In streaming mode, avidemux is not supposed to send an EOS event downstream but
it is supposed to return UNEXPECTED from the chain function instead so that
upstream can do the right EOS handling.
Fix the duration query so that it also works with formats other than
TIME, such as DEFAULT to get the number of frames.
Add a convert function.
Fixes#578052.
In the sequence of header lengths, for headers >127 bytes, we use
multiple bytes to encode the length. Bytes other than the last must have
the top (flag) bit set.
The on-npt-stop signals was added only recently to rtpjitterbuffer in
-bad, so check if the signal exists before g_signal_connect()ing to
it, to avoid warnings.
Some clips (trailers) may have (length-wise) unbalanced streams,
which stalls the pipeline if seeking into that region.
Additional stream synchronization can handle this, as well as
sparse (subtitle) streams (at some later time ?)
Try to post a decent error message when it looks like we're failing
because the Real RTSP extension plugin is missing. Also add i18n
bits for rtspsrc so our error messages get translated.
Cater for DELTA_UNIT flag on buffers, keep track of current
position, remove and warn about edit lists if any (as those
as are de facto discarded anyway), add some debug statements
and indent fixes.
The audioMuxVersion structure is packed in such a way that the codec
data does not start byte-aligned, which means there's an extra bit of
padding at the end. We don't want that bit in the codec data, since
some decoders seem get confused when they're fed with an extra codec
data byte (also it's just not right of course).
Add network interface selection when joining multicast groups.
Useful when using the udpsrc on multihomed hosts.
Fixes#575234.
API: GstUDPSrc::multicast-iface
Non-ok flow returns may happen for a variety of perfectly legitimate and expected reasons
(temporarily not linked, seeking, pipeline shutdown), so we really shouldn't spew ERROR
debug messages to stderr in those cases. Fixes#570781. (Seems like someone already took
care of some of these.)
Standard pull mode loop based SEEK handling fails in push mode,
so convert the SEEK event appropriately and dispatch to upstream.
Also cater for NEWSEGMENT event handling, and properly inform
downstream and application of SEEKABLE capabilities, depending
on scheduling mode and upstream.
Previously the sockaddr length used for recvfrom() was calculated as
sizeof (struct sockaddr). However, this is too little to hold an IPv6
address, so the full size of the gst_sockaddr union should be used
instead.
MS RTSP spec states that the UDP port pair used in subsequent SETUP
requests for various streams must be identical (since there will actually
be only 1 stream of muxed asf packets). Following traditional specs and
using different port pairs in the SETUPs for separate streams will result
in all but the first one failing and only one stream being streamed.
So, in appropriate circumstances, retry UDP SETUP using previously used
port pair. Fixes#552650.
When we are dealing with connected sockets shared between a udpsrc and a udpsink
we might receive ICMP connection refused error messages in udpsrc that will
cause it to go into a bursty loop because the poll returns right away without a
message to read.
Instead of looping, read the error message from the error queue in udpsrc.
Fixes#567857.
Reading integers from random memory addresses will result
in SIGBUS on some architectures if the memory address
is not correctly aligned. This can happen at two
places in avidemux so we should use GST_READ_UINT32_LE
and friends here. Fixes bug #572256.
stps atoms contain "partial sync" information, which means that it's
a sync point where pts != dts. This is needed to properly handle
MPEG2, H.264, Dirac, etc., in quicktime.
Not all Matroska files have a Tags element which contains
information about the title among other things. Most video
Matroska files only contain the Title element so we
should parse this too. Fixes bug #570435.
Move reallocating the history buffer out of _compute_frequencies() and call the
right function as needed. Add some logging and tweak the formatting of existing
logging. Simplify setting need_new_coefficients when changing properties.
Link to properties. Correct titles for examples. Document a few trivial cases. Keep lists in section file and docs/plugins/Makefile.am alphabetically ordered. Fix warnings that gtk-doc points out.
Introduce a new max-delay property that can only
be set before going to PLAYING or PAUSED. This
is used to limit the maximum delay and is set
to the current delay by default.
Using this will make sure that we have enough data
in our internal ringbuffer for the echo. With dynamic
reallocation of the ringbuffer as used before silence
could've been used as the echo directly after setting
a new delay.
Link to properties. Correct titles for examples. Document a few trivial cases. Keep lists in section file and docs/plugins/Makefile.am alphabetically ordered.
Save some allocations if the echo delay is increased often
during playback by always allocating enough memory to hold
data up to the next complete second, i.e. in the worst case
allocate memory for one additional second.
Add a note to the docs that audioecho's reverb will
sound metallic. This happens because for a real
reverb filter additional filtering is necessary.
Also note which values should be used for the delay
property to get an echo effect.
The element can add an echo and a simple reverb effect to
an audio stream but for a real reverb filter it would need
some additional filtering to prevent a metallic-sounding
result.
Original commit message from CVS:
Patch by: Luotao Fu <l dot fu at pengutronix dot de>
* gst/videocrop/gstvideocrop.c:
(gst_video_crop_get_image_details_from_caps):
Add 8bit grayscale support to videocrop plugin. Fixes#567952.
Original commit message from CVS:
* gst/spectrum/gstspectrum.c: (gst_spectrum_reset_state),
(gst_spectrum_transform_ip):
* gst/spectrum/gstspectrum.h:
Implement a simple compensation algorithm for rounding errors.
This makes sure that a spectrum message is posted on the bus
every interval nanoseconds. Fixes bug #567955.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_parse_segments):
Catch invalid and commonly wrong playback rates in the elst atoms.
Fixes#567800.
Original commit message from CVS:
* gst/spectrum/gstspectrum.c: (gst_spectrum_reset_state):
Don't call gst_fft_f32_free() with NULL to prevent a
crash. Fixes bug #567642.
Original commit message from CVS:
* gst/spectrum/gstspectrum.c: (gst_spectrum_transform_ip):
* gst/spectrum/gstspectrum.h:
Use correct types for frame/fft counters and some minor
cleanup.
Original commit message from CVS:
* gst/spectrum/Makefile.am:
* gst/spectrum/README:
* gst/spectrum/gstspectrum.c: (gst_spectrum_base_init),
(gst_spectrum_class_init), (gst_spectrum_init),
(gst_spectrum_reset_state), (gst_spectrum_finalize),
(gst_spectrum_set_property), (gst_spectrum_start),
(gst_spectrum_stop), (gst_spectrum_setup),
(gst_spectrum_transform_ip):
* gst/spectrum/gstspectrum.h:
Post a spectrum message on the bus for every interval, even
if the interval is small than the length of the FFT.
Fixes bug #567642.
Major cleanup of the spectrum element.
Original commit message from CVS:
Patch by: Thiago Sousa Santos <thiagoss@embedded.ufcg.edu.br>
* gst/qtdemux/qtdemux.c:
Fix format string for guint64.
Original commit message from CVS:
* gst/audiofx/audiochebband.c: (gst_audio_cheb_band_class_init),
(gst_audio_cheb_band_init), (gst_audio_cheb_band_finalize),
(gst_audio_cheb_band_set_property):
* gst/audiofx/audiochebband.h:
* gst/audiofx/audiocheblimit.c: (gst_audio_cheb_limit_class_init),
(gst_audio_cheb_limit_init), (gst_audio_cheb_limit_finalize),
(gst_audio_cheb_limit_set_property):
* gst/audiofx/audiocheblimit.h:
* gst/audiofx/audiowsincband.c: (gst_audio_wsincband_class_init),
(gst_audio_wsincband_init), (gst_audio_wsincband_finalize),
(gst_audio_wsincband_set_property):
* gst/audiofx/audiowsincband.h:
* gst/audiofx/audiowsinclimit.c: (gst_audio_wsinclimit_class_init),
(gst_audio_wsinclimit_init), (gst_audio_wsinclimit_finalize),
(gst_audio_wsinclimit_set_property):
* gst/audiofx/audiowsinclimit.h:
Use a custom mutex for protecting the instance fields instead of
the GstObject lock. Using the latter can lead to deadlocks, especially
with the FIR filters when updating the latency.
Original commit message from CVS:
* gst/audiofx/Makefile.am:
* gst/audiofx/audiofxbasefirfilter.c:
(gst_audio_fx_base_fir_filter_dispose),
(gst_audio_fx_base_fir_filter_base_init),
(gst_audio_fx_base_fir_filter_class_init),
(gst_audio_fx_base_fir_filter_init),
(gst_audio_fx_base_fir_filter_push_residue),
(gst_audio_fx_base_fir_filter_setup),
(gst_audio_fx_base_fir_filter_transform),
(gst_audio_fx_base_fir_filter_start),
(gst_audio_fx_base_fir_filter_stop),
(gst_audio_fx_base_fir_filter_query),
(gst_audio_fx_base_fir_filter_query_type),
(gst_audio_fx_base_fir_filter_event),
(gst_audio_fx_base_fir_filter_set_kernel):
* gst/audiofx/audiofxbasefirfilter.h:
* gst/audiofx/audiofxbaseiirfilter.c:
Implement a base class for generic audio FIR filters.
* gst/audiofx/audiowsincband.c:
(gst_gst_audio_wsincband_mode_get_type),
(gst_gst_audio_wsincband_window_get_type),
(gst_audio_wsincband_base_init), (gst_audio_wsincband_class_init),
(gst_audio_wsincband_init), (gst_audio_wsincband_build_kernel),
(gst_audio_wsincband_setup), (gst_audio_wsincband_set_property),
(gst_audio_wsincband_get_property):
* gst/audiofx/audiowsincband.h:
* gst/audiofx/audiowsinclimit.c:
(gst_audio_wsinclimit_mode_get_type),
(gst_audio_wsinclimit_window_get_type),
(gst_audio_wsinclimit_base_init),
(gst_audio_wsinclimit_class_init), (gst_audio_wsinclimit_init),
(gst_audio_wsinclimit_build_kernel), (gst_audio_wsinclimit_setup),
(gst_audio_wsinclimit_set_property),
(gst_audio_wsinclimit_get_property):
* gst/audiofx/audiowsinclimit.h:
* tests/check/elements/audiowsincband.c: (GST_START_TEST):
* tests/check/elements/audiowsinclimit.c: (GST_START_TEST):
Use this new base class for audiowsincband and audiowsinclimit.
Also cleanup both elements.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c:
In push mode, error out if we get EOS before we've created any srcpads.
Handle (in pull mode) some files that have a truncated moov atom where
the final sub-atom is a 'free' atom and the contents of that are not
present in the file.
Original commit message from CVS:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_video_caps):
* gst/matroska/matroska-mux.c: (gst_matroska_mux_video_pad_setcaps):
Some cleanups, refactoring and minor enhancements in caps handling.
* gst/matroska/matroska-mux.c: (gst_matroska_mux_class_init),
(gst_matroska_mux_init), (gst_matroska_pad_reset),
(gst_matroska_pad_free), (gst_matroska_mux_reset),
(gst_matroska_mux_video_pad_setcaps),
(gst_matroska_mux_request_new_pad):
* tests/check/elements/matroskamux.c: (teardown_src_pad):
Only remove, release or reset what is appropriate upon state change.
Original commit message from CVS:
* gst/matroska/matroska-mux.c: (gst_matroska_mux_reset),
(gst_matroska_mux_handle_sink_event), (gst_matroska_mux_finish):
* gst/matroska/matroska-mux.h:
Remove internal taglist and fully use tagsetter interface.
Original commit message from CVS:
* gst/avi/gstavimux.c: (gst_avi_mux_reset),
(gst_avi_mux_riff_get_avi_header):
* gst/avi/gstavimux.h:
Ensure header size invariance during subsequent rewrite by using
tags snapshot.
Original commit message from CVS:
* gst/audiofx/Makefile.am:
* gst/audiofx/audiofxbaseiirfilter.c:
(gst_audio_fx_base_iir_filter_base_init),
(gst_audio_fx_base_iir_filter_dispose),
(gst_audio_fx_base_iir_filter_class_init),
(gst_audio_fx_base_iir_filter_init),
(gst_audio_fx_base_iir_filter_calculate_gain),
(gst_audio_fx_base_iir_filter_set_coefficients),
(gst_audio_fx_base_iir_filter_setup), (process),
(gst_audio_fx_base_iir_filter_transform_ip),
(gst_audio_fx_base_iir_filter_stop):
* gst/audiofx/audiofxbaseiirfilter.h:
Implement a base class for IIR filters.
* gst/audiofx/audiochebband.c: (gst_audio_cheb_band_base_init),
(gst_audio_cheb_band_class_init), (gst_audio_cheb_band_init),
(generate_coefficients), (gst_audio_cheb_band_set_property),
(gst_audio_cheb_band_setup):
* gst/audiofx/audiochebband.h:
* gst/audiofx/audiocheblimit.c: (gst_audio_cheb_limit_base_init),
(gst_audio_cheb_limit_class_init), (gst_audio_cheb_limit_init),
(generate_coefficients), (gst_audio_cheb_limit_set_property),
(gst_audio_cheb_limit_setup):
* gst/audiofx/audiocheblimit.h:
Use the IIR filter base class for the chebyshev filters.
Original commit message from CVS:
Patch by: j^ <j at oil21.org>
* gst/qtdemux/qtdemux.c: (qtdemux_video_caps),
(qtdemux_audio_caps):
Add codec mapping for xvid, fmp4 and ac3 tracks.
Fixes#565850
Original commit message from CVS:
* ext/pulse/pulsemixerctrl.c:
And remove temporary comment pointing to the bug ticket.
* gst/avi/gstavimux.c:
Move reoccuring logging to LOG and log instance too.
Original commit message from CVS:
* gst/avi/gstavimux.c:
Instead of filtering wrongly just use the mergemode. Applications is
use KEEP_ALL if they want to supress tag-events. Fixes#563221 for
avi for real (I hope). Everyone chime in, before I fix the others.
Original commit message from CVS:
* gst/avi/gstavidemux.c:
More logging.
* gst/avi/gstavimux.c:
Handle more metadata fields. Better estimate of metadata size. Don't
merge received tags, if application has specified tags using
GST_TAG_MERGE_REPLACE_ALL. Fixes#563221 for avi.
Original commit message from CVS:
* gst/rtp/gstrtpjpegdepay.c: (gst_rtp_jpeg_depay_process):
Add an EOI marker at the end of the jpeg frame when it's missing.
Fixes#563056.
Original commit message from CVS:
* gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_flush),
(gst_rtp_mp4v_pay_event):
Don't try to push packets before we could find a valid config
startcode. Fixes#563509.
Original commit message from CVS:
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_render):
Provide the parameters that are required for the format string
to fix a compiler warning.
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_render):
Make gst_multiudpsink_render() ignore errors from sendto() instead of
breaking streaming. Emit a warning instead. Fixes#562572.
Original commit message from CVS:
Patch by: 이문형 <iwings at gmail dot com>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_udp):
Prevent further read/write actions taken to the connect-failed socket by
erroring out quickly. See #562258.
Original commit message from CVS:
2008-11-25 Julien Moutte <julien@fluendo.com>
* gst/qtdemux/qtdemux.c: (qtdemux_video_caps): Add MPG1 and MPG2
fourcc
to supported qtdemux video codecs as I found some video clips
using
those.
Original commit message from CVS:
* gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_detect):
* gst/autodetect/gstautoaudiosrc.c: (gst_auto_audio_src_detect):
* gst/autodetect/gstautovideosink.c: (gst_auto_video_sink_reset),
(gst_auto_video_sink_detect):
* gst/autodetect/gstautovideosrc.c: (gst_auto_video_src_detect):
Post an error when we can't set the internal ghostpad target.
Original commit message from CVS:
* gst/videocrop/gstvideocrop.c: (gst_video_crop_init),
(gst_video_crop_transform), (gst_video_crop_transform_caps),
(gst_video_crop_set_caps), (gst_video_crop_set_property):
* gst/videocrop/gstvideocrop.h:
Fix renegotiation when changing properties using the new basetransform
features. Fixes#561502.
* tests/icles/Makefile.am:
* tests/icles/videocrop2-test.c: (make_pipeline), (main):
Add crazy interactive test unit for dynamically changing properties.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (new_session_pad),
(gst_rtspsrc_parse_range):
Add some more debugging.
Use the reanges received from the server unconditionally.
Fixes#561625.
Original commit message from CVS:
Patch by: Tal Shalif <tshalif at nargila dot org>
* gst/qtdemux/qtdemux.c: (qtdemux_audio_caps):
Use G_{BIG,LITTLE}_ENDIAN instead of the non-GLib variants as
the latter don't exist on some systems (mingw). Fixes bug #561992.
Original commit message from CVS:
* gst/udp/gstmultiudpsink.c:
* gst/udp/gstudpnetutils.c:
* gst/udp/gstudpnetutils.h:
* gst/udp/gstudpsrc.c:
Fix multiudpsink on OSX by passing the specific length of the socket,
refactor that into a function shared with the same thing in udpsrc.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (uint64_ceiling_scale_int),
(uint64_ceiling_scale), (gst_wavparse_calculate_duration),
(gst_wavparse_stream_headers):
Fix the scaling code.
Fix parsing of the INFO chunks, we were reading the wrong number of
bytes. Fixes#561580.
Original commit message from CVS:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_audio_caps):
* gst/matroska/matroska-ids.h:
Make mkvdemux aware of E-AC3.
Original commit message from CVS:
* gst/rtsp/Makefile.am:
* gst/rtsp/gstrtsp.c: (plugin_init):
* gst/rtsp/gstrtspgoogle.c:
* gst/rtsp/gstrtspgoogle.h:
Remove google extension again, it's not needed anymore because we never
send multiple transports anymore.
Original commit message from CVS:
Based on patch by: Eric Zhang <chao.zhang at access-company dot com>
* gst/rtsp/gstrtspsrc.c: (gst_rtsp_nat_method_get_type),
(gst_rtspsrc_class_init), (gst_rtspsrc_set_property),
(gst_rtspsrc_get_property), (gst_rtspsrc_create_stream),
(gst_rtspsrc_stream_free),
(gst_rtspsrc_stream_configure_udp_sinks),
(gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_send_dummy_packets),
(gst_rtspsrc_create_transports_string),
(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Add property to configure NAT traversal method.
Ignore EOS from the internal sinks.
Implement sending dummy packets as a (simple) method to open up
some firewalls.
Send PLAY request to the server after we started the udp sources.
Fixes#559545.
Original commit message from CVS:
Patch by: Yotam <sh dot yotam at gmail dot com>
* gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_event):
Flush the remaining frames on EOS. Fixes#560641.
Original commit message from CVS:
* gst/qtdemux/qtdemux.h (struct _GstQTDemux):
* gst/qtdemux/qtdemux.c (gst_qtdemux_do_seek): Queue up new
segment events instead of sending them from the seeking thread.
Fixes#559288.
(gst_qtdemux_push_pending_newsegment): New helper, sends out
queued newsegment events.
(gst_qtdemux_loop_state_movie): Voilà, call it here. Only need to
call it here, as we only seek when looping, and only push in the
movie state.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_tag_add_tmpo),
(qtdemux_tag_add_covr), (qtdemux_parse_udta):
* gst/qtdemux/qtdemux_fourcc.h:
* gst/qtdemux/qtdemux_types.c:
Add cover and alternative copyright tag, and enhance some existing
ones by marking them as container atoms.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_transports_string),
(gst_rtspsrc_change_state):
Only send one transport at a time for improved compatibility with some
broken servers. See #537832.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_do_seek),
(gst_rtspsrc_perform_seek):
Only pause/play in the seek handler when the source was playing.
Fixes#529379.
Original commit message from CVS:
* gst/matroska/matroska-mux.c:
(gst_matroska_mux_handle_dirac_packet):
Fix muxing of Dirac streams if the input already has the format
we need, i.e. is the output of matroskademux.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_parse_trak),
(qtdemux_video_caps), (qtdemux_audio_caps):
Refactor some raw audio caps building, and handle >16-bit cases.
Fix/replace building caps from a string description.
Original commit message from CVS:
* gst/audiofx/audiowsincband.c:
* gst/audiofx/audiowsinclimit.c:
* gst/cutter/gstcutter.c:
Make author name consistent with others.
Original commit message from CVS:
Based on patch by: Eric Zhang <chao.zhang at access-company dot com>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_perform_seek),
(gst_rtspsrc_stream_configure_udp_sink):
Pause the RTSP stream before doing a new play request.
Make sure that adding the udpsinks does not cause the rtspsrc to become
a sink. Fixes#559547.
Original commit message from CVS:
* gst/matroska/matroska-ids.h:
* gst/matroska/matroska-mux.c: (gst_matroska_pad_free),
(gst_matroska_mux_handle_dirac_packet),
(gst_matroska_mux_write_data):
Implement Dirac muxing into Matroska comforming to the spec, i.e.
put all Dirac packages up to a picture into a Matroska block.
TODO: Implement writing of the ReferenceBlock Matroska elements,
currently the Dirac muxing is only 100% correct if Matroska version 2
is selected for muxing.
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:
Add support for float/double as input and remove the (nowadays)
useless parsing of the depth as we require width==depth.
Original commit message from CVS:
* gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_setcaps):
* gst/rtp/gstrtpmpapay.c:
Narrow down the caps of the mpeg audio pay/depayloaders to only accept
mpeg version 1. Fixes#558427.
Original commit message from CVS:
* gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_flush),
(gst_rtp_L16_pay_getcaps):
Only put an integral amount of samples in the RTP packet.
Fixes#556641.
Original commit message from CVS:
* gst/rtp/gstrtpchannels.c: (gst_rtp_channels_get_by_index):
* gst/rtp/gstrtpchannels.h:
Add method to get possible channel positions.
Original commit message from CVS:
* gst/wavenc/gstwavenc.c: (gst_wavenc_chain):
Don't allow width=32,depth=24 as input. WAV requires that the width
is the next integer multiply of 8 from the depth.
Original commit message from CVS:
* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16_depay_setcaps):
* gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_setcaps),
(gst_rtp_L16_pay_getcaps):
* gst/rtp/gstrtpchannels.c: (check_channels),
(gst_rtp_channels_get_by_pos), (gst_rtp_channels_get_by_order),
(gst_rtp_channels_create_default):
* gst/rtp/gstrtpchannels.h:
Add mappings for multichannel support. Does not completely just work
because the getcaps function does not yet return the allowed channel
mappings. See #556641.
Original commit message from CVS:
* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16_depay_setcaps),
(gst_rtp_L16_depay_process):
Check if clock-rate and channels are valid.
Don't ignore the return value of setcaps.
No need to validate the buffer, the base class does that for us.
Use the marker bit to set the DISCONT flag on outgoing buffers.
* gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_setcaps):
Don't ignore the return value of set_outcaps.
* gst/rtp/gstrtpac3depay.c: (gst_rtp_ac3_depay_setcaps),
(gst_rtp_ac3_depay_process):
Don't ignore the return value of set_caps.
No need to validate the buffer, the base class does that for us.
* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_setcaps),
(gst_rtp_amr_depay_process):
* gst/rtp/gstrtpamrdepay.h:
Don't ignore the return value of setcaps.
No need to validate the buffer, the base class does that for us.
No need to set output caps on the buffers, the base class does that for
us.
The subclass will make sure we are negotiated.
* gst/rtp/gstrtpdvdepay.c: (gst_rtp_dv_depay_setcaps),
(gst_rtp_dv_depay_process), (gst_rtp_dv_depay_reset):
* gst/rtp/gstrtpdvdepay.h:
Clean up caps negotiation.
The subclass will make sure we are negotiated.
* gst/rtp/gstrtpg726depay.c: (gst_rtp_g726_depay_setcaps),
(gst_rtp_g726_depay_process):
Clean up caps negotiation.
Use the marker bit to set the DISCONT flag on outgoing buffers.
* gst/rtp/gstrtpg729depay.c: (gst_rtp_g729_depay_init),
(gst_rtp_g729_depay_setcaps), (gst_rtp_g729_depay_process):
* gst/rtp/gstrtpg729depay.h:
The subclass will make sure we are negotiated.
Use the marker bit to set the DISCONT flag on outgoing buffers.
* gst/rtp/gstrtpgsmdepay.c: (gst_rtp_gsm_depay_setcaps),
(gst_rtp_gsm_depay_process):
Clean up caps negotiation.
Use the marker bit to set the DISCONT flag on outgoing buffers.
* gst/rtp/gstrtpgsmpay.c: (gst_rtp_gsm_pay_setcaps):
Clean up caps negotiation.
Don't ignore the return value of set_outcaps.
* gst/rtp/gstrtph263depay.c: (gst_rtp_h263_depay_setcaps),
(gst_rtp_h263_depay_process):
Clean up caps negotiation.
No need to validate the buffer, the base class does that for us.
* gst/rtp/gstrtph263pay.c: (gst_rtp_h263_pay_setcaps),
(gst_rtp_h263_pay_flush), (gst_rtp_h263_pay_handle_buffer):
* gst/rtp/gstrtph263pay.h:
Don't ignore the return value of set_outcaps.
Do some more timestamps.
* gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_setcaps),
(gst_rtp_h263p_depay_process):
Clean up caps negotiation.
Don't ignore the return value of setcaps.
No need to validate the buffer, the base class does that for us.
* gst/rtp/gstrtph263ppay.c: (gst_rtp_h263p_pay_class_init),
(gst_rtp_h263p_pay_setcaps), (gst_rtp_h263p_pay_flush),
(gst_rtp_h263p_pay_handle_buffer):
* gst/rtp/gstrtph263ppay.h:
Don't ignore the return value of set_outcaps.
Do some more timestamps.
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_setcaps),
(gst_rtp_h264_depay_process):
Clean up caps negotiation.
Don't ignore the return value of setcaps.
Fix possible caps leak.
No need to validate the buffer, the base class does that for us.
* gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_setcaps):
Add some more debug info.
* gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_setcaps),
(gst_rtp_ilbc_depay_process):
Clean up caps negotiation.
Use the marker bit to set the DISCONT flag on outgoing buffers.
* gst/rtp/gstrtpilbcpay.c: (gst_rtpilbcpay_sink_setcaps):
Clean up caps negotiation.
* gst/rtp/gstrtpmp1sdepay.c: (gst_rtp_mp1s_depay_setcaps),
(gst_rtp_mp1s_depay_process):
Clean up caps negotiation.
Don't ignore the return value of setcaps.
No need to validate the buffer, the base class does that for us.
No need to set caps on buffers, subclass does that for us.
* gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_setcaps),
(gst_rtp_mp2t_depay_process):
Clean up caps negotiation.
Don't ignore the return value of setcaps.
No need to validate the buffer, the base class does that for us.
No need to set caps on buffers, subclass does that for us.
* gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_setcaps),
(gst_rtp_mp4a_depay_process):
Clean up caps negotiation.
Don't ignore the return value of setcaps.
No need to validate the buffer, the base class does that for us.
* gst/rtp/gstrtpmp4apay.c: (gst_rtp_mp4a_pay_new_caps),
(gst_rtp_mp4a_pay_setcaps):
Don't ignore the return value of set_outcaps.
* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_setcaps),
(gst_rtp_mp4g_depay_process):
Clean up caps negotiation.
Don't ignore the return value of setcaps.
No need to validate the buffer, the base class does that for us.
No need to set caps on buffers, subclass does that for us.
* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_finalize),
(gst_rtp_mp4g_pay_new_caps), (gst_rtp_mp4g_pay_setcaps):
Don't ignore the return value of set_outcaps.
* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_setcaps),
(gst_rtp_mp4v_depay_process):
Clean up caps negotiation.
Don't ignore the return value of setcaps.
No need to validate the buffer, the base class does that for us.
No need to set caps on buffers, subclass does that for us.
* gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_new_caps),
(gst_rtp_mp4v_pay_setcaps):
Don't ignore the return value of set_outcaps.
* gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_setcaps),
(gst_rtp_mpa_depay_process):
Clean up caps negotiation.
Don't ignore the return value of setcaps.
No need to validate the buffer, the base class does that for us.
Use the marker bit to set the DISCONT flag on outgoing buffers.
* gst/rtp/gstrtpmpapay.c: (gst_rtp_mpa_pay_setcaps):
Don't ignore the return value of set_outcaps.
* gst/rtp/gstrtpmpvdepay.c: (gst_rtp_mpv_depay_setcaps),
(gst_rtp_mpv_depay_process):
Clean up caps negotiation.
Actually set output caps.
No need to validate the buffer, the base class does that for us.
* gst/rtp/gstrtpmpvpay.c: (gst_rtp_mpv_pay_setcaps):
Don't ignore the return value of set_outcaps.
* gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_setcaps),
(gst_rtp_pcma_depay_process):
Clean up caps negotiation.
Set output buffer duration because we can.
Use the marker bit to set the DISCONT flag on outgoing buffers.
* gst/rtp/gstrtppcmapay.c: (gst_rtp_pcma_pay_setcaps):
Don't ignore the return value of set_outcaps.
* gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_setcaps),
(gst_rtp_pcmu_depay_process):
Clean up caps negotiation.
Use the marker bit to set the DISCONT flag on outgoing buffers.
* gst/rtp/gstrtppcmupay.c: (gst_rtp_pcmu_pay_setcaps):
Don't ignore the return value of set_outcaps.
* gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_init),
(gst_rtp_speex_depay_setcaps), (gst_rtp_speex_depay_process):
Clean up caps negotiation.
Set output caps on the pad and header buffers.
Set duration on output buffers because we can.
* gst/rtp/gstrtpspeexpay.c: (gst_rtp_speex_pay_parse_ident):
Don't ignore the return value of set_outcaps.
* gst/rtp/gstrtpsv3vdepay.c: (gst_rtp_sv3v_depay_setcaps),
(gst_rtp_sv3v_depay_process):
Clean up caps negotiation.
No need to validate the buffer, the base class does that for us.
No need to set caps out output buffers, subclass does that.
* gst/rtp/gstrtptheoradepay.c: (gst_rtp_theora_depay_setcaps),
(gst_rtp_theora_depay_process):
Don't ignore the return value of setcaps.
No need to validate the buffer, the base class does that for us.
* gst/rtp/gstrtptheorapay.c: (gst_rtp_theora_pay_class_init),
(gst_rtp_theora_pay_flush_packet), (encode_base64),
(gst_rtp_theora_pay_finish_headers), (gst_rtp_theora_pay_parse_id),
(gst_rtp_theora_pay_handle_buffer):
Don't ignore the return value of set_outcaps.
* gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_setcaps),
(gst_rtp_vorbis_depay_process):
Don't ignore the return value of setcaps.
No need to validate the buffer, the base class does that for us.
* gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_finish_headers):
Don't ignore the return value of set_outcaps.
* gst/rtp/gstrtpvrawdepay.c: (gst_rtp_vraw_depay_setcaps):
Clean up caps negotiation, don't ignore setcaps return.
* gst/rtp/gstrtpvrawpay.c: (gst_rtp_vraw_pay_setcaps):
Don't ignore the return value of set_outcaps.
Original commit message from CVS:
* gst/autodetect/gstautoaudiosrc.c:
(gst_auto_audio_src_class_init):
* gst/autodetect/gstautovideosrc.c:
(gst_auto_video_src_class_init):
Fix "Since" tags in the documentation.
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/matroska/matroska-mux.c: (gst_matroska_mux_reset),
(gst_matroska_mux_request_new_pad), (gst_matroska_mux_release_pad):
Fix a memory leak when pads are requested but the pipeline never
goes into PLAYING.
Correctly remove request pads, no matter if they have collected
data or not.
Fixes bug #557710.
Original commit message from CVS:
Patch by: <lrn1986 at gmail dot com>
* gst/udp/gstudpnetutils.h:
Define the correct WINVER so getaddinfo() can be used when using
mingw32. Fixes bug #557294.
Original commit message from CVS:
* gst/equalizer/gstiirequalizer.c: (update_coefficients):
Don't calculate the filter coefficients for every single buffer
but only when it's needed. Fixes bug #557260.
Original commit message from CVS:
* gst/avi/gstavimux.c:
Fix VPRP chunk setup in avimux.
Fixes: #556010
Patch By: Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
Original commit message from CVS:
* gst/videobox/gstvideobox.c:
support dynamically changing properties in videobox
Fixed: #557085
Patch By: Wim Taymans <wim.taymans@collabora.co.uk>
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_stream_scan):
Skip entries for streams that don't have a output pad yet, thereby
avoiding calling pad functions with a NULL pad.
Fixes#556424
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_handle_src_event):
Return TRUE instead of FALSE from the event handler when we swallowed the
event.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_parse_subindex),
(gst_avi_demux_parse_stream), (gst_avi_demux_parse_index):
* gst/avi/gstavidemux.h:
For timestamping audio packets we need to take into account the
amount of blocks in one entry using the blockalign. Fixes some sync
issues with zero-padded audio blocks in the beginning of avi files.
Original commit message from CVS:
* gst/multifile/gstmultifilesrc.c: (gst_multi_file_src_class_init),
(gst_multi_file_src_query):
Implement DEFAULT and BUFFER position queries. See #555260.
Original commit message from CVS:
* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_class_init),
(gst_rtp_amr_depay_process):
Mark DISCONT on output buffers when the marker bit signals a new talk
spurt.
* gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_handle_buffer):
Set the marker bit for buffers with a DISCONT flag to signal a talk
spurt.
Original commit message from CVS:
* gst/videomixer/videomixer.c: (gst_videomixer_fill_queues),
(gst_videomixer_sink_event):
Handle segments a little better. Fixes#537361.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_methods):
Don't assume the server supports PAUSE by default. Fixes#551048.
Original commit message from CVS:
* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init),
(gst_udpsrc_set_uri), (gst_udpsrc_start):
Switch on the socket family to get the addrlen size right.
Original commit message from CVS:
Patch by: Daniel Franke <df at dfranke dot us>
* gst/udp/gstudpsrc.c: (gst_udpsrc_create), (gst_udpsrc_start):
OS X's bind() implementation is picky about its addrlen parameter and
fails with EINVAL if it is larger than expected for the socket's address
family. Set the length to the expected length instead. Fixes#553191.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_open):
Handle the case where we cannot do desribe or when the describe result
does not contain a valid SDP message.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_state_header),
(gst_qtdemux_chain):
Some 'broken' files out there have atom lengths of zero...
which basically results in qtdemux consuming that atom again and again
until the *end of night* !
Detect that and emits an adequate element error message.
Original commit message from CVS:
* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_init),
(gst_rtp_mp4g_depay_finalize), (gst_rtp_mp4g_depay_setcaps),
(gst_rtp_mp4g_depay_clear_queue), (gst_rtp_mp4g_depay_flush_queue),
(gst_rtp_mp4g_depay_queue), (gst_rtp_mp4g_depay_process),
(gst_rtp_mp4g_depay_change_state):
* gst/rtp/gstrtpmp4gdepay.h:
Handle interleaved streams by reordering AU in a queue.
Original commit message from CVS:
* gst/rtp/gstrtpmp4gdepay.c: (gst_bs_parse_init),
(gst_bs_parse_read), (gst_rtp_mp4g_depay_process):
Change some of the ranges in the caps, mostly for the amount of bits we
can use.
Added a little bitstream parse and use it to parse the AU header fields.
Check for malformed and wrongly sized packets better.
Implement more header field parsing.
Handle the size of fragmented packets correctly.
Original commit message from CVS:
Patch by: Jonathan Matthew <notverysmart@gmail.com>
* gst/qtdemux/qtdemux.c: (qtdemux_video_caps):
Add mapping for 'tiff' => image/tiff
Fixes#552213
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state),
(gst_qtdemux_loop_state_header), (qtdemux_parse_node),
(qtdemux_parse_trak), (qtdemux_video_caps):
* gst/qtdemux/qtdemux.h:
* gst/qtdemux/qtdemux_fourcc.h:
* gst/qtdemux/qtdemux_types.c:
Add support for video/mj2 mime-type and its additional atoms/boxes.
Fixes#550646.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_audio_caps):
Add mapping for IMA Loki SDL MJPEG ADPCM codec.
Add some alternative byteswapped mappings that seem to pop up sometimes.
Fixes#550288.
Original commit message from CVS:
Patch by: Mersad Jelacic <mersad at axis dot com>
* gst/multipart/multipartdemux.c:
* gst/multipart/multipartmux.c: (gst_multipart_mux_get_mime):
Convert audio/x-adpcm to and from the audio/G726-X in the muxer and
demuxer. Fixes#549551.
Original commit message from CVS:
* gst/icydemux/gsticydemux.c:
Small docs fix: in the example pipeline, we need to pass
iradio-mode=true to the source, so the server actually sends
an ICY stream.
Original commit message from CVS:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_send_event),
(gst_matroska_demux_video_caps), (gst_matroska_demux_audio_caps):
* gst/matroska/matroska-mux.c:
(gst_matroska_mux_video_pad_setcaps),
(gst_matroska_mux_audio_pad_setcaps), (gst_matroska_mux_finish):
Add Real[Audio|Video] support to Matroska containers.
It works fine for:
* decoding real audio/video streams contained in mkv
* 'transmuxing' real (.rm) files into .mkv files
It will not work though for encoding real[audio/video] streams that
don't contain the 'mdpr_data' extra data on the caps.
The reason why this will not work is because I never intended to
duplicate virtually all the 'mdpr' block creation into mkvmux.
Fixes#536067
Original commit message from CVS:
* gst/law/alaw-encode.c: (gst_alaw_enc_init), (gst_alaw_enc_chain):
* gst/law/mulaw-conversion.c:
* gst/law/mulaw-encode.c: (gst_mulawenc_init),
(gst_mulawenc_chain):
The encoder can't really renegotiate at the time they perform a
pad-alloc so make the srcpads use fixed caps.
Check the buffer size after a pad-alloc because the returned size might
not be right when the downstream element does not know the size of the
new buffer (capsfilter). Fixes#549073.
Original commit message from CVS:
* gst/autodetect/Makefile.am:
Don't link the autodetect plugin with GConf as it doesn't
use GConf. Fixes bug #545463.
Original commit message from CVS:
* gst/matroska/ebml-read.c: (gst_ebml_read_element_id),
(gst_ebml_read_element_length), (gst_ebml_read_uint),
(gst_ebml_read_sint), (gst_ebml_read_float),
(gst_ebml_read_header):
Change some GST_ELEMENT_ERRORs to GST_ERROR_OBJECT to make it
possible to ignore errors and not post any ERROR messages on
the bus.
* gst/matroska/matroska-demux.c:
(gst_matroska_demux_parse_contents):
Ignore any errors and not just EOS when parsing the contents of
a SeekHead. Errors here are usually caused by truncated files
and playback of the file works fine. Fixes playback of the
audio_only_chapter_seekbroken.mka file from the MPlayer samples
archive.
Original commit message from CVS:
* gst/multipart/multipartdemux.c:
* gst/multipart/multipartmux.c:
Conform to RFC2046. audio/basic is mulaw 8000Hz mono.
Original commit message from CVS:
* gst/wavenc/gstwavenc.c: (gst_wavenc_chain):
Revert the last commit. wavenc still supports width!=depth for 32 bit
width. Thanks Tim.
Original commit message from CVS:
* gst/matroska/matroska-demux.c:
(gst_matroska_demux_parse_blockgroup_or_simpleblock):
If the duration of a block is unknown only use the timestamp for the
first lace and use GST_CLOCK_TIME_NONE as duration for the following
laces. Otherwise every lace has the same timestamp which leads to
various problems. Really fixes bug #548831.
Original commit message from CVS:
* gst/wavenc/gstwavenc.c: (gst_wavenc_chain):
If we're not allowing width!=depth in wavenc we should also disable
the code that was added to support width!=depth.
Original commit message from CVS:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream):
Don't calculate the default duration of a frame from the audio sampling
rate. This only works for raw audio if every frame contains a single
sample and results in broken buffer durations for other formats
if no specified default duration is given or the blocks have no
duration. Fixes bug #548831.
Original commit message from CVS:
* gst/matroska/matroska-demux.c:
(gst_matroska_demux_parse_blockgroup_or_simpleblock):
Allow zero sized blocks instead of returning GST_FLOW_OK. Such blocks
are used for text/plain subtitles as a gap-filler in some files.
Original commit message from CVS:
* gst/rtsp/gstrtspgoogle.c:
Things that can happen when your brain is in google mode trying to
deal with their google rtsp server extensions and trying to type your
google mail account.
Original commit message from CVS:
* gst/rtsp/Makefile.am:
* gst/rtsp/gstrtsp.c: (plugin_init):
* gst/rtsp/gstrtspgoogle.c: (gst_rtsp_google_before_send),
(gst_rtsp_google_after_send), (gst_rtsp_google_get_transports),
(_do_init), (gst_rtsp_google_base_init),
(gst_rtsp_google_class_init), (gst_rtsp_google_init),
(gst_rtsp_google_finalize), (gst_rtsp_google_change_state),
(gst_rtsp_google_extension_init):
* gst/rtsp/gstrtspgoogle.h:
Add google RTSP extension, it can only handle udp and responds with
unsupported if we do anything else. Fixes#546465.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_connection_send),
(gst_rtspsrc_connection_receive), (gst_rtspsrc_loop_send_cmd),
(gst_rtspsrc_create_transports_string),
(gst_rtspsrc_setup_streams), (gst_rtspsrc_open),
(gst_rtspsrc_close), (gst_rtspsrc_pause):
Make transport setup code a bit better using GString.
Add some more debug.
Check for closed connections before doing anything on them.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_udp_sink):
Don't try to configure RTCP back to the server when the server did not
give us a valid port number.
Original commit message from CVS:
* gst/videobox/gstvideobox.c: (gst_video_box_set_property):
Use new basetransform method to renegotiate. Fixes#544956.
* tests/icles/Makefile.am:
* tests/icles/videobox-test.c: (make_pipeline), (main):
Add videobox renegotiation example.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_read_subindexes_pull),
(gst_avi_demux_read_subindexes_push):
Some AVI 2.0 (ODML) files don't respect the 'specifications' completely
and instead of using the 'ix##' nomenclature, use '##ix'.
They're still valid though, this fixes the duration and indexes for
virtually all the ODML files I have.
Original commit message from CVS:
* gst/level/gstlevel.c: (gst_level_message_new):
Fix compilation (also known as the classic 'fix code that someone
committed without compiling it first').
Original commit message from CVS:
* gst/level/gstlevel.c:
Little renaming (l -> level).
* gst/spectrum/gstspectrum.c:
* gst/spectrum/gstspectrum.h:
Also send full timestamp/duration details here.
Original commit message from CVS:
* gst/level/gstlevel.c:
* gst/level/gstlevel.h:
Send same timestamp/duration details as videoanalysis. This gives
applications better chance to sync analysis results with playback.
Original commit message from CVS:
* gst/matroska/matroska-mux.c:
(gst_matroska_mux_handle_sink_event),
(flac_streamheader_to_codecdata):
We need to drop one additional buffer for FLAC as the fLaC
marker and STREAMINFO block are merged into one buffer in the caps.
Also don't pretend to support NEWSEGMENT events, otherwise we
will most probably write some invalid data.
Original commit message from CVS:
* gst/matroska/matroska-mux.c: (flac_streamheader_to_codecdata),
(gst_matroska_mux_audio_pad_setcaps):
Add support for muxing FLAC into Matroska containers.
Fixes bug #311586.
