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Author SHA1 Message Date
Mirko Bonadei
99eea42fc1 Reland "Reland "Export symbols needed by the Chromium component build (part 1)."" 
This reverts commit b49520bfc08f5c5832dda1d642125f0bb898f974.

Reason for revert: Problem fixed in https://chromium-review.googlesource.com/c/chromium/src/+/1261398.

Original change's description:
> Revert "Reland "Export symbols needed by the Chromium component build (part 1).""
> 
> This reverts commit 588f4642d1a29f7beaf28265dbd08728191b4c52.
> 
> Reason for revert: Breaks WebRTC Chromium FYI Win Builder (dbg).
> lld-link: error: undefined symbol: "__declspec(dllimport) __thiscall webrtc::Config::Config(void)" (__imp_??0Config@webrtc@@QAE@XZ)
> [...]
> 
> Original change's description:
> > Reland "Export symbols needed by the Chromium component build (part 1)."
> > 
> > This reverts commit 2ea9af227517556136fd629dd2663c0d75d77c7b.
> > 
> > Reason for revert: The problem will be fixed by
> > https://chromium-review.googlesource.com/c/chromium/src/+/1261122.
> > 
> > Original change's description:
> > > Revert "Export symbols needed by the Chromium component build (part 1)."
> > > 
> > > This reverts commit 9e24dcff167c4eb3555bf0ce6eaba090c10fbe53.
> > > 
> > > Reason for revert: Breaks chromium.webrtc.fyi bots.
> > > 
> > > Original change's description:
> > > > Export symbols needed by the Chromium component build (part 1).
> > > > 
> > > > This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h)
> > > > to mark WebRTC symbols as visible from a shared library, this doesn't
> > > > mean these symbols are part of the public API (please continue to refer
> > > > to [1] for info about what is considered public WebRTC API).
> > > > 
> > > > [1] - https://webrtc.googlesource.com/src/+/HEAD/native-api.md
> > > > 
> > > > Bug: webrtc:9419
> > > > Change-Id: I802abd32874d42d3aa5ecd3c8022e7cf5e043d99
> > > > Reviewed-on: https://webrtc-review.googlesource.com/c/103505
> > > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> > > > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > > > Cr-Commit-Position: refs/heads/master@{#24969}
> > > 
> > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org
> > > 
> > > Change-Id: I01f6e18f0d2c0f0309cdaa6c943c3927e1f1f49f
> > > No-Presubmit: true
> > > No-Tree-Checks: true
> > > No-Try: true
> > > Bug: webrtc:9419
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/103720
> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#24974}
> > 
> > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org
> > 
> > Change-Id: I83bbc7f550fc23e823c4d055e0a6f60c828960dd
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: webrtc:9419
> > Reviewed-on: https://webrtc-review.googlesource.com/c/103740
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#24980}
> 
> TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org
> 
> Change-Id: I4b7cfe492f2c8eeda5c8ac52520e0cfc95ade9b0
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9419
> Reviewed-on: https://webrtc-review.googlesource.com/c/103801
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24983}

TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org

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

Bug: webrtc:9419
Change-Id: Id986a0a03cdc2818690337784396882af067f7fa
Reviewed-on: https://webrtc-review.googlesource.com/c/104602
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25049}
 2018-10-08 12:54:06 +00:00
Sebastian Jansson
2e068e8b6f Adds RTT based backoff trial to SendSideBandwidthEstimation. 
Bug: webrtc:9718
Change-Id: Ic94dcd7612524d350f54d1907f843577b890badf
Reviewed-on: https://webrtc-review.googlesource.com/c/104122
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25048}
 2018-10-08 12:36:06 +00:00
Joel Sutherland
d2fb1bfeba Generate module.modulemap file when building Mac Framework 
Without this file, the Framework can't be used by Swift projects.

Bug: webrtc:9142
Change-Id: I8803ec8b194dc116e133257e205f4620bb34a692
Reviewed-on: https://webrtc-review.googlesource.com/c/103340
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25047}
 2018-10-08 12:32:45 +00:00
Oleh Prypin
e6708f31a0 Notify a rotation about autoroll CLs 
Also don't keep dry-run-only CLs as work-in-progress.

