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Reland "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received"
This is a reland of fbde32e596
The chromium problem should be fixed with
https://chromium-review.googlesource.com/c/chromium/src/+/1862437
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
Bug: webrtc:10525
Change-Id: I3b61f9535aa3f1fca2ed84f068233803d4ec9fe2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157045
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29485}
This commit is contained in:
Niels Möller
Commit Bot
parent
e5d0fe0dff
commit
ac0a4cbbd8
25 changed files with 198 additions and 123 deletions
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@ -413,6 +413,7 @@ class RTC_EXPORT RTCInboundRTPStreamStats final : public RTCRTPStreamStats {
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RTCStatsMember<uint64_t> fec_packets_received;
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RTCStatsMember<uint64_t> fec_packets_discarded;
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RTCStatsMember<uint64_t> bytes_received;
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RTCStatsMember<uint64_t> header_bytes_received;
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RTCStatsMember<int32_t> packets_lost; // Signed per RFC 3550
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RTCStatsMember<double> last_packet_received_timestamp;
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// TODO(hbos): Collect and populate this value for both "audio" and "video",
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@ -466,6 +467,7 @@ class RTC_EXPORT RTCOutboundRTPStreamStats final : public RTCRTPStreamStats {
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RTCStatsMember<uint32_t> packets_sent;
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RTCStatsMember<uint64_t> retransmitted_packets_sent;
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RTCStatsMember<uint64_t> bytes_sent;
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RTCStatsMember<uint64_t> header_bytes_sent;
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RTCStatsMember<uint64_t> retransmitted_bytes_sent;
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// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7066
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RTCStatsMember<double> target_bitrate;
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@ -188,7 +188,11 @@ webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
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return stats;
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}
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stats.bytes_rcvd = call_stats.bytesReceived;
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stats.payload_bytes_rcvd = call_stats.payload_bytes_rcvd;
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stats.header_and_padding_bytes_rcvd =
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call_stats.header_and_padding_bytes_rcvd;
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stats.bytes_rcvd =
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stats.payload_bytes_rcvd + stats.header_and_padding_bytes_rcvd;
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stats.packets_rcvd = call_stats.packetsReceived;
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stats.packets_lost = call_stats.cumulativeLost;
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stats.capture_start_ntp_time_ms = call_stats.capture_start_ntp_time_ms_;
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@ -63,7 +63,7 @@ const unsigned int kSpeechOutputLevel = 99;
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const double kTotalOutputEnergy = 0.25;
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const double kTotalOutputDuration = 0.5;
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const CallReceiveStatistics kCallStats = {678, 234, -12, 567, 890, 123};
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const CallReceiveStatistics kCallStats = {678, 234, -12, 567, 78, 890, 123};
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const std::pair<int, SdpAudioFormat> kReceiveCodec = {
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123,
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{"codec_name_recv", 96000, 0}};
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@ -266,7 +266,9 @@ TEST(AudioReceiveStreamTest, GetStats) {
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helper.SetupMockForGetStats();
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AudioReceiveStream::Stats stats = recv_stream->GetStats();
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EXPECT_EQ(kRemoteSsrc, stats.remote_ssrc);
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EXPECT_EQ(static_cast<int64_t>(kCallStats.bytesReceived), stats.bytes_rcvd);
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EXPECT_EQ(kCallStats.payload_bytes_rcvd, stats.payload_bytes_rcvd);
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EXPECT_EQ(kCallStats.header_and_padding_bytes_rcvd,
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stats.header_and_padding_bytes_rcvd);
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EXPECT_EQ(static_cast<uint32_t>(kCallStats.packetsReceived),
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stats.packets_rcvd);
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EXPECT_EQ(kCallStats.cumulativeLost, stats.packets_lost);
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@ -440,7 +440,11 @@ webrtc::AudioSendStream::Stats AudioSendStream::GetStats(
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stats.target_bitrate_bps = channel_send_->GetBitrate();
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webrtc::CallSendStatistics call_stats = channel_send_->GetRTCPStatistics();
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stats.bytes_sent = call_stats.bytesSent;
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stats.payload_bytes_sent = call_stats.payload_bytes_sent;
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stats.header_and_padding_bytes_sent =
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call_stats.header_and_padding_bytes_sent;
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stats.bytes_sent =
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stats.payload_bytes_sent + stats.header_and_padding_bytes_sent;
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stats.retransmitted_bytes_sent = call_stats.retransmitted_bytes_sent;
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stats.packets_sent = call_stats.packetsSent;
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stats.retransmitted_packets_sent = call_stats.retransmitted_packets_sent;
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@ -64,7 +64,7 @@ const double kEchoReturnLoss = -65;
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const double kEchoReturnLossEnhancement = 101;
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const double kResidualEchoLikelihood = -1.0f;
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const double kResidualEchoLikelihoodMax = 23.0f;
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const CallSendStatistics kCallStats = {112, 13456, 17890};
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const CallSendStatistics kCallStats = {112, 12, 13456, 17890};
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const ReportBlock kReportBlock = {456, 780, 123, 567, 890, 132, 143, 13354};
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const int kTelephoneEventPayloadType = 123;
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const int kTelephoneEventPayloadFrequency = 65432;
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@ -414,7 +414,9 @@ TEST(AudioSendStreamTest, GetStats) {
