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";
// https://w3c.github.io/webrtc-stats/#dom-rtcdtlsrole
const char* const RTCDtlsRole::kUnknown = "unknown";
const char* const RTCDtlsRole::kClient = "client";
const char* const RTCDtlsRole::kServer = "server";
// https://www.w3.org/TR/webrtc/#rtcicerole
const char* const RTCIceRole::kUnknown = "unknown";
const char* const RTCIceRole::kControlled = "controlled";
const char* const RTCIceRole::kControlling = "controlling";
// https://www.w3.org/TR/webrtc/#dom-rtcicetransportstate
const char* const RTCIceTransportState::kNew = "new";
const char* const RTCIceTransportState::kChecking = "checking";
const char* const RTCIceTransportState::kConnected = "connected";
const char* const RTCIceTransportState::kCompleted = "completed";
const char* const RTCIceTransportState::kDisconnected = "disconnected";
const char* const RTCIceTransportState::kFailed = "failed";
const char* const RTCIceTransportState::kClosed = "closed";
// clang-format off
WEBRTC_RTCSTATS_IMPL(RTCCertificateStats, RTCStats, "certificate",
&fingerprint,
&fingerprint_algorithm,
&base64_certificate,
&issuer_certificate_id)
// clang-format on
RTCCertificateStats::RTCCertificateStats(std::string id, Timestamp timestamp)
: RTCStats(std::move(id), timestamp),
fingerprint("fingerprint"),
fingerprint_algorithm("fingerprintAlgorithm"),
base64_certificate("base64Certificate"),
issuer_certificate_id("issuerCertificateId") {}
RTCCertificateStats::RTCCertificateStats(const RTCCertificateStats& other) =
default;
RTCCertificateStats::~RTCCertificateStats() {}
// clang-format off
WEBRTC_RTCSTATS_IMPL(RTCCodecStats, RTCStats, "codec",
&transport_id,
&payload_type,
&mime_type,
&clock_rate,
&channels,
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&sdp_fmtp_line)
// clang-format on
RTCCodecStats::RTCCodecStats(std::string id, Timestamp timestamp)
: RTCStats(std::move(id), timestamp),
transport_id("transportId"),
payload_type("payloadType"),
mime_type("mimeType"),
clock_rate("clockRate"),
channels("channels"),
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sdp_fmtp_line("sdpFmtpLine") {}
RTCCodecStats::RTCCodecStats(const RTCCodecStats& other) = default;
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(std::string id, Timestamp timestamp)
: RTCStats(std::move(id), timestamp),
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) =
default;
RTCDataChannelStats::~RTCDataChannelStats() {}
// clang-format off
WEBRTC_RTCSTATS_IMPL(RTCIceCandidatePairStats, RTCStats, "candidate-pair",
&transport_id,
&local_candidate_id,
&remote_candidate_id,
&state,
&priority,
&nominated,
&writable,
&packets_sent,
&packets_received,
&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,
&consent_requests_sent,
&packets_discarded_on_send,
&bytes_discarded_on_send,
&last_packet_received_timestamp,
&last_packet_sent_timestamp)
// clang-format on
RTCIceCandidatePairStats::RTCIceCandidatePairStats(std::string id,
Timestamp timestamp)
: RTCStats(std::move(id), timestamp),
transport_id("transportId"),
local_candidate_id("localCandidateId"),
remote_candidate_id("remoteCandidateId"),
state("state"),
priority("priority"),
nominated("nominated"),
writable("writable"),
packets_sent("packetsSent"),
packets_received("packetsReceived"),
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"),
consent_requests_sent("consentRequestsSent"),
packets_discarded_on_send("packetsDiscardedOnSend"),
bytes_discarded_on_send("bytesDiscardedOnSend"),
last_packet_received_timestamp("lastPacketReceivedTimestamp"),
last_packet_sent_timestamp("lastPacketSentTimestamp") {}
RTCIceCandidatePairStats::RTCIceCandidatePairStats(
