Commit Graph

9525 Commits

Author SHA1 Message Date
ed4224fbda Android GlRectDrawer: Add fragment shader for RGB(A) textures
Add third shader type for RGB(A) and refactor according to the Rule of three.

BUG=webrtc:4742
R=hbos@webrtc.org

Review URL: https://codereview.webrtc.org/1311093005 .

Cr-Commit-Position: refs/heads/master@{#9843}
2015-09-02 10:52:10 +00:00
d4563f4dde Revert of Excluding two troublesome trybots from the CQ config. (patchset #1 id:1 of https://codereview.webrtc.org/1310953006/ )
Reason for revert:
Both bots are now back online. Let's try enabling them again.

Original issue's description:
> Excluding two troublesome trybots from the CQ config.
>
> Temporary measurement to allow landing of a queue of CLs.
>
> BUG=524352, 526625
> R=phoglund@webrtc.org
>
> Committed: fa7cb8eaa7

TBR=phoglund@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=524352, 526625

Review URL: https://codereview.webrtc.org/1305983005

Cr-Commit-Position: refs/heads/master@{#9842}
2015-09-02 08:28:45 +00:00
71cfe690b7 For TestResolverShutdown, use address that can't be resolved.
This test only currently works because stun.l.google.com has an IPv4
address and the TURN port is created with an IPv6 address. But the test
would start failing if/when it starts providing an IPv6 address. Which
may already be happening, as indicated by a recent test failure.

Review URL: https://codereview.webrtc.org/1290233003

Cr-Commit-Position: refs/heads/master@{#9841}
2015-09-02 02:01:40 +00:00
e63d2a1c62 Add JNI/java wrapper for the file rotating logging class.
BUG=
R=glaznev@webrtc.org

Review URL: https://codereview.webrtc.org/1309073004 .

Cr-Commit-Position: refs/heads/master@{#9840}
2015-09-01 23:11:28 +00:00
abd0d1a3f7 Handle all RTCICEConnectionState values in ARDVideoCallViewController
BUG=

Review URL: https://codereview.webrtc.org/1318343005

Cr-Commit-Position: refs/heads/master@{#9839}
2015-09-01 22:35:42 +00:00
4d2f4d1c69 - Make shared EGL context used for HW video decoding member
of decoder factory class.
- Add new Peer connection factory method to initialize shared
EGL context.

This provides an option to use single peer connection factory
in the application and create peer connections from the same
factory and reinitialize shared EGL context for video
decoding HW acceleration.

R=wzh@webrtc.org

Review URL: https://codereview.webrtc.org/1304063011 .

Cr-Commit-Position: refs/heads/master@{#9838}
2015-09-01 22:04:21 +00:00
c36d4df250 Use committer list from chrome-infra-auth
R=kjellander@webrtc.org, kjellander@chromium.org
BUG=chromium:511311

Review URL: https://codereview.webrtc.org/1318023003 .

Patch from Sergiy Byelozyorov <sergiyb@chromium.org>.

Cr-Commit-Position: refs/heads/master@{#9837}
2015-09-01 19:51:17 +00:00
098c1de578 Fixes for PNaCl build of remoting client plugin with GN.
1. Added support for PNaCl in base.
2. Separated parts of desktop_capture module that need to be
   compiled for PNaCl.

BUG=512899
R=jiayl@chromium.org, jiayl@webrtc.org

Review URL: https://codereview.webrtc.org/1323453003 .

Cr-Commit-Position: refs/heads/master@{#9836}
2015-09-01 18:36:50 +00:00
97579a4e12 Add option to enable ECDSA key for Java API.
Review URL: https://codereview.webrtc.org/1312293003

Cr-Commit-Position: refs/heads/master@{#9835}
2015-09-01 18:31:34 +00:00
eebc0996bf Add magjed@ as owner for talk/app/webrtc/java/android/org/webrtc/
magjed@ has done a lot of work in this folder.

