Commit Graph

98 Commits

Author SHA1 Message Date
ff9f6461b6 Revert "Reland "Refactor NetEq delay manager logic.""
This reverts commit 2a7c57c34f323ee1977f6e7809ee23bfcf9a7459.

Reason for revert: unexpected big changes in behavior.

Original change's description:
> Reland "Refactor NetEq delay manager logic."
>
> This is a reland of f8e62fcb14e37a5be4f1e4f599d34c8483fea8e9
>
> Original change's description:
> > Refactor NetEq delay manager logic.
> >
> > - Removes dependence on sequence number for calculating target delay.
> > - Changes target delay unit to milliseconds instead of number of
> >   packets.
> > - Moves acceleration/preemptive expand thresholds to decision logic.
> >   Tests for this will be added in a follow up cl.
> >
> > Bug: webrtc:10333
> > Change-Id: If690aae4abf41ef1d9353f0ff01fb7d121cf8a26
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186265
> > Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> > Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#32326}
>
> Bug: webrtc:10333
> Change-Id: Iad5e7063f63b84762959ee5b412f5f14a7b2cd06
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186943
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32332}

TBR=ivoc@webrtc.org,jakobi@webrtc.org

Change-Id: Iffda0e8a7b647392d8dfc6724d49439fa13d71b2
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10333
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187100
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32341}
2020-10-07 12:47:01 +00:00
2a7c57c34f Reland "Refactor NetEq delay manager logic."
This is a reland of f8e62fcb14e37a5be4f1e4f599d34c8483fea8e9

Original change's description:
> Refactor NetEq delay manager logic.
>
> - Removes dependence on sequence number for calculating target delay.
> - Changes target delay unit to milliseconds instead of number of
>   packets.
> - Moves acceleration/preemptive expand thresholds to decision logic.
>   Tests for this will be added in a follow up cl.
>
> Bug: webrtc:10333
> Change-Id: If690aae4abf41ef1d9353f0ff01fb7d121cf8a26
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186265
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32326}

Bug: webrtc:10333
Change-Id: Iad5e7063f63b84762959ee5b412f5f14a7b2cd06
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186943
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32332}
2020-10-06 18:33:53 +00:00
b1ae5ccd16 Revert "Refactor NetEq delay manager logic."
This reverts commit f8e62fcb14e37a5be4f1e4f599d34c8483fea8e9.

Reason for revert: breaks downstream test.

Original change's description:
> Refactor NetEq delay manager logic.
>
> - Removes dependence on sequence number for calculating target delay.
> - Changes target delay unit to milliseconds instead of number of
>   packets.
> - Moves acceleration/preemptive expand thresholds to decision logic.
>   Tests for this will be added in a follow up cl.
>
> Bug: webrtc:10333
> Change-Id: If690aae4abf41ef1d9353f0ff01fb7d121cf8a26
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186265
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32326}

TBR=ivoc@webrtc.org,jakobi@webrtc.org

Change-Id: I1bdeacce61b902a0003a40c740f6acccf1443e3e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10333
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186942
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32329}
2020-10-06 15:37:45 +00:00
f8e62fcb14 Refactor NetEq delay manager logic.
- Removes dependence on sequence number for calculating target delay.
- Changes target delay unit to milliseconds instead of number of
  packets.
- Moves acceleration/preemptive expand thresholds to decision logic.
  Tests for this will be added in a follow up cl.

Bug: webrtc:10333
Change-Id: If690aae4abf41ef1d9353f0ff01fb7d121cf8a26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186265
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32326}
2020-10-06 13:22:45 +00:00
a43178c871 Reland "Activating AVX2 support by default"
This is a reland of ad148272b89394978915cb00e1c1be552d908a42

Original change's description:
> Activating AVX2 support by default
>
> This CL activates the newly added AVX2 support by default.
> The activation is done beneath a kill-switch.
>
> Beyond the above, the CL also changes an incorrect DCHECK_GT
> to a DCHECK_GE.
>
> Bug: webrtc:11663
> Change-Id: I231ccb2f5efabf74cd8190411daa954b2b94a2a0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183042
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32193}

Bug: webrtc:11663, chromium:1134234
Change-Id: I0cb34cf08d4d14bc3aee055254493c9c9ee8faa0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186401
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32303}
2020-10-03 22:37:13 +00:00
082fac73be Revert "Reland "Activating AVX2 support by default""
This reverts commit a0ad0bbf8f4b7a20c6efb6950ede2a44205b2b74.