const RTCIceCandidatePairStats& other) = default;
RTCIceCandidatePairStats::~RTCIceCandidatePairStats() {}
// clang-format off
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WEBRTC_RTCSTATS_IMPL(RTCIceCandidateStats, RTCStats, "abstract-ice-candidate",
&transport_id,
&is_remote,
&network_type,
&ip,
&address,
&port,
&protocol,
&relay_protocol,
&candidate_type,
&priority,
&url,
&foundation,
&related_address,
&related_port,
&username_fragment,
&tcp_type,
&vpn,
&network_adapter_type)
// clang-format on
RTCIceCandidateStats::RTCIceCandidateStats(std::string id,
Timestamp timestamp,
bool is_remote)
: RTCStats(std::move(id), timestamp),
transport_id("transportId"),
is_remote("isRemote", is_remote),
network_type("networkType"),
ip("ip"),
address("address"),
port("port"),
protocol("protocol"),
relay_protocol("relayProtocol"),
candidate_type("candidateType"),
priority("priority"),
url("url"),
foundation("foundation"),
related_address("relatedAddress"),
related_port("relatedPort"),
username_fragment("usernameFragment"),
tcp_type("tcpType"),
vpn("vpn"),
network_adapter_type("networkAdapterType") {}
RTCIceCandidateStats::RTCIceCandidateStats(const RTCIceCandidateStats& other) =
default;
RTCIceCandidateStats::~RTCIceCandidateStats() {}
const char RTCLocalIceCandidateStats::kType[] = "local-candidate";
RTCLocalIceCandidateStats::RTCLocalIceCandidateStats(std::string id,
Timestamp timestamp)
: RTCIceCandidateStats(std::move(id), timestamp, false) {}
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std::unique_ptr<RTCStats> RTCLocalIceCandidateStats::copy() const {
return std::make_unique<RTCLocalIceCandidateStats>(*this);
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}
const char* RTCLocalIceCandidateStats::type() const {
return kType;
}
const char RTCRemoteIceCandidateStats::kType[] = "remote-candidate";
RTCRemoteIceCandidateStats::RTCRemoteIceCandidateStats(std::string id,
Timestamp timestamp)
: RTCIceCandidateStats(std::move(id), timestamp, true) {}
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std::unique_ptr<RTCStats> RTCRemoteIceCandidateStats::copy() const {
return std::make_unique<RTCRemoteIceCandidateStats>(*this);
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}
const char* RTCRemoteIceCandidateStats::type() const {
return kType;
}
// clang-format off
WEBRTC_RTCSTATS_IMPL(DEPRECATED_RTCMediaStreamStats, RTCStats, "stream",
&stream_identifier,
&track_ids)
// clang-format on
DEPRECATED_RTCMediaStreamStats::DEPRECATED_RTCMediaStreamStats(
std::string id,
Timestamp timestamp)
: RTCStats(std::move(id), timestamp),
stream_identifier("streamIdentifier"),
track_ids("trackIds") {}
DEPRECATED_RTCMediaStreamStats::DEPRECATED_RTCMediaStreamStats(
const DEPRECATED_RTCMediaStreamStats& other) = default;
DEPRECATED_RTCMediaStreamStats::~DEPRECATED_RTCMediaStreamStats() {}
// clang-format off
WEBRTC_RTCSTATS_IMPL(DEPRECATED_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_sent,
&huge_frames_sent,
&frames_received,
&frames_decoded,
&frames_dropped,
&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)
// clang-format on
DEPRECATED_RTCMediaStreamTrackStats::DEPRECATED_RTCMediaStreamTrackStats(
std::string id,
Timestamp timestamp,
const char* kind)
: RTCStats(std::move(id), timestamp),
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_sent("framesSent"),
huge_frames_sent("hugeFramesSent"),
frames_received("framesReceived"),
frames_decoded("framesDecoded"),
frames_dropped("framesDropped"),
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") {
RTC_DCHECK(kind == RTCMediaStreamTrackKind::kAudio ||
