Files
platform-external-webrtc/webrtc/modules/audio_processing/audio_processing.gypi
andrew@webrtc.org 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

220 lines
6.8 KiB
Python

# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'targets': [
{
'target_name': 'audio_processing',
'type': 'static_library',
'variables': {
# Outputs some low-level debug files.
'aec_debug_dump%': 0,
# Disables the usual mode where we trust the reported system delay
# values the AEC receives. The corresponding define is set appropriately
# in the code, but it can be force-enabled here for testing.
'aec_untrusted_delay_for_testing%': 0,
},
'dependencies': [
'<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
'<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
],
'include_dirs': [
'../interface',
'aec/include',
'aecm/include',
'agc/include',
'include',
'ns/include',
'utility',
],
'direct_dependent_settings': {
'include_dirs': [
'../interface',
'include',
],
},
'sources': [
'aec/include/echo_cancellation.h',
'aec/echo_cancellation.c',
'aec/echo_cancellation_internal.h',
'aec/aec_core.h',
'aec/aec_core.c',
'aec/aec_core_internal.h',
'aec/aec_rdft.h',
'aec/aec_rdft.c',
'aec/aec_resampler.h',
'aec/aec_resampler.c',
'aecm/include/echo_control_mobile.h',
'aecm/echo_control_mobile.c',
'aecm/aecm_core.c',
'aecm/aecm_core.h',
'agc/include/gain_control.h',
'agc/analog_agc.c',
'agc/analog_agc.h',
'agc/digital_agc.c',
'agc/digital_agc.h',
'audio_buffer.cc',
'audio_buffer.h',
'audio_processing_impl.cc',
'audio_processing_impl.h',
'echo_cancellation_impl.cc',
'echo_cancellation_impl.h',
'echo_cancellation_impl_wrapper.h',
'echo_control_mobile_impl.cc',
'echo_control_mobile_impl.h',
'gain_control_impl.cc',
'gain_control_impl.h',
'high_pass_filter_impl.cc',
'high_pass_filter_impl.h',
'include/audio_processing.h',
'level_estimator_impl.cc',
'level_estimator_impl.h',
'noise_suppression_impl.cc',
'noise_suppression_impl.h',
'splitting_filter.cc',
'splitting_filter.h',
'processing_component.cc',
'processing_component.h',
'utility/delay_estimator.c',
'utility/delay_estimator.h',
'utility/delay_estimator_internal.h',
'utility/delay_estimator_wrapper.c',
'utility/delay_estimator_wrapper.h',
'utility/fft4g.c',
'utility/fft4g.h',
'utility/ring_buffer.c',
'utility/ring_buffer.h',
'voice_detection_impl.cc',
'voice_detection_impl.h',
],
'conditions': [
['aec_debug_dump==1', {
'defines': ['WEBRTC_AEC_DEBUG_DUMP',],
}],
['aec_untrusted_delay_for_testing==1', {
'defines': ['WEBRTC_UNTRUSTED_DELAY',],
}],
['enable_protobuf==1', {
'dependencies': ['audioproc_debug_proto'],
'defines': ['WEBRTC_AUDIOPROC_DEBUG_DUMP'],
}],
['prefer_fixed_point==1', {
'defines': ['WEBRTC_NS_FIXED'],
'sources': [
'ns/include/noise_suppression_x.h',
'ns/noise_suppression_x.c',
'ns/nsx_core.c',
'ns/nsx_core.h',
'ns/nsx_defines.h',
],
}, {
'defines': ['WEBRTC_NS_FLOAT'],
'sources': [
'ns/defines.h',
'ns/include/noise_suppression.h',
'ns/noise_suppression.c',
'ns/ns_core.c',
'ns/ns_core.h',
'ns/windows_private.h',
],
}],
['target_arch=="ia32" or target_arch=="x64"', {
'dependencies': ['audio_processing_sse2',],
}],
['(target_arch=="arm" and armv7==1) or target_arch=="armv7"', {
'dependencies': ['audio_processing_neon',],
}],
],
# TODO(jschuh): Bug 1348: fix size_t to int truncations.
'msvs_disabled_warnings': [ 4267, ],
},
],
'conditions': [
['enable_protobuf==1', {
'targets': [
{
'target_name': 'audioproc_debug_proto',
'type': 'static_library',
'sources': ['debug.proto',],
'variables': {
'proto_in_dir': '.',
# Workaround to protect against gyp's pathname relativization when
# this file is included by modules.gyp.
'proto_out_protected': 'webrtc/audio_processing',
'proto_out_dir': '<(proto_out_protected)',
},
'includes': ['../../build/protoc.gypi',],
},
],
}],
['target_arch=="ia32" or target_arch=="x64"', {
'targets': [
{
'target_name': 'audio_processing_sse2',
'type': 'static_library',
'sources': [
'aec/aec_core_sse2.c',
'aec/aec_rdft_sse2.c',
],
'cflags': ['-msse2',],
'xcode_settings': {
'OTHER_CFLAGS': ['-msse2',],
},
},
],
}],
['(target_arch=="arm" and armv7==1) or target_arch=="armv7"', {
'targets': [{
'target_name': 'audio_processing_neon',
'type': 'static_library',
'includes': ['../../build/arm_neon.gypi',],
'dependencies': [
'<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
],
'sources': [
'aecm/aecm_core_neon.c',
'ns/nsx_core_neon.c',
],
'conditions': [
['OS=="android" or OS=="ios"', {
'dependencies': [
'audio_processing_offsets',
],
'sources': [
'aecm/aecm_core_neon.S',
'ns/nsx_core_neon.S',
],
'sources!': [
'aecm/aecm_core_neon.c',
'ns/nsx_core_neon.c',
],
'includes!': ['../../build/arm_neon.gypi',],
}],
],
}],
'conditions': [
['OS=="android" or OS=="ios"', {
'targets': [{
'target_name': 'audio_processing_offsets',
'type': 'none',
'sources': [
'aecm/aecm_core_neon_offsets.c',
'ns/nsx_core_neon_offsets.c',
],
'variables': {
'asm_header_dir': 'asm_offsets',
},
'includes': ['../../build/generate_asm_header.gypi',],
}],
}],
],
}],
],
}