171 Commits

Author SHA1 Message Date
peah
9e69abf85e Added logging using the raw variant of the new aec logging macros
Replaced the wav file dumping functionality in aec_core.c with the newly added corresponding macros

Added macros for logging of AEC internal data

BUG=

Review URL: https://codereview.webrtc.org/1272403003

Cr-Commit-Position: refs/heads/master@{#9808}
2015-08-28 11:41:30 +00:00
Peter Kasting
dce40cf804 Update a ton of audio code to use size_t more correctly and in general reduce
use of int16_t/uint16_t.

This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects.

This was be reviewed and approved in pieces:
https://codereview.webrtc.org/1224093003
https://codereview.webrtc.org/1224123002
https://codereview.webrtc.org/1224163002
https://codereview.webrtc.org/1225133003
https://codereview.webrtc.org/1225173002
https://codereview.webrtc.org/1227163003
https://codereview.webrtc.org/1227203003
https://codereview.webrtc.org/1227213002
https://codereview.webrtc.org/1227893002
https://codereview.webrtc.org/1228793004
https://codereview.webrtc.org/1228803003
https://codereview.webrtc.org/1228823002
https://codereview.webrtc.org/1228823003
https://codereview.webrtc.org/1228843002
https://codereview.webrtc.org/1230693002
https://codereview.webrtc.org/1231713002

The change is being landed as TBR to all the folks who reviewed the above.

BUG=chromium:81439
TEST=none
R=andrew@webrtc.org, pbos@webrtc.org
TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher

Review URL: https://codereview.webrtc.org/1230503003 .

Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 21:52:45 +00:00
henrik.lundin
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
Bjorn Volcker
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
Bjorn Volcker
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
Bjorn Volcker
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
Peter Kasting
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
Bjorn Volcker
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
Henrik Lundin
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
Henrik Lundin
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
Henrik Lundin
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
Bjorn Volcker
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
Andrew MacDonald
cb7f8ce2df Clear ARM NEON flag
Merge WEBRTC_ARCH_ARM64_NEON and WEBRTC_ARCH_ARM_NEON into one
WEBRTC_HAS_NEON.
Replace WEBRTC_DETECT_ARM_NEON by WEBRTC_DETECT_NEON.
Replace WEBRTC_ARCH_ARM by WEBRTC_ARCH_ARM64 for arm64 cpu.

BUG=4002
R=andrew@webrtc.org, jridges@masque.com, kjellander@webrtc.org

Change-Id: I870a4d0682b80633b671c9aab733153f6d95a980

Review URL: https://webrtc-codereview.appspot.com/49309004

Cr-Commit-Position: refs/heads/master@{#9228}
2015-05-20 05:20:04 +00:00
Bjorn Volcker
1ff218fac3 audio_processing/aec: Do not scale target delay at startup when on Android
When running AEC in extended_filter mode there is no startup phase to evaluate the reported system delay values.
Instead we simply use the first value and scale by two to avoid over compensating when synchronizing render and capture.
We don't need to be too accurate since we have extended the filter length.

On Android we use fixed (measured) reported delay values.
There is no need to be extra conservative here, because that is already built-in in the measured value.
In fact, the difference between devices is large and with such an extra conservative approach the true delay can not be caught by the filter length.
With this change we can improve performance on some devices.

BUG=4472
TESTED=offline on recordings from various devices
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49909004

Cr-Commit-Position: refs/heads/master@{#9144}
2015-05-06 10:08:50 +00:00
Bjorn Volcker
3cfa756f37 audio_processing/aec: Fixes an incorrect sampling rate multiplier when processing in 48 kHz
In AEC a fixed fft size is used, but processing can in the lower band be in either 8 or 16 kHz.
Therefore we need a multiplier/rate factor to, for example, map frequency bands in Hz to frequency bins.

The multiplier/rate factor can only be either 1 or 2, but when 48 kHz support was added it was assigned 3.

BUG=crbug.com/482424
R=aluebs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43329004

Cr-Commit-Position: refs/heads/master@{#9117}
2015-04-29 18:22:50 +00:00
Bjorn Volcker
f6a99e63b6 Refactor audio_processing: Free functions return void
There is no point in returning an error when Free() fails. In fact it can only happen if we have a null pointer as object. There is further no place where the return value is used.

Affected components are
- aec
- aecm
- agc
- ns

BUG=441
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50579004

Cr-Commit-Position: refs/heads/master@{#8966}
2015-04-10 05:56:59 +00:00
Bjorn Volcker
bf395c1fc0 Add WebRTC Media Constraint to force using Delay Agnostic AEC on Android
If built-in Echo Cancellation is available on a device it is automatically enabled. The reason is that it in most cases performs better than the WebRTC software echo control for mobile. The drawback is that we can not develop, test and rollout the delay agnostic AEC (DA-AEC) on Android as for desktops.

