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Reformatted rtp_sender: made lint clean.

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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
This commit is contained in:
phoglund@webrtc.org
2013-01-25 10:53:38 +00:00
parent 3e47a0a611
commit 43da54a458
4 changed files with 652 additions and 752 deletions
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18
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
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@ -123,11 +123,10 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
simulcast_(false), simulcast_(false),
key_frame_req_method_(kKeyFrameReqFirRtp), key_frame_req_method_(kKeyFrameReqFirRtp),
remote_bitrate_(configuration.remote_bitrate_estimator), remote_bitrate_(configuration.remote_bitrate_estimator),
rtt_observer_(configuration.rtt_observer)
#ifdef MATLAB #ifdef MATLAB
, plot1_(NULL) , plot1_(NULL),
#endif #endif
{ rtt_observer_(configuration.rtt_observer) {
RTPReceiverStrategy* rtp_receiver_strategy; RTPReceiverStrategy* rtp_receiver_strategy;
if (configuration.audio) { if (configuration.audio) {
// If audio, we need to be able to handle telephone events too, so stash // If audio, we need to be able to handle telephone events too, so stash
@ -1645,7 +1644,7 @@ bool ModuleRtpRtcpImpl::SendTelephoneEventActive(
id_, id_,
"SendTelephoneEventActive()"); "SendTelephoneEventActive()");
return rtp_sender_.SendTelephoneEventActive(telephone_event); return rtp_sender_.SendTelephoneEventActive(&telephone_event);
} }
// Set audio packet size, used to determine when it's time to send a DTMF // Set audio packet size, used to determine when it's time to send a DTMF
@ -1689,7 +1688,7 @@ WebRtc_Word32 ModuleRtpRtcpImpl::GetRTPAudioLevelIndicationStatus(
kTraceRtpRtcp, kTraceRtpRtcp,
id_, id_,
"GetRTPAudioLevelIndicationStatus()"); "GetRTPAudioLevelIndicationStatus()");
return rtp_sender_.AudioLevelIndicationStatus(enable, id); return rtp_sender_.AudioLevelIndicationStatus(&enable, &id);
} }
WebRtc_Word32 ModuleRtpRtcpImpl::SetAudioLevel( WebRtc_Word32 ModuleRtpRtcpImpl::SetAudioLevel(
@ -1719,7 +1718,7 @@ WebRtc_Word32 ModuleRtpRtcpImpl::SendREDPayloadType(
WebRtc_Word8& payload_type) const { WebRtc_Word8& payload_type) const {
WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SendREDPayloadType()"); WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SendREDPayloadType()");
return rtp_sender_.RED(payload_type); return rtp_sender_.RED(&payload_type);
} }
RtpVideoCodecTypes ModuleRtpRtcpImpl::ReceivedVideoCodec() const { RtpVideoCodecTypes ModuleRtpRtcpImpl::ReceivedVideoCodec() const {
@ -1882,9 +1881,9 @@ WebRtc_Word32 ModuleRtpRtcpImpl::GenericFECStatus(
it++; it++;
} }
} }
WebRtc_Word32 ret_val = rtp_sender_.GenericFECStatus(enable, WebRtc_Word32 ret_val = rtp_sender_.GenericFECStatus(&enable,
payload_type_red, &payload_type_red,
payload_type_fec); &payload_type_fec);
if (child_enabled) { if (child_enabled) {
// Returns true if enabled for any child module. // Returns true if enabled for any child module.
enable = child_enabled; enable = child_enabled;
@ -2083,5 +2082,4 @@ int64_t ModuleRtpRtcpImpl::RtcpReportInterval() {
else else
return RTCP_INTERVAL_VIDEO_MS; return RTCP_INTERVAL_VIDEO_MS;
} }
} // Namespace webrtc } // Namespace webrtc

4
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
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@ -515,11 +515,11 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
RemoteBitrateEstimator* remote_bitrate_; RemoteBitrateEstimator* remote_bitrate_;
RtcpRttObserver* rtt_observer_;
#ifdef MATLAB #ifdef MATLAB
MatlabPlot* plot1_; MatlabPlot* plot1_;
#endif #endif
RtcpRttObserver* rtt_observer_;
}; };
} // namespace webrtc } // namespace webrtc

1099
webrtc/modules/rtp_rtcp/source/rtp_sender.cc
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File diff suppressed because it is too large Load Diff

283
webrtc/modules/rtp_rtcp/source/rtp_sender.h
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@ -23,9 +23,10 @@
#include "webrtc/modules/rtp_rtcp/source/ssrc_database.h" #include "webrtc/modules/rtp_rtcp/source/ssrc_database.h"
#include "webrtc/modules/rtp_rtcp/source/video_codec_information.h" #include "webrtc/modules/rtp_rtcp/source/video_codec_information.h"
#define MAX_INIT_RTP_SEQ_NUMBER 32767 // 2^15 -1 #define MAX_INIT_RTP_SEQ_NUMBER 32767 // 2^15 -1.
