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Issue 3000273002: Reverse |rtx_payload_types| map, and rename. (Closed)
Patch Set: Created 3 years, 4 months ago
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1 /* 1 /*
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2013 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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80 } 80 }
81 81
82 bool UseSendSideBwe(const AudioReceiveStream::Config& config) { 82 bool UseSendSideBwe(const AudioReceiveStream::Config& config) {
83 return UseSendSideBwe(config.rtp.extensions, config.rtp.transport_cc); 83 return UseSendSideBwe(config.rtp.extensions, config.rtp.transport_cc);
84 } 84 }
85 85
86 bool UseSendSideBwe(const FlexfecReceiveStream::Config& config) { 86 bool UseSendSideBwe(const FlexfecReceiveStream::Config& config) {
87 return UseSendSideBwe(config.rtp_header_extensions, config.transport_cc); 87 return UseSendSideBwe(config.rtp_header_extensions, config.transport_cc);
88 } 88 }
89 89
90 template <typename K, typename V>
danilchap 2017/08/21 14:11:26 may be do not use template until you'll use it for
nisse-webrtc 2017/08/22 07:11:01 Done.
91 const K* find_by_value(const std::map<K, V>& m, V v) {
danilchap 2017/08/21 14:11:26 FindByValue https://google.github.io/styleguide/cp
nisse-webrtc 2017/08/22 07:11:01 Done.
92 for (const auto& kv : m) {
93 if (kv.second == v)
94 return &kv.first;
95 }
96 return nullptr;
97 }
98
90 rtclog::StreamConfig CreateRtcLogStreamConfig( 99 rtclog::StreamConfig CreateRtcLogStreamConfig(
91 const VideoReceiveStream::Config& config) { 100 const VideoReceiveStream::Config& config) {
92 rtclog::StreamConfig rtclog_config; 101 rtclog::StreamConfig rtclog_config;
93 rtclog_config.remote_ssrc = config.rtp.remote_ssrc; 102 rtclog_config.remote_ssrc = config.rtp.remote_ssrc;
94 rtclog_config.local_ssrc = config.rtp.local_ssrc; 103 rtclog_config.local_ssrc = config.rtp.local_ssrc;
95 rtclog_config.rtx_ssrc = config.rtp.rtx_ssrc; 104 rtclog_config.rtx_ssrc = config.rtp.rtx_ssrc;
96 rtclog_config.rtcp_mode = config.rtp.rtcp_mode; 105 rtclog_config.rtcp_mode = config.rtp.rtcp_mode;
97 rtclog_config.remb = config.rtp.remb; 106 rtclog_config.remb = config.rtp.remb;
98 rtclog_config.rtp_extensions = config.rtp.extensions; 107 rtclog_config.rtp_extensions = config.rtp.extensions;
99 108
100 for (const auto& d : config.decoders) { 109 for (const auto& d : config.decoders) {
101 auto search = config.rtp.rtx_payload_types.find(d.payload_type); 110 auto search = find_by_value(config.rtp.media_pt_by_rtx_pt, d.payload_type);
danilchap 2017/08/21 14:11:26 since search is no longer a long-named iterator, u
nisse-webrtc 2017/08/22 07:11:01 Done.
102 rtclog_config.codecs.emplace_back( 111 rtclog_config.codecs.emplace_back(d.payload_name, d.payload_type,
103 d.payload_name, d.payload_type, 112 search ? *search : 0);
104 search != config.rtp.rtx_payload_types.end() ? search->second : 0);
105 } 113 }
106 return rtclog_config; 114 return rtclog_config;
107 } 115 }
108 116
109 rtclog::StreamConfig CreateRtcLogStreamConfig( 117 rtclog::StreamConfig CreateRtcLogStreamConfig(
110 const VideoSendStream::Config& config, 118 const VideoSendStream::Config& config,
111 size_t ssrc_index) { 119 size_t ssrc_index) {
112 rtclog::StreamConfig rtclog_config; 120 rtclog::StreamConfig rtclog_config;
113 rtclog_config.local_ssrc = config.rtp.ssrcs[ssrc_index]; 121 rtclog_config.local_ssrc = config.rtp.ssrcs[ssrc_index];
114 if (ssrc_index < config.rtp.rtx.ssrcs.size()) { 122 if (ssrc_index < config.rtp.rtx.ssrcs.size()) {
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1422 (use_send_side_bwe && header.extension.hasTransportSequenceNumber)) { 1430 (use_send_side_bwe && header.extension.hasTransportSequenceNumber)) {
1423 receive_side_cc_.OnReceivedPacket( 1431 receive_side_cc_.OnReceivedPacket(
1424 packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(), 1432 packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(),
1425 header); 1433 header);
1426 } 1434 }
1427 } 1435 }
1428 1436
1429 } // namespace internal 1437 } // namespace internal
1430 1438
1431 } // namespace webrtc 1439 } // namespace webrtc
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