After the removal of field_trial_default, metrics_default and
runtime_enabled_features_default, this build target doesn't build
anything and can be safely removed.
Bug: webrtc:9631
Change-Id: Iee1111e065ffefe0b4b9a695ee67a594e6d82caa
Reviewed-on: https://webrtc-review.googlesource.com/c/103702
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24976}
It will soon lose the ability to do so.
Bug: webrtc:8396
Change-Id: I1d7c62fbc2585233cf1656fdcc4bb5380c2f41a5
Reviewed-on: https://webrtc-review.googlesource.com/c/100980
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24947}
It will soon lose the ability to do so.
Bug: webrtc:8396
Change-Id: I06dce417bba855b57130bd1a052988b2f235dcbd
Reviewed-on: https://webrtc-review.googlesource.com/102882
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24921}
This CL removes some deprecated build targets (and their headers)
from system_wrappers:
- field_trial_api
- field_trial_default
- metrics_api
- metrics_default
It also refreshes all the dependencies on field_trial.h and metrics.h.
A nice side effect is that it is finally possible to remove 'nogncheck'
from the following files (when it was used with field_trial_default
and metrics_default):
- sdk/objc/api/peerconnection/RTCMetricsSampleInfo+Private.h
- sdk/android/src/jni/pc/peerconnectionfactory.cc
- sdk/objc/api/peerconnection/RTCFieldTrials.mm
Bug: webrtc:9631
Change-Id: Ib621f41ef8ad0aba4fe1c1d7e749c044afc956c3
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/100524
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24878}
It will soon lose the ability to do so.
Bug: webrtc:8396
Change-Id: I7d8e1549c44628fc9bdf2480468a0f1d3ae812f2
Reviewed-on: https://webrtc-review.googlesource.com/102062
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24853}
The tests only use the automatic mode, and I'd rather not maintain
(and test!) the rest.
Bug: webrtc:8396
Change-Id: I4cd1096e088d2ea8807a605b8448bd44ff9e88ed
Reviewed-on: https://webrtc-review.googlesource.com/102060
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24852}
It will soon lose the ability to do so.
Bug: webrtc:8396
Change-Id: Ifca101fce0c349dba8c402ab2b6ad614061a88f6
Reviewed-on: https://webrtc-review.googlesource.com/101281
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24836}
It will soon lose the ability to do so.
Bug: webrtc:8396
Change-Id: Id413204e53afec28495dff0873f027a56caed80f
Reviewed-on: https://webrtc-review.googlesource.com/101861
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24834}
It hasn't been changed in any meaningful way since 2013, the same year
it was created.
Bug: webrtc:8396
Change-Id: I5633188134f71f24311fbd3098d046632fc4ee3a
Reviewed-on: https://webrtc-review.googlesource.com/101563
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24816}
It will soon lose the ability to do so.
Also, the ACM no longer creates comfort noise encoders for us, so
don't bother testing that.
Bug: webrtc:8396
Change-Id: I24a12e26bef142f9f8e7532b764f28572e0c6ace
Reviewed-on: https://webrtc-review.googlesource.com/101640
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24803}
Replace calls to .str() which copies with .Release which moves in cases where that's safe.
This CL was generated by this command:
git grep -l 'StringBuilder' |
xargs perl -i -0 -pe "s/(rtc::StringBuilder (\S+);.*?return )\\g2.str\(\)/\$1\$2.Release\(\)/sg"
Bug: webrtc:8982
Change-Id: If4dadbeb039df010aaaa9e58da81c1971a84fe8f
Reviewed-on: https://webrtc-review.googlesource.com/100307
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24790}
Uninline RTPFragmentaion functions
fix RTPFragmentation move constructor and assign operators (was recursive for win)
replace assert with rtc::dchecked_cast
Remove unused includes and dependencies.
Fix other targets that used those includes transitively instead of directly
Bug: None
Change-Id: I647cb1eda107dc7d87d25234095545bc2842fa40
Reviewed-on: https://webrtc-review.googlesource.com/100500
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24759}
This change enables NetEq to use the packet concealment audio (aka
PLC) produced by a decoder. The change also includes a new API to the
AudioDecoder interface, which lets the decoder implementation generate
and deliver concealment audio.
