Thomas Vander Stichele [Thu, 12 Apr 2007 11:41:11 +0000 (11:41 +0000)]
gst/rtp/Makefile.am: gst/rtp/fnv1hash.c (MASK_24, FNV1_HASH_32_INIT, FNV1_HASH_32_PRIME, fnv1_hash_32_new, fnv1_hash_...
Original commit message from CVS:
* gst/rtp/Makefile.am:
* gst/rtp/fnv1hash.c (MASK_24, FNV1_HASH_32_INIT, FNV1_HASH_32_PRIME,
fnv1_hash_32_new, fnv1_hash_32_update, fnv1_hash_32_to_24):
* gst/rtp/fnv1hash.h (__GST_FNV1_HASH_H__):
Add a simple hashing implementation that we can use to generate
a 24-bit ident value based on the codebooks for vorbis and theora.
* gst/rtp/gstrtptheorapay.c (gst_rtp_theora_pay_finish_headers,
gst_rtp_theora_pay_handle_buffer):
* gst/rtp/gstrtpvorbisdepay.c
(gst_rtp_vorbis_depay_parse_configuration,
gst_rtp_vorbis_depay_switch_codebook, gst_rtp_vorbis_depay_process):
* gst/rtp/gstrtpvorbispay.c (gst_rtp_vorbis_pay_reset_packet,
gst_rtp_vorbis_pay_init_packet, gst_rtp_vorbis_pay_flush_packet,
gst_rtp_vorbis_pay_finish_headers, gst_rtp_vorbis_pay_handle_buffer):
Use the hashing function, ensuring that the same codebooks result
in the same ident and thus the same SDP description.
Various log fixes/changes.
jerry tan [Thu, 12 Apr 2007 11:37:50 +0000 (11:37 +0000)]
sys/sunaudio/gstsunaudiosrc.c: it is the application's responsibility to make sure it open the device once.
Original commit message from CVS:
Patch by: jerry tan <jerry dot tan at sun dot com>
* sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_open):
remove the call of ioctl (fd, AUDIO_MIXER_MULTIPLE_OPEN), it is the
application's responsibility to make sure it open the device once.
Remove a careless error if AUDIODEV is set. Fixes #392620.
Wim Taymans [Thu, 12 Apr 2007 10:52:02 +0000 (10:52 +0000)]
gst/qtdemux/qtdemux.c: Make timescale 32 bits again so we don't screw up the pts_offset calculations.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c:
Make timescale 32 bits again so we don't screw up the pts_offset
calculations.
Wim Taymans [Thu, 12 Apr 2007 08:21:28 +0000 (08:21 +0000)]
gst/rtsp/gstrtpdec.*: Make backward compat with rtpbin by adding the request-pt-map signals.
Original commit message from CVS:
* gst/rtsp/gstrtpdec.c: (gst_rtp_dec_marshal_BOXED__UINT_UINT),
(gst_rtp_dec_class_init), (gst_rtp_dec_chain_rtp):
* gst/rtsp/gstrtpdec.h:
Make backward compat with rtpbin by adding the request-pt-map signals.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
(new_session_pad), (request_pt_map),
(gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_stream_configure_caps),
(gst_rtspsrc_activate_streams):
* gst/rtsp/gstrtspsrc.h:
Implement request-pt-map signals instead of setting caps on the buffers
for the session manager.
Wim Taymans [Wed, 11 Apr 2007 10:25:25 +0000 (10:25 +0000)]
gst/udp/gstudp.c: Register GstNetBuffer in plugin_init so that the type can be used from multiple threads without races.
Original commit message from CVS:
* gst/udp/gstudp.c: (plugin_init):
Register GstNetBuffer in plugin_init so that the type can be used from
multiple threads without races.
Christian Schaller [Wed, 11 Apr 2007 10:19:06 +0000 (10:19 +0000)]
update to spec file
Original commit message from CVS:
update to spec file
Wim Taymans [Wed, 11 Apr 2007 09:53:38 +0000 (09:53 +0000)]
gst/qtdemux/: Handle version 1 mdhd atoms to get extended precision durations.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_get_duration),
(qtdemux_parse_samples), (qtdemux_parse_segments),
(qtdemux_parse_trak), (qtdemux_parse_tree):
* gst/qtdemux/qtdemux_dump.c: (qtdemux_dump_mdhd):
Handle version 1 mdhd atoms to get extended precision durations.
Fixes #426972.
Wim Taymans [Tue, 10 Apr 2007 17:06:05 +0000 (17:06 +0000)]
gst/rtp/gstrtpamrdepay.c: Fix depayloader clock_rate and some cleanups.
Original commit message from CVS:
* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_setcaps),
(gst_rtp_amr_depay_process):
Fix depayloader clock_rate and some cleanups.
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_finalize),
(gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process):
* gst/rtp/gstrtph264depay.h:
Don't push codec_data in the adapter because it might get flushed when
we get a discont.
* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_process):
Handle multiple AU per packet.
* gst/rtp/gstrtpsv3vdepay.c: (gst_rtp_sv3v_depay_process),
(gst_rtp_sv3v_depay_plugin_init):
Disable rank, this one does not work.
Remove timestamping, base class does that.
Stefan Kost [Tue, 10 Apr 2007 12:01:33 +0000 (12:01 +0000)]
gst/auparse/gstauparse.c: limit caps to the formats we announce in the template
Original commit message from CVS:
* gst/auparse/gstauparse.c: (gst_au_parse_parse_header):
limit caps to the formats we announce in the template
* gst/wavparse/gstwavparse.c: (uint64_ceiling_scale_int),
(gst_wavparse_perform_seek), (gst_wavparse_stream_headers),
(gst_wavparse_add_src_pad), (gst_wavparse_stream_data):
fix some crashers/asserts when dealing with broken files
Peter Kjellerstedt [Tue, 10 Apr 2007 10:01:14 +0000 (10:01 +0000)]
gst/: Fix some compiler warnings. Fixes #428182.
