This CL fixes the problem of misfired negotiationneeded notifications due
to the lack of a NegotiationNeeded slot and the proper procedure to
update it.
Change-Id: Ie273c691f11316c9846606446f6cf838226b5d5c
Bug: chromium:740501
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131283
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27594}
This is a reland of 7ac0d5f348
Original change's description:
> Replace usage of old SetRates/SetRateAllocation methods
>
> This rather large CL replaces all relevant usage of the old
> VideoEncoder::SetRates()/SetRateAllocation() methods in WebRTC.
> API is unchanged to allow downstream projects to update without
> breakage.
>
> Bug: webrtc:10481
> Change-Id: Iab8f292ce6be6c3f5056a239d26361962b14bb38
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131949
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27554}
TBR=brandtr@webrtc.org,sakal@webrtc.org,perkj@webrtc.org
Bug: webrtc:10481
Change-Id: I2978d5c527a18e885b7845c4e53a2424e8ad5b4b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132551
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27593}
This reverts commit 7ac0d5f348.
Reason for revert: <INSERT REASONING HERE>
Original change's description:
> Replace usage of old SetRates/SetRateAllocation methods
>
> This rather large CL replaces all relevant usage of the old
> VideoEncoder::SetRates()/SetRateAllocation() methods in WebRTC.
> API is unchanged to allow downstream projects to update without
> breakage.
>
> Bug: webrtc:10481
> Change-Id: Iab8f292ce6be6c3f5056a239d26361962b14bb38
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131949
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27554}
TBR=brandtr@webrtc.org,sakal@webrtc.org,nisse@webrtc.org,sprang@webrtc.org,perkj@webrtc.org
Change-Id: I576760b584e3f258013b0279c0c173c895bbb37e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10481
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132561
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27559}
This rather large CL replaces all relevant usage of the old
VideoEncoder::SetRates()/SetRateAllocation() methods in WebRTC.
API is unchanged to allow downstream projects to update without
breakage.
Bug: webrtc:10481
Change-Id: Iab8f292ce6be6c3f5056a239d26361962b14bb38
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131949
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27554}
Add base class NetworkPredictor and NetworkPredictorFactory in /api, make it possible to inject customized NetworkPredictor in PeerConnectionFactory level. The NetworkPredictor object will be pass down to GoogCCNetworkControl and DelayBasedBwe.
Bug: webrtc:10492
Change-Id: Iceeadbe1c9388b11ce4ac01ee56554cb0bf64d04
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130201
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27543}
This is a reland of 177670afd6
Fixing failing tests.
TBR=magjed@webrtc.org
Original change's description:
> Add bindings for simulcast and RIDs in Android SDK.
>
> This adds the bindings for rid in RtpParameters.Encoding and bindings
> for send_encodings in RtpTransceiverInit to allow creating a transceiver
> with multiple send encodings.
>
> Bug: webrtc:10464
> Change-Id: I4c205dc0f466768c63b7efcb3c68e93277236da0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128960
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Seth Hampson <shampson@webrtc.org>
> Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27323}
Bug: webrtc:10464
Change-Id: I95fac3967217c20a9fdddb490aea30eca2061ef0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130362
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27402}
And delete corresponding dependencies on :webrtc_common. After this
change, common_types.h is included directly only from code in the
following directories:
api/
api/video/
api/video_codecs/
common_video/libyuv/include/
media/base/
modules/remote_bitrate_estimator/
modules/rtp_rtcp/source/
modules/video_coding/codecs/vp9/
There remains plenty of indirect dependencies on the types declared in
common_types.h, but the fewer direct dependencies should make it
easier to find the proper place for each type.
Bug: webrtc:5876
Change-Id: I93e8f214025ecb613c19fdec2015bd3f96c59aae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130501
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27376}
This reverts commit 177670afd6.
Reason for revert: Fails android_instrumentation_test_apk:
https://ci.chromium.org/p/webrtc/builders/ci/Android64%20(M%20Nexus5X)/11553
Original change's description:
> Add bindings for simulcast and RIDs in Android SDK.
>
> This adds the bindings for rid in RtpParameters.Encoding and bindings
> for send_encodings in RtpTransceiverInit to allow creating a transceiver
> with multiple send encodings.
