Bug 1270066 - Update the WebRTC histograms with an alert email value. r=dexter f=jesup
authorGeorg Fritzsche <georg.fritzsche@googlemail.com>
Thu, 05 May 2016 15:17:43 +0200
changeset 296365 06355277b0ac0e7d2d155453a2ca671fd57737f9
parent 296233 29662e28a9c93ac67ee0b8ddfb65a9f29bbf73f5
child 296366 f6a21128e6ac685e12cb39e691b86e8f48189ab9
push id76311
push usercbook@mozilla.com
push dateFri, 06 May 2016 12:26:12 +0000
treeherdermozilla-inbound@84a3e5716801 [default view] [failures only]
perfherder[talos] [build metrics] [platform microbench] (compared to previous push)
reviewersdexter
bugs1270066
milestone49.0a1
first release with
nightly linux32
nightly linux64
nightly mac
nightly win32
nightly win64
last release without
nightly linux32
nightly linux64
nightly mac
nightly win32
nightly win64
Bug 1270066 - Update the WebRTC histograms with an alert email value. r=dexter f=jesup
toolkit/components/telemetry/Histograms.json
toolkit/components/telemetry/histogram-whitelists.json
--- a/toolkit/components/telemetry/Histograms.json
+++ b/toolkit/components/telemetry/Histograms.json
@@ -6456,233 +6456,265 @@
   "WEBRTC_STUN_RATE_LIMIT_EXCEEDED_BY_TYPE_GIVEN_FAILURE": {
     "alert_emails": ["webrtc-ice-telemetry-alerts@mozilla.com"],
     "expires_in_version": "53",
     "kind": "enumerated",
     "n_values": 4,
     "description": "For each failed PeerConnection, bit 0 indicates the short-duration rate limit was reached, bit 1 indicates the long-duration rate limit was reached"
   },
   "WEBRTC_AVSYNC_WHEN_AUDIO_LAGS_VIDEO_MS": {
+    "alert_emails": ["webrtc-telemetry-alerts@mozilla.com"],
     "expires_in_version": "never",
     "kind": "exponential",
     "high": 60000,
     "n_buckets": 1000,
     "description": "The delay (in milliseconds) when audio is behind video. Zero delay is counted. Measured every second of a call."
   },
   "WEBRTC_AVSYNC_WHEN_VIDEO_LAGS_AUDIO_MS": {
+    "alert_emails": ["webrtc-telemetry-alerts@mozilla.com"],
     "expires_in_version": "never",
     "kind": "exponential",
     "high": 60000,
     "n_buckets": 1000,
     "description": "The delay (in milliseconds) when video is behind audio. Zero delay is not counted. Measured every second of a call."
   },
   "WEBRTC_VIDEO_QUALITY_INBOUND_BANDWIDTH_KBITS": {
+    "alert_emails": ["webrtc-telemetry-alerts@mozilla.com"],
     "expires_in_version": "never",
     "kind": "exponential",
     "high": 1000000,
     "n_buckets": 1000,
     "description": "Locally measured data rate of inbound video (kbit/s). Computed every second of a call."
   },
   "WEBRTC_AUDIO_QUALITY_INBOUND_BANDWIDTH_KBITS": {
+    "alert_emails": ["webrtc-telemetry-alerts@mozilla.com"],
     "expires_in_version": "never",
     "kind": "exponential",
     "high": 1000000,
     "n_buckets": 1000,
     "description": "Locally measured data rate on inbound audio (kbit/s). Computed every second of a call."
   },
   "WEBRTC_VIDEO_QUALITY_OUTBOUND_BANDWIDTH_KBITS": {
+    "alert_emails": ["webrtc-telemetry-alerts@mozilla.com"],
     "expires_in_version": "never",
     "kind": "exponential",
     "high": 1000000,
     "n_buckets": 1000,
     "description": "Data rate deduced from RTCP from remote recipient of outbound video (kbit/s). Computed every second of a call (for easy comparison)."
   },
   "WEBRTC_AUDIO_QUALITY_OUTBOUND_BANDWIDTH_KBITS": {
+    "alert_emails": ["webrtc-telemetry-alerts@mozilla.com"],
     "expires_in_version": "never",
     "kind": "exponential",
     "high": 1000000,
     "n_buckets": 1000,
     "description": "Data rate deduced from RTCP from remote recipient of outbound audio (kbit/s). Computed every second of a call (for easy comparison)."
