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2714 Commits

Author SHA1 Message Date
bjornv@webrtc.org
b5ab52d010 common_audio/signal_processing: Remove unused macros WEBRTC_SPL_GET_BYTE and WEBRTC_SPL_SET_BYTE 
These two macros are not used anywhere in webrtc. Previously used in old neteq (I think).

BUG=3348,3353
TESTED=manually on linux and trybots
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6916 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-18 12:01:02 +00:00
braveyao@webrtc.org
8a2c84f59d Log the Android Audio API choice correctly. 
BUG=3699
TEST=Manual Test
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6915 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-18 03:02:42 +00:00
kjellander@webrtc.org
d235eaef25 Suppress deprecation warnings in video_capture for iOS 
The chromium_revision roll in r6913 broke the iOS build since the
videoMinFrameDuration and videoMaxFrameDuration properties
have been deprecated in iOS 7.0, which is now the default target
platform for iOS.

BUG=3705
TEST=Passing ios and ios_rel trybots.
TBR=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6914 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-16 20:47:16 +00:00
kjellander@webrtc.org
34a865a038 Roll chromium_revision 288251:289723 
Mainly to pick up the libvpx.gyp change in r288724
to unblock https://webrtc-codereview.appspot.com/16229005/

Overview of changes in Chrome DEPS:
$ svn diff http://src.chromium.org/chrome/trunk/src/DEPS -r 288251:289723
which can be compared with the output of:
$ svn cat http://webrtc.googlecode.com/svn/trunk/DEPS | grep chromium_deps | sed 's/^ *//' | sort | uniq

In a WebRTC checkout, that sums up to the following relevant changes:
* src/buildtools 59b932:567f0a
* testing/gtest 643:692
* testing/gmock 410:485
* third_party/boringssl/src 533cbe:c3d796
* third_party/libvpx 287125:289332
* third_party/libyuv 1035:1038
* third_party/nss 287121:289430
* third_party/opus/src 256783:289085
* tools/gyp 1959:1964

BUG=2863, chromium:339647
TEST=Local testing as trybots currently cannot handle DEPS changes properly due to http://crbug.com/385594
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6913 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-16 18:49:55 +00:00
sergeyu@chromium.org
d402875fa5 Set updated_rect for frames generated by WindowCapturer implementationsw 
Previous updated_rect wasn't set for frames generated by WindowCapturer
implementation. That makes them unustable with chromoting host that
uses update_rect. With that change the frames will always contain
updated_rect that coveras the whole frame.

Change by Ronak Vora <ronakvora@google.com>

R=wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/22079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6912 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-15 23:13:23 +00:00
henrike@webrtc.org
fb1eb43377 Rename linuxwindowpicker to x11windowpicker & only use it with use_x11 
These days we have Linux chromium builds that don't use X11. We don't
want webrtc to add an X11 dependency to those builds.

BUG=3625
R=henrike@webrtc.org, tnakamura@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6909 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-15 14:44:13 +00:00
bjornv@webrtc.org
1e3ef4b999 common_audio/signal_processing: Remove macro WEBRTC_SPL_UMUL_32_16_RSFT16 
Macros should in general be avoided. WEBRTC_SPL_UMUL_32_16_RSFT16 is only used in iSAC fixed point as part of multiplying with LSB and MSB. A better approach is to have one function for that complete operation in iSAC.

This CL removes the macro and replace the operation locally.

BUG=3148, 3353
TESTED=locally on Linux and trybots
R=tina.legrand@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6907 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-15 05:17:20 +00:00
andresp@webrtc.org
a84b0a6dab Small refactor on ViE to remove redudant conditions and long ifdefs. 
BUG=3694
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6905 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-14 16:46:46 +00:00
stefan@webrtc.org
58e2d262fc Return an aggregated report from ViERtpRtcp::GetSentRTCPStatistics(). 
Fixes issues where statistics only was reported for the first stream if
configured with simulcast, and in case of RTX the reported statistics was
depending on the order of the report blocks.

Also fixes issues with multiple report blocks in the SendStatisticsProxy and the
RtcpStatisticsCallback. SendStatisticsProxy is now aware of RTX ssrcs, and the
RTCPReceiver is calling the RtcpStatisticsCallback with the correct SSRCs, and
not only the primary stream SSRC.

R=mflodman@webrtc.org, sprang@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6903 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-14 15:10:49 +00:00
minyue@webrtc.org
e8018b0b24 Adding a 5% as packet loss level for Opus 
This is a follow up of
https://webrtc-codereview.appspot.com/16979004/

The purpose of this CL is to add 5% as a level for optimizing the packet loss rate to report to Opus. Adding such a level makes the grid finer.

