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2015-01-21audiosrc: Fill in the correct silenceSebastian Dröge1-1/+8
For unsigned raw formats this is not all zeroes, and for non-raw formats we just continue to assume all zeroes for now. https://bugzilla.gnome.org/show_bug.cgi?id=739446
2014-04-26docs: remove outdated and pointless 'Last reviewed' lines from docsTim-Philipp Müller1-2/+0
They are very confusing for people, and more often than not also just not very accurate. Seeing 'last reviewed: 2005' in your docs is not very confidence-inspiring. Let's just remove those comments.
2013-12-27audiobase*: Drop trailing withespacesReynaldo H. Verdejo Pinochet1-1/+1
2012-12-12libs: Use foo/foo.h as single-include header consistently everywhereSebastian Dröge1-0/+1
https://bugzilla.gnome.org/show_bug.cgi?id=688785
2012-11-12audio: Use new GType for GThread instead of just G_TYPE_POINTERSebastian Dröge1-3/+6
2012-11-03Fix FSF addressTim-Philipp Müller1-2/+2
https://bugzilla.gnome.org/show_bug.cgi?id=687520
2012-09-10ringbuffer: add support for timestampsPontus Oldberg1-4/+8
Make it possible for subclasses to provide the timestamp (as an absolute time against the pipeline clock) of the last read data. Fix up alsa to provide the timestamp received from alsa. Because the alsa timestamps are in monotonic time, we can only do this when the monotonic clock has been selected as the pipeline clock. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=635256
2012-09-09audio: port to the new GLib thread APIThibault Saunier1-7/+4
2012-09-09Remove glib-compat-private.h stuff we don't need any moreTim-Philipp Müller1-2/+0
It's all been ported to the latest GLib API now.
2012-01-27audiosrc: wait on the right cond variableWim Taymans1-1/+1
This broke with a merge commit
2012-01-18Remove compatibility code cruft for old GLib versionsTim-Philipp Müller1-6/+0
2011-12-08Merge remote-tracking branch 'origin/master' into 0.11Tim-Philipp Müller1-1/+10
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.
2011-12-04Work around deprecated thread API in glib masterTim-Philipp Müller1-0/+9
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().
2011-11-11rename baseaudio* -> audiobase*Wim Taymans1-3/+3
2011-11-11rename GstBaseAudio* ->GstAudioBase*Wim Taymans1-7/+7
2011-11-11audio: GstRingBuffer -> GstAudioRingBufferWim Taymans1-32/+35
2011-11-11audio: rename internal audio ringbufferWim Taymans1-77/+83
2011-11-10audiosrc: avoid deadlockWim Taymans1-0/+5
2011-04-19audio: Use G_DEFINE_TYPE instead of GST_BOILERPLATESebastian Dröge1-9/+4
2011-03-27libs: port to new data APIWim Taymans1-4/+4
2010-01-27audiosrc: add gratuitious FIXME for use of generic G_TYPE_POINTER typeTim-Philipp Müller1-0/+1
2009-10-28Remove GST_DEBUG_FUNCPTR where they're pointlessTim-Philipp Müller1-2/+2
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.
2009-08-08Revert inlines that cause compiler warnings and are not needed anywayTim-Philipp Müller1-1/+1
2009-08-08gst-libs: Remove dead assignments and resulting unused variables.Edward Hervey1-15/+1
2009-07-20audiosrc: Add stream-status messagesOlivier Crête1-0/+16
See #587695
2009-06-17audio: correctly handle short read/writesWim Taymans1-3/+4
2009-06-16audiosink, audiosrc: do the class_ref()s in the right class_init functionsTim-Philipp Müller1-4/+2
Spotted by Philip Jägenstedt. Hopefully fixes #585970 for real.
2009-06-15audiosink,audiosrc: ref the audio ring buffer class and type in class_initTim-Philipp Müller1-0/+4
Hack around thread-safety issues in GObject and our racy _get_type() functions (we could easily fix the _get_type() functions, but we still need to hack around the GObject class races until we require a newer GLib version, I think).
