#define FIXED_16_16_TO_DOUBLE(v) ((double) ((v & 0xffff0000) >> 16) + ((v & 0xffff) / 65536.0))
#define FIXED_32_32_TO_DOUBLE(v) ((double) ((v & G_GUINT64_CONSTANT (0xffffffff00000000)) >> 32) + ((v & G_GUINT64_CONSTANT (0xffffffff)) / 4294967296.0))
+/* https://www.w3.org/TR/webrtc-stats/#remoteinboundrtpstats-dict* */
+static gboolean
+_get_stats_from_remote_rtp_source_stats (GstWebRTCBin * webrtc,
+ TransportStream * stream, const GstStructure * source_stats,
+ guint ssrc, guint clock_rate, const gchar * codec_id,
+ const gchar * transport_id, GstStructure * s)
+{
+ gboolean have_rb = FALSE, internal = FALSE;
+ int lost;
+ GstStructure *r_in;
+ gchar *r_in_id, *out_id;
+ guint32 rtt;
+ guint fraction_lost, jitter;
+ double ts;
+
+ gst_structure_get_double (s, "timestamp", &ts);
+ gst_structure_get (source_stats, "internal", G_TYPE_BOOLEAN, &internal,
+ "have-rb", G_TYPE_BOOLEAN, &have_rb, NULL);
+
+ /* This isn't what we're looking for */
+ if (internal == TRUE || have_rb == FALSE)
+ return FALSE;
+
+ r_in_id = g_strdup_printf ("rtp-remote-inbound-stream-stats_%u", ssrc);
+ out_id = g_strdup_printf ("rtp-outbound-stream-stats_%u", ssrc);
+
+ r_in = gst_structure_new_empty (r_in_id);
+ _set_base_stats (r_in, GST_WEBRTC_STATS_REMOTE_INBOUND_RTP, ts, r_in_id);
+
+ /* RTCRtpStreamStats */
+ gst_structure_set (r_in, "local-id", G_TYPE_STRING, out_id, NULL);
+ gst_structure_set (r_in, "ssrc", G_TYPE_UINT, ssrc, NULL);
+ gst_structure_set (r_in, "codec-id", G_TYPE_STRING, codec_id, NULL);
+ gst_structure_set (r_in, "transport-id", G_TYPE_STRING, transport_id, NULL);
+ /* To be added: kind */
+
+ /* RTCReceivedRtpStreamStats */
+
+ if (gst_structure_get_int (source_stats, "rb-packetslost", &lost))
+ gst_structure_set (r_in, "packets-lost", G_TYPE_INT, lost, NULL);
+
+ if (clock_rate && gst_structure_get_uint (source_stats, "rb-jitter", &jitter))
+ gst_structure_set (r_in, "jitter", G_TYPE_DOUBLE,
+ CLOCK_RATE_VALUE_TO_SECONDS (jitter, clock_rate), NULL);
+
+ /* RTCReceivedRtpStreamStats:
+
+ unsigned long long packetsReceived;
+ unsigned long packetsDiscarded;
+ unsigned long packetsRepaired;
+ unsigned long burstPacketsLost;
+ unsigned long burstPacketsDiscarded;
+ unsigned long burstLossCount;
+ unsigned long burstDiscardCount;
+ double burstLossRate;
+ double burstDiscardRate;
+ double gapLossRate;
+ double gapDiscardRate;
+
+ Can't be implemented frame re-assembly happens after rtpbin:
+
+ unsigned long framesDropped;
+ unsigned long partialFramesLost;
+ unsigned long fullFramesLost;
+ */
+
+ /* RTCRemoteInboundRTPStreamStats */
+
+ if (gst_structure_get_uint (source_stats, "rb-fractionlost", &fraction_lost))
+ gst_structure_set (r_in, "fraction-lost", G_TYPE_DOUBLE,
+ (double) fraction_lost / 256.0, NULL);
+
+ if (gst_structure_get_uint (source_stats, "rb-round-trip", &rtt)) {
+ /* 16.16 fixed point to double */
+ double val = FIXED_16_16_TO_DOUBLE (rtt);
+ gst_structure_set (r_in, "round-trip-time", G_TYPE_DOUBLE, val, NULL);
+ }
+
+ /* RTCRemoteInboundRTPStreamStats:
+
+ To be added:
+
+ DOMString localId;
+ double totalRoundTripTime;
+ unsigned long long reportsReceived;
+ unsigned long long roundTripTimeMeasurements;