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/avi/gstavidemux.c: (gst_avi_demux_massage_index):
* gst/rtp/gstrtpL16depay.c:
* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_process):
* gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_get_mode),
(gst_rtp_speex_depay_setcaps):
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps),
(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_loop_udp):
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_init_send):
Fix some compiler warnings. Fixes #428182.
Wim Taymans [Fri, 6 Apr 2007 12:54:16 +0000 (12:54 +0000)]
gst/rtsp/: Morph RTPDec into something compatible with RTPBin as a fallback.
Original commit message from CVS:
* gst/rtsp/Makefile.am:
* gst/rtsp/gstrtpdec.c: (find_session_by_id), (create_session),
(free_session), (gst_rtp_dec_base_init), (gst_rtp_dec_class_init),
(gst_rtp_dec_init), (gst_rtp_dec_finalize),
(gst_rtp_dec_query_src), (gst_rtp_dec_chain_rtp),
(gst_rtp_dec_chain_rtcp), (gst_rtp_dec_set_property),
(gst_rtp_dec_get_property), (gst_rtp_dec_provide_clock),
(gst_rtp_dec_change_state), (create_recv_rtp), (create_recv_rtcp),
(create_rtcp), (gst_rtp_dec_request_new_pad),
(gst_rtp_dec_release_pad):
* gst/rtsp/gstrtpdec.h:
* gst/rtsp/gstrtsp.c: (plugin_init):
Morph RTPDec into something compatible with RTPBin as a fallback.
Various other style fixes.
* gst/rtsp/gstrtspsrc.c: (find_stream_by_id),
(find_stream_by_udpsrc), (gst_rtspsrc_stream_free),
(gst_rtspsrc_cleanup), (gst_rtspsrc_media_to_caps),
(new_session_pad), (gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_loop_udp), (gst_rtspsrc_setup_auth),
(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Implement RTPBin session manager handling.
Don't try to add empty properties to caps.
Implement fallback session manager, handling.
Don't combine errors from RTCP streams, just ignore them.
* gst/rtsp/rtsptransport.c: (rtsp_transport_get_manager):
* gst/rtsp/rtsptransport.h:
Implement fallback session manager.
Make RTPBin the default one when available.
Wim Taymans [Thu, 5 Apr 2007 15:05:24 +0000 (15:05 +0000)]
gst/qtdemux/gstrtpxqtdepay.*: Try to recover from packet loss a little better.
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.
Wim Taymans [Thu, 5 Apr 2007 13:56:44 +0000 (13:56 +0000)]
gst/rtp/gstrtpmp4adepay.c: This element is ready to be autoplugged.
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.
Julien Moutte [Thu, 5 Apr 2007 11:26:25 +0000 (11:26 +0000)]
gst/avi/gstavidemux.c: Don't leave the offsets defined by upstream element on the compressed data buffer we are pushi...
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.
Stefan Kost [Wed, 4 Apr 2007 12:39:41 +0000 (12:39 +0000)]
gst/avi/: Don't abort on out-of-memory. Use stream-nr as unsigned integer only.
Original commit message from CVS:
* gst/avi/README:
* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
(gst_avi_demux_parse_stream), (gst_avi_demux_parse_index),
(gst_avi_demux_stream_index), (gst_avi_demux_sync),
(gst_avi_demux_stream_scan), (gst_avi_demux_massage_index),
(gst_avi_demux_calculate_durations_from_index),
(gst_avi_demux_stream_header_push),
(gst_avi_demux_stream_header_pull), (gst_avi_demux_combine_flows),
(gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data):
Don't abort on out-of-memory. Use stream-nr as unsigned integer only.
Wim Taymans [Tue, 3 Apr 2007 09:55:45 +0000 (09:55 +0000)]
gst/smpte/barboxwipes.c:
Original commit message from CVS:
* gst/smpte/barboxwipes.c:
Fix error as spotted by Snaik <snaik32 at gmail dot com>
Sebastian Dröge [Fri, 30 Mar 2007 17:19:34 +0000 (17:19 +0000)]
gst/wavparse/gstwavparse.c: Support audio/x-raw-float in wav files. This only works with plugins-base CVS, using an o...
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.
Sebastian Dröge [Fri, 30 Mar 2007 15:59:27 +0000 (15:59 +0000)]
Revert last change as we don't want plugins-good to depend on plugins-base CVS now.
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.
Sebastian Dröge [Fri, 30 Mar 2007 04:50:11 +0000 (04:50 +0000)]
ext/wavpack/: Don't play audioconvert. As wavpack wants/outputs all samples with width==32 and depth=[1,32] accept th...
Original commit message from CVS:
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_reset),
(gst_wavpack_dec_init), (gst_wavpack_dec_sink_set_caps),
(gst_wavpack_dec_clip_outgoing_buffer),
(gst_wavpack_dec_post_tags), (gst_wavpack_dec_chain):
* ext/wavpack/gstwavpackdec.h:
* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_reset),
(gst_wavpack_enc_sink_set_caps), (gst_wavpack_enc_set_wp_config),
(gst_wavpack_enc_chain):
* ext/wavpack/gstwavpackenc.h:
* ext/wavpack/gstwavpackparse.c:
Don't play audioconvert. As wavpack wants/outputs all samples with
width==32 and depth=[1,32] accept this and let audioconvert convert
to accepted formats instead of doing it in the element for n*8 depths.
This also adds support for non-n*8 depths and prevents some useless
memory allocations. Fixes #421598
Also add a workaround for bug #421542 in wavpackenc for now...
* tests/check/elements/wavpackdec.c: (GST_START_TEST):
* tests/check/elements/wavpackenc.c: (GST_START_TEST):
* tests/check/elements/wavpackparse.c: (GST_START_TEST):
Consider the change above in the unit tests and test if the correct
caps are accepted and set. Also check for GST_BUFFER_OFFSET_END in
the wavpackparse unit test.