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helper.SetupMockForGetStats();
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AudioSendStream::Stats stats = send_stream->GetStats(true);
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EXPECT_EQ(kSsrc, stats.local_ssrc);
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EXPECT_EQ(static_cast<int64_t>(kCallStats.bytesSent), stats.bytes_sent);
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EXPECT_EQ(kCallStats.payload_bytes_sent, stats.payload_bytes_sent);
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EXPECT_EQ(kCallStats.header_and_padding_bytes_sent,
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stats.header_and_padding_bytes_sent);
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EXPECT_EQ(kCallStats.packetsSent, stats.packets_sent);
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EXPECT_EQ(kReportBlock.cumulative_num_packets_lost, stats.packets_lost);
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EXPECT_EQ(Q8ToFloat(kReportBlock.fraction_lost), stats.fraction_lost);
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@ -43,7 +43,6 @@
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#include "rtc_base/race_checker.h"
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#include "rtc_base/thread_checker.h"
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#include "rtc_base/time_utils.h"
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#include "system_wrappers/include/field_trial.h"
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#include "system_wrappers/include/metrics.h"
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namespace webrtc {
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@ -57,11 +56,6 @@ constexpr double kAudioSampleDurationSeconds = 0.01;
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constexpr int kVoiceEngineMinMinPlayoutDelayMs = 0;
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constexpr int kVoiceEngineMaxMinPlayoutDelayMs = 10000;
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// Field trial which controls whether to report standard-compliant bytes
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// sent/received per stream. If enabled, padding and headers are not included
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// in bytes sent or received.
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constexpr char kUseStandardBytesStats[] = "WebRTC-UseStandardBytesStats";
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RTPHeader CreateRTPHeaderForMediaTransportFrame(
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const MediaTransportEncodedAudioFrame& frame,
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uint64_t channel_id) {
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@ -278,8 +272,6 @@ class ChannelReceive : public ChannelReceiveInterface,
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// E2EE Audio Frame Decryption
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rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor_;
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webrtc::CryptoOptions crypto_options_;
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const bool use_standard_bytes_stats_;
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};
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void ChannelReceive::OnReceivedPayloadData(
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@ -484,9 +476,7 @@ ChannelReceive::ChannelReceive(
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associated_send_channel_(nullptr),
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media_transport_config_(media_transport_config),
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frame_decryptor_(frame_decryptor),
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crypto_options_(crypto_options),
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use_standard_bytes_stats_(
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webrtc::field_trial::IsEnabled(kUseStandardBytesStats)) {
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crypto_options_(crypto_options) {
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// TODO(nisse): Use _moduleProcessThreadPtr instead?
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module_process_thread_checker_.Detach();
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@ -734,16 +724,17 @@ CallReceiveStatistics ChannelReceive::GetRTCPStatistics() const {
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// --- Data counters
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if (statistician) {
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if (use_standard_bytes_stats_) {
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stats.bytesReceived = rtp_stats.packet_counter.payload_bytes;
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} else {
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stats.bytesReceived = rtp_stats.packet_counter.TotalBytes();
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}
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stats.payload_bytes_rcvd = rtp_stats.packet_counter.payload_bytes;
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stats.header_and_padding_bytes_rcvd =
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rtp_stats.packet_counter.header_bytes +
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rtp_stats.packet_counter.padding_bytes;
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stats.packetsReceived = rtp_stats.packet_counter.packets;
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stats.last_packet_received_timestamp_ms =
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rtp_stats.last_packet_received_timestamp_ms;
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} else {
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stats.bytesReceived = 0;
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stats.payload_bytes_rcvd = 0;
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stats.header_and_padding_bytes_rcvd = 0;
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stats.packetsReceived = 0;
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stats.last_packet_received_timestamp_ms = absl::nullopt;
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}
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@ -54,7 +54,8 @@ struct CallReceiveStatistics {
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unsigned int cumulativeLost;
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unsigned int jitterSamples;
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int64_t rttMs;
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size_t bytesReceived;
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int64_t payload_bytes_rcvd = 0;
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int64_t header_and_padding_bytes_rcvd = 0;
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int packetsReceived;
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// The capture ntp time (in local timebase) of the first played out audio
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// frame.
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@ -52,11 +52,6 @@ namespace {
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constexpr int64_t kMaxRetransmissionWindowMs = 1000;
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constexpr int64_t kMinRetransmissionWindowMs = 30;
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// Field trial which controls whether to report standard-compliant bytes
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// sent/received per stream. If enabled, padding and headers are not included
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// in bytes sent or received.
