summaryrefslogtreecommitdiffstats
path: root/gst
Commit message (Collapse)AuthorAgeFilesLines
* gst/avi/gstavidemux.*: Save some memory (8%) by repacking the index entry ↵Stefan Kost2007-02-122-202/+302
| | | | | | | | | | | | | | | | | | | | 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.
* gst/goom/gstgoom.*: Improved docs and use GST_DEBUG_FUNCPTR.Stefan Kost2007-02-124-18/+30
| | | | | | | | | | | | | 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 Müller2007-02-124-1/+274
| | | | | | | | | | | | | | 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.
* gst/avi/gstavimux.c: Comment a #if 0 in caps template definition as VS6 ↵Sébastien Moutte2007-02-118-9/+28
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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.
* gst/avi/gstavimux.c: Explicitly cast result of pointer arithmetic to integer ↵Tim-Philipp Müller2007-02-091-2/+2
| | | | | | | | | 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 Müller2007-02-081-0/+12
| | | | | | Original commit message from CVS: * gst/debug/progressreport.c: Some more docs.
* docs/plugins/inspect/plugin-rtp.xml: Update for new elements.Tim-Philipp Müller2007-02-071-0/+68
| | | | | | | | 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 Müller2007-02-072-34/+84
| | | | | | | | | | | | | | | 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.
* gst/smpte/gstsmpte.c: Let's try this again and use the right cast this time.Tim-Philipp Müller2007-02-061-2/+3
| | | | | | 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 Müller2007-02-061-2/+2
| | | | | | | | | | 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.
* gst/audiofx/: Some small cleanups and port both elements to the new ↵Sebastian Dröge2007-02-065-148/+101
| | | | | | | | | | | | | | | | | | 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).
* gst/videocrop/gstvideocrop.c: Fix cropping for packed 4:2:2 formats ↵Tim-Philipp Müller2007-01-281-12/+14
| | | | | | | | | | | | | | | 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.
* gst/rtsp/gstrtspsrc.c: Convert SDP fields to upper/lowercase following the ↵Wim Taymans2007-01-251-7/+19
| | | | | | | | | | 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 Taymans2007-01-2512-19/+22
| | | | | | | | | | | | | | | | | | | 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
* gst/rtp/gstrtpL16pay.*: Fill up to MTU using adapter.Wim Taymans2007-01-252-21/+62
| | | | | | | | | 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 Hervey2007-01-251-2/+4
| | | | | | | | 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 Taymans2007-01-246-454/+303
| | | | | | | | | | | | | | | | | | | | | | | 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 Taymans2007-01-242-9/+22
| | | | | | | | | | | | 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 Taymans2007-01-245-6/+388
| | | | | | | | | | | | | | | 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 Dröge2007-01-244-3/+520
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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 Taymans2007-01-241-4/+7
| | | | | | | | | 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 Taymans2007-01-2422-19/+49
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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 Dröge2007-01-234-4/+347
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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 Taymans2007-01-231-35/+62
| | | | | | | | 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
* gst/smpte/: constify some static structs.Wim Taymans2007-01-235-40/+82
| | | | | | | | | | | | | | Original commit message from CVS: * gst/smpte/barboxwipes.c: (gst_wipe_boxes_draw), (gst_wipe_triangles_clock_draw), (gst_wipe_triangles_draw): * gst/smpte/gstmask.c: (_gst_mask_register): * gst/smpte/gstmask.h: * gst/smpte/gstsmpte.c: (gst_smpte_update_mask): * gst/smpte/paint.c: (gst_smpte_paint_hbox), (draw_bresenham_line), (gst_smpte_paint_triangle_clock): constify some static structs. Don't update the mask if nothing changed to the params. Make sure we never draw outside of the picture. Fixes #398325.
* gst/avi/gstavidemux.c: Error out properly when pull_range fails while we're ↵Tim-Philipp Müller2007-01-221-9/+13
| | | | | | | | | reading the headers, instead of just paus... Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_stream_header_pull): Error out properly when pull_range fails while we're reading the headers, instead of just pausing the task silently. Fixes #399338.
