98f53510b2
system_wrappers: rename interface -> include
...
BUG=webrtc:5095
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1413333002 .
Cr-Commit-Position: refs/heads/master@{#10438}
2015-10-28 17:17:50 +00:00
d4cec15c75
Resolved Rebase Conflicts
...
This is just https://webrtc-codereview.appspot.com/53629004/
Remove a constructor of VCMJitterBuffer.
Remove unnecessary factory use
Comment Fix
Move frame incoming simulation to the clock
DCHECK typo fix
Coding Style Fix
Rephrased some comments, and removed some virtual for override function.
Coding Style Fix
Coding Style Fix
Add a unittest for VCMReceiver::FrameForDecoding. Mainly test the time control algorithm.
BUG=
TBR=holmer@chromium.org
Review URL: https://codereview.webrtc.org/1173253008 .
Cr-Commit-Position: refs/heads/master@{#9470}
2015-06-19 16:17:10 +00:00
b1825a4038
Change JitterBuffer::GetNackList to return a std::vector<uint16_t>.
...
This fixed the problem with returning a pointer to an internal buffer
of a JitterBuffer.
R=stefan@webrtc.org
BUG=none
TEST=none
Review URL: https://webrtc-codereview.appspot.com/53639004
Cr-Commit-Position: refs/heads/master@{#9365}
2015-06-03 22:03:46 +00:00
92d9489881
Miscellaneous cleanups in VCMReceiver and its unit tests.
...
The most important change is to prevent a potential buffer overflow in
NackList(). It cannot happen if the |size| argument passed to NackList()
is consistent with the |max_nack_list_size| argument passed to
SetNackSettings(), and there is an assertion to check that. But it is
good to defend against this in the release build because assert() is
compiled away in the release build.
Remove the unused |master| parameter to the VCMReceiver constructor.
Remove the unused State() getter method and the corresponding state_
member.
Remove the declarations for the nonexistent GenerateReceiverId()
method and the receiver_id_counter_ member.
Remove the unneeded data_buffer_ member of TestVCMReceiver. It was
assigned to packet.dataPtr and then immediately overwritten by
stream_generator_->GetPacket() or stream_generator_->PopPacket().
R=stefan@webrtc.org
BUG=none
TEST=none
Review URL: https://webrtc-codereview.appspot.com/51119004
Cr-Commit-Position: refs/heads/master@{#9318}
2015-05-28 20:36:22 +00:00
5464a6e548
Remove VideoCodingModule::InitializeReceiver.
...
This code is no longer used to reset, so we can just initialize the
object in the constructor.
BUG=4391
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43249004
Cr-Commit-Position: refs/heads/master@{#9044}
2015-04-21 14:35:34 +00:00
00b8f6b364
Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away
...
BUG=
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36229004
Cr-Commit-Position: refs/heads/master@{#8517}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8517 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-26 14:43:50 +00:00
4dd40d6b88
Signal threads for faster receiver destruction.
...
Unblocks pending threads (render thread + decoder thread) when
destroying renderers and shutting down decoders.
Speeds up SetLocalDescription significantly (10x or so) under
WebRtcVideoEngine2 but also shutdown times in ~ViEChannel and
~ViEReceiver in general.
BUG=1788
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41959004
Cr-Commit-Position: refs/heads/master@{#8387}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8387 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-17 13:23:27 +00:00
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
5570769210
Remove the last getters from VideoReceiveStream stats.
...
R=stefan@webrtc.org
BUG=1667
Review URL: https://webrtc-codereview.appspot.com/32899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7965 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-19 15:45:03 +00:00
ce4e9a3562
Refactor some receive-side stats.
...
Removes polling of CName as well as receive codec statistics in favor of
internal callbacks keeping a statistics struct up to date.
R=mflodman@webrtc.org , stefan@webrtc.org
BUG=1667
Review URL: https://webrtc-codereview.appspot.com/28259005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7950 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-18 13:50:16 +00:00
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
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
a7e360e89b
Removed lines preventing simultaneous kHardNack and decoding with errors. Also made changes recommended by gcl lint (with the exception of changing non-const references to pointers).
