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1 /* | 1 /* |
2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2016 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 |
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87 } | 87 } |
88 | 88 |
89 if (metrics.target_audio_bitrate_bps) { | 89 if (metrics.target_audio_bitrate_bps) { |
90 dump_metrics->set_target_audio_bitrate_bps( | 90 dump_metrics->set_target_audio_bitrate_bps( |
91 *metrics.target_audio_bitrate_bps); | 91 *metrics.target_audio_bitrate_bps); |
92 } | 92 } |
93 | 93 |
94 if (metrics.rtt_ms) | 94 if (metrics.rtt_ms) |
95 dump_metrics->set_rtt_ms(*metrics.rtt_ms); | 95 dump_metrics->set_rtt_ms(*metrics.rtt_ms); |
96 | 96 |
| 97 if (metrics.uplink_recoverable_packet_loss_fraction) { |
| 98 dump_metrics->set_uplink_recoverable_packet_loss_fraction( |
| 99 *metrics.uplink_recoverable_packet_loss_fraction); |
| 100 } |
| 101 |
97 DumpEventToFile(event, dump_file_.get()); | 102 DumpEventToFile(event, dump_file_.get()); |
98 #endif // WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP | 103 #endif // WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP |
99 } | 104 } |
100 | 105 |
101 void DebugDumpWriterImpl::DumpEncoderRuntimeConfig( | 106 void DebugDumpWriterImpl::DumpEncoderRuntimeConfig( |
102 const AudioNetworkAdaptor::EncoderRuntimeConfig& config, | 107 const AudioNetworkAdaptor::EncoderRuntimeConfig& config, |
103 int64_t timestamp) { | 108 int64_t timestamp) { |
104 #ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP | 109 #ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP |
105 Event event; | 110 Event event; |
106 event.set_timestamp(timestamp); | 111 event.set_timestamp(timestamp); |
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129 | 134 |
130 DumpEventToFile(event, dump_file_.get()); | 135 DumpEventToFile(event, dump_file_.get()); |
131 #endif // WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP | 136 #endif // WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP |
132 } | 137 } |
133 | 138 |
134 std::unique_ptr<DebugDumpWriter> DebugDumpWriter::Create(FILE* file_handle) { | 139 std::unique_ptr<DebugDumpWriter> DebugDumpWriter::Create(FILE* file_handle) { |
135 return std::unique_ptr<DebugDumpWriter>(new DebugDumpWriterImpl(file_handle)); | 140 return std::unique_ptr<DebugDumpWriter>(new DebugDumpWriterImpl(file_handle)); |
136 } | 141 } |
137 | 142 |
138 } // namespace webrtc | 143 } // namespace webrtc |
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