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Bug: webrtc:10173 Change-Id: Icf22901824fc85cc390e9450c480d3b7a728dc34 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165385 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30194}
Bug: webrtc:11152 Change-Id: Ic50f2dc49ca420b3406d4dea11ed20328aa59136 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165382 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30195}
…et rate" This is a reland of 63db770 that was broken as I flipped != and == :( Luckily this made a test flaky, and hence was the original change reverted. Original change's description: > Add field trial to base stable target rate on loss based target rate > > I.e not the pushback_rate that includes the congestion window pushback > (if enabled). > > Bug: None > Change-Id: I413d011004a95da03dd62f5b423abc3c8b66b333 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165383 > Commit-Queue: Jonas Oreland <jonaso@webrtc.org> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#30189} Bug: None Change-Id: Ia637d0498e6c0c2708eba659e2a30f3235944d4c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165391 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30196}
Tbr: henrik.lundin Bug: webrtc:11270 Change-Id: I1b3ad0afe3f5072ea4529e89729b087a4bd29fec Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165396 Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30197}
Split out of https://webrtc-review.googlesource.com/c/src/+/165389. I disentangled the plottable counter printer from the perf result printer so it will work for both future implementations of the perf test JSON writers. The only thing plottable counters and the results writer had in common was that both wrote JSON anyway. Bug: chromium:1029452 Change-Id: I041c3096641eda42542e8d994b246eb313940b4b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165397 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30198}
The current camera switch API sequentially cycles through each camera name for each method invocation. This policy provides reasonable behavior for devices with 2 or 3 cameras, but presents challenges with devices that contain several cameras. For example in a scenario where the current camera is oriented on the same side as the next camera name, a developer would need to call switchCamera multiple times to capture from a camera oriented on a different side of the device. This commit allows a developer to specify a camera name when switching cameras. This flexibility allows developers to have more control over which device they switch to in cases where a device contains several cameras. Bug: webrtc:11261 Change-Id: I93d46d70b2c7cf735a411a4ef4f33e926bf3a5ea Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165040 Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> Commit-Queue: Sami Kalliomäki <sakal@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30199}
Change log: https://chromium.googlesource.com/chromium/src/+log/efba8d2927..b57c714b1d Full diff: https://chromium.googlesource.com/chromium/src/+/efba8d2927..b57c714b1d Changed dependencies * src/third_party/android_build_tools/aapt2: by7YdhjwRQYtrv0Q_q_fPsqptrm5ib-SXmiNfgJYp50C..TM6ESkOFwhdEwjsIxbY3m6j7BIhg8mpY_X9Pg0nwb1AC * src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/52175631d3..85c4a438f6 DEPS diff: https://chromium.googlesource.com/chromium/src/+/efba8d2927..b57c714b1d/DEPS No update to Clang. TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com, BUG=None Change-Id: I220738c4f274949f951a392c12bd1b42671903da Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165540 Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com> Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com> Cr-Commit-Position: refs/heads/master@{#30200}
Bug: webrtc:11237 Change-Id: I83360b2608a58c7ab9f0cb050aa289df178eb66f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165560 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Joe Chen <jsphchn@google.com> Cr-Commit-Position: refs/heads/master@{#30201}
This change improves performance under high load by processing all pending tasks each time the thread is woken up by libevent. Additionally, the pipe used to wake up the TaskQueue thread now not be written to if there's already a pending write on the pipe. This fixes a bug where under high load the pipe write buffer can fill and cause tasks to get dropped. Bug: webrtc:11259, webrtc:8876 Change-Id: Ic82978c71bf9e9a25f281ca4775d46168d161d4e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165420 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30202}
Change log: https://chromium.googlesource.com/chromium/src/+log/b57c714b1d..d63380b813 Full diff: https://chromium.googlesource.com/chromium/src/+/b57c714b1d..d63380b813 Changed dependencies * src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/85c4a438f6..ae4bbcda1a * src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/081c5b5979..797d74a266 DEPS diff: https://chromium.googlesource.com/chromium/src/+/b57c714b1d..d63380b813/DEPS No update to Clang. TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com, BUG=None Change-Id: I48bd305729082e7b4ea053a42ec710c1ec28042f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165620 Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com> Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com> Cr-Commit-Position: refs/heads/master@{#30203}
