This CL includes the patch from oprypin@webrtc.org, which is also applied
to the MIPS code (also affected), and the protobuf for ApmTest.Process
(audio_processing_unittest.cc), which used when WEBRTC_AUDIOPROC_FIXED_PROFILE
is set.
This change has been tested on mobile platforms.
Bug: webrtc:8200
Change-Id: Ic50a5ab57c16551397756b1fb473e1067b8e7ece
Reviewed-on: https://webrtc-review.googlesource.com/10811
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20394}
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
Reason for revert:
Resetting the estimate means that we need to start gathering data from scratch again. The combination of
1) DelayBasedEstimator not reacting to overuse unless there is a valid estimate of the acknowledged bitrate, and
2) AcknowledgedBitrateEstimator needing a significant amount of time/data to obtain an provide an estimate
causes poor performance in simulations/tests. It is not clear whether this will affect real networks negatively, but I suggest reverting this to be on the safe side.
See also https://bugs.chromium.org/p/webrtc/issues/detail?id=7884
Original issue's description:
> Test and fix for huge bwe drop after alr state.
>
> BUG=webrtc:7746
>
> Review-Url: https://codereview.webrtc.org/2931873002
> Cr-Commit-Position: refs/heads/master@{#18692}
> Committed: 37aa8ba616TBR=solenberg@webrtc.org,kwiberg@webrtc.org,minyue@webrtc.org,holmer@chromium.org,philipel@webrtc.org,oprypin@webrtc.org,holmer@google.com,stefan@webrtc.org,tschumim@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7746
Review-Url: https://codereview.webrtc.org/2964213002
Cr-Commit-Position: refs/heads/master@{#18866}
This CL corrects the initialization of the AEC3, as well
as for the other submodules in the whole audio processing module
in the sense that it properly update the submodule states also
for the case when reinitialization is trigger from the render
side of the audio processing module.
Bug: chromium:736889,webrtc:7879
Change-Id: I423e963835d0c3227caa8e186b29031bcb912515
Reviewed-on: https://chromium-review.googlesource.com/549315
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18784}
Publicly available dataset of conversational speech audio recordings.
This CL includes the following:
- README.md: dataset description file, it also includes the scripts
- *.wav.sha1: hash files for each audio track in the dataset
The overall size of the wav files is ~36MB.
The primary intended use of this dataset is in combination with the conversational speech tool (see https://chromium.googlesource.com/external/webrtc/+/master/webrtc/modules/audio_processing/test/conversational_speech/), using which longer recordings with custom turn switch timing can be created.
BUG=webrtc:7218
Review-Url: https://codereview.webrtc.org/2869833002
Cr-Commit-Position: refs/heads/master@{#18068}
Reason for revert:
The blocker in upstreaming has been removed.
Original issue's description:
> Revert of Reland "Add first part of the network_tester functionality" (patchset #3 id:40001 of https://codereview.chromium.org/2808203003/ )
>
> Reason for revert:
> Break downstream bots.
>
> Original issue's description:
> > Reland "Add first part of the network_tester functionality"
> >
> > BUG=webrtc:7426
> >
> > Review-Url: https://codereview.webrtc.org/2808203003
> > Cr-Commit-Position: refs/heads/master@{#17666}
> > Committed: 1c223b2f75
>
> TBR=stefan@webrtc.org,minyue@webrtc.org,nisse@webrtc.org,terelius@webrtc.org,michaelt@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7426
>
> Review-Url: https://codereview.webrtc.org/2813193002
> Cr-Commit-Position: refs/heads/master@{#17672}
> Committed: e5fd38989dTBR=stefan@webrtc.org,nisse@webrtc.org,terelius@webrtc.org,michaelt@webrtc.org,philipel@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7426
Review-Url: https://codereview.webrtc.org/2811253005
Cr-Commit-Position: refs/heads/master@{#17688}
Reason for revert:
Break downstream bots.