2008-09-04gst-libs/gst/audio/gstaudiosrc.c: Disable a code path that is now called but ↵Wim Taymans1-1/+2
causes a deadlock for some reason and is... Original commit message from CVS: * gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_stop): Disable a code path that is now called but causes a deadlock for some reason and is unneeded.
2008-05-09gst-libs/gst/audio/gstaudiosink.c: Choose to allocate one less segment but ↵Wim Taymans1-3/+0
require one additional segment as latency. Original commit message from CVS: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_acquire): Choose to allocate one less segment but require one additional segment as latency. * gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_acquire): No need to increment the number of segments in the source. * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_get_time), (clock_convert_external), (gst_base_audio_sink_resample_slaving), (gst_base_audio_sink_skew_slaving), (gst_base_audio_sink_none_slaving), (gst_base_audio_sink_render), (gst_base_audio_sink_async_play): Remove adding latency when returning the internal time while subtracting it again when we use the value a little later. When calculating the end timestamp, we are making a rounding error with the current algorithm. Ensure that we don't accumulate these rounding errors when aligning samples by not resampling at all if we don't need to. Fixes #419351. Make the initial calibration of the clock slaving a little more predictable and accurate. Also handle the case where we don't do clock slaving.
2008-05-04Remove some unused code.Sebastian Dröge1-1/+1
Original commit message from CVS: * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func): * gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func): * gst/tcp/gsttcp.c: (gst_tcp_socket_write): * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_get_fps_list): Remove some unused code. * gst/audioconvert/gstaudioquantize.c: (gst_audio_quantize_free_noise_shaping): Don't return before freeing the noise shaping history.
2007-03-01gst-libs/gst/audio/: Improve debugging.Wim Taymans1-9/+10
Original commit message from CVS: * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func): * gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func): Improve debugging. * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_query), (gst_base_audio_sink_event), (gst_base_audio_sink_render), (gst_base_audio_sink_async_play): Improve latency and clock slaving calculations. Improve slave clock calibration. * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_commit_full): When we are asked to render N sample to 0 bytes, return N.
2006-11-06gst-libs/gst/audio/: Use g_strerror instead of strerror so we get UTF-8.Tim-Philipp Müller1-2/+2
Original commit message from CVS: * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func): * gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func): Use g_strerror instead of strerror so we get UTF-8.
2006-09-27Added docs for the audio libs.Wim Taymans1-0/+47
Original commit message from CVS: * docs/libs/gst-plugins-base-libs-docs.sgml: * docs/libs/gst-plugins-base-libs-sections.txt: * gst-libs/gst/audio/gstaudioclock.c: * gst-libs/gst/audio/gstaudioclock.h: * gst-libs/gst/audio/gstaudiosink.c: * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstaudiosrc.c: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_render): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init): * gst-libs/gst/audio/gstbaseaudiosrc.h: * gst-libs/gst/audio/gstringbuffer.h: Added docs for the audio libs.
2006-09-16ext/gnomevfs/gstgnomevfssrc.c: Add docs about icydemux usage in connection ↵Stefan Kost1-1/+1
with gnomevfssrc Original commit message from CVS: * ext/gnomevfs/gstgnomevfssrc.c: Add docs about icydemux usage in connection with gnomevfssrc * ext/libvisual/visual.c: * ext/ogg/gstoggaviparse.c: * ext/ogg/gstoggdemux.c: * ext/ogg/gstoggmux.c: * ext/ogg/gstoggparse.c: * gst-libs/gst/audio/gstaudiofiltertemplate.c: * gst-libs/gst/audio/gstaudiosink.c: * gst-libs/gst/audio/gstaudiosrc.c: * gst/audiorate/gstaudiorate.c: More G_OBJECT macro fixing. * gst/audiotestsrc/gstaudiotestsrc.h: Fix wrong info in header due to copy & paste
2006-04-28gst-libs/gst/audio/gstbaseaudiosrc.c: GstBaseAudioSrc must be live or it ↵Wim Taymans1-1/+0
does not work. Original commit message from CVS: * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init), (gst_base_audio_src_get_time), (gst_base_audio_src_create): GstBaseAudioSrc must be live or it does not work. * gst-libs/gst/audio/gstaudiosrc.c: (gst_audio_src_init): Don't set live to TRUE as this is the default in the parentclass.