Review URL: https://codereview.webrtc.org/1322133002

Cr-Commit-Position: refs/heads/master@{#9834}
2015-09-01 18:09:28 +00:00
194cceadae Do not use HW H.264 encoder on Nexus 7.
H.264 HW encoder on some Nexus 7 models have
poor bitrate control.

R=jiayl@webrtc.org

Review URL: https://codereview.webrtc.org/1311893009 .

Cr-Commit-Position: refs/heads/master@{#9833}
2015-09-01 17:36:49 +00:00
4edc39c569 Set the IceConnectionReceivingTimeout as a RTCConfiguration parameter.
BUG= 4901

Review URL: https://codereview.webrtc.org/1315503003

Cr-Commit-Position: refs/heads/master@{#9832}
2015-09-01 16:54:02 +00:00
0f9af01456 Added send stream test case for VP9 header.
BUG=webrtc:4914

Review URL: https://codereview.webrtc.org/1288363002

Cr-Commit-Position: refs/heads/master@{#9831}
2015-09-01 14:01:54 +00:00
fa7cb8eaa7 Excluding two troublesome trybots from the CQ config.
Temporary measurement to allow landing of a queue of CLs.

BUG=524352, 526625
R=phoglund@webrtc.org

Review URL: https://codereview.webrtc.org/1310953006 .

Cr-Commit-Position: refs/heads/master@{#9830}
2015-09-01 13:58:14 +00:00
02d283a6ff Speculative revert of "- Move test cases for more natural ordering."
This reverts commit c20a5dc9305b988ca173cd63e606124b02e6d54c.

BUG=webrtc:4959
R=solenberg@webrtc.org
TBR=solenberg@webrtc.org

Review URL: https://codereview.webrtc.org/1309313008 .

Cr-Commit-Position: refs/heads/master@{#9829}
2015-09-01 13:48:31 +00:00
05f71fcb61 NetEq: Fixing a corner case with depleted sync buffer
In some cases, the number of samples (per channel) in NetEq's sync
buffer could fall below the allowed minimum (5 samples for narrowband,
scaling for other rates). If the number of samples extracted from the
buffer was smaller than the desired number, an error is
returned. However, if the decoder returns fewer samples than expected,
it could happen that the sync buffer level falls under the minimum,
but enough samples are extracted. This triggered an assert. With this
change, the minimum level of the sync buffer is always enforced.

A test is implemented to trigger the problem. It made the assert fire
without this fix, but it now passes.

BUG=webrtc:4840
R=minyue@webrtc.org

Review URL: https://codereview.webrtc.org/1324453002 .

Cr-Commit-Position: refs/heads/master@{#9828}
2015-09-01 09:52:06 +00:00
521875a9a4 Use RtcpPacket to send APP in RtcpSender
BUG=webrtc:2450
R=asapersson@webrtc.org

Review URL: https://codereview.webrtc.org/1311453002 .

Cr-Commit-Position: refs/heads/master@{#9827}
2015-09-01 08:11:36 +00:00
e551f12a41 Revert "Adding unittests to AudioConferenceMixer."
This reverts commit 22c2729607964aa38d6cb4e521994453b6a271c4.

TBR=henrik.lundin@webrtc.org,
BUG=

Review URL: https://codereview.webrtc.org/1326583002 .

Cr-Commit-Position: refs/heads/master@{#9826}
2015-09-01 07:54:49 +00:00
22c2729607 Adding unittests to AudioConferenceMixer.
Unit tests around AudioConferenceMixer was severely missing. This CL is to add some tests.

BUG=
R=ajm@chromium.org, andrew@webrtc.org

Review URL: https://codereview.webrtc.org/1257583011 .

Cr-Commit-Position: refs/heads/master@{#9825}
2015-09-01 07:33:37 +00:00
b7306ae6fe Revert "Avoiding size_t in MIPS."
This reverts commit 32e2f461b13c530d34f9c434e7e76da6ff3eda83.

BUG=526716
TBR=minyue@webrtc.org

Review URL: https://codereview.webrtc.org/1314873004 .