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constexpr char kUseStandardBytesStats[] = "WebRTC-UseStandardBytesStats";
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MediaTransportEncodedAudioFrame::FrameType
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MediaTransportFrameTypeForWebrtcFrameType(webrtc::AudioFrameType frame_type) {
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switch (frame_type) {
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@ -263,7 +258,6 @@ class ChannelSend : public ChannelSendInterface,
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rtc::ThreadChecker construction_thread_;
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const bool use_twcc_plr_for_ana_;
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const bool use_standard_bytes_stats_;
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bool encoder_queue_is_active_ RTC_GUARDED_BY(encoder_queue_) = false;
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@ -609,8 +603,6 @@ ChannelSend::ChannelSend(Clock* clock,
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new RateLimiter(clock, kMaxRetransmissionWindowMs)),
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use_twcc_plr_for_ana_(
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webrtc::field_trial::FindFullName("UseTwccPlrForAna") == "Enabled"),
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use_standard_bytes_stats_(
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webrtc::field_trial::IsEnabled(kUseStandardBytesStats)),
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media_transport_config_(media_transport_config),
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frame_encryptor_(frame_encryptor),
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crypto_options_(crypto_options),
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@ -1019,17 +1011,12 @@ CallSendStatistics ChannelSend::GetRTCPStatistics() const {
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StreamDataCounters rtp_stats;
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StreamDataCounters rtx_stats;
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_rtpRtcpModule->GetSendStreamDataCounters(&rtp_stats, &rtx_stats);
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if (use_standard_bytes_stats_) {
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stats.bytesSent = rtp_stats.transmitted.payload_bytes +
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rtx_stats.transmitted.payload_bytes;
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} else {
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stats.bytesSent = rtp_stats.transmitted.payload_bytes +
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rtp_stats.transmitted.padding_bytes +
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rtp_stats.transmitted.header_bytes +
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rtx_stats.transmitted.payload_bytes +
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rtx_stats.transmitted.padding_bytes +
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rtx_stats.transmitted.header_bytes;
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}
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stats.payload_bytes_sent =
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rtp_stats.transmitted.payload_bytes + rtx_stats.transmitted.payload_bytes;
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stats.header_and_padding_bytes_sent =
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rtp_stats.transmitted.padding_bytes + rtp_stats.transmitted.header_bytes +
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rtx_stats.transmitted.padding_bytes + rtx_stats.transmitted.header_bytes;
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// TODO(https://crbug.com/webrtc/10555): RTX retransmissions should show up in
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// separate outbound-rtp stream objects.
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stats.retransmitted_bytes_sent = rtp_stats.retransmitted.payload_bytes;
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@ -36,7 +36,8 @@ class RtpTransportControllerSendInterface;
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struct CallSendStatistics {
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int64_t rttMs;
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size_t bytesSent;
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int64_t payload_bytes_sent;
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int64_t header_and_padding_bytes_sent;
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// https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-retransmittedbytessent
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uint64_t retransmitted_bytes_sent;
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int packetsSent;
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@ -46,7 +46,7 @@ class NoLossTest : public AudioEndToEndTest {
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void OnStreamsStopped() override {
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AudioSendStream::Stats send_stats = send_stream()->GetStats();
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EXPECT_PRED2(IsNear, kBytesSent, send_stats.bytes_sent);
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EXPECT_PRED2(IsNear, kBytesSent, send_stats.payload_bytes_sent);
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EXPECT_PRED2(IsNear, kPacketsSent, send_stats.packets_sent);
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EXPECT_EQ(0, send_stats.packets_lost);
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EXPECT_EQ(0.0f, send_stats.fraction_lost);
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@ -66,7 +66,7 @@ class NoLossTest : public AudioEndToEndTest {
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EXPECT_EQ(false, send_stats.typing_noise_detected);
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AudioReceiveStream::Stats recv_stats = receive_stream()->GetStats();
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EXPECT_PRED2(IsNear, kBytesSent, recv_stats.bytes_rcvd);
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EXPECT_PRED2(IsNear, kBytesSent, recv_stats.payload_bytes_rcvd);
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EXPECT_PRED2(IsNear, kPacketsSent, recv_stats.packets_rcvd);
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EXPECT_EQ(0u, recv_stats.packets_lost);
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EXPECT_EQ("opus", send_stats.codec_name);
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@ -36,7 +36,11 @@ class AudioReceiveStream {
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Stats();
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~Stats();
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uint32_t remote_ssrc = 0;
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int64_t bytes_rcvd = 0;
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// TODO(nisse): Sum of below two values. Deprecated, delete as soon as
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// downstream applications are updated.
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int64_t bytes_rcvd;
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int64_t payload_bytes_rcvd = 0;
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int64_t header_and_padding_bytes_rcvd = 0;
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uint32_t packets_rcvd = 0;
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uint64_t fec_packets_received = 0;
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uint64_t fec_packets_discarded = 0;
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@ -43,7 +43,11 @@ class AudioSendStream {
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// TODO(solenberg): Harmonize naming and defaults with receive stream stats.
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uint32_t local_ssrc = 0;
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int64_t bytes_sent = 0;
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// TODO(nisse): Sum of below two values. Deprecated, delete as soon as
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// downstream applications are updated.