BUG=
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6902 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-14 12:16:12 +00:00
minyue@webrtc.org
4521e2d0bd Adding online bitrate change to voe_cmd_test 
This is to verify a way of changing the bitrate on-the-fly under current WebRTC implementation.

TEST=changing bit rate for different codecs. sound quality changed when bit rate was set successful. catched error when bit rate is invalid for a running codec.

BUG=
R=andrew@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17059004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6901 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-14 12:15:27 +00:00
andresp@webrtc.org
817a034cf2 Fix TimeToSendPadding return to be 0 if no padding bytes are sent. 
BUG=3694
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15149005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6900 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-14 08:24:47 +00:00
bjornv@webrtc.org
8434dbe284 common_audio/signal_processing: Remove macro WEBRTC_SPL_SUB_SAT_W32 
This macro is literally using the function WebRtcSpl_SubSatW32(), hence there is no need for a macro.

BUG=3348, 3353
TESTED=locally on Linux and trybots
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6899 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-14 07:26:28 +00:00
asapersson@webrtc.org
23a4d8522e Decreased kMaxOverusesBeforeApplyRampupDelay (from 7 to 4). 
Increased kStandardRampUpDelayMs (30 to 40s).

BUG=1577
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6886 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-13 14:33:49 +00:00
henrik.lundin@webrtc.org
5af76aedcd Removing TODOs related to AcmReceiverBitExactness checksums 
Should have been part of r6883.

BUG=3519
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6884 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-13 13:02:00 +00:00
henrik.lundin@webrtc.org
388bd79a76 Update checksums for AcmReceiverBitExactness on android 
This should have been a part of r6882.

BUG=3519
TBR=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22059004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6883 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-13 10:38:15 +00:00
henrik.lundin@webrtc.org
023f12fb6e NetEq background noise generation off by default 
This CL turns the background noise generation in NetEq off by default. The noise generation used to kick in during long-duration packet losses, when there was no point in extrapolating the latest audio any longer. However, this sometimes produces annoying noise in situations where silence would have been preferable.

With this change, a long packet-loss concealment will be faded out to zeros instead of a low noise.

Reference files are updated where needed.

BUG=3519
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6882 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-13 09:45:40 +00:00
stefan@webrtc.org
c27543d297 Fix STAP-A bug where we might overflow the packet buffer due to not accounting for the length of the length field. 
BUG=3679
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6881 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-13 07:40:45 +00:00
fbarchard@google.com
c891fee7ab Make a int64 constant use ULL suffix so it wont get truncated. 
BUG=3690
TESTED=try bots
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6878 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-12 22:39:06 +00:00
phoglund@webrtc.org
40995c7fd0 Fixing uninitialized variable in file_audio_device.cc. 
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6872 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-12 11:09:12 +00:00
bjornv@webrtc.org
0a3cbb3906 common_audio/signal_processing: Removes macro WEBRTC_SPL_MUL_32_32_RSFT32 
The macro is only used at four places in iSAC fixed point and the macro have been replaced at those places.
In addition, it is used in a unit test, but throws a warning treated as error (issue3674).

The macro has both MIPS and armv7 optimizations. Removing them impacts only MIPS platforms without DSP ASE. This may cause a very small increase in complexity when using iSAC fix.
The armv7 optimizations are not used anywhere, since specific ones are used inline in iSAC fix.

BUG=3348,3353,3674
TESTED=locally and trybots
R=ljubomir.papuga@gmail.com, tina.legrand@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6871 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-12 10:54:50 +00:00
bjornv@webrtc.org
cf8f33a6d6 Removes mismatching signs in signal_processing_unittests 
Negative inputs was used in WebRtcSpl_NormU32() causing warnings.

BUG=3674
TESTED=locally and trybots
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6870 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-12 10:27:21 +00:00
minyue@webrtc.org
6aac93bd9c Adding SetOpusMaxBandwidth in VoE and ACM 
This is a step to solve
https://code.google.com/p/webrtc/issues/detail?id=1906

In particular, we add an API in VoE and ACM to call Opus's API of setting maximum bandwidth.

TEST = added a test in voe_cmd_test and listened to the result

BUG=
R=henrika@google.com, henrika@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6869 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-12 08:13:33 +00:00
henrike@webrtc.org
6ac22e6b47 Remove more dependencies on openssl, add dependency on boringssl. Continues on r6798 
R=andrew@webrtc.org, fbarchard@chromium.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6867 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-11 21:06:30 +00:00
bjornv@webrtc.org
820f8e9ca7 modules/audio_processing: Moves declaration of kDelayDiffOffsetSamples 
audio_processing did not compile when aec_untrusted_delay_for_testing=1 was set. The constant kDelayDiffOffsetSamples was declared only for Mac when WEBRTC_UNTRUSTED_DELAY was automatically turned on.

