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* gst/avi/gstavimux.c: Allow muxing video/x-h264 (was already in the caps). ↵zhangfei gao2007-02-142-2/+11
| | | | | | | | | Fixes #407780. Original commit message from CVS: Patch by: zhangfei gao <gaozhangfei@yahoo.com.cn> * gst/avi/gstavimux.c: (gst_avi_mux_vidsink_set_caps): Allow muxing video/x-h264 (was already in the caps). Fixes #407780.
* gst/rtsp/rtspurl.c: Fix parsing of password field in url. Fixes #407797.jp.liu2007-02-142-1/+8
| | | | | | | Original commit message from CVS: Patch by: jp.liu <jp_liu at astrocom dot cn> * gst/rtsp/rtspurl.c: (rtsp_url_parse): Fix parsing of password field in url. Fixes #407797.
* gst/wavparse/gstwavparse.*: Update docs.Wim Taymans2007-02-143-240/+281
| | | | | | | | | | | | | | | | | | | | | | | | Original commit message from CVS: * gst/wavparse/gstwavparse.c: (gst_wavparse_class_init), (gst_wavparse_reset), (gst_wavparse_init), (gst_wavparse_destroy_sourcepad), (gst_wavparse_fmt), (gst_wavparse_parse_file_header), (gst_wavparse_stream_init), (gst_wavparse_perform_seek), (gst_wavparse_peek_chunk_info), (gst_wavparse_stream_headers), (gst_wavparse_parse_stream_init), (gst_wavparse_add_src_pad), (gst_wavparse_stream_data), (gst_wavparse_loop), (gst_wavparse_chain), (gst_wavparse_pad_convert), (gst_wavparse_pad_query), (gst_wavparse_srcpad_event), (gst_wavparse_change_state), (plugin_init): * gst/wavparse/gstwavparse.h: Update docs. Use boilerplate. Various code cleanups. When the bitrate is not known (bps == 0 or compressed formats) let downstream element guestimate the duration and position and don't generate timestamps or durations. Fixes #405213. Fix EOS and ERROR conditions in chain mode, we just need to forward the error flowreturn upstream.
* Re-factor the gconfaudiosink into a "GstSwitchSink" base class and a child ↵Jan Schmidt2007-02-1310-156/+533
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | that implements the GConf key monitoring. ... Original commit message from CVS: * ext/gconf/Makefile.am: * ext/gconf/gconf.c: (gst_gconf_get_string), (gst_gconf_get_key_for_sink_profile), (gst_gconf_set_string), (gst_gconf_render_bin_with_default): * ext/gconf/gconf.h: * ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_base_init), (gst_gconf_audio_sink_reset), (gst_gconf_audio_sink_init), (gst_gconf_audio_sink_dispose), (do_change_child), (gst_gconf_switch_profile), (gst_gconf_audio_sink_set_property), (cb_change_child), (gst_gconf_audio_sink_change_state): * ext/gconf/gstgconfaudiosink.h: * ext/gconf/gstswitchsink.c: (gst_switch_sink_base_init), (gst_switch_sink_class_init), (gst_switch_sink_reset), (gst_switch_sink_init), (gst_switch_sink_dispose), (gst_switch_commit_new_kid), (gst_switch_sink_set_child), (gst_switch_sink_set_property), (gst_switch_sink_handle_event), (gst_switch_sink_get_property), (gst_switch_sink_change_state): * ext/gconf/gstswitchsink.h: * gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_class_init), (gst_auto_audio_sink_dispose), (gst_auto_audio_sink_clear_kid), (gst_auto_audio_sink_reset), (gst_auto_audio_sink_detect): * gst/autodetect/gstautovideosink.c: (gst_auto_video_sink_class_init), (gst_auto_video_sink_dispose), (gst_auto_video_sink_clear_kid), (gst_auto_video_sink_reset), (gst_auto_video_sink_detect): Re-factor the gconfaudiosink into a "GstSwitchSink" base class and a child that implements the GConf key monitoring. The end goal of this is an audio sink that can be changed on the fly, but at the moment it still only changes on the next READY transition.
