This fix the mismatch of resolution VP9 wrapper set for coded frame with
its actual resolution.
Bug: webm:1485, webrtc:5749
Change-Id: Ie1225d8f3a3d00e66229a1a79858d0a89b3d5fae
Reviewed-on: https://webrtc-review.googlesource.com/46040
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21819}
This CL adds a GN build flag to include builtin software codecs
(enabled by default).
When setting the flag to false, libvpx can also be excluded. The
benefit is that the resulting binary is smaller.
Replaces https://webrtc-review.googlesource.com/c/src/+/29203
Bug: webrtc:7925
Change-Id: Id330ea8a43169e449ee139eca18e4557cc932e10
Reviewed-on: https://webrtc-review.googlesource.com/36340
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21818}
This change list contains the structured logging module for ICE using
the RtcEventLog infrastructure, and also extension to the log parser and
analyzer.
Bug: None
Change-Id: I6539cf282155c2cde4d3161c53500c0746671a02
Reviewed-on: https://webrtc-review.googlesource.com/34622
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21816}
Previously, if a media section is rejected by the answerer, the TransportInfo
of that section will not be added to the answer but when answer SDP is
deserialized by the offerer, the rejected TransportInfo will be added.
This CL fixes this inconsistency by adding the TransportInfo of all the m=
sections including the rejected ones.
Bug: webrtc:8818
Change-Id: I3b163245979c4ac4df31db39262e499f1c4901c4
Reviewed-on: https://webrtc-review.googlesource.com/46380
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21815}
The reasons behind this change:
1. In OpenSSL 1.1.0. BIO will be an opaque object. We won't have direct access to the `num` field.
2. `num` is only used by OpenSSL provided BIOs and different types of BIOs use num differently.
WebRTC is providing its own customized BIO implementation, it probably shouldn't piggyback into
this internal field to store the stream/socket state.
4. We can access the stream/socket state directly using the underlying object anyway.
Bug: webrtc:8817
Change-Id: I41cdd2920fba378e312e8436a7b9733381555522
Reviewed-on: https://webrtc-review.googlesource.com/46360
Commit-Queue: Jiawei Ou <ouj@fb.com>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21814}
This happens when pc.close() is called.
As a stopgap measure, we return zeroes instead, leading to stats
being omitted.
Bug: chromium:807174
Change-Id: I36f342adcd038822afb75d8593de808591eb9c4b
Reviewed-on: https://webrtc-review.googlesource.com/46161
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21813}
This is a temporary measure until the synchronization method
used in the class, gets fixed.
Bug: chromium:796889, chromium:795340
Change-Id: Ie3d394ae42f005e8e0f353d04ea9c1d053ea9fd2
Reviewed-on: https://webrtc-review.googlesource.com/40460
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21812}
The original rtc_event_log_api is refactored to a pure API target plus
multiple targets coupled with WebRTC implementations.
Bug: None
Change-Id: Iab9eee3f7bf4228c52d94a5f26fc39bb99b5033f
Reviewed-on: https://webrtc-review.googlesource.com/43247
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@google.com>
Cr-Commit-Position: refs/heads/master@{#21811}
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}
