Commit graph

2903 commits

Author SHA1 Message Date
Wim Taymans
9e37243eca gst/rtsp/gstrtspsrc.c: Let more error state trickle down so that we can catch more error cases.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send_keep_alive),
(gst_rtspsrc_loop_udp), (gst_rtspsrc_try_send), (gst_rtspsrc_send),
(gst_rtspsrc_setup_streams), (gst_rtspsrc_open),
(gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause),
(gst_rtspsrc_change_state):
Let more error state trickle down so that we can catch more error
cases.
Handle keep-alive a little smarter by selecting a method the server
actually supports.
Fix a race in UDP streaming shutdown.
2007-05-04 15:17:14 +00:00
Wim Taymans
5f2fbbd76b gst/rtsp/gstrtspsrc.c: Ignore errors when trying to use the keep-alive messages.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send_keep_alive):
Ignore errors when trying to use the keep-alive messages.
2007-05-04 13:04:31 +00:00
Wim Taymans
fb80e57990 gst/rtsp/gstrtspsrc.c: Send RTCP messages back to the server over the TCP connection.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_handle_src_event),
(gst_rtspsrc_handle_src_query), (gst_rtspsrc_sink_chain),
(gst_rtspsrc_stream_configure_manager),
(gst_rtspsrc_stream_free_udp), (gst_rtspsrc_stream_configure_tcp),
(gst_rtspsrc_stream_configure_mcast),
(gst_rtspsrc_stream_configure_udp),
(gst_rtspsrc_stream_configure_udp_sink),
(gst_rtspsrc_stream_configure_transport):
Send RTCP messages back to the server over the TCP connection.
* gst/rtsp/rtspconnection.c: (rtsp_connection_write),
(rtsp_connection_send), (rtsp_connection_read), (read_body),
(rtsp_connection_receive):
* gst/rtsp/rtspconnection.h:
Factor out and expose lowlevel _write and _read methods.
Implement sending data messages to the server.
2007-05-04 12:31:32 +00:00
Wim Taymans
4d42c097a6 gst/multipart/multipartmux.c: Fix timestamps on outgoing buffers.
Original commit message from CVS:
* gst/multipart/multipartmux.c: (gst_multipart_mux_queue_pads),
(gst_multipart_mux_collected):
Fix timestamps on outgoing buffers.
2007-05-03 15:55:06 +00:00
Wim Taymans
5ba2fa6e3f gst/multipart/multipartmux.c: Emit NEWSEGMENT events before pushing the first buffer.
Original commit message from CVS:
* gst/multipart/multipartmux.c:
(gst_multipart_mux_request_new_pad), (gst_multipart_mux_collected),
(gst_multipart_mux_change_state):
Emit NEWSEGMENT events before pushing the first buffer.
2007-05-03 14:39:09 +00:00
Wim Taymans
17011e9a41 gst/rtsp/gstrtspsrc.c: Refactor transport configuration code.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_finalize),
(gst_rtspsrc_alloc_udp_ports), (gst_rtspsrc_handle_src_event),
(gst_rtspsrc_handle_src_query),
(gst_rtspsrc_stream_configure_manager),
(gst_rtspsrc_stream_free_udp), (gst_rtspsrc_stream_configure_tcp),
(gst_rtspsrc_stream_configure_mcast),
(gst_rtspsrc_stream_configure_udp),
(gst_rtspsrc_stream_configure_udp_sink),
(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_push_event),
(gst_rtspsrc_loop_udp), (gst_rtspsrc_open),
(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play),
(gst_rtspsrc_pause):
Refactor transport configuration code.
Create internal pads for TCP transport so that we can implement events
and queries.
Handle events and queries.
Parse range from the SDP.
Fix race in pause handler where the connection could still be flushing.
