Commit graph

129 commits

Author SHA1 Message Date
Rashad Sookram
147fdb9f46 Merge branch 'm112' into 5615 2023-04-27 12:45:13 -04:00
Rashad Sookram
03ddb5df82 Merge branch 'm110' into 5481 2023-02-17 11:35:29 -05:00
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
Per Kjellander
e0b4cab69c Remove default enabled field trial WebRTC-SendSideBwe-WithOverhead
Bug: webrtc:6762
Change-Id: I520188a13ee5f50c441226574ccb3df54f842835
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285300
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38783}
2022-11-30 20:19:36 +00:00
Jakob Ivarsson
918eb19303 Fix crash when Opus maxptime < 20ms.
A follow up cl will be created to better handle nullopt frame length range in AudioSendStream.

Note that maxptime is still not used for setting the frame length (only ptime is).

Bug: chromium:1109337
Change-Id: Id21fd8c76a6c4a0c85719a955116f8d16001a3d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284501
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38702}
2022-11-22 01:21:24 +00:00
Rashad Sookram
2e9c66e1b1 Finish resolving merge conflicts 2022-11-11 19:10:59 -05:00
Rashad Sookram
bd151651d3 Merge in M108 2022-11-11 17:02:35 -05:00
Artem Titov
e39115a0ca Migrate audio perf tests on new perf metrics export API
Bug: b/246095034
Change-Id: Id659e43c116428cab47d334c93a6036f74dbb8e1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276626
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38192}
2022-09-25 18:55:50 +00:00
Rashad Sookram
9ed4194e49
Update to 5005 (M102) (#86) 2022-08-24 11:07:33 -04:00
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
Oleh Prypin
cc7bd85748 Don't add libopus to public_deps, its headers are only used directly
Bug: webrtc:8603
Change-Id: I2ce1f96a80dd23e420b3693b899d2b14382fd2d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266765
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Oleh Prypin <oprypin@google.com>
Cr-Commit-Position: refs/heads/main@{#37363}
2022-06-28 19:13:14 +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
Florent Castelli
c3e6e3a3e8 Remove dependency on rtc_base_approved from most targets
Bug: webrtc:9838
Change-Id: Ibd0199803597eff48ca139a5cecdc3209c62c5d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259873
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36643}
2022-04-25 12:15:30 +00:00
Peter Thatcher
4a2e0e5d45
Update to 4896 (M100) (#72) 2022-04-15 17:13:23 -06:00
Florent Castelli
4467ad7835 Remove //rtc_base:macromagic from public deps
Bug: webrtc:8603
Change-Id: I9708df48c9bde9f86ba2d1a92a278bb0d09f3865
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257909
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36444}
2022-04-05 12:36:12 +00:00
Byoungchan Lee
604fd2f1ab Remove RTC_DISALLOW_COPY_AND_ASSIGN from modules/
Bug: webrtc:13555, webrtc:13082
Change-Id: I2c2cbcbd918f0cfa970c1a964893220ba11d4b41
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247960
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: (Daniel.L) Byoungchan Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#35771}
2022-01-24 11:50:20 +00:00
Ivo Creusen
deb1b1bc70 Always call IsOk() to ensure audio codec configuration is valid when negotiating.
We should avoid creating codecs with invalid parameters, since this can
expose security issues. For many codecs the IsOk() method to check the
codec config is only called in DCHECKs. This CL ensures IsOk() is always
called, also in non-debug builds.