+ */
+
+ gst_structure_set (r_in, "gst-rtpsource-stats", GST_TYPE_STRUCTURE,
+ source_stats, NULL);
+
+ _gst_structure_take_structure (s, r_in_id, &r_in);
+
+ g_free (r_in_id);
+ g_free (out_id);
+
+ return TRUE;
+}
+
/* https://www.w3.org/TR/webrtc-stats/#inboundrtpstats-dict*
https://www.w3.org/TR/webrtc-stats/#outboundrtpstats-dict* */
static void
const gchar * codec_id, const gchar * transport_id, GstStructure * s)
{
guint ssrc, fir, pli, nack, jitter;
- int lost, clock_rate;
+ int clock_rate;
guint64 packets, bytes;
gboolean internal;
double ts;
G_TYPE_INT, &clock_rate, "internal", G_TYPE_BOOLEAN, &internal, NULL);
if (internal) {
- GstStructure *r_in, *out;
+ GstStructure *out;
gchar *out_id, *r_in_id;
- gboolean have_rb = FALSE;
out_id = g_strdup_printf ("rtp-outbound-stream-stats_%u", ssrc);
- r_in_id = g_strdup_printf ("rtp-remote-inbound-stream-stats_%u", ssrc);
-
- gst_structure_get (source_stats, "have-rb", G_TYPE_BOOLEAN, &have_rb, NULL);
-
- r_in = gst_structure_new_empty (r_in_id);
- _set_base_stats (r_in, GST_WEBRTC_STATS_REMOTE_INBOUND_RTP, ts, r_in_id);
-
- /* RTCRtpStreamStats */
- gst_structure_set (r_in, "local-id", G_TYPE_STRING, out_id, NULL);
- gst_structure_set (r_in, "ssrc", G_TYPE_UINT, ssrc, NULL);
- gst_structure_set (r_in, "codec-id", G_TYPE_STRING, codec_id, NULL);
- gst_structure_set (r_in, "transport-id", G_TYPE_STRING, transport_id, NULL);
- /* To be added: kind */
-
- /* RTCReceivedRtpStreamStats:
-
- To be added:
-
- unsigned long long packetsReceived;
- long long packetsLost;
- double jitter;
- unsigned long packetsDiscarded;
- unsigned long packetsRepaired;
- unsigned long burstPacketsLost;
- unsigned long burstPacketsDiscarded;
- unsigned long burstLossCount;
- unsigned long burstDiscardCount;
- double burstLossRate;
- double burstDiscardRate;
- double gapLossRate;
- double gapDiscardRate;
-
- Can't be implemented frame re-assembly happens after rtpbin:
-
- unsigned long framesDropped;
- unsigned long partialFramesLost;
- unsigned long fullFramesLost;
- */
-
-
- /* RTCRemoteInboundRTPStreamStats */
-
- if (have_rb) {
- guint32 rtt;
- if (gst_structure_get_uint (source_stats, "rb-round-trip", &rtt)) {
- /* 16.16 fixed point to double */
- double val = FIXED_16_16_TO_DOUBLE (rtt);
- gst_structure_set (r_in, "round-trip-time", G_TYPE_DOUBLE, val, NULL);
- }
- }
-
- /* RTCRemoteInboundRTPStreamStats:
-
- To be added:
-
- DOMString localId;
- double totalRoundTripTime;
- unsigned long long reportsReceived;
- unsigned long long roundTripTimeMeasurements;
- */
out = gst_structure_new_empty (out_id);
_set_base_stats (out, GST_WEBRTC_STATS_OUTBOUND_RTP, ts, out_id);
gst_structure_set (out, "nack-count", G_TYPE_UINT, nack, NULL);
/* XXX: mediaType, trackId, sliCount, qpSum */
- gst_structure_set (out, "remote-id", G_TYPE_STRING, r_in_id, NULL);
-
+ r_in_id = g_strdup_printf ("rtp-remote-inbound-stream-stats_%u", ssrc);
+ if (gst_structure_has_field (s, r_in_id))
+ gst_structure_set (out, "remote-id", G_TYPE_STRING, r_in_id, NULL);
+ g_free (r_in_id);
/* RTCOutboundRTPStreamStats:
gst_structure_set (out, "gst-rtpsource-stats", GST_TYPE_STRUCTURE,
source_stats, NULL);
-
