Commit Graph

506 Commits

Author SHA1 Message Date
4bc66fc387 Fix data race in AMP.
R=aluebs@webrtc.org, aluebs-webrtc
BUG=516637

Review URL: https://codereview.webrtc.org/1279953003 .

Cr-Commit-Position: refs/heads/master@{#9694}
2015-08-10 22:26:43 +00:00
8381b37488 Removed bjornv from OWNERS and added two new owners
BUG=

Review URL: https://codereview.webrtc.org/1272603002

Cr-Commit-Position: refs/heads/master@{#9685}
2015-08-06 13:25:37 +00:00
ee66016930 Added IsInBeam to mock_nonlinear_beamformer.h
Review URL: https://codereview.webrtc.org/1262853003

Cr-Commit-Position: refs/heads/master@{#9655}
2015-07-30 00:24:42 +00:00
b3b79b6115 Clean up the Config to enable 48kHz support in AudioProcessing
Now 48kHz is enabled by default.

BUG=webrtc:3146

Review URL: https://codereview.webrtc.org/1233393003

Cr-Commit-Position: refs/heads/master@{#9643}
2015-07-27 17:18:05 +00:00
3ab2f14d56 Remove C++11 calls from intelligibility_utils
The C++11 here was overkill. This replaces it with simpler logic that
covers all cases encountered so far in practice.

The problem was previously brought up here: https://codereview.webrtc.org/1250663007/

BUG=427718, 487341, webrtc:4866
R=andrew@webrtc.org

Review URL: https://codereview.webrtc.org/1242943008

Cr-Commit-Position: refs/heads/master@{#9627}
2015-07-23 19:15:32 +00:00
86c6d33aec Allow more than 2 input channels in AudioProcessing.
The number of output channels is constrained to be equal to either 1 or the
number of input channels.

An earlier version of this commit caused a crash on AEC dump.

TBR=aluebs@webrtc.org,pbos@webrtc.org

Review URL: https://codereview.webrtc.org/1248393003 .

Cr-Commit-Position: refs/heads/master@{#9626}
2015-07-23 18:41:45 +00:00
64e753c399 Revert of Allow more than 2 input channels in AudioProcessing. (patchset #13 id:240001 of https://codereview.webrtc.org/1226093007/)
Reason for revert:
Breaks Chromium FYI content_browsertest on all platforms. The testcase that fails is WebRtcAecDumpBrowserTest.CallWithAecDump.

https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux/builds/19388

Sample output:
[ RUN      ] WebRtcAecDumpBrowserTest.CallWithAecDump
Xlib:  extension "RANDR" missing on display ":9".
[4:14:0722/211548:1282124453:WARNING:webrtcvoiceengine.cc(472)] Unexpected codec: ISAC/48000/1 (105)
[4:14:0722/211548:1282124593:WARNING:webrtcvoiceengine.cc(472)] Unexpected codec: PCMU/8000/2 (110)
[4:14:0722/211548:1282124700:WARNING:webrtcvoiceengine.cc(472)] Unexpected codec: PCMA/8000/2 (118)
[4:14:0722/211548:1282124815:WARNING:webrtcvoiceengine.cc(472)] Unexpected codec: G722/8000/2 (119)
[19745:19745:0722/211548:1282133667:INFO:CONSOLE(64)] "Looking at video in element remote-view-1", source: http://127.0.0.1:48819/media/webrtc_test_utilities.js (64)
[19745:19745:0722/211548:1282136892:INFO:CONSOLE(64)] "Looking at video in element remote-view-2", source: http://127.0.0.1:48819/media/webrtc_test_utilities.js (64)
../../content/test/webrtc_content_browsertest_base.cc:62: Failure
Value of: ExecuteScriptAndExtractString( shell()->web_contents(), javascript, &result)
  Actual: false
Expected: true
Failed to execute javascript call({video: true, audio: true});.
From javascript: (nothing)
When executing 'call({video: true, audio: true});'
../../content/test/webrtc_content_browsertest_base.cc:75: Failure
Failed
../../content/browser/media/webrtc_aecdump_browsertest.cc:26: Failure
Expected: (base::kNullProcessId) != (*id), actual: 0 vs 0
../../content/browser/media/webrtc_aecdump_browsertest.cc:95: Failure
Value of: GetRenderProcessHostId(&render_process_id)
  Actual: false
Expected: true
../../content/browser/media/webrtc_aecdump_browsertest.cc:99: Failure
Value of: base::PathExists(dump_file)
  Actual: false
Expected: true
../../content/browser/media/webrtc_aecdump_browsertest.cc:101: Failure
Value of: base::GetFileSize(dump_file, &file_size)
  Actual: false
Expected: true
../../content/browser/media/webrtc_aecdump_browsertest.cc:102: Failure
Expected: (file_size) > (0), actual: 0 vs 0
[  FAILED  ] WebRtcAecDumpBrowserTest.CallWithAecDump, where TypeParam =  and GetParam() =  (361 ms)

