Commit graph

105 commits

Author SHA1 Message Date
Jared Siskin
c018bae807 Format /modules
git ls-files | grep -e  "\(\.h\|\.cc\)$" | grep -e  "^modules/" | xargs clang-format -i ; git cl format
after landing: add to .git-blame-ignore-revs

Bug: webrtc:15082
Change-Id: I2c3cd28740062794f8c10e39d8406aadb9e9a35a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301620
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Jared Siskin <jtsiskin@meta.com>
Cr-Commit-Position: refs/heads/main@{#39901}
2023-04-20 02:02:45 +00:00
Henrik Lundin
84f75699c6 Break apart AudioCodingModule and AcmReceiver
This change makes AudioCodingModule a pure sender and AcmReceiver a pure
receiver.

The Config struct is in practice no longer used by AudioCodingModule,
so a new definition is included in AcmReceiver. The old definition
remains in AudioCodingModule while downstream clients are being
updated.

Bug: webrtc:14867
Change-Id: If0d0b4214c5aa278cf6c85c5b62c6da644de20e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291533
Reviewed-by: Tomas Lundqvist <tomasl@google.com>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39244}
2023-02-01 16:09:26 +00:00
Jakob Ivarsson
757da3cf70 Stop setting OPUS_SIGNAL_VOICE when DTX is enabled.
This was done in crbug.com/webrtc/4559 since "CELT-only mode does not have DTX", but that should not be the case anymore (support was added in Opus v1.2.1).

One exception where DTX does not work is with OPUS_APPLICATION_AUDIO (used with stereo) and low complexity settings. This should not be a common config.

Bug: None
Change-Id: I1476083b836bcabeb73df83d5bf06c3878146d28
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288420
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38923}
2022-12-20 11:06:48 +00:00
Alessio Bazzica
17887eb04a Reland "[ACM] iSAC audio codec removed"
This is a reland of commit b46c4bf27b

Original change's description:
> [ACM] iSAC audio codec removed
>
> Note: this CL has to leave behind one part of iSAC, which is its VAD
> currently used by AGC1 in APM. The target visibility has been
> restricted and the VAD will be removed together with AGC1 when the
> time comes.
>
> Tested: see https://chromium-review.googlesource.com/c/chromium/src/+/4013319
>
> Bug: webrtc:14450
> Change-Id: I69cc518b16280eae62a1f1977cdbfa24c08cf5f9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282421
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38652}

Bug: webrtc:14450
Change-Id: Ia22c4d7724b6022238235fede93e36e570a49376
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283843
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38665}
2022-11-17 12:52:35 +00:00
Alessio Bazzica
fbeb76ab51 Revert "[ACM] iSAC audio codec removed"
This reverts commit b46c4bf27b.

Reason for revert: breaks a downstream project

Original change's description:
> [ACM] iSAC audio codec removed
>
> Note: this CL has to leave behind one part of iSAC, which is its VAD
> currently used by AGC1 in APM. The target visibility has been
> restricted and the VAD will be removed together with AGC1 when the
> time comes.
>
> Tested: see https://chromium-review.googlesource.com/c/chromium/src/+/4013319
>
> Bug: webrtc:14450
> Change-Id: I69cc518b16280eae62a1f1977cdbfa24c08cf5f9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282421
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38652}

Bug: webrtc:14450
Change-Id: Ice138004e84e8c5f896684e8d01133d4b2a77bb7
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283800
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#38655}
2022-11-16 20:40:52 +00:00
Alessio Bazzica
b46c4bf27b [ACM] iSAC audio codec removed
Note: this CL has to leave behind one part of iSAC, which is its VAD
currently used by AGC1 in APM. The target visibility has been
restricted and the VAD will be removed together with AGC1 when the
time comes.

Tested: see https://chromium-review.googlesource.com/c/chromium/src/+/4013319

Bug: webrtc:14450
Change-Id: I69cc518b16280eae62a1f1977cdbfa24c08cf5f9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282421
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38652}
2022-11-16 16:42:55 +00:00
Ali Tofigh
714e3cbb48 Adopt absl::string_view in modules/audio_coding/
Bug: webrtc:13579
Change-Id: Ifec66fb6ba9724d18539de7245a358c2d13c7939
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268547
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37573}
2022-07-20 13:34:23 +00:00
Byoungchan Lee
f65d735e7d Remove ACMTestTimer in iSACTest
It hasn't been used in years.

