webrtc_m130/stats/rtcstats_objects.cc

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/*
* Copyright 2016 The WebRTC Project Authors. All rights reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "api/stats/rtcstats_objects.h"
#include <utility>
#include "api/stats/rtc_stats.h"
#include "rtc_base/checks.h"
namespace webrtc {
const char* const RTCDataChannelState::kConnecting = "connecting";
const char* const RTCDataChannelState::kOpen = "open";
const char* const RTCDataChannelState::kClosing = "closing";
const char* const RTCDataChannelState::kClosed = "closed";
const char* const RTCStatsIceCandidatePairState::kFrozen = "frozen";
const char* const RTCStatsIceCandidatePairState::kWaiting = "waiting";
const char* const RTCStatsIceCandidatePairState::kInProgress = "in-progress";
const char* const RTCStatsIceCandidatePairState::kFailed = "failed";
const char* const RTCStatsIceCandidatePairState::kSucceeded = "succeeded";
// Strings defined in https://tools.ietf.org/html/rfc5245.
const char* const RTCIceCandidateType::kHost = "host";
const char* const RTCIceCandidateType::kSrflx = "srflx";
const char* const RTCIceCandidateType::kPrflx = "prflx";
const char* const RTCIceCandidateType::kRelay = "relay";
const char* const RTCDtlsTransportState::kNew = "new";
const char* const RTCDtlsTransportState::kConnecting = "connecting";
const char* const RTCDtlsTransportState::kConnected = "connected";
const char* const RTCDtlsTransportState::kClosed = "closed";
const char* const RTCDtlsTransportState::kFailed = "failed";
const char* const RTCMediaStreamTrackKind::kAudio = "audio";
const char* const RTCMediaStreamTrackKind::kVideo = "video";
// https://w3c.github.io/webrtc-stats/#dom-rtcnetworktype
const char* const RTCNetworkType::kBluetooth = "bluetooth";
const char* const RTCNetworkType::kCellular = "cellular";
const char* const RTCNetworkType::kEthernet = "ethernet";
const char* const RTCNetworkType::kWifi = "wifi";
const char* const RTCNetworkType::kWimax = "wimax";
const char* const RTCNetworkType::kVpn = "vpn";
const char* const RTCNetworkType::kUnknown = "unknown";
// https://w3c.github.io/webrtc-stats/#dom-rtcqualitylimitationreason
const char* const RTCQualityLimitationReason::kNone = "none";
const char* const RTCQualityLimitationReason::kCpu = "cpu";
const char* const RTCQualityLimitationReason::kBandwidth = "bandwidth";
const char* const RTCQualityLimitationReason::kOther = "other";
// https://webrtc.org/experiments/rtp-hdrext/video-content-type/
const char* const RTCContentType::kUnspecified = "unspecified";
const char* const RTCContentType::kScreenshare = "screenshare";
// clang-format off
WEBRTC_RTCSTATS_IMPL(RTCCertificateStats, RTCStats, "certificate",
&fingerprint,
&fingerprint_algorithm,
&base64_certificate,
&issuer_certificate_id)
// clang-format on
RTCCertificateStats::RTCCertificateStats(const std::string& id,
int64_t timestamp_us)
: RTCCertificateStats(std::string(id), timestamp_us) {}
RTCCertificateStats::RTCCertificateStats(std::string&& id, int64_t timestamp_us)
: RTCStats(std::move(id), timestamp_us),
fingerprint("fingerprint"),
fingerprint_algorithm("fingerprintAlgorithm"),
base64_certificate("base64Certificate"),
issuer_certificate_id("issuerCertificateId") {}
RTCCertificateStats::RTCCertificateStats(const RTCCertificateStats& other)
: RTCStats(other.id(), other.timestamp_us()),
fingerprint(other.fingerprint),
fingerprint_algorithm(other.fingerprint_algorithm),
base64_certificate(other.base64_certificate),
issuer_certificate_id(other.issuer_certificate_id) {}
RTCCertificateStats::~RTCCertificateStats() {}
// clang-format off
WEBRTC_RTCSTATS_IMPL(RTCCodecStats, RTCStats, "codec",
&payload_type,
&mime_type,
&clock_rate,
&channels,
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&sdp_fmtp_line)
// clang-format on
RTCCodecStats::RTCCodecStats(const std::string& id, int64_t timestamp_us)
: RTCCodecStats(std::string(id), timestamp_us) {}
RTCCodecStats::RTCCodecStats(std::string&& id, int64_t timestamp_us)
: RTCStats(std::move(id), timestamp_us),
payload_type("payloadType"),
mime_type("mimeType"),
clock_rate("clockRate"),
channels("channels"),
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sdp_fmtp_line("sdpFmtpLine") {}
RTCCodecStats::RTCCodecStats(const RTCCodecStats& other)
: RTCStats(other.id(), other.timestamp_us()),
payload_type(other.payload_type),
mime_type(other.mime_type),
clock_rate(other.clock_rate),
channels(other.channels),
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sdp_fmtp_line(other.sdp_fmtp_line) {}
RTCCodecStats::~RTCCodecStats() {}
// clang-format off
WEBRTC_RTCSTATS_IMPL(RTCDataChannelStats, RTCStats, "data-channel",
&label,
&protocol,
&data_channel_identifier,
&state,
&messages_sent,
&bytes_sent,
&messages_received,
&bytes_received)
// clang-format on
RTCDataChannelStats::RTCDataChannelStats(const std::string& id,
int64_t timestamp_us)
: RTCDataChannelStats(std::string(id), timestamp_us) {}
RTCDataChannelStats::RTCDataChannelStats(std::string&& id, int64_t timestamp_us)
: RTCStats(std::move(id), timestamp_us),
label("label"),
protocol("protocol"),
data_channel_identifier("dataChannelIdentifier"),
state("state"),
messages_sent("messagesSent"),
bytes_sent("bytesSent"),
messages_received("messagesReceived"),
bytes_received("bytesReceived") {}
RTCDataChannelStats::RTCDataChannelStats(const RTCDataChannelStats& other)
: RTCStats(other.id(), other.timestamp_us()),
label(other.label),
protocol(other.protocol),
data_channel_identifier(other.data_channel_identifier),
state(other.state),
messages_sent(other.messages_sent),
bytes_sent(other.bytes_sent),
messages_received(other.messages_received),
bytes_received(other.bytes_received) {}
RTCDataChannelStats::~RTCDataChannelStats() {}
// clang-format off
WEBRTC_RTCSTATS_IMPL(RTCIceCandidatePairStats, RTCStats, "candidate-pair",
&transport_id,
&local_candidate_id,
&remote_candidate_id,
&state,
&priority,
&nominated,
&writable,
&readable,
&bytes_sent,
&bytes_received,
&total_round_trip_time,
&current_round_trip_time,
&available_outgoing_bitrate,
&available_incoming_bitrate,
&requests_received,
&requests_sent,
&responses_received,
&responses_sent,
&retransmissions_received,
&retransmissions_sent,
&consent_requests_received,
&consent_requests_sent,
&consent_responses_received,
