Original commit message from CVS:
* sys/ximage/gstximagesrc.c (gst_ximage_src_start,
gst_ximage_src_ximage_get):
* sys/ximage/gstximagesrc.h (last_ximage):
When using Damage actually keep the last frame, and not assume
that the buffer we get already has the last frame on it.
Copy the cursor over if we specify a non-zero start x and
start y.
Original commit message from CVS:
* gst/level/gstlevel.c: (gst_level_calculate_##TYPE),
(gst_level_transform_ip):
Use guint8 * instead of gpointer then vs6 know the size of data
pointed when moving the pointer.
* gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_handle_buffer):
Move instructions after variables declaration.
* win32/vs6/autogen.dsp:
* win32/vs6/libgstrtp.dsp:
* win32/vs6/libgstrtsp.dsp:
Update vs6 project files.
Original commit message from CVS:
* gst/rtsp/Makefile.am:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_handle_src_query),
(gst_rtspsrc_send_keep_alive), (gst_rtspsrc_open):
* gst/rtsp/rtsprange.c: (parse_npt_time), (parse_npt_range),
(parse_clock_range), (parse_smpte_range), (rtsp_range_parse),
(rtsp_range_free):
* gst/rtsp/rtsprange.h:
Add code to parse time ranges.
Report DURATION on the stream when possible.
Original commit message from CVS:
* gst/videomixer/videomixer.c: (gst_videomixer_blend_ayuv_ayuv),
(gst_videomixer_fill_checker), (gst_videomixer_fill_color),
(gst_videomixer_collected):
Fix strides calculation for AYUV (it's just width*4) (#436910).
Original commit message from CVS:
* gst/audiofx/audioamplify.c: (gst_audio_amplify_transform_ip):
* gst/audiofx/audiodynamic.c: (gst_audio_dynamic_transform_ip):
* gst/audiofx/audioinvert.c: (gst_audio_invert_transform_ip):
Sync the GObject properties before each processing step to properly
work with the controller.
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.
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.
Original commit message from CVS:
* gst/multipart/multipartmux.c: (gst_multipart_mux_queue_pads),
(gst_multipart_mux_collected):
Fix timestamps on outgoing buffers.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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).
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.
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.
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).
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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
Original commit message from CVS:
* gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_class_init),
(gst_rtp_mp4a_depay_setcaps), (gst_rtp_mp4a_depay_plugin_init):
This element is ready to be autoplugged.
Original commit message from CVS:
2007-04-05 Julien MOUTTE <julien@moutte.net>
* gst/avi/gstavidemux.c: (gst_avi_demux_process_next_entry):
Don't leave the offsets defined by upstream element on the
compressed data buffer we are pushing downstream. Make them
GST_BUFFER_OFFSET_NONE.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c:
Support audio/x-raw-float in wav files. This only works with
plugins-base CVS, using an older version doesn't have any
disadvantages though.
Original commit message from CVS:
* configure.ac:
* gst/auparse/gstauparse.c: (gst_au_parse_reset),
(gst_au_parse_parse_header), (gst_au_parse_chain):
* gst/auparse/gstauparse.h:
Revert last change as we don't want plugins-good to depend on
plugins-base CVS now.
Original commit message from CVS:
* configure.ac:
Require gst-plugins-base CVS for audioconvert with non-native
float support and width/depth fix in libgstriff.
Patch by: René Stadler <mail at renestadler dot de>
* gst/auparse/gstauparse.c: (gst_au_parse_reset),
(gst_au_parse_parse_header), (gst_au_parse_chain):
* gst/auparse/gstauparse.h:
Don't swap the floats ourself if they're not in native endianness.
Instead let audioconvert handle this. Fixes#339838.
Original commit message from CVS:
* gst/wavenc/gstwavenc.c: (gst_wavenc_create_header_buf),
(gst_wavenc_sink_setcaps):
Correctly handle width!=depth input.