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int64_t bytes_sent;
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int64_t payload_bytes_sent = 0;
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int64_t header_and_padding_bytes_sent = 0;
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// https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-retransmittedbytessent
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uint64_t retransmitted_bytes_sent = 0;
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int32_t packets_sent = 0;
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@ -393,7 +393,13 @@ struct MediaSenderInfo {
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return 0;
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}
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}
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int64_t bytes_sent = 0;
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// TODO(nisse): Sum of below two values. Deprecated, delete as soon as
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// downstream applications are updated.
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int64_t bytes_sent;
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// https://w3c.github.io/webrtc-stats/#dom-rtcsentrtpstreamstats-bytessent
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int64_t payload_bytes_sent = 0;
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// https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-headerbytessent
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int64_t header_and_padding_bytes_sent = 0;
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// https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-retransmittedbytessent
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uint64_t retransmitted_bytes_sent = 0;
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int packets_sent = 0;
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@ -447,7 +453,13 @@ struct MediaReceiverInfo {
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}
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}
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int64_t bytes_rcvd = 0;
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// TODO(nisse): Sum of below two values. Deprecated, delete as soon as
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// downstream applications are updated.
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int64_t bytes_rcvd;
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// https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-bytesreceived
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int64_t payload_bytes_rcvd = 0;
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// https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-headerbytesreceived
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int64_t header_and_padding_bytes_rcvd = 0;
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int packets_rcvd = 0;
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int packets_lost = 0;
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// TODO(bugs.webrtc.org/10679): Unused, delete as soon as downstream code is
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@ -48,11 +48,6 @@ namespace {
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const int kMinLayerSize = 16;
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// Field trial which controls whether to report standard-compliant bytes
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// sent/received per stream. If enabled, padding and headers are not included
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// in bytes sent or received.
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constexpr char kUseStandardBytesStats[] = "WebRTC-UseStandardBytesStats";
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// If this field trial is enabled, we will enable sending FlexFEC and disable
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// sending ULPFEC whenever the former has been negotiated in the SDPs.
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bool IsFlexfecFieldTrialEnabled() {
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@ -1808,9 +1803,7 @@ WebRtcVideoChannel::WebRtcVideoSendStream::WebRtcVideoSendStream(
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encoder_sink_(nullptr),
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parameters_(std::move(config), options, max_bitrate_bps, codec_settings),
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rtp_parameters_(CreateRtpParametersWithEncodings(sp)),
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sending_(false),
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use_standard_bytes_stats_(
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webrtc::field_trial::IsEnabled(kUseStandardBytesStats)) {
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sending_(false) {
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// Maximum packet size may come in RtpConfig from external transport, for
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// example from QuicTransportInterface implementation, so do not exceed
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// given max_packet_size.
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@ -2379,13 +2372,10 @@ VideoSenderInfo WebRtcVideoChannel::WebRtcVideoSendStream::GetVideoSenderInfo(
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it != stats.substreams.end(); ++it) {
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// TODO(pbos): Wire up additional stats, such as padding bytes.
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webrtc::VideoSendStream::StreamStats stream_stats = it->second;
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if (use_standard_bytes_stats_) {
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info.bytes_sent += stream_stats.rtp_stats.transmitted.payload_bytes;
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} else {
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info.bytes_sent += stream_stats.rtp_stats.transmitted.payload_bytes +
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stream_stats.rtp_stats.transmitted.header_bytes +
|
||||
stream_stats.rtp_stats.transmitted.padding_bytes;
|
||||
}
|
||||
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;
|
||||
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
|
||||
|
|
@ -2409,6 +2399,8 @@ 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.
|
||||
|
|
@ -2501,9 +2493,7 @@ WebRtcVideoChannel::WebRtcVideoReceiveStream::WebRtcVideoReceiveStream(
|
|||
decoder_factory_(decoder_factory),
|
||||
sink_(NULL),
|
||||
first_frame_timestamp_(-1),
|
||||
estimated_remote_start_ntp_time_ms_(0),
|
||||
use_standard_bytes_stats_(
|
||||
webrtc::field_trial::IsEnabled(kUseStandardBytesStats)) {
|
||||
estimated_remote_start_ntp_time_ms_(0) {
|
||||
config_.renderer = this;
|
||||
ConfigureCodecs(recv_codecs);
|
||||
ConfigureFlexfecCodec(flexfec_config.payload_type);
|
||||
|
|
@ -2799,11 +2789,12 @@ WebRtcVideoChannel::WebRtcVideoReceiveStream::GetVideoReceiverInfo(
|
|||
if (stats.current_payload_type != -1) {
|
||||
info.codec_payload_type = stats.current_payload_type;
|
||||
}
|
||||
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.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;
|
||||
info.packets_rcvd = stats.rtp_stats.packet_counter.packets;
|
||||
info.packets_lost = stats.rtp_stats.packets_lost;
|
||||
|
||||
|
|
|
|||
|
|
@ -380,8 +380,6 @@ 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.
