Commit Graph

4245 Commits

Author SHA1 Message Date
e9b493e763 Removing macro in acm_opus.cc
Remove it since macros are not recommended to use according to code style guide.

BUG=
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6917 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-18 12:06:31 +00:00
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
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
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
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
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
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
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
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
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
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
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
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
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
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