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76 Commits

Author SHA1 Message Date
henrik.lundin@webrtc.org
859626570a VP8 RTP work 
Fixing the plumbing to get the KEYIDX between VP8 wrapper and
rtp_rtcp module. Also fixing a missing pipe for temporalIdx

Review URL: http://webrtc-codereview.appspot.com/295004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1024 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-25 10:17:00 +00:00
kjellander@webrtc.org
543c3eaa46 Fixing Release compilation errors 
Review URL: http://webrtc-codereview.appspot.com/267026

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1000 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-23 12:20:35 +00:00
mikhal@webrtc.org
2b838b4121 video_coding: updating the session info unit test following recent changes 
Review URL: http://webrtc-codereview.appspot.com/290002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@996 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-23 00:20:19 +00:00
mikhal@webrtc.org
425b377973 video_coding: Updating internal_defines to resolve latest build error. Refers to JB flush update. 
Review URL: http://webrtc-codereview.appspot.com/289001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@995 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-22 23:41:29 +00:00
mikhal@webrtc.org
f13388f134 video_coding: Requesting a key frame after a JB flush 
Review URL: http://webrtc-codereview.appspot.com/280006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@994 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-22 22:57:51 +00:00
mikhal@webrtc.org
6b9a7f8704 video_coding: Allowing for a decodable state independent of selective nacking 
Review URL: http://webrtc-codereview.appspot.com/263001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@993 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-22 22:48:20 +00:00
phoglund@webrtc.org
1144ba2268 Base and codec tests now run verify output and render to file instead of to screen. 
Rewrote the codec test to render to file and do video comparisons.

Refactored the coded tests somewhat. I still need to figure out how to do comparison in the automated case.

Added video analysis to the test. This will make sure that the system output roughly the right thing.

Moved the video metrics library into the test_support library. Made the metrics library available in the automated tests.

Made sure no one passes in too large YUV videos into the autotest.

The standard test's output now gets captured for both the left and right windows.

Wrote a rendering device which just writes the raw frames to file, for analysis. Updated the base standard test to dump its left window output to file. We don't do anything with it yet though.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/249001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@931 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-11 09:01:03 +00:00
mflodman@webrtc.org
637a59e68e jitter buffer update: waiting for key frame when Nack is enabled and continuity cannot be determined. 
Review URL: http://webrtc-codereview.appspot.com/266010
Patch from mikhals <mikhal@webrtc.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@924 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-10 12:50:48 +00:00
kjellander@webrtc.org
ad79d6f164 Removing exceptions since not used 
Review URL: http://webrtc-codereview.appspot.com/267002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@910 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-09 16:11:14 +00:00
kjellander@webrtc.org
1a8d08ad76 Changing usage of gtest_main target, to use test_support_main instead. 
Review URL: http://webrtc-codereview.appspot.com/252002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@884 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-03 23:28:47 +00:00
mikhal@webrtc.org
e203de7ba2 jitter_buffer updates: 
1. Determining continuity based on pictureId and not seq. numbers when available.
2. Hybrid bug fix: Don't set to decodable when the nack list is empty.
Review URL: http://webrtc-codereview.appspot.com/255001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@878 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-03 00:42:52 +00:00
henrik.lundin@webrtc.org
679cb07980 Fix build error for release build 
Review URL: http://webrtc-codereview.appspot.com/252003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@874 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-02 19:52:27 +00:00
henrik.lundin@webrtc.org
baf6db5ead Making dual decoder work again in VCM 
Changing the assignment operator in VCMJitterBuffer to a named
function (CopyFrom) instead, since it is not a straight
assignment. Also fixing two bugs in the jitter copy function.

Bug fix in VCMEncodedFrame: The copy constructor did not
copy _length.

In VCM codec database, make sure that the callback object is
preserved when copying back from secondary to primary decoder.

In VP8 wrapper, adding code to copy the _decodedImage to the
Copy() method.

Bugfix in video_coding_test rtp_player:
The retransmissions where made in reverse order. Now new items are
appended to the end of the LostPackets list, which makes the order
correct when retransmitting.

Handling the case when cloning an unused decoder state:
When the decoder has not successfully decoded a frame yet,
it cannot be cloned. A NULL pointer will be returned all
the way out to VideoCodingModuleImpl::Decode(). When this
happens, the VCM will call Reset() for the dual receiver,
in order to reset the state to kPassive.

