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gst-libs/gst/audio/gstbaseaudiosink.c: Undo previous commit, it breaks resume after pause.
Original commit message from CVS: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_render), (gst_base_audio_sink_change_state): Undo previous commit, it breaks resume after pause.
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2 changed files with 8 additions and 7 deletions
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@ -1,3 +1,10 @@
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2006-01-25 Wim Taymans <wim@fluendo.com>
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* gst-libs/gst/audio/gstbaseaudiosink.c:
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(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
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Undo previous commit that returned WRONG_STATE sooner, it breaks
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resume after pause.
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2006-01-25 Wim Taymans <wim@fluendo.com>
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* gst-libs/gst/audio/gstbaseaudiosink.c:
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@ -536,8 +536,6 @@ no_sync:
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sink->next_sample = render_offset + samples;
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samples = gst_ring_buffer_commit (ringbuf, render_offset, data, samples);
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if (samples == -1)
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goto stopping;
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if (GST_CLOCK_TIME_IS_VALID (stop) && stop >= bsink->segment.stop) {
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GST_DEBUG_OBJECT (sink,
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@ -570,11 +568,6 @@ wrong_size:
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("sink received buffer of wrong size."));
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return GST_FLOW_ERROR;
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}
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stopping:
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{
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GST_DEBUG_OBJECT (sink, "ringbuffer is stopping");
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return GST_FLOW_WRONG_STATE;
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}
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}
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GstRingBuffer *
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@ -625,6 +618,7 @@ gst_base_audio_sink_change_state (GstElement * element,
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{
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GstClock *clock;
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/* FIXME, only start slaving when we really start the ringbuffer */
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GST_OBJECT_LOCK (sink);
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clock = GST_ELEMENT_CLOCK (sink);
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/* if we are slaved to a clock, we need to set the initial
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