* gst/avi/gstavidemux.c: Put debug stuff into #ifndef GST_DISABLE_DEBUG #endifStefan Kost2007-02-132-7/+26
| | | | | | | | | | | | Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query), (gst_avi_demux_parse_index), (gst_avi_demux_stream_index), (gst_avi_demux_sync), (gst_avi_demux_massage_index), (gst_avi_demux_calculate_durations_from_index), (gst_avi_demux_push_event), (gst_avi_demux_stream_header_pull), (gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data), (gst_avi_demux_loop): Put debug stuff into #ifndef GST_DISABLE_DEBUG #endif
* Add crossreferences to glib/gobject/gstream docs.Stefan Kost2007-02-133-3/+20
| | | | | | | Original commit message from CVS: * configure.ac: * docs/plugins/Makefile.am: Add crossreferences to glib/gobject/gstream docs.
* gst/monoscope/: Fix copy'n'paste-o in docs chunk. Also add some missing ↵Tim-Philipp Müller2007-02-123-3/+10
| | | | | | | | | | CFLAGS (but no LIBS, since we only use define... Original commit message from CVS: * gst/monoscope/Makefile.am: * gst/monoscope/gstmonoscope.c: Fix copy'n'paste-o in docs chunk. Also add some missing CFLAGS (but no LIBS, since we only use defines from the headers).
* gst/wavparse/gstwavparse.c: Fix massive memory leak when operating in ↵Jonathan Matthew2007-02-122-7/+13
| | | | | | | | | | | | streaming mode due to Original commit message from CVS: Based on patch by: Jonathan Matthew <jonathan at kaolin wh9 net> * gst/wavparse/gstwavparse.c: (gst_wavparse_parse_stream_init), (gst_wavparse_stream_data): Fix massive memory leak when operating in streaming mode due to GST_BUFFER_MALLOCDATA() not being set on newly-created buffers. Fixes #407057.
* gst/avi/gstavidemux.*: Save some memory (8%) by repacking the index entry ↵Stefan Kost2007-02-123-202/+319
| | | | | | | | | | | | | | | | | | | | structure (more to come). Add more FIXMEs t... Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_class_init), (gst_avi_demux_reset), (gst_avi_demux_index_entry_for_time), (gst_avi_demux_handle_src_query), (gst_avi_demux_parse_superindex), (gst_avi_demux_parse_subindex), (gst_avi_demux_parse_stream), (gst_avi_demux_parse_index), (gst_avi_demux_stream_index), (gst_avi_demux_sync), (gst_avi_demux_next_data_buffer), (gst_avi_demux_stream_scan), (gst_avi_demux_massage_index), (gst_avi_demux_calculate_durations_from_index), (gst_avi_demux_push_event), (gst_avi_demux_stream_header_pull), (gst_avi_demux_do_seek), (gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data), (gst_avi_demux_loop): * gst/avi/gstavidemux.h: Save some memory (8%) by repacking the index entry structure (more to come). Add more FIXMEs to questionable parts.
* sys/v4l2/: More FIXME comments and messaging changes.Stefan Kost2007-02-122-0/+19
| | | | | | | | | | Original commit message from CVS: * sys/v4l2/gstv4l2src.c: (gst_v4l2src_v4l2fourcc_to_caps), (gst_v4l2src_get_caps): * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list), (gst_v4l2src_grab_frame), (gst_v4l2src_set_capture), (gst_v4l2src_capture_init): More FIXME comments and messaging changes.
* gst/goom/gstgoom.*: Improved docs and use GST_DEBUG_FUNCPTR.Stefan Kost2007-02-125-18/+46
| | | | | | | | | | | | | Original commit message from CVS: * gst/goom/gstgoom.c: (gst_goom_class_init), (gst_goom_init), (gst_goom_change_state): * gst/goom/gstgoom.h: Improved docs and use GST_DEBUG_FUNCPTR. * gst/level/gstlevel.c: (gst_level_class_init): Use GST_DEBUG_FUNCPTR. * gst/monoscope/gstmonoscope.c: (gst_monoscope_init), (gst_monoscope_chain), (gst_monoscope_change_state): Improved docs source cleanups.
