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.

diff --git a/ChangeLog b/ChangeLog
index a95a79d5f7fe6b9d4bafc1bfade7c77c294ef4cd..35f529407fb4c58f5a16d84dc719b2b0d6fe11a3 100644 (file)
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,3 +1,18 @@
+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  Tim-Philipp Müller  <tim at centricular dot net>
 
        * gst-libs/gst/utils/missing-plugins.c: (copy_and_clean_caps):

diff --git a/gst-libs/gst/audio/gstbaseaudiosink.c b/gst-libs/gst/audio/gstbaseaudiosink.c
index 4eb8fea9712964bd1f2ddc8e3d1acf515f9c2b12..ce314021249577cfb817086991dfeee3c0689e8e 100644 (file)
--- a/gst-libs/gst/audio/gstbaseaudiosink.c
+++ b/gst-libs/gst/audio/gstbaseaudiosink.c
@@ -104,6 +104,7 @@ static void gst_base_audio_sink_get_times (GstBaseSink * bsink,
     GstBuffer * buffer, GstClockTime * start, GstClockTime * end);
 static gboolean gst_base_audio_sink_setcaps (GstBaseSink * bsink,
     GstCaps * caps);
+static void gst_base_audio_sink_fixate (GstBaseSink * bsink, GstCaps * caps);
 
 /* static guint gst_base_audio_sink_signals[LAST_SIGNAL] = { 0 }; */
 
@@ -155,6 +156,7 @@ gst_base_audio_sink_class_init (GstBaseAudioSinkClass * klass)
   gstbasesink_class->get_times =
       GST_DEBUG_FUNCPTR (gst_base_audio_sink_get_times);
   gstbasesink_class->set_caps = GST_DEBUG_FUNCPTR (gst_base_audio_sink_setcaps);
+  gstbasesink_class->fixate = GST_DEBUG_FUNCPTR (gst_base_audio_sink_fixate);
   gstbasesink_class->async_play =
       GST_DEBUG_FUNCPTR (gst_base_audio_sink_async_play);
   gstbasesink_class->activate_pull =
@@ -375,6 +377,32 @@ acquire_error:
   }
 }
 
+static void
+gst_base_audio_sink_fixate (GstBaseSink * bsink, GstCaps * caps)
+{
+  GstStructure *s;
+  gint width, depth;
+
+  s = gst_caps_get_structure (caps, 0);
+
+  /* fields for all formats */
+  gst_structure_fixate_field_nearest_int (s, "rate", 44100);
+  gst_structure_fixate_field_nearest_int (s, "channels", 2);
+  gst_structure_fixate_field_nearest_int (s, "width", 16);
+
+  /* fields for int */
+  if (gst_structure_has_field (s, "depth")) {
+    gst_structure_get_int (s, "width", &width);
+    /* round width to nearest multiple of 8 for the depth */
+    depth = GST_ROUND_UP_8 (width);
+    gst_structure_fixate_field_nearest_int (s, "depth", depth);
+  }
+  if (gst_structure_has_field (s, "signed"))
+    gst_structure_fixate_field_boolean (s, "signed", TRUE);
+  if (gst_structure_has_field (s, "endianness"))
+    gst_structure_fixate_field_nearest_int (s, "endianness", G_BYTE_ORDER);
+}
+
 static void
 gst_base_audio_sink_get_times (GstBaseSink * bsink, GstBuffer * buffer,
     GstClockTime * start, GstClockTime * end)
@@ -824,29 +852,9 @@ gst_base_audio_sink_activate_pull (GstBaseSink * basesink, gboolean active)
   GstBaseAudioSink *sink = GST_BASE_AUDIO_SINK (basesink);
 
