platform-external-webrtc

Files

kthelgason ebc34e78db [GN] Add rtc_sdk_framework_objc target to GN

The build artifacts don't look completely identical to the ones generated
by the GYP targets, but manual review shows the same symbols are exported.

On iOS,  the version generated by the GN follows convention, including
a "Headers" directory, and the .modulemap file. I think this is preferred
over the gyp version.

BUG=webrtc:6320
NOTRY=True
TESTED=Run AppRTCDemo on iOS + Mac and verified with nm that they export the same symbols.

Review-Url: https://codereview.webrtc.org/2340633003
Cr-Commit-Position: refs/heads/master@{#14228}

2016-09-15 11:30:21 +00:00

androidjunit

Re-order and remove unused Java imports.

2016-08-02 13:44:19 +00:00

api

GN: Change group deps to public_deps.

2016-09-15 07:53:34 +00:00

audio

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

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

base

Revert of Add arraysize to rtc_base_approved. Remove dependency of audio_device on rtc_base. (patchset #1 id:1 of https://codereview.webrtc.org/2346763002/ )

2016-09-15 08:45:33 +00:00

build

MB: Change Android Clang to build shared instead of static.

2016-09-15 10:42:34 +00:00

call

Remove RTC_LOGGED_* macro.

2016-09-10 05:40:34 +00:00

common_audio

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

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

common_video

GN: Declare resources for targets.

2016-09-14 12:10:06 +00:00

examples

GN: Change group deps to public_deps.

2016-09-15 07:53:34 +00:00

libjingle

GN: Change group deps to public_deps.

2016-09-15 07:53:34 +00:00

media

The VoE functionality to apply receive-side processing to VoE channels is unused. I'm removing it so we can avoid instantiating a full APM per channel (and thus also for webrtc::AudioSendStream and webrtc::AudioReceiveStream), and then never use it.

2016-09-15 11:29:21 +00:00

modules

Fix undefined reference to log2 on android

2016-09-15 08:56:30 +00:00

p2p

GN: Change group deps to public_deps.

2016-09-15 07:53:34 +00:00

GN: Change group deps to public_deps.

2016-09-15 07:53:34 +00:00

sdk

[GN] Add rtc_sdk_framework_objc target to GN

2016-09-15 11:30:21 +00:00

stats

GN: Change group deps to public_deps.

2016-09-15 07:53:34 +00:00

system_wrappers

Remove RTC_LOGGED_* macro.

2016-09-10 05:40:34 +00:00

test

2016-09-15 11:29:21 +00:00

tools

GN: Declare resources for targets.

2016-09-14 12:10:06 +00:00

video

Add periodic logging of number of captured and dropped frames in VideoCaptureInput. Logged every minute.

2016-09-12 07:10:53 +00:00

voice_engine

2016-09-15 11:29:21 +00:00

.gitignore

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BUILD.gn

GN: Declare resources for targets.

2016-09-14 12:10:06 +00:00

call.h

This reverts commit 8eb37a39e79fe1098d3503dcb8c8c2d196203fed. Chrome now have its own implementation of TaskQueues that is based on Chrome threads.

2016-09-01 08:17:43 +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

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.gyp

Moving/renaming webrtc/common.h.

2016-09-07 14:34:45 +00:00

config.cc

Avoid unnecessary HW video encoder reconfiguration

2016-06-16 19:08:11 +00:00

config.h

Moving/renaming webrtc/common.h.

2016-09-07 14:34:45 +00:00

DEPS

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

engine_configurations.h

Reland of move all reference to carbon api (patchset #1 id:1 of https://codereview.webrtc.org/2317343003/ )

2016-09-09 10:19:54 +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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rtc_unittests_apk.isolate

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rtc_unittests.isolate

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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

Add codec name to CodecSpecificInfo and get the codec name stats from there instead.

2016-09-01 07:21:18 +00:00

video_engine_tests_apk.isolate

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video_engine_tests.isolate

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video_frame.h

Introduce webrtc::VideoFrame::timestamp_us, and corresponding constructor.

2016-09-05 07:51:25 +00:00

video_receive_stream.h

Add periodic logging of video stats.

2016-08-11 15:41:26 +00:00

video_send_stream.h

This reverts commit 8eb37a39e79fe1098d3503dcb8c8c2d196203fed. Chrome now have its own implementation of TaskQueues that is based on Chrome threads.

2016-09-01 08:17:43 +00:00

webrtc_examples.gyp

Delete talk directory, and references from build_ios_libs.sh.

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

webrtc_nonparallel_tests_apk.isolate

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webrtc_nonparallel_tests.isolate

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webrtc_perf_tests_apk.isolate

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webrtc_perf_tests.isolate

Add an HD resolution perf test.

2016-08-11 14:01:03 +00:00

webrtc_tests.gypi

Added build flag around the Intelligibility enhancer performance test code

2016-09-14 04:42:43 +00:00

webrtc.gyp

Move webrtc/audio_*.h to webrtc/api/call

2016-08-31 14:33:14 +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.