gst-libs/gst/audio/gstaudiosink.c: Implement pause that does not wait for completion.

Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c:
(gst_audioringbuffer_class_init), (gst_audioringbuffer_release),
(gst_audioringbuffer_pause):
Implement pause that does not wait for completion.

* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
Don't drop buffers when going to PAUSED but perform preroll on
remaining samples now that core base class supports this.

* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_release),
(gst_ring_buffer_pause_unlocked), (gst_ring_buffer_stop),
(gst_ring_buffer_commit):
Pause should not signal waiters.
Implement return value of _commit correctly.

ChangeLog

@@ -1,3 +1,21 @@
+2006-01-30  Wim Taymans  <wim@fluendo.com>
+
+	* gst-libs/gst/audio/gstaudiosink.c:
+	(gst_audioringbuffer_class_init), (gst_audioringbuffer_release),
+	(gst_audioringbuffer_pause):
+	Implement pause that does not wait for completion.
+
+	* gst-libs/gst/audio/gstbaseaudiosink.c:
+	(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
+	Don't drop buffers when going to PAUSED but perform preroll on
+	remaining samples now that core base class supports this.
+
+	* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_release),
+	(gst_ring_buffer_pause_unlocked), (gst_ring_buffer_stop),
+	(gst_ring_buffer_commit):
+	Pause should not signal waiters.
+	Implement return value of _commit correctly.
+
 2006-01-30  Andy Wingo  <wingo@pobox.com>
 
 	* tests/check/Makefile.am (check_vorbis): Add pipelines/vorbisenc.

common

@@ -1 +1 @@
-Subproject commit bc4325349e8d0ec90aa5c5e74566880cc2e82527
+Subproject commit 0b93085188e83fe678ec5ded2823cd7c24dfa843

gst-libs/gst/audio/gstaudiosink.c

@@ -77,6 +77,7 @@ static gboolean gst_audioringbuffer_acquire (GstRingBuffer * buf,
     GstRingBufferSpec * spec);
 static gboolean gst_audioringbuffer_release (GstRingBuffer * buf);
 static gboolean gst_audioringbuffer_start (GstRingBuffer * buf);
+static gboolean gst_audioringbuffer_pause (GstRingBuffer * buf);
 static gboolean gst_audioringbuffer_stop (GstRingBuffer * buf);
 static guint gst_audioringbuffer_delay (GstRingBuffer * buf);
 
@@ -132,7 +133,7 @@ gst_audioringbuffer_class_init (GstAudioRingBufferClass * klass)
   gstringbuffer_class->release =
       GST_DEBUG_FUNCPTR (gst_audioringbuffer_release);
   gstringbuffer_class->start = GST_DEBUG_FUNCPTR (gst_audioringbuffer_start);
-  gstringbuffer_class->pause = GST_DEBUG_FUNCPTR (gst_audioringbuffer_stop);
+  gstringbuffer_class->pause = GST_DEBUG_FUNCPTR (gst_audioringbuffer_pause);
   gstringbuffer_class->resume = GST_DEBUG_FUNCPTR (gst_audioringbuffer_start);
   gstringbuffer_class->stop = GST_DEBUG_FUNCPTR (gst_audioringbuffer_stop);
 
@@ -348,6 +349,7 @@ gst_audioringbuffer_release (GstRingBuffer * buf)
   abuf = GST_AUDIORING_BUFFER (buf);
 
   abuf->running = FALSE;
+  GST_DEBUG ("signal wait");
   GST_AUDIORING_BUFFER_SIGNAL (buf);
   GST_OBJECT_UNLOCK (buf);
 
@@ -388,6 +390,25 @@ gst_audioringbuffer_start (GstRingBuffer * buf)
   return TRUE;
 }
 
+static gboolean
+gst_audioringbuffer_pause (GstRingBuffer * buf)
+{
+  GstAudioSink *sink;
+  GstAudioSinkClass *csink;
+
+  sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf));
+  csink = GST_AUDIO_SINK_GET_CLASS (sink);
+
+  /* unblock any pending writes to the audio device */
+  if (csink->reset) {
+    GST_DEBUG ("reset...");
+    csink->reset (sink);
+    GST_DEBUG ("reset done");
+  }
+
+  return TRUE;
+}
+
 static gboolean
 gst_audioringbuffer_stop (GstRingBuffer * buf)
 {

gst-libs/gst/audio/gstbaseaudiosink.c

@@ -406,7 +406,7 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
   gint64 diff, ctime, cstop;
   guint8 *data;
   guint size;
-  guint samples;
+  guint samples, written;
   gint bps;
   gdouble crate = 1.0;
   GstClockTime crate_num;
@@ -458,6 +458,8 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
    * arriving before the segment.start or after segment.stop are to be 
    * thrown away. All samples should also be clipped to the segment 
    * boundaries */
+  /* let's calc stop based on the number of samples in the buffer instead
+   * of trusting the DURATION */
   stop =
       time + gst_util_uint64_scale_int (samples, GST_SECOND,
       ringbuf->spec.rate);
@@ -536,7 +538,27 @@ no_sync:
   /* the next sample should be current sample and its length */
   sink->next_sample = render_offset + samples;
 
