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389 Commits

Author SHA1 Message Date
stefan@webrtc.org
5b7120c81b Fix two issues where we might end up busy looping in decoder_render mode. 
This happens if
- Next frame is far into the future (> 200 ms).
- Next frame is ready for decode/render but incomplete.

BUG=1696
TESTS=trybots

Review URL: https://webrtc-codereview.appspot.com/1354005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3914 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-29 16:41:30 +00:00
pwestin@webrtc.org
b0061f94b2 Enable Nack pacing. 
Review URL: https://webrtc-codereview.appspot.com/1357004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3912 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-27 00:41:08 +00:00
andrew@webrtc.org
1acb3b33bc Add comfort noise disabling and routing mode selection to audioproc. 
TBR=bjornv

Review URL: https://webrtc-codereview.appspot.com/1358004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3907 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-26 00:39:27 +00:00
mikhal@webrtc.org
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
mikhal@webrtc.org
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
mikhal@webrtc.org
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
pwestin@webrtc.org
52aa019e98 Avoid adding duplicates in pacer lists. 
Review URL: https://webrtc-codereview.appspot.com/1329007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3899 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-25 17:35:56 +00:00
stefan@webrtc.org
cb60fb2e6c Make sure timestamps are monotonically increasing. 
BUG=1563

Review URL: https://webrtc-codereview.appspot.com/1325008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3898 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-24 23:54:31 +00:00
andrew@webrtc.org
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
mikhal@webrtc.org
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
stefan@webrtc.org
8ca8a71de2 Revert "Add a default RTT to CallStats and use different values for buffered/real-time mode." 
This reverts commit aae26db1da5803482b094357c546b8454ab1c26d.

BUG=1613
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1327008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3890 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-23 16:48:32 +00:00
turaj@webrtc.org
a942692725 Buf fix for r3883. 
Review URL: https://webrtc-codereview.appspot.com/1319012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3889 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-23 16:08:29 +00:00
stefan@webrtc.org
ccd4b2aec8 Add a default RTT to CallStats and use different values for buffered/real-time mode. 
BUG=1613

Review URL: https://webrtc-codereview.appspot.com/1326007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3888 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-23 15:58:23 +00:00
pbos@webrtc.org
d25b602dc0 VP8: Avoid copying the codec struct on Reset(). 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1319013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3887 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-23 13:08:04 +00:00
mflodman@webrtc.org
efdf778d3f BUG=1351 
Propose of this CL: Close the camera properly on MacOS in order to allow other apps to use it.

Changes in this CL:
1. video_capture_qtkit_info_objc.mm _captureDevicesInfo is never released. I have found this memory leak using Instruments from XCode. The patch is releasing it in dealloc.

2. In video_capture_qtkit_objc.h:
a) _captureDeviceName is not needed. Is allocated in the class but never used.
b) I don't see the role of the  NSAutoreleasePool. also if you use it you have to release it when the class is destroyed. Otherwise you will leak memory. Libjingle has for each thread a pool on mac os.

3. In video_capture_qtkit_objc.mm
a) the camera is not stopped properly . See the changes from dealloc. NOTE : If you don't call [[_captureVideoDeviceInput device] close] other apps will not be able to use the camera since you are not closing your app

b) Removed QTCaptureDevice* videoDevice = (QTCaptureDevice*)[_captureDevices objectAtIndex:0]; I don't know why this because the desired camera is opened in setCaptureDeviceById and can be different than position 0 in the camera array. At this moment if you have two cameras and user want to pick the one on index 1 the app also locks the one on 0 .

Other changes I have done to improve (and are not in this CL):
a) I have set the FPS properly to the desired. I have succeeded to reduce the CPU with 3 % doing this. The current code for setting FPS is commented in webrtc
b) I have removed _rLock from the equation. I don't know if it's good or not but I hadn't understood what exactly we are trying to protect with this. Anyway in the current implementation is never released.

