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265a5d298aeff39a31defb51a80569844a232831
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114 Commits

Author SHA1 Message Date
pbos@webrtc.org
265a5d298a Remove traces of deprecated WebRtc_Word types. 
BUG=314
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1385004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3933 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-02 09:11:20 +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
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
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
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
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
marpan@webrtc.org
557e92515d Reapply the reverted r3747. 
https://code.google.com/p/webrtc/source/detail?r=3747

r3747 timed-out on a tsan test. Verified that it passes
the test and reduced the execution time of that test (r3782).

TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1292006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3807 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 21:21:32 +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
marpan@webrtc.org
6ff76c7404 Reduce execution time of rate control test. 
TBR=mikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1289005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3782 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-08 20:32:48 +00:00
pbos@webrtc.org
034f004a4f WebRtc_Word32 => int32_t in video_coding/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1203008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3778 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-08 11:13:29 +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
henrike@webrtc.org
cfc07c943f Revert of r3747. 
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1277005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3752 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-02 14:55:44 +00:00
fbarchard@google.com
747c4cc96e For VGA (640x360), currently 1 thread is used. This change increases it to 2 threads. For HD, 4 threads are enabled. 
BUG=none
TEST=run a hangout and screencast high framerate, high resolution windows of youtube.  Observe that 1 cpu is insufficient to maintain high framerate with complex content.
Review URL: https://webrtc-codereview.appspot.com/1203006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3747 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-01 22:16:45 +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
pbos@webrtc.org
8685090060 Account for header inside I420Encoder::InitEncode. 
Also verify that the header is part of the received payload inside
I420Decoder::Decode.

BUG=

Review URL: https://webrtc-codereview.appspot.com/1211005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3690 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-19 11:39:03 +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
pbos@webrtc.org
8911ce46a4 Generic video-codec support. 
Labels frames as key/delta, also marks the first RTP packet of a frame as such,
to allow proper reconstruction even if packets are received out of order.

BUG=1442
TBR=ajm@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1207004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3680 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-18 16:39:03 +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
dwkang@webrtc.org
7473f89f63 Revert "Internal clean up: removing unused include line." 
(reverting https://webrtc-codereview.appspot.com/1177004)

BUG=none

Review URL: https://webrtc-codereview.appspot.com/1181005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3655 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-12 01:43:00 +00:00
dwkang@webrtc.org
25316b2a09 Internal clean up: removing unused include line. 
BUG=none
TESTED=passed try server

Review URL: https://webrtc-codereview.appspot.com/1177004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3654 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-12 01:10:02 +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
7c16c3c4a1 Move video_coding OWNERS to video_coding/. 
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1171004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3628 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-07 13:11:32 +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