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| author | Henrik Boström <hbos@webrtc.org> | 2020-05-05 12:20:01 +0200 |
|---|---|---|
| committer | Commit Bot <commit-bot@chromium.org> | 2020-05-05 13:38:51 +0000 |
| commit | 9a925c9ce33a6ccdd11b545b11ba68e985c2a65d (patch) | |
| tree | a78863f258000505a13a05dfb8a8a77efb50b5da /api | |
| parent | 49f574b3b33915405e03e02ca47472677ea4a448 (diff) | |
| download | webrtc-9a925c9ce33a6ccdd11b545b11ba68e985c2a65d.tar.gz | |
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}
Diffstat (limited to 'api')
| -rw-r--r-- | api/peer_connection_interface.h | 2 | ||||
| -rw-r--r-- | api/stats/rtcstats_objects.h | 6 |
2 files changed, 0 insertions, 8 deletions
diff --git a/api/peer_connection_interface.h b/api/peer_connection_interface.h index 1d81de74d8..0ae47b2a2f 100644 --- a/api/peer_connection_interface.h +++ b/api/peer_connection_interface.h @@ -666,8 +666,6 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface { // Whether network condition based codec switching is allowed. absl::optional<bool> allow_codec_switching; - bool enable_simulcast_stats = true; - // // Don't forget to update operator== if adding something. // diff --git a/api/stats/rtcstats_objects.h b/api/stats/rtcstats_objects.h index 28d841db09..e83c83d97e 100644 --- a/api/stats/rtcstats_objects.h +++ b/api/stats/rtcstats_objects.h @@ -469,7 +469,6 @@ class RTC_EXPORT RTCOutboundRTPStreamStats final : public RTCRTPStreamStats { RTCStatsMember<std::string> media_source_id; RTCStatsMember<std::string> remote_id; - RTCStatsMember<std::string> rid; RTCStatsMember<uint32_t> packets_sent; RTCStatsMember<uint64_t> retransmitted_packets_sent; RTCStatsMember<uint64_t> bytes_sent; @@ -481,11 +480,6 @@ class RTC_EXPORT RTCOutboundRTPStreamStats final : public RTCRTPStreamStats { RTCStatsMember<uint32_t> key_frames_encoded; RTCStatsMember<double> total_encode_time; RTCStatsMember<uint64_t> total_encoded_bytes_target; - RTCStatsMember<uint32_t> frame_width; - RTCStatsMember<uint32_t> frame_height; - RTCStatsMember<double> frames_per_second; - RTCStatsMember<uint32_t> frames_sent; - RTCStatsMember<uint32_t> huge_frames_sent; // TODO(https://crbug.com/webrtc/10635): This is only implemented for video; // implement it for audio as well. RTCStatsMember<double> total_packet_send_delay; |