No-Try: True
Bug: chromium:836566
Change-Id: I3008845396d09c751443755f96a5d571389e1d2e
Reviewed-on: https://webrtc-review.googlesource.com/c/104620
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25046}
 2018-10-08 12:20:41 +00:00
Artem Titov
75e3647a76 Switch usages of DefaultNetworkSimulationConfig to BuiltInNetworkBehaviorConfig 
Bug: webrtc:9630
Change-Id: Ia0e0b5b4e1e3a8e687d1e7fe3bb600dbdda09efa
Reviewed-on: https://webrtc-review.googlesource.com/c/104561
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25045}
 2018-10-08 12:19:31 +00:00
Niels Möller
3a74239091 Fix compilation issues on media_transport_interface.h 
Include api/video/encoded_image.h, and move constructors and
destructors to .cc file.

Bug: webrtc:9719
Change-Id: Ibecdc1151bf672155d3c09e13749ac39c921c3aa
Reviewed-on: https://webrtc-review.googlesource.com/c/104560
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25044}
 2018-10-08 11:50:29 +00:00
Bjorn Terelius
788c51c858 Pass HeaderExtensionMap by reference in rtc_event_log2rtp_dump. 
Bug: webrtc:8111
Change-Id: I0a64f92227e54aa1c180375bf406daaeac638fba
Reviewed-on: https://webrtc-review.googlesource.com/c/102640
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25043}
 2018-10-08 10:52:27 +00:00
Yves Gerey
b6a8942fb4 Fix race condition for GetContributingSources test. 
Bug: webrtc:9813
Change-Id: I44f50f9858217c8303862f3820db11dbd8736b6c
Reviewed-on: https://webrtc-review.googlesource.com/c/104121
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25042}
 2018-10-08 10:22:51 +00:00
Artem Titov
666fb32d1f Rename DefaultNetworkSimulationConfig into BuiltInNetworkBehaviorConfig. 
It is done to better show what for this class exists and also restore
correspondence between config and interface, that is implemented by
configurable object.

Bug: webrtc:9630
Change-Id: I28456d1c792d67d9b2a405c8599054137a5d596a
Reviewed-on: https://webrtc-review.googlesource.com/c/104003
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25041}
 2018-10-08 10:16:08 +00:00
chromium-webrtc-autoroll
6a8327fc82 Roll chromium_revision ccb83d4a55..0cdd2e3eab (597330:597498) 
Change log: ccb83d4a55..0cdd2e3eab
Full diff: ccb83d4a55..0cdd2e3eab

Changed dependencies
* src/base: 0db61379d8..226b8bac1f
* src/build: 8bb9fef862..63f397a2df
* src/ios: d4e0105161..02ce4e5b29
* src/testing: 677e1ef262..a8da99ff4c
* src/third_party: dcfac1c5ed..503cf2d8b5
* src/tools: 731f63f312..8cfdea14be
DEPS diff: ccb83d4a55..0cdd2e3eab/DEPS

No update to Clang.

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

Change-Id: Ia37f07f965b85a76cec0cddaec10dc4d1535834e
Reviewed-on: https://webrtc-review.googlesource.com/c/104444
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#25040}
 2018-10-08 09:43:55 +00:00
Sebastian Jansson
7c1744d3dc Reland "Reland "Using units in SendSideBandwidthEstimation."" 
This reverts commit a4de9c8b04ce9f7b1c4be639edd6cc4d3a9f07a1.

Reason for revert: <INSERT REASONING HERE>

Original change's description:
> Revert "Reland "Using units in SendSideBandwidthEstimation.""
> 
> This reverts commit e2cb26cb4fa2a3ce7c12636225ba9c720d7c7e56.
> 
> Reason for revert: <INSERT REASONING HERE>
> 
> Original change's description:
> > Reland "Using units in SendSideBandwidthEstimation."
> > 
> > This reverts commit 917e5967a597fa8d6e6cae9ffccb21e3d35d553b.
> > 
> > Reason for revert: Handling downstream use case.
> > 
> > Original change's description:
> > > Revert "Using units in SendSideBandwidthEstimation."
> > > 
> > > This reverts commit 35b5e5f3b0dc409bf571b3609860ad5bb8e00c29.
> > > 
> > > Reason for revert: Breaks downstream project
> > > 
> > > Original change's description:
> > > > Using units in SendSideBandwidthEstimation.
> > > >
> > > > This CL moves SendSideBandwidthEstimation to use the unit types
> > > > DataRate, TimeDelta and Timestamp. This prepares for upcoming changes.
> > > >
> > > > Bug: webrtc:9718
> > > > Change-Id: If10e329920dda037b53055ff3352ae7f8d7e32b8
> > > > Reviewed-on: https://webrtc-review.googlesource.com/c/104021
> > > > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > > > Reviewed-by: Björn Terelius <terelius@webrtc.org>
> > > > Cr-Commit-Position: refs/heads/master@{#25029}
> > > 
> > > TBR=terelius@webrtc.org,srte@webrtc.org
> > > 
> > > No-Try: True
> > > Bug: webrtc:9718
> > > Change-Id: Iaf470f1eec9911ee6fc7c1b4f5db9675d89d3780
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/104480
> > > Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
> > > Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#25035}
> > 
> > TBR=oprypin@webrtc.org,terelius@webrtc.org,srte@webrtc.org
> > 
> > Change-Id: I0940791fcd1e196598b0f0a2ec779c49931ee5df
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: webrtc:9718
> > Reviewed-on: https://webrtc-review.googlesource.com/c/104520
> > Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#25036}
> 
> TBR=oprypin@webrtc.org,terelius@webrtc.org,srte@webrtc.org
> 
> Change-Id: I6628771c79fc78dfd856649ae92232e95df63495
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9718
> Reviewed-on: https://webrtc-review.googlesource.com/c/104540
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25037}

