audiomixer: Port to GstAggregator

https://bugzilla.gnome.org/show_bug.cgi?id=737183

Co-Authored by: Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>

gst/audiomixer/Makefile.am

@@ -6,11 +6,15 @@ include $(top_srcdir)/common/orc.mak
 
 libgstaudiomixer_la_SOURCES = gstaudiomixer.c
 nodist_libgstaudiomixer_la_SOURCES = $(ORC_NODIST_SOURCES)
-libgstaudiomixer_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS) $(ORC_CFLAGS)
+libgstaudiomixer_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) \
+	$(GST_CFLAGS) $(ORC_CFLAGS) \
+	-I$(top_srcdir)/gst-libs \
+	-I$(top_builddir)/gst-libs
 libgstaudiomixer_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
 libgstaudiomixer_la_LIBADD =  \
 		$(GST_PLUGINS_BASE_LIBS) \
 	        -lgstaudio-@GST_API_VERSION@ \
+		$(top_builddir)/gst-libs/gst/base/libgstbadbase-$(GST_API_VERSION).la \
 		$(GST_BASE_LIBS) $(GST_LIBS) $(ORC_LIBS)
 libgstaudiomixer_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
 

gst/audiomixer/gstaudiomixer.h

@@ -25,7 +25,7 @@
 #define __GST_AUDIO_MIXER_H__
 
 #include <gst/gst.h>
-#include <gst/base/gstcollectpads.h>
+#include <gst/base/gstaggregator.h>
 #include <gst/audio/audio.h>
 
 G_BEGIN_DECLS
@@ -49,12 +49,7 @@ typedef struct _GstAudioMixerPadClass GstAudioMixerPadClass;
  * The audiomixer object structure.
  */
 struct _GstAudioMixer {
-  GstElement      element;
+  GstAggregator      element;
-
-  GstPad         *srcpad;
-  GstCollectPads *collect;
-  /* pad counter, used for creating unique request pads */
-  gint            padcount;
 
   /* the next are valid for both int and float */
   GstAudioInfo    info;
@@ -64,13 +59,9 @@ struct _GstAudioMixer {
   /* Buffer starting at offset containing block_size samples */
   GstBuffer      *current_buffer;
 
-  /* sink event handling */
-  GstSegment      segment;
-  volatile gboolean segment_pending;
-  volatile gboolean flush_stop_pending;
-
   /* current caps */
   GstCaps *current_caps;
+  gboolean send_caps;
 
   /* target caps (set via property) */
   GstCaps *filter_caps;
@@ -83,16 +74,10 @@ struct _GstAudioMixer {
 
   /* Size in samples that is output per buffer */
   guint blocksize;
-
-  /* Pending inline events */
-  GList *pending_events;
-  
-  gboolean send_stream_start;
-  gboolean send_caps;
 };
 
 struct _GstAudioMixerClass {
-  GstElementClass parent_class;
+  GstAggregatorClass parent_class;
 };
 
 GType    gst_audiomixer_get_type (void);
@@ -105,17 +90,30 @@ GType    gst_audiomixer_get_type (void);
 #define GST_AUDIO_MIXER_PAD_GET_CLASS(obj)  (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_AUDIO_MIXER_PAD,GstAudioMixerPadClass))
 
 struct _GstAudioMixerPad {
-  GstPad parent;
+  GstAggregatorPad parent;
 
   gdouble volume;
   gint volume_i32;
   gint volume_i16;
   gint volume_i8;
   gboolean mute;
+
+  /* < private > */
+  GstBuffer *buffer;            /* current buffer we're mixing,
+                                   for comparison with collect.buffer
+                                   to see if we need to update our
+                                   cached values. */
+  guint position, size;
+
+  guint64 output_offset;        /* Offset in output segment that
+                                   collect.pos refers to in the
+                                   current buffer. */
+
+  guint64 next_offset;          /* Next expected offset in the input segment */
 };
 
 struct _GstAudioMixerPadClass {
-  GstPadClass parent_class;
+  GstAggregatorPadClass parent_class;
 };
 
