blob: 2d977e0499609982e0109c95bde9ad7b6d126e89 [file] [log] [blame]
/*
* Copyright (C) 2018 Apple Inc. All rights reserved.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
* 1. Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS'' AND ANY
* EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
* WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
* DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS BE LIABLE FOR ANY
* DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
* (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
* LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON
* ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
* (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
* SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "config.h"
#include "LibWebRTCStatsCollector.h"
#if USE(LIBWEBRTC)
#include "JSRTCStatsReport.h"
#include "Performance.h"
#include <wtf/MainThread.h>
namespace WebCore {
LibWebRTCStatsCollector::LibWebRTCStatsCollector(CollectorCallback&& callback)
: m_callback(WTFMove(callback))
{
}
LibWebRTCStatsCollector::~LibWebRTCStatsCollector()
{
if (!m_callback)
return;
callOnMainThread([callback = WTFMove(m_callback)]() mutable {
callback();
});
}
static inline String fromStdString(const std::string& value)
{
return String::fromUTF8(value.data(), value.length());
}
static inline void fillRTCStats(RTCStatsReport::Stats& stats, const webrtc::RTCStats& rtcStats)
{
stats.timestamp = Performance::reduceTimeResolution(Seconds::fromMicroseconds(rtcStats.timestamp_us())).milliseconds();
stats.id = fromStdString(rtcStats.id());
}
static inline void fillRTCRTPStreamStats(RTCStatsReport::RTCRTPStreamStats& stats, const webrtc::RTCRTPStreamStats& rtcStats)
{
fillRTCStats(stats, rtcStats);
if (rtcStats.ssrc.is_defined())
stats.ssrc = *rtcStats.ssrc;
if (rtcStats.associate_stats_id.is_defined())
stats.associateStatsId = fromStdString(*rtcStats.associate_stats_id);
if (rtcStats.is_remote.is_defined())
stats.isRemote = *rtcStats.is_remote;
if (rtcStats.media_type.is_defined())
stats.mediaType = fromStdString(*rtcStats.media_type);
if (rtcStats.track_id.is_defined())
stats.trackId = fromStdString(*rtcStats.track_id);
if (rtcStats.transport_id.is_defined())
stats.transportId = fromStdString(*rtcStats.transport_id);
if (rtcStats.codec_id.is_defined())
stats.codecId = fromStdString(*rtcStats.codec_id);
if (rtcStats.fir_count.is_defined())
stats.firCount = *rtcStats.fir_count;
if (rtcStats.pli_count.is_defined())
stats.pliCount = *rtcStats.pli_count;
if (rtcStats.nack_count.is_defined())
stats.nackCount = *rtcStats.nack_count;
if (rtcStats.sli_count.is_defined())
stats.sliCount = *rtcStats.sli_count;
if (rtcStats.qp_sum.is_defined())
stats.qpSum = *rtcStats.qp_sum;
stats.qpSum = 0;
}
static inline void fillInboundRTPStreamStats(RTCStatsReport::InboundRTPStreamStats& stats, const webrtc::RTCInboundRTPStreamStats& rtcStats)
{
fillRTCRTPStreamStats(stats, rtcStats);
if (rtcStats.packets_received.is_defined())
stats.packetsReceived = *rtcStats.packets_received;
if (rtcStats.bytes_received.is_defined())
stats.bytesReceived = *rtcStats.bytes_received;
if (rtcStats.packets_lost.is_defined())
stats.packetsLost = *rtcStats.packets_lost;
if (rtcStats.jitter.is_defined())
stats.jitter = *rtcStats.jitter;
if (rtcStats.fraction_lost.is_defined())
stats.fractionLost = *rtcStats.fraction_lost;
if (rtcStats.packets_discarded.is_defined())
stats.packetsDiscarded = *rtcStats.packets_discarded;
if (rtcStats.packets_repaired.is_defined())
stats.packetsRepaired = *rtcStats.packets_repaired;
if (rtcStats.burst_packets_lost.is_defined())
stats.burstPacketsLost = *rtcStats.burst_packets_lost;
if (rtcStats.burst_packets_discarded.is_defined())
stats.burstPacketsDiscarded = *rtcStats.burst_packets_discarded;
if (rtcStats.burst_loss_count.is_defined())
stats.burstLossCount = *rtcStats.burst_loss_count;
if (rtcStats.burst_discard_count.is_defined())
stats.burstDiscardCount = *rtcStats.burst_discard_count;
if (rtcStats.burst_loss_rate.is_defined())
stats.burstLossRate = *rtcStats.burst_loss_rate;
if (rtcStats.burst_discard_rate.is_defined())
stats.burstDiscardRate = *rtcStats.burst_discard_rate;
if (rtcStats.gap_loss_rate.is_defined())
stats.gapLossRate = *rtcStats.gap_loss_rate;
if (rtcStats.gap_discard_rate.is_defined())
stats.gapDiscardRate = *rtcStats.gap_discard_rate;
if (rtcStats.frames_decoded.is_defined())
stats.framesDecoded = *rtcStats.frames_decoded;
}
static inline void fillOutboundRTPStreamStats(RTCStatsReport::OutboundRTPStreamStats& stats, const webrtc::RTCOutboundRTPStreamStats& rtcStats)
{
fillRTCRTPStreamStats(stats, rtcStats);
if (rtcStats.packets_sent.is_defined())
stats.packetsSent = *rtcStats.packets_sent;
if (rtcStats.bytes_sent.is_defined())
stats.bytesSent = *rtcStats.bytes_sent;
if (rtcStats.target_bitrate.is_defined())
stats.targetBitrate = *rtcStats.target_bitrate;
if (rtcStats.frames_encoded.is_defined())
stats.framesEncoded = *rtcStats.frames_encoded;
}
static inline void fillRTCMediaStreamTrackStats(RTCStatsReport::MediaStreamTrackStats& stats, const webrtc::RTCMediaStreamTrackStats& rtcStats)
{
fillRTCStats(stats, rtcStats);
if (rtcStats.track_identifier.is_defined())
stats.trackIdentifier = fromStdString(*rtcStats.track_identifier);
if (rtcStats.remote_source.is_defined())
stats.remoteSource = *rtcStats.remote_source;
if (rtcStats.ended.is_defined())
stats.ended = *rtcStats.ended;
if (rtcStats.detached.is_defined())
stats.detached = *rtcStats.detached;
if (rtcStats.frame_width.is_defined())
stats.frameWidth = *rtcStats.frame_width;
if (rtcStats.frame_height.is_defined())
stats.frameHeight = *rtcStats.frame_height;
if (rtcStats.frames_per_second.is_defined())
stats.framesPerSecond = *rtcStats.frames_per_second;
if (rtcStats.frames_sent.is_defined())
stats.framesSent = *rtcStats.frames_sent;
if (rtcStats.frames_received.is_defined())
stats.framesReceived = *rtcStats.frames_received;
if (rtcStats.frames_decoded.is_defined())
stats.framesDecoded = *rtcStats.frames_decoded;
if (rtcStats.frames_dropped.is_defined())
stats.framesDropped = *rtcStats.frames_dropped;
if (rtcStats.partial_frames_lost.is_defined())
stats.partialFramesLost = *rtcStats.partial_frames_lost;
if (rtcStats.full_frames_lost.is_defined())
stats.fullFramesLost = *rtcStats.full_frames_lost;
if (rtcStats.audio_level.is_defined())
stats.audioLevel = *rtcStats.audio_level;
if (rtcStats.echo_return_loss.is_defined())
stats.echoReturnLoss = *rtcStats.echo_return_loss;
if (rtcStats.echo_return_loss_enhancement.is_defined())
stats.echoReturnLossEnhancement = *rtcStats.echo_return_loss_enhancement;
}
static inline void fillRTCDataChannelStats(RTCStatsReport::DataChannelStats& stats, const webrtc::RTCDataChannelStats& rtcStats)
{
fillRTCStats(stats, rtcStats);
if (rtcStats.label.is_defined())
stats.label = fromStdString(*rtcStats.label);
if (rtcStats.protocol.is_defined())
stats.protocol = fromStdString(*rtcStats.protocol);
if (rtcStats.datachannelid.is_defined())
stats.datachannelid = *rtcStats.datachannelid;
if (rtcStats.state.is_defined())
stats.state = fromStdString(*rtcStats.state);
if (rtcStats.messages_sent.is_defined())
stats.messagesSent = *rtcStats.messages_sent;
if (rtcStats.bytes_sent.is_defined())
stats.bytesSent = *rtcStats.bytes_sent;
if (rtcStats.messages_received.is_defined())
stats.messagesReceived = *rtcStats.messages_received;
if (rtcStats.bytes_received.is_defined())
stats.bytesReceived = *rtcStats.bytes_received;
}
static inline RTCStatsReport::IceCandidatePairState iceCandidatePairState(const std::string& state)
{
if (state == "frozen")
return RTCStatsReport::IceCandidatePairState::Frozen;
if (state == "waiting")
return RTCStatsReport::IceCandidatePairState::Waiting;
if (state == "in-progress")
return RTCStatsReport::IceCandidatePairState::Inprogress;
if (state == "failed")
return RTCStatsReport::IceCandidatePairState::Failed;
if (state == "succeeded")
return RTCStatsReport::IceCandidatePairState::Succeeded;
if (state == "cancelled")
return RTCStatsReport::IceCandidatePairState::Cancelled;
ASSERT_NOT_REACHED();
return RTCStatsReport::IceCandidatePairState::Frozen;
}
static inline void fillRTCIceCandidatePairStats(RTCStatsReport::IceCandidatePairStats& stats, const webrtc::RTCIceCandidatePairStats& rtcStats)
{
fillRTCStats(stats, rtcStats);
if (rtcStats.transport_id.is_defined())
stats.transportId = fromStdString(*rtcStats.transport_id);
if (rtcStats.local_candidate_id.is_defined())
stats.localCandidateId = fromStdString(*rtcStats.local_candidate_id);
if (rtcStats.remote_candidate_id.is_defined())
stats.remoteCandidateId = fromStdString(*rtcStats.remote_candidate_id);
if (rtcStats.state.is_defined())
stats.state = iceCandidatePairState(*rtcStats.state);
if (rtcStats.priority.is_defined())
stats.priority = *rtcStats.priority;
if (rtcStats.nominated.is_defined())
stats.nominated = *rtcStats.nominated;
if (rtcStats.writable.is_defined())
stats.writable = *rtcStats.writable;
if (rtcStats.readable.is_defined())
stats.readable = *rtcStats.readable;
if (rtcStats.bytes_sent.is_defined())
stats.bytesSent = *rtcStats.bytes_sent;
if (rtcStats.bytes_received.is_defined())
stats.bytesReceived = *rtcStats.bytes_received;
if (rtcStats.total_round_trip_time.is_defined())
stats.totalRoundTripTime = *rtcStats.total_round_trip_time;
if (rtcStats.current_round_trip_time.is_defined())
stats.currentRoundTripTime = *rtcStats.current_round_trip_time;
if (rtcStats.available_outgoing_bitrate.is_defined())
stats.availableOutgoingBitrate = *rtcStats.available_outgoing_bitrate;
if (rtcStats.available_incoming_bitrate.is_defined())
stats.availableIncomingBitrate = *rtcStats.available_incoming_bitrate;
if (rtcStats.requests_received.is_defined())
stats.requestsReceived = *rtcStats.requests_received;
if (rtcStats.requests_sent.is_defined())
stats.requestsSent = *rtcStats.requests_sent;
if (rtcStats.responses_received.is_defined())
stats.responsesReceived = *rtcStats.responses_received;
if (rtcStats.responses_sent.is_defined())
stats.responsesSent = *rtcStats.responses_sent;
if (rtcStats.requests_received.is_defined())
stats.retransmissionsReceived = *rtcStats.requests_received;
if (rtcStats.requests_sent.is_defined())
stats.retransmissionsSent = *rtcStats.requests_sent;
if (rtcStats.responses_received.is_defined())
stats.consentRequestsReceived = *rtcStats.responses_received;
if (rtcStats.responses_sent.is_defined())
stats.consentRequestsSent = *rtcStats.responses_sent;
if (rtcStats.responses_received.is_defined())
stats.consentResponsesReceived = *rtcStats.responses_received;
if (rtcStats.responses_sent.is_defined())
stats.consentResponsesSent = *rtcStats.responses_sent;
}
static inline Optional<RTCStatsReport::IceCandidateType> iceCandidateState(const std::string& state)
{
if (state == "host")
return RTCStatsReport::IceCandidateType::Host;
if (state == "srflx")
return RTCStatsReport::IceCandidateType::Srflx;
if (state == "prflx")
return RTCStatsReport::IceCandidateType::Prflx;
if (state == "relay")
return RTCStatsReport::IceCandidateType::Relay;
return { };
}
static inline void fillRTCIceCandidateStats(RTCStatsReport::IceCandidateStats& stats, const webrtc::RTCIceCandidateStats& rtcStats)
{
stats.type = rtcStats.type() == webrtc::RTCRemoteIceCandidateStats::kType ? RTCStatsReport::Type::RemoteCandidate : RTCStatsReport::Type::LocalCandidate;
fillRTCStats(stats, rtcStats);
if (rtcStats.transport_id.is_defined())
stats.transportId = fromStdString(*rtcStats.transport_id);
if (rtcStats.ip.is_defined())
stats.address = fromStdString(*rtcStats.ip);
if (rtcStats.port.is_defined())
stats.port = *rtcStats.port;
if (rtcStats.protocol.is_defined())
stats.protocol = fromStdString(*rtcStats.protocol);
if (rtcStats.candidate_type.is_defined())
stats.candidateType = iceCandidateState(*rtcStats.candidate_type);
if (!stats.candidateType || stats.candidateType == RTCStatsReport::IceCandidateType::Prflx || stats.candidateType == RTCStatsReport::IceCandidateType::Host)
stats.address = { };
if (rtcStats.priority.is_defined())
stats.priority = *rtcStats.priority;
if (rtcStats.url.is_defined())
stats.url = fromStdString(*rtcStats.url);
if (rtcStats.deleted.is_defined())
stats.deleted = *rtcStats.deleted;
}
static inline void fillRTCCertificateStats(RTCStatsReport::CertificateStats& stats, const webrtc::RTCCertificateStats& rtcStats)
{
fillRTCStats(stats, rtcStats);
if (rtcStats.fingerprint.is_defined())
stats.fingerprint = fromStdString(*rtcStats.fingerprint);
if (rtcStats.fingerprint_algorithm.is_defined())
stats.fingerprintAlgorithm = fromStdString(*rtcStats.fingerprint_algorithm);
if (rtcStats.base64_certificate.is_defined())
stats.base64Certificate = fromStdString(*rtcStats.base64_certificate);
if (rtcStats.issuer_certificate_id.is_defined())
stats.issuerCertificateId = fromStdString(*rtcStats.issuer_certificate_id);
}
static inline void fillRTCCodecStats(RTCStatsReport::CodecStats& stats, const webrtc::RTCCodecStats& rtcStats)
{
fillRTCStats(stats, rtcStats);
if (rtcStats.payload_type.is_defined())
stats.payloadType = *rtcStats.payload_type;
if (rtcStats.mime_type.is_defined())
stats.mimeType = fromStdString(*rtcStats.mime_type);
if (rtcStats.clock_rate.is_defined())
stats.clockRate = *rtcStats.clock_rate;
if (rtcStats.channels.is_defined())
stats.channels = *rtcStats.channels;
if (rtcStats.sdp_fmtp_line.is_defined())
stats.sdpFmtpLine = fromStdString(*rtcStats.sdp_fmtp_line);
if (rtcStats.implementation.is_defined())
stats.implementation = fromStdString(*rtcStats.implementation);
}
static inline void fillRTCTransportStats(RTCStatsReport::TransportStats& stats, const webrtc::RTCTransportStats& rtcStats)
{
fillRTCStats(stats, rtcStats);
if (rtcStats.bytes_sent.is_defined())
stats.bytesSent = *rtcStats.bytes_sent;
if (rtcStats.bytes_received.is_defined())
stats.bytesReceived = *rtcStats.bytes_received;
if (rtcStats.rtcp_transport_stats_id.is_defined())
stats.rtcpTransportStatsId = fromStdString(*rtcStats.rtcp_transport_stats_id);
if (rtcStats.selected_candidate_pair_id.is_defined())
stats.selectedCandidatePairId = fromStdString(*rtcStats.selected_candidate_pair_id);
if (rtcStats.local_certificate_id.is_defined())
stats.localCertificateId = fromStdString(*rtcStats.local_certificate_id);
if (rtcStats.remote_certificate_id.is_defined())
stats.remoteCertificateId = fromStdString(*rtcStats.remote_certificate_id);
}
static inline void fillRTCPeerConnectionStats(RTCStatsReport::PeerConnectionStats& stats, const webrtc::RTCPeerConnectionStats& rtcStats)
{
fillRTCStats(stats, rtcStats);
if (rtcStats.data_channels_opened.is_defined())
stats.dataChannelsOpened = *rtcStats.data_channels_opened;
if (rtcStats.data_channels_closed.is_defined())
stats.dataChannelsClosed = *rtcStats.data_channels_closed;
}
void LibWebRTCStatsCollector::OnStatsDelivered(const rtc::scoped_refptr<const webrtc::RTCStatsReport>& rtcReport)
{
callOnMainThread([protectedThis = rtc::scoped_refptr<LibWebRTCStatsCollector>(this), rtcReport] {
auto report = protectedThis->m_callback();
if (!report)
return;
ASSERT(report->backingMap());
for (const auto& rtcStats : *rtcReport) {
if (rtcStats.type() == webrtc::RTCInboundRTPStreamStats::kType) {
RTCStatsReport::InboundRTPStreamStats stats;
fillInboundRTPStreamStats(stats, static_cast<const webrtc::RTCInboundRTPStreamStats&>(rtcStats));
report->addStats<IDLDictionary<RTCStatsReport::InboundRTPStreamStats>>(WTFMove(stats));
} else if (rtcStats.type() == webrtc::RTCOutboundRTPStreamStats::kType) {
RTCStatsReport::OutboundRTPStreamStats stats;
fillOutboundRTPStreamStats(stats, static_cast<const webrtc::RTCOutboundRTPStreamStats&>(rtcStats));
report->addStats<IDLDictionary<RTCStatsReport::OutboundRTPStreamStats>>(WTFMove(stats));
} else if (rtcStats.type() == webrtc::RTCMediaStreamTrackStats::kType) {
RTCStatsReport::MediaStreamTrackStats stats;
fillRTCMediaStreamTrackStats(stats, static_cast<const webrtc::RTCMediaStreamTrackStats&>(rtcStats));
report->addStats<IDLDictionary<RTCStatsReport::MediaStreamTrackStats>>(WTFMove(stats));
} else if (rtcStats.type() == webrtc::RTCDataChannelStats::kType) {
RTCStatsReport::DataChannelStats stats;
fillRTCDataChannelStats(stats, static_cast<const webrtc::RTCDataChannelStats&>(rtcStats));
report->addStats<IDLDictionary<RTCStatsReport::DataChannelStats>>(WTFMove(stats));
} else if (rtcStats.type() == webrtc::RTCIceCandidatePairStats::kType) {
RTCStatsReport::IceCandidatePairStats stats;
fillRTCIceCandidatePairStats(stats, static_cast<const webrtc::RTCIceCandidatePairStats&>(rtcStats));
report->addStats<IDLDictionary<RTCStatsReport::IceCandidatePairStats>>(WTFMove(stats));
} else if (rtcStats.type() == webrtc::RTCRemoteIceCandidateStats::kType || rtcStats.type() == webrtc::RTCLocalIceCandidateStats::kType) {
RTCStatsReport::IceCandidateStats stats;
fillRTCIceCandidateStats(stats, static_cast<const webrtc::RTCIceCandidateStats&>(rtcStats));
report->addStats<IDLDictionary<RTCStatsReport::IceCandidateStats>>(WTFMove(stats));
} else if (rtcStats.type() == webrtc::RTCCertificateStats::kType) {
RTCStatsReport::CertificateStats stats;
fillRTCCertificateStats(stats, static_cast<const webrtc::RTCCertificateStats&>(rtcStats));
report->addStats<IDLDictionary<RTCStatsReport::CertificateStats>>(WTFMove(stats));
} else if (rtcStats.type() == webrtc::RTCCodecStats::kType) {
RTCStatsReport::CodecStats stats;
fillRTCCodecStats(stats, static_cast<const webrtc::RTCCodecStats&>(rtcStats));
report->addStats<IDLDictionary<RTCStatsReport::CodecStats>>(WTFMove(stats));
} else if (rtcStats.type() == webrtc::RTCTransportStats::kType) {
RTCStatsReport::TransportStats stats;
fillRTCTransportStats(stats, static_cast<const webrtc::RTCTransportStats&>(rtcStats));
report->addStats<IDLDictionary<RTCStatsReport::TransportStats>>(WTFMove(stats));
} else if (rtcStats.type() == webrtc::RTCPeerConnectionStats::kType) {
RTCStatsReport::PeerConnectionStats stats;
fillRTCPeerConnectionStats(stats, static_cast<const webrtc::RTCPeerConnectionStats&>(rtcStats));
report->addStats<IDLDictionary<RTCStatsReport::PeerConnectionStats>>(WTFMove(stats));
}
}
});
}
}; // namespace WTF
#endif // USE(LIBWEBRTC)