Commit graph

24 commits

Author SHA1 Message Date
Mersad Jelacic
ed814cbaed gst-libs/gst/rtp/gstbasertpaudiopayload.*: API: add gst_base_rtp_audio_payload_set_samplebits_options() to make it po...
Original commit message from CVS:
Patch by: Mersad Jelacic  <mersad at axis dot com>
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
API: add gst_base_rtp_audio_payload_set_samplebits_options() to make it
possible to specify the sample size in bits. (#509637)
2008-03-03 16:11:50 +00:00
Wim Taymans
d8c28a99c8 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Fix potential leaks.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_handle_frame_based_buffer),
(gst_base_rtp_audio_payload_handle_sample_based_buffer):
Fix potential leaks.
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_chain):
Fix leak when there is no function configured.
2008-02-12 14:51:26 +00:00
Wim Taymans
56e39e7c1c gst-libs/gst/rtp/gstbasertpaudiopayload.c: Return FALSE from the event handler to let the parent class handle the event.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_payload_audio_handle_event):
Return FALSE from the event handler to let the parent class handle the
event.
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_chain), (gst_base_rtp_depayload_push_full):
Mark outgoing buffers as DISCONT if the incomming buffer was DISCONT.
* gst-libs/gst/rtp/gstbasertppayload.c:
Bump the MTU to 1400.
2007-09-04 16:18:48 +00:00
Stefan Kost
aac0353ce6 gst-libs/gst/: Make gtk-doc happy.
Original commit message from CVS:
* gst-libs/gst/interfaces/propertyprobe.c:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/tag/gstvorbistag.c:
Make gtk-doc happy.
2007-07-10 20:46:41 +00:00
Wim Taymans
7ace85992a gst-libs/gst/rtp/gstbasertpaudiopayload.c: Some cleanups, remove minptime property as it is now in the parent class.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_class_init),
(gst_base_rtp_audio_payload_init),
(gst_base_rtp_audio_payload_finalize),
(gst_base_rtp_audio_payload_handle_frame_based_buffer),
(gst_base_rtp_audio_payload_handle_sample_based_buffer),
(gst_base_rtp_payload_audio_handle_event):
Some cleanups, remove minptime property as it is now in the parent
class.
Override parent class event function.
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_class_init), (gst_basertppayload_init),
(gst_basertppayload_event), (gst_basertppayload_set_property),
(gst_basertppayload_get_property):
* gst-libs/gst/rtp/gstbasertppayload.h:
Add min-ptime property.
Add handle-event vmethod. Fixes #415001.
2007-05-21 09:45:28 +00:00
Sébastien Moutte
c88306fe26 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Move variable declaration before the first instruction.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_handle_frame_based_buffer):
Move variable declaration before the first instruction.
* gst/videotestsrc/videotestsrc.c:
Define M_PI if it's not defined yet.
* win32/common/libgstrtp.def:
Add new exported functions.
2007-05-09 21:17:40 +00:00
Stefan Kost
64a9674bd2 gst/: gst/audiotestsrc/gstaudiotestsrc.c
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst/adder/gstadder.c:
* gst/audiotestsrc/gstaudiotestsrc.c
(gst_audio_test_src_create_white_noise):
* gst/videotestsrc/gstvideotestsrc.c:
* gst/volume/gstvolume.c (VOLUME_UNITY_INT16,
VOLUME_UNITY_INT16_BIT_SHIFT, VOLUME_MAX_DOUBLE,
volume_sink_template, volume_src_template, gst_volume_init,
volume_process_double, volume_process_int16,
volume_process_int16_clamp):
Doc fixes and formatting.
2007-05-04 13:10:07 +00:00
Olivier Crete
e3ff444d30 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Chain up to parent class in dispose function; get rid of unnecessary 'dipo...
Original commit message from CVS:
Patch by: Olivier Crete  <tester at tester ca>
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_class_init),
(gst_base_rtp_audio_payload_init),
(gst_base_rtp_audio_payload_dispose):
Chain up to parent class in dispose function; get rid of
unnecessary 'diposed' flag in private structure (#415001).
2007-04-21 15:25:22 +00:00
Tim-Philipp Müller
71d77fbecc Some minor docs fixes and additions; also add missing 'Since' bits.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs.types:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_class_init):
* gst-libs/gst/rtp/gstbasertpdepayload.c:
* gst-libs/gst/rtp/gstbasertppayload.c:
Some minor docs fixes and additions; also add missing 'Since' bits.
2007-04-21 15:10:25 +00:00
Zeeshan Ali
80ebb9eb42 gst-libs/gst/rtp/gstbasertpaudiopayload.*: The recently-added gst_base_rtp_audio_payload_push() should take an object...
Original commit message from CVS:
Patch by: Zeeshan Ali  <zeenix gmail com>
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_handle_frame_based_buffer),
(gst_base_rtp_audio_payload_handle_sample_based_buffer),
(gst_base_rtp_audio_payload_push):
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
The recently-added gst_base_rtp_audio_payload_push() should take an
object of type GstBaseRTPAudioPayload as first argument (#431672).
2007-04-21 14:40:45 +00:00
Wim Taymans
0a39f494b5 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.
2007-03-29 16:20:31 +00:00
Philippe Kalaf
b6d7f65463 gst-libs/gst/rtp/gstbasertpaudiopayload.*: olivier.crete@collabora.co.uk.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
Add min-ptime property to RTP base audio payloader. Patch by
olivier.crete@collabora.co.uk.
Fixes #415001

