wu@webrtc.org 4f5da030f1 Use _numMixedParticipants instead of audioFrameList->size() to determine if there're more than one participants. ...

There are two audioFrameLists. The previous check wouldn't work correctly if each list had a single member.

TEST=chrome https://apprtc.appspot.com/?debug=loopback&video=false and verify e2e delay stats
BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6723 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-07-17 22:19:21 +00:00

..

audio_coding

Landing pkasting's webrtc fixes for MSVC level 4 warnings in WebRTC.

2014-07-11 19:09:59 +00:00

audio_conference_mixer

Use _numMixedParticipants instead of audioFrameList->size() to determine if there're more than one participants.

2014-07-17 22:19:21 +00:00

audio_device

After an audio interruption the audio unit no longer invokes its render callback, which results in a loss of audio. Restarting the audio unit post interruption fixes the issue.

2014-07-15 20:20:47 +00:00

audio_processing

Remove unused ExperimentalNS API in AudioProcessing

2014-07-17 11:32:09 +00:00

bitrate_controller

GN: Add BUILD.gn files + kjellander to OWNERS

2014-06-23 19:21:07 +00:00

desktop_capture

Fix an invalid memory access due to typo in win/cursor.cc.

2014-07-15 20:32:03 +00:00

interface

Landing pkasting's webrtc fixes for MSVC level 4 warnings in WebRTC.

2014-07-11 19:09:59 +00:00

media_file

Add unit test for MediaFile WAV file writing

2014-07-17 08:11:32 +00:00

pacing

Improvements to the pacer where it lost some budget due to truncation errors.

2014-07-15 16:40:38 +00:00

remote_bitrate_estimator

GN: Fix include paths for WebRTC in Chromium build.

2014-07-13 09:02:54 +00:00

rtp_rtcp

Fix issue where padding is sent before media with undefined timestamps if not abs-send-time is enabled.

2014-07-17 16:10:14 +00:00

utility

Some refactoring inside rtp_rtcp/.

2014-07-08 12:10:51 +00:00

video_capture

Fixes up rtc so that it compiles on iOS 8 SDK.

2014-07-17 00:21:59 +00:00

video_coding

r6709 lacks a change in BUILD.gn

2014-07-16 22:18:49 +00:00

video_processing

Remove the VPM denoiser.

2014-07-15 09:50:40 +00:00

video_render

Add tkchin@ to OWNERS.

2014-07-01 18:38:28 +00:00

module_common_types_unittest.cc

Fix issues with incorrect wrap checks when having big buffers and high bitrate.

2013-04-11 17:48:02 +00:00

modules_java_chromium.gyp

Fixes build breakage when building WebRTC in Chromium and having include_tests=1.

2013-07-08 18:14:58 +00:00

modules_java.gyp

Build all java files into jar for each module on Android

2013-07-01 17:52:39 +00:00

modules_tests.isolate

Pass GYP DEPTH variable to isolate.

2014-06-13 09:02:15 +00:00

modules_unittests.isolate

Pass GYP DEPTH variable to isolate.

2014-06-13 09:02:15 +00:00

modules.gyp

Remove the VPM denoiser.

2014-07-15 09:50:40 +00:00

OWNERS

Add kjellander@webrtc.org as OWNER for *.isolate

2014-06-10 05:42:53 +00:00