| 6955870806624479723addfae6dcf5d13968796c |
|
13-Jan-2016 |
Peter Kasting <pkasting@google.com> |
Convert channel counts to size_t.
IIRC, this was originally requested by ajm during review of the other size_t conversions I did over the past year, and I agreed it made sense, but wanted to do it separately since those changes were already gargantuan.
BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, henrika@webrtc.org, kjellander@webrtc.org, minyue@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/1316523002 .
Cr-Commit-Position: refs/heads/master@{#11229}
/external/webrtc/webrtc/modules/audio_processing/echo_cancellation_impl.cc
|
| b2328d11dcc86fba1661ee3fa0d51fc126939764 |
|
12-Jan-2016 |
aluebs <aluebs@webrtc.org> |
Remove additional channel constraints when Beamforming is enabled in AudioProcessing
The general constraints on number of channels for AudioProcessing is:
num_in_channels == num_out_channels || num_out_channels == 1
When Beamforming is enabled and additional constraint was added forcing:
num_out_channels == 1
This artificial constraint was removed by adding upmixing support in CopyTo, since it was already supported for the AudioFrame interface using InterleaveTo.
Review URL: https://codereview.webrtc.org/1571013002
Cr-Commit-Position: refs/heads/master@{#11215}
/external/webrtc/webrtc/modules/audio_processing/echo_cancellation_impl.cc
|
| 25702cb1628941427fa55e528f53483f239ae011 |
|
08-Jan-2016 |
pkasting <pkasting@chromium.org> |
Misc. small cleanups.
* Better param names
* Avoid using negative values for (bogus) placeholder channel counts (mostly in tests). Since channels will be changing to size_t, negative values will be illegal; it's sufficient to use 0 in these cases.
* Use arraysize()
* Use size_t for counting frames, samples, blocks, buffers, and bytes -- most of these are already size_t in most places, this just fixes some stragglers
* reinterpret_cast<int64_t>(void*) is not necessarily safe; use uintptr_t instead
* Remove unnecessary code, e.g. dead code, needlessly long/repetitive code, or function overrides that exactly match the base definition
* Fix indenting
* Use uint32_t for timestamps (matching how it's already a uint32_t in most places)
* Spelling
* RTC_CHECK_EQ(expected, actual)
* Rewrap
* Use .empty()
* Be more pedantic about matching int/int32_t/
* Remove pointless consts on input parameters to functions
* Add missing sanity checks
All this was found in the course of constructing https://codereview.webrtc.org/1316523002/ , and is being landed separately first.
BUG=none
TEST=none
Review URL: https://codereview.webrtc.org/1534193008
Cr-Commit-Position: refs/heads/master@{#11191}
/external/webrtc/webrtc/modules/audio_processing/echo_cancellation_impl.cc
|
| df3efa8c079294857a8b8e0a02634d06a6d6b6d6 |
|
28-Nov-2015 |
peah <peah@webrtc.org> |
Introduced the new locking scheme
BUG=webrtc:5099
Review URL: https://codereview.webrtc.org/1424663003
Cr-Commit-Position: refs/heads/master@{#10836}
/external/webrtc/webrtc/modules/audio_processing/echo_cancellation_impl.cc
|
| 2446e5a2de9d5d78117a02ff2b206194bb8f4e8e |
|
18-Nov-2015 |
peah <peah@webrtc.org> |
Fixed the render queue item size preallocation and verification, moved constant declarations, removed redundant queue allocation
BUG=
Review URL: https://codereview.webrtc.org/1454683002
Cr-Commit-Position: refs/heads/master@{#10689}
/external/webrtc/webrtc/modules/audio_processing/echo_cancellation_impl.cc
|
| fa6228e221d818af55e3d8343c792f2c1ecc7252 |
|
17-Nov-2015 |
peah <peah@webrtc.org> |
Introduced the render sample queue for the aec and aecm.
BUG=webrtc:5099
Review URL: https://codereview.webrtc.org/1410833002
Cr-Commit-Position: refs/heads/master@{#10662}
/external/webrtc/webrtc/modules/audio_processing/echo_cancellation_impl.cc
|
| 9b72af94cd61782ada88f777b07854daf9657bb2 |
|
11-Nov-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
Remove webrtc/modules/audio_processing/{aec,aecm,ns}/include
BUG=webrtc:5095
TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
R=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1440523002 .