* gst/debug/: Add code for a pushfilesrc element that implements a pushfile:// ↵Tim-Philipp Müller2007-02-125-1/+285
| | | | | | | | | | | | | | URI handler, to make debugging push-mode... Original commit message from CVS: * gst/debug/Makefile.am: * gst/debug/gstdebug.c: (plugin_init): * gst/debug/gstpushfilesrc.c: * gst/debug/gstpushfilesrc.h: Add code for a pushfilesrc element that implements a pushfile:// URI handler, to make debugging push-mode operation of demuxer/decoders that support both easier in connection with seek/playbin/etc. The element isn't registered at the moment.
* Makefile.am: Add win32 MANIFESTSébastien Moutte2007-02-115-261/+409
| | | | | | | | | | | | | | | | | | | | | | | | | Original commit message from CVS: * Makefile.am: Add win32 MANIFEST * sys/directdraw/gstdirectdrawsink.c: * sys/directdraw/gstdirectdrawsink.h: Clear unused code and add comments. Remove yuv from template caps, it only supports RGB actually. Implement XOverlay interface and remove window and fullscreen properties. Add debug logs. Test for blit capabilities to return only the current colorspace if the hardware can't blit for one colorspace to another. * sys/directsound/gstdirectsoundsink.c: Add some debugs. * win32/MANIFEST: Add VS7 project files and solution. * win32/vs6/gst_plugins_bad.dsw: * win32/vs6/libgstdirectdraw.dsp: * win32/vs6/libgstdirectsound.dsp: * win32/vs6/libgstqtdemux.dsp: Update project files.
* gst/avi/gstavimux.c: Comment a #if 0 in caps template definition as VS6 ↵Sébastien Moutte2007-02-1119-21/+684
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | seems to do not support it. Original commit message from CVS: * gst/avi/gstavimux.c: Comment a #if 0 in caps template definition as VS6 seems to do not support it. * gst/rtsp/gstrtspsrc.c:(gst_rtspsrc_loop_udp): Use gst_guint64_to_gdouble for conversion. * gst/rtsp/rtspconnection.c:(rtsp_connection_send): Move variables declaration before the first instruction. * gst/rtsp/rtspdefs.c:(rtsp_strresult): Don't use hstrerror for error log on G_OS_WIN32 build as it's not supported. And don't include netdb.h for G_OS_WIN32 * gst/rtsp/sdpmessage.c:(sdp_parse_line): This initialization SDPMedia nmedia = {.media = NULL }; is not supported by VS6 then use an other way to initialize SDPMedia structure. * gst/udp/gstdynudpsink.h: * gst/udp/gstdynudpnetutils.h: Do not include <sys/time.h> for G_OS_WIN32 * gst/udp/gstudpsrc.c: Define socklen_t as int for G_OS_WIN32 * win/common/config.h.in: Undef HAVE_NETINET_IN_H * win32/vs6/gst_plugins_good.dsw: * win32/vs6/libgstrtp.dsp: * win32/vs6/libgstrtsp.dsp: * win32/vs6/libgstautogen.dsp: * win32/vs6/libgstaudiofx.dsp: * win32/vs6/libgstudp.dsp: Add and update project files. * win32/common/gstudp-enumtypes.c: * win32/common/gstudp-enumtypes.h: Add a copy of udp enumtypes to win32/common as in core and base.
* configure.ac: Activate monoscope when building with --enable-experimental. FixStefan Kost2007-02-115-15/+24
| | | | | | | | | | Original commit message from CVS: * configure.ac: Activate monoscope when building with --enable-experimental. Fix --enable-external configure switch description. * sys/sunaudio/gstsunaudiomixer.c: (gst_sunaudiomixer_base_init): * sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_dispose): Help gst-indent.
* gst/avi/gstavimux.c: Explicitly cast result of pointer arithmetic to integer ↵Tim-Philipp Müller2007-02-092-2/+8
| | | | | | | | | in order to avoid compiler warnings on s... Original commit message from CVS: * gst/avi/gstavimux.c: (gst_avi_mux_riff_get_avi_header): Explicitly cast result of pointer arithmetic to integer in order to avoid compiler warnings on some 64-bit systems. Should fix #406018.
