This reverts commit 80c4cca4915dbc6094a5bfae749f85f7371eadd1.
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}
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}
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}
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}
Only specially taggged targets may transitively depend on poisonous
targets. We first apply it to audio codecs.
This makes it much clearer exactly what parts of the code still have
dependencies on the audio codecs (and we want to eventually get rid of
pretty much all of them).
Bug: webrtc:8396, webrtc:9121
Change-Id: Iba5c2e806c702b5cfe881022674705f647896d43
Reviewed-on: https://webrtc-review.googlesource.com/69520
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22979}
The target modules/audio_coding:isac_neon needs to link with
transform_tables.c but adding a dependency between isac_neon and
isac_fix_c creates a circular dependency.
This CL moves transform_tables.c to isac_fix_common (which is already a
dependency of isac_neon).
Bug: None
Change-Id: I4135ec772b0017e77f1411e9a8093b495220c636
Reviewed-on: https://webrtc-review.googlesource.com/71581
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22968}
PCMFile.cc uses RTC_DCHECK. include and depend on rtc_base:checks target directly
change usage of value_or by using explicit constructor instead of implicit
Bug: webrtc:9078
Change-Id: I63c596b8a05b387e56df846b15c33a605fbad4e6
Reviewed-on: https://webrtc-review.googlesource.com/69985
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22897}
- Directly include api/audio/audio_frame.h everywhere AudioFrame is used.
- This *will* remove transient dependencies on libjpeg and a bunch of other things from the e.g. APM.
- audio_frame.h still included from module_common_types.h for backwards compatibility with clients.
Bug: webrtc:9139, webrtc:7504
Change-Id: Id96f9268c01667fbcc29a01f5c1dd25a37836897
Reviewed-on: https://webrtc-review.googlesource.com/62464
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22845}
These two new histograms relate to the packet-loss concealment that
happens when audio packets are lost or late for decoding, and the
NetEq must resort to extrapolating audio from the previously
decoded data.
Bug: webrtc:9126
Change-Id: I99cc97e653169fb742da0092653ab99fd10e5d7b
Reviewed-on: https://webrtc-review.googlesource.com/67861
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22812}
So that we don't have to be capable of creating one ourselves, which
requires a dependency on the audio decoders.
BUG=webrtc:5801, webrtc:8396
Change-Id: I80749ec3b86cba73994307046d05964f59167d44
Reviewed-on: https://webrtc-review.googlesource.com/18440
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22774}
This moves it from an API directory (system_wrappers/include/) to a
non-API directory, which is exactly what we want for utilities like
this.
BUG=webrtc:8445
NOPRESUBMIT=true
Change-Id: Ie2879aca5fc1667e4222499d2a8fc2bba9ae2425
Reviewed-on: https://webrtc-review.googlesource.com/21328
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22587}
This moves it from an API directory (system_wrappers/include/) to a
non-API directory, which is exactly what we want for utilities like
this.
BUG=webrtc:8445
Change-Id: I440974da4d347b09ff042478720d7983056b62b9
Reviewed-on: https://webrtc-review.googlesource.com/21226
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22579}
This moves it from an API directory (system_wrappers/include/) to a
non-API directory, which is exactly what we want for utilities like
this.
Bug: webrtc:8445
NOPRESUBMIT=true
Change-Id: I30d01fcb9cbe1427a7703a3cdd7befae751066b5
Reviewed-on: https://webrtc-review.googlesource.com/21982
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22550}
WebRTC internal code should always used include paths that starts
from the root of the project and that clearly identify the header file.
This allows 'gn check' to actually keep dependencies under control
because 'gn check' cannot enforce anything if the include path
is not fully qualified (starting from the root of the project).
Bug: webrtc:8815
Change-Id: I23fb4fed0c27a4d98bea360315b959af843587bc
Reviewed-on: https://webrtc-review.googlesource.com/46101
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21847}
The original rtc_event_log_api is refactored to a pure API target plus
multiple targets coupled with WebRTC implementations.
Bug: None
Change-Id: Iab9eee3f7bf4228c52d94a5f26fc39bb99b5033f
Reviewed-on: https://webrtc-review.googlesource.com/43247
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@google.com>
Cr-Commit-Position: refs/heads/master@{#21811}
If a WebRTC build target requires gmock it has to include
test/gmock.h and just depend on //test:test_support.
Unfortunately //testtest_support was a leaky abstraction because it
wasn't propagating the correct -I compiler flag. To make everything
work, all the targets that use gmock started also to depend on
//testing/gmock (even if they were not including any gmock header
directly).
