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94 commits

Author SHA1 Message Date
Andoni Morales Alastruey
2434f2932b alsasink: check for spdif support only in the current device 2012-05-18 12:01:06 +02:00
Sebastian Dröge
69b18ab09d gst-libs: Remove interfaces libs and mixer/tuner interfaces 
The navigation interface is now in the video library.
 2012-04-13 13:14:13 +02:00
Tim-Philipp Müller
3c6a3ad629 Use new gst_element_class_set_static_metadata() 2012-04-10 00:45:16 +01:00
Sebastian Dröge
75f91ebea0 ext: Add new layout field to the raw audio caps 2012-01-05 10:34:25 +01:00
Sebastian Dröge
2fc75efdce alsa: Port to the new multichannel caps 2012-01-05 10:34:20 +01:00
Tim-Philipp Müller
3dfdd6be9d audioringbuffer: rename GST_BUFTYPE_* to GST_AUDIO_RING_BUFFER_FORMAT_TYPE_* 
Bit unwieldy, but more appropriate. Could also be moved into
audio.h as GstAudioFormatType.
 2011-12-25 21:38:21 +00:00
Wim Taymans
dde5e5a248 alsa: remove more property probe stuff 2011-12-22 16:37:29 +01:00
Tim-Philipp Müller
fb6d09055a Merge remote-tracking branch 'origin/master' into 0.11 
Conflicts:
	ext/alsa/gstalsadeviceprobe.c
	ext/alsa/gstalsamixer.c
	ext/pango/gsttextoverlay.c
	ext/pango/gsttextoverlay.h
	gst-libs/gst/audio/gstaudiobasesink.c
	gst-libs/gst/audio/gstaudioringbuffer.c
	gst-libs/gst/audio/gstaudiosrc.c
	gst-libs/gst/video/Makefile.am
	gst-libs/gst/video/video.c
	gst/encoding/gststreamcombiner.c
	gst/encoding/gststreamsplitter.c
	gst/playback/gstplaybasebin.c
	gst/playback/gststreamsynchronizer.c
	gst/playback/gstsubtitleoverlay.c
	gst/playback/gsturidecodebin.c
	sys/xvimage/xvimagesink.c
	tests/examples/Makefile.am
	win32/common/libgstvideo.def

Video overlay composition disabled for now, needs
porting to buffer meta.
 2011-12-08 01:19:03 +00:00
Tim-Philipp Müller
0d98aa25b8 Work around deprecated thread API in glib master 
Add private replacements for deprecated functions such as
g_mutex_new(), g_mutex_free(), g_cond_new() etc., mostly
to avoid the deprecation warnings. We'll change these
over to the new API once we depend on glib >= 2.32.

Replace g_thread_create() with g_thread_try_new().
 2011-12-04 17:16:30 +00:00
Tim-Philipp Müller
ec0d3566bf Merge remote-tracking branch 'origin/master' into 0.11 
Conflicts:
	ext/alsa/gstalsasrc.c
	ext/alsa/gstalsasrc.h
	gst/adder/gstadder.c
	gst/playback/gstplaybin2.c
	gst/playback/gstplaysinkconvertbin.c
	win32/common/libgstvideo.def
 2011-12-02 00:07:39 +00:00
Tim-Philipp Müller
e88e47cd24 Revert "alsasrc: Improve timestamp accuracy" 
This reverts commit 0b774e0b7c.
 2011-11-30 23:15:35 +00:00
Tim-Philipp Müller
e5ae553850 Revert "alsasrc: Fix some compilation errors" 
This reverts commit 2b84f5bd74.
 2011-11-30 23:15:22 +00:00
Tim-Philipp Müller
4cc8920db4 Revert "alsa: Remove unused but set variable" 
This reverts commit e9aed7f31c.
 2011-11-30 23:15:12 +00:00
Tim-Philipp Müller
1290f7de0e Revert "alsasrc: fail gracefully when ALSA does not give timestamps" 
This reverts commit c7282a5718.