* gst/debug/progressreport.c: Some more docs.Tim-Philipp Müller2007-02-082-0/+17
| | | | | | Original commit message from CVS: * gst/debug/progressreport.c: Some more docs.
* docs/plugins/inspect/plugin-rtp.xml: Update for new elements.Tim-Philipp Müller2007-02-073-0/+97
| | | | | | | | Original commit message from CVS: * docs/plugins/inspect/plugin-rtp.xml: Update for new elements. * gst/debug/progressreport.h: Commit newly-created header file as well.
* Make progressreport element post messages with the current progress on the ↵Tim-Philipp Müller2007-02-076-34/+112
| | | | | | | | | | | | | | | bus. Also add some basic docs for it. Original commit message from CVS: * 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.hierarchy: * gst/debug/Makefile.am: * gst/debug/progressreport.c: (gst_progress_report_post_progress), (gst_progress_report_do_query), (gst_progress_report_report): Make progressreport element post messages with the current progress on the bus. Also add some basic docs for it.
* ext/hal/hal.*: Some small cleanups; deal with errors when parsing the HAL ↵Tim-Philipp Müller2007-02-074-21/+31
| | | | | | | | | | ALSA capabilities a bit better. Original commit message from CVS: * ext/hal/hal.c: (gst_hal_get_string): * ext/hal/hal.h: Some small cleanups; deal with errors when parsing the HAL ALSA capabilities a bit better.
* gst/smpte/gstsmpte.c: Let's try this again and use the right cast this time.Tim-Philipp Müller2007-02-062-2/+8
| | | | | | Original commit message from CVS: * gst/smpte/gstsmpte.c: (gst_smpte_transition_type_get_type): Let's try this again and use the right cast this time.
* gst/smpte/gstsmpte.c: Add cast to avoid compiler warnings with older GLib ↵Tim-Philipp Müller2007-02-062-2/+9
| | | | | | | | | | versions where the nick/name members in GEn... Original commit message from CVS: * gst/smpte/gstsmpte.c: (gst_smpte_transition_type_get_type): Add cast to avoid compiler warnings with older GLib versions where the nick/name members in GEnumValue are not declared as constant strings.
* ext/gconf/: In gconfaudiosink, get the right key as the old key in do_toggle ↵Tim-Philipp Müller2007-02-064-31/+54
| | | | | | | | | | | | | | | | (ie. one dependent on the profile select... Original commit message from CVS: * ext/gconf/gconf.c: (gst_gconf_get_key_for_sink_profile), (gst_gconf_render_bin_from_key), (gst_gconf_get_default_audio_sink): * ext/gconf/gconf.h: * ext/gconf/gstgconfaudiosink.c: (get_gconf_key_for_profile), (do_toggle_element), (gst_gconf_audio_sink_set_property), (gst_gconf_audio_sink_get_property): In gconfaudiosink, get the right key as the old key in do_toggle (ie. one dependent on the profile selected). Log some more stuff so we can see what's actually going on.
* gst/audiofx/: Some small cleanups and port both elements to the new ↵Sebastian Dröge2007-02-066-148/+116
| | | | | | | | | | | | | | | | | | GstAudioFilter base class to save a few lines of ... Original commit message from CVS: * gst/audiofx/audioamplify.c: (gst_audio_amplify_base_init), (gst_audio_amplify_class_init), (gst_audio_amplify_init), (gst_audio_amplify_set_process_function), (gst_audio_amplify_setup): * gst/audiofx/audioamplify.h: * gst/audiofx/audioinvert.c: (gst_audio_invert_base_init), (gst_audio_invert_class_init), (gst_audio_invert_setup): * gst/audiofx/audioinvert.h: Some small cleanups and port both elements to the new GstAudioFilter base class to save a few lines of common code. * gst/audiofx/Makefile.am: Link against libgstaudio for the above changes
* Fix up to use the newly ported (actually working) GstAudioFilter.Tim-Philipp Müller2007-02-033-149/+178
| | | | | | | | | | | | | | | | | | | | | | | Original commit message from CVS: * configure.ac: * gst/equalizer/Makefile.am: * gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_base_init), (gst_iir_equalizer_class_init), (gst_iir_equalizer_init), (setup_filter), (gst_iir_equalizer_compute_frequencies), (gst_iir_equalizer_set_property), (gst_iir_equalizer_get_property), (gst_iir_equalizer_transform_ip), (gst_iir_equalizer_setup), (plugin_init): * gst/equalizer/gstiirequalizer.h: Fix up to use the newly ported (actually working) GstAudioFilter. Bump core/base requirements to CVS for this. * tests/icles/.cvsignore: * tests/icles/Makefile.am: * tests/icles/equalizer-test.c: (check_bus), (equalizer_set_band_value), (equalizer_set_all_band_values), (equalizer_set_band_value_and_wait), (equalizer_set_all_band_values_and_wait), (do_slider_fiddling), (main): Add brain-dead interactive test for equalizer.
