-Make ACM1 to depend on ACM2.
-Remove APIs to set and get background noise mode. There is no VoE call to these APIs.
-Remove APIs to set and get receive side VAD mode. There is no VoE call to these APIs, and NetEq 4, doesn't support them.
-Remove callback for in-band DTMF detection. ACM doesn't support in-band DTMF detection.
-Use acm_common_defs.h everywhere required.
-Complete ACM factory method.
-Update ACMCodecDatabase of ACM2. CNG full-band need to be define-guarded. Remove dynamic payload-type assignment.
BUG=
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2237004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4772 4adac7df-926f-26a2-2b94-8c16560cd09d
I mistakenly ommitted the checks when logging.h was ported from
libjingle to webrtc. This caused a significant CPU cost for logs which
were later filtered out anyway.
Verified with LS_VERBOSE logging in neteq4, running:
$ out/Release/modules_unittests \
--gtest_filter=NetEqDecodingTest.TestBitExactness \
--gtest_repeat=50 > time.txt
$ grep "case ran" time.txt | grep "[0-9]* ms" -o | sort
Results on a MacBook Retina, averaged over 5 runs:
Verbose logs disabled: 666 ms
Exisiting implementation, verbose logs enabled: 944 ms (1.42x)
New implementation, verbose logs enabled: 673 ms (1.01x)
BUG=2314
R=henrik.lundin@webrtc.org, henrike@webrtc.org, kjellander@webrtc.org, turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2160005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4682 4adac7df-926f-26a2-2b94-8c16560cd09d
Tests enabled in r4671 failed:
build.chromium.org/p/client.webrtc/builders/Android%20Tests/builds/31/steps/slave_steps/logs/stdio
> Enable SetInitialPlayoutDelay on Android.
>
> Background:
> In Chrome mirroring which uses 500ms buffering mode,
> audio video mismatch happens in the begining because of the lack of the api.
>
> BUG=b/10538425
> TEST=pass 'git try' except tests which is aleady broken in the bot. pass 'build/android/test_runner.py gtest -s modules_tests --verbose --release -f *InitialPlayoutDelayTest*'
> R=henrika@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/2144004TBR=dwkang@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2160006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4672 4adac7df-926f-26a2-2b94-8c16560cd09d
Background:
In Chrome mirroring which uses 500ms buffering mode,
audio video mismatch happens in the begining because of the lack of the api.
BUG=b/10538425
TEST=pass 'git try' except tests which is aleady broken in the bot. pass 'build/android/test_runner.py gtest -s modules_tests --verbose --release -f *InitialPlayoutDelayTest*'
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2144004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4671 4adac7df-926f-26a2-2b94-8c16560cd09d
The ACM tests needed re-writing, because all tests were not individual gtests, and the result was difficult to interpret.
While doing the re-write, I discovered a bug related to 48 kHz CNG. We can't have the 48 kHz CNG active at the moment. The bug is fixed in this CL.
I also needed to rewrite parts of the VAD/DTX implementation, so that the status of VAD and DTX (enabled or not) is propagated back from the function SetVAD().
BUG=issue2173
R=minyue@webrtc.org, turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1961004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4625 4adac7df-926f-26a2-2b94-8c16560cd09d
This is a re-land attempt of http://review.webrtc.org/1673004/
It now includes a build/isolate.gypi in WebRTC that includes the same
file as the one that would be included when WebRTC is used in a Chromium
checkout. It is needed since it is not possible to use variables in GYP's
includes sections.
Implemented according to the instructions at
http://www.chromium.org/developers/testing/isolated-testing
Workflow has been like this:
1. create _run GYP target
2. create a stripped down .isolate file
3. export GYP_DEFINES="$GYP_DEFINES test_isolation_mode=check"
4. runhooks
5. compile
6. test if the test would run (i.e. find it's dependencies) without
actually executing it:
tools/swarm_client/isolate.py run --isolated out/Release/testname.isolated
7. If failing, run the fix_test_cases.py script like this:
tools/swarm_client/googletest/fix_test_cases.py --isolated out/Release/testname.isolated
All tests that run on the bots for WebRTC has got _run target
and .isolate file created.
"Normal tests" that run fine on any machine:
* audio_decoder_unittests
* common_audio_unittests
* common_video_unittests
* metrics_unittests
* modules_tests
* modules_unittests
* neteq_unittests
* system_wrappers_unittests
* test_support_unittests
* tools_unittests
* video_engine_core_unittests
* voice_engine_unittests
Tests that requires bare-metal and audio/video devices:
* audio_device_tests
* video_capture_tests
I also added the isolate boilerplate code for the following
tests that are not yet pure gtest binaries (which means they
cannot run isolated yet):
* video_render_tests
* vie_auto_test
* voe_auto_test
TEST=running isolate.py as described above. WebRTC trybots passing. Created a Chromium checkout with third_party/webrtc ToT and this patch applied, passing the runhooks step.
