webrtc/talk/session/media
Karl Wiberg 9478437fde rtc::Buffer improvements
1. Constructors, SetData(), and AppendData() now accept uint8_t*,
     int8_t*, and char*. Previously, they accepted void*, meaning that
     any kind of pointer was accepted. I think requiring an explicit
     cast in cases where the input array isn't already of a byte-sized
     type is a better compromise between convenience and safety.

  2. data() can now return a uint8_t* instead of a char*, which seems
     more appropriate for a byte array, and is harder to mix up with
     zero-terminated C strings. data<int8_t>() is also available so
     that callers that want that type instead won't have to cast, as
     is data<char>() (which remains the default until all existing
     callers have been fixed).

  3. Constructors, SetData(), and AppendData() now accept arrays
     natively, not just decayed to pointers. The advantage of this is
     that callers don't have to pass the size separately.

  4. There are new constructors that allow setting size and capacity
     without initializing the array. Previously, this had to be done
     separately after construction.

  5. Instead of TransferTo(), Buffer now supports swap(), and move
     construction and assignment, and has a Pass() method that works
     just like std::move(). (The Pass method is modeled after
     scoped_ptr::Pass().)

R=jmarusic@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42989004

Cr-Commit-Position: refs/heads/master@{#9033}
2015-04-20 12:03:00 +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 Avoid implicit type truncations by inserting explicit casts or modifying prototypes to avoid needless up- and then down-casting. 2015-02-02 23:54:40 +00:00
channel.cc rtc::Buffer improvements 2015-04-20 12:03:00 +00:00
channel.h Remove a dependency of BaseChannel on WebRtcSession by having WebRtcSession push down new media descriptions to BaseChannel rather than having BaseChannel listen to the description changes from WebRtcSession. 2015-03-16 21:16:23 +00:00
channelmanager_unittest.cc Delete NullVideoRenderer 2015-03-24 15:11:24 +00:00
channelmanager.cc Remove SignalCaptureStateChange from MediaEngine. 2015-04-14 00:17:36 +00:00
channelmanager.h Remove GetStartCaptureFormat and some related code. 2015-04-13 17:44:56 +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 Use a NULL session in unit tests that don't actually use the session. 2015-03-13 20:05:46 +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 Check associated payload type when negotiate RTX codecs. 2015-03-16 04:15:23 +00:00
mediasession.cc Check associated payload type when negotiate RTX codecs. 2015-03-16 04:15:23 +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 Update libjingle license statements at top of talk files for consistency 2015-01-20 21:36:13 +00:00
rtcpmuxfilter.cc (Auto)update libjingle 72097588-> 72159069 2014-07-29 17:36:52 +00:00
rtcpmuxfilter.h move xmpp and p2p to webrtc 2014-10-28 22:20:11 +00:00
soundclip.cc (Auto)update libjingle 72097588-> 72159069 2014-07-29 17:36:52 +00:00
soundclip.h (Auto)update libjingle 73222930-> 73226398 2014-08-13 17:26:08 +00:00
srtpfilter_unittest.cc Use WebRTC API to convert byteorder in srtpfilter. 2015-03-31 22:02:50 +00:00
srtpfilter.cc rtc::Buffer improvements 2015-04-20 12:03:00 +00:00
srtpfilter.h Remove or rename typedefs with _t prefixes. 2014-12-17 13:43:55 +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