Cr-Commit-Position: refs/heads/master@{#10608}
/external/webrtc/webrtc/modules/audio_processing/echo_cancellation_impl.cc
|
| c12be3984ff49f202f873f8ecd83f0e5b9cb36c9 |
|
10-Nov-2015 |
peah <peah@webrtc.org> |
-Removed the indirect error message reporting in aec and aecm.
-Made the component error messages generic to be an unspecified error message.
BUG=webrtc:5099
Review URL: https://codereview.webrtc.org/1404743003
Cr-Commit-Position: refs/heads/master@{#10570}
/external/webrtc/webrtc/modules/audio_processing/echo_cancellation_impl.cc
|
| 98f53510b222f71fdd8b799b2f33737ceeb28c61 |
|
28-Oct-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
system_wrappers: rename interface -> include
BUG=webrtc:5095
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1413333002 .
Cr-Commit-Position: refs/heads/master@{#10438}
/external/webrtc/webrtc/modules/audio_processing/echo_cancellation_impl.cc
|
| 13b96ba90f72164134019cbfc07d4a47cf1fd091 |
|
03-Oct-2015 |
Minyue <minyue@webrtc.org> |
Adding APM configuration in AEC dump.
The AEC dump was not self-contented enough in the sense that APM configuration is missing, and therefore, given an AEC dump, it is sometimes not clear how to reproduce problems.
This CL tries to address the problem.
Note that this cannot guarantee a perfect reproduction in all cases. Dumping from the middle of a call makes the initial states unknown and thus may make the result non-reproducible.
BUG=
TEST= 1. new dump in Chromium and unpack
2. unpack old dump
R=andrew@webrtc.org, peah@webrtc.org
Review URL: https://codereview.webrtc.org/1348903004 .
Cr-Commit-Position: refs/heads/master@{#10155}
/external/webrtc/webrtc/modules/audio_processing/echo_cancellation_impl.cc
|
| dce40cf804019a9898b6ab8d8262466b697c56e0 |
|
24-Aug-2015 |
Peter Kasting <pkasting@google.com> |
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}
/external/webrtc/webrtc/modules/audio_processing/echo_cancellation_impl.cc
|
| 366e95252aba863f886aa692459bf267e5525d86 |
|
03-Jul-2015 |
henrik.lundin <henrik.lundin@webrtc.org> |
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}
/external/webrtc/webrtc/modules/audio_processing/echo_cancellation_impl.cc
|
| 0f133b99c655cbdb347b4a71ac872c071532189f |
|
02-Jul-2015 |
henrik.lundin <henrik.lundin@webrtc.org> |
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}
/external/webrtc/webrtc/modules/audio_processing/echo_cancellation_impl.cc
|
| b02af18c5cb6d6c3def7f44d27a63068360f4f29 |
|
16-Jun-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
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}
/external/webrtc/webrtc/modules/audio_processing/echo_cancellation_impl.cc
|
| 9345e86551a0e59e77dfee6eed3e2fe2a833b269 |
|
10-Jun-2015 |
Bjorn Volcker <bjornv@webrtc.org> |
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}
/external/webrtc/webrtc/modules/audio_processing/echo_cancellation_impl.cc
|
| 441f6347311bcf2079435c3888d67e1fb321f9f8 |
|
09-Jun-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
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}
/external/webrtc/webrtc/modules/audio_processing/echo_cancellation_impl.cc
|
| 3fbf3f8841b5460503fb646eaedcb063620434a8 |
|
05-Jun-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
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}
/external/webrtc/webrtc/modules/audio_processing/echo_cancellation_impl.cc
|
| 5f4b7e2873864c61e2ad6d88679dcd5d321bfd16 |
|
05-Jun-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
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}
/external/webrtc/webrtc/modules/audio_processing/echo_cancellation_impl.cc
|
| d35a5c350617cc9d60ce45201764a99229b7299a |
|
10-Feb-2015 |
aluebs@webrtc.org <aluebs@webrtc.org> |
Make ChannelBuffer aware of frequency bands
Now the ChannelBuffer has 2 separate arrays, one for the full-band data and one for the splitted one. The corresponding accessors are added to the ChannelBuffer.
