Original commit message from CVS:
* ext/alsa/Makefile.am:
There is no GST_PLUGINS_BASE_LIBS defined.
* ext/alsa/gstalsa.c:
* ext/alsa/gstalsasink.c: (gst_alsasink_delay):
* ext/alsa/gstalsasrc.c: (gst_alsasrc_delay):
Add support for ALSA 24-bit formats.
snd_pcm_delay can return an error code, especially
during XRUNS. In that case, the best we can do is assume
delay = 0.
* gst/audioconvert/Makefile.am:
Add flags from -base before any more-remote dependencies.
Original commit message from CVS:
* ext/alsa/gstalsamixer.c (source, n_poll_fds, poll_fds,
gst_alsa_mixer_check, gst_alsa_mixer_dispatch,
gst_alsa_mixer_finalize, gst_alsa_mixer_handle_source_callback,
gst_alsa_mixer_handle_callback, gst_alsa_mixer_elem_handle_callback,
gst_alsa_mixer_ensure_track_list, gst_alsa_mixer_update_option,
gst_alsa_mixer_update_track):
Apply some of the cleanup Tim suggested in #152864 afterwards.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_open):
Extra log line.
* ext/pango/gstclockoverlay.c: (gst_clock_overlay_init):
* ext/pango/gsttimeoverlay.c: (gst_time_overlay_init):
Use pango_font_description_set_family_static instead of
pango_font_description_set_family to save a string copy (it was
leaking due to the strdup anyway)
* gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_finalize):
* gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_finalize):
* gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_finalize):
* gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_finalize):
Chain up in finalize.
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:
Patch by: Loïc Minier <lool+gnome at via ecp fr>
* configure.ac:
* ext/alsa/Makefile.am:
* gst/audiotestsrc/Makefile.am:
Fix compilation with LDFLAGS='-Wl,-z,defs' (#410963).
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_reset):
Ignore errors in reset, these are not fatal. They also grab the element
lock which is already taking when this function is called. Fixes
#405451.
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_find_device_name):
Try to get devic-name from device string first, and from handle only
as fallback (seems to yield better results and is more robust
against buggy probing code on the application side).
Original commit message from CVS:
Based on patch by: Julien Puydt <julien.puydt at laposte net>
* ext/alsa/gstalsa.c: (gst_alsa_find_device_name_no_handle),
(gst_alsa_find_device_name):
* ext/alsa/gstalsa.h:
* ext/alsa/gstalsasink.c: (gst_alsasink_get_property):
* ext/alsa/gstalsasrc.c: (gst_alsasrc_get_property):
Improve device-name detection a bit, especially in the case where
the device is not actually open (#405020, #405024). Move common code
into gstalsa.c instead of duplicating it.
Original commit message from CVS:
* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_find_master_mixer),
(gst_alsa_mixer_ensure_track_list):
Try harder to guess which mixer track is the master mixer
track (instead of just taking the first one that has a pvolume).
Fixes#342228.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (xrun_recovery), (gst_alsasink_write):
* ext/alsa/gstalsasrc.c: (xrun_recovery), (gst_alsasrc_read):
Use DEBUG_OBJECT more.
Original commit message from CVS:
Patch by: Viktor Peters <viktor dot peters at gmail dot com>
* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_ensure_track_list),
(gst_alsa_mixer_update), (gst_alsa_mixer_get_volume),
(gst_alsa_mixer_set_volume), (gst_alsa_mixer_set_mute),
(gst_alsa_mixer_set_record):
* ext/alsa/gstalsamixertrack.c:
(gst_alsa_mixer_track_update_alsa_capabilities),
(alsa_track_has_cap), (gst_alsa_mixer_track_new),
(gst_alsa_mixer_track_update):
* ext/alsa/gstalsamixertrack.h:
Improve and fix mixer track handling, in particular better handling
of alsa's pvolume/pswitch/cvolume/cswitch capabilities; create separate
track objects for tracks that have both capture and playback volume
(and label them differently as well so they're not mistakenly
assumed to be duplicates); classify mixer tracks that only affect
the audible volume of something (rather than the capture volume)
as playback tracks. Redefine/fix meaning of RECORD and MUTE flags
for capture tracks to correspond to alsa-pswitch alsa-cswitch
(following the meaning documented in the mixer interface header
file); add support for alsa's exclusive cswitch groups; update/sync
state/flags better if mixer settings are changed by another
application. Fixes#336075.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_set_property),
(gst_alsasink_open):
* ext/alsa/gstalsasrc.c: (gst_alsasrc_set_property),
(gst_alsasrc_open):
Avoid setting and using a NULL device name.
Print more info when we fail to open a device.
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_detect_rates),
(gst_alsa_detect_channels), (gst_alsa_probe_supported_formats):
Small code cleanup.
* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open),
(gst_alsa_mixer_new):
Remove hack that always set the device to hw:0*.
Properly find the card name for whatever device was configured.
Do some better debugging.
Fixes#350784.
* ext/alsa/gstalsamixerelement.c:
(gst_alsa_mixer_element_set_property),
(gst_alsa_mixer_element_change_state):
Cleanups.
Handle setting of a NULL device name better.
Original commit message from CVS:
* ext/alsa/gstalsa.c: (caps_add_channel_configuration),
(gst_alsa_detect_channels):
* ext/alsa/gstalsasink.c:
Add support for cards that (only) do more than 8 channels,
like the Delta 44 (#345188).
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions):
* gst-libs/gst/audio/multichannel.h:
API: add GST_AUDIO_CHANNEL_POSITION_NONE, which stands for an
unspecified channel position and cannot be combined with any
of the other audio channel positions; adjust position layout
checks accordingly (#345188).
