Added functionality on the FakeNetworkPipe to introduce arbitrary
clock offsets. This offset is added to the reported receive time of
all packets. This prepares for a later CL using this to test correction
of receive time stamps.
Bug: webrtc:9054
Change-Id: I811b3aa8359bc917f59443088d8a418368242db9
Reviewed-on: https://webrtc-review.googlesource.com/64726
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22763}
This will enable changing thresholds when switching between hardware
and software encoders. It is also a partial revert of
https://webrtc-review.googlesource.com/33340: construction of the
OveruseFrameDetector is still in VideoSendStream, but configuration is
moved back to VideoStreamEncoder.
Longer term, information about HW vs SW, or generally, about resources
consumed by the encoder, should be passed in the per-frame callbacks
to OveruseFrameDetector, and then the CpuOveruseOptions could move
back to construction time.
Bug: webrtc:8504, webrtc:8830
Change-Id: I44577519d4e05356730cac9bd9ae3c74bfc17ed7
Reviewed-on: https://webrtc-review.googlesource.com/65163
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22761}
A downstream bug ocurred because of a lack of symmetry when adding and
removing a remote sender in Plan B that specifies SSRCs, but doesn't
specify stream IDs. The issue when the first remote description is
applied "default" for the stream ID on the remote sender, but the
second time it's applied the current remote sender's "default" stream
ID does not match the new remote description's empty stream ID. This
was incorrectly interpreted as a new remote sender (which removed/added
the sender).
Bug: webrtc:7933
Change-Id: I87191b9e887b3450ef15111b5e867023c723a86e
Reviewed-on: https://webrtc-review.googlesource.com/67191
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Noah Richards <noahric@chromium.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22760}
This changeset refactors the OpenSSLSessionCache out of the Factory. Instead of
directly injecting a pointer to the factory to each OpenSSLAdapter instead just
a pointer to the OpenSSLSessionCache is submitted which the Factory is the sole
owner of. This provides a cleaner dependency injection interface and allows the
OpenSSLSessionCache to be tested independently of the factory that uses it. It
also allows for the factories role to be more clearly defined allowing for
additional dependency injection in future updates.
This change also removes the habit of having OpenSSL typedefs around certain
functions and instead uses the standardised ossl_typ.h header which contains
these typedefs. This makes the headers more directly tied to just what they are
responsible for doing.
Bug: webrtc:9085
Change-Id: I7938178b70acc613856139d387a1b46928dca6ad
Reviewed-on: https://webrtc-review.googlesource.com/66941
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22758}
This profile will now not be used unless the application explicitly
sets the flag in CryptoOptions to true. As a result, an 80-bit
authentication tag will be used instead of a 32-bit one. See bug for
more details.
Bug: webrtc:7670
Change-Id: I7c0a118fd7b1e7aac23b9eb8717099f055de0441
Reviewed-on: https://webrtc-review.googlesource.com/66600
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22757}
This changeset addresses concerns about how the OpenSSLAdapter does certificate
name matching. The current approach has a number of issues which are outlined
in the bug description. The approach taken in this changeset is to use the
standard function X509_check_host which should correctly parse the wildcard
expansions and is directly supported in OpenSSL instead of attempting my own
implementation. This changeset uses this as an opportunity to add additional
parameter checking and refactoring logging code out of the main code path.
Bug: webrtc:8888
Change-Id: Iaffe1daddcd52193ba674489f613ce8515b81e91
Reviewed-on: https://webrtc-review.googlesource.com/65022
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Emad Omara <emadomara@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22755}
With the latest usrsctp roll, the MTU value you provide is the space
avaiable for chunks in the packet. We previously specified this to be the
MTU for the entire SCTP packet, so we were logging errors when the SCTP
packets were 12 bytes larger than expected (the size of the SCTP header).
This fix updates our MTU specified to account for the SCTP header size
as well.
Bug: webrtc:9082
Change-Id: Id3bfa839d4e7662230111ebbdf33bd81ccdc7cf4
Reviewed-on: https://webrtc-review.googlesource.com/66943
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22754}
|is_posix| will be switched to false for Fuchsia, this is a preliminary change.
Bug: chromium:812974
Change-Id: I3bfda3e056ad1e5229834286ce5d095d9204a428
Reviewed-on: https://webrtc-review.googlesource.com/65782
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Fabrice de Gans-Riberi <fdegans@chromium.org>
Cr-Commit-Position: refs/heads/master@{#22753}
This CL implements the functions related to decoding.
Bug: webrtc:8909
Change-Id: Iefa3c1565a9b9ae93f14992b4a1cca141b7c5193
Reviewed-on: https://webrtc-review.googlesource.com/66403
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22747}
Implement debugQuickLookObject for RTCI420Buffers and RTCCVPixelBuffers.
Also draw gradients consistently regardless of endianness in the unit
tests for RTCCVPixelBuffers and ObjCVideoTrackSource.
Bug: webrtc:9007
Change-Id: Ia5a3d0905a763efc190165471983061fc07551f2
Reviewed-on: https://webrtc-review.googlesource.com/64987
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22746}
The settings struct specifies bitrate in kbps, but we are
treating it as bps.
Bug: webrtc:9113
Change-Id: I27745da93aaec68041ea4283b45eccb35d820793
Reviewed-on: https://webrtc-review.googlesource.com/66960
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22743}
AGC2 component that computes and applies the digital gain.
