82a846f0cb
Adding Ami to the video renderer and capturer modules.
...
TBR=fischman@webrtc.org ,wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2202006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4735 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-12 17:43:17 +00:00
36cf4d2309
The video render module for iOS.
...
BUG=2105, 2028
R=fischman@webrtc.org , mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2064004
Patch from SeungJae Lee <sjlee@webrtc.org >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4734 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-12 17:39:53 +00:00
e509f943ed
This issue is related to
...
https://chromereviews.googleplex.com/9908014/
I was thinking about shipping ACM2 from the signal repository. There seems to be too many changes in one CL.
BUG=
R=andrew@webrtc.org , turaj@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2171004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4733 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-12 17:03:00 +00:00
8fa03a15ab
Make PCM16 available in Chromium builds.
...
PCM16 can be useful for unit tests in Chromium. In particular Mikhal
would like to use it for ChromeCast.
This currently (r222592) has no impact on Chrome binary size, presumably
because PCM16 is unused and the linker strips the symbols.
To measure the potential impact, I looked at the size (bytes) of
out/Release/vie_auto_test on Linux with various codecs removed:
r4724 : 4567384
No PCM16 : 4565936
No ILBC : 4500424
No G722 : 4555800
No RED : 4565880
Giving the following size increases of adding each codec:
PCM16 : 1.4 kB (0.03%)
ILBC : 70.0 kB (1.49%)
G722 : 11.6 kB (0.25%)
RED : 1.5 kB (0.03%)
R=mikhal@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2195005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4732 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-12 01:30:30 +00:00
89df092807
Make the destructor of AudioCodingModule public.
...
This allows the type to be used with a scoped_ptr. Remove all calls to
the deprecated Destroy() from tests.
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2200006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4731 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-12 01:27:43 +00:00
5eb997a2fd
Fix unsigned/signed comparison error due to r4729.
...
Fix it for real by switching to ints rather than casting.
TBR=xians
Review URL: https://webrtc-codereview.appspot.com/2191009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4730 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-12 01:01:42 +00:00
8f94013651
Reduce frequency of high audio delay warning logs.
...
This will log the warning every 5 seconds instead of every 10 ms.
BUG=b/10674993
TESTED=Ran voe_cmd_test with hard-coded high delay. Observed a log
every 5 seconds.
R=noahric@chromium.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2184009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4729 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-11 22:35:00 +00:00
256b83146c
Removes function that is not used anywhere but somehow still causing library load issues on Android Release build.
...
BUG=2364
R=andrew@webrtc.org , fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2190008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4728 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-11 20:43:13 +00:00
5c678eabd9
Implement 'abs-send-time' extension in VideoSendStream.
...
BUG=2229
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2184010
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4727 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-11 19:00:39 +00:00
6138c5cfa4
OpenSl: fixes crashes externally reported in issue 2361 and 2362.
...
BUG=2361,2362
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2196008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4726 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-11 18:50:06 +00:00
036b7436df
Adding APIs. These APIs are not implemented yet, they are to help developement of ACM.
...
Un-implemented APIs.
TBR=henrik.lundin@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/2191008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4725 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-11 18:45:02 +00:00
d4d59ac871
Remove FrameForStorage:Follow up on r4688
...
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2201004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4723 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-11 15:18:15 +00:00
2902328cce
Implement 'toffset' extension in VideoSendStream.
...
BUG=2229
R=holmer@google.com , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4722 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-11 10:14:56 +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
835ef67d14
Remove repeated conditions key.
...
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2196007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4720 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-11 00:16:00 +00:00
82f014aa0b
OpenSL (not default): Enables low latency audio on Android.
...
BUG=1669
R=andrew@webrtc.org , fischman@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2032004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4719 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-10 18:24:07 +00:00
319c98d663
Fix format string in video_quality_analysis.cc.
...
Fixes compilation errors on Android and Linux32 targets.
TBR=kjellander@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/2196005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4717 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-10 15:23:50 +00:00
182d025d94
Remove include_dirs from voice_engine.gyp.
...
BUG=1662
R=xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2193004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4716 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-10 15:01:26 +00:00
df531a2eee
Test that VideoSendStream responds to NACK.