2007-05-03 13:48:54 +00:00
Wim Taymans
24e51b3c73 gst/rtsp/gstrtspsrc.*: Fix race when multiple udp sources post timeouts, just act on the first received timeout.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_finalize), (new_session_pad), (request_pt_map),
(gst_rtspsrc_loop_send_cmd), (gst_rtspsrc_try_send),
(gst_rtspsrc_send), (gst_rtspsrc_async_open), (gst_rtspsrc_close),
(gst_rtspsrc_play), (gst_rtspsrc_handle_message),
(gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Fix race when multiple udp sources post timeouts, just act on the first
received timeout.
Protect stream list with a recursive lock to fix some races.
Flush connection when we need to do a reconnect or stop.
Make state lock recursive.
* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
(rtsp_connection_close):
Some small cleanups.
2007-05-02 19:32:58 +00:00
Wim Taymans
6991907036 gst/wavparse/gstwavparse.c: Only set DISCONT when there actually is a discont or when we just started.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek),
(gst_wavparse_stream_headers), (gst_wavparse_stream_data):
Only set DISCONT when there actually is a discont or when we just
started.
2007-05-02 18:25:09 +00:00
Sebastian Dröge
09b83eac48 ext/flac/gstflac.c: Call bindtextdomain() to get localized strings.
Original commit message from CVS:
* ext/flac/gstflac.c: (plugin_init):
Call bindtextdomain() to get localized strings.
2007-05-02 18:01:52 +00:00
Wim Taymans
64e0ee90f6 gst/wavparse/gstwavparse.*: Be a bit more clever when dealing with VBR files with FACT tags, we don't want to timesta...
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek),
(gst_wavparse_stream_headers), (gst_wavparse_add_src_pad),
(gst_wavparse_stream_data):
* gst/wavparse/gstwavparse.h:
Be a bit more clever when dealing with VBR files with FACT tags, we
don't want to timestamp buffers in that case but the estimated BPS can
be used for seeking.
Only send close segment in the streaming thread.
2007-05-02 17:19:36 +00:00
Sebastian Dröge
b64fd034a5 ext/flac/gstflacdec.c: Correctly post an error on the bus if something went wrong in the loop function. This fixes a ...
Original commit message from CVS:
* ext/flac/gstflacdec.c: (gst_flac_dec_loop):
Correctly post an error on the bus if something went wrong in the loop
function. This fixes a few cases where the task was paused and nothing
happened anymore.
2007-05-02 17:08:09 +00:00
Wim Taymans
8281f6c054 gst/rtsp/test.c: Fix compilation of deprecated test just because I'm too lazy to delete it.
Original commit message from CVS:
* gst/rtsp/test.c: (main):
Fix compilation of deprecated test just because I'm too lazy to delete
it.
2007-05-02 14:27:28 +00:00
Wim Taymans
92396be152 gst/rtsp/gstrtspsrc.*: Fix sending RTCP to the right place.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_finalize), (gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_handle_request), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_send_keep_alive), (gst_rtspsrc_loop_udp),
(gst_rtspsrc_loop_send_cmd), (gst_rtspsrc_try_send),
(gst_rtspsrc_open), (gst_rtspsrc_handle_message):
* gst/rtsp/gstrtspsrc.h:
Fix sending RTCP to the right place.
Fix bug in reffing the wrong UDP element.
Use new pad names for the session manager.
Implement handling server requests in interleaved and UDP modes.
Handle session keep-alive in UDP modes.
Remove GCond for handling UDP timeouts.
* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
(rtsp_connection_send), (rtsp_connection_read), (read_body),
(rtsp_connection_receive), (rtsp_connection_close):
* gst/rtsp/rtspconnection.h:
Store connection IP address for later.
Add timeout args to all operations that might block forever.
Parse session timeout.
Only close sockets when not already closed.
* gst/rtsp/rtspdefs.c:
* gst/rtsp/rtspdefs.h:
Add timeout return value and error string.
* gst/rtsp/rtspmessage.c: (rtsp_message_init_response):
Add small comment.
2007-05-02 13:32:57 +00:00
Sjoerd Simons
f34fce9df4 gst/rtp/gstrtpmp4vpay.*: Handle NEWSEGMENT and FLUSH events. Fixes #434824.