Original commit message from CVS:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_reset),
(gst_matroska_demux_element_send_event),
(gst_matroska_demux_handle_seek_event), (gst_matroska_demux_loop):
* gst/matroska/matroska-demux.h:
Close the current segment if we're doing a non-flushing seek and send
the close-segment and the new segment of the seek from the streaming
thread.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c:
Use audio/x-qdm for caps. Collect some info - mplayer has a decoder
for it but ffmpeg does not.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c:
Handle the acid chunk and send tempo as part of tags. Other fields are
interesting too, but need more tag-definitions. Fixes#545433.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c:
Refactor wavparse. Call _reset() from dispose() and move old code from
dispose into reset. This way we don't leak taglists when we abort
parsing. Fix some comments. Move code for skipping a chunk into extra
function. Replace chunk sizes with a const to ease readability.
Original commit message from CVS:
* gst/rtsp/URLS:
Add another URL.
* tests/check/elements/id3v2mux.c: (test_taglib_id3mux_with_tags):
* tests/check/elements/rglimiter.c: (GST_START_TEST):
Add some more debug info.
Original commit message from CVS:
* gst/avi/gstavimux.c: (gst_avi_mux_riff_get_avi_header):
Provide cbSize field for audio extra_data size, and take care to
pad extra_data.
Original commit message from CVS:
* gst/matroska/matroska-demux.c: (gst_matroskademux_do_index_seek),
(gst_matroska_demux_element_send_event),
(gst_matroska_demux_handle_seek_event),
(gst_matroska_demux_handle_src_event):
When receiving a SEEK event on a specific pad first search for a seek
table entry for the stream of the pad and then fall back to an entry
for a different stream.
Original commit message from CVS:
* configure.ac:
* gst/matroska/matroska-ids.c: (gst_matroska_register_tags):
* gst/matroska/matroska-ids.h:
Build depend on core CVS for the attachment tag.
Original commit message from CVS:
* configure.ac:
* gst/matroska/Makefile.am:
* gst/matroska/lzo.c: (get_byte), (get_len), (copy),
(copy_backptr), (lzo1x_decode), (main):
* gst/matroska/lzo.h:
* gst/matroska/matroska-demux.c:
(gst_matroska_demux_read_track_encoding),
(gst_matroska_decompress_data), (gst_matroska_decode_data),
(gst_matroska_decode_buffer),
(gst_matroska_decode_content_encodings),
(gst_matroska_demux_read_track_encodings),
(gst_matroska_demux_add_stream),
(gst_matroska_demux_parse_blockgroup_or_simpleblock):
* gst/matroska/matroska-ids.h:
Decode the codec private data and following ContentEncoding if
necessary.
Support bzip2, lzo and header stripped compression. For lzo use the
ffmpeg lzo implementation as liblzo is GPL licensed.
Fix zlib decompression.
Original commit message from CVS:
* gst/matroska/matroska-mux.c:
(gst_matroska_mux_audio_pad_setcaps):
Fix muxing of MP3/MP2 with different MPEG versions by calculating the
duration of a frame with the new mpegaudioversion caps field.
Original commit message from CVS:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_finalize),
(gst_matroska_demux_class_init), (gst_matroska_demux_init),
(gst_matroska_demux_combine_flows), (gst_matroska_demux_reset),
(gst_matroska_demux_stream_from_num),
(gst_matroska_demux_tracknumber_unique),
(gst_matroska_demux_add_stream), (gst_matroska_demux_send_event),
(gst_matroska_demux_handle_seek_event),
(gst_matroska_demux_sync_streams),
(gst_matroska_demux_parse_blockgroup_or_simpleblock),
(gst_matroska_demux_loop):
* gst/matroska/matroska-demux.h:
Allow an infinite number of stream inside Matroska containers and use
a GPtrArray for storing them instead of allowing "only" 127 streams.
Original commit message from CVS:
* gst/matroska/matroska-demux.c:
(gst_matroska_demux_loop_stream_parse_id):
If no Tracks are found error out instead of trying it again until the
end of time.
Original commit message from CVS:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_audio_caps):
Fix demuxing of raw integer audio. The samples are unsigned only for 8
bit and signed otherwise, not the other way around.
Original commit message from CVS:
* gst/matroska/matroska-mux.c:
(gst_matroska_mux_audio_pad_setcaps):
Add support for muxing raw float audio now that the spec defines the
endianness and add support for muxing raw integer audio with 24 and
32 bits.
Allow muxing of more than 8 audio channels.
Original commit message from CVS:
* gst/matroska/matroska-mux.c: (gst_matroska_mux_create_uid),
(gst_matroska_mux_reset), (gst_matroska_mux_start):
Add locking to the global array of used track UIDs to prevent random
crashes if more than a single matrosmux instance is used.
Use 64 bit values for the track UIDs.
Use the global GRandom of GLib instead of creating our own one
for the few random numbers we need every single time.
Original commit message from CVS:
* gst/goom/convolve_fx.c:
* gst/goom/filters.c:
* gst/goom/goom_config.h:
* gst/goom/goom_core.c:
* gst/goom/goom_tools.h:
Fix build with MSVC: include glib.h to define inline appropriately,
use header guards where needed.
* gst/udp/gstudpnetutils.c:
* gst/udp/gstudpsrc.c:
Fix build with MSVC: use WSA* constants/functions where appropriate, use
g_snprintf rather than snprintf.
Fixes#544433.
Original commit message from CVS:
* gst/debug/gsttaginject.c:
* gst/debug/gsttaginject.h:
Sent tags in _transform_ip() instead of _start(). Fixes#543404
partially.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_add_stream),
(qtdemux_parse_trak):
Revert ISO base media spec based pixel-aspect-ratio calculation.
Fixes#543300.
Original commit message from CVS:
* gst/udp/gstudpnetutils.c:
EAI_ADDRFAMILY was obsoleted in BSD at some point. Define it to the
old value (1) if it's not defined which should not cause any problems
as we're using it internal only anyway.
Original commit message from CVS:
Patch by: Alessandro Decina <alessandro at nnva dot org>
* gst/avi/gstavidemux.c: (gst_avi_demux_riff_parse_vprp):
Fix build of avidemux on big endian architectures.
Original commit message from CVS:
Patch by: Thiago Sousa Santos <thiagoss at lcc dot ufcg dot edu dot br>
* gst/qtdemux/qtdemux.c: (qtdemux_audio_caps):
Correctly distinguish 8bit vs 16bit raw audio. Fixes#542410.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_add_stream),
(qtdemux_parse_trak):
Set pixel-aspect-ratio in caps using display width and height
provided in track.
Original commit message from CVS:
* gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_init),
(gst_rtp_speex_depay_process):
* gst/rtp/gstrtpspeexdepay.h:
Revert last change: Only the jitterbuffer is able to convert RTP to
Gstreamer timestamps and normal (de)payloaders should simply copy it.
Reopens bug #541787.
Original commit message from CVS:
* gst/rtp/gstrtpvrawdepay.c:
Include stdlib.h for atoi().
* gst/rtsp/gstrtspsrc.c:
Use floating point math for latencies < 0 sec in log output.
Original commit message from CVS:
Patch by: Tomasz Grobelny <tomasz at grobelny dot oswiecenia dot net>
* gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_init),
(gst_rtp_speex_depay_process):
* gst/rtp/gstrtpspeexdepay.h:
Take timestamp from the RTP packet as a first step to fix problems
with transmission over RTP when the network is not reliable.
Fixes bug #541787.
Original commit message from CVS:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_class_init),
(gst_matroska_demux_add_stream), (gst_matroska_demux_query),
(gst_matroska_demux_element_query),
(gst_matroska_demux_handle_src_query),
(gst_matroska_demux_handle_seek_event):
Handle position and duration query in DEFAULT format if the
pad's track has a default frame duration set.
Fix seeking now that the segment's duration doesn't contain the
(possibly wrong or inaccurate) duration of the Matroska file.
Original commit message from CVS:
* gst/matroska/ebml-read.c: (_ext2dbl):
Use NAN constant instead of 0.0/0.0 if possible. NAN is defined
in math.h except on MSVC where it is defined in xmath.h.
Fixes compilation with MSVC.
Original commit message from CVS:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_reset),
(gst_matroska_demux_handle_src_query),
(gst_matroska_demux_parse_info),
(gst_matroska_demux_loop_stream_parse_id):
* gst/matroska/matroska-demux.h:
Don't set the segment duration to the duration from the Matroska
header as this value could be wrong and is just informational.
Original commit message from CVS:
* gst/matroska/matroska-demux.c:
(gst_matroska_demux_loop_stream_parse_id):
If no Tracks element is found until the first Cluster is found
search it and error out if none is found in the complete file.
Original commit message from CVS:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_sync_streams):
Resync non-subtitle tracks too if a too large gap compared to other
tracks is detected.
Original commit message from CVS:
* gst/avi/gstavimux.c: (gst_avi_mux_riff_get_avi_header):
* gst/avi/gstavimux.h:
Add 8 bytes to current streamheader to make for a complete one
and to make more players happy. Fixes#519460.
Original commit message from CVS:
* gst/udp/gstudpsrc.c: (gst_udpsrc_start):
Call getsockname() after the call to bind() to get updated values
for the port, etc. This fixes the usage of udpsrc on anonymous
binding and it's usage by rtspsrc. Fixes bugs #539372, #539548.
Thanks to Aurelien Grimaud for pointing out the obvious fix.
Original commit message from CVS:
2008-06-23 Julien Moutte <julien@fluendo.com>
* gst/matroska/matroska-demux.c:
(gst_matroska_demux_read_track_encoding),
(gst_matroska_demux_parse_blockgroup_or_simpleblock): Fix buggy
format strings in macros. (makes it build on OS X again...)
Original commit message from CVS:
* gst/rtp/gstrtpg726pay.c: (gst_rtp_g726_pay_setcaps):
No need to check for audio/G723 and audio/32KADPCM here as they are
no longer supported.
Original commit message from CVS:
* gst/matroska/matroska-demux.c:
(gst_matroska_demux_add_wvpk_header),
(gst_matroska_demux_audio_caps):
* gst/matroska/matroska-ids.h:
Fix demuxing of WavPack files. Muxing is still broken.
Original commit message from CVS:
* gst/matroska/matroska-demux.c: (gst_matroska_track_free),
(gst_matroska_demux_add_mpeg_seq_header),
(gst_matroska_demux_add_wvpk_header),
(gst_matroska_demux_check_subtitle_buffer),
(gst_matroska_demux_parse_blockgroup_or_simpleblock),
(gst_matroska_demux_video_caps), (gst_matroska_demux_audio_caps),
(gst_matroska_demux_subtitle_caps):
* gst/matroska/matroska-ids.h:
Add a "vfunc" to the track context for postprocessing frames and
convert the wavpack and subtitle postprocessing to this vfunc.
Copy buffer flags in those functions to the new buffers too.
Parse CodecState elements of Blocks.
Add a postprocessing function for MPEG video that adds the sequence
header from the codec private data or codec state to the frames if
it's not already there.
Original commit message from CVS:
* gst/matroska/matroska-demux.c:
(gst_matroska_demux_parse_blockgroup_or_simpleblock):
If a gap of more than 1/2 second is found in one stream send a
NEWSEGMENT event to not stall the pipeline if the gap is too large.
This also fixes Matroska files where the first buffer doesn't start
at timestamp 0. Fixes bug #429322.
The duration of a block is the default duration multiplied with the
number of laces. Every lace is one frame and the default duration
is the duration of one frame. This fixes playback of files that use
lacing for some tracks.
Original commit message from CVS:
* gst/matroska/matroska-demux.c:
(gst_matroska_demux_parse_contents_seekentry):
Update FIXME/TODOs and only ignore EOS at the central, important place
instead of several places.
Original commit message from CVS:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_reset),
(gst_matroska_demux_stream_from_num),
(gst_matroska_demux_encoding_cmp),
(gst_matroska_demux_encoding_order_unique),
(gst_matroska_demux_read_track_encoding),
(gst_matroska_demux_read_track_encodings),
(gst_matroska_demux_tracknumber_unique),
(gst_matroska_demux_add_stream), (gst_matroska_demux_init_stream),
(gst_matroska_demux_parse_tracks),
(gst_matroska_demux_parse_index_cuetrack),
(gst_matroska_demux_parse_index_pointentry),
(gst_matroska_demux_parse_index), (gst_matroska_demux_parse_info),
(gst_matroska_demux_parse_metadata_id_simple_tag),
(gst_matroska_demux_parse_metadata_id_tag),
(gst_matroska_demux_parse_metadata),
(gst_matroska_demux_parse_attached_file),
(gst_matroska_demux_parse_attachments),
(gst_matroska_demux_parse_chapters),
(gst_matroska_demux_sync_streams), (gst_matroska_decode_buffer),
(gst_matroska_demux_parse_blockgroup_or_simpleblock),
(gst_matroska_demux_parse_cluster),
(gst_matroska_demux_parse_contents_seekentry),
(gst_matroska_demux_parse_contents),
(gst_matroska_demux_loop_stream_parse_id),
(gst_matroska_demux_loop):
Improve debug output everywhere and fix the EOS logic.
Check the values of the ContentEncoding elements more strictly and
don't use tracks for which it's invalid.
Check that the track number is unique for this stream.
Check that seek positions are below G_MAXINT64 as our seeks are
int64-based and overflows will fail badly.
After seeks also don't push SimpleBlocks until the first one
containing a keyframe is found. Before this was done only for normal
Blocks.
Update some FIXME/TODOs.
* gst/matroska/ebml-read.c: (gst_ebml_read_peek_bytes),
(gst_ebml_read_utf8), (gst_ebml_read_header):
Improve debug output.
* gst/matroska/matroska-ids.c:
(gst_matroska_track_init_video_context):
* gst/matroska/matroska-ids.h:
* gst/matroska/matroska-mux.c:
(gst_matroska_mux_video_pad_setcaps):
Remove eye mode and don't parse it anymore. We can't use that
information in GStreamer yet so it's useless.
Original commit message from CVS:
* gst/rtsp/URLS:
Some more urls.
* gst/smpte/barboxwipes.c:
Add a comment
* tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh:
Fix typo, add audioresample to the pipeline.
Original commit message from CVS:
* gst/matroska/matroska-demux.c: (gst_matroska_index_compare):
When comparing index elements with the same time compare their
block number.
Original commit message from CVS:
* gst/matroska/matroska-demux.c:
(gst_matroska_demux_parse_attached_file)
Init variable to NULL to avoid compiler warning.
Original commit message from CVS:
* gst/matroska/Makefile.am:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_reset),
(gst_matroska_demux_parse_attached_file),
(gst_matroska_demux_parse_attachments),
(gst_matroska_demux_parse_contents_seekentry),
(gst_matroska_demux_loop_stream_parse_id):
* gst/matroska/matroska-demux.h:
* gst/matroska/matroska-ids.c: (gst_matroska_register_tags):
* gst/matroska/matroska-ids.h:
* gst/matroska/matroska.c: (plugin_init):
Parse Attachments and post them as GST_TAG_IMAGE if we detect
it as image and otherwise as GST_TAG_ATTACHMENT. Include filename
and description of the attachments in the caps. Fixes bug #537622.
Original commit message from CVS:
* gst/matroska/ebml-read.c: (gst_ebml_read_peek_bytes):
Return GST_FLOW_UNEXPECTED instead of GST_FLOW_ERROR on short reads.
If we get less bytes than requested we can't do anything except doing
our EOS logic.
Original commit message from CVS:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_reset),
(gst_matroskademux_do_index_seek),
(gst_matroska_demux_parse_index_cuetrack),
(gst_matroska_demux_parse_index_pointentry),
(gst_matroska_index_compare), (gst_matroska_demux_parse_index),
(gst_matroska_demux_parse_metadata):
* gst/matroska/matroska-demux.h:
* gst/matroska/matroska-ids.h:
Use a GArray for storing the Cue (i.e. seek) information, store
the CueTrackPositions for every track, store the block number
and optimize searching in the array by sorting it after the last
element was added.
Fix a small memory leak when trying to parse a tags element that was
already parsed.
Original commit message from CVS:
* gst/matroska/matroska-mux.c: (gst_matroska_mux_reset),
(gst_matroska_mux_start), (gst_matroska_mux_finish),
(gst_matroska_mux_write_data):
* gst/matroska/matroska-mux.h:
Don't write another SeekHead which indexes all Clusters to the end of
the file. This isn't useful for anything and just increases filesize.
Original commit message from CVS:
* gst/matroska/ebml-read.c:
* gst/matroska/ebml-read.h:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_reset),
(gst_matroska_demux_parse_metadata):
* gst/matroska/matroska-demux.h:
Make sure that every Tags element is only parsed once and it's
containing tags are only posted once.
Original commit message from CVS:
* gst/matroska/ebml-read.c: (gst_ebml_peek_id),
(gst_ebml_read_header):
* gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream),
(gst_matroska_demux_parse_tracks),
(gst_matroska_demux_parse_index_cuetrack),
(gst_matroska_demux_parse_index_pointentry),
(gst_matroska_demux_parse_index), (gst_matroska_demux_parse_info),
(gst_matroska_demux_parse_metadata_id_simple_tag),
(gst_matroska_demux_parse_metadata_id_tag),
(gst_matroska_demux_parse_metadata),
(gst_matroska_demux_parse_attachments),
(gst_matroska_demux_parse_chapters),
(gst_matroska_demux_parse_blockgroup_or_simpleblock),
(gst_matroska_demux_parse_cluster),
(gst_matroska_demux_parse_contents_seekentry),
(gst_matroska_demux_parse_contents),
(gst_matroska_demux_loop_stream_parse_id):
Handle EBML elements like Void or CRC32 in the EbmlRead base class
already. They're not useful in the matroska parser and only cause
additional code.
Original commit message from CVS:
* gst/matroska/ebml-read.c: (gst_ebml_level_free),
(gst_ebml_finalize), (gst_ebml_read_change_state),
(gst_ebml_read_element_level_up), (gst_ebml_read_master):
* gst/matroska/matroska-demux.c:
(gst_matroska_demux_parse_contents_seekentry):
Reverse the level list as we usually are only interested in the
first element or want to add a new first element. Having the
first element stored at the end and calling g_list_last() and
g_list_append() is more expensive.
Also use GSlice for allocating the GstEbmlLevel structs.
Original commit message from CVS:
* gst/debug/gsttaginject.c: (gst_tag_inject_finalize),
(gst_tag_inject_class_init), (gst_tag_inject_init):
Don't unref NULL taglist in finalize. Don't use c++ style
comments.
Original commit message from CVS:
* gst/matroska/matroska-demux.c:
(gst_matroska_demux_parse_metadata_id_simple_tag):
* gst/matroska/matroska-mux.c: (gst_matroska_mux_write_simple_tag),
(gst_matroska_mux_write_data):
Use gst_value_serialize() and gst_value_deserialize() for transforming
tags from some GType to a string and the other way around. The default
transformations in GLib don't include transformations from string to
number types.
Original commit message from CVS:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_reset),
(gst_matroska_demux_parse_tracks),
(gst_matroska_demux_parse_index), (gst_matroska_demux_parse_info),
(gst_matroska_demux_parse_attachments),
(gst_matroska_demux_parse_chapters),
(gst_matroska_demux_parse_contents_seekentry),
(gst_matroska_demux_loop_stream_parse_id):
* gst/matroska/matroska-demux.h:
Only parse Tracks, SeekHead and SegmentInfo elements once but allow
Tags multiple times. The first ones can appear more than once but must
contain the same content as the first for backup purposes so we ignore
all but the first one. Tags can appear multiple times with different
content.
Jump to all elements except Clusters that are available from a
SeekHead to make it more likely to have all required informations
before getting to the first Clusters.
Add dummy functions for parsing Attachments and Chapters.
Original commit message from CVS:
* 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:
Add property to control automatic join/leave of multicast groups.
Add G_LIKELY.
Remove setting caps on buffers explicitly, basesrc does that for us now.
Improve debug info.
Convert some non-fatal error into warnings.
Use g_ntohs for better portability.
Leave multicast groups when stopping.
When using external sockets, use getsockname() on them to fill up the
addr structure before calling methods that use the structure.
Should all fix#536903.
API: GstUDPSrc::auto-multicast property
Original commit message from CVS:
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_init_send),
(gst_multiudpsink_remove):
Fix a typo and do some small cleanups.
Original commit message from CVS:
Patch by: Olivier Crete <tester at tester dot ca>
* gst/rtp/gstrtptheoradepay.c: (gst_rtp_theora_depay_setcaps):
Make the delivery-method mandatory on the caps and only accept inline
for now.
Reverse strcmp checks for delivery-method.
* gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_setcaps):
Make delivery method optional when parsing caps and note this in the
caps.
Reverse strcmp checks for delivery-method.
* gst/rtp/gstrtpvorbispay.c:
Update a comment to note that the delivery-method is optional,
Fixes#537675.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_mcast):
Set udpsrc for receiving data from multicast groups to PAUSED instead of
leaving them in READY. Fixes#537832.
Original commit message from CVS:
* gst/avi/gstavimux.c:
Simplify code. gst_tag_list_merge() does the NULL checks. Add a FIXME
for a random constant in tagmuxing code.
Original commit message from CVS:
* gst/matroska/matroska-mux.c: (gst_matroska_mux_request_new_pad),
(gst_matroska_mux_release_pad), (gst_matroska_mux_write_data):
Update the counter for the number of streams when pads are added or
removed. This will make sure that a seek table is generated for
files with just one audio stream.
Original commit message from CVS:
* gst/matroska/matroska-demux.c:
(gst_matroska_demux_parse_metadata_id_simple_tag):
* gst/matroska/matroska-ids.h:
* gst/matroska/matroska-mux.c: (gst_matroska_mux_write_simple_tag):
Add some more tags, improve debugging a bit and make sure that
GValue transformation has succeeded before using the result
as a tag.
Original commit message from CVS:
Patch by: Olivier Crete <tester at tester dot ca>
* gst/rtp/gstrtptheorapay.c:
The Theora RTP payloader only supports the "inline" delievery method
so let's declare this on the caps of the static pad template.
Fixes bug #537675.
Original commit message from CVS:
* gst/videomixer/videomixer.c: (gst_videomixer_fill_queues),
(gst_videomixer_blend_buffers):
Use stream_time to synchronize the object properties.
Use running_time of the master pad to timestamp outgoing buffers.
Fix the initial segment event to extend an unknown amount of time.
Fixes#537361.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
(gst_avi_demux_parse_index), (gst_avi_demux_massage_index),
(gst_avi_demux_calculate_durations_from_index),
(gst_avi_demux_stream_header_push),
(gst_avi_demux_stream_header_pull):
Try to ignore unparsable/unknown streams and give a warning instead of
erroring out. Fixes#537377.
Original commit message from CVS:
* gst/matroska/ebml-write.c: (gst_ebml_write_float):
Use GDOUBLE_TO_BE() instead of (probably slower) custom code.
* gst/matroska/matroska-demux.c: (gst_matroska_demux_base_init),
(gst_matroska_demux_class_init), (gst_matroska_demux_init),
(gst_matroska_track_free), (gst_matroska_demux_encoding_cmp),
(gst_matroska_demux_read_track_encodings),
(gst_matroska_demux_add_stream),
(gst_matroska_demux_handle_src_query),
(gst_matroska_demux_init_stream),
(gst_matroska_demux_parse_index_cuetrack),
(gst_matroska_demux_parse_index_pointentry),
(gst_matroska_demux_parse_info),
(gst_matroska_demux_parse_metadata_id_simple_tag),
(gst_matroska_demux_parse_metadata),
(gst_matroska_demux_add_wvpk_header), (gst_matroska_decode_buffer),
(gst_matroska_demux_parse_blockgroup_or_simpleblock),
(gst_matroska_demux_parse_cluster),
(gst_matroska_demux_parse_contents_seekentry),
(gst_matroska_demux_loop_stream_parse_id),
(gst_matroska_demux_loop), (gst_matroska_demux_video_caps),
(gst_matroska_demux_audio_caps),
(gst_matroska_demux_subtitle_caps):
* gst/matroska/matroska-demux.h:
* gst/matroska/matroska-ids.c:
(gst_matroska_track_init_subtitle_context):
* gst/matroska/matroska-ids.h:
* gst/matroska/matroska-mux.c: (gst_matroska_mux_base_init),
(gst_matroska_mux_class_init), (gst_matroska_mux_init),
(gst_matroska_mux_create_uid), (gst_matroska_mux_reset),
(gst_matroska_mux_video_pad_setcaps),
(gst_matroska_mux_audio_pad_setcaps),
(gst_matroska_mux_subtitle_pad_setcaps),
(gst_matroska_mux_request_new_pad),
(gst_matroska_mux_track_header), (gst_matroska_mux_start),
(gst_matroska_mux_write_simple_tag), (gst_matroska_mux_finish),
(gst_matroska_mux_write_data), (gst_matroska_mux_collected),
(gst_matroska_mux_set_property):
Add many FIXMEs/TODOs all over the matroska muxer and demuxer
elements, do some checks for valid values in the demuxer, handle
tracktimecodescale in the demuxer, set correct default values for all
settings in the demuxer, review and add all missing matroska
IDs and some more raw YUV formats, and some trivial cleanup.
Original commit message from CVS:
* gst/interleave/deinterleave.c: (gst_deinterleave_add_new_pads),
(gst_deinterleave_src_query):
* gst/interleave/interleave.c: (gst_interleave_src_query_duration),
(gst_interleave_src_query):
Properly implement duration and position queries in bytes format. We
have to take the upstream reply and divide/multiply it by the number
of channels to get the correct result.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_udp_sink):
Use the new gst_rtsp_connection_get_ip() to access the IP address
of a GstRTSPConnection since it is a private member.
Original commit message from CVS:
* ext/flac/gstflacdec.c: (gst_flac_extract_picture_buffer):
* gst/id3demux/id3v2frames.c: (parse_picture_frame):
Use new utility functions in libgsttag to process coverart (#512333).
Original commit message from CVS:
* gst/matroska/ebml-write.c: (gst_ebml_write_finalize),
(gst_ebml_write_set_cache):
Unref the write cache in finalize if it was set and add add "FIXME"
to a comment that needs it.
Original commit message from CVS:
* gst/interleave/interleave.c: (gst_interleave_pad_get_type),
(gst_interleave_pad_get_property), (gst_interleave_pad_class_init),
(gst_interleave_request_new_pad), (gst_interleave_release_pad):
* gst/interleave/interleave.h:
Use an always increasing integer for the number in the name of the
requested sink pads to guarantuee a unique name. Add a "channel"
property to GstInterleavePad to make it possible for applications
to retrieve the channel number in the output for every pad.
Use g_type_register_static_simple() instead of
g_type_register_static() to save some relocations.
Original commit message from CVS:
* gst/interleave/interleave.c: (gst_interleave_pad_get_type),
(gst_interleave_change_state):
Stop GstCollectPads before calling the parent's state change function
when going from PAUSED to READY as we otherwise deadlock.
Fixes bug #536258.
Original commit message from CVS:
* gst/interleave/interleave.c:
(gst_interleave_check_channel_positions),
(gst_interleave_set_channel_positions),
(gst_interleave_class_init):
Use new gst_audio_check_channel_positions() function and register
the GstInterleavePad type from a threadsafe context.
Original commit message from CVS:
* gst/interleave/interleave.c: (gst_interleave_pad_get_type),
(gst_interleave_finalize), (gst_audio_check_channel_positions),
(gst_interleave_set_channel_positions),
(gst_interleave_class_init), (gst_interleave_init),
(gst_interleave_set_property), (gst_interleave_get_property),
(gst_interleave_request_new_pad), (gst_interleave_release_pad),
(gst_interleave_sink_setcaps), (gst_interleave_src_query_duration),
(gst_interleave_src_query_latency), (gst_interleave_collected):
* gst/interleave/interleave.h:
Allow setting channel positions via a property and allow using the
channel positions on the input as the channel positions of the output.
Fix some broken logic and memory leaks.
* tests/check/Makefile.am:
* tests/check/elements/interleave.c: (src_handoff_float32),
(sink_handoff_float32), (GST_START_TEST), (interleave_suite):
Add unit tests for checking correct handling of channel positions.
Original commit message from CVS:
* gst/videomixer/videomixer.c: (gst_videomixer_query_duration),
(gst_videomixer_query_latency):
When using gst_element_iterate_pads() one has to unref every pad
after usage.
Original commit message from CVS:
2008-05-31 Julien Moutte <julien@fluendo.com>
* gst/qtdemux/qtdemux.c: (gst_qtdemux_find_keyframe),
(gst_qtdemux_find_segment), (gst_qtdemux_perform_seek),
(gst_qtdemux_seek_to_previous_keyframe),
(gst_qtdemux_activate_segment), (gst_qtdemux_loop): Make sure we
we don't clip the segment's stop using the main segment duration
as
that could crop quite some video frames. Make reverse playback
support
more robust and support edit lists. Support seeking to the last
frame,
and fix reverse looping playback. Add some debugging.
* win32/common/config.h: Updated.
Original commit message from CVS:
* gst/equalizer/gstiirequalizer.c:
(gst_iir_equalizer_transform_ip):
Don't clip float/double samples, correctly unset passthrough mode
and use better rounding for integer samples.
Original commit message from CVS:
* gst/equalizer/gstiirequalizer.c:
(gst_iir_equalizer_band_set_property), (gst_iir_equalizer_init),
(setup_filter), (set_passthrough), (update_coefficients),
(gst_iir_equalizer_compute_frequencies),
(gst_iir_equalizer_transform_ip):
* gst/equalizer/gstiirequalizer.h:
Update the filter coefficients only when needed in the transform_ip
function and correctly set the element into passthrough mode if the
gain of all bands is 0.
Original commit message from CVS:
Based on patch by: Sebastian Keller <sebastian-keller at gmx dot de>
* gst/alpha/gstalpha.c: (gst_alpha_class_init), (gst_alpha_init),
(gst_alpha_set_property), (gst_alpha_get_property),
(gst_alpha_chroma_key_ayuv), (gst_alpha_chromakey_row_i420):
Try to skip pixels or areas that are too dark or too bright for us to do
meaningfull color detection.
Added properties to control the sensitivity to light and darkness.
Added some small cleanups. Fixes#512345.
Original commit message from CVS:
* gst/videomixer/videomixer.c: (gst_videomixer_reset),
(gst_videomixer_init), (gst_videomixer_query_duration),
(gst_videomixer_query_latency), (gst_videomixer_query),
(gst_videomixer_blend_buffers):
* gst/videomixer/videomixer.h:
Implement position (in time), duration and latency queries.
Original commit message from CVS:
Patch by: j^ <j at oil21 dot org>
* gst/qtdemux/qtdemux.c: (qtdemux_video_caps):
Add caps for DVCPRO50 and DVCPRO HD PAL/NTSC. See #526481.
Original commit message from CVS:
* gst/interleave/deinterleave.c:
Add another example launch line.
* gst/interleave/interleave.c: (interleave_24),
(gst_interleave_finalize), (gst_interleave_base_init),
(gst_interleave_class_init), (gst_interleave_init),
(gst_interleave_request_new_pad), (gst_interleave_release_pad),
(gst_interleave_change_state), (__remove_channels),
(__set_channels), (gst_interleave_sink_getcaps),
(gst_interleave_set_process_function),
(gst_interleave_sink_setcaps), (gst_interleave_sink_event),
(gst_interleave_src_query_duration), (gst_interleave_src_query),
(forward_event_func), (forward_event), (gst_interleave_src_event),
(gst_interleave_collected):
* gst/interleave/interleave.h:
Major rewrite of interleave using GstCollectpads. This new version
also supports almost all raw audio formats and has better caps
negotiation. Fixes bug #506594.
Also update docs and add some more examples.
* tests/check/elements/interleave.c: (interleave_chain_func),
(GST_START_TEST), (src_handoff_float32), (sink_handoff_float32),
(interleave_suite):
Add some more extensive unit tests for interleave.
Original commit message from CVS:
* gst/interleave/deinterleave.c: (gst_deinterleave_base_init),
(gst_deinterleave_class_init), (gst_deinterleave_init),
(gst_deinterleave_add_new_pads), (gst_deinterleave_sink_getcaps):
* gst/interleave/deinterleave.h:
Don't set a getcaps() function on the src pads as it's not required
and the default getcaps() function returns the correct results for
our src pads.
Complete documentation and add myself to the authors of the element.
Original commit message from CVS:
* gst/udp/Makefile.am:
Add -D_GNU_SOURCE to CFLAGS so we get things like EAI_ADDRFAMILY
when including netdb.h when building against glibc >= 2.8.
Original commit message from CVS:
2008-05-22 Julien Moutte <julien@fluendo.com>
* gst/smpte/gstsmptealpha.c: (gst_smpte_alpha_setcaps): Fix
debug statement arguments.
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_setup_qos_dscp):
* gst/udp/gstudpnetutils.c: (gst_udp_join_group),
(gst_udp_leave_group): Fix IP and IPV6 options to make it work
on more platforms.
Original commit message from CVS:
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_init_send),
(gst_multiudpsink_add_internal):
* gst/udp/gstudpnetutils.c: (gst_udp_set_loop_ttl),
(gst_udp_join_group):
* gst/udp/gstudpnetutils.h:
* gst/udp/gstudpsrc.c: (gst_udpsrc_start):
Joining a multicast group and setting the loop/ttl properties are
totally unrelated tasks are must be separated.
Original commit message from CVS:
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init),
(gst_multiudpsink_setup_qos_dscp), (gst_multiudpsink_add_internal):
* gst/udp/gstmultiudpsink.h:
Add a fixme for the auto-multicast property.
Fix some confusing debug messages.
Disable setting a qos value by default.
Original commit message from CVS:
Patch by: Gustaf Räntilä <g dot rantila at gmail dot com>
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_render):
Ignore EPERM errors from sendto. Fixes#533619.
Original commit message from CVS:
Patch by: Henrik Eriksson <henriken at axis dot com>
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init),
(gst_multiudpsink_init), (gst_multiudpsink_setup_qos_dscp),
(gst_multiudpsink_set_property), (gst_multiudpsink_get_property),
(gst_multiudpsink_init_send), (gst_multiudpsink_add_internal):
* gst/udp/gstmultiudpsink.h:
Add qos-dscp property to manage the Quality of service.
Original commit message from CVS:
Patch by: Bruno Santos <brunof at ua dot pt>
* gst/udp/gstudpnetutils.c: (gst_udp_get_addr),
(gst_udp_join_group), (gst_udp_leave_group),
(gst_udp_is_multicast):
* gst/udp/gstudpnetutils.h:
Provide a bunch of helper methods to deal with IPv4 and IPv6
transparently.
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init),
(gst_multiudpsink_init), (gst_multiudpsink_set_property),
(gst_multiudpsink_get_property), (join_multicast),
(gst_multiudpsink_init_send), (gst_multiudpsink_add_internal),
(gst_multiudpsink_remove):
* gst/udp/gstmultiudpsink.h:
Add multicast TTL and loopback properties.
Use the helper methods to implement ip4 and ip6.
* gst/udp/gstudpsrc.c: (gst_udpsrc_create), (gst_udpsrc_start):
* gst/udp/gstudpsrc.h:
Use the helper methods to implement ip4 and ip6.
Fixes#515962.
Original commit message from CVS:
Patch by: Patrick Radizi <patrick dot radizi at axis dot com>
* gst/multipart/multipartdemux.c: (gst_multipart_demux_class_init),
(gst_multipart_demux_get_gstname),
(gst_multipart_find_pad_by_mime), (gst_multipart_demux_chain):
* gst/multipart/multipartdemux.h:
Don't blindly copy the mime-type as the caps name because they not
always map directly. Instead use a hashtable with common mappings.
Fixes#533287.
Original commit message from CVS:
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_class_init),
(gst_rtp_h264_depay_init), (gst_rtp_h264_depay_set_property),
(gst_rtp_h264_depay_get_property), (gst_rtp_h264_depay_setcaps),
(gst_rtp_h264_depay_process):
* gst/rtp/gstrtph264depay.h:
Add experimental support for outputting quicktime-like AVC output in
addition to the existing bytestream output.
* gst/rtp/gstrtph264pay.c: (gst_h264_scan_mode_get_type),
(gst_rtp_h264_pay_class_init), (gst_rtp_h264_pay_init),
(gst_rtp_h264_pay_setcaps), (gst_rtp_h264_pay_payload_nal),
(gst_rtp_h264_pay_handle_buffer), (gst_rtp_h264_pay_set_property),
(gst_rtp_h264_pay_get_property):
* gst/rtp/gstrtph264pay.h:
Make the parsing mode configurable, for some inputs we don't need to
scan every byte for start codes.
Only set the marker bit on ACCESS units.
Original commit message from CVS:
* gst/equalizer/gstiirequalizer.c:
Use a bigger type in integer mode for the intermediate results to
prevent overflows. This fixes the crippled sound when using the
equalizer in integer mode. Fixes bug #510865.
Original commit message from CVS:
* gst/videomixer/videomixer.c:
* gst/videomixer/videomixer.h:
Instead of a random number for the request pad id's,
use a counter.
Register the videomixerpad class from the element's class_init
where it's safer, and allows the docs generator to scan it.
Original commit message from CVS:
* gst/smpte/Makefile.am:
* gst/smpte/gstsmpte.c: (gst_smpte_plugin_init):
* gst/smpte/gstsmpte.h:
* gst/smpte/gstsmptealpha.c:
(gst_smpte_alpha_transition_type_get_type),
(gst_smpte_alpha_get_type), (gst_smpte_alpha_base_init),
(gst_smpte_alpha_class_init), (gst_smpte_alpha_update_mask),
(gst_smpte_alpha_setcaps), (gst_smpte_alpha_get_unit_size),
(gst_smpte_alpha_init), (gst_smpte_alpha_finalize),
(gst_smpte_alpha_do_ayuv), (gst_smpte_alpha_do_i420),
(gst_smpte_alpha_transform), (gst_smpte_alpha_set_property),
(gst_smpte_alpha_get_property), (gst_smpte_alpha_plugin_init):
* gst/smpte/gstsmptealpha.h:
* gst/smpte/plugin.c: (plugin_init):
Add new plugin that adds the SMPTE transition in the alpha channel of
I420 and AYUV frames so that they can be blended with videomixer later
on. Uses all niceties such as using base transform for efficient alloc
and negotiation. It currently requires GstController to control the
position in the transition effect.
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.args:
* docs/plugins/gst-plugins-good-plugins.hierarchy:
* docs/plugins/gst-plugins-good-plugins.interfaces:
* docs/plugins/gst-plugins-good-plugins.types:
* gst/videomixer/videomixer.c:
Try using thaytans new mechanism to get extra classes into plugin
docs. Aparently works for the Eq. For VideoMixer the GObject stuff is
missing still.
Original commit message from CVS:
* gst/interleave/deinterleave.c: (gst_deinterleave_class_init),
(gst_deinterleave_init), (gst_deinterleave_add_new_pads),
(gst_deinterleave_set_pads_caps), (gst_deinterleave_set_property),
(gst_deinterleave_get_property):
* gst/interleave/deinterleave.h:
Add a property to select whether channel positions should be kept on
the mono output buffers or should be dropped.
Original commit message from CVS:
* gst/avi/gstavimux.c: (gst_avi_mux_audsink_set_caps):
Set proper rate in avi stream header for PCM audio, and also do some
more sanity checks on caps in this case. Fixes#511489.
Original commit message from CVS:
* gst/interleave/deinterleave.c: (gst_deinterleave_finalize),
(gst_deinterleave_init), (gst_deinterleave_sink_event),
(gst_deinterleave_process), (gst_deinterleave_sink_activate_push):
* gst/interleave/deinterleave.h:
Queue events until src pads were added and they can be sent. Otherwise
downstream will never get the first newsegment event.
Original commit message from CVS:
* gst/interleave/deinterleave.c: (gst_deinterleave_sink_setcaps),
(gst_deinterleave_getcaps):
Always set the channel positions when gst_audio_get_channel_positions()
returns something, even if they're not set in the caps. This makes
sure that the output channels can be interleaved again correctly
in the mono/stereo cases too.
Don't ask for the peercaps of the current pad in getcaps() as this
might call getcaps() again and deadlock.
Original commit message from CVS:
* gst/interleave/Makefile.am:
* gst/interleave/deinterleave.c: (deinterleave_24),
(gst_deinterleave_finalize), (gst_deinterleave_base_init),
(gst_deinterleave_class_init), (gst_deinterleave_init),
(gst_deinterleave_add_new_pads), (gst_deinterleave_set_pads_caps),
(gst_deinterleave_set_process_function),
(gst_deinterleave_sink_setcaps), (__remove_channels),
(__set_channels), (gst_deinterleave_getcaps),
(gst_deinterleave_process), (gst_deinterleave_chain),
(gst_deinterleave_sink_activate_push):
* gst/interleave/deinterleave.h:
Add support for all raw audio formats and provide better negotiation
if the caps are changing.
Don't allow changes of the channel positions and set the position of
the corresponding channel on the src pad caps.
General cleanup and smaller bugfixes.
* tests/check/elements/deinterleave.c: (float_buffer_check_probe):
Check the channel positions on the output buffer caps.
Original commit message from CVS:
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_process):
Small comment added.
* gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_class_init),
(gst_rtp_h264_pay_decode_nal), (gst_rtp_h264_pay_parse_sps_pps),
(gst_rtp_h264_pay_payload_nal), (gst_rtp_h264_pay_handle_buffer):
Debug string cleanups (remove trailing \n)
Refactor and clean up the payloader a bit and make sure that we only
put one NAL unit in an RTP packet even if the input buffer contains
multiple NAL units.
Add suport for AVC format input.
Original commit message from CVS:
* gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_class_init),
(gst_rtp_h264_pay_finalize), (gst_rtp_h264_pay_handle_buffer),
(gst_rtp_h264_pay_set_property), (gst_rtp_h264_pay_get_property):
* gst/rtp/gstrtph264pay.h:
Make it possible to specify profile-level-id and sprop-parameter-sets
using properties in case they are not available in-stream.
Original commit message from CVS:
* gst/interleave/Makefile.am:
* gst/interleave/deinterleave.h:
* gst/interleave/interleave.h:
* gst/interleave/plugin.h:
Split definitions into separate header files for better documentation
generation.
* gst/interleave/deinterleave.c: (gst_deinterleave_base_init),
(gst_deinterleave_class_init), (gst_deinterleave_sink_setcaps),
(gst_deinterleave_process):
Don't use alloca, allow caps changes as long as the number of channels
does not change, don't use g_warning, return NOT_NEGOTIATED as early
as possible and some other cleanup.
* gst/interleave/interleave.c: (gst_interleave_base_init),
(gst_interleave_class_init):
Do some random cleanup.
* tests/check/Makefile.am:
* tests/check/elements/deinterleave.c: (GST_START_TEST),
(deinterleave_chain_func), (deinterleave_pad_added),
(deinterleave_suite):
Add unit tests for the deinterleave element.
Original commit message from CVS:
* gst/avi/gstavimux.c: (gst_avi_mux_start_file):
Send an initial BYTE segment to inform downstream of later seeking,
and to forego sync attempts.
Original commit message from CVS:
* gst/matroska/matroska-demux.c:
(gst_matroska_demux_push_dvd_clut_change_event):
Convert subtitle palette info in VobSub private data from VobSub's
(buggy) RGB to YUV.
Original commit message from CVS:
* gst/avi/gstavimux.c: (gst_avi_mux_pad_reset):
Do not leave fourcc stream header field empty upon reset.
