694 Commits

Author SHA1 Message Date
Danil Chapovalov
59f3b71c04 Automate conversion from c++ VideoCodeType to java VideoCodecType
Bug: b/148146536
Change-Id: I030c7c6c2a1a9d002bcc60f45c8d6025bd0935b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167301
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30751}
2020-03-11 08:02:36 +00:00
Danil Chapovalov
4e1d6ce384 Rename java VideoCodecType to VideoCodecMimeType
to avoid collission and confusion with VideoCodeType based on
c++ enum with the same name.

Bug: b/148146536
Change-Id: I049cce21d59f454c7ce507fdfc3a85d168f96223
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170048
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30728}
2020-03-09 15:27:45 +00:00
Florent Castelli
b05ca4b616 Implement new specification for degradation preference
The degradation preference is now based on the content hint of the track
if it's unspecified.

Bug: webrtc:11164
Change-Id: Iaa0dbf1c1bf68a46fc5131e534d423c30c5439c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161233
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30691}
2020-03-05 14:24:25 +00:00
Courtney Edwards
134c6996c8 Fix Chromium Roll failing because of -Wrange-loop-construct
Bug: webrtc:11398
Change-Id: I51f6f9968b3a94b5fec325e8b5d29fd2bb290ee1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169553
Commit-Queue: Courtney Edwards <courtneyfe@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30669}
2020-03-03 13:04:25 +00:00
Mirta Dvornicic
4f34d78c85 Report available instead of encoding bitrate to VideoEncoderSelector.
The encoding bitrate might be limited depending on the current encoder.

Bug: webrtc:11341
Change-Id: I734fce12734b1e703e7948847cdb1365c08a137b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169123
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30619}
2020-02-26 15:56:36 +00:00
philipel
cddfc46db6 Added java interface VideoEncoderFactory.VideoEncoderSelector and implemented VideoEncoderSelectorWrapper.
Bug: webrtc:11341
Change-Id: Ic15658e09643aec119a97ddfaebfdb72ba3407c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168487
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30519}
2020-02-13 17:23:15 +00:00
Sami Kalliomäki
0f6bcd18b2 Hold a reference to AndroidVideoTrackSource while calling onFrameCaptured.
This makes it safe to deliver frames to the sink from VideoProcessor
even after setSink has been called with null reference without danger
of use after free.

Bug: b/148063550
Change-Id: Ib78f75ac49fc6117f744c55da1a4e671bbdcdf22
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168160
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30455}
2020-02-04 15:00:05 +00:00
Steve Anton
f417238217 Remove iceRegatherIntervalRange
This was an ICE configuration experiment added a couple years ago that did not end up being used.

Bug: webrtc:11316
Change-Id: Iafb7e1c4f7b4598815f045808dbf6e470172f119
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167680
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30395}
2020-01-28 19:16:18 +00:00
Jonas Olsson
b2b2031457 Concatenate string literals at compile time.
This CL was generated by running:
git ls-files | grep ".cc" | xargs perl -i -ne 'BEGIN {undef $/}; s/("[\s\n]*<<[\s\n]*")/" "/g; print;'; git cl format

After that I manually edited modules/audio_processing/gain_controller2.cc to preserve its original
formatting.

This primary benefit of this change is a small reduction in binary size.

Bug: None
Change-Id: I689fa7ba9c717c314bb167e5d592c3c4e0871e29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165961
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30251}
2020-01-14 14:47:48 +00:00
Sebastian Jansson
6ea2c6ae87 Cleanup: Merges Thread and MessageQueue.
Since rtc::Thread is the only class inheriting from rtc::MessageQueue
and most members of MessageQueue are public or protected the split is
not adding much value. In preparation for future cleanup, this cl merges
the two classes.

Bug: webrtc:9883
Change-Id: Ia0efb4349f66f653aa34fa4d244998f187e3ce36
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165340
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30235}
2020-01-13 13:53:20 +00:00
Sebastian Jansson
290de82b2a Cleanup: Replace MessageQueue pointers with Thread pointers.
This is part of a CL series merging rtc::MessageQueue into rtc::Thread.

Bug: webrtc:9883
Change-Id: I4a1bcd44c9523b6402b3f05b50597bdc2e6615e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165345
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30216}
2020-01-10 19:03:12 +00:00
Aaron Alaniz
415e39da56 Update Android camera switch API to allow specifying a name
The current camera switch API sequentially cycles through each
camera name for each method invocation. This policy provides
reasonable behavior for devices with 2 or 3 cameras, but
presents challenges with devices that contain several cameras.
For example in a scenario where the current camera is oriented
on the same side as the next camera name, a developer would need to
call switchCamera multiple times to capture from a camera oriented on
a different side of the device.

