Original commit message from CVS:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/interfaces/interfaces-marshal.list:
* gst-libs/gst/interfaces/rtspextension.c:
* gst-libs/gst/interfaces/rtspextension.h:
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/rtsp/gstrtsp.h:
* gst-libs/gst/rtsp/gstrtspextension.c:
(gst_rtsp_extension_get_type), (gst_rtsp_extension_iface_init),
(gst_rtsp_extension_detect_server),
(gst_rtsp_extension_before_send), (gst_rtsp_extension_after_send),
(gst_rtsp_extension_parse_sdp), (gst_rtsp_extension_setup_media),
(gst_rtsp_extension_configure_stream),
(gst_rtsp_extension_get_transports),
(gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
* gst-libs/gst/rtsp/gstrtspextension.h:
* gst-libs/gst/rtsp/rtsp-marshal.list:
Move the rtspextension.h interface into gstrtspextension.h
as part of libgstrtsp instead of libgstinterfaces, because it's
only for use within plugins, not applications.
Add stuff to do the enum & marshal generation needed in libgstrtsp now.
Use the GST_TYPE_RTSP_RESULT enum type for the return value of the
signal that the GstRTSPExtension interface emits, since G_TYPE_ENUM
is abstract.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_decode_ip):
* gst-libs/gst/rtsp/gstrtspbase64.h:
API: gst_rtsp_base64_decode_ip()
Added function to decode Base64 in-place.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/audio.c: (gst_audio_buffer_clip):
* gst-libs/gst/audio/audio.h:
* tests/check/libs/audio.c: (GST_START_TEST), (audio_suite):
API: Add buffer clipping function for raw audio buffers. Fixes#456656.
Also add deprecation guards for gst_audio_structure_set_int() to the
header.
Original commit message from CVS:
* gst-libs/gst/interfaces/mixer.h:
Add padding vars in place of the signal pointers
when building with DISABLE_DEPRECATED so that the
interface structure doesn't change size.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_class_init), (gst_basertppayload_init),
(gst_basertppayload_set_property),
(gst_basertppayload_get_property):
Don't break ABI, restore previous ranges. Keep the default random
selection of timestamp and seqnum offset but as soon as the app sets a
specific value, use that one.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_class_init), (gst_basertppayload_init),
(gst_basertppayload_event), (gst_basertppayload_push),
(gst_basertppayload_set_property),
(gst_basertppayload_get_property),
(gst_basertppayload_change_state):
* gst-libs/gst/rtp/gstbasertppayload.h:
Fix ranges of rtp payloader properties so that the full range can be
used in addition to -1 (random).
Fix wrong seqnum reporting in caps.
Fixes#420326.
Original commit message from CVS:
* gst-libs/gst/interfaces/propertyprobe.c:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/tag/gstvorbistag.c:
Make gtk-doc happy.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_callback):
Quick hack to make audiosinks stop at EOS when operating in
pull-mode; needs to be fixed properly some day.
Original commit message from CVS:
* gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
Don't pass trailing zeroes in fixed-size string arrays in ID3v1 tags
to utf8-validate; fixes recognition of ID3v1 tags in UTF-8 encoding
(#451707); also, output some debugging info when dealing with
freeform strings.
* tests/check/libs/tag.c: (GST_START_TEST), (tag_suite):
Add unit test for the above.
Original commit message from CVS:
* gst-libs/gst/pbutils/descriptions.c: (caps_are_rtp_caps):
Add description for Windows Media RTP caps.
* gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps):
Remove RTP fields that don't define the format from caps.
Original commit message from CVS:
2007-06-19 Andy Wingo <wingo@pobox.com>
* gst-libs/gst/audio/gstbaseaudiosink.c
(gst_base_audio_sink_init): Enable pull-mode operation.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Change minimum rate back to 1000 to allow low-sample-rate wav files
to play back.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_set_gst_timestamp):
Use G_GINT64_CONSTANT macro for int64 constant.
* win32/common/libgstinterfaces.def:
* win32/common/libgsttag.def:
Add new exported functions.