 GType gst_audiomixer_pad_get_type (void);

tests/check/elements/audiomixer.c

@@ -870,12 +870,35 @@ GST_END_TEST;
 
 
 static GstBuffer *handoff_buffer = NULL;
+static gboolean
+_quit (GMainLoop * ml)
+{
+  g_main_loop_quit (ml);
+
+  return G_SOURCE_REMOVE;
+}
+
 static void
 handoff_buffer_cb (GstElement * fakesink, GstBuffer * buffer, GstPad * pad,
-    gpointer user_data)
+    GstClockTime * wanted_end)
 {
-  GST_DEBUG ("got buffer %p", buffer);
+  GST_DEBUG ("got buffer -- SIZE: %ld -- %p DURATION is %" GST_TIME_FORMAT
+      " -- WANTED END %" GST_TIME_FORMAT, gst_buffer_get_size (buffer), buffer,
+      GST_TIME_ARGS (GST_BUFFER_PTS (buffer) + GST_BUFFER_DURATION (buffer)),
+      GST_TIME_ARGS (*wanted_end));
+
   gst_buffer_replace (&handoff_buffer, buffer);
+
+
+  /* Buffers we push in will be 'cut' into different smaller buffers,
+   * we make sure that the last chunck was pushes before we concider the buffer
+   * we pushed as being done */
+  if (main_loop && *wanted_end
+      && *wanted_end <=
+      GST_BUFFER_PTS (buffer) + GST_BUFFER_DURATION (buffer)) {
+    *wanted_end = 0;
+    g_idle_add ((GSourceFunc) _quit, main_loop);
+  }
 }
 
 /* check if clipping works as expected */
@@ -891,9 +914,14 @@ GST_START_TEST (test_clip)
   GstEvent *event;
   GstBuffer *buffer;
   GstCaps *caps;
+  GMainLoop *local_mainloop;
+  GstClockTime wanted_end = 0;
 
   GST_INFO ("preparing test");
 
+  local_mainloop = g_main_loop_new (NULL, FALSE);
+  main_loop = NULL;
+
   /* build pipeline */
   bin = gst_pipeline_new ("pipeline");
   bus = gst_element_get_bus (bin);
@@ -907,7 +935,8 @@ GST_START_TEST (test_clip)
   audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
   sink = gst_element_factory_make ("fakesink", "sink");
   g_object_set (sink, "signal-handoffs", TRUE, NULL);
-  g_signal_connect (sink, "handoff", (GCallback) handoff_buffer_cb, NULL);
+  g_signal_connect (sink, "handoff", (GCallback) handoff_buffer_cb,
+      &wanted_end);
   gst_bin_add_many (GST_BIN (bin), audiomixer, sink, NULL);
 
   res = gst_element_link (audiomixer, sink);
@@ -948,8 +977,19 @@ GST_START_TEST (test_clip)
   buffer = gst_buffer_new_and_alloc (44100);
   GST_BUFFER_TIMESTAMP (buffer) = 0;
   GST_BUFFER_DURATION (buffer) = 250 * GST_MSECOND;
-  GST_DEBUG ("pushing buffer %p", buffer);
+  GST_DEBUG ("pushing buffer %p END is %" GST_TIME_FORMAT,
+      buffer,
+      GST_TIME_ARGS (GST_BUFFER_PTS (buffer) + GST_BUFFER_DURATION (buffer)));
   ret = gst_pad_chain (sinkpad, buffer);
+  main_loop = local_mainloop;
+
+  /* The aggregation is done in a dedicated thread, so we can't
+   * not know when it is actually going to happen, so we just add\
+   * a 100 ms timeout to be able to then check that the aggregation
+   * did happen as we do not have much other choice.
+   */
+  g_timeout_add (100, (GSourceFunc) _quit, main_loop);
+  g_main_loop_run (main_loop);
   ck_assert_int_eq (ret, GST_FLOW_OK);
   fail_unless (handoff_buffer == NULL);
 