TBR=oprypin@webrtc.org,terelius@webrtc.org,srte@webrtc.org

Change-Id: If5473859cea725420afce11b6683fa0c70a29b0a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9718
Reviewed-on: https://webrtc-review.googlesource.com/c/104501
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25039}
 2018-10-08 09:02:03 +00:00
Per Kjellander
841c912ddd Changed FakeVp8Encoder to write dimensions in payload. 
Add FakeVp8Decoder that parse width and height from the payload.
Add unit test for testing that width and height is set when decoding frames.


Bug: none
Change-Id: Ifbfff4f62f99625309ce0ef21cf89c76448769d8
Reviewed-on: https://webrtc-review.googlesource.com/c/103140
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25038}
 2018-10-08 08:37:38 +00:00
Sebastian Jansson
a4de9c8b04 Revert "Reland "Using units in SendSideBandwidthEstimation."" 
This reverts commit e2cb26cb4fa2a3ce7c12636225ba9c720d7c7e56.

Reason for revert: <INSERT REASONING HERE>

Original change's description:
> Reland "Using units in SendSideBandwidthEstimation."
> 
> This reverts commit 917e5967a597fa8d6e6cae9ffccb21e3d35d553b.
> 
> Reason for revert: Handling downstream use case.
> 
> Original change's description:
> > Revert "Using units in SendSideBandwidthEstimation."
> > 
> > This reverts commit 35b5e5f3b0dc409bf571b3609860ad5bb8e00c29.
> > 
> > Reason for revert: Breaks downstream project
> > 
> > Original change's description:
> > > Using units in SendSideBandwidthEstimation.
> > >
> > > This CL moves SendSideBandwidthEstimation to use the unit types
> > > DataRate, TimeDelta and Timestamp. This prepares for upcoming changes.
> > >
> > > Bug: webrtc:9718
> > > Change-Id: If10e329920dda037b53055ff3352ae7f8d7e32b8
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/104021
> > > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > > Reviewed-by: Björn Terelius <terelius@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#25029}
> > 
> > TBR=terelius@webrtc.org,srte@webrtc.org
> > 
> > No-Try: True
> > Bug: webrtc:9718
> > Change-Id: Iaf470f1eec9911ee6fc7c1b4f5db9675d89d3780
> > Reviewed-on: https://webrtc-review.googlesource.com/c/104480
> > Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
> > Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#25035}
> 
> TBR=oprypin@webrtc.org,terelius@webrtc.org,srte@webrtc.org
> 
> Change-Id: I0940791fcd1e196598b0f0a2ec779c49931ee5df
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9718
> Reviewed-on: https://webrtc-review.googlesource.com/c/104520
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25036}

TBR=oprypin@webrtc.org,terelius@webrtc.org,srte@webrtc.org

Change-Id: I6628771c79fc78dfd856649ae92232e95df63495
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9718
Reviewed-on: https://webrtc-review.googlesource.com/c/104540
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25037}
 2018-10-08 08:27:29 +00:00
Sebastian Jansson
e2cb26cb4f Reland "Using units in SendSideBandwidthEstimation." 
This reverts commit 917e5967a597fa8d6e6cae9ffccb21e3d35d553b.

Reason for revert: Handling downstream use case.

