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Bug: webrtc:10633 Change-Id: I835686f58f9edcf0c7cec8f0b3d54eb93f2920df Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143176 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28349}
227 lines
8.9 KiB
C++
227 lines
8.9 KiB
C++
/*
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* Copyright (c) 2012 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 MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_
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#define MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_
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#include <map>
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#include <memory>
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#include <vector>
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#include "absl/strings/string_view.h"
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#include "absl/types/optional.h"
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#include "api/array_view.h"
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#include "modules/rtp_rtcp/include/flexfec_sender.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "modules/rtp_rtcp/source/playout_delay_oracle.h"
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#include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
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#include "modules/rtp_rtcp/source/rtp_sender.h"
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#include "modules/rtp_rtcp/source/rtp_sequence_number_map.h"
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#include "modules/rtp_rtcp/source/ulpfec_generator.h"
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#include "rtc_base/critical_section.h"
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#include "rtc_base/one_time_event.h"
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#include "rtc_base/rate_statistics.h"
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#include "rtc_base/synchronization/sequence_checker.h"
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#include "rtc_base/thread_annotations.h"
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namespace webrtc {
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class FrameEncryptorInterface;
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class RtpPacketizer;
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class RtpPacketToSend;
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// kConditionallyRetransmitHigherLayers allows retransmission of video frames
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// in higher layers if either the last frame in that layer was too far back in
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// time, or if we estimate that a new frame will be available in a lower layer
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// in a shorter time than it would take to request and receive a retransmission.
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enum RetransmissionMode : uint8_t {
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kRetransmitOff = 0x0,
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kRetransmitBaseLayer = 0x2,
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kRetransmitHigherLayers = 0x4,
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kConditionallyRetransmitHigherLayers = 0x8,
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};
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class RTPSenderVideo {
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public:
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static constexpr int64_t kTLRateWindowSizeMs = 2500;
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RTPSenderVideo(Clock* clock,
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RTPSender* rtpSender,
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FlexfecSender* flexfec_sender,
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PlayoutDelayOracle* playout_delay_oracle,
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FrameEncryptorInterface* frame_encryptor,
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bool require_frame_encryption,
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bool need_rtp_packet_infos,
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const WebRtcKeyValueConfig& field_trials);
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virtual ~RTPSenderVideo();
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// expected_retransmission_time_ms.has_value() -> retransmission allowed.
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bool SendVideo(VideoFrameType frame_type,
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int8_t payload_type,
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uint32_t capture_timestamp,
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int64_t capture_time_ms,
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const uint8_t* payload_data,
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size_t payload_size,
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const RTPFragmentationHeader* fragmentation,
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const RTPVideoHeader* video_header,
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absl::optional<int64_t> expected_retransmission_time_ms);
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void RegisterPayloadType(int8_t payload_type,
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absl::string_view payload_name,
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bool raw_payload);
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// Set RED and ULPFEC payload types. A payload type of -1 means that the
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// corresponding feature is turned off. Note that we DO NOT support enabling
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// ULPFEC without enabling RED, and RED is only ever used when ULPFEC is
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// enabled.
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void SetUlpfecConfig(int red_payload_type, int ulpfec_payload_type);
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// FlexFEC/ULPFEC.
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// Set FEC rates, max frames before FEC is sent, and type of FEC masks.
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// Returns false on failure.
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void SetFecParameters(const FecProtectionParams& delta_params,
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const FecProtectionParams& key_params);
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// FlexFEC.
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absl::optional<uint32_t> FlexfecSsrc() const;
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uint32_t VideoBitrateSent() const;
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uint32_t FecOverheadRate() const;
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// Returns the current packetization overhead rate, in bps. Note that this is
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// the payload overhead, eg the VP8 payload headers, not the RTP headers
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// or extension/
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uint32_t PacketizationOverheadBps() const;
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// For each sequence number in |sequence_number|, recall the last RTP packet
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// which bore it - its timestamp and whether it was the first and/or last
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// packet in that frame. If all of the given sequence numbers could be
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// recalled, return a vector with all of them (in corresponding order).
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// If any could not be recalled, return an empty vector.
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std::vector<RtpSequenceNumberMap::Info> GetSentRtpPacketInfos(
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rtc::ArrayView<const uint16_t> sequence_numbers) const;
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protected:
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static uint8_t GetTemporalId(const RTPVideoHeader& header);
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StorageType GetStorageType(uint8_t temporal_id,
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int32_t retransmission_settings,
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int64_t expected_retransmission_time_ms);
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private:
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struct TemporalLayerStats {
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TemporalLayerStats()
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: frame_rate_fp1000s(kTLRateWindowSizeMs, 1000 * 1000),
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last_frame_time_ms(0) {}
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// Frame rate, in frames per 1000 seconds. This essentially turns the fps
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// value into a fixed point value with three decimals. Improves precision at
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// low frame rates.
