6e95d7afab
This CL makes the padding packets get their own RTP timestamps, rather than having the same timestamp as the last sent video packet. The purpose is to solve Issue 2611, where the overuse- detector does not react to padding packets. A test was implemented to verify that the padding packets do get their own timestamps. BUG=2611 R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3869004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5125 4adac7df-926f-26a2-2b94-8c16560cd09d
347 lines
11 KiB
C++
347 lines
11 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 WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
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#include <assert.h>
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#include <math.h>
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#include <map>
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#include "webrtc/common_types.h"
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#include "webrtc/modules/pacing/include/paced_sender.h"
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#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
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#include "webrtc/modules/rtp_rtcp/source/bitrate.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
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#include "webrtc/modules/rtp_rtcp/source/ssrc_database.h"
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#include "webrtc/modules/rtp_rtcp/source/video_codec_information.h"
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#define MAX_INIT_RTP_SEQ_NUMBER 32767 // 2^15 -1.
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namespace webrtc {
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class CriticalSectionWrapper;
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class RTPPacketHistory;
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class RTPSenderAudio;
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class RTPSenderVideo;
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class RTPSenderInterface {
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public:
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RTPSenderInterface() {}
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virtual ~RTPSenderInterface() {}
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virtual uint32_t SSRC() const = 0;
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virtual uint32_t Timestamp() const = 0;
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virtual int32_t BuildRTPheader(
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uint8_t *data_buffer, const int8_t payload_type,
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const bool marker_bit, const uint32_t capture_time_stamp,
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int64_t capture_time_ms,
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const bool time_stamp_provided = true,
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const bool inc_sequence_number = true) = 0;
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virtual uint16_t RTPHeaderLength() const = 0;
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virtual uint16_t IncrementSequenceNumber() = 0;
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virtual uint16_t SequenceNumber() const = 0;
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virtual uint16_t MaxPayloadLength() const = 0;
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virtual uint16_t MaxDataPayloadLength() const = 0;
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virtual uint16_t PacketOverHead() const = 0;
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virtual uint16_t ActualSendBitrateKbit() const = 0;
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virtual int32_t SendToNetwork(
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uint8_t *data_buffer, int payload_length, int rtp_header_length,
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int64_t capture_time_ms, StorageType storage,
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PacedSender::Priority priority) = 0;
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};
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class RTPSender : public Bitrate, public RTPSenderInterface {
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public:
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RTPSender(const int32_t id, const bool audio, Clock *clock,
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Transport *transport, RtpAudioFeedback *audio_feedback,
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PacedSender *paced_sender);
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virtual ~RTPSender();
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void ProcessBitrate();
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virtual uint16_t ActualSendBitrateKbit() const OVERRIDE;
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uint32_t VideoBitrateSent() const;
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uint32_t FecOverheadRate() const;
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uint32_t NackOverheadRate() const;
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void SetTargetSendBitrate(const uint32_t bits);
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virtual uint16_t MaxDataPayloadLength() const
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OVERRIDE; // with RTP and FEC headers.
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int32_t RegisterPayload(
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const char payload_name[RTP_PAYLOAD_NAME_SIZE],
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const int8_t payload_type, const uint32_t frequency,
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const uint8_t channels, const uint32_t rate);
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int32_t DeRegisterSendPayload(const int8_t payload_type);
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int8_t SendPayloadType() const;
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int SendPayloadFrequency() const;
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void SetSendingStatus(bool enabled);
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void SetSendingMediaStatus(const bool enabled);
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bool SendingMedia() const;
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// Number of sent RTP packets.
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uint32_t Packets() const;
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// Number of sent RTP bytes.
