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1 /* | 1 /* |
2 * Copyright (c) 2004 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2004 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
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233 void SetupRecording(); | 233 void SetupRecording(); |
234 | 234 |
235 rtc::ThreadChecker worker_thread_checker_; | 235 rtc::ThreadChecker worker_thread_checker_; |
236 | 236 |
237 WebRtcVoiceEngine* const engine_ = nullptr; | 237 WebRtcVoiceEngine* const engine_ = nullptr; |
238 std::vector<AudioCodec> send_codecs_; | 238 std::vector<AudioCodec> send_codecs_; |
239 std::vector<AudioCodec> recv_codecs_; | 239 std::vector<AudioCodec> recv_codecs_; |
240 int max_send_bitrate_bps_ = 0; | 240 int max_send_bitrate_bps_ = 0; |
241 AudioOptions options_; | 241 AudioOptions options_; |
242 rtc::Optional<int> dtmf_payload_type_; | 242 rtc::Optional<int> dtmf_payload_type_; |
| 243 int dtmf_payload_freq_ = -1; |
243 bool recv_transport_cc_enabled_ = false; | 244 bool recv_transport_cc_enabled_ = false; |
244 bool recv_nack_enabled_ = false; | 245 bool recv_nack_enabled_ = false; |
245 bool playout_ = false; | 246 bool playout_ = false; |
246 bool send_ = false; | 247 bool send_ = false; |
247 webrtc::Call* const call_ = nullptr; | 248 webrtc::Call* const call_ = nullptr; |
248 | 249 |
249 // SSRC of unsignalled receive stream, or -1 if there isn't one. | 250 // SSRC of unsignalled receive stream, or -1 if there isn't one. |
250 int64_t default_recv_ssrc_ = -1; | 251 int64_t default_recv_ssrc_ = -1; |
251 // Volume for unsignalled stream, which may be set before the stream exists. | 252 // Volume for unsignalled stream, which may be set before the stream exists. |
252 double default_recv_volume_ = 1.0; | 253 double default_recv_volume_ = 1.0; |
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265 std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_; | 266 std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_; |
266 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; | 267 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; |
267 | 268 |
268 webrtc::AudioSendStream::Config::SendCodecSpec send_codec_spec_; | 269 webrtc::AudioSendStream::Config::SendCodecSpec send_codec_spec_; |
269 | 270 |
270 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); | 271 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); |
271 }; | 272 }; |
272 } // namespace cricket | 273 } // namespace cricket |
273 | 274 |
274 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ | 275 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ |
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