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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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