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1 /* | 1 /* |
2 * Copyright (c) 2004 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2004 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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92 bool CheckNoRtp() { return rtp_packets_.empty(); } | 92 bool CheckNoRtp() { return rtp_packets_.empty(); } |
93 bool CheckNoRtcp() { return rtcp_packets_.empty(); } | 93 bool CheckNoRtcp() { return rtcp_packets_.empty(); } |
94 void set_fail_set_send_codecs(bool fail) { fail_set_send_codecs_ = fail; } | 94 void set_fail_set_send_codecs(bool fail) { fail_set_send_codecs_ = fail; } |
95 void set_fail_set_recv_codecs(bool fail) { fail_set_recv_codecs_ = fail; } | 95 void set_fail_set_recv_codecs(bool fail) { fail_set_recv_codecs_ = fail; } |
96 virtual bool AddSendStream(const StreamParams& sp) { | 96 virtual bool AddSendStream(const StreamParams& sp) { |
97 if (std::find(send_streams_.begin(), send_streams_.end(), sp) != | 97 if (std::find(send_streams_.begin(), send_streams_.end(), sp) != |
98 send_streams_.end()) { | 98 send_streams_.end()) { |
99 return false; | 99 return false; |
100 } | 100 } |
101 send_streams_.push_back(sp); | 101 send_streams_.push_back(sp); |
102 rtp_parameters_[sp.first_ssrc()] = CreateRtpParametersWithOneEncoding(); | 102 rtp_send_parameters_[sp.first_ssrc()] = |
| 103 CreateRtpParametersWithOneEncoding(); |
103 return true; | 104 return true; |
104 } | 105 } |
105 virtual bool RemoveSendStream(uint32_t ssrc) { | 106 virtual bool RemoveSendStream(uint32_t ssrc) { |
106 auto parameters_iterator = rtp_parameters_.find(ssrc); | 107 auto parameters_iterator = rtp_send_parameters_.find(ssrc); |
107 if (parameters_iterator != rtp_parameters_.end()) { | 108 if (parameters_iterator != rtp_send_parameters_.end()) { |
108 rtp_parameters_.erase(parameters_iterator); | 109 rtp_send_parameters_.erase(parameters_iterator); |
109 } | 110 } |
110 return RemoveStreamBySsrc(&send_streams_, ssrc); | 111 return RemoveStreamBySsrc(&send_streams_, ssrc); |
111 } | 112 } |
112 virtual bool AddRecvStream(const StreamParams& sp) { | 113 virtual bool AddRecvStream(const StreamParams& sp) { |
113 if (std::find(receive_streams_.begin(), receive_streams_.end(), sp) != | 114 if (std::find(receive_streams_.begin(), receive_streams_.end(), sp) != |
114 receive_streams_.end()) { | 115 receive_streams_.end()) { |
115 return false; | 116 return false; |
116 } | 117 } |
117 receive_streams_.push_back(sp); | 118 receive_streams_.push_back(sp); |
| 119 rtp_receive_parameters_[sp.first_ssrc()] = |
| 120 CreateRtpParametersWithOneEncoding(); |
118 return true; | 121 return true; |
119 } | 122 } |
120 virtual bool RemoveRecvStream(uint32_t ssrc) { | 123 virtual bool RemoveRecvStream(uint32_t ssrc) { |
| 124 auto parameters_iterator = rtp_receive_parameters_.find(ssrc); |
| 125 if (parameters_iterator != rtp_receive_parameters_.end()) { |
| 126 rtp_receive_parameters_.erase(parameters_iterator); |
| 127 } |
121 return RemoveStreamBySsrc(&receive_streams_, ssrc); | 128 return RemoveStreamBySsrc(&receive_streams_, ssrc); |
122 } | 129 } |
123 | 130 |
124 virtual webrtc::RtpParameters GetRtpParameters(uint32_t ssrc) const { | 131 virtual webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const { |
125 auto parameters_iterator = rtp_parameters_.find(ssrc); | 132 auto parameters_iterator = rtp_send_parameters_.find(ssrc); |
126 if (parameters_iterator != rtp_parameters_.end()) { | 133 if (parameters_iterator != rtp_send_parameters_.end()) { |
127 return parameters_iterator->second; | 134 return parameters_iterator->second; |
128 } | 135 } |
129 return webrtc::RtpParameters(); | 136 return webrtc::RtpParameters(); |
130 } | 137 } |
131 virtual bool SetRtpParameters(uint32_t ssrc, | 138 virtual bool SetRtpSendParameters(uint32_t ssrc, |
132 const webrtc::RtpParameters& parameters) { | 139 const webrtc::RtpParameters& parameters) { |
133 auto parameters_iterator = rtp_parameters_.find(ssrc); | 140 auto parameters_iterator = rtp_send_parameters_.find(ssrc); |
134 if (parameters_iterator != rtp_parameters_.end()) { | 141 if (parameters_iterator != rtp_send_parameters_.end()) { |