Fixes#519301.
Original commit message from CVS:
Based on patch by: Wouter Cloetens <wouter at mind be>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_skip_lws), (gst_rtspsrc_unskip_lws),
(gst_rtspsrc_skip_commas), (gst_rtspsrc_skip_item),
(gst_rtsp_decode_quoted_string),
(gst_rtspsrc_parse_digest_challenge), (gst_rtspsrc_parse_auth_hdr),
(gst_rtspsrc_setup_auth):
Support Digest authentication. Fixes#532065.
Original commit message from CVS:
* gst/level/gstlevel.c:
Also support 32bit (e.g. whe having it after 'mad'). Add more notes
about whats needed for liboil acceleration. Simplify docs a bit.
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/matroska/matroska-mux.c: (gst_matroska_mux_collected):
Update the track duration if the old one was invalid.
Fixes bug #532117.
Original commit message from CVS:
* gst/rtp/gstrtph264pay.c (gst_rtp_h264_pay_parse_sps_pps):
Use GST_STR_NULL when trying to print sps and pps strings that could
be NULL, as this might crash on some platforms.
Original commit message from CVS:
* gst/rtp/gstrtpilbcpay.c:
Added missing stdlib.h include for strtol(), and made include ordering and
style consistent with the corresponding depayloader.
Original commit message from CVS:
* gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_process):
Add some more debug info and guard against small payloads.
* gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_process):
Set duration on outgoing buffers because we can.
Original commit message from CVS:
Patch by: Olivier Crete <tester at tester dot ca>
* gst/rtp/gstrtpspeexpay.c: (gst_rtp_speex_pay_class_init),
(gst_rtp_speex_pay_getcaps):
Add negotiation for the speec channels and rate. See #465146.
Original commit message from CVS:
Patch by: Olivier Crete <tester at tester dot ca>
* gst/rtp/gstrtpilbcpay.c: (gst_rtpilbcpay_class_init),
(gst_rtpilbcpay_sink_setcaps), (gst_rtpilbcpay_sink_getcaps):
Add negotiation for the ILBC mode. See #465146.
Original commit message from CVS:
Patch by: Youness Alaoui <youness.alaoui at collabora co uk>
* gst/udp/gstudpsrc.c: (gst_udpsrc_create):
Don't error out if we get an ICMP destination-unreachable
message when trying to read packets on win32 (#529454).
Original commit message from CVS:
* configure.ac:
* gst/qtdemux/qtdemux.c: (qtdemux_parse_trak):
Use new error code for encrypted streams (which requires core CVS).
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_videosrc_template),
(gst_qtdemux_audiosrc_template):
Fix swapped pad template names, spotted by Thiago Sousa Santos.
Original commit message from CVS:
2008-04-28 Julien Moutte <julien@fluendo.com>
* gst/debug/rndbuffersize.c: (gst_rnd_buffer_size_loop): Fix printf
format to pacify Mac OSX's gcc.
Original commit message from CVS:
* gst/debug/rndbuffersize.c: (DEFAULT_SEED), (DEFAULT_MIN),
(DEFAULT_MAX), (src_template), (sink_template),
(gst_rnd_buffer_size_base_init), (gst_rnd_buffer_size_class_init),
(gst_rnd_buffer_size_init), (gst_rnd_buffer_size_activate),
(gst_rnd_buffer_size_loop), (gst_rnd_buffer_size_plugin_init):
Bring rndbuffersize element into a state that doesn't require us
to move it to -bad immediately. For one, fix up default min/max
values so that the element actuall works using the default values.
Also, don't ignore flow return values and do some kind of minimal
eos logic. Allow min=max to pull fixed-sized buffers. Bunch of
other gratuitious clean-ups.
Original commit message from CVS:
* gst/rtp/gstasteriskh263.c: (gst_asteriskh263_chain):
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_add_internal):
* gst/udp/gstudpsrc.c: (gst_udpsrc_start):
Use GLib versions of htonl, htons, ntohl and ntohs in order
to avoid problems on win32 (#529707).
Original commit message from CVS:
Patch by: Jesús Corrius <jesus at softcatala org>
* gst/goom/filters.c: (zoomVector):
* gst/goom/goom_core.c: (init_buffers):
Fix build with mingw32: use rand() instead of random() and
replace bzero() with memset(). Fixes#529692.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init), (request_pt_map),
(gst_rtspsrc_configure_caps):
Ref caps as the return value for the request_pt_map signal.
Remove some caps weirdness when configuring a stream. See #528245.
Original commit message from CVS:
* gst/goom/plugin_info.c: (setOptimizedMethods):
Disable altivec optimisations for 32-bit PPC as well to make
things build properly on all PPC systems. Fixes#528143
Original commit message from CVS:
* gst/goom/ppc_drawings.s:
* gst/goom/ppc_zoom_ultimate.s:
Change license of these files to LGPL, as permitted by the
author, Guillaume Borios. See #515073.
Original commit message from CVS:
* gst/goom/convolve_fx.c:
* gst/goom/motif_goom1.h:
* gst/goom/motif_goom2.h:
As hinted in Bug #518213, revert one change and fix warnings properly.
This fixes both #518213 and #520073 for me.
Original commit message from CVS:
* gst/matroska/ebml-read.c: (gst_ebml_read_seek):
* gst/matroska/matroska-demux.c:
(gst_matroska_demux_handle_seek_event),
(gst_matroska_demux_parse_contents_seekentry),
(gst_matroska_demux_loop):
Fix the Forte build by making function declaration signatures
match the implementations.
Original commit message from CVS:
2008-04-07 Julien Moutte <julien@fluendo.com>
* gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_setcaps): Fix build
because of a bad argument number.
Original commit message from CVS:
* gst/rtp/gstrtph264pay.c: (encode_base64),
(gst_rtp_h264_pay_setcaps), (gst_rtp_h264_pay_handle_buffer):
* gst/rtp/gstrtph264pay.h:
Parse codec_data for future AVC compatibility.
Fail when we encounter AVC data for now.
Original commit message from CVS:
* gst/spectrum/gstspectrum.c: (gst_spectrum_class_init),
(gst_spectrum_init), (gst_spectrum_set_property),
(gst_spectrum_get_property), (gst_spectrum_message_new):
Rename property enums and default defines for the properties to match
the property names and rephrase property descriptions to make them a
bit clearer (hopefully). See #518188.
Original commit message from CVS:
Based on patch by: mersad <mersad at axis dot com>
* gst/law/alaw-decode.c: (gst_alaw_dec_sink_setcaps),
(gst_alaw_dec_chain), (gst_alaw_dec_change_state):
* gst/law/alaw-decode.h:
* gst/law/alaw-encode.c: (gst_alaw_enc_chain):
* gst/law/mulaw-decode.c: (mulawdec_sink_setcaps),
(gst_mulawdec_chain), (gst_mulawdec_change_state):
* gst/law/mulaw-decode.h:
* gst/law/mulaw-encode.c: (gst_mulawenc_chain):
Make negotiation a bit modern.
Use pad_alloc. Fixes#525359.
Original commit message from CVS:
* gst/goom/Makefile.am:
Remove ppc assembler optimisations from the build until they
actually build (they also seem to have GPL headers).
Original commit message from CVS:
* gst/freeze/FAQ:
* gst/freeze/Makefile.am:
* gst/freeze/gstfreeze.c:
Add example to source code documentation blob and remove the 3 line
FAQ.
* gst/interleave/interleave.c:
Add a source code documentation blob.
Original commit message from CVS:
Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_finalize):
Call WSAStartup() and WSACleanup before using the Winsock API.
See #520808.
Original commit message from CVS:
* gst/goom/plugin_info.c:
* gst/goom/ppc_zoom_ultimate.h:
Small fixes to build more on PPC: ifdef out code that uses unknown
define; add newline at end of header file to avoid compiler warning.
Assembler code still doesn't build though.
Original commit message from CVS:
* gst/avi/gstavidemux.c:
Fix up my last commit. Use G_GUINT32_FORMAT for the guint32 debug log.
Also downgrade a GST_WARNING to GST_DEBUG and add a comment.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_perform_seek),
(gst_qtdemux_activate_segment),
(gst_qtdemux_prepare_current_sample),
(gst_qtdemux_loop_state_movie), (qtdemux_parse_trak):
Make sure we always send a DISCONT after a seek by setting the sample
index to an undefined value after a seek.
Original commit message from CVS:
* gst/avi/gstavisubtitle.h: (GST_IS_AVI_SUBTITLE),
(GST_IS_AVI_SUBTITLE_CLASS):
Fix up IS_FOO macros, which makes gtk-doc much happier.
Original commit message from CVS:
* gst/matroska/ebml-ids.h:
Add ID for EBML CRC32 elements.
* gst/matroska/Makefile.am:
* gst/matroska/ebml-read.c: (gst_ebml_finalize),
(gst_ebml_read_class_init), (gst_ebml_read_peek_bytes),
(gst_ebml_read_get_length), (_ext2dbl), (gst_ebml_read_float),
(gst_ebml_read_header):
Support reading 80bit floats, add finalize method to clean up
in any case, support reading length/id elements with any length
as long as it's smaller than our supported maximum, don't leak
buffers if reading as much data as we wanted failed and some
smaller cleanup.
Original commit message from CVS:
Patch by: Olivier Crete <tester at tester dot ca>
* gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_process):
Check that a buffer is large enough before reading from it.
Fixes bug #521102.
Original commit message from CVS:
* configure.ac:
* gst/audiofx/Makefile.am:
* gst/audiofx/audiochebband.c:
* gst/audiofx/audiocheblimit.c:
* gst/audiofx/math_compat.h:
Check for sinh(), cosh() and asinh() and define our own
implementations if they're not available. Fixes bug #520880.
Original commit message from CVS:
Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
* gst/udp/gstudpsrc.c: (gst_udpsrc_finalize), (gst_udpsrc_start),
(gst_udpsrc_stop):
Properly balance WSA_Cleanup with WSA_Startup.
Also make the poll controllable on windows. Fixes#520888.
Original commit message from CVS:
* gst/matroska/ebml-read.c:
Use GINT64 formatting constants from GLIB.
* gst/matroska/matroska-demux.c:
Add some guards to avoid a possible division by 0 and crashing
with NULL events on some systems.
Use gst_gdouble_to_guint64 somewhere instead of an implicit
conversion.
* gst/matroska/matroska-mux.c:
Check for invalid timestamps in a bunch of places to avoid
writing bogus durations into the output file.
Fix some double<->gint64 conversions that weren't using
gst_guint64_to_gdouble
Original commit message from CVS:
* gst/videomixer/videomixer.c: (gst_videomixer_blend_buffers):
Don't call gst_object_sync_values() unless we have a valid timestamp.
Original commit message from CVS:
* gst/matroska/matroska-demux.c:
* gst/matroska/matroska-ids.h:
* gst/matroska/matroska-mux.c:
Fix Dirac mapping. I had previously added a VfW-type
mapping, but it looks like Dirac will get a native Matroska
mapping, and this is the most likely method.
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/udp/gstudpsrc.c: (gst_udpsrc_init), (gst_udpsrc_create),
(gst_udpsrc_get_property), (gst_udpsrc_start), (gst_udpsrc_unlock),
(gst_udpsrc_unlock_stop), (gst_udpsrc_stop):
* gst/udp/gstudpsrc.h:
Port to GstPoll. See #505417.
Original commit message from CVS:
* gst/law/mulaw-decode.c: (gst_mulawdec_chain):
Return GST_FLOW_NOT_NEGOTIATED when the caps are not set
yet on the srcpad. We need rate and channels before we
can do any processing. Fixes bug #519088.
Original commit message from CVS:
* configure.ac:
Detect and indicate if GCC inline assembly syntax is
available.
* gst/goom/Makefile.am:
* gst/goom/convolve_fx.c:
* gst/goom/flying_stars_fx.c:
* gst/goom/goom_config.h:
* gst/goom/goom_core.c:
* gst/goom/goomsl.c:
* gst/goom/ifs.c:
* gst/goom/mmx.c:
* gst/goom/plugin_info.c:
* gst/goom/xmmx.c:
Fix various GCC-isms, and only build the inline assembly
with compilers that support GCC inline assembly.
Fix a couple of other warnings shown with Forte.
Original commit message from CVS:
* gst/goom/xmmx.c:
Use 'emms' instead of 'femms' to not crash on cpus that do not
implement this 3dnow specific instruction.
Original commit message from CVS:
* gst/goom/plugin_info.c: (setOptimizedMethods):
Use extended MMX for draw_line() too if available, not only
normal MMX.
Original commit message from CVS:
* gst/goom2k1/Makefile.am:
* gst/goom2k1/gstgoom.c:
Rename the installed library, and don't register the same
GType name as the new goom.
Original commit message from CVS:
* gst/goom/gstgoom.c: (goom_debug), (plugin_init):
* gst/goom/plugin_info.c: (goom_debug), (GST_CAT_DEFAULT),
(setOptimizedMethods):
Call oil_init(), otherwise oil_get_cpu_flags() won't return
anything useful. Export goom debug category so we can get
rid of the VERBOSE define and the printfs.
Original commit message from CVS:
* gst/goom/Makefile.am: Don't compile lex or yacc outputs
with warnings, but add other CFLAGS
* gst/goom/goomsl.c (gsl_instr_set_namespace),
(gsl_instr_add_param), (iflow_execute), (gsl_enternamespace),
(calculate_labels), (gsl_read_file):
* gst/goom/goomsl_lex.l:
* gst/goom/goomsl_yacc.y:
* gst/goom/plugin_info.c: Remove a few live printf, and
fprintf, replace exit() calls with g_assert_not_reached()
if it not optimal for a library
Original commit message from CVS:
* gst/goom/Makefile.am: Remove the warnings being disabled,
fix linkage on x86, spotted by Sebastian Dröge
<slomo@circular-chaos.org>
* gst/goom/convolve_fx.c (convolve_init),
(create_output_with_brightness), (convolve_apply):
* gst/goom/filters.c (zoomFilterVisualFXWrapper_create):
* gst/goom/goomsl.c:
* gst/goom/ifs.c (ifs_update), (ifs_visualfx_create):
* gst/goom/plugin_info.c:
* gst/goom/tentacle3d.c (tentacle_fx_create):
Fix warnings, and disable the motifs in the convolve_fx
plugin (they were causing warnings, and they were just
"Goom" in funny letterring)
Original commit message from CVS:
2008-02-23 Bastien Nocera <hadess@hadess.net>
* configure.ac: Add checks for Flex/Yacc/Bison and other
furry animals, for the new goom 2k4 based plugin
* gst/goom/*: Update to use goom 2k4, uses liboil to detect
CPU optimisations (not working yet), move the old plugin to...
* gst/goom2k1/*: ... here, in case somebody is sick enough
Fixes#515073
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_streams):
Post the server response code in an error message instead of a generic
'error' message. Fixes#517237.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream):
Init values to -1 instead of the default 0 value.
Fixes#516524.
Original commit message from CVS:
patch by: Wim Taymans <wim.taymans@collabora.co.uk>
fixes: #514889
* gst/rtp/gstrtph264pay.c:
* gst/rtp/gstrtpmp4gdepay.c:
* gst/rtp/gstrtpmp4gpay.c:
* gst/rtp/gstrtpmp4gpay.h:
* gst/rtp/gstrtptheorapay.c:
* gst/rtp/gstrtpvorbispay.c:
Fix various leaks shown up in valgrind
- free sprops and buffer in error cases in H264 payloader
- fix leak in mp4g depayloader when construction the caps
- don't leak config string in the mp4g payloader
- don't leak buffers and headers in theora and vorbis payloaders
* tests/check/elements/rtp-payloading.c:
Fix the RTP data test
- Actually send valid amr data to the payloader instead of 20
zero-bytes
- The mp4g payloader expects codec_data on the caps
Original commit message from CVS:
* gst/avi/gstavidemux.c:
Revert patch which sends timestamps only on keyframes, as it
breaks playback with current gst-ffmpeg.
Fixes: #515562
Original commit message from CVS:
* gst/multifile/gstmultifilesrc.c: (gst_multi_file_src_create):
* tests/check/elements/multifile.c: (GST_START_TEST):
Close some memory leaks spotted by the unit test. Fixes bug #515697.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* gst/spectrum/gstspectrum.c:
* tests/examples/spectrum/.cvsignore:
* tests/examples/spectrum/Makefile.am:
* tests/examples/spectrum/spectrum-example.c:
Add a simple example application for the spectrum element, include it
in the docs, and fix some documentation ambiguities.
Fixes: #348085
Original commit message from CVS:
* gst/equalizer/Makefile.am:
* gst/spectrum/Makefile.am:
Fix includes order
* tests/check/Makefile.am:
Exclude v4l2src from the states test - it takes too long to start.
* tests/check/elements/spectrum.c:
Make the test run properly with CK_FORK=no
Original commit message from CVS:
2008-02-08 Julien Moutte <julien@fluendo.com>
* gst/matroska/matroska-demux.c:
(gst_matroska_demux_parse_blockgroup_or_simpleblock): Flag
keyframe and delta units correctly when dealign with a
BlockGroup.
Fixes: #514397
Original commit message from CVS:
* gst/multifile/gstmultifilesrc.c: (gst_multi_file_src_create):
Need to use gsize here for the size, fixes compiler warning.
* tests/examples/equalizer/.cvsignore:
* tests/examples/equalizer/Makefile.am:
* tests/examples/spectrum/.cvsignore:
* tests/examples/spectrum/Makefile.am:
Add missing files to fix the build.
Original commit message from CVS:
* gst/multifile/gstmultifilesink.c:
* gst/multifile/gstmultifilesrc.c:
Use g_file_[sg]et_contents() instead of using stdio functions.
Should be less error prone.
* tests/check/elements/multifile.c:
Create a temporary directory using standard functions instead of
creating a directory in the current dir.
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.args:
* docs/plugins/inspect/plugin-audiofx.xml:
* gst/audiofx/Makefile.am:
* gst/audiofx/audiofx.c:
* gst/audiofx/audiowsincband.c:
* gst/audiofx/audiowsincband.h:
* gst/audiofx/audiowsinclimit.c:
* gst/audiofx/audiowsinclimit.h:
* tests/check/Makefile.am:
* tests/check/elements/audiowsincband.c:
* tests/check/elements/audiowsinclimit.c:
Move the lpwsinc and bpwsinc elements from gst-plugins-bad into
the audiofx plugin, and rename to audiowsinclimit and audiowsincband
respectively.
Fixes: #467666
Original commit message from CVS:
* gst/icydemux/gsticydemux.c: (gst_icydemux_chain):
* tests/check/elements/icydemux.c:
Return GST_FLOW_NOT_NEGOTIATED if we get a buffer without
caps, and add a somewhat useful debug message. Plus test.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c:
Include unistd.h only if HAVE_UNISTD_H is defined
* win32/common/config.h.in:
* win32/common/config.h:
Define socklen_t as it seems it's not defined in default
Visual Studio headers.
* win32/vs6/libgstalpha.dsp:
* win32/vs6/libgstapetag.dsp:
* win32/vs6/libgstavi.dsp:
* win32/vs6/libgstrtp.dsp:
* win32/vs6/libgstrtsp.dsp:
* win32/vs6/libgstvideomixer.dsp:
Update project file dependencies and add new source files
Original commit message from CVS:
Patch by: Bjarne Rosengren <bjarne at axis dot com>
* gst/matroska/ebml-write.c: (gst_ebml_write_element_push):
Don't leak buffers when we don't push them downstream.
Fixes bug #514965.
Original commit message from CVS:
* gst/multifile/gstmultifilesink.c:
Add a fixme comment.
* gst/selector/gstoutputselector.c:
Fix same leak as in input-selector.
* tests/icles/output-selector-test.c:
Improve the test.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_parse_subindex):
If there's no entries in the subindex, don't try to do anything stupid,
just return.
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:
* gst/avi/gstavisubtitle.c:
Add documentation for avisubtitle and change class to
Codec/Parser/Subtitle
Original commit message from CVS:
* gst/alpha/Makefile.am:
* gst/alpha/gstalpha.c:
Re-write the 'alpha' plugin to be BaseTransform based, simplifying
some stuff, and making buffer-alloc and resizing work automatically.
No longer crashes on odd frame widths and heights, although there
seems to be a disagreement with ffmpegcolorspace about what size
an AYUV frame with odd height should be.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_process_next_entry),
(gst_avi_demux_stream_data):
GStreamer timestamps are PTS values while AVI only knows about DTS
timestamps. Make sure we only copy the DTS as the buffer timestamp when
we are dealing with a key frame.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_rtpinfo):
Use g_ascii_strtoll() instead of atoll, which is only
available in C99.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init):
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init):
Don't implement get_unit_size() ourselves, the GstAudioFilter base
class already does this for us.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
Add symbols from -unused.txt to the right place.
* gst/dvdspu/gstdvdspu.c:
* gst/dvdspu/gstdvdspu.h:
Coherent namespace usage.
* gst/spectrum/gstspectrum.c:
Fix broken XML fragment in doc snippet even more.
Original commit message from CVS:
Based on a patch by:
Victor STINNER <victor dot stinner at haypocalc dot com>
* gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers):
Set variable to NULL after freeing it to prevent double frees
or make failures by another use of it afterwards more obvious
and fix use of it after the freeing.
Original commit message from CVS:
* gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_finalize):
Unparent all bands from the equalizer when finalizing to stop
leaking them.
Original commit message from CVS:
Patch by: Olivier Crete <tester at tester dot ca>
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init),
(gst_multiudpsink_init), (gst_multiudpsink_set_property),
(gst_multiudpsink_get_property), (gst_multiudpsink_init_send),
(gst_multiudpsink_add_internal), (gst_multiudpsink_remove):
* gst/udp/gstmultiudpsink.h:
Add property to automatically join a multicast group or not. This can be
useful when sharing a socket between multiple elements.
Fixes#509531.
Original commit message from CVS:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* gst/multipart/Makefile.am:
* gst/multipart/multipartdemux.c:
* gst/multipart/multipartdemux.h:
* gst/multipart/multipartmux.c:
* gst/multipart/multipartmux.h:
Re-add multipartdemux to the docs. Last round of section cleanup.
Original commit message from CVS:
As found by: Tommi Myöhänen <ext-tommi.myohanen nokia com>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_rtpinfo):
Use atoll to parse the rtptime with enough precision. Fixes#509329.
Original commit message from CVS:
* gst/avi/gstavisubtitle.c: (gst_avi_subtitle_extract_file):
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send):
Initialise variables to work around (false) 'foo might be used
uninitialized in this function' warnings by gcc-3.3.3 (#509298).
Original commit message from CVS:
* ext/flac/gstflacdec.c: (gst_flac_extract_picture_buffer):
* gst/id3demux/id3v2frames.c: (parse_picture_frame):
Generate the image-type values correctly. Leave them out of the caps
when outputting a "preview image" tag, since it only makes sense
to have one of those - the type is irrelevant.
* sys/sunaudio/gstsunaudiomixerctrl.c:
(gst_sunaudiomixer_ctrl_open):
If we can, mark the mixer multiple open when we use it, in case
(for some reason) the process wants to open it again elsewhere.
Original commit message from CVS:
Patch by: Olivier Crete <tester at tester dot ca>
* gst/rtp/gstrtptheoradepay.c: (gst_rtp_theora_depay_setcaps):
* gst/rtp/gstrtptheorapay.c:
Fix the clock rate to 90000 as required by the RFC.
Fixes#508644.
Original commit message from CVS:
Based on patch by: Tommi Myöhänen <ext-tommi.myohanen nokia com>
* gst/id3demux/id3v2frames.c: (parse_comment_frame):
Make sure the ISO 639-X language code in ID3v2 COMM frames
is actually valid UTF-8 (or rather: ASCII), so we don't end
up with non-UTF8 strings in tags if there's garbage in the
language field. Also make sure the language code is always
lower case. Fixes: #508291.
Original commit message from CVS:
reviewed by: Edward Hervey <edward.hervey@collabora.co.uk>
* gst/videomixer/videomixer.c:
(gst_videomixer_set_master_geometry), (_do_init),
(gst_videomixer_child_proxy_get_child_by_index),
(gst_videomixer_child_proxy_get_children_count),
(gst_videomixer_child_proxy_init), (gst_videomixer_reset),
(gst_videomixer_init), (gst_videomixer_request_new_pad),
(gst_videomixer_release_pad), (gst_videomixer_fill_queues):
Implement GstChildProxy interface.
Send newsegment at the right moment
Fixes#488879
Original commit message from CVS:
* gst/alpha/Makefile.am:
* gst/alpha/gstalpha.c: (gst_alpha_class_init), (gst_alpha_init),
(gst_alpha_sink_event), (gst_alpha_chain),
(gst_alpha_change_state), (plugin_init):
Make the various properties of 'alpha' controllable. This allows doing
niceties like fade-in/fade-out.
Original commit message from CVS:
* gst/matroska/matroska-mux.c: (COMMON_VIDEO_CAPS_NO_FRAMERATE),
(videosink_templ):
Also fix up pad templates to indicate that image/jpeg doesn't
absolutely require the framerate property to be set (#504081).
Original commit message from CVS:
Based on patch by: Wouter Cloetens <wouter at mind be>
* gst/matroska/matroska-mux.c: (gst_matroska_mux_video_pad_setcaps),
(gst_matroska_mux_request_new_pad), (gst_matroska_mux_release_pad),
(gst_matroska_mux_finish), (gst_matroska_mux_collected):
* gst/matroska/matroska-mux.h:
Keep track of first and last timestamps for each incoming stream,
so we can calculate the total duration for live sources and other
input where we can't query the duration from the start or where
there's no constant framerate from which we can deduce the
duration; also use calculated/observed duration if it is bigger
than the previously queried duration. Furthermore, use
gst_pad_query_peer_duration() and take into account that it may
return TRUE but still a duration of CLOCK_TIME_NONE, which easily
screws up comparisons when using unsigned integers. Fixes#504081.
Original commit message from CVS:
* configure.ac:
* gst/audiofx/audioamplify.c:
(gst_audio_amplify_clipping_method_get_type),
(gst_audio_amplify_init), (gst_audio_amplify_transform_ip):
* gst/audiofx/audiodynamic.c: (gst_audio_dynamic_init),
(gst_audio_dynamic_transform_ip):
* gst/audiofx/audioinvert.c: (gst_audio_invert_init),
(gst_audio_invert_transform_ip):
* gst/audiofx/audiopanorama.c: (gst_audio_panorama_init),
(gst_audio_panorama_transform):
* gst/level/gstlevel.c: (gst_level_init):
Make elements GST_BUFFER_FLAG_GAP aware and call
gst_base_transform_set_gap_aware for this.
Bump core requirement to CVS.
* gst/audiofx/audiochebyshevfreqband.c:
(gst_audio_chebyshev_freq_band_transform_ip):
* gst/audiofx/audiochebyshevfreqlimit.c:
(gst_audio_chebyshev_freq_limit_transform_ip):
Also sync GObject properties to the controller if operating
in passthrough mode.
Original commit message from CVS:
* gst/avi/gstavi.c:
increase rank because no known issues anymore ...
* gst/avi/gstavisubtitle.c:
send subtitle name to the srcpad
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send), (gst_rtspsrc_open):
* gst/rtsp/gstrtspsrc.h:
Implement redirect for the DESCRIBE reply. Fixes#506025.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_loop):
* gst/wavparse/gstwavparse.c: (gst_wavparse_chain):
* sys/ximage/gstximagesrc.c: (composite_pixel):
Fix 'xyz may be used uninitialized' compiler warnings caused
by broken g_assert_not_reached() macro in GLib-2.15.x (it's
not really nice to abort in any case). Fixes#505745.
Original commit message from CVS:
* gst/avi/gstavisubtitle.c: (IS_BOM_UTF8), (IS_BOM_UTF16_BE),
(IS_BOM_UTF16_LE), (IS_BOM_UTF32_BE), (IS_BOM_UTF32_LE),
(gst_avi_subtitle_extract_file), (gst_avi_subtitle_parse_gab2_chunk):
Detect other UTF byte order markers and convert to UTF-8 as
appropriate.
Original commit message from CVS:
* gst/avi/gstavisubtitle.c: (src_template),
(gst_avi_subtitle_extract_utf8_file),
(gst_avi_subtitle_parse_gab2_chunk), (gst_avi_subtitle_chain),
(gst_avi_subtitle_base_init), (gst_avi_subtitle_class_init),
(gst_avi_subtitle_init), (gst_avi_subtitle_change_state):
* gst/avi/gstavisubtitle.h:
Refactor a bit; fix name extraction; don't assume all the data
in the chunk is actually subtitle data, there may be padding at
the end; fix GST_ELEMENT_ERROR usage; store extracted subtitle
file so it's there to send again after a seek (for future use).
Original commit message from CVS:
* gst/avi/Makefile.am:
* gst/avi/gstavi.c:
* gst/avi/gstavisubtitle.c:
* gst/avi/gstavisubtitle.h:
* tests/check/Makefile.am:
* tests/check/elements/avisubtitle.c:
* win32/common/config.h:
Add avi subtitle element for bug #442034. Need seeking support
and more support for character conversion.
Original commit message from CVS:
* gst/multifile/gstmultifilesrc.c:
* gst/multifile/gstmultifilesrc.h:
When subsequent files are read, if the file doesn't exist, send
an EOS instead of causing an error.
Original commit message from CVS:
* gst/id3demux/id3v2frames.c: (parse_url_link_frame):
Parse WOAF frames and put the result into GST_TAG_CONTACT,
which is where it would end up if the same information was
put in a vorbis comment (don't think it's worth adding a
new URI tag for this). Fixes#488112.
Original commit message from CVS:
Patch by: Wai-Ming Ho <webregbox at yahoo dot co dot uk>
* gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_init),
(gst_rtp_h264_pay_finalize), (gst_rtp_h264_pay_setcaps),
(next_start_code), (is_nal_equal), (gst_rtp_h264_pay_decode_nal),
(encode_base64), (gst_rtp_h264_pay_parse_sps_pps),
(gst_rtp_h264_pay_handle_buffer):
* gst/rtp/gstrtph264pay.h:
Use higher performance start-code searching.
Parse NALs and store SPS, PPS and profile in the caps so that they can
be used in the SDP. Fixes#502814.
Original commit message from CVS:
Patch by: Wouter Cloetens <wouter at mind dot be>
* gst/multipart/multipartdemux.c: (gst_multipart_demux_chain):
Copy timestamp from input to output. Not very perfect yet but better
than nothing. Fixes#503023.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (bpwsinc_transform):
* gst/filter/gstlpwsinc.c: (lpwsinc_transform):
The transform() methods are not called in passthrough mode so
there's no need for checking if the element is in passthrough mode.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (bpwsinc_transform):
* gst/filter/gstlpwsinc.c: (lpwsinc_transform):
Sync the GObject properties with the controller even in passthrough
mode to get consistent property values.
Original commit message from CVS:
* gst/audiofx/audioamplify.c: (gst_audio_amplify_transform_ip):
* gst/audiofx/audiochebyshevfreqband.c:
(gst_audio_chebyshev_freq_band_transform_ip):
* gst/audiofx/audiochebyshevfreqlimit.c:
(gst_audio_chebyshev_freq_limit_transform_ip):
* gst/audiofx/audiodynamic.c: (gst_audio_dynamic_transform_ip):
* gst/audiofx/audioinvert.c: (gst_audio_invert_transform_ip):
The transform_ip() methods should do nothing if in passthrough mode.
It might get non-writable buffers in that case but the buffer might
as well be writable.
* gst/audiofx/audiopanorama.c: (gst_audio_panorama_transform):
The transform() methods won't be called in passthrough mode and
otherwise the buffer is always writable so don't check here.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_srcpad_event):
Fix seeking in .wav files again (#501775). Some people seem to think
they don't need to test their changes when they're just 'reflowing'
some code.
Original commit message from CVS:
* gst/autodetect/gstautovideosink.c:
(gst_auto_video_sink_class_init), (gst_auto_video_sink_dispose),
(gst_auto_video_sink_init),
(gst_auto_video_sink_create_element_with_pretty_name),
(gst_auto_video_sink_find_best),
(gst_auto_video_sink_set_property),
(gst_auto_video_sink_get_property):
* gst/autodetect/gstautovideosink.h:
Fix docs.
Use same error reporting code as autoaudiosink.
Add property to filter sinks based on caps. Only select raw video sinks
by default for backwards compat.
API: GstAutoVideoSink::filter-caps
Original commit message from CVS:
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
* gst/autodetect/gstautoaudiosink.c:
(gst_auto_audio_sink_class_init), (gst_auto_audio_sink_dispose),
(gst_auto_audio_sink_init), (gst_auto_audio_sink_find_best),
(gst_auto_audio_sink_set_property),
(gst_auto_audio_sink_get_property):
* gst/autodetect/gstautoaudiosink.h:
Add property to filter sinks based on caps. Only select raw audio sinks
by default for backwards compat. Fixes#417420.
API: GstAutoAudioSink::filter-caps
Original commit message from CVS:
* gst/rtp/gstrtph263depay.c: (gst_rtp_h263_depay_class_init),
(gst_rtp_h263_depay_process):
Code beautification.
Added debug statements.
Don't bit-shift everything, just do operations on last/first byte
instead.
Original commit message from CVS:
Patch by: Jayarama S. Santana <sundarsantana at gmail dot com>
* gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_process):
Fix wrong comparison in overrun check. Fixes#499239 some more.
Original commit message from CVS:
* gst/rtp/gstrtph263depay.c: (gst_rtp_h263_depay_init),
(gst_rtp_h263_depay_process):
* gst/rtp/gstrtph263depay.h:
Fix h263 depayloader so that ANY h263 decoder can handle the outgoing
stream.
Original commit message from CVS:
Based on Path by: Jayarama S. Santana <sundarsantana at gmail dot com>
* gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_setcaps),
(gst_rtp_mp4a_depay_process):
* gst/rtp/gstrtpmp4adepay.h:
Fix depayloading when multiple frames are inside one RTP packet.
Fixes#499239.
Original commit message from CVS:
* gst/rtp/gstrtph263depay.c: (gst_rtp_h263_depay_process):
Read the I flag for Mode A h263 rtp stream and set the
GST_BUFFER_FLAG_DELTA_UNIT accordingly.
Fixes#499383
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (bpwsinc_set_property):
* gst/filter/gstlpwsinc.c: (lpwsinc_set_property):
Post a GST_MESSAGE_LATENCY if the latency changes.
Original commit message from CVS:
* gst/equalizer/gstiirequalizer10bands.c:
* gst/equalizer/gstiirequalizer3bands.c:
Remove preset iface again. We'll re-add this after its been released
in -good.
Original commit message from CVS:
2007-11-20 Julien MOUTTE <julien@moutte.net>
* ext/taglib/gsttaglibmux.c: (gst_tag_lib_mux_render_tag),
(gst_tag_lib_mux_adjust_event_offsets):
* gst/qtdemux/qtdemux.c: (qtdemux_parse_theora_extension):
* sys/osxaudio/Makefile.am:
* sys/osxvideo/cocoawindow.h:
* sys/osxvideo/cocoawindow.m: Fix build on Mac OS X 10.5
Original commit message from CVS:
* gst/equalizer/gstiirequalizer10bands.c:
* gst/equalizer/gstiirequalizer3bands.c:
Activate preset iface and upload two presets here.
Original commit message from CVS:
Patch by: Jordi Jaen Pallares <jordijp at gmail dot com>
* gst/rtp/gstrtpmp2tpay.c: (gst_rtp_mp2t_pay_class_init),
(gst_rtp_mp2t_pay_init), (gst_rtp_mp2t_pay_finalize),
(gst_rtp_mp2t_pay_flush), (gst_rtp_mp2t_pay_handle_buffer):
* gst/rtp/gstrtpmp2tpay.h:
Fill the MTU with as many packets as possible. Fixes#491323.
Original commit message from CVS:
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_streams):
Fix some more leaks. Fixes#497007.
Original commit message from CVS:
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_free),
(gst_rtspsrc_stream_configure_tcp):
Fix 3 pad leaks. Fixes#496983.
Original commit message from CVS:
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_setcaps):
Fix small leak. Fixes#497017.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state),
(gst_qtdemux_prepare_current_sample),
(gst_qtdemux_loop_state_movie), (qtdemux_parse_theora_extension),
(qtdemux_parse_node), (qtdemux_parse_trak), (qtdemux_video_caps):
* gst/qtdemux/qtdemux_fourcc.h:
* gst/qtdemux/qtdemux_types.c:
Add suppport for theora in quicktime according to XiphQT.
Original commit message from CVS:
* gst/id3demux/id3tags.c:
* gst/id3demux/id3tags.h:
* gst/id3demux/id3v2frames.c: (id3v2_tag_to_taglist):
We don't want the same string multiple times in a tag list for the
same tag ever, for any tag, not just for GST_TAG_GENRE, so make sure
this doesn't happen and remove special-case code for GST_TAG_GENRE.
Original commit message from CVS:
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_handle_src_event),
(gst_rtspsrc_parse_range):
Don't leak event, don't leak range (fixes#496752).
Original commit message from CVS:
Patch by: Arek Korbik <arkadini@gmail.com>
* gst/alpha/gstalphacolor.c: (gst_alpha_color_set_caps):
Detect RGBA/BGRA correctly on little endian systems.
Original commit message from CVS:
Patch by: Mark Nauwelaerts <manauw skynet be>
* gst/matroska/matroska-demux.c:
(gst_matroska_demux_push_dvd_clut_change_event),
(gst_matroska_demux_parse_blockgroup_or_simpleblock),
(gst_matroska_demux_subtitle_caps):
* gst/matroska/matroska-ids.h:
Extract palette data for dvd subpicture streams and send it
downstream as custom gstreamer dvd event (fixes#453417).
Original commit message from CVS:
* gst/avi/gstavidemux.c:
* gst/wavparse/gstwavparse.c:
Return the result in _activate_pull(). Don't ref element there.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers),
(gst_wavparse_pad_convert), (gst_wavparse_pad_query),
(gst_wavparse_srcpad_event):
Ref the element when we should, but not when we its not needed. Reflow
the event_handling to not leak the event.
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler dot de>
* gst/replaygain/rganalysis.c: (yule_filter):
Avoid slowdown from denormals when processing near-silence input data.
Spotted by Gabriel Bouvigne. Fixes#494499.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state),
(qtdemux_parse_samples):
Properly free QTDemuxSamples array.
Protect table write with a sensible check, some files apparently DO contain
stts values starting with 0 :(
Original commit message from CVS:
* gst/avi/gstavidemux.c:
* gst/qtdemux/qtdemux.c:
Drop EOS in _handle_src_event(). Fix the refcount in qtdemux that
previous commit messed up.
Original commit message from CVS:
* gst/avi/gstavidemux.c:
* gst/qtdemux/qtdemux.c:
Sync _handle_src_event() with oggdemux. In avidemux also ref the
element when we should, but not when we its not needed.
Original commit message from CVS:
* gst/equalizer/demo.c: (draw_spectrum):
* gst/spectrum/demo-audiotest.c: (draw_spectrum):
* gst/spectrum/demo-osssrc.c: (draw_spectrum):
* gst/spectrum/gstspectrum.c: (gst_spectrum_class_init):
Change the meaning of the magnitude values given in the
GstMessages by spectrum to decibel instead of
decibel+threshold.
Original commit message from CVS:
* gst/equalizer/gstiirequalizer10bands.c:
* gst/equalizer/gstiirequalizer3bands.c:
* gst/equalizer/gstiirequalizernbands.c:
And continue to update docs. Also include some sample code
for the n-band equalizer in the docs.
Original commit message from CVS:
* gst/equalizer/gstiirequalizer10bands.c:
(gst_iir_equalizer_10bands_class_init):
* gst/equalizer/gstiirequalizer3bands.c:
(gst_iir_equalizer_3bands_class_init):
* gst/equalizer/gstiirequalizernbands.c:
Update docs and property ranges to the real values.
Original commit message from CVS:
* gst/spectrum/gstspectrum.c:
Now do the scaling right for real. Also initialize a previously
uninitialized variable.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c:
Return FALSE if we can't handle a query instead of changing the
format. Ignore fact when dealing with mpeg audio.
Original commit message from CVS:
* gst/spectrum/demo-audiotest.c: (main):
Use autoaudiosink instead of alsasink and use a sine wave.
* gst/spectrum/gstspectrum.c:
Fix the magnitude calculation.
Original commit message from CVS:
* gst/equalizer/demo.c: (main):
* gst/equalizer/gstiirequalizer.c:
(gst_iir_equalizer_band_class_init), (setup_filter):
Allow setting 0 as bandwidth and handle this correctly.
Also handle a bandwidth of rate/2 properly.
* gst/equalizer/gstiirequalizernbands.c:
(gst_iir_equalizer_nbands_class_init):
Make it possible to generate a N-band equalizer with 1 bands. The
previous limit of 2 was caused by a nowadays replaced calculation
doing a division by zero if number of bands was 1.
Original commit message from CVS:
Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
* configure.ac:
* gst/udp/gstdynudpsink.c:
* gst/udp/gstdynudpsink.h:
* gst/udp/gstmultiudpsink.c:
* gst/udp/gstmultiudpsink.h:
* gst/udp/gstudpsink.c:
* gst/udp/gstudpsink.h:
Fix includes for MSVC and GLib-2.14.0 (#492388).
* gst/udp/gstudpsrc.c: (gst_udpsrc_start):
No more pipe define since GLib-2.14.0, need to use _pipe() directly.
Original commit message from CVS:
* gst/law/mulaw-decode.c: (mulawdec_sink_setcaps),
(gst_mulawdec_chain):
* gst/law/mulaw-decode.h:
Calculate outgoing buffer duration if incoming buffer didn't have a
valid duration.
Original commit message from CVS:
* gst/equalizer/Makefile.am:
* gst/equalizer/demo.c: (on_window_destroy), (on_configure_event),
(on_gain_changed), (on_bandwidth_changed), (on_freq_changed),
(draw_spectrum), (message_handler), (main):
Add small demo application based on the spectrum demo applications
that gets white noise as input, pushes it through an equalizer and
paints the spectrum. For every equalizer band it's possible to set
gain, bandwidth and frequency.
* gst/equalizer/gstiirequalizer.c: (setup_filter):
Add some guarding against too large or too small frequencies and
bandwidths. Also improve debugging a bit.
Original commit message from CVS:
* gst/equalizer/gstiirequalizer.c:
(gst_iir_equalizer_band_set_property),
(gst_iir_equalizer_band_get_property),
(gst_iir_equalizer_band_class_init), (arg_to_scale),
(setup_filter), (gst_iir_equalizer_compute_frequencies):
Replace filters with a bit better filters for which we can actually
find documentation, which don't change anything on zero gain, etc.
Make the frequency property of the bands writable, rename the
band-width property to bandwidth and change the meaning to the
frequency difference between bandedges, change the meaning of the
gain property to dB instead of a weird scale between -1 and 1 that
has no real meaning.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_activate_segment),
(gst_qtdemux_combine_flows), (gst_qtdemux_loop_state_movie):
Smarter combine_flow code that also deals with downstream elements
returning UNEXPECTED when they receive data out of the segment
boundaries. Fixes#491305.
Original commit message from CVS:
* gst/interleave/interleave.c: (gst_interleave_request_new_pad):
Let's not call every request pad we create "sink%d", that'll
create problems if there's to be more than one pad. Fixes#490682.