|
||||
|
|
@ -471,8 +469,6 @@ 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);
|
||||
|
|
|
|||
|
|
@ -1599,8 +1599,6 @@ 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;
|
||||
|
|
@ -1613,7 +1611,7 @@ 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].bytes_sent,
|
||||
EXPECT_EQ(info.senders[0].payload_bytes_sent,
|
||||
NumRtpBytes() - kRtpHeaderSize * NumRtpPackets());
|
||||
EXPECT_EQ(NumRtpPackets(), info.senders[0].packets_sent);
|
||||
EXPECT_EQ(0.0, info.senders[0].fraction_lost);
|
||||
|
|
@ -1638,7 +1636,7 @@ 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].bytes_rcvd);
|
||||
info.receivers[0].payload_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.
|
||||
|
|
@ -1659,8 +1657,6 @@ 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;
|
||||
|
|
@ -1694,7 +1690,7 @@ 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).bytes_sent, kTimeout);
|
||||
GetSenderStats(0).payload_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);
|
||||
|
|
@ -1704,7 +1700,7 @@ 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).bytes_rcvd, kTimeout);
|
||||
GetReceiverStats(i).payload_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);
|
||||
|
|
@ -5282,9 +5278,6 @@ 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;
|
||||
|
|
@ -5297,7 +5290,7 @@ 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].bytes_rcvd));
|
||||
rtc::checked_cast<size_t>(info.receivers[0].payload_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);
|
||||
|
|
|
|||
|
|
@ -2158,7 +2158,10 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
|
|||
stream.second->GetStats(recv_streams_.size() > 0);
|
||||
VoiceSenderInfo sinfo;
|
||||
sinfo.add_ssrc(stats.local_ssrc);
|
||||
sinfo.bytes_sent = stats.bytes_sent;
|
||||
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.retransmitted_bytes_sent = stats.retransmitted_bytes_sent;
|
||||
sinfo.packets_sent = stats.packets_sent;
|
||||
sinfo.retransmitted_packets_sent = stats.retransmitted_packets_sent;
|
||||
|
|
@ -2201,7 +2204,10 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
|
|||
webrtc::AudioReceiveStream::Stats stats = stream.second->GetStats();
|
||||
VoiceReceiverInfo rinfo;
|
||||
rinfo.add_ssrc(stats.remote_ssrc);
|
||||
rinfo.bytes_rcvd = stats.bytes_rcvd;
|
||||
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.packets_rcvd = stats.packets_rcvd;
|
||||
rinfo.fec_packets_received = stats.fec_packets_received;
|
||||
rinfo.fec_packets_discarded = stats.fec_packets_discarded;
|
||||
|
|
|
|||
|
|
@ -566,7 +566,8 @@ class WebRtcVoiceEngineTestFake : public ::testing::Test {
|
|||
webrtc::AudioSendStream::Stats GetAudioSendStreamStats() const {
|
||||
webrtc::AudioSendStream::Stats stats;
|
||||
stats.local_ssrc = 12;
|
||||
stats.bytes_sent = 345;
|
||||
stats.payload_bytes_sent = 345;
|
||||
stats.header_and_padding_bytes_sent = 56;
|
||||
stats.packets_sent = 678;
|
||||
stats.packets_lost = 9012;
|
||||
stats.fraction_lost = 34.56f;
|
||||
|
|
@ -600,7 +601,9 @@ class WebRtcVoiceEngineTestFake : public ::testing::Test {
|
|||
bool is_sending) {
|
||||
const auto stats = GetAudioSendStreamStats();
|
||||
EXPECT_EQ(info.ssrc(), stats.local_ssrc);
|
||||
EXPECT_EQ(info.bytes_sent, stats.bytes_sent);
|
||||
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.packets_sent, stats.packets_sent);
|
||||
EXPECT_EQ(info.packets_lost, stats.packets_lost);
|
||||
EXPECT_EQ(info.fraction_lost, stats.fraction_lost);
|
||||
|
|
@ -642,7 +645,8 @@ class WebRtcVoiceEngineTestFake : public ::testing::Test {
|
|||
webrtc::AudioReceiveStream::Stats GetAudioReceiveStreamStats() const {
|
||||
webrtc::AudioReceiveStream::Stats stats;
|
||||
stats.remote_ssrc = 123;
|
||||
stats.bytes_rcvd = 456;
|
||||
stats.payload_bytes_rcvd = 456;
|
||||
stats.header_and_padding_bytes_rcvd = 67;
|
||||
stats.packets_rcvd = 768;
|
||||
stats.packets_lost = 101;
|
||||
stats.codec_name = "codec_name_recv";
|
||||
|
|
@ -682,7 +686,9 @@ 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.bytes_rcvd, stats.bytes_rcvd);