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_init),
(gst_wavpack_dec_sink_set_caps):
Set caps on the src pad as soon as possible.
* ext/wavpack/gstwavpackdec.h:
* ext/wavpack/gstwavpackcommon.h:
* ext/wavpack/gstwavpackenc.h:
* ext/wavpack/gstwavpackparse.h:
Fix indention. gst-indent is now called by cicl.
René Stadler [Thu, 29 Mar 2007 18:51:33 +0000 (18:51 +0000)]
configure.ac: Require gst-plugins-base CVS for audioconvert with non-native float support and width/depth fix in libg...
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.
Wim Taymans [Thu, 29 Mar 2007 14:40:35 +0000 (14:40 +0000)]
gst/rtp/: Flush adapter on disconts.
Original commit message from CVS:
* gst/rtp/gstasteriskh263.h:
* gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_process),
(gst_rtp_h263p_depay_change_state):
* gst/rtp/gstrtph263pdepay.h:
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_class_init),
(gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process),
(gst_rtp_h264_depay_change_state):
* gst/rtp/gstrtph264depay.h:
* gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_class_init),
(gst_rtp_mp4a_depay_setcaps), (gst_rtp_mp4a_depay_process):
* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_process):
Flush adapter on disconts.
Wim Taymans [Thu, 29 Mar 2007 14:03:21 +0000 (14:03 +0000)]
gst/rtp/: Use more efficient adapter and rtpbuffer methods when possible.
Original commit message from CVS:
* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16_depay_process):
* gst/rtp/gstrtpgsmdepay.c: (gst_rtp_gsm_depay_process):
* gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_process):
* gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_process):
* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_process):
* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_flush):
* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_setcaps),
(gst_rtp_mp4v_depay_process):
* gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_flush):
* gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_process):
* gst/rtp/gstrtpmpapay.c: (gst_rtp_mpa_pay_flush):
* gst/rtp/gstrtpmpvdepay.c: (gst_rtp_mpv_depay_process):
* gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_process):
* gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_process):
* gst/rtp/gstrtpsv3vdepay.c: (gst_rtp_sv3v_depay_process):
Use more efficient adapter and rtpbuffer methods when possible.
Sebastian Dröge [Thu, 29 Mar 2007 12:14:22 +0000 (12:14 +0000)]
gst/wavenc/gstwavenc.c: Correctly handle width!=depth input.
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.
Laurent Glayal [Thu, 29 Mar 2007 09:59:23 +0000 (09:59 +0000)]
gst/udp/: Rework the socket allocation a bit based on the sockfd argument so that it becomes usable.
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
Laurent Glayal [Thu, 29 Mar 2007 08:08:49 +0000 (08:08 +0000)]
gst/rtp/: Added H264 payloader. Fixes #423782.
Original commit message from CVS:
Patch by: Laurent Glayal <spglegle at yahoo dot fr>
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_base_init),
(gst_rtp_h264_pay_class_init), (gst_rtp_h264_pay_init),
(gst_rtp_h264_pay_finalize), (gst_rtp_h264_pay_setcaps),
(gst_rtp_h264_pay_handle_buffer), (gst_rtp_h264_pay_set_property),
(gst_rtp_h264_pay_get_property), (gst_rtp_h264_pay_change_state),
(gst_rtp_h264_pay_plugin_init):
* gst/rtp/gstrtph264pay.h:
Added H264 payloader. Fixes #423782.
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_class_init),
(gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process):
Small fixes.
Sebastian Dröge [Wed, 28 Mar 2007 22:27:36 +0000 (22:27 +0000)]
gst/wavparse/gstwavparse.c: Actually support depths from 1 to 32, not only 8 to 32.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c:
Actually support depths from 1 to 32, not only 8 to 32.
Sebastian Dröge [Wed, 28 Mar 2007 22:23:43 +0000 (22:23 +0000)]
gst/wavparse/gstwavparse.c: Add support for wav files containing audio/x-raw-int with random depths between 1 and 32 ...
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.
Stefan Kost [Wed, 28 Mar 2007 18:40:12 +0000 (18:40 +0000)]
gst/rtp/: Added MP4A-LATM depayloader. Fixes #417792.
Original commit message from CVS:
Based on patch by: Stefan Kost <ensonic@users.sf.net>
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_base_init),
(gst_rtp_mp4a_depay_class_init), (gst_rtp_mp4a_depay_init),
(gst_rtp_mp4a_depay_finalize), (gst_rtp_mp4a_depay_setcaps),
(gst_rtp_mp4a_depay_process), (gst_rtp_mp4a_depay_set_property),
(gst_rtp_mp4a_depay_get_property),
(gst_rtp_mp4a_depay_change_state),
(gst_rtp_mp4a_depay_plugin_init):
* gst/rtp/gstrtpmp4adepay.h:
Added MP4A-LATM depayloader. Fixes #417792.
* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_setcaps),
(gst_rtp_mp4v_depay_process):
Fixup depayloader, setting codec_data, using more efficient adaptor and
rtpbuffer handling.
* gst/rtsp/URLS:
Add url to test above.
Edward Hervey [Wed, 28 Mar 2007 15:17:27 +0000 (15:17 +0000)]
gst/qtdemux/: Process 'ctts' atoms, which are present in AVC ISO files (.mov files with h264 video).
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
Wim Taymans [Sun, 25 Mar 2007 15:34:42 +0000 (15:34 +0000)]
gst/rtsp/gstrtspsrc.*: Handle default clock-rates for static payload types, rearrange stuff so that the rtpmap field ...
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.
Tim-Philipp Müller [Sat, 24 Mar 2007 19:46:59 +0000 (19:46 +0000)]
gst/interleave/deinterleave.c: Remove 'channel-positions' field when munging input caps into 1-channel output caps (I...
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).
Tim-Philipp Müller [Thu, 22 Mar 2007 22:14:29 +0000 (22:14 +0000)]
gst/interleave/deinterleave.c: Don't leak input buffer in chain function; maintain our own list of source pads - ther...
