43da54a458
TESTED=rtp_rtcp_unittests BUG= Review URL: https://webrtc-codereview.appspot.com/1062004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3412 4adac7df-926f-26a2-2b94-8c16560cd09d
320 lines
11 KiB
C++
320 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 <cassert>
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#include <cmath>
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#include <map>
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#include "webrtc/common_types.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 PacedSender;
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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 WebRtc_UWord32 SSRC() const = 0;
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virtual WebRtc_UWord32 Timestamp() const = 0;
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virtual WebRtc_Word32 BuildRTPheader(
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WebRtc_UWord8 *data_buffer, const WebRtc_Word8 payload_type,
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const bool marker_bit, const WebRtc_UWord32 capture_time_stamp,
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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 WebRtc_UWord16 RTPHeaderLength() const = 0;
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virtual WebRtc_UWord16 IncrementSequenceNumber() = 0;
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virtual WebRtc_UWord16 SequenceNumber() const = 0;
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virtual WebRtc_UWord16 MaxPayloadLength() const = 0;
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virtual WebRtc_UWord16 MaxDataPayloadLength() const = 0;
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virtual WebRtc_UWord16 PacketOverHead() const = 0;
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virtual WebRtc_UWord16 ActualSendBitrateKbit() const = 0;
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virtual WebRtc_Word32 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) = 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 WebRtc_Word32 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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WebRtc_UWord16 ActualSendBitrateKbit() const;
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WebRtc_UWord32 VideoBitrateSent() const;
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WebRtc_UWord32 FecOverheadRate() const;
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WebRtc_UWord32 NackOverheadRate() const;
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void SetTargetSendBitrate(const WebRtc_UWord32 bits);
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WebRtc_UWord16 MaxDataPayloadLength() const; // with RTP and FEC headers.
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WebRtc_Word32 RegisterPayload(
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const char payload_name[RTP_PAYLOAD_NAME_SIZE],
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const WebRtc_Word8 payload_type, const WebRtc_UWord32 frequency,
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const WebRtc_UWord8 channels, const WebRtc_UWord32 rate);
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WebRtc_Word32 DeRegisterSendPayload(const WebRtc_Word8 payload_type);
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WebRtc_Word8 SendPayloadType() const;
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int SendPayloadFrequency() const;
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void SetSendingStatus(const 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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WebRtc_UWord32 Packets() const;
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// Number of sent RTP bytes.
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WebRtc_UWord32 Bytes() const;
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void ResetDataCounters();
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WebRtc_UWord32 StartTimestamp() const;
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void SetStartTimestamp(WebRtc_UWord32 timestamp, bool force);
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WebRtc_UWord32 GenerateNewSSRC();
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void SetSSRC(const WebRtc_UWord32 ssrc);
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WebRtc_UWord16 SequenceNumber() const;
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void SetSequenceNumber(WebRtc_UWord16 seq);
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WebRtc_Word32 CSRCs(WebRtc_UWord32 arr_of_csrc[kRtpCsrcSize]) const;
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void SetCSRCStatus(const bool include);
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void SetCSRCs(const WebRtc_UWord32 arr_of_csrc[kRtpCsrcSize],
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const WebRtc_UWord8 arr_length);
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WebRtc_Word32 SetMaxPayloadLength(const WebRtc_UWord16 length,
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const WebRtc_UWord16 packet_over_head);
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WebRtc_Word32 SendOutgoingData(
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const FrameType frame_type, const WebRtc_Word8 payload_type,
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const WebRtc_UWord32 time_stamp, int64_t capture_time_ms,
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const WebRtc_UWord8 *payload_data, const WebRtc_UWord32 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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WebRtc_Word32 SendPadData(WebRtc_Word8 payload_type,
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WebRtc_UWord32 capture_timestamp,
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int64_t capture_time_ms, WebRtc_Word32 bytes);
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// RTP header extension
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WebRtc_Word32 SetTransmissionTimeOffset(
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const WebRtc_Word32 transmission_time_offset);
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WebRtc_Word32 RegisterRtpHeaderExtension(const RTPExtensionType type,
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const WebRtc_UWord8 id);
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WebRtc_Word32 DeregisterRtpHeaderExtension(const RTPExtensionType type);
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WebRtc_UWord16 RtpHeaderExtensionTotalLength() const;
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WebRtc_UWord16 BuildRTPHeaderExtension(WebRtc_UWord8 *data_buffer) const;
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WebRtc_UWord8 BuildTransmissionTimeOffsetExtension(
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WebRtc_UWord8 *data_buffer) const;
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bool UpdateTransmissionTimeOffset(WebRtc_UWord8 *rtp_packet,
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const WebRtc_UWord16 rtp_packet_length,
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const WebRtcRTPHeader &rtp_header,
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const WebRtc_Word64 time_diff_ms) const;
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void TimeToSendPacket(uint16_t sequence_number, int64_t capture_time_ms);
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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 WebRtc_UWord16 nack_sequence_numbers_length,
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const WebRtc_UWord16 *nack_sequence_numbers,
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const WebRtc_UWord16 avg_rtt);
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void SetStorePacketsStatus(const bool enable,
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const WebRtc_UWord16 number_to_store);
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bool StorePackets() const;
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WebRtc_Word32 ReSendPacket(WebRtc_UWord16 packet_id,
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WebRtc_UWord32 min_resend_time = 0);
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WebRtc_Word32 ReSendToNetwork(const WebRtc_UWord8 *packet,
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const WebRtc_UWord32 size);
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bool ProcessNACKBitRate(const WebRtc_UWord32 now);
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// RTX.