Cr-Commit-Position: refs/heads/master@{#9824}
2015-08-31 18:44:05 +00:00
32e2f461b1 Avoiding size_t in MIPS.
TBR=pkasting@chromium.org

BUG=526716

Review URL: https://codereview.webrtc.org/1310553008 .

Cr-Commit-Position: refs/heads/master@{#9823}
2015-08-31 15:40:17 +00:00
2c27430545 Print some output in long perf tests, to keep them alive
At least Android try bots seem to have a timeout that will forcibly shut
down the executable if no output has been observed for 60s. Since full
stack test typically run for 60s we need to output give some to avoid
racy shutdown.

BUG=chromium:513170
R=pbos@webrtc.org

Review URL: https://codereview.webrtc.org/1288453003 .

Cr-Commit-Position: refs/heads/master@{#9822}
2015-08-31 15:21:22 +00:00
0f4b3731c3 Stylizing AudioConferenceMixer.
Cleaning AudioConferenceMixer APIs to match Chromium style guide.

Main changes:
1. change all mutable references to pointers
2. add const to all non-mutable references
3. add const to as many methods as possible

BUG=
R=andrew@webrtc.org

Review URL: https://codereview.webrtc.org/1311733003 .

Cr-Commit-Position: refs/heads/master@{#9821}
2015-08-31 14:04:44 +00:00
ca28fdcf9f Use RtcpPacket to send XR (RTRR, DLRR, VOIP) in RtcpSender
BUG=webrtc:2450
R=asapersson@webrtc.org

Review URL: https://codereview.webrtc.org/1304123003 .

Cr-Commit-Position: refs/heads/master@{#9820}
2015-08-31 12:01:08 +00:00
c252dabbd6 CameraEnumerationAndroid: Make getSupportedFormats() an interface
Enumerating camera capabilities in the deprecated android.hardware.Camera interface is really slow because of the need to open and release the camera. By making getSupportedFormats() an interface, we allow apps the opportunity to inject their own implementation, such as storing the supported formats offline in the device's internal storage. It will also be possible to add an implementation of getSupportedFormats() using the new android.hardware.Camera2 interface in a follow-up CL.

R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1321903002 .

Cr-Commit-Position: refs/heads/master@{#9819}
2015-08-31 11:19:20 +00:00
c92c23d99a Roll chromium_revision f8d6ba9..a28d8d5 (337800:346100)
Relevant changes:
* src/buildtools: ecc8e25..565d04e
* src/testing/gmock: 2976396..0421b6f
* src/testing/gtest: 23574bf..9855a87
* src/third_party/android_tools: 21f4bcb..4238a28
* src/third_party/boringssl/src: de24aad..12fe1b2
* src/third_party/icu: c81a1a3..6b3ce81
* src/third_party/libjpeg_turbo: f4631b6..631e2dd
* src/third_party/libsrtp: 9c53f85..502e81a
* src/third_party/libvpx: aa9b5f1..a208eca
* src/third_party/libyuv: 6dde4f1..3c4f573
* src/third_party/openmax_dl: 22bb108..2eb98d8
* src/tools/grit: 1dac9ae..15d48e3
* src/tools/gyp: 5122240..6ee91ad
* src/tools/swarming_client: b39a448..2866a22
Details: f8d6ba9..a28d8d5/DEPS

Clang version changed 245402:245965
Details: f8d6ba9..a28d8d5/tools/clang/scripts/update.sh

BUG=None
R=glaznev@webrtc.org
TBR=glaznev@chromium.org, henrika@chromium.org

Review URL: https://codereview.webrtc.org/1308693010 .

Cr-Commit-Position: refs/heads/master@{#9818}
2015-08-31 09:30:29 +00:00
c20a5dc930 - Move test cases for more natural ordering.
- Get rid of the CoInitialize tests for WVoE/WViE.

BUG=webrtc:4690
R=pbos@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1319163002 .