Original issue's description:
> Allow more than 2 input channels in AudioProcessing.
>
> The number of output channels is constrained to be equal to either 1 or the
> number of input channels.
>
> R=aluebs@webrtc.org, andrew@webrtc.org, pbos@webrtc.org
>
> Committed: c204754b7a

TBR=andrew@webrtc.org,aluebs@webrtc.org,ajm@chromium.org,pbos@chromium.org,pbos@webrtc.org,mgraczyk@chromium.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review URL: https://codereview.webrtc.org/1253573005

Cr-Commit-Position: refs/heads/master@{#9621}
2015-07-23 11:30:14 +00:00
c204754b7a Allow more than 2 input channels in AudioProcessing.
The number of output channels is constrained to be equal to either 1 or the
number of input channels.

R=aluebs@webrtc.org, andrew@webrtc.org, pbos@webrtc.org

Review URL: https://codereview.webrtc.org/1226093007 .

Cr-Commit-Position: refs/heads/master@{#9619}
2015-07-23 04:06:16 +00:00
b297c5a01f Miscellaneous changes split from https://codereview.webrtc.org/1230503003 .
These are mostly trivial changes and are separated out just to reduce the
diff on that change to the minimum possible.

Note explanatory comments on patch set 1.

BUG=none
TEST=none

Review URL: https://codereview.webrtc.org/1235643003

Cr-Commit-Position: refs/heads/master@{#9617}
2015-07-22 22:17:26 +00:00
7c5304c791 Allow webrtc compilation with stlport
Android has not yet finalized its libc++ build. Allow compilation with
stlport by removing several C++11 library usages.

BUG=427718,487341,webrtc:4866
R=andrew@webrtc.org

Review URL: https://codereview.webrtc.org/1250663007 .

Patch from Jared Duke <jdduke@chromium.org>.

Cr-Commit-Position: refs/heads/master@{#9616}
2015-07-22 20:04:30 +00:00
2981945bcf Moved arrray_util include to beamformer.h
Review URL: https://codereview.webrtc.org/1244813003

Cr-Commit-Position: refs/heads/master@{#9604}
2015-07-20 20:22:27 +00:00
ekm
35b72fbceb Add new variance update option and unittests for intelligibility
- New option for computing variance that is more adaptive with lower complexity.
- Fixed related off-by-one errors.
- Added intelligibility unittests.
- Do not enhance if experiencing variance underflow.

R=andrew@webrtc.org, henrik.lundin@webrtc.org

Review URL: https://codereview.webrtc.org/1207353002 .

Cr-Commit-Position: refs/heads/master@{#9567}
2015-07-10 21:11:57 +00:00
bb36fdf95f Remove empty-string comparisons.
Use .empty() and !.empty() in favor of == "" or != "".

