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* gst/icydemux/gsticydemux.c: If the metadata strings we get in the stream are ↵Tim-Philipp Müller2007-04-122-30/+97
| | | | | | | | | | | | | not UTF-8, try to interpret them accordi... 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.
* gst/rtp/gstrtph264depay.c: Use the proper sync word for SPS and PPS.Wim Taymans2007-04-122-2/+7
| | | | | | Original commit message from CVS: * gst/rtp/gstrtph264depay.c: Use the proper sync word for SPS and PPS.
* gst/rtp/Makefile.am: gst/rtp/fnv1hash.c (MASK_24, FNV1_HASH_32_INIT, ↵Thomas Vander Stichele2007-04-127-13/+151
| | | | | | | | | | | | | | | | | | | | | | | 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.
* sys/sunaudio/gstsunaudiosrc.c: it is the application's responsibility to ↵jerry tan2007-04-122-4/+12
| | | | | | | | | | | 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.
* gst/qtdemux/qtdemux.c: Make timescale 32 bits again so we don't screw up the ↵Wim Taymans2007-04-121-1/+1
| | | | | | | | | 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.
* gst/rtsp/gstrtpdec.*: Make backward compat with rtpbin by adding the ↵Wim Taymans2007-04-125-5/+134
| | | | | | | | | | | | | | | | | | 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.
* gst/udp/gstudp.c: Register GstNetBuffer in plugin_init so that the type can ↵Wim Taymans2007-04-112-0/+12
| | | | | | | | | 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.
* update to spec fileChristian Schaller2007-04-111-0/+1
| | | | | Original commit message from CVS: update to spec file
* gst/qtdemux/: Handle version 1 mdhd atoms to get extended precision durations.Wim Taymans2007-04-112-21/+52
| | | | | | | | | | 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.
* gst/rtp/gstrtpamrdepay.c: Fix depayloader clock_rate and some cleanups.Wim Taymans2007-04-106-85/+166
| | | | | | | | | | | | | | | | | | 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.
* gst/auparse/gstauparse.c: limit caps to the formats we announce in the templateStefan Kost2007-04-103-25/+65
| | | | | | | | | | 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 Kjellerstedt2007-04-108-3/+23
| | | | | | | | | | | | | | 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 Taymans2007-04-069-212/+811
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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 Taymans2007-04-052-1/+7
| | | | | | | 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 Moutte2007-04-052-0/+9
| | | | | | | | | | | | 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 Kost2007-04-043-41/+90
| | | | | | | | | | | | | | 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 Taymans2007-04-032-1/+6
| | | | | | 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 Dröge2007-03-302-0/+12
| | | | | | | | | | 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 Dröge2007-03-304-3/+51
| | | | | | | | | | | | 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.
* ext/wavpack/: Don't play audioconvert. As wavpack wants/outputs all samples ↵Sebastian Dröge2007-03-3010-198/+138
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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.
* configure.ac: Require gst-plugins-base CVS for audioconvert with non-native ↵René Stadler2007-03-294-42/+17
| | | | | | | | | | | | | | | 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 Taymans2007-03-298-13/+76
| | | | | | | | | | | | | | | | 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 Taymans2007-03-2915-75/+43
| | | | | | | | | | | | | | | | | | | | 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 Dröge2007-03-293-12/+26
| | | | | | | | | | 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 Glayal2007-03-295-34/+114
| | | | | | | | | | | | | | | | | | | | | | 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 Glayal2007-03-296-10/+423
| | | | | | | | | | | | | | | | | | 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 Dröge2007-03-282-1/+6
| | | | | | | | 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 Dröge2007-03-282-1/+7
| | | | | | | | | 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 Kost2007-03-287-44/+484
| | | | | | | | | | | | | | | | | | | | | | 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 Hervey2007-03-286-4/+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 Taymans2007-03-253-32/+207
| | | | | | | | | | | | | | | | | | | | 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.
* ext/wavpack/gstwavpackparse.c: Revert last commit, preventing infinite ↵Sebastian Dröge2007-03-221-2/+4
| | | | | | | | | | 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.
* ext/wavpack/gstwavpackenc.c: Send the new segment event in time format ↵Sebastian Dröge2007-03-222-5/+3
| | | | | | | | | | | | | | 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.
* ext/wavpack/gstwavpackdec.c: Revert to use gst_pad_alloc_buffer() here. We ↵Sebastian Dröge2007-03-221-6/+6
| | | | | | | | | 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.
* ext/gconf/gconf.c: Accept complex pipeline descriptions as an audio profile ↵Christophe Dehais2007-03-222-7/+16
| | | | | | | | | | 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.
* ext/wavpack/gstwavpackenc.*: Put the write helpers into the GstWavpackEnc ↵Sebastian Dröge2007-03-223-25/+10
| | | | | | | | | | | | | | | | 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.
* ext/wavpack/gstwavpackdec.c: Don't use gst_pad_alloc_buffer() as we might ↵Sebastian Dröge2007-03-211-10/+8
| | | | | | | | | | | 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.
* gst/apetag/gsttagdemux.c: Rename registered type in preparation of ↵Tim-Philipp Müller2007-03-212-2/+7
| | | | | | | | | 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 Müller2007-03-192-6/+7
| | | | | | | | | 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.
* REQUIREMENTS: Change the format to key/value, add a bunch of information, ↵David Schleef2007-03-182-85/+137
| | | | | | | | | 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.
* REQUIREMENTS: Fix a few things. This file really needs a good once-over.David Schleef2007-03-182-14/+7
| | | | | | Original commit message from CVS: * REQUIREMENTS: Fix a few things. This file really needs a good once-over.
* sys/osxvideo/osxvideosink.m: Fix previous commit, we want to pass the NSView ↵Edward Hervey2007-03-161-1/+1
| | | | | | | | in the message. Original commit message from CVS: * sys/osxvideo/osxvideosink.m: Fix previous commit, we want to pass the NSView in the message.
* sys/osxvideo/osxvideosink.m: Emit 'have-ns-view' message when working in ↵Edward Hervey2007-03-161-5/+17
| | | | | | | | | 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.
* 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.
* sys/Makefile.am: Don't forget to distribute the sys/osxaudio/ directory.Edward Hervey2007-03-152-1/+6
| | | | | | Original commit message from CVS: * sys/Makefile.am: Don't forget to distribute the sys/osxaudio/ directory.
* Activate osxaudio in gst-plugins-good with proper build setup.Edward Hervey2007-03-159-54/+129
| | | | | | | | | | | | | | | | | | | | | | | | | 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
* gst/rtp/: Ported mulaw and alaw payloaders to use new base classPhilippe Kalaf2007-03-145-278/+31
| | | | | | | | | 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
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