inter: new intermediate surface plugin

This set of elements allows easily rendering audio and video to
an intermediate surface that is then used as a source in a different
pipeline.

1 files changed, 481 insertions, 0 deletions

diff --git a/gst/inter/gstinteraudiosrc.c b/gst/inter/gstinteraudiosrc.c
new file mode 100644
index 000000000..df7c16f70
--- /dev/null
+++ b/gst/inter/gstinteraudiosrc.c
@@ -0,0 +1,481 @@
+/* GStreamer
+ * Copyright (C) 2011 David A. Schleef <ds@schleef.org>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin Street, Suite 500,
+ * Boston, MA 02110-1335, USA.
+ */
+/**
+ * SECTION:element-gstinteraudiosrc
+ *
+ * The interaudiosrc element does FIXME stuff.
+ *
+ * <refsect2>
+ * <title>Example launch line</title>
+ * |[
+ * gst-launch -v fakesrc ! interaudiosrc ! FIXME ! fakesink
+ * ]|
+ * FIXME Describe what the pipeline does.
+ * </refsect2>
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <gst/gst.h>
+#include <gst/base/gstbasesrc.h>
+#include "gstinteraudiosrc.h"
+#include <string.h>
+
+GST_DEBUG_CATEGORY_STATIC (gst_inter_audio_src_debug_category);
+#define GST_CAT_DEFAULT gst_inter_audio_src_debug_category
+
+/* prototypes */
+
+
+static void gst_inter_audio_src_set_property (GObject * object,
+    guint property_id, const GValue * value, GParamSpec * pspec);
+static void gst_inter_audio_src_get_property (GObject * object,
+    guint property_id, GValue * value, GParamSpec * pspec);
+static void gst_inter_audio_src_dispose (GObject * object);
+static void gst_inter_audio_src_finalize (GObject * object);
+
+static GstCaps *gst_inter_audio_src_get_caps (GstBaseSrc * src);
+static gboolean gst_inter_audio_src_set_caps (GstBaseSrc * src, GstCaps * caps);
+static gboolean gst_inter_audio_src_negotiate (GstBaseSrc * src);
+static gboolean gst_inter_audio_src_newsegment (GstBaseSrc * src);
+static gboolean gst_inter_audio_src_start (GstBaseSrc * src);
+static gboolean gst_inter_audio_src_stop (GstBaseSrc * src);
+static void
+gst_inter_audio_src_get_times (GstBaseSrc * src, GstBuffer * buffer,
+    GstClockTime * start, GstClockTime * end);
+static gboolean gst_inter_audio_src_is_seekable (GstBaseSrc * src);
+static gboolean gst_inter_audio_src_unlock (GstBaseSrc * src);
+static gboolean gst_inter_audio_src_event (GstBaseSrc * src, GstEvent * event);
+static GstFlowReturn
+gst_inter_audio_src_create (GstBaseSrc * src, guint64 offset, guint size,
+    GstBuffer ** buf);
+static gboolean gst_inter_audio_src_do_seek (GstBaseSrc * src,
+    GstSegment * segment);
+static gboolean gst_inter_audio_src_query (GstBaseSrc * src, GstQuery * query);
+static gboolean gst_inter_audio_src_check_get_range (GstBaseSrc * src);
+static void gst_inter_audio_src_fixate (GstBaseSrc * src, GstCaps * caps);
+static gboolean gst_inter_audio_src_unlock_stop (GstBaseSrc * src);
+static gboolean
+gst_inter_audio_src_prepare_seek_segment (GstBaseSrc * src, GstEvent * seek,
+    GstSegment * segment);
+
+enum
+{
+  PROP_0
+};
+
+/* pad templates */
+
+static GstStaticPadTemplate gst_inter_audio_src_src_template =
+GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-raw-int, "
+        "endianness = (int) BYTE_ORDER, "
+        "signed = (boolean) true, "
+        "width = (int) 16, "
+        "depth = (int) 16, "
+        "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]")
+    );
+
+
+/* class initialization */
+
+#define DEBUG_INIT(bla) \
+  GST_DEBUG_CATEGORY_INIT (gst_inter_audio_src_debug_category, "interaudiosrc", 0, \
