blob: ed89028b1e92c0973ce88636f5882ef7a5975c33 [file] [log] [blame]
* 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.
#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/rtc_event_log/rtc_event_log.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_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 "modules/utility/include/process_thread.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/critical_section.h"
#include "rtc_base/rate_limiter.h"
#include "rtc_base/thread_annotations.h"
#include "rtc_base/thread_checker.h"
namespace webrtc {
class FrameEncryptorInterface;
class RTPFragmentationHeader;
class RtpRtcp;
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<RtpRtcp> rtp_rtcp,
std::unique_ptr<RTPSenderVideo> sender_video,
std::unique_ptr<VideoFecGenerator> fec_generator);
RtpStreamSender(RtpStreamSender&&) = default;
RtpStreamSender& operator=(RtpStreamSender&&) = default;
// Note: Needs pointer stability.
std::unique_ptr<RtpRtcp> 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 OverheadObserver,
public VCMProtectionCallback,
public StreamFeedbackObserver {
// Rtp modules are assumed to be sorted in simulcast index order.
Clock* clock,
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
~RtpVideoSender() override;
// RegisterProcessThread register |module_process_thread| with those objects
// that use it. Registration has to happen on the thread were
// |module_process_thread| was created (libjingle's worker thread).
// TODO(perkj): Replace the use of |module_process_thread| with a TaskQueue,
// maybe |worker_queue|.
void RegisterProcessThread(ProcessThread* module_process_thread) override;
void DeRegisterProcessThread() override;
// RtpVideoSender will only route packets if being active, all packets will be
// dropped otherwise.
void SetActive(bool active) override;
// 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) override;
bool IsActive() override;
void OnNetworkAvailability(bool network_available) override;
std::map<uint32_t, RtpState> GetRtpStates() const override;
std::map<uint32_t, RtpPayloadState> GetRtpPayloadStates() const override;
void DeliverRtcp(const uint8_t* packet, size_t length) 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) override;
// Implements FecControllerOverride.
void SetFecAllowed(bool fec_allowed) 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,
const RTPFragmentationHeader* fragmentation) override;
void OnBitrateAllocationUpdated(
const VideoBitrateAllocation& bitrate) override;
void OnTransportOverheadChanged(
size_t transport_overhead_bytes_per_packet) override;
// Implements OverheadObserver.
void OnOverheadChanged(size_t overhead_bytes_per_packet) override;
void OnBitrateUpdated(BitrateAllocationUpdate update, int framerate) override;
uint32_t GetPayloadBitrateBps() const override;
uint32_t GetProtectionBitrateBps() const override;
void SetEncodingData(size_t width,
size_t height,
size_t num_temporal_layers) override;
std::vector<RtpSequenceNumberMap::Info> GetSentRtpPacketInfos(
uint32_t ssrc,
rtc::ArrayView<const uint16_t> sequence_numbers) const override;
// From StreamFeedbackObserver.
void OnPacketFeedbackVector(
std::vector<StreamPacketInfo> packet_feedback_vector) override;
void UpdateModuleSendingState() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
void ConfigureProtection();
void ConfigureSsrcs();
void ConfigureRids();
bool NackEnabled() const;
uint32_t GetPacketizationOverheadRate() const;
const bool send_side_bwe_with_overhead_;
const bool account_for_packetization_overhead_;
const bool use_early_loss_detection_;
const bool has_packet_feedback_;
// TODO(holmer): Remove crit_ once RtpVideoSender runs on the
// transport task queue.
rtc::CriticalSection crit_;
bool active_ RTC_GUARDED_BY(crit_);
ProcessThread* module_process_thread_;
rtc::ThreadChecker module_process_thread_checker_;
std::map<uint32_t, RtpState> suspended_ssrcs_;
const std::unique_ptr<FecController> fec_controller_;
bool fec_allowed_ RTC_GUARDED_BY(crit_);
// Rtp modules are assumed to be sorted in simulcast index order.
const std::vector<webrtc_internal_rtp_video_sender::RtpStreamSender>
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(crit_);
size_t transport_overhead_bytes_per_packet_ RTC_GUARDED_BY(crit_);
size_t overhead_bytes_per_packet_ RTC_GUARDED_BY(crit_);
uint32_t protection_bitrate_bps_;
uint32_t encoder_target_rate_bps_;
std::vector<bool> loss_mask_vector_ RTC_GUARDED_BY(crit_);
std::vector<FrameCounts> frame_counts_ RTC_GUARDED_BY(crit_);
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, RtpRtcp*> ssrc_to_rtp_module_;
} // namespace webrtc