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Magnus Jedvert 7640fcf6ed Android VideoSource: Add adaptOutputFormat function 
The Java VideoSource class wraps the C++ AndroidVideoTrackSource.
AndroidVideoTrackSource is the object actually owning the VideoAdapter.
We currently control the VideoAdapter through the Java VideoCapturer,
but it is more natural and direct to control it through the Java
VideoSource class. This CL adds the necessary function to do this, and
the function in VideoCapturer is deprecated.

BUG=webrtc:6391
R=sakal@webrtc.org

Review URL: https://codereview.webrtc.org/2350933006 .

Cr-Commit-Position: refs/heads/master@{#14332}
2016-09-21 14:20:16 +00:00
..
androidjunit
Make android_junit_tests use LocalRobolectricTestRunner
2016-09-18 19:28:15 +00:00
api
Android VideoSource: Add adaptOutputFormat function
2016-09-21 14:20:16 +00:00
audio
Set the event log in Channel from AudioSendStream. This will re-enable logging of outgoing audio packets.
2016-09-21 13:51:52 +00:00
base
Revert of Allow the DTLS fingerprint verification to occur after the handshake. (patchset #11 id:200001 of https://codereview.webrtc.org/2163683003/ )
2016-09-20 00:21:00 +00:00
build
Roll chromium_revision cf9457edb7..cede888c27 (416297:419407)
2016-09-20 20:23:12 +00:00
call
Set the event log in Channel from AudioSendStream. This will re-enable logging of outgoing audio packets.
2016-09-21 13:51:52 +00:00
common_audio
Compensate for the IntelligibilityEnhancer processing delay in high bands
2016-09-20 21:52:08 +00:00
common_video
Revert of Move MutableDataY{,U,V} methods to I420Buffer only. (patchset #14 id:260001 of https://codereview.webrtc.org/2278883002/ )
2016-09-21 10:52:21 +00:00
examples
Android VideoSource: Add adaptOutputFormat function
2016-09-21 14:20:16 +00:00
libjingle
GN: Change group deps to public_deps.
2016-09-15 07:53:34 +00:00
media
Revert of Move MutableDataY{,U,V} methods to I420Buffer only. (patchset #14 id:260001 of https://codereview.webrtc.org/2278883002/ )
2016-09-21 10:52:21 +00:00
modules
Moved legacy_encoded_audio_frame into audio_decoder_interface.
2016-09-21 12:50:45 +00:00
p2p
Remove dead code branches from P2PtransportChannel unittest.
2016-09-20 07:23:33 +00:00
pc
GYP: Remove targets inside include_tests==1 that are converted to GN.
2016-09-15 11:57:39 +00:00
sdk
GN: Enable rtc_common_config for more targets.
2016-09-18 15:12:36 +00:00
stats
RTCTestStats moved into webrtc/stats/test (rtc_stats_test_utils)
2016-09-19 13:06:05 +00:00
system_wrappers
Roll chromium_revision cf9457edb7..cede888c27 (416297:419407)
2016-09-20 20:23:12 +00:00
test
Generalize FEC unit tests and rename GenerateFec.
2016-09-21 06:16:36 +00:00
tools
GYP: Remove targets inside include_tests==1 that are converted to GN.
2016-09-15 11:57:39 +00:00
video
Revert of Move MutableDataY{,U,V} methods to I420Buffer only. (patchset #14 id:260001 of https://codereview.webrtc.org/2278883002/ )
2016-09-21 10:52:21 +00:00
voice_engine
Add voe_cmd_test to voice_engine/BUILD.gn (and remove it from voice_engine.gyp, together with the channel_transport gyp target)
2016-09-15 20:12:10 +00:00
.gitignore
…
BUILD.gn
Reenabled the RtcEventLog unittests
2016-09-19 21:42:10 +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
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
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
Add new decoding statistics for muted output
2016-09-20 08:47:19 +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
…
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
…
PRESUBMIT.py
…
README.chromium
…
supplement.gypi
Cleanup MIPS specific link configuration
2016-06-20 05:39:25 +00:00
transport.h
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
2015-12-09 11:13:40 +00:00
typedefs.h
[MIPS] Fix build issue for mips64el
2016-06-23 10:52:35 +00:00
video_decoder.h
Replace scoped_ptr with unique_ptr in webrtc/common_video/
2016-03-02 11:41:40 +00:00
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_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
Replace VideoCapturerInput with VideoSinkInterface.
2016-09-16 14:53:48 +00:00
webrtc_examples.gyp
Ignores warning for SecRandomCopyBytes() using Xcode 8
2016-09-21 11:13:07 +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
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.