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8 commits

Author SHA1 Message Date
Jonas Olsson
a4d873786f Format almost everything. 
This CL was generated by running

git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \
grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \
grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \
grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \
grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \
| xargs clang-format -i ; git cl format

Most of these changes are clang-format grouping and reordering includes
differently.

Bug: webrtc:9340
Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28505}
 2019-07-08 13:45:15 +00:00
Jakob Ivarsson
8fc92e640a Add lifetime concealment stats to NetEqStatsPlotter. 
Bug: None
Change-Id: Iaf91218e3ebedf301e991083fe32cb26ba5b7476
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135562
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27875}
 2019-05-08 08:40:08 +00:00
Henrik Lundin
e360c09c86 NetEq: Minor change to print-out format for interruption stats 
Going back to a ratio in [0.0, 1.0] instead of a % number. Also changed
the format of the tag to match the others.

Bug: webrtc:10549
Change-Id: I03216718156843e345f8d0a76258a15f1a355fbe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135104
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27840}
 2019-05-03 10:31:35 +00:00
Henrik Lundin
44125faba5 Reland "Piping audio interruption metrics to API layer" 
The metrics are now added as RTCNonStandardStatsMember objects in
RTCMediaStreamTrackStats. Unit tests are updated.

This is a reland of https://webrtc-review.googlesource.com/c/src/+/134303,
with fixes.

TBR=kwiberg@webrtc.org

Bug: webrtc:10549
Change-Id: I29dcc6fbfc69156715664e71acfa054c1b2d9038
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134500
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27806}
 2019-04-29 15:39:50 +00:00
Henrik Andreassson
fc02a793c2 Revert "Piping audio interruption metrics to API layer" 
This reverts commit 299c4e6846.

Reason for revert: https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Win10%20Tester/2753

../../chrome/browser/media/webrtc/webrtc_browsertest_base.cc(539): error: Expected equality of these values:
  "ok-got-stats"
  ExecuteJavascript("verifyLegacyStatsGenerated()", tab)
    Which is: "Test failed: Error: \"googInterruptionCount\" is not a whitelisted stat. Exposing new metrics in the legacy getStats() API is not allowed. Please follow the standardization process: https://docs.google.com/document/d/1q1CJVUqJ6YW9NNRc0tENkLNny8AHrKZfqjy3SL89zjc/edit?usp=sharing\n    at failTest (http://127.0.0.1:50650/webrtc/test_functions.js:46:15)\n    at http://127.0.0.1:50650/webrtc/peerconnection.js:481:19"
With diff:
@@ -1,1 +1,3 @@
-ok-got-stats
+Test failed: Error: \"googInterruptionCount\" is not a whitelisted stat. Exposing new metrics in the legacy getStats() API is not allowed. Please follow the standardization process: https://docs.google.com/document/d/1q1CJVUqJ6YW9NNRc0tENkLNny8AHrKZfqjy3SL89zjc/edit?usp=sharing
+    at failTest (http://127.0.0.1:50650/webrtc/test_functions.js:46:15)
+    at http://127.0.0.1:50650/webrtc/peerconnection.js:481:19

Original change's description:
> Piping audio interruption metrics to API layer
>
> Bug: webrtc:10549
> Change-Id: Ie6abe5819c5b73dc5f5f89bdc375bad77f44ce97
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134303
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27788}

TBR=henrik.lundin@webrtc.org,kwiberg@webrtc.org,ivoc@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:10549
Change-Id: I345306ba9758c0a3b1597724fa860d3e3fdb8995
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134464
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27802}
 2019-04-29 11:23:16 +00:00
Henrik Lundin
299c4e6846 Piping audio interruption metrics to API layer 
Bug: webrtc:10549
Change-Id: Ie6abe5819c5b73dc5f5f89bdc375bad77f44ce97
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134303
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27788}
 2019-04-26 13:32:34 +00:00
Henrik Lundin
2a8bd090a3 NetEq: Create an audio interruption metric 
This CL adds a new metric to NetEq, which logs whenever a loss
concealment event has lasted longer than 150 ms (an "interruption").
The number of such events, as well as the sum length of them, is kept
in a SampleCounter, which can be queried at any time.

Any initial PLC at the beginning of a call, before the first packet is
decoded, is ignored.

Unit tests and piping to neteq_rtpplay are included.

Bug: webrtc:10549
Change-Id: I8a224a34254c47c74317617f420f6de997232d88
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132796
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27781}
 2019-04-26 09:48:05 +00:00
Ivo Creusen
55de08e7ef Restructure neteq_rtpplay into a library with small executable wrapper. 
Most of the code in neteq_rtpplay is moved into a factory class for
NetEqTest. The factory method takes the same argc and argv arguments as
neteq_rtpplay.
This CL also adds a small public API for neteq_test to allow easy
integration into external software.

Bug: webrtc:9667
Change-Id: I5241c1f51736cb6fbe47b0ad25f4bc83dabd727d
Reviewed-on: https://webrtc-review.googlesource.com/96100
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24531}
 2018-09-03 10:42:40 +00:00