Commit graph

2752 commits

Author SHA1 Message Date
Jan Schmidt
825cf238bb gst/rtsp/: g_base64_encode is a GLib 2.12 function. Use an equivalent taken from icecast to replace it. Relicensed fr...
Original commit message from CVS:
* gst/rtsp/Makefile.am:
* gst/rtsp/rtspconnection.c: (append_auth_header),
(rtsp_connection_send), (rtsp_connection_set_auth):
g_base64_encode is a GLib 2.12 function. Use an equivalent taken
from icecast to replace it. Relicensed from GPL courtesy of Mike
Smith.
2007-02-23 19:12:52 +00:00
Jan Schmidt
66df66daa2 gst/rtsp/: Implement simple Basic Authentication support so that urls like rtsp://user:pass@hostname/rtspstream work ...
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_finalize),
(gst_rtspsrc_create_stream), (rtsp_auth_method_to_string),
(gst_rtspsrc_parse_auth_hdr), (gst_rtspsrc_setup_auth),
(gst_rtspsrc_send), (gst_rtspsrc_try_send), (gst_rtspsrc_open),
(gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause),
(gst_rtspsrc_uri_set_uri):
* gst/rtsp/gstrtspsrc.h:
* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
(append_auth_header), (rtsp_connection_send),
(rtsp_connection_free), (rtsp_connection_set_auth):
* gst/rtsp/rtspconnection.h:
* gst/rtsp/rtspdefs.h:
* gst/rtsp/rtspurl.c: (rtsp_url_get_request_uri):
* gst/rtsp/rtspurl.h:
Implement simple Basic Authentication support so that urls like
rtsp://user:pass@hostname/rtspstream work on hosts that require
authentication.
2007-02-23 18:12:27 +00:00
Edgard Lima
fff672f930 Fix segfault when oppening a radio device.
Original commit message from CVS:
Fix segfault when oppening a radio device.
2007-02-22 17:53:26 +00:00
Stefan Kost
5c1b116dc8 Fix level for multi-channel case.
Original commit message from CVS:
* gst/level/gstlevel.c: (gst_level_set_caps),
(gst_level_transform_ip):
* sys/v4l2/README:
* tests/check/elements/level.c: (GST_START_TEST):
Fix level for multi-channel case.
2007-02-22 14:35:28 +00:00
Stefan Kost
6e44a9c618 gst/level/gstlevel.*: Use function pointer for process function and add process functions for float audio.
Original commit message from CVS:
* gst/level/gstlevel.c: (gst_level_init), (gst_level_set_caps),
(gst_level_transform_ip):
* gst/level/gstlevel.h:
Use function pointer for process function and add process functions
for float audio.
2007-02-21 10:18:12 +00:00
Stefan Kost
2d1b42029a sys/v4l2/v4l2src_calls.c: Readd GST_ELEMENT_ERROR if we can't reenque buffers after EIO, fixes #407369
Original commit message from CVS:
* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list),
(gst_v4l2src_grab_frame), (gst_v4l2src_set_capture),
(gst_v4l2src_capture_init):
Readd GST_ELEMENT_ERROR if we can't reenque buffers after EIO,
fixes #407369
2007-02-19 12:22:43 +00:00
Wim Taymans
bd4b1f680c gst/rtp/: Added simple mpeg transport stream payloader.
Original commit message from CVS:
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtpmp2tpay.c: (gst_rtp_mp2t_pay_base_init),
(gst_rtp_mp2t_pay_class_init), (gst_rtp_mp2t_pay_init),
(gst_rtp_mp2t_pay_setcaps), (gst_rtp_mp2t_pay_handle_buffer),
(gst_rtp_mp2t_pay_plugin_init):
* gst/rtp/gstrtpmp2tpay.h:
Added simple mpeg transport stream payloader.
2007-02-18 13:24:26 +00:00
Wim Taymans
7fd025043d gst/rtsp/URLS: Add example H264 rtsp url.
Original commit message from CVS:
* gst/rtsp/URLS:
Add example H264 rtsp url.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps),
(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
Don't convert values to lowercase or we might mess up base64 encoded
properties.
