Commit graph

338 commits

Author SHA1 Message Date
Sam Zackrisson
c496d58882 Add flag for fast jitter buffer playout in neteq simulation
It is currently not possible to run e.g. neteq_rtpplay in the fast
accelerate mode.

Bug: None
Change-Id: I5e0ce3fae2ad5585fe9fb545109bb0c9a87fd201
Reviewed-on: https://webrtc-review.googlesource.com/c/110162
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25561}
2018-11-08 14:32:48 +00:00
Sam Zackrisson
f0e7440a35 Add missing conditional defines to neteq test and tools targets
The .cc source files listed below #ifdef for WEBRTC_CODEC_OPUS and
WEBRTC_CODEC_ILBC but the build files don't include the defines.

modules/audio_coding/neteq/tools/neteq_test.cc
modules/audio_coding/neteq/tools/neteq_test_factory.cc

Bug: None
Change-Id: I6065021f68e58d0e5663acd006a9865bf265adc0
Reviewed-on: https://webrtc-review.googlesource.com/c/109925
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25555}
2018-11-08 11:25:10 +00:00
Niels Möller
c572ff3c71 Add default constructor for rtc::Event
Bug: webrtc:9962
Change-Id: Icaa91e657e6881fcb1553f354c07866109a0ea68
Reviewed-on: https://webrtc-review.googlesource.com/c/109500
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25535}
2018-11-07 08:57:50 +00:00
tzik
254d3db59a Add missing #include to absl/memory/memory.h from audio_encoder_cng.cc
absl::make_unique is used in this file without absl/memory/memory.h
#include, that causes a build error on C++17 build of Chromium.

Bug: chromium:752720
Change-Id: I78fe9f76a6ea670a4250b4cf25c3c02cf4c4beb6
Reviewed-on: https://webrtc-review.googlesource.com/c/109540
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25514}
2018-11-06 10:57:47 +00:00
Henrik Lundin
007065522a Removing ancient and unused test scripts and data files
None of these scripts or files have been used in a very long time. They
are removed for the same reason, and since the data files add to the
weight of the resources folder.

Bug: webrtc:5289
Change-Id: Ia14a46aed180f286fa881fe5f60da6973a5fe027
Reviewed-on: https://webrtc-review.googlesource.com/c/109022
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25502}
2018-11-05 16:08:46 +00:00
Karl Wiberg
2365936b87 Hide the AudioEncoderCng class behind a create function
And put codecs/cng/webrtc_cng.h in a non-public build target while
we're at it.

Bug: webrtc:8396
Change-Id: I9f51dffadfb645cd1454617fad30e09d639ff53c
Reviewed-on: https://webrtc-review.googlesource.com/c/108782
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25486}
2018-11-02 13:00:05 +00:00
Yves Gerey
69807e8871 Depend directly on destination targets.
Makes 'gn check' happy.
Followup to https://webrtc-review.googlesource.com/c/src/+/106820

Bug: webrtc:5876, webrtc:9855
Change-Id: I33fa2c31ba26dc10c9a9c17da0ffed255c1f4d5b
Reviewed-on: https://webrtc-review.googlesource.com/c/108760
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25447}
2018-10-31 10:21:40 +00:00
Yves Gerey
21cddffd99 Harmonize paths to dependent targets.
This CL consistently use:
 * relative paths for WebRTC dependent targets (test_support)
 * absolute paths for shared dependent targets (abseil)
This is a necessary (but insufficient) step to build WebRTC tests
from Chromium tree (rtc_include_tests=true), since test/ doesn't
sit anymore in the top level directory.

We also make sure that target declarations and uses are
consistent in regard to build_with_chromium flag.

