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
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
architecture with intrinsics and assembly code. The total iSAC codec speech improved
about 3~5%.
Notes
(1) The Neon version after this optimization is not bit-exact with the generic
C version. The out quality, however, is not worse as verified by test vectors ouput,
and undertandably in theory (32bit x 32bit in Neon is more accurate than the approximation
C code in the generic version).
(2) In Android, a isac neon library will be built. Along with some new function structures,
it is partly for preparation of introducing a run time detection of Neon architecture soon.
Review URL: http://webrtc-codereview.appspot.com/268016
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1192 4adac7df-926f-26a2-2b94-8c16560cd09d
With some compiler settings, a warning was issued for NetEQ,
saying that pw16_randVec was accessed out of bounds.
This did never happen in practice, but this change makes the
compiler understand this.
Review URL: http://webrtc-codereview.appspot.com/309001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1121 4adac7df-926f-26a2-2b94-8c16560cd09d
Implementing helper function for new unit test
NetEqDecodingTest::TestNetworkStatistics. The test itself
remains to be defined. (Will be added in a coming CL.)
This change required some refactoring of the test code
to avoid excessive code duplication.
Review URL: http://webrtc-codereview.appspot.com/295009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1049 4adac7df-926f-26a2-2b94-8c16560cd09d
The special cases for packet sizes <= 10 ms (one case for each
sample rate) resulted in reading outside of the pw16_decoded
vector. This is now fixed by making sure that WebRtcSpl_DownsampleFast
gets correct input and output vector lengths.
Review URL: http://webrtc-codereview.appspot.com/295008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1027 4adac7df-926f-26a2-2b94-8c16560cd09d
Fixing a bug when creating new NetEQ slave instances in ACM.
The old code called WebRtcNetEQ_GetCurrentDelay() for the
master instance to get a delay value for WebRtcNetEQ_SetExtraDelay().
This is wrong, since WebRtcNetEQ_GetCurrentDelay() reports on the
current total buffer length, while WebRtcNetEQ_SetExtraDelay() is
the extra delay that is desired to in order to sync with video.
The fix includes keeping the extra delay value in a member variable
in the ACMNetEQ class.
Review URL: http://webrtc-codereview.appspot.com/295001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1001 4adac7df-926f-26a2-2b94-8c16560cd09d
Removing WebRtcNetEQ_GetJitterStatistics(),
WebRtcNetEQ_ResetJitterStatistics(), and the associated
struct WebRtcNetEQ_JitterStatistics. The change ripples
through all the way to the VoiceEngine API.
Review URL: http://webrtc-codereview.appspot.com/285002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@998 4adac7df-926f-26a2-2b94-8c16560cd09d
(1) For generic platforms, code was changed to remove the shifting within loops.
Basically, it's just change a loop from
for() {
sum += (a*b) >> scale;
}
to:
for() {
sum += (a*b);
}
sum >> scale;
Type int64_t is used for sum to make sure no information is not lost.
Performance is about the same as before the change. Bits are not exact,
although in theory the change should have preserved more information. The purpose
of this change is to make the generic code and ARM code bit exact, simpify the code,
while keep the speech quality at least not lower. (Some speech tests might be good.)
(2) For ARM platform, used assembly to optimize the performance. iSAC runs faster
with this change. (Reduced run time of an offline file test from 10.16ms to 8.81ms)
Review URL: http://webrtc-codereview.appspot.com/267014
git-svn-id: http://webrtc.googlecode.com/svn/trunk@972 4adac7df-926f-26a2-2b94-8c16560cd09d
Removed TODO from webrtc.gyp since it is done.
Tabs -> spaces.
Tabs -> spaces.
Tabs -> spaces.
Fixed compilation on Windows.
Added missing file.
Merge branch 'master' into fix_mac_modules
Fixed compilation errors for the modules.gyp on Mac. This included some pretty large refactorings.
Please enter the commit message for your changes. Lines starting
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/269005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@957 4adac7df-926f-26a2-2b94-8c16560cd09d
I also fixed compilation on Mac (by enabling exceptions for the NetEqTestTools target). Executing the test fails on Mac, but I assume this is because it checks bit exactness, similar to the issue we had with audio_coding_module (see issue 114)
Review URL: http://webrtc-codereview.appspot.com/255004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@895 4adac7df-926f-26a2-2b94-8c16560cd09d
