Commit Graph

80 Commits

Author SHA1 Message Date
d5ce2e63df Remove EventWrapper::Reset().
This simplifies the event wrapper which we've recently found issues in.
Also refactoring EndToEndTest.RespectsNetworkState to not depend on it.

BUG=
R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41939004

Cr-Commit-Position: refs/heads/master@{#8366}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8366 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-13 14:58:38 +00:00
7d2b6a9346 Enable Clang warning implicit-fallthrough and annotate the code.
BUG=4242
R=henrik.lundin@webrtc.org, stefan@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34899004

Cr-Commit-Position: refs/heads/master@{#8187}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8187 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-28 18:38:13 +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
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
70e2d11ea8 Reduce jitter delay for low fps streams.
Enabled by finch flag.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31389005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7288 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-24 14:06:56 +00:00
2ec560606b Add H.264 packetization.
This also includes:
- Creating new packetizer and depacketizer interfaces.
- Moved VP8 packetization was H264 packetization and depacketization to these interfaces. This is a work in progress and should be continued to get this 100% generic. This also required changing the return type for RtpFormatVp8::NextPacket(), which now returns bool instead of the index of the first partition.
- Created a Create() factory method for packetizers and depacketizers.

R=niklas.enbom@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6804 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-31 14:59:24 +00:00
ebb467fdc8 Avoid NACK-list flush error on keyframe packets.
Receiver code used to indicate a flush error even if the incoming packet
is a keyframe, forcing a request of a keyframe. Now it takes this
keyframe into account and doesn't error as the stream is decodable from
this point.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15549005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6188 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-19 15:28:02 +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
f31a47abdc VCM:Accounting for bounds when inserting packets. We currently receive indicators to the first and last packets of the frame, but not have any sanity to verify that all packets are indeed within the bounds (when available). This cl attempts to fix that,
BUG=
R=marpan@google.com

Review URL: https://webrtc-codereview.appspot.com/2077004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4614 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-26 17:10:11 +00:00
b2c28c3699 Relanding 4597 - Don't force key frame when decoding with errors.
Makes sure that incomplete key frame or delta frames will be released from the JB when decoding with errors.
The decoder in turn will trigger a PLI until a complete key frame is received in order to start a session.

TBR=stefan@webrtc.org

BUG=

Review URL: https://webrtc-codereview.appspot.com/2097004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4607 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-23 21:54:50 +00:00
ceea41d135 Revert 4597 "Don't force key frame when decoding with errors"
> Don't force key frame when decoding with errors
> 
> BUG=2241
> R=stefan@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/2036004

TBR=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2093004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4600 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-23 00:53:24 +00:00
44af55cc44 Don't force key frame when decoding with errors
BUG=2241
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2036004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4597 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-22 23:29:43 +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
12dc1a38ca Switch C++-style C headers with their C equivalents.
The C++ headers define the C functions within the std:: namespace, but
we mainly don't use the std:: namespace for C functions. Therefore we
should include the C headers.

BUG=1833
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1917004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4486 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-05 16:22:53 +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
1a7b9b94be Cleanup WebRTC tracing
The goal of this change is to:
1. Remove unused tracing events.
2. Organize tracing events to facilitate measurement of end to end latency.

The major change in this CL is to use ASYNC_STEP such that operation
flow can be traced for the same frame.

R=marpan@webrtc.org, pwestin@webrtc.org, turaj@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1761004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4308 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-08 21:31:18 +00:00
5616abadf5 Suppress excessive logging in video_coding
Only prints the warning message if a frame was dropped.

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1735004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4278 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-27 19:47:40 +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
b1bba167f4 Prevent excessive logging in jitter buffer
Jitter buffer logs a message when it is going to recycle frames. This adds a
lot of noise even in normal operation. This change make sure only critical
cases are logged.

R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1580007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4150 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-31 18:52:16 +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
c74c3c2447 Adds integration test for RTX and fixes bugs found.
BUG=1811
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4096 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-23 13:48:22 +00:00
a7dc37d568 Log the type of recycled frames.
Also correct the logging of incoming key frame packets.

BUG=1814
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1537004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4090 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-23 07:21:05 +00:00
8c49c1eab3 Log a message when a key frame packet is received
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1518004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4089 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-22 21:18:59 +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
fe307e1332 Add one unit test for NACKing a key frame
Adding a test case that wasn't covered. This new test is passing.

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1475004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4051 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-16 21:19:59 +00:00
2038214c77 Log too long non-decodable duration events.
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1488004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4043 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-16 11:39:06 +00:00
cb20a5b2d7 VCM/JB: Bug fix in ExtractAndSetDecode
BUG=1771
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1466005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4035 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-15 17:10:44 +00:00
1673481ed7 Fixes a bug where the render buffer size (and indirectly the non-continuous duration) was computed incorrectly.
BUG=1769
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1473004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4026 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-14 12:00:47 +00:00
7bfb3a3227 Add more tracing for key frames.
R=mallinath@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1428004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4015 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-13 22:59:00 +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
9c7685f9a6 VCM/JB: Break and skip to key if possible
BUG=1734
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1421004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3969 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-07 16:07:52 +00:00
a5dee33639 Correctly add packets to nack list when sequence number wraps.
BUG=1737
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1427004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3966 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-07 11:11:17 +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