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/interleave.c:
Add unit test for the above.
Original commit message from CVS:
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_interleaved):
Fix race when pausing a RTSP stream in interleaved.
Fixes#475784.
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_finalize):
Use correct unref function for buffers. #488844.
Original commit message from CVS:
Based on patch by: Laurent Glayal <spglegle yahoo fr>
* gst/udp/gstudpsrc.c: (gst_udpsrc_create):
When the socket is used by the app for other purposes, don't generate an
error if there is activaty on the socket that is not data related.
Fixes#487488.
Original commit message from CVS:
Patch by: Anders Skargren <anders dot skargren at axis dot com>
* gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_handle_buffer):
Set marker bit correctly.
Original commit message from CVS:
* gst/equalizer/gstiirequalizer.c:
(gst_iir_equalizer_band_set_property),
(gst_iir_equalizer_band_get_property),
(gst_iir_equalizer_band_class_init), (gst_iir_equalizer_band_init),
(gst_iir_equalizer_band_get_type), (gst_iir_equalizer_class_init),
(setup_filter), (gst_iir_equalizer_setup):
* gst/equalizer/gstiirequalizer.h:
Move bandwidth property to the separate bands and add float64 support.
Original commit message from CVS:
Based on patch by: Jason Kivlighn <jkivlighn gmail com>
* gst/id3demux/id3v2frames.c:
Extract license/copyright URIs from ID3v2 WCOP frames
(Fixes#447000).
* tests/check/elements/id3demux.c:
* tests/files/Makefile.am:
* tests/files/id3-447000-wcop.tag:
Add simple unit test.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_flush):
Fix compiler warning by using GST_CLOCK_TIME_NONE to initialise
a GstClockTime.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_flush),
(gst_rtspsrc_do_seek), (gst_rtspsrc_perform_seek),
(gst_rtspsrc_configure_caps), (gst_rtspsrc_loop_udp),
(gst_rtspsrc_parse_range), (gst_rtspsrc_open), (gst_rtspsrc_play),
(gst_rtspsrc_change_state):
More seeking fixes, mostly passing around the new playback segment in
order to configure it properly.
Also reset base_time of udp sources when setting them back to PLAYING as
a temporary hack until core supports seek in live sources properly.
Original commit message from CVS:
* gst/id3demux/gstid3demux.c:
* gst/id3demux/gstid3demux.h:
* gst/id3demux/id3tags.c:
* gst/id3demux/id3tags.h:
* gst/id3demux/id3v2frames.c:
Port ID3 tag demuxer over to the new GstTagDemux in -base
(now would be a good time to test re-importing your music
collection).
Original commit message from CVS:
* gst/apetag/Makefile.am:
* gst/apetag/gstapedemux.c:
* gst/apetag/gstapedemux.h:
* gst/apetag/gsttagdemux.c:
* gst/apetag/gsttagdemux.h:
Port APE tag demuxer over to the new GstTagDemux in -base.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_flush),
(gst_rtspsrc_perform_seek), (gst_rtspsrc_handle_src_event),
(gst_rtspsrc_handle_internal_src_query),
(gst_rtspsrc_handle_src_query), (new_session_pad),
(gst_rtspsrc_stream_configure_tcp),
(gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_loop_send_cmd):
Improve flushing behaviour.
Set state of the udp sources to PAUSE/PLAYING correctly.
Handle events and queries for UDP and TCP transport now.
Original commit message from CVS:
* gst/avi/gstavimux.c:
Fix "Index entry has invalid stream nr 1".
Add support for muxing aac - work in progress (see #482495).
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_get_bandwidth),
(gst_rtspsrc_collect_bandwidth), (gst_rtspsrc_create_stream),
(gst_rtspsrc_media_to_caps), (gst_rtspsrc_loop_interleaved):
* gst/rtsp/gstrtspsrc.h:
Parse bandwidth modifiers, they are not yet configured in the session
manager because we don't have an API for that yet.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_rtpmap),
(gst_rtspsrc_media_to_caps), (gst_rtspsrc_loop_interleaved):
Use shiny new function in -base to get the default clock-rate.
Update some docs.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_play):
* gst/rtsp/gstrtspsrc.h:
In TCP mode, only timestamp the first buffer. TCP is not real time and
it does not make sense to try to skew compensate, also some servers send
the first batch of data in a burst.
Original commit message from CVS:
* gst/matroska/matroska-demux.c:
Fix setting the discont flag on the first buffer
pushed downstream for formats with private codec
data that needs to be deserialised into buffers
(such as vorbis and FLAC when in a matroska container).
Original commit message from CVS:
Patch by: Antoine Tremblay <hexa00 at gmail dot com>
* gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_init),
(gst_rtp_mp4v_pay_finalize), (gst_rtp_mp4v_pay_flush),
(gst_rtp_mp4v_pay_handle_buffer):
* gst/rtp/gstrtpmp4vpay.h:
Free the config string. Fixes#480707.
Clean up the timestamp code a little.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
(gst_rtspsrc_handle_src_query), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_loop_udp), (gst_rtspsrc_close):
* gst/rtsp/gstrtspsrc.h:
Set timestamps on RTP buffers in interleaved mode.
Mark first buffers with a DISCONT.
Remove flush hack now that sync for live sources has been figured out.
Original commit message from CVS:
* gst/qtdemux/gstrtpxqtdepay.c: (gst_rtp_xqt_depay_process),
(gst_rtp_xqt_depay_change_state):
* gst/qtdemux/gstrtpxqtdepay.h:
Fail if we don't know the quicktime format.
Original commit message from CVS:
* ext/taglib/gstapev2mux.cc:
* ext/taglib/gstid3v2mux.cc:
* gst/apetag/gstapedemux.c:
Add support for the new GST_TAG_COMPOSER (#459809).
Original commit message from CVS:
* gst/law/alaw-decode.c:
* gst/law/alaw-decode.h:
* gst/law/alaw-encode.c:
* gst/law/alaw-encode.h:
* gst/law/alaw.c:
* gst/law/mulaw-conversion.h:
Compulsive clean-ups: use boilerplate macros, add debug
categories, fix up things to conform to symbol nomenklatura,
etc.
Original commit message from CVS:
Based on patch by: Laurent Glayal <spglegle yahoo fr>
* gst/law/alaw-decode.c:
* gst/law/alaw-encode.c:
Use static tables for A-Law decoding and encoding; this makes
A-Law decoding and encoding less CPU-intensive, but increases
the binary size a bit. Leaving old code around for now,
selectable by a define in the code. Fixes#435435.
Original commit message from CVS:
* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_process):
Set outgoing packet duration because we can. Fixes#478244 some more.
Original commit message from CVS:
* ChangeLog:
Add missing newline.
* gst/librfb/rfbdecoder.c:
Fix the build (missing stdlib.h).
* gst/spectrum/gstspectrum.c:
* gst/spectrum/gstspectrum.h:
Use basetransform segment so that it is correctly managed on flushes
and start/stop. Report message timestamp as stream time, which is what
an application can understand. (Yes these are adapted from wim recent
level element changes)
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_streams),
(gst_rtspsrc_dup_printf):
Give meaningfull error when all streams failed to configure for some
reason.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_loop),
(gst_wavparse_chain):
Don't push EOS from the chain function, the element
driving the pipeline is responsible for this. The bug
this was meant to fix seems to be queue not forwarding
EOS in all cases (see #476514).
Original commit message from CVS:
* gst/level/gstlevel.c: (gst_level_class_init), (gst_level_start),
(gst_level_transform_ip):
* gst/level/gstlevel.h:
Use basetransform segment so that it is correctly managed on flushes and
start/stop.
Report message timestamp as stream time, which is what an application
can understand.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_perform_eos),
(gst_wavparse_loop), (gst_wavparse_chain):
Add EOS logic for the push-based mode too. Fixes#476514.
Original commit message from CVS:
* gst/udp/gstudpsrc.c: (gst_udpsrc_init), (gst_udpsrc_create):
Make udpsrc timestamp outgoing buffers based on when they were received.
Also make it output a segment in time.
Original commit message from CVS:
Patch by: Haakon Sporsheim <haakon.sporsheim at tandberg com>
* gst/rtp/gstrtph263pay.c:
Fix up header structure so that compilers don't add padding
between the structure fields, since that would lead to us
sending RTP packets with broken headers (as is currently the
case when compiling with MSVC). Also see similar fixes in
libgstrtp in gst-plugins-base. (#474616; #471194)
Original commit message from CVS:
* gst/spectrum/gstspectrum.c: (gst_spectrum_start),
(gst_spectrum_transform_ip):
Use the correct parameter order for the memset calls.
Thanks to Christian Schaller for noticing.
Original commit message from CVS:
* configure.ac:
* gst/spectrum/Makefile.am:
* gst/spectrum/demo-audiotest.c: (draw_spectrum),
(message_handler), (main):
* gst/spectrum/demo-osssrc.c: (draw_spectrum), (message_handler):
* gst/spectrum/gstspectrum.c: (gst_spectrum_base_init),
(gst_spectrum_class_init), (gst_spectrum_init),
(gst_spectrum_dispose), (gst_spectrum_set_property),
(gst_spectrum_get_property), (gst_spectrum_start),
(gst_spectrum_setup), (gst_spectrum_message_new),
(gst_spectrum_transform_ip):
* gst/spectrum/gstspectrum.h:
Port GstSpectrum to GstAudioFilter and libgstfft, add support
for int32, float and double, use floats for the message contents,
average all FFTs done in one interval for better results, use
a better windowing function, allow posting the phase in the message
and actually do an FFT with the requested number of bands instead
of interpolating.
* tests/check/elements/spectrum.c: (GST_START_TEST),
(spectrum_suite):
Improve the units tests by checking for a 11025Hz sine wave
and add unit tests for all 4 supported sample types.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c:
* gst/wavparse/gstwavparse.h:
Implement seek-query. Refactor duration calculations. Appropriate use
of uint64_scale_int and uint64_scale. Move repeadedly calculated stuff
out of loops.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_udp_sink),
(gst_rtspsrc_dup_printf):
Use new basesink async property to make sparse RTCP packet not wait for
preroll.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_dup_printf),
(gst_rtspsrc_get_float), (gst_rtspsrc_play):
Make sure we generate and parse floating point values in the POSIX
locale instead of the current locale.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_do_seek),
(gst_rtspsrc_parse_methods), (gst_rtspsrc_open),
(gst_rtspsrc_play):
* gst/rtsp/gstrtspsrc.h:
Fix method detection again.
Keep track of when we must send a Range header.
Use segment values for Range, Speed and Scale headers.
Parse Speed and Scale headers to update the segment values.
Original commit message from CVS:
* gst/audiofx/audiochebyshevfreqband.c:
* gst/audiofx/audiochebyshevfreqlimit.c:
Add small comparision with the windowed sinc filters in the docs.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_loop_udp), (gst_rtspsrc_loop_send_cmd),
(gst_rtspsrc_loop):
Refactor the udp and interleaved loop function a bit.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_finalize), (gst_rtspsrc_connection_send),
(gst_rtspsrc_connection_receive), (gst_rtspsrc_sink_chain),
(gst_rtspsrc_handle_request), (gst_rtspsrc_send_keep_alive),
(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp),
(gst_rtspsrc_try_send), (gst_rtspsrc_pause):
* gst/rtsp/gstrtspsrc.h:
Protect connection activity with a new lock, avoids deadlocks when going
to PAUSED. Fixes#455808.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init),
(bpwsinc_set_property), (bpwsinc_get_property):
* gst/filter/gstbpwsinc.h:
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init),
(gst_lpwsinc_init), (lpwsinc_build_kernel), (lpwsinc_set_property),
(lpwsinc_get_property):
* gst/filter/gstlpwsinc.h:
* tests/check/elements/lpwsinc.c: (GST_START_TEST):
Use generator macros for the process functions for the different
sample types, add lower upper boundaries for the GObject properties
so automatically generated UIs can use sliders and change frequency
properties to floats to save a bit of memory, even ints would in
theory be enough. Also rename frequency to cutoff for consistency
reasons.
* docs/plugins/gst-plugins-bad-plugins.args:
* docs/plugins/gst-plugins-bad-plugins.signals:
* docs/plugins/inspect/plugin-gstrtpmanager.xml:
Regenerated for the above changes.
Original commit message from CVS:
* gst/audiofx/audiochebyshevfreqband.c:
(gst_audio_chebyshev_freq_band_class_init):
* gst/audiofx/audiochebyshevfreqlimit.c:
(gst_audio_chebyshev_freq_limit_class_init):
Use generator macros for the process functions for the different
sample types, add lower upper boundaries for the GObject properties
so automatically generated UIs can use sliders and add a note about
the number of poles as a too high number of poles combined with
very low or very high frequencies will produce only noise.
* docs/plugins/gst-plugins-good-plugins.args:
Regenerated for the property changes.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_set_property),
(gst_rtspsrc_flush), (gst_rtspsrc_sink_chain),
(gst_rtspsrc_stream_configure_udp_sink),
(gst_rtspsrc_send_keep_alive), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_loop_udp), (gst_rtspsrc_loop_send_cmd),
(gst_rtspsrc_try_send), (gst_rtspsrc_send),
(gst_rtspsrc_parse_methods), (gst_rtspsrc_parse_range),
(gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_pause),
(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Improve timeout handling.
Use the same socket for sending and receiving RTCP packets so that some
servers can track clients better.
Improve connection closed handling. Try to reconnect.
Don't overwrite our content base with NULL.
Improve debugging.
Improve range parsing and handling.
Remove flushing hack now that core does the right thing.
Original commit message from CVS:
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init),
(gst_multiudpsink_init), (gst_multiudpsink_set_property),
(gst_multiudpsink_get_property), (gst_multiudpsink_init_send),
(gst_multiudpsink_close), (gst_multiudpsink_add):
* gst/udp/gstmultiudpsink.h:
Add support for getting and setting the socket to use.
* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init),
(gst_udpsrc_create), (gst_udpsrc_get_property):
Add support for getting the currently used socket.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init),
(gst_bpwsinc_init), (process_32), (process_64),
(bpwsinc_build_kernel), (bpwsinc_push_residue),
(bpwsinc_transform), (bpwsinc_start), (bpwsinc_query),
(bpwsinc_query_type), (bpwsinc_event), (bpwsinc_set_property):
* gst/filter/gstbpwsinc.h:
Implement latency query and only forward those samples downstream
that actually contain the data we want, i.e. drop kernel_length/2
in the beginning and append kernel_length/2 (created by convolving
the filter kernel with zeroes) to the end.
* tests/check/elements/bpwsinc.c: (GST_START_TEST):
Adjust the unit test for this slightly changed behaviour.
* gst/filter/gstlpwsinc.c: (lpwsinc_build_kernel):
Reset residue length only when actually creating a residue.
Original commit message from CVS:
* gst/udp/gstudpsrc.c: (gst_udpsrc_create):
Improve UDP performance by avoiding a select() when we have data
available immediatly.
Original commit message from CVS:
* gst/rtsp/gstrtpdec.c: (gst_rtp_dec_marshal_VOID__UINT_UINT),
(gst_rtp_dec_class_init):
* gst/rtsp/gstrtpdec.h:
Add (dummy) SSRC management signals.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
(gst_rtspsrc_set_property), (gst_rtspsrc_get_property),
(find_stream), (gst_rtspsrc_create_stream), (new_session_pad),
(request_pt_map), (gst_rtspsrc_do_stream_eos), (on_bye_ssrc),
(on_timeout), (gst_rtspsrc_stream_configure_manager),
(gst_rtspsrc_stream_push_event), (gst_rtspsrc_push_event),
(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_parse_rtpinfo),
(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Add connection-speed property.
Add find_stream helper functions.
Handle stream EOS based on BYE messages or SSRC timeout.
Returns SUCCESS from the state change function as we hide our async
elements from the parent.
Original commit message from CVS:
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init),
(gst_lpwsinc_init), (process_32), (process_64),
(lpwsinc_build_kernel), (lpwsinc_push_residue),
(lpwsinc_transform), (lpwsinc_start), (lpwsinc_query),
(lpwsinc_query_type), (lpwsinc_event), (lpwsinc_set_property):
* gst/filter/gstlpwsinc.h:
Implement latency query and only forward those samples downstream
that actually contain the data we want, i.e. drop kernel_length/2
in the beginning and append kernel_length/2 (created by convolving
the filter kernel with zeroes) to the end.
* tests/check/elements/lpwsinc.c: (GST_START_TEST):
Adjust the unit test for this slightly changed behaviour.
Original commit message from CVS:
* gst/debug/Makefile.am:
* gst/debug/breakmydata.c:
* gst/debug/gstdebug.c:
* gst/debug/negotiation.c:
* gst/debug/progressreport.c:
* gst/debug/rndbuffersize.c:
* gst/debug/testplugin.c:
Add new test element and clean-up the others a little.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (bpwsinc_build_kernel):
* gst/filter/gstlpwsinc.c: (lpwsinc_build_kernel):
Improve debugging a bit.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init),
(bpwsinc_start):
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init),
(lpwsinc_start):
Reset the residue in BaseTransform::start to get a clean residue
on stream changes.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (process_32), (process_64):
* gst/filter/gstlpwsinc.c: (process_32), (process_64):
Fix processing with buffer sizes that are larger than the filter
kernel size.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (bpwsinc_build_kernel):
Fix a segfault with more than one channel and don't rebuild
the kernel & residue with every buffer.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (gst_bpwsinc_mode_get_type),
(gst_bpwsinc_window_get_type), (gst_bpwsinc_class_init),
(gst_bpwsinc_init), (bpwsinc_build_kernel), (bpwsinc_set_property),
(bpwsinc_get_property):
* gst/filter/gstbpwsinc.h:
Add support for a bandreject mode and allow specifying the window
function that should be used.
* gst/filter/gstlpwsinc.c:
And another small formatting fix.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init),
(gst_bpwsinc_init), (process_32), (process_64),
(bpwsinc_build_kernel), (bpwsinc_setup), (bpwsinc_get_unit_size),
(bpwsinc_transform), (bpwsinc_set_property),
(bpwsinc_get_property):
* gst/filter/gstbpwsinc.h:
Apply the same changes to the bandpass filter:
- Support double input
- Fix processing for input with >1 channels
- Specify frequency in Hz
- Specify actual filter kernel length
- Use transform instead of transform_ip as we're working
out of place anyway
- Factor out filter kernel generation and update the filter
kernel when the properties are set
Fix bandpass filter kernel generation to actually generate
a bandpass filter by creating a highpass instead of a second
lowpass.
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init):
Small formatting fix.
Original commit message from CVS:
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init),
(gst_lpwsinc_init), (process_32), (process_64),
(lpwsinc_build_kernel), (lpwsinc_set_property),
(lpwsinc_get_property):
* gst/filter/gstlpwsinc.h:
Specify the actual filter length instead of a weird
2N+1. Setting the property will round to the next odd number.
Also remove now obsolete FIXMEs.
Original commit message from CVS:
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_window_get_type),
(gst_lpwsinc_class_init), (gst_lpwsinc_init),
(lpwsinc_build_kernel), (lpwsinc_set_property),
(lpwsinc_get_property):
* gst/filter/gstlpwsinc.h:
Allow choosing between hamming and blackman window. The blackman
window provides a better stopband attenuation but a bit slower
rolloff.
Original commit message from CVS:
* gst/filter/gstlpwsinc.c: (process_32), (process_64),
(lpwsinc_build_kernel):
Fix processing if the input has more than one channel.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (gst_bpwsinc_dispose),
(gst_bpwsinc_init), (bpwsinc_setup), (bpwsinc_transform_ip),
(bpwsinc_set_property), (bpwsinc_get_property):
"this" is a C++ keyword, use "self" instead.
Add TODOs and FIXMEs and remove two wrong FIXMEs.
* gst/filter/gstlpwsinc.c:
Add FIXMEs and a new TODO.
Original commit message from CVS:
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_dispose),
(gst_lpwsinc_class_init), (gst_lpwsinc_init), (process_32),
(process_64), (lpwsinc_build_kernel), (lpwsinc_setup),
(lpwsinc_get_unit_size), (lpwsinc_transform),
(lpwsinc_set_property), (lpwsinc_get_property):
* gst/filter/gstlpwsinc.h:
Add double support, replace "this" with "self" as the former
is a C++ keyword.
Implement the frequency property in Hz instead of fraction
of sampling frequency.
Remove some unecessary FIXMEs and add some TODOs, add some
required locking and refactor the kernel generation into a
separate function that is also called when the properties
change now.
And use BaseTransform::transform instead of transform_ip
as the convolution is done out of place anyway. Should
be done in place later.
Original commit message from CVS:
* gst/filter/Makefile.am:
* gst/filter/gstbpwsinc.c: (gst_bpwsinc_dispose),
(gst_bpwsinc_base_init), (gst_bpwsinc_class_init),
(gst_bpwsinc_init), (bpwsinc_setup):
* gst/filter/gstbpwsinc.h:
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_dispose),
(gst_lpwsinc_base_init), (gst_lpwsinc_class_init),
(gst_lpwsinc_init), (lpwsinc_setup):
* gst/filter/gstlpwsinc.h:
Use GstAudioFilter as base class and don't leak the memory
of the filter kernel and residue.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (get_default_rate_for_pt),
(gst_rtspsrc_parse_rtpmap), (gst_rtspsrc_media_to_caps),
(gst_rtspsrc_stream_configure_tcp),
(gst_rtspsrc_stream_configure_udp_sink):
Fix default clock-rate for realmedia.
Fix parsing of transport.
Don't try to link NULL pads.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_state_movie):
If the buffer was entirely clipped ... don't try sending it :)
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_activate_streams),
(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_parse_methods),
(gst_rtspsrc_create_transports_string),
(gst_rtspsrc_prepare_transports):
If we don't hav a session manager, set the caps on outgoing buffers
ourselves.
Force PAUSE/PLAY methods for now until the extensions can overwrite.
Append final bit of the transport string even when it does not contain a
placeholder.
Original commit message from CVS:
* gst/rtsp/gstrtspext.c: (gst_rtsp_ext_list_free),
(gst_rtsp_ext_list_connect):
* gst/rtsp/gstrtspext.h:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_finalize), (gst_rtspsrc_send_cb):
Clean up the interface list.
Allow connecting to interface signals for the extensions.
Remove old extension code.
Free list on cleanup.
Allow extensions to send additional RTSP messages.
Original commit message from CVS:
* gst/audiofx/audioamplify.c: (gst_audio_amplify_init),
(gst_audio_amplify_setup), (gst_audio_amplify_transform_ip):
* gst/audiofx/audiodynamic.c:
(gst_audio_dynamic_set_process_function), (gst_audio_dynamic_init),
(gst_audio_dynamic_setup), (gst_audio_dynamic_transform_ip):
* gst/audiofx/audiodynamic.h:
* gst/audiofx/audioinvert.c: (gst_audio_invert_init),
(gst_audio_invert_setup), (gst_audio_invert_transform_ip):
* gst/audiofx/audioinvert.h:
Don't save format information ourselves, this is already saved in
GstAudioFilter.
Original commit message from CVS:
* gst/rtsp/gstrtspext.c: (gst_rtsp_ext_list_filter),
(gst_rtsp_ext_list_stream_select):
* gst/rtsp/gstrtspext.h:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_streams):
Use rank to filter out extensions.
Add url to stream_select interface call.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_activate_segment),
(gst_qtdemux_clip_buffer), (gst_qtdemux_loop_state_movie),
(qtdemux_parse_trak), (qtdemux_video_caps), (qtdemux_audio_caps):
Clip raw audio and video when we can, keep track of current output
segment.
Don't leak buffers and events when there is no output pad.
Improve debugging here and there.
Original commit message from CVS:
* gst/equalizer/gstiirequalizer.c:
(gst_iir_equalizer_band_set_property),
(gst_iir_equalizer_child_proxy_get_child_by_index),
(gst_iir_equalizer_child_proxy_get_children_count),
(gst_iir_equalizer_child_proxy_interface_init),
(gst_iir_equalizer_class_init), (arg_to_scale), (setup_filter),
(gst_iir_equalizer_compute_frequencies):
* gst/equalizer/gstiirequalizer10bands.c:
(gst_iir_equalizer_10bands_class_init):
* gst/equalizer/gstiirequalizer3bands.c:
(gst_iir_equalizer_3bands_class_init):
* gst/equalizer/gstiirequalizernbands.c:
Better algorith for the center frequencies. Subtract band filters from
input for negative gains. Rework the gain mapping.
Original commit message from CVS:
* gst/multifile/gstmultifilesrc.c: (gst_multi_file_src_create):
Add example to the docs. Fix buffer-offset-end and add some debug.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_parse_node):
Fix parsing of esds atoms inside mp4a atoms so that we can set correct
codec_info for AAC audio. Fixes#457097 along with a whole other bunch
of qt/aac files.
Original commit message from CVS:
* gst/id3demux/gstid3demux.c: (gst_id3demux_chain):
Don't return GST_FLOW_ERROR when pushing an event returns FALSE. We
don't have enough granularity to convert that boolean into a
GstFlowReturn.
Original commit message from CVS:
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
* gst/rtp/gstrtpilbcpay.c: (gst_rtpilbcpay_setcaps):
Set the encoding-name in the rtp caps to all uppercase, as required by
the caps spec.
Some small cleanups in the error paths. Fixes#453037.
Original commit message from CVS:
* gst/multifile/Makefile.am:
* gst/multifile/gstmultifile.c:
* gst/multifile/gstmultifilesink.c:
* gst/multifile/gstmultifilesink.h:
* gst/multifile/gstmultifilesrc.c:
* gst/multifile/gstmultifilesrc.h:
Add .h files to be able to add it to the docs.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
(new_session_pad), (gst_rtspsrc_setup_streams):
* gst/rtsp/gstrtspsrc.h:
For container formats we only need to activate one of the streams so
that we correctly signal no-more-pads. Fixes#451015.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_parse_samples),
(qtdemux_video_caps):
* gst/qtdemux/qtdemux_fourcc.h:
Add MJPG to the variants of motion jpeg.
Original commit message from CVS:
* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
(rtsp_connection_close), (rtsp_connection_free):
Use threadsafe inet_ntop to convert an ip number to a string.
Fixes#447961.
Don't leak fd (and ip) when freeing a connection without first closing
it.
Original commit message from CVS:
* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
(rtsp_connection_free):
Revert previous commit again, since we are frozen (sorry).
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
(rtsp_connection_free):
inet_ntoa() uses a static buffer internally, so we need to copy the
returned string if we want to store it for later (#447961).
Original commit message from CVS:
* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
(rtsp_connection_connect):
Fix the MingW build.
Patch By: Vincent Torri <vtorri at univ-evry dot fr>
Fixes: #446981
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_parse_trak):
For AMR-NB streams, export the AMRSpecificBox as codec_data on the
caps.
Fixes#447458
Original commit message from CVS:
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_setcaps):
Make sure we allocate enough memory for the codec_data.
Fixes#447210.
Original commit message from CVS:
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
* gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_setcaps):
* gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_setcaps):
Add missing rate fields to caps. Fixes#441118.
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:
* 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:
* 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:
* 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:
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:
* 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:
* 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:
* 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.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send),
(gst_rtspsrc_setup_streams):
Ignore streams that fail the setup command, we will retry with a
different transport later on.
* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_parse_sdp),
(rtsp_ext_wms_configure_stream):
Fix encoding name case.
Original commit message from CVS:
* gst/rtp/gstrtptheoradepay.c: (decode_base64),
(gst_rtp_theora_depay_parse_configuration):
* gst/rtp/gstrtptheorapay.c: (encode_base64),
(gst_rtp_theora_pay_finish_headers),
(gst_rtp_theora_pay_handle_buffer):
Update theora pay/depayloader in a similar to vorbis.
* gst/rtp/gstrtpvorbisdepay.c:
(gst_rtp_vorbis_depay_parse_configuration):
Update docs.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send):
When we try to execute a method that is not supported by the server,
don't error out but remove the method from the accepted methods so that
we never try to perform this method again.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_range):
Parse range correctly.
* gst/rtsp/rtspurl.c: (rtsp_url_get_request_uri):
The baseurl now always has a '/' at the start.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_configure_caps),
(gst_rtspsrc_parse_range), (gst_rtspsrc_open),
(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play):
Factor out caps configuration and configure more stuff such as the time
ranges and speed/scale values.
* gst/rtsp/rtsptransport.c:
Add Copyright after non-trival fixes.
Original commit message from CVS:
* gst/level/gstlevel.c: (gst_level_transform_ip):
Use guint8 * instead of gpointer then vs6 can build
in_data += (filter->width / 8).
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/gstrtspsrc.h:
* gst/rtsp/rtspconnection.c: (rtsp_connection_receive):
* gst/rtsp/rtspmessage.c: (rtsp_message_init_data),
(rtsp_message_get_header):
* gst/rtsp/rtspmessage.h:
Make channel guint8 where possible.
Make rtsp_message_init_data() take the channel as a guint8.
* gst/rtsp/rtspdefs.c:
Fixed a typo: Timout -> Timeout
* gst/rtsp/rtspdefs.h:
Make RTSP_CHECK() behave as a statement.
* gst/rtsp/sdpmessage.c:
Avoid a compiler warning in INIT_ARRAY().
Fixes#437692.
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/rtspurl.c: (rtsp_url_parse), (rtsp_url_free),
(rtsp_url_get_request_uri):
* gst/rtsp/rtspurl.h:
Add support for query parameters to RTSP URLs.
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
(parse_range), (range_as_text), (rtsp_transport_mode_as_text),
(rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
(rtsp_transport_parse), (rtsp_transport_as_text):
* gst/rtsp/rtsptransport.h:
Add validation to rtsp_transport_parse().
Add rtsp_transport_as_text() to generate an RTSP header from an
RTSPTransport.
Change ssrc to guint (was a string) since that is what it is, even
though it is sent as a hex string.
Correctly identify PLAY|RECORD mode parameters (the syntax in the RFC is
incorrect, which can be seen when looking at the examples in the RFC).
Fixes#437670.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers):
Skip LIST chunks before the fmt chunk (fixes#437499). Also fix
streaming mode regression for file from #343837 with 'bext' chunk
before the 'fmt' chunk.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_flush),
(gst_rtspsrc_do_seek), (gst_rtspsrc_perform_seek),
(gst_rtspsrc_handle_src_event),
(gst_rtspsrc_stream_configure_manager),
(gst_rtspsrc_stream_configure_tcp), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_send_keep_alive), (gst_rtspsrc_open),
(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play):
* gst/rtsp/gstrtspsrc.h:
* gst/rtsp/rtspdefs.h:
Preliminary seek support.
Activate internal pads so that we can receive events on them.
Don't try to parse a range string when it's NULL.
Original commit message from CVS:
* gst/rtp/README:
Update README with new RTP variables that will be used for
synchronisation.
* gst/rtp/gstrtpvorbisdepay.c: (decode_base64),
(gst_rtp_vorbis_depay_parse_configuration),
(gst_rtp_vorbis_depay_process):
* gst/rtp/gstrtpvorbispay.c: (encode_base64),
(gst_rtp_vorbis_pay_finish_headers),
(gst_rtp_vorbis_pay_handle_buffer):
Update vorbis pay and depayloader to draft-04.
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:
* 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:
* 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:
* 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:
* 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/qtdemux/gstrtpxqtdepay.c: (gst_rtp_xqt_depay_process),
(gst_rtp_xqt_depay_change_state):
* gst/qtdemux/gstrtpxqtdepay.h:
Try to recover from packet loss a little better.
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/qtdemux/qtdemux.c: (gst_qtdemux_prepare_current_sample),
(gst_qtdemux_chain), (qtdemux_parse_samples):
* gst/qtdemux/qtdemux_dump.c: (qtdemux_dump_ctts):
* gst/qtdemux/qtdemux_dump.h:
* gst/qtdemux/qtdemux_fourcc.h:
* gst/qtdemux/qtdemux_types.c:
Process 'ctts' atoms, which are present in AVC ISO files (.mov files
with h264 video).
Use the offset present in 'ctts' to calculate the PTS for each packet
and set the PTS on outgoing buffers.
Fixes#423283
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:
* gst/interleave/deinterleave.c: (gst_deinterleave_sink_setcaps):
Remove 'channel-positions' field when munging input caps into
1-channel output caps (I guess technically we should set the
position for each channel on the output caps if it's non-NONE,
but I'll save that as a task for another day).
Original commit message from CVS:
* gst/interleave/deinterleave.c: (gst_deinterleave_add_new_pads),
(gst_deinterleave_remove_pads), (gst_deinterleave_process),
(gst_deinterleave_chain):
Don't leak input buffer in chain function; maintain our own list of
source pads - there are no guarantees about the order of the list
in the GstElement struct, and we want a very specific order; lastly,
some more debugging.
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:
* gst/equalizer/gstiirequalizer10bands.c:
(gst_iir_equalizer_10bands_init):
A 10 band EQ should be initialized to 1 bands and not to 3.
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:
* 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:
* gst/equalizer/Makefile.am:
* gst/equalizer/gstiirequalizer.c: (_do_init),
(gst_iir_equalizer_band_set_property),
(gst_iir_equalizer_band_get_property),
(gst_iir_equalizer_band_class_init),
(gst_iir_equalizer_band_get_type),
(gst_iir_equalizer_child_proxy_get_child_by_index),
(gst_iir_equalizer_child_proxy_get_children_count),
(gst_iir_equalizer_child_proxy_interface_init),
(gst_iir_equalizer_base_init), (gst_iir_equalizer_class_init),
(gst_iir_equalizer_finalize), (setup_filter),
(gst_iir_equalizer_compute_frequencies),
(gst_iir_equalizer_set_property), (gst_iir_equalizer_get_property),
(gst_iir_equalizer_setup), (plugin_init):
* gst/equalizer/gstiirequalizer.h:
* gst/equalizer/gstiirequalizernbands.c:
(gst_iir_equalizer_nbands_base_init),
(gst_iir_equalizer_nbands_class_init),
(gst_iir_equalizer_nbands_init),
(gst_iir_equalizer_nbands_set_property),
(gst_iir_equalizer_nbands_get_property):
* gst/equalizer/gstiirequalizernbands.h:
Refactor plugin into a base class and a first subclass (nband eq). The
nband eq uses GstChildProxy and is controlable. More subclasses will
follow.
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:
* gst/qtdemux/qtdemux.c:
* gst/qtdemux/qtdemux.h:
Share qtdemux debug category across all files, otherwise all debugging
in files other than qtdemux.c would end up in the default category.
Original commit message from CVS:
* gst/spectrum/gstspectrum.c: (gst_spectrum_start),
(gst_spectrum_event), (gst_spectrum_transform_ip):
* gst/spectrum/gstspectrum.h:
One FIXME less, by resolving message timestamps against the playback
segment.
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:
* gst/spectrum/gstspectrum.c: (gst_spectrum_class_init),
(gst_spectrum_init), (gst_spectrum_set_property),
(gst_spectrum_transform_ip):
Fix and cleanup default property values.
Add FIXMEs for stuff that looks rather wrong.
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:
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:
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:
* 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:
* 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:
* 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:
* 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:
* 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:
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:
* 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:
* 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:
* 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/equalizer/gstiirequalizer.c: (gst_iir_equalizer_class_init),
(gst_iir_equalizer_set_property), (gst_iir_equalizer_get_property),
(gst_iir_equalizer_filter_inplace):
Rename "values" property to "band-values" and change type into a
GValueArray, so it's more easily bindable and the range of the
values passed in is defined and checked etc.; also do some
locking.
Original commit message from CVS:
* gst/videocrop/gstvideocrop.c:
(gst_video_crop_get_image_details_from_caps),
(gst_video_crop_transform_packed_complex):
Fix cropping for packed 4:2:2 formats YUYV/YUY2 and UYVY.
* tests/icles/videocrop-test.c: (check_bus_for_errors),
(test_with_caps), (main):
Block streaming thread before changing filter caps while the
pipeline is running so that we don't get random not-negotiated
errors just because GStreamer can't handle that yet.
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.
Original commit message from CVS:
* gst/multifile/gstmultifilesink.c:
(gst_multi_file_sink_class_init):
* gst/multifile/gstmultifilesrc.c: (gst_multi_file_src_class_init):
* gst/mve/gstmvedemux.c: (gst_mve_video_create_buffer),
(gst_mve_video_palette), (gst_mve_video_code_map),
(gst_mve_audio_init), (gst_mve_audio_data), (gst_mve_timer_create),
(gst_mve_demux_chain):
* gst/mve/gstmvemux.c: (gst_mve_mux_push_chunk):
* gst/mve/mveaudioenc.c: (mve_compress_audio):
* gst/mve/mvevideodec16.c: (ipvideo_copy_block):
* gst/mve/mvevideodec8.c: (ipvideo_copy_block):
* gst/mve/mvevideoenc16.c: (mve_encode_frame16):
* gst/mve/mvevideoenc8.c: (mve_encode_frame8):
Use proper print statements.
Fixes build on mac os x.
<wingo> oo look at me my name is edward i'm hacking on macos wooo
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.
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.
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.
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.
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
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.
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.
Original commit message from CVS:
* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_parse_int),
(gst_rtp_mp4g_depay_setcaps), (gst_rtp_mp4g_depay_process):
Parse config params as string and int.
Parse and use AU header length
Original commit message from CVS:
* gst/smpte/barboxwipes.c: (gst_wipe_boxes_draw),
(gst_wipe_triangles_clock_draw), (gst_wipe_triangles_draw):
* gst/smpte/gstmask.c: (_gst_mask_register):
* gst/smpte/gstmask.h:
* gst/smpte/gstsmpte.c: (gst_smpte_update_mask):
* gst/smpte/paint.c: (gst_smpte_paint_hbox), (draw_bresenham_line),
(gst_smpte_paint_triangle_clock):
constify some static structs.
Don't update the mask if nothing changed to the params.
Make sure we never draw outside of the picture. Fixes#398325.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_stream_header_pull):
Error out properly when pull_range fails while we're reading the
headers, instead of just pausing the task silently. Fixes#399338.
Original commit message from CVS:
* gst/smpte/gstsmpte.c: (gst_smpte_collected):
Some more sanity checks to make sure the input formats match and the
input pads are actually negotiated, in case someone tries to feed
buffers from fakesrc or filesrc. Fixes#398299.
Also const-ify an array, just because we can.
Original commit message from CVS:
* gst/smpte/gstsmpte.c: (fill_i420), (gst_smpte_collected):
Ignore previous commit, that was only valid for widths and heights
that are multiples of 4.
Copy over size/stride macros from jpegdec. This allows the element
to work with any width,height...
... but puts in evidence that the actual transformations only work
with width/height that are multiples of 4.
Original commit message from CVS:
* gst/smpte/gstsmpte.c: (gst_smpte_collected):
Allocate buffers of the right size.
The proper size of a I420 buffer in bytes is:
width * height * 3
------------------
2
Original commit message from CVS:
* gst/smpte/gstsmpte.c: (gst_smpte_init):
Proxy getcaps on sink pads too, so that we either end up with the
same dimensions on all pads or error out if that's not possible
(seems to work even!). Fixes#398086, I think.
Original commit message from CVS:
Patch by: Sebastian Dröge <slomo circular-chaos org>
* gst/audiofx/audiopanorama.c: (gst_audio_panorama_class_init),
(gst_audio_panorama_set_process_function):
Use a function array for process methods, add more docs and define the
startindex of enums.
Original commit message from CVS:
Patch by: Mark Nauwelaerts <manauw at skynet be>
* gst/avi/gstavimux.c: (gst_avi_mux_finalize),
(gst_avi_mux_pad_reset), (gst_avi_mux_reset), (gst_avi_mux_init),
(gst_avi_mux_vidsink_set_caps), (gst_avi_mux_audsink_set_caps),
(gst_avi_mux_request_new_pad), (gst_avi_mux_release_pad),
(gst_avi_mux_riff_get_avi_header),
(gst_avi_mux_riff_get_avix_header), (gst_avi_mux_riff_get_header),
(gst_avi_mux_write_avix_index), (gst_avi_mux_add_index),
(gst_avi_mux_bigfile), (gst_avi_mux_start_file),
(gst_avi_mux_stop_file), (gst_avi_mux_handle_event),
(gst_avi_mux_do_buffer), (gst_avi_mux_do_one_buffer),
(gst_avi_mux_change_state):
* gst/avi/gstavimux.h:
* tests/check/elements/avimux.c: (teardown_src_pad):
Add support for more than one audio stream; write better AVIX
header; refactor code a bit; don't announce vorbis caps on our audio
sink pads since we don't support it anyway. Closes#379298.
Original commit message from CVS:
2007-01-13 Andy Wingo <wingo@pobox.com>
* gst/interleave/deinterleave.c (gst_deinterleave_add_new_pads):
Use fixed caps on src pads.
(gst_deinterleave_remove_pads): Remove src pads, not sink pads. I
seem to have reverse midas disease!
(gst_deinterleave_process): Proxy timestamps, offsets, durations,
and set caps on outgoing buffers. Fixes#395597, I think.
Original commit message from CVS:
2007-01-13 Andy Wingo <wingo@pobox.com>
* gst/interleave/interleave.c (gst_interleave_init): Init the
activation mode properly.
(gst_interleave_src_setcaps, gst_interleave_src_getcaps)
(gst_interleave_init): Set a setcaps and getcaps function on the
src pad, so that we can implement pull-mode negotiation.
(gst_interleave_sink_setcaps): Renamed from
gst_interleave_setcaps, as it only does the sink logic now.
Implement both for pull-mode and push-mode.
(gst_interleave_process): Set caps on our outgoing buffer.
(gst_interleave_src_activate_pull): Fix some more bogus casts.
What is up with this.
Original commit message from CVS:
* gst/apetag/gsttagdemux.c: (gst_tag_demux_read_range):
* gst/id3demux/gstid3demux.c: (gst_id3demux_read_range):
Set correct caps on outgoing pulled buffers, or things blow up
after recent core changes.
Original commit message from CVS:
Patch by: Lutz Mueller <lutz at topfrose dot de>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_activate_streams):
Check for stream pad before activating.
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/COPYING.MIT:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
(gst_rtspsrc_stream_free), (gst_rtspsrc_cleanup),
(gst_rtspsrc_alloc_udp_ports), (pad_unblocked), (pad_blocked),
(gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_loop_udp), (gst_rtspsrc_send),
(gst_rtspsrc_parse_methods),
(gst_rtspsrc_create_transports_string),
(gst_rtspsrc_prepare_transports), (gst_rtspsrc_setup_streams),
(gst_rtspsrc_open), (gst_rtspsrc_close):
* gst/rtsp/gstrtspsrc.h:
* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
(rtsp_connection_connect), (rtsp_connection_send), (read_line),
(parse_request_line), (parse_line), (rtsp_connection_read),
(rtsp_connection_close):
* gst/rtsp/rtspdefs.c: (rtsp_init_status), (rtsp_strresult),
(rtsp_method_as_text), (rtsp_header_as_text),
(rtsp_status_as_text), (rtsp_find_header_field),
(rtsp_find_method):
* gst/rtsp/rtspdefs.h:
* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_after_send),
(rtsp_ext_wms_configure_stream):
* gst/rtsp/rtspmessage.c: (rtsp_message_new), (rtsp_message_init),
(rtsp_message_new_request), (rtsp_message_init_request),
(rtsp_message_new_response), (rtsp_message_init_response),
(rtsp_message_init_data), (rtsp_message_unset),
(rtsp_message_free), (rtsp_message_add_header),
(rtsp_message_get_header), (rtsp_message_set_body),
(rtsp_message_get_body), (dump_mem), (rtsp_message_dump):
* gst/rtsp/rtspmessage.h:
* gst/rtsp/sdpmessage.c: (sdp_message_get_attribute_val_n),
(sdp_media_get_attribute_val_n), (read_string), (read_string_del),
(sdp_parse_line), (sdp_message_parse_buffer), (print_media),
(sdp_message_dump):
Allow url to be NULL to be able to use it for server connections.