Bug: chromium:1265806
Change-Id: Ibd3c6c65d3bb547cd2603e11808ac40ac693a8b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238801
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35422}
2021-11-26 10:11:21 +00:00
Peter Thatcher
4af16890e5 Fix AudioEncoder 2021-10-29 09:07:06 -06:00
Peter Thatcher
1a0b210a9d ks 2021-10-29 07:49:26 -06:00
Artem Titov
cfea2182f8 Use backticks not vertical bars to denote variables in comments
Bug: webrtc:12338
Change-Id: I89c8b3a328d04203177522cbdfd9e606fd4bce4c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228246
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34696}
2021-08-10 10:40:03 +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
Jesús de Vicente Peña
d674ec77af Not dropping the refresh DTX packets but substituting them by 1 byte packets.
Bug: webrtc:12380
Change-Id: I27029c591ac2555d6ae61b706adcf97c9498a9fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217880
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33983}
2021-05-11 19:47:34 +00:00
Yura Yaroshevich
d46a174f0c Expose adaptive_ptime from iOS SDK.
Bug: None
Change-Id: I48fd0937f51dc972b3eccd66f99ae80378e32fe1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214968
Commit-Queue: Yura Yaroshevich <yura.yaroshevich@gmail.com>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33766}
2021-04-18 21:53:32 +00:00
Peter Thatcher
13cb722ac9
Update to WebRTC 4389 (e7d9f74)
Contains changes for M86 (4240), M87 (4280), M88 (4324), and M89 (4389).
2021-04-16 13:26:31 -07:00
Jesús de Vicente Peña
3b9abd8dee Avoiding the noise pumping during DTX regions by just forwarding the refresh DTX packets that decrease the comfort noise level at the decoder.
Bug: webrtc:12380
Change-Id: I60e4684150cb4880224f402a9bf42a72811863b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202920
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33174}
2021-02-05 10:05:25 +00:00
Peter Thatcher
52871c1f98
Avoid CHECK when decreasing the ptime (#44) 2021-01-22 11:08:46 -08:00
Peter Thatcher
22490bf0b9
Fix configuration of ptime so it's not overridden in the future (#43) 2021-01-22 09:41:43 -08:00
Peter Thatcher
84775e0ab6
Allow control of min and max bitrates (#42) 2021-01-21 10:49:55 -08:00
Peter Thatcher
f3ce387c3b
Make audio encoder changes more resilient to BWE changes (#41) 2021-01-20 16:58:31 -08:00
Peter Thatcher
1ff88760c8
Add PeerConnection::ConfigureAudioEncoders (#40)
Allows more dynamic, easy, and precise control of OPUS from PeerConnection.
2021-01-20 11:52:27 -08:00
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
Jakob Ivarsson
36274f9158 Reland "Reland "Default enable WebRTC-SendSideBwe-WithOverhead.""
This is a reland of 1dbe30c7e8

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

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

Reason for revert: Speculative revert due to failing tests.

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

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

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

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

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

Reason for revert: breaks downstream tests

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

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

Change-Id: If59fd41dcd8f6db76ea297c34c25fe19ae2ae973
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:6762
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189973
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32474}
2020-10-22 16:57:18 +00:00
Jakob Ivarsson
87c1950841 Default enable WebRTC-SendSideBwe-WithOverhead.
Bug: webrtc:6762
Change-Id: I18ace06a33b3b60d5a19796d4769f70cd977d604
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188801
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32472}
2020-10-22 13:37:18 +00:00
Jeremy Leconte
c8850cbf55 Change gtest name to allow filtering based on the story name.
It is meant for Pinpoint to run only the relevant tests when running a bisection.
The Pinpoint side of this change can be found here:
https://crrev.com/c/2404161

Bug: webrtc:11084
Change-Id: I466f39816b83e2f83a3a49845c99605f4d5a857b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183763
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32082}
2020-09-11 14:11:27 +00:00
Peter Thatcher
e502386463
Merge with 4183 (M85) (#25) 2020-09-01 14:43:30 -07:00
Peter Thatcher
8f41923c9d
Update to M84 (4147) (#14)
Merge branch-heads/4147 into master
2020-07-21 19:48:32 -07:00
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
Peter Thatcher
9aeaa25d4c
Update to WebRTC 4103 (M83) (#12)
* Merge in branch-heads/4103 (M83)

* Disable legacy DTLS protocols (before 1.2)

* Update sdk/objc modifications for upstream changes

* Update ios and mac deployment targets

Co-authored-by: Jim Gustafson <jim@signal.org>
2020-06-25 11:14:34 -07:00
Jakob Ivarsson
d95138b684 Make stable target adaptation enabled by default.
This will result in slightly higher encode bitrates and longer frame
lengths compared to using the smoothing filter.