   if (active) {
-    GstCaps *sinkcaps, *peercaps, *caps;
-
     gst_ring_buffer_set_callback (sink->ringbuffer,
         gst_base_audio_sink_callback, sink);
-
-    /* need to spawn a thread to start pulling. that's the ring buffer thread,
-       which is started in ring_buffer_acquire(), which is called due to a sink
-       setcaps(). So we need to setcaps, which is tough because we don't know
-       exactly what caps we'll be getting. We can guess, though, and that's as
-       good as we're going to get without the user telling us explicitly e.g.
-       via a capsfilter before the sink. */
-    sinkcaps = gst_pad_get_caps (basesink->sinkpad);
-    peercaps = gst_pad_peer_get_caps (basesink->sinkpad);
-    caps = gst_caps_intersect (sinkcaps, peercaps);
-    gst_caps_unref (sinkcaps);
-    gst_caps_unref (peercaps);
-    gst_caps_truncate (caps);
-    gst_pad_fixate_caps (basesink->sinkpad, caps);
-
-    GST_DEBUG_OBJECT (sink, "initializing pull-mode ringbuffer with caps %"
-        GST_PTR_FORMAT, caps);
-    ret = gst_pad_set_caps (basesink->sinkpad, caps);
-    gst_caps_unref (caps);
+    ret = gst_ring_buffer_start (sink->ringbuffer);
   } else {
     gst_ring_buffer_set_callback (sink->ringbuffer, NULL, NULL);
     /* stop thread */
@@ -865,8 +873,8 @@ gst_base_audio_sink_callback (GstRingBuffer * rbuf, guint8 * data, guint len,
   GstBuffer *buf;
   GstFlowReturn ret;
 
-  basesink = GST_BASE_SINK (data);
-  sink = GST_BASE_AUDIO_SINK (data);
+  basesink = GST_BASE_SINK (user_data);
+  sink = GST_BASE_AUDIO_SINK (user_data);
 
   /* would be nice to arrange for pad_alloc_buffer to return data -- as it is we
      will copy twice, once into data, once into DMA */

diff --git a/gst-libs/gst/audio/gstbaseaudiosrc.c b/gst-libs/gst/audio/gstbaseaudiosrc.c
index 7cc750c52d64767153bfc8c8cfe6c2cb58b711c1..992c5d61e07e846d465e9f9cf6d9204db0354b02 100644 (file)
--- a/gst-libs/gst/audio/gstbaseaudiosrc.c
+++ b/gst-libs/gst/audio/gstbaseaudiosrc.c
@@ -67,8 +67,6 @@ static void gst_base_audio_src_set_property (GObject * object, guint prop_id,
 static void gst_base_audio_src_get_property (GObject * object, guint prop_id,
     GValue * value, GParamSpec * pspec);
 
-static void gst_base_audio_src_fixate (GstPad * pad, GstCaps * caps);
-
 static GstStateChangeReturn gst_base_audio_src_change_state (GstElement *
     element, GstStateChange transition);
 
@@ -86,6 +84,7 @@ static gboolean gst_base_audio_src_event (GstBaseSrc * bsrc, GstEvent * event);
 static void gst_base_audio_src_get_times (GstBaseSrc * bsrc,
     GstBuffer * buffer, GstClockTime * start, GstClockTime * end);
 static gboolean gst_base_audio_src_setcaps (GstBaseSrc * bsrc, GstCaps * caps);
+static void gst_base_audio_src_fixate (GstBaseSrc * bsrc, GstCaps * caps);
 
 /* static guint gst_base_audio_src_signals[LAST_SIGNAL] = { 0 }; */
 
@@ -136,6 +135,7 @@ gst_base_audio_src_class_init (GstBaseAudioSrcClass * klass)
   gstbasesrc_class->create = GST_DEBUG_FUNCPTR (gst_base_audio_src_create);
   gstbasesrc_class->check_get_range =
       GST_DEBUG_FUNCPTR (gst_base_audio_src_check_get_range);
+  gstbasesrc_class->fixate = GST_DEBUG_FUNCPTR (gst_base_audio_src_fixate);
 }
 
 static void
@@ -151,9 +151,6 @@ gst_base_audio_src_init (GstBaseAudioSrc * baseaudiosrc,
   baseaudiosrc->clock = gst_audio_clock_new ("clock",
       (GstAudioClockGetTimeFunc) gst_base_audio_src_get_time, baseaudiosrc);
 
-  gst_pad_set_fixatecaps_function (GST_BASE_SRC_PAD (baseaudiosrc),
-      gst_base_audio_src_fixate);
-
   /* we are always a live source */
   gst_base_src_set_live (GST_BASE_SRC (baseaudiosrc), TRUE);
   /* we operate in time */
@@ -292,7 +289,7 @@ gst_base_audio_src_get_property (GObject * object, guint prop_id,
 }
 
 static void
-gst_base_audio_src_fixate (GstPad * pad, GstCaps * caps)
+gst_base_audio_src_fixate (GstBaseSrc * bsrc, GstCaps * caps)
 {
   GstStructure *s;
   gint width, depth;