Bug: webrtc:9943
Bug: webrtc:9855
Change-Id: I21dea98894df2fd4bfe2fd7ee7b71ba971e0ab5b
Reviewed-on: https://webrtc-review.googlesource.com/c/108720
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25445}
2018-10-31 10:04:59 +00:00
Niels Möller
2c16cc61c2 Replace some usage of EventWrapper with rtc::Event.
Bug: webrtc:3380
Change-Id: Id33b19bf107273e6f838aa633784db73d02ae2c2
Reviewed-on: https://webrtc-review.googlesource.com/c/107888
Reviewed-by: Henrik Grunell <henrikg@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25407}
2018-10-29 09:37:24 +00:00
Yves Gerey
988cc0870b [Cleanup] Add missing #include. Remove useless ones.
This CL is the result of running include-what-you-use tool on part
of the code base (audio target and dependencies) plus manual fixes.

bug: webrtc:8311
Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604
Reviewed-on: https://webrtc-review.googlesource.com/c/106280
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25311}
2018-10-23 11:32:56 +00:00
Niels Möller
2edab4c026 Delete use of STR_CASE_CMP, replaced with absl::EqualsIgnoreCase.
Bug: webrtc:5876
Change-Id: Ica2d47ca45b8ef01a548d8dbe31dbed740a0ebda
Reviewed-on: https://webrtc-review.googlesource.com/c/106820
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25306}
2018-10-23 09:24:15 +00:00
Mirko Bonadei
2dfa998be2 Reland "Prefix flag macros with WEBRTC_."
This is a reland of 5ccdc1331f

Original change's description:
> Prefix flag macros with WEBRTC_.
>
> Macros defined in rtc_base/flags.h are intended to be used to define
> flags in WebRTC's binaries (e.g. tests).
>
> They are currently not prefixed and this could cause problems with
> downstream clients since these names are quite common.
>
> This CL adds the 'WEBRTC_' prefix to them.
>
> Generated with:
>
> for x in DECLARE DEFINE; do
>   for y in bool int float string FLAG; do
>     git grep -l "\b$x\_$y\b" | \
>     xargs sed -i "s/\b$x\_$y\b/WEBRTC_$x\_$y/g"
>   done
> done
> git cl format
>
> Bug: webrtc:9884
> Change-Id: I7b524762b6a3e5aa5b2fc2395edd3e1a0fe72591
> Reviewed-on: https://webrtc-review.googlesource.com/c/106682
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25270}

TBR=kwiberg@webrtc.org

Bug: webrtc:9884
Change-Id: I5ba5368a231a334d135ed5e6fd7a279629ced8a3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/107161
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25277}
2018-10-19 15:06:43 +00:00
Mirko Bonadei
c538fc77b0 Revert "Prefix flag macros with WEBRTC_."
This reverts commit 5ccdc1331f.

Reason for revert: Breaks downstream project.

Original change's description:
> Prefix flag macros with WEBRTC_.
> 
> Macros defined in rtc_base/flags.h are intended to be used to define
> flags in WebRTC's binaries (e.g. tests).
> 
> They are currently not prefixed and this could cause problems with
> downstream clients since these names are quite common.
> 
> This CL adds the 'WEBRTC_' prefix to them.
> 
> Generated with:
> 
> for x in DECLARE DEFINE; do
>   for y in bool int float string FLAG; do
>     git grep -l "\b$x\_$y\b" | \
>     xargs sed -i "s/\b$x\_$y\b/WEBRTC_$x\_$y/g"
>   done
> done
> git cl format
> 
> Bug: webrtc:9884
> Change-Id: I7b524762b6a3e5aa5b2fc2395edd3e1a0fe72591
> Reviewed-on: https://webrtc-review.googlesource.com/c/106682
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25270}

TBR=mbonadei@webrtc.org,kwiberg@webrtc.org

Change-Id: Ia79cd6066ecfd1511c34f1b30fd423e560ed6854
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9884
Reviewed-on: https://webrtc-review.googlesource.com/c/107160
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25276}
2018-10-19 15:04:13 +00:00
Mirko Bonadei
5ccdc1331f Prefix flag macros with WEBRTC_.
Macros defined in rtc_base/flags.h are intended to be used to define
flags in WebRTC's binaries (e.g. tests).

They are currently not prefixed and this could cause problems with
downstream clients since these names are quite common.

This CL adds the 'WEBRTC_' prefix to them.