&consent_responses_sent)
// clang-format on
RTCIceCandidatePairStats::RTCIceCandidatePairStats(const std::string& id,
int64_t timestamp_us)
: RTCIceCandidatePairStats(std::string(id), timestamp_us) {}
RTCIceCandidatePairStats::RTCIceCandidatePairStats(std::string&& id,
int64_t timestamp_us)
: RTCStats(std::move(id), timestamp_us),
transport_id("transportId"),
local_candidate_id("localCandidateId"),
remote_candidate_id("remoteCandidateId"),
state("state"),
priority("priority"),
nominated("nominated"),
writable("writable"),
readable("readable"),
bytes_sent("bytesSent"),
bytes_received("bytesReceived"),
total_round_trip_time("totalRoundTripTime"),
current_round_trip_time("currentRoundTripTime"),
available_outgoing_bitrate("availableOutgoingBitrate"),
available_incoming_bitrate("availableIncomingBitrate"),
requests_received("requestsReceived"),
requests_sent("requestsSent"),
responses_received("responsesReceived"),
responses_sent("responsesSent"),
retransmissions_received("retransmissionsReceived"),
retransmissions_sent("retransmissionsSent"),
consent_requests_received("consentRequestsReceived"),
consent_requests_sent("consentRequestsSent"),
consent_responses_received("consentResponsesReceived"),
consent_responses_sent("consentResponsesSent") {}
RTCIceCandidatePairStats::RTCIceCandidatePairStats(
const RTCIceCandidatePairStats& other)
: RTCStats(other.id(), other.timestamp_us()),
transport_id(other.transport_id),
local_candidate_id(other.local_candidate_id),
remote_candidate_id(other.remote_candidate_id),
state(other.state),
priority(other.priority),
nominated(other.nominated),
writable(other.writable),
readable(other.readable),
bytes_sent(other.bytes_sent),
bytes_received(other.bytes_received),
total_round_trip_time(other.total_round_trip_time),
current_round_trip_time(other.current_round_trip_time),
available_outgoing_bitrate(other.available_outgoing_bitrate),
available_incoming_bitrate(other.available_incoming_bitrate),
requests_received(other.requests_received),
requests_sent(other.requests_sent),
responses_received(other.responses_received),
responses_sent(other.responses_sent),
retransmissions_received(other.retransmissions_received),
retransmissions_sent(other.retransmissions_sent),
consent_requests_received(other.consent_requests_received),
consent_requests_sent(other.consent_requests_sent),
consent_responses_received(other.consent_responses_received),
consent_responses_sent(other.consent_responses_sent) {}
RTCIceCandidatePairStats::~RTCIceCandidatePairStats() {}
// clang-format off
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WEBRTC_RTCSTATS_IMPL(RTCIceCandidateStats, RTCStats, "abstract-ice-candidate",
&transport_id,
&is_remote,
&network_type,
&ip,
&port,
&protocol,
&relay_protocol,
&candidate_type,
&priority,
&url,
&deleted)
// clang-format on
RTCIceCandidateStats::RTCIceCandidateStats(const std::string& id,
int64_t timestamp_us,
bool is_remote)
: RTCIceCandidateStats(std::string(id), timestamp_us, is_remote) {}
RTCIceCandidateStats::RTCIceCandidateStats(std::string&& id,
int64_t timestamp_us,
bool is_remote)
: RTCStats(std::move(id), timestamp_us),
transport_id("transportId"),
is_remote("isRemote", is_remote),
network_type("networkType"),
ip("ip"),
port("port"),
protocol("protocol"),
relay_protocol("relayProtocol"),
candidate_type("candidateType"),
priority("priority"),
url("url"),
deleted("deleted", false) {}
RTCIceCandidateStats::RTCIceCandidateStats(const RTCIceCandidateStats& other)
: RTCStats(other.id(), other.timestamp_us()),
transport_id(other.transport_id),
is_remote(other.is_remote),
network_type(other.network_type),
ip(other.ip),
port(other.port),
protocol(other.protocol),
relay_protocol(other.relay_protocol),
candidate_type(other.candidate_type),
priority(other.priority),
url(other.url),
deleted(other.deleted) {}
RTCIceCandidateStats::~RTCIceCandidateStats() {}
const char RTCLocalIceCandidateStats::kType[] = "local-candidate";
RTCLocalIceCandidateStats::RTCLocalIceCandidateStats(const std::string& id,
int64_t timestamp_us)
: RTCIceCandidateStats(id, timestamp_us, false) {}
RTCLocalIceCandidateStats::RTCLocalIceCandidateStats(std::string&& id,
int64_t timestamp_us)
: RTCIceCandidateStats(std::move(id), timestamp_us, false) {}
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std::unique_ptr<RTCStats> RTCLocalIceCandidateStats::copy() const {
return std::unique_ptr<RTCStats>(new RTCLocalIceCandidateStats(*this));
}
const char* RTCLocalIceCandidateStats::type() const {
return kType;
}
const char RTCRemoteIceCandidateStats::kType[] = "remote-candidate";
RTCRemoteIceCandidateStats::RTCRemoteIceCandidateStats(const std::string& id,
int64_t timestamp_us)
: RTCIceCandidateStats(id, timestamp_us, true) {}
RTCRemoteIceCandidateStats::RTCRemoteIceCandidateStats(std::string&& id,
int64_t timestamp_us)
: RTCIceCandidateStats(std::move(id), timestamp_us, true) {}
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std::unique_ptr<RTCStats> RTCRemoteIceCandidateStats::copy() const {
return std::unique_ptr<RTCStats>(new RTCRemoteIceCandidateStats(*this));
}
const char* RTCRemoteIceCandidateStats::type() const {
return kType;
}
// clang-format off
WEBRTC_RTCSTATS_IMPL(RTCMediaStreamStats, RTCStats, "stream",
&stream_identifier,
&track_ids)
// clang-format on
RTCMediaStreamStats::RTCMediaStreamStats(const std::string& id,
int64_t timestamp_us)
: RTCMediaStreamStats(std::string(id), timestamp_us) {}
RTCMediaStreamStats::RTCMediaStreamStats(std::string&& id, int64_t timestamp_us)
: RTCStats(std::move(id), timestamp_us),
stream_identifier("streamIdentifier"),
track_ids("trackIds") {}
RTCMediaStreamStats::RTCMediaStreamStats(const RTCMediaStreamStats& other)
: RTCStats(other.id(), other.timestamp_us()),
stream_identifier(other.stream_identifier),
track_ids(other.track_ids) {}
RTCMediaStreamStats::~RTCMediaStreamStats() {}
// clang-format off
WEBRTC_RTCSTATS_IMPL(RTCMediaStreamTrackStats, RTCStats, "track",
&track_identifier,
&media_source_id,
&remote_source,
&ended,
&detached,
&kind,
&jitter_buffer_delay,
&jitter_buffer_emitted_count,
&frame_width,
&frame_height,
&frames_per_second,
&frames_sent,
&huge_frames_sent,
&frames_received,
&frames_decoded,
&frames_dropped,
&frames_corrupted,
&partial_frames_lost,
&full_frames_lost,
&audio_level,
&total_audio_energy,
