admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Revert "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received"

Browse Source 
This reverts commit fbde32e596f06893d6dda13eb7d29f4c251cf08b.

Reason for revert: It seems to break WebRTC FYI tests in Chromium.

https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Linux%20Tester/4763

Original change's description:
> Fix GetStats bytesSent/Received, wireup headerBytesSent/Received
> 
> Changes the standard GetStats, legacy GetStats unchanged.
> 
> Bug: webrtc:10525
> Change-Id: Ie10fe8079f1d8b4cc6bbe513f6a2fc91477b5441
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156084
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29462}

TBR=kwiberg@webrtc.org,hbos@webrtc.org,nisse@webrtc.org,hta@webrtc.org

Change-Id: I6a983ea4d5ff38e49f096a8ff5cd9b426768f955
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10525
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157043
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29478}
This commit is contained in:
Mirko Bonadei 2019-10-15 08:54:49 +00:00 committed by Commit Bot
parent 55c7694a9f
commit ef0627fb50
25 changed files with 123 additions and 198 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

2
api/stats/rtcstats_objects.h
View File

@ -413,7 +413,6 @@ class RTC_EXPORT RTCInboundRTPStreamStats final : public RTCRTPStreamStats {
RTCStatsMember<uint64_t> fec_packets_received;
RTCStatsMember<uint64_t> fec_packets_discarded;
RTCStatsMember<uint64_t> bytes_received;
RTCStatsMember<uint64_t> header_bytes_received;
RTCStatsMember<int32_t> packets_lost; // Signed per RFC 3550
RTCStatsMember<double> last_packet_received_timestamp;
// TODO(hbos): Collect and populate this value for both "audio" and "video",
@ -467,7 +466,6 @@ class RTC_EXPORT RTCOutboundRTPStreamStats final : public RTCRTPStreamStats {
RTCStatsMember<uint32_t> packets_sent;
RTCStatsMember<uint64_t> retransmitted_packets_sent;
RTCStatsMember<uint64_t> bytes_sent;
RTCStatsMember<uint64_t> header_bytes_sent;
RTCStatsMember<uint64_t> retransmitted_bytes_sent;
// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7066
RTCStatsMember<double> target_bitrate;

6
audio/audio_receive_stream.cc
View File

@ -188,11 +188,7 @@ webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
return stats;
}
stats.payload_bytes_rcvd = call_stats.payload_bytes_rcvd;
stats.header_and_padding_bytes_rcvd =
call_stats.header_and_padding_bytes_rcvd;
stats.bytes_rcvd =
stats.payload_bytes_rcvd + stats.header_and_padding_bytes_rcvd;
stats.bytes_rcvd = call_stats.bytesReceived;
stats.packets_rcvd = call_stats.packetsReceived;
stats.packets_lost = call_stats.cumulativeLost;
stats.capture_start_ntp_time_ms = call_stats.capture_start_ntp_time_ms_;

6
audio/audio_receive_stream_unittest.cc
View File

@ -63,7 +63,7 @@ const unsigned int kSpeechOutputLevel = 99;
const double kTotalOutputEnergy = 0.25;
const double kTotalOutputDuration = 0.5;
const CallReceiveStatistics kCallStats = {678, 234, -12, 567, 78, 890, 123};
const CallReceiveStatistics kCallStats = {678, 234, -12, 567, 890, 123};
const std::pair<int, SdpAudioFormat> kReceiveCodec = {
123,
{"codec_name_recv", 96000, 0}};
@ -266,9 +266,7 @@ TEST(AudioReceiveStreamTest, GetStats) {
helper.SetupMockForGetStats();
AudioReceiveStream::Stats stats = recv_stream->GetStats();
EXPECT_EQ(kRemoteSsrc, stats.remote_ssrc);
EXPECT_EQ(kCallStats.payload_bytes_rcvd, stats.payload_bytes_rcvd);
EXPECT_EQ(kCallStats.header_and_padding_bytes_rcvd,
stats.header_and_padding_bytes_rcvd);
EXPECT_EQ(static_cast<int64_t>(kCallStats.bytesReceived), stats.bytes_rcvd);
EXPECT_EQ(static_cast<uint32_t>(kCallStats.packetsReceived),
stats.packets_rcvd);
EXPECT_EQ(kCallStats.cumulativeLost, stats.packets_lost);

6
audio/audio_send_stream.cc
View File

@ -440,11 +440,7 @@ webrtc::AudioSendStream::Stats AudioSendStream::GetStats(
stats.target_bitrate_bps = channel_send_->GetBitrate();
webrtc::CallSendStatistics call_stats = channel_send_->GetRTCPStatistics();
stats.payload_bytes_sent = call_stats.payload_bytes_sent;
stats.header_and_padding_bytes_sent =
call_stats.header_and_padding_bytes_sent;
stats.bytes_sent =
stats.payload_bytes_sent + stats.header_and_padding_bytes_sent;
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;

6
audio/audio_send_stream_unittest.cc
View File

@ -64,7 +64,7 @@ const double kEchoReturnLoss = -65;
const double kEchoReturnLossEnhancement = 101;
const double kResidualEchoLikelihood = -1.0f;
const double kResidualEchoLikelihoodMax = 23.0f;
const CallSendStatistics kCallStats = {112, 12, 13456, 17890};
const CallSendStatistics kCallStats = {112, 13456, 17890};
const ReportBlock kReportBlock = {456, 780, 123, 567, 890, 132, 143, 13354};
const int kTelephoneEventPayloadType = 123;
const int kTelephoneEventPayloadFrequency = 65432;
@ -414,9 +414,7 @@ TEST(AudioSendStreamTest, GetStats) {
helper.SetupMockForGetStats();
AudioSendStream::Stats stats = send_stream->GetStats(true);
EXPECT_EQ(kSsrc, stats.local_ssrc);
EXPECT_EQ(kCallStats.payload_bytes_sent, stats.payload_bytes_sent);
EXPECT_EQ(kCallStats.header_and_padding_bytes_sent,
stats.header_and_padding_bytes_sent);
EXPECT_EQ(static_cast<int64_t>(kCallStats.bytesSent), stats.bytes_sent);
EXPECT_EQ(kCallStats.packetsSent, stats.packets_sent);
EXPECT_EQ(kReportBlock.cumulative_num_packets_lost, stats.packets_lost);
EXPECT_EQ(Q8ToFloat(kReportBlock.fraction_lost), stats.fraction_lost);

