This method is an anti-pattern. Removes usage of the method from
CameraCapturer and deletes it.
Bug: webrtc:8456
Change-Id: I8a70ce968af412fa6e6b9308a9e05d6a8a1ba05d
Reviewed-on: https://webrtc-review.googlesource.com/46140
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21808}
Part of Android SDK is in this directory.
Bug: None
Change-Id: If5d7e2625e7b1461229850d4b40b05a49066b5fc
Reviewed-on: https://webrtc-review.googlesource.com/46200
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21807}
This is a reland of bbdabe50db0cf09f6007dda12a6476dc4602b174.
Original change's description:
> Rename stereo video codec to multiplex
>
> This CL only does the rename from"stereo" to multiplex". With this we have a
> better name that doesn't clash with audio's usage of stereo.
>
> Bug: webrtc:7671
> Change-Id: Iebc3fc20839025f1bc8bcf0e16141bf9744ef652
> Reviewed-on: https://webrtc-review.googlesource.com/43242
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21769}
TBR=niklas.enbom@webrtc.org
Bug: webrtc:7671
Change-Id: I6f38dc46126f279f334d52b56339b40acdc30511
Reviewed-on: https://webrtc-review.googlesource.com/45820
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21794}
This is the time to wait after creating the server to ensure it's
listening before trying to connect to it. The previous value of 10 was
not enough; tests occasionally failed.
Bug: webrtc:8711
Change-Id: I67d592fdb9a863d574f2a33096b7050935693f4e
Reviewed-on: https://webrtc-review.googlesource.com/44521
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21793}
Use rtc::ToString instead of std::to_string.
std::to_string isn't usable in some versions of the Android NDK.
Most of the webrtc code (except test code) is using rtc::ToString(). This is the only instance that is using std::to_string()
Bug: None
Change-Id: Id8e234c3e48269dd115c6dc50867121f52cdc508
Reviewed-on: https://webrtc-review.googlesource.com/45560
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Jiawei Ou <ouj@fb.com>
Cr-Commit-Position: refs/heads/master@{#21792}
Delete the subdirectory modules/media_file, and all references to it.
Bug: none
Change-Id: I19d86420a7d1d51cb6174c914a90484918106c5a
Reviewed-on: https://webrtc-review.googlesource.com/40540
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21790}
When using VideoFrames, expect_encode_from_texture is true even for
ByteBuffer frames. This causes the encoder to sometimes initialize
itself in surface mode even when egl_context_ is not available.
This leads to a crash.
Bug: webrtc:8776
Change-Id: I8cac36514725b8f430d7bf456d481a4b0c6fcd42
Reviewed-on: https://webrtc-review.googlesource.com/43861
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21789}
Implements the dynamic permission model required by the newer SDK and
changes the theme.
Bug: webrtc:8803
Change-Id: I3ea23a25b27f196fcffd018c7cdd2ff6255b62d9
Reviewed-on: https://webrtc-review.googlesource.com/44400
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21788}
This change makes the fix for too long delays during Opus DTX periods
permanent. The fix has up until now been under an experiment, named
WebRTC-NetEqOpusDtxDelayFix.
Bug: webrtc:8488,chromium:780849
Change-Id: I006abb67f96d9d7880bf2215d7d6b52db6cbbfbc
Reviewed-on: https://webrtc-review.googlesource.com/44420
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21786}
This method returns the DTLS SSL certificate chain associated with the
audio transport on the remote side. This will become populated once the
DTLS connection with the peer has been completed.
TBR=deadbeef@webrtc.org
Bug: webrtc:8800
Change-Id: Ib90ccb3463415e798c17c187c5bdbfc4da26f11f
Reviewed-on: https://webrtc-review.googlesource.com/44140
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21785}
This implements the WebRTC specification for handling
the legacy offer options offer_to_receive_audio and
offer_to_receive_video. They are not implemented for CreateAnswer.
With Unified Plan semantics, clients should switch to the
RtpTransceiver API for ensuring the correct media sections are
offered.