+      "debug category for interaudiosrc element");
+
+GST_BOILERPLATE_FULL (GstInterAudioSrc, gst_inter_audio_src, GstBaseSrc,
+    GST_TYPE_BASE_SRC, DEBUG_INIT);
+
+static void
+gst_inter_audio_src_base_init (gpointer g_class)
+{
+  GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
+
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&gst_inter_audio_src_src_template));
+
+  gst_element_class_set_details_simple (element_class, "FIXME Long name",
+      "Generic", "FIXME Description", "FIXME <fixme@example.com>");
+}
+
+static void
+gst_inter_audio_src_class_init (GstInterAudioSrcClass * klass)
+{
+  GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
+  GstBaseSrcClass *base_src_class = GST_BASE_SRC_CLASS (klass);
+
+  gobject_class->set_property = gst_inter_audio_src_set_property;
+  gobject_class->get_property = gst_inter_audio_src_get_property;
+  gobject_class->dispose = gst_inter_audio_src_dispose;
+  gobject_class->finalize = gst_inter_audio_src_finalize;
+  base_src_class->get_caps = GST_DEBUG_FUNCPTR (gst_inter_audio_src_get_caps);
+  base_src_class->set_caps = GST_DEBUG_FUNCPTR (gst_inter_audio_src_set_caps);
+  if (0)
+    base_src_class->negotiate =
+        GST_DEBUG_FUNCPTR (gst_inter_audio_src_negotiate);
+  base_src_class->newsegment =
+      GST_DEBUG_FUNCPTR (gst_inter_audio_src_newsegment);
+  base_src_class->start = GST_DEBUG_FUNCPTR (gst_inter_audio_src_start);
+  base_src_class->stop = GST_DEBUG_FUNCPTR (gst_inter_audio_src_stop);
+  base_src_class->get_times = GST_DEBUG_FUNCPTR (gst_inter_audio_src_get_times);
+  if (0)
+    base_src_class->is_seekable =
+        GST_DEBUG_FUNCPTR (gst_inter_audio_src_is_seekable);
+  base_src_class->unlock = GST_DEBUG_FUNCPTR (gst_inter_audio_src_unlock);
+  base_src_class->event = GST_DEBUG_FUNCPTR (gst_inter_audio_src_event);
+  base_src_class->create = GST_DEBUG_FUNCPTR (gst_inter_audio_src_create);
+  if (0)
+    base_src_class->do_seek = GST_DEBUG_FUNCPTR (gst_inter_audio_src_do_seek);
+  base_src_class->query = GST_DEBUG_FUNCPTR (gst_inter_audio_src_query);
+  if (0)
+    base_src_class->check_get_range =
+        GST_DEBUG_FUNCPTR (gst_inter_audio_src_check_get_range);
+  base_src_class->fixate = GST_DEBUG_FUNCPTR (gst_inter_audio_src_fixate);
+  if (0)
+    base_src_class->unlock_stop =
+        GST_DEBUG_FUNCPTR (gst_inter_audio_src_unlock_stop);
+  if (0)
+    base_src_class->prepare_seek_segment =
+        GST_DEBUG_FUNCPTR (gst_inter_audio_src_prepare_seek_segment);
+
+
+}
+
+static void
+gst_inter_audio_src_init (GstInterAudioSrc * interaudiosrc,
+    GstInterAudioSrcClass * interaudiosrc_class)
+{
+
+  interaudiosrc->srcpad =
+      gst_pad_new_from_static_template (&gst_inter_audio_src_src_template,
+      "src");
+
+  gst_base_src_set_live (GST_BASE_SRC (interaudiosrc), TRUE);
+  gst_base_src_set_blocksize (GST_BASE_SRC (interaudiosrc), -1);
+
+  interaudiosrc->surface = gst_inter_surface_get ("default");
+}
+
+void
+gst_inter_audio_src_set_property (GObject * object, guint property_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  /* GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (object); */
+
+  switch (property_id) {
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
+      break;
+  }
+}
+
+void
+gst_inter_audio_src_get_property (GObject * object, guint property_id,
+    GValue * value, GParamSpec * pspec)
+{
+  /* GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (object); */
+
+  switch (property_id) {
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