kind == RTCMediaStreamTrackKind::kVideo);
}
DEPRECATED_RTCMediaStreamTrackStats::DEPRECATED_RTCMediaStreamTrackStats(
const DEPRECATED_RTCMediaStreamTrackStats& other) = default;
DEPRECATED_RTCMediaStreamTrackStats::~DEPRECATED_RTCMediaStreamTrackStats() {}
// clang-format off
WEBRTC_RTCSTATS_IMPL(RTCPeerConnectionStats, RTCStats, "peer-connection",
&data_channels_opened,
&data_channels_closed)
// clang-format on
RTCPeerConnectionStats::RTCPeerConnectionStats(std::string id,
Timestamp timestamp)
: RTCStats(std::move(id), timestamp),
data_channels_opened("dataChannelsOpened"),
data_channels_closed("dataChannelsClosed") {}
RTCPeerConnectionStats::RTCPeerConnectionStats(
const RTCPeerConnectionStats& other) = default;
RTCPeerConnectionStats::~RTCPeerConnectionStats() {}
// clang-format off
WEBRTC_RTCSTATS_IMPL(RTCRTPStreamStats, RTCStats, "rtp",
&ssrc,
&kind,
&track_id,
&transport_id,
&codec_id,
&media_type)
// clang-format on
RTCRTPStreamStats::RTCRTPStreamStats(std::string id, Timestamp timestamp)
: RTCStats(std::move(id), timestamp),
ssrc("ssrc"),
kind("kind"),
track_id("trackId"),
transport_id("transportId"),
codec_id("codecId"),
media_type("mediaType") {}
RTCRTPStreamStats::RTCRTPStreamStats(const RTCRTPStreamStats& other) = default;
RTCRTPStreamStats::~RTCRTPStreamStats() {}
// clang-format off
WEBRTC_RTCSTATS_IMPL(
RTCReceivedRtpStreamStats, RTCRTPStreamStats, "received-rtp",
&jitter,
&packets_lost)
// clang-format on
RTCReceivedRtpStreamStats::RTCReceivedRtpStreamStats(std::string id,
Timestamp timestamp)
: RTCRTPStreamStats(std::move(id), timestamp),
jitter("jitter"),
packets_lost("packetsLost") {}
RTCReceivedRtpStreamStats::RTCReceivedRtpStreamStats(
const RTCReceivedRtpStreamStats& other) = default;
RTCReceivedRtpStreamStats::~RTCReceivedRtpStreamStats() {}
// clang-format off
WEBRTC_RTCSTATS_IMPL(
RTCSentRtpStreamStats, RTCRTPStreamStats, "sent-rtp",
&packets_sent,
&bytes_sent)
// clang-format on
RTCSentRtpStreamStats::RTCSentRtpStreamStats(std::string id,
Timestamp timestamp)
: RTCRTPStreamStats(std::move(id), timestamp),
packets_sent("packetsSent"),
bytes_sent("bytesSent") {}
RTCSentRtpStreamStats::RTCSentRtpStreamStats(
const RTCSentRtpStreamStats& other) = default;
RTCSentRtpStreamStats::~RTCSentRtpStreamStats() {}
// clang-format off
WEBRTC_RTCSTATS_IMPL(
RTCInboundRTPStreamStats, RTCReceivedRtpStreamStats, "inbound-rtp",
&track_identifier,
&mid,
&remote_id,
&packets_received,
&packets_discarded,
&fec_packets_received,
&fec_packets_discarded,
&bytes_received,
&header_bytes_received,
&last_packet_received_timestamp,
&jitter_buffer_delay,
&jitter_buffer_target_delay,
&jitter_buffer_minimum_delay,
&jitter_buffer_emitted_count,
&total_samples_received,
&concealed_samples,
&silent_concealed_samples,
&concealment_events,
&inserted_samples_for_deceleration,
&removed_samples_for_acceleration,
&audio_level,
&total_audio_energy,
&total_samples_duration,
&playout_id,
&frames_received,
&frame_width,
&frame_height,
&frames_per_second,
&frames_decoded,
&key_frames_decoded,
&frames_dropped,
&total_decode_time,
&total_processing_delay,
stats: add frame assembly time stats implements a total frame assembly time statistic that measures the cumulative time between the arrival of the first packet of a frame (the lowest reception time) and the time all packets of the frame have been received (i.e. the highest reception time) This is similar to totalProcessingDelay https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totalprocessingdelay in particular with respect to only being incremented for frames that are being decoded but does not include the amount of time spent decoding