Reason for revert: Speculative revert. I suspect it breaks downstream project

Original change's description:
> Reland "Activating AVX2 support by default"
>
> This is a reland of ad148272b89394978915cb00e1c1be552d908a42
>
> Original change's description:
> > Activating AVX2 support by default
> >
> > This CL activates the newly added AVX2 support by default.
> > The activation is done beneath a kill-switch.
> >
> > Beyond the above, the CL also changes an incorrect DCHECK_GT
> > to a DCHECK_GE.
> >
> > Bug: webrtc:11663
> > Change-Id: I231ccb2f5efabf74cd8190411daa954b2b94a2a0
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183042
> > Commit-Queue: Per Åhgren <peah@webrtc.org>
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#32193}
>
> Bug: webrtc:11663
> Change-Id: Ib41dc1d1c5865f2828699c462939d15d5562df47
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186262
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32270}

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

Change-Id: I1305fad8d19ba0bd69a38b9e2959af54f900535d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11663
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186304
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32273}
2020-10-01 13:39:07 +00:00
a0ad0bbf8f Reland "Activating AVX2 support by default"
This is a reland of ad148272b89394978915cb00e1c1be552d908a42

Original change's description:
> Activating AVX2 support by default
>
> This CL activates the newly added AVX2 support by default.
> The activation is done beneath a kill-switch.
>
> Beyond the above, the CL also changes an incorrect DCHECK_GT
> to a DCHECK_GE.
>
> Bug: webrtc:11663
> Change-Id: I231ccb2f5efabf74cd8190411daa954b2b94a2a0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183042
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32193}

Bug: webrtc:11663
Change-Id: Ib41dc1d1c5865f2828699c462939d15d5562df47
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186262
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32270}
2020-10-01 12:19:19 +00:00
80907be915 Revert "Activating AVX2 support by default"
This reverts commit ad148272b89394978915cb00e1c1be552d908a42.

Reason for revert: Speculative revert to investigate test failures

Original change's description:
> Activating AVX2 support by default
>
> This CL activates the newly added AVX2 support by default.
> The activation is done beneath a kill-switch.
>
> Beyond the above, the CL also changes an incorrect DCHECK_GT
> to a DCHECK_GE.
>
> Bug: webrtc:11663
> Change-Id: I231ccb2f5efabf74cd8190411daa954b2b94a2a0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183042
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32193}

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

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:11663
Change-Id: Ibb019e8c702dce45ebf47f1c1e8db19069b4964d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186081
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32259}
2020-09-30 15:30:04 +00:00
e5d669ed28 Reland "Activating AVX2 support by default"
This is a reland of ad148272b89394978915cb00e1c1be552d908a42

Original change's description:
> Activating AVX2 support by default
>
> This CL activates the newly added AVX2 support by default.
> The activation is done beneath a kill-switch.
>
> Beyond the above, the CL also changes an incorrect DCHECK_GT
> to a DCHECK_GE.
>
> Bug: webrtc:11663
> Change-Id: I231ccb2f5efabf74cd8190411daa954b2b94a2a0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183042
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32193}

Bug: webrtc:11663
Change-Id: I669435c2f4e451ee0766d809443484f2dde09d8d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185482
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32200}
2020-09-25 14:46:20 +00:00
9ccbe17abb Revert "Activating AVX2 support by default"
This reverts commit ad148272b89394978915cb00e1c1be552d908a42.

Reason for revert: Causing test failures downstream.

