OLD | NEW |
1 /* | 1 /* |
2 * libjingle | 2 * libjingle |
3 * Copyright 2004 Google Inc. | 3 * Copyright 2004 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, |
(...skipping 174 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
185 const AudioOptions& options() const { return options_; } | 185 const AudioOptions& options() const { return options_; } |
186 | 186 |
187 bool SetSendParameters(const AudioSendParameters& params) override; | 187 bool SetSendParameters(const AudioSendParameters& params) override; |
188 bool SetRecvParameters(const AudioRecvParameters& params) override; | 188 bool SetRecvParameters(const AudioRecvParameters& params) override; |
189 bool SetPlayout(bool playout) override; | 189 bool SetPlayout(bool playout) override; |
190 bool PausePlayout(); | 190 bool PausePlayout(); |
191 bool ResumePlayout(); | 191 bool ResumePlayout(); |
192 bool SetSend(SendFlags send) override; | 192 bool SetSend(SendFlags send) override; |
193 bool PauseSend(); | 193 bool PauseSend(); |
194 bool ResumeSend(); | 194 bool ResumeSend(); |
195 bool SetAudioSend(uint32 ssrc, bool enable, const AudioOptions* options, | 195 bool SetAudioSend(uint32_t ssrc, |
| 196 bool enable, |
| 197 const AudioOptions* options, |
196 AudioRenderer* renderer) override; | 198 AudioRenderer* renderer) override; |
197 bool AddSendStream(const StreamParams& sp) override; | 199 bool AddSendStream(const StreamParams& sp) override; |
198 bool RemoveSendStream(uint32 ssrc) override; | 200 bool RemoveSendStream(uint32_t ssrc) override; |
199 bool AddRecvStream(const StreamParams& sp) override; | 201 bool AddRecvStream(const StreamParams& sp) override; |
200 bool RemoveRecvStream(uint32 ssrc) override; | 202 bool RemoveRecvStream(uint32_t ssrc) override; |
201 bool SetRemoteRenderer(uint32 ssrc, AudioRenderer* renderer) override; | 203 bool SetRemoteRenderer(uint32_t ssrc, AudioRenderer* renderer) override; |
202 bool GetActiveStreams(AudioInfo::StreamList* actives) override; | 204 bool GetActiveStreams(AudioInfo::StreamList* actives) override; |
203 int GetOutputLevel() override; | 205 int GetOutputLevel() override; |
204 int GetTimeSinceLastTyping() override; | 206 int GetTimeSinceLastTyping() override; |
205 void SetTypingDetectionParameters(int time_window, | 207 void SetTypingDetectionParameters(int time_window, |
206 int cost_per_typing, | 208 int cost_per_typing, |
207 int reporting_threshold, | 209 int reporting_threshold, |
208 int penalty_decay, | 210 int penalty_decay, |
209 int type_event_delay) override; | 211 int type_event_delay) override; |
210 bool SetOutputScaling(uint32 ssrc, double left, double right) override; | 212 bool SetOutputScaling(uint32_t ssrc, double left, double right) override; |
211 | 213 |
212 bool CanInsertDtmf() override; | 214 bool CanInsertDtmf() override; |
213 bool InsertDtmf(uint32 ssrc, int event, int duration, int flags) override; | 215 bool InsertDtmf(uint32_t ssrc, int event, int duration, int flags) override; |
214 | 216 |
215 void OnPacketReceived(rtc::Buffer* packet, | 217 void OnPacketReceived(rtc::Buffer* packet, |
216 const rtc::PacketTime& packet_time) override; | 218 const rtc::PacketTime& packet_time) override; |
217 void OnRtcpReceived(rtc::Buffer* packet, | 219 void OnRtcpReceived(rtc::Buffer* packet, |
218 const rtc::PacketTime& packet_time) override; | 220 const rtc::PacketTime& packet_time) override; |
219 void OnReadyToSend(bool ready) override {} | 221 void OnReadyToSend(bool ready) override {} |