Bug: webrtc:9180
Change-Id: Icaacebccf645d4694b0d2d6310f6f2c7132881c4
Reviewed-on: https://webrtc-review.googlesource.com/96340
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24738}
Currently we use the NetworkStatistics to monitor these metrics, but because these get reset on every call, this makes it impossible to use them for other purposes.
Bug: webrtc:9667
Change-Id: If648085f04d2d58aae263cff5b9491bcad373a96
Reviewed-on: https://webrtc-review.googlesource.com/99740
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24727}
It is problematic to set field trials more than once, so to avoid running into problems, this functionality has been placed in the main function of neteq_rtpplay.
Bug: webrtc:9667
Change-Id: Ib9b9990f30a1715b50889dbfc4d74787bcbe5dae
Reviewed-on: https://webrtc-review.googlesource.com/98541
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24673}
The NetEqTest class was recently refactored. In the process, the
functionality for measuring the simulation time suffered a bug. This
CL fixes it.
Bug: webrtc:9667
Change-Id: I139e697ede21584ef77ae23cfa8e77f6dac65b51
Reviewed-on: https://webrtc-review.googlesource.com/98982
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24658}
This CL adds more useful information to NetEqState, and implements setting action_times_ms, which can be used to get a better idea of what actually happened during a timestep.
Bug: webrtc:9667
Change-Id: I789a3e1ad852066fdf4e9b4c96b8fb6033dacb27
Reviewed-on: https://webrtc-review.googlesource.com/98163
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24643}
This CL replaces std::o?stringstream with rtc::StringBuilder where that's possible to do without changing any of the surrounding code. It also updates includes and build files as appropriate.
The CL was generated by running 'git grep -l -P std::o?stringstream | xargs perl -pi -e "s/std::o?stringstream/rtc::StringBuilder/g"'. Then I've manually updated the #includes and BUILD files, run 'git cl format' and unstaged any file that would need more complex fixes.
Bug: webrtc:8982
Change-Id: Ibc32153f4a3fd177e260b6ad05ce393972549357
Reviewed-on: https://webrtc-review.googlesource.com/98460
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24605}
This CL enables -Wexit-time-destructors and -Wglobal-constructors on
rtc_static_library and rtc_source_set build targets.
It also adds the possibility to suppress these warnings because
they trigger in a few places.
The long term goal is to avoid regressions on this and remove all the
suppressions.
Bug: webrtc:9693
Change-Id: I4c1ecc137ef9e87ec5e66981ce95d96fb082727c
Reviewed-on: https://webrtc-review.googlesource.com/98380
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24604}
Only user, iSACTest, refactored to use a sleep instead.
Bug: webrtc:3380
Change-Id: I683a5a05349f75a17e5d2a02d4a20a9cf059a28f
Reviewed-on: https://webrtc-review.googlesource.com/96802
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24541}
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}
LappedTransform is only used in BandwidthAdaptationTest and therefore it
should not be anymore a visible target under common_audio.
This CL moves LappedTransform and other two classes it depends on (and which
are not used elsewhere) to modules/audio_coding/codecs/opus/test.
Bug: webrtc:9577, webrtc:5298
Change-Id: I1aa8052c2df2b2b150c279c0c9b1001474aed47a
Reviewed-on: https://webrtc-review.googlesource.com/96440
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24509}
Specifically, don't expect the ACM to be able to create encoders; we
have to give it an encoder that we make ourselves.
The new way of creating encoders used a 32 kbit/s bitrate
unconditionally for iSAC; I had to change it to 32 kbit/s for 16 kHz
and 56 kbit/s for 32 kHz, which is what the old way of creating
encoders has used since forever.
I also had to change some test expectations on Opus, because the new
way defaults to 32 kbit/s for mono and 64 kbit/s for stereo (which I
believe to be correct), while the old way defaults to 64 kbit/s in
both cases.
Bug: webrtc:8396
Change-Id: I3aab944175a8e27f4c63380e822b27e839bba7f2
Reviewed-on: https://webrtc-review.googlesource.com/94540
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24375}
Specifically, don't expect the ACM to be able to create encoders; we
have to give it an encoder that we make ourselves.