namespace webrtc { namespace webrtc {
class CriticalSectionWrapper; class CriticalSectionWrapper;
class PacedSender; class PacedSender;
class RTPPacketHistory; class RTPPacketHistory;
@ -40,36 +41,30 @@ class RTPSenderInterface {
virtual WebRtc_UWord32 SSRC() const = 0; virtual WebRtc_UWord32 SSRC() const = 0;
virtual WebRtc_UWord32 Timestamp() const = 0; virtual WebRtc_UWord32 Timestamp() const = 0;
virtual WebRtc_Word32 BuildRTPheader(WebRtc_UWord8* dataBuffer, virtual WebRtc_Word32 BuildRTPheader(
const WebRtc_Word8 payloadType, WebRtc_UWord8 *data_buffer, const WebRtc_Word8 payload_type,
const bool markerBit, const bool marker_bit, const WebRtc_UWord32 capture_time_stamp,
const WebRtc_UWord32 captureTimeStamp, const bool time_stamp_provided = true,
const bool timeStampProvided = true, const bool inc_sequence_number = true) = 0;
const bool incSequenceNumber = true) = 0;
virtual WebRtc_UWord16 RTPHeaderLength() const = 0; virtual WebRtc_UWord16 RTPHeaderLength() const = 0;
virtual WebRtc_UWord16 IncrementSequenceNumber() = 0; virtual WebRtc_UWord16 IncrementSequenceNumber() = 0;
virtual WebRtc_UWord16 SequenceNumber() const = 0; virtual WebRtc_UWord16 SequenceNumber() const = 0;
virtual WebRtc_UWord16 MaxPayloadLength() const = 0; virtual WebRtc_UWord16 MaxPayloadLength() const = 0;
virtual WebRtc_UWord16 MaxDataPayloadLength() const = 0; virtual WebRtc_UWord16 MaxDataPayloadLength() const = 0;
virtual WebRtc_UWord16 PacketOverHead() const = 0; virtual WebRtc_UWord16 PacketOverHead() const = 0;
virtual WebRtc_UWord16 ActualSendBitrateKbit() const = 0; virtual WebRtc_UWord16 ActualSendBitrateKbit() const = 0;
virtual WebRtc_Word32 SendToNetwork(uint8_t* data_buffer, virtual WebRtc_Word32 SendToNetwork(
int payload_length, uint8_t *data_buffer, int payload_length, int rtp_header_length,
int rtp_header_length, int64_t capture_time_ms, StorageType storage) = 0;
int64_t capture_time_ms,
StorageType storage) = 0;
}; };
class RTPSender : public Bitrate, public RTPSenderInterface { class RTPSender : public Bitrate, public RTPSenderInterface {
public: public:
RTPSender(const WebRtc_Word32 id, RTPSender(const WebRtc_Word32 id, const bool audio, Clock *clock,
const bool audio, Transport *transport, RtpAudioFeedback *audio_feedback,
Clock* clock, PacedSender *paced_sender);
Transport* transport,
RtpAudioFeedback* audio_feedback,
PacedSender* paced_sender);
virtual ~RTPSender(); virtual ~RTPSender();
void ProcessBitrate(); void ProcessBitrate();
@ -82,16 +77,14 @@ class RTPSender : public Bitrate, public RTPSenderInterface {
void SetTargetSendBitrate(const WebRtc_UWord32 bits); void SetTargetSendBitrate(const WebRtc_UWord32 bits);
WebRtc_UWord16 MaxDataPayloadLength() const; // with RTP and FEC headers WebRtc_UWord16 MaxDataPayloadLength() const; // with RTP and FEC headers.