This CL makes //testtest_support propagate the include path up in the
dependency chain so it is possible to remove unused dependencies.
Note: all_dependent_configs should probably be used in the original
gmock target. There is an ongoing discussion about it. This CL solves
the problem on WebRTC side and it is forward compatible.
TBR=phoglund@webrtc.org
Bug: webrtc:8603
Change-Id: If08daf2ce9a6431a6e881a236743b4ec33b59ea7
Reviewed-on: https://webrtc-review.googlesource.com/44340
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21776}
This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613.
Reason for revert:
The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660
Original change's description:
> Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""
>
> This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16.
>
> Reason for revert:
>
> Breaks Chrome FYI:
>
> /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check
> -> returned 1
> ERROR at //build/split_static_library.gni:12:5: Dependency not allowed.
> static_library(target_name) {
> ^----------------------------
> The item //content/renderer:renderer
> can not depend on //third_party/webrtc/media:rtc_internal_video_codecs
> because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [
> //third_party/webrtc/*
> //third_party/webrtc_overrides/*
> ]
>
> https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout
>
> Original change's description:
> > GN rtc_* templates: Set default visibility to webrtc_root + "/*"
> >
> > This means that by default, targets are visible to everything under
> > the WebRTC root, but not visible to anything else.
> >
> > API targets are manually tagged with visibility "*", so that targets
> > outside the WebRTC tree can see them.
> >
> > BUG=webrtc:8254
> >
> > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509
> > Reviewed-on: https://webrtc-review.googlesource.com/24140
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21548}
>
> TBR=mbonadei@webrtc.org,kwiberg@webrtc.org
>
> Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8254
> Reviewed-on: https://webrtc-review.googlesource.com/38760
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21555}
TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org
Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8254
Reviewed-on: https://webrtc-review.googlesource.com/38860
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21558}
This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16.
Reason for revert:
Breaks Chrome FYI:
/b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check
-> returned 1
ERROR at //build/split_static_library.gni:12:5: Dependency not allowed.
static_library(target_name) {
^----------------------------
The item //content/renderer:renderer
can not depend on //third_party/webrtc/media:rtc_internal_video_codecs
because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [
//third_party/webrtc/*
//third_party/webrtc_overrides/*
]
https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout
Original change's description:
> GN rtc_* templates: Set default visibility to webrtc_root + "/*"
>
> This means that by default, targets are visible to everything under
> the WebRTC root, but not visible to anything else.
>
> API targets are manually tagged with visibility "*", so that targets
> outside the WebRTC tree can see them.
>
> BUG=webrtc:8254
>
> Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509
> Reviewed-on: https://webrtc-review.googlesource.com/24140
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21548}
TBR=mbonadei@webrtc.org,kwiberg@webrtc.org
Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8254
Reviewed-on: https://webrtc-review.googlesource.com/38760
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21555}
This means that by default, targets are visible to everything under
the WebRTC root, but not visible to anything else.
API targets are manually tagged with visibility "*", so that targets
outside the WebRTC tree can see them.
BUG=webrtc:8254
Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509
Reviewed-on: https://webrtc-review.googlesource.com/24140
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21548}
This allows removing the specific classes in a later CL.
Bug: webrtc:8677
Change-Id: I3b9c1f3191c38e6d31a3de990e2d882505e79adc
Reviewed-on: https://webrtc-review.googlesource.com/35040
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Joachim Bauch <jbauch@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21412}
One reason for the circular deps is that common_types.h is a
historical dumping ground for various structs and defines that
are believed to be generally useful. I tried moving things out
that did not appear to be used downstream (StreamCounters,
RtpCounters etc) and moved the things that seemed used
(RtpHeader + supporting structs) to a new file api/rtp_headers.h.
This makes their place in the api more clear while moving out
the things that don't belong in the API in the first place.
I had to extract out typedefs.h from webrtc_common to resolve
another circular dependency. I believe checks includes typedefs,
but common depends on checks.
Bug: webrtc:7745
Change-Id: I725d49616b1ec0cdc8b74be7c078f7a4d46f084b
Reviewed-on: https://webrtc-review.googlesource.com/33001
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21295}
This splits things out of rtc_base and makes dependencies explicit.
Bug: webrtc:6828
Change-Id: Id521896c3c43595349021c857bec216e429a0c8d
Reviewed-on: https://webrtc-review.googlesource.com/32780
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21264}
