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Implement RTCOutboundRtpStreamStats.retransmitted[Bytes/Packets]Sent.

Browse source 
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
Show stats Download patch file Download diff file Expand all files Collapse all files

2
api/stats/rtcstats_objects.h
View file

@ -442,7 +442,9 @@ class RTC_EXPORT RTCOutboundRTPStreamStats final : public RTCRTPStreamStats {
~RTCOutboundRTPStreamStats() override;
RTCStatsMember<uint32_t> packets_sent;
RTCStatsMember<uint64_t> retransmitted_packets_sent;
RTCStatsMember<uint64_t> bytes_sent;
RTCStatsMember<uint64_t> retransmitted_bytes_sent;
// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7066
RTCStatsMember<double> target_bitrate;
RTCStatsMember<uint32_t> frames_encoded;

2
audio/audio_send_stream.cc
View file

@ -389,7 +389,9 @@ webrtc::AudioSendStream::Stats AudioSendStream::GetStats(
webrtc::CallSendStatistics call_stats = channel_send_->GetRTCPStatistics();
stats.bytes_sent = call_stats.bytesSent;
stats.retransmitted_bytes_sent = call_stats.retransmitted_bytes_sent;
stats.packets_sent = call_stats.packetsSent;
stats.retransmitted_packets_sent = call_stats.retransmitted_packets_sent;
// RTT isn't known until a RTCP report is received. Until then, VoiceEngine
// returns 0 to indicate an error value.
if (call_stats.rttMs > 0) {

27
audio/channel_send.cc
View file

@ -1070,17 +1070,22 @@ CallSendStatistics ChannelSend::GetRTCPStatistics() const {
CallSendStatistics stats = {0};
stats.rttMs = GetRTT();
size_t bytesSent(0);
uint32_t packetsSent(0);
if (_rtpRtcpModule->DataCountersRTP(&bytesSent, &packetsSent) != 0) {
RTC_DLOG(LS_WARNING)
<< "GetRTPStatistics() failed to retrieve RTP datacounters"
<< " => output will not be complete";
}
stats.bytesSent = bytesSent;
stats.packetsSent = packetsSent;
StreamDataCounters rtp_stats;
StreamDataCounters rtx_stats;
_rtpRtcpModule->GetSendStreamDataCounters(&rtp_stats, &rtx_stats);
// TODO(https://crbug.com/webrtc/10525): Bytes sent should only include
// payload bytes, not header and padding bytes.
stats.bytesSent =
rtp_stats.transmitted.payload_bytes +
rtp_stats.transmitted.padding_bytes + rtp_stats.transmitted.header_bytes +
rtx_stats.transmitted.payload_bytes +
rtx_stats.transmitted.padding_bytes + rtx_stats.transmitted.header_bytes;
// TODO(https://crbug.com/webrtc/10555): RTX retransmissions should show up in
// separate outbound-rtp stream objects.
stats.retransmitted_bytes_sent = rtp_stats.retransmitted.payload_bytes;
stats.packetsSent =
rtp_stats.transmitted.packets + rtx_stats.transmitted.packets;
stats.retransmitted_packets_sent = rtp_stats.retransmitted.packets;
return stats;
}

4
audio/channel_send.h
View file

@ -35,7 +35,11 @@ class RtpTransportControllerSendInterface;
struct CallSendStatistics {
int64_t rttMs;
size_t bytesSent;
// https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-retransmittedbytessent
uint64_t retransmitted_bytes_sent;
int packetsSent;
// https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-retransmittedpacketssent
uint64_t retransmitted_packets_sent;
};
// See section 6.4.2 in http://www.ietf.org/rfc/rfc3550.txt for details.

4
call/audio_send_stream.h
View file

@ -42,7 +42,11 @@ class AudioSendStream {
// TODO(solenberg): Harmonize naming and defaults with receive stream stats.
uint32_t local_ssrc = 0;
int64_t bytes_sent = 0;
// https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-retransmittedbytessent
uint64_t retransmitted_bytes_sent = 0;
int32_t packets_sent = 0;
// https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-retransmittedpacketssent
uint64_t retransmitted_packets_sent = 0;
int32_t packets_lost = -1;
float fraction_lost = -1.0f;
std::string codec_name;

4
media/base/media_channel.h
View file

@ -387,7 +387,11 @@ struct MediaSenderInfo {
}
}
int64_t bytes_sent = 0;
// https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-retransmittedbytessent
uint64_t retransmitted_bytes_sent = 0;
int packets_sent = 0;
// https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-retransmittedpacketssent
uint64_t retransmitted_packets_sent = 0;
int packets_lost = 0;
float fraction_lost = 0.0f;
int64_t rtt_ms = 0;