This should be done according to the C++ style guide. Bug: none Change-Id: I3f8d36339bbc7175bd67631e38820b5883e875d5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165386 Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30204}
Change log: https://chromium.googlesource.com/chromium/src/+log/d63380b813..aa827d6534 Full diff: https://chromium.googlesource.com/chromium/src/+/d63380b813..aa827d6534 Changed dependencies * src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/ae4bbcda1a..ce9e11f024 * src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/797d74a266..12f8d69f12 DEPS diff: https://chromium.googlesource.com/chromium/src/+/d63380b813..aa827d6534/DEPS No update to Clang. TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com, BUG=None Change-Id: I899224e29d8727fb1a73b6782d1b1e2e3e0e9608 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165641 Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com> Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com> Cr-Commit-Position: refs/heads/master@{#30205}
Bug: None Change-Id: Idbee56e8c6ed25fb90b2456c243e30ef72a0b68d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165642 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30206}
Change log: https://chromium.googlesource.com/chromium/src/+log/aa827d6534..54a7cb4bda Full diff: https://chromium.googlesource.com/chromium/src/+/aa827d6534..54a7cb4bda Changed dependencies * src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/ce9e11f024..9dbcda8385 * src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/12f8d69f12..13928b7e7f DEPS diff: https://chromium.googlesource.com/chromium/src/+/aa827d6534..54a7cb4bda/DEPS No update to Clang. TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com, BUG=None Change-Id: Ib03d9edc7e52303c9c6c01e566940c05e7f2a010 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165662 Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com> Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com> Cr-Commit-Position: refs/heads/master@{#30207}
This will make RESULT lines still come out after we add a second JSON writer implementation. Bug: chromium:1029452 Change-Id: I5cba3151c21df2901f19305e9b71bc5c9638a0ae Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165399 Reviewed-by: Artem Titov <titovartem@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30208}
Bug: webrtc:11268 Change-Id: I764eb37a386075838e981c6d5b820e25d77f1a80 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165395 Commit-Queue: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30209}
…rame_types parameter If in simulcast case some streams are disabled (especially the first one), the key-frame requests might be ignorred by e.g. libvpx vp8 encoder wrapper. Before this CL SimulcastEncoderAdapter always passes single frame type in Encode() call. However, if underlying encoder used simulcast, it would've expected as many frame types as there are streams. Bug: none Change-Id: I7f56a6540b67273b7d3cf9fa86dc76015b92d271 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165681 Reviewed-by: Evan Shrubsole <eshr@google.com> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30210}
Long term goal is to use the VideoStreamDecoder in the VideoReceiveStream so that we can stop using legacy VideoCodingModule components and classes. This CL is one of several in preparation for that. Bug: webrtc:7408, webrtc:9378 Change-Id: Ifd7e4c3c7d38dbb7c4b0636aaad318c571a29158 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164525 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30211}
Bug: webrtc:11152 Change-Id: Iab4975e9f378b177a2abf34559f9b74752e69843 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165582 Reviewed-by: Markus Handell <handellm@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30212}
…efault values." This reverts commit bcbdeed. Reason for revert: Speculative revert after a perf regression. Original change's description: > In RtpBitrateConfigurator ignore new parameters when set to default values. > > Bug: webrtc:11263 > Change-Id: Ia7539c7c142b059d0295849b916439bb647f112d > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162207 > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#30191} TBR=danilchap@webrtc.org,srte@webrtc.org # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:11263 Change-Id: I17804655465b27523c462d2aba44519c820b8e04 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165687 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30213}
Some messages were processed after involved objects were destructed, a.k.a. 'use after free'. This CL fixes that by disconnecting signals before fixture destruction, honoring CreateChannel/DestroyChannel symmetry and following what is done in similar test cases. Bug: webrtc:11269 Change-Id: I122aca70a9978b752edc01e5f31583f4425f3624 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165685 Reviewed-by: Qingsi Wang <qingsi@webrtc.org> Commit-Queue: Yves Gerey <yvesg@google.com> Cr-Commit-Position: refs/heads/master@{#30214}
