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1 /* | 1 /* |
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2013 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 #include <algorithm> | 10 #include <algorithm> |
| 11 #include <memory> |
11 #include <sstream> | 12 #include <sstream> |
12 #include <string> | 13 #include <string> |
13 | 14 |
14 #include "testing/gtest/include/gtest/gtest.h" | 15 #include "testing/gtest/include/gtest/gtest.h" |
15 | 16 |
16 #include "webrtc/base/checks.h" | 17 #include "webrtc/base/checks.h" |
17 #include "webrtc/base/scoped_ptr.h" | |
18 #include "webrtc/base/thread_annotations.h" | 18 #include "webrtc/base/thread_annotations.h" |
19 #include "webrtc/call.h" | 19 #include "webrtc/call.h" |
20 #include "webrtc/call/transport_adapter.h" | 20 #include "webrtc/call/transport_adapter.h" |
21 #include "webrtc/config.h" | 21 #include "webrtc/config.h" |
22 #include "webrtc/modules/audio_coding/include/audio_coding_module.h" | 22 #include "webrtc/modules/audio_coding/include/audio_coding_module.h" |
23 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" | 23 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" |
24 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" | 24 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" |
25 #include "webrtc/system_wrappers/include/critical_section_wrapper.h" | 25 #include "webrtc/system_wrappers/include/critical_section_wrapper.h" |
26 #include "webrtc/system_wrappers/include/rtp_to_ntp.h" | 26 #include "webrtc/system_wrappers/include/rtp_to_ntp.h" |
27 #include "webrtc/test/call_test.h" | 27 #include "webrtc/test/call_test.h" |
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228 } else { | 228 } else { |
229 ret = voe_network_->ReceivedRTPPacket(channel_, packet, length, | 229 ret = voe_network_->ReceivedRTPPacket(channel_, packet, length, |
230 PacketTime()); | 230 PacketTime()); |
231 } | 231 } |
232 return ret == 0 ? DELIVERY_OK : DELIVERY_PACKET_ERROR; | 232 return ret == 0 ? DELIVERY_OK : DELIVERY_PACKET_ERROR; |
233 } | 233 } |
234 | 234 |
235 private: | 235 private: |
236 int channel_; | 236 int channel_; |
237 VoENetwork* voe_network_; | 237 VoENetwork* voe_network_; |
238 rtc::scoped_ptr<RtpHeaderParser> parser_; | 238 std::unique_ptr<RtpHeaderParser> parser_; |
239 }; | 239 }; |
240 | 240 |
241 VoiceEngine* voice_engine = VoiceEngine::Create(); | 241 VoiceEngine* voice_engine = VoiceEngine::Create(); |
242 VoEBase* voe_base = VoEBase::GetInterface(voice_engine); | 242 VoEBase* voe_base = VoEBase::GetInterface(voice_engine); |
243 VoECodec* voe_codec = VoECodec::GetInterface(voice_engine); | 243 VoECodec* voe_codec = VoECodec::GetInterface(voice_engine); |
244 VoENetwork* voe_network = VoENetwork::GetInterface(voice_engine); | 244 VoENetwork* voe_network = VoENetwork::GetInterface(voice_engine); |
245 VoEVideoSync* voe_sync = VoEVideoSync::GetInterface(voice_engine); | 245 VoEVideoSync* voe_sync = VoEVideoSync::GetInterface(voice_engine); |
246 const std::string audio_filename = | 246 const std::string audio_filename = |
247 test::ResourcePath("voice_engine/audio_long16", "pcm"); | 247 test::ResourcePath("voice_engine/audio_long16", "pcm"); |
248 ASSERT_STRNE("", audio_filename.c_str()); | 248 ASSERT_STRNE("", audio_filename.c_str()); |
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776 int encoder_inits_; | 776 int encoder_inits_; |
777 uint32_t last_set_bitrate_; | 777 uint32_t last_set_bitrate_; |
778 VideoSendStream* send_stream_; | 778 VideoSendStream* send_stream_; |
779 VideoEncoderConfig encoder_config_; | 779 VideoEncoderConfig encoder_config_; |
780 } test; | 780 } test; |
781 | 781 |
782 RunBaseTest(&test); | 782 RunBaseTest(&test); |
783 } | 783 } |
784 | 784 |
785 } // namespace webrtc | 785 } // namespace webrtc |
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