25
audio/channel_receive.cc
View File

@ -43,6 +43,7 @@
#include "rtc_base/race_checker.h"
#include "rtc_base/thread_checker.h"
#include "rtc_base/time_utils.h"
#include "system_wrappers/include/field_trial.h"
#include "system_wrappers/include/metrics.h"
namespace webrtc {
@ -56,6 +57,11 @@ constexpr double kAudioSampleDurationSeconds = 0.01;
constexpr int kVoiceEngineMinMinPlayoutDelayMs = 0;
constexpr int kVoiceEngineMaxMinPlayoutDelayMs = 10000;
// Field trial which controls whether to report standard-compliant bytes
// sent/received per stream. If enabled, padding and headers are not included
// in bytes sent or received.
constexpr char kUseStandardBytesStats[] = "WebRTC-UseStandardBytesStats";
RTPHeader CreateRTPHeaderForMediaTransportFrame(
const MediaTransportEncodedAudioFrame& frame,
uint64_t channel_id) {
@ -272,6 +278,8 @@ class ChannelReceive : public ChannelReceiveInterface,
// E2EE Audio Frame Decryption
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor_;
webrtc::CryptoOptions crypto_options_;
const bool use_standard_bytes_stats_;
};
void ChannelReceive::OnReceivedPayloadData(
@ -476,7 +484,9 @@ ChannelReceive::ChannelReceive(
associated_send_channel_(nullptr),
media_transport_config_(media_transport_config),
frame_decryptor_(frame_decryptor),
crypto_options_(crypto_options) {
crypto_options_(crypto_options),
use_standard_bytes_stats_(
webrtc::field_trial::IsEnabled(kUseStandardBytesStats)) {
// TODO(nisse): Use _moduleProcessThreadPtr instead?
module_process_thread_checker_.Detach();
@ -724,17 +734,16 @@ CallReceiveStatistics ChannelReceive::GetRTCPStatistics() const {
// --- Data counters
if (statistician) {
stats.payload_bytes_rcvd = rtp_stats.packet_counter.payload_bytes;
stats.header_and_padding_bytes_rcvd =
rtp_stats.packet_counter.header_bytes +
rtp_stats.packet_counter.padding_bytes;
if (use_standard_bytes_stats_) {
stats.bytesReceived = rtp_stats.packet_counter.payload_bytes;
} else {
stats.bytesReceived = rtp_stats.packet_counter.TotalBytes();
}
stats.packetsReceived = rtp_stats.packet_counter.packets;
stats.last_packet_received_timestamp_ms =
rtp_stats.last_packet_received_timestamp_ms;
} else {
stats.payload_bytes_rcvd = 0;
stats.header_and_padding_bytes_rcvd = 0;
stats.bytesReceived = 0;
stats.packetsReceived = 0;
stats.last_packet_received_timestamp_ms = absl::nullopt;
}

3
audio/channel_receive.h
View File

@ -54,8 +54,7 @@ struct CallReceiveStatistics {
unsigned int cumulativeLost;
unsigned int jitterSamples;
int64_t rttMs;
int64_t payload_bytes_rcvd = 0;
int64_t header_and_padding_bytes_rcvd = 0;
size_t bytesReceived;
int packetsReceived;
// The capture ntp time (in local timebase) of the first played out audio
// frame.

25
audio/channel_send.cc
View File

@ -52,6 +52,11 @@ namespace {
constexpr int64_t kMaxRetransmissionWindowMs = 1000;
constexpr int64_t kMinRetransmissionWindowMs = 30;
// Field trial which controls whether to report standard-compliant bytes
// sent/received per stream. If enabled, padding and headers are not included
// in bytes sent or received.
constexpr char kUseStandardBytesStats[] = "WebRTC-UseStandardBytesStats";
MediaTransportEncodedAudioFrame::FrameType
MediaTransportFrameTypeForWebrtcFrameType(webrtc::AudioFrameType frame_type) {
switch (frame_type) {
@ -258,6 +263,7 @@ class ChannelSend : public ChannelSendInterface,
rtc::ThreadChecker construction_thread_;
const bool use_twcc_plr_for_ana_;
const bool use_standard_bytes_stats_;
bool encoder_queue_is_active_ RTC_GUARDED_BY(encoder_queue_) = false;
@ -603,6 +609,8 @@ ChannelSend::ChannelSend(Clock* clock,
new RateLimiter(clock, kMaxRetransmissionWindowMs)),
use_twcc_plr_for_ana_(
webrtc::field_trial::FindFullName("UseTwccPlrForAna") == "Enabled"),
use_standard_bytes_stats_(
webrtc::field_trial::IsEnabled(kUseStandardBytesStats)),
media_transport_config_(media_transport_config),
frame_encryptor_(frame_encryptor),
crypto_options_(crypto_options),
@ -1011,12 +1019,17 @@ CallSendStatistics ChannelSend::GetRTCPStatistics() const {
StreamDataCounters rtp_stats;
StreamDataCounters rtx_stats;
_rtpRtcpModule->GetSendStreamDataCounters(&rtp_stats, &rtx_stats);
stats.payload_bytes_sent =
rtp_stats.transmitted.payload_bytes + rtx_stats.transmitted.payload_bytes;
stats.header_and_padding_bytes_sent =
rtp_stats.transmitted.padding_bytes + rtp_stats.transmitted.header_bytes +
rtx_stats.transmitted.padding_bytes + rtx_stats.transmitted.header_bytes;
if (use_standard_bytes_stats_) {
stats.bytesSent = rtp_stats.transmitted.payload_bytes +
rtx_stats.transmitted.payload_bytes;
} else {
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;

3
audio/channel_send.h
View File

@ -36,8 +36,7 @@ class RtpTransportControllerSendInterface;
struct CallSendStatistics {
int64_t rttMs;
int64_t payload_bytes_sent;
int64_t header_and_padding_bytes_sent;
size_t bytesSent;
// https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-retransmittedbytessent
uint64_t retransmitted_bytes_sent;
int packetsSent;