* gst/wavparse/gstwavparse.c:
Already export in the caps that width==8 uses unsigned samples and
everything else uses signed samples.
Original commit message from CVS:
Patch by: Laurent Glayal <spglegle at yahoo dot fr>
* gst/udp/gstdynudpsink.c: (gst_dynudpsink_class_init),
(gst_dynudpsink_init), (gst_dynudpsink_set_property),
(gst_dynudpsink_get_property), (gst_dynudpsink_init_send),
(gst_dynudpsink_close):
* gst/udp/gstdynudpsink.h:
* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init),
(gst_udpsrc_create), (gst_udpsrc_set_property),
(gst_udpsrc_get_property), (gst_udpsrc_start), (gst_udpsrc_stop):
* gst/udp/gstudpsrc.h:
Rework the socket allocation a bit based on the sockfd argument so that
it becomes usable.
Add a closefd property to instruct the udp elements to close the custom
file descriptors when going to READY. Fixes#423304.
API:GstUDPSrc::closefd property
API:GstDynUDPSink::closefd property
Original commit message from CVS:
* gst/wavparse/gstwavparse.c:
Add support for wav files containing audio/x-raw-int with random
depths between 1 and 32 bits.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (find_stream_by_setup),
(gst_rtspsrc_create_stream), (gst_rtspsrc_stream_free),
(get_default_rate_for_pt), (gst_rtspsrc_parse_rtpmap),
(gst_rtspsrc_media_to_caps),
(gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_stream_configure_caps),
(gst_rtspsrc_activate_streams), (gst_rtspsrc_parse_rtpinfo):
* gst/rtsp/gstrtspsrc.h:
Handle default clock-rates for static payload types, rearrange stuff so
that the rtpmap field in the sdp can override the defaults.
Parse RTP-Info field to get the seqnum and timebase fields that should
go in the caps.
Delay configuring caps after we got the RTP-Info from the PLAY reply from
the server.
Original commit message from CVS:
Patch by: Christophe Dehais <christophe dot dehais at gmail dot com>
* ext/gconf/gconf.c: (gst_gconf_render_bin_with_default):
Accept complex pipeline descriptions as an audio profile instead of just
a single element. Fixes#420658.
Original commit message from CVS:
* gst/apetag/gsttagdemux.c: (gst_tag_demux_get_type):
Rename registered type in preparation of GstTagDemux moving to
-base at some point in the future.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers):
Streaming mode fixes: don't unref buffer we don't own any longer;
remove bogus adapter flush. Fixes#419338.
Original commit message from CVS:
* REQUIREMENTS: Change the format to key/value, add a bunch of
information, remove a bunch of requirements that are for
other GStreamer packages.
Original commit message from CVS:
* gst/rtp/gstrtppcmapay.c:
* gst/rtp/gstrtppcmapay.h:
* gst/rtp/gstrtppcmupay.c:
* gst/rtp/gstrtppcmupay.h:
Ported mulaw and alaw payloaders to use new base class
Original commit message from CVS:
* gst/apetag/gsttagdemux.c: (gst_tag_demux_srcpad_event):
Fix handling of -1 values for start and stop values when seeking,
and SEEK_CUR+SEEK_END here as well.
Original commit message from CVS:
* gst/id3demux/gstid3demux.c: (gst_id3demux_srcpad_event):
Fix handling of -1 values for start and stop values when seeking,
and SEEK_CUR+SEEK_END.
Original commit message from CVS:
* gst/id3demux/id3v2frames.c: (parse_picture_frame):
Fix parsing of ID3 v2.2.0 PIC frames. Only in version >= 2.3.0 is
the image format a variable-length NUL-terminated string; in
versions before that the image format is a fixed-length string of
3 characters (see #348644 for a sample tag).
