new enums are added as below.
for webrtc_stats_type_e
: WEBRTC_STATS_TYPE_CANDIDATE_PAIR
: WEBRTC_STATS_TYPE_LOCAL_CANDIDATE
: WEBRTC_STATS_TYPE_REMOTE_CANDIDATE
for webrtc_stats_prop_e
: WEBRTC_STATS_PROP_KIND
: WEBRTC_STATS_PROP_ADDRESS
: WEBRTC_STATS_PROP_PORT
: WEBRTC_STATS_PROP_CANDIDATE_TYPE
: WEBRTC_STATS_PROP_PRIORITY
: WEBRTC_STATS_PROP_PROTOCOL
: WEBRTC_STATS_PROP_RELAY_PROTOCOL
: WEBRTC_STATS_PROP_URL
: WEBRTC_STATS_PROP_LOCAL_CANDIDATE_ID
: WEBRTC_STATS_PROP_REMOTE_CANDIDATE_ID
Some enum values are changed.
Some definitions are deprecated.
[Version] 0.4.1
[Issue type] New feature
Change-Id: I021af299a1b3ddb048f29a3888b5dc86d8191ab8
WEBRTC_STATS_TYPE_INBOUND_RTP = 0x0002, /**< Inbound RTP */
WEBRTC_STATS_TYPE_OUTBOUND_RTP = 0x0004, /**< Outbound RTP */
WEBRTC_STATS_TYPE_REMOTE_INBOUND_RTP = 0x0008, /**< Remote inbound RTP */
- WEBRTC_STATS_TYPE_REMOTE_OUTBOUND_RTP = 0x0010 /**< Remote outbound RTP */
+ WEBRTC_STATS_TYPE_REMOTE_OUTBOUND_RTP = 0x0010, /**< Remote outbound RTP */
+ WEBRTC_STATS_TYPE_CANDIDATE_PAIR = 0x0020, /**< Candidate pair (Since 7.5) */
+ WEBRTC_STATS_TYPE_LOCAL_CANDIDATE = 0x0040, /**< Local candidate (Since 7.5) */
+ WEBRTC_STATS_TYPE_REMOTE_CANDIDATE = 0x0080, /**< Remote candidate (Since 7.5) */
} webrtc_stats_type_e;
/**
* @since_tizen 7.0
* @see webrtc_foreach_stats()
*/
-#define WEBRTC_STATS_TYPE_ALL \
- WEBRTC_STATS_TYPE_CODEC | \
- WEBRTC_STATS_TYPE_INBOUND_RTP | \
- WEBRTC_STATS_TYPE_OUTBOUND_RTP | \
- WEBRTC_STATS_TYPE_REMOTE_INBOUND_RTP | \
- WEBRTC_STATS_TYPE_REMOTE_OUTBOUND_RTP
+#define WEBRTC_STATS_TYPE_ALL 0
/**
+ * @deprecated Deprecated since 7.5.
* @brief Definition for mask value used by #webrtc_stats_prop_e that represents properties of RTC stats.
* @since_tizen 7.0
* @remarks It corresponds with the values described in https://www.w3.org/TR/webrtc-stats/#dom-rtcstats.
#define WEBRTC_STATS_COMMON 0x00000100
/**
+ * @deprecated Deprecated since 7.5.
* @brief Definition for mask value used by #webrtc_stats_prop_e that represents properties of RTC codec stats.
* @since_tizen 7.0
* @remarks It corresponds with the values described in https://www.w3.org/TR/webrtc-stats/#dom-rtccodecstats.
#define WEBRTC_STATS_CODEC 0x00000200
/**
+ * @deprecated Deprecated since 7.5.
* @brief Definition for mask value used by #webrtc_stats_prop_e that represents properties of RTC RTP stream stats.
* @since_tizen 7.0
* @remarks It corresponds with the values described in https://www.w3.org/TR/webrtc-stats/#dom-rtcrtpstreamstats.