Generated with:

for x in DECLARE DEFINE; do
  for y in bool int float string FLAG; do
    git grep -l "\b$x\_$y\b" | \
    xargs sed -i "s/\b$x\_$y\b/WEBRTC_$x\_$y/g"
  done
done
git cl format

Bug: webrtc:9884
Change-Id: I7b524762b6a3e5aa5b2fc2395edd3e1a0fe72591
Reviewed-on: https://webrtc-review.googlesource.com/c/106682
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25270}
2018-10-19 10:55:20 +00:00
Sebastian Jansson
b9972fa37b Adds AudioNetworkAdaptation support to Scenario tests.
Bug: webrtc:9718
Change-Id: I6cb976df5767797fec670134d29e030ec0f9d3a2
Reviewed-on: https://webrtc-review.googlesource.com/c/106340
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25236}
2018-10-17 15:42:58 +00:00
Minyue Li
34d990fef9 Adding NetEq buffer full metric to UMA.
BUG: webrtc:9882
Change-Id: Idbcbbbd99855b2251fbb66629efeab4f2d1f6498
Reviewed-on: https://webrtc-review.googlesource.com/c/106400
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25230}
2018-10-17 12:54:19 +00:00
Ivo Creusen
ed04912ccd Stop simulations when a LOG_END event is reached.
When a LOG_END event is reached, it makes no sense to continue simulating NetEq.

Bug: webrtc:9667
Change-Id: Ie4f6811cdec0d0632f6e7906059e0e74e9f10438
Reviewed-on: https://webrtc-review.googlesource.com/c/105643
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25176}
2018-10-15 16:06:40 +00:00
Ivo Creusen
d8a52b3ff4 Make ivoc owner of audio_coding.
Bug: None
Change-Id: I9e20031cd292b3459d5bead1a5763af9af18a325
Reviewed-on: https://webrtc-review.googlesource.com/c/106021
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25174}
2018-10-15 15:08:28 +00:00
Artem Titov
40a7a35eaa Extract functionality of test_main into separate library.
Extract functionality of test_main into separate library to be able to
reuse it if another main will be required.

Bug: webrtc:5996
Change-Id: I2925b4240bd0e4fb884b43bb16667ca2d6216bbd
Reviewed-on: https://webrtc-review.googlesource.com/c/105921
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25172}
2018-10-15 14:13:06 +00:00
Ivo Creusen
d2d2ecb4a8 Add command-line flag for setting the max number of packets in the buffer.
There is currently no way to set this for simulations in neteq_rtpplay.

Bug: webrtc:9667
Change-Id: I34f34565538bd3c378cdb9d355f5173c3517d59a
Reviewed-on: https://webrtc-review.googlesource.com/c/105982
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25171}
2018-10-15 14:10:24 +00:00
Jakob Ivarsson
83bd37cda4 Add field trials for configuring Opus encoder packet loss rate.
Add options to:
1. Bypass optimization (use reported packet loss).
2. Set a maximum value.
3. Set a coefficient.

Bug: webrtc:9866
Change-Id: I3fef43e5186a4f0f50fda3506e445860518cfbd7
Reviewed-on: https://webrtc-review.googlesource.com/c/105304
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25161}
2018-10-15 08:59:43 +00:00
Bjorn Terelius
5350d1cafd RtcEventLogSource no longer uses deprecated parsing functions.
Also remove header extension map from NetEqEventLogInput and RtcEventLogSource.

Bug: webrtc:8111
Change-Id: Ic9be7b03e32ab8aa12284596e21e53b6763f483a
Reviewed-on: https://webrtc-review.googlesource.com/c/102622
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25122}
2018-10-11 16:13:17 +00:00
Jakob Ivarsson
88b68ace17 Create field trial for setting a minimum value for Opus encoder packet loss rate
Bug: webrtc:9848
Change-Id: I0663ee3af7729a220de7aff08cd74545e1a7409a
Reviewed-on: https://webrtc-review.googlesource.com/c/104800
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25081}
2018-10-10 08:52:56 +00:00
Patrik Höglund
7730193a49 Remove SetExecutablePath, simplify ResourcePath
SetExecutablePath isn't used anymore.

Nobody was using the fancy select-per-platform functionality, and the
documentation was wrong anyway. In the cases somebody needed an
override per platform, they were using defines in their own test
instead. I think that is more verbose but more predictable and easy
to understand (see how it's done in audio_processing_unittest.cc
when loading output_data_mac, for instance).