BUG=
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1228913003

Cr-Commit-Position: refs/heads/master@{#9559}
2015-07-09 14:48:27 +00:00
cbd44e6d73 Use Resampler default constructor in VAD
Review URL: https://codereview.webrtc.org/1224693013

Cr-Commit-Position: refs/heads/master@{#9551}
2015-07-08 03:21:58 +00:00
4e7aa43ea0 audio_processing: Adds two UMA histograms logging delay jumps in AEC
We have two histograms today that trigger on large jumps in either platform reported stream delays (WebRTC.Audio.PlatformReportedStreamDelayJump) or the system delay in the AEC (WebRTC.Audio.AecSystemDelayJump). The latter is the internal buffer size in the AEC.
The sizes of such jumps are of relevance since it can harm the AEC and even put it in a complete failure state. It is hard, not to say impossible, to tell how frequent it is.
Therefore, two complementary histograms are added; number of jumps in each metric.
This way we get a quick way to determine how often a jump occurs in general and also how frequent it is within a call.

This is solved by adding a counter for each metric.
The counter is activated either upon an event trigger or if we know for sure when the AEC is running.
Unfortunately, we can't rely on the destructor at the end of a call so we add a public API for the user to take on the action of calling it at the end of a call.

Tested locally by building ToT chromium including changes and three triggered jumps (200, 50 and 60 ms).
The stats picked up the 60 and 200 ms jumps as expected.

BUG=488124
R=asapersson@webrtc.org, pbos@webrtc.org

Review URL: https://codereview.webrtc.org/1229443003.

Cr-Commit-Position: refs/heads/master@{#9544}
2015-07-07 09:50:16 +00:00
d92f2674d7 audio_processing: Changed kMinDiffDelayMs from 50 to 60 ms
The UMA histograms WebRTC.Audio.AecSystemDelayJump and WebRTC.Audio.PlatformReportedStreamDelayJump triggers if the jump is larger than kMinDiffDelayMs.
Especially WebRTC.Audio.AecSystemDelayJump is sensitive around 50 ms differences, since the granularity is 4 ms and we can get a significant amount of hits at 52 ms.
Therefore, a change to 60 ms can make the logging more robust. The effect of not logging jumps in the interval 50-60 ms is of minor importance since they are not likely to affect the AEC performance. It's when we get values from ~100 ms and above that we should be worried.

Tested with a local ToT Chromium build where 52, 64 and 200 ms jumps were forced.

BUG=488124
TBR=henrik.lundin@webrtc.org

Review URL: https://codereview.webrtc.org/1208313003.

Cr-Commit-Position: refs/heads/master@{#9540}
2015-07-05 08:46:10 +00:00
894ad94302 Fix occurrences of const typed declaration without initialization
This fixes compilation errors as the following:

error: constructor must explicitly initialize the const member

BUG=506663
R=aluebs@webrtc.org, tommi@webrtc.org

Signed-off-by: Eduardo Lima (Etrunko) <eduardo.lima@intel.com>

Review URL: https://codereview.webrtc.org/1222233002

Cr-Commit-Position: refs/heads/master@{#9538}
2015-07-03 15:34:40 +00:00
366e95252a Follow-up: Remove old ReportedDelay AEC config
This is a follow-up to r9531, where the configuration ReportedDelay
was replaced by DelayAgnostic. The config was kept in the code to
avoid API breakages. In https://codereview.chromium.org/1219263003/
depending code has been updated to avoid breakages.

BUG=webrtc:4651
R=bjornv@webrtc.org

Review URL: https://codereview.webrtc.org/1212653012

Cr-Commit-Position: refs/heads/master@{#9536}
2015-07-03 07:50:13 +00:00
0f133b99c6 Rename APM Config ReportedDelay to DelayAgnostic
We use this Config struct for enabling/disabling the delay agnostic
AEC. This change renames it to DelayAgnostic for readability reasons.

NOTE: The logic is reversed in this CL. The old ReportedDelay config
turned DA-AEC off, while the new DelayAgnostic turns it on.