Original change's description:
> Revert "Using units in SendSideBandwidthEstimation."
> 
> This reverts commit 35b5e5f3b0dc409bf571b3609860ad5bb8e00c29.
> 
> Reason for revert: Breaks downstream project
> 
> Original change's description:
> > Using units in SendSideBandwidthEstimation.
> >
> > This CL moves SendSideBandwidthEstimation to use the unit types
> > DataRate, TimeDelta and Timestamp. This prepares for upcoming changes.
> >
> > Bug: webrtc:9718
> > Change-Id: If10e329920dda037b53055ff3352ae7f8d7e32b8
> > Reviewed-on: https://webrtc-review.googlesource.com/c/104021
> > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > Reviewed-by: Björn Terelius <terelius@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#25029}
> 
> TBR=terelius@webrtc.org,srte@webrtc.org
> 
> No-Try: True
> Bug: webrtc:9718
> Change-Id: Iaf470f1eec9911ee6fc7c1b4f5db9675d89d3780
> Reviewed-on: https://webrtc-review.googlesource.com/c/104480
> Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
> Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25035}

TBR=oprypin@webrtc.org,terelius@webrtc.org,srte@webrtc.org

Change-Id: I0940791fcd1e196598b0f0a2ec779c49931ee5df
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9718
Reviewed-on: https://webrtc-review.googlesource.com/c/104520
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25036}
 2018-10-08 08:26:09 +00:00
Oleh Prypin
917e5967a5 Revert "Using units in SendSideBandwidthEstimation." 
This reverts commit 35b5e5f3b0dc409bf571b3609860ad5bb8e00c29.

Reason for revert: Breaks downstream project

Original change's description:
> Using units in SendSideBandwidthEstimation.
>
> This CL moves SendSideBandwidthEstimation to use the unit types
> DataRate, TimeDelta and Timestamp. This prepares for upcoming changes.
>
> Bug: webrtc:9718
> Change-Id: If10e329920dda037b53055ff3352ae7f8d7e32b8
> Reviewed-on: https://webrtc-review.googlesource.com/c/104021
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25029}

TBR=terelius@webrtc.org,srte@webrtc.org

No-Try: True
Bug: webrtc:9718
Change-Id: Iaf470f1eec9911ee6fc7c1b4f5db9675d89d3780
Reviewed-on: https://webrtc-review.googlesource.com/c/104480
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25035}
 2018-10-08 08:21:18 +00:00
Yves Gerey
2e00abc98e Reland "[cleanup] Remove useless includes." 
Reason for reland: Downstream project fixed.

Original change's description:

> [cleanup] Remove useless includes.
>
> Manual cleanup guided by include-what-you-use diagnostic.
>
> Bug: webrtc:8311
> Change-Id: I00be03392cc7ee005101427ea7dc701621ccea68
> Reviewed-on: https://webrtc-review.googlesource.com/c/103320
> Commit-Queue: Yves Gerey <yvesg@webrtc.org>
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25013}

Bug: webrtc:8311
Change-Id: Id6ec4aeb798886a90ace640a190eaf16497ba31b
Reviewed-on: https://webrtc-review.googlesource.com/c/104120
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25034}
 2018-10-08 07:44:19 +00:00
Niels Möller
4dc66c53d0 Move EncodedImage class to api/video/ 
Bug: webrtc:9378
Change-Id: I8fb3b19cad0ad428abc6c8e6b507180d461882ba
Reviewed-on: https://webrtc-review.googlesource.com/c/104002
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25033}
 2018-10-08 07:37:10 +00:00
Mirko Bonadei
38537edf59 Fix visibility of api/units build targets. 
Bug: None
Change-Id: Ie5f44ba9f5565b1a51175de2f4514829edb3dd03
Reviewed-on: https://webrtc-review.googlesource.com/c/104161
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25032}
 2018-10-08 07:27:30 +00:00
Sebastian Jansson
343f4144be Allows copy and assignment of field trial parameters. 
Bug: webrtc:9346
Change-Id: I7734fbe60f95e554a636946cadc1940a78419951
Reviewed-on: https://webrtc-review.googlesource.com/c/104020
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25031}
 2018-10-06 15:32:04 +00:00
chromium-webrtc-autoroll
e53341c1dc Roll chromium_revision 7099444bc9..ccb83d4a55 (597172:597330) 
Change log: 7099444bc9..ccb83d4a55
Full diff: 7099444bc9..ccb83d4a55

Changed dependencies
* src/base: b7130f4871..0db61379d8
* src/build: c71b994c1b..8bb9fef862
* src/ios: 4fec31cb4a..d4e0105161
* src/testing: a844ba6377..677e1ef262
* src/third_party: 980e505656..dcfac1c5ed
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/de52d9ad29..4fc4281d21
* src/third_party/depot_tools: 2fb6310237..71e3be7a50
* src/tools: dd53e662ac..731f63f312
DEPS diff: 7099444bc9..ccb83d4a55/DEPS

No update to Clang.

