225 lines
8.9 KiB
C++
225 lines
8.9 KiB
C++
/*
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* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef CALL_RTP_VIDEO_SENDER_H_
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#define CALL_RTP_VIDEO_SENDER_H_
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#include <map>
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#include <memory>
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#include <unordered_set>
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#include <vector>
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#include "absl/types/optional.h"
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#include "api/array_view.h"
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#include "api/call/transport.h"
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#include "api/fec_controller.h"
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#include "api/fec_controller_override.h"
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#include "api/rtc_event_log/rtc_event_log.h"
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#include "api/transport/field_trial_based_config.h"
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#include "api/video_codecs/video_encoder.h"
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#include "call/rtp_config.h"
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#include "call/rtp_payload_params.h"
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#include "call/rtp_transport_controller_send_interface.h"
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#include "call/rtp_video_sender_interface.h"
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#include "modules/rtp_rtcp/include/flexfec_sender.h"
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#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
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#include "modules/rtp_rtcp/source/rtp_sender.h"
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#include "modules/rtp_rtcp/source/rtp_sender_video.h"
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#include "modules/rtp_rtcp/source/rtp_sequence_number_map.h"
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#include "modules/rtp_rtcp/source/rtp_video_header.h"
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#include "modules/utility/include/process_thread.h"
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#include "rtc_base/constructor_magic.h"
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#include "rtc_base/rate_limiter.h"
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#include "rtc_base/synchronization/mutex.h"
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#include "rtc_base/thread_annotations.h"
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#include "rtc_base/thread_checker.h"
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namespace webrtc {
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class FrameEncryptorInterface;
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class RtpTransportControllerSendInterface;
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namespace webrtc_internal_rtp_video_sender {
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// RTP state for a single simulcast stream. Internal to the implementation of
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// RtpVideoSender.
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struct RtpStreamSender {
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RtpStreamSender(std::unique_ptr<ModuleRtpRtcpImpl2> rtp_rtcp,
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std::unique_ptr<RTPSenderVideo> sender_video,
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std::unique_ptr<VideoFecGenerator> fec_generator);
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~RtpStreamSender();
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RtpStreamSender(RtpStreamSender&&) = default;
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RtpStreamSender& operator=(RtpStreamSender&&) = default;
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// Note: Needs pointer stability.
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std::unique_ptr<ModuleRtpRtcpImpl2> rtp_rtcp;
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std::unique_ptr<RTPSenderVideo> sender_video;
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std::unique_ptr<VideoFecGenerator> fec_generator;
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};
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} // namespace webrtc_internal_rtp_video_sender
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// RtpVideoSender routes outgoing data to the correct sending RTP module, based
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// on the simulcast layer in RTPVideoHeader.
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class RtpVideoSender : public RtpVideoSenderInterface,
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public VCMProtectionCallback,
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public StreamFeedbackObserver {
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public:
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// Rtp modules are assumed to be sorted in simulcast index order.
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RtpVideoSender(
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Clock* clock,
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std::map<uint32_t, RtpState> suspended_ssrcs,
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const std::map<uint32_t, RtpPayloadState>& states,
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const RtpConfig& rtp_config,
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int rtcp_report_interval_ms,
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Transport* send_transport,
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const RtpSenderObservers& observers,
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RtpTransportControllerSendInterface* transport,
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RtcEventLog* event_log,
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RateLimiter* retransmission_limiter, // move inside RtpTransport
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std::unique_ptr<FecController> fec_controller,
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FrameEncryptorInterface* frame_encryptor,
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const CryptoOptions& crypto_options, // move inside RtpTransport
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rtc::scoped_refptr<FrameTransformerInterface> frame_transformer);
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~RtpVideoSender() override;
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// RegisterProcessThread register |module_process_thread| with those objects
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// that use it. Registration has to happen on the thread were
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// |module_process_thread| was created (libjingle's worker thread).
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// TODO(perkj): Replace the use of |module_process_thread| with a TaskQueue,
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// maybe |worker_queue|.
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void RegisterProcessThread(ProcessThread* module_process_thread)
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RTC_LOCKS_EXCLUDED(mutex_) override;
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void DeRegisterProcessThread() RTC_LOCKS_EXCLUDED(mutex_) override;
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// RtpVideoSender will only route packets if being active, all packets will be
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// dropped otherwise.
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void SetActive(bool active) RTC_LOCKS_EXCLUDED(mutex_) override;
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// Sets the sending status of the rtp modules and appropriately sets the
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// payload router to active if any rtp modules are active.
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void SetActiveModules(const std::vector<bool> active_modules)
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RTC_LOCKS_EXCLUDED(mutex_) override;
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bool IsActive() RTC_LOCKS_EXCLUDED(mutex_) override;
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void OnNetworkAvailability(bool network_available)
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RTC_LOCKS_EXCLUDED(mutex_) override;
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std::map<uint32_t, RtpState> GetRtpStates() const
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RTC_LOCKS_EXCLUDED(mutex_) override;
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std::map<uint32_t, RtpPayloadState> GetRtpPayloadStates() const
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RTC_LOCKS_EXCLUDED(mutex_) override;
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void DeliverRtcp(const uint8_t* packet, size_t length)
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RTC_LOCKS_EXCLUDED(mutex_) override;
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// Implements webrtc::VCMProtectionCallback.
