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1 /*
2  *  Copyright (c) 2014 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 #include "webrtc/config.h"
11 
12 #include <sstream>
13 #include <string>
14 
15 namespace webrtc {
ToString() const16 std::string FecConfig::ToString() const {
17   std::stringstream ss;
18   ss << "{ulpfec_payload_type: " << ulpfec_payload_type;
19   ss << ", red_payload_type: " << red_payload_type;
20   ss << '}';
21   return ss.str();
22 }
23 
ToString() const24 std::string RtpExtension::ToString() const {
25   std::stringstream ss;
26   ss << "{name: " << name;
27   ss << ", id: " << id;
28   ss << '}';
29   return ss.str();
30 }
31 
32 const char* RtpExtension::kTOffset = "urn:ietf:params:rtp-hdrext:toffset";
33 const char* RtpExtension::kAbsSendTime =
34     "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time";
35 const char* RtpExtension::kVideoRotation = "urn:3gpp:video-orientation";
36 const char* RtpExtension::kAudioLevel =
37     "urn:ietf:params:rtp-hdrext:ssrc-audio-level";
38 const char* RtpExtension::kTransportSequenceNumber =
39     "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions";
40 
IsSupportedForAudio(const std::string & name)41 bool RtpExtension::IsSupportedForAudio(const std::string& name) {
42   return name == webrtc::RtpExtension::kAbsSendTime ||
43          name == webrtc::RtpExtension::kAudioLevel ||
44          name == webrtc::RtpExtension::kTransportSequenceNumber;
45 }
46 
IsSupportedForVideo(const std::string & name)47 bool RtpExtension::IsSupportedForVideo(const std::string& name) {
48   return name == webrtc::RtpExtension::kTOffset ||
49          name == webrtc::RtpExtension::kAbsSendTime ||
50          name == webrtc::RtpExtension::kVideoRotation ||
51          name == webrtc::RtpExtension::kTransportSequenceNumber;
52 }
53 
VideoStream()54 VideoStream::VideoStream()
55     : width(0),
56       height(0),
57       max_framerate(-1),
58       min_bitrate_bps(-1),
59       target_bitrate_bps(-1),
60       max_bitrate_bps(-1),
61       max_qp(-1) {}
62 
63 VideoStream::~VideoStream() = default;
64 
ToString() const65 std::string VideoStream::ToString() const {
66   std::stringstream ss;
67   ss << "{width: " << width;
68   ss << ", height: " << height;
69   ss << ", max_framerate: " << max_framerate;
70   ss << ", min_bitrate_bps:" << min_bitrate_bps;
71   ss << ", target_bitrate_bps:" << target_bitrate_bps;
72   ss << ", max_bitrate_bps:" << max_bitrate_bps;
73   ss << ", max_qp: " << max_qp;
74 
75   ss << ", temporal_layer_thresholds_bps: [";
76   for (size_t i = 0; i < temporal_layer_thresholds_bps.size(); ++i) {
77     ss << temporal_layer_thresholds_bps[i];
78     if (i != temporal_layer_thresholds_bps.size() - 1)
79       ss << ", ";
80   }
81   ss << ']';
82 
83   ss << '}';
84   return ss.str();
85 }
86 
VideoEncoderConfig()87 VideoEncoderConfig::VideoEncoderConfig()
88     : content_type(ContentType::kRealtimeVideo),
89       encoder_specific_settings(NULL),
90       min_transmit_bitrate_bps(0) {
91 }
92 
93 VideoEncoderConfig::~VideoEncoderConfig() = default;
94 
ToString() const95 std::string VideoEncoderConfig::ToString() const {
96   std::stringstream ss;
97 
98   ss << "{streams: [";
99   for (size_t i = 0; i < streams.size(); ++i) {
100     ss << streams[i].ToString();
101     if (i != streams.size() - 1)
102       ss << ", ";
103   }
104   ss << ']';
105   ss << ", content_type: ";
106   switch (content_type) {
107     case ContentType::kRealtimeVideo:
108       ss << "kRealtimeVideo";
109       break;
110     case ContentType::kScreen:
111       ss << "kScreenshare";
112       break;
113   }
114   ss << ", encoder_specific_settings: ";
115   ss << (encoder_specific_settings != NULL ? "(ptr)" : "NULL");
116 
117   ss << ", min_transmit_bitrate_bps: " << min_transmit_bitrate_bps;
118   ss << '}';
119   return ss.str();
120 }
121 
122 }  // namespace webrtc
123