This is done to avoid having to refresh the bands pointers in AudioBuffer. It will also allow us to have a general accessor like data()[band][channel][sample].
All the files using the ChannelBuffer needed to be re-factored.
Tested with modules_unittests, common_audio_unittests, audioproc, audioproc_f, voe_cmd_test.
R=andrew@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36999004
Cr-Commit-Position: refs/heads/master@{#8318}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8318 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/echo_cancellation_impl.cc
|
| b1786dbab00dd66a9e59a68414e85b2b2615a24f |
|
03-Feb-2015 |
bjornv@webrtc.org <bjornv@webrtc.org> |
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
/external/webrtc/webrtc/modules/audio_processing/echo_cancellation_impl.cc
|
| c78d81ae895510edc217d1598d9111e3c8987b09 |
|
21-Jan-2015 |
aluebs@webrtc.org <aluebs@webrtc.org> |
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
/external/webrtc/webrtc/modules/audio_processing/echo_cancellation_impl.cc
|
| ee0c100d5495cd8c440b767a7852532afbbcefb2 |
|
20-Jan-2015 |
tina.legrand@webrtc.org <tina.legrand@webrtc.org> |
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
/external/webrtc/webrtc/modules/audio_processing/echo_cancellation_impl.cc
|
| 64d3c4b9ac862c682d297851083f81942a035652 |
|
15-Jan-2015 |
aluebs@webrtc.org <aluebs@webrtc.org> |
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
/external/webrtc/webrtc/modules/audio_processing/echo_cancellation_impl.cc
|
| c5ebbd98f5996db0defbbfc14f5ca41e620bd7e4 |
|
10-Dec-2014 |
aluebs@webrtc.org <aluebs@webrtc.org> |
Support 48kHz in Noise Suppression
Doing the same for the 16-24kHz band than was done in the 8-16kHz.
Results look and sound as nice.
BUG=webrtc:3146
R=andrew@webrtc.org, bjornv@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29139004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7865 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/echo_cancellation_impl.cc
|
| a7384a1126cda7ce726f73b023bad997627fc138 |
|
03-Dec-2014 |
aluebs@webrtc.org <aluebs@webrtc.org> |
Simplify audio_buffer APIs
Now there is only one API to get the data or the channels (one const and one no const) merged or by band.
The band is passed in as a parameter, instead of calling different methods.
BUG=webrtc:3146
R=andrew@webrtc.org, bjornv@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27249004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7790 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/echo_cancellation_impl.cc
|
| 38214d53db5f19228b38424700db8b90cf8b0234 |
|
03-Jul-2014 |
kwiberg@webrtc.org <kwiberg@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
EchoCancellationImpl::ProcessRenderAudio: Use float samples directly
This patch lets EchoCancellationImpl::ProcessRenderAudio ask the given
AudioBuffer for float sample data directly, instead of asking for
int16 samples and then converting manually.
Since EchoCancellationImpl::ProcessRenderAudio takes a const
AudioBuffer*, it was necessary to add some const accessors for float
data to AudioBuffer.
R=aluebs@webrtc.org, andrew@webrtc.org, bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6590 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/echo_cancellation_impl.cc
|
| 3f83072c26969a667f7d1d07603559c0ed0f3ece |
|
11-Jun-2014 |
bjornv@webrtc.org <bjornv@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
modules/audio_processing: Adds a config for reported delays
There are platforms and devices where the reported delays are untrusted and we currently solve that with an extended filter length and a slightly more conservative delay handling.
With this change we give the user the possibility to turn off reported system delay values completely.
- Includes new unit tests.
TESTED=trybots and manual testing
R=aluebs@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13629004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6391 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/echo_cancellation_impl.cc
|
| d5da25063c46678f50882da5c1bad21ca3cfe271 |
|
15-May-2014 |
mflodman@webrtc.org <mflodman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert "Revert "Audio processing: Feed each processing step its choice
of int or float data"
This reverts commit 6142.
