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2008-05-14gst/audioresample/gstaudioresample.c: Revert previous change which made ↵Tim-Philipp Müller1-0/+4
basetransform handle buffer_alloc and which b... Original commit message from CVS: * gst/audioresample/gstaudioresample.c: (gst_audioresample_init): Revert previous change which made basetransform handle buffer_alloc and which breaks things badly in the non-passthrough case since it returned buffers with a different (ie. sometimes smaller) size than the size requested.
2008-05-08gst/audioresample/gstaudioresample.c: Let audioresample use the buffer ↵Sjoerd Simons1-4/+0
allocation of basetransform instead of it's ow... Original commit message from CVS: Patch by: Sjoerd Simons <sjoerd at luon dot net> * gst/audioresample/gstaudioresample.c: (gst_audioresample_init): Let audioresample use the buffer allocation of basetransform instead of it's own stuff. * tests/check/elements/audioresample.c: (alloc_only_48000), (GST_START_TEST), (audioresample_suite): Add unit test for the recent basetransform bugfix, where upstream changes caps to something that can't be passed through anymore.
2008-03-22Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static ↵Sebastian Dröge1-3/+3
strings (i.e. all). This gives us less memory u... Original commit message from CVS: * configure.ac: * ext/alsa/gstalsamixerelement.c: (gst_alsa_mixer_element_class_init): * ext/alsa/gstalsasink.c: (gst_alsasink_class_init): * ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init): * ext/cdparanoia/gstcdparanoiasrc.c: (gst_cd_paranoia_src_class_init): * ext/gio/gstgiosink.c: (gst_gio_sink_class_init): * ext/gio/gstgiosrc.c: (gst_gio_src_class_init): * ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_class_init): * ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_class_init): * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init): * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init): * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init): * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init): * ext/pango/gsttextrender.c: (gst_text_render_class_init): * ext/theora/theoradec.c: (gst_theora_dec_class_init): * ext/theora/theoraenc.c: (gst_theora_enc_class_init): * ext/theora/theoraparse.c: (gst_theora_parse_class_init): * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_class_init): * gst-libs/gst/audio/gstaudiofiltertemplate.c: (gst_audio_filter_template_class_init): * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_class_init): * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_class_init): * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_class_init): * gst-libs/gst/interfaces/mixertrack.c: (gst_mixer_track_class_init): * gst-libs/gst/rtp/gstbasertpdepayload.c: (gst_base_rtp_depayload_class_init): * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_class_init): * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_class_init): * gst/audiorate/gstaudiorate.c: (gst_audio_rate_class_init): * gst/audioresample/gstaudioresample.c: (gst_audioresample_class_init): * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_class_init): * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init): * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init): * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init), (preroll_unlinked): * gst/playback/gstplaybin.c: (gst_play_bin_class_init): * gst/playback/gstplaybin2.c: (gst_play_bin_class_init): * gst/playback/gstplaysink.c: (gst_play_sink_class_init): * gst/playback/gstqueue2.c: (gst_queue_class_init): * gst/playback/gststreaminfo.c: (gst_stream_info_class_init): * gst/playback/gststreamselector.c: (gst_selector_pad_class_init), (gst_stream_selector_class_init): * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init): * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init): * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init): * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init): * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init): * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init): * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init): * gst/videorate/gstvideorate.c: (gst_video_rate_class_init): * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init): * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_class_init): * gst/volume/gstvolume.c: (gst_volume_class_init): * sys/v4l/gstv4lelement.c: (gst_v4lelement_class_init): * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init): * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init): * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init): * sys/ximage/ximagesink.c: (gst_ximagesink_class_init): * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init): Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory usage, fewer allocations and thus less memory defragmentation. Depend on core CVS for this. Fixes bug #523806.
