36 Commits

Author SHA1 Message Date
Fredrik Solenberg
55900fd416 Move APM initialization into WebRtcVoiceEngine
TBR=kwiberg@webrtc.org

Bug: webrtc:4690
Change-Id: Icd8590d3f7476c1a841c7e2425d1134d224b1a53
Reviewed-on: https://webrtc-review.googlesource.com/23480
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20855}
2017-11-23 21:20:18 +00:00
Bjorn Terelius
de939432dc Revert "Revert "Encode log events periodically instead of for every event.""
This reverts commit 33c5c7f5e4f018a5103770021328fc530d451d75.

Reason for revert: Fix broken API change.

TBR=sprang@webrtc.org,solenberg@webrtc.org
TBRing because only pc/ and api/ have changed since last LGTMed version.

Original change's description:
> Revert "Encode log events periodically instead of for every event."
>
> This reverts commit b154c27e72fddb6c0d7cac69a9c68fff22154519.
>
> Reason for revert: Broke the internal project.
>
> Original change's description:
> > Encode log events periodically instead of for every event.
> >
> > Updated unit test to take output_period and random seed as parameters.
> > Updated the peerconnection interface to allow passing in an output_period.
> >
> > This is in preparation of some upcoming CLs that will change the format
> > to store batches of delta-encoded values.
> >
> >
> > Bug: webrtc:8111
> > Change-Id: Id5d9844dfad8d8edad346cd7cbebc7eadaaa5416
> > Reviewed-on: https://webrtc-review.googlesource.com/22600
> > Commit-Queue: Björn Terelius <terelius@webrtc.org>
> > Reviewed-by: Elad Alon <eladalon@webrtc.org>
> > Reviewed-by: Tommi <tommi@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#20736}
>
> Change-Id: I2257c46c014adb8c7c4fb28538635cabed1f2229
> Bug: webrtc:8111
> Reviewed-on: https://webrtc-review.googlesource.com/24160
> Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
> Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20738}

Bug: webrtc:8111
Change-Id: Ie69862cd52d11c1e15adeb6e2caacafe16863c80
Reviewed-on: https://webrtc-review.googlesource.com/24620
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20811}
2017-11-21 10:58:57 +00:00
Zhi Huang
33c5c7f5e4 Revert "Encode log events periodically instead of for every event."
This reverts commit b154c27e72fddb6c0d7cac69a9c68fff22154519.

Reason for revert: Broke the internal project.

Original change's description:
> Encode log events periodically instead of for every event.
> 
> Updated unit test to take output_period and random seed as parameters.
> Updated the peerconnection interface to allow passing in an output_period.
> 
> This is in preparation of some upcoming CLs that will change the format
> to store batches of delta-encoded values.
> 
> 
> Bug: webrtc:8111
> Change-Id: Id5d9844dfad8d8edad346cd7cbebc7eadaaa5416
> Reviewed-on: https://webrtc-review.googlesource.com/22600
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Elad Alon <eladalon@webrtc.org>
> Reviewed-by: Tommi <tommi@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20736}

TBR=solenberg@webrtc.org,eladalon@webrtc.org,terelius@webrtc.org,tommi@webrtc.org,sprang@webrtc.org,pthatcher@webrtc.org

Change-Id: I2257c46c014adb8c7c4fb28538635cabed1f2229
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8111
Reviewed-on: https://webrtc-review.googlesource.com/24160
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20738}
2017-11-17 21:02:02 +00:00
Bjorn Terelius
b154c27e72 Encode log events periodically instead of for every event.
Updated unit test to take output_period and random seed as parameters.
Updated the peerconnection interface to allow passing in an output_period.

This is in preparation of some upcoming CLs that will change the format
to store batches of delta-encoded values.


Bug: webrtc:8111
Change-Id: Id5d9844dfad8d8edad346cd7cbebc7eadaaa5416
Reviewed-on: https://webrtc-review.googlesource.com/22600
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20736}
2017-11-17 19:15:11 +00:00
Mirko Bonadei
675513b96a Stop using LOG macros in favor of RTC_ prefixed macros.
This CL has been generated with the following script:

for m in PLOG \
  LOG_TAG \
  LOG_GLEM \
  LOG_GLE_EX \
  LOG_GLE \
  LAST_SYSTEM_ERROR \
  LOG_ERRNO_EX \
  LOG_ERRNO \
  LOG_ERR_EX \
  LOG_ERR \
  LOG_V \
  LOG_F \
  LOG_T_F \
  LOG_E \
  LOG_T \
  LOG_CHECK_LEVEL_V \
  LOG_CHECK_LEVEL \
  LOG
do
  git grep -l $m | xargs sed -i "s,\b$m\b,RTC_$m,g"
done
git checkout rtc_base/logging.h
git cl format

