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/*
* Copyright (c) 2015 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 CALL_RTP_VIDEO_SENDER_H_
#define CALL_RTP_VIDEO_SENDER_H_
#include <map>
#include <memory>
#include <unordered_set>
#include <vector>
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/call/transport.h"
#include "api/fec_controller.h"
#include "api/fec_controller_override.h"
#include "api/field_trials_view.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/sequence_checker.h"
#include "api/task_queue/task_queue_base.h"
#include "api/task_queue/task_queue_factory.h"
#include "api/video_codecs/video_encoder.h"
#include "call/rtp_config.h"
#include "call/rtp_payload_params.h"
#include "call/rtp_transport_controller_send_interface.h"
#include "call/rtp_video_sender_interface.h"
#include "modules/rtp_rtcp/include/flexfec_sender.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
#include "modules/rtp_rtcp/source/rtp_sender.h"
#include "modules/rtp_rtcp/source/rtp_sender_video.h"
#include "modules/rtp_rtcp/source/rtp_sequence_number_map.h"
#include "modules/rtp_rtcp/source/rtp_video_header.h"
#include "rtc_base/rate_limiter.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/thread_annotations.h"
namespace webrtc {
class FrameEncryptorInterface;
class RtpTransportControllerSendInterface;
namespace webrtc_internal_rtp_video_sender {
// RTP state for a single simulcast stream. Internal to the implementation of
// RtpVideoSender.
struct RtpStreamSender {
RtpStreamSender(std::unique_ptr<ModuleRtpRtcpImpl2> rtp_rtcp,
std::unique_ptr<RTPSenderVideo> sender_video,
std::unique_ptr<VideoFecGenerator> fec_generator);
~RtpStreamSender();
RtpStreamSender(RtpStreamSender&&) = default;
RtpStreamSender& operator=(RtpStreamSender&&) = default;
// Note: Needs pointer stability.
std::unique_ptr<ModuleRtpRtcpImpl2> rtp_rtcp;
std::unique_ptr<RTPSenderVideo> sender_video;
std::unique_ptr<VideoFecGenerator> fec_generator;
};
} // namespace webrtc_internal_rtp_video_sender
// RtpVideoSender routes outgoing data to the correct sending RTP module, based
// on the simulcast layer in RTPVideoHeader.
class RtpVideoSender : public RtpVideoSenderInterface,
public VCMProtectionCallback,
public StreamFeedbackObserver {
public:
// Rtp modules are assumed to be sorted in simulcast index order.
RtpVideoSender(
Clock* clock,
const std::map<uint32_t, RtpState>& suspended_ssrcs,
const std::map<uint32_t, RtpPayloadState>& states,
const RtpConfig& rtp_config,
int rtcp_report_interval_ms,
Transport* send_transport,
const RtpSenderObservers& observers,
RtpTransportControllerSendInterface* transport,
RtcEventLog* event_log,
RateLimiter* retransmission_limiter, // move inside RtpTransport
std::unique_ptr<FecController> fec_controller,
FrameEncryptorInterface* frame_encryptor,
const CryptoOptions& crypto_options, // move inside RtpTransport
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
const FieldTrialsView& field_trials,
TaskQueueFactory* task_queue_factory);
~RtpVideoSender() override;
RtpVideoSender(const RtpVideoSender&) = delete;
RtpVideoSender& operator=(const RtpVideoSender&) = delete;
// Sets the sending status of the rtp modules and appropriately sets the
// payload router to active if any rtp modules are active.
void SetActiveModules(const std::vector<bool>& active_modules)
RTC_LOCKS_EXCLUDED(mutex_) override;
void Stop() RTC_LOCKS_EXCLUDED(mutex_) override;
bool IsActive() RTC_LOCKS_EXCLUDED(mutex_) override;
void OnNetworkAvailability(bool network_available)
RTC_LOCKS_EXCLUDED(mutex_) override;
std::map<uint32_t, RtpState> GetRtpStates() const
RTC_LOCKS_EXCLUDED(mutex_) override;
std::map<uint32_t, RtpPayloadState> GetRtpPayloadStates() const
RTC_LOCKS_EXCLUDED(mutex_) override;
void DeliverRtcp(const uint8_t* packet, size_t length)
RTC_LOCKS_EXCLUDED(mutex_) override;
// Implements webrtc::VCMProtectionCallback.
int ProtectionRequest(const FecProtectionParams* delta_params,
const FecProtectionParams* key_params,
uint32_t* sent_video_rate_bps,
uint32_t* sent_nack_rate_bps,
uint32_t* sent_fec_rate_bps)
RTC_LOCKS_EXCLUDED(mutex_) override;
// Implements FecControllerOverride.
void SetFecAllowed(bool fec_allowed) RTC_LOCKS_EXCLUDED(mutex_) override;
// Implements EncodedImageCallback.