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

Change-Id: Id56449f4cf136cf9febc90555d14ed2f9bc6362f
Reviewed-on: https://webrtc-review.googlesource.com/c/104220
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#25030}
 2018-10-05 22:38:34 +00:00
Sebastian Jansson
35b5e5f3b0 Using units in SendSideBandwidthEstimation. 
This CL moves SendSideBandwidthEstimation to use the unit types
DataRate, TimeDelta and Timestamp. This prepares for upcoming changes.

Bug: webrtc:9718
Change-Id: If10e329920dda037b53055ff3352ae7f8d7e32b8
Reviewed-on: https://webrtc-review.googlesource.com/c/104021
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25029}
 2018-10-05 18:37:13 +00:00
Jonathan Metzman
9f80b97309 Fix fuzzer build failures on Windows 
Fix the following issues with fuzz targets when built on Windows:
1. Fix audio_processing_fuzzer by making types match in
invocations of CheckedDivExact by explicitly casting to size_t.
2. Fix packet_buffer_fuzzer by including "frame_object.h" for
declaration of RtpFrameObject.
3. Fix rtcp_receiver_fuzzer by including "tmmb_item.h" for declaration
of TmmbItem.

Bug: chromium:891867
Change-Id: Iddc338360ca37d5fc31488ec908eb4cdb5cc7b94
Reviewed-on: https://webrtc-review.googlesource.com/c/103844
Commit-Queue: Jonathan Metzman <metzman@chromium.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25028}
 2018-10-05 18:13:32 +00:00
chromium-webrtc-autoroll
4d6f605123 Roll chromium_revision d62b62d830..7099444bc9 (597059:597172) 
Change log: d62b62d830..7099444bc9
Full diff: d62b62d830..7099444bc9

Changed dependencies
* src/base: 8a7fcd34a1..b7130f4871
* src/build: 3ffc6a0feb..c71b994c1b
* src/ios: f6a4666b1b..4fec31cb4a
* src/testing: ea96478eee..a844ba6377
* src/third_party: d8d78ffc4f..980e505656
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/56216d7832..de52d9ad29
* src/third_party/depot_tools: f98905e8f0..2fb6310237
* src/tools: b580bb107d..dd53e662ac
DEPS diff: d62b62d830..7099444bc9/DEPS

No update to Clang.

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

Change-Id: Ida1c06739e0a1f620545d9f4ffb3b2f39874a8b5
Reviewed-on: https://webrtc-review.googlesource.com/c/104141
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#25027}
 2018-10-05 17:48:42 +00:00
Niklas Enbom
ef8a3eb522 Include NTP value in playout path. 
Change-Id: Icc612d6779474b56394e5d4b3afe8aee592ef192

BUG: webrtc:9832
Change-Id: Icc612d6779474b56394e5d4b3afe8aee592ef192
Reviewed-on: https://webrtc-review.googlesource.com/c/103883
Commit-Queue: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25026}
 2018-10-05 16:46:38 +00:00
Sebastian Jansson
a23dc78c7d Removes initial window field trial. 
Bug: webrtc:9718
Change-Id: Ia1cc352bde1d8994cce7eb7e3bdcbc04e03fd718
Reviewed-on: https://webrtc-review.googlesource.com/c/104041
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25025}
 2018-10-05 15:41:50 +00:00
Paulina Hensman
a6471eb589 Reland "Tidy up and increase exception handling in compare_videos" 
This is a reland of 1c60ff521eda26c80fa53097d9c614f10200f651

Original change's description:
> Tidy up and increase exception handling in compare_videos
> 
> Bug: webrtc:9642
> Change-Id: I5c8b252de3b285f81a5437af99d789b5a28ce646
> Reviewed-on: https://webrtc-review.googlesource.com/102880
> Commit-Queue: Paulina Hensman <phensman@webrtc.org>
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24909}

Bug: webrtc:9642
Change-Id: I11078a358297ae5046991ac3b0680df468bb413b
Reviewed-on: https://webrtc-review.googlesource.com/c/102941
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25024}
 2018-10-05 15:10:17 +00:00
Sebastian Jansson
6c19decc14 Adds Clamping functions for DataRate. 
This is quite useful in many places where we need to restrict the range
of a DataRate. It makes it easier to read the intention than with:
value_ = std::max(some_lower_limit, std::min(value_, some_upper_limit));

The naming follows the naming for rtc::SafeClamp.