* gst/smpte/gstsmpte.c: Some more sanity checks to make sure the input formats ↵Tim-Philipp Müller2007-01-191-5/+18
| | | | | | | | | | | match and the input pads are actually ne... Original commit message from CVS: * gst/smpte/gstsmpte.c: (gst_smpte_collected): Some more sanity checks to make sure the input formats match and the input pads are actually negotiated, in case someone tries to feed buffers from fakesrc or filesrc. Fixes #398299. Also const-ify an array, just because we can.
* gst/smpte/gstsmpte.c: Ignore previous commit, that was only valid for widths ↵Edward Hervey2007-01-191-6/+18
| | | | | | | | | | | | | and heights that are multiples of 4. Original commit message from CVS: * gst/smpte/gstsmpte.c: (fill_i420), (gst_smpte_collected): Ignore previous commit, that was only valid for widths and heights that are multiples of 4. Copy over size/stride macros from jpegdec. This allows the element to work with any width,height... ... but puts in evidence that the actual transformations only work with width/height that are multiples of 4.
* gst/smpte/gstsmpte.c: Allocate buffers of the right size.Edward Hervey2007-01-191-3/+3
| | | | | | | | | | Original commit message from CVS: * gst/smpte/gstsmpte.c: (gst_smpte_collected): Allocate buffers of the right size. The proper size of a I420 buffer in bytes is: width * height * 3 ------------------ 2
* gst/smpte/gstsmpte.c: Proxy getcaps on sink pads too, so that we either end ↵Tim-Philipp Müller2007-01-181-2/+8
| | | | | | | | | | up with the same dimensions on all pads o... Original commit message from CVS: * gst/smpte/gstsmpte.c: (gst_smpte_init): Proxy getcaps on sink pads too, so that we either end up with the same dimensions on all pads or error out if that's not possible (seems to work even!). Fixes #398086, I think.
* gst/audiofx/audiopanorama.c: Fix doc section name (Fixes #397946)Stefan Kost2007-01-181-1/+1
| | | | | | Original commit message from CVS: * gst/audiofx/audiopanorama.c: Fix doc section name (Fixes #397946)
* gst/audiofx/audiopanorama.c: Use a function array for process methods, add ↵Sebastian Dröge2007-01-161-50/+40
| | | | | | | | | | | more docs and define the startindex of enums. Original commit message from CVS: Patch by: Sebastian Dröge <slomo circular-chaos org> * gst/audiofx/audiopanorama.c: (gst_audio_panorama_class_init), (gst_audio_panorama_set_process_function): Use a function array for process methods, add more docs and define the startindex of enums.
* Add support for more than one audio stream; write better AVIX header; ↵Mark Nauwelaerts2007-01-142-538/+536
| | | | | | | | | | | | | | | | | | | | | | | refactor code a bit; don't announce vorbis caps... Original commit message from CVS: Patch by: Mark Nauwelaerts <manauw at skynet be> * gst/avi/gstavimux.c: (gst_avi_mux_finalize), (gst_avi_mux_pad_reset), (gst_avi_mux_reset), (gst_avi_mux_init), (gst_avi_mux_vidsink_set_caps), (gst_avi_mux_audsink_set_caps), (gst_avi_mux_request_new_pad), (gst_avi_mux_release_pad), (gst_avi_mux_riff_get_avi_header), (gst_avi_mux_riff_get_avix_header), (gst_avi_mux_riff_get_header), (gst_avi_mux_write_avix_index), (gst_avi_mux_add_index), (gst_avi_mux_bigfile), (gst_avi_mux_start_file), (gst_avi_mux_stop_file), (gst_avi_mux_handle_event), (gst_avi_mux_do_buffer), (gst_avi_mux_do_one_buffer), (gst_avi_mux_change_state): * gst/avi/gstavimux.h: * tests/check/elements/avimux.c: (teardown_src_pad): Add support for more than one audio stream; write better AVIX header; refactor code a bit; don't announce vorbis caps on our audio sink pads since we don't support it anyway. Closes #379298.