2007-02-16 12:32:01 +00:00
Wim Taymans
dc325990e0 gst/rtp/README: Fix case of string params.
Original commit message from CVS:
* gst/rtp/README:
Fix case of string params.
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_class_init),
(gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process):
Fix depayloader, support more packet types.
Add sync codes to make sure the packetizer can do its job.
* gst/rtp/gstrtpmp4gdepay.c:
* gst/rtp/gstrtpmp4gpay.c:
* gst/rtp/gstrtpmpvdepay.c: (gst_rtp_mpv_depay_process):
Fix caps case again.
2007-02-16 12:30:22 +00:00
Wim Taymans
e4b3dce677 gst/rtp/gstrtph264depay.c: Set right caps on output buffers.
Original commit message from CVS:
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_process):
Set right caps on output buffers.
2007-02-15 12:26:28 +00:00
Wim Taymans
df5916db2f gst/rtsp/sdpmessage.c: Clear stack allocated SDPMedia struct before calling _init() on it.
Original commit message from CVS:
* gst/rtsp/sdpmessage.c: (sdp_parse_line):
As spotted by: Peter Kjellerstedt  <pkj at axis com>:
Clear stack allocated SDPMedia struct before calling _init() on it.
Clarify this in the docs as well.
2007-02-14 17:04:47 +00:00
Jan Schmidt
3b5868a9e0 ext/gconf/gstgconfaudiosink.c: Don't reset the profile when going switching states, as it makes the element non-reusa...
Original commit message from CVS:
* ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_reset),
(do_change_child):
Don't reset the profile when going switching states, as it makes
the element non-reusable.
2007-02-14 17:01:25 +00:00
jp.liu
6021b92465 gst/rtsp/sdpmessage.*: Fix memory management of SDP messages. Fixes #407793.
Original commit message from CVS:
* gst/rtsp/sdpmessage.c: (sdp_origin_init), (sdp_connection_init),
(sdp_bandwidth_init), (sdp_time_init), (sdp_zone_init),
(sdp_key_init), (sdp_attribute_init), (sdp_message_init),
(sdp_message_uninit), (sdp_message_free), (sdp_media_init),
(sdp_media_uninit), (sdp_media_free), (sdp_message_add_media),
(sdp_parse_line):
* gst/rtsp/sdpmessage.h:
Based on patch by: jp.liu <jp_liu at astrocom dot cn>
Fix memory management of SDP messages. Fixes #407793.
2007-02-14 15:24:50 +00:00
zhangfei gao
d08a7da76b gst/avi/gstavimux.c: Allow muxing video/x-h264 (was already in the caps). Fixes #407780.
Original commit message from CVS:
Patch by: zhangfei gao <gaozhangfei@yahoo.com.cn>
* gst/avi/gstavimux.c: (gst_avi_mux_vidsink_set_caps):
Allow muxing video/x-h264 (was already in the caps). Fixes #407780.
2007-02-14 12:07:01 +00:00
jp.liu
a8f72c67d1 gst/rtsp/rtspurl.c: Fix parsing of password field in url. Fixes #407797.
Original commit message from CVS:
Patch by: jp.liu <jp_liu at astrocom dot cn>
* gst/rtsp/rtspurl.c: (rtsp_url_parse):
Fix parsing of password field in url. Fixes #407797.
2007-02-14 10:09:12 +00:00
Wim Taymans
2644d7178b gst/wavparse/gstwavparse.*: Update docs.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_class_init),
(gst_wavparse_reset), (gst_wavparse_init),
(gst_wavparse_destroy_sourcepad), (gst_wavparse_fmt),
(gst_wavparse_parse_file_header), (gst_wavparse_stream_init),
(gst_wavparse_perform_seek), (gst_wavparse_peek_chunk_info),
(gst_wavparse_stream_headers), (gst_wavparse_parse_stream_init),
(gst_wavparse_add_src_pad), (gst_wavparse_stream_data),
(gst_wavparse_loop), (gst_wavparse_chain),
(gst_wavparse_pad_convert), (gst_wavparse_pad_query),
(gst_wavparse_srcpad_event), (gst_wavparse_change_state),
(plugin_init):
* gst/wavparse/gstwavparse.h:
Update docs.