Bug: webrtc:9709
Change-Id: I08e05197acec325d85babd2a06806a8667f2fcb1
Reviewed-on: https://webrtc-review.googlesource.com/c/104040
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25023}
 2018-10-05 15:06:17 +00:00
Sebastian Jansson
b88fe025b7 Removes logging spam from congestion window. 
Bug: webrtc:9830
Change-Id: I20c84b757de03f2bcc010b19f256297ca9722fa6
Reviewed-on: https://webrtc-review.googlesource.com/c/104066
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25022}
 2018-10-05 15:00:17 +00:00
Alex Loiko
f2637a8d6f Reland of 'Bug in histogram metric reporting.' 
Original CL: https://webrtc-review.googlesource.com/c/src/+/101340

A (actually several weeks) while ago, we noticed an error with the
WebRTC.Audio.Agc2.EstimatedNoiseLevel histogram. It always reported
the value 0. Here is why:

The histogram bins go from 0 to 100. But the value logged is dBFS. It
is always less than or equal to 0. This CL changes inverts the value
logged. The noise level value should be somewhere between -90 and 0
dBFS.

The histogram description is updated in
https://chromium-review.googlesource.com/c/chromium/src/+/1264578

Bug: webrtc:7494
Change-Id: I0b53630d4284ce1078fd453e05e89ee53ca9f6c7
Reviewed-on: https://webrtc-review.googlesource.com/c/104063
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25021}
 2018-10-05 14:47:13 +00:00
Rasmus Brandt
e28dedf100 Remove old data files. 
Bug: None
Change-Id: I52385b1248eb19c6e9247cc28b06d215174ddb87
Reviewed-on: https://webrtc-review.googlesource.com/c/103040
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25020}
 2018-10-05 14:40:21 +00:00
Stefan Holmer
64be7fa7d8 Move FecController to RtpVideoSender. 
This also moves the packet feedback tracking to RtpVideoSender.

Bug: webrtc:9517
Change-Id: Ifb1ff85051730108a0b0d1dd30f6f8595ad2af6e
Reviewed-on: https://webrtc-review.googlesource.com/c/95920
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25019}
 2018-10-05 14:39:01 +00:00
Rasmus Brandt
8e87852cbe Remove old video_bitrate_allocator.h 
Bug: webrtc:9513
Change-Id: If44e14fbb5d9ace5aadb325b766b596f8217bb9b
Reviewed-on: https://webrtc-review.googlesource.com/c/103001
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25018}
 2018-10-05 14:36:41 +00:00
Patrik Höglund
8434aeb3a7 Use Chromium's code for locating the src dir. 
This code is much more sophisticated in that it doesn't rely
on argv[0], but rather asks the OS where our executable is.
We can then simply go two steps up since we count on running
in out/Whatever relative to the src dir. This is how Chromium
does it.

The aim here is to get rid of SetExecutablePath, which will
be the next CL.

Bug: webrtc:9792
Change-Id: I7da027b7391e759b1f612de12f27a244fe884c09
Reviewed-on: https://webrtc-review.googlesource.com/c/103121
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25017}
 2018-10-05 14:09:57 +00:00
Stefan Holmer
0a5792e907 Add UMA metric and logging of frames dropped in the render queue. 
Bug: b/80195113
Change-Id: I7a696fe58ccf4e2bc7502438c2f58beb65848d25
Reviewed-on: https://webrtc-review.googlesource.com/c/104062
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25016}
 2018-10-05 13:19:11 +00:00
Oleh Prypin
96a0f61917 Revert "[cleanup] Remove useless includes." 
This reverts commit be8b5348c76105f8fe869b0cae4065ddca106419.

Reason for revert: Breaks downstream project

Original change's description:
> [cleanup] Remove useless includes.
> 
> Manual cleanup guided by include-what-you-use diagnostic.
> 
> Bug: webrtc:8311
> Change-Id: I00be03392cc7ee005101427ea7dc701621ccea68
> Reviewed-on: https://webrtc-review.googlesource.com/c/103320
> Commit-Queue: Yves Gerey <yvesg@webrtc.org>
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25013}

TBR=phoglund@google.com,phoglund@webrtc.org,yvesg@webrtc.org

Change-Id: I7a6e1cdfef685173b76f234ad598083043dcd9a0
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8311
Reviewed-on: https://webrtc-review.googlesource.com/c/104022
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25015}
 2018-10-05 13:13:45 +00:00
Erik Språng
1cd7391c00 Turning off a stream should results in target bitrate 0 signal 
Bug: webrtc:9734, b:116850043
Change-Id: Ia7b4a8ecf2099c3026c83b06febca833d428d0a2
Reviewed-on: https://webrtc-review.googlesource.com/c/103981
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25014}
 2018-10-05 12:00:26 +00:00
Yves Gerey
be8b5348c7 [cleanup] Remove useless includes. 
Manual cleanup guided by include-what-you-use diagnostic.

