Implement RTCOutboundRtpStreamStats.retransmitted[Bytes/Packets]Sent.

Spec: https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-retransmittedpacketssent

These are already existed in StreamDataCounters. This CL takes care of
the plumbing of these values to the standard stats collector.

TBR=solenberg@webrtc.org

Bug: webrtc:10447
Change-Id: I27d6c3ee3ab627d306303e6ee67e586ddf31cc81
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132012
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27663}
This commit is contained in:
Henrik Boström
2019-04-17 13:51:53 +02:00
committed by Commit Bot
parent decc07679d
commit cf96e0f87d
13 changed files with 72 additions and 11 deletions

View File

@ -1771,7 +1771,9 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Audio) {
voice_media_info.senders[0].local_stats.push_back(cricket::SsrcSenderInfo());
voice_media_info.senders[0].local_stats[0].ssrc = 1;
voice_media_info.senders[0].packets_sent = 2;
voice_media_info.senders[0].retransmitted_packets_sent = 20;
voice_media_info.senders[0].bytes_sent = 3;
voice_media_info.senders[0].retransmitted_bytes_sent = 30;
voice_media_info.senders[0].codec_payload_type = 42;
RtpCodecParameters codec_parameters;
@ -1799,7 +1801,9 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Audio) {
expected_audio.transport_id = "RTCTransport_TransportName_1";
expected_audio.codec_id = "RTCCodec_AudioMid_Outbound_42";
expected_audio.packets_sent = 2;
expected_audio.retransmitted_packets_sent = 20;
expected_audio.bytes_sent = 3;
expected_audio.retransmitted_bytes_sent = 30;
ASSERT_TRUE(report->Get(expected_audio.id()));
EXPECT_EQ(
@ -1825,7 +1829,9 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Video) {
video_media_info.senders[0].plis_rcvd = 3;
video_media_info.senders[0].nacks_rcvd = 4;
video_media_info.senders[0].packets_sent = 5;
video_media_info.senders[0].retransmitted_packets_sent = 50;
video_media_info.senders[0].bytes_sent = 6;
video_media_info.senders[0].retransmitted_bytes_sent = 60;
video_media_info.senders[0].codec_payload_type = 42;
video_media_info.senders[0].frames_encoded = 8;
video_media_info.senders[0].total_encode_time_ms = 9000;
@ -1865,7 +1871,9 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Video) {
expected_video.pli_count = 3;
expected_video.nack_count = 4;
expected_video.packets_sent = 5;
expected_video.retransmitted_packets_sent = 50;
expected_video.bytes_sent = 6;
expected_video.retransmitted_bytes_sent = 60;
expected_video.frames_encoded = 8;
expected_video.total_encode_time = 9.0;
// |expected_video.content_type| should be undefined.
@ -2038,7 +2046,9 @@ TEST_F(RTCStatsCollectorTest, CollectNoStreamRTCOutboundRTPStreamStats_Audio) {
voice_media_info.senders[0].local_stats.push_back(cricket::SsrcSenderInfo());
voice_media_info.senders[0].local_stats[0].ssrc = 1;
voice_media_info.senders[0].packets_sent = 2;
voice_media_info.senders[0].retransmitted_packets_sent = 20;
voice_media_info.senders[0].bytes_sent = 3;
voice_media_info.senders[0].retransmitted_bytes_sent = 30;
voice_media_info.senders[0].codec_payload_type = 42;
RtpCodecParameters codec_parameters;
@ -2067,7 +2077,9 @@ TEST_F(RTCStatsCollectorTest, CollectNoStreamRTCOutboundRTPStreamStats_Audio) {
expected_audio.transport_id = "RTCTransport_TransportName_1";
expected_audio.codec_id = "RTCCodec_AudioMid_Outbound_42";
expected_audio.packets_sent = 2;
expected_audio.retransmitted_packets_sent = 20;
expected_audio.bytes_sent = 3;
expected_audio.retransmitted_bytes_sent = 30;
ASSERT_TRUE(report->Get(expected_audio.id()));
EXPECT_EQ(