2008-03-03Correct all relevant warnings found by the sparse semantic code analyzer. ↵BRANCH-RELEASE-0_10_19Sebastian Dröge1-1/+1
This include marking several symbols static... Original commit message from CVS: * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_get_type): * ext/alsa/gstalsasink.c: (set_hwparams): * ext/alsa/gstalsasrc.c: (set_hwparams): * ext/gio/gstgio.c: (gst_gio_uri_handler_get_uri): * ext/ogg/gstoggmux.h: * ext/ogg/gstogmparse.c: * gst-libs/gst/audio/audio.c: * gst-libs/gst/fft/kiss_fft_f64.c: (kiss_fft_f64_alloc): * gst-libs/gst/pbutils/missing-plugins.c: (gst_missing_uri_sink_message_new), (gst_missing_element_message_new), (gst_missing_decoder_message_new), (gst_missing_encoder_message_new): * gst-libs/gst/rtp/gstbasertppayload.c: * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_bye_get_reason): * gst/audioconvert/gstaudioconvert.c: * gst/audioresample/gstaudioresample.c: * gst/ffmpegcolorspace/imgconvert.c: * gst/playback/test.c: (gen_video_element), (gen_audio_element): * gst/typefind/gsttypefindfunctions.c: * gst/videoscale/vs_4tap.c: * gst/videoscale/vs_4tap.h: * sys/v4l/gstv4lelement.c: * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_get_any_caps): * sys/v4l/v4l_calls.c: * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_capture_init), (gst_v4lsrc_try_capture): * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls), (gst_ximagesink_ximage_new): * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new): * tests/check/elements/audioconvert.c: * tests/check/elements/audioresample.c: (fail_unless_perfect_stream): * tests/check/elements/audiotestsrc.c: (setup_audiotestsrc): * tests/check/elements/decodebin.c: * tests/check/elements/gdpdepay.c: (setup_gdpdepay), (setup_gdpdepay_streamheader): * tests/check/elements/gdppay.c: (setup_gdppay), (GST_START_TEST), (setup_gdppay_streamheader): * tests/check/elements/gnomevfssink.c: (setup_gnomevfssink): * tests/check/elements/multifdsink.c: (setup_multifdsink): * tests/check/elements/textoverlay.c: * tests/check/elements/videorate.c: (setup_videorate): * tests/check/elements/videotestsrc.c: (setup_videotestsrc): * tests/check/elements/volume.c: (setup_volume): * tests/check/elements/vorbisdec.c: (setup_vorbisdec): * tests/check/elements/vorbistag.c: * tests/check/generic/clock-selection.c: * tests/check/generic/states.c: (setup), (teardown): * tests/check/libs/cddabasesrc.c: * tests/check/libs/video.c: * tests/check/pipelines/gio.c: * tests/check/pipelines/oggmux.c: * tests/check/pipelines/simple-launch-lines.c: (simple_launch_lines_suite): * tests/check/pipelines/streamheader.c: * tests/check/pipelines/theoraenc.c: * tests/check/pipelines/vorbisdec.c: * tests/check/pipelines/vorbisenc.c: * tests/examples/seek/scrubby.c: * tests/examples/seek/seek.c: (query_positions_elems), (query_positions_pads): * tests/icles/stress-xoverlay.c: (myclock): Correct all relevant warnings found by the sparse semantic code analyzer. This include marking several symbols static, using NULL instead of 0 for pointers and using "foo (void)" instead of "foo ()" for declarations. * win32/common/libgstrtp.def: Add gst_rtp_buffer_set_extension_data to the symbol definition file.
2007-11-23gst/audioresample/gstaudioresample.c: Implement latency query.Sebastian Dröge1-0/+78
Original commit message from CVS: * gst/audioresample/gstaudioresample.c: (gst_audioresample_init), (audioresample_query), (audioresample_query_type), (gst_audioresample_set_property): Implement latency query.
2007-04-27ext/theora/theoradec.c: Calculate buffer duration correctly to generate a ↵Julien Moutte1-3/+1
perfect stream (#433888). Original commit message from CVS: 2007-04-27 Julien MOUTTE <julien@moutte.net> * ext/theora/theoradec.c: (_theora_granule_time), (theora_dec_push_forward), (theora_handle_data_packet), (theora_dec_decode_buffer): Calculate buffer duration correctly to generate a perfect stream (#433888). * gst/audioresample/gstaudioresample.c: (audioresample_check_discont): Glib provides ABS.
