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kwiberg@webrtc.org decd9306ae AudioEncoder: num_10ms_frames_per_packet -> Num10MsFramesInNextPacket 
Rename this accessor function to reflect its new, slightly changed
meaning. The reason for the change is that some codecs (iSAC) vary the
number of 10 ms frames from packet to packet, and so can't return a
truly constant value.

BUG=3926
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7556 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-29 08:38:50 +00:00
..
base
Only configure the SSL library in one place.
2014-10-27 18:13:40 +00:00
build
move xmpp and p2p to webrtc
2014-10-28 22:20:11 +00:00
common_audio
common_audio: Removed trivial macro WEBRTC_SPL_UMUL_16_16
2014-10-28 13:03:10 +00:00
common_video
GN: Add common configs to all targets.
2014-09-28 17:37:22 +00:00
examples
Split video engine android initialization into each internal module initialization.
2014-09-17 11:44:51 +00:00
libjingle
move xmpp and p2p to webrtc
2014-10-28 22:20:11 +00:00
modules
AudioEncoder: num_10ms_frames_per_packet -> Num10MsFramesInNextPacket
2014-10-29 08:38:50 +00:00
overrides
webrtc/overrides: add OWNERS-file.
2014-09-17 08:04:28 +00:00
p2p
move xmpp and p2p to webrtc
2014-10-28 22:20:11 +00:00
sound
rtc_unittest: turned sound's test gyp into gypi to speed up GYP generation.
2014-10-01 16:33:03 +00:00
system_wrappers
scoped_ptr.h: Renames function and change namespace scope to fix conflicts with Chromium not detected by the FYI bots.
2014-10-28 18:06:42 +00:00
test
Move (test) RtpFileReader to a lightweight target.
2014-10-27 18:01:03 +00:00
tools
Adding the subtool rtcBot report visualizer
2014-10-24 09:26:16 +00:00
video
Move min transmit bitrate to VideoEncoderConfig.
2014-10-24 09:23:21 +00:00
video_engine
Add macros and APIs for webrtc histograms.
2014-10-23 12:57:56 +00:00
voice_engine
Add a simple AudioConverter class.
2014-10-27 18:18:17 +00:00
.gitignore
.gitignore: Add *.mk, created as part of ChromiumOS build
2013-01-04 21:25:42 +00:00
BUILD.gn
GN: Enable libvpx, add link target and convert some test targets
2014-09-30 18:05:02 +00:00
call.h
Remove -1 from Call::Config::start_bitrate_bps.
2014-10-14 11:52:10 +00:00
common_types.h
Revert cls (original cl + fixes) 7422-7424 "Add VP9 codec to VCM..."
2014-10-17 18:54:46 +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
Move min transmit bitrate to VideoEncoderConfig.
2014-10-24 09:23:21 +00:00
config.h
Move min transmit bitrate to VideoEncoderConfig.
2014-10-24 09:23:21 +00:00
engine_configurations.h
Enable G.722 for Chromium builds
2014-10-29 08:32:44 +00:00
experiments.h
Remove no longer used SkipEncodingUnusedStreams.
2014-07-22 07:17:17 +00:00
frame_callback.h
Wire up statistics in video receive stream of new API
2014-02-07 12:06:29 +00:00
LICENSE
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
LICENSE_THIRD_PARTY
Consolidate all third party licenses in LICENSE_THIRD_PARTY.
2013-05-05 18:54:10 +00:00
OWNERS
GN: Add BUILD.gn files + kjellander to OWNERS
2014-06-23 19:21:07 +00:00
PATENTS
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
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
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
rtc_unittests.isolate
Adds isolate for rtc_unittests and moves sound's unittests to rtc_unittest.
2014-09-30 14:21:10 +00:00
supplement.gypi
Roll chromium_revision 289723:291647
2014-08-25 14:16:32 +00:00
transport.h
Rename newapi::Transport::SendRTP()->SendRtp().
2013-11-20 12:17:04 +00:00
typedefs.h
Add CHECK and friends from Chromium.
2014-08-28 16:28:26 +00:00
video_encoder.h
Revert cls (original cl + fixes) 7422-7424 "Add VP9 codec to VCM..."
2014-10-17 18:54:46 +00:00
video_engine_tests.isolate
Pass GYP DEPTH variable to isolate.
2014-06-13 09:02:15 +00:00
video_frame.h
Revert 7151 "Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h.""
2014-09-17 09:02:25 +00:00
video_receive_stream.h
Change return value for number of discarded packets to be int.
2014-09-04 07:07:44 +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
Move min transmit bitrate to VideoEncoderConfig.
2014-10-24 09:23:21 +00:00
webrtc_examples.gyp
Add macros and APIs for webrtc histograms.
2014-10-23 12:57:56 +00:00
webrtc_perf_tests.isolate
Pass GYP DEPTH variable to isolate.
2014-06-13 09:02:15 +00:00
webrtc_tests.gypi
move xmpp and p2p to webrtc
2014-10-28 22:20:11 +00:00
webrtc.gyp
move xmpp and p2p to webrtc
2014-10-28 22:20:11 +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.