Bug: chromium:1331345
Change-Id: I8fdc1952fa1114f7f78e2535ffb76e9678e53d1d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265520
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37164}
2022-06-09 12:14:02 +00:00
Byoungchan Lee
1abcb1106c Remove usage of sprintf in modules
sprintf is marked as deprecated with Xcode 14.

Bug: chromium:1331345
Change-Id: I834f392bee96e6b6725d5aee469a243dbc6e272e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265521
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37162}
2022-06-09 11:57:33 +00:00
Niels Möller
e66b83f8ad Never pass a signed char to ctype macros like isdigit()
Bug: None
Change-Id: I451bb2c1f175a77aefbc8363009bf35a769fe941
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264442
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37037}
2022-05-30 13:05:03 +00:00
Niels Möller
ea1e6f44f8 Delete rtc_base/format_macros.h
It defined RTC_PRIuS, which was needed for compatibility with MSVC
prior to version 2015.

Bug: webrtc:6424
Change-Id: I5668d473376201cad3e8da65927c967fc397804b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261314
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36814}
2022-05-09 12:03:21 +00:00
Jonas Oreland
a943e730b2 WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 7/inf
Convert audio/ and collateral (audio encoder copy red).

Bug: webrtc:10335
Change-Id: Iac54c0cfd2f62f4402f3deec35ae2725ec35b81a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255820
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36229}
2022-03-17 07:11:44 +00:00
Ali Tofigh
62238097c9 Remove top-level const from parameters in function declarations.
This is a safe cleanup change since top-level const applied to
parameters in function declarations (that are not also
definitions) are ignored by the compiler. Hence, such changes do
not change the type of the declared functions and are simply
no-ops.

Bug: webrtc:13610
Change-Id: Ibafb92c45119a6d8bdb6f9109aa8dad6385163a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249086
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35802}
2022-01-26 11:05:25 +00:00
Anton Bikineev
7abf45fe2c LSC: Apply clang-tidy's modernize-use-bool-literals
The check finds implicit conversions of integer literals to bools:
  bool b1 = 1;
  bool b2 = static_cast<bool>(1);
and transforms them to:
  bool b1 = true;
  bool b2 = true;

Bug: chromium:1290142
Change-Id: I6819a0bd2ca84ecadae08ed9389c17d2652589f4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/248166
Auto-Submit: Anton Bikineev <bikineev@chromium.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Anton Bikineev <bikineev@chromium.org>
Cr-Commit-Position: refs/heads/main@{#35778}
2022-01-24 20:42:01 +00:00
Mirko Bonadei
e5e78c4521 Fix -Wunused-but-set-variable.
Bug: None
Change-Id: I8943227108e46c4c942895e4bd8fb276947502e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236525
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35272}
2021-10-28 12:53:49 +00:00
Artem Titov
d00ce747c7 Use backticks not vertical bars to denote variables in comments for /modules/audio_coding
Bug: webrtc:12338
Change-Id: I02613d9fca45d00e2477f334b7a0416e7912e26b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227037
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34621}
2021-08-02 10:45:40 +00:00
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
Minyue Li
c940870b72 Revert "opus: take SILK vad result into account for voice detection"
This reverts commit 686a3709ac.

Reason for revert: crbug.com/1144220

Original change's description:
> opus: take SILK vad result into account for voice detection
>
> BUG=webrtc:11643
>
> Change-Id: Idc3a9b6bb7bd1a33f905843e5d6067ae19d5172c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176508
> Commit-Queue: Minyue Li <minyue@webrtc.org>
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31743}

TBR=devicentepena@webrtc.org,minyue@webrtc.org,fippo@sip-communicator.org

Bug: webrtc:11643
Change-Id: I9c77e4f6e919c4b648a5783edf4188e1f8114602
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191485
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32542}
2020-11-04 07:29:48 +00:00
Niels Möller
84d360771b Fix locking in RTPFile class
This code used to have a reader-writer lock, and call
std::queue::pop() with only a reader lock, which appears unsafe. Code
changed to use a plain webrtc::Mutex.

