This is a reland of 47836b4ebb8a5c71695b5ec07bffd5ee4e3bc2ff
Internal tests are synced with the fix.
Original change's description:
> Keep spatial_idx=kNoSpatialIdx(255) if there is no layer indices.
>
> spatial_idx is not present in RTP header if there is no temporal or
> spatial layering. But the parser sets spatial_idx to 0 in this case.
> When reflector repacketizes such packets it writes layering indices
> into outgoing packets. When packets arrive to receiver it thinks that
> it deals with multi layer stream and passes it through special path
> in Vp9 reference frame finder which never outputs inter frames.
>
> I modified the parser such that it keeps spatial_idx=kNoSpatialIdx(255)
> when there is no layer indices in RTP header. Related unit tests have
> been modified as well.
>
> Bug: none
> Change-Id: I14498cafb4e57797577dc873298c35b243479f88
> Reviewed-on: https://webrtc-review.googlesource.com/17980
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20560}
TBR=brandtr@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org
Bug: none
Change-Id: I6087a8b20a926296b30432d69251670120b2a20c
Reviewed-on: https://webrtc-review.googlesource.com/20940
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20591}
An energy value is calculated by summing squares of processed audio
samples. The expression 'out*out >> 6' could overflow. In this CL we
change it to 'out*(out>>6) + out*(out*(out%(1<<6))>>6)'.
The which is verified and proven to be equal, but doesn't
overflow. The change also passes our change-detection tests in
GainControlBitExactnessTest.*
We verified with Godbolt that the modulo and divisions are converted
into branch-free bitwise operations.
NOTRY=True # changing comment, tests just passed.
Bug: chromium:780638, chromium:780376
Change-Id: I415535193433a2fbc275c643fb4e4026ba3e0bdd
Reviewed-on: https://webrtc-review.googlesource.com/20867
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20589}
The first example CL for generating JNI code
(https://webrtc-review.googlesource.com/c/src/+/4500) seems to stick, so
this CL updates the rest of the VideoEncoder. The JNI code for
Java -> C++ is still done manually.
This CL puts the necessary helper Java methods in a class called
VideoEncoderWrapper.
Bug: webrtc:8278
Change-Id: Ic3a6defe59c094f67ffd8ea86d6c272c676980ae
Reviewed-on: https://webrtc-review.googlesource.com/20871
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20587}
This CL adds a replacement for JNIEnv::FindClass that works from any
thread, i.e. from native C++ threads as well. This function will be used
from the generated JNI code. Long term, we should stop using
classreferenceholder that relies on a hardcoded list of WebRTC classes.
Bug: webrtc:8278
Change-Id: I4f40c744325ac02b73bd8fa479ab50b684429dc2
Reviewed-on: https://webrtc-review.googlesource.com/20223
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20583}
It had a large-ish number of non-standard-formatted lines. This caused
a problem in another CL where I try to move this file: `git cl format`
wants to reformat all these lines, which in turn causes the rename
detection to treat it as a delete+add, making reviewing hard.
Better to just reformat it in a separate CL and get it over with.
BUG=none
Change-Id: I619f0454546e8a55fba58da08073da9bb1d06207
Reviewed-on: https://webrtc-review.googlesource.com/20865
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20582}
Added possibility to extract audio_processing VAD annotations in the Quality Assessment tool.
Annotations are extracted into compressed Numpy 'annotations.npz' files.
Annotations contain information about VAD, speech level, speech probabilities etc.
TBR=alessiob@webrtc.org
Bug: webrtc:7494
Change-Id: I0e54bb67132ae4e180f89959b8bca3ea7f259458
Reviewed-on: https://webrtc-review.googlesource.com/17840
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20581}
On Mac OSX system, if retina screen is used, the GetWindowBounds() returns
pre-scaled values instead of system coordinates. So this fix considers
per-monitor scale-factor, and stretchs the DesktopRect.
Bug: chromium:778049
Change-Id: I9dc51e08235eba9b3ef6378eaa15737aa444b0c8
Reviewed-on: https://webrtc-review.googlesource.com/17600
Commit-Queue: Zijie He <zijiehe@chromium.org>
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/master@{#20578}
This literally copies & pastes the code from WebRtcSession into
PeerConnection as private methods. The only other changes were to
inline the WebRtcSession construction/initialization/destruction
into PeerConnection and fix issues using rtc::Bind on the
reference-counted PeerConnection.