Can now send responses as well as requests.
No longer hangs in an endless loop if EOF is received.
Can now convert a status code to a text string.
Return RTSP_HDR_INVALID for unknown headers.
Return RTSP_INVALID for unknown methods.
Copy CSeq and Session headers from the request.
Only free memory corresponding to the currently set message type.
Added const to function arguments as appropriate.
Avoid a compiler warning when initializing nmedia.
Use guint rather than gint to avoid compiler warnings.
Fix crasher in wms extension.
Factor out stream setup from open_connection.
Delay activation of streams when actual data is received from the
server, this prepares us to do proper protocol switching.
Added new license.
Fixes#380895.
Original commit message from CVS:
Patch by: Sebastian Dröge <slomo ubuntu com>
* docs/plugins/Makefile.am:
* gst/audiofx/audiopanorama.c:
Some small docs fixes (#394851).
Original commit message from CVS:
Patch by: Vincent Torri <vtorri at univ-evry fr>
* ext/jpeg/gstjpegdec.c:
* ext/jpeg/gstjpegenc.c:
* ext/jpeg/smokecodec.c:
These libjpeg callbacks should return a 'boolean' (unsigned char
apparently) and not a 'gboolean' (which maps to gint). Fixes
warnings when compiling with MingW (#393427).
* gst/rtsp/rtspconnection.c: (rtsp_connection_read):
Use ioctlsocket on win32.
* gst/udp/gstudpsrc.c: (gst_udpsrc_create):
Some printf format fixes for win32.
Original commit message from CVS:
2007-01-07 Andy Wingo <wingo@pobox.com>
* configure.ac:
* gst/interleave/Makefile.am:
* gst/interleave/plugin.h:
* gst/interleave/plugin.c:
* gst/interleave/interleave.c:
* gst/interleave/deinterleave.c: New elements interleave and
deinterleave, implement channel interleaving and deinterleaving.
The interleaver can operate in pull or push mode but the
deinterleaver is more like a demuxer and can only operate in push
mode.
Original commit message from CVS:
* gst/cutter/gstcutter.c: (gst_cutter_chain):
Use gst_guint64_to_gdouble for conversion.
* win32/vs6/libgstmatroska.dsp:
Add zlib to the link.
* win32/vs6/libgstvideobox.dsp:
Update liboil library name (project is linked to liboil-0.3-0.lib now).
Original commit message from CVS:
* configure.ac:
* gst/qtdemux/Makefile.am:
* gst/qtdemux/qtdemux.c: (qtdemux_parse_moov):
Check for zlib and if available pass it explicitly to the linker
when linking qtdemux. If not available (or --disable-external has
been specified!), disable the bits in qtdemux that use it. Fixes
build on MingW (#392856).
Original commit message from CVS:
* gst/matroska/Makefile.am:
If zlib is available and used, we must link it explicitly for
things to work on MingW (fixes#392855).
Original commit message from CVS:
* gst/matroska/matroska-mux.c:
(gst_matroska_mux_audio_pad_setcaps):
The "signed" field in audio caps is of boolean type, trying to use
gst_structure_get_int() to extract it will fail. Fixing this makes
matroskamux accept raw audio input (#387121) (use at your own risk
though, due to the matroska spec being not entirely useful in this
respect).
Also fix up raw audio structures in template caps so that they
represent what our setcaps function will actually accept, so that
converters know what to convert to.
Finally, don't fail if there isn't an "endianness" field in 8-bit
PCM caps.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_post_progress),
(gst_qtdemux_chain):
Don't post BUFFERING messages in streaming mode if the stream
headers are behind the movie data; instead, post "progress" element
messages as a temporary solution. Apps might get confused and do
silly things to the pipeline state if they see buffering messages
from different sources and don't realize they come from different
sources (#387160).
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_chain),
(gst_qtdemux_add_stream):
Don't output g_warning for an unsupported format, just send a
GST_ELEMENT_WARNING and don't add the pad.
Fix the case where it doesn't check for a NULL pad in streaming mode.
Fixes#387137
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_parse_trak):
Fix crash dereferencing NULL pointer if there's no stco atom.
Fixes#387122.
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/videomixer/videomixer.c: (gst_videomixer_pad_set_property),
(gst_videomixer_set_master_geometry),
(gst_videomixer_pad_sink_setcaps), (gst_videomixer_collect_free),
(gst_videomixer_reset), (gst_videomixer_init),
(gst_videomixer_finalize), (gst_videomixer_request_new_pad),
(gst_videomixer_release_pad), (gst_videomixer_collected),
(gst_videomixer_change_state):
Introduce some locking around the videomixer state so that it does not
crash when adding/removing pads. Fixes#383043.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_get_src_query_types),
(gst_qtdemux_handle_src_query), (gst_qtdemux_handle_src_event):
We don't support seeking in streaming mode, so don't even try.
Implement seeking query so apps can query seekability properly
(see #365414). Fix duration query.
Original commit message from CVS:
* gst/effectv/gstquark.c: (gst_quarktv_transform),
(gst_quarktv_planetable_clear):
Add some NULL pointer checks (possibly related to #385623).
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_get_duration),
(gst_qtdemux_handle_src_query), (qtdemux_parse_trak),
(qtdemux_audio_caps):
Add AMR-WB to the list of supported formats.
Original commit message from CVS:
* gst/apetag/gsttagdemux.c: (gst_tag_demux_chain_parse_tag),
(gst_tag_demux_chain):
* gst/id3demux/gstid3demux.c: (gst_id3demux_chain):
In streaming mode, if the first buffer we get doesn't have an
offset, fix it up to be 0, otherwise trimming won't work later on
and we'll be typefinding application/x-id3, which may result in
decodebin plugging an endless number of id3demux elements as a
consequence. Fixes#385031.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_parse_tree):
Fix non-working redirects from inetfilm.com (handle 'alis' reference
data type as well). Fixes#378613.
Original commit message from CVS:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream),
(gst_matroska_demux_video_caps):
* gst/matroska/matroska-ids.c:
(gst_matroska_track_init_video_context):
* gst/matroska/matroska-ids.h:
Try harder to extract the framerate for video tracks correctly and
save it directly instead of converting it back and forth a few
times. Mostly makes a difference for very small framerates (<1).
Fixes#380199.
Original commit message from CVS:
Patch by: Sebastian Dröge <mail at slomosnail de>
* gst/apetag/gstapedemux.c: (ape_demux_parse_tags):
We need to be able to read and parse any possible floating point string
format ("1,234" or "1.234") irrespective of the current locale. g_strod()
will parse the former only in certain locales though, so we really need
to canonicalise the separator to '.' and then use g_ascii_strtod() to
make sure we can parse either version at all times.
Fixes#382982 for real.
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler de>
* gst/qtdemux/qtdemux.c: (gst_qtdemux_get_duration),
(gst_qtdemux_handle_src_query), (qtdemux_parse_trak),
(qtdemux_audio_caps):
Fix caps for 24 bit raw PCM audio (2).
Fixes#383471.
Original commit message from CVS:
Patch by: Sebastian Dröge <mail at slomosnail de >
* gst/audiofx/audiopanorama.c: (gst_audio_panorama_init),
(gst_audio_panorama_set_caps), (gst_audio_panorama_transform):
* gst/audiofx/audiopanorama.h:
Fix audiopanorame with float samples. Fixes#383726.
Original commit message from CVS:
* gst/smpte/gstmask.h:
* gst/smpte/gstsmpte.c: (gst_smpte_class_init),
(gst_smpte_setcaps), (gst_smpte_init), (gst_smpte_reset),
(gst_smpte_collected), (gst_smpte_set_property),
(gst_smpte_get_property), (gst_smpte_change_state), (plugin_init):
* gst/smpte/gstsmpte.h:
Port to 0.10 some more.
Added duration property to specify the duration of the transition.
Make framerate a fraction.
Deprecate fps property, we only use negotiated fps.
Added docs.
Fix collectpad usage.
Reset state in READY.
Send NEWSEGMENT event.
Fix racy updates of object properties.
Added debug category.
Fixes#383323.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_get_duration),
(gst_qtdemux_handle_src_query), (qtdemux_parse_trak),
(qtdemux_video_caps):
Handle more H263 variants.
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/videomixer/videomixer.c:
(gst_videomixer_set_master_geometry),
(gst_videomixer_pad_sink_setcaps), (gst_videomixer_collect_free):
Don't reset xpos and ypos in the setcaps function because causes
unexpected behaviour.
Fixes#382179.
Original commit message from CVS:
* gst/multipart/multipartmux.c: (gst_multipart_mux_compare_pads),
(gst_multipart_mux_queue_pads), (gst_multipart_mux_collected):
Keep track of the buffer timestamp in the collectdata member instead
of modifying the buffer without making the metadata writable first.
Fixes#382277.
Original commit message from CVS:
Patch by: Rob Taylor <robtaylor at floopily dot org>
* gst/udp/gstudpsrc.c: (gst_udpsrc_start):
If using multicast in udpsrc, bind to the multicast address rather than
IN_ADDR_ANY.
This allows the simultanous use of multiple udpsrcs listening on
different multicat addresses. Without this all udpsrcs will receive all
packets from all subscribed multicast addresses.
Fixes#383001.
Original commit message from CVS:
* gst/apetag/gstapedemux.c: (ape_demux_parse_tags):
Use g_strtod() instead of sscanf to parse doubles, so that it will
try parsing in the C locale if the current locale fails.
Fixes: #382982
Patch by: Sebastian Dröge <mail at slomosnail de >
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler de>
* gst/replaygain/gstrganalysis.c: (gst_rg_analysis_event):
Call the base class handler. Fixes#380610.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_open):
* gst/rtsp/gstrtspsrc.h:
* gst/rtsp/rtspext.h:
* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_configure_stream),
(rtsp_ext_wms_get_context):
Add method so that extensions can choose to disable the setup of
a stream.
Make the WMS extension skip setup of x-wms-rtx streams. Fixes#377792.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_get_duration),
(gst_qtdemux_handle_src_query), (qtdemux_parse_trak):
Remove some asserts and replace them with a proper error
message. Fixes#379261.
Original commit message from CVS:
Patch by: Jonas Holmberg <jonas dot holmberg at axis dot com>
* gst/multipart/multipartmux.c: (gst_multipart_mux_collected):
Push header in a separate buffer instead of memcpy:ing all data
Change LF => CRLF in headers
Move trailing LF to header
Original commit message from CVS:
* gst/videobox/gstvideobox.c: (gst_video_box_class_init):
Minor clean-ups: const-ify static array, remove trailing comma from
last enum (gcc-2.9x trips over that), use GST_DEBUG_FUNCPTR.
Original commit message from CVS:
* gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame):
Make sure that g_free always gets called on the same pointer that was
returned by g_malloc. Fixes#376594.
Do not leak memory if decompressed size is wrong.
Remove unneeded check of return value of g_malloc.
Patch by: René Stadler <mail@renestadler.de>
Original commit message from CVS:
* gst/matroska/matroska-mux.c: (gst_matroska_mux_class_init),
(gst_matroska_mux_request_new_pad):
Use GST_DEBUG_FUNCPTR; activate request pad before returning it.
* tests/check/elements/matroskamux.c: (setup_src_pad),
(setup_sink_pad), (GST_START_TEST):
Activate pads before using them.
Original commit message from CVS:
Patch by: Ville Syrjala <ville.syrjala@movial.fi>
* gst/rtp/gstrtph263pay.c:
* gst/rtp/gstrtph263pdepay.c:
* gst/rtp/gstrtph263ppay.c:
Specify H.263 variant and version in the caps (fixes#361637)
Original commit message from CVS:
* gst/rtsp/rtspconnection.c: (read_body):
Don't set a data pointer to NULL and a size > 0 when we deal
with empty packets.
* gst/rtsp/rtspmessage.c: (rtsp_message_new_response),
(rtsp_message_init_response), (rtsp_message_init_data),
(rtsp_message_unset), (rtsp_message_free),
(rtsp_message_take_body):
Check that we can't create invalid empty packets.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_massage_index):
Disable init_frames delay timestamp adjustment, it does not
seem to be needed at all. Fixes#369621.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_get_duration),
(gst_qtdemux_handle_src_query), (qtdemux_parse_trak):
Don't parse extra sample params for raw pcm. Fixes#374914.
Original commit message from CVS:
* gst/flx/gstflxdec.c: (gst_flxdec_class_init):
fix categorisation, make short desc more explicit, remove unused code
Fixes#372021
Original commit message from CVS:
* gst/rtp/gstrtpvorbisdepay.c:
* gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_handle_buffer):
Fix description.
Small cleanup in the payloader.
Original commit message from CVS:
* gst/rtp/Makefile.am:
We depend on gsttag to generate the vorbis comments.
* gst/rtp/gstrtpvorbisdepay.c:
(gst_rtp_vorbis_depay_parse_configuration),
(gst_rtp_vorbis_depay_setcaps),
(gst_rtp_vorbis_depay_switch_codebook),
(gst_rtp_vorbis_depay_process):
* gst/rtp/gstrtpvorbisdepay.h:
Parse configuration string in the depayloader.
Implement selecting and switching to a new codebook.
Receiving vorbis over RTP now works.
* gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_reset_packet),
(gst_rtp_vorbis_pay_init_packet),
(gst_rtp_vorbis_pay_finish_headers),
(gst_rtp_vorbis_pay_handle_buffer):
* gst/rtp/gstrtpvorbispay.h:
Set timestamps on outgoing buffers and RTP packets.
Fix configuration string, prepend number of Packet headers.
Fix encoding of ident string.
Add delivery-method to caps.
Streaming vorbis over RTP now works.
Original commit message from CVS:
* gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_setcaps),
(gst_rtp_vorbis_pay_finish_headers), (gst_rtp_vorbis_pay_parse_id),
(gst_rtp_vorbis_pay_handle_buffer):
* gst/rtp/gstrtpvorbispay.h:
Generate a valid configuration string in the caps based on the
vorbis headers.
Original commit message from CVS:
* gst/id3demux/id3v2frames.c: (parse_picture_frame):
We require a -base more recent than 0.10.9, so it's safe to use
GST_TYPE_TAG_IMAGE_TYPE unconditionally now.
* ext/dv/gstdvdec.c: (gst_dvdec_sink_event):
* ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_sink_event):
Use _newsegment_full() now that we depend on a recent enough core.
* gst/wavparse/gstwavparse.c:
Remove cruft that we don't need any longer now that we depend on
a recent enough -base.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_get_duration),
(gst_qtdemux_handle_src_query), (qtdemux_parse_tree),
(qtdemux_parse_trak):
Handle unbounded length streams a bit better. Fixes#367696.
Original commit message from CVS:
Patch by: Michal Benes <michal dot benes at itonis tv>
* gst/matroska/matroska-demux.c: (gst_matroska_demux_encoding_cmp),
(gst_matroska_demux_read_track_encodings),
(gst_matroska_decode_buffer):
Fix several issues with encoded/compressed/encrypted/signed tracks;
also, remove superfluous newline characters from some debug
statements. (#366155)
Original commit message from CVS:
Patch by: Mark Nauwelaerts <manauw at skynet be>
* gst/videomixer/videomixer.c: (gst_videomixer_update_queues):
Fix videomixer so that it can handle any combination of framerates.
Fixes#367221.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
(gst_avi_demux_parse_file_header),
(gst_avi_demux_stream_init_push), (gst_avi_demux_parse_stream),
(gst_avi_demux_stream_header_push), (gst_avi_demux_stream_data),
(gst_avi_demux_chain):
Fix position query for audio. also fixes timestamps in streaming
mode and bug #364958.
Small cleanups.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek),
(gst_wavparse_stream_data):
Fix seeking some more, mostly for speed changes.
Original commit message from CVS:
Patch by: Josep Torra Valles <josep at fluendo com>
* gst/qtdemux/qtdemux.c: (gst_qtdemux_handle_sink_event),
(next_entry_size), (qtdemux_inflate), (qtdemux_parse_moov),
(qtdemux_parse_tree), (qtdemux_parse_trak), (qtdemux_tag_add_str),
(qtdemux_tag_add_num), (qtdemux_tag_add_date),
(qtdemux_tag_add_gnre):
Make compile with Forte compiler, mostly don't do pointer arithmetic
with void pointers (#362626).
Original commit message from CVS:
* gst/rtsp/URLS:
Added some other URL.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_udp),
(gst_rtspsrc_handle_request), (gst_rtspsrc_send),
(gst_rtspsrc_open), (gst_rtspsrc_play),
(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Work on fallback to TCP connection when the UDP socket times out.
Handler server requests, just reply with OK for now.
* gst/rtsp/rtspdefs.c: (rtsp_strresult):
* gst/rtsp/rtspdefs.h:
Added some more Real extension headers.
* gst/rtsp/rtspurl.c: (rtsp_url_parse):
Fix parsing of urls with a ':' that is not part of the hostname:port
part of the url.
Original commit message from CVS:
* gst/apetag/gsttagdemux.c: (gst_tag_demux_add_srcpad):
* gst/icydemux/gsticydemux.c: (gst_icydemux_add_srcpad):
* gst/id3demux/gstid3demux.c: (gst_id3demux_add_srcpad):
Activate pad before adding it to the already-running element.
* tests/check/elements/icydemux.c: (icydemux_found_pad):
Activate newly-created pad too.
Original commit message from CVS:
Patch by: Sebastien Cote <sebas642 at yahoo dot ca>
* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init),
(gst_udpsrc_finalize), (gst_udpsrc_create), (gst_udpsrc_set_uri),
(gst_udpsrc_start):
Fix some leaks in caps and uris. Fixes#361252.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (extract_initial_length_and_fourcc),
(gst_qtdemux_loop_state_header):
Printf format fixes.
* sys/dvb/gstdvbsrc.c:
Use "_stdint.h".
Original commit message from CVS:
* ext/faad/gstfaad.c: (gst_faad_setcaps), (gst_faad_chain),
(gst_faad_close_decoder):
Some cleanups.
Added some more debugging.
Don't ever ignore unlinked, we're not a demuxer.
* gst/qtdemux/qtdemux.c: (gst_qtdemux_add_stream):
Activate pad before adding it to the element.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtsp_proto_get_type),
(gst_rtspsrc_class_init), (gst_rtspsrc_init),
(gst_rtspsrc_create_stream), (gst_rtspsrc_media_to_caps),
(gst_rtspsrc_alloc_udp_ports),
(gst_rtspsrc_stream_configure_transport), (find_stream_by_channel),
(gst_rtspsrc_push_event), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_create_transports_string),
(gst_rtspsrc_configure_transports), (gst_rtspsrc_open),
(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Rework how the transport string is constructed, try to share channels
and udp ports.
Make most of the stuff less dependant on RTP as we are also going to use
it for RDT.
Add support for transport specific session managers.
* gst/rtsp/rtspconnection.c: (rtsp_connection_flush):
Implement _flush().
* gst/rtsp/rtspdefs.c: (rtsp_strresult):
* gst/rtsp/rtspdefs.h:
Add generic error return code.
* gst/rtsp/rtspext.h:
Add support for pluggable tranport strings.
* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_before_send),
(rtsp_ext_wms_after_send), (rtsp_ext_wms_parse_sdp),
(rtsp_ext_wms_get_context):
Detect WMServer and activate the extension.
* gst/rtsp/rtsptransport.c: (rtsp_transport_get_mime),
(rtsp_transport_get_manager), (rtsp_transport_parse):
* gst/rtsp/rtsptransport.h:
Added methods to get mime/manager for certain transports.
Original commit message from CVS:
* gst/spectrum/gstspectrum.c: (gst_spectrum_init),
(gst_spectrum_transform_ip):
Removed cruft code that was just commented out. Removed some obsolete
debug logs statements.
Original commit message from CVS:
* gst/rtsp/Makefile.am:
* gst/rtsp/gstrtpdec.c: (gst_rtpdec_getcaps),
(gst_rtpdec_chain_rtp), (gst_rtpdec_chain_rtcp):
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_finalize), (gst_rtspsrc_create_stream),
(gst_rtspsrc_parse_rtpmap),
(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_send),
(gst_rtspsrc_parse_methods), (gst_rtspsrc_open),
(gst_rtspsrc_play), (gst_rtspsrc_handle_message):
* gst/rtsp/gstrtspsrc.h:
* gst/rtsp/rtspdefs.c: (rtsp_strresult):
* gst/rtsp/rtspdefs.h:
* gst/rtsp/rtspext.h:
* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_parse_sdp),
(rtsp_ext_wms_get_context):
* gst/rtsp/rtspextwms.h:
* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
(rtsp_transport_parse):
* gst/rtsp/rtsptransport.h:
Factor out extension in separate module.
Fix getcaps to filter against the padtemplate.
Use Content-Base if the server gives one.
Rework the transport parsing a bit for future extensions.
Added some Real Header field definitions.
Original commit message from CVS:
* gst/apetag/gstapedemux.c: (ape_demux_parse_tags):
Extract disc/album/medium number and count and try harder
to extract track number/count.
Original commit message from CVS:
* gst/rtsp/URLS:
Add some more URLs.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
(gst_rtspsrc_init), (gst_rtspsrc_finalize),
(gst_rtspsrc_set_property), (gst_rtspsrc_get_property),
(gst_rtspsrc_stream_setup_rtp), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_loop_udp), (gst_rtspsrc_loop_send_cmd),
(gst_rtspsrc_loop), (gst_rtspsrc_send),
(gst_rtspsrc_parse_methods), (gst_rtspsrc_open),
(gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause),
(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Add timeout property to control UDP timeouts.
Fix error messages.
Also start a loop function when operating in UDP mode so that we can
do some more stuff async.
Handle element messages from udpsrc to detect timeouts. If a timeout
happens we currently generate an error.
API: rtspsrc::timeout property.
* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init),
(gst_udpsrc_create):
Really implement the timeout in microseconds and not milliseconds.
Original commit message from CVS:
* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init),
(gst_udpsrc_create), (gst_udpsrc_set_property),
(gst_udpsrc_get_property), (gst_udpsrc_unlock), (gst_udpsrc_stop):
* gst/udp/gstudpsrc.h:
Added property to post a message on timeout.
Updated docs.
When restarting the select, initialize the fdsets again.
Init control sockets so we don't accidentally close a random socket.
API: GstUDPSrc::timeout property
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtsp_proto_get_type):
Fix flag registration.
* gst/rtsp/rtspconnection.c: (rtsp_connection_read):
Reading 0 also means 'no more commands'
Original commit message from CVS:
Patch by: Antoine Tremblay <hexa00 at gmail dot com>
* gst/udp/gstudpsrc.c: (gst_udpsrc_create):
Fix possible infinite loop when shutting down, a read can also return
0 to indicate no more messages are available. Fixes#358156.
Original commit message from CVS:
* gst/autodetect/gstautoaudiosink.c:
(gst_auto_audio_sink_base_init), (gst_auto_audio_sink_class_init),
(gst_auto_audio_sink_find_best):
* gst/autodetect/gstautovideosink.c: (gst_auto_video_sink_detect):
Small cleanups.
don't try to set "sync" property when it is not available.
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/alpha/gstalpha.c:
* gst/rtp/gstrtpamrdepay.c:
* gst/rtsp/gstrtspsrc.c:
* gst/udp/gstudpsrc.c:
* gst/videomixer/videomixer.c:
Include stdlib.h in some more places, makes things compile
with uClibc and -Werror (#357592).
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_stream_index):
Don't check for a tag that is never there and check if we read the
correct tag. Fixes seeking again.
We must post an error when all pads are unlinked.
Original commit message from CVS:
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_process):
* gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_setcaps),
(gst_rtp_vorbis_pay_reset_packet),
(gst_rtp_vorbis_pay_init_packet),
(gst_rtp_vorbis_pay_flush_packet), (gst_rtp_vorbis_pay_parse_id),
(gst_rtp_vorbis_pay_handle_buffer):
More fixage, set endoder-params correctly in the payloader.
Original commit message from CVS:
* gst/autodetect/gstautoaudiosink.c:
(gst_auto_audio_sink_base_init):
* gst/autodetect/gstautovideosink.c:
(gst_auto_video_sink_base_init):
Make static pad templates static to appease valgrind's leak
detector.
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/autodetect.c: (GST_START_TEST),
(autodetect_suite):
Add simple test for the ghostpad lockup on shutdown fixed in core
CVS (audio bit disabled because it would need dozens of alsa
suppressions and I'm too lazy to add those now).
Original commit message from CVS:
* gst/rtp/README:
Update README with some examples.
* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_init),
(gst_rtp_mp4g_pay_finalize), (gst_rtp_mp4g_pay_parse_audio_config),
(gst_rtp_mp4g_pay_parse_video_config), (gst_rtp_mp4g_pay_new_caps),
(gst_rtp_mp4g_pay_setcaps):
* gst/rtp/gstrtpmp4gpay.h:
Make optional RTP parameters of type STRING, as required by the
application/x-rtp caps specification.
Original commit message from CVS:
* gst/rtp/gstrtph263pdepay.c:
* gst/rtp/gstrtph263ppay.c:
Correctly calculate size of each H263+ RTP buffer taking into account MTU and
RTP header.
Original commit message from CVS:
* gst/rtsp/URLS:
Added some test URLS.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
(gst_rtspsrc_loop), (gst_rtspsrc_open):
* gst/rtsp/gstrtspsrc.h:
When creating streams, give access to the complete SDP.
Fix some leaks.
Collect and merge global stream properties in stream caps.
Preliminary support for WMServer.
* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
(rtsp_connection_connect), (rtsp_connection_read), (read_body),
(rtsp_connection_receive):
* gst/rtsp/rtspconnection.h:
Make connection interruptable.
Refactor to make it reconnectable.
Don't fail on short reads when reading data packets.
* gst/rtsp/rtspurl.c: (rtsp_url_parse), (rtsp_url_set_port),
(rtsp_url_get_port):
* gst/rtsp/rtspurl.h:
Add methods for getting/setting the port.
* gst/rtsp/sdpmessage.c: (sdp_message_get_attribute_val_n),
(sdp_message_get_attribute_val), (sdp_media_get_attribute),
(sdp_media_get_attribute_val_n), (sdp_media_get_attribute_val),
(sdp_media_get_format), (sdp_parse_line),
(sdp_message_parse_buffer):
Fix headers.
Add methods for getting multiple attributes with the same name.
Increase buffer size when parsing.
Fix parsing of a=foo fields.
* gst/rtsp/test.c: (main):
Update to new connection API.
* gst/rtsp/rtspmessage.c: (rtsp_message_new_response),
(rtsp_message_init_response), (rtsp_message_init_data),
(rtsp_message_unset), (rtsp_message_free), (rtsp_message_dump):
* gst/rtsp/rtspmessage.h:
* gst/rtsp/rtsptransport.c: (rtsp_transport_free):
* gst/rtsp/rtsptransport.h:
* gst/rtsp/sdp.h:
* gst/rtsp/sdpmessage.h:
* gst/rtsp/gstrtsp.c:
* gst/rtsp/gstrtsp.h:
* gst/rtsp/gstrtpdec.c:
* gst/rtsp/gstrtpdec.h:
* gst/rtsp/rtsp.h:
* gst/rtsp/rtspdefs.c:
* gst/rtsp/rtspdefs.h:
Dual licensed under MIT and LGPL now.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (find_stream_by_pt),
(gst_rtspsrc_create_stream), (gst_rtspsrc_free_stream),
(gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_stream_configure_transport), (find_stream_by_channel),
(gst_rtspsrc_push_event), (gst_rtspsrc_loop), (gst_rtspsrc_send),
(gst_rtspsrc_parse_methods), (gst_rtspsrc_open),
(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play):
* gst/rtsp/gstrtspsrc.h:
Reorganize stream parsing and creation.
Detect container formats in interleaved mode.
Keep more state about the streams.
Assume a server also supports PLAY if it does not say.
Add unicast and interleaved properties to TCP transport requests to make
some servers happy (WMServer).
* gst/rtsp/sdpmessage.h:
Add some defines for the standard Bandwidth types.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_base_init),
(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_push_event),
(gst_rtspsrc_loop), (gst_rtspsrc_uri_set_uri):
* gst/rtsp/gstrtspsrc.h:
Export sometimes source pad with correct caps on the template, create
the ghostpad from the template.
Remove RTCP template as we never expose RTCP.
Protect against invalid body size.
Avoid memcpy when creating the output buffer.
Properly post an error and send EOS when the loop function is shut down.
Original commit message from CVS:
Based on patch by: Lutz Mueller <lutz at topfrose dot de>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
(gst_rtspsrc_init), (gst_rtspsrc_set_property), (gst_rtspsrc_open),
(gst_rtspsrc_uri_get_uri), (gst_rtspsrc_uri_set_uri):
* gst/rtsp/gstrtspsrc.h:
Make sure we can never set an invalid location.
* gst/rtsp/rtspmessage.c: (rtsp_message_steal_body):
* gst/rtsp/rtspmessage.h:
Added _steal_body method for future use.
* gst/rtsp/rtspurl.c: (rtsp_url_parse), (rtsp_url_free):
Make freeing of NULL url return immediatly.
Original commit message from CVS:
Based on patch by: Lutz Mueller <lutz at topfrose dot de>
* gst/rtsp/gstrtspsrc.c: (_do_init), (gst_rtspsrc_class_init),
(gst_rtspsrc_init), (gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_play),
(gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Use boilerplate.
Make rtspsrc subclass GstBin to make state changes easier.
Add Range header field on the PLAY request.
Original commit message from CVS:
Based on patch by: Thijs Vermeir <thijs dot vermeir at barco dot com>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_set_state),
(gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_open),
(gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause):
* gst/rtsp/rtspconnection.c: (inet_aton):
Small cleanups.
when multicast is selected as the transport, create UDP sources and
connect to the multicast group.
Move parsing and setting of caps to a common place.
Fixes#349894.
Original commit message from CVS:
Patch by: Yves Lefebvre <ivanohe at abacom dot com>
* gst/avi/gstavimux.c: (gst_avi_mux_stop_file):
Correctly set the dwLength in strh.
With this patch, the file duration is now displayed correctly in window
media player and the AVI plays completely. Fixes#356147
Original commit message from CVS:
* gst/multipart/multipartdemux.c: (gst_multipart_demux_chain):
Fix documentation, it is not possible to control the framerate of jpegdec
using filtered caps yet. Fixes#355210.
Return the downstream GstFlowReturn instead of GST_FLOW_OK so that we
stop when there is an error.
Original commit message from CVS:
* gst/apetag/gsttagdemux.c: (gst_tag_demux_chain_parse_tag):
* gst/id3demux/gstid3demux.c: (gst_id3demux_chain):
Don't interpret a first buffer with an offset of NONE as
'from the middle of the stream', but only a first buffer
that has a valid buffer offset that's non-zero (see #345449).
Original commit message from CVS:
* gst/icydemux/gsticydemux.c: (gst_icydemux_reset),
(gst_icydemux_typefind_or_forward):
* gst/icydemux/gsticydemux.h:
When we merge/collect multiple incoming buffers for typefinding
purposes, keep an initial 0 offset on the first outgoing buffer
as well (otherwise id3demux won't work right). Fixes#345449.
Also Make buffer metadata writable before setting buffer caps.
* tests/check/elements/icydemux.c: (typefind_succeed),
(cleanup_icydemux), (push_data), (GST_START_TEST),
(icydemux_suite):
Small test case for the above.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_peek_chunk),
(gst_avi_demux_stream_index), (gst_avi_demux_sync),
(gst_avi_demux_stream_header_push),
(gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data),
(gst_avi_demux_loop):
More code reuse and better logging in _peek_chunk(). Reintroduce check
for chunk sizes before reading them (avoid oom). Better handling for
invalid chunksizes when streaming.
Original commit message from CVS:
* gst/spectrum/gstspectrum.c: (gst_spectrum_class_init),
(gst_spectrum_start), (gst_spectrum_stop), (gst_spectrum_event):
Implements stop() to clear the adapter and event() to clear the
adapter on FLUSH_STOP and EOS.
Original commit message from CVS:
* gst/level/gstlevel.c: (gst_level_set_property):
* gst/level/gstlevel.h:
Fix type mixup in level->interval (gdouble<->guint64). Spotted by
René Stadler
Original commit message from CVS:
* gst/spectrum/gstspectrum.c: (gst_spectrum_init),
(gst_spectrum_set_property):
* gst/spectrum/gstspectrum.h:
Fix type mixup in spectrum->interval (gdouble<->guint64). Spotted by
René Stadler
Original commit message from CVS:
* gst/spectrum/demo-osssrc.c: (draw_spectrum), (main):
Use more defines
* gst/spectrum/gstspectrum.c: (gst_spectrum_init),
(gst_spectrum_dispose), (gst_spectrum_set_caps),
(gst_spectrum_transform_ip):
* gst/spectrum/gstspectrum.h:
Apply some of the spectrum cleanup changes suggested in #348085.
Original commit message from CVS:
* configure.ac:
Bump requirements of -base (videocrop test case needs this).
* gst/videocrop/gstvideocrop.c:
Document sloppy handling of subsampled chroma planes if
left/top cropping is an odd number.
* tests/check/elements/videocrop.c: (handoff_cb),
(videocrop_test_cropping_init_context),
(videocrop_test_cropping_deinit_context),
(videocrop_test_cropping), (check_1x1_buffer), (GST_START_TEST),
(videocrop_suite), (main):
Add another unit test that crops the input to 1x1 (and checks
that that pixel has the expected values in a number of formats).
Original commit message from CVS:
* gst/videocrop/Makefile.am:
* gst/videocrop/gstvideocrop.c: (gst_video_crop_class_init),
(gst_video_crop_transform_packed),
(gst_video_crop_transform_planar):
Some quick tests indicate that it doesn't make a great deal
of sense to use liboil here, at least not for the memcpy()s
we do, so remove liboil usage until there is clear evidence
it actually makes a positive difference somewhere.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_read_subindexes_pull),
(gst_avi_demux_sync), (gst_avi_demux_stream_header_push),
(gst_avi_demux_stream_data):
Revert one change to fix streaming avi (adapter size != data size).
Original commit message from CVS:
Patch by: Frédéric Riss <frederic.riss at gmail dot com>
* gst/matroska/matroska-demux.c: (gst_matroska_track_free),
(gst_matroska_demux_reset),
(gst_matroska_demux_read_track_encodings),
(gst_matroska_demux_add_stream), (gst_matroska_decode_buffer),
(gst_matroska_demux_parse_blockgroup_or_simpleblock),
(gst_matroska_demux_subtitle_caps):
* gst/matroska/matroska-ids.h:
Add support for VOBSUB subtitle tracks and zlib-compressed
tracks. Make sure we start on a keyframe after a seek. (#343348)
Original commit message from CVS:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_push_hdr_buf),
(gst_matroska_demux_push_flac_codec_priv_data),
(gst_matroska_demux_push_xiph_codec_priv_data),
(gst_matroska_demux_parse_blockgroup_or_simpleblock),
(gst_matroska_demux_video_caps), (gst_matroska_demux_audio_caps):
* gst/matroska/matroska-ids.h:
Add basic FLAC support (#311586), not perfect yet though, needs some
tweaking in flacdec; also, seeking could be better.
Do better bounds checking when deserialising vorbis stream headers
to make sure we don't read beyond the end of the buffer on bad input.
Original commit message from CVS:
* configure.ac:
* gst/videocrop/Makefile.am:
* gst/videocrop/gstvideocrop.c: (gst_video_crop_base_init),
(gst_video_crop_class_init), (gst_video_crop_init),
(gst_video_crop_get_image_details_from_caps),
(gst_video_crop_get_unit_size), (gst_video_crop_transform_packed),
(gst_video_crop_transform_planar), (gst_video_crop_transform),
(gst_video_crop_transform_dimension),
(gst_video_crop_transform_dimension_value),
(gst_video_crop_transform_caps), (gst_video_crop_set_caps),
(gst_video_crop_set_property), (gst_video_crop_get_property),
(plugin_init):
Port/rewrite videocrop from scratch for GStreamer-0.10, and make
it support all formats videoscale supports (#345653).
Original commit message from CVS:
* gst/qtdemux/qtdemux.c:
(gst_qtdemux_do_seek):
Reset each streams last_flow to GST_FLOW_OK.
(gst_qtdemux_activate_segment):
Removing mystic modifications for good.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_activate_segment),
(qtdemux_parse_tree):
put back 'segment start<=stop' change that was mystically reverted by
the last commit
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_activate_segment),
(gst_qtdemux_add_stream), (qtdemux_parse_trak),
(qtdemux_video_caps):
Make sure segment start<=stop in weird quicktime files.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_class_init),
(gst_avi_demux_init), (gst_avi_demux_finalize),
(gst_avi_demux_reset), (gst_avi_demux_index_last),
(gst_avi_demux_index_next), (gst_avi_demux_index_entry_for_time),
(gst_avi_demux_parse_subindex), (gst_avi_demux_parse_index),
(gst_avi_demux_stream_index), (gst_avi_demux_peek_tag),
(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_stream_header_pull), (gst_avi_demux_do_seek),
(gst_avi_demux_process_next_entry), (gst_avi_demux_loop),
(gst_avi_demux_chain), (gst_avi_demux_sink_activate),
(gst_avi_demux_change_state):
* gst/avi/gstavidemux.h:
More attempts to turn this into readable code.
Don't leak adapters.
Calculate duration according to index more efficiently.
Don't try to act like we drive the pipeline in chain mode.
Original commit message from CVS:
* gst/audiofxgood/audiopanorama.c: (gst_audio_panorama_class_init),
(gst_audio_panorama_set_property),
(gst_audio_panorama_get_property),
(gst_audio_panorama_transform_m2s_int),
(gst_audio_panorama_transform_s2s_int),
(gst_audio_panorama_transform_m2s_float),
(gst_audio_panorama_transform_s2s_float):
* gst/audiofxgood/audiopanorama.h:
* tests/check/elements/audiopanorama.c: (GST_START_TEST):
Make also the pan-property float (saves scaling and yields better
resolution)
Original commit message from CVS:
* gst/audiofxgood/audiopanorama.c: (gst_audio_panorama_set_caps),
(gst_audio_panorama_transform_m2s_float),
(gst_audio_panorama_transform_s2s_float):
ChangeLog surgery to add cymax's real name
Original commit message from CVS:
* gst/audiofxgood/audiopanorama.c:
(gst_audio_panorama_transform_m2s):
Fix docs & debug category. Add Fixme for volume pan levels.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_read_subindexes_pull),
(gst_avi_demux_sync), (gst_avi_demux_stream_header_push),
(gst_avi_demux_stream_header_pull),
(gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data),
(gst_avi_demux_chain):
unbreak AVI index handling, some more debug, remove an obsolete
adapter_flush that caused streaming to wander off in the wild
Original commit message from CVS:
* gst/avi/gstavidemux.c: (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_odml),
(gst_avi_demux_parse_index), (gst_avi_demux_stream_index),
(gst_avi_demux_calculate_durations_from_index),
(gst_avi_demux_stream_header_push),
(gst_avi_demux_stream_header_pull):
* gst/avi/gstavidemux.h:
Some more cleanups.
Fix totalFrames parsing in ODML.
Disable use of index for length calculation in case of ODML as this is
broken now.
Original commit message from CVS:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
There is no taglibmux element ...
* gst/rtsp/gstrtspsrc.c:
Use '%' rather than '&perc;' in gtk-doc blurb, docs build
was complaining about unknown entity here.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
(gst_avi_demux_do_seek), (gst_avi_demux_handle_seek),
(gst_avi_demux_process_next_entry):
* gst/avi/gstavidemux.h:
Mark DISCONT.
Remove old unused fields and reorder the struct a bit.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_reset),
(gst_avi_demux_index_entry_for_time),
(gst_avi_demux_handle_src_query), (gst_avi_demux_handle_src_event),
(gst_avi_demux_stream_init), (gst_avi_demux_parse_stream),
(gst_avi_demux_stream_index), (gst_avi_demux_peek_tag),
(gst_avi_demux_next_data_buffer),
(gst_avi_demux_calculate_durations_from_index),
(gst_avi_demux_stream_header), (gst_avi_demux_do_seek),
(gst_avi_demux_handle_seek), (gst_avi_demux_aggregated_flow),
(gst_avi_demux_process_next_entry), (gst_avi_demux_loop),
(gst_avi_demux_sink_activate_pull), (gst_avi_demux_change_state):
* gst/avi/gstavidemux.h:
Precalc most of the duration query for each stream.
Make seeking more correct.
Use GstSegment to track position and duration.
Code cleanups and leak fixes.
Calculate correct total duration based on index length.
Original commit message from CVS:
* gst/id3demux/id3v2frames.c: (parse_text_identification_frame),
(parse_insert_string_field):
If strings in text fields are marked ISO8859-1, but contain
valid UTF-8 already, then handle them as UTF-8 and ignore
the encoding. (#351794)
Original commit message from CVS:
* gst/monoscope/gstmonoscope.c: (gst_monoscope_chain):
Don't unref buffers of which we've already given away
ownership to the adapter.
Original commit message from CVS:
* gst/audiopanorama/.cvsignore:
* gst/audiopanorama/Makefile.am:
* gst/audiopanorama/audiofx.c:
* gst/audiopanorama/audiopanorama.c:
* gst/audiopanorama/audiopanorama.h:
die! die! die! you should never have been there
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_add_stream), (qtdemux_parse),
(qtdemux_node_dump_foreach), (qtdemux_parse_trak),
(qtdemux_video_caps), (qtdemux_audio_caps):
Some more constification.
Fix some paletted data formats again.
Fix ulaw/alaw in qt.
Set correct caps for raw RGB.
Add support for yuv2, which is like Yuv2.
Add support for raw audio with the NONE fourcc, which is like raw.
Original commit message from CVS:
* ext/gdk_pixbuf/pixbufscale.c: (gst_pixbufscale_get_unit_size):
* gst/videobox/gstvideobox.c: (gst_video_box_get_unit_size):
use g_assert in _get_unit_size
Original commit message from CVS:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* docs/plugins/inspect/plugin-audiofxgood.xml:
cleanup -unused.txt to make it useful, add previously missing docs
* ext/Makefile.am:
* ext/esd/esdmon.c:
* ext/esd/esdsink.c:
* ext/esd/gstesd.c: (plugin_init):
reflow to get rid of two external symbols
* gst/audiofxgood/audiofx.c: (plugin_init):
re-add
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/multipart/multipartdemux.c: (multipart_parse_header):
Accept leading whitespace before the boundary
This patch makes the demuxer allow some whitespace before the actual
boundary. This makes the demuxer work with the ``old'' gstreamer
multipartmuxer again (which placed an extra \n before the start
of the stream) Fixes#349068.
