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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 |
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38 rtc_source_set("rtp_interfaces") { | 38 rtc_source_set("rtp_interfaces") { |
39 sources = [ | 39 sources = [ |
40 "rtcp_packet_sink_interface.h", | 40 "rtcp_packet_sink_interface.h", |
41 "rtp_config.cc", | 41 "rtp_config.cc", |
42 "rtp_config.h", | 42 "rtp_config.h", |
43 "rtp_packet_sink_interface.h", | 43 "rtp_packet_sink_interface.h", |
44 "rtp_stream_receiver_controller_interface.h", | 44 "rtp_stream_receiver_controller_interface.h", |
45 "rtp_transport_controller_send_interface.h", | 45 "rtp_transport_controller_send_interface.h", |
46 ] | 46 ] |
47 deps = [ | 47 deps = [ |
| 48 "../api:array_view", |
48 "../rtc_base:rtc_base_approved", | 49 "../rtc_base:rtc_base_approved", |
49 ] | 50 ] |
50 } | 51 } |
51 | 52 |
52 rtc_source_set("rtp_receiver") { | 53 rtc_source_set("rtp_receiver") { |
53 sources = [ | 54 sources = [ |
54 "rtcp_demuxer.cc", | 55 "rtcp_demuxer.cc", |
55 "rtcp_demuxer.h", | 56 "rtcp_demuxer.h", |
56 "rtp_demuxer.cc", | 57 "rtp_demuxer.cc", |
57 "rtp_demuxer.h", | 58 "rtp_demuxer.h", |
58 "rtp_rtcp_demuxer_helper.cc", | 59 "rtp_rtcp_demuxer_helper.cc", |
59 "rtp_rtcp_demuxer_helper.h", | 60 "rtp_rtcp_demuxer_helper.h", |
60 "rtp_stream_receiver_controller.cc", | 61 "rtp_stream_receiver_controller.cc", |
61 "rtp_stream_receiver_controller.h", | 62 "rtp_stream_receiver_controller.h", |
62 "rtx_receive_stream.cc", | 63 "rtx_receive_stream.cc", |
63 "rtx_receive_stream.h", | 64 "rtx_receive_stream.h", |
64 "ssrc_binding_observer.h", | 65 "ssrc_binding_observer.h", |
65 ] | 66 ] |
66 deps = [ | 67 deps = [ |
67 ":rtp_interfaces", | 68 ":rtp_interfaces", |
68 "..:webrtc_common", | 69 "..:webrtc_common", |
| 70 "../api:array_view", |
69 "../modules/rtp_rtcp", | 71 "../modules/rtp_rtcp", |
70 "../rtc_base:rtc_base_approved", | 72 "../rtc_base:rtc_base_approved", |
71 ] | 73 ] |
72 } | 74 } |
73 | 75 |
74 rtc_source_set("rtp_sender") { | 76 rtc_source_set("rtp_sender") { |
75 sources = [ | 77 sources = [ |
76 "rtp_transport_controller_send.cc", | 78 "rtp_transport_controller_send.cc", |
77 "rtp_transport_controller_send.h", | 79 "rtp_transport_controller_send.h", |
78 ] | 80 ] |
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168 "rtp_rtcp_demuxer_helper_unittest.cc", | 170 "rtp_rtcp_demuxer_helper_unittest.cc", |
169 "rtx_receive_stream_unittest.cc", | 171 "rtx_receive_stream_unittest.cc", |
170 ] | 172 ] |
171 deps = [ | 173 deps = [ |
172 ":call", | 174 ":call", |
173 ":mock_rtp_interfaces", | 175 ":mock_rtp_interfaces", |
174 ":rtp_interfaces", | 176 ":rtp_interfaces", |
175 ":rtp_receiver", | 177 ":rtp_receiver", |
176 ":rtp_sender", | 178 ":rtp_sender", |
177 "..:webrtc_common", | 179 "..:webrtc_common", |
| 180 "../api:array_view", |
178 "../api:mock_audio_mixer", | 181 "../api:mock_audio_mixer", |
179 "../logging:rtc_event_log_api", | 182 "../logging:rtc_event_log_api", |
180 "../modules/audio_device:mock_audio_device", | 183 "../modules/audio_device:mock_audio_device", |
181 "../modules/audio_mixer", | 184 "../modules/audio_mixer", |
182 "../modules/bitrate_controller", | 185 "../modules/bitrate_controller", |
183 "../modules/congestion_controller:mock_congestion_controller", | 186 "../modules/congestion_controller:mock_congestion_controller", |
184 "../modules/pacing", | 187 "../modules/pacing", |
185 "../modules/rtp_rtcp", | 188 "../modules/rtp_rtcp", |
186 "../modules/rtp_rtcp:mock_rtp_rtcp", | 189 "../modules/rtp_rtcp:mock_rtp_rtcp", |
187 "../modules/utility:mock_process_thread", | 190 "../modules/utility:mock_process_thread", |
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250 sources = [ | 253 sources = [ |
251 "test/mock_rtp_packet_sink_interface.h", | 254 "test/mock_rtp_packet_sink_interface.h", |
252 ] | 255 ] |
253 deps = [ | 256 deps = [ |
254 ":rtp_interfaces", | 257 ":rtp_interfaces", |
255 "../test:test_support", | 258 "../test:test_support", |
256 "//testing/gmock", | 259 "//testing/gmock", |
257 ] | 260 ] |
258 } | 261 } |
259 } | 262 } |
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