 2011-11-30 23:15:03 +00:00
Tim-Philipp Müller
d11849114c Revert "alsasrc: handle the case where the drivers don't supply timestamps" 
This reverts commit 8154b69112.
 2011-11-30 23:14:54 +00:00
Stefan Sauer
6d167abdfa Revert "alsasrc: style fix" 
This reverts commit f70ca6d4cb.
 2011-11-30 23:14:44 +00:00
Tim-Philipp Müller
0c056a04fe Merge commit '4a58223e4c824fedc024af435337a769e8ce593e' into 0.11 2011-11-28 21:20:10 +00:00
Vincent Penquerc'h
96374054ac various: fix pad template leaks 
https://bugzilla.gnome.org/show_bug.cgi?id=662664
 2011-11-28 13:09:02 +00:00
Stefan Sauer
f70ca6d4cb alsasrc: style fix 
Use timestamp==0 instead of mixing it with !timestamp style checks.
 2011-11-28 10:55:39 +01:00
Stefan Sauer
8154b69112 alsasrc: handle the case where the drivers don't supply timestamps 
If highres-timestamp is 0, try lowres and if that fails fallback to system clock
timestamps.
 2011-11-28 09:13:29 +01:00
Wim Taymans
ee7072fe7e rename GstBaseAudio* ->GstAudioBase* 2011-11-11 11:52:47 +01:00
Wim Taymans
6511f36fdb audio: GstRingBuffer -> GstAudioRingBuffer 2011-11-11 11:21:41 +01:00
Wim Taymans
3254e79f04 alsa: fix negotiation 
Don't assume the format is a string because now it is a list of string in the
template.
Chain up to the parent class implementation of get_caps.
 2011-11-10 16:05:19 +01:00
Wim Taymans
a00927ad03 Merge branch 'master' into 0.11 2011-10-04 17:58:49 +02:00
Vincent Penquerc'h
c7282a5718 alsasrc: fail gracefully when ALSA does not give timestamps 
https://bugzilla.gnome.org/show_bug.cgi?id=660170
 2011-10-03 11:14:09 +02:00
Wim Taymans
33196cdd2c audio: change audio format syntax a little 
Remove the _ in front of the endianness prefix.
Remove the _3 postfix for the 24 bits formats.
Add a _32 postfix after the formats that occupy extra space beyond their
natural size.
The result is that the GST_AUDIO_NE() macro can simply append the endianness
after all formats and that we only specify a different sample width when it is
different from the natural size of the sample. This makes things more consistent
and follows the pulseaudio conventions instead of the alsa ones.
 2011-09-06 12:06:39 +02:00
Wim Taymans
dae848818d audio: rework audio caps. 
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.
 2011-08-18 19:15:03 +02:00
Tim-Philipp Müller
c16e7321b9 alsa: don't use GstImplementsInterface 2011-06-26 22:58:17 +01:00
Wim Taymans
489eed9bb8 Merge branch 'master' into 0.11 2011-05-19 11:31:53 +02:00
Robert Swain
e9aed7f31c alsa: Remove unused but set variable 
Unused but set variables cause warnings in GCC 4.6.x and newer.
 2011-05-18 09:34:52 +02:00
Sebastian Dröge
c255019b90 ext: Update for caps/pad template related API changes 2011-05-17 13:06:01 +02:00
Sebastian Dröge
d0362c2b87 Merge branch 'master' into 0.11 
Conflicts:
	configure.ac
	ext/alsa/gstalsasrc.c
	gst-libs/gst/audio/gstbaseaudiosink.c
	gst-libs/gst/tag/gstxmptag.c
	gst/playback/gstsubtitleoverlay.c
	gst/videorate/gstvideorate.c
	sys/xvimage/xvimagesink.c
 2011-05-16 17:06:22 +02:00
Sebastian Dröge
0415b90e99 alsa: Update for negotiation related API changes 2011-05-16 15:35:41 +02:00
Sebastian Dröge
2b84f5bd74 alsasrc: Fix some compilation errors 2011-05-14 11:42:32 +02:00
Pontus Oldberg
0b774e0b7c alsasrc: Improve timestamp accuracy 
Fixes bug #635256.