+      break;
+  }
+}
+
+void
+gst_inter_audio_src_dispose (GObject * object)
+{
+  /* GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (object); */
+
+  /* clean up as possible.  may be called multiple times */
+
+  G_OBJECT_CLASS (parent_class)->dispose (object);
+}
+
+void
+gst_inter_audio_src_finalize (GObject * object)
+{
+  /* GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (object); */
+
+  /* clean up object here */
+
+  G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+
+static GstCaps *
+gst_inter_audio_src_get_caps (GstBaseSrc * src)
+{
+  GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
+
+  GST_DEBUG_OBJECT (interaudiosrc, "get_caps");
+
+  return NULL;
+}
+
+static gboolean
+gst_inter_audio_src_set_caps (GstBaseSrc * src, GstCaps * caps)
+{
+  GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
+  const GstStructure *structure;
+  gboolean ret;
+  int sample_rate;
+
+  GST_DEBUG_OBJECT (interaudiosrc, "set_caps");
+
+  structure = gst_caps_get_structure (caps, 0);
+
+  ret = gst_structure_get_int (structure, "rate", &sample_rate);
+  if (ret) {
+    interaudiosrc->sample_rate = sample_rate;
+  }
+
+  return ret;
+}
+
+static gboolean
+gst_inter_audio_src_negotiate (GstBaseSrc * src)
+{
+  GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
+
+  GST_DEBUG_OBJECT (interaudiosrc, "negotiate");
+
+  return TRUE;
+}
+
+static gboolean
+gst_inter_audio_src_newsegment (GstBaseSrc * src)
+{
+  GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
+
+  GST_DEBUG_OBJECT (interaudiosrc, "newsegment");
+
+  return TRUE;
+}
+
+static gboolean
+gst_inter_audio_src_start (GstBaseSrc * src)
+{
+  GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
+
+  GST_DEBUG_OBJECT (interaudiosrc, "start");
+
+  return TRUE;
+}
+
+static gboolean
+gst_inter_audio_src_stop (GstBaseSrc * src)
+{
+  GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
+
+  GST_DEBUG_OBJECT (interaudiosrc, "stop");
+
+  return TRUE;
+}
+
+static void
+gst_inter_audio_src_get_times (GstBaseSrc * src, GstBuffer * buffer,
+    GstClockTime * start, GstClockTime * end)
+{
+  GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
+
+  GST_DEBUG_OBJECT (interaudiosrc, "get_times");
+
+  /* for live sources, sync on the timestamp of the buffer */
+  if (gst_base_src_is_live (src)) {
+    GstClockTime timestamp = GST_BUFFER_TIMESTAMP (buffer);
+
+    if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
+      /* get duration to calculate end time */
+      GstClockTime duration = GST_BUFFER_DURATION (buffer);
+
+      if (GST_CLOCK_TIME_IS_VALID (duration)) {
+        *end = timestamp + duration;
+      }
+      *start = timestamp;
+    }
+  } else {
+    *start = -1;
+    *end = -1;
+  }
+}
+
+static gboolean
+gst_inter_audio_src_is_seekable (GstBaseSrc * src)
+{
+  GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
+
+  GST_DEBUG_OBJECT (interaudiosrc, "is_seekable");
+
+  return FALSE;
+}
+
+static gboolean
+gst_inter_audio_src_unlock (GstBaseSrc * src)
+{
+  GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
+
+  GST_DEBUG_OBJECT (interaudiosrc, "unlock");
+
+  return TRUE;
+}
+
+static gboolean
+gst_inter_audio_src_event (GstBaseSrc * src, GstEvent * event)
+{
+  GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
+
+  GST_DEBUG_OBJECT (interaudiosrc, "event");
+
+  return TRUE;
+}
+
+static GstFlowReturn
+gst_inter_audio_src_create (GstBaseSrc * src, guint64 offset, guint size,
+    GstBuffer ** buf)
+{
+  GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
