These are the changes made to WebRtcVoiceEngine and surrounding
code. It still contains some things that are inelegant, like how
AudioCodecSpec and AudioFormatInfo is ferried around in
SendCodecSpec. This should probably be resolved before landing.
There are also a few test still that are disabled. They should be
removed or fixed, as the case may be.
I've put this CL up to get a better overview of the changes made and
how reviewable they are.
BUG=webrtc:5806
Review-Url: https://codereview.webrtc.org/2705093002
Cr-Commit-Position: refs/heads/master@{#17904}
With the upcoming changes to WebRtcVoiceEngine due to injectable audio
encoders, codec names will no longer be normalized against the ACM's
codec database. This not only breaks a couple of downstream tests
(which I've written fixes for) but also runs the risk of breaking
other external usage, like SDP mangling etc.
Eventually, we should change to using the spelling from the relevant
RFCs, although not as part of this other, large change.
BUG=webrtc:5806
Review-Url: https://codereview.webrtc.org/2845603002
Cr-Commit-Position: refs/heads/master@{#17903}
AudioDecoder and AudioDecoderFactory are in webrtc/api/ now, so move
their mocks to someplace central where tests from all over WebRTC are
allowed to #include them.
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2798063004
Cr-Commit-Position: refs/heads/master@{#17619}
Reason for revert:
Trying to re-land after solving some related issues.
There are no changes compared to the original CL.
Original issue's description:
> Revert of Loosening the coupling between WebRTC and //third_party/protobuf (patchset #16 id:300001 of https://codereview.webrtc.org/2747863003/ )
>
> Reason for revert:
> I will try to reland next week because it is causing some problems.
>
> Original issue's description:
> > To accommodate some downstream WebRTC users we need to loosen
> > the coupling between our code and the //third_party/protobuf.
> >
> > This includes using typedefs to define strings instead of
> > assuming std::string.
> >
> > After this refactoring it will be possible to link with other
> > protobuf implementations than the current one.
> >
> > We moved the PRESUBMIT check to another CL [1]. The goal of this
> > presubmit is to avoid the direct usage of google::protobuf outside
> > of the webrtc/base/protobuf_utils.h header file.
> >
> > [1] - https://codereview.webrtc.org/2753823003/
> >
> > BUG=webrtc:7340
> > NOTRY=True
> >
> > Review-Url: https://codereview.webrtc.org/2747863003
> > Cr-Commit-Position: refs/heads/master@{#17466}
> > Committed: 16ab93b952
>
> TBR=kjellander@webrtc.org,henrik.lundin@webrtc.org,kwiberg@webrtc.org,michaelt@webrtc.org,peah@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:7340
>
> Review-Url: https://codereview.webrtc.org/2786363002
> Cr-Commit-Position: refs/heads/master@{#17483}
> Committed: d00aad5eb2TBR=kjellander@webrtc.org,henrik.lundin@webrtc.org,kwiberg@webrtc.org,michaelt@webrtc.org,peah@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7340
NOTRY=True
Review-Url: https://codereview.webrtc.org/2791963003
Cr-Commit-Position: refs/heads/master@{#17584}
This CL contains all the changes made to audio_coding while making
audio encoders injectable. Apart from some small changes to
webrtcvoiceengine, nothing here is hooked up to the outside
world. Those changes will be added to a follow-up CL.
BUG=webrtc:5806
Review-Url: https://codereview.webrtc.org/2695243005
Cr-Commit-Position: refs/heads/master@{#17569}
Because on gcc, cast to void doesn't silence the warning. See
https://gcc.gnu.org/bugzilla/show_bug.cgi?id=66425
Also add an RTC_ prefix to the macro instead of only defining it if it
wasn't already defined, to ensure that we always get our own version.
BUG=none
Review-Url: https://codereview.webrtc.org/2797983003
Cr-Commit-Position: refs/heads/master@{#17563}
Reason for revert:
I will try to reland next week because it is causing some problems.
Original issue's description:
> To accommodate some downstream WebRTC users we need to loosen
> the coupling between our code and the //third_party/protobuf.
>
> This includes using typedefs to define strings instead of
> assuming std::string.
>
> After this refactoring it will be possible to link with other
> protobuf implementations than the current one.
>
> We moved the PRESUBMIT check to another CL [1]. The goal of this
> presubmit is to avoid the direct usage of google::protobuf outside
> of the webrtc/base/protobuf_utils.h header file.
