d4fe824862
audio_processing: Replaced macro WEBRTC_SPL_RSHIFT_W16 with >>
...
The implementation of WEBRTC_SPL_RSHIFT_W16 is simply >>. This CL removes the macro usage in audio_processing and signal_processing.
Affected components:
* aecm
* agc
* nsx
Indirectly affecting (through signal_processing changes)
* codecs/cng
* codecs/isac/fix
* codecs/isac/main
BUG=3348,3353
TESTED=locally on Linux and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28699005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7432 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-13 13:01:13 +00:00
396a5e0001
WebRtcIsac_Decode et al.: Type encoded data as uint8[], not uint16[]
...
This patch changes WebRtcIsac_Decode, WebRtcIsac_DecodeRcu, and
WebRtcIsacfix_Decode so that they read the encoded data from a uint8
array instead of a uint16 array.
BUG=909
R=aluebs@webrtc.org , bjornv@webrtc.org , henrik.lundin@webrtc.org , turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25739004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7431 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-13 11:23:24 +00:00
3f7f899a15
WebRtcIsac_UpdateBwEstimate et al.: Type byte streams as uint8, not uint16
...
This patch changes the signature of WebRtcIsac_UpdateBwEstimate,
WebRtcIsacfix_UpdateBwEstimate, and WebRtcIsacfix_UpdateBwEstimate1 so
that they expect the encoded data to be uint8 arrays, not uint16,
which is more natural. The implementations of the functions are left
unchanged for now.
BUG=909
R=aluebs@webrtc.org , bjornv@webrtc.org , henrik.lundin@webrtc.org , turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7430 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-13 11:07:06 +00:00
1172988c79
Some WebRtcIsac_* and WebRtcIsacfix_* functions: type encoded stream as uint8[]
...
The affected functions are
WebRtcIsacfix_ReadFrameLen
WebRtcIsacfix_GetNewBitStream
WebRtcIsacfix_ReadBwIndex
and
WebRtcIsac_ReadFrameLen
WebRtcIsac_GetNewBitStream
WebRtcIsac_ReadBwIndex
WebRtcIsac_GetRedPayload
BUG=909
R=aluebs@webrtc.org , henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22979004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7429 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-13 10:53:42 +00:00
c502df54f8
Merge the supporting to UYVY on Linux video capture in crbug/410202 to webrtc standalone.
...
BUG=3765
TEST=Manual
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7427 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-13 02:13:00 +00:00
651c05e4fc
Release _inputSendPin & _outputCapturePin before _captureFilter & _sinkFilter since they should depend on the filters.
...
The previous steps work fine for all the webcam, but have problem on SplitCam driver as in the issue report.
Anyway it's always good to de-initial with the reversing order to initialization.
BUG=3845
TEST=Manual
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25659004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7426 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-13 02:11:55 +00:00
c87b74717b
Adjust/increase rate control thresold for a vp9 test.
...
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27689004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7423 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-10 17:55:57 +00:00
573c78e31c
Add VP9 codec to VCM and vie_auto_test.
...
Include VP9 tests in videoprocessor_integrationtests.
Include end-to-end send/receiveVP9 test.
Passes trybots.
R=kjellander@webrtc.org , mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29449004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7422 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-10 16:44:47 +00:00
3cefbc99f4
Mark all virtual overrides in the hierarchy of Transport as virtual + OVERRIDE.
...
This also marks all virtual overrides of other classes in the same files.
This will make a subsequent change I intend to do safer, where I'll change the
argument types of the base Transport functions, by breaking the compile if I
miss any overrides.
This also highlighted a number of unused functions. I've removed some of these.
TBR=mflodman@webrtc.org , pkasting@chromium.org
BUG=none
TEST=none
Review URL: https://webrtc-codereview.appspot.com/28709004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7421 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-10 09:42:53 +00:00
e46bc77e94
Reland 28629004: adding new AEC dump start interface for chrome.
...