Bug: webrtc:8452
Change-Id: I1a53ef3e0a5ef6e244e62b2e012b864914784600
Reviewed-on: https://webrtc-review.googlesource.com/21325
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20617}
2017-11-09 11:56:32 +00:00
Henrik Lundin
9bde6b7698 Add new UMA metric for the audio receiver delay
The UMA metric will log the same information that goes into the
googCurrentDelayMs stat.

Bug: webrtc:8488
Change-Id: I26abb3d86a07e8c0ddb4168540a8e2458115f004
Reviewed-on: https://webrtc-review.googlesource.com/18201
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20557}
2017-11-06 08:39:56 +00:00
Alex Narest
78609d5b6b Reland of BWE allocation strategy
TBR=stefan@webrtc.org,alexnarest@webrtc.org

Bug: webrtc:8243
Change-Id: Ie68e4f414b2ac32ba4e64877cb250fabcb089a07
Reviewed-on: https://webrtc-review.googlesource.com/13940
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Alex Narest <alexnarest@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20369}
2017-10-20 10:16:15 +00:00
Alex Narest
dc9ca9329b Revert "BWE allocation strategy"
This reverts commit a5fbc23379823d74b8cf4bc18887ff40237989e8.

Reason for revert: <INSERT REASONING HERE>

Original change's description:
> BWE allocation strategy
> 
> This is reland of https://webrtc-review.googlesource.com/c/src/+/4860 with the fixed RampUpTest test
> 
> Bug: webrtc:8243
> Change-Id: I4b90a449b00dd05feee974001e08fb40710b59ac
> Reviewed-on: https://webrtc-review.googlesource.com/13124
> Commit-Queue: Alex Narest <alexnarest@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20345}

TBR=stefan@webrtc.org,alexnarest@webrtc.org

Change-Id: I8ed12cd2115ef63204e384cc93c9f4473daa54d1
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8243
Reviewed-on: https://webrtc-review.googlesource.com/14020
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Alex Narest <alexnarest@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20361}
2017-10-19 15:34:52 +00:00
Karl Wiberg
8818237538 voe::Channel: Don't use CodecManager and RentACodec
As a bonus, we also get rid of most uses of CodecInst in voe::Channel.

BUG=webrtc:8396

Change-Id: I20f82274dcec2ad8be6e046b987d7d65a19d0e46
Reviewed-on: https://webrtc-review.googlesource.com/10802
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20354}
2017-10-19 13:12:51 +00:00
Alex Narest
a5fbc23379 BWE allocation strategy
This is reland of https://webrtc-review.googlesource.com/c/src/+/4860 with the fixed RampUpTest test

Bug: webrtc:8243
Change-Id: I4b90a449b00dd05feee974001e08fb40710b59ac
Reviewed-on: https://webrtc-review.googlesource.com/13124
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20345}
2017-10-19 09:30:00 +00:00
Lu Liu
39260c4a6b Revert "BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic."
This reverts commit 54d1da13a584680ae80a1f229291e5bb7e76e6e1.

Reason for revert: Breaking tests

Original change's description:
> BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic.
> 
> This CL implements the main logic and IOS appRTC integration.
> 
> Unit tests and Android appRTC will be in separate CL.
> 
> Bug: webrtc:8243
> Change-Id: If8e5195294046a47316e9fade1b0dfec211155e1
> Reviewed-on: https://webrtc-review.googlesource.com/4860
> Commit-Queue: Alex Narest <alexnarest@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20329}

TBR=deadbeef@webrtc.org,nisse@webrtc.org,stefan@webrtc.org,alexnarest@webrtc.org

Change-Id: I5be1da78f360f72be66f9d56dd6b88c1cc13e963
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8243
Reviewed-on: https://webrtc-review.googlesource.com/12560
Reviewed-by: Lu Liu <lliuu@webrtc.org>
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20330}
2017-10-17 19:59:04 +00:00
Alex Narest
54d1da13a5 BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic.
This CL implements the main logic and IOS appRTC integration.

