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webrtc_m130/webrtc/modules/rtp_rtcp/source/rtp_sender.h
Danil Chapovalov 5e57b17283 Introduce helpers to RtpSender to propagate RtpPacketToSend. 
The helpers intended to replace and deprecate BuildRtpHeader when
RtpSenderAudio/RtpSenderVideo will be updated to pass RtpPacket class
instead of raw buffer for sending.

BUG=webrtc:5261
R=sprang@webrtc.org

Review URL: https://codereview.webrtc.org/2303283002 .

Cr-Commit-Position: refs/heads/master@{#14051}
2016-09-02 17:16:08 +00:00

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/*
* Copyright (c) 2012 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 WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
#include <map>
#include <memory>
#include <utility>
#include <vector>
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/random.h"
#include "webrtc/base/rate_statistics.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/playout_delay_oracle.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_packet_history.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "webrtc/modules/rtp_rtcp/source/ssrc_database.h"
#include "webrtc/transport.h"
namespace webrtc {
class RateLimiter;
class RtcEventLog;
class RtpPacketToSend;
class RTPSenderAudio;
class RTPSenderVideo;
class RTPSender {
public:
RTPSender(bool audio,
Clock* clock,
Transport* transport,
RtpPacketSender* paced_sender,
TransportSequenceNumberAllocator* sequence_number_allocator,
TransportFeedbackObserver* transport_feedback_callback,
BitrateStatisticsObserver* bitrate_callback,
FrameCountObserver* frame_count_observer,
SendSideDelayObserver* send_side_delay_observer,
RtcEventLog* event_log,
SendPacketObserver* send_packet_observer,
RateLimiter* nack_rate_limiter);
~RTPSender();
void ProcessBitrate();
uint16_t ActualSendBitrateKbit() const;
uint32_t VideoBitrateSent() const;
uint32_t FecOverheadRate() const;
uint32_t NackOverheadRate() const;
// Includes size of RTP and FEC headers.
size_t MaxDataPayloadLength() const;
int32_t RegisterPayload(const char* payload_name,
const int8_t payload_type,
const uint32_t frequency,
const size_t channels,
const uint32_t rate);
int32_t DeRegisterSendPayload(const int8_t payload_type);
void SetSendPayloadType(int8_t payload_type);
int8_t SendPayloadType() const;
int SendPayloadFrequency() const;
void SetSendingStatus(bool enabled);
void SetSendingMediaStatus(bool enabled);
bool SendingMedia() const;
void GetDataCounters(StreamDataCounters* rtp_stats,
StreamDataCounters* rtx_stats) const;
uint32_t TimestampOffset() const;
void SetTimestampOffset(uint32_t timestamp);
uint32_t GenerateNewSSRC();
void SetSSRC(uint32_t ssrc);
uint16_t SequenceNumber() const;
void SetSequenceNumber(uint16_t seq);
void SetCsrcs(const std::vector<uint32_t>& csrcs);
void SetMaxPayloadLength(size_t max_payload_length);
bool SendOutgoingData(FrameType frame_type,
int8_t payload_type,
uint32_t timestamp,
int64_t capture_time_ms,
const uint8_t* payload_data,
size_t payload_size,
const RTPFragmentationHeader* fragmentation,
const RTPVideoHeader* rtp_header,
uint32_t* transport_frame_id_out);
// RTP header extension
int32_t SetTransmissionTimeOffset(int32_t transmission_time_offset);
int32_t SetAbsoluteSendTime(uint32_t absolute_send_time);
void SetVideoRotation(VideoRotation rotation);
int32_t SetTransportSequenceNumber(uint16_t sequence_number);
int32_t RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id);
bool IsRtpHeaderExtensionRegistered(RTPExtensionType type);
int32_t DeregisterRtpHeaderExtension(RTPExtensionType type);
size_t RtpHeaderExtensionLength() const;
uint16_t BuildRtpHeaderExtension(uint8_t* data_buffer, bool marker_bit) const
EXCLUSIVE_LOCKS_REQUIRED(send_critsect_);
uint8_t BuildTransmissionTimeOffsetExtension(uint8_t* data_buffer) const
