Original commit message from CVS:
* common/gtk-doc-plugins.mak:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
renamed to actual element names, so much nicer to look at
* docs/plugins/tmpl/gstmultifdsink.sgml:
remove
* docs/plugins/tmpl/multifdsink.sgml:
* docs/plugins/tmpl/tcpserversink.sgml:
add
* ext/alsa/gstalsa.c:
* ext/cdparanoia/gstcdparanoia.c: (cdparanoia_get_property):
* ext/ogg/gstoggmux.c:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain):
* gst/playback/gstdecodebin.c:
* gst/tcp/gstmultifdsink.c: (gst_multifdsink_class_init):
* gst/tcp/gsttcpserversink.c:
various fixes and documentation additions
Original commit message from CVS:
* TODO:
delete this file, it is by far outdated
* ext/alsa/gstalsa.1: remove
* ext/alsa/gstalsa.c: (add_rates), (add_channels), (gst_alsa_caps),
(gst_alsa_check_sample_rates), (gst_alsa_rates_probe),
(gst_alsa_get_caps):
Add HW probing for supported sample rates. Fixes#161704
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_class_init), (gst_alsa_get_caps):
* ext/alsa/gstalsa.h:
Add HW probing for period_count/size and buffer_size MIX/MAX
Adjust default/user defined value if out of bounds
Should fix bug #162024
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_change_state):
Reset variables on READY.
* gst/matroska/matroska-mux.c: (gst_matroska_mux_request_new_pad),
(gst_matroska_mux_loop):
Require data before writing header.
Original commit message from CVS:
* configure.ac:
Fix indentation, fix v4l2 plugin detection.
* ext/Makefile.am:
Fix libmms location (Maciej, use diff -u!).
* ext/alsa/gstalsa.c: (gst_alsa_init):
Initialize caps cache to NULL.
* gst/playback/gstplaybin.c: (gst_play_bin_change_state):
Only change state on audiosink if it exists.
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_get_caps), (gst_alsa_close_audio):
* ext/alsa/gstalsa.h:
refactor big chunks of the core caps negotiation code to make it
a lot faster, because people claim it's really slow
(actually, just cache the getcaps when the device is opened)
Original commit message from CVS:
2004-11-28 Martin Soto <martinsoto@users.sourceforge.net>
* ext/alsa/gstalsasink.c (gst_alsa_sink_loop):
* ext/alsa/gstalsa.h:
* ext/alsa/gstalsa.c (gst_alsa_set_clock):
Make alsasink actually honor gst_element_set_clock and use that
clock instead of ist internal one.
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_get_caps):
Don't omit the last (which incase of dmix is the only :) )
channel count. Don't set channels if <= 2.
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_pcm_wait):
add debugging
* ext/alsa/gstalsasink.c: (gst_alsa_sink_loop):
do a wait when we enter the loop func with no data available to
write instead of getting into an 100% CPU loop by just returning and
being called again by the scheduler
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_change_state):
Only set hardware parameters *after* negotiation. Before
negotiation, it will set ANY and that seems to cause crashes
(see e.g. #151288, #153227).
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:
* ext/alsa/gstalsa.c: (gst_alsa_open_audio),
(gst_alsa_sw_params_dump), (gst_alsa_hw_params_dump),
(gst_alsa_close_audio):
disable some of the debugging code for now. Writing debugging to a
buffer is broken in current alsalib releases.
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_xrun_recovery):
only restart audio when we indeed have an xrun to fix repeated
xruns. Fix suggested by Giuliano Pochini.
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_xrun_recovery):
use our own functions for restarting the alsa device.
* ext/alsa/gstalsasink.c: (gst_alsa_sink_check_event):
I should apply patches myself - use MIN for the third argument, not
the second, this fixes seeking
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_start), (gst_alsa_xrun_recovery):
* ext/alsa/gstalsa.h:
* ext/alsa/gstalsasrc.c: (gst_alsa_src_init),
(gst_alsa_src_update_avail), (gst_alsa_src_loop):
Use alsa trigger_tstamp to get the timestamp of the first
sample in the buffer for more precise sync. Some cleanups.
