Commit Graph

340 Commits

Author SHA1 Message Date
3f2634eadc Reland "Keep spatial_idx=kNoSpatialIdx(255) if there is no layer indices."
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}
2017-11-07 16:34:20 +00:00
7cfbf3a0ff Make energy calculation in AGC not overflow.
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}
2017-11-07 15:37:55 +00:00
fbb3b7d004 Reland: "Make javac warnings errors for WebRTC targets."
This reverts commit 2bad72a27329ff30ceb9479253f5eb3d21888d25.

Reason for revert: Fixing downstream projects (take 2).

Original change's description:
> Reland "Revert "Make javac warnings errors for WebRTC targets.""
> 
> This is a reland of 098d24c3c18f4b1fd043d7ba716d7601f0ce2b74
> Original change's description:
> > Revert "Make javac warnings errors for WebRTC targets."
> > 
> > This reverts commit 19b761403c3522902d69d61179f4d184e3632f79.
> > 
> > Reason for revert: Breaking internal builds
> > 
> > Original change's description:
> > > Make javac warnings errors for WebRTC targets.
> > > 
> > > Adds new rtc_* templates for Android targets to allow specifying
> > > default values that affect WebRTC targets.
> > > 
> > > Bug: webrtc:6597
> > > Change-Id: Ie529bfc8500d1e785b8a59dba7078b5f88ccfcd1
> > > Reviewed-on: https://webrtc-review.googlesource.com/15103
> > > Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> > > Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#20567}
> > 
> TBR=phoglund@webrtc.org,sakal@webrtc.org
> > 
> > Change-Id: I6d3ff5604b3d4307765d3a65adb783f89fcc974c
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: webrtc:6597
> > Reviewed-on: https://webrtc-review.googlesource.com/20740
> > Reviewed-by: Lu Liu <lliuu@webrtc.org>
> > Commit-Queue: Lu Liu <lliuu@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#20571}
> 
> Bug: webrtc:6597
> Change-Id: Icfb5ded46ce76b674bae67bfa02054b4ec52bb0f
> Reviewed-on: https://webrtc-review.googlesource.com/20800
> Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20577}

TBR=phoglund@webrtc.org,ehmaldonado@webrtc.org,mbonadei@webrtc.org,sakal@webrtc.org,lliuu@webrtc.org

Change-Id: Id3713c1885318741711987ae642a269a9ca5bb85
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:6597
Reviewed-on: https://webrtc-review.googlesource.com/18441
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20588}
2017-11-07 15:36:46 +00:00
731d1cabeb Reduce flakiness of asynchronous RtcpTransceiver tests
Restructure tests to never wait for no packets,
Greatly increase wait timeout.
(Reduce expectation of synchronous primitives precision)

Bug: webrtc:8494
Change-Id: I9a80fda3a2bf527d8b7337ecabaf625e543b8c62
Reviewed-on: https://webrtc-review.googlesource.com/20502
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20584}
2017-11-07 13:08:05 +00:00
3e83b7fe8d audio_processing VAD annotations in APM-qa.
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}
2017-11-07 10:37:00 +00:00
411582b13a [Window Capturer] Implement scaling in GetWindowBounds()
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}
2017-11-07 01:49:35 +00:00
2bad72a273 Reland "Revert "Make javac warnings errors for WebRTC targets.""
This is a reland of 098d24c3c18f4b1fd043d7ba716d7601f0ce2b74
Original change's description:
> Revert "Make javac warnings errors for WebRTC targets."
> 
> This reverts commit 19b761403c3522902d69d61179f4d184e3632f79.
> 
> Reason for revert: Breaking internal builds
> 
> Original change's description:
> > Make javac warnings errors for WebRTC targets.
> > 
> > Adds new rtc_* templates for Android targets to allow specifying
> > default values that affect WebRTC targets.
> > 
> > Bug: webrtc:6597
> > Change-Id: Ie529bfc8500d1e785b8a59dba7078b5f88ccfcd1
> > Reviewed-on: https://webrtc-review.googlesource.com/15103
> > Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> > Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#20567}
> 
TBR=phoglund@webrtc.org,sakal@webrtc.org
> 
> Change-Id: I6d3ff5604b3d4307765d3a65adb783f89fcc974c
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:6597
> Reviewed-on: https://webrtc-review.googlesource.com/20740
> Reviewed-by: Lu Liu <lliuu@webrtc.org>
> Commit-Queue: Lu Liu <lliuu@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20571}

Bug: webrtc:6597
Change-Id: Icfb5ded46ce76b674bae67bfa02054b4ec52bb0f
Reviewed-on: https://webrtc-review.googlesource.com/20800
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20577}
2017-11-07 01:31:45 +00:00
a1a475a5b6 Revert "Revert "Make javac warnings errors for WebRTC targets.""
This reverts commit 098d24c3c18f4b1fd043d7ba716d7601f0ce2b74.