EXCLUSIVE_LOCKS_REQUIRED(send_critsect_);
uint8_t BuildAudioLevelExtension(uint8_t* data_buffer) const
EXCLUSIVE_LOCKS_REQUIRED(send_critsect_);
uint8_t BuildAbsoluteSendTimeExtension(uint8_t* data_buffer) const
EXCLUSIVE_LOCKS_REQUIRED(send_critsect_);
uint8_t BuildVideoRotationExtension(uint8_t* data_buffer) const
EXCLUSIVE_LOCKS_REQUIRED(send_critsect_);
uint8_t BuildTransportSequenceNumberExtension(uint8_t* data_buffer,
uint16_t sequence_number) const
EXCLUSIVE_LOCKS_REQUIRED(send_critsect_);
uint8_t BuildPlayoutDelayExtension(uint8_t* data_buffer,
uint16_t min_playout_delay_ms,
uint16_t max_playout_delay_ms) const
EXCLUSIVE_LOCKS_REQUIRED(send_critsect_);
// Verifies that the specified extension is registered, and that it is
// present in rtp packet. If extension is not registered kNotRegistered is
// returned. If extension cannot be found in the rtp header, or if it is
// malformed, kError is returned. Otherwise *extension_offset is set to the
// offset of the extension from the beginning of the rtp packet and kOk is
// returned.
enum class ExtensionStatus {
kNotRegistered,
kOk,
kError,
};
ExtensionStatus VerifyExtension(RTPExtensionType extension_type,
uint8_t* rtp_packet,
size_t rtp_packet_length,
const RTPHeader& rtp_header,
size_t extension_length_bytes,
size_t* extension_offset) const
EXCLUSIVE_LOCKS_REQUIRED(send_critsect_);
bool UpdateAudioLevel(uint8_t* rtp_packet,
size_t rtp_packet_length,
const RTPHeader& rtp_header,
bool is_voiced,
uint8_t dBov) const;
bool UpdateVideoRotation(uint8_t* rtp_packet,
size_t rtp_packet_length,
const RTPHeader& rtp_header,
VideoRotation rotation) const;
bool TimeToSendPacket(uint16_t sequence_number,
int64_t capture_time_ms,
bool retransmission,
int probe_cluster_id);
size_t TimeToSendPadding(size_t bytes, int probe_cluster_id);
// NACK.
int SelectiveRetransmissions() const;
int SetSelectiveRetransmissions(uint8_t settings);
void OnReceivedNack(const std::vector<uint16_t>& nack_sequence_numbers,
int64_t avg_rtt);
void SetStorePacketsStatus(bool enable, uint16_t number_to_store);
bool StorePackets() const;
int32_t ReSendPacket(uint16_t packet_id, int64_t min_resend_time = 0);
// Feedback to decide when to stop sending playout delay.
void OnReceivedRtcpReportBlocks(const ReportBlockList& report_blocks);
// RTX.
void SetRtxStatus(int mode);
int RtxStatus() const;
uint32_t RtxSsrc() const;
void SetRtxSsrc(uint32_t ssrc);
void SetRtxPayloadType(int payload_type, int associated_payload_type);
// Create empty packet, fills ssrc, csrcs and reserve place for header
// extensions RtpSender updates before sending.
std::unique_ptr<RtpPacketToSend> AllocatePacket() const;
// Allocate sequence number for provided packet.
// Save packet's fields to generate padding that doesn't break media stream.
// Return false if sending was turned off.
bool AssignSequenceNumber(RtpPacketToSend* packet);
// Functions wrapping RTPSenderInterface.
int32_t BuildRTPheader(uint8_t* data_buffer,
int8_t payload_type,
bool marker_bit,
uint32_t capture_timestamp,
int64_t capture_time_ms,
bool timestamp_provided = true,
bool inc_sequence_number = true);
int32_t BuildRtpHeader(uint8_t* data_buffer,
int8_t payload_type,
bool marker_bit,
uint32_t capture_timestamp,
int64_t capture_time_ms);
size_t RtpHeaderLength() const;
uint16_t AllocateSequenceNumber(uint16_t packets_to_send);
size_t MaxPayloadLength() const;
uint32_t SSRC() const;
// Deprecated. Create RtpPacketToSend instead and use next function.
bool SendToNetwork(uint8_t* data_buffer,
size_t payload_length,
size_t rtp_header_length,
int64_t capture_time_ms,
StorageType storage,
RtpPacketSender::Priority priority);
bool SendToNetwork(std::unique_ptr<RtpPacketToSend> packet,
StorageType storage,
RtpPacketSender::Priority priority);
// Audio.