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_init),
(gst_rtp_mp4v_pay_empty), (gst_rtp_mp4v_pay_event):
* gst/rtp/gstrtpmp4vpay.h:
Handle NEWSEGMENT and FLUSH events. Fixes #434824.
2007-05-01 16:13:58 +00:00
Tim-Philipp Müller
baa94a9b42 docs/plugins/gst-plugins-good-plugins-docs.sgml: Remove v4l2src from docs, since it breaks the docs build, and the pl...
Original commit message from CVS:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
Remove v4l2src from docs, since it breaks the docs build, and the
plugin is only built if --enable-experimental is used anyway.
* docs/plugins/Makefile.am:
Spaces => tab.
2007-04-30 11:15:58 +00:00
Wim Taymans
066598d8de gst/udp/gstmultiudpsink.c: Add code to drop membership of a multicast group.
Original commit message from CVS:
* gst/udp/gstmultiudpsink.c: (leave_multicast),
(gst_multiudpsink_add), (gst_multiudpsink_remove):
Add code to drop membership of a multicast group.
* gst/udp/gstudpsink.c: (gst_udpsink_update_uri),
(gst_udpsink_set_uri):
Implement URI handler.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_parse_rtpinfo):
Use URI handler to make udpsink instace.
Improve code to configure port and destination.
2007-04-29 14:43:37 +00:00
Wim Taymans
589b8282e8 gst/udp/gstmultiudpsink.c: Fix multicast detection.
Original commit message from CVS:
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_add):
Fix multicast detection.
Don't try to join a multicast group if the address is not multicast.
* gst/udp/gstudpsrc.c: (gst_udpsrc_update_uri):
Small debug improvement.
2007-04-29 12:19:21 +00:00
Wim Taymans
6a790cb75a gst/rtsp/gstrtspsrc.c: Ignore ASYNC state messages from the udpsink, it's irrelevant for the parent.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play),
(gst_rtspsrc_handle_message):
Ignore ASYNC state messages from the udpsink, it's irrelevant for the
parent.
2007-04-27 16:44:17 +00:00
Wim Taymans
7fe2138eea gst/rtp/gstrtpilbcdepay.h: Fix mode property when specified as an arg.
Original commit message from CVS:
* gst/rtp/gstrtpilbcdepay.h:
Fix mode property when specified as an arg.
2007-04-27 15:30:39 +00:00
Edward Hervey
a9a843b340 docs/plugins/: Add documentation for osxaudio plugin.
Original commit message from CVS:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* docs/plugins/gst-plugins-good-plugins.hierarchy:
* docs/plugins/inspect/plugin-osxaudio.xml:
Add documentation for osxaudio plugin.
2007-04-26 15:08:20 +00:00
Wim Taymans
530f214bd5 gst/rtsp/gstrtspsrc.*: Protect state changes with a lock.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_finalize), (gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_open), (gst_rtspsrc_close),
(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play),
(gst_rtspsrc_pause):
* gst/rtsp/gstrtspsrc.h:
Protect state changes with a lock.
* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
(parse_line):
* gst/rtsp/rtspconnection.h:
Remove some unused stuff.
2007-04-26 10:08:27 +00:00
Wim Taymans
45b77c57b4 gst/udp/gstudpsrc.c: Handle the case where there are exactly 0 bytes to read and the ioctl did not report an error. F...
Original commit message from CVS:
* gst/udp/gstudpsrc.c: (gst_udpsrc_create):
Handle the case where there are exactly 0 bytes to read and the ioctl
did not report an error. Fixes #433530.
2007-04-26 08:48:30 +00:00
Wim Taymans
88bf47c911 gst/wavparse/gstwavparse.*: Apply DISCONT to buffers.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek),
(gst_wavparse_stream_headers), (gst_wavparse_stream_data):
* gst/wavparse/gstwavparse.h:
Apply DISCONT to buffers.
Only apply timestamp to the first sample after a DISCONT, too many VBR
files cause random jitter in the timestamps. Fixes #433119.
