Commit Graph

3424 Commits

Author SHA1 Message Date
turaj@webrtc.org
6388c3e2fd Implement initial delay. This CL allows clients of VoE to set an initial delay. Playout of audio is delayed and the extra playout delay is maintained during the call. While packets are buffered (in NetEq) to acheive the desired delay. ACM will playout silence (zeros). Initial delay has to be set before any packet is pushed into ACM.
TEST=ACM unit test is added, also a manual integration test is writen. 
Review URL: https://webrtc-codereview.appspot.com/1097009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3506 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-12 21:42:18 +00:00
andrew@webrtc.org
e6e344a7dc Sync libvpx and its gyp wrapper from Chromium.
TBR=kjellander
BUG=webrtc:1213

Review URL: https://webrtc-codereview.appspot.com/1096007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3505 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-12 19:35:18 +00:00
fbarchard@google.com
0ee57c2436 Increase maximum resolution to 4k x 3k.
BUG=1375
TESTED=try bots
Review URL: https://webrtc-codereview.appspot.com/1097008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3503 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-12 04:57:56 +00:00
mikhal@webrtc.org
57a0049e25 VCM: Removing frame drop enable from Reset call
BUG = 1387

Review URL: https://webrtc-codereview.appspot.com/1097010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3500 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-11 21:23:23 +00:00
kjellander@webrtc.org
18a21a03c6 Android NDK build tools
This CL enables building with Android NDK in the way that Chromium buildbots do it.

== Overview ==
* Add Android dependencies to DEPS (SDK, NDK, Android test runner). This also makes it possible to use Android's build/android/run_tests.py script to execute tests on Android devices.
* Add a Python script to build the WebRTC Video demo for Android using ndk-build and Ant. This is designed as an annotation script for Buildbots but is also fine to run locally.
* Update Android.mk so it works with the compiler output from a build performed by build/android/buildbot/bb_run_bot.py (which is how Chrome buildbots build).

== Syncing Android dependencies ==
To get the dependencies added in DEPS synced out, you must change the last line
of your .gclient file to look like this:
];target_os = ["android"]

That will append another variable to the .gclient file that causes these
dependencies to be synced during gclient sync.
If you want to get additional platform-specific dependencies in the same
checkout, add them to the list too, e.g. target_os = ["android", "unix"].

== Android.mk ==
The fix in Android.mk is needed since Chrome is building using build/android/buildbot/bb_run_bot.py, which only output the libraries into out/Debug. With the change it works for both that and a normal build (which copies the library files from out/Debug/obj.target/subpath to out/Debug anyway as a part of the build).

== svn:ignore ==
NOTICE: Before submitting, the following directories should be added to svn:ignore in third_party to avoid them from being removed and re-synced for every build:
* android_testrunner
* android_tools
* WebKit
This has to be done in a manual SVN commit since it's not possible to include in a git-svn CL (and I don't want to migrate this to a SVN CL).

BUG=none
TEST=local builds

Review URL: https://webrtc-codereview.appspot.com/1024009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3497 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-11 17:43:19 +00:00
kjellander@webrtc.org
00ab7cf4fd Fix perf output for audioproc and iSAC fixed-point tests
The measurement and trace entries had been mixed up in the calls to webrtc::test::PrintResult, resulting in the plotted graphs were named after the metric. The parameter names are quite confusing which probably led to this.

BUG=none
TEST=none

Review URL: https://webrtc-codereview.appspot.com/1093007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3496 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-11 12:33:03 +00:00
stefan@webrtc.org
0cb48a0a18 Set SingleStream BWE in unittests.
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/1094004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3494 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-11 08:30:23 +00:00
stefan@webrtc.org
63066f7200 Set qpMax to 56 in for all VP8 tests. Fixes buildbot breakage.
TBR=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1098010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3493 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-11 08:27:33 +00:00
mikhal@webrtc.org
3d305c64b4 Updates to send side streaming mode:
1. Disabling frame-droppers from the vie encoder and not the channel.
2. Accounting for qpMax in the VP8 wrapper.

Review URL: https://webrtc-codereview.appspot.com/1101007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3492 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-10 18:42:55 +00:00
tnakamura@webrtc.org
79481474ad Update version number to 3.23
TBR=niklas.enbom
Review URL: https://webrtc-codereview.appspot.com/1105004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3489 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-08 19:56:09 +00:00
kjellander@webrtc.org
7c850745d3 Adding third_party/directx and winsdk_samples to svn:ignore
This avoids them getting wiped during sync for every build, which saves
build time on Windows.

I also removed no longer present google-visualization-python dir.