Bug: webrtc:12102
Change-Id: Icbea17a824c91975dfebd4d05bbd0c21e1abeadc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190700
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32511}
2020-10-28 15:19:06 +00:00
Niels Möller
7c85d395d7 Delete unneeded includes of system_wrappers/include/sleep.h
Non-test usage is in modules/audio_device and modules/desktop_capture.

Bug: None
Change-Id: Ie7dd89aa40e6dcfa9e49e1956b87b50fd9f1c227
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190140
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32484}
2020-10-26 09:55:26 +00:00
Philipp Hancke
fc4668dae2 configure target bitrate in opus dtx tests
This avoids a difference in behaviour between mobile and
desktop platforms since the bitrate is now too low for
CELT mode.

BUG=webrtc:11643

Change-Id: I9ac1439bea0ccbbfee7388516932e30d6cb06bf4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179522
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#31757}
2020-07-17 11:43:45 +00:00
Philipp Hancke
686a3709ac opus: take SILK vad result into account for voice detection
BUG=webrtc:11643

Change-Id: Idc3a9b6bb7bd1a33f905843e5d6067ae19d5172c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176508
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31743}
2020-07-16 11:37:35 +00:00
Markus Handell
0df0faefd5 Migrate modules/audio_coding, audio_mixer/ and audio_processing/ to webrtc::Mutex.
Bug: webrtc:11567
Change-Id: I03b78bd2e411e9bcca199f85e4457511826cd17e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176745
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31649}
2020-07-07 14:35:58 +00:00
Per Åhgren
d82a02c837 ACM: Corrected temporary buffer size
This CL corrects the temporary buffers size in the
pre-processing of the capture audio before encoding.

As part of this it removes the ACM-specific hardcoding
of the size and instead ensures that the size of the
temporary buffer matches that of the AudioFrame.

Bug: webrtc:11242
Change-Id: I56dd6cadfd4e140e8e159966c33d1027383ea9fa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170340
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30775}
2020-03-12 12:23:20 +00:00
Niels Möller
3bdc5e9a5f Delete ACMVADCallback
This callback is enabled via the method
AudioCodingModule::RegisterVADCallback, which is unused, and deleted
in this cl.

Bug: None
Change-Id: I04c8690fbb673305e69fe5b1c32d88efd6c72d1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148420
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30735}
2020-03-10 09:53:46 +00:00
Minyue Li
ff0e4dbd1f Reland "Send absolute capture time through audio coding module."
This is a reland of 48655cfdbf

Original change's description:
> Send absolute capture time through audio coding module.
>
> Bug: webrtc:10739
> Change-Id: I44e0305be7c84b253172511c2977b1d700e40c1b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167067
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Chen Xing <chxg@google.com>
> Commit-Queue: Minyue Li <minyue@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30363}

Bug: webrtc:10739
Change-Id: I10086d239ad3f1efb8485098bf3b0ad23110962b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167213
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Chen Xing <chxg@google.com>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30380}
2020-01-27 13:18:27 +00:00
Minyue Li
4175914f41 Revert "Send absolute capture time through audio coding module."
This reverts commit 48655cfdbf.

Reason for revert: failing upstream tests

Original change's description:
> Send absolute capture time through audio coding module.
> 
> Bug: webrtc:10739
> Change-Id: I44e0305be7c84b253172511c2977b1d700e40c1b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167067
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Chen Xing <chxg@google.com>
> Commit-Queue: Minyue Li <minyue@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30363}

TBR=danilchap@webrtc.org,ossu@webrtc.org,minyue@webrtc.org,chxg@google.com

Change-Id: Ia36b9ae899563c9afd8612ffd83871b8a5778a2b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10739
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167212
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30364}
2020-01-23 16:21:06 +00:00
Minyue Li
48655cfdbf Send absolute capture time through audio coding module.
Bug: webrtc:10739
Change-Id: I44e0305be7c84b253172511c2977b1d700e40c1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167067
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Chen Xing <chxg@google.com>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30363}
2020-01-23 16:06:12 +00:00
Minyue Li
8e83c7ac09 Make Opus PLC always output 10ms audio.
BUG: b/143582588
Change-Id: I41ad5f4f91d9af3f595666a8f32b7ab5382605bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158672
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29733}
2019-11-07 21:15:58 +00:00
Minyue Li
fb075d558d Removing unused Opus wrapper API: WebRTCOpus_DecodePlc.
Bug: None
Change-Id: I5b613b4c13ec5f6ad13d8430043d006f6d83c11f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158671
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29664}
2019-10-31 12:01:31 +00:00
Niels Möller
834a554962 Include module_common_types.h only where needed
Bug: None
Change-Id: I73d493f8f186b429c7be808f4dfac0398f150931
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153891
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29277}
2019-09-24 08:22:38 +00:00
Mirko Bonadei
317a1f09ed Use std::make_unique instead of absl::make_unique.
WebRTC is now using C++14 so there is no need to use the Abseil version
of std::make_unique.