* gst/interleave/deinterleave.c (gst_deinterleave_add_new_pads): Use fixed ↵Andy Wingo2007-01-131-5/+11
| | | | | | | | | | | | | | caps on src pads. Original commit message from CVS: 2007-01-13 Andy Wingo <wingo@pobox.com> * gst/interleave/deinterleave.c (gst_deinterleave_add_new_pads): Use fixed caps on src pads. (gst_deinterleave_remove_pads): Remove src pads, not sink pads. I seem to have reverse midas disease! (gst_deinterleave_process): Proxy timestamps, offsets, durations, and set caps on outgoing buffers. Fixes #395597, I think.
* gst/interleave/interleave.c (gst_interleave_init): Init the activation mode ↵Andy Wingo2007-01-131-17/+133
| | | | | | | | | | | | | | | | | | | properly. Original commit message from CVS: 2007-01-13 Andy Wingo <wingo@pobox.com> * gst/interleave/interleave.c (gst_interleave_init): Init the activation mode properly. (gst_interleave_src_setcaps, gst_interleave_src_getcaps) (gst_interleave_init): Set a setcaps and getcaps function on the src pad, so that we can implement pull-mode negotiation. (gst_interleave_sink_setcaps): Renamed from gst_interleave_setcaps, as it only does the sink logic now. Implement both for pull-mode and push-mode. (gst_interleave_process): Set caps on our outgoing buffer. (gst_interleave_src_activate_pull): Fix some more bogus casts. What is up with this.
* gst/audiofx/audiopanorama.*: Add 'method' property and provide a simple ↵Sebastian Dröge2007-01-132-31/+229
| | | | | | | | | | | | | | | | | | | | | | | (non-psychoacustic) processing method (#394859). Original commit message from CVS: Patch by: Sebastian Dröge <slomo circular-chaos org> * gst/audiofx/audiopanorama.c: (gst_audio_panorama_method_get_type), (gst_audio_panorama_class_init), (gst_audio_panorama_init), (gst_audio_panorama_set_process_function), (gst_audio_panorama_set_property), (gst_audio_panorama_get_property), (gst_audio_panorama_set_caps), (gst_audio_panorama_transform_m2s_int_simple), (gst_audio_panorama_transform_s2s_int_simple), (gst_audio_panorama_transform_m2s_float_simple), (gst_audio_panorama_transform_s2s_float_simple): * gst/audiofx/audiopanorama.h: Add 'method' property and provide a simple (non-psychoacustic) processing method (#394859). * tests/check/elements/audiopanorama.c: (GST_START_TEST), (panorama_suite): Tests for new method.
* gst/qtdemux/: Add X-QT depayloader that will eventually share code with the ↵Wim Taymans2007-01-126-25/+839
| | | | | | | | | | | | | | | | | | | | | | | demuxer. Original commit message from CVS: * gst/qtdemux/Makefile.am: * gst/qtdemux/gstrtpxqtdepay.c: (gst_rtp_xqt_depay_base_init), (gst_rtp_xqt_depay_class_init), (gst_rtp_xqt_depay_init), (gst_rtp_xqt_depay_finalize), (gst_rtp_quicktime_parse_sd), (gst_rtp_xqt_depay_setcaps), (gst_rtp_xqt_depay_process), (gst_rtp_xqt_depay_set_property), (gst_rtp_xqt_depay_get_property), (gst_rtp_xqt_depay_change_state), (gst_rtp_xqt_depay_plugin_init): * gst/qtdemux/gstrtpxqtdepay.h: * gst/qtdemux/qtdemux.c: (gst_qtdemux_base_init), (gst_qtdemux_loop_state_header), (gst_qtdemux_loop), (qtdemux_parse_moov), (qtdemux_parse_container), (qtdemux_parse_node), (gst_qtdemux_add_stream), (qtdemux_parse_trak), (qtdemux_audio_caps): * gst/qtdemux/qtdemux.h: * gst/qtdemux/quicktime.c: (plugin_init): Add X-QT depayloader that will eventually share code with the demuxer. Make new plugin entry point with quicktime releated stuff.