Change log: https://chromium.googlesource.com/chromium/src/+log/54a7cb4bda..bd2395cd43 Full diff: https://chromium.googlesource.com/chromium/src/+/54a7cb4bda..bd2395cd43 Changed dependency * src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/9dbcda8385..f6f813d450 DEPS diff: https://chromium.googlesource.com/chromium/src/+/54a7cb4bda..bd2395cd43/DEPS No update to Clang. TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com, BUG=None Change-Id: I50405f17a60be878e906f03e05605b5581f70578 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165666 Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com> Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com> Cr-Commit-Position: refs/heads/master@{#30215}
This is part of a CL series merging rtc::MessageQueue into rtc::Thread. Bug: webrtc:9883 Change-Id: I4a1bcd44c9523b6402b3f05b50597bdc2e6615e3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165345 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30216}
Given that we already have Thread:.Invoke that can be used with lambda, SynchronousMethodCall doesn't add any value. This simplification prepares for simulated time peer connection tests. Bug: webrtc:11255 Change-Id: I478a11f15e30e009dae4a3fee2120f6d7a03355f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165683 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30217}
Bug: None Change-Id: I572b65107797da8494f1956ab0a08a3221be4bb7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165002 Commit-Queue: Seth Hampson <shampson@webrtc.org> Reviewed-by: Taylor <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30218}
…decs""" This is a reland of 4e64e60 This CL lands all code except the code that activates the change, see media/engine/webrtc_video_engine.cc Once downstream projects are fixed, there will be a one-line change to activate the change to distinguish between send and receive video codecs. Original change's description: > Reland "Reland "Distinguish between send and receive video codecs"" > > This is a reland of 77eb338 > > Original change's description: > > Reland "Distinguish between send and receive video codecs" > > > > This reverts commit f2d6fe6. > > > > Reason for revert: Downstream test updated. > > > > Original change's description: > > > Revert "Reland "Distinguish between send and receive video codecs"" > > > > > > This reverts commit 26e6afe. > > > > > > Reason for revert: Breaks another downstream test. > > > > > > Original change's description: > > > > Reland "Distinguish between send and receive video codecs" > > > > > > > > This reverts commit f22af3c. > > > > > > > > Reason for revert: Downstream tests have been updated. > > > > > > > > Original change's description: > > > > > Revert "Distinguish between send and receive video codecs" > > > > > > > > > > This reverts commit 18314bd. > > > > > > > > > > Reason for revert: Breaks downstream test. > > > > > > > > > > Original change's description: > > > > > > Distinguish between send and receive video codecs > > > > > > > > > > > > Even though send and receive codecs are the same, > > > > > > they might have different support in HW. > > > > > > Distinguish between send and receive codecs to be able to keep > > > > > > track of which codecs have HW support. > > > > > > > > > > > > Bug: chromium:1029737 > > > > > > Change-Id: I16a80da44c5061ca42f2aabda76e6bf0b879bf7b > > > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161306 > > > > > > Reviewed-by: Anders Carlsson <andersc@webrtc.org> > > > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > > > > > Commit-Queue: Johannes Kron <kron@webrtc.org> > > > > > > Cr-Commit-Position: refs/heads/master@{#30041} > > > > > > > > > > TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org > > > > > > > > > > Change-Id: I7e5807460006db613e9b3b369ec6036b88f164fd > > > > > No-Presubmit: true > > > > > No-Tree-Checks: true > > > > > No-Try: true > > > > > Bug: chromium:1029737 > > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161662 > > > > > Reviewed-by: Johannes Kron <kron@webrtc.org> > > > > > Commit-Queue: Johannes Kron <kron@webrtc.org> > > > > > Cr-Commit-Position: refs/heads/master@{#30042} > > > > > > > > TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org > > > > > > > > # Not skipping CQ checks because original CL landed > 1 day ago. > > > > > > > > Bug: chromium:1029737 > > > > Change-Id: Ia70b11376b43888e2495ef21838c2d2e3c68d735 > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161734 > > > > Reviewed-by: Johannes Kron <kron@webrtc.org> > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > > > Reviewed-by: Anders Carlsson <andersc@webrtc.org> > > > > Commit-Queue: Johannes Kron <kron@webrtc.org> > > > > Cr-Commit-Position: refs/heads/master@{#30078} > > > > > > TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org > > > > > > Change-Id: Ia4971b898c9209a3736a916a1c2c48d392dfdad6 > > > No-Presubmit: true > > > No-Tree-Checks: true > > > No-Try: true > > > Bug: chromium:1029737 > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162140 > > > Reviewed-by: Johannes Kron <kron@webrtc.org> > > > Commit-Queue: Johannes Kron <kron@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#30079} > > > > TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org > > > > # Not skipping CQ checks because original CL landed > 1 day ago. > > > > Bug: chromium:1029737 > > Change-Id: If2c3c5b5e7d86cb852a1f20f02b6ceae62b2e0c8 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162186 > > Reviewed-by: Johannes Kron <kron@webrtc.org> > > Reviewed-by: Anders Carlsson <andersc@webrtc.org> > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > Commit-Queue: Johannes Kron <kron@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#30097} > > Bug: chromium:1029737 > Change-Id: I5912822df8169fbb3097c0f440f7924527fa950b > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162483 > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Reviewed-by: Anders Carlsson <andersc@webrtc.org> > Commit-Queue: Johannes Kron <kron@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#30120} Bug: chromium:1029737 Change-Id: Id4f1c6f6f0cf7b96fe93dd22d14310d286af31f3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165682 Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Anders Carlsson <andersc@webrtc.org> Commit-Queue: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30219}
Change log: https://chromium.googlesource.com/chromium/src/+log/bd2395cd43..d794106d9d Full diff: https://chromium.googlesource.com/chromium/src/+/bd2395cd43..d794106d9d Changed dependencies * src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/f6f813d450..32c9791b8a * src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/13928b7e7f..fc132e61db DEPS diff: https://chromium.googlesource.com/chromium/src/+/bd2395cd43..d794106d9d/DEPS No update to Clang. TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com, BUG=None Change-Id: I6cc34f75c049bc75a92eddaf00e6dc0694d64837 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165669 Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com> Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com> Cr-Commit-Position: refs/heads/master@{#30220}
This way we can rely on existing task scheduling and execution functionality, reducing the required functionality to support the fake socket server. This prepares for support simulated time execution of peer connection level tests. Bug: webrtc:11255 Change-Id: I7de64a099c2e355c70929ecff79b8ea3b98b70b8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165398 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30221}
Change log: https://chromium.googlesource.com/chromium/src/+log/d794106d9d..b581de5b1b Full diff: https://chromium.googlesource.com/chromium/src/+/d794106d9d..b581de5b1b Changed dependencies * src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/32c9791b8a..71813e2ccf * src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/fc132e61db..7a8bf94894 * src/third_party/sqlite4java: 889660698187baa7c8b0d79f7bf58563125fbd66..LofjKH9dgXIAJhRYCPQlMFywSwxYimrfDeBmaHc-Z5EC DEPS diff: https://chromium.googlesource.com/chromium/src/+/d794106d9d..b581de5b1b/DEPS No update to Clang. TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com, BUG=None Change-Id: I7c06ddf990c474892f71ef81e45d1520b8798e6f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165730 Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com> Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com> Cr-Commit-Position: refs/heads/master@{#30222}
BUG: webrtc:11100 Change-Id: I37a23b32b339c000cc2e88793c31732f7f1d259d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165686 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30223}
A previous refactoring introduced an issues in SimulatedProcessThread causing stalls when task are posted. This CL fixes this and cleans up the code to make it easier to see that it's correct. Bug: webrtc:11255 Change-Id: I33d7daa993ad2a4cfe2b63f674692455c2e09d05 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167380 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Ali Tofigh <alito@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30429}
As part of this, we also use TaskQueue and RepeatedTask rather than rtc::Thread + rtc::MessageHandler. With the ultimate goal of deprecating rtc::Thread. Bug: webrtc:9883 Change-Id: I2fb851ac31ee2431435d51de78ff446572512201 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167528 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30430}
This reverts commit 0e96535. Reason for revert: Downstream test failure Original change's description: > Inlines NullAudioPoller functionality into AudioState class. > > As part of this, we also use TaskQueue and RepeatedTask rather > than rtc::Thread + rtc::MessageHandler. With the ultimate goal of > deprecating rtc::Thread. > > Bug: webrtc:9883 > Change-Id: I2fb851ac31ee2431435d51de78ff446572512201 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167528 > Commit-Queue: Sebastian Jansson <srte@webrtc.org> > Reviewed-by: Sam Zackrisson <saza@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#30430} TBR=saza@webrtc.org,srte@webrtc.org Change-Id: I4c77259f7b6477fc1cb79350f2d47817f106770d No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:9883 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168046 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30431}