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17519004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6172 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/echo_cancellation_impl.cc
|
| b1a66d166c137aef483b1a24540b00b0d45bdf27 |
|
14-May-2014 |
mflodman@webrtc.org <mflodman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert "Audio processing: Feed each processing step its choice of int or float data"
This reverts r6138.
tbr=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6142 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/echo_cancellation_impl.cc
|
| 934a265a47f71f06fea5878bd50ddcc629ca455a |
|
14-May-2014 |
kwiberg@webrtc.org <kwiberg@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Audio processing: Feed each processing step its choice of int or float data
Each audio processing step is given a pointer to an AudioBuffer, where
it can read and write int data. This patch adds corresponding
AudioBuffer methods to read and write float data; the buffer will
automatically convert the stored data between int and float as
necessary.
This patch also modifies the echo cancellation step to make use of the
new methods (it was already using floats internally; now it doesn't
have to convert from and to ints anymore).
(The reference data to the ApmTest.Process test had to be modified
slightly; this is because the echo canceller no longer unnecessarily
converts float data to int and then immediately back to float for each
iteration in the loop in EchoCancellationImpl::ProcessCaptureAudio.)
BUG=
R=aluebs@webrtc.org, andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18399005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6138 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/echo_cancellation_impl.cc
|
| e9d3760d5cd0f4bb1821897dc995dcbe3cfdccaa |
|
23-Apr-2014 |
bjornv@webrtc.org <bjornv@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
AEC: Adds a reported_delay_enabled_ flag
Adds a feature to completely turn on or off buffer handling based on reported delay values. During startup, reported delays are controlled differently through, e.g., WEBRTC_UNTRUSTED_DELAY. By default, the feature is enabled giving the same output as before this change.
TESTED=trybots, modules_unittest
R=aluebs@webrtc.org, andrew@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12349005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5965 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/echo_cancellation_impl.cc
|
| ddbb8a2c243f9d54cb0ce0092e341dfc6e126bb3 |
|
22-Apr-2014 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Support arbitrary input/output rates and downmixing in AudioProcessing.
Select "processing" rates based on the input and output sampling rates.
Resample the input streams to those rates, and if necessary to the
output rate.
- Remove deprecated stream format APIs.
- Remove deprecated device sample rate APIs.
- Add a ChannelBuffer class to help manage deinterleaved channels.
- Clean up the splitting filter state.
- Add a unit test which verifies the output against known-working
native format output.
BUG=2894
R=aluebs@webrtc.org, bjornv@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5959 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/echo_cancellation_impl.cc
|
| 5964fe0f86a4f33831d1f4994dbde1b42c93bd81 |
|
22-Apr-2014 |
bjornv@webrtc.org <bjornv@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
audio_processing: DestroyHandle() now returns void
The return value was not used anyhow and there is no proper action to be taken if we would have received an error. Hence, in line with issue441 we should return void upon free.
BUG=441
TESTED=trybots,modules_unittest
R=andrew@webrtc.org, aluebs@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12269004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5949 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/echo_cancellation_impl.cc
|
| 56e4a05053d6addc7dbbe2b4d07271305fdbea75 |
|
27-Feb-2014 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove ProcessingComponent's dependence on AudioProcessingImpl.
- Move needed accessors to AudioProcessing.
- Inject the crit directly as a dependency.
- Remove the now unneeded EchoCancellationImplWrapper.
BUG=2894
R=aluebs@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5620 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/echo_cancellation_impl.cc
|
| 60730cfe3ce80e4023cd678373456cb703f000a4 |
|
07-Jan-2014 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove the requirement to call set_sample_rate_hz and friends.
Instead have ProcessStream transparently handle changes to the stream
audio parameters (sample rate and channels). This removes two locks
per 10 ms ProcessStream call taken by VoiceEngine (four total with the
audio level indicator.)
Also, prepare future improvements by having the splitting filter take
a length parameter. This will allow it to work at different sample
rates. Remove the useless splitting_filter wrapper.
TESTED=voe_cmd_test with audio processing enabled and switching between
codecs; unit tests.
R=aluebs@webrtc.org, bjornv@webrtc.org, turaj@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5346 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/echo_cancellation_impl.cc
|
| 1760a17b8e176f38bcc203bd0afbc9c59df9b502 |
|
26-Sep-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add an extended filter mode to AEC.
