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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
fdbe1442c5
Use C functions in aec for MIPS
...
With GCC 4.9, the MIPS NDK toolchain has been changed to only support 16 spregs by default - the even-numbered ones. This has been changed to support the R6 MIPS architecture. While the old behaviour could be restored by adding "-modd-spreg", this would come with a performance hit because the kernel would emulate odd-numbered spregs and missing R2 instructions.
As a result of this change, the functions removed in this CL no longer compile as there are no longer enough spregs for the compiler to assign. So we are removing these functions and they will use the C implementation until the MIPS code is rewritten.
R=andrew@webrtc.org , ljubomir.papuga@gmail.com , pasko@chromium.org
Review URL: https://webrtc-codereview.appspot.com/16159005
Patch from Fabrice de Gans-Riberi <fdegans@chromium.org >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6797 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-29 14:39:10 +00:00
e75d78d32d
Integrate rtcp packet class to rtcp receiver tests.
...
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6795 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-29 08:21:50 +00:00
f9460688a6
Make sure padding is sent on the first sending RTP module.
...
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6774 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-24 16:41:25 +00:00
194fea7640
The lastest commit on this file was in
...
https://webrtc-codereview.appspot.com/15529004/
The final patch set should have included this, but was missed.
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18839004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6755 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-22 09:55:51 +00:00
5ab7616983
Remove remains of WEBRTC_NO_STL.
...
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12959004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6752 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-22 06:48:58 +00:00
ceafa8cce9
MIPS optimizations for ISAC (patch #2 )
...
Implemented functions:
- WebRtcIsacfix_CalculateResidualEnergy
- WebRtcIsacfix_Spec2Time
- WebRtcIsacfix_Time2Spec
- WebRtcIsacfix_HighpassFilterFixDec32
- WebRtcIsacfix_PCorr2Q32
Gain achieved: aprox. further 5% on top of patch#1 on ISAC encoding path.
The optimizations are bit-exact to the C code, with the excception of the
MIPS DSPr2 variant of the WebRtcIsacfix_Time2Spec function (the accuracy of
the WebRtcIsacfix_Time2Spec MIPS DSPr2 variant is same or better than C
variant). Code verification and improvement achieved have been determined
using the iSACFixtest application.
R=andrew@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19749004
Patch from Ljubomir Papuga <lpapuga@mips.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6749 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-21 16:43:13 +00:00
f563e85ab0
This is to re-open an earlier CL
...
https://webrtc-codereview.appspot.com/16619005/
which is reverted due to an issue in audio conference mixer.
This issue has been solved in
https://webrtc-codereview.appspot.com/20779004/
BUG=webrtc:3155
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18819005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6736 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-18 21:11:27 +00:00
ff50debd37
Runtime guard for iOS7 property.
...
BUG=3487
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6733 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-18 17:17:59 +00:00
9343cf67a9
Fix crash in AudioDeviceUtilityIOS::~AudioDeviceUtilityIOS.
...
BUG=3581
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16109004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6732 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-18 17:13:28 +00:00
026859b983
This is related to an earlier CL of enabling Opus 48 kHz.
...
https://webrtc-codereview.appspot.com/16619005/
It was reverted due to a build bot error, which this CL is to fix. The problem was that when audio conference mixer receives audio frames all at 48 kHz and mixed them, it uses Audio Processing Module (APM) to do a post-processing. However the APM cannot handle 48 kHz input. The current solution is not to allow the mixer to output 48 kHz.
TEST=locally solved https://webrtc-codereview.appspot.com/16619005/
BUG=
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6730 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-18 12:28:28 +00:00
e364ac902f
AudioBuffer: Optimize const accesses to arrays that autoconvert int16<->float
...
Specifically, when someone asks for a const pointer to the int16
version of the array, there's no need to invalidate the float version
of that array, and vice versa. (But obviously, invalidation still has
to happen when someone asks for a non-const pointer.)
R=aluebs@webrtc.org , andrew@webrtc.org , minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6725 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-18 07:50:29 +00:00
c145668dc8
Reduce runtime of RingBufferTest by a factor of 100.
...
This test was needlessly long.
