Commit graph

114 commits

Author SHA1 Message Date
Wim Taymans
0a4c1bc64c gst-libs/gst/audio/gstbaseaudiosink.c: Avoid holding the OBJECT_LOCK when calling ringbuffer functions that take the ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_change_state):
Avoid holding the OBJECT_LOCK when calling ringbuffer functions that
take the ringbuffer lock because rinbuffer lock > OBJECT_LOCK. We can do
this because the async_play method is deprecated and usually not called
anymore.
2009-01-05 17:13:13 +00:00
Wim Taymans
a579eba73d gst-libs/gst/audio/gstbaseaudiosink.c: Pause the write thread before deactivating and releasing the ringbuffer to avo...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_setcaps):
Pause the write thread before deactivating and releasing the ringbuffer
to avoid a deadlock when we do gapless playback with different sample
rates in playbin2.  Fixes #564929.
2008-12-20 12:45:03 +00:00
Sebastian Dröge
04d9ff9a24 gst-libs/gst/audio/: API: Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to the public API. This is needed for the C...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_slave_method_get_type),
(gst_base_audio_sink_class_init):
* gst-libs/gst/audio/gstbaseaudiosink.h:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_slave_method_get_type),
(gst_base_audio_src_class_init):
* gst-libs/gst/audio/gstbaseaudiosrc.h:
API: Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to the
public API. This is needed for the C++ bindings to be able
to use this base classes. Fixes bug #564200, #564206.
2008-12-13 06:57:09 +00:00
Wim Taymans
6983c1c85b gst-libs/gst/audio/gstbaseaudiosink.c: Really fix audiosink drain handling by keeping track of the running_time of th...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_drain), (gst_base_audio_sink_event),
(gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
(gst_base_audio_sink_change_state):
Really fix audiosink drain handling by keeping track of the running_time
of the last sample.
2008-11-25 10:32:49 +00:00
Stefan Kost
a8264f66c7 gst-libs/gst/audio/gstbaseaudiosink.c: Time is already in running_time. Remove base_time handling. Fixes audiosinks n...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
Time is already in running_time. Remove base_time handling. Fixes
audiosinks not draining and thus chopping some audio in the end.
2008-11-24 20:11:52 +00:00
Stefan Kost
7f937c99d4 gst-libs/gst/audio/gstbaseaudiosink.c: Add one log message to check for audio_drained. Sync one log message with the ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
Add one log message to check for audio_drained. Sync one log message
with the condition. Send EOS after draining audio in pull mode.
2008-11-24 12:56:54 +00:00
Wim Taymans
e701e64005 gst-libs/gst/audio/gstbaseaudiosink.c: Use gst_base_sink_do_preroll() to wait for PLAYING and before waiting for the ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
(gst_base_audio_sink_callback):
Use gst_base_sink_do_preroll() to wait for PLAYING and before waiting
for the latency to expire, fixes #559567.
2008-11-10 14:22:09 +00:00
Wim Taymans
6eed8ca285 gst-libs/gst/audio/gstaudiosink.c: Implement a separate activate functions to start monitoring the segments or, in pu...
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c:
(gst_audioringbuffer_class_init), (gst_audioringbuffer_acquire),
(gst_audioringbuffer_activate), (gst_audioringbuffer_release),
(gst_audioringbuffer_stop):
Implement a separate activate functions to start monitoring the segments
or, in pull mode, pulling in data.
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_init), (gst_base_audio_sink_dispose),
(gst_base_audio_sink_query_pad), (gst_base_audio_sink_query),
(gst_base_audio_sink_setcaps), (gst_base_audio_sink_callback),
(gst_base_audio_sink_activate_pull),
(gst_base_audio_sink_async_play),
(gst_base_audio_sink_change_state):
Implement pad and element convert query function.
Activate the ringbuffer.
Use the segment last_stop value as the offset to pull.
Use new basesink _do_preroll() method to preroll in the pulling thread.
Take appropriate locking in the pulling thread.
* gst-libs/gst/audio/gstringbuffer.h:
Update some docs.
2008-10-20 15:35:37 +00:00
Wim Taymans
d2f328f55b gst-libs/gst/audio/gstbaseaudiosink.c: Report latency even if we are not live instead of hiding it.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_query), (gst_base_audio_sink_skew_slaving),
(gst_base_audio_sink_render):
Report latency even if we are not live instead of hiding it.