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.
Sebastian Dröge [Thu, 22 Mar 2007 16:25:56 +0000 (16:25 +0000)]
ext/wavpack/gstwavpackparse.c: Revert last commit, preventing infinite plugging loops with ranks is no clean solution...
Original commit message from CVS:
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_plugin_init):
Revert last commit, preventing infinite plugging loops with ranks
is no clean solution and in general there's no reason why one wants
to parse framed wavpack data again.
Sebastian Dröge [Thu, 22 Mar 2007 15:52:51 +0000 (15:52 +0000)]
ext/wavpack/gstwavpackenc.c: Send the new segment event in time format instead of bytes. This allows "wavpackenc ! wa...
Original commit message from CVS:
* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_push_block):
Send the new segment event in time format instead of bytes. This
allows "wavpackenc ! wavpackdec ! someaudiosink" pipelines.
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_plugin_init):
Accept framed and non-framed input, wavpackparse doesn't care. To
prevent "wavpackparse ! wavpackparse ! ..." pipelines lower the
rank of wavpackparse by one. This allows "wavpackenc ! wavpackparse !
..." pipelines.
Sebastian Dröge [Thu, 22 Mar 2007 11:08:03 +0000 (11:08 +0000)]
ext/wavpack/gstwavpackdec.c: Revert to use gst_pad_alloc_buffer() here. We can and should use it.
Original commit message from CVS:
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_chain):
Revert to use gst_pad_alloc_buffer() here. We can and should use it.
Thanks to Jan and Mike for noticing my mistake.
Christophe Dehais [Thu, 22 Mar 2007 09:44:17 +0000 (09:44 +0000)]
ext/gconf/gconf.c: Accept complex pipeline descriptions as an audio profile instead of just a single element. Fixes #...
Original commit message from CVS:
Patch by: Christophe Dehais <christophe dot dehais at gmail dot com>
* ext/gconf/gconf.c: (gst_gconf_render_bin_with_default):
Accept complex pipeline descriptions as an audio profile instead of just
a single element. Fixes #420658.
Sebastian Dröge [Thu, 22 Mar 2007 00:17:41 +0000 (00:17 +0000)]
ext/wavpack/gstwavpackenc.*: Put the write helpers into the GstWavpackEnc struct directly and not as a pointer to sav...
Original commit message from CVS:
* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_class_init),
(gst_wavpack_enc_init), (gst_wavpack_enc_chain),
(gst_wavpack_enc_rewrite_first_block):
* ext/wavpack/gstwavpackenc.h:
Put the write helpers into the GstWavpackEnc struct directly and not
as a pointer to save two small, but useless mallocs. This also makes
it possible to drop the finalize method.
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_push_buffer):
For consistency reasons also set GST_BUFFER_OFFSET_END on the outgoing
buffers the same way wavpackenc does it.
Sebastian Dröge [Wed, 21 Mar 2007 23:50:09 +0000 (23:50 +0000)]
ext/wavpack/gstwavpackdec.c: Don't use gst_pad_alloc_buffer() as we might clip the buffer later and
Original commit message from CVS:
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_chain):
Don't use gst_pad_alloc_buffer() as we might clip the buffer later and
BaseTransform-based elements will likely break because of wrong
unit-size. Also plug a possible memleak that happens when decoding
fails for some reason.
Tim-Philipp Müller [Wed, 21 Mar 2007 11:49:32 +0000 (11:49 +0000)]
gst/apetag/gsttagdemux.c: Rename registered type in preparation of GstTagDemux moving to
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.
Tim-Philipp Müller [Mon, 19 Mar 2007 10:29:19 +0000 (10:29 +0000)]
gst/wavparse/gstwavparse.c: Streaming mode fixes: don't unref buffer we don't own any longer; remove bogus adapter fl...
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.
David Schleef [Sun, 18 Mar 2007 04:21:28 +0000 (04:21 +0000)]
REQUIREMENTS: Change the format to key/value, add a bunch of information, remove a bunch of requirements that are for...
Original commit message from CVS:
* REQUIREMENTS: Change the format to key/value, add a bunch of
information, remove a bunch of requirements that are for
other GStreamer packages.
David Schleef [Sun, 18 Mar 2007 02:00:54 +0000 (02:00 +0000)]
REQUIREMENTS: Fix a few things. This file really needs a good once-over.
Original commit message from CVS:
* REQUIREMENTS: Fix a few things. This file really needs a
good once-over.
Edward Hervey [Fri, 16 Mar 2007 18:38:18 +0000 (18:38 +0000)]
sys/osxvideo/osxvideosink.m: Fix previous commit, we want to pass the NSView in the message.
Original commit message from CVS:
* sys/osxvideo/osxvideosink.m:
Fix previous commit, we want to pass the NSView in the message.
Edward Hervey [Fri, 16 Mar 2007 16:27:20 +0000 (16:27 +0000)]
sys/osxvideo/osxvideosink.m: Emit 'have-ns-view' message when working in embedded mode. The message will contain a po...
Original commit message from CVS:
* sys/osxvideo/osxvideosink.m:
Emit 'have-ns-view' message when working in embedded mode. The message
will contain a pointer to the newly created NSView.
Stefan Kost [Fri, 16 Mar 2007 09:57:40 +0000 (09:57 +0000)]
gst/equalizer/gstiirequalizer10bands.c: A 10 band EQ should be initialized to 1 bands and not to 3.
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.
Edward Hervey [Thu, 15 Mar 2007 12:05:01 +0000 (12:05 +0000)]
sys/Makefile.am: Don't forget to distribute the sys/osxaudio/ directory.
Original commit message from CVS:
* sys/Makefile.am:
Don't forget to distribute the sys/osxaudio/ directory.
Edward Hervey [Thu, 15 Mar 2007 11:39:53 +0000 (11:39 +0000)]
Activate osxaudio in gst-plugins-good with proper build setup.