135 parameters_iterator->second = parameters; | 142 parameters_iterator->second = parameters; |
136 return true; | 143 return true; |
137 } | 144 } |
| 145 // Replicate the behavior of the real media channel: return false |
| 146 // when setting parameters for unknown SSRCs. |
| 147 return false; |
| 148 } |
| 149 |
| 150 virtual webrtc::RtpParameters GetRtpReceiveParameters(uint32_t ssrc) const { |
| 151 auto parameters_iterator = rtp_receive_parameters_.find(ssrc); |
| 152 if (parameters_iterator != rtp_receive_parameters_.end()) { |
| 153 return parameters_iterator->second; |
| 154 } |
| 155 return webrtc::RtpParameters(); |
| 156 } |
| 157 virtual bool SetRtpReceiveParameters( |
| 158 uint32_t ssrc, |
| 159 const webrtc::RtpParameters& parameters) { |
| 160 auto parameters_iterator = rtp_receive_parameters_.find(ssrc); |
| 161 if (parameters_iterator != rtp_receive_parameters_.end()) { |
| 162 parameters_iterator->second = parameters; |
| 163 return true; |
| 164 } |
138 // Replicate the behavior of the real media channel: return false | 165 // Replicate the behavior of the real media channel: return false |
139 // when setting parameters for unknown SSRCs. | 166 // when setting parameters for unknown SSRCs. |
140 return false; | 167 return false; |
141 } | 168 } |
142 | 169 |
143 bool IsStreamMuted(uint32_t ssrc) const { | 170 bool IsStreamMuted(uint32_t ssrc) const { |
144 bool ret = muted_streams_.find(ssrc) != muted_streams_.end(); | 171 bool ret = muted_streams_.find(ssrc) != muted_streams_.end(); |
145 // If |ssrc = 0| check if the first send stream is muted. | 172 // If |ssrc = 0| check if the first send stream is muted. |
146 if (!ret && ssrc == 0 && !send_streams_.empty()) { | 173 if (!ret && ssrc == 0 && !send_streams_.empty()) { |
147 return muted_streams_.find(send_streams_[0].first_ssrc()) != | 174 return muted_streams_.find(send_streams_[0].first_ssrc()) != |
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236 private: | 263 private: |
237 bool sending_; | 264 bool sending_; |
238 bool playout_; | 265 bool playout_; |
239 std::vector<RtpHeaderExtension> recv_extensions_; | 266 std::vector<RtpHeaderExtension> recv_extensions_; |
240 std::vector<RtpHeaderExtension> send_extensions_; | 267 std::vector<RtpHeaderExtension> send_extensions_; |
241 std::list<std::string> rtp_packets_; | 268 std::list<std::string> rtp_packets_; |
242 std::list<std::string> rtcp_packets_; | 269 std::list<std::string> rtcp_packets_; |
243 std::vector<StreamParams> send_streams_; | 270 std::vector<StreamParams> send_streams_; |
244 std::vector<StreamParams> receive_streams_; | 271 std::vector<StreamParams> receive_streams_; |
245 std::set<uint32_t> muted_streams_; | 272 std::set<uint32_t> muted_streams_; |
246 std::map<uint32_t, webrtc::RtpParameters> rtp_parameters_; | 273 std::map<uint32_t, webrtc::RtpParameters> rtp_send_parameters_; |
| 274 std::map<uint32_t, webrtc::RtpParameters> rtp_receive_parameters_; |
247 bool fail_set_send_codecs_; | 275 bool fail_set_send_codecs_; |
248 bool fail_set_recv_codecs_; | 276 bool fail_set_recv_codecs_; |
249 uint32_t send_ssrc_; | 277 uint32_t send_ssrc_; |
250 std::string rtcp_cname_; | 278 std::string rtcp_cname_; |
251 bool ready_to_send_; | 279 bool ready_to_send_; |
252 rtc::NetworkRoute last_network_route_; | 280 rtc::NetworkRoute last_network_route_; |
253 int num_network_route_changes_ = 0; | 281 int num_network_route_changes_ = 0; |
254 }; | 282 }; |
255 | 283 |
256 class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> { | 284 class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> { |
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911 | 939 |
912 private: | 940 private: |
913 std::vector<FakeDataMediaChannel*> channels_; | 941 std::vector<FakeDataMediaChannel*> channels_; |
914 std::vector<DataCodec> data_codecs_; | 942 std::vector<DataCodec> data_codecs_; |
915 DataChannelType last_channel_type_; | 943 DataChannelType last_channel_type_; |
916 }; | 944 }; |
917 | 945 |
918 } // namespace cricket | 946 } // namespace cricket |
919 | 947 |
920 #endif // WEBRTC_MEDIA_BASE_FAKEMEDIAENGINE_H_ | 948 #endif // WEBRTC_MEDIA_BASE_FAKEMEDIAENGINE_H_ |
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