+  GstBuffer *buffer;
+  int n;
+
+  GST_DEBUG_OBJECT (interaudiosrc, "create");
+
+  buffer = NULL;
+
+  g_mutex_lock (interaudiosrc->surface->mutex);
+  n = gst_adapter_available (interaudiosrc->surface->audio_adapter) / 4;
+  if (n > 1600 * 2) {
+    GST_DEBUG ("flushing %d samples", 800);
+    gst_adapter_flush (interaudiosrc->surface->audio_adapter, 800 * 4);
+    n -= 800;
+  }
+  if (n > 1600)
+    n = 1600;
+  if (n > 0) {
+    buffer = gst_adapter_take_buffer (interaudiosrc->surface->audio_adapter,
+        n * 4);
+  }
+  g_mutex_unlock (interaudiosrc->surface->mutex);
+
+  if (n < 1600) {
+    GstBuffer *newbuf = gst_buffer_new_and_alloc (1600 * 4);
+
+    GST_DEBUG ("creating %d samples of silence", 1600 - n);
+    memset (GST_BUFFER_DATA (newbuf) + n * 4, 0, 1600 * 4 - n * 4);
+    if (buffer) {
+      memcpy (GST_BUFFER_DATA (newbuf), GST_BUFFER_DATA (buffer), n * 4);
+      gst_buffer_unref (buffer);
+    }
+    buffer = newbuf;
+  }
+  n = 1600;
+
+  GST_BUFFER_OFFSET (buffer) = interaudiosrc->n_samples;
+  GST_BUFFER_OFFSET_END (buffer) = interaudiosrc->n_samples + n;
+  GST_BUFFER_TIMESTAMP (buffer) =
+      gst_util_uint64_scale_int (interaudiosrc->n_samples, GST_SECOND,
+      interaudiosrc->sample_rate);
+  GST_DEBUG_OBJECT (interaudiosrc, "create ts %" GST_TIME_FORMAT,
+      GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
+  GST_BUFFER_DURATION (buffer) =
+      gst_util_uint64_scale_int (interaudiosrc->n_samples + n, GST_SECOND,
+      interaudiosrc->sample_rate) - GST_BUFFER_TIMESTAMP (buffer);
+  GST_BUFFER_OFFSET (buffer) = interaudiosrc->n_samples;
+  GST_BUFFER_OFFSET_END (buffer) = -1;
+  GST_BUFFER_FLAG_UNSET (buffer, GST_BUFFER_FLAG_DISCONT);
+  if (interaudiosrc->n_samples == 0) {
+    GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DISCONT);
+  }
+  gst_buffer_set_caps (buffer, GST_PAD_CAPS (GST_BASE_SRC_PAD (interaudiosrc)));
+  interaudiosrc->n_samples += n;
+
+  *buf = buffer;
+
+  return GST_FLOW_OK;
+}
+
+static gboolean
+gst_inter_audio_src_do_seek (GstBaseSrc * src, GstSegment * segment)
+{
+  GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
+
+  GST_DEBUG_OBJECT (interaudiosrc, "do_seek");
+
+  return FALSE;
+}
+
+static gboolean
+gst_inter_audio_src_query (GstBaseSrc * src, GstQuery * query)
+{
+  GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
+
+  GST_DEBUG_OBJECT (interaudiosrc, "query");
+
+  return TRUE;
+}
+
+static gboolean
+gst_inter_audio_src_check_get_range (GstBaseSrc * src)
+{
+  GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
+
+  GST_DEBUG_OBJECT (interaudiosrc, "get_range");
+
+  return FALSE;
+}
+
+static void
+gst_inter_audio_src_fixate (GstBaseSrc * src, GstCaps * caps)
+{
+  GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
+  GstStructure *structure;
+
+  structure = gst_caps_get_structure (caps, 0);
+
+  GST_DEBUG_OBJECT (interaudiosrc, "fixate");
+
+  gst_structure_fixate_field_nearest_int (structure, "channels", 2);
+  gst_structure_fixate_field_nearest_int (structure, "rate", 48000);
+
+}
+
+static gboolean
+gst_inter_audio_src_unlock_stop (GstBaseSrc * src)
+{
+  GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
+
+  GST_DEBUG_OBJECT (interaudiosrc, "stop");
+
+  return TRUE;
+}
+
+static gboolean
+gst_inter_audio_src_prepare_seek_segment (GstBaseSrc * src, GstEvent * seek,
+    GstSegment * segment)
+{
+  GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
+
+  GST_DEBUG_OBJECT (interaudiosrc, "seek_segment");
+
+  return FALSE;
+}