Commit Graph

34730 Commits

Author SHA1 Message Date
7cca016721 Hard code rtx handling option in NetEq.
This allows NetEq to adapt to late reordered packets which are common when using retransmissions.

Remaining cleanup of the plumbing from WebRTC API will be done in a follow-up cl.

Bug: webrtc:10178
Change-Id: Ia9911eaafdabd3b69441dc089116d79e24f1b2b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231002
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34898}
2021-09-01 18:18:59 +00:00
2c41cbae37 Revert "Wire up non-sender RTT for audio, and implement related standardized stats."
This reverts commit fb0dca6c055cbf9e43af665d3c437eba6f43372e.

Reason for revert: Speculative revert due to failing stats test in chromium. Possibly because the chromium test expected the metrics to not be supported, and now they are. Reverting just to unblock the webrtc roll into chromium.

Original change's description:
> Wire up non-sender RTT for audio, and implement related standardized stats.
>
> The implemented stats are:
> - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptime
> - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-totalroundtriptime
> - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptimemeasurements
>
> Bug: webrtc:12951, webrtc:12714
> Change-Id: Ia362d5c4b0456140e32da79d40edc06ab9ce2a2c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226956
> Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#34861}

# Not skipping CQ checks because original CL landed > 1 day ago.

TBR=hta,hbos,minyue

Bug: webrtc:12951, webrtc:12714
Change-Id: If07ad63286eea9cdde88271e61cc28f4b268b290
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231001
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34897}
2021-09-01 17:32:00 +00:00
78a8ce0a4c Delete decoder specific buffer_pool_size from webrtc::VideoCodec
VideoDecoder no longer uses this VideoCodec class,
thus this member is unused.

Bug: webrtc:13045
Change-Id: I6e46a563e90f2538bf288995a3837d95c00ba9cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230941
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34896}
2021-09-01 15:20:31 +00:00
84d1595d01 Rename VirtualSocketServer::SetDefaultRoute --> SetDefaultSourceAddress
and make docs a bit clearer.

Bug: None
Change-Id: I73504de96384012d18c00c527835fabab03a3791
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230544
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34895}
2021-09-01 14:27:29 +00:00
5b231de486 Make RtpPayloadParams::MinimalisticVp9Structure codec agnostic.
Bug: none
Change-Id: I97f603aad53933b09c761da954130b06ea5a5501
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230760
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34894}
2021-09-01 14:15:59 +00:00
b8a19df71c AGC2: removed unused noise estimator implementation
This CL also includes the following changes:
- `AudioProcessing::Config::GainController2::noise_estimator`
  deprecated
- `EnergyToDbfs()` optimized by removing unnecessary `sqrt`
- Unit test minor fix, incorrect type was used

Bug: webrtc:7494
Change-Id: I88a6672d6f7cd03fcf6a3031883522d256880140
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230940
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34893}
2021-09-01 12:45:20 +00:00
0d51a5fc00 Revert "PipeWire capturer: implement proper DMA-BUFs support"
This reverts commit f2177f6612079ccce9c320ea7e77bc934c684f5c.

Reason for revert: Broke WebRTC to Chrome rolls:
https://chromium-review.googlesource.com/c/chromium/src/+/3135220
example: https://ci.chromium.org/ui/p/chromium/builders/try/fuchsia-x64-cast/431230/overview

ERROR at //third_party/webrtc/modules/desktop_capture/linux/egl_dmabuf.cc:26:11: Include not allowed.
#include "rtc_base/sanitizer.h"
          ^-------------------
It is not in any dependency of
  //third_party/webrtc/modules/desktop_capture:desktop_capture_generic
The include file is in the target(s):
  //third_party/webrtc/rtc_base:sanitizer
which should somehow be reachable.



