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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 "voice_engine/channel.h" | 11 #include "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 "api/array_view.h" | 19 #include "api/array_view.h" |
17 #include "audio/utility/audio_frame_operations.h" | 20 #include "audio/utility/audio_frame_operations.h" |
18 #include "call/rtp_transport_controller_send_interface.h" | 21 #include "call/rtp_transport_controller_send_interface.h" |
19 #include "logging/rtc_event_log/rtc_event_log.h" | 22 #include "logging/rtc_event_log/rtc_event_log.h" |
20 #include "modules/audio_coding/codecs/audio_format_conversion.h" | 23 #include "modules/audio_coding/codecs/audio_format_conversion.h" |
21 #include "modules/audio_device/include/audio_device.h" | 24 #include "modules/audio_device/include/audio_device.h" |
22 #include "modules/audio_processing/include/audio_processing.h" | 25 #include "modules/audio_processing/include/audio_processing.h" |
23 #include "modules/include/module_common_types.h" | 26 #include "modules/include/module_common_types.h" |
24 #include "modules/pacing/packet_router.h" | 27 #include "modules/pacing/packet_router.h" |
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91 } | 94 } |
92 | 95 |
93 void LogAudioSendStreamConfig( | 96 void LogAudioSendStreamConfig( |
94 const webrtc::rtclog::StreamConfig& config) override { | 97 const webrtc::rtclog::StreamConfig& config) override { |
95 rtc::CritScope lock(&crit_); | 98 rtc::CritScope lock(&crit_); |
96 if (event_log_) { | 99 if (event_log_) { |
97 event_log_->LogAudioSendStreamConfig(config); | 100 event_log_->LogAudioSendStreamConfig(config); |
98 } | 101 } |
99 } | 102 } |
100 | 103 |
101 void LogRtpHeader(webrtc::PacketDirection direction, | 104 void LogIncomingRtpHeader(const RtpPacketReceived& packet) override { |
102 const uint8_t* header, | |
103 size_t packet_length) override { | |
104 LogRtpHeader(direction, header, packet_length, PacedPacketInfo::kNotAProbe); | |
105 } | |
106 | |
107 void LogRtpHeader(webrtc::PacketDirection direction, | |
108 const uint8_t* header, | |
109 size_t packet_length, | |
110 int probe_cluster_id) override { | |
111 rtc::CritScope lock(&crit_); | 105 rtc::CritScope lock(&crit_); |
112 if (event_log_) { | 106 if (event_log_) { |
113 event_log_->LogRtpHeader(direction, header, packet_length, | 107 event_log_->LogIncomingRtpHeader(packet); |
114 probe_cluster_id); | |
115 } | 108 } |
116 } | 109 } |
117 | 110 |
118 void LogRtcpPacket(webrtc::PacketDirection direction, | 111 void LogOutgoingRtpHeader(const RtpPacketToSend& packet, |
119 const uint8_t* packet, | 112 int probe_cluster_id) override { |
120 size_t length) override { | |
121 rtc::CritScope lock(&crit_); | 113 rtc::CritScope lock(&crit_); |
122 if (event_log_) { | 114 if (event_log_) { |
123 event_log_->LogRtcpPacket(direction, packet, length); | 115 event_log_->LogOutgoingRtpHeader(packet, probe_cluster_id); |
124 } | 116 } |
125 } | 117 } |
126 | 118 |
| 119 void LogIncomingRtcpPacket(rtc::ArrayView<const uint8_t> packet) override { |
| 120 rtc::CritScope lock(&crit_); |
| 121 if (event_log_) { |
| 122 event_log_->LogIncomingRtcpPacket(packet); |
| 123 } |
| 124 } |
| 125 |
| 126 void LogOutgoingRtcpPacket(rtc::ArrayView<const uint8_t> packet) override { |
| 127 rtc::CritScope lock(&crit_); |
| 128 if (event_log_) { |
| 129 event_log_->LogOutgoingRtcpPacket(packet); |
| 130 } |
| 131 } |
| 132 |
127 void LogAudioPlayout(uint32_t ssrc) override { | 133 void LogAudioPlayout(uint32_t ssrc) override { |
128 rtc::CritScope lock(&crit_); | 134 rtc::CritScope lock(&crit_); |
129 if (event_log_) { | 135 if (event_log_) { |
130 event_log_->LogAudioPlayout(ssrc); | 136 event_log_->LogAudioPlayout(ssrc); |
131 } | 137 } |
132 } | 138 } |
133 | 139 |
134 void LogLossBasedBweUpdate(int32_t bitrate_bps, | 140 void LogLossBasedBweUpdate(int32_t bitrate_bps, |
135 uint8_t fraction_loss, | 141 uint8_t fraction_loss, |
136 int32_t total_packets) override { | 142 int32_t total_packets) override { |
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2029 int64_t min_rtt = 0; | 2035 int64_t min_rtt = 0; |
2030 if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) != | 2036 if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) != |
2031 0) { | 2037 0) { |
2032 return 0; | 2038 return 0; |
2033 } | 2039 } |
2034 return rtt; | 2040 return rtt; |
2035 } | 2041 } |
2036 | 2042 |
2037 } // namespace voe | 2043 } // namespace voe |
2038 } // namespace webrtc | 2044 } // namespace webrtc |
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