#define WEBRTC_STATS_RTP_STREAM 0x00000400
/**
+ * @deprecated Deprecated since 7.5.
* @brief Definition for mask value used by #webrtc_stats_prop_e that represents properties of RTC received RTP stream stats.
* @since_tizen 7.0
* @remarks It corresponds with the values described in https://www.w3.org/TR/webrtc-stats/#dom-rtcreceivedrtpstreamstats.
#define WEBRTC_STATS_RECEIVED_RTP_STREAM 0x00000800
/**
+ * @deprecated Deprecated since 7.5.
* @brief Definition for mask value used by #webrtc_stats_prop_e that represents properties of RTC inbound RTP stream stats.
* @since_tizen 7.0
* @remarks It corresponds with the values described in https://www.w3.org/TR/webrtc-stats/#dom-rtcinboundrtpstreamstats.
#define WEBRTC_STATS_INBOUND_RTP_STREAM 0x00001000
/**
+ * @deprecated Deprecated since 7.5.
* @brief Definition for mask value used by #webrtc_stats_prop_e that represents properties of RTC sent RTP stream stats.
* @since_tizen 7.0
* @remarks It corresponds with the values described in https://www.w3.org/TR/webrtc-stats/#dom-rtcsentrtpstreamstats.
#define WEBRTC_STATS_SENT_RTP_STREAM 0x00002000
/**
+ * @deprecated Deprecated since 7.5.
* @brief Definition for mask value used by #webrtc_stats_prop_e that represents properties of RTC outbound RTP stream stats.
* @since_tizen 7.0
* @remarks It corresponds with the values described in https://www.w3.org/TR/webrtc-stats/#dom-rtcoutboundrtpstreamstats.
#define WEBRTC_STATS_OUTBOUND_RTP_STREAM 0x00004000
/**
+ * @deprecated Deprecated since 7.5.
* @brief Definition for mask value used by #webrtc_stats_prop_e that represents properties of RTC remote inbound RTP stream stats.
* @since_tizen 7.0
* @remarks It corresponds with the values described in https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteinboundrtpstreamstats.
#define WEBRTC_STATS_REMOTE_INBOUND_RTP_STREAM 0x00008000
/**
+ * @deprecated Deprecated since 7.5.
* @brief Definition for mask value used by #webrtc_stats_prop_e that represents properties of RTC remote outbound RTP stream stats.
* @since_tizen 7.0
* @remarks It corresponds with the values described in https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats.
* @since_tizen 7.0
*/
typedef enum {
- WEBRTC_STATS_PROP_TIMESTAMP = WEBRTC_STATS_COMMON | 0x01, /**< Timestamp */
- WEBRTC_STATS_PROP_ID = WEBRTC_STATS_COMMON | 0x02, /**< Id */
- WEBRTC_STATS_PROP_PAYLOAD_TYPE = WEBRTC_STATS_CODEC | 0x01, /**< Payload type */
- WEBRTC_STATS_PROP_CLOCK_RATE = WEBRTC_STATS_CODEC | 0x02, /**< Clock rate */
- WEBRTC_STATS_PROP_CHANNELS = WEBRTC_STATS_CODEC | 0x03, /**< Channels */
- WEBRTC_STATS_PROP_MIME_TYPE = WEBRTC_STATS_CODEC | 0x04, /**< MIME type */