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int ProtectionRequest(const FecProtectionParams* delta_params,
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const FecProtectionParams* key_params,
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uint32_t* sent_video_rate_bps,
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uint32_t* sent_nack_rate_bps,
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uint32_t* sent_fec_rate_bps)
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RTC_LOCKS_EXCLUDED(mutex_) override;
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// Implements FecControllerOverride.
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void SetFecAllowed(bool fec_allowed) RTC_LOCKS_EXCLUDED(mutex_) override;
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// Implements EncodedImageCallback.
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// Returns 0 if the packet was routed / sent, -1 otherwise.
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EncodedImageCallback::Result OnEncodedImage(
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const EncodedImage& encoded_image,
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const CodecSpecificInfo* codec_specific_info)
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RTC_LOCKS_EXCLUDED(mutex_) override;
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void OnBitrateAllocationUpdated(const VideoBitrateAllocation& bitrate)
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RTC_LOCKS_EXCLUDED(mutex_) override;
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void OnTransportOverheadChanged(size_t transport_overhead_bytes_per_packet)
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RTC_LOCKS_EXCLUDED(mutex_) override;
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void OnBitrateUpdated(BitrateAllocationUpdate update, int framerate)
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RTC_LOCKS_EXCLUDED(mutex_) override;
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uint32_t GetPayloadBitrateBps() const RTC_LOCKS_EXCLUDED(mutex_) override;
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uint32_t GetProtectionBitrateBps() const RTC_LOCKS_EXCLUDED(mutex_) override;
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void SetEncodingData(size_t width, size_t height, size_t num_temporal_layers)
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RTC_LOCKS_EXCLUDED(mutex_) override;
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std::vector<RtpSequenceNumberMap::Info> GetSentRtpPacketInfos(
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uint32_t ssrc,
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rtc::ArrayView<const uint16_t> sequence_numbers) const
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RTC_LOCKS_EXCLUDED(mutex_) override;
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// From StreamFeedbackObserver.
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void OnPacketFeedbackVector(
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std::vector<StreamPacketInfo> packet_feedback_vector)
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RTC_LOCKS_EXCLUDED(mutex_) override;
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private:
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bool IsActiveLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
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void SetActiveModulesLocked(const std::vector<bool> active_modules)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
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void UpdateModuleSendingState() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
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void ConfigureProtection();
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void ConfigureSsrcs();
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void ConfigureRids();
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bool NackEnabled() const;
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uint32_t GetPacketizationOverheadRate() const;
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const FieldTrialBasedConfig field_trials_;
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const bool send_side_bwe_with_overhead_;
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const bool account_for_packetization_overhead_;
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const bool use_early_loss_detection_;
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const bool has_packet_feedback_;
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const bool use_deferred_fec_;
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// TODO(holmer): Remove mutex_ once RtpVideoSender runs on the
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// transport task queue.
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mutable Mutex mutex_;
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bool active_ RTC_GUARDED_BY(mutex_);
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ProcessThread* module_process_thread_;
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rtc::ThreadChecker module_process_thread_checker_;
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std::map<uint32_t, RtpState> suspended_ssrcs_;
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const std::unique_ptr<FecController> fec_controller_;
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bool fec_allowed_ RTC_GUARDED_BY(mutex_);
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// Rtp modules are assumed to be sorted in simulcast index order.
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const std::vector<webrtc_internal_rtp_video_sender::RtpStreamSender>
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rtp_streams_;
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const RtpConfig rtp_config_;
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const absl::optional<VideoCodecType> codec_type_;
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RtpTransportControllerSendInterface* const transport_;
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// When using the generic descriptor we want all simulcast streams to share
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// one frame id space (so that the SFU can switch stream without having to
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// rewrite the frame id), therefore |shared_frame_id| has to live in a place
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// where we are aware of all the different streams.
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int64_t shared_frame_id_ = 0;
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std::vector<RtpPayloadParams> params_ RTC_GUARDED_BY(mutex_);
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size_t transport_overhead_bytes_per_packet_ RTC_GUARDED_BY(mutex_);
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uint32_t protection_bitrate_bps_;
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uint32_t encoder_target_rate_bps_;
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std::vector<bool> loss_mask_vector_ RTC_GUARDED_BY(mutex_);
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std::vector<FrameCounts> frame_counts_ RTC_GUARDED_BY(mutex_);
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FrameCountObserver* const frame_count_observer_;
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// Effectively const map from SSRC to RtpRtcp, for all media SSRCs.
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// This map is set at construction time and never changed, but it's
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// non-trivial to make it properly const.
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std::map<uint32_t, RtpRtcpInterface*> ssrc_to_rtp_module_;
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RTC_DISALLOW_COPY_AND_ASSIGN(RtpVideoSender);
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};
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} // namespace webrtc
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#endif // CALL_RTP_VIDEO_SENDER_H_
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