>
> [1] - https://codereview.webrtc.org/2753823003/
>
> BUG=webrtc:7340
> NOTRY=True
>
> Review-Url: https://codereview.webrtc.org/2747863003
> Cr-Commit-Position: refs/heads/master@{#17466}
> Committed: 16ab93b952TBR=kjellander@webrtc.org,henrik.lundin@webrtc.org,kwiberg@webrtc.org,michaelt@webrtc.org,peah@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7340
Review-Url: https://codereview.webrtc.org/2786363002
Cr-Commit-Position: refs/heads/master@{#17483}
the coupling between our code and the //third_party/protobuf.
This includes using typedefs to define strings instead of
assuming std::string.
After this refactoring it will be possible to link with other
protobuf implementations than the current one.
We moved the PRESUBMIT check to another CL [1]. The goal of this
presubmit is to avoid the direct usage of google::protobuf outside
of the webrtc/base/protobuf_utils.h header file.
[1] - https://codereview.webrtc.org/2753823003/
BUG=webrtc:7340
NOTRY=True
Review-Url: https://codereview.webrtc.org/2747863003
Cr-Commit-Position: refs/heads/master@{#17466}
The command
tools/checklicenses/checklicenses.py --ignore-suppressions ./webrtc
previously produced this output:
'webrtc/modules/audio_processing/test/apmtest.m' has non-whitelisted license 'UNKNOWN'
'webrtc/modules/audio_processing/transient/test/readDetection.m' has non-whitelisted license 'UNKNOWN'
'webrtc/modules/audio_processing/transient/test/readPCM.m' has non-whitelisted license 'UNKNOWN'
...
This CL adds the WebRTC licence with appropriate year to all our
MATLAB files. All these files were contributed by WebRTC project
members hlundin@, pbos@, niklase@.
BUG=chromium:98592
NOTRY=True
Review-Url: https://codereview.webrtc.org/2781663005
Cr-Commit-Position: refs/heads/master@{#17428}
This is part of a series of CLs. Next CLs:
1. CL for RPLR-based FecController
2. CL for allowing experiment-driven configuration of the above (through both field-trials and protobuf)
BUG=webrtc:7058
Review-Url: https://codereview.webrtc.org/2661043003
Cr-Commit-Position: refs/heads/master@{#17368}
A DCHECK added in a recent bugfix, which asserted that a signed 64->32
bit cast did not overflow, has been found to not always pass. We fix
this by saturating.
BUG=chromium:693868
Review-Url: https://codereview.webrtc.org/2746903002
Cr-Commit-Position: refs/heads/master@{#17209}
In an attempt to help Kenny locate the evil Dr. Deo.
BUG=webrtc:7307
Review-Url: https://codereview.webrtc.org/2732193006
Cr-Commit-Position: refs/heads/master@{#17135}
New Opus version starts to support 120ms frame lengths. AudioEncoderOpusTest had an unnecessary check on the available frame lengths.
It is removed in this CL.
BUG=b/35415318
Review-Url: https://codereview.webrtc.org/2731583003
Cr-Commit-Position: refs/heads/master@{#17011}
In short, what I did was to
* Remove acm_common_defs.h (the stuff in it was used only by
acm_codec_database.cc).
* Move audio_coding_module_typedefs.h to a new build target.
* Move the NetEqDecoder enum (and the associated
NetEqDecoderToSdpAudioFormat function) to a new file in a new
build target.
BUG=webrtc:7243, webrtc:7244
Review-Url: https://codereview.webrtc.org/2723253005
Cr-Commit-Position: refs/heads/master@{#17005}
A fuzzer run caused the operands of this multiplication to be 512 and
5000000, resulting in a product about 20% too large for int32_t. So
change this from a 16x32->32 to a 16x32->64 multiplication. Since we
right shift by 2 at the end, the end result will still fit in int32_t.
I also had to fix a few follow-on add/sub overflows found by the same
fuzzer input once the multiplication was fixed. I chose to saturate
these, since it wasn't just an intermediate value that overflowed.
BUG=chromium:693868
Review-Url: https://codereview.webrtc.org/2729573002
Cr-Commit-Position: refs/heads/master@{#17003}
It's the faster, less strict cousin of checked_cast.
BUG=none
Review-Url: https://codereview.webrtc.org/2714063002
Cr-Commit-Position: refs/heads/master@{#16958}
It's currently only used to ensure transport-cc is enabled for the format in question. It might be used to toggle more things in future.
BUG=webrtc:5806
Review-Url: https://codereview.webrtc.org/2669583002
Cr-Commit-Position: refs/heads/master@{#16514}
BUG=webrtc:7097
TEST=Set "ptime=120", try WebRTC calls over custom build Chromium with and without Opus 120ms. Try both Chromium w <-> Chromium w and Chromium w <-> Chromium w/o
Review-Url: https://codereview.webrtc.org/2668633004
Cr-Commit-Position: refs/heads/master@{#16408}
We read past the end of the initialized part of the buffer, seemingly
on purpose (no one knows the details of this code anymore). The right
thing to do is probably to zero that part of the buffer.