220 bool GetStats(VoiceMediaInfo* info) override; | 222 bool GetStats(VoiceMediaInfo* info) override; |
221 | 223 |
222 // implements Transport interface | 224 // implements Transport interface |
223 bool SendRtp(const uint8_t* data, | 225 bool SendRtp(const uint8_t* data, |
224 size_t len, | 226 size_t len, |
225 const webrtc::PacketOptions& options) override { | 227 const webrtc::PacketOptions& options) override { |
226 rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len, | 228 rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len, |
227 kMaxRtpPacketLen); | 229 kMaxRtpPacketLen); |
228 return VoiceMediaChannel::SendPacket(&packet); | 230 return VoiceMediaChannel::SendPacket(&packet); |
229 } | 231 } |
230 | 232 |
231 bool SendRtcp(const uint8_t* data, size_t len) override { | 233 bool SendRtcp(const uint8_t* data, size_t len) override { |
232 rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len, | 234 rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len, |
233 kMaxRtpPacketLen); | 235 kMaxRtpPacketLen); |
234 return VoiceMediaChannel::SendRtcp(&packet); | 236 return VoiceMediaChannel::SendRtcp(&packet); |
235 } | 237 } |
236 | 238 |
237 void OnError(int error); | 239 void OnError(int error); |
238 | 240 |
239 int GetReceiveChannelId(uint32 ssrc) const; | 241 int GetReceiveChannelId(uint32_t ssrc) const; |
240 int GetSendChannelId(uint32 ssrc) const; | 242 int GetSendChannelId(uint32_t ssrc) const; |
241 | 243 |
242 private: | 244 private: |
243 bool SetSendCodecs(const std::vector<AudioCodec>& codecs); | 245 bool SetSendCodecs(const std::vector<AudioCodec>& codecs); |
244 bool SetSendRtpHeaderExtensions( | 246 bool SetSendRtpHeaderExtensions( |
245 const std::vector<RtpHeaderExtension>& extensions); | 247 const std::vector<RtpHeaderExtension>& extensions); |
246 bool SetOptions(const AudioOptions& options); | 248 bool SetOptions(const AudioOptions& options); |
247 bool SetMaxSendBandwidth(int bps); | 249 bool SetMaxSendBandwidth(int bps); |
248 bool SetRecvCodecs(const std::vector<AudioCodec>& codecs); | 250 bool SetRecvCodecs(const std::vector<AudioCodec>& codecs); |
249 bool SetRecvRtpHeaderExtensions( | 251 bool SetRecvRtpHeaderExtensions( |
250 const std::vector<RtpHeaderExtension>& extensions); | 252 const std::vector<RtpHeaderExtension>& extensions); |
251 bool SetLocalRenderer(uint32 ssrc, AudioRenderer* renderer); | 253 bool SetLocalRenderer(uint32_t ssrc, AudioRenderer* renderer); |
252 bool MuteStream(uint32 ssrc, bool mute); | 254 bool MuteStream(uint32_t ssrc, bool mute); |
253 | 255 |
254 WebRtcVoiceEngine* engine() { return engine_; } | 256 WebRtcVoiceEngine* engine() { return engine_; } |
255 int GetLastEngineError() { return engine()->GetLastEngineError(); } | 257 int GetLastEngineError() { return engine()->GetLastEngineError(); } |
256 int GetOutputLevel(int channel); | 258 int GetOutputLevel(int channel); |
257 bool GetRedSendCodec(const AudioCodec& red_codec, | 259 bool GetRedSendCodec(const AudioCodec& red_codec, |
258 const std::vector<AudioCodec>& all_codecs, | 260 const std::vector<AudioCodec>& all_codecs, |
259 webrtc::CodecInst* send_codec); | 261 webrtc::CodecInst* send_codec); |
260 bool EnableRtcp(int channel); | 262 bool EnableRtcp(int channel); |
261 bool ResetRecvCodecs(int channel); | 263 bool ResetRecvCodecs(int channel); |
262 bool SetPlayout(int channel, bool playout); | 264 bool SetPlayout(int channel, bool playout); |
263 static uint32 ParseSsrc(const void* data, size_t len, bool rtcp); | 265 static uint32_t ParseSsrc(const void* data, size_t len, bool rtcp); |