Original commit message from CVS:
* configure.ac:
* sys/Makefile.am:
* sys/osxaudio/Makefile.am:
* sys/osxaudio/gstosxaudio.c:
* sys/osxaudio/gstosxaudiosink.c:
(gst_osx_audio_sink_osxelement_do_init), (gst_osx_audio_sink_init),
(gst_osx_audio_sink_getcaps),
(gst_osx_audio_sink_create_ringbuffer), (plugin_init):
* sys/osxaudio/gstosxaudiosrc.c:
(gst_osx_audio_src_osxelement_do_init), (gst_osx_audio_src_init),
(gst_osx_audio_src_create_ringbuffer):
* sys/osxaudio/gstosxringbuffer.c: (gst_osx_ring_buffer_get_type),
(gst_osx_ring_buffer_class_init), (gst_osx_ring_buffer_init),
(gst_osx_ring_buffer_acquire), (gst_osx_ring_buffer_start),
(gst_osx_ring_buffer_pause), (gst_osx_ring_buffer_stop):
* sys/osxaudio/gstosxringbuffer.h:
Activate osxaudio in gst-plugins-good with proper build setup.
Add inlined documentation.
Fix debug statements
Fix ringbuffer when pausing.
Fixes #323471
Philippe Kalaf [Wed, 14 Mar 2007 22:21:26 +0000 (22:21 +0000)]
gst/rtp/: Ported mulaw and alaw payloaders to use new base class
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
Edward Hervey [Wed, 14 Mar 2007 16:30:19 +0000 (16:30 +0000)]
sys/osxvideo/: Fix leaks when running a NSApp.
Original commit message from CVS:
* sys/osxvideo/cocoawindow.h:
* sys/osxvideo/cocoawindow.m:
* sys/osxvideo/osxvideosink.h:
* sys/osxvideo/osxvideosink.m:
Fix leaks when running a NSApp.
Accept any kind of resolutions.
Works in fullscreen. Can maximize.
Only thing left before being able to move this to -good is documentation
and embedded window support.
Thomas Vander Stichele [Wed, 14 Mar 2007 15:25:10 +0000 (15:25 +0000)]
po/: Update translations.
Original commit message from CVS:
* po/af.po:
* po/az.po:
* po/cs.po:
* po/en_GB.po:
* po/it.po:
* po/nl.po:
* po/or.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/uk.po:
* po/vi.po:
Update translations.
Tim-Philipp Müller [Wed, 14 Mar 2007 14:49:45 +0000 (14:49 +0000)]
configure.ac: Fix string replace error (AG_AG_GST_* => AG_GST_*).
Original commit message from CVS:
* configure.ac:
Fix string replace error (AG_AG_GST_* => AG_GST_*).
Stefan Kost [Wed, 14 Mar 2007 14:48:08 +0000 (14:48 +0000)]
gst/equalizer/: Add 3 and 10 band version and add missing gst_object_sync_values.
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_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), (setup_filter),
(gst_iir_equalizer_compute_frequencies),
(gst_iir_equalizer_transform_ip), (plugin_init):
* gst/equalizer/gstiirequalizer10bands.c:
(gst_iir_equalizer_10bands_base_init),
(gst_iir_equalizer_10bands_class_init),
(gst_iir_equalizer_10bands_init),
(gst_iir_equalizer_10bands_set_property),
(gst_iir_equalizer_10bands_get_property):
* gst/equalizer/gstiirequalizer10bands.h:
* gst/equalizer/gstiirequalizer3bands.c:
(gst_iir_equalizer_3bands_base_init),
(gst_iir_equalizer_3bands_class_init),
(gst_iir_equalizer_3bands_init),
(gst_iir_equalizer_3bands_set_property),
(gst_iir_equalizer_3bands_get_property):
* gst/equalizer/gstiirequalizer3bands.h:
* gst/equalizer/gstiirequalizernbands.c:
(gst_iir_equalizer_nbands_base_init),
(gst_iir_equalizer_nbands_init):
Add 3 and 10 band version and add missing gst_object_sync_values.
* gst/spectrum/gstspectrum.c: (gst_spectrum_event),
(gst_spectrum_transform_ip):
Add some comments about float support.
Tim-Philipp Müller [Mon, 12 Mar 2007 17:56:54 +0000 (17:56 +0000)]
gst/apetag/gsttagdemux.c: Fix handling of -1 values for start and stop values when seeking, and SEEK_CUR+SEEK_END her...
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.
Jan Schmidt [Mon, 12 Mar 2007 17:24:23 +0000 (17:24 +0000)]
gst/id3demux/gstid3demux.c: Fix handling of -1 values for start and stop values when seeking, and SEEK_CUR+SEEK_END.
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.
Jan Schmidt [Mon, 12 Mar 2007 15:49:02 +0000 (15:49 +0000)]
I'm too lazy to comment this
Original commit message from CVS:
Add Patch by: line for wim, since he's away
Tim-Philipp Müller [Mon, 12 Mar 2007 13:28:29 +0000 (13:28 +0000)]
gst/id3demux/id3v2frames.c: Fix parsing of ID3 v2.2.0 PIC frames. Only in version >= 2.3.0 is the image format a vari...
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'.
Sébastien Moutte [Sun, 11 Mar 2007 22:23:04 +0000 (22:23 +0000)]
sys/directdraw/gstdirectdrawsink.*: Handle display mode changes during playback.
Original commit message from CVS:
* sys/directdraw/gstdirectdrawsink.c:
* sys/directdraw/gstdirectdrawsink.h:
Handle display mode changes during playback.
Sébastien Moutte [Sat, 10 Mar 2007 16:07:31 +0000 (16:07 +0000)]
win32/MANIFEST: Add new project files to MANIFEST.
Original commit message from CVS:
* win32/MANIFEST:
Add new project files to MANIFEST.