Original change's description:
> Activating AVX2 support by default
>
> This CL activates the newly added AVX2 support by default.
> The activation is done beneath a kill-switch.
>
> Beyond the above, the CL also changes an incorrect DCHECK_GT
> to a DCHECK_GE.
>
> Bug: webrtc:11663
> Change-Id: I231ccb2f5efabf74cd8190411daa954b2b94a2a0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183042
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32193}

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

Change-Id: If2287a0a4b37931ce5f85baae093a66b19d0a78b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11663
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185481
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32196}
2020-09-25 12:13:46 +00:00
ad148272b8 Activating AVX2 support by default
This CL activates the newly added AVX2 support by default.
The activation is done beneath a kill-switch.

Beyond the above, the CL also changes an incorrect DCHECK_GT
to a DCHECK_GE.

Bug: webrtc:11663
Change-Id: I231ccb2f5efabf74cd8190411daa954b2b94a2a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183042
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32193}
2020-09-25 11:30:22 +00:00
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
909f3a5339 Rename several more tests that use EXPECT_DEATH to *DeathTest.
Bug: webrtc:11577
Change-Id: I0397ee933464496e4885bb0f8030f3d669e5e612
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175641
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31309}
2020-05-18 16:10:04 +00:00
16ddae924e Update Opus tests for Opus 1.3
This updates various bitexactness tests and other tests that no longer
pass.

Bug: webrtc:11325
Change-Id: Ifa3e4b42e303f5573e028dfdf8a108a76f6318ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168952
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30688}
2020-03-05 08:53:37 +00:00
c31a4ec66a Disable opus tests to allow upgrade to opus 1.3
The upgrade to opus 1.3 is easier to carry out while the opus
bitexactness tests are temporarily disabled.

Bug: webrtc:11325
Change-Id: I96eecdbc93a01da88b92ae7f6473034c9795f3a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167726
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30425}
2020-01-30 14:57:15 +00:00
ff0e4dbd1f Reland "Send absolute capture time through audio coding module."
This is a reland of 48655cfdbfd99e0b6331e59201bcb8514f8b2a0a

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
4175914f41 Revert "Send absolute capture time through audio coding module."
This reverts commit 48655cfdbfd99e0b6331e59201bcb8514f8b2a0a.

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
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
9dc209a23a Add ability to disable detailed error message in RTC_CHECKs
Bug: webrtc:11133
Change-Id: I989654f1fb97b476a17956d69ee374406439ea8f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160653
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29952}
2019-11-28 17:51:00 +00:00
4dd56a3830 ACM: Adding unittests for the remixing functionality
On top of adding unittests for the remixing, the CL
moves the code tested to a separate file in order
to allow it to be tested.

Bug: webrtc:11007
Change-Id: I531736517bbcc715b3c1bf3a4256c42208c5b778
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155740
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29839}
2019-11-20 06:20:22 +00:00
4f2e9406c9 ACM: Adding support for more than 2 channels in the send pipeline
This CL adds support in the audio coding module for sending more than
2 channels to the encoder.

Bug: webrtc:11007
Change-Id: I0909b5c37a54c9d2e1353b864e55008cda50ffae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155583
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29385}
2019-10-04 14:38:59 +00:00
ef14f072a9 Delete AudioDecoder method IncomingPacket
Only the ISAC codec had an non-trivial implementation, for its unused
adaptive mode. This cl deletes that implementation, and the call
from NetEq, and the interface method.

Bug: webrtc:10098
Change-Id: Iaf7667e0ae867fc9d64286dff4c01a8ce0b6e2a4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153882
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29279}
2019-09-24 08:30:24 +00:00
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
507f43465b Reland "Make relative arrival delay mode default in NetEq delay manager."
This is a reland of 77c71d1488b1c821b2b3481f23a3264f1b1d37a5

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
75e2290af2 Rollback to strict audio codec tests for libopus on android (neon).
This a revert of the manual accommodation done in [1].
The lenient tests are no longer needed since a proper libopus fix [2]
has been rolled in [3].