   },
   "WEBRTC_VIDEO_QUALITY_INBOUND_PACKETLOSS_RATE": {
+    "alert_emails": ["webrtc-telemetry-alerts@mozilla.com"],
     "expires_in_version": "never",
     "kind": "exponential",
     "high": 1000,
     "n_buckets": 100,
     "description": "Locally measured packet loss on inbound video (permille). Sampled every second of a call."
   },
   "WEBRTC_AUDIO_QUALITY_INBOUND_PACKETLOSS_RATE": {
+    "alert_emails": ["webrtc-telemetry-alerts@mozilla.com"],
     "expires_in_version": "never",
     "kind": "exponential",
     "high": 1000,
     "n_buckets": 100,
     "description": "Locally measured packet loss on inbound audio (permille). Sampled every second of a call."
   },
   "WEBRTC_VIDEO_QUALITY_OUTBOUND_PACKETLOSS_RATE": {
+    "alert_emails": ["webrtc-telemetry-alerts@mozilla.com"],
     "expires_in_version": "never",
     "kind": "exponential",
     "high": 1000,
     "n_buckets": 100,
     "description": "RTCP-reported packet loss by remote recipient of outbound video (permille). Sampled every second of a call (for easy comparison)."
   },
   "WEBRTC_AUDIO_QUALITY_OUTBOUND_PACKETLOSS_RATE": {
+    "alert_emails": ["webrtc-telemetry-alerts@mozilla.com"],
     "expires_in_version": "never",
     "kind": "exponential",
     "high": 1000,
     "n_buckets": 100,
     "description": "RTCP-reported packet loss by remote recipient of outbound audio (permille). Sampled every second of a call (for easy comparison)."
   },
   "WEBRTC_VIDEO_QUALITY_INBOUND_JITTER": {
+    "alert_emails": ["webrtc-telemetry-alerts@mozilla.com"],
     "expires_in_version": "never",
     "kind": "exponential",
     "high": 10000,
     "n_buckets": 100,
     "description": "Locally measured jitter on inbound video (ms). Sampled every second of a call."
   },
   "WEBRTC_AUDIO_QUALITY_INBOUND_JITTER": {
+    "alert_emails": ["webrtc-telemetry-alerts@mozilla.com"],
     "expires_in_version": "never",
     "kind": "exponential",
     "high": 10000,
     "n_buckets": 1000,
     "description": "Locally measured jitter on inbound audio (ms). Sampled every second of a call."
   },
   "WEBRTC_VIDEO_QUALITY_OUTBOUND_JITTER": {
+    "alert_emails": ["webrtc-telemetry-alerts@mozilla.com"],
     "expires_in_version": "never",
     "kind": "exponential",
     "high": 10000,
     "n_buckets": 1000,
     "description": "RTCP-reported jitter by remote recipient of outbound video (ms). Sampled every second of a call (for easy comparison)."
   },
   "WEBRTC_AUDIO_QUALITY_OUTBOUND_JITTER": {
+    "alert_emails": ["webrtc-telemetry-alerts@mozilla.com"],
     "expires_in_version": "never",
     "kind": "exponential",
     "high": 10000,
     "n_buckets": 1000,
     "description": "RTCP-reported jitter by remote recipient of outbound audio (ms). Sampled every second of a call (for easy comparison)."
