Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(499)

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

Issue 2709723003: Initial implementation of RtpTransportControllerReceive and related interfaces.
Patch Set: Rename foo_audio --> audio_foo. 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/audio/audio_receive_stream_unittest.cc ('k') | webrtc/call/call.cc » ('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("../webrtc.gni") 9 import("../webrtc.gni")
10 10
11 rtc_source_set("call_interfaces") { 11 rtc_source_set("call_interfaces") {
12 sources = [ 12 sources = [
13 "audio_receive_stream.h", 13 "audio_receive_stream.h",
14 "audio_send_stream.cc", 14 "audio_send_stream.cc",
15 "audio_send_stream.h", 15 "audio_send_stream.h",
16 "audio_state.h", 16 "audio_state.h",
17 "call.h", 17 "call.h",
18 "flexfec_receive_stream.h", 18 "flexfec_receive_stream.h",
19 "rtp_transport_controller_receive.h",
19 "rtp_transport_controller_send_interface.h", 20 "rtp_transport_controller_send_interface.h",
20 "syncable.cc", 21 "syncable.cc",
21 "syncable.h", 22 "syncable.h",
22 ] 23 ]
23 deps = [ 24 deps = [
24 "..:webrtc_common", 25 "..:webrtc_common",
25 "../api:audio_mixer_api", 26 "../api:audio_mixer_api",
26 "../api:libjingle_peerconnection_api", 27 "../api:libjingle_peerconnection_api",
27 "../api:transport_api", 28 "../api:transport_api",
28 "../api/audio_codecs:audio_codecs_api", 29 "../api/audio_codecs:audio_codecs_api",
29 "../base:rtc_base", 30 "../base:rtc_base",
30 "../base:rtc_base_approved", 31 "../base:rtc_base_approved",
31 "../modules/audio_coding:audio_encoder_interface", 32 "../modules/audio_coding:audio_encoder_interface",
32 ] 33 ]
33 } 34 }
34 35
35 rtc_static_library("call") { 36 rtc_static_library("call") {
36 sources = [ 37 sources = [
37 "bitrate_allocator.cc", 38 "bitrate_allocator.cc",
38 "call.cc", 39 "call.cc",
39 "flexfec_receive_stream_impl.cc", 40 "flexfec_receive_stream_impl.cc",
40 "flexfec_receive_stream_impl.h", 41 "flexfec_receive_stream_impl.h",
42 "rtp_transport_controller_receive.cc",
41 "rtp_transport_controller_send.cc", 43 "rtp_transport_controller_send.cc",
42 "rtp_transport_controller_send.h", 44 "rtp_transport_controller_send.h",
43 ] 45 ]
44 46
45 if (!build_with_chromium && is_clang) { 47 if (!build_with_chromium && is_clang) {
46 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). 48 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
47 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] 49 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
48 } 50 }
49 51
50 public_deps = [ 52 public_deps = [
(...skipping 92 matching lines...) Expand 10 before | Expand all | Expand 10 after
143 "//testing/gtest", 145 "//testing/gtest",
144 "//webrtc/test:field_trial", 146 "//webrtc/test:field_trial",
145 "//webrtc/test:test_common", 147 "//webrtc/test:test_common",
146 ] 148 ]
147 if (!build_with_chromium && is_clang) { 149 if (!build_with_chromium && is_clang) {
148 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). 150 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
149 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] 151 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
150 } 152 }
151 } 153 }
152 } 154 }
OLDNEW
« no previous file with comments | « webrtc/audio/audio_receive_stream_unittest.cc ('k') | webrtc/call/call.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698