Bug: webrtc:8396
Change-Id: I032b12f3813af6ac3ea0dfb688006899dffe4855
Reviewed-on: https://webrtc-review.googlesource.com/94150
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24323}
This CL also excludes several codec mappings depending on compile-time flags.
Bug: webrtc:9289
Change-Id: I1a9183f88378307925b747576a5513e54be3782e
Reviewed-on: https://webrtc-review.googlesource.com/93462
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24259}
Specifically, don't expect the ACM to be able to create encoders; we
have to give it an encoder that we make ourselves.
To make it work, I had to add support for the "ptime" parameter to the
L16 codec.
Bug: webrtc:8396
Change-Id: I3869422882611d2eed65d6c849ea7cd3ad6bd126
Reviewed-on: https://webrtc-review.googlesource.com/87423
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24217}
This CL removes //build/config/clang:find_bad_constructs from the
suppressed_configs list, which means that clang:find_bad_constructs
is now enabled on these translation units.
Bug: webrtc:9251, webrtc:163
Change-Id: I6a7d4964723a5e195189aac30a83d9e924e61dd7
Reviewed-on: https://webrtc-review.googlesource.com/89743
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24053}
WebRTC internal headers are always included starting from the root
(e.g. #include "modules/audio_coding/..."), so there is no need to
specify the include_dirs removed by this CL.
Bug: webrtc:9538
Change-Id: I91e70508c67020bbf70304df5e48ca757ad43221
Reviewed-on: https://webrtc-review.googlesource.com/89385
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24026}
This is an reland of 6f5b0f920a
Relanded after speculative revert without any changes.
TBR=ilnik@webrtc.org
Original change's description:
> Remove rtc::Optional alias and api:optional target
>
> Update left-overs where old target still was used.
>
> Bug: webrtc:9078
> Change-Id: I2162c928091fc4ff1dea33a3f03adbe47207d206
> Reviewed-on: https://webrtc-review.googlesource.com/84740
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23913}
Bug: webrtc:9078
Change-Id: Ia33c6438253c6ec49f45d938e8a3607b51c418be
Reviewed-on: https://webrtc-review.googlesource.com/88160
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23941}
Update left-overs where old target still was used.
Bug: webrtc:9078
Change-Id: I2162c928091fc4ff1dea33a3f03adbe47207d206
Reviewed-on: https://webrtc-review.googlesource.com/84740
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23913}
This CL makes NetEq handle nested RED packets without crashing. Nested
RED packets mean that the block PT (see
https://tools.ietf.org/html/rfc2198.html#section-3) in the RED packet
is also set to the RED PT. This implies a nested RED packet, which is
not supported. Instead, all payloads in a RED packet that have the RED
PT will be discarded.
Bug: chromium:851662
Change-Id: I86ec257e60fb8076e3574ac5a4a1ca50196f6b34
Reviewed-on: https://webrtc-review.googlesource.com/86824
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23825}
Now that there is only one implementation of the decision logic, there
is no longer any need to have GetDecisionSpecialized being separate.
Bug: webrtc:9421
Change-Id: Id364ce09ac05d106652d749502058056f11bba27
Reviewed-on: https://webrtc-review.googlesource.com/86604
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23804}
Building NetEqEventLogInput requires protobuf support, while building
NetEqRtpDumpInput located in the same file does not. This makes both
classes unusable when protobuf support is lacking. With this CL, the
NetEqEventLogInput is broken out into separate files, to allow usage
of NetEqRtpDumpInput even when protobufs are not supported.
Bug: webrtc:9421
Change-Id: I55efec4ec259713654566cdaa00d2e34c5e9a60f
Reviewed-on: https://webrtc-review.googlesource.com/84587
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23803}
This is a reland of 80c4cca491
Original change's description:
> NetEq: Deprecate playout modes Fax, Off and Streaming
>
> The playout modes other than Normal have not been reachable for a long
> time, other than through tests. It is time to deprecate them.