 2011-05-14 11:42:32 +02:00
Sebastian Dröge
353186aec8 ext: Use G_DEFINE_TYPE instead of GST_BOILERPLATE 2011-04-19 14:22:42 +02:00
Wim Taymans
7b310c6a03 alsasrc/sink: add property to get the card name 
fixes #627203
 2010-08-18 16:45:37 +02:00
Stefan Kost
0fee4ed3d0 alsa: remove 'dir' out variable 
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
 2010-08-12 15:41:59 +03:00
Tim-Philipp Müller
930f72c6b0 alsa: don't pass non-constant strings as printf format strings 
Fixes 'format not a string literal and no format arguments' compiler
warning when compiling with -DGST_DISABLE_PRINTF_EXTENSION.
 2010-04-08 01:26:55 +01:00
Sebastian Dröge
44e474f76d alsa: Ignore errors when unpreparing or closing the device 
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.
 2010-04-04 21:18:04 +02:00
Benjamin Otte
5e21fa5e0e gst_element_class_set_details => gst_element_class_set_details_simple 
Also change my email from the old university one to the current one.
 2010-03-16 17:41:50 +01:00
Wim Taymans
1f601e12dc alsasrc: return right number of bytes that we wrote 2010-03-08 11:25:01 +01:00
Tim-Philipp Müller
6f4c1ac583 Remove GST_DEBUG_FUNCPTR where they're pointless 
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.
 2009-10-28 00:59:35 +00:00
Edward Hervey
76044dce6d ext: Remove dead assignments and resulting unused variables. 2009-08-08 15:54:41 +02:00
Balachandran C
01e0fdd86c alsasrc: set alsasrc->handle back to NULL when closing device 
Fixes crashes in gst_alsa_find_device_name() when probing or
reading the device-name property (e.g. when doing a dot-file
dump). Fixes #589797.
 2009-07-27 14:18:27 +01:00
Stefan Kost
2b33c755b6 Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipe... 
Original commit message from CVS:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-overrides.txt:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/gst-plugins-base-plugins.interfaces:
* docs/plugins/gst-plugins-base-plugins.prerequisites:
* docs/plugins/gst-plugins-base-plugins.signals:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-queue2.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* ext/alsa/gstalsamixer.c:
* ext/alsa/gstalsasink.c:
* ext/alsa/gstalsasrc.c:
* ext/gio/gstgiosink.c:
* ext/gio/gstgiosrc.c:
* ext/gio/gstgiostreamsink.c:
* ext/gio/gstgiostreamsrc.c:
* ext/gnomevfs/gstgnomevfssink.c:
* ext/gnomevfs/gstgnomevfssrc.c:
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggmux.c:
* ext/pango/gstclockoverlay.c:
* ext/pango/gsttextoverlay.c:
* ext/pango/gsttextrender.c:
* ext/pango/gsttimeoverlay.c:
* ext/theora/theoradec.c:
* ext/theora/theoraenc.c:
* ext/theora/theoraparse.c:
* ext/vorbis/vorbisdec.c:
* ext/vorbis/vorbisenc.c:
* ext/vorbis/vorbisparse.c:
* ext/vorbis/vorbistag.c:
* gst/adder/gstadder.c:
* gst/audioconvert/gstaudioconvert.c:
* gst/audioresample/gstaudioresample.c:
* gst/audiotestsrc/gstaudiotestsrc.c:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/gdp/gstgdpdepay.c:
* gst/gdp/gstgdppay.c:
* gst/playback/gstdecodebin2.c:
* gst/playback/gstplaybin.c:
* gst/playback/gstplaybin2.c:
* gst/playback/gstqueue2.c:
* gst/playback/gsturidecodebin.c:
* gst/tcp/gstmultifdsink.c:
* gst/tcp/gsttcpserversink.c:
* gst/videorate/gstvideorate.c:
* gst/videoscale/gstvideoscale.c:
* gst/videotestsrc/gstvideotestsrc.c:
* gst/volume/gstvolume.c:
* sys/ximage/ximagesink.c:
* sys/xvimage/xvimagesink.c:
Cleanup Plugin docs. Link to signals and properties. Fix sub-section
titles. Drop mentining that all our example pipelines are "simple"
pipelines.