// Returns 0 if the packet was routed / sent, -1 otherwise.
EncodedImageCallback::Result OnEncodedImage(
const EncodedImage& encoded_image,
const CodecSpecificInfo* codec_specific_info)
RTC_LOCKS_EXCLUDED(mutex_) override;
void OnBitrateAllocationUpdated(const VideoBitrateAllocation& bitrate)
RTC_LOCKS_EXCLUDED(mutex_) override;
void OnVideoLayersAllocationUpdated(
const VideoLayersAllocation& layers) override;
void OnTransportOverheadChanged(size_t transport_overhead_bytes_per_packet)
RTC_LOCKS_EXCLUDED(mutex_) override;
void OnBitrateUpdated(BitrateAllocationUpdate update, int framerate)
RTC_LOCKS_EXCLUDED(mutex_) override;
uint32_t GetPayloadBitrateBps() const RTC_LOCKS_EXCLUDED(mutex_) override;
uint32_t GetProtectionBitrateBps() const RTC_LOCKS_EXCLUDED(mutex_) override;
void SetEncodingData(size_t width, size_t height, size_t num_temporal_layers)
RTC_LOCKS_EXCLUDED(mutex_) override;
std::vector<RtpSequenceNumberMap::Info> GetSentRtpPacketInfos(
uint32_t ssrc,
rtc::ArrayView<const uint16_t> sequence_numbers) const
RTC_LOCKS_EXCLUDED(mutex_) override;
// From StreamFeedbackObserver.
void OnPacketFeedbackVector(
std::vector<StreamPacketInfo> packet_feedback_vector)
RTC_LOCKS_EXCLUDED(mutex_) override;
private:
bool IsActiveLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
void SetActiveModulesLocked(const std::vector<bool>& active_modules)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
void UpdateModuleSendingState() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
void ConfigureProtection();
void ConfigureSsrcs(const std::map<uint32_t, RtpState>& suspended_ssrcs);
bool NackEnabled() const;
uint32_t GetPacketizationOverheadRate() const;
DataRate CalculateOverheadRate(DataRate data_rate,
DataSize packet_size,
DataSize overhead_per_packet,
Frequency framerate) const;
const FieldTrialsView& field_trials_;
const bool use_frame_rate_for_overhead_;
const bool has_packet_feedback_;
// Semantically equivalent to checking for `transport_->GetWorkerQueue()`
// but some tests need to be updated to call from the correct context.
RTC_NO_UNIQUE_ADDRESS SequenceChecker transport_checker_;
// TODO(bugs.webrtc.org/13517): Remove mutex_ once RtpVideoSender runs on the
// transport task queue.
mutable Mutex mutex_;
bool active_ RTC_GUARDED_BY(mutex_);
bool registered_for_feedback_ RTC_GUARDED_BY(transport_checker_) = false;
const std::unique_ptr<FecController> fec_controller_;
bool fec_allowed_ RTC_GUARDED_BY(mutex_);
// Rtp modules are assumed to be sorted in simulcast index order.
const std::vector<webrtc_internal_rtp_video_sender::RtpStreamSender>
rtp_streams_;
const RtpConfig rtp_config_;
const absl::optional<VideoCodecType> codec_type_;
RtpTransportControllerSendInterface* const transport_;
// When using the generic descriptor we want all simulcast streams to share
// one frame id space (so that the SFU can switch stream without having to
// rewrite the frame id), therefore `shared_frame_id` has to live in a place
// where we are aware of all the different streams.
int64_t shared_frame_id_ = 0;
std::vector<RtpPayloadParams> params_ RTC_GUARDED_BY(mutex_);
size_t transport_overhead_bytes_per_packet_ RTC_GUARDED_BY(mutex_);
uint32_t protection_bitrate_bps_;
uint32_t encoder_target_rate_bps_;
std::vector<bool> loss_mask_vector_ RTC_GUARDED_BY(mutex_);
std::vector<FrameCounts> frame_counts_ RTC_GUARDED_BY(mutex_);
FrameCountObserver* const frame_count_observer_;
// Effectively const map from SSRC to RtpRtcp, for all media SSRCs.
// This map is set at construction time and never changed, but it's
// non-trivial to make it properly const.
std::map<uint32_t, RtpRtcpInterface*> ssrc_to_rtp_module_;
};
} // namespace webrtc
#endif // CALL_RTP_VIDEO_SENDER_H_
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