Macro incorrectly displays DISABLED_ON_ANDROID in test names for
parameterized tests under --gtest_list_tests, causing tests to be
disabled on all platforms since they contain the DISABLED_ prefix rather
than their expanded variants.
This expands the macro variants to inline if they're disabled or not,
and removes building some tests under configurations where they should
fail, instead of building them but disabling them by default.
The change also removes gtest_disable.h as an unused include from many
other files.
BUG=webrtc:5387, webrtc:5400
R=kjellander@webrtc.org, phoglund@webrtc.orgTBR=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1547343002 .
Cr-Commit-Position: refs/heads/master@{#11150}
Renames CreateFakeNativeHandleFrame to FakeNativeHandle::CreateFrame and
moves into test.gyp target 'fake_video_frames' which contains previous
frame_generator target.
Removes unused warnings from includers of
webrtc/test/fake_texture_frame.h which did not use the function above.
BUG=webrtc:5398
R=kjellander@webrtc.orgTBR=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1554223002 .
Cr-Commit-Position: refs/heads/master@{#11149}
public function RtpHeaderParser::Parse with old signature restored as deprecated.
BUG=webrtc:5277
TBR=åsapersson
NOTRY=True
Review URL: https://codereview.webrtc.org/1550283002
Cr-Commit-Position: refs/heads/master@{#11135}
Opus has become the mostly used codec in WebRTC. There, however, is no bit exactness test for Opus decoding in NetEq.
The new RTP file is generated by the following steps:
1. Encode a clean RTP file with Opus
RTPencode resources/audio_coding/speech_mono_32_48kHz.pcm neteq_opus_raw.rtp 960 opus 1
2. Adding jitter to the clean RTP file
RTPjitter neteq_opus_raw.rtp jitter.dat neteq_opus.rtp
(Note: jitter.dat does not exist in WebRTC resources folder. Check the source code for RTPjitter to know how to define such a file.)
BUG=webrtc:3987
TEST=observed Opus normal decoding and FEC decoding were used, listened to the reference output.
Review URL: https://codereview.webrtc.org/1515113002
Cr-Commit-Position: refs/heads/master@{#11113}
PrepareReportBlock and AddReportBlock private functions merged:
PrepareReportBlock moved report block from statistic to temporary structure
AddReportBlock copied that temporary structure into temporary map right after.
Thanks to rtcp packet classes that temporary structure is now unneccesary.
BUG=webrtc:5260
R=åsapersson
Review URL: https://codereview.webrtc.org/1538833002
Cr-Commit-Position: refs/heads/master@{#11112}
This implementation will be replaced by a faster one and sparse will be removed.
BUG=webrtc:5283
Review URL: https://codereview.webrtc.org/1530913002
Cr-Commit-Position: refs/heads/master@{#11099}
Reason for revert:
Compile error on Android needs to be fixed before relanding.
Original issue's description:
> Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12afc8cc29fe9c59dea432b697b871a87.
>
> The revert of the original CL was commit 36d4c545007129446e551c45c17b25377dce89a4.
> Original review: https://codereview.webrtc.org/1413483003/
>
> The original CL changes a function on audio_processing.h that is used by Chrome, this CL adds back the old function.
>
> NOTRY=true
> TBR=glaznev@webrtc.org, henrik.lundin@webrtc.org, solenberg@google.com, henrikg@webrtc.org, perkj@webrtc.org
> BUG=webrtc:4741
>
> Committed: https://crrev.com/f4f5cb09277d5ef6aeac8341e5f54a055867803a
> Cr-Commit-Position: refs/heads/master@{#11093}
TBR=glaznev@webrtc.org,henrik.lundin@webrtc.org,solenberg@google.com,henrikg@webrtc.org,perkj@webrtc.org,kwiberg@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4741
Review URL: https://codereview.webrtc.org/1537213002
Cr-Commit-Position: refs/heads/master@{#11094}
Currently, FFT is performance when AEC buffers farend signal. This has some drawbacks
1. memory inefficiency: two ring buffers are needed;
2. computation inefficiency: if ringbuffer gets wrapped around, some FFT computation will be wasted;
3. accessibility: the main AEC function looses accessibility to the time-domain signal.
Therefore, this CL tries to buffer time domain data, which is buffered any way if a debugging macro is defined, and calculate the FFTs where they are actually used.
BUG=
Review URL: https://codereview.webrtc.org/1512573003
Cr-Commit-Position: refs/heads/master@{#11091}
Reason for revert:
Breaks Chrome-FYI bots because of a change in the StartDebugRecording function in audio_processing.h, that is called from Chrome.
Original issue's description:
> Added option to specify a maximum file size when recording an AEC dump.
>
> For applications with a strict filesize limit for debug files,
> I added an option to specify a maximum filesize for AEC dumps. An
> existing unit test is extended to check that the feature works as
> advertised.
>
> BUG=webrtc:4741
> TBR=glaznev@webrtc.org
>
> Committed: https://crrev.com/ae2c5ad12afc8cc29fe9c59dea432b697b871a87
> Cr-Commit-Position: refs/heads/master@{#11081}
TBR=pthatcher@webrtc.org,henrik.lundin@webrtc.org,henrikg@webrtc.org,solenberg@webrtc.org,andrew@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org,glaznev@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4741
Review URL: https://codereview.webrtc.org/1533913004
Cr-Commit-Position: refs/heads/master@{#11087}
This should solve a problem discovered when converting from GYP to
other project formats, where the source files weren't included correctly
for each platform.
