Revert "stats: remove RTCRtpInboundRTPStream and RTCRtpoutboundRTPStream aliases"
This reverts commit 9671d60925b81baefd4a0d6b05ad539fa4a782d7. Reason for revert: Breaks dependencies, will re-land after fixes Original change's description: > stats: remove RTCRtpInboundRTPStream and RTCRtpoutboundRTPStream aliases > > after upgrading downstream projects > > BUG=webrtc:14973 > > Change-Id: I5df8e95a1c70b1d6078e255166c36ed01f868b6a > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296820 > Reviewed-by: Christoffer Jansson <jansson@webrtc.org> > Reviewed-by: Henrik Boström <hbos@webrtc.org> > Commit-Queue: Philipp Hancke <phancke@microsoft.com> > Cr-Commit-Position: refs/heads/main@{#39526} Bug: webrtc:14973 Change-Id: I50878526566660d9772f7c8664970eec8bd86341 No-Presubmit: true No-Tree-Checks: true No-Try: true Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296940 Reviewed-by: Philipp Hancke <phancke@microsoft.com> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Auto-Submit: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39530}
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@ -493,6 +493,8 @@ class RTC_EXPORT RTCInboundRtpStreamStats final
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// The former googMinPlayoutDelayMs (in seconds).
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RTCNonStandardStatsMember<double> min_playout_delay;
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};
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// TODO(bugs.webrtc.org/14973): remove name alias.
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using RTCInboundRTPStreamStats = RTCInboundRtpStreamStats;
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// https://w3c.github.io/webrtc-stats/#outboundrtpstats-dict*
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class RTC_EXPORT RTCOutboundRtpStreamStats final
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@ -544,6 +546,8 @@ class RTC_EXPORT RTCOutboundRtpStreamStats final
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power_efficient_encoder;
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RTCStatsMember<std::string> scalability_mode;
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};
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// TODO(bugs.webrtc.org/14973): remove name alias.
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using RTCOutboundRTPStreamStats = RTCOutboundRtpStreamStats;
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// https://w3c.github.io/webrtc-stats/#remoteinboundrtpstats-dict*
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class RTC_EXPORT RTCRemoteInboundRtpStreamStats final
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@ -48,7 +48,7 @@ class PeerConnectionMediaChannelSplitTest
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int NacksReceivedCount(PeerConnectionIntegrationWrapper& pc) {
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rtc::scoped_refptr<const webrtc::RTCStatsReport> report = pc.NewGetStats();
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auto sender_stats = report->GetStatsOfType<RTCOutboundRtpStreamStats>();
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auto sender_stats = report->GetStatsOfType<RTCOutboundRTPStreamStats>();
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if (sender_stats.size() != 1) {
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ADD_FAILURE();
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return 0;
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@ -61,7 +61,7 @@ int NacksReceivedCount(PeerConnectionIntegrationWrapper& pc) {
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int NacksSentCount(PeerConnectionIntegrationWrapper& pc) {
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rtc::scoped_refptr<const webrtc::RTCStatsReport> report = pc.NewGetStats();
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auto receiver_stats = report->GetStatsOfType<RTCInboundRtpStreamStats>();
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auto receiver_stats = report->GetStatsOfType<RTCInboundRTPStreamStats>();
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if (receiver_stats.size() != 1) {
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ADD_FAILURE();
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return 0;
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