BUG=1916
R=henrike@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2056004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4590 4adac7df-926f-26a2-2b94-8c16560cd09d
As this breaks the FYI bots in
http://build.chromium.org/p/chromium.webrtc.fyi/waterfall
due to different path to isolate.gypi (which cannot easily
be resolved due to limitations in GYP)
> Isolate GYP target and .isolate files for tests
>
> Implemented according to the instructions at
> http://www.chromium.org/developers/testing/isolated-testing
>
> Workflow has been like this:
> 1. create _run GYP target
> 2. create a stripped down .isolate file
> 3. export GYP_DEFINES="$GYP_DEFINES test_isolation_mode=check"
> 4. runhooks
> 5. compile
> 6. test if the test would run (i.e. find it's dependencies) without
> actually executing it:
> tools/swarm_client/isolate.py run --isolated out/Release/testname.isolated
> 7. If failing, run the fix_test_cases.py script like this:
> tools/swarm_client/fix_test_cases.py --isolated out/Release/testname.isolated
>
> All tests that run on the bots for WebRTC has got _run target
> and .isolate file created.
>
> "Normal tests" that run fine on any machine:
> * audio_decoder_unittests
> * common_audio_unittests
> * common_video_unittests
> * metrics_unittests
> * modules_integrationtests
> * modules_unittests
> * neteq_unittests
> * system_wrappers_unittests
> * test_support_unittests
> * tools_unittests
> * video_engine_core_unittests
> * voice_engine_unittests
>
> Tests that requires bare-metal and audio/video devices:
> * audio_device_integrationtests
> * video_capture_integrationtests
>
> I also added the isolate boilerplate code for the following
> tests that are not yet pure gtest binaries (which means they
> cannot run isolated yet):
> * video_render_integrationtests
> * vie_auto_test
> * voe_auto_test
>
> TEST=running isolate.py as described above.
> BUG=1916
> R=tommi@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/1673004TBR=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2040004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4548 4adac7df-926f-26a2-2b94-8c16560cd09d
Implemented according to the instructions at
http://www.chromium.org/developers/testing/isolated-testing
Workflow has been like this:
1. create _run GYP target
2. create a stripped down .isolate file
3. export GYP_DEFINES="$GYP_DEFINES test_isolation_mode=check"
4. runhooks
5. compile
6. test if the test would run (i.e. find it's dependencies) without
actually executing it:
tools/swarm_client/isolate.py run --isolated out/Release/testname.isolated
7. If failing, run the fix_test_cases.py script like this:
tools/swarm_client/fix_test_cases.py --isolated out/Release/testname.isolated
All tests that run on the bots for WebRTC has got _run target
and .isolate file created.
"Normal tests" that run fine on any machine:
* audio_decoder_unittests
* common_audio_unittests
* common_video_unittests
* metrics_unittests
* modules_integrationtests
* modules_unittests
* neteq_unittests
* system_wrappers_unittests
* test_support_unittests
* tools_unittests
* video_engine_core_unittests
* voice_engine_unittests
Tests that requires bare-metal and audio/video devices:
* audio_device_integrationtests
* video_capture_integrationtests
I also added the isolate boilerplate code for the following
tests that are not yet pure gtest binaries (which means they
cannot run isolated yet):
* video_render_integrationtests
* vie_auto_test
* voe_auto_test
TEST=running isolate.py as described above.
BUG=1916
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1673004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4547 4adac7df-926f-26a2-2b94-8c16560cd09d
This is the conversation I had with Henrik Lundin regarding this problem.
Me:
In Expand::AnalyseSignal() we compute correlation and distortion, then calculate the ratio of correlation to distortion. There if distortion is zero we expect that correlation to be zero. Although in practice this might be true, I suppose we rarely hit into absolutely periodic signal, but in one of the tests the assertion in line 455 of expand.cc was triggered. The distortion is computed over a shorter length of the signal, while correlation is computed over longer segments. Therefore, I guess, if the signal has just enough zeros at the beginning we can end up in situation that distortion is zero but not the correlation. Do you agree? I didn't have time to attempt to solve this, but if my line of thought is correct, we should not have that assert. Perhaps, if correlation is zero we set the ratio to 0, otherwise, ratio would be the largest value of its own type. Any thoughts?
Henrik:
I agree with you. Go ahead with your solution.
R=minyue@google.com
Review URL: https://webrtc-codereview.appspot.com/1888006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4448 4adac7df-926f-26a2-2b94-8c16560cd09d