the frame. This statistic is useful for evaluating mechanisms like NACK and FEC and gives some insight into the behavior of the pacer sending the packets. Note that for frames with just a single packet the assembly time will be zero. In order to calculate an average assembly time an additional frames_assembled_from_multiple_packets counter for frames with more than a single packet is added. Currently this is a nonstandard stat so will only show up in webrtc-internals and not in getStats. Formally it can be defined as totalAssemblyTime of type double Only exists for video. The sum of the time, in seconds, each video frame takes from the time the first RTP packet is received (reception timestamp) and to the time the last RTP packet of a frame is received. Given the complexities involved, the time of arrival or the reception timestamp is measured as close to the network layer as possible. This metric is not incremented for frames that are not decoded, i.e., framesDropped, partialFramesLost or frames that fail decoding for other reasons (if any). Only incremented for frames consisting of more than one RTP packet. The average frame assembly time can be calculated by dividing the totalAssemblyTime with framesAssembledFromMultiplePacket. framesAssembledFromMultiplePacket of type unsigned long Only exists for video. It represents the total number of frames correctly decoded for this RTP stream that consist of more than one RTP packet. For such frames the totalAssemblyTime is incremented. BUG=webrtc:13986 Change-Id: Ie0ae431d72a57a0001c3240daba8eda35955f04e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260920 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36922}
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&total_assembly_time,
&frames_assembled_from_multiple_packets,
&total_inter_frame_delay,
&total_squared_inter_frame_delay,
&pause_count,
&total_pauses_duration,
&freeze_count,
&total_freezes_duration,
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&content_type,
&estimated_playout_timestamp,
&decoder_implementation,
&fir_count,
&pli_count,
&nack_count,
&qp_sum,
&goog_timing_frame_info,
&power_efficient_decoder,
&jitter_buffer_flushes,
&delayed_packet_outage_samples,
&relative_packet_arrival_delay,
&interruption_count,
&total_interruption_duration,
&min_playout_delay)
// clang-format on
RTCInboundRTPStreamStats::RTCInboundRTPStreamStats(std::string id,
Timestamp timestamp)
: RTCReceivedRtpStreamStats(std::move(id), timestamp),
playout_id("playoutId"),
track_identifier("trackIdentifier"),
mid("mid"),
remote_id("remoteId"),
packets_received("packetsReceived"),
packets_discarded("packetsDiscarded"),
fec_packets_received("fecPacketsReceived"),
fec_packets_discarded("fecPacketsDiscarded"),
bytes_received("bytesReceived"),
header_bytes_received("headerBytesReceived"),
last_packet_received_timestamp("lastPacketReceivedTimestamp"),
jitter_buffer_delay("jitterBufferDelay"),
jitter_buffer_target_delay("jitterBufferTargetDelay"),
jitter_buffer_minimum_delay("jitterBufferMinimumDelay"),
jitter_buffer_emitted_count("jitterBufferEmittedCount"),
total_samples_received("totalSamplesReceived"),
concealed_samples("concealedSamples"),
silent_concealed_samples("silentConcealedSamples"),
concealment_events("concealmentEvents"),
inserted_samples_for_deceleration("insertedSamplesForDeceleration"),
removed_samples_for_acceleration("removedSamplesForAcceleration"),
audio_level("audioLevel"),
total_audio_energy("totalAudioEnergy"),
total_samples_duration("totalSamplesDuration"),
frames_received("framesReceived"),
frame_width("frameWidth"),
frame_height("frameHeight"),