This is required because we are not allow to pass CRT objects across dll boundaries, that says, when we pass a file descriptor from chrome dll to libpeerconnection dll, the file descriptor will become invalid immediate, more information can be found here:
http://msdn.microsoft.com/en-us/library/ms235460.aspx
R=andresp@webrtc.org , andrew@webrtc.org , bjornv@webrtc.org , henrike@webrtc.org , henrikg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30629004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7418 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-10 08:36:56 +00:00
4bd2db9a55
Opus wrapper: Use const for inputs and uint8[] for byte streams
...
About half of the functions already followed the desired pattern; this
patch fixes the other half.
BUG=909
R=aluebs@webrtc.org , bjornv@webrtc.org , henrik.lundin@webrtc.org , minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7409 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-09 11:21:10 +00:00
2c0cdbce22
Estimating NTP time with a given RTT.
...
RemoteNtpTimeEstimator needed user to give a remote SSRC and it intended to call RtpRtcp module to obtain RTT, to be able to calculate Ntp time.
When RTT cannot be directly obtained from the RtpRtcp module with the specified SSRC, RemoteNtpTimeEstimator would fail.
This change allows RemoteNtpTimeEstimator to calculate NTP with an external RTT estimate.
An immediate benefit is that capture_start_ntp_time_ms_ can be obtained in a Google hangout call.
BUG=
TEST=chromium + hangout call
R=stefan@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7407 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-09 10:52:43 +00:00
c803907d87
Removing useless packets when inserting them (NetEq)
...
This is to save the buffer.
Some old code may become unnecessary, and will be removed in a separate CL.
BUG=
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7406 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-09 10:49:54 +00:00
3ea35fdb1b
common_audio: Removed macro WEBRTC_SPL_LSHIFT_W16
...
The macro was a trivial << operation and where used has been replaced by <<. Affected components are
* ilbc
* isacfix
BUG=3348,3353
TESTED=locally on linux and trybots
R=henrik.lundin@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22919005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7404 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-09 08:47:02 +00:00
f71785cd3b
audio_coding: Replaced macro WEBRTC_SPL_RSHIFT_W16 with >>
...
Replaced trivial shift macro with >>. The actual implementation of the macro is simply >>.
Affected codecs:
* ilbc
* isac/fix
BUG=3348,3353
TESTED=locally on linux and trybots
R=henrik.lundin@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23889004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7396 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-08 15:36:30 +00:00
5e3d7c78de
Change name of a NetEq internal member variable
...
In the StatisticsCalculator class, the member last_report_timestamp_
was unfortunately named. It's now been changed to
timestamps_since_last_report_, which describes it more accurately.
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7393 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-08 12:10:53 +00:00
a9e363e721
Roll chromium_revision c264a05..fc668e2 (297113:298195)
...
Mainly to pick up https://codereview.chromium.org/619723006/
to fix our MSan bot.
Summary of changes (git diff c264a05..fc668e2 DEPS):
* third_party/boringssl 01fe820..c7dd5f30
* third_party/usrsctp/usrsctplib 8975bd5..d5685d4
* tools/swarming_client 79940aee..33d442a
Clang updated 216630:217949 (git diff c264a05..fc668e2 tools/clang/scripts/update.sh)
This caused TSan v2 to hit an assert in libjingle_peerconnection_unittest
and libjingle_media_unittest, which is why the dlclose call
had to be disabled for now (webrtc:3895).
BUG=3895
R=henrika@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28659004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7385 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-07 12:49:34 +00:00
9103953b58
Fix neteq_rtpplay so that empty SSRC is valid
...
In r7380, the command line flag --ssrc was added to neteq_rtpplay.
However, it was not possible to omit that flag, since the validation
did not accept an empty string. This CL fixes that.
TBR=kwiberg@webrtc.org
BUG=2692
Review URL: https://webrtc-codereview.appspot.com/24869004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7382 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-07 07:18:36 +00:00
7cbc4f969a
Set NetEq playout mode through the Config struct
...