WebRtc_Word32 RegisterPayload( WebRtc_Word32 RegisterPayload(
const char payloadName[RTP_PAYLOAD_NAME_SIZE], const char payload_name[RTP_PAYLOAD_NAME_SIZE],
const WebRtc_Word8 payloadType, const WebRtc_Word8 payload_type, const WebRtc_UWord32 frequency,
const WebRtc_UWord32 frequency, const WebRtc_UWord8 channels, const WebRtc_UWord32 rate);
const WebRtc_UWord8 channels,
const WebRtc_UWord32 rate);
WebRtc_Word32 DeRegisterSendPayload(const WebRtc_Word8 payloadType); WebRtc_Word32 DeRegisterSendPayload(const WebRtc_Word8 payload_type);
WebRtc_Word8 SendPayloadType() const; WebRtc_Word8 SendPayloadType() const;
@ -102,10 +95,10 @@ class RTPSender : public Bitrate, public RTPSenderInterface {
void SetSendingMediaStatus(const bool enabled); void SetSendingMediaStatus(const bool enabled);
bool SendingMedia() const; bool SendingMedia() const;
// number of sent RTP packets // Number of sent RTP packets.
WebRtc_UWord32 Packets() const; WebRtc_UWord32 Packets() const;
// number of sent RTP bytes // Number of sent RTP bytes.
WebRtc_UWord32 Bytes() const; WebRtc_UWord32 Bytes() const;
void ResetDataCounters(); void ResetDataCounters();
@ -119,35 +112,30 @@ class RTPSender : public Bitrate, public RTPSenderInterface {
WebRtc_UWord16 SequenceNumber() const; WebRtc_UWord16 SequenceNumber() const;
void SetSequenceNumber(WebRtc_UWord16 seq); void SetSequenceNumber(WebRtc_UWord16 seq);
WebRtc_Word32 CSRCs(WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize]) const; WebRtc_Word32 CSRCs(WebRtc_UWord32 arr_of_csrc[kRtpCsrcSize]) const;
void SetCSRCStatus(const bool include); void SetCSRCStatus(const bool include);
void SetCSRCs(const WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize], void SetCSRCs(const WebRtc_UWord32 arr_of_csrc[kRtpCsrcSize],
const WebRtc_UWord8 arrLength); const WebRtc_UWord8 arr_length);
WebRtc_Word32 SetMaxPayloadLength(const WebRtc_UWord16 length, WebRtc_Word32 SetMaxPayloadLength(const WebRtc_UWord16 length,
const WebRtc_UWord16 packetOverHead); const WebRtc_UWord16 packet_over_head);
WebRtc_Word32 SendOutgoingData(const FrameType frameType, WebRtc_Word32 SendOutgoingData(
const WebRtc_Word8 payloadType, const FrameType frame_type, const WebRtc_Word8 payload_type,
const WebRtc_UWord32 timeStamp, const WebRtc_UWord32 time_stamp, int64_t capture_time_ms,
int64_t capture_time_ms, const WebRtc_UWord8 *payload_data, const WebRtc_UWord32 payload_size,
const WebRtc_UWord8* payloadData, const RTPFragmentationHeader *fragmentation,
const WebRtc_UWord32 payloadSize, VideoCodecInformation *codec_info = NULL,
const RTPFragmentationHeader* fragmentation, const RTPVideoTypeHeader * rtp_type_hdr = NULL);
VideoCodecInformation* codecInfo = NULL,
const RTPVideoTypeHeader* rtpTypeHdr = NULL);
WebRtc_Word32 SendPadData(WebRtc_Word8 payload_type, WebRtc_Word32 SendPadData(WebRtc_Word8 payload_type,
WebRtc_UWord32 capture_timestamp, WebRtc_UWord32 capture_timestamp,
int64_t capture_time_ms, int64_t capture_time_ms, WebRtc_Word32 bytes);
WebRtc_Word32 bytes); // RTP header extension
/*
* RTP header extension
*/
WebRtc_Word32 SetTransmissionTimeOffset( WebRtc_Word32 SetTransmissionTimeOffset(
const WebRtc_Word32 transmissionTimeOffset); const WebRtc_Word32 transmission_time_offset);
WebRtc_Word32 RegisterRtpHeaderExtension(const RTPExtensionType type, WebRtc_Word32 RegisterRtpHeaderExtension(const RTPExtensionType type,
const WebRtc_UWord8 id); const WebRtc_UWord8 id);
@ -156,109 +144,97 @@ class RTPSender : public Bitrate, public RTPSenderInterface {
WebRtc_UWord16 RtpHeaderExtensionTotalLength() const; WebRtc_UWord16 RtpHeaderExtensionTotalLength() const;
WebRtc_UWord16 BuildRTPHeaderExtension(WebRtc_UWord8* dataBuffer) const; WebRtc_UWord16 BuildRTPHeaderExtension(WebRtc_UWord8 *data_buffer) const;
WebRtc_UWord8 BuildTransmissionTimeOffsetExtension( WebRtc_UWord8 BuildTransmissionTimeOffsetExtension(
WebRtc_UWord8* dataBuffer) const; WebRtc_UWord8 *data_buffer) const;
bool UpdateTransmissionTimeOffset(WebRtc_UWord8* rtp_packet, bool UpdateTransmissionTimeOffset(WebRtc_UWord8 *rtp_packet,
const WebRtc_UWord16 rtp_packet_length, const WebRtc_UWord16 rtp_packet_length,
const WebRtcRTPHeader& rtp_header, const WebRtcRTPHeader &rtp_header,
const WebRtc_Word64 time_diff_ms) const; const WebRtc_Word64 time_diff_ms) const;
void TimeToSendPacket(uint16_t sequence_number, int64_t capture_time_ms); void TimeToSendPacket(uint16_t sequence_number, int64_t capture_time_ms);
/* // NACK.