 2008-07-10 21:06:06 +00:00
Sebastian Dröge
49deb0c05d Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory u... 
Original commit message from CVS:
* configure.ac:
* ext/alsa/gstalsamixerelement.c:
(gst_alsa_mixer_element_class_init):
* ext/alsa/gstalsasink.c: (gst_alsasink_class_init):
* ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init):
* ext/cdparanoia/gstcdparanoiasrc.c:
(gst_cd_paranoia_src_class_init):
* ext/gio/gstgiosink.c: (gst_gio_sink_class_init):
* ext/gio/gstgiosrc.c: (gst_gio_src_class_init):
* ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_class_init):
* ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_class_init):
* ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init):
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init):
* ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init):
* ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
* ext/pango/gsttextrender.c: (gst_text_render_class_init):
* ext/theora/theoradec.c: (gst_theora_dec_class_init):
* ext/theora/theoraenc.c: (gst_theora_enc_class_init):
* ext/theora/theoraparse.c: (gst_theora_parse_class_init):
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_class_init):
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
(gst_audio_filter_template_class_init):
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init):
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_class_init):
* gst-libs/gst/interfaces/mixertrack.c:
(gst_mixer_track_class_init):
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_class_init):
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_class_init):
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_class_init):
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_class_init):
* gst/audioresample/gstaudioresample.c:
(gst_audioresample_class_init):
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audio_test_src_class_init):
* gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init):
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
(preroll_unlinked):
* gst/playback/gstplaybin.c: (gst_play_bin_class_init):
* gst/playback/gstplaybin2.c: (gst_play_bin_class_init):
* gst/playback/gstplaysink.c: (gst_play_sink_class_init):
* gst/playback/gstqueue2.c: (gst_queue_class_init):
* gst/playback/gststreaminfo.c: (gst_stream_info_class_init):
* gst/playback/gststreamselector.c: (gst_selector_pad_class_init),
(gst_stream_selector_class_init):
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init):
* gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
* gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
* gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init):
* gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init):
* gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init):
* gst/videorate/gstvideorate.c: (gst_video_rate_class_init):
* gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init):
* gst/videotestsrc/gstvideotestsrc.c:
(gst_video_test_src_class_init):
* gst/volume/gstvolume.c: (gst_volume_class_init):
* sys/v4l/gstv4lelement.c: (gst_v4lelement_class_init):
* sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init):
* sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init):
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init):
* sys/ximage/ximagesink.c: (gst_ximagesink_class_init):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use
static strings (i.e. all). This gives us less memory usage,
fewer allocations and thus less memory defragmentation. Depend
on core CVS for this. Fixes bug #523806.
 2008-03-22 15:00:53 +00:00
Sebastian Dröge
ec7afb6f84 Correct all relevant warnings found by the sparse semantic code analyzer. This include marking several symbols static... 