Also make supplied mime type lower-case and fix up 'jpg' to 'jpeg'.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_parse_subindex),
(gst_avi_demux_parse_index):
* sys/v4l2/v4l2_calls.c: (gst_v4l2_fill_lists):
* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_grab_frame):
Printf format fixes; also add some missing quotes in translated
strings. Fixes#416728 and #416727.
Original commit message from CVS:
* gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_find_best):
Tim and I can't think of any reason the child audio sink needs to
be set back to NULL after successfully determining that it can
reach READY - it gets immediately set back to READY by the caller
anyway, causing an unnecessary close/open of any audio devices
involved.
Original commit message from CVS:
* sys/sunaudio/gstsunaudio.c: (plugin_init):
* sys/sunaudio/gstsunaudiomixertrack.c:
(gst_sunaudiomixer_track_new):
Actually translate sunaudio mixer track labels instead of just
marking the strings as translatable (#377306); clean up weird
label string mapping code that serves no apparent purpose. Also
set the 'untranslated-label' property when creating mixer tracks
if the GstMixerTrack base class supports this.
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/sunaudio.c: (GST_START_TEST),
(sunaudio_suite):
Very minimalistic unit test for sunaudiomixer element (compiles, but not
actually tested on a system where sunaudiomixer is available).
Original commit message from CVS:
* ext/dv/gstdvdec.c: (gst_dvdec_init), (gst_dvdec_sink_setcaps),
(gst_dvdec_src_negotiate), (gst_dvdec_chain),
(gst_dvdec_change_state):
* ext/dv/gstdvdec.h:
Infer pixel-aspect-ratio from the video frame format if it isn't
provided by the container, as happens when playing DV from AVI
or Quicktime containers.
Patch by: Wim Taymans <wim@fluendo.com>
Fixes#380944
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_activate_streams):
When activated, remove the udpsrc timeout, we have dataflow and timeouts
will later be handled by the jitterbuffer.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
(gst_avi_demux_push_event), (gst_avi_demux_do_seek),
(gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data),
(gst_avi_demux_chain):
Fix stream position reporting after a seek. Fixes#416445.
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler dot de>
* gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
(gst_avi_demux_push_event), (gst_avi_demux_process_next_entry),
(gst_avi_demux_stream_data), (gst_avi_demux_chain):
Make avidemux accept optional header chunks in any order.
Fixes#415446.
Original commit message from CVS:
reviewed by: Stefan Kost <ensonic@users.sf.net>
* gst/audiofx/Makefile.am:
* gst/audiofx/audiodynamic.c:
(gst_audio_dynamic_characteristics_get_type),
(gst_audio_dynamic_mode_get_type),
(gst_audio_dynamic_set_process_function),
(gst_audio_dynamic_base_init), (gst_audio_dynamic_class_init),
(gst_audio_dynamic_init), (gst_audio_dynamic_set_property),
(gst_audio_dynamic_get_property), (gst_audio_dynamic_setup),
(gst_audio_dynamic_transform_hard_knee_compressor_int),
(gst_audio_dynamic_transform_hard_knee_compressor_float),
(gst_audio_dynamic_transform_soft_knee_compressor_int),
(gst_audio_dynamic_transform_soft_knee_compressor_float),
(gst_audio_dynamic_transform_hard_knee_expander_int),
(gst_audio_dynamic_transform_hard_knee_expander_float),
(gst_audio_dynamic_transform_soft_knee_expander_int),
(gst_audio_dynamic_transform_soft_knee_expander_float),
(gst_audio_dynamic_transform_ip):
* gst/audiofx/audiodynamic.h:
* gst/audiofx/audiofx.c: (plugin_init):
Add new audiodynamic element which can act as a compressor or
expander. Supported are hard-knee and soft-knee operation modes with
user-specified ratio and threshold.
Attack and release parameters are not yet implemented but will follow.
* 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:
Integrate audiodynamic into the docs.
* tests/check/Makefile.am:
* tests/check/elements/audiodynamic.c: (setup_dynamic),
(cleanup_dynamic), (GST_START_TEST), (dynamic_suite), (main):
Add unit test for audiodynamic.