TBR=pbos
Review URL: https://webrtc-codereview.appspot.com/15029004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6724 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-17 23:16:44 +00:00
4f5da030f1
Use _numMixedParticipants instead of audioFrameList->size() to determine if there're more than one participants.
...
There are two audioFrameLists. The previous check wouldn't work correctly if each list had a single member.
TEST=chrome https://apprtc.appspot.com/?debug=loopback&video=false and verify e2e delay stats
BUG=
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15019004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6723 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-17 22:19:21 +00:00
8b94e3da0f
Fix issue where padding is sent before media with undefined timestamps if not abs-send-time is enabled.
...
This broke bandwidth estimation for calls without abs-send-time is enabled, but where RTX was.
BUG=
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16929004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6719 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-17 16:10:14 +00:00
4065988108
Remove unused ExperimentalNS API in AudioProcessing
...
R=andrew@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12909004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6718 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-17 11:32:09 +00:00
2b6bc8d84f
AudioBuffer: Eliminate the SplitChannelBuffer class
...
It's just a container for two IFChannelBuffers, and doesn't earn its
keep. The main problem is that the number of methods it needs that
just forward calls to either of its two IFChannelBuffers was already
large, and was about to grow.
R=aluebs@webrtc.org , minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6717 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-17 09:46:37 +00:00
2561d52460
Simplify AudioBuffer::mixed_low_pass_data API
...
R=andrew@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21869004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6715 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-17 08:27:39 +00:00
af93fc08a1
AudioBuffer: Let ChannelBuffer handle bounds checking of channel parameter
...
R=aluebs@webrtc.org , minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13019004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6714 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-17 08:18:33 +00:00
2ade42bd96
Add unit test for MediaFile WAV file writing
...
R=aluebs@webrtc.org , andrew@webrtc.org , minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16029004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6713 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-17 08:11:32 +00:00
4a472fb18d
Fixes up rtc so that it compiles on iOS 8 SDK.
...
Adds support for UIInterfaceOrientationUnknown (new with in SDK) and makes it the same as
UIInterfaceOrientationPortrait.
R=noahric@google.com , tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13029004
Patch from David Maclachlan <dmaclach@chromium.org >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6712 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-17 00:21:59 +00:00
c56ae63ea6
r6709 lacks a change in BUILD.gn
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BUG=
R=marpan@google.com , marpan@webrtc.org , pbos@webrtc.org , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6710 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-16 22:18:49 +00:00
74aaf29a0f
Raw packet loss rate reported by RTP_RTCP module may vary too drastically over time. This CL is to add a filter to the value in VoE before lending it to audio coding module.
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The filter is an exponential filter borrowed from video coding module.
The method is written in a new class called PacketLossProtector (not sure if the name is nice), which can be used in the future for more sophisticated logic.
BUG=
R=henrika@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6709 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-16 21:28:26 +00:00
2e3c97ddf5
Compile-time guard for iOS7 specific property.
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BUG=3487
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6706 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-16 19:59:05 +00:00
63c60ed224
Remove old padding path in RTPSender.
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Removing RTPSender::SendPaddingAccordingToBitrate() as well as a couple
of arguments from SendPadData().
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6703 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-16 09:37:29 +00:00
e8ea33ccb1
nrsh1 is written before tmp321 is read, so needs to be earlyclobber
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Otherwise, the compiler is allowed to put them in the same register
under the assumption that all inputs are read before any
(non-earlyclobber) output is written, which in this case would result
in nrsh2 being corrupted.
BUG=3439
R=aluebs@webrtc.org , ljubomir.papuga@gmail.com
Review URL: https://webrtc-codereview.appspot.com/16089004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6700 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-16 08:26:48 +00:00
bac5f0fb56
Fix an invalid memory access due to typo in win/cursor.cc.
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BUG=crbug/391468
R=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/19949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6698 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-15 20:32:03 +00:00
122caa51b1
After an audio interruption the audio unit no longer invokes its render callback, which results in a loss of audio. Restarting the audio unit post interruption fixes the issue.
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CL also replaces deprecated AudioSession calls with equivalent AVAudioSession ones.
BUG=3487
R=glaznev@webrtc.org , noahric@chromium.org
Review URL: https://webrtc-codereview.appspot.com/21769004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6697 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-15 20:20:47 +00:00