2007-04-26 08:39:49 +00:00
Wim Taymans
6937be1a09 gst/rtsp/gstrtpdec.*: Add dummy latency property to be backwards compat with rtpbin.
Original commit message from CVS:
* gst/rtsp/gstrtpdec.c: (gst_rtp_dec_class_init),
(gst_rtp_dec_init), (gst_rtp_dec_set_property),
(gst_rtp_dec_get_property):
* gst/rtsp/gstrtpdec.h:
Add dummy latency property to be backwards compat with rtpbin.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
(gst_rtspsrc_set_property), (gst_rtspsrc_get_property),
(gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_parse_rtpinfo):
* gst/rtsp/gstrtspsrc.h:
Add latency property and configure in the session manager.
Don't set invalid clock-base and seqnum-base on caps, some servers
sometimes don't send them.
2007-04-25 15:55:32 +00:00
Tim-Philipp Müller
e53a24511b gst/alpha/gstalphacolor.c: Double-check that RGB input caps are really RGBA caps (apparently the core doesn't always ...
Original commit message from CVS:
* gst/alpha/gstalphacolor.c: (gst_alpha_color_base_init),
(gst_alpha_color_transform_caps), (gst_alpha_color_set_caps):
Double-check that RGB input caps are really RGBA caps (apparently
the core doesn't always catch it if those caps aren't a subset of
our template caps, also see #421543). Fixes #429319 in a way.
Also, don't leak the pad template in the transform_caps function.
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/alphacolor.c: (setup_alphacolor),
(cleanup_alphacolor), (create_caps_rgb24), (create_caps_rgba32),
(create_buffer_rgb24_3x4), (create_buffer_rgba32_3x4),
(GST_START_TEST), (alphacolor_suite):
Add some basic unit tests for alphacolor.
2007-04-25 15:31:53 +00:00
Tim-Philipp Müller
3f55b6e912 ext/libpng/gstpngdec.c: If we get a fatal flow return in the loop function, first post the error message and only the...
Original commit message from CVS:
* ext/libpng/gstpngdec.c: (gst_pngdec_task):
If we get a fatal flow return in the loop function, first post the
error message and only then send the EOS event downstream, otherwise
applications might get an eos message before the error message and
think everything was ok (related to #429319).
2007-04-25 15:08:22 +00:00
Wim Taymans
a7531984c3 gst/rtsp/rtspconnection.c: Read the channel byte as an unsigned byte.
Original commit message from CVS:
* gst/rtsp/rtspconnection.c: (rtsp_connection_receive):
Read the channel byte as an unsigned byte.
2007-04-25 10:07:12 +00:00
Wim Taymans
24c5812d65 gst/rtp/: Make sure we configure the clock_rate in the baseclass in the setcaps function. Fixes #431282.
Original commit message from CVS:
* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16_depay_set_property):
* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_init),
(gst_rtp_amr_depay_setcaps), (gst_rtp_amr_depay_process):
* gst/rtp/gstrtpgsmdepay.c: (gst_rtp_gsm_depay_init),
(gst_rtp_gsm_depay_setcaps):
* gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_setcaps):
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_setcaps):
* gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_class_init),
(gst_rtp_ilbc_depay_init), (gst_rtp_ilbc_depay_setcaps),
(gst_rtp_ilbc_depay_process), (gst_ilbc_depay_set_property),
(gst_ilbc_depay_get_property):
* gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_setcaps):
* gst/rtp/gstrtpmp4adepay.c:
* gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_init),
(gst_rtp_pcma_depay_setcaps):
* gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_init),
(gst_rtp_pcmu_depay_setcaps):
Make sure we configure the clock_rate in the baseclass in the setcaps
function. Fixes #431282.
2007-04-25 09:47:48 +00:00
Wim Taymans
1beeda3ff2 gst/rtsp/gstrtspsrc.*: Parse server address from SDP.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_finalize),
(gst_rtspsrc_stream_free), (request_pt_map),
(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_open):
* gst/rtsp/gstrtspsrc.h:
Parse server address from SDP.