4
audio/test/audio_stats_test.cc
View File

@ -46,7 +46,7 @@ class NoLossTest : public AudioEndToEndTest {
void OnStreamsStopped() override {
AudioSendStream::Stats send_stats = send_stream()->GetStats();
EXPECT_PRED2(IsNear, kBytesSent, send_stats.payload_bytes_sent);
EXPECT_PRED2(IsNear, kBytesSent, send_stats.bytes_sent);
EXPECT_PRED2(IsNear, kPacketsSent, send_stats.packets_sent);
EXPECT_EQ(0, send_stats.packets_lost);
EXPECT_EQ(0.0f, send_stats.fraction_lost);
@ -66,7 +66,7 @@ class NoLossTest : public AudioEndToEndTest {
EXPECT_EQ(false, send_stats.typing_noise_detected);
AudioReceiveStream::Stats recv_stats = receive_stream()->GetStats();
EXPECT_PRED2(IsNear, kBytesSent, recv_stats.payload_bytes_rcvd);
EXPECT_PRED2(IsNear, kBytesSent, recv_stats.bytes_rcvd);
EXPECT_PRED2(IsNear, kPacketsSent, recv_stats.packets_rcvd);
EXPECT_EQ(0u, recv_stats.packets_lost);
EXPECT_EQ("opus", send_stats.codec_name);

6
call/audio_receive_stream.h
View File

@ -36,11 +36,7 @@ class AudioReceiveStream {
Stats();
~Stats();
uint32_t remote_ssrc = 0;
// TODO(nisse): Sum of below two values. Deprecated, delete as soon as
// downstream applications are updated.
int64_t bytes_rcvd;
int64_t payload_bytes_rcvd = 0;
int64_t header_and_padding_bytes_rcvd = 0;
int64_t bytes_rcvd = 0;
uint32_t packets_rcvd = 0;
uint64_t fec_packets_received = 0;
uint64_t fec_packets_discarded = 0;

6
call/audio_send_stream.h
View File

@ -43,11 +43,7 @@ class AudioSendStream {
// TODO(solenberg): Harmonize naming and defaults with receive stream stats.
uint32_t local_ssrc = 0;
// TODO(nisse): Sum of below two values. Deprecated, delete as soon as
// downstream applications are updated.
int64_t bytes_sent;
int64_t payload_bytes_sent = 0;
int64_t header_and_padding_bytes_sent = 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;

16
media/base/media_channel.h
View File

@ -393,13 +393,7 @@ struct MediaSenderInfo {
return 0;
}
}
// TODO(nisse): Sum of below two values. Deprecated, delete as soon as
// downstream applications are updated.
int64_t bytes_sent;
// https://w3c.github.io/webrtc-stats/#dom-rtcsentrtpstreamstats-bytessent
int64_t payload_bytes_sent = 0;
// https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-headerbytessent
int64_t header_and_padding_bytes_sent = 0;
int64_t bytes_sent = 0;
// https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-retransmittedbytessent
uint64_t retransmitted_bytes_sent = 0;
int packets_sent = 0;
@ -453,13 +447,7 @@ struct MediaReceiverInfo {
}
}
// TODO(nisse): Sum of below two values. Deprecated, delete as soon as
// downstream applications are updated.
int64_t bytes_rcvd;
// https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-bytesreceived
int64_t payload_bytes_rcvd = 0;
// https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-headerbytesreceived
int64_t header_and_padding_bytes_rcvd = 0;
int64_t bytes_rcvd = 0;
int packets_rcvd = 0;
int packets_lost = 0;
// TODO(bugs.webrtc.org/10679): Unused, delete as soon as downstream code is

37
media/engine/webrtc_video_engine.cc
View File

@ -48,6 +48,11 @@ namespace {
const int kMinLayerSize = 16;
// Field trial which controls whether to report standard-compliant bytes
// sent/received per stream. If enabled, padding and headers are not included
// in bytes sent or received.
constexpr char kUseStandardBytesStats[] = "WebRTC-UseStandardBytesStats";
// If this field trial is enabled, we will enable sending FlexFEC and disable
// sending ULPFEC whenever the former has been negotiated in the SDPs.
bool IsFlexfecFieldTrialEnabled() {
@ -1803,7 +1808,9 @@ WebRtcVideoChannel::WebRtcVideoSendStream::WebRtcVideoSendStream(
encoder_sink_(nullptr),
parameters_(std::move(config), options, max_bitrate_bps, codec_settings),
rtp_parameters_(CreateRtpParametersWithEncodings(sp)),
sending_(false) {
sending_(false),
use_standard_bytes_stats_(
webrtc::field_trial::IsEnabled(kUseStandardBytesStats)) {
// Maximum packet size may come in RtpConfig from external transport, for
// example from QuicTransportInterface implementation, so do not exceed
// given max_packet_size.
@ -2372,10 +2379,13 @@ 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;
info.payload_bytes_sent += stream_stats.rtp_stats.transmitted.payload_bytes;
info.header_and_padding_bytes_sent +=
stream_stats.rtp_stats.transmitted.header_bytes +
stream_stats.rtp_stats.transmitted.padding_bytes;
if (use_standard_bytes_stats_) {
info.bytes_sent += stream_stats.rtp_stats.transmitted.payload_bytes;
} else {
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;
info.total_packet_send_delay_ms += stream_stats.total_packet_send_delay_ms;
// TODO(https://crbug.com/webrtc/10555): RTX retransmissions should show up
@ -2399,8 +2409,6 @@ VideoSenderInfo WebRtcVideoChannel::WebRtcVideoSendStream::GetVideoSenderInfo(
info.report_block_datas.push_back(stream_stats.report_block_data.value());
}
}
info.bytes_sent =
info.payload_bytes_sent + info.header_and_padding_bytes_sent;
if (!stats.substreams.empty()) {
// TODO(pbos): Report fraction lost per SSRC.
@ -2493,7 +2501,9 @@ WebRtcVideoChannel::WebRtcVideoReceiveStream::WebRtcVideoReceiveStream(
decoder_factory_(decoder_factory),
sink_(NULL),
first_frame_timestamp_(-1),
estimated_remote_start_ntp_time_ms_(0) {
estimated_remote_start_ntp_time_ms_(0),
use_standard_bytes_stats_(
webrtc::field_trial::IsEnabled(kUseStandardBytesStats)) {
config_.renderer = this;
ConfigureCodecs(recv_codecs);
ConfigureFlexfecCodec(flexfec_config.payload_type);
@ -2789,12 +2799,11 @@ WebRtcVideoChannel::WebRtcVideoReceiveStream::GetVideoReceiverInfo(
if (stats.current_payload_type != -1) {
info.codec_payload_type = stats.current_payload_type;
}
info.payload_bytes_rcvd = stats.rtp_stats.packet_counter.payload_bytes;
info.header_and_padding_bytes_rcvd =
stats.rtp_stats.packet_counter.header_bytes +
stats.rtp_stats.packet_counter.padding_bytes;
info.bytes_rcvd =
info.payload_bytes_rcvd + info.header_and_padding_bytes_rcvd;
if (use_standard_bytes_stats_) {
info.bytes_rcvd = stats.rtp_stats.packet_counter.payload_bytes;
} else {
info.bytes_rcvd = stats.rtp_stats.packet_counter.TotalBytes();
}
info.packets_rcvd = stats.rtp_stats.packet_counter.packets;
info.packets_lost = stats.rtp_stats.packets_lost;

