This CL introduces a bit-exactness test for the receive-side of the
AudioCoding Module. The main part of the test is done in the helper
class AcmReceiveTest. The test is executed from the test fixture
AcmReceiverBitExactness.
The test inserts packets from a pre-encoded RTP file. The output is
summed up into a checksum, which is verified versus a reference at the
end of the test. Alternatively, if the flag --generate_output is given,
the output is written to a file for subjective verification.
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13769004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6549 4adac7df-926f-26a2-2b94-8c16560cd09d
The design of VoeNetEqStatsTest in voice_engine_unittests depended on
being able to inject a factory for the audio coding module into
voice engine. This functionality is now likely going away, which would
make this test fail to compile. Further, the functionality under test
is mostly ACM functionality, wherefore it makes better sense to test it
at ACM level.
BUG=2996
R=henrika@webrtc.org, turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12029004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5912 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
Instead have ProcessStream transparently handle changes to the stream
audio parameters (sample rate and channels). This removes two locks
per 10 ms ProcessStream call taken by VoiceEngine (four total with the
audio level indicator.)
Also, prepare future improvements by having the splitting filter take
a length parameter. This will allow it to work at different sample
rates. Remove the useless splitting_filter wrapper.
TESTED=voe_cmd_test with audio processing enabled and switching between
codecs; unit tests.
R=aluebs@webrtc.org, bjornv@webrtc.org, turaj@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5346 4adac7df-926f-26a2-2b94-8c16560cd09d
Optionally prevents doing a frame copy when putting frames into a
VideoSendStream. PutFrame() is still there, which copies the frame.
Also removes time_since_capture_ms as a parameter, since
I420VideoFrame::render_time_ms() denotes when the frame was captured.
BUG=2657
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5265 4adac7df-926f-26a2-2b94-8c16560cd09d
Recent changes in GYP seem to have broken our previous
"hack" for getting the GYP rule for .isolate files
imported from the Chromium build/isolate.gypi.
The best solution for now is to remove the hack
and check in a copy of Chromium's src/build/isolate.gypi
in WebRTC's build/ dir instead. A similar approach is
used for our build/protoc.gypi file.
TEST=On Linux, I successfully ran:
gclient runhooks
ninja -C out/Release
and verified a bunch of .isolated files were created in
out/Release (which didn't happen before this patch).
I also renamed the build/isolate.gypi from Chromium to
ensure that our own is used and not that one (in case any
paths would be incorrect).
I also ran build/gyp_chromium in a Chromium checkout
with WebRTC in third_party/webrtc having this patch applied
to ensure GYP processing was still working.
Finally, I verified that the same project generation and
compilation from a Chromium checkout worked the way we build
our Android native tests, using:
. build/android/envsetup.sh
GYP_DEFINES="$GYP_DEFINES include_tests=1 enable_tracing=1" gclient runhooks
ninja -C out/Release android_builder_webrtc
BUG=1916
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2338004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4907 4adac7df-926f-26a2-2b94-8c16560cd09d
We encounter a sample-underrun if NetEq is initialized with a sampling rate fs =16000 and receive Opus packets with frame-size less than 5 ms. The reason is as follows.
Let say NetEq buffer has 4 packets of Opus each of size 2.5ms this means that internally timestamp of packets incremented by 80 (internally Opus treated as 32 kHz codec). Given the initial sampling rate of NetEq, at the first time that it wants to fetch packets, it targets to fetch 160 samples. Therefore, it will only extracts 2 packets. Decoding these packets give us exactly 160 samples (5 ms at 32 kHz), however, upon decoding the first packet the internal sampling rate will be updated to 32 kHz. So it is expected that sync buffer to deliver 320 samples while it does only have 160 samples (or maybe few more as it starts with some zeros). And we encounter and under-run.
Even if we ignore the under-run "assert(sync_buffer_->FutureLength() >= expand_->overlap_length())" (neteq_impl.cc::811) is trigered. I'm not sure what happens if we remove this assert perhaps NetEq will work fine in subsequent calls. However the first under-run is blocking ACM2 test to pass.
Here I have a solution to update sample rate as soon as a packet is inserted, if required. It not a very efficient approach as we do the same reset in NetEqImpl::Decode().
It is a bit tricky to reproduce this because the TOT ACM tests do not run ACM2. In https://webrtc-codereview.appspot.com/2192005/ I have a patch to run both ACMs. To reproduce the problem, one can patch that CL and run
$ out/Debug/modules_tests --gtest_filter=AudioCodingModuleTest.TestOpus
Note that we would not encounter any problem if NetEq4 is initiated with 32000 Hz sampling rate. You can test this by setting |kNeteqInitSampleRateHz| to 32000 in webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
BUG=
R=andrew@webrtc.org, henrik.lundin@webrtc.org, kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2306004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4896 4adac7df-926f-26a2-2b94-8c16560cd09d
Re-land: http://review.webrtc.org/2151007/TBR=bjornv@webrtc.org
Original change description:
This mode extends the filter length from the current 48 ms to 128 ms.
It is runtime selectable which allows it to be enabled through
experiment. We reuse the DelayCorrection infrastructure to avoid having
to replumb everything up to libjingle.
Increases AEC complexity by ~50% on modern x86 CPUs.