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RateStatistics frame_rate_fp1000s;
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int64_t last_frame_time_ms;
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};
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size_t CalculateFecPacketOverhead() const RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
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void SendVideoPacket(std::unique_ptr<RtpPacketToSend> packet,
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StorageType storage);
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void SendVideoPacketAsRedMaybeWithUlpfec(
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std::unique_ptr<RtpPacketToSend> media_packet,
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StorageType media_packet_storage,
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bool protect_media_packet);
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// TODO(brandtr): Remove the FlexFEC functions when FlexfecSender has been
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// moved to PacedSender.
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void SendVideoPacketWithFlexfec(std::unique_ptr<RtpPacketToSend> media_packet,
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StorageType media_packet_storage,
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bool protect_media_packet);
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bool LogAndSendToNetwork(std::unique_ptr<RtpPacketToSend> packet,
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StorageType storage);
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bool red_enabled() const RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_) {
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return red_payload_type_ >= 0;
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}
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bool ulpfec_enabled() const RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_) {
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return ulpfec_payload_type_ >= 0;
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}
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bool flexfec_enabled() const { return flexfec_sender_ != nullptr; }
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bool UpdateConditionalRetransmit(uint8_t temporal_id,
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int64_t expected_retransmission_time_ms)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(stats_crit_);
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RTPSender* const rtp_sender_;
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Clock* const clock_;
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// Maps payload type to codec type, for packetization.
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// TODO(nisse): Set on construction, to avoid lock.
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rtc::CriticalSection payload_type_crit_;
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std::map<int8_t, absl::optional<VideoCodecType>> payload_type_map_
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RTC_GUARDED_BY(payload_type_crit_);
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// Should never be held when calling out of this class.
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rtc::CriticalSection crit_;
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int32_t retransmission_settings_ RTC_GUARDED_BY(crit_);
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VideoRotation last_rotation_ RTC_GUARDED_BY(crit_);
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absl::optional<ColorSpace> last_color_space_ RTC_GUARDED_BY(crit_);
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bool transmit_color_space_next_frame_ RTC_GUARDED_BY(crit_);
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// Tracks the current request for playout delay limits from application
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// and decides whether the current RTP frame should include the playout
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// delay extension on header.
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PlayoutDelayOracle* const playout_delay_oracle_;
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// Maps sent packets' sequence numbers to a tuple consisting of:
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// 1. The timestamp, without the randomizing offset mandated by the RFC.
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// 2. Whether the packet was the first in its frame.
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// 3. Whether the packet was the last in its frame.
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const std::unique_ptr<RtpSequenceNumberMap> rtp_sequence_number_map_
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RTC_PT_GUARDED_BY(crit_);
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// RED/ULPFEC.
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int red_payload_type_ RTC_GUARDED_BY(crit_);
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int ulpfec_payload_type_ RTC_GUARDED_BY(crit_);
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UlpfecGenerator ulpfec_generator_ RTC_GUARDED_BY(crit_);
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// FlexFEC.
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FlexfecSender* const flexfec_sender_;
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// FEC parameters, applicable to either ULPFEC or FlexFEC.
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FecProtectionParams delta_fec_params_ RTC_GUARDED_BY(crit_);
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FecProtectionParams key_fec_params_ RTC_GUARDED_BY(crit_);
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rtc::CriticalSection stats_crit_;
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// Bitrate used for FEC payload, RED headers, RTP headers for FEC packets
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// and any padding overhead.
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RateStatistics fec_bitrate_ RTC_GUARDED_BY(stats_crit_);
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// Bitrate used for video payload and RTP headers.
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RateStatistics video_bitrate_ RTC_GUARDED_BY(stats_crit_);
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RateStatistics packetization_overhead_bitrate_ RTC_GUARDED_BY(stats_crit_);
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std::map<int, TemporalLayerStats> frame_stats_by_temporal_layer_
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RTC_GUARDED_BY(stats_crit_);
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OneTimeEvent first_frame_sent_;
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// E2EE Custom Video Frame Encryptor (optional)
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FrameEncryptorInterface* const frame_encryptor_ = nullptr;
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// If set to true will require all outgoing frames to pass through an
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// initialized frame_encryptor_ before being sent out of the network.
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// Otherwise these payloads will be dropped.
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bool require_frame_encryption_;
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// Set to true if the generic descriptor should be authenticated.
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const bool generic_descriptor_auth_experiment_;
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};
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} // namespace webrtc
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#endif // MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_
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