>
> The only meaningful use was that Fax mode was sometimes set from
> tests, in order to avoid time-stretching operations (accelerate and
> pre-emptive expand) from messing with the test results. With this CL,
> a new config is added instead, which lets the user specify exactly
> this: don't do time-stretching.
>
> As a result of Fax and Off modes being removed, the following code
> clean-up was done:
> - Fold DecisionLogicNormal into DecisionLogic.
> - Remove AudioRepetition and AlternativePlc operations, since they can
> no longer be reached.
>
> Bug: webrtc:9421
> Change-Id: I651458e9c1931a99f3b07e242817d303bac119df
> Reviewed-on: https://webrtc-review.googlesource.com/84123
> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23704}
Bug: webrtc:9421
Change-Id: Ice351b635788167f2971b26470f73a5e5fa1a240
Reviewed-on: https://webrtc-review.googlesource.com/86543
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23799}
fft.c is third party library and have to be moved to proper third_party
directory. So this CL will extract it to separate gn target to be able
then to move it to proper location.
Bug: webrtc:8366
Change-Id: I228ebab3c821aa7095f7aa460c23c2ea0fb98f01
Reviewed-on: https://webrtc-review.googlesource.com/85640
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23753}
This reverts commit 80c4cca491.
Reason for revert: Breaks downstream tests.
Original change's description:
> NetEq: Deprecate playout modes Fax, Off and Streaming
>
> The playout modes other than Normal have not been reachable for a long
> time, other than through tests. It is time to deprecate them.
>
> The only meaningful use was that Fax mode was sometimes set from
> tests, in order to avoid time-stretching operations (accelerate and
> pre-emptive expand) from messing with the test results. With this CL,
> a new config is added instead, which lets the user specify exactly
> this: don't do time-stretching.
>
> As a result of Fax and Off modes being removed, the following code
> clean-up was done:
> - Fold DecisionLogicNormal into DecisionLogic.
> - Remove AudioRepetition and AlternativePlc operations, since they can
> no longer be reached.
>
> Bug: webrtc:9421
> Change-Id: I651458e9c1931a99f3b07e242817d303bac119df
> Reviewed-on: https://webrtc-review.googlesource.com/84123
> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23704}
TBR=henrik.lundin@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org
Change-Id: I555aae8850fc4ac1ea919bfa72c11b5218066f30
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9421
Reviewed-on: https://webrtc-review.googlesource.com/84680
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23706}
The playout modes other than Normal have not been reachable for a long
time, other than through tests. It is time to deprecate them.
The only meaningful use was that Fax mode was sometimes set from
tests, in order to avoid time-stretching operations (accelerate and
pre-emptive expand) from messing with the test results. With this CL,
a new config is added instead, which lets the user specify exactly
this: don't do time-stretching.
As a result of Fax and Off modes being removed, the following code
clean-up was done:
- Fold DecisionLogicNormal into DecisionLogic.
- Remove AudioRepetition and AlternativePlc operations, since they can
no longer be reached.
Bug: webrtc:9421
Change-Id: I651458e9c1931a99f3b07e242817d303bac119df
Reviewed-on: https://webrtc-review.googlesource.com/84123
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23704}
By removing it we can in turn (next CL) get rid of RTPTypeHeader, which is a
union that cause some problems.
Bug: none
Change-Id: I9246ecbfe2c8b7eda27497cccbc5f438958b64bf
Reviewed-on: https://webrtc-review.googlesource.com/83985
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23666}
Running clang-format with chromium's style guide.
The goal is n-fold:
* providing consistency and readability (that's what code guidelines are for)
* preventing noise with presubmit checks and git cl format
* building on the previous point: making it easier to automatically fix format issues
* you name it
Please consider using git-hyper-blame to ignore this commit.
Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script with parameter 'modules/audio_coding'
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"[\./api]*:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: Ic980ee605148fdb160666d4aa03cc87175e48fe8
Reviewed-on: https://webrtc-review.googlesource.com/84130
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23659}
This test is so old, it used to be interactive with an automated mode
bolted on to the side. That automatic mode is the only one that's used
nowadays.