Bug: webrtc:8311
Change-Id: I00be03392cc7ee005101427ea7dc701621ccea68
Reviewed-on: https://webrtc-review.googlesource.com/c/103320
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25013}
 2018-10-05 11:51:06 +00:00
chromium-webrtc-autoroll
a1134509c9 Roll chromium_revision 618ddbcb7f..d62b62d830 (596951:597059) 
Change log: 618ddbcb7f..d62b62d830
Full diff: 618ddbcb7f..d62b62d830

Changed dependencies
* src/build: d36c5ed010..3ffc6a0feb
* src/testing: 8589b2b984..ea96478eee
* src/third_party: fdab02bc50..d8d78ffc4f
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/3f7d74f7cd..56216d7832
* src/third_party/depot_tools: 5b1fa949bb..f98905e8f0
* src/tools: 6cdf067f3e..b580bb107d
DEPS diff: 618ddbcb7f..d62b62d830/DEPS

No update to Clang.

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

Change-Id: If93e8a6c15303fb67a97029db2f1806d7acfb2f0
Reviewed-on: https://webrtc-review.googlesource.com/c/103967
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#25012}
 2018-10-05 11:06:23 +00:00
Artem Titov
8ea1e9def1 Switch webrtc from deprecated usages of NetworkSimulationInterface 
Bug: webrtc:9630
Change-Id: I42222261676b0c260c1aab81523a23988d3cd1c1
Reviewed-on: https://webrtc-review.googlesource.com/c/103780
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25011}
 2018-10-05 11:01:42 +00:00
Niels Möller
ae4237e5db Set ChannelReceive transport at construction time. 
Followup to cl https://webrtc-review.googlesource.com/c/src/+/103640.
Set the rtcp_send_transport at construction time, delete
RegisterTransport, and the proxying of transport methods.

In addition, delete the unused RtcpRtpStats argument from the
constructor.

Bug: webrtc:9801
Change-Id: I80f25bc08dc2130386053568ddce4ef91654caeb
Reviewed-on: https://webrtc-review.googlesource.com/c/103803
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25010}
 2018-10-05 10:56:40 +00:00
Paulina Hensman
6c966eaf17 Remove @SuppressLint(NewApi) and guard @TargetApi methods 
Also rename runningOnLollipopOrHigher() etc in WebRtcAudioUtils
to runningOnApi21OrHigher() etc since mapping API numbers to
names is error prone.

Bug: webrtc:9818
Change-Id: I4a71de72e3891ca2b6fc2341db9131bb2db4cce7
Reviewed-on: https://webrtc-review.googlesource.com/c/103820
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25009}
 2018-10-05 10:36:14 +00:00
Mirko Bonadei
97c65b76c2 Make modules/audio_mixer:audio_mixer_impl publicly visible. 
Follow-up CL after offline discussion with aleloi@.

Bug: None
Change-Id: I24623df7d821a44656336d5623a31cde4436c94f
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/c/103982
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25008}
 2018-10-05 10:11:17 +00:00
Niels Möller
44b384d013 Delete support for VoIP metrics (RFC 3611 4.7) 
Bug: None
Change-Id: I2f3cd622d3863fa88a9e1971894eced8eeb777e6
Reviewed-on: https://webrtc-review.googlesource.com/c/103805
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25007}
 2018-10-05 10:07:57 +00:00
Alex Loiko
4bb1e4a1d5 Lower gain parameters for AGC2. 
The AdaptiveAgc often boosts the signal outside of Float S16 range. It
is expected, which is why we have a limiter after it in the process
chain. But it turns out that this happens regularly even for simple
input examples. The output signal peaks can be as high as +4 dBFs for a
single speaker example (which should be easy). It leads to excessive
gain modulation by the limiter.

This CL is a new tuning designed to produce a safer gain. After this,
we shouldn't hit the saturation region of the limiter as often. But we
will still maintain a high gain.

We have a 'configurable kill-switch': the settings can be changed via
field trials WebRTC-Audio-Agc2Force(Initial|Extra)SaturationMargin.