   },
   "WEBRTC_VIDEO_ERROR_RECOVERY_MS": {
+    "alert_emails": ["webrtc-telemetry-alerts@mozilla.com"],
     "expires_in_version": "never",
     "kind": "exponential",
     "high": 10000,
     "n_buckets": 500,
     "description": "Time to recover from a video error in ms"
   },
   "WEBRTC_VIDEO_RECOVERY_BEFORE_ERROR_PER_MIN": {
+    "alert_emails": ["webrtc-telemetry-alerts@mozilla.com"],
     "expires_in_version": "never",
     "kind": "exponential",
     "high": 1000,
     "n_buckets": 200,
     "description": "Number of losses recovered before error per min"
   },
   "WEBRTC_VIDEO_RECOVERY_AFTER_ERROR_PER_MIN": {
+    "alert_emails": ["webrtc-telemetry-alerts@mozilla.com"],
     "expires_in_version": "never",
     "kind": "exponential",
     "high": 1000,
     "n_buckets": 200,
     "description": "Number of losses recovered after error per min"
   },
   "WEBRTC_VIDEO_DECODE_ERROR_TIME_PERMILLE": {
+    "alert_emails": ["webrtc-telemetry-alerts@mozilla.com"],
     "expires_in_version": "never",
     "kind": "exponential",
     "high": 1000,
     "n_buckets": 100,
     "description": "Percentage*10 (permille) of call decoding with errors or frozen due to errors"
   },
   "WEBRTC_VIDEO_QUALITY_OUTBOUND_RTT": {
+    "alert_emails": ["webrtc-telemetry-alerts@mozilla.com"],
     "expires_in_version": "never",
     "kind": "exponential",
     "high": 10000,
     "n_buckets": 1000,
     "description": "Roundtrip time of outbound video (ms). Sampled every second of a call."
   },
   "WEBRTC_AUDIO_QUALITY_OUTBOUND_RTT": {
+    "alert_emails": ["webrtc-telemetry-alerts@mozilla.com"],
     "expires_in_version": "never",
     "kind": "exponential",
     "high": 10000,
     "n_buckets": 1000,
     "description": "Roundtrip time of outbound audio (ms). Sampled every second of a call."
   },
   "WEBRTC_VIDEO_ENCODER_BITRATE_AVG_PER_CALL_KBPS": {
+    "alert_emails": ["webrtc-telemetry-alerts@mozilla.com"],
     "expires_in_version": "never",
     "kind": "exponential",
     "high": 10000,
     "n_buckets": 100,
     "description": "Video encoder's average bitrate (in kbits/s) over an entire call"
   },
   "WEBRTC_VIDEO_ENCODER_BITRATE_STD_DEV_PER_CALL_KBPS": {
+    "alert_emails": ["webrtc-telemetry-alerts@mozilla.com"],
     "expires_in_version": "never",
     "kind": "exponential",
     "high": 5000,
     "n_buckets": 100,
     "description": "Standard deviation from video encoder's average bitrate (in kbits/s) over an entire call"
   },
   "WEBRTC_VIDEO_ENCODER_FRAMERATE_AVG_PER_CALL": {
+    "alert_emails": ["webrtc-telemetry-alerts@mozilla.com"],
     "expires_in_version": "never",
     "kind": "exponential",
     "high": 200,
     "n_buckets": 50,
     "description": "Video encoder's average framerate (in fps) over an entire call"
   },
   "WEBRTC_VIDEO_ENCODER_FRAMERATE_10X_STD_DEV_PER_CALL": {
+    "alert_emails": ["webrtc-telemetry-alerts@mozilla.com"],
     "expires_in_version": "never",
     "kind": "exponential",
     "high": 200,
     "n_buckets": 50,
     "description": "Standard deviation from video encoder's average framerate (in 1/10 fps) over an entire call"
   },
   "WEBRTC_VIDEO_ENCODER_DROPPED_FRAMES_PER_CALL_FPM": {
+    "alert_emails": ["webrtc-telemetry-alerts@mozilla.com"],
     "expires_in_version": "never",
     "kind": "exponential",
     "high": 50000,
     "n_buckets": 100,
     "description": "Video encoder's number of frames dropped (in frames/min) over an entire call"
   },
   "WEBRTC_VIDEO_DECODER_BITRATE_AVG_PER_CALL_KBPS": {
+    "alert_emails": ["webrtc-telemetry-alerts@mozilla.com"],
     "expires_in_version": "never",
     "kind": "exponential",
     "high": 10000,
     "n_buckets": 100,
     "description": "Video decoder's average bitrate (in kbits/s) over an entire call"
   },