This CL includes
- adding a media constraint to enable/disable DA-AEC.
- automatically turning on echo cancellation if DA-AEC is enabled.
- a fix in the AEC that enables delay estimation when DA-AEC is enabled, but delay metrics is disabled.
- sets the Config struct ReportedDelay, which controls DA-AEC internally in the AEC.

The test code to verify that it works in AppRTCDemo can be found here:
https://webrtc-codereview.appspot.com/50479004/

BUG=4472
TESTED=locally on N7, N6, Android One
R=glaznev@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48699004

Cr-Commit-Position: refs/heads/master@{#8861}
2015-03-25 21:46:10 +00:00
bjornv@webrtc.org
d7a212e8b9 audio_processing/aec: Increased delay metrics aggregation window to five seconds
The known clients (GetStats and UMA histogram in Chrome) use at least 5 second aggregation window. There is no particular value in calculating the metrics more often.

The CL also includes a small refactoring moving a declaration inside an if statement.

BUG=2994
TEST=N/A
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40219004

Cr-Commit-Position: refs/heads/master@{#8619}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8619 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-05 16:14:58 +00:00
bjornv@webrtc.org
976c0f3043 audio_processing/aec: NEON code should not be invoked if it is detectable, but is not NEON
There exist devices with runtime checks for NEON, but where the device is not NEON. One such device is Tegra2 on which currently NEON code is running.

This fix adds a missing feature check when initializing the AEC.

BUG=4304
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42159004

Cr-Commit-Position: refs/heads/master@{#8559}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8559 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-02 16:25:51 +00:00
bjornv@webrtc.org
cc64a9cc4f voice_engine: Updates GetEcDelayMetrics() w.r.t. new metric
As of r8230 (https://webrtc-codereview.appspot.com/39739004/) a new Echo Delay Metric was added calculating the fraction of poor values that may cause the AEC to fail. There are currently two methods for GetDelayMetrics() in webrtc::AutioProcessing and one is deprecated.

This CL updates
- GetEcDelayMetrics()
- voe_auto_test
- talk/media/(fake)webrtcvoiceengine

BUG=N/A
TESTED=locally and trybots
R=pbos@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41749004

Cr-Commit-Position: refs/heads/master@{#8251}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8251 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-05 12:53:24 +00:00
bjornv@webrtc.org
b1786dbab0 audio_processing: Added a new AEC delay metric value that gives the amount of poor delays
To more easily determine if for example the AEC is not working properly one could monitor how often the estimated delay is out of bounds. With out of bounds we mean either being negative or too large, where both cases will break the AEC.

A new delay metric is added telling the user how often poor delay values were estimated. This is measured in percentage since last time the metrics were calculated.

All APIs have been updated with a third parameter with EchoCancellation::GetDelayMetrics() giving the option to exclude the new metric not to break existing code.

The new metric has been added to audio_processing_unittests with an additional protobuf member, and reference files accordingly updated.
voe_auto_test has not been updated to display the new metric.

BUG=4246
TESTED=audioproc on files
R=aluebs@webrtc.org, andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39739004

Cr-Commit-Position: refs/heads/master@{#8230}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8230 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-03 06:07:21 +00:00
bjornv@webrtc.org
5614cf16e7 audio_processing: Use fixed aggregation window in delay metrics
Previously, the delay estimate history was reset every time the metrics were pulled. This required all clients to be on the same thread and make use of one call.

Now we use a fixed aggregation window of one second and when a client pulls the metrics you get the latest value.
Under certain circumstances like tests you would like to have the aggregation window set to the recording length. We therefore turn on the fixed aggregation window after the first call.

BUG=2994
TESTED=locally on Mac and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38759004

Cr-Commit-Position: refs/heads/master@{#8170}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8170 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-27 18:10:27 +00:00
bjornv@webrtc.org
70117a83d4 AEC: Implements a new function for calculating delay metrics
Two new member variables have been added and the code for calculating the delay metrics have been moved to a function.

BUG=2994
TESTED=locally on Mac and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8163 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-27 11:30:54 +00:00
andrew@webrtc.org
e65d9d974c Fix an unitialized variable warning.
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35819004

Patch from Sebastien Marchand <sebmarchand@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8118 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-21 22:05:12 +00:00
aluebs@webrtc.org
c78d81ae89 Re-land "Support 48kHz in AEC"
Doing something similar for the band 16-24kHz to what is done for the band 8-16kHz. The only difference is that there is no comfort noise added in this band. Could not test how this sounds because there are no aecdumps with 48kHz sample rate as nfar as I know.
Tested for 32kHz sample rate and the output is bitexact with how it was before this CL.