* gst/equalizer/gstiirequalizer.c: Rename "values" property to "band-values" ↵Tim-Philipp Müller2007-02-021-17/+35
| | | | | | | | | | | | | and change type into a Original commit message from CVS: * gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_class_init), (gst_iir_equalizer_set_property), (gst_iir_equalizer_get_property), (gst_iir_equalizer_filter_inplace): Rename "values" property to "band-values" and change type into a GValueArray, so it's more easily bindable and the range of the values passed in is defined and checked etc.; also do some locking.
* Port equalizer plugin to 0.10 (#403572).James Doc Livingston2007-02-022-25/+22
| | | | | | | | | | | | | | Original commit message from CVS: Patch by: James "Doc" Livingston <doclivingston at gmail com> * configure.ac: * gst/equalizer/Makefile.am: * gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_get_type), (gst_iir_equalizer_base_init), (gst_iir_equalizer_class_init), (gst_iir_equalizer_compute_frequencies), (gst_iir_equalizer_set_property), (gst_iir_equalizer_filter_inplace), (gst_iir_equalizer_setup), (plugin_init): Port equalizer plugin to 0.10 (#403572).
* ext/wavpack/gstwavpackparse.c: Fix a off by one that leads to the duration ↵Sebastian Dröge2007-01-311-6/+4
| | | | | | | | | | | reported as one sample less than it is Original commit message from CVS: * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_src_query), (gst_wavpack_parse_handle_seek_event), (gst_wavpack_parse_create_src_pad): Fix a off by one that leads to the duration reported as one sample less than it is
* configure.ac: Check for an Objective C compilerEdward Hervey2007-01-305-0/+1156
| | | | | | | | | | | | | | Original commit message from CVS: * configure.ac: Check for an Objective C compiler * sys/Makefile.am: * sys/osxvideo/Makefile.am: * sys/osxvideo/cocoawindow.h: * sys/osxvideo/cocoawindow.m: * sys/osxvideo/osxvideosink.h: * sys/osxvideo/osxvideosink.m: Port of osxvideo plugin to 0.10. Do NOT consider 100% stable ! Fixes #402470
* tests/check/elements/.cvsignore: Some more ignores.Wim Taymans2007-01-292-0/+7
| | | | | | Original commit message from CVS: * tests/check/elements/.cvsignore: Some more ignores.
* gst/videocrop/gstvideocrop.c: Fix cropping for packed 4:2:2 formats ↵Tim-Philipp Müller2007-01-282-50/+69
| | | | | | | | | | | | | | | YUYV/YUY2 and UYVY. Original commit message from CVS: * gst/videocrop/gstvideocrop.c: (gst_video_crop_get_image_details_from_caps), (gst_video_crop_transform_packed_complex): Fix cropping for packed 4:2:2 formats YUYV/YUY2 and UYVY. * tests/icles/videocrop-test.c: (check_bus_for_errors), (test_with_caps), (main): Block streaming thread before changing filter caps while the pipeline is running so that we don't get random not-negotiated errors just because GStreamer can't handle that yet.