   "WEBRTC_VIDEO_DECODER_BITRATE_STD_DEV_PER_CALL_KBPS": {
+    "alert_emails": ["webrtc-telemetry-alerts@mozilla.com"],
     "expires_in_version": "never",
     "kind": "exponential",
     "high": 5000,
     "n_buckets": 100,
     "description": "Standard deviation from video decoder's average bitrate (in kbits/s) over an entire call"
   },
   "WEBRTC_VIDEO_DECODER_FRAMERATE_AVG_PER_CALL": {
+    "alert_emails": ["webrtc-telemetry-alerts@mozilla.com"],
     "expires_in_version": "never",
     "kind": "exponential",
     "high": 200,
     "n_buckets": 50,
     "description": "Video decoder's average framerate (in fps) over an entire call"
   },
   "WEBRTC_VIDEO_DECODER_FRAMERATE_10X_STD_DEV_PER_CALL": {
+    "alert_emails": ["webrtc-telemetry-alerts@mozilla.com"],
     "expires_in_version": "never",
     "kind": "exponential",
     "high": 200,
     "n_buckets": 50,
     "description": "Standard deviation from video decoder's average framerate (in 1/10 fps) over an entire call"
   },
   "WEBRTC_VIDEO_DECODER_DISCARDED_PACKETS_PER_CALL_PPM": {
+    "alert_emails": ["webrtc-telemetry-alerts@mozilla.com"],
     "expires_in_version": "never",
     "kind": "exponential",
     "high": 50000,
     "n_buckets": 100,
     "description": "Video decoder's number of discarded packets (in packets/min) over an entire call"
   },
   "WEBRTC_CALL_DURATION": {
+    "alert_emails": ["webrtc-telemetry-alerts@mozilla.com"],
     "expires_in_version": "never",
     "kind": "exponential",
     "high": 10000,
     "n_buckets": 1000,
     "description": "The length of time (in seconds) that a call lasted."
   },
   "WEBRTC_CALL_COUNT": {
+    "alert_emails": ["webrtc-telemetry-alerts@mozilla.com"],
     "expires_in_version": "48",
     "kind": "exponential",
     "high": 500,
     "n_buckets": 50,
     "description": "The number of calls made during a session."
   },
   "WEBRTC_CALL_COUNT_2": {
     "alert_emails": ["webrtc-telemetry-alerts@mozilla.com"],
@@ -6711,104 +6743,118 @@
     "alert_emails": ["seceng@mozilla.org"],
     "expires_in_version": "50",
     "kind": "enumerated",
     "n_values": 15,
     "description": "Origins for getUserMedia calls (0=other, 1=HTTPS, 2=file, 3=app, 4=localhost, 5=loop, 6=privileged)",
     "releaseChannelCollection": "opt-out"
   },
   "WEBRTC_GET_USER_MEDIA_TYPE": {
+    "alert_emails": ["webrtc-telemetry-alerts@mozilla.com"],
     "expires_in_version": "never",
     "kind": "enumerated",
     "n_values": 8,
     "description": "Type for media in getUserMedia calls (0=Camera, 1=Screen, 2=Application, 3=Window, 4=Browser, 5=Microphone, 6=AudioCapture, 7=Other)"
   },
   "WEBRTC_LOAD_STATE_RELAXED": {
+    "alert_emails": ["webrtc-telemetry-alerts@mozilla.com"],
     "expires_in_version": "never",
     "kind": "linear",
     "high": 100,
     "n_buckets": 25,
     "description": "Percentage of time spent in the Relaxed load state in calls over 30 seconds."
   },
   "WEBRTC_LOAD_STATE_RELAXED_SHORT": {
+    "alert_emails": ["webrtc-telemetry-alerts@mozilla.com"],
     "expires_in_version": "never",
     "kind": "linear",
     "high": 100,
     "n_buckets": 25,
     "description": "Percentage of time spent in the Relaxed load state in calls 5-30 seconds."
   },
   "WEBRTC_LOAD_STATE_NORMAL": {
+    "alert_emails": ["webrtc-telemetry-alerts@mozilla.com"],
     "expires_in_version": "never",
     "kind": "linear",
     "high": 100,
     "n_buckets": 25,
     "description": "Percentage of time spent in the Normal load state in calls over 30 seconds."
   },
   "WEBRTC_LOAD_STATE_NORMAL_SHORT": {
+    "alert_emails": ["webrtc-telemetry-alerts@mozilla.com"],
     "expires_in_version": "never",
     "kind": "linear",
     "high": 100,
     "n_buckets": 25,
     "description": "Percentage of time spent in the Normal load state in calls over 5-30 seconds."