|
||||
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(rtc::checked_cast<unsigned int>(info.packets_rcvd),
|
||||
stats.packets_rcvd);
|
||||
EXPECT_EQ(rtc::checked_cast<unsigned int>(info.packets_lost),
|
||||
|
|
|
|||
|
|
@ -256,7 +256,9 @@ 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.bytes_rcvd);
|
||||
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);
|
||||
inbound_stats->packets_lost =
|
||||
static_cast<int32_t>(media_receiver_info.packets_lost);
|
||||
}
|
||||
|
|
@ -343,7 +345,9 @@ void SetOutboundRTPStreamStatsFromMediaSenderInfo(
|
|||
outbound_stats->retransmitted_packets_sent =
|
||||
media_sender_info.retransmitted_packets_sent;
|
||||
outbound_stats->bytes_sent =
|
||||
static_cast<uint64_t>(media_sender_info.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);
|
||||
outbound_stats->retransmitted_bytes_sent =
|
||||
media_sender_info.retransmitted_bytes_sent;
|
||||
}
|
||||
|
|
|
|||
|
|
@ -1739,7 +1739,8 @@ 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].bytes_rcvd = 3;
|
||||
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].codec_payload_type = 42;
|
||||
voice_media_info.receivers[0].jitter_ms = 4500;
|
||||
voice_media_info.receivers[0].last_packet_received_timestamp_ms =
|
||||
|
|
@ -1776,6 +1777,7 @@ 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;
|
||||
|
|
@ -1809,7 +1811,8 @@ 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].bytes_rcvd = 3;
|
||||
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].codec_payload_type = 42;
|
||||
video_media_info.receivers[0].firs_sent = 5;
|
||||
video_media_info.receivers[0].plis_sent = 6;
|
||||
|
|
@ -1852,6 +1855,7 @@ 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;
|
||||
|
|
@ -1896,7 +1900,8 @@ 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].bytes_sent = 3;
|
||||
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].retransmitted_bytes_sent = 30;
|
||||
voice_media_info.senders[0].codec_payload_type = 42;
|
||||
|
||||
|
|
@ -1929,6 +1934,7 @@ 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()));
|
||||
|
|
@ -1956,7 +1962,8 @@ 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].bytes_sent = 6;
|
||||
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].retransmitted_bytes_sent = 60;
|
||||
video_media_info.senders[0].codec_payload_type = 42;
|
||||
video_media_info.senders[0].frames_encoded = 8;
|
||||
|
|
@ -2008,6 +2015,7 @@ 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;
|
||||
|
|
@ -2196,7 +2204,8 @@ 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].bytes_sent = 3;
|
||||
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].retransmitted_bytes_sent = 30;
|
||||
voice_media_info.senders[0].codec_payload_type = 42;
|
||||
|
||||
|
|
@ -2230,6 +2239,7 @@ 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()));
|
||||
|
|
|
|||
|
|
@ -797,6 +797,8 @@ 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);
|
||||
|
|
@ -855,6 +857,8 @@ 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);
|
||||
|
|
|
|||
|
|
@ -19,10 +19,16 @@
|
|||
#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.
|
||||
|
|
@ -82,9 +88,14 @@ void CreateTrackReports(const TrackVector& tracks,
|
|||
}
|
||||
|
||||
void ExtractCommonSendProperties(const cricket::MediaSenderInfo& info,
|
||||
StatsReport* report) {
|
||||
StatsReport* report,
|
||||
bool use_standard_bytes_stats) {
|
||||
report->AddString(StatsReport::kStatsValueNameCodecName, info.codec_name);
|
||||
report->AddInt64(StatsReport::kStatsValueNameBytesSent, info.bytes_sent);
|
||||
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);
|
||||
if (info.rtt_ms >= 0) {
|
||||
report->AddInt64(StatsReport::kStatsValueNameRtt, info.rtt_ms);
|
||||
}
|
||||
|
|
@ -131,7 +142,9 @@ void SetAudioProcessingStats(StatsReport* report,
|
|||
}
|
||||
}
|
||||
|
||||
void ExtractStats(const cricket::VoiceReceiverInfo& info, StatsReport* report) {
|
||||
void ExtractStats(const cricket::VoiceReceiverInfo& info,
|
||||
StatsReport* report,
|
||||
bool use_standard_bytes_stats) {