Original commit message from CVS:
* configure.ac:
Require CVS of GStreamer core and -base (for
GST_TAG_EXTENDED_COMMENT and gst_tag_parse_extended_comment()).
* ext/taglib/gstid3v2mux.cc:
Write extended comment tags properly (#348762).
* gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame),
(parse_comment_frame):
Extract COMM frames into extended comments, which makes it
easier to properly retain the description bit of the tag
and maintain this information when re-tagging (#348762).
Original commit message from CVS:
* gst/autodetect/gstautoaudiosink.c:
(gst_auto_audio_sink_find_best):
When we can't find a usable audiosink, don't error out,
but use a fake sink instead and post a warning message
on the bus (#341278).
Original commit message from CVS:
* gst/rtp/gstrtpamrdepay.c:
* gst/rtp/gstrtpmp4gdepay.c:
Caps extra properties must be defined as strings for
depayloaders because they are generated from an SDP.
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_base_init),
(gst_rtp_h264_depay_class_init), (gst_rtp_h264_depay_init),
(gst_rtp_h264_depay_finalize), (decode_base64),
(gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process),
(gst_rtp_h264_depay_set_property),
(gst_rtp_h264_depay_get_property),
(gst_rtp_h264_depay_change_state),
(gst_rtp_h264_depay_plugin_init):
* gst/rtp/gstrtph264depay.h:
Added basic, not completely functional RFC 3984 H264 depayloader.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_redirects_sort_func),
(qtdemux_process_redirects), (qtdemux_parse_tree):
Extract all references/redirections if there is more
than one and sort them; also extract minimum required
bitrate information if available. (#350399)
Original commit message from CVS:
Patch by: Edward Hervey <edward@fluendo.com>
* configure.ac:
* gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek),
(gst_wavparse_stream_data):
Send the newsegment event in the streaming thread.
Fixes#347529
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/multipart/multipartdemux.c: (gst_multipart_demux_base_init),
(gst_multipart_demux_class_init), (gst_multipart_demux_init),
(gst_multipart_demux_finalize), (get_line_end),
(multipart_parse_header), (multipart_find_boundary),
(gst_multipart_demux_chain), (gst_multipart_demux_change_state),
(gst_multipart_set_property), (gst_multipart_get_property):
Uses GstAdapter instead of own buffering.
Actually parses the mime-type correctly (In tests the mime-type was
always "" with the old version).
Uses the Content-length header if available to speed up things.
Reliably autoscans the boundary name by default.
Fixes#349068.
* gst/multipart/multipartmux.c: (gst_multipart_mux_collected):
Don't start the stream with a \n.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_stream_index),
(gst_avi_demux_massage_index):
* gst/avi/gstavidemux.h:
Whitespace fixes and more debug
Original commit message from CVS:
* gst/autodetect/gstautoaudiosink.c:
(gst_auto_audio_sink_create_element_with_pretty_name),
(gst_auto_audio_sink_find_best),
(gst_auto_audio_sink_change_state):
Get rid of old and unused magic sound-server properties stuff.
Add suffix to child sink's name that makes it easy to see from
the name alone which type it actually is (alsa, oss, esd, etc.).
Original commit message from CVS:
* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init),
(gst_udpsrc_set_property), (gst_udpsrc_get_property),
(gst_udpsrc_start):
* gst/udp/gstudpsrc.h:
Rename "buffer" to "buffer-size" to make clear it is a size we set and
not some sort of feature we enable.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c:
#define 'fact' RIFF chunk if we are not compiling against
-base CVS (we don't want to depend on -base CVS for this
one define only, and also not for release order reasons).
Original commit message from CVS:
* gst/id3demux/id3tags.c:
(id3demux_add_id3v2_frame_blob_to_taglist):
Extract frames for ID3v2 versions prior to ID3v2.3.0 properly as
well, and add the version to the blob's buffer caps, since that
information will be needed for deserialisation later on (#348644).
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_read_subindexes),
(gst_avi_demux_parse_stream):
Moved win32 variant of GST_DEBUG_CATEGORY_EXTERN to gstinfo.h. Fixed
indentation and spacing.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_reset),
(gst_wavparse_other), (gst_wavparse_perform_seek),
(gst_wavparse_get_upstream_size), (gst_wavparse_stream_headers),
(gst_wavparse_add_src_pad), (gst_wavparse_stream_data),
(gst_wavparse_pad_query):
* gst/wavparse/gstwavparse.h:
Use information from 'fact' chunk for length calculation of compressed
samples. Calculate bps if bogus value is found in wav header (embeded
mp2/mp3).
Original commit message from CVS:
* gst/id3demux/gstid3demux.c: (plugin_init):
* gst/id3demux/id3tags.c:
(id3demux_add_id3v2_frame_blob_to_taglist):
* gst/id3demux/id3tags.h:
On second thought, it might be wiser and more efficient
not to do tag registration from a streaming thread.
Original commit message from CVS:
* gst/id3demux/id3tags.c:
(id3demux_add_id3v2_frame_blob_to_taglist),
(id3demux_id3v2_frames_to_tag_list):
Put ID3v2 frames we can't parse as binary blobs into private
tags, so that they are not lost when retagging, at least once
id3v2mux has been taught to re-inject those frames again.
See bug #334375.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
(gst_avi_demux_process_next_entry):
Fix some leaks.
* gst/id3demux/id3tags.c: (id3demux_id3v2_frames_to_tag_list):
Don't use \n in debug lines.
Original commit message from CVS:
* gst/avi/gstavimux.c: (gst_avi_mux_vidsink_set_caps):
Respect mpegversion for "video/mpeg" and give message in case of
unhandled versions.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_perform_seek),
(gst_qtdemux_prepare_current_sample),
(gst_qtdemux_loop_state_movie):
Store duration in uint64 too instead of clipping.
When we do a keyframe seek and the requested time is at the
keyframe, don't seek back to the beginning of the keyframe.
Fixes#347439.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_reset),
(gst_avi_demux_stream_header), (push_tag_lists):
* gst/avi/gstavidemux.h:
Don't push tag events found by gst_riff_parse_info() before outputting
GST_EVENT_NEWSEGMENT.
Original commit message from CVS:
* gst/rtsp/Makefile.am:
* gst/rtsp/rtspconnection.c: (rtsp_connection_send),
(rtsp_connection_close):
* gst/rtsp/rtspdefs.h:
replaced closesocket and close in code with one CLOSE_SOCKET.
Some more cleanups. Fixes#345301.
Original commit message from CVS:
Patch by: Rob Taylor <robtaylor at floopily dot org>
* gst/udp/gstmultiudpsink.c: (join_multicast),
(gst_multiudpsink_init_send), (gst_multiudpsink_add):
If a destination is added before the stream is set to PAUSED, the
multicast group is not joined as the socket is not created yet.
Also TTL and LOOP should also be set. Fixes#346921.
Original commit message from CVS:
* gst/apetag/gsttagdemux.c: (gst_tag_demux_trim_buffer),
(gst_tag_demux_read_range):
* gst/id3demux/gstid3demux.c: (gst_id3demux_trim_buffer),
(gst_id3demux_read_range):
Don't return FLOW_UNEXPECTED when a buffer is before
the start of the stream (which might happen with
large ID3v2 tags if the tag reading was done pullrange
based and we then switched to push mode later on).
Fixes regression introduced by commit from June 29th.
Original commit message from CVS:
* gst/apetag/gsttagdemux.c: (gst_tag_demux_chain):
* gst/id3demux/gstid3demux.c: (gst_id3demux_chain):
Return FLOW_UNEXPECTED when at the end of the file, not
FLOW_ERROR. Fixes 'internal stream error' errors that
would sometimes occur in totem when scrubbing to the
end of an ID3v1 tagged mp3 file.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_reset),
(gst_avi_demux_read_subindexes), (gst_avi_demux_parse_stream),
(gst_avi_demux_handle_seek), (gst_avi_demux_aggregated_flow),
(gst_avi_demux_process_next_entry), (push_tag_lists),
(gst_avi_demux_stream_data), (gst_avi_demux_loop):
* gst/avi/gstavidemux.h:
Proper aggregation of each stream's GstFlowReturn in order to figure out
whether the task should stop or not.
Don't send inline events before pushing out a NEW_SEGMENT, more
specifically for GST_TAG_EVENT.
Change a GST_ERROR to a GST_WARNING for a non-fatal situation in reading
sub-indexes.
Original commit message from CVS:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream),
(gst_matroska_demux_send_event),
(gst_matroska_demux_loop_stream_parse_id):
* gst/matroska/matroska-ids.h:
Send tag event after newsegment event.
Original commit message from CVS:
* gst/id3demux/gstid3demux.c: (gst_id3demux_trim_buffer),
(gst_id3demux_read_range):
Make sure we don't return GST_FLOW_OK with a NULL buffer in
certain cases where a read beyond the end of the file is
requested. Fixes#345930.
* gst/apetag/gsttagdemux.c: (gst_tag_demux_trim_buffer),
(gst_tag_demux_read_range):
Fix same issue here as well.
Original commit message from CVS:
* gst/matroska/matroska-demux.c:
(gst_matroska_demux_check_subtitle_buffer),
(gst_matroska_demux_parse_blockgroup_or_simpleblock),
(gst_matroska_demux_subtitle_caps):
* gst/matroska/matroska-ids.c:
(gst_matroska_track_init_subtitle_context):
* gst/matroska/matroska-ids.h:
Try to fix up broken matroska files containing subtitle
streams with non-UTF8 character encodings (courtesy of
mkvmerge) using either the encoding specified in the
GST_SUBTITLE_ENCODING environment variable or the
current locale's character set if it is non-UTF8.
Fixes#337076.
Original commit message from CVS:
* gst/id3demux/id3v2frames.c: (parse_picture_frame):
Set image type from APIC frame as "image-type" field
of GST_TAG_IMAGE buffer caps (#344605).
Original commit message from CVS:
* gst/rtsp/rtspconnection.c: (inet_aton), (rtsp_connection_send),
(rtsp_connection_close), (rtsp_connection_free):
Use better G_OS_* macros. Fixes#345301 some more.
Original commit message from CVS:
Patch by: Joni Valtanen <joni dot valtanen at movial dot fi>
* gst/rtsp/rtspconnection.c: (inet_aton), (rtsp_connection_send),
(rtsp_connection_close):
Make RTSP plugin compile on windows. Fixes#345301.
Some changes to original patch to catch errors better.
use ifdef WIN32 instead of ifndef.
Original commit message from CVS:
Patch by: Mark Nauwelaerts <manauw at skynet be>
* gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek):
Make sure we don't read beyond the end of the file (#345232).
Original commit message from CVS:
* gst/spectrum/demo-audiotest.c: (on_window_destroy),
(draw_spectrum), (message_handler), (main):
* gst/spectrum/demo-osssrc.c: (on_window_destroy), (draw_spectrum),
(message_handler), (main):
port to use message to get results, cleanly exit when closing the window
* gst/spectrum/gstspectrum.c: (gst_spectrum_class_init),
(gst_spectrum_init), (gst_spectrum_dispose),
(gst_spectrum_set_property), (gst_spectrum_get_property),
(gst_spectrum_set_caps), (gst_spectrum_start),
(gst_spectrum_message_new), (gst_spectrum_transform_ip):
* gst/spectrum/gstspectrum.h:
port to derive from basetransform and send results via messages
(like level element)
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_perform_seek),
(gst_qtdemux_combine_flows), (gst_qtdemux_loop_state_movie),
(gst_qtdemux_loop), (gst_qtdemux_chain), (qtdemux_parse_trak):
Combine return values from src pad pushes.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_state_header),
(gst_qtdemux_prepare_current_sample), (gst_qtdemux_advance_sample),
(gst_qtdemux_add_stream):
Don't crash on files with 0 samples, EOS immediatly instead.
Fixes#344944.
Original commit message from CVS:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream):
No language specified means the implied language is English
according to the matroska spec (partially fixes#344708);
add some more debug output.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_peek_chunk_info),
(gst_wavparse_peek_chunk), (gst_wavparse_stream_headers),
(gst_wavparse_chain):
When operating chain-based, don't make any assumptions about the
chunking of the incoming data and make streaming work on days other
than the second Thursday after a full moon. Also fix up debug
messages here and there and make use of the most excellent new
gst_pad_query_peer_duration() utility function.
Skip any 'bext' chunks in front of the 'fmt ' chunk. Fixes#343837.
* gst/wavparse/gstwavparse.h:
Remove trailing comma after last enum value, some compilers don't
like that.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_handle_seek):
Prevent out of bounds array access when scrubbing towards
the end of the file between the last index entry and the
end. Fixes occasional 'start <= stop' newsegment event
assertions when scrubbing in MJPEG files.
Original commit message from CVS:
* gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame),
(scan_encoded_string), (parse_picture_frame):
Extract images from ID3v2 tags (APIC frames). Fixes#339704.
* configure.ac:
Require core >= 0.10.8 (for GST_TAG_IMAGE and
GST_TAG_PPEVIEW_IMAGE used in the patch above).
Original commit message from CVS:
* gst/apetag/gsttagdemux.c: (gst_tag_demux_get_upstream_size):
* gst/id3demux/gstid3demux.c: (id3demux_get_upstream_size):
Use gst_pad_query_peer_duration() utility function here.
Original commit message from CVS:
* gst/avi/gstavidemux.c:
add an explicit dll imported declaration for GST_CAT_EVENT+WIN32
* win32/MANIFEST:
sort file listing
* win32/vs6/libgstavi.dsp:
add gstavimux.c to the project
* win32/vs6/libgstid3demux.dsp:
add link to zlib library
* win32/vs6/libgstmatroska.dsp:
add matroska-ids.c to the project
Original commit message from CVS:
* gst/alpha/gstalphacolor.c: (gst_alpha_color_transform_caps):
* gst/debug/negotiation.c: (gst_negotiation_update_caps):
* gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_setcaps):
GST_PTR_FORMAT should be used to print caps in debug statements.
Original commit message from CVS:
Patch by: Sebastian Dröge <slomo at ubuntu dot com>
* gst/apetag/gstapedemux.c: (ape_demux_get_gst_tag_from_tag),
(ape_demux_parse_tags):
Some clean-ups and additions: map APE 'file' tag to
GST_TAG_LOCATION (#343123); add support for extracting
the track count and clean up parsing a bit (#343127).
Original commit message from CVS:
* gst/id3demux/id3v2frames.c: (id3v2_tag_to_taglist):
A track/volume number or count of 0 does not make sense,
just ignore it along with negative numbers (a tag might
only contain a track count without a track number).
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_activate_segment):
Clip the outputed NEWSEGMENT stop time to the configured segment stop
time.
Original commit message from CVS:
Patch by: Mark Nauwelaerts <manauw at skynet be>
* gst/matroska/matroska-mux.c: (gst_matroska_mux_change_state):
gst_collect_pads_stop() needs to be called before chaining up
to the parent class (#342734).
Original commit message from CVS:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream),
(gst_matroska_demux_push_xiph_codec_priv_data),
(gst_matroska_demux_parse_blockgroup_or_simpleblock),
(gst_matroska_demux_video_caps), (gst_matroska_demux_audio_caps):
* gst/matroska/matroska-ids.h:
* gst/matroska/matroska-mux.c: (gst_matroska_mux_base_init),
(gst_matroska_mux_video_pad_setcaps),
(xiph3_streamheader_to_codecdata),
(vorbis_streamheader_to_codecdata),
(theora_streamheader_to_codecdata),
(gst_matroska_mux_audio_pad_setcaps),
(gst_matroska_mux_write_data):
Add support for muxing/demuxing theora video (#342448; too bad
none of the usual linux players can actually play this). Playback
in GStreamer will require additional changes to theoradec in -base.
Refactor streamheaders <=> CodecPrivateData code a bit; some small
cleanups.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_parse_trak), (plugin_init):
po/POTFILES.in:
Throw an error when the file is encrypted. Move plugin_init stuff
to the end of the file, add stuff for i18n, make debug category
static.
Original commit message from CVS:
* gst/spectrum/gstspectrum.c: (gst_spectrum_class_init),
(gst_spectrum_init), (gst_spectrum_set_sink_caps),
(gst_spectrum_get_sink_caps), (gst_spectrum_chain):
Use boilerplate macro, fix strings to match plugin-moval-requirements
Original commit message from CVS:
* gst/spectrum/Makefile.am:
Link to base libraries
* gst/spectrum/demo-osssrc.c: (main):
use new threshhold property
* gst/spectrum/gstspectrum.c: (gst_spectrum_class_init),
(gst_spectrum_init), (gst_spectrum_dispose),
(gst_spectrum_set_property), (gst_spectrum_set_sink_caps),
(gst_spectrum_get_sink_caps), (gst_spectrum_chain),
(gst_spectrum_change_state):
* gst/spectrum/gstspectrum.h:
Use gst_adapter, support multiple-channels, add threshold property for
result, add docs, fix resulting spectrum range (was including mirrored
results)
Original commit message from CVS:
* gst/id3demux/id3v2frames.c: (id3v2_tag_to_taglist):
Don't output any tag when we encounter a negative track number - the
tag type is uint, so we end up outputting huge positive numbers
instead. (Fixes: #342029)
Original commit message from CVS:
2006-05-18 Philippe Kalaf <philippe.kalaf at collabora.co.uk>
* rtp/gst/gstrtph263pay.c:
Properly set static caps for H263 at 34.
Original commit message from CVS:
* gst/law/mulaw-decode.c: (mulawdec_getcaps):
We can only do caps intersection if the othercaps are non-empty and not
ANY. Else we return the pad template (base_caps).
Original commit message from CVS:
* gst/autodetect/gstautoaudiosink.c:
(gst_auto_audio_sink_find_best):
* gst/autodetect/gstautovideosink.c:
(gst_auto_video_sink_find_best):
Make the name of the child element be based on the name of the
parent, so that debug output is more useful.
* gst/id3demux/id3v2frames.c: (find_utf16_bom),
(parse_insert_string_field), (parse_split_strings):
Rework string parsing to always walk over BOM markers in UTF16
strings, using the endianness indicated by the innermost one,
then trying the opposite endianness if that fails to convert
to valid UTF-8. Fixes#341774
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_fmt),
(gst_wavparse_perform_seek), (gst_wavparse_stream_headers):
Fix use of uninitialised values if we're NOT seeking in ready.
Fix typos.
Original commit message from CVS:
* gst/id3demux/id3v2frames.c: (parse_insert_string_field):
Some more debug info. No need to check whether the string
returned by g_convert() is really UTF-8 - either it is or
we get NULL returned.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_parse), (qtdemux_parse_trak),
(gst_qtdemux_handle_esds):
Figure out the real audio type in mp4a boxes by parsing the
optional descriptors in the optional esds box. Promote the
default AAC to mp3 when indicated. Fixes#330632.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_parse), (qtdemux_dump_unknown),
(qtdemux_parse_trak), (gst_qtdemux_handle_esds):
Parse version 2 sample descriptions.
Don't #define gst_util_dump_mem(), use something more
specific instead to avoid confusion.
Original commit message from CVS:
* gst/id3demux/id3v2frames.c: (id3v2_genre_fields_to_taglist):
Fix parsing of numeric genre strings some more, by ensuring that
we only try and parse strings that a) Start with '(' and b) Consist
only of digits.
Also, when finding an escaping '((' sequence, bust it back to '(' by
swallowing the first parenthesis
Original commit message from CVS:
Patch by: Mark Nauwelaerts <manauw at skynet bet>
* gst/avi/gstavimux.c: (gst_avi_mux_do_audio_buffer),
(gst_avi_mux_do_video_buffer):
Work around gst_buffer_make_metadata_writable() bug that
results in avimux marking all frames in the index as
keyframes (#340859).
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_handle_src_query),
(qtdemux_dump_mvhd):
Don't cause side effects in a debugging function.
Also report duration in push mode since we can.
Original commit message from CVS:
Patch by: Michal Benes <michal dot benes at xeris dot cz>
* gst/matroska/matroska-demux.c: (gst_matroska_demux_reset):
Don't leak caps when freeing the stream context (#340623).
Original commit message from CVS:
* gst/matroska/matroska-mux.c:
(gst_matroska_mux_stream_is_vorbis_header),
(gst_matroska_mux_write_data):
Don't strcmp() NULL strings.
Only start new clusters on video keyframes, not on any
random audio buffer that doesn't have the DELTA_UNIT
flag set (fixes 'make check' again).
Original commit message from CVS:
Patch by: Mark Nauwelaerts <manauw at skynet be>
* gst/matroska/matroska-mux.c: (gst_matroska_mux_best_pad),
(gst_matroska_mux_stream_is_vorbis_header),
(gst_matroska_mux_write_data):
Don't misinterpret GST_CLOCK_TIME_NONE as very high timestamp
value and then dead-lock when muxing vorbis audio streams
(the three vorbis header buffers carry no timestamp, and it
would try to mux these after all video buffers). Fixes#340346.
Improve clustering: start a new cluster also whenever we get
a keyframe.
Original commit message from CVS:
* gst/id3demux/gstid3demux.c: (gst_id3demux_chain),
(gst_id3demux_sink_activate):
Let core insert default error message for TYPE_NOT_FOUND
errors, it's just as good as our own and has the added
bonus of being translated.
Original commit message from CVS:
* gst/apetag/gsttagdemux.c: (gst_tag_demux_init),
(gst_tag_demux_sink_event):
* gst/id3demux/gstid3demux.c: (gst_id3demux_init),
(gst_id3demux_sink_event):
Post an error message when we get an EOS event and were not
able to find out the type of stream.
* tests/check/elements/id3v2mux.c: (fill_mp3_buffer), (got_buffer),
(test_taglib_id3mux_with_tags):
Decrease num-buffers to 16 per iteration again, otherwise the
many memcpy()s and reallocations in the test will hammer slow
CPUs completely and make the test timeout.
Original commit message from CVS:
* gst/id3demux/id3v2frames.c: (has_utf16_bom),
(parse_split_strings):
Recognise and skip any byte order marker (BOM) in
UTF-16 strings.
Original commit message from CVS:
* gst/auparse/gstauparse.c: (gst_au_parse_base_init),
(gst_au_parse_class_init), (gst_au_parse_init),
(gst_au_parse_reset), (gst_au_parse_add_srcpad),
(gst_au_parse_remove_srcpad), (gst_au_parse_parse_header),
(gst_au_parse_chain), (gst_au_parse_src_convert),
(gst_au_parse_src_query), (gst_au_parse_handle_seek),
(gst_au_parse_sink_event), (gst_au_parse_src_event),
(gst_au_parse_change_state):
* gst/auparse/gstauparse.h:
Rewrite auparse to suck a little bit less: make source pad
dynamic, so decodebin/playbin work with non-raw formats
like alaw/mulaw; add query function for duration/position
queries; check whether we have enough data before attempting
to parse the header (instead of crashing when that is not the
case); work around audioconvert sucking by swapping endianness
to the native endianness ourselves for float formats; send
initial newsegment event. Fixes#161712.
Original commit message from CVS:
* gst/matroska/Makefile.am:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream),
(gst_matroska_demux_handle_src_event):
* gst/matroska/matroska-ids.c:
(gst_matroska_track_init_video_context),
(gst_matroska_track_init_audio_context),
(gst_matroska_track_init_subtitle_context),
(gst_matroska_track_init_complex_context):
* gst/matroska/matroska-ids.h:
Handle case where the TrackType ebml chunk does not come before the
TrackInfoAudio or TrackInfoVideo ebml chunk (#339446). Ignore QoS
events.
Original commit message from CVS:
* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_setcaps):
* gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_setcaps):
It's codec_data, not codec_info.
Original commit message from CVS:
Patch by: Mark Nauwelaerts <manauw at skynet dot be>
* gst/matroska/matroska-demux.c: (gst_matroska_demux_video_caps):
Handle codec_data for VfW compatibility codec IDs (#339451)
* gst/matroska/matroska-mux.c:
(gst_matroska_mux_video_pad_setcaps):
Same here, handle codec_data and add additional caps we can handle
now to the pad template (huffyuv, dv and h263 video) (#339451)
Original commit message from CVS:
Patch by: Josef Zlomek <josef dot zlomek at itonis dot tv>
* gst/matroska/matroska-mux.c:
(gst_matroska_mux_create_buffer_header),
(gst_matroska_mux_write_data):
Fix timestamping of B-frames, use signed integers, do
some rounding (#339678).
Original commit message from CVS:
* gst/matroska/matroska-demux.c: (gst_matroskademux_do_index_seek):
Fix a bad conversion using gst_guint64_to_gdouble.
fabs ((gdouble) demux->index[entry].time - (gdouble) seek_pos) can not be
replaced by fabs (gst_guint64_to_gdouble (demux->index[entry].time - seek_pos)) as the
difference could be negative. fabs (gst_guint64_to_gdouble (demux->index[entry].time) -
gst_guint64_to_gdouble (seek_pos)) is the good solution. Thanks to Tim who has seen my
mistake.
Original commit message from CVS:
* gst/matroska/matroska-demux.c: (gst_matroskademux_do_index_seek):
Use gst_guint64_to_gdouble for conversions
* win32/vs6/gst_plugins_good.dsw:
* win32/vs6/libgsticydemux.dsp:
Add a project file for icydemux
Original commit message from CVS:
Patch by: Fabrizio Gennari <fabrizio dot ge at tiscali dot it>
* gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
(gst_avi_demux_parse_index), (gst_avi_demux_massage_index):
When splitting audio chunks, the block alignment is not taken in
consideration, so the smaller chunks could be of size which is
not a multiple of the block alignment. Fixes#336904
Original commit message from CVS:
* gst/debug/progressreport.c: (gst_progress_report_finalize),
(gst_progress_report_class_init), (gst_progress_report_init),
(gst_progress_report_do_query), (gst_progress_report_report),
(gst_progress_report_set_property),
(gst_progress_report_get_property):
Add 'format' property to force querying to a particular format.
Original commit message from CVS:
Patch by: Fabrizio Gennari <fabrizio dot ge at tiscali dot it>
* gst/avi/gstavidemux.c: (gst_avi_demux_peek_tag),
(gst_avi_demux_next_data_buffer), (gst_avi_demux_stream_scan):
Fix index creation when we have to scan the file to create
an index. There may be other types of RIFF 'LIST' chunks than
'movi' and we need to skip them properly as well or we'll end up
reading garbage (#336889). Some other cosmetic changes.
Original commit message from CVS:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream),
(gst_matroskademux_do_index_seek),
(gst_matroska_demux_handle_seek_event),
(gst_matroska_demux_parse_blockgroup_or_simpleblock):
* gst/matroska/matroska-ids.h:
Set DISCONT flag on first buffer after a discontinuity.
Fix newsegment events sent when seeking and honour KEY_UNIT
seek flag. Create pad with bogus caps if we don't recognise
the stream codec id.
* gst/matroska/matroska-demux.h:
Fix GObject macros.
Original commit message from CVS:
Patch by: Mark Nauwelaerts <manauw at skynet dot be>
* gst/matroska/matroska-demux.c:
(gst_matroska_demux_handle_seek_event), (gst_matroska_demux_loop):
Handle end of segment properly when set; don't dead-lock when
posting start of segment message when doing a segment seek.
Fixes#338810.
Original commit message from CVS:
Patch by: j^ <j at bootlab dot org>
* gst/qtdemux/qtdemux.c: (qtdemux_parse_trak),
(qtdemux_video_caps):
Never treat video streams as an audio stream.
Add qtdrw mime type.
Fixes#339041
Original commit message from CVS:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_audio_caps),
(gst_matroska_demux_plugin_init):
Make mpeg2 aac audio work: create artificial private codec data
chunk which faad2 seems to require, just as we do for mpeg4 aac.
Also call gst_riff_init(). Partially fixes#338767.
Original commit message from CVS:
* gst/wavenc/gstwavenc.c: (gst_wavenc_base_init),
(gst_wavenc_class_init), (gst_wavenc_init),
(gst_wavenc_create_header_buf), (gst_wavenc_push_header),
(gst_wavenc_sink_setcaps), (get_id_from_name), (gst_wavenc_event),
(gst_wavenc_chain), (gst_wavenc_change_state):
* gst/wavenc/gstwavenc.h:
Set caps on first outgoing buffer, so that it doesn't error out
immediately with a non-negotiated error (#338716). Rewrite and
clean up a bit; fix setcaps function to parse things properly;
fix sink caps (8bit audio is unsigned and doesn't have depth);
use boilerplate macros; remove unused properties stuff.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_parse_trak):
For VBR audio, don't try to calculate the samples_per_frame.
Fixes#338935.
Original commit message from CVS:
2006-04-12 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
* 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
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c:
* gst/rtp/gstrtpilbcpay.c:
* gst/rtp/gstrtpilbcpay.h:
* gst/rtp/gstrtpilbcdepay.c:
* gst/rtp/gstrtpilbcdepay.h:
Added new iLBC payloader/depayloader. Payloader uses new audio payload base
class.
Original commit message from CVS:
* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_process):
Don't leak memory allocated by gst_buffer_new_and_alloc() by
overwriting GST_BUFFER_MALLOCDATA.
Original commit message from CVS:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_base_init),
(gst_matroska_demux_handle_seek_event),
(gst_matroska_demux_video_caps), (gst_matroska_demux_audio_caps),
(gst_matroska_demux_subtitle_caps),
(gst_matroska_demux_plugin_init):
Use static pad templates with ANY caps for audio and video
source pads and get rid of a lot of unnecessary (and partially
broken) code for the template caps. Clean up caps finding
functions. Fixes playback of audio files/streams that do not
contain the sample rate and/or number of channels in the audio
context (happens a lot with vorbis/mp3 .mka files it seems).
Fixes#337183.
Also add myself to copyright holders.
Original commit message from CVS:
Patch by: Ryan Lortie (desrt) <desrt at destr dot ca>
* gst/avi/gstavidemux.c: (gst_avi_demux_parse_superindex),
(gst_avi_demux_parse_stream), (gst_avi_demux_parse_index),
(gst_avi_demux_stream_header):
Fix some crashers with empty chunks. (Fixes#337749)
Original commit message from CVS:
* gst/level/gstlevel.c: (gst_level_set_caps),(gst_level_transform_ip):
use G_GINT64_CONSTANT for INT64 constants
* gst/videofilter/gstvideobalance.c:
define rint for WIN32 #define rint(x) (floor((x)+0.5))
* win32/vs6/libgstavi.dsp:
add missing libraries for the link and remove avimux.c from
the project as it isn't ported to 0.10 yet
Original commit message from CVS:
* gst/matroska/ebml-read.c: (gst_ebml_read_sint):
Even better would be if we actually did the right thing
here (also, G_GUINT64_CONSTANT only exists since GLib-2.10).
Original commit message from CVS:
* gst/matroska/ebml-read.c: (gst_ebml_read_sint):
Can't just replace 1LL with 1L here just because MSVC doesn't
support it, as it might lead to incorrect results when doing the
bitshifting here. Using GLib's G_GUINT64_CONSTANT() macro to
force a 64-bit constant in a way that all compilers are happy with.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_prepare_current_sample),
(gst_qtdemux_chain), (gst_qtdemux_add_stream), (qtdemux_dump_stsz),
(qtdemux_dump_stco), (qtdemux_parse_trak):
Don't make rounding errors in timestamp/duration calculations.
Fix timestamps for AMR and IMA4. Fixes (#337436).
Create a dummy segment even when there is no edit list.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream):
Don't unref the GstPadTemplate returned by
gst_element_class_get_pad_template().
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_perform_seek),
(gst_qtdemux_do_seek), (gst_qtdemux_loop_state_movie),
(gst_qtdemux_loop):
Use duration as segment stop position if none is
explicitly configured.
Also perform EOS when we run past the segment stop.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_go_back),
(gst_qtdemux_perform_seek), (gst_qtdemux_do_seek),
(gst_qtdemux_loop_state_movie), (gst_qtdemux_loop),
(gst_qtdemux_chain), (qtdemux_parse_tree), (qtdemux_parse_trak):
More cleanups, added comments.
Mark discontinuities on outgoing buffers.
Post better errors when something goes wrong.
Handle EOS and segment end properly.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_init),
(gst_qtdemux_push_event), (gst_qtdemux_go_back),
(gst_qtdemux_perform_seek), (gst_qtdemux_do_seek),
(gst_qtdemux_handle_src_event), (plugin_init),
(gst_qtdemux_change_state), (gst_qtdemux_loop_state_movie),
(gst_qtdemux_loop), (gst_qtdemux_chain),
(qtdemux_sink_activate_pull), (gst_qtdemux_add_stream),
(qtdemux_parse), (qtdemux_parse_tree), (qtdemux_parse_trak),
(qtdemux_parse_udta), (qtdemux_tag_add_str), (qtdemux_tag_add_num),
(qtdemux_tag_add_gnre), (gst_qtdemux_handle_esds):
* gst/qtdemux/qtdemux.h:
Handle stss boxes so we can mark and find keyframes.
Implement correct accurate and keyframe seeking.
Use _DEBUG_OBJECT when possible.
Original commit message from CVS:
* ext\jpeg\smokecodec.c:
use of GST_DEBUG instead of DEBUG(a...) for WIN32
* ext\speex\gstspeexenc.c: (gst_speexenc_set_header_on_caps):
move first instruction after all variables declarations
* gst\alpha\gstalpha.c:
* gst\effectv\gstshagadelic.c:
* gst\smpte\paint.c:
* gst\videofilter\gstvideobalance.c:
define M_PI if it's not defined (it's not defined on WIN32)
* gst\cutter\gstcutter.c: (gst_cutter_chain):
* gst\id3demux\id3v2frames.c: (parse_relative_volume_adjustment_two):
* gst\level\gstlevel.c: (gst_level_set_property), (gst_level_transform_ip):
* gst\matroska\matroska-demux.c: (gst_matroska_demux_parse_info),
(gst_matroska_demux_video_caps):
* gst\matroska\matroska-mux.c: (gst_matroska_mux_start), (gst_matroska_mux_finish):
* gst\wavparse\gstwavparse.c: (gst_wavparse_stream_data):
use gst_guint64_to_gdouble for conversions
* gst\goom\filters.c: (setPixelRGB_):
fix a debug which was using undefined variable
* gst\level\gstlevel.c: (gst_level_set_caps), (gst_level_transform_ip):
* gst\matroska\ebml-read.c: (gst_ebml_read_sint):
replace LL suffix with L suffix (LL isn't supported by MSVC6.0)
* win32/vs6:
add vs6 projects files for most of plugins-good
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_parse_index),
(gst_avi_demux_stream_index), (gst_avi_demux_stream_scan),
(gst_avi_demux_massage_index), (gst_avi_demux_handle_seek):
this patch combines the global init_frames with the stream
init_frames. Rationale being that the global delay should
be subtracted from any stream delay.
Fixes#335858.
Original commit message from CVS:
* ext/speex/gstspeexdec.c: (speex_dec_change_state):
* gst/interleave/deinterleave.c: (deinterleave_change_state):
* gst/interleave/interleave.c: (interleave_change_state):
* gst/wavenc/gstwavenc.c: (gst_wavenc_change_state):
More state change function fixes.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_get_upstream_size),
(gst_wavparse_stream_headers), (gst_wavparse_stream_data):
Don't try to read beyond the end of the file just because
the header claims a bigger size (like with truncated files).
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_loop):
* gst/wavparse/gstwavparse.h:
Delay source pad creation until we have the first chunk of
media data, so the we can examine the data and adjust the
caps accordingly if required. This makes playback of .wav
files with DTS-declared-as-PCM content work (#313266).
Original commit message from CVS:
* gst/apetag/gsttagdemux.c: (gst_tag_demux_chain):
* gst/id3demux/gstid3demux.c: (gst_id3demux_chain):
Don't attempt typefinding on too-short buffers that have been
completely trimmed away.
* gst/id3demux/id3tags.c: (id3demux_read_id3v2_tag):
Improve the debug output
Original commit message from CVS:
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
* gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek):
Fix block alignment calculation. Alignment should be done before
adding the byte offset where the data starts (#335231).
Original commit message from CVS:
* gst/matroska/ebml-write.c: (gst_ebml_write_element_push):
Ensure that we set correct caps on buffers that are transferred
direct from the input.
Original commit message from CVS:
* gst/goom/filters.c: (zoomFilterDestroy):
* gst/goom/goom_core.c: (goom_close):
Free filter data when cleaning up. (Fixes: #334995)
Original commit message from CVS:
* gst/id3demux/id3v2frames.c:
(parse_relative_volume_adjustment_two):
We only care about gain and peak data for the master volume.
Original commit message from CVS:
* gst/apetag/gsttagdemux.c: (gst_tag_demux_chain):
Ensure that we set caps on the buffers we pass.
* gst/id3demux/gstid3demux.c: (gst_id3demux_chain),
(gst_id3demux_sink_activate):
Ensure that we set caps on the buffers we pass.
Use STREAM, TYPE_NOT_FOUND as the error class when
typefinding fails.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_class_init),
(gst_qtdemux_init), (gst_qtdemux_dispose),
(gst_qtdemux_add_stream), (qtdemux_parse_trak):
Series of memleak fixes:
- Unref the GstAdapter in finalize.
- Use gst_pad_new_from_static_template(), shorter and safer.
- Free unused QtDemuxStream when not used.
Original commit message from CVS:
* configure.ac:
Bump -base requirement to 0.10.5 for gst_tag_from_id3_user_tag(),
used by id3demux.
* gst/id3demux/gstid3demux.c: (plugin_init):
* gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame),
(parse_user_text_identification_frame),
(parse_unique_file_identifier):
Add support for UFID and TXXX frames and extract musicbrainz tags.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_process_next_entry):
Catch short reads, like they might happen with truncated
files (see #305279); remove unnecessary indentation.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_reset),
(gst_wavparse_change_state):
Implement seek in READY (re-fixes #327658)
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_parse_udta):
Extract disc number and count from files that use
'disk' instead of 'disc' as node identifier for that
(fixes#332066).
Original commit message from CVS:
* gst/id3demux/Makefile.am:
* gst/id3demux/gstid3demux.c: (gst_id3demux_add_srcpad),
(gst_id3demux_chain), (gst_id3demux_sink_activate):
Use new typefind helper functions here as well, and
do typefinding in pull-mode if upstream supports that.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
(gst_avi_demux_handle_src_event), (gst_avi_demux_parse_stream),
(gst_avi_demux_calculate_durations_from_index),
(gst_avi_demux_stream_header):
* gst/avi/gstavidemux.h:
If we have an index, use a duration based on the index instead
of blindly trusting the information in the stream headers
(fixes#331817).
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_parse), (qtdemux_parse_trak):
Use GST_WARNING instead of GST_ERROR for all the too short/long atoms
when parsing.
Also let's be a bit less vulgar in our warning messages :)
Original commit message from CVS:
* configure.ac:
Bump requirements to current core and -base CVS
(core for new typefind helper API, and -base for the
WAVFORMATEX support that was added to libgstriff and
is needed by wavparse).
* gst/apetag/Makefile.am:
* gst/apetag/gsttagdemux.c: (gst_tag_demux_chain),
(gst_tag_demux_sink_activate):
Use new typefind helpers for typefinding instead of our
home-grown stuff; also, do typefinding in pull-mode if
upstream supports that.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_parse_trak):
Can't divide through zero (suppress warning in case of
stream with one single still picture) (see #327083)
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek),
(gst_wavparse_stream_headers), (gst_wavparse_stream_data),
(gst_wavparse_pad_convert), (gst_wavparse_srcpad_event),
(gst_wavparse_sink_activate), (gst_wavparse_sink_activate_pull):
Use DEBUG_OBJECT more.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_add_stream),
(qtdemux_parse_trak), (qtdemux_video_caps):
Add support for palettised Apple SMC videos (#327075, based on
patch by: Fabrizio Gennari <fabrizio dot ge at tiscali dot it>).
Original commit message from CVS:
* ext/cdio/Makefile.am:
Add GST_BASE_CFLAGS and GST_BASE_LIBS (seems to be
required for Cygwin, see #317048)
* gst/rtp/gstasteriskh263.c:
Cygwin has includes for both the unix network socket API
and the windows API, but only one can be included, so fix
includes to only use one or the other, prefering the unxi
one (#317048).
Original commit message from CVS:
2006-02-23 Philippe Kalaf <philippe.kalaf at collabora.co.uk>
* rtp/gst/gstrtppcmadepay.c:
* rtp/gst/gstrtppcmadepay.h:
* rtp/gst/gstgstrtppcmapay.c:
* rtp/gst/gstgstrtppcmapay.h:
* rtp/gst/gstrtppcmudepay.c:
* rtp/gst/gstrtppcmudepay.h:
* rtp/gst/gstrtppcmupay.c:
* rtp/gst/gstrtppcmupay.h:
* rtp/gst/Makefile.am:
* rtp/gst/gstrtp.c:
* rtp/gst/README:
Separated the G711 payloaders/depayloaders into separate elements for
mulaw/alaw. Also removed the old g711 payloaders/depayloaders.
Original commit message from CVS:
Reviewed by : Edward Hervey <edward@fluendo.com>
* gst/qtdemux/qtdemux.c: (qtdemux_video_caps):
Add 'dvsd' and 'dv25' to list of possible fourcc values for DV Video.
Add image/png for fourcc 'png '
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_src_convert),
(gst_avi_demux_handle_src_query), (gst_avi_demux_handle_src_event),
(gst_avi_demux_parse_file_header), (gst_avi_demux_stream_init),
(gst_avi_demux_parse_avih), (gst_avi_demux_parse_superindex),
(gst_avi_demux_parse_subindex), (gst_avi_demux_parse_stream),
(gst_avi_demux_stream_header), (gst_avi_demux_change_state):
Use scaling code for added precission and more correct stop
position in case scale==0.
Original commit message from CVS:
* gst/flx/flx_color.h:
* gst/flx/flx_fmt.h:
* gst/flx/gstflxdec.c: (gst_flxdec_init),
(gst_flxdec_src_query_handler), (flx_decode_color),
(gst_flxdec_chain):
* gst/flx/gstflxdec.h:
Set MALLOCDATA for the temp buffers so we don't leak.
Some debug cleanups.
Consume all data in the adapter before leaving the chain
function. Fixes#330678.
Original commit message from CVS:
* gst/id3demux/id3tags.c: (id3demux_id3v2_frames_to_tag_list):
* gst/id3demux/id3v2frames.c: (id3v2_genre_fields_to_taglist):
Handle 0 data size in otherwise valid frames.
Handle numeric strings in 2.4.0 even when not in parentheses
Original commit message from CVS:
* gst/matroska/matroska-demux.c:
(gst_matroska_demux_subtitle_caps),
(gst_matroska_demux_plugin_init):
* gst/matroska/matroska-ids.h:
Recognise SSA/ASS and USF subtitle formats and
set proper caps when they are found.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_state_movie):
Don't GST_LOG timestamps from nonexistent index
entries (#331582).
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_state_header):
Check that the size of the returned buffer is of the correct size
because the parser assumes that.
Fixes#331543.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers):
Pass extra_data to gst_riff_create_audio_caps(), so that
WAVEFORMATEX stuff works. Post audio codec name and post
it as taglist on the bus. Allow up to 8 channesl for raw
PCM in the source pad template caps.