The gain is computed from an estimated speech and noise level.
This component decides how fast the gain can change and what it
should be.
Bug: webrtc:7494
Change-Id: If55b6e5c765f958e433730cd9e3b2b93c14a7910
Reviewed-on: https://webrtc-review.googlesource.com/64985
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22741}
We put back the old noise estimator from LevelController. We add a few
new unit tests. We also re-arrange the code so that it fits with how
it is used in AGC2. The differences are:
1. The NoiseLevelEstimator is now fully self-contained.
2. The NoiseLevelEstimator is responsible for calling SignalClassifier
and computing the signal energy. Previously the signal type and
energy were used in several places. It made sense to compute the
values independently of the noise calculation.
3. Re-initialization doesn't have to be done by the caller.
4. The interface is AudioFrameView instead of AudioBuffer.
# Bots are green, nothing should break internal stuff
NOTRY=True
Bug: webrtc:7494
Change-Id: I442bdbbeb3796eb2518e96000aec9dc5a039ae6d
Reviewed-on: https://webrtc-review.googlesource.com/66380
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22738}
Change the arrays to be continuous uint8_t arrays instead
being of arrays of arrays (of arrays).
Code size difference is 17K for arm, ~42K for arm64.
New lookup algorithm, tailored for these two tables + tests.
Instead of returning a raw pointer into the table, the algorithm
returns an ArrayView, which includes size information for how much
memory can be read.
Change-Id: I000b094520bac944e518eb8b51d8dbef4670f5d7
Bug: webrtc:9102
Reviewed-on: https://webrtc-review.googlesource.com/65920
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22736}
This ensures that the callback will be called if total max bit rate
changes even if min bitrate or padding bitrate has not changed.
Bug: None
Change-Id: I616e95b1f9f5a30733f1d0acb86e18c93001d3db
Reviewed-on: https://webrtc-review.googlesource.com/63642
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22734}
The original cl broke some downstream project because some internal source
encoders do not call OnBitrateChanged on GenericEncoder.
Bug: webrtc:9058
Change-Id: I7841c65059fb4fc9e1ab9754bb1d232ce660a990
Reviewed-on: https://webrtc-review.googlesource.com/66342
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22733}
Another submodule of the Automatic Gain Controller 2. It refines the
biased estimate of the Adaptive Mode Level Estimator. It works by
generating a delayed stream of peak levels. The delayed peaks are
compared to the level estimate.
Bug: webrtc:7494
Change-Id: If4c2c19088d1ca73fb93511dad4e1c8ccabcaf03
Reviewed-on: https://webrtc-review.googlesource.com/65461
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22732}
This parameter is being removed from the C++ API, remove it from the
ObjC API also. It was never used for anything by the H264 decoder.
Bug: webrtc:9107
Change-Id: I5222eac932a4e7d4129d803f8126b5e8d0b027b6
Reviewed-on: https://webrtc-review.googlesource.com/66740
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22730}
This CL splits out the audio device module Java code into a separate
target, and also splits up the audio device module implementations into
three different build targets, one for OpenSLES, AAudio, and the Java
based implementation.
Bug: webrtc:7452, webrtc:9048
Change-Id: I8ec09c73580b468837223ddd420fb29ca61fdea5
Reviewed-on: https://webrtc-review.googlesource.com/66461
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22727}
In order to handle per-layer frame dropping both VP9 encoder wrapper
and RTP packetizer were modified.
- Encoder wrapper buffers last encoded frame and passes it to
packetizer after frame of next layer is encoded or encoding of
superframe is finished.
- Encoder wrapper sets end_of_superframe flag on last encoded frame of
superframe before passing it to packetizer.
- If end_of_superframe is True then packetizer sets marker bit on last
packet of frame.
Bug: webrtc:9066
Change-Id: I1d45319fbe6bc63d01721ea67bfb7440d4c29275
Reviewed-on: https://webrtc-review.googlesource.com/65540
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22722}
Measure time spent in frame encode callback, accumulate it for layers
and subtract it from measured encode time of next layer frame.
Bug: none
Change-Id: Ifc3baae2f9a49913a55a7de2de9507102edd0295
Reviewed-on: https://webrtc-review.googlesource.com/65981
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22720}
This CL removes PushCaptureData(), which is unused.
The reason I'm removing it is since this method is cauing chromium-style
violations for all files that includes
modules/audio_device/include/audio_device_defines.h, and it's annoying
to suppress it everywhere.
Bug: webrtc:8659
Change-Id: I9133d05259075d8e8ec89b764be934f37b5fa77e
Reviewed-on: https://webrtc-review.googlesource.com/66404
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22717}
Fixes target bitrate calculation for framerate based adjuster. Adds new
API to bitrate adjuster - getCodecConfigFramerate() - that returns the
FPS that should be passed to MediaCodec on initialization.
Bug: b/73741487, cl/186656928
Change-Id: Ia4a5e99d302de67fbee0c132ab8e9392bc205b44
Reviewed-on: https://webrtc-review.googlesource.com/65162
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22716}
To make setup_apprtc.py work on Windows
Bug: webrtc:7602
Change-Id: I17c19c1cb8b2b71dafd90ae5f8be80e50c3397e9
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/66660
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22715}