Bug: webrtc:9792
Change-Id: I7289bf5883fe43852638922d7c7583eae0c08601
Reviewed-on: https://webrtc-review.googlesource.com/c/104482
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25062}
2018-10-09 14:01:16 +00:00
Niels Möller
433eafe1f5 Delete unused includes of assert.h
Bug: None
Change-Id: Iadc531710dca0ba34a00ac363bfe0784355bb6f9
Reviewed-on: https://webrtc-review.googlesource.com/c/103501
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24995}
2018-10-04 14:01:44 +00:00
Niels Möller
eddd3665a2 Delete unused method AudioCodingModuleImpl::SetOpusApplication.
Bug: None
Change-Id: I8fc1b4b9a4521444867c8b34ee54187c86dd6027
Reviewed-on: https://webrtc-review.googlesource.com/c/102040
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24993}
2018-10-04 13:46:31 +00:00
Ivo Creusen
dc6d5533e1 Add more NetEq information to NetEqState.
Some important NetEq information was not available in NetEqState, which
meant it was not available on the API. This CL adds additional
information.

Bug: webrtc:9667
Change-Id: I702707c7d60472f488047d48fb286f839c5608dc
Reviewed-on: https://webrtc-review.googlesource.com/c/102300
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@{#24985}
2018-10-04 11:50:29 +00:00
Mirko Bonadei
8ca5c5216d Temporarily increase visibility of publicly used build targets.
Bug: webrtc:9808
Change-Id: I4ad2402dc288766732a2d81a289f717deec56629
Reviewed-on: https://webrtc-review.googlesource.com/c/103460
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24981}
2018-10-04 11:29:21 +00:00
Karl Wiberg
c2c4d042ae AudioCodingModuleTest.TestRedFec: Don't let the ACM create audio encoders
It will soon lose the ability to do so.

Bug: webrtc:8396
Change-Id: If120afa37325c00ae2c3e9a9bd75bf89c8897f8c
Reviewed-on: https://webrtc-review.googlesource.com/c/103441
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24979}
2018-10-04 11:20:57 +00:00
Pablo Barrera González
bc2959072d NetEq: Fix an UBSan error
UBSan will trigger when time_stretched_samples overflows using a
big number. This change avoids this problem by storing the
intermediate result into a int64_t.

