Original commit message from CVS:
* sys/oss/gstosshelper.c:
Verify that the format returned after the ioctl is the one
we requested. It is valid for the ioctl to succeed while
substituting an alternate 'supported' sample format.
Original commit message from CVS:
* sys/oss/gstossaudio.c: (plugin_init):
* sys/oss/gstosssink.c: (gst_oss_sink_open):
* sys/oss/gstosssrc.c: (gst_oss_src_open):
Post decent (and translated) error message when we can't
open the audio device for some reason.
Original commit message from CVS:
* sys/oss/gstosssink.c:
* sys/oss/gstosssrc.c:
Allow the AUDIODEV environment variable to redirect us
to a different default OSS device, like sunaudiosink does
on Solaris (makes audio play automatically on SunRays).
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (bpwsinc_transform):
* gst/filter/gstlpwsinc.c: (lpwsinc_transform):
The transform() methods are not called in passthrough mode so
there's no need for checking if the element is in passthrough mode.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (bpwsinc_transform):
* gst/filter/gstlpwsinc.c: (lpwsinc_transform):
Sync the GObject properties with the controller even in passthrough
mode to get consistent property values.
Original commit message from CVS:
* gst/audiofx/audioamplify.c: (gst_audio_amplify_transform_ip):
* gst/audiofx/audiochebyshevfreqband.c:
(gst_audio_chebyshev_freq_band_transform_ip):
* gst/audiofx/audiochebyshevfreqlimit.c:
(gst_audio_chebyshev_freq_limit_transform_ip):
* gst/audiofx/audiodynamic.c: (gst_audio_dynamic_transform_ip):
* gst/audiofx/audioinvert.c: (gst_audio_invert_transform_ip):
The transform_ip() methods should do nothing if in passthrough mode.
It might get non-writable buffers in that case but the buffer might
as well be writable.
* gst/audiofx/audiopanorama.c: (gst_audio_panorama_transform):
The transform() methods won't be called in passthrough mode and
otherwise the buffer is always writable so don't check here.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_srcpad_event):
Fix seeking in .wav files again (#501775). Some people seem to think
they don't need to test their changes when they're just 'reflowing'
some code.
Original commit message from CVS:
* gst/autodetect/gstautovideosink.c:
(gst_auto_video_sink_class_init), (gst_auto_video_sink_dispose),
(gst_auto_video_sink_init),
(gst_auto_video_sink_create_element_with_pretty_name),
(gst_auto_video_sink_find_best),
(gst_auto_video_sink_set_property),
(gst_auto_video_sink_get_property):
* gst/autodetect/gstautovideosink.h:
Fix docs.
Use same error reporting code as autoaudiosink.
Add property to filter sinks based on caps. Only select raw video sinks
by default for backwards compat.
API: GstAutoVideoSink::filter-caps
Original commit message from CVS:
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
* gst/autodetect/gstautoaudiosink.c:
(gst_auto_audio_sink_class_init), (gst_auto_audio_sink_dispose),
(gst_auto_audio_sink_init), (gst_auto_audio_sink_find_best),
(gst_auto_audio_sink_set_property),
(gst_auto_audio_sink_get_property):
* gst/autodetect/gstautoaudiosink.h:
Add property to filter sinks based on caps. Only select raw audio sinks
by default for backwards compat. Fixes#417420.
API: GstAutoAudioSink::filter-caps
Original commit message from CVS:
Patch by: Wouter Cloetens <wouter at mind dot be>
* configure.ac:
Bump libsoup requirement as libsoup does not support async client
operation prior to version 2.2.104 and it has some leaks.
* ext/soup/gstsouphttpsrc.c: (gst_souphttp_src_class_init),
(gst_souphttp_src_init), (gst_souphttp_src_dispose),
(gst_souphttp_src_set_property), (gst_souphttp_src_create),
(gst_souphttp_src_start), (gst_souphttp_src_stop),
(gst_souphttp_src_unlock), (gst_souphttp_src_unlock_stop),
(gst_souphttp_src_get_size), (soup_got_headers), (soup_got_body),
(soup_finished), (soup_got_chunk), (soup_response),
(soup_session_close):
* ext/soup/gstsouphttpsrc.h:
Implement unlock().
Picks up the size from the Content-Length header and emit a duration
message.
Don't leak the GMainContext object.
Fixes#500099.
Original commit message from CVS:
* gst/rtp/gstrtph263depay.c: (gst_rtp_h263_depay_class_init),
(gst_rtp_h263_depay_process):
Code beautification.
Added debug statements.
Don't bit-shift everything, just do operations on last/first byte
instead.
Original commit message from CVS:
Patch by: Jayarama S. Santana <sundarsantana at gmail dot com>
* gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_process):
Fix wrong comparison in overrun check. Fixes#499239 some more.
Original commit message from CVS:
* gst/rtp/gstrtph263depay.c: (gst_rtp_h263_depay_init),
(gst_rtp_h263_depay_process):
* gst/rtp/gstrtph263depay.h:
Fix h263 depayloader so that ANY h263 decoder can handle the outgoing
stream.
Original commit message from CVS:
Based on Path by: Jayarama S. Santana <sundarsantana at gmail dot com>
* gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_setcaps),
(gst_rtp_mp4a_depay_process):
* gst/rtp/gstrtpmp4adepay.h:
Fix depayloading when multiple frames are inside one RTP packet.
Fixes#499239.