// Send a DTMF tone using RFC 2833 (4733).
int32_t SendTelephoneEvent(uint8_t key, uint16_t time_ms, uint8_t level);
// Set audio packet size, used to determine when it's time to send a DTMF
// packet in silence (CNG).
int32_t SetAudioPacketSize(uint16_t packet_size_samples);
// Store the audio level in d_bov for
// header-extension-for-audio-level-indication.
int32_t SetAudioLevel(uint8_t level_d_bov);
RtpVideoCodecTypes VideoCodecType() const;
uint32_t MaxConfiguredBitrateVideo() const;
// FEC.
void SetGenericFECStatus(bool enable,
uint8_t payload_type_red,
uint8_t payload_type_fec);
void GenericFECStatus(bool* enable,
uint8_t* payload_type_red,
uint8_t* payload_type_fec) const;
int32_t SetFecParameters(const FecProtectionParams *delta_params,
const FecProtectionParams *key_params);
size_t SendPadData(size_t bytes,
bool timestamp_provided,
uint32_t timestamp,
int64_t capture_time_ms);
size_t SendPadData(size_t bytes,
bool timestamp_provided,
uint32_t timestamp,
int64_t capture_time_ms,
int probe_cluster_id);
// Called on update of RTP statistics.
void RegisterRtpStatisticsCallback(StreamDataCountersCallback* callback);
StreamDataCountersCallback* GetRtpStatisticsCallback() const;
uint32_t BitrateSent() const;
void SetRtpState(const RtpState& rtp_state);
RtpState GetRtpState() const;
void SetRtxRtpState(const RtpState& rtp_state);
RtpState GetRtxRtpState() const;
bool ActivateCVORtpHeaderExtension();
protected:
int32_t CheckPayloadType(int8_t payload_type, RtpVideoCodecTypes* video_type);
private:
// Maps capture time in milliseconds to send-side delay in milliseconds.
// Send-side delay is the difference between transmission time and capture
// time.
typedef std::map<int64_t, int> SendDelayMap;
size_t CreateRtpHeader(uint8_t* header,
int8_t payload_type,
uint32_t ssrc,
bool marker_bit,
uint32_t timestamp,
uint16_t sequence_number,
const std::vector<uint32_t>& csrcs) const
EXCLUSIVE_LOCKS_REQUIRED(send_critsect_);
bool PrepareAndSendPacket(std::unique_ptr<RtpPacketToSend> packet,
bool send_over_rtx,
bool is_retransmit,
int probe_cluster_id);
// Return the number of bytes sent. Note that both of these functions may
// return a larger value that their argument.
size_t TrySendRedundantPayloads(size_t bytes, int probe_cluster_id);
std::unique_ptr<RtpPacketToSend> BuildRtxPacket(
const RtpPacketToSend& packet);
bool SendPacketToNetwork(const RtpPacketToSend& packet,
const PacketOptions& options);
void UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms);
void UpdateOnSendPacket(int packet_id,
int64_t capture_time_ms,
uint32_t ssrc);
// Find the byte position of the RTP extension as indicated by |type| in
// |rtp_packet|. Return false if such extension doesn't exist.