* NACK
*/
int SelectiveRetransmissions() const; int SelectiveRetransmissions() const;
int SetSelectiveRetransmissions(uint8_t settings); int SetSelectiveRetransmissions(uint8_t settings);
void OnReceivedNACK(const WebRtc_UWord16 nackSequenceNumbersLength, void OnReceivedNACK(const WebRtc_UWord16 nack_sequence_numbers_length,
const WebRtc_UWord16* nackSequenceNumbers, const WebRtc_UWord16 *nack_sequence_numbers,
const WebRtc_UWord16 avgRTT); const WebRtc_UWord16 avg_rtt);
void SetStorePacketsStatus(const bool enable, void SetStorePacketsStatus(const bool enable,
const WebRtc_UWord16 numberToStore); const WebRtc_UWord16 number_to_store);
bool StorePackets() const; bool StorePackets() const;
WebRtc_Word32 ReSendPacket(WebRtc_UWord16 packet_id, WebRtc_Word32 ReSendPacket(WebRtc_UWord16 packet_id,
WebRtc_UWord32 min_resend_time = 0); WebRtc_UWord32 min_resend_time = 0);
WebRtc_Word32 ReSendToNetwork(const WebRtc_UWord8* packet, WebRtc_Word32 ReSendToNetwork(const WebRtc_UWord8 *packet,
const WebRtc_UWord32 size); const WebRtc_UWord32 size);
bool ProcessNACKBitRate(const WebRtc_UWord32 now); bool ProcessNACKBitRate(const WebRtc_UWord32 now);
/* // RTX.
* RTX void SetRTXStatus(const bool enable, const bool set_ssrc,
*/
void SetRTXStatus(const bool enable,
const bool setSSRC,
const WebRtc_UWord32 SSRC); const WebRtc_UWord32 SSRC);
void RTXStatus(bool* enable, WebRtc_UWord32* SSRC) const; void RTXStatus(bool *enable, WebRtc_UWord32 *SSRC) const;
/* // Functions wrapping RTPSenderInterface.
* Functions wrapping RTPSenderInterface virtual WebRtc_Word32 BuildRTPheader(
*/ WebRtc_UWord8 *data_buffer, const WebRtc_Word8 payload_type,
virtual WebRtc_Word32 BuildRTPheader(WebRtc_UWord8* dataBuffer, const bool marker_bit, const WebRtc_UWord32 capture_time_stamp,
const WebRtc_Word8 payloadType, const bool time_stamp_provided = true,
const bool markerBit, const bool inc_sequence_number = true);
const WebRtc_UWord32 captureTimeStamp,
const bool timeStampProvided = true,
const bool incSequenceNumber = true);
virtual WebRtc_UWord16 RTPHeaderLength() const ; virtual WebRtc_UWord16 RTPHeaderLength() const;
virtual WebRtc_UWord16 IncrementSequenceNumber(); virtual WebRtc_UWord16 IncrementSequenceNumber();
virtual WebRtc_UWord16 MaxPayloadLength() const; virtual WebRtc_UWord16 MaxPayloadLength() const;
virtual WebRtc_UWord16 PacketOverHead() const; virtual WebRtc_UWord16 PacketOverHead() const;
// current timestamp // Current timestamp.
virtual WebRtc_UWord32 Timestamp() const; virtual WebRtc_UWord32 Timestamp() const;
virtual WebRtc_UWord32 SSRC() const; virtual WebRtc_UWord32 SSRC() const;
virtual WebRtc_Word32 SendToNetwork(uint8_t* data_buffer, virtual WebRtc_Word32 SendToNetwork(
int payload_length, uint8_t *data_buffer, int payload_length, int rtp_header_length,
int rtp_header_length, int64_t capture_time_ms, StorageType storage);
int64_t capture_time_ms,
StorageType storage); // Audio.
