2091 Commits

Author SHA1 Message Date
erikvarga@webrtc.org
0f08d227c2 Add a function for enabling the congestion window and pushback controller in the webrtc::SendSideCongestionController.
Bug: webrtc:9923
Change-Id: Id01ebd7237ba33f34003aa9560405a13da7580e2
Reviewed-on: https://webrtc-review.googlesource.com/c/107893
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Commit-Queue: Erik Varga <erikvarga@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25393}
2018-10-26 17:19:32 +00:00
Danil Chapovalov
99b71dfd4a Use function_video_(en|de)coder_factory from api
Remove them from test.
It is completion of the move started with
https://webrtc-review.googlesource.com/c/src/+/107705

Bug: None
Change-Id: Ib0b26db04a1ee814322851280ba1e59b4b3f7ce6
Reviewed-on: https://webrtc-review.googlesource.com/c/107891
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25392}
2018-10-26 14:54:19 +00:00
Ilya Nikolaevskiy
88c2c50dbd Use monotonic clock to derive NTP timestamps in RTCP module
Use helper TimeMicrosToNtp() on clock TimeInMicroseconds()
instead of CurrentNtpTime() and CurrentNtpTimeMillis()

Also update TimeMicrosToNtp() to not introduce fractional in
milliseconds offset. Expose that offset in time_utils.h

Add test showing indended behavior.

Bug: webrtc:9919
Change-Id: I8b019e11ae5b79d0b8ba113a84066b0369cd2575
Reviewed-on: https://webrtc-review.googlesource.com/c/107889
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25391}
2018-10-26 14:30:28 +00:00
Sebastian Jansson
916ae08a04 Makes critsect_.Leave() more visible in PacedSender.
This means that the PacedSender::Process function becomes slightly
larger, however, it makes it much more obvious to the reader where
the locks are held or not. Confusion over this has previously caused
bugs.

Bug: webrtc:9870
Change-Id: I63257eae59ecf5e7dd28ea24f63157cefe9f81bd
Reviewed-on: https://webrtc-review.googlesource.com/c/105460
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25389}
2018-10-26 13:09:45 +00:00
Danil Chapovalov
6dd7f9120e Remove deprecated deregistration functions in RtcpTransceiver
These functions were deprecated in
https://webrtc-review.googlesource.com/c/src/+/107305

Bug: None
Change-Id: I3246f1e2f16be7591daec520e90c6b75071eb92d
Reviewed-on: https://webrtc-review.googlesource.com/c/107730
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25388}
2018-10-26 12:31:37 +00:00
Niels Möller
7d76a31f3d Use MediaTransportInterface, for audio streams.
Bug: webrtc:9719
Change-Id: I6d3db66b781173b207de51d84193fbd34a7f3239
Reviewed-on: https://webrtc-review.googlesource.com/c/104642
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25385}
2018-10-26 11:40:57 +00:00
Jamie Walch
9c8ae4b9a1 Disable probe delay warning in release builds.
This log is triggering many times a second for Chrome Remote Desktop on some
browsers. This CL just turns it off for release builds to avoid log files
filling up users' disks until we figure out what's going on.

Bug: chromium:888038
Change-Id: Ibbe9d47295b3633314feb28e155e3f59b878dbdb
Reviewed-on: https://webrtc-review.googlesource.com/c/107688
Commit-Queue: Jamie Walch <jamiewalch@google.com>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25372}
2018-10-25 16:00:16 +00:00
Niels Möller
039743e066 Reland "Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase"
This is a reland of 80cd25bcfb2264fa0f1192de942a6f063879dd42

Original change's description:
> Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase
>
> Bug: None
> Change-Id: I225fe1e16a3c96e5a03e3ae8fe975f368be7e6ad
> Reviewed-on: https://webrtc-review.googlesource.com/c/107303
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25312}