Bug: webrtc:10981
Change-Id: I64704196c56b0ad910895c908baad38c994a971b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177425
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31556}
2020-06-24 11:22:21 +00:00
Jakob Ivarsson
af0d5bca34 Remove ANA FEC control in Opus encoder.
This has been proven to not be useful.

Bug: chromium:1086942
Change-Id: Ib71b194f59301851791a1a056f5f10b98c5a1d57
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177520
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31548}
2020-06-22 11:18:26 +00:00
Philipp Hancke
0fd1ef135c opus: add helper function to extract voice activity information
BUG=webrtc:11643

Change-Id: I3cebc40916de0e4b0f5e41f5fda97dd53f76e4e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176740
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31490}
2020-06-10 14:21:01 +00:00
Jakob Ivarsson
7649006692 Remove packet loss rate optimization and minimum field trial.
Bug: webrtc:11664
Change-Id: I63fab70e5ae85e2971bed4998ab3b15f61f9e1c0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176752
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31480}
2020-06-10 08:37:30 +00:00
Mirko Bonadei
9621377730 Remove WebRTC-Audio-NewOpusPacketLossRateOptimization.
This field trial is unused.

Bug: webrtc:11503
Change-Id: I34262ea4ab169479ceded820c1aa309981731f1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173338
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31059}
2020-04-14 10:02:52 +00:00
Ali Tofigh
7e5dfdbca3 Implement AudioEncoder::GetFrameLengthRange() for all audio encoders.
The WebRTC-SendSideBwe-WithOverhead field trial requires audio
encoders to properly implement the
AudioEncoder::GetFrameLengthRange() function. Thic CL implements
the function for all audio encoders in WebRTC in preparation for
making that function pure virtual in the interface.


Bug: webrtc:11427
Change-Id: Ieab6b6c72c62af6ac9525a20fcb39bd477079551
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171503
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30890}
2020-03-25 22:19:21 +00:00
Ivo Creusen
16ddae924e Update Opus tests for Opus 1.3
This updates various bitexactness tests and other tests that no longer
pass.

Bug: webrtc:11325
Change-Id: Ifa3e4b42e303f5573e028dfdf8a108a76f6318ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168952
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30688}
2020-03-05 08:53:37 +00:00
Tom Anderson
422f9dd5df Conditionally use OPUS_GET_IN_DTX if available
OPUS_GET_IN_DTX was added 2019-04-15, but we still need to support
building on systems with older versions of the Opus headers (eg. Debian
Jessie, released 2015-04-25).  This is needed to fix the "Build From
Tarball" bot [1].

[1] https://ci.chromium.org/p/infra/builders/cron/Build%20From%20Tarball

BUG=chromium:1047860,webrtc:11085
R=minyue@webrtc.org,henrick.lundin@webrtc.org

Change-Id: I5418c3caf4d2c7da9b9ba43ce85879b1e0eec6e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168560
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Thomas Anderson <thomasanderson@chromium.org>
Cr-Commit-Position: refs/heads/master@{#30612}
2020-02-25 18:10:42 +00:00
Niels Möller
dbf5416a80 Delete header file rtc_base/memory/aligned_array.h
Move definition of AlignedArray to the only code using it, the
test-only LappedTransform class, and delete unused methods.

Bug: webrtc:6424, webrtc:9577
Change-Id: I1bb5f57400f7217345b7ec7376235ad4c4bae858
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168701
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30576}
2020-02-20 14:55:25 +00:00