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Karl Wiberg 88de4792d0 AudioEncoderIsac: Print error code if CHECK for successful encoding fails 
This will hopefully make the crash in bug 4577 easier to understand if
it happens again.

BUG=4577
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/52389004

Cr-Commit-Position: refs/heads/master@{#9100}
2015-04-28 13:43:43 +00:00
..
base
Move CriticalSection into rtc_base_approved.
2015-04-27 15:39:00 +00:00
build
GYP variables for building expat, icu, libsrtp, usrsctp
2015-04-27 12:38:37 +00:00
common_audio
Add 3-band filter-bank implementation
2015-04-27 18:34:16 +00:00
common_video
Remove i420_video_frame.h from common_video.gyp
2015-04-10 12:40:21 +00:00
examples/android
Use short include paths for icu headers.
2015-04-23 06:58:02 +00:00
libjingle
GYP variables for building expat, icu, libsrtp, usrsctp
2015-04-27 12:38:37 +00:00
modules
AudioEncoderIsac: Print error code if CHECK for successful encoding fails
2015-04-28 13:43:43 +00:00
overrides
Cleanup: Remove DISALLOW_EVIL_CONSTRUCTORS macro.
2015-04-04 23:56:56 +00:00
p2p
STUN allocation should not be disabled when using shared port and TURN servers are provided.
2015-04-27 18:46:58 +00:00
sound
Remove henrike@ from OWNERS
2015-04-01 15:08:49 +00:00
system_wrappers
Remove webrtc/base/move.h, and make types move-only manually
2015-04-22 17:43:59 +00:00
test
Make sure histograms in jitter buffer are only updated if running.
2015-04-24 06:51:52 +00:00
tools
Disables mic bump-up level if not built with chromium
2015-04-22 04:39:47 +00:00
video
Fix dangling pointer in screenshare_loopback
2015-04-28 12:00:29 +00:00
video_engine
Prevent zero-timestamps in captured_frame_.
2015-04-28 08:48:13 +00:00
voice_engine
VoE: VoEBase unit test
2015-04-21 09:39:41 +00:00
.gitignore
…
BUILD.gn
Let Chromium declare the mips_dsp_rev build variable.
2015-03-04 09:51:17 +00:00
call.h
Remove ViE interface usage from VideoReceiveStream.
2015-04-09 12:35:46 +00:00
codereview.settings
Add codereview.settings to the /webrtc subdirectory
2014-12-05 13:43:35 +00:00
common_types.cc
Parsing of transport wide sequence number rtp extension header.
2015-03-17 14:33:46 +00:00
common_types.h
Parsing of transport wide sequence number rtp extension header.
2015-03-17 14:33:46 +00:00
common.gyp
Fix style violations in common_types.h and config.h
2015-02-26 14:01:28 +00:00
common.h
…
config.cc
Set correct encoder-specific settings for vpx in the new API.
2015-04-28 08:01:14 +00:00
config.h
Set correct encoder-specific settings for vpx in the new API.
2015-04-28 08:01:14 +00:00
engine_configurations.h
Delete all codec-specific subclasses of ACMGenericCodec
2015-02-23 09:26:51 +00:00
experiments.h
…
frame_callback.h
Changing include guard in frame_callback.h.
2015-02-03 14:51:39 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Add the Ooura FFT to RealFourier.
2015-03-19 20:07:43 +00:00
OWNERS
…
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests.isolate
…
supplement.gypi
Move target_subarch from gyp_webrtc to supplement.gypi
2015-04-16 07:24:23 +00:00
transport.h
…
typedefs.h
Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
2015-03-04 13:04:54 +00:00
video_decoder.h
Revert "Changed argument occurences of const I420VideoFrame* to const I420VideoFrame& and non-const I420VideoFrame& to I420VideoFrame*."
2015-03-16 13:48:18 +00:00
video_encoder.h
Dynamic resolution change for VP8 HW encode.
2015-04-21 22:29:53 +00:00
video_engine_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
video_frame.h
Reland "Remove usage of webrtc::NativeHandle since is just adds an extra level of indirection.""
2015-04-09 11:44:19 +00:00
video_receive_stream.h
Use IncomingVideoStream in VideoReceiveStream.
2015-04-24 13:15:40 +00:00
video_renderer.h
Use VideoReceiveStream as an ExternalRenderer.
2015-02-09 15:15:24 +00:00
video_send_stream.h
Reland "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame.""
2015-03-18 09:51:44 +00:00
webrtc_examples.gyp
GYP variables for building expat, icu, libsrtp, usrsctp
2015-04-27 12:38:37 +00:00
webrtc_perf_tests.isolate
Offline screenshare quality test, plus loopback.
2015-02-18 12:46:44 +00:00
webrtc_tests.gypi
Add a BWE fairness test.
2015-04-16 18:20:26 +00:00
webrtc.gyp
Rename GYP and GN targets for video capture+render.
2015-02-11 07:47:47 +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.