Tbr: kwiberg@webrtc.org
Bug: None
Change-Id: Id43a93bada9d6d66a4d0f0286f583066156aa2fc
Reviewed-on: https://webrtc-review.googlesource.com/c/107716
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25368}
2018-10-25 14:13:44 +00:00
Sebastian Jansson
3eb1c72bb6 Removes deprecated BitrateAllocation alias.
Bug: webrtc:9883
Change-Id: Ia14727a43c31241590889e48aded63dd8b30e181
Reviewed-on: https://webrtc-review.googlesource.com/c/107734
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25357}
2018-10-25 11:02:58 +00:00
Per Åhgren
370bae466c APM: Adding more explicit handling of failures in the json config data
This CL creates a new API for the parser of APM json config that
that provides an explicit way for the user to know when there has
been an issue in the parsing of the json config data.

Bug: webrtc:9921
Change-Id: Idd8f40529f40ab6871efb5b356c0fd2cea21b7d9
Reviewed-on: https://webrtc-review.googlesource.com/c/107841
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25355}
2018-10-25 10:31:54 +00:00
Per Åhgren
7a95e0fcf4 APM: Add ability to turn on/off dumping of internal data
This CL modifies the internal data logging and the audioproc_f tool
to allow controlling that via the command line, rather than solely via a
build flag. The logging of internal data is by default off.

Bug: webrtc:5298
Change-Id: I96d1b4f990582938527b9039d6c2ecbb6f76e9ca
Reviewed-on: https://webrtc-review.googlesource.com/c/107713
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25352}
2018-10-25 09:03:53 +00:00
Erik Språng
e2fd86a79c Move encoder metadata into EncoderInfo struct.
This deprecates the following methods in VideoEncoder:
  virtual ScalingSettings GetScalingSettings() const;
  virtual bool SupportsNativeHandle() const;
  virtual const char* ImplementationName() const;

Though they are not marked RTC_DEPRECATED since we still want to call
them from within the default GetEncoderInfo() until downstream
projects have been updated.

Furthmore, implementation name is changed from const char* to
std:string, which prevents some lifetime issues with dynamic encoder
names, and CodecSpecificInfo.codec_name is removed in favor of getting
the implementation name via GetEncoderInfo().

This CL removes calls to these deprecated methods, follow-ups will also
remove implementations of the methods and replace them with new
GetEncoderInfo() substitutions.

Bug: webrtc:9890
Change-Id: I6fd6e531480c0b952f53dbd5105e0b0adc3e3b0c
Reviewed-on: https://webrtc-review.googlesource.com/c/106905
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25351}
2018-10-25 08:51:53 +00:00
Ilya Nikolaevskiy
5e58bcbf29 Forward audio rtp frequency to Rtcp sender and use it for SR packets
Process video rtp frequency in the same way.

Bug: webrtc:6458
Change-Id: Ia22768e1242d686c2b3e2b911f3e5e492cf8b895
Reviewed-on: https://webrtc-review.googlesource.com/c/107651
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25334}
2018-10-24 12:27:09 +00:00
Per Åhgren
f0c449e3ff APM: Correct includes required for the data dumping functionality
Bug: webrtc:5298
Change-Id: Ia8b8e6a308f1812216651efaf0e2249e9d0cbfd8
Reviewed-on: https://webrtc-review.googlesource.com/c/107631
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25327}
2018-10-24 07:38:28 +00:00
Per Åhgren
700b4a4e65 AEC3: Allow limiting dominant nearend to the non-initial phase
This CL allows control over the dominant nearend functionality so that
it is not active during the initial phase, when estimates are less
certain.

Bug: webrtc:9906,chromium:898273
Change-Id: I5f61dac806ec3b1ebc1a3ec72f0a16d07a67f14a
Reviewed-on: https://webrtc-review.googlesource.com/c/107632
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25326}
2018-10-24 07:15:49 +00:00
Alessio Bazzica
087e9bed41 AGC2 Limiter class renamed.
Limiter has been renamed to LimiterDbGainCurve, which is a more correct name
and will allow in a follow-up CL to reuse the Limiter name for GainCurveApplier.
This is done to allow to use the limiter without instancing the fixed digital
gain controller and then to fix an AGC2 issue (namely, fixed gain applied after
the adaptive one).

