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noahric d4badbcb6d Fix SetRates for encoders with internal sources. 
An earlier change moved SetRates to happen on every input frame, but
encoders with internal sources don't receive input frames, so they
weren't getting updated bitrate (and framerate) signals.

BUG=

Review URL: https://codereview.webrtc.org/1682253005

Cr-Commit-Position: refs/heads/master@{#12354}
2016-04-13 21:59:57 +00:00
..
api
Removing preference field from cricket::Codec.
2016-04-13 17:07:24 +00:00
audio
Reland "Add check_deps rules in DEPS files."
2016-03-17 11:05:50 +00:00
base
Add WriteUVarint to ByteBufferWriter and ReadUVarint to ByteBufferReader
2016-04-11 23:11:47 +00:00
build
Fix paths to Android tests .isolate files.
2016-04-12 20:15:18 +00:00
call
Add macros for ability to log samples that are added to histograms (RTC_LOGGED_*).
2016-03-29 09:15:11 +00:00
common_audio
Moved struct definition to the header file for the ring buffer.
2016-04-11 05:51:10 +00:00
common_video
Add CoreVideoFrameBuffer.
2016-04-04 21:10:47 +00:00
examples
Add test annotations to AppRTCDemoTest.
2016-04-12 04:27:40 +00:00
libjingle
Reland of move {media,p2p,pc,xmllite,xmpp}_tests.gypi files. (patchset #1 id:1 of https://codereview.webrtc.org/1846693002/ )
2016-03-31 14:23:52 +00:00
media
Revert of Use microsecond timestamp in cricket::VideoFrame. (patchset #13 id:240001 of https://codereview.webrtc.org/1865283002/ )
2016-04-13 17:45:51 +00:00
modules
Fix SetRates for encoders with internal sources.
2016-04-13 21:59:57 +00:00
p2p
Add QuicTransportChannel methods for QUIC streams
2016-04-11 23:56:26 +00:00
pc
Revert of Accept all the media profiles required by JSEP. (patchset #5 id:80001 of https://codereview.webrtc.org/1880913002/ )
2016-04-13 17:48:32 +00:00
system_wrappers
Implement CPU feature detection for ARM Linux.
2016-04-06 21:02:34 +00:00
test
Fixed rtcp rpsi parsing of invalid packets.
2016-04-11 17:05:06 +00:00
tools
Reland "Add check_deps rules in DEPS files."
2016-03-17 11:05:50 +00:00
video
Add missing tracing to RtpSender objects.
2016-04-11 09:45:43 +00:00
voice_engine
Switch to using new ACM methods for encoder management
2016-04-06 19:22:45 +00:00
.gitignore
…
audio_receive_stream.h
Replace scoped_ptr with unique_ptr in webrtc/audio/
2016-02-23 18:46:39 +00:00
audio_send_stream.h
Relanding https://codereview.webrtc.org/1715883002/ in pieces.
2016-03-11 11:06:48 +00:00
audio_sink.h
Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies."
2016-02-26 21:46:22 +00:00
audio_state.h
…
BUILD.gn
GN: Update audio_sink.h location
2016-02-29 05:49:54 +00:00
call.h
Make the audio channel communicate network state changes to the call.
2016-03-22 22:32:31 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
…
common_types.h
Tune BWE to be a bit less sensitive to spurious delay events.
2016-03-10 13:13:26 +00:00
common.gyp
Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies."
2016-02-26 21:46:22 +00:00
common.h
Changed name for the upcoming AEC from NextGenerationAec to AEC3.
2016-03-08 00:59:43 +00:00
config.cc
Change transport sequence number extension strings to specify what revision is implemented.
2016-03-01 15:27:27 +00:00
config.h
…
DEPS
Delete class webrtc::VideoRenderer and its header file.
2016-03-23 11:48:17 +00:00
engine_configurations.h
Remove the VoEDtmf interface.
2016-03-08 12:11:00 +00:00
frame_callback.h
Measure encoding time on encode callbacks.
2016-02-05 10:13:41 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
Make pbos owner of additional video files.
2016-02-26 12:29:17 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests.isolate
…
stream.h
…
supplement.gypi
Roll chromium_revision 7542f07..38664e7 (377632:377790) + set SDK 10.11 on Mac
2016-02-26 11:21:18 +00:00
transport.h
…
typedefs.h
…
video_decoder.h
Replace scoped_ptr with unique_ptr in webrtc/common_video/
2016-03-02 11:41:40 +00:00
video_encoder.h
Replace scoped_ptr with unique_ptr in webrtc/common_video/
2016-03-02 11:41:40 +00:00
video_engine_tests.isolate
…
video_frame.h
Cleanup of webrtc::VideoFrame.
2016-02-29 12:11:57 +00:00
video_receive_stream.h
Delete class webrtc::VideoRenderer and its header file.
2016-03-23 11:48:17 +00:00
video_send_stream.h
Delete class webrtc::VideoRenderer and its header file.
2016-03-23 11:48:17 +00:00
webrtc_examples.gyp
Add test runner scripts for instrumentation tests
2016-04-08 13:54:32 +00:00
webrtc_nonparallel_tests.isolate
…
webrtc_perf_tests.isolate
…
webrtc_tests.gypi
Reland of move {media,p2p,pc,xmllite,xmpp}_tests.gypi files. (patchset #1 id:1 of https://codereview.webrtc.org/1846693002/ )
2016-03-31 14:23:52 +00:00
webrtc.gyp
GYP: Add webrtc/pc/pc.gyp:* to 'All' target.
2016-04-01 09:15:05 +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.