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| 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 "webrtc/api/call/audio_send_stream.h" |
| 12 |
| 13 #include <string> |
| 14 |
| 15 namespace { |
| 16 |
| 17 std::string ToString(const webrtc::CodecInst& codec_inst) { |
| 18 std::stringstream ss; |
| 19 ss << "{pltype: " << codec_inst.pltype; |
| 20 ss << ", plname: \"" << codec_inst.plname << "\""; |
| 21 ss << ", plfreq: " << codec_inst.plfreq; |
| 22 ss << ", pacsize: " << codec_inst.pacsize; |
| 23 ss << ", channels: " << codec_inst.channels; |
| 24 ss << ", rate: " << codec_inst.rate; |
| 25 ss << '}'; |
| 26 return ss.str(); |
| 27 } |
| 28 } // namespace |
| 29 |
| 30 namespace webrtc { |
| 31 |
| 32 AudioSendStream::Stats::Stats() = default; |
| 33 |
| 34 AudioSendStream::Config::Config(Transport* send_transport) |
| 35 : send_transport(send_transport) {} |
| 36 |
| 37 std::string AudioSendStream::Config::ToString() const { |
| 38 std::stringstream ss; |
| 39 ss << "{rtp: " << rtp.ToString(); |
| 40 ss << ", send_transport: " << (send_transport ? "(Transport)" : "nullptr"); |
| 41 ss << ", voe_channel_id: " << voe_channel_id; |
| 42 ss << ", min_bitrate_kbps: " << min_bitrate_kbps; |
| 43 ss << ", max_bitrate_kbps: " << max_bitrate_kbps; |
| 44 ss << ", send_codec_spec: " << send_codec_spec.ToString(); |
| 45 ss << '}'; |
| 46 return ss.str(); |
| 47 } |
| 48 |
| 49 AudioSendStream::Config::Rtp::Rtp() = default; |
| 50 |
| 51 AudioSendStream::Config::Rtp::~Rtp() = default; |
| 52 |
| 53 std::string AudioSendStream::Config::Rtp::ToString() const { |
| 54 std::stringstream ss; |
| 55 ss << "{ssrc: " << ssrc; |
| 56 ss << ", extensions: ["; |
| 57 for (size_t i = 0; i < extensions.size(); ++i) { |
| 58 ss << extensions[i].ToString(); |
| 59 if (i != extensions.size() - 1) { |
| 60 ss << ", "; |
| 61 } |
| 62 } |
| 63 ss << ']'; |
| 64 ss << ", nack: " << nack.ToString(); |
| 65 ss << ", c_name: " << c_name; |
| 66 ss << '}'; |
| 67 return ss.str(); |
| 68 } |
| 69 |
| 70 AudioSendStream::Config::SendCodecSpec::SendCodecSpec() { |
| 71 webrtc::CodecInst empty_inst = {0}; |
| 72 codec_inst = empty_inst; |
| 73 codec_inst.pltype = -1; |
| 74 } |
| 75 |
| 76 std::string AudioSendStream::Config::SendCodecSpec::ToString() const { |
| 77 std::stringstream ss; |
| 78 ss << "{nack_enabled: " << (nack_enabled ? "true" : "false"); |
| 79 ss << ", transport_cc_enabled: " << (transport_cc_enabled ? "true" : "false"); |
| 80 ss << ", enable_codec_fec: " << (enable_codec_fec ? "true" : "false"); |
| 81 ss << ", enable_opus_dtx: " << (enable_opus_dtx ? "true" : "false"); |
| 82 ss << ", opus_max_playback_rate: " << opus_max_playback_rate; |
| 83 ss << ", cng_payload_type: " << cng_payload_type; |
| 84 ss << ", cng_plfreq: " << cng_plfreq; |
| 85 ss << ", codec_inst: " << ::ToString(codec_inst); |
| 86 ss << '}'; |
| 87 return ss.str(); |
| 88 } |
| 89 |
| 90 bool AudioSendStream::Config::SendCodecSpec::operator==( |
| 91 const AudioSendStream::Config::SendCodecSpec& rhs) const { |
| 92 if (nack_enabled != rhs.nack_enabled) { |
| 93 return false; |
| 94 } |
| 95 if (transport_cc_enabled != rhs.transport_cc_enabled) { |
| 96 return false; |
| 97 } |
| 98 if (enable_codec_fec != rhs.enable_codec_fec) { |
| 99 return false; |
| 100 } |
| 101 if (enable_opus_dtx != rhs.enable_opus_dtx) { |
| 102 return false; |
| 103 } |
| 104 if (opus_max_playback_rate != rhs.opus_max_playback_rate) { |
| 105 return false; |
| 106 } |
| 107 if (cng_payload_type != rhs.cng_payload_type) { |
| 108 return false; |
| 109 } |
| 110 if (cng_plfreq != rhs.cng_plfreq) { |
| 111 return false; |
| 112 } |
| 113 if (codec_inst != rhs.codec_inst) { |
| 114 return false; |
| 115 } |
| 116 return true; |
| 117 } |
| 118 } // namespace webrtc |
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