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Author SHA1 Message Date
mikhal@webrtc.org
8f86cc8712 VCM/Receiver: Return null when can't extract frame. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1435004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3974 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-07 18:05:21 +00:00
mikhal@webrtc.org
474e915272 Relanding 3962: VCM/JB: Porting jitter_buffer_test to gtest 
TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1434004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3971 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-07 16:55:03 +00:00
mikhal@webrtc.org
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
mikhal@webrtc.org
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
pbos@webrtc.org
3004c79c6a Fix clang errors in non-GYP_DEFINES=clang=1 build 
BUG=1623
R=stefan@webrtc.org, tina.legrand@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1368004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3968 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-07 12:36:21 +00:00
stefan@webrtc.org
d3a1959678 Fix jitter buffer unittest. 
TBR=mflodman@webrtc.org
BUG=1737

Review URL: https://webrtc-codereview.appspot.com/1430005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3967 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-07 12:35:58 +00:00
stefan@webrtc.org
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
stefan@webrtc.org
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
stefan@webrtc.org
273759048c Revert r3956 "VCM/JB: Porting jitter_buffer_test to gtest." 
TBR=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1408005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3962 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-06 13:12:58 +00:00
mikhal@webrtc.org
45f2da0920 VCM/JB: Porting jitter_buffer_test to gtest. 
Tests were not modified, but ported as is.

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1391004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3956 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-03 22:22:46 +00:00
mikhal@webrtc.org
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
stefan@webrtc.org
4980679d35 Fixes two bugs in receive statistics. 
- Reported bitrate wasn't reset correctly when no frames had been received.
- Internal framerate estimate wasn't reset when no frames had been received.

BUG=1713

Review URL: https://webrtc-codereview.appspot.com/1377004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3924 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-30 22:05:07 +00:00
pwestin@webrtc.org
d35964a1ce Fixing AV sync. 
Increased 2 const to allow for a bigger difference in AV sync.

BUG=1711

Re-wrote the ComputeDelays to be readable and remove the possibilities of returning values lower than base_target_delay_ms

R=mflodman@webrtc.org, mikhal@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1367004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3922 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-30 16:06:10 +00:00
mikhal@webrtc.org
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
mikhal@webrtc.org
865ada3a52 Don't reset the last je value and mode 
Review URL: https://webrtc-codereview.appspot.com/1369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3916 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-29 19:09:41 +00:00
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
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
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
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
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
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
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
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
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
hclam@chromium.org
806dc3b0e6 More trace events 
The goal of this change is to unify tracing events styles
and add trace events for all RTP traffic.

BUG=1555
Review URL: https://webrtc-codereview.appspot.com/1290007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3806 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 19:54:10 +00:00
stefan@webrtc.org
4d2f5de67a Improve how NACK lists are generated before a frame has been decoded. 
BUG=1598

Review URL: https://webrtc-codereview.appspot.com/1295004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3805 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 18:24:41 +00:00
edjee@google.com
79b0289bfc Adds event traces and counters for WebRTC receive side. 
Review URL: https://webrtc-codereview.appspot.com/1279005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3766 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-04 19:43:34 +00:00
pbos@webrtc.org
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
marpan@webrtc.org
29f34b8727 Fix for issue: https://code.google.com/p/webrtc/issues/detail?id=1549 
Review URL: https://webrtc-codereview.appspot.com/1270004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3741 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-28 18:57:46 +00:00
stefan@webrtc.org
bfacda60be Add interface to signal a network down event. 
- In real-time mode encoding will be paused until the network is back up.
- In buffering mode the encoder will keep encoding, and packets will be
  buffered at the sender. When the buffer grows above the target delay
  encoding will be paused.
- Fixes a couple of issues related to pacing which was found with the new test.
- Introduces different max bitrates for pacing and for encoding. This allows
  the pacer to faster get rid of the queue after a network down event.

(Work based on issue 1237004)

BUG=1524
TESTS=trybots,vie_auto_test

Review URL: https://webrtc-codereview.appspot.com/1258004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3730 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-27 16:36:01 +00:00
stefan@webrtc.org
836af79f58 Remove incorrect asserts. 
BUG=1527

Review URL: https://webrtc-codereview.appspot.com/1214006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3698 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-21 12:15:44 +00:00
stefan@webrtc.org
3d0b0d6902 Follow-up fix for r3681. 
TESTS=trybots and vie_auto_test
BUG=1514

Review URL: https://webrtc-codereview.appspot.com/1216006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3689 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-19 10:04:57 +00:00
stefan@webrtc.org
f4944d49cf Fix framerate sent to account for actually sent frames. 
TESTS=trybots
BUG=1481

Review URL: https://webrtc-codereview.appspot.com/1195005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3682 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-18 17:04:52 +00:00
stefan@webrtc.org
abc9d5b6aa Change VCM interface to take target bitrate in bits per second. 
This also solves issue 1469.

TESTS=trybots
BUG=1469

Review URL: https://webrtc-codereview.appspot.com/1215004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3681 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-18 17:00:51 +00:00
stefan@webrtc.org
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
kjellander@webrtc.org
971278a962 Splitting out video_coding_test executable again. 
This CL undoes the merge of the developer test tool and the gtest tests
that was merged in https://code.google.com/p/webrtc/source/detail?r=3176

Doing that, we get a pure gtest executable of
video_coding_integrationtests which can run properly on the bots.

BUG=none
TEST=Trybots passing + local execution on Linux with:
out/Debug/video_coding_integrationtests --gtest_print_time (to ensure it will be possible to run with runtest.py)

Review URL: https://webrtc-codereview.appspot.com/1171007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3638 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-08 10:20:53 +00:00
stefan@webrtc.org
84cd8e39cf Disable frame dropper for screenshare mode. 
BUG=1466

Review URL: https://webrtc-codereview.appspot.com/1170004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3629 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-07 13:12:32 +00:00
stefan@webrtc.org
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
stefan@webrtc.org
9e254133ad Rewrite the jitter buffer statistics test and put make it robust under valgrind. 
BUG=1158

Review URL: https://webrtc-codereview.appspot.com/1116008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3579 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-28 08:45:23 +00:00
stefan@webrtc.org
eb91792cfd Refactoring temporal layers implementation and adding VideoCodecMode for easier control of codec settings. 
Review URL: https://webrtc-codereview.appspot.com/1105007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3528 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-18 14:40:18 +00:00