Bug: webrtc:8396
Change-Id: I3b473f53ff6afa363b9691e8471a5754f46d3d3f
Reviewed-on: https://webrtc-review.googlesource.com/83583
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23620}
It appears to have been created in mid-2013, and hasn't been changed
since except to keep the compiler happy when surrounding code changed.
It crashes when I try to run it without arguments, and no one
remembers how to use it.
Bug: webrtc:8396
Change-Id: I2eae36cf468f28c5bf05c85e6a3aaeebc48a1ffc
Reviewed-on: https://webrtc-review.googlesource.com/83581
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23618}
Since it isn't being run by the bots, it has bit rotted; when I try to
run it manually, it fails with a long list of error messages:
Error Calling API in file ../../modules/audio_coding/test/APITest.cc at line 995
Error Calling API in file ../../modules/audio_coding/test/APITest.cc at line 996
>>> Error Enabling VAD <<<
Error Calling API in file ../../modules/audio_coding/test/APITest.cc at line 995
Error Calling API in file ../../modules/audio_coding/test/APITest.cc at line 996
>>> Error Enabling DTX <<<
>>> Error Enabling VAD <<<
Error Calling API in file ../../modules/audio_coding/test/APITest.cc at line 995
Error Calling API in file ../../modules/audio_coding/test/APITest.cc at line 996
>>> Error Enabling VAD <<<
Error Calling API in file ../../modules/audio_coding/test/APITest.cc at line 995
Error Calling API in file ../../modules/audio_coding/test/APITest.cc at line 996
Error Calling API in file ../../modules/audio_coding/test/APITest.cc at line 985
...and so on.
Bug: webrtc:8396
Change-Id: Id8f1e01a751b4bb3527702b7b7a4986ce0abb378
Reviewed-on: https://webrtc-review.googlesource.com/81745
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23542}
It's always been there, and there's no security risk.
Bug: chromium:843477
Change-Id: I6121943f23b477300cf60ffc4858ef0ab43466dc
Reviewed-on: https://webrtc-review.googlesource.com/78782
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23393}
This CL implements a fix behind a field trial for a NetEq issue. NetEq restarts audio too quickly after a buffer underrun, which can quickly lead to another underrun in some circumstances. The fix changes NetEq's behavior to wait with restarting playback until sufficient audio is buffered.
Bug: webrtc:9289
Change-Id: I5968c9478ce8d84caf77f00b8d0a39156b47fc8d
Reviewed-on: https://webrtc-review.googlesource.com/77423
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23347}
NetEq tapers down the audio produced through loss concealment when the
expansion has been going on for some time. When the audio packets starts
coming in again, there is a ramp-up that happens. This ramp-up could
before this change extend over more than one 10 ms block, which made
keeping track of the scaling factor necessary. With this change, we make
this ramp-up quicker in the rare cases when it lasted more than 10 ms,
so that it always ramps up to 100% within one block. This way, we can
remove the mute_factor_array.
This change breaks bit-exactness, but careful listening could not reveal
an audible difference.
This change is a part of a larger refactoring of NetEq's PLC code.
Bug: webrtc:9180
Change-Id: I4c513ce3ed8d66f9beec2abfb1f0c7ffaac7a21e
Reviewed-on: https://webrtc-review.googlesource.com/77180
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23342}
Instead of checking for an exact bitrate check that the bitrate is between
the min and max values.
Also relax a threshold in a bandwith adaptation test.
Bug: webrtc:9280
Change-Id: I465d785a53759f73242198ee1ccd7da1a26c48b7
Reviewed-on: https://webrtc-review.googlesource.com/78041
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23330}
The overflow currently does not cause any problems, but it has been
found that it can cause crashes after a refactoring that is coming in
the near future.
Bug: webrtc:9180
Change-Id: Ia2c4e545c062c4f8ad13cbc47b8796c6e8a4e906
Reviewed-on: https://webrtc-review.googlesource.com/77667
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23327}
This is a reland of 79ded653fe
Original change's description:
> Update expected bitrate in Opus tests
>
> Upstream changes to Opus DTX behavior changes the bitrates of Opus. This
> CL re-enables recently disabled unittests and updates the expected bitrates.