Re-land: http://review.webrtc.org/2151007/
TBR=bjornv@webrtc.org
Original change description:
This mode extends the filter length from the current 48 ms to 128 ms.
It is runtime selectable which allows it to be enabled through
experiment. We reuse the DelayCorrection infrastructure to avoid having
to replumb everything up to libjingle.
Increases AEC complexity by ~50% on modern x86 CPUs.
Measurements (in percent of usage on one core):
Machine/CPU Normal Extended
MacBook Retina (Early 2013),
Core i7 Ivy Bridge (2.7 GHz, hyperthreaded) 0.6% 0.9%
MacBook Air (Late 2010), Core 2 Duo (2.13 GHz) 1.4% 2.7%
Chromebook Pixel, Core i5 Ivy Bridge (1.8 GHz) 0.6% 1.0%
Samsung ARM Chromebook,
Samsung Exynos 5 Dual (1.7 GHz) 3.2% 5.6%
The relative value is large of course but the absolute should be
acceptable in order to have a working AEC on some platforms.
Detailed changes to the algorithm:
- The filter length is changed from 48 to 128 ms. This comes with tuning
of several parameters: i) filter adaptation stepsize and error
threshold; ii) non-linear processing smoothing and overdrive.
- Option to ignore the reported delays on platforms which we deem
sufficiently unreliable. Currently this will be enabled in Chromium for
Mac.
- Faster startup times by removing the excessive "startup phase"
processing of reported delays.
- Much more conservative adjustments to the far-end read pointer. We
smooth the delay difference more heavily, and back off from the
difference more. Adjustments force a readaptation of the filter, so they
should be avoided except when really necessary.
Corresponds to these changes:
https://chromereviews.googleplex.com/9412014
https://chromereviews.googleplex.com/9514013
https://chromereviews.googleplex.com/9960013
BUG=454,827,1261
Review URL: https://webrtc-codereview.appspot.com/2295006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4848 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/echo_cancellation_impl.cc
|
| ce014d97cd9bcc893ce60897227e8a5147d8741c |
|
25-Sep-2013 |
asapersson@webrtc.org <asapersson@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 4837 "Add an extended filter mode to AEC."
> Add an extended filter mode to AEC.
>
> This mode extends the filter length from the current 48 ms to 128 ms.
> It is runtime selectable which allows it to be enabled through
> experiment. We reuse the DelayCorrection infrastructure to avoid having
> to replumb everything up to libjingle.
>
> Increases AEC complexity by ~50% on modern x86 CPUs.
> Measurements (in percent of usage on one core):
>
> Machine/CPU Normal Extended
> MacBook Retina (Early 2013),
> Core i7 Ivy Bridge (2.7 GHz, hyperthreaded) 0.6% 0.9%
>
> MacBook Air (Late 2010), Core 2 Duo (2.13 GHz) 1.4% 2.7%
>
> Chromebook Pixel, Core i5 Ivy Bridge (1.8 GHz) 0.6% 1.0%
>
> Samsung ARM Chromebook,
> Samsung Exynos 5 Dual (1.7 GHz) 3.2% 5.6%
>
> The relative value is large of course but the absolute should be
> acceptable in order to have a working AEC on some platforms.
>
> Detailed changes to the algorithm:
> - The filter length is changed from 48 to 128 ms. This comes with tuning
> of several parameters: i) filter adaptation stepsize and error
> threshold; ii) non-linear processing smoothing and overdrive.
> - Option to ignore the reported delays on platforms which we deem
> sufficiently unreliable. Currently this will be enabled in Chromium for
> Mac.
> - Faster startup times by removing the excessive "startup phase"
> processing of reported delays.
> - Much more conservative adjustments to the far-end read pointer. We
> smooth the delay difference more heavily, and back off from the
> difference more. Adjustments force a readaptation of the filter, so they
> should be avoided except when really necessary.
>
> Corresponds to these changes:
> https://chromereviews.googleplex.com/9412014
> https://chromereviews.googleplex.com/9514013
> https://chromereviews.googleplex.com/9960013
>
> BUG=454,827,1261
> R=bjornv@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/2151007
TBR=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2296005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4839 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/echo_cancellation_impl.cc
|
| 26e02f0ee44735697d575e35d2e5f3cd43dde512 |
|
25-Sep-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add an extended filter mode to AEC.
