platform-external-webrtc

Files

bjornv@webrtc.org 758d6d431e audio_processing/aecm: Removed usage of macro WEBRTC_SPL_MUL_16_16

The macro is in C defined as
#define WEBRTC_SPL_MUL_16_16(a, b) ((int32_t) (((int16_t)(a)) * ((int16_t)(b))))
(For definition on ARMv7 and MIPS, see common_audio/signal_processing/include/spl_inl_armv7.h and common_audio/signal_processing/include/spl_inl_mips.h)

The replacement consists of
- avoiding casts to int16_t if inputs already are int16_t
- adding explicit cast to <type> if result is assigned to <type> (other than int or int32_t)

BUG=3348, 3353
TESTED=locally on Mac and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8025 4adac7df-926f-26a2-2b94-8c16560cd09d

2015-01-08 17:52:56 +00:00

base

Add a dummy implemenation of SChannelAdapter::SetMode that makes sure that StartSSL fails if the mode is set to DTLS.

2015-01-07 18:01:07 +00:00

build

Suppress races in event tracing code.

2015-01-08 12:38:42 +00:00

common_audio

Remove WebRtcSpl_ScaleAndAddVectorsWithRoundNeon

2015-01-07 21:34:23 +00:00

common_video

Use size_t more consistently for packet/payload lengths.

2014-11-20 22:28:14 +00:00

examples

Update Android projects to API level 21.

2014-10-31 23:26:10 +00:00

libjingle

Move the Jingle-specific network code into webrtc/libjingle.

2014-12-23 22:14:15 +00:00

modules

audio_processing/aecm: Removed usage of macro WEBRTC_SPL_MUL_16_16

2015-01-08 17:52:56 +00:00

overrides

Remove or rename typedefs with _t prefixes.

2014-12-17 13:43:55 +00:00

p2p

Disable flaky RelayServerTest.TestExpiration on all platforms.

2015-01-05 23:56:19 +00:00

sound

rtc_unittest: turned sound's test gyp into gypi to speed up GYP generation.

2014-10-01 16:33:03 +00:00

system_wrappers

Hack clock_unittest fix for parallel execution.

2015-01-05 19:08:58 +00:00

test

Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess.

2014-12-15 22:09:40 +00:00

tools

Merge in AGC manager and AGC tools.

2014-12-15 16:33:16 +00:00

video

Fix the ramp-up-down-up test which was using ts-offset extension with the abs-send-time estimator.

2014-12-19 18:00:21 +00:00

video_engine

Add histograms stats for sent/received fraction loss for a stream:

2015-01-08 07:50:56 +00:00

voice_engine

Rtp-Rtcp sender cleanup.

2014-12-19 13:49:55 +00:00

.gitignore

…

BUILD.gn

Enabling building with NEON on ARM64

2014-11-26 17:01:40 +00:00

call.h

Wire up RTT statistics to webrtc::Call.

2014-12-11 13:26:09 +00:00

codereview.settings

Add codereview.settings to the /webrtc subdirectory

2014-12-05 13:43:35 +00:00

common_types.h

Refactor some receive-side stats.

2014-12-18 13:50:16 +00:00

common.gyp

Add ToString() to VideoSendStream::Config.

2014-05-15 09:35:06 +00:00

common.h

…

config.cc

Log formatting fix for VideoEncoderConfig.

2014-11-06 09:35:08 +00:00

config.h

Refactor some receive-side stats.

2014-12-18 13:50:16 +00:00

engine_configurations.h

Add VP9 codec to VCM and vie_auto_test.

2014-11-01 06:10:48 +00:00

experiments.h

Remove no longer used SkipEncodingUnusedStreams.

2014-07-22 07:17:17 +00:00

frame_callback.h

Wire up statistics in video receive stream of new API

2014-02-07 12:06:29 +00:00

LICENSE

…

LICENSE_THIRD_PARTY

…

OWNERS

GN: Add BUILD.gn files + kjellander to OWNERS

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

PATENTS

…

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

…

rtc_unittests.isolate

Update all .isolate files for the new format.

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

supplement.gypi

Roll chromium_revision 289723:291647

2014-08-25 14:16:32 +00:00

transport.h

Rename newapi::Transport::SendRTP()->SendRtp().

2013-11-20 12:17:04 +00:00

typedefs.h

Annotate COMPILE_ASSERT with __attribute__((unused)).

2014-11-17 13:47:38 +00:00

video_decoder.h

Add support for VP9 in webrtc::Call and video_loopback.

2014-11-04 19:41:15 +00:00

video_encoder.h

Report encoded frame size in VideoSendStream.

2014-12-01 15:23:21 +00:00

video_engine_tests.isolate

Update isolate files for Android APK tests.

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

video_frame.h

Report encoded frame size in VideoSendStream.

2014-12-01 15:23:21 +00:00

video_receive_stream.h

Wire up bandwidth stats to the new API and webrtcvideoengine2.

2014-11-05 14:05:29 +00:00

video_renderer.h

Separate Call API/build files from video_engine/.

2013-10-28 16:32:01 +00:00

video_send_stream.h

Enable payload-based padding by default and remove the API.

2014-12-19 15:33:17 +00:00

webrtc_examples.gyp

Add macros and APIs for webrtc histograms.

2014-10-23 12:57:56 +00:00

webrtc_perf_tests.isolate

Update isolate files for Android APK tests.

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

webrtc_tests.gypi

Add AGC manager tests.

2014-12-16 14:48:47 +00:00

webrtc.gyp

Merge in AGC manager and AGC tools.

2014-12-15 16:33:16 +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.