commit 2b9fa09fa3e3379fd8e76490c394f25670352ef2
author Henrik Boström <hbos@webrtc.org> Fri Aug 16 13:09:51 2019 +0200
committer Commit Bot <commit-bot@chromium.org> Fri Aug 16 14:10:46 2019 +0000
tree 7129633350d6a856b41e63fcb7108e5daf1a15cd
parent cc96db6de20131be1248e4a237e30a5a6d59cae8

[GetStats] Expose video codec implementation in standardized metrics.

Spec issue: https://github.com/w3c/webrtc-stats/issues/445
Spec PR: https://github.com/w3c/webrtc-stats/pull/473

Now that the spec's RTCCodecStats.implementation has moved to
RTCOutboundRtpStreamStats.encoderImplementation and
RTCInboundRtpStreamStats.decoderImplementation, this CL implements them
using the same string that the legacy getStats() API used.

Bug: webrtc:10890
Change-Id: Ic43ce44735453626791959df3061ee253356015a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149168
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28877}

api/stats/rtcstats_objects.h

diff --git a/api/stats/rtcstats_objects.h b/api/stats/rtcstats_objects.h
index 8955c4e..b492203 100644
--- a/api/stats/rtcstats_objects.h
+++ b/api/stats/rtcstats_objects.h
@@ -122,8 +122,6 @@
   RTCStatsMember<uint32_t> channels;
   // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7061
   RTCStatsMember<std::string> sdp_fmtp_line;
-  // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7061
-  RTCStatsMember<std::string> implementation;
 };
 
 // https://w3c.github.io/webrtc-stats/#dcstats-dict*
@@ -447,6 +445,9 @@
   RTCStatsMember<double> total_decode_time;
   // https://henbos.github.io/webrtc-provisional-stats/#dom-rtcinboundrtpstreamstats-contenttype
   RTCStatsMember<std::string> content_type;
+  // TODO(hbos): This is only implemented for video; implement it for audio as
+  // well.
+  RTCStatsMember<std::string> decoder_implementation;
 };
 
 // https://w3c.github.io/webrtc-stats/#outboundrtpstats-dict*
@@ -482,6 +483,9 @@
   RTCStatsMember<std::string> quality_limitation_reason;
   // https://henbos.github.io/webrtc-provisional-stats/#dom-rtcoutboundrtpstreamstats-contenttype
   RTCStatsMember<std::string> content_type;
+  // TODO(hbos): This is only implemented for video; implement it for audio as
+  // well.
+  RTCStatsMember<std::string> encoder_implementation;
 };
 
 // TODO(https://crbug.com/webrtc/10671): Refactor the stats dictionaries to have