This CL has been created with the following steps:

git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt
git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt
git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt

diff --new-line-format="" --unchanged-line-format="" \
  /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \
  uniq > /tmp/only_make_unique.txt
diff --new-line-format="" --unchanged-line-format="" \
  /tmp/only_make_unique.txt /tmp/memory.txt | \
  xargs grep -l "absl/memory" > /tmp/add-memory.txt

git grep -l "\babsl::make_unique\b" | \
  xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g"

git checkout PRESUBMIT.py abseil-in-webrtc.md

cat /tmp/add-memory.txt | \
  xargs sed -i \
  's/#include "absl\/memory\/memory.h"/#include <memory>/g'
git cl format
# Manual fix order of the new inserted #include <memory>

cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \
  xargs sed -i '/#include "absl\/memory\/memory.h"/d'

git ls-files | grep BUILD.gn | \
  xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d'

python tools_webrtc/gn_check_autofix.py \
  -m tryserver.webrtc -b linux_rel

# Repead the gn_check_autofix step for other platforms

git ls-files | grep BUILD.gn | \
  xargs sed -i 's/absl\/memory:memory/absl\/memory/g'
git cl format

Bug: webrtc:10945
Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 15:47:29 +00:00
Jakob Ivarsson
507f43465b Reland "Make relative arrival delay mode default in NetEq delay manager."
This is a reland of 77c71d1488

Original change's description:
> Make relative arrival delay mode default in NetEq delay manager.
> 
> Bug: webrtc:10333
> Change-Id: I9b1e0bec0b1813cf31259492f83eb2ca86a44d3f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150782
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29075}

Bug: webrtc:10333
Change-Id: I9c726cec1afc1147a4618fc224404a83962e6ae2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152281
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29136}
2019-09-10 14:05:48 +00:00
Alessio Bazzica
5b728cca77 Revert "Make relative arrival delay mode default in NetEq delay manager."
This reverts commit 77c71d1488.

Reason for revert: breaking downstream projects

Original change's description:
> Make relative arrival delay mode default in NetEq delay manager.
> 
> Bug: webrtc:10333
> Change-Id: I9b1e0bec0b1813cf31259492f83eb2ca86a44d3f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150782
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29075}

TBR=henrik.lundin@webrtc.org,srte@webrtc.org,minyue@webrtc.org,jakobi@webrtc.org

Change-Id: I67c5b9c7a6e854d3aac379aa4d98bfeb5425d312
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10333
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151642
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29078}
2019-09-05 11:59:53 +00:00
Jakob Ivarsson
77c71d1488 Make relative arrival delay mode default in NetEq delay manager.
Bug: webrtc:10333
Change-Id: I9b1e0bec0b1813cf31259492f83eb2ca86a44d3f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150782
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29075}
2019-09-05 09:15:47 +00:00
Niels Möller
5ceb4ac5ed Delete some unused AudioCodingModule methods
Methods deleted:

  ReceiveFrequency, PlayoutFrequency, ReceiveCodec,
  SetMinimumPlayoutDelay, SetMaximumPlayoutDelay,
  SetBaseMinimumPlayoutDelayMs, GetBaseMinimumPlayoutDelayMs,
  PlayoutTimestamp, FilteredCurrentDelayMs, TargetDelayMs.

