generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity
4e7aa43ea0
webrtc/talk/session/media
History
Jelena Marusic c28a896a7b VoE: Initialize WebRtcVoiceMediaChannel with AudioOptions during creation 
BUG=4690

Changes:
1. In MediaEngineInterface changed CreateChannel() to CreateChannel(const AudioOptions&). Plan is to eventually remove Get/SetAudioOptions and the cousins SetDelayOffset and SetDevices.
2. In ChannelManager changed CreateVoiceChannel(...) to CreateVoiceChannel(..., const AudioOptions&).
3. In ChannelManager removed SetEngineAudioOptions, because it is not used and we want to eventually remove SetAudioOptions.
4. Updated MediaEngineInterface implementations and unit tests accordingly.
5. In WebRtcVoiceEngine changed access of Set/ClearOptionOverrides to protected. These are only used by WebRtcVoiceMediaChannel (now a friend). Plan is to rethink the logic behind option overrides.
6. Cosmetics: replaced NULL with nullptr in touched code

R=solenberg@google.com, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/56499004

Cr-Commit-Position: refs/heads/master@{#9330}
2015-05-29 13:05:52 +00:00
..
audiomonitor.cc (Auto)update libjingle 73222930-> 73226398 2014-08-13 17:26:08 +00:00
audiomonitor.h move xmpp and p2p to webrtc 2014-10-28 22:20:11 +00:00
bundlefilter_unittest.cc (Auto)update libjingle 73222930-> 73226398 2014-08-13 17:26:08 +00:00
bundlefilter.cc Change GetStreamBySsrc to not copy StreamParams. 2015-01-22 23:00:41 +00:00
bundlefilter.h (Auto)update libjingle 73222930-> 73226398 2014-08-13 17:26:08 +00:00
channel_unittest.cc Add RtcpMuxPolicy support to PeerConnection. 2015-05-21 14:48:19 +00:00
channel.cc Add RtcpMuxPolicy support to PeerConnection. 2015-05-21 14:48:19 +00:00
channel.h Add RtcpMuxPolicy support to PeerConnection. 2015-05-21 14:48:19 +00:00
channelmanager_unittest.cc VoE: Initialize WebRtcVoiceMediaChannel with AudioOptions during creation 2015-05-29 13:05:52 +00:00
channelmanager.cc VoE: Initialize WebRtcVoiceMediaChannel with AudioOptions during creation 2015-05-29 13:05:52 +00:00
channelmanager.h VoE: Initialize WebRtcVoiceMediaChannel with AudioOptions during creation 2015-05-29 13:05:52 +00:00
currentspeakermonitor_unittest.cc Move Jingle-specific files from talk/session/media to webrtc/libjingle/session/media. This is part of an ongoing effort to remove Jingle-specific files from the WebRTC repository. 2014-12-19 22:29:55 +00:00
currentspeakermonitor.cc Move ViewRequest and MediaStreams to streamparams.h, and remove dependency on mediasessionclient.h and mediamessages.h. This is part of the effort to remove Jingle-specific code from WebRTC and into its own repository. 2014-12-16 21:09:08 +00:00
currentspeakermonitor.h Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository. 2014-12-18 20:31:29 +00:00
externalhmac.cc Update libsrtp includes in preparation of roll into Chromium. 2015-03-23 22:12:19 +00:00
externalhmac.h Update libsrtp includes in preparation of roll into Chromium. 2015-03-23 22:12:19 +00:00
mediamonitor.cc (Auto)update libjingle 73222930-> 73226398 2014-08-13 17:26:08 +00:00
mediamonitor.h Thread annotation of rtc::CriticalSection. 2014-09-24 07:10:57 +00:00
mediarecorder_unittest.cc Remove unused voice channel argument from cricket::VideoChannel ctor and corresponding field in class. 2015-04-22 13:30:33 +00:00
mediarecorder.cc (Auto)update libjingle 73222930-> 73226398 2014-08-13 17:26:08 +00:00
mediarecorder.h (Auto)update libjingle 73222930-> 73226398 2014-08-13 17:26:08 +00:00
mediasession_unittest.cc Ensure mediasession generated offers with RTX contain an RTX ssrc for each video ssrc. 2015-05-18 21:02:40 +00:00
mediasession.cc Ensure mediasession generated offers with RTX contain an RTX ssrc for each video ssrc. 2015-05-18 21:02:40 +00:00
mediasession.h After another round of reviews. 2015-02-24 20:20:19 +00:00
mediasink.h Adds trunk/talk folder of revision 359 from libjingles google code to 2013-07-10 00:45:36 +00:00
planarfunctions_unittest.cc Roll chromium_revision 8af41b3..dcb0929 (324854:325030) 2015-04-15 15:22:19 +00:00
rtcpmuxfilter_unittest.cc Add RtcpMuxPolicy support to PeerConnection. 2015-05-21 14:48:19 +00:00
rtcpmuxfilter.cc Add RtcpMuxPolicy support to PeerConnection. 2015-05-21 14:48:19 +00:00
rtcpmuxfilter.h Add RtcpMuxPolicy support to PeerConnection. 2015-05-21 14:48:19 +00:00
soundclip.cc Remove Soundclip handling from libjingle. 2015-05-19 09:37:39 +00:00
soundclip.h Remove Soundclip handling from libjingle. 2015-05-19 09:37:39 +00:00
srtpfilter_unittest.cc Use WebRTC API to convert byteorder in srtpfilter. 2015-03-31 22:02:50 +00:00
srtpfilter.cc Protect access to shared list of SRTP sessions. 2015-05-28 23:21:00 +00:00
srtpfilter.h Prevent potential double-free if srtp_create fails. 2015-05-27 21:41:52 +00:00
typewrapping.h.pump (Auto)update libjingle 72097588-> 72159069 2014-07-29 17:36:52 +00:00
typingmonitor_unittest.cc move xmpp and p2p to webrtc 2014-10-28 22:20:11 +00:00
typingmonitor.cc (Auto)update libjingle 73222930-> 73226398 2014-08-13 17:26:08 +00:00
typingmonitor.h (Auto)update libjingle 73222930-> 73226398 2014-08-13 17:26:08 +00:00
voicechannel.h Adds trunk/talk folder of revision 359 from libjingles google code to 2013-07-10 00:45:36 +00:00
yuvscaler_unittest.cc (Auto)update libjingle 73222930-> 73226398 2014-08-13 17:26:08 +00:00