&echo_return_loss,
&echo_return_loss_enhancement,
&total_samples_received,
&total_samples_duration,
&concealed_samples,
&silent_concealed_samples,
&concealment_events,
&inserted_samples_for_deceleration,
&removed_samples_for_acceleration,
&jitter_buffer_flushes,
&delayed_packet_outage_samples,
&relative_packet_arrival_delay,
&jitter_buffer_target_delay,
&interruption_count,
&total_interruption_duration,
&freeze_count,
&pause_count,
&total_freezes_duration,
&total_pauses_duration,
&total_frames_duration,
&sum_squared_frame_durations)
// clang-format on
RTCMediaStreamTrackStats::RTCMediaStreamTrackStats(const std::string& id,
int64_t timestamp_us,
const char* kind)
: RTCMediaStreamTrackStats(std::string(id), timestamp_us, kind) {}
RTCMediaStreamTrackStats::RTCMediaStreamTrackStats(std::string&& id,
int64_t timestamp_us,
const char* kind)
: RTCStats(std::move(id), timestamp_us),
track_identifier("trackIdentifier"),
media_source_id("mediaSourceId"),
remote_source("remoteSource"),
ended("ended"),
detached("detached"),
kind("kind", kind),
jitter_buffer_delay("jitterBufferDelay"),
jitter_buffer_emitted_count("jitterBufferEmittedCount"),
frame_width("frameWidth"),
frame_height("frameHeight"),
frames_per_second("framesPerSecond"),
frames_sent("framesSent"),
huge_frames_sent("hugeFramesSent"),
frames_received("framesReceived"),
frames_decoded("framesDecoded"),
frames_dropped("framesDropped"),
frames_corrupted("framesCorrupted"),
partial_frames_lost("partialFramesLost"),
full_frames_lost("fullFramesLost"),
audio_level("audioLevel"),
total_audio_energy("totalAudioEnergy"),
echo_return_loss("echoReturnLoss"),
echo_return_loss_enhancement("echoReturnLossEnhancement"),
total_samples_received("totalSamplesReceived"),
total_samples_duration("totalSamplesDuration"),
concealed_samples("concealedSamples"),
silent_concealed_samples("silentConcealedSamples"),
concealment_events("concealmentEvents"),
inserted_samples_for_deceleration("insertedSamplesForDeceleration"),
removed_samples_for_acceleration("removedSamplesForAcceleration"),
jitter_buffer_flushes(
"jitterBufferFlushes",
{NonStandardGroupId::kRtcAudioJitterBufferMaxPackets}),
delayed_packet_outage_samples(
"delayedPacketOutageSamples",
{NonStandardGroupId::kRtcAudioJitterBufferMaxPackets,
NonStandardGroupId::kRtcStatsRelativePacketArrivalDelay}),
relative_packet_arrival_delay(
"relativePacketArrivalDelay",
{NonStandardGroupId::kRtcStatsRelativePacketArrivalDelay}),
jitter_buffer_target_delay("jitterBufferTargetDelay"),
interruption_count("interruptionCount"),
total_interruption_duration("totalInterruptionDuration"),
freeze_count("freezeCount"),
pause_count("pauseCount"),
total_freezes_duration("totalFreezesDuration"),
total_pauses_duration("totalPausesDuration"),
total_frames_duration("totalFramesDuration"),
sum_squared_frame_durations("sumOfSquaredFramesDuration") {
RTC_DCHECK(kind == RTCMediaStreamTrackKind::kAudio ||
kind == RTCMediaStreamTrackKind::kVideo);
}
RTCMediaStreamTrackStats::RTCMediaStreamTrackStats(
const RTCMediaStreamTrackStats& other)
: RTCStats(other.id(), other.timestamp_us()),
track_identifier(other.track_identifier),
media_source_id(other.media_source_id),
remote_source(other.remote_source),
ended(other.ended),
detached(other.detached),
kind(other.kind),
jitter_buffer_delay(other.jitter_buffer_delay),
jitter_buffer_emitted_count(other.jitter_buffer_emitted_count),
frame_width(other.frame_width),
frame_height(other.frame_height),
frames_per_second(other.frames_per_second),
frames_sent(other.frames_sent),
huge_frames_sent(other.huge_frames_sent),
frames_received(other.frames_received),
frames_decoded(other.frames_decoded),
frames_dropped(other.frames_dropped),
frames_corrupted(other.frames_corrupted),
partial_frames_lost(other.partial_frames_lost),
full_frames_lost(other.full_frames_lost),
audio_level(other.audio_level),
total_audio_energy(other.total_audio_energy),
echo_return_loss(other.echo_return_loss),
echo_return_loss_enhancement(other.echo_return_loss_enhancement),
total_samples_received(other.total_samples_received),
total_samples_duration(other.total_samples_duration),
concealed_samples(other.concealed_samples),
silent_concealed_samples(other.silent_concealed_samples),
concealment_events(other.concealment_events),
inserted_samples_for_deceleration(
other.inserted_samples_for_deceleration),
removed_samples_for_acceleration(other.removed_samples_for_acceleration),
jitter_buffer_flushes(other.jitter_buffer_flushes),
delayed_packet_outage_samples(other.delayed_packet_outage_samples),
relative_packet_arrival_delay(other.relative_packet_arrival_delay),
jitter_buffer_target_delay(other.jitter_buffer_target_delay),
interruption_count(other.interruption_count),
total_interruption_duration(other.total_interruption_duration),
freeze_count(other.freeze_count),
pause_count(other.pause_count),
total_freezes_duration(other.total_freezes_duration),
total_pauses_duration(other.total_pauses_duration),
total_frames_duration(other.total_frames_duration),
sum_squared_frame_durations(other.sum_squared_frame_durations) {}
RTCMediaStreamTrackStats::~RTCMediaStreamTrackStats() {}
// clang-format off
WEBRTC_RTCSTATS_IMPL(RTCPeerConnectionStats, RTCStats, "peer-connection",
&data_channels_opened,
&data_channels_closed)
// clang-format on
RTCPeerConnectionStats::RTCPeerConnectionStats(const std::string& id,
int64_t timestamp_us)
: RTCPeerConnectionStats(std::string(id), timestamp_us) {}
RTCPeerConnectionStats::RTCPeerConnectionStats(std::string&& id,
int64_t timestamp_us)
: RTCStats(std::move(id), timestamp_us),
data_channels_opened("dataChannelsOpened"),
data_channels_closed("dataChannelsClosed") {}
RTCPeerConnectionStats::RTCPeerConnectionStats(
const RTCPeerConnectionStats& other)
: RTCStats(other.id(), other.timestamp_us()),
data_channels_opened(other.data_channels_opened),
data_channels_closed(other.data_channels_closed) {}
RTCPeerConnectionStats::~RTCPeerConnectionStats() {}
// clang-format off
WEBRTC_RTCSTATS_IMPL(RTCRTPStreamStats, RTCStats, "rtp",
&ssrc,
&is_remote,
&media_type,
&kind,
&track_id,
&transport_id,
&codec_id,
&fir_count,
&pli_count,
&nack_count,
&sli_count,
&qp_sum)
// clang-format on
RTCRTPStreamStats::RTCRTPStreamStats(const std::string& id,
int64_t timestamp_us)
: RTCRTPStreamStats(std::string(id), timestamp_us) {}