Measurements (in percent of usage on one core):
Machine/CPU Normal Extended
MacBook Retina (Early 2013),
Core i7 Ivy Bridge (2.7 GHz, hyperthreaded) 0.6% 0.9%
MacBook Air (Late 2010), Core 2 Duo (2.13 GHz) 1.4% 2.7%
Chromebook Pixel, Core i5 Ivy Bridge (1.8 GHz) 0.6% 1.0%
Samsung ARM Chromebook,
Samsung Exynos 5 Dual (1.7 GHz) 3.2% 5.6%
The relative value is large of course but the absolute should be
acceptable in order to have a working AEC on some platforms.
Detailed changes to the algorithm:
- The filter length is changed from 48 to 128 ms. This comes with tuning
of several parameters: i) filter adaptation stepsize and error
threshold; ii) non-linear processing smoothing and overdrive.
- Option to ignore the reported delays on platforms which we deem
sufficiently unreliable. Currently this will be enabled in Chromium for
Mac.
- Faster startup times by removing the excessive "startup phase"
processing of reported delays.
- Much more conservative adjustments to the far-end read pointer. We
smooth the delay difference more heavily, and back off from the
difference more. Adjustments force a readaptation of the filter, so they
should be avoided except when really necessary.
Corresponds to these changes:
https://chromereviews.googleplex.com/9412014https://chromereviews.googleplex.com/9514013https://chromereviews.googleplex.com/9960013
BUG=454,827,1261
Review URL: https://webrtc-codereview.appspot.com/2295006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4848 4adac7df-926f-26a2-2b94-8c16560cd09d
> Add an extended filter mode to AEC.
>
> This mode extends the filter length from the current 48 ms to 128 ms.
> It is runtime selectable which allows it to be enabled through
> experiment. We reuse the DelayCorrection infrastructure to avoid having
> to replumb everything up to libjingle.
>
> Increases AEC complexity by ~50% on modern x86 CPUs.
> Measurements (in percent of usage on one core):
>
> Machine/CPU Normal Extended
> MacBook Retina (Early 2013),
> Core i7 Ivy Bridge (2.7 GHz, hyperthreaded) 0.6% 0.9%
>
> MacBook Air (Late 2010), Core 2 Duo (2.13 GHz) 1.4% 2.7%
>
> Chromebook Pixel, Core i5 Ivy Bridge (1.8 GHz) 0.6% 1.0%
>
> Samsung ARM Chromebook,
> Samsung Exynos 5 Dual (1.7 GHz) 3.2% 5.6%
>
> The relative value is large of course but the absolute should be
> acceptable in order to have a working AEC on some platforms.
>
> Detailed changes to the algorithm:
> - The filter length is changed from 48 to 128 ms. This comes with tuning
> of several parameters: i) filter adaptation stepsize and error
> threshold; ii) non-linear processing smoothing and overdrive.
> - Option to ignore the reported delays on platforms which we deem
> sufficiently unreliable. Currently this will be enabled in Chromium for
> Mac.
> - Faster startup times by removing the excessive "startup phase"
> processing of reported delays.
> - Much more conservative adjustments to the far-end read pointer. We
> smooth the delay difference more heavily, and back off from the
> difference more. Adjustments force a readaptation of the filter, so they
> should be avoided except when really necessary.
>
> Corresponds to these changes:
> https://chromereviews.googleplex.com/9412014
> https://chromereviews.googleplex.com/9514013
> https://chromereviews.googleplex.com/9960013
>
> BUG=454,827,1261
> R=bjornv@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/2151007TBR=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2296005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4839 4adac7df-926f-26a2-2b94-8c16560cd09d
This mode extends the filter length from the current 48 ms to 128 ms.
It is runtime selectable which allows it to be enabled through
experiment. We reuse the DelayCorrection infrastructure to avoid having
to replumb everything up to libjingle.
Increases AEC complexity by ~50% on modern x86 CPUs.
Measurements (in percent of usage on one core):
Machine/CPU Normal Extended
MacBook Retina (Early 2013),
Core i7 Ivy Bridge (2.7 GHz, hyperthreaded) 0.6% 0.9%
MacBook Air (Late 2010), Core 2 Duo (2.13 GHz) 1.4% 2.7%
Chromebook Pixel, Core i5 Ivy Bridge (1.8 GHz) 0.6% 1.0%
Samsung ARM Chromebook,
Samsung Exynos 5 Dual (1.7 GHz) 3.2% 5.6%
The relative value is large of course but the absolute should be
acceptable in order to have a working AEC on some platforms.
Detailed changes to the algorithm:
- The filter length is changed from 48 to 128 ms. This comes with tuning
of several parameters: i) filter adaptation stepsize and error
threshold; ii) non-linear processing smoothing and overdrive.
- Option to ignore the reported delays on platforms which we deem
sufficiently unreliable. Currently this will be enabled in Chromium for
Mac.
- Faster startup times by removing the excessive "startup phase"
processing of reported delays.
- Much more conservative adjustments to the far-end read pointer. We
smooth the delay difference more heavily, and back off from the
difference more. Adjustments force a readaptation of the filter, so they
should be avoided except when really necessary.
Corresponds to these changes:
https://chromereviews.googleplex.com/9412014https://chromereviews.googleplex.com/9514013https://chromereviews.googleplex.com/9960013
BUG=454,827,1261
R=bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2151007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4837 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