This change opens up the possibility to set the playout mode when
creating the NetEq object. The old methods SetPlayoutMode and
PlayoutMode are still available, but are deprecated.
BUG=3520
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23869004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7381 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-07 06:37:39 +00:00
8b65d511a0
Add an SSRC filter to neteq_rtpplay
...
This allows the user to set the desired SSRC if the input file
contains multiple streams.
BUG=2692
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30609004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7380 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-07 05:30:04 +00:00
532ed43e85
Prevent reading outside iSAC bitstream, if the stream is corrupted.
...
BUG=chrome_373312(#24 )
R=henrik.lundin@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20089004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7379 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-07 00:21:02 +00:00
4b133da5fd
Let RtpFileSource use RtpFileReader
...
RtpFileSource used to implement it's own RTP dump file reader, but
with this change it simply uses RtpFileReader. One benefit is that
pcap files are now also supported.
All NetEq test tools that use RtpFileSource are updated.
BUG=2692
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22839004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7367 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-02 08:19:38 +00:00
348eac641e
audio_processing: Replaced WEBRTC_SPL_RSHIFT_U32 with >>
...
A trivial macro that is replaced. Affected components:
* AGC
* NSx
BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29599005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7366 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-02 08:07:05 +00:00
5fa8c458d8
Remove mouse cursor capturer from the ScreenCapturer interface
...
Mouse can be captured using MouseCursorMonitor and all code in chromium
already uses it instead of ScreenCapturer.
R=jiayl@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=7363
Review URL: https://webrtc-codereview.appspot.com/31529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7365 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-02 01:47:10 +00:00
6138f0f89d
Revert "Remove mouse cursor capturer from the ScreenCapturer interface"
...
This reverts commit 0adc4953512ee0a57cf7f3c0591b024c2316554a. It broke
FYI bots
TBR=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/27649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7364 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-02 01:36:43 +00:00
1fced0f2aa
Remove mouse cursor capturer from the ScreenCapturer interface
...
Mouse can be captured using MouseCursorMonitor and all code in chromium
already uses it instead of ScreenCapturer.
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7363 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-02 00:18:10 +00:00
76819d315d
Add error trap for XFixesGetCursorImage()
...
BUG=https://code.google.com/p/webrtc/issues/detail?id=3245
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31519004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7362 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-01 23:07:12 +00:00
c86e45d7c4
Fix parallelizability in modules_tests.
...
R=henrik.lundin@webrtc.org
BUG=3873
TEST=third_party/gtest-parallel/gtest-parallel -r 10 -w 64 out/Debug/modules_tests
Review URL: https://webrtc-codereview.appspot.com/24799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7354 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-01 10:05:40 +00:00
b8caf6a504
GN: Enable libvpx, add link target and convert some test targets
...
Libvpx now supports GN and this CL turns on compiling it.
I also introduced an executable target 'webrtc_tests'
that depends on all in WeBRTC + tests in order to get a full
linking step executed (since we've seen link problems for GN
when rolling WebRTC into Chromium).
I also converted a few test targets and made a GN file for
third_party/gflags.
BUG=3441
TESTED=Trybots + full Chromium build with a symlinked src/third_party/webrtc
dir to a workspace wit this CL applied.
R=brettw@chromium.org
TBR=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25569004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7344 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30 18:05:02 +00:00
79a7148108
Revert 7337 "Reland 28629004: adding new AEC dump start interfac..."
...
> Reland 28629004: adding new AEC dump start interface for chrome
>
> adding new AEC dump start interface for chrome.
>
> This is required because we are not allow to pass CRT objects across dll boundaries, that says, when we pass a file descriptor from chrome dll to libpeerconnection dll, the file descriptor will become invalid immediate, more information can be found here:
> http://msdn.microsoft.com/en-us/library/ms235460.aspx
>
> Chromium bug:crbug/415935
> TEST=bots
> R=bjornv@webrtc.org , kwiberg@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/27639004
TBR=xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22849004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7342 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30 15:29:13 +00:00
7aad5e5cce
Revert 7338 "Fixed the android build by making the interface pur..."