2007-04-21gst/audioresample/gstaudioresample.c: Make more functions static, just ↵Tim-Philipp Müller1-9/+9
because we can. Original commit message from CVS: * gst/audioresample/gstaudioresample.c: Make more functions static, just because we can.
2007-04-16ext/pango/gstclockoverlay.c: Fix unused variable warning if HAVE_LOCALTIME_R ↵Vincent Torri1-1/+1
is undefinied Original commit message from CVS: Patch by: Vincent Torri <vtorri at univ-evry dot fr> * ext/pango/gstclockoverlay.c: (gst_clock_overlay_render_time): Fix unused variable warning if HAVE_LOCALTIME_R is undefinied * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps): * gst/audioresample/gstaudioresample.c: (audioresample_do_output): Use the correct format strings for integer formats.
2007-03-15gst/audioresample/gstaudioresample.c: Don't trigger discontinuities for very ↵Michael Smith1-16/+34
small imperfections; a filter flush will... Original commit message from CVS: * gst/audioresample/gstaudioresample.c: (audioresample_check_discont), (audioresample_transform): Don't trigger discontinuities for very small imperfections; a filter flush will sound bad, and many plugins have rounding errors leading to these.
2007-03-14gst/audioresample/gstaudioresample.c: Handle discontinuous streams.Julien Moutte2-9/+48
Original commit message from CVS: 2007-03-14 Julien MOUTTE <julien@moutte.net> * gst/audioresample/gstaudioresample.c: (gst_audioresample_init), (audioresample_transform_size), (audioresample_do_output), (audioresample_transform), (audioresample_pushthrough): Handle discontinuous streams. * gst/audioresample/gstaudioresample.h: * tests/check/elements/audioresample.c: (test_discont_stream_instance), (GST_START_TEST), (audioresample_suite): Add a test for discontinuous streams. * win32/common/config.h: Updated.
2007-03-14gst/audioresample/: Since I really am not interested in a debug line for ↵Thomas Vander Stichele2-2/+6
each sample being processed, move the librar... Original commit message from CVS: * gst/audioresample/debug.h: * gst/audioresample/resample.c: (resample_init): Since I really am not interested in a debug line for each sample being processed, move the library's debugging to its own category, libaudioresample
2007-03-14add debugging and reformat docsThomas Vander Stichele1-8/+21
Original commit message from CVS: add debugging and reformat docs
2007-01-04configure.ac: split out GST_CFLAGS into GST_PLUGINS_BASE_CFLAGS and ↵Thomas Vander Stichele1-1/+1
GST_CFLAGS so that GST_BASE_CFLAGS can go inbetwe... Original commit message from CVS: * configure.ac: split out GST_CFLAGS into GST_PLUGINS_BASE_CFLAGS and GST_CFLAGS so that GST_BASE_CFLAGS can go inbetween them, making sure we use uninstalled gst-libs headers * docs/libs/Makefile.am: * ext/alsa/Makefile.am: * ext/cdparanoia/Makefile.am: * ext/gnomevfs/Makefile.am: * ext/libvisual/Makefile.am: * ext/ogg/Makefile.am: * ext/theora/Makefile.am: * ext/vorbis/Makefile.am: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/cdda/Makefile.am: * gst-libs/gst/interfaces/Makefile.am: * gst-libs/gst/riff/Makefile.am: * gst-libs/gst/rtp/Makefile.am: * gst-libs/gst/tag/Makefile.am: * gst/adder/Makefile.am: * gst/audioconvert/Makefile.am: * gst/audiorate/Makefile.am: * gst/audioresample/Makefile.am: * gst/playback/Makefile.am: * gst/tcp/Makefile.am: * gst/videoscale/Makefile.am: * gst/volume/Makefile.am: * sys/ximage/Makefile.am: * sys/xvimage/Makefile.am: * tests/icles/Makefile.am: adapt
2006-10-28gst/audioresample/gstaudioresample.c: Another typo fix (#366212).Tim-Philipp Müller1-1/+1
Original commit message from CVS: * gst/audioresample/gstaudioresample.c: (gst_audioresample_init): Another typo fix (#366212).