...
BUG=2228
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2194006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4715 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-10 14:56:33 +00:00
f880f863dd
Convert printing in video quality tests to Chromium's perf format.
...
Add support for --label flag to the frame_analyzer, that
decides what label shall be used for the perf output.
BUG=none
TEST=
Make sure to have zxing and ffmpeg in the PATH.
Create a captured video (from running vie_auto_test custom call)
webrtc/tools/compare_videos.py --ref_video=reference_video.yuv --test_video=captured_output.yuv --frame_analyzer=out/Release/frame_analyzer --label=TEST_VGA
And then inspecting the output that is prefixed with RESULT.
R=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2190005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4714 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-10 12:10:01 +00:00
e07049f19f
Lock RTPSender statistics.
...
Suppressing these errors in TSan has become tedious. It's better to just
lock them.
BUG=2349
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2197004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4713 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-10 11:29:17 +00:00
744fbc7fe4
Split up EngineTests and RampupTests.
...
This allows having one group of tests per file, the test files are
long enough as they are.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2196004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4712 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-10 09:26:25 +00:00
eda189be14
Remove redundant STR_CASE_CMP macro definitions.
...
R=minyue@webrtc.org , turaj@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2187005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4711 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-09 17:50:10 +00:00
a19c9f4173
Updated WebRTC version to 3.41
...
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2190007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4709 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-09 17:23:44 +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
7ebf0e7f44
Remove include_dirs from video_engine_core.gypi.
...
BUG=1662
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2181005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4707 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-09 16:56:31 +00:00
59f20bb735
Break out RTCPSender dependency on ModuleRtpRtcpImpl.
...
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2191004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4706 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-09 16:02:19 +00:00
841c8a44bb
Rename VideoCall to Call.
...
Call should encompass more than video, there's no point in calling it
VideoCall.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2191005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4704 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-09 15:04:25 +00:00
86136a0e8f
Re-enable tests for Remote Bitrate Estimator
...
BUG=
R=stefan@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2179004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4703 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-09 13:06:52 +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
30e055c4dd
Handle empty RTP video packets agnostic to codec.
...
Sending empty RTP packets caused a crash when using a generic codec
instead of VP8. This fix moves handling of empty RTP packets out of
ReceiveVp8Codec and into ParseRtpPacket.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2185004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4701 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-08 11:15:00 +00:00
b159c2e3dd
Reduce cost of PushSincResampler::Resample().
...
Ideally, PushSincResampler would have very little overhead on
SincResampler. This gets closer to that ideal.
Replace std::min/max and floor with inline functions. Add a benchmark
test to verify the improvement.
On a MacBook Retina, this results in PushSincResampler::Resample()
accounting for ~1% of CPU usage on voe_cmd_test vs the earlier ~2%
(with ISAC16 and 48 kHz audio devices).
Using the new benchmark, this results in a performance improvement of:
16 -> 44.1 : 1.7x
16 -> 48 : 1.9x
32 -> 44.1 : 1.6x
32 -> 48 : 1.7x
44.1 -> 16 : 1.5x
44.1 -> 32 : 1.7x
44.1 -> 48 : 1.7x
48 -> 16 : 1.5x
48 -> 32 : 1.5x
48 -> 44.1 : 1.8x
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2157005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4695 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-06 21:15:55 +00:00
c7f708679d
Clamp camera id to legal values.
...
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2184004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4694 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-06 18:17:45 +00:00
b2c8a952a7
Improving padding rules and breaking out bw allocation to ViEEncoder.
...
BUG=1837
TESTS=vie_auto_test --automated, trybots
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2170004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4693 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-06 13:58:01 +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
5500d93fe5
Add temporal layer factory.
...
R=marpan@google.com , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2180004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4691 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-06 11:26:15 +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
aa3d1c8169
Make unittest log printouts opt-in with a --logs flag.
...
TESTED=Using modules_unittests, no logs are printed by default.
Specifying --logs prints logs. gtest flags work correctly.
R=henrike@webrtc.org , kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2181004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4686 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-05 21:16:29 +00:00
bebf3995ce
Pre-multiply images for MouseCursorShape.