The old Config is kept in parallel with the new during a transition
period. This is to avoid problems with API breakages. During this
period, ReportedDelay is disabled or DelayAgnostic is enabled, DA-AEC
is engaged in APM.

BUG=webrtc:4651
R=bjornv@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1211053006

Cr-Commit-Position: refs/heads/master@{#9531}
2015-07-02 07:17:59 +00:00
1adbacb19d Adding method IsInBeam to beamformer class.
This was previously reviewed at:
https://webrtc-codereview.appspot.com/53729004/

Review URL: https://codereview.webrtc.org/1211613005

Cr-Commit-Position: refs/heads/master@{#9517}
2015-06-29 23:15:23 +00:00
1ca324f237 Adds UMA histogram for system delay jumps
Sudden platform system delay jumps can hurt AEC and we have no stats that monitor these jumps. How often do they occur, and when they are reported are they accurate?

This CL logs all jumps in both the reported and actual delay.

The histogram has been tested with a chromium build where a fake jump of 200 ms was applied after 5 seconds and it was registered correctly in chrome://histograms

BUG=488124
R=henrik.lundin@webrtc.org, peah@webrtc.org

Review URL: https://codereview.webrtc.org/1213733004.

Cr-Commit-Position: refs/heads/master@{#9513}
2015-06-29 12:57:42 +00:00
ebe7422372 Created SphericalPoint in array_util.h
Review URL: https://codereview.webrtc.org/1211703002

Cr-Commit-Position: refs/heads/master@{#9507}
2015-06-25 21:24:11 +00:00
ecf6b81644 Pull the Voice Activity Detector out from the AGC
This change generates bit-exact values when running through audioproc_f than before.

This change was originally uploaded here:
* https://codereview.webrtc.org/1181933002/
* https://codereview.webrtc.org/1177043017/

And reverted because of an ASAN problem in Chrome here:
* https://codereview.webrtc.org/1192863006/
* https://codereview.webrtc.org/1194963003/

TBR=andrew@webrtc.org

Review URL: https://codereview.webrtc.org/1212543002

Cr-Commit-Position: refs/heads/master@{#9505}
2015-06-25 19:28:55 +00:00
51c7cbb86a Revert "Pull the Voice Activity Detector out from the AGC"
This reverts commit 518c683f3e413523a458a94b533274bd7f29992d.

Breaks Linux-Asan bot
https://uberchromegw.corp.google.com/i/client.webrtc/builders/Linux%20Asan/builds/4348/steps/libjingle_peerconnection_unittest/logs/stdio

BUG=
TBR=aluebs@webrtc.org

Review URL: https://codereview.webrtc.org/1208793002.

Cr-Commit-Position: refs/heads/master@{#9503}
2015-06-25 06:46:14 +00:00
518c683f3e Pull the Voice Activity Detector out from the AGC
This change generates bit-exact values when running through audioproc_f than before.

This change was originally uploaded here:
* https://codereview.webrtc.org/1181933002/
* https://codereview.webrtc.org/1177043017/

And reverted because of an ASAN problem in Chrome here:
* https://codereview.webrtc.org/1192863006/
* https://codereview.webrtc.org/1194963003/

TBR=andrew@webrtc.org

Review URL: https://codereview.webrtc.org/1211563003

Cr-Commit-Position: refs/heads/master@{#9502}
2015-06-25 01:46:03 +00:00
ac4234ccfc Add a [rtc_]build_with_neon variable to unify conditions.
Also consolidate ARM options for gn in an arm_neon_config.

R=jridges@masque.com, kjellander@webrtc.org, zhongwei.yao@chromium.org

Review URL: https://codereview.webrtc.org/1181373004.