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
Don't try to calculate silence samples, base class does this much
better now.
* gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
(gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps),
(gst_ring_buffer_acquire):
Calculate silence samples correctly.
* gst-libs/gst/audio/gstringbuffer.h:
Add _CAST macro.
Original commit message from CVS:
* ext/alsa/gstalsamixerelement.c:
(gst_alsa_mixer_element_change_state):
Make state change fail if the specified device can't be opened
for some reason.
Original commit message from CVS:
* ext/alsa/gstalsa.c: (caps_add_channel_configuration):
Fix typo, so that alsasink also advertises 8 channels
if that's supported (tags: can, worms, open, alsa, ph34r).
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
* gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
(gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps):
Revert last two changes that broke the freeze.
Original commit message from CVS:
Patch by: Michael Sheldon <webmaster at mikeasoft com>
* ext/alsa/gstalsasrc.c:
Add 32 bps to template caps and increase channels range
from [1,2] to [1,MAX]. See #346326.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (set_hwparams):
If we fail to set the buffer_time and period_time alsa
parameters, post a warning and leave alsa select a
default instead of failing. Fixes#342085
Original commit message from CVS:
* ext/alsa/gstalsadeviceprobe.c: (gst_alsa_get_device_list),
(gst_alsa_device_property_probe_probe_property),
(gst_alsa_device_property_probe_needs_probe),
(gst_alsa_device_property_probe_get_values),
(gst_alsa_type_add_device_property_probe_interface):
* ext/alsa/gstalsadeviceprobe.h:
* ext/alsa/gstalsamixerelement.c:
(gst_alsa_mixer_element_init_interfaces):
* ext/alsa/gstalsamixerelement.h:
Clean up and simplify alsa device probing. Make it actually work
for multiple classes. Don't cache results any longer.
* ext/alsa/gstalsasink.c: (gst_alsasink_init_interfaces),
(gst_alsasink_init):
* ext/alsa/gstalsasrc.c: (gst_alsasrc_dispose),
(gst_alsasrc_interface_supported), (gst_implements_interface_init),
(gst_alsasrc_init_interfaces), (gst_alsasrc_set_property):
Make alsasink and alsasrc implement the GstPropertyProbe interface
for device probing (#342181).
Patch by: Martin Szulecki <gnomebugzilla at sukimashita com>
Original commit message from CVS:
* ext/alsa/Makefile.am:
* ext/alsa/gstalsa.c: (gst_alsa_detect_rates),
(gst_alsa_detect_formats), (get_channel_free_structure),
(caps_add_channel_configuration), (gst_alsa_detect_channels),
(gst_alsa_probe_supported_formats):
* ext/alsa/gstalsa.h:
* ext/alsa/gstalsasink.c: (gst_alsasink_getcaps):
Refactor and improve caps probing code: probe signedness
when we probe the supported formats/widths; set endianness
to the one we actually probed for (ie. cpu endianness).
* ext/alsa/gstalsasrc.c: (gst_alsasrc_init), (gst_alsasrc_getcaps),
(gst_alsasrc_close):
* ext/alsa/gstalsasrc.h:
Implement caps probing for alsasrc.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_finalise),
(gst_alsasink_init):
* ext/alsa/gstalsasink.h:
Don't leak allocated snd_output_t structure if there's
more than one alsasink instance at a time (#341873).
Also fix GObject macros in header file.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_getcaps),
(alsasink_parse_spec):
query witdh capabilities from alsa, fixes#338919
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (set_hwparams), (alsasink_parse_spec):
More debug to trace why my USB headset is not working with gst
Original commit message from CVS:
* gst/playback/gststreamselector.c:
(gst_stream_selector_set_property),
(gst_stream_selector_bufferalloc):
Preserve the existing buggy streamselector behaviour by performing
a fallback buffer allocation when downstream isn't linked yet.
This should really be fixed in playbin by blocking pads until it's
linked them.
Also, use gst_pad_alloc_buffer instead of
gst_pad_alloc_buffer_and_set.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_finalise):
Chain up to the parent finalize method.
Add 32-bit sample size to the template caps.
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
(gst_riff_create_video_template_caps):
Add the fourcc that the VMWare codec uses.
* gst/playback/gststreamselector.c:
(gst_stream_selector_set_property),
(gst_stream_selector_bufferalloc),
(gst_stream_selector_request_new_pad):
For the active pad, forward buffer-alloc requests, otherwise
return GST_FLOW_NOT_LINKED. This also prevents xvimagesink
having to memcpy every frame when used by playbin.
* gst/tcp/gstmultifdsink.c:
(gst_multi_fd_sink_handle_client_write):
Get negotiated caps from the sink pad, rather than the sink
pad's peer.
Original commit message from CVS:
* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_get_volume):
Fix issues with mixer keeping state when muting/unmuting
and when changing the volume whilst muted (see #331763
and #331765).
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_open),
(gst_alsasink_reset):
Also release lock when we get an error in _reset();
fix an error message.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_class_init),
(gst_alsasink_init), (get_channel_free_structure),
(caps_add_channel_configuration), (gst_alsasink_getcaps),
(gst_alsasink_close):
* ext/alsa/gstalsasink.h:
Add support for more than 2 channels (#326720).
Original commit message from CVS:
2006-02-09 Andy Wingo <wingo@pobox.com>
* gst-libs/gst/audio/gstringbuffer.c
(gst_ring_buffer_samples_done): Cast to guint64, fixes an integer
overflow after 13.5 hours of recording. Kapow!
* ext/alsa/gstalsasrc.c (gst_alsasrc_delay): Clamp the delay to
the buffer size -- we don't care about underrun/overrun reporting
right now, just need to return a useful value.