RTCRTPStreamStats::RTCRTPStreamStats(std::string&& id, int64_t timestamp_us)
: RTCStats(std::move(id), timestamp_us),
ssrc("ssrc"),
is_remote("isRemote", false),
media_type("mediaType"),
kind("kind"),
track_id("trackId"),
transport_id("transportId"),
codec_id("codecId"),
fir_count("firCount"),
pli_count("pliCount"),
nack_count("nackCount"),
sli_count("sliCount"),
qp_sum("qpSum") {}
RTCRTPStreamStats::RTCRTPStreamStats(const RTCRTPStreamStats& other)
: RTCStats(other.id(), other.timestamp_us()),
ssrc(other.ssrc),
is_remote(other.is_remote),
media_type(other.media_type),
kind(other.kind),
track_id(other.track_id),
transport_id(other.transport_id),
codec_id(other.codec_id),
fir_count(other.fir_count),
pli_count(other.pli_count),
nack_count(other.nack_count),
sli_count(other.sli_count),
qp_sum(other.qp_sum) {}
RTCRTPStreamStats::~RTCRTPStreamStats() {}
// clang-format off
WEBRTC_RTCSTATS_IMPL(
RTCInboundRTPStreamStats, RTCRTPStreamStats, "inbound-rtp",
&packets_received,
&fec_packets_received,
&fec_packets_discarded,
&bytes_received,
&header_bytes_received,
&packets_lost,
&last_packet_received_timestamp,
&jitter,
&round_trip_time,
&packets_discarded,
&packets_repaired,
&burst_packets_lost,
&burst_packets_discarded,
&burst_loss_count,
&burst_discard_count,
&burst_loss_rate,
&burst_discard_rate,
&gap_loss_rate,
&gap_discard_rate,
&frames_decoded,
&key_frames_decoded,
&total_decode_time,
&total_inter_frame_delay,
&total_squared_inter_frame_delay,
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&content_type,
&estimated_playout_timestamp,
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&decoder_implementation)
// clang-format on
RTCInboundRTPStreamStats::RTCInboundRTPStreamStats(const std::string& id,
int64_t timestamp_us)
: RTCInboundRTPStreamStats(std::string(id), timestamp_us) {}
RTCInboundRTPStreamStats::RTCInboundRTPStreamStats(std::string&& id,
int64_t timestamp_us)
: RTCRTPStreamStats(std::move(id), timestamp_us),
packets_received("packetsReceived"),
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"),
round_trip_time("roundTripTime"),
packets_discarded("packetsDiscarded"),
packets_repaired("packetsRepaired"),
burst_packets_lost("burstPacketsLost"),
burst_packets_discarded("burstPacketsDiscarded"),
burst_loss_count("burstLossCount"),
burst_discard_count("burstDiscardCount"),
burst_loss_rate("burstLossRate"),
burst_discard_rate("burstDiscardRate"),
gap_loss_rate("gapLossRate"),
gap_discard_rate("gapDiscardRate"),
frames_decoded("framesDecoded"),
key_frames_decoded("keyFramesDecoded"),
total_decode_time("totalDecodeTime"),
total_inter_frame_delay("totalInterFrameDelay"),
total_squared_inter_frame_delay("totalSquaredInterFrameDelay"),
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content_type("contentType"),
estimated_playout_timestamp("estimatedPlayoutTimestamp"),
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decoder_implementation("decoderImplementation") {}
RTCInboundRTPStreamStats::RTCInboundRTPStreamStats(
const RTCInboundRTPStreamStats& other)
: RTCRTPStreamStats(other),
packets_received(other.packets_received),
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),
round_trip_time(other.round_trip_time),
packets_discarded(other.packets_discarded),
packets_repaired(other.packets_repaired),
burst_packets_lost(other.burst_packets_lost),
burst_packets_discarded(other.burst_packets_discarded),
burst_loss_count(other.burst_loss_count),
burst_discard_count(other.burst_discard_count),
burst_loss_rate(other.burst_loss_rate),
burst_discard_rate(other.burst_discard_rate),
gap_loss_rate(other.gap_loss_rate),
gap_discard_rate(other.gap_discard_rate),
frames_decoded(other.frames_decoded),
key_frames_decoded(other.key_frames_decoded),
total_decode_time(other.total_decode_time),
total_inter_frame_delay(other.total_inter_frame_delay),
total_squared_inter_frame_delay(other.total_squared_inter_frame_delay),
2019-08-16 13:09:51 +02:00
content_type(other.content_type),
estimated_playout_timestamp(other.estimated_playout_timestamp),
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decoder_implementation(other.decoder_implementation) {}
RTCInboundRTPStreamStats::~RTCInboundRTPStreamStats() {}
// clang-format off
WEBRTC_RTCSTATS_IMPL(
RTCOutboundRTPStreamStats, RTCRTPStreamStats, "outbound-rtp",
&media_source_id,
&remote_id,
Reland "Improve outbound-rtp statistics for simulcast" This reverts commit 9a925c9ce33a6ccdd11b545b11ba68e985c2a65d. Reason for revert: The original CL is updated in PS #2 to fix the googRtt issue which was that when the legacy sender stats were put in "aggregated_senders" we forgot to update rtt_ms the same way that we do it for "senders". Original change's description: > Revert "Improve outbound-rtp statistics for simulcast" > > This reverts commit da6cda839dac7d9d18eba8d365188fa94831e0b1. > > Reason for revert: Breaks googRtt in legacy getStats API > > Original change's description: > > Improve outbound-rtp statistics for simulcast > > > > Bug: webrtc:9547 > > Change-Id: Iec4eb976aa11ee743805425bedb77dcea7c2c9be > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168120 > > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Eldar Rello <elrello@microsoft.com> > > Cr-Commit-Position: refs/heads/master@{#31097} > > TBR=hbos@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,hta@webrtc.org,elrello@microsoft.com > > # Not skipping CQ checks because original CL landed > 1 day ago. > > Bug: webrtc:9547 > Change-Id: I06673328c2a5293a7eef03b3aaf2ded9d13df1b3 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174443 > Reviewed-by: Henrik Boström <hbos@webrtc.org> > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#31165} TBR=hbos@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,hta@webrtc.org,elrello@microsoft.com # Not skipping CQ checks because this is a reland. Bug: webrtc:9547 Change-Id: I723744c496c3c65f95ab6a8940862c8b9f544338 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174480 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31169}
2020-05-05 15:54:46 +02:00
&rid,
&packets_sent,
&retransmitted_packets_sent,
&bytes_sent,
&header_bytes_sent,
&retransmitted_bytes_sent,
&target_bitrate,
&frames_encoded,
&key_frames_encoded,
&total_encode_time,
&total_encoded_bytes_target,
