Commit graph

25 commits

Author SHA1 Message Date
Mirko Bonadei
190244bb59 Remove all #include <assert.h>/<cassert> and usage in Obj-C code.
This CL completes the removal of assert() and relative headers from
the codebase (excluded
//examples/objc/AppRTCMobile/third_party/SocketRocket which is in a
third_party sub-directory).

Bug: webrtc:6779
Change-Id: I93ed57168d2c0e011626873d66529488c5f484f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225546
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34528}
2021-07-22 14:00:26 +00:00
Mirko Bonadei
25ab3228f3 Replace assert() with RTC_DCHECK().
CL partially auto-generated with:

git grep -l "\bassert(" | grep "\.[c|h]" | \
  xargs sed -i 's/\bassert(/RTC_DCHECK(/g'

And with:

git grep -l "RTC_DCHECK(false)" |  \
  xargs sed -i 's/RTC_DCHECK(false)/RTC_NOTREACHED()/g'

With some manual changes to include "rtc_base/checks.h" where
needed.

A follow-up CL will remove assert() from Obj-C code as well
and remove the #include of <assert.h>.

The choice to replace with RTC_DCHECK is because assert()
is because RTC_DCHECK has similar behavior as assert()
based on NDEBUG.

This CL also contains manual changes to switch from
basic RTC_DCHECK to other (preferred) versions like
RTC_DCHECK_GT (and similar).

Bug: webrtc:6779
Change-Id: I00bed8886e03d685a2f42324e34aef2c9b7a63b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/224846
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34442}
2021-07-09 07:49:43 +00:00
Christoffer Jansson
46d002cb36 Add M1 Mac expected results for AudioDecoderIsacFixTest
Bug: webrtc:12882
Change-Id: I56c1fcdd85fab88924b9a9f53a1a20485633f840
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223660
Commit-Queue: Christoffer Jansson <jansson@google.com>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34389}
2021-06-30 07:03:52 +00:00
Christoffer Jansson
7208457e80 Same length for all ARM64 platforms
Update more audio checksums for M1

Bug: webrtc:12882
Change-Id: I527a43a01afe2b2e4af137852174159bf3111652
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/224081
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Christoffer Jansson <jansson@google.com>
Cr-Commit-Position: refs/heads/master@{#34372}
2021-06-28 11:18:40 +00:00
Jakob Ivarsson
213dc2cfc5 Temporarily disable Opus decode test.
Bug: webrtc:12518, webrtc:12543
Change-Id: I5481ee96fe2a3f9fd549e17cd9424441223a8b63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211245
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33417}
2021-03-10 12:47:18 +00:00
Jakob Ivarsson
36274f9158 Reland "Reland "Default enable WebRTC-SendSideBwe-WithOverhead.""
This is a reland of 1dbe30c7e8

Original change's description:
> Reland "Default enable WebRTC-SendSideBwe-WithOverhead."
>
> This is a reland of 87c1950841
>
> Original change's description:
> > Default enable WebRTC-SendSideBwe-WithOverhead.
> >
> > Bug: webrtc:6762
> > Change-Id: I18ace06a33b3b60d5a19796d4769f70cd977d604
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188801
> > Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Reviewed-by: Ali Tofigh <alito@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#32472}
>
> Bug: webrtc:6762
> Change-Id: Icf096a8755d29600a13bd08b1f22f5a79de21e90
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190143
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32492}

Bug: webrtc:6762
Change-Id: I6d79894a213fc42d2338409e7513247725881b1a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191221
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32534}
2020-11-02 11:05:56 +00:00
Björn Terelius
d546186b89 Revert "Reland "Default enable WebRTC-SendSideBwe-WithOverhead.""
This reverts commit 1dbe30c7e8.

Reason for revert: Speculative revert due to failing tests.

Original change's description:
> Reland "Default enable WebRTC-SendSideBwe-WithOverhead."
>
> This is a reland of 87c1950841
>
> Original change's description:
> > Default enable WebRTC-SendSideBwe-WithOverhead.
> >
> > Bug: webrtc:6762
> > Change-Id: I18ace06a33b3b60d5a19796d4769f70cd977d604
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188801
> > Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Reviewed-by: Ali Tofigh <alito@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#32472}
>
> Bug: webrtc:6762
> Change-Id: Icf096a8755d29600a13bd08b1f22f5a79de21e90
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190143
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32492}

TBR=stefan@webrtc.org,jakobi@webrtc.org,alito@webrtc.org

Change-Id: I7e0378788576236059627cf8c3bad58cd70aff7e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:6762
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190500
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32504}
2020-10-27 10:51:46 +00:00
Jakob Ivarsson
1dbe30c7e8 Reland "Default enable WebRTC-SendSideBwe-WithOverhead."
This is a reland of 87c1950841

Original change's description:
> Default enable WebRTC-SendSideBwe-WithOverhead.
>
> Bug: webrtc:6762
> Change-Id: I18ace06a33b3b60d5a19796d4769f70cd977d604
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188801
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32472}

Bug: webrtc:6762
Change-Id: Icf096a8755d29600a13bd08b1f22f5a79de21e90
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190143
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32492}
2020-10-26 12:35:47 +00:00
Jakob Ivarsson
27af3c4c24 Revert "Default enable WebRTC-SendSideBwe-WithOverhead."
This reverts commit 87c1950841.