[1] https://webrtc-review.googlesource.com/c/src/+/148700
[2] https://chromium-review.googlesource.com/c/chromium/src/+/1785648
[3] https://webrtc-review.googlesource.com/c/src/+/151721/

Bug: webrtc:9995, chromium:986727
Change-Id: I7f64a45ccbe2c4d985ba663cf77c6fa0efebd528
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151781
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#29089}
2019-09-06 07:48:28 +00:00
5b728cca77 Revert "Make relative arrival delay mode default in NetEq delay manager."
This reverts commit 77c71d1488b1c821b2b3481f23a3264f1b1d37a5.

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
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
32472449f1 Delete unused method AudioCodingModule::GetDecodingCallStatistics
Bug: None
Change-Id: I2804e241251d2faa421169085cd3f63972cc395f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151123
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29060}
2019-09-04 10:08:16 +00:00
25eb47ccf1 Make the RtpHeaderParserImpl available to tests and tools only.
There are a few reasons for making this test only:
* The code is only used by tests and utilities.
* The pure interface has only a single implementation so an interface isn't really needed.
  (a followup change could remove it altogether)
* The implementation always incorporates locking regardless of how the class gets used.
  See e.g. previous use in the Packet class.
* The implementation is a layer on top of RtpUtility::RtpHeaderParser which is
  sufficient for most production cases.

Change-Id: Ide6d50567cf8ae5127a2eb04cceeb10cf317ec36
Bug: none
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150658
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29010}
2019-08-29 15:56:40 +00:00
110a4de4e2 Roll chromium_revision 8f0166a59b..f0fd984a31 (685582:685691)
!! **Manual change** Less strict audio codec tests to accommodate opus fix [1].
!! This is meant to be a temporary mitigation.
[1] https://chromium-review.googlesource.com/c/chromium/src/+/1746617

Change log: 8f0166a59b..f0fd984a31
Full diff: 8f0166a59b..f0fd984a31

Changed dependencies
* src/base: 17d8ac209c..f6cc884505
* src/build: d6837de8f1..956965a6ea
* src/ios: 76e0b0bc60..6780db9c3e
* src/testing: 5d328647a1..48823ed18a
* src/third_party: d70201c684..82063e79f0
* src/third_party/depot_tools: 1b4c7e9f38..6d98232fde
* src/tools: b8953a5bf5..2aa12eadc5
DEPS diff: 8f0166a59b..f0fd984a31/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: I9869cc3f493bc82361e4f93ad846b32390edb340
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148700
Reviewed-by: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28833}
2019-08-12 15:53:01 +00:00
5b5d97c938 Reland of "Reporting of decoding_codec_plc events""
This is a reland of 0a88ea050cda58de81d624cf2764d46929447ed5.

The new stat will not be reported unless it is GT 0.

Reporting of decoding_codec_plc events

Bug: webrtc:10838
Change-Id: Ic8585b4eeae9a2643374f15bc2578d1141e59683
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148448
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Alex Narest <alexnarest@google.com>
Cr-Commit-Position: refs/heads/master@{#28797}
2019-08-07 18:41:46 +00:00
c653172e74 Delete obsolete method AudioCodingModule::SetBitRate
Bug: None
Change-Id: I2291f7b4b46d269592eacad67a126010b750fac2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148079
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28784}
2019-08-07 08:37:25 +00:00
bedb7a8aea Revert "Reporting of decoding_codec_plc events"
This reverts commit 0a88ea050cda58de81d624cf2764d46929447ed5.

Reason for revert: This CL breaks Chromium's FYI bots (example: https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Linux%20Tester/4033).

Original change's description:
> Reporting of decoding_codec_plc events
> 
> Change-Id: Id71b37244bc98bffaf25131a519127b3d2b86a8f
> 
> Bug: webrtc:10838
> Change-Id: Id71b37244bc98bffaf25131a519127b3d2b86a8f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147263
> Commit-Queue: Alex Narest <alexnarest@google.com>
> Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28700}

TBR=mflodman@webrtc.org,alexnarest@google.com

Change-Id: I5e5dd29ee375ba422f79932d4b8c3fd028a53db4
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10838
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147269
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28707}
2019-07-30 14:39:09 +00:00
0a88ea050c Reporting of decoding_codec_plc events
Change-Id: Id71b37244bc98bffaf25131a519127b3d2b86a8f