bool FindHeaderExtensionPosition(RTPExtensionType type,
const uint8_t* rtp_packet,
size_t rtp_packet_length,
const RTPHeader& rtp_header,
size_t* position) const
EXCLUSIVE_LOCKS_REQUIRED(send_critsect_);
bool UpdateTransportSequenceNumber(RtpPacketToSend* packet,
int* packet_id) const;
void UpdatePlayoutDelayLimits(uint8_t* rtp_packet,
size_t rtp_packet_length,
const RTPHeader& rtp_header,
uint16_t min_playout_delay,
uint16_t max_playout_delay) const;
void UpdateRtpStats(const RtpPacketToSend& packet,
bool is_rtx,
bool is_retransmit);
bool IsFecPacket(const RtpPacketToSend& packet) const;
Clock* const clock_;
const int64_t clock_delta_ms_;
Random random_ GUARDED_BY(send_critsect_);
const bool audio_configured_;
const std::unique_ptr<RTPSenderAudio> audio_;
const std::unique_ptr<RTPSenderVideo> video_;
RtpPacketSender* const paced_sender_;
TransportSequenceNumberAllocator* const transport_sequence_number_allocator_;
TransportFeedbackObserver* const transport_feedback_observer_;
int64_t last_capture_time_ms_sent_;
rtc::CriticalSection send_critsect_;
Transport *transport_;
bool sending_media_ GUARDED_BY(send_critsect_);
size_t max_payload_length_;
int8_t payload_type_ GUARDED_BY(send_critsect_);
std::map<int8_t, RtpUtility::Payload*> payload_type_map_;
RtpHeaderExtensionMap rtp_header_extension_map_ GUARDED_BY(send_critsect_);
int32_t transmission_time_offset_;
uint32_t absolute_send_time_;
VideoRotation rotation_;
bool video_rotation_active_;
uint16_t transport_sequence_number_;
// Tracks the current request for playout delay limits from application
// and decides whether the current RTP frame should include the playout
// delay extension on header.
PlayoutDelayOracle playout_delay_oracle_;
bool playout_delay_active_ GUARDED_BY(send_critsect_);
RtpPacketHistory packet_history_;
// Statistics
rtc::CriticalSection statistics_crit_;
SendDelayMap send_delays_ GUARDED_BY(statistics_crit_);
FrameCounts frame_counts_ GUARDED_BY(statistics_crit_);
StreamDataCounters rtp_stats_ GUARDED_BY(statistics_crit_);
StreamDataCounters rtx_rtp_stats_ GUARDED_BY(statistics_crit_);
StreamDataCountersCallback* rtp_stats_callback_ GUARDED_BY(statistics_crit_);
RateStatistics total_bitrate_sent_ GUARDED_BY(statistics_crit_);
RateStatistics nack_bitrate_sent_ GUARDED_BY(statistics_crit_);
FrameCountObserver* const frame_count_observer_;
SendSideDelayObserver* const send_side_delay_observer_;
RtcEventLog* const event_log_;
SendPacketObserver* const send_packet_observer_;
BitrateStatisticsObserver* const bitrate_callback_;
// RTP variables
uint32_t timestamp_offset_ GUARDED_BY(send_critsect_);
SSRCDatabase* const ssrc_db_;
uint32_t remote_ssrc_ GUARDED_BY(send_critsect_);
bool sequence_number_forced_ GUARDED_BY(send_critsect_);
uint16_t sequence_number_ GUARDED_BY(send_critsect_);
uint16_t sequence_number_rtx_ GUARDED_BY(send_critsect_);
bool ssrc_forced_ GUARDED_BY(send_critsect_);
uint32_t ssrc_ GUARDED_BY(send_critsect_);
uint32_t last_rtp_timestamp_ GUARDED_BY(send_critsect_);
int64_t capture_time_ms_ GUARDED_BY(send_critsect_);
int64_t last_timestamp_time_ms_ GUARDED_BY(send_critsect_);
bool media_has_been_sent_ GUARDED_BY(send_critsect_);
bool last_packet_marker_bit_ GUARDED_BY(send_critsect_);
std::vector<uint32_t> csrcs_ GUARDED_BY(send_critsect_);
int rtx_ GUARDED_BY(send_critsect_);
uint32_t ssrc_rtx_ GUARDED_BY(send_critsect_);
// Mapping rtx_payload_type_map_[associated] = rtx.
std::map<int8_t, int8_t> rtx_payload_type_map_ GUARDED_BY(send_critsect_);
RateLimiter* const retransmission_rate_limiter_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSender);
};
} // namespace webrtc
#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
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