BUG=none
TEST=none



git-svn-id: http://webrtc.googlecode.com/svn/trunk@3488 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-07 16:12:25 +00:00
kjellander@webrtc.org
687efe3ba8 Adding third_party/opus to svn:ignore
This avoids the directory getting wiped before it's synced out again
for every build.

BUG=none
TEST=none



git-svn-id: http://webrtc.googlecode.com/svn/trunk@3486 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-07 11:16:09 +00:00
henrikg@webrtc.org
b64732abfc Fix Win64 build breakage
This is for landing https://webrtc-codereview.appspot.com/1096006/ by Justin Schuh.

Stable will be updated after this has landed.

Review URL: https://webrtc-codereview.appspot.com/1091008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3484 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-07 10:14:05 +00:00
phoglund@webrtc.org
147c73ea60 Made it possible to render custom call output to file.
This is to enable quality tests using the custom call.

BUG=
TESTED=locally

Review URL: https://webrtc-codereview.appspot.com/1093005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3483 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-07 08:52:08 +00:00
kma@webrtc.org
d83b9fdf45 Fixed a bug in iSAC transform functions on ARM-Neon platform. Performance unchanged.
Bugs=none
Test=trybots, and file bit-exact tests; passed.

Description of the bug: Neon registers q4-q7 not saved before calling the outside FFT routines in the assembly functions.
Review URL: https://webrtc-codereview.appspot.com/1097006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3480 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-06 23:53:13 +00:00
mflodman@webrtc.org
4fd5527ab1 Don't report an error for GetEstimatedReceiveBandwidth if there is no valid
estimate.

BUG=1377

Review URL: https://webrtc-codereview.appspot.com/1095005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3479 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-06 17:46:39 +00:00
kjellander@webrtc.org
fe3d606f15 Enable indefinitely running vie_auto_test option
When doing test automation, the prompt in vie_auto_test is not working as expected on Windows when the test is run from a Buildbot. As soon a prompt is presented to the test runner, vie_auto_test exits, assuming the user pressed Ctrl-D.

By adding a third option for the Stop/Modify call prompt that allows running the call indefinitely (and making that the default), no prompt is displayed when the --auto_custom_call flag is used.

BUG=none
TEST=Execution with vie_auto_test.exe --auto_custom_call --override "Enter destination IP.=192.168.3.11" and by running vie_auto_test in interactive mode.
+ Trybots passing.

Review URL: https://webrtc-codereview.appspot.com/1099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3478 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-06 09:36:37 +00:00
andrew@webrtc.org
1e7ed7afe9 Use LOG_F interface for unsupported functions.
This will provide the function name in the log.

BUG=b/8115521
TESTED=enabled ANDROID_NOT_SUPPORTED on Linux and observed log lines as expected

Review URL: https://webrtc-codereview.appspot.com/1096005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3474 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-05 21:23:39 +00:00
kma@webrtc.org
959da8d286 Added labels in transform_neon.S in iSAC-fix, so the tables be shared with other files in iOS build. Also, moved several code lines in the same file, in case register values cannot be preserved after a function call which could cause a crash in some platforms (e.g. iOS etc.).
Bugs: none
Review URL: https://webrtc-codereview.appspot.com/1072007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3473 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-05 20:46:55 +00:00
phoglund@webrtc.org
a7303bdfb5 Lint-cleaned video and audio receivers.
BUG=
TESTED=trybots

Review URL: https://webrtc-codereview.appspot.com/1093004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3471 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-05 15:12:39 +00:00
elham@webrtc.org
c4e45f67c0 Updated version number to 3.22
Review URL: https://webrtc-codereview.appspot.com/1096004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3469 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-04 21:34:05 +00:00
tina.legrand@webrtc.org
23e3559507 Updating Perf numbers for Win Large Test.
Due to a bug in the RTP module, which appeared during packet loss, we have had too short delay in the Win Large Test. When the bug was fixed we had a regression error that should be fixed with this update.

Review URL: https://webrtc-codereview.appspot.com/1091005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3466 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-04 13:25:11 +00:00
phoglund@webrtc.org
244251a9cd Moved almost all payload-related stuff to the payload registry.
The big benefit is we no longer have a circular dependency between the media receiver and the payload registry. The payload registry is starting to take a bit more place on the stage, and now knows how to do different things depending on audio or video.

BUG=
TESTED=rtp_rtcp_unittests, vie_auto_test, voe_auto_test

Review URL: https://webrtc-codereview.appspot.com/1078004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3465 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-04 13:23:07 +00:00
kjellander@webrtc.org
fa53d8717c Fixing/disabling Windows x64 warnings
Disabled MSVC #4267 warnings in common.gypi to enable x64 builds
for Windows.
Fixed MSVC #4267 warnings in test/testsupport.
Added third_party/directxsdk to .gitignore.

