webrtc_m130

History

peah 1f1912d1f0 Added unittest of the locking functionality in the audio processing module

The test is currently disabled as it takes too long to run in a coffe-cup manner

BUG=webrtc:5099

Review URL: https://codereview.webrtc.org/1394803002

Cr-Commit-Position: refs/heads/master@{#10560}

2015-11-09 11:13:25 +00:00

audio

Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats().

2015-11-06 23:34:58 +00:00

base

Pass audio to AudioEncoder::Encode() in an ArrayView

2015-11-06 09:21:39 +00:00

build

Remove ICU usage from jni_helpers.cc.

2015-11-06 21:56:11 +00:00

call

Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats().

2015-11-06 23:34:58 +00:00

common_audio

Revert of Add aecdump support to audioproc_f. (patchset #8 id:200001 of https://codereview.webrtc.org/1409943002/ )

2015-11-05 20:33:25 +00:00

common_video

modules: more interface -> include renames

2015-11-04 07:32:04 +00:00

examples

Remove ICU usage from jni_helpers.cc.

2015-11-06 21:56:11 +00:00

libjingle

Switch usage of _DEBUG macro to NDEBUG.

2015-10-30 23:08:54 +00:00

modules

Added unittest of the locking functionality in the audio processing module

2015-11-09 11:13:25 +00:00

p2p

Stop a session when a new connection becomes writable.

2015-11-04 20:07:49 +00:00

sound

Fix chromium-style warnings in webrtc/sound/.

2015-10-29 11:50:15 +00:00

system_wrappers

Remove old system_wrappers event_tracer.h.

2015-11-06 10:16:56 +00:00

test

Change to use local Random object instead of global rand() in the RtcEventLog unit test.

2015-11-06 13:14:01 +00:00

tools

modules: more interface -> include renames

2015-11-04 07:32:04 +00:00

video

Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats().

2015-11-06 23:34:58 +00:00

video_engine

Removed vie_defines.h

2015-11-06 12:33:56 +00:00

voice_engine

Removing trace checks in VoETestManager.

2015-11-04 15:11:52 +00:00

.gitignore

.gitignore: Add *.mk, created as part of ChromiumOS build

2013-01-04 21:25:42 +00:00

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

audio_state.h

Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats().

2015-11-06 23:34:58 +00:00

BUILD.gn

Remove unused crypto.gni import.

2015-11-04 22:44:04 +00:00

call.h

Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats().

2015-11-06 23:34:58 +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

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

LICENSE_THIRD_PARTY

Add the Ooura FFT to RealFourier.

2015-03-19 20:07:43 +00:00

OWNERS

GN: Add BUILD.gn files + kjellander to OWNERS

2014-06-23 19:21:07 +00:00

PATENTS

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

PRESUBMIT.py

PRESUBMIT.py: accept variants on the copyright message that are present in the codebase.

2014-05-23 17:27:18 +00:00

README.chromium

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

rtc_unittests.isolate

Update all .isolate files for the new format.

2014-10-31 18:08:09 +00:00

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

Update isolate files for Android APK tests.

2014-11-13 08:35:05 +00:00

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

Remove ICU usage from jni_helpers.cc.

2015-11-06 21:56:11 +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

Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats().

2015-11-06 23:34:58 +00:00

webrtc.gyp

Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats().

2015-11-06 23:34:58 +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.