10
media/engine/webrtc_video_engine.cc
View file

@ -2267,10 +2267,20 @@ VideoSenderInfo WebRtcVideoChannel::WebRtcVideoSendStream::GetVideoSenderInfo(
it != stats.substreams.end(); ++it) {
// TODO(pbos): Wire up additional stats, such as padding bytes.
webrtc::VideoSendStream::StreamStats stream_stats = it->second;
// TODO(http://crbug.com/webrtc/10525): Bytes sent should only include
// payload bytes, not header and padding bytes.
info.bytes_sent += stream_stats.rtp_stats.transmitted.payload_bytes +
stream_stats.rtp_stats.transmitted.header_bytes +
stream_stats.rtp_stats.transmitted.padding_bytes;
info.packets_sent += stream_stats.rtp_stats.transmitted.packets;
// TODO(https://crbug.com/webrtc/10555): RTX retransmissions should show up
// in separate outbound-rtp stream objects.
if (!stream_stats.is_rtx && !stream_stats.is_flexfec) {
info.retransmitted_bytes_sent +=
stream_stats.rtp_stats.retransmitted.payload_bytes;
info.retransmitted_packets_sent +=
stream_stats.rtp_stats.retransmitted.packets;
}
info.packets_lost += stream_stats.rtcp_stats.packets_lost;
if (stream_stats.width > info.send_frame_width)
info.send_frame_width = stream_stats.width;

2
media/engine/webrtc_voice_engine.cc
View file

@ -2192,7 +2192,9 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
VoiceSenderInfo sinfo;
sinfo.add_ssrc(stats.local_ssrc);
sinfo.bytes_sent = stats.bytes_sent;
sinfo.retransmitted_bytes_sent = stats.retransmitted_bytes_sent;
sinfo.packets_sent = stats.packets_sent;
sinfo.retransmitted_packets_sent = stats.retransmitted_packets_sent;
sinfo.packets_lost = stats.packets_lost;
sinfo.fraction_lost = stats.fraction_lost;
sinfo.codec_name = stats.codec_name;

2
modules/rtp_rtcp/source/rtp_rtcp_impl.cc
View file

@ -542,6 +542,8 @@ int32_t ModuleRtpRtcpImpl::DataCountersRTP(size_t* bytes_sent,
rtp_sender_->GetDataCounters(&rtp_stats, &rtx_stats);
if (bytes_sent) {
// TODO(http://crbug.com/webrtc/10525): Bytes sent should only include
// payload bytes, not header and padding bytes.
*bytes_sent = rtp_stats.transmitted.payload_bytes +
rtp_stats.transmitted.padding_bytes +
rtp_stats.transmitted.header_bytes +

4
pc/rtc_stats_collector.cc
View file

@ -295,8 +295,12 @@ void SetOutboundRTPStreamStatsFromMediaSenderInfo(
outbound_stats->is_remote = false;
outbound_stats->packets_sent =
static_cast<uint32_t>(media_sender_info.packets_sent);
outbound_stats->retransmitted_packets_sent =
media_sender_info.retransmitted_packets_sent;
outbound_stats->bytes_sent =
static_cast<uint64_t>(media_sender_info.bytes_sent);
outbound_stats->retransmitted_bytes_sent =
media_sender_info.retransmitted_bytes_sent;
}
void SetOutboundRTPStreamStatsFromVoiceSenderInfo(

12
pc/rtc_stats_collector_unittest.cc
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(

4
pc/rtc_stats_integrationtest.cc
View file

@ -762,7 +762,11 @@ class RTCStatsReportVerifier {
verifier.TestMemberIsUndefined(outbound_stream.qp_sum);
}
verifier.TestMemberIsNonNegative<uint32_t>(outbound_stream.packets_sent);
verifier.TestMemberIsNonNegative<uint64_t>(
outbound_stream.retransmitted_packets_sent);
verifier.TestMemberIsNonNegative<uint64_t>(outbound_stream.bytes_sent);
verifier.TestMemberIsNonNegative<uint64_t>(
outbound_stream.retransmitted_bytes_sent);
verifier.TestMemberIsUndefined(outbound_stream.target_bitrate);
if (outbound_stream.media_type.is_defined() &&
*outbound_stream.media_type == "video") {

6
stats/rtcstats_objects.cc
View file

@ -652,7 +652,9 @@ RTCInboundRTPStreamStats::~RTCInboundRTPStreamStats() {}
WEBRTC_RTCSTATS_IMPL(
RTCOutboundRTPStreamStats, RTCRTPStreamStats, "outbound-rtp",
&packets_sent,
&retransmitted_packets_sent,
&bytes_sent,
&retransmitted_bytes_sent,
&target_bitrate,
&frames_encoded,
&total_encode_time,
@ -667,7 +669,9 @@ RTCOutboundRTPStreamStats::RTCOutboundRTPStreamStats(std::string&& id,
int64_t timestamp_us)
: RTCRTPStreamStats(std::move(id), timestamp_us),
packets_sent("packetsSent"),
retransmitted_packets_sent("retransmittedPacketsSent"),
bytes_sent("bytesSent"),
retransmitted_bytes_sent("retransmittedBytesSent"),
target_bitrate("targetBitrate"),
frames_encoded("framesEncoded"),
total_encode_time("totalEncodeTime"),
@ -677,7 +681,9 @@ RTCOutboundRTPStreamStats::RTCOutboundRTPStreamStats(
const RTCOutboundRTPStreamStats& other)
: RTCRTPStreamStats(other),
packets_sent(other.packets_sent),
retransmitted_packets_sent(other.retransmitted_packets_sent),
bytes_sent(other.bytes_sent),
retransmitted_bytes_sent(other.retransmitted_bytes_sent),
target_bitrate(other.target_bitrate),
frames_encoded(other.frames_encoded),
total_encode_time(other.total_encode_time),