Bug: webrtc:7494
Change-Id: Icd7050e3e51b832bfbf35e5cc61109215c5b1ca6
Reviewed-on: https://webrtc-review.googlesource.com/c/106901
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25322}
2018-10-23 15:20:52 +00:00
Oleh Prypin
6e8e2993dd Revert "Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase"
This reverts commit 80cd25bcfb2264fa0f1192de942a6f063879dd42.

Reason for revert: Breaks downstream project

Original change's description:
> Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase
>
> Bug: None
> Change-Id: I225fe1e16a3c96e5a03e3ae8fe975f368be7e6ad
> Reviewed-on: https://webrtc-review.googlesource.com/c/107303
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25312}

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

No-Try: true
Bug: None
Change-Id: I77b66bc032e2d95d1bd408c6cdeceb4dcd511699
Reviewed-on: https://webrtc-review.googlesource.com/c/107643
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25317}
2018-10-23 13:21:27 +00:00
Niels Möller
80cd25bcfb Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase
Bug: None
Change-Id: I225fe1e16a3c96e5a03e3ae8fe975f368be7e6ad
Reviewed-on: https://webrtc-review.googlesource.com/c/107303
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25312}
2018-10-23 12:13:02 +00:00
Yves Gerey
988cc0870b [Cleanup] Add missing #include. Remove useless ones.
This CL is the result of running include-what-you-use tool on part
of the code base (audio target and dependencies) plus manual fixes.

bug: webrtc:8311
Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604
Reviewed-on: https://webrtc-review.googlesource.com/c/106280
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25311}
2018-10-23 11:32:56 +00:00
Gustavo Garcia
f7a7c8a8bc Stop adding RTT delay if there was not packet loss for enough time
Bug: none
Change-Id: I9105c0ee6a4f75381e133a496c3728dece9aca4b
Reviewed-on: https://webrtc-review.googlesource.com/c/23261
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25309}
2018-10-23 10:38:39 +00:00
Johannes Kron
9581bc4c52 Rename too long variable name to extmap_allow_mixed
Bug: webrtc:7990
Change-Id: I990111e473553163cecb9c73fec90d07c24aca02
Reviewed-on: https://webrtc-review.googlesource.com/c/107362
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25307}
2018-10-23 09:25:19 +00:00
Niels Möller
2edab4c026 Delete use of STR_CASE_CMP, replaced with absl::EqualsIgnoreCase.
Bug: webrtc:5876
Change-Id: Ica2d47ca45b8ef01a548d8dbe31dbed740a0ebda
Reviewed-on: https://webrtc-review.googlesource.com/c/106820
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25306}
2018-10-23 09:24:15 +00:00
Danil Chapovalov
98f5f6cdea In RtcpTransceiver functions with callback avoid relying on PostTaskAndReply
deprecated version guarantees (using PostTaskAndReply) callback task will run on the task queue,
and thus doesn't guarantee to run it if task queue is destroyed,

new callback versions instead guarantee callback will always run,
but may run off the task queue if task queue is destroyed.

Both keep guarantee observer callbacks will not run after on_destroyed/on_removed is called.

