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minyue 275d255e21 Adding debug dump test. 
This test is to verify that the debug dump can perfectly reproduce APM states if the recording is made from the first input sample.

BUG=

Review URL: https://codereview.webrtc.org/1393353003

Cr-Commit-Position: refs/heads/master@{#10506}
2015-11-04 14:24:02 +00:00
..
audio
modules: more interface -> include renames
2015-11-04 07:32:04 +00:00
base
Remove WEBRTC_ANDROID from hardcoded gtest relative path usage.
2015-11-04 07:47:46 +00:00
build
Suppress data races in AudioDeviceLinuxPulse::Init.
2015-11-02 12:42:48 +00:00
call
modules: more interface -> include renames
2015-11-04 07:32:04 +00:00
common_audio
Add aecdump support to audioproc_f.
2015-10-30 06:43:00 +00:00
common_video
modules: more interface -> include renames
2015-11-04 07:32:04 +00:00
examples
modules: more interface -> include renames
2015-11-04 07:32:04 +00:00
libjingle
Switch usage of _DEBUG macro to NDEBUG.
2015-10-30 23:08:54 +00:00
modules
Adding debug dump test.
2015-11-04 14:24:02 +00:00
p2p
Remove deprecated IsUnresolved() method from SocketAddress API.
2015-11-03 00:20:28 +00:00
sound
Fix chromium-style warnings in webrtc/sound/.
2015-10-29 11:50:15 +00:00
system_wrappers
Remove system_wrappers dep from field_trial_default and metrics_default.
2015-11-03 17:36:56 +00:00
test
modules: more interface -> include renames
2015-11-04 07:32:04 +00:00
tools
modules: more interface -> include renames
2015-11-04 07:32:04 +00:00
video
Add histograms for send-side delay stats for a sent video stream:
2015-11-04 08:59:10 +00:00
video_engine
modules: more interface -> include renames
2015-11-04 07:32:04 +00:00
voice_engine
modules: more interface -> include renames
2015-11-04 07:32:04 +00:00
.gitignore
…
audio_receive_stream.h
Re-Land: Implement AudioReceiveStream::GetStats().
2015-10-22 08:49:39 +00:00
audio_send_stream.h
Implement AudioSendStream::GetStats().
2015-10-27 10:35:30 +00:00
BUILD.gn
Revert "Add GN Build file for rtc_sound target."
2015-10-26 16:39:21 +00:00
call.h
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +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
Adding support for simulcast and spatial layers into VideoQualityTest
2015-11-02 15:23:24 +00:00
common.gyp
Fix style violations in common_types.h and config.h
2015-02-26 14:01:28 +00:00
common.h
Add RTC_ prefix to contructormagic macros.
2015-09-16 12:37:52 +00:00
config.cc
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +00:00
config.h
Adding support for simulcast and spatial layers into VideoQualityTest
2015-11-02 15:23:24 +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
Unify FrameType and VideoFrameType.
2015-10-19 09:39:15 +00:00
libjingle_examples.gyp
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Add the Ooura FFT to RealFourier.
2015-03-19 20:07:43 +00:00
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
Move target_subarch from gyp_webrtc to supplement.gypi
2015-04-16 07:24:23 +00:00
transport.h
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +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
Remove redudant encoder rate calls.
2015-10-29 15:30:29 +00:00
video_engine_tests.isolate
…
video_frame.h
Remove VideoFrameType aliases for FrameType.
2015-10-23 13:58:27 +00:00
video_receive_stream.h
Fix RTPPayloadRegistry to correctly restore RTX, if a valid mapping exists.
2015-10-14 18:29:56 +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_nonparallel_tests.isolate
Create isolate files for nonparallel tests.
2015-10-29 14:29:15 +00:00
webrtc_perf_tests.isolate
Enable GoogleWifiTrace3Mbps simulations.
2015-05-04 12:26:26 +00:00
webrtc_tests.gypi
Create isolate files for nonparallel tests.
2015-10-29 14:29:15 +00:00
webrtc.gyp
Remove references to libpeerconnection.
2015-10-08 12:41:05 +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.