Cr-Commit-Position: refs/heads/master@{#9501}
2015-06-25 01:25:59 +00:00
1c7075f076 Ensure transient suppression is never enabled on mobile.
Review URL: https://codereview.webrtc.org/1209653002

Cr-Commit-Position: refs/heads/master@{#9500}
2015-06-25 01:14:17 +00:00
ekm
db4fecfb01 Attempt to reland: Allow intelligibility to compile in apm (https://codereview.webrtc.org/1182323005/)
Revert of original: https://codereview.webrtc.org/1187033005/

Changes in original:
- Added files to gyp and BUILD
- Made minor fixes to get everything to compile
    and intelligibility_proc to run
- Added comments
- Auto-reformatting

New Changes:
- Added <numeric> header to intelligibility_enhancer.cc to address buildbot errors
- Switched to use WAV for i/o in intelligibility_proc.cc to address windows errors
- clean up

Note: Patch 1 duplicates Patch 7 of https://codereview.webrtc.org/1182323005/

R=andrew@webrtc.org

Review URL: https://codereview.webrtc.org/1190733004.

Cr-Commit-Position: refs/heads/master@{#9486}
2015-06-23 00:49:14 +00:00
f260fc2136 Revert "Pull the Voice Activity Detector out from the AGC"
This reverts commit 34be126c1b3ee60ecdb86b1de41a0648347450b2.

It breaks Chromium ASAN.

TBR=niklas.enbom@webrtc.org

Review URL: https://codereview.webrtc.org/1192863006.

Cr-Commit-Position: refs/heads/master@{#9472}
2015-06-19 18:24:01 +00:00
f5f8f52a4b Revert "Increase the kMaxNoiseProbability in voice_activity_detector_test"
This reverts commit c9b0f675687d318b9367b1d6764182b9411355de.

It breaks Chromium ASAN.

TBR=niklas.enbom@webrtc.org
BUG=

Review URL: https://codereview.webrtc.org/1194963003.

Cr-Commit-Position: refs/heads/master@{#9471}
2015-06-19 18:18:02 +00:00
c9b0f67568 Increase the kMaxNoiseProbability in voice_activity_detector_test
Because it breaks on Android.

TBR=ajm

Review URL: https://codereview.webrtc.org/1177043017.

Cr-Commit-Position: refs/heads/master@{#9467}
2015-06-18 21:48:09 +00:00
34be126c1b Pull the Voice Activity Detector out from the AGC
This change generates bit-exact values when running through audioproc_f than before.

R=andrew@webrtc.org, bloch@google.com

Review URL: https://codereview.webrtc.org/1181933002.

Cr-Commit-Position: refs/heads/master@{#9465}
2015-06-18 19:34:00 +00:00
7101269c61 Reland "Revert "audio_processing/aec: make delay estimator aware of starving farend buffer""
Original review at https://codereview.webrtc.org/1180423006

SystemDelayTests was not updated w.r.t. extended_filter mode and some tests were disabled on Android since DA-AEC is automatically set.
All tests have now been updated for both extended_filter mode as well as DA-AEC, hence are now enabled on Android.

Also
* Moves default settings of extended_filter and DA-AEC form Init() to Create() to avoid unintentional loss of state during a reset.
* Fixes a potential bug of starting from scratch in extended_filter mode + DA-AEC.

This reverts commit 01c9b012e9171c813ace9e405c32fc75f4262bf6.

BUG=
R=henrik.lundin@webrtc.org

Review URL: https://codereview.webrtc.org/1187943005.

Cr-Commit-Position: refs/heads/master@{#9458}
2015-06-18 09:05:03 +00:00
c555b99c13 Revert of Allow intelligibility to compile in apm (patchset #1 id:1 of https://codereview.webrtc.org/1182323005/)
Reason for revert:
Breaking the build bots: http://build.chromium.org/p/client.webrtc/builders/Mac32%20Release%20%5Blarge%20tests%5D/builds/4544

Fails to compile with this error:

../../webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.cc:218:25: error: no member named 'accumulate' in namespace 'std'
    power_target = std::accumulate(clear_variance_.variance(),