Original commit message from CVS:
* ext/raw1394/gstdv1394src.c: (gst_dv1394src_start):
Free handles that we allocated when exiting via the error paths.
Original commit message from CVS:
* gst/id3demux/gstid3demux.c: (gst_id3demux_add_srcpad),
(gst_id3demux_sink_activate):
Don't leak caps: make gst_id3demux_add_srcpad() not take ownership of the
caps passed to it (previouslly one code path assumes it takes ownership
while another one assumes it doesn't).
* configure.ac:
* tests/files/Makefile.am:
* tests/files/id3-407349-1.tag:
* tests/files/id3-407349-2.tag:
Add directory where data for unit tests can be stored.
* tests/Makefile.am:
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/id3demux.c: (pad_added_cb), (error_cb),
(read_tags_from_file), (run_check_for_file),
(check_date_1977_06_23), (GST_START_TEST), (id3demux_suite):
Add unit test for id3demux, and in particular for bug #407349. Only
testing pull-mode for now; push mode doesn't work yet because the test
files are smaller than ID3_TYPE_FIND_MIN_SIZE.
Original commit message from CVS:
* gst/id3demux/id3tags.c: (id3demux_id3v2_frames_to_tag_list):
* gst/id3demux/id3tags.h:
* gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame),
(parse_obsolete_tdat_frame):
Do not convert obsolete TDA/TDAT frames to TDRC frames, otherwise
the four-digit number will be interpreted as a year, whereas it is
month and day in DDMM format. Instead, parse TDAT frames and fix up
the date in the GST_TAG_DATE tag later if we also extracted a year.
Fixes#407349.
Original commit message from CVS:
* ext/gconf/gstswitchsink.c: (gst_switch_sink_dispose),
(gst_switch_commit_new_kid):
Fix up the dispose logic so it doesn't leak, and fix setting of
the child state so that we don't set a child to our current state
just as we are changing it to something else.
Original commit message from CVS:
* ext/gconf/gstswitchsink.c: (gst_switch_sink_dispose),
(gst_switch_commit_new_kid):
Fix up the reference counting of the child elements.
Original commit message from CVS:
* ext/gconf/gstswitchsink.c: (gst_switch_sink_reset),
(gst_switch_commit_new_kid), (gst_switch_sink_set_child):
Install fakesink in NULL by fixing some broken logic. This obviates
the need to manually set _IS_SINK.
Add some comments and remove a little cruft while I'm at it.
Original commit message from CVS:
* ext/gconf/gstswitchsink.c: (gst_switch_sink_reset):
Mark us as a sink when we have no fakesink in NULL. Fixes#414887.
Original commit message from CVS:
* tests/check/Makefile.am:
Gah! Also disable gconfvideosink from the tests, otherwise
it will instantiate autovideosink, and dfbvideosink and
leak on the buildbots.
Original commit message from CVS:
* ext/cdio/gstcdiocddasrc.c: (gst_cdio_cdda_src_open),
(gst_cdio_cdda_src_finalize):
Make sure we always destroy our libcdio handle.
Original commit message from CVS:
* tests/check/Makefile.am:
Disable autovideosink so the buildbots don't barf over memory
leaked in the directfb sink.
Original commit message from CVS:
* gst/multipart/multipartdemux.c: (gst_multipart_demux_init),
(gst_multipart_find_pad_by_mime):
Use gst_pad_new_from_static_template instead of
static_pad_template_get+pad_new.
Original commit message from CVS:
* ext/dv/gstdvdec.c: (gst_dvdec_init):
Use gst_pad_new_from_static_template instead of
static_pad_template_get+pad_new.
Original commit message from CVS:
Patch by: Loïc Minier <lool+gnome at via ecp fr>
* ext/libcaca/Makefile.am:
* gst/debug/Makefile.am:
Don't mix tabs and spaces (#414168).