Review URL: https://webrtc-codereview.appspot.com/1097014

Patch from Silviu Caragea <silviu.cpp@gmail.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3886 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-23 11:57:56 +00:00
mikhal@webrtc.org
c1f243f8e7 VCM/JB: Skip to the next complete key frame 
Review URL: https://webrtc-codereview.appspot.com/1317006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3885 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-22 22:24:38 +00:00
turaj@webrtc.org
28d54ab18f Improve AV-sync when initial delay is set and NetEq has long buffer. 
Review URL: https://webrtc-codereview.appspot.com/1324006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3883 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-22 18:53:35 +00:00
kjellander@webrtc.org
1b427719dc emove desktop_capture.gypi from modules.gyp 
When adding this in
we started getting linking problems on the mac_asan bot due to
the empty list of source files for the library target.

Please re-add it into modules.gyp when the library has source files
to compile.

BUG=none
TEST=Passing mac_asan trybot.

Review URL: https://webrtc-codereview.appspot.com/1313009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3882 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-22 13:13:49 +00:00
mikhal@webrtc.org
e0e029e8cb Revert 3876 
Review URL: https://webrtc-codereview.appspot.com/1341005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3877 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-19 21:11:41 +00:00
mikhal@webrtc.org
ee184b9520 VCM/Receiver: Only update render time when decoding 
Review URL: https://webrtc-codereview.appspot.com/1336004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3876 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-19 19:15:47 +00:00
mikhal@webrtc.org
a73d52ca52 revert r3871 
TBR= solenberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1331004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3872 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-18 20:26:28 +00:00
solenberg@webrtc.org
9756017717 - Replace the BWE_MIN and BWE_MAX macros with std::min and std::max 
- Add 'virtual' to a bunch of overridden methods of RemoteBitrateEstimatorMultiStream and RemoteBitrateEstimatorSingleStream.

BUG=

Review URL: https://webrtc-codereview.appspot.com/1324005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3871 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-18 19:12:42 +00:00
solenberg@webrtc.org
d26457fdeb Apply Chromium C++ style to BitRateStats. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1325006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3870 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-18 12:25:32 +00:00
braveyao@webrtc.org
c14b728b71 Add lock to prevent possible rare race condition in Win coreAudio capture implementation. 
BUG = 
TEST = voe_auto_test

Review URL: https://webrtc-codereview.appspot.com/1320011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3868 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-18 09:32:07 +00:00
sergeyu@chromium.org
a0cd9182aa Add desktop_capture directory for screen and window capturers. 
The screen captures will be moved from chromium to WebRTC to make it easy
to share this code with other projects. This CL adds a new directory where
the current screen capturer code will be moved to.

Review URL: https://webrtc-codereview.appspot.com/1297005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3866 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-17 18:21:42 +00:00
mikhal@webrtc.org
dbd6a6d653 Updating delay for first value 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1327005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3865 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-17 16:23:22 +00:00
andresp@webrtc.org
48c5882f2a Remove libvpx pre-processor conditions and conditional compile of default temporal layers files. 
R=stefan@webrtc.org,marpan@webrtc.org
BUG=201

Review URL: https://webrtc-codereview.appspot.com/1323005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3864 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-17 15:31:40 +00:00
tina.legrand@webrtc.org
db11fab49e Adding Opus unit test 
This CL adds a unit test for Opus, as well as new APIs for true stereo decoding (skipping master/slave approach).

BUG=

Review URL: https://webrtc-codereview.appspot.com/1222006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3860 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-17 10:39:41 +00:00
turaj@webrtc.org
f1a3b4bc0c Issue 1647. Avoid unsequenced modification. 
issue=1647
test=trybots,manual

Review URL: https://webrtc-codereview.appspot.com/1327004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3858 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-16 17:01:35 +00:00
pbos@webrtc.org
6e788df19e Remove vim/emacs modelines from .gypi files 
BUG=1655

Review URL: https://webrtc-codereview.appspot.com/1326005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3857 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-16 12:40:34 +00:00
solenberg@webrtc.org
56b5f77a2b Add support for multiple streams to RtpPlayer: 
- Tests video_rtp_play.cc, video_rtp_play_mt.cc, decode_from_storage.cc rewritten
 - rtp_player.cc/.h rewritten; added interfaces for externally setting up sinks
 - Support for reading .rtp files pulled out into rtp_file_reader namespace
 - Added support for reading .pcap (libpcap/wireshark/tcpdump) files, see pcap_file_reader

