platform-external-webrtc

Files

minyue 9846845da6 Calculate audio levels in AEC in time domain.

In AEC, audio levels are calculated in frequency domain. This makes the calculation dependent on FFT. We now make the calculation performed in time domain. The complexity is the same, but the dependence on FFT is removed.

BUG=

Review URL: https://codereview.webrtc.org/1542573002

Cr-Commit-Position: refs/heads/master@{#11357}

2016-01-22 13:46:47 +00:00

api

Update API for Objective-C RTCMediaStream.

2016-01-22 00:14:23 +00:00

audio

Enable transport seq num extension on receive channel to suppress log warning.

2016-01-21 14:32:48 +00:00

base

CriticalSection: Use types+methods from base/platform_thread*.*.

2016-01-22 13:13:38 +00:00

build

rtc_use_h264 flag (replacing use_third_party_h264 flag) for building OpenH264/FFmpeg, false by default but can be overridden in supplement.gypi and build_overrides/webrtc.gni.

2016-01-21 11:34:47 +00:00

call

Fixes a bug which incorrectly logs incoming RTCP as outgoing.

2016-01-21 13:42:10 +00:00

common_audio

Convert channel counts to size_t.

2016-01-13 00:26:55 +00:00

common_video

Added EncodedImage::GetBufferPaddingBytes.

2016-01-21 13:43:18 +00:00

examples

Add a new interface for creating a udp socket in which it binds the socket to a network if the network handle is set.

2016-01-15 22:49:15 +00:00

libjingle

Remove DISABLED_ON_ macros.

2016-01-04 21:44:16 +00:00

modules

Calculate audio levels in AEC in time domain.

2016-01-22 13:46:47 +00:00

p2p

Remove two dead 'using' instances.

2016-01-18 23:24:36 +00:00

sound

OWNERS: Add * to .gyp{i,} everywhere.

2015-12-16 19:44:39 +00:00

system_wrappers

Remove implementation of CriticalSectionWrapper and use rtc::CriticalSection

2016-01-22 13:26:19 +00:00

test

Add stefan@webrtc.org to webrtc/test/OWNERS.

2016-01-21 14:33:03 +00:00

tools

Convert channel counts to size_t.

2016-01-13 00:26:55 +00:00

video

Swap use of CriticalSectionWrapper for rtc::CriticalSection in webrtc/video.

2016-01-21 22:25:12 +00:00

voice_engine

Swap use of CriticalSectionWrapper with rtc::CriticalSection in voice_engine/

2016-01-21 18:37:44 +00:00

.gitignore

Flesh out webrtc/.gitignore

2015-11-16 19:02:02 +00:00

audio_receive_stream.h

Storing raw audio sink for default audio track.

2016-01-15 17:20:08 +00:00

audio_send_stream.h

Prepare the AudioSendStream to be hooked up to send-side BWE.

2015-12-07 09:26:32 +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

Re-land: "Use an explicit identifier in Config"

2016-01-14 12:32:51 +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

Improve informative message in codereview.settings.

2015-11-10 10:08:36 +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

Remove extra_options from VideoCodec.

2016-01-19 15:26:24 +00:00

common.gyp

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common.h

Re-land: "Use an explicit identifier in Config"

2016-01-14 12:32:51 +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

Re-land: "Use an explicit identifier in Config"

2016-01-14 12:32:51 +00:00

engine_configurations.h

Remove VIDEOCODEC_* from engine_configurations.h.

2015-11-24 12:56:05 +00:00

frame_callback.h

Unify FrameType and VideoFrameType.

2015-10-19 09:39:15 +00:00

LICENSE

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LICENSE_THIRD_PARTY

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OWNERS

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PATENTS

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PRESUBMIT.py

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README.chromium

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rtc_unittests.isolate

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stream.h

Wire up PacketTime to ReceiveStreams.

2015-09-08 12:36:23 +00:00

supplement.gypi

Roll chromium_revision 68cf0b8..aa8e58a (361406:361601)

2015-11-25 21:20:11 +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

Update attributes to match gclibc's ansidecl.h

2016-01-08 12:51:47 +00:00

video_decoder.h

Expose codec implementation names in stats.

2015-12-18 15:01:23 +00:00

video_encoder.h

Expose codec implementation names in stats.

2015-12-18 15:01:23 +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

Added EncodedImage::GetBufferPaddingBytes.

2016-01-21 13:43:18 +00:00

video_receive_stream.h

Expose codec implementation names in stats.

2015-12-18 15:01:23 +00:00

video_renderer.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

video_send_stream.h

Expose codec implementation names in stats.

2015-12-18 15:01:23 +00:00

webrtc_examples.gyp

Disable warnings failing when using Clang on Windows.

2015-12-16 22:05:36 +00:00

webrtc_nonparallel_tests.isolate

Create isolate files for nonparallel tests.

2015-10-29 14:29:15 +00:00

webrtc_perf_tests.isolate

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webrtc_tests.gypi

Remove extra_options from VideoCodec.

2016-01-19 15:26:24 +00:00

webrtc.gyp

Remove 'video_engine_core_unittests' binary.

2015-12-04 09:59:02 +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.