frames_per_second("framesPerSecond"),
frames_decoded("framesDecoded"),
key_frames_decoded("keyFramesDecoded"),
frames_dropped("framesDropped"),
total_decode_time("totalDecodeTime"),
total_processing_delay("totalProcessingDelay"),
stats: add frame assembly time stats implements a total frame assembly time statistic that measures the cumulative time between the arrival of the first packet of a frame (the lowest reception time) and the time all packets of the frame have been received (i.e. the highest reception time) This is similar to totalProcessingDelay https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totalprocessingdelay in particular with respect to only being incremented for frames that are being decoded but does not include the amount of time spent decoding the frame. This statistic is useful for evaluating mechanisms like NACK and FEC and gives some insight into the behavior of the pacer sending the packets. Note that for frames with just a single packet the assembly time will be zero. In order to calculate an average assembly time an additional frames_assembled_from_multiple_packets counter for frames with more than a single packet is added. Currently this is a nonstandard stat so will only show up in webrtc-internals and not in getStats. Formally it can be defined as totalAssemblyTime of type double Only exists for video. The sum of the time, in seconds, each video frame takes from the time the first RTP packet is received (reception timestamp) and to the time the last RTP packet of a frame is received. Given the complexities involved, the time of arrival or the reception timestamp is measured as close to the network layer as possible. This metric is not incremented for frames that are not decoded, i.e., framesDropped, partialFramesLost or frames that fail decoding for other reasons (if any). Only incremented for frames consisting of more than one RTP packet. The average frame assembly time can be calculated by dividing the totalAssemblyTime with framesAssembledFromMultiplePacket. framesAssembledFromMultiplePacket of type unsigned long Only exists for video. It represents the total number of frames correctly decoded for this RTP stream that consist of more than one RTP packet. For such frames the totalAssemblyTime is incremented. BUG=webrtc:13986 Change-Id: Ie0ae431d72a57a0001c3240daba8eda35955f04e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260920 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36922}
2022-05-05 15:55:36 +02:00
total_assembly_time("totalAssemblyTime"),
frames_assembled_from_multiple_packets(
"framesAssembledFromMultiplePackets"),
total_inter_frame_delay("totalInterFrameDelay"),
total_squared_inter_frame_delay("totalSquaredInterFrameDelay"),
pause_count("pauseCount"),
total_pauses_duration("totalPausesDuration"),
freeze_count("freezeCount"),
total_freezes_duration("totalFreezesDuration"),
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content_type("contentType"),
estimated_playout_timestamp("estimatedPlayoutTimestamp"),
decoder_implementation("decoderImplementation"),
fir_count("firCount"),
pli_count("pliCount"),
nack_count("nackCount"),
qp_sum("qpSum"),
goog_timing_frame_info("googTimingFrameInfo"),
power_efficient_decoder("powerEfficientDecoder"),
jitter_buffer_flushes(
"jitterBufferFlushes",
{NonStandardGroupId::kRtcAudioJitterBufferMaxPackets}),
delayed_packet_outage_samples(
"delayedPacketOutageSamples",
{NonStandardGroupId::kRtcAudioJitterBufferMaxPackets,
NonStandardGroupId::kRtcStatsRelativePacketArrivalDelay}),
relative_packet_arrival_delay(
"relativePacketArrivalDelay",