>
> Bug: webrtc:9280
> Change-Id: I668a0b6a8b82cbbb70d795db4546cb5469266bf2
> Reviewed-on: https://webrtc-review.googlesource.com/77766
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23306}
TBR=henrik.lundin@webrtc.org
Bug: webrtc:9280
Change-Id: I6bfcd1c5e1d5298543024a0faa6a695026434df3
Reviewed-on: https://webrtc-review.googlesource.com/77980
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23318}
This reverts commit 79ded653fe.
Reason for revert: Different repos have different Opus
Original change's description:
> Update expected bitrate in Opus tests
>
> Upstream changes to Opus DTX behavior changes the bitrates of Opus. This
> CL re-enables recently disabled unittests and updates the expected bitrates.
>
> Bug: webrtc:9280
> Change-Id: I668a0b6a8b82cbbb70d795db4546cb5469266bf2
> Reviewed-on: https://webrtc-review.googlesource.com/77766
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23306}
TBR=henrik.lundin@webrtc.org,gustaf@webrtc.org
Change-Id: I3c18db2d6052c4049d836c3e595b00189aebcbc8
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9280
Reviewed-on: https://webrtc-review.googlesource.com/77800
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23307}
Upstream changes to Opus DTX behavior changes the bitrates of Opus. This
CL re-enables recently disabled unittests and updates the expected bitrates.
Bug: webrtc:9280
Change-Id: I668a0b6a8b82cbbb70d795db4546cb5469266bf2
Reviewed-on: https://webrtc-review.googlesource.com/77766
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23306}
In order to roll the new version of Opus in WebRTC, this CL disables
some tests that will fail because of [1].
They will be re-enabled and fixed as soon as the new Opus revision is
rolled.
[1] - https://chromium-review.googlesource.com/1061499TBR=henrik.lundin@webrtc.org
Bug: webrtc:9280
Change-Id: I84870ced66d554f75c2d093dac8103ad7860cae5
Reviewed-on: https://webrtc-review.googlesource.com/77640
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23293}
The current implementation carefully shifts down the energy so as not to overflow.
The fuzzer audio_decoder_ilbc_fuzzer found an integer overflow anyway.
The energy is only used for a threshold check.
This fix stops the energy computation when the threshold is reached, before it can overflow.
Bug: chromium:837922
Change-Id: I45e84d2d271a37e6476b08433a2cbd5a8f6e6f26
Reviewed-on: https://webrtc-review.googlesource.com/76122
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23242}
A bug was introduced to NetEq RTP player in a recent CL:
https://webrtc-review.googlesource.com/c/src/+/69806
This is to fix it.
Bug: webrtc:9147
Change-Id: I949fd6b220d7c7f08c6e2940468232d1d955a3dc
Reviewed-on: https://webrtc-review.googlesource.com/75321
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23181}
This change adds flags and default values for two more RTP header
extensions: VideoContentType and VideoTiming.
This will silence a number of annoying warnings when running with
application logs.
Bug: none
Change-Id: I9bb01ea2519813d3c47553ecff384141fbede23e
Reviewed-on: https://webrtc-review.googlesource.com/75300
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23178}
This removes the redundant type and replaces all usages. A slight change
in behavior is that we no longer get nanosecond resolution. This should
not matter since no current code requires nanosecond resolution.
Bug: webrtc:9155
Change-Id: I04334e08c686d95731621a6c8a7e40400d0ae3b2
Reviewed-on: https://webrtc-review.googlesource.com/71163
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23174}
Only half of the array was initialized. Now all of it is.
Bug: chromium:839960
Change-Id: If8bbe12c4c4c0dc0d529c93b22e49a94ecb09919
Reviewed-on: https://webrtc-review.googlesource.com/74820
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23167}
The audio processing code is using parts of the iSAC codec to do voice
activity detection (VAD), but it's undesirable for it to pull in the
entire iSAC codec as a dependency. So this CL factors out the parts of
iSAC that's needed for VAD to a separate build target.
Bug: webrtc:8396
Change-Id: I884e25d8fd0bc815fca664352b0573b4b173880e
Reviewed-on: https://webrtc-review.googlesource.com/69640
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23110}
