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1 /* | 1 /* |
2 * libjingle | 2 * libjingle |
3 * Copyright 2010 Google Inc. | 3 * Copyright 2010 Google Inc. |
4 * | 4 * |
5 * Redistribution and use in source and binary forms, with or without | 5 * Redistribution and use in source and binary forms, with or without |
6 * modification, are permitted provided that the following conditions are met: | 6 * modification, are permitted provided that the following conditions are met: |
7 * | 7 * |
8 * 1. Redistributions of source code must retain the above copyright notice, | 8 * 1. Redistributions of source code must retain the above copyright notice, |
9 * this list of conditions and the following disclaimer. | 9 * this list of conditions and the following disclaimer. |
10 * 2. Redistributions in binary form must reproduce the above copyright notice, | 10 * 2. Redistributions in binary form must reproduce the above copyright notice, |
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250 bool nack; | 250 bool nack; |
251 bool media_processor_registered; | 251 bool media_processor_registered; |
252 bool rx_agc_enabled; | 252 bool rx_agc_enabled; |
253 webrtc::AgcModes rx_agc_mode; | 253 webrtc::AgcModes rx_agc_mode; |
254 webrtc::AgcConfig rx_agc_config; | 254 webrtc::AgcConfig rx_agc_config; |
255 int cn8_type; | 255 int cn8_type; |
256 int cn16_type; | 256 int cn16_type; |
257 int dtmf_type; | 257 int dtmf_type; |
258 int red_type; | 258 int red_type; |
259 int nack_max_packets; | 259 int nack_max_packets; |
260 uint32 send_ssrc; | 260 uint32_t send_ssrc; |
261 int send_audio_level_ext_; | 261 int send_audio_level_ext_; |
262 int receive_audio_level_ext_; | 262 int receive_audio_level_ext_; |
263 int send_absolute_sender_time_ext_; | 263 int send_absolute_sender_time_ext_; |
264 int receive_absolute_sender_time_ext_; | 264 int receive_absolute_sender_time_ext_; |
265 int associate_send_channel; | 265 int associate_send_channel; |
266 DtmfInfo dtmf_info; | 266 DtmfInfo dtmf_info; |
267 std::vector<webrtc::CodecInst> recv_codecs; | 267 std::vector<webrtc::CodecInst> recv_codecs; |
268 webrtc::CodecInst send_codec; | 268 webrtc::CodecInst send_codec; |
269 webrtc::PacketTime last_rtp_packet_time; | 269 webrtc::PacketTime last_rtp_packet_time; |
270 std::list<std::string> packets; | 270 std::list<std::string> packets; |
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309 i != channels_.end(); ++i) { | 309 i != channels_.end(); ++i) { |
310 delete i->second; | 310 delete i->second; |
311 } | 311 } |
312 } | 312 } |
313 | 313 |
314 bool IsExternalMediaProcessorRegistered() const { | 314 bool IsExternalMediaProcessorRegistered() const { |
315 return media_processor_ != NULL; | 315 return media_processor_ != NULL; |
316 } | 316 } |
317 bool IsInited() const { return inited_; } | 317 bool IsInited() const { return inited_; } |
318 int GetLastChannel() const { return last_channel_; } | 318 int GetLastChannel() const { return last_channel_; } |
319 int GetChannelFromLocalSsrc(uint32 local_ssrc) const { | 319 int GetChannelFromLocalSsrc(uint32_t local_ssrc) const { |
320 for (std::map<int, Channel*>::const_iterator iter = channels_.begin(); | 320 for (std::map<int, Channel*>::const_iterator iter = channels_.begin(); |
321 iter != channels_.end(); ++iter) { | 321 iter != channels_.end(); ++iter) { |
322 if (local_ssrc == iter->second->send_ssrc) | 322 if (local_ssrc == iter->second->send_ssrc) |
323 return iter->first; | 323 return iter->first; |
324 } | 324 } |
325 return -1; | 325 return -1; |
326 } | 326 } |
327 int GetNumChannels() const { return static_cast<int>(channels_.size()); } | 327 int GetNumChannels() const { return static_cast<int>(channels_.size()); } |
328 bool GetPlayout(int channel) { | 328 bool GetPlayout(int channel) { |
329 return channels_[channel]->playout; | 329 return channels_[channel]->playout; |
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953 block.extended_highest_sequence_number = kIntStatValue; | 953 block.extended_highest_sequence_number = kIntStatValue; |
954 receive_blocks->push_back(block); | 954 receive_blocks->push_back(block); |
955 } | 955 } |
956 return 0; | 956 return 0; |
957 } | 957 } |
958 WEBRTC_STUB(GetRTPStatistics, (int channel, unsigned int& averageJitterMs, | 958 WEBRTC_STUB(GetRTPStatistics, (int channel, unsigned int& averageJitterMs, |
959 unsigned int& maxJitterMs, | 959 unsigned int& maxJitterMs, |
960 unsigned int& discardedPackets)); | 960 unsigned int& discardedPackets)); |
961 WEBRTC_FUNC(GetRTCPStatistics, (int channel, webrtc::CallStatistics& stats)) { | 961 WEBRTC_FUNC(GetRTCPStatistics, (int channel, webrtc::CallStatistics& stats)) { |
962 WEBRTC_CHECK_CHANNEL(channel); | 962 WEBRTC_CHECK_CHANNEL(channel); |
963 stats.fractionLost = static_cast<int16>(kIntStatValue); | 963 stats.fractionLost = static_cast<int16_t>(kIntStatValue); |
964 stats.cumulativeLost = kIntStatValue; | 964 stats.cumulativeLost = kIntStatValue; |
965 stats.extendedMax = kIntStatValue; | 965 stats.extendedMax = kIntStatValue; |
966 stats.jitterSamples = kIntStatValue; | 966 stats.jitterSamples = kIntStatValue; |
967 stats.rttMs = kIntStatValue; | 967 stats.rttMs = kIntStatValue; |
968 stats.bytesSent = kIntStatValue; | 968 stats.bytesSent = kIntStatValue; |
969 stats.packetsSent = kIntStatValue; | 969 stats.packetsSent = kIntStatValue; |
970 stats.bytesReceived = kIntStatValue; | 970 stats.bytesReceived = kIntStatValue; |
971 stats.packetsReceived = kIntStatValue; | 971 stats.packetsReceived = kIntStatValue; |
972 return 0; | 972 return 0; |
973 } | 973 } |
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1284 DtmfInfo dtmf_info_; | 1284 DtmfInfo dtmf_info_; |
1285 webrtc::VoEMediaProcess* media_processor_; | 1285 webrtc::VoEMediaProcess* media_processor_; |
1286 FakeAudioProcessing audio_processing_; | 1286 FakeAudioProcessing audio_processing_; |
1287 }; | 1287 }; |
1288 | 1288 |
1289 #undef WEBRTC_CHECK_HEADER_EXTENSION_ID | 1289 #undef WEBRTC_CHECK_HEADER_EXTENSION_ID |
1290 | 1290 |
1291 } // namespace cricket | 1291 } // namespace cricket |
1292 | 1292 |
1293 #endif // TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_ | 1293 #endif // TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_ |
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