Removing platform limitations for NetEqDecodingTest:TestBitExactness
and NetEqDecodingTest:TestNetworkStatistics. New reference files
where provided in revision 6 of the resources, which allows us
to enable these tests.
BUG=
TEST=neteq_unittests linux32/64, win32/64, mac32
Review URL: http://webrtc-codereview.appspot.com/329027
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1332 4adac7df-926f-26a2-2b94-8c16560cd09d
Clock-drift (in parts-per-million) and peaky-jitter mode status.
Both metrics are propagated to the VoE API. Tests are added
in the NetEQ unittests, and to some extent in ACM unittests
and VoE tests.
Introducing a proper translation between structs NetworkStatistics
and ACMNetworkStatistics.
Note: The file neteq_network_stats.dat in resources must be updated
for the unittests to pass.
Review URL: http://webrtc-codereview.appspot.com/337005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1328 4adac7df-926f-26a2-2b94-8c16560cd09d
Removed old PSNR/SSIM implementations in:
* test/testsupport/metrics/video_metrics.cc
* src/modules/video_coding/codecs/test_framework/test.cc
The functions in video_metrics.cc is now using code in libyuv instead. Old code in test.cc is using the same functions.
The code for video_metrics.h had to be moved into a separate GYP file to avoid circular dependency error on Mac (see issue 160 for more details). The reason for this is that libyuv's unittest target depends on test_support_main.
BUG=
TEST=metrics_unittests in Debug+Release on Linux, Mac and Windows.
Review URL: http://webrtc-codereview.appspot.com/333025
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1325 4adac7df-926f-26a2-2b94-8c16560cd09d
Added temporal layers number flag for video_quality_measurement tool.
This tool now also uses webrtc::VideoCodingModule::Codec() to get its
VideoCodec struct configuration instead of filling it in manually.
Updated paths for header files to use full directory paths.
Tested in Debug+Release on Linux, Mac and Windows. Passes Valgrind memcheck on Linux.
BUG=
TEST=video_codecs_test_framework_integrationtests. Also executed out/Debug/video_quality_measurement --input_filename=resources/foreman_cif.yuv --width=352 --height=288
Review URL: http://webrtc-codereview.appspot.com/339001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1310 4adac7df-926f-26a2-2b94-8c16560cd09d
Robustness improvements to the delay estimator used in AECM and AEC. In AEC only for logging. Faster convergence.
TEST=audioproc_unittest + offline file tests.
output_data_fixed.pb updated despite unverified changes in r1112.
Review URL: http://webrtc-codereview.appspot.com/337006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1306 4adac7df-926f-26a2-2b94-8c16560cd09d
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
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
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
This change avoids having old packets end up on the current packet list for FEC decoding, and so they are immediately sent out to jitter buffer.
The current list of packets for FEC decoding are sent out only when new packet arrives (with time-stamp greater than current).
Review URL: http://webrtc-codereview.appspot.com/322009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1222 4adac7df-926f-26a2-2b94-8c16560cd09d
Removed some Valgrind warnings by closing output files. There are still some Valgrind warnings left, that needs to be fixed by a developer with more insight.
Updated all include paths to contain full paths to header files.
Tested in Debug+Release on Linux, Mac and Windows.
All tests ran successfully except the VideoProcessingModuleTest.ContentAnalysis test that fails on Windows with the following error:
unknown file: error: SEH exception with code 0xc0000005
thrown in the test body.
Fixing that is out of scope for this CL.
Review URL: http://webrtc-codereview.appspot.com/266011
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1217 4adac7df-926f-26a2-2b94-8c16560cd09d
- Also rename _dummy -> no_op which states its purpose more clearly.
- Always use exclusion lists (i.e. sources! instead of sources)
TEST=builds and passes system_wrapper_unittest on Linux, Mac, Win
Review URL: http://webrtc-codereview.appspot.com/317007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1199 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
- r1156 fixed a check on the _text member of FileWrapper. Turns out this
was incompatibile with the RTP dumps, which want to write both binary
and text data. Writing text data to a file open as "b" isn't actually
an error, so I simply removed the check.
- Also cleans up the interface, most notably removing all WebRtc types.
TEST=vie_auto_test
Review URL: http://webrtc-codereview.appspot.com/317005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1175 4adac7df-926f-26a2-2b94-8c16560cd09d
Updated RtpFormatVp8, ModuleRTPUtility, VP8Encoder and VP8Decoder
to support a new LayerSync ("Y") bit. Note, in VP8Encoder the bit
must be used together with a non-negative value for temporalIdx.
Fixing the plumbing between RTP module and and from VP8 wrapper.
Updating unit tests; all pass.
The new bit is yet to be used by the VP8 wrapper.
Review URL: http://webrtc-codereview.appspot.com/323008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1169 4adac7df-926f-26a2-2b94-8c16560cd09d