4
media/engine/webrtc_video_engine.h
View File

@ -380,6 +380,8 @@ class WebRtcVideoChannel : public VideoMediaChannel,
bool sending_ RTC_GUARDED_BY(&thread_checker_);
const bool use_standard_bytes_stats_;
// In order for the |invoker_| to protect other members from being
// destructed as they are used in asynchronous tasks it has to be destructed
// first.
@ -469,6 +471,8 @@ class WebRtcVideoChannel : public VideoMediaChannel,
// Start NTP time is estimated as current remote NTP time (estimated from
// RTCP) minus the elapsed time, as soon as remote NTP time is available.
int64_t estimated_remote_start_ntp_time_ms_ RTC_GUARDED_BY(sink_lock_);
const bool use_standard_bytes_stats_;
};
void Construct(webrtc::Call* call, WebRtcVideoEngine* engine);

17
media/engine/webrtc_video_engine_unittest.cc
View File

@ -1599,6 +1599,8 @@ TEST_F(WebRtcVideoChannelBaseTest, InvalidRecvBufferSize) {
// Test that stats work properly for a 1-1 call.
TEST_F(WebRtcVideoChannelBaseTest, GetStats) {
webrtc::test::ScopedFieldTrials field_trials(
"WebRTC-UseStandardBytesStats/Enabled/");
SetUp();
const int kDurationSec = 3;
@ -1611,7 +1613,7 @@ TEST_F(WebRtcVideoChannelBaseTest, GetStats) {
ASSERT_EQ(1U, info.senders.size());
// TODO(whyuan): bytes_sent and bytes_rcvd are different. Are both payload?
// For webrtc, bytes_sent does not include the RTP header length.
EXPECT_EQ(info.senders[0].payload_bytes_sent,
EXPECT_EQ(info.senders[0].bytes_sent,
NumRtpBytes() - kRtpHeaderSize * NumRtpPackets());
EXPECT_EQ(NumRtpPackets(), info.senders[0].packets_sent);
EXPECT_EQ(0.0, info.senders[0].fraction_lost);
@ -1636,7 +1638,7 @@ TEST_F(WebRtcVideoChannelBaseTest, GetStats) {
ASSERT_TRUE(info.receivers[0].codec_payload_type);
EXPECT_EQ(DefaultCodec().id, *info.receivers[0].codec_payload_type);
EXPECT_EQ(NumRtpBytes() - kRtpHeaderSize * NumRtpPackets(),
info.receivers[0].payload_bytes_rcvd);
info.receivers[0].bytes_rcvd);
EXPECT_EQ(NumRtpPackets(), info.receivers[0].packets_rcvd);
EXPECT_EQ(0, info.receivers[0].packets_lost);
// TODO(asapersson): Not set for webrtc. Handle missing stats.
@ -1657,6 +1659,8 @@ TEST_F(WebRtcVideoChannelBaseTest, GetStats) {
// Test that stats work properly for a conf call with multiple recv streams.
TEST_F(WebRtcVideoChannelBaseTest, GetStatsMultipleRecvStreams) {
webrtc::test::ScopedFieldTrials field_trials(
"WebRTC-UseStandardBytesStats/Enabled/");
SetUp();
cricket::FakeVideoRenderer renderer1, renderer2;
@ -1690,7 +1694,7 @@ TEST_F(WebRtcVideoChannelBaseTest, GetStatsMultipleRecvStreams) {
// TODO(whyuan): bytes_sent and bytes_rcvd are different. Are both payload?
// For webrtc, bytes_sent does not include the RTP header length.
EXPECT_EQ_WAIT(NumRtpBytes() - kRtpHeaderSize * NumRtpPackets(),
GetSenderStats(0).payload_bytes_sent, kTimeout);
GetSenderStats(0).bytes_sent, kTimeout);
EXPECT_EQ_WAIT(NumRtpPackets(), GetSenderStats(0).packets_sent, kTimeout);
EXPECT_EQ(kVideoWidth, GetSenderStats(0).send_frame_width);
EXPECT_EQ(kVideoHeight, GetSenderStats(0).send_frame_height);
@ -1700,7 +1704,7 @@ TEST_F(WebRtcVideoChannelBaseTest, GetStatsMultipleRecvStreams) {
EXPECT_EQ(1U, GetReceiverStats(i).ssrcs().size());
EXPECT_EQ(i + 1, GetReceiverStats(i).ssrcs()[0]);
EXPECT_EQ_WAIT(NumRtpBytes() - kRtpHeaderSize * NumRtpPackets(),
GetReceiverStats(i).payload_bytes_rcvd, kTimeout);
GetReceiverStats(i).bytes_rcvd, kTimeout);
EXPECT_EQ_WAIT(NumRtpPackets(), GetReceiverStats(i).packets_rcvd, kTimeout);
EXPECT_EQ_WAIT(kVideoWidth, GetReceiverStats(i).frame_width, kTimeout);
EXPECT_EQ_WAIT(kVideoHeight, GetReceiverStats(i).frame_height, kTimeout);
@ -5278,6 +5282,9 @@ TEST_F(WebRtcVideoChannelTest, GetStatsTranslatesDecodeStatsCorrectly) {
}
TEST_F(WebRtcVideoChannelTest, GetStatsTranslatesReceivePacketStatsCorrectly) {
webrtc::test::ScopedFieldTrials field_trials(
"WebRTC-UseStandardBytesStats/Enabled/");
FakeVideoReceiveStream* stream = AddRecvStream();
webrtc::VideoReceiveStream::Stats stats;
stats.rtp_stats.packet_counter.payload_bytes = 2;
@ -5290,7 +5297,7 @@ TEST_F(WebRtcVideoChannelTest, GetStatsTranslatesReceivePacketStatsCorrectly) {
cricket::VideoMediaInfo info;
ASSERT_TRUE(channel_->GetStats(&info));
EXPECT_EQ(stats.rtp_stats.packet_counter.payload_bytes,
rtc::checked_cast<size_t>(info.receivers[0].payload_bytes_rcvd));
rtc::checked_cast<size_t>(info.receivers[0].bytes_rcvd));
EXPECT_EQ(stats.rtp_stats.packet_counter.packets,
rtc::checked_cast<unsigned int>(info.receivers[0].packets_rcvd));
EXPECT_EQ(stats.rtp_stats.packets_lost, info.receivers[0].packets_lost);

