platform-external-webrtc

Files

henrik.lundin 0d8385770b NetEq: Convert AverageIAT from int to float calculations

With this change, the calculations inside AverageIAT are changed to be
in double-precision floating point instead of in fixed point. Also,
the method's name is changed to EstimatedClockDriftPpm to better
reflect what it returns.

A few unit tests had to be updated because of minor numerical
differences.

Also removing the UBSan suppression related to this issue.

BUG=webrtc:5889

Review-Url: https://codereview.webrtc.org/2408653002
Cr-Commit-Position: refs/heads/master@{#14628}

2016-10-13 10:35:58 +00:00

androidjunit

Format all Java in WebRTC.

2016-09-29 11:12:51 +00:00

api

DataChannelInterface default impl of [messages/bytes]_[sent/received].

2016-10-12 21:48:20 +00:00

audio

Add RtcpRttStats to AudioStream

2016-10-11 07:29:34 +00:00

base

Remove deprecated Gestalt methods.

2016-10-10 18:19:22 +00:00

build

Landmine due to corrupt .pdb files on Windows.

2016-10-13 04:07:15 +00:00

call

Add RtcpRttStats to AudioStream

2016-10-11 07:29:34 +00:00

common_audio

Fix "left shift of negative value" bug

2016-10-07 14:38:54 +00:00

common_video

Revert of Delete webrtc::VideoFrame::CopyFrame. (patchset #2 id:20001 of https://codereview.webrtc.org/2371363003/ )

2016-10-06 13:29:34 +00:00

examples

Roll chromium_revision 316b880c55..2cabef4e7d (421519:424936)

2016-10-13 03:25:44 +00:00

libjingle

GN: Change rtc_source_set targets --> rtc_static_library

2016-09-23 07:38:58 +00:00

logging

Added logging for audio send/receive stream configs.

2016-10-10 12:12:57 +00:00

media

Removed the legacy behavior of stopping playout when setting new receive codecs.

2016-10-12 10:20:34 +00:00

modules

NetEq: Convert AverageIAT from int to float calculations

2016-10-13 10:35:58 +00:00

p2p

Revert of Prune connections based on network name. (patchset #3 id:130001 of https://codereview.webrtc.org/2395243005/ )

2016-10-11 13:43:36 +00:00

Remove useless debugging code

2016-10-10 22:57:00 +00:00

sdk

Using relative path for GN for iOS.

2016-10-11 23:12:47 +00:00

stats

RTCIceCandidatePairStats[1] added.

2016-10-11 21:54:55 +00:00

system_wrappers

Fixed flaky clock_unittest by using relative comparison.

2016-10-05 08:55:10 +00:00

test

Add RtcpRttStats to AudioStream

2016-10-11 07:29:34 +00:00

tools

Fix delay plot crash in event_log_visualizer.

2016-10-10 15:16:33 +00:00

video

Set screenshare.enabled parameter to false when running video_loopback test.

2016-10-11 23:20:40 +00:00

voice_engine

Voe::Channel: Turned GetPlayoutFrequency into GetRtpTimestampRateHz.

2016-10-12 18:04:16 +00:00

.gitignore

…

BUILD.gn

Revert of Remove tools dir from root webrtc target (patchset #1 id:1 of https://codereview.webrtc.org/2412353004/ )

2016-10-13 03:50:25 +00:00

call.h

Refactoring: move ownership of RtcEventLog from Call to PeerConnection

2016-10-07 18:53:15 +00:00

codereview.settings

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common_types.cc

Reland of Ignore Camera and Flip bits in CVO when parsing video rotation (patchset #1 id:1 of https://codereview.webrtc.org/2300323002/ )

2016-09-08 10:25:05 +00:00

common_types.h

Enable the -Wundef warning for clang

2016-09-29 00:42:08 +00:00

common.gyp

Delete old video defines in engine config.

2016-10-07 05:07:36 +00:00

config.cc

Rename FecConfig to UlpfecConfig in config.h.

2016-10-05 06:28:43 +00:00

config.h

Rename FecConfig to UlpfecConfig in config.h.

2016-10-05 06:28:43 +00:00

DEPS

Delete old video defines in engine config.

2016-10-07 05:07:36 +00:00

LICENSE

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LICENSE_THIRD_PARTY

Delete talk directory, and references from build_ios_libs.sh.

2016-09-02 10:07:13 +00:00

OWNERS

OWNERS: Make everyone able to change *.gn,*.gni files.

2016-09-09 12:51:48 +00:00

PATENTS

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PRESUBMIT.py

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README.chromium

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supplement.gypi

Cleanup MIPS specific link configuration

2016-06-20 05:39:25 +00:00

transport.h

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typedefs.h

[MIPS] Fix build issue for mips64el

2016-06-23 10:52:35 +00:00

video_decoder.h

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video_encoder.h

Make MediaCodecEncoder fallback to a software encoder on failure.

2016-10-06 12:55:19 +00:00

video_frame.h

Revert of Delete webrtc::VideoFrame::CopyFrame. (patchset #2 id:20001 of https://codereview.webrtc.org/2371363003/ )

2016-10-06 13:29:34 +00:00

video_receive_stream.h

Rename FecConfig to UlpfecConfig in config.h.

2016-10-05 06:28:43 +00:00

video_send_stream.h

Rename FecConfig to UlpfecConfig in config.h.

2016-10-05 06:28:43 +00:00

voice_engine_configurations.h

Delete old video defines in engine config.

2016-10-07 05:07:36 +00:00

webrtc_examples.gyp

Reland of name AppRTCDemo on Android and iOS to AppRTCMobile (patchset #1 id:1 of https://codereview.webrtc.org/2358133003/ )

2016-09-27 08:10:52 +00:00

webrtc_tests.gypi

GYP: Remove targets inside include_tests==1 that are converted to GN.

2016-09-15 11:57:39 +00:00

webrtc.gyp

Moved RtcEventLog files from call/ to logging/

2016-10-04 01:31:32 +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.