Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(125)

Side by Side Diff: webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc

Issue 2661043003: Allow ANA to receive RPLR (recoverable packet loss rate) indications (Closed)
Patch Set: Uncomment thread-checker to fix UT Created 3 years, 9 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
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
(...skipping 76 matching lines...) Expand 10 before | Expand all | Expand 10 after
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);
(...skipping 22 matching lines...) Expand all
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
OLDNEW

Powered by Google App Engine
This is Rietveld 408576698