Original change's description:
> PipeWire capturer: implement proper DMA-BUFs support
>
> Currently both KWin (KDE) and Mutter (GNOME) window managers don't
> use DMA-BUFs by default, but only when client asks specifically for
> them (KWin) or when experimental DMA-BUF support is enabled (Mutter).
> While current implementation works just fine on integrated graphics
> cards, it causes issues on dedicated GPUs (AMD and NVidia) where the
> code either crashes or screensharing is slow and unusable.
>
> To fix this, DMA-BUFs has to be opened using OpenGL context and not
> being directly mmaped(). This implementation requires to use DMA-BUF
> modifiers, as they are now mandatory for DMA-BUFs usage.
>
> Documentation for this behavior can be found here:
> https://gitlab.freedesktop.org/pipewire/pipewire/-/blob/master/doc/dma-buf.dox
>
> Bug: chromium:1233417, webrtc:13137
> Change-Id: I0cecf16d6bb0f576954b9e8f071cab526f7baf2c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227022
> Commit-Queue: Tommi <tommi@webrtc.org>
> Reviewed-by: Tommi <tommi@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#34889}

TBR=mbonadei@webrtc.org,tommi@webrtc.org,sprang@webrtc.org,mfoltz@chromium.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com,grulja@gmail.com

Change-Id: I2c573f17adbb216156cd72f62f4dbb7328f8fb6a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1233417, webrtc:13137
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230944
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Commit-Queue: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34892}
2021-09-01 11:32:42 +00:00
eb89027733 Revert "frame transformer: make GetPayloadType pure virtual again"
This reverts commit 209ac5fd95594ab3834dad3e3dbd14c8196637bc.

Reason for revert: Breaks WebRTC autoroll presubmit:
https://chromium-review.googlesource.com/c/chromium/src/+/3134502
Example failure https://ci.chromium.org/ui/p/chromium/builders/try/mac-rel/775468/overview

../../buildtools/third_party/libc++/trunk/include/__memory/unique_ptr.h:725:32: error: allocating an object of abstract class type 'testing::NiceMock<blink::(anonymous namespace)::MockTransformableVideoFrame>'
    return unique_ptr<_Tp>(new _Tp(_VSTD::forward<_Args>(__args)...));
                               ^
../../third_party/blink/renderer/platform/peerconnection/rtc_encoded_video_stream_transformer_test.cc:69:26: note: in instantiation of function template specialization 'std::make_unique<testing::NiceMock<blink::(anonymous namespace)::MockTransformableVideoFrame>>' requested here
  auto mock_frame = std::make_unique<NiceMock<MockTransformableVideoFrame>>();
                         ^
../../third_party/webrtc/api/frame_transformer_interface.h:36:19: note: unimplemented pure virtual method 'GetPayloadType' in 'NiceMock'
  virtual uint8_t GetPayloadType() const = 0;
                  ^


Original change's description:
> frame transformer: make GetPayloadType pure virtual again
>
> after chrome was updated in
>   https://chromium-review.googlesource.com/c/chromium/src/+/3103323
>
> BUG=webrtc:13077
>
> Change-Id: I7e5ff6aaae81c5dcfbaa41b09ef01bc95bb7251a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230143
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
> Cr-Commit-Position: refs/heads/main@{#34877}

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:13077
Change-Id: I6b2e4e2804890c857f1f832a6a4faa614ec026c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230920
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34891}
2021-09-01 10:24:10 +00:00
b062829311 Reland "Enable WebRTC-Vp9DependencyDescriptor by default"
This is a reland of 472707150662bc4e174072e445938e5c405aa884

Original change's description:
> Enable WebRTC-Vp9DependencyDescriptor by default
>
> Bug: chromium:1178444
> Change-Id: I420e1e9b3c557b8b186cb08c15b962a779e1ca17
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226941
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34584}

Bug: chromium:1178444
Change-Id: I874412b41e657179be6ffbe399617e18a29ec804
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230121
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34890}
2021-09-01 09:27:39 +00:00
f2177f6612 PipeWire capturer: implement proper DMA-BUFs support
Currently both KWin (KDE) and Mutter (GNOME) window managers don't
use DMA-BUFs by default, but only when client asks specifically for
them (KWin) or when experimental DMA-BUF support is enabled (Mutter).
While current implementation works just fine on integrated graphics
cards, it causes issues on dedicated GPUs (AMD and NVidia) where the
code either crashes or screensharing is slow and unusable.

