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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, |
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308 bool nack_enabled_ = false; | 308 bool nack_enabled_ = false; |
309 bool playout_ = false; | 309 bool playout_ = false; |
310 SendFlags desired_send_ = SEND_NOTHING; | 310 SendFlags desired_send_ = SEND_NOTHING; |
311 SendFlags send_ = SEND_NOTHING; | 311 SendFlags send_ = SEND_NOTHING; |
312 webrtc::Call* const call_ = nullptr; | 312 webrtc::Call* const call_ = nullptr; |
313 | 313 |
314 // SSRC of unsignalled receive stream, or -1 if there isn't one. | 314 // SSRC of unsignalled receive stream, or -1 if there isn't one. |
315 int64_t default_recv_ssrc_ = -1; | 315 int64_t default_recv_ssrc_ = -1; |
316 // Volume for unsignalled stream, which may be set before the stream exists. | 316 // Volume for unsignalled stream, which may be set before the stream exists. |
317 double default_recv_volume_ = 1.0; | 317 double default_recv_volume_ = 1.0; |
318 // SSRC to use for RTCP receiver reports; default to 1 in case of no signaled | 318 // Default SSRC to use for RTCP receiver reports in case of no signaled |
319 // send streams. See: https://code.google.com/p/webrtc/issues/detail?id=4740 | 319 // send streams. See: https://code.google.com/p/webrtc/issues/detail?id=4740 |
320 uint32_t receiver_reports_ssrc_ = 1; | 320 // and https://code.google.com/p/chromium/issues/detail?id=547661 |
321 // This should really be fixed in the RTP/RTCP module. | |
pbos-webrtc
2015/11/12 12:48:12
Why in the RTP/RTCP module?
| |
322 uint32_t receiver_reports_ssrc_ = 0xFA17FA17u; | |
321 | 323 |
322 class WebRtcAudioSendStream; | 324 class WebRtcAudioSendStream; |
323 std::map<uint32_t, WebRtcAudioSendStream*> send_streams_; | 325 std::map<uint32_t, WebRtcAudioSendStream*> send_streams_; |
324 std::vector<RtpHeaderExtension> send_extensions_; | 326 std::vector<RtpHeaderExtension> send_extensions_; |
325 | 327 |
326 class WebRtcAudioReceiveStream; | 328 class WebRtcAudioReceiveStream; |
327 std::map<uint32_t, WebRtcAudioReceiveStream*> receive_channels_; | 329 std::map<uint32_t, WebRtcAudioReceiveStream*> receive_channels_; |
328 std::map<uint32_t, webrtc::AudioReceiveStream*> receive_streams_; | 330 std::map<uint32_t, webrtc::AudioReceiveStream*> receive_streams_; |
329 std::map<uint32_t, StreamParams> receive_stream_params_; | 331 std::map<uint32_t, StreamParams> receive_stream_params_; |
330 // receive_channels_ can be read from WebRtc callback thread. Access from | 332 // receive_channels_ can be read from WebRtc callback thread. Access from |
331 // the WebRtc thread must be synchronized with edits on the worker thread. | 333 // the WebRtc thread must be synchronized with edits on the worker thread. |
332 // Reads on the worker thread are ok. | 334 // Reads on the worker thread are ok. |
333 std::vector<RtpHeaderExtension> receive_extensions_; | 335 std::vector<RtpHeaderExtension> receive_extensions_; |
334 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; | 336 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; |
335 | 337 |
336 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); | 338 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); |
337 }; | 339 }; |
338 } // namespace cricket | 340 } // namespace cricket |
339 | 341 |
340 #endif // TALK_MEDIA_WEBRTCVOICEENGINE_H_ | 342 #endif // TALK_MEDIA_WEBRTCVOICEENGINE_H_ |
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