audio_send_stream.cc (3737B)
1 /* 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 3 * 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 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #include "call/audio_send_stream.h" 12 13 #include <cstddef> 14 #include <string> 15 16 #include "absl/strings/str_cat.h" 17 #include "api/audio_codecs/audio_format.h" 18 #include "api/call/transport.h" 19 #include "rtc_base/strings/string_builder.h" 20 21 namespace webrtc { 22 23 AudioSendStream::Stats::Stats() = default; 24 AudioSendStream::Stats::~Stats() = default; 25 26 AudioSendStream::Config::Config(Transport* send_transport) 27 : send_transport(send_transport) {} 28 29 AudioSendStream::Config::~Config() = default; 30 31 std::string AudioSendStream::Config::ToString() const { 32 StringBuilder ss; 33 ss << "{rtp: " << rtp.ToString(); 34 ss << ", rtcp_report_interval_ms: " << rtcp_report_interval_ms; 35 ss << ", send_transport: " << (send_transport ? "(Transport)" : "null"); 36 ss << ", min_bitrate_bps: " << min_bitrate_bps; 37 ss << ", max_bitrate_bps: " << max_bitrate_bps; 38 ss << ", has audio_network_adaptor_config: " 39 << (audio_network_adaptor_config ? "true" : "false"); 40 ss << ", has_dscp: " << (has_dscp ? "true" : "false"); 41 ss << ", send_codec_spec: " 42 << (send_codec_spec ? send_codec_spec->ToString() : "<unset>"); 43 ss << "}"; 44 return ss.Release(); 45 } 46 47 AudioSendStream::Config::Rtp::Rtp() = default; 48 49 AudioSendStream::Config::Rtp::~Rtp() = default; 50 51 std::string AudioSendStream::Config::Rtp::ToString() const { 52 char buf[1024]; 53 SimpleStringBuilder ss(buf); 54 ss << "{ssrc: " << ssrc; 55 if (!rid.empty()) { 56 ss << ", rid: " << rid; 57 } 58 if (!mid.empty()) { 59 ss << ", mid: " << mid; 60 } 61 ss << ", csrcs: ["; 62 for (size_t i = 0; i < csrcs.size(); ++i) { 63 ss << csrcs[i]; 64 if (i != csrcs.size() - 1) { 65 ss << ", "; 66 } 67 } 68 ss << ']'; 69 ss << ", extmap-allow-mixed: " << (extmap_allow_mixed ? "true" : "false"); 70 ss << ", extensions: ["; 71 for (size_t i = 0; i < extensions.size(); ++i) { 72 ss << extensions[i].ToString(); 73 if (i != extensions.size() - 1) { 74 ss << ", "; 75 } 76 } 77 ss << ']'; 78 ss << ", c_name: " << c_name; 79 ss << '}'; 80 return ss.str(); 81 } 82 83 AudioSendStream::Config::SendCodecSpec::SendCodecSpec( 84 int payload_type, 85 const SdpAudioFormat& format) 86 : payload_type(payload_type), format(format) {} 87 AudioSendStream::Config::SendCodecSpec::~SendCodecSpec() = default; 88 89 std::string AudioSendStream::Config::SendCodecSpec::ToString() const { 90 char buf[1024]; 91 SimpleStringBuilder ss(buf); 92 ss << "{nack_enabled: " << (nack_enabled ? "true" : "false"); 93 ss << ", enable_non_sender_rtt: " 94 << (enable_non_sender_rtt ? "true" : "false"); 95 ss << ", cng_payload_type: " 96 << (cng_payload_type ? absl::StrCat(*cng_payload_type) : "<unset>"); 97 ss << ", red_payload_type: " 98 << (red_payload_type ? absl::StrCat(*red_payload_type) : "<unset>"); 99 ss << ", payload_type: " << payload_type; 100 ss << ", format: " << absl::StrCat(format); 101 ss << '}'; 102 return ss.str(); 103 } 104 105 bool AudioSendStream::Config::SendCodecSpec::operator==( 106 const AudioSendStream::Config::SendCodecSpec& rhs) const { 107 if (nack_enabled == rhs.nack_enabled && 108 enable_non_sender_rtt == rhs.enable_non_sender_rtt && 109 cng_payload_type == rhs.cng_payload_type && 110 red_payload_type == rhs.red_payload_type && 111 payload_type == rhs.payload_type && format == rhs.format && 112 target_bitrate_bps == rhs.target_bitrate_bps) { 113 return true; 114 } 115 return false; 116 } 117 } // namespace webrtc