Use boilerplate.
Various code cleanups.
When the bitrate is not known (bps == 0 or compressed formats) let
downstream element guestimate the duration and position and don't
generate timestamps or durations. Fixes #405213.
Fix EOS and ERROR conditions in chain mode, we just need to forward the
error flowreturn upstream.
2007-02-14 09:55:47 +00:00
Jan Schmidt
b1aa8fef18 Re-factor the gconfaudiosink into a "GstSwitchSink" base class and a child that implements the GConf key monitoring. ...
Original commit message from CVS:
* ext/gconf/Makefile.am:
* ext/gconf/gconf.c: (gst_gconf_get_string),
(gst_gconf_get_key_for_sink_profile), (gst_gconf_set_string),
(gst_gconf_render_bin_with_default):
* ext/gconf/gconf.h:
* ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_base_init),
(gst_gconf_audio_sink_reset), (gst_gconf_audio_sink_init),
(gst_gconf_audio_sink_dispose), (do_change_child),
(gst_gconf_switch_profile), (gst_gconf_audio_sink_set_property),
(cb_change_child), (gst_gconf_audio_sink_change_state):
* ext/gconf/gstgconfaudiosink.h:
* ext/gconf/gstswitchsink.c: (gst_switch_sink_base_init),
(gst_switch_sink_class_init), (gst_switch_sink_reset),
(gst_switch_sink_init), (gst_switch_sink_dispose),
(gst_switch_commit_new_kid), (gst_switch_sink_set_child),
(gst_switch_sink_set_property), (gst_switch_sink_handle_event),
(gst_switch_sink_get_property), (gst_switch_sink_change_state):
* ext/gconf/gstswitchsink.h:
* gst/autodetect/gstautoaudiosink.c:
(gst_auto_audio_sink_class_init), (gst_auto_audio_sink_dispose),
(gst_auto_audio_sink_clear_kid), (gst_auto_audio_sink_reset),
(gst_auto_audio_sink_detect):
* gst/autodetect/gstautovideosink.c:
(gst_auto_video_sink_class_init), (gst_auto_video_sink_dispose),
(gst_auto_video_sink_clear_kid), (gst_auto_video_sink_reset),
(gst_auto_video_sink_detect):
Re-factor the gconfaudiosink into a "GstSwitchSink" base class
and a child that implements the GConf key monitoring. The end goal of
this is an audio sink that can be changed on the fly, but at the
moment it still only changes on the next READY transition.
2007-02-13 16:01:29 +00:00
Stefan Kost
5116ff603e gst/avi/gstavidemux.c: Put debug stuff into #ifndef GST_DISABLE_DEBUG #endif
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
(gst_avi_demux_parse_index), (gst_avi_demux_stream_index),
(gst_avi_demux_sync), (gst_avi_demux_massage_index),
(gst_avi_demux_calculate_durations_from_index),
(gst_avi_demux_push_event), (gst_avi_demux_stream_header_pull),
(gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data),
(gst_avi_demux_loop):
Put debug stuff into #ifndef GST_DISABLE_DEBUG #endif
2007-02-13 11:57:18 +00:00
Stefan Kost
15075edea2 Add crossreferences to glib/gobject/gstream docs.
Original commit message from CVS:
* configure.ac:
* docs/plugins/Makefile.am:
Add crossreferences to glib/gobject/gstream docs.
2007-02-13 09:46:26 +00:00
Tim-Philipp Müller
ecc16f3e31 gst/monoscope/: Fix copy'n'paste-o in docs chunk. Also add some missing CFLAGS (but no LIBS, since we only use define...
Original commit message from CVS:
* gst/monoscope/Makefile.am:
* gst/monoscope/gstmonoscope.c:
Fix copy'n'paste-o in docs chunk. Also add some missing CFLAGS
(but no LIBS, since we only use defines from the headers).
2007-02-12 23:35:16 +00:00
Jonathan Matthew
9c49fa7113 gst/wavparse/gstwavparse.c: Fix massive memory leak when operating in streaming mode due to
Original commit message from CVS:
Based on patch by: Jonathan Matthew  <jonathan at kaolin wh9 net>
* gst/wavparse/gstwavparse.c: (gst_wavparse_parse_stream_init),
(gst_wavparse_stream_data):
Fix massive memory leak when operating in streaming mode due to
GST_BUFFER_MALLOCDATA() not being set on newly-created buffers.