>
> Bug: webrtc:10464
> Change-Id: I4c205dc0f466768c63b7efcb3c68e93277236da0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128960
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Seth Hampson <shampson@webrtc.org>
> Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27323}
TBR=magjed@webrtc.org,shampson@webrtc.org,amithi@webrtc.org
Change-Id: Id6c4e2d41c3c2fbfad31baed907cfa73d82ef14a
No-Tree-Checks: True
No-Try: True
Bug: webrtc:10464
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130466
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27354}
Since it is a WebRTC-only macro, let's prefix it with WEBRTC_.
Bug: None
Change-Id: I309666858ea898dc7cd1a68c21be190f98c87b11
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129935
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27327}
This adds the bindings for rid in RtpParameters.Encoding and bindings
for send_encodings in RtpTransceiverInit to allow creating a transceiver
with multiple send encodings.
Bug: webrtc:10464
Change-Id: I4c205dc0f466768c63b7efcb3c68e93277236da0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128960
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27323}
Even if neither frame height nor frame width is <=0 we can end up
with <=0 dimensions in renderHeight or renderWidth. With this change,
we perform the check on the latter.
Bug: webrtc:10367
Change-Id: I9672672659ad7d12cf1e7ccab5b5cde583ae7e8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129760
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27307}
Injection is made possible through VP8Encoder::Create.
According to native-api.md, it is a defacto public API despite
not being in the api/ folder.
Bug: webrtc:10259, webrtc:10382
Change-Id: Ifc5d55aa99613cfee0fcb4f0c6690121c85b2e3e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128883
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27281}
After being stuck "forever" (3 seconds) waiting for an event to
trigger, log the stack trace of the current thread to aid debugging of
the deadlock.
Bug: webrtc:10308
Change-Id: I04852f191027294d7e7a7f5e63de4c6c7fdd6326
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128342
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27263}
Use rtc::scoped_refptr instead of std::unique_ptr to hold the instance
of OpenSLEngineManager; this makes it safe to share it between
OpenSLESRecorder and OpenSLESPlayer.
Bug: webrtc:10436
Change-Id: Ibd0717e5410020c89a40bfdb05953a02378a6a4b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128651
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27253}
We may sometimes want to override only input or only output, or
override them with different values. This change allows setting the
overrides separately.
Change-Id: Ib0c44cb7a3cfa834f997fb6cd54f7cad68705f41
Bug: webrtc:10441
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128763
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27236}
This CL adds a way to extract the underlying android.opengl.EGLContext
and javax.microedition.khronos.egl.EGLContext for EglBase14 and
EglBase10 respectively. The reason is that clients can't be expected to
use only WebRTC's OpenGL code and might need to integrate with their
own GL code.
Bug: None
Change-Id: Ie00a564de45a090683542a52005da7e43c586ced
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127888
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27205}
The stacktrace unit test was flaking on arm32; my theory is that this
happened when the thread whose stack we were dumping was doing a
system call inside `params->deadlock_start_event.Set();` in
ThreadFunction(). (This would be a problem because, according to the
comment at the bottom of the file, "stack traces originating from
kernel space do not include user space stack traces for ARM32.")
Attempt to solve this problem by spinning on an atomic flag instead,
since this involve no system calls. And add a short sleep to the main
thread, to give the other thread time to get from the barrier to the
thing it's actually supposed to deadlock on.
Bug: webrtc:10420
Change-Id: I4c6392157c8a06c64cb11146ffe9368e5bade6fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128340
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27158}
The video decoder thread is the pilot user.
For now this is an Android-only feature, since that's the only
platform we can print stack traces on.
Bug: webrtc:9987
Change-Id: Ie638c619673b5f159d91a32683fd787baf46479a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126222
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27127}
This cl deprecates the FrameType enum, and adds aliases AudioFrameType
and VideoFrameType.
After downstream usage is updated, the enums will be separated
and be moved out of common_types.h.