Bug: chromium:886904
Change-Id: Id09dc4b468f841f03b523d5f21763f610b163a42
Reviewed-on: https://webrtc-review.googlesource.com/c/103123
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Pablo Barrera González <barrerap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24977}
2018-10-04 10:51:08 +00:00
Mirko Bonadei
311c13b3c2 Remove noop system_wrappers_default build target.
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}
2018-10-04 10:25:37 +00:00
Minyue Li
002fbb8c7d Adding field trial to force target level percentile in NetEQ.
Bug: webrtc:9822
Change-Id: I636f75de10851729825311ee5783e836f3b583cd
Reviewed-on: https://webrtc-review.googlesource.com/c/101220
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24975}
2018-10-04 10:00:54 +00:00
Minyue Li
7f6417f480 Restricting NetEq postpone decoding after expand.
Bug: webrtc:9289
Change-Id: I923f304e6c12423fe5323c62484a27346033b19a
Reviewed-on: https://webrtc-review.googlesource.com/c/98320
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24966}
2018-10-04 08:01:09 +00:00
Karl Wiberg
5cc8e14586 audio_coding_module_unittest: Don't rely on the ACM to create encoders
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}
2018-10-03 09:47:10 +00:00
Karl Wiberg
895ce82cab VAD/DTX tests: Don't let the ACM create audio encoders
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}
2018-10-02 08:19:32 +00:00
Karl Wiberg
3ff52ffa22 Remove the useless ACMTest base class
Bug: webrtc:8396
Change-Id: I021a2429910b21ffe4829e0ed51b9290bc715c0c
Reviewed-on: https://webrtc-review.googlesource.com/102884
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24907}
2018-10-01 12:01:44 +00:00
Mirko Bonadei
17f4878419 Remove deprecated field_trial_default and metrics_default.
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}
2018-09-28 07:21:07 +00:00
Karl Wiberg
91957c1540 AudioCodingModuleTest.TwoWayCommunication: Don't let the ACM create encoders
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}
2018-09-27 00:27:26 +00:00
Karl Wiberg
3a6b6bda17 AudioCodingModuleTest.TwoWayCommunication: Remove non-automatic mode
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}
2018-09-27 00:18:46 +00:00
Karl Wiberg
377a231c0d acm_receiver_unittest: Don't rely on the ACM to create encoders
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}
2018-09-25 17:16:56 +00:00
Karl Wiberg
bf7a0463da AudioCodingModuleTest.TestIsac: Don't rely on the ACM to create encoders
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}
2018-09-25 16:54:36 +00:00
Karl Wiberg
9a60e9a5b0 Remove the delay_test binary
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}
2018-09-25 08:25:56 +00:00
Karl Wiberg
d363db1907 TestStereo: Don't rely on the ACM to create encoders
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}
2018-09-24 15:11:05 +00:00
Karl Wiberg
36b37dce8f AudioCodingModuleTest.TestStereo: Delete write-only variables
Bug: webrtc:8396
Change-Id: I96c744c39ed15a2e20a45b120db9304dff486b76
Reviewed-on: https://webrtc-review.googlesource.com/101542
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24792}
2018-09-24 10:46:36 +00:00
Jonas Olsson
84df1c724e Make fewer copies when using StringBuilder.
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}
2018-09-24 09:39:19 +00:00
Danil Chapovalov
db1285676b Cleanup modules_common_types
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}
2018-09-18 08:08:33 +00:00
Henrik Lundin
00eb12a20c Let NetEq use the PLC output from a decoder
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}
2018-09-14 07:05:20 +00:00
Ivo Creusen
d1c2f78bfe Implement new stats interface on NetEq to monitor the operations and internal state.
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}
2018-09-13 14:03:47 +00:00
Minyue Li
1a80018a3c Avoid wrong parsing of padding length and its use in NetEq simulation.
Bug: b/113648474, webrtc:9730
Change-Id: Ieff7ab8697f5c8742548897a9b452a20b0bd2e7c
Reviewed-on: https://webrtc-review.googlesource.com/98461
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24703}
2018-09-12 11:23:03 +00:00
Ivo Creusen
f81b0f11a6 Move code for setting field trials from NetEqTestFactory to the main function in neteq_rtpplay.
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}
2018-09-11 09:27:11 +00:00
Mirko Bonadei
8b0aed1dd6 Fix no_global_constructors/no_exit_time_destructors in Neteq.
Bug: webrtc:9693
Change-Id: I0135e934c638ec391546928ba1e623d137b27b75
Reviewed-on: https://webrtc-review.googlesource.com/98600
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24668}
2018-09-11 06:39:14 +00:00
Henrik Lundin
9be7745509 NetEq tools: Fixing an issue with measuring the simulation time
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}
2018-09-10 16:16:22 +00:00
Ivo Creusen
4384f53285 Add more useful information to NetEqState and implement action_times_ms
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}
2018-09-10 09:10:53 +00:00
Niels Möller
fe3240aeae Reland "Delete class EventTimerWrapper."
This is a reland of a421775a6d

Original change's description:
> Delete class EventTimerWrapper.
>
> 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}

Tbr: henrik.lundin@webrtc.org
Bug: webrtc:3380
Change-Id: I541473b9c3ce2020f76d420598a7b10766f1d2a9
Reviewed-on: https://webrtc-review.googlesource.com/98481
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24620}
2018-09-07 09:54:55 +00:00
Jonas Olsson
366a50c4ef Remove simple stringstream usages.
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}
2018-09-06 12:53:19 +00:00
Mirko Bonadei
96ede16a4e Enable -Wexit-time-destructors and -Wglobal-constructors.
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}
2018-09-06 12:43:20 +00:00
Niels Moller
85e6e826a2 Revert "Delete class EventTimerWrapper."
This reverts commit a421775a6d.

Reason for revert: Depends on https://webrtc-review.googlesource.com/c/src/+/97320, which will be reverted due to breakage in video_engine_tests.

