Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(342)

Side by Side Diff: webrtc/modules/audio_processing/audio_processing.gypi

Issue 2424173003: Move functionality out from AudioFrame and into AudioFrameOperations. (Closed)
Patch Set: Created 4 years, 2 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
1 # Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 1 # Copyright (c) 2012 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 { 9 {
10 'variables': { 10 'variables': {
11 'shared_generated_dir': '<(SHARED_INTERMEDIATE_DIR)/audio_processing/asm_off sets', 11 'shared_generated_dir': '<(SHARED_INTERMEDIATE_DIR)/audio_processing/asm_off sets',
12 }, 12 },
13 'targets': [ 13 'targets': [
14 { 14 {
15 'target_name': 'audio_processing', 15 'target_name': 'audio_processing',
16 'type': 'static_library', 16 'type': 'static_library',
17 'variables': { 17 'variables': {
18 # Outputs some low-level debug files. 18 # Outputs some low-level debug files.
19 'agc_debug_dump%': 0, 19 'agc_debug_dump%': 0,
20 20
21 # Disables the usual mode where we trust the reported system delay 21 # Disables the usual mode where we trust the reported system delay
22 # values the AEC receives. The corresponding define is set appropriately 22 # values the AEC receives. The corresponding define is set appropriately
23 # in the code, but it can be force-enabled here for testing. 23 # in the code, but it can be force-enabled here for testing.
24 'aec_untrusted_delay_for_testing%': 0, 24 'aec_untrusted_delay_for_testing%': 0,
25 }, 25 },
26 'dependencies': [ 26 'dependencies': [
27 '<(webrtc_root)/base/base.gyp:rtc_base_approved', 27 '<(webrtc_root)/base/base.gyp:rtc_base_approved',
28 '<(webrtc_root)/common.gyp:webrtc_common', 28 '<(webrtc_root)/common.gyp:webrtc_common',
29 '<(webrtc_root)/common_audio/common_audio.gyp:common_audio', 29 '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
30 '<(webrtc_root)/modules/modules.gyp:audio_frame_operations',
30 '<(webrtc_root)/modules/modules.gyp:isac', 31 '<(webrtc_root)/modules/modules.gyp:isac',
31 '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers', 32 '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
32 ], 33 ],
33 'sources': [ 34 'sources': [
34 'aec/aec_core.cc', 35 'aec/aec_core.cc',
35 'aec/aec_core.h', 36 'aec/aec_core.h',
36 'aec/aec_core_optimized_methods.h', 37 'aec/aec_core_optimized_methods.h',
37 'aec/aec_resampler.cc', 38 'aec/aec_resampler.cc',
38 'aec/aec_resampler.h', 39 'aec/aec_resampler.h',
39 'aec/echo_cancellation.cc', 40 'aec/echo_cancellation.cc',
(...skipping 275 matching lines...) Expand 10 before | Expand all | Expand 10 after
315 'defines': ['WEBRTC_APM_DEBUG_DUMP=1',], 316 'defines': ['WEBRTC_APM_DEBUG_DUMP=1',],
316 }], 317 }],
317 ['apm_debug_dump==0', { 318 ['apm_debug_dump==0', {
318 'defines': ['WEBRTC_APM_DEBUG_DUMP=0',], 319 'defines': ['WEBRTC_APM_DEBUG_DUMP=0',],
319 }], 320 }],
320 ], 321 ],
321 }], 322 }],
322 }], 323 }],
323 ], 324 ],
324 } 325 }
OLDNEW

Powered by Google App Engine
This is Rietveld 408576698