Cr-Commit-Position: refs/heads/master@{#9817}
2015-08-31 09:14:05 +00:00
3c4ef29140 NetEq: Allow negative shift in BackgroundNoise::SaveParameters
This change allows a shift factor to be negative.  This is the way it
was in the old (NetEq3) code; see
4d363ae305/webrtc/modules/audio_coding/neteq/bgn_update.c,
lines 183-188.

Some input signals can lead to negative shifts, and would then trigger
an assert. The assert is now removed.

BUG=webrtc:4840
R=minyue@webrtc.org

Review URL: https://codereview.webrtc.org/1315833003 .

Cr-Commit-Position: refs/heads/master@{#9816}
2015-08-31 08:18:45 +00:00
3a14bf311f Replacing SSLIdentity* with scoped_refptr<RTCCertificate> in the cricket::TransportDescriptionFactory layers.
Updates TransportDescriptionFactory, calls and unittests.

BUG=webrtc:4927
R=tommi@webrtc.org, torbjorng@webrtc.org

Review URL: https://codereview.webrtc.org/1311903004 .

Cr-Commit-Position: refs/heads/master@{#9815}
2015-08-31 07:28:13 +00:00
a6cba3ab5c Java VideoRenderer.Callbacks: Make renderFrame() interface asynchronous
This CL makes the Java render interface asynchronous by requiring every call to renderFrame() to be followed by an explicit renderFrameDone() call. In JNI, this is implemented with cricket::VideoFrame::Copy() before calling renderFrame(), and a corresponding call to delete in renderFrameDone(). This CL is primarily done to prepare for a new renderer implementation.

BUG=webrtc:4742, webrtc:4909
R=glaznev@webrtc.org

Review URL: https://codereview.webrtc.org/1313563002 .

Cr-Commit-Position: refs/heads/master@{#9814}
2015-08-29 13:57:56 +00:00
1380e266ff Convert some more things to size_t.
These changes stem from requests by Andrew on https://codereview.webrtc.org/1228823002/ to eliminate some "return -1"s and change to using asserts plus returning size_ts.  I then also converted the relevant connected bits.

This also cleans up a bunch of style issues, e.g. no spaces around operators.

BUG=chromium:81439
TEST=none
R=andrew@webrtc.org, henrik.lundin@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://codereview.webrtc.org/1305983003 .

Cr-Commit-Position: refs/heads/master@{#9813}
2015-08-29 00:31:15 +00:00
e8386d2199 Added send-thresholding and fixed text packet dumping. Also a little squelch for the over-max-MTU log spam we see in there.
BUG=https://code.google.com/p/webrtc/issues/detail?id=4468
R=pthatcher@chromium.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1304063006 .

Cr-Commit-Position: refs/heads/master@{#9812}
2015-08-28 18:54:45 +00:00
79de90b110 Do not explicitly delete OpenGL shaders in VideoRendererGui.
This is handled by Android itself and may result in GL errors
when trying to release shaders when Activity is destroyed.

R=wzh@webrtc.org

Review URL: https://codereview.webrtc.org/1322703004 .

Cr-Commit-Position: refs/heads/master@{#9811}
2015-08-28 18:47:46 +00:00
f42376c601 Wire up currently-received video codec to stats.
BUG=webrtc:1844, webrtc:4808
R=mflodman@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1315413002

Cr-Commit-Position: refs/heads/master@{#9810}
2015-08-28 14:35:40 +00:00
6813ec84fb VideoCapturerAndroid: Move to android folder and split out camera enumeration into separate file
Pure code move of:
talk/app/webrtc/java/src/org/webrtc/VideoCapturerAndroid.java
into:
talk/app/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java
talk/app/webrtc/java/android/org/webrtc/CameraEnumerationAndroid.java

NOPRESUBMIT=true

Review URL: https://codereview.webrtc.org/1323453002

Cr-Commit-Position: refs/heads/master@{#9809}
2015-08-28 12:22:27 +00:00
9e69abf85e Added logging using the raw variant of the new aec logging macros
Replaced the wav file dumping functionality in aec_core.c with the newly added corresponding macros

