Original commit message from CVS:
* docs/plugins/gst-plugins-bad-plugins.args:
* sys/directdraw/gstdirectdrawsink.c:
(gst_directdraw_sink_class_init):
Rename the keep-aspect-ratio property to force-aspect-ratio to make
it consistent with xvimagesink and ximagesink.
Original commit message from CVS:
* ext/dv/gstdvdemux.c: (gst_dvdemux_loop):
* ext/libpng/gstpngdec.c: (user_read_data), (gst_pngdec_task):
When operating in pull mode, error out correct on not-linked.
Original commit message from CVS:
* tests/icles/videocrop-test.c: (main):
Default to xvimagesink instead of autovideosink while
autovideosink/ghostpads/whatever don't handle the way we use it in
the way we expect it to.
Original commit message from CVS:
2007-06-06 Andy Wingo <wingo@pobox.com>
* sys/v4l2/v4l2src_calls.c (gst_v4l2src_probe_caps_for_format)
(gst_v4l2src_probe_caps_for_format_and_size): Only probe for
format and size if the ioctls are defined; should fix compilation
on Linux < 2.16.19.
Original commit message from CVS:
* gst/videobox/gstvideobox.c: (gst_video_box_ayuv_i420):
Printf fixes in debug statements; use LOG level for debug statements
that are printed for each and every frame; convert c++ comments to
C-style comments; not much point using g_try_malloc() if we then not
even check the return value.
Original commit message from CVS:
* configure.ac:
Bump requirements to released versions (core and base 0.10.13).
* gst/icydemux/gsticydemux.c: (gst_icydemux_unicodify):
Use gst_tag_utf8_from_freeform_string() from libgsttag instead of
own implementation.
Original commit message from CVS:
2007-06-05 Andy Wingo <wingo@pobox.com>
* sys/v4l2/gstv4l2src.c (gst_v4l2src_start, gst_v4l2src_stop): Add
some useless comments.
* sys/v4l2/v4l2src_calls.c (gst_v4l2src_capture_init): Don't queue
frames before calling STREAMON, that might leave them in a state
where they can't be dequeued if we go back to NULL without calling
STREAMON, according to the docs.
(gst_v4l2src_capture_start): Enqueue buffers here instead, right
before we call STREAMON.
(gst_v4l2src_capture_deinit): Remove crack to work around dequeue
failures. (For me this code hung.) The pool refcounting is still
crack; added a note to that effect.
Original commit message from CVS:
* gst/multipart/multipartmux.c: (gst_multipart_mux_class_init),
(gst_multipart_mux_get_mime), (gst_multipart_mux_collected):
Add support for mapping gst structure names to the MIME type equivalent.
Implemented for audio/x-mulaw->audio/basic. Fixes#442874.
Original commit message from CVS:
* gst/wavenc/gstwavenc.c: (gst_wavenc_create_header_buf),
(gst_wavenc_sink_setcaps), (gst_wavenc_format_samples),
(gst_wavenc_chain), (gst_wavenc_change_state):
* gst/wavenc/gstwavenc.h:
Properly write wav files with width!=depth by having the depth most
significant bytes set and all others zero. Fixes#442535.
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
(rtsp_connection_connect), (add_date_header),
(rtsp_connection_send), (parse_response_status),
(parse_request_line), (parse_line), (rtsp_connection_receive):
* gst/rtsp/rtspdefs.c: (rtsp_version_as_text):
* gst/rtsp/rtspdefs.h:
* gst/rtsp/rtspmessage.c: (key_value_foreach),
(rtsp_message_init_request), (rtsp_message_init_response),
(rtsp_message_remove_header), (rtsp_message_append_headers),
(rtsp_message_dump):
* gst/rtsp/rtspmessage.h:
Improves version checking, allowing an RTSP server to reply with "505
RTSP Version not supported.
Adds a Date header to all messages.
Replies with RTSP_EPARSE rather than RTSP_EINVALID in cases where we
want to be able to send a response even if something in the request was
invalid. EINVAL is only used when passing wrong arguments to functions.
Do not handle an invalid method in parse_request_line(). Defer this to
the caller so it can respond with "405 Method Not Allowed".
Improves parsing of the timeout parameter to the Session header,
allowing whitespace after the semicolon.
Avoids a compiler warning due to variables shadowing a function argument.
