This CL adds wrappers for the following PeerConnection native
APIs to the Objective C API:
- SdpSemantics enum added to the RTCConfiguration
- RTCRtpTransceiver
- RTCPeerConnection.addTrack
- RTCPeerConnection.removeTrack
- RTCPeerConnection.addTransceiver
- RTCPeerConnection.transceivers
Bug: webrtc:8870
Change-Id: I9449df9742a59e90894712dc7749ca30b569d94b
Reviewed-on: https://webrtc-review.googlesource.com/54780
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22214}
This allows clients to move to these new accessors and off of the
sync_label field which is deprecated.
Bug: webrtc:7932
Change-Id: I32b30087bf1be380d607a649bd90fa9617dafeb9
Reviewed-on: https://webrtc-review.googlesource.com/58020
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22213}
To be able to safely move over to the new code, the revised
code is added alongside the old code. Most of the files added
in this CL are more or less direct copies of the previous code.
This new version of send side congestion controller will be
activated under a field trial in a followup CL.
Bug: webrtc:8415
Change-Id: I034e583cf891a8f6357119739a1517cc0a4abe88
Reviewed-on: https://webrtc-review.googlesource.com/53322
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22212}
The flag is passed as --isolated-script-test-perf-output=/b/whatever
on the bots, but this code expected a blank space instead of =.
Bug: webrtc:8932
Change-Id: I9ca48c9b285e365ac23a04ea2e89d9a8e75f5540
Reviewed-on: https://webrtc-review.googlesource.com/58088
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22211}
Now VCMTiming::MaxWaitingTime will not clip negative values. Thus frame
buffer will be able to distinguish between late frames and when waiting
cycle was simply interrupted by a new inserted frame right before the
waiting timer would expire.
Bug: webrtc:8917
Change-Id: I6b253f459fcb3a346064a103cc92ee332b074e1b
Reviewed-on: https://webrtc-review.googlesource.com/57741
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22210}
Allows mixing up to 4 input streams. Useful for profiling and manual
tests. Allows testing different combinations of input/output rates and
number of channels. Reads and writes WAV files. Can also configure
whether to use the Limiter component of the AudioMixer.
Bug: webrtc:8925
Change-Id: Iaf4fee5284980f6ed01f4bb721e49bb1af8dd392
Reviewed-on: https://webrtc-review.googlesource.com/56842
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22209}
This is a reland of 12dc1842d62ee8df1e462f9b6a617fef9ab8b3b7.
Original change's description:
> Some cleanup for the logging code:
>
> * Only include 'tag' for Android. Before there was an
> extra std::string variable per log statement for all
> platforms.
> * Remove unused logging macro for Windows and 'module' ctor argument.
> * Move httpcommon code out of logging and to where it's used.
>
> Change-Id: I347805d3df2cee2840c0d2eef5bfefaff1cdbf37
> Bug: webrtc:8928
> Reviewed-on: https://webrtc-review.googlesource.com/57183
> Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Tommi <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22184}
Bug: webrtc:8928
Change-Id: Ib97895aaeb376e19f136d258c0259a340235a5d1
Reviewed-on: https://webrtc-review.googlesource.com/58200
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22208}
The webrtc::AudioMixer uses a limiter component. This CL allows
changes the APM-AGC limiter to the APM-AGC2 limiter though a Chrome
field trial.
The AGC2 limiter has a float interface. We plan to eventually switch
to the AGC2 limiter. Therefore, we will now mix in de-interleaved
floats. Float mixing will happen both when using the old limiter and
when using the new one.
After this CL the mixer will support two limiters. The limiters have
different interfaces and need different processing steps. Because of
that, we make (rather big) changes to the control flow in
FrameCombiner. For a short while, we will mix in deinterleaved floats
when using any limiter.
Originally landed in https://webrtc-review.googlesource.com/c/src/+/56141/
Reverted in https://webrtc-review.googlesource.com/c/src/+/57940
because of both breaking compilation and having a severe error. The
error is fixed and a test is added. The compilation issue is fixed.
Bug: webrtc:8925
Change-Id: Ieba138dee9652c826459fe637ae2dccbbc06bcf0
Reviewed-on: https://webrtc-review.googlesource.com/58085
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22207}
This reverts commit 63e83c77ae81730a78ec4d5bf0465f25970f867a.
Reason for revert: JNI generator is not using the heap profiler
anymore.
Original change's description:
> Forward fix jni_generator_helper.h.