This commit allows a developer to specify a camera name when switching
cameras. This flexibility allows developers to have more control over
which device they switch to in cases where a device contains several cameras.

Bug: webrtc:11261
Change-Id: I93d46d70b2c7cf735a411a4ef4f33e926bf3a5ea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165040
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30199}
2020-01-09 16:04:09 +00:00
henrika
b40f75e427 Avoids crash in ADM for Android
Tbr: henrik.lundin
Bug: webrtc:11270
Change-Id: I1b3ad0afe3f5072ea4529e89729b087a4bd29fec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165396
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30197}
2020-01-09 15:02:16 +00:00
Florent Castelli
266021dfa2 Add support for DegradationPreference in Android SDK
This wires the current degradation preference in the SDK, it will later
be nullable in a follow up change once the native API supports it.

Bug: webrtc:11164
Change-Id: I8324e6e0af996dfddfa07e3aff4ba242d9533388
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161321
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30170}
2020-01-07 17:20:41 +00:00
Raman Budny
ec7b36cddf Added exception handling to EncodedImage's release callback.
Bug: webrtc:11230
Change-Id: Iad5bb4470891fbaea6b83ba647c8b4bbc4e38c72
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162803
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30165}
2020-01-07 12:57:52 +00:00
Alex Narest
d2fb5f510f Fixes WebRtcAudioTrack crash while stopping
TBR=henrika@webrtc.org

Bug: webrtc:11248
Change-Id: I5b829b5193d2accdfbf1e06c5317a5cd441c48c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163990
Commit-Queue: Alex Narest <alexnarest@google.com>
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Alex Narest <alexnarest@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30133}
2020-01-02 16:03:54 +00:00
Niels Möller
82f33c566a Delete transitional method EncodedImage.maybeRetain
Bug: webrtc:9378
Change-Id: Ibe3d5bad835d1725faa38f8e2a804efc9272776e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155661
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30072}
2019-12-12 14:11:14 +00:00
Byoungchan Lee
5f728fc04f Fix nullablity on CameraCapturer
Both cameraThreadHandler and surfaceHelper shouldn't be null.

Bug: None
Change-Id: I3c239c4275c53b836bbc2e9d6af71bf2b1b65387
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161480
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30047}
2019-12-10 08:33:15 +00:00
Niels Möller
5b030cabcc Change jni VideoEncoderWrapper to not use the encoder task queue
If the task to call OnEncodedImage is posted to the encoder task queue
just after VideoStreamEncoder::Stop post the task to release the
encoder, the destruction sequence of java HardwareVideoEncoder
deadlocks in outputBuffersBusyCount.waitForZero();

Encoders are generally allowed to call OnEncodedImage on any internal
encoder thread, so posting to the encoder task queue seems unnecessary.

Bug: webrtc:9378
Change-Id: Iee14f151d9efdc5ab348f9c86069fdb762e6a0dc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161447
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30035}
2019-12-09 10:11:00 +00:00
Saurav Das
934afc6ba1 Deprecate RtpReceiver's SetParameters method
This removes the SetParameters method from AudioRtpReceiver and Video
RtpReceiver, which is currently not used and is not part of the
specifications.


Bug: webrtc:11111
Change-Id: I6f67773bfef2d4b51e9ab670bde17b5fbf5f94c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159307
Reviewed-by: Patrik Höglund <phoglund@google.com>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Daniela Jovanoska Petrenko <denicija@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Saurav Das <dinosaurav@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29995}
2019-12-03 19:50:42 +00:00
Mirko Bonadei
fe7ce1c3bc Fix ErrorProne MultiVariableDeclaration.
This check has been turned on in [1] and it is now preventing the
Chromium Roll into WebRTC.

[1] - https://chromium-review.googlesource.com/c/chromium/src/+/1939956

TBR: sakal@webrtc.org
Bug: None
Change-Id: I43372eb3b3987bdf91bc717a6f50be3d8b1db56c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161006
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@google.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29954}
2019-11-28 18:49:20 +00:00
Mirko Bonadei
9f9e20a3dc Fix errorprone issues preventing Chromium Roll.
Some ErrorProne warnings have been enabled by [1], that broke the
Chromium Roll into WebRTC, this CL should have taken care of all the
problems.