Original commit message from CVS:
Based on Patch by: Daniel Charles <dcharles at ti dot com>
* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_setcaps),
(gst_rtp_amr_depay_process):
* gst/rtp/gstrtpamrdepay.h:
* gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_base_init),
(gst_rtp_amr_pay_class_init), (gst_rtp_amr_pay_init),
(gst_rtp_amr_pay_setcaps), (gst_rtp_amr_pay_handle_buffer):
* gst/rtp/gstrtpamrpay.h:
Add support for AMR-WB.
Small cleanups such as using BOILERPLATE.
Original commit message from CVS:
* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_configure_stream):
Fix compile warning when debug is disabled as spotted bu Saur on IRC.
Original commit message from CVS:
2007-05-30 Andy Wingo <wingo@pobox.com>
* sys/v4l2/gstv4l2object.h:
* sys/v4l2/gstv4l2object.c (gst_v4l2_object_new): Revert some
unintended changes.
Original commit message from CVS:
2007-05-30 Andy Wingo <wingo@pobox.com>
* sys/v4l2/v4l2src_calls.h:
* sys/v4l2/v4l2src_calls.c (gst_v4l2src_fill_format_list): Store
the format list in the order that the driver gives it to us.
(gst_v4l2src_probe_caps_for_format_and_size)
(gst_v4l2src_probe_caps_for_format): New functions, fill GstCaps
based on the capabilities of the device.
(gst_v4l2src_grab_frame): Update for object variable renaming.
(gst_v4l2src_set_capture): Update to be strict in its parameters,
as in the set_caps below.
(gst_v4l2src_capture_init): Update for object variable renaming,
and reflow.
(gst_v4l2src_capture_start, gst_v4l2src_capture_stop)
(gst_v4l2src_capture_deinit): Update for object variable renaming.
(gst_v4l2src_update_fps, gst_v4l2src_set_fps)
(gst_v4l2src_get_fps): Remove; these functions don't have much
meaning outside of an atomic set_caps method.
(gst_v4l2src_buffer_new): Don't set buffer duration, it is not
known.
* sys/v4l2/gstv4l2tuner.c (gst_v4l2_tuner_set_channel): Remove
call to update_fps; not sure about this change.
(gst_v4l2_tuner_set_norm): Work around the fact that for the
moment we don't have an update_fps_func.
* sys/v4l2/gstv4l2src.h (struct _GstV4l2Src): Don't put v4l2
structures in the object, just store what we need. Do store the
probed caps of the device. Don't store the current frame rate.
* sys/v4l2/gstv4l2src.c (gst_v4l2src_init): Remove the
update_fps_function, for now. Update for new object variable
naming.
(gst_v4l2src_set_property, gst_v4l2src_get_property): Update for
new object variable naming.
(gst_v4l2src_v4l2fourcc_to_structure): Rename from ..._to_caps.
(gst_v4l2_structure_to_v4l2fourcc): Rename from ...caps_to_....
(gst_v4l2src_get_caps): Rework to probe the device for supported
frame sizes and frame rates.
(gst_v4l2src_set_caps): Rework to be strict in the given
parameters: if someone asks us to have a certain size and rate,
that is what we configure.
(gst_v4l2src_get_read): Update for object variable naming. Don't
leak buffers on short reads.
(gst_v4l2src_get_mmap): Update for object variable naming, and add
comments.
(gst_v4l2src_create): Update for object variable naming.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_base_init),
(gst_avi_demux_reset), (gst_avi_demux_parse_stream):
* gst/avi/gstavidemux.h:
Parse subtitle text streams instead of erroring out (#442034). Still
needs a parser for the subtitles to actually show up.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_push_event),
(gst_avi_demux_loop):
Make _push_event() return TRUE if the event could be pushed on at
least one pad and not only if it could be pushed on all pads,
otherwise we'll end up posting an error message on EOS if one or
more source pads are not connected.
Original commit message from CVS:
* ext/speex/gstspeexdec.c: (speex_dec_chain_parse_data):
Use different variables for nested for loops so that the outer loop
functions properly and speex files with multiple frames per buffer work
properly.
Fixes#441408.
Original commit message from CVS:
* ext/flac/gstflacenc.c: (gst_flac_enc_init),
(notgst_value_array_append_buffer),
(gst_flac_enc_process_stream_headers),
(gst_flac_enc_write_callback), (gst_flac_enc_chain),
(gst_flac_enc_change_state):
* ext/flac/gstflacenc.h:
Collect headers, add "streamheader" field to output caps and set
BUFFER_IN_CAPS flag on pushed header buffers. That way oggmux
produces output according to the official FLAC-to-Ogg mapping
instead of completely broken files. Fixes#426044.