Bug: None
Change-Id: I61bf52127f3084c0186aa8bc89037bf9296801d8
Reviewed-on: https://webrtc-review.googlesource.com/c/107305
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25301}
2018-10-23 08:29:34 +00:00
Sam Zackrisson
b0ab2ce256 Reland "Remove the HighPassFilter interface"
Downstream Chromium dependencies fixed here:
https://chromium-review.googlesource.com/c/chromium/src/+/1286449

This is a reland of e2405c1a823f3baf90a9c72f2e058f91eb659c20

Original change's description:
> Remove the HighPassFilter interface
>
> The functionality remains unaffected.
> Filter toggling is still available via webrtc::AudioProcessing::Config.
> Example:
> webrtc::AudioProcessing::Config config = apm.GetConfig();
> // Read settings
> if (config.high_pass_filter.enabled) { ... }
> // Apply setting
> config.high_pass_filter.enabled = true;
> apm.ApplyConfig();
>
> Bug: webrtc:9535
> Change-Id: Ib4c4b04078bbb490ebdab9721b8c7811d73777a8
> Reviewed-on: https://webrtc-review.googlesource.com/c/102541
> Commit-Queue: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25198}

Bug: webrtc:9535
Change-Id: I0017193ad3ca1762e186f3ad79f29d33ef468202
Reviewed-on: https://webrtc-review.googlesource.com/c/106681
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25300}
2018-10-23 07:44:09 +00:00
Gustaf Ullberg
c9f9b8711f AEC3: Improve dominant nearend detection
This change makes the dominant nearend detection more accurate.
- The hangover is increased not leave nearend state between words.
- The SNR requirement is increased to not enter nearend state without
  speech activity.
- An early exit mechanism has been added to leave nearend state quickly
  when the echo is strong.

Bug: chromium:897701,webrtc:9897
Change-Id: I9e0f3e6ecb80eee1c0c917d4835f110555f74acf
Reviewed-on: https://webrtc-review.googlesource.com/c/107347
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25299}
2018-10-23 07:05:46 +00:00
Mirko Bonadei
ac19414512 Export symbols needed by the Chromium component build (part 6).
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: I67a4d016a11deca5ac5459826741dd2d3f7931d5
Reviewed-on: https://webrtc-review.googlesource.com/c/107400
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25298}
2018-10-23 06:48:51 +00:00
Jonathan Yu
9a5da497b5 Split out a separate target for SimulcastEncoderAdapter
This will allow downstream projects to use it to construct their own
injected codecs without pulling in dependencies on the software codecs.

Bug: webrtc:7925
Change-Id: If8628fedd18e57a51a8b6e5baf4f63a686bf52e8
Reviewed-on: https://webrtc-review.googlesource.com/c/107027
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Jonathan Yu <yujo@chromium.org>
Cr-Commit-Position: refs/heads/master@{#25297}
2018-10-22 22:36:59 +00:00
Ying Wang
68b2df7cf1 Make protection_overhead_rate configurable through field trial.
Bug: None
Change-Id: I0a4e03d5f8134809c44d2d0ff88c4b4c8b8a9eaa
Reviewed-on: https://webrtc-review.googlesource.com/c/107341
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25293}
2018-10-22 15:44:23 +00:00
philipel
6d2165036c Don't decode frames with an older timestamp than the last decoded timestamp.
This change prevents decoding corruption by not allowing keyframes with a
newer frame id but an older timestamp to be decoded. This does not handle
reordering well.

Bug: none
Change-Id: I4a67ca84ee86a782da74a10530c531d893d3bd3c
Reviewed-on: https://webrtc-review.googlesource.com/c/107304
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25292}
2018-10-22 13:11:46 +00:00
Sebastian Jansson
8d33c0c104 Adds field trial to do safer reset on route change.
Bug: webrtc:9718
Change-Id: I71143a9616981a24bca7bd5c663a9dae9fc9692e
Reviewed-on: https://webrtc-review.googlesource.com/c/106903
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25286}
2018-10-22 10:46:49 +00:00
Jesús de Vicente Peña
c98849cf92 AEC3: changes the signal used for deciding when to update the erle so the reverb render signal is now used
In this CL we change the signal that controls the updates of the ERLE estimator. Until now, the render signal was used which is not optimum for reverberant signals. In this CL, a reverberation has been added to the the render signal and this new signal has been used for controlling when to update the ERLE estimator.

