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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-23 08:46:40 +00:00
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>
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a0067316e7
commit
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4 changed files with 176 additions and 1 deletions
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@ -14854,7 +14854,20 @@
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"presence": "always"
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}
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},
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"properties": {},
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"properties": {
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"dtx": {
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"blurb": "If enabled, the payloader will not transmit empty packets",
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"conditionally-available": false,
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"construct": false,
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"construct-only": false,
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"controllable": false,
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"default": "false",
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"mutable": "playing",
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"readable": true,
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"type": "gboolean",
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"writable": true
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}
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},
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"rank": "primary"
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},
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"rtppcmadepay": {
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@ -58,6 +58,13 @@
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GST_DEBUG_CATEGORY_STATIC (rtpopuspay_debug);
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#define GST_CAT_DEFAULT (rtpopuspay_debug)
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enum
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{
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PROP_0,
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PROP_DTX,
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};
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#define DEFAULT_DTX FALSE
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static GstStaticPadTemplate gst_rtp_opus_pay_sink_template =
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GST_STATIC_PAD_TEMPLATE ("sink",
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@ -90,24 +97,77 @@ G_DEFINE_TYPE (GstRtpOPUSPay, gst_rtp_opus_pay, GST_TYPE_RTP_BASE_PAYLOAD);
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GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpopuspay, "rtpopuspay",
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GST_RANK_PRIMARY, GST_TYPE_RTP_OPUS_PAY, rtp_element_init (plugin));
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#define GST_RTP_OPUS_PAY_CAST(obj) ((GstRtpOPUSPay *)(obj))
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static void
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gst_rtp_opus_pay_set_property (GObject * object,
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guint prop_id, const GValue * value, GParamSpec * pspec)
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{
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GstRtpOPUSPay *self = GST_RTP_OPUS_PAY (object);
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switch (prop_id) {
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case PROP_DTX:
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self->dtx = g_value_get_boolean (value);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_rtp_opus_pay_get_property (GObject * object,
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guint prop_id, GValue * value, GParamSpec * pspec)
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{
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GstRtpOPUSPay *self = GST_RTP_OPUS_PAY (object);
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switch (prop_id) {
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case PROP_DTX:
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g_value_set_boolean (value, self->dtx);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_rtp_opus_pay_class_init (GstRtpOPUSPayClass * klass)
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{
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GstRTPBasePayloadClass *gstbasertppayload_class;
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GstElementClass *element_class;
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GObjectClass *gobject_class;
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gstbasertppayload_class = (GstRTPBasePayloadClass *) klass;
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element_class = GST_ELEMENT_CLASS (klass);
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gobject_class = (GObjectClass *) klass;
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gstbasertppayload_class->set_caps = gst_rtp_opus_pay_setcaps;
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gstbasertppayload_class->get_caps = gst_rtp_opus_pay_getcaps;
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gstbasertppayload_class->handle_buffer = gst_rtp_opus_pay_handle_buffer;
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gobject_class->set_property = gst_rtp_opus_pay_set_property;
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gobject_class->get_property = gst_rtp_opus_pay_get_property;
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gst_element_class_add_static_pad_template (element_class,
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&gst_rtp_opus_pay_src_template);
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gst_element_class_add_static_pad_template (element_class,
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&gst_rtp_opus_pay_sink_template);
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/**
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* GstRtpOPUSPay:dtx:
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*
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* If enabled, the payloader will not transmit empty packets.