BUG=
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/1201009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3856 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-16 10:31:56 +00:00
stefan@webrtc.org
885cd13356 Start NACKing as soon as we have the first packet of a key frame. 
BUG=1605

Review URL: https://webrtc-codereview.appspot.com/1307007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3855 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-16 09:38:26 +00:00
stefan@webrtc.org
bdb9b971be Change receive statistics bitrate to be provided in bps instead of kbps. 
BUG=1469

Review URL: https://webrtc-codereview.appspot.com/1326004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3854 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-16 09:02:03 +00:00
turaj@webrtc.org
92d1f07551 Elevate NetEq short-term activity statistics to ACM level for logging. 
Review URL: https://webrtc-codereview.appspot.com/1313004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3850 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-15 16:52:04 +00:00
kjellander@webrtc.org
4b8de90dce Disable -Wunsequenced warning in audio_coding_module 
BUG=1647
TEST=Compile locally on Linux with clang enabled.
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1316005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3848 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-15 06:38:56 +00:00
mikhal@webrtc.org
c2a3aa7926 Partial revert of r3844 
Review URL: https://webrtc-codereview.appspot.com/1320004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3845 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-12 19:53:30 +00:00
mikhal@webrtc.org
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
mflodman@webrtc.org
9f5ebb5251 Adding a payload type for RTX. 
BUG=736
TEST=Modified RTP unittests.

Review URL: https://webrtc-codereview.appspot.com/1278004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3843 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-12 14:55:46 +00:00
stefan@webrtc.org
b8e7f4cc97 Change capture interface to use NTP capture time. 
Move NTP functionality to Clock.

BUG=1563
TEST=trybots and vie_auto_test --automated

Review URL: https://webrtc-codereview.appspot.com/1313005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3842 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-12 11:56:23 +00:00
mikhal@webrtc.org
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
stefan@webrtc.org
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
stefan@webrtc.org
122d209e67 Fixes an issue where the start bitrate is stored in kbps instead of bps. 
BUG=1638
TEST=trybots and vie_auto_test loopback with nack.

Review URL: https://webrtc-codereview.appspot.com/1312004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3831 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-11 17:21:40 +00:00
wu@webrtc.org
eac36b8561 Fix -Wstring-conversion warnings. 
Review URL: https://webrtc-codereview.appspot.com/1299007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3830 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-11 15:37:46 +00:00
andresp@webrtc.org
523f93729b Re-write the build of the nacklist. 
Review URL: https://webrtc-codereview.appspot.com/1304008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3822 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-11 11:30:39 +00:00
fischman@webrtc.org
f2a97fc2b4 WebRTCDemo: handle stride!=width from first frame. 
Previously only mid-stream frames handled stride!=width correctly.

BUG=1615

Review URL: https://webrtc-codereview.appspot.com/1304009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3821 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-10 23:21:10 +00:00
pbos@webrtc.org
e4b6064f8e Replace legacy G_CONST with const. 
BUG=1608

Review URL: https://webrtc-codereview.appspot.com/1310005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3814 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-10 18:06:57 +00:00
pbos@webrtc.org
ab9202b673 Removing remaining WebRtc_Word32 not in typedefs.h 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1306006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3813 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-10 17:59:17 +00:00
fischman@webrtc.org
77d59fe408 WebRTCDemo: no-op out instead of NPEing on destroyed camera. 
BUG=1617

Review URL: https://webrtc-codereview.appspot.com/1310004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3812 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-10 17:11:51 +00:00
pbos@webrtc.org
dfc5bb9c97 WebRtc_Word32 -> int32_t in video_capture/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1298005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3811 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-10 08:23:13 +00:00
pbos@webrtc.org
ddf94e71e5 WebRtc_Word32 -> int32_t in video_render/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1304006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3810 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-10 08:09:04 +00:00