Bug: webrtc:9873
Change-Id: I0ebea3fc208f97aa237af015ba543015d49ed978
Reviewed-on: https://webrtc-review.googlesource.com/c/105660
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25285}
2018-10-22 10:30:12 +00:00
Sebastian Jansson
ecdd432a32 Routing unacknowledged data in TransportFeedbackAdapter.
Bug: webrtc:9518
Change-Id: Ie5d016fb5e41645560502af8b8bff324f477229e
Reviewed-on: https://webrtc-review.googlesource.com/c/107302
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25284}
2018-10-22 09:32:11 +00:00
Mirko Bonadei
2dfa998be2 Reland "Prefix flag macros with WEBRTC_."
This is a reland of 5ccdc1331fcc3cd78eaa14408fe0c38d37a5a51d

Original change's description:
> Prefix flag macros with WEBRTC_.
>
> Macros defined in rtc_base/flags.h are intended to be used to define
> flags in WebRTC's binaries (e.g. tests).
>
> They are currently not prefixed and this could cause problems with
> downstream clients since these names are quite common.
>
> This CL adds the 'WEBRTC_' prefix to them.
>
> Generated with:
>
> for x in DECLARE DEFINE; do
>   for y in bool int float string FLAG; do
>     git grep -l "\b$x\_$y\b" | \
>     xargs sed -i "s/\b$x\_$y\b/WEBRTC_$x\_$y/g"
>   done
> done
> git cl format
>
> Bug: webrtc:9884
> Change-Id: I7b524762b6a3e5aa5b2fc2395edd3e1a0fe72591
> Reviewed-on: https://webrtc-review.googlesource.com/c/106682
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25270}

TBR=kwiberg@webrtc.org

Bug: webrtc:9884
Change-Id: I5ba5368a231a334d135ed5e6fd7a279629ced8a3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/107161
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25277}
2018-10-19 15:06:43 +00:00
Mirko Bonadei
c538fc77b0 Revert "Prefix flag macros with WEBRTC_."
This reverts commit 5ccdc1331fcc3cd78eaa14408fe0c38d37a5a51d.

Reason for revert: Breaks downstream project.

Original change's description:
> Prefix flag macros with WEBRTC_.
> 
> Macros defined in rtc_base/flags.h are intended to be used to define
> flags in WebRTC's binaries (e.g. tests).
> 
> They are currently not prefixed and this could cause problems with
> downstream clients since these names are quite common.
> 
> This CL adds the 'WEBRTC_' prefix to them.
> 
> Generated with:
> 
> for x in DECLARE DEFINE; do
>   for y in bool int float string FLAG; do
>     git grep -l "\b$x\_$y\b" | \
>     xargs sed -i "s/\b$x\_$y\b/WEBRTC_$x\_$y/g"
>   done
> done
> git cl format
> 
> Bug: webrtc:9884
> Change-Id: I7b524762b6a3e5aa5b2fc2395edd3e1a0fe72591
> Reviewed-on: https://webrtc-review.googlesource.com/c/106682
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25270}

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

Change-Id: Ia79cd6066ecfd1511c34f1b30fd423e560ed6854
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9884
Reviewed-on: https://webrtc-review.googlesource.com/c/107160
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25276}
2018-10-19 15:04:13 +00:00
Mirko Bonadei
5ccdc1331f Prefix flag macros with WEBRTC_.
Macros defined in rtc_base/flags.h are intended to be used to define
flags in WebRTC's binaries (e.g. tests).

They are currently not prefixed and this could cause problems with
downstream clients since these names are quite common.

This CL adds the 'WEBRTC_' prefix to them.