Original: https://webrtc-codereview.appspot.com/28319004/
Reverted: https://webrtc-codereview.appspot.com/33949004/

BUG=webrtc:3146
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8116 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-21 19:10:55 +00:00
tina.legrand@webrtc.org
ee0c100d54 Revert 8080 "Support 48kHz in AEC"
> Support 48kHz in AEC
> 
> Doing something similar for the band 16-24kHz to what is done for the band 8-16kHz. The only difference is that there is no comfort noise added in this band. Could not test how this sounds because there are no aecdumps with 48kHz sample rate as nfar as I know.
> Tested for 32kHz sample rate and the output is bitexact with how it was before this CL.
> 
> BUG=webrtc:3146
> R=andrew@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/28319004

TBR=aluebs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8100 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-20 10:22:49 +00:00
aluebs@webrtc.org
64d3c4b9ac Support 48kHz in AEC
Doing something similar for the band 16-24kHz to what is done for the band 8-16kHz. The only difference is that there is no comfort noise added in this band. Could not test how this sounds because there are no aecdumps with 48kHz sample rate as nfar as I know.
Tested for 32kHz sample rate and the output is bitexact with how it was before this CL.

BUG=webrtc:3146
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8080 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-15 19:52:05 +00:00
andrew@webrtc.org
6b6301588e Move ring_buffer to common_audio.
In preparation for adding a C++ wrapper in common_audio. Also, change
the return type of Init to void and call it from Create.

R=aluebs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8068 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-15 00:09:53 +00:00
bjornv@webrtc.org
bac0012120 Extend delay estimation window in AEC to 500 ms on all platforms
On non-Android the delay estimator in audio_processing/aec has solely been used for logging purposes. The maximum possible observed delay has been 236 ms. We have seen longer delays for which the delay estimate at best ends up at 236 ms, but can also be 'random'. reported delays are clamped to 500 ms.
This cl extends the delay estimation window to match that.

BUG=4086, 3504, 4113
TESTED=locally on Linux and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7989 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-02 09:23:49 +00:00
bjornv@webrtc.org
3a70625caf audio_processing: Added back ATTRIBUTE_UNUSED lost in r7877
BUG=N/A
TESTED=Now it builds with aec_debug_dump=1 on Mac
TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7986 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-01 22:04:12 +00:00
pbos@webrtc.org
f832a6d090 Remove _t from function pointer typedefs.
_t are reserved in POSIX.

R=bjornv@webrtc.org
BUG=162

Review URL: https://webrtc-codereview.appspot.com/34539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7947 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-18 09:56:09 +00:00
pbos@webrtc.org
e468bc9e60 Rename _t struct types in audio_processing.
_t names are reserved in POSIX.

R=bjornv@webrtc.org
BUG=162

Review URL: https://webrtc-codereview.appspot.com/34509005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7943 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-18 09:11:33 +00:00
pbos@webrtc.org
5f162c8509 Merge AEC changes.
R=bjornv@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/34459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7877 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-11 13:46:59 +00:00
andrew@webrtc.org
a56a2c57cf Enabling building with NEON on ARM64
This patch enables NEON on ARM64 platform. Passed building both on
Android ARMv7 and Android ARM64.

BUG=3580
R=andrew@webrtc.org, jridges@masque.com

Review URL: https://webrtc-codereview.appspot.com/25069004

Patch from Zhongwei Yao <zhongwei.yao@arm.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7751 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-26 17:01:40 +00:00
pbos@webrtc.org
f5b56fbc41 Annotate COMPILE_ASSERT with __attribute__((unused)).
Also renames UNUSED -> ATTRIBUTE_UNUSED to be able to use this when
building peerconnection_jni.cc which apparently has this defined to
something else.

R=kjellander@webrtc.org
TBR=mflodman@webrtc.org
BUG=4018

Review URL: https://webrtc-codereview.appspot.com/28039005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7711 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-17 13:47:38 +00:00
andrew@webrtc.org
a3ed713dad Add a WavReader counterpart to WavWriter.
Don't bother with a C interface as we currently have no need to call
this from C code. The first use will be in the audioproc tool.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7585 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31 21:51:03 +00:00
andrew@webrtc.org
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
bjornv@webrtc.org
c75f607042 audio_processing/aec: Ported NEON optimizations of SubbandCoherence() and its sub-functions to SSE2
These optimizations were originally committed in r6860, but reverted in r6861, since it broke a bitexactness test (ApmTest.Process) in modules_unittests. That test has now been updated in r7149, hence this CL now pass the test.

