webrtcdsp: Add WebRTC Audio Processing support

This DSP library can be used to enhance voice signal for real time
communication call. In implements multiple filters like noise reduction,
high pass filter, echo cancellation, automatic gain control, etc.

The webrtcdsp element can be used along, or with the help of the
webrtcechoprobe if echo cancellation is enabled. The echo probe should
be placed as close as possible to the audio sink, while the DSP is
generally place close to the audio capture. For local testing, one can
use an echo loop pipeline like the following:

  autoaudiosrc ! webrtcdsp ! webrtcechoprobe ! autoaudiosink

This pipeline should produce a single echo rather then repeated echo.
Those elements works if they are placed in the same top level pipeline.

https://bugzilla.gnome.org/show_bug.cgi?id=767800

1 files changed, 86 insertions, 0 deletions

diff --git a/ext/webrtcdsp/gstwebrtcechoprobe.h b/ext/webrtcdsp/gstwebrtcechoprobe.h
new file mode 100644
index 000000000..a18547d47
--- /dev/null
+++ b/ext/webrtcdsp/gstwebrtcechoprobe.h
@@ -0,0 +1,86 @@
+/*
+ * WebRTC Audio Processing Elements
+ *
+ *  Copyright 2016 Collabora Ltd
+ *    @author: Nicolas Dufresne <nicolas.dufresne@collabora.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 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
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA  02110-1301 USA
+ *
+ */
+
+#ifndef __GST_WEBRTC_ECHO_PROBE_H__
+#define __GST_WEBRTC_ECHO_PROBE_H__
+
+#include <gst/gst.h>
+#include <gst/base/gstadapter.h>
+#include <gst/base/gstbasetransform.h>
+#include <gst/audio/audio.h>
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_WEBRTC_ECHO_PROBE            (gst_webrtc_echo_probe_get_type())
+#define GST_WEBRTC_ECHO_PROBE(obj)            (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_ECHO_PROBE,GstWebrtcEchoProbe))
+#define GST_IS_WEBRTC_ECHO_PROBE(obj)         (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WEBRTC_ECHO_PROBE))
+#define GST_WEBRTC_ECHO_PROBE_CLASS(klass)    (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_ECHO_PROBE,GstWebrtcEchoProbeClass))
+#define GST_IS_WEBRTC_ECHO_PROBE_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_ECHO_PROBE))
+#define GST_WEBRTC_ECHO_PROBE_GET_CLASS(obj)  (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_ECHO_PROBE,GstWebrtcEchoProbeClass))
+
+#define GST_WEBRTC_ECHO_PROBE_LOCK(obj) g_mutex_lock (&GST_WEBRTC_ECHO_PROBE (obj)->lock)
+#define GST_WEBRTC_ECHO_PROBE_UNLOCK(obj) g_mutex_unlock (&GST_WEBRTC_ECHO_PROBE (obj)->lock)
+
+typedef struct _GstWebrtcEchoProbe GstWebrtcEchoProbe;
+typedef struct _GstWebrtcEchoProbeClass GstWebrtcEchoProbeClass;
+
+/**
+ * GstWebrtcEchoProbe:
+ *
+ * The adder object structure.
+ */
+struct _GstWebrtcEchoProbe
+{
+  GstAudioFilter parent;
+
+  /* This lock is required as the DSP may need to lock itself using it's
+   * object lock and also lock the probe. The natural order for the DSP is
+   * to lock the DSP and then the echo probe. If we where using the probe
+   * object lock, we'd be racing with GstBin which will lock sink to src,
+   * and may accidently reverse the order. */
+  GMutex lock;
+
+  /* Protected by the lock */
+  GstAudioInfo info;
+  guint period_size;
+  gint latency;
+  gboolean synchronized;
+
+  GstSegment segment;
+  GstAdapter *adapter;
+
+  /* Private */
+  gboolean acquired;
+};
+
+struct _GstWebrtcEchoProbeClass
+{
+  GstAudioFilterClass parent_class;
+};
+
+GType gst_webrtc_echo_probe_get_type (void);
+
+GstWebrtcEchoProbe *gst_webrtc_acquire_echo_probe (const gchar * name);
+void gst_webrtc_release_echo_probe (GstWebrtcEchoProbe * probe);
+
+G_END_DECLS
+#endif /* __GST_WEBRTC_ECHO_PROBE_H__ */