This reverts commit 8fb22e71ee9bd77676838c5723f7e89a74a64aa9.
Reason for revert: breaks downstream projects
Original change's description:
> Updates tests for turning simulcast streams on/off.
>
> Due to libvpx we were restricted to always turning the low simulcast
> stream on, or else the encoder would always label the active streams'
> encoded frames as key frames. Now that libvpx has been updated and
> rolled in, this change updates tests to reflect that it is working.
>
> Bug: webrtc:8653
> Change-Id: I065ef817ace2292605e27e135802cf4a3e90647e
> Reviewed-on: https://webrtc-review.googlesource.com/46340
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Seth Hampson <shampson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21831}
TBR=deadbeef@webrtc.org,sprang@webrtc.org,shampson@webrtc.org
Change-Id: If14074a7fc56c83b75584d8e9a6a913a40514bad
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8653
Reviewed-on: https://webrtc-review.googlesource.com/46840
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21832}
Due to libvpx we were restricted to always turning the low simulcast
stream on, or else the encoder would always label the active streams'
encoded frames as key frames. Now that libvpx has been updated and
rolled in, this change updates tests to reflect that it is working.
Bug: webrtc:8653
Change-Id: I065ef817ace2292605e27e135802cf4a3e90647e
Reviewed-on: https://webrtc-review.googlesource.com/46340
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21831}
It was enabled only when building with libstdc++ (the C++ library
bundled with gcc), which we rarely do these days. And it's unclear if
it ever worked well.
Bug: none
Change-Id: I1c4b3e498fb240ba946542afd194b254fcd2da19
Reviewed-on: https://webrtc-review.googlesource.com/46102
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21828}
Due to the growing number of arguments, these functions are being replaced by the AudioProcessingBuilder class.
Bug: webrtc:8668
Change-Id: Ic3936fbd47d92eac22a857a678dca5fd8c029d8b
Reviewed-on: https://webrtc-review.googlesource.com/46241
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21826}
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 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}
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}