BUG=3767
TESTED=manually on linux and trybots
TBR=aluebs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7189 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-16 05:01:42 +00:00
bjornv@webrtc.org
7bb2586c55 audio_processing: Correct sample rate in aec_debug_dump
When writing to wav files in the low level flag aec_debug_dump incorrect sample rates were used for recordings using rates from 32 kHz and above. This since internally inside the AEC we process the data using 16 kHz. Any upper band is processed and combined later on.

This CL adds the correct sample rate to the recording.

BUG=3359
TESTED=locally on 44.1 kHz recordings on Linux
R=aluebs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7182 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-15 13:23:07 +00:00
andrew@webrtc.org
a22485eaf0 MIPS optimizations for AEC audio processing module
Added new optimizations for MIPS that were removed in r6797.
For more information about this see https://code.google.com/p/webrtc/source/detail?r=6797

R=andrew@webrtc.org, djordje.pesut@imgtec.com

Review URL: https://webrtc-codereview.appspot.com/15259004

Patch from Ljubomir Papuga <ljubomir.papuga@mips.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7010 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-29 17:51:28 +00:00
pbos@webrtc.org
047a46f8b4 Remove Android.mk build files.
These files are generally not maintained and break, some contain files
that don't exist anymore and do not build anymore. If we need to add
some of these back we should really set up a bot for them.

R=andrew@webrtc.org, glaznev@webrtc.org, henrike@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/15249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6974 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-26 08:48:51 +00:00
kwiberg@webrtc.org
584cd8da4b Fix WEBRTC_AEC_DEBUG_DUMP (broken by int16->float conversion)
And in the process, make it dump WAV files instead of raw PCM.

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

Review URL: https://webrtc-codereview.appspot.com/19089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6959 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-25 06:26:04 +00:00
andrew@webrtc.org
71d9572e9c Minor bug fix and cosmetic changes in AEC MIPS optimizations.
Minor bug fix in WebRtcAec_FilterAdaptation_mips, which did not manifest with
gcc 4.7.2, but it did with version 4.9.0. While there, also made some cosmetic
changes to comply with Chromium coding style.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22399004

Patch from Ljubomir Papuga <lpapuga@mips.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6931 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-19 15:42:50 +00:00
andrew@webrtc.org
a1ad844229 Precompute the AEC FFT tables, rather than initializing at run-time.
These global arrays are shared amongst all AEC instances, and were at
serious risk of data races. A Chromium TSAN bot recently caught this.

Also move the function pointer selection for optimization to
create-time. (Ideally this would only be done once.)

BUG=chromium:404133,1503
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6922 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-18 19:02:51 +00:00
bjornv@webrtc.org
820f8e9ca7 modules/audio_processing: Moves declaration of kDelayDiffOffsetSamples
audio_processing did not compile when aec_untrusted_delay_for_testing=1 was set. The constant kDelayDiffOffsetSamples was declared only for Mac when WEBRTC_UNTRUSTED_DELAY was automatically turned on.

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

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

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6866 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-11 15:39:00 +00:00
bjornv@webrtc.org
96d8b0e69f Revert 6860 "SSE2 version of SubbandCoherence()"
> SSE2 version of SubbandCoherence()
> 
> The performance gain on a x86 laptop (Intel(R) Core(TM) i5-2520M CPU @ 2.50GHz)
> reported by audioproc is ~3.3%
> 
> The output is bit exact.
> 
> R=bjornv@webrtc.org, cd@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/18779004
> 
> Patch from Scott LaVarnway <slavarnw@gmail.com>.

TBR=bjornv@webrtc.org

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

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

The output is bit exact.

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

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

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6860 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-11 10:38:31 +00:00
andrew@webrtc.org
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
bjornv@webrtc.org
5bde66e913 audio_processing: Updates aec_core_sse2.c with changes made to aec_common.h
The change of definitions moved to aec_common.h was done in CL17839005.

BUG=3131
TBR=kwiberg@webrtc.org
TESTED=builds locally

Review URL: https://webrtc-codereview.appspot.com/16859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6648 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-10 08:09:50 +00:00
bjornv@webrtc.org
555fc78f27 Neon version of SubbandCoherence()
The performance gain on a Nexus 7 reported by audioproc is ~1.4%

The output is NOT bit exact.  Any difference seen is +-1.

BUG=3131
R=bjornv@webrtc.org, cd@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17839005

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6647 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-10 08:03:11 +00:00