Original issue's description:
> Reland "Add first part of the network_tester functionality"
>
> BUG=webrtc:7426
>
> Review-Url: https://codereview.webrtc.org/2808203003
> Cr-Commit-Position: refs/heads/master@{#17666}
> Committed: 1c223b2f75TBR=stefan@webrtc.org,minyue@webrtc.org,nisse@webrtc.org,terelius@webrtc.org,michaelt@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7426
Review-Url: https://codereview.webrtc.org/2813193002
Cr-Commit-Position: refs/heads/master@{#17672}
Reason for revert:
Tasn test failure.
Original issue's description:
> Add first part of the network_tester functionality.
>
> BUG=webrtc:7426
>
> Review-Url: https://codereview.webrtc.org/2779233002
> Cr-Commit-Position: refs/heads/master@{#17635}
> Committed: 333d0ff631TBR=stefan@webrtc.org,minyue@webrtc.org,nisse@webrtc.org,terelius@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7426
Review-Url: https://codereview.webrtc.org/2800403003
Cr-Commit-Position: refs/heads/master@{#17636}
This CL adds the ability to _create_ HW codecs (Android and iOS) in the
VideoProcessor integration tests. Since the VideoProcessor class is not thread
safe yet, this CL does not add the ability to _use_ HW codecs in the tests. A
follow-up CL is planned that will add this ability.
This CL further adds a separate build target which is used to separate the
"plot" versions of the integration tests from the "correctness" versions. The
former will be run manually on devices, whereas the latter are used on the
trybots/buildbots to find regressions in the SW codecs. The underlying test
is the same, however.
BUG=webrtc:6634
Review-Url: https://codereview.webrtc.org/2695653002
Cr-Commit-Position: refs/heads/master@{#16716}
The test case is put inside a new test target. That test target will be started from a test script to asses video quality.
BUG=webrtc:6545
Review-Url: https://codereview.webrtc.org/2585813002
Cr-Commit-Position: refs/heads/master@{#16088}
This file shows up with whitespace changes when importing the code
into a downstream project
BUG=None
NOTRY=True
NOPRESUBMIT=True
R=mbonadei@webrtc.org
Review-Url: https://codereview.webrtc.org/2592913003 .
Cr-Commit-Position: refs/heads/master@{#15758}
Create a .gitignore file in resources/ that is responsible for
preventing the downloaded binary files from being added to source control.
BUG=None
NOTRY=True
Review-Url: https://codereview.webrtc.org/2587163002
Cr-Commit-Position: refs/heads/master@{#15691}
Issue: This API was calculating the file_header and frame_header offset only for the first frame which is not the right logic. We need to skip the file and frame header every time we extract a new frame.
Also added a unit test case which compares the extracted frame with the frame stored in text file.
NOPRESUBMIT=true
NOTRY=true
BUG=webrtc:6761
Review-Url: https://codereview.webrtc.org/2532963002
Cr-Commit-Position: refs/heads/master@{#15288}
Reason for revert:
Breaking some trybots due to memory error.
Original issue's description:
> Fixed bug in ExtractFrameFromY4mFile API which was not extracting the frames correctly.
>
> Issue: This API was calculating the file_header and frame_header offset only for the first frame which is not the right logic. We need to skip the file and frame header every time we extract a new frame.
>
> Also added a unit test case which compares the extracted frame with the frame stored in text file.
>
> BUG=webrtc:6761
>
> NOPRESUBMIT=true
> NOTRY=true
>
> Committed: https://crrev.com/b7636b4656d7f8c368963f2256dc2ef7b7ba89c8
> Cr-Commit-Position: refs/heads/master@{#15260}
TBR=phoglund@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6761
Review-Url: https://codereview.webrtc.org/2535783002
Cr-Commit-Position: refs/heads/master@{#15262}
Issue: This API was calculating the file_header and frame_header offset only for the first frame which is not the right logic. We need to skip the file and frame header every time we extract a new frame.
