Rework the audio caps similar to the video caps. Remove
width/depth/endianness/signed fields and replace with a simple string
format and media type audio/x-raw.
Create a GstAudioInfo and some helper methods to parse caps.
Remove duplicate code from the ringbuffer and replace with audio info.
Use AudioInfo in the base audio filter class.
Port elements to new API.
Alsa seems to expect that we initialize it. Remove the variable and pass NULL
as we actually don't use it. In alsasink also #ifdef one section that is
grabing diagnostics to be disabled, when logging is disabled (the code was
using the out parameter as well).
Fixes#626125
Errors could happen here when the device was removed already
or when something is broken anyway. If errors happen here and
they're propagated, the element can't shutdown cleanly.
Fixes bug #614545.
There's not much point in using GST_DEBUG_FUNCPTR with GObject
virtual functions such as get_property, set_propery, finalize and
dispose, since they'll never be used by anyone anyway. Saves a
few bytes and possibly a sixteenth of a polar bear.
This is due to race conditions between functions that
modified the mixer like set_volume and
snd_mixer_handle_events since the handle_events
can now be called at any time.
Fixed by adding locking around any snd_mixer call
since even read functions can modify the mixer stucture, since
alsa likes to clear it's values before reading new ones.
The favorite race condition seemed to be that set_volume
called read_elem (in alsalib) that reset the volumes to
0 and then read them with read_x_volume. This read looped
on each channel and as the race condition occured the
channels value could be anything , most of the time
it was 0. Thus no value was read or only the value of
one channel was and the volume was reset to 0.
Fixes bug #478512.
Original commit message from CVS:
Based on patch by: Matthias Kretz <kretz at kde dot org>
* ext/alsa/gstalsasink.c: (gst_alsasink_open),
(gst_alsasink_prepare), (gst_alsasink_unprepare),
(gst_alsasink_write):
Make all access non-blocking so that we can better handle unplugging
of usb devices. Fixes#559111
Original commit message from CVS:
Patch by: Sam Morris <sam at robots dot org to uk>
* gst-libs/gst/interfaces/mixertrack.c:
(gst_mixer_track_class_init), (gst_mixer_track_get_property),
(gst_mixer_track_set_property):
API: Add "index" property to GstMixerTrack to differantiate between
multiple mixer tracks with the same label.
* ext/alsa/gstalsamixeroptions.c: (gst_alsa_mixer_options_new):
* ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
Set the "index" property of GstMixerTrack to the index given by ALSA.
Fixes bug #528299.
Original commit message from CVS:
* ext/alsa/gstalsamixer.c:
Also consider "speaker" as a name for master volume. If that doesn't
help look for the first non-mono volume control that also has a
playback switch.
Original commit message from CVS:
* ext/alsa/gstalsamixertrack.c:
(gst_alsa_mixer_track_update_alsa_capabilities):
Make sure playback volumes aren't accidentally overwritten by
capture volumes if an alsa mixer track has both playback and
capture capabilities: we create two GstMixerTracks in that
case, so make sure we query only the alsa capabilities that
refer to the type of GstMixerTrack we created from the dual
capability alsa element. Should fix issues with Audigy2 sound
cards (#518082).
Original commit message from CVS:
* ext/alsa/gstalsadeviceprobe.c:
(gst_alsa_get_device_list): Don't return before freeing up
the allocated structures.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps):
* gst-libs/gst/audio/gstringbuffer.h:
Rename recently added buffer types to make more sense.
* ext/alsa/gstalsasink.c: (alsasink_parse_spec),
(gst_alsasink_write):
Adapt for above API changes.
Fixes bug #520523.
Original commit message from CVS:
2008-02-29 Julien Moutte <julien@fluendo.com>
* ext/alsa/gstalsa.c: (gst_alsa_open_iec958_pcm),
(gst_alsa_probe_supported_formats): Probe for IEC958 pcm to
detect
if we can do SPDIF output.
* ext/alsa/gstalsa.h:
* ext/alsa/gstalsasink.c: (set_hwparams), (alsasink_parse_spec),
(gst_alsasink_prepare), (gst_alsasink_close),
(gst_alsasink_write):
* ext/alsa/gstalsasink.h: Initial support for SPDIF.
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_parse_caps):
* gst-libs/gst/audio/gstringbuffer.h: Add non linear buffer
types
to support AC3, EC3 and IEC958 buffers.
Original commit message from CVS:
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
* ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_delay):
Add some more debug info.
Make sure we never return a negative delay. Fixes#516246.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_delay):
Revert patch that makes the sink hold the object lock when
calling snd_pcm_delay(), since it breaks playback for me.
Original commit message from CVS:
* ext/alsa/gstalsa.h: (GST_CHECK_ALSA_VERSION):
* ext/alsa/gstalsasink.c: (set_swparams):
* ext/alsa/gstalsasrc.c: (set_swparams), (gst_alsasrc_open):
Don't use snd_pcm_sw_params_set_xfer_align() if we're compiling
against libasound >= 1.0.16, since it's been deprecated in
0.10.16, and alignment is always 1 then, apparently. (#512899)
Original commit message from CVS:
2008-02-11 Julien Moutte <julien@fluendo.com>
Patch by: Alan Peevers <peeves@pacbell.net>
* ext/alsa/gstalsasink.c: (gst_alsasink_delay): Take appropriate
lock when calling alsa methods.
Original commit message from CVS:
Patch by: Bastien Nocera <hadess at hadess net>
* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_get_volume),
(check_if_volumes_are_the_same), (gst_alsa_mixer_set_volume):
Use snd_mixer_selem_set_{playback|capture}_volume_all() if
the volume is the same for all channels. This works around
some problem in alsa that leaves us with inconsistent state
for some reason (#486840).
Original commit message from CVS:
Patch by: Jerone Young <jerone at gmail com>
* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_find_master_mixer):
If there's no mixer track by the name of 'Master' or 'Front',
check if there's one called 'PCM' before trying the generic
fallback logic (fixes#506928, where we pick 'Mic' as master
track for the AD1984 card in a Thinkpad T61/X61 laptop).