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
83b5200f95
Add framerate for complete received frames to histogram stats:
...
"WebRTC.Video.CompleteFramesReceivedPerSecond".
BUG=crbug/419657
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32089004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7762 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-28 10:17:13 +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
96dc685143
Add stats for video:
...
- number of sent/received RTCP NACK/FIR/PLI per minute
- percentage of unique sent/received NACK requests
- percentage of discarded/duplicated packets by the jitter buffer
- permille of sent/received key frames
BUG=crbug/419657
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24959004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7592 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03 14:40:38 +00:00
88fbb2d86b
Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
...
Same as https://webrtc-codereview.appspot.com/19519004 . The issue in
http://chromegw.corp.google.com/i/internal.chromium.webrtc.fyi/builders/Linux ...
is solved by this change
http://src.chromium.org/viewvc/chrome/trunk/src/third_party/libjingle/libjing ...
(tested locally).
BUG=3380
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17619005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6218 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-21 21:18:46 +00:00
2fa7f79094
Revert 6202 "Switch to using base/constructormagic.h and remove ..."
...
> Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
>
> BUG=N/A
> R=andrew@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/19519004
TBR=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14579007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6210 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-21 11:07:29 +00:00
125ffd709d
Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
...
BUG=N/A
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19519004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6202 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-20 15:20:44 +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
71f055fb41
Add send frame rate statistics callback
...
BUG=2235
R=mflodman@webrtc.org , pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4479005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5213 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-04 15:09:27 +00:00
3c5a9242fe
Don't force cont' when enabling kWithErrors
...
BUG=
R=pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2047004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4669 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-03 20:45:36 +00:00
2b810bf77b
Removing non decodable count from session info: This value isn't used, and therfore can be removed. This is a step towards the refactor of the session info to use maps.
...
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2143004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4667 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-03 19:09:49 +00:00
dbf6a81cb5
Follow-up changes to kSelectiveErrors
...
Committing cl for agalusza (cl 1992004)
TEST = trybots
R=marpan@google.com
Review URL: https://webrtc-codereview.appspot.com/2085004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4587 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-21 20:40:47 +00:00
d177c10e2d
Added logic for kSelectiveErrors to VCMJitterBuffer and corresponding unit tests.
...
R=marpan@google.com , mikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1943004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4503 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-08 01:12:33 +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
d818dcb939
Sets up framework for decoding with errors: collects frame sizes (in number of packets) in JB and passes this information to VCMSessionInfo with rtt_ms as FrameData.
...
R=marpan@google.com , mikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1841004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4424 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-29 21:48:11 +00:00
4cf1a8af69
Removes kStateFree and kStateDecoding, added a free_frames_ list which simplifies finding a free frame.
...
The idea is to have all frames not in use be stored in free_frames_, and whenever a packet from a new frame arrives we can just pop a frame from free_frames_. When a frame is grabbed for decoding it will be removed from all lists, and will be added to free_frames_ when it's returned to the jitter buffer.
We should be able to remove the state enum completely later, as their state is defined by the list they are in. But I'll keep it around for now to simplify the cl.
TEST=try bots and vie_auto_test --automated
R=mikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1721004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4273 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-27 15:20:14 +00:00
9ca7360b97
VCM: removing max jitter estimate
...
BUG= 1921
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1690004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4249 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-20 20:13:07 +00:00
50fb4afade
Revert r4145 "Revert 4127 "Switch frame list implementation to std::map.""
...
TBR=tnakamura@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1678004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4233 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-17 07:33:58 +00:00
c8b29a2feb
Revert r4146 "Revert 4104 "Refactor jitter buffer to use separate lists for de...""
...
TBR=tnakamura@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1677004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4232 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-17 07:13:16 +00:00
694cdc6e84
Revert 4104 "Refactor jitter buffer to use separate lists for de..."
...
Reason - leading suspect of video frame corruption tracked in http://b/9216252
Note that if this turns out to not be the cause, be sure to re-revert both this change and r4145.
> Refactor jitter buffer to use separate lists for decodable and incomplete frames.
>
> This changes the design of the jitter buffer to keeping track of decodable frames from the point when packets are inserted in the buffer, instead of searching for decodable frames when they are needed.
>
> To accomplish this the frame_list_, which previously contained all frames (incomplete or complete, continuous or not), is split into a list of decodable_frames_ (complete, continuous) and a list of incomplete_frames_ (either incomplete or non-continuous). These frame lists are updated every time a packet is inserted.