Reland "Improve outbound-rtp statistics for simulcast" This reverts commit 9a925c9ce33a6ccdd11b545b11ba68e985c2a65d. Reason for revert: The original CL is updated in PS #2 to fix the googRtt issue which was that when the legacy sender stats were put in "aggregated_senders" we forgot to update rtt_ms the same way that we do it for "senders". Original change's description: > Revert "Improve outbound-rtp statistics for simulcast" > > This reverts commit da6cda839dac7d9d18eba8d365188fa94831e0b1. > > Reason for revert: Breaks googRtt in legacy getStats API > > Original change's description: > > Improve outbound-rtp statistics for simulcast > > > > Bug: webrtc:9547 > > Change-Id: Iec4eb976aa11ee743805425bedb77dcea7c2c9be > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168120 > > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Eldar Rello <elrello@microsoft.com> > > Cr-Commit-Position: refs/heads/master@{#31097} > > TBR=hbos@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,hta@webrtc.org,elrello@microsoft.com > > # Not skipping CQ checks because original CL landed > 1 day ago. > > Bug: webrtc:9547 > Change-Id: I06673328c2a5293a7eef03b3aaf2ded9d13df1b3 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174443 > Reviewed-by: Henrik Boström <hbos@webrtc.org> > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#31165} TBR=hbos@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,hta@webrtc.org,elrello@microsoft.com # Not skipping CQ checks because this is a reland. Bug: webrtc:9547 Change-Id: I723744c496c3c65f95ab6a8940862c8b9f544338 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174480 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31169}
2020-05-05 15:54:46 +02:00
&frame_width,
&frame_height,
&frames_per_second,
&frames_sent,
&huge_frames_sent,
&total_packet_send_delay,
&quality_limitation_reason,
&quality_limitation_resolution_changes,
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&content_type,
&encoder_implementation)
// clang-format on
RTCOutboundRTPStreamStats::RTCOutboundRTPStreamStats(const std::string& id,
int64_t timestamp_us)
: RTCOutboundRTPStreamStats(std::string(id), timestamp_us) {}
RTCOutboundRTPStreamStats::RTCOutboundRTPStreamStats(std::string&& id,
int64_t timestamp_us)
: RTCRTPStreamStats(std::move(id), timestamp_us),
media_source_id("mediaSourceId"),
remote_id("remoteId"),
Reland "Improve outbound-rtp statistics for simulcast" This reverts commit 9a925c9ce33a6ccdd11b545b11ba68e985c2a65d. Reason for revert: The original CL is updated in PS #2 to fix the googRtt issue which was that when the legacy sender stats were put in "aggregated_senders" we forgot to update rtt_ms the same way that we do it for "senders". Original change's description: > Revert "Improve outbound-rtp statistics for simulcast" > > This reverts commit da6cda839dac7d9d18eba8d365188fa94831e0b1. > > Reason for revert: Breaks googRtt in legacy getStats API > > Original change's description: > > Improve outbound-rtp statistics for simulcast > > > > Bug: webrtc:9547 > > Change-Id: Iec4eb976aa11ee743805425bedb77dcea7c2c9be > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168120 > > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Eldar Rello <elrello@microsoft.com> > > Cr-Commit-Position: refs/heads/master@{#31097} > > TBR=hbos@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,hta@webrtc.org,elrello@microsoft.com > > # Not skipping CQ checks because original CL landed > 1 day ago. > > Bug: webrtc:9547 > Change-Id: I06673328c2a5293a7eef03b3aaf2ded9d13df1b3 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174443 > Reviewed-by: Henrik Boström <hbos@webrtc.org> > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#31165} TBR=hbos@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,hta@webrtc.org,elrello@microsoft.com # Not skipping CQ checks because this is a reland. Bug: webrtc:9547 Change-Id: I723744c496c3c65f95ab6a8940862c8b9f544338 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174480 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31169}
2020-05-05 15:54:46 +02:00
rid("rid"),
packets_sent("packetsSent"),
retransmitted_packets_sent("retransmittedPacketsSent"),
bytes_sent("bytesSent"),
header_bytes_sent("headerBytesSent"),
retransmitted_bytes_sent("retransmittedBytesSent"),
target_bitrate("targetBitrate"),
frames_encoded("framesEncoded"),
key_frames_encoded("keyFramesEncoded"),
total_encode_time("totalEncodeTime"),
total_encoded_bytes_target("totalEncodedBytesTarget"),
Reland "Improve outbound-rtp statistics for simulcast" This reverts commit 9a925c9ce33a6ccdd11b545b11ba68e985c2a65d. Reason for revert: The original CL is updated in PS #2 to fix the googRtt issue which was that when the legacy sender stats were put in "aggregated_senders" we forgot to update rtt_ms the same way that we do it for "senders". Original change's description: > Revert "Improve outbound-rtp statistics for simulcast" > > This reverts commit da6cda839dac7d9d18eba8d365188fa94831e0b1. > > Reason for revert: Breaks googRtt in legacy getStats API > > Original change's description: > > Improve outbound-rtp statistics for simulcast > > > > Bug: webrtc:9547 > > Change-Id: Iec4eb976aa11ee743805425bedb77dcea7c2c9be > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168120 > > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Eldar Rello <elrello@microsoft.com> > > Cr-Commit-Position: refs/heads/master@{#31097} > > TBR=hbos@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,hta@webrtc.org,elrello@microsoft.com > > # Not skipping CQ checks because original CL landed > 1 day ago. > > Bug: webrtc:9547 > Change-Id: I06673328c2a5293a7eef03b3aaf2ded9d13df1b3 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174443 > Reviewed-by: Henrik Boström <hbos@webrtc.org> > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#31165} TBR=hbos@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,hta@webrtc.org,elrello@microsoft.com # Not skipping CQ checks because this is a reland. Bug: webrtc:9547 Change-Id: I723744c496c3c65f95ab6a8940862c8b9f544338 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174480 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31169}
2020-05-05 15:54:46 +02:00
frame_width("frameWidth"),
frame_height("frameHeight"),
frames_per_second("framesPerSecond"),
