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

Issue 2534173002: Wire up x-google-{min,start,max}-bitrate to WebRtcVoiceMediaChannel. (Closed)
Patch Set: . Created 4 years 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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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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