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Stefan Holmer 8bffba7107 Fix BWE bug where audio has timestamps in us. 
The BWE expects arrival timestamps in ms, while the audio path delivered
them in us, causing the BWE to break down under the combined audio/video
BWE experiment. This was introduced in r9892 (68786d2040).

BUG=webrtc:4758
R=mflodman@webrtc.org, sprang@webrtc.org

Review URL: https://codereview.webrtc.org/1360913004 .

Cr-Commit-Position: refs/heads/master@{#10032}
2015-09-23 13:54:04 +00:00
..
base
Revert of TransportController refactoring. (patchset #6 id:100001 of https://codereview.webrtc.org/1350523003/ )
2015-09-23 09:24:27 +00:00
build
Purge nss files and dependencies.
2015-09-22 18:58:13 +00:00
common_audio
Add RTC_ prefix to (D)CHECKs and related macros.
2015-09-17 07:24:51 +00:00
common_video
Add RTC_ prefix to (D)CHECKs and related macros.
2015-09-17 07:24:51 +00:00
examples
Remove the video capture module on Android.
2015-09-21 08:46:37 +00:00
libjingle
Add RTC_ prefix to contructormagic macros.
2015-09-16 12:37:52 +00:00
modules
Returning correct duration estimate on Opus DTX packets.
2015-09-23 13:20:56 +00:00
overrides
Remove overridden basictypes.h.
2015-09-18 08:48:39 +00:00
p2p
Revert of TransportController refactoring. (patchset #6 id:100001 of https://codereview.webrtc.org/1350523003/ )
2015-09-23 09:24:27 +00:00
sound
Add RTC_ prefix to contructormagic macros.
2015-09-16 12:37:52 +00:00
system_wrappers
Add RTC_ prefix to (D)CHECKs and related macros.
2015-09-17 07:24:51 +00:00
test
This relands "Tool to convert RtcEventLog files to RtpDump format.", commit 35624c2c3686a2ad40daffe073aa78507b0ef88e.
2015-09-18 13:41:18 +00:00
tools
Add RTC_ prefix to (D)CHECKs and related macros.
2015-09-17 07:24:51 +00:00
video
Fix BWE bug where audio has timestamps in us.
2015-09-23 13:54:04 +00:00
video_engine
Revert of Wire up send-side bandwidth estimation. (patchset #8 id:140001 of https://codereview.webrtc.org/1338203003/ )
2015-09-23 11:52:01 +00:00
voice_engine
Improving support for Android Audio Effects in WebRTC.
2015-09-23 12:09:40 +00:00
.gitignore
.gitignore: Add *.mk, created as part of ChromiumOS build
2013-01-04 21:25:42 +00:00
audio_receive_stream.h
Control combined_audio_video_bwe with config bool.
2015-07-24 14:10:25 +00:00
audio_send_stream.h
Add send transports to individual webrtc::Call streams.
2015-08-28 11:07:15 +00:00
BUILD.gn
Revert "Android: Enable C99 mode instead of C89 (default)."
2015-09-21 14:50:56 +00:00
call.h
Wire up PacketTime to ReceiveStreams.
2015-09-08 12:36:23 +00:00
codereview.settings
Update deeper codereview.settings files to match the root.
2015-06-24 19:40:39 +00:00
common_types.cc
Separating voice activity flag from audio level in RtpHeaderExtension.
2015-08-10 13:08:46 +00:00
common_types.h
Remove channel ids from various interfaces.
2015-09-17 21:06:02 +00:00
common.gyp
Fix style violations in common_types.h and config.h
2015-02-26 14:01:28 +00:00
common.h
Add RTC_ prefix to contructormagic macros.
2015-09-16 12:37:52 +00:00
config.cc
Add support for transport wide sequence numbers
2015-08-03 11:38:48 +00:00
config.h
Add support for transport wide sequence numbers
2015-08-03 11:38:48 +00:00
engine_configurations.h
Remove ACM AudioCodingFeedback callback object and derived classes
2015-09-18 10:56:15 +00:00
frame_callback.h
Rename I420VideoFrame to VideoFrame.
2015-05-30 00:21:56 +00:00
libjingle_examples.gyp
Fix build when using Xcode 7 which contains .tbd files instead of .dylib
2015-09-05 03:01:08 +00:00
LICENSE
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
LICENSE_THIRD_PARTY
Add the Ooura FFT to RealFourier.
2015-03-19 20:07:43 +00:00
OWNERS
GN: Add BUILD.gn files + kjellander to OWNERS
2014-06-23 19:21:07 +00:00
PATENTS
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
PRESUBMIT.py
PRESUBMIT.py: accept variants on the copyright message that are present in the codebase.
2014-05-23 17:27:18 +00:00
README.chromium
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
rtc_unittests.isolate
Update all .isolate files for the new format.
2014-10-31 18:08:09 +00:00
stream.h
Wire up PacketTime to ReceiveStreams.
2015-09-08 12:36:23 +00:00
supplement.gypi
Move target_subarch from gyp_webrtc to supplement.gypi
2015-04-16 07:24:23 +00:00
transport.h
Rename newapi::Transport::SendRTP()->SendRtp().
2013-11-20 12:17:04 +00:00
typedefs.h
Clear ARM NEON flag
2015-05-20 05:20:04 +00:00
video_decoder.h
iOS HW H264 support.
2015-06-29 21:35:08 +00:00
video_encoder.h
Allow for framerate reduction for HW encoder.
2015-07-13 23:26:40 +00:00
video_engine_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
video_frame.h
Add RTC_ prefix to (D)CHECKs and related macros.
2015-09-17 07:24:51 +00:00
video_receive_stream.h
Wire up currently-received video codec to stats.
2015-08-28 14:35:40 +00:00
video_renderer.h
Rename I420VideoFrame to VideoFrame.
2015-05-30 00:21:56 +00:00
video_send_stream.h
Make LoadObserver settable per video send stream. Gives client flexibility and makes the implementation slightly simpler. See discussion in: https://codereview.webrtc.org/1269863005/
2015-09-08 12:13:25 +00:00
webrtc_examples.gyp
Adding support for OpenSL ES output in native WebRTC
2015-05-18 14:49:04 +00:00
webrtc_perf_tests.isolate
Enable GoogleWifiTrace3Mbps simulations.
2015-05-04 12:26:26 +00:00
webrtc_tests.gypi
Fix BWE bug where audio has timestamps in us.
2015-09-23 13:54:04 +00:00
webrtc.gyp
This relands "Tool to convert RtcEventLog files to RtpDump format.", commit 35624c2c3686a2ad40daffe073aa78507b0ef88e.
2015-09-18 13:41:18 +00:00

README.chromium 

Name: WebRTC
URL: http://www.webrtc.org
Version: 90
License: BSD
License File: LICENSE

Description:
WebRTC provides real time voice and video processing
functionality to enable the implementation of 
PeerConnection/MediaStream.

Third party code used in this project is described 
in the file LICENSE_THIRD_PARTY.