Original commit message from CVS:
* gst-libs/gst/audio/audio.h: (GST_CLOCK_TIME_TO_FRAMES)
* gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_render)
use of gst_guint64_to_gdouble to be compliant with vs6
* gst/playback/gstdecodebin.c: (try_to_link_1)
* gst/videorate/videorate.c: (gst_video_rate_blank_data)
use of G_GINT64_CONSTANT for int64 constants
* win32/common/libgstinterfaces.def:
export some symbols (gst_mixer_get_type,gst_mixer_track_get_type)
* win32/vs6:
update and add new project files
Original commit message from CVS:
* gst-libs/gst/audio/audio.c: (gst_audio_duration_from_pad_buffer):
* gst-libs/gst/audio/audio.h:
fix prototype - wondering why the test worked regardless
Original commit message from CVS:
Make ringbuffer faster and more simple by removing the locks
in the playback thread.
Add sample accurate playback based on buffer sample offsets.
Make the baseaudiosink provide a clock.
Parse caps in the base class.
Correctly handle seeking, flushing and state changes.
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_get_caps_internal):
buffer-frames property was missing
* ext/arts/gst_arts.c:
rate missing from sinkcaps
* ext/audiofile/gstafparse.c:
* ext/audiofile/gstafsink.c:
* ext/audiofile/gstafsrc.c:
* ext/swfdec/gstswfdec.c:
int audio doesn't know buffer-frames
* ext/cdparanoia/gstcdparanoia.c:
int audio doesn't know chunksize either
* ext/nas/nassink.c:
it's endianness, not endianess
* gst-libs/gst/audio/audio.h:
make float standard pad template caps really describe float
* gst/law/mulaw.c: (linear_factory):
signed only, please
* gst/mpegstream/gstdvddemux.c:
widths of 20 are not valid
Original commit message from CVS:
* gst/auparse/gstauparse.c:
fixes a-law, adds mu-law, linear pcm (8,16,24,32), ieee (32, 64)
only unsupported formats are ADPCM/CCITT G.72x
reviewed by Ronald
* gst-libs/gst/audio/audio.h:
adds 24bit depth to PCM (x-raw-int)
Original commit message from CVS:
2004-02-27 Benjamin Otte <otte@gnome.org>
* gst-libs/gst/audio/audio.h:
add macro to make sure header isn't included twice
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_process_chunk):
don't use gst_buffer_free
* gst/playondemand/filter.func:
don't usae gst_data_free. Free data only once.
Original commit message from CVS:
2004-01-26 Benjamin Otte <in7y118@public.uni-hamburg.de>
* gst-libs/gst/audio/audio.h:
remove buffer-frames from audio caps
* gst/audioconvert/gstaudioconvert.c:
fix plugin to really work.
Original commit message from CVS:
* actually recurse into sndfile if we are able
* big ladspa cleanups, mainly to comply with the buffer-frames caps property, but also general
cleanups
- the samplerate prop is gone, if you want to set it explicitly (as in for get-based plugins)
you need to use a filtered connection, just like with buffer-frames
* big float2int and int2float changes for buffer-frames compatibility - I think it's quite a bit
simpler
* make the ossclock general, add it to gstaudio, and use it in sndfile as well
i need to update mimetypes, but that's coming soon. there are some other plugins that don't
support buffer-frames, i guess i need to get around to fixing them as well.
Original commit message from CVS:
New mimetypes gone into effect today - this commit changes all old mimetypes over to the new mimetypes spec as described in the previous commit's document. Note: some plugins will break, some pipelines will break, expect HEAD to be broken or at least not 100% working for a few days, but don't forget to report bugs
Original commit message from CVS:
moving and renaming
we put the libs in the source in gst-libs/gst/(dir)
the headers get installed in prefix/include/gst/(dir)
the libs are installed in prefix/lib/gst
with a libgst prefix
the sources should be without the gst prefix
as per irc agreement
please comment if this sounds like a bad idea ;)