Generated with:

for x in DECLARE DEFINE; do
  for y in bool int float string FLAG; do
    git grep -l "\b$x\_$y\b" | \
    xargs sed -i "s/\b$x\_$y\b/WEBRTC_$x\_$y/g"
  done
done
git cl format

Bug: webrtc:9884
Change-Id: I7b524762b6a3e5aa5b2fc2395edd3e1a0fe72591
Reviewed-on: https://webrtc-review.googlesource.com/c/106682
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25270}
2018-10-19 10:55:20 +00:00
Benjamin Wright
192eeec14d Enable End-to-End Encrypted Video Frames.
This change integrates the FrameDecryptorInterface and the FrameEncryptorInterface into
the video send and receive path. If a FrameEncryptorInterface is set on an outgoing video RTPSender
then each outgoing video frame will first pass through the provided FrameEncryptor which
will have a chance to modify the payload contents for the purposes of encryption. In addition to
this the new GenericFrameDescriptor will be added as additional data.

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

Bug: webrtc:9795
Change-Id: I9f743ce0cb63df0cf070f6144be7ada078b4e5d2
Reviewed-on: https://webrtc-review.googlesource.com/c/103920
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25258}
2018-10-18 16:05:13 +00:00
Sebastian Jansson
78416b6e18 Adds time to initial config in analyzer code.
Bug: webrtc:9586
Change-Id: Ib5cbcdcf2cce3bea24d8c03a25f6cd415feb97ad
Reviewed-on: https://webrtc-review.googlesource.com/c/106880
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25254}
2018-10-18 12:43:31 +00:00
Per Åhgren
65faede3b0 AEC3: Introduce partial adaptive filter resets at echo path changes
With this CL, the main and shadow filters are no longer fully reset to
0 as the delay changes. This allows for more robust echo removal for
some scenarios.

Bug: webrtc:9879,chromium:895838
Change-Id: I859aa3df3ae41648bc8efde01ec2e2a5cb392279
Reviewed-on: https://webrtc-review.googlesource.com/c/106345
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25251}
2018-10-18 10:46:06 +00:00
Per Åhgren
1ffee36cb9 AEC3: Remove ERLE uncertainty code that has no effect
Removing code that has no audible effect.

Bug: webrtc:8671
Change-Id: Ibd7d0d19d760ae16b09285498c2ee09b42eb5968
Reviewed-on: https://webrtc-review.googlesource.com/c/106301
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25250}
2018-10-18 10:08:27 +00:00
Sebastian Jansson
b9972fa37b Adds AudioNetworkAdaptation support to Scenario tests.
Bug: webrtc:9718
Change-Id: I6cb976df5767797fec670134d29e030ec0f9d3a2
Reviewed-on: https://webrtc-review.googlesource.com/c/106340
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25236}
2018-10-17 15:42:58 +00:00
philipel
fab9129e94 Get frame type, width and height from the generic descriptor.
Bug: webrtc:9361
Change-Id: I5558ba02f921880f9c4677b85830c7c18faffea4
Reviewed-on: https://webrtc-review.googlesource.com/c/106382
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25231}
2018-10-17 13:31:09 +00:00
Minyue Li
34d990fef9 Adding NetEq buffer full metric to UMA.
BUG: webrtc:9882
Change-Id: Idbcbbbd99855b2251fbb66629efeab4f2d1f6498
Reviewed-on: https://webrtc-review.googlesource.com/c/106400
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25230}
2018-10-17 12:54:19 +00:00
Danil Chapovalov
5a464d3ee5 Add resolution to generic frame descriptor extension
Bug: None
Change-Id: Ifb5c5f4099d346b673032f41fa13d4ac65439e5d
Reviewed-on: https://webrtc-review.googlesource.com/c/106680
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25228}
2018-10-17 11:28:05 +00:00
Niklas Enbom
d895f42bfb Revert "Remove the HighPassFilter interface"
This reverts commit e2405c1a823f3baf90a9c72f2e058f91eb659c20.