To fix this, DMA-BUFs has to be opened using OpenGL context and not
being directly mmaped(). This implementation requires to use DMA-BUF
modifiers, as they are now mandatory for DMA-BUFs usage.

Documentation for this behavior can be found here:
https://gitlab.freedesktop.org/pipewire/pipewire/-/blob/master/doc/dma-buf.dox

Bug: chromium:1233417, webrtc:13137
Change-Id: I0cecf16d6bb0f576954b9e8f071cab526f7baf2c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227022
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34889}
2021-09-01 08:10:40 +00:00
338d31435d Add possibility to specify a realm with mb.py
Bug: webrtc:13134
Change-Id: I886b8b7612d4f1c59abe2c2484ab9e556bcd27ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230784
Reviewed-by: Christoffer Jansson <jansson@google.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34888}
2021-08-31 20:16:42 +00:00
126d0b932f Avoid windows.h in shared_memory.h
shared_memory.h ends up being included by consumers of webrtc such as
Chromium so it causes namespace pollution. Specifically, it causes
SendMessageCallback to be defined as SendMessageCallbackW partway
through compilation of security_key_auth_handler_win_unittest.cc,
leading to renaming of SendMessageCallback when it is used but not when
it is defined.

Bug: Chromium:796644
Change-Id: Ib1acc0d736a0a6cf97e318e773b20d9a432f6b77
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229901
Commit-Queue: Bruce Dawson <brucedawson@chromium.org>
Reviewed-by: Joe Downing <joedow@chromium.org>
Cr-Commit-Position: refs/heads/main@{#34887}
2021-08-31 16:40:32 +00:00
5af152c214 Inroduce BitstreamReader class to parse sequences of bits
With intent to replace BitBuffer.
This version is optimised for binary size and readability of the parsers
at the cost of slower parsing of invalid bitstreams

Bug: None
Change-Id: Ib054e2a7758b9a69cbf2559e739465b104a7dcf3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230244
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34886}
2021-08-31 15:08:49 +00:00
b7a74c3805 Remove inactive owners.
Bug: None
Change-Id: I7f2ccf6986077a3748727311d5cd0e5dac9dbf70
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230801
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34885}
2021-08-31 14:27:49 +00:00
00dd5ced24 Delete deprecated VideoDecoder::InitDecode
Bug: webrtc:13045
Change-Id: Id1ca822c3be5a4f496dd67b59eab31a79a74bf67
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228949
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34884}
2021-08-31 14:13:49 +00:00
25edb62a94 Calculate relative arrival delay for reordered packets.
This changes behavior slightly but results in a better delay estimate and cleaner code.

Bug: webrtc:10178
Change-Id: If150258bc1ea58149940f17c5660733ff61159c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230740
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34883}
2021-08-31 12:26:41 +00:00
9680d29e8d dcsctp: Support unlimited max_retransmissions
The restart limit for timers can already be limitless, but the
RetransmissionErrorCounter didn't support this. With this change, the
max_retransmissions and max_init_retransmits can be absl::nullopt to
indicate that there should be infinite retries.

Bug: webrtc:13129
Change-Id: Ia6e91cccbc2e1bb77b3fdd7f37436290adc2f483
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230701
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34882}
2021-08-31 10:57:48 +00:00
4281208fbd Update CallPerfTest.TestEncodeFramerateVp8Simulcast
Only verify simulcast layers with reduced framerate (FramerateController used) and not input fps for now.
Input framerate varies in some tests.

Bug: webrtc:13031
Change-Id: I19b14b9fba70da2df49c0471b67e4c3a5fea4a2e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230782
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34881}
2021-08-31 10:26:38 +00:00
9adbbebde9 Removing usage of std::unordered_set
In Chromium, they are discouraged.