Original change's description:
> Delete class EventTimerWrapper.
> 
> 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}

TBR=henrika@webrtc.org,henrik.lundin@webrtc.org,nisse@webrtc.org

Change-Id: Iea92618c87cb4eb4595f22674528920171a9defb
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:3380
Reviewed-on: https://webrtc-review.googlesource.com/97681
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24551}
2018-09-04 11:50:03 +00:00
Mirko Bonadei
01cd853d2e Add nogncheck for headers of codecs not used in Chromium.
iLBC [1] and Red [2] are not used in Chromium, this means that WebRTC
doeesn't add the GN dependency on them but the include checker
complains because when it parses the code it sees iLBC and Red headers
included (the GN checker doesn't run the c preprocessor).

[1] - https://cs.chromium.org/chromium/src/.gn?l=62-65&rcl=3f6c31f0fdba128d810c4e3e391fae3b1aca7e7c
[2] - https://cs.chromium.org/chromium/src/third_party/webrtc/modules/audio_coding/audio_coding.gni?l=28-30&rcl=eb254b40b33380fcec43028dd89f3f6bab3d08a7

Bug: chromium:824831
Change-Id: I5059c1773fbe35f568c2fe3d8db3807f29973e7e
Reviewed-on: https://webrtc-review.googlesource.com/97620
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24546}
2018-09-04 09:45:56 +00:00
Niels Möller
a421775a6d Delete class EventTimerWrapper.
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}
2018-09-04 06:53:28 +00:00
Ivo Creusen
55de08e7ef Restructure neteq_rtpplay into a library with small executable wrapper.
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}
2018-09-03 10:42:40 +00:00
Alessio Bazzica
d4161a3c9d Moving LappedTransform, Blocker and AudioRingBuffer.
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}
2018-08-31 15:27:50 +00:00
Niels Möller
d941c09bc0 Delete unimplemented methods from the NetEq interface.
Bug: None
Change-Id: I51949a096c445813acc6649676e32c575732ef40
Reviewed-on: https://webrtc-review.googlesource.com/95643
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24469}
2018-08-28 15:48:26 +00:00
Niels Möller
764c14c87d Delete unused AudioCodingModule methods.
Methods deleted: IsCodecValid (static), QueryEncoder, SendFrequency.

Bug: None
Change-Id: Id63ea7cdc364583e896d3301d04fa9caae1e4d94
Reviewed-on: https://webrtc-review.googlesource.com/95486
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24440}
2018-08-27 10:00:59 +00:00
Niels Möller
18f1adc0da Delete AudioCodingModule::LeastRequiredDelayMs and related NetEq code.
Bug: None
Change-Id: I2f68502d19415899b3694f7bf5da523da831b223
Reviewed-on: https://webrtc-review.googlesource.com/95640
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24439}
2018-08-27 09:58:19 +00:00
Niels Möller
ec93075c00 Delete deprecated methods on AudioCodingModule
Bug: None
Change-Id: I05f1ab6cdd6ac52972835af7ea94aacf5f64d673
Reviewed-on: https://webrtc-review.googlesource.com/95482
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24388}
2018-08-22 13:26:17 +00:00
Karl Wiberg
801500cf99 Audio encoder tests: Create audio encoders the new way
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}
2018-08-22 07:48:55 +00:00
Karl Wiberg
658a552fd5 Audio encoder tests: Create audio encoders the new way
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}
2018-08-17 06:38:09 +00:00
Minyue Li
c97933fb82 Clean up code regarding jitter buffer plot in event log visualizer.
Bug: webrtc:9147
Change-Id: I2c1f0b383706ae9a788eb8b5d308d4c7fe612730
Reviewed-on: https://webrtc-review.googlesource.com/92390
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24261}
2018-08-10 11:19:56 +00:00
Ivo Creusen
80006b9922 Add command-line flag to enable the bugfix to postpone decoding after expand.
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}
2018-08-10 10:06:56 +00:00
Karl Wiberg
133cff009b AudioCodingModuleTest.TestAllCodecs: Create audio encoders the new way
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}
2018-08-08 01:38:05 +00:00
Oleh Prypin
d2f4e8bd90 Explicitly add -mfpu=neon to all targets that use NEON
Remove obsolete comment about Chromium not defining NEON for Android.

