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Side by Side Diff: talk/media/webrtc/fakewebrtcvoiceengine.h

Issue 1362503003: Use suffixed {uint,int}{8,16,32,64}_t types. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: google::int32 Created 5 years, 3 months ago
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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;
(...skipping 38 matching lines...) Expand 10 before | Expand all | Expand 10 after
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;
(...skipping 623 matching lines...) Expand 10 before | Expand all | Expand 10 after
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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