Reason for revert: Fixing downstream projects.

Original change's description:
> Revert "Make javac warnings errors for WebRTC targets."
> 
> This reverts commit 19b761403c3522902d69d61179f4d184e3632f79.
> 
> Reason for revert: Breaking internal builds
> 
> Original change's description:
> > Make javac warnings errors for WebRTC targets.
> > 
> > Adds new rtc_* templates for Android targets to allow specifying
> > default values that affect WebRTC targets.
> > 
> > Bug: webrtc:6597
> > Change-Id: Ie529bfc8500d1e785b8a59dba7078b5f88ccfcd1
> > Reviewed-on: https://webrtc-review.googlesource.com/15103
> > Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> > Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#20567}
> 
> TBR=phoglund@webrtc.org,sakal@webrtc.org
> 
> Change-Id: I6d3ff5604b3d4307765d3a65adb783f89fcc974c
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:6597
> Reviewed-on: https://webrtc-review.googlesource.com/20740
> Reviewed-by: Lu Liu <lliuu@webrtc.org>
> Commit-Queue: Lu Liu <lliuu@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20571}

TBR=phoglund@webrtc.org,sakal@webrtc.org,lliuu@webrtc.org

Change-Id: I3f0289c6ddc1930b1c92f653a61eff3f6a2bba30
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:6597
Reviewed-on: https://webrtc-review.googlesource.com/20741
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20572}
2017-11-06 20:30:58 +00:00
098d24c3c1 Revert "Make javac warnings errors for WebRTC targets."
This reverts commit 19b761403c3522902d69d61179f4d184e3632f79.

Reason for revert: Breaking internal builds

Original change's description:
> Make javac warnings errors for WebRTC targets.
> 
> Adds new rtc_* templates for Android targets to allow specifying
> default values that affect WebRTC targets.
> 
> Bug: webrtc:6597
> Change-Id: Ie529bfc8500d1e785b8a59dba7078b5f88ccfcd1
> Reviewed-on: https://webrtc-review.googlesource.com/15103
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20567}

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

Change-Id: I6d3ff5604b3d4307765d3a65adb783f89fcc974c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:6597
Reviewed-on: https://webrtc-review.googlesource.com/20740
Reviewed-by: Lu Liu <lliuu@webrtc.org>
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20571}
2017-11-06 19:58:38 +00:00
12251b6386 Adding @SuppressWarnings(NoSynchronizedMethodCheck).
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}
2017-11-06 17:48:38 +00:00
19b761403c Make javac warnings errors for WebRTC targets.
Adds new rtc_* templates for Android targets to allow specifying
default values that affect WebRTC targets.

Bug: webrtc:6597
Change-Id: Ie529bfc8500d1e785b8a59dba7078b5f88ccfcd1
Reviewed-on: https://webrtc-review.googlesource.com/15103
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20567}
2017-11-06 15:59:06 +00:00
ae29428489 Revert "Keep spatial_idx=kNoSpatialIdx(255) if there is no layer indices."
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}
2017-11-06 15:27:48 +00:00
fd6c0914c5 Delete deprecated constructor of SendSideCongestionController.
Move packet_router #include to where it's needed, and delete unused
MockPacketRouter.