- WEBRTC_STATS_PROP_CODEC_TYPE = WEBRTC_STATS_CODEC | 0x05, /**< Codec type */
- WEBRTC_STATS_PROP_SDP_FMTP_LINE = WEBRTC_STATS_CODEC | 0x06, /**< SDP FMTP line */
- WEBRTC_STATS_PROP_SSRC = WEBRTC_STATS_RTP_STREAM | 0x01, /**< SSRC */
- WEBRTC_STATS_PROP_TRANSPORT_ID = WEBRTC_STATS_RTP_STREAM | 0x02, /**< Transport id */
- WEBRTC_STATS_PROP_CODEC_ID = WEBRTC_STATS_RTP_STREAM | 0x03, /**< Codec id */
- WEBRTC_STATS_PROP_PACKETS_RECEIVED = WEBRTC_STATS_RECEIVED_RTP_STREAM | 0x01, /**< Packets received */
- WEBRTC_STATS_PROP_PACKETS_LOST = WEBRTC_STATS_RECEIVED_RTP_STREAM | 0x02, /**< Packets lost */
- WEBRTC_STATS_PROP_PACKETS_DISCARDED = WEBRTC_STATS_RECEIVED_RTP_STREAM | 0x03, /**< Packets discarded */
- WEBRTC_STATS_PROP_JITTER = WEBRTC_STATS_RECEIVED_RTP_STREAM | 0x05, /**< Jitter */
- WEBRTC_STATS_PROP_BYTES_RECEIVED = WEBRTC_STATS_INBOUND_RTP_STREAM | 0x01, /**< Bytes received */
- WEBRTC_STATS_PROP_PACKETS_DUPLICATED = WEBRTC_STATS_INBOUND_RTP_STREAM | 0x02, /**< Packets duplicated */
- WEBRTC_STATS_PROP_BYTES_SENT = WEBRTC_STATS_SENT_RTP_STREAM | 0x01, /**< Bytes sent */
- WEBRTC_STATS_PROP_PACKETS_SENT = WEBRTC_STATS_SENT_RTP_STREAM | 0x02, /**< Packets sent */
- WEBRTC_STATS_PROP_REMOTE_ID = WEBRTC_STATS_INBOUND_RTP_STREAM | WEBRTC_STATS_OUTBOUND_RTP_STREAM | 0x01, /**< Remote id */
- WEBRTC_STATS_PROP_FIR_COUNT = WEBRTC_STATS_INBOUND_RTP_STREAM | WEBRTC_STATS_OUTBOUND_RTP_STREAM | 0x02, /**< FIR packets */
- WEBRTC_STATS_PROP_PLI_COUNT = WEBRTC_STATS_INBOUND_RTP_STREAM | WEBRTC_STATS_OUTBOUND_RTP_STREAM | 0x03, /**< PLI packets */
- WEBRTC_STATS_PROP_NACK_COUNT = WEBRTC_STATS_INBOUND_RTP_STREAM | WEBRTC_STATS_OUTBOUND_RTP_STREAM | 0x04, /**< NACK packets */
- WEBRTC_STATS_PROP_ROUND_TRIP_TIME = WEBRTC_STATS_REMOTE_INBOUND_RTP_STREAM | 0x01, /**< Round trip time */
- WEBRTC_STATS_PROP_FRACTION_LOST = WEBRTC_STATS_REMOTE_INBOUND_RTP_STREAM | 0x02, /**< Fraction lost */
- WEBRTC_STATS_PROP_REMOTE_TIMESTAMP = WEBRTC_STATS_REMOTE_OUTBOUND_RTP_STREAM | 0x01, /**< Remote timestamp */
- WEBRTC_STATS_PROP_LOCAL_ID = WEBRTC_STATS_REMOTE_INBOUND_RTP_STREAM | WEBRTC_STATS_REMOTE_OUTBOUND_RTP_STREAM | 0x01, /**< Local id */
+ WEBRTC_STATS_PROP_TIMESTAMP = 1, /**< Timestamp */
+ WEBRTC_STATS_PROP_ID, /**< Id */
+ WEBRTC_STATS_PROP_PAYLOAD_TYPE, /**< Payload type */
+ WEBRTC_STATS_PROP_CLOCK_RATE, /**< Clock rate */
+ WEBRTC_STATS_PROP_CHANNELS, /**< Channels */
+ WEBRTC_STATS_PROP_MIME_TYPE, /**< MIME type */
+ WEBRTC_STATS_PROP_CODEC_TYPE, /**< Codec type */
+ WEBRTC_STATS_PROP_SDP_FMTP_LINE, /**< SDP FMTP line */
+ WEBRTC_STATS_PROP_SSRC, /**< SSRC */
+ WEBRTC_STATS_PROP_TRANSPORT_ID, /**< Transport id */
+ WEBRTC_STATS_PROP_CODEC_ID, /**< Codec id */