Original commit message from CVS:
* gst/id3demux/gstid3demux.c: (gst_id3demux_reset),
(gst_id3demux_send_new_segment), (gst_id3demux_chain),
(gst_id3demux_sink_event):
* gst/id3demux/gstid3demux.h:
* gst/apetag/gsttagdemux.c: (gst_tag_demux_reset),
(gst_tag_demux_chain), (gst_tag_demux_sink_event),
(gst_tag_demux_send_new_segment):
Handle and adjust new-segment events so that downstream really
sees a stream with the tag pieces stripped off the front and back.
Fixes strangeness in seeking when mp3 decoders use the new-segment
byte position to estimate their current playback position timestamp
and then the arriving buffers don't match up.
Original commit message from CVS:
* gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_detect):
Don't unnecessarily perform a READY->NULL->READY transition on the
detected audio sink when starting up. Fixes: #440127
Original commit message from CVS:
* ext/flac/gstflacenc.c: (gst_flac_enc_sink_setcaps),
(gst_flac_enc_chain):
Don't crash in chain function if setcaps hasn't been called.
Original commit message from CVS:
Patch by: Vincent Torri <vtorri at univ-evry fr>
* sys/directdraw/gstdirectdrawsink.c:
(gst_directdraw_sink_buffer_alloc),
(gst_directdraw_sink_show_frame),
(gst_directdraw_sink_check_primary_surface),
(gst_directdraw_sink_check_offscreen_surface),
(EnumModesCallback2), (gst_directdraw_sink_get_ddrawcaps),
(gst_directdraw_sink_surface_create):
* sys/directdraw/gstdirectdrawsink.h:
Fix more warnings when compiling with MingW (#439914).
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_auth),
(gst_rtspsrc_try_send), (gst_rtspsrc_parse_methods),
(gst_rtspsrc_setup_streams), (gst_rtspsrc_open),
(gst_rtspsrc_play):
(rtsp_connection_send), (rtsp_connection_receive):
* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_after_send):
Fix for new API.
* gst/rtsp/rtspconnection.c: (add_auth_header),
Only add authorisation and session headers when sending messages.
* gst/rtsp/rtspmessage.c: (key_value_foreach), (rtsp_message_init),
(rtsp_message_init_request), (rtsp_message_init_response),
(rtsp_message_unset), (rtsp_message_add_header),
(rtsp_message_remove_header), (rtsp_message_get_header),
(rtsp_message_append_headers), (dump_key_value),
(rtsp_message_dump):
* gst/rtsp/rtspmessage.h:
Add support for multiple headers of the same type by storing the parsed
headers in a GArray instaed of a hashtable.
Original commit message from CVS:
* docs/plugins/gst-plugins-bad-plugins.args:
Remove directsoundsink property doc as this sink use the mixer
interface now.
* docs/plugins/gst-plugins-bad-plugins.interfaces:
Add interfaces implemented by Windows sinks.
* sys/directsound/gstdirectsoundsink.c:
* sys/directsound/gstdirectsoundsink.h:
Remove directsoundsink property and implement the mixer interface.
* win32/vs6/gst_plugins_bad.dsw:
* win32/vs6/libgstdirectsound.dsp:
Update project files.
* gst-libs/gst/dshow/gstdshow.cpp:
* gst-libs/gst/dshow/gstdshow.h:
* gst-libs/gst/dshow/gstdshowfakesink.cpp:
* gst-libs/gst/dshow/gstdshowfakesink.h:
* gst-libs/gst/dshow/gstdshowfakesrc.cpp:
* gst-libs/gst/dshow/gstdshowfakesrc.h:
* gst-libs/gst/dshow/gstdshowinterface.cpp:
* gst-libs/gst/dshow/gstdshowinterface.h:
* win32/common/libgstdshow.def:
* win32/vs6/libgstdshow.dsp:
Add a new gst library which allow to create internal Direct Show
graph (pipelines) to wrap Windows sources, decoders or encoders.
It includes a DirectShow fake source and sink and utility functions.
* sys/dshowsrcwrapper/gstdshowaudiosrc.c:
* sys/dshowsrcwrapper/gstdshowaudiosrc.h:
* sys/dshowsrcwrapper/gstdshowsrcwrapper.c:
* sys/dshowsrcwrapper/gstdshowsrcwrapper.h:
* sys/dshowsrcwrapper/gstdshowvideosrc.c:
* sys/dshowsrcwrapper/gstdshowvideosrc.h:
* win32/vs6/libdshowsrcwrapper.dsp:
Add a new plugin to wrap DirectShow sources on Windows.
