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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_RECEIVER_H_
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_H_
#include <map>
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/bitrate.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_payload_registry.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class RtpRtcpFeedback;
class ModuleRtpRtcpImpl;
class Trace;
class RTPReceiverAudio;
class RTPReceiverVideo;
class RTPReceiverStrategy;
class RTPReceiver : public Bitrate {
public:
// Callbacks passed in here may not be NULL (use Null Object callbacks if you
// want callbacks to do nothing). This class takes ownership of the media
// receiver but nothing else.
RTPReceiver(const int32_t id,
Clock* clock,
ModuleRtpRtcpImpl* owner,
RtpAudioFeedback* incoming_audio_messages_callback,
RtpData* incoming_payload_callback,
RtpFeedback* incoming_messages_callback,
RTPReceiverStrategy* rtp_media_receiver,
RTPPayloadRegistry* rtp_payload_registry);
virtual ~RTPReceiver();
RtpVideoCodecTypes VideoCodecType() const;
uint32_t MaxConfiguredBitrate() const;
int32_t SetPacketTimeout(const uint32_t timeout_ms);
void PacketTimeout();
void ProcessDeadOrAlive(const bool RTCPalive, const int64_t now);
void ProcessBitrate();
int32_t RegisterReceivePayload(
const char payload_name[RTP_PAYLOAD_NAME_SIZE],
const int8_t payload_type,
const uint32_t frequency,
const uint8_t channels,
const uint32_t rate);
int32_t DeRegisterReceivePayload(const int8_t payload_type);
int32_t ReceivePayloadType(
const char payload_name[RTP_PAYLOAD_NAME_SIZE],
const uint32_t frequency,
const uint8_t channels,
const uint32_t rate,
int8_t* payload_type) const;
int32_t IncomingRTPPacket(
RTPHeader* rtpheader,
const uint8_t* incoming_rtp_packet,
const uint16_t incoming_rtp_packet_length);
NACKMethod NACK() const ;
// Turn negative acknowledgement requests on/off.
int32_t SetNACKStatus(const NACKMethod method, int max_reordering_threshold);
// Returns the last received timestamp.
virtual uint32_t TimeStamp() const;
int32_t LastReceivedTimeMs() const;
virtual uint16_t SequenceNumber() const;
int32_t EstimatedRemoteTimeStamp(uint32_t& timestamp) const;
uint32_t SSRC() const;
int32_t CSRCs(uint32_t array_of_csrc[kRtpCsrcSize]) const;
int32_t Energy(uint8_t array_of_energy[kRtpCsrcSize]) const;
// Get the currently configured SSRC filter.
int32_t SSRCFilter(uint32_t& allowed_ssrc) const;
// Set a SSRC to be used as a filter for incoming RTP streams.
int32_t SetSSRCFilter(const bool enable, const uint32_t allowed_ssrc);
int32_t Statistics(uint8_t* fraction_lost,
uint32_t* cum_lost,
uint32_t* ext_max,
uint32_t* jitter, // Will be moved from JB.
uint32_t* max_jitter,
uint32_t* jitter_transmission_time_offset,
bool reset) const;
int32_t Statistics(uint8_t* fraction_lost,
uint32_t* cum_lost,
uint32_t* ext_max,
uint32_t* jitter, // Will be moved from JB.
uint32_t* max_jitter,
uint32_t* jitter_transmission_time_offset,
int32_t* missing,
bool reset) const;
int32_t DataCounters(uint32_t* bytes_received,
uint32_t* packets_received) const;
int32_t ResetStatistics();
int32_t ResetDataCounters();
uint16_t PacketOHReceived() const;
uint32_t PacketCountReceived() const;
uint32_t ByteCountReceived() const;
int32_t RegisterRtpHeaderExtension(const RTPExtensionType type,
const uint8_t id);
int32_t DeregisterRtpHeaderExtension(const RTPExtensionType type);
void GetHeaderExtensionMapCopy(RtpHeaderExtensionMap* map) const;
// RTX.
void SetRTXStatus(bool enable, uint32_t ssrc);
void RTXStatus(bool* enable, uint32_t* ssrc, int* payload_type) const;
void SetRtxPayloadType(int payload_type);
virtual int8_t REDPayloadType() const;
bool HaveNotReceivedPackets() const;
virtual bool RetransmitOfOldPacket(const uint16_t sequence_number,
const uint32_t rtp_time_stamp) const;
void UpdateStatistics(const RTPHeader* rtp_header,
const uint16_t bytes,
const bool old_packet);
private:
// Returns whether RED is configured with payload_type.
bool REDPayloadType(const int8_t payload_type) const;
bool InOrderPacket(const uint16_t sequence_number) const;
void CheckSSRCChanged(const RTPHeader* rtp_header);
void CheckCSRC(const WebRtcRTPHeader* rtp_header);
int32_t CheckPayloadChanged(const RTPHeader* rtp_header,
const int8_t first_payload_byte,
bool& isRED,
ModuleRTPUtility::PayloadUnion* payload);
void UpdateNACKBitRate(int32_t bytes, uint32_t now);
bool ProcessNACKBitRate(uint32_t now);
RTPPayloadRegistry* rtp_payload_registry_;
scoped_ptr<RTPReceiverStrategy> rtp_media_receiver_;
int32_t id_;
ModuleRtpRtcpImpl& rtp_rtcp_;
RtpFeedback* cb_rtp_feedback_;
CriticalSectionWrapper* critical_section_rtp_receiver_;
mutable int64_t last_receive_time_;
uint16_t last_received_payload_length_;
uint32_t packet_timeout_ms_;
// SSRCs.
uint32_t ssrc_;
uint8_t num_csrcs_;
uint32_t current_remote_csrc_[kRtpCsrcSize];
uint8_t num_energy_;
uint8_t current_remote_energy_[kRtpCsrcSize];
bool use_ssrc_filter_;
uint32_t ssrc_filter_;
// Stats on received RTP packets.
uint32_t jitter_q4_;
mutable uint32_t jitter_max_q4_;
mutable uint32_t cumulative_loss_;
uint32_t jitter_q4_transmission_time_offset_;
uint32_t local_time_last_received_timestamp_;
int64_t last_received_frame_time_ms_;
uint32_t last_received_timestamp_;
uint16_t last_received_sequence_number_;
int32_t last_received_transmission_time_offset_;
uint16_t received_seq_first_;
uint16_t received_seq_max_;
uint16_t received_seq_wraps_;
// Current counter values.
uint16_t received_packet_oh_;
uint32_t received_byte_count_;
uint32_t received_old_packet_count_;
uint32_t received_inorder_packet_count_;
// Counter values when we sent the last report.
mutable uint32_t last_report_inorder_packets_;
mutable uint32_t last_report_old_packets_;
mutable uint16_t last_report_seq_max_;
mutable uint8_t last_report_fraction_lost_;
mutable uint32_t last_report_cumulative_lost_; // 24 bits valid.
mutable uint32_t last_report_extended_high_seq_num_;
mutable uint32_t last_report_jitter_;
mutable uint32_t last_report_jitter_transmission_time_offset_;
NACKMethod nack_method_;
int max_reordering_threshold_;
bool rtx_;
uint32_t ssrc_rtx_;
int payload_type_rtx_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_H_