Original commit message from CVS:
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/riff/riff-read.c: (gst_riff_parse_info):
Use gst_tag_utf8_from_freeform_string() from libgsttag instead of
our own implementation.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
In riff, the depth is stored in the size field but it just means that
the least significant bits are cleared. We can therefore just play
the sample as if it had a depth == width. Fixes: #440997
Patch by: Wim Taymans <wim@fluendo.com>
Patch by: Sebastian Dröge <slomo@circular-chaos.org>
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
After an interrupt (PAUSED/flush) assume that the next sample should not
be aligned to the previous sample. Fixes#417992.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Don't add channels and rate fields to the template caps for
audio/x-dts, as wavparse might not always be able to set them,
which would then lead to 'caps are not a real subset of the
template caps' warnings.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Specify the full valid range for MP3 samplerates. Fixes a regression
caused by extra header checks since the last release.
Original commit message from CVS:
* ext/cdparanoia/gstcdparanoiasrc.c:
(gst_cd_paranoia_src_read_sector):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
Small cleanups.
* ext/theora/theoradec.c: (theora_dec_sink_event):
Fix typo.
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_set_gst_timestamp):
Add some FIXME
* gst/playback/gstdecodebin.c: (queue_underrun_cb):
And some debug info when a FIXME path is hit.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_class_init),
(gst_base_rtp_audio_payload_init),
(gst_base_rtp_audio_payload_finalize),
(gst_base_rtp_audio_payload_handle_frame_based_buffer),
(gst_base_rtp_audio_payload_handle_sample_based_buffer),
(gst_base_rtp_payload_audio_handle_event):
Some cleanups, remove minptime property as it is now in the parent
class.
Override parent class event function.
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_class_init), (gst_basertppayload_init),
(gst_basertppayload_event), (gst_basertppayload_set_property),
(gst_basertppayload_get_property):
* gst-libs/gst/rtp/gstbasertppayload.h:
Add min-ptime property.
Add handle-event vmethod. Fixes#415001.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_setcaps),
(gst_base_rtp_depayload_set_gst_timestamp):
Parse and use additional caps fields as described in updated
application/x-rtp caps spec.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_handle_frame_based_buffer):
Move variable declaration before the first instruction.
* gst/videotestsrc/videotestsrc.c:
Define M_PI if it's not defined yet.
* win32/common/libgstrtp.def:
Add new exported functions.
Original commit message from CVS:
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_add_track):
Fix it so that it (a) makes sense and (b) doesn't break
everything cdda-related including the unit test.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c: (_do_init),
(gst_base_audio_src_create):
* po/POTFILES.in:
When posting a warning message because samples were dropped, post
something more intelligible than he default error message for clock
errors which is just confusing in this context (#432984).
Original commit message from CVS:
Patch by: Olivier Crete <tester at tester ca>
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_class_init),
(gst_base_rtp_audio_payload_init),
(gst_base_rtp_audio_payload_dispose):
Chain up to parent class in dispose function; get rid of
unnecessary 'diposed' flag in private structure (#415001).
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs.types:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_class_init):
* gst-libs/gst/rtp/gstbasertpdepayload.c:
* gst-libs/gst/rtp/gstbasertppayload.c:
Some minor docs fixes and additions; also add missing 'Since' bits.
Original commit message from CVS:
Patch by: Zeeshan Ali <zeenix gmail com>
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_handle_frame_based_buffer),
(gst_base_rtp_audio_payload_handle_sample_based_buffer),
(gst_base_rtp_audio_payload_push):
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
The recently-added gst_base_rtp_audio_payload_push() should take an
object of type GstBaseRTPAudioPayload as first argument (#431672).
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Allow random depths between 1 and 32 instead of only multiplies of 8.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Set the maximum number of channels for PCM and float in the correct
place to have it also used when creating the template caps.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Correctly support 4, 6 and 8 channels with normal PCM and float
wav files.
Fix the depth and signedness calculation in extensible wav files and
also handle 1, 2, 4, 6, 8 channels here when a file without channel
mask is found.
Add support for float, alaw and mulaw in extensible wav files.
This allows correct playback of all but 5 files from
http://www-mmsp.ece.mcgill.ca/Documents/AudioFormats/WAVE/Samples.html
(gst_riff_create_audio_template_caps):
Add voxware and float formats to the template caps.
