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1 /* | 1 /* |
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
11 #include <string> | 11 #include <string> |
12 #include <utility> | 12 #include <utility> |
13 #include <vector> | 13 #include <vector> |
14 | 14 |
15 #include "webrtc/audio/audio_send_stream.h" | 15 #include "webrtc/audio/audio_send_stream.h" |
16 #include "webrtc/audio/audio_state.h" | 16 #include "webrtc/audio/audio_state.h" |
17 #include "webrtc/audio/conversion.h" | 17 #include "webrtc/audio/conversion.h" |
18 #include "webrtc/base/task_queue.h" | 18 #include "webrtc/base/task_queue.h" |
19 #include "webrtc/call/rtp_transport_controller_send_interface.h" | 19 #include "webrtc/call/rtp_transport_controller_send_interface.h" |
20 #include "webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h" | 20 #include "webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h" |
21 #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h" | |
22 #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder_factory.h" | |
23 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h" | 21 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h" |
24 #include "webrtc/modules/audio_processing/include/mock_audio_processing.h" | 22 #include "webrtc/modules/audio_processing/include/mock_audio_processing.h" |
25 #include "webrtc/modules/congestion_controller/include/mock/mock_congestion_obse
rver.h" | 23 #include "webrtc/modules/congestion_controller/include/mock/mock_congestion_obse
rver.h" |
26 #include "webrtc/modules/congestion_controller/include/send_side_congestion_cont
roller.h" | 24 #include "webrtc/modules/congestion_controller/include/send_side_congestion_cont
roller.h" |
27 #include "webrtc/modules/pacing/paced_sender.h" | 25 #include "webrtc/modules/pacing/paced_sender.h" |
28 #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h" | 26 #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h" |
29 #include "webrtc/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h" | 27 #include "webrtc/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h" |
30 #include "webrtc/test/gtest.h" | 28 #include "webrtc/test/gtest.h" |
| 29 #include "webrtc/test/mock_audio_encoder.h" |
| 30 #include "webrtc/test/mock_audio_encoder_factory.h" |
31 #include "webrtc/test/mock_voe_channel_proxy.h" | 31 #include "webrtc/test/mock_voe_channel_proxy.h" |
32 #include "webrtc/test/mock_voice_engine.h" | 32 #include "webrtc/test/mock_voice_engine.h" |
33 #include "webrtc/voice_engine/transmit_mixer.h" | 33 #include "webrtc/voice_engine/transmit_mixer.h" |
34 | 34 |
35 namespace webrtc { | 35 namespace webrtc { |
36 namespace test { | 36 namespace test { |
37 namespace { | 37 namespace { |
38 | 38 |
39 using testing::_; | 39 using testing::_; |
40 using testing::Eq; | 40 using testing::Eq; |
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554 stream_config.send_codec_spec->cng_payload_type = rtc::Optional<int>(105); | 554 stream_config.send_codec_spec->cng_payload_type = rtc::Optional<int>(105); |
555 internal::AudioSendStream send_stream( | 555 internal::AudioSendStream send_stream( |
556 stream_config, helper.audio_state(), helper.worker_queue(), | 556 stream_config, helper.audio_state(), helper.worker_queue(), |
557 helper.transport(), helper.bitrate_allocator(), helper.event_log(), | 557 helper.transport(), helper.bitrate_allocator(), helper.event_log(), |
558 helper.rtcp_rtt_stats()); | 558 helper.rtcp_rtt_stats()); |
559 send_stream.Reconfigure(stream_config); | 559 send_stream.Reconfigure(stream_config); |
560 } | 560 } |
561 | 561 |
562 } // namespace test | 562 } // namespace test |
563 } // namespace webrtc | 563 } // namespace webrtc |
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