Bug: webrtc:6883
Change-Id: I2aaf660169da45f22574b4cbb16aea8522cc07a6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123184
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27011}
This is preparing for upstream removing the alias java_groups for the
individual support library targets: https://crrev.com/c/1500780
Bug: chromium:937987
Change-Id: I1c9efd83f6997288b334f3dc2f41233fa4e2ab61
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125961
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Peter Wen <wnwen@chromium.org>
Cr-Commit-Position: refs/heads/master@{#26995}
Ignore rtc_link_task_queue_impl flag,
instead use build_with_chromium for custom chromium implementation injection
This changes TaskQueue implementation used in webrtc fuzzers in chromium:
from own webrtc implementation to chromium's.
Bug: webrtc:10191
Change-Id: I63be28b680ae8ea8ee1dbf0c699263c392ce29d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125196
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26977}
Previously, we have created a Legacy ADM when no ADM is supplied.
With this change we will start creating a Java ADM instead.
The end goal is to make injection mandatory, and never creating ADMs.
This is one step on the way, and will allow us to clean up the Legacy
ADM code.
Bug: webrtc:7452
Change-Id: Ib99adc50346fe6b748f9435d2fc6321a50c3ee4e
Reviewed-on: https://webrtc-review.googlesource.com/c/123887
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26949}
CodecSpecificInfo has a default constructor, so initializing by memset is not necessary and is in the way of adding non-trivial members.
Related chromium CL: https://chromium-review.googlesource.com/c/chromium/src/+/1495533
Bug: webrtc:10342
Change-Id: I36046f919f5fc34ea51de7288ff5c9cc0f2950b8
Reviewed-on: https://webrtc-review.googlesource.com/c/125093
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26924}
Since there is no way to enable/disable these diagnostics at runtime,
this CL moves the suppression into the rtc_* templates in order to
remove the need to explicitly add the snippet of code needed to
suppress it (currently copy/pasted in 144 locations).
The diagnostic that causes the most problems is the one about "complex
class/struct explicit ctor/dtor" [1] because WebRTC doesn't find
it useful enough.
Other diagnostics are good (for example the one that warns about
using "virtual" instead of "override", but that will be covered by
this clang-tidy check [2]) while others are Chromium related so
they have never triggered.
[1] - https://cs.chromium.org/chromium/src/tools/clang/plugins/FindBadConstructsConsumer.cpp?l=147-167&rcl=b4bebe1aa15dba7ca5fcc6456a81a55665327c3a
[2] - https://clang.llvm.org/extra/clang-tidy/checks/modernize-use-override.html
Bug: webrtc:163
Change-Id: Icbf27efa5b369100a31e6a32df1a0913729b3b34
Reviewed-on: https://webrtc-review.googlesource.com/c/125088
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26918}
While passing negative delta is an error it is not fatal and recovered next line.
Bug: None
Change-Id: I3b9ce234a7763ba92bd158c9eda8ba4bd7a06f4b
Reviewed-on: https://webrtc-review.googlesource.com/c/124702
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26916}
There has been some crashes due to frames having illegal sizes, most
likely 0x0. Probably these frames are created as a workaround for
something.
It would be best to stop 0x0 frames from being created in the first
place, but a reasonable quick fix is to just not draw those frames.
Bug: webrtc:10367
Change-Id: Ib93057c4de7285773c99614b4e7d9bd4b099c4dc
Reviewed-on: https://webrtc-review.googlesource.com/c/124988
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26897}
It was previously possible to escape the sandbox by calling
rtc::Thread::SetAllowBlockingCalls(true).
This CL only removes the loophole on non-Android builds, because we
still have old Android code that relies on it. We expect that code to
go away soon-ish, though.
Bug: webrtc:9987
Change-Id: Ida96400d0abe430af4c2046284795d37d64f6613
Reviewed-on: https://webrtc-review.googlesource.com/c/123523
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26792}
Added audio format field and set method to Builder. - WebRTCAudioRecord. Added audio format field, added to constructor. Default audio format value AudioFormat.ENCODING_PCM_16BIT. initRecord(), added how to calculate bytesPerFrame, depends on audioFormat.
First commit and contribution, updated AUTHORS file
Bug: None
Change-Id: I16f660d42350ec9ce2e329b239bd7f6324e76dfe
Reviewed-on: https://webrtc-review.googlesource.com/c/122302
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26775}