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