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jianjun.zhu e1b4d243e2 Skip AUD while extracting SPS and PPS on iOS. 
H.264 frames may have AUD before SPS. This change detects AUD and try to extract SPS and PPS behind AUD.

BUG=webrtc:6173

Review-Url: https://codereview.webrtc.org/2209143002
Cr-Commit-Position: refs/heads/master@{#13765}
2016-08-16 01:56:23 +00:00
..
androidjunit
Re-order and remove unused Java imports.
2016-08-02 13:44:19 +00:00
api
Remove old methods in AudioTransport, make it pass a gn build
2016-08-15 18:46:28 +00:00
audio
Removed calls to VoE::SetPlayout() from WebRTCVoiceEngine.
2016-08-04 12:28:28 +00:00
base
FakeTiming added, an implementation of Timing that can be used for
2016-08-15 14:00:06 +00:00
build
Add Mac bots to MB.
2016-08-11 08:04:39 +00:00
call
Add periodic logging of video stats.
2016-08-11 15:41:26 +00:00
common_audio
GN: Update tests to have the correct shard timeout value on Android.
2016-08-15 09:29:19 +00:00
common_video
GN: Update tests to have the correct shard timeout value on Android.
2016-08-15 09:29:19 +00:00
examples
Remove LooperExecutor and replace it with built-in HandlerThread.
2016-08-09 09:03:41 +00:00
libjingle
GN: Update xmpp and p2p to match Chromium build
2016-06-18 11:23:08 +00:00
media
Remove old methods in AudioTransport, make it pass a gn build
2016-08-15 18:46:28 +00:00
modules
Skip AUD while extracting SPS and PPS on iOS.
2016-08-16 01:56:23 +00:00
p2p
Fixing problems with ICE candidate pair prioritization.
2016-08-11 23:05:15 +00:00
pc
Add support for GCM cipher suites from RFC 7714.
2016-08-04 12:20:38 +00:00
sdk
remove unnecessary double allocation
2016-08-15 10:42:08 +00:00
system_wrappers
GN: Update tests to have the correct shard timeout value on Android.
2016-08-15 09:29:19 +00:00
test
GN: Update tests to have the correct shard timeout value on Android.
2016-08-15 09:29:19 +00:00
tools
Clarify some function names in visualization tool.
2016-08-15 15:47:21 +00:00
video
Make variable for selecting if intervals without samples should be included in stats configurable (for rate counters).
2016-08-15 08:20:37 +00:00
voice_engine
Remove old methods in AudioTransport, make it pass a gn build
2016-08-15 18:46:28 +00:00
.gitignore
…
audio_receive_stream.h
Add AudioReceiveStream::SetGain() method and use that in WVoMC::SetOutputVolume().
2016-06-17 15:30:58 +00:00
audio_send_stream.h
Variable audio bitrate.
2016-07-26 11:44:12 +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
Add video_loopback to gn.
2016-08-15 13:29:19 +00:00
call.h
Add periodic logging of video stats.
2016-08-11 15:41:26 +00:00
codereview.settings
…
common_types.cc
FileWrapper[Impl] modifications and actually remove the "Impl" class.
2016-06-15 17:30:18 +00:00
common_types.h
Reland of actor NACK bitrate allocation (patchset #1 id:1 of https://codereview.webrtc.org/2131913003/ )
2016-07-13 16:11:38 +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
New module for the adaptive level controlling functionality in the audio processing module
2016-06-29 22:26:19 +00:00
config.cc
Avoid unnecessary HW video encoder reconfiguration
2016-06-16 19:08:11 +00:00
config.h
Add decoder-specific settings with proper lifetime.
2016-08-05 16:19:33 +00:00
DEPS
Moved creation of AudioDecoderFactory to inside PeerConnectionFactory.
2016-06-13 14:35:01 +00:00
engine_configurations.h
…
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
Add stefan@ to webrtc/OWNERS.
2016-08-10 15:49:59 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests_apk.isolate
…
rtc_unittests.isolate
…
supplement.gypi
Cleanup MIPS specific link configuration
2016-06-20 05:39:25 +00:00
transport.h
…
typedefs.h
[MIPS] Fix build issue for mips64el
2016-06-23 10:52:35 +00:00
video_decoder.h
…
video_encoder.h
Add EncodedImageCallback::OnEncodedImage().
2016-08-03 00:46:47 +00:00
video_engine_tests_apk.isolate
…
video_engine_tests.isolate
…
video_frame.h
Add TODO comments on deprecated VideoFrame methods.
2016-06-22 15:47:52 +00:00
video_receive_stream.h
Add periodic logging of video stats.
2016-08-11 15:41:26 +00:00
video_send_stream.h
Add periodic logging of video stats.
2016-08-11 15:41:26 +00:00
webrtc_examples.gyp
Add GN targets for AppRTC Demo on Android.
2016-06-29 12:55:12 +00:00
webrtc_nonparallel_tests_apk.isolate
…
webrtc_nonparallel_tests.isolate
…
webrtc_perf_tests_apk.isolate
…
webrtc_perf_tests.isolate
Add an HD resolution perf test.
2016-08-11 14:01:03 +00:00
webrtc_tests.gypi
Add video_loopback to gn.
2016-08-15 13:29:19 +00:00
webrtc.gyp
Remove webrtc_all target
2016-06-14 09:09:24 +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.