Moving the declaration outside the ifdef makes it build with the flag on for any platform.

BUG=3673
TESTED=locally and trybots
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6866 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-11 15:39:00 +00:00
henrik.lundin@webrtc.org
4e4b0984da Merge NetEqDecodingTest.TestBitExactnesst and .TestNetworkStatistics 
The two tests both read and process the same (rather long) RTP input
file, and simply look at different outputs. This change merges the two
tests into one, in order to reduce testing time.

BUG=
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6865 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-11 14:48:49 +00:00
henrike@webrtc.org
065247b5b7 Rebase webrtc/base with r6863 version of talk/base: 
cls integrated: r6809
svn diff -r 6808:6809 http://webrtc.googlecode.com/svn/trunk/talk/base > 6809.diff
patch -p0 -i 6809.diff

BUG=3379
TBR=solenberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6864 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-11 14:32:13 +00:00
henrik.lundin@webrtc.org
1c8391205e Use test::Packet test::PacketSource classes in neteq_rtpplay 
This change replaces the old NETEQTEST_RTPpacket and
NETEQTEST_DummyRTPpacket with the new test::Packet class. Note that the
Packet class automatically handles "dummy" packets (i.e., packets for
which only the header and a length field was stored to file)
automatically. There is no need to explicitly signal this to the
application any longer. The RTP input file is now handled as a
test::PacketSource object.

Also adding a new ConvertHeader method to the Packet class. This is
needed to extract the header information as an alternative data type.

Finally, some dead code was deleted from rtp_analyze.cc (unrelated to
the reset of this change).

BUG=2692
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6862 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-11 12:29:38 +00:00
bjornv@webrtc.org
96d8b0e69f Revert 6860 "SSE2 version of SubbandCoherence()" 
> SSE2 version of SubbandCoherence()
> 
> The performance gain on a x86 laptop (Intel(R) Core(TM) i5-2520M CPU @ 2.50GHz)
> reported by audioproc is ~3.3%
> 
> The output is bit exact.
> 
> R=bjornv@webrtc.org, cd@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/18779004
> 
> Patch from Scott LaVarnway <slavarnw@gmail.com>.

TBR=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6861 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-11 12:09:13 +00:00
bjornv@webrtc.org
0db82f337f SSE2 version of SubbandCoherence() 
The performance gain on a x86 laptop (Intel(R) Core(TM) i5-2520M CPU @ 2.50GHz)
reported by audioproc is ~3.3%

The output is bit exact.

R=bjornv@webrtc.org, cd@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18779004

Patch from Scott LaVarnway <slavarnw@gmail.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6860 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-11 10:38:31 +00:00
henrike@webrtc.org
3763b9bda0 webrtc/base: removes linkage of crypto 
BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6853 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 21:26:18 +00:00
stefan@webrtc.org
59a2f9f584 Remove the old H264 code now that a new H.264 packetizer has been implemented. 
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6847 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 15:09:24 +00:00
stefan@webrtc.org
9d74f7ce8c Fix single nalu packetization bug. 
Nalus which had the same size as the max payload size would cause the payload size accounting to wrap.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6846 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 15:02:16 +00:00
pbos@webrtc.org
e8c84bf4de Fix so video_replay logs aren't spammed. 
Add unknown-SSRC counters instead and log number of unknown packets at
end of session.

R=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/13119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6845 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 14:42:45 +00:00
minyue@webrtc.org
1d956dd1a7 Since the packet loss rate cannot be estimated accurately, there is always a mismatch between the estimated packet loss rate and the true one. Such a mismatch will make Opus FEC suboptimal. 
It is advisable to set the packet loss rate of FEC conservatively. Say, if the estimated loss rate is 5%, we can set it to 1%. The risk of degradation in quality is small and the overall performance is good.

BUG=
R=henrik.lundin@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6844 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 12:31:36 +00:00
henrik.lundin@webrtc.org
ea25784107 Change how background noise mode in NetEq is set 
This change prepares for switching default background noise (bgn) mode
from on to off. The actual switch will be done later.

In this change, the bgn mode is included as a setting in NetEq's config
struct. We're also removing the connection between playout modes and
bgn modes in ACM. In practice this means that bgn mode will change from
off to on for streaming mode, but since the playout modes are not used
it does not matter.

