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* gst/wavparse/gstwavparse.c: Relax the audio/mpeg caps again and add FIXME: ↵Stefan Kost2007-04-131-0/+7
| | | | | | | | | | comment. Original commit message from CVS: * gst/wavparse/gstwavparse.c: (uint64_ceiling_scale_int), (gst_wavparse_perform_seek), (gst_wavparse_stream_headers), (gst_wavparse_stream_data): Relax the audio/mpeg caps again and add FIXME: comment.
* gst/wavparse/gstwavparse.*: More sanity check for the header fields. Fix ↵Stefan Kost2007-04-131-0/+9
| | | | | | | | | | | | type for 'rate' header field. Original commit message from CVS: * gst/wavparse/gstwavparse.c: (uint64_ceiling_scale_int), (gst_wavparse_perform_seek), (gst_wavparse_stream_headers), (gst_wavparse_stream_data): * gst/wavparse/gstwavparse.h: More sanity check for the header fields. Fix type for 'rate' header field.
* gst/icydemux/gsticydemux.c: If the metadata strings we get in the stream are ↵Tim-Philipp Müller2007-04-121-0/+10
| | | | | | | | | | | | | 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 Taymans2007-04-121-0/+5
| | | | | | 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 Stichele2007-04-121-0/+20
| | | | | | | | | | | | | | | | | | | | | | | 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 tan2007-04-121-0/+9
| | | | | | | | | | | 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/rtsp/gstrtpdec.*: Make backward compat with rtpbin by adding the ↵Wim Taymans2007-04-121-0/+16
| | | | | | | | | | | | | | | | | | 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 Taymans2007-04-111-0/+6
| | | | | | | | | 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.
* gst/rtp/gstrtpamrdepay.c: Fix depayloader clock_rate and some cleanups.Wim Taymans2007-04-101-0/+20
| | | | | | | | | | | | | | | | | | 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 Kost2007-04-101-0/+10
| | | | | | | | | | 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 Kjellerstedt2007-04-101-0/+14
| | | | | | | | | | | | | | 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 Taymans2007-04-061-0/+35
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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/rtp/gstrtpmp4adepay.c: This element is ready to be autoplugged.Wim Taymans2007-04-051-0/+6
| | | | | | | 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/+7
| | | | | | | | | | | | 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 Kost2007-04-041-0/+13
| | | | | | | | | | | | | | 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 Taymans2007-04-031-0/+5
| | | | | | 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/+7
| | | | | | | | | | 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 Dröge2007-03-301-0/+9
| | | | | | | | | | | | 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 Stadler2007-03-291-0/+14
| | | | | | | | | | | | | | | 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 Taymans2007-03-291-0/+15
| | | | | | | | | | | | | | | | 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 Taymans2007-03-291-0/+19
| | | | | | | | | | | | | | | | | | | | 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 Dröge2007-03-291-0/+9
| | | | | | | | | | 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 Glayal2007-03-291-0/+20
| | | | | | | | | | | | | | | | | | | | | | 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 Glayal2007-03-291-0/+19
| | | | | | | | | | | | | | | | | | 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 Dröge2007-03-281-0/+5
| | | | | | | | 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 Dröge2007-03-281-0/+6
| | | | | | | | | 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 Kost2007-03-281-0/+24
| | | | | | | | | | | | | | | | | | | | | | 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/rtsp/gstrtspsrc.*: Handle default clock-rates for static payload types, ↵Wim Taymans2007-03-251-0/+17
| | | | | | | | | | | | | | | | | | | | 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.
* ext/gconf/gconf.c: Accept complex pipeline descriptions as an audio profile ↵Christophe Dehais2007-03-221-0/+8
| | | | | | | | | | 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.
* gst/apetag/gsttagdemux.c: Rename registered type in preparation of ↵Tim-Philipp Müller2007-03-211-0/+6
| | | | | | | | | 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 Müller2007-03-191-0/+6
| | | | | | | | | 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 Schleef2007-03-181-0/+6
| | | | | | | | | 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 Schleef2007-03-181-0/+5
| | | | | | Original commit message from CVS: * REQUIREMENTS: Fix a few things. This file really needs a good once-over.
* sys/Makefile.am: Don't forget to distribute the sys/osxaudio/ directory.Edward Hervey2007-03-151-0/+5
| | | | | | 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 Hervey2007-03-151-0/+24
| | | | | | | | | | | | | | | | | | | | | | | | | 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 Kalaf2007-03-141-0/+7
| | | | | | | | | 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
* po/: Update translations.Thomas Vander Stichele2007-03-141-0/+16
| | | | | | | | | | | | | | | | | 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.
* configure.ac: Fix string replace error (AG_AG_GST_* => AG_GST_*).Tim-Philipp Müller2007-03-141-0/+5
| | | | | | Original commit message from CVS: * configure.ac: Fix string replace error (AG_AG_GST_* => AG_GST_*).
* gst/apetag/gsttagdemux.c: Fix handling of -1 values for start and stop ↵Tim-Philipp Müller2007-03-121-0/+6
| | | | | | | | | 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 Schmidt2007-03-121-0/+6
| | | | | | | | | 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.
* I'm too lazy to comment thisJan Schmidt2007-03-121-0/+2
| | | | | | Original commit message from CVS: Add Patch by: line for wim, since he's away
* gst/id3demux/id3v2frames.c: Fix parsing of ID3 v2.2.0 PIC frames. Only in ↵Tim-Philipp Müller2007-03-121-0/+9
| | | | | | | | | | | | 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'.
* win32/MANIFEST: Add new project files to MANIFEST.Sébastien Moutte2007-03-101-0/+9
| | | | | | | | | | 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.
* Printf format fixes; also add some missing quotes in translated strings. ↵Tim-Philipp Müller2007-03-101-0/+9
| | | | | | | | | | | | 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 Schmidt2007-03-091-0/+9
| | | | | | | | | | | | 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.
* po/: Add ja.po file from #377306.Tim-Philipp Müller2007-03-091-0/+6
| | | | | | | Original commit message from CVS: * po/LINGUAS: * po/ja.po: Add ja.po file from #377306.
* sys/sunaudio/: Actually translate sunaudio mixer track labels instead of ↵Tim-Philipp Müller2007-03-091-0/+18
| | | | | | | | | | | | | | | | | | | | 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).
* tests/check/Makefile.am: Re-enable the states test and see if it works on ↵Jan Schmidt2007-03-091-0/+5
| | | | | | | | the buildbots. Original commit message from CVS: * tests/check/Makefile.am: Re-enable the states test and see if it works on the buildbots.
* ext/dv/gstdvdec.*: Infer pixel-aspect-ratio from the video frame format if ↵Wim Taymans2007-03-091-0/+13
| | | | | | | | | | | | | | | 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
* gst/rtsp/gstrtspsrc.c: When activated, remove the udpsrc timeout, we have ↵Wim Taymans2007-03-091-0/+6
| | | | | | | | | 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.
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