Original issue's description:
> Allow intelligibility to compile in apm
>
> - Added files to gyp and BUILD
> - Made minor fixes to get everything to compile
>     and intelligibility_proc to run
> - Added comments
> - Auto-reformatting
>
> Original cl is at: https://webrtc-codereview.appspot.com/57579004/
>
> TBR=aluebs@webrtc.org
>
> Committed: b7553dfdbb

TBR=ekmeyerson@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review URL: https://codereview.webrtc.org/1187033005

Cr-Commit-Position: refs/heads/master@{#9455}
2015-06-17 03:26:20 +00:00
ekm
b7553dfdbb Allow intelligibility to compile in apm
- Added files to gyp and BUILD
- Made minor fixes to get everything to compile
    and intelligibility_proc to run
- Added comments
- Auto-reformatting

Original cl is at: https://webrtc-codereview.appspot.com/57579004/

TBR=aluebs@webrtc.org

Review URL: https://codereview.webrtc.org/1182323005.

Cr-Commit-Position: refs/heads/master@{#9454}
2015-06-17 01:57:37 +00:00
01c9b012e9 Revert "audio_processing/aec: make delay estimator aware of starving farend buffer"
The code only affects DA-AEC, but since DA-AEC is the default AEC if run on Android tests failed. Reverting to fix that test.

This reverts commit 9002cc426dab7a576f5247f45ba888cd081a39f0.

BUG=
TBR=henrik.lundin@webrtc.org

Review URL: https://codereview.webrtc.org/1183243003.

Cr-Commit-Position: refs/heads/master@{#9453}
2015-06-16 21:09:51 +00:00
9002cc426d audio_processing/aec: make delay estimator aware of starving farend buffer
We've seen that if we get a buffer underrun followed by a sudden buffer build up the DA-AEC can't really catch up even though it should be possible to estimate the upcoming difference. We have a feature for this already, but that is only used in the regular AEC. This CL turns that feature on also for DA-AEC.

- Adds a helper function MoveFarReadPtrWithoutSystemDelayUpdate()
- Only apply conservative correction for positive delays, where we can put the AEC into a non-causal state
- Stuff the farend buffer if we don't have enough data to process w.r.t. to current nearend buffer.
- Always run delay estimation based on reported delays to catch buffer starvation.

BUG=
R=henrik.lundin@webrtc.org

Review URL: https://codereview.webrtc.org/1180423006.

Cr-Commit-Position: refs/heads/master@{#9452}
2015-06-16 20:29:52 +00:00
1d34fe979c Adds support for webrtc::test::ResourcePath on iOS
BUG=webrtc:4752
R=tkchin@webrtc.org

Review URL: https://codereview.webrtc.org/1178843002.

Cr-Commit-Position: refs/heads/master@{#9445}
2015-06-16 08:04:24 +00:00
b02af18c5c Follow-up: Remove old DelayCorrection AEC config
This is a follow-up to r9401, where the configuration DelayCorrection
was replaced by ExtendedFilter.

This change also removes the media constraint
kExperimentalEchoCancellation which was replaced by
kExtendedFilterEchoCancellation in the same CL.

Both settings that are now being removed were kept in the code to avoid
API breakages. In https://codereview.chromium.org/1167343004,
depending code has been updated to avoid breakages.

BUG=webrtc:4696
R=bjornv@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1181413004.

Cr-Commit-Position: refs/heads/master@{#9444}
2015-06-16 07:53:32 +00:00
ekm
030249dd24 Initial SIE commit: migrating existing code
Moved exact existing intelligibility enhancement implementation into new
repository for reference when making further changes.

Note: this cl does not add these files to any gyp.

Original cl is at https://webrtc-codereview.appspot.com/52719004/ .

TBR=aluebs@webrtc.org

Review URL: https://codereview.webrtc.org/1177953006.