264 static Error WebRtcErrorToChannelError(int err_code); | 266 static Error WebRtcErrorToChannelError(int err_code); |
265 | 267 |
266 class WebRtcVoiceChannelRenderer; | 268 class WebRtcVoiceChannelRenderer; |
267 // Map of ssrc to WebRtcVoiceChannelRenderer object. A new object of | 269 // Map of ssrc to WebRtcVoiceChannelRenderer object. A new object of |
268 // WebRtcVoiceChannelRenderer will be created for every new stream and | 270 // WebRtcVoiceChannelRenderer will be created for every new stream and |
269 // will be destroyed when the stream goes away. | 271 // will be destroyed when the stream goes away. |
270 typedef std::map<uint32, WebRtcVoiceChannelRenderer*> ChannelMap; | 272 typedef std::map<uint32_t, WebRtcVoiceChannelRenderer*> ChannelMap; |
271 typedef int (webrtc::VoERTP_RTCP::* ExtensionSetterFunction)(int, bool, | 273 typedef int (webrtc::VoERTP_RTCP::* ExtensionSetterFunction)(int, bool, |
272 unsigned char); | 274 unsigned char); |
273 | 275 |
274 void SetNack(int channel, bool nack_enabled); | 276 void SetNack(int channel, bool nack_enabled); |
275 void SetNack(const ChannelMap& channels, bool nack_enabled); | 277 void SetNack(const ChannelMap& channels, bool nack_enabled); |
276 bool SetSendCodec(const webrtc::CodecInst& send_codec); | 278 bool SetSendCodec(const webrtc::CodecInst& send_codec); |
277 bool SetSendCodec(int channel, const webrtc::CodecInst& send_codec); | 279 bool SetSendCodec(int channel, const webrtc::CodecInst& send_codec); |
278 bool ChangePlayout(bool playout); | 280 bool ChangePlayout(bool playout); |
279 bool ChangeSend(SendFlags send); | 281 bool ChangeSend(SendFlags send); |
280 bool ChangeSend(int channel, SendFlags send); | 282 bool ChangeSend(int channel, SendFlags send); |
281 void ConfigureSendChannel(int channel); | 283 void ConfigureSendChannel(int channel); |
282 bool ConfigureRecvChannel(int channel); | 284 bool ConfigureRecvChannel(int channel); |
283 bool DeleteChannel(int channel); | 285 bool DeleteChannel(int channel); |
284 bool InConferenceMode() const { | 286 bool InConferenceMode() const { |
285 return options_.conference_mode.GetWithDefaultIfUnset(false); | 287 return options_.conference_mode.GetWithDefaultIfUnset(false); |
286 } | 288 } |
287 bool IsDefaultChannel(int channel_id) const { | 289 bool IsDefaultChannel(int channel_id) const { |
288 return channel_id == voe_channel(); | 290 return channel_id == voe_channel(); |
289 } | 291 } |
290 bool SetSendCodecs(int channel, const std::vector<AudioCodec>& codecs); | 292 bool SetSendCodecs(int channel, const std::vector<AudioCodec>& codecs); |
291 bool SetSendBitrateInternal(int bps); | 293 bool SetSendBitrateInternal(int bps); |
292 | 294 |
293 bool SetHeaderExtension(ExtensionSetterFunction setter, int channel_id, | 295 bool SetHeaderExtension(ExtensionSetterFunction setter, int channel_id, |
294 const RtpHeaderExtension* extension); | 296 const RtpHeaderExtension* extension); |
295 void RecreateAudioReceiveStreams(); | 297 void RecreateAudioReceiveStreams(); |
296 void AddAudioReceiveStream(uint32 ssrc); | 298 void AddAudioReceiveStream(uint32_t ssrc); |
297 void RemoveAudioReceiveStream(uint32 ssrc); | 299 void RemoveAudioReceiveStream(uint32_t ssrc); |
298 bool SetRecvCodecsInternal(const std::vector<AudioCodec>& new_codecs); | 300 bool SetRecvCodecsInternal(const std::vector<AudioCodec>& new_codecs); |
299 | 301 |
300 bool SetChannelRecvRtpHeaderExtensions( | 302 bool SetChannelRecvRtpHeaderExtensions( |