…ld trial. Bug: webrtc:10274 Change-Id: I94a8c200947c66277d67812bc1d0acc9e1f40e7a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168045 Commit-Queue: Björn Terelius <terelius@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30432}
…f references in gof number of references can't be invalid if gof was correctly parsed from a vp9 packet, but RtpFrameReferenceFinder still better be protected from the invalid data. (cherry picked from commit a118702) Bug: chromium:1048013 Change-Id: I548f5c87199421b7736409cbcacbec760ad799ae Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168124 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Original-Commit-Position: refs/heads/master@{#30444} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168241 Cr-Commit-Position: refs/branch-heads/4044@{signalapp#1} Cr-Branched-From: be99ee8-refs/heads/master@{#30432}
…order (cherry picked from commit 72859e5) Bug: webrtc:11319, chromium:1049539 Change-Id: If63db02d282ee622c12405f85c0fbae1ba13fcb2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168196 Reviewed-by: Florent Castelli <orphis@webrtc.org> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Original-Commit-Position: refs/heads/master@{#30459} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168301 Cr-Commit-Position: refs/branch-heads/4044@{signalapp#2} Cr-Branched-From: be99ee8-refs/heads/master@{#30432}
…nceFinder (cherry picked from commit ef0d76a) Bug: chromium:1049129 Change-Id: I133673d86aadd6a87b3420a04bbf45ed53841a96 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168240 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Original-Commit-Position: refs/heads/master@{#30466} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168300 Cr-Commit-Position: refs/branch-heads/4044@{signalapp#3} Cr-Branched-From: be99ee8-refs/heads/master@{#30432}
…nnects. This fixes a bug where transport_overhead_bytes_per_packet_ is sometimes not set and therefore not included in the BWE. (cherry picked from commit b4cdd62) Bug: webrtc:11359, chromium:1053421 Change-Id: Id3593299c6bcd7ce3c44a7b148c202240b3f1981 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168525 Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org> Cr-Original-Commit-Position: refs/heads/master@{#30522} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168723 Cr-Commit-Position: refs/branch-heads/4044@{signalapp#4} Cr-Branched-From: be99ee8-refs/heads/master@{#30432}
…solution If e.g. CPU adaptation reduces input video size too much, video pipeline would reduce the number of used simulcast streams/spatial layers. This may result in disabled video if some streams are disabled by Rtp encoding parameters API. (cherry picked from commit 03d9096) Bug: webrtc:11319, chromium:1052313 Change-Id: Id7f157255599dcb6f494129b83477cda4bea982a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168480 Reviewed-by: Evan Shrubsole <eshr@google.com> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Original-Commit-Position: refs/heads/master@{#30498} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168600 Cr-Commit-Position: refs/branch-heads/4044@{signalapp#5} Cr-Branched-From: be99ee8-refs/heads/master@{#30432}
This CL avoids the head-allocations done in a sum of the squared values in a nested vector. (cherry picked from commit 0618cbc) No-Try: True TBR: saza@webrtc.org Bug: webrtc:11361, chromium:1052086 Change-Id: I698b855bdd54df2147ef3b6d5e3d401401228d76 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168543 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Per Åhgren <peah@webrtc.org> Cr-Original-Commit-Position: refs/heads/master@{#30520} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168965 Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/branch-heads/4044@{signalapp#6} Cr-Branched-From: be99ee8-refs/heads/master@{#30432}
…layers No-try, because IOS bots are broken on release branch. (cherry picked from commit 1dea1ea) No-Try: True Bug: chromium:1051476 Change-Id: Iaf2b6ab6640cd314a620dbdf1547d8f1b2f40693 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168921 Reviewed-by: Evan Shrubsole <eshr@google.com> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Original-Commit-Position: refs/heads/master@{#30581} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168923 Cr-Commit-Position: refs/branch-heads/4044@{signalapp#7} Cr-Branched-From: be99ee8-refs/heads/master@{#30432}
Loosen the restrictions for ice-char by allowing '-' and '='. Being spec-compliant breaks interoperability. The spec-behaviour will be restored with a notice period. BUG=chromium:1053756,chromium:1044521 (cherry picked from commit 48e849f) No-Try: True Change-Id: I880babd0869302bd713912ddfcfa48866fad32c1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168820 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Original-Commit-Position: refs/heads/master@{#30560} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169663 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/branch-heads/4044@{signalapp#8} Cr-Branched-From: be99ee8-refs/heads/master@{#30432}