/*
* Audio // Send a DTMF tone using RFC 2833 (4733).
*/
// Send a DTMF tone using RFC 2833 (4733)
WebRtc_Word32 SendTelephoneEvent(const WebRtc_UWord8 key, WebRtc_Word32 SendTelephoneEvent(const WebRtc_UWord8 key,
const WebRtc_UWord16 time_ms, const WebRtc_UWord16 time_ms,
const WebRtc_UWord8 level); const WebRtc_UWord8 level);
bool SendTelephoneEventActive(WebRtc_Word8& telephoneEvent) const; bool SendTelephoneEventActive(WebRtc_Word8 *telephone_event) const;
// Set audio packet size, used to determine when it's time to send a DTMF // Set audio packet size, used to determine when it's time to send a DTMF
// packet in silence (CNG) // packet in silence (CNG).
WebRtc_Word32 SetAudioPacketSize(const WebRtc_UWord16 packetSizeSamples); WebRtc_Word32 SetAudioPacketSize(const WebRtc_UWord16 packet_size_samples);
// Set status and ID for header-extension-for-audio-level-indication. // Set status and ID for header-extension-for-audio-level-indication.
WebRtc_Word32 SetAudioLevelIndicationStatus(const bool enable, WebRtc_Word32 SetAudioLevelIndicationStatus(const bool enable,
const WebRtc_UWord8 ID); const WebRtc_UWord8 ID);
// Get status and ID for header-extension-for-audio-level-indication. // Get status and ID for header-extension-for-audio-level-indication.
WebRtc_Word32 AudioLevelIndicationStatus(bool& enable, WebRtc_Word32 AudioLevelIndicationStatus(bool *enable,
WebRtc_UWord8& ID) const; WebRtc_UWord8 *id) const;
// Store the audio level in dBov for // Store the audio level in d_bov for
// header-extension-for-audio-level-indication. // header-extension-for-audio-level-indication.
WebRtc_Word32 SetAudioLevel(const WebRtc_UWord8 level_dBov); WebRtc_Word32 SetAudioLevel(const WebRtc_UWord8 level_d_bov);
// Set payload type for Redundant Audio Data RFC 2198 // Set payload type for Redundant Audio Data RFC 2198.
WebRtc_Word32 SetRED(const WebRtc_Word8 payloadType); WebRtc_Word32 SetRED(const WebRtc_Word8 payload_type);
// Get payload type for Redundant Audio Data RFC 2198 // Get payload type for Redundant Audio Data RFC 2198.
WebRtc_Word32 RED(WebRtc_Word8& payloadType) const; WebRtc_Word32 RED(WebRtc_Word8 *payload_type) const;
/* // Video.
* Video VideoCodecInformation *CodecInformationVideo();
*/
VideoCodecInformation* CodecInformationVideo();
RtpVideoCodecTypes VideoCodecType() const; RtpVideoCodecTypes VideoCodecType() const;
@ -266,81 +242,78 @@ class RTPSender : public Bitrate, public RTPSenderInterface {
WebRtc_Word32 SendRTPIntraRequest(); WebRtc_Word32 SendRTPIntraRequest();
// FEC // FEC.