Original commit message from CVS:
* gst/multipart/multipartdemux.c: (gst_multipart_demux_base_init),
(gst_multipart_demux_class_init), (gst_multipart_demux_init),
(gst_multipart_demux_finalize), (gst_multipart_find_pad_by_mime),
(gst_multipart_demux_chain), (gst_multipart_demux_change_state),
(gst_multipart_set_property), (gst_multipart_get_property):
Applied #318663. Gives quite a few false positives in
autoscan mode, but it's better than nothing. Not closing yet.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_handle_src_event),
(gst_qtdemux_loop), (qtdemux_sink_activate_pull):
Don't stop the task if the pad isn't linked.
Original commit message from CVS:
* gst/id3demux/id3tags.c: (id3demux_id3v2_frames_to_tag_list):
ID3 2.3.0 used synch-safe integers for the tag size, but not for the
frame size. (Fixes#331368)
Original commit message from CVS:
* gst/rtsp/README:
Updated README.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_get_type),
(gst_rtspsrc_class_init), (gst_rtspsrc_set_property),
(gst_rtspsrc_get_property), (gst_rtspsrc_stream_setup_rtp):
* gst/rtsp/gstrtspsrc.h:
Make sure the RTP port is an even port an try to allocate
another if not.
Added retry property to control max retries for port allocation.
Make sure RTCP port is RTP port+1.
Cleanup when port allocation fails.
Fixes#319183.
Original commit message from CVS:
* gst/alpha/gstalpha.c: (gst_alpha_change_state):
Don't ignore return value of the parent class's state
change function (#331385, patch by: Wouter Paesen).
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_event):
Add comment in a fultile attempt to stop the copy-and-paste
paradigm leading to duplication of bad code.
* gst/rtsp/rtsptransport.c: (rtsp_transport_parse):
Mime parameters have to be checked case insensitive
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_post_buffering),
(gst_qtdemux_chain):
When buffering MDAT data, show the user something is
happening by posting 'buffering' messages on the bus.
Original commit message from CVS:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_sync_streams):
Advance stream time for lagging subtitle streams by sending
newsegment events with the update flag set.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_init),
(gst_qtdemux_handle_src_query), (gst_qtdemux_change_state),
(next_entry_size), (gst_qtdemux_chain):
* gst/qtdemux/qtdemux.h:
Make push-based work if mdat atom is before moov atom.
Don't answer duration query. This should be transformed into replying
FALSE to seek events.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_stream_header):
There can be bogus data before the hdrl LIST tag in the RIFF header.
It's hard to say if it's not respecting the AVI specifications or not,
but since Google Video is producing AVIs like that and the other player
don't seem to complain, I guess we should do the same.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (next_entry_size), (gst_qtdemux_chain):
Handle the case where data atoms are before moov atoms in push-based mode.
Errors out gracefully.
Original commit message from CVS:
* gst/qtdemux/Makefile.am:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_init),
(gst_qtdemux_handle_sink_event), (gst_qtdemux_change_state),
(extract_initial_length_and_fourcc),
(gst_qtdemux_loop_state_header), (gst_qtdemux_loop_state_movie),
(gst_qtdemux_loop_header), (next_entry_size), (gst_qtdemux_chain),
(qtdemux_sink_activate), (qtdemux_sink_activate_pull),
(qtdemux_sink_activate_push), (qtdemux_parse_trak):
* gst/qtdemux/qtdemux.h:
QtDemux can now work push-based.
It still needs some love for seeking.
Original commit message from CVS:
* gst/id3demux/id3v2frames.c: (parse_insert_string_field),
(parse_split_strings):
Add more validation to ensure that a char encoding conversion
produced a valid UTF-8 string.
Original commit message from CVS:
Reviewed by: Edward Hervey <edward@fluendo.com>
* gst/avi/gstavidemux.c: (gst_avi_demux_process_next_entry):
Properly handle end of segment. Closes#330885.
Original commit message from CVS:
* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_class_init),
(gst_rtp_mp4g_pay_init), (gst_rtp_mp4g_pay_parse_audio_config),
(gst_rtp_mp4g_pay_parse_video_config), (gst_rtp_mp4g_pay_new_caps),
(gst_rtp_mp4g_pay_setcaps), (gst_rtp_mp4g_pay_flush):
* gst/rtp/gstrtpmp4gpay.h:
Make more things work.
Handle ACC config strings.
Original commit message from CVS:
* gst/apetag/gsttagdemux.c: (gst_tag_demux_get_upstream_size),
(gst_tag_demux_do_typefind):
... and fix the very same leaks in GstTagDemux.
Original commit message from CVS:
* gst/id3demux/gstid3demux.c: (id3demux_get_upstream_size),
(gst_id3demux_do_typefind):
Fix a couple of mem leaks. (Patch by Jonathan Matthew
<jonathan at kaolin dot wh9 dot net>)
Original commit message from CVS:
* gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_setcaps):
First set options, then set caps or else the baseclass
will not know about the options, duh.
Original commit message from CVS:
* gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_class_init),
(gst_rtp_mp4v_pay_setcaps):
Don't waste time looking for a config string if we have codec_info
on the incomming caps.
Original commit message from CVS:
* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_chain):
Added more meaningfull warnings when something goes wrong.
Clear F bit on outgoing AMR packets.
* gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_class_init),
(gst_rtp_amr_pay_handle_buffer):
Added debugging category
Support payloading of multiple AMR frames.
* gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_depay_data):
Added some debugging.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header),
(qtdemux_parse_trak):
use the correct variable to check if we can calculate
the last chunk. Looks like an obvious bug, and makes
the dump of offsets comparable to other tools
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header),
(qtdemux_parse_trak):
clean up some debugging, using _OBJECT, moving recurring
messages to LOG level
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_handle_src_query),
(gst_qtdemux_handle_src_event), (gst_qtdemux_loop_header),
(qtdemux_inflate), (qtdemux_parse), (qtdemux_parse_trak),
(qtdemux_parse_udta), (qtdemux_tag_add_str), (qtdemux_tag_add_num),
(qtdemux_tag_add_gnre), (gst_qtdemux_handle_esds),
(qtdemux_video_caps), (qtdemux_audio_caps):
* gst/qtdemux/qtdemux.h:
Some QT demux loving.
Handle seeking in a less broken way.
Fix AMR caps to match the AMR decoder.
Set first timestamp on AMR samples to 0 for now.
Remove some \n in DEBUG strings.
Use _scale_int for maximum precision.
Original commit message from CVS:
* gst/matroska/ebml-write.c: (gst_ebml_write_reset),
(gst_ebml_write_flush_cache), (gst_ebml_write_element_push),
(gst_ebml_write_seek):
* gst/matroska/ebml-write.h:
Make sure we send a newsegment event in BYTES format
before sending buffers (#328531).
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
(gst_avi_demux_all_source_pads_unlinked),
(gst_avi_demux_process_next_entry):
* gst/avi/gstavidemux.h:
Third attempt, use gst_pad_is_linked() this time.
Original commit message from CVS:
* gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame),
(parse_split_strings):
Adjust for data length indicators when parsing (Fixes#329810)
Fix stupid bug parsing UTF-8 tag text.
Output tag strings with multiple fields as multiple tags, so the
app gets all the data.
Original commit message from CVS:
* gst/id3demux/id3v2frames.c: (parse_text_identification_frame),
(id3v2_tag_to_taglist), (id3v2_genre_string_to_taglist),
(id3v2_genre_fields_to_taglist):
Never output a tag with a null contents string.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_all_source_pads_unlinked):
Only pause if all pads are unlinked AND we've tried to send data
on all of them at least once.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_all_source_pads_unlinked),
(gst_avi_demux_process_next_entry), (gst_avi_demux_loop):
Make loop function/task pause itself when all source pads are
unlinked.
Original commit message from CVS:
* gst/auparse/gstauparse.c: (gst_au_parse_chain):
Don't push buffers into the adapter that we are going to
push downstream again without framing anyway. Also, the
adaptor takes ownership of buffers put into it (fixes
auparse pushing invalid buffers for .au files with
ADPCM contents). Finally, set caps on all outgoing buffers.
Original commit message from CVS:
* gst/id3demux/gstid3demux.c: (gst_id3demux_chain),
(gst_id3demux_read_id3v1), (gst_id3demux_sink_activate),
(gst_id3demux_send_tag_event):
* gst/id3demux/id3tags.c: (id3demux_read_id3v1_tag):
Someone should kick my butt. Remove ID3v1 tags from the end of the
file.
Improve error messages. Send the TAG message as soon as we complete
typefinding, instead of waiting until we send the first buffer.
Downstream tag event is still sent before the first buffer.
Original commit message from CVS:
* gst/id3demux/gstid3demux.c: (gst_id3demux_remove_srcpad):
Don't put function calls in g_return_if_fail() statements,
or they'll be replaced with NOOPs if someone compiles with
G_DISABLE_CHECKS defined.
Original commit message from CVS:
* gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame):
Never trust ANY information encoded in a media file, especially
when it's giving you sizes. (Fixes#328452)
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header),
(gst_qtdemux_add_stream), (qtdemux_parse_trak):
More coherent framerate setting on caps.
If sample_size is available, use that for the samples' duration in
the index. This enables single frame streams to work (and I imagine
fixes some other cases).
Tested on testsuite, no regression.
Original commit message from CVS:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_video_caps),
(gst_matroska_demux_audio_caps), (gst_matroska_demux_plugin_init):
* gst/matroska/matroska-ids.h:
Added recognition of Real Audio and Video streams in matroska demuxer.
Original commit message from CVS:
* gst/id3demux/id3v2frames.c: (id3v2_tag_to_taglist):
Remove errant break statement, and fix compilation with
older GCC.
Original commit message from CVS:
* gst/id3demux/id3tags.c: (id3demux_read_id3v2_tag):
* gst/id3demux/id3tags.h:
* gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame),
(parse_comment_frame), (parse_text_identification_frame),
(id3v2_tag_to_taglist), (id3v2_are_digits),
(id3v2_genre_string_to_taglist), (id3v2_genre_fields_to_taglist),
(parse_split_strings), (free_tag_strings):
Rewrite parsing of text tags to handle multiple NULL terminated
strings. Parse numeric genre strings and ID3v2 type
"(3)(6)Alternative" style genre strings.
Parse dates that are only YYYY or YYYY-mm format.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_parse_trak),
(qtdemux_audio_caps):
'twos' and 'sowt' fourcc can be 16bit or 8bit audio.
Fix 8bit case (#327133, based on patch by: Fabrizio
Gennari <fabrizio dot ge at tiscali dot it>).
Also, "G_LITTLE_ENDIAN" and "G_BIG_ENDIAN" are not
valid literals for endianness in caps strings,
only "LITTLE_ENDIAN" and "BIG_ENDIAN" are valid.
Original commit message from CVS:
* gst/videobox/gstvideobox.c: (gst_video_box_class_init):
Don't forget to initialize liboil, otherwise our oil functions
will crash (fixes#327871; patch by: Christoph Burghardt
<hawkes at web dot de>).
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_pad_convert):
Fix conversion from TIME to BYTES format (fixes#326864;
patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>)
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_init),
(gst_qtdemux_send_event), (gst_qtdemux_handle_src_event),
(gst_qtdemux_change_state), (gst_qtdemux_loop_header):
* gst/qtdemux/qtdemux.h:
Fix seeking for quicktime files. Could still use some more
love and sophistication.
Original commit message from CVS:
* gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame):
Fix compilation of id3demux when zlib is not present.
(Fixes#326602; patch by: Sergey Scobich)
Original commit message from CVS:
2006-01-13 Thomas Vander Stichele <thomas at apestaart dot org>
patch by: Mike Smith
* gst/level/gstlevel.c: (gst_level_message_new),
(gst_level_message_append_channel):
Fix memleak. Fixes#326612
Original commit message from CVS:
reviewed by: Edward Hervey <edward@fluendo.com>
* gst/qtdemux/qtdemux.c: (qtdemux_video_caps):
Add support for Indeo3 video in Quicktime files.
Closes#326524
Original commit message from CVS:
2005-01-07 Philippe Khalaf <philippe.kalaf@collabora.co.uk>
* gst-plugins-good/gst/udp/gstdynudpsink.c:
* gst-plugins-good/gst/udp/gstudpsrc.c:
Allow udpsrc and dynudpsink to take a sockfd as a parameter. For udpsrc,
overrides the port or multicast parameters. Fixes bugs #323021.
Original commit message from CVS:
* gst/id3demux/gstid3demux.c: (gst_id3demux_add_srcpad):
Add gst_element_no_more_pads() for proper decodebin behaviour.
* gst/id3demux/id3v2frames.c: (parse_comment_frame),
(parse_text_identification_frame), (parse_split_strings):
Failure to decode some tags is not a GST_ERROR() but a
GST_WARNING()
When iterating over a chunk of text, check that we haven't gone too
far.
Original commit message from CVS:
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_render),
(gst_multiudpsink_remove), (gst_multiudpsink_get_stats):
* gst/udp/gstmultiudpsink.h:
Track packets sent per client in addition to bytes sent; provide
this info through get-stats signal
Original commit message from CVS:
* gst/id3demux/id3tags.c: (id3demux_read_id3v2_tag):
If a broken tag has 0 bytes payload, at least still skip
the 10 byte header
Original commit message from CVS:
2005-12-22 Philippe Khalaf <burger@speedy.org>
* gst-plugins-good/gst/rtp/gstrtph263pdepay.h:
* gst-plugins-good/gst/rtp/gstrtph263pdepay.c:
* gst-plugins-good/gst/rtp/gstrtpmp4vdepay.h:
* gst-plugins-good/gst/rtp/gstrtpmp4vdepay.c:
Making these depayloaders (H263+ and mpeg4 video) inherit from
RtpBaseDepayloaderClass. Fixes bugs #323922 and #323908.
Original commit message from CVS:
2005-12-21 Jan Schmidt <thaytan@mad.scientist.com>
* 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:
* gst/id3demux/gstid3demux.c: (gst_id3demux_get_type),
(gst_id3demux_base_init), (gst_id3demux_class_init),
(gst_id3demux_chain):
* gst/id3demux/gstid3demux.h:
Add documentation for id3demux.
Don't fail if the first buffer is not at offset 0, just
attempt to typefind and do pass through
Rename the gst_type function from gst_gst_id3demux..
Original commit message from CVS:
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_render),
(gst_multiudpsink_add), (gst_multiudpsink_remove),
(gst_multiudpsink_get_stats):
* gst/udp/gstmultiudpsink.h:
Collect statistics; return them from get_stats.
Original commit message from CVS:
* ext/swfdec/gstswfdec.c: (gst_swfdec_class_init),
(gst_swfdec_chain), (gst_swfdec_render):
Add debugging category and return GstFlowReturn in the right places
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_link):
Get something from the peer pad once we've checked if there is a peer pad.
* gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state),
(qtdemux_tree_get_child_by_type), (qtdemux_parse_trak),
(qtdemux_video_caps):
Couple of fixes
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_reset),
(gst_avi_demux_handle_src_event), (gst_avi_demux_parse_stream),
(gst_avi_demux_parse_odml), (gst_avi_demux_peek_tag),
(gst_avi_demux_next_data_buffer), (gst_avi_demux_stream_scan),
(gst_avi_demux_stream_header), (gst_avi_demux_loop):
Construct index for indexless files.
Make sure pad/buffers are correctly reset to NULL once we don't need
them anymore, else we get lovely segfaults/assertions.
* gst/wavparse/gstwavparse.c:
Yes, you can have 96KHz audio and wma in wav :(
Original commit message from CVS:
* ext/esd/esdmon.c: (gst_esdmon_open_audio):
* ext/esd/esdsink.c: (gst_esdsink_prepare):
* gst/multipart/multipartdemux.c:
change some char* into char[]
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_reset),
(gst_wavparse_other), (gst_wavparse_perform_seek),
(gst_wavparse_stream_headers), (gst_wavparse_stream_data),
(gst_wavparse_loop), (gst_wavparse_pad_convert),
(gst_wavparse_srcpad_event), (gst_wavparse_sink_activate_pull):
* gst/wavparse/gstwavparse.h:
Use GstSegment to implement more seeking features.
Original commit message from CVS:
* gst/rtsp/rtspconnection.c:
Add <netinet/in.h> include and move <arpa/inet.h> include
to make things work on OpenBSD as well (fixes#323717;
patch by: Benjamin Pineau)
Original commit message from CVS:
2005-12-14 Philippe Khalaf <burger@speedy.org>
* gst-plugins-good/gst/rtp/gstasteriskh263.c:
* gst-plugins-good/gst/rtp/gstrtpamrdepay.c:
* gst-plugins-good/gst/rtp/gstrtpamrpay.c:
* gst-plugins-good/gst/rtp/gstrtpg711depay.c:
* gst-plugins-good/gst/rtp/gstrtpg711depay.c:
* gst-plugins-good/gst/rtp/gstrtpgsmdepay.c:
* gst-plugins-good/gst/rtp/gstrtph263pay.c:
* gst-plugins-good/gst/rtp/gstrtph263pdepay.c:
* gst-plugins-good/gst/rtp/gstrtph263ppay.c:
* gst-plugins-good/gst/rtp/gstrtpmp4vdepay.c:
* gst-plugins-good/gst/rtp/gstrtpmp4vpay.c:
* gst-plugins-good/gst/rtp/gstrtpmpadepay.c:
* gst-plugins-good/gst/rtp/gstrtpmpapay.c:
* gst-plugins-good/gst/rtp/README:
Fixed payload range in payloder caps. Removed payload range completly from
depayloaders as they don't require payload type in their caps. In effect,
there isn't any specific payload type for any given codec, only suggestions.
Fixes bug #324011.
Original commit message from CVS:
2005-12-13 Julien MOUTTE <julien@moutte.net>
* gst/videomixer/videomixer.c: (gst_videomixer_init),
(gst_videomixer_fill_queues), (gst_videomixer_blend_buffers),
(gst_videomixer_collected): Code cleanup and re-enabling
queued time validity check for correct EOS handling.
Original commit message from CVS:
* gst/flx/gstflxdec.c: (gst_flxdec_chain):
If the speed of the file is null in the header, set the frame_time to the default
setting of GST_SECOND / 70. Which is the default frame_delay for .fli files as
stated in this document : http://www.compuphase.com/flic.htm
Would be nice to have the time conversion done properly too
(duration = flxh->frames * flxdec->frame_time)
Original commit message from CVS:
2005-12-12 Julien MOUTTE <julien@moutte.net>
* 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/videomixer/videomixer.c:
(gst_videomixer_pad_sink_setcaps),
(gst_videomixer_getcaps), (gst_videomixer_fill_queues),
(gst_videomixer_update_queues), (gst_videomixer_collected):
Adding
documentation for videomixer on my way with a funny sample
pipeline.
Original commit message from CVS:
* gst/auparse/gstauparse.c: (gst_au_parse_base_init),
(gst_au_parse_class_init), (gst_au_parse_init),
(gst_au_parse_dispose), (gst_au_parse_chain),
(gst_au_parse_change_state), (plugin_init):
* gst/auparse/gstauparse.h:
Use gst_object_unref() for GstObjects instead of
g_object_unref() and fix a mem leak in a debug
statement; while we're at it, also borgify, use
boilerplate macros and clean up a little bit.
Original commit message from CVS:
* gst/goom/gstgoom.c:
* gst/level/level-example.c: (main):
* gst/smoothwave/demo-osssrc.c: (main):
Use audiotestsrc instead of sinesrc (#323798).
Original commit message from CVS:
* gst/debug/efence.c: (gst_efence_init), (gst_efence_chain),
(gst_fenced_buffer_copy):
Make sure GST_BUFFER_DATA is set on fenced copied buffers; fix
GST_DEBUG crasher where GST_TIME_FORMAT was not used in
conjunction with GST_TIME_ARGS. Also, don't leak pad templates
and use GST_DEBUG_FUNCPTR for pad functions.
Original commit message from CVS:
* ext/faad/gstfaad.c: (gst_faad_chanpos_to_gst),
(gst_faad_update_caps):
Assume that an unknown channel mapping with 2 channels
is stereo and play it that way instead of erroring.
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header),
(gst_qtdemux_add_stream), (qtdemux_parse_trak):
Handle e.g. jpeg streams with 0 duration frames as having 0 framerate.
Debug fixes. Some 64 bit variable fixes
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header),
(gst_qtdemux_add_stream):
Memleak fixes.
Send out EOS for valid reasons (couldn't pull_range() from upstream
for example).
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_process_next_entry):
Useless check now we're setting the current entry correctly.
Original commit message from CVS:
* ext/faad/gstfaad.c: (gst_faad_srcgetcaps):
Handle gracefully the consequence of "Maximum number of scalefactor
bands exceeded", which results in 0 channels with samplerates of 0.
* gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state):
Do upward transitions, then call parent state_change, then do
downward transitions.
Original commit message from CVS:
* gst/matroska/matroska-mux.c:
(gst_matroska_mux_video_pad_setcaps):
Look for pixel-aspect-ratio in caps, not pixel_width and
pixel_height (Fixes: #322645)
Original commit message from CVS:
* gst/matroska/matroska-mux.c:
(gst_matroska_mux_video_pad_setcaps):
From Michal Benes:
frame duration should be GST_SECOND / framerate, not
GST_SECOND * framerate. (Fixes: #322643)
Original commit message from CVS:
* configure.ac:
fix up GST_PLUGIN_LDFLAGS
* gst/rtsp/rtspconnection.c:
fix includes (see #317043)
* gst/videofilter/Makefile.am:
stop installing this library
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:
* gst/autodetect/gstautoaudiosink.c:
(gst_auto_audio_sink_base_init):
* gst/autodetect/gstautovideosink.c:
(gst_auto_video_sink_base_init),
(gst_auto_video_sink_factory_filter):
documenting auto*sink
using strstr for the video sink lookup, class field is not ordered
update other plugins
Original commit message from CVS:
2005-11-24 Julien MOUTTE <julien@moutte.net>
* gst/effectv/gstquark.c: (gst_quarktv_set_caps),
(gst_quarktv_get_unit_size), (gst_quarktv_transform),
(gst_quarktv_planetable_clear), (gst_quarktv_change_state),
(gst_quarktv_base_init), (gst_quarktv_class_init),
(gst_quarktv_init): Flush the planes list on reverse caps
negotiation. This was crashing because of differently sized
buffers.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header),
(gst_qtdemux_add_stream), (qtdemux_dump_mvhd),
(qtdemux_parse_trak):
Convert to fractional framerates
Original commit message from CVS:
Fix for #321430: unresolved symbols due to incorrect linkage on inline functions
in goom.
Does not, however, fix the general crackheadedness of goom (global variables,
oh my!); this should be moved to -bad.
Original commit message from CVS:
* gst/autodetect/gstautoaudiosink.c:
(gst_auto_audio_sink_find_best), (gst_auto_audio_sink_detect):
* gst/autodetect/gstautovideosink.c:
(gst_auto_video_sink_find_best), (gst_auto_video_sink_detect):
Use gst_plugin_feature_list_free() to free feature list and
in the case of autovideosink free the list at all. Also
miscellaneous cosmetic fixes.
Original commit message from CVS:
* gst/cutter/gstcutter.c: (gst_cutter_chain),
(gst_cutter_set_property), (gst_cutter_get_caps):
copy calculation code from level; remove use of some audio
functions
Original commit message from CVS:
* ext/raw1394/gstdv1394src.c: (gst_dv1394src_class_init),
(gst_dv1394src_init), (gst_dv1394src_dispose),
(gst_dv1394src_set_property), (gst_dv1394src_discover_avc_node),
(gst_dv1394src_uri_set_uri):
* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init),
(gst_udpsrc_update_uri), (gst_udpsrc_set_uri),
(gst_udpsrc_set_property), (gst_udpsrc_uri_get_uri):
URIHandler interface and element properties are now properly
synchronized for DV1394src and UDPSrc
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_init),
(gst_wavparse_create_sourcepad), (gst_wavparse_sink_activate):
Use GST_DEBUG_FUNCPTR; add debug message in pad activate function.
Original commit message from CVS:
* gst/auparse/Makefile.am:
* gst/auparse/gstauparse.c: (gst_auparse_class_init),
(gst_auparse_init), (gst_auparse_dispose), (gst_auparse_chain),
(gst_auparse_change_state):
* gst/auparse/gstauparse.h:
Partially fix#161712. playbin still doesn't work on these files,
(on the bug report, Andy says we aren't typefinding it for some
reason?) but at least auparse isn't totally busted like it was before.
Original commit message from CVS:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_sync_streams):
Filler events are gone for now, comment out section generating
them.
Original commit message from CVS:
2005-11-20 Julien MOUTTE <julien@moutte.net>
* gst/matroska/matroska-mux.c: (gst_matroska_mux_start): Replace
GST_PAD_IS_USABLE by something approaching it.
Original commit message from CVS:
2005-11-20 Julien MOUTTE <julien@moutte.net>
* gst/matroska/matroska-mux.c: (gst_matroska_mux_start): Fix for
API changes.
* gst/wavenc/gstwavenc.c: (gst_wavenc_chain): Fix for API
changes,
but also fix the code that was not checking return values from
pad_push neither using pad_alloc_buffer.
Original commit message from CVS:
Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
* gst/matroska/matroska-demux.c: (gst_matroskademux_do_index_seek):
When seeking, seek to closest index entry at or before the requested
seek position, not just the closest one (#321001).
Original commit message from CVS:
* gst/avi/gstavidemux.c: (swap_line), (gst_avi_demux_invert),
(gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data):
Invert DIB images again (see #132341).
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_reset),
(gst_avi_demux_src_convert), (gst_avi_demux_handle_src_event),
(gst_avi_demux_stream_header), (gst_avi_demux_handle_seek),
(gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data),
(gst_avi_demux_loop):
* gst/avi/gstavidemux.h:
Yeah, implement proper seeking. Exact seeking and segment seeking.
Still need to do some checks for segment_stop.
Original commit message from CVS:
Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
* gst/qtdemux/qtdemux.c: (qtdemux_parse_udta):
Add support for custom genre tags.
Original commit message from CVS:
Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
* gst/matroska/matroska-mux.c: (gst_matroska_mux_write_data):
Don't try to ready buffer duration from buffer that we don't
own any longer and that might already have been unreffed.
(#321136)
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
(gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data),
(gst_avi_demux_loop):
No need to take the STREAM_LOCK in the loop function. Improve
some debug messages. Don't leak pad names in debug messages.
Original commit message from CVS:
* gst/matroska/matroska-demux.c:
(gst_matroska_demux_push_vorbis_codec_priv_data),
(gst_matroska_demux_add_wvpk_header):
Don't error out when the source pad isn't linked.
Original commit message from CVS:
* gst/wavenc/gstwavenc.c: (gst_wavenc_stop_file),
(gst_wavenc_init), (gst_wavenc_event), (gst_wavenc_chain):
Added proper event handlind,
made downstream newsegment event use GST_FORMAT_BYTES (otherwise it's
ignored),
and don't set a duration of 0 for buffers otherwise they are discarded
by GstBaseSink.
GstWavEnc needs some serious loving, after going through the code I'm
really wondering how this can stay in -good ...
Original commit message from CVS:
Patch by: Michal Benes <michal.benes@xeris.cz>
* check/Makefile.am:
* gst/matroska/ebml-write.c: (gst_ebml_write_seek):
* gst/matroska/matroska-mux.c: (gst_matroska_mux_handle_src_event),
(gst_matroska_mux_audio_pad_setcaps), (gst_matroska_mux_start):
add a unit test for matroskamux
fix the bugs that the unit test exposed
Original commit message from CVS:
* gst/autodetect/gstautoaudiosink.c:
(gst_auto_audio_sink_class_init),
(gst_auto_audio_sink_change_state):
* gst/autodetect/gstautovideosink.c:
(gst_auto_video_sink_class_init),
(gst_auto_video_sink_change_state):
Fix state change function and use GST_DEBUG_FUNCPTR in
class_init.
Original commit message from CVS:
Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
* gst/matroska/ebml-write.c: (gst_ebml_write_new),
(gst_ebml_write_reset), (gst_ebml_write_element_new):
* gst/matroska/ebml-write.h:
* gst/matroska/matroska-mux.c: (gst_matroska_mux_write_data):
Set timestamps on outgoing ebml headers as well, so that the
element after matroskamux can get the timestamp already when
reading the first ebml element and doesn't have to wait for
the actual data buffer for that (#320308).
Original commit message from CVS:
Payloader now sets some default caps on the srcpad if caps on the sinkpad are never set. This is important for the g711 to work with burger's rtpbin element.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_class_init),
(qtdemux_parse_tree):
Remove 'got-redirect' signal and post element message
on the bus instead.
Original commit message from CVS:
2005-10-27 Julien MOUTTE <julien@moutte.net>
* gst/videobox/gstvideobox.c: (gst_video_box_class_init),
(gst_video_box_transform_caps), (gst_video_box_set_caps),
(gst_video_box_get_unit_size), (gst_video_box_copy_plane_i420),
(gst_video_box_i420), (gst_video_box_ayuv): Use liboil for
I420 rendering as well, doesn't bring much for my platform.
Might help on some other platforms.
Original commit message from CVS:
2005-10-26 Julien MOUTTE <julien@moutte.net>
* gst/videobox/Makefile.am: Use liboil.
* gst/videobox/gstvideobox.c: (gst_video_box_class_init),
(gst_video_box_set_property), (gst_video_box_transform_caps),
(gst_video_box_set_caps), (gst_video_box_get_unit_size),
(gst_video_box_ayuv): Lot of optimization in AYUV rendering
using liboil. Will dot the same to I420 border generation
tomorrow.
Original commit message from CVS:
2005-10-26 Julien MOUTTE <julien@moutte.net>
* gst/videobox/gstvideobox.c: (gst_video_box_class_init),
(gst_video_box_transform_caps), (gst_video_box_get_unit_size),
(gst_video_box_ayuv): Fix the stride issue when boxing to AYUV.
Original commit message from CVS:
2005-10-26 Julien MOUTTE <julien@moutte.net>
* gst/videomixer/videomixer.c:
(gst_videomixer_pad_set_property),
(gst_videomixer_pad_sink_setcaps), (gst_videomixer_getcaps):
Use gst_pad_get_parent and drop the ref that was added through
that call.
Original commit message from CVS:
2005-10-24 Julien MOUTTE <julien@moutte.net>
* gst/videomixer/videomixer.c: Don't restrict video geometry
from 16 to 4096.
Original commit message from CVS:
* ext/speex/gstspeexenc.c: (gst_speexenc_get_query_types),
(gst_speexenc_src_query):
Add position and duration query, fix query type function.
* gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream),
(gst_matroska_demux_video_caps), (gst_matroska_demux_audio_caps):
Let's not set non-fixed caps on source pads.
Original commit message from CVS:
* ext/dv/gstdvdemux.c: (gst_dvdemux_demux_frame):
* gst/avi/gstavidemux.c: (gst_avi_demux_get_src_query_types),
(gst_avi_demux_handle_seek):
Set correct stream_time in newsegment event.
avi can also handle a duration query now.
Original commit message from CVS:
* gst/matroska/matroska-demux.c:
(gst_matroska_demux_handle_src_query),
(gst_matroska_demux_handle_seek_event),
(gst_matroska_demux_loop_stream_parse_id):
Fix duration query; fix basetime in newsegment event after
seek; fix duration in initial newsegment event.
* gst/matroska/matroska-mux.c:
(gst_matroska_mux_audio_pad_setcaps), (gst_matroska_mux_start):
Extract number of channels and samplerate from vorbis headers;
add some debug messages when querying the durations of the
input streams.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_handle_seek),
(gst_wavparse_stream_headers), (gst_wavparse_stream_data),
(gst_wavparse_pad_convert), (gst_wavparse_srcpad_event):
Set stream time correctly in newsegment.
Original commit message from CVS:
* gst/goom/filters.c:
* gst/goom/graphic.h:
* gst/goom/lines.c:
Make inline functions either 'static inline' or 'extern inline',
otherwise the Forte compiler apparently won't inline them (#317300).
Original commit message from CVS:
2005-10-19 Julien MOUTTE <julien@moutte.net>
* ext/libpng/gstpngdec.c: (gst_pngdec_class_init),
(gst_pngdec_init), (user_error_fn), (user_warning_fn),
(user_info_callback), (user_endrow_callback),
(user_end_callback),
(user_read_data), (gst_pngdec_caps_create_and_set),
(gst_pngdec_task), (gst_pngdec_chain), (gst_pngdec_sink_event),
(gst_pngdec_libpng_clear), (gst_pngdec_libpng_init),
(gst_pngdec_change_state), (gst_pngdec_sink_activate_push),
(gst_pngdec_sink_activate_pull), (gst_pngdec_sink_activate):
* ext/libpng/gstpngdec.h: Complete rewrite of pngdec. It's now
very nice and handle push/pull based model. if you have filesrc
connected to it, it will do random access to load the png file.
If you have a network source that can't do _getrange, it does
progressive loading through the chain function.
* gst/alpha/gstalphacolor.c: (gst_alpha_color_transform_caps),
(transform_rgb), (transform_bgr): Fix caps negotiation correctly
thanks to Master Wim Taymans ;-)
Original commit message from CVS:
Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
* gst/matroska/matroska-mux.c:
(gst_matroska_mux_video_pad_setcaps),
(gst_matroska_mux_audio_pad_setcaps):
Fix mpeg4 input handling (#318847); also, while we're at it,
fix media type for Motion-JPEG: should be image/jpeg.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_handle_seek),
(gst_wavparse_stream_headers), (gst_wavparse_stream_data),
(gst_wavparse_pad_convert), (gst_wavparse_srcpad_event):
Fix for segment-start/stop API change.
Original commit message from CVS:
2005-10-17 Julien MOUTTE <julien@moutte.net>
* gst/videobox/gstvideobox.c: (gst_video_box_transform_caps),
(gst_video_box_get_unit_size): Fix caps nego some more to get
AYUV
output declared in transform_caps.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header):
* gst/tta/gstttaparse.c: (gst_tta_parse_src_event),
(gst_tta_parse_parse_header):
newsegment API update.
Original commit message from CVS:
* gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_reset),
(gst_auto_audio_sink_find_best), (gst_auto_audio_sink_detect):
* gst/autodetect/gstautovideosink.c: (gst_auto_video_sink_reset),
(gst_auto_video_sink_find_best), (gst_auto_video_sink_detect):
Set state of elements to NULL before removing from bins.
Set state of test element to NULL if we failed to move it to READY
Original commit message from CVS:
* gst/rtp/gstrtpamrenc.c: (gst_rtpamrenc_setcaps):
* gst/rtp/gstrtpgsmparse.c:
* gst/rtp/gstrtph263penc.c:
* gst/rtp/gstrtpmp4venc.c: (gst_rtpmp4venc_class_init),
(gst_rtpmp4venc_parse_data), (gst_rtpmp4venc_handle_buffer),
(gst_rtpmp4venc_set_property):
* gst/rtp/gstrtpmpaenc.c: (gst_rtpmpaenc_handle_buffer):
Various class and caps fixes from Andre Magalhaes (andrunko)
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_base_init),
(gst_avi_demux_class_init), (gst_avi_demux_parse_stream),
(gst_avi_demux_change_state):
Don't crash when encountering a stream with an unknown fourcc or
codec id. Instead, create a pad of type video/x-avi-unknown or
audio/x-avi-unknown, which as a side-effect also results in less
confusing error messages in players ('no decoder' vs. 'no streams');
minor fixes to state change function and class_init function.
Original commit message from CVS:
* gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_init):
* gst/autodetect/gstautovideosink.c: (gst_auto_video_sink_init):
These are sinks.
Original commit message from CVS:
* check/elements/level.c: (GST_START_TEST):
fix test for new GstClockTime use
* gst/level/gstlevel.c: (gst_level_init), (gst_level_set_caps),
(gst_level_transform_ip):
* gst/level/gstlevel.h:
fix up the decay peak, ensuring the decay peak is never lower
than the peak for that interval
Original commit message from CVS:
* gst/rtp/TODO:
* gst/rtp/gstrtpdec.c: (gst_rtpdec_getcaps):
* gst/rtp/gstrtpmp4venc.c: (gst_rtpmp4venc_class_init),
(gst_rtpmp4venc_parse_data), (gst_rtpmp4venc_handle_buffer),
(gst_rtpmp4venc_set_property):
* gst/rtp/gstrtpmp4venc.h:
* gst/rtp/gstrtpmpaenc.c: (gst_rtpmpaenc_handle_buffer):
* gst/rtp/gstrtpmpaenc.h:
Use is_filled to both check MTU and max-ptime of base class.
Original commit message from CVS:
* gst/rtp/gstrtpmp4venc.c: (gst_rtpmp4venc_class_init),
(gst_rtpmp4venc_parse_data), (gst_rtpmp4venc_handle_buffer),
(gst_rtpmp4venc_set_property):
Don't fragment packets with multiple frames.
Original commit message from CVS:
* gst/rtp/TODO:
* gst/rtp/gstrtpmp4vdec.c: (gst_rtpmp4vdec_setcaps):
* gst/rtp/gstrtpmp4venc.c: (gst_rtpmp4venc_class_init),
(gst_rtpmp4venc_init), (gst_rtpmp4venc_parse_data),
(gst_rtpmp4venc_handle_buffer), (gst_rtpmp4venc_set_property),
(gst_rtpmp4venc_get_property):
* gst/rtp/gstrtpmp4venc.h:
Remove g_print.
Update TODO
Make payload encoder a bit smarter and more correct with
timestamps.
Added option in payloader to include config string in-band.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_rtpmap),
(gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_send),
(gst_rtspsrc_change_state):
More SDP parsing and caps setting.
Do NO_PREROLL differently.
add pads only after negotiated.
* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init),
(gst_udpsrc_getcaps):
Implement the getcaps function.
Original commit message from CVS:
* gst/rtp/README:
Update README
* gst/rtp/gstrtpamrdec.c: (gst_rtpamrdec_sink_setcaps):
Make extra params as strings.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_set_state),
(gst_rtspsrc_parse_rtpmap), (gst_rtspsrc_media_to_caps),
(gst_rtspsrc_stream_setup_rtp), (gst_rtspsrc_send):
Make state change return NO_PREROLL as this is a live
source.
* gst/udp/gstudpsrc.c: (gst_udpsrc_set_property):
Don't unref old caps when NULL.
Original commit message from CVS:
* gst/level/level-example.c: (main):
Fix for new bus API.
* gst/udp/gstudpsrc.c: (gst_udpsrc_set_property):
Set caps on pads.
Original commit message from CVS:
Updates to payloader/depayloaders, make payloaders use
the base classes.
Updated README with suggested RTP caps and how to convert
to/from SDP.
Added config descriptor in mp4v payloader.
Original commit message from CVS:
2005-09-15 Andy Wingo <wingo@pobox.com>
* gst/autodetect/gstautoaudiosink.c (gst_auto_audio_sink_find_best):
* gst/autodetect/gstautovideosink.c
(gst_auto_video_sink_find_best): Update for new registry API.
Original commit message from CVS:
* common/c-to-xml.py:
* common/gtk-doc-plugins.mak:
a simple py script to generate valid xml from a C example
probably also need to strip an MIT license when we decide
* docs/plugins/Makefile.am:
* gst/level/Makefile.am:
* gst/level/gstlevel.c: (gst_level_init):
* gst/level/level-example.c: (message_handler), (main):
add an example to level that will show up in the docs
* gst/rtp/TODO:
add a note for the future
Original commit message from CVS:
* gst/wavenc/gstwavenc.c: (gst_wavenc_class_init):
Actually define the debug object being used in wavenc. Fixes#316205
Original commit message from CVS:
Link smpte plugin against GST_BASE_LIBS, to get libgstbase; needed to
build on win32 as this plugin uses collectpads (bug 316204)
Original commit message from CVS:
* gst-plugins-good.spec.in:
spec file fixes
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init),
(gst_multiudpsink_render), (gst_multiudpsink_add),
(gst_multiudpsink_clear):
it actually helps to actually stream if we hook up the
add signal to an actual implementation
* gst/udp/gstudpsrc.c: (gst_udpsrc_start):
some debugging
Original commit message from CVS:
Andrewio Patrickoforus Wingonymus - 5 additional tests for your sins
Add a regression test for level and fix a casting bug that made the additional
channels turn out wrong
Original commit message from CVS:
2005-08-28 Andy Wingo <wingo@pobox.com>
* Updates for two-arg init from GST_BOILERPLATE.
* ext/ladspa/gstsignalprocessor.c (gst_signal_processor_init): Use
the second arg for the class, because G_OBJECT_GET_CLASS (self)
returns the wrong thing.
(gst_signal_processor_add_pad_from_template): Make pads of the
right type.
* ext/ladspa/gstladspa.c (gst_ladspa_class_get_param_spec): Make
writable param specs G_PARAM_CONSTRUCT so default values work.
(gst_ladspa_init): Use the second arg for the class.
Original commit message from CVS:
* gst/rtp/gstrtph263penc.c: (gst_rtph263penc_class_init),
(gst_rtph263penc_flush), (gst_rtph263penc_chain),
(gst_rtph263penc_set_property), (gst_rtph263penc_get_property):
* gst/rtp/gstrtph263penc.h:
Added configurable pt and ssrc, to be merged in the caps or
a base class...
Original commit message from CVS:
* gst/rtp/gstrtph263pdec.c: (gst_rtph263pdec_init),
(gst_rtph263pdec_chain):
* gst/rtp/gstrtph263penc.c: (gst_rtph263penc_class_init),
(gst_rtph263penc_flush), (gst_rtph263penc_chain):
Some cleanups in the h263p (de)payloaders.
Original commit message from CVS:
* ext/amrnb/amrnbdec.c:
* ext/amrnb/amrnbenc.c: (gst_amrnbenc_setcaps):
* ext/amrnb/amrnbparse.c:
Update caps with audio/AMR.
* gst/rtp/gstrtpamrdec.c: (gst_rtpamrdec_init),
(gst_rtpamrdec_sink_setcaps), (gst_rtpamrdec_chain),
(gst_rtpamrdec_change_state):
* gst/rtp/gstrtpamrdec.h:
* gst/rtp/gstrtpamrenc.c: (gst_rtpamrenc_class_init),
(gst_rtpamrenc_init), (gst_rtpamrenc_chain):
Dont set FT headers twice, it was already in the encoded
bitstream.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send), (gst_rtspsrc_open),
(gst_rtspsrc_close), (gst_rtspsrc_play):
* gst/rtsp/rtspconnection.c: (parse_line):
Cleanups
* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init),
(gst_udpsrc_create), (gst_udpsrc_set_property),
(gst_udpsrc_get_property):
* gst/udp/gstudpsrc.h:
Added caps property, we need this soon to type the buffers.
Original commit message from CVS:
* gst/rtp/gstrtpamrdec.c: (gst_rtpamrdec_init),
(gst_rtpamrdec_chain):
Fix up amr depayloader a bit.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send), (gst_rtspsrc_open),
(gst_rtspsrc_close), (gst_rtspsrc_play):
Look for options result in Public and Allow header fields..
spec says Allow but some servers return Public...
Original commit message from CVS:
* gst/rtp/gstrtpamrenc.c: (gst_rtpamrenc_class_init),
(gst_rtpamrenc_init), (gst_rtpamrenc_chain),
(gst_rtpamrenc_set_property), (gst_rtpamrenc_get_property):
* gst/rtp/gstrtpamrenc.h:
Added payload_type and ssrc properties to the payloader.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send), (gst_rtspsrc_open),
(gst_rtspsrc_close), (gst_rtspsrc_play):
Options need to be stripped and are in the Public header field.