{NonStandardGroupId::kRtcStatsRelativePacketArrivalDelay}),
interruption_count("interruptionCount"),
total_interruption_duration("totalInterruptionDuration"),
min_playout_delay("minPlayoutDelay") {}
RTCInboundRTPStreamStats::RTCInboundRTPStreamStats(
const RTCInboundRTPStreamStats& other) = default;
RTCInboundRTPStreamStats::~RTCInboundRTPStreamStats() {}
// clang-format off
WEBRTC_RTCSTATS_IMPL(
RTCOutboundRTPStreamStats, RTCRTPStreamStats, "outbound-rtp",
&media_source_id,
&remote_id,
&mid,
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_durations,
&quality_limitation_resolution_changes,
2019-08-16 13:09:51 +02:00
&content_type,
&encoder_implementation,
&fir_count,
&pli_count,
&nack_count,
&qp_sum,
&active,
&power_efficient_encoder,
&scalability_mode)
// clang-format on
RTCOutboundRTPStreamStats::RTCOutboundRTPStreamStats(std::string id,
Timestamp timestamp)
: RTCRTPStreamStats(std::move(id), timestamp),
media_source_id("mediaSourceId"),
remote_id("remoteId"),
mid("mid"),
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_durations("qualityLimitationDurations"),
quality_limitation_resolution_changes(
"qualityLimitationResolutionChanges"),
2019-08-16 13:09:51 +02:00
content_type("contentType"),
encoder_implementation("encoderImplementation"),
fir_count("firCount"),
pli_count("pliCount"),
nack_count("nackCount"),
qp_sum("qpSum"),
active("active"),
power_efficient_encoder("powerEfficientEncoder"),
scalability_mode("scalabilityMode") {}
RTCOutboundRTPStreamStats::RTCOutboundRTPStreamStats(
const RTCOutboundRTPStreamStats& other) = default;
RTCOutboundRTPStreamStats::~RTCOutboundRTPStreamStats() {}
// clang-format off
WEBRTC_RTCSTATS_IMPL(
RTCRemoteInboundRtpStreamStats, RTCReceivedRtpStreamStats,
"remote-inbound-rtp",
&local_id,
&round_trip_time,
&fraction_lost,
&total_round_trip_time,
&round_trip_time_measurements)
// clang-format on
RTCRemoteInboundRtpStreamStats::RTCRemoteInboundRtpStreamStats(
std::string id,
Timestamp timestamp)
: RTCReceivedRtpStreamStats(std::move(id), timestamp),
local_id("localId"),
round_trip_time("roundTripTime"),
fraction_lost("fractionLost"),
total_round_trip_time("totalRoundTripTime"),
round_trip_time_measurements("roundTripTimeMeasurements") {}
RTCRemoteInboundRtpStreamStats::RTCRemoteInboundRtpStreamStats(
const RTCRemoteInboundRtpStreamStats& other) = default;
RTCRemoteInboundRtpStreamStats::~RTCRemoteInboundRtpStreamStats() {}
// clang-format off
WEBRTC_RTCSTATS_IMPL(
RTCRemoteOutboundRtpStreamStats, RTCSentRtpStreamStats,
"remote-outbound-rtp",
&local_id,
&remote_timestamp,
Reland "Wire up non-sender RTT for audio, and implement related standardized stats." This reverts commit 2c41cbae37cac548a1133589b9d2c2e8614fa6cb. Reason for revert: The breaking test in Chromium has been temporarily disabled in https://chromium-review.googlesource.com/c/chromium/src/+/3139794/2. Original change's description: > Revert "Wire up non-sender RTT for audio, and implement related standardized stats." > > This reverts commit fb0dca6c055cbf9e43af665d3c437eba6f43372e. > > Reason for revert: Speculative revert due to failing stats test in chromium. Possibly because the chromium test expected the metrics to not be supported, and now they are. Reverting just to unblock the webrtc roll into chromium. > > Original change's description: > > Wire up non-sender RTT for audio, and implement related standardized stats. > > > > The implemented stats are: > > - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptime > > - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-totalroundtriptime > > - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptimemeasurements > > > > Bug: webrtc:12951, webrtc:12714 > > Change-Id: Ia362d5c4b0456140e32da79d40edc06ab9ce2a2c > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226956 > > Commit-Queue: Ivo Creusen <ivoc@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#34861} > > # Not skipping CQ checks because original CL landed > 1 day ago. > > TBR=hta,hbos,minyue > > Bug: webrtc:12951, webrtc:12714 > Change-Id: If07ad63286eea9cdde88271e61cc28f4b268b290 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231001 > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > Reviewed-by: Ivo Creusen <ivoc@webrtc.org> > Reviewed-by: Olga Sharonova <olka@webrtc.org> > Commit-Queue: Björn Terelius <terelius@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#34897} # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:12951, webrtc:12714 Change-Id: I786b06933d85bdffc5e879bf52436bb3469b7f3a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231181 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/main@{#34930}
2021-09-03 14:51:22 +00:00
&reports_sent,
&round_trip_time,
&round_trip_time_measurements,
&total_round_trip_time)
// clang-format on
RTCRemoteOutboundRtpStreamStats::RTCRemoteOutboundRtpStreamStats(
std::string id,
Timestamp timestamp)
: RTCSentRtpStreamStats(std::move(id), timestamp),
local_id("localId"),
remote_timestamp("remoteTimestamp"),
Reland "Wire up non-sender RTT for audio, and implement related standardized stats." This reverts commit 2c41cbae37cac548a1133589b9d2c2e8614fa6cb. Reason for revert: The breaking test in Chromium has been temporarily disabled in https://chromium-review.googlesource.com/c/chromium/src/+/3139794/2. Original change's description: > Revert "Wire up non-sender RTT for audio, and implement related standardized stats." > > This reverts commit fb0dca6c055cbf9e43af665d3c437eba6f43372e. > > Reason for revert: Speculative revert due to failing stats test in chromium. Possibly because the chromium test expected the metrics to not be supported, and now they are. Reverting just to unblock the webrtc roll into chromium. > > Original change's description: > > Wire up non-sender RTT for audio, and implement related standardized stats. > > > > The implemented stats are: > > - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptime > > - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-totalroundtriptime > > - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptimemeasurements > > > > Bug: webrtc:12951, webrtc:12714 > > Change-Id: Ia362d5c4b0456140e32da79d40edc06ab9ce2a2c > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226956 > > Commit-Queue: Ivo Creusen <ivoc@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#34861} > > # Not skipping CQ checks because original CL landed > 1 day ago. > > TBR=hta,hbos,minyue > > Bug: webrtc:12951, webrtc:12714 > Change-Id: If07ad63286eea9cdde88271e61cc28f4b268b290 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231001 > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > Reviewed-by: Ivo Creusen <ivoc@webrtc.org> > Reviewed-by: Olga Sharonova <olka@webrtc.org> > Commit-Queue: Björn Terelius <terelius@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#34897} # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:12951, webrtc:12714 Change-Id: I786b06933d85bdffc5e879bf52436bb3469b7f3a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231181 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/main@{#34930}
2021-09-03 14:51:22 +00:00
reports_sent("reportsSent"),
round_trip_time("roundTripTime"),
round_trip_time_measurements("roundTripTimeMeasurements"),
total_round_trip_time("totalRoundTripTime") {}