   },
   "WEBRTC_LOAD_STATE_STRESSED": {
+    "alert_emails": ["webrtc-telemetry-alerts@mozilla.com"],
     "expires_in_version": "never",
     "kind": "linear",
     "high": 100,
     "n_buckets": 25,
     "description": "Percentage of time spent in the Stressed load state in calls over 30 seconds."
   },
   "WEBRTC_LOAD_STATE_STRESSED_SHORT": {
+    "alert_emails": ["webrtc-telemetry-alerts@mozilla.com"],
     "expires_in_version": "never",
     "kind": "linear",
     "high": 100,
     "n_buckets": 25,
     "description": "Percentage of time spent in the Stressed load state in calls 5-30 seconds."
   },
   "WEBRTC_RENEGOTIATIONS": {
+    "alert_emails": ["webrtc-telemetry-alerts@mozilla.com"],
     "expires_in_version": "never",
     "kind": "linear",
     "high": 21,
     "n_buckets": 20,
     "description": "Number of Renegotiations during each call"
   },
   "WEBRTC_MAX_VIDEO_SEND_TRACK": {
+    "alert_emails": ["webrtc-telemetry-alerts@mozilla.com"],
     "expires_in_version": "never",
     "kind": "linear",
     "high": 10,
     "n_buckets": 9,
     "description": "Number of Video tracks sent simultaneously"
   },
   "WEBRTC_MAX_VIDEO_RECEIVE_TRACK": {
+    "alert_emails": ["webrtc-telemetry-alerts@mozilla.com"],
     "expires_in_version": "never",
     "kind": "linear",
     "high": 20,
     "n_buckets": 19,
     "description": "Number of Video tracks received simultaneously"
   },
   "WEBRTC_MAX_AUDIO_SEND_TRACK": {
+    "alert_emails": ["webrtc-telemetry-alerts@mozilla.com"],
     "expires_in_version": "never",
     "kind": "linear",
     "high": 20,
     "n_buckets": 19,
     "description": "Number of Audio tracks sent simultaneously"
   },
   "WEBRTC_MAX_AUDIO_RECEIVE_TRACK": {
+    "alert_emails": ["webrtc-telemetry-alerts@mozilla.com"],
     "expires_in_version": "never",
     "kind": "linear",
     "high": 30,
     "n_buckets": 29,
     "description": "Number of Audio tracks received simultaneously"
   },
   "WEBRTC_DATACHANNEL_NEGOTIATED": {
+    "alert_emails": ["webrtc-telemetry-alerts@mozilla.com"],
     "expires_in_version": "never",
     "kind": "boolean",
     "description": "Was DataChannels negotiated"
   },
   "WEBRTC_CALL_TYPE": {
+    "alert_emails": ["webrtc-telemetry-alerts@mozilla.com"],
     "expires_in_version": "never",
     "kind": "enumerated",
     "n_values": 8,
     "description": "Type of call: (Bitmask) Audio = 1, Video = 2, DataChannels = 4"
   },
   "DEVTOOLS_DEBUGGER_RDP_LOCAL_TRACERDETACH_MS": {
     "expires_in_version": "never",
     "kind": "exponential",
--- a/toolkit/components/telemetry/histogram-whitelists.json
+++ b/toolkit/components/telemetry/histogram-whitelists.json