Original commit message from CVS:
* gst/rtp/gstrtph263depay.c: (gst_rtp_h263_depay_process):
Read the I flag for Mode A h263 rtp stream and set the
GST_BUFFER_FLAG_DELTA_UNIT accordingly.
Fixes#499383
Original commit message from CVS:
* tests/check/elements/wavpackenc.c: (GST_START_TEST):
Don't check the caps of the output buffer if they're equal some
other caps. The caps can change in a backward compatible way
and did at this point.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (bpwsinc_set_property):
* gst/filter/gstlpwsinc.c: (lpwsinc_set_property):
Post a GST_MESSAGE_LATENCY if the latency changes.
Original commit message from CVS:
* gst/equalizer/gstiirequalizer10bands.c:
* gst/equalizer/gstiirequalizer3bands.c:
Remove preset iface again. We'll re-add this after its been released
in -good.
Original commit message from CVS:
* ext/wavpack/gstwavpackcommon.c: (gst_wavpack_set_channel_layout):
Also set the channel layout on the Wavpack caps if we're having
a mono layout. Of course only do it for "audio/x-wavpack".
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler dot de>
* ext/libpng/gstpngenc.c:
Don't leak buffer data memory. Fixes#498395.
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler dot de>
* tests/check/pipelines/simple-launch-lines.c:
Tests for #498395.
Original commit message from CVS:
2007-11-20 Julien MOUTTE <julien@moutte.net>
* ext/taglib/gsttaglibmux.c: (gst_tag_lib_mux_render_tag),
(gst_tag_lib_mux_adjust_event_offsets):
* gst/qtdemux/qtdemux.c: (qtdemux_parse_theora_extension):
* sys/osxaudio/Makefile.am:
* sys/osxvideo/cocoawindow.h:
* sys/osxvideo/cocoawindow.m: Fix build on Mac OS X 10.5
Original commit message from CVS:
* gst/equalizer/gstiirequalizer10bands.c:
* gst/equalizer/gstiirequalizer3bands.c:
Activate preset iface and upload two presets here.
Original commit message from CVS:
Patch by: Jordi Jaen Pallares <jordijp at gmail dot com>
* gst/rtp/gstrtpmp2tpay.c: (gst_rtp_mp2t_pay_class_init),
(gst_rtp_mp2t_pay_init), (gst_rtp_mp2t_pay_finalize),
(gst_rtp_mp2t_pay_flush), (gst_rtp_mp2t_pay_handle_buffer):
* gst/rtp/gstrtpmp2tpay.h:
Fill the MTU with as many packets as possible. Fixes#491323.
Original commit message from CVS:
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_streams):
Fix some more leaks. Fixes#497007.
Original commit message from CVS:
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_free),
(gst_rtspsrc_stream_configure_tcp):
Fix 3 pad leaks. Fixes#496983.
Original commit message from CVS:
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_setcaps):
Fix small leak. Fixes#497017.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state),
(gst_qtdemux_prepare_current_sample),
(gst_qtdemux_loop_state_movie), (qtdemux_parse_theora_extension),
(qtdemux_parse_node), (qtdemux_parse_trak), (qtdemux_video_caps):
* gst/qtdemux/qtdemux_fourcc.h:
* gst/qtdemux/qtdemux_types.c:
Add suppport for theora in quicktime according to XiphQT.
Original commit message from CVS:
Always copy buffers by default (handle safer with bugged drivers) and added a property to make it possible to use mmap effectively (no copy if possible) when application wants to. Fixes: #480557.
Original commit message from CVS:
* gst/id3demux/id3tags.c:
* gst/id3demux/id3tags.h:
* gst/id3demux/id3v2frames.c: (id3v2_tag_to_taglist):
We don't want the same string multiple times in a tag list for the
same tag ever, for any tag, not just for GST_TAG_GENRE, so make sure
this doesn't happen and remove special-case code for GST_TAG_GENRE.
Original commit message from CVS:
* ext/taglib/gstid3v2mux.cc: (add_musicbrainz_tag), (add_funcs):
Write GST_TAG_MUSICBRAINZ_DISCID and GST_TAG_CDDA_CDDB_DISCID
into ID3v2 TXXX frames (fixes#347848).
Original commit message from CVS:
Patch by: Julien Puydt <julien dot puydt at laposte net>
* ext/raw1394/Makefile.am:
* ext/raw1394/gst1394probe.c: (gst_1394_get_guid_array),
(gst_1394_property_probe_get_properties),
(gst_1394_property_probe_probe_property),
(gst_1394_property_probe_needs_probe),
(gst_1394_property_probe_get_values),
(gst_1394_property_probe_interface_init),
(gst_1394_type_add_property_probe_interface):
* ext/raw1394/gst1394probe.h: (GST_1394_PROBE_H):
* ext/raw1394/gstdv1394src.c: (_do_init), (gst_dv1394src_class_init),
(gst_dv1394src_init), (gst_dv1394src_dispose),
(gst_dv1394src_set_property), (gst_dv1394src_get_property),
(gst_dv1394src_discover_avc_node), (gst_dv1394src_query),
(gst_dv1394src_update_device_name):
* ext/raw1394/gstdv1394src.h:
Implement GstPropertyProbe interface and add "device-name" property,
so applications can use this to probe for available devices in the
same way they can already with v4lsrc and v4l2src (however horrible
this property probe interface may be). Fixes#358841.
Original commit message from CVS:
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_handle_src_event),
(gst_rtspsrc_parse_range):
Don't leak event, don't leak range (fixes#496752).