Loosen the restrictions for ice-char by also allowing '#' (known to break) and '_' (urlsafe base64) in addition to the existing exceptions for '-' and '='. Also fixes typo in log message. BUG=chromium:1053756 (cherry picked from commit 98d5bbb) Change-Id: I8f254a7c25f780276452fa3e27245b6b7ad1a3ce Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168943 Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Steve Anton <steveanton@webrtc.org> Cr-Original-Commit-Position: refs/heads/master@{#30596} No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169664 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/branch-heads/4044@{signalapp#9} Cr-Branched-From: be99ee8-refs/heads/master@{#30432}
This CL corrects the temporary buffers size in the pre-processing of the capture audio before encoding. As part of this it removes the ACM-specific hardcoding of the size and instead ensures that the size of the temporary buffer matches that of the AudioFrame. (cherry picked from commit d82a02c) No-Try: True TBR: kwiberg@webrtc.org Bug: webrtc:11242, chromium:1060647 Change-Id: I56dd6cadfd4e140e8e159966c33d1027383ea9fa Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170340 Commit-Queue: Per Åhgren <peah@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Cr-Original-Commit-Position: refs/heads/master@{#30775} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170780 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/branch-heads/4044@{signalapp#10} Cr-Branched-From: be99ee8-refs/heads/master@{#30432}
peter-signal
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Jul 18, 2020
…ors on key-frames TBR=brandtr@webrtc.org (cherry picked from commit 35fc153) Bug: webrtc:11575, chromium:1084963 Change-Id: I09be17ab5155e9f610c8f7c451ca52d7d65e24d1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175222 Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Cr-Original-Commit-Position: refs/heads/master@{#31295} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175902 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/branch-heads/4147@{#1} Cr-Branched-From: 2b7d969-refs/heads/master@{#31262}
peter-signal
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Apr 16, 2021
Some clients will not count audio packets into the bandwidth estimate despite negotiating e.g. abs-send-time for that SSRC. If padding is sent on such an RTP module, we might get stuck in a low resolution. This CL works around that by preferring to send padding on video SSRCs. Originally reviewed on: https://webrtc-review.googlesource.com/c/src/+/161941 (cherry picked from commit 1e51a38) Bug: webrtc:11196, chromium:1033411 Change-Id: I04efb8caeafd856bbd71bdc1e305b3dad270930c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162180 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/branch-heads/3987@{#1} Cr-Branched-From: 1256d9b-refs/heads/master@{#30022}
peter-signal
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…the tails. (cherry picked from commit 5eb5bb5) Bug: chromium:1249867 Change-Id: Ic469f6226fe079c306cec6f941eeb70d6d9094f3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231682 Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Original-Commit-Position: refs/heads/main@{#34966} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232322 Cr-Commit-Position: refs/branch-heads/4638@{#1} Cr-Branched-From: fb50179-refs/heads/main@{#34960}
rashad-signal
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This interface will be implemented by P2PTransportChannel in a follow-up CL. It will allow an ICE controller to request actions to manipulate the connection used by the transport. Bug: webrtc:14367, webrtc:1413 Change-Id: I5cd171bd09c8dfc88588f8fc06e87d74a90b5216 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271290 Reviewed-by: Jonas Oreland <jonaso@webrtc.org> Commit-Queue: Sameer Vijaykar <samvi@google.com> Cr-Commit-Position: refs/heads/main@{#38062}
inaqui-signal
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Jul 24, 2023
(cherry picked from commit eec1810) Bug: chromium:1454086 Change-Id: I39573b706c4031d091c45a182b13cb3b2dba6233 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/309920 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Original-Commit-Position: refs/heads/main@{#40332} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/310920 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/branch-heads/5845@{#1} Cr-Branched-From: f80cf81-refs/heads/main@{#40319}
inaqui-signal
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The FrameCadenceAdapter starts a delayed task to request a new refresh frame on receiving frame drop. However, the resulting RepeatingTaskHandle was not Stop()ed on destruction, leading to UAF. (cherry picked from commit fb98b01) Fixed: chromium:1478944 Change-Id: Iba441420953e989cfc7fcfd2f358b5b30f375786 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320200 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Original-Commit-Position: refs/heads/main@{#40747} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320420 Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/branch-heads/5993@{#1} Cr-Branched-From: 5afcec0-refs/heads/main@{#40703}
jim-signal
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Jan 17, 2024