Take ts-offset and render-delay of the basesink into account when
scheduling samples.
Rework the clipping code so that we can take the various offsets into
account and still do correct clipping.
2008-06-20 09:09:37 +00:00
Wim Taymans
35e4b75b86 gst-libs/gst/audio/gstaudioclock.*: Add method to inform the clock that the time starts from 0 again. We use this inf...
Original commit message from CVS:
* gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
(gst_audio_clock_reset), (gst_audio_clock_get_internal_time):
* gst-libs/gst/audio/gstaudioclock.h:
Add method to inform the clock that the time starts from 0 again. We use
this info to calculate a clock offset so that the time we report in
internal_time is monotonically increasing, as required by the clock base
class. Fixes #521761.
API: GstAudioClock::gst_audio_clock_reset()
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_skew_slaving),
(gst_base_audio_sink_change_state):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create), (gst_base_audio_src_change_state):
Reset reported time when we (re)create the ringbuffer.
2008-05-27 16:20:17 +00:00
Wim Taymans
f36d9d6b08 gst-libs/gst/audio/gstbaseaudiosink.c: We can only use our optimal calibration if we prerolled before the latency exp...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_sync_latency):
We can only use our optimal calibration if we prerolled before the
latency expired.
2008-05-20 16:26:53 +00:00
Wim Taymans
95d162fb71 gst-libs/gst/audio/gstbaseaudiosink.c: Change the way in which the ringbuffer is started when dealing with a slaved c...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_resample_slaving),
(gst_base_audio_sink_skew_slaving),
(gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
(gst_base_audio_sink_async_play),
(gst_base_audio_sink_change_state):
Change the way in which the ringbuffer is started when dealing with a
slaved clock and latency. We now sync to the clock until we reach
upstream latency before starting the ringbuffer. This has the effect
that we can accurately align the master and slave clocks and let the
rate correction code take care of the initial drift or rounding errors
instead of leaving them uncorrected with the old approach.
2008-05-20 11:09:06 +00:00
Wim Taymans
0c9b13988c gst-libs/gst/audio/gstbaseaudiosink.c: Revert previous patch that attempted to more accurately calculate the initial ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render), (gst_base_audio_sink_async_play):
Revert previous patch that attempted to more accurately calculate the
initial offset between master and slave clock. The best thing we can do
in general is take the time of both clocks as the diff since we don't
know when the actual preroll happened.
2008-05-12 08:45:11 +00:00
Wim Taymans
fc523e047c gst-libs/gst/audio/gstaudiosink.c: Choose to allocate one less segment but 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.
2008-05-09 16:38:10 +00:00
Wim Taymans
09f7dee84d gst-libs/gst/audio/gstbaseaudiosink.c: Report the latency with the new seglatency parameter.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_query):
Report the latency with the new seglatency parameter.
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_debug_spec_buff), (gst_ring_buffer_parse_caps),
(gst_ring_buffer_acquire):
* gst-libs/gst/audio/gstringbuffer.h:
Add new field to the ringbufferspec to specify the expected latency
between the underlying device read/write pointer, this is needed
when writing sinks that sit a little closer to the hardware.
Add some more docs for other fields.
2008-05-07 15:47:03 +00:00
Tim-Philipp Müller
7a29d716bd gst/: Work around missing bits of thread-safety on older GLibs some more to avoid assertions when starting up multipl...
Original commit message from CVS:
* 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/playback/gstplayback.c: (plugin_init):
* gst/volume/gstvolume.c: (plugin_init):
Work around missing bits of thread-safety on older GLibs some
more to avoid assertions when starting up multiple playbin
objects concurrently (see #512382).
2008-04-06 20:16:27 +00:00
Wim Taymans
ce67ac6373 gst-libs/gst/audio/gstbaseaudiosink.c: Guard against over and underflows because of clock slaving.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_skew_slaving), (gst_base_audio_sink_render):
Guard against over and underflows because of clock slaving.
When we are using our own clock, still compensate for any calibrations
that we might have done to our clock.
2008-04-03 10:37:03 +00:00
Wim Taymans
877a45b791 gst-libs/gst/audio/gstaudiosink.c: Small debug improvement.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_release):
Small debug improvement.
