webrtc_m130

History

andrew@webrtc.org 28e82bfec6 Replace Resampler with PushResampler in transmit_mixer.

* VoE can now exchange 44.1 kHz audio with AudioDevice.
* Changes still required in AudioDevice to remove the 44 kHz workarounds and
enable native 44.1 kHz.

BUG=webrtc:1395
TESTED=voe_cmd_test loopback running through codecs using all combinations of {8, 16, 32} kHz and {1, 2} channels, and Opus (48 kHz, stereo)
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1373004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3930 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-05-02 00:30:36 +00:00

build

Move WEBRTC_THREAD_RR and WEBRTC_CLOCK_TYPE_REALTIME to system_wrappers.gypi .

2013-04-30 22:53:51 +00:00

common_audio

Consolidate common_audio into a single target.

2013-04-30 23:43:26 +00:00

common_video

WebRtc_Word32 -> int32_t in common_video.

2013-04-09 13:36:08 +00:00

modules

Consolidate common_audio into a single target.

2013-04-30 23:43:26 +00:00

system_wrappers

Move WEBRTC_THREAD_RR and WEBRTC_CLOCK_TYPE_REALTIME to system_wrappers.gypi .

2013-04-30 22:53:51 +00:00

test

Revert "Updating test file contents to emmastjernloef"

2013-04-17 14:59:28 +00:00

tools

Fix frame_editing_unittest reference file handling.

2013-03-11 15:30:33 +00:00

video_engine

Consolidate common_audio into a single target.

2013-04-30 23:43:26 +00:00

voice_engine

Replace Resampler with PushResampler in transmit_mixer.

2013-05-02 00:30:36 +00:00

.gitignore

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

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

common_types.h

Adding extra options to interact with external encoder/decoder.

2013-04-23 22:50:53 +00:00

engine_configurations.h

Remove WEBRTC_*_ENGINE_NETWORK_API use

2013-04-04 19:51:42 +00:00

LICENSE

Move src/ -> webrtc/

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

LICENSE_THIRD_PARTY

Move src/ -> webrtc/

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

PATENTS

Move src/ -> webrtc/

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

README.chromium

Move src/ -> webrtc/

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

typedefs.h

Replace legacy G_CONST with const.

2013-04-10 18:06:57 +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.