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uint32_t Bytes() const;
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void ResetDataCounters();
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uint32_t StartTimestamp() const;
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void SetStartTimestamp(uint32_t timestamp, bool force);
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uint32_t GenerateNewSSRC();
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void SetSSRC(const uint32_t ssrc);
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virtual uint16_t SequenceNumber() const OVERRIDE;
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void SetSequenceNumber(uint16_t seq);
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int32_t CSRCs(uint32_t arr_of_csrc[kRtpCsrcSize]) const;
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void SetCSRCStatus(const bool include);
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void SetCSRCs(const uint32_t arr_of_csrc[kRtpCsrcSize],
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const uint8_t arr_length);
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int32_t SetMaxPayloadLength(const uint16_t length,
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const uint16_t packet_over_head);
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int32_t SendOutgoingData(
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const FrameType frame_type, const int8_t payload_type,
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const uint32_t time_stamp, int64_t capture_time_ms,
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const uint8_t *payload_data, const uint32_t payload_size,
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const RTPFragmentationHeader *fragmentation,
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VideoCodecInformation *codec_info = NULL,
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const RTPVideoTypeHeader * rtp_type_hdr = NULL);
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int BuildPaddingPacket(uint8_t* packet, int header_length, int32_t bytes);
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int SendPadData(int payload_type, uint32_t timestamp, int64_t capture_time_ms,
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int32_t bytes, StorageType store,
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bool force_full_size_packets, bool only_pad_after_markerbit);
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// RTP header extension
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int32_t SetTransmissionTimeOffset(
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const int32_t transmission_time_offset);
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int32_t SetAbsoluteSendTime(
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const uint32_t absolute_send_time);
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int32_t RegisterRtpHeaderExtension(const RTPExtensionType type,
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const uint8_t id);
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int32_t DeregisterRtpHeaderExtension(const RTPExtensionType type);
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uint16_t RtpHeaderExtensionTotalLength() const;
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uint16_t BuildRTPHeaderExtension(uint8_t* data_buffer) const;
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uint8_t BuildTransmissionTimeOffsetExtension(
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uint8_t *data_buffer) const;
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uint8_t BuildAbsoluteSendTimeExtension(
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uint8_t* data_buffer) const;
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bool UpdateTransmissionTimeOffset(uint8_t *rtp_packet,
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const uint16_t rtp_packet_length,
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const RTPHeader &rtp_header,
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const int64_t time_diff_ms) const;
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bool UpdateAbsoluteSendTime(uint8_t *rtp_packet,
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const uint16_t rtp_packet_length,
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const RTPHeader &rtp_header,
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const int64_t now_ms) const;
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bool TimeToSendPacket(uint16_t sequence_number, int64_t capture_time_ms,
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bool retransmission);
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int TimeToSendPadding(int bytes);
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// NACK.
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int SelectiveRetransmissions() const;
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int SetSelectiveRetransmissions(uint8_t settings);
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void OnReceivedNACK(const std::list<uint16_t>& nack_sequence_numbers,
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const uint16_t avg_rtt);
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void SetStorePacketsStatus(const bool enable,
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const uint16_t number_to_store);
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bool StorePackets() const;
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int32_t ReSendPacket(uint16_t packet_id, uint32_t min_resend_time = 0);
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bool ProcessNACKBitRate(const uint32_t now);
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// RTX.
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void SetRTXStatus(RtxMode mode, bool set_ssrc, uint32_t ssrc);
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void RTXStatus(RtxMode* mode, uint32_t* ssrc, int* payload_type) const;
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void SetRtxPayloadType(int payloadType);
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// Functions wrapping RTPSenderInterface.
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virtual int32_t BuildRTPheader(
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uint8_t *data_buffer, const int8_t payload_type,
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const bool marker_bit, const uint32_t capture_time_stamp,
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int64_t capture_time_ms,
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const bool time_stamp_provided = true,
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const bool inc_sequence_number = true) OVERRIDE;
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virtual uint16_t RTPHeaderLength() const OVERRIDE;
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virtual uint16_t IncrementSequenceNumber() OVERRIDE;
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virtual uint16_t MaxPayloadLength() const OVERRIDE;
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virtual uint16_t PacketOverHead() const OVERRIDE;
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// Current timestamp.
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virtual uint32_t Timestamp() const OVERRIDE;
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virtual uint32_t SSRC() const OVERRIDE;
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virtual int32_t SendToNetwork(
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uint8_t *data_buffer, int payload_length, int rtp_header_length,
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int64_t capture_time_ms, StorageType storage,
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PacedSender::Priority priority) OVERRIDE;
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// Audio.
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// Send a DTMF tone using RFC 2833 (4733).
