webrtc_m130/webrtc/modules/audio_processing/audio_processing.gypi
andrew@webrtc.org 382c0c209d Allow the RTP level indicator computation to work at any sample rate.
Break out the computation to a separate class, and call directly into
this from channel.cc rather than going through AudioProcessing. This
circumvents AudioProcessing's sample rate limitations.

We now compute the RMS over all samples rather than downmixing to a
single channel. This makes the call point in channel.cc easier, is
more "correct" and should have similar (negligible) complexity.

This caused slight changes in the RMS output, so the ApmTest.Process
reference has been updated. Snippet of the failing output:

[ RUN      ] ApmTest.Process
Running test 4 of 12...
Value of: rms_level
  Actual: 27
Expected: test->rms_level()
Which is: 28
Running test 5 of 12...
Value of: rms_level
  Actual: 26
Expected: test->rms_level()
Which is: 27
Running test 6 of 12...
Value of: rms_level
  Actual: 26
Expected: test->rms_level()
Which is: 27
Running test 10 of 12...
Value of: rms_level
  Actual: 27
Expected: test->rms_level()
Which is: 28
Running test 11 of 12...
Value of: rms_level
  Actual: 26
Expected: test->rms_level()
Which is: 27
Running test 12 of 12...
Value of: rms_level
  Actual: 26
Expected: test->rms_level()
Which is: 27

BUG=3290
TESTED=Chrome assert is avoided and both voe_cmd_test and apprtc
produce reasonable printed out results from RMS().

R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6056 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-05 18:22:21 +00:00

228 lines
7.0 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.
{
'variables': {
'audio_processing_dependencies': [
'<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
'<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
],
'shared_generated_dir': '<(SHARED_INTERMEDIATE_DIR)/audio_processing/asm_offsets',
},
'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': [
'<@(audio_processing_dependencies)',
],
'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',
'common.h',
'echo_cancellation_impl.cc',
'echo_cancellation_impl.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',
'processing_component.cc',
'processing_component.h',
'rms_level.cc',
'rms_level.h',
'typing_detection.cc',
'typing_detection.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',
],
'conditions': [
['target_arch=="mipsel"', {
'sources': [
'ns/nsx_core_mips.c',
],
}, {
'sources': [
'ns/nsx_core_c.c',
],
}],
],
}, {
'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 arm_version==7) or target_arch=="armv7"', {
'dependencies': ['audio_processing_neon',],
}],
['target_arch=="mipsel"', {
'sources': [
'aecm/aecm_core_mips.c',
],
'conditions': [
['mips_fpu==1', {
'sources': [
'aec/aec_core_mips.c',
'aec/aec_rdft_mips.c',
],
}],
],
}, {
'sources': [
'aecm/aecm_core_c.c',
],
}],
],
# 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 arm_version==7) 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': [
'<(gen_core_neon_offsets_gyp):*',
],
'sources': [
'aecm/aecm_core_neon.S',
'ns/nsx_core_neon.S',
],
'include_dirs': [
'<(shared_generated_dir)',
],
'sources!': [
'aecm/aecm_core_neon.c',
'ns/nsx_core_neon.c',
],
'includes!': ['../../build/arm_neon.gypi',],
}],
],
}],
}],
],
}