* win32/vs6/libgstaudiofx.dsp:
* win32/vs6/libgstrtp.dsp:
* win32/vs6/libgstrtsp.dsp:
Update project files.
Tim-Philipp Müller [Sat, 10 Mar 2007 12:30:48 +0000 (12:30 +0000)]
Printf format fixes; also add some missing quotes in translated strings. Fixes #416728 and #416727.
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.
Jan Schmidt [Fri, 9 Mar 2007 20:12:08 +0000 (20:12 +0000)]
gst/autodetect/gstautoaudiosink.c: Tim and I can't think of any reason the child audio sink needs to be set back to N...
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.
Tim-Philipp Müller [Fri, 9 Mar 2007 19:51:27 +0000 (19:51 +0000)]
po/: Add ja.po file from #377306.
Original commit message from CVS:
* po/LINGUAS:
* po/ja.po:
Add ja.po file from #377306.
Tim-Philipp Müller [Fri, 9 Mar 2007 19:44:30 +0000 (19:44 +0000)]
sys/sunaudio/: Actually translate sunaudio mixer track labels instead of just marking the strings as translatable (#3...
Original commit message from CVS:
* sys/sunaudio/gstsunaudio.c: (plugin_init):
* sys/sunaudio/gstsunaudiomixertrack.c:
(gst_sunaudiomixer_track_new):
Actually translate sunaudio mixer track labels instead of just
marking the strings as translatable (#377306); clean up weird
label string mapping code that serves no apparent purpose. Also
set the 'untranslated-label' property when creating mixer tracks
if the GstMixerTrack base class supports this.
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/sunaudio.c: (GST_START_TEST),
(sunaudio_suite):
Very minimalistic unit test for sunaudiomixer element (compiles, but not
actually tested on a system where sunaudiomixer is available).
Jan Schmidt [Fri, 9 Mar 2007 18:49:37 +0000 (18:49 +0000)]
tests/check/Makefile.am: Re-enable the states test and see if it works on the buildbots.
Original commit message from CVS:
* tests/check/Makefile.am:
Re-enable the states test and see if it works on the buildbots.
Wim Taymans [Fri, 9 Mar 2007 17:32:32 +0000 (17:32 +0000)]
ext/dv/gstdvdec.*: Infer pixel-aspect-ratio from the video frame format if it isn't provided by the container, as hap...
Original commit message from CVS:
* ext/dv/gstdvdec.c: (gst_dvdec_init), (gst_dvdec_sink_setcaps),
(gst_dvdec_src_negotiate), (gst_dvdec_chain),
(gst_dvdec_change_state):
* ext/dv/gstdvdec.h:
Infer pixel-aspect-ratio from the video frame format if it isn't
provided by the container, as happens when playing DV from AVI
or Quicktime containers.
Patch by: Wim Taymans <wim@fluendo.com>
Fixes #380944
Wim Taymans [Fri, 9 Mar 2007 17:05:17 +0000 (17:05 +0000)]
gst/rtsp/gstrtspsrc.c: When activated, remove the udpsrc timeout, we have dataflow and timeouts will later be handled...
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.
Wim Taymans [Fri, 9 Mar 2007 16:53:39 +0000 (16:53 +0000)]
ext/taglib/gstid3v2mux.cc: Add write support for GST_TAG_MUSICBRAINZ_SORTNAME (TSOP) tag.
Original commit message from CVS:
* ext/taglib/gstid3v2mux.cc:
Add write support for GST_TAG_MUSICBRAINZ_SORTNAME (TSOP) tag.
Fixes #414496.
Wim Taymans [Fri, 9 Mar 2007 15:04:45 +0000 (15:04 +0000)]
gst/avi/gstavidemux.c: Fix stream position reporting after a seek. Fixes #416445.
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.
Stefan Kost [Fri, 9 Mar 2007 08:58:26 +0000 (08:58 +0000)]
gst/equalizer/: Refactor plugin into a base class and a first subclass (nband eq). The nband eq uses GstChildProxy an...
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.
René Stadler [Thu, 8 Mar 2007 16:01:42 +0000 (16:01 +0000)]
gst/avi/gstavidemux.c: Make avidemux accept optional header chunks in any order.
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.
Jan Schmidt [Thu, 8 Mar 2007 12:23:57 +0000 (12:23 +0000)]
tests/check/Makefile.am: Disable the states check until the remaining Valgrind errors are fixed or suppressed.
Original commit message from CVS:
* tests/check/Makefile.am:
Disable the states check until the remaining Valgrind errors
are fixed or suppressed.
Sebastian Dröge [Thu, 8 Mar 2007 10:24:43 +0000 (10:24 +0000)]
tests/check/elements/.cvsignore: Add audiodynamic check to .cvsignore
Original commit message from CVS:
* tests/check/elements/.cvsignore:
Add audiodynamic check to .cvsignore
Sebastian Dröge [Thu, 8 Mar 2007 10:02:12 +0000 (10:02 +0000)]
gst/audiofx/: Add new audiodynamic element which can act as a compressor or expander. Supported are hard-knee and sof...
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.
Jan Schmidt [Wed, 7 Mar 2007 19:48:03 +0000 (19:48 +0000)]
ext/raw1394/gstdv1394src.c: Free handles that we allocated when exiting via the error paths.
Original commit message from CVS:
* ext/raw1394/gstdv1394src.c: (gst_dv1394src_start):
Free handles that we allocated when exiting via the error paths.
Sebastian Dröge [Wed, 7 Mar 2007 12:07:07 +0000 (12:07 +0000)]
ext/wavpack/: Use a general wavpack debug category for common code.
Original commit message from CVS:
* ext/wavpack/gstwavpack.c: (plugin_init):
* ext/wavpack/gstwavpackcommon.c:
Use a general wavpack debug category for common code.
* ext/wavpack/gstwavpackstreamreader.c:
(gst_wavpack_stream_reader_set_pos_abs),
(gst_wavpack_stream_reader_set_pos_rel),
(gst_wavpack_stream_reader_write_bytes):
Use the general wavpack debug category here too and add debug
output to the functions that should not be called at all by
the wavpack library.
