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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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262 bool nack_enabled_ = false; | 262 bool nack_enabled_ = false; |
263 bool playout_ = false; | 263 bool playout_ = false; |
264 SendFlags desired_send_ = SEND_NOTHING; | 264 SendFlags desired_send_ = SEND_NOTHING; |
265 SendFlags send_ = SEND_NOTHING; | 265 SendFlags send_ = SEND_NOTHING; |
266 webrtc::Call* const call_ = nullptr; | 266 webrtc::Call* const call_ = nullptr; |
267 | 267 |
268 // SSRC of unsignalled receive stream, or -1 if there isn't one. | 268 // SSRC of unsignalled receive stream, or -1 if there isn't one. |
269 int64_t default_recv_ssrc_ = -1; | 269 int64_t default_recv_ssrc_ = -1; |
270 // Volume for unsignalled stream, which may be set before the stream exists. | 270 // Volume for unsignalled stream, which may be set before the stream exists. |
271 double default_recv_volume_ = 1.0; | 271 double default_recv_volume_ = 1.0; |
| 272 // Sink for unsignalled stream, which may be set before the stream exists. |
| 273 rtc::scoped_ptr<webrtc::AudioSinkInterface> default_sink_; |
272 // Default SSRC to use for RTCP receiver reports in case of no signaled | 274 // Default SSRC to use for RTCP receiver reports in case of no signaled |
273 // send streams. See: https://code.google.com/p/webrtc/issues/detail?id=4740 | 275 // send streams. See: https://code.google.com/p/webrtc/issues/detail?id=4740 |
274 // and https://code.google.com/p/chromium/issues/detail?id=547661 | 276 // and https://code.google.com/p/chromium/issues/detail?id=547661 |
275 uint32_t receiver_reports_ssrc_ = 0xFA17FA17u; | 277 uint32_t receiver_reports_ssrc_ = 0xFA17FA17u; |
276 | 278 |
277 class WebRtcAudioSendStream; | 279 class WebRtcAudioSendStream; |
278 std::map<uint32_t, WebRtcAudioSendStream*> send_streams_; | 280 std::map<uint32_t, WebRtcAudioSendStream*> send_streams_; |
279 std::vector<webrtc::RtpExtension> send_rtp_extensions_; | 281 std::vector<webrtc::RtpExtension> send_rtp_extensions_; |
280 | 282 |
281 class WebRtcAudioReceiveStream; | 283 class WebRtcAudioReceiveStream; |
282 std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_; | 284 std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_; |
283 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; | 285 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; |
284 | 286 |
285 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); | 287 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); |
286 }; | 288 }; |
287 } // namespace cricket | 289 } // namespace cricket |
288 | 290 |
289 #endif // TALK_MEDIA_WEBRTCVOICEENGINE_H_ | 291 #endif // TALK_MEDIA_WEBRTCVOICEENGINE_H_ |
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