Original commit message from CVS:
* ext/alsa/gstalsa.c: (add_channels):
handle min <= max correctly
* ext/alsa/gstalsa.c: (gst_alsa_fixate_to_mimetype),
(gst_alsa_fixate_field_nearest_int), (gst_alsa_fixate):
add fixation functions so we fixate correctly. No preferring of alaw
anymore because it's the first structure.
* ext/alsa/gstalsa.h:
* ext/alsa/gstalsa.c: (gst_alsa_sw_params_dump),
(gst_alsa_hw_params_dump):
add functions to ease debugging in alsalib
* ext/alsa/gstalsa.c: (gst_alsa_probe_hw_params),
(gst_alsa_set_hw_params), (gst_alsa_set_sw_params),
(gst_alsa_start_audio):
only specify hw params if we really setup a format (fixes#134007 -
or at least works around it)
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_probe_hw_params),
(gst_alsa_set_hw_params), (gst_alsa_set_sw_params),
(gst_alsa_start_audio):
- don't call set_periods_integer anymore, it breaks the
configuration randomly
- call snd_pcm_hw_params_set_access directly instead of using masks
- don't fail if the sw_params can't be set, just use the default
params and hope it works. Alsalib has weird issues when you touch
sw_params and does no proper error reporting about what failed.
* ext/alsa/gstalsa.c: (gst_alsa_open_audio),
(gst_alsa_close_audio):
make our alsa debugging go via gst debugging and not conditionally
defined
* ext/alsa/gstalsa.h:
add ALSA_DEBUG_FLUSH macro
* ext/alsa/gstalsaplugin.c: (gst_alsa_error_wrapper),
(plugin_init):
wrap alsa errors to be printed via the gst debugging system and not
spammed to stderr
Original commit message from CVS:
second batch :
remove ',' at end of enums as they could confuse older gcc, foreign compilers (forte) and gtk-doc
(in gst-plugins/ext/ this time)
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_get_type), (gst_alsa_class_init),
(gst_alsa_get_property), (gst_alsa_probe_get_properties),
(gst_alsa_class_probe_devices), (gst_alsa_class_list_devices),
(gst_alsa_probe_probe_property), (gst_alsa_probe_needs_probe),
(gst_alsa_probe_get_values), (gst_alsa_probe_interface_init),
(gst_alsa_open_audio), (gst_alsa_close_audio):
* ext/alsa/gstalsa.h:
Add propertyprobe interface implementation, add some device-name
property, all this so that it looks good in gnome-volume-control.
Original commit message from CVS:
2004-02-14 Benjamin Otte <in7y118@public.uni-hamburg.de>
* ext/alsa/gstalsa.c: (gst_alsa_pcm_wait),
(gst_alsa_xrun_recovery):
* ext/alsa/gstalsa.h:
try xrun recovery when wait failed. Make xrun recovery function
return TRUE/FALSE to indicate success. (might fix#134354)
Original commit message from CVS:
2003-12-27 Benjamin Otte <in7y118@public.uni-hamburg.de>
* ext/alsa/gstalsa.c: (gst_alsa_open_audio):
Don't send ALSA debugging to stderr.
* ext/alsa/gstalsa.h:
Use GST_WARNING instead of g_warning when ALSA functions fail.
Original commit message from CVS:
next big bunch of stuff:
- proper caps setting in alsasrc
- query / conversion functions
WARNING: Alsa 0.9.2 is heavily borked wrt recording - expect segfaults
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:
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:
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:
rewrote the caps nego / state change stuff once again, new features:
- bugfixes
- get_caps function to report better caps when device is opened
- better _link function
Original commit message from CVS:
ALSA cleanup step 3:
- make caps nego work, when caps are already set
- rewriting lots of caps nego while doing so
- start stream explicitly now (will probably stay that way because of sync)
- random bugfixes
alsasrc is probably broken again.
alsasink should now be stable enough to be used with gst-player or rhythmbox (seeking works)
Original commit message from CVS:
Bugfixing in alsa again:
- Leif's commit reverted an earlier patch
(stupid diff)
- Some comment from Leif made me clean up his code
- Moved wait() directly in front of mmap
- Assorted fixes
- fixed newbie bug: DON'T EVER USE STATIC VARIABLES WHEN YOU'RE NOT ABSOLUTELY SURE WHAT YOU'RE DOING, Leif *slap* ;)
I hope I didn't break the src now...
Original commit message from CVS:
+ alsasrc compiles and runs in "alsasrc ! fakesink" and "alsasrc ! osssink"
pipelines. seems to have a 100% cpu issue at the moment.
Original commit message from CVS:
fixing alsa step 2: complete rewrite of data transfer. The whole stuff is clean enough to go from there now.
License change to LGPL, since no copied code is left now.
Missing:
- alsasrc
- resetting format
- corner cases
- testsuite
Original commit message from CVS:
+ removed the access_addr crap from GstAlsaPad ... just use
this->access_addr[channel] instead
+ completely reorganized and reindented code
+ removed the gst_alsa_sink_silence_on_channel function, needs to be completely
redone anyway
+ got alsasink to work on my machine finally ! yay !
Original commit message from CVS:
Alsasink is no longer bitrotten anymore, yay!
Alsasrc untested.
Also, fixed a logic error in the main loop regarding proper interpretation of avail_update. This fix came from jack.
Original commit message from CVS:
some jack fixes, alsa touchups, and add rtp by default to the build
if there are any problems building rtp, we're moving it back to experimental ;)
