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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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