* tests/icles/videocrop-test.c: Catch errors while the test is running.Tim-Philipp Müller2007-01-271-16/+36
| | | | | | Original commit message from CVS: * tests/icles/videocrop-test.c: (test_with_caps): Catch errors while the test is running.
* ext/shout2/gstshout2.*: Properly handle tags in shout2send. Fixes #399825.charles2007-01-263-21/+46
| | | | | | | | | Original commit message from CVS: Patch by: charles <charlesg3 at gmail dot com> * ext/shout2/gstshout2.c: (gst_shout2send_init), (set_shout_metadata), (gst_shout2send_event): * ext/shout2/gstshout2.h: Properly handle tags in shout2send. Fixes #399825.
* ext/wavpack/gstwavpackparse.c: Fix the SEEKING query. We can seek if we are ↵Sebastian Dröge2007-01-251-3/+6
| | | | | | | | | | | in pull mode, not the other way around. A... Original commit message from CVS: * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_src_query): Fix the SEEKING query. We can seek if we are in pull mode, not the other way around. Also set the correct format in the seeking query and handle the case where the headers are not read yet and we can't say anything about our seeking capabilities.
* ext/wavpack/: Fix spelling in 2 places: It's called Wavpack, not WavePack.Sebastian Dröge2007-01-252-3/+3
| | | | | | | Original commit message from CVS: * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_base_init): * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_base_init): Fix spelling in 2 places: It's called Wavpack, not WavePack.
* gst/rtsp/gstrtspsrc.c: Convert SDP fields to upper/lowercase following the ↵Wim Taymans2007-01-252-7/+26
| | | | | | | | | | rules in the SDP to caps document. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps), (gst_rtspsrc_activate_streams): Convert SDP fields to upper/lowercase following the rules in the SDP to caps document.
* gst/rtp/: Fix case of encoding-name and key/value pairs to match the document.Wim Taymans2007-01-2513-19/+40
| | | | | | | | | | | | | | | | | | | Original commit message from CVS: * gst/rtp/README: * gst/rtp/gstrtpilbcdepay.c: * gst/rtp/gstrtpilbcpay.c: * gst/rtp/gstrtpmp4gdepay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpspeexdepay.c: * gst/rtp/gstrtpspeexpay.c: * gst/rtp/gstrtpsv3vdepay.c: * gst/rtp/gstrtptheoradepay.c: * gst/rtp/gstrtptheorapay.c: * gst/rtp/gstrtpvorbisdepay.c: * gst/rtp/gstrtpvorbispay.c: Fix case of encoding-name and key/value pairs to match the document. This is to make interoperation with SDP case-insensitive as required by the relevant RFCs.
* gst/: Use proper print statements.Edward Hervey2007-01-252-2/+4
| | | | | | | | | | | | | | | | | | | | Original commit message from CVS: * gst/multifile/gstmultifilesink.c: (gst_multi_file_sink_class_init): * gst/multifile/gstmultifilesrc.c: (gst_multi_file_src_class_init): * gst/mve/gstmvedemux.c: (gst_mve_video_create_buffer), (gst_mve_video_palette), (gst_mve_video_code_map), (gst_mve_audio_init), (gst_mve_audio_data), (gst_mve_timer_create), (gst_mve_demux_chain): * gst/mve/gstmvemux.c: (gst_mve_mux_push_chunk): * gst/mve/mveaudioenc.c: (mve_compress_audio): * gst/mve/mvevideodec16.c: (ipvideo_copy_block): * gst/mve/mvevideodec8.c: (ipvideo_copy_block): * gst/mve/mvevideoenc16.c: (mve_encode_frame16): * gst/mve/mvevideoenc8.c: (mve_encode_frame8): Use proper print statements. Fixes build on mac os x. <wingo> oo look at me my name is edward i'm hacking on macos wooo
* configure.ac: Bump required -core/-base to CVSWim Taymans2007-01-252-2/+7
| | | | | | Original commit message from CVS: * configure.ac: Bump required -core/-base to CVS
* gst/rtp/gstrtpL16pay.*: Fill up to MTU using adapter.Wim Taymans2007-01-253-21/+70
| | | | | | | | | Original commit message from CVS: * gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_setcaps), (gst_rtp_L16_pay_flush), (gst_rtp_L16_pay_handle_buffer): * gst/rtp/gstrtpL16pay.h: Fill up to MTU using adapter. Timestamp rtp packets.