>
> This is another step in the direction of doing most of the work in the jitter buffer only once, when packets are inserted, instead of doing it every time we look for a frame or try to get a nack list.
>
> BUG=1798
> TEST=vie_auto_test, trybots
> R=mikhal@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/1522005
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1586007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4146 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-31 16:09:48 +00:00
4d9c07ad6d
Revert 4127 "Switch frame list implementation to std::map."
...
We want to revert r4104 for b/9216252, but because r4127 was built on top of r4104, we need to revert r4127 first. We'll un/re-revert this if we discover that r4104 is not to blame.
> Switch frame list implementation to std::map.
>
> This reduces the complexity of insert and find (by timestamp) from linear to logarithmic, which has a big impact on large frame lists.
>
> BUG=1726
> TEST=trybots, vie_auto_test --automated
> R=mikhal@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/1561005
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1590005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4145 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-31 16:06:01 +00:00
ace7ad2302
Switch frame list implementation to std::map.
...
This reduces the complexity of insert and find (by timestamp) from linear to logarithmic, which has a big impact on large frame lists.
BUG=1726
TEST=trybots, vie_auto_test --automated
R=mikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1561005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4127 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-29 07:41:48 +00:00
7f3f8bc5a6
Refactor jitter buffer to use separate lists for decodable and incomplete frames.
...
This changes the design of the jitter buffer to keeping track of decodable frames from the point when packets are inserted in the buffer, instead of searching for decodable frames when they are needed.
To accomplish this the frame_list_, which previously contained all frames (incomplete or complete, continuous or not), is split into a list of decodable_frames_ (complete, continuous) and a list of incomplete_frames_ (either incomplete or non-continuous). These frame lists are updated every time a packet is inserted.
This is another step in the direction of doing most of the work in the jitter buffer only once, when packets are inserted, instead of doing it every time we look for a frame or try to get a nack list.
BUG=1798
TEST=vie_auto_test, trybots
R=mikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1522005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4104 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-27 07:02:45 +00:00
3417eb49f6
Consolidate GetFrame and InsertPacket and move NACK list processing to after a packet has been successfully inserted.
...
TEST=trybots
BUG=1799
R=mikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4080 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-21 15:25:53 +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
6faba6edc9
VCM: Setting buffering delay in timing
...
Review URL: https://webrtc-codereview.appspot.com/1338004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3921 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-30 15:39:34 +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
8392cd9edd
VCM/JB: Using last decoded state for waiting for key
...
relanding 1323006
BUG=
Review URL: https://webrtc-codereview.appspot.com/1354004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3902 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-25 21:30:50 +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
df9c0e5ec9
Revert 3892 "VCM/JB: Using last decoded state for waiting for key"
...
> VCM/JB: Using last decoded state for waiting for key
>
> Review URL: https://webrtc-codereview.appspot.com/1323006
Although I have no idea why, it appears this might be causing failures in ViEStandardIntegrationTest.RunsFileTestWithoutErrors. I was unable to reproduce locally. This is a trial revert to verify. If the errors continue to happen, I will restore this change.
TBR=mikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1321010
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3896 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-24 02:13:18 +00:00
1248d4effc
VCM/JB: Using last decoded state for waiting for key
...
Review URL: https://webrtc-codereview.appspot.com/1323006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3892 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-23 20:57:06 +00:00
d6bd7cd2b1
removing redundant calls to cleanframes
...
Review URL: https://webrtc-codereview.appspot.com/1318004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3844 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-12 17:09:51 +00:00
9da751715f
VCM/JB:Removing hybrid and setting a decodable state.
...
Review URL: https://webrtc-codereview.appspot.com/1283004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3834 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-11 18:49:13 +00:00
7bc465bd21
Fix issues with incorrect wrap checks when having big buffers and high bitrate.
...
Introduces shared functions for timestamp and sequence number wrap checks.
BUG=1607
TESTS=trybots
Review URL: https://webrtc-codereview.appspot.com/1291005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3833 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-11 17:48:02 +00:00
7b859cc1e9
Webrtc_Word32 => int32_t in video_coding/main/
...
BUG=
Review URL: https://webrtc-codereview.appspot.com/1279004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3753 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-02 15:54:38 +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
119c67df36
Adding a max jitter filter to the JB estimate - allowing two modes, one will return the last estimate (current setting), and another will return the max value seen, and allow setting an initial value.
...
This cl also includes tests and some clean up.
Review URL: https://webrtc-codereview.appspot.com/1019007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3445 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-31 17:18:02 +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
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