BUG=3519
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6843 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 12:27:37 +00:00
pbos@webrtc.org
4b5625e5ac RTP video playback tool using Call APIs. 
Plays back rtpdump files from Wireshark in realtime as well as save the
resulting raw video to file. Unlike the RTP playback tool it doesn't
support faster-than-realtime playback/rendering, but it instead utilizes
the same path as production code and also contains support for playing
back FEC.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6838 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-06 16:26:56 +00:00
stefan@webrtc.org
1ccff349ee Fix crashing fake network pipe tests. 
These tests are not included in bots, this will be fixed in a follow-up by pbos@webrtc.org.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6837 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-06 15:41:58 +00:00
minyue@webrtc.org
2a8df7c375 Fixing two bugs in voe_cmd_test. 
I am trying to add a new functionality to voe_cmd_test, and I found two bugs:

1. in r5928, a functionality was removed but the item in the menu was not. Functionalities after it are offset.

r5928: https://code.google.com/p/webrtc/source/detail?r=5928&path=/trunk/webrtc/voice_engine/test/cmd_test/voe_cmd_test.cc

2. in r6736, opus are set to output 48 kHz audio. When mixing Opus output with an audio file, channel.cc may go wrong.

r6736: https://code.google.com/p/webrtc/source/detail?r=6736

BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6836 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-06 10:05:19 +00:00
stefan@webrtc.org
79c3359e67 Add end-to-end H.264 packetization test. 
Also correctly wires up H.264 packetization in the new Call api.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6835 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-06 09:24:53 +00:00
stefan@webrtc.org
8b033adb19 Change the way we reference enumerators in H.264 packetization code to be standard C++ compliant. 
R=kjellander@webrtc.org, phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6833 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-06 08:06:53 +00:00
fbarchard@google.com
d7b4dea801 initialize packet len in NETEQTEST_DummyRTPpacket.cc and NETEQTEST_RTPpacket.cc to fix build error on vs2013 
BUG=3660
TESTED=set DEPOT_TOOLS_WIN_TOOLCHAIN=0 & set GYP_DEFINES=target_arch=ia32 & call python webrtc\build\gyp_webrtc -G msvs_version=2013 &ninja -C out\Debug
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21109005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6831 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-05 23:46:42 +00:00
pbos@webrtc.org
dde16f19e3 Fix some code styles. 
BUG=
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22009004

Patch from Changbin Shao <changbin.shao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6830 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-05 23:35:43 +00:00
fbarchard@google.com
e086af0fa3 Fix implicite cast from signed int to unsigned int in unittest.cc 
BUG=3636
TESTED=set GYP_DEFINES=target_arch=ia32 & call python webrtc\build\gyp_webrtc -G msvs_version=2013 & ninja -C out\Debug
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6827 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-05 17:10:52 +00:00
stefan@webrtc.org
fdcb42dac4 Fix potential crash when depacketizing VP8. 
Caused by a missing check for H264 when reading the RTPVideoTypeHeader union.

R=asapersson@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6825 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-05 13:21:18 +00:00
henrike@webrtc.org
d6542852f3 Unbreaks linux.cc in Chromium. 
BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6823 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-04 21:51:14 +00:00
minyue@webrtc.org
0040a6ef97 This is a setup to solve 
https://code.google.com/p/webrtc/issues/detail?id=1906

In particular, we add an API to call Opus's set maximum bandwidth to prevent the encoder from coding audio content beyond this bandwidth so as to increase computation and transmission efficiency (without affecting sampling rate).

BUG=
R=henrik.lundin@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6817 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-04 14:41:57 +00:00
asapersson@webrtc.org
84b9e1e9d9 Fix for retransmission. Base layer packets were not retransmitted. 
Issue introduced in r6669.

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6816 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-04 11:59:42 +00:00
stefan@webrtc.org
e1c9caf6ee Fix mistake in rtp/rtcp/BUILD.gn introduced with r6804. 
TEST=buildtools/linux64/gn gen out/Default
TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6805 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-31 15:07:59 +00:00
stefan@webrtc.org
2ec560606b Add H.264 packetization. 
This also includes:
- Creating new packetizer and depacketizer interfaces.
- Moved VP8 packetization was H264 packetization and depacketization to these interfaces. This is a work in progress and should be continued to get this 100% generic. This also required changing the return type for RtpFormatVp8::NextPacket(), which now returns bool instead of the index of the first partition.
- Created a Create() factory method for packetizers and depacketizers.

R=niklas.enbom@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6804 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-31 14:59:24 +00:00