Reason for revert: breaks downstream tests

Original change's description:
> Default enable WebRTC-SendSideBwe-WithOverhead.
>
> Bug: webrtc:6762
> Change-Id: I18ace06a33b3b60d5a19796d4769f70cd977d604
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188801
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32472}

TBR=stefan@webrtc.org,jakobi@webrtc.org,alito@webrtc.org

Change-Id: If59fd41dcd8f6db76ea297c34c25fe19ae2ae973
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:6762
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189973
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32474}
2020-10-22 16:57:18 +00:00
Jakob Ivarsson
87c1950841 Default enable WebRTC-SendSideBwe-WithOverhead.
Bug: webrtc:6762
Change-Id: I18ace06a33b3b60d5a19796d4769f70cd977d604
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188801
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32472}
2020-10-22 13:37:18 +00:00
Karl Wiberg
30a3e78794 iSAC encoder: Make it possible to change target bitrate at any time
Not just at construction time.

Bug: webrtc:11704
Change-Id: I952c7dbe20774cc976065c7d2f992a80074ebf63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177663
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31550}
2020-06-22 14:59:22 +00:00
Ivo Creusen
16ddae924e Update Opus tests for Opus 1.3
This updates various bitexactness tests and other tests that no longer
pass.

Bug: webrtc:11325
Change-Id: Ifa3e4b42e303f5573e028dfdf8a108a76f6318ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168952
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30688}
2020-03-05 08:53:37 +00:00
Alessio Bazzica
b28e57e725 NetEQ audio decoder unit test: use ParsePayload
AudioDecoder::Decode() is obsolete. This CL replaces it with
ParsePayload() in the audio decoder NetEQ unit tests.

Bug: webrtc:10098
Change-Id: I602b0330adbe1d0921b0c4524aa7305b500f2ebf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168486
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30511}
2020-02-13 09:05:55 +00:00
Ivo Creusen
c31a4ec66a Disable opus tests to allow upgrade to opus 1.3
The upgrade to opus 1.3 is easier to carry out while the opus
bitexactness tests are temporarily disabled.

Bug: webrtc:11325
Change-Id: I96eecdbc93a01da88b92ae7f6473034c9795f3a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167726
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30425}
2020-01-30 14:57:15 +00:00
Niels Möller
48b32b748e Delete support for enabling adaptive isac mode
This appears unused. If deleted, other code related to isac bandwidth
estimation becomes unused and may be deleted in followup cls.

Bug: webrtc:10098
Change-Id: Ifeac2e90de895b12c337ea28cc33704350b9abf4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153667
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29252}
2019-09-20 10:41:09 +00:00
Jiawei Ou
608e6ba394 Add AudioDecoderIsacT::Config to include sampling rate and BWInfo object
This CL will make AudioDecoderIsacT symmetrical to AudioEncoderIsacT.

Bug: webrtc:10826
Change-Id: I78d1cf7bc2245bf4a282aabd81c8ece6ca23f285
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146683
Commit-Queue: Jiawei Ou <ouj@fb.com>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28847}
2019-08-14 00:40:19 +00:00
Karl Wiberg
7eb0a5e210 AudioDecoderOpus: Add support for 16 kHz output sample rate
In addition to the 48 kHz that we've always used.

Bug: webrtc:10631
Change-Id: If73bf7ff9c1c0d22e0d1caa245128612850f8e41
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138268
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28104}
2019-05-29 12:42:38 +00:00
Steve Anton
10542f21c8 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
Mechanically generated by running this command:

tools_webrtc/do-renames.sh update all-renames.txt && git cl format

Then manually updating:

tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc

Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833
Reviewed-on: https://webrtc-review.googlesource.com/c/115653
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 17:11:39 +00:00
Mirko Bonadei
682aac5103 Enable clang::find_bad_constructs for audio_coding (part 1/2).
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}
2018-07-20 13:07:47 +00:00
Yves Gerey
665174fdbb Reformat the WebRTC code base
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}
2018-06-19 14:00:39 +00:00
Danil Chapovalov
b602123a5a Replace rtc::Optional with absl::optional in modules/audio_coding
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}
2018-06-19 12:46:20 +00:00
Oskar Sundbom
12ab00b4d8 Optional: Use nullopt and implicit construction in /modules/audio_coding
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.

This CL was uploaded by git cl split.

R=kwiberg@webrtc.org

Bug: None
Change-Id: I055411a3e521964c81100869a197dd92f5608f1b
Reviewed-on: https://webrtc-review.googlesource.com/23619
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20728}
2017-11-17 11:58:37 +00:00
Karl Wiberg
31fbb5425e NetEq: Drop unnecessary dependency on the audio decoder implementations
BUG=webrtc:8396

Change-Id: I7524dae93b43b656a13fdd535e48373bc29b405e
Reviewed-on: https://webrtc-review.googlesource.com/10804
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20310}
2017-10-16 12:57:47 +00:00
Mirko Bonadei
92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 05:02:56 +00:00
Mirko Bonadei
bb547203bf Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 04:25:06 +00:00
Renamed from webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc (Browse further)