Bug: webrtc:6847
Change-Id: I03c86c6fb8b413f5a535a237fa1724cc10960ffa
Reviewed-on: https://webrtc-review.googlesource.com/17320
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20564}
2017-11-06 15:02:36 +00:00
e3a4da9f44 AGC: Change default clipping level min to 70
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}
2017-11-06 14:16:06 +00:00
47836b4ebb 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}
2017-11-06 12:15:16 +00:00
88f080ae9a Move SPS/PPS/IDR requirement from RtpFrameObject to PacketBuffer.
BUG=webrtc:8423

Change-Id: I0f0d59461afead700c20c9a2ed9b2bc991590b4a
Reviewed-on: https://webrtc-review.googlesource.com/15101
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20559}
2017-11-06 12:04:46 +00:00
a750333372 Remove support for SW H264 High profile decoding
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}
2017-11-02 21:22:49 +00:00
180362842a NetEq: Fix a problem with too large delay during codec-internal DTX/CNG
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}
2017-11-02 13:09:07 +00:00
eb254b40b3 Don't select audio codecs depending on GN vars build_with_{chromium|mozilla}
BUG=webrtc:8343

Change-Id: I5943006a4da17f72eb88eae9d7ea57574d54f680
Reviewed-on: https://webrtc-review.googlesource.com/9401
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20540}
2017-11-01 18:59:27 +00:00
8686077102 [Window Capturer] Inaccurate cursor position on cinnamon
When cinnamon is used, it always wraps the application window with its own
window. Instead of (0, 0), the DesktopRect from XWindowAttributes starts from
(10, 36).
So this change considers this difference when translating the DesktopRect in
GetWindowRect() function.

Bug: chromium:778035
Change-Id: I4944b2d1e13a4c379e114fd1749d74e95a63003b
Reviewed-on: https://webrtc-review.googlesource.com/17660
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Zijie He <zijiehe@chromium.org>
Cr-Commit-Position: refs/heads/master@{#20538}
2017-11-01 18:17:07 +00:00
f0cc814343 Support writing network timestamp delta fields into VideoTimingExtension
Bug: None
Change-Id: I17b9ba0eb8095cfd8e6bc5bf97b2949d5d3edd24
Reviewed-on: https://webrtc-review.googlesource.com/17500
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20524}
2017-11-01 10:15:56 +00:00
96acb43b2a Fix Chromium compile of StereoEncoderAdapter.
WebRTC rolls into Chromium are failing, we should fix it ASAP.

Log:
FAILED:
obj/third_party/webrtc/modules/video_coding/webrtc_stereo/stereo_encoder_adapter.obj
ninja -t msvc -e environment.x64 -- E:\b\c\goma_client/gomacc.exe
"e:\b\c\win_toolchain\vs_files\88c3b62e1eb0893b8cd57e3f4859c3af27907f64\vc\tools\msvc\14.11.25503\bin\hostx64\x64/cl.exe"
/nologo /showIncludes
@obj/third_party/webrtc/modules/video_coding/webrtc_stereo/stereo_encoder_adapter.obj.rsp
/c
../../third_party/webrtc/modules/video_coding/codecs/stereo/stereo_encoder_adapter.cc
/Foobj/third_party/webrtc/modules/video_coding/webrtc_stereo/stereo_encoder_adapter.obj
/Fd"obj/third_party/webrtc/modules/video_coding/webrtc_stereo_cc.pdb"
../../third_party/webrtc/modules/video_coding/codecs/stereo/stereo_encoder_adapter.cc(134):
error C2220: warning treated as error - no 'object' file generated
../../third_party/webrtc/modules/video_coding/codecs/stereo/stereo_encoder_adapter.cc(134):
warning C4267: 'argument': conversion from 'size_t' to 'uint32_t',
possible loss of data

Bug: chromium:780411
Change-Id: Ia80f4551d0efeebc6d084e951f5c25e8b9401250
Reviewed-on: https://webrtc-review.googlesource.com/17781
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Max Morin <maxmorin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20522}
2017-11-01 09:53:16 +00:00
13a8f201e4 Add CHECKs to FlexfecReceiver.
There is a crash happening in this neighbourhood, so adding
CHECKs to tease it out explicitly.

BUG=webrtc:8481

Change-Id: I79a2ec8fd838f4a4735a04496e363b72975919ec
Reviewed-on: https://webrtc-review.googlesource.com/17361
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20520}
2017-11-01 08:53:36 +00:00
ae3981a998 Removes experimental sleep in ADM initialization for Android
Bug: b/63010674
Change-Id: I744fa9be1031784431685a90f5c36d4a37e6a989
Reviewed-on: https://webrtc-review.googlesource.com/17441
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20518}
2017-11-01 08:09:56 +00:00
9d4af0130e New PacedSender constructor with injected PacketQueue
Intended to enable unit testing of the pacer with a mock PacketQueue.