Bug: webrtc:12689
Change-Id: I0e2a03b909d8a6d239e11969659e4fdc1a89766c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229188
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34880}
2021-08-31 08:59:41 +00:00
7bf22e574a Update WebRTC code version (2021-08-31T04:06:38).
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com,mbonadei@webrtc.org

Bug: None
Change-Id: Icb2c414a0745a9e6342a77f8a945d3f2a236ca7a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230727
Reviewed-by: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#34879}
2021-08-31 05:20:27 +00:00
933afdf197 Restrict WGC screen capture to Windows version 20H1 and greater.
The Windows.Graphics.Capture API CreateForMonitor has a bug that was
fixed in 20H1 that causes an exception to be thrown when an HMONITOR
with a value of 0 is provided. This is a valid input used to request
capture of all monitors. To avoid this issue, we can restrict screen
capture using WGC to versions of Windows >=20H1.

Bug: webrtc:13078
Change-Id: Ia66bf2b2738c29813d41e214fdfc1eb96e0a1312
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229140
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Austin Orion <auorion@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#34878}
2021-08-30 19:59:03 +00:00
209ac5fd95 frame transformer: make GetPayloadType pure virtual again
after chrome was updated in
  https://chromium-review.googlesource.com/c/chromium/src/+/3103323

BUG=webrtc:13077

Change-Id: I7e5ff6aaae81c5dcfbaa41b09ef01bc95bb7251a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230143
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#34877}
2021-08-30 15:35:42 +00:00
9fdcfe90f1 In RtcpTransceiver add support for receiving network generic messages
These message suppose to extract all information
NetworkControllerInterface may need from rtcp.

Bug: webrtc:8239
Change-Id: I21d9081ad147ca8abe1ae05ca7201568c6ff77d1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230421
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34876}
2021-08-30 13:58:18 +00:00
10721227e4 Revert "Propagate socket write errors for DtlsTransport"
There is a suspicion that it causes OpenSSL errors:
  [openssl_stream_adapter.cc(961)]
       OpenSSLStreamAdapter::Error(SSL_write, 5, 0)

This commit does change the interaction with OpenSSL as propagating the
socket write errors as SR_BLOCK results in calling BIO_set_retry_write,
as part of current implementation of OpenSSLStreamAdapter.

Testing this regression has proven to be hard to do manually.

This reverts commit edfaaef086ccff2dbff29d64c9a8d9f633637c57.

Bug: webrtc:12943
Change-Id: Ib6767bd4af68c59fd3b7cb051341876f175bb921
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230420
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34875}
2021-08-30 12:28:25 +00:00
260400d37f Fix NPE when setting the camera2 stabilization mode
Fatal Exception: java.lang.NullPointerException: Attempt to get length of null array
       at org.webrtc.Camera2Session$CaptureSessionCallback.chooseStabilizationMode(Camera2Session.java:234)
       at org.webrtc.Camera2Session$CaptureSessionCallback.onConfigured(Camera2Session.java:172)
       at android.hardware.camera2.impl.CallbackProxies$SessionStateCallbackProxy.lambda$onConfigured$0(CallbackProxies.java:53)
       at android.hardware.camera2.impl.-$$Lambda$CallbackProxies$SessionStateCallbackProxy$soW0qC12Osypoky6AfL3P2-TeDw.run(-.java:4)
       at android.os.Handler.handleCallback(Handler.java:873)
       at android.os.Handler.dispatchMessage(Handler.java:99)
       at android.os.Looper.loop(Looper.java:193)
       at android.os.HandlerThread.run(HandlerThread.java:65)

Bug: webrtc:13032
Change-Id: Ifb6ef920b700ca03d37c64803c0b34230785846f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227292
Reviewed-by: Xavier Lepaul‎ <xalep@webrtc.org>
Commit-Queue: Xavier Lepaul‎ <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34874}
2021-08-30 12:25:15 +00:00
424b420a22 Fix NPE when accessing Android camera focus modes
Looks like getSupportedFocusModes() may return null, despite the documentation stating otherwise.