Bug: webrtc:7494, chromium:892043
Change-Id: I5014377050c74c32ae8998282991141eae31cf58
Reviewed-on: https://webrtc-review.googlesource.com/c/102922
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25006}
 2018-10-05 09:55:25 +00:00
Erik Språng
5fbc0e0b33 Hide libvpx vp8 encoder behind an interface and add mock for testing. 
Bug: webrtc:9809
Change-Id: I27baa0309511cbd849c09c5d063a64d1fb1fcebf
Reviewed-on: https://webrtc-review.googlesource.com/c/103442
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25005}
 2018-10-05 08:51:52 +00:00
Johannes Kron
78cdde3df6 Add support for sending RTP two-byte header extensions. 
Automatic detection if one-byte header or two-byte header should be used based
on extension ID and extension length.

Bug: webrtc:7990
Change-Id: I9fc848ecc59458d1ca97bace0e57ea04d3d0ced6
Reviewed-on: https://webrtc-review.googlesource.com/c/103422
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25004}
 2018-10-05 08:45:52 +00:00
Mirko Bonadei
7880be1799 Don't include <memory.h> in aligned_malloc.cc. 
The inclusion of <memory.h> creates problems when building with Chromium
third_party/webrtc/rtc_base/memory/aligned_malloc.cc:13:10:
  fatal error: 'memory.h' file not found
    #include <memory.h>

It seems the code doesn't need to include <memory.h> but <cstring>.

Bug: None
Change-Id: Ib6591711aa7cfea49a2ff08321cfb3bd3689797a
Reviewed-on: https://webrtc-review.googlesource.com/c/103980
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25003}
 2018-10-05 07:44:46 +00:00
chromium-webrtc-autoroll
0a74e09b39 Roll chromium_revision 0af97cea37..618ddbcb7f (596847:596951) 
Change log: 0af97cea37..618ddbcb7f
Full diff: 0af97cea37..618ddbcb7f

Changed dependencies
* src/base: c832cca29d..8a7fcd34a1
* src/build: dc7ee415d7..d36c5ed010
* src/ios: 818ce397ac..f6a4666b1b
* src/testing: 912d2ff7c3..8589b2b984
* src/third_party: de96cb90e3..fdab02bc50
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/1b6e81b579..3f7d74f7cd
* src/third_party/depot_tools: e8f574a216..5b1fa949bb
* src/tools: 646382ba94..6cdf067f3e
DEPS diff: 0af97cea37..618ddbcb7f/DEPS

No update to Clang.

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

Change-Id: Ie49fe61a6630d71306dff295b89a90866b0ef914
Reviewed-on: https://webrtc-review.googlesource.com/c/103885
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#25002}
 2018-10-05 01:30:26 +00:00
Benjamin Wright
84583f6183 Enable End-to-End Encrypted Audio Payloads. 
This change integrates the FrameDecryptorInterface and the FrameEncryptorInterface into
the audio media path. If a FrameEncryptorInterface is set on an outgoing audio RTPSender
then each outgoing audio payload will first pass through the provided FrameEncryptor which
will have a chance to modify the payload contents for the purposes of encryption.

If a FrameDecryptorInterface is set on an incoming audio RtpReceiver then each incoming
audio payload will first pass through the provided FrameDecryptor which have a chance to
modify the payload contents for the purpose of decryption.

While AEAD is supported by the FrameDecryptor/FrameEncryptor interfaces this CL does not
use it and so it is left as null.

Bug: webrtc:9681
Change-Id: Ic383a9dce280528739f9d271357c2220e0a0dccf
Reviewed-on: https://webrtc-review.googlesource.com/c/101702
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Emad Omara <emadomara@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25001}
 2018-10-04 22:08:34 +00:00
chromium-webrtc-autoroll
aa43b7bb2f Roll chromium_revision c5c13a1e38..0af97cea37 (596716:596847) 
Change log: c5c13a1e38..0af97cea37
Full diff: c5c13a1e38..0af97cea37

Changed dependencies
* src/base: efcf82194a..c832cca29d
* src/build: c3c521dfb4..dc7ee415d7
* src/ios: dafcdbac33..818ce397ac
* src/testing: 75227fd89d..912d2ff7c3
* src/third_party: e0cbf41eb5..de96cb90e3
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/e3b37a4725..1b6e81b579
* src/third_party/depot_tools: b250ec16d3..e8f574a216
* src/tools: 7562397f66..646382ba94
DEPS diff: c5c13a1e38..0af97cea37/DEPS

No update to Clang.

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

Change-Id: I642e46d736a83be184854bf5a467e02b9988a495
Reviewed-on: https://webrtc-review.googlesource.com/c/103880
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#25000}
 2018-10-04 21:32:23 +00:00