>
> In crrev.com/531028, the JNI generator starts to add heap profiler
> events to JNI generated functions.
>
> This will cause a ~80KiB regression and at the moment it is breaking
> the Chromium Roll into WebRTC.
>
> This CL defines a void macro to re-enable the Chromium Roll avoiding
> the size regression.
>
> Bug: chromium:801260
> Change-Id: I9543299199c4e14b6b9b235c5cb98c0d53cf29ea
> Reviewed-on: https://webrtc-review.googlesource.com/43021
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21730}
TBR=mbonadei@webrtc.org,magjed@webrtc.org,sakal@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: chromium:801260
Change-Id: I7dac211b89d8206dc461af0a17b6d53cc8661b2a
Reviewed-on: https://webrtc-review.googlesource.com/58040
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22206}
Summary:
The implementation of H264AnnexBBufferHasVideoFormatDescription was
assuming that the SPS NALU is either the first NALU in the stream, or
the second one, in case an AUD NALU is present in the first location.
This change removes this assumption and instead searches for the SPS
NALU, failing only if we can't find one.
In addition, it cleans up some binary buffer manipulation code, using the
the parsed NALU indices we already have in AnnexBBufferReader instead.
Test Plan: Unit tests
Change-Id: Id9715aa1d751f0ba1a1992def2b690607896df56
bug: webrtc:8922
Change-Id: Id9715aa1d751f0ba1a1992def2b690607896df56
Reviewed-on: https://webrtc-review.googlesource.com/49982
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22205}
The AEC3 factory is now part of the WebRTC API.
Bug: webrtc:8844
Change-Id: If6f419b4ca0354e2d346c0e6474086e456ba747e
Reviewed-on: https://webrtc-review.googlesource.com/57141
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22204}
This is a fairly minimalistic string building class that
can be used instead of stringstream, which is discouraged
but tempting to use due to its convenient interface and
familiarity for anyone using our logging macros.
As a starter, I'm changing the string building code in
ReceiveStatisticsProxy and SendStatisticsProxy from using
stringstream and using SimpleStringBuilder instead.
In the case of SimpleStringBuilder, there's a single allocation,
it's done on the stack (fast), and minimal code is required for
each concatenation. The developer is responsible for ensuring
that the buffer size is adequate but the class won't overflow
the buffer. In dcheck-enabled builds, a check will go off if
we run out of buffer space.
As part of using SimpleStringBuilder for a small part of
rtc::LogMessage, a few more changes were made:
- SimpleStringBuilder is used for formatting errors instead of ostringstream.
- A new 'noop' state has been introduced for log messages that will be dropped.
- Use a static (singleton) noop ostream object for noop logging messages
instead of building up an actual ostringstream object that will be dropped.
- Add a LogMessageForTest class for better state inspection/testing.
- Fix benign bug in LogTest.Perf, change the test to not use File IO and
always enable it.
- Ensure that minimal work is done for noop messages.
- Remove dependency on rtc::Thread.
- Add tests for the extra_ field, correctly parsed paths and noop handling.
Bug: webrtc:8529, webrtc:4364, webrtc:8933
Change-Id: Ifa258c135135945e4560d9e24315f7d96f784acb
Reviewed-on: https://webrtc-review.googlesource.com/55520
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22203}
is configured.
An immediate (re)sorting of candidate paris reduces the latency of
network switching when it is necessary in ICE after (re)configuring the
network preference. A fix of comment and boilerplate code is also
included.
Bug: None
Change-Id: I8685235172d97193ffa6b53d4d2c7796fd01f861
Reviewed-on: https://webrtc-review.googlesource.com/57340
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22197}
This reverts commit 12dc1842d62ee8df1e462f9b6a617fef9ab8b3b7.
Reason for revert: Some internal tests keeps failing with this change.
Original change's description:
> Reland "Some cleanup for the logging code:"
>
> This is a reland of 9ecdcdf2b527bdb1d097782d92817c9866d6d18b.
>
> Original change's description:
> > Some cleanup for the logging code:
> >
> > * Only include 'tag' for Android. Before there was an
> > extra std::string variable per log statement for all
> > platforms.
> > * Remove unused logging macro for Windows and 'module' ctor argument.
> > * Move httpcommon code out of logging and to where it's used.
