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12 Commits

Author SHA1 Message Date
Stefan Holmer
9131efdb30 Read recv timestamps from socket (posix only). 
This helps a lot on Android devices where the user threads can be scheduled with low priority when the app is in the background, causing spurious significantly delayed before a packet can be read from the socket. With this patch the timestamp is taken by the kernel when the packet actually arrives.

R=juberti@chromium.org
TBR=juberti@webrtc.org

BUG=webrtc:5773

Review URL: https://codereview.webrtc.org/1944683002 .

Cr-Commit-Position: refs/heads/master@{#12850}
 2016-05-23 16:19:37 +00:00
Honghai Zhang
82d7862fe7 Change default timestamp to 64 bits in all webrtc directories. 
BUG=
R=pbos@webrtc.org, pthatcher@webrtc.org, solenberg@webrtc.org

Review URL: https://codereview.webrtc.org/1835053002 .

Cr-Commit-Position: refs/heads/master@{#12646}
 2016-05-06 18:29:27 +00:00
jbauch
555604a746 Replace scoped_ptr with unique_ptr in webrtc/base/ 
This propagated into various other places. Also had to #include headers that
were implicitly pulled by "scoped_ptr.h".

BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1920043002

Cr-Commit-Position: refs/heads/master@{#12501}
 2016-04-26 10:13:28 +00:00
jbauch
313afba2eb Lazily allocate input buffer for AsyncTCPSocket. 
As a follow-up to https://codereview.webrtc.org/1737053006/ this CL further
improves memory usage by lazily allocating input buffers up to the passed
maximum size. This also changes the input buffer to a Buffer object.

BUG=

Review URL: https://codereview.webrtc.org/1744293002

Cr-Commit-Position: refs/heads/master@{#11859}
 2016-03-03 11:41:14 +00:00
jbauch
250fc658c5 Lazily allocate output buffer for AsyncTCPSocket. 
As a follow-up to https://codereview.webrtc.org/1737053006/ this CL further
improves memory usage by lazily allocating output buffers up to the passed
maximum size. This also changes the output buffer to a Buffer object.

BUG=

Review URL: https://codereview.webrtc.org/1741413002

Cr-Commit-Position: refs/heads/master@{#11801}
 2016-02-28 23:06:47 +00:00
jbauch
3c1657658d Don't allocate buffers for listening sockets. 
Listening sockets will not read/write directly, so they don't need buffers.

BUG=

Review URL: https://codereview.webrtc.org/1737053006

Cr-Commit-Position: refs/heads/master@{#11791}
 2016-02-26 17:31:41 +00:00
stefan
c1aeaf0dc3 Wire up packet_id / send time callbacks to webrtc via libjingle. 
BUG=webrtc:4173

Review URL: https://codereview.webrtc.org/1363573002

Cr-Commit-Position: refs/heads/master@{#10289}
 2015-10-15 14:26:17 +00:00
Peter Boström
0c4e06b4c6 Use suffixed {uint,int}{8,16,32,64}_t types. 
Removes the use of uint8, etc. in favor of uint8_t.

BUG=webrtc:5024
R=henrik.lundin@webrtc.org, henrikg@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://codereview.webrtc.org/1362503003 .

Cr-Commit-Position: refs/heads/master@{#10196}
 2015-10-07 10:23:32 +00:00
honghaiz
b19eba3d4b Fix Turn TCP port issue. 
Sometimes the port still try to send stun packet when the connection is disconnected,
causing an assertion error.

BUG=4859

Review URL: https://codereview.webrtc.org/1247573002

Cr-Commit-Position: refs/heads/master@{#9671}
 2015-08-03 17:23:40 +00:00
henrike@webrtc.org
f048872e91 Adds a modified copy of talk/base to webrtc/base. It is the first step in 
migrating talk/base to webrtc/base.

BUG=N/A
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17479005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6129 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-13 18:00:26 +00:00
perkj@webrtc.org
e9a604accd Revert 6107 "Adds a modified copy of talk/base to webrtc/base. I..." 
This breaks Chromium FYI builds and prevent roll of webrtc/libjingle to Chrome.

http://chromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Win%20Builder/builds/457


> Adds a modified copy of talk/base to webrtc/base. It is the first step in migrating talk/base to webrtc/base.
> 
> BUG=N/A
> R=andrew@webrtc.org, wu@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/12199004

TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6116 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-13 08:15:48 +00:00
henrike@webrtc.org
2c7d1b39b9 Adds a modified copy of talk/base to webrtc/base. It is the first step in migrating talk/base to webrtc/base. 
BUG=N/A
R=andrew@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6107 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-12 18:03:09 +00:00