Bug: webrtc:13032
Change-Id: I0119b8a97be9ef4340c3e93f16e2dcaa899f2f3c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227288
Reviewed-by: Xavier Lepaul‎ <xalep@webrtc.org>
Commit-Queue: Xavier Lepaul‎ <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34873}
2021-08-30 12:21:35 +00:00
1460e15a45 Fix NPE when converting camera2 supported sizes
StreamConfigurationMap.getOutputSizes() may return null:
https://developer.android.com/reference/android/hardware/camera2/params/StreamConfigurationMap#getOutputSizes(java.lang.Class%3CT%3E)

Fixes:

Fatal Exception: java.lang.NullPointerException: Attempt to get length of null array
       at org.webrtc.Camera2Enumerator.convertSizes(Camera2Enumerator.java:234)
       at org.webrtc.Camera2Enumerator.getSupportedSizes(Camera2Enumerator.java:147)
       at org.webrtc.Camera2Session.findCaptureFormat(Camera2Session.java:325)
       at org.webrtc.Camera2Session.start(Camera2Session.java:313)
       at org.webrtc.Camera2Session.<init>(Camera2Session.java:296)
       at org.webrtc.Camera2Session.create(Camera2Session.java:274)
       at org.webrtc.Camera2Capturer.createCameraSession(Camera2Capturer.java:35)
       at org.webrtc.CameraCapturer$5.run(CameraCapturer.java:272)
       at android.os.Handler.handleCallback(Handler.java:883)
       at android.os.Handler.dispatchMessage(Handler.java:100)
       at android.os.Looper.loop(Looper.java:237)
       at android.os.HandlerThread.run(HandlerThread.java:67)

Bug: webrtc:13032
Change-Id: I9154be567cd12c066087818ba22e9cd69e75a22f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227291
Reviewed-by: Xavier Lepaul‎ <xalep@webrtc.org>
Commit-Queue: Xavier Lepaul‎ <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34872}
2021-08-30 12:18:45 +00:00
cd0a4f5ff5 Handle camera2 session creation errors
openCamera may throw IllegalArgumentException:
    https://developer.android.com/reference/android/hardware/camera2/CameraManager#openCamera(java.lang.String,%20android.hardware.camera2.CameraDevice.StateCallback,%20android.os.Handler)

Bug: webrtc:13032
Change-Id: I9d094691ca38f9baf312168cd67c323fd4ed5d37
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227293
Reviewed-by: Xavier Lepaul‎ <xalep@webrtc.org>
Commit-Queue: Xavier Lepaul‎ <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34871}
2021-08-30 10:52:55 +00:00
75b0f5575e Replace legacy getStats with standard getStats in the Android example
Bug: webrtc:12688
Change-Id: I7e2e10ab1b1ce994bbfbcfad377a77b39239d3d0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221760
Reviewed-by: Xavier Lepaul‎ <xalep@webrtc.org>
Commit-Queue: Xavier Lepaul‎ <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34870}
2021-08-30 10:45:15 +00:00
3cd7a0ffdd Remove media/base/h264_profile_level_id.* and media/base/vp9_profile.h
The content of these files was moved to api/video_codecs in
https://webrtc.googlesource.com/src.git/+/c3fcee7c3a7714afc3e37d4753b40f4fdbc3653e
but the original files could not be removed due to dependencies
in downstream projects.

Bug: chromium:1187565
Change-Id: I414efa22102bfdea0765fa72a8cf8b0bd5c090db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229303
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34869}
2021-08-30 10:31:08 +00:00
9ad972d4fb Remove deprecated signature of VideoDecoderFactory::QueryCodecSupport
This function was deprecated in this CL
https://webrtc-review.googlesource.com/c/src/+/229184

Bug: chromium:1187565
Change-Id: Ic0e18af69185b48accc441c4bbe1a2d8926db383
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230241
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34868}
2021-08-30 10:26:05 +00:00
59947d2871 SimulcastEncoderAdapter: Use FramerateController instead of FramerateControllerDeprecated.
Results from test (CallPerfTest.TestEncodeFramerateVp8Simulcast):
Simulcast streams:
0: max_fps:20 -> StreamStats.encode_frame_rate:15 (before), 20 (after)
1: max_fps:30