Cr-Commit-Position: refs/heads/master@{#9441}
2015-06-15 20:02:33 +00:00
728d9037c0 Reformat existing code. There should be no functional effects.
This includes changes like:
* Attempt to break lines at better positions
* Use "override" in more places, don't use "virtual" with it
* Use {} where the body is more than one line
* Make declaration and definition arg names match
* Eliminate unused code
* EXPECT_EQ(expected, actual) (but use (actual, expected) for e.g. _GT)
* Correct #include order
* Use anonymous namespaces in preference to "static" for file-scoping
* Eliminate unnecessary casts
* Update reference code in comments of ARM assembly sources to match actual current C code
* Fix indenting to be more style-guide compliant
* Use arraysize() in more places
* Use bool instead of int for "boolean" values (0/1)
* Shorten and simplify code
* Spaces around operators
* 80 column limit
* Use const more consistently
* Space goes after '*' in type name, not before
* Remove unnecessary return values
* Use "(var == const)", not "(const == var)"
* Spelling
* Prefer true, typed constants to "enum hack" constants
* Avoid "virtual" on non-overridden functions
* ASSERT(x == y) -> ASSERT_EQ(y, x)

BUG=none
R=andrew@webrtc.org, asapersson@webrtc.org, henrika@webrtc.org, juberti@webrtc.org, kjellander@webrtc.org, kwiberg@webrtc.org

Review URL: https://codereview.webrtc.org/1172163004

Cr-Commit-Position: refs/heads/master@{#9420}
2015-06-11 21:31:48 +00:00
b7e5054414 Match existing type usage better.
This makes a variety of small changes to synchronize bits of code using different types, remove useless code or casts, and add explicit casts in some places previously doing implicit ones.  For example:

* Change a few type declarations to better match how the majority of code uses those objects.
* Eliminate "< 0" check for unsigned values.
* Replace "(float)sin(x)", where |x| is also a float, with "sinf(x)", and similar.
* Add casts to uint32_t in many places timestamps were used and the existing code stored signed values into the unsigned objects.
* Remove downcasts when the results would be passed to a larger type, e.g. calling "foo((int16_t)x)" with an int |x| when foo() takes an int instead of an int16_t.
* Similarly, add casts when passing a larger type to a function taking a smaller one.
* Add casts to int16_t when doing something like "int16_t = int16_t + int16_t" as the "+" operation would implicitly upconvert to int, and similar.
* Use "false" instead of "0" for setting a bool.
* Shift a few temp types when doing a multi-stage calculation involving typecasts, so as to put the most logical/semantically correct type possible into the temps.  For example, when doing "int foo = int + int; size_t bar = (size_t)foo + size_t;", we might change |foo| to a size_t and move the cast if it makes more sense for |foo| to be represented as a size_t.

BUG=none
R=andrew@webrtc.org, asapersson@webrtc.org, henrika@webrtc.org, juberti@webrtc.org, kwiberg@webrtc.org
TBR=andrew, asapersson, henrika

Review URL: https://codereview.webrtc.org/1168753002

Cr-Commit-Position: refs/heads/master@{#9419}
2015-06-11 19:56:03 +00:00
f045e4da43 Prepare to convert various types to size_t.
This makes some behaviorally-invariant changes to make certain code that
currently only works correctly with signed types work safely regardless of the
signedness of the types in question.  This is preparation for a future change
that will convert a variety of types to size_t.

There are also some formatting changes (e.g. converting "enum hack" usage to real consts) to make it simpler to just change "int" to "size_t" in the future to change the types of those constants.

BUG=none
R=andrew@webrtc.org, juberti@webrtc.org, kwiberg@webrtc.org
TBR=ajm

Review URL: https://codereview.webrtc.org/1174813003

Cr-Commit-Position: refs/heads/master@{#9413}
2015-06-11 04:15:51 +00:00
786dbdcc38 Rename targets to use lower case format.
It makes writing a build script for merging libraries
across architectures easier. See talk/build/build_ios_libs.sh.

BUG=
R=andrew@webrtc.org, kjellander@webrtc.org

Review URL: https://codereview.webrtc.org/1171793002.