301 int channel_id, | 303 int channel_id, |
302 const std::vector<RtpHeaderExtension>& extensions); | 304 const std::vector<RtpHeaderExtension>& extensions); |
303 bool SetChannelSendRtpHeaderExtensions( | 305 bool SetChannelSendRtpHeaderExtensions( |
304 int channel_id, | 306 int channel_id, |
305 const std::vector<RtpHeaderExtension>& extensions); | 307 const std::vector<RtpHeaderExtension>& extensions); |
306 | 308 |
307 rtc::ThreadChecker thread_checker_; | 309 rtc::ThreadChecker thread_checker_; |
(...skipping 13 matching lines...) Expand all Loading... |
321 bool typing_noise_detected_; | 323 bool typing_noise_detected_; |
322 SendFlags desired_send_; | 324 SendFlags desired_send_; |
323 SendFlags send_; | 325 SendFlags send_; |
324 webrtc::Call* const call_; | 326 webrtc::Call* const call_; |
325 | 327 |
326 // send_channels_ contains the channels which are being used for sending. | 328 // send_channels_ contains the channels which are being used for sending. |
327 // When the default channel (voe_channel) is used for sending, it is | 329 // When the default channel (voe_channel) is used for sending, it is |
328 // contained in send_channels_, otherwise not. | 330 // contained in send_channels_, otherwise not. |
329 ChannelMap send_channels_; | 331 ChannelMap send_channels_; |
330 std::vector<RtpHeaderExtension> send_extensions_; | 332 std::vector<RtpHeaderExtension> send_extensions_; |
331 uint32 default_receive_ssrc_; | 333 uint32_t default_receive_ssrc_; |
332 // Note the default channel (voe_channel()) can reside in both | 334 // Note the default channel (voe_channel()) can reside in both |
333 // receive_channels_ and send_channels_ in non-conference mode and in that | 335 // receive_channels_ and send_channels_ in non-conference mode and in that |
334 // case it will only be there if a non-zero default_receive_ssrc_ is set. | 336 // case it will only be there if a non-zero default_receive_ssrc_ is set. |
335 ChannelMap receive_channels_; // for multiple sources | 337 ChannelMap receive_channels_; // for multiple sources |
336 std::map<uint32, webrtc::AudioReceiveStream*> receive_streams_; | 338 std::map<uint32_t, webrtc::AudioReceiveStream*> receive_streams_; |
337 std::map<uint32, StreamParams> receive_stream_params_; | 339 std::map<uint32_t, StreamParams> receive_stream_params_; |
338 // receive_channels_ can be read from WebRtc callback thread. Access from | 340 // receive_channels_ can be read from WebRtc callback thread. Access from |
339 // the WebRtc thread must be synchronized with edits on the worker thread. | 341 // the WebRtc thread must be synchronized with edits on the worker thread. |
340 // Reads on the worker thread are ok. | 342 // Reads on the worker thread are ok. |
341 std::vector<RtpHeaderExtension> receive_extensions_; | 343 std::vector<RtpHeaderExtension> receive_extensions_; |
342 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; | 344 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; |
343 | 345 |
344 // Do not lock this on the VoE media processor thread; potential for deadlock | 346 // Do not lock this on the VoE media processor thread; potential for deadlock |
345 // exists. | 347 // exists. |
346 mutable rtc::CriticalSection receive_channels_cs_; | 348 mutable rtc::CriticalSection receive_channels_cs_; |
347 }; | 349 }; |
348 | 350 |
349 } // namespace cricket | 351 } // namespace cricket |
350 | 352 |
351 #endif // TALK_MEDIA_WEBRTCVOICEENGINE_H_ | 353 #endif // TALK_MEDIA_WEBRTCVOICEENGINE_H_ |
OLD | NEW |