WebRtc_Word32 SetGenericFECStatus(const bool enable, WebRtc_Word32 SetGenericFECStatus(const bool enable,
const WebRtc_UWord8 payloadTypeRED, const WebRtc_UWord8 payload_type_red,
const WebRtc_UWord8 payloadTypeFEC); const WebRtc_UWord8 payload_type_fec);
WebRtc_Word32 GenericFECStatus(bool& enable, WebRtc_Word32 GenericFECStatus(bool *enable, WebRtc_UWord8 *payload_type_red,
WebRtc_UWord8& payloadTypeRED, WebRtc_UWord8 *payload_type_fec) const;
WebRtc_UWord8& payloadTypeFEC) const;
WebRtc_Word32 SetFecParameters( WebRtc_Word32 SetFecParameters(const FecProtectionParams *delta_params,
const FecProtectionParams* delta_params, const FecProtectionParams *key_params);
const FecProtectionParams* key_params);
protected: protected:
WebRtc_Word32 CheckPayloadType(const WebRtc_Word8 payloadType, WebRtc_Word32 CheckPayloadType(const WebRtc_Word8 payload_type,
RtpVideoCodecTypes& videoType); RtpVideoCodecTypes *video_type);
private: private:
void UpdateNACKBitRate(const WebRtc_UWord32 bytes, void UpdateNACKBitRate(const WebRtc_UWord32 bytes, const WebRtc_UWord32 now);
const WebRtc_UWord32 now);
WebRtc_Word32 SendPaddingAccordingToBitrate( WebRtc_Word32 SendPaddingAccordingToBitrate(WebRtc_Word8 payload_type,
WebRtc_Word8 payload_type, WebRtc_UWord32 capture_timestamp,
WebRtc_UWord32 capture_timestamp, int64_t capture_time_ms);
int64_t capture_time_ms);
WebRtc_Word32 _id; WebRtc_Word32 id_;
const bool _audioConfigured; const bool audio_configured_;
RTPSenderAudio* _audio; RTPSenderAudio *audio_;
RTPSenderVideo* _video; RTPSenderVideo *video_;
PacedSender* paced_sender_; PacedSender *paced_sender_;
CriticalSectionWrapper* _sendCritsect; CriticalSectionWrapper *send_critsect_;
Transport* _transport; Transport *transport_;
bool _sendingMedia; bool sending_media_;
WebRtc_UWord16 _maxPayloadLength; WebRtc_UWord16 max_payload_length_;
WebRtc_UWord16 _targetSendBitrate; WebRtc_UWord16 target_send_bitrate_;
WebRtc_UWord16 _packetOverHead; WebRtc_UWord16 packet_over_head_;
WebRtc_Word8 _payloadType; WebRtc_Word8 payload_type_;
std::map<WebRtc_Word8, ModuleRTPUtility::Payload*> _payloadTypeMap; std::map<WebRtc_Word8, ModuleRTPUtility::Payload *> payload_type_map_;
RtpHeaderExtensionMap _rtpHeaderExtensionMap; RtpHeaderExtensionMap rtp_header_extension_map_;
WebRtc_Word32 _transmissionTimeOffset; WebRtc_Word32 transmission_time_offset_;
// NACK // NACK
WebRtc_UWord32 _nackByteCountTimes[NACK_BYTECOUNT_SIZE]; WebRtc_UWord32 nack_byte_count_times_[NACK_BYTECOUNT_SIZE];
WebRtc_Word32 _nackByteCount[NACK_BYTECOUNT_SIZE]; WebRtc_Word32 nack_byte_count_[NACK_BYTECOUNT_SIZE];
Bitrate _nackBitrate; Bitrate nack_bitrate_;
RTPPacketHistory* _packetHistory; RTPPacketHistory *packet_history_;
// Statistics // Statistics
WebRtc_UWord32 _packetsSent; WebRtc_UWord32 packets_sent_;
WebRtc_UWord32 _payloadBytesSent; WebRtc_UWord32 payload_bytes_sent_;
// RTP variables // RTP variables
bool _startTimeStampForced; bool start_time_stamp_forced_;
WebRtc_UWord32 _startTimeStamp; WebRtc_UWord32 start_time_stamp_;
SSRCDatabase& _ssrcDB; SSRCDatabase &ssrc_db_;
WebRtc_UWord32 _remoteSSRC; WebRtc_UWord32 remote_ssrc_;
bool _sequenceNumberForced; bool sequence_number_forced_;
WebRtc_UWord16 _sequenceNumber; WebRtc_UWord16 sequence_number_;
WebRtc_UWord16 _sequenceNumberRTX; WebRtc_UWord16 sequence_number_rtx_;
bool _ssrcForced; bool ssrc_forced_;
WebRtc_UWord32 _ssrc; WebRtc_UWord32 ssrc_;
WebRtc_UWord32 _timeStamp; WebRtc_UWord32 time_stamp_;
WebRtc_UWord8 _CSRCs; WebRtc_UWord8 csrcs_;
WebRtc_UWord32 _CSRC[kRtpCsrcSize]; WebRtc_UWord32 csrc_[kRtpCsrcSize];
bool _includeCSRCs; bool include_csrcs_;
bool _RTX; bool rtx_;
WebRtc_UWord32 _ssrcRTX; WebRtc_UWord32 ssrc_rtx_;
}; };
} // namespace webrtc
#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_ } // namespace webrtc
#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_