Commit Graph

16 Commits

Author SHA1 Message Date
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
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
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
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AudioDecoder::Decode() is obsolete.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
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
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
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
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
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
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
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
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
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
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
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
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