10
media/engine/webrtc_voice_engine.cc
View File

@ -2158,10 +2158,7 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
stream.second->GetStats(recv_streams_.size() > 0);
VoiceSenderInfo sinfo;
sinfo.add_ssrc(stats.local_ssrc);
sinfo.payload_bytes_sent = stats.payload_bytes_sent;
sinfo.header_and_padding_bytes_sent = stats.header_and_padding_bytes_sent;
sinfo.bytes_sent =
sinfo.payload_bytes_sent + sinfo.header_and_padding_bytes_sent;
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;
@ -2204,10 +2201,7 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
webrtc::AudioReceiveStream::Stats stats = stream.second->GetStats();
VoiceReceiverInfo rinfo;
rinfo.add_ssrc(stats.remote_ssrc);
rinfo.payload_bytes_rcvd = stats.payload_bytes_rcvd;
rinfo.header_and_padding_bytes_rcvd = stats.header_and_padding_bytes_rcvd;
rinfo.bytes_rcvd =
rinfo.payload_bytes_rcvd + rinfo.header_and_padding_bytes_rcvd;
rinfo.bytes_rcvd = stats.bytes_rcvd;
rinfo.packets_rcvd = stats.packets_rcvd;
rinfo.fec_packets_received = stats.fec_packets_received;
rinfo.fec_packets_discarded = stats.fec_packets_discarded;

14
media/engine/webrtc_voice_engine_unittest.cc
View File

@ -566,8 +566,7 @@ class WebRtcVoiceEngineTestFake : public ::testing::Test {
webrtc::AudioSendStream::Stats GetAudioSendStreamStats() const {
webrtc::AudioSendStream::Stats stats;
stats.local_ssrc = 12;
stats.payload_bytes_sent = 345;
stats.header_and_padding_bytes_sent = 56;
stats.bytes_sent = 345;
stats.packets_sent = 678;
stats.packets_lost = 9012;
stats.fraction_lost = 34.56f;
@ -601,9 +600,7 @@ class WebRtcVoiceEngineTestFake : public ::testing::Test {
bool is_sending) {
const auto stats = GetAudioSendStreamStats();
EXPECT_EQ(info.ssrc(), stats.local_ssrc);
EXPECT_EQ(info.payload_bytes_sent, stats.payload_bytes_sent);
EXPECT_EQ(info.header_and_padding_bytes_sent,
stats.header_and_padding_bytes_sent);
EXPECT_EQ(info.bytes_sent, stats.bytes_sent);
EXPECT_EQ(info.packets_sent, stats.packets_sent);
EXPECT_EQ(info.packets_lost, stats.packets_lost);
EXPECT_EQ(info.fraction_lost, stats.fraction_lost);
@ -645,8 +642,7 @@ class WebRtcVoiceEngineTestFake : public ::testing::Test {
webrtc::AudioReceiveStream::Stats GetAudioReceiveStreamStats() const {
webrtc::AudioReceiveStream::Stats stats;
stats.remote_ssrc = 123;
stats.payload_bytes_rcvd = 456;
stats.header_and_padding_bytes_rcvd = 67;
stats.bytes_rcvd = 456;
stats.packets_rcvd = 768;
stats.packets_lost = 101;
stats.codec_name = "codec_name_recv";
@ -686,9 +682,7 @@ class WebRtcVoiceEngineTestFake : public ::testing::Test {
void VerifyVoiceReceiverInfo(const cricket::VoiceReceiverInfo& info) {
const auto stats = GetAudioReceiveStreamStats();
EXPECT_EQ(info.ssrc(), stats.remote_ssrc);
EXPECT_EQ(info.payload_bytes_rcvd, stats.payload_bytes_rcvd);
EXPECT_EQ(info.header_and_padding_bytes_rcvd,
stats.header_and_padding_bytes_rcvd);
EXPECT_EQ(info.bytes_rcvd, stats.bytes_rcvd);
EXPECT_EQ(rtc::checked_cast<unsigned int>(info.packets_rcvd),
stats.packets_rcvd);
EXPECT_EQ(rtc::checked_cast<unsigned int>(info.packets_lost),

8
pc/rtc_stats_collector.cc
View File

@ -256,9 +256,7 @@ void SetInboundRTPStreamStatsFromMediaReceiverInfo(
inbound_stats->packets_received =
static_cast<uint32_t>(media_receiver_info.packets_rcvd);
inbound_stats->bytes_received =
static_cast<uint64_t>(media_receiver_info.payload_bytes_rcvd);
inbound_stats->header_bytes_received =
static_cast<uint64_t>(media_receiver_info.header_and_padding_bytes_rcvd);
static_cast<uint64_t>(media_receiver_info.bytes_rcvd);
inbound_stats->packets_lost =
static_cast<int32_t>(media_receiver_info.packets_lost);
}
@ -345,9 +343,7 @@ void SetOutboundRTPStreamStatsFromMediaSenderInfo(
outbound_stats->retransmitted_packets_sent =
media_sender_info.retransmitted_packets_sent;
outbound_stats->bytes_sent =
static_cast<uint64_t>(media_sender_info.payload_bytes_sent);
outbound_stats->header_bytes_sent =
static_cast<uint64_t>(media_sender_info.header_and_padding_bytes_sent);
static_cast<uint64_t>(media_sender_info.bytes_sent);
outbound_stats->retransmitted_bytes_sent =
media_sender_info.retransmitted_bytes_sent;
}