Added macros for logging of AEC internal data

BUG=

Review URL: https://codereview.webrtc.org/1272403003

Cr-Commit-Position: refs/heads/master@{#9808}
2015-08-28 11:41:30 +00:00
4fbae2b791 Add send transports to individual webrtc::Call streams.
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1273363005

Cr-Commit-Position: refs/heads/master@{#9807}
2015-08-28 11:07:15 +00:00
6480d03f17 Make jni_helpers build on arm32.
BUG=None

Review URL: https://codereview.webrtc.org/1311753002

Cr-Commit-Position: refs/heads/master@{#9806}
2015-08-28 09:58:45 +00:00
6ec1f921b1 AndroidVideoCapturer: Delegate framerate choice to VideoCapturerAndroid.java
webrtc::VideoSource resolves the kMaxFrameRate constraint by capping the desired framerate to kMaxFrameRate. That framerate is then passed into cricket::VideoCapturer::GetBestCaptureFormat(). The default implementation will choose a format from the supported formats list. Instead, we should override this function in AndroidVideoCapturer to give VideoCapturerAndroid.java the opportunity to choose a suitable framerate range.

BUG=webrtc:4938
R=glaznev@webrtc.org

Review URL: https://codereview.webrtc.org/1308953004 .

Cr-Commit-Position: refs/heads/master@{#9805}
2015-08-28 09:41:06 +00:00
98f3cc54da NetEq: Removing two asserts
These asserts cover error cases that are also handled by the code
after the assert. Should not have both assert and error handling.

BUG=webrtc:4840

Review URL: https://codereview.webrtc.org/1321023002

Cr-Commit-Position: refs/heads/master@{#9804}
2015-08-28 08:12:26 +00:00
1e346b20c4 NetEq: Minor follow-up fix in StatisticsCalculator
This change follows the recommendation of a post-commit comment in
r9778:
https://codereview.webrtc.org/1296633002/diff/100001/webrtc/modules/audio_coding/neteq/statistics_calculator.cc#newcode198

TBR=pkasting@chromium.org

Review URL: https://codereview.webrtc.org/1319953002

Cr-Commit-Position: refs/heads/master@{#9803}
2015-08-27 20:41:10 +00:00
116c84e1b0 NetEq: Fixing a bug that caused rtc::checked_cast to trigger
This is a bug that was introduced in
https://codereview.webrtc.org/1230503003, where the variable "int
temp_bufsize" was changed to a size_t. If the packet buffer was
flushed while inserting a packet, temp_bufsize became negative, which
was tested later in an if-statement. Now, with size_t instead, it
would just become very large, and the if-statement would never see a
negative value. The effect was that the packet size in samples could
be updated with a very large positive number, causing an overflow
which triggered rtc::checked_cast in
StatisticsCalculator::GetNetworkStatistics, line 220.

Also adding a test to reproduce the crash. Without the fix, the test
results in the above mentioned checked_cast to trigger. With the fix,
everything works fine.

BUG=chromium:525260

Review URL: https://codereview.webrtc.org/1307893004

Cr-Commit-Position: refs/heads/master@{#9802}
2015-08-27 20:14:54 +00:00
9c3efd0052 Reland: Implement NetEq's CurrentDelay function
This was not implemented before. It returns the current total delay
(packet buffer and sync buffer) of NetEq. This is the same information
that was already available in
NetEqNetworkStatistics::current_buffer_size_ms, that can be obtained
through NetEq::NetworkStatistics(). But, since the current delay is a
key metric of NetEq, it is convenient to have it available in a
simpler way.

This is a re-landing of r9359,
https://webrtc-codereview.appspot.com/51149004, which was reverted in
r9360. The refactoring made in r9669 facilitated the relanding.