+ WEBRTC_STATS_PROP_PACKETS_RECEIVED, /**< Packets received */
+ WEBRTC_STATS_PROP_PACKETS_LOST, /**< Packets lost */
+ WEBRTC_STATS_PROP_PACKETS_DISCARDED, /**< Packets discarded */
+ WEBRTC_STATS_PROP_JITTER, /**< Jitter */
+ WEBRTC_STATS_PROP_BYTES_RECEIVED, /**< Bytes received */
+ WEBRTC_STATS_PROP_PACKETS_DUPLICATED, /**< Packets duplicated */
+ WEBRTC_STATS_PROP_BYTES_SENT, /**< Bytes sent */
+ WEBRTC_STATS_PROP_PACKETS_SENT, /**< Packets sent */
+ WEBRTC_STATS_PROP_REMOTE_ID, /**< Remote id */
+ WEBRTC_STATS_PROP_FIR_COUNT, /**< FIR packets */
+ WEBRTC_STATS_PROP_PLI_COUNT, /**< PLI packets */
+ WEBRTC_STATS_PROP_NACK_COUNT, /**< NACK packets */
+ WEBRTC_STATS_PROP_ROUND_TRIP_TIME, /**< Round trip time */
+ WEBRTC_STATS_PROP_FRACTION_LOST, /**< Fraction lost */
+ WEBRTC_STATS_PROP_REMOTE_TIMESTAMP, /**< Remote timestamp */
+ WEBRTC_STATS_PROP_LOCAL_ID, /**< Local id */
+ WEBRTC_STATS_PROP_KIND, /**< Kind (Since 7.5) */
+ WEBRTC_STATS_PROP_ADDRESS, /**< Address of the candidate (Since 7.5) */
+ WEBRTC_STATS_PROP_PORT, /**< Port number of the candidate (Since 7.5) */
+ WEBRTC_STATS_PROP_CANDIDATE_TYPE, /**< Candidate type (Since 7.5) */
+ WEBRTC_STATS_PROP_PRIORITY, /**< Priority (Since 7.5) */
+ WEBRTC_STATS_PROP_PROTOCOL, /**< Protocol, 'udp' or 'tcp' (Since 7.5) */
+ WEBRTC_STATS_PROP_RELAY_PROTOCOL, /**< Relay protocol (Since 7.5) */
+ WEBRTC_STATS_PROP_URL, /**< URL (Since 7.5) */
+ WEBRTC_STATS_PROP_LOCAL_CANDIDATE_ID, /**< Local candidate id associated with the candidate pair (Since 7.5) */
+ WEBRTC_STATS_PROP_REMOTE_CANDIDATE_ID, /**< Remote candidate id associated with the candidate pair (Since 7.5) */
} webrtc_stats_prop_e;
/**
Name: capi-media-webrtc
Summary: A WebRTC library in Tizen Native API
-Version: 0.3.289
+Version: 0.4.1
Release: 0
Group: Multimedia/API
License: Apache-2.0
RET_VAL_IF(_webrtc->state != WEBRTC_STATE_PLAYING, WEBRTC_ERROR_INVALID_STATE, "the state should be PLAYING");
+ if (type_mask == WEBRTC_STATS_TYPE_ALL) {
+ /* exported all types */
+ type_mask = WEBRTC_STATS_TYPE_CODEC | WEBRTC_STATS_TYPE_INBOUND_RTP | WEBRTC_STATS_TYPE_OUTBOUND_RTP |
+ WEBRTC_STATS_TYPE_REMOTE_INBOUND_RTP | WEBRTC_STATS_TYPE_REMOTE_OUTBOUND_RTP | WEBRTC_STATS_TYPE_CANDIDATE_PAIR |
+ WEBRTC_STATS_TYPE_LOCAL_CANDIDATE | WEBRTC_STATS_TYPE_REMOTE_CANDIDATE;
+ }
_webrtcbin_foreach_stats(_webrtc, type_mask, callback, user_data);
LOG_INFO("webrtc[%p] type_mask[0x%x] callback[%p] user_data[%p]", webrtc, type_mask, callback, user_data);
* "ssrc" G_TYPE_STRING the rtp sequence src in use
* "transport-id" G_TYPE_STRING identifier for the associated RTCTransportStats for this stream
* "codec-id" G_TYPE_STRING identifier for the associated RTCCodecStats for this stream