Cr-Commit-Position: refs/heads/master@{#9412}
2015-06-10 20:45:12 +00:00
9345e86551 audio_processing: Create now returns a pointer to the object
Affects
* NS
* AGC
* AEC

BUG=441
TESTED=locally on Linux and trybots
R=kwiberg@webrtc.org

Review URL: https://codereview.webrtc.org/1175903002.

Cr-Commit-Position: refs/heads/master@{#9411}
2015-06-10 19:43:46 +00:00
a2c79405b4 Ensures that modules_unittests runs on iOS
BUG=4752
R=tkchin@chromium.org

Review URL: https://codereview.webrtc.org/1171033002.

Cr-Commit-Position: refs/heads/master@{#9408}
2015-06-10 11:24:58 +00:00
441f634731 Re-land r9378 "Rename APM Config DelayCorrection to ExtendedFilter"
(This reverts commit 3fbf3f8841b5460503fb646eaedcb063620434a8.)

The original submission was reverted because it broke the Chrome build. This is fixed in patch set 2 of this change by keeping the old MediaConstraintsInterface string kExperimentalEchoCancellation. It will be removed once the Chrome code has been updated.

Original description:
"We use this Config struct for enabling/disabling Extended filter mode in AEC. This change renames it to ExtendedFilter for readability reasons. The corresponding media constraint is also renamed to kExtendedFilterEchoCancellation.

The old Config is kept in parallel with the new during a transition period. This is to avoid problems with API breakages. During this period, if any of the two Configs are enabled, the extended filter mode is engaged in APM. That is, the two Configs are combined with an "OR" operation.

This change also renames experimental_aec in AudioOptions to extended_filter_aec."

BUG=webrtc:4696
R=bjornv@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1151573021.

Cr-Commit-Position: refs/heads/master@{#9401}
2015-06-09 14:03:23 +00:00
3fbf3f8841 Revert r9378 "Rename APM Config DelayCorrection to ExtendedFilter"
This reverts commit 5f4b7e2873864c61e2ad6d88679dcd5d321bfd16, since it
broke some of the build bots.

BUG=4696
TBR=bjornv@webrtc.org

Review URL: https://codereview.webrtc.org/1166463006

Cr-Commit-Position: refs/heads/master@{#9380}
2015-06-05 09:04:20 +00:00
5f4b7e2873 Rename APM Config DelayCorrection to ExtendedFilter
We use this Config struct for enabling/disabling Extended filter mode
in AEC. This change renames it to ExtendedFilter for readability
reasons. The corresponding media constraint is also renamed to
kExtendedFilterEchoCancellation.

The old Config is kept in parallel with the new during a transition
period. This is to avoid problems with API breakages. During this
period, if any of the two Configs are enabled, the extended filter
mode is engaged in APM. That is, the two Configs are combined with an
"OR" operation.

This change also renames experimental_aec in AudioOptions to extended_filter_aec.

BUG=4696
R=bjornv@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/54659004

Cr-Commit-Position: refs/heads/master@{#9378}
2015-06-05 07:55:40 +00:00
7dbc076f34 audio_processing/aec: Turn SignalBasedDelayCorrection to after 15 seconds
The delay agnostic AEC uses a signal based delay correction method to adjust buffer synchronization between loudspeaker and microphone. On Mac in particular we have seen deviations in UMA stats that point towards an echo already at startup. This is likely due to an early and incorrect correction based on poor audio data.
By waiting 15 seconds before we turn on the ability to correct we can avoid a majority of these.
The reported delay values are in general accurate enough and relying on them in the beginning is fine. The value 15 seconds is chosen because we have seen from UMA data that a significant amount of calls tend to end before 15 seconds when being in the UseDelayAgnosticAEC Finch experiment.

We turn this "feature" on for all platforms but Android, where the reported system delays are inaccurate and we want to take action as soon as possible.
In addition, the set of "good" delay values has been increased from 25% to 75% of the filter length.

BUG=webrtc:3504
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50199004

Cr-Commit-Position: refs/heads/master@{#9376}
2015-06-05 07:40:45 +00:00