@@ -1028,62 +1028,16 @@
     "WEAVE_START_COUNT",
     "WEBCRYPTO_ALG",
     "WEBCRYPTO_EXTRACTABLE_ENC",
     "WEBCRYPTO_EXTRACTABLE_GENERATE",
     "WEBCRYPTO_EXTRACTABLE_IMPORT",
     "WEBCRYPTO_EXTRACTABLE_SIG",
     "WEBCRYPTO_METHOD",
     "WEBCRYPTO_RESOLVED",
-    "WEBRTC_AUDIO_QUALITY_INBOUND_BANDWIDTH_KBITS",
-    "WEBRTC_AUDIO_QUALITY_INBOUND_JITTER",
-    "WEBRTC_AUDIO_QUALITY_INBOUND_PACKETLOSS_RATE",
-    "WEBRTC_AUDIO_QUALITY_OUTBOUND_BANDWIDTH_KBITS",
-    "WEBRTC_AUDIO_QUALITY_OUTBOUND_JITTER",
-    "WEBRTC_AUDIO_QUALITY_OUTBOUND_PACKETLOSS_RATE",
-    "WEBRTC_AUDIO_QUALITY_OUTBOUND_RTT",
-    "WEBRTC_AVSYNC_WHEN_AUDIO_LAGS_VIDEO_MS",
-    "WEBRTC_AVSYNC_WHEN_VIDEO_LAGS_AUDIO_MS",
-    "WEBRTC_CALL_COUNT",
-    "WEBRTC_CALL_DURATION",
-    "WEBRTC_CALL_TYPE",
-    "WEBRTC_DATACHANNEL_NEGOTIATED",
-    "WEBRTC_GET_USER_MEDIA_TYPE",
-    "WEBRTC_LOAD_STATE_NORMAL",
-    "WEBRTC_LOAD_STATE_NORMAL_SHORT",
-    "WEBRTC_LOAD_STATE_RELAXED",
-    "WEBRTC_LOAD_STATE_RELAXED_SHORT",
-    "WEBRTC_LOAD_STATE_STRESSED",
-    "WEBRTC_LOAD_STATE_STRESSED_SHORT",
-    "WEBRTC_MAX_AUDIO_RECEIVE_TRACK",
-    "WEBRTC_MAX_AUDIO_SEND_TRACK",
-    "WEBRTC_MAX_VIDEO_RECEIVE_TRACK",
-    "WEBRTC_MAX_VIDEO_SEND_TRACK",
-    "WEBRTC_RENEGOTIATIONS",
-    "WEBRTC_VIDEO_DECODER_BITRATE_AVG_PER_CALL_KBPS",
-    "WEBRTC_VIDEO_DECODER_BITRATE_STD_DEV_PER_CALL_KBPS",
-    "WEBRTC_VIDEO_DECODER_DISCARDED_PACKETS_PER_CALL_PPM",
-    "WEBRTC_VIDEO_DECODER_FRAMERATE_10X_STD_DEV_PER_CALL",
-    "WEBRTC_VIDEO_DECODER_FRAMERATE_AVG_PER_CALL",
-    "WEBRTC_VIDEO_DECODE_ERROR_TIME_PERMILLE",
-    "WEBRTC_VIDEO_ENCODER_BITRATE_AVG_PER_CALL_KBPS",
-    "WEBRTC_VIDEO_ENCODER_BITRATE_STD_DEV_PER_CALL_KBPS",
-    "WEBRTC_VIDEO_ENCODER_DROPPED_FRAMES_PER_CALL_FPM",
-    "WEBRTC_VIDEO_ENCODER_FRAMERATE_10X_STD_DEV_PER_CALL",
-    "WEBRTC_VIDEO_ENCODER_FRAMERATE_AVG_PER_CALL",
-    "WEBRTC_VIDEO_ERROR_RECOVERY_MS",
-    "WEBRTC_VIDEO_QUALITY_INBOUND_BANDWIDTH_KBITS",
-    "WEBRTC_VIDEO_QUALITY_INBOUND_JITTER",
-    "WEBRTC_VIDEO_QUALITY_INBOUND_PACKETLOSS_RATE",
-    "WEBRTC_VIDEO_QUALITY_OUTBOUND_BANDWIDTH_KBITS",
-    "WEBRTC_VIDEO_QUALITY_OUTBOUND_JITTER",
-    "WEBRTC_VIDEO_QUALITY_OUTBOUND_PACKETLOSS_RATE",
-    "WEBRTC_VIDEO_QUALITY_OUTBOUND_RTT",
-    "WEBRTC_VIDEO_RECOVERY_AFTER_ERROR_PER_MIN",
-    "WEBRTC_VIDEO_RECOVERY_BEFORE_ERROR_PER_MIN",
     "WEBSOCKETS_HANDSHAKE_TYPE",
     "WORD_CACHE_HITS_CHROME",
     "WORD_CACHE_HITS_CONTENT",
     "WORD_CACHE_MISSES_CHROME",
     "WORD_CACHE_MISSES_CONTENT",
     "XMLHTTPREQUEST_ASYNC_OR_SYNC",
     "XUL_BACKGROUND_REFLOW_MS",
     "XUL_CACHE_DISABLED",