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* gst/auparse/gstauparse.c: limit caps to the formats we announce in the templateStefan Kost2007-04-102-25/+55
| | | | | | | | | | 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
* gst/: Fix some compiler warnings. Fixes #428182.Peter Kjellerstedt2007-04-106-3/+9
| | | | | | | | | | | | | | 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.
* gst/rtsp/: Morph RTPDec into something compatible with RTPBin as a fallback.Wim Taymans2007-04-068-212/+776
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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.
* gst/qtdemux/gstrtpxqtdepay.*: Try to recover from packet loss a little better.Wim Taymans2007-04-052-1/+25
| | | | | | | | 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.
* gst/rtp/gstrtpmp4adepay.c: This element is ready to be autoplugged.Wim Taymans2007-04-051-1/+1
| | | | | | | 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.
* gst/avi/gstavidemux.c: Don't leave the offsets defined by upstream element ↵Julien Moutte2007-04-051-0/+2
| | | | | | | | | | | | 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.
* gst/avi/: Don't abort on out-of-memory. Use stream-nr as unsigned integer only.Stefan Kost2007-04-042-41/+77
| | | | | | | | | | | | | | 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.
* gst/smpte/barboxwipes.c: Wim Taymans2007-04-031-1/+1
| | | | | | Original commit message from CVS: * gst/smpte/barboxwipes.c: Fix error as spotted by Snaik <snaik32 at gmail dot com>
* gst/wavparse/gstwavparse.c: Support audio/x-raw-float in wav files. This ↵Sebastian Dröge2007-03-301-0/+5
| | | | | | | | | | 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.
* Revert last change as we don't want plugins-good to depend on plugins-base ↵Sebastian Dröge2007-03-302-2/+41
| | | | | | | | | | | | 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.
* configure.ac: Require gst-plugins-base CVS for audioconvert with non-native ↵René Stadler2007-03-292-41/+2
| | | | | | | | | | | | | | | 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.
* gst/rtp/: Flush adapter on disconts.Wim Taymans2007-03-297-13/+61
| | | | | | | | | | | | | | | | 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.
* gst/rtp/: Use more efficient adapter and rtpbuffer methods when possible.Wim Taymans2007-03-2914-75/+24
| | | | | | | | | | | | | | | | | | | | 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.
* gst/wavenc/gstwavenc.c: Correctly handle width!=depth input.Sebastian Dröge2007-03-292-12/+17
| | | | | | | | | | 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.
* gst/udp/: Rework the socket allocation a bit based on the sockfd argument so ↵Laurent Glayal2007-03-294-34/+94
| | | | | | | | | | | | | | | | | | | | | | 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
* gst/rtp/: Added H264 payloader. Fixes #423782.Laurent Glayal2007-03-295-10/+404
| | | | | | | | | | | | | | | | | | 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.
* gst/wavparse/gstwavparse.c: Actually support depths from 1 to 32, not only 8 ↵Sebastian Dröge2007-03-281-1/+1
| | | | | | | | to 32. Original commit message from CVS: * gst/wavparse/gstwavparse.c: Actually support depths from 1 to 32, not only 8 to 32.
* gst/wavparse/gstwavparse.c: Add support for wav files containing ↵Sebastian Dröge2007-03-281-1/+1
| | | | | | | | | 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.
* gst/rtp/: Added MP4A-LATM depayloader. Fixes #417792.Stefan Kost2007-03-286-44/+460
| | | | | | | | | | | | | | | | | | | | | | 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.
* gst/qtdemux/: Process 'ctts' atoms, which are present in AVC ISO files (.mov ↵Edward Hervey2007-03-285-4/+53
| | | | | | | | | | | | | | | | | 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
* gst/rtsp/gstrtspsrc.*: Handle default clock-rates for static payload types, ↵Wim Taymans2007-03-252-32/+190
| | | | | | | | | | | | | | | | | | | | 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.
* gst/interleave/deinterleave.c: Remove 'channel-positions' field when munging ↵Tim-Philipp Müller2007-03-241-0/+1
| | | | | | | | | | | 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).
* gst/interleave/deinterleave.c: Don't leak input buffer in chain function; ↵Tim-Philipp Müller2007-03-221-28/+26
| | | | | | | | | | | | | 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.
* gst/apetag/gsttagdemux.c: Rename registered type in preparation of ↵Tim-Philipp Müller2007-03-211-2/+1
| | | | | | | | | 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.