Review URL: http://webrtc-codereview.appspot.com/239010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@873 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-02 18:58:39 +00:00
stefan@webrtc.org
f960211f8b Fixes two jitter buffer bugs related to NACK. 
Avoid decoding delta frames after a Flush() and after requesting
a key frame due to full NACK list.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/247011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@837 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-28 16:00:49 +00:00
stefan@webrtc.org
eb65860720 Reverts the workaround in r823 and solves a macro bug. 
The macro bug caused frames to be dropped after being grabbed
for decoding.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/248004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@831 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-28 12:25:34 +00:00
mikhal@webrtc.org
5200a05500 video_coding/jitter_buffer Updating condition on which we return a frame. 
Review URL: http://webrtc-codereview.appspot.com/240011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@825 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-27 16:54:54 +00:00
stefan@webrtc.org
2d28aff785 Workaround for an issue where frames are grabbed for decoding prematurely. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/240013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@823 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-27 16:13:18 +00:00
stefan@webrtc.org
d855c1a4e8 Reverts r807 and fixes the real issue in the VCM. 
This fixes an issue in the VCM where we don't wait for a packet to arrive
if the jitter buffer is empty. This also fixes an issue where an old
packet can trigger a packet event signal for a future frame.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/248001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@814 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-25 11:52:48 +00:00
wjia@webrtc.org
f0cd394a2e Put fwrite calls under corresponding macros since they shouldn't show up in release build. 
This also make chromeos build happy.
BUG=none
TEST=compile
Review URL: http://webrtc-codereview.appspot.com/247006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@808 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-25 00:40:43 +00:00
mikhal@webrtc.org
f31826e17b adding a wait on the decode thread when no frames are available 
Review URL: http://webrtc-codereview.appspot.com/246009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@807 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-25 00:20:54 +00:00
stefan@webrtc.org
ead87b5051 Fix potential issue where frame buffers might be freed while being decoded. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/243004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@791 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-21 06:46:37 +00:00
mikhal@webrtc.org
ee3dfa6f43 Review URL: http://webrtc-codereview.appspot.com/241007 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@789 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-21 00:46:09 +00:00
kma@webrtc.org
d75889e2eb Change of Android makefiles to build latest video coding code. 
Review URL: http://webrtc-codereview.appspot.com/239008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@786 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-20 16:28:56 +00:00
stefan@webrtc.org
ffd28f95c5 Request key frames to battle error propagation. 
The VP8 decoder wrapper will request key frames 30 frames after seeing
a packet loss, if it hasn't received a state refresh (only possible
through key frames in this version).

For this to be possible the jitter buffer has been made aware of
picture ids to be able to detect frame losses. Legacy JB code to
handle streams without marker bits was also removed since it
conflicts with streams with FEC.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/239002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@774 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-19 15:55:39 +00:00
mikhal@webrtc.org
d0752c370d video_coding: Update to hybrid mode: Set FEC values for zero below a threshold. 
Review URL: http://webrtc-codereview.appspot.com/245001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@773 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-19 15:48:30 +00:00
pwestin@webrtc.org
1da1ce0da5 First implementation of simulcast, adds VP8 simulcast to video engine. 
Changed API to RTP module
Expanded Auto test with a test for simulcast
Made the video codec tests compile
Added the vp8_simulcast files to this cl
Added missing auto test file
Review URL: http://webrtc-codereview.appspot.com/188001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@736 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-13 15:19:55 +00:00
stefan@webrtc.org
4c059d87b3 Add metric for number of packets discarded by JB due to not being decodable 
Also fixes a couple of bugs related to sequence number wrap found while
testing.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/218001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@732 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-13 07:35:37 +00:00
stefan@webrtc.org
791eec7424 Add API to get the number of packets discarded by the video jitter buffer due to being too late. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/200001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@723 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-11 07:53:43 +00:00
stefan@webrtc.org
06887aebae Fixes two bugs when decoding with packet losses. 
Disable _missingFrame bit since we can't set it correctly with FEC.

No longer return more than one decoded frame per Decode() call.
This is a work-around for a bug where the frame info map was popped more often than items were added to the map.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/215001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@722 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-10 14:17:46 +00:00
stefan@webrtc.org
4b6f747373 Fixes a newly introduced bug in the jitter buffer where buffer reallocation 
causes corrupt pointers.
Review URL: http://webrtc-codereview.appspot.com/186003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@688 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-04 06:58:39 +00:00
stefan@webrtc.org
93d216c23f Fixed bug in jitter buffer which caused the missingFrames bit to never be set. 
Also updated the VP8 wrapper to return fully concealed frames (for rendering).