(The *right* right thing to would be to rewrite this so that it was
easier to see what data was supposed to be where when, but
priorities...)
BUG=chromium:683040
Review-Url: https://codereview.webrtc.org/2659383002
Cr-Commit-Position: refs/heads/master@{#16365}
The test verifies that the hysteresis window in the configuration works
as intended.
BUG=webrtc:6708
Review-Url: https://codereview.webrtc.org/2594563002
Cr-Commit-Position: refs/heads/master@{#15700}
This effectively reverts commit c3e1cabc696240e4b5a128653264785292878205
(https://codereview.webrtc.org/2589703002/).
The reason the test was failing before was missing resource
dependencies in the GN file. This is now fixed.
Furthermore, the test did not trigger the complexity adaptation that
it was supposed to test, since the hysteresis window of the bitrate
was not taken into account. This is also fixed.
Finally, a percent label was added to a printout, to match the same
printout in the other test.
BUG=webrtc:6708
Review-Url: https://codereview.webrtc.org/2580383002
Cr-Commit-Position: refs/heads/master@{#15679}
Reason for revert:
This fails on perf bots.
See for instance https://uberchromegw.corp.google.com/i/client.webrtc.perf/builders/Android32%20Tests%20%28K%20Nexus5%29/builds/105/steps/webrtc_perf_tests/logs/stdio:
I 643.319s run_tests_on_device(03832b22f0e944a0) Note: Google Test filter = AudioEncoderOpusComplexityAdaptationTest.AdaptationOff
I 643.319s run_tests_on_device(03832b22f0e944a0) [==========] Running 1 test from 1 test case.
I 643.319s run_tests_on_device(03832b22f0e944a0) [----------] Global test environment set-up.
I 643.319s run_tests_on_device(03832b22f0e944a0) [----------] 1 test from AudioEncoderOpusComplexityAdaptationTest
I 643.319s run_tests_on_device(03832b22f0e944a0) [ RUN ] AudioEncoderOpusComplexityAdaptationTest.AdaptationOff
I 643.319s run_tests_on_device(03832b22f0e944a0) ../../webrtc/modules/audio_coding/codecs/opus/opus_complexity_unittest.cc:36: Failure
I 643.319s run_tests_on_device(03832b22f0e944a0) Value of: audio_loop.Init(kInputFileName, kMaxLoopLengthSamples, kInputBlockSizeSamples)
I 643.320s run_tests_on_device(03832b22f0e944a0) Actual: false
I 643.320s run_tests_on_device(03832b22f0e944a0) Expected: true
I 643.320s run_tests_on_device(03832b22f0e944a0) [ CRASHED ]
I 643.405s run_tests_on_device(0384888cf0e945d1) >>ScopedMainEntryLogger
I 643.405s run_tests_on_device(0384888cf0e945d1) Note: Google Test filter = AudioEncoderOpusComplexityAdaptationTest.AdaptationOn
I 643.405s run_tests_on_device(0384888cf0e945d1) [==========] Running 1 test from 1 test case.
I 643.405s run_tests_on_device(0384888cf0e945d1) [----------] Global test environment set-up.
I 643.405s run_tests_on_device(0384888cf0e945d1) [----------] 1 test from AudioEncoderOpusComplexityAdaptationTest
I 643.405s run_tests_on_device(0384888cf0e945d1) [ RUN ] AudioEncoderOpusComplexityAdaptationTest.AdaptationOn
I 643.405s run_tests_on_device(0384888cf0e945d1) ../../webrtc/modules/audio_coding/codecs/opus/opus_complexity_unittest.cc:36: Failure
I 643.405s run_tests_on_device(0384888cf0e945d1) Value of: audio_loop.Init(kInputFileName, kMaxLoopLengthSamples, kInputBlockSizeSamples)
I 643.405s run_tests_on_device(0384888cf0e945d1) Actual: false
I 643.405s run_tests_on_device(0384888cf0e945d1) Expected: true
I 643.405s run_tests_on_device(0384888cf0e945d1) [ CRASHED ]
Original issue's description:
> Re-enable Opus complexity tests on Android
>
> BUG=webrtc:6708
> TBR=minyue@webrtc.org
>
> Review-Url: https://codereview.webrtc.org/2589673002
> Cr-Commit-Position: refs/heads/master@{#15667}
> Committed: 14479ddfd4
TBR=
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6708
Review-Url: https://codereview.webrtc.org/2589703002
Cr-Commit-Position: refs/heads/master@{#15668}
