admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/webrtc
History
Henrik Lundin 3e89dbf458 Add AudioEncoder::GetTargetBitrate 
The GetTargetBitrate implementation will return the
target bitrate of the codec. This may differ from the
desired target bitrate, as set by SetTargetBitrate, depending on implementation.

Tests are updated to exercise the new functionality.

R=kwiberg@webrtc.org

Review URL: https://codereview.webrtc.org/1184313002.

Cr-Commit-Position: refs/heads/master@{#9461}
2015-06-18 12:58:46 +00:00
..
base
Define uint64 and int64 using long long.
2015-06-16 14:13:40 +00:00
build
Roll chromium_revision e937e5f..c2239a8 (333350:334133)
2015-06-15 09:10:25 +00:00
common_audio
Add missing include guards for audio_ring_buffer.h. Yikes.
2015-06-17 18:39:44 +00:00
common_video
Re-land "Convert native handles to buffers before encoding."
2015-06-05 09:08:12 +00:00
examples/android/media_demo
VoE: Remove unused interfaces
2015-05-26 08:25:00 +00:00
libjingle
Remove webrtc/libjingle/{examples,session}.
2015-06-12 11:13:46 +00:00
modules
Add AudioEncoder::GetTargetBitrate
2015-06-18 12:58:46 +00:00
overrides
Attempt at fixing error on the Chrome Windows FYI bots.
2015-05-25 19:22:23 +00:00
p2p
Connection resurrected with incorrect candidate type.
2015-06-12 17:12:54 +00:00
sound
Remove henrike@ from OWNERS
2015-04-01 15:08:49 +00:00
system_wrappers
Reformat existing code. There should be no functional effects.
2015-06-11 21:31:48 +00:00
test
Adds support for webrtc::test::ResourcePath on iOS
2015-06-16 08:04:24 +00:00
tools
Reformat existing code. There should be no functional effects.
2015-06-11 21:31:48 +00:00
video
Add sent framerates to histogram stats:
2015-06-16 08:17:09 +00:00
video_engine
Moved check for native frame to VideoReceiveStream::FrameCallback.
2015-06-08 07:26:40 +00:00
voice_engine
Reformat existing code. There should be no functional effects.
2015-06-11 21:31:48 +00:00
.gitignore
…
audio_receive_stream.h
VoE2 API draft
2015-06-08 11:05:07 +00:00
audio_send_stream.h
VoE2 API draft
2015-06-08 11:05:07 +00:00
BUILD.gn
PRESUBMIT: Improve PyLint check and add GN format check.
2015-05-25 10:55:50 +00:00
call.h
C++11 in-class member initialization in Call configs.
2015-06-11 10:38:46 +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
Make Config::default_value leak instead of having an exit-time destructor.
2015-05-23 00:50:33 +00:00
config.cc
Add AudioReceiveStream to Call API.
2015-04-29 13:24:10 +00:00
config.h
Add options for NetEq fast accelerate mode through libjingle
2015-06-01 08:29:55 +00:00
engine_configurations.h
Remove VideoEngine interfaces.
2015-05-12 14:51:08 +00:00
experiments.h
…
frame_callback.h
Rename I420VideoFrame to VideoFrame.
2015-05-30 00:21:56 +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
Update all .isolate files for the new format.
2014-10-31 18:08:09 +00:00
supplement.gypi
Move target_subarch from gyp_webrtc to supplement.gypi
2015-04-16 07:24:23 +00:00
transport.h
…
typedefs.h
Clear ARM NEON flag
2015-05-20 05:20:04 +00:00
video_decoder.h
Rename I420VideoFrame to VideoFrame.
2015-05-30 00:21:56 +00:00
video_encoder.h
Re-land "Convert native handles to buffers before encoding."
2015-06-05 09:08:12 +00:00
video_engine_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
video_frame.h
Re-land "Convert native handles to buffers before encoding."
2015-06-05 09:08:12 +00:00
video_receive_stream.h
C++11 in-class member initialization in Call configs.
2015-06-11 10:38:46 +00:00
video_renderer.h
Rename I420VideoFrame to VideoFrame.
2015-05-30 00:21:56 +00:00
video_send_stream.h
C++11 in-class member initialization in Call configs.
2015-06-11 10:38:46 +00:00
webrtc_examples.gyp
Adding support for OpenSL ES output in native WebRTC
2015-05-18 14:49:04 +00:00
webrtc_perf_tests.isolate
Enable GoogleWifiTrace3Mbps simulations.
2015-05-04 12:26:26 +00:00
webrtc_tests.gypi
Add HW fallback option to software encoding.
2015-05-19 21:09:17 +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.