9 Commits

Author SHA1 Message Date
ehmaldonado
f98dc105ba GN: Add target for modules_tests.
Additional changes I needed to make it work:

- Modified a header in RTPFile.cc. Every other file is
  using "webrtc/engine_configurations.h" instead.

- Disabled flag 4373 for msvs because it was disabled
  in build/common.gypi.

BUG=webrtc:6038
TBR=kwiberg@webrtc.org
NOTRY=True

Review-Url: https://codereview.webrtc.org/2187563005
Cr-Commit-Position: refs/heads/master@{#13628}
2016-08-03 17:46:51 +00:00
aleloi
09f45108c2 Removed callback in old AudioConferenceMixer.
OutputMixer and AudioConferenceMixer communicated via a callback. OutputMixer implemented an AudioMixerOutputReceiver interface, which defines the callback function NewMixedAudio. This has been removed and replaced by a simple function in the new mixer. The audio frame with mixed audio is now copied one time less. I have also removed one forward declaration.

Review-Url: https://codereview.webrtc.org/2111293003
Cr-Commit-Position: refs/heads/master@{#13550}
2016-07-28 10:52:23 +00:00
ehmaldonado
529f83c521 Add webrtc_perf_tests to BUILD.gn
Updated the sources in audio_processing:audioproc_test_utils to match the configuration on
"webrtc/modules/audio_processing/audio_processing_tests.gypi"

Removed audio_buffer_tools from modules_unittests to match the gyp file.

BUG=webrtc:6041

Review-Url: https://codereview.webrtc.org/2178963002
Cr-Commit-Position: refs/heads/master@{#13541}
2016-07-27 15:14:42 +00:00
stefan
5e12d36ba7 Reset InterArrival if arrival time clock makes a jump.
Also adds a copy of the BWE test suite to the new DelayBasedBwe class.

BUG=webrtc:6079

Review-Url: https://codereview.webrtc.org/2126793002
Cr-Commit-Position: refs/heads/master@{#13428}
2016-07-11 08:44:12 +00:00
mflodman
e15032c750 Remove all old suspension logic.
I'm also removing media_optimization_unittest.cc, since it only tested the
suspension logic and nothing else.

R=pbos@webrtc.org

Review URL: https://codereview.webrtc.org/2119503002 .

Cr-Commit-Position: refs/heads/master@{#13355}
2016-07-01 07:00:19 +00:00
Alejandro Luebs
a181c9ad17 Keep track of the user-facing number of channels in a ChannelBuffer
Before this change the ChannelBuffer had a fixed number of channels. This meant for example that when the Beamformer would reduce the number of channels to one, the merging filter bank was still merging all the channels, which was unnecessary since they were not processed and just discarded later. This change doesn't change the signal at all. It just reflects the number of channels in the ChannelBuffer, reducing the complexity.

R=henrik.lundin@webrtc.org, peah@webrtc.org, tina.legrand@webrtc.org

Review URL: https://codereview.webrtc.org/2053773002 .

Cr-Commit-Position: refs/heads/master@{#13352}
2016-06-30 22:33:47 +00:00
kwiberg
e7edea9759 Revert of Move FilePlayer and FileRecorder to Voice Engine (patchset #5 id:80001 of https://codereview.chromium.org/2037623002/ )
Reason for revert:
voice_engine_unittests: FilePlayerTest.PlayWavPcm16File and FilePlayerTest.PlayWavPcmuFile fail on 32-bit android (android_rel and android-dbg try bots, Android32 Tests (L Nexus5) and Android32 Tests (L Nexus7.2) build bots).

Not sure why this would happen, since I just moved the test without modifying it. Some test filtering that no longer manages to disable them? Anyway, reverting until I know how to fix.

This was actually caught by the try bots, but I missed it because I was manually ignoring them because of an error with the bots. :-(

Original issue's description:
> Move FilePlayer and FileRecorder to Voice Engine
>
> Because Voice Engine was the only user.
>
> R=perkj@webrtc.org, solenberg@webrtc.org
>
> Committed: 65874b163e

TBR=perkj@webrtc.org,solenberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review-Url: https://codereview.webrtc.org/2092633002
Cr-Commit-Position: refs/heads/master@{#13267}
2016-06-22 23:29:58 +00:00
Karl Wiberg
65874b163e Move FilePlayer and FileRecorder to Voice Engine
Because Voice Engine was the only user.

R=perkj@webrtc.org, solenberg@webrtc.org

Review URL: https://codereview.webrtc.org/2037623002 .

Cr-Commit-Position: refs/heads/master@{#13261}
2016-06-22 21:47:53 +00:00
kjellander
fb11424551 GN: Add modules_unittests
Changes:
* Enabled protobuf for iOS globally.
* Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global
scope similar to GYP since tests depend on it.
* Added missing rtc_libvpx_build_vp9 variable.
* Moved out audio_coding defines into .gni file to avoid code duplication
* Renamed files to avoid object naming conflicts that GN disallows:
  * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc}
  * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc}

BUG=webrtc:5949
TESTED=Built and ran the tests on Mac. Also ran:
gn gen out/Default --args="rtc_enable_bwe_test_logging=true"
and verified that more objects are being built (1885 vs 1883)
when compiling modules_unittests.

NOTRY=True
NOPRESUBMIT=True

Review-Url: https://codereview.webrtc.org/2041233006
Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 07:19:53 +00:00