rtpopuspay: add DTX support

If enabled, the payloader won't transmit empty frames.

Can be tested using:
  opusenc dtx=true bitrate-type=vbr ! rtpopuspay dtx=true

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/967>

docs/gst_plugins_cache.json

@@ -14854,7 +14854,20 @@
                         "presence": "always"
                     }
                 },
-                "properties": {},
+                "properties": {
+                    "dtx": {
+                        "blurb": "If enabled, the payloader will not transmit empty packets",
+                        "conditionally-available": false,
+                        "construct": false,
+                        "construct-only": false,
+                        "controllable": false,
+                        "default": "false",
+                        "mutable": "playing",
+                        "readable": true,
+                        "type": "gboolean",
+                        "writable": true
+                    }
+                },
                 "rank": "primary"
             },
             "rtppcmadepay": {

gst/rtp/gstrtpopuspay.c

@@ -58,6 +58,13 @@
 GST_DEBUG_CATEGORY_STATIC (rtpopuspay_debug);
 #define GST_CAT_DEFAULT (rtpopuspay_debug)
 
+enum
+{
+  PROP_0,
+  PROP_DTX,
+};
+
+#define DEFAULT_DTX FALSE
 
 static GstStaticPadTemplate gst_rtp_opus_pay_sink_template =
     GST_STATIC_PAD_TEMPLATE ("sink",
@@ -90,24 +97,77 @@ G_DEFINE_TYPE (GstRtpOPUSPay, gst_rtp_opus_pay, GST_TYPE_RTP_BASE_PAYLOAD);
 GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpopuspay, "rtpopuspay",
     GST_RANK_PRIMARY, GST_TYPE_RTP_OPUS_PAY, rtp_element_init (plugin));
 
+#define GST_RTP_OPUS_PAY_CAST(obj) ((GstRtpOPUSPay *)(obj))
+
+static void
+gst_rtp_opus_pay_set_property (GObject * object,
+    guint prop_id, const GValue * value, GParamSpec * pspec)
+{
+  GstRtpOPUSPay *self = GST_RTP_OPUS_PAY (object);
+
+  switch (prop_id) {
+    case PROP_DTX:
+      self->dtx = g_value_get_boolean (value);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_rtp_opus_pay_get_property (GObject * object,
+    guint prop_id, GValue * value, GParamSpec * pspec)
+{
+  GstRtpOPUSPay *self = GST_RTP_OPUS_PAY (object);
+
+  switch (prop_id) {
+    case PROP_DTX:
+      g_value_set_boolean (value, self->dtx);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
 static void
 gst_rtp_opus_pay_class_init (GstRtpOPUSPayClass * klass)
 {
   GstRTPBasePayloadClass *gstbasertppayload_class;
   GstElementClass *element_class;
+  GObjectClass *gobject_class;
 
   gstbasertppayload_class = (GstRTPBasePayloadClass *) klass;
   element_class = GST_ELEMENT_CLASS (klass);
+  gobject_class = (GObjectClass *) klass;
 
   gstbasertppayload_class->set_caps = gst_rtp_opus_pay_setcaps;
   gstbasertppayload_class->get_caps = gst_rtp_opus_pay_getcaps;
   gstbasertppayload_class->handle_buffer = gst_rtp_opus_pay_handle_buffer;
 
+  gobject_class->set_property = gst_rtp_opus_pay_set_property;
+  gobject_class->get_property = gst_rtp_opus_pay_get_property;
+
   gst_element_class_add_static_pad_template (element_class,
       &gst_rtp_opus_pay_src_template);
   gst_element_class_add_static_pad_template (element_class,
       &gst_rtp_opus_pay_sink_template);
 
+  /**
+   * GstRtpOPUSPay:dtx:
+   *
+   * If enabled, the payloader will not transmit empty packets.
+   *
+   * Since: 1.20
+   */
+  g_object_class_install_property (gobject_class, PROP_DTX,
+      g_param_spec_boolean ("dtx", "Discontinuous Transmission",
+          "If enabled, the payloader will not transmit empty packets",
+          DEFAULT_DTX,
+          G_PARAM_READWRITE | GST_PARAM_MUTABLE_PLAYING |
+          G_PARAM_STATIC_STRINGS));
+
   gst_element_class_set_static_metadata (element_class,
       "RTP Opus payloader",
       "Codec/Payloader/Network/RTP",
@@ -121,6 +181,7 @@ gst_rtp_opus_pay_class_init (GstRtpOPUSPayClass * klass)
 static void
 gst_rtp_opus_pay_init (GstRtpOPUSPay * rtpopuspay)
 {
+  rtpopuspay->dtx = DEFAULT_DTX;
 }
 
 static gboolean
@@ -236,9 +297,18 @@ static GstFlowReturn
 gst_rtp_opus_pay_handle_buffer (GstRTPBasePayload * basepayload,
     GstBuffer * buffer)
 {
+  GstRtpOPUSPay *self = GST_RTP_OPUS_PAY_CAST (basepayload);
   GstBuffer *outbuf;
   GstClockTime pts, dts, duration;
 