frames_sent("framesSent"),
huge_frames_sent("hugeFramesSent"),
total_packet_send_delay("totalPacketSendDelay"),
quality_limitation_reason("qualityLimitationReason"),
quality_limitation_resolution_changes(
"qualityLimitationResolutionChanges"),
2019-08-16 13:09:51 +02:00
content_type("contentType"),
encoder_implementation("encoderImplementation") {}
RTCOutboundRTPStreamStats::RTCOutboundRTPStreamStats(
const RTCOutboundRTPStreamStats& other)
: RTCRTPStreamStats(other),
media_source_id(other.media_source_id),
remote_id(other.remote_id),
Reland "Improve outbound-rtp statistics for simulcast" This reverts commit 9a925c9ce33a6ccdd11b545b11ba68e985c2a65d. Reason for revert: The original CL is updated in PS #2 to fix the googRtt issue which was that when the legacy sender stats were put in "aggregated_senders" we forgot to update rtt_ms the same way that we do it for "senders". Original change's description: > Revert "Improve outbound-rtp statistics for simulcast" > > This reverts commit da6cda839dac7d9d18eba8d365188fa94831e0b1. > > Reason for revert: Breaks googRtt in legacy getStats API > > Original change's description: > > Improve outbound-rtp statistics for simulcast > > > > Bug: webrtc:9547 > > Change-Id: Iec4eb976aa11ee743805425bedb77dcea7c2c9be > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168120 > > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Eldar Rello <elrello@microsoft.com> > > Cr-Commit-Position: refs/heads/master@{#31097} > > TBR=hbos@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,hta@webrtc.org,elrello@microsoft.com > > # Not skipping CQ checks because original CL landed > 1 day ago. > > Bug: webrtc:9547 > Change-Id: I06673328c2a5293a7eef03b3aaf2ded9d13df1b3 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174443 > Reviewed-by: Henrik Boström <hbos@webrtc.org> > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#31165} TBR=hbos@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,hta@webrtc.org,elrello@microsoft.com # Not skipping CQ checks because this is a reland. Bug: webrtc:9547 Change-Id: I723744c496c3c65f95ab6a8940862c8b9f544338 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174480 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31169}
2020-05-05 15:54:46 +02:00
rid(other.rid),
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),
key_frames_encoded(other.key_frames_encoded),
total_encode_time(other.total_encode_time),
total_encoded_bytes_target(other.total_encoded_bytes_target),
Reland "Improve outbound-rtp statistics for simulcast" This reverts commit 9a925c9ce33a6ccdd11b545b11ba68e985c2a65d. Reason for revert: The original CL is updated in PS #2 to fix the googRtt issue which was that when the legacy sender stats were put in "aggregated_senders" we forgot to update rtt_ms the same way that we do it for "senders". Original change's description: > Revert "Improve outbound-rtp statistics for simulcast" > > This reverts commit da6cda839dac7d9d18eba8d365188fa94831e0b1. > > Reason for revert: Breaks googRtt in legacy getStats API > > Original change's description: > > Improve outbound-rtp statistics for simulcast > > > > Bug: webrtc:9547 > > Change-Id: Iec4eb976aa11ee743805425bedb77dcea7c2c9be > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168120 > > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Eldar Rello <elrello@microsoft.com> > > Cr-Commit-Position: refs/heads/master@{#31097} > > TBR=hbos@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,hta@webrtc.org,elrello@microsoft.com > > # Not skipping CQ checks because original CL landed > 1 day ago. > > Bug: webrtc:9547 > Change-Id: I06673328c2a5293a7eef03b3aaf2ded9d13df1b3 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174443 > Reviewed-by: Henrik Boström <hbos@webrtc.org> > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#31165} TBR=hbos@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,hta@webrtc.org,elrello@microsoft.com # Not skipping CQ checks because this is a reland. Bug: webrtc:9547 Change-Id: I723744c496c3c65f95ab6a8940862c8b9f544338 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174480 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31169}
2020-05-05 15:54:46 +02:00
frame_width(other.frame_width),
frame_height(other.frame_height),
frames_per_second(other.frames_per_second),
frames_sent(other.frames_sent),
huge_frames_sent(other.huge_frames_sent),
total_packet_send_delay(other.total_packet_send_delay),
quality_limitation_reason(other.quality_limitation_reason),
quality_limitation_resolution_changes(
other.quality_limitation_resolution_changes),
2019-08-16 13:09:51 +02:00
content_type(other.content_type),
encoder_implementation(other.encoder_implementation) {}
RTCOutboundRTPStreamStats::~RTCOutboundRTPStreamStats() {}
// clang-format off
WEBRTC_RTCSTATS_IMPL(
RTCRemoteInboundRtpStreamStats, RTCStats, "remote-inbound-rtp",
&ssrc,
&kind,
&transport_id,
&codec_id,
&packets_lost,
&jitter,
&local_id,
&round_trip_time)
// clang-format on
RTCRemoteInboundRtpStreamStats::RTCRemoteInboundRtpStreamStats(
const std::string& id,
int64_t timestamp_us)
: RTCRemoteInboundRtpStreamStats(std::string(id), timestamp_us) {}
RTCRemoteInboundRtpStreamStats::RTCRemoteInboundRtpStreamStats(
std::string&& id,
int64_t timestamp_us)
: RTCStats(std::move(id), timestamp_us),
ssrc("ssrc"),
kind("kind"),
transport_id("transportId"),
codec_id("codecId"),
packets_lost("packetsLost"),
jitter("jitter"),
local_id("localId"),
round_trip_time("roundTripTime") {}
RTCRemoteInboundRtpStreamStats::RTCRemoteInboundRtpStreamStats(
const RTCRemoteInboundRtpStreamStats& other)
: RTCStats(other),
ssrc(other.ssrc),
kind(other.kind),
transport_id(other.transport_id),
codec_id(other.codec_id),
packets_lost(other.packets_lost),
jitter(other.jitter),
local_id(other.local_id),
round_trip_time(other.round_trip_time) {}
RTCRemoteInboundRtpStreamStats::~RTCRemoteInboundRtpStreamStats() {}
// clang-format off
WEBRTC_RTCSTATS_IMPL(RTCMediaSourceStats, RTCStats, "parent-media-source",
&track_identifier,
&kind)
// clang-format on
RTCMediaSourceStats::RTCMediaSourceStats(const std::string& id,
int64_t timestamp_us)
: RTCMediaSourceStats(std::string(id), timestamp_us) {}
RTCMediaSourceStats::RTCMediaSourceStats(std::string&& id, int64_t timestamp_us)
