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webrtc_m130/webrtc/modules
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Danil Chapovalov 4466782ae4 StartTimestamp generated randomly in RtpSender constructor 
instead of not-randomly at SetSendingState(true)
Renamed to timestamp_offset_ to better match meaning of the variable.

R=asapersson@webrtc.org, terelius@webrtc.org

Review URL: https://codereview.webrtc.org/2241193002 .

Cr-Commit-Position: refs/heads/master@{#13796}
2016-08-17 13:07:49 +00:00
..
audio_coding
WebRtcVoiceEngine: Use AudioDecoderFactory to generate recv codecs.
2016-08-17 09:45:47 +00:00
audio_conference_mixer
Propagate muted info from VoE Channel to AudioConferenceMixer
2016-05-18 12:36:07 +00:00
audio_device
Add a gyp/gn option to use dummy audio file devices.
2016-08-17 01:21:23 +00:00
audio_mixer
Revert of Move FilePlayer and FileRecorder to Voice Engine (patchset #3 id:40001 of https://codereview.webrtc.org/2247033003/ )
2016-08-16 11:08:39 +00:00
audio_processing
Adding GN targets for audioproc, audioproc_f and unpack_aecdump
2016-08-08 06:19:35 +00:00
bitrate_controller
Added NiceMock for MockRtcEventLog in several unittests where we don't care about the event logging.
2016-07-06 07:12:21 +00:00
congestion_controller
CongestionController::SetBweBitrates may now trigger probing.
2016-08-17 09:12:14 +00:00
desktop_capture
Currently there is not way to programmically test whether a ScreenCapturer
2016-08-17 00:34:00 +00:00
include
Add sender controlled playout delay limits
2016-06-08 07:24:30 +00:00
media_file
FileWrapper[Impl] modifications and actually remove the "Impl" class.
2016-06-15 17:30:18 +00:00
pacing
CongestionController::SetBweBitrates may now trigger probing.
2016-08-17 09:12:14 +00:00
remote_bitrate_estimator
Tune BWE to be more sensitive on low capacity networks.
2016-08-04 13:42:14 +00:00
rtp_rtcp
StartTimestamp generated randomly in RtpSender constructor
2016-08-17 13:07:49 +00:00
utility
FileRecorderImpl and FilePlayerImpl don't need their own .h and .cc files
2016-08-17 09:46:57 +00:00
video_capture
GN: Add video_capture_tests for Mac
2016-08-15 16:21:36 +00:00
video_coding
Skip AUD while extracting SPS and PPS on iOS.
2016-08-16 01:56:23 +00:00
video_processing
Rewrite CreateBlackFrame in webrtcvideoengine.
2016-06-20 10:39:00 +00:00
audio_codec_speed_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
audio_codec_speed_tests.isolate
Update isolate.gypi to support Swarming + move .isolate files
2015-09-25 20:19:21 +00:00
audio_decoder_unittests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
audio_decoder_unittests.isolate
Update isolate.gypi to support Swarming + move .isolate files
2015-09-25 20:19:21 +00:00
audio_device_tests.isolate
Update isolate.gypi to support Swarming + move .isolate files
2015-09-25 20:19:21 +00:00
BUILD.gn
Currently there is not way to programmically test whether a ScreenCapturer
2016-08-17 00:34:00 +00:00
module_common_types_unittest.cc
modules: more interface -> include renames
2015-11-04 07:32:04 +00:00
modules_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
modules_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
modules_unittests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
modules_unittests.isolate
Revert of Move FilePlayer and FileRecorder to Voice Engine (patchset #3 id:40001 of https://codereview.webrtc.org/2247033003/ )
2016-08-16 11:08:39 +00:00
modules.gyp
Currently there is not way to programmically test whether a ScreenCapturer
2016-08-17 00:34:00 +00:00
OWNERS
Add kjellander@webrtc.org as OWNER for *.isolate
2014-06-10 05:42:53 +00:00
video_render_tests.isolate
Update isolate.gypi to support Swarming + move .isolate files
2015-09-25 20:19:21 +00:00