Two other targets in WebRTC have similar source files, which are correctly
generated for each platform:
* video_render_module_internal_impl
* video_capture_module_internal_impl
They both list the sources as it's changed to in this CL.
NOTRY=True
Review URL: https://codereview.webrtc.org/1536923003
Cr-Commit-Position: refs/heads/master@{#11083}
This removes a dependency on Chromium's build/build_config.h
(which is not allowed).
The added defines are identical to the ones in build/build_config.h.
NOTRY=True
Review URL: https://codereview.webrtc.org/1532333002
Cr-Commit-Position: refs/heads/master@{#11082}
For applications with a strict filesize limit for debug files,
I added an option to specify a maximum filesize for AEC dumps. An
existing unit test is extended to check that the feature works as
advertised.
BUG=webrtc:4741
TBR=glaznev@webrtc.org
Review URL: https://codereview.webrtc.org/1413483003
Cr-Commit-Position: refs/heads/master@{#11081}
At speed 8, vp9 on ARM is currently ~2x times slower than vp8 on ARM (speed -12).
Update some parameters in videoprocessor_integrationtest.cc
to make tests pass on android (which uses the new speed setting).
TBR=stefan@webrtc.org
BUG=
Review URL: https://codereview.webrtc.org/1526973004 .
Cr-Commit-Position: refs/heads/master@{#11072}
We can now use std::move instead!
This CL leaves the Pass methods in place; a follow-up CL will add deprecation annotations to them.
Review URL: https://codereview.webrtc.org/1460043002
Cr-Commit-Position: refs/heads/master@{#11064}
-Extended the InverseFft function to be more generally
applicable.
-Included the previous external extra scaling into the
preexisting InverseFft call.
-Moved the updating of aec->delayEstCtr to where it is
actually used.
-Refactored the output production and comfort noise
addition using the InverseFft function.
-Removed the if-statements checking the value of the
constant flagHbandCn as any value different from 1 would
crash the program. Also removed the constant
The changes have been tested for bitexactness.
BUG=webrtc:5201
Review URL: https://codereview.webrtc.org/1492343002
Cr-Commit-Position: refs/heads/master@{#11054}
We should only account for audio packets in the pacer budget if we also
are allocating bandwidth for the audio streams.
BUG=chromium:567659,webrtc:5263
R=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1524763002 .
Cr-Commit-Position: refs/heads/master@{#11053}
NetEqNetworkStatistics has been updated some time ago. A bit exactness test in neteq unittests is still using the old NetEqNetworkStatistics.
New neteq4_network_stats.dat generated by running TestBitExactness with flag "genref"
BUG=
Review URL: https://codereview.webrtc.org/1522103002
Cr-Commit-Position: refs/heads/master@{#11052}
All encoders already handle the "Opus-specific" requests sanely (by
failing nicely), so we don't need extra checks to protect them.
BUG=webrtc:5028
Review URL: https://codereview.webrtc.org/1527453005
Cr-Commit-Position: refs/heads/master@{#11051}
-Changed the type for the frequency estimate of the comfort noise for the
higher band to be a two dimensional float array instead of a complex_t array.
This makes sense since all the other frequency estimate (apart from the
coherence) use this format and doing this change allows bundling the
IFFT operations into using the InverseFFT method.
-Moved the memset of the frequency estimate of the comfort noise to where it is used and made it conditional so that it is only performed when used.
-Harmonized the if-statements for when the frequency estimate of the comfort noise is computed in the different optimized ComfortNoise computation methods.
The changes have been tested for bitexactness.
BUG=webrtc:5201
Review URL: https://codereview.webrtc.org/1494133002
Cr-Commit-Position: refs/heads/master@{#11050}
So that the two of them sit next to each other at the top level of
AudioCodingModuleImpl. CodecManager now manages the specifications for
Rent-A-Codec, rather than managing encoders directly.
BUG=webrtc:5028
Review URL: https://codereview.webrtc.org/1520283006
Cr-Commit-Position: refs/heads/master@{#11048}
using the wrong sample rate for the render signal.
The CL is basically a partial revert of the related changes done on
output_mixer.cc in the CL https://codereview.webrtc.org/1234463003.
The CL also reverts the removal of the input_sample_rate_hz() method
that was removed as part of the CL
https://codereview.webrtc.org/1379123002 (as it was at that point
no longer used).
It should be noted that this CL turns off the effect of the
IntelligibilityEnhancer when the AudioFrame AudioProcessing APIs are
used. While it may be possible to solve that by adding upsampling after
the API call, that approach was discarded due to that:
-That would add extra processing in the echo path, leading to possible
AEC performance reduction.
-That would add extra complexity for the mobile case.
-That would only patch the intelligibility enhancer operation as the
proper way to do such an operation is within APM.
-The intelligibility enhancer is not active by default anywhere.
BUG=webrtc:5237
Review URL: https://codereview.webrtc.org/1525173002
Cr-Commit-Position: refs/heads/master@{#11045}