Bug: webrtc:8422
Change-Id: I142386b2d91ad0d5ba8f3f9d876e67972c490de4
Reviewed-on: https://webrtc-review.googlesource.com/17300
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20498}
2017-10-31 11:39:22 +00:00
996eb9e353 Fix typo in VideoSendTiming header extension structure
Bug: None
Change-Id: Ic6c5613bea1fad3ac7456a691eb8e87efb6eeb2c
Reviewed-on: https://webrtc-review.googlesource.com/16980
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20497}
2017-10-31 11:20:22 +00:00
dbcac7fefe Add StereoCodecAdapter classes
This CL is the step 1 for adding alpha channel support over the wire in webrtc.
- Add the footprint for adapter classes that wraps actual codecs.
- This CL does not add a webrtc::VideoFrame container that can carry alpha to 
make the CL shorter for an easier review. Therefore, it exercises a code path
for when we receive no alpha input, just regular I420 frames.
- Unittest sends a video frame for encode/decode through these adapters and 
checks the output PSNR.
- See https://webrtc-review.googlesource.com/c/src/+/7800 for the experimental 
CL that gives an idea about how it will come together.
Design Doc: https://goo.gl/sFeSUT

Bug: webrtc:7671
Change-Id: I9d3be13647a0a958feceb8d7a9aa93852fc6a1fa
Reviewed-on: https://webrtc-review.googlesource.com/11841
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20490}
2017-10-31 06:39:52 +00:00
8c8d49ea0f Add periodic compound packet sending to RtcpTransceiver
Bug: webrtc:8239
Change-Id: I1511db63a15e8c5101a933e55e66d3877ff963be
Reviewed-on: https://webrtc-review.googlesource.com/15440
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20480}
2017-10-30 16:51:29 +00:00
bde473e4fa Fix/suppress new warnings introduced in Chromium roll.
TBR=henrika@webrtc.org

Bug: webrtc:6597
Change-Id: Id26945a7be05250673b58de8220f78bc62886688
Reviewed-on: https://webrtc-review.googlesource.com/16860
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20477}
2017-10-30 16:10:29 +00:00
90397d9aa9 Remove use of RTPFragmentationHeader from RTPSenderAudio
The RTPFragmentationHeader was used when sending audio using RED
for loss protection. This feature has been deprecated and
gradually removed. This cl removes remnants of support from
the RTP send path.

Bug: webrtc:6471
Change-Id: Ia1249047b09c16f79498827f74c2ce07aa38b8f7
Reviewed-on: https://webrtc-review.googlesource.com/16427
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20473}
2017-10-30 09:56:19 +00:00
c7b18fef19 Shifted value doesn't fit in 'int32_t'.
This CL replaces one 'int32_t' with 'uint32_t'. The value is a
non-negative energy, and the number of leading zeros is
computed. During computation, a shift can cause it to overflow.

Issue was found by the Audio Processing fuzzer.

Bug: chromium:778939, chromium:778921, chromium:778919
Change-Id: I3d7e0b547e6b0edcd9995903517ea851142a08c1
Reviewed-on: https://webrtc-review.googlesource.com/16433
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20470}
2017-10-28 10:22:32 +00:00
fb1a8661db Add support for H.264 constrained high profile in VideoProcessor.
BUG=webrtc:8448

Change-Id: I968d6cd78dd4f3c19a7944ae4cc73c5eddb9a949
Reviewed-on: https://webrtc-review.googlesource.com/16160
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20466}
2017-10-27 13:30:34 +00:00
5d7fd19c20 Don't build windows core audio if using dummy file devices.
If WEBRTC_DUMMY_FILE_DEVICES is set, WEBRTC_CORE_AUDIO_BUILD should not.
Otherwise audio_device_core_win.h will be included [1] when it shouldn't
(according to [2]).

[1] https://webrtc.googlesource.com/src/+/master/modules/audio_device/audio_device_impl.cc#22
[2] https://webrtc.googlesource.com/src/+/master/modules/audio_device/BUILD.gn#177

Bug: webrtc:6265
Change-Id: Ia6ccb9dda39f411c0d8a548a0501408e87d11a40
Reviewed-on: https://webrtc-review.googlesource.com/16430
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20464}
2017-10-27 12:53:34 +00:00
b0576ecc71 Reland of Improves native Android audio implementations
Second attempt to land https://webrtc-review.googlesource.com/c/src/+/15481.
This time with an extra (dummy) interface to ensure that we don't
break downstream clients.

