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ff0e4dbd1f0b5549c9cd90687b8599131e909b10
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23 Commits

Author SHA1 Message Date
Minyue Li
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
Minyue Li
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
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
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
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
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
Fredrik Solenberg
657b296ff5 Reland "Remove CodecInst pt.1" 
This is a reland of 056f9738bf7a3d16da45398239656e165c4e0851

Original change's description:
> Remove CodecInst pt.1
> 
> Update audio_coding tests to not use CodecInst.
> 
> Bug: webrtc:7626
> Change-Id: I880fb8d72d7d0a915d274e67feb6106f023697c2
> Reviewed-on: https://webrtc-review.googlesource.com/c/112594
> Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25879}

Bug: webrtc:7626
Change-Id: I5d6ca0baf6230bfe9bf95c2c25496d2a56812d90
Reviewed-on: https://webrtc-review.googlesource.com/c/112942
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25902}
 2018-12-05 10:38:23 +00:00
Fredrik Solenberg
ec0f45be11 Revert "Remove CodecInst pt.1" 
This reverts commit 056f9738bf7a3d16da45398239656e165c4e0851.

Reason for revert: breaks downstream

Original change's description:
> Remove CodecInst pt.1
> 
> Update audio_coding tests to not use CodecInst.
> 
> Bug: webrtc:7626
> Change-Id: I880fb8d72d7d0a915d274e67feb6106f023697c2
> Reviewed-on: https://webrtc-review.googlesource.com/c/112594
> Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25879}

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

Change-Id: I51d666969bcd63e2b7cb7d669ec2f59b5f8f9dde
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7626
Reviewed-on: https://webrtc-review.googlesource.com/c/112906
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25881}
 2018-12-03 15:50:51 +00:00
Fredrik Solenberg
056f9738bf Remove CodecInst pt.1 
Update audio_coding tests to not use CodecInst.

Bug: webrtc:7626
Change-Id: I880fb8d72d7d0a915d274e67feb6106f023697c2
Reviewed-on: https://webrtc-review.googlesource.com/c/112594
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25879}
 2018-12-03 15:16:20 +00:00
Niels Möller
433eafe1f5 Delete unused includes of assert.h 
Bug: None
Change-Id: Iadc531710dca0ba34a00ac363bfe0784355bb6f9
Reviewed-on: https://webrtc-review.googlesource.com/c/103501
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24995}
 2018-10-04 14:01:44 +00:00
Niels Möller
a12c42a6b2 Delete root header file typedef.h. 
Usage replaced with stdint.h, rtc_base/system/arch.h and
rtc_base/system/unused.h, as appropriate.

Bug: webrtc:6854
Change-Id: I97225465d14b969903d92979e2df3c3c05d35f18
Reviewed-on: https://webrtc-review.googlesource.com/90249
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24100}
 2018-07-25 14:59:26 +00:00
Yves Gerey
665174fdbb Reformat the WebRTC code base 
Running clang-format with chromium's style guide.

The goal is n-fold:
 * providing consistency and readability (that's what code guidelines are for)
 * preventing noise with presubmit checks and git cl format
 * building on the previous point: making it easier to automatically fix format issues
 * you name it

Please consider using git-hyper-blame to ignore this commit.

Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
 2018-06-19 14:00:39 +00:00
Karl Wiberg
5817d3dfaa AudioCodingModule::Create(): Require caller to supply an AudioDecoderFactory 
So that we don't have to be capable of creating one ourselves, which
requires a dependency on the audio decoders.

BUG=webrtc:5801, webrtc:8396

Change-Id: I80749ec3b86cba73994307046d05964f59167d44
Reviewed-on: https://webrtc-review.googlesource.com/18440
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22774}
 2018-04-06 15:10:27 +00:00
solenberg
c7b4a45594 Remove various IDs: 
- AudioFrame
- AudioCodingModule

BUG=webrtc:4690
TBR=kwiberg@webrtc.org

Review-Url: https://codereview.webrtc.org/3019543002
Cr-Original-Commit-Position: refs/heads/master@{#20005}
Committed: https://webrtc.googlesource.com/src/+/2d0f77585d556d8b11d6269d35149ae9ca14c472
Review-Url: https://codereview.webrtc.org/3019543002
Cr-Commit-Position: refs/heads/master@{#20019}
 2017-09-28 14:37:11 +00:00
solenberg
e423a9de93 Revert of Remove various IDs (patchset #7 id:120001 of https://codereview.webrtc.org/3019543002/ ) 
Reason for revert:
Breaks downstream

Original issue's description:
> Remove various IDs:
>
> - AudioFrame
> - AudioCodingModule
>
> BUG=webrtc:4690
> TBR=kwiberg@webrtc.org
>
> Review-Url: https://codereview.webrtc.org/3019543002
> Cr-Commit-Position: refs/heads/master@{#20005}
> Committed: https://webrtc.googlesource.com/src/+/2d0f77585d556d8b11d6269d35149ae9ca14c472

TBR=henrik.lundin@webrtc.org,kwiberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/3014683002
Cr-Commit-Position: refs/heads/master@{#20008}
 2017-09-27 18:28:14 +00:00
solenberg
2d0f77585d Remove various IDs: 
- AudioFrame
- AudioCodingModule

BUG=webrtc:4690
TBR=kwiberg@webrtc.org

Review-Url: https://codereview.webrtc.org/3019543002
Cr-Commit-Position: refs/heads/master@{#20005}
 2017-09-27 17:33:57 +00:00
Mirko Bonadei
7120742701 Adding NOLINT for typedefs.h and common_types.h 
Now that we have moved WebRTC from src/webrtc to src/, common_types.h
and typedefs.h are triggering a cpplint error.

The cpplint complaint is:
Include the directory when naming .h files  [build/include] [4]

This CL disables the error but we have to remove these two headers
from the root directory.

NOPRESUBMIT=true

Bug: webrtc:5876
Change-Id: I08e1b69aadcc4b28ab83bf25e3819d135d41d333
Reviewed-on: https://webrtc-review.googlesource.com/1577
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19859}
 2017-09-15 13:03:51 +00:00
Mirko Bonadei
92ea95e34a Fixing WebRTC after moving from src/webrtc to src/ 
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
 2017-09-15 05:02:56 +00:00
Mirko Bonadei
bb547203bf Moving src/webrtc into src/. 
In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
 2017-09-15 04:25:06 +00:00