> >
> > Change-Id: I347805d3df2cee2840c0d2eef5bfefaff1cdbf37
> > Bug: webrtc:8928
> > Reviewed-on: https://webrtc-review.googlesource.com/57183
> > Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Commit-Queue: Tommi <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22184}
>
> Bug: webrtc:8928
> Change-Id: Id062a5b61917e66561f6c8441c2defd525e38f16
> Reviewed-on: https://webrtc-review.googlesource.com/57880
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Tommi <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22191}
TBR=kwiberg@webrtc.org,tommi@webrtc.org,jonasolsson@webrtc.org
Change-Id: I2b04e361459926a503552a0e1fcf3d1da3ddb643
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8928
Reviewed-on: https://webrtc-review.googlesource.com/58101
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22195}
This is a reland of 9ecdcdf2b527bdb1d097782d92817c9866d6d18b.
Original change's description:
> Some cleanup for the logging code:
>
> * Only include 'tag' for Android. Before there was an
> extra std::string variable per log statement for all
> platforms.
> * Remove unused logging macro for Windows and 'module' ctor argument.
> * Move httpcommon code out of logging and to where it's used.
>
> Change-Id: I347805d3df2cee2840c0d2eef5bfefaff1cdbf37
> Bug: webrtc:8928
> Reviewed-on: https://webrtc-review.googlesource.com/57183
> Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Tommi <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22184}
Bug: webrtc:8928
Change-Id: Id062a5b61917e66561f6c8441c2defd525e38f16
Reviewed-on: https://webrtc-review.googlesource.com/57880
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22191}
This reverts commit bd7b461f16b53363e2c510893d8d3aade5737f3a.
Reason for revert: Broke the internal project. The issue maybe related to the apm_debug_dump configuration.
Original change's description:
> Choose between APM-AGC-Limiter and Apm-AGC2-fixed-gain_controller.
>
> The webrtc::AudioMixer uses a limiter component. This CL changes the
> APM-AGC limiter to the APM-AGC2 limiter though a Chrome field trial.
>
> The new limiter has a float interface. Since we're moving to it, we
> now mix in floats as well. After this CL the mixer will support two
> limiters. The limiters have different interfaces and need different
> processing steps. Because of that, we make (rather big) changes to the
> control flow in FrameCombiner. For a short while, we will mix in
> deinterleaved floats when using any limiter.
>
> NOTRY=true
>
> Bug: webrtc:8925
> Change-Id: Ie296c2b0d94f3f0078811a2a58f6fbf0f3e6e4a8
> Reviewed-on: https://webrtc-review.googlesource.com/56141
> Commit-Queue: Alex Loiko <aleloi@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22185}
TBR=gustaf@webrtc.org,aleloi@webrtc.org
Change-Id: I3dd1a2b1fca32c4dd046e6fc325744079e3ac5ca
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8925
Reviewed-on: https://webrtc-review.googlesource.com/57940
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22189}
This reverts commit 9ecdcdf2b527bdb1d097782d92817c9866d6d18b.
Reason for revert: Breaks downstream project.
Original change's description:
> Some cleanup for the logging code:
>
> * Only include 'tag' for Android. Before there was an
> extra std::string variable per log statement for all
> platforms.
> * Remove unused logging macro for Windows and 'module' ctor argument.
> * Move httpcommon code out of logging and to where it's used.
>
> Change-Id: I347805d3df2cee2840c0d2eef5bfefaff1cdbf37
> Bug: webrtc:8928
> Reviewed-on: https://webrtc-review.googlesource.com/57183
> Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Tommi <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22184}
TBR=kwiberg@webrtc.org,tommi@webrtc.org,jonasolsson@webrtc.org
Change-Id: I37a13d766fbdee2adb7f45231cf8be6b2b456bec
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8928
Reviewed-on: https://webrtc-review.googlesource.com/57720
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22187}
The webrtc::AudioMixer uses a limiter component. This CL changes the
APM-AGC limiter to the APM-AGC2 limiter though a Chrome field trial.
The new limiter has a float interface. Since we're moving to it, we
now mix in floats as well. After this CL the mixer will support two
limiters. The limiters have different interfaces and need different
processing steps. Because of that, we make (rather big) changes to the
control flow in FrameCombiner. For a short while, we will mix in
deinterleaved floats when using any limiter.
NOTRY=true
Bug: webrtc:8925
Change-Id: Ie296c2b0d94f3f0078811a2a58f6fbf0f3e6e4a8
Reviewed-on: https://webrtc-review.googlesource.com/56141
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22185}
* Only include 'tag' for Android. Before there was an
extra std::string variable per log statement for all
platforms.
* Remove unused logging macro for Windows and 'module' ctor argument.
* Move httpcommon code out of logging and to where it's used.
