You might overrun the 32 byte fixed-size string "receiveCodec.plname" by copying "payloadName" without checking the length.
Note: This defect has an elevated risk because the source argument is a parameter of the current function.
Review URL: http://webrtc-codereview.appspot.com/352009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1428 4adac7df-926f-26a2-2b94-8c16560cd09d
In addition to our naming guidelines, this will cause these files to get parsed by Sonar, and to make searching/grepping the source using file extensions easier in the future.
BUG=
TEST=Compiling on Linux.
Review URL: http://webrtc-codereview.appspot.com/348005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1405 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
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
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
Restructured the test hierarchy somewhat - there is now a fixture for before-voe-init time and one for after-voe-init time.
Rewrote the hardware-before-streaming test.
Separated unrelated tests out from the rtp_rtcp tests.
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/323009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1184 4adac7df-926f-26a2-2b94-8c16560cd09d
Fixed broken build.
Nit fix.
Fixed style issues.
Removed accidental comment-out.
Removed test that no longer makes sense.
Rewrote hardware-before-init and rtp-rtcp-before-streaming test code to gtest.
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/320009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1162 4adac7df-926f-26a2-2b94-8c16560cd09d
Started extracting methods out of the main test, which will hopefully make us able to make the tests independent.
Merge branch 'master' into voe_split_methods
Conflicts:
src/voice_engine/main/test/auto_test/voe_extended_test.cc
src/voice_engine/main/test/auto_test/voe_extended_test.h
src/voice_engine/main/test/auto_test/voe_standard_test.cc
src/voice_engine/main/test/auto_test/voe_standard_test.h
Extracted methods out of the standard test.
Added space before inheritance colons.
Rolled back some header file changes.
Fixed long lines.
Fixed long lines.
Fixed indentation. There is nothing but whitespace changes here, except for removing some extraneous semicolons in .h files and fixing a spelling error in a comment.
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/313001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1131 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
- Compute RMS over a packet's worth of audio to be sent in Channel, rather than the captured audio in TransmitMixer.
- We now use the entire packet rather than the last 10 ms frame.
- Restore functionality to LevelEstimator.
- Fix a bug in the splitting filter.
- Fix a number of bugs in process_test related to a poorly named
AudioFrame member.
- Update the unittest protobuf and float reference output.
- Add audioproc unittests.
- Reenable voe_extended_tests, and add a real function test.
- Use correct minimum level of 127.
TEST=audioproc_unittest, audioproc, voe_extended_test, voe_auto_test
Review URL: http://webrtc-codereview.appspot.com/279003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@950 4adac7df-926f-26a2-2b94-8c16560cd09d
A single participant is not processed at all. With multiple
participants, we divide-by-2 as before when mixing. Afterwards,
the mixed signal is limited by the AGC to -7 dBFS and then doubled to
restore the original level.
This preserves the level while guaranteeing good saturation protection.
Add a test to voe_auto_test. Hijack and improve the existing mixing test
for this.
TEST=voe_auto_test, voe_cmd_test
Review URL: http://webrtc-codereview.appspot.com/241013
git-svn-id: http://webrtc.googlecode.com/svn/trunk@920 4adac7df-926f-26a2-2b94-8c16560cd09d