2006-10-10ext/gnomevfs/: Fix URI interface implementation return type.Josep Torre Valles1-1/+1
Original commit message from CVS: 2006-10-10 Zaheer Abbas Merali <zaheerabbas at merali dot org> Patch by: Josep Torre Valles <josep@fluendo.com> * ext/gnomevfs/gstgnomevfssink.c: * ext/gnomevfs/gstgnomevfssrc.c: Fix URI interface implementation return type. * ext/pango/gsttextoverlay.c: (gst_text_overlay_set_property): Fix what looks like a copy/paste issue when assigning values. * gst-libs/gst/audio/gstaudiofiltertemplate.c: (gst_audio_filter_template_get_type): Cast to prevent Forte warnings. * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create): Fix URI interface implementation return type. gst_pad_query_position requires a signed integer pointer as 3rd parameter, GstClockTime is unsigned. * gst/audioconvert/audioconvert.c: Fix integer overflow when treated as signed. * gst/audioresample/resample.c: (resample_add_input_data): Cast to prevent warnings on Forte. * gst/ffmpegcolorspace/imgconvert.c: (build_rgb_palette): Fix integer overflow when treated as signed. * gst/ffmpegcolorspace/imgconvert_template.h: Fix integer overflow when treated as signed. RGBA_OUT shifts bits. * gst/playback/gstdecodebin.c: (queue_filled_cb), (cleanup_decodebin): Who initialises a guint to -1! Cast function pointers to prevent warnings on Forte. * gst/playback/gstplaybasebin.c: (queue_deadlock_check), (queue_threshold_reached): Cast function pointers correctly to prevent warnings on Forte. * gst/playback/gststreaminfo.c: (gst_stream_info_dispose): Cast function pointers correctly to prevent warnings on Forte. * gst/subparse/gstssaparse.c: (gst_ssa_parse_setcaps): Obvious change to unsigned, 0xEF > max signed char. * gst/tcp/gstmultifdsink.c: (get_buffers_max), (count_burst_unit): GstClockTime is unsigned, initialise correctly. * gst/tcp/gsttcp.c: (gst_tcp_socket_write): Cast so pointer arithemetic doesn't cause warnings on Forte. * gst/videorate/gstvideorate.c: Use correct return value. * tests/examples/seek/scrubby.c: GstClockTime is unsigned, initialise correctly.
2006-10-05Printf format fixes.Tim-Philipp Müller1-1/+1
Original commit message from CVS: * ext/alsa/gstalsadeviceprobe.c: (gst_alsa_device_property_probe_get_values): * ext/alsa/gstalsasink.c: (set_hwparams): * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_elem_pad), (gst_ogg_chain_new_stream), (gst_ogg_demux_read_chain): * ext/ogg/gstoggmux.c: (gst_ogg_mux_send_headers), (gst_ogg_mux_process_best_pad): * ext/ogg/gstoggparse.c: (gst_ogg_parse_new_stream), (gst_ogg_parse_chain): * ext/ogg/gstogmparse.c: (gst_ogm_parse_stream_header): * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet): * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_setup), (gst_vorbis_enc_buffer_check_discontinuous): * ext/vorbis/vorbisparse.c: (vorbis_parse_src_query): * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_render): * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_handle_track_seek): * gst-libs/gst/rtp/gstbasertpdepayload.c: (gst_base_rtp_depayload_push_full): * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push): * gst/audioresample/resample.c: (resample_input_pushthrough): * gst/playback/gstplaybasebin.c: (queue_out_of_data): * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_handle_clients): * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset), (wavpack_type_find): * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_create): * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy), (gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new): * tests/check/elements/volume.c: (GST_START_TEST): Printf format fixes.