: RTCStats(std::move(id), timestamp_us),
track_identifier("trackIdentifier"),
kind("kind") {}
RTCMediaSourceStats::RTCMediaSourceStats(const RTCMediaSourceStats& other)
: RTCStats(other.id(), other.timestamp_us()),
track_identifier(other.track_identifier),
kind(other.kind) {}
RTCMediaSourceStats::~RTCMediaSourceStats() {}
// clang-format off
[getStats] Implement "media-source" audio levels, fixing Chrome bug. Implements RTCAudioSourceStats members: - audioLevel - totalAudioEnergy - totalSamplesDuration In this CL description these are collectively referred to as the audio levels. The audio levels are removed from sending "track" stats (in Chrome, these are now reported as undefined instead of 0). Background: For sending tracks, audio levels were always reported as 0 in Chrome (https://crbug.com/736403), while audio levels were correctly reported for receiving tracks. This problem affected the standard getStats() but not the legacy getStats(), blocking some people from migrating. This was likely not a problem in native third_party/webrtc code because the delivery of audio frames from device to send-stream uses a different code path outside of chromium. A recent PR (https://github.com/w3c/webrtc-stats/pull/451) moved the send-side audio levels to the RTCAudioSourceStats, while keeping the receive-side audio levels on the "track" stats. This allows an implementation to report the audio levels even if samples are not sent onto the network (such as if an ICE connection has not been established yet), reflecting some of the current implementation. Changes: 1. Audio levels are added to RTCAudioSourceStats. Send-side audio "track" stats are left undefined. Receive-side audio "track" stats are not changed in this CL and continue to work. 2. Audio level computation is moved from the AudioState and AudioTransportImpl to the AudioSendStream. This is because a) the AudioTransportImpl::RecordedDataIsAvailable() code path is not exercised in chromium, and b) audio levels should, per-spec, not be calculated on a per-call basis, for which the AudioState is defined. 3. The audio level computation is now performed in AudioSendStream::SendAudioData(), a code path used by both native and chromium code. 4. Comments are added to document behavior of existing code, such as AudioLevel and AudioSendStream::SendAudioData(). Note: In this CL, just like before this CL, audio level is only calculated after an AudioSendStream has been created. This means that before an O/A negotiation, audio levels are unavailable. According to spec, if we have an audio source, we should have audio levels. An immediate solution to this would have been to calculate the audio level at pc/rtp_sender.cc. The problem is that the LocalAudioSinkAdapter::OnData() code path, while exercised in chromium, is not exercised in native code. The issue of calculating audio levels on a per-source bases rather than on a per-send stream basis is left to https://crbug.com/webrtc/10771, an existing "media-source" bug. This CL can be verified manually in Chrome at: https://codepen.io/anon/pen/vqRGyq Bug: chromium:736403, webrtc:10771 Change-Id: I8036cd9984f3b187c3177470a8c0d6670a201a5a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143789 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28480}
2019-07-03 17:11:10 +02:00
WEBRTC_RTCSTATS_IMPL(RTCAudioSourceStats, RTCMediaSourceStats, "media-source",
&audio_level,
&total_audio_energy,
&total_samples_duration)
// clang-format on
RTCAudioSourceStats::RTCAudioSourceStats(const std::string& id,
int64_t timestamp_us)
: RTCAudioSourceStats(std::string(id), timestamp_us) {}
RTCAudioSourceStats::RTCAudioSourceStats(std::string&& id, int64_t timestamp_us)
[getStats] Implement "media-source" audio levels, fixing Chrome bug. Implements RTCAudioSourceStats members: - audioLevel - totalAudioEnergy - totalSamplesDuration In this CL description these are collectively referred to as the audio levels. The audio levels are removed from sending "track" stats (in Chrome, these are now reported as undefined instead of 0). Background: For sending tracks, audio levels were always reported as 0 in Chrome (https://crbug.com/736403), while audio levels were correctly reported for receiving tracks. This problem affected the standard getStats() but not the legacy getStats(), blocking some people from migrating. This was likely not a problem in native third_party/webrtc code because the delivery of audio frames from device to send-stream uses a different code path outside of chromium. A recent PR (https://github.com/w3c/webrtc-stats/pull/451) moved the send-side audio levels to the RTCAudioSourceStats, while keeping the receive-side audio levels on the "track" stats. This allows an implementation to report the audio levels even if samples are not sent onto the network (such as if an ICE connection has not been established yet), reflecting some of the current implementation. Changes: 1. Audio levels are added to RTCAudioSourceStats. Send-side audio "track" stats are left undefined. Receive-side audio "track" stats are not changed in this CL and continue to work. 2. Audio level computation is moved from the AudioState and AudioTransportImpl to the AudioSendStream. This is because a) the AudioTransportImpl::RecordedDataIsAvailable() code path is not exercised in chromium, and b) audio levels should, per-spec, not be calculated on a per-call basis, for which the AudioState is defined. 3. The audio level computation is now performed in AudioSendStream::SendAudioData(), a code path used by both native and chromium code. 4. Comments are added to document behavior of existing code, such as AudioLevel and AudioSendStream::SendAudioData(). Note: In this CL, just like before this CL, audio level is only calculated after an AudioSendStream has been created. This means that before an O/A negotiation, audio levels are unavailable. According to spec, if we have an audio source, we should have audio levels. An immediate solution to this would have been to calculate the audio level at pc/rtp_sender.cc. The problem is that the LocalAudioSinkAdapter::OnData() code path, while exercised in chromium, is not exercised in native code. The issue of calculating audio levels on a per-source bases rather than on a per-send stream basis is left to https://crbug.com/webrtc/10771, an existing "media-source" bug. This CL can be verified manually in Chrome at: https://codepen.io/anon/pen/vqRGyq Bug: chromium:736403, webrtc:10771 Change-Id: I8036cd9984f3b187c3177470a8c0d6670a201a5a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143789 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28480}