Also added a unit test case which compares the extracted frame with the frame stored in text file.
BUG=webrtc:6761
NOPRESUBMIT=true
NOTRY=true
Review-Url: https://codereview.webrtc.org/2529923002
Cr-Commit-Position: refs/heads/master@{#15260}
Move the resources to //resources and upload them to Google Storage.
BUG=webrtc:6727
Review-Url: https://codereview.webrtc.org/2508943004
Cr-Commit-Position: refs/heads/master@{#15152}
Also update existing perf tests to use send side bwe.
BUG=webrtc:4604, chromium:522001
Review-Url: https://codereview.webrtc.org/2227733004
Cr-Commit-Position: refs/heads/master@{#13726}
NetEq bitexactness test depended on reference files which differs from platform to platform. This makes it very hard to update Neteq.
New method maintains the ability to save output into files. But it verifies the checksum only. With this, when bitexactness test fails, we can still check closely to the output file if need, but the test becomes much easier to modify.
BUG=
Review-Url: https://codereview.webrtc.org/1928923002
Cr-Commit-Position: refs/heads/master@{#12567}
Reason for revert:
There seems an error made in this patch.
Hi Henrik,
I think the bit shift returned by CrossCorrelationWithAutoShift may be wrongly used by DotProduct.
We'd better revert this CL.
Doing another fix (and future fixes) will be paintful. I will work on a easy-to-modify bitexactness test first.
Original issue's description:
> Avoiding overflow in cross correlation in NetEq.
>
> BUG=
TBR=henrik.lundin@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=
Review-Url: https://codereview.webrtc.org/1925053002
Cr-Commit-Position: refs/heads/master@{#12543}
In some rare occations (very low energy signal), a shift value happened
to be negative. This is now fixed by using the WEBRTC_SPL_SHIFT_W32,
which in essence checks the sign of the number of shifts and performs a
right or left shift accordingly.
The fix reverts to how the code was written in old NetEq; see
4d363ae305/webrtc/modules/audio_coding/neteq/normal.c (165).
BUG=webrtc:5490
Review URL: https://codereview.webrtc.org/1675293002
Cr-Commit-Position: refs/heads/master@{#11546}
I removed the 'libjingle' target in talk/libjingle.gyp and replaced
all users of it with base/base.gyp:rtc_base. It seems the jsoncpp
and expat dependencies were not used by it's previous references.
The files in talk/media/testdata were uploaded to Google Storage and
added .sha1 files in resources/media instead of simply moving them.
The previously disabled warnings that were inherited from
talk/build/common.gypi are now replaced by target-specific disabling
of only the failing warnings. Additional disabling was needed since the stricter
compilation warnings that applies to code in webrtc/.
License headers will be updated in a follow-up CL in order to not
break Git history.
Other modifications:
* Updated the header guards.
* Sorted the includes using chromium/src/tools/sort-headers.py
except for these files:
talk/app/webrtc/peerconnectionendtoend_unittest.cc
talk/app/webrtc/java/jni/androidmediadecoder_jni.cc
talk/app/webrtc/java/jni/androidmediaencoder_jni.cc
webrtc/media/devices/win32devicemanager.cc.
* Unused GYP reference to libjingle_tests_additional_deps was removed.
* Removed duplicated GYP entries of
webrtc/base/testutils.cc
webrtc/base/testutils.h
The HAVE_WEBRTC_VIDEO and HAVE_WEBRTC_VOICE defines were used by only talk/media,
so they were moved to the media.gyp.
I also checked that none of
EXPAT_RELATIVE_PATH,
FEATURE_ENABLE_VOICEMAIL,
GTEST_RELATIVE_PATH,
JSONCPP_RELATIVE_PATH,
LOGGING=1,
SRTP_RELATIVE_PATH,
FEATURE_ENABLE_SSL,
FEATURE_ENABLE_VOICEMAIL,
FEATURE_ENABLE_PSTN,
HAVE_SCTP,
HAVE_SRTP,
are used by the talk/media code.