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*
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* Since: 1.20
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*/
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g_object_class_install_property (gobject_class, PROP_DTX,
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g_param_spec_boolean ("dtx", "Discontinuous Transmission",
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"If enabled, the payloader will not transmit empty packets",
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DEFAULT_DTX,
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G_PARAM_READWRITE | GST_PARAM_MUTABLE_PLAYING |
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G_PARAM_STATIC_STRINGS));
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gst_element_class_set_static_metadata (element_class,
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"RTP Opus payloader",
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"Codec/Payloader/Network/RTP",
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@ -121,6 +181,7 @@ gst_rtp_opus_pay_class_init (GstRtpOPUSPayClass * klass)
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static void
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gst_rtp_opus_pay_init (GstRtpOPUSPay * rtpopuspay)
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{
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rtpopuspay->dtx = DEFAULT_DTX;
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}
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static gboolean
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@ -236,9 +297,18 @@ static GstFlowReturn
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gst_rtp_opus_pay_handle_buffer (GstRTPBasePayload * basepayload,
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GstBuffer * buffer)
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{
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GstRtpOPUSPay *self = GST_RTP_OPUS_PAY_CAST (basepayload);
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GstBuffer *outbuf;
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GstClockTime pts, dts, duration;
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/* DTX packets are zero-length frames, with a 1 or 2-bytes header */
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if (self->dtx && gst_buffer_get_size (buffer) <= 2) {
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GST_LOG_OBJECT (self,
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"discard empty buffer as DTX is enabled: %" GST_PTR_FORMAT, buffer);
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gst_buffer_unref (buffer);
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return GST_FLOW_OK;
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}
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pts = GST_BUFFER_PTS (buffer);
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dts = GST_BUFFER_DTS (buffer);
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duration = GST_BUFFER_DURATION (buffer);
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@ -44,6 +44,8 @@ typedef struct _GstRtpOPUSPayClass GstRtpOPUSPayClass;
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struct _GstRtpOPUSPay
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{
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GstRTPBasePayload payload;
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gboolean dtx;
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};
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struct _GstRtpOPUSPayClass
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@ -21,6 +21,7 @@
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#include <gst/check/gstharness.h>
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#include <gst/audio/audio.h>
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#include <gst/base/base.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include <stdlib.h>
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#define RELEASE_ELEMENT(x) if(x) {gst_object_unref(x); x = NULL;}
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@ -1670,6 +1671,93 @@ GST_START_TEST (rtp_vorbis_renegotiate)
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GST_END_TEST;
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static guint16
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pull_rtp_buffer (GstHarness * h)
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{
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gint16 seq;
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GstBuffer *buf;
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GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
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buf = gst_harness_try_pull (h);
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fail_unless (buf);
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fail_unless (gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp));
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seq = gst_rtp_buffer_get_seq (&rtp);
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gst_rtp_buffer_unmap (&rtp);
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gst_buffer_unref (buf);
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return seq;
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}
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static void
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test_rtp_opus_dtx (gboolean dtx)
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{
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GstHarness *h;
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GstBuffer *buf;
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/* generated with a muted mic using:
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* gst-launch-1.0 pulsesrc ! opusenc dtx=true bitrate-type=vbr ! fakesink silent=false dump=true -v
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*/
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static const guint8 opus_empty[] = { 0xf8 };
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static const guint8 opus_frame[] = { 0xf8, 0xff, 0xfe };
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guint16 seq, expected_seq;
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h = gst_harness_new_parse ("rtpopuspay");
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fail_unless (h);
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gst_harness_set (h, "rtpopuspay", "dtx", dtx, NULL);
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gst_harness_set_caps_str (h,
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"audio/x-opus, rate=48000, channels=1, channel-mapping-family=0",
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"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");
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/* push first opus frame */
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buf =
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gst_buffer_new_wrapped (g_memdup (opus_frame, sizeof (opus_frame)),
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sizeof (opus_frame));
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fail_unless_equals_int (gst_harness_push (h, buf), GST_FLOW_OK);
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seq = pull_rtp_buffer (h);
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expected_seq = seq + 1;
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/* push empty frame */
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buf =
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gst_buffer_new_wrapped (g_memdup (opus_empty, sizeof (opus_empty)),
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sizeof (opus_empty));
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fail_unless_equals_int (gst_harness_push (h, buf), GST_FLOW_OK);
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if (dtx) {
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/* buffer is not transmitted if dtx is enabled */
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buf = gst_harness_try_pull (h);
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fail_if (buf);
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} else {
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seq = pull_rtp_buffer (h);
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fail_unless_equals_int (seq, expected_seq);
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expected_seq++;
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}
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/* push second opus frame */
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buf =
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gst_buffer_new_wrapped (g_memdup (opus_frame, sizeof (opus_frame)),
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sizeof (opus_frame));
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fail_unless_equals_int (gst_harness_push (h, buf), GST_FLOW_OK);
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seq = pull_rtp_buffer (h);
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fail_unless_equals_int (seq, expected_seq);
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gst_harness_teardown (h);
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}
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GST_START_TEST (rtp_opus_dtx_disabled)
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{
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test_rtp_opus_dtx (FALSE);
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}
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GST_END_TEST;
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GST_START_TEST (rtp_opus_dtx_enabled)
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{
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test_rtp_opus_dtx (TRUE);
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}
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GST_END_TEST;
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/*
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* Creates the test suite.
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*
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@ -1734,6 +1822,8 @@ rtp_payloading_suite (void)
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tcase_add_test (tc_chain, rtp_g729);
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tcase_add_test (tc_chain, rtp_gst_custom_event);
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tcase_add_test (tc_chain, rtp_vorbis_renegotiate);
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tcase_add_test (tc_chain, rtp_opus_dtx_disabled);
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tcase_add_test (tc_chain, rtp_opus_dtx_enabled);
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return s;
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}
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