Bug: webrtc:13031
Change-Id: I30e6b2dcb2746859bd3e21b098bfa7b0fb3b2dda
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230120
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34867}
2021-08-30 10:20:55 +00:00
8177f58dde [PCLF] Add support for dumping video with multiple receivers
Bug: b/197896468
Change-Id: I7896246eedb2e9efe847df4dddfc8ef05f7d152b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230424
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34866}
2021-08-30 10:09:05 +00:00
6eb30e40af [DVQA] Tolerate receiving frames which were considerer as dropped before
It can happen that SFU will resend the frame which was before
considered as dropped during stream switching.

Bug: b/197740434
Change-Id: I95a67e6e637f6005a24df15875b50133a6e8eaaf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230423
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34865}
2021-08-30 10:04:36 +00:00
44b919c10a Remove use of UiThreadTestRule and migrate to UiThreadTest in chromium
Remove android.support.test.rule.UiThreadTestRule as chromium did in [1] and
Replace android.support.test.annotation.UiThreadTest
with org.chromium.base.test.UiThreadTest.

Also remove unused uiThreadHandler from NetworkMonitorTest.

[1] https://crrev.com/c/2332301

Bug: webrtc:11962
Change-Id: I8f3781d43d4d53d8158c39c81568d8b09b2bec6a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230220
Reviewed-by: Xavier Lepaul‎ <xalep@webrtc.org>
Commit-Queue: Byoungchan Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#34864}
2021-08-30 10:01:16 +00:00
0d175356cb Revert "Always unwrap VP9 TL0PicIdx forward if the frame is newer."
This reverts commit dbab1be1d13060666b303209eded45c55cb46856.

Reason for revert: Breaks VP9 media performance under heavy packet loss.

Original change's description:
> Always unwrap VP9 TL0PicIdx forward if the frame is newer.
>
> Bug: webrtc:12979
> Change-Id: Idcc14f8f61b04f9eb194b55ffa40fb95319a881c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226463
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34513}

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:12979
Change-Id: Id315db8d67143372724448b8801a86aee9a2f0aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230422
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34863}
2021-08-30 09:23:47 +00:00
e5b4e941a0 Surface audio unit errors.
With this change, we catch audio unit start errors and pipe them to the
audio session. The audio session notifies its delegate, which can then
take appropriate action based on the error code.
The signal follows the same path as the playout glitch detection.

Bug: webrtc:13119
Change-Id: I8c9f9d2a1e3457447d0ce61ad197f7e1c6392837
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230240
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Xavier Lepaul‎ <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34862}
2021-08-30 09:06:25 +00:00
fb0dca6c05 Wire up non-sender RTT for audio, and implement related standardized stats.
The implemented stats are:
- https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptime
- https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-totalroundtriptime
- https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptimemeasurements

Bug: webrtc:12951, webrtc:12714
Change-Id: Ia362d5c4b0456140e32da79d40edc06ab9ce2a2c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226956
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34861}
2021-08-30 09:03:50 +00:00
58157b5cd2 Update WebRTC code version (2021-08-30T04:04:31).
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com,mbonadei@webrtc.org

Bug: None
Change-Id: I5b68b7aa0e3da01d280c781905979ac8dcb76e1f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230601
Reviewed-by: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#34860}
2021-08-30 09:02:49 +00:00
575498ffc2 Tweak VP8 payload to comply with RFC 7741
This updates the VP8 payload diagrams to be compliant with RFC 7741. It
also fixes some minor inconsistencies with PID, previously referred to
as PartID.

Bug: None
Change-Id: I33eb57d96f3d95b01ef5f0afa21a9dc54b41db2d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230243
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34859}
2021-08-30 09:01:47 +00:00
66528d7e90 Export DxgiDuplicatorController when building as shared lib
This class is accessed by Electron for its desktop capture support,
but it breaks with component builds on Windows because the symbols
aren't exported by the dll.
No behavior change at runtime, only modifies the generated .lib
when building as a shared library (static builds are unchanged).