For Chromium, the following changes will need to be applied to the roll CL that updates the
DEPS for WebRTC and libjingle: https://codereview.chromium.org/1604303002/
BUG=webrtc:5420
NOPRESUBMIT=True
TBR=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1587193006
Cr-Commit-Position: refs/heads/master@{#11495}
Several unittests were disabled on android, this CL will reenable them. One of
the tests was accidentally disabled on all platforms, and now no longer gives a
bitexact result.
BUG=webrtc:3343,webrtc:5349
Review URL: https://codereview.webrtc.org/1532903002
Cr-Commit-Position: refs/heads/master@{#11323}
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}
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}
Adding original_length to the Packet struct. This is populated with
the plen value from the RTP dump file. In the case of reading a
pcap file, original_length will be equal to length.
Also increasing the maximum packet size to 3500 bytes. This is to
accomodate some test files that contain PCM16b audio encoding.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28609004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7333 4adac7df-926f-26a2-2b94-8c16560cd09d
This CL turns the background noise generation in NetEq off by default. The noise generation used to kick in during long-duration packet losses, when there was no point in extrapolating the latest audio any longer. However, this sometimes produces annoying noise in situations where silence would have been preferable.
With this change, a long packet-loss concealment will be faded out to zeros instead of a low noise.
Reference files are updated where needed.
BUG=3519
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20109004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6882 4adac7df-926f-26a2-2b94-8c16560cd09d
A new reference file (neteq4_universal_ref_win_64.pcm) was generated and
uploaded.
Also removing the old hack to have different reference files
for different version of Visual Studio. The test is now only supporting
VS 2012 and later (_MSC_VER >= 1700). This makes the windows 32-bit
output identical to the generic reference file
(neteq4_universal_ref.pcm), so the specialized one
(neteq4_universal_ref_win_32.pcm) could have been removed. However,
since the resources sync mechanism does not include removing of old
files, a client could pick up the old reference and fail. Therefore,
this cl also updates neteq4_universal_ref_win_32.pcm to be identical to
neteq4_universal_ref.pcm.
BUG=1458
R=kjellander@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14569005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6204 4adac7df-926f-26a2-2b94-8c16560cd09d
- Add an Initialize() overload to allow specification of format
parameters. This is mainly useful for testing, but could be used in
the cases where a consumer knows the format before the streams arrive.
- Add a reverse_sample_rate_hz_ parameter to prepare for mismatched
capture and render rates. There is no functional change as it is
currently constrained to match the capture rate.
- Fix a bug in the float dump: we need to use add_ rather than set_.
- Add a debug dump test for both int and float interfaces.
- Enable unpacking of float dumps.
- Enable audioproc to read float dumps.
- Move more shared functionality to test_utils.h, and generally tidy up
a bit by consolidating repeated code.
BUG=2894
TESTED=Verified that the output produced by the float debug dump test is
correct. Processed the resulting debug dump file with audioproc and
ensured that we get identical output. (This is crucial, as we need to
be able to exactly reproduce online results offline.)
R=aluebs@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5676 4adac7df-926f-26a2-2b94-8c16560cd09d
As a by-product, a generic tool for testing and comparing the complexity of codecs is added, and new audio files are added to the resources.
Three complexity tests are included
1. Default Opus complexity
2. Opus complexity knob
3. Default iSAC complexity (to compare with Opus)
The complexity tests are only meant for development reasons
and not to be run at bots.
The .isolate file is only needed for the APK packaging and test execution on Android.
TEST=passes all trybots
BUG=
R=kjellander@webrtc.org, tina.legrand@webrtc.org, turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5655 4adac7df-926f-26a2-2b94-8c16560cd09d
