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kwiberg 98cc88c873 Correctly handle the error case where the CodecId has a negative value 
Negative values should be treated the same as too-large positive values: by returning an empty Maybe.

TBR=henrik.lundin@webrtc.org

Review URL: https://codereview.webrtc.org/1407383010

Cr-Commit-Position: refs/heads/master@{#10509}
2015-11-04 17:56:22 +00:00
..
audio
modules: more interface -> include renames
2015-11-04 07:32:04 +00:00
base
Remove WEBRTC_ANDROID from hardcoded gtest relative path usage.
2015-11-04 07:47:46 +00:00
build
Suppress data races in AudioDeviceLinuxPulse::Init.
2015-11-02 12:42:48 +00:00
call
modules: more interface -> include renames
2015-11-04 07:32:04 +00:00
common_audio
Add aecdump support to audioproc_f.
2015-10-30 06:43:00 +00:00
common_video
modules: more interface -> include renames
2015-11-04 07:32:04 +00:00
examples
modules: more interface -> include renames
2015-11-04 07:32:04 +00:00
libjingle
Switch usage of _DEBUG macro to NDEBUG.
2015-10-30 23:08:54 +00:00
modules
Correctly handle the error case where the CodecId has a negative value
2015-11-04 17:56:22 +00:00
p2p
Remove deprecated IsUnresolved() method from SocketAddress API.
2015-11-03 00:20:28 +00:00
sound
Fix chromium-style warnings in webrtc/sound/.
2015-10-29 11:50:15 +00:00
system_wrappers
Remove system_wrappers dep from field_trial_default and metrics_default.
2015-11-03 17:36:56 +00:00
test
modules: more interface -> include renames
2015-11-04 07:32:04 +00:00
tools
modules: more interface -> include renames
2015-11-04 07:32:04 +00:00
video
Add histograms for send-side delay stats for a sent video stream:
2015-11-04 08:59:10 +00:00
video_engine
modules: more interface -> include renames
2015-11-04 07:32:04 +00:00
voice_engine
Removing trace checks in VoETestManager.
2015-11-04 15:11:52 +00:00
.gitignore
…
audio_receive_stream.h
Re-Land: Implement AudioReceiveStream::GetStats().
2015-10-22 08:49:39 +00:00
audio_send_stream.h
Implement AudioSendStream::GetStats().
2015-10-27 10:35:30 +00:00
BUILD.gn
Revert "Add GN Build file for rtc_sound target."
2015-10-26 16:39:21 +00:00
call.h
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +00:00
codereview.settings
Update lower-level codereview.settings files.
2015-10-06 21:19:34 +00:00
common_types.cc
Separating voice activity flag from audio level in RtpHeaderExtension.
2015-08-10 13:08:46 +00:00
common_types.h
Adding support for simulcast and spatial layers into VideoQualityTest
2015-11-02 15:23:24 +00:00
common.gyp
…
common.h
Add RTC_ prefix to contructormagic macros.
2015-09-16 12:37:52 +00:00
config.cc
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +00:00
config.h
Adding support for simulcast and spatial layers into VideoQualityTest
2015-11-02 15:23:24 +00:00
engine_configurations.h
Don't link with audio codecs that we don't use
2015-10-01 04:54:29 +00:00
frame_callback.h
Unify FrameType and VideoFrameType.
2015-10-19 09:39:15 +00:00
libjingle_examples.gyp
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
…
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests.isolate
…
stream.h
Wire up PacketTime to ReceiveStreams.
2015-09-08 12:36:23 +00:00
supplement.gypi
…
transport.h
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +00:00
typedefs.h
Use suffixed {uint,int}{8,16,32,64}_t types.
2015-10-07 10:23:32 +00:00
video_decoder.h
…
video_encoder.h
Remove redudant encoder rate calls.
2015-10-29 15:30:29 +00:00
video_engine_tests.isolate
…
video_frame.h
Remove VideoFrameType aliases for FrameType.
2015-10-23 13:58:27 +00:00
video_receive_stream.h
Fix RTPPayloadRegistry to correctly restore RTX, if a valid mapping exists.
2015-10-14 18:29:56 +00:00
video_renderer.h
…
video_send_stream.h
Reland of Collecting encode_time_ms for each frame (patchset #1 id:1 of https://codereview.webrtc.org/1383283005/ )
2015-10-07 09:43:25 +00:00
webrtc_examples.gyp
…
webrtc_nonparallel_tests.isolate
Create isolate files for nonparallel tests.
2015-10-29 14:29:15 +00:00
webrtc_perf_tests.isolate
…
webrtc_tests.gypi
Create isolate files for nonparallel tests.
2015-10-29 14:29:15 +00:00
webrtc.gyp
Remove references to libpeerconnection.
2015-10-08 12:41:05 +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.