Semi-related fix: don't use `rtc_remove_configs` directly, `suppressed_configs` is the "public interface".

Bug: webrtc:9579
Change-Id: I512628feb462a29432f1356cfef00efe1ddaf84f
Reviewed-on: https://webrtc-review.googlesource.com/91761
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24165}
2018-08-01 13:15:42 +00:00
Artem Titov
75caa597a3 Untangle fft third party lib from dependon WebRTC
TBR=phoglund

Bug: webrtc:9558
Change-Id: I6cc1936549f008694c3617c1d990524c34da16e3
Reviewed-on: https://webrtc-review.googlesource.com/90411
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24115}
2018-07-26 13:44:30 +00:00
Niels Möller
a15fd0dee6 Add missing include of stdint.h in MIPS code.
Needed after cl https://webrtc-review.googlesource.com/c/src/+/90249,
which deleted the include of typedefs.h.

Bug: webrtc:6854
Change-Id: I4ab86fae40843613a76da378658343198a800d0c
Reviewed-on: https://webrtc-review.googlesource.com/90414
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24112}
2018-07-26 11:30:19 +00:00
Niels Möller
a12c42a6b2 Delete root header file typedef.h.
Usage replaced with stdint.h, rtc_base/system/arch.h and
rtc_base/system/unused.h, as appropriate.

Bug: webrtc:6854
Change-Id: I97225465d14b969903d92979e2df3c3c05d35f18
Reviewed-on: https://webrtc-review.googlesource.com/90249
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24100}
2018-07-25 14:59:26 +00:00
Artem Titov
52b9000380 Move g722 to proper third_party directory
Bug: webrtc:8366
Change-Id: I81b051dd25da2d7eaa2902af284d8b669ad8e3c9
Reviewed-on: https://webrtc-review.googlesource.com/85620
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24096}
2018-07-25 11:56:59 +00:00
Artem Titov
e095b81940 Move g711 to proper third_party directory
Bug: webrtc:8366
Change-Id: Ic57bd5c5c01871aee2956b2a098a79b106f54c9e
Reviewed-on: https://webrtc-review.googlesource.com/85375
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24095}
2018-07-25 10:27:08 +00:00
Artem Titov
8a838fd207 Move fft to proper third_party directory
Bug: webrtc:8366
Change-Id: I741a381fe1cf18909baefd89743b2ff4fe0a6bae
Reviewed-on: https://webrtc-review.googlesource.com/86822
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24091}
2018-07-25 08:39:28 +00:00
Artem Titov
5d7a4c6692 Fixing py lint errors
Bug: webrtc:9548
Change-Id: I0daf8dc06fdaac1637c32994ef6ad542ed52202a
Reviewed-on: https://webrtc-review.googlesource.com/90045
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24068}
2018-07-23 15:28:48 +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
Mirko Bonadei
216664ab13 Cleanup unneeded includes in audio_coding/BUILD.gn.
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}
2018-07-18 15:16:29 +00:00
Sam Zackrisson
3f84f498e4 Remove useless import of arm.gni
Bug: None
Change-Id: I439410d9edf306b664ef21157216870d6e1c8207
Reviewed-on: https://webrtc-review.googlesource.com/87436
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23953}
2018-07-12 14:39:00 +00:00
Danil Chapovalov
065a52a655 Reland "Remove rtc::Optional alias and api:optional target"
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}
2018-07-11 19:02:51 +00:00
Ilya Nikolaevskiy
b661c658da Revert "Remove rtc::Optional alias and api:optional target"
This reverts commit 6f5b0f920a.

Reason for revert: Breaks internal project.

