/* * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. * * Use of this source code is governed by a BSD-style license * that can be found in the LICENSE file in the root of the source * tree. An additional intellectual property rights grant can be found * in the file PATENTS. All contributing project authors may * be found in the AUTHORS file in the root of the source tree. */ #ifndef MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_ #define MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_ #include #include #include "absl/strings/string_view.h" #include "absl/types/optional.h" #include "common_types.h" // NOLINT(build/include) #include "modules/rtp_rtcp/include/flexfec_sender.h" #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "modules/rtp_rtcp/source/playout_delay_oracle.h" #include "modules/rtp_rtcp/source/rtp_rtcp_config.h" #include "modules/rtp_rtcp/source/rtp_sender.h" #include "modules/rtp_rtcp/source/ulpfec_generator.h" #include "rtc_base/critical_section.h" #include "rtc_base/one_time_event.h" #include "rtc_base/rate_statistics.h" #include "rtc_base/sequenced_task_checker.h" #include "rtc_base/thread_annotations.h" namespace webrtc { class FrameEncryptorInterface; class RtpPacketizer; class RtpPacketToSend; // kConditionallyRetransmitHigherLayers allows retransmission of video frames // in higher layers if either the last frame in that layer was too far back in // time, or if we estimate that a new frame will be available in a lower layer // in a shorter time than it would take to request and receive a retransmission. enum RetransmissionMode : uint8_t { kRetransmitOff = 0x0, kRetransmitBaseLayer = 0x2, kRetransmitHigherLayers = 0x4, kConditionallyRetransmitHigherLayers = 0x8, }; class RTPSenderVideo { public: static constexpr int64_t kTLRateWindowSizeMs = 2500; RTPSenderVideo(Clock* clock, RTPSender* rtpSender, FlexfecSender* flexfec_sender, FrameEncryptorInterface* frame_encryptor, bool require_frame_encryption, const WebRtcKeyValueConfig& field_trials); virtual ~RTPSenderVideo(); bool SendVideo(FrameType frame_type, int8_t payload_type, uint32_t capture_timestamp, int64_t capture_time_ms, const uint8_t* payload_data, size_t payload_size, const RTPFragmentationHeader* fragmentation, const RTPVideoHeader* video_header, int64_t expected_retransmission_time_ms); void RegisterPayloadType(int8_t payload_type, absl::string_view payload_name); // ULPFEC. void SetUlpfecConfig(int red_payload_type, int ulpfec_payload_type); // FlexFEC/ULPFEC. void SetFecParameters(const FecProtectionParams& delta_params, const FecProtectionParams& key_params); // FlexFEC. absl::optional FlexfecSsrc() const; uint32_t VideoBitrateSent() const; uint32_t FecOverheadRate() const; uint32_t PacketizationOverheadBps() const; void OnReceivedAck(int64_t extended_highest_sequence_number) { playout_delay_oracle_.OnReceivedAck(extended_highest_sequence_number); } protected: static uint8_t GetTemporalId(const RTPVideoHeader& header); StorageType GetStorageType(uint8_t temporal_id, int32_t retransmission_settings, int64_t expected_retransmission_time_ms); private: struct TemporalLayerStats { TemporalLayerStats() : frame_rate_fp1000s(kTLRateWindowSizeMs, 1000 * 1000), last_frame_time_ms(0) {} // Frame rate, in frames per 1000 seconds. This essentially turns the fps // value into a fixed point value with three decimals. Improves precision at // low frame rates. RateStatistics frame_rate_fp1000s; int64_t last_frame_time_ms; }; size_t CalculateFecPacketOverhead() const RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_); void SendVideoPacket(std::unique_ptr packet, StorageType storage); void SendVideoPacketAsRedMaybeWithUlpfec( std::unique_ptr media_packet, StorageType media_packet_storage, bool protect_media_packet); // TODO(brandtr): Remove the FlexFEC functions when FlexfecSender has been // moved to PacedSender. void SendVideoPacketWithFlexfec(std::unique_ptr media_packet, StorageType media_packet_storage, bool protect_media_packet); bool LogAndSendToNetwork(std::unique_ptr packet, StorageType storage, RtpPacketSender::Priority priority); bool red_enabled() const RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_) { return red_payload_type_ >= 0; } bool ulpfec_enabled() const RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_) { return ulpfec_payload_type_ >= 0; } bool flexfec_enabled() const { return flexfec_sender_ != nullptr; } bool UpdateConditionalRetransmit(uint8_t temporal_id, int64_t expected_retransmission_time_ms) RTC_EXCLUSIVE_LOCKS_REQUIRED(stats_crit_); RTPSender* const rtp_sender_; Clock* const clock_; // Maps payload type to codec type, for packetization. // TODO(nisse): Set on construction, to avoid lock. rtc::CriticalSection payload_type_crit_; std::map payload_type_map_ RTC_GUARDED_BY(payload_type_crit_); // Should never be held when calling out of this class. rtc::CriticalSection crit_; int32_t retransmission_settings_ RTC_GUARDED_BY(crit_); VideoRotation last_rotation_ RTC_GUARDED_BY(crit_); absl::optional last_color_space_ RTC_GUARDED_BY(crit_); bool transmit_color_space_next_frame_ RTC_GUARDED_BY(crit_); // Tracks the current request for playout delay limits from application // and decides whether the current RTP frame should include the playout // delay extension on header. PlayoutDelayOracle playout_delay_oracle_; // RED/ULPFEC. int red_payload_type_ RTC_GUARDED_BY(crit_); int ulpfec_payload_type_ RTC_GUARDED_BY(crit_); UlpfecGenerator ulpfec_generator_ RTC_GUARDED_BY(crit_); // FlexFEC. FlexfecSender* const flexfec_sender_; // FEC parameters, applicable to either ULPFEC or FlexFEC. FecProtectionParams delta_fec_params_ RTC_GUARDED_BY(crit_); FecProtectionParams key_fec_params_ RTC_GUARDED_BY(crit_); rtc::CriticalSection stats_crit_; // Bitrate used for FEC payload, RED headers, RTP headers for FEC packets // and any padding overhead. RateStatistics fec_bitrate_ RTC_GUARDED_BY(stats_crit_); // Bitrate used for video payload and RTP headers. RateStatistics video_bitrate_ RTC_GUARDED_BY(stats_crit_); RateStatistics packetization_overhead_bitrate_ RTC_GUARDED_BY(stats_crit_); std::map frame_stats_by_temporal_layer_ RTC_GUARDED_BY(stats_crit_); OneTimeEvent first_frame_sent_; // E2EE Custom Video Frame Encryptor (optional) FrameEncryptorInterface* const frame_encryptor_ = nullptr; // If set to true will require all outgoing frames to pass through an // initialized frame_encryptor_ before being sent out of the network. // Otherwise these payloads will be dropped. bool require_frame_encryption_; // Set to true if the generic descriptor should be authenticated. const bool generic_descriptor_auth_experiment_; }; } // namespace webrtc #endif // MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_