Became unused with cl
https://webrtc-review.googlesource.com/c/src/+/111504

Bug: None
Change-Id: Ie50e8e86a622661c3daa9db83a2e66489dcd2d98
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148071
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28918}
2019-08-20 13:42:36 +00:00
Niels Möller
b90d38a978 Delete unused Opus-specific methods of AudioCodingModule
Bug: None
Change-Id: Ib191e4beadf85cd57e765bc52d305e274e50a473
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148400
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28815}
2019-08-09 07:06:36 +00:00
Jonas Olsson
a4d873786f Format almost everything.
This CL was generated by running

git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \
grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \
grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \
grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \
grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \
| xargs clang-format -i ; git cl format

Most of these changes are clang-format grouping and reordering includes
differently.

Bug: webrtc:9340
Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28505}
2019-07-08 13:45:15 +00:00
Karl Wiberg
a1d1a1e976 WebRTC Opus C interface: Add support for non-48 kHz decode sample rate
Plus tests for 16 kHz.

Bug: webrtc:10631
Change-Id: I2d89bc6d0d9548f0ad7bb1e36d6dfde6b6b31f83
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138072
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28099}
2019-05-29 10:33:03 +00:00
Karl Wiberg
7e7c5c3c25 WebRTC Opus C interface: Add support for non-48 kHz encode sample rate
Plus tests fo 16 kHz.

Bug: webrtc:10631
Change-Id: I162c40b6120d7e308e535faba7501e437b0b5dc4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137047
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28029}
2019-05-22 22:56:58 +00:00
Niels Möller
c35b6e675a Deprecate RTPFragmentationHeader argument to AudioPacketizationCallback::SendData
It appears unused everywhere. It will be deleted in a followup cl.

Bug: webrtc:6471
Change-Id: Ief992db6e52aee3cf1bc77ffd659ffbc072672ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134212
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27787}
2019-04-26 12:58:14 +00:00
Niels Möller
8f7ce222e7 Make VideoFrameType an enum class, and move to separate file and target
Bug: webrtc:5876, webrtc:6883
Change-Id: I1435cfa9e8e54c4ba2978261048ff3fbb993ce0e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126225
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27239}
2019-03-22 12:44:51 +00:00
Niels Möller
c936cb6a86 Make AudioFrameType an enum class, and move to audio_coding_module_typedefs.h
Bug: webrtc:5876
Change-Id: I0c92f9410fcf0832bfa321229b3437134255dba6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128085
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27190}
2019-03-19 16:59:27 +00:00
Niels Möller
87e2d785a0 Prepare for splitting FrameType into AudioFrameType and VideoFrameType
This cl deprecates the FrameType enum, and adds aliases AudioFrameType
and VideoFrameType.

After downstream usage is updated, the enums will be separated
and be moved out of common_types.h.

Bug: webrtc:6883
Change-Id: I2aaf660169da45f22574b4cbb16aea8522cc07a6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123184
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27011}
2019-03-07 10:12:57 +00:00
Mirko Bonadei
c4dd730765 Fix -Wextra-semi warnings.
Starting from https://chromium-review.googlesource.com/c/1485012,
-Wextra-semi is enabled and WebRTC has some violations to fix.

This is a follow-up of https://webrtc-review.googlesource.com/c/123560.

Bug: webrtc:10355
Change-Id: I012b7497fc8991037fd77aa98f1579c22e08206f
Reviewed-on: https://webrtc-review.googlesource.com/c/124126
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26831}
2019-02-25 09:22:51 +00:00
Niels Möller
bf47495979 Update remaining audio test code to not use WebRtcRTPHeader.
Bug: webrtc:5876
Change-Id: I5b1abcec4a0ef52b6dd36d1fe94dbfd3f88f28a7
Reviewed-on: https://webrtc-review.googlesource.com/c/123235
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26736}
2019-02-18 13:29:35 +00:00
Niels Möller
afb5dbbf4e Update ACM to use RTPHeader instead of WebRtcRTPHeader
Bug: webrtc:5876
Change-Id: Id3311dcf508cca34495349197eeac2edf8783772
Reviewed-on: https://webrtc-review.googlesource.com/c/123188
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26729}
2019-02-18 08:01:31 +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
Steve Anton
40d55331d7 Include absl/memory/memory.h if absl::make_unique is used
Tbr: kwiberg@webrtc.org
Bug: None
Change-Id: Iaf4533d2ce0e80b351a8a664ef8cf7ba0e5ec583
Reviewed-on: https://webrtc-review.googlesource.com/c/115746
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@google.com>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26168}
2019-01-08 20:08:32 +00:00