...
Propagated orthogonal API for decoding with errors from VideoCodingModule to VCMJitterBuffer.
Modified VCMJitterBuffer to allow three error modes: kNoErrors, kSelectiveErrors, kWithErrors.
R=marpan@google.com , mikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1846004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4463 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-01 03:15:08 +00:00
a4407329d4
Include files from webrtc/.. paths in video_coding/.
...
BUG=1662
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1783006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4348 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-16 12:32:05 +00:00
ef14488d03
Trigger a PLI if the duration of non-decodable frames exceeds a threshold.
...
BUG=1663
R=mikhal@webrtc.org , ronghuawu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/1359004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3975 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-07 19:16:33 +00:00
759b041019
Relanding r3952: VCM: Updating receiver logic
...
BUG=r1734
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1433004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3970 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-07 16:36:00 +00:00
4ce19b1664
Revert r3952 "VCM: Updating receiver logic"
...
TBR=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1410005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3963 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-06 13:16:51 +00:00
d3cd565ecf
VCM: Updating receiver logic
...
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1363005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3952 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-03 17:54:18 +00:00
381da4be9c
VCM: Adding API for the size(duration) of the jitter buffer.
...
Refers to the duration in time of the frames which are ready to be sent to the decoder.
Review URL: https://webrtc-codereview.appspot.com/1319004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3903 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-25 21:45:29 +00:00
dc3cd217b2
VCM/JB: FrameForDecoding->IncompleteFrameForDecoding
...
- Update complete frame for decoding
- Remove FrameForDecodingNack
This CL should only be committed after issue http://webrtc-codereview.appspot.com/1313007/
Review URL: https://webrtc-codereview.appspot.com/1316007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3901 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-25 20:27:04 +00:00
2baf5f5fa0
Refactor webrtc specific Event implementation to an EventFactory.
...
Review URL: https://webrtc-codereview.appspot.com/1187005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3664 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-13 08:46:25 +00:00
a64300af50
Refactor NACK list creation to build the NACK list as packets arrive.
...
Also fixes a timer bug related to NACKing in the RTP module which could cause packets to only be NACKed twice if there's frequent packet losses.
Note that I decided to remove any selective NACKing for now as I don't think the gain of doing it is big enough compared to the added complexity. The same reasoning for empty packets. None of them will be retransmitted by a smart sender since the sender would know that they aren't needed.
BUG=1420
TEST=video_coding_unittests, vie_auto_test, video_coding_integrationtests, trybots
Review URL: https://webrtc-codereview.appspot.com/1115006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3599 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-04 15:24:40 +00:00
ef9f76a59d
Adding a receive side API for buffering mode.
...
At the same time, renaming the send side API.
Review URL: https://webrtc-codereview.appspot.com/1104004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3525 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-15 23:22:18 +00:00
becf9c897c
Fix mismatch between different NACK list lengths and packet buffers.
...
This is a second version of http://review.webrtc.org/1065006/ which passes the parameters via methods instead of via constructors.
BUG=1289
Review URL: https://webrtc-codereview.appspot.com/1065007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3456 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-01 15:09:57 +00:00
a678a3baee
Move video_coding to new Clock interface and remove fake clock implementations from RTP module tests.
...
TEST=video_coding_unittests, video_coding_integrationtests, rtp_rtcp_unittests, trybots
Review URL: https://webrtc-codereview.appspot.com/1044004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3393 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-21 07:42:11 +00:00
1ea4b502ef
Refactor receiver.h/.cc.
...
TEST=video_coding_unittests, vie_auto_test --automated
Review URL: https://webrtc-codereview.appspot.com/994008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3336 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-07 08:49:41 +00:00
14b43beb7c
Move src/ -> webrtc/
...
TBR=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/915006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-22 18:19:23 +00:00
