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Side by Side Diff: talk/media/webrtc/webrtcvideoengine2.h

Issue 1376673004: Add a PacketOptions struct to webrtc::Transport. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Comment added Created 5 years, 2 months ago
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1 /* 1 /*
2 * libjingle 2 * libjingle
3 * Copyright 2014 Google Inc. 3 * Copyright 2014 Google Inc.
4 * 4 *
5 * Redistribution and use in source and binary forms, with or without 5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met: 6 * modification, are permitted provided that the following conditions are met:
7 * 7 *
8 * 1. Redistributions of source code must retain the above copyright notice, 8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer. 9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice, 10 * 2. Redistributions in binary form must reproduce the above copyright notice,
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457 GUARDED_BY(renderer_lock_); 457 GUARDED_BY(renderer_lock_);
458 int64_t first_frame_timestamp_ GUARDED_BY(renderer_lock_); 458 int64_t first_frame_timestamp_ GUARDED_BY(renderer_lock_);
459 // Start NTP time is estimated as current remote NTP time (estimated from 459 // Start NTP time is estimated as current remote NTP time (estimated from
460 // RTCP) minus the elapsed time, as soon as remote NTP time is available. 460 // RTCP) minus the elapsed time, as soon as remote NTP time is available.
461 int64_t estimated_remote_start_ntp_time_ms_ GUARDED_BY(renderer_lock_); 461 int64_t estimated_remote_start_ntp_time_ms_ GUARDED_BY(renderer_lock_);
462 }; 462 };
463 463
464 void Construct(webrtc::Call* call, WebRtcVideoEngine2* engine); 464 void Construct(webrtc::Call* call, WebRtcVideoEngine2* engine);
465 void SetDefaultOptions(); 465 void SetDefaultOptions();
466 466
467 bool SendRtp(const uint8_t* data, size_t len) override; 467 bool SendRtp(const uint8_t* data,
468 size_t len,
469 const webrtc::PacketOptions& options) override;
468 bool SendRtcp(const uint8_t* data, size_t len) override; 470 bool SendRtcp(const uint8_t* data, size_t len) override;
469 471
470 void StartAllSendStreams(); 472 void StartAllSendStreams();
471 void StopAllSendStreams(); 473 void StopAllSendStreams();
472 474
473 static std::vector<VideoCodecSettings> MapCodecs( 475 static std::vector<VideoCodecSettings> MapCodecs(
474 const std::vector<VideoCodec>& codecs); 476 const std::vector<VideoCodec>& codecs);
475 std::vector<VideoCodecSettings> FilterSupportedCodecs( 477 std::vector<VideoCodecSettings> FilterSupportedCodecs(
476 const std::vector<VideoCodecSettings>& mapped_codecs) const; 478 const std::vector<VideoCodecSettings>& mapped_codecs) const;
477 static bool ReceiveCodecsHaveChanged(std::vector<VideoCodecSettings> before, 479 static bool ReceiveCodecsHaveChanged(std::vector<VideoCodecSettings> before,
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516 WebRtcVideoDecoderFactory* const external_decoder_factory_; 518 WebRtcVideoDecoderFactory* const external_decoder_factory_;
517 std::vector<VideoCodecSettings> recv_codecs_; 519 std::vector<VideoCodecSettings> recv_codecs_;
518 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; 520 std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
519 webrtc::Call::Config::BitrateConfig bitrate_config_; 521 webrtc::Call::Config::BitrateConfig bitrate_config_;
520 VideoOptions options_; 522 VideoOptions options_;
521 }; 523 };
522 524
523 } // namespace cricket 525 } // namespace cricket
524 526
525 #endif // TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_H_ 527 #endif // TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_H_
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