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

Issue 2392883002: Multi frequency DTMF support - sender side (Closed)
Patch Set: WVoMC unittests for multi rate DTMF send Created 4 years, 2 months 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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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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