|
||||
ExtractCommonReceiveProperties(info, report);
|
||||
const FloatForAdd floats[] = {
|
||||
{StatsReport::kStatsValueNameExpandRate, info.expand_rate},
|
||||
|
|
@ -179,7 +192,11 @@ void ExtractStats(const cricket::VoiceReceiverInfo& info, StatsReport* report) {
|
|||
report->AddInt(StatsReport::kStatsValueNameDecodingCodecPLC,
|
||||
info.decoding_codec_plc);
|
||||
|
||||
report->AddInt64(StatsReport::kStatsValueNameBytesReceived, info.bytes_rcvd);
|
||||
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);
|
||||
if (info.capture_start_ntp_time_ms >= 0) {
|
||||
report->AddInt64(StatsReport::kStatsValueNameCaptureStartNtpTimeMs,
|
||||
info.capture_start_ntp_time_ms);
|
||||
|
|
@ -187,8 +204,10 @@ void ExtractStats(const cricket::VoiceReceiverInfo& info, StatsReport* report) {
|
|||
report->AddString(StatsReport::kStatsValueNameMediaType, "audio");
|
||||
}
|
||||
|
||||
void ExtractStats(const cricket::VoiceSenderInfo& info, StatsReport* report) {
|
||||
ExtractCommonSendProperties(info, report);
|
||||
void ExtractStats(const cricket::VoiceSenderInfo& info,
|
||||
StatsReport* report,
|
||||
bool use_standard_bytes_stats) {
|
||||
ExtractCommonSendProperties(info, report, use_standard_bytes_stats);
|
||||
|
||||
SetAudioProcessingStats(report, info.typing_noise_detected,
|
||||
info.apm_statistics);
|
||||
|
|
@ -246,11 +265,17 @@ void ExtractStats(const cricket::VoiceSenderInfo& info, StatsReport* report) {
|
|||
}
|
||||
}
|
||||
|
||||
void ExtractStats(const cricket::VideoReceiverInfo& info, StatsReport* report) {
|
||||
void ExtractStats(const cricket::VideoReceiverInfo& info,
|
||||
StatsReport* report,
|
||||
bool use_standard_bytes_stats) {
|
||||
ExtractCommonReceiveProperties(info, report);
|
||||
report->AddString(StatsReport::kStatsValueNameCodecImplementationName,
|
||||
info.decoder_implementation_name);
|
||||
report->AddInt64(StatsReport::kStatsValueNameBytesReceived, info.bytes_rcvd);
|
||||
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);
|
||||
if (info.capture_start_ntp_time_ms >= 0) {
|
||||
report->AddInt64(StatsReport::kStatsValueNameCaptureStartNtpTimeMs,
|
||||
info.capture_start_ntp_time_ms);
|
||||
|
|
@ -301,8 +326,10 @@ void ExtractStats(const cricket::VideoReceiverInfo& info, StatsReport* report) {
|
|||
webrtc::videocontenttypehelpers::ToString(info.content_type));
|
||||
}
|
||||
|
||||
void ExtractStats(const cricket::VideoSenderInfo& info, StatsReport* report) {
|
||||
ExtractCommonSendProperties(info, report);
|
||||
void ExtractStats(const cricket::VideoSenderInfo& info,
|
||||
StatsReport* report,
|
||||
bool use_standard_bytes_stats) {
|
||||
ExtractCommonSendProperties(info, report, use_standard_bytes_stats);
|
||||
|
||||
report->AddString(StatsReport::kStatsValueNameCodecImplementationName,
|
||||
info.encoder_implementation_name);
|
||||
|
|
@ -417,7 +444,7 @@ void ExtractStatsFromList(
|
|||
StatsReport* report =
|
||||
collector->PrepareReport(true, ssrc, track_id, transport_id, direction);
|
||||
if (report)
|
||||
ExtractStats(d, report);
|
||||
ExtractStats(d, report, collector->UseStandardBytesStats());
|
||||
|
||||
if (!d.remote_stats.empty()) {
|
||||
report = collector->PrepareReport(false, ssrc, track_id, transport_id,
|
||||
|
|
@ -470,7 +497,10 @@ const char* AdapterTypeToStatsType(rtc::AdapterType type) {
|
|||
}
|
||||
|
||||
StatsCollector::StatsCollector(PeerConnectionInternal* pc)
|
||||
: pc_(pc), stats_gathering_started_(0) {
|
||||
: pc_(pc),
|
||||
stats_gathering_started_(0),
|
||||
use_standard_bytes_stats_(
|
||||
webrtc::field_trial::IsEnabled(kUseStandardBytesStats)) {
|
||||
RTC_DCHECK(pc_);
|
||||
}
|
||||
|
||||
|
|
|
|||
|
|
@ -94,6 +94,8 @@ class StatsCollector {
|
|||
// ignored.
|
||||
void ClearUpdateStatsCacheForTest();
|
||||
|
||||
bool UseStandardBytesStats() const { return use_standard_bytes_stats_; }
|
||||
|
||||
private:
|
||||
friend class StatsCollectorTest;
|
||||
|
||||
|
|
@ -143,6 +145,7 @@ 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.