...
> Fixed the android build by making the interface pure virtual.
>
> TBR=asapersson@webrtc.org , bjornv@webrtc.org ,
>
> Review URL: https://webrtc-codereview.appspot.com/24789004
TBR=xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7341 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30 15:26:15 +00:00
db75a66b0f
Minor code change to fix some warnings in MIPS build.
...
R=andrew@webrtc.org , bjornv@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26619004
Patch from Ljubomir Papuga <lpapuga@mips.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7339 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30 15:17:50 +00:00
90d1979d77
Fixed the android build by making the interface pure virtual.
...
TBR=asapersson@webrtc.org , bjornv@webrtc.org ,
Review URL: https://webrtc-codereview.appspot.com/24789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7338 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30 15:15:22 +00:00
14092e00f1
Reland 28629004: adding new AEC dump start interface for chrome
...
adding new AEC dump start interface for chrome.
This is required because we are not allow to pass CRT objects across dll boundaries, that says, when we pass a file descriptor from chrome dll to libpeerconnection dll, the file descriptor will become invalid immediate, more information can be found here:
http://msdn.microsoft.com/en-us/library/ms235460.aspx
Chromium bug:crbug/415935
TEST=bots
R=bjornv@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27639004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7337 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30 14:35:15 +00:00
875206196c
Revert 7334 "adding new AEC dump start interface for chrome."
...
> adding new AEC dump start interface for chrome.
>
> This is required because we are not allow to pass CRT objects across dll boundaries, that says, when we pass a file descriptor from chrome dll to libpeerconnection dll, the file descriptor will become invalid immediate, more information can be found here:
> http://msdn.microsoft.com/en-us/library/ms235460.aspx
>
> Chromium bug:crbug/415935
> TEST=bots
> R=bjornv@webrtc.org , kwiberg@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/28629004
TBR=xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26659004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7335 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30 13:30:05 +00:00
2e417d6428
adding new AEC dump start interface for chrome.
...
This is required because we are not allow to pass CRT objects across dll boundaries, that says, when we pass a file descriptor from chrome dll to libpeerconnection dll, the file descriptor will become invalid immediate, more information can be found here:
http://msdn.microsoft.com/en-us/library/ms235460.aspx
Chromium bug:crbug/415935
TEST=bots
R=bjornv@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28629004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7334 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30 13:11:27 +00:00
65e56dba53
audio_processing/aecm: Added help function for calculating log of energy
...
The same operation of calculating log of the energy was executed four times. This CL adds a help function LogOfEnergyInQ8() for that.
BUG=N/A
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7331 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30 09:31:28 +00:00
23ec8372a6
audio_processing: Removed usage of macro WEBRTC_SPL_MUL
...
WEBRTC_SPL_MUL is a trivial multiplication after casting to int32_t. This is already taken care of by the compiler which makes the macro unnecessary.
Affected components:
* AGC
* NSx
BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7330 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30 09:29:28 +00:00
750423c722
audio_processing: Replaced trivial macro WEBRTC_SPL_LSHIFT_W32 with <<
...
Affected components:
* AECM
* AGC
* HPF
* NSx
BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27629004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7329 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30 09:26:36 +00:00
d71118194f
audio_processing: Replaced macro WEBRTC_SPL_LSHIFT_W16 with <<
...
A trivial macro that serves no purpose. Affected components are:
* audio_processing/nsx
* audio_processing/agc
* audio_processing/aecm
* common_audio/LpcToReflCoef
BUG=3348,3353
TESTED=locally on linux
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7321 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-29 10:56:27 +00:00
7c15510f38
common_audio refactoring: Removed macro WEBRTC_SPL_LSHIFT_U32
...
The macro is a trivial shift operator including a cast before shift. There is no guard against negative shifts. Replaced with << at place and added casts when necessary.
Affects both fixed and float point versions of iSAC
BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7320 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-29 09:40:38 +00:00
730d270771
Remove callback from RtpDepacketizer::Parse().