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.
Original commit message from CVS:
* gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
Try encodings from all environment variables, not just those in the
first environment variable that is set.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_skew_slaving):
Don't try to create invalid calibration parameters by making the
internal time go backwards, instead make external time go forward.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Add audio/x-raw-float support, now that audioconvert support
non-native endianness floats.
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler dot de>
with some minor changes
* gst-libs/gst/floatcast/floatcast.h:
Use more efficient float endianness conversion functions that don't
involve 2 function calls per value.
* gst/audioconvert/audioconvert.c: (audio_convert_get_func_index),
(check_default), (audio_convert_prepare_context):
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_parse_caps), (make_lossless_changes):
Support non-native endianness floats as input and output.
Fixes#339838.
* tests/check/elements/audioconvert.c: (verify_convert),
(GST_START_TEST):
Add unit tests for the non-native endianness float conversions.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_base_init),
(gst_base_rtp_depayload_class_init), (gst_base_rtp_depayload_init),
(gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
(gst_base_rtp_depayload_set_gst_timestamp),
(gst_base_rtp_depayload_change_state),
(gst_base_rtp_depayload_set_property),
(gst_base_rtp_depayload_get_property):
* gst-libs/gst/rtp/gstbasertpdepayload.h:
Add Private structure.
Bring element code to 2007.
Parse clock-base caps param and use it when generating the
newsegment.
Reset variables before going to PAUSED.
Fix some docs.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
PCM samples with width=8 must be always unsigned, no matter what
depth they have.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtpbuffer.c:
(gst_rtp_buffer_default_clock_rate):
* gst-libs/gst/rtp/gstrtpbuffer.h:
Fix fixed payload names and docs.
Added method to get the default clock rates of fixed payload types.
API: GstRTPBuffer::gst_rtp_buffer_default_clock_rate()
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c: (slave_method_get_type),
(gst_base_audio_sink_class_init), (gst_base_audio_sink_init),
(gst_base_audio_sink_query), (gst_base_audio_sink_get_time),
(gst_base_audio_sink_set_property),
(gst_base_audio_sink_get_property), (gst_base_audio_sink_event),
(clock_convert_external), (gst_base_audio_sink_resample_slaving),
(gst_base_audio_sink_skew_slaving),
(gst_base_audio_sink_handle_slaving), (gst_base_audio_sink_render),
(gst_base_audio_sink_async_play):
* gst-libs/gst/audio/gstbaseaudiosink.h:
Store private stuff in GstBaseAudioSinkPrivate.
Add configurable clock slaving modes property.
API:: GstBaseAudioSink::slave-method property
Some more latency reporting tweaks.
Added skew based clock slaving correction and make it the default until
the resampling method is more robust.
Original commit message from CVS:
* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
* tests/check/libs/tag.c: (GST_START_TEST):
Make sure we parse floating-point numbers in vorbis comments
correctly with either '.' or ',' as separator, no matter what
the current locale is. Add unit test for this too.
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler de>
* gst-libs/gst/tag/gstvorbistag.c: (gst_tag_to_vorbis_comments):
When writing out floating-point numbers to vorbis comment tags, always
use the same character as separator no matter what the current locale is
(fixes#423051).
* tests/check/libs/tag.c: (GST_START_TEST):
Add unit tests for replaygain tags in vorbis comments (closes#423055).
Original commit message from CVS:
* gst-libs/gst/app/Makefile.am:
Use GST_ALL_LDFLAGS, which actually exists, but maybe David
can confirm that was what he wanted.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
Add min-ptime property to RTP base audio payloader. Patch by
olivier.crete@collabora.co.uk.
Fixes#415001
Indentation/whitespace/documentation fixes.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:(gst_base_audio_sink_render):
Use gst_guint64_to_gdouble for conversion.
* win32/MANIFEST:
Add new files to the win32 MANIFEST.
* win32/common/libgstaudio.def:
* win32/common/libgstpbutils.def:
Add new exported functions.
* win32/vs6/gst_plugins_base.dsw:
* win32/vs6/libgstdecodebin.dsp:
* win32/vs6/libgstplaybin.dsp:
Change the link to libgstpbutils.lib.
