webrtc_stats: Update description as per the GStreamer's update

[Version] 0.3.84
[Issue Type] Documentation

Change-Id: I4ad0c0acd8bb3004a8ef860c9e448403b0af3b2d
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index 2e3aa64616a69ad3d8f6953ba12d9fac3537604e..3a73cfe73b11e6d4b49435b3270640966bc86ce5 100644 (file)
--- a/packaging/capi-media-webrtc.spec
+++ b/packaging/capi-media-webrtc.spec
@@ -1,6 +1,6 @@
 Name:       capi-media-webrtc
 Summary:    A WebRTC library in Tizen Native API
-Version:    0.3.83
+Version:    0.3.84
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0

diff --git a/src/webrtc_stats.c b/src/webrtc_stats.c
index ccbc36e7adb85a1f3a1fe2052c8795ce47054518..b81501d13c3266b3c3199ce8f827a4f81cbb8511 100644 (file)
--- a/src/webrtc_stats.c
+++ b/src/webrtc_stats.c
@@ -86,7 +86,7 @@ static stats_field_s __stats_rtp_stream_fields[] = {
  * RTCReceivedStreamStats supported fields (https://w3c.github.io/webrtc-stats/#receivedrtpstats-dict*)
  *
  *  "packets-received"    G_TYPE_UINT64               number of packets received (only for local inbound)
- *  "packets-lost"        G_TYPE_UINT64               number of packets lost
+ *  "packets-lost"        G_TYPE_INT64                number of packets lost
  *  "packets-discarded"   G_TYPE_UINT64               number of packets discarded
  *  "packets-repaired"    G_TYPE_UINT64               number of packets repaired
  *  "jitter"              G_TYPE_DOUBLE               packet jitter measured in seconds
@@ -107,9 +107,9 @@ static stats_field_s __stats_received_rtp_stream_fields[] = {
  *  "remote-id"           G_TYPE_STRING               identifier for the associated RTCRemoteOutboundRTPStreamStats
  *  "bytes-received"      G_TYPE_UINT64               number of bytes received (only for local inbound)
  *  "packets-duplicated"  G_TYPE_UINT64               number of packets duplicated
- *  "fir-count"           G_TYPE_UINT                 FIR requests sent by the receiver
- *  "pli-count"           G_TYPE_UINT                 PLI requests sent by the receiver
- *  "nack-count"          G_TYPE_UINT                 NACK requests sent by the receiver
+ *  "fir-count"           G_TYPE_UINT                 FIR packets sent by the receiver
+ *  "pli-count"           G_TYPE_UINT                 PLI packets sent by the receiver
+ *  "nack-count"          G_TYPE_UINT                 NACK packets sent by the receiver
  */
 static stats_field_s __stats_inbound_rtp_stream_fields[] = {
        { "remote-id", WEBRTC_STATS_PROP_REMOTE_ID, 0 },
@@ -157,7 +157,7 @@ static stats_field_s __stats_outbound_rtp_stream_fields[] = {
  *
  *  "local-id"            G_TYPE_STRING               identifier for the associated RTCOutboundRTPSTreamStats
  *  "round-trip-time"     G_TYPE_DOUBLE               round trip time of packets measured in seconds
- *  "fraction-lost"       G_TYPE_DOUBLE               the fraction packet loss reported for this SSRC
+ *  "fraction-lost"       G_TYPE_DOUBLE               fraction packet loss
  */
 static stats_field_s __stats_remote_inbound_rtp_stream_fields[] = {
        { "local-id", WEBRTC_STATS_PROP_LOCAL_ID, 0 },
@@ -171,7 +171,7 @@ static stats_field_s __stats_remote_inbound_rtp_stream_fields[] = {
  * RTCRemoteOutboundRTPStreamStats supported fields (https://w3c.github.io/webrtc-stats/#remoteoutboundrtpstats-dict*)
  *
  *  "local-id"            G_TYPE_STRING               identifier for the associated RTCInboundRTPSTreamStats
- *  "remote-timestamp"    G_TYPE_DOUBLE               the remote timestamp at which these statistics were sent by the remote endpoint
+ *  "remote-timestamp"    G_TYPE_DOUBLE               remote timestamp the statistics were sent by the remote
  */
 static stats_field_s __stats_remote_outbound_rtp_stream_fields[] = {
        { "local-id", WEBRTC_STATS_PROP_LOCAL_ID, 0 },