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_plugin_init):
* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_plugin_init):
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_plugin_init):
Change debugging category names to conform to the conventions.
Edward Hervey [Wed, 7 Mar 2007 11:37:23 +0000 (11:37 +0000)]
gst/qtdemux/qtdemux.*: Share qtdemux debug category across all files, otherwise all debugging in files other than qtd...
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.
Stefan Kost [Wed, 7 Mar 2007 11:24:05 +0000 (11:24 +0000)]
gst/level/gstlevel.*: Resolve message timestamps against the playback segment.
Original commit message from CVS:
* gst/level/gstlevel.c: (gst_level_class_init),
(gst_level_set_caps), (gst_level_start), (gst_level_event),
(gst_level_transform_ip):
* gst/level/gstlevel.h:
Resolve message timestamps against the playback segment.
Stefan Kost [Wed, 7 Mar 2007 11:23:20 +0000 (11:23 +0000)]
gst/spectrum/gstspectrum.*: One FIXME less, by resolving message timestamps against the playback segment.
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.
Tim-Philipp Müller [Tue, 6 Mar 2007 23:21:41 +0000 (23:21 +0000)]
Fix ChangeLog message
Original commit message from CVS:
Fix ChangeLog message
Tim-Philipp Müller [Tue, 6 Mar 2007 23:19:30 +0000 (23:19 +0000)]
gst/id3demux/gstid3demux.c: Don't leak caps: make gst_id3demux_add_srcpad() not take ownership of the caps passed to ...
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.
Tim-Philipp Müller [Tue, 6 Mar 2007 22:14:59 +0000 (22:14 +0000)]
tests/check/Makefile.am: Add missing backslash at end of line.
Original commit message from CVS:
* tests/check/Makefile.am:
Add missing backslash at end of line.
Jan Schmidt [Tue, 6 Mar 2007 18:36:09 +0000 (18:36 +0000)]
Trigger rebuild.
Original commit message from CVS:
Trigger rebuild.
Tim-Philipp Müller [Tue, 6 Mar 2007 18:16:49 +0000 (18:16 +0000)]
gst/id3demux/: Do not convert obsolete TDA/TDAT frames to TDRC frames, otherwise the four-digit number will be interp...
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.
Jan Schmidt [Tue, 6 Mar 2007 14:53:04 +0000 (14:53 +0000)]
ext/gconf/gstswitchsink.c: Fix up the dispose logic so it doesn't leak, and fix setting of the child state so that we...
Original commit message from CVS:
* ext/gconf/gstswitchsink.c: (gst_switch_sink_dispose),
(gst_switch_commit_new_kid):
Fix up the dispose logic so it doesn't leak, and fix setting of
the child state so that we don't set a child to our current state
just as we are changing it to something else.
Wim Taymans [Tue, 6 Mar 2007 13:57:55 +0000 (13:57 +0000)]
gst/spectrum/gstspectrum.c: Fix and cleanup default property values.
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.
Wim Taymans [Tue, 6 Mar 2007 13:21:23 +0000 (13:21 +0000)]
gst/goom/gstgoom.*: Document, fix and improve goom adapter behaviour.
Original commit message from CVS:
* gst/goom/gstgoom.c: (gst_goom_src_setcaps), (get_buffer),
(gst_goom_chain):
* gst/goom/gstgoom.h:
Document, fix and improve goom adapter behaviour.
Fixes #407006.
Jan Schmidt [Mon, 5 Mar 2007 18:43:29 +0000 (18:43 +0000)]
ext/esd/esdsink.c: Unref static pad template after using it.
Original commit message from CVS:
* ext/esd/esdsink.c: (gst_esdsink_open):
Unref static pad template after using it.
Jan Schmidt [Mon, 5 Mar 2007 17:17:04 +0000 (17:17 +0000)]
ext/gconf/gstswitchsink.c: Fix up the reference counting of the child elements.
Original commit message from CVS:
* ext/gconf/gstswitchsink.c: (gst_switch_sink_dispose),
(gst_switch_commit_new_kid):
Fix up the reference counting of the child elements.
Wim Taymans [Mon, 5 Mar 2007 17:08:32 +0000 (17:08 +0000)]
gst/rtp/: Fix encoding-name case.
Original commit message from CVS:
* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_setcaps):
* gst/rtp/gstrtptheorapay.c: (gst_rtp_theora_pay_finish_headers):
* gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_finish_headers):
Fix encoding-name case.
Wim Taymans [Mon, 5 Mar 2007 16:39:29 +0000 (16:39 +0000)]
gst/rtp/: Fix speex (de)payloader. Fixes #358040.
Original commit message from CVS:
* gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_class_init),
(gst_rtp_speex_depay_get_mode), (gst_rtp_speex_depay_setcaps),
(gst_rtp_speex_depay_process):
* gst/rtp/gstrtpspeexpay.c: (gst_rtp_speex_pay_base_init),
(gst_rtp_speex_pay_class_init), (gst_rtp_speex_pay_setcaps),
(gst_rtp_speex_pay_parse_ident), (gst_rtp_speex_pay_handle_buffer),
(gst_rtp_speex_pay_change_state):
* gst/rtp/gstrtpspeexpay.h:
Fix speex (de)payloader. Fixes #358040.
Jan Schmidt [Mon, 5 Mar 2007 15:42:58 +0000 (15:42 +0000)]
ext/gconf/gstswitchsink.c: Install fakesink in NULL by fixing some broken logic. This obviates the need to manually s...
Original commit message from CVS:
* ext/gconf/gstswitchsink.c: (gst_switch_sink_reset),
(gst_switch_commit_new_kid), (gst_switch_sink_set_child):
Install fakesink in NULL by fixing some broken logic. This obviates
the need to manually set _IS_SINK.
Add some comments and remove a little cruft while I'm at it.