[1] - https://chromium-review.googlesource.com/c/chromium/src/+/1935889

Bug: None
Change-Id: I2670e948c320984a122fdb774b891c98e05f582e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160862
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29933}
2019-11-27 12:52:48 +00:00
Ivo Creusen
fba448178c Make it possible to inject a custom NetEqFactory from the java interface.
Bug: webrtc:11005
Change-Id: I18b17847a6e066335f96ca1b718af2388805f8fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160183
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29900}
2019-11-25 12:39:08 +00:00
Xavier Lepaul
6e9d0d38ef Make base classes for native video encoder/decoder public
Implementers of Java wrappers for native encoders need to have the same
implementation of all the unsupported methods, as mentioned in the
documentation of VideoEncoder.createNativeVideoEncoder (and its decoder
equivalent).

This simplifies implementation of such encoders/decoders, and also make sure
they don’t override unsupported methods, as they are guaranteed not to be
called.

Bug: None
Change-Id: Iaa8499eda1b52cc14b04622bea2766cd09ba43e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160186
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Xavier Lepaul <xalep@google.com>
Cr-Commit-Position: refs/heads/master@{#29866}
2019-11-21 17:04:50 +00:00
Raman Budny
ac7fd87375 Force alignment of generated JVM called functions.
This CL effectively expands the zone of influence of
https://webrtc-review.googlesource.com/64160,
forcing 16-byte stack alignment of generated JNI methods
for the Android x86 platform.

Bug: webrtc:9085
Change-Id: Idc40c00ea3fb52dbbbeac7b58ceda2a9a44733d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159928
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29858}
2019-11-21 12:34:35 +00:00
Sami Kalliomäki
b86a1770ee Expose ABGRToI420 in YuvHelper.
Bug: None
Change-Id: I59947339a3a4bb683211ec3c00713ccfbf35bc40
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160182
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29855}
2019-11-21 12:02:30 +00:00
Yves Gerey
29e07e5080 Add @Nullable annotations to quiet errorprone.
Those are preventive annotations to prepare for incoming android update
(coming with Chromium roll).
Currently the roll is blocked partly because errorprone complains!

Bug: webrtc:11095, chromium:1003532
Change-Id: If4e2879a522e895ce7fb1f2a9ad36d06f98f2a61
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160002
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#29830}
2019-11-19 12:50:30 +00:00
Jakob Ivarsson
017c84f3ea Synchronize is_screencast_ state in AndroidVideoTrackSource.
Follow up to https://webrtc-review.googlesource.com/c/src/+/159689.

Bug: None
Change-Id: I3f2b481db091d405c1b00ca18c2e7ce5f3375607
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159702
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29790}
2019-11-13 14:03:09 +00:00
Jakob Ivarsson
c5ec54e51b Add SetIsScreencast method to VideoSource.
Bug: None
Change-Id: Iec0bb066b8100fa1d4bd095f78a0473933d1e30d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159689
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29785}
2019-11-13 10:30:36 +00:00
Honghai Zhang
3c0e86a87d Add a field trial to use only the higher 64 bits to find network handle from an ipv6 address.
Bug: webrtc:11067
Change-Id: Ib4f069981f7641f67436757a8592ab0f168a9a6e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158800
Commit-Queue: Honghai Zhang <honghaiz@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29697}
2019-11-05 20:27:50 +00:00
philipel
16cec3be2c Added allow_codec_switching parameter to RTCConfig.
Bug: webrtc:10795
Change-Id: I5507f1d801e262223bd18198c685b5fffa644b0b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157891
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29612}
2019-10-25 11:06:31 +00:00
Honghai Zhang
f8998cf8c4 Add a turn port prune policy to keep the first ready turn port.
Bug: webrtc:11026
Change-Id: I6222e9613ee4ce2dcfbb717e2430ea833c0dc373
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155542
Commit-Queue: Honghai Zhang <honghaiz@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29470}
2019-10-14 19:08:23 +00:00
Cyril Lashkevich
fa77ba6af1 SetStreams API of RtpSender wrapped for iOS and Android
Bug: webrtc:10129
Change-Id: I36ea0110de655bbffa2bd18a024abd15a2136838
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155983
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29405}
2019-10-08 13:51:19 +00:00
Niels Möller
7c2bed8337 Avoid memcpy in JavaToNativeEncodedImage
Followup to https://webrtc-review.googlesource.com/c/src/+/142160

Bug: webrtc:9378
Change-Id: If790cd628433046d6819a92449fcc68106535df4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154561
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29359}
2019-10-01 12:55:44 +00:00
Niels Möller
ef3dbad49a New class ScopedJavaRefCounted
Intended to be used for holding on to references to the java
EncodedImage and call its release method when no longer used by C++.

