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Side by Side Diff: webrtc/modules/audio_processing/BUILD.gn

Issue 1949803004: Moved the functionality in aec_core_internal.h into other files (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@ObjectifyEchoCancellation_CL
Patch Set: Corrected a bad merge Created 4 years, 7 months ago
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1 # Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 1 # Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
2 # 2 #
3 # Use of this source code is governed by a BSD-style license 3 # Use of this source code is governed by a BSD-style license
4 # that can be found in the LICENSE file in the root of the source 4 # that can be found in the LICENSE file in the root of the source
5 # tree. An additional intellectual property rights grant can be found 5 # tree. An additional intellectual property rights grant can be found
6 # in the file PATENTS. All contributing project authors may 6 # in the file PATENTS. All contributing project authors may
7 # be found in the AUTHORS file in the root of the source tree. 7 # be found in the AUTHORS file in the root of the source tree.
8 8
9 import("//build/config/arm.gni") 9 import("//build/config/arm.gni")
10 import("//third_party/protobuf/proto_library.gni") 10 import("//third_party/protobuf/proto_library.gni")
11 import("../../build/webrtc.gni") 11 import("../../build/webrtc.gni")
12 12
13 declare_args() { 13 declare_args() {
14 # Outputs some low-level debug files. 14 # Outputs some low-level debug files.
15 aec_debug_dump = false 15 aec_debug_dump = false
16 16
17 # Disables the usual mode where we trust the reported system delay 17 # Disables the usual mode where we trust the reported system delay
18 # values the AEC receives. The corresponding define is set appropriately 18 # values the AEC receives. The corresponding define is set appropriately
19 # in the code, but it can be force-enabled here for testing. 19 # in the code, but it can be force-enabled here for testing.
20 aec_untrusted_delay_for_testing = false 20 aec_untrusted_delay_for_testing = false
21 } 21 }
22 22
23 source_set("audio_processing") { 23 source_set("audio_processing") {
24 sources = [ 24 sources = [
25 "aec/aec_core.cc", 25 "aec/aec_core.cc",
26 "aec/aec_core.h", 26 "aec/aec_core.h",
27 "aec/aec_core_internal.h", 27 "aec/aec_core_optimized_methods.h",
28 "aec/aec_rdft.cc", 28 "aec/aec_rdft.cc",
29 "aec/aec_rdft.h", 29 "aec/aec_rdft.h",
30 "aec/aec_resampler.cc", 30 "aec/aec_resampler.cc",
31 "aec/aec_resampler.h", 31 "aec/aec_resampler.h",
32 "aec/echo_cancellation.cc", 32 "aec/echo_cancellation.cc",
33 "aec/echo_cancellation.h", 33 "aec/echo_cancellation.h",
34 "aecm/aecm_core.cc", 34 "aecm/aecm_core.cc",
35 "aecm/aecm_core.h", 35 "aecm/aecm_core.h",
36 "aecm/echo_control_mobile.cc", 36 "aecm/echo_control_mobile.cc",
37 "aecm/echo_control_mobile.h", 37 "aecm/echo_control_mobile.h",
(...skipping 254 matching lines...) Expand 10 before | Expand all | Expand 10 after
292 "../../common_audio", 292 "../../common_audio",
293 ] 293 ]
294 294
295 if (aec_debug_dump) { 295 if (aec_debug_dump) {
296 defines = [ "WEBRTC_AEC_DEBUG_DUMP=1" ] 296 defines = [ "WEBRTC_AEC_DEBUG_DUMP=1" ]
297 } else { 297 } else {
298 defines = [ "WEBRTC_AEC_DEBUG_DUMP=0" ] 298 defines = [ "WEBRTC_AEC_DEBUG_DUMP=0" ]
299 } 299 }
300 } 300 }
301 } 301 }
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