platform-external-webrtc

Files

henrik.lundin@webrtc.org 74767401f2 Fix a bug preventing FilePlayer from playing encoded wav files

A bug in ACM2 prevented decoding and playout of wav files where the
audio data was encoded (i.e., not just linear PCM 16 bit data).

This CL fixes the issue, and adds a unit test for the FilePlayer.

BUG=3386
R=henrike@webrtc.org, tina.legrand@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21499005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6248 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-05-26 13:37:45 +00:00

base

Rebase webrtc/base with r6232:

2014-05-23 18:40:46 +00:00

build

removed webrtc_base_tests_utils from merge libs as it was breaking some builds.

2014-05-15 21:45:09 +00:00

common_audio

Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.

2014-05-21 21:18:46 +00:00

common_video

Added include of assert.h for files calling assert but missing the include.

2014-04-29 17:54:17 +00:00

examples

WebRTCDemo: clean the error message due to API clean up and add ability to route the audio through all three outputs, headset/earpiece/loudspeaker

2014-05-21 03:37:45 +00:00

modules

Fix a bug preventing FilePlayer from playing encoded wav files

2014-05-26 13:37:45 +00:00

overrides/webrtc/base

Rename webrtc/base's IS_ALIGNED macro to RTC_IS_ALIGNED to avoid conflict between webrtc/base/basictypes.h and third_party/.../vpx_codec.h.

2014-05-21 16:52:14 +00:00

system_wrappers

Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.

2014-05-21 21:18:46 +00:00

test

Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.

2014-05-21 21:18:46 +00:00

tools

Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.

2014-05-21 21:18:46 +00:00

video

First incoming packet was not accounted for in receive stats. Changed call order for incoming packet to receive statistics class.

2014-05-26 13:06:04 +00:00

video_engine

First incoming packet was not accounted for in receive stats. Changed call order for incoming packet to receive statistics class.

2014-05-26 13:06:04 +00:00

voice_engine

1. Make a clear distinction between codec internal FEC and RED, confusing mentioning of FEC in the old codes is replaced by RED

2014-05-23 15:16:51 +00:00

.gitignore

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

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

call.h

Add DeliveryStatus enum to DeliverPacket().

2014-05-14 13:57:12 +00:00

common_types.h

Add ToString() to VideoSendStream::Config.

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

common.gyp

Add ToString() to VideoSendStream::Config.

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

common.h

Add a Config class interface to AudioProcessing for passing options.

2013-07-25 18:28:29 +00:00

config.cc

Add ToString() to VideoSendStream::Config.

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

config.h

Add ToString() to VideoSendStream::Config.

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

engine_configurations.h

Removes parts of the webrtc::VoEExternalMedia sub API as part of a clean-up operation where the goal is to remove unused APIs.

2014-05-12 12:19:19 +00:00

experiments.h

Adding API for setting bandwidth estimation configurations.

2014-03-25 10:37:31 +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

Consolidate all third party licenses in LICENSE_THIRD_PARTY.

2013-05-05 18:54:10 +00:00

OWNERS

Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.

2014-04-14 20:08:03 +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

…

supplement.gypi

Roll chromium_revision 260462:266514

2014-04-29 09:36:40 +00:00

transport.h

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

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

typedefs.h

Remove ALLOW_UNUSED.

2014-05-05 18:18:02 +00:00

video_engine_tests.isolate

Merge metrics_unittests into video_engine_tests.

2013-12-13 14:31:47 +00:00

video_receive_stream.h

Rename Start/Stop in Video{Send,Receive}Streams.

2014-04-24 11:13:21 +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

Revert "Revert "Remove VideoSendStreamInput::PutFrame.""

2014-05-23 13:03:45 +00:00

webrtc_examples.gyp

Add webrtc field trials API.

2014-05-14 12:24:04 +00:00

webrtc_perf_tests.isolate

Move realtime tests to webrtc_perf_tests.

2013-12-13 12:48:05 +00:00

webrtc_tests.gypi

Add webrtc field trials API.

2014-05-14 12:24:04 +00:00

webrtc.gyp

Add ToString() to VideoSendStream::Config.

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