Bug: webrtc:9378
Change-Id: I40d917c2bb4217419ef2d609e517566c8466a274
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154740
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29347}
2019-09-30 14:43:56 +00:00
henrika
ee8ee2f103 Avoids update of WebRTC.Audio.SourceMatchesRecordingSession for Android < N
Before this change we always logged false in WebRTC.Audio.SourceMatchesRecordingSession
even when a test had not been executed (happens e.g. for Android < N).

This issue is now fixed and we only update WebRTC.Audio.SourceMatchesRecordingSession
if a valid test has been performed.

No-Try: True
TBR: glaznev
Bug: webrtc:10971
Change-Id: I907197476f00b812c67bb71e8fdcd6f297cfbdee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154563
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29324}
2019-09-26 14:59:12 +00:00
Niels Möller
67309ef93c Add release callback and reference count to java EncodedImage class
Callback set by HardwareVideoEncoder, and wired to the codec's
releaseOutputBuffer. Intention is to move call of this method to the
destructor of a corresponding C++ class in a followup cl, and
eliminate an allocation and memcpy in the process.

Bug: webrtc:9378
Change-Id: I578480b63b68e6ac7a96cdde36379b3c50f05c3f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142160
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29283}
2019-09-24 12:26:09 +00:00
henrika
14137a1064 Adds logging of audio sessions status on the recording side in ADM for Android.
Goal is to be able to retrieve more details about possible microphone conflicts in
cases where Init/Start of audio recording fails.

Only supported on Android N and higher.

Also adds new boolean UMA histogram called WebRTC.Audio.SourceMatchesRecordingSession.
Its value is stored after the recording session has been stopped.

Does not affect the media flow or functionality of the ADM. Time to start audio should
not be affected either since the new check and logging takes place on a separate
ExecutorService thread.

See go/webrtc-adm-android for more details and examples.

Bug: webrtc:10971
Change-Id: Ia80c1534e326907a1582824225d5f58caa016922
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150793
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29236}
2019-09-19 11:35:10 +00:00
Mirko Bonadei
317a1f09ed Use std::make_unique instead of absl::make_unique.
WebRTC is now using C++14 so there is no need to use the Abseil version
of std::make_unique.

This CL has been created with the following steps:

git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt
git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt
git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt

diff --new-line-format="" --unchanged-line-format="" \
  /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \
  uniq > /tmp/only_make_unique.txt
diff --new-line-format="" --unchanged-line-format="" \
  /tmp/only_make_unique.txt /tmp/memory.txt | \
  xargs grep -l "absl/memory" > /tmp/add-memory.txt

git grep -l "\babsl::make_unique\b" | \
  xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g"

git checkout PRESUBMIT.py abseil-in-webrtc.md

cat /tmp/add-memory.txt | \
  xargs sed -i \
  's/#include "absl\/memory\/memory.h"/#include <memory>/g'
git cl format
# Manual fix order of the new inserted #include <memory>

cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \
  xargs sed -i '/#include "absl\/memory\/memory.h"/d'

git ls-files | grep BUILD.gn | \
  xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d'

python tools_webrtc/gn_check_autofix.py \
  -m tryserver.webrtc -b linux_rel

# Repead the gn_check_autofix step for other platforms

git ls-files | grep BUILD.gn | \
  xargs sed -i 's/absl\/memory:memory/absl\/memory/g'
git cl format

Bug: webrtc:10945
Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 15:47:29 +00:00
henrika
69f8c42d2c [RELAND] Add support of AudioRecord.Builder in the ADM for Android
Now fixed issue which caused http://b/140707892

First version was reverted in https://webrtc-review.googlesource.com/c/src/+/152526.
The mistake I had done in the original version was that I missed that the new
builder could throw a different type of exception and it was never caught.

TBR: glaznev@webrtc.org
Bug: webrtc:10942
Change-Id: I0e11511936d2d25681a1ffae3bbd367095fee7a1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152664
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29164}
2019-09-12 11:44:20 +00:00
Hari Molabanti
a1727db1ac Revert "Add support of AudioRecord.Builder in the ADM for Android"
This reverts commit 24b945d60526f8074d0db1329ba20e9b49602794.