2006-08-20gst/audioconvert/gstaudioconvert.c: Lower debug, use g_assert in _get_unit_sizeStefan Kost1-1/+1
Original commit message from CVS: * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_get_unit_size), (set_structure_widths): Lower debug, use g_assert in _get_unit_size * gst/audioresample/gstaudioresample.c: (audioresample_get_unit_size): * gst/ffmpegcolorspace/gstffmpegcolorspace.c: (gst_ffmpegcsp_get_unit_size): * gst/videoscale/gstvideoscale.c: (gst_video_scale_get_unit_size): use g_assert in _get_unit_size
2006-07-28gst/audioresample/gstaudioresample.c: Don't leak references to the incoming ↵Jan Schmidt1-3/+3
caps. Clean them up when stopping. Original commit message from CVS: * gst/audioresample/gstaudioresample.c: (audioresample_stop), (audioresample_set_caps): Don't leak references to the incoming caps. Clean them up when stopping. * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init), (gst_video_scale_finalize): Don't leak our temporary pixel buffer. * tests/check/Makefile.am: * tests/check/pipelines/simple-launch-lines.c: (run_pipeline), (GST_START_TEST), (simple_launch_lines_suite): Fix leaks and re-enable the test for valgrind checking.
2006-06-22gst/: Avoid unnecessary class cast check in class_init functions (#337747).Cody Russell1-1/+1
Original commit message from CVS: Patch by: Cody Russell <bratsche at gnome org> * gst/audioresample/gstaudioresample.c: (gst_audioresample_class_init): * gst/playback/gststreamselector.c: (gst_stream_selector_class_init): * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init): * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init): * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init): * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init): * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init): * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init): * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init): * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_class_init): * gst/volume/gstvolume.c: (gst_volume_class_init): Avoid unnecessary class cast check in class_init functions (#337747).
2006-06-16gst/audioresample/gstaudioresample.c: Implement GstBaseTransform::start and ↵Tim-Philipp Müller1-25/+28
::stop so that audioresample can clear it... Original commit message from CVS: * gst/audioresample/gstaudioresample.c: (gst_audioresample_class_init), (gst_audioresample_init), (audioresample_start), (audioresample_stop), (gst_audioresample_set_property), (gst_audioresample_get_property): Implement GstBaseTransform::start and ::stop so that audioresample can clear its internal state properly and be reused insted of causing non-negotiated errors with playbin under some circumstances (#342789). * tests/check/elements/audioresample.c: (setup_audioresample), (cleanup_audioresample): Need to set element state here so that ::start and ::stop are called.
2006-06-01Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClassStefan Kost1-2/+2
Original commit message from CVS: * ext/alsa/gstalsamixerelement.h: * ext/alsa/gstalsamixeroptions.h: * ext/alsa/gstalsamixertrack.h: * ext/gnomevfs/gstgnomevfssink.h: * ext/gnomevfs/gstgnomevfssrc.h: * ext/theora/gsttheoradec.h: * ext/theora/gsttheoraenc.h: * ext/theora/gsttheoraparse.h: * ext/vorbis/vorbisparse.h: * gst-libs/gst/audio/gstaudioclock.h: * gst-libs/gst/audio/gstaudiofilter.h: * gst-libs/gst/rtp/gstbasertpaudiopayload.h: * gst/audioconvert/gstaudioconvert.h: * gst/audioresample/gstaudioresample.h: * gst/audiotestsrc/gstaudiotestsrc.h: * gst/ffmpegcolorspace/gstffmpegcolorspace.h: * gst/playback/gststreamselector.h: * gst/tcp/gstmultifdsink.h: * gst/tcp/gsttcpclientsink.h: * gst/tcp/gsttcpclientsrc.h: * gst/tcp/gsttcpserversink.h: * gst/tcp/gsttcpserversrc.h: * gst/videorate/gstvideorate.h: * gst/videoscale/gstvideoscale.h: * gst/videotestsrc/gstvideotestsrc.h: * gst/volume/gstvolume.h: * sys/v4l/gstv4ljpegsrc.h: * sys/v4l/gstv4lmjpegsink.h: * sys/v4l/gstv4lmjpegsrc.h: * sys/v4l/gstv4lsrc.h: * sys/ximage/ximagesink.h: * sys/xvimage/xvimagesink.h: * tests/old/testsuite/alsa/sinesrc.h: Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