Original commit message from CVS:
* ext/cdio/gstcdiocddasrc.c: (gst_cdio_cdda_src_probe_devices),
(gst_cdio_cdda_src_read_sector), (gst_cdio_cdda_src_open),
(gst_cdio_cdda_src_finalize):
Small code cleanups.
Don't use pad_alloc as the base class cannot deal with the error codes.
Original commit message from CVS:
Patch by: René Stadler <mail@renestadler.de>
* gst/wavparse/gstwavparse.c: (uint64_ceiling_scale_int),
(gst_wavparse_perform_seek), (gst_wavparse_stream_headers),
(gst_wavparse_stream_data):
Handle rounding better to not drop last sample frame. Fixes#356692
Original commit message from CVS:
* tests/check/Makefile.am:
Disable cacasink from the states check too - it also calls exit(1)
on us when it can't find a terminal to talk to.
Original commit message from CVS:
* ext/hal/gsthalaudiosink.c: (do_toggle_element):
* ext/hal/gsthalaudiosrc.c: (do_toggle_element):
Having NULL as UDI previously selected the default sink/src. Change
this back but mention it in the debug output.
* ext/hal/hal.c: (gst_hal_get_alsa_element),
(gst_hal_get_oss_element), (gst_hal_get_string),
(gst_hal_render_bin_from_udi), (gst_hal_get_audio_sink),
(gst_hal_get_audio_src):
* ext/hal/hal.h:
Refactor a bit, check all error conditions, greatly improve debugging
and fix some possible memory leaks. Also implement OSS support
and allow specifying an UDI that points to a real device. For this the
child device which supports ALSA (preferred) or OSS is used.
As a side effect this makes it impossible now to get a alsasink in
halaudiosrc and a alsasrc in halaudiosink.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (find_stream_by_channel),
(find_stream_by_udpsrc), (gst_rtspsrc_handle_message):
Errors from the udp sources are not fatal unless all of them are in
error.
Original commit message from CVS:
* tests/check/Makefile.am:
Disable aasink in the states test. I suspect this is the element that
is calling exit(1) when it can't proceed.
Original commit message from CVS:
* tests/check/Makefile.am:
Draw plugins in from the build tree sys/ dir, rather than picking
up the already installed versions.
Original commit message from CVS:
2007-03-01 Zaheer Abbas Merali <zaheerabbas at merali dot org>
* sys/ximage/gstximagesrc.c: (gst_ximage_src_open_display):
Error out correctly when getting xcontext fails.
Original commit message from CVS:
* gst/rtsp/gstrtpdec.c: (gst_rtpdec_change_state):
Make state change to PAUSED NO_PREROLL because that's what it will be in
the future and rtspsrc relies on it.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_change_state):
Don't error out when we don't get an error from the state change
function.
Original commit message from CVS:
* ext/hal/gsthalaudiosink.c: (do_toggle_element):
* ext/hal/gsthalaudiosrc.c: (do_toggle_element):
Check if the device UDI is set before trying to query HAL
about it and give a useful error message if it wasn't set.
* ext/hal/hal.c: (gst_hal_get_string):
Don't query HAL for NULL UDIs. Passing NULL as UDI to HAL
gives an assertion failure in D-Bus when running with
DBUS_FATAL_WARNINGS=1.
Original commit message from CVS:
* gst/rtsp/URLS:
Add another interesting test url.
* gst/rtsp/rtspmessage.c: (rtsp_message_get_header):
Don't allow getting header fields from data packets.
Original commit message from CVS:
* ext/shout2/gstshout2.c: (gst_shout2send_class_init),
(gst_shout2send_init), (gst_shout2send_start),
(gst_shout2send_set_property), (gst_shout2send_get_property):
* ext/shout2/gstshout2.h:
Add a property for username.
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.
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.
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
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.
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.
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.
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.
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.
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.
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.
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).
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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
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.
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.
