Original commit message from CVS:
Make vis_video_thread play when connected afterwards
This generates some segfaults in gst_thread but we need to fix that
Original commit message from CVS:
total code reorganization as a start to get alsasrc working - sink and src are now really different classes, not just on paper - includes a fix that makes the testsuite work that might be an older bug
Original commit message from CVS:
Rewrote much of videoscale. Now handles most common YUV formats
as well as 32 and 24 bit RGB. Only handles "nearest" scaling.
Original commit message from CVS:
Adds divx/xvid encoders.
* divx encoder is based on divx4linux (commercial, closed-source)
* xvid encoder is based on xvidcore (http://www.xvid.org/, GPL - Christian? ;) )
Both use a GstCaps that doesn't conform with what we currently use, I might fix that later on or so. For now, it doesn't matter, it's just a test. We're also missing corresponding decoders (ffmpeg can decoded this too, but that's not the point), these might come later too.
Original commit message from CVS:
Made a theorical libgstplay which refs/unrefs elements before putting them in AsyncQueue.
Added a "pipeline_error" signal which will later allow the player and apps to detect that pipeline was unable to play and why...
This version is NOT STABLE AT ALL. it will need fixes in core but i commit it as is so that we fix those problems
Original commit message from CVS:
fix clock - seeking, xruns etc should be handled correctly now
includes bugfix to not play the rest of the audio buffer when going PAUSED => READY
Original commit message from CVS:
- revert change to use a newly added gst_buffer_is_readonly (which wasn't newly added before)
- walk buffer backwards when it might be possible that data is read out of overwritten parts when going forwards
Original commit message from CVS:
Initialize various variables so gcc won't complain.
Use GST_BUFFER_FLAG_IS_SET instead of unknown function gst_buffer_is_readonly.
Original commit message from CVS:
Added initial version of audioconvert, a generic converter of integer audio/raw formats.
It currently supports conversion of
- channels (mono/stereo only, until someone tells me how to mix other channels)
- endianness (little/bi endian)
- signedness
- width (8, 1, 24 and 32 bits)
- depth (1 - width bits)
missing:
- enough testing (I intend to write a testsuite for this, but that's pending)
- samplerate conversion
- other goodies like format conversion etc
Expect bugs when using it.
problems this should solve:
- encoding wav files on big endian machines
- goom working with mono audio files in gst-player
- Iain's soundcard (that one is a problem in itself)
- complaints about missing conversion
- too many age old, nearly unmaintained plugins (stereo2mono etc.)
Have fun.
Original commit message from CVS:
make Company happy : Changed visualisation pipeline structure.. audio sink is directly connected to tee so no queue between volume and audio sink...
Original commit message from CVS:
fix timestamp syncing
timestamps are only guessed so add a (big) threshold before starting to drop/insert
fix some clocking madness
Original commit message from CVS:
ALSA rewrite, part 5:
- sync to timestamps (which breaks a _lot_, because most plugins send out wrong timestamps)
- clocking support (A/V sync is superb as long as you don't sync and don't get wrong timestamps)
- 1/2 of format conversion
- assorted bugfixes
I'd like to get people to check the timestamps the plugins send out.
mpegdemux seems to be pretty broken, mad works (I just patched it...), avidemux works at least sometimes.
Haven't checked more so far.
Original commit message from CVS:
Port some changes from 0.6.0/0.6.1 back to HEAD (.pc files moved to pkg-config/, .m4 file moved from plugins to core)