+ * "kind" G_TYPE_STRING either "audio" or "video", depending on the associated transceiver (Since: GStreamer 1.22)
*/
static stats_field_s __stats_rtp_stream_fields[] = {
{ "ssrc", WEBRTC_STATS_PROP_SSRC, 0 },
{ "transport-id", WEBRTC_STATS_PROP_TRANSPORT_ID, 0 },
{ "codec-id", WEBRTC_STATS_PROP_CODEC_ID, 0 },
+ { "kind", WEBRTC_STATS_PROP_KIND, 0 },
{ NULL, 0, 0 }
};
{ "packets-received", WEBRTC_STATS_PROP_PACKETS_RECEIVED, 0 },
{ "packets-lost", WEBRTC_STATS_PROP_PACKETS_LOST, 0 },
{ "packets-discarded", WEBRTC_STATS_PROP_PACKETS_DISCARDED, 0 },
- { "packets-repaired", WEBRTC_STATS_PROP_NOT_EXPORTED, 0 }, /* It'll be exported after RTX verficiation */
+ { "packets-repaired", WEBRTC_STATS_PROP_NOT_EXPORTED, 0 }, /* It'll be exported after RTX verification */
{ "jitter", WEBRTC_STATS_PROP_JITTER, 0 },
{ NULL, 0, 0 }
};
{ NULL, 0, 0 }
};
+/*
+ * Description below is extracted from GstWebRTCBin::get-stats:
+ * RTCIceCandidateStats supported fields (https://www.w3.org/TR/webrtc-stats/#icecandidate-dict*) (Since: GStreamer 1.22)
+ *
+ * "transport-id" G_TYPE_STRING identifier for the associated RTCTransportStats for this stream
+ * "address" G_TYPE_STRING address of the candidate, allowing for IPv4, IPv6 and FQDNs
+ * "port" G_TYPE_UINT port number of the candidate
+ * "candidate-type" G_TYPE_STRING RTCIceCandidateType
+ * "priority" G_TYPE_UINT64 calculated as defined in RFC 5245
+ * "protocol" G_TYPE_STRING Either "udp" or "tcp". Based on the "transport" defined in RFC 5245
+ * "relay-protocol" G_TYPE_STRING protocol used by the endpoint to communicate with the TURN server. Only present for local candidates. Either "udp", "tcp" or "tls"
+ * "url" G_TYPE_STRING URL of the ICE server from which the candidate was obtained. Only present for local candidates
+ */
+static stats_field_s __stats_ice_candidate_fields[] = {
+ { "transport-id", WEBRTC_STATS_PROP_TRANSPORT_ID, 0 },
+ { "address", WEBRTC_STATS_PROP_ADDRESS, 0 },
+ { "port", WEBRTC_STATS_PROP_PORT, 0 },
+ { "candidate-type", WEBRTC_STATS_PROP_CANDIDATE_TYPE, 0 },
+ { "priority", WEBRTC_STATS_PROP_PRIORITY, 0 },
+ { "protocol", WEBRTC_STATS_PROP_PROTOCOL, 0 },
+ { "relay-protocol", WEBRTC_STATS_PROP_RELAY_PROTOCOL, 0 },
+ { "url", WEBRTC_STATS_PROP_URL, 0 },
+ { NULL, 0, 0 }
+};
+
+/*
+ * Description below is extracted from GstWebRTCBin::get-stats:
+ * RTCIceCandidatePairStats supported fields (https://www.w3.org/TR/webrtc-stats/#candidatepair-dict*) (Since: GStreamer 1.22)
+ *
+ * "local-candidate-id" G_TYPE_STRING unique identifier that is associated to the object that was inspected to produce the RTCIceCandidateStats for the local candidate associated with this candidate pair.
