16 Commits

Author SHA1 Message Date
Peter Kasting
6955870806 Convert channel counts to size_t.
IIRC, this was originally requested by ajm during review of the other size_t conversions I did over the past year, and I agreed it made sense, but wanted to do it separately since those changes were already gargantuan.

BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, henrika@webrtc.org, kjellander@webrtc.org, minyue@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://codereview.webrtc.org/1316523002 .

Cr-Commit-Position: refs/heads/master@{#11229}
2016-01-13 00:26:55 +00:00
pkasting
25702cb162 Misc. small cleanups.
* Better param names
* Avoid using negative values for (bogus) placeholder channel counts (mostly in tests).  Since channels will be changing to size_t, negative values will be illegal; it's sufficient to use 0 in these cases.
* Use arraysize()
* Use size_t for counting frames, samples, blocks, buffers, and bytes -- most of these are already size_t in most places, this just fixes some stragglers
* reinterpret_cast<int64_t>(void*) is not necessarily safe; use uintptr_t instead
* Remove unnecessary code, e.g. dead code, needlessly long/repetitive code, or function overrides that exactly match the base definition
* Fix indenting
* Use uint32_t for timestamps (matching how it's already a uint32_t in most places)
* Spelling
* RTC_CHECK_EQ(expected, actual)
* Rewrap
* Use .empty()
* Be more pedantic about matching int/int32_t/
* Remove pointless consts on input parameters to functions
* Add missing sanity checks

All this was found in the course of constructing https://codereview.webrtc.org/1316523002/ , and is being landed separately first.

BUG=none
TEST=none

Review URL: https://codereview.webrtc.org/1534193008

Cr-Commit-Position: refs/heads/master@{#11191}
2016-01-08 21:50:32 +00:00
stefan
43edf0ffb9 Require negotiation to send transport cc feedback over RTCP.
BUG=4312

Review URL: https://codereview.webrtc.org/1452883002

Cr-Commit-Position: refs/heads/master@{#10735}
2015-11-21 02:05:53 +00:00
Shao Changbin
e62202fedf Support handling multiple RTX but only generate SDP with RTX associated with VP8.
This implementation registers RTX-APT map inside RTP sender and receiver.
While it only generates SDP with RTX associated with VP8 to make it
compatible with previous Chrome versions.

Should add following changes after reaches stable,
* Use RTX-APT map for building and restoring RTP packets.
* Add RTX support for RED or VP9 in Video engine.
* Set RTX payload type for RED inside FecConfig in EndToEndTest.

BUG=4024
R=mflodman@webrtc.org, pbos@webrtc.org, pthatcher@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36889004

Cr-Commit-Position: refs/heads/master@{#9040}
2015-04-21 12:25:42 +00:00
pkasting@chromium.org
d324546ced Misc. cleanup split out of https://webrtc-codereview.appspot.com/37699004/ :
* Move constants into the files/functions that use them
* Declare variables in the narrowest scope possible
* Use correct (expected, actual) order for gtest macros
* Remove unused functions
* Untabify
* 80-column limit
* Avoid C-style casts
* Prefer true typed constants to "enum hack" constants
* Print size_t using the right format macro
* Shorten and simplify code
* Other random cleanup bits and style fixes

BUG=none
TEST=none
R=henrik.lundin@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36179004

Cr-Commit-Position: refs/heads/master@{#8467}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8467 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-23 21:29:45 +00:00
guoweis@webrtc.org
cce874b8d2 Fix libjingle_media_unittest codec comparison issue
Missing one comparison of AudioCodec

TBR=juberti@webrtc.org

BUG=

Review URL: https://webrtc-codereview.appspot.com/42409005

Cr-Commit-Position: refs/heads/master@{#8437}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8437 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-19 18:14:52 +00:00
guoweis@webrtc.org
bc6961fe32 Make webrtc 50 KB smaller by not inlining Codec.
The Codec class is a big class and objects of the Codec class are passed
around by value. That means that inlined operations would be duplicated
at many places, in particular inside STL.