Hook up a udpsink to send RTCP back to the server.
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* gst/rtsp/rtsptransport.h:
Add some docs.
2007-04-25 08:36:46 +00:00
Stefan Kost
fa7454bda2 gst/wavparse/gstwavparse.c: Make header field check conditional. Fixes #433135
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers):
Make header field check conditional. Fixes #433135
2007-04-25 06:52:09 +00:00
Tim-Philipp Müller
7002f0336b Add minimal docs blurb to alphacolor; split out headers into separate header file for gtk-doc.
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/inspect/plugin-alphacolor.xml:
* gst/alpha/Makefile.am:
* gst/alpha/gstalphacolor.c:
* gst/alpha/gstalphacolor.h:
Add minimal docs blurb to alphacolor; split out headers into
separate header file for gtk-doc.
2007-04-24 09:12:42 +00:00
Tim-Philipp Müller
106db1b2eb gst/debug/progressreport.c: Don't try to post NULL message (in case we can't query upstream position or duration).
Original commit message from CVS:
* gst/debug/progressreport.c: (gst_progress_report_report):
Don't try to post NULL message (in case we can't query upstream
position or duration).
2007-04-20 17:25:50 +00:00
Michael Smith
4a1ceda8df gst/cutter/gstcutter.*: Fix some of the most obvious bugs in cutter. Now doesn't leak everything if input is silent.
Original commit message from CVS:
* gst/cutter/gstcutter.c: (gst_cutter_init), (gst_cutter_chain),
(gst_cutter_get_caps):
* gst/cutter/gstcutter.h:
Fix some of the most obvious bugs in cutter. Now doesn't leak
everything if input is silent.
2007-04-18 12:36:37 +00:00
Sebastian Dröge
1723d916dd gst/wavenc/gstwavenc.*: everything else results in a invalid block align and invalid files.
Original commit message from CVS:
* gst/wavenc/gstwavenc.c: (gst_wavenc_create_header_buf),
(gst_wavenc_sink_setcaps), (gst_wavenc_change_state):
* gst/wavenc/gstwavenc.h:
Wav apparently only supports width==GST_ROUND_UP(depth), everything
else results in a invalid block align and invalid files.
2007-04-18 09:48:25 +00:00
Snaik
b5cfe36ab7 gst/smpte/barboxwipes.c: Add missing break statement for BOX_HORIZONTAL case.
Original commit message from CVS:
Patch by: Snaik <snaik32 gmail com>
* gst/smpte/barboxwipes.c: (gst_wipe_boxes_draw):
Add missing break statement for BOX_HORIZONTAL case.
2007-04-17 16:39:02 +00:00
Vincent Torri
188cc7a9e0 gst/wavparse/gstwavparse.c: Use correct format strings for integer types.
Original commit message from CVS:
Patch by: Vincent Torri <vtorri at univ-evry dot fr>
* gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers):
Use correct format strings for integer types.
2007-04-17 10:14:43 +00:00
Sebastian Dröge
c383f21c10 gst/wavparse/gstwavparse.c: Use gst_riff_create_audio_template_caps () instead of the local caps.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_base_init),
(gst_wavparse_create_sourcepad):
Use gst_riff_create_audio_template_caps () instead of the local caps.
This makes updates of the local caps unecessary whenever libgstriff
gets support for new formats.
2007-04-17 02:51:02 +00:00
Brian Cameron
f520911b6f sys/sunaudio/: Fix and/or update copyright attributions (#430228).
Original commit message from CVS:
Patch by: Brian Cameron  <brian.cameron at sun dot com>
* sys/sunaudio/gstsunaudio.c:
* sys/sunaudio/gstsunaudiomixer.c:
* sys/sunaudio/gstsunaudiomixer.h:
* sys/sunaudio/gstsunaudiomixerctrl.c:
* sys/sunaudio/gstsunaudiomixerctrl.h:
* sys/sunaudio/gstsunaudiomixertrack.h:
* sys/sunaudio/gstsunaudiosink.c:
* sys/sunaudio/gstsunaudiosink.h:
* sys/sunaudio/gstsunaudiosrc.c:
* sys/sunaudio/gstsunaudiosrc.h:
Fix and/or update copyright attributions (#430228).