+ * "remote-candidate-id" G_TYPE_STRING unique identifier that is associated to the object that was inspected to produce the RTCIceCandidateStats for the remote candidate associated with this candidate pair.
+ */
+static stats_field_s __stats_ice_candidate_pair_fields[] = {
+ { "local-candidate-id", WEBRTC_STATS_PROP_LOCAL_CANDIDATE_ID, 0 },
+ { "remote-candidate-id", WEBRTC_STATS_PROP_REMOTE_CANDIDATE_ID, 0 },
+ { NULL, 0, 0 }
+};
+
static stats_field_s *__stats_all_fields_list[] = {
__stats_common_fields,
__stats_codec_fields,
__stats_remote_inbound_rtp_stream_fields,
__stats_remote_outbound_rtp_stream_fields,
__stats_peer_connection_fields,
+ __stats_ice_candidate_fields,
+ __stats_ice_candidate_pair_fields,
NULL
};
NULL
};
+static stats_field_s *__stats_candidate_pair_fields_list[] = {
+ __stats_common_fields,
+ __stats_ice_candidate_pair_fields,
+ NULL
+};
+
+static stats_field_s *__stats_local_candidate_fields_list[] = {
+ __stats_common_fields,
+ __stats_ice_candidate_fields,
+ NULL
+};
+
+static stats_field_s *__stats_remote_candidate_fields_list[] = {
+ __stats_common_fields,
+ __stats_ice_candidate_fields,
+ NULL
+};
+
typedef struct _promise_userdata_s {
webrtc_s *webrtc;
int type_mask;
static gboolean __stats_candidate_pair_invoke_callback(const GstStructure *s, webrtc_stats_type_e type, stats_field_s **fields_list, promise_userdata_s *user_data)
{
- stats_userdata_s stats_userdata = { .p_userdata = user_data, .type = type, .fields_list = fields_list };
+ stats_userdata_s stats_userdata = { .p_userdata = user_data, .type = type, .fields_list = fields_list, .export = true };
+ gboolean ret;
+
RET_VAL_IF(user_data == NULL, FALSE, "user_data is NULL");
LOG_DEBUG_ENTER();
- return gst_structure_foreach(s, __stats_field_foreach_cb, &stats_userdata);
- /* not implemented */
+ if (user_data->dump)
+ stats_userdata.sub_dump = json_object_new();
+
+ ret = gst_structure_foreach(s, __stats_field_foreach_cb, &stats_userdata);
+
+ if (user_data->dump)
+ json_object_set_object_member(user_data->dump, "candidate-pair", stats_userdata.sub_dump);
+
+ return ret;
}
static gboolean __stats_local_candidate_invoke_callback(const GstStructure *s, webrtc_stats_type_e type, stats_field_s **fields_list, promise_userdata_s *user_data)
{
- stats_userdata_s stats_userdata = { .p_userdata = user_data, .type = type, .fields_list = fields_list };
+ stats_userdata_s stats_userdata = { .p_userdata = user_data, .type = type, .fields_list = fields_list, .export = true };
+ gboolean ret;
+
RET_VAL_IF(user_data == NULL, FALSE, "user_data is NULL");
LOG_DEBUG_ENTER();
- return gst_structure_foreach(s, __stats_field_foreach_cb, &stats_userdata);
- /* not implemented */
+ if (user_data->dump)
+ stats_userdata.sub_dump = json_object_new();
+