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
Fix bug in determining the sample start/stop position, we want to base
this decision on the fact that we are going forwards or backwards, not
slower or faster. This fixes some ugly resync warnings when playing at
very slow speeds.
2008-03-24 11:24:22 +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
Tim-Philipp Müller
3feb4bc8c5 gst-libs/gst/audio/: Ref audio clock class from a thread-safe context to make sure however unlikely that may be in pr...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init):
Ref audio clock class from a thread-safe context to make sure
we're not bit by GObjects lack of thread-safety here (#349410),
however unlikely that may be in practice.
2008-01-10 17:55:53 +00:00
Wim Taymans
ac1cc82165 gst-libs/gst/audio/gstbaseaudiosink.c: Our EOS time contains the base_time, _wait_eos() expects a running_time so we ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_drain):
Our EOS time contains the base_time, _wait_eos() expects a running_time
so we have to subtract the base_time again before calling the function.
This fixes an EOS regression where the base_time was added twice and EOS
took longer and longer in certain situations.
Fixes #498767.
2007-11-21 18:02:21 +00:00
Wim Taymans
157a65b15e Expose methods for some object properties so that subclasses can more easily configure them.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/gstbaseaudiosink.c: (slave_method_get_type),
(gst_base_audio_sink_set_provide_clock),
(gst_base_audio_sink_get_provide_clock),
(gst_base_audio_sink_set_slave_method),
(gst_base_audio_sink_get_slave_method),
(gst_base_audio_sink_set_property),
(gst_base_audio_sink_get_property), (gst_base_audio_sink_drain),
(gst_base_audio_sink_none_slaving),
(gst_base_audio_sink_handle_slaving):
* gst-libs/gst/audio/gstbaseaudiosink.h:
Expose methods for some object properties so that subclasses can more
easily configure them.
Added slave method none, that completely disables slaving to the
internal clock.
API: gst_base_audio_sink_set_provide_clock()
API: gst_base_audio_sink_get_provide_clock()
API: gst_base_audio_sink_set_slave_method()
API: gst_base_audio_sink_get_slave_method()
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_set_provide_clock),
(gst_base_audio_src_get_provide_clock),
(gst_base_audio_src_set_property),
(gst_base_audio_src_get_property), (gst_base_audio_src_create):
* gst-libs/gst/audio/gstbaseaudiosrc.h:
Expose methods for some object properties so that subclasses can more
easily configure them.
API: gst_base_audio_src_set_provide_clock()
API: gst_base_audio_src_get_provide_clock()
2007-11-21 13:04:17 +00:00
Wim Taymans
6a20747e83 gst-libs/gst/audio/gstbaseaudiosink.c: Also explicitly release the ringbuffer when going to NULL because it is requir...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_drain), (gst_base_audio_sink_change_state):
Also explicitly release the ringbuffer when going to NULL because it
is required in the setcaps function, before the state change to PAUSED
completes.
2007-10-16 15:33:31 +00:00
Wim Taymans
02f280a9a0 gst-libs/gst/audio/gstbaseaudiosink.c: Use new basesink method to make our EOS drain interruptable.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_drain):
Use new basesink method to make our EOS drain interruptable.
2007-10-10 15:36:56 +00:00
Wim Taymans
5ba1ed3a21 gst-libs/gst/audio/gstbaseaudiosink.c: When slaved to the clock, don't try to align a sample with the previous one wh...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_async_play):
When slaved to the clock, don't try to align a sample with the previous
one when going to PLAYING again.
2007-10-02 11:11:13 +00:00
Jan Schmidt
d5996e9c37 Fix a bunch of compile warnings shown with Forte.
Original commit message from CVS:
* ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
(gst_text_overlay_set_property):
* ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_ntp_to_unix),
(gst_rtcp_unix_to_ntp):
* gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_get_type):
* gst/playback/gstqueue2.c:
* tests/examples/seek/seek.c: (set_scale):
Fix a bunch of compile warnings shown with Forte.
* gst/audiorate/gstaudiorate.c:
Always pull in config.h before including any system headers.
2007-09-17 17:24:55 +00:00
Wim Taymans
4764e6044f gst-libs/gst/audio/gstbaseaudiosink.c: Disable pull mode scheduling, we're not ready for it yet and it subtly breaks ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_init):
Disable pull mode scheduling, we're not ready for it yet and it subtly
breaks a lot of things.