Bug: webrtc:10838
Change-Id: Id71b37244bc98bffaf25131a519127b3d2b86a8f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147263
Commit-Queue: Alex Narest <alexnarest@google.com>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28700}
2019-07-29 16:40:23 +00:00
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
a36c591c09 Reland "Reland "Change buffer level filter to store current level in number of samples.""
This is a reland of 0ded32d5a3d7acb9a00c3a1d9941e539aa94eee5

Original change's description:
> Reland "Change buffer level filter to store current level in number of samples."
> 
> This is a reland of 87977dd06e702ed517f26235c12e37bd927527c7
> 
> Original change's description:
> > Change buffer level filter to store current level in number of samples.
> > 
> > The buffer level should not be converted back and forth between samples and packets in case of variable packet lengths.
> > 
> > Bug: webrtc:10736
> > Change-Id: Ia08dcfac3d8104dc79fbad0704a5f6f12a050a01
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142178
> > Reviewed-by: Minyue Li <minyue@webrtc.org>
> > Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#28368}
> 
> Bug: webrtc:10736
> Change-Id: I1ff603e65cdd31c7429f36b035dcc00a17b68f3b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143787
> Commit-Queue: Minyue Li <minyue@webrtc.org>
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28393}

Bug: webrtc:10736
Change-Id: I251b8321e5a5fd870e018bc7c8083ec0a41de81b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144023
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28398}
2019-06-27 09:16:27 +00:00
b93af8543d Revert "Reland "Change buffer level filter to store current level in number of samples.""
This reverts commit 0ded32d5a3d7acb9a00c3a1d9941e539aa94eee5.

Reason for revert: breaks downstream projects.

Original change's description:
> Reland "Change buffer level filter to store current level in number of samples."
> 
> This is a reland of 87977dd06e702ed517f26235c12e37bd927527c7
> 
> Original change's description:
> > Change buffer level filter to store current level in number of samples.
> > 
> > The buffer level should not be converted back and forth between samples and packets in case of variable packet lengths.
> > 
> > Bug: webrtc:10736
> > Change-Id: Ia08dcfac3d8104dc79fbad0704a5f6f12a050a01
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142178
> > Reviewed-by: Minyue Li <minyue@webrtc.org>
> > Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#28368}
> 
> Bug: webrtc:10736
> Change-Id: I1ff603e65cdd31c7429f36b035dcc00a17b68f3b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143787
> Commit-Queue: Minyue Li <minyue@webrtc.org>
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28393}

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

Change-Id: I570c83ec3a88a24d7a1f883a351748dd71bea015
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10736
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144022
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28397}
2019-06-27 08:07:21 +00:00
0ded32d5a3 Reland "Change buffer level filter to store current level in number of samples."
This is a reland of 87977dd06e702ed517f26235c12e37bd927527c7

Original change's description:
> Change buffer level filter to store current level in number of samples.
> 
> The buffer level should not be converted back and forth between samples and packets in case of variable packet lengths.
> 
> Bug: webrtc:10736
> Change-Id: Ia08dcfac3d8104dc79fbad0704a5f6f12a050a01
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142178
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28368}

Bug: webrtc:10736
Change-Id: I1ff603e65cdd31c7429f36b035dcc00a17b68f3b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143787
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28393}
2019-06-26 20:30:05 +00:00
d3fc161c16 Revert "Change buffer level filter to store current level in number of samples."
This reverts commit 87977dd06e702ed517f26235c12e37bd927527c7.

Reason for revert: Breaks downstream project

Original change's description:
> Change buffer level filter to store current level in number of samples.
> 
> The buffer level should not be converted back and forth between samples and packets in case of variable packet lengths.
> 
> Bug: webrtc:10736
> Change-Id: Ia08dcfac3d8104dc79fbad0704a5f6f12a050a01
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142178
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28368}

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

Change-Id: I3900c9f6071fce51d13fb3b7c886157304d7a5c3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10736
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143786
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28369}
2019-06-25 12:33:01 +00:00
87977dd06e Change buffer level filter to store current level in number of samples.
The buffer level should not be converted back and forth between samples and packets in case of variable packet lengths.

