| OLD | NEW |
| 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 |
| (...skipping 227 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
| 238 void SetupRecording(); | 238 void SetupRecording(); |
| 239 | 239 |
| 240 rtc::ThreadChecker worker_thread_checker_; | 240 rtc::ThreadChecker worker_thread_checker_; |
| 241 | 241 |
| 242 WebRtcVoiceEngine* const engine_ = nullptr; | 242 WebRtcVoiceEngine* const engine_ = nullptr; |
| 243 std::vector<AudioCodec> send_codecs_; | 243 std::vector<AudioCodec> send_codecs_; |
| 244 std::vector<AudioCodec> recv_codecs_; | 244 std::vector<AudioCodec> recv_codecs_; |
| 245 int max_send_bitrate_bps_ = 0; | 245 int max_send_bitrate_bps_ = 0; |
| 246 AudioOptions options_; | 246 AudioOptions options_; |
| 247 rtc::Optional<int> dtmf_payload_type_; | 247 rtc::Optional<int> dtmf_payload_type_; |
| 248 bool desired_playout_ = false; | 248 int dtmf_payload_freq_ = -1; |
| 249 bool recv_transport_cc_enabled_ = false; | 249 bool recv_transport_cc_enabled_ = false; |
| 250 bool recv_nack_enabled_ = false; | 250 bool recv_nack_enabled_ = false; |
| 251 bool desired_playout_ = false; |
| 251 bool playout_ = false; | 252 bool playout_ = false; |
| 252 bool send_ = false; | 253 bool send_ = false; |
| 253 webrtc::Call* const call_ = nullptr; | 254 webrtc::Call* const call_ = nullptr; |
| 254 | 255 |
| 255 // SSRC of unsignalled receive stream, or -1 if there isn't one. | 256 // SSRC of unsignalled receive stream, or -1 if there isn't one. |
| 256 int64_t default_recv_ssrc_ = -1; | 257 int64_t default_recv_ssrc_ = -1; |
| 257 // Volume for unsignalled stream, which may be set before the stream exists. | 258 // Volume for unsignalled stream, which may be set before the stream exists. |
| 258 double default_recv_volume_ = 1.0; | 259 double default_recv_volume_ = 1.0; |
| 259 // Sink for unsignalled stream, which may be set before the stream exists. | 260 // Sink for unsignalled stream, which may be set before the stream exists. |
| 260 std::unique_ptr<webrtc::AudioSinkInterface> default_sink_; | 261 std::unique_ptr<webrtc::AudioSinkInterface> default_sink_; |
| (...skipping 10 matching lines...) Expand all Loading... |
| 271 std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_; | 272 std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_; |
| 272 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; | 273 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; |
| 273 | 274 |
| 274 webrtc::AudioSendStream::Config::SendCodecSpec send_codec_spec_; | 275 webrtc::AudioSendStream::Config::SendCodecSpec send_codec_spec_; |
| 275 | 276 |
| 276 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); | 277 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); |
| 277 }; | 278 }; |
| 278 } // namespace cricket | 279 } // namespace cricket |
| 279 | 280 |
| 280 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ | 281 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ |
| OLD | NEW |