Reason for revert: Caused http://b/140707892

Original change's description:
> Add support of AudioRecord.Builder in the ADM for Android
> 
> Use the latest builder class for AudioRecord instead of the old
> constructor. AudioTrack has been updated for a while now.
> 
> Bug: webrtc:10942
> Change-Id: Ia68b12e5aaf1525cfa630650fbaaa02d70ada15f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151305
> Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
> Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29072}

TBR=henrika@webrtc.org,glaznev@webrtc.org

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

Bug: webrtc:10942
Change-Id: Idbc487cf8d42e76f6a3435be6fef6634aa0cd62b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152526
Reviewed-by: Daixiang Mou <dmou@webrtc.org>
Commit-Queue: Daixiang Mou <dmou@webrtc.org>
Commit-Queue: Hari Molabanti <harimb@google.com>
Cr-Commit-Position: refs/heads/master@{#29159}
2019-09-11 18:37:03 +00:00
henrika
4d6b2691bd Adds setAudio[Track/Record]StateCallback interfaces to the Java ADM
Bug: webrtc:10950
Change-Id: Ifa7bd7eb003bf97812ce0dfa5a0192ee8955419c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151648
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29107}
2019-09-09 08:10:41 +00:00
henrika
24b945d605 Add support of AudioRecord.Builder in the ADM for Android
Use the latest builder class for AudioRecord instead of the old
constructor. AudioTrack has been updated for a while now.

Bug: webrtc:10942
Change-Id: Ia68b12e5aaf1525cfa630650fbaaa02d70ada15f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151305
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29072}
2019-09-05 07:59:30 +00:00
Qingsi Wang
7cdcda9dd5 Use the sanitized pair when surfacing the candidate pair change event.
TBR=andersc@webrtc.org

Bug: None
Change-Id: Ie2c389fe966dada2768e3222e1f8da74e1715568
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150762
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Alex Drake <alexdrake@webrtc.org>
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29052}
2019-09-03 17:17:49 +00:00
Alex Narest
44dc241ae8 Allows configuration of playout audio buffer
Playout audio buffer length in Java audio device configuration with fieldtrial.

Bug: webrtc:10928
Change-Id: I79286f09591f4b2c6a6146f23d3dce92a29f6b21
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150657
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Alex Narest <alexnarest@google.com>
Cr-Commit-Position: refs/heads/master@{#29005}
2019-08-29 12:57:14 +00:00
Jonas Oreland
228900f8b1 Add TURN_LOGGING_ID to android sdk
This patch adds support for setting the TURN_LOGGING_ID
in RTCConfig using the android SDK.

TURN_LOGGING_ID was added to webrtc in
https://webrtc-review.googlesource.com/c/src/+/149829

The intended usage of this attribute is to correlate client and
backend logs.

bug: webrtc:10897
Change-Id: Ifd62e0f1dac396942c76a794bf7a75553d3244b7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150538
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28996}
2019-08-29 06:55:42 +00:00
Sami Kalliomäki
fdd2340311 Revert "Detect leaks of TextureBufferImpl objects."
This reverts commit 44bd29a3b068363e013cd425c68fd00dba21d633.

Reason for revert:
Going for an alternative implementation that makes this unnecessary
https://webrtc-review.googlesource.com/c/src/+/150649

Original change's description:
> Detect leaks of TextureBufferImpl objects.
>
> The performance cost is not trivial but according to my profiling,
> it is acceptable.
>
> Bug: b/139745386
> Change-Id: I0e63221ccf22e9f6fb32c630ff63a279e765994a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150539
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28973}

TBR=sakal@webrtc.org,kthelgason@webrtc.org

Change-Id: Ic6266e5fd24389d41a6d5dbfe51de6505b861b12
Bug: b/139745386
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150650
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28983}
2019-08-28 12:35:04 +00:00
Sami Kalliomäki
44bd29a3b0 Detect leaks of TextureBufferImpl objects.
The performance cost is not trivial but according to my profiling,
it is acceptable.

Bug: b/139745386
Change-Id: I0e63221ccf22e9f6fb32c630ff63a279e765994a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150539
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28973}
2019-08-27 13:53:48 +00:00
Niels Möller
2579f0c584 RTCError as return type for PeerConnectionInterface::SetConfiguration
Bug: None
Change-Id: I6dd7378ceac617e29945d72906cb8e2e0bd49538
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149166
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28910}
2019-08-20 06:52:05 +00:00
Alex Narest
bbeb10925e Reporting audio device underrun counter
Bug: webrtc:10884
Change-Id: I35636fcbc1e2a19a89242379cdff6ec5c12fd21a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149200
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Alex Narest <alexnarest@google.com>
Cr-Commit-Position: refs/heads/master@{#28874}
2019-08-16 11:49:55 +00:00