Wim Taymans [Mon, 5 Mar 2007 14:46:43 +0000 (14:46 +0000)]
ext/gconf/gstswitchsink.c: Mark us as a sink when we have no fakesink in NULL. Fixes #414887.
Original commit message from CVS:
* ext/gconf/gstswitchsink.c: (gst_switch_sink_reset):
Mark us as a sink when we have no fakesink in NULL. Fixes #414887.
Stefan Kost [Mon, 5 Mar 2007 08:30:52 +0000 (08:30 +0000)]
gst/spectrum/: Remove two obsolete and confusing comments.
Original commit message from CVS:
* gst/spectrum/demo-audiotest.c: (message_handler):
* gst/spectrum/demo-osssrc.c: (message_handler):
Remove two obsolete and confusing comments.
Tim-Philipp Müller [Sun, 4 Mar 2007 18:52:12 +0000 (18:52 +0000)]
po/POTFILES.in: Update.
Original commit message from CVS:
* po/POTFILES.in:
Update.
Jan Schmidt [Sun, 4 Mar 2007 17:33:34 +0000 (17:33 +0000)]
tests/check/Makefile.am: Gah! Also disable gconfvideosink from the tests, otherwise it will instantiate autovideosink...
Original commit message from CVS:
* tests/check/Makefile.am:
Gah! Also disable gconfvideosink from the tests, otherwise
it will instantiate autovideosink, and dfbvideosink and
leak on the buildbots.
Jan Schmidt [Sun, 4 Mar 2007 17:13:19 +0000 (17:13 +0000)]
ext/cdio/gstcdiocddasrc.c: Make sure we always destroy our libcdio handle.
Original commit message from CVS:
* ext/cdio/gstcdiocddasrc.c: (gst_cdio_cdda_src_open),
(gst_cdio_cdda_src_finalize):
Make sure we always destroy our libcdio handle.
Jan Schmidt [Sun, 4 Mar 2007 17:05:58 +0000 (17:05 +0000)]
tests/check/Makefile.am: Disable autovideosink so the buildbots don't barf over memory leaked in the directfb sink.
Original commit message from CVS:
* tests/check/Makefile.am:
Disable autovideosink so the buildbots don't barf over memory
leaked in the directfb sink.
Jan Schmidt [Sun, 4 Mar 2007 15:28:30 +0000 (15:28 +0000)]
sys/ximage/gstximagesrc.c: Chain up in dispose
Original commit message from CVS:
* sys/ximage/gstximagesrc.c: (gst_ximage_src_dispose):
Chain up in dispose
Jan Schmidt [Sun, 4 Mar 2007 15:07:15 +0000 (15:07 +0000)]
gst/multipart/multipartdemux.c: Use gst_pad_new_from_static_template instead of static_pad_template_get+pad_new.
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.
Jan Schmidt [Sun, 4 Mar 2007 14:56:53 +0000 (14:56 +0000)]
sys/ximage/gstximagesrc.c: Catch the case where no clock has been set.
Original commit message from CVS:
* sys/ximage/gstximagesrc.c: (gst_ximage_src_create):
Catch the case where no clock has been set.
Jan Schmidt [Sun, 4 Mar 2007 13:52:03 +0000 (13:52 +0000)]
Fix a bunch of leaks shown by the newly-added states test.
Original commit message from CVS:
* ext/flac/gstflacenc.c: (gst_flac_enc_finalize):
* ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_class_init),
(gst_gconf_audio_sink_dispose), (gst_gconf_audio_sink_finalize):
* ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_base_init),
(gst_gconf_audio_src_class_init), (gst_gconf_audio_src_dispose),
(gst_gconf_audio_src_finalize), (do_toggle_element):
* ext/gconf/gstgconfvideosink.c: (gst_gconf_video_sink_base_init),
(gst_gconf_video_sink_class_init), (gst_gconf_video_sink_finalize),
(do_toggle_element):
* ext/gconf/gstgconfvideosrc.c: (gst_gconf_video_src_base_init),
(gst_gconf_video_src_class_init), (gst_gconf_video_src_dispose),
(gst_gconf_video_src_finalize), (do_toggle_element):
* ext/gconf/gstswitchsink.c: (gst_switch_sink_class_init),
(gst_switch_sink_reset), (gst_switch_sink_set_child):
* ext/hal/gsthalaudiosink.c: (gst_hal_audio_sink_base_init):
* ext/hal/gsthalaudiosrc.c: (gst_hal_audio_src_base_init):
* ext/shout2/gstshout2.c: (gst_shout2send_class_init),
(gst_shout2send_init), (gst_shout2send_finalize):
* gst/debug/testplugin.c: (gst_test_class_init),
(gst_test_finalize):
* gst/flx/gstflxdec.c: (gst_flxdec_class_init),
(gst_flxdec_dispose):
* gst/multipart/multipartmux.c: (gst_multipart_mux_finalize):
* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_finalize):
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_finalize):
* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_free_context):
* gst/rtsp/rtspextwms.h:
* gst/smpte/gstsmpte.c: (gst_smpte_class_init),
(gst_smpte_finalize):
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_finalize):
* gst/udp/gstudpsink.c: (gst_udpsink_class_init),
(gst_udpsink_finalize):
* gst/wavparse/gstwavparse.c: (gst_wavparse_dispose),
(gst_wavparse_sink_activate):
* sys/oss/gstosssink.c: (gst_oss_sink_finalise):
* sys/oss/gstosssrc.c: (gst_oss_src_class_init),
(gst_oss_src_finalize):
* sys/v4l2/gstv4l2object.c: (gst_v4l2_object_destroy):
* sys/v4l2/gstv4l2object.h:
* sys/v4l2/gstv4l2src.c: (gst_v4l2src_class_init),
(gst_v4l2src_finalize):
* sys/ximage/gstximagesrc.c: (gst_ximage_src_ximage_get):
Fix a bunch of leaks shown by the newly-added states test.