This CL is a follow-up of work done in https://webrtc-review.googlesource.com/c/src/+/323882 where the goal was to reduce the amount of FrameDropped error logs in WebRTC.DesktopCapture.Win.WgcCaptureSessionGetFrameResult. The previous work avoids FrameDropped logs for a minimized window being captured with WGC but we still se a large amount of these error (or rather warning) logs. See [1] which comes from Canary. This CL does two different things to improve the situation: 1) It adds kFramePoolEmpty to the existing GetFrameResult::kFrameDropped enum to point out that the warning comes from the frame pool not being able to return a valid new frame. It also makes it more clear that it does not cause an outer/final error as WgcCapturerResult::kFrameDropped. We still keep the inner GetFrameResult::kFrameDropped but it is only produced when the frame pool returns NULL and our external queue is empty. Hence, a real frame-drop error. Note that, it is still easy to provoke kFramePoolEmpty simply by asking for a high resolution at a high rate. The example in [2] comes from a 4K screen @30fps. Hence, we have not fixed anything yet. 2) It also increases the size of the internal frame pool from 1 to 2. This does lead to an almost zero rate of kFramePoolEmpt warnings at the expense of a slightly reduced max capture rate. BUT, with 1 as size, we can "see" a higher max capture rate but it is not a true rate since it comes with a high rate of kFramePoolEmpty errors. Hence, we "emulate" a high capture rate by simply feeding copies of the last frame that we had stored in the external queue. Using 2 leads to a more "true" rate of what we actually can capture in terms of *new* frames and also a substantially lower rate of kFramePoolEmpty. In addition, with 1 as size, if we ask at a too high rate and provide a copy of the last frame, our CPU adaptation will not reduce its rate since we think that things are OK when it is actually not. Also, the samples in [3] and [4] both use 2 as numberOfBuffers as well. Let me also mention that with this small change, I a have not been able to provoke any kFramePoolEmpty error messages. Finally, geDisplayMedia can be called called with constraints where min and max framerate is defined. The mechanism which maintains the min rate is implemented via the RequestRefreshFrame API and it can be sent to the source (DesktopCaptureDevice) back to back with a previous timer interrupt for a capture request. Without this change, these RRFs were also a source of a large amount of kFramePoolEmpty error logs. With 2 buffers instead; this is no longer the case. [1] https://screenshot.googleplex.com/7sfv6HdGXLwyxdj [2] https://paste.googleplex.com/4795680001359872 [3] https://github.com/robmikh/Win32CaptureSample/blob/master/Win32CaptureSample/SimpleCapture.cpp [4] https://learn.microsoft.com/en-us/windows/uwp/audio-video-camera/screen-capture#add-the-screen-capture-capability (cherry picked from commit 4be5927) Bug: chromium:1314868 Change-Id: I73b823b31a993fd2cd6e007b212826dfe1a80012 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325521 Commit-Queue: Alexander Cooper <alcooper@chromium.org> Reviewed-by: Alexander Cooper <alcooper@chromium.org> Cr-Original-Commit-Position: refs/heads/main@{#41079} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326960 Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Cr-Commit-Position: refs/branch-heads/6099@{#1} Cr-Branched-From: 507f1cc-refs/heads/main@{#41042}
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If SVC is used, the minimum bitrate would be 30kbps, instead of 49, as configured in svc_config.h, because the overall stream will get min_bitrate from the default VP8 simulcast configuration, and this 30kbps will be allocated to the stream for svc_rate_allocator to divide between layers. However, with the configuration before this change, 49kbps would be always allocated to the lowest simulcast stream. (cherry picked from commit f49a826) Bug: webrtc:15852, chromium:330672089 Change-Id: I1c77f45654af8850180a83f8e3f4428cc42d086e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343760 Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Cr-Original-Commit-Position: refs/heads/main@{#41940} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343981 Cr-Commit-Position: refs/branch-heads/6367@{#1} Cr-Branched-From: 802552a-refs/heads/main@{#41921}
jim-signal
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(cherry picked from commit 74a4038) Bug: chromium:325284120 Change-Id: Iea0aea0a17bb0b1f73b3c1cbd408b7a6cd2b216e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/340180 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Original-Commit-Position: refs/heads/main@{#41776} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/340600 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/branch-heads/6312@{#1} Cr-Branched-From: 0355f45-refs/heads/main@{#41763}
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Jun 25, 2024
