RTCInboundRTPStreamStats's [fir/pli/nack]_count are collected for video.
Previously this was only collected for RTCOutboundRTPStreamStats video,
with no comment saying it was missing for Inbound. (nack_count should be
collected vor audio as well but this is currently not available - there
is already an existing comment about this in rtcstats_objects.h.)
BUG=chromium:657855, chromium:657854, chromium:627816
Review-Url: https://codereview.webrtc.org/2515293002
Cr-Commit-Position: refs/heads/master@{#15185}
diff --git a/webrtc/api/rtcstatscollector.cc b/webrtc/api/rtcstatscollector.cc
index fbc6dbc..b866b68 100644
--- a/webrtc/api/rtcstatscollector.cc
+++ b/webrtc/api/rtcstatscollector.cc
@@ -107,6 +107,7 @@
track_stats->ended = (track.state() == MediaStreamTrackInterface::kEnded);
}
+// Provides the media independent counters (both audio and video).
void SetInboundRTPStreamStatsFromMediaReceiverInfo(
const cricket::MediaReceiverInfo& media_receiver_info,
RTCInboundRTPStreamStats* inbound_stats) {
@@ -126,23 +127,32 @@
void SetInboundRTPStreamStatsFromVoiceReceiverInfo(
const cricket::VoiceReceiverInfo& voice_receiver_info,
- RTCInboundRTPStreamStats* inbound_stats) {
+ RTCInboundRTPStreamStats* inbound_audio) {
SetInboundRTPStreamStatsFromMediaReceiverInfo(
- voice_receiver_info, inbound_stats);
- inbound_stats->media_type = "audio";
- inbound_stats->jitter =
+ voice_receiver_info, inbound_audio);
+ inbound_audio->media_type = "audio";
+ inbound_audio->jitter =
static_cast<double>(voice_receiver_info.jitter_ms) /
rtc::kNumMillisecsPerSec;
+ // |fir_count|, |pli_count| and |sli_count| are only valid for video and are
+ // purposefully left undefined for audio.
}
void SetInboundRTPStreamStatsFromVideoReceiverInfo(
const cricket::VideoReceiverInfo& video_receiver_info,
- RTCInboundRTPStreamStats* inbound_stats) {
+ RTCInboundRTPStreamStats* inbound_video) {
SetInboundRTPStreamStatsFromMediaReceiverInfo(
- video_receiver_info, inbound_stats);
- inbound_stats->media_type = "video";
+ video_receiver_info, inbound_video);
+ inbound_video->media_type = "video";
+ inbound_video->fir_count =
+ static_cast<uint32_t>(video_receiver_info.firs_sent);
+ inbound_video->pli_count =
+ static_cast<uint32_t>(video_receiver_info.plis_sent);
+ inbound_video->nack_count =
+ static_cast<uint32_t>(video_receiver_info.nacks_sent);
}
+// Provides the media independent counters (both audio and video).
void SetOutboundRTPStreamStatsFromMediaSenderInfo(
const cricket::MediaSenderInfo& media_sender_info,
RTCOutboundRTPStreamStats* outbound_stats) {
diff --git a/webrtc/api/rtcstatscollector_unittest.cc b/webrtc/api/rtcstatscollector_unittest.cc
index e032384..64e7e0a 100644
--- a/webrtc/api/rtcstatscollector_unittest.cc
+++ b/webrtc/api/rtcstatscollector_unittest.cc
@@ -1383,6 +1383,9 @@
video_media_info.receivers[0].packets_rcvd = 2;
video_media_info.receivers[0].bytes_rcvd = 3;
video_media_info.receivers[0].fraction_lost = 4.5f;
+ video_media_info.receivers[0].firs_sent = 5;
+ video_media_info.receivers[0].plis_sent = 6;
+ video_media_info.receivers[0].nacks_sent = 7;
EXPECT_CALL(*video_media_channel, GetStats(_))
.WillOnce(DoAll(SetArgPointee<0>(video_media_info), Return(true)));
@@ -1404,23 +1407,26 @@
rtc::scoped_refptr<const RTCStatsReport> report = GetStatsReport();
- RTCInboundRTPStreamStats expected_audio(
+ RTCInboundRTPStreamStats expected_video(
"RTCInboundRTPVideoStream_1", report->timestamp_us());
- expected_audio.ssrc = "1";
- expected_audio.is_remote = false;
- expected_audio.media_type = "video";
- expected_audio.transport_id = "RTCTransport_TransportName_" +
+ expected_video.ssrc = "1";
+ expected_video.is_remote = false;
+ expected_video.media_type = "video";
+ expected_video.transport_id = "RTCTransport_TransportName_" +
rtc::ToString<>(cricket::ICE_CANDIDATE_COMPONENT_RTP);
- expected_audio.packets_received = 2;
- expected_audio.bytes_received = 3;
- expected_audio.fraction_lost = 4.5;
+ expected_video.fir_count = 5;
+ expected_video.pli_count = 6;
+ expected_video.nack_count = 7;
+ expected_video.packets_received = 2;
+ expected_video.bytes_received = 3;
+ expected_video.fraction_lost = 4.5;
- ASSERT(report->Get(expected_audio.id()));
- const RTCInboundRTPStreamStats& audio = report->Get(
- expected_audio.id())->cast_to<RTCInboundRTPStreamStats>();
- EXPECT_EQ(audio, expected_audio);
+ ASSERT(report->Get(expected_video.id()));
+ const RTCInboundRTPStreamStats& video = report->Get(
+ expected_video.id())->cast_to<RTCInboundRTPStreamStats>();
+ EXPECT_EQ(video, expected_video);
- EXPECT_TRUE(report->Get(*expected_audio.transport_id));
+ EXPECT_TRUE(report->Get(*expected_video.transport_id));
}
TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Audio) {