+ ret = gst_structure_foreach(s, __stats_field_foreach_cb, &stats_userdata);
+
+ if (user_data->dump)
+ json_object_set_object_member(user_data->dump, "local-candidate", stats_userdata.sub_dump);
+
+ return ret;
}
static gboolean __stats_remote_candidate_invoke_callback(const GstStructure *s, webrtc_stats_type_e type, stats_field_s **fields_list, promise_userdata_s *user_data)
{
- stats_userdata_s stats_userdata = { .p_userdata = user_data, .type = type, .fields_list = fields_list };
+ stats_userdata_s stats_userdata = { .p_userdata = user_data, .type = type, .fields_list = fields_list, .export = true };
+ gboolean ret;
+
RET_VAL_IF(user_data == NULL, FALSE, "user_data is NULL");
LOG_DEBUG_ENTER();
- return gst_structure_foreach(s, __stats_field_foreach_cb, &stats_userdata);
- /* not implemented */
+ if (user_data->dump)
+ stats_userdata.sub_dump = json_object_new();
+
+ ret = gst_structure_foreach(s, __stats_field_foreach_cb, &stats_userdata);
+
+ if (user_data->dump)
+ json_object_set_object_member(user_data->dump, "remote-candidate", stats_userdata.sub_dump);
+
+ return ret;
}
static gboolean __stats_certificate_invoke_callback(const GstStructure *s, webrtc_stats_type_e type, stats_field_s **fields_list, promise_userdata_s *user_data)
#define WEBRTC_STATS_TYPE_DATA_CHANNEL 0x0040 /**< Data Channel */
#define WEBRTC_STATS_TYPE_STREAM 0x0080 /**< Stream */
#define WEBRTC_STATS_TYPE_TRANSPORT 0x0100 /**< Transport */
-#define WEBRTC_STATS_TYPE_CANDIDATE_PAIR 0x0200 /**< Candidate Pair */
-#define WEBRTC_STATS_TYPE_LOCAL_CANDIDATE 0x0400 /**< Local Candidate */
-#define WEBRTC_STATS_TYPE_REMOTE_CANDIDATE 0x0800 /**< Remote Candidate */
#define WEBRTC_STATS_TYPE_CERTIFICATE 0x1000 /**< Certificate */
/* Refer to GstWebRTCStatsType of webrtc_fwd.h */
[GST_WEBRTC_STATS_TRANSPORT] = {
__stats_transport_invoke_callback, WEBRTC_STATS_TYPE_TRANSPORT, __stats_transport_fields_list },
[GST_WEBRTC_STATS_CANDIDATE_PAIR] = {
- __stats_candidate_pair_invoke_callback, WEBRTC_STATS_TYPE_CANDIDATE_PAIR, NULL },
+ __stats_candidate_pair_invoke_callback, WEBRTC_STATS_TYPE_CANDIDATE_PAIR, __stats_candidate_pair_fields_list },
[GST_WEBRTC_STATS_LOCAL_CANDIDATE] = {
- __stats_local_candidate_invoke_callback, WEBRTC_STATS_TYPE_LOCAL_CANDIDATE, NULL },
+ __stats_local_candidate_invoke_callback, WEBRTC_STATS_TYPE_LOCAL_CANDIDATE, __stats_local_candidate_fields_list },
[GST_WEBRTC_STATS_REMOTE_CANDIDATE] = {
- __stats_remote_candidate_invoke_callback, WEBRTC_STATS_TYPE_REMOTE_CANDIDATE, NULL },
+ __stats_remote_candidate_invoke_callback, WEBRTC_STATS_TYPE_REMOTE_CANDIDATE, __stats_remote_candidate_fields_list },
[GST_WEBRTC_STATS_CERTIFICATE] = {
__stats_certificate_invoke_callback, WEBRTC_STATS_TYPE_CERTIFICATE, NULL }
};
[WEBRTC_STATS_TYPE_OUTBOUND_RTP] = "outbound-rtp",