Fixes #407057.
2007-02-12 23:27:31 +00:00
Stefan Kost
114afecd8d gst/avi/gstavidemux.*: Save some memory (8%) by repacking the index entry structure (more to come). Add more FIXMEs t...
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_class_init),
(gst_avi_demux_reset), (gst_avi_demux_index_entry_for_time),
(gst_avi_demux_handle_src_query), (gst_avi_demux_parse_superindex),
(gst_avi_demux_parse_subindex), (gst_avi_demux_parse_stream),
(gst_avi_demux_parse_index), (gst_avi_demux_stream_index),
(gst_avi_demux_sync), (gst_avi_demux_next_data_buffer),
(gst_avi_demux_stream_scan), (gst_avi_demux_massage_index),
(gst_avi_demux_calculate_durations_from_index),
(gst_avi_demux_push_event), (gst_avi_demux_stream_header_pull),
(gst_avi_demux_do_seek), (gst_avi_demux_process_next_entry),
(gst_avi_demux_stream_data), (gst_avi_demux_loop):
* gst/avi/gstavidemux.h:
Save some memory (8%) by repacking the index entry structure (more to
come). Add more FIXMEs to questionable parts.
2007-02-12 15:29:44 +00:00
Stefan Kost
77790aa2dc sys/v4l2/: More FIXME comments and messaging changes.
Original commit message from CVS:
* sys/v4l2/gstv4l2src.c: (gst_v4l2src_v4l2fourcc_to_caps),
(gst_v4l2src_get_caps):
* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list),
(gst_v4l2src_grab_frame), (gst_v4l2src_set_capture),
(gst_v4l2src_capture_init):
More FIXME comments and messaging changes.
2007-02-12 12:57:22 +00:00
Stefan Kost
14d79a36f3 gst/goom/gstgoom.*: Improved docs and use GST_DEBUG_FUNCPTR.
Original commit message from CVS:
* gst/goom/gstgoom.c: (gst_goom_class_init), (gst_goom_init),
(gst_goom_change_state):
* gst/goom/gstgoom.h:
Improved docs and use GST_DEBUG_FUNCPTR.
* gst/level/gstlevel.c: (gst_level_class_init):
Use GST_DEBUG_FUNCPTR.
* gst/monoscope/gstmonoscope.c: (gst_monoscope_init),
(gst_monoscope_chain), (gst_monoscope_change_state):
Improved docs source cleanups.
2007-02-12 12:43:00 +00:00
Tim-Philipp Müller
84c6815cf7 gst/debug/: Add code for a pushfilesrc element that implements a pushfile:// URI handler, to make debugging push-mode...
Original commit message from CVS:
* gst/debug/Makefile.am:
* gst/debug/gstdebug.c: (plugin_init):
* gst/debug/gstpushfilesrc.c:
* gst/debug/gstpushfilesrc.h:
Add code for a pushfilesrc element that implements a pushfile:// URI
handler, to make debugging push-mode operation of demuxer/decoders
that support both easier in connection with seek/playbin/etc.
The element isn't registered at the moment.
2007-02-12 10:29:57 +00:00
Sébastien Moutte
9c8ea35617 gst/avi/gstavimux.c: Comment a #if 0 in caps template definition as VS6 seems to do not support it.
Original commit message from CVS:
* gst/avi/gstavimux.c:
Comment a #if 0 in caps template definition as VS6 seems to
do not support it.
* gst/rtsp/gstrtspsrc.c:(gst_rtspsrc_loop_udp):
Use gst_guint64_to_gdouble for conversion.
* gst/rtsp/rtspconnection.c:(rtsp_connection_send):
Move variables declaration before the first instruction.
* gst/rtsp/rtspdefs.c:(rtsp_strresult):
Don't use hstrerror for error log on G_OS_WIN32 build as it's not supported.
And don't include netdb.h for G_OS_WIN32
* gst/rtsp/sdpmessage.c:(sdp_parse_line):
This initialization SDPMedia nmedia = {.media = NULL }; is not supported
by VS6 then use an other way to initialize SDPMedia structure.