It gets data from any webcam, dv cam, micro. We could add
tv tunner card later.
Original commit message from CVS:
* docs/plugins/Makefile.am:
Also look for .m (objectivec) files.
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/gst-plugins-bad-plugins.args:
* sys/osxvideo/osxvideosink.m:
Add documentation for element and properties.
Original commit message from CVS:
* gst/spectrum/gstspectrum.c (gst_spectrum_set_property,
gst_spectrum_event, gst_spectrum_transform_ip):
Use lock to protect from concurrent access.
Original commit message from CVS:
* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_class_init),
(gst_wavpack_enc_set_property), (gst_wavpack_enc_get_property):
Specify and use properties as unsigned int that are an unsigned int.
Original commit message from CVS:
* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_class_init),
(gst_wavpack_enc_init), (gst_wavpack_enc_set_wp_config),
(gst_wavpack_enc_set_property), (gst_wavpack_enc_get_property):
* ext/wavpack/gstwavpackenc.h:
Fixup docs, make the bitrate property an int as it should be and
allow to set the different extra processing modes instead of only
allowing none and the default one.
Original commit message from CVS:
* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init),
(gst_udpsrc_create), (gst_udpsrc_unlock), (gst_udpsrc_unlock_stop):
Since we depend on 0.10.13 -core, override the unlock_stop vmethod for
safer shutdown.
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler dot de>
* configure.ac:
* gst/auparse/gstauparse.c: (gst_au_parse_reset),
(gst_au_parse_parse_header), (gst_au_parse_chain):
* gst/auparse/gstauparse.h:
Use audioconvert for converting from non-native endianness floats
in auparse instead of doing it ourself. Fixes#424527.
This needs the audioconvert from plugins-base CVS.
Original commit message from CVS:
Patch by: Antoine Tremblay <hexa00 at gmail dot com>
* gst/rtp/gstrtph263ppay.c: (gst_fragmentation_mode_get_type),
(gst_rtp_h263p_pay_class_init), (gst_rtp_h263p_pay_init),
(gst_rtp_h263p_pay_set_property), (gst_rtp_h263p_pay_get_property),
(gst_rtp_h263p_pay_flush):
* gst/rtp/gstrtph263ppay.h:
Add new fragmentation mode base on GOB headers. Fixes#438940.
Original commit message from CVS:
* ext/wavpack/gstwavpackenc.c:
Add missing audioconverts in the example pipelines of wavpackenc. As
the wavpack stuff now needs input with 32 bit width (and random depth)
this is needed now. The example pipelines for the parser and decoder
are still fine.
Original commit message from CVS:
* sys/directdraw/gstdirectdrawsink.c: (gst_ddrawsurface_finalize),
(gst_directdraw_sink_buffer_alloc),
(gst_directdraw_sink_get_ddrawcaps),
(gst_directdraw_sink_surface_create):
Bunch of small fixes: remove static function that doesn't exist;
declare another one that does; printf format fix; use right macro
when specifying debug category; remove a bunch of unused variables;
#if 0 out an unused chunk of code (partially fixes#439914).
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_streams):
Don't crash when an unsupported transport error was returned by the
server, just try to configure the next stream. Fixes#439255.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
(gst_rtspsrc_set_property), (gst_rtspsrc_get_property),
(gst_rtspsrc_stream_configure_udp), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_loop_udp), (gst_rtspsrc_try_send), (gst_rtspsrc_send),
(gst_rtspsrc_setup_streams), (gst_rtspsrc_open):
* gst/rtsp/gstrtspsrc.h:
Add TCP timeout property and use it for all TCP connection.
* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
(rtsp_connection_write), (rtsp_connection_next_timeout),
(rtsp_connection_reset_timeout):
Make connect and writes cancelable and make them use the timeout.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send_keep_alive),
(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp),
(gst_rtspsrc_try_send), (gst_rtspsrc_send),
(gst_rtspsrc_setup_streams):
Refactor timeout handling.
Also send keep-alive when dealing with TCP transport.
* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
(rtsp_connection_free), (rtsp_connection_next_timeout),
(rtsp_connection_reset_timeout):
* gst/rtsp/rtspconnection.h:
Use a timer to handle the session timeouts, add some methods to deal
with timeouts.
