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
class VoEAudioProcessing
-API renaming:
SetEchoMetricsStatus() to SetEcMetricsStatus()
GetEchoMetricsStatus() to GetEcMetricsStatus()
since delay logging is not strictly an echo metric.
-New API:
GetEcDelayMetrics()
-Implementations
--SetEcMetricsStatus() sets same status to all EC related metrics, currently Echo Metrics and Delay Logging.
--GetEcMetricsStatus() gets an error if all EC related metrics don't have the same status.
--GetEcDelayMetrics() gets the median and standard deviation of AEC internal delay (on a block by block basis).
class VoECallReport
The changes above leads to changes in the Call Report.
-New API:
GetEcDelaySummary()
-API updates:
ResetCallReportStatistics()
WriteReportToFile()
auto_tests updates:
-Standard test, with new Call Report calls and APM calls
-Extended test, with new Call Report calls and APM calls
Review URL: http://webrtc-codereview.appspot.com/187004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@754 4adac7df-926f-26a2-2b94-8c16560cd09d
The main reason is to depend on all ("*") targets in voice_engine.gyp and video_engine.gyp. We don't want the merge_lib targets building by default, since they do funny stuff like delete some libraries.
Review URL: http://webrtc-codereview.appspot.com/191003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@699 4adac7df-926f-26a2-2b94-8c16560cd09d
There's no reason to try to continue if these simple settings fail; better to know about it immediately.
Also, readjusting the indentation to avoid breaking strings over several lines. This bends GStyle a bit, but it's well worth it to avoid the common "forgot to add a space" error.
Review URL: http://webrtc-codereview.appspot.com/173003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@676 4adac7df-926f-26a2-2b94-8c16560cd09d