Bug: None
Change-Id: I5dc606846de990c1bf4d375ddbb1c73dfc512762
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230200
Reviewed-by: Joe Downing <joedow@chromium.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/main@{#34858}
2021-08-26 17:34:45 +00:00
09fb787f9a Use absl instead of self-made function for low-level bit counting
to reduce code duplication and rely on better optimized code.

Bug: None
Change-Id: Ie2f1ff680ff702aae84132229ae0e1743478424f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229385
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34857}
2021-08-26 08:56:37 +00:00
c80c566134 Update WebRTC code version (2021-08-26T04:03:38).
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com,mbonadei@webrtc.org

Bug: None
Change-Id: I9f52ad581d8fc102f035d33b35628dca2ad4dd84
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230203
Reviewed-by: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#34856}
2021-08-26 06:06:56 +00:00
27edde3182 Handle camera1 session creation errors more gracefully
Specifically, defer getting the camera index so the error can be
reported instead of crashing:

Fatal Exception: java.lang.IllegalArgumentException: No such camera: Camera 1, Facing front, Orientation 270
       at org.webrtc.Camera1Enumerator.getCameraIndex(Camera1Enumerator.java:170)
       at org.webrtc.Camera1Capturer.createCameraSession(Camera1Capturer.java:31)
       at org.webrtc.CameraCapturer$5.run(CameraCapturer.java:272)
       at android.os.Handler.handleCallback(Handler.java:790)
       at android.os.Handler.dispatchMessage(Handler.java:99)
       at android.os.Looper.loop(Looper.java:214)
       at android.os.HandlerThread.run(HandlerThread.java:65)

Bug: webrtc:13032
Change-Id: Ida6bc65046770c11c2b3ee832906e8454cec10df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227290
Reviewed-by: Xavier Lepaul‎ <xalep@webrtc.org>
Commit-Queue: Xavier Lepaul‎ <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34855}
2021-08-25 17:04:40 +00:00
68952fed31 Handle camera2 session start error
getCameraCharacteristics() may throw IllegalArgumentException:

Fatal Exception: java.lang.IllegalArgumentException: supportsCameraApi:2569: Unknown camera ID 1
       at android.hardware.camera2.CameraManager.throwAsPublicException(CameraManager.java:1119)
       at android.hardware.camera2.CameraManager.getCameraCharacteristics(CameraManager.java:531)
       at org.webrtc.Camera2Session.start(Camera2Session.java:304)
       at org.webrtc.Camera2Session.<init>(Camera2Session.java:296)
       at org.webrtc.Camera2Session.create(Camera2Session.java:274)
       at org.webrtc.Camera2Capturer.createCameraSession(Camera2Capturer.java:35)
       at org.webrtc.CameraCapturer$5.run(CameraCapturer.java:272)
       at android.os.Handler.handleCallback(Handler.java:883)
       at android.os.Handler.dispatchMessage(Handler.java:100)
       at android.os.Looper.loop(Looper.java:237)
       at android.os.HandlerThread.run(HandlerThread.java:67)

Bug: webrtc:13032
Change-Id: I30b6d6da40bc90a94c0c3c79f9dff523182d3da4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227289
Reviewed-by: Xavier Lepaul‎ <xalep@webrtc.org>
Commit-Queue: Xavier Lepaul‎ <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34854}
2021-08-25 17:01:51 +00:00
0f549f908c Catch RuntimeException on Camera.setDisplayOrientation
Bug: webrtc:13032
Change-Id: I3736e61b8f49ae058851d7f5d60858454e5d5b09
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227287
Reviewed-by: Xavier Lepaul‎ <xalep@webrtc.org>
Commit-Queue: Xavier Lepaul‎ <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34853}
2021-08-25 15:30:51 +00:00
2ee0e64696 Add support for manually configuring subnets as VPN
This patch adds support for manually setting subnets that
should be handled as VPN, i.e be subject to VpnPreference,
in case webrtc fails to auto-detect VPNs.