@@ -957,9 +997,16 @@ GST_START_TEST (test_clip)
   buffer = gst_buffer_new_and_alloc (44100);
   GST_BUFFER_TIMESTAMP (buffer) = 900 * GST_MSECOND;
   GST_BUFFER_DURATION (buffer) = 250 * GST_MSECOND;
-  GST_DEBUG ("pushing buffer %p", buffer);
+
+  wanted_end = 135 * GST_MSECOND;
+  GST_DEBUG ("pushing buffer %p START %" GST_TIME_FORMAT " -- DURATION is %"
+      GST_TIME_FORMAT, buffer, GST_TIME_ARGS (GST_BUFFER_PTS (buffer)),
+      GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)));
+
+  main_loop = local_mainloop;
   ret = gst_pad_chain (sinkpad, buffer);
   ck_assert_int_eq (ret, GST_FLOW_OK);
+  g_main_loop_run (main_loop);
   fail_unless (handoff_buffer != NULL);
   gst_buffer_replace (&handoff_buffer, NULL);
 
@@ -967,18 +1014,28 @@ GST_START_TEST (test_clip)
   buffer = gst_buffer_new_and_alloc (44100);
   GST_BUFFER_TIMESTAMP (buffer) = 1 * GST_SECOND;
   GST_BUFFER_DURATION (buffer) = 250 * GST_MSECOND;
-  GST_DEBUG ("pushing buffer %p", buffer);
+
+  wanted_end = 390 * GST_MSECOND;
+  GST_DEBUG ("pushing buffer %p END is %" GST_TIME_FORMAT,
+      buffer,
+      GST_TIME_ARGS (GST_BUFFER_PTS (buffer) + GST_BUFFER_DURATION (buffer)));
   ret = gst_pad_chain (sinkpad, buffer);
   ck_assert_int_eq (ret, GST_FLOW_OK);
+  g_main_loop_run (main_loop);
   fail_unless (handoff_buffer != NULL);
   gst_buffer_replace (&handoff_buffer, NULL);
+  fail_unless (handoff_buffer == NULL);
 
   /* should be clipped and ok */
   buffer = gst_buffer_new_and_alloc (44100);
   GST_BUFFER_TIMESTAMP (buffer) = 2 * GST_SECOND;
   GST_BUFFER_DURATION (buffer) = 250 * GST_MSECOND;
-  GST_DEBUG ("pushing buffer %p", buffer);
+  GST_DEBUG ("pushing buffer %p END is %" GST_TIME_FORMAT,
+      buffer,
+      GST_TIME_ARGS (GST_BUFFER_PTS (buffer) + GST_BUFFER_DURATION (buffer)));
   ret = gst_pad_chain (sinkpad, buffer);
+  g_timeout_add (100, (GSourceFunc) _quit, main_loop);
+  g_main_loop_run (main_loop);
   ck_assert_int_eq (ret, GST_FLOW_OK);
   fail_unless (handoff_buffer == NULL);
 
@@ -1245,9 +1302,11 @@ GST_START_TEST (test_flush_start_flush_stop)
   audiomixer_src = gst_element_get_static_pad (audiomixer, "src");
   fail_if (GST_PAD_IS_FLUSHING (audiomixer_src));
   gst_pad_send_event (sinkpad1, gst_event_new_flush_start ());
-  fail_unless (GST_PAD_IS_FLUSHING (audiomixer_src));
+  fail_if (GST_PAD_IS_FLUSHING (audiomixer_src));
+  fail_unless (GST_PAD_IS_FLUSHING (sinkpad1));
   gst_pad_send_event (sinkpad1, gst_event_new_flush_stop (TRUE));
   fail_if (GST_PAD_IS_FLUSHING (audiomixer_src));
+  fail_if (GST_PAD_IS_FLUSHING (sinkpad1));
   gst_object_unref (audiomixer_src);
 
   gst_element_release_request_pad (audiomixer, sinkpad1);