* gst/udp/gstdynudpsink.h:
* gst/udp/gstdynudpnetutils.h:
Do not include <sys/time.h> for G_OS_WIN32
* gst/udp/gstudpsrc.c:
Define socklen_t as int for G_OS_WIN32
* win/common/config.h.in:
Undef HAVE_NETINET_IN_H
* win32/vs6/gst_plugins_good.dsw:
* win32/vs6/libgstrtp.dsp:
* win32/vs6/libgstrtsp.dsp:
* win32/vs6/libgstautogen.dsp:
* win32/vs6/libgstaudiofx.dsp:
* win32/vs6/libgstudp.dsp:
Add and update project files.
* win32/common/gstudp-enumtypes.c:
* win32/common/gstudp-enumtypes.h:
Add a copy of udp enumtypes to win32/common as in core
and base.
2007-02-11 12:57:47 +00:00
Stefan Kost
e687b50f62 configure.ac: Activate monoscope when building with --enable-experimental. Fix
Original commit message from CVS:
* configure.ac:
Activate monoscope when building with --enable-experimental. Fix
--enable-external configure switch description.
* sys/sunaudio/gstsunaudiomixer.c: (gst_sunaudiomixer_base_init):
* sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_dispose):
Help gst-indent.
2007-02-11 10:53:21 +00:00
Tim-Philipp Müller
d8f5483d85 gst/avi/gstavimux.c: Explicitly cast result of pointer arithmetic to integer in order to avoid compiler warnings on s...
Original commit message from CVS:
* gst/avi/gstavimux.c: (gst_avi_mux_riff_get_avi_header):
Explicitly cast result of pointer arithmetic to integer in order to
avoid compiler warnings on some 64-bit systems. Should fix #406018.
2007-02-09 09:24:58 +00:00
Tim-Philipp Müller
6bbee3202a gst/debug/progressreport.c: Some more docs.
Original commit message from CVS:
* gst/debug/progressreport.c:
Some more docs.
2007-02-08 11:09:15 +00:00
Tim-Philipp Müller
ba2af9fa12 docs/plugins/inspect/plugin-rtp.xml: Update for new elements.
Original commit message from CVS:
* docs/plugins/inspect/plugin-rtp.xml:
Update for new elements.
* gst/debug/progressreport.h:
Commit newly-created header file as well.
2007-02-07 21:09:45 +00:00
Tim-Philipp Müller
b5ee422546 Make progressreport element post messages with the current progress on the bus. Also add some basic docs for it.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* docs/plugins/gst-plugins-good-plugins.hierarchy:
* gst/debug/Makefile.am:
* gst/debug/progressreport.c: (gst_progress_report_post_progress),
(gst_progress_report_do_query), (gst_progress_report_report):
Make progressreport element post messages with the current progress
on the bus. Also add some basic docs for it.
2007-02-07 20:39:16 +00:00
Tim-Philipp Müller
784a4689e0 ext/hal/hal.*: Some small cleanups; deal with errors when parsing the HAL ALSA capabilities a bit better.
Original commit message from CVS:
* ext/hal/hal.c: (gst_hal_get_string):
* ext/hal/hal.h:
Some small cleanups; deal with errors when parsing the HAL ALSA
capabilities a bit better.
2007-02-07 13:08:34 +00:00
Tim-Philipp Müller
2a873dd98e gst/smpte/gstsmpte.c: Let's try this again and use the right cast this time.
Original commit message from CVS:
* gst/smpte/gstsmpte.c: (gst_smpte_transition_type_get_type):
Let's try this again and use the right cast this time.
2007-02-06 16:29:30 +00:00
Tim-Philipp Müller
7dd530e6c4 gst/smpte/gstsmpte.c: Add cast to avoid compiler warnings with older GLib versions where the nick/name members in GEn...
Original commit message from CVS:
* gst/smpte/gstsmpte.c: (gst_smpte_transition_type_get_type):
Add cast to avoid compiler warnings with older GLib versions
where the nick/name members in GEnumValue are not declared as
constant strings.