Bug: webrtc:13097
Change-Id: I42514f0677a35cfe30ad053570fa9c2a5b4a856b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230122
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34852}
2021-08-25 14:49:11 +00:00
c5cb7f1fad Fix NPE if no compatible capture format was found
Fatal Exception: java.lang.NullPointerException: Attempt to read from field 'int org.webrtc.CameraEnumerationAndroid$CaptureFormat.width' on a null object reference
       at org.webrtc.Camera2Session$CameraStateCallback.onOpened(Camera2Session.java:122)
       at android.hardware.camera2.impl.CameraDeviceImpl$1.run(CameraDeviceImpl.java:151)
       at android.os.Handler.handleCallback(Handler.java:938)
       at android.os.Handler.dispatchMessage(Handler.java:99)
       at android.os.Looper.loop(Looper.java:246)
       at android.os.HandlerThread.run(HandlerThread.java:67)


Fix NPE when setting the camera2 stabilization mode

Fatal Exception: java.lang.NullPointerException: Attempt to get length of null array
       at org.webrtc.Camera2Session$CaptureSessionCallback.chooseStabilizationMode(Camera2Session.java:234)
       at org.webrtc.Camera2Session$CaptureSessionCallback.onConfigured(Camera2Session.java:172)
       at android.hardware.camera2.impl.CallbackProxies$SessionStateCallbackProxy.lambda$onConfigured$0(CallbackProxies.java:53)
       at android.hardware.camera2.impl.-$$Lambda$CallbackProxies$SessionStateCallbackProxy$soW0qC12Osypoky6AfL3P2-TeDw.run(-.java:4)
       at android.os.Handler.handleCallback(Handler.java:873)
       at android.os.Handler.dispatchMessage(Handler.java:99)
       at android.os.Looper.loop(Looper.java:193)
       at android.os.HandlerThread.run(HandlerThread.java:65)

Bug: webrtc:13032
Change-Id: I6edd9f0061c445f90ab0881d78183077f89e391f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227294
Reviewed-by: Xavier Lepaul‎ <xalep@webrtc.org>
Commit-Queue: Xavier Lepaul‎ <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34851}
2021-08-25 13:35:11 +00:00
75bbd1fbe6 Revert "red: generate and parse the red fmtp format"
This reverts commit 9d0730942677a520ce7e184d081b4c5a2469fc48.

Reason for revert: Speculative revert due to failing downstream test. If the test recovers, I'll assign the issue to the tests owners.

Original change's description:
> red: generate and parse the red fmtp format
>
> generates a fmtp line like
>   a=fmtp:<red payloadtype> <opus payloadtype>/<opus payloadtype>
> and matches the incoming redundant payload types against the
> send codec one. Offers without an FMTP line will not use RED.
> Redundancy levels of 1 (plus main packet ) to 32 are accepted but
> this is not wired up to the encoder since the O/A semantic of
> RFC 2198 is not clear.
>
> This decreases the chance of a collision with the SATIN codec
> which also runs on 48khz (but so far does not specify a channelCount of 2)
>
> BUG=webrtc:11640
>
> Change-Id: I8755e5b1e944d105212f1bbe4f330cf4e0753e67
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229583
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#34848}

TBR=henrik.lundin@webrtc.org,hta@webrtc.org,philipp.hancke@googlemail.com,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com

Change-Id: I5a0816a22a2a213679ab047c61e3b1dda40c4f59
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11640
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230140
Reviewed-by: Björn Terelius <terelius@google.com>
Commit-Queue: Björn Terelius <terelius@google.com>
Cr-Commit-Position: refs/heads/main@{#34850}
2021-08-25 11:46:34 +00:00
b7aac6f5f4 Update SdpOfferAnswerHandler to use rtc::make_ref_counted
Also change return type of FinalRefCountedObject::Release() to
RefCountReleaseStatus, for consistency with other refcount classes.

Bug: webrtc:12701
Change-Id: I37c325e78ba7ae3e220b618da02cb243604ca4cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229590
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34849}
2021-08-25 11:00:12 +00:00