2007-09-13 22:52:09 +00:00
Wim Taymans
c2460052b3 gst-libs/gst/audio/gstbaseaudiosink.c: When skew slaving, try to hover around the middle of a segment so that we at m...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_skew_slaving), (gst_base_audio_sink_render),
(gst_base_audio_sink_change_state):
When skew slaving, try to hover around the middle of a segment so that
we at most drift by half a segment.
If we are aligning in the oposite direction of the clock skew, we don't
have to resync.
2007-09-03 19:17:33 +00:00
Tim-Philipp Müller
8a499651b9 gst-libs/gst/audio/gstbaseaudiosink.c: Quick hack to make audiosinks stop at EOS when operating in pull-mode; needs t...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_callback):
Quick hack to make audiosinks stop at EOS when operating in
pull-mode; needs to be fixed properly some day.
2007-07-08 13:07:38 +00:00
Andy Wingo
ae6fd1b3f2 gst-libs/gst/audio/gstbaseaudiosink.c
Original commit message from CVS:
2007-06-19  Andy Wingo  <wingo@pobox.com>

* gst-libs/gst/audio/gstbaseaudiosink.c
(gst_base_audio_sink_init): Enable pull-mode operation.
2007-06-19 19:13:04 +00:00
Wim Taymans
b2fdf703c9 gst-libs/gst/audio/gstbaseaudiosink.c: After an interrupt (PAUSED/flush) assume that the next sample should not be al...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
After an interrupt (PAUSED/flush) assume that the next sample should not
be aligned to the previous sample. Fixes #417992.
2007-05-24 16:22:23 +00:00
Stefan Kost
e7c3ddf3fc gst-libs/gst/audio/gstbaseaudiosink.c
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c
(gst_base_audio_sink_change_state):
Fix typo in comment.
* gst/playback/gstdecodebin.c (gst_decode_bin_class_init,
free_dynamics, pad_probe, close_pad_link, try_to_link_1,
get_our_ghost_pad, remove_element_chain, queue_underrun_cb,
close_link):
* gst/playback/gstplaybin.c (gst_play_bin_set_property,
gen_audio_element, remove_sinks, gst_play_bin_send_event_to_sink):
Remove trailing whitespaces in comments.
* gst/volume/Makefile.am:
Fix tabs.
2007-05-18 15:23:43 +00:00
Wim Taymans
b802dea831 gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to create invalid calibration parameters by making the internal time...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_skew_slaving):
Don't try to create invalid calibration parameters by making the
internal time go backwards, instead make external time go forward.
2007-04-05 15:44:40 +00:00
Wim Taymans
450030ebaf gst-libs/gst/audio/gstbaseaudiosink.*: Store private stuff in GstBaseAudioSinkPrivate.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c: (slave_method_get_type),
(gst_base_audio_sink_class_init), (gst_base_audio_sink_init),
(gst_base_audio_sink_query), (gst_base_audio_sink_get_time),
(gst_base_audio_sink_set_property),
(gst_base_audio_sink_get_property), (gst_base_audio_sink_event),
(clock_convert_external), (gst_base_audio_sink_resample_slaving),
(gst_base_audio_sink_skew_slaving),
(gst_base_audio_sink_handle_slaving), (gst_base_audio_sink_render),
(gst_base_audio_sink_async_play):
* gst-libs/gst/audio/gstbaseaudiosink.h:
Store private stuff in GstBaseAudioSinkPrivate.
Add configurable clock slaving modes property.
API:: GstBaseAudioSink::slave-method property
Some more latency reporting tweaks.
Added skew based clock slaving correction and make it the default until
the resampling method is more robust.
2007-03-28 14:50:47 +00:00
Sébastien Moutte
1596dd263c gst-libs/gst/audio/gstbaseaudiosink.c: Use gst_guint64_to_gdouble for conversion.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:(gst_base_audio_sink_render):
Use gst_guint64_to_gdouble for conversion.
* win32/MANIFEST:
Add new files to the win32 MANIFEST.
* win32/common/libgstaudio.def:
* win32/common/libgstpbutils.def:
Add new exported functions.
* win32/vs6/gst_plugins_base.dsw:
* win32/vs6/libgstdecodebin.dsp:
* win32/vs6/libgstplaybin.dsp:
Change the link to libgstpbutils.lib.
* win32/vs6/libgstdecodebin2.dsp:
Add a new project for decodebin2.