Bug: webrtc:7600
Change-Id: I6ced00b86b165a352bd0ca3d64b48fadcfd12235
Reviewed-on: https://webrtc-review.googlesource.com/41341
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21784}
This reverts commit 4954a77cf81e6793245f52d485834acd3e6eab1c.
Reason for revert: Breaks downstream build which was depending on the name "kVideoCodecStereo". Will need to do some sort of trickery to make this change without breaking the relevant code. Sorry. :(
Original change's description:
> Reland "Rename stereo video codec to multiplex"
>
> This is a reland of bbdabe50db0cf09f6007dda12a6476dc4602b174.
> This was reverted because of breaking internal build. I contacted sheriff
> and looked at logs but cannot find anything related to this CL. This was landed
> with #3850 build which caused exception, but 3847-3855 seem to all have failed.
> I am relanding to see if it will work this time or it will give some related
> error message that can guide me.
>
> Original change's description:
> > Rename stereo video codec to multiplex
> >
> > This CL only does the rename from"stereo" to multiplex". With this we have a
> > better name that doesn't clash with audio's usage of stereo.
> >
> > Bug: webrtc:7671
> > Change-Id: Iebc3fc20839025f1bc8bcf0e16141bf9744ef652
> > Reviewed-on: https://webrtc-review.googlesource.com/43242
> > Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> > Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21769}
>
> TBR=niklas.enbom@webrtc.org
>
> Bug: webrtc:7671
> Change-Id: I5934abad1ce28acf02842ea8ee2af7768a826eb8
> Reviewed-on: https://webrtc-review.googlesource.com/44520
> Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21780}
TBR=sprang@webrtc.org,niklas.enbom@webrtc.org,qiangchen@chromium.org,emircan@webrtc.org
Change-Id: I0a71327c2ddfdd030b1e058cd6a41b1689836719
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7671
Reviewed-on: https://webrtc-review.googlesource.com/44621
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21783}
These tests verify the behavior between Plan B and
Unified Plan PeerConnections.
Bug: webrtc:7600
Change-Id: Ic41a0e692d32cde6fe7719ada2dbffd4281c008c
Reviewed-on: https://webrtc-review.googlesource.com/43244
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21782}
This reverts commit 65c0a60302202189c37af91fca6abf092f022b1c.
Reason for revert: Breaking downstream test which was calling CreateAnswer in stable state. Will reland after fixing test.
Original change's description:
> Parameterize PeerConnection signaling tests for Unified Plan
>
> This also changes the behavior of CreateAnswer to fail unless
> the signaling state is kHaveRemoteOffer or kHaveLocalPranswer,
> as per the WebRTC specification.
>
> Bug: webrtc:8765
> Change-Id: I60ac67cd92b17fcbff964afc14d049481e816a28
> Reviewed-on: https://webrtc-review.googlesource.com/41042
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21779}
TBR=steveanton@webrtc.org,deadbeef@webrtc.org,pthatcher@webrtc.org
Change-Id: I90eacadb217353a7e098826563f5aeaaced52452
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8765
Reviewed-on: https://webrtc-review.googlesource.com/44581
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21781}
This is a reland of bbdabe50db0cf09f6007dda12a6476dc4602b174.
This was reverted because of breaking internal build. I contacted sheriff
and looked at logs but cannot find anything related to this CL. This was landed
with #3850 build which caused exception, but 3847-3855 seem to all have failed.
I am relanding to see if it will work this time or it will give some related
error message that can guide me.
Original change's description:
> Rename stereo video codec to multiplex
>
> This CL only does the rename from"stereo" to multiplex". With this we have a
> better name that doesn't clash with audio's usage of stereo.
>
> Bug: webrtc:7671
> Change-Id: Iebc3fc20839025f1bc8bcf0e16141bf9744ef652
> Reviewed-on: https://webrtc-review.googlesource.com/43242
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21769}
TBR=niklas.enbom@webrtc.org
Bug: webrtc:7671
Change-Id: I5934abad1ce28acf02842ea8ee2af7768a826eb8
Reviewed-on: https://webrtc-review.googlesource.com/44520
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21780}
This also changes the behavior of CreateAnswer to fail unless
the signaling state is kHaveRemoteOffer or kHaveLocalPranswer,
as per the WebRTC specification.
