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| 1 /* | 1 /* |
| 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 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 "webrtc/voice_engine/channel.h" | 11 #include "webrtc/voice_engine/channel.h" |
| 12 | 12 |
| 13 #include <algorithm> | 13 #include <algorithm> |
| 14 #include <utility> | 14 #include <utility> |
| 15 | 15 |
| 16 #include "webrtc/audio/utility/audio_frame_operations.h" | 16 #include "webrtc/audio/utility/audio_frame_operations.h" |
| 17 #include "webrtc/base/array_view.h" | |
| 18 #include "webrtc/base/checks.h" | |
| 19 #include "webrtc/base/criticalsection.h" | |
| 20 #include "webrtc/base/format_macros.h" | |
| 21 #include "webrtc/base/location.h" | |
| 22 #include "webrtc/base/logging.h" | |
| 23 #include "webrtc/base/rate_limiter.h" | |
| 24 #include "webrtc/base/task_queue.h" | |
| 25 #include "webrtc/base/thread_checker.h" | |
| 26 #include "webrtc/base/timeutils.h" | |
| 27 #include "webrtc/call/rtp_transport_controller_send_interface.h" | 17 #include "webrtc/call/rtp_transport_controller_send_interface.h" |
| 28 #include "webrtc/config.h" | 18 #include "webrtc/config.h" |
| 29 #include "webrtc/logging/rtc_event_log/rtc_event_log.h" | 19 #include "webrtc/logging/rtc_event_log/rtc_event_log.h" |
| 30 #include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h" | 20 #include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h" |
| 31 #include "webrtc/modules/audio_device/include/audio_device.h" | 21 #include "webrtc/modules/audio_device/include/audio_device.h" |
| 32 #include "webrtc/modules/audio_processing/include/audio_processing.h" | 22 #include "webrtc/modules/audio_processing/include/audio_processing.h" |
| 33 #include "webrtc/modules/include/module_common_types.h" | 23 #include "webrtc/modules/include/module_common_types.h" |
| 34 #include "webrtc/modules/pacing/packet_router.h" | 24 #include "webrtc/modules/pacing/packet_router.h" |
| 35 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h" | 25 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h" |
| 36 #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h" | 26 #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h" |
| 37 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h" | 27 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h" |
| 38 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h" | 28 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h" |
| 39 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h" | 29 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h" |
| 40 #include "webrtc/modules/utility/include/process_thread.h" | 30 #include "webrtc/modules/utility/include/process_thread.h" |
| 31 #include "webrtc/rtc_base/array_view.h" |
| 32 #include "webrtc/rtc_base/checks.h" |
| 33 #include "webrtc/rtc_base/criticalsection.h" |
| 34 #include "webrtc/rtc_base/format_macros.h" |
| 35 #include "webrtc/rtc_base/location.h" |
| 36 #include "webrtc/rtc_base/logging.h" |
| 37 #include "webrtc/rtc_base/rate_limiter.h" |
| 38 #include "webrtc/rtc_base/task_queue.h" |
| 39 #include "webrtc/rtc_base/thread_checker.h" |
| 40 #include "webrtc/rtc_base/timeutils.h" |
| 41 #include "webrtc/system_wrappers/include/field_trial.h" | 41 #include "webrtc/system_wrappers/include/field_trial.h" |
| 42 #include "webrtc/system_wrappers/include/trace.h" | 42 #include "webrtc/system_wrappers/include/trace.h" |
| 43 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h" | 43 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h" |
| 44 #include "webrtc/voice_engine/output_mixer.h" | 44 #include "webrtc/voice_engine/output_mixer.h" |
| 45 #include "webrtc/voice_engine/statistics.h" | 45 #include "webrtc/voice_engine/statistics.h" |
| 46 #include "webrtc/voice_engine/utility.h" | 46 #include "webrtc/voice_engine/utility.h" |
| 47 | 47 |
| 48 namespace webrtc { | 48 namespace webrtc { |
| 49 namespace voe { | 49 namespace voe { |
| 50 | 50 |
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| 3140 int64_t min_rtt = 0; | 3140 int64_t min_rtt = 0; |
| 3141 if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) != | 3141 if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) != |
| 3142 0) { | 3142 0) { |
| 3143 return 0; | 3143 return 0; |
| 3144 } | 3144 } |
| 3145 return rtt; | 3145 return rtt; |
| 3146 } | 3146 } |
| 3147 | 3147 |
| 3148 } // namespace voe | 3148 } // namespace voe |
| 3149 } // namespace webrtc | 3149 } // namespace webrtc |
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