Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(220)

Side by Side Diff: webrtc/voice_engine/BUILD.gn

Issue 2799033006: Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. (Closed)
Patch Set: More backwards-compatibility! Created 3 years, 7 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
« no previous file with comments | « webrtc/test/mock_audio_encoder_factory.h ('k') | webrtc/voice_engine/channel.h » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
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("../webrtc.gni") 9 import("../webrtc.gni")
10 10
11 rtc_static_library("audio_coder") { 11 rtc_static_library("audio_coder") {
12 sources = [ 12 sources = [
13 "coder.cc", 13 "coder.cc",
14 "coder.h", 14 "coder.h",
15 ] 15 ]
16 deps = [ 16 deps = [
17 "..:webrtc_common", 17 "..:webrtc_common",
18 "../api/audio_codecs:builtin_audio_decoder_factory", 18 "../api/audio_codecs:builtin_audio_decoder_factory",
19 "../api/audio_codecs:builtin_audio_encoder_factory",
19 "../modules:module_api", 20 "../modules:module_api",
20 "../modules/audio_coding", 21 "../modules/audio_coding",
21 "../modules/audio_coding:audio_encoder_factory_interface",
22 "../modules/audio_coding:audio_format_conversion", 22 "../modules/audio_coding:audio_format_conversion",
23 "../modules/audio_coding:builtin_audio_encoder_factory",
24 "../modules/audio_coding:rent_a_codec", 23 "../modules/audio_coding:rent_a_codec",
25 ] 24 ]
26 25
27 if (!build_with_chromium && is_clang) { 26 if (!build_with_chromium && is_clang) {
28 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). 27 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
29 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] 28 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
30 } 29 }
31 } 30 }
32 31
33 rtc_static_library("file_player") { 32 rtc_static_library("file_player") {
(...skipping 100 matching lines...) Expand 10 before | Expand all | Expand 10 after
134 ":audio_level", 133 ":audio_level",
135 ":file_player", 134 ":file_player",
136 ":file_recorder", 135 ":file_recorder",
137 "..:webrtc_common", 136 "..:webrtc_common",
138 "../api:audio_mixer_api", 137 "../api:audio_mixer_api",
139 "../api:call_api", 138 "../api:call_api",
140 "../api:libjingle_peerconnection_api", 139 "../api:libjingle_peerconnection_api",
141 "../api:transport_api", 140 "../api:transport_api",
142 "../api/audio_codecs:audio_codecs_api", 141 "../api/audio_codecs:audio_codecs_api",
143 "../api/audio_codecs:builtin_audio_decoder_factory", 142 "../api/audio_codecs:builtin_audio_decoder_factory",
143 "../api/audio_codecs:builtin_audio_encoder_factory",
144 "../audio/utility:audio_frame_operations", 144 "../audio/utility:audio_frame_operations",
145 "../base:rtc_base_approved", 145 "../base:rtc_base_approved",
146 "../base:rtc_task_queue", 146 "../base:rtc_task_queue",
147 "../modules:module_api", 147 "../modules:module_api",
148 148
149 # TODO(nisse): Delete when declaration of RtpTransportController 149 # TODO(nisse): Delete when declaration of RtpTransportController
150 # and related interfaces move to api/. 150 # and related interfaces move to api/.
151 "../call:call_interfaces", 151 "../call:call_interfaces",
152 "../common_audio", 152 "../common_audio",
153 "../logging:rtc_event_log_api", 153 "../logging:rtc_event_log_api",
154 "../modules/audio_coding:audio_encoder_interface",
155 "../modules/audio_coding:audio_format_conversion", 154 "../modules/audio_coding:audio_format_conversion",
156 "../modules/audio_coding:rent_a_codec", 155 "../modules/audio_coding:rent_a_codec",
157 "../modules/audio_conference_mixer", 156 "../modules/audio_conference_mixer",
158 "../modules/audio_device", 157 "../modules/audio_device",
159 "../modules/audio_processing", 158 "../modules/audio_processing",
160 "../modules/bitrate_controller", 159 "../modules/bitrate_controller",
161 "../modules/media_file", 160 "../modules/media_file",
162 "../modules/pacing", 161 "../modules/pacing",
163 "../modules/rtp_rtcp", 162 "../modules/rtp_rtcp",
164 "../modules/utility", 163 "../modules/utility",
(...skipping 143 matching lines...) Expand 10 before | Expand all | Expand 10 after
308 ] 307 ]
309 } 308 }
310 309
311 if (!build_with_chromium && is_clang) { 310 if (!build_with_chromium && is_clang) {
312 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163) . 311 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163) .
313 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] 312 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
314 } 313 }
315 } 314 }
316 } 315 }
317 } 316 }
OLDNEW
« no previous file with comments | « webrtc/test/mock_audio_encoder_factory.h ('k') | webrtc/voice_engine/channel.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698