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/190003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@687 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-04 06:48:11 +00:00
mikhal@webrtc.org
ae7a0522c5 video_coding robustness: Updating hybrid mode's settings 
1. Disabling adjustment factor - temporary update. 
2. Enabling a windowed filtered loss for the hybrid mode.  
Review URL: http://webrtc-codereview.appspot.com/192003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@684 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-03 22:54:34 +00:00
marpan@google.com
f1f3fb33b5 Update to rate-mismatch factor in media_opt_util. 
Review URL: http://webrtc-codereview.appspot.com/193003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@678 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-03 19:09:45 +00:00
stefan@webrtc.org
5b91464edf Allow an aggregated partition to spill over to a new packet. 
Adds support for the case where the partition 0 and parts of partition 1
are transmitted in packet 1, and the end of partition 2 is transmitted
in packet 2.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/181003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@675 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-03 10:26:12 +00:00
mikhal@webrtc.org
e185e9f68a video_coding: updates to jitter buffer logic: Make sure that every frame is inserted only once to the list. 
Review URL: http://webrtc-codereview.appspot.com/165001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@648 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-23 22:02:40 +00:00
mikhal@webrtc.org
105ff39dec video coding: updating offline tests. 
Additional clean-up to the offline test: Placing test callbacks in a designated file. 
Review URL: http://webrtc-codereview.appspot.com/167002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@642 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-23 16:41:11 +00:00
marpan@google.com
45fa141f0a qm_select: changed default settings for uep. 
Review URL: http://webrtc-codereview.appspot.com/132015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@584 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-12 16:53:19 +00:00
kjellander@webrtc.org
f0a8464b74 Added more statistics during SSIM/PSNR calculation, including calculation of min/max value. 
Moved video_metrics.h into a GYP library so it can be used from other projects.

Review URL: http://webrtc-codereview.appspot.com/132013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@582 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-12 13:45:39 +00:00
xians@google.com
d3185fe219 refactor the gyp file to gypi file. 
Basically, the gypi file is a copy of gyp file, but has some difference on the
path of the dependencies.
Review URL: http://webrtc-codereview.appspot.com/137020

git-svn-id: http://webrtc.googlecode.com/svn/trunk@581 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-12 12:24:39 +00:00
marpan@google.com
30ecda146a media_opt_util: Added comment and lowered window size parameter. 
Review URL: http://webrtc-codereview.appspot.com/135018

git-svn-id: http://webrtc.googlecode.com/svn/trunk@575 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-09 17:15:12 +00:00
marpan@google.com
3f28061f3a media_opt_util: Modification to correction factor in FEC overhead. 
Review URL: http://webrtc-codereview.appspot.com/133019

git-svn-id: http://webrtc.googlecode.com/svn/trunk@573 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-09 16:39:40 +00:00
mikhal@webrtc.org
6f54c20703 video coding test: Adding MT functionality 
Review URL: http://webrtc-codereview.appspot.com/135008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@570 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-09 14:38:59 +00:00
stefan@webrtc.org
c3d891059e Adds support for VP8 partitions 
This change adds support for VP8 partitions in the video jitter buffer and 
the VP8 encoder and decoder wrappers. The feature is currently disabled by
default since it requires a later version of libvpx.

With this change the jitter buffer will also start keeping track of each
packet header until decoding, and the VCMSessionInfo and VCMPacket objects 
will keep pointers into the encoded frame buffers.
Review URL: http://webrtc-codereview.appspot.com/137021

git-svn-id: http://webrtc.googlecode.com/svn/trunk@558 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-08 06:50:28 +00:00
andrew@webrtc.org
413b993166 Put some table size information in one place. 
Motivated by fixing an unused variable warning in release mode.
Review URL: http://webrtc-codereview.appspot.com/132007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@523 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-02 22:03:56 +00:00
marpan@google.com
243db12616 media_opt_util: Fixed an assert and some code cleanup for AvgRecoveryFEC function. 
Review URL: http://webrtc-codereview.appspot.com/139007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@502 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-31 22:14:52 +00:00
henrik.lundin@webrtc.org
8571af7be6 Updating to new VP8 rtp format 
The VP8 packetizer and tests have been updated to the new
RTP draft (http://tools.ietf.org/html/draft-ietf-payload-vp8-01).
The receive-side parser is also updated, and a new unit test
is implemented for it. Finally, some data traversing work to
get the parsed information into the decoder.
Review URL: http://webrtc-codereview.appspot.com/116011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@482 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-29 15:37:12 +00:00
mikhal@webrtc.org
a057a9561c video_coding: Updating protection logic in media optimization utility: 
1. Changing protection logic structure: Accepts only one method (not a list)
2. Removed unused code (unreferenced protection methods)
3. Removed inline constructors/destructors.  
Review URL: http://webrtc-codereview.appspot.com/120005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@467 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-26 21:17:34 +00:00
mikhal@webrtc.org
552f173979 video_coding: Moving video metrics computation to a designated file. 
This is the first stage of a general clean-up to test_util. Will try to divide this clean-up to small changes, so it will be easier to review. 
Review URL: http://webrtc-codereview.appspot.com/120004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@466 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-26 17:38:09 +00:00
stefan@webrtc.org
49cbc512ae Fix unused variable warning in video_coding. 
Issue 57: [Patch] Fix unused variable warnings in the video_coding module
Review URL: http://webrtc-codereview.appspot.com/126004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@435 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-24 08:51:08 +00:00
mikhal@webrtc.org
06ad81fd58 video_coding: changing the UpdateMethod function (protection settings). 
Review URL: http://webrtc-codereview.appspot.com/126002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@423 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-23 16:03:59 +00:00