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void SetRTXStatus(const bool enable, const bool set_ssrc,
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const WebRtc_UWord32 SSRC);
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void RTXStatus(bool *enable, WebRtc_UWord32 *SSRC) const;
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// Functions wrapping RTPSenderInterface.
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virtual WebRtc_Word32 BuildRTPheader(
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WebRtc_UWord8 *data_buffer, const WebRtc_Word8 payload_type,
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const bool marker_bit, const WebRtc_UWord32 capture_time_stamp,
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const bool time_stamp_provided = true,
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const bool inc_sequence_number = true);
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virtual WebRtc_UWord16 RTPHeaderLength() const;
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virtual WebRtc_UWord16 IncrementSequenceNumber();
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virtual WebRtc_UWord16 MaxPayloadLength() const;
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virtual WebRtc_UWord16 PacketOverHead() const;
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// Current timestamp.
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virtual WebRtc_UWord32 Timestamp() const;
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virtual WebRtc_UWord32 SSRC() const;
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virtual WebRtc_Word32 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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// Audio.
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// Send a DTMF tone using RFC 2833 (4733).
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WebRtc_Word32 SendTelephoneEvent(const WebRtc_UWord8 key,
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const WebRtc_UWord16 time_ms,
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const WebRtc_UWord8 level);
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bool SendTelephoneEventActive(WebRtc_Word8 *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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WebRtc_Word32 SetAudioPacketSize(const WebRtc_UWord16 packet_size_samples);
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// Set status and ID for header-extension-for-audio-level-indication.
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WebRtc_Word32 SetAudioLevelIndicationStatus(const bool enable,
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const WebRtc_UWord8 ID);
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// Get status and ID for header-extension-for-audio-level-indication.
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WebRtc_Word32 AudioLevelIndicationStatus(bool *enable,
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WebRtc_UWord8 *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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WebRtc_Word32 SetAudioLevel(const WebRtc_UWord8 level_d_bov);
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// Set payload type for Redundant Audio Data RFC 2198.
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WebRtc_Word32 SetRED(const WebRtc_Word8 payload_type);
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// Get payload type for Redundant Audio Data RFC 2198.
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WebRtc_Word32 RED(WebRtc_Word8 *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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WebRtc_UWord32 MaxConfiguredBitrateVideo() const;
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WebRtc_Word32 SendRTPIntraRequest();
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// FEC.
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WebRtc_Word32 SetGenericFECStatus(const bool enable,
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const WebRtc_UWord8 payload_type_red,
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const WebRtc_UWord8 payload_type_fec);
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WebRtc_Word32 GenericFECStatus(bool *enable, WebRtc_UWord8 *payload_type_red,
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WebRtc_UWord8 *payload_type_fec) const;
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WebRtc_Word32 SetFecParameters(const FecProtectionParams *delta_params,
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const FecProtectionParams *key_params);
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protected:
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WebRtc_Word32 CheckPayloadType(const WebRtc_Word8 payload_type,
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RtpVideoCodecTypes *video_type);
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private:
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void UpdateNACKBitRate(const WebRtc_UWord32 bytes, const WebRtc_UWord32 now);
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WebRtc_Word32 SendPaddingAccordingToBitrate(WebRtc_Word8 payload_type,
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WebRtc_UWord32 capture_timestamp,
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int64_t capture_time_ms);
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WebRtc_Word32 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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WebRtc_UWord16 max_payload_length_;
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WebRtc_UWord16 target_send_bitrate_;
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WebRtc_UWord16 packet_over_head_;
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WebRtc_Word8 payload_type_;
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std::map<WebRtc_Word8, ModuleRTPUtility::Payload *> payload_type_map_;
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RtpHeaderExtensionMap rtp_header_extension_map_;
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WebRtc_Word32 transmission_time_offset_;
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// NACK
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WebRtc_UWord32 nack_byte_count_times_[NACK_BYTECOUNT_SIZE];
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WebRtc_Word32 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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WebRtc_UWord32 packets_sent_;
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WebRtc_UWord32 payload_bytes_sent_;
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// RTP variables
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bool start_time_stamp_forced_;
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WebRtc_UWord32 start_time_stamp_;
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SSRCDatabase &ssrc_db_;
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WebRtc_UWord32 remote_ssrc_;
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bool sequence_number_forced_;
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WebRtc_UWord16 sequence_number_;
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WebRtc_UWord16 sequence_number_rtx_;
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bool ssrc_forced_;
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WebRtc_UWord32 ssrc_;
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WebRtc_UWord32 time_stamp_;
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WebRtc_UWord8 csrcs_;
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WebRtc_UWord32 csrc_[kRtpCsrcSize];
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bool include_csrcs_;
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bool rtx_;
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WebRtc_UWord32 ssrc_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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