Unit tests and Android appRTC will be in separate CL.

Bug: webrtc:8243
Change-Id: If8e5195294046a47316e9fade1b0dfec211155e1
Reviewed-on: https://webrtc-review.googlesource.com/4860
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20329}
2017-10-17 18:22:15 +00:00
Elad Alon
604c14df58 Reland "Remove deprecated functions from RtcEventLog"
The unified Log() interface replaces the many old LogX() functions. This helps hide dependencies between the modules which log different events.

TBR=stefan@webrtc.org

Bug: webrtc:8111
Change-Id: I36c8b6c4cf03d738c9033af2e98db6dc200eede9
Reviewed-on: https://webrtc-review.googlesource.com/6940
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20170}
2017-10-05 15:07:53 +00:00
Danil Chapovalov
d312a9149f Revert "Remove deprecated functions from RtcEventLog"
This reverts commit 5fd6e5ec1feed9e937efbe27ba9924cee3be2b81.

Reason for revert: breaks downstream project
please still keep default arguments for CreateEventLog

Original change's description:
> Remove deprecated functions from RtcEventLog
> 
> The unified Log() interface replaces the many old LogX() functions. This helps hide dependencies between the modules which log different events.
> 
> TBR=stefan@webrtc.org
> 
> Bug: webrtc:8111
> Change-Id: I5ea9fd50ba6da87d5867513c81c5e3bdb0524a32
> Reviewed-on: https://webrtc-review.googlesource.com/2689
> Commit-Queue: Elad Alon <eladalon@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Elad Alon <eladalon@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20159}

TBR=danilchap@webrtc.org,eladalon@webrtc.org,terelius@webrtc.org,stefan@webrtc.org

Change-Id: Iefc195f5804dabc0f76b87f889ff55481f4d285b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8111
Reviewed-on: https://webrtc-review.googlesource.com/6842
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20164}
2017-10-05 12:25:31 +00:00
Elad Alon
5fd6e5ec1f Remove deprecated functions from RtcEventLog
The unified Log() interface replaces the many old LogX() functions. This helps hide dependencies between the modules which log different events.

TBR=stefan@webrtc.org

Bug: webrtc:8111
Change-Id: I5ea9fd50ba6da87d5867513c81c5e3bdb0524a32
Reviewed-on: https://webrtc-review.googlesource.com/2689
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20159}
2017-10-05 10:00:20 +00:00
Niels Möller
22ec952829 Delete in_order argument to RtpReceiver::IncomingRtpPacket
Bug: webrtc:7135
Change-Id: I35fbc76a5ca8d50caff918bbfd2cb13dce4cbd21
Reviewed-on: https://webrtc-review.googlesource.com/4141
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20154}
2017-10-05 07:19:20 +00:00
Karl Wiberg
c62f6c7121 RTPPayloadRegistry: Use SdpAudioFormat to represent audio codecs
This is needed in the general case, now that we aim to support codecs
other than those built-in to WebRTC.

BUG=webrtc:8159

Change-Id: I40a41252bf69ad5d4d0208e3c1e8918da7394706
Reviewed-on: https://webrtc-review.googlesource.com/5380
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20136}
2017-10-04 11:30:14 +00:00
Elad Alon
83ccca1864 Create and use RtcEventLogOutput for output
We need to support two modes of writing to the output:
1. Current way - the application lets lets WebRTC know which file to write to, and WebRTC is then in charge of the writing.
2. New way - the application would receive indications from WebRTC about (encoded) RTC events, and would itself be in charge of processing them (be it writing it to a file, uploading it somewhere, etc.).

We achieve this by creating an interface for output - RtcEventLogOutput. By providing an instance of the subclass, RtcEventLogOutputFile, the old behavior is achieved. The subclass of the new behavior is to be added by a later CL.

TBR=stefan@webrtc.org

Bug: webrtc:8111
Change-Id: I9c50521a7f7144d86d8353a65995795862e19c44
Reviewed-on: https://webrtc-review.googlesource.com/2686
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20135}
2017-10-04 11:18:47 +00:00
Elad Alon
4a87e1c211 Remove encoding code from RtcEventLogImpl and use RtcEventLogEncoder instead
RtcEventLogImpl no longer hard-codes the way encoding is done. It now relies on RtcEventEncoder for it. This gives two benefits:
1. We can decide between the current encoding and the new encoding (which is still WIP) without code duplication (no need for RtcEventLogImplNew).
2. Encoding is done only when the event needs to be written to a file. This both avoids unnecessary encoding of events which don't end up getting written to a file, as well as is useful for the new, delta-based encoding, which is stateful.