* Use G_GSIZE_FORMAT in print statements for portability.Edward Hervey2007-01-253-3/+13
| | | | | | | | Original commit message from CVS: * gst/multipart/multipartmux.c: (gst_multipart_mux_collected): * sys/ximage/ximageutil.c: (ximageutil_check_xshm_calls): Use G_GSIZE_FORMAT in print statements for portability. Fixes build on macosx.
* gst/rtp/: Port and enable raw audio payloader/depayloader. Needs a bit more ↵Wim Taymans2007-01-247-454/+323
| | | | | | | | | | | | | | | | | | | | | | | work on the payloader side. Original commit message from CVS: * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: (plugin_init): * gst/rtp/gstrtpL16depay.c: (gst_rtp_L16_depay_base_init), (gst_rtp_L16_depay_class_init), (gst_rtp_L16_depay_init), (gst_rtp_L16_depay_parse_int), (gst_rtp_L16_depay_setcaps), (gst_rtp_L16_depay_process), (gst_rtp_L16_depay_set_property), (gst_rtp_L16_depay_get_property), (gst_rtp_L16_depay_change_state), (gst_rtp_L16_depay_plugin_init): * gst/rtp/gstrtpL16depay.h: * gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_get_type), (gst_rtp_L16_pay_base_init), (gst_rtp_L16_pay_class_init), (gst_rtp_L16_pay_init), (gst_rtp_L16_pay_finalize), (gst_rtp_L16_pay_setcaps), (gst_rtp_L16_pay_handle_buffer), (gst_rtp_L16_pay_plugin_init): * gst/rtp/gstrtpL16pay.h: Port and enable raw audio payloader/depayloader. Needs a bit more work on the payloader side.
* gst/rtsp/gstrtspsrc.*: Only unblock the udp pads when we linked and ↵Wim Taymans2007-01-243-9/+31
| | | | | | | | | | | | activated them all. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (pad_blocked), (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_udp): * gst/rtsp/gstrtspsrc.h: Only unblock the udp pads when we linked and activated them all. Fixes #395688.
* gst/rtp/: Added simple AC3 depayloader (RFC 4184).Wim Taymans2007-01-246-6/+403
| | | | | | | | | | | | | | | Original commit message from CVS: * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: (plugin_init): * gst/rtp/gstrtpac3depay.c: (gst_rtp_ac3_depay_base_init), (gst_rtp_ac3_depay_class_init), (gst_rtp_ac3_depay_init), (gst_rtp_ac3_depay_setcaps), (gst_rtp_ac3_depay_process), (gst_rtp_ac3_depay_set_property), (gst_rtp_ac3_depay_get_property), (gst_rtp_ac3_depay_change_state), (gst_rtp_ac3_depay_plugin_init): * gst/rtp/gstrtpac3depay.h: Added simple AC3 depayloader (RFC 4184). * gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_setcaps): Fix a leak.