_gst_structure_take_structure (s, out_id, &out);
- _gst_structure_take_structure (s, r_in_id, &r_in);
g_free (out_id);
- g_free (r_in_id);
} else {
GstStructure *in, *r_out;
gchar *r_out_id, *in_id;
/* RTCSentRtpStreamStats */
if (have_sr) {
- if (gst_structure_get_uint64 (source_stats, "sr-octet-count", &bytes))
- gst_structure_set (r_out, "bytes-sent", G_TYPE_UINT64, bytes, NULL);
- if (gst_structure_get_uint64 (source_stats, "sr-packet-count", &packets))
- gst_structure_set (r_out, "packets-sent", G_TYPE_UINT64, packets, NULL);
+ guint sr_bytes, sr_packets;
+
+ if (gst_structure_get_uint (source_stats, "sr-octet-count", &sr_bytes))
+ gst_structure_set (r_out, "bytes-sent", G_TYPE_UINT, sr_bytes, NULL);
+ if (gst_structure_get_uint (source_stats, "sr-packet-count", &sr_packets))
+ gst_structure_set (r_out, "packets-sent", G_TYPE_UINT, sr_packets,
+ NULL);
}
/* RTCSentRtpStreamStats:
static void
_get_stats_from_transport_channel (GstWebRTCBin * webrtc,
TransportStream * stream, const gchar * codec_id, guint ssrc,
- GstStructure * s)
+ guint clock_rate, GstStructure * s)
{
GstWebRTCDTLSTransport *transport;
GObject *rtp_session;
/* skip foreign sources */
gst_structure_get (stats, "ssrc", G_TYPE_UINT, &stats_ssrc, NULL);
- if (ssrc && stats_ssrc && ssrc != stats_ssrc)
- continue;
-
- _get_stats_from_rtp_source_stats (webrtc, stream, stats, codec_id,
- transport_id, s);
+ if (gst_structure_get_uint (stats, "ssrc", &stats_ssrc) &&
+ ssrc == stats_ssrc)
+ _get_stats_from_rtp_source_stats (webrtc, stream, stats, codec_id,
+ transport_id, s);
+ else if (gst_structure_get_uint (stats, "rb-ssrc", &stats_ssrc) &&
+ ssrc == stats_ssrc)
+ _get_stats_from_remote_rtp_source_stats (webrtc, stream, stats, ssrc,
+ clock_rate, codec_id, transport_id, s);
}
g_object_unref (rtp_session);
/* https://www.w3.org/TR/webrtc-stats/#codec-dict* */
static void
_get_codec_stats_from_pad (GstWebRTCBin * webrtc, GstPad * pad,
- GstStructure * s, gchar ** out_id, guint * out_ssrc)
+ GstStructure * s, gchar ** out_id, guint * out_ssrc, guint * out_clock_rate)
{
GstStructure *stats;
GstCaps *caps;
gchar *id;
double ts;
guint ssrc = 0;
+ gint clock_rate = 0;
gst_structure_get_double (s, "timestamp", &ts);
caps = gst_pad_get_current_caps (pad);
if (caps && gst_caps_is_fixed (caps)) {
GstStructure *caps_s = gst_caps_get_structure (caps, 0);
- gint pt, clock_rate;
+ gint pt;
if (gst_structure_get_int (caps_s, "payload", &pt))
gst_structure_set (stats, "payload-type", G_TYPE_UINT, pt, NULL);
if (out_ssrc)
*out_ssrc = ssrc;
+
+ if (out_clock_rate)
+ *out_clock_rate = clock_rate;
}
static gboolean
GstWebRTCBinPad *wpad = GST_WEBRTC_BIN_PAD (pad);
TransportStream *stream;
gchar *codec_id;
- guint ssrc;
+ guint ssrc, clock_rate;
- _get_codec_stats_from_pad (webrtc, pad, s, &codec_id, &ssrc);
+ _get_codec_stats_from_pad (webrtc, pad, s, &codec_id, &ssrc, &clock_rate);
if (!wpad->trans)
goto out;
if (!stream)
goto out;
- _get_stats_from_transport_channel (webrtc, stream, codec_id, ssrc, s);
+ _get_stats_from_transport_channel (webrtc, stream, codec_id, ssrc,
+ clock_rate, s);
out:
g_free (codec_id);
projects / platform / upstream / gst-plugins-bad.git / commitdiff