TBR=minyue@webrtc.org

Review URL: https://codereview.webrtc.org/1313873003

Cr-Commit-Position: refs/heads/master@{#9801}
2015-08-27 20:12:27 +00:00
a567bf3292 Rename local variable to avoid shadowing
See comment here: https://codereview.webrtc.org/1208993010/diff/180001/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h#newcode189

TBR=minyue@webrtc.org

Review URL: https://codereview.webrtc.org/1315333003

Cr-Commit-Position: refs/heads/master@{#9800}
2015-08-27 18:17:40 +00:00
31545685c7 Using 'override' keyword in dtlstransport.h.
Chromium FYI trybots were complaining about virtual methods not being marked override.

R=henrikg@webrtc.org
TBR=henrikg@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1318983002 .

Cr-Commit-Position: refs/heads/master@{#9799}
2015-08-27 13:44:25 +00:00
4376648df0 AudioDecoder: Replace Init() with Reset()
The Init() method was previously used to initialize and reset
decoders, and returned an error code. The new Reset() method is used
for reset only; the constructor is now responsible for fully
initializing the AudioDecoder.

Reset() doesn't return an error code; it turned out that none of the
functions it ended up calling could actually fail, so this CL removes
their error return codes as well.

R=henrik.lundin@webrtc.org

Review URL: https://codereview.webrtc.org/1319683002 .

Cr-Commit-Position: refs/heads/master@{#9798}
2015-08-27 13:22:21 +00:00
1c3dd38cb8 Android: Fix memory leak for remote MediaStream
BUG=webrtc:4892
R=glaznev@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1308733004 .

Cr-Commit-Position: refs/heads/master@{#9797}
2015-08-27 11:40:09 +00:00
7391881f97 Revert of Added send-thresholding and fixed text packet dumping. (patchset #4 id:160001 of https://codereview.webrtc.org/1266033005/ )
Reason for revert:
The CL adds a global variable.

Original issue's description:
> Added send-thresholding and fixed text packet dumping.  Also a little squelch for the over-max-MTU log spam we see in there.
>
> BUG=https://code.google.com/p/webrtc/issues/detail?id=4468
> R=pthatcher@chromium.org, pthatcher@webrtc.org
>
> Committed: d838d27919

TBR=pthatcher@webrtc.org,bemasc@webrtc.org,pthatcher@chromium.org,thakis@chromium.org,lally@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=https://code.google.com/p/webrtc/issues/detail?id=4468

Review URL: https://codereview.webrtc.org/1315923003

Cr-Commit-Position: refs/heads/master@{#9796}
2015-08-27 11:30:04 +00:00
fdac516510 Disallow simulcast for H.264.
BUG=

Review URL: https://codereview.webrtc.org/1291673006

Cr-Commit-Position: refs/heads/master@{#9795}
2015-08-27 08:59:32 +00:00
d82819892a Replaces SSLIdentity* with scoped_refptr<RTCCertificate> in the cricket::Transport layer.
Why the replacements? Mainly two reasons:
1) RTCCertificate owns the identity and as long as things are referencing the identity there should be a scoped_refptr reference to the RTCCertificate. Handing out raw pointers is less memory safe.
2) With the latest RFC, an RTCCertificate should be sufficient for specifying a crypto cert and the code should be updated to use RTCCertificate instead of SSLIdentity directly.

This replace work is split up into multiple CLs. In this CL...
- WebRtcSessionDescriptionFactory is updated to use RTCCertificate over SSLIdentity.
- WebRtcSessionDescriptionFactory::SignalCertificateReady is connected to WebRtcSession::OnCertificateReady and WebRtcSession is updated to use RTCCertificate.
- The cricket::Transport and related classes are updated to use RTCCertificate. These are called from WebRtcSession::OnCertificateReady.

BUG=webrtc:4927
R=tommi@webrtc.org, torbjorng@webrtc.org

Review URL: https://codereview.webrtc.org/1312643004 .

Cr-Commit-Position: refs/heads/master@{#9794}
2015-08-27 08:12:37 +00:00