2006-04-28make GstElementDetails constStefan Kost1-1/+1
Original commit message from CVS: * ext/alsa/gstalsamixerelement.c: * ext/alsa/gstalsasrc.c: * ext/cdparanoia/gstcdparanoiasrc.c: * ext/gnomevfs/gstgnomevfssink.c: * ext/gnomevfs/gstgnomevfssrc.c: * ext/ogg/gstoggdemux.c: * ext/ogg/gstoggmux.c: * ext/ogg/gstoggparse.c: * ext/ogg/gstogmparse.c: * ext/pango/gstclockoverlay.c: * ext/pango/gsttextoverlay.c: * ext/pango/gsttextrender.c: * ext/pango/gsttimeoverlay.c: * ext/theora/theoradec.c: * ext/theora/theoraenc.c: * ext/vorbis/vorbisdec.c: * ext/vorbis/vorbisenc.c: * gst-libs/gst/audio/gstaudiofilter.c: * gst-libs/gst/audio/gstaudiofiltertemplate.c: * gst/audioconvert/gstaudioconvert.c: * gst/audiorate/gstaudiorate.c: * gst/audioresample/gstaudioresample.c: * gst/audiotestsrc/gstaudiotestsrc.c: * gst/ffmpegcolorspace/gstffmpegcolorspace.c: * gst/playback/gstdecodebin.c: * gst/playback/gstplaybin.c: * gst/playback/gststreamselector.c: * gst/subparse/gstsubparse.c: * gst/tcp/gstmultifdsink.c: * gst/tcp/gsttcpclientsink.c: * gst/tcp/gsttcpclientsrc.c: * gst/tcp/gsttcpserversink.c: * gst/tcp/gsttcpserversrc.c: * gst/typefind/gsttypefindfunctions.c: (plugin_init): * gst/videorate/gstvideorate.c: * gst/videoscale/gstvideoscale.c: * gst/videotestsrc/gstvideotestsrc.c: * gst/volume/gstvolume.c: * sys/v4l/gstv4ljpegsrc.c: * sys/v4l/gstv4lmjpegsink.c: * sys/v4l/gstv4lmjpegsrc.c: * sys/v4l/gstv4lsrc.c: * sys/ximage/ximagesink.c: * sys/xvimage/xvimagesink.c: * tests/check/libs/cddabasesrc.c: make GstElementDetails const
2006-04-28gst/audioresample/gstaudioresample.c: Add support for other formats ↵Wim Taymans1-17/+75
audioresample can handle such as 32 bits in and f... Original commit message from CVS: * gst/audioresample/gstaudioresample.c: (gst_audioresample_init), (resample_set_state_from_caps): Add support for other formats audioresample can handle such as 32 bits in and float and 64 bits float. Fixes #301759
2006-04-09gst/audioresample/debug.h: replace debug macros with variable number of ↵Sébastien Moutte2-5/+13
parameters by a simple alias to gstreamer sta... Original commit message from CVS: * gst/audioresample/debug.h: replace debug macros with variable number of parameters by a simple alias to gstreamer standard debug macros (#define RESAMPLE_ERROR GST_ERROR, __VA_ARGS__ is not supported by MSVC 6.0 and 7.1) * gst/audioresample/resample.h: define M_PI and rint for WIN32 * win32/common/libgstaudio.def: * win32/common/libgstriff.def: * win32/common/libgsttag.def: * win32/common/libgstvideo.def: add new exported functions * win32/vs6: update project files
2006-03-02docs/plugins/: Add audioresample to docs.Wim Taymans2-69/+95
Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-base-plugins-docs.sgml: * docs/plugins/gst-plugins-base-plugins-sections.txt: Add audioresample to docs. * gst/audioconvert/gstaudioconvert.c: Add revision date. * gst/audioresample/gstaudioresample.c: (gst_audioresample_base_init), (gst_audioresample_class_init), (gst_audioresample_init), (gst_audioresample_dispose), (audioresample_get_unit_size), (audioresample_transform_caps), (resample_set_state_from_caps), (audioresample_transform_size), (audioresample_set_caps), (audioresample_event), (audioresample_do_output), (audioresample_transform), (audioresample_pushthrough), (gst_audioresample_set_property), (gst_audioresample_get_property), (plugin_init): * gst/audioresample/gstaudioresample.h: Added docs. Small code cleanups.
