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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 <map> |
| 15 #include <string> |
14 #include <utility> | 16 #include <utility> |
| 17 #include <vector> |
15 | 18 |
16 #include "webrtc/api/array_view.h" | 19 #include "webrtc/api/array_view.h" |
17 #include "webrtc/audio/utility/audio_frame_operations.h" | 20 #include "webrtc/audio/utility/audio_frame_operations.h" |
18 #include "webrtc/call/rtp_transport_controller_send_interface.h" | 21 #include "webrtc/call/rtp_transport_controller_send_interface.h" |
19 #include "webrtc/logging/rtc_event_log/rtc_event_log.h" | 22 #include "webrtc/logging/rtc_event_log/rtc_event_log.h" |
20 #include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h" | 23 #include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h" |
21 #include "webrtc/modules/audio_device/include/audio_device.h" | 24 #include "webrtc/modules/audio_device/include/audio_device.h" |
22 #include "webrtc/modules/audio_processing/include/audio_processing.h" | 25 #include "webrtc/modules/audio_processing/include/audio_processing.h" |
23 #include "webrtc/modules/include/module_common_types.h" | 26 #include "webrtc/modules/include/module_common_types.h" |
24 #include "webrtc/modules/pacing/packet_router.h" | 27 #include "webrtc/modules/pacing/packet_router.h" |
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93 } | 96 } |
94 | 97 |
95 void LogAudioSendStreamConfig( | 98 void LogAudioSendStreamConfig( |
96 const webrtc::rtclog::StreamConfig& config) override { | 99 const webrtc::rtclog::StreamConfig& config) override { |
97 rtc::CritScope lock(&crit_); | 100 rtc::CritScope lock(&crit_); |
98 if (event_log_) { | 101 if (event_log_) { |
99 event_log_->LogAudioSendStreamConfig(config); | 102 event_log_->LogAudioSendStreamConfig(config); |
100 } | 103 } |
101 } | 104 } |
102 | 105 |
103 void LogRtpHeader(webrtc::PacketDirection direction, | 106 void LogIncomingRtpHeader(const RtpPacketReceived& packet) override { |
104 const uint8_t* header, | |
105 size_t packet_length) override { | |
106 LogRtpHeader(direction, header, packet_length, PacedPacketInfo::kNotAProbe); | |
107 } | |
108 | |
109 void LogRtpHeader(webrtc::PacketDirection direction, | |
110 const uint8_t* header, | |
111 size_t packet_length, | |
112 int probe_cluster_id) override { | |
113 rtc::CritScope lock(&crit_); | 107 rtc::CritScope lock(&crit_); |
114 if (event_log_) { | 108 if (event_log_) { |
115 event_log_->LogRtpHeader(direction, header, packet_length, | 109 event_log_->LogIncomingRtpHeader(packet); |
116 probe_cluster_id); | |
117 } | 110 } |
118 } | 111 } |
119 | 112 |
120 void LogRtcpPacket(webrtc::PacketDirection direction, | 113 void LogOutgoingRtpHeader(const RtpPacketToSend& packet, |
121 const uint8_t* packet, | 114 int probe_cluster_id) override { |
122 size_t length) override { | |
123 rtc::CritScope lock(&crit_); | 115 rtc::CritScope lock(&crit_); |
124 if (event_log_) { | 116 if (event_log_) { |
125 event_log_->LogRtcpPacket(direction, packet, length); | 117 event_log_->LogOutgoingRtpHeader(packet, probe_cluster_id); |
126 } | 118 } |
127 } | 119 } |
128 | 120 |
| 121 void LogIncomingRtcpPacket(rtc::ArrayView<const uint8_t> packet) override { |
| 122 rtc::CritScope lock(&crit_); |
| 123 if (event_log_) { |
| 124 event_log_->LogIncomingRtcpPacket(packet); |
| 125 } |
| 126 } |
| 127 |
| 128 void LogOutgoingRtcpPacket(rtc::ArrayView<const uint8_t> packet) override { |
| 129 rtc::CritScope lock(&crit_); |
| 130 if (event_log_) { |
| 131 event_log_->LogOutgoingRtcpPacket(packet); |
| 132 } |
| 133 } |
| 134 |
129 void LogAudioPlayout(uint32_t ssrc) override { | 135 void LogAudioPlayout(uint32_t ssrc) override { |
130 rtc::CritScope lock(&crit_); | 136 rtc::CritScope lock(&crit_); |
131 if (event_log_) { | 137 if (event_log_) { |
132 event_log_->LogAudioPlayout(ssrc); | 138 event_log_->LogAudioPlayout(ssrc); |
133 } | 139 } |
134 } | 140 } |
135 | 141 |
136 void LogLossBasedBweUpdate(int32_t bitrate_bps, | 142 void LogLossBasedBweUpdate(int32_t bitrate_bps, |
137 uint8_t fraction_loss, | 143 uint8_t fraction_loss, |
138 int32_t total_packets) override { | 144 int32_t total_packets) override { |
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3132 int64_t min_rtt = 0; | 3138 int64_t min_rtt = 0; |
3133 if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) != | 3139 if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) != |
3134 0) { | 3140 0) { |
3135 return 0; | 3141 return 0; |
3136 } | 3142 } |
3137 return rtt; | 3143 return rtt; |
3138 } | 3144 } |
3139 | 3145 |
3140 } // namespace voe | 3146 } // namespace voe |
3141 } // namespace webrtc | 3147 } // namespace webrtc |
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