* gst/audiofx/: Add new element "audioamplify". This allows scaling of raw ↵Sebastian Dröge2007-01-2412-3/+1065
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | audio samples, similar to the "volume" eleme... Original commit message from CVS: reviewed by: Stefan Kost <ensonic@users.sf.net> * gst/audiofx/Makefile.am: * gst/audiofx/audioamplify.c: (gst_audio_amplify_clipping_method_get_type), (gst_audio_amplify_base_init), (gst_audio_amplify_class_init), (gst_audio_amplify_init), (gst_audio_amplify_set_process_function), (gst_audio_amplify_set_property), (gst_audio_amplify_get_property), (gst_audio_amplify_set_caps), (gst_audio_amplify_transform_int_clip), (gst_audio_amplify_transform_int_wrap_negative), (gst_audio_amplify_transform_int_wrap_positive), (gst_audio_amplify_transform_float_clip), (gst_audio_amplify_transform_float_wrap_negative), (gst_audio_amplify_transform_float_wrap_positive), (gst_audio_amplify_transform_ip): * gst/audiofx/audioamplify.h: * gst/audiofx/audiofx.c: (plugin_init): Add new element "audioamplify". This allows scaling of raw audio samples, similar to the "volume" element, but provides different modes for clipping and allows unlimited amplification. It's mainly targeted for creative sound design and not as a replacement of the "volume" element. Fixes #397162 * 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: Add docs for audioamplify and integrate them into the build system * tests/check/Makefile.am: * tests/check/elements/audioamplify.c: (setup_amplify), (cleanup_amplify), (GST_START_TEST), (amplify_suite), (main): Add fairly extensive unit test suite for audioamplify
* gst/rtsp/gstrtspsrc.c: Unblock pads after adding the pads to the element so ↵Wim Taymans2007-01-242-4/+13
| | | | | | | | | that autopluggers get a change to link so... Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (pad_unblocked), (pad_blocked): Unblock pads after adding the pads to the element so that autopluggers get a change to link something. Possibly fixes #395688.
* gst/rtp/: Fix caps with payload numbers.Wim Taymans2007-01-2423-19/+79
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Original commit message from CVS: * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpgsmdepay.c: * gst/rtp/gstrtph263pdepay.c: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtpilbcdepay.c: * gst/rtp/gstrtpmp2tdepay.c: * gst/rtp/gstrtpmp4gdepay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_class_init): * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_class_init), (gst_rtp_mpa_depay_init), (gst_rtp_mpa_depay_setcaps), (gst_rtp_mpa_depay_process): * gst/rtp/gstrtpmpvdepay.c: (gst_rtp_mpv_depay_class_init), (gst_rtp_mpv_depay_init), (gst_rtp_mpv_depay_process): * gst/rtp/gstrtppcmadepay.c: * gst/rtp/gstrtppcmudepay.c: * gst/rtp/gstrtpspeexdepay.c: * gst/rtp/gstrtpspeexpay.c: * gst/rtp/gstrtpsv3vdepay.c: * gst/rtp/gstrtptheoradepay.c: * gst/rtp/gstrtptheorapay.c: * gst/rtp/gstrtpvorbisdepay.c: * gst/rtp/gstrtpvorbispay.c: Fix caps with payload numbers. Add some fixed payload numbers to caps when possible.
* gst/qtdemux/gstrtpxqtdepay.c: Fix caps on the depayloader.Wim Taymans2007-01-241-6/+4
| | | | | | Original commit message from CVS: * gst/qtdemux/gstrtpxqtdepay.c: Fix caps on the depayloader.
* gst/audiofx/: Add new audiofx element "audioinvert". This element swaps the ↵Sebastian Dröge2007-01-2312-4/+690
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | upper and lower half of samples and can b... Original commit message from CVS: reviewed by: Stefan Kost <ensonic@users.sf.net> * gst/audiofx/Makefile.am: * gst/audiofx/audiofx.c: (plugin_init): * gst/audiofx/audioinvert.c: (gst_audio_invert_base_init), (gst_audio_invert_class_init), (gst_audio_invert_init), (gst_audio_invert_set_property), (gst_audio_invert_get_property), (gst_audio_invert_set_caps), (gst_audio_invert_transform_int), (gst_audio_invert_transform_float), (gst_audio_invert_transform_ip): * gst/audiofx/audioinvert.h: Add new audiofx element "audioinvert". This element swaps the upper and lower half of samples and can be used for example for a wide-stereo effect. Fixes #396057 * 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: Add docs for the audioinvert element and add them to the build system. * tests/check/Makefile.am: * tests/check/elements/audioinvert.c: (setup_invert), (cleanup_invert), (GST_START_TEST), (invert_suite), (main): Add unit test suite for the audioinvert element.
* gst/rtp/gstrtpmp4gdepay.c: Parse config params as string and int.Wim Taymans2007-01-232-35/+69
| | | | | | | | Original commit message from CVS: * gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_parse_int), (gst_rtp_mp4g_depay_setcaps), (gst_rtp_mp4g_depay_process): Parse config params as string and int. Parse and use AU header length
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