2007-02-06 16:24:57 +00:00
Tim-Philipp Müller
881308d5c5 ext/gconf/: In gconfaudiosink, get the right key as the old key in do_toggle (ie. one dependent on the profile select...
Original commit message from CVS:
* ext/gconf/gconf.c: (gst_gconf_get_key_for_sink_profile),
(gst_gconf_render_bin_from_key),
(gst_gconf_get_default_audio_sink):
* ext/gconf/gconf.h:
* ext/gconf/gstgconfaudiosink.c: (get_gconf_key_for_profile),
(do_toggle_element), (gst_gconf_audio_sink_set_property),
(gst_gconf_audio_sink_get_property):
In gconfaudiosink, get the right key as the old key in do_toggle
(ie. one dependent on the profile selected). Log some more stuff so
we can see what's actually going on.
2007-02-06 15:56:14 +00:00
Sebastian Dröge
cdba2c4219 gst/audiofx/: Some small cleanups and port both elements to the new GstAudioFilter base class to save a few lines of ...
Original commit message from CVS:
* gst/audiofx/audioamplify.c: (gst_audio_amplify_base_init),
(gst_audio_amplify_class_init), (gst_audio_amplify_init),
(gst_audio_amplify_set_process_function),
(gst_audio_amplify_setup):
* gst/audiofx/audioamplify.h:
* gst/audiofx/audioinvert.c: (gst_audio_invert_base_init),
(gst_audio_invert_class_init), (gst_audio_invert_setup):
* gst/audiofx/audioinvert.h:
Some small cleanups and port both elements to the new GstAudioFilter
base class to save a few lines of common code.
* gst/audiofx/Makefile.am:
Link against libgstaudio for the above changes
2007-02-06 11:16:49 +00:00
Wim Taymans
10294f3849 tests/check/elements/.cvsignore: Some more ignores.
Original commit message from CVS:
* tests/check/elements/.cvsignore:
Some more ignores.
2007-01-29 10:59:48 +00:00
charles
8d70a788fa ext/shout2/gstshout2.*: Properly handle tags in shout2send. Fixes #399825.
Original commit message from CVS:
Patch by: charles <charlesg3 at gmail dot com>
* ext/shout2/gstshout2.c: (gst_shout2send_init),
(set_shout_metadata), (gst_shout2send_event):
* ext/shout2/gstshout2.h:
Properly handle tags in shout2send. Fixes #399825.
2007-01-26 12:21:41 +00:00
Wim Taymans
2de7376aaf gst/rtsp/gstrtspsrc.c: Convert SDP fields to upper/lowercase following the rules in the SDP to caps document.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps),
(gst_rtspsrc_activate_streams):
Convert SDP fields to upper/lowercase following the rules in the SDP to
caps document.
2007-01-25 14:40:15 +00:00
Wim Taymans
22eb34e2fe gst/rtp/: Fix case of encoding-name and key/value pairs to match the document.
Original commit message from CVS:
* gst/rtp/README:
* gst/rtp/gstrtpilbcdepay.c:
* gst/rtp/gstrtpilbcpay.c:
* gst/rtp/gstrtpmp4gdepay.c:
* gst/rtp/gstrtpmp4gpay.c:
* gst/rtp/gstrtpspeexdepay.c:
* gst/rtp/gstrtpspeexpay.c:
* gst/rtp/gstrtpsv3vdepay.c:
* gst/rtp/gstrtptheoradepay.c:
* gst/rtp/gstrtptheorapay.c:
* gst/rtp/gstrtpvorbisdepay.c:
* gst/rtp/gstrtpvorbispay.c:
Fix case of encoding-name and key/value pairs to match the document.
This is to make interoperation with SDP case-insensitive as required by
the relevant RFCs.
2007-01-25 14:22:53 +00:00
Wim Taymans
a95df52763 configure.ac: Bump required -core/-base to CVS
Original commit message from CVS:
* configure.ac:
Bump required -core/-base to CVS
2007-01-25 11:02:01 +00:00
Wim Taymans
40d06b6a55 gst/rtp/gstrtpL16pay.*: Fill up to MTU using adapter.