Improves native Android audio implementations.

Bug: webrtc:8453
Change-Id: I659a3013ae523a2588e4c41ca44b7d0d2d65efb7
Reviewed-on: https://webrtc-review.googlesource.com/16425
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20462}
2017-10-27 10:53:20 +00:00
e4c6915b87 Remove verbose setting and reorder some print statements in VideoProcessor.
Always enabling verbose mode means about 100% more text is printed,
but this should not be a problem as the only time that we explicitly
look at the logs is when the bots are failing, or when we want to save
all output for plotting.

BUG=webrtc:8448

Change-Id: Ia5feab5220d047440d15cddb7d3fbca1c5a4aaf5
Reviewed-on: https://webrtc-review.googlesource.com/16140
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20461}
2017-10-27 10:52:14 +00:00
34fa295d47 Delete unused VP8 packetization modes.
Always use the packetization formely known as kEqualSize.
The RTPFragementation header is ignored, which is no change 
in behaviour, since the caller previously always passed null.

Bug: webrtc:6471
Change-Id: Id9e2f985280c2ee8cc33fcf0e5c1fc3ee61c1aff
Reviewed-on: https://webrtc-review.googlesource.com/15222
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20459}
2017-10-27 09:18:17 +00:00
e87cfe2315 Remove unused method PacketLossModeToStr.
Add method FrameType for frame to TestConfig.

Bug: none
Change-Id: Icfeb12fcb961559c9b36a3aedb081a840b9d8556
Reviewed-on: https://webrtc-review.googlesource.com/16120
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20458}
2017-10-27 08:51:27 +00:00
0af34ad3fa The onWebRtcAudioTrackStartError is changed in this CL which breaks the internal projects.
Revert "Improves native Android audio implementations."

This reverts commit 92b1ffd0f655e88532cb7313707f300fec911b46.

Reason for revert: <INSERT REASONING HERE>

Original change's description:
> Improves native Android audio implementations.
> 
> Summary:
> 
> Adds AudioTrackStartErrorCode to separate different types of error
> codes in combination with StartPlayout.
> 
> Harmonizes WebRtcAudioRecord and WebRtcAudioTrack implementations
> to ensure that init/start/stop is performed identically.
> 
> Adds thread checking in WebRtcAudio track.
> 
> Bug: webrtc:8453
> Change-Id: Ic913e888ff9493c9cc748a7b4dae43eb6b37fa85
> Reviewed-on: https://webrtc-review.googlesource.com/15481
> Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20448}

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

Change-Id: If1d1d9717387a4a8f6d9d6acf7e86ded4c655b5e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8453
Reviewed-on: https://webrtc-review.googlesource.com/16321
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20452}
2017-10-26 21:58:39 +00:00
c2a0eb2699 [Window Capture] Mouse cursor missing during window sharing on Mac OSX
CGWindowID is 32-bit, WindowId is 64-bit, using WindowId to receive int value
from CFNumberGetValue() causes the top 32 bits to be random. WindowFinderMac is
impacted by this issue and returns a random number. WindowCapturerMac cannot
match the window_id_ with the the random number.

Meanwhile MouseCursorMonitorMac uses window title to match "Dock" window. See,
https://cs.chromium.org/chromium/src/third_party/webrtc/modules/desktop_capture/mouse_cursor_monitor_mac.mm?rcl=a194e58e799ccab6c999998e5d0f75725aa3f748&l=174

This logic should not be necessary on 10.12 or upper, the name of dock window
is not "Dock" anymore. But to ensure the consistency on old platforms, I have
also added this logic back into GetWindowList() function.

Bug: chromium:778049
Change-Id: Ie827bcd5d31f2ca69ff24c24cf640cb7cc50d419
Reviewed-on: https://webrtc-review.googlesource.com/15782
Commit-Queue: Zijie He <zijiehe@chromium.org>
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/master@{#20451}
2017-10-26 21:14:57 +00:00
92b1ffd0f6 Improves native Android audio implementations.
Summary:

Adds AudioTrackStartErrorCode to separate different types of error
codes in combination with StartPlayout.