* win32/vs6/libgstdecodebin2.dsp:
Add a new project for decodebin2.
* win32/vs6/libgstpbutils.dsp:
Add a new project for pbutils.
Original commit message from CVS:
* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
Also accept partial dates with only year and month,
like 1999-12-00 (fixes#410396 even more).
* tests/check/libs/tag.c: (GST_START_TEST):
Add unit test for the above.
Original commit message from CVS:
* gst-libs/gst/interfaces/mixertrack.c:
(gst_mixer_track_class_init), (gst_mixer_track_get_property),
(gst_mixer_track_set_property):
API: add "untranslated-label" property which should be set by
implementations at construct time (#414645).
* ext/alsa/gstalsamixeroptions.c: (gst_alsa_mixer_options_new):
* ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
Set "untranslated-label" when constructing mixer track objects.
* tests/check/elements/alsa.c: (GST_START_TEST), (alsa_suite):
Unit test to check the above.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_async_play):
Fix regression that made GStreamer skip the first samples of audio.
Fixes#414684.
Original commit message from CVS:
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/app/gstappbuffer.c:
* gst-libs/gst/app/gstappbuffer.h:
* gst-libs/gst/app/gstappsrc.c:
Add GstAppBuffer that includes a callback and closure for
proper handling of data chunks.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
* gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
Improve debugging.
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_query), (gst_base_audio_sink_event),
(gst_base_audio_sink_render), (gst_base_audio_sink_async_play):
Improve latency and clock slaving calculations.
Improve slave clock calibration.
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_commit_full):
When we are asked to render N sample to 0 bytes, return N.
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler de>
* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
Parse date strings in vorbis comments that have an invalid (zero)
month or day (#410396).
* tests/check/libs/tag.c: (GST_START_TEST):
Test case for the above.
Original commit message from CVS:
* gst-libs/gst/utils/install-plugins.c:
* gst-libs/gst/utils/missing-plugins.c:
* tests/check/libs/utils.c: (missing_msg_check_getters):
Change GStreamer marker prefix in detail string from 'gstreamer.net'
to just 'gstreamer'. Document the caps string component of the
decoder/encoder detail a bit better, since not everyone will be
familiar with the GStreamer media type/caps system (but they better
enjoy nested itemized lists).
Original commit message from CVS:
* gst-libs/gst/netbuffer/gstnetbuffer.c:
(notgst_buffer_copy_fields_in_place), (gst_netbuffer_copy):
Fix copying of GstNetBuffer (would crash before, or at least lead to
invalid memory access, #410772), for now by copying the GstBuffer copy
code from the core over here so we can copy the GstBuffer fields on a
provided buffer instance (of type GstNetBuffer in this case). Would be
better to fix this with some support by the core though (and in the long
run change the broken GstBuffer/GstMiniObject copy semantics, #393099).
* tests/check/Makefile.am:
Enable unit test for GstNetBuffer.
Original commit message from CVS:
2007-02-22 Andy Wingo <wingo@pobox.com>
* gst-libs/gst/audio/gstbaseaudiosink.c
(gst_base_audio_sink_init): Disable pull-mode activation until we
figure out how to make audio sinks go to PLAYING.
Original commit message from CVS:
* gst-libs/gst/utils/base-utils.c:
* gst-libs/gst/utils/descriptions.c:
* gst-libs/gst/utils/install-plugins.c:
* gst-libs/gst/utils/missing-plugins.c:
Some more docs (and descriptions for two subtitle formats).
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler de>
* gst-libs/gst/tag/gstvorbistag.c:
Add vorbis/FLAC-tag mapping for new GST_TAG_REFERENCE_LEVEL
(#403597).
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiofilter.c:
(gst_audio_filter_class_init), (gst_audio_filter_change_state):
Clear our formats structure and free the caps contained in it when
shutting down.
Original commit message from CVS:
2007-02-05 Andy Wingo <wingo@pobox.com>
* gst-libs/gst/audio/gstbaseaudiosink.c
(gst_base_audio_sink_callback): Update basesink->offset so that we
pull monotonically increasing offsets instead of, um, seeking back
to 0 each time. Fixes alsasrc ! alsasink!
