Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(368)

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

Issue 2452643002: Revert of Clean up logging in AudioSendStream::SetupSendCodec(). (Closed)
Patch Set: Created 4 years, 1 month 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 | « no previous file | webrtc/api/call/audio_send_stream.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) 2015 The WebRTC project authors. All Rights Reserved. 1 # Copyright (c) 2015 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/webrtc.gni") 9 import("../build/webrtc.gni")
10 if (is_android) { 10 if (is_android) {
11 import("//build/config/android/config.gni") 11 import("//build/config/android/config.gni")
12 import("//build/config/android/rules.gni") 12 import("//build/config/android/rules.gni")
13 } 13 }
14 14
15 group("api") { 15 group("api") {
16 public_deps = [ 16 public_deps = [
17 ":libjingle_peerconnection", 17 ":libjingle_peerconnection",
18 ] 18 ]
19 } 19 }
20 20
21 rtc_source_set("call_api") { 21 rtc_source_set("call_api") {
22 sources = [ 22 sources = [
23 "call/audio_receive_stream.h", 23 "call/audio_receive_stream.h",
24 "call/audio_send_stream.cc",
25 "call/audio_send_stream.h", 24 "call/audio_send_stream.h",
26 "call/audio_sink.h", 25 "call/audio_sink.h",
27 "call/audio_state.h", 26 "call/audio_state.h",
28 "call/flexfec_receive_stream.h", 27 "call/flexfec_receive_stream.h",
29 ] 28 ]
30 29
31 deps = [ 30 deps = [
32 # TODO(kjellander): Add remaining dependencies when webrtc:4243 is done. 31 # TODO(kjellander): Add remaining dependencies when webrtc:4243 is done.
33 "..:webrtc_common", 32 "..:webrtc_common",
34 "../base:rtc_base_approved", 33 "../base:rtc_base_approved",
(...skipping 450 matching lines...) Expand 10 before | Expand all | Expand 10 after
485 484
486 shared_libraries = [ ":libjingle_peerconnection_so" ] 485 shared_libraries = [ ":libjingle_peerconnection_so" ]
487 } 486 }
488 487
489 android_resources("libjingle_peerconnection_android_unittest_resources") { 488 android_resources("libjingle_peerconnection_android_unittest_resources") {
490 resource_dirs = [ "androidtests/res" ] 489 resource_dirs = [ "androidtests/res" ]
491 custom_package = "org.webrtc" 490 custom_package = "org.webrtc"
492 } 491 }
493 } 492 }
494 } 493 }
OLDNEW
« no previous file with comments | « no previous file | webrtc/api/call/audio_send_stream.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698