* gst/qtdemux/Makefile.am: Dist all new files.Tim-Philipp Müller2007-01-121-1/+6
| | | | | | Original commit message from CVS: * gst/qtdemux/Makefile.am: Dist all new files.
* gst/qtdemux/: Cleanup and refactor to make the code more readable.Wim Taymans2007-01-129-1871/+2037
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Original commit message from CVS: * gst/qtdemux/Makefile.am: * gst/qtdemux/qtdemux.c: (extract_initial_length_and_fourcc), (gst_qtdemux_loop_state_header), (gst_qtdemux_combine_flows), (gst_qtdemux_loop_state_movie), (gst_qtdemux_loop), (gst_qtdemux_chain), (qtdemux_sink_activate_pull), (qtdemux_inflate), (qtdemux_parse_moov), (qtdemux_parse_container), (qtdemux_parse_node), (qtdemux_tree_get_child_by_type), (qtdemux_tree_get_sibling_by_type), (gst_qtdemux_add_stream), (qtdemux_parse_samples), (qtdemux_parse_segments), (qtdemux_parse_trak), (qtdemux_tag_add_str), (qtdemux_tag_add_num), (qtdemux_tag_add_date), (qtdemux_tag_add_gnre), (qtdemux_parse_udta), (qtdemux_redirects_sort_func), (qtdemux_process_redirects), (qtdemux_parse_redirects), (qtdemux_parse_tree), (gst_qtdemux_handle_esds), (qtdemux_video_caps), (qtdemux_audio_caps): * gst/qtdemux/qtdemux.h: * gst/qtdemux/qtdemux_dump.c: (qtdemux_dump_mvhd), (qtdemux_dump_tkhd), (qtdemux_dump_elst), (qtdemux_dump_mdhd), (qtdemux_dump_hdlr), (qtdemux_dump_vmhd), (qtdemux_dump_dref), (qtdemux_dump_stsd), (qtdemux_dump_stts), (qtdemux_dump_stss), (qtdemux_dump_stsc), (qtdemux_dump_stsz), (qtdemux_dump_stco), (qtdemux_dump_co64), (qtdemux_dump_dcom), (qtdemux_dump_cmvd), (qtdemux_dump_unknown), (qtdemux_node_dump_foreach), (qtdemux_node_dump): * gst/qtdemux/qtdemux_dump.h: * gst/qtdemux/qtdemux_fourcc.h: * gst/qtdemux/qtdemux_types.c: (qtdemux_type_get): * gst/qtdemux/qtdemux_types.h: * gst/qtdemux/qtpalette.h: Cleanup and refactor to make the code more readable. Move debugging/tables into separate files. Add 2/4/16 color palletee support. Fix raw 15 bit RGB handling. Use more FOURCC constants. Add some docs.
* gst/: Set correct caps on outgoing pulled buffers, or things blow up after ↵Tim-Philipp Müller2007-01-112-0/+4
| | | | | | | | | | recent core changes. Original commit message from CVS: * gst/apetag/gsttagdemux.c: (gst_tag_demux_read_range): * gst/id3demux/gstid3demux.c: (gst_id3demux_read_range): Set correct caps on outgoing pulled buffers, or things blow up after recent core changes.
* gst/multipart/multipartmux.c: Return FLOW errors ASAP. Fixes #394977.Jonas Holmberg2007-01-111-40/+70
| | | | | | | | | | | Original commit message from CVS: Based on patch by: Jonas Holmberg <jonas dot holmberg at axis dot com> * gst/multipart/multipartmux.c: (gst_multipart_mux_init), (gst_multipart_mux_request_new_pad), (gst_multipart_mux_queue_pads), (gst_multipart_mux_collected), (gst_multipart_mux_change_state): Return FLOW errors ASAP. Fixes #394977. Misc cleanups.