With http://review.webrtc.org/1070008 landed, this should make it possible
to build for x64 on Windows.

BUG=1348
TEST=Compiling with http://review.webrtc.org/1070008 applied:
set GYP_DEFINES="target_arch=x64"
set GYP_GENERATORS=ninja
gclient sync
ninja -C out\Debug_x64

Review URL: https://webrtc-codereview.appspot.com/1060008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3464 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-04 10:07:17 +00:00
braveyao@webrtc.org
254d85af54 Exchange TRY by enumerating image formats in Linux video capture
ISSUE = issue 529
TEST  = unittest on Linux
Review URL: https://webrtc-codereview.appspot.com/1066011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3463 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-04 07:53:53 +00:00
andrew@webrtc.org
6ed8ebcef9 Fix MaxChannels test; 32 -> 100.
TBR=henrika

Review URL: https://webrtc-codereview.appspot.com/1060010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3460 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-02 00:05:58 +00:00
andrew@webrtc.org
4a6f62d4dc Remove (in practice) the voice engine channel limit.
There's really no reason for this limit. I've bumped it to a
practically unreachable ceiling, with a TODO for removing it
entirely.

TBR=henrika
BUG=b/8122300

Review URL: https://webrtc-codereview.appspot.com/1070014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3459 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-01 23:42:44 +00:00
mikhal@webrtc.org
dbe97d2550 Adding a send side API for streaming
Review URL: https://webrtc-codereview.appspot.com/1070009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3457 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-01 19:33:21 +00:00
stefan@webrtc.org
becf9c897c Fix mismatch between different NACK list lengths and packet buffers.
This is a second version of http://review.webrtc.org/1065006/ which passes the parameters via methods instead of via constructors.

BUG=1289

Review URL: https://webrtc-codereview.appspot.com/1065007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3456 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-01 15:09:57 +00:00
stefan@webrtc.org
b586507986 Break out RemoteBitrateEstimator from RtpRtcp module and make RemoteBitrateEstimator::Process trigger new REMB messages.
Also make sure RTT is computed independently of whether it's time to send RTCP messages or not.

BUG=1298

Review URL: https://webrtc-codereview.appspot.com/1060005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3455 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-01 14:33:42 +00:00
tina.legrand@webrtc.org
46d90dcd74 Adding three frame sizes to Opus
Adding support for 10, 40 and 60 ms packet sizes for Opus.

BUG=issue1015

Review URL: https://webrtc-codereview.appspot.com/1086004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3454 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-01 14:20:06 +00:00
phoglund@webrtc.org
d087789b9c Adjusted net_50_5_plr_5 on Linux, removed all gilbert_elliot metrics (too flaky), added mac expectations.
BUG=

Review URL: https://webrtc-codereview.appspot.com/1075006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3453 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-01 13:28:58 +00:00
henrik.lundin@webrtc.org
aaad6134b9 Implementing stereo support for G.722
This CL implements stereo support for G.722 through a new class
AudioDecoderG722Stereo derived from AudioDecoderG722.

Also implementing tests for G.722 stereo.

Review URL: https://webrtc-codereview.appspot.com/1073006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3452 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-01 11:49:28 +00:00
braveyao@webrtc.org
7050f96bff Set frame length for frame converting in external renderer
ISSUE = Issue 1342
TEST  = Manual Test
Review URL: https://webrtc-codereview.appspot.com/1083005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3451 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-01 02:49:29 +00:00
bjornv@webrtc.org
ac46c6dac3 Replaced relative path to reference from <(webrtc_root).
Changed to proper include paths in AECM and NSX.
Tested on trybots.

BUG=None

Review URL: https://webrtc-codereview.appspot.com/1063014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3450 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-31 21:06:16 +00:00
turaj@webrtc.org
9d532fd275 Fix propagating RED paylaod-type to ACM.
BUG=issue1322
TBR=henrika@google.com
Review URL: https://webrtc-codereview.appspot.com/1086005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3449 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-31 18:34:19 +00:00
turaj@webrtc.org
763faeab4e Removing a codec from NetEq database has a bug. |funcDurationEst| is not updated.
This is discovered during a test for controlling delay. It is not simple to reproduce it. 