Reason for revert: Breaks Chrome compile: https://logs.chromium.org/logs/chromium/buildbucket/cr-buildbucket.appspot.com/8932502586827763408/+/steps/compile__with_patch_/0/stdout 
Original change's description:
> Remove the HighPassFilter interface
> 
> The functionality remains unaffected.
> Filter toggling is still available via webrtc::AudioProcessing::Config.
> Example:
> webrtc::AudioProcessing::Config config = apm.GetConfig();
> // Read settings
> if (config.high_pass_filter.enabled) { ... }
> // Apply setting
> config.high_pass_filter.enabled = true;
> apm.ApplyConfig();
> 
> Bug: webrtc:9535
> Change-Id: Ib4c4b04078bbb490ebdab9721b8c7811d73777a8
> Reviewed-on: https://webrtc-review.googlesource.com/c/102541
> Commit-Queue: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25198}

TBR=solenberg@webrtc.org,saza@webrtc.org,peah@webrtc.org

Change-Id: Ieb34d5c573c4ab22eefbb54aeaa2f72844740b89
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9535
Reviewed-on: https://webrtc-review.googlesource.com/c/106421
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Commit-Queue: Niklas Enbom <niklas.enbom@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25215}
2018-10-16 15:51:45 +00:00
Danil Chapovalov
6026f05ef1 Calculate max payload size for an rtp packet to fit full video frame
instead of sometimes incorrectly guessing it

Bug: webrtc:9868
Change-Id: I8b15ecca4c660d83ea129dc9df6ec174ad83b4c6
Reviewed-on: https://webrtc-review.googlesource.com/c/106281
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25213}
2018-10-16 15:32:37 +00:00
Sebastian Jansson
f5e767dbbc Don't send max allocation probe unless allocation changed.
This changes the behavior to a probe only gets trigged if
the total max allocated bitrate  actually changed.

Also adding helpful log dump flag to ramp up tests that
was used to investigate the issue.

Bug: chromium:894434
Change-Id: I907675b8fd5a339f838b07d433ecf837e312def1
Reviewed-on: https://webrtc-review.googlesource.com/c/105981
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25212}
2018-10-16 15:13:57 +00:00
Per Åhgren
3e7b7b154b AEC3: Changes to initial behavior and handling of saturated echo
This CL introduces two related changes
1) It changes the way that the AEC3 determines whether the linear
filter is sufficiently good for its output to be used. The new scheme
achieves this much earlier than what was done in the legacy scheme.
2) It changes the way that saturated echo is and handled so that the
impact of the nearend speech is lower.

Bug: webrtc:9835,webrtc:9843,chromium:895435,chromium:895431
Change-Id: I0b493676886e2134205e9992bbe4badac7e414cc
Reviewed-on: https://webrtc-review.googlesource.com/c/104380
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25208}
2018-10-16 13:22:44 +00:00
Mirko Bonadei
276827cbdb Export symbols needed by the Chromium component build (part 3).
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).

Bug: webrtc:9419
Change-Id: I4d4e2ae52ee01de68147fd0f2cfe4c92d600ad94
Reviewed-on: https://webrtc-review.googlesource.com/c/106343
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25207}
2018-10-16 12:57:04 +00:00
Danil Chapovalov
6c78ff486a Always verify packet wasn't resend recently before resending it.
Pacer may accept same packet serveral time for resending,
packet may spend non-zero time in pacer queue.
As a result packet can be resend several time within one rtt
wasting bandwidth.

Bug: None
Change-Id: I753a5400b47d3804735e66e539a1b103916d0c94
Reviewed-on: https://webrtc-review.googlesource.com/c/106260
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25205}
2018-10-16 11:26:10 +00:00
Rasmus Brandt
2d0c68744c Remove |hw_encoder| and |hw_decoder| from VideoCodecTestFixture::Config.
Only used for output filename nowadays. Previously, it was used for
selecting the codec implementation. That is now done by injecting
the appropriate codec factory.

Bug: webrtc:9317
Change-Id: Ia2bf28f7df165fb65410ecd1f5d646ee6604e1be
Reviewed-on: https://webrtc-review.googlesource.com/c/106023
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25204}
2018-10-16 10:59:23 +00:00