Change-Id: I347805d3df2cee2840c0d2eef5bfefaff1cdbf37
Bug: webrtc:8928
Reviewed-on: https://webrtc-review.googlesource.com/57183
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22184}
The length of the fuzzer input can sometimes be really long (more than
1000000 bytes), and this take a very long time to execute. Typically,
the fuzzer times out instead. This change limits the used length of
the fuzzer to 200000 bytes.
NOTRY=TRUE
Bug: chromium:802149
Change-Id: Ia9d2f080602bba8ff70c5f0575bb9ecfa99c537c
Reviewed-on: https://webrtc-review.googlesource.com/57581
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22183}
The length of the fuzzer input can sometimes be really long (more than
600000 bytes), and this take a very long time to execute. Typically,
the fuzzer times out instead. This change limits the used length of
the fuzzer to 100000 bytes.
NOTRY=TRUE
Bug: chromium:802193
Change-Id: Id32174611fadb480f4e2c6b4f553a2ba0fa5b493
Reviewed-on: https://webrtc-review.googlesource.com/57580
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22182}
The length of the fuzzer input can sometimes be really long (more than
600000 bytes), and this take a very long time to execute. Typically,
the fuzzer times out instead. This change limits the used length of
the fuzzer to 100000 bytes.
NOTRY=TRUE
Bug: chromium:802245
Change-Id: Ibe02b6de932d900408f870d9ba440b7b8e08dc0e
Reviewed-on: https://webrtc-review.googlesource.com/57180
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22181}
In rare and pathological circumstances, it could happen that the input
length to the merge function is very short. This CL will avoid one of
the problems with out-of-bounds read that could result from this.
Bug: chromium:799499
Change-Id: I6bde105ae88f9d130764b6dfb3d25443d07e214b
Reviewed-on: https://webrtc-review.googlesource.com/57582
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22180}
There was an implementation, but it relied on SSLCertificate::GetChain,
which was never implemented. Except in the fake certificate classes
used by the stats collector tests, hence the tests were passing.
Instead of implementing GetChain, we decided (in
https://webrtc-review.googlesource.com/c/src/+/6500) to add
methods that return a SSLCertChain directly, since it results in a
somewhat cleaner object model.
So this CL switches everything to use the "chain" methods, and gets
rid of the obsolete methods and member variables.
Bug: webrtc:8920
Change-Id: Ie9d7d53654ba859535462521b54c788adec7badf
Reviewed-on: https://webrtc-review.googlesource.com/56961
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22177}
This CL fixes the bug that we must have some padding for H264 codec
when used in Multiplex encoder/decoder, otherwise H264 decoder will
crash.
And this CL fixes a bug that potential infinite loop exists in
MultiplexEncoderAdapter
Bug: webrtc:8921
Change-Id: I4124579c31ee69f72e29d118378aa1f8b3f05eb4
Reviewed-on: https://webrtc-review.googlesource.com/56960
Commit-Queue: Qiang Chen <qiangchen@chromium.org>
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22175}
Pass pointer to application_data from RtpPacketToSend arriving via RtpSender::SendToNetwork through to Transport::SendRtp, in PacketOptions.
Bug: webrtc:8906
Change-Id: Ie75013ed472710f4efcfbcc160e46a6119a1f41d
Reviewed-on: https://webrtc-review.googlesource.com/55600
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Dino Radaković <dinor@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22174}
Overriding implementations of VideoEncoder::GetScalingSettings that
want to enable quality scaling must now provide the thresholds.
Bug: webrtc:8830
Change-Id: I75c47cb56ac1b9cf77401684980b3167e485f51c
Reviewed-on: https://webrtc-review.googlesource.com/46622
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22172}
The bit rate target for ramp down in was set equal to the simulated
capacity. Expected behavior of an estimator is to achieve an estimate
near the true value but not always the exact value. Adding a margin
allows from noise in the measurement while still testing for the desired
behavor.
Bug: webrtc:8878
Change-Id: I18fb6c9704bf08e58ee08ce6c85abee2eaa08356
Reviewed-on: https://webrtc-review.googlesource.com/57080
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22171}
FALLBACK_SOFTWARE is now treated as a critical error and results in
immediate fallback to software coding if available. If ERROR is
returned, codec reset is attempted. If that fails, software fallback
is used.
Bug: b/73498933
Change-Id: I7fe163efd09e6f27c72491e9595954ddc59b1448
Reviewed-on: https://webrtc-review.googlesource.com/54901
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22169}