Indentation/whitespace/documentation fixes.
2007-03-14 21:11:18 +00:00
Sébastien Moutte
9caee48ed4 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Use gst_gdouble_to_guint64 for conversions.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:(gst_base_rtp_audio_payload_handle_frame_based_buffer):
Use gst_gdouble_to_guint64 for conversions.
* win32/common/config.h.in:
Add a define for GST_INSTALL_PLUGINS_HELPER
* win32/common/libgstaudio.def:
* win32/common/libgstcdda.def:
* win32/common/libgstnetbuffer.def:
* win32/common/libgstrtp.def:
* win32/common/libgutils.def:
Add new exported functions.
* win32/vs6/gst_plugins_base.dsw:
* win32/vs6/libgstdecodebin.dsp:
* win32/vs6/libgstnetbuffer.dsp:
* win32/vs6/libgstplaybin.dsp:
* win32/vs6/libgstrtp.dsp:
* win32/vs6/libgstvorbis.dsp:
* win32/vs6/libgstcdda.dsp:
* win32/vs6/libgstgdp.dsp:
* win32/vs6/libgstutils.dsp:
Update and add new project files.
2007-02-10 19:27:48 +00:00
Jens Granseuer
595217e840 Declare variables at the beginning of a block. Fixes #383195.
Original commit message from CVS:
Patch by: Jens Granseuer <jensgr at gmx net>
* gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_handle_frame_based_buffer),
(gst_base_rtp_audio_payload_handle_sample_based_buffer):
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_fixate):
Declare variables at the beginning of a block. Fixes #383195.
2006-12-09 15:12:38 +00:00
Wim Taymans
351622d028 gst-libs/gst/rtp/: Fix and activate base audio payloader.
Original commit message from CVS:
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_init):
Fix and activate base audio payloader.
2006-10-31 10:49:19 +00:00
Philippe Kalaf
306ab03865 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Removed empty * between paragraphs
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
Removed empty * between paragraphs
2006-09-30 00:14:20 +00:00
Philippe Kalaf
5ba46c0866 gst-libs/gst/rtp/: Moved some documentation into .c file
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/rtp/README:
Moved some documentation into .c file
2006-09-29 23:50:53 +00:00
Philippe Kalaf
214a128382 gst-libs/gst/rtp/gstbasertpaudiopayload.*: Moved AudioCodecType into priv
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
Moved AudioCodecType into priv
Renamed all gst_basertpaudiopayload to gst_base_rtp_audio_payload prefixes
2006-09-27 00:13:29 +00:00
Philippe Kalaf
7e52276a83 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Fixed nasty memory leak
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
Fixed nasty memory leak
2006-06-29 12:21:06 +00:00
Tim-Philipp Müller
114a273f27 Use GST_DEBUG_CATEGORY_STATIC where possible (#342503).
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
* gst-libs/gst/rtp/gstbasertppayload.c:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/tcp/gstmultifdsink.c:
* gst/tcp/gsttcpclientsink.c:
* gst/tcp/gsttcpclientsrc.c:
* gst/tcp/gsttcpserversink.c:
* gst/tcp/gsttcpserversrc.c:
* gst/videorate/gstvideorate.c:
* gst/videotestsrc/gstvideotestsrc.c:
* sys/v4l/gstv4ljpegsrc.c:
* sys/v4l/gstv4lmjpegsink.c:
* sys/v4l/gstv4lsrc.c:
* tests/examples/seek/scrubby.c:
* tests/examples/seek/seek.c:
Use GST_DEBUG_CATEGORY_STATIC where possible (#342503).
2006-06-23 09:53:09 +00:00
Philippe Kalaf
0e710f94cc gst-libs/gst/rtp/gstbasertpaudiopayload.c: 80 line columns
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
80 line columns
Removed redundant floor()
2006-05-19 17:57:56 +00:00
Philippe Kalaf
8675bc89e4 gst-libs/gst/rtp/README: Some new documentation
Original commit message from CVS:
2006-05-18 Philippe Kalaf <philippe.kalaf@collabora.co.uk>

* gst-libs/gst/rtp/README:
Some new documentation
* gst-libs/gst/rtp/gstrtpbuffer.h:
Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
New RTP audio base payloader class. Supports frame or sample based codecs.
Not enabled in Makefile.am until approved.
2006-05-18 23:00:02 +00:00
Thomas Vander Stichele
cccfb086bc doc reparagraphing and DEBUG_FUNCPTRing
Original commit message from CVS:
doc reparagraphing and DEBUG_FUNCPTRing
2006-05-08 15:51:15 +00:00
Philippe Kalaf
a916af7c48 gst-libs/gst/rtp/gstrtpbuffer.h: Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children
Original commit message from CVS:
2006-04-12 Philippe Kalaf <philippe.kalaf@collabora.co.uk>

* gst-libs/gst/rtp/gstrtpbuffer.h:
Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
New RTP audio base payloader class. Supports frame or sample based codecs
2006-04-13 03:55:12 +00:00