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}

TBR=danilchap@webrtc.org,kwiberg@webrtc.org

Change-Id: I95f5ec33520b823c3d0c9cb83d945d6a15355367
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9078
Reviewed-on: https://webrtc-review.googlesource.com/88140
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23921}
2018-07-11 07:41:41 +00:00
Danil Chapovalov
6f5b0f920a 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}
2018-07-10 18:02:23 +00:00
Minyue Li
99fb004f0d Remove a legacy DCHEC in FakeDecodeFromFile.
Bug: None
Change-Id: Ia76bf18eb228b658d0a7146cdb6e46586b3507a0
Reviewed-on: https://webrtc-review.googlesource.com/87435
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23890}
2018-07-09 19:56:58 +00:00
Minyue Li
9a94057a79 Making PacketDuration always consistent with Decode in FakeDecodeFromFile.
Bug: None
Change-Id: Ib34efd629009075fdc793ab041296d2814c9677e
Reviewed-on: https://webrtc-review.googlesource.com/87380
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23874}
2018-07-06 13:30:47 +00:00
Karl Wiberg
918f50c5d1 Use absl::make_unique and absl::WrapUnique directly
Instead of going through our wrappers in ptr_util.h.

This CL was generated by the following script:

  git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",'
  git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g'
  git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g'
  git checkout -- rtc_base/ptr_util{.h,_unittest.cc}
  git cl format

Followed by manually adding dependencies on
//third_party/abseil-cpp/absl/memory until `gn check` stopped
complaining.

Bug: webrtc:9473
Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c
Reviewed-on: https://webrtc-review.googlesource.com/86600
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 10:59:49 +00:00
Henrik Lundin
defa7a8049 NetEq: Handle nested RED packets
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}
2018-07-03 20:27:57 +00:00
Henrik Lundin
5afa61cf15 NetEq: Fold GetDecisionSpecialized into GetDecision
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}
2018-07-02 14:51:09 +00:00
Henrik Lundin
9f2e624024 Break out NetEqEventLogInput to separate source files
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}
2018-07-02 14:15:29 +00:00
Henrik Lundin
7687ad58b2 Reland "NetEq: Deprecate playout modes Fax, Off and Streaming"
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}
2018-07-02 10:20:33 +00:00
Minyue Li
a91decab4f Implement PacketDuration() for FakeDecoderFromFile.
Bug: None
Change-Id: Ie4ab1ce737608706f12f298f793f76571805ca91
Reviewed-on: https://webrtc-review.googlesource.com/86160
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23780}
2018-06-29 08:32:36 +00:00
Artem Titov
d9711098b0 Extract fft to separate target to be able to move it to third_party
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}
2018-06-27 09:08:19 +00:00
Minyue Li
c9ac93fabb Adding NetEq lifetime stats to event log visualizer.
Bug: webrtc:9147
Change-Id: I798f8ac41192182d50df6fe98fbe56c8cb7f294c
Reviewed-on: https://webrtc-review.googlesource.com/85340
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23738}
2018-06-26 11:27:09 +00:00
Minyue Li
f7789c6e89 Limiting increment in timestamps with neteq simulation.
Bug: None
Change-Id: I9a0688bcf1c887793b5c94ea023b025aed7366a5
Reviewed-on: https://webrtc-review.googlesource.com/74840
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23733}
2018-06-26 08:07:38 +00:00
Artem Titov
91280e4d04 Extract third party part of g722 codec into separate target
Bug: webrtc:8366
Change-Id: I7e08aa53424afd3001f4c22be270a8b0ff7af565
Reviewed-on: https://webrtc-review.googlesource.com/84744
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23725}
2018-06-25 11:30:59 +00:00
Artem Titov
3ecec176a8 Extract third party part of g711 codec into separate target
Bug: webrtc:8366
Change-Id: I34c7ea707213e0c1a50826896da01f70c072eae5
Reviewed-on: https://webrtc-review.googlesource.com/84741
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23724}
2018-06-25 11:26:59 +00:00
Minyue Li
45fc6dfaaa Aligning time in audio jitter buffer plot to other plots in rtc event log visualizer.
Bug: webrtc:9147
Change-Id: I4ddb3e93ea04a11a68e097ecad731d6d9d6842a9
Reviewed-on: https://webrtc-review.googlesource.com/75322
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23712}
2018-06-21 14:23:53 +00:00
Henrik Lundin
1ff41eb784 Revert "NetEq: Deprecate playout modes Fax, Off and Streaming"
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}
2018-06-21 12:36:44 +00:00
Henrik Lundin
80c4cca491 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}
2018-06-21 11:51:21 +00:00