload_len
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gst_rtp_buffer_calc_payload_len
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gst_rtp_buffer_default_clock_rate
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gst_rtp_buffer_get_csrc
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gst_rtp_buffer_get_csrc
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gst_rtp_buffer_get_csrc_count
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gst_rtp_buffer_get_csrc_count
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gst_rtp_buffer_get_extension
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gst_rtp_buffer_get_extension
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@ -857,6 +919,26 @@ GST_RTP_PAYLOAD_PCMA_STRING
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GST_RTP_PAYLOAD_PCMU_STRING
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GST_RTP_PAYLOAD_PCMU_STRING
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GST_RTP_PAYLOAD_TS41_STRING
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GST_RTP_PAYLOAD_TS41_STRING
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GST_RTP_PAYLOAD_TS48_STRING
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GST_RTP_PAYLOAD_TS48_STRING
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GST_RTP_PAYLOAD_1016_STRING
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GST_RTP_PAYLOAD_CELLB_STRING
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GST_RTP_PAYLOAD_CN_STRING
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GST_RTP_PAYLOAD_DVI4_11025_STRING
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GST_RTP_PAYLOAD_DVI4_16000_STRING
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GST_RTP_PAYLOAD_DVI4_22050_STRING
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GST_RTP_PAYLOAD_DVI4_8000_STRING
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GST_RTP_PAYLOAD_G721_STRING
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GST_RTP_PAYLOAD_G722_STRING
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GST_RTP_PAYLOAD_G723_53
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GST_RTP_PAYLOAD_G723_63
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GST_RTP_PAYLOAD_G723_STRING
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GST_RTP_PAYLOAD_H261_STRING
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GST_RTP_PAYLOAD_JPEG_STRING
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GST_RTP_PAYLOAD_LPC_STRING
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GST_RTP_PAYLOAD_MP2T_STRING
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GST_RTP_PAYLOAD_NV_STRING
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GST_RTP_PAYLOAD_QCELP_STRING
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GST_RTP_PAYLOAD_TS41
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GST_RTP_PAYLOAD_TS48
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</SECTION>
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</SECTION>
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# tag
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# tag
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@ -1,6 +1,7 @@
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libgstrtpincludedir = $(includedir)/gstreamer-@GST_MAJORMINOR@/gst/rtp
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libgstrtpincludedir = $(includedir)/gstreamer-@GST_MAJORMINOR@/gst/rtp
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libgstrtpinclude_HEADERS = gstrtpbuffer.h \
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libgstrtpinclude_HEADERS = gstrtpbuffer.h \
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gstrtcpbuffer.h \
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gstbasertpaudiopayload.h \
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gstbasertpaudiopayload.h \
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gstbasertppayload.h \
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gstbasertppayload.h \
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gstbasertpdepayload.h
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gstbasertpdepayload.h
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@ -8,6 +9,7 @@ libgstrtpinclude_HEADERS = gstrtpbuffer.h \
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lib_LTLIBRARIES = libgstrtp-@GST_MAJORMINOR@.la
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lib_LTLIBRARIES = libgstrtp-@GST_MAJORMINOR@.la
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libgstrtp_@GST_MAJORMINOR@_la_SOURCES = gstrtpbuffer.c \
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libgstrtp_@GST_MAJORMINOR@_la_SOURCES = gstrtpbuffer.c \
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gstrtcpbuffer.c \
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gstbasertpaudiopayload.c \
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gstbasertpaudiopayload.c \
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gstbasertppayload.c \
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gstbasertppayload.c \
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gstbasertpdepayload.c
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gstbasertpdepayload.c
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@ -598,6 +598,19 @@ gst_base_rtp_audio_payload_handle_sample_based_buffer (GstBaseRTPPayload *
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return ret;
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return ret;
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}
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}
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/**
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* gst_base_rtp_audio_payload_push:
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* @basepayload: a #GstBaseRTPPayload
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* @data: data to set as payload
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* @payload_len: length of payload
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* @timestamp: a #GstClockTime
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*
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* Create an RTP buffer and store @payload_len bytes of @data as the
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* payload. Set the timestamp on the new buffer to @timestamp before pushing
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* the buffer downstream.
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*
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* Returns: a #GstFlowReturn
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*/
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GstFlowReturn
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GstFlowReturn
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gst_base_rtp_audio_payload_push (GstBaseRTPPayload * basepayload,
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gst_base_rtp_audio_payload_push (GstBaseRTPPayload * basepayload,
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const guint8 * data, guint payload_len, GstClockTime timestamp)
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const guint8 * data, guint payload_len, GstClockTime timestamp)
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@ -712,14 +725,22 @@ gst_base_rtp_payload_audio_handle_event (GstPad * pad, GstEvent * event,
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return res;
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return res;
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}
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}
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/**
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* gst_base_rtp_audio_payload_get_adapter:
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* @basertpaudiopayload: a #GstBaseRTPAudioPayload
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*
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* Gets the internal adapter used by the depayloader.
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*
|
||||||
|
* Returns: a #GstAdapter.
|
||||||
|
*/
|
||||||
GstAdapter *
|
GstAdapter *
|
||||||
gst_base_rtp_audio_payload_get_adapter (GstBaseRTPAudioPayload
|
gst_base_rtp_audio_payload_get_adapter (GstBaseRTPAudioPayload
|
||||||
* basertpaudiopayload)
|
* basertpaudiopayload)
|
||||||
{
|
{
|
||||||
if (basertpaudiopayload->priv->adapter == NULL) {
|
GstAdapter *adapter;
|
||||||
return NULL;
|
|
||||||
}
|
|
||||||
|
|
||||||
g_object_ref (G_OBJECT (basertpaudiopayload->priv->adapter));
|
if ((adapter = basertpaudiopayload->priv->adapter))
|
||||||
return basertpaudiopayload->priv->adapter;
|
g_object_ref (adapter);
|
||||||
|
|
||||||
|
return adapter;
|
||||||
}
|
}
|
||||||
|
|
|
||||||
1182
gst-libs/gst/rtp/gstrtcpbuffer.c
Normal file
1182
gst-libs/gst/rtp/gstrtcpbuffer.c
Normal file
File diff suppressed because it is too large
Load diff
206
gst-libs/gst/rtp/gstrtcpbuffer.h
Normal file
206
gst-libs/gst/rtp/gstrtcpbuffer.h
Normal file
|
|
@ -0,0 +1,206 @@
|
||||||
|
/* 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__ */
|
||||||
|
|
||||||
Loading…
Reference in a new issue