Bug: webrtc:8765
Change-Id: I60ac67cd92b17fcbff964afc14d049481e816a28
Reviewed-on: https://webrtc-review.googlesource.com/41042
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21779}
This removes use of the MockPeerConnection and replaces it with a fake
implementation of PeerConnection tailored to the needs of
StatsCollector and (soon) RTCStatsCollector.
The stats collector tests really care about the PeerConnection only so
much as to set up scenarios to test the StatsCollector with. Since each
scenario (e.g., adding a track) affects the results of multiple methods
(e.g., voice_channel and SessionStats), the tests were needing to
manually configure these dependent operations which was tedious, error
prone and difficult to change. The new fake lets the tests express the
scenario more concisely (e.g., AddVoiceChannel) while filling in all
the affected methods on the PeerConnection automatically. Furthermore,
this can be expanded to use with the RTCStatsCollector and to cover
more scenarios in the future.
Bug: webrtc:8764
Change-Id: I195074174684256543f7cdc27c3834e5ff0b4702
Reviewed-on: https://webrtc-review.googlesource.com/43521
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21778}
If a WebRTC build target requires gmock it has to include
test/gmock.h and just depend on //test:test_support.
Unfortunately //testtest_support was a leaky abstraction because it
wasn't propagating the correct -I compiler flag. To make everything
work, all the targets that use gmock started also to depend on
//testing/gmock (even if they were not including any gmock header
directly).
This CL makes //testtest_support propagate the include path up in the
dependency chain so it is possible to remove unused dependencies.
Note: all_dependent_configs should probably be used in the original
gmock target. There is an ongoing discussion about it. This CL solves
the problem on WebRTC side and it is forward compatible.
TBR=phoglund@webrtc.org
Bug: webrtc:8603
Change-Id: If08daf2ce9a6431a6e881a236743b4ec33b59ea7
Reviewed-on: https://webrtc-review.googlesource.com/44340
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21776}
Various files in webrtc codebase use scanf/sscanf function without
including stdio.h header file which is supposed to define it. This
somehow works when using glibc, but fails with uClibc.
Bug: webrtc:8641
Change-Id: Ie4ae17af32b32ed8cea567166b6b0e5193966995
Reviewed-on: https://webrtc-review.googlesource.com/32261
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21775}
This reverts commit bbdabe50db0cf09f6007dda12a6476dc4602b174.
Reason for revert: This breaks the internal build.
Original change's description:
> Rename stereo video codec to multiplex
>
> This CL only does the rename from"stereo" to multiplex". With this we have a
> better name that doesn't clash with audio's usage of stereo.
>
> Bug: webrtc:7671
> Change-Id: Iebc3fc20839025f1bc8bcf0e16141bf9744ef652
> Reviewed-on: https://webrtc-review.googlesource.com/43242
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21769}
TBR=sprang@webrtc.org,niklas.enbom@webrtc.org,qiangchen@chromium.org,emircan@webrtc.org
Change-Id: Icf019cb09e07de45821d31d7d8ea7707d01346ee
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7671
Reviewed-on: https://webrtc-review.googlesource.com/44360
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21774}
This reverts commit 4dc891f5e3a4bcad4db31e1af0ad45b6c471eef2.
Reason for revert: Reverting this CL to make it possible to revert https://webrtc-review.googlesource.com/c/src/+/43242
Original change's description:
> Add unit tests covering MultiplexImageComponent
>
> This CL changes some types in MultiplexImage and MultiplexImageComponent. Also,
> adds unit test coverage in TestMultiplexAdapter for these structs.