* win32/vs6/libgstpbutils.dsp:
Add a new project for pbutils.
2007-03-10 15:59:33 +00:00
Wim Taymans
a2a8b1b8ce gst-libs/gst/audio/gstbaseaudiosink.c: Fix regression that made GStreamer skip the first samples of audio.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_async_play):
Fix regression that made GStreamer skip the first samples of audio.
Fixes #414684.
2007-03-06 12:10:08 +00:00
Wim Taymans
5ee0a694a6 gst-libs/gst/audio/gstbaseaudiosink.c: base time is irrelevant here.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_async_play):
base time is irrelevant here.
2007-03-01 17:29:55 +00:00
Wim Taymans
85c7eeecc3 gst-libs/gst/audio/: Improve debugging.
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.
2007-03-01 17:01:43 +00:00
Wim Taymans
3c94c06c5a gst-libs/gst/audio/gstaudioclock.c: Fix clock name.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
(gst_audio_clock_new):
Fix clock name.
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_init), (gst_base_audio_sink_query):
* gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
(gst_base_audio_src_query), (gst_base_audio_src_get_offset),
(gst_base_audio_src_create):
Improve latency query code.
Use proper clock names.
2007-02-28 15:02:25 +00:00
Andy Wingo
d9b6796d91 gst-libs/gst/audio/gstbaseaudiosink.c
Original commit message from CVS:
2007-02-22  Andy Wingo  <wingo@pobox.com>

* gst-libs/gst/audio/gstbaseaudiosink.c
(gst_base_audio_sink_init): Disable pull-mode activation until we
figure out how to make audio sinks go to PLAYING.
2007-02-22 11:04:10 +00:00
Wim Taymans
a43d0f57eb gst-libs/gst/audio/gstbaseaudiosink.c: Answer latency query.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init), (gst_base_audio_sink_query),
(gst_base_audio_sink_render), (gst_base_audio_sink_callback),
(gst_base_audio_sink_async_play),
(gst_base_audio_sink_change_state):
Answer latency query.
Use configured latency when syncing.
Fix clock slaving.
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init), (gst_base_audio_src_dispose),
(gst_base_audio_src_query), (gst_base_audio_src_change_state):
Fix possible memleak.
Implement latency query.
Small cleanups.
2007-02-15 12:06:25 +00:00
Andy Wingo
451ff2f992 gst-libs/gst/audio/gstbaseaudiosink.c
Original commit message from CVS:
2007-02-05  Andy Wingo  <wingo@pobox.com>

* gst-libs/gst/audio/gstbaseaudiosink.c
(gst_base_audio_sink_callback): Update basesink->offset so that we
pull monotonically increasing offsets instead of, um, seeking back
to 0 each time. Fixes alsasrc ! alsasink!
2007-02-05 18:39:51 +00:00
Andy Wingo
d853b23819 gst-libs/gst/audio/gstbaseaudiosink.c
Original commit message from CVS:
2007-01-12  Andy Wingo  <wingo@pobox.com>

* gst-libs/gst/audio/gstbaseaudiosink.c
(gst_base_audio_sink_fixate): Implement, stolen from baseaudiosrc.
(gst_base_audio_sink_activate_pull): Remove the handwavey nego
stuff, as the base class handles this now. Actually tell the ring
buffer to start.
(gst_base_audio_sink_callback): Cast the ring buffer correctly.
How did this work before? Maybe I'm not as awesome a programmer as
I think.

* gst-libs/gst/audio/gstbaseaudiosrc.c
(gst_base_audio_src_fixate): Rework as a basesrc vmethod instead
of a pad function.
2007-01-12 21:19:35 +00:00
Andy Wingo
85aee8e273 gst-libs/gst/audio/gstbaseaudiosink.c (gst_base_audio_sink_class_init)
Original commit message from CVS:
2007-01-06  Andy Wingo  <wingo@pobox.com>

* gst-libs/gst/audio/gstbaseaudiosink.c
(gst_base_audio_sink_class_init)
(gst_base_audio_sink_init):
(gst_base_audio_sink_activate_pull): Add an activate_pull function
to baseaudiosink, and tell basesink that we can work in pull mode.
This way the ring buffer thread drives the pipeline directly, if
pull mode is possible. There is some lingering nastiness regarding
capsnego, however.