RTCRemoteOutboundRtpStreamStats::RTCRemoteOutboundRtpStreamStats(
const RTCRemoteOutboundRtpStreamStats& other) = default;
RTCRemoteOutboundRtpStreamStats::~RTCRemoteOutboundRtpStreamStats() {}
// clang-format off
WEBRTC_RTCSTATS_IMPL(RTCMediaSourceStats, RTCStats, "parent-media-source",
&track_identifier,
&kind)
// clang-format on
RTCMediaSourceStats::RTCMediaSourceStats(std::string id, Timestamp timestamp)
: RTCStats(std::move(id), timestamp),
track_identifier("trackIdentifier"),
kind("kind") {}
RTCMediaSourceStats::RTCMediaSourceStats(const RTCMediaSourceStats& other) =
default;
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,
&echo_return_loss,
&echo_return_loss_enhancement)
// clang-format on
RTCAudioSourceStats::RTCAudioSourceStats(std::string id, Timestamp timestamp)
: RTCMediaSourceStats(std::move(id), timestamp),
[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
audio_level("audioLevel"),
total_audio_energy("totalAudioEnergy"),
total_samples_duration("totalSamplesDuration"),
echo_return_loss("echoReturnLoss"),
echo_return_loss_enhancement("echoReturnLossEnhancement") {}
RTCAudioSourceStats::RTCAudioSourceStats(const RTCAudioSourceStats& other) =
default;
RTCAudioSourceStats::~RTCAudioSourceStats() {}
// clang-format off
WEBRTC_RTCSTATS_IMPL(RTCVideoSourceStats, RTCMediaSourceStats, "media-source",
&width,
&height,
&frames,
&frames_per_second)
// clang-format on
RTCVideoSourceStats::RTCVideoSourceStats(std::string id, Timestamp timestamp)
: RTCMediaSourceStats(std::move(id), timestamp),
width("width"),
height("height"),
frames("frames"),
frames_per_second("framesPerSecond") {}
RTCVideoSourceStats::RTCVideoSourceStats(const RTCVideoSourceStats& other) =
default;
RTCVideoSourceStats::~RTCVideoSourceStats() {}
// clang-format off
WEBRTC_RTCSTATS_IMPL(RTCTransportStats, RTCStats, "transport",
&bytes_sent,
&packets_sent,
&bytes_received,
&packets_received,
&rtcp_transport_stats_id,
&dtls_state,
&selected_candidate_pair_id,
&local_certificate_id,
&remote_certificate_id,
&tls_version,
&dtls_cipher,
&dtls_role,
&srtp_cipher,
&selected_candidate_pair_changes,
&ice_role,
&ice_local_username_fragment,
&ice_state)
// clang-format on
RTCTransportStats::RTCTransportStats(std::string id, Timestamp timestamp)
: RTCStats(std::move(id), timestamp),
bytes_sent("bytesSent"),
packets_sent("packetsSent"),
bytes_received("bytesReceived"),
packets_received("packetsReceived"),
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"),
dtls_role("dtlsRole"),
srtp_cipher("srtpCipher"),
selected_candidate_pair_changes("selectedCandidatePairChanges"),
ice_role("iceRole"),
ice_local_username_fragment("iceLocalUsernameFragment"),
ice_state("iceState") {}
RTCTransportStats::RTCTransportStats(const RTCTransportStats& other) = default;
RTCTransportStats::~RTCTransportStats() {}
RTCAudioPlayoutStats::RTCAudioPlayoutStats(const std::string& id,
Timestamp timestamp)
: RTCStats(std::move(id), timestamp),
kind("kind", "audio"),
synthesized_samples_duration("synthesizedSamplesDuration"),
synthesized_samples_events("synthesizedSamplesEvents"),
total_samples_duration("totalSamplesDuration"),
total_playout_delay("totalPlayoutDelay"),
total_samples_count("totalSamplesCount") {}
RTCAudioPlayoutStats::RTCAudioPlayoutStats(const RTCAudioPlayoutStats& other) =
default;
RTCAudioPlayoutStats::~RTCAudioPlayoutStats() {}
// clang-format off
WEBRTC_RTCSTATS_IMPL(RTCAudioPlayoutStats, RTCStats, "media-playout",
&kind,
&synthesized_samples_duration,
&synthesized_samples_events,
&total_samples_duration,
&total_playout_delay,
&total_samples_count)
// clang-format on
} // namespace webrtc