By not inlining Codec methods, webrtc shrinks by 50 KB in
a Linux x64 clang build.

Total change: -54147 bytes
==========================
 +2810 - Source: /home/bratell/src/chromium/src/third_party/libjingle/source/talk/media/base/codec.cc - (gained 2920, lost 110)
 -1003 - Source: /home/bratell/src/chromium/src/third_party/libjingle/source/talk/media/base/codec.h - (gained 0, lost 1003)
 -1129 - Source: /home/bratell/src/chromium/src/third_party/libjingle/source/talk/media/sctp/sctpdataengine.cc - (gained 1660, lost 2789)
 -1190 - Source: /home/bratell/src/chromium/src/third_party/libjingle/source/talk/media/base/rtpdataengine.cc - (gained 1408, lost 2598)
 -1747 - Source: /home/bratell/src/chromium/src/third_party/libjingle/source/talk/session/media/mediasession.cc - (gained 803, lost 2550)
 -2141 - Source: /home/bratell/src/chromium/src/third_party/libjingle/source/talk/media/webrtc/webrtcvideoengine.cc - (gained 1679, lost 3820)
 -2250 - Source: /home/bratell/src/chromium/src/third_party/libjingle/source/talk/app/webrtc/webrtcsdp.cc - (gained 1224, lost 3474)
 -2927 - Source: /usr/include/c++/4.8/bits/stl_vector.h - (gained 0, lost 2927)
 -3729 - Source: /home/bratell/src/chromium/src/third_party/libjingle/source/talk/media/webrtc/webrtcvideoengine2.cc - (gained 10925, lost 14654)
 -6369 - Source: /usr/include/c++/4.8/bits/vector.tcc - (gained 0, lost 6369)
 -10582 - Source: /usr/include/c++/4.8/bits/stl_heap.h - (gained 0, lost 10582)
 -19324 - Source: /usr/include/c++/4.8/bits/stl_algo.h - (gained 743, lost 20067)

BUG=
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40729005

Cr-Commit-Position: refs/heads/master@{#8436}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8436 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-19 17:55:50 +00:00
buildbot@webrtc.org
d4e598d57a (Auto)update libjingle 72097588-> 72159069
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6799 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-29 17:36:52 +00:00
wu@webrtc.org
ff1b1bf094 When creating an answer, takes the codec preference from the offer.
This change is based on RFC3264:

"Although the answerer MAY list the formats in their desired order of preference, it is RECOMMENDED that unless there is a specific reason, the answerer list formats in the same relative order they were present in the offer."

BUG=2868
TEST=unit tests and manually with munge-sdp test
R=juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/14589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6514 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-20 20:57:42 +00:00
buildbot@webrtc.org
fbd13286dc (Auto)update libjingle 69555283-> 69567902
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6497 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-19 19:50:55 +00:00
pbos@webrtc.org
b5a22b1464 Revert r6110 and r6109.
Effectively re-landing r6104 as well as r6108 which includes a fix to a
compile error that caused r6104 to be reverted in r6110.

BUG=
TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6119 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-13 11:07:01 +00:00
buildbot@webrtc.org
17911dca80 (Auto)update libjingle 66798415-> 66813165
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6110 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-12 18:42:49 +00:00
henrike@webrtc.org
0df2ea064f Rollback of r6108
BUG=N/A
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6109 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-12 18:41:12 +00:00
pbos@webrtc.org
a7f70a487f Initialize bitrates in ValidateCodecFormat.
Attempt to un-break a Visual Studio build (unknown version) that
incorrectly reports that these are potentially uninitialized.

BUG=
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15469005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6108 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-12 18:20:40 +00:00
pbos@webrtc.org
d266a2020f Initial wiring of new webrtc API in libjingle.
BUG=1788
R=pthatcher@google.com, pthatcher@webrtc.org
TBR=juberti@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8549005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6104 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-12 14:32:01 +00:00
henrike@webrtc.org
28e2075280 Adds trunk/talk folder of revision 359 from libjingles google code to
trunk/talk


git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-10 00:45:36 +00:00