Original commit message from CVS:
* gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_setcaps),
(gst_rtp_L16_pay_flush), (gst_rtp_L16_pay_handle_buffer):
* gst/rtp/gstrtpL16pay.h:
Fill up to MTU using adapter.
Timestamp rtp packets.
2007-01-25 10:54:19 +00:00
Edward Hervey
d7666d033c Use G_GSIZE_FORMAT in print statements for portability.
Original commit message from CVS:
* gst/multipart/multipartmux.c: (gst_multipart_mux_collected):
* sys/ximage/ximageutil.c: (ximageutil_check_xshm_calls):
Use G_GSIZE_FORMAT in print statements for portability.
Fixes build on macosx.
2007-01-25 10:36:35 +00:00
Wim Taymans
85420195b2 gst/rtp/: Port and enable raw audio payloader/depayloader. Needs a bit more work on the payloader side.
Original commit message from CVS:
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16_depay_base_init),
(gst_rtp_L16_depay_class_init), (gst_rtp_L16_depay_init),
(gst_rtp_L16_depay_parse_int), (gst_rtp_L16_depay_setcaps),
(gst_rtp_L16_depay_process), (gst_rtp_L16_depay_set_property),
(gst_rtp_L16_depay_get_property), (gst_rtp_L16_depay_change_state),
(gst_rtp_L16_depay_plugin_init):
* gst/rtp/gstrtpL16depay.h:
* gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_get_type),
(gst_rtp_L16_pay_base_init), (gst_rtp_L16_pay_class_init),
(gst_rtp_L16_pay_init), (gst_rtp_L16_pay_finalize),
(gst_rtp_L16_pay_setcaps), (gst_rtp_L16_pay_handle_buffer),
(gst_rtp_L16_pay_plugin_init):
* gst/rtp/gstrtpL16pay.h:
Port and enable raw audio payloader/depayloader. Needs a bit more work
on the payloader side.
2007-01-24 18:20:14 +00:00
Wim Taymans
a6a9207c42 gst/rtsp/gstrtspsrc.*: Only unblock the udp pads when we linked and activated them all.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (pad_blocked),
(gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_udp):
* gst/rtsp/gstrtspsrc.h:
Only unblock the udp pads when we linked and activated them all.
Fixes #395688.
2007-01-24 16:25:55 +00:00
Wim Taymans
f083178741 gst/rtp/: Added simple AC3 depayloader (RFC 4184).
Original commit message from CVS:
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtpac3depay.c: (gst_rtp_ac3_depay_base_init),
(gst_rtp_ac3_depay_class_init), (gst_rtp_ac3_depay_init),
(gst_rtp_ac3_depay_setcaps), (gst_rtp_ac3_depay_process),
(gst_rtp_ac3_depay_set_property), (gst_rtp_ac3_depay_get_property),
(gst_rtp_ac3_depay_change_state), (gst_rtp_ac3_depay_plugin_init):
* gst/rtp/gstrtpac3depay.h:
Added simple AC3 depayloader (RFC 4184).
* gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_setcaps):
Fix a leak.
2007-01-24 15:18:34 +00:00
Sebastian Dröge
54b10ebf2a gst/audiofx/: Add new element "audioamplify". This allows scaling of raw audio samples, similar to the "volume" eleme...
Original commit message from CVS:
reviewed by: Stefan Kost  <ensonic@users.sf.net>
* gst/audiofx/Makefile.am:
* gst/audiofx/audioamplify.c:
(gst_audio_amplify_clipping_method_get_type),
(gst_audio_amplify_base_init), (gst_audio_amplify_class_init),
(gst_audio_amplify_init), (gst_audio_amplify_set_process_function),
(gst_audio_amplify_set_property), (gst_audio_amplify_get_property),
(gst_audio_amplify_set_caps),
(gst_audio_amplify_transform_int_clip),
(gst_audio_amplify_transform_int_wrap_negative),
(gst_audio_amplify_transform_int_wrap_positive),
(gst_audio_amplify_transform_float_clip),
(gst_audio_amplify_transform_float_wrap_negative),
(gst_audio_amplify_transform_float_wrap_positive),
(gst_audio_amplify_transform_ip):
* gst/audiofx/audioamplify.h:
* gst/audiofx/audiofx.c: (plugin_init):
Add new element "audioamplify". This allows scaling of raw audio
samples, similar to the "volume" element, but provides different modes
for clipping and allows unlimited amplification. It's mainly targeted
for creative sound design and not as a replacement of the "volume"
element. Fixes #397162
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* docs/plugins/gst-plugins-good-plugins.args:
* docs/plugins/inspect/plugin-audiofx.xml:
Add docs for audioamplify and integrate them into the build system
* tests/check/Makefile.am:
* tests/check/elements/audioamplify.c: (setup_amplify),
(cleanup_amplify), (GST_START_TEST), (amplify_suite), (main):
Add fairly extensive unit test suite for audioamplify
2007-01-24 12:41:03 +00:00
Wim Taymans
1f51fd9785 gst/rtsp/gstrtspsrc.c: Unblock pads after adding the pads to the element so that autopluggers get a change to link so...
