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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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237 void SetupRecording(); | 237 void SetupRecording(); |
238 | 238 |
239 rtc::ThreadChecker worker_thread_checker_; | 239 rtc::ThreadChecker worker_thread_checker_; |
240 | 240 |
241 WebRtcVoiceEngine* const engine_ = nullptr; | 241 WebRtcVoiceEngine* const engine_ = nullptr; |
242 std::vector<AudioCodec> send_codecs_; | 242 std::vector<AudioCodec> send_codecs_; |
243 std::vector<AudioCodec> recv_codecs_; | 243 std::vector<AudioCodec> recv_codecs_; |
244 int max_send_bitrate_bps_ = 0; | 244 int max_send_bitrate_bps_ = 0; |
245 AudioOptions options_; | 245 AudioOptions options_; |
246 rtc::Optional<int> dtmf_payload_type_; | 246 rtc::Optional<int> dtmf_payload_type_; |
247 bool desired_playout_ = false; | 247 int dtmf_payload_freq_ = -1; |
248 bool recv_transport_cc_enabled_ = false; | 248 bool recv_transport_cc_enabled_ = false; |
249 bool recv_nack_enabled_ = false; | 249 bool recv_nack_enabled_ = false; |
| 250 bool desired_playout_ = false; |
250 bool playout_ = false; | 251 bool playout_ = false; |
251 bool send_ = false; | 252 bool send_ = false; |
252 webrtc::Call* const call_ = nullptr; | 253 webrtc::Call* const call_ = nullptr; |
253 | 254 |
254 // SSRC of unsignalled receive stream, or -1 if there isn't one. | 255 // SSRC of unsignalled receive stream, or -1 if there isn't one. |
255 int64_t default_recv_ssrc_ = -1; | 256 int64_t default_recv_ssrc_ = -1; |
256 // Volume for unsignalled stream, which may be set before the stream exists. | 257 // Volume for unsignalled stream, which may be set before the stream exists. |
257 double default_recv_volume_ = 1.0; | 258 double default_recv_volume_ = 1.0; |
258 // Sink for unsignalled stream, which may be set before the stream exists. | 259 // Sink for unsignalled stream, which may be set before the stream exists. |
259 std::unique_ptr<webrtc::AudioSinkInterface> default_sink_; | 260 std::unique_ptr<webrtc::AudioSinkInterface> default_sink_; |
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270 std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_; | 271 std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_; |
271 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; | 272 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; |
272 | 273 |
273 webrtc::AudioSendStream::Config::SendCodecSpec send_codec_spec_; | 274 webrtc::AudioSendStream::Config::SendCodecSpec send_codec_spec_; |
274 | 275 |
275 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); | 276 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); |
276 }; | 277 }; |
277 } // namespace cricket | 278 } // namespace cricket |
278 | 279 |
279 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ | 280 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ |
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