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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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| 252 int max_send_bitrate_bps_ = 0; | 252 int max_send_bitrate_bps_ = 0; |
| 253 AudioOptions options_; | 253 AudioOptions options_; |
| 254 rtc::Optional<int> dtmf_payload_type_; | 254 rtc::Optional<int> dtmf_payload_type_; |
| 255 int dtmf_payload_freq_ = -1; | 255 int dtmf_payload_freq_ = -1; |
| 256 bool recv_transport_cc_enabled_ = false; | 256 bool recv_transport_cc_enabled_ = false; |
| 257 bool recv_nack_enabled_ = false; | 257 bool recv_nack_enabled_ = false; |
| 258 bool desired_playout_ = false; | 258 bool desired_playout_ = false; |
| 259 bool playout_ = false; | 259 bool playout_ = false; |
| 260 bool send_ = false; | 260 bool send_ = false; |
| 261 webrtc::Call* const call_ = nullptr; | 261 webrtc::Call* const call_ = nullptr; |
| 262 webrtc::Call::Config::BitrateConfig bitrate_config_; |
| 262 | 263 |
| 263 // SSRC of unsignalled receive stream, or -1 if there isn't one. | 264 // SSRC of unsignalled receive stream, or -1 if there isn't one. |
| 264 int64_t default_recv_ssrc_ = -1; | 265 int64_t default_recv_ssrc_ = -1; |
| 265 // Volume for unsignalled stream, which may be set before the stream exists. | 266 // Volume for unsignalled stream, which may be set before the stream exists. |
| 266 double default_recv_volume_ = 1.0; | 267 double default_recv_volume_ = 1.0; |
| 267 // Sink for unsignalled stream, which may be set before the stream exists. | 268 // Sink for unsignalled stream, which may be set before the stream exists. |
| 268 std::unique_ptr<webrtc::AudioSinkInterface> default_sink_; | 269 std::unique_ptr<webrtc::AudioSinkInterface> default_sink_; |
| 269 // Default SSRC to use for RTCP receiver reports in case of no signaled | 270 // Default SSRC to use for RTCP receiver reports in case of no signaled |
| 270 // send streams. See: https://code.google.com/p/webrtc/issues/detail?id=4740 | 271 // send streams. See: https://code.google.com/p/webrtc/issues/detail?id=4740 |
| 271 // and https://code.google.com/p/chromium/issues/detail?id=547661 | 272 // and https://code.google.com/p/chromium/issues/detail?id=547661 |
| 272 uint32_t receiver_reports_ssrc_ = 0xFA17FA17u; | 273 uint32_t receiver_reports_ssrc_ = 0xFA17FA17u; |
| 273 | 274 |
| 274 class WebRtcAudioSendStream; | 275 class WebRtcAudioSendStream; |
| 275 std::map<uint32_t, WebRtcAudioSendStream*> send_streams_; | 276 std::map<uint32_t, WebRtcAudioSendStream*> send_streams_; |
| 276 std::vector<webrtc::RtpExtension> send_rtp_extensions_; | 277 std::vector<webrtc::RtpExtension> send_rtp_extensions_; |
| 277 | 278 |
| 278 class WebRtcAudioReceiveStream; | 279 class WebRtcAudioReceiveStream; |
| 279 std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_; | 280 std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_; |
| 280 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; | 281 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; |
| 281 | 282 |
| 282 webrtc::AudioSendStream::Config::SendCodecSpec send_codec_spec_; | 283 webrtc::AudioSendStream::Config::SendCodecSpec send_codec_spec_; |
| 283 | 284 |
| 284 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); | 285 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); |
| 285 }; | 286 }; |
| 286 } // namespace cricket | 287 } // namespace cricket |
| 287 | 288 |
| 288 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ | 289 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ |
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