Original commit message from CVS:
* ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_get_type):
* ext/alsa/gstalsasink.c: (set_hwparams):
* ext/alsa/gstalsasrc.c: (set_hwparams):
* ext/gio/gstgio.c: (gst_gio_uri_handler_get_uri):
* ext/ogg/gstoggmux.h:
* ext/ogg/gstogmparse.c:
* gst-libs/gst/audio/audio.c:
* gst-libs/gst/fft/kiss_fft_f64.c: (kiss_fft_f64_alloc):
* gst-libs/gst/pbutils/missing-plugins.c:
(gst_missing_uri_sink_message_new),
(gst_missing_element_message_new),
(gst_missing_decoder_message_new),
(gst_missing_encoder_message_new):
* gst-libs/gst/rtp/gstbasertppayload.c:
* gst-libs/gst/rtp/gstrtcpbuffer.c:
(gst_rtcp_packet_bye_get_reason):
* gst/audioconvert/gstaudioconvert.c:
* gst/audioresample/gstaudioresample.c:
* gst/ffmpegcolorspace/imgconvert.c:
* gst/playback/test.c: (gen_video_element), (gen_audio_element):
* gst/typefind/gsttypefindfunctions.c:
* gst/videoscale/vs_4tap.c:
* gst/videoscale/vs_4tap.h:
* sys/v4l/gstv4lelement.c:
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_get_any_caps):
* sys/v4l/v4l_calls.c:
* sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_capture_init),
(gst_v4lsrc_try_capture):
* sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
(gst_ximagesink_ximage_new):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls),
(gst_xvimagesink_xvimage_new):
* tests/check/elements/audioconvert.c:
* tests/check/elements/audioresample.c:
(fail_unless_perfect_stream):
* tests/check/elements/audiotestsrc.c: (setup_audiotestsrc):
* tests/check/elements/decodebin.c:
* tests/check/elements/gdpdepay.c: (setup_gdpdepay),
(setup_gdpdepay_streamheader):
* tests/check/elements/gdppay.c: (setup_gdppay), (GST_START_TEST),
(setup_gdppay_streamheader):
* tests/check/elements/gnomevfssink.c: (setup_gnomevfssink):
* tests/check/elements/multifdsink.c: (setup_multifdsink):
* tests/check/elements/textoverlay.c:
* tests/check/elements/videorate.c: (setup_videorate):
* tests/check/elements/videotestsrc.c: (setup_videotestsrc):
* tests/check/elements/volume.c: (setup_volume):
* tests/check/elements/vorbisdec.c: (setup_vorbisdec):
* tests/check/elements/vorbistag.c:
* tests/check/generic/clock-selection.c:
* tests/check/generic/states.c: (setup), (teardown):
* tests/check/libs/cddabasesrc.c:
* tests/check/libs/video.c:
* tests/check/pipelines/gio.c:
* tests/check/pipelines/oggmux.c:
* tests/check/pipelines/simple-launch-lines.c:
(simple_launch_lines_suite):
* tests/check/pipelines/streamheader.c:
* tests/check/pipelines/theoraenc.c:
* tests/check/pipelines/vorbisdec.c:
* tests/check/pipelines/vorbisenc.c:
* tests/examples/seek/scrubby.c:
* tests/examples/seek/seek.c: (query_positions_elems),
(query_positions_pads):
* tests/icles/stress-xoverlay.c: (myclock):
Correct all relevant warnings found by the sparse semantic code
analyzer. This include marking several symbols static, using
NULL instead of 0 for pointers and using "foo (void)" instead
of "foo ()" for declarations.
* win32/common/libgstrtp.def:
Add gst_rtp_buffer_set_extension_data to the symbol definition file.
 2008-03-03 06:04:31 +00:00
Tim-Philipp Müller
5f1cc19bef ext/alsa/: Don't use snd_pcm_sw_params_set_xfer_align() if we're compiling against libasound >= 1.0.16, since it's be... 
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)
 2008-02-11 20:23:44 +00:00
Tim-Philipp Müller
5c279f449a ext/alsa/: 'Could not open resource for writing' is not an acceptable even less so when we're trying to open it to re... 
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_open):
* ext/alsa/gstalsasrc.c: (gst_alsasrc_open):
'Could not open resource for writing' is not an acceptable
error message when we can't open the audio device (see #492334),
even less so when we're trying to open it to record something.
 2007-11-03 10:39:21 +00:00