2007-04-16 21:29:40 +00:00
Wim Taymans
b752470823 docs/plugins/gst-plugins-good-plugins-sections.txt: Fix docs.
Original commit message from CVS:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
Fix docs.
* gst/rtsp/URLS:
Add some more example urls.
* gst/rtsp/gstrtpdec.c: (gst_rtp_dec_marshal_BOXED__UINT_UINT),
(gst_rtp_dec_chain_rtp):
Better debugging.
* gst/rtsp/gstrtspsrc.c: (request_pt_map),
(gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_parse_rtpinfo):
Remove unused code.
2007-04-13 09:32:21 +00:00
Stefan Kost
3bf1b5ecf7 gst/wavparse/gstwavparse.c: Relax the audio/mpeg caps again and add FIXME: comment.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (uint64_ceiling_scale_int),
(gst_wavparse_perform_seek), (gst_wavparse_stream_headers),
(gst_wavparse_stream_data):
Relax the audio/mpeg caps again and add FIXME: comment.
2007-04-13 08:19:35 +00:00
Stefan Kost
0722106b57 gst/wavparse/gstwavparse.*: More sanity check for the header fields. Fix type for 'rate' header field.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (uint64_ceiling_scale_int),
(gst_wavparse_perform_seek), (gst_wavparse_stream_headers),
(gst_wavparse_stream_data):
* gst/wavparse/gstwavparse.h:
More sanity check for the header fields. Fix type for 'rate' header
field.
2007-04-13 06:20:28 +00:00
Tim-Philipp Müller
ef7c18817f gst/icydemux/gsticydemux.c: If the metadata strings we get in the stream are not UTF-8, try to interpret them accordi...
Original commit message from CVS:
* gst/icydemux/gsticydemux.c: (notgst_tag_freeform_string_to_utf8),
(gst_icydemux_unicodify):
If the metadata strings we get in the stream are not UTF-8, try to
interpret them according to the character encodings specified in the
GST_ICY_TAG_ENCODING and GST_TAG_ENCODING environment variables, and
only fall back to locale/ISO-8859-1 if those aren't set or don't
work. Should fix #428901.
2007-04-12 16:06:31 +00:00
Wim Taymans
f5e4a8b028 gst/rtp/gstrtph264depay.c: Use the proper sync word for SPS and PPS.
Original commit message from CVS:
* gst/rtp/gstrtph264depay.c:
Use the proper sync word for SPS and PPS.
2007-04-12 14:20:56 +00:00
Thomas Vander Stichele
2fc868841f gst/rtp/Makefile.am: gst/rtp/fnv1hash.c (MASK_24, FNV1_HASH_32_INIT, FNV1_HASH_32_PRIME, fnv1_hash_32_new, fnv1_hash_...
Original commit message from CVS:
* gst/rtp/Makefile.am:
* gst/rtp/fnv1hash.c (MASK_24, FNV1_HASH_32_INIT, FNV1_HASH_32_PRIME,
fnv1_hash_32_new, fnv1_hash_32_update, fnv1_hash_32_to_24):
* gst/rtp/fnv1hash.h (__GST_FNV1_HASH_H__):
Add a simple hashing implementation that we can use to generate
a 24-bit ident value based on the codebooks for vorbis and theora.
* gst/rtp/gstrtptheorapay.c (gst_rtp_theora_pay_finish_headers,
gst_rtp_theora_pay_handle_buffer):
* gst/rtp/gstrtpvorbisdepay.c
(gst_rtp_vorbis_depay_parse_configuration,
gst_rtp_vorbis_depay_switch_codebook, gst_rtp_vorbis_depay_process):
* gst/rtp/gstrtpvorbispay.c (gst_rtp_vorbis_pay_reset_packet,
gst_rtp_vorbis_pay_init_packet, gst_rtp_vorbis_pay_flush_packet,
gst_rtp_vorbis_pay_finish_headers, gst_rtp_vorbis_pay_handle_buffer):
Use the hashing function, ensuring that the same codebooks result
in the same ident and thus the same SDP description.
