henrik.lundin@webrtc.org c8e98187d1 Receiver bit-exactness test for AudioCoding Module ...

This CL introduces a bit-exactness test for the receive-side of the
AudioCoding Module. The main part of the test is done in the helper
class AcmReceiveTest. The test is executed from the test fixture
AcmReceiverBitExactness.

The test inserts packets from a pre-encoded RTP file. The output is
summed up into a checksum, which is verified versus a reference at the
end of the test. Alternatively, if the flag --generate_output is given,
the output is written to a file for subjective verification.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6549 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-06-26 19:07:04 +00:00

data

Noise suppression: Change signature to work on floats instead of ints

2014-06-10 11:13:09 +00:00

resources

Remove unused resource

2014-06-11 13:59:44 +00:00

talk

Implement SetSendCodecs() unit tests for WebRtcVideoChannel2.

2014-06-26 08:49:03 +00:00

third_party

Added mirror of gtest-parallel.

2014-05-23 09:37:29 +00:00

tools

talk/base and webrtc/base suppression had the same names for their suppressions which is not allowed. Renamed the talk/base ones as they are going away.

2014-06-24 14:24:37 +00:00

webrtc

Receiver bit-exactness test for AudioCoding Module

2014-06-26 19:07:04 +00:00

.clang-format

Style-option file for clang-format.

2013-11-25 15:30:37 +00:00

.gitignore

Add Chromium's src/buildtools to DEPS.

2014-06-26 14:02:16 +00:00

all.gyp

Add a check to all.gyp to respect the include_tests variable.

2014-06-17 16:10:20 +00:00

android-webrtc.mk

Changed a flag in ARM build for platform build.

2012-09-08 00:32:59 +00:00

Android.mk

Added buildbot benchmarking in iSAC and APM into Android platform build.

2012-12-07 15:26:28 +00:00

AUTHORS

AppRTCDemo was blocking the main thread for network requests. This fixes it by making the background queue serial instead of using @synchronize to make the background operations serial.

2014-04-30 00:17:47 +00:00

BUILD.gn

Initial GN work for WebRTC

2014-06-17 08:54:03 +00:00

codereview.settings

Add a root codereview.settings file.

2013-04-24 02:08:50 +00:00

COPYING

Libjingle in webrtc needs updated AUTHORS, COPYING, LICENSE_THIRD_PARTY AND README.

2013-07-16 18:04:56 +00:00

DEPS

Add Chromium's src/buildtools to DEPS.

2014-06-26 14:02:16 +00:00

drover.properties

Add a drover.properties file for reference.

2013-05-30 18:15:54 +00:00

libvpx.mk

Change android makefile to reflect recent file structure change

2011-07-13 17:49:49 +00:00

LICENSE

Updated license path in LICENSE and LICENSE_THIRD_PARTY.

2012-11-22 12:06:21 +00:00

license_template.txt

Update template to follow chromium copyright style

2013-04-24 01:01:28 +00:00

LICENSE_THIRD_PARTY

Libjingle in webrtc needs updated AUTHORS, COPYING, LICENSE_THIRD_PARTY AND README.

2013-07-16 18:04:56 +00:00

OWNERS

GN: Add BUILD.gn files + kjellander to OWNERS

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

PATENTS

Add references to src/ copies for LICENSE etc.

2011-10-26 01:05:07 +00:00

PRESUBMIT.py

PRESUBMIT: Add Android ARM64 and remove Linux TSan

2014-06-08 17:55:51 +00:00

WATCHLISTS

Adding the new video folder and pacer to the wathclist.

2014-06-10 13:59:37 +00:00

Description

The idea is to make CMake build for WebRTC m130 version - for audio processing module

BSD-3-Clause 446 MiB

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