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1 /* | 1 /* |
2 * Copyright 2004 The WebRTC project authors. All Rights Reserved. | 2 * Copyright 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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457 const DataCodecs& data_codecs() const { return data_codecs_; } | 457 const DataCodecs& data_codecs() const { return data_codecs_; } |
458 void set_data_codecs(const DataCodecs& codecs) { data_codecs_ = codecs; } | 458 void set_data_codecs(const DataCodecs& codecs) { data_codecs_ = codecs; } |
459 SecurePolicy secure() const { return secure_; } | 459 SecurePolicy secure() const { return secure_; } |
460 void set_secure(SecurePolicy s) { secure_ = s; } | 460 void set_secure(SecurePolicy s) { secure_ = s; } |
461 // Decides if a StreamParams shall be added to the audio and video media | 461 // Decides if a StreamParams shall be added to the audio and video media |
462 // content in SessionDescription when CreateOffer and CreateAnswer is called | 462 // content in SessionDescription when CreateOffer and CreateAnswer is called |
463 // even if |options| don't include a Stream. This is needed to support legacy | 463 // even if |options| don't include a Stream. This is needed to support legacy |
464 // applications. |add_legacy_| is true per default. | 464 // applications. |add_legacy_| is true per default. |
465 void set_add_legacy_streams(bool add_legacy) { add_legacy_ = add_legacy; } | 465 void set_add_legacy_streams(bool add_legacy) { add_legacy_ = add_legacy; } |
466 | 466 |
467 void set_enable_rtp_header_encryption(bool enable) { | |
468 enable_rtp_header_encryption_ = enable; | |
469 } | |
470 | |
467 SessionDescription* CreateOffer( | 471 SessionDescription* CreateOffer( |
468 const MediaSessionOptions& options, | 472 const MediaSessionOptions& options, |
469 const SessionDescription* current_description) const; | 473 const SessionDescription* current_description) const; |
470 SessionDescription* CreateAnswer( | 474 SessionDescription* CreateAnswer( |
471 const SessionDescription* offer, | 475 const SessionDescription* offer, |
472 const MediaSessionOptions& options, | 476 const MediaSessionOptions& options, |
473 const SessionDescription* current_description) const; | 477 const SessionDescription* current_description) const; |
474 | 478 |
475 private: | 479 private: |
476 const AudioCodecs& GetAudioCodecsForOffer( | 480 const AudioCodecs& GetAudioCodecsForOffer( |
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556 | 560 |
557 AudioCodecs audio_send_codecs_; | 561 AudioCodecs audio_send_codecs_; |
558 AudioCodecs audio_recv_codecs_; | 562 AudioCodecs audio_recv_codecs_; |
559 AudioCodecs audio_sendrecv_codecs_; | 563 AudioCodecs audio_sendrecv_codecs_; |
560 RtpHeaderExtensions audio_rtp_extensions_; | 564 RtpHeaderExtensions audio_rtp_extensions_; |
561 VideoCodecs video_codecs_; | 565 VideoCodecs video_codecs_; |
562 RtpHeaderExtensions video_rtp_extensions_; | 566 RtpHeaderExtensions video_rtp_extensions_; |
563 DataCodecs data_codecs_; | 567 DataCodecs data_codecs_; |
564 SecurePolicy secure_; | 568 SecurePolicy secure_; |
565 bool add_legacy_; | 569 bool add_legacy_; |
570 bool enable_rtp_header_encryption_ = false; | |
pthatcher1
2017/03/21 07:07:06
"encrypted_rtp_extensions_" to match existing name
joachim
2017/03/23 00:04:33
The "XXX_rtp_extensions_" members are lists of ext
| |
566 std::string lang_; | 571 std::string lang_; |
567 const TransportDescriptionFactory* transport_desc_factory_; | 572 const TransportDescriptionFactory* transport_desc_factory_; |
568 }; | 573 }; |
569 | 574 |
570 // Convenience functions. | 575 // Convenience functions. |
571 bool IsMediaContent(const ContentInfo* content); | 576 bool IsMediaContent(const ContentInfo* content); |
572 bool IsAudioContent(const ContentInfo* content); | 577 bool IsAudioContent(const ContentInfo* content); |
573 bool IsVideoContent(const ContentInfo* content); | 578 bool IsVideoContent(const ContentInfo* content); |
574 bool IsDataContent(const ContentInfo* content); | 579 bool IsDataContent(const ContentInfo* content); |
575 const ContentInfo* GetFirstMediaContent(const ContentInfos& contents, | 580 const ContentInfo* GetFirstMediaContent(const ContentInfos& contents, |
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615 void GetSupportedVideoCryptoSuiteNames(const rtc::CryptoOptions& crypto_options, | 620 void GetSupportedVideoCryptoSuiteNames(const rtc::CryptoOptions& crypto_options, |
616 std::vector<std::string>* crypto_suite_names); | 621 std::vector<std::string>* crypto_suite_names); |
617 void GetSupportedDataCryptoSuiteNames(const rtc::CryptoOptions& crypto_options, | 622 void GetSupportedDataCryptoSuiteNames(const rtc::CryptoOptions& crypto_options, |
618 std::vector<std::string>* crypto_suite_names); | 623 std::vector<std::string>* crypto_suite_names); |
619 void GetDefaultSrtpCryptoSuiteNames(const rtc::CryptoOptions& crypto_options, | 624 void GetDefaultSrtpCryptoSuiteNames(const rtc::CryptoOptions& crypto_options, |
620 std::vector<std::string>* crypto_suite_names); | 625 std::vector<std::string>* crypto_suite_names); |
621 | 626 |
622 } // namespace cricket | 627 } // namespace cricket |
623 | 628 |
624 #endif // WEBRTC_PC_MEDIASESSION_H_ | 629 #endif // WEBRTC_PC_MEDIASESSION_H_ |
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