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int32_t SendTelephoneEvent(const uint8_t key,
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const uint16_t time_ms,
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const uint8_t level);
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bool SendTelephoneEventActive(int8_t *telephone_event) const;
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// Set audio packet size, used to determine when it's time to send a DTMF
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// packet in silence (CNG).
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int32_t SetAudioPacketSize(const uint16_t packet_size_samples);
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// Set status and ID for header-extension-for-audio-level-indication.
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int32_t SetAudioLevelIndicationStatus(const bool enable, const uint8_t ID);
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// Get status and ID for header-extension-for-audio-level-indication.
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int32_t AudioLevelIndicationStatus(bool *enable, uint8_t *id) const;
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// Store the audio level in d_bov for
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// header-extension-for-audio-level-indication.
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int32_t SetAudioLevel(const uint8_t level_d_bov);
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// Set payload type for Redundant Audio Data RFC 2198.
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int32_t SetRED(const int8_t payload_type);
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// Get payload type for Redundant Audio Data RFC 2198.
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int32_t RED(int8_t *payload_type) const;
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// Video.
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VideoCodecInformation *CodecInformationVideo();
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RtpVideoCodecTypes VideoCodecType() const;
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uint32_t MaxConfiguredBitrateVideo() const;
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int32_t SendRTPIntraRequest();
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// FEC.
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int32_t SetGenericFECStatus(const bool enable,
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const uint8_t payload_type_red,
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const uint8_t payload_type_fec);
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int32_t GenericFECStatus(bool *enable, uint8_t *payload_type_red,
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uint8_t *payload_type_fec) const;
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int32_t SetFecParameters(const FecProtectionParams *delta_params,
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const FecProtectionParams *key_params);
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protected:
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int32_t CheckPayloadType(const int8_t payload_type,
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RtpVideoCodecTypes *video_type);
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private:
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int CreateRTPHeader(uint8_t* header, int8_t payload_type,
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uint32_t ssrc, bool marker_bit,
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uint32_t timestamp, uint16_t sequence_number,
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const uint32_t* csrcs, uint8_t csrcs_length) const;
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void UpdateNACKBitRate(const uint32_t bytes, const uint32_t now);
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bool SendPaddingAccordingToBitrate(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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void BuildRtxPacket(uint8_t* buffer, uint16_t* length,
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uint8_t* buffer_rtx);
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bool SendPacketToNetwork(const uint8_t *packet, uint32_t size);
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int32_t id_;
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const bool audio_configured_;
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RTPSenderAudio *audio_;
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RTPSenderVideo *video_;
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PacedSender *paced_sender_;
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CriticalSectionWrapper *send_critsect_;
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Transport *transport_;
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bool sending_media_;
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uint16_t max_payload_length_;
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uint16_t target_send_bitrate_;
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uint16_t packet_over_head_;
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int8_t payload_type_;
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std::map<int8_t, ModuleRTPUtility::Payload *> payload_type_map_;
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RtpHeaderExtensionMap rtp_header_extension_map_;
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int32_t transmission_time_offset_;
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uint32_t absolute_send_time_;
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// NACK
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uint32_t nack_byte_count_times_[NACK_BYTECOUNT_SIZE];
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int32_t nack_byte_count_[NACK_BYTECOUNT_SIZE];
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Bitrate nack_bitrate_;
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RTPPacketHistory *packet_history_;
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// Statistics
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scoped_ptr<CriticalSectionWrapper> statistics_crit_;
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uint32_t packets_sent_;
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uint32_t payload_bytes_sent_;
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// RTP variables
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bool start_time_stamp_forced_;
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uint32_t start_time_stamp_;
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SSRCDatabase &ssrc_db_;
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uint32_t remote_ssrc_;
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bool sequence_number_forced_;
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uint16_t sequence_number_;
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uint16_t sequence_number_rtx_;
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bool ssrc_forced_;
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uint32_t ssrc_;
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uint32_t timestamp_;
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int64_t capture_time_ms_;
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int64_t last_timestamp_time_ms_;
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bool last_packet_marker_bit_;
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uint8_t num_csrcs_;
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uint32_t csrcs_[kRtpCsrcSize];
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bool include_csrcs_;
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RtxMode rtx_;
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uint32_t ssrc_rtx_;
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int payload_type_rtx_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
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