* gst/rtsp/gstrtspsrc.c: Check for stream pad before activating.Lutz Mueller2007-01-111-5/+7
| | | | | | | Original commit message from CVS: Patch by: Lutz Mueller <lutz at topfrose dot de> * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_activate_streams): Check for stream pad before activating.
* gst/rtsp/: Allow url to be NULL to be able to use it for server connections.Peter Kjellerstedt2007-01-1010-299/+644
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Original commit message from CVS: Patch by: Peter Kjellerstedt <pkj at axis com> * gst/rtsp/COPYING.MIT: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream), (gst_rtspsrc_stream_free), (gst_rtspsrc_cleanup), (gst_rtspsrc_alloc_udp_ports), (pad_unblocked), (pad_blocked), (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp), (gst_rtspsrc_send), (gst_rtspsrc_parse_methods), (gst_rtspsrc_create_transports_string), (gst_rtspsrc_prepare_transports), (gst_rtspsrc_setup_streams), (gst_rtspsrc_open), (gst_rtspsrc_close): * gst/rtsp/gstrtspsrc.h: * gst/rtsp/rtspconnection.c: (rtsp_connection_create), (rtsp_connection_connect), (rtsp_connection_send), (read_line), (parse_request_line), (parse_line), (rtsp_connection_read), (rtsp_connection_close): * gst/rtsp/rtspdefs.c: (rtsp_init_status), (rtsp_strresult), (rtsp_method_as_text), (rtsp_header_as_text), (rtsp_status_as_text), (rtsp_find_header_field), (rtsp_find_method): * gst/rtsp/rtspdefs.h: * gst/rtsp/rtspextwms.c: (rtsp_ext_wms_after_send), (rtsp_ext_wms_configure_stream): * gst/rtsp/rtspmessage.c: (rtsp_message_new), (rtsp_message_init), (rtsp_message_new_request), (rtsp_message_init_request), (rtsp_message_new_response), (rtsp_message_init_response), (rtsp_message_init_data), (rtsp_message_unset), (rtsp_message_free), (rtsp_message_add_header), (rtsp_message_get_header), (rtsp_message_set_body), (rtsp_message_get_body), (dump_mem), (rtsp_message_dump): * gst/rtsp/rtspmessage.h: * gst/rtsp/sdpmessage.c: (sdp_message_get_attribute_val_n), (sdp_media_get_attribute_val_n), (read_string), (read_string_del), (sdp_parse_line), (sdp_message_parse_buffer), (print_media), (sdp_message_dump): Allow url to be NULL to be able to use it for server connections. Can now send responses as well as requests. No longer hangs in an endless loop if EOF is received. Can now convert a status code to a text string. Return RTSP_HDR_INVALID for unknown headers. Return RTSP_INVALID for unknown methods. Copy CSeq and Session headers from the request. Only free memory corresponding to the currently set message type. Added const to function arguments as appropriate. Avoid a compiler warning when initializing nmedia. Use guint rather than gint to avoid compiler warnings. Fix crasher in wms extension. Factor out stream setup from open_connection. Delay activation of streams when actual data is received from the server, this prepares us to do proper protocol switching. Added new license. Fixes #380895.
* Some small docs fixes (#394851).Sebastian Dröge2007-01-101-4/+4
| | | | | | | | Original commit message from CVS: Patch by: Sebastian Dröge <slomo ubuntu com> * docs/plugins/Makefile.am: * gst/audiofx/audiopanorama.c: Some small docs fixes (#394851).
* gst/avi/gstavidemux.c: Fix docs.Wim Taymans2007-01-091-2/+4
| | | | | | Original commit message from CVS: * gst/avi/gstavidemux.c: Fix docs.
generated by cgit (git 2.34.1) at 2025-06-27 22:18:46 +0000