-  samples = gst_ring_buffer_commit (ringbuf, render_offset, data, samples);
+  do {
+    written = gst_ring_buffer_commit (ringbuf, render_offset, data, samples);
+    GST_DEBUG_OBJECT (sink, "wrote %u of %u", written, samples);
+    /* if we wrote all, we're done */
+    if (written == samples)
+      break;
+
+    /* else something interrupted us */
+    GST_DEBUG_OBJECT (sink, "wait for preroll...");
+    bsink->have_preroll = TRUE;
+    GST_PAD_PREROLL_WAIT (bsink->sinkpad);
+    bsink->have_preroll = FALSE;
+    GST_DEBUG_OBJECT (sink, "preroll done");
+    if (G_UNLIKELY (bsink->flushing))
+      goto stopping;
+    GST_DEBUG_OBJECT (sink, "continue after preroll");
+
+    render_offset += written;
+    samples -= written;
+    data += written * bps;
+  } while (TRUE);
 
   if (GST_CLOCK_TIME_IS_VALID (stop) && stop >= bsink->segment.stop) {
     GST_DEBUG_OBJECT (sink,
@@ -569,6 +591,11 @@ wrong_size:
         ("sink received buffer of wrong size."));
     return GST_FLOW_ERROR;
   }
+stopping:
+  {
+    GST_DEBUG_OBJECT (sink, "ringbuffer is stopping");
+    return GST_FLOW_WRONG_STATE;
+  }
 }
 
 GstRingBuffer *
@@ -644,6 +671,9 @@ gst_base_audio_sink_change_state (GstElement * element,
       GST_OBJECT_UNLOCK (sink);
       break;
     }
+    case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
+      gst_ring_buffer_pause (sink->ringbuffer);
+      break;
     case GST_STATE_CHANGE_PAUSED_TO_READY:
       gst_ring_buffer_set_flushing (sink->ringbuffer, TRUE);
       break;
@@ -655,7 +685,6 @@ gst_base_audio_sink_change_state (GstElement * element,
 
   switch (transition) {
     case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
-      gst_ring_buffer_pause (sink->ringbuffer);
       /* slop slaving ourselves to the master, if any */
       gst_clock_set_master (sink->provided_clock, NULL);
       break;

gst-libs/gst/audio/gstringbuffer.c

@@ -654,6 +654,7 @@ gst_ring_buffer_release (GstRingBuffer * buf)
     res = rclass->release (buf);
 
   /* signal any waiters */
+  GST_DEBUG_OBJECT (buf, "signal waiter");
   GST_RING_BUFFER_SIGNAL (buf);
 
   if (!res)
@@ -820,9 +821,6 @@ gst_ring_buffer_pause_unlocked (GstRingBuffer * buf)
     goto done;
   }
 
-  /* signal any waiters */
-  GST_RING_BUFFER_SIGNAL (buf);
-
   rclass = GST_RING_BUFFER_GET_CLASS (buf);
   if (rclass->pause)
     res = rclass->pause (buf);
@@ -906,6 +904,7 @@ gst_ring_buffer_stop (GstRingBuffer * buf)
   }
 
   /* signal any waiters */
+  GST_DEBUG_OBJECT (buf, "signal waiter");
   GST_RING_BUFFER_SIGNAL (buf);
 
   rclass = GST_RING_BUFFER_GET_CLASS (buf);
@@ -1127,6 +1126,7 @@ gst_ring_buffer_commit (GstRingBuffer * buf, guint64 sample, guchar * data,
   gint segdone;
   gint segsize, segtotal, bps, sps;
   guint8 *dest;
+  guint to_write;
 
   g_return_val_if_fail (buf != NULL, -1);
   g_return_val_if_fail (buf->data != NULL, -1);
@@ -1138,8 +1138,9 @@ gst_ring_buffer_commit (GstRingBuffer * buf, guint64 sample, guchar * data,
   bps = buf->spec.bytes_per_sample;
   sps = buf->samples_per_seg;
 
+  to_write = len;
   /* write out all samples */
-  while (len > 0) {
+  while (to_write > 0) {
     gint sampleslen;
     gint writeseg, sampleoff;
 
@@ -1158,14 +1159,14 @@ gst_ring_buffer_commit (GstRingBuffer * buf, guint64 sample, guchar * data,
       diff = writeseg - segdone;
 
       GST_DEBUG
-          ("pointer at %d, sample %llu, write to %d-%d, len %d, diff %d, segtotal %d, segsize %d",
+          ("pointer at %d, sample %llu, write to %d-%d, to_write %d, diff %d, segtotal %d, segsize %d",
-          segdone, sample, writeseg, sampleoff, len, diff, segtotal, sps);
+          segdone, sample, writeseg, sampleoff, to_write, diff, segtotal, sps);
 
       /* segment too far ahead, we need to drop */
       if (diff < 0) {
         /* we need to drop one segment at a time, pretend we wrote a
          * segment. */
-        sampleslen = MIN (sps, len);
+        sampleslen = MIN (sps, to_write);
         goto next;
       }
 
@@ -1181,27 +1182,27 @@ gst_ring_buffer_commit (GstRingBuffer * buf, guint64 sample, guchar * data,
 
     /* we can write now */
     writeseg = writeseg % segtotal;
-    sampleslen = MIN (sps - sampleoff, len);
+    sampleslen = MIN (sps - sampleoff, to_write);
 
-    GST_DEBUG_OBJECT (buf, "write @%p seg %d, off %d, len %d",
+    GST_DEBUG_OBJECT (buf, "write @%p seg %d, off %d, sampleslen %d",
         dest + writeseg * segsize, writeseg, sampleoff, sampleslen);
 
     memcpy (dest + (writeseg * segsize) + (sampleoff * bps), data,
         (sampleslen * bps));
 
   next:
-    len -= sampleslen;
+    to_write -= sampleslen;
     sample += sampleslen;
     data += sampleslen * bps;
   }
 
-  return len;
+  return len - to_write;
 
   /* ERRORS */
 not_started:
   {
     GST_DEBUG_OBJECT (buf, "stopped processing");
-    return -1;
+    return len - to_write;
   }
 }