...
BUG=chromium:267270
R=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/2173004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4685 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-05 19:32:46 +00:00
31b4a5ac82
Recognize armv7 target_arch for ios support in webrtc common.gyp
...
BUG=2343
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2176004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4684 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-05 16:46:36 +00:00
9080518a39
Restore severity precondition to logging.h.
...
I mistakenly ommitted the checks when logging.h was ported from
libjingle to webrtc. This caused a significant CPU cost for logs which
were later filtered out anyway.
Verified with LS_VERBOSE logging in neteq4, running:
$ out/Release/modules_unittests \
--gtest_filter=NetEqDecodingTest.TestBitExactness \
--gtest_repeat=50 > time.txt
$ grep "case ran" time.txt | grep "[0-9]* ms" -o | sort
Results on a MacBook Retina, averaged over 5 runs:
Verbose logs disabled: 666 ms
Exisiting implementation, verbose logs enabled: 944 ms (1.42x)
New implementation, verbose logs enabled: 673 ms (1.01x)
BUG=2314
R=henrik.lundin@webrtc.org , henrike@webrtc.org , kjellander@webrtc.org , turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2160005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4682 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-05 16:40:43 +00:00
95e51f509c
Remove send and receive streams when destroyed.
...
Fixes crash where packets were sent to a receive stream that had been
destroyed but not removed from the ssrc mapping from call to receiver.
Added a repro case that reliably crashed before the fix.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2161007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4681 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-05 12:38:54 +00:00
164c4f71ba
Add clockdrift to RtpGenerator
...
RtpGenerator is a help class for NetEq testing. This change
add the possibility to simulate clockdrift. If no clockdrift is
set, the default is 0 (i.e., no drift).
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2175005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4680 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-05 12:16:38 +00:00
7e1bf318bf
Allow unknown flags in test_main.cc.
...
Adds AllowCommandLineParsing to allow us to ignore "--no-sandbox" given
by new TSanV2 bots. Not ignoring this flag prevents the test from
running on this machine. Also removing unnecessary asserts that clutter
code.
BUG=
TEST=Locally running video_engine_tests with --no-sandbox.
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2178004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4679 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-05 10:27:46 +00:00
36439bf906
NetEq4: Small change to reduce allocs in AudioMultiVector
...
This change reduced the allocation count by 20000 in the bit-exactness
test.
BUG=Issue 1363
TEST=out/Debug/modules_unittests --gtest_filter=NetEqDecodingTest.TestBitExactness
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2157004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4678 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-05 06:02:56 +00:00
e2d4da6586
Enable EngineTest.ReceivesPliAndRecoversWithNack and fix memcheck suppression filter.
...
BUG=2346
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2169004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4677 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-04 14:21:57 +00:00
77bf5c28c8
Clean capture timestamp code.
...
BUG=
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2134004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4675 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-04 11:35:43 +00:00
06f1f74331
Disable EngineTest.ReceivesPliAndRecoversWithNack.
...
The test times out on Linux memcheck bot at times.
BUG=2348
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2159007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4674 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-04 11:00:07 +00:00
b21e528c60
Protecting Bitrate to avoid data race found by tsan.
...
TEST=try and vie_auto_test with tsan.
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2163004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4673 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-04 08:42:44 +00:00
65abb6b1ed
Revert 4671 "Enable SetInitialPlayoutDelay on Android."
...
Tests enabled in r4671 failed:
build.chromium.org/p/client.webrtc/builders/Android%20Tests/builds/31/steps/slave_steps/logs/stdio
> Enable SetInitialPlayoutDelay on Android.
>
> Background:
> In Chrome mirroring which uses 500ms buffering mode,
> audio video mismatch happens in the begining because of the lack of the api.
>
> BUG=b/10538425
> TEST=pass 'git try' except tests which is aleady broken in the bot. pass 'build/android/test_runner.py gtest -s modules_tests --verbose --release -f *InitialPlayoutDelayTest*'
> R=henrika@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/2144004
TBR=dwkang@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2160006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4672 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-04 07:47:39 +00:00