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_change_state), (gst_alsa_start),
(gst_alsa_xrun_recovery):
* ext/alsa/gstalsa.h:
* ext/alsa/gstalsasink.c: (gst_alsa_sink_check_event),
(gst_alsa_sink_loop), (gst_alsa_sink_get_time):
* ext/alsa/gstalsasrc.c: (gst_alsa_src_init),
(gst_alsa_src_get_time), (gst_alsa_src_update_avail),
(gst_alsa_src_loop):
Add clock to alsasrc. Take new capture timestamp when
restarting after an overrun. Split up some functions between
alsasrc ans alsasink.
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_get_time), (gst_alsa_clock_update),
(gst_alsa_change_state), (gst_alsa_update_avail),
(gst_alsa_xrun_recovery):
* ext/alsa/gstalsa.h:
* ext/alsa/gstalsasrc.c: (gst_alsa_src_loop):
Cleanups, take queued samples into account when reporting
the time.
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_init), (gst_alsa_dispose),
(gst_alsa_get_time), (gst_alsa_xrun_recovery):
* ext/alsa/gstalsa.h:
* ext/alsa/gstalsaclock.c: (gst_alsa_clock_get_type):
* ext/alsa/gstalsasrc.c: (gst_alsa_src_init), (gst_alsa_src_loop),
(gst_alsa_src_change_state):
* ext/alsa/gstalsasrc.h:
Make the xrun code timestamp and offset the buffers correctly.
moved the clock to the base class, use alsa methods to get time.
Do correct timestamping on outgoing buffers.
Original commit message from CVS:
2004-06-14 Benjamin Otte <otte@gnome.org>
* ext/alsa/gstalsa.c: Use snd_pcm_hw_params_set_rate _near instead of
snd_pcm_hw_params_set_rate since the latter fails for no good
reason on some setups.<
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_samples_to_timestamp):
cast to GstClockTime to get higher granularity
* ext/alsa/gstalsasink.c: (gst_alsa_sink_check_event):
use gst_element_set_time_delay to get the exact time
* ext/mad/gstmad.c: (gst_mad_chain):
use the negotiated rate instead of the current frame's rate which
might be wrong because of bit errors. This avoids emitting totally
bogus timestamps and screwing sync.
(fixes#143454)
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:
* ext/alsa/gstalsa.c: (device_list),
(gst_alsa_class_probe_devices):
* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open):
Fix alsa oddness in mixer after the combination of using mixer
in source/sink elements and using hw:x,y instead of just hw:x.
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_class_probe_devices):
Don't probe for playback device if we're a source element. Fixes
#139658.
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_change_state),
(gst_alsa_close_audio):
handle case better where a soundcard can't pause
* ext/ogg/gstoggdemux.c:
don't crash when we get events but don't have pads yet
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_fixate): Don't fixate fields that
aren't in the caps.
* gst/sine/gstsinesrc.c: change rate caps to [1,MAX]
* gst/videocrop/gstvideocrop.c: (plugin_init): Change rank to NONE.
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_get_property),
(gst_alsa_open_audio), (gst_alsa_close_audio):
* ext/alsa/gstalsa.c:
Don't open the device if we're a mixer (= padless).
* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_class_init),
(gst_alsa_mixer_init), (gst_alsa_mixer_open),
(gst_alsa_mixer_close), (gst_alsa_mixer_change_state):
Open mixer during state change rather than during object
initialization. Also, get a device name. Currently in a somewhat
hackish fashion, but I didn't really find something better.
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_class_probe_devices):
* sys/oss/gstosselement.c: (gst_osselement_class_probe_devices):
Don't block during probing...
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)