2006-01-03gst/audioresample/resample.h: As before, but for o_bufRELEASE-0_10_2Michael Smith1-1/+1
Original commit message from CVS: * gst/audioresample/resample.h: As before, but for o_buf
2006-01-03gst/audioresample/resample.h: Declare struct _ResampleState.buffer as ↵Michael Smith1-1/+1
unsigned char *, not void *, since we do arithm... Original commit message from CVS: * gst/audioresample/resample.h: Declare struct _ResampleState.buffer as unsigned char *, not void *, since we do arithmetic on it.
2005-12-15gst/audioresample/gstaudioresample.c: Don't leak all input buffers to ↵Michael Smith1-2/+3
audioresample. Original commit message from CVS: * gst/audioresample/gstaudioresample.c: Don't leak all input buffers to audioresample.
2005-12-06expand tabsThomas Vander Stichele4-46/+46
Original commit message from CVS: expand tabs
2005-12-02gst/audioresample/: Fix audioresample, seek torture, new segments, reverse ↵Wim Taymans7-76/+293
negotiation etc.. work fine. Original commit message from CVS: * gst/audioresample/buffer.c: (audioresample_buffer_queue_flush): * gst/audioresample/buffer.h: * gst/audioresample/gstaudioresample.c: * gst/audioresample/gstaudioresample.h: * gst/audioresample/resample.c: (resample_input_flush), (resample_input_pushthrough), (resample_input_eos), (resample_get_output_size_for_input), (resample_get_input_size_for_output), (resample_get_output_size), (resample_get_output_data): * gst/audioresample/resample.h: * gst/audioresample/resample_ref.c: (resample_scale_ref): Fix audioresample, seek torture, new segments, reverse negotiation etc.. work fine.
2005-11-21gst/: Segment update fix.Wim Taymans1-2/+2
Original commit message from CVS: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_class_init), (gst_base_audio_sink_init), (gst_base_audio_sink_provide_clock), (gst_base_audio_sink_set_clock), (gst_base_audio_sink_render), (gst_base_audio_sink_change_state): * gst/audioresample/gstaudioresample.c: Segment update fix.
2005-10-16restructure configure.ac, use correct libtool LDFLAGS, fix up definesThomas Vander Stichele1-1/+2
Original commit message from CVS: restructure configure.ac, use correct libtool LDFLAGS, fix up defines
2005-09-23gst/audioresample/: Convert to using gst debuggingDavid Schleef4-8/+18
Original commit message from CVS: * gst/audioresample/Makefile.am: * gst/audioresample/debug.h: * gst/audioresample/gstaudioresample.c: * gst/audioresample/resample.c: Convert to using gst debugging
2005-09-09check/: Add extra tests for basetransform based components.Jan Schmidt1-0/+2
Original commit message from CVS: * check/Makefile.am: * check/pipelines/simple_launch_lines.c: (setup_pipeline), (run_pipeline), (GST_START_TEST), (simple_launch_lines_suite): Add extra tests for basetransform based components. Comment out the test_element_negotiation test until we decide if it's testing correct behaviour. * ext/libvisual/visual.c: (gst_visual_init), (get_buffer), (gst_visual_chain), (gst_visual_change_state): Slightly more correct but still bogus timestamping. Fix state change function. * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_class_init): * gst/audioresample/gstaudioresample.c: * gst/ffmpegcolorspace/gstffmpegcolorspace.c: (gst_ffmpegcsp_class_init): * gst/videoscale/gstvideoscale.c: (gst_videoscale_class_init), (gst_videoscale_prepare_size), (gst_videoscale_set_caps), (gst_videoscale_prepare_image): * gst/volume/gstvolume.c: (gst_volume_class_init), (volume_transform_ip): Basetransform updates. Enable passthrough modes. * sys/ximage/ximagesink.c: (gst_ximage_buffer_init), (gst_ximagesink_renegotiate_size), (gst_ximagesink_xcontext_get), (gst_ximagesink_setcaps), (gst_ximagesink_buffer_alloc): Negotiation fix that allows the window to return to the original size and renegotiate passthrough upstream. Extra debug output.