Bug=
test=manual test verified that |functionDurationEst| is correctly updated.
Review URL: https://webrtc-codereview.appspot.com/1074013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3448 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-31 18:21:58 +00:00
turaj@webrtc.org
c0ada864b2 fix for issue 281.
A reverse copy is removed. The index to src buffer could be -1, this happens very often. The reverse copy is not needed as the content of the destination is overwritten further down in "WebRtcIlbcfix_CbConstruct()" 


Bug=issue281
TEST=manual test over 1600 files TIMIT database, all outputs are bit-exact with the ones generated from head revision. Local run of asan does not generate any warning.
 
Review URL: https://webrtc-codereview.appspot.com/1063013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3447 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-31 18:21:06 +00:00
turaj@webrtc.org
8c8ad85c5d fix issue 1322, accept -1 as default payload-type for redundant coding (FEC).
issue=1322 
test=trybot, voe auto-tes
Review URL: https://webrtc-codereview.appspot.com/1043007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3446 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-31 18:20:17 +00:00
mikhal@webrtc.org
119c67df36 Adding a max jitter filter to the JB estimate - allowing two modes, one will return the last estimate (current setting), and another will return the max value seen, and allow setting an initial value.
This cl also includes tests and some clean up.

Review URL: https://webrtc-codereview.appspot.com/1019007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3445 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-31 17:18:02 +00:00
mikhal@webrtc.org
e07c661a29 VP8: Making key frame interval a tunnable parameter
Review URL: https://webrtc-codereview.appspot.com/1070006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3444 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-31 16:37:13 +00:00
henrik.lundin@webrtc.org
6e3968f62a Fix NetEq4 unit tests for VS2012
This merges the changes from r3199.

Review URL: https://webrtc-codereview.appspot.com/1078010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3443 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-31 15:07:30 +00:00
henrik.lundin@webrtc.org
73deaadd0e Removing a hack for CNG
However, two other "hacks" had to be added to maintain bit-exactness
with legacy.

Note that this change requires a new version of the universal.rtp test
input, although the output reference stays the same.

Moving reference files, and using a new input vector for NetEq4.
The new input vector neteq_universal_new.rtp is identical to the old
neteq_universal.rtp, except that the payload type for CNG packets that
follows a wideband codec is changed to 98.

Update to resources revision 15 where the new reference files are.

Also changing a faulty log error.

Review URL: https://webrtc-codereview.appspot.com/1078009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3442 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-31 13:32:51 +00:00
phoglund@webrtc.org
96a08cef68 Fixed stale regression values and calibrated some vie_auto_test values.
BUG=

Review URL: https://webrtc-codereview.appspot.com/1066010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3441 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-31 10:06:44 +00:00
henrik.lundin@webrtc.org
ac59dba3f7 Adding iSAC-fb support
Adding tests, too.

Review URL: https://webrtc-codereview.appspot.com/1070011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3440 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-31 09:55:24 +00:00
kjellander@webrtc.org
3d13d9f0a0 Fix audio_e2e_test command line arguments
The changes in r1068 that moved over to the webrtc/test/buildbot_tests.py launch script was not properly tested on the real machine for the audio_e2e_test. Due to that it contained a few syntax errors and paths that were not resolved as expected. This CL fixes this and has been tested more thorougly.

BUG=none
TEST=Ran, standing in the checkout dir:
out/Release/buildbot_tests.py -t audio_e2e_test
with successful result.

Review URL: https://webrtc-codereview.appspot.com/1070012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3438 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-30 21:39:10 +00:00
andrew@webrtc.org
73a702c979 This is a change in the iOS audio device to use VoiceProcessingIO API instead of RemoteIO. This way we don't need to use WebRTC EC and NS because it happens on the device hardware.
Review URL: https://webrtc-codereview.appspot.com/1061007
Patch from Gil Osher <gil.osher@vonage.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3437 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-30 21:18:31 +00:00
bjornv@webrtc.org
7ded92b71e Re-committing r3428
TBR=ajm
BUG=None

Review URL: https://webrtc-codereview.appspot.com/1066008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3436 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-30 16:16:59 +00:00
henrik.lundin@webrtc.org
51f11eb5ae Fixing problems in audio_decoder_unittests
The tests did not work in Release mode because of the asserts.

Review URL: https://webrtc-codereview.appspot.com/1062010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3435 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-30 13:00:33 +00:00
henrik.lundin@webrtc.org
ddf981c789 Disable iSAC fix test in audio_decoder_unittests
The test AudioDecoderIsacFixTest.EncodeDecode was disabled since it
triggers a valgrind warning. The issue is tracked in
https://code.google.com/p/webrtc/issues/detail?id=1353

BUG=1353

Review URL: https://webrtc-codereview.appspot.com/1084004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3434 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-30 12:29:48 +00:00