2019-07-03 17:11:10 +02:00
: RTCMediaSourceStats(std::move(id), timestamp_us),
audio_level("audioLevel"),
total_audio_energy("totalAudioEnergy"),
total_samples_duration("totalSamplesDuration") {}
RTCAudioSourceStats::RTCAudioSourceStats(const RTCAudioSourceStats& other)
[getStats] Implement "media-source" audio levels, fixing Chrome bug. Implements RTCAudioSourceStats members: - audioLevel - totalAudioEnergy - totalSamplesDuration In this CL description these are collectively referred to as the audio levels. The audio levels are removed from sending "track" stats (in Chrome, these are now reported as undefined instead of 0). Background: For sending tracks, audio levels were always reported as 0 in Chrome (https://crbug.com/736403), while audio levels were correctly reported for receiving tracks. This problem affected the standard getStats() but not the legacy getStats(), blocking some people from migrating. This was likely not a problem in native third_party/webrtc code because the delivery of audio frames from device to send-stream uses a different code path outside of chromium. A recent PR (https://github.com/w3c/webrtc-stats/pull/451) moved the send-side audio levels to the RTCAudioSourceStats, while keeping the receive-side audio levels on the "track" stats. This allows an implementation to report the audio levels even if samples are not sent onto the network (such as if an ICE connection has not been established yet), reflecting some of the current implementation. Changes: 1. Audio levels are added to RTCAudioSourceStats. Send-side audio "track" stats are left undefined. Receive-side audio "track" stats are not changed in this CL and continue to work. 2. Audio level computation is moved from the AudioState and AudioTransportImpl to the AudioSendStream. This is because a) the AudioTransportImpl::RecordedDataIsAvailable() code path is not exercised in chromium, and b) audio levels should, per-spec, not be calculated on a per-call basis, for which the AudioState is defined. 3. The audio level computation is now performed in AudioSendStream::SendAudioData(), a code path used by both native and chromium code. 4. Comments are added to document behavior of existing code, such as AudioLevel and AudioSendStream::SendAudioData(). Note: In this CL, just like before this CL, audio level is only calculated after an AudioSendStream has been created. This means that before an O/A negotiation, audio levels are unavailable. According to spec, if we have an audio source, we should have audio levels. An immediate solution to this would have been to calculate the audio level at pc/rtp_sender.cc. The problem is that the LocalAudioSinkAdapter::OnData() code path, while exercised in chromium, is not exercised in native code. The issue of calculating audio levels on a per-source bases rather than on a per-send stream basis is left to https://crbug.com/webrtc/10771, an existing "media-source" bug. This CL can be verified manually in Chrome at: https://codepen.io/anon/pen/vqRGyq Bug: chromium:736403, webrtc:10771 Change-Id: I8036cd9984f3b187c3177470a8c0d6670a201a5a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143789 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28480}
2019-07-03 17:11:10 +02:00
: RTCMediaSourceStats(other),
audio_level(other.audio_level),
total_audio_energy(other.total_audio_energy),
total_samples_duration(other.total_samples_duration) {}
RTCAudioSourceStats::~RTCAudioSourceStats() {}
// clang-format off
WEBRTC_RTCSTATS_IMPL(RTCVideoSourceStats, RTCMediaSourceStats, "media-source",
&width,
&height,
&frames,
&frames_per_second)
// clang-format on
RTCVideoSourceStats::RTCVideoSourceStats(const std::string& id,
int64_t timestamp_us)
: RTCVideoSourceStats(std::string(id), timestamp_us) {}
RTCVideoSourceStats::RTCVideoSourceStats(std::string&& id, int64_t timestamp_us)
: RTCMediaSourceStats(std::move(id), timestamp_us),
width("width"),
height("height"),
frames("frames"),
frames_per_second("framesPerSecond") {}
RTCVideoSourceStats::RTCVideoSourceStats(const RTCVideoSourceStats& other)
: RTCMediaSourceStats(other),
width(other.width),
height(other.height),
frames(other.frames),
frames_per_second(other.frames_per_second) {}
RTCVideoSourceStats::~RTCVideoSourceStats() {}
// clang-format off
WEBRTC_RTCSTATS_IMPL(RTCTransportStats, RTCStats, "transport",
&bytes_sent,
&bytes_received,
&rtcp_transport_stats_id,
&dtls_state,
&selected_candidate_pair_id,
&local_certificate_id,
&remote_certificate_id,
&tls_version,
&dtls_cipher,
&srtp_cipher,
&selected_candidate_pair_changes)
// clang-format on
RTCTransportStats::RTCTransportStats(const std::string& id,
int64_t timestamp_us)
: RTCTransportStats(std::string(id), timestamp_us) {}
RTCTransportStats::RTCTransportStats(std::string&& id, int64_t timestamp_us)
: RTCStats(std::move(id), timestamp_us),
bytes_sent("bytesSent"),
bytes_received("bytesReceived"),
rtcp_transport_stats_id("rtcpTransportStatsId"),
dtls_state("dtlsState"),
selected_candidate_pair_id("selectedCandidatePairId"),
local_certificate_id("localCertificateId"),
remote_certificate_id("remoteCertificateId"),
tls_version("tlsVersion"),
dtls_cipher("dtlsCipher"),
srtp_cipher("srtpCipher"),
selected_candidate_pair_changes("selectedCandidatePairChanges") {}
RTCTransportStats::RTCTransportStats(const RTCTransportStats& other)
: RTCStats(other.id(), other.timestamp_us()),
bytes_sent(other.bytes_sent),
bytes_received(other.bytes_received),
rtcp_transport_stats_id(other.rtcp_transport_stats_id),
dtls_state(other.dtls_state),
selected_candidate_pair_id(other.selected_candidate_pair_id),
local_certificate_id(other.local_certificate_id),
remote_certificate_id(other.remote_certificate_id),
tls_version(other.tls_version),
dtls_cipher(other.dtls_cipher),
srtp_cipher(other.srtp_cipher),
selected_candidate_pair_changes(other.selected_candidate_pair_changes) {}
RTCTransportStats::~RTCTransportStats() {}
} // namespace webrtc