Various log fixes/changes.
2007-04-12 11:41:11 +00:00
jerry tan
a7efc5ceb7 sys/sunaudio/gstsunaudiosrc.c: it is the application's responsibility to make sure it open the device once.
Original commit message from CVS:
Patch by: jerry tan <jerry dot tan at sun dot com>
* sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_open):
remove the call of  ioctl (fd, AUDIO_MIXER_MULTIPLE_OPEN), it is the
application's responsibility to make sure it open the device once.
Remove a careless error if AUDIODEV is set. Fixes #392620.
2007-04-12 11:37:50 +00:00
Wim Taymans
86a4c1c6b0 gst/rtsp/gstrtpdec.*: Make backward compat with rtpbin by adding the request-pt-map signals.
Original commit message from CVS:
* gst/rtsp/gstrtpdec.c: (gst_rtp_dec_marshal_BOXED__UINT_UINT),
(gst_rtp_dec_class_init), (gst_rtp_dec_chain_rtp):
* gst/rtsp/gstrtpdec.h:
Make backward compat with rtpbin by adding the request-pt-map signals.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
(new_session_pad), (request_pt_map),
(gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_stream_configure_caps),
(gst_rtspsrc_activate_streams):
* gst/rtsp/gstrtspsrc.h:
Implement request-pt-map signals instead of setting caps on the buffers
for the session manager.
2007-04-12 08:21:28 +00:00
Wim Taymans
bd11d3c9d2 gst/udp/gstudp.c: Register GstNetBuffer in plugin_init so that the type can be used from multiple threads without races.
Original commit message from CVS:
* gst/udp/gstudp.c: (plugin_init):
Register GstNetBuffer in plugin_init so that the type can be used from
multiple threads without races.
2007-04-11 10:25:25 +00:00
Wim Taymans
acddbd83ff gst/rtp/gstrtpamrdepay.c: Fix depayloader clock_rate and some cleanups.
Original commit message from CVS:
* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_setcaps),
(gst_rtp_amr_depay_process):
Fix depayloader clock_rate and some cleanups.
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_finalize),
(gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process):
* gst/rtp/gstrtph264depay.h:
Don't push codec_data in the adapter because it might get flushed when
we get a discont.
* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_process):
Handle multiple AU per packet.
* gst/rtp/gstrtpsv3vdepay.c: (gst_rtp_sv3v_depay_process),
(gst_rtp_sv3v_depay_plugin_init):
Disable rank, this one does not work.
Remove timestamping, base class does that.
2007-04-10 17:06:05 +00:00
Stefan Kost
497d589d56 gst/auparse/gstauparse.c: limit caps to the formats we announce in the template
Original commit message from CVS:
* gst/auparse/gstauparse.c: (gst_au_parse_parse_header):
limit caps to the formats we announce in the template
* gst/wavparse/gstwavparse.c: (uint64_ceiling_scale_int),
(gst_wavparse_perform_seek), (gst_wavparse_stream_headers),
(gst_wavparse_add_src_pad), (gst_wavparse_stream_data):
fix some crashers/asserts when dealing with broken files
2007-04-10 12:01:33 +00:00
Peter Kjellerstedt
50f88db3ad gst/: Fix some compiler warnings. Fixes #428182.
Original commit message from CVS:
Patch by: Peter Kjellerstedt  <pkj at axis com>
* gst/avi/gstavidemux.c: (gst_avi_demux_massage_index):
* gst/rtp/gstrtpL16depay.c:
* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_process):
* gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_get_mode),
(gst_rtp_speex_depay_setcaps):
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps),
(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_loop_udp):
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_init_send):
Fix some compiler warnings. Fixes #428182.
2007-04-10 10:01:14 +00:00