[WEBRTC_STATS_TYPE_REMOTE_INBOUND_RTP] = "remote-inbound-rtp",
[WEBRTC_STATS_TYPE_REMOTE_OUTBOUND_RTP] = "remote-outbound-rtp",
+ [WEBRTC_STATS_TYPE_CANDIDATE_PAIR] = "candidate-pair",
+ [WEBRTC_STATS_TYPE_LOCAL_CANDIDATE] = "local-candidate",
+ [WEBRTC_STATS_TYPE_REMOTE_CANDIDATE] = "remote-candidate",
};
static appdata_s g_ad;
switch (prop_info->type) {
case WEBRTC_STATS_PROP_TYPE_BOOL:
- g_print(" prop[%23s, 0x%08x, type:%d, value:%21d], user_data[%p]\n",
+ g_print(" prop[%23s, %2d, type:%d, value:%21d], user_data[%p]\n",
prop_info->name, prop_info->prop, prop_info->type, prop_info->v_bool, user_data);
break;
case WEBRTC_STATS_PROP_TYPE_INT:
- g_print(" prop[%23s, 0x%08x, type:%d, value:%21d], user_data[%p]\n",
+ g_print(" prop[%23s, %2d, type:%d, value:%21d], user_data[%p]\n",
prop_info->name, prop_info->prop, prop_info->type, prop_info->v_int, user_data);
break;
case WEBRTC_STATS_PROP_TYPE_INT64:
- g_print(" prop[%23s, 0x%08x, type:%d, value:%21"PRId64"], user_data[%p]\n",
+ g_print(" prop[%23s, %2d, type:%d, value:%21"PRId64"], user_data[%p]\n",
prop_info->name, prop_info->prop, prop_info->type, prop_info->v_int64, user_data);
break;
case WEBRTC_STATS_PROP_TYPE_UINT:
- g_print(" prop[%23s, 0x%08x, type:%d, value:%21u], user_data[%p]\n",
+ g_print(" prop[%23s, %2d, type:%d, value:%21u], user_data[%p]\n",
prop_info->name, prop_info->prop, prop_info->type, prop_info->v_uint, user_data);
break;
case WEBRTC_STATS_PROP_TYPE_UINT64:
- g_print(" prop[%23s, 0x%08x, type:%d, value:%21"PRIu64"], user_data[%p]\n",
+ g_print(" prop[%23s, %2d, type:%d, value:%21"PRIu64"], user_data[%p]\n",
prop_info->name, prop_info->prop, prop_info->type, prop_info->v_uint64, user_data);
break;
case WEBRTC_STATS_PROP_TYPE_FLOAT:
- g_print(" prop[%23s, 0x%08x, type:%d, value:%21f], user_data[%p]\n",
+ g_print(" prop[%23s, %2d, type:%d, value:%21f], user_data[%p]\n",
prop_info->name, prop_info->prop, prop_info->type, prop_info->v_float, user_data);
break;
case WEBRTC_STATS_PROP_TYPE_DOUBLE:
- g_print(" prop[%23s, 0x%08x, type:%d, value:%21lf], user_data[%p]\n",
+ g_print(" prop[%23s, %2d, type:%d, value:%21lf], user_data[%p]\n",
prop_info->name, prop_info->prop, prop_info->type, prop_info->v_double, user_data);
break;
case WEBRTC_STATS_PROP_TYPE_STRING:
- g_print(" prop[%23s, 0x%08x, type:%d, value:%21s], user_data[%p]\n",
+ g_print(" prop[%23s, %2d, type:%d, value:%21s], user_data[%p]\n",
prop_info->name, prop_info->prop, prop_info->type, prop_info->v_string, user_data);
break;
default:
- g_printerr("invalid prop_info->type, prop[%23s, 0x%08x, type:%d], user_data[%p]\n",
+ g_printerr("invalid prop_info->type, prop[%23s, %2d, type:%d], user_data[%p]\n",
prop_info->name, prop_info->prop, prop_info->type, user_data);
break;
}
projects / platform / core / api / webrtc.git / commitdiff