f298855981
Removes all unused features, reducing the exposed interface surface. This makes refactoring and maintenance simpler as we can change TransportFeedbackAdapter without making corresponding changes to RtpVideoSender. Bug: webrtc:9883 Change-Id: If372a868e0765e94df52b4de52d3bb619ce11471 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156943 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29649}
216 lines
8.2 KiB
C++
216 lines
8.2 KiB
C++
/*
|
|
* 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/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<PlayoutDelayOracle> playout_delay_oracle,
|
|
std::unique_ptr<RtpRtcp> rtp_rtcp,
|
|
std::unique_ptr<RTPSenderVideo> sender_video);
|
|
~RtpStreamSender();
|
|
|
|
RtpStreamSender(RtpStreamSender&&) = default;
|
|
RtpStreamSender& operator=(RtpStreamSender&&) = default;
|
|
|
|
// Note: Needs pointer stability.
|
|
std::unique_ptr<PlayoutDelayOracle> playout_delay_oracle;
|
|
std::unique_ptr<RtpRtcp> rtp_rtcp;
|
|
std::unique_ptr<RTPSenderVideo> sender_video;
|
|
};
|
|
|
|
} // 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 {
|
|
public:
|
|
// Rtp modules are assumed to be sorted in simulcast index order.
|
|
RtpVideoSender(
|
|
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;
|
|
|
|
private:
|
|
void UpdateModuleSendingState() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
|
|
void ConfigureProtection();
|
|
void ConfigureSsrcs();
|
|
void ConfigureRids();
|
|
bool FecEnabled() const;
|
|
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_;
|
|
|
|
// 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_;
|
|
|
|
std::unique_ptr<FlexfecSender> flexfec_sender_;
|
|
|
|
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>
|
|
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(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_;
|
|
|
|
RTC_DISALLOW_COPY_AND_ASSIGN(RtpVideoSender);
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // CALL_RTP_VIDEO_SENDER_H_
|