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Side by Side Diff: webrtc/media/engine/webrtcvoiceengine.h

Issue 2392883002: Multi frequency DTMF support - sender side (Closed)
Patch Set: rebase Created 4 years, 1 month ago
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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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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_;
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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_
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