* gst/rtsp/rtspurl.c: (rtsp_url_parse):
Fix url / parsing...
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send), (gst_rtspsrc_open),
(gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause),
(gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
* gst/rtsp/rtspdefs.c: (rtsp_method_as_text), (rtsp_find_method):
* gst/rtsp/rtspdefs.h:
* gst/rtsp/rtsptransport.c: (rtsp_transport_parse):
Handle RTSP defaults better.
Issue OPTIONS request to figure out what we are allowed to do.
Make the methods a bitfield so we can easily collect supported
options.
Fix rtsp_find_method.
Do proper RTSP connection shutdown.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
(gst_avi_demux_stream_header):
* gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state),
(gst_qtdemux_add_stream), (qtdemux_parse_tree):
Uncomment metadata and codec-name handling.
Original commit message from CVS:
* ext/mad/Makefile.am:
* gst/avi/Makefile.am:
* gst/effectv/Makefile.am:
* gst/udp/Makefile.am:
* gst/wavparse/Makefile.am:
Use -lgstfoo-@GST_MAJORMINOR@ instead of -lgstfoo-0.9
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers),
(gst_wavparse_stream_data):
Add some fixes from 0.8 branch: allow 24/32bps songs and
blockalign samples to the header-specified size, if any
(#311070); error out on channels==0 or bitrate==0
(#309043, #304588).
Original commit message from CVS:
* ext/faad/gstfaad.c: (gst_faad_class_init), (gst_faad_setcaps):
Add debug category, remove Close() call that made it crash
whenever reusing, renegotiating or anything; Close() actually
free()s the handle and should only be called on READY->NULL.
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header):
Actually set caps on buffer (in addition to pad), also.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_stream_header):
Fix AVI header parsing: add missing break statement after
GST_RIFF_INFO_LIST parsing code; gst_riff_read_chunk() has
already advanced the avi->offset, no need to do it twice
(fixes MovieOfMovies.avi).
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_event),
(gst_avi_demux_handle_seek):
Fix seeking (or, well, fix threading issue where a variable was
set before a lock was taken and was already unset before that
same lock was taken and was thus no longer in existance when it
actually had to be used).
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_process_next_entry):
Mixing binary and logical operators is not going to work; fix
position-querying in Totem.
Original commit message from CVS:
2005-08-04 Andy Wingo <wingo@pobox.com>
* sys/oss/gstossaudio.c (plugin_init): Second-class citizen.
* gst/videobox/gstvideobox.c (gst_video_box_get_size): Update for
API changes.
* configure.ac (DEFAULT_AUDIOSINK, DEFAULT_VIDEOSINK): Set to
autoaudiosink and autovideosink.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_reset),
(gst_avi_demux_parse_stream), (gst_avi_demux_process_next_entry):
You need to allocatate (len+1) characters to store a len size string.
Also don't stop the processing task if the output pad is not linked.
Original commit message from CVS:
* configure.ac:
* gst/wavparse/gstwavparse.c:
* gst/wavparse/gstwavparse.h:
* gst/wavparse/Makefile.am:
Ported wavparse to 0.9 . Playing, seeking and state changes work.
Could need more loving on the headers though.
Original commit message from CVS:
2005-07-16 Philippe Khalaf <burger@speedy.org>
* gst/fdsrc/gstfdsrc.c:
* gst/fdsrc/gstfdsrc.h:
* gst/fdsrc/Makefile.am:
Moved fdsrc 0.9 port from gstreamer/gst/elements to here.
Original commit message from CVS:
* gst/level/gstlevel.c: (gst_level_transform):
* gst/videobox/gstvideobox.c: (gst_video_box_class_init),
(gst_video_box_get_size), (gst_video_box_transform):
Port to new base class.
Original commit message from CVS:
2005-07-08 Andy Wingo <wingo@pobox.com>
* gst/avi/Makefile.am (libgstavi_la_CFLAGS): No gettext hacks, the
defines come from config.h.
* autogen.sh: Run autopoint, etc.
* Makefile.am (DIST_SUBDIRS, SUBDIRS): Go into po/.
* configure.ac: Add gettext stuff.
Original commit message from CVS:
* gst/videobox/gstvideobox.c: (gst_video_box_init),
(gst_video_box_transform_caps), (gst_video_box_set_caps):
Logic was reversed. Needs some more fixes in the transform
function to include AYUV output.
Moved AYUV as prefered format.
Original commit message from CVS:
* gst/base/gstbasesrc.c: (gst_base_src_get_range),
(gst_base_src_default_negotiate), (gst_base_src_negotiate):
Allow subclasses to implement their own negotiation.
Original commit message from CVS:
2005-07-05 Andy Wingo <wingo@pobox.com>
* gst/videobox/gstvideobox.c: Clean up, port to 0.9, use
BaseTransform.
* gst/videobox/Makefile.am: Link to base libs, include
plugins-base cflags, dist the README.
* configure.ac (GST_PLUGIN_ALL, AC_CONFIG_FILES): Add videobox to
the build.
Original commit message from CVS:
2005-07-04 Andy Wingo <wingo@pobox.com>
* examples/level/:
* examples/level/Makefile.am:
* examples/level/README:
* examples/level/demo.c:
* examples/level/plot.c: Examples moved out of the source dir. Not
updated tho.
* configure.ac: Add level to the build.
* gst/level/Makefile.am:
* gst/level/gstlevel.h:
* gst/level/gstlevel.c: Cleaned up, ported to 0.9.
Original commit message from CVS:
* gst/udp/Makefile.am:
* gst/udp/gstudp.c:
* gst/udp/gstdynudpsink.c: (new)
* gst/udp/gstdynudpsink.h: (new)
Added new element (udpdynsink) that receives GstNetBuffers and sends the
udp packets to the source given in the buffer. It's used by rtpsession
element for now.
* gst/udp/gstudpsrc.c:
Fixed memory leak.
Original commit message from CVS:
2005-07-01 Jan Schmidt <thaytan@mad.scientist.com>
* ext/libcaca/Makefile.am:
* ext/mad/Makefile.am:
* gst/effectv/Makefile.am:
* gst/udp/Makefile.am:
Replace GST_PLUGINS_LIBS_* with GST_PLUGINS_BASE_*
* ext/mad/gstid3tag.c: (gst_id3_tag_src_query),
(gst_id3_tag_src_event), (gst_id3_tag_sink_event),
(gst_id3_tag_chain), (plugin_init):
* ext/mad/gstmad.c: (gst_mad_src_query), (gst_mad_chain):
Signedness warning fix, use gst_pad_get_peer instead of GST_PAD_PEER
in querying and event handling, because we're not holding the pad
lock and the peer may disappear.
* gst/avi/gstavidemux.c: (gst_avi_demux_parse_subindex),
(gst_avi_demux_parse_index), (gst_avi_demux_massage_index):
Signedness warning fixes.
* gst/videofilter/gstvideotemplate.c: (plugin_init):
Remove gst_library_load
Original commit message from CVS:
* gst/avi/Makefile.am: (libgstavi_la_LIBADD):
Added linking to libgstriff-0.9
* ext/mad/gstmad.c: (gst_mad_src_query):
check the format of the upstream query and return query if it's the
same format as the requested one.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtsp_proto_get_type),
(gst_rtspsrc_class_init), (gst_rtspsrc_create_stream),
(gst_rtspsrc_add_element), (gst_rtspsrc_set_state),
(gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_stream_configure_transport), (find_stream),
(gst_rtspsrc_loop), (gst_rtspsrc_open), (gst_rtspsrc_play),
(gst_rtspsrc_change_state):
Fix case where outpad could not be decided.
Original commit message from CVS:
* ext/mad/gstmad.c: (gst_mad_chain), (gst_mad_change_state):
* ext/sidplay/gstsiddec.cc:
* gst/alpha/gstalpha.c: (gst_alpha_chain):
* gst/goom/gstgoom.c: (gst_goom_chain):
No need to take the lock anymore, core already did
that before calling us.
Original commit message from CVS:
* Makefile.am:
* ext/Makefile.am:
* sys/Makefile.am:
Make my automake version shut up about undefined variables
* gst/goom/gstgoom.c:
GstAdapter moved to base objects.
Original commit message from CVS:
* configure.ac:
* gst/alpha/Makefile.am:
* gst/alpha/gstalpha.c: (gst_alpha_class_init), (gst_alpha_init),
(gst_alpha_sink_setcaps), (gst_alpha_chain):
Ported alpha, remove alphacolor as functionality is in
ffmpegcolorspace.
Original commit message from CVS:
Move core plugins out of core. I don't mind fdsrc/fdsink
going back into the core; they were just disabled there, so
I moved them. Some of this stuff could (should) be deleted.
* gst/oldcore/Makefile.am:
* gst/oldcore/gstaggregator.c:
* gst/oldcore/gstaggregator.h:
* gst/oldcore/gstelements.c:
* gst/oldcore/gstfdsink.c:
* gst/oldcore/gstfdsink.h:
* gst/oldcore/gstfdsrc.c:
* gst/oldcore/gstfdsrc.h:
* gst/oldcore/gstmd5sink.c:
* gst/oldcore/gstmd5sink.h:
* gst/oldcore/gstmultifilesrc.c:
* gst/oldcore/gstmultifilesrc.h:
* gst/oldcore/gstpipefilter.c:
* gst/oldcore/gstpipefilter.h:
* gst/oldcore/gstshaper.c:
* gst/oldcore/gstshaper.h:
* gst/oldcore/gststatistics.c:
* gst/oldcore/gststatistics.h:
Original commit message from CVS:
Fixed a few things to enable the mad and effectv to be able to find the headers in the gst-plugins-base/gst-libs and to link against the libs in there.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_audio_caps_with_data):
Do actually fix invalid RIFF fmt header values for alaw
and mulaw audio instead of just saying so.
* gst/wavparse/gstwavparse.c: (gst_wavparse_fmt):
Give gst_riff_create_audio_caps_with_data() a chance to
fix up broken format header fields before extracting any
parameters from the header. (fixes#167633)
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_invert):
Declare variables at beginning of block and make gcc-2.95 happy
(fixes # 167482, patch by Gergely Nagy).
* gst/tcp/gsttcpclientsrc.c:
* gst/tcp/gsttcpclientsrc.h:
Move some includes into the header, so that struct sockaddr_in is
defined when it should be defined on FreeBSD as well (fixes
#167483).
* gst/tcp/gsttcpserversrc.c: (gst_tcpserversrc_init_receive):
Don't pass uninitialised values to setsockopt() here either.
Original commit message from CVS:
* ext/mpeg2dec/gstmpeg2dec.c:
Don't send things to NULL PAD_PEERs
* gst/deinterlace/gstdeinterlace.c: (gst_deinterlace_chain):
Copy-on-write the incoming buffer.
* gst/mpegstream/gstdvddemux.h:
* gst/mpegstream/gstmpegclock.h:
* gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_parse_syshead),
(normal_seek), (gst_mpeg_demux_handle_src_event):
* gst/mpegstream/gstmpegdemux.h:
* gst/mpegstream/gstmpegpacketize.h:
* gst/mpegstream/gstmpegparse.c:
(gst_mpeg_parse_update_streaminfo), (gst_mpeg_parse_reset),
(gst_mpeg_parse_handle_discont), (gst_mpeg_parse_parse_packhead),
(gst_mpeg_parse_loop), (gst_mpeg_parse_get_rate),
(gst_mpeg_parse_convert_src), (gst_mpeg_parse_handle_src_query),
(gst_mpeg_parse_handle_src_event), (gst_mpeg_parse_change_state):
* gst/mpegstream/gstmpegparse.h:
* gst/mpegstream/gstrfc2250enc.h:
Various changes to the way time is computed that make seeking and
total time estimation much better here.
Use G_BEGIN/END_DECLS instead of __cplusplus
* gst/videocrop/gstvideocrop.c: (gst_video_crop_chain):
Use gst_buffer_stamp instead of only copying the TIMESTAMP
Original commit message from CVS:
Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state),
(gst_qtdemux_loop_header):
Re-apply patch from #142272 that allows non-seekable sources,
re-proposed by Daniel Drake <dsd@gentoo.org>.
Original commit message from CVS:
* gst/matroska/matroska-demux.c:
(gst_matroska_demux_parse_blockgroup):
Fix logic error in timing of subtitle stream synchronization.
* gst/typefind/gsttypefindfunctions.c: (qt_type_find):
Add skip-chunk, which is found in kodak-camera streams.
Original commit message from CVS:
Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* ext/mad/gstmad.c: (gst_mad_src_event):
* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_event):
Allow seeks on audio pad, make mad forward those (#164826).
* gst/audioscale/gstaudioscale.c: (gst_audioscale_chain):
Set duration (#165335).
Original commit message from CVS:
* ext/a52dec/gsta52dec.c: (gst_a52dec_push),
(gst_a52dec_handle_event), (gst_a52dec_chain):
Add some debug output. Check that a discont has a valid
time associated.
* ext/alsa/gstalsasink.c: (gst_alsa_sink_check_event),
(gst_alsa_sink_loop):
Ignore TAG events. A little extra debug for broken timestamps.
* ext/dvdnav/dvdnavsrc.c: (dvdnavsrc_init), (dvdnavsrc_loop),
(dvdnavsrc_change_state):
Ensure we send a discont to engage the link before we send any
other events.
* ext/dvdread/dvdreadsrc.c: (dvdreadsrc_init),
(dvdreadsrc_finalize), (_close), (_open), (_seek_title),
(_seek_chapter), (seek_sector), (dvdreadsrc_get),
(dvdreadsrc_uri_get_uri), (dvdreadsrc_uri_set_uri):
Handle URI of the form dvd://title[,chapter[,angle]]. Currently only
dvd://title works in totem because typefinding sends a seek that ends
up going back to chapter 1 regardless.
* ext/mpeg2dec/gstmpeg2dec.c:
* ext/mpeg2dec/gstmpeg2dec.h:
Output correct timestamps and handle disconts.
* ext/ogg/gstoggdemux.c: (get_relative):
Small guard against a null dereference.
* ext/pango/gsttextoverlay.c: (gst_textoverlay_finalize),
(gst_textoverlay_set_property):
Free memory when done. Don't call gst_event_filler_get_duration on
EOS events. Use GST_LOG and GST_WARNING instead of g_message and
g_warning.
* ext/smoothwave/gstsmoothwave.c: (gst_smoothwave_init),
(draw_line), (gst_smoothwave_dispose), (gst_sw_sinklink),
(gst_sw_srclink), (gst_smoothwave_chain):
Draw solid lines, prettier colours.
* gst/mpeg2sub/gstmpeg2subt.c: (gst_mpeg2subt_init):
Add a default palette that'll work for some movies.
* gst/mpegstream/gstdvddemux.c: (gst_dvd_demux_init),
(gst_dvd_demux_handle_dvd_event), (gst_dvd_demux_send_discont),
(gst_dvd_demux_send_subbuffer), (gst_dvd_demux_reset):
* gst/mpegstream/gstdvddemux.h:
* gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_send_discont),
(gst_mpeg_demux_parse_syshead), (gst_mpeg_demux_parse_pes):
* gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_init),
(gst_mpeg_parse_handle_discont), (gst_mpeg_parse_parse_packhead):
* gst/mpegstream/gstmpegparse.h:
Use PTM/NAV events when for timestamp adjustment when connected to
dvdnavsrc. Don't use many discont events where one suffices.
* gst/playback/gstplaybasebin.c: (group_destroy),
(gen_preroll_element), (gst_play_base_bin_add_element):
* gst/playback/gstplaybasebin.h:
Make sure we remove subtitles from the same bin we put them in.
* gst/subparse/gstsubparse.c: (convert_encoding), (parse_subrip),
(gst_subparse_buffer_format_autodetect),
(gst_subparse_change_state):
Fix some memleaks and invalid accesses.
* gst/typefind/gsttypefindfunctions.c: (ogganx_type_find),
(oggskel_type_find), (cmml_type_find), (plugin_init):
Some typefind functions for Annodex v3.0 files
* gst/wavparse/gstwavparse.h:
GstRiffReadClass is the correct parent class.
Original commit message from CVS:
* gst/matroska/ebml-read.c: (gst_ebml_read_buffer):
Allow for 0-sized buffers. Fixes length query problems in
starwars.mkv from the testsuite.
Original commit message from CVS:
* ext/mad/gstmad.c: (gst_mad_check_caps_reset), (gst_mad_chain):
Fail if caps negotiation fails. Should fix#162184, and should
definately be in there regardless of it fixing the actual bug.
* gst/avi/gstavimux.c: (gst_avimux_get_type), (gst_avimux_init),
(gst_avimux_write_tag), (gst_avimux_riff_get_avi_header),
(gst_avimux_riff_get_avix_header),
(gst_avimux_riff_get_video_header),
(gst_avimux_riff_get_audio_header), (gst_avimux_write_index),
(gst_avimux_start_file), (gst_avimux_handle_event),
(gst_avimux_change_state):
* gst/avi/gstavimux.h:
Refactor structure writing to use GST_WRITE_UINT macros, add
metadata writing support.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state),
(gst_qtdemux_loop_header), (gst_qtdemux_handle_esds):
More memory leak fixes (#149162).
Original commit message from CVS:
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_use_event):
Don't bail on unknown events.
* gst/audioscale/gstaudioscale.c: (gst_audioscale_chain):
Don't crash on events before negotiation.
* gst/avi/gstavidemux.c: (gst_avi_demux_add_stream):
Send tags on pads, too.
* gst/playback/gststreamselector.c:
(gst_stream_selector_request_new_pad):
Forward events on first pad if no input was selected yet.
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_change_state):
Reset variables on READY.
* gst/matroska/matroska-mux.c: (gst_matroska_mux_request_new_pad),
(gst_matroska_mux_loop):
Require data before writing header.
Original commit message from CVS:
* ext/dv/gstdvdec.c:
Fix audio caps i just broke (missing ',')
* gst/matroska/matroska-mux.c: (gst_matroska_mux_get_type),
(gst_matroska_mux_reset):
Fix typo + add FIXME about old "x-gst-metadata" crap
Original commit message from CVS:
* gst-libs/gst/riff/riff-ids.h:
* gst/wavenc/riff.h:
Add AMR (VBR and CBR) ids to riff.h audio codec list
* gst/asfdemux/gstasfdemux.c:
(gst_asf_demux_process_ext_content_desc),
(gst_asf_demux_process_object):
Retrieve more tags from ASF files (Genre, AlbumTitle, Artist)
Original commit message from CVS:
Remove time-based check for first vorbis packet altogether, as it
was a hack since day one (Arwed who wrote it says so)...
Original commit message from CVS:
Fix Vorbis streams failing to decode in some files, where cluster_time isn't 0,
because then it doesn't send codec_priv before actual data.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_type_get), (qtdemux_audio_caps):
* gst/typefind/gsttypefindfunctions.c: (q3gp_type_find),
(plugin_init):
Add 3GP (variables name Q3GP because they can't start with a
number). Add samr audio fourcc (used in .3gp files), decoder
is work in progress. Also do a GST_WARNING instead of ERROR
in case of unknown nodes, to decrease output.
Original commit message from CVS:
* gst/matroska/matroska-demux.c:
(gst_matroska_demux_parse_blockgroup):
Save position, so that queries give proper return values. Don't
know how this could ever have worked before...
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_stream_index),
(gst_avi_demux_stream_scan):
Add some more debug. Fix logic error when setting movi offset
while reading index.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_stream_index),
(gst_avi_demux_stream_scan), (gst_avi_demux_handle_seek),
(gst_avi_demux_process_next_entry):
Add some debugging. Better detection of broken indexes and the
accompanying index recovery. No infinite loops on state changes
when we're still in our loopfunction.
Original commit message from CVS:
* gst/matroska/matroska-demux.c: (gst_matroska_ebmlnum_uint),
(gst_matroska_ebmlnum_sint), (gst_matroska_demux_parse_blockgroup):
Lace sizes can be zero.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_stream_index):
Work for truncated (unfinished download etc.) files. Fixes#160514.
Original commit message from CVS:
* gst/matroska/matroska-demux.c: (aac_rate_idx), (aac_profile_idx),
(gst_matroska_demux_audio_caps):
Some MPEG-AAC hacks, because else it doesn't work...
Original commit message from CVS:
* gst/asfdemux/README
* gst/wavenc/riff.h
* gst-libs/gst/riff/riff-ids.h
* gst-libs/gst/riff/riff-media.c
add new 4CC codes for h263 related codecs
fixes partially bug #155163
Original commit message from CVS:
* gst/matroska/matroska-demux.c:
(gst_matroska_demux_handle_src_query):
Don't set DEFAULT, unsupported - makes length display incorrectly
in some cases.
Original commit message from CVS:
* ext/a52dec/gsta52dec.c: (gst_a52dec_init),
(gst_a52dec_handle_event), (gst_a52dec_update_streaminfo),
(gst_a52dec_handle_frame), (gst_a52dec_chain),
(gst_a52dec_change_state), (plugin_init):
* ext/a52dec/gsta52dec.h:
Do something useful with timestamps. Make chain-based (since
there's really no reason to be loopbased).
* gst/avi/gstavidemux.c: (gst_avi_demux_process_next_entry):
Update current_byte/frame correctly.
Original commit message from CVS:
* gst/matroska/ebml-read.c: (gst_ebml_read_class_init),
(gst_ebml_read_init), (gst_ebml_read_use_event),
(gst_ebml_read_element_id), (gst_ebml_peek_id),
(gst_ebml_read_seek), (gst_ebml_read_skip),
(gst_ebml_read_reserve), (gst_ebml_read_buffer),
(gst_ebml_read_master):
* gst/matroska/ebml-read.h:
* gst/matroska/matroska-demux.c:
(gst_matroska_demux_parse_contents),
(gst_matroska_demux_loop_stream), (gst_matroska_demux_audio_caps):
Disgustingly evil hack for working around INTERRUPT events and
their extremely annoying habit of being a pain in the ass. We
simply peek a cluster before reading any of it.
Original commit message from CVS:
Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* gst/law/alaw-decode.c: (alawdec_getcaps):
* gst/law/mulaw-decode.c: (mulawdec_getcaps):
Prevent warnings when negotiating caps (fixes#159338).
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_stream_index),
(gst_avi_demux_massage_index):
Fix quite humiliating bug in omitting 0-sized index chunks but
forgetting to count them for timestamps.
Original commit message from CVS:
* configure.ac:
* ext/libvisual/visual.c: (gst_visual_get_type),
(libvisual_log_handler), (gst_visual_getcaps),
(gst_visual_srclink), (gst_visual_change_state), (make_valid_name),
(plugin_init):
Update libvisual to 0.1.7. Link in the debug handling to gstreamer
* ext/smoothwave/Makefile.am:
* ext/smoothwave/demo-osssrc.c: (main):
* ext/smoothwave/gstsmoothwave.c: (gst_smoothwave_class_init),
(gst_smoothwave_init), (gst_smoothwave_dispose), (gst_sw_sinklink),
(gst_sw_srclink), (gst_smoothwave_chain), (gst_sw_change_state),
(plugin_init):
* ext/smoothwave/gstsmoothwave.h:
Make gstsmoothwave a working element in the 20th century.
* gst/chart/gstchart.c: (gst_chart_init), (gst_chart_srcconnect):
Fix incorrect link function
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_get_caps_internal):
buffer-frames property was missing
* ext/arts/gst_arts.c:
rate missing from sinkcaps
* ext/audiofile/gstafparse.c:
* ext/audiofile/gstafsink.c:
* ext/audiofile/gstafsrc.c:
* ext/swfdec/gstswfdec.c:
int audio doesn't know buffer-frames
* ext/cdparanoia/gstcdparanoia.c:
int audio doesn't know chunksize either
* ext/nas/nassink.c:
it's endianness, not endianess
* gst-libs/gst/audio/audio.h:
make float standard pad template caps really describe float
* gst/law/mulaw.c: (linear_factory):
signed only, please
* gst/mpegstream/gstdvddemux.c:
widths of 20 are not valid
Original commit message from CVS:
* gst/videomixer/videomixer.c: (gst_videomixer_blend_ayuv_ayuv),
(gst_videomixer_fill_checker), (gst_videomixer_fill_color),
(gst_videomixer_blend_buffers), (gst_videomixer_loop):
Only mix AYUV for maximum quality.
Original commit message from CVS:
Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* gst/rtp/gstrtpgsmparse.c: (gst_rtpgsm_caps_nego):
Add missing NULL terminator (#157543).
Original commit message from CVS:
* gst/videomixer/videomixer.c: (gst_videomixer_blend_ayuv_i420),
(gst_videomixer_fill_checker), (gst_videomixer_fill_color),
(gst_videomixer_blend_buffers), (gst_videomixer_loop):
Fix stride issues. Does not completely work for odd
heights.
Original commit message from CVS:
* gst/alpha/gstalpha.c: (gst_alpha_method_get_type),
(gst_alpha_chroma_key), (gst_alpha_chain):
Fix stride issues. Does not completely work for odd
heights.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header),
(gst_qtdemux_add_stream), (qtdemux_parse), (qtdemux_parse_tree),
(qtdemux_parse_udta), (qtdemux_tag_add), (gst_qtdemux_handle_esds):
Change all g_print()s to debugging. Add a bunch of consistency
checks.
Original commit message from CVS:
Reviewd by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* gst/wavparse/gstwavparse.c: (gst_wavparse_fmt):
Fix memleak (#155223).
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiofilter.c: (gst_audiofilter_link):
fix link function to always query channels and query width for
floats
* configure.ac:
add equalizer dir
* gst/equalizer/Makefile.am:
* gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_get_type),
(gst_iir_equalizer_base_init), (gst_iir_equalizer_class_init),
(gst_iir_equalizer_init), (gst_iir_equalizer_finalize),
(arg_to_scale), (setup_filter),
(gst_iir_equalizer_compute_frequencies),
(gst_iir_equalizer_set_property), (gst_iir_equalizer_get_property),
(gst_iir_equalizer_filter_inplace), (gst_iir_equalizer_setup),
(plugin_init):
add an equalizer
Original commit message from CVS:
* gst/avi/gstavimux.c: (gst_avimux_audsinkconnect),
(gst_avimux_stop_file):
First calculate the rate, and only then use it. Hdr.rate is a
multiple and not a derivative of hdr.scale. Scale is not the
same as blockalign but is solely related to rate.
Original commit message from CVS:
* gst/spectrum/Makefile.am:
* gst/spectrum/demo-osssrc.c: (spectrum_chain), (main),
(idle_func):
Fix demo and reenable it. Yes, I'm currently playing with audio
analysis tools
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header),
(qtdemux_parse), (gst_qtdemux_handle_esds):
An esds box is not a container.
Fix parsing of mp4v boxes.
Do not try to renegotiate fps for each frame. Need to
find a better method. This should fix mp4 playback.
Original commit message from CVS:
* configure.ac: update for swfdec-0.3 and liboil-0.2
* ext/swfdec/gstswfdec.c: update for swfdec-0.3
* ext/swfdec/gstswfdec.h: same
* gst/videofilter/gstvideobalance.c: update for liboil-0.2
* gst/videotestsrc/videotestsrc.c: same
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnomevfssrc_get),
(gst_gnomevfssrc_srcpad_query), (gst_gnomevfssrc_srcpad_event):
Some debug.
* gst/avi/gstavidemux.c: (gst_avi_demux_reset),
(gst_avi_demux_handle_src_event), (gst_avi_demux_read_superindex),
(gst_avi_demux_read_subindexes), (gst_avi_demux_add_stream),
(gst_avi_demux_stream_index), (gst_avi_demux_skip),
(gst_avi_demux_sync), (gst_avi_demux_stream_scan),
(gst_avi_demux_massage_index), (gst_avi_demux_stream_header):
* gst/avi/gstavidemux.h:
Support for openDML-2.0 indx/ix## chunks. Support for broken index
recovery (where, if part of the index is broken, we will still read
the rest of the index and recover the broken part by stream
scanning). More broken media support. EOS workarounds. General AVI
braindamage headache recovery. Aspirin included.
Original commit message from CVS:
Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* gst/wavenc/gstwavenc.c: (gst_wavenc_stop_file):
Fix wrong discont event setup (fixes#154967).
Original commit message from CVS:
Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* gst/auparse/gstauparse.c: (gst_auparse_chain):
Error out on invalid data (fixes#154807).
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_massage_index):
OK, so the original code was too strict. It makes random AVI files
hang for seconds upon opening, which is unacceptable and is far
beyond the original goal of getting multiple chunks for one-chunk
sounc stream files. So now do just that.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c:
add ATRAC3 to STATIC CAPS to fix a warning
* gst/matroska/ebml-read.c:
* gst-libs/gst/riff/riff-read.c:
fix typos
Original commit message from CVS:
* gst/wavparse/Makefile.am
* gst/wavparse/riff.h
* gst/wavparse/wavparse.vcproj
riff.h removal (unused and duplication with riff-ids.h)
Original commit message from CVS:
* gst/flx/gstflxdec.c: (gst_flxdec_init), (gst_flxdec_loop):
Actually _do_ negotiation. Pass gdouble as arg instead
of guint64 for the framerate.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_event):
Fix seeking in some files. All this code is no longer needed (and
actually breaks stuff) because we now synchronize the full index
right when reading the header.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_stream_index),
(gst_avi_demux_stream_scan), (sort), (gst_avi_demux_massage_index),
(gst_avi_demux_stream_header), (gst_avi_demux_stream_data):
Improve allocation, cutting and sorting of the index. How takes a
few seconds instead of minutes.
Original commit message from CVS:
gstwavparse.c: it did not build in system with Glib < 2.4 because it
used the macro G_MAXUINT32. Now we define the macro if it is not yet
defined.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data),
(gst_riff_create_video_template_caps):
Add wing commander format mimetype/fourccs.
* gst/avi/gstavidemux.c: (gst_avi_demux_massage_index):
Don't crash if some value is 0.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data),
(gst_riff_create_video_template_caps):
Add DIB fourcc (raw, palettized 8-bit RGB).
* gst-libs/gst/riff/riff-read.c:
(gst_riff_read_strf_vids_with_data):
Oops, fix strf_data reading bug.
* gst/avi/gstavidemux.c: (gst_avi_demux_add_stream):
Use a non-NULL tag.
* gst/qtdemux/qtdemux.c: (qtdemux_parse_trak):
Time for hacks. Sorry Dave. At least one quicktime movie (a
trailer) that I've encountered contains multiple video tracks.
One of those is the actual video track, the other are one-frame
tracks (images). Unfortunately, the number of frames according
to the trak header is 1 for each, so that doesn't help. So
instead, I look at the duration and discard tracks with a
duration shorter than 20% of the length of the stream. Better
than nothing.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_class_init),
(gst_wavparse_stream_init), (gst_wavparse_fmt),
(gst_wavparse_other), (gst_wavparse_loop),
(gst_wavparse_pad_convert), (gst_wavparse_pad_query),
(gst_wavparse_srcpad_event):
* gst/wavparse/gstwavparse.h:
Added some more debugging info.
Fix the case where the length of the file is 0.
Make sure we seek to sample borders.
Original commit message from CVS:
Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* gst/wavparse/gstwavparse.c: (gst_wavparse_fmt):
Throw error if we didn't recognize the stream. Fixes#152289.
Original commit message from CVS:
Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* gst/wavparse/gstwavparse.c: (gst_wavparse_fmt):
Fix memleak.
Original commit message from CVS:
* ext/dvdread/dvdreadsrc.c: (dvdreadsrc_class_init),
(dvdreadsrc_init), (dvdreadsrc_dispose), (dvdreadsrc_set_property),
(dvdreadsrc_get_property), (_open), (_seek), (_read),
(dvdreadsrc_get), (dvdreadsrc_open_file),
(dvdreadsrc_change_state):
Fix. Don't do one big huge loop around the whole DVD, that will
cache all data and thus eat sizeof(dvd) (several GB) before we
see something.
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_seek):
Actually NULL'ify event after using it.
* gst/matroska/ebml-read.c: (gst_ebml_read_use_event),
(gst_ebml_read_handle_event), (gst_ebml_read_element_id),
(gst_ebml_read_element_length), (gst_ebml_read_element_data),
(gst_ebml_read_seek), (gst_ebml_read_skip):
Handle events.
* gst/mpegstream/gstdvddemux.c: (gst_dvd_demux_base_init),
(gst_dvd_demux_init), (gst_dvd_demux_get_audio_stream),
(gst_dvd_demux_get_subpicture_stream), (gst_dvd_demux_plugin_init):
Fix timing (this will probably break if I seek using menus, but
I didn't get there yet). VOBs and normal DVDs should now work.
Add a mpeg2-only pad with high rank so this get autoplugged for
MPEG-2 movies.
* gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_base_init),
(gst_mpeg_demux_class_init), (gst_mpeg_demux_init),
(gst_mpeg_demux_new_output_pad), (gst_mpeg_demux_get_video_stream),
(gst_mpeg_demux_get_audio_stream),
(gst_mpeg_demux_get_private_stream), (gst_mpeg_demux_parse_packet),
(gst_mpeg_demux_parse_pes), (gst_mpeg_demux_plugin_init):
Use this as second rank for MPEG-1 and MPEG-2. Still use this for
MPEG-1 but use dvddemux for MPEG-2.
* gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_class_init),
(gst_mpeg_parse_init), (gst_mpeg_parse_new_pad),
(gst_mpeg_parse_parse_packhead):
Timing. Only add pad template if it exists. Add sink template from
class and not from ourselves. This means we will always use the
correct sink template even if it is not the one defined in this
file.
Original commit message from CVS:
* ext/flac/gstflacdec.c: (gst_flacdec_src_query):
Only return true if we actually filled something in. Prevents
player applications from showing a random length for flac files.
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_class_init),
(gst_riff_read_use_event), (gst_riff_read_handle_event),
(gst_riff_read_seek), (gst_riff_read_skip), (gst_riff_read_strh),
(gst_riff_read_strf_vids_with_data),
(gst_riff_read_strf_auds_with_data), (gst_riff_read_strf_iavs):
OK, ok, so I implemented event handling. Apparently it's normal
that we receive random events at random points without asking
for it.
* gst/avi/gstavidemux.c: (gst_avi_demux_reset),
(gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query),
(gst_avi_demux_handle_src_event), (gst_avi_demux_stream_index),
(gst_avi_demux_sync), (gst_avi_demux_stream_scan),
(gst_avi_demux_massage_index), (gst_avi_demux_stream_header),
(gst_avi_demux_handle_seek), (gst_avi_demux_process_next_entry),
(gst_avi_demux_stream_data), (gst_avi_demux_loop):
* gst/avi/gstavidemux.h:
Implement non-lineair chunk handling and subchunk processing.
The first solves playback of AVI files where the audio and video
data of individual buffers that we read are not synchronized.
This should not happen according to the wonderful AVI specs, but
of course it does happen in reality. It is also a prerequisite for
the second. Subchunk processing allows us to cut chunks in small
pieces and process each of these pieces separately. This is
required because I've seen several AVI files with incredibly large
audio chunks, even some files with only one audio chunk for the
whole file. This allows for proper playback including seeking.
This patch is supposed to fix all AVI A/V sync issues.
* gst/flx/gstflxdec.c: (gst_flxdec_class_init),
(flx_decode_chunks), (flx_decode_color), (gst_flxdec_loop):
Work.
* gst/modplug/gstmodplug.cc:
Proper return value setting for the query() function.
* gst/playback/gstplaybasebin.c: (setup_source):
Being in non-playing state (after, e.g., EOS) is not necessarily
a bad thing. Allow for that. This fixes playback of short files.
They don't actually playback fully now, because the clock already
runs. This means that small files (<500kB) with a small length
(<2sec) will still not or barely play. Other files, such as mod
or flx, will work correctly, however.
Original commit message from CVS:
Company's wisdom:
Events should be passed on using the sinkpad's default handler not the src
Seek events only go upstream, so send a discont downstream instead.
Original commit message from CVS:
* ext/dirac/Makefile.am:
* ext/dirac/gstdirac.cc:
* ext/dirac/gstdiracdec.cc:
* ext/dirac/gstdiracdec.h:
Do something. Don't actually know if this works because I don't
have a demuxer yet.
* ext/gsm/gstgsmdec.c: (gst_gsmdec_getcaps):
Add channels=1 to caps returned from _getcaps().
* ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_get_type),
(gst_ogm_video_parse_get_type), (gst_ogm_audio_parse_base_init),
(gst_ogm_video_parse_base_init), (gst_ogm_parse_init),
(gst_ogm_audio_parse_init), (gst_ogm_video_parse_init),
(gst_ogm_parse_sink_convert), (gst_ogm_parse_chain),
(gst_ogm_parse_change_state):
Separate between audio/video so ogmaudioparse actually uses the
audio pad templates. Both audio and video work now, including
autoplugging. Also use sometimes-srcpad hack.
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_seek):
Handle events better. Don't hang on infinite loops.
* gst/avi/gstavidemux.c: (gst_avi_demux_class_init),
(gst_avi_demux_init), (gst_avi_demux_reset),
(gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query),
(gst_avi_demux_stream_header), (gst_avi_demux_stream_data),
(gst_avi_demux_change_state):
* gst/avi/gstavidemux.h:
Improve A/V sync. Still not perfect.
* gst/matroska/ebml-read.c: (gst_ebml_read_seek),
(gst_ebml_read_skip):
Handle events better.
* gst/qtdemux/qtdemux.c: (gst_qtdemux_handle_sink_event),
(gst_qtdemux_loop_header), (qtdemux_parse_trak),
(qtdemux_audio_caps):
Add IMA4. Improve event handling. Save offset after a seek when
the headers are at the end of the file so that we don't end up in
an infinite loop.
* gst/typefind/gsttypefindfunctions.c: (qt_type_find):
Add low-priority typefind support for files with no length.
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c: (_read_var_length), (_read_guid),
(gst_asf_demux_process_segment), (gst_asf_demux_handle_data),
(gst_asf_demux_process_chunk), (gst_asf_demux_handle_sink_event):
Prevent infinite loops. More correct error reporting.
* gst/auparse/gstauparse.c: (gst_auparse_chain):
Error out if negotiation fails.
* gst/playback/gstplaybasebin.c: (setup_source),
(gst_play_base_bin_change_state), (gst_play_base_bin_error),
(gst_play_base_bin_found_tag):
Error/tag forwarding. Pre-roll fixes for source errors on state
changes (e.g. "file does not exist") to prevent hangs.
Original commit message from CVS:
2004-09-19 Zaheer Abbas Merali <zaheerabbas at merali dot org>
* gst/wavenc/gstwavenc.c: (gst_wavenc_init), (gst_wavenc_chain):
* gst/wavenc/gstwavenc.h:
Added newmedia support to wavenc
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
(gst_avi_demux_stream_data):
Just hardcode for raw audio then. AVI audio sucks.
Original commit message from CVS:
* configure.ac: remove NASM check, since we don't use it. Update
dirac check to 0.4
* ext/dirac/gstdiracdec.cc: update to current 0.4 API
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link):
Initialized variables.
* gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state),
(gst_qtdemux_loop_header), (qtdemux_parse), (qtdemux_parse_trak),
(gst_qtdemux_handle_esds), (qtdemux_audio_caps): Fix seeking, add
SVQ3 format
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
(gst_avi_demux_add_stream), (gst_avi_demux_stream_data):
* gst/avi/gstavidemux.h:
Fix for compressed audio (mp3) timestamp generation. How did this
ever work?
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_add_stream),
(qtdemux_parse_trak):
Don't crash by dividing by zero (see sample movie in #126922).
Original commit message from CVS:
* gst/videomixer/videomixer.c: (gst_videomixer_blend_buffers):
Copy timestamps from the master pad to the output buffers.
Original commit message from CVS:
Write track and segment UIDs, write muxing date, write TRACKDEFAULTDURATION for TTA audio, write BLOCKDURATION if known.
Original commit message from CVS:
Fix byte order reversion for writing ebml floats.
Write segment duration and muxing application in matroska.
Added TTA codec to the list of supported codecs to mux into matroska.
Original commit message from CVS:
Interpret BLOCKDURATION and set buffer duration accordingly, enable demuxing
of TTA audio from matroska, fixes bugs #148950 and #148951.
Original commit message from CVS:
* gst/udp/gstudpsrc.c: (gst_udpsrc_init), (gst_udpsrc_get):
* gst/udp/gstudpsrc.h:
Don't call gst_pad_push in a get function. Fixes#150449
Original commit message from CVS:
2004-07-28 Zaheer Abbas Merali <zaheerabbas at merali dot org>
* ext/lame/gstlame.c: (gst_lame_chain): send tag events downstream
* ext/shout2/gstshout2.c: (gst_shout2send_protocol_get_type),
(gst_shout2send_get_type), (gst_shout2send_set_clock),
(gst_shout2send_class_init), (gst_shout2send_init),
(set_shout_metadata), (gst_shout2send_set_metadata),
(gst_shout2send_chain), (gst_shout2send_set_property),
(gst_shout2send_get_property), (gst_shout2send_connect),
(gst_shout2send_change_state):
* ext/shout2/gstshout2.h:
- fix for sending mp3 audio to icecast2 server, if pad link function not
called before PAUSED state
- added option to use GStreamer clock sync (as opposed to libshout's own sync)
- added tagging support for mp3 audio broadcasted
* gst/monoscope/gstmonoscope.c: (gst_monoscope_class_init):
debug info
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_fmt),
(gst_wavparse_handle_seek), (gst_wavparse_srcpad_event):
Add the pad to the element after setting up the caps. This
makes it a lot easier to autoplug.
Original commit message from CVS:
* gst/videomixer/videomixer.c: (gst_videomixer_pad_get_type),
(gst_videomixer_pad_class_init), (gst_videomixer_pad_get_property),
(gst_videomixer_pad_set_property),
(gst_videomixer_pad_sinkconnect), (gst_videomixer_pad_init),
(gst_video_mixer_background_get_type), (gst_videomixer_get_type),
(gst_videomixer_class_init), (gst_videomixer_init),
(gst_videomixer_getcaps), (gst_videomixer_request_new_pad),
(gst_videomixer_blend_ayuv_i420), (pad_zorder_compare),
(gst_videomixer_sort_pads), (gst_videomixer_fill_checker),
(gst_videomixer_fill_color), (gst_videomixer_fill_queues),
(gst_videomixer_blend_buffers), (gst_videomixer_update_queues),
(gst_videomixer_loop), (plugin_init):
Be a nicer negotiation citizen and provide a getcaps function on
the srcpad. This also fixes a crash when resizing.
Original commit message from CVS:
Set the explicit caps on the pad when the file is parsed as explicit caps get wiped during state changes. fixes bug #148043
Original commit message from CVS:
* gst/alpha/Makefile.am:
* gst/alpha/gstalphacolor.c: (gst_alpha_color_get_type),
(gst_alpha_color_base_init), (gst_alpha_color_class_init),
(gst_alpha_color_init), (gst_alpha_color_set_property),
(gst_alpha_color_get_property), (gst_alpha_color_sink_link),
(transform), (gst_alpha_color_chain),
(gst_alpha_color_change_state), (plugin_init):
Stupid plugin to to RGBA to AYUV conversion because none of
the colorspace plugins can handle that yet.
