Commit Graph

21 Commits

Author SHA1 Message Date
c5ebbd98f5 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
2014-12-10 19:30:57 +00:00
a7384a1126 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
2014-12-03 01:06:35 +00:00
38214d53db 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
2014-07-03 09:47:33 +00:00
3f83072c26 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
2014-06-11 04:48:11 +00:00
d5da25063c 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
2014-05-15 11:17:21 +00:00
b1a66d166c 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
2014-05-14 09:39:56 +00:00
934a265a47 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
2014-05-14 09:01:35 +00:00
e9d3760d5c 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
2014-04-23 13:20:07 +00:00
ddbb8a2c24 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
2014-04-22 21:00:04 +00:00
5964fe0f86 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
2014-04-22 06:52:28 +00:00
56e4a05053 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
2014-02-27 22:23:17 +00:00
60730cfe3c 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
2014-01-07 17:45:09 +00:00
1760a17b8e 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
2013-09-25 23:17:38 +00:00
ce014d97cd 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
2013-09-25 12:27:27 +00:00
26e02f0ee4 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
2013-09-25 02:17:47 +00:00
61e596fc49 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
2013-07-25 18:28:29 +00:00
b7192b8247 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
2013-04-10 07:50:54 +00:00
91d11b3cdd 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
2013-03-05 16:53:09 +00:00
6be1e934ad 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
2013-03-01 18:47:28 +00:00
21a2fc902d 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
2013-02-15 17:01:03 +00:00
14b43beb7c 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
2012-10-22 18:19:23 +00:00