2005-09-04fix distcheckRELEASE-0_9_2Thomas Vander Stichele1-1/+1
Original commit message from CVS: * common/gtk-doc-plugins.mak: * docs/plugins/Makefile.am: fix distcheck * gst/audioresample/resample.c: fix wrong docstring
2005-08-28Updates for two-arg init from GST_BOILERPLATE_FULL.Andy Wingo1-4/+2
Original commit message from CVS: 2005-08-28 Andy Wingo <wingo@pobox.com> * Updates for two-arg init from GST_BOILERPLATE_FULL.
2005-08-26use base class' newsegment to properly timestampThomas Vander Stichele1-2/+5
Original commit message from CVS: use base class' newsegment to properly timestamp
2005-08-25check/: add a test for audioconvertThomas Vander Stichele2-4/+10
Original commit message from CVS: * check/Makefile.am: * check/elements/audioconvert.c: (setup_audioconvert), (cleanup_audioconvert), (get_int_caps), (verify_convert), (GST_START_TEST), (audioconvert_suite), (main): add a test for audioconvert * gst/audioresample/gstaudioresample.c: * gst/audioresample/gstaudioresample.h: set DURATION so that TIMESTAMP(a) + DURATION(a) == TIMESTAMP(b); note that for buffers of 1/3 sec this means DURATION(c) is one nanosecond more than for a and b
2005-08-25add a check for audioresampleThomas Vander Stichele2-2/+13
Original commit message from CVS: add a check for audioresample
2005-08-25show some info on what's left in the queueThomas Vander Stichele1-2/+6
Original commit message from CVS: show some info on what's left in the queue
2005-08-25gst/audioresample/: add room for extra overlap samples when asked to ↵Thomas Vander Stichele6-35/+73
transform size protect against possible mem corr... Original commit message from CVS: * gst/audioresample/debug.c: * gst/audioresample/gstaudioresample.c: add room for extra overlap samples when asked to transform size protect against possible mem corruption and check for discrepancies between written size and outbuffer's size so we can warn for potential problems * gst/audioresample/resample.c: (resample_init), (resample_get_output_size_for_input), (resample_get_output_size), (resample_set_n_channels), (resample_set_format): set debug level based on RESAMPLE_DEBUG env var make sure that get_output_size* returns a whole number of sample_size set sample_size each time either channel or format is set * gst/audioresample/resample_chunk.c: (resample_scale_chunk): * gst/audioresample/resample_functable.c: (resample_scale_functable): * gst/audioresample/resample_ref.c: (resample_scale_ref): remove r->sample_size, it's done in resample.c now add some debugging to the ref implementation make sure we only give back bytes that are wholes of the sample size
2005-08-24fix broken header setup in Makefile.amChristian Schaller1-1/+1
Original commit message from CVS: fix broken header setup in Makefile.am
2005-08-24dist moreThomas Vander Stichele2-5/+6
Original commit message from CVS: dist more
2005-08-24port audioresample to basetransformThomas Vander Stichele10-236/+271
Original commit message from CVS: port audioresample to basetransform
2005-08-23gst/audioresample/Makefile.am: Leet audioresampling codeDavid Schleef14-0/+2254
Original commit message from CVS: * gst/audioresample/Makefile.am: Leet audioresampling code * gst/audioresample/buffer.c: * gst/audioresample/buffer.h: * gst/audioresample/debug.c: * gst/audioresample/debug.h: * gst/audioresample/functable.c: * gst/audioresample/functable.h: * gst/audioresample/gstaudioresample.c: * gst/audioresample/gstaudioresample.h: * gst/audioresample/resample.c: * gst/audioresample/resample.h: * gst/audioresample/resample_chunk.c: * gst/audioresample/resample_functable.c: * gst/audioresample/resample_ref.c: