зеркало из https://github.com/mozilla/gecko-dev.git
Backed out changeset 99c66f2c4097 (bug 1790508) for causing wpt failures. CLOSED TREE
This commit is contained in:
Butkovits Atila
Родитель
f81a096eb8
Коммит
8ef8248f4a
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@ -301,33 +301,16 @@ struct ParamTraits<mozilla::dom::RTCInboundRtpStreamStats> {
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WriteParam(aWriter, aParam.mFrameWidth);
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WriteParam(aWriter, aParam.mFrameHeight);
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WriteParam(aWriter, aParam.mFramesPerSecond);
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WriteParam(aWriter, aParam.mQpSum);
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WriteParam(aWriter, aParam.mTotalDecodeTime);
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WriteParam(aWriter, aParam.mTotalInterFrameDelay);
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WriteParam(aWriter, aParam.mTotalSquaredInterFrameDelay);
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WriteParam(aWriter, aParam.mLastPacketReceivedTimestamp);
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WriteParam(aWriter, aParam.mHeaderBytesReceived);
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WriteParam(aWriter, aParam.mFecPacketsReceived);
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WriteParam(aWriter, aParam.mFecPacketsDiscarded);
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WriteParam(aWriter, aParam.mBytesReceived);
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WriteParam(aWriter, aParam.mNackCount);
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WriteParam(aWriter, aParam.mFirCount);
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WriteParam(aWriter, aParam.mPliCount);
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WriteParam(aWriter, aParam.mTotalProcessingDelay);
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// Always missing from libwebrtc stats
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// WriteParam(aWriter, aParam.mEstimatedPlayoutTimestamp);
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WriteParam(aWriter, aParam.mFramesReceived);
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WriteParam(aWriter, aParam.mJitterBufferDelay);
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WriteParam(aWriter, aParam.mJitterBufferEmittedCount);
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WriteParam(aWriter, aParam.mTotalSamplesReceived);
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WriteParam(aWriter, aParam.mConcealedSamples);
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WriteParam(aWriter, aParam.mSilentConcealedSamples);
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WriteParam(aWriter, aParam.mConcealmentEvents);
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WriteParam(aWriter, aParam.mInsertedSamplesForDeceleration);
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WriteParam(aWriter, aParam.mRemovedSamplesForAcceleration);
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WriteParam(aWriter, aParam.mAudioLevel);
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WriteParam(aWriter, aParam.mTotalAudioEnergy);
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WriteParam(aWriter, aParam.mTotalSamplesDuration);
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WriteRTCReceivedRtpStreamStats(aWriter, aParam);
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}
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@ -337,33 +320,16 @@ struct ParamTraits<mozilla::dom::RTCInboundRtpStreamStats> {
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ReadParam(aReader, &(aResult->mFrameWidth)) &&
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ReadParam(aReader, &(aResult->mFrameHeight)) &&
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ReadParam(aReader, &(aResult->mFramesPerSecond)) &&
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ReadParam(aReader, &(aResult->mQpSum)) &&
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ReadParam(aReader, &(aResult->mTotalDecodeTime)) &&
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ReadParam(aReader, &(aResult->mTotalInterFrameDelay)) &&
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ReadParam(aReader, &(aResult->mTotalSquaredInterFrameDelay)) &&
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ReadParam(aReader, &(aResult->mLastPacketReceivedTimestamp)) &&
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ReadParam(aReader, &(aResult->mHeaderBytesReceived)) &&
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ReadParam(aReader, &(aResult->mFecPacketsReceived)) &&
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ReadParam(aReader, &(aResult->mFecPacketsDiscarded)) &&
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ReadParam(aReader, &(aResult->mBytesReceived)) &&
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ReadParam(aReader, &(aResult->mNackCount)) &&
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ReadParam(aReader, &(aResult->mFirCount)) &&
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ReadParam(aReader, &(aResult->mPliCount)) &&
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ReadParam(aReader, &(aResult->mTotalProcessingDelay)) &&
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// Always missing from libwebrtc
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// ReadParam(aReader, &(aResult->mEstimatedPlayoutTimestamp)) &&
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ReadParam(aReader, &(aResult->mFramesReceived)) &&
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ReadParam(aReader, &(aResult->mJitterBufferDelay)) &&
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ReadParam(aReader, &(aResult->mJitterBufferEmittedCount)) &&
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ReadParam(aReader, &(aResult->mTotalSamplesReceived)) &&
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ReadParam(aReader, &(aResult->mConcealedSamples)) &&
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ReadParam(aReader, &(aResult->mSilentConcealedSamples)) &&
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ReadParam(aReader, &(aResult->mConcealmentEvents)) &&
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ReadParam(aReader, &(aResult->mInsertedSamplesForDeceleration)) &&
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ReadParam(aReader, &(aResult->mRemovedSamplesForAcceleration)) &&
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ReadParam(aReader, &(aResult->mAudioLevel)) &&
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ReadParam(aReader, &(aResult->mTotalAudioEnergy)) &&
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ReadParam(aReader, &(aResult->mTotalSamplesDuration)) &&
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ReadRTCReceivedRtpStreamStats(aReader, aResult);
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}
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};
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@ -432,18 +398,12 @@ struct ParamTraits<mozilla::dom::RTCRemoteInboundRtpStreamStats> {
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static void Write(MessageWriter* aWriter, const paramType& aParam) {
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WriteParam(aWriter, aParam.mLocalId);
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WriteParam(aWriter, aParam.mRoundTripTime);
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WriteParam(aWriter, aParam.mTotalRoundTripTime);
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WriteParam(aWriter, aParam.mFractionLost);
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WriteParam(aWriter, aParam.mRoundTripTimeMeasurements);
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WriteRTCReceivedRtpStreamStats(aWriter, aParam);
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}
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static bool Read(MessageReader* aReader, paramType* aResult) {
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return ReadParam(aReader, &(aResult->mLocalId)) &&
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ReadParam(aReader, &(aResult->mRoundTripTime)) &&
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ReadParam(aReader, &(aResult->mTotalRoundTripTime)) &&
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ReadParam(aReader, &(aResult->mFractionLost)) &&
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ReadParam(aReader, &(aResult->mRoundTripTimeMeasurements)) &&
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ReadRTCReceivedRtpStreamStats(aReader, aResult);
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}
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};
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@ -324,13 +324,6 @@ nsTArray<RefPtr<RTCStatsPromise>> RTCRtpReceiver::GetStatsInternal() {
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local.mPacketsReceived.Construct(audioStats->packets_rcvd);
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local.mPacketsDiscarded.Construct(audioStats->packets_discarded);
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local.mBytesReceived.Construct(audioStats->payload_bytes_rcvd);
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// Always missing from libwebrtc stats
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// if (audioStats->estimated_playout_ntp_timestamp_ms) {
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// local.mEstimatedPlayoutTimestamp.Construct(
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// aConduit->GetTimestampMaker().ConvertNtpToDomTime(
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// webrtc::Timestamp::Millis(
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// *audioStats->estimated_playout_ntp_timestamp_ms)));
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// }
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local.mJitterBufferDelay.Construct(
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audioStats->jitter_buffer_delay_seconds);
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local.mJitterBufferEmittedCount.Construct(
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@ -340,6 +333,8 @@ nsTArray<RefPtr<RTCStatsPromise>> RTCRtpReceiver::GetStatsInternal() {
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local.mConcealedSamples.Construct(audioStats->concealed_samples);
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local.mSilentConcealedSamples.Construct(
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audioStats->silent_concealed_samples);
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/*
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* Potential new stats that are now available upstream.
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if (audioStats->last_packet_received_timestamp_ms) {
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local.mLastPacketReceivedTimestamp.Construct(
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aConduit->GetTimestampMaker().ConvertNtpToDomTime(
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@ -353,9 +348,14 @@ nsTArray<RefPtr<RTCStatsPromise>> RTCRtpReceiver::GetStatsInternal() {
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audioStats->fec_packets_received);
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local.mFecPacketsDiscarded.Construct(
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audioStats->fec_packets_discarded);
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if (audioStats->estimated_playout_ntp_timestamp_ms) {
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local.mEstimatedPlayoutTimestamp.Construct(
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aConduit->GetTimestampMaker().ConvertNtpToDomTime(
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webrtc::Timestamp::Millis(
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*audioStats->estimated_playout_ntp_timestamp_ms)));
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}
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local.mConcealmentEvents.Construct(
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audioStats->concealment_events);
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local.mInsertedSamplesForDeceleration.Construct(
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audioStats->inserted_samples_for_deceleration);
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local.mRemovedSamplesForAcceleration.Construct(
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@ -368,7 +368,7 @@ nsTArray<RefPtr<RTCStatsPromise>> RTCRtpReceiver::GetStatsInternal() {
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audioStats->total_output_energy);
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local.mTotalSamplesDuration.Construct(
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audioStats->total_output_duration);
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*/
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if (!report->mInboundRtpStreamStats.AppendElement(
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std::move(local), fallible)) {
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mozalloc_handle_oom(0);
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@ -449,9 +449,10 @@ nsTArray<RefPtr<RTCStatsPromise>> RTCRtpReceiver::GetStatsInternal() {
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videoStats->jitter_buffer_delay_seconds);
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local.mJitterBufferEmittedCount.Construct(
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videoStats->jitter_buffer_emitted_count);
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/*
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* Potential new stats that are now available upstream.
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if (videoStats->qp_sum) {
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local.mQpSum.Construct(videoStats->qp_sum.value());
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local.mQpSum.Construct(*videoStats->qp_sum.value);
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}
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local.mTotalDecodeTime.Construct(
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double(videoStats->total_decode_time_ms) / 1000);
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@ -460,7 +461,7 @@ nsTArray<RefPtr<RTCStatsPromise>> RTCRtpReceiver::GetStatsInternal() {
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local.mTotalSquaredInterFrameDelay.Construct(
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videoStats->total_squared_inter_frame_delay);
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if (videoStats->rtp_stats.last_packet_received_timestamp_ms) {
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local.mLastPacketReceivedTimestamp.Construct(
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local.mLastPacketReceiveTimestamp.Construct(
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aConduit->GetTimestampMaker().ConvertNtpToDomTime(
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webrtc::Timestamp::Millis(
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*videoStats->rtp_stats
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@ -470,10 +471,6 @@ nsTArray<RefPtr<RTCStatsPromise>> RTCRtpReceiver::GetStatsInternal() {
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local.mHeaderBytesReceived.Construct(
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videoStats->rtp_stats.packet_counter.header_bytes +
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videoStats->rtp_stats.packet_counter.padding_bytes);
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local.mTotalProcessingDelay.Construct(
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videoStats->total_processing_delay.seconds<double>());
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/*
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* Potential new stats that are now available upstream.
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if (videoStats->estimated_playout_ntp_timestamp_ms) {
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local.mEstimatedPlayoutTimestamp.Construct(
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aConduit->GetTimestampMaker().ConvertNtpToDomTime(
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@ -257,11 +257,14 @@ nsTArray<RefPtr<dom::RTCStatsPromise>> RTCRtpSender::GetStatsInternal() {
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remote.mRoundTripTime.Construct(
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static_cast<double>(audioStats->rtt_ms) / 1000.0);
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}
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/*
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* Potential new stats that are now available upstream.
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remote.mFractionLost.Construct(audioStats->fraction_lost);
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remote.mTotalRoundTripTime.Construct(
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double(aReportBlockData.sum_rtt_ms()) / 1000);
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remote.mRoundTripTimeMeasurements.Construct(
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aReportBlockData.num_rtts());
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*/
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if (!report->mRemoteInboundRtpStreamStats.AppendElement(
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std::move(remote), fallible)) {
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mozalloc_handle_oom(0);
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@ -333,14 +336,16 @@ nsTArray<RefPtr<dom::RTCStatsPromise>> RTCRtpSender::GetStatsInternal() {
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static_cast<double>(rtcpReportData.last_rtt_ms()) / 1000.0);
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}
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constructCommonRemoteInboundRtpStats(remote, rtcpReportData);
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/*
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* Potential new stats that are now available upstream.
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remote.mTotalRoundTripTime.Construct(
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streamStats->report_block_data->sum_rtt_ms() / 1000.0);
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remote.mFractionLost.Construct(
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static_cast<float>(
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rtcpReportData.report_block().fraction_lost) /
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static_cast<float>(streamStats->rtcp_stats.fraction_lost) /
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(1 << 8));
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remote.mRoundTripTimeMeasurements.Construct(
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streamStats->report_block_data->num_rtts());
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streamStats->report_block_data.num_rtts());
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*/
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if (!report->mRemoteInboundRtpStreamStats.AppendElement(
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std::move(remote), fallible)) {
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mozalloc_handle_oom(0);
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@ -19,14 +19,10 @@ const statsExpectedByType = {
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"packetsDiscarded",
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"bytesReceived",
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"jitter",
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"lastPacketReceivedTimestamp",
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"headerBytesReceived",
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// Always missing from libwebrtc stats
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// "estimatedPlayoutTimestamp",
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"jitterBufferDelay",
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"jitterBufferEmittedCount",
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],
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optional: ["remoteId", "nackCount", "qpSum"],
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optional: ["remoteId", "nackCount"],
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localVideoOnly: [
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"firCount",
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"pliCount",
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@ -36,24 +32,11 @@ const statsExpectedByType = {
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"frameWidth",
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"frameHeight",
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"framesReceived",
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"totalDecodeTime",
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"totalInterFrameDelay",
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"totalProcessingDelay",
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"totalSquaredInterFrameDelay",
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],
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localAudioOnly: [
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"totalSamplesReceived",
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// libwebrtc doesn't seem to do FEC for video
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"fecPacketsReceived",
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"fecPacketsDiscarded",
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"concealedSamples",
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"silentConcealedSamples",
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"concealmentEvents",
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"insertedSamplesForDeceleration",
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"removedSamplesForAcceleration",
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"audioLevel",
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"totalAudioEnergy",
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"totalSamplesDuration",
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],
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unimplemented: [
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"mediaTrackId",
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@ -67,6 +50,8 @@ const statsExpectedByType = {
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"burstDiscardCount",
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"gapDiscardRate",
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"gapLossRate",
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// Not yet implemented for inbound media, see bug 1519590
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"qpSum",
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],
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deprecated: ["mozRtt", "isRemote"],
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},
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@ -114,9 +99,6 @@ const statsExpectedByType = {
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"packetsLost",
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"jitter",
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"localId",
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"totalRoundTripTime",
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"fractionLost",
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"roundTripTimeMeasurements",
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],
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optional: ["roundTripTime", "nackCount", "packetsReceived"],
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unimplemented: [
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@ -126,6 +108,7 @@ const statsExpectedByType = {
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"associateStatsId",
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"sliCount",
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"packetsRepaired",
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"fractionLost",
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"burstPacketsLost",
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"burstLossCount",
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"burstDiscardCount",
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@ -470,21 +453,6 @@ function pedanticChecks(report) {
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`${stat.type}.pliCount is a sane number for a short ` +
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`${stat.kind} test. value=${stat.pliCount}`
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);
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// qpSum
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if (stat.qpSum !== undefined) {
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ok(
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stat.qpSum > 0,
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`${stat.type}.qpSum is at least 0 ` +
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`${stat.kind} test. value=${stat.qpSum}`
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);
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}
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} else {
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is(
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stat.qpSum,
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undefined,
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`${stat.type}.qpSum does not exist when stat.kind != video`
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);
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}
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}
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@ -506,6 +474,7 @@ function pedanticChecks(report) {
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`${stat.type}.packetsDiscarded is sane number for a short test. ` +
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`value=${stat.packetsDiscarded}`
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);
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// bytesReceived
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ok(
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stat.bytesReceived >= 0 && stat.bytesReceived < 10 ** 9, // Not a magic number, just a guess
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@ -529,26 +498,6 @@ function pedanticChecks(report) {
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`local only test. value=${stat.jitter}`
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);
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// lastPacketReceivedTimestamp
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ok(
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stat.lastPacketReceivedTimestamp !== undefined,
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`${stat.type}.lastPacketReceivedTimestamp has a value`
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);
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// headerBytesReceived
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ok(
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stat.headerBytesReceived >= 0 && stat.headerBytesReceived < 50000,
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`${stat.type}.headerBytesReceived is sane for a short test. ` +
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`value=${stat.headerBytesReceived}`
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);
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// Always missing from libwebrtc stats
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// estimatedPlayoutTimestamp
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// ok(
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// stat.estimatedPlayoutTimestamp !== undefined,
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// `${stat.type}.estimatedPlayoutTimestamp has a value`
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// );
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// jitterBufferEmittedCount
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let expectedJitterBufferEmmitedCount = stat.kind == "video" ? 7 : 1000;
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ok(
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@ -595,19 +544,6 @@ function pedanticChecks(report) {
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`${stat.kind} test. value=${stat.totalSamplesReceived}`
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);
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// fecPacketsReceived
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ok(
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stat.fecPacketsReceived >= 0 && stat.fecPacketsReceived < 10 ** 5,
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`${stat.type}.fecPacketsReceived is a sane number for a short ` +
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`${stat.kind} test. value=${stat.fecPacketsReceived}`
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);
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// fecPacketsDiscarded
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ok(
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stat.fecPacketsDiscarded >= 0 && stat.fecPacketsDiscarded < 100,
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`${stat.type}.fecPacketsDiscarded is sane number for a short test. ` +
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`value=${stat.fecPacketsDiscarded}`
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);
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// concealedSamples
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ok(
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stat.concealedSamples >= 0 &&
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@ -623,51 +559,6 @@ function pedanticChecks(report) {
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`${stat.type}.silentConcealedSamples is a sane number for a short ` +
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`${stat.kind} test. value=${stat.silentConcealedSamples}`
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);
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// concealmentEvents
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ok(
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stat.concealmentEvents >= 0 &&
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stat.concealmentEvents <= stat.packetsReceived,
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`${stat.type}.concealmentEvents is a sane number for a short ` +
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`${stat.kind} test. value=${stat.concealmentEvents}`
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);
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// insertedSamplesForDeceleration
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ok(
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stat.insertedSamplesForDeceleration >= 0 &&
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stat.insertedSamplesForDeceleration <= stat.totalSamplesReceived,
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`${stat.type}.insertedSamplesForDeceleration is a sane number for a short ` +
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`${stat.kind} test. value=${stat.insertedSamplesForDeceleration}`
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);
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// removedSamplesForAcceleration
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ok(
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stat.removedSamplesForAcceleration >= 0 &&
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stat.removedSamplesForAcceleration <= stat.totalSamplesReceived,
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`${stat.type}.removedSamplesForAcceleration is a sane number for a short ` +
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`${stat.kind} test. value=${stat.removedSamplesForAcceleration}`
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);
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// audioLevel
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ok(
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stat.audioLevel >= 0 && stat.audioLevel <= 128,
|
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`${stat.type}.bytesReceived is a sane number for a short ` +
|
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`${stat.kind} test. value=${stat.audioLevel}`
|
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);
|
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|
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// totalAudioEnergy
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ok(
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stat.totalAudioEnergy >= 0 && stat.totalAudioEnergy <= 128,
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`${stat.type}.totalAudioEnergy is a sane number for a short ` +
|
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`${stat.kind} test. value=${stat.totalAudioEnergy}`
|
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);
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// totalSamplesDuration
|
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ok(
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stat.totalSamplesDuration >= 0 && stat.totalSamplesDuration <= 300,
|
||||
`${stat.type}.totalSamplesDuration is a sane number for a short ` +
|
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`${stat.kind} test. value=${stat.totalSamplesDuration}`
|
||||
);
|
||||
}
|
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|
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//
|
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|
|
@ -700,7 +591,6 @@ function pedanticChecks(report) {
|
|||
`${stat.type}.framesPerSecond is a sane number for a short ` +
|
||||
`${stat.kind} test. value=${stat.framesPerSecond}`
|
||||
);
|
||||
|
||||
// framesDecoded
|
||||
ok(
|
||||
stat.framesDecoded > 0 && stat.framesDecoded < 1000000,
|
||||
|
|
@ -722,35 +612,6 @@ function pedanticChecks(report) {
|
|||
`${stat.kind} test. value=${stat.frameHeight}`
|
||||
);
|
||||
|
||||
// totalDecodeTime
|
||||
ok(
|
||||
stat.totalDecodeTime >= 0 && stat.totalDecodeTime < 300,
|
||||
`${stat.type}.totalDecodeTime is sane for a short test. ` +
|
||||
`value=${stat.totalDecodeTime}`
|
||||
);
|
||||
|
||||
// totalProcessingDelay
|
||||
ok(
|
||||
stat.totalProcessingDelay < 10,
|
||||
`${stat.type}.totalProcessingDelay is sane number for a short test ` +
|
||||
`local only test. value=${stat.totalProcessingDelay}`
|
||||
);
|
||||
|
||||
// totalInterFrameDelay
|
||||
ok(
|
||||
stat.totalInterFrameDelay >= 0 && stat.totalInterFrameDelay < 100,
|
||||
`${stat.type}.totalInterFrameDelay is sane for a short test. ` +
|
||||
`value=${stat.totalInterFrameDelay}`
|
||||
);
|
||||
|
||||
// totalSquaredInterFrameDelay
|
||||
ok(
|
||||
stat.totalSquaredInterFrameDelay >= 0 &&
|
||||
stat.totalSquaredInterFrameDelay < 100,
|
||||
`${stat.type}.totalSquaredInterFrameDelay is sane for a short test. ` +
|
||||
`value=${stat.totalSquaredInterFrameDelay}`
|
||||
);
|
||||
|
||||
// framesReceived
|
||||
ok(
|
||||
stat.framesReceived >= 0 && stat.framesReceived < 100000,
|
||||
|
|
@ -795,28 +656,6 @@ function pedanticChecks(report) {
|
|||
`${stat.kind} test. value=${stat.packetsReceived}`
|
||||
);
|
||||
}
|
||||
|
||||
// totalRoundTripTime
|
||||
ok(
|
||||
stat.totalRoundTripTime < 50000,
|
||||
`${stat.type}.totalRoundTripTime is a sane number for a short ` +
|
||||
`${stat.kind} test. value=${stat.totalRoundTripTime}`
|
||||
);
|
||||
|
||||
// fractionLost
|
||||
ok(
|
||||
stat.fractionLost < 0.2,
|
||||
`${stat.type}.fractionLost is a sane number for a short ` +
|
||||
`${stat.kind} test. value=${stat.fractionLost}`
|
||||
);
|
||||
|
||||
// roundTripTimeMeasurements
|
||||
ok(
|
||||
stat.roundTripTimeMeasurements >= 1 &&
|
||||
stat.roundTripTimeMeasurements < 500,
|
||||
`${stat.type}.roundTripTimeMeasurements is a sane number for a short ` +
|
||||
`${stat.kind} test. value=${stat.roundTripTimeMeasurements}`
|
||||
);
|
||||
} else if (stat.type == "outbound-rtp") {
|
||||
//
|
||||
// Required fields
|
||||
|
|
|
|||
|
|
@ -68,41 +68,21 @@ dictionary RTCInboundRtpStreamStats : RTCReceivedRtpStreamStats {
|
|||
unsigned long frameWidth;
|
||||
unsigned long frameHeight;
|
||||
double framesPerSecond;
|
||||
unsigned long long qpSum;
|
||||
double totalDecodeTime;
|
||||
double totalInterFrameDelay;
|
||||
double totalSquaredInterFrameDelay;
|
||||
DOMHighResTimeStamp lastPacketReceivedTimestamp;
|
||||
unsigned long long headerBytesReceived;
|
||||
unsigned long long fecPacketsReceived;
|
||||
unsigned long long fecPacketsDiscarded;
|
||||
unsigned long long bytesReceived;
|
||||
unsigned long nackCount;
|
||||
unsigned long firCount;
|
||||
unsigned long pliCount;
|
||||
double totalProcessingDelay;
|
||||
// Always missing from libwebrtc
|
||||
// DOMHighResTimeStamp estimatedPlayoutTimestamp;
|
||||
double jitterBufferDelay;
|
||||
unsigned long long jitterBufferEmittedCount;
|
||||
unsigned long long totalSamplesReceived;
|
||||
unsigned long long concealedSamples;
|
||||
unsigned long long silentConcealedSamples;
|
||||
unsigned long long concealmentEvents;
|
||||
unsigned long long insertedSamplesForDeceleration;
|
||||
unsigned long long removedSamplesForAcceleration;
|
||||
double audioLevel;
|
||||
double totalAudioEnergy;
|
||||
double totalSamplesDuration;
|
||||
unsigned long framesReceived;
|
||||
};
|
||||
|
||||
dictionary RTCRemoteInboundRtpStreamStats : RTCReceivedRtpStreamStats {
|
||||
DOMString localId;
|
||||
double roundTripTime;
|
||||
double totalRoundTripTime;
|
||||
double fractionLost;
|
||||
unsigned long long roundTripTimeMeasurements;
|
||||
};
|
||||
|
||||
dictionary RTCSentRtpStreamStats : RTCRtpStreamStats {
|
||||
|
|
|
|||
|
|
@ -8,7 +8,49 @@
|
|||
[inbound-rtp's qpSum]
|
||||
expected: FAIL
|
||||
|
||||
[inbound-rtp's estimatedPlayoutTimestamp]
|
||||
[inbound-rtp's totalDecodeTime]
|
||||
expected: FAIL
|
||||
|
||||
[inbound-rtp's totalInterFrameDelay]
|
||||
expected: FAIL
|
||||
|
||||
[inbound-rtp's totalSquaredInterFrameDelay]
|
||||
expected: FAIL
|
||||
|
||||
[inbound-rtp's lastPacketReceivedTimestamp]
|
||||
expected: FAIL
|
||||
|
||||
[inbound-rtp's headerBytesReceived]
|
||||
expected: FAIL
|
||||
|
||||
[inbound-rtp's fecPacketsReceived]
|
||||
expected: FAIL
|
||||
|
||||
[inbound-rtp's fecPacketsDiscarded]
|
||||
expected: FAIL
|
||||
|
||||
[inbound-rtp's totalProcessingDelay]
|
||||
expected: FAIL
|
||||
|
||||
[inbound-rtp's estimatedPlayoutTimestamp]
|
||||
expected: FAIL
|
||||
|
||||
[inbound-rtp's concealmentEvents]
|
||||
expected: FAIL
|
||||
|
||||
[inbound-rtp's insertedSamplesForDeceleration]
|
||||
expected: FAIL
|
||||
|
||||
[inbound-rtp's removedSamplesForAcceleration]
|
||||
expected: FAIL
|
||||
|
||||
[inbound-rtp's audioLevel]
|
||||
expected: FAIL
|
||||
|
||||
[inbound-rtp's totalAudioEnergy]
|
||||
expected: FAIL
|
||||
|
||||
[inbound-rtp's totalSamplesDuration]
|
||||
expected: FAIL
|
||||
|
||||
[inbound-rtp's decoderImplementation]
|
||||
|
|
@ -53,6 +95,15 @@
|
|||
[outbound-rtp's transportId]
|
||||
expected: FAIL
|
||||
|
||||
[remote-inbound-rtp's totalRoundTripTime]
|
||||
expected: FAIL
|
||||
|
||||
[remote-inbound-rtp's fractionLost]
|
||||
expected: FAIL
|
||||
|
||||
[remote-inbound-rtp's roundTripTimeMeasurements]
|
||||
expected: FAIL
|
||||
|
||||
[remote-inbound-rtp's framesDropped]
|
||||
expected: FAIL
|
||||
|
||||
|
|
|
|||
|
|
@ -59,6 +59,14 @@
|
|||
bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1728357
|
||||
expected: FAIL
|
||||
|
||||
[RTCInboundRtpStreamStats's totalAudioEnergy]
|
||||
bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1728361
|
||||
expected: FAIL
|
||||
|
||||
[RTCInboundRtpStreamStats's totalSamplesDuration]
|
||||
bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1728362
|
||||
expected: FAIL
|
||||
|
||||
[RTCMediaSourceStats's trackIdentifier]
|
||||
bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1728364
|
||||
expected: FAIL
|
||||
|
|
|
|||
Загрузка…
Ссылка в новой задаче