|
||||
|
|
|
|||
|
|
@ -324,7 +324,9 @@ 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.bytes_rcvd), value_in_report);
|
||||
EXPECT_EQ(rtc::ToString(info.payload_bytes_rcvd +
|
||||
info.header_and_padding_bytes_rcvd),
|
||||
value_in_report);
|
||||
EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNameJitterReceived,
|
||||
&value_in_report));
|
||||
EXPECT_EQ(rtc::ToString(info.jitter_ms), value_in_report);
|
||||
|
|
@ -397,7 +399,9 @@ 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.bytes_sent), value_in_report);
|
||||
EXPECT_EQ(rtc::ToString(sinfo.payload_bytes_sent +
|
||||
sinfo.header_and_padding_bytes_sent),
|
||||
value_in_report);
|
||||
EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNamePacketsSent,
|
||||
&value_in_report));
|
||||
EXPECT_EQ(rtc::ToString(sinfo.packets_sent), value_in_report);
|
||||
|
|
@ -528,7 +532,8 @@ 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->bytes_sent = 100;
|
||||
voice_sender_info->payload_bytes_sent = 88;
|
||||
voice_sender_info->header_and_padding_bytes_sent = 12;
|
||||
voice_sender_info->packets_sent = 101;
|
||||
voice_sender_info->rtt_ms = 102;
|
||||
voice_sender_info->fraction_lost = 103;
|
||||
|
|
@ -563,7 +568,8 @@ void UpdateVoiceSenderInfoFromAudioTrack(
|
|||
|
||||
void InitVoiceReceiverInfo(cricket::VoiceReceiverInfo* voice_receiver_info) {
|
||||
voice_receiver_info->add_ssrc(kSsrcOfTrack);
|
||||
voice_receiver_info->bytes_rcvd = 110;
|
||||
voice_receiver_info->payload_bytes_rcvd = 98;
|
||||
voice_receiver_info->header_and_padding_bytes_rcvd = 12;
|
||||
voice_receiver_info->packets_rcvd = 111;
|
||||
voice_receiver_info->packets_lost = 114;
|
||||
voice_receiver_info->jitter_ms = 116;
|
||||
|
|
@ -904,7 +910,8 @@ TEST_P(StatsCollectorTrackTest, BytesCounterHandles64Bits) {
|
|||
|
||||
VideoSenderInfo video_sender_info;
|
||||
video_sender_info.add_ssrc(1234);
|
||||
video_sender_info.bytes_sent = kBytesSent;
|
||||
video_sender_info.payload_bytes_sent = kBytesSent;
|
||||
video_sender_info.header_and_padding_bytes_sent = 0;
|
||||
VideoMediaInfo video_info;
|
||||
video_info.senders.push_back(video_sender_info);
|
||||
|
||||
|
|
@ -936,7 +943,8 @@ TEST_P(StatsCollectorTrackTest, AudioBandwidthEstimationInfoIsReported) {
|
|||
|
||||
VoiceSenderInfo voice_sender_info;
|
||||
voice_sender_info.add_ssrc(1234);
|
||||
voice_sender_info.bytes_sent = kBytesSent;
|
||||
voice_sender_info.payload_bytes_sent = kBytesSent - 12;
|
||||
voice_sender_info.header_and_padding_bytes_sent = 12;
|
||||
VoiceMediaInfo voice_info;
|
||||
voice_info.senders.push_back(voice_sender_info);
|
||||
|
||||
|
|
@ -984,7 +992,9 @@ TEST_P(StatsCollectorTrackTest, VideoBandwidthEstimationInfoIsReported) {
|
|||
|
||||
VideoSenderInfo video_sender_info;
|
||||
video_sender_info.add_ssrc(1234);
|
||||
video_sender_info.bytes_sent = kBytesSent;
|
||||
video_sender_info.payload_bytes_sent = kBytesSent - 12;
|
||||
video_sender_info.header_and_padding_bytes_sent = 12;
|
||||
|
||||
VideoMediaInfo video_info;
|
||||
video_info.senders.push_back(video_sender_info);
|
||||
|
||||
|
|
@ -1081,7 +1091,8 @@ TEST_P(StatsCollectorTrackTest, TrackAndSsrcObjectExistAfterUpdateSsrcStats) {
|
|||
|
||||
VideoSenderInfo video_sender_info;
|
||||
video_sender_info.add_ssrc(1234);
|
||||
video_sender_info.bytes_sent = kBytesSent;
|
||||
video_sender_info.payload_bytes_sent = kBytesSent - 12;
|
||||
video_sender_info.header_and_padding_bytes_sent = 12;
|
||||
VideoMediaInfo video_info;
|
||||
video_info.senders.push_back(video_sender_info);
|
||||
|
||||
|
|
@ -1135,7 +1146,8 @@ TEST_P(StatsCollectorTrackTest, TransportObjectLinkedFromSsrcObject) {
|
|||
|
||||
VideoSenderInfo video_sender_info;
|
||||
video_sender_info.add_ssrc(1234);
|
||||
video_sender_info.bytes_sent = kBytesSent;
|
||||
video_sender_info.payload_bytes_sent = kBytesSent - 12;
|
||||
video_sender_info.header_and_padding_bytes_sent = 12;
|
||||
VideoMediaInfo video_info;
|
||||
video_info.senders.push_back(video_sender_info);
|
||||
|
||||
|
|
|
|||
|
|
@ -598,6 +598,7 @@ WEBRTC_RTCSTATS_IMPL(
|
|||
RTCInboundRTPStreamStats, RTCRTPStreamStats, "inbound-rtp",
|
||||
&packets_received,
|
||||
&bytes_received,
|
||||
&header_bytes_received,
|
||||
&packets_lost,
|
||||
&last_packet_received_timestamp,
|
||||
&jitter,
|
||||
|
|
@ -630,6 +631,7 @@ 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"),
|
||||
|
|
@ -657,6 +659,7 @@ 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),
|
||||
|
|
@ -686,6 +689,7 @@ WEBRTC_RTCSTATS_IMPL(
|
|||
&packets_sent,
|
||||
&retransmitted_packets_sent,
|
||||
&bytes_sent,
|
||||
&header_bytes_sent,
|
||||
&retransmitted_bytes_sent,
|
||||
&target_bitrate,
|
||||
&frames_encoded,
|
||||
|
|
@ -710,6 +714,7 @@ 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"),
|
||||
|
|
@ -730,6 +735,7 @@ 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),
|
||||
|
|
|
|||
Loading…
Reference in a new issue