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BUG=
R=pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30489004
Patch from Changbin Shao <changbin.shao@intel.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7318 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-29 08:00:22 +00:00
f21ea918ad
GN: Add common configs to all targets.
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This is needed to ensure we have the same build with GN
as with GYP, since GYP includes the common.gypi on a global level.
Several fixes has been needed in the past because some code have
been built without the right defines.
BUG=3441
R=brettw@chromium.org
Review URL: https://webrtc-codereview.appspot.com/28589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7317 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-28 17:37:22 +00:00
0a256acb67
Getting orientation is not working properly. VideoCaptureImpl::RotationFromDegrees returns -1 in case fails not 0. So we need to change the if statement.
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BUG=3869
TEST=Manual
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25699004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7311 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-26 22:50:06 +00:00
384d05f362
Remove the different block lengths in ns_core
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Relanding the CL: https://webrtc-codereview.appspot.com/30539004/
It had to be reverted because some development code was uploaded by mistake.
TBR=bjornv@webrtc.org
BUG=webrtc:3811
Review URL: https://webrtc-codereview.appspot.com/28589005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7307 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-26 14:41:19 +00:00
5088377d70
Revert 7297 "Remove the different block lengths in ns_core"
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> Remove the different block lengths in ns_core
>
> This CL has bit-exact output.
>
> What it does:
> * Remove the blockLen10Ms, as it is hardcoded to be equal to blockLen.
> * This makes outLen to be always zero, so it can be removed too.
> * It also avoids the need to have an outBuf, because it is not used, so it is also removed
> * Replaced blockLen10Ms by blockLen everywhere, since they were hardcoded to be equal.
> * We don't need to check if outLen is zero, because it always is, so it was removed.
> * Of course, the outBuf needs no initial set or copying around, because it is not used.
>
> BUG=webrtc:3811
> R=bjornv@webrtc.org , kwiberg@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/30539004
TBR=aluebs@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26629004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7306 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-26 14:33:08 +00:00
bfacaabfce
Add accessors for array of channel pointers in AudioBuffer. They are
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needed as arguments to any multichannel audio processing unit.
R=andrew@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30499004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7303 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-25 20:52:08 +00:00
60fbd65482
Removing error triggered for disabling FEC on non-opus
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A failure was triggered when one sets FEC status on a codec that does not support FEC. While it is definitely logical when one wants to enable it, it makes no good sense if one tries to disable it.
BUG=
R=tina.legrand@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7298 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-25 14:36:30 +00:00
5f3965783b
Remove the different block lengths in ns_core
...
This CL has bit-exact output.
What it does:
* Remove the blockLen10Ms, as it is hardcoded to be equal to blockLen.
* This makes outLen to be always zero, so it can be removed too.
* It also avoids the need to have an outBuf, because it is not used, so it is also removed
* Replaced blockLen10Ms by blockLen everywhere, since they were hardcoded to be equal.
* We don't need to check if outLen is zero, because it always is, so it was removed.
* Of course, the outBuf needs no initial set or copying around, because it is not used.
BUG=webrtc:3811
R=bjornv@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7297 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-25 13:53:43 +00:00
741711a861
Revert r7049/r7123, which added unnecessary "u"s to "return 0"s.
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r7049 added some unnecessary casts ("return 0" -> "return static_cast<uint16_t>(0)"). r7123 converted these to "return 0u". The original impetus for this was to eliminate type conversion warnings. However, the 'u's are unnecessary; Visual Studio can return "0" from a function returning an unsigned value without producing a warning. The real reason for the original warnings was that the code was returning -1 from a function returning an unsigned value, which does need a cast; the -1s were eliminated before the above two revisions landed.
Also reverse the order of some conditionals to prevent possible underflow.
While the underflow wouldn't have changed the behavior of the code, it's easier
to reason about the code when such underflow can't happen, and possibly safer
against future modifications as well.
BUG=3663
TEST=none
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22599004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7296 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-25 07:38:14 +00:00