20
pc/rtc_stats_collector_unittest.cc
View File

@ -1739,8 +1739,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Audio) {
voice_media_info.receivers[0].packets_rcvd = 2;
voice_media_info.receivers[0].fec_packets_discarded = 5566;
voice_media_info.receivers[0].fec_packets_received = 6677;
voice_media_info.receivers[0].payload_bytes_rcvd = 3;
voice_media_info.receivers[0].header_and_padding_bytes_rcvd = 4;
voice_media_info.receivers[0].bytes_rcvd = 3;
voice_media_info.receivers[0].codec_payload_type = 42;
voice_media_info.receivers[0].jitter_ms = 4500;
voice_media_info.receivers[0].last_packet_received_timestamp_ms =
@ -1777,7 +1776,6 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Audio) {
expected_audio.fec_packets_discarded = 5566;
expected_audio.fec_packets_received = 6677;
expected_audio.bytes_received = 3;
expected_audio.header_bytes_received = 4;
expected_audio.packets_lost = -1;
// |expected_audio.last_packet_received_timestamp| should be undefined.
expected_audio.jitter = 4.5;
@ -1811,8 +1809,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Video) {
video_media_info.receivers[0].local_stats[0].ssrc = 1;
video_media_info.receivers[0].packets_rcvd = 2;
video_media_info.receivers[0].packets_lost = 42;
video_media_info.receivers[0].payload_bytes_rcvd = 3;
video_media_info.receivers[0].header_and_padding_bytes_rcvd = 12;
video_media_info.receivers[0].bytes_rcvd = 3;
video_media_info.receivers[0].codec_payload_type = 42;
video_media_info.receivers[0].firs_sent = 5;
video_media_info.receivers[0].plis_sent = 6;
@ -1855,7 +1852,6 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Video) {
expected_video.nack_count = 7;
expected_video.packets_received = 2;
expected_video.bytes_received = 3;
expected_video.header_bytes_received = 12;
expected_video.packets_lost = 42;
expected_video.frames_decoded = 8;
expected_video.key_frames_decoded = 3;
@ -1900,8 +1896,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Audio) {
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].payload_bytes_sent = 3;
voice_media_info.senders[0].header_and_padding_bytes_sent = 12;
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;
@ -1934,7 +1929,6 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Audio) {
expected_audio.packets_sent = 2;
expected_audio.retransmitted_packets_sent = 20;
expected_audio.bytes_sent = 3;
expected_audio.header_bytes_sent = 12;
expected_audio.retransmitted_bytes_sent = 30;
ASSERT_TRUE(report->Get(expected_audio.id()));
@ -1962,8 +1956,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Video) {
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].payload_bytes_sent = 6;
video_media_info.senders[0].header_and_padding_bytes_sent = 12;
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;
@ -2015,7 +2008,6 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Video) {
expected_video.packets_sent = 5;
expected_video.retransmitted_packets_sent = 50;
expected_video.bytes_sent = 6;
expected_video.header_bytes_sent = 12;
expected_video.retransmitted_bytes_sent = 60;
expected_video.frames_encoded = 8;
expected_video.key_frames_encoded = 3;
@ -2204,8 +2196,7 @@ TEST_F(RTCStatsCollectorTest, CollectNoStreamRTCOutboundRTPStreamStats_Audio) {
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].payload_bytes_sent = 3;
voice_media_info.senders[0].header_and_padding_bytes_sent = 4;
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;
@ -2239,7 +2230,6 @@ TEST_F(RTCStatsCollectorTest, CollectNoStreamRTCOutboundRTPStreamStats_Audio) {
expected_audio.packets_sent = 2;
expected_audio.retransmitted_packets_sent = 20;
expected_audio.bytes_sent = 3;
expected_audio.header_bytes_sent = 4;
expected_audio.retransmitted_bytes_sent = 30;
ASSERT_TRUE(report->Get(expected_audio.id()));

4
pc/rtc_stats_integrationtest.cc
View File

@ -797,8 +797,6 @@ class RTCStatsReportVerifier {
inbound_stream.fec_packets_discarded);
}
verifier.TestMemberIsNonNegative<uint64_t>(inbound_stream.bytes_received);
verifier.TestMemberIsNonNegative<uint64_t>(
inbound_stream.header_bytes_received);
// packets_lost is defined as signed, but this should never happen in
// this test. See RFC 3550.
verifier.TestMemberIsNonNegative<int32_t>(inbound_stream.packets_lost);
@ -857,8 +855,6 @@ class RTCStatsReportVerifier {
verifier.TestMemberIsNonNegative<uint64_t>(
outbound_stream.retransmitted_packets_sent);
verifier.TestMemberIsNonNegative<uint64_t>(outbound_stream.bytes_sent);
verifier.TestMemberIsNonNegative<uint64_t>(
outbound_stream.header_bytes_sent);
verifier.TestMemberIsNonNegative<uint64_t>(
outbound_stream.retransmitted_bytes_sent);
verifier.TestMemberIsUndefined(outbound_stream.target_bitrate);