* gst/wavparse/gstwavparse.c: Streaming mode fixes: don't unref buffer we ↵Tim-Philipp Müller2007-03-191-6/+1
| | | | | | | | | 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.
* gst/equalizer/gstiirequalizer10bands.c: A 10 band EQ should be initialized ↵Stefan Kost2007-03-161-1/+18
| | | | | | | | | 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.
* gst/rtp/: Ported mulaw and alaw payloaders to use new base classPhilippe Kalaf2007-03-144-278/+24
| | | | | | | | | 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
* gst/equalizer/: Add 3 and 10 band version and add missing ↵Stefan Kost2007-03-148-24/+541
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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.
* gst/apetag/gsttagdemux.c: Fix handling of -1 values for start and stop ↵Tim-Philipp Müller2007-03-121-3/+16
| | | | | | | | | 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.
* gst/id3demux/gstid3demux.c: Fix handling of -1 values for start and stop ↵Jan Schmidt2007-03-121-3/+14
| | | | | | | | | 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.
* gst/id3demux/id3v2frames.c: Fix parsing of ID3 v2.2.0 PIC frames. Only in ↵Tim-Philipp Müller2007-03-121-7/+32
| | | | | | | | | | | | 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'.
* Printf format fixes; also add some missing quotes in translated strings. ↵Tim-Philipp Müller2007-03-101-10/+10
| | | | | | | | | | | | 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.
* gst/autodetect/gstautoaudiosink.c: Tim and I can't think of any reason the ↵Jan Schmidt2007-03-091-1/+0
| | | | | | | | | | | | 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.
* gst/rtsp/gstrtspsrc.c: When activated, remove the udpsrc timeout, we have ↵Wim Taymans2007-03-091-0/+5
| | | | | | | | | 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.
* gst/avi/gstavidemux.c: Fix stream position reporting after a seek. Fixes ↵Wim Taymans2007-03-091-0/+2
| | | | | | | | | | | #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.
* gst/equalizer/: Refactor plugin into a base class and a first subclass ↵Stefan Kost2007-03-095-133/+487
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | (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.
* gst/avi/gstavidemux.c: Make avidemux accept optional header chunks in any order.René Stadler2007-03-081-58/+71
| | | | | | | | | | 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.
* gst/audiofx/: Add new audiodynamic element which can act as a compressor or ↵Sebastian Dröge2007-03-084-3/+787
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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.
* gst/qtdemux/qtdemux.*: Share qtdemux debug category across all files, ↵Edward Hervey2007-03-072-3/+4
| | | | | | | | | | 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.
* gst/level/gstlevel.*: Resolve message timestamps against the playback segment.Stefan Kost2007-03-072-5/+49
| | | | | | | | | 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.
* gst/spectrum/gstspectrum.*: One FIXME less, by resolving message timestamps ↵Stefan Kost2007-03-072-3/+21
| | | | | | | | | | | 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.
* gst/id3demux/gstid3demux.c: Don't leak caps: make gst_id3demux_add_srcpad() ↵Tim-Philipp Müller2007-03-061-6/+7
| | | | | | | | | | | | | | | | | | | | | | | | | 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.
* gst/id3demux/: Do not convert obsolete TDA/TDAT frames to TDRC frames, ↵Tim-Philipp Müller2007-03-063-6/+44
| | | | | | | | | | | | | | | 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.
* gst/spectrum/gstspectrum.c: Fix and cleanup default property values.Wim Taymans2007-03-061-6/+17
| | | | | | | | | 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.
* gst/goom/gstgoom.*: Document, fix and improve goom adapter behaviour.Wim Taymans2007-03-062-15/+25
| | | | | | | | | 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.
* gst/rtp/: Fix encoding-name case.Wim Taymans2007-03-053-3/+3
| | | | | | | | 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.
* gst/rtp/: Fix speex (de)payloader. Fixes #358040.Wim Taymans2007-03-053-23/+242
| | | | | | | | | | | | | 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.
* gst/multipart/multipartdemux.c: Use gst_pad_new_from_static_template instead ↵Jan Schmidt2007-03-041-4/+5
| | | | | | | | | | 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.
* Fix a bunch of leaks shown by the newly-added states test.Jan Schmidt2007-03-0411-6/+86
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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.
* Don't mix tabs and spaces (#414168).Loïc Minier2007-03-031-8/+8
| | | | | | | | 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).
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