Harmonizes WebRtcAudioRecord and WebRtcAudioTrack implementations
to ensure that init/start/stop is performed identically.

Adds thread checking in WebRtcAudio track.

Bug: webrtc:8453
Change-Id: Ic913e888ff9493c9cc748a7b4dae43eb6b37fa85
Reviewed-on: https://webrtc-review.googlesource.com/15481
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20448}
2017-10-26 13:45:36 +00:00
78161ca59d Add sending sdes to RtcpTransceiver.
Bug: webrtc:8239
Change-Id: Icff1528e177e0bb39dd82bd4f8533e1ed2736c40
Reviewed-on: https://webrtc-review.googlesource.com/15540
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20447}
2017-10-26 13:33:16 +00:00
a0565999db Delete VCMSendStatisticsCallback and corresponding use of ProcessThread
Bug: webrtc:8422
Change-Id: I5863266a0226d475c4fdd810f2f6f1acdf922df3
Reviewed-on: https://webrtc-review.googlesource.com/14880
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20440}
2017-10-26 08:13:55 +00:00
fb6d32602c Delete unused PredictivePacketManipulator.
BUG=webrtc:8448

Change-Id: I07ff9db5cb49f84d98b6076e748a990aa560b5b5
Reviewed-on: https://webrtc-review.googlesource.com/15400
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20437}
2017-10-26 07:17:24 +00:00
f7eea2a6bc Change key frame mismatch threshold for VP9 in unit test.
Change the threshold in ProcessNoLossChangeBitRateVP9.

Bug: webrtc:8442
Change-Id: Ic924a60f60c57cc2c990430cb6c70fdbefec97f4
Reviewed-on: https://webrtc-review.googlesource.com/15840
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20436}
2017-10-26 07:01:15 +00:00
74e72c8c9b Lowering the threshold for delay change detection in AEC3
This CL lowers the threshold for delay change detection in AEC3.
This makes the delay decisions more stable.

TBR=gustaf@webrtc.org

Bug: chromium:778396,webrtc:8451
Change-Id: I8b015455399d696172b7c0beb033caf508f426e9
Reviewed-on: https://webrtc-review.googlesource.com/15541
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20433}
2017-10-25 21:56:30 +00:00
173fd91b56 [Window Capture] Inaccurate cursor position during window sharing on X11
{root_x, root_y} should be used to report the absolute cursor position in
MouseCursorMonitorX11.

Bug: chromium:778035
Change-Id: I421005d52786a57da8e8c3901bdf4afa2843ff24
Reviewed-on: https://webrtc-review.googlesource.com/15680
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Zijie He <zijiehe@chromium.org>
Cr-Commit-Position: refs/heads/master@{#20432}
2017-10-25 20:41:29 +00:00
84634b8634 Temporarily disabled failing death test.
Some death tests for AEC3 cause memory leaks on trybots. This CL
temporarily disables BlockProcessor.VerifyRenderBlockSizeCheck.

Bug: webrtc:8449,webrtc:6985
Change-Id: I2900a73f7c7d5bf0e8b58a20f9a40bd5d654629a
Reviewed-on: https://webrtc-review.googlesource.com/15500
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20431}
2017-10-25 15:24:46 +00:00
b9f536167c Removing undefined left shifts in AudioProcessing
This CL replaces 5 left shifts where the shifted value may be 
negative. The shifts are replaced with equivalent multiplications.

Bug: chromium:777231, chromium:776719, chromium:776624, chromium:776286
Change-Id: Ifb27d5506eac779e60f238432bdf9e4bc5b2da4c
Reviewed-on: https://webrtc-review.googlesource.com/14800
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20430}
2017-10-25 13:35:36 +00:00
a194e58e79 Move sequence_number_utils.h to rtc_base/
Bug: webrtc:8440
Change-Id: I36e70da6ce70b95db7d3fce8b0013bff5c795bfc
Reviewed-on: https://webrtc-review.googlesource.com/14860
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20429}
2017-10-25 12:33:57 +00:00
47d3a0197f Reenable some supressed warnings for the objc SDK.
Bug: webrtc:8441
Change-Id: I6b427dfc1fe275e274d042766e0850628cf19994
Reviewed-on: https://webrtc-review.googlesource.com/15000
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20425}
2017-10-25 11:17:36 +00:00