54
pc/stats_collector.cc
View File

@ -19,16 +19,10 @@
#include "pc/peer_connection.h"
#include "rtc_base/checks.h"
#include "rtc_base/third_party/base64/base64.h"
#include "system_wrappers/include/field_trial.h"
namespace webrtc {
namespace {
// Field trial which controls whether to report standard-compliant bytes
// sent/received per stream. If enabled, padding and headers are not included
// in bytes sent or received.
constexpr char kUseStandardBytesStats[] = "WebRTC-UseStandardBytesStats";
// The following is the enum RTCStatsIceCandidateType from
// http://w3c.github.io/webrtc-stats/#rtcstatsicecandidatetype-enum such that
// our stats report for ice candidate type could conform to that.
@ -88,14 +82,9 @@ void CreateTrackReports(const TrackVector& tracks,
}
void ExtractCommonSendProperties(const cricket::MediaSenderInfo& info,
StatsReport* report,
bool use_standard_bytes_stats) {
StatsReport* report) {
report->AddString(StatsReport::kStatsValueNameCodecName, info.codec_name);
int64_t bytes_sent = info.payload_bytes_sent;
if (!use_standard_bytes_stats) {
bytes_sent += info.header_and_padding_bytes_sent;
}
report->AddInt64(StatsReport::kStatsValueNameBytesSent, bytes_sent);
report->AddInt64(StatsReport::kStatsValueNameBytesSent, info.bytes_sent);
if (info.rtt_ms >= 0) {
report->AddInt64(StatsReport::kStatsValueNameRtt, info.rtt_ms);
}
@ -142,9 +131,7 @@ void SetAudioProcessingStats(StatsReport* report,
}
}
void ExtractStats(const cricket::VoiceReceiverInfo& info,
StatsReport* report,
bool use_standard_bytes_stats) {
void ExtractStats(const cricket::VoiceReceiverInfo& info, StatsReport* report) {
ExtractCommonReceiveProperties(info, report);
const FloatForAdd floats[] = {
{StatsReport::kStatsValueNameExpandRate, info.expand_rate},
@ -192,11 +179,7 @@ void ExtractStats(const cricket::VoiceReceiverInfo& info,
report->AddInt(StatsReport::kStatsValueNameDecodingCodecPLC,
info.decoding_codec_plc);
int64_t bytes_rcvd = info.payload_bytes_rcvd;
if (!use_standard_bytes_stats) {
bytes_rcvd += info.header_and_padding_bytes_rcvd;
}
report->AddInt64(StatsReport::kStatsValueNameBytesReceived, bytes_rcvd);
report->AddInt64(StatsReport::kStatsValueNameBytesReceived, info.bytes_rcvd);
if (info.capture_start_ntp_time_ms >= 0) {
report->AddInt64(StatsReport::kStatsValueNameCaptureStartNtpTimeMs,
info.capture_start_ntp_time_ms);
@ -204,10 +187,8 @@ void ExtractStats(const cricket::VoiceReceiverInfo& info,
report->AddString(StatsReport::kStatsValueNameMediaType, "audio");
}
void ExtractStats(const cricket::VoiceSenderInfo& info,
StatsReport* report,
bool use_standard_bytes_stats) {
ExtractCommonSendProperties(info, report, use_standard_bytes_stats);
void ExtractStats(const cricket::VoiceSenderInfo& info, StatsReport* report) {
ExtractCommonSendProperties(info, report);
SetAudioProcessingStats(report, info.typing_noise_detected,
info.apm_statistics);
@ -265,17 +246,11 @@ void ExtractStats(const cricket::VoiceSenderInfo& info,
}
}
void ExtractStats(const cricket::VideoReceiverInfo& info,
StatsReport* report,
bool use_standard_bytes_stats) {
void ExtractStats(const cricket::VideoReceiverInfo& info, StatsReport* report) {
ExtractCommonReceiveProperties(info, report);
report->AddString(StatsReport::kStatsValueNameCodecImplementationName,
info.decoder_implementation_name);
int64_t bytes_rcvd = info.payload_bytes_rcvd;
if (!use_standard_bytes_stats) {
bytes_rcvd += info.header_and_padding_bytes_rcvd;
}
report->AddInt64(StatsReport::kStatsValueNameBytesReceived, bytes_rcvd);
report->AddInt64(StatsReport::kStatsValueNameBytesReceived, info.bytes_rcvd);
if (info.capture_start_ntp_time_ms >= 0) {
report->AddInt64(StatsReport::kStatsValueNameCaptureStartNtpTimeMs,
info.capture_start_ntp_time_ms);
@ -326,10 +301,8 @@ void ExtractStats(const cricket::VideoReceiverInfo& info,
webrtc::videocontenttypehelpers::ToString(info.content_type));
}
void ExtractStats(const cricket::VideoSenderInfo& info,
StatsReport* report,
bool use_standard_bytes_stats) {
ExtractCommonSendProperties(info, report, use_standard_bytes_stats);
void ExtractStats(const cricket::VideoSenderInfo& info, StatsReport* report) {
ExtractCommonSendProperties(info, report);
report->AddString(StatsReport::kStatsValueNameCodecImplementationName,
info.encoder_implementation_name);
@ -444,7 +417,7 @@ void ExtractStatsFromList(
StatsReport* report =
collector->PrepareReport(true, ssrc, track_id, transport_id, direction);
if (report)
ExtractStats(d, report, collector->UseStandardBytesStats());
ExtractStats(d, report);
if (!d.remote_stats.empty()) {
report = collector->PrepareReport(false, ssrc, track_id, transport_id,
@ -497,10 +470,7 @@ const char* AdapterTypeToStatsType(rtc::AdapterType type) {
}
StatsCollector::StatsCollector(PeerConnectionInternal* pc)
: pc_(pc),
stats_gathering_started_(0),
use_standard_bytes_stats_(
webrtc::field_trial::IsEnabled(kUseStandardBytesStats)) {
: pc_(pc), stats_gathering_started_(0) {
RTC_DCHECK(pc_);
}

3
pc/stats_collector.h
View File

@ -94,8 +94,6 @@ class StatsCollector {
// ignored.
void ClearUpdateStatsCacheForTest();
bool UseStandardBytesStats() const { return use_standard_bytes_stats_; }
private:
friend class StatsCollectorTest;
@ -145,7 +143,6 @@ class StatsCollector {
// Raw pointer to the peer connection the statistics are gathered from.
PeerConnectionInternal* const pc_;
double stats_gathering_started_;
const bool use_standard_bytes_stats_;
// TODO(tommi): We appear to be holding on to raw pointers to reference
// counted objects? We should be using scoped_refptr here.