>
> Bug: webrtc:7671
> Change-Id: I832d0466dc67d3b6b7fa0d3fb76f02c0190e474f
> Reviewed-on: https://webrtc-review.googlesource.com/44081
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Reviewed-by: Qiang Chen <qiangchen@chromium.org>
> Cr-Commit-Position: refs/heads/master@{#21770}
TBR=qiangchen@chromium.org,emircan@webrtc.org
Change-Id: I9cce6ed5f2990a2f443e04a9e5913cbd296242e4
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7671
Reviewed-on: https://webrtc-review.googlesource.com/44341
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21773}
Incorrect length parameter was passed to memset (lenght of array in
elements instead of length in bytes, which is 8 times more since int64
is used).
Bug: none
Change-Id: I9100d1986377a8b3b9e475d1fbc215f4a1dedfb1
Reviewed-on: https://webrtc-review.googlesource.com/44280
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21771}
This CL changes some types in MultiplexImage and MultiplexImageComponent. Also,
adds unit test coverage in TestMultiplexAdapter for these structs.
Bug: webrtc:7671
Change-Id: I832d0466dc67d3b6b7fa0d3fb76f02c0190e474f
Reviewed-on: https://webrtc-review.googlesource.com/44081
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Qiang Chen <qiangchen@chromium.org>
Cr-Commit-Position: refs/heads/master@{#21770}
This CL only does the rename from"stereo" to multiplex". With this we have a
better name that doesn't clash with audio's usage of stereo.
Bug: webrtc:7671
Change-Id: Iebc3fc20839025f1bc8bcf0e16141bf9744ef652
Reviewed-on: https://webrtc-review.googlesource.com/43242
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21769}
Other headers, such as the libc++ headers, may depend on the
definition.
Bug: chromium:801780
Change-Id: I81e00708e08ab21b9456a8ed46ca7a1c1d4f7e50
Reviewed-on: https://webrtc-review.googlesource.com/43501
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Peter Collingbourne <pcc@google.com>
Cr-Commit-Position: refs/heads/master@{#21768}
"tun", "utun" and "tap" interfaces will now be assumed to be VPNs, if
the type is otherwise unknown.
This CL also moves GetAdapterTypeFromName out of BasicNetworkManager,
so that other network manager classes (e.g., the one in Chromium) can
use it too.
Bug: chromium:805759
Change-Id: I9988619666e2a9449cf5c089d24cf7d3afde8239
Reviewed-on: https://webrtc-review.googlesource.com/43580
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21767}
The AecMobile struct contains a ::farendOld field. It's type is 'short [2][80]'.
The field was initialized by
memset(&aecm->farendOld[0][0], 0, 160);
But sizeof(short) is not guaranteed to be 1. This causes use of
unititialized memory on some platforms. According to MSAN, it can
affect the output of the echo canceller.
The issue was found by the MSAN fuzzer.
This change initializes the array properly.
Bug: chromium:805396
Change-Id: Ibcaca2185cfa153e8fd826e9addfc04d7b65e417
Reviewed-on: https://webrtc-review.googlesource.com/43860
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21764}
This makes the binary around 8000 bytes smaller.
Bug: webrtc:8529
Change-Id: Ic59b16e300dd4dd5471e1079103982300cb5d660
Reviewed-on: https://webrtc-review.googlesource.com/43300
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21762}
A coherence vector cohxd is computed in
WebRtcAec_ComputeCoherence. The coherence values should theoretically
be 0 <= x <= 1. Due to the way they are computed that is not always
the case.
The coherence values are used to update an error signal
estimate hNl in webrtc::EchoSuppression. 'hNl[i]' should contain an
error magnitude for frequency 'i'.
The error magnitudes are used as a basis for exponentiation. If a
magnitude is negative, the result is NaN.
The NaNs will then spread to the output signal.
This change caps the hNl values at 0. I considered capping the
coherence values at 1. The coherence values are calculated differently
for MIPS, NEON and SSE. Therefore it's simpler to cap the hNl values
instead.
The issue was found by the AudioProcessing fuzzer.
Bug: chromium:804634
Change-Id: I8ebaa441d77c3f79d9c194a850cb2b9eed1c2024
Reviewed-on: https://webrtc-review.googlesource.com/43740
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21761}
