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Alex Glaznev fddf6e526c Use WebRTC logging in MediaCodec JNI code. 
Also enable HW encoder scaling in AppRTCDemo.

R=wzh@webrtc.org

Review URL: https://codereview.webrtc.org/1396653002 .

Cr-Commit-Position: refs/heads/master@{#10205}
2015-10-07 23:51:20 +00:00
..
audio
Split webrtc/video into webrtc/{audio,call,video}.
2015-09-25 11:58:39 +00:00
base
Get rid of SCHANNEL code.
2015-10-07 11:58:00 +00:00
build
Don't link with audio codecs that we don't use
2015-10-01 04:54:29 +00:00
call
Add a PacketOptions struct to webrtc::Transport.
2015-10-02 10:39:40 +00:00
common_audio
Add RTC_ prefix to (D)CHECKs and related macros.
2015-09-17 07:24:51 +00:00
common_video
Add RTC_ prefix to (D)CHECKs and related macros.
2015-09-17 07:24:51 +00:00
examples
Use WebRTC logging in MediaCodec JNI code.
2015-10-07 23:51:20 +00:00
libjingle
Use suffixed {uint,int}{8,16,32,64}_t types.
2015-10-07 10:23:32 +00:00
modules
Read the number of TLs for VP9 too + cleanup
2015-10-07 13:43:43 +00:00
p2p
Use suffixed {uint,int}{8,16,32,64}_t types.
2015-10-07 10:23:32 +00:00
sound
Add RTC_ prefix to contructormagic macros.
2015-09-16 12:37:52 +00:00
system_wrappers
Add RTC_ prefix to (D)CHECKs and related macros.
2015-09-17 07:24:51 +00:00
test
Remove kSkipFrame usage.
2015-10-06 12:05:03 +00:00
tools
Fix minor GYP error in webrtc/tools/internal_tools.gyp
2015-10-07 12:51:40 +00:00
video
Read the number of TLs for VP9 too + cleanup
2015-10-07 13:43:43 +00:00
video_engine
Add delay metric (includes network delay (rtt/2) + jitter delay + decode time + render delay):
2015-10-06 11:08:22 +00:00
voice_engine
Use suffixed {uint,int}{8,16,32,64}_t types.
2015-10-07 10:23:32 +00:00
.gitignore
…
audio_receive_stream.h
Align new VoE API with design.
2015-10-01 15:13:46 +00:00
audio_send_stream.h
Align new VoE API with design.
2015-10-01 15:13:46 +00:00
BUILD.gn
Update build files to use webrtc_overrides in Chromium instead of overrides.
2015-09-30 10:49:11 +00:00
call.h
Align new VoE API with design.
2015-10-01 15:13:46 +00:00
codereview.settings
Update lower-level codereview.settings files.
2015-10-06 21:19:34 +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
Unify newapi::RtcpMode and RTCPMethod.
2015-10-02 09:37:18 +00:00
common.gyp
…
common.h
Add RTC_ prefix to contructormagic macros.
2015-09-16 12:37:52 +00:00
config.cc
Add support for transport wide sequence numbers
2015-08-03 11:38:48 +00:00
config.h
Add support for transport wide sequence numbers
2015-08-03 11:38:48 +00:00
engine_configurations.h
Don't link with audio codecs that we don't use
2015-10-01 04:54:29 +00:00
frame_callback.h
Rename I420VideoFrame to VideoFrame.
2015-05-30 00:21:56 +00:00
libjingle_examples.gyp
Fix build when using Xcode 7 which contains .tbd files instead of .dylib
2015-09-05 03:01:08 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
…
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests.isolate
…
stream.h
Wire up PacketTime to ReceiveStreams.
2015-09-08 12:36:23 +00:00
supplement.gypi
…
transport.h
Add a PacketOptions struct to webrtc::Transport.
2015-10-02 10:39:40 +00:00
typedefs.h
Use suffixed {uint,int}{8,16,32,64}_t types.
2015-10-07 10:23:32 +00:00
video_decoder.h
iOS HW H264 support.
2015-06-29 21:35:08 +00:00
video_encoder.h
Allow for framerate reduction for HW encoder.
2015-07-13 23:26:40 +00:00
video_engine_tests.isolate
…
video_frame.h
Remove kSkipFrame usage.
2015-10-06 12:05:03 +00:00
video_receive_stream.h
Unify newapi::RtcpMode and RTCPMethod.
2015-10-02 09:37:18 +00:00
video_renderer.h
Rename I420VideoFrame to VideoFrame.
2015-05-30 00:21:56 +00:00
video_send_stream.h
Reland of Collecting encode_time_ms for each frame (patchset #1 id:1 of https://codereview.webrtc.org/1383283005/ )
2015-10-07 09:43:25 +00:00
webrtc_examples.gyp
Adding support for OpenSL ES output in native WebRTC
2015-05-18 14:49:04 +00:00
webrtc_perf_tests.isolate
Enable GoogleWifiTrace3Mbps simulations.
2015-05-04 12:26:26 +00:00
webrtc_tests.gypi
Remove AgcManager.
2015-09-29 22:45:23 +00:00
webrtc.gyp
Split webrtc/video into webrtc/{audio,call,video}.
2015-09-25 11:58:39 +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.