Edward Lemur 7e3b5697d9 Ignore swarming arguments in low_bandwidth_audio_test.py ...

Needed because swarming adds --isolated-script-test-output and --isolated-script-test-perf-output

See for example:
https://chromium-swarm.appspot.com/task?id=39006c763bebf710

No-Try: true
Bug: chromium:755660
Change-Id: Iff9fb3441200f760c511a67211fbc4a1272717b4
Reviewed-on: https://webrtc-review.googlesource.com/6362
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20146}

2017-10-04 16:19:44 +00:00

..

test

Ignore swarming arguments in low_bandwidth_audio_test.py

2017-10-04 16:19:44 +00:00

utility

Adding NOLINT for typedefs.h and common_types.h

2017-09-15 13:03:51 +00:00

audio_receive_stream_unittest.cc

Added RTCMediaStreamTrackStats.jitterBufferDelay for audio

2017-10-02 10:47:00 +00:00

audio_receive_stream.cc

New method RtpReceiver::GetLatestTimestamps.

2017-10-03 16:14:29 +00:00

audio_receive_stream.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

audio_send_stream_tests.cc

Remove voe_auto_test and add new tests to cover the missing cases.

2017-09-15 16:56:08 +00:00

audio_send_stream_unittest.cc

Remove voe::Statistics.

2017-09-29 13:00:28 +00:00

audio_send_stream.cc

Fix potentional race in AudioSendStream constructor

2017-10-03 16:15:33 +00:00

audio_send_stream.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

audio_state_unittest.cc

Remove the VoiceEngineObserver callback interface.

2017-09-26 16:35:01 +00:00

audio_state.cc

Remove the VoiceEngineObserver callback interface.

2017-09-26 16:35:01 +00:00

audio_state.h

Remove the VoiceEngineObserver callback interface.

2017-09-26 16:35:01 +00:00

audio_transport_proxy.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

audio_transport_proxy.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

BUILD.gn

MB: Add support for isolating scripts + isolate low_bandwidth_audio_test.py.

2017-10-02 16:57:09 +00:00

conversion.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

DEPS

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

OWNERS

Moving src/webrtc into src/.

2017-09-15 04:25:06 +00:00

scoped_voe_interface.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

time_interval_unittest.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

time_interval.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

time_interval.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00