+  /* DTX packets are zero-length frames, with a 1 or 2-bytes header */
+  if (self->dtx && gst_buffer_get_size (buffer) <= 2) {
+    GST_LOG_OBJECT (self,
+        "discard empty buffer as DTX is enabled: %" GST_PTR_FORMAT, buffer);
+    gst_buffer_unref (buffer);
+    return GST_FLOW_OK;
+  }
+
   pts = GST_BUFFER_PTS (buffer);
   dts = GST_BUFFER_DTS (buffer);
   duration = GST_BUFFER_DURATION (buffer);

gst/rtp/gstrtpopuspay.h

@@ -44,6 +44,8 @@ typedef struct _GstRtpOPUSPayClass GstRtpOPUSPayClass;
 struct _GstRtpOPUSPay
 {
   GstRTPBasePayload payload;
+
+  gboolean dtx;
 };
 
 struct _GstRtpOPUSPayClass

tests/check/elements/rtp-payloading.c

@@ -21,6 +21,7 @@
 #include <gst/check/gstharness.h>
 #include <gst/audio/audio.h>
 #include <gst/base/base.h>
+#include <gst/rtp/gstrtpbuffer.h>
 #include <stdlib.h>
 
 #define RELEASE_ELEMENT(x) if(x) {gst_object_unref(x); x = NULL;}
@@ -1670,6 +1671,93 @@ GST_START_TEST (rtp_vorbis_renegotiate)
 
 GST_END_TEST;
 
+static guint16
+pull_rtp_buffer (GstHarness * h)
+{
+  gint16 seq;
+  GstBuffer *buf;
+  GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
+
+  buf = gst_harness_try_pull (h);
+  fail_unless (buf);
+
+  fail_unless (gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp));
+  seq = gst_rtp_buffer_get_seq (&rtp);
+  gst_rtp_buffer_unmap (&rtp);
+
+  gst_buffer_unref (buf);
+  return seq;
+}
+
+static void
+test_rtp_opus_dtx (gboolean dtx)
+{
+  GstHarness *h;
+  GstBuffer *buf;
+  /* generated with a muted mic using:
+   * gst-launch-1.0 pulsesrc ! opusenc dtx=true bitrate-type=vbr ! fakesink silent=false dump=true -v
+   */
+  static const guint8 opus_empty[] = { 0xf8 };
+  static const guint8 opus_frame[] = { 0xf8, 0xff, 0xfe };
+  guint16 seq, expected_seq;
+
+  h = gst_harness_new_parse ("rtpopuspay");
+  fail_unless (h);
+
+  gst_harness_set (h, "rtpopuspay", "dtx", dtx, NULL);
+
+  gst_harness_set_caps_str (h,
+      "audio/x-opus, rate=48000, channels=1, channel-mapping-family=0",
+      "application/x-rtp, media=audio, clock-rate=48000, encoding-name=OPUS, sprop-stereo=(string)0, encoding-params=(string)2, sprop-maxcapturerate=(string)48000, payload=96");
+
+  /* push first opus frame */
+  buf =
+      gst_buffer_new_wrapped (g_memdup (opus_frame, sizeof (opus_frame)),
+      sizeof (opus_frame));
+  fail_unless_equals_int (gst_harness_push (h, buf), GST_FLOW_OK);
+  seq = pull_rtp_buffer (h);
+  expected_seq = seq + 1;
+
+  /* push empty frame */
+  buf =
+      gst_buffer_new_wrapped (g_memdup (opus_empty, sizeof (opus_empty)),
+      sizeof (opus_empty));
+  fail_unless_equals_int (gst_harness_push (h, buf), GST_FLOW_OK);
+  if (dtx) {
+    /* buffer is not transmitted if dtx is enabled */
+    buf = gst_harness_try_pull (h);
+    fail_if (buf);
+  } else {
+    seq = pull_rtp_buffer (h);
+    fail_unless_equals_int (seq, expected_seq);
+    expected_seq++;
+  }
+
+  /* push second opus frame */
+  buf =
+      gst_buffer_new_wrapped (g_memdup (opus_frame, sizeof (opus_frame)),
+      sizeof (opus_frame));
+  fail_unless_equals_int (gst_harness_push (h, buf), GST_FLOW_OK);
+  seq = pull_rtp_buffer (h);
+  fail_unless_equals_int (seq, expected_seq);
+
+  gst_harness_teardown (h);
+}
+
+GST_START_TEST (rtp_opus_dtx_disabled)
+{
+  test_rtp_opus_dtx (FALSE);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (rtp_opus_dtx_enabled)
+{
+  test_rtp_opus_dtx (TRUE);
+}
+
+GST_END_TEST;
+
 /*
  * Creates the test suite.
  *
@@ -1734,6 +1822,8 @@ rtp_payloading_suite (void)
   tcase_add_test (tc_chain, rtp_g729);
   tcase_add_test (tc_chain, rtp_gst_custom_event);
   tcase_add_test (tc_chain, rtp_vorbis_renegotiate);
+  tcase_add_test (tc_chain, rtp_opus_dtx_disabled);
+  tcase_add_test (tc_chain, rtp_opus_dtx_enabled);
   return s;
 }