This mode extends the filter length from the current 48 ms to 128 ms.
It is runtime selectable which allows it to be enabled through
experiment. We reuse the DelayCorrection infrastructure to avoid having
to replumb everything up to libjingle.
Increases AEC complexity by ~50% on modern x86 CPUs.
Measurements (in percent of usage on one core):
Machine/CPU Normal Extended
MacBook Retina (Early 2013),
Core i7 Ivy Bridge (2.7 GHz, hyperthreaded) 0.6% 0.9%
MacBook Air (Late 2010), Core 2 Duo (2.13 GHz) 1.4% 2.7%
Chromebook Pixel, Core i5 Ivy Bridge (1.8 GHz) 0.6% 1.0%
Samsung ARM Chromebook,
Samsung Exynos 5 Dual (1.7 GHz) 3.2% 5.6%
The relative value is large of course but the absolute should be
acceptable in order to have a working AEC on some platforms.
Detailed changes to the algorithm:
- The filter length is changed from 48 to 128 ms. This comes with tuning
of several parameters: i) filter adaptation stepsize and error
threshold; ii) non-linear processing smoothing and overdrive.
- Option to ignore the reported delays on platforms which we deem
sufficiently unreliable. Currently this will be enabled in Chromium for
Mac.
- Faster startup times by removing the excessive "startup phase"
processing of reported delays.
- Much more conservative adjustments to the far-end read pointer. We
smooth the delay difference more heavily, and back off from the
difference more. Adjustments force a readaptation of the filter, so they
should be avoided except when really necessary.
Corresponds to these changes:
https://chromereviews.googleplex.com/9412014
https://chromereviews.googleplex.com/9514013
https://chromereviews.googleplex.com/9960013
BUG=454,827,1261
R=bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2151007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4837 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/echo_cancellation_impl.cc
|
| 61e596fc49443971219aa3989b200407d919f6c5 |
|
25-Jul-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add a Config class interface to AudioProcessing for passing options.
Pass the Config down to all AudioProcessing components.
Also add an EchoCancellationImplWrapper to optionally create different
EchoCancellationImpls.
BUG=2117
TBR=turaj@webrtc.org
TESTED=git try
Review URL: https://webrtc-codereview.appspot.com/1843004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4400 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/echo_cancellation_impl.cc
|
| b7192b82476d00384fdc153e6a09a6ac53cef67b |
|
10-Apr-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
WebRtc_Word32 -> int32_t in audio_processing/
BUG=314
Review URL: https://webrtc-codereview.appspot.com/1307004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3809 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/echo_cancellation_impl.cc
|
| 91d11b3cdd3ab20600aadb92929e4d48ff4f7422 |
|
05-Mar-2013 |
bjornv@webrtc.org <bjornv@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adds new AEC API to audio_processing.
One unit test added.
Tested with audioproc_unittest and trybots
TEST=none
BUG=none
Review URL: https://webrtc-codereview.appspot.com/1154004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3613 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/echo_cancellation_impl.cc
|
| 6be1e934ad48ccdac734b5cbd50528235ec93816 |
|
01-Mar-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Properly error check calls to AudioProcessing.
Checks must be made with "!= 0", not "== -1". Additionally:
* Clean up the function calling into AudioProcessing.
* Remove the unused _noiseWarning.
* Make the other warnings bool.
BUG=chromium:178040
Review URL: https://webrtc-codereview.appspot.com/1147004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3590 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/echo_cancellation_impl.cc
|
| 21a2fc902d37fb9607698dc0770c477b149163fe |
|
15-Feb-2013 |
bjornv@webrtc.org <bjornv@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
This Cl includes
* A getter for echo_state
* Style changes, such as changes to int where appropriate
TEST=audioproc_unittest, trybots
BUG=None
Review URL: https://webrtc-codereview.appspot.com/1093011
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3522 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/echo_cancellation_impl.cc
|
| 14b43beb7ce4440b30dcea31196de5b4a529cb6b |
|
22-Oct-2012 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Move src/ -> webrtc/
TBR=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/915006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/echo_cancellation_impl.cc
|