Bug: webrtc:10736
Change-Id: Ia08dcfac3d8104dc79fbad0704a5f6f12a050a01
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142178
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28368}
2019-06-25 11:21:51 +00:00
4731f0062e Delete deprecated PlatformThread looping
Bug: webrtc:10594, webrtc:7187
Change-Id: Icba3a5cf6dbe817ead427c27645b3ad7bc8819be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134642
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27833}
2019-05-03 08:35:42 +00:00
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
44c21f48ee Encoder side of Multistream Opus.
Follows https://webrtc-review.googlesource.com/c/src/+/129768 closely.
Adds an ENCODER and sets it up to parse SDP config for multistream
opus.

E.g. this is the new SDP syntax for 6.1 surround sound:
"multiopus/48000/6 channel_mapping=0,4,1,2,3,5 num_streams=4 coupled_streams=2"


Bug: webrtc:8649
Change-Id: I3fc341e76f5c41dab0243cf65f6461e4c3d9d67d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132001
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27775}
2019-04-25 15:07:38 +00:00
e5b94160b5 Decoder for multistream Opus.
See https://webrtc-review.googlesource.com/c/src/+/121764 for the
overall vision.

This CL adds a multistream Opus decoder. It's a new code-path to not
interfere with the standard Opus decoder. We introduce new SDP syntax,
which uses terminology of RFC 7845. We also set up the decoder side to
parse it. The encoder part will come in a later CL.

E.g. this is the new SDP syntax for 6.1 surround sound:
"multiopus/48000/6 channel_mapping=0,4,1,2,3,5 num_streams=4 coupled_streams=2"

Bug: webrtc:8649
Change-Id: Ifbc584cbb6d07aed373f223512a20d6d72cec5ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129768
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27493}
2019-04-08 16:15:37 +00:00
50b8c399c9 Generalize the C-language Opus interface.
Switch to explicit channel mappings (RFC 7845) when creating
multi-stream Opus en/de-coders. The responsibility of setting up the
channel mappings will shift from WebRTC to the WebRTC user.

See https://webrtc-review.googlesource.com/c/src/+/121764 for the
current vision. See also the first child CL
https://webrtc-review.googlesource.com/c/src/+/129768
that sets up the Decoder to use this code.

Bug: webrtc:8649
Change-Id: I55959a293d54bb4c982eff68ec107c5ef8666c5c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129767
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27452}
2019-04-04 14:06:44 +00:00
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
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
d3a780b476 Cleanup NetEqPostponeDecodingAfterExpand field trial.
Change-Id: Ie96e9b35ced4b6ca8daa78f1fa80816386a6643b
Bug: webrtc:9289
Reviewed-on: https://webrtc-review.googlesource.com/c/124127
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26899}
2019-02-28 14:45:59 +00:00
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
65438812ba 2nd reland of https://webrtc-review.googlesource.com/c/src/+/114883
The difference to the original is new bitexactness strings.  The
reason for reland is breaking downstream projects.

Original CL description:

Tests for multi-stream Opus.

This CL (mainly) adds bit-exactness tests for multi-stream Opus. The
tests are in audio_coding_unittest.cc. Some refactoring of
AcmSendTestOldApi, AcmSenderBitExactnessOldApi is done to make it
possible. A few checks for "channels \in {1, 2}" are replaced with
"channels \in {1, 2, 4, 6, 8}" in the WebRTC Opus codec wrapper. A few
other changes are made to be able to write and read multi-channel WAV
files.

The SDP changes are NOT included; as of this CL there is no way to set
up a multi-stream opus en/de-coder from SDP strings.

TBR=ossu@webrtc.org

Bug: webrtc:8649
Change-Id: I6261b18c69fd666d43ab34ed8f1bc9d5cc82b21f
Reviewed-on: https://webrtc-review.googlesource.com/c/123882
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26809}
2019-02-22 09:59:01 +00:00