Bug: webrtc:8323
Change-Id: Ib3f071ac10d18566a21a3b04813b1d4ec691ef3c
Reviewed-on: https://webrtc-review.googlesource.com/15160
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20574}
This commit prepares WebRtcSession so that it can be cleanly
copy & pasted into PeerConnection in the next commit. To accomplish
this, the following was done:
1. Added a pointer to the owning PeerConnection to WebRtcSession.
2. Replace WebRtcSession state enum with signaling state.
3. All signals/observers only observed by PeerConnection were
replaced with direct calls to PeerConnection methods.
4. All duplicated fields were moved to PeerConnection.
5. The remaining tests that still use WebRtcSession for mocks were
updated to minimize dependence on WebRtcSession construction.
Bug: webrtc:8323
Change-Id: Ifc1a4ee819dcc9beca5363291012f7d5563ff7b1
Reviewed-on: https://webrtc-review.googlesource.com/9020
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20573}
EXPECT_EQ(0u, caller()->ice_connection_state_history().size());
Was failing since it cleared the connection state history before
the caller had finished transitioning to Completed, so the initial
transitions would be mistaken for later transitions.
Bug: None
Change-Id: I7043638f077ac5dcaeeca0d3ea6accc93c920364
Reviewed-on: https://webrtc-review.googlesource.com/16261
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20570}
In https://chromium-review.googlesource.com/c/chromium/src/+/750645
Chromium started to use an ErrorProne plugin to discourage synchronized
public methods (an encourage the usage of synchronized blocks).
In order to unblock the Chromium Roll we can suppress these warnings
and decide if we want to align with Chromium on this check or ask
them to make it optional.
More details in the bug.
TBR=magjed@webrtc.org
Bug: webrtc:8491
Change-Id: Ie77a324e54aab44a4f59853959549f1d21f884a0
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/20060
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20569}
Starting from https://webrtc-review.googlesource.com/c/src/+/18140
(which includes 3296e11b37)
//third_party/protobuf starts to depend on //third_party/zlib.
To fix the Chromium Roll WebRTC has add the license file of
//third_party/zlib to its generate_licenses.py script in order to add
it to markdown generated license file.
Bug: None
Change-Id: If504ef00b166fdbcbe22acb0a2721bfb55624d3e
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/18244
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20566}
This reverts commit 47836b4ebb8a5c71695b5ec07bffd5ee4e3bc2ff.
Reason for revert: This breaks internal tests, reverting to check if they recover.
Original change's description:
> Keep spatial_idx=kNoSpatialIdx(255) if there is no layer indices.
>
> spatial_idx is not present in RTP header if there is no temporal or
> spatial layering. But the parser sets spatial_idx to 0 in this case.
> When reflector repacketizes such packets it writes layering indices
> into outgoing packets. When packets arrive to receiver it thinks that
> it deals with multi layer stream and passes it through special path
> in Vp9 reference frame finder which never outputs inter frames.
>
> I modified the parser such that it keeps spatial_idx=kNoSpatialIdx(255)
> when there is no layer indices in RTP header. Related unit tests have
> been modified as well.
>
> Bug: none
> Change-Id: I14498cafb4e57797577dc873298c35b243479f88
> Reviewed-on: https://webrtc-review.googlesource.com/17980
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20560}
TBR=brandtr@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,ssilkin@webrtc.org
Change-Id: I67d083cf769974d8df8bd5d70942af97db578db9
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: none
Reviewed-on: https://webrtc-review.googlesource.com/20501
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20565}
The old value was 170, but experiments have shown that 70 is better.
This will let the AGC reduce the gain further when input clipping is
detected. The effect should be less clipping, but sometimes slightly
lower signals.
In Chrome, the value 70 has already been used since June (see
https://codereview.chromium.org/2928133002).
Bug: webrtc:6622, chromium:672476
Change-Id: Ie5a60bb875eef71f303b28e096b22a8cd4b449d4
Reviewed-on: https://webrtc-review.googlesource.com/20222
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20563}
The new SurfaceEglRenderer helper class extends EglRenderer and
implements rendering on a SurfaceView.
Bug: webrtc:8242
Change-Id: Ic532fe487755d3b54c6bd03f239d714e1ecb10ad
Reviewed-on: https://webrtc-review.googlesource.com/2940
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20562}
This can occur if there are callbacks in-flight when the compression
session is destroyed. Has been observed but is rare.