BUG=webrtc:8111

Change-Id: I9517132e5f96b8059002a66fde8d42d3a678c3bb
Reviewed-on: https://webrtc-review.googlesource.com/1365
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20118}
2017-10-03 15:26:56 +00:00
Sam Zackrisson
ecc51e96db Change to LOG(...) logging in most of voice_engine/channel.cc
First patch set runs a script to directly convert log statements.
Second patch sets manually fixes smaller errors.

Due to the size of this change, the remaining WEBRTC_TRACE statements will
be handled in a different CL.

Bug: webrtc:5118
Change-Id: Ic39c3a6310a2b461b47a7b4757210d98637e8acd
Reviewed-on: https://webrtc-review.googlesource.com/1228
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20091}
2017-10-02 19:10:00 +00:00
Bjorn Terelius
440216fcf3 Split LogRtpHeader and LogRtcpPacket into separate versions for incoming and outgoing packets.
Change LogIncomingRtcpPacket and LogOutgoingRtcpPacket to take ArrayView<uint8_t>.
Split LogSessionAndReadBack into three functions and create class to share state between them.
Split VerifyRtpEvent into one incoming and one outgoing version.

Originally uploaded as https://codereview.webrtc.org/2997973002/

Bug: webrtc:8111
Change-Id: I22bdc35163bef60bc8293679226b19e41e8f49b3
Reviewed-on: https://webrtc-review.googlesource.com/5020
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20063}
2017-10-02 08:44:20 +00:00
solenberg
1c239d476e Remove voe::Statistics.
BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/3020473002
Cr-Commit-Position: refs/heads/master@{#20042}
2017-09-29 13:00:28 +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
henrika
4580217b56 Adds WebRTC.Audio.EncodingTaskQueueLatencyMs
Part II of https://webrtc-review.googlesource.com/c/src/+/1584

Bug: webrtc:8206
Change-Id: I71ff164a884c61404d1c542d943dd12a5ee2de6f
Reviewed-on: https://webrtc-review.googlesource.com/4180
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20012}
2017-09-28 09:48:19 +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
solenberg
6df16bf46d Remove unnecessary send codec initialization from voe::Channel.
BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/3012403002
Cr-Commit-Position: refs/heads/master@{#19981}
2017-09-27 00:13:19 +00:00
solenberg
fc3a2e3393 Remove the VoiceEngineObserver callback interface.
BUG=webrtc:4690
TBR=kwiberg@webrtc.org

Review-Url: https://codereview.webrtc.org/3019513002
Cr-Commit-Position: refs/heads/master@{#19976}
2017-09-26 16:35:01 +00:00
solenberg
2397b9a114 Remove voe::OutputMixer and AudioConferenceMixer.
This code path is not used anymore.

BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/3015553002
Cr-Commit-Position: refs/heads/master@{#19929}
2017-09-22 13:48:10 +00:00
Karl Wiberg
73b60b82ee Remove the redundant method GetPayloadSpecifics
It's in the way of a refactoring.

Also change PayloadTypeToPayload---the method all callers can use instead---to return Optional<Payload> instead of const Payload* (for thread safety reasons: an object that protects itself with a mutex shouldn't be handing out pointers to parts of itself). 

BUG=webrtc:8159

Change-Id: I7ef0d545077ffdea016b309f2165e3c4955a2928
Reviewed-on: https://webrtc-review.googlesource.com/2360
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19917}
2017-09-21 20:19:55 +00:00
solenberg
946d886187 Remove VoENetwork
BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/3016543002
Cr-Commit-Position: refs/heads/master@{#19912}
2017-09-21 11:02:53 +00:00
solenberg
dd3abbb532 Remove VoERTP_RTCP.
BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/3006383002
Cr-Commit-Position: refs/heads/master@{#19892}
2017-09-18 14:05:30 +00:00
solenberg
6dc2038d0d Remove VoECodec.
BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/3019433002
Cr-Commit-Position: refs/heads/master@{#19889}
2017-09-18 12:22:39 +00:00
solenberg
b63310a256 Remove VoEFile and things it uses.
BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/3013033002
Cr-Commit-Position: refs/heads/master@{#19885}
2017-09-18 10:04:12 +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