30
pc/stats_collector_unittest.cc
View File

@ -324,9 +324,7 @@ void VerifyVoiceReceiverInfoReport(const StatsReport* report,
EXPECT_EQ(rtc::ToString(info.audio_level), value_in_report);
EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNameBytesReceived,
&value_in_report));
EXPECT_EQ(rtc::ToString(info.payload_bytes_rcvd +
info.header_and_padding_bytes_rcvd),
value_in_report);
EXPECT_EQ(rtc::ToString(info.bytes_rcvd), value_in_report);
EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNameJitterReceived,
&value_in_report));
EXPECT_EQ(rtc::ToString(info.jitter_ms), value_in_report);
@ -399,9 +397,7 @@ void VerifyVoiceSenderInfoReport(const StatsReport* report,
EXPECT_EQ(sinfo.codec_name, value_in_report);
EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNameBytesSent,
&value_in_report));
EXPECT_EQ(rtc::ToString(sinfo.payload_bytes_sent +
sinfo.header_and_padding_bytes_sent),
value_in_report);
EXPECT_EQ(rtc::ToString(sinfo.bytes_sent), value_in_report);
EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNamePacketsSent,
&value_in_report));
EXPECT_EQ(rtc::ToString(sinfo.packets_sent), value_in_report);
@ -532,8 +528,7 @@ void InitVoiceSenderInfo(cricket::VoiceSenderInfo* voice_sender_info,
uint32_t ssrc = kSsrcOfTrack) {
voice_sender_info->add_ssrc(ssrc);
voice_sender_info->codec_name = "fake_codec";
voice_sender_info->payload_bytes_sent = 88;
voice_sender_info->header_and_padding_bytes_sent = 12;
voice_sender_info->bytes_sent = 100;
voice_sender_info->packets_sent = 101;
voice_sender_info->rtt_ms = 102;
voice_sender_info->fraction_lost = 103;
@ -568,8 +563,7 @@ void UpdateVoiceSenderInfoFromAudioTrack(
void InitVoiceReceiverInfo(cricket::VoiceReceiverInfo* voice_receiver_info) {
voice_receiver_info->add_ssrc(kSsrcOfTrack);
voice_receiver_info->payload_bytes_rcvd = 98;
voice_receiver_info->header_and_padding_bytes_rcvd = 12;
voice_receiver_info->bytes_rcvd = 110;
voice_receiver_info->packets_rcvd = 111;
voice_receiver_info->packets_lost = 114;
voice_receiver_info->jitter_ms = 116;
@ -910,8 +904,7 @@ TEST_P(StatsCollectorTrackTest, BytesCounterHandles64Bits) {
VideoSenderInfo video_sender_info;
video_sender_info.add_ssrc(1234);
video_sender_info.payload_bytes_sent = kBytesSent;
video_sender_info.header_and_padding_bytes_sent = 0;
video_sender_info.bytes_sent = kBytesSent;
VideoMediaInfo video_info;
video_info.senders.push_back(video_sender_info);
@ -943,8 +936,7 @@ TEST_P(StatsCollectorTrackTest, AudioBandwidthEstimationInfoIsReported) {
VoiceSenderInfo voice_sender_info;
voice_sender_info.add_ssrc(1234);
voice_sender_info.payload_bytes_sent = kBytesSent - 12;
voice_sender_info.header_and_padding_bytes_sent = 12;
voice_sender_info.bytes_sent = kBytesSent;
VoiceMediaInfo voice_info;
voice_info.senders.push_back(voice_sender_info);
@ -992,9 +984,7 @@ TEST_P(StatsCollectorTrackTest, VideoBandwidthEstimationInfoIsReported) {
VideoSenderInfo video_sender_info;
video_sender_info.add_ssrc(1234);
video_sender_info.payload_bytes_sent = kBytesSent - 12;
video_sender_info.header_and_padding_bytes_sent = 12;
video_sender_info.bytes_sent = kBytesSent;
VideoMediaInfo video_info;
video_info.senders.push_back(video_sender_info);
@ -1091,8 +1081,7 @@ TEST_P(StatsCollectorTrackTest, TrackAndSsrcObjectExistAfterUpdateSsrcStats) {
VideoSenderInfo video_sender_info;
video_sender_info.add_ssrc(1234);
video_sender_info.payload_bytes_sent = kBytesSent - 12;
video_sender_info.header_and_padding_bytes_sent = 12;
video_sender_info.bytes_sent = kBytesSent;
VideoMediaInfo video_info;
video_info.senders.push_back(video_sender_info);
@ -1146,8 +1135,7 @@ TEST_P(StatsCollectorTrackTest, TransportObjectLinkedFromSsrcObject) {
VideoSenderInfo video_sender_info;
video_sender_info.add_ssrc(1234);
video_sender_info.payload_bytes_sent = kBytesSent - 12;
video_sender_info.header_and_padding_bytes_sent = 12;
video_sender_info.bytes_sent = kBytesSent;
VideoMediaInfo video_info;
video_info.senders.push_back(video_sender_info);

6
stats/rtcstats_objects.cc
View File

@ -598,7 +598,6 @@ WEBRTC_RTCSTATS_IMPL(
RTCInboundRTPStreamStats, RTCRTPStreamStats, "inbound-rtp",
&packets_received,
&bytes_received,
&header_bytes_received,
&packets_lost,
&last_packet_received_timestamp,
&jitter,
@ -631,7 +630,6 @@ RTCInboundRTPStreamStats::RTCInboundRTPStreamStats(std::string&& id,
fec_packets_received("fecPacketsReceived"),
fec_packets_discarded("fecPacketsDiscarded"),
bytes_received("bytesReceived"),
header_bytes_received("headerBytesReceived"),
packets_lost("packetsLost"),
last_packet_received_timestamp("lastPacketReceivedTimestamp"),
jitter("jitter"),
@ -659,7 +657,6 @@ RTCInboundRTPStreamStats::RTCInboundRTPStreamStats(
fec_packets_received(other.fec_packets_received),
fec_packets_discarded(other.fec_packets_discarded),
bytes_received(other.bytes_received),
header_bytes_received(other.header_bytes_received),
packets_lost(other.packets_lost),
last_packet_received_timestamp(other.last_packet_received_timestamp),
jitter(other.jitter),
@ -689,7 +686,6 @@ WEBRTC_RTCSTATS_IMPL(
&packets_sent,
&retransmitted_packets_sent,
&bytes_sent,
&header_bytes_sent,
&retransmitted_bytes_sent,
&target_bitrate,
&frames_encoded,
@ -714,7 +710,6 @@ RTCOutboundRTPStreamStats::RTCOutboundRTPStreamStats(std::string&& id,
packets_sent("packetsSent"),
retransmitted_packets_sent("retransmittedPacketsSent"),
bytes_sent("bytesSent"),
header_bytes_sent("headerBytesSent"),
retransmitted_bytes_sent("retransmittedBytesSent"),
target_bitrate("targetBitrate"),
frames_encoded("framesEncoded"),
@ -735,7 +730,6 @@ RTCOutboundRTPStreamStats::RTCOutboundRTPStreamStats(
packets_sent(other.packets_sent),
retransmitted_packets_sent(other.retransmitted_packets_sent),
bytes_sent(other.bytes_sent),
header_bytes_sent(other.header_bytes_sent),
retransmitted_bytes_sent(other.retransmitted_bytes_sent),
target_bitrate(other.target_bitrate),
frames_encoded(other.frames_encoded),