Commit Graph

161 Commits

Author SHA1 Message Date
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
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
724947b8ef Add SwapFrame() to VideoSendStreamInput.
Optionally prevents doing a frame copy when putting frames into a
VideoSendStream. PutFrame() is still there, which copies the frame.

Also removes time_since_capture_ms as a parameter, since
I420VideoFrame::render_time_ms() denotes when the frame was captured.

BUG=2657
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5265 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11 16:26:16 +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
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
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
0aeb22e32c Adding tl0idx consideration for continuity
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5046 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-28 22:26:14 +00:00
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
37bb4974e7 Expose VideoCodingModule's decoder stats up the stack from VCMTiming to chrome://webrtc-internals.
R=juberti@google.com, mikhal@webrtc.org, stefan@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5027 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-23 23:59:45 +00:00
be9c560aab Revert r4913 that reverts r4911. Original CL description:
"Adding temporal layer strategy that keeps base layer framerate at an acceptable value."

R=holmer@google.com, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2351006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4920 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-04 13:11:31 +00:00
44db9d1a57 Revert 4911 "Adding temporal layer strategy that keeps base laye..."
> Adding temporal layer strategy that keeps base layer framerate at an acceptable value.
> 
> R=stefan@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/2272005

TBR=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4913 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-03 17:42:07 +00:00
26f78f7ecb Adding temporal layer strategy that keeps base layer framerate at an acceptable value.
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2272005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4911 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-03 14:06:14 +00:00
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
30377c7f71 Change the parameters of calculating maximum decode time.
- Reduce the window size from 20 to 10 seconds. If there is
  any spurious high decode time, it will be faster to pass it.
- Ignore more samples at first because HW decoder has higher
  initialization latency.

BUG=crbug.com/298176
TEST=Run apprtc loopback on Chromebook Daisy.
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2315005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4874 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-28 06:06:18 +00:00
544b17c6a9 Implemented AutoMuter in MediaOptimization
Also added a unittest. This is the first step towards creating an
AutoMuter function in WebRTC.

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

Review URL: https://webrtc-codereview.appspot.com/2294005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4857 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-26 12:05:15 +00:00
054ccd2e35 Remove include_dirs from video_coding.
BUG=1662
TEST=compile on trybots
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2294007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4853 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-26 09:22:09 +00:00
641587f938 Disable some VP8 tests on Android.
DecodeWithACompleteKeyFrame and FixedTemporalLayersStrategy.

TBR=andresp
BUG=2037

Review URL: https://webrtc-codereview.appspot.com/2283004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4829 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-24 18:43:28 +00:00
b426c469b9 MediaOptimization: Converting a few members to scoped_ptrs
For consistency with other parts of the code.

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2275006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4822 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-24 07:41:53 +00:00
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
98fcd2d4c3 Adding unit tests for default temporal layer strategy.
R=mflodman@webrtc.org, stefan@webrtc.org, stefan

Review URL: https://webrtc-codereview.appspot.com/2235005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4810 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-23 11:12:59 +00:00
4f3624d39e Avoid recursively taking critical section.
TEST=trybots
BUG=2261
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2258006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4800 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-20 07:43:17 +00:00
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
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
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
e401c2e391 Split video coding module unit tests into sender and receiver unit tests.
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2199005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4753 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-16 20:29:13 +00:00
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
1f09dbe353 Moving test-only code (stream_generator) out of vcm implemention.
R=niklas.enbom@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2207004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4740 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-13 19:17:54 +00:00
554d158ce6 Reset jitter buffer and timing if frames are getting too much delay.
BUG=chromium/263867
TEST=trybots
R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2177005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4721 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-11 08:45:26 +00:00
021c42bfa8 Lock use of _packetRequestCallback in VCM.
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2190006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4708 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-09 17:18:31 +00:00
0181b5f8dd ExternalVideoDecoder for new VideoEngine API.
Implements the ExternalVideoDecoder interface for VideoReceiveStream.
Also adds a FakeDecoder used in tests, removing the overhead of running
the EngineTest tests with VP8 under Memcheck/TSan, allowing us to enable
them under Memcheck/TSan as well.

BUG=2346,2312
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2172004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4702 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-09 08:26:30 +00:00
7bb8f02274 Adds support for combining RTX and FEC/RED.
This is accomplished by breaking out RTX and FEC/RED functionality from the RTP module and keeping track of the base payload type, that is the payload type received when not receiving RTX.

Enables retransmissions over RTX by default in the loopback test.

BUG=1811
TESTS=voe/vie_auto_test --automated and trybots.
R=mflodman@webrtc.org, pbos@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2154004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4692 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-06 13:40:11 +00:00
f1e807c0e5 Removing FrameForStorage
R=pwestin@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2142004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4688 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-05 22:34:41 +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
822fbd8b68 Update talk to 50918584.
Together with Stefan's http://review.webrtc.org/1960004/.

R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2048004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4556 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-15 23:38:54 +00:00
dde7d4c6ed Roll chromium_revision 214260:217707 and gflags 45:84
gflags roll is needed mostly to pick up fixes for warnings triggered by newer
compiler/settings pulled in by the chromium roll.  Had to switch from the old
google-gflags project the current gflags project to pick up this fix (see
https://code.google.com/p/gflags/source/detail?r=74 for details).

Update android build.xml file to reflect tools moves in new SDK pulled in by the chromium_revision roll.

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

Review URL: https://webrtc-codereview.appspot.com/2043004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4555 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-15 23:31:30 +00:00
0d94c2f81c Avoid acquiring VCM::_receiveCritSect during decode callback.
When VideoDecoder::Decode, Reset, or Release is called,
VideoCodingModuleImpl::_receiveCritSect may have been
acquired. Decode callback needs to acquire the same lock
in ViEChannel::FrameToRender. It is not a problem for
SW decode because decode callback is run on the same
WebRTC decoding thread and the lock is re-entrant. But
for HW decode, decode callback is run on a thread different
from WebRTC decoding thread. Decode callback gets the locks
in the opposite order. Deadlock can happen.

BUG=http://crbug.com/170345
TEST=Try apprtc.appspot.com/?debug=loopback on ARM Chromebook Daisy.
     Run libjingle_peerconnection_unittest.

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1997005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4523 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-12 14:20:49 +00:00
64799da6c6 Allowing decoding with errors, when disabling nack.
BUG=1897
R=stefan@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1982004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4508 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-08 22:45:33 +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
aa4d96a134 Revert r4301
R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4357 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-16 19:25:04 +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
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