Bug: webrtc:8489
Change-Id: I5d4b35c555f6ff68af48edfcc7acf53395fa86fe
Reviewed-on: https://webrtc-review.googlesource.com/18220
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20561}
spatial_idx is not present in RTP header if there is no temporal or
spatial layering. But the parser sets spatial_idx to 0 in this case.
When reflector repacketizes such packets it writes layering indices
into outgoing packets. When packets arrive to receiver it thinks that
it deals with multi layer stream and passes it through special path
in Vp9 reference frame finder which never outputs inter frames.
I modified the parser such that it keeps spatial_idx=kNoSpatialIdx(255)
when there is no layer indices in RTP header. Related unit tests have
been modified as well.
Bug: none
Change-Id: I14498cafb4e57797577dc873298c35b243479f88
Reviewed-on: https://webrtc-review.googlesource.com/17980
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20560}
The UMA metric will log the same information that goes into the
googCurrentDelayMs stat.
Bug: webrtc:8488
Change-Id: I26abb3d86a07e8c0ddb4168540a8e2458115f004
Reviewed-on: https://webrtc-review.googlesource.com/18201
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20557}
We need to get rid of the ones that don't take audio codec factory
arguments in order to eliminate the dependency on audio codec
implementations.
BUG=webrtc:8396
Change-Id: Id0c1c3b70c2b3479da81ba1056cc69e857e454bd
Reviewed-on: https://webrtc-review.googlesource.com/12281
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20555}
Also put Baseline profile in front of Constrained Baseline profile. The
reason is that the HW encoders are mostly BP, and we want this to be the
first codec in the list so that HW is preferred by default.
The H264 tests in chromium needs to be updated again with this change,
which was changed here: https://codereview.chromium.org/2985263002/.
Bug: webrtc:8317
Change-Id: Ief75683962b79b6664143d73b9259729c66ce082
Reviewed-on: https://webrtc-review.googlesource.com/17780
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20554}
So that we don't have to be capable of creating one ourselves, which
requires a dependency on the audio decoders.
In the process, remove the default values for the VoEBase::Init() arguments,
since there isn't a sensible default value for the audio decoder factory
anymore.
BUG=webrtc:8396
Change-Id: Idb433efa49e1a68e8206d369d27b3c255185777a
Reviewed-on: https://webrtc-review.googlesource.com/18200
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20552}
The length of the generated comfort noise is measured with a
counter. A bug in the implementation caused the counter to be reset
not only when a new packet was decoded, but also when NetEq asked the
decoder for more comfort noise without giving it a new packet to
decode. This means that the counter was reset once every 20 ms (in the
case of Opus), and it would never match the gap in timestamps that is
the exit criterion for CNG. This would have resulted in perpetual CNG,
but there is a stop-gap in NetEq. If the buffer level exceeds 4 times
the target level, CNG mode is exited anyway. This is what happens at
the end of every silence period.
With this CL, the bug should be fixed. The fix is wrapped in an
experiment, to allow verifying the fix and the impact of it with real
world data.
Bug: webrtc:8488
Change-Id: Idfc24df780eb2c55dbf08de840e6644e8557a0af
Reviewed-on: https://webrtc-review.googlesource.com/18181
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20551}
Previously input buffers would be filled incorrectly for sparsely
packed buffers where stride is not equal to the plane width.
Bug: webrtc:8478
Change-Id: I080fa3c354a27982bb996be8c1e41b103384e4bc
Reviewed-on: https://webrtc-review.googlesource.com/17321
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20550}
This is a reland of ba78b5a905bffa05933a135673996df02328f2a4
Original change's description:
> Android: Generate JNI code for VideoSink and VideoEncoder
>
> This is the first CL to start generating JNI code. It has updated two of
> the most recent classes to use JNI code generation.
>
> Bug: webrtc:8278
> Change-Id: I1b19ee78c273346ceeaa0401dbdf8696803f16c7
> Reviewed-on: https://webrtc-review.googlesource.com/3820
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#19994}
Bug: webrtc:8278
Change-Id: Id3e6513736eb87d7c234be3b0d13c5d30435201c
Reviewed-on: https://webrtc-review.googlesource.com/4500
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20548}
This ensures that on service outages we get a clear early error message instead of something cryptic later down the line
Bug: None
Change-Id: Ib637ed97144284e3744aaa948f594f5795fa9c72
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/18040
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20545}
