This CL includes a larger structural change in how we handle buffers in AEC. We now perform FFT at once and move within blocks to compensate for system delays.
TEST=audioproc_unittest(float and fix), voe_auto_test
Review URL: http://webrtc-codereview.appspot.com/335012
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1299 4adac7df-926f-26a2-2b94-8c16560cd09d
We can't reset the complete last decoded state when we recycle until a
key frame because that will allow any delta frame to be decoded afterwards,
and since the decoder isn't reset we will get decode errors.
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/330003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1295 4adac7df-926f-26a2-2b94-8c16560cd09d
Only changed:
- Name of some of the critsects.
- All critsects (but one) are now scoped_ptr.
- Use of ptr constructor of CriticalSectionScoped instead of reference version.
BUG=184
TEST=vie_auto_test
Review URL: http://webrtc-codereview.appspot.com/330015
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1291 4adac7df-926f-26a2-2b94-8c16560cd09d
The packetizer class is changed so that the max payload size is
provided on construction of the class rather than for each packet.
The tests are re-written to comply with the new design.
Also fixing a few errors in the tests.
Review URL: http://webrtc-codereview.appspot.com/335010
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1280 4adac7df-926f-26a2-2b94-8c16560cd09d
This CL defines and starts to implement a new robustness API for
video coding module. The API is partly implemented. Some of the
modes and methods are still TBD.
Also including a new unittest with mocking of decoder and callbacks,
and faking of system clock.
Review URL: http://webrtc-codereview.appspot.com/333006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1276 4adac7df-926f-26a2-2b94-8c16560cd09d
The number of capture channels can only be determined upon receiving the
first captured frame. We now assume stereo capture by default and set the
number of AudioProcessing input channels based on captured frames.
TEST=Windows mono-only device now runs AudioProcessing correctly (NS etc.), voe_auto_test (though some new, seemingly unrelated, tests are failing)
Review URL: http://webrtc-codereview.appspot.com/330013
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1273 4adac7df-926f-26a2-2b94-8c16560cd09d
- Keep up to date with the clang rev Chrome is using.
- tools/clang/update.sh now updates clang on Linux if it has been
manually run once.
- We can remove clang_revision from DEPS; now maintained in update.sh.
- Also keep the gyp rev in sync.
TEST=build on Linux, Mac, Win
Review URL: http://webrtc-codereview.appspot.com/331010
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1271 4adac7df-926f-26a2-2b94-8c16560cd09d
The recorded frame timestamps are used to modify the output video on a frame-per-frame so it can be compared with the reference video using PSNR. This code will make it possible to use vie_auto_test for full stack comparisons with network interference and similar interesting simulations.
There's some refactoring done in vie_comparison_test.cc to make it fit to the new test.
Compiled and executed in Debug+Release on Linux, Mac and Windows.
BUG=
TEST=vie_auto_test --automated --gtest_filter=ViEVideoVerificationTest.*
Review URL: http://webrtc-codereview.appspot.com/320002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1269 4adac7df-926f-26a2-2b94-8c16560cd09d
This change basicly re-enables the change of r1220, which was
reverted in r1235 due to Clang issues.
The difference from r1220 is that the TickTimeInterface was
renamed to TickTimeClass, and no longer inherits from TickTime.
Review URL: http://webrtc-codereview.appspot.com/335006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1267 4adac7df-926f-26a2-2b94-8c16560cd09d
This test now runs and fails as a gtest should (previously it always
exited with 0 even if the tests failed).
The audio_coding_module_test target no longer uses exceptions in the generated project.
Output files are written to our global output folder, using
testsupport/fileutils.h.
BUG=
TEST=audio_coding_module_test on all platforms, in Debug+Release
Review URL: http://webrtc-codereview.appspot.com/334004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1266 4adac7df-926f-26a2-2b94-8c16560cd09d
This is the first change in a series of changes to get new functionality
into the VP8 packetizer.
This first refactors the RtpFormatVp8Test class, without changing the
operation of the tested RtpFormatVp8 class. A test helper class
RtpFormatVp8TestHelper is introduced to reduce code duplication.
Review URL: http://webrtc-codereview.appspot.com/304009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1258 4adac7df-926f-26a2-2b94-8c16560cd09d
Adding new statistics API to NetEQ, reporting the waiting time
for each frame. The output is raw waiting time for the frames
that have been decoded since the last statistics report (or
maximum 100 frames). The statistics are reset on each query.
Implemented functionality in ACM to query NetEQ for the raw
waiting times, and process it to produce max, average and
median.
Updating common_types.h and VoiceEngine tests to include the
new metrics.
Unit tests are also added for NetEQ and AcmNetEq.
Review URL: http://webrtc-codereview.appspot.com/328011
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1251 4adac7df-926f-26a2-2b94-8c16560cd09d
