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16 Commits

Author SHA1 Message Date
pkasting@chromium.org
16825b1a82 Use int64_t more consistently for times, in particular for RTT values. 
Existing code was inconsistent about whether to use uint16_t, int, unsigned int,
or uint32_t, and sometimes silently truncated one to another, or truncated
int64_t.  Because most core time-handling functions use int64_t, being
consistent about using int64_t unless otherwise necessary minimizes the number
of explicit or implicit casts.

BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, holmer@google.com, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8045 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-12 21:51:21 +00:00
pkasting@chromium.org
0b1534c52e Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess. 
This fixes a variety of MSVC warnings about value truncations when implicitly
storing the 64-bit values we get back from e.g. TimeTicks in 32-bit objects, and
removes the need for a number of explicit casts.

This also moves a number of constants so they're declared right where they're used, which is easier to read and maintain, and makes some of them of integral type rather than using the "enum hack".

BUG=chromium:81439
TEST=none
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7905 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-15 22:09:40 +00:00
pbos@webrtc.org
4f16c874c6 Simplifying VideoReceiver and JitterBuffer. 
Removing frame_buffers_ array and dual-receiver mechanism. Also adding
some thread annotations to VCMJitterBuffer.

R=stefan@webrtc.org
BUG=4014

Review URL: https://webrtc-codereview.appspot.com/27239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7735 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-24 09:06:48 +00:00
stefan@webrtc.org
34c5da6b5e Cleaned up logging in video_coding. 
Converted all calls to WEBRTC_TRACE to LOG(). Also removed a large number of less useful logs.

BUG=3153
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5887 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-11 14:08:35 +00:00
andresp@webrtc.org
1df9dc3957 Isolate register post encode callback in video coding module to simplify code and critical sections. 
R=marpan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5357 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-09 08:01:57 +00:00
andresp@webrtc.org
b08a12d6e8 Isolate debug recording from video sender into a thread safe small class. 
R=marpan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5353 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-08 12:38:22 +00:00
andresp@webrtc.org
e682aa5077 Refactoring MediaOptimization so it can easily be turned into a thread-safe class. 
BUG=2732
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5322 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-19 10:59:48 +00:00
sprang@webrtc.org
4070935f4f Implement and test EncodedImageCallback in new ViE API. 
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4059004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5179 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-26 11:41:59 +00:00
henrik.lundin@webrtc.org
ce8e0936d9 Rename AutoMute to SuspendBelowMinBitrate 
Changes all instances throughout the WebRTC stack.

BUG=2436
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5130 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-18 12:18:43 +00:00
henrik.lundin@webrtc.org
1a3a6e5340 Removing the threshold from the auto-mute APIs 
The threshold is now set equal to the minimum bitrate of the
encoder. The test is also changed to have the REMB values
depend on the minimum bitrate from the encoder.

BUG=2436
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5040 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-28 10:16:14 +00:00
henrik.lundin@webrtc.org
572699d3eb Propagate AutoMuter interface out to VideoCodingModule 
BUG=2436
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2311004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4878 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-30 12:16:08 +00:00
henrik.lundin@webrtc.org
bec11ef632 Reformatting media_optimization.cc and .h 
Ran both tools/refactoring/webrtc_reformat.py and clang-format.
Changing VCMMediaOptimization -> MediaOptimization and
VCMEncodedFrameSample -> EncodedFrameSample.
Aligning the order of methods in .h and .cc files and fixing comments.

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2265007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4816 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-23 19:54:25 +00:00
stefan@webrtc.org
8db81c5112 Fix races in vcm::Process(). 
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2241004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4775 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-18 11:57:34 +00:00
pbos@webrtc.org
32d640e03d Fix typo in r4765. 
Fixes compile error on all platforms.

BUG=
TEST=compile on tryboys
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2231004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4766 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-17 10:36:30 +00:00
pbos@webrtc.org
da2c4cede0 Fix dangling pointer _encoder in video_sender.cc. 
When _codecDataBase.SetSendCodec() fails, the encoder may be deleted.
This is however not reflected in _encoder, which then becomes a dangling
pointer to the deleted object.

BUG=2384
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4765 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-17 09:38:41 +00:00
andresp@webrtc.org
f7eb75be1a Split VideoCodingModuleImpl into VideoSender and VideoReceiver. 
Only implmentation is changed the interface to the module is unchanged for now.

R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2200008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4746 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-14 00:25:28 +00:00