Revert "Complete migration from "track" to "inbound-rtp" stats"
This reverts commit 94fe0d3de5e8162d1a105fd1a3ec4bd2da97f43b. Reason for revert: Causes an assert in this line during a call: https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/sdk/objc/api/peerconnection/RTCStatisticsReport.mm;drc=87a6e5ab4d8f0baf4e2a9b7752b43d825f9c0ce1;l=122?originalUrl=https:%2F%2Fcs.chromium.org%2F where frameWidth appears more than once Original change's description: > Complete migration from "track" to "inbound-rtp" stats > > Bug: webrtc:11683 > Change-Id: I4c4a4fa0a7d6a20976922aca41d57540aa27fd1d > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178611 > 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@{#31683} TBR=hbos@webrtc.org,hta@webrtc.org,elrello@microsoft.com Change-Id: I0ded36a40c8808dac5a777ed41815e52ab9a2573 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:11683 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179020 Reviewed-by: Zeke Chin <tkchin@webrtc.org> Commit-Queue: Zeke Chin <tkchin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31692}
This commit is contained in:
@ -419,18 +419,6 @@ class RTC_EXPORT RTCInboundRTPStreamStats final : public RTCRTPStreamStats {
|
||||
// TODO(hbos): Collect and populate this value for both "audio" and "video",
|
||||
// currently not collected for "video". https://bugs.webrtc.org/7065
|
||||
RTCStatsMember<double> jitter;
|
||||
RTCStatsMember<double> jitter_buffer_delay;
|
||||
RTCStatsMember<uint64_t> jitter_buffer_emitted_count;
|
||||
RTCStatsMember<uint64_t> total_samples_received;
|
||||
RTCStatsMember<uint64_t> concealed_samples;
|
||||
RTCStatsMember<uint64_t> silent_concealed_samples;
|
||||
RTCStatsMember<uint64_t> concealment_events;
|
||||
RTCStatsMember<uint64_t> inserted_samples_for_deceleration;
|
||||
RTCStatsMember<uint64_t> removed_samples_for_acceleration;
|
||||
RTCStatsMember<double> audio_level;
|
||||
RTCStatsMember<double> total_audio_energy;
|
||||
RTCStatsMember<double> total_samples_duration;
|
||||
RTCStatsMember<int32_t> frames_received;
|
||||
// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
|
||||
RTCStatsMember<double> round_trip_time;
|
||||
// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
|
||||
@ -453,13 +441,8 @@ class RTC_EXPORT RTCInboundRTPStreamStats final : public RTCRTPStreamStats {
|
||||
RTCStatsMember<double> gap_loss_rate;
|
||||
// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
|
||||
RTCStatsMember<double> gap_discard_rate;
|
||||
RTCStatsMember<uint32_t> frame_width;
|
||||
RTCStatsMember<uint32_t> frame_height;
|
||||
RTCStatsMember<uint32_t> frame_bit_depth;
|
||||
RTCStatsMember<double> frames_per_second;
|
||||
RTCStatsMember<uint32_t> frames_decoded;
|
||||
RTCStatsMember<uint32_t> key_frames_decoded;
|
||||
RTCStatsMember<uint32_t> frames_dropped;
|
||||
RTCStatsMember<double> total_decode_time;
|
||||
RTCStatsMember<double> total_inter_frame_delay;
|
||||
RTCStatsMember<double> total_squared_inter_frame_delay;
|
||||
|
||||
Reference in New Issue
Block a user