(gst_base_audio_sink_callback): Implement the callback to pull
data. This interface is a bit light, though -- it should get a
GstFlowReturn return value at least.
2007-01-06 17:28:40 +00:00
Wim Taymans
0990cbf274 gst-libs/gst/audio/gstbaseaudiosink.*: Make the clock sync code more accurate wrt resampling and playback at differen...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_event), (gst_base_audio_sink_render):
* gst-libs/gst/audio/gstbaseaudiosink.h:
Make the clock sync code more accurate wrt resampling and playback
at different rates.
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_commit_full), (gst_ring_buffer_commit):
* gst-libs/gst/audio/gstringbuffer.h:
Use better algorithm to interpolate sample rates.
2006-11-13 17:30:17 +00:00
Wim Taymans
1166abbc99 gst-libs/gst/audio/gstbaseaudiosink.*: Extract rate from the NEWSEGMENT event.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_event), (gst_base_audio_sink_render):
* gst-libs/gst/audio/gstbaseaudiosink.h:
Extract rate from the NEWSEGMENT event.
Use commit_full to also take rate adjustment into account when writing
samples to the ringbuffer.
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_commit_full), (gst_ring_buffer_commit),
(gst_ring_buffer_read):
* gst-libs/gst/audio/gstringbuffer.h:
Added _commit_full() to also take rate into account.
Use simple interpolation algorithm to resample audio.
API: gst_ring_buffer_commit_full()
* tests/examples/seek/scrubby.c: (speed_cb), (do_seek):
* tests/examples/seek/seek.c: (segment_done):
Don't try to seek with 0.0 rate, just pause instead.
Remove bogus debug line.
2006-10-18 13:42:49 +00:00
Ville Syrjala
9b139e41fb gst-libs/gst/audio/gstbaseaudiosink.c: Don't crash when ringbuffer is not yet created.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_setcaps):
Don't crash when ringbuffer is not yet created.
Patch by: Ville Syrjala <ville dot syrjala at movial dot fi>
Fixes #361634.
* gst/playback/gstplaybasebin.c: (new_decoded_pad_full):
* gst/playback/gststreamselector.c:
(gst_stream_selector_request_new_pad):
Activate pads befre adding them to running elements.
2006-10-13 14:15:42 +00:00
Tim-Philipp Müller
9e107d670a Printf format fixes.
Original commit message from CVS:
* ext/alsa/gstalsadeviceprobe.c:
(gst_alsa_device_property_probe_get_values):
* ext/alsa/gstalsasink.c: (set_hwparams):
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_elem_pad),
(gst_ogg_chain_new_stream), (gst_ogg_demux_read_chain):
* ext/ogg/gstoggmux.c: (gst_ogg_mux_send_headers),
(gst_ogg_mux_process_best_pad):
* ext/ogg/gstoggparse.c: (gst_ogg_parse_new_stream),
(gst_ogg_parse_chain):
* ext/ogg/gstogmparse.c: (gst_ogm_parse_stream_header):
* ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_setup),
(gst_vorbis_enc_buffer_check_discontinuous):
* ext/vorbis/vorbisparse.c: (vorbis_parse_src_query):
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_handle_track_seek):
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_push_full):
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
* gst/audioresample/resample.c: (resample_input_pushthrough):
* gst/playback/gstplaybasebin.c: (queue_out_of_data):
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_handle_clients):
* gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
(wavpack_type_find):
* gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_create):
* sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
(gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new):
* tests/check/elements/volume.c: (GST_START_TEST):
Printf format fixes.
2006-10-05 15:55:21 +00:00
Wim Taymans
9945d7a468 gst-libs/gst/audio/gstbaseaudiosink.c: When we have a timestamp, we can still perform clipping.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
When we have a timestamp, we can still perform clipping.
When we have no clock, we must play the sample ASAP.
2006-09-28 15:08:15 +00:00
Wim Taymans
1980f16731 gst-libs/gst/audio/gstbaseaudiosink.c: Add some more info in a WARNING.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
Add some more info in a WARNING.
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
Handle PAUSE in create function, use new -core addition to
wait for playing. Fixes pausing and resuming capture from an
audiosrc.
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_commit),
(gst_ring_buffer_read):
Constify some more.
Caller supports interrupted reads now.
2006-09-27 13:52:14 +00:00