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noahric 57779104f0 Forward Encode failure codes from sub encoders. 
Previously, failure codes were ignored, which meant simulcasted codecs
couldn't e.g. trigger software fallback. This stops the simulcasted
Encode call at the first faiulre and returns that code.

Another option is to continue sending the frame to the other encoders
but still return the first failure code. It's not clear that that is any
better than just quitting as soon as a failure happens.

BUG=

Review-Url: https://codereview.webrtc.org/2008723002
Cr-Commit-Position: refs/heads/master@{#12892}
2016-05-25 13:48:54 +00:00
..
api
RTCCertificateGeneratorInterface and RTCCertificateGeneratorStoreWrapper added.
2016-05-24 18:46:42 +00:00
audio
Remove SendPacer from ViEEncoder and make sure SendPacer starts at a valid bitrate
2016-05-11 13:01:19 +00:00
base
Increase the rtccertificategenerator_unittest.cc certificate generation timeout
2016-05-25 12:06:43 +00:00
build
Remove xmllite and xmpp dependencies from merge_libs.gyp.
2016-05-24 12:46:25 +00:00
call
Improved error checking for file errors in RtcEventLog.
2016-05-24 08:04:02 +00:00
common_audio
Revert of Adding a some checks and switching out a few assert for RTC_[D]CHECK. (patchset #6 id:100001 of https://codereview.webrtc.org/2007563002/ )
2016-05-24 13:44:36 +00:00
common_video
Let VideoRenderFrames::FrameToRender return a rtc::Optional.
2016-05-25 13:25:26 +00:00
examples
Improve IP_PATTERN in DirectRTCClient in Android AppRTC Demo
2016-05-25 09:26:29 +00:00
libjingle
Remove unused libjingle_xmpphelp target.
2016-05-24 18:33:39 +00:00
media
Remove xmllite and xmpp dependencies from media.gyp
2016-05-24 09:33:55 +00:00
modules
Forward Encode failure codes from sub encoders.
2016-05-25 13:48:54 +00:00
p2p
Disabled BasicPortAllocatorTest.TestDisableUdpTurn (flaky)
2016-05-25 09:13:03 +00:00
pc
Do not create a temporary transport channel when using max-bundle
2016-05-18 00:49:58 +00:00
sdk
iOS: Replace find with actual filenames listed.
2016-05-24 05:13:51 +00:00
system_wrappers
Adjust max number of samples that can be stored for a histogram (2000->300).
2016-05-24 13:03:45 +00:00
test
Add a default implementation in metrics_default.cc of histograms methods in system_wrappers/interface/metrics.h.
2016-05-20 13:29:50 +00:00
tools
Replace scoped_ptr with unique_ptr everywhere
2016-05-01 21:53:55 +00:00
video
Add class which periodically computes statistics.
2016-05-23 13:08:05 +00:00
voice_engine
Revert of Adding a some checks and switching out a few assert for RTC_[D]CHECK. (patchset #6 id:100001 of https://codereview.webrtc.org/2007563002/ )
2016-05-24 13:44:36 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
audio_receive_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
audio_send_stream.h
Remove webrtc/base/scoped_ptr.h
2016-05-15 02:44:18 +00:00
audio_sink.h
Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies."
2016-02-26 21:46:22 +00:00
audio_state.h
Add comments about the Audio parts of the public Call API being WIP.
2015-12-03 12:06:31 +00:00
BUILD.gn
Revert of Add missing headers and fix some missing dependencies (patchset #1 id:20001 of https://codereview.webrtc.org/1990593002/ )
2016-05-18 12:00:55 +00:00
call.h
Update the call when the network route changes
2016-04-19 22:41:53 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +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
Add histogram stats for average send delay of sent packets for a sent video stream. The delay is measured from a packet is sent to the transport until leaving the socket.
2016-05-03 06:44:11 +00:00
common.gyp
Revert of Add missing headers and fix some missing dependencies (patchset #1 id:20001 of https://codereview.webrtc.org/1990593002/ )
2016-05-18 12:00:55 +00:00
common.h
#include "webrtc/base/constructormagic.h" where appropriate
2016-04-26 15:14:48 +00:00
config.cc
Change transport sequence number extension strings to specify what revision is implemented.
2016-03-01 15:27:27 +00:00
config.h
Re-land: "Use an explicit identifier in Config"
2016-01-14 12:32:51 +00:00
DEPS
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
engine_configurations.h
Remove the VoEDtmf interface.
2016-03-08 12:11:00 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
Add root owners to webrtc/OWNERS
2016-05-11 12:47:43 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
rtc_unittests.isolate
…
supplement.gypi
Disable libyuv jpeg support on Android
2016-05-17 15:39:15 +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
Remove runtime NEON detection
2016-05-17 17:56:48 +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
Remove QualityScaler framerate reduction.
2016-04-26 11:37:20 +00:00
video_engine_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
video_engine_tests.isolate
Add audio streams to CallTest and a first A/V call test.
2016-01-07 16:43:31 +00:00
video_frame.h
Reland of Delete webrtc::VideoFrame methods buffer and stride. (patchset #1 id:1 of https://codereview.webrtc.org/1983583002/ )
2016-05-17 11:05:51 +00:00
video_receive_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
video_send_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
webrtc_examples.gyp
UI for AppRTC Android demo that doesn't require Design Support Library
2016-05-24 07:18:18 +00:00
webrtc_nonparallel_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_nonparallel_tests.isolate
Create isolate files for nonparallel tests.
2015-10-29 14:29:15 +00:00
webrtc_perf_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_perf_tests.isolate
Enable GoogleWifiTrace3Mbps simulations.
2015-05-04 12:26:26 +00:00
webrtc_tests.gypi
Split TaskQueue out into a separate build target.
2016-05-24 18:40:44 +00:00
webrtc.gyp
Revert of Add missing headers and fix some missing dependencies (patchset #1 id:20001 of https://codereview.webrtc.org/1990593002/ )
2016-05-18 12:00:55 +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.