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gst/videorate/gstvideorate.c (gst_video_rate_reset)
Original commit message from CVS: 2006-04-06 Andy Wingo <wingo@pobox.com> * gst/videorate/gstvideorate.c (gst_video_rate_reset) (gst_video_rate_init): Caps-related parameters should not be reset by a flush -- move their inits to the instance init function. (gst_video_rate_flush_prev): Don't complain if gst_pad_push is not OK, just return the result. * gst/audiotestsrc/gstaudiotestsrc.c (gst_audio_test_src_class_init) (gst_audio_test_src_get_times): Re-enable is-live=true, as was broken by Stefan's commit on 24 March.
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3 changed files with 19 additions and 21 deletions
11
ChangeLog
11
ChangeLog
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@ -1,5 +1,16 @@
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2006-04-06 Andy Wingo <wingo@pobox.com>
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* gst/videorate/gstvideorate.c (gst_video_rate_reset)
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(gst_video_rate_init): Caps-related parameters should not be reset
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by a flush -- move their inits to the instance init function.
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(gst_video_rate_flush_prev): Don't complain if gst_pad_push
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is not OK, just return the result.
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* gst/audiotestsrc/gstaudiotestsrc.c
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(gst_audio_test_src_class_init)
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(gst_audio_test_src_get_times): Re-enable is-live=true, as was
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broken by Stefan's commit on 24 March.
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* ext/ogg/gstoggmux.c (gst_ogg_mux_push_buffer): Set caps on
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buffers being pushed out. Fixes oggmux ! multifdsink.
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@ -139,10 +139,8 @@ static gboolean gst_audio_test_src_query (GstBaseSrc * basesrc,
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static void gst_audio_test_src_change_wave (GstAudioTestSrc * src);
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/*
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static void gst_audio_test_src_get_times (GstBaseSrc * basesrc,
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GstBuffer * buffer, GstClockTime * start, GstClockTime * end);
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*/
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static GstFlowReturn gst_audio_test_src_create (GstBaseSrc * basesrc,
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guint64 offset, guint length, GstBuffer ** buffer);
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@ -196,10 +194,8 @@ gst_audio_test_src_class_init (GstAudioTestSrcClass * klass)
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GST_DEBUG_FUNCPTR (gst_audio_test_src_is_seekable);
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gstbasesrc_class->do_seek = GST_DEBUG_FUNCPTR (gst_audio_test_src_do_seek);
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gstbasesrc_class->query = GST_DEBUG_FUNCPTR (gst_audio_test_src_query);
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/*
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gstbasesrc_class->get_times =
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GST_DEBUG_FUNCPTR (gst_audio_test_src_get_times);
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*/
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gstbasesrc_class->get_times =
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GST_DEBUG_FUNCPTR (gst_audio_test_src_get_times);
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gstbasesrc_class->create = GST_DEBUG_FUNCPTR (gst_audio_test_src_create);
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}
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@ -579,7 +575,6 @@ gst_audio_test_src_change_volume (GstAudioTestSrc * src)
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}
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}
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#ifdef __DISABLE_NO_LIVE__
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static void
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gst_audio_test_src_get_times (GstBaseSrc * basesrc, GstBuffer * buffer,
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GstClockTime * start, GstClockTime * end)
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@ -602,7 +597,6 @@ gst_audio_test_src_get_times (GstBaseSrc * basesrc, GstBuffer * buffer,
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*end = -1;
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}
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}
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#endif
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static gboolean
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gst_audio_test_src_do_seek (GstBaseSrc * basesrc, GstSegment * segment)
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@ -360,10 +360,6 @@ gst_video_rate_reset (GstVideoRate * videorate)
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{
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GST_DEBUG ("resetting data");
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videorate->from_rate_numerator = 0;
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videorate->from_rate_denominator = 0;
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videorate->to_rate_numerator = 0;
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videorate->to_rate_denominator = 0;
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videorate->in = 0;
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videorate->out = 0;
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videorate->drop = 0;
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@ -395,6 +391,11 @@ gst_video_rate_init (GstVideoRate * videorate)
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gst_video_rate_reset (videorate);
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videorate->silent = DEFAULT_SILENT;
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videorate->new_pref = DEFAULT_NEW_PREF;
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videorate->from_rate_numerator = 0;
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videorate->from_rate_denominator = 0;
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videorate->to_rate_numerator = 0;
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videorate->to_rate_denominator = 0;
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}
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/* flush the oldest buffer */
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@ -433,8 +434,7 @@ gst_video_rate_flush_prev (GstVideoRate * videorate)
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"old is best, dup, pushing buffer outgoing ts %" GST_TIME_FORMAT,
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GST_TIME_ARGS (push_ts));
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if ((res = gst_pad_push (videorate->srcpad, outbuf)) != GST_FLOW_OK)
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goto push_error;
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res = gst_pad_push (videorate->srcpad, outbuf);
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return res;
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@ -444,13 +444,6 @@ eos_before_buffers:
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GST_INFO_OBJECT (videorate, "got EOS before any buffer was received");
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return GST_FLOW_OK;
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}
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/* ERRORS */
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push_error:
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{
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GST_WARNING_OBJECT (videorate, "couldn't push buffer on srcpad, reason %s",
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gst_flow_get_name (res));
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return res;
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}
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}
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static void
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