2006-04-08Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)Stefan Kost1-1/+1
Original commit message from CVS: * ext/alsa/gstalsamixeroptions.c: (gst_alsa_mixer_options_class_init): * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_class_init): * ext/ogg/gstoggdemux.c: (gst_ogg_pad_class_init): * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init): * ext/ogg/gstoggparse.c: (gst_ogg_parse_class_init): * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_class_init): * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_class_init): * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_class_init): * gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_class_init): * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_class_init): * gst-libs/gst/interfaces/colorbalancechannel.c: (gst_color_balance_channel_class_init): * gst-libs/gst/interfaces/mixeroptions.c: (gst_mixer_options_class_init): * gst-libs/gst/interfaces/mixertrack.c: (gst_mixer_track_class_init): * gst-libs/gst/interfaces/tunerchannel.c: (gst_tuner_channel_class_init): * gst-libs/gst/interfaces/tunernorm.c: (gst_tuner_norm_class_init): * gst-libs/gst/netbuffer/gstnetbuffer.c: (gst_netbuffer_class_init): * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_class_init): * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init): * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init): * gst/playback/gstplaybin.c: (gst_play_bin_class_init): * gst/playback/gststreaminfo.c: (gst_stream_info_class_init): * gst/playback/gststreamselector.c: (gst_stream_selector_class_init): * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init): * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init): * sys/v4l/gstv4lcolorbalance.c: (gst_v4l_color_balance_channel_class_init): * sys/v4l/gstv4ljpegsrc.c: (gst_v4ljpegsrc_class_init): * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init): * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init): * sys/v4l/gstv4ltuner.c: (gst_v4l_tuner_channel_class_init), (gst_v4l_tuner_norm_class_init): * sys/ximage/ximagesink.c: (gst_ximagesink_class_init): * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init): * tests/old/testsuite/alsa/sinesrc.c: (sinesrc_class_init): Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
2006-01-10gst-libs/gst/audio/gstaudiosrc.c: Don't leak GCond in audio sources.Michael Smith1-0/+7
Original commit message from CVS: * gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_dispose): Don't leak GCond in audio sources.
2005-12-06expand tabsThomas Vander Stichele1-9/+9
Original commit message from CVS: expand tabs
2005-11-21*.*: Ran scripts/update-macros. Oh yes.Andy Wingo1-6/+6
Original commit message from CVS: 2005-11-21 Andy Wingo <wingo@pobox.com> * *.h: * *.c: Ran scripts/update-macros. Oh yes.
2005-08-28Updates for two-arg init from GST_BOILERPLATE_FULL.Andy Wingo1-3/+5
Original commit message from CVS: 2005-08-28 Andy Wingo <wingo@pobox.com> * Updates for two-arg init from GST_BOILERPLATE_FULL.
2005-08-22gst-libs/gst/audio/gstaudiosrc.*: Implement open_device and close_device in ↵Andy Wingo1-4/+58
the ring buffer, like gstaudiosink. Original commit message from CVS: 2005-08-22 Andy Wingo <wingo@pobox.com> * gst-libs/gst/audio/gstaudiosrc.h: * gst-libs/gst/audio/gstaudiosrc.c: Implement open_device and close_device in the ring buffer, like gstaudiosink. * ext/alsa/gstalsamixer.h: * ext/alsa/gstalsamixer.c: Not a GObject any more. Include a nifty macro to implement the interface without much code. Cleanups. * ext/alsa/gstalsasrc.h: * ext/alsa/gstalsasrc.c: Be a mixer. Open device and mixer in READY. * ext/alsa/Makefile.am: Add new files. * ext/alsa/gstalsamixerelement.c: * ext/alsa/gstalsamixerelement.c: Split element code out from mixer code so that alsasrc can be a mixer too.
2005-07-10use overridable ERROR_CFLAGS; more macro splittingThomas Vander Stichele1-51/+51
Original commit message from CVS: use overridable ERROR_CFLAGS; more macro splitting
2005-07-06Added audiosource base classes.Wim Taymans1-0/+418
Original commit message from CVS: Added audiosource base classes. Ported alsasrc, still very basic.
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