Use SPA_CHUNK_FLAG_CORRUPTED and SPA_META_HEADER_FLAG_CORRUPTED flags to determine corrupted buffers or corrupted buffer data. We used to only rely on compositors setting chunk->size, but this doesn't make sense for dmabufs where they have to make up arbitrary values. It also looks this is not reliable and can cause glitches as we end up processing corrupted buffers. (cherry picked from commit cfbd6b0) Bug: chromium:341928670 Change-Id: Ida0c6a5e7a37e19598c6d5884726200f81b94962 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349881 Commit-Queue: Mark Foltz <mfoltz@chromium.org> Reviewed-by: Mark Foltz <mfoltz@chromium.org> Reviewed-by: Alexander Cooper <alcooper@chromium.org> Cr-Original-Commit-Position: refs/heads/main@{#42292} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351563 Commit-Queue: Alexander Cooper <alcooper@chromium.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Cr-Commit-Position: refs/branch-heads/6478@{#1} Cr-Branched-From: 16fb790-refs/heads/main@{#42290}
jim-signal
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Sep 5, 2024
The new version of MSan (rolled by [1]) detects the following:
```
==39908==WARNING: MemorySanitizer: use-of-uninitialized-value
#0 0x5591400a52ef in GetPlayoutDelayMs ./../../modules/audio_coding/neteq/decision_logic.cc:466:35
#1 0x5591400a52ef in webrtc::DecisionLogic::ExpectedPacketAvailable(webrtc::NetEqController::NetEqStatus) ./../../modules/audio_coding/neteq/decision_logic.cc:311:36
#2 0x5591400a39e9 in webrtc::DecisionLogic::GetDecision(webrtc::NetEqController::NetEqStatus const&, bool*) ./../../modules/audio_coding/neteq/decision_logic.cc:0:0
#3 0x55913cf590c9 in webrtc::DecisionLogicTest_PreemptiveExpand_Test::TestBody() ./../../modules/audio_coding/neteq/decision_logic_unittest.cc:139:3
#4 0x55913ef28283 in HandleExceptionsInMethodIfSupported<testing::Test, void> ./../../third_party/googletest/src/googletest/src/gtest.cc:0:3
#5 0x55913ef28283 in testing::Test::Run() ./../../third_party/googletest/src/googletest/src/gtest.cc:2710:5
#6 0x55913ef2ab46 in testing::TestInfo::Run() ./../../third_party/googletest/src/googletest/src/gtest.cc:2856:11
#7 0x55913ef2da34 in testing::TestSuite::Run() ./../../third_party/googletest/src/googletest/src/gtest.cc:3034:30
#8 0x55913ef621e8 in testing::internal::UnitTestImpl::RunAllTests() ./../../third_party/googletest/src/googletest/src/gtest.cc:5964:44
#9 0x55913ef60f54 in HandleExceptionsInMethodIfSupported<testing::internal::UnitTestImpl, bool> ./../../third_party/googletest/src/googletest/src/gtest.cc:0:0
#10 0x55913ef60f54 in testing::UnitTest::Run() ./../../third_party/googletest/src/googletest/src/gtest.cc:5543:10
#11 0x55913ee1a944 in RUN_ALL_TESTS ./../../third_party/googletest/src/googletest/include/gtest/gtest.h:2334:73
#12 0x55913ee1a944 in webrtc::(anonymous namespace)::TestMainImpl::Run(int, char**) ./../../test/test_main_lib.cc:203:21
#13 0x55913cbd36b8 in main ./../../test/test_main.cc:72:16
#14 0x7fdb18c73082 in __libc_start_main /build/glibc-LcI20x/glibc-2.31/csu/../csu/libc-start.c:308:16
#15 0x55913cb3e1a9 in _start ??:0:0
```
[1] - https://webrtc-review.googlesource.com/c/src/+/353620
Bug: b/344970813
Change-Id: I9b5d7791e68b4c494168ba9f007a3099ae21fed4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353581
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42433}
jim-signal
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that referenced
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Sep 5, 2024
M128 merge approval: https://issues.chromium.org/issues/354143228#comment11 This reverts commit 46b43e0. Reason for revert: chromium:354143228 Original change's description: > Update support for missing HIGH profiles and 1080p > > The High and ConstrainedHigh profiles are missing from the decoder capabilities. Also level 3.1 doesn't allow 1080p > > Bug: webrtc:347724928 > Change-Id: I3f33468327d2aaf352fc80f69d2ee31481bafcb5 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355001 > Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> > Reviewed-by: Åsa Persson <asapersson@webrtc.org> > Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#42528} (cherry picked from commit 12f9d5c) Bug: chromium:354143228, webrtc:347724928 Change-Id: I4d55b2982aca2e94ec983473336c4fa2a72d842f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357861 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Cr-Original-Commit-Position: refs/heads/main@{#42675} No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358021 Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/branch-heads/6613@{#1} Cr-Branched-From: 1ac162e-refs/heads/main@{#42664}
jim-signal
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that referenced
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Oct 24, 2024
Merge approval: https://g-issues.chromium.org/issues/367752722#comment5 (cherry picked from commit e81ba3089755e88292c135733ea187fdd278d858) Bug: chromium:328598314, chromium:367752722 Change-Id: I132b4c30f132ace2bbef6359edd994c1ad75c9ad Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362620 Reviewed-by: Johannes Kron <kron@webrtc.org> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Cr-Original-Commit-Position: refs/heads/main@{#43035} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362981 Commit-Queue: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/branch-heads/6723@{#1} Cr-Branched-From: 13e377b-refs/heads/main@{#43019}
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