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (pad_unblocked), (pad_blocked):
Unblock pads after adding the pads to the element so that autopluggers
get a change to link something. Possibly fixes #395688.
2007-01-24 12:26:41 +00:00
Wim Taymans
3df533de2c gst/rtp/: Fix caps with payload numbers.
Original commit message from CVS:
* gst/rtp/gstrtpamrdepay.c:
* gst/rtp/gstrtpgsmdepay.c:
* gst/rtp/gstrtph263pdepay.c:
* gst/rtp/gstrtph263ppay.c:
* gst/rtp/gstrtph264depay.c:
* gst/rtp/gstrtpilbcdepay.c:
* gst/rtp/gstrtpmp2tdepay.c:
* gst/rtp/gstrtpmp4gdepay.c:
* gst/rtp/gstrtpmp4gpay.c:
* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_class_init):
* gst/rtp/gstrtpmp4vpay.c:
* gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_class_init),
(gst_rtp_mpa_depay_init), (gst_rtp_mpa_depay_setcaps),
(gst_rtp_mpa_depay_process):
* gst/rtp/gstrtpmpvdepay.c: (gst_rtp_mpv_depay_class_init),
(gst_rtp_mpv_depay_init), (gst_rtp_mpv_depay_process):
* gst/rtp/gstrtppcmadepay.c:
* gst/rtp/gstrtppcmudepay.c:
* gst/rtp/gstrtpspeexdepay.c:
* gst/rtp/gstrtpspeexpay.c:
* gst/rtp/gstrtpsv3vdepay.c:
* gst/rtp/gstrtptheoradepay.c:
* gst/rtp/gstrtptheorapay.c:
* gst/rtp/gstrtpvorbisdepay.c:
* gst/rtp/gstrtpvorbispay.c:
Fix caps with payload numbers.
Add some fixed payload numbers to caps when possible.
2007-01-24 12:22:51 +00:00
Sebastian Dröge
447ae144c2 gst/audiofx/: Add new audiofx element "audioinvert". This element swaps the upper and lower half of samples and can b...
Original commit message from CVS:
reviewed by: Stefan Kost  <ensonic@users.sf.net>
* gst/audiofx/Makefile.am:
* gst/audiofx/audiofx.c: (plugin_init):
* gst/audiofx/audioinvert.c: (gst_audio_invert_base_init),
(gst_audio_invert_class_init), (gst_audio_invert_init),
(gst_audio_invert_set_property), (gst_audio_invert_get_property),
(gst_audio_invert_set_caps), (gst_audio_invert_transform_int),
(gst_audio_invert_transform_float),
(gst_audio_invert_transform_ip):
* gst/audiofx/audioinvert.h:
Add new audiofx element "audioinvert". This element swaps the upper
and lower half of samples and can be used for example for a
wide-stereo effect. Fixes #396057
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* docs/plugins/gst-plugins-good-plugins.args:
* docs/plugins/inspect/plugin-audiofx.xml:
Add docs for the audioinvert element and add them to the build system.
* tests/